- FreeSWITCH-users - FreeSWITCH Mailing Lists

fail2ban assistance
by clive engelberg 28 Nov '17

28 Nov '17

Hi. I am experiencing this strange behavior when using Zoiper android client where it sends register attempts with a username "yes". I suspect this is to keep open the NAT. Anyway, fail2ban does not like this, as you can imagine, and it bans the user. Can someone help me to add an ignoreregex for fail2ban please. The strings that are causing it to fail are: 2017-11-14 17:19:41.716109 [WARNING] sofia_reg.c:1775 SIP auth challenge (REGISTER) on sofia profile 'default' for [yes(a)196.22.1.22] from ip 105.184.98.99 2017-11-14 17:19:41.756109 [WARNING] sofia_reg.c:2889 Can't find user [yes(a)196.22.1.22] from 105.184.98.99 maybe this would work... ignoreregex= \[WARNING\] sofia_reg.c:\d+ SIP auth challenge $REGISTER$ on sofia profile \'[^']+\' for \[yes@.*\] from ip <HOST> my regex skills are lacking... Thanks in advance Clive

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Fork a call on registering
by Danyi Lin 28 Nov '17

28 Nov '17

How can I fork a call to a new registered endpoint? Here is the scenario: Callee has 2 devices: dev1 online (registered), dev2 offline 1. An incoming call to the user, dev1 is ringing 2. dev2 register now 3. fork call to dev2 Call flow as below ( https://gist.github.com/dylin/5ef157f2bc4da8b1c0b145c03e9efcfd) Caller | FS | dev1 | dev2 ----------- ----------- ----------- ----------- | INVITE | | | |------------>| | | | | | | | | INVITE | | | |----------->| | | | | | | | | REGISTER | | |<-----------+------------| | | | | | | INVITE | | | |------------+----------->| | | | | I tried freeswitch.Session("sofia/to/dev2", lega) to originate call to dev2, and dev2 got call. However if dev1 answered call, it won't cancel INVITE to dev2 as a normal fork. Any suggestions are welcome. Best regards, Danyi Lin

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SLA not working across multiple FS profiles
by sagar malam 28 Nov '17

28 Nov '17

Hello, We are trying to implement SLA for users registered in different profiles We have setup of freeswitch as following : - There are two profiles for registration of users named *register1* (listening on 5070 on ipv4) and *register2* (listening on 5071 on ipv6). - One user for e.g.* 1001(a)example.com <1001(a)example.com>* is registered on register1 profile and register2 profile as well. - We have followed below link for configurations : https://wiki.freeswitch.org/wiki/Shared_Line_Appearance - For Shared Line appearance we have following dialplan : <extension name = Demo> <condition field="${destination_number}" expression="(.*)"> <action application="set" data="hangup_after_bridge=true"/> * <action application="export" data="nolocal:sip_invite_domain=example.com <http://example.com/>"/>* <action application="set" data="continue_on_fail=true"/> *<action application="export" data="nolocal:presence_id=1002@example.com <1002(a)example.com>"/>* <action application="bridge" data="${sofia_contact( 1001(a)example.com)}"/ </condition> </extension> Now one user for e.g. 1002(a)example.com is dialing extension 1001, presence is shown on both registered 1001 extensions, call is answered from the extension registered on register1 profile but SLA is not working for 1001 registered from profile register2, same issue arises for vice versa scenario as well. So, If you can please elaborate that whether this scenario should work or we are missing something or this is default functionality of freeswitch. Thank you in advance.

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Maximum Number of Extensions in XML
by Alexander Perkins 27 Nov '17

27 Nov '17

Hi. Does anybody know the maximum number of XML files I can have for extensions? Or the number of extensions per XML file? I ask because I once tried loading an extensions file in asterisk with around 25K extensions and it took around 20 minutes. Just curious. Thanks, Alex

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Re: [Freeswitch-users] Fork a call on registering
by itadebayo 27 Nov '17

27 Nov '17

Use comma, pipe or enterprise operator to send call to multiple end points in parallel, sequential or enterprise mode. Ismail Sent from my Samsung Galaxy smartphone. -------- Original message --------From: Danyi Lin <danyi.lin(a)gmail.com> Date: 27/11/2017 7:11 p.m. (GMT+01:00) To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Fork a call on registering How can I fork a call to a new registered endpoint? Here is the scenario: Callee has 2 devices: dev1 online (registered), dev2 offline 1. An incoming call to the user, dev1 is ringing 2. dev2 register now 3. fork call to dev2 Call flow as below (https://gist.github.com/dylin/5ef157f2bc4da8b1c0b145c03e9efcfd) Caller | FS | dev1 | dev2 ----------- ----------- ----------- ----------- | INVITE | | | |------------>| | | | | | | | | INVITE | | | |----------->| | | | | | | | | REGISTER | | |<-----------+------------| | | | | | | INVITE | | | |------------+----------->| | | | | I tried freeswitch.Session("sofia/to/dev2", lega) to originate call to dev2,and dev2 got call. However if dev1 answered call, it won't cancel INVITEto dev2 as a normal fork. Any suggestions are welcome. Best regards, Danyi Lin

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Compiling ESL as a DLL
by Alex Pierry 27 Nov '17

27 Nov '17

Hi guys, could anybody give me a step by step of how to compile the ESL project to produce a DLL rather than a LIB? I'm using visual studio community 2017 with the head of version 1.7 of Freeswitch. Compiling as a LIB (project default) is working fine, but I need use it with ManagedEsl.dll wrapper inside a C# Project. Thank you in advance. Alex Pierry <http://pt.slideshare.net/marketingteclan> <http://blog.teclan.com.br/>

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verto output device ring is always the default audio
by António Silva 27 Nov '17

27 Nov '17

Hi, I'm use dynamic media tags with verto my problem is that when i set a specific device for output the ring is always played on the default device, during the call switching to different outputs is done perfectly. I've try forcing in verto.js: case $.verto.enum.state.ringing: var speaker = dialog.useSpeak; console.info("Using Speaker: ", speaker); if (speaker && speaker !== "any" && speaker !== "none") { setTimeout(function() { dialog.setAudioPlaybackDevice(speaker); }, 500); } break; I do see the message changing it, but doesn't work... any idea what i'm doing wrong? Log changing the device when ringing: jquery.verto.js:2214 Using Speaker: e6208da870143078295d9e309123f2d53ddd5025ea44c915e2133ab7d1aaff36 jquery.verto.js:2160 Dialog: 134c1362-77af-4f53-9aa8-b4a98655d779 Setting speaker: <video id="incoming-media-element1511478310478" autoplay="autoplay" style="width: 640px; height: 480px; object-fit:contain;"></video> Built-in Audio Digital Stereo (HDMI 2) Thanks. -- Saludos / Regards / Cumprimentos António Silva

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txfax unexpected message received
by António Silva 27 Nov '17

27 Nov '17

Hi, when sending a fax to a particular destination it fails with "Fax processing not successful - result (13) Unexpected message received." To other destination is working ok, i don't know what is the particularity on this one that makes it fail... From the logs i see: 2017-11-24 18:30:05.625309 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Rx: CFR with final frame tag 2017-11-24 18:30:05.625309 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Rx: ff 13 84 2017-11-24 18:30:05.625309 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Rx final frame in state D 2017-11-24 18:30:05.625309 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Unexpected CFR frame in state D I guess that is not suppose to receive confirmed to received.. because the training is not yet done. Can i do anything from my side to make it work? I've tried with different resolutions but no luck.. i try with t38 but this gw the t38 is not well implemented and also didn't work. To send the fax i just do: originate {fax_verbose=true}sofia/gateway/CS2K/939165135 &txfax(/root/txfax.tiff) Attach full log. Thanks for the help. -- Saludos / Regards / Cumprimentos António Silva

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mod_skypopen no longer functional?
by Markus 24 Nov '17

24 Nov '17

Dear list, I'm new here. And I'm looking for help (surprise!). In 2015 the only available SIP-to-Skype gateway that was left was mod_skypopen for FreeSWITCH. I had it running just fine until 12th/13th November 2017, based on the instructions provided at https://freeswitch.org/confluence/display/FREESWITCH/mod_skypopen, under 1.4.15+git~20141229T185951Z~507a0f22c5~64bit, then it simply stopped working all of a sudden. Stopped working meaning: It seems the Skype clients are "flapping", sometimes one (out of 3) is running, sometimes more, but something is wrong with them. FreeSWITCH console says the following when a call comes in: "2017-11-24 16:39:27.219382 [INFO] mod_dialplan_xml.c:635 Processing 49myCLI <49myCLI>->my_skype_user in context my_context 2017-11-24 16:39:27.219382 [ERR] switch_core_session.c:510 Could not locate channel type skypopen" I'm somehow guessing Skype has changed something on their end? But that is just a wild guess. I am totally clueless on how to debug this thing. I would like to get a look at the Skype client(s) that is/are running on the machine, but I simply do not know how. Any help? This is being used in a commercial project, so I'm ready to pay for your time. Please contact me offlist if you are providing paid help. I tried to contact the author G. M. directly, but there was no answer. Help! :) Thank you! Markus

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video recorded as "video muted"?
by Mundkowsky, Robert 24 Nov '17

24 Nov '17

Hi, We are using Verto to record audio and video for a conference. The users do not have options displayed for muting anything, but we are seeing some recordings where the audio is fine, but the video is not there or rather there is an image showing "Freeswitch Video Muted (audio not affected)". I looked thru the logs for one call that had this issue and I do not see any problems. I was suspecting that the video codec sent by the user could not be transcoded, but it seems like it was VP8 and logs seem to show this codec was successfully negotiated. Any suggestions? Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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