October 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Incorrect example for 'originate' with socket and async full
by . 29 Oct '18

29 Oct '18

Hi, I wanted to report an error/typo in the confluence wiki related to 'originate', and its use together with the socket application. Apologies, I don't have access to the wiki, but thought this was worth passing on, so here it is: - https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_comma… The wiki example for ring_ready also includes the socket application and is written as: - originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 async full) however this hangs forever if used as written. Eventually, after several hours of hair tearing and googling, I stumbled on this thread: - https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.h… Indeed this is the problem. The correct syntax for the socket component is: - &socket('127.0.0.1:8082 async full') - Specifically single quotes around the command so that it is treated as a single string. After making this change there is no more hanging on the outbound calls and all events come through in a timely fashion. Cheers, Joe

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transfer_fallback_extension + transfer_on_fail
by Paul Muaddib 28 Oct '18

28 Oct '18

Hi, I setup the following: When someone calls an extensions the phone rings for 30 seconds. After that the phone keeps ringing but also the phones at front desk, here 10 and 11 When the user is busy (USER_BUSY) or when the extensions does not exist (NO_PICKUP) the caller will be transferred to the front desk. This is working with transfer_on_fail Now I would like to accomplish the following: When someone does a blind transfer and the callee does not answer after 30 seconds, how can I transfer the callee back to the extension that initiated the transfer? In the documentation I found the channel variable transfer_fallback_extension. But there is not explanation to it and how does this work together with transfer_on_fail? Best regards, Paul <extension name="Extension"> <condition field="destination_number" expression="^([1-9][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="transfer_on_fail=NO_PICKUP,USER_BUSY,NO_ANSWER,SUBSCRIBER_ABSENT BRIDGE_FAILED_EXTENSION xml"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bind_digit_action" data="get_digits,~^([1-9][0-9])$,exec:lua,get_digits.lua,peer,self"/> <action application="digit_action_set_realm" data="get_digits"/> <action application="bridge" data="<ignore_early_media=true>[leg_timeout=30]user/${dialed_extension}, pickup/global, pickup/${dialed_extension}"/> <action application="bridge" data="<originate_timeout=90,ignore_early_media=true> user/${dialed_extension}, user/10, user/11, pickup/global, pickup/${dialed_extension}"/> <action application="hangup"/> </condition> </extension> <extension name="BRIDGE_FAILED_EXTENSION"> <condition> <action application="log" data="INFO ######## Extension failed - ${originate_disposition} ########" /> <condition field="${originate_failed_cause}" expression="(NO_PICKUP)|(USER_BUSY)|(NO_ANSWER)" break="on-true"> <action application="transfer" data="operator"/> </condition> <condition field="${originate_failed_cause}" expression=".*" break="on-true"> <action application="transfer" data="operator"/> </condition> </condition> </extension>

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enterprise originate + originate_delay_start
by Paul Muaddib 28 Oct '18

28 Oct '18

Hi, I am trying to accomplish this bridge command but I am facing two problems. <action application="bridge" data="<originate_timeout=90,ignore_early_media=true>${group_call(werkstatt+A)},pickup/global,pickup/werkstatt:_:{originate_delay_start=25000}${group_call(buero+A)}“/> 1. When I set originate_delay_start to 25000 the group gets not called at all. If I set it to 10000, it takes about 18 seconds and not 10 seconds. 2. Even though I set ignore_early_media=true I don't hear any ringing. Best regards, Paul PS: Why is the value for originate_delay_start millisconds and for leg_delay_start it is seconds?

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Identify voice quality issues using CDR records
by sukitha jayasinghe 28 Oct '18

28 Oct '18

Dear all, I have serious voice quality issues in my freeswitch 1.6.20 installation. In CDRs I found that the voice quality percentage is 100 and mos is 4.5, but the quality is very bad for one side. I have pasted the audio part of the CDR below. What mean by the skip packet count. Is it possible to identify the issue by looking at these statistics? "audio" : { "inbound" : { "raw_bytes" : 90298, "media_bytes" : 90298, "packet_count" : 1235, "media_packet_count" : 1235, "skip_packet_count" : 1184, "jitter_packet_count" : 0, "dtmf_packet_count" : 0, "cng_packet_count" : 0, "flush_packet_count" : 0, "largest_jb_size" : 0, "jitter_min_variance" : 137.097046, "jitter_max_variance" : 315.2, "jitter_loss_rate" : 0, "jitter_burst_rate" : 0, "mean_interval" : 27.274788, "flaw_total" : 0, "quality_percentage" : 100, "mos" : 4.5 },

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Mod_spy part2
by Alex Goulis 27 Oct '18

27 Oct '18

Hi all...Now that I've managed to get mod_spy to work, I've noticed it only allows a perpetual listen and gives none of the dynamic dtmf or the ability to whisper on the eavesdrop app when it initiates it. I've set all the correct eavesdrop parameters before calling userspy, but they dont have any effect.Is there a way to get the eavesdrop to whisper or enable the dtmf functions to work using userspy?Thanks!

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mod_spy, does it only work for registered users?
by Alex Goulis 27 Oct '18

27 Oct '18

Hi all... Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?

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XML unescaping
by Sergey Safarov 27 Oct '18

27 Oct '18

Hello When need to assign variable value with XML symbols requires escaping is suggested to use dialplan contractions like example <action application="set"><![CDATA[ sip_h_Call-Info=<urn:nena:uid:callid:72270801514820004831: lng.001.experient.co.us>;purpose=nena-CallId]]></action> But more elegant will be <action application="set" data="sip_h_Call-Info=<urn:nena:uid:callid:72270801514820004831: lng.001.experient.co.us>;purpose=nena-CallId"/> I want add this functionality to FS. Could you give me reference to FS code where is used escaping/unescaping. Sergey

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Prevent FS from changing # to %23 in the SIP URI
by Lyubo Popov 26 Oct '18

26 Oct '18

Hello all, I am facing a problem where FS will convert a "#" character to "% 23" (no spaces) in the SIP URI. The problem is that the call will be recorded like that in CDRs or send in RADIUS packets ( I am using mod_xml_radius) and my billing will have problems accounting the call as it will expect # and not "% 23". Is there a way to tel the switch to not encode ( changes) such SIP URIs? Thank you for your help! Re. Koki

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Playing file into videoconference via mod_av freezes after server reboot
by Chad Phillips 25 Oct '18

25 Oct '18

FreeSWITCH 1.8.2 on latest Debian Stretch, compiled stock from source, deps installed via FS repo. I have FreeSWITCH managed by systemd to start at boot, which works well. However, I'm hitting a weird issue where, after initial system boot, if I try to play a video file into a conference via mod_av, the video is frozen on the first frame (audio is fine). If I restart FreeSWITCH, then the video file plays fine. Any thoughts on what might be causing this? My only guess: on boot systemd starts FreeSWITCH before some other service or setup that is needed to play video files via mod_av, and restarting FreeSWITCH fixes the issue because said service is now running.

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Re: [Freeswitch-users] [username password with @][http cache]
by Bilal Abbasi 25 Oct '18

25 Oct '18

Yeah that did helped, i was just wondering can i use ftp URL in mod http cache? FTP URL seems to be not working in my case, i was looking for any best solution to download ftp sound files. Regards Abbasi On Wed, 24 Oct 2018 at 4:08 PM, António Silva via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "António Silva" <asilva(a)wirelessmundi.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Wed, 24 Oct 2018 11:21:23 +0200 > Subject: Re: [Freeswitch-users] [username password with @][http cache] > > you could try url encode de @, to %40 > > Example: > > username:password%40@url > > > On 24/10/2018 09:27, Bilal Abbasi wrote: > > Hey users, > I have to download a sound file from FTP server having password with @ in > it. > The mod_http_cache has the way username:password@url, i was looking to > escape the @ from it. > This might be very basic question to ask, may be some one has done that > already. > > Regards > Abbasi > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales@freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://clu… > > FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > -- > Saludos / Regards / Cumprimentos > António Silva > > > > > ---------- Forwarded message ---------- > From: "António Silva via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Wed, 24 Oct 2018 04:08:06 -0700 (PDT) > Subject: Re: [Freeswitch-users] [username password with @][http cache] > _________________________________________________________________________ > 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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