From martin at pattersong.co.uk Mon Nov 1 09:47:53 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Mon, 1 Nov 2021 09:47:53 +0000 Subject: [Freeswitch-users] log format change breaking fail2ban? percentage added to logs? how to revert? In-Reply-To: References: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl> Message-ID: Yikes! That change silently breaks the fail2ban log line match, meaning no IP addresses will get banned any more. I've raised an issue on the fail2ban project (which holds the config files that match these - and many other products' - log lines) and added a note to the release notes in Confluence. Watch out that debian buster packages only an earlier version of fail2ban, so even when fail2ban release an updated version, you may have to fetch it manually. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Fri, 29 Oct 2021 at 11:20, Dragos Oancea wrote: > > It adds idle cpu to log lines. > check commits 03d5730ffcdd0dc69247693c58239757623b9319 and 241d397e39fad336c13b0f43fecfa697182a4a3b if you wish to revert. > > > On Thu, Oct 28, 2021 at 1:42 PM Jelle de Jong wrote: >> >> Hello everybody, >> >> The log format seem to have changed, how can I get it back to what it >> used to without the percentage thing? A search on Google failed me to >> provide an answer... >> >> # fs_cli -x version >> FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit >> (-release-19-883d2cb662 64bit) >> >> # grep pref_line /etc/fail2ban/filter.d/freeswitch.conf >> _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)? >> failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth >> (failure|challenge) \((REGISTER|INVITE)\) on sofia profile \'[^']+\' for >> \[[^\]]*\] from ip $ >> %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find user >> \[[^@]+@[^\]]+\] from $ >> >> 2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth >> failure (REGISTER) on sofia profile 'internal' for [30640 at 82.94.245.xxx] >> from ip 217.138.220.158 >> >> 2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure >> (REGISTER) on sofia profile 'internal' for [123456 at 82.94.245.xxx] from >> ip 193.107.216.182 >> >> # fgrep percentage /etc/freeswitch/ -R >> >> What is the use case of this new percentage in the log files? >> >> Kind regards, >> >> Jelle de Jong >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 martin at pattersong.co.uk Mon Nov 1 10:46:37 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Mon, 1 Nov 2021 10:46:37 +0000 Subject: [Freeswitch-users] log format change breaking fail2ban? percentage added to logs? how to revert? In-Reply-To: <963bec80-c0d1-b4a2-2a60-45748e0d0959@powercraft.nl> References: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl> <963bec80-c0d1-b4a2-2a60-45748e0d0959@powercraft.nl> Message-ID: _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)?( \d+\.\d\d%%)? works for me. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Fri, 29 Oct 2021 at 21:21, Jelle de Jong wrote: > > Thank you Dragos, looking at the commits there is no configure option to > change the new behaviour. > > I tried creating a new regex line based on the added percentage > > _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+ \d+\.\d+%%)? > > I can not get it to work successfully any-more, if someone has a updated > fail2ban filter for the new log format please share. > > Kind regards, > > Jelle de Jong > > On 10/29/21 12:20 PM, Dragos Oancea wrote: > > It adds idle cpu to log lines. > > check commits 03d5730ffcdd0dc69247693c58239757623b9319 > > and 241d397e39fad336c13b0f43fecfa697182a4a3b if you wish to revert. > > > > > > On Thu, Oct 28, 2021 at 1:42 PM Jelle de Jong > > wrote: > > > > Hello everybody, > > > > The log format seem to have changed, how can I get it back to what it > > used to without the percentage thing? A search on Google failed me to > > provide an answer... > > > > # fs_cli -x version > > FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit > > (-release-19-883d2cb662 64bit) > > > > # grep pref_line /etc/fail2ban/filter.d/freeswitch.conf > > _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)? > > failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth > > (failure|challenge) \((REGISTER|INVITE)\) on sofia profile \'[^']+\' > > for > > \[[^\]]*\] from ip $ > > %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find > > user > > \[[^@]+@[^\]]+\] from $ > > > > 2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth > > failure (REGISTER) on sofia profile 'internal' for > > [30640 at 82.94.245.xxx] > > from ip 217.138.220.158 > > > > 2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure > > (REGISTER) on sofia profile 'internal' for [123456 at 82.94.245.xxx] from > > ip 193.107.216.182 > > > > # fgrep percentage /etc/freeswitch/ -R > > > > What is the use case of this new percentage in the log files? > > > > Kind regards, > > > > Jelle de Jong > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > > https://signalwire.community > > > > 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 > > > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time https://signalwire.community > > > > 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 > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 rahman.duran at erzurum.edu.tr Mon Nov 1 06:15:52 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Mon, 1 Nov 2021 09:15:52 +0300 Subject: [Freeswitch-users] Codec problem, Freeswitch does not regard absolute_codec_string? In-Reply-To: References: Message-ID: Hi, The problem was single quotes around ep_codec string: Once I removed single quotes, it worked. Regards, Rahman Dragos Oancea , 29 Eki 2021 Cum, 14:40 tarihinde şunu yazdı: > maybe the comments on this issue will help: > https://github.com/signalwire/freeswitch/issues/1340 . > > On Wed, Oct 27, 2021 at 8:13 PM Rahman Duran > wrote: > >> Hi, >> >> I am struggling tuning Freeswitch for late codec negotiation. Here is >> local profile settings: >> >> >> >> >> >> >> As you see I enabled late-negotiation and set codecs prefs to allow G722 >> and PCMA. My goal is to use G722 if both sides support it and fall back to >> PCMA if G722 is not an option for A leg or B leg. >> >> Here is the local dial plan config where I bridge calls: >> >> >> >> >> > expression="^true$|^TRUE$|^True$"/> >> >> > data="call_pickup_group=${user_data(${destination_number}@${domain_name} >> var call_pickup_group)}"/> >> > data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >> > data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >> > data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >> >> >> > data="nolocal:absolute_codec_string='${ep_codec_string}'"/> >> > data="callee_id_name=${user_data(${destination_number}@${domain_name} >> var effective_caller_id_name)}"/> >> >> >> >> >> > data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >> var effective_caller_id_name)}}user/${destination_number}@ >> ${domain_name}"/> >> > data="hata-${originate_disposition} XML hata_anonslari"/> >> >> >> >> So I export absolute_codec_string to B leg using ep_codec_string. When I >> test a call from 2730->2110, the call fails with >> "incompetible_destination". Call logs here: >> https://pastebin.freeswitch.org/view/b2978e79 The 2730 is a Cisco phone >> with "G722,PCMU,PCMA,G729,L16,iLBC" codec support. The 2110 is a Samsung >> phone that only supports "PCMA,PCMU,G729" codecs. As you see cisco invites >> with all codecs it supports and freeswitch sets absolute_codec_string with >> "G722,PCMA" as expected from the config. But as you see in the call logs >> Freeswitch only sends "G722" in SDP to Samsung phone so Samsung rejects the >> call. >> >> Why does Freeswitch only send G722? How can I fix this? What am I doing >> wrong? >> >> P.S. using debian packages: FreeSWITCH Version >> 1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit) >> >> Regards, >> >> Rahman Duran >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Antony.Stone at freeswitch.open.source.it Mon Nov 1 16:17:42 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Mon, 1 Nov 2021 17:17:42 +0100 Subject: [Freeswitch-users] log format change breaking fail2ban? percentage added to logs? how to revert? In-Reply-To: References: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl> Message-ID: <202111011717.43010.Antony.Stone@freeswitch.open.source.it> On Friday 29 October 2021 at 12:20:36, Dragos Oancea wrote: > On Thu, Oct 28, 2021 at 1:42 PM Jelle de Jong wrote: > > Hello everybody, > > > > The log format seem to have changed, how can I get it back to what it > > used to without the percentage thing? > > What is the use case of this new percentage in the log files? > It adds idle cpu to log lines. > check commits 03d5730ffcdd0dc69247693c58239757623b9319 > and 241d397e39fad336c13b0f43fecfa697182a4a3b if you wish to revert. That seems a very strange way of doing it to me - to report the *idle* percentage, instead of how *busy* the system is. I believe people generally think of disks as "how full?", memory as "how much used?", network bandwidth as "how much data?" and CPUs as "how busy?" Can anyone explain why it was chosen to report "percentage idle"? Antony. [PS: This message being sent at 16:17 UTC 2021-11-01; this is the fifth email I've sent to the list since I saw my last one get posted on 2021-10-28; I'm still trying to work out why this mailman is so slow / unreliable.] -- "In fact I wanted to be John Cleese and it took me some time to realise that the job was already taken." - Douglas Adams Please reply to the list; please *don't* CC me. From s.safarov at gmail.com Tue Nov 2 05:41:36 2021 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 2 Nov 2021 08:41:36 +0300 Subject: [Freeswitch-users] SIP over DTLS support In-Reply-To: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com> References: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com> Message-ID: WebRTC clients use DTLS encryption. FreeSwitch supports WebRTC. https://nickvsnetworking.com/freeswitch-webrtc-with-sipml5/ On Tue, Nov 2, 2021 at 12:55 AM Telium Technical Support wrote: > Does FreeSwitch support SIP of DTLS? Not clear from the docs (though > mentions of webrtc possibly using this) > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 christian.berger at foncloud.net Tue Nov 2 10:54:59 2021 From: christian.berger at foncloud.net (Christian Berger) Date: Tue, 2 Nov 2021 11:54:59 +0100 Subject: [Freeswitch-users] Growing latency in conferences Message-ID: <3ca249bf-ec81-c9c9-5750-5006c3cb6b5f@foncloud.net> Hi, we have recently come across a problem with high and growing latency during conferences. In order to make sure it has nothing to do with the telephone terminals, we have traced the calls directly at the FreeSwitches. What we see in this experiment is that latency caused by FreeSwitch seems to grow consistently at roughly 200ms per hour. It seems like this is a problem caused by a Jitter Buffer. Does the Conferencing Module use the Jitter Buffer for Bridging or does it use its own buffer. If it uses its own buffer, how can we configure it? Regards, Christian Christian Berger   foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: Christian.Berger at foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon?  Unter https://www.foncloud.net/wissen  finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services.   Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer 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. 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Name: 172.31.27.219-20912-172.31.23.173-34550.dat.diff.png Type: image/png Size: 6740 bytes Desc: not available URL: From jelledejong at powercraft.nl Tue Nov 2 13:10:42 2021 From: jelledejong at powercraft.nl (Jelle de Jong) Date: Tue, 2 Nov 2021 14:10:42 +0100 Subject: [Freeswitch-users] log format change breaking fail2ban? percentage added to logs? how to revert? In-Reply-To: References: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl> <963bec80-c0d1-b4a2-2a60-45748e0d0959@powercraft.nl> Message-ID: Thank You Martin! _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)?( \d+\.\d\d%%)? # fail2ban-client status freeswitch Status for the jail: freeswitch |- Filter | |- Currently failed: 5 | |- Total failed: 3789 | `- File list: /var/log/freeswitch/freeswitch.log `- Actions |- Currently banned: 1 |- Total banned: 114 `- Banned IP list: 217.138.199.182 Kind regards, Jelle de Jong On 11/1/21 11:46 AM, Martin Paterson wrote: > _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)?( \d+\.\d\d%%)? > works for me. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > On Fri, 29 Oct 2021 at 21:21, Jelle de Jong wrote: >> >> Thank you Dragos, looking at the commits there is no configure option to >> change the new behaviour. >> >> I tried creating a new regex line based on the added percentage >> >> _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+ \d+\.\d+%%)? >> >> I can not get it to work successfully any-more, if someone has a updated >> fail2ban filter for the new log format please share. >> >> Kind regards, >> >> Jelle de Jong >> >> On 10/29/21 12:20 PM, Dragos Oancea wrote: >>> It adds idle cpu to log lines. >>> check commits 03d5730ffcdd0dc69247693c58239757623b9319 >>> and 241d397e39fad336c13b0f43fecfa697182a4a3b if you wish to revert. >>> >>> >>> On Thu, Oct 28, 2021 at 1:42 PM Jelle de Jong >> > wrote: >>> >>> Hello everybody, >>> >>> The log format seem to have changed, how can I get it back to what it >>> used to without the percentage thing? A search on Google failed me to >>> provide an answer... >>> >>> # fs_cli -x version >>> FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit >>> (-release-19-883d2cb662 64bit) >>> >>> # grep pref_line /etc/fail2ban/filter.d/freeswitch.conf >>> _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)? >>> failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth >>> (failure|challenge) \((REGISTER|INVITE)\) on sofia profile \'[^']+\' >>> for >>> \[[^\]]*\] from ip $ >>> %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find >>> user >>> \[[^@]+@[^\]]+\] from $ >>> >>> 2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth >>> failure (REGISTER) on sofia profile 'internal' for >>> [30640 at 82.94.245.xxx] >>> from ip 217.138.220.158 >>> >>> 2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure >>> (REGISTER) on sofia profile 'internal' for [123456 at 82.94.245.xxx] from >>> ip 193.107.216.182 >>> >>> # fgrep percentage /etc/freeswitch/ -R >>> >>> What is the use case of this new percentage in the log files? >>> >>> Kind regards, >>> >>> Jelle de Jong >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire >>> https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >>> >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time https://signalwire.community >>> >>> 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 >>> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 rahman.duran at erzurum.edu.tr Mon Nov 1 06:15:52 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Mon, 1 Nov 2021 09:15:52 +0300 Subject: [Freeswitch-users] Codec problem, Freeswitch does not regard absolute_codec_string? In-Reply-To: References: Message-ID: Hi, The problem was single quotes around ep_codec string: Once I removed single quotes, it worked. Regards, Rahman Dragos Oancea , 29 Eki 2021 Cum, 14:40 tarihinde şunu yazdı: > maybe the comments on this issue will help: > https://github.com/signalwire/freeswitch/issues/1340 . > > On Wed, Oct 27, 2021 at 8:13 PM Rahman Duran > wrote: > >> Hi, >> >> I am struggling tuning Freeswitch for late codec negotiation. Here is >> local profile settings: >> >> >> >> >> >> >> As you see I enabled late-negotiation and set codecs prefs to allow G722 >> and PCMA. My goal is to use G722 if both sides support it and fall back to >> PCMA if G722 is not an option for A leg or B leg. >> >> Here is the local dial plan config where I bridge calls: >> >> >> >> >> > expression="^true$|^TRUE$|^True$"/> >> >> > data="call_pickup_group=${user_data(${destination_number}@${domain_name} >> var call_pickup_group)}"/> >> > data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >> > data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >> > data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >> >> >> > data="nolocal:absolute_codec_string='${ep_codec_string}'"/> >> > data="callee_id_name=${user_data(${destination_number}@${domain_name} >> var effective_caller_id_name)}"/> >> >> >> >> >> > data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >> var effective_caller_id_name)}}user/${destination_number}@ >> ${domain_name}"/> >> > data="hata-${originate_disposition} XML hata_anonslari"/> >> >> >> >> So I export absolute_codec_string to B leg using ep_codec_string. When I >> test a call from 2730->2110, the call fails with >> "incompetible_destination". Call logs here: >> https://pastebin.freeswitch.org/view/b2978e79 The 2730 is a Cisco phone >> with "G722,PCMU,PCMA,G729,L16,iLBC" codec support. The 2110 is a Samsung >> phone that only supports "PCMA,PCMU,G729" codecs. As you see cisco invites >> with all codecs it supports and freeswitch sets absolute_codec_string with >> "G722,PCMA" as expected from the config. But as you see in the call logs >> Freeswitch only sends "G722" in SDP to Samsung phone so Samsung rejects the >> call. >> >> Why does Freeswitch only send G722? How can I fix this? What am I doing >> wrong? >> >> P.S. using debian packages: FreeSWITCH Version >> 1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit) >> >> Regards, >> >> Rahman Duran >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Antony.Stone at freeswitch.open.source.it Tue Nov 2 15:21:39 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 2 Nov 2021 16:21:39 +0100 Subject: [Freeswitch-users] Documentation for hold? Message-ID: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> Hi. I want to put a channel on hold in my dialplan. https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools tells me that there is a command "hold" to do this, but the link for the documentation is https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold&action=edit&redlink=1 which does not take me anywhere useful. Can someone point me at the correct page for this command? Thanks, Antony. -- Police have found a cartoonist dead in his house. They say that details are currently sketchy. Please reply to the list; please *don't* CC me. From freeswitch13 at mailbox.org Tue Nov 2 20:03:22 2021 From: freeswitch13 at mailbox.org (Gregor Maier) Date: Tue, 2 Nov 2021 21:03:22 +0100 Subject: Question regarding continuous SIP trace with Freeswitch Message-ID: <9c55457a-62db-5f6c-1720-7980aa1f3325@mailbox.org> Hello! I'm currently struggling how to continuously trace all SIP packages from and to Freeswitch. Tcpdump or sngrep solely or similar tools aren't a solution, because SIP is encrypted (TLS). Asterisk provides an internal interface which logs all (or only restricted hosts e.g.) SIP packages to pcap (after decryption or before encryption of course). I found https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture - but there is no solution which handles the case "registering as client to a providers SIP server" (there is no local private key involved - the provider is the owner of the private key). Voipmonitor is pretty oversized for the planned use case (home server running on an APU 4 device (https://pcengines.ch/apu4d4.htm) on a sd card). Is there a similar and practical way how Freeswitch can log SIP packages to pcap? Thanks Gregor From gmaruzz at gmail.com Wed Nov 3 10:10:03 2021 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 3 Nov 2021 11:10:03 +0100 Subject: [Freeswitch-users] disable video from being automatically added to sdp In-Reply-To: References: Message-ID: On Wed, Oct 27, 2021 at 8:38 PM Jimmy Simpson wrote: > I can not seem to disable and stop Freeswitch from adding the m=video to > the SDP: > you can use channel variable "absolute_codec_string" (check the variable name, I'm going from memory) in dialstring or before bridging, or you can change "outbound_codec_string" (again, check the var name) in general freeswitch config -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From martin at pattersong.co.uk Wed Nov 3 10:24:14 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Wed, 3 Nov 2021 10:24:14 +0000 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> Message-ID: The Wayback machine reveals that there has never been content at that page (https://web.archive.org/web/20150912221709/https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hold), however the applications 'hold' and 'unhold' do exist (do 'show applications' in fs_cli, and they're listed, along with the command syntax). I will add Confluence pages for them, so at least the syntax from the 'show applications' command is listed. I've never used the commands, so I'm not sure I can put much content in the pages, so feel free to add with whatever other information you discover when you use the command. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > Hi. > > I want to put a channel on hold in my dialplan. > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools tells me that > there is a command "hold" to do this, but the link for the documentation is > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold&action=edit&redlink=1 > which does not take me anywhere useful. > > Can someone point me at the correct page for this command? > > > Thanks, > > > Antony. > > -- > Police have found a cartoonist dead in his house. They say that details are > currently sketchy. > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 david.villasmil.work at gmail.com Wed Nov 3 11:13:41 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 3 Nov 2021 11:13:41 +0000 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: Homer? On Wed, 3 Nov 2021 at 03:18, Gregor Maier via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Gregor Maier > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Tue, 2 Nov 2021 21:03:22 +0100 > Subject: Question regarding continuous SIP trace with Freeswitch > > Hello! > > I'm currently struggling how to continuously trace all SIP packages from > and to Freeswitch. Tcpdump or sngrep solely or similar tools aren't a > solution, because SIP is > encrypted (TLS). Asterisk provides an internal interface which logs all > (or only restricted hosts e.g.) SIP packages to pcap (after decryption or > before encryption of course). > > I found > https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture - but > there is no solution which handles the case "registering as client to a > providers SIP > server" (there is no local private key involved - the provider is the > owner of the private key). > > Voipmonitor is pretty oversized for the planned use case (home server > running on an APU 4 device (https://pcengines.ch/apu4d4.htm) on a sd > card). > > Is there a similar and practical way how Freeswitch can log SIP packages > to pcap? > > > Thanks > Gregor > > > > > ---------- Forwarded message ---------- > From: Gregor Maier via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Tue, 02 Nov 2021 20:18:40 -0700 (PDT) > Subject: [Freeswitch-users] Question regarding continuous SIP trace with > Freeswitch > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Wed Nov 3 11:15:14 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Wed, 3 Nov 2021 12:15:14 +0100 Subject: [Freeswitch-users] Sofia and ODBC Message-ID: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> Hi. https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Databases tells me that "Instead of using the default SQLite database as the internal FreeSWITCH database, you can use any core database type. This works for: mod_db mod_voicemail mod_sofia mod_dingaling mod_cidlookup mod_lcr" Each of those is a link, and if I click on the mod_sofia link I get to the page https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia which tells me nothing about using ODBC. If I scroll further down the first page I started from, the heading "Configuring FreeSWITCH to use databases" has a sub-section "mod_sofia" with a link pointing to https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files whose only mention of ODBC is: odbc-dsn If you have ODBC support and a working dsn you can use it instead of SQLite So, where can find the information telling me _how_ I can put my sofia configuration into ODBC, instead of XML files? Basically I'm looking for the equivalent of Asterisk's "Realtime" configuration option, where SIP connections can be held in a database table instead of a flat file, meaning that (a) it's easier to maintain dynamically, and (b) new entries get used without needing a reload. I'm liking FreeSwitch (after using it for about a month), but still finding it sometimes hard to discover the definitive documentation for what it can do (or how to make it do it). Thanks, Antony. -- This sentence contains exacly three erors. Please reply to the list; please *don't* CC me. From brian at freeswitch.com Wed Nov 3 14:46:01 2021 From: brian at freeswitch.com (Brian West) Date: Wed, 3 Nov 2021 09:46:01 -0500 Subject: [Freeswitch-users] Sofia and ODBC In-Reply-To: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> References: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> Message-ID: See XML Curl, not anything related to ODBC. /b On Wed, Nov 3, 2021 at 7:43 AM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > Hi. > > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Databases > tells > me that "Instead of using the default SQLite database as the internal > FreeSWITCH database, you can use any core database type. This works for: > mod_db > mod_voicemail > mod_sofia > mod_dingaling > mod_cidlookup > mod_lcr" > > Each of those is a link, and if I click on the mod_sofia link I get to the > page > https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia which > tells me > nothing about using ODBC. > > If I scroll further down the first page I started from, the heading > "Configuring > FreeSWITCH to use databases" has a sub-section "mod_sofia" with a link > pointing > to > https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files > whose only mention of ODBC is: > > odbc-dsn > If you have ODBC support and a working dsn you can use it instead of SQLite > > > So, where can find the information telling me _how_ I can put my sofia > configuration into ODBC, instead of XML files? > > Basically I'm looking for the equivalent of Asterisk's "Realtime" > configuration > option, where SIP connections can be held in a database table instead of a > flat > file, meaning that (a) it's easier to maintain dynamically, and (b) new > entries > get used without needing a reload. > > I'm liking FreeSwitch (after using it for about a month), but still > finding it > sometimes hard to discover the definitive documentation for what it can do > (or > how to make it do it). > > > Thanks, > > > Antony. > > -- > This sentence contains exacly three erors. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 brian at freeswitch.com Wed Nov 3 14:46:54 2021 From: brian at freeswitch.com (Brian West) Date: Wed, 3 Nov 2021 09:46:54 -0500 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> Message-ID: You just park it and play music, there is NO HOLD/UNHOLD in the manner you're needing. /b On Wed, Nov 3, 2021 at 6:11 AM Martin Paterson wrote: > The Wayback machine reveals that there has never been content at that > page ( > https://web.archive.org/web/20150912221709/https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hold > ), > however the applications 'hold' and 'unhold' do exist (do 'show > applications' in fs_cli, and they're listed, along with the command > syntax). I will add Confluence pages for them, so at least the syntax > from the 'show applications' command is listed. I've never used the > commands, so I'm not sure I can put much content in the pages, so feel > free to add with whatever other information you discover when you use > the command. > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > On Tue, 2 Nov 2021 at 15:21, Antony Stone > wrote: > > > > Hi. > > > > I want to put a channel on hold in my dialplan. > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools tells > me that > > there is a command "hold" to do this, but the link for the documentation > is > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold&action=edit&redlink=1 > > which does not take me anywhere useful. > > > > Can someone point me at the correct page for this command? > > > > > > Thanks, > > > > > > Antony. > > > > -- > > Police have found a cartoonist dead in his house. They say that details > are > > currently sketchy. > > > > Please reply to the > list; > > please *don't* > CC me. > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 martin at pattersong.co.uk Wed Nov 3 15:33:45 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Wed, 3 Nov 2021 15:33:45 +0000 Subject: [Freeswitch-users] Sofia and ODBC In-Reply-To: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> References: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> Message-ID: I think you are confusing two different things here. The docs pages you link to are about choosing the internal database that FreeSWITCH uses for its internal state information. This is not the same as serving configuration, which is by default provided in XML files. You can dynamically serve the XML configuration with the module: mod_xml_curl and also control call flow with mod_httapi. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Wed, 3 Nov 2021 at 11:15, Antony Stone wrote: > > Hi. > > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Databases tells > me that "Instead of using the default SQLite database as the internal > FreeSWITCH database, you can use any core database type. This works for: > mod_db > mod_voicemail > mod_sofia > mod_dingaling > mod_cidlookup > mod_lcr" > > Each of those is a link, and if I click on the mod_sofia link I get to the page > https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia which tells me > nothing about using ODBC. > > If I scroll further down the first page I started from, the heading "Configuring > FreeSWITCH to use databases" has a sub-section "mod_sofia" with a link pointing > to https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files > whose only mention of ODBC is: > > odbc-dsn > If you have ODBC support and a working dsn you can use it instead of SQLite > > > So, where can find the information telling me _how_ I can put my sofia > configuration into ODBC, instead of XML files? > > Basically I'm looking for the equivalent of Asterisk's "Realtime" configuration > option, where SIP connections can be held in a database table instead of a flat > file, meaning that (a) it's easier to maintain dynamically, and (b) new entries > get used without needing a reload. > > I'm liking FreeSwitch (after using it for about a month), but still finding it > sometimes hard to discover the definitive documentation for what it can do (or > how to make it do it). > > > Thanks, > > > Antony. > > -- > This sentence contains exacly three erors. > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 mike at freeswitch.org Wed Nov 3 20:38:12 2021 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 3 Nov 2021 15:38:12 -0500 Subject: [Freeswitch-users] SIP over DTLS support In-Reply-To: References: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com> Message-ID: DTLS is used for media, not for signaling. The question you intended to be asking the answer is likely yes, the question you actually asked the answer is no. > On Nov 2, 2021, at 12:41 AM, Sergey Safarov wrote: > > WebRTC clients use DTLS encryption. > FreeSwitch supports WebRTC. > > https://nickvsnetworking.com/freeswitch-webrtc-with-sipml5/ > > > On Tue, Nov 2, 2021 at 12:55 AM Telium Technical Support > wrote: > Does FreeSwitch support SIP of DTLS? Not clear from the docs (though mentions of webrtc possibly using this) > > _ -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Wed Nov 3 21:00:14 2021 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 3 Nov 2021 16:00:14 -0500 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> Message-ID: It’s a module abandoned 7-8 years ago in the contrib repo that we have never maintained and has never been part of the FS packaging. We generally don’t package unmaintained code but you are welcome to use it if you choose and it still works. The db layers have gone through some evolution since then, what exactly are you trying to accomplish, it may already be available elsewhere. > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > Hi. > > Can anyone tell me why > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query is not > built as part of the standard Debian packages available from > https://files.freeswitch.org/repo/deb/ ? > > Thanks, > > > Antony. > > (PS: This message sent at 15:21 UTC 2021-10-29, just wondering when it will > appear on the list.) From support at telium.io Wed Nov 3 22:14:45 2021 From: support at telium.io (Telium Technical Support) Date: Wed, 3 Nov 2021 22:14:45 +0000 Subject: [Freeswitch-users] SIP over DTLS support In-Reply-To: References: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com> Message-ID: <0100017ce7de67b8-fb49b6d0-f637-43a5-8bb9-33f196ae5f79-000000@email.amazonses.com> I actually meant SIP over DTLS, here is an overview: https://www.ietf.org/archive/id/draft-jennings-sip-dtls-05.html I have to dig for an IETF number (assuming it was ratified). But I think the answer is still NO.. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Jerris Sent: Wednesday, November 3, 2021 4:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP over DTLS support DTLS is used for media, not for signaling. The question you intended to be asking the answer is likely yes, the question you actually asked the answer is no. On Nov 2, 2021, at 12:41 AM, Sergey Safarov > wrote: WebRTC clients use DTLS encryption. FreeSwitch supports WebRTC. https://nickvsnetworking.com/freeswitch-webrtc-with-sipml5/ On Tue, Nov 2, 2021 at 12:55 AM Telium Technical Support > wrote: Does FreeSwitch support SIP of DTLS? Not clear from the docs (though mentions of webrtc possibly using this) _ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jerry.Kendall at BishopHosting.com Wed Nov 3 12:47:29 2021 From: Jerry.Kendall at BishopHosting.com (Jerry Kendall) Date: Wed, 03 Nov 2021 08:47:29 -0400 Subject: [Freeswitch-users] Sofia and ODBC In-Reply-To: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> Message-ID: Lookup "ODBC in the core"Sent from my Galaxy -------- Original message --------From: Antony Stone Date: 2021-11-03 8:41 AM (GMT-05:00) To: FreeSWITCH Users Help Subject: [Freeswitch-users] Sofia and ODBC Hi.https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Databases tells me that "Instead of using the default SQLite database as the internal FreeSWITCH database, you can use any core database type. This works for: mod_db mod_voicemail mod_sofia mod_dingaling mod_cidlookup mod_lcr"Each of those is a link, and if I click on the mod_sofia link I get to the page https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia which tells me nothing about using ODBC.If I scroll further down the first page I started from, the heading "Configuring FreeSWITCH to use databases" has a sub-section "mod_sofia" with a link pointing to https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files whose only mention of ODBC is:odbc-dsn If you have ODBC support and a working dsn you can use it instead of SQLiteSo, where can find the information telling me _how_ I can put my sofia configuration into ODBC, instead of XML files?Basically I'm looking for the equivalent of Asterisk's "Realtime" configuration option, where SIP connections can be held in a database table instead of a flat file, meaning that (a) it's easier to maintain dynamically, and (b) new entries get used without needing a reload.I'm liking FreeSwitch (after using it for about a month), but still finding it sometimes hard to discover the definitive documentation for what it can do (or how to make it do it).Thanks,Antony.-- This sentence contains exacly three erors.                                                   Please reply to the list;                                                         please *don't* CC me._________________________________________________________________________The FreeSWITCH project is sponsored by SignalWire https://signalwire.comEnhance your FreeSWITCH install with disruptive priced SMS and PSTN services.Build your next product on our scalable cloud platform.Join our online community to chat in real time https://signalwire.communityProfessional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.comOfficial FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Wed Nov 3 15:49:01 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Wed, 3 Nov 2021 16:49:01 +0100 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111031649.01531.Antony.Stone@freeswitch.open.source.it> On Wednesday 03 November 2021 at 15:46:54, Brian West wrote: > You just park it and play music, there is NO HOLD/UNHOLD in the manner > you're needing. No, this is not for placing the call on hold inside FreeSwitch - this is for sending a re-Invite out to an upstream PBX to put the call on hold there (a re-Invite with SDP containing a=sendonly). > > On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > > Hi. > > > > > > I want to put a channel on hold in my dialplan. > > > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools tells > > > me that there is a command "hold" to do this, but the link for the > > > documentation is > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold > > > &action=edit&redlink=1 > > > > > which does not take me anywhere useful. > > > > > > Can someone point me at the correct page for this command? > > > > > > > > > Thanks, > > > > > > > > > Antony. -- Normal people think "If it ain't broke, don't fix it". Engineers think "If it ain't broke, it doesn't have enough features yet". Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Wed Nov 3 16:01:16 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Wed, 3 Nov 2021 17:01:16 +0100 Subject: [Freeswitch-users] Sofia and ODBC In-Reply-To: References: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111031701.16988.Antony.Stone@freeswitch.open.source.it> On Wednesday 03 November 2021 at 16:33:45, Martin Paterson wrote: > I think you are confusing two different things here. The docs pages > you link to are about choosing the internal database that FreeSWITCH > uses for its internal state information. This is not the same as > serving configuration, which is by default provided in XML files. > > You can dynamically serve the XML configuration with the module: > mod_xml_curl and also control call flow with mod_httapi. Aha - that clarifies greatly - thank you. I'm surprised that mod_xml_curl is not mentioned on the first page I found about FreeSwitch and databases in general: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Databases Antony. -- I bought a book on memory techniques, but I've forgotten where I put it. Please reply to the list; please *don't* CC me. From gmaruzz at gmail.com Thu Nov 4 09:12:39 2021 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 4 Nov 2021 10:12:39 +0100 Subject: [Freeswitch-users] Sofia and ODBC In-Reply-To: <202111031701.16988.Antony.Stone@freeswitch.open.source.it> References: <202111031215.14875.Antony.Stone@freeswitch.open.source.it> <202111031701.16988.Antony.Stone@freeswitch.open.source.it> Message-ID: On Thu, Nov 4, 2021 at 3:11 AM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Wednesday 03 November 2021 at 16:33:45, Martin Paterson wrote: > > > I'm surprised that mod_xml_curl is not mentioned on the first page I found > about FreeSwitch and databases in general: > > because has nothing to do with DBs, and all to do with xml and curl (http/s)? ;) -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From remi.marand at odigo.com Thu Nov 4 09:58:07 2021 From: remi.marand at odigo.com (=?iso-8859-1?Q?R=E9mi_MARAND?=) Date: Thu, 4 Nov 2021 09:58:07 +0000 Subject: [Freeswitch-users] RE - Internal Interface suddenly freezes In-Reply-To: References: <9CDD4C05-0E51-4CFC-838B-763828D563F0@wnt.at> Message-ID: Hello, We have upgraded to 1.10.6 and use mod_mariadb as advised. Unfortunately, we still have these registrations stops. it is not frequent and difficult to reproduce. I would like to test the multi-thread registration as preconized, I tried but there is no corresponding info in logs. Should you confirm that the parameters : Must be inserted in the "global_settings" part of the sofia.conf.xml file ?? How should I check the effective multithread registration process ?? Best regards. Rémi Marand. De : FreeSWITCH-users De la part de Stefan Kainz Envoyé : mardi 25 mai 2021 13:15 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] RE - Internal Interface suddenly freezes Thank you. We will try 1.10.6 with mod_mariadb this week! Thanks! Von: FreeSWITCH-users > Im Auftrag von Rémi MARAND Gesendet: Montag, 24. Mai 2021 19:24 An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] RE - Internal Interface suddenly freezes Hello, Thank you for the various information about this incident on the userlist and github. We are using Postgresql on Freeswitch server itself. The version of Freeswitch is 1.10.5-release-17-25569c1631~64bit On Debian 10. I cannot reproduce the problem on our development environment with thousands of registrations and thousands of calls, for this reason I suspect some unexpected behavior of the web client, and I note that a few TCP connections: on wss:7443 remain after the Register they carry is completed. I am looking for a solution on the Freeswitch side to close them on inactivity timeout. If we are successful in causing the problem, we will release the core as you recommend. As for the transition to version 1.10.6, this is a track that we will try. Best Regards. remi.marand at odigo.com De : FreeSWITCH-users > De la part de Stefan Kainz Envoyé : lundi 24 mai 2021 17:42 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] RE - Internal Interface suddenly freezes Another one :) At this point we have a script that restarts freeswitch every time this happens. We don't really have a solution for this so far, and are out of things to test ... May I ask if you have an external interface in your setup? Does that interface freeze too? We have a external interface, and it looks like it doesn't happen to that one. We also have a third interface called "alternate_routing", which doesn't handle registrations or presence. Like the external interface it doesn't freeze. Do you use Mariadb/ODBC for odbc-core? Im hoping the GitHub-issue gets some attention soon. Maybe it helps if some of you also post a backtrace when this happens. To your problem, can you confirm that this isn't some sort of SYN-Flood-Attack? Maybe check the IP's that are connected. I think its pretty easy to find out if those are real-users or hackers. Regards, On 22.05.2021, at 08:22, Rémi MARAND > wrote: Hello, It seems we have the same problem, it doesn't happen often, but the Register part of internal profile does crash. I observe that the number of connections established on the WSS port (7443) sometimes exceeds the number of active registrations. (netstat -tpa).. It could be really important (2 or 3 times the number of Registrations), it could of course be a software problem on the WebRTC (Front) client side. Is there a way on the Freeswitch side in the configuration to disconnect TLS connections that are no longer registered? On timeout for example. Best regards. Rémi Marand. remi.marand at odigo.com De : FreeSWITCH-users > De la part de Kakiman Expert Envoyé : vendredi 30 avril 2021 15:34 À : freeswitch-users at lists.freeswitch.org Objet : [Freeswitch-users] RE - Internal Interface suddenly freezes hello Did you find any solution to this issue ? I am in production with 1.10.5 version and since a week, I have the same issue. The internal profile is out of service and registration is impossible. and we don't have any answer to "sofia status profile internal" , the prompt in fs_cli keeps waiting. the only way to solve is to restart freeswitch thanks regards _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 covici at ccs.covici.com Thu Nov 4 12:10:19 2021 From: covici at ccs.covici.com (John Covici) Date: Thu, 04 Nov 2021 08:10:19 -0400 Subject: [Freeswitch-users] RE - Internal Interface suddenly freezes In-Reply-To: References: <9CDD4C05-0E51-4CFC-838B-763828D563F0@wnt.at> Message-ID: Those sound like profile parameters to me -- are you sure about where they go? On Thu, 04 Nov 2021 05:58:07 -0400, Rémi MARAND wrote: > > [1 ] > [1.1 ] > Hello, > > We have upgraded to 1.10.6 and use mod_mariadb as advised. > > Unfortunately, we still have these registrations stops. it is not frequent and difficult to reproduce. > > I would like to test the multi-thread registration as preconized, I tried but there is no corresponding info in logs. > > Should you confirm that the parameters : > > > Must be inserted in the "global_settings" part of the sofia.conf.xml file ?? > How should I check the effective multithread registration process ?? > > Best regards. > > Rémi Marand. > > De : FreeSWITCH-users De la part de Stefan Kainz > Envoyé : mardi 25 mai 2021 13:15 > À : FreeSWITCH Users Help > Objet : Re: [Freeswitch-users] RE - Internal Interface suddenly freezes > > Thank you. > > We will try 1.10.6 with mod_mariadb this week! > > Thanks! > > Von: FreeSWITCH-users > Im Auftrag von Rémi MARAND > Gesendet: Montag, 24. Mai 2021 19:24 > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] RE - Internal Interface suddenly freezes > > Hello, > > Thank you for the various information about this incident on the userlist and github. > > We are using Postgresql on Freeswitch server itself. > The version of Freeswitch is 1.10.5-release-17-25569c1631~64bit On Debian 10. > > I cannot reproduce the problem on our development environment with thousands of registrations and thousands of calls, for this reason I suspect some unexpected behavior of the web client, and I note that a few TCP connections: on wss:7443 remain after the Register they carry is completed. > I am looking for a solution on the Freeswitch side to close them on inactivity timeout. > > If we are successful in causing the problem, we will release the core as you recommend. > As for the transition to version 1.10.6, this is a track that we will try. > > Best Regards. > > remi.marand at odigo.com > > De : FreeSWITCH-users > De la part de Stefan Kainz > Envoyé : lundi 24 mai 2021 17:42 > À : FreeSWITCH Users Help > Objet : Re: [Freeswitch-users] RE - Internal Interface suddenly freezes > > Another one :) > > At this point we have a script that restarts freeswitch every time this happens. > We don't really have a solution for this so far, and are out of things to test ... > > May I ask if you have an external interface in your setup? Does that interface freeze too? > We have a external interface, and it looks like it doesn't happen to that one. > We also have a third interface called "alternate_routing", which doesn't handle registrations or presence. Like the external interface it doesn't freeze. > > Do you use Mariadb/ODBC for odbc-core? > > Im hoping the GitHub-issue gets some attention soon. > Maybe it helps if some of you also post a backtrace when this happens. > > To your problem, can you confirm that this isn't some sort of SYN-Flood-Attack? > Maybe check the IP's that are connected. I think its pretty easy to find out if those are real-users or hackers. > > Regards, > > On 22.05.2021, at 08:22, Rémi MARAND > wrote: > > Hello, > > It seems we have the same problem, it doesn't happen often, but the Register part of internal profile does crash. > > I observe that the number of connections established on the WSS port (7443) sometimes exceeds the number of active registrations. (netstat -tpa).. > It could be really important (2 or 3 times the number of Registrations), it could of course be a software problem on the WebRTC (Front) client side. > > Is there a way on the Freeswitch side in the configuration to disconnect TLS connections that are no longer registered? On timeout for example. > > Best regards. > > Rémi Marand. > remi.marand at odigo.com > > De : FreeSWITCH-users > De la part de Kakiman Expert > Envoyé : vendredi 30 avril 2021 15:34 > À : freeswitch-users at lists.freeswitch.org > Objet : [Freeswitch-users] RE - Internal Interface suddenly freezes > > hello > > Did you find any solution to this issue ? > > I am in production with 1.10.5 version and since a week, I have the same issue. > The internal profile is out of service and registration is impossible. > > and we don't have any answer to "sofia status profile internal" , the prompt in fs_cli keeps waiting. > > the only way to solve is to restart freeswitch > > thanks > regards > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 > > [1.2 ] > [2 ] > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From kaiduanx at yahoo.ca Thu Nov 4 13:28:14 2021 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 4 Nov 2021 13:28:14 +0000 (UTC) Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: <202111031649.01531.Antony.Stone@freeswitch.open.source.it> References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> <202111031649.01531.Antony.Stone@freeswitch.open.source.it> Message-ID: <1556615656.300528.1636032494651@mail.yahoo.com> Antony, For hold FS does not send reINVITE with SDP containing a=sendonly to upstream PBX. You need to change the source code to do that. /Kaiduan On Wednesday, November 3, 2021, 10:11:37 p.m. EDT, Antony Stone wrote: On Wednesday 03 November 2021 at 15:46:54, Brian West wrote: > You just park it and play music, there is NO HOLD/UNHOLD in the manner > you're needing. No, this is not for placing the call on hold inside FreeSwitch - this is for sending a re-Invite out to an upstream PBX to put the call on hold there (a re-Invite with SDP containing a=sendonly). > > On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > > Hi. > > > > > > I want to put a channel on hold in my dialplan. > > > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools tells > > > me that there is a command "hold" to do this, but the link for the > > > documentation is > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold > > > &action=edit&redlink=1 > > > > > which does not take me anywhere useful. > > > > > > Can someone point me at the correct page for this command? > > > > > > > > > Thanks, > > > > > > > > > Antony. -- Normal people think "If it ain't broke, don't fix it". Engineers think "If it ain't broke, it doesn't have enough features yet".                                                   Please reply to the list;                                                         please *don't* CC me. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 kaiduanx at yahoo.ca Thu Nov 4 13:29:53 2021 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 4 Nov 2021 13:29:53 +0000 (UTC) Subject: [Freeswitch-users] SIP over DTLS support In-Reply-To: <0100017ce7de67b8-fb49b6d0-f637-43a5-8bb9-33f196ae5f79-000000@email.amazonses.com> References: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com> <0100017ce7de67b8-fb49b6d0-f637-43a5-8bb9-33f196ae5f79-000000@email.amazonses.com> Message-ID: <327275729.371981.1636032593903@mail.yahoo.com> Freeswitch does not support SIP over DTLS. Is SIP over TLS is not good enough for your case? /Kaiduan On Wednesday, November 3, 2021, 10:09:47 p.m. EDT, Telium Technical Support wrote: I actually meant SIP over DTLS, here is an overview: https://www.ietf.org/archive/id/draft-jennings-sip-dtls-05.html   I have to dig for an IETF number (assuming it was ratified).  But I think the answer is still NO….   From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Jerris Sent: Wednesday, November 3, 2021 4:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP over DTLS support   DTLS is used for media, not for signaling.  The question you intended to be asking the answer is likely yes, the question you actually asked the answer is no. On Nov 2, 2021, at 12:41 AM, Sergey Safarov wrote:   WebRTC clients use DTLS encryption. FreeSwitch supports WebRTC.   https://nickvsnetworking.com/freeswitch-webrtc-with-sipml5/     On Tue, Nov 2, 2021 at 12:55 AM Telium Technical Support wrote: Does FreeSwitch support SIP of DTLS?  Not clear from the docs (though mentions of webrtc possibly using this) _ _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 support at telium.io Thu Nov 4 14:00:37 2021 From: support at telium.io (Telium Technical Support) Date: Thu, 4 Nov 2021 14:00:37 +0000 Subject: [Freeswitch-users] SIP over DTLS support In-Reply-To: References: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com> <0100017ce7de67b8-fb49b6d0-f637-43a5-8bb9-33f196ae5f79-000000@email.amazonses.com> Message-ID: <0100017ceb405fab-f2421b77-65a0-461a-9c86-681d25fdd3d9-000000@email.amazonses.com> How does FS resume the call legs (after recovery) if the SIP session is closed ? (if using TLS) is only the RTP resumed and the next SIP communication will cause the call to drop? -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kaiduan xie via FreeSWITCH-users Sent: None To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SIP over DTLS support _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 Antony.Stone at freeswitch.open.source.it Thu Nov 4 09:47:06 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 4 Nov 2021 10:47:06 +0100 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111041047.06813.Antony.Stone@freeswitch.open.source.it> On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: > It’s a module abandoned 7-8 years ago in the contrib repo that we have > never maintained and has never been part of the FS packaging. We > generally don’t package unmaintained code but you are welcome to use it if > you choose and it still works. The db layers have gone through some > evolution since then, what exactly are you trying to accomplish, it may > already be available elsewhere. I have a few use cases: 1. Inbound calls come in to a DDI range I have, and need to be forwarded on to users' mobile phones. This list of DDI --> mobile number mappings is fairly large, and maintained externally to the telephony platform, therefore I want to do a dynamic lookup of the DDI to find out which mobile number to forward a call on to. 2. Users dialling out have different destinations they're allowed to call, and again the details about who is allowed to call where are maintained by people nothing to do with operating the telephony platform, so I want to do a database lookup of each user as they place a call to see whether the destination they're dialling is in their allowed list. 3. Outbound providers charge us for placing calls to different destinations, and the prices change (typically monthly, but sometimes more often), and I need to do accurate live per-call accounting to keep track of users' and groups' spend levels. I've done all the above in Asterisk using its ODBC connectivity, so I'm looking for the best way to execute arbitrary SQL statements in a FreeSwitch dialplan. > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > > > Hi. > > > > Can anyone tell me why > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query is > > not built as part of the standard Debian packages available from > > https://files.freeswitch.org/repo/deb/ ? > > > > Thanks, > > > > > > Antony. -- "640 kilobytes (of RAM) should be enough for anybody." - Bill Gates Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Thu Nov 4 13:07:09 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 4 Nov 2021 14:07:09 +0100 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: <202111031649.01531.Antony.Stone@freeswitch.open.source.it> References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> <202111031649.01531.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111041407.09601.Antony.Stone@freeswitch.open.source.it> On Wednesday 03 November 2021 at 16:49:01, Antony Stone wrote: > On Wednesday 03 November 2021 at 15:46:54, Brian West wrote: > > You just park it and play music, there is NO HOLD/UNHOLD in the manner > > you're needing. > > No, this is not for placing the call on hold inside FreeSwitch - this is > for sending a re-Invite out to an upstream PBX to put the call on hold > there (a re-Invite with SDP containing a=sendonly). For reference, I've achieved what I needed using ${uuid_hold(channel)} https://freeswitch.org/confluence/display/FREESWITCH/mod_commands > > > On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > > > Hi. > > > > > > > > I want to put a channel on hold in my dialplan. > > > > > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools > > > > tells me that there is a command "hold" to do this, but the link for > > > > the documentation is > > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold > > > > &action=edit&redlink=1 > > > > > > > > which does not take me anywhere useful. > > > > > > > > Can someone point me at the correct page for this command? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Antony. -- Some things the German language doesn't easily distinguish between: - slugs and snails - cucumbers and gherkins - snakes and queues - wearing something, or carrying it - mothers and nuts - driving a car, riding a bicycle, or travelling by train - a man and a husband - a woman and a wife - changing clothes and moving house - pockets and bags Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Thu Nov 4 14:12:07 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 4 Nov 2021 15:12:07 +0100 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> <202111031649.01531.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111041512.07310.Antony.Stone@freeswitch.open.source.it> On Thursday 04 November 2021 at 14:59:53, kaiduan xie via FreeSWITCH-users wrote: > Antony, > For hold FS does not send reINVITE with SDP containing a=sendonly to > upstream PBX. You need to change the source code to do that. Er, yes it does. I now have a working solution. Thanks to all who helped me in the right direction. > On Wednesday, November 3, 2021, 10:11:37 p.m. EDT, Antony Stone wrote: > > On Wednesday 03 November 2021 at 15:46:54, Brian West wrote: > > You just park it and play music, there is NO HOLD/UNHOLD in the manner > > you're needing. > > No, this is not for placing the call on hold inside FreeSwitch - this is > for sending a re-Invite out to an upstream PBX to put the call on hold > there (a re-Invite with SDP containing a=sendonly). > > > > On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > > > Hi. > > > > > > > > I want to put a channel on hold in my dialplan. > > > > > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools > > > > tells me that there is a command "hold" to do this, but the link for > > > > the documentation is > > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold > > > > &action=edit&redlink=1 > > > > > > > > which does not take me anywhere useful. > > > > > > > > Can someone point me at the correct page for this command? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Antony. -- "Remember: the S in IoT stands for Security." - Jan-Piet Mens Please reply to the list; please *don't* CC me. From covici at ccs.covici.com Thu Nov 4 19:24:03 2021 From: covici at ccs.covici.com (John Covici) Date: Thu, 04 Nov 2021 15:24:03 -0400 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <202111041047.06813.Antony.Stone@freeswitch.open.source.it> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> Message-ID: I think you should be able to dl all this with mod_xml_curl. This module will give you the xml you need after searching whatever database you need -- and it can be on a per-call basis. One thing though, you might be better using a real programming language for these use cases. Freeswitch is more of a toolkit whereas asterisk is more of a pbx specific arrangement. On Thu, 04 Nov 2021 05:47:06 -0400, Antony Stone wrote: > > On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: > > > It’s a module abandoned 7-8 years ago in the contrib repo that we have > > never maintained and has never been part of the FS packaging. We > > generally don’t package unmaintained code but you are welcome to use it if > > you choose and it still works. The db layers have gone through some > > evolution since then, what exactly are you trying to accomplish, it may > > already be available elsewhere. > > I have a few use cases: > > 1. Inbound calls come in to a DDI range I have, and need to be forwarded on to > users' mobile phones. This list of DDI --> mobile number mappings is fairly > large, and maintained externally to the telephony platform, therefore I want > to do a dynamic lookup of the DDI to find out which mobile number to forward a > call on to. > > 2. Users dialling out have different destinations they're allowed to call, and > again the details about who is allowed to call where are maintained by people > nothing to do with operating the telephony platform, so I want to do a > database lookup of each user as they place a call to see whether the > destination they're dialling is in their allowed list. > > 3. Outbound providers charge us for placing calls to different destinations, > and the prices change (typically monthly, but sometimes more often), and I > need to do accurate live per-call accounting to keep track of users' and > groups' spend levels. > > I've done all the above in Asterisk using its ODBC connectivity, so I'm > looking for the best way to execute arbitrary SQL statements in a FreeSwitch > dialplan. > > > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > > > > > Hi. > > > > > > Can anyone tell me why > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query is > > > not built as part of the standard Debian packages available from > > > https://files.freeswitch.org/repo/deb/ ? > > > > > > Thanks, > > > > > > > > > Antony. > > -- > "640 kilobytes (of RAM) should be enough for anybody." > > - Bill Gates > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From avi at avimarcus.net Thu Nov 4 19:41:43 2021 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 4 Nov 2021 19:41:43 +0000 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> Message-ID: <0100017cec78aa7d-a9b8fdb3-9cf8-482c-84d0-3140f6b33686-000000@email.amazonses.com> mod_xml_curl seems a bit heavy for this. You can write a short js or lua script to do the odbc query and assign it to a channel variable. I'm not sure about accessing a _foreign_ odbc source though, or if it only accesses the freeswitch core db. The info isn't loading for me... https://freeswitch.org/confluence/display/FREESWITCH/Lua+FreeSWITCH+Dbh -Avi Marcus BestFone On Thu, Nov 4, 2021 at 9:24 PM John Covici wrote: > I think you should be able to dl all this with mod_xml_curl. This > module will give you the xml you need after searching whatever > database you need -- and it can be on a per-call basis. One thing > though, you might be better using a real programming language for > these use cases. Freeswitch is more of a toolkit whereas asterisk is > more of a pbx specific arrangement. > > On Thu, 04 Nov 2021 05:47:06 -0400, > Antony Stone wrote: > > > > On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: > > > > > It’s a module abandoned 7-8 years ago in the contrib repo that we have > > > never maintained and has never been part of the FS packaging. We > > > generally don’t package unmaintained code but you are welcome to use > it if > > > you choose and it still works. The db layers have gone through some > > > evolution since then, what exactly are you trying to accomplish, it may > > > already be available elsewhere. > > > > I have a few use cases: > > > > 1. Inbound calls come in to a DDI range I have, and need to be forwarded > on to > > users' mobile phones. This list of DDI --> mobile number mappings is > fairly > > large, and maintained externally to the telephony platform, therefore I > want > > to do a dynamic lookup of the DDI to find out which mobile number to > forward a > > call on to. > > > > 2. Users dialling out have different destinations they're allowed to > call, and > > again the details about who is allowed to call where are maintained by > people > > nothing to do with operating the telephony platform, so I want to do a > > database lookup of each user as they place a call to see whether the > > destination they're dialling is in their allowed list. > > > > 3. Outbound providers charge us for placing calls to different > destinations, > > and the prices change (typically monthly, but sometimes more often), and > I > > need to do accurate live per-call accounting to keep track of users' and > > groups' spend levels. > > > > I've done all the above in Asterisk using its ODBC connectivity, so I'm > > looking for the best way to execute arbitrary SQL statements in a > FreeSwitch > > dialplan. > > > > > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > > > > > > > Hi. > > > > > > > > Can anyone tell me why > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query > is > > > > not built as part of the standard Debian packages available from > > > > https://files.freeswitch.org/repo/deb/ ? > > > > > > > > Thanks, > > > > > > > > > > > > Antony. > > > > -- > > "640 kilobytes (of RAM) should be enough for anybody." > > > > - Bill Gates > > > > Please reply to the > list; > > please *don't* > CC me. > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Thu Nov 4 23:13:28 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 4 Nov 2021 23:13:28 +0000 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <0100017cec78aa7d-a9b8fdb3-9cf8-482c-84d0-3140f6b33686-000000@email.amazonses.com> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> <0100017cec78aa7d-a9b8fdb3-9cf8-482c-84d0-3140f6b33686-000000@email.amazonses.com> Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Nov 4, 2021 at 8:15 PM Avi Marcus wrote: > mod_xml_curl seems a bit heavy for this. > > You can write a short js or lua script to do the odbc query and assign it > to a channel variable. > > I'm not sure about accessing a _foreign_ odbc source though, or if it only > accesses the freeswitch core db. > > The info isn't loading for me... > https://freeswitch.org/confluence/display/FREESWITCH/Lua+FreeSWITCH+Dbh > > > > -Avi Marcus > BestFone > > On Thu, Nov 4, 2021 at 9:24 PM John Covici wrote: > >> I think you should be able to dl all this with mod_xml_curl. This >> module will give you the xml you need after searching whatever >> database you need -- and it can be on a per-call basis. One thing >> though, you might be better using a real programming language for >> these use cases. Freeswitch is more of a toolkit whereas asterisk is >> more of a pbx specific arrangement. >> >> On Thu, 04 Nov 2021 05:47:06 -0400, >> Antony Stone wrote: >> > >> > On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: >> > >> > > It’s a module abandoned 7-8 years ago in the contrib repo that we have >> > > never maintained and has never been part of the FS packaging. We >> > > generally don’t package unmaintained code but you are welcome to use >> it if >> > > you choose and it still works. The db layers have gone through some >> > > evolution since then, what exactly are you trying to accomplish, it >> may >> > > already be available elsewhere. >> > >> > I have a few use cases: >> > >> > 1. Inbound calls come in to a DDI range I have, and need to be >> forwarded on to >> > users' mobile phones. This list of DDI --> mobile number mappings is >> fairly >> > large, and maintained externally to the telephony platform, therefore I >> want >> > to do a dynamic lookup of the DDI to find out which mobile number to >> forward a >> > call on to. >> > >> > 2. Users dialling out have different destinations they're allowed to >> call, and >> > again the details about who is allowed to call where are maintained by >> people >> > nothing to do with operating the telephony platform, so I want to do a >> > database lookup of each user as they place a call to see whether the >> > destination they're dialling is in their allowed list. >> > >> > 3. Outbound providers charge us for placing calls to different >> destinations, >> > and the prices change (typically monthly, but sometimes more often), >> and I >> > need to do accurate live per-call accounting to keep track of users' >> and >> > groups' spend levels. >> > >> > I've done all the above in Asterisk using its ODBC connectivity, so I'm >> > looking for the best way to execute arbitrary SQL statements in a >> FreeSwitch >> > dialplan. >> > >> > > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: >> > > > >> > > > Hi. >> > > > >> > > > Can anyone tell me why >> > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query >> is >> > > > not built as part of the standard Debian packages available from >> > > > https://files.freeswitch.org/repo/deb/ ? >> > > > >> > > > Thanks, >> > > > >> > > > >> > > > Antony. >> > >> > -- >> > "640 kilobytes (of RAM) should be enough for anybody." >> > >> > - Bill Gates >> > >> > Please reply to the >> list; >> > please *don't* >> CC me. >> > >> > >> _________________________________________________________________________ >> > >> > The FreeSWITCH project is sponsored by SignalWire >> https://signalwire.com >> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> > Build your next product on our scalable cloud platform. >> > >> > Join our online community to chat in real time >> https://signalwire.community >> > >> > 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 >> > >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici wb2una >> covici at ccs.covici.com >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Fri Nov 5 00:49:48 2021 From: krice at freeswitch.org (krice at freeswitch.org) Date: Thu, 4 Nov 2021 19:49:48 -0500 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <202111041047.06813.Antony.Stone@freeswitch.open.source.it> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> Message-ID: <604401d7d1df$095497d0$1bfdc770$@freeswitch.org> This is specifically what the curl extensions are for. you bind the dialplan there and ask the DB for exactly what to do with the call. -----Original Message----- From: FreeSWITCH-users On Behalf Of Antony Stone Sent: Thursday, November 4, 2021 4:47 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Developers / packagers here? On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: > It’s a module abandoned 7-8 years ago in the contrib repo that we have > never maintained and has never been part of the FS packaging. We > generally don’t package unmaintained code but you are welcome to use > it if you choose and it still works. The db layers have gone through > some evolution since then, what exactly are you trying to accomplish, > it may already be available elsewhere. I have a few use cases: 1. Inbound calls come in to a DDI range I have, and need to be forwarded on to users' mobile phones. This list of DDI --> mobile number mappings is fairly large, and maintained externally to the telephony platform, therefore I want to do a dynamic lookup of the DDI to find out which mobile number to forward a call on to. 2. Users dialling out have different destinations they're allowed to call, and again the details about who is allowed to call where are maintained by people nothing to do with operating the telephony platform, so I want to do a database lookup of each user as they place a call to see whether the destination they're dialling is in their allowed list. 3. Outbound providers charge us for placing calls to different destinations, and the prices change (typically monthly, but sometimes more often), and I need to do accurate live per-call accounting to keep track of users' and groups' spend levels. I've done all the above in Asterisk using its ODBC connectivity, so I'm looking for the best way to execute arbitrary SQL statements in a FreeSwitch dialplan. > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > > > Hi. > > > > Can anyone tell me why > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query > > is not built as part of the standard Debian packages available from > > https://files.freeswitch.org/repo/deb/ ? > > > > Thanks, > > > > > > Antony. -- "640 kilobytes (of RAM) should be enough for anybody." - Bill Gates Please reply to the list; please *don't* CC me. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 krice at freeswitch.org Fri Nov 5 00:51:07 2021 From: krice at freeswitch.org (krice at freeswitch.org) Date: Thu, 4 Nov 2021 19:51:07 -0500 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <0100017cec78aa7d-a9b8fdb3-9cf8-482c-84d0-3140f6b33686-000000@email.amazonses.com> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> <0100017cec78aa7d-a9b8fdb3-9cf8-482c-84d0-3140f6b33686-000000@email.amazonses.com> Message-ID: <604501d7d1df$3822b5b0$a8682110$@freeswitch.org> Mod_xml_curl isn’t a bit heavy for this. This is exactly what it was designed to do. It also makes it where you are returning a much smaller dialplan. The curl request then eliminates a potential pile of regex hits that are quite expensive to parse thru. From: FreeSWITCH-users On Behalf Of Avi Marcus Sent: Thursday, November 4, 2021 2:42 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Developers / packagers here? mod_xml_curl seems a bit heavy for this. You can write a short js or lua script to do the odbc query and assign it to a channel variable. I'm not sure about accessing a _foreign_ odbc source though, or if it only accesses the freeswitch core db. The info isn't loading for me... https://freeswitch.org/confluence/display/FREESWITCH/Lua+FreeSWITCH+Dbh -Avi Marcus BestFone On Thu, Nov 4, 2021 at 9:24 PM John Covici > wrote: I think you should be able to dl all this with mod_xml_curl. This module will give you the xml you need after searching whatever database you need -- and it can be on a per-call basis. One thing though, you might be better using a real programming language for these use cases. Freeswitch is more of a toolkit whereas asterisk is more of a pbx specific arrangement. On Thu, 04 Nov 2021 05:47:06 -0400, Antony Stone wrote: > > On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: > > > It’s a module abandoned 7-8 years ago in the contrib repo that we have > > never maintained and has never been part of the FS packaging. We > > generally don’t package unmaintained code but you are welcome to use it if > > you choose and it still works. The db layers have gone through some > > evolution since then, what exactly are you trying to accomplish, it may > > already be available elsewhere. > > I have a few use cases: > > 1. Inbound calls come in to a DDI range I have, and need to be forwarded on to > users' mobile phones. This list of DDI --> mobile number mappings is fairly > large, and maintained externally to the telephony platform, therefore I want > to do a dynamic lookup of the DDI to find out which mobile number to forward a > call on to. > > 2. Users dialling out have different destinations they're allowed to call, and > again the details about who is allowed to call where are maintained by people > nothing to do with operating the telephony platform, so I want to do a > database lookup of each user as they place a call to see whether the > destination they're dialling is in their allowed list. > > 3. Outbound providers charge us for placing calls to different destinations, > and the prices change (typically monthly, but sometimes more often), and I > need to do accurate live per-call accounting to keep track of users' and > groups' spend levels. > > I've done all the above in Asterisk using its ODBC connectivity, so I'm > looking for the best way to execute arbitrary SQL statements in a FreeSwitch > dialplan. > > > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > > > > > Hi. > > > > > > Can anyone tell me why > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query is > > > not built as part of the standard Debian packages available from > > > https://files.freeswitch.org/repo/deb/ ? > > > > > > Thanks, > > > > > > > > > Antony. > > -- > "640 kilobytes (of RAM) should be enough for anybody." > > - Bill Gates > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 Fri Nov 5 01:47:16 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 5 Nov 2021 01:47:16 +0000 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <604501d7d1df$3822b5b0$a8682110$@freeswitch.org> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> <0100017cec78aa7d-a9b8fdb3-9cf8-482c-84d0-3140f6b33686-000000@email.amazonses.com> <604501d7d1df$3822b5b0$a8682110$@freeswitch.org> Message-ID: This is true. You _can_ do it all via sql queries in the dialplan it self, but you're going to need a lot of regexp'ing, not to mention variable manipulation, etc. it could really get convoluted. It really is simpler just to do xml_curl. A simple php would suffice. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, Nov 5, 2021 at 1:34 AM wrote: > Mod_xml_curl isn’t a bit heavy for this. This is exactly what it was > designed to do. It also makes it where you are returning a much smaller > dialplan. The curl request then eliminates a potential pile of regex hits > that are quite expensive to parse thru. > > > > > > > > *From:* FreeSWITCH-users *On > Behalf Of *Avi Marcus > *Sent:* Thursday, November 4, 2021 2:42 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Developers / packagers here? > > > > mod_xml_curl seems a bit heavy for this. > > > > You can write a short js or lua script to do the odbc query and assign it > to a channel variable. > > > > I'm not sure about accessing a _foreign_ odbc source though, or if it only > accesses the freeswitch core db. > > The info isn't loading for me... > https://freeswitch.org/confluence/display/FREESWITCH/Lua+FreeSWITCH+Dbh > > > > > > > -Avi Marcus > BestFone > > > > On Thu, Nov 4, 2021 at 9:24 PM John Covici wrote: > > I think you should be able to dl all this with mod_xml_curl. This > module will give you the xml you need after searching whatever > database you need -- and it can be on a per-call basis. One thing > though, you might be better using a real programming language for > these use cases. Freeswitch is more of a toolkit whereas asterisk is > more of a pbx specific arrangement. > > On Thu, 04 Nov 2021 05:47:06 -0400, > Antony Stone wrote: > > > > On Wednesday 03 November 2021 at 22:00:14, Mike Jerris wrote: > > > > > It’s a module abandoned 7-8 years ago in the contrib repo that we have > > > never maintained and has never been part of the FS packaging. We > > > generally don’t package unmaintained code but you are welcome to use > it if > > > you choose and it still works. The db layers have gone through some > > > evolution since then, what exactly are you trying to accomplish, it may > > > already be available elsewhere. > > > > I have a few use cases: > > > > 1. Inbound calls come in to a DDI range I have, and need to be forwarded > on to > > users' mobile phones. This list of DDI --> mobile number mappings is > fairly > > large, and maintained externally to the telephony platform, therefore I > want > > to do a dynamic lookup of the DDI to find out which mobile number to > forward a > > call on to. > > > > 2. Users dialling out have different destinations they're allowed to > call, and > > again the details about who is allowed to call where are maintained by > people > > nothing to do with operating the telephony platform, so I want to do a > > database lookup of each user as they place a call to see whether the > > destination they're dialling is in their allowed list. > > > > 3. Outbound providers charge us for placing calls to different > destinations, > > and the prices change (typically monthly, but sometimes more often), and > I > > need to do accurate live per-call accounting to keep track of users' and > > groups' spend levels. > > > > I've done all the above in Asterisk using its ODBC connectivity, so I'm > > looking for the best way to execute arbitrary SQL statements in a > FreeSwitch > > dialplan. > > > > > > On Oct 29, 2021, at 10:21 AM, Antony Stone wrote: > > > > > > > > Hi. > > > > > > > > Can anyone tell me why > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query > is > > > > not built as part of the standard Debian packages available from > > > > https://files.freeswitch.org/repo/deb/ ? > > > > > > > > Thanks, > > > > > > > > > > > > Antony. > > > > -- > > "640 kilobytes (of RAM) should be enough for anybody." > > > > - Bill Gates > > > > Please reply to the > list; > > please *don't* > CC me. > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Antony.Stone at freeswitch.open.source.it Thu Nov 4 20:00:41 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 4 Nov 2021 21:00:41 +0100 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111042100.41872.Antony.Stone@freeswitch.open.source.it> On Thursday 04 November 2021 at 20:24:03, John Covici wrote: > I think you should be able to dl all this with mod_xml_curl. This > module will give you the xml you need after searching whatever > database you need -- and it can be on a per-call basis. One thing > though, you might be better using a real programming language for > these use cases. Freeswitch is more of a toolkit whereas asterisk is > more of a pbx specific arrangement. Thanks. When you say "use a real programming language", do you mean "instead of FreeSwitch / Asterisk" or do you mean instead of XML / curl? Just wondering whether you regard FreeSwitch / Asterisk as a sensible starting point here, and if so, what you would recommend using alongside them as a "real language". Antony. -- A user interface is like a joke. If you have to explain it, it means it doesn't work. Please reply to the list; please *don't* CC me. From covici at ccs.covici.com Fri Nov 5 07:38:09 2021 From: covici at ccs.covici.com (John Covici) Date: Fri, 05 Nov 2021 03:38:09 -0400 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: <202111042100.41872.Antony.Stone@freeswitch.open.source.it> References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> <202111042100.41872.Antony.Stone@freeswitch.open.source.it> Message-ID: I mean instead of using the dialplan, you could use c sharp or even lua for a portion of your workflow. Freeswitch does support doing this kind of thing and then you get more realtime control instead of a static dialplan. I have used mod_managed to write something to do what asterisk's controlplayback does because I could not find anything in fs to do what I wanted to do. On Thu, 04 Nov 2021 16:00:41 -0400, Antony Stone wrote: > > On Thursday 04 November 2021 at 20:24:03, John Covici wrote: > > > I think you should be able to dl all this with mod_xml_curl. This > > module will give you the xml you need after searching whatever > > database you need -- and it can be on a per-call basis. One thing > > though, you might be better using a real programming language for > > these use cases. Freeswitch is more of a toolkit whereas asterisk is > > more of a pbx specific arrangement. > > Thanks. > > When you say "use a real programming language", do you mean "instead of > FreeSwitch / Asterisk" or do you mean instead of XML / curl? > > Just wondering whether you regard FreeSwitch / Asterisk as a sensible starting > point here, and if so, what you would recommend using alongside them as a > "real language". > > > Antony. > > -- > A user interface is like a joke. > If you have to explain it, it means it doesn't work. > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From martin at pattersong.co.uk Fri Nov 5 09:29:00 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Fri, 5 Nov 2021 09:29:00 +0000 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: <202111041512.07310.Antony.Stone@freeswitch.open.source.it> References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> <202111031649.01531.Antony.Stone@freeswitch.open.source.it> <202111041512.07310.Antony.Stone@freeswitch.open.source.it> Message-ID: It turns out from looking at the code that both the command uuid_hold and the dptools:hold application actually call the same underlying function (switch_ivr_hold_uuid) to achieve the holding. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Thu, 4 Nov 2021 at 14:12, Antony Stone wrote: > > On Thursday 04 November 2021 at 14:59:53, kaiduan xie via FreeSWITCH-users > wrote: > > > Antony, > > For hold FS does not send reINVITE with SDP containing a=sendonly to > > upstream PBX. You need to change the source code to do that. > > Er, yes it does. I now have a working solution. > > Thanks to all who helped me in the right direction. > > > On Wednesday, November 3, 2021, 10:11:37 p.m. EDT, Antony Stone wrote: > > > > On Wednesday 03 November 2021 at 15:46:54, Brian West wrote: > > > You just park it and play music, there is NO HOLD/UNHOLD in the manner > > > you're needing. > > > > No, this is not for placing the call on hold inside FreeSwitch - this is > > for sending a re-Invite out to an upstream PBX to put the call on hold > > there (a re-Invite with SDP containing a=sendonly). > > > > > > On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > > > > Hi. > > > > > > > > > > I want to put a channel on hold in my dialplan. > > > > > > > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools > > > > > tells me that there is a command "hold" to do this, but the link for > > > > > the documentation is > > > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold > > > > > &action=edit&redlink=1 > > > > > > > > > > which does not take me anywhere useful. > > > > > > > > > > Can someone point me at the correct page for this command? > > > > > > > > > > > > > > > Thanks, > > > > > > > > > > > > > > > Antony. > > -- > "Remember: the S in IoT stands for Security." > > - Jan-Piet Mens > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 david.villasmil.work at gmail.com Fri Nov 5 11:28:55 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 5 Nov 2021 11:28:55 +0000 Subject: [Freeswitch-users] Developers / packagers here? In-Reply-To: References: <202110291721.46635.Antony.Stone@freeswitch.open.source.it> <202111041047.06813.Antony.Stone@freeswitch.open.source.it> <202111042100.41872.Antony.Stone@freeswitch.open.source.it> Message-ID: In FS there’s the embedded lua. You can do almost anything with it. On Fri, 5 Nov 2021 at 08:18, John Covici wrote: > I mean instead of using the dialplan, you could use c sharp or even > lua for a portion of your workflow. Freeswitch does support doing > this kind of thing and then you get more realtime control instead of a > static dialplan. I have used mod_managed to write something to do > what asterisk's controlplayback does because I could not find anything > in fs to do what I wanted to do. > > On Thu, 04 Nov 2021 16:00:41 -0400, > Antony Stone wrote: > > > > On Thursday 04 November 2021 at 20:24:03, John Covici wrote: > > > > > I think you should be able to dl all this with mod_xml_curl. This > > > module will give you the xml you need after searching whatever > > > database you need -- and it can be on a per-call basis. One thing > > > though, you might be better using a real programming language for > > > these use cases. Freeswitch is more of a toolkit whereas asterisk is > > > more of a pbx specific arrangement. > > > > Thanks. > > > > When you say "use a real programming language", do you mean "instead of > > FreeSwitch / Asterisk" or do you mean instead of XML / curl? > > > > Just wondering whether you regard FreeSwitch / Asterisk as a sensible > starting > > point here, and if so, what you would recommend using alongside them as > a > > "real language". > > > > > > Antony. > > > > -- > > A user interface is like a joke. > > If you have to explain it, it means it doesn't work. > > > > Please reply to the > list; > > please *don't* > CC me. > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Fri Nov 5 13:21:28 2021 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Fri, 5 Nov 2021 13:21:28 +0000 (UTC) Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> <202111031649.01531.Antony.Stone@freeswitch.open.source.it> <202111041512.07310.Antony.Stone@freeswitch.open.source.it> Message-ID: <669801552.578858.1636118488984@mail.yahoo.com> Antony, What is your Freeswitch version? uuid_hold in FS 1.8.5 does not send out reINVITE with sendonly. /Kaiduan On Friday, November 5, 2021, 05:30:06 a.m. EDT, Martin Paterson wrote: It turns out from looking at the code that both the command uuid_hold and the dptools:hold application actually call the same underlying function (switch_ivr_hold_uuid) to achieve the holding. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Thu, 4 Nov 2021 at 14:12, Antony Stone wrote: > > On Thursday 04 November 2021 at 14:59:53, kaiduan xie via FreeSWITCH-users > wrote: > > >  Antony, > > For hold FS does not send reINVITE with SDP containing a=sendonly to > > upstream PBX. You need to change the source code to do that. > > Er, yes it does.  I now have a working solution. > > Thanks to all who helped me in the right direction. > > > On Wednesday, November 3, 2021, 10:11:37 p.m. EDT, Antony Stone wrote: > > > >  On Wednesday 03 November 2021 at 15:46:54, Brian West wrote: > > > You just park it and play music, there is NO HOLD/UNHOLD in the manner > > > you're needing. > > > > No, this is not for placing the call on hold inside FreeSwitch - this is > > for sending a re-Invite out to an upstream PBX to put the call on hold > > there (a re-Invite with SDP containing a=sendonly). > > > > > > On Tue, 2 Nov 2021 at 15:21, Antony Stone wrote: > > > > > Hi. > > > > > > > > > > I want to put a channel on hold in my dialplan. > > > > > > > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools > > > > > tells me that there is a command "hold" to do this, but the link for > > > > > the documentation is > > > > > https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold > > > > > &action=edit&redlink=1 > > > > > > > > > > which does not take me anywhere useful. > > > > > > > > > > Can someone point me at the correct page for this command? > > > > > > > > > > > > > > > Thanks, > > > > > > > > > > > > > > > Antony. > > -- > "Remember: the S in IoT stands for Security." > >  - Jan-Piet Mens > >                                                    Please reply to the list; >                                                          please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 support at naud.io Fri Nov 5 14:09:45 2021 From: support at naud.io (Support from NetworkedAudio LLC) Date: Fri, 5 Nov 2021 14:09:45 +0000 Subject: [Freeswitch-users] Examples of mod_portaudio_stream? Message-ID: Hi, There seems to be a mod_portaudio_stream that could be used as MOH for a conference with no other members. Does anyone have any examples of how to use it? Erik -------------- next part -------------- An HTML attachment was scrubbed... URL: From dgreenwald at gmail.com Fri Nov 5 19:08:20 2021 From: dgreenwald at gmail.com (Daniel Greenwald) Date: Fri, 5 Nov 2021 15:08:20 -0400 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: If homer is overkill. You could turn on sip trace which will log sip to fs log. Or configure sip capture to send HEP directly into sngrep. On Wed, Nov 3, 2021 at 7:36 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Homer? > > On Wed, 3 Nov 2021 at 03:18, Gregor Maier via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: Gregor Maier >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Tue, 2 Nov 2021 21:03:22 +0100 >> Subject: Question regarding continuous SIP trace with Freeswitch >> >> Hello! >> >> I'm currently struggling how to continuously trace all SIP packages from >> and to Freeswitch. Tcpdump or sngrep solely or similar tools aren't a >> solution, because SIP is >> encrypted (TLS). Asterisk provides an internal interface which logs all >> (or only restricted hosts e.g.) SIP packages to pcap (after decryption or >> before encryption of course). >> >> I found >> https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture - >> but there is no solution which handles the case "registering as client to a >> providers SIP >> server" (there is no local private key involved - the provider is the >> owner of the private key). >> >> Voipmonitor is pretty oversized for the planned use case (home server >> running on an APU 4 device (https://pcengines.ch/apu4d4.htm) on a sd >> card). >> >> Is there a similar and practical way how Freeswitch can log SIP packages >> to pcap? >> >> >> Thanks >> Gregor >> >> >> >> >> ---------- Forwarded message ---------- >> From: Gregor Maier via FreeSWITCH-users < >> freeswitch-users at lists.freeswitch.org> >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Tue, 02 Nov 2021 20:18:40 -0700 (PDT) >> Subject: [Freeswitch-users] Question regarding continuous SIP trace with >> Freeswitch >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 rahman.duran at erzurum.edu.tr Fri Nov 5 05:29:48 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Fri, 5 Nov 2021 08:29:48 +0300 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: Hi, Bump. Any hints on this? Regards, Rahman Rahman Duran , 26 Eki 2021 Sal, 09:14 tarihinde şunu yazdı: > Hi, > > I am using announcements for fail hungup causes like busy, no_answer etc. > Here is my dial plan > > > > > expression="^true$|^TRUE$|^True$"/> > > data="call_pickup_group=${user_data(${destination_number}@${domain_name} > var call_pickup_group)}"/> > data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> > data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> > data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> > > > data="nolocal:absolute_codec_string=${ep_codec_string}"/> > data="callee_id_name=${user_data(${destination_number}@${domain_name} var > effective_caller_id_name)}"/> > > > > > data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} > var effective_caller_id_name)}}user/${destination_number}@ > ${domain_name}"/> > data="hata-${originate_disposition} XML hata_anonslari"/> > > > > And here is announcements context that I handle hangup causes: > > > > > > > > > > data="$${anons_dosya_yolu}/user_busy.wav"/> > > > > > > > > > data="$${anons_dosya_yolu}/no_answer.wav"/> > > > > > > expression="^hata-USER_NOT_REGISTERED$"> > data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> > > data="$${anons_dosya_yolu}/user_not_registered.wav"/> > > > > > > > data="cdr_hata_anonsu=${originate_disposition}"/> > > > > > > > > > The problem is if the calling leg does not listen the announcement to the > end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I > already know the real hangup cause, how can I override the CDR hangup cause > with the real one? For now I am setting another variable (cdr_hata_anonsu) > and added it to CDR logs, but if possible I want to fix this with a real > solution. > > Regards, > > Rahman Duran > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jennifer.hashimoto at caztel.com Fri Nov 5 17:17:54 2021 From: jennifer.hashimoto at caztel.com (Jennifer Hashimoto) Date: Fri, 5 Nov 2021 13:17:54 -0400 Subject: [Freeswitch-users] reinvite too quickly causing 491 Message-ID: <8007B73A-611C-4319-9727-7C30F6650D57@caztel.com> Hi, I’m wondering if someone can help me figure out this problem One of our carriers is sometimes sending a re-invite within milliseconds of sending the OK on answer. This causes Freeswitch to send a 491 to the outbound leg without informing inbound leg that anything has happened. This causes the call on the outbound leg to die. After getting the 491 the carrier does nothing - does not retry after 1-2 seconds or anything. The inbound leg thinks it is answered, but there is no audio. the inbound caller hangs up at some point and the carrier says leg does not exist when we send them the bye. Any help would be appreciated! FreeSWITCH version: 1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit) --------------------------------------------------- Jennifer Akemi Hashimoto Caztel Communications jennifer.hashimoto at caztel.com 418-655-5546 -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch13 at mailbox.org Fri Nov 5 23:48:16 2021 From: freeswitch13 at mailbox.org (Gregor Maier) Date: Sat, 6 Nov 2021 00:48:16 +0100 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: HEP directly to sngrep is a good idea. Unfortunately writing pcap files doesn't work (known problem https://github.com/irontec/sngrep/issues/270). Maybe there is another tool which is able to read HEP data stream and write it to a "real" pcap file? Thanks Gregor On 05.11.21 at 20:08 Daniel Greenwald wrote: > If homer is overkill. You could turn on sip trace which will log sip to fs > log. Or configure sip capture to send HEP directly into sngrep. > > On Wed, Nov 3, 2021 at 7:36 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Homer? >> >> On Wed, 3 Nov 2021 at 03:18, Gregor Maier via FreeSWITCH-users < >> freeswitch-users at lists.freeswitch.org> wrote: >> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gregor Maier >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Tue, 2 Nov 2021 21:03:22 +0100 >>> Subject: Question regarding continuous SIP trace with Freeswitch >>> >>> Hello! >>> >>> I'm currently struggling how to continuously trace all SIP packages from >>> and to Freeswitch. Tcpdump or sngrep solely or similar tools aren't a >>> solution, because SIP is >>> encrypted (TLS). Asterisk provides an internal interface which logs all >>> (or only restricted hosts e.g.) SIP packages to pcap (after decryption or >>> before encryption of course). >>> >>> I found >>> https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture - >>> but there is no solution which handles the case "registering as client to a >>> providers SIP >>> server" (there is no local private key involved - the provider is the >>> owner of the private key). >>> >>> Voipmonitor is pretty oversized for the planned use case (home server >>> running on an APU 4 device (https://pcengines.ch/apu4d4.htm) on a sd >>> card). >>> >>> Is there a similar and practical way how Freeswitch can log SIP packages >>> to pcap? >>> >>> >>> Thanks >>> Gregor >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gregor Maier via FreeSWITCH-users < >>> freeswitch-users at lists.freeswitch.org> >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Tue, 02 Nov 2021 20:18:40 -0700 (PDT) >>> Subject: [Freeswitch-users] Question regarding continuous SIP trace with >>> Freeswitch >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 Antony.Stone at freeswitch.open.source.it Fri Nov 5 13:41:53 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Fri, 5 Nov 2021 14:41:53 +0100 Subject: [Freeswitch-users] Documentation for hold? In-Reply-To: References: <202111021621.39622.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111051441.53888.Antony.Stone@freeswitch.open.source.it> On Friday 05 November 2021 at 14:39:27, kaiduan xie via FreeSWITCH-users wrote: > Antony, > What is your Freeswitch version? 1.10.7 - current Debian package from https://files.freeswitch.org/repo/deb/ > uuid_hold in FS 1.8.5 does not send out reINVITE with sendonly. > > /Kaiduan > On Friday, November 5, 2021, 05:30:06 a.m. EDT, Martin Paterson wrote: > > It turns out from looking at the code that both the command uuid_hold > and the dptools:hold application actually call the same underlying > function (switch_ivr_hold_uuid) to achieve the holding. > > Martin. > On Thu, 4 Nov 2021 at 14:12, Antony Stone wrote: > > On Thursday 04 November 2021 at 14:59:53, kaiduan xie wrote: > > > Antony, > > > > > > For hold FS does not send reINVITE with SDP containing a=sendonly to > > > upstream PBX. You need to change the source code to do that. > > > > Er, yes it does. I now have a working solution. > > > > Thanks to all who helped me in the right direction. Antony. -- https://tools.ietf.org/html/rfc6890 - providing 16 million IPv4 addresses for talking to yourself. Please reply to the list; please *don't* CC me. From david.villasmil.work at gmail.com Sat Nov 6 02:02:27 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 6 Nov 2021 02:02:27 +0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: The fact is A is hanging up the call. I don’t think you can actually change this without changing FS source code to override it. On Sat, 6 Nov 2021 at 01:20, Rahman Duran wrote: > Hi, > > Bump. Any hints on this? > > Regards, > > Rahman > > Rahman Duran , 26 Eki 2021 Sal, 09:14 > tarihinde şunu yazdı: > >> Hi, >> >> I am using announcements for fail hungup causes like busy, no_answer etc. >> Here is my dial plan >> >> >> >> >> > expression="^true$|^TRUE$|^True$"/> >> >> > data="call_pickup_group=${user_data(${destination_number}@${domain_name} >> var call_pickup_group)}"/> >> > data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >> > data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >> > data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >> >> >> > data="nolocal:absolute_codec_string=${ep_codec_string}"/> >> > data="callee_id_name=${user_data(${destination_number}@${domain_name} >> var effective_caller_id_name)}"/> >> >> >> >> >> > data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >> var effective_caller_id_name)}}user/${destination_number}@ >> ${domain_name}"/> >> > data="hata-${originate_disposition} XML hata_anonslari"/> >> >> >> >> And here is announcements context that I handle hangup causes: >> >> >> >> >> >> >> >> >> >> > data="$${anons_dosya_yolu}/user_busy.wav"/> >> >> >> >> >> >> >> >> >> > data="$${anons_dosya_yolu}/no_answer.wav"/> >> >> >> >> >> >> > expression="^hata-USER_NOT_REGISTERED$"> >> > data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >> >> > data="$${anons_dosya_yolu}/user_not_registered.wav"/> >> >> >> >> >> >> >> > data="cdr_hata_anonsu=${originate_disposition}"/> >> >> >> >> >> >> >> >> >> The problem is if the calling leg does not listen the announcement to the >> end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I >> already know the real hangup cause, how can I override the CDR hangup cause >> with the real one? For now I am setting another variable (cdr_hata_anonsu) >> and added it to CDR logs, but if possible I want to fix this with a real >> solution. >> >> Regards, >> >> Rahman Duran >> > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Nov 6 02:03:16 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 6 Nov 2021 02:03:16 +0000 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: At the risk of repressing myself: Homer? :) On Sat, 6 Nov 2021 at 01:41, Gregor Maier via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Gregor Maier > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Sat, 6 Nov 2021 00:48:16 +0100 > Subject: Re: [Freeswitch-users] Question regarding continuous SIP trace > with Freeswitch > HEP directly to sngrep is a good idea. Unfortunately writing pcap files > doesn't work (known problem https://github.com/irontec/sngrep/issues/270). > > Maybe there is another tool which is able to read HEP data stream and > write it to a "real" pcap file? > > > Thanks > Gregor > > > On 05.11.21 at 20:08 Daniel Greenwald wrote: > > If homer is overkill. You could turn on sip trace which will log sip to > fs > > log. Or configure sip capture to send HEP directly into sngrep. > > > > On Wed, Nov 3, 2021 at 7:36 AM David Villasmil < > > david.villasmil.work at gmail.com> wrote: > > > >> Homer? > >> > >> On Wed, 3 Nov 2021 at 03:18, Gregor Maier via FreeSWITCH-users < > >> freeswitch-users at lists.freeswitch.org> wrote: > >> > >>> > >>> > >>> > >>> ---------- Forwarded message ---------- > >>> From: Gregor Maier > >>> To: FreeSWITCH Users Help > >>> Cc: > >>> Bcc: > >>> Date: Tue, 2 Nov 2021 21:03:22 +0100 > >>> Subject: Question regarding continuous SIP trace with Freeswitch > >>> > >>> Hello! > >>> > >>> I'm currently struggling how to continuously trace all SIP packages > from > >>> and to Freeswitch. Tcpdump or sngrep solely or similar tools aren't a > >>> solution, because SIP is > >>> encrypted (TLS). Asterisk provides an internal interface which logs all > >>> (or only restricted hosts e.g.) SIP packages to pcap (after decryption > or > >>> before encryption of course). > >>> > >>> I found > >>> https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture - > >>> but there is no solution which handles the case "registering as client > to a > >>> providers SIP > >>> server" (there is no local private key involved - the provider is the > >>> owner of the private key). > >>> > >>> Voipmonitor is pretty oversized for the planned use case (home server > >>> running on an APU 4 device (https://pcengines.ch/apu4d4.htm) on a sd > >>> card). > >>> > >>> Is there a similar and practical way how Freeswitch can log SIP > packages > >>> to pcap? > >>> > >>> > >>> Thanks > >>> Gregor > >>> > >>> > >>> > >>> > >>> ---------- Forwarded message ---------- > >>> From: Gregor Maier via FreeSWITCH-users < > >>> freeswitch-users at lists.freeswitch.org> > >>> To: FreeSWITCH Users Help > >>> Cc: > >>> Bcc: > >>> Date: Tue, 02 Nov 2021 20:18:40 -0700 (PDT) > >>> Subject: [Freeswitch-users] Question regarding continuous SIP trace > with > >>> Freeswitch > >>> > _________________________________________________________________________ > >>> > >>> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > >>> services. > >>> Build your next product on our scalable cloud platform. > >>> > >>> Join our online community to chat in real time > >>> https://signalwire.community > >>> > >>> 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 > >> > >> -- > >> Regards, > >> > >> David Villasmil > >> email: david.villasmil.work at gmail.com > >> phone: +34669448337 > >> > _________________________________________________________________________ > >> > >> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > >> services. > >> Build your next product on our scalable cloud platform. > >> > >> Join our online community to chat in real time > >> https://signalwire.community > >> > >> 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 > > > > > > > ---------- Forwarded message ---------- > From: Gregor Maier via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Fri, 05 Nov 2021 18:41:08 -0700 (PDT) > Subject: Re: [Freeswitch-users] Question regarding continuous SIP trace > with Freeswitch > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From martin at pattersong.co.uk Mon Nov 8 14:04:16 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Mon, 8 Nov 2021 14:04:16 +0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: David is right, the hangup cause for the a-leg is ORIGINATOR_CANCEL because that's what actually happened to the a-leg. However the CDR for the b-leg will show the hangup cause that you expect. I think by default b-leg CDRs aren't generated, but you can turn them on in config. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Sat, 6 Nov 2021 at 02:02, David Villasmil wrote: > > The fact is A is hanging up the call. I don’t think you can actually change this without changing FS source code to override it. > > > On Sat, 6 Nov 2021 at 01:20, Rahman Duran wrote: >> >> Hi, >> >> Bump. Any hints on this? >> >> Regards, >> >> Rahman >> >> Rahman Duran , 26 Eki 2021 Sal, 09:14 tarihinde şunu yazdı: >>> >>> Hi, >>> >>> I am using announcements for fail hungup causes like busy, no_answer etc. Here is my dial plan >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> And here is announcements context that I handle hangup causes: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> The problem is if the calling leg does not listen the announcement to the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I already know the real hangup cause, how can I override the CDR hangup cause with the real one? For now I am setting another variable (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix this with a real solution. >>> >>> Regards, >>> >>> Rahman Duran >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 gregor at infomedia.si Mon Nov 8 14:12:09 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 8 Nov 2021 15:12:09 +0100 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: If you are using xml_cdr and creating your own CDRs, you can set variable and read it later in your code and override it. On Mon, 8 Nov 2021 at 15:06, Martin Paterson wrote: > David is right, the hangup cause for the a-leg is ORIGINATOR_CANCEL > because that's what actually happened to the a-leg. However the CDR > for the b-leg will show the hangup cause that you expect. I think by > default b-leg CDRs aren't generated, but you can turn them on in > config. > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > On Sat, 6 Nov 2021 at 02:02, David Villasmil > wrote: > > > > The fact is A is hanging up the call. I don’t think you can actually > change this without changing FS source code to override it. > > > > > > On Sat, 6 Nov 2021 at 01:20, Rahman Duran > wrote: > >> > >> Hi, > >> > >> Bump. Any hints on this? > >> > >> Regards, > >> > >> Rahman > >> > >> Rahman Duran , 26 Eki 2021 Sal, 09:14 > tarihinde şunu yazdı: > >>> > >>> Hi, > >>> > >>> I am using announcements for fail hungup causes like busy, no_answer > etc. Here is my dial plan > >>> > >>> > >>> > >>> > >>> expression="^true$|^TRUE$|^True$"/> > >>> > >>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} > var call_pickup_group)}"/> > >>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> > >>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> > >>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> > >>> > >>> > >>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> > >>> data="callee_id_name=${user_data(${destination_number}@${domain_name} var > effective_caller_id_name)}"/> > >>> > >>> > >>> > >>> > >>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} > var effective_caller_id_name)}}user/${destination_number}@ > ${domain_name}"/> > >>> data="hata-${originate_disposition} XML hata_anonslari"/> > >>> > >>> > >>> > >>> And here is announcements context that I handle hangup causes: > >>> > >>> > >>> > >>> > >>> > >>> > >>> expression="^hata-USER_BUSY$"> > >>> > >>> > >>> data="$${anons_dosya_yolu}/user_busy.wav"/> > >>> > >>> > >>> > >>> > >>> > >>> expression="^hata-NO_ANSWER$"> > >>> > >>> > >>> data="$${anons_dosya_yolu}/no_answer.wav"/> > >>> > >>> > >>> > >>> > >>> > >>> expression="^hata-USER_NOT_REGISTERED$"> > >>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> > >>> > >>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> > >>> > >>> > >>> > >>> > >>> > >>> > >>> data="cdr_hata_anonsu=${originate_disposition}"/> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> The problem is if the calling leg does not listen the announcement to > the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As > I already know the real hangup cause, how can I override the CDR hangup > cause with the real one? For now I am setting another variable > (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix > this with a real solution. > >>> > >>> Regards, > >>> > >>> Rahman Duran > >> > >> > _________________________________________________________________________ > >> > >> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > >> Build your next product on our scalable cloud platform. > >> > >> Join our online community to chat in real time > https://signalwire.community > >> > >> 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 > > > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 rahman.duran at erzurum.edu.tr Mon Nov 8 05:24:04 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Mon, 8 Nov 2021 08:24:04 +0300 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: Hi David, You are right but I don't want or need to change any freeswitch internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR template has any dynamic mechanism to use on variables. For example can I say "if variable A is not empty use A, else use variable B" in the CDR template? Regards, Rahman David Villasmil , 6 Kas 2021 Cmt, 05:07 tarihinde şunu yazdı: > The fact is A is hanging up the call. I don’t think you can actually > change this without changing FS source code to override it. > > > On Sat, 6 Nov 2021 at 01:20, Rahman Duran > wrote: > >> Hi, >> >> Bump. Any hints on this? >> >> Regards, >> >> Rahman >> >> Rahman Duran , 26 Eki 2021 Sal, 09:14 >> tarihinde şunu yazdı: >> >>> Hi, >>> >>> I am using announcements for fail hungup causes like busy, no_answer >>> etc. Here is my dial plan >>> >>> >>> >>> >>> >> expression="^true$|^TRUE$|^True$"/> >>> >>> >> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>> var call_pickup_group)}"/> >>> >> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>> >> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>> >> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>> >>> >>> >> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>> >> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>> var effective_caller_id_name)}"/> >>> >>> >>> >>> >>> >> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>> var effective_caller_id_name)}}user/${destination_number}@ >>> ${domain_name}"/> >>> >> data="hata-${originate_disposition} XML hata_anonslari"/> >>> >>> >>> >>> And here is announcements context that I handle hangup causes: >>> >>> >>> >>> >>> >>> >>> >> expression="^hata-USER_BUSY$"> >>> >>> >>> >> data="$${anons_dosya_yolu}/user_busy.wav"/> >>> >>> >>> >>> >>> >>> >> expression="^hata-NO_ANSWER$"> >>> >>> >>> >> data="$${anons_dosya_yolu}/no_answer.wav"/> >>> >>> >>> >>> >>> >>> >> expression="^hata-USER_NOT_REGISTERED$"> >>> >> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>> >>> >> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>> >>> >>> >>> >>> >>> >>> >> data="cdr_hata_anonsu=${originate_disposition}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> The problem is if the calling leg does not listen the announcement to >>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>> I already know the real hangup cause, how can I override the CDR hangup >>> cause with the real one? For now I am setting another variable >>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>> this with a real solution. >>> >>> Regards, >>> >>> Rahman Duran >>> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 freeswitch13 at mailbox.org Sat Nov 6 09:16:41 2021 From: freeswitch13 at mailbox.org (Gregor Maier) Date: Sat, 6 Nov 2021 10:16:41 +0100 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: <63e9ca13-f74e-fe80-0cfa-1d242c1e4c4c@mailbox.org> On 06.11.21 at 03:03 David Villasmil wrote: > At the risk of repressing myself: Homer? > :) Yes, David, I already heard it - but I don't know the way from Homer to pcap - I want to analyze SIP packages with sngrep! Because it's extremely handy and powerful! Especially if it comes to compare different messages and measure time between arbitrary packages. And Homer is bloatware *compared to my requirement*. sngrep would be the desired and best solution *for me* if it would be able to write correct pcap files while acting as HEP server. Thanks Gregor From freeswitch13 at mailbox.org Sat Nov 6 15:28:03 2021 From: freeswitch13 at mailbox.org (Gregor Maier) Date: Sat, 6 Nov 2021 16:28:03 +0100 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: <0b5c6910-fa33-a2e6-0b15-d10932be3445@mailbox.org> On 06.11.21 at 03:03 David Villasmil wrote: > At the risk of repressing myself: Homer? > :) Ok, as there seems to be no other solution, I tried it. Thank Good, it isn't too big. I feared it would be more. I saw it's possible to export pcap traces on the UI for the selected calls. Therefore it's anyway possible to use sngrep to better analyze calls. Getting rtp quality info is a nice side effect. Thanks for being persistent! Gregor From avi at avimarcus.net Mon Nov 8 20:18:42 2021 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 8 Nov 2021 20:18:42 +0000 Subject: [Freeswitch-users] Bridge to other FS server has no audio until DTMF In-Reply-To: <0100017c5ba82bd9-711151e4-7603-4eea-a6d9-a9029c06383c-000000@email.amazonses.com> References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com> <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com> <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com> <0100017c5ba82bd9-711151e4-7603-4eea-a6d9-a9029c06383c-000000@email.amazonses.com> Message-ID: <0100017d0133f552-473a0842-c954-40c8-972a-f7c562fd2ed7-000000@email.amazonses.com> Is there a way to "fix" the standoff? Perhaps a header to send or a channel variable to set? I'd like to do bypass_media to cut one of my servers out of the media path, but then I won't be able to queue an rfc2833 digit press. Thanks, -Avi Marcus BestFone On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus wrote: > I had to do this to get it to execute on the B leg: > > > ... but it didn't help. Only DTMF worked... either manually dialed or via > queue_dtmf > Freeswitch A waited for my DTMF to actually send the silence. > Version 1.10.6 -release-18-1ff9d0a60e 64bit > > > 2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort > noise payload to 13 > 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/ > JOIN_CLASS_7229999 at voip.bestfone.com] has been answered > EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com > playback(silence_stream://100) > 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec > Activated L16 at 8000hz 1 channels 20ms > > -- 20 seconds later when I pressed a button -- > > 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done > playing file silence_stream://100 > 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/ > JOIN_CLASS_7229999 at voip.bestfone.com) Callstate Change DOWN -> ACTIVE > 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 > (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State Change > CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 > (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) Running State > Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) > 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 > (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State EXCHANGE_MEDIA > 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA > 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet > for [5] ts=960 dur=160/160/2000 seq=26795 lw=960 > > > > This seemingly shouldn't be an issue. FS1 already has active media from > the A leg, so it should initiate to the B leg. The B leg has been > instructed to play a file, so it should initiate to the A leg... > But if this is somehow unavoidable, perhaps we need a workaround config, > where we have a simple variable in the bridge string to avoid the standoff? > > -Avi Marcus > > > > On Thu, Oct 7, 2021 at 6:01 PM Brian West wrote: > >> execure_on_answer=playback::silence_stream://100 should solve it. >> >> /b >> PS, the non pc term that this has been said to be is >> https://en.wikipedia.org/wiki/Mexican_standoff >> >> On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus wrote: >> >>> I meant there's audio from pstn to fs1, but indeed I'm observing no >>> audio between fs1 and fs2. >>> >>> What api should I call with api on answer..? >>> >>> On Thu, Oct 7, 2021, 3:19 PM David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> If you see rtp glowing both ways, then this is not the stalemate I was >>>> talking about. The scenario I’m referring to is about FS not starting >>>> sending rtp waiting for the other side to start sending, and the other side >>>> doing the same thing, thus going into a stalemate. This is solved by >>>> injecting a silence (I would do api_on_answer). >>>> >>>> What you’re describing seems different to me. >>>> >>>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus wrote: >>>> >>>>> I'm using dialplan bridge, so then the dialplan is over. How do I send >>>>> silence after the bridge...? An api_on_answer with a uuid_broadcast.. >>>>> seems overly complicated. >>>>> >>>>> >>>>> >>>>> >>>>> (And I don't know why there isn't audio - I had to set up an audio to >>>>> get to this options in the IVR... so there's already audio. And Server B >>>>> also started a file playback so should have initiated audio.) >>>>> >>>>> >>>>> -Avi Marcus >>>>> >>>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> I seem to remember Brian saying this was because FS is waiting for >>>>>> the remote end to send audio before starting itself. I believe he >>>>>> recommended sending an empty (silence) to force the audio stream to be sent >>>>>> even if fs hasn’t received anything. >>>>>> >>>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus wrote: >>>>>> >>>>>>> I started a new thread in case anyone muted it... it wasn't simply a >>>>>>> network issue. >>>>>>> >>>>>>> It seems the bridging occurs and dialplan processes, but no media >>>>>>> flows - until DTMF from the A-leg. >>>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> >>>>>>> freeswitch B. >>>>>>> >>>>>>> Calls directly from carrier to Freeswitch B are fine. >>>>>>> Calls from a different carrier to Freeswitch A -> media and IVR -> >>>>>>> Freeswitch B are also fine. >>>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in >>>>>>> the media path, it's an FS issue... >>>>>>> >>>>>>> >>>>>>> I actually mcguyvered this right now with a queue_dtmf before the >>>>>>> bridge, to force the audio stream to update. >>>>>>> >>>>>>> Here's the log on freeswitch B: >>>>>>> >>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>> log(DEBUG class chosen: 1234567) >>>>>>> 2021-10-07 09:16:24.343175 [DEBUG >>>>>>> ] mod_dptools.c:1879 class chosen: 1234567 >>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>> javascript(conference/lookupAndJoinConference.js 1234567) >>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>> playback(class/hold-wait-teacher.wav) >>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123 >>>>>>> entering state [completed][200] >>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123 >>>>>>> entering state [ready][200] >>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz >>>>>>> 1 channels 20ms >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2021-10-07 09:16:34.903283 [DEBUG >>>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed. >>>>>>> 2021-10-07 09:16:34.923190 [DEBUG >>>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 >>>>>>> 2021-10-07 09:16:34.923190 [INFO >>>>>>> ] switch_channel.c:522 RECV DTMF 3:2080 >>>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 >>>>>>> 2021-10-07 09:16:37.143169 [DEBUG >>>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav >>>>>>> >>>>>>> >>>>>>> You can see a 10 second gap between call ready 200 and correct >>>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't >>>>>>> happen automatically, only when I choose to press something. >>>>>>> >>>>>>> >>>>>>> Any ideas as to the root cause of this? >>>>>>> >>>>>>> >>>>>>> -Avi Marcus >>>>>>> >>>>>>> ---------- Forwarded message --------- >>>>>>> From: Avi Marcus >>>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM >>>>>>> Subject: Bridge to other FS server has no audio ??? >>>>>>> To: FreeSWITCH Users Help >>>>>>> >>>>>>> >>>>>>> Any ideas on why a call doesn't have media? It used to work, but I >>>>>>> think my upstream changed his SDP again. >>>>>>> >>>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS >>>>>>> server B. Media works. >>>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B >>>>>>> (not on bypass). Media works. >>>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges >>>>>>> to FS server B. Call looks OK, but no media is flowing (I don't hear >>>>>>> anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All >>>>>>> the same codecs are set in the json cdrs (PCMU). >>>>>>> >>>>>>> FS server B is to join a conference if that matters. >>>>>>> >>>>>>> I was assuming it had to do with codecs, but setting >>>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs - >>>>>>> it's already always PCMU. >>>>>>> >>>>>>> I have NO clue what further could cause this other than codecs, >>>>>>> which seem to be fine. Any ideas please? >>>>>>> >>>>>>> >>>>>>> -Avi Marcus >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> >>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>> https://signalwire.com >>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>>> services. >>>>>>> Build your next product on our scalable cloud platform. >>>>>>> >>>>>>> Join our online community to chat in real time >>>>>>> https://signalwire.community >>>>>>> >>>>>>> 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 >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> >>>>>> David Villasmil >>>>>> email: david.villasmil.work at gmail.com >>>>>> phone: +34669448337 >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>> https://signalwire.com >>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>> services. >>>>>> Build your next product on our scalable cloud platform. >>>>>> >>>>>> Join our online community to chat in real time >>>>>> https://signalwire.community >>>>>> >>>>>> 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> 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 >>>> >>>> -- >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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 >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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] >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 support at naud.io Mon Nov 8 21:01:37 2021 From: support at naud.io (Support from NetworkedAudio LLC) Date: Mon, 8 Nov 2021 21:01:37 +0000 Subject: [Freeswitch-users] Examples of mod_portaudio_stream? In-Reply-To: References: Message-ID: So I added a Visual C++ project for it and added pablio.h and similar. The dll builds and loads, and for now I’m just testing it with: It’s crashing my test 1.10.7 with EXCEPTION_ACCESS_VIOLATION_READ: Fatal Error: EXCEPTION_ACCESS_VIOLATION_READ I just want to use one of my PortAudio channels as Music on Hold (moh) – using a soundcard for audio. Anyone got any pointers? From: FreeSWITCH-users on behalf of Support from NetworkedAudio LLC Date: Friday, November 5, 2021 at 9:36 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Examples of mod_portaudio_stream? Hi, There seems to be a mod_portaudio_stream that could be used as MOH for a conference with no other members. Does anyone have any examples of how to use it? Erik -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Nov 8 23:19:19 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 8 Nov 2021 23:19:19 +0000 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: Then why not just run tcpdump filtering for the signaling port and rotate every N bytes or Mb or whatever? And then simply pushing those files somewhere you can later use them? Maybe I misunderstood something If what you want is to go at some on-demand time to trace, then just do it like that. On Mon, 8 Nov 2021 at 18:46, Gregor Maier via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Gregor Maier > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Sat, 6 Nov 2021 10:16:41 +0100 > Subject: Re: [Freeswitch-users] Question regarding continuous SIP trace > with Freeswitch > > On 06.11.21 at 03:03 David Villasmil wrote: > > At the risk of repressing myself: Homer? > > :) > > Yes, David, I already heard it - but I don't know the way from Homer to > pcap - I > want to analyze SIP packages with sngrep! Because it's extremely handy and > powerful! Especially if it comes to compare different messages and measure > time > between arbitrary packages. > > And Homer is bloatware *compared to my requirement*. > > sngrep would be the desired and best solution *for me* if it would be able > to > write correct pcap files while acting as HEP server. > > > Thanks > Gregor > > > > > ---------- Forwarded message ---------- > From: Gregor Maier via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Mon, 08 Nov 2021 10:46:08 -0800 (PST) > Subject: Re: [Freeswitch-users] Question regarding continuous SIP trace > with Freeswitch > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Nov 8 23:20:11 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 8 Nov 2021 23:20:11 +0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: I’ve never tried actually manually setting the reason after hangup, you may want to try that. On Mon, 8 Nov 2021 at 19:03, Rahman Duran wrote: > Hi David, > > You are right but I don't want or need to change any freeswitch internals. > All I need is to fiddle with cdr. So I wonder if the CSV CDR template has > any dynamic mechanism to use on variables. For example can I say "if > variable A is not empty use A, else use variable B" in the CDR template? > > Regards, > > Rahman > > David Villasmil , 6 Kas 2021 Cmt, 05:07 > tarihinde şunu yazdı: > >> The fact is A is hanging up the call. I don’t think you can actually >> change this without changing FS source code to override it. >> >> >> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >> wrote: >> >>> Hi, >>> >>> Bump. Any hints on this? >>> >>> Regards, >>> >>> Rahman >>> >>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>> tarihinde şunu yazdı: >>> >>>> Hi, >>>> >>>> I am using announcements for fail hungup causes like busy, no_answer >>>> etc. Here is my dial plan >>>> >>>> >>>> >>>> >>>> >>> expression="^true$|^TRUE$|^True$"/> >>>> >>>> >>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>> var call_pickup_group)}"/> >>>> >>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>> >>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>> >>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>> >>>> >>>> >>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>> >>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>> var effective_caller_id_name)}"/> >>>> >>>> >>>> >>>> >>>> >>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>> var effective_caller_id_name)}}user/${destination_number}@ >>>> ${domain_name}"/> >>>> >>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>> >>>> >>>> >>>> And here is announcements context that I handle hangup causes: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> expression="^hata-USER_BUSY$"> >>>> >>>> >>>> >>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>> >>>> >>>> >>>> >>>> >>>> >>> expression="^hata-NO_ANSWER$"> >>>> >>>> >>>> >>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>> >>>> >>>> >>>> >>>> >>>> >>> expression="^hata-USER_NOT_REGISTERED$"> >>>> >>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>> >>>> >>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The problem is if the calling leg does not listen the announcement to >>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>> I already know the real hangup cause, how can I override the CDR hangup >>>> cause with the real one? For now I am setting another variable >>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>> this with a real solution. >>>> >>>> Regards, >>>> >>>> Rahman Duran >>>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From martin at pattersong.co.uk Tue Nov 9 09:23:53 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Tue, 9 Nov 2021 09:23:53 +0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: Rahman, CDRs don't have a mechanism like you describe, but variables do. You can put any variable into the CDR and looking back at your original post, you are doing exactly the right thing here by setting a variable (cdr_hata anonsu) with the information you require and putting it in the CDR. Your request was for a real solution - but I think you have it already. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Mon, 8 Nov 2021 at 23:20, David Villasmil wrote: > > I’ve never tried actually manually setting the reason after hangup, you may want to try that. > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran wrote: >> >> Hi David, >> >> You are right but I don't want or need to change any freeswitch internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR template has any dynamic mechanism to use on variables. For example can I say "if variable A is not empty use A, else use variable B" in the CDR template? >> >> Regards, >> >> Rahman >> >> David Villasmil , 6 Kas 2021 Cmt, 05:07 tarihinde şunu yazdı: >>> >>> The fact is A is hanging up the call. I don’t think you can actually change this without changing FS source code to override it. >>> >>> >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran wrote: >>>> >>>> Hi, >>>> >>>> Bump. Any hints on this? >>>> >>>> Regards, >>>> >>>> Rahman >>>> >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 tarihinde şunu yazdı: >>>>> >>>>> Hi, >>>>> >>>>> I am using announcements for fail hungup causes like busy, no_answer etc. Here is my dial plan >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> And here is announcements context that I handle hangup causes: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The problem is if the calling leg does not listen the announcement to the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I already know the real hangup cause, how can I override the CDR hangup cause with the real one? For now I am setting another variable (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix this with a real solution. >>>>> >>>>> Regards, >>>>> >>>>> Rahman Duran >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time https://signalwire.community >>>> >>>> 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 >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time https://signalwire.community >>> >>> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 david.villasmil.work at gmail.com Tue Nov 9 09:41:51 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 9 Nov 2021 09:41:51 +0000 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: <988b2f6d-48d6-199a-c9cb-24167bb51397@mailbox.org> References: <988b2f6d-48d6-199a-c9cb-24167bb51397@mailbox.org> Message-ID: If you have the certs, sure you can decrypt it all. (And you must have, since FS can manage the traffic). Otherwise, look at the profile’s https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/sofia.conf.xml Capture-server to some ip fs will Simply send there a copy of all messages. You can then catch them on that side. On Tue, 9 Nov 2021 at 05:49, Gregor Maier wrote: > > Hello David! > > On 09.11.21 at 00:19 David Villasmil wrote: > > Then why not just run tcpdump filtering for the signaling port and rotate > > every N bytes or Mb or whatever? And then simply pushing those files > > somewhere you can later use them? > > > > Maybe I misunderstood something > > Yes. See my initial post: SIP (and RTP) is TLS'd and I don't have any > certificate > because the VoIP provider owns it. > > > > > If what you want is to go at some on-demand time to trace, then just do > it > > like that. > > That's exactly what I'm doing with Asterisk. Asterisk is able to write > pcap files. > Therefore no problem. > > Just out of curiosity, I tested the actual Homer version (with Asterisk). > But I > gave up, because: > > - correlation does work only partly (maybe an Asterisk problem) > - correlation of ongoing calls doesn't work (~5 minutes e.g. and much more) > - for me, the basic list of result is highly confusing (single call legs > aren't correlated. Even Invite and answer 401 isn't correlated) > - registers aren't correlated > - search for something like CID's doesn't work reliably (it's working 2 or > 3 times > - afterwards no more - tested w/ FF and Chromium - I wasn't able to > search for 2 > CIDs at the same time) > - The GUI makes the browser going crazy (-> one CPU is used 100% after > some time) > > > sngrep is able to correlate single call legs over hours *out of the box* - > why > can't homer do the same? I don't think, a result list based on single > methods is a > good solution - or did I miss something? This could be very possible, > because I > wasn't able to find any reliable and complete and actual documentation. > > If sngrep could safe regular pcap files based on HEP data, sngrep would be > the way > to go - unfortunately, sngrep writes broken pcap files if the input data > stream is > HEP. That's a known missing feature at this time. Even sngrep itself can't > read > those pcap files any more. > > > Thanks > Gregor > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Nov 9 09:42:30 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 9 Nov 2021 09:42:30 +0000 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: <988b2f6d-48d6-199a-c9cb-24167bb51397@mailbox.org> Message-ID: Bye that “other side” can be the same box. On Tue, 9 Nov 2021 at 09:41, David Villasmil wrote: > If you have the certs, sure you can decrypt it all. (And you must have, > since FS can manage the traffic). > Otherwise, look at the profile’s > > > https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/sofia.conf.xml > > Capture-server to some ip fs will > Simply send there a copy of all messages. You can then catch them on that > side. > > On Tue, 9 Nov 2021 at 05:49, Gregor Maier > wrote: > >> >> Hello David! >> >> On 09.11.21 at 00:19 David Villasmil wrote: >> > Then why not just run tcpdump filtering for the signaling port and >> rotate >> > every N bytes or Mb or whatever? And then simply pushing those files >> > somewhere you can later use them? >> > >> > Maybe I misunderstood something >> >> Yes. See my initial post: SIP (and RTP) is TLS'd and I don't have any >> certificate >> because the VoIP provider owns it. >> >> > >> > If what you want is to go at some on-demand time to trace, then just do >> it >> > like that. >> >> That's exactly what I'm doing with Asterisk. Asterisk is able to write >> pcap files. >> Therefore no problem. >> >> Just out of curiosity, I tested the actual Homer version (with Asterisk). >> But I >> gave up, because: >> >> - correlation does work only partly (maybe an Asterisk problem) >> - correlation of ongoing calls doesn't work (~5 minutes e.g. and much >> more) >> - for me, the basic list of result is highly confusing (single call legs >> aren't correlated. Even Invite and answer 401 isn't correlated) >> - registers aren't correlated >> - search for something like CID's doesn't work reliably (it's working 2 >> or 3 times >> - afterwards no more - tested w/ FF and Chromium - I wasn't able to >> search for 2 >> CIDs at the same time) >> - The GUI makes the browser going crazy (-> one CPU is used 100% after >> some time) >> >> >> sngrep is able to correlate single call legs over hours *out of the box* >> - why >> can't homer do the same? I don't think, a result list based on single >> methods is a >> good solution - or did I miss something? This could be very possible, >> because I >> wasn't able to find any reliable and complete and actual documentation. >> >> If sngrep could safe regular pcap files based on HEP data, sngrep would >> be the way >> to go - unfortunately, sngrep writes broken pcap files if the input data >> stream is >> HEP. That's a known missing feature at this time. Even sngrep itself >> can't read >> those pcap files any more. >> >> >> Thanks >> Gregor >> > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rahman.duran at erzurum.edu.tr Tue Nov 9 05:49:49 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Tue, 9 Nov 2021 08:49:49 +0300 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: I already tried to set hangup_cause in my "error anouncement" dialplan extension. But as A leg hangup before playback ends, Freeswitch does not execute remaining actions in the "error anouncement" dialplan extension so it does not work. In "freeradius" configuration, I can write dynamic templates for access and accounting logs like this "%{%{Aruba-Location-Id}:-%{%{Siemens-AP-Name}:-none}}" This will evaluates first "Aruba-Location-Id" and use its value. If it is empty or not exists, then it lookup for "Siemens-AP-Name". If it is empty than it uses string "none". I hoped for some dynamic markup language in Freeswitch csv_cdr module so I can write a template for my needs. I don't want to use xml_cdr because all I need is to key-value based cdr logs sent to syslog (graylog) and analysed there. If this is not possible I will try to use xml_cdr but this will add more complexity and I fear I will loose cdr records is http server is down and Freeswitch continue to operate. P.S. I already log both a and b legs so this is not about legs. Rahman Duran David Villasmil , 9 Kas 2021 Sal, 02:46 tarihinde şunu yazdı: > I’ve never tried actually manually setting the reason after hangup, you > may want to try that. > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran > wrote: > >> Hi David, >> >> You are right but I don't want or need to change any freeswitch >> internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR >> template has any dynamic mechanism to use on variables. For example can I >> say "if variable A is not empty use A, else use variable B" in the CDR >> template? >> >> Regards, >> >> Rahman >> >> David Villasmil , 6 Kas 2021 Cmt, 05:07 >> tarihinde şunu yazdı: >> >>> The fact is A is hanging up the call. I don’t think you can actually >>> change this without changing FS source code to override it. >>> >>> >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >>> wrote: >>> >>>> Hi, >>>> >>>> Bump. Any hints on this? >>>> >>>> Regards, >>>> >>>> Rahman >>>> >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>>> tarihinde şunu yazdı: >>>> >>>>> Hi, >>>>> >>>>> I am using announcements for fail hungup causes like busy, no_answer >>>>> etc. Here is my dial plan >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^true$|^TRUE$|^True$"/> >>>>> >>>>> >>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>>> var call_pickup_group)}"/> >>>>> >>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>>> >>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>>> >>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>>> >>>>> >>>>> >>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>>> >>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}}user/${destination_number}@ >>>>> ${domain_name}"/> >>>>> >>>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>>> >>>>> >>>>> >>>>> And here is announcements context that I handle hangup causes: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_BUSY$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-NO_ANSWER$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_NOT_REGISTERED$"> >>>>> >>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The problem is if the calling leg does not listen the announcement to >>>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>>> I already know the real hangup cause, how can I override the CDR hangup >>>>> cause with the real one? For now I am setting another variable >>>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>>> this with a real solution. >>>>> >>>>> Regards, >>>>> >>>>> Rahman Duran >>>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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 >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 David Villasmil , 9 Kas 2021 Sal, 02:46 tarihinde şunu yazdı: > I’ve never tried actually manually setting the reason after hangup, you > may want to try that. > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran > wrote: > >> Hi David, >> >> You are right but I don't want or need to change any freeswitch >> internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR >> template has any dynamic mechanism to use on variables. For example can I >> say "if variable A is not empty use A, else use variable B" in the CDR >> template? >> >> Regards, >> >> Rahman >> >> David Villasmil , 6 Kas 2021 Cmt, 05:07 >> tarihinde şunu yazdı: >> >>> The fact is A is hanging up the call. I don’t think you can actually >>> change this without changing FS source code to override it. >>> >>> >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >>> wrote: >>> >>>> Hi, >>>> >>>> Bump. Any hints on this? >>>> >>>> Regards, >>>> >>>> Rahman >>>> >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>>> tarihinde şunu yazdı: >>>> >>>>> Hi, >>>>> >>>>> I am using announcements for fail hungup causes like busy, no_answer >>>>> etc. Here is my dial plan >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^true$|^TRUE$|^True$"/> >>>>> >>>>> >>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>>> var call_pickup_group)}"/> >>>>> >>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>>> >>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>>> >>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>>> >>>>> >>>>> >>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>>> >>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}}user/${destination_number}@ >>>>> ${domain_name}"/> >>>>> >>>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>>> >>>>> >>>>> >>>>> And here is announcements context that I handle hangup causes: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_BUSY$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-NO_ANSWER$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_NOT_REGISTERED$"> >>>>> >>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The problem is if the calling leg does not listen the announcement to >>>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>>> I already know the real hangup cause, how can I override the CDR hangup >>>>> cause with the real one? For now I am setting another variable >>>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>>> this with a real solution. >>>>> >>>>> Regards, >>>>> >>>>> Rahman Duran >>>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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 >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 freeswitch13 at mailbox.org Tue Nov 9 05:49:35 2021 From: freeswitch13 at mailbox.org (Gregor Maier) Date: Tue, 9 Nov 2021 06:49:35 +0100 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: Message-ID: <988b2f6d-48d6-199a-c9cb-24167bb51397@mailbox.org> Hello David! On 09.11.21 at 00:19 David Villasmil wrote: > Then why not just run tcpdump filtering for the signaling port and rotate > every N bytes or Mb or whatever? And then simply pushing those files > somewhere you can later use them? > > Maybe I misunderstood something Yes. See my initial post: SIP (and RTP) is TLS'd and I don't have any certificate because the VoIP provider owns it. > > If what you want is to go at some on-demand time to trace, then just do it > like that. That's exactly what I'm doing with Asterisk. Asterisk is able to write pcap files. Therefore no problem. Just out of curiosity, I tested the actual Homer version (with Asterisk). But I gave up, because: - correlation does work only partly (maybe an Asterisk problem) - correlation of ongoing calls doesn't work (~5 minutes e.g. and much more) - for me, the basic list of result is highly confusing (single call legs aren't correlated. Even Invite and answer 401 isn't correlated) - registers aren't correlated - search for something like CID's doesn't work reliably (it's working 2 or 3 times - afterwards no more - tested w/ FF and Chromium - I wasn't able to search for 2 CIDs at the same time) - The GUI makes the browser going crazy (-> one CPU is used 100% after some time) sngrep is able to correlate single call legs over hours *out of the box* - why can't homer do the same? I don't think, a result list based on single methods is a good solution - or did I miss something? This could be very possible, because I wasn't able to find any reliable and complete and actual documentation. If sngrep could safe regular pcap files based on HEP data, sngrep would be the way to go - unfortunately, sngrep writes broken pcap files if the input data stream is HEP. That's a known missing feature at this time. Even sngrep itself can't read those pcap files any more. Thanks Gregor From rahman.duran at erzurum.edu.tr Tue Nov 9 09:57:14 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Tue, 9 Nov 2021 12:57:14 +0300 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: Martin, I was looking for a way to reduce cdr vars :) So I can search or generate reports on the logging server with the same fields. But know I have to consider two separate fields and it does not aggregate nicely on the log server. I also export some variables for "destination_number" as it changes when I do dialplan transfers, and I want to see the original destination number and the reason it changed with exporting lots of other variables. I hopped if I find a way in the cdr template I can combine many of them too. But I see there is no way to do what I want csv_cdr so I will focus on the log parsing side to generate accurate CDRs. Regards, Rahman Martin Paterson , 9 Kas 2021 Sal, 12:43 tarihinde şunu yazdı: > Rahman, > > CDRs don't have a mechanism like you describe, but variables do. You > can put any variable into the CDR and looking back at your original > post, you are doing exactly the right thing here by setting a variable > (cdr_hata anonsu) with the information you require and putting it in > the CDR. Your request was for a real solution - but I think you have > it already. > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > On Mon, 8 Nov 2021 at 23:20, David Villasmil > wrote: > > > > I’ve never tried actually manually setting the reason after hangup, you > may want to try that. > > > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran > wrote: > >> > >> Hi David, > >> > >> You are right but I don't want or need to change any freeswitch > internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR > template has any dynamic mechanism to use on variables. For example can I > say "if variable A is not empty use A, else use variable B" in the CDR > template? > >> > >> Regards, > >> > >> Rahman > >> > >> David Villasmil , 6 Kas 2021 Cmt, > 05:07 tarihinde şunu yazdı: > >>> > >>> The fact is A is hanging up the call. I don’t think you can actually > change this without changing FS source code to override it. > >>> > >>> > >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran > wrote: > >>>> > >>>> Hi, > >>>> > >>>> Bump. Any hints on this? > >>>> > >>>> Regards, > >>>> > >>>> Rahman > >>>> > >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 > tarihinde şunu yazdı: > >>>>> > >>>>> Hi, > >>>>> > >>>>> I am using announcements for fail hungup causes like busy, no_answer > etc. Here is my dial plan > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> expression="^true$|^TRUE$|^True$"/> > >>>>> > >>>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} > var call_pickup_group)}"/> > >>>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> > >>>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> > >>>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> > >>>>> > >>>>> > >>>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> > >>>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} var > effective_caller_id_name)}"/> > >>>>> > >>>>> > >>>>> > >>>>> data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> > >>>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} > var effective_caller_id_name)}}user/${destination_number}@ > ${domain_name}"/> > >>>>> data="hata-${originate_disposition} XML hata_anonslari"/> > >>>>> > >>>>> > >>>>> > >>>>> And here is announcements context that I handle hangup causes: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> expression="^hata-USER_BUSY$"> > >>>>> data="cdr_hata_anonsu=USER_BUSY"/> > >>>>> > >>>>> data="$${anons_dosya_yolu}/user_busy.wav"/> > >>>>> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> expression="^hata-NO_ANSWER$"> > >>>>> data="cdr_hata_anonsu=NO_ANSWER"/> > >>>>> > >>>>> data="$${anons_dosya_yolu}/no_answer.wav"/> > >>>>> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> expression="^hata-USER_NOT_REGISTERED$"> > >>>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> > >>>>> > >>>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> > >>>>> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> data="cdr_hata_anonsu=${originate_disposition}"/> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> The problem is if the calling leg does not listen the announcement > to the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. > As I already know the real hangup cause, how can I override the CDR hangup > cause with the real one? For now I am setting another variable > (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix > this with a real solution. > >>>>> > >>>>> Regards, > >>>>> > >>>>> Rahman Duran > >>>> > >>>> > _________________________________________________________________________ > >>>> > >>>> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > >>>> Build your next product on our scalable cloud platform. > >>>> > >>>> Join our online community to chat in real time > https://signalwire.community > >>>> > >>>> 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 > >>> > >>> -- > >>> Regards, > >>> > >>> David Villasmil > >>> email: david.villasmil.work at gmail.com > >>> phone: +34669448337 > >>> > _________________________________________________________________________ > >>> > >>> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > >>> Build your next product on our scalable cloud platform. > >>> > >>> Join our online community to chat in real time > https://signalwire.community > >>> > >>> 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 > >> > >> > _________________________________________________________________________ > >> > >> The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > >> Build your next product on our scalable cloud platform. > >> > >> Join our online community to chat in real time > https://signalwire.community > >> > >> 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 > > > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 support at telium.io Mon Nov 8 22:38:41 2021 From: support at telium.io (Telium Technical Support) Date: Mon, 8 Nov 2021 22:38:41 +0000 Subject: [Freeswitch-users] Looking for libesl, headers, etc. Message-ID: <0100017d01b41e0c-d932e15f-f9d9-4bd1-b529-36b84016059c-000000@email.amazonses.com> I built a Debian 9 based FreeSwitch+FusionPBX from packages. I now want to create a simple C++ program talking to FS using the ESL. I can't find libesl on the system, nor header files. Is this available as a separate download package? Do I have to DL the whole FS source, compile, and pull that out? -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Tue Nov 9 17:37:54 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Tue, 9 Nov 2021 19:37:54 +0200 Subject: [Freeswitch-users] Looking for libesl, headers, etc. In-Reply-To: <0100017d01b41e0c-d932e15f-f9d9-4bd1-b529-36b84016059c-000000@email.amazonses.com> References: <0100017d01b41e0c-d932e15f-f9d9-4bd1-b529-36b84016059c-000000@email.amazonses.com> Message-ID: it's in the FS repo. libs/esl . yes, you'll have to download/fetch whole FS source. On Tue, Nov 9, 2021 at 4:10 PM Telium Technical Support wrote: > I built a Debian 9 based FreeSwitch+FusionPBX from packages. I now want > to create a simple C++ program talking to FS using the ESL. > > > > I can’t find libesl on the system, nor header files. Is this available as > a separate download package? Do I have to DL the whole FS source, compile, > and pull that out? > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 9 22:49:29 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 9 Nov 2021 22:49:29 +0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: You won't lose cdrs if done right. If fs can't post the CDR it can save the files locally wherever you instruct it to. try this: then do your dialplan. at the end, AFTER the bridge, add let's see what happens... Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Nov 9, 2021 at 3:06 PM Rahman Duran wrote: > I already tried to set hangup_cause in my "error anouncement" dialplan > extension. But as A leg hangup before playback ends, Freeswitch does not > execute remaining actions in the "error anouncement" dialplan extension so > it does not work. > > In "freeradius" configuration, I can write dynamic templates for access > and accounting logs like this > "%{%{Aruba-Location-Id}:-%{%{Siemens-AP-Name}:-none}}" This will evaluates > first "Aruba-Location-Id" and use its value. If it is empty or not exists, > then it lookup for "Siemens-AP-Name". If it is empty than it uses string > "none". > > I hoped for some dynamic markup language in Freeswitch csv_cdr module so I > can write a template for my needs. I don't want to use xml_cdr because all > I need is to key-value based cdr logs sent to syslog (graylog) and analysed > there. If this is not possible I will try to use xml_cdr but this will add > more complexity and I fear I will loose cdr records is http server is down > and Freeswitch continue to operate. > > P.S. I already log both a and b legs so this is not about legs. > > Rahman Duran > > > > David Villasmil , 9 Kas 2021 Sal, 02:46 > tarihinde şunu yazdı: > >> I’ve never tried actually manually setting the reason after hangup, you >> may want to try that. >> >> On Mon, 8 Nov 2021 at 19:03, Rahman Duran >> wrote: >> >>> Hi David, >>> >>> You are right but I don't want or need to change any freeswitch >>> internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR >>> template has any dynamic mechanism to use on variables. For example can I >>> say "if variable A is not empty use A, else use variable B" in the CDR >>> template? >>> >>> Regards, >>> >>> Rahman >>> >>> David Villasmil , 6 Kas 2021 Cmt, 05:07 >>> tarihinde şunu yazdı: >>> >>>> The fact is A is hanging up the call. I don’t think you can actually >>>> change this without changing FS source code to override it. >>>> >>>> >>>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >>>> wrote: >>>> >>>>> Hi, >>>>> >>>>> Bump. Any hints on this? >>>>> >>>>> Regards, >>>>> >>>>> Rahman >>>>> >>>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>>>> tarihinde şunu yazdı: >>>>> >>>>>> Hi, >>>>>> >>>>>> I am using announcements for fail hungup causes like busy, no_answer >>>>>> etc. Here is my dial plan >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^true$|^TRUE$|^True$"/> >>>>>> >>>>>> >>>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>>>> var call_pickup_group)}"/> >>>>>> >>>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>>>> >>>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>>>> >>>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>>>> >>>>>> >>>>>> >>>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>>>> >>>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>>>> var effective_caller_id_name)}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> >>>>>> >>>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>>>> var effective_caller_id_name)}}user/${destination_number}@ >>>>>> ${domain_name}"/> >>>>>> >>>>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>>>> >>>>>> >>>>>> >>>>>> And here is announcements context that I handle hangup causes: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^hata-USER_BUSY$"> >>>>>> >>>>> data="cdr_hata_anonsu=USER_BUSY"/> >>>>>> >>>>>> >>>>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^hata-NO_ANSWER$"> >>>>>> >>>>> data="cdr_hata_anonsu=NO_ANSWER"/> >>>>>> >>>>>> >>>>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^hata-USER_NOT_REGISTERED$"> >>>>>> >>>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>>>> >>>>>> >>>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> The problem is if the calling leg does not listen the announcement to >>>>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>>>> I already know the real hangup cause, how can I override the CDR hangup >>>>>> cause with the real one? For now I am setting another variable >>>>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>>>> this with a real solution. >>>>>> >>>>>> Regards, >>>>>> >>>>>> Rahman Duran >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> 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 >>>> >>>> -- >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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 >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > > David Villasmil , 9 Kas 2021 Sal, 02:46 > tarihinde şunu yazdı: > >> I’ve never tried actually manually setting the reason after hangup, you >> may want to try that. >> >> On Mon, 8 Nov 2021 at 19:03, Rahman Duran >> wrote: >> >>> Hi David, >>> >>> You are right but I don't want or need to change any freeswitch >>> internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR >>> template has any dynamic mechanism to use on variables. For example can I >>> say "if variable A is not empty use A, else use variable B" in the CDR >>> template? >>> >>> Regards, >>> >>> Rahman >>> >>> David Villasmil , 6 Kas 2021 Cmt, 05:07 >>> tarihinde şunu yazdı: >>> >>>> The fact is A is hanging up the call. I don’t think you can actually >>>> change this without changing FS source code to override it. >>>> >>>> >>>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >>>> wrote: >>>> >>>>> Hi, >>>>> >>>>> Bump. Any hints on this? >>>>> >>>>> Regards, >>>>> >>>>> Rahman >>>>> >>>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>>>> tarihinde şunu yazdı: >>>>> >>>>>> Hi, >>>>>> >>>>>> I am using announcements for fail hungup causes like busy, no_answer >>>>>> etc. Here is my dial plan >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^true$|^TRUE$|^True$"/> >>>>>> >>>>>> >>>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>>>> var call_pickup_group)}"/> >>>>>> >>>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>>>> >>>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>>>> >>>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>>>> >>>>>> >>>>>> >>>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>>>> >>>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>>>> var effective_caller_id_name)}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> >>>>>> >>>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>>>> var effective_caller_id_name)}}user/${destination_number}@ >>>>>> ${domain_name}"/> >>>>>> >>>>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>>>> >>>>>> >>>>>> >>>>>> And here is announcements context that I handle hangup causes: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^hata-USER_BUSY$"> >>>>>> >>>>> data="cdr_hata_anonsu=USER_BUSY"/> >>>>>> >>>>>> >>>>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^hata-NO_ANSWER$"> >>>>>> >>>>> data="cdr_hata_anonsu=NO_ANSWER"/> >>>>>> >>>>>> >>>>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^hata-USER_NOT_REGISTERED$"> >>>>>> >>>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>>>> >>>>>> >>>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> The problem is if the calling leg does not listen the announcement to >>>>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>>>> I already know the real hangup cause, how can I override the CDR hangup >>>>>> cause with the real one? For now I am setting another variable >>>>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>>>> this with a real solution. >>>>>> >>>>>> Regards, >>>>>> >>>>>> Rahman Duran >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> 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 >>>> >>>> -- >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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 >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Antony.Stone at freeswitch.open.source.it Tue Nov 9 14:51:18 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 9 Nov 2021 14:51:18 +0000 Subject: [Freeswitch-users] Looking for libesl, headers, etc. In-Reply-To: <0100017d01b41e0c-d932e15f-f9d9-4bd1-b529-36b84016059c-000000@email.amazonses.com> References: <0100017d01b41e0c-d932e15f-f9d9-4bd1-b529-36b84016059c-000000@email.amazonses.com> Message-ID: <202111091451.18584.Antony.Stone@freeswitch.open.source.it> On Monday 08 November 2021 at 22:38:41, Telium Technical Support wrote: > I built a Debian 9 based FreeSwitch+FusionPBX from packages. I now want to > create a simple C++ program talking to FS using the ESL. Why are you building a Debian 9 system? Debian 11 is the current stable release; security support for Debian 9 ended in July 202, and long-term support will end in June 2022. > I can't find libesl on the system, nor header files. Is this available as > a separate download package? Do I have to DL the whole FS source, > compile, and pull that out? According to the documentation at https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library "ESL (libesl) is a C library that has no dependencies on FreeSWITCH, and it can be built, and moved to client/remote machines." Therefore I would expect you to be able to build it directly from https://github.com/signalwire/freeswitch/blob/master/libs/esl/src/esl.c Antony. -- "Measuring average network latency is about as useful as measuring the mean temperature of patients in a hospital." - Stéphane Bortzmeyer Please reply to the list; please *don't* CC me. From freeswitch13 at mailbox.org Tue Nov 9 15:53:49 2021 From: freeswitch13 at mailbox.org (Gregor Maier) Date: Tue, 9 Nov 2021 16:53:49 +0100 Subject: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch In-Reply-To: References: <988b2f6d-48d6-199a-c9cb-24167bb51397@mailbox.org> Message-ID: <27260914-4a62-080b-31a5-c6d2f7247f24@mailbox.org> Hello David, On 09.11.21 at 10:41 David Villasmil wrote: > If you have the certs, sure you can decrypt it all. (And you must have, > since FS can manage the traffic). => No, if FS doesn't act as server (from a TCP perspective - not SIP), FS doesn't need and has no certificates at all. Connecting to a trunk via *flows* (this is standard (here) for trunks for home / private or small business internet customers (usually behind NAT)) doesn't need any server part at all. > Otherwise, look at the profile’s > > https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/sofia.conf.xml > > Capture-server to some ip fs will > Simply send there a copy of all messages. You can then catch them on that > side. That's exactly what's used by homer (heplify-server). Or sngrep. See my other post regarding those tools. Thanks Gregor From Jerry.Kendall at BishopHosting.com Tue Nov 9 17:31:21 2021 From: Jerry.Kendall at BishopHosting.com (Jerry Kendall) Date: Tue, 9 Nov 2021 12:31:21 -0500 Subject: [Freeswitch-users] Meanings of channel status variables In-Reply-To: References: Message-ID: <0185201c-29fc-fe09-5e9e-4a5ca962f1d9@BishopHosting.com> Hey there.... Looking at the end of call stats... I can find/figure out most of the stats but, the ones below are not documented well or at all. can anyone shed some light on what these channel variables represent? variable_rtp_audio_in_flaw_total variable_rtp_audio_in_flush_packet_count variable_rtp_audio_in_jitter_burst_rate variable_rtp_audio_in_jitter_loss_rate variable_rtp_audio_in_jitter_max_variance variable_rtp_audio_in_jitter_min_variance variable_rtp_audio_in_jitter_packet_count variable_rtp_audio_in_largest_jb_size variable_rtp_audio_in_mean_interval variable_rtp_audio_in_mos variable_rtp_audio_in_skip_packet_count Jerry -------------- next part -------------- _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 support at telium.io Wed Nov 10 14:26:08 2021 From: support at telium.io (Telium Technical Support) Date: Wed, 10 Nov 2021 14:26:08 +0000 Subject: [Freeswitch-users] ESL asynchronous use Message-ID: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> I'm trying to figure out asynchronous use of the ESL (from C). If I call esl_connect_timeout with a timeout of 0, what happens if the connection is not made immediately? Is there some callback function indicating success/failure? If the esl connection is lost (FreeSwitch crashes), is there a callback function indicating the lost connection? Or do I need to issue a status/keepalive to ensure function succeeds? Looking a bit at the esl.c code I'm wondering if the above are synchronous only.(polling for status) -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jerry.Kendall at BishopHosting.com Wed Nov 10 15:59:48 2021 From: Jerry.Kendall at BishopHosting.com (Jerry Kendall) Date: Wed, 10 Nov 2021 10:59:48 -0500 Subject: [Freeswitch-users] Meanings of channel status variables In-Reply-To: References: Message-ID: Hey there.... Looking at the end of call stats... I can find/figure out most of the stats but, the ones below are not documented well or at all. can anyone shed some light on what these channel variables represent? variable_rtp_audio_in_flaw_total variable_rtp_audio_in_flush_packet_count variable_rtp_audio_in_jitter_burst_rate variable_rtp_audio_in_jitter_loss_rate variable_rtp_audio_in_jitter_max_variance variable_rtp_audio_in_jitter_min_variance variable_rtp_audio_in_jitter_packet_count variable_rtp_audio_in_largest_jb_size variable_rtp_audio_in_mean_interval variable_rtp_audio_in_mos variable_rtp_audio_in_skip_packet_count Jerry -------------- next part -------------- _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 Thu Nov 11 13:45:45 2021 From: brian at freeswitch.com (Brian West) Date: Thu, 11 Nov 2021 05:45:45 -0800 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> Message-ID: If you have a crash you should perform a back trace and submit an issue on github. /b On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support wrote: > I’m trying to figure out asynchronous use of the ESL (from C). If I call > esl_connect_timeout with a timeout of 0, what happens if the connection is > not made immediately? Is there some callback function indicating > success/failure? > > > > If the esl connection is lost (FreeSwitch crashes), is there a callback > function indicating the lost connection? Or do I need to issue a > status/keepalive to ensure function succeeds? > > > > Looking a bit at the esl.c code I’m wondering if the above are synchronous > only…(polling for status) > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 support at telium.io Thu Nov 11 12:56:20 2021 From: support at telium.io (Telium Technical Support) Date: Thu, 11 Nov 2021 12:56:20 +0000 Subject: [Freeswitch-users] Looking for libesl, headers, etc. In-Reply-To: <202111091451.18584.Antony.Stone@freeswitch.open.source.it> References: <0100017d01b41e0c-d932e15f-f9d9-4bd1-b529-36b84016059c-000000@email.amazonses.com> <202111091451.18584.Antony.Stone@freeswitch.open.source.it> Message-ID: <0100017d0f12095b-8ca18eb4-5bfb-491d-85c2-a5e0d3b2617a-000000@email.amazonses.com> I wanted to try the FusionPBX GUI with FS, and according to the docs (https://docs.fusionpbx.com/en/latest/getting_started/quick_install.html) Debian 9 is the recommended base. Since FS was available as a precompiled package I hoped that the ESL was available as a package as well, but I ended up grabbing the source. -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antony Stone Sent: Tuesday, November 9, 2021 9:51 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Looking for libesl, headers, etc. On Monday 08 November 2021 at 22:38:41, Telium Technical Support wrote: > I built a Debian 9 based FreeSwitch+FusionPBX from packages. I now > want to create a simple C++ program talking to FS using the ESL. Why are you building a Debian 9 system? Debian 11 is the current stable release; security support for Debian 9 ended in July 202, and long-term support will end in June 2022. > I can't find libesl on the system, nor header files. Is this > available as a separate download package? Do I have to DL the whole > FS source, compile, and pull that out? According to the documentation at https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library "ESL (libesl) is a C library that has no dependencies on FreeSWITCH, and it can be built, and moved to client/remote machines." Therefore I would expect you to be able to build it directly from https://github.com/signalwire/freeswitch/blob/master/libs/esl/src/esl.c Antony. -- "Measuring average network latency is about as useful as measuring the mean temperature of patients in a hospital." - Stéphane Bortzmeyer Please reply to the list; please *don't* CC me. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 support at telium.io Thu Nov 11 14:01:51 2021 From: support at telium.io (Telium Technical Support) Date: Thu, 11 Nov 2021 14:01:51 +0000 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> Message-ID: <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> No crash, just learning how to use the ESL. I strongly prefer async design, so wondering if connect and disconnect have some sort of success/failure callback. Or is the only way to check success to send a command over the socket? Same with disconnect, do we need to wait for a send failure? From: Brian West [mailto:brian at freeswitch.com] Sent: Thursday, November 11, 2021 8:46 AM To: Telium Technical Support Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ESL asynchronous use If you have a crash you should perform a back trace and submit an issue on github. /b On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support > wrote: I’m trying to figure out asynchronous use of the ESL (from C). If I call esl_connect_timeout with a timeout of 0, what happens if the connection is not made immediately? Is there some callback function indicating success/failure? If the esl connection is lost (FreeSwitch crashes), is there a callback function indicating the lost connection? Or do I need to issue a status/keepalive to ensure function succeeds? Looking a bit at the esl.c code I’m wondering if the above are synchronous only…(polling for status) _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 344 bytes Desc: not available URL: From krice at freeswitch.org Fri Nov 12 06:05:59 2021 From: krice at freeswitch.org (Ken Rice) Date: Fri, 12 Nov 2021 00:05:59 -0600 Subject: [Freeswitch-users] Looking for libesl, headers, etc. In-Reply-To: <0100017d0f12095b-8ca18eb4-5bfb-491d-85c2-a5e0d3b2617a-000000@email.amazonses.com> References: <0100017d0f12095b-8ca18eb4-5bfb-491d-85c2-a5e0d3b2617a-000000@email.amazonses.com> Message-ID: <00558986-256F-4152-8AF3-487CFCB244E6@freeswitch.org> you should really really use the current version of Debian (11). Debian 9 is past EOL Sent from my iPhone > On Nov 11, 2021, at 17:16, Telium Technical Support wrote: > > I wanted to try the FusionPBX GUI with FS, and according to the docs (https://docs.fusionpbx.com/en/latest/getting_started/quick_install.html) Debian 9 is the recommended base. > > Since FS was available as a precompiled package I hoped that the ESL was available as a package as well, but I ended up grabbing the source. > > -----Original Message----- > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antony Stone > Sent: Tuesday, November 9, 2021 9:51 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Looking for libesl, headers, etc. > >> On Monday 08 November 2021 at 22:38:41, Telium Technical Support wrote: >> >> I built a Debian 9 based FreeSwitch+FusionPBX from packages. I now >> want to create a simple C++ program talking to FS using the ESL. > > Why are you building a Debian 9 system? Debian 11 is the current stable release; security support for Debian 9 ended in July 202, and long-term support will end in June 2022. > >> I can't find libesl on the system, nor header files. Is this >> available as a separate download package? Do I have to DL the whole >> FS source, compile, and pull that out? > > According to the documentation at > https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library "ESL > (libesl) is a C library that has no dependencies on FreeSWITCH, and it can be built, and moved to client/remote machines." > > Therefore I would expect you to be able to build it directly from https://github.com/signalwire/freeswitch/blob/master/libs/esl/src/esl.c > > Antony. > > -- > "Measuring average network latency is about as useful as measuring the mean temperature of patients in a hospital." > > - Stéphane Bortzmeyer > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 grcamauer at gmail.com Fri Nov 12 13:31:16 2021 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 12 Nov 2021 10:31:16 -0300 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> Message-ID: There is a SESSION_HEARTBEAT event that you can monitor to see if FS has stopped responding. It also has a lot of other useful information. Guillermo On Thu, Nov 11, 2021 at 8:56 PM Telium Technical Support wrote: > No crash, just learning how to use the ESL. I strongly prefer async > design, so wondering if connect and disconnect have some sort of > success/failure callback. > > > > Or is the only way to check success to send a command over the socket? > > > > Same with disconnect, do we need to wait for a send failure? > > > > *From:* Brian West [mailto:brian at freeswitch.com] > *Sent:* Thursday, November 11, 2021 8:46 AM > *To:* Telium Technical Support > *Cc:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL asynchronous use > > > > If you have a crash you should perform a back trace and submit an issue on > github. > > > > /b > > > > > > On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support < > support at telium.io> wrote: > > I’m trying to figure out asynchronous use of the ESL (from C). If I call > esl_connect_timeout with a timeout of 0, what happens if the connection is > not made immediately? Is there some callback function indicating > success/failure? > > > > If the esl connection is lost (FreeSwitch crashes), is there a callback > function indicating the lost connection? Or do I need to issue a > status/keepalive to ensure function succeeds? > > > > Looking a bit at the esl.c code I’m wondering if the above are synchronous > only…(polling for status) > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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: Image removed by sender. > https://www.facebook.com/signalwireinc?src=email] > [image: Image removed by sender. > https://twitter.com/freeswitch] > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image001.jpg Type: image/jpeg Size: 344 bytes Desc: not available URL: From brian at freeswitch.com Fri Nov 12 20:38:44 2021 From: brian at freeswitch.com (Brian West) Date: Fri, 12 Nov 2021 12:38:44 -0800 Subject: [Freeswitch-users] Bridge to other FS server has no audio until DTMF In-Reply-To: <0100017d0133f552-473a0842-c954-40c8-972a-f7c562fd2ed7-000000@email.amazonses.com> References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com> <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com> <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com> <0100017c5ba82bd9-711151e4-7603-4eea-a6d9-a9029c06383c-000000@email.amazonses.com> <0100017d0133f552-473a0842-c954-40c8-972a-f7c562fd2ed7-000000@email.amazonses.com> Message-ID: First off, don't use export and don't use nolocal, just set it and see what happens. /b On Mon, Nov 8, 2021 at 12:36 PM Avi Marcus wrote: > Is there a way to "fix" the standoff? Perhaps a header to send or a > channel variable to set? > > I'd like to do bypass_media to cut one of my servers out of the media > path, but then I won't be able to queue an rfc2833 digit press. > > Thanks, > -Avi Marcus > BestFone > > > On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus wrote: > >> I had to do this to get it to execute on the B leg: >> >> >> ... but it didn't help. Only DTMF worked... either manually dialed or via >> queue_dtmf >> Freeswitch A waited for my DTMF to actually send the silence. >> Version 1.10.6 -release-18-1ff9d0a60e 64bit >> >> >> 2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort >> noise payload to 13 >> 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/ >> JOIN_CLASS_7229999 at voip.bestfone.com] has been answered >> EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com >> playback(silence_stream://100) >> 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec >> Activated L16 at 8000hz 1 channels 20ms >> >> -- 20 seconds later when I pressed a button -- >> >> 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done >> playing file silence_stream://100 >> 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/ >> JOIN_CLASS_7229999 at voip.bestfone.com) Callstate Change DOWN -> ACTIVE >> 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 >> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State Change >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >> 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 >> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) Running State >> Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) >> 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 >> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State >> EXCHANGE_MEDIA >> 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA >> 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet >> for [5] ts=960 dur=160/160/2000 seq=26795 lw=960 >> >> >> >> This seemingly shouldn't be an issue. FS1 already has active media from >> the A leg, so it should initiate to the B leg. The B leg has been >> instructed to play a file, so it should initiate to the A leg... >> But if this is somehow unavoidable, perhaps we need a workaround config, >> where we have a simple variable in the bridge string to avoid the standoff? >> >> -Avi Marcus >> >> >> >> On Thu, Oct 7, 2021 at 6:01 PM Brian West wrote: >> >>> execure_on_answer=playback::silence_stream://100 should solve it. >>> >>> /b >>> PS, the non pc term that this has been said to be is >>> https://en.wikipedia.org/wiki/Mexican_standoff >>> >>> On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus wrote: >>> >>>> I meant there's audio from pstn to fs1, but indeed I'm observing no >>>> audio between fs1 and fs2. >>>> >>>> What api should I call with api on answer..? >>>> >>>> On Thu, Oct 7, 2021, 3:19 PM David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> If you see rtp glowing both ways, then this is not the stalemate I was >>>>> talking about. The scenario I’m referring to is about FS not starting >>>>> sending rtp waiting for the other side to start sending, and the other side >>>>> doing the same thing, thus going into a stalemate. This is solved by >>>>> injecting a silence (I would do api_on_answer). >>>>> >>>>> What you’re describing seems different to me. >>>>> >>>>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus wrote: >>>>> >>>>>> I'm using dialplan bridge, so then the dialplan is over. How do I >>>>>> send silence after the bridge...? An api_on_answer with a uuid_broadcast.. >>>>>> seems overly complicated. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> (And I don't know why there isn't audio - I had to set up an audio to >>>>>> get to this options in the IVR... so there's already audio. And Server B >>>>>> also started a file playback so should have initiated audio.) >>>>>> >>>>>> >>>>>> -Avi Marcus >>>>>> >>>>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < >>>>>> david.villasmil.work at gmail.com> wrote: >>>>>> >>>>>>> I seem to remember Brian saying this was because FS is waiting for >>>>>>> the remote end to send audio before starting itself. I believe he >>>>>>> recommended sending an empty (silence) to force the audio stream to be sent >>>>>>> even if fs hasn’t received anything. >>>>>>> >>>>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus wrote: >>>>>>> >>>>>>>> I started a new thread in case anyone muted it... it wasn't simply >>>>>>>> a network issue. >>>>>>>> >>>>>>>> It seems the bridging occurs and dialplan processes, but no media >>>>>>>> flows - until DTMF from the A-leg. >>>>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> >>>>>>>> freeswitch B. >>>>>>>> >>>>>>>> Calls directly from carrier to Freeswitch B are fine. >>>>>>>> Calls from a different carrier to Freeswitch A -> media and IVR -> >>>>>>>> Freeswitch B are also fine. >>>>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is >>>>>>>> in the media path, it's an FS issue... >>>>>>>> >>>>>>>> >>>>>>>> I actually mcguyvered this right now with a queue_dtmf before the >>>>>>>> bridge, to force the audio stream to update. >>>>>>>> >>>>>>>> Here's the log on freeswitch B: >>>>>>>> >>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>>> log(DEBUG class chosen: 1234567) >>>>>>>> 2021-10-07 09:16:24.343175 [DEBUG >>>>>>>> ] mod_dptools.c:1879 class chosen: 1234567 >>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>>> javascript(conference/lookupAndJoinConference.js 1234567) >>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>>> playback(class/hold-wait-teacher.wav) >>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123 >>>>>>>> entering state [completed][200] >>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123 >>>>>>>> entering state [ready][200] >>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz >>>>>>>> 1 channels 20ms >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> 2021-10-07 09:16:34.903283 [DEBUG >>>>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed. >>>>>>>> 2021-10-07 09:16:34.923190 [DEBUG >>>>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 >>>>>>>> 2021-10-07 09:16:34.923190 [INFO >>>>>>>> ] switch_channel.c:522 RECV DTMF 3:2080 >>>>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 >>>>>>>> 2021-10-07 09:16:37.143169 [DEBUG >>>>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav >>>>>>>> >>>>>>>> >>>>>>>> You can see a 10 second gap between call ready 200 and correct >>>>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't >>>>>>>> happen automatically, only when I choose to press something. >>>>>>>> >>>>>>>> >>>>>>>> Any ideas as to the root cause of this? >>>>>>>> >>>>>>>> >>>>>>>> -Avi Marcus >>>>>>>> >>>>>>>> ---------- Forwarded message --------- >>>>>>>> From: Avi Marcus >>>>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM >>>>>>>> Subject: Bridge to other FS server has no audio ??? >>>>>>>> To: FreeSWITCH Users Help >>>>>>>> >>>>>>>> >>>>>>>> Any ideas on why a call doesn't have media? It used to work, but I >>>>>>>> think my upstream changed his SDP again. >>>>>>>> >>>>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to >>>>>>>> FS server B. Media works. >>>>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B >>>>>>>> (not on bypass). Media works. >>>>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges >>>>>>>> to FS server B. Call looks OK, but no media is flowing (I don't hear >>>>>>>> anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All >>>>>>>> the same codecs are set in the json cdrs (PCMU). >>>>>>>> >>>>>>>> FS server B is to join a conference if that matters. >>>>>>>> >>>>>>>> I was assuming it had to do with codecs, but setting >>>>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs - >>>>>>>> it's already always PCMU. >>>>>>>> >>>>>>>> I have NO clue what further could cause this other than codecs, >>>>>>>> which seem to be fine. Any ideas please? >>>>>>>> >>>>>>>> >>>>>>>> -Avi Marcus >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> >>>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>>> https://signalwire.com >>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>>>> services. >>>>>>>> Build your next product on our scalable cloud platform. >>>>>>>> >>>>>>>> Join our online community to chat in real time >>>>>>>> https://signalwire.community >>>>>>>> >>>>>>>> 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 >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work at gmail.com >>>>>>> phone: +34669448337 >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> >>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>> https://signalwire.com >>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>>> services. >>>>>>> Build your next product on our scalable cloud platform. >>>>>>> >>>>>>> Join our online community to chat in real time >>>>>>> https://signalwire.community >>>>>>> >>>>>>> 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 >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>> https://signalwire.com >>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>> services. >>>>>> Build your next product on our scalable cloud platform. >>>>>> >>>>>> Join our online community to chat in real time >>>>>> https://signalwire.community >>>>>> >>>>>> 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 >>>>> >>>>> -- >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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] >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 support at telium.io Fri Nov 12 22:57:57 2021 From: support at telium.io (Telium Technical Support) Date: Fri, 12 Nov 2021 22:57:57 +0000 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> Message-ID: <0100017d165f327a-6143e888-a0cf-4347-88db-2cfa3c3603d2-000000@email.amazonses.com> I looked at the ESL C code and I *THINK* that I have to poll using the esl_recv_event_timed function (to look for the SESSION_HEARTBEAT event). Is there already a callback function that I can hook into to be notified of events? If not, at which frequency should I poll for events (sleep inbetween) based on anyones experience? I don’t like blocking code so I’m hoping there is already a callback function setup. Or is there some other way to handle this? From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer Sent: Friday, November 12, 2021 8:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ESL asynchronous use There is a SESSION_HEARTBEAT event that you can monitor to see if FS has stopped responding. It also has a lot of other useful information. Guillermo On Thu, Nov 11, 2021 at 8:56 PM Telium Technical Support > wrote: No crash, just learning how to use the ESL. I strongly prefer async design, so wondering if connect and disconnect have some sort of success/failure callback. Or is the only way to check success to send a command over the socket? Same with disconnect, do we need to wait for a send failure? From: Brian West [mailto:brian at freeswitch.com ] Sent: Thursday, November 11, 2021 8:46 AM To: Telium Technical Support > Cc: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] ESL asynchronous use If you have a crash you should perform a back trace and submit an issue on github. /b On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support > wrote: I’m trying to figure out asynchronous use of the ESL (from C). If I call esl_connect_timeout with a timeout of 0, what happens if the connection is not made immediately? Is there some callback function indicating success/failure? If the esl connection is lost (FreeSwitch crashes), is there a callback function indicating the lost connection? Or do I need to issue a status/keepalive to ensure function succeeds? Looking a bit at the esl.c code I’m wondering if the above are synchronous only…(polling for status) _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 344 bytes Desc: not available URL: From support at telium.io Sat Nov 13 01:31:42 2021 From: support at telium.io (Telium Technical Support) Date: Sat, 13 Nov 2021 01:31:42 +0000 Subject: [Freeswitch-users] Use of esl_recv_event Message-ID: <0100017d16ebf3f2-f82406dc-f3e6-4a50-99f6-2b7061435fd6-000000@email.amazonses.com> I'm looking at how to use the esl_recv_event function in esl.c and the docs are missing some details. Could someone tell me: 1. Do I have to lock a mutex before accessing this function (since it dequeues an event) 2. The third parameter is a ptr to a ptr to a struct. So, do I have to allocate a struct before calling (to get the data)? Or will the lib allocate the struct and I have to free it? Comments from esl.c: /*! \brief Poll the handle's socket until an event is received or a connection error occurs \param handle Handle to poll \param check_q If set to 1, will check the handle queue (handle->race_event) and return the last event from it \param[out] save_event If this is not NULL, will return the event received */ ESL_DECLARE(esl_status_t) esl_recv_event(esl_handle_t *handle, int check_q, esl_event_t **save_event); /*! -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sun Nov 14 16:02:10 2021 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 14 Nov 2021 10:02:10 -0600 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: <0100017d165f327a-6143e888-a0cf-4347-88db-2cfa3c3603d2-000000@email.amazonses.com> References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> <0100017d165f327a-6143e888-a0cf-4347-88db-2cfa3c3603d2-000000@email.amazonses.com> Message-ID: You might like SignalWire RELAY which is a serverless call control API https://docs.signalwire.com/topics/relay/#relay-documentation On Sun, Nov 14, 2021 at 2:27 AM Telium Technical Support wrote: > I looked at the ESL C code and I **THINK** that I have to poll using the > esl_recv_event_timed function (to look for the SESSION_HEARTBEAT event). > > > > Is there already a callback function that I can hook into to be *notified* > of events? If not, at which frequency should I poll for events (sleep > inbetween) based on anyones experience? I don’t like blocking code so I’m > hoping there is already a callback function setup. > > > > Or is there some other way to handle this? > > > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Guillermo > Ruiz Camauer > *Sent:* Friday, November 12, 2021 8:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL asynchronous use > > > > There is a SESSION_HEARTBEAT event that you can monitor to see if FS has > stopped responding. It also has a lot of other useful information. > > > > Guillermo > > > > > > > > > On Thu, Nov 11, 2021 at 8:56 PM Telium Technical Support < > support at telium.io> wrote: > > No crash, just learning how to use the ESL. I strongly prefer async > design, so wondering if connect and disconnect have some sort of > success/failure callback. > > > > Or is the only way to check success to send a command over the socket? > > > > Same with disconnect, do we need to wait for a send failure? > > > > *From:* Brian West [mailto:brian at freeswitch.com] > *Sent:* Thursday, November 11, 2021 8:46 AM > *To:* Telium Technical Support > *Cc:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL asynchronous use > > > > If you have a crash you should perform a back trace and submit an issue on > github. > > > > /b > > > > > > On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support < > support at telium.io> wrote: > > I’m trying to figure out asynchronous use of the ESL (from C). If I call > esl_connect_timeout with a timeout of 0, what happens if the connection is > not made immediately? Is there some callback function indicating > success/failure? > > > > If the esl connection is lost (FreeSwitch crashes), is there a callback > function indicating the lost connection? Or do I need to issue a > status/keepalive to ensure function succeeds? > > > > Looking a bit at the esl.c code I’m wondering if the above are synchronous > only…(polling for status) > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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: Image removed by sender. > https://www.facebook.com/signalwireinc?src=email] > [image: Image removed by sender. > https://twitter.com/freeswitch] > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > > > > > -- > > Guillermo Ruiz Camauer > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 344 bytes Desc: not available URL: From avi at avimarcus.net Sun Nov 14 16:59:49 2021 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 14 Nov 2021 16:59:49 +0000 Subject: [Freeswitch-users] Bridge to other FS server has no audio until DTMF In-Reply-To: References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com> <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com> <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com> <0100017c5ba82bd9-711151e4-7603-4eea-a6d9-a9029c06383c-000000@email.amazonses.com> <0100017d0133f552-473a0842-c954-40c8-972a-f7c562fd2ed7-000000@email.amazonses.com> Message-ID: <0100017d1f640ab3-0a751041-50c3-4678-a002-77940a77ce82-000000@email.amazonses.com> ... I'm not seeing it executing in the logs for leg A or leg B. Maybe because leg A is already answered? I tried it again as an export, and upping it to 500ms, but it didn't force the audio to flow. So far only DTMF has worked. -Avi Marcus On Fri, Nov 12, 2021 at 10:38 PM Brian West wrote: > First off, don't use export and don't use nolocal, just set it and see > what happens. > > /b > > > On Mon, Nov 8, 2021 at 12:36 PM Avi Marcus wrote: > >> Is there a way to "fix" the standoff? Perhaps a header to send or a >> channel variable to set? >> >> I'd like to do bypass_media to cut one of my servers out of the media >> path, but then I won't be able to queue an rfc2833 digit press. >> >> Thanks, >> -Avi Marcus >> BestFone >> >> >> On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus wrote: >> >>> I had to do this to get it to execute on the B leg: >>> >>> >>> ... but it didn't help. Only DTMF worked... either manually dialed or >>> via queue_dtmf >>> Freeswitch A waited for my DTMF to actually send the silence. >>> Version 1.10.6 -release-18-1ff9d0a60e 64bit >>> >>> >>> 2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort >>> noise payload to 13 >>> 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel >>> [sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com] has been answered >>> EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com >>> playback(silence_stream://100) >>> 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec >>> Activated L16 at 8000hz 1 channels 20ms >>> >>> -- 20 seconds later when I pressed a button -- >>> >>> 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done >>> playing file silence_stream://100 >>> 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 >>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) Callstate Change >>> DOWN -> ACTIVE >>> 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 >>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State Change >>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>> 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 >>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) Running State >>> Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) >>> 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 >>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State >>> EXCHANGE_MEDIA >>> 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA >>> 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet >>> for [5] ts=960 dur=160/160/2000 seq=26795 lw=960 >>> >>> >>> >>> This seemingly shouldn't be an issue. FS1 already has active media from >>> the A leg, so it should initiate to the B leg. The B leg has been >>> instructed to play a file, so it should initiate to the A leg... >>> But if this is somehow unavoidable, perhaps we need a workaround config, >>> where we have a simple variable in the bridge string to avoid the standoff? >>> >>> -Avi Marcus >>> >>> >>> >>> On Thu, Oct 7, 2021 at 6:01 PM Brian West wrote: >>> >>>> execure_on_answer=playback::silence_stream://100 should solve it. >>>> >>>> /b >>>> PS, the non pc term that this has been said to be is >>>> https://en.wikipedia.org/wiki/Mexican_standoff >>>> >>>> On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus wrote: >>>> >>>>> I meant there's audio from pstn to fs1, but indeed I'm observing no >>>>> audio between fs1 and fs2. >>>>> >>>>> What api should I call with api on answer..? >>>>> >>>>> On Thu, Oct 7, 2021, 3:19 PM David Villasmil < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>>> If you see rtp glowing both ways, then this is not the stalemate I >>>>>> was talking about. The scenario I’m referring to is about FS not starting >>>>>> sending rtp waiting for the other side to start sending, and the other side >>>>>> doing the same thing, thus going into a stalemate. This is solved by >>>>>> injecting a silence (I would do api_on_answer). >>>>>> >>>>>> What you’re describing seems different to me. >>>>>> >>>>>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus wrote: >>>>>> >>>>>>> I'm using dialplan bridge, so then the dialplan is over. How do I >>>>>>> send silence after the bridge...? An api_on_answer with a uuid_broadcast.. >>>>>>> seems overly complicated. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> (And I don't know why there isn't audio - I had to set up an audio >>>>>>> to get to this options in the IVR... so there's already audio. And Server B >>>>>>> also started a file playback so should have initiated audio.) >>>>>>> >>>>>>> >>>>>>> -Avi Marcus >>>>>>> >>>>>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < >>>>>>> david.villasmil.work at gmail.com> wrote: >>>>>>> >>>>>>>> I seem to remember Brian saying this was because FS is waiting for >>>>>>>> the remote end to send audio before starting itself. I believe he >>>>>>>> recommended sending an empty (silence) to force the audio stream to be sent >>>>>>>> even if fs hasn’t received anything. >>>>>>>> >>>>>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus wrote: >>>>>>>> >>>>>>>>> I started a new thread in case anyone muted it... it wasn't simply >>>>>>>>> a network issue. >>>>>>>>> >>>>>>>>> It seems the bridging occurs and dialplan processes, but no media >>>>>>>>> flows - until DTMF from the A-leg. >>>>>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> >>>>>>>>> freeswitch B. >>>>>>>>> >>>>>>>>> Calls directly from carrier to Freeswitch B are fine. >>>>>>>>> Calls from a different carrier to Freeswitch A -> media and IVR -> >>>>>>>>> Freeswitch B are also fine. >>>>>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is >>>>>>>>> in the media path, it's an FS issue... >>>>>>>>> >>>>>>>>> >>>>>>>>> I actually mcguyvered this right now with a queue_dtmf before the >>>>>>>>> bridge, to force the audio stream to update. >>>>>>>>> >>>>>>>>> Here's the log on freeswitch B: >>>>>>>>> >>>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>>>> log(DEBUG class chosen: 1234567) >>>>>>>>> 2021-10-07 09:16:24.343175 [DEBUG >>>>>>>>> ] mod_dptools.c:1879 class chosen: 1234567 >>>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>>>> javascript(conference/lookupAndJoinConference.js 1234567) >>>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123 >>>>>>>>> playback(class/hold-wait-teacher.wav) >>>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123 >>>>>>>>> entering state [completed][200] >>>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123 >>>>>>>>> entering state [ready][200] >>>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG >>>>>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz >>>>>>>>> 1 channels 20ms >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> 2021-10-07 09:16:34.903283 [DEBUG >>>>>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed. >>>>>>>>> 2021-10-07 09:16:34.923190 [DEBUG >>>>>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 >>>>>>>>> 2021-10-07 09:16:34.923190 [INFO >>>>>>>>> ] switch_channel.c:522 RECV DTMF 3:2080 >>>>>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 >>>>>>>>> 2021-10-07 09:16:37.143169 [DEBUG >>>>>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav >>>>>>>>> >>>>>>>>> >>>>>>>>> You can see a 10 second gap between call ready 200 and correct >>>>>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't >>>>>>>>> happen automatically, only when I choose to press something. >>>>>>>>> >>>>>>>>> >>>>>>>>> Any ideas as to the root cause of this? >>>>>>>>> >>>>>>>>> >>>>>>>>> -Avi Marcus >>>>>>>>> >>>>>>>>> ---------- Forwarded message --------- >>>>>>>>> From: Avi Marcus >>>>>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM >>>>>>>>> Subject: Bridge to other FS server has no audio ??? >>>>>>>>> To: FreeSWITCH Users Help >>>>>>>>> >>>>>>>>> >>>>>>>>> Any ideas on why a call doesn't have media? It used to work, but I >>>>>>>>> think my upstream changed his SDP again. >>>>>>>>> >>>>>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to >>>>>>>>> FS server B. Media works. >>>>>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B >>>>>>>>> (not on bypass). Media works. >>>>>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges >>>>>>>>> to FS server B. Call looks OK, but no media is flowing (I don't hear >>>>>>>>> anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All >>>>>>>>> the same codecs are set in the json cdrs (PCMU). >>>>>>>>> >>>>>>>>> FS server B is to join a conference if that matters. >>>>>>>>> >>>>>>>>> I was assuming it had to do with codecs, but setting >>>>>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs - >>>>>>>>> it's already always PCMU. >>>>>>>>> >>>>>>>>> I have NO clue what further could cause this other than codecs, >>>>>>>>> which seem to be fine. Any ideas please? >>>>>>>>> >>>>>>>>> >>>>>>>>> -Avi Marcus >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> >>>>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>>>> https://signalwire.com >>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and >>>>>>>>> PSTN services. >>>>>>>>> Build your next product on our scalable cloud platform. >>>>>>>>> >>>>>>>>> Join our online community to chat in real time >>>>>>>>> https://signalwire.community >>>>>>>>> >>>>>>>>> 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 >>>>>>>> >>>>>>>> -- >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work at gmail.com >>>>>>>> phone: +34669448337 >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> >>>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>>> https://signalwire.com >>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>>>> services. >>>>>>>> Build your next product on our scalable cloud platform. >>>>>>>> >>>>>>>> Join our online community to chat in real time >>>>>>>> https://signalwire.community >>>>>>>> >>>>>>>> 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 >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> >>>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>>> https://signalwire.com >>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>>> services. >>>>>>> Build your next product on our scalable cloud platform. >>>>>>> >>>>>>> Join our online community to chat in real time >>>>>>> https://signalwire.community >>>>>>> >>>>>>> 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 >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> >>>>>> David Villasmil >>>>>> email: david.villasmil.work at gmail.com >>>>>> phone: +34669448337 >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>> https://signalwire.com >>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>> services. >>>>>> Build your next product on our scalable cloud platform. >>>>>> >>>>>> Join our online community to chat in real time >>>>>> https://signalwire.community >>>>>> >>>>>> 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> 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] >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> 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 >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 support at telium.io Mon Nov 15 16:08:36 2021 From: support at telium.io (Telium Technical Support) Date: Mon, 15 Nov 2021 16:08:36 +0000 Subject: [Freeswitch-users] fs_cli command syntax Message-ID: <0100017d245b81a7-fcc42c3f-3ab6-430d-bf47-2481dd4ef915-000000@email.amazonses.com> I was using the cli trying to enable events. After much playing and searching I discovered that I must prefix my command with slash; eg: /event plain all Why is this? I can't find any reference to why this command needs a slash prefix? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Nov 15 16:55:18 2021 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 15 Nov 2021 17:55:18 +0100 Subject: [Freeswitch-users] fs_cli command syntax In-Reply-To: <0100017d245b81a7-fcc42c3f-3ab6-430d-bf47-2481dd4ef915-000000@email.amazonses.com> References: <0100017d245b81a7-fcc42c3f-3ab6-430d-bf47-2481dd4ef915-000000@email.amazonses.com> Message-ID: On Mon, Nov 15, 2021 at 5:52 PM Telium Technical Support wrote: > I was using the cli trying to enable events. After much playing and > searching I discovered that I must prefix my command with slash; eg: > > > > /event plain all > > > > Why is this? I can’t find any reference to why this command needs a slash > prefix? > https://freeswitch.org/confluence/display/FREESWITCH/Event+System https://freeswitch.org/confluence/pages/viewpage.action?pageId=1048948#CommandLineInterface(fs_cli)-4.2Forwardslash(/)commands -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From support at telium.io Tue Nov 16 21:47:00 2021 From: support at telium.io (Telium Technical Support) Date: Tue, 16 Nov 2021 21:47:00 +0000 Subject: [Freeswitch-users] How to retrieve all waiting events via ESL in C/C++ Message-ID: <0100017d2ab7ac83-845aedfc-51de-4086-b61a-0e332bc48278-000000@email.amazonses.com> I am trying to retrieve all waiting events via ESL, using the esl_recv_event_timed function. The documentation doesn't offer much help. My code below is intended to retrieve all events waiting. But it doesn't work...I just gather an increasing number of headers (always appending onto the last), and the body is the same, for each event. As if the events are not dequeuing. As well, events I expect (eg HEARTBEAT) are not appearing. So it seems I'm not really iterating through the list of waiting events: QStringList body; esl_event_t *save_event = NULL; while (true) { esl_status_t connectionStatus = esl_recv_event_timed(&m_EslHandle, 0, 0, &save_event); if (connectionStatus == ESL_SUCCESS) { if (save_event) { esl_event_header_t* nextHeader = save_event->headers; while ((nextHeader != NULL) && (nextHeader != save_event->last_header)) { headerList << TL_KeyValuePair(nextHeader->name,nextHeader->value); nextHeader = nextHeader->next; } body = QString(m_EslHandle.last_sr_event->body).split("\n",Qt::SkipEmptyParts); } else break; } else { return false; } } Can someone offer a simple code (using C ESL) on how to retrieve all waiting events? -------------- next part -------------- An HTML attachment was scrubbed... URL: From support at telium.io Tue Nov 16 21:48:51 2021 From: support at telium.io (Telium Technical Support) Date: Tue, 16 Nov 2021 21:48:51 +0000 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> <0100017d165f327a-6143e888-a0cf-4347-88db-2cfa3c3603d2-000000@email.amazonses.com> Message-ID: <0100017d2ab95d9d-2d773d5f-24b6-46b1-8454-be8fca1649ae-000000@email.amazonses.com> I didn’t see a C/C++ interface to this, also adds another layer (REST)…and I think is commercial which I can’t do at this point since I’m just doing proof of concept. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, November 14, 2021 11:02 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ESL asynchronous use You might like SignalWire RELAY which is a serverless call control API https://docs.signalwire.com/topics/relay/#relay-documentation On Sun, Nov 14, 2021 at 2:27 AM Telium Technical Support > wrote: I looked at the ESL C code and I *THINK* that I have to poll using the esl_recv_event_timed function (to look for the SESSION_HEARTBEAT event). Is there already a callback function that I can hook into to be notified of events? If not, at which frequency should I poll for events (sleep inbetween) based on anyones experience? I don’t like blocking code so I’m hoping there is already a callback function setup. Or is there some other way to handle this? From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Guillermo Ruiz Camauer Sent: Friday, November 12, 2021 8:31 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] ESL asynchronous use There is a SESSION_HEARTBEAT event that you can monitor to see if FS has stopped responding. It also has a lot of other useful information. Guillermo On Thu, Nov 11, 2021 at 8:56 PM Telium Technical Support > wrote: No crash, just learning how to use the ESL. I strongly prefer async design, so wondering if connect and disconnect have some sort of success/failure callback. Or is the only way to check success to send a command over the socket? Same with disconnect, do we need to wait for a send failure? From: Brian West [mailto:brian at freeswitch.com ] Sent: Thursday, November 11, 2021 8:46 AM To: Telium Technical Support > Cc: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] ESL asynchronous use If you have a crash you should perform a back trace and submit an issue on github. /b On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support > wrote: I’m trying to figure out asynchronous use of the ESL (from C). If I call esl_connect_timeout with a timeout of 0, what happens if the connection is not made immediately? Is there some callback function indicating success/failure? If the esl connection is lost (FreeSwitch crashes), is there a callback function indicating the lost connection? Or do I need to issue a status/keepalive to ensure function succeeds? Looking a bit at the esl.c code I’m wondering if the above are synchronous only…(polling for status) _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 -- Guillermo Ruiz Camauer _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 344 bytes Desc: not available URL: From david.villasmil.work at gmail.com Wed Nov 17 00:54:31 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 17 Nov 2021 00:54:31 +0000 Subject: [Freeswitch-users] ESL asynchronous use In-Reply-To: <0100017d2ab95d9d-2d773d5f-24b6-46b1-8454-be8fca1649ae-000000@email.amazonses.com> References: <0100017d0a3de44a-a73e125f-bf04-4897-80b9-79289f8dbf0f-000000@email.amazonses.com> <0100017d0f4e07b8-9aba0145-4139-4b8e-88b8-ce16c3422603-000000@email.amazonses.com> <0100017d165f327a-6143e888-a0cf-4347-88db-2cfa3c3603d2-000000@email.amazonses.com> <0100017d2ab95d9d-2d773d5f-24b6-46b1-8454-be8fca1649ae-000000@email.amazonses.com> Message-ID: XD On Tue, 16 Nov 2021 at 23:13, Telium Technical Support wrote: > I didn’t see a C/C++ interface to this, also adds another layer (REST)…and > I think is commercial which I can’t do at this point since I’m just doing > proof of concept. > > > > > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Sunday, November 14, 2021 11:02 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL asynchronous use > > > > You might like SignalWire RELAY which is a serverless call control API > > > > https://docs.signalwire.com/topics/relay/#relay-documentation > > > > On Sun, Nov 14, 2021 at 2:27 AM Telium Technical Support < > support at telium.io> wrote: > > I looked at the ESL C code and I **THINK** that I have to poll using the > esl_recv_event_timed function (to look for the SESSION_HEARTBEAT event). > > > > Is there already a callback function that I can hook into to be *notified* > of events? If not, at which frequency should I poll for events (sleep > inbetween) based on anyones experience? I don’t like blocking code so I’m > hoping there is already a callback function setup. > > > > Or is there some other way to handle this? > > > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Guillermo > Ruiz Camauer > *Sent:* Friday, November 12, 2021 8:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL asynchronous use > > > > There is a SESSION_HEARTBEAT event that you can monitor to see if FS has > stopped responding. It also has a lot of other useful information. > > > > Guillermo > > > > > > > > > On Thu, Nov 11, 2021 at 8:56 PM Telium Technical Support < > support at telium.io> wrote: > > No crash, just learning how to use the ESL. I strongly prefer async > design, so wondering if connect and disconnect have some sort of > success/failure callback. > > > > Or is the only way to check success to send a command over the socket? > > > > Same with disconnect, do we need to wait for a send failure? > > > > *From:* Brian West [mailto:brian at freeswitch.com] > *Sent:* Thursday, November 11, 2021 8:46 AM > *To:* Telium Technical Support > *Cc:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL asynchronous use > > > > If you have a crash you should perform a back trace and submit an issue on > github. > > > > /b > > > > > > On Wed, Nov 10, 2021 at 7:50 PM Telium Technical Support < > support at telium.io> wrote: > > I’m trying to figure out asynchronous use of the ESL (from C). If I call > esl_connect_timeout with a timeout of 0, what happens if the connection is > not made immediately? Is there some callback function indicating > success/failure? > > > > If the esl connection is lost (FreeSwitch crashes), is there a callback > function indicating the lost connection? Or do I need to issue a > status/keepalive to ensure function succeeds? > > > > Looking a bit at the esl.c code I’m wondering if the above are synchronous > only…(polling for status) > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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: Image removed by sender. > https://www.facebook.com/signalwireinc?src=email] > [image: Image removed by sender. > https://twitter.com/freeswitch] > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > > > > > -- > > Guillermo Ruiz Camauer > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 344 bytes Desc: not available URL: From support at telium.io Tue Nov 16 22:52:39 2021 From: support at telium.io (Telium Technical Support) Date: Tue, 16 Nov 2021 22:52:39 +0000 Subject: [Freeswitch-users] How to retrieve all waiting events via ESL in C/C++ In-Reply-To: <0100017d2ab7ac83-845aedfc-51de-4086-b61a-0e332bc48278-000000@email.amazonses.com> References: <0100017d2ab7ac83-845aedfc-51de-4086-b61a-0e332bc48278-000000@email.amazonses.com> Message-ID: <0100017d2af3c95b-3447a8d6-b76b-499c-b148-6152b042eb96-000000@email.amazonses.com> Well, I got the code working, but still unsure about the event struct. So I'm freeing the struct with esl_safe_free which I found used in some fs code. Hope that's correct. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Telium Technical Support Sent: Tuesday, November 16, 2021 4:47 PM To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] How to retrieve all waiting events via ESL in C/C++ I am trying to retrieve all waiting events via ESL, using the esl_recv_event_timed function. The documentation doesn't offer much help. My code below is intended to retrieve all events waiting. But it doesn't work...I just gather an increasing number of headers (always appending onto the last), and the body is the same, for each event. As if the events are not dequeuing. As well, events I expect (eg HEARTBEAT) are not appearing. So it seems I'm not really iterating through the list of waiting events: QStringList body; esl_event_t *save_event = NULL; while (true) { esl_status_t connectionStatus = esl_recv_event_timed(&m_EslHandle, 0, 0, &save_event); if (connectionStatus == ESL_SUCCESS) { if (save_event) { esl_event_header_t* nextHeader = save_event->headers; while ((nextHeader != NULL) && (nextHeader != save_event->last_header)) { headerList << TL_KeyValuePair(nextHeader->name,nextHeader->value); nextHeader = nextHeader->next; } body = QString(m_EslHandle.last_sr_event->body).split("\n",Qt::SkipEmptyParts); } else break; } else { return false; } } Can someone offer a simple code (using C ESL) on how to retrieve all waiting events? -------------- next part -------------- An HTML attachment was scrubbed... URL: From glenchristie at hotmail.com Wed Nov 17 04:26:15 2021 From: glenchristie at hotmail.com (Glen Christie) Date: Wed, 17 Nov 2021 04:26:15 +0000 Subject: [Freeswitch-users] mod_xml_cdr & centos 7 via yum install Message-ID: So I installed freeswitch via yum on an update to date centos 7 system using the following instructions from https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 yum install -y https://files.freeswitch.org/repo/yum/centos-release/freeswitch-release-repo-0-1.noarch.rpm epel-release yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-* mod_xml_cdr doesn't seem to be installed by default, it's not in /usr/lib64/freeswitch/mod autoload_configs has xml_cdr.conf.xml and I uncommented in modules.conf.xml but doing a load mod_xml_cdr in the cli just produces the below 2021-11-17 14:43:30.825959 98.40% [CRIT] switch_loadable_module.c:1750 Error Loading module /usr/lib64/freeswitch/mod/mod_xml_cdr.so **/usr/lib64/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file: No such file or directory** Is mod_xml_cdr depreciated now? I cant seem to find any info on whether it is or not. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Wed Nov 17 16:33:09 2021 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 17 Nov 2021 21:33:09 +0500 Subject: [Freeswitch-users] [Not sending re-invite on 5XX message] Message-ID: Hi Users, We are using freeswitch to send calls to carrier through SRV records, what happens is if carrier sends 503 response FreeSWITCH thinks of that call as rejection and sends to other available IPs under SRV record. That's not leading us to non 5xx error, as the carrier told us that if you get 503 don't expect any other response from other available IPs. So is there anyway i can restrict FS to not send re-invites in case of a 503 response? Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From kashif at kashifbukhari.com Wed Nov 17 17:57:06 2021 From: kashif at kashifbukhari.com (Kashif Ali) Date: Wed, 17 Nov 2021 22:57:06 +0500 Subject: [Freeswitch-users] [Not sending re-invite on 5XX message] In-Reply-To: References: Message-ID: check dns srv record you should have setup weight for srv record. On Wed, Nov 17, 2021 at 9:34 PM Bilal Abbasi wrote: > Hi Users, > > We are using freeswitch to send calls to carrier through SRV records, what > happens is if carrier sends 503 response FreeSWITCH thinks of that call as > rejection and sends to other available IPs under SRV record. > That's not leading us to non 5xx error, as the carrier told us that if you > get 503 don't expect any other response from other available IPs. > So is there anyway i can restrict FS to not send re-invites in case of a > 503 response? > > Regards > Bilal Abbasi > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 bilaln018 at gmail.com Wed Nov 17 20:29:47 2021 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 18 Nov 2021 01:29:47 +0500 Subject: [Freeswitch-users] [Not sending re-invite on 5XX message] In-Reply-To: References: Message-ID: Thanks for reply Kashif, but those SRV records are not under my control, they are carrier’s records. Regards On Wed, 17 Nov 2021 at 11:15 PM, Kashif Ali wrote: > check dns srv record you should have setup weight for srv record. > > On Wed, Nov 17, 2021 at 9:34 PM Bilal Abbasi wrote: > >> Hi Users, >> >> We are using freeswitch to send calls to carrier through SRV records, >> what happens is if carrier sends 503 response FreeSWITCH thinks of that >> call as rejection and sends to other available IPs under SRV record. >> That's not leading us to non 5xx error, as the carrier told us that if >> you get 503 don't expect any other response from other available IPs. >> So is there anyway i can restrict FS to not send re-invites in case of a >> 503 response? >> >> Regards >> Bilal Abbasi >> > _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 dajanjua at gmail.com Fri Nov 19 13:37:02 2021 From: dajanjua at gmail.com (Danish Daud) Date: Fri, 19 Nov 2021 18:37:02 +0500 Subject: [Freeswitch-users] S3 Upload Issue Message-ID: Hi Users, We have a scenario, in which we are uploading recording file on "S3" (AWS) after completion of a call through python script. We encountered error in uploading recording file on S3 for some calls and for some it worked fine. After debugging the issue we have this scenario: Currently when the calls hangup after being answered (before we get the below mentioned logs in fs_cli), the script returns us an error (logs attached). But if we get the below mentioned [Debug] logs and call hangup after these, no error occurs and script successfully upload the file to s3. We have not encountered that issue before. [Debug Logs] 2021-11-19 02:58:08.748629 [DEBUG] switch_ivr_async.c:1427 Record session sample rate: 8000 -> 8000 2021-11-19 02:58:08.748629 [DEBUG] switch_core_media_bug.c:970 Attaching BUG to sofia/external/+18162394364 2021-11-19 02:58:08.768629 [DEBUG] switch_ivr_async.c:1717 No silence detection configured; assuming start of speech 2021-11-19 02:58:09.588628 [DEBUG] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-11-19 02:58:09.588628 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2021-11-19 02:59:02.468628 [DEBUG] sofia.c:7406 Channel sofia/internal/3175520010 at 3.210.46.94:12000 entering state [calling][0] 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7406 Channel sofia/internal/3175520010 at 3.210.46.94:12000 entering state [ready][200] 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7416 Remote SDP: v=0 o=Z 0 2 IN IP4 44.227.145.86 s=Z c=IN IP4 44.227.145.86 t=0 0 m=audio 57578 RTP/AVP 0 106 9 8 3 101 98 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5656 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [opus:106:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5517 Set telephone-event payload to 101 at 8000 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5860 Set telephone-event payload to 101 at 8000 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5918 sofia/internal/3175520010 at 3.210.46.94:12000 Set 2833 dtmf send payload to 101 recv payload to 101 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:8427 Processing updated SDP 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:8657 Audio params are unchanged for sofia/internal/3175520010 at 3.210.46.94:12000. ------------------------------------------------------------------------------------------------------------------------------------------ [Upload Error logs] 2021-11-19 02:47:03.348630 [ERR] mod_python.c:317 Error calling python script 2021-11-19 02:47:03.348630 [ERR] mod_python.c:165 Python Error by calling script "s3_upload": Message: Failed to upload /tmp/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav to storage.rvm.sas2.co/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav: An error occurred (BadDigest) when calling the PutObject operation (reached max retries: 4): The Content-MD5 you specified did not match what we received. Exception: None Traceback (most recent call last) File: "/usr/local/freeswitch/scripts/s3_upload.py", line 22, in fsapi File: "/usr/local/lib/python2.7/dist-packages/boto3/s3/inject.py", line 131, in upload_file File: "/usr/local/lib/python2.7/dist-packages/boto3/s3/transfer.py", line 287, in upload_file ------------------------------------------------------------------------------------------------------------------------------------ Any help in this regard is highly appreciated. Regards, Danish Daud -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Sat Nov 20 14:28:18 2021 From: lists at telium.io (TTI) Date: Sat, 20 Nov 2021 14:28:18 +0000 Subject: [Freeswitch-users] Which is right mailing list? Message-ID: <0100017d3dbf7a1c-3244709c-f24d-4763-b054-4b8639ee6322-000000@email.amazonses.com> I'm starting to interact with FS programmatically (in C) and need some advice. Is this the right mailing list? I see a freeswitch-dev list, is that for developers using FS, or developers creating FS? -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Sat Nov 20 14:32:14 2021 From: lists at telium.io (TTI) Date: Sat, 20 Nov 2021 14:32:14 +0000 Subject: [Freeswitch-users] Preserving recovery information during restart Message-ID: <0100017d3dc31398-136a8700-b087-4318-b8d7-b5bcfb211895-000000@email.amazonses.com> I have a FS running and due to a hardware problem need to reboot the machine. I would like to preserve recovery information (so calls can resume). Is it possible to use the shutdown command to gracefully shutdown FS without clearing the recovery data? The manual shows shutdown options as: shutdown [asap|asap restart|cancel|elegant|now|restart|restart asap|restart elegant] and I'm wondering if the 'now' parameters does this? If not I assume I have to kill -SIGKILL the process, to ensure that recovery data is not cleared out? -------------- next part -------------- An HTML attachment was scrubbed... URL: From udy786 at gmail.com Sun Nov 21 01:03:38 2021 From: udy786 at gmail.com (Uday kumar) Date: Sun, 21 Nov 2021 06:33:38 +0530 Subject: [Freeswitch-users] S3 Upload Issue In-Reply-To: References: Message-ID: I will suggest you to use Python 3 and change the code accordingly and try. As I was boto3 in other process project to upload recording to S3 and had same random issue. After shifting to Python 3, never had any problem. On Sun, 21 Nov 2021, 4:05 am Danish Daud, wrote: > Hi Users, > > We have a scenario, in which we are uploading recording file on "S3" (AWS) > after completion of a call through python script. > We encountered error in uploading recording file on S3 for some calls and > for some it worked fine. After debugging the issue we have this scenario: > Currently when the calls hangup after being answered (before we get the > below mentioned logs in fs_cli), the script returns us an error (logs > attached). But if we get the below mentioned [Debug] logs and call hangup > after these, no error occurs and script successfully upload the file to s3. > We have not encountered that issue before. > > [Debug Logs] > 2021-11-19 02:58:08.748629 [DEBUG] switch_ivr_async.c:1427 Record session > sample rate: 8000 -> 8000 > 2021-11-19 02:58:08.748629 [DEBUG] switch_core_media_bug.c:970 Attaching > BUG to sofia/external/+18162394364 > 2021-11-19 02:58:08.768629 [DEBUG] switch_ivr_async.c:1717 No silence > detection configured; assuming start of speech > 2021-11-19 02:58:09.588628 [DEBUG] switch_rtp.c:7793 Correct audio ip/port > confirmed. > 2021-11-19 02:58:09.588628 [DEBUG] switch_core_io.c:448 Setting BUG Codec > PCMU:0 > 2021-11-19 02:59:02.468628 [DEBUG] sofia.c:7406 Channel > sofia/internal/3175520010 at 3.210.46.94:12000 entering state [calling][0] > 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7406 Channel > sofia/internal/3175520010 at 3.210.46.94:12000 entering state [ready][200] > 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7416 Remote SDP: > v=0 > o=Z 0 2 IN IP4 44.227.145.86 > s=Z > c=IN IP4 44.227.145.86 > t=0 0 > m=audio 57578 RTP/AVP 0 106 9 8 3 101 98 > a=rtpmap:106 opus/48000/2 > a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:98 telephone-event/48000 > a=fmtp:98 0-16 > > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5656 Audio Codec > Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec > Compare [opus:106:48000:20:0:2]/[PCMU:0:8000:20:64000:1] > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec > Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec > Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec > Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5517 Set > telephone-event payload to 101 at 8000 > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5860 Set > telephone-event payload to 101 at 8000 > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5918 > sofia/internal/3175520010 at 3.210.46.94:12000 Set 2833 dtmf send payload to > 101 recv payload to 101 > 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:8427 Processing updated SDP > 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:8657 Audio params > are unchanged for sofia/internal/3175520010 at 3.210.46.94:12000. > > ------------------------------------------------------------------------------------------------------------------------------------------ > > [Upload Error logs] > > 2021-11-19 02:47:03.348630 [ERR] mod_python.c:317 Error calling python > script > 2021-11-19 02:47:03.348630 [ERR] mod_python.c:165 Python Error by calling > script "s3_upload": > Message: Failed to upload > /tmp/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav to > storage.rvm.sas2.co/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav: > An error occurred (BadDigest) when calling the PutObject operation (reached > max retries: 4): The Content-MD5 you specified did not match what we > received. > Exception: None > > Traceback (most recent call last) > File: "/usr/local/freeswitch/scripts/s3_upload.py", line 22, in > fsapi > File: "/usr/local/lib/python2.7/dist-packages/boto3/s3/inject.py", > line 131, in upload_file > File: > "/usr/local/lib/python2.7/dist-packages/boto3/s3/transfer.py", line 287, in > upload_file > > ------------------------------------------------------------------------------------------------------------------------------------ > > Any help in this regard is highly appreciated. > > Regards, > Danish Daud > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 udy786 at gmail.com Sun Nov 21 01:16:18 2021 From: udy786 at gmail.com (Uday kumar) Date: Sun, 21 Nov 2021 06:46:18 +0530 Subject: [Freeswitch-users] mod_xml_cdr & centos 7 via yum install In-Reply-To: References: Message-ID: Its not deprecated. I think when you use rpm to install then xml_curl not getting installed default. You need to install Try yum install freeswitch-xml-cdr On Wed, 17 Nov 2021, 10:07 am Glen Christie, wrote: > So I installed freeswitch via yum on an update to date centos 7 system > using the following instructions from > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 > > > > yum install -y > https://files.freeswitch.org/repo/yum/centos-release/freeswitch-release-repo-0-1.noarch.rpm > epel-release > > yum install -y freeswitch-config-vanilla freeswitch-lang-* > freeswitch-sounds-* > > > > mod_xml_cdr doesn’t seem to be installed by default, it’s not in > /usr/lib64/freeswitch/mod > > > > autoload_configs has xml_cdr.conf.xml and I uncommented module="mod_xml_cdr"/> in modules.conf.xml > > > > but doing a load mod_xml_cdr in the cli just produces the below > > > > > > 2021-11-17 14:43:30.825959 98.40% [CRIT] switch_loadable_module.c:1750 > Error Loading module /usr/lib64/freeswitch/mod/mod_xml_cdr.so > > **/usr/lib64/freeswitch/mod/mod_xml_cdr.so: cannot open shared object > file: No such file or directory** > > > > Is mod_xml_cdr depreciated now? I cant seem to find any info on whether > it is or not. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 bilaln018 at gmail.com Mon Nov 22 18:00:02 2021 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Mon, 22 Nov 2021 23:00:02 +0500 Subject: [Freeswitch-users] S3 Upload Issue In-Reply-To: References: Message-ID: Thanks Uday, We did this change and issue is resolved now. Writing here for future reference. Regards Abbasi On Sun, 21 Nov 2021 at 6:08 AM, Uday kumar wrote: > I will suggest you to use Python 3 and change the code accordingly and > try. > > As I was boto3 in other process project to upload recording to S3 and had > same random issue. After shifting to Python 3, never had any problem. > > On Sun, 21 Nov 2021, 4:05 am Danish Daud, wrote: > >> Hi Users, >> >> We have a scenario, in which we are uploading recording file on "S3" >> (AWS) after completion of a call through python script. >> We encountered error in uploading recording file on S3 for some calls and >> for some it worked fine. After debugging the issue we have this scenario: >> Currently when the calls hangup after being answered (before we get the >> below mentioned logs in fs_cli), the script returns us an error (logs >> attached). But if we get the below mentioned [Debug] logs and call hangup >> after these, no error occurs and script successfully upload the file to s3. >> We have not encountered that issue before. >> >> [Debug Logs] >> 2021-11-19 02:58:08.748629 [DEBUG] switch_ivr_async.c:1427 Record session >> sample rate: 8000 -> 8000 >> 2021-11-19 02:58:08.748629 [DEBUG] switch_core_media_bug.c:970 Attaching >> BUG to sofia/external/+18162394364 >> 2021-11-19 02:58:08.768629 [DEBUG] switch_ivr_async.c:1717 No silence >> detection configured; assuming start of speech >> 2021-11-19 02:58:09.588628 [DEBUG] switch_rtp.c:7793 Correct audio >> ip/port confirmed. >> 2021-11-19 02:58:09.588628 [DEBUG] switch_core_io.c:448 Setting BUG Codec >> PCMU:0 >> 2021-11-19 02:59:02.468628 [DEBUG] sofia.c:7406 Channel >> sofia/internal/3175520010 at 3.210.46.94:12000 entering state [calling][0] >> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7406 Channel >> sofia/internal/3175520010 at 3.210.46.94:12000 entering state [ready][200] >> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7416 Remote SDP: >> v=0 >> o=Z 0 2 IN IP4 44.227.145.86 >> s=Z >> c=IN IP4 44.227.145.86 >> t=0 0 >> m=audio 57578 RTP/AVP 0 106 9 8 3 101 98 >> a=rtpmap:106 opus/48000/2 >> a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:98 telephone-event/48000 >> a=fmtp:98 0-16 >> >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5656 Audio Codec >> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [opus:106:48000:20:0:2]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5517 Set >> telephone-event payload to 101 at 8000 >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5860 Set >> telephone-event payload to 101 at 8000 >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5918 >> sofia/internal/3175520010 at 3.210.46.94:12000 Set 2833 dtmf send payload >> to 101 recv payload to 101 >> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:8427 Processing updated SDP >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:8657 Audio params >> are unchanged for sofia/internal/3175520010 at 3.210.46.94:12000. >> >> ------------------------------------------------------------------------------------------------------------------------------------------ >> >> [Upload Error logs] >> >> 2021-11-19 02:47:03.348630 [ERR] mod_python.c:317 Error calling python >> script >> 2021-11-19 02:47:03.348630 [ERR] mod_python.c:165 Python Error by calling >> script "s3_upload": >> Message: Failed to upload >> /tmp/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav to >> storage.rvm.sas2.co/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav: >> An error occurred (BadDigest) when calling the PutObject operation (reached >> max retries: 4): The Content-MD5 you specified did not match what we >> received. >> Exception: None >> >> Traceback (most recent call last) >> File: "/usr/local/freeswitch/scripts/s3_upload.py", line 22, in >> fsapi >> File: >> "/usr/local/lib/python2.7/dist-packages/boto3/s3/inject.py", line 131, in >> upload_file >> File: >> "/usr/local/lib/python2.7/dist-packages/boto3/s3/transfer.py", line 287, in >> upload_file >> >> ------------------------------------------------------------------------------------------------------------------------------------ >> >> Any help in this regard is highly appreciated. >> >> Regards, >> Danish Daud >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 dujinfang at gmail.com Mon Nov 22 23:18:31 2021 From: dujinfang at gmail.com (Seven Du) Date: Tue, 23 Nov 2021 07:18:31 +0800 Subject: [Freeswitch-users] Which is right mailing list? In-Reply-To: <0100017d3dbf7a1c-3244709c-f24d-4763-b054-4b8639ee6322-000000@email.amazonses.com> References: <0100017d3dbf7a1c-3244709c-f24d-4763-b054-4b8639ee6322-000000@email.amazonses.com> Message-ID: yes. dev is for developers working on FS code. On Sun, Nov 21, 2021 at 7:15 AM TTI wrote: > I’m starting to interact with FS programmatically (in C) and need some > advice. Is this the right mailing list? > > > > I see a freeswitch-dev list, is that for developers using FS, or > developers creating FS? > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From dajanjua at gmail.com Mon Nov 22 04:27:29 2021 From: dajanjua at gmail.com (Danish Daud) Date: Mon, 22 Nov 2021 09:27:29 +0500 Subject: [Freeswitch-users] S3 Upload Issue In-Reply-To: References: Message-ID: Thanks! Uday, for your suggestions. Actually I also have not faced that issue while I tried to upload a file to S3 through python script locally (outside of freeswitch). I have tested successfully both with python 2.7 & 3.7. We are using FreeSwitch 1.10.6 & we have tried to build mod_python in a way that it should use python 3 instead of python 2, but not succeed. If someone can guide us regarding that. It might help. Also Can someone understand the logic I send above regarding the logs for successful upload and without these we receive errors ? On Sun, Nov 21, 2021 at 6:17 AM Uday kumar wrote: > I will suggest you to use Python 3 and change the code accordingly and > try. > > As I was boto3 in other process project to upload recording to S3 and had > same random issue. After shifting to Python 3, never had any problem. > > On Sun, 21 Nov 2021, 4:05 am Danish Daud, wrote: > >> Hi Users, >> >> We have a scenario, in which we are uploading recording file on "S3" >> (AWS) after completion of a call through python script. >> We encountered error in uploading recording file on S3 for some calls and >> for some it worked fine. After debugging the issue we have this scenario: >> Currently when the calls hangup after being answered (before we get the >> below mentioned logs in fs_cli), the script returns us an error (logs >> attached). But if we get the below mentioned [Debug] logs and call hangup >> after these, no error occurs and script successfully upload the file to s3. >> We have not encountered that issue before. >> >> [Debug Logs] >> 2021-11-19 02:58:08.748629 [DEBUG] switch_ivr_async.c:1427 Record session >> sample rate: 8000 -> 8000 >> 2021-11-19 02:58:08.748629 [DEBUG] switch_core_media_bug.c:970 Attaching >> BUG to sofia/external/+18162394364 >> 2021-11-19 02:58:08.768629 [DEBUG] switch_ivr_async.c:1717 No silence >> detection configured; assuming start of speech >> 2021-11-19 02:58:09.588628 [DEBUG] switch_rtp.c:7793 Correct audio >> ip/port confirmed. >> 2021-11-19 02:58:09.588628 [DEBUG] switch_core_io.c:448 Setting BUG Codec >> PCMU:0 >> 2021-11-19 02:59:02.468628 [DEBUG] sofia.c:7406 Channel >> sofia/internal/3175520010 at 3.210.46.94:12000 entering state [calling][0] >> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7406 Channel >> sofia/internal/3175520010 at 3.210.46.94:12000 entering state [ready][200] >> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7416 Remote SDP: >> v=0 >> o=Z 0 2 IN IP4 44.227.145.86 >> s=Z >> c=IN IP4 44.227.145.86 >> t=0 0 >> m=audio 57578 RTP/AVP 0 106 9 8 3 101 98 >> a=rtpmap:106 opus/48000/2 >> a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:98 telephone-event/48000 >> a=fmtp:98 0-16 >> >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5656 Audio Codec >> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [opus:106:48000:20:0:2]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >> Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5517 Set >> telephone-event payload to 101 at 8000 >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5860 Set >> telephone-event payload to 101 at 8000 >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5918 >> sofia/internal/3175520010 at 3.210.46.94:12000 Set 2833 dtmf send payload >> to 101 recv payload to 101 >> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:8427 Processing updated SDP >> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:8657 Audio params >> are unchanged for sofia/internal/3175520010 at 3.210.46.94:12000. >> >> ------------------------------------------------------------------------------------------------------------------------------------------ >> >> [Upload Error logs] >> >> 2021-11-19 02:47:03.348630 [ERR] mod_python.c:317 Error calling python >> script >> 2021-11-19 02:47:03.348630 [ERR] mod_python.c:165 Python Error by calling >> script "s3_upload": >> Message: Failed to upload >> /tmp/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav to >> storage.rvm.sas2.co/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav: >> An error occurred (BadDigest) when calling the PutObject operation (reached >> max retries: 4): The Content-MD5 you specified did not match what we >> received. >> Exception: None >> >> Traceback (most recent call last) >> File: "/usr/local/freeswitch/scripts/s3_upload.py", line 22, in >> fsapi >> File: >> "/usr/local/lib/python2.7/dist-packages/boto3/s3/inject.py", line 131, in >> upload_file >> File: >> "/usr/local/lib/python2.7/dist-packages/boto3/s3/transfer.py", line 287, in >> upload_file >> >> ------------------------------------------------------------------------------------------------------------------------------------ >> >> Any help in this regard is highly appreciated. >> >> Regards, >> Danish Daud >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 dajanjua at gmail.com Mon Nov 22 12:15:05 2021 From: dajanjua at gmail.com (Danish Daud) Date: Mon, 22 Nov 2021 17:15:05 +0500 Subject: [Freeswitch-users] S3 Upload Issue In-Reply-To: References: Message-ID: [SOLVED] By switching to FreeSwitch 1.10.7 & then building mod_python3 instead of mod_python resolve the issue. (Please run the "ldconfig" command before configuring FreeSwitch 1.10.7 using "./configure" command while installing from source) Thank you !!! On Mon, Nov 22, 2021 at 9:27 AM Danish Daud wrote: > Thanks! Uday, for your suggestions. > Actually I also have not faced that issue while I tried to upload a file > to S3 through python script locally (outside of freeswitch). I have tested > successfully both with python 2.7 & 3.7. > We are using FreeSwitch 1.10.6 & we have tried to build mod_python in a > way that it should use python 3 instead of python 2, but not succeed. If > someone can guide us regarding that. It might help. > Also Can someone understand the logic I send above regarding the logs for > successful upload and without these we receive errors ? > > > On Sun, Nov 21, 2021 at 6:17 AM Uday kumar wrote: > >> I will suggest you to use Python 3 and change the code accordingly and >> try. >> >> As I was boto3 in other process project to upload recording to S3 and had >> same random issue. After shifting to Python 3, never had any problem. >> >> On Sun, 21 Nov 2021, 4:05 am Danish Daud, wrote: >> >>> Hi Users, >>> >>> We have a scenario, in which we are uploading recording file on "S3" >>> (AWS) after completion of a call through python script. >>> We encountered error in uploading recording file on S3 for some calls >>> and for some it worked fine. After debugging the issue we have this >>> scenario: >>> Currently when the calls hangup after being answered (before we get the >>> below mentioned logs in fs_cli), the script returns us an error (logs >>> attached). But if we get the below mentioned [Debug] logs and call hangup >>> after these, no error occurs and script successfully upload the file to s3. >>> We have not encountered that issue before. >>> >>> [Debug Logs] >>> 2021-11-19 02:58:08.748629 [DEBUG] switch_ivr_async.c:1427 Record >>> session sample rate: 8000 -> 8000 >>> 2021-11-19 02:58:08.748629 [DEBUG] switch_core_media_bug.c:970 Attaching >>> BUG to sofia/external/+18162394364 >>> 2021-11-19 02:58:08.768629 [DEBUG] switch_ivr_async.c:1717 No silence >>> detection configured; assuming start of speech >>> 2021-11-19 02:58:09.588628 [DEBUG] switch_rtp.c:7793 Correct audio >>> ip/port confirmed. >>> 2021-11-19 02:58:09.588628 [DEBUG] switch_core_io.c:448 Setting BUG >>> Codec PCMU:0 >>> 2021-11-19 02:59:02.468628 [DEBUG] sofia.c:7406 Channel >>> sofia/internal/3175520010 at 3.210.46.94:12000 entering state [calling][0] >>> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7406 Channel >>> sofia/internal/3175520010 at 3.210.46.94:12000 entering state [ready][200] >>> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7416 Remote SDP: >>> v=0 >>> o=Z 0 2 IN IP4 44.227.145.86 >>> s=Z >>> c=IN IP4 44.227.145.86 >>> t=0 0 >>> m=audio 57578 RTP/AVP 0 106 9 8 3 101 98 >>> a=rtpmap:106 opus/48000/2 >>> a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=rtpmap:98 telephone-event/48000 >>> a=fmtp:98 0-16 >>> >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >>> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5656 Audio Codec >>> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >>> Compare [opus:106:48000:20:0:2]/[PCMU:0:8000:20:64000:1] >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >>> Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >>> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec >>> Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5517 Set >>> telephone-event payload to 101 at 8000 >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5860 Set >>> telephone-event payload to 101 at 8000 >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5918 >>> sofia/internal/3175520010 at 3.210.46.94:12000 Set 2833 dtmf send payload >>> to 101 recv payload to 101 >>> 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:8427 Processing updated SDP >>> 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:8657 Audio params >>> are unchanged for sofia/internal/3175520010 at 3.210.46.94:12000. >>> >>> ------------------------------------------------------------------------------------------------------------------------------------------ >>> >>> [Upload Error logs] >>> >>> 2021-11-19 02:47:03.348630 [ERR] mod_python.c:317 Error calling python >>> script >>> 2021-11-19 02:47:03.348630 [ERR] mod_python.c:165 Python Error by >>> calling script "s3_upload": >>> Message: Failed to upload >>> /tmp/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav to >>> storage.rvm.sas2.co/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav: >>> An error occurred (BadDigest) when calling the PutObject operation (reached >>> max retries: 4): The Content-MD5 you specified did not match what we >>> received. >>> Exception: None >>> >>> Traceback (most recent call last) >>> File: "/usr/local/freeswitch/scripts/s3_upload.py", line 22, in >>> fsapi >>> File: >>> "/usr/local/lib/python2.7/dist-packages/boto3/s3/inject.py", line 131, in >>> upload_file >>> File: >>> "/usr/local/lib/python2.7/dist-packages/boto3/s3/transfer.py", line 287, in >>> upload_file >>> >>> ------------------------------------------------------------------------------------------------------------------------------------ >>> >>> Any help in this regard is highly appreciated. >>> >>> Regards, >>> Danish Daud >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 glenchristie at hotmail.com Sun Nov 21 05:48:58 2021 From: glenchristie at hotmail.com (Glen Christie) Date: Sun, 21 Nov 2021 05:48:58 +0000 Subject: [Freeswitch-users] mod_xml_cdr & centos 7 via yum install In-Reply-To: References: Message-ID: That did the trick. Thank you very much. From: FreeSWITCH-users On Behalf Of Uday kumar Sent: Sunday, 21 November 2021 11:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_xml_cdr & centos 7 via yum install Its not deprecated. I think when you use rpm to install then xml_curl not getting installed default. You need to install Try yum install freeswitch-xml-cdr On Wed, 17 Nov 2021, 10:07 am Glen Christie, > wrote: So I installed freeswitch via yum on an update to date centos 7 system using the following instructions from https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 yum install -y https://files.freeswitch.org/repo/yum/centos-release/freeswitch-release-repo-0-1.noarch.rpm epel-release yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-* mod_xml_cdr doesn’t seem to be installed by default, it’s not in /usr/lib64/freeswitch/mod autoload_configs has xml_cdr.conf.xml and I uncommented in modules.conf.xml but doing a load mod_xml_cdr in the cli just produces the below 2021-11-17 14:43:30.825959 98.40% [CRIT] switch_loadable_module.c:1750 Error Loading module /usr/lib64/freeswitch/mod/mod_xml_cdr.so **/usr/lib64/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file: No such file or directory** Is mod_xml_cdr depreciated now? I cant seem to find any info on whether it is or not. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 glenchristie at hotmail.com Tue Nov 23 01:38:21 2021 From: glenchristie at hotmail.com (Glen Christie) Date: Tue, 23 Nov 2021 01:38:21 +0000 Subject: [Freeswitch-users] mod_perl coredump Message-ID: Hi, I'm running centos 7 and trying to enable mod_perl (it's required astpp to run). I've installed both via yum and built from source with the latest freeswitch. When I enable mod_perl freeswitch crashes with a coredump (see below) freeswitch at localhost.localdomain> load mod_perl 2021-11-23 11:26:01.754985 99.70% [INFO] switch_stun.c:897 External ip address detected using STUN: removed 2021-11-23 11:26:01.974836 99.70% [INFO] switch_stun.c:897 External ip address detected using STUN: removed 2021-11-23 11:26:02.054927 99.70% [INFO] mod_enum.c:887 ENUM Reloaded 2021-11-23 11:26:02.054927 99.70% [INFO] switch_time.c:1430 Timezone reloaded 1750 definitions 2021-11-23 11:26:02.074919 99.70% [INFO] mod_perl.c:518 Allocated perl intrepreter. Segmentation fault (core dumped) It does this with both the yum install version and building from source. Perl version is:- This is perl 5, version 16, subversion 3 (v5.16.3) built for x86_64-linux-thread-multi What can I do to solve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From sm at opencluster.com.au Tue Nov 23 06:48:30 2021 From: sm at opencluster.com.au (Shane Mitchell) Date: Tue, 23 Nov 2021 06:48:30 +0000 Subject: [Freeswitch-users] originate api: Escaping variable names/values Message-ID: Using the originate API and enterprise origination: I am attempting to specify various variables, such as custom headers and digest credentials, and am trying to understand how escaping is implemented, as I am inserting user-specified values into the string. I need to set it all up in one API call due to the various ways originate is being used. Here's a simple example: originate [sip_h_X-Custom=ESCAPED]sofia/public/user at example.com $park() I have searched the docs and source code, and have a solution that works most of the time, but there are edge cases I can't seem to resolve. I've tried URL encoding but it seems to have no effect here. Escaping with back-slash seems to have an effect, so I've experimented with it below. For example, this works: Escaped: \'123 (inserted in above API command) Actual: '123 (the value of X-Custom observed in the generated INVITE) Expected: '123 While this doesn't: Escaped: \'1\'23 Actual: 123 Expected: '1'23 In fact, for an even number of quotes, none survive, while for an odd number of quotes, only the last quote ever survives: Escaped: \'1\'2\'3 Actual: 12'3 Expected: '1'2'3 Escaped: \'1\'2\'3\'4\'5\'6\'7\'8\'9 Actual: 12345678'9 Expected: '1'2'3'4'5'6'7'8'9 Wrapping the entire argument with single quotes does not change the behavior: Escaped: '\'1\'2\'3\'4\'5\'6\'7\'8\'9' Actual: 12345678'9 Expected: '1'2'3'4'5'6'7'8'9 Besides single quotes, backslashes also act strangely. Many times they work properly: Escaped: a\\b\\c Actual: a\b\c Expected: a\b\c But when escape sequences are in a sequence, they act unexpectedly: Escaped: \\\\ Actual: \ Expected: \\ Escaped: a\\\'b Actual: a'b Expected: a\'b The next example becomes truncated. My guess is that it is interpreting "\n" as a new line. Escaped: a\\nice\\day Actual: a Expected: a\nice\day Can anyone advise if I'm using the correct syntax/this is an issue, or if there's something I'm missing/doing wrong? Version: FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) From k4kaleem at gmail.com Wed Nov 24 22:38:54 2021 From: k4kaleem at gmail.com (kaleem rehman) Date: Wed, 24 Nov 2021 22:38:54 +0000 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Message-ID: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end. any ideas of achieving this Thanks, Kaleem -------------- next part -------------- An HTML attachment was scrubbed... URL: From ehtasham.malik at expertflow.com Thu Nov 25 10:28:31 2021 From: ehtasham.malik at expertflow.com (Ehtasham Ul-Haq) Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI In-Reply-To: References: Message-ID: Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:[image: domain2.png].expertflow.com FB: [image: FB-f-Logo__blue_29.png]/Expertflow LinkedIn: [image: linkedIn.png] /company/expertflow Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow Twitter: [image: twitter.JPG] /Expertflow 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik at expertflow.com ; Skype: *shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: > Hi All, > > our requirement is simple, we will have CALL US button on website > > when they click, we want a call generated to our FS Server via WebRTC (no > need for calls from FS to Users, it will be one way only from User to > Server. > > With call we want to send additional data like URL of page they on, login > if they are logged in. > we can get data like URL and userlogin but want to sent it with SIP call > as SIP Message (Probably as USER to USER Information) so we can pull at > other end. > > any ideas of achieving this > Thanks, > Kaleem > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 k4kaleem at gmail.com Thu Nov 25 20:31:04 2021 From: k4kaleem at gmail.com (kaleem rehman) Date: Thu, 25 Nov 2021 20:31:04 +0000 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI In-Reply-To: References: Message-ID: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards, Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq To: FreeSWITCH Users Help Cc: Ahmed Hasan Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:[image: domain2.png].expertflow.com FB: [image: FB-f-Logo__blue_29.png]/Expertflow LinkedIn: [image: linkedIn.png] /company/expertflow Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow Twitter: [image: twitter.JPG] /Expertflow 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik at expertflow.com ; Skype: *shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: > Hi All, > > our requirement is simple, we will have CALL US button on website > > when they click, we want a call generated to our FS Server via WebRTC (no > need for calls from FS to Users, it will be one way only from User to > Server. > > With call we want to send additional data like URL of page they on, login > if they are logged in. > we can get data like URL and userlogin but want to sent it with SIP call > as SIP Message (Probably as USER to USER Information) so we can pull at > other end. > > any ideas of achieving this > Thanks, > Kaleem > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 On Thu, Nov 25, 2021 at 2:14 PM kaleem rehman wrote: > Salaam Ehtasham, > > we are looking to use JSSIP or SIPJS, we are flexible and can look into > SIPML if for any reason we have to. > > Regards, > Kaleem > > > > Hi > Which Library you are using to start a call from Website ? > > Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) > > WWW:[image: domain2.png].expertflow.com > FB: [image: > FB-f-Logo__blue_29.png]/Expertflow > LinkedIn: [image: linkedIn.png] /company/expertflow > Youtube: [image: > YouTube-social-square_red_128px.png]/user/expertflow > Twitter: [image: twitter.JPG] > /Expertflow > 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark > and Google Talk: ehtasham.malik at expertflow.com > >; Skype: > *shani.awan3* > > > On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman > wrote: > > >* Hi All, > *>>* our requirement is simple, we will have CALL US button on website > *>>* when they click, we want a call generated to our FS Server via WebRTC (no > *>* need for calls from FS to Users, it will be one way only from User to > *>* Server. > *>>* With call we want to send additional data like URL of page they on, login > *>* if they are logged in. > *>* we can get data like URL and userlogin but want to sent it with SIP call > *>* as SIP Message (Probably as USER to USER Information) so we can pull at > *>* other end. > *>>* any ideas of achieving this > *>* Thanks, > *>* Kaleem > *>* _________________________________________________________________________ > *>>* The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > *>* Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > *>* services. > *>* Build your next product on our scalable cloud platform. > *>>* Join our online community to chat in real time > *>* https://signalwire.community > *>>* 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 kaiduanx at yahoo.ca Fri Nov 26 02:57:01 2021 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI In-Reply-To: References: Message-ID: <1168298075.936412.1637895421247@mail.yahoo.com> You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS? /Kaiduan On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman wrote:  Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards,Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq To: FreeSWITCH Users Help Cc: Ahmed Hasan Bcc:  Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ?  Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:.expertflow.com    FB: /Expertflow   LinkedIn:    /company/expertflow Youtube:  /user/expertflow Twitter:  /Expertflow 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik at expertflow.com; Skype: shani.awan3 On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in.we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information)  so we can pull at other end. any ideas of achieving thisThanks,Kaleem_________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 On Thu, Nov 25, 2021 at 2:14 PM kaleem rehman wrote: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards,Kaleem Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:[image: domain2.png].expertflow.com FB: [image: FB-f-Logo__blue_29.png]/Expertflow LinkedIn: [image: linkedIn.png] /company/expertflow Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow Twitter: [image: twitter.JPG] /Expertflow 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik at expertflow.com ; Skype: *shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: > Hi All, > > our requirement is simple, we will have CALL US button on website > > when they click, we want a call generated to our FS Server via WebRTC (no > need for calls from FS to Users, it will be one way only from User to > Server. > > With call we want to send additional data like URL of page they on, login > if they are logged in. > we can get data like URL and userlogin but want to sent it with SIP call > as SIP Message (Probably as USER to USER Information) so we can pull at > other end. > > any ideas of achieving this > Thanks, > Kaleem > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 glenganderton at gmail.com Fri Nov 26 02:50:16 2021 From: glenganderton at gmail.com (Glen G) Date: Fri, 26 Nov 2021 13:50:16 +1100 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: References: Message-ID: Hi, I'm running FusionPBX 4.4 and have recently starting getting the following message: *[ERR] switch_cpp.cpp:48 Cannot queue any more events* This system system load is quite low max 20 calls per second and max 60 calls, hosted on AWS (very low CPU and RAM utilization) I've read on other posts that the issue here would be with a script that is consuming events but not processing them quickly enough. I'd like to get pointed in the right direction to track down what script this might be, is there some debug or logging mode I can enable to track this down? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Nov 26 03:57:32 2021 From: krice at freeswitch.org (Ken Rice) Date: Thu, 25 Nov 2021 21:57:32 -0600 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: References: Message-ID: <548D9113-B132-4708-8948-9F0DE3248AB1@freeswitch.org> your dialer script is probably not fast enough to handle the events is monitoring. Sent from my iPhone > On Nov 25, 2021, at 21:21, Glen G wrote: > >  > Hi, > > I'm running FusionPBX 4.4 and have recently starting getting the following message: > > [ERR] switch_cpp.cpp:48 Cannot queue any more events > > This system system load is quite low max 20 calls per second and max 60 calls, hosted on AWS (very low CPU and RAM utilization) > > I've read on other posts that the issue here would be with a script that is consuming events but not processing them quickly enough. I'd like to get pointed in the right direction to track down what script this might be, is there some debug or logging mode I can enable to track this down? > > Thanks. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 k4kaleem at gmail.com Fri Nov 26 17:35:37 2021 From: k4kaleem at gmail.com (kaleem rehman) Date: Fri, 26 Nov 2021 17:35:37 +0000 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Message-ID: Hi Kaiduan, thanks for looking into it. Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent. no need for user to login to enter credentials, we want simple "*call us"* type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS. Regards, K ---------- Forwarded message ---------- From: kaiduan xie To: "freeswitch-users at lists.freeswitch.org" Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS? /Kaiduan On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < k4kaleem at gmail.com> wrote: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards, Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq To: FreeSWITCH Users Help Cc: Ahmed Hasan Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:[image: domain2.png].expertflow.com FB: [image: FB-f-Logo__blue_29.png]/Expertflow LinkedIn: [image: linkedIn.png] /company/expertflow Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow Twitter: [image: twitter.JPG] /Expertflow 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik at expertflow.com ; Skype: *shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end. any ideas of achieving this Thanks, Kaleem _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community 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 lewis.bergman at gmail.com Fri Nov 26 23:55:20 2021 From: lewis.bergman at gmail.com (Lewis Bergman) Date: Fri, 26 Nov 2021 17:55:20 -0600 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: <548D9113-B132-4708-8948-9F0DE3248AB1@freeswitch.org> References: <548D9113-B132-4708-8948-9F0DE3248AB1@freeswitch.org> Message-ID: fs_cli is the place to start. I will show the exact line of the code that's reporting the issue. You can then look into the code to see what might be going on. On Thu, Nov 25, 2021, 10:09 PM Ken Rice wrote: > your dialer script is probably not fast enough to handle the events is > monitoring. > > > Sent from my iPhone > > On Nov 25, 2021, at 21:21, Glen G wrote: > >  > Hi, > > I'm running FusionPBX 4.4 and have recently starting getting the following > message: > > *[ERR] switch_cpp.cpp:48 Cannot queue any more events* > > This system system load is quite low max 20 calls per second and max 60 > calls, hosted on AWS (very low CPU and RAM utilization) > > I've read on other posts that the issue here would be with a script that > is consuming events but not processing them quickly enough. I'd like to get > pointed in the right direction to track down what script this might be, is > there some debug or logging mode I can enable to track this down? > > Thanks. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 davidswalkabout at gmail.com Sat Nov 27 01:40:35 2021 From: davidswalkabout at gmail.com (David P) Date: Sat, 27 Nov 2021 14:40:35 +1300 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages In-Reply-To: References: Message-ID: If you use Verto, you will need to embed the userID and password of a Freeswitch user in the page resources so it can be passed in the $.verto.init(...) call. You can protect this signaling channel somewhat by rotating the password frequently. However, I don't know if there is any way to protect the large range of ports that FS needs to be open to handle exchange of audio and video. I asked here a few weeks ago if it would be possible to configure FS to ignore requests on these ports from all addresses except those that have an active login on the signaling channel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From glenganderton at gmail.com Fri Nov 26 04:55:26 2021 From: glenganderton at gmail.com (Glen G) Date: Fri, 26 Nov 2021 15:55:26 +1100 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: <548D9113-B132-4708-8948-9F0DE3248AB1@freeswitch.org> References: <548D9113-B132-4708-8948-9F0DE3248AB1@freeswitch.org> Message-ID: Hi Ken, We are using FusionPBX just as a business phone system so utilization isn't too high at all. There are many users of FusionPBX with much much higher loads than me and they don't experience this issue. Any tips on tracking down the script that may be causing this? On Fri, Nov 26, 2021 at 3:14 PM Ken Rice wrote: > your dialer script is probably not fast enough to handle the events is > monitoring. > > > Sent from my iPhone > > On Nov 25, 2021, at 21:21, Glen G wrote: > >  > Hi, > > I'm running FusionPBX 4.4 and have recently starting getting the following > message: > > *[ERR] switch_cpp.cpp:48 Cannot queue any more events* > > This system system load is quite low max 20 calls per second and max 60 > calls, hosted on AWS (very low CPU and RAM utilization) > > I've read on other posts that the issue here would be with a script that > is consuming events but not processing them quickly enough. I'd like to get > pointed in the right direction to track down what script this might be, is > there some debug or logging mode I can enable to track this down? > > Thanks. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 glenganderton at gmail.com Fri Nov 26 04:59:22 2021 From: glenganderton at gmail.com (Glen G) Date: Fri, 26 Nov 2021 15:59:22 +1100 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: References: <548D9113-B132-4708-8948-9F0DE3248AB1@freeswitch.org> Message-ID: Also note the following: This ERROR takes a couple of weeks to appear after rebooting the system. When the error occurs it doesn't seem to affect the performance of the system at all. On Fri, Nov 26, 2021 at 3:55 PM Glen G wrote: > Hi Ken, > > We are using FusionPBX just as a business phone system so utilization > isn't too high at all. There are many users of FusionPBX with much much > higher loads than me and they don't experience this issue. Any tips on > tracking down the script that may be causing this? > > On Fri, Nov 26, 2021 at 3:14 PM Ken Rice wrote: > >> your dialer script is probably not fast enough to handle the events is >> monitoring. >> >> >> Sent from my iPhone >> >> On Nov 25, 2021, at 21:21, Glen G wrote: >> >>  >> Hi, >> >> I'm running FusionPBX 4.4 and have recently starting getting the >> following message: >> >> *[ERR] switch_cpp.cpp:48 Cannot queue any more events* >> >> This system system load is quite low max 20 calls per second and max 60 >> calls, hosted on AWS (very low CPU and RAM utilization) >> >> I've read on other posts that the issue here would be with a script that >> is consuming events but not processing them quickly enough. I'd like to get >> pointed in the right direction to track down what script this might be, is >> there some debug or logging mode I can enable to track this down? >> >> Thanks. >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 steve.aubrey-bradshaw at pipcall.com Fri Nov 26 09:43:10 2021 From: steve.aubrey-bradshaw at pipcall.com (Steve Aubrey-Bradshaw) Date: Fri, 26 Nov 2021 09:43:10 +0000 Subject: [Freeswitch-users] Lua to create gateway XML Message-ID: Hi, I'm trying to use a Lua script to manage a list of SIP gateways on FreeSwitch that are stored in a mysql database. I have a script that I can run from the CLI that prints out the correct XML and I've added the lines below to autoload_configs/lua.conf.xml However, nothing from the script is loading. I suspect that I'm not returning the XML from the script correctly. I'm currently using stream:write Does anyone have any experience with this? Best Regards, Steve From krice at freeswitch.org Sat Nov 27 17:45:26 2021 From: krice at freeswitch.org (Ken Rice) Date: Sat, 27 Nov 2021 11:45:26 -0600 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: References: Message-ID: the script thats originating the calls. 20cps at a mac of 60 concurrent calls can only be dialer traffic. (20cps is enough dialer traffic to light up almost 200 ports of dialer or several hundred ports of conversational traffic) If its pure Fusion I would contact their dev to as Fusion configs while they are running on top of FS add a bunch of things that is FusionPBX specific Sent from my iPhone > On Nov 26, 2021, at 23:11, Glen G wrote: > >  > Hi Ken, > > We are using FusionPBX just as a business phone system so utilization isn't too high at all. There are many users of FusionPBX with much much higher loads than me and they don't experience this issue. Any tips on tracking down the script that may be causing this? > >> On Fri, Nov 26, 2021 at 3:14 PM Ken Rice wrote: >> your dialer script is probably not fast enough to handle the events is monitoring. >> >> >> Sent from my iPhone >> >>>> On Nov 25, 2021, at 21:21, Glen G wrote: >>>> >>>  >>> Hi, >>> >>> I'm running FusionPBX 4.4 and have recently starting getting the following message: >>> >>> [ERR] switch_cpp.cpp:48 Cannot queue any more events >>> >>> This system system load is quite low max 20 calls per second and max 60 calls, hosted on AWS (very low CPU and RAM utilization) >>> >>> I've read on other posts that the issue here would be with a script that is consuming events but not processing them quickly enough. I'd like to get pointed in the right direction to track down what script this might be, is there some debug or logging mode I can enable to track this down? >>> >>> Thanks. >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time https://signalwire.community >>> >>> 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 >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> 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 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > 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 dujinfang at gmail.com Mon Nov 29 03:07:00 2021 From: dujinfang at gmail.com (Seven Du) Date: Mon, 29 Nov 2021 11:07:00 +0800 Subject: [Freeswitch-users] Lua to create gateway XML In-Reply-To: References: Message-ID: not stream:write, you should return a XML_STRING = .... On Sat, Nov 27, 2021 at 1:39 PM Steve Aubrey-Bradshaw < steve.aubrey-bradshaw at pipcall.com> wrote: > Hi, > I'm trying to use a Lua script to manage a list of SIP gateways on > FreeSwitch that are stored in a mysql database. > > I have a script that I can run from the CLI that prints out the correct > XML and I've added the lines below to autoload_configs/lua.conf.xml > > > > > However, nothing from the script is loading. > > I suspect that I'm not returning the XML from the script correctly. I'm > currently using stream:write > > Does anyone have any experience with this? > > Best Regards, > Steve > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From glenganderton at gmail.com Sun Nov 28 23:05:00 2021 From: glenganderton at gmail.com (Glen G) Date: Mon, 29 Nov 2021 10:05:00 +1100 Subject: [Freeswitch-users] FusionPBX - Cannot queue any more events In-Reply-To: References: Message-ID: Thanks for the response Ken. I was getting the 20cps from the show status command (see below). I always thought that was because I have a couple of clients with large ring groups 10-15 extensions in a ring group. With the clients I have it's very unlikely that would be making too many outbound calls. One last question for you though Ken, if I see this ERROR and it doesn't seem to have any effect on the the performance or functionality of the system, do you think I could safely ignore it for now until I upgrade from to the latest version of freeswitch (and fusionPBX)? 1 session(s) per Sec out of max 1000, peak 22, last 5min 3 On Sun, Nov 28, 2021 at 5:12 AM Ken Rice wrote: > the script thats originating the calls. 20cps at a mac of 60 concurrent > calls can only be dialer traffic. (20cps is enough dialer traffic to light > up almost 200 ports of dialer or several hundred ports of conversational > traffic) > > If its pure Fusion I would contact their dev to as Fusion configs while > they are running on top of FS add a bunch of things that is FusionPBX > specific > > Sent from my iPhone > > On Nov 26, 2021, at 23:11, Glen G wrote: > >  > Hi Ken, > > We are using FusionPBX just as a business phone system so utilization > isn't too high at all. There are many users of FusionPBX with much much > higher loads than me and they don't experience this issue. Any tips on > tracking down the script that may be causing this? > > On Fri, Nov 26, 2021 at 3:14 PM Ken Rice wrote: > >> your dialer script is probably not fast enough to handle the events is >> monitoring. >> >> >> Sent from my iPhone >> >> On Nov 25, 2021, at 21:21, Glen G wrote: >> >>  >> Hi, >> >> I'm running FusionPBX 4.4 and have recently starting getting the >> following message: >> >> *[ERR] switch_cpp.cpp:48 Cannot queue any more events* >> >> This system system load is quite low max 20 calls per second and max 60 >> calls, hosted on AWS (very low CPU and RAM utilization) >> >> I've read on other posts that the issue here would be with a script that >> is consuming events but not processing them quickly enough. I'd like to get >> pointed in the right direction to track down what script this might be, is >> there some debug or logging mode I can enable to track this down? >> >> Thanks. >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Mon Nov 29 20:31:02 2021 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Mon, 29 Nov 2021 21:31:02 +0100 Subject: [Freeswitch-users] Freeswitch video playback re-invite Message-ID: Hi, I have a question. Is there any way to trigger re-invite, if video playback is requested? for example I have following extension: I think Freeswitch should do re-invite with video codecs in SDP, but it does not. Maybe I missed something? I checked examples of config: https://github.com/signalwire/freeswitch/blob/master/conf/testing/dialplan/default/0011_play_video_local_stream.xml And so far did not find an answer to how to trigger re-invite. Thanks! -- Best regards, Volodymyr From Antony.Stone at freeswitch.open.source.it Tue Nov 30 09:34:41 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 30 Nov 2021 10:34:41 +0100 Subject: [Freeswitch-users] Serving sofia.conf via lua Message-ID: <202111301034.41664.Antony.Stone@freeswitch.open.source.it> Hi. I am trying to get Freeswitch to get its sofia configuation from lua instead of static XML files. https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl tells me for example that "Section: configuration - This is used to send back configuration files such as sofia.conf." So, I'm pretty sure what I want to do should be possible, but nowhere have I so far been able to find a fully working example telling me how to do it. My understanding so far is that: 1. I need to create a lua script which generates the *complete* XML as would be served from /etc/freeswitch/autoload_configs/sofia.conf.xml and all its included sub-directories/files 2. This XML as output from the script should be enclosed within tags:
... XML goes here ...
3. I should place this lua script in freeswitch's scripting directory, which on my (Debian) machine is /usr/share/freeswitch/scripts 4. I need to enable this script to be run by including in /etc/freeswitch/autoload_configs/lua.conf.xml: The first thing which is not clear to me is what I should do about /etc/freeswitch/autoload_configs/sofia.conf.xml - if I leave it as it is, it appears to be used in the normal way, and my lua script is ignored - if I delete it (or rename it to sofia.conf.xml.noload for example) then sofia doesn't get loaded *at all* and when I go into fs_cli I do not even have a "sofia" command - if I "touch" the filename so that /etc/reeswitch/autoload_configs/sofia.conf.xml exists but is empty, I get the same result egain - the "sofia" command does not even exist in the cli So, can anyone point me at how to join these things up correctly so that I can serve sofia configuration settings from a lua script? I've been told here that: On Friday 05 November 2021 at 12:28:55, David Villasmil wrote: > In FS there’s the embedded lua. You can do almost anything with it. I'm struggling to find the documentation telling me how, though :( Thanks in advance, Antony. -- "It would appear we have reached the limits of what it is possible to achieve with computer technology, although one should be careful with such statements; they tend to sound pretty silly in five years." - John von Neumann (1949) Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Tue Nov 30 10:20:02 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 30 Nov 2021 11:20:02 +0100 Subject: [Freeswitch-users] Serving sofia.conf via lua (updated) Message-ID: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> Hi. Sending again with a bit of clarification - my original message hasn't hit the list yet, so I'm hoping people can reply to just this one... I am trying to get Freeswitch to get its sofia configuation from lua instead of static XML files. https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl tells me for example that "Section: configuration - This is used to send back configuration files such as sofia.conf." Yes, I know that that page refers to mod_xml_curl and not mod_xml_lua but I'm pretty certain that the principle is the same, and https://freeswitch.org/confluence/display/FREESWITCH/Serving+Configuration+with+Lua seems to confirm this, although is it much thinner on examples. I do not that this page also says "there is no Lua page yet, and Lua-related docs are scattered all over the place..." :( So, anyway, I'm pretty sure what I want to do should be possible, but nowhere have I so far been able to find a fully working example telling me how to do it. My understanding so far is that: 1. I need to create a lua script which generates the *complete* XML as would be served from /etc/freeswitch/autoload_configs/sofia.conf.xml and all its included sub-directories/files 2. This XML as output from the script should be enclosed within tags:
... XML goes here ...
3. I should place this lua script in freeswitch's scripting directory, which on my (Debian) machine is /usr/share/freeswitch/scripts 4. I need to enable this script to be run by including in /etc/freeswitch/autoload_configs/lua.conf.xml: The first thing which is not clear to me is what I should do about /etc/freeswitch/autoload_configs/sofia.conf.xml - if I leave it as it is, it appears to be used in the normal way, and my lua script is ignored - if I delete it (or rename it to sofia.conf.xml.noload for example) then sofia doesn't get loaded *at all* and when I go into fs_cli I do not even have a "sofia" command - if I "touch" the filename so that /etc/reeswitch/autoload_configs/sofia.conf.xml exists but is empty, I get the same result egain - the "sofia" command does not even exist in the cli So, can anyone point me at how to join these things up correctly so that I can serve sofia configuration settings from a lua script? I've been told here that: On Friday 05 November 2021 at 12:28:55, David Villasmil wrote: > In FS there’s the embedded lua. You can do almost anything with it. I'm struggling to find the documentation telling me how, though :( Thanks in advance, Antony. -- "It would appear we have reached the limits of what it is possible to achieve with computer technology, although one should be careful with such statements; they tend to sound pretty silly in five years." - John von Neumann (1949) Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Tue Nov 30 13:17:47 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 30 Nov 2021 14:17:47 +0100 Subject: [Freeswitch-users] Serving sofia.conf via lua (updated, partly resolved) In-Reply-To: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> References: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111301417.48250.Antony.Stone@freeswitch.open.source.it> On Tuesday 30 November 2021 at 11:20:02, Antony Stone wrote: > I am trying to get Freeswitch to get its sofia configuation from lua > instead of static XML files. I've worked out what the problem was, even before my question/s appear to have reached the list!? It turns out that I needed to move: much higher up in /etc/freeswitch/autoload_configs/modules.conf.xml so that it got loaded before: After that, my lua script's configuration does get used to configure sofia :) (This seems like either missing documentation, or even an inappropriate default setting to me - after all, in what circumstance would it be better to load lua after nearly everything else, given that it can be used to actually configure everything else and is then needed first?) Anyway, I now have another problem. If I take my (static, nothing fancy yet) and monolithic sofia configuration file and put it into /etc/freeswitch/autoload_configs/sofia.conf.xml (and disable lua) then it does what I expect and all works. If I take the same file and put it into my lua script (correctly enclosed within the extra and
tags), then it does get loaded, but it produces errors relating to some of the variables pre-defined in vars.xml. So, for example, one of my SIP profiles contains: This produces (when sofia gets loaded) the error messages: -------- tport.c:1844 tport_server_addrinfo() tport_server_addrinfo(0x56318f75ee20): su_getaddrinfo($${local_ip_v4}, 5080) for *: Name or service not known nta.c:2276 nta_agent_add_tport() nta: bind($${local_ip_v4}:5080;transport=*;maddr=$${local_ip_v4}): No such file or directory nua_stack.c:195 nua_stack_init() nua: initializing SIP stack failed -------- Inside the console fs_cli I can do: eval $${local_ip_v4} and it gives me the correct answer. If I replace $${local_ip_v4} in the xml configuration delivered by lua with the numeric IP address, then sofia loads and works as it did with the plain XML file in /etc/freeswitch/autoload_configs/sofia.conf.xml However, I see that some other variables, also defined in vars.xml, are being correctly interpreted and used from the XML provided by lua, so this isn't a consistent "you can't reference variables" problem. Does anyone have a clue what I need to do to fix this? Thanks, Antony. -- I bought a book about anti-gravity. The reviews say you can't put it down. Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Tue Nov 30 14:55:11 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 30 Nov 2021 15:55:11 +0100 Subject: [Freeswitch-users] Serving sofia.conf via lua (updated, partly resolved) In-Reply-To: <202111301417.48250.Antony.Stone@freeswitch.open.source.it> References: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> <202111301417.48250.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111301555.11423.Antony.Stone@freeswitch.open.source.it> On Tuesday 30 November 2021 at 14:17:47, Antony Stone wrote: > If I take my (static, nothing fancy yet) and monolithic sofia configuration > file and put it into /etc/freeswitch/autoload_configs/sofia.conf.xml (and > disable lua) then it does what I expect and all works. > > If I take the same file and put it into my lua script (correctly enclosed > within the extra and
tags), then it does get loaded, > but it produces errors relating to some of the variables pre-defined in > vars.xml. <...snip...> > However, I see that some other variables, also defined in vars.xml, are > being correctly interpreted and used from the XML provided by lua, so this > isn't a consistent "you can't reference variables" problem. I'm no longer so sure about *any* of the variables working correctly. I've now converted the static XML being served by lua into something which does the database lookups which I needed lua for in the first place, and that is working very nicely, however any variables included in the XML output are not being interpreted when the configuration is actually used by Freeswitch. Does that mean that variables (for example, defined in vars.xml) simply cannot be used in XML when that XML comes from lua instead of being in a standard file? What's the way round this, so that I can use the values variables as normal in a configuration file, when the configuration is being generated by lua? Thanks, Antony. -- Pavlov is in the pub enjoying a pint. The barman rings for last orders, and Pavlov jumps up exclaiming "Damn! I forgot to feed the dog!" Please reply to the list; please *don't* CC me. From martin at pattersong.co.uk Tue Nov 30 17:57:13 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Tue, 30 Nov 2021 17:57:13 +0000 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. Message-ID: I have an issue that I’m struggling to resolve. I’ve tried this out in the vanilla config. If a bridged destination in the default context: but it has the effect that if a destination returns 302 Moved Temporarily (1002 forwards to 1003 here), then it doesn’t run through the dialplan the same context, it goes to the public context and fails because the destination (1003) is an extension in the default context. The log looks like it’s handling the 302 as if it’s a new call: [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 in context public EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) I must be missing something in my understanding here – I don’t understand why the behaviour is different, but more interestingly, is there a way of getting the enterprise dial to process the 302s in the same way as when dialling one destination? Best wishes, Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S From brian at freeswitch.com Tue Nov 30 18:24:33 2021 From: brian at freeswitch.com (Brian West) Date: Tue, 30 Nov 2021 12:24:33 -0600 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. In-Reply-To: References: Message-ID: You will want to set outbound_redirect_fatal=true, or the enterprise originate will follow the 302, you can safely ignore those if needed. On Tue, Nov 30, 2021 at 12:09 PM Martin Paterson wrote: > I have an issue that I’m struggling to resolve. I’ve tried this out in > the vanilla config. > > If a bridged destination in the default context: > data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> > > but it has the effect that if a destination returns 302 Moved > Temporarily (1002 forwards to 1003 here), then it doesn’t run through > the dialplan the same context, it goes to the public context and fails > because the destination (1003) is an extension in the default context. > The log looks like it’s handling the 302 as if it’s a new call: > > [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving > invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 > 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 > [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 > in context public > EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) > > I must be missing something in my understanding here – I don’t > understand why the behaviour is different, but more interestingly, is > there a way of getting the enterprise dial to process the 302s in the > same way as when dialling one destination? > > Best wishes, > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > 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 Antony.Stone at freeswitch.open.source.it Tue Nov 30 15:55:09 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Tue, 30 Nov 2021 16:55:09 +0100 Subject: [Freeswitch-users] Serving sofia.conf via lua (updated, partly resolved) In-Reply-To: <202111301555.11423.Antony.Stone@freeswitch.open.source.it> References: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> <202111301417.48250.Antony.Stone@freeswitch.open.source.it> <202111301555.11423.Antony.Stone@freeswitch.open.source.it> Message-ID: <202111301655.09341.Antony.Stone@freeswitch.open.source.it> On Tuesday 30 November 2021 at 15:55:11, Antony Stone wrote: > I've now converted the static XML being served by lua into something which > does the database lookups which I needed lua for in the first place, and > that is working very nicely, however any variables included in the XML > output are not being interpreted when the configuration is actually used > by Freeswitch. > > Does that mean that variables (for example, defined in vars.xml) simply > cannot be used in XML when that XML comes from lua instead of being in a > standard file? I found http://lists.freeswitch.org/pipermail/freeswitch-users/2011- October/077171.html which explains how to do this, and I couldn't find this clue anywhere in the documentation at https://freeswitch.org/confluence/ Antony. -- Warum können Seeräuber nicht den Umfang eines Kreises berechnen? Weil sie Piraten... Please reply to the list; please *don't* CC me.