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Overriding hangup cause in CDR logs
by Rahman Duran 02 Dec '21

02 Dec '21

Hi, I am using announcements for fail hungup causes like busy, no_answer etc. Here is my dial plan <extension name="local_accounts_freeswitch"> <condition regex="all"> <regex field="destination_number" expression="^[0-9]{4}$"/> <regex field="${local_user_exists}" expression="^true$|^TRUE$|^True$"/> <regex field="${santral_kontrol}" expression="^freeswitch$"/> <action application="set" inline="true" data="call_pickup_group=${user_data(${destination_number}@${domain_name} var call_pickup_group)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> <action application="hash" data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> <action application="log" data="ep_codec_string -- ${ep_codec_string}" /> <action application="set" data="inherit_codec=true"/> <action application="export" data="nolocal:absolute_codec_string=${ep_codec_string}"/> <action application="set_profile_var" data="callee_id_name=${user_data(${destination_number}@${domain_name} var effective_caller_id_name)}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=30"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> <action application="bridge" data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/> <action application="execute_extension" data="hata-${originate_disposition} XML hata_anonslari"/> </condition> </extension> And here is announcements context that I handle hangup causes: <include>  <context name="hata_anonslari">  <extension name="error-causes-user_busy"> <condition field="destination_number" expression="^hata-USER_BUSY$"> <action application="export" data="cdr_hata_anonsu=USER_BUSY"/> <action application="pre_answer"/> <action application="playback" data="$${anons_dosya_yolu}/user_busy.wav"/> <action application="sleep" data="1000"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> <extension name="error-causes-no_answer"> <condition field="destination_number" expression="^hata-NO_ANSWER$"> <action application="export" data="cdr_hata_anonsu=NO_ANSWER"/> <action application="pre_answer"/> <action application="playback" data="$${anons_dosya_yolu}/no_answer.wav"/> <action application="sleep" data="1000"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> <extension name="error-causes-user_not_registered"> <condition field="destination_number" expression="^hata-USER_NOT_REGISTERED$"> <action application="export" data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> <action application="pre_answer"/> <action application="playback" data="$${anons_dosya_yolu}/user_not_registered.wav"/> <action application="sleep" data="1000"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> <extension name="error-causes-any"> <condition field="destination_number" expression="^hata-(.*)$"> <action application="export" data="cdr_hata_anonsu=${originate_disposition}"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension>  </context> </include> 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

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Serving sofia.conf via lua
by Antony Stone 30 Nov '21

30 Nov '21

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: <document type="freeswitch/xml"> <section name="configuration"> ... XML goes here ... </section> </document> 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: <param name="xml-handler-script" value="sofia.conf.lua"/> <param name="xml-handler-bindings" value="configuration"/> 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.

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Freeswitch video playback re-invite
by Volodymyr Fedorov 29 Nov '21

29 Nov '21

Hi, I have a question. Is there any way to trigger re-invite, if video playback is requested? for example I have following extension: <extension name="videoplayback"> <condition field="destination_number" expression="^6054$"> <action application="answer"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="playback" data="/opt/video/lipsync.mp4"/> </condition> </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/… And so far did not find an answer to how to trigger re-invite. Thanks! -- Best regards, Volodymyr

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Lua to create gateway XML
by Steve Aubrey-Bradshaw 29 Nov '21

29 Nov '21

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 <param name="xml-handler-script" value="gateways.lua"/> <param name="xml-handler-bindings" value="configuration"/> 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

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FusionPBX - Cannot queue any more events
by Glen G 28 Nov '21

28 Nov '21

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.

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Re: [Freeswitch-users] WebRTC calls one way with custom sip messages
by David P 27 Nov '21

27 Nov '21

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.

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Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI
by kaleem rehman 26 Nov '21

26 Nov '21

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 <ehtasham.malik(a)expertflow.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Ahmed Hasan <ahmad.hasan(a)expertflow.com> 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 <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik(a)expertflow.com <andreas.stuber(a)expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem(a)gmail.com> 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(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 _________________________________________________________________________ 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 On Thu, Nov 25, 2021 at 2:14 PM kaleem rehman <k4kaleem(a)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 > > > > 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 <http://www.expertflow.com/> > FB: [image: > FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> > LinkedIn: [image: linkedIn.png] /company/expertflow > <https://www.linkedin.com/company/expertflow> Youtube: [image: > YouTube-social-square_red_128px.png]/user/expertflow > <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] > /Expertflow <https://twitter.com/Expertflow> > 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark > and Google Talk: ehtasham.malik at expertflow.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > <andreas.stuber at expertflow.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>; Skype: > <http://andreas.stuber.expertflow.com/>*shani.awan3* > > > On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> 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 <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 <https://signalwire.community/> > *>>* Professional FreeSWITCH Services > *>* sales at freeswitch.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > *>* https://freeswitch.com <https://freeswitch.com/> > *>>* Official FreeSWITCH Sites > *>* https://freeswitch.com/oss <https://freeswitch.com/oss> > *>* https://freeswitch.org/confluence <https://freeswitch.org/confluence> > *>* https://cluecon.com <https://cluecon.com/> > *>>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > *>* https://freeswitch.com <https://freeswitch.com/>* > >

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WebRTC calls one way with custom sip messages UUI
by kaleem rehman 25 Nov '21

25 Nov '21

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

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originate api: Escaping variable names/values
by Shane Mitchell 23 Nov '21

23 Nov '21

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(a)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)

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mod_perl coredump
by Glen Christie 23 Nov '21

23 Nov '21

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(a)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?

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