November 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch
by David Villasmil 09 Nov '21

09 Nov '21

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_… 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 <freeswitch13(a)mailbox.org> 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(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch
by David Villasmil 09 Nov '21

09 Nov '21

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(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Gregor Maier <freeswitch13(a)mailbox.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Examples of mod_portaudio_stream?
by Support from NetworkedAudio LLC 08 Nov '21

08 Nov '21

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

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Re: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch
by David Villasmil 06 Nov '21

06 Nov '21

At the risk of repressing myself: Homer? :) On Sat, 6 Nov 2021 at 01:41, Gregor Maier via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Gregor Maier <freeswitch13(a)mailbox.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)gmail.com> wrote: > > > >> Homer? > >> > >> On Wed, 3 Nov 2021 at 03:18, Gregor Maier via FreeSWITCH-users < > >> freeswitch-users(a)lists.freeswitch.org> wrote: > >> > >>> > >>> > >>> > >>> ---------- Forwarded message ---------- > >>> From: Gregor Maier <freeswitch13(a)mailbox.org> > >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > >>> 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(a)lists.freeswitch.org> > >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > >>> 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(a)freeswitch.com > >>> https://freeswitch.com > >>> > >>> Official FreeSWITCH Sites > >>> https://freeswitch.com/oss > >>> https://freeswitch.org/confluence > >>> https://cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users(a)lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> https://freeswitch.com > >> > >> -- > >> Regards, > >> > >> David Villasmil > >> email: david.villasmil.work(a)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(a)freeswitch.com > >> https://freeswitch.com > >> > >> Official FreeSWITCH Sites > >> https://freeswitch.com/oss > >> https://freeswitch.org/confluence > >> https://cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> https://freeswitch.com > > > > > > > ---------- Forwarded message ---------- > From: Gregor Maier via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] Question regarding continuous SIP trace with Freeswitch
by David Villasmil 05 Nov '21

05 Nov '21

Homer? On Wed, 3 Nov 2021 at 03:18, Gregor Maier via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Gregor Maier <freeswitch13(a)mailbox.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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reinvite too quickly causing 491
by Jennifer Hashimoto 05 Nov '21

05 Nov '21

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(a)caztel.com <mailto:jennifer.hashimoto@caztel.com> 418-655-5546

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Re: [Freeswitch-users] Documentation for hold?
by Antony Stone 05 Nov '21

05 Nov '21

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.

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Re: [Freeswitch-users] Documentation for hold?
by Antony Stone 05 Nov '21

05 Nov '21

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.

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Developers / packagers here?
by Antony Stone 05 Nov '21

05 Nov '21

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.) -- 90% of networking problems are routing problems. 9 of the remaining 10% are routing problems in the other direction. The remaining 1% might be something else, but check the routing anyway. Please reply to the list; please *don't* CC me.

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Re: [Freeswitch-users] SIP over DTLS support
by Telium Technical Support 04 Nov '21

04 Nov '21

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@lists.freeswitch.org] On Behalf Of kaiduan xie via FreeSWITCH-users Sent: None To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> 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(a)freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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