May 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Auto start freeswitch license server
by Abhilash Rathod 10 May '19

10 May '19

Hello Team, I have purchased License for the Mod_com_G729 and transcoding works fine. But I want to know can I start this license server automatically when any transcoding command run. the command is like below. $ /usr/local/freeswitch/bin/fs_encode -l mod_com_g729 tmp_dg_en.wav.wav tmp_dg_en.wav.G729 Advance thanks for the help. -- *Thanks & Regards* *Abhilash Rathod* [image: Ecosmob Technologies Pvt. Ltd.] <https://htmlsig.com/t/000001C50ZHN> Ecosmob Technologies Pvt. Ltd. https://www.ecosmob.com -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

1 0

Re: [Freeswitch-users] redirect + swapping of ports
by Vrajendra VT 09 May '19

09 May '19

+ freeswitch-users(a)lists.freeswitch.org On Thursday, May 9, 2019, 8:51:16 AM GMT+5:30, Vrajendra VT <vrajendra.vt(a)yahoo.com> wrote: Thank you Jose for the reply. The reason for doing this:Lets just say FS2 will be replaced with N no of FSs in future So, FS1 will redirect to FS2 .... FSN, depending upon which is free (whose load is less).As of now, will take FS1 and FS2 as an example. Since I have swapped the ports. the incoming call (on 5060) is handled by external profile (rather than internal) since I have swapped the ports in vars.xml. Since I am only redirecting from FS1, I have changed only public.xml something like this on FS1 <extension name="public_extensions" continue="true"> <condition field="destination_number" expression="^(\d+)$"> <action application="redirect" data="sip:1011@[IP of FS2]"/> </condition> </extension> After this, I am seeing the following in the logs, 2019-05-09 03:05:29.267994 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 302 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:60 sofia/external/+16617480240@[IP from SIP PROVIDER] Standard HANGUP, cause: NORMAL_UNSPECIFIED 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:852 (sofia/external/+16617480240@[IP from SIP PROVIDER]) State HANGUP going to sleep 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:619 (sofia/external/+16617480240@[IP from SIP PROVIDER]) State Change CS_HANGUP -> CS_REPORTING 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+16617480240@[IP from SIP PROVIDER]) Running State Change CS_REPORTING (Cur 2 Tot 7538) 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:938 (sofia/external/+16617480240@[IP from SIP PROVIDER]) State REPORTING 2019-05-09 03:05:29.428022 [ERR] mod_cdr_csv.c:156 Error opening /var/log/freeswitch/cdr-csv/Master.csv 2019-05-09 03:05:29.428022 [DEBUG] switch_core_state_machine.c:174 sofia/external/16617480240@[IP from SIP PROVIDER] Standard REPORTING, cause: NORMAL_UNSPECIFIED 2019-05-09 03:05:29.428022 [DEBUG] switch_core_state_machine.c:938 (sofia/external/16617480240@[IP from SIP PROVIDER]) State REPORTING going to sleep 2019-05-09 03:05:29.428022 [DEBUG] switch_core_state_machine.c:610 (sofia/external/16617480240@[IP from SIP PROVIDER]) State Change CS_REPORTING -> CS_DESTROY 2019-05-09 03:05:29.428022 [DEBUG] switch_core_session.c:1665 Session 7537 (sofia/external/16617480240@[IP from SIP PROVIDER]) Locked, Waiting on external entities 2019-05-09 03:05:29.428022 [NOTICE] switch_core_session.c:1683 Session 7537 (sofia/external/16617480240@[IP from SIP PROVIDER]) Ended 2019-05-09 03:05:29.428022 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/16617480240@[IP from SIP PROVIDER] [CS_DESTROY] One thing I observed, "+" missing in source number for second half of logs. After corresponding redirect, I am not seeing any logs on FS2.Could there be issue in FS2 configs? Thanks. On Thursday, May 9, 2019, 12:22:44 AM GMT+5:30, Jose Figueroa <josen.figueroa(a)unixmexico.org> wrote: Hello Vrajendra, There are several ways to achieve this. The question is what is the reason you're doing this? It would be good to understand the scenario.In VoIP, you can use prefixes to redirect incoming calls or outgoing calls to specific gateways or even use sip proxy strategies. In addition, if you share your config here I am pretty sure someone from the mailing list can help you. ThanksJose N Figueroa On Wed, May 8, 2019 at 12:40 PM Vrajendra VT via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: ---------- Forwarded message ---------- From: Vrajendra VT <vrajendra.vt(a)yahoo.com> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Wed, 8 May 2019 14:19:04 +0000 (UTC) Subject: redirect + swapping of ports Hi, I am a novice in Freeswitch. I have a provider sending me calls on 5060 port only. I have 2 FS's. Calls are landing on FS1. From FS1, I need to redirect to FS2 (lets assume all calls to be redirected) With normal config's (I had done few config changes in default and public), I was able to get some logs albeit (Proxy Authentication error). This may be because by default internal calls are authenticated. Thats why I swapped ports 5060 and 5080 of internal and external profiles. But I am getting 480 error Any help is appreciated. Thanks. Thanks. ---------- Forwarded message ---------- From: Vrajendra VT via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Wed, 08 May 2019 10:40:28 -0700 (PDT) Subject: [Freeswitch-users] redirect + swapping of ports _________________________________________________________________________ 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

1 0

Re: [Freeswitch-users] redirect + swapping of ports
by Jose Figueroa 08 May '19

08 May '19

Hello Vrajendra, There are several ways to achieve this. The question is what is the reason you're doing this? It would be good to understand the scenario. In VoIP, you can use prefixes to redirect incoming calls or outgoing calls to specific gateways or even use sip proxy strategies. In addition, if you share your config here I am pretty sure someone from the mailing list can help you. Thanks Jose N Figueroa On Wed, May 8, 2019 at 12:40 PM Vrajendra VT via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Vrajendra VT <vrajendra.vt(a)yahoo.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 8 May 2019 14:19:04 +0000 (UTC) > Subject: redirect + swapping of ports > Hi, > > I am a novice in Freeswitch. > > I have a provider sending me calls on 5060 port only. > I have 2 FS's. Calls are landing on FS1. From FS1, I need to redirect to > FS2 > (lets assume all calls to be redirected) > With normal config's (I had done few config changes in default and public), > I was able to get some logs albeit (Proxy Authentication error). This may > be > because by default internal calls are authenticated. > > Thats why I swapped ports 5060 and 5080 of internal and external profiles. > But I am getting 480 error > > Any help is appreciated. Thanks. > > > Thanks. > > > > ---------- Forwarded message ---------- > From: Vrajendra VT via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 08 May 2019 10:40:28 -0700 (PDT) > Subject: [Freeswitch-users] redirect + swapping of ports > _________________________________________________________________________ > > 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

1 0

About Call Transfer
by 王聡 08 May '19

08 May '19

Hey all, I’m trying to add a new function to my callcenter server which allow peers transfer their call to other operator. I added the following configure into my dialplan: <action application="bind_meta_app" data="4 b s execute_extension:att_xfer XML features”/> And features.xml showed like: <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="3 6 'tone_stream://%(10000,0,350,440)' digits 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="user/AA-${digits}@$${domain}"/> </condition> </extension> This should works when *4 was pressed and att_xfer activated, however when I test this dialplan, freeswitch showed the log as: 2019-05-08 18:31:46.546807 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF *:1280 2019-05-08 18:31:46.546807 [INFO] switch_channel.c:515 RECV DTMF *:1280 2019-05-08 18:31:46.986810 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 4:1280 2019-05-08 18:31:46.986810 [INFO] switch_channel.c:515 RECV DTMF 4:1280 2019-05-08 18:31:46.986810 [DEBUG] switch_ivr_async.c:4233 sofia/internal/anonymous(a)192.168.121.11 Processing meta digit '4' [execute_extension:att_xfer XML features] 2019-05-08 18:31:47.226816 [DEBUG] switch_ivr.c:625 sofia/internal/anonymous(a)192.168.121.11 Command Execute playback(local_stream://moh/8000) EXECUTE sofia/internal/anonymous(a)192.168.121.11 playback(local_stream://moh/8000) 2019-05-08 18:31:47.226816 [DEBUG] mod_local_stream.c:907 Opening Stream [moh/8000] 8000hz 2019-05-08 18:31:47.226816 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms 2019-05-08 18:31:47.226816 [DEBUG] switch_ivr.c:625 sofia/internal/BB-000004@192.168.121.3:54129 Command Execute playback(execute_extension:att_xfer XML features) EXECUTE sofia/internal/BB-000004@192.168.121.3:54129 playback(execute_extension:att_xfer XML features) 2019-05-08 18:31:47.226816 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/execute_extension:att_xfer XML features.PCMU 2019-05-08 18:31:47.226816 [ERR] mod_sndfile.c:204 Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/execute_extension:att_xfer XML features.wav] [System error : No such file or directory.] 2019-05-08 18:31:47.246814 [DEBUG] switch_ivr_play_say.c:1941 done playing file local_stream://moh/8000 The anonymous is my test phone and BB-000004 is operator. I pressed *4 on operator sip phone and got the error above. Freeswitch seems to execute the playback on bind_meta_app. Is there any advice on this problem? System info: FreeSWITCH: FreeSWITCH Version 1.8.5+git~20190124T225359Z~31281a0bf1~64bit (git 31281a0 2019-01-24 22:53:59Z 64bit) OS: Debian Stretch 9.8 Client: Bria 5 Regards. C.Wang

2 1

redirect + swapping of ports
by Vrajendra VT 08 May '19

08 May '19

Hi, I am a novice in Freeswitch. I have a provider sending me calls on 5060 port only. I have 2 FS's. Calls are landing on FS1. From FS1, I need to redirect to FS2 (lets assume all calls to be redirected) With normal config's (I had done few config changes in default and public), I was able to get some logs albeit (Proxy Authentication error). This may be because by default internal calls are authenticated. Thats why I swapped ports 5060 and 5080 of internal and external profiles. But I am getting 480 error Any help is appreciated. Thanks. Thanks.

1 0

cannot buy G729 licences: site is not responding
by António Silva 08 May '19

08 May '19

Hi, Is not possible to buy G.729 licences the site is not responding: https://endor.freeswitch.com/cart.php?a=add&pid=33 I follow the link from https://freeswitch.com/index.php/modules-more/ -- Saludos / Regards / Cumprimentos António Silva

1 0

Welcome text for participants joining conference??
by Ramesh Kandasamy 07 May '19

07 May '19

Hi all, Is there a way in freeswitch, we can display a welcome text for participants joining the video conference? Also, is there a way to prompt with a customized text before the end of conference? Thanks Ramesh

1 1

ESL connection lost to FS
by Neil Patel 07 May '19

07 May '19

Hi all, I have a fresh server running FS 1.8 from package on Debian 9. It runs a python app connecting to FS using ESL that executes this code per call attempt: c = ESLconnection(str(settings.ESL_HOSTNAME), str(settings.ESL_PORT), str(settings.ESL_PASSWORD)) if not c.connected(): abort() dial_command = "originate …" ev = c.api("bgapi", str(dial_command)) c.disconnect() if ev: result = ev.serialize() logger.debug('API result = {}'.format(result)) When we ran 350K call test, 70K calls got to this code and failed the final if condition, meaning ev was None. No CDR generated. This is totally an unexpected error case for us after tens of millions of calls over last 3 years on our other servers. Any idea why this is happening? The only difference I can see is we’ve not seen this issue on FS 1.6 (built from source), so maybe it’s introduced in 1.8 (run from package)? FYI we've built pyESL from source, which is what's recommended here. Any chance there is an incompatibility between pyESL built from source and FS 1.8 installed from package? Thanks, Neil This question reproduced from Slack <https://signalwire-community.slack.com/archives/CDC8H14PN/p1555508784091900> where I didn't get a response

3 5

RTP Packet Loss
by 王聡 07 May '19

07 May '19

Hey all, I established a freeswitch server on a cloud server for video chat based on master branch(1.9.0), and got a problem recently. Last week I found the video was lagged in several hours, and log showed: Correcting calculated ptime value from [1660] to [20] to compensate for [82] lost packet(s) I supposed it was a network trouble, but network monitor showed no huge traffic(no more than 100Mbps), but the inbound traffic was larger than outbound, This problem was gone after I executed the VMotion for this cloud server(without storage, network interface not changed). Is there any suggestions for this problem? Regards. C.Wang

1 0

Gateways and DNS and TTL
by David Villasmil 06 May '19

06 May '19

Hello all, When creating a new gateway with a FQDN instead of an IP Address, does freeswitch make a DNS Query on every call or is there a cache or does it follow the TTL? I'm on Debian 9, installed FS from freeswitch's repo. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0