- FreeSWITCH-users - FreeSWITCH Mailing Lists

user context being ignored
by Peter Villeneuve 10 Apr '17

10 Apr '17

Hi all, I'm confused by some strange behavior I'm seeing on my vanilla FS installation. If I have <param name="apply-inbound-acl" value="domains"/> set in my internal sip profile, even my registered user is unable to make any calls since he gets rejected by the 2017-04-07 17:13:10.085105 [WARNING] sofia.c:9983 IP xx.xx.xx.xx Rejected by acl "domains" message in the logs. Shouldn't FS fall back to regular digest authentication like it used to? I know the user (let's call him 1000) is registered properly since running sofia status profile internal reg in the cli shows that he is indeed registered as expected. I made sure in user 1000's directory entry that <variable name="user_context" value="default"/> is set. If 1) I I apply <param name="apply-inbound-acl" value="domains"/> to the internal profile then calls from user 1000 get rejected by acl domains as explained above, with no fallback to digest authentication (don't know why it doesn't challenge the user) if 2) I disable <param name="apply-inbound-acl" value="domains"/> then the call hits the public dialplan instead of the default one even though <variable name="user_context" value="default"/> is set in user 1000's directory xml entry. I haven't used FS in awhile but I recall in the past not having this much trouble trying to get it working as expected. I suppose I could set the context in the internal profile to default but then all kinds of bad guys will be able to make calls on my dime (it's happened before). I want authenticated users only to make calls but this seems to be eluding me. Is this a known bug (my debian installation was built from git 02c0860 2017-03-03 23:35:25Z 32bit) or am I doing something silly? Thanks, Peter

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Re: [Freeswitch-users] [Freeswitch-dev] I'm so grateful
by Giovanni Maruzzelli 10 Apr '17

10 Apr '17

On 10 April 2017 at 08:26, Mohammed Habib <habib(a)alexcoder.com> wrote: > Hi, > > > > I so grateful to you for all the good things you've done, please read my > message here read more <http://lexion-consultants.com/vs.php?8b8a> > > > > Rushing, Mohammed Habib > You're so kind ! I would say: touching! <3 -giovanni > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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getting error/user_not_registered when it should go to the presence_id
by David Villasmil 09 Apr '17

09 Apr '17

Hello all, If i understand properly, by defining a user like: <include> <user id="54321"> <params> <param name="password" value="mypass"/> <param name="directory-exten-visible" value="true"/> <param name="dial-string" value="{presence_id=12345@mydomain ,instant_ringback=true}loopback/12345"/> </params> <variables> <variable name="domain_uuid" value="mydomain/> <variable name="extension_uuid" value="f477242d-d629-483d-845c-c5126b072508"/> <variable name="toll_allow" value=""/> <variable name="accountcode" value=""/> <variable name="user_context" value="default"/> <variable name="directory-visible" value="true"/> <variable name="limit_max" value="5"/> </variables> </user> </include> When the user 54321 is not registered, call should go to 12345@mydomain (which is the same domain) to be handled by further dialplan... But when i try to (on my public profile): <extension name="54321_test" continue="false"> <condition field="context" expression="public" /> <condition field="network_addr" expression="^10\.1\.1\.1$"/> <condition field="destination_number" expression="^(543.*)$" > <action application="info"/> <action application="log" data="ERR DIAL-STRING: ${sofia_contact(*/${dialed_user}@mydomain)}"/> <action application="bridge" data="user/${sofia_contact(*/${destination_number}@${dialed_domain})}" /> </condition> </extension> I invariably get "error/user_not_registered"... Help is greatly appreciated... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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log file not being populated
by David Villasmil 08 Apr '17

08 Apr '17

hello guys, my logfile.conf.xml is as follows: <configuration name="logfile.conf" description="File Logging"> <settings>  <param name="rotate-on-hup" value="true"/> </settings> <profiles> <profile name="default"> <settings>  <param name="logfile" value="/usr/local/freeswitch/log/freeswitch.log"/>  <param name="rollover" value="10485760"/>     <param name="uuid" value="true" /> </settings> </profile> </profiles> </configuration> the logfile _is_ being created but nothing is added to it... i wonder if this ever happened to anyone... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Not proxying session refresh INVITEs when in bypass media mode
by Spencer Thomason 08 Apr '17

08 Apr '17

Hello all, Apologies if this is the wrong place for this type of discussion as its a little dev oriented. JIRA/PR to follow once we get a little better direction. We’re interested in adding a feature to ignore (i.e. not proxy across the bridge) session refresh INVITEs where the SDP has not changed when in bypass media mode, similarly to how ReINVITEs are handled in “normal” mode. So a few questions: - Does this already exist and I missed it? - Is there any recommendation on when to consider the SDP “the same” as the previous SDP? The scenarios I can think of where we would definitely need proxy the ReINVITE are T38, holds and video attribute changes. My thought is to just 200 OK codec changes that keep the same connection IP and port to keep things simpler but I’d like some feedback here. Are there other scenarios I’ve missed? - Where should we store the address and port for later comparison? SWITCH_REMOTE_MEDIA_IP_VARIABLE (remote_media_ip) and SWITCH_REMOTE_MEDIA_PORT_VARIABLE (remote_media_port)? Are these only used for audio currently? I see there are another set for video... - As we would need to parse the connection address and port in the SDP, what would be the preferred way? Just use the sofia sdp parser? Or do it manually? Doing it manually might be faster but I’m sure there would be edge cases we would need to work out i.e. multiple c= lines, etc.. Or is there a “builtin” way I overlooked? - Has anyone tried this and are there any horror stories? - Does anyone have any feedback for the best user interface handles? I.e. potential channel variables and/or SIP profile params? I was thinking something along the lines of a param called “nomedia-ignore-refresh-invite” but that might be confusing and hopefully someone has a better name :-) Thanks! Spencer

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Re: [Freeswitch-users] Changing mode for CDRs
by DanB 07 Apr '17

07 Apr '17

Rick, have used it when we were testing the .csv processing of FreeSWITCH (like a couple of years back). Should be still actual AFAIK, otherwise let me know if you run into any kind of issues. Regarding maintaining some kind of real-time, you can call the script from cron every minute or so and I think that is the closest real-time you can get since until FreeSWITCH will start writing a new file (rotate old one), you cannot process it externally since you will get concurrency on write/read. In CGRateS you will process it instantly based on inotify events but the CDR file needs to be first released by FreeSWITCH. If you want closer to real-time, maybe mod_json_cdr is a better option for you? DanB On 07.04.2017 19:57, freeswitch-users-request(a)lists.freeswitch.org wrote: > Message: 4 > Date: Fri, 7 Apr 2017 18:43:32 +0100 > From: Rick Jarvis<rick(a)magicmail.mooo.com> > Subject: Re: [Freeswitch-users] Changing mode for CDRs > To: FreeSWITCH Users Help<freeswitch-users(a)lists.freeswitch.org> > Message-ID:<1518A107-A9E3-46BA-9838-2D2D141AB01E(a)magicmail.mooo.com> > Content-Type: text/plain; charset="utf-8" > > Thanks Dan - did you use this in conjunction with a file watcher, presumably, to maintain some kind of realtime? > > You say you ?used to? use it, does that mean it?s FS version thing? What I can?t work out is that there are older cdrs in the directory that have 770 permissions, yet I haven?t updated FS or anything like that, so I can?t work out why they would now be writing out with 700 perms!?

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Re: [Freeswitch-users] Changing mode for CDRs
by DanB 07 Apr '17

07 Apr '17

Rick, I don't want to abuse FreeSWITCH mailing list with CGRateS related questions nor I think it is useful for the rest of the people out here. There is a dedicated ML for CGRateS where you can address these details and will be more than happy to assist you. DanB On 07.04.2017 21:09, freeswitch-users-request(a)lists.freeswitch.org wrote: > Message: 4 > Date: Fri, 7 Apr 2017 20:09:24 +0100 > From: Rick Jarvis<rick(a)magicmail.mooo.com> > Subject: Re: [Freeswitch-users] Changing mode for CDRs > To: FreeSWITCH Users Help<freeswitch-users(a)lists.freeswitch.org> > Message-ID:<CC486338-3636-4440-BF94-D2F1306C25B1(a)magicmail.mooo.com> > Content-Type: text/plain; charset="us-ascii" > > Sounds like cdr_json might be a good option. Would that write straight into mongodb in a way that cgrates can monitor it?

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Re: [Freeswitch-users] Changing mode for CDRs
by DanB 07 Apr '17

07 Apr '17

Hi Rick, We used to use this script: https://github.com/cgrates/cgrates/blob/master/data/scripts/freeswitch_cdr_… Let me know if any issues with it (there is some time since I have last checked it). DanB On 07.04.2017 19:26, freeswitch-users-request(a)lists.freeswitch.org wrote: > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 7 Apr 2017 18:25:56 +0100 > From: Rick Jarvis<rick(a)magicmail.mooo.com> > Subject: Re: [Freeswitch-users] Changing mode for CDRs > To: FreeSWITCH Users Help<freeswitch-users(a)lists.freeswitch.org> > Message-ID:<5305262A-7CF9-40FC-A5B8-EB3950F8676E(a)magicmail.mooo.com> > Content-Type: text/plain; charset="utf-8" > > Thanks Ken, yes good point! It?s cdr_csv, trying to get cgrates to read the csv files.

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getting a list of jJOB UUIDs currently running in background
by Nuno Reis 07 Apr '17

07 Apr '17

Hello all. Sometime ago, I don't remember exactly where, don't know if here in the mailing list or jira or somewhere else, I do remember seeing someone showing how to run a lua script in background with 'luarun' and obtaining a job-uuid for that background thread/script which could be used If I'm not mistaken to control that background job including killing it using 'uuid_kill'. Is this achievable? At present time I want to be able to launch a lua script in background a get a uuid for it something similar to a process id which I can use to check If the script is still running and eventually use uuid_kill on it. If someone can elaborate on how to achieve this I'll be very pleased and thankful. Looking forward to hear from you. Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

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Gateway - Configure non standard SIP port
by Dmitry Mordovin 07 Apr '17

07 Apr '17

Hello, My SIP provider use non standard SIP port - 5093. I configure gateway like: /<gateway name="azeko">// // <param name="realm" value="1.2.3.4:5093"/>// // <param name="sip-port" value="5093"/>// // <param name="proxy" value="1.2.3.4:5093"/>// // <param name="expire-seconds" value="3600"/>// // <param name="register" value="false"/>/ /</gateway>/ But any way, FreeSWITCH continue send SIP messages to 5060 port. LOG: /send 1240 bytes to udp/[1.2.3.4]:5060 at 09:09:09.116502:// //------------------------------------------------------------------------// // INVITE sip:92214090@1.2.3.4 SIP/2.0// // Via: SIP/2.0/UDP 1.1.1.1:5080;rport;branch=z9hG4bKrtrNe4BZ0Kprc// // Max-Forwards: 70// //.../ How can I configure non-standard SIP port for gateway? BR, Dmitry

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