May 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_lua
by Hossein Gholizadeh 30 May '18

30 May '18

hi ... after freeswitch version 1.4 , we don't need mod_lua?

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Mini IVR for caller inside of voicemail
by Preston Hagar 29 May '18

29 May '18

We've just finished replacing an older Mitel system with our new FreeSWITCH system. One feature of the old Mitel system we are missing that a few of our users have asked for worked like this: Person calls user's number (DID or internal). Caller gets user's voicemail. Either the users greeting could be recorded to say something like "To reach my cell phone, press 2, To reach my assistant, press 3, To leave me a message, wait for the beep". The caller could then decide to either press a key to be transferred to somewhere else, or just wait for the beep and leave a message. The user could setup their forwarding number(s) to prompt for through the voicemail menu prompts, it didn't require an admin or any other access. I know I could generally do this with an IVR setup, but it would be nice to have it like the old system where the user could add/remove the forwarding numbers at will and I wouldn't have to build out custom IVRs for multiple extensions. Does anyone have any pointers on how to achieve something like this? Is it even possible with mod_voicemail, or do I need to be looking into doing VM completely differently somehow? Thanks. Preston

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Re: [Freeswitch-users] Resuming Verto
by Michael Jerris 29 May '18

29 May '18

in jquery.verto.js: if (verto.options.sessid) { verto.sessid = verto.options.sessid; } else { verto.sessid = localStorage.getItem("verto_session_uuid") || generateGUID(); localStorage.setItem("verto_session_uuid", verto.sessid); } > On May 29, 2018, at 5:06 PM, Rick Jarvis <rick(a)magicmail.mooo.com> wrote: > > Thanks Mike. I wonder how a key/value store can report the correct state in that case? I realise this is beyond the scope of general Freeswitch stuff and more in the depths of coding, but if anyone has tackled this I'd love any pointers. I've looked through the Verto Communicator source and it didn't clue me up exactly (probably partly cos it's in Angular which I have no knowledge of).... > > > On 29/05/2018 19:59, Michael Jerris wrote: >> verto communicator stuffs the session id down in local storage for this. >> >> >>> On May 29, 2018, at 2:29 PM, Rick Jarvis <rick(a)magicmail.mooo.com> wrote: >>> >>> My Verto app currently checks: >>> >>> vertoHandle (to see if Verto has been initialised >>> currentCall.state.name (to see if a call is connected) >>> >>> The problem with this of course is that if the browser is refreshed, those variables are lost and my app is assuming it needs to reconnect (which often, due to the magic of Verto, it doesn't actually need to). >>> >>> So my question is how best to check the session (and call) states? Would, say, loading those variables into the browser's localstorage be a good method, or is there a better way of querying the current sates? >>>

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(with corrections) Can FS avoid sending re-INVITE to client? Or bridge on its behalf?
by M Yudkowsky 29 May '18

29 May '18

Comrades! (with some corrections, IN UPPER CASE, as some words went astray in editing.) I'd like to find out of if FS, which is making calls on behalf of a client, can intercept timer-based re-INVITEs and respond to them without sending the INVITE to the client first. Alternatively, can FS be made smart enough to bridge "in the cloud" on behalf a client, even if the client uses internal RTP port numbers? Details: I've got an internal server (IS) that uses FS as a firewall. The IS asks for leg A, then leg B, and then bridges leg A and leg B "in the cloud." FS (working in media bypass mode) handles this without any problems. For many non-NANP and some NANP calls, some SIP providers insist on a re-INVITE every 5 to 15 minutes. Unfortunately, the IS has a bug: if you send it a re-INVITE -- WHEN THE SIP PROVIDER SENDS A RE-INVITE TO FS, AND THEN FS SENDS IT ON TO THE IS -- the IS will respond correctly and continue the call in the cloud. But at the same time, the IS will open an internal RTP port (which is not used for anything) and then fail 60s later when no RTP reaches that port. THIS ERROR IS NOT RELATED TO FS AT ALL AND HAPPENS EVEN WHEN THE IS INTERACTS DIRECTLY WITH THE SIP PROVIDER. At this point, the IS sends a BYE for one of the legs, requests the other leg back from the cloud, and continues to handle the remaining leg. Note that the IS in the course of normal processing does break the cloud conference, talk to each individual leg, bridges them together again or each with some other new leg... it's quite a dance of INVITEs and re-INVITEs that the IS sends. But inbound re-INVITEs related to timers is the problem at the moment. Two possible solutions: Solution A. Is there a way for FS to be made smart enough and aggressive enough to act on a routine, timer-related re-INVITE? THIS WOULD MEAN FS WOULD AUTOMATICALLY RESPOND TO THE RE-INVITE AND NOT SEND IT ON TO THE IS. This would bypass the bug in our IS. Solution B. When FS is set to bypass RTP media, the bridge happens up in the cloud. In normal operating mode, without bypass or proxy of RTP set, this is what happens: When the IS attempts to bridge the calls it sends the bridging re-INVITEs to FS, and those re-INVITEs are for _internal_ ports that FS offered to the IS. FS bridges these internal ports but (and perhaps I'm reading Wireshark wrong) does not attempt to bridge the external ports and take the media into the cloud. The bridge is done on FS itself. A possible solution, then, would be to configure FS to determine that the media can be bridged in the cloud and have it do so. -- Moshe Yudkowsky Disaggregate Corporation 2952 W Fargo Chicago, IL 60645 USA +1 773 764 8727 speech(a)pobox.com http://www.Disaggregate.com http://www.PebbleAndAvalanche.com

1 0

Resuming Verto
by Rick Jarvis 29 May '18

29 May '18

My Verto app currently checks: vertoHandle (to see if Verto has been initialised currentCall.state.name (to see if a call is connected) The problem with this of course is that if the browser is refreshed, those variables are lost and my app is assuming it needs to reconnect (which often, due to the magic of Verto, it doesn't actually need to). So my question is how best to check the session (and call) states? Would, say, loading those variables into the browser's localstorage be a good method, or is there a better way of querying the current sates?

2 1

Connecting freeswitch to postgresql
by Blackhold 29 May '18

29 May '18

Hi, I have just installed freeswitch 1.9 freeswitch@freeswitch-capa8> version FreeSWITCH Version 1.9.0+git~20180516T211208Z~3c7db639fd~64bit (git 3c7db63 2018-05-16 21:12:08Z 64bit) And I'm trying to freeswitch use postgresql instead sqlitedb. I have configured the file switch.conf.xml of /usr/local/freeswitch/conf/autoload_configs with uncomenting and configuring this line <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch_db user=freeswitch_user password='****' options='-c client_min_messages=NOTICE'" /> then I exported all sqlitedbs to sql and then imported to a database that I have created in postgresql I have run again /etc/init.d/freeswitch.sh start But seems it still is taking sqlitedb. I tried to move the sqlitedbs files to other side and turned on again freeswitch, but it creates sqlitedbs again. What I'm missing? what I'm doing wrong? what could I check? Thanks you much! - Blackhold http://blackhold.nusepas.com @blackhold_ ~> cal lluitar contra el fort per deixar de ser febles, i contra nosaltres mateixos quan siguem forts (Xirinacs) <°((( ><

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Crash on 1.6 when using ICE
by 王聡 29 May '18

29 May '18

Hey all, Recently I upgraded my FS server from 1.4 to 1.6.20 and enabled video record, but frequently crash occurred. The coredump file shows the crash occurred in handle_ice() function by a memory error like this: #0 switch_sockaddr_get_port (sa=0x0) at src/switch_apr.c:924 No locals. #1 0x00007f5aa524e662 in handle_ice (rtp_session=0x7f57e01a2138, ice=0x7f57e01a0cf0, data=0x0, len=140015398758072) at src/switch_rtp.c:1231 ipbuf = “58.93.127.1", '\000' <repeats 37 times> cmp = -1521403865 stunbuf = "\001\001\000,!\022\244B\001o\025\177\236\006\262X\226Q\321$\000 \000\b\000\001:\266\033O\023`\000\b\000\024z\204%(\307>\352'\254jn\"K\255\227\376텠6\200(\000\004\366\065\062\177", '\000' <repeats 447 times> rpacket = 0x7f59defcc2d0 remote_ip = 0x1ba4 <error: Cannot access memory at address 0x1ba4> from_addr = 0x7f57e01a0cf0 bytes = 64 buf = “58.93.127.1", '\000' <repeats 67 times> port = 7076 buf2 = '\000' <repeats 79 times> packet = 0x1ba4 attr = 0x7f57e01b26b8 end_buf = 0x0 username = "76MHJVvjVqM37tvJ:b94edc3e\000\000\000\000\000\000\000\000" buf = "\001\000T\000!\022\244B\001o\025\177\236\006\262X\226Q\321$\006\000\031\000\067\066MHJVvjVqM37tvJ:b94edc3e\000\000\000$\000\004\000n\377\377\377*\200\b\000n\033\004H0:9\254\b\000\024\000K\203Ə\264\242\063\303G\367\265\061\226v\233\206F\210\001G(\200\004\000iU\316\354", '\000' <repeats 407 times> xlen = -553860400 __func__ = "handle_ice" #2 0x00007f5aa5252428 in read_rtp_packet (rtp_session=0x7f57e01a2138, bytes=0x7f59defcc740, flags=0x0, pmapP=0x0, poll_status=SWITCH_STATUS_SUCCESS, return_jb_packet=SWITCH_FALSE) at src/switch_rtp.c:5691 ts = 3759842008 sync = -535125288 now = 140027297347623 __PRETTY_FUNCTION__ = "read_rtp_packet" __func__ = "read_rtp_packet" More test shows that the crash occurred when client switched its network during a call. It always crash at the same point when I switched from wifi to carrier network. And the crash disappeared after I disabled the ICE settings in client. However, the same setting works well on previous server (1.4), so is there any changes on ICE solution between 1.4 and 1.6.20? Or the video record is the reason? Server: Old: FreeSWITCH Version 1.4.26+git~20160205T175853Z~ca9207aa32~64bit (git ca9207a 2016-02-05 17:58:53Z 64bit) New: FreeSWITCH Version 1.6.20+git~20180123T214909Z~987c9b9a2a~64bit (git 987c9b9 2018-01-23 21:49:09Z 64bit) Client: Custom SIP phone based on LibLinphone Regards.

2 1

Domains and contexts with single dialplan
by Rick Jarvis 28 May '18

28 May '18

This is something I really should understand by now, but for some reason I have always struggled with contexts vs dialplans Is it possible to share the same *dialplan*, but be ring fenced from other *domains*? So there would be a single default dialplan, but 201@domain1 calling into conference 3200 is talking to others@domain1 but no-one from @domain2? R

2 2

Freeswitch fax by loopback
by Hossein Gholizadeh 27 May '18

27 May '18

Hi i want to send fax by loopback to my self ... here is my configurations of fax in dialplan : <extension name="fax_detect"> <condition field="destination_number" expression="^2010$"> <action application="answer" /> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="spandsp_start_fax_detect" data="transfer '9978 XML default' 6"/> <action application="bridge" data="user/$1@${domain_name}"/> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="$${base_dir}/faxes/${uuid}.tif"/> <action application="hangup"/> </condition> </extension> <extension name="fax_transmit"> <condition field="destination_number" expression="^9179$"> <action application="txfax" data="$${base_dir}/txfax.tif"/> <action application="hangup"/> </condition> </extension> after configuring these i want to send a fax file by loopback in fs_cli i run this command: originate loopback/9978 (/tmp/test.tif) but i just get this error: cannot send non-existant fax file . but my fax file is there already. is my loopback sending correct ? am i right that every fax files which received will be saved in this directory ( $${base_dir}/faxes/${uuid}.tif)?

1 0

[mod_unimrcp][Speech Recognition start/stop]
by Bilal Abbasi 25 May '18

25 May '18

Hi Users, I am using mod_unimrcp, i have successfully done the Speech to Text using below function <action application="play_and_detect_speech" data="ivr/ivr-welcome_to_freeswitch.wav detect:*unimrcp:uni2* {start-input-timers=false}*builtin:speech/transcribe* "/> But i don't need the play and detect, i need to put the caller in a conference so i really need something like this action application="start_detect:*unimrcp:uni2* {start-input-timers=false} *builtin:speech/transcribe*"/> <action application="conference" data="confname@default]"> action application="stop_detect:*unimrcp:uni2* {start-input-timers=false} *builtin:speech/transcribe*"/> I could not find a way to start the unimrcp client and then stop at hangup, can somebody please help me out with this. Regards

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