July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread
by Mohd Kamal Bin Mustafa 25 Sep '17

25 Sep '17

I have this python code:- import time import json import logging import threading from freeswitchESL import ESL class EventThread(threading.Thread): def __init__(self, target, args): super(EventThread, self).__init__() self.target = target self.args = args def run(self): print("Running target") self.target(*self.args) def on_channel_park(e): uuid = e.getHeader('Unique-ID') print("Ringing") conn.execute('ring_ready ', '', uuid) conn.execute('sleep', '5500', uuid) conn.execute('answer', '', uuid) print("answered, playing media ...") media_url = 'http://www.noiseaddicts.com/samples_1w72b820/17.mp3' resp = conn.execute("playback", media_url, uuid) print(resp.getHeader("Reply-Text")) print("playback done") print(resp) if __name__ == '__main__': conn = ESL.ESLconnection('127.0.0.1', '8021', 'mypass') conn.events('plain', 'all') while True: e = conn.recvEvent() uuid = e.getHeader('Unique-ID') events_message = json.loads(e.serialize('json')) event_name = events_message['Event-Name'].lower() method_name = 'on_%s' % event_name print(method_name) if event_name == 'channel_park': print("spawning thread") t = EventThread(on_channel_park, args=(e,)) t.start() if event_name == 'playback_stop': conn.execute("hangup", "", uuid) print("Hanging up") Above, the code will stuck at conn.execute('ring_ready ', '', uuid) in on_channel_park(). It never went pass through until I hangup the call. In freeswitch log, it stopped at executing park(). But if I moved ring_ready, sleep, answer outside of the thread it will work, i.e I can hear a playback:- def on_channel_park(e): uuid = e.getHeader('Unique-ID') print("answered, playing media ...") media_url = 'http://www.noiseaddicts.com/samples_1w72b820/17.mp3' resp = conn.execute("playback", media_url, uuid) print(resp.getHeader("Reply-Text")) print("playback done") print(resp) if __name__ == '__main__': conn = ESL.ESLconnection('127.0.0.1', '8021', 'mypass') conn.events('plain', 'all') while True: e = conn.recvEvent() uuid = e.getHeader('Unique-ID') events_message = json.loads(e.serialize('json')) event_name = events_message['Event-Name'].lower() method_name = 'on_%s' % event_name print(method_name) if event_name == 'channel_park': print("Ringing") conn.execute('ring_ready ', '', uuid) conn.execute('sleep', '5500', uuid) conn.execute('answer', '', uuid) print("spawning thread") t = EventThread(on_channel_park, args=(e,)) t.start() Any idea why this happen ?

3 2

gsmopen one-way voice issue
by Jeff Chua 24 Sep '17

24 Sep '17

I'm configured freeswitch with gsmopen but can't seem to figure where why I only get voice out from SIP-client to mobile-phone but not voice from mobile to SIP client. When I tried to call from mobile-phone to IVR extension 5000 and press 2 for the standard echo test (9196), I could hear the echo back. When I tried to call 9195 for the five second delay echo test, I couldn't anything. I've add these settings in vars.xml as well. <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/> <X-PRE-PROCESS cmd="set" data="inbound-late-negotiation=false"/> <X-PRE-PROCESS cmd="set" data="inbound-zrtp-passthru=false"/> <X-PRE-PROCESS cmd="set" data="disable-transcoding=false"/> <X-PRE-PROCESS cmd="set" data="inherit_codec=false"/> <X-PRE-PROCESS cmd="set" data="bypass_media=false"/> <X-PRE-PROCESS cmd="set" data="proxy_media=false"/> What am I missing? Thanks, Jeff

4 13

Bridge - user not registered - verto
by Gregor Nanger 22 Sep '17

22 Sep '17

When I make bridge to user/extension to verto user, I get user is not registered. Call is transfered, but it looks like freeswitch tries to transfer to verto and sip endpoint. And since sip endpoint is not registered, I get User_not-registered. Everything works, but I get 2 calllogs because of this. I also try to modify dial-string in user params, but same behavior Is this by design? Best regards, Gregor -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si

3 11

FreeSWITCH on Raspberry board
by youssef elouam 20 Sep '17

20 Sep '17

hi all ; I am trying install freeSWITCH in raspberry board (Raspbian OS). but when I run "make && make install" I get as result : make: *** No targets. Stop. could you please help fix this issue. thanks in advance Bestregards --------------------------Youssef ELOUAM.

13 23

Greenswitch ESL for Python
by Michael Avers 08 Sep '17

08 Sep '17

Hello, I'm looking at greenswitch as an alternative Python inbound ESL connector. Other than Italo's company (thanks for releasing this BTW), anyone else has seen success using it in a moderate to heavy call environment? Project is at https://github.com/EvoluxBR/greenswitch Thanks Mike

3 3

Dockerfile for FreeSWITCH
by Brian West 18 Aug '17

18 Aug '17

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docker If anyone wishes to submit pull requests to improve these please do so at freeswitch.org/jira I would like to work with the community to improve and document the use cases correct. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

7 15

Forward fails with deflect and REFER to null
by Mario G 12 Aug '17

12 Aug '17

What are the bold deflect and refer to lines below telling me? This worked until last year, not sure if it’s related to 1.6, I can’t nail down a start date. A ring group rings. When one phone’s FWD button is pressed prior to a call or during a call the call is forwarded to VM extension 2921, and the other phones stopped ringing. This no longer works, the set phone does not ring, the call is not transferred to VM and the other phones keep ringing. I couldn’t find anything on the wiki (it only described using codes to turn FWD on/off), etc. to explain why this stopped working. Any help to get the phone forwarding working again is greatly appreciated. Thanks! Mario G YEALINK T48G DSS button is set to forward to 2921, pressing again turns it off. All other DSS keys such as call park, etc work fine. <extension name="vmonly"> <condition field="destination_number" expression="^2921$">  <action application="lua" data="/usr/local/FreeSwitch-scripts/known-people.lua CALLIN"/> <action application="answer"/> <action application="playback" data="${lua_vmmsg}"/>  <action application="set" data="skip_greeting=true"/> <action application="voicemail" data="default ${domain_name} 100"/> </condition> </extension> This log shows the only lines that contain 2921 in bold: 2016-11-19 16:27:26.673609 [INFO] mod_dialplan_xml.c:637 Processing Mario G <15051234567>->2921 in context public Dialplan: sofia/internal/15051234567(a)11.123.1.10 parsing [public->unloop] continue=false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Regex (PASS) [unloop] ${sip_looped_call}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Action deflect(${destination_number}) 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/15051234567(a)11.123.1.10) State Change CS_ROUTING -> CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/15051234567(a)11.123.1.10) State ROUTING going to sleep 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15051234567(a)11.123.1.10) Running State Change CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/15051234567(a)11.123.1.10) State EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] mod_sofia.c:198 sofia/internal/15051234567(a)11.123.1.10 SOFIA EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:328 sofia/internal/15051234567(a)11.123.1.10 Standard EXECUTE EXECUTE sofia/internal/15051234567(a)11.123.1.10 deflect(2921) 2016-11-19 16:27:26.673609 [DEBUG] sofia.c:8349 Process REFER to [(null)@2921]

3 11

FS AWS does not call between mobile phones connected to the same WiFi
by Jose David Jurado Alonso 07 Aug '17

07 Aug '17

Hi, I have an FS server running on an AWS instance and it does not call between mobiles connected to the same WiFi. The call is successful if each of the mobiles is connected to different Wifi or by mobile network (4G). Obviously FS is always in a different network and I use a domain/URL to routing the calls. I also do not see anything strange in the logs. I think that call does not occur because the server does not find the destination number as registered and hangup/abandoned the call by timeout. It always works except when the two mobiles are connected to the same wifi. My current configuration is the next: vars.xml: <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomain.us"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:mydomain.us"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> internal.xml: <param name="apply-nat-acl" value="rfc1918.auto"/> <param name="local-network-acl" value="rfc1918.auto"/> <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/> <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/> internal-ipv6.xml:    external.xml: <param name="local-network-acl" value="rfc1918.auto"/> <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/> <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/> external-ipv6.xml: <param name="local-network-acl" value="rfc1918.auto"/>    One call example is "sip:1009@mydomain.us:5080"

3 4

TURN Configuration under CGN (RFC6598)
by Nandy Dagondon 05 Aug '17

05 Aug '17

Hi folks, I just learned that our Internet provider implemented CGN when I'm about to experiment WebRTC/mod_verto. I already signed up a free TURN server account at http://numb.viagenie.ca/ to proceed. Where do I place turn: numb.viagenie.ca in the config files? Is this correct? sip_profiles/external.xml: <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/> /Nandy

4 19

SIM7100 LTE modem with USB audio
by Stanislav Sinyagin 05 Aug '17

05 Aug '17

Simcom has recently released a new 4G/LTE modem, and it has USB audio support. You can find sim7100_usb_audio_application_note_v0.01.pdf with details at the vendor site, or at techship.com after registration. It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation. So, in theory, gsmopen module may be adapted to it (or maybe a new module is worth starting). I ordered a sample, will check it out soon. cheers, stanislav

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