- FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod Verto - Chat Version 1.6.19
by Jonathan Hunter 14 Feb '18

14 Feb '18

Hi Guys, I am testing verto on version 1.6.19 and it works great for Video and Audio, however when using the function setupChat I am seeing some strange behaviour. I am using extension 1002 and I connect to conference conference_1. I then send a chat message (to test functionality) and it sends it fine, however in the chat history window the same message is repeated, I assume by the server, and in the console I see; 2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456 [{ "jsonrpc": "2.0", "id": 3450, "method": "verto.info", "params": { "msg": { "from": "1002(a)172.31.12.49", "to": "1002(a)172.31.12.49", "body": "hello", "from_user": "1002", "from_host": "172.31.12.49", "from_full": "1002", "from_msg_name": "Jon H", "from_msg_number": "Consumer123456" } } }] 2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456 [{ "jsonrpc": "2.0", "id": 3451, "method": "verto.info", "params": { "msg": { "from": "conference_1(a)172.31.12.49", "to": "1002(a)172.31.12.49", "body": "Hello, you said: ", "to_user": "1002", "to_host": "172.31.12.49", "to_full": "1002", "to_msg_name": "Jon H", "to_msg_number": "Consumer123456" } } }] 2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456 [{ "jsonrpc": "2.0", "id": 3452, "method": "verto.info", "params": { "msg": { "from": "1002(a)172.31.12.49", "to": "1002(a)172.31.12.49", "body": "Hello, you said: ", "to_user": "1002", "to_host": "172.31.12.49", "to_full": "1002", "to_msg_name": "JonH", "to_msg_number": "Consumer123456" } } }] Where are these messages coming with "Hello, you said:" from? Is this a configuration issue on my part? Many thanks! Jon

1 0

Calls recording issue
by Igor Olhovskiy 14 Feb '18

14 Feb '18

Hi! I’ve faced an issue with call recording. I’m recording with uuid_record in wav (also tried with mp3) with these legs A: WebRTC/SIPJS/Opus@8000 B: SIP/PCMA In this case I get clear recording only on leg B. Leg A is sounds like tape on SLOW (really SLOW) playing. Mostly from leg A i hear smth like (A-R-R….!) In a case if leg A is SIP with any codec, it’s ok. FS 1.6.20 Could it be a bug in FS? Regards, Igor

3 2

"received=" in path header
by Tihomir Culjaga 14 Feb '18

14 Feb '18

hi, i have UA > Kamailio > Freeswitch UA is using sip for websockets and registers via kamailio to freeswitch. When i try to originate a call from FS to the registered endpoint, the call fails. freeswitch@FS01> bgapi originate {origination_caller_id_number=1002}sofia/internal/1001%mydomain &echo() I think FS is trying to use transport from received parematar instead of path uri. any advice how to handle this ? recv 964 bytes from udp/[192.168.50.60]:5060 at 22:18:57.909700: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKe929.b6aa43dd1eefa9e4756af7d31d65066e.0 Via: SIP/2.0/WSS 192.0.2.110;rport=61744;received=192.168.200.77;branch=z9hG4bK3179897 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 192 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="22c52b4b-f795-4caa-bb9b-15bbd87564f7", uri="sip:192.168.50.60", response="a073277bc9c1fcd681de661cb418d838", qop=auth, cnonce="euqeokf2d1gr", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=sip:192.168.200.77:61744 %3Btransport%3Dws> ------------------------------------------------------------------------ send 738 bytes to udp/[192.168.50.60]:5060 at 22:18:57.950968: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKe929.b6aa43dd1eefa9e4756af7d31d65066e.0 Via: SIP/2.0/WSS 192.0.2.110;rport=61744;received=192.168.200.77;branch=z9hG4bK3179897 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=eK8SmZmcU7UBr Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 192 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:18:57 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=sip:192.168.200.77:61744 %3Btransport%3Dws Content-Length: 0 ------------------------------------------------------------------------ in freeswitch i the registration: Call-ID: dlnrna9o4ngn25fb9d1vi2 User: 1001@mydomain Contact: "" <sip:1001@192.0.2.110 ;transport=ws;fs_nat=yes;fs_path=sip%3A192.168.50.60%3Blr%3Breceived%3Dsip%3A192.168.200.77%3A61744%3Btransport%3Dws> Agent: SIP.js/0.7.0 BB Status: Registered(WS-NAT)(unknown) EXP(2018-02-09 22:29:57) EXPSECS(510) Ping-Status: Reachable Ping-Time: 0.00 Host: FS01 IP: 192.168.50.60 Port: 5060 Auth-User: 1001 Auth-Realm: 192.168.50.60 MWI-Account: 1001@mydomain the register contain a path header with received=<original uri>. when i try to originate a call to this registered user the call goes nowhere :=) freeswitch@FS01> bgapi originate {origination_caller_id_number=1002}sofia/internal/1001%mydomain &echo() +OK Job-UUID: 200e2197-a6de-42d9-b4bb-63917223bc53 2018-02-09 22:25:40.500100 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables 2018-02-09 22:25:40.500100 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001 [327826f9-8a8c-46c7-b3ed-8bf22852df88] 2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:4819 (sofia/internal/1001) State Change CS_NEW -> CS_INIT 2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1001) Running State Change CS_INIT (Cur 1 Tot 217) 2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/1001) State INIT 2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:90 sofia/internal/1001 SOFIA INIT 2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1264 sip:192.168.50.60;lr;received=sip:192.168.200.77:61744;transport=ws Setting proxy route to sofia/internal/1001 2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1295 sofia/internal/1001 sending invite version: 1.6.19 git b1b21d0 2017-11-20 16:34:16Z 64bit

2 2

Originate with Dialplan
by Sonnie 14 Feb '18

14 Feb '18

Hello, I have a current scenerio were a user registers with freeswitch and makes a call out - Works Perfect! <include> <user id="1000">  <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> <variable name="cgr_reqtype" value="*prepaid"/> </variables> </user></include> and then in the default dialplan, i have this   <extension name="CGRateS_Auth"> <condition field="${cgr_notify}" expression="^$">  <action application="park" /> </condition> </extension> Now, i would love to send a call out with the originate command I've tried this - originate {origination_caller_id_number='123456789',ignore_early_media=true,originate_timeout=45}sofia/gateway/provider_one/7774262712 &playback(/usr/share/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello.wav) default CGRates_Auth That sends the call out and then hangs it up(expectedly) Is there a way i can replicate my working scenerio(maybe pass in those channel variable) but with the originate command? I'm naively thinking, maybe sending the call to a dialplan first before sending it to a gateway? Please pardon me if the answers are obvious.

1 0

DTMF relay in 3-way call
by Michael Avers 14 Feb '18

14 Feb '18

Hello, I have the following scenario: Deskphone dials PSTN Destination A. Once answered, deskphone 3-ways a second PSTN Destination B. From that point, DTMF pressed by Destination A needs to be heard by B. Is there a simple way to have that DTMF relay over from A to B? Currently the 3-way is happening on the deskphone (Yealink T42). Is there perhaps a different way they should be connected? Thanks Mike

1 0

Freeswitch support for text messages/commands between Verto/webRTC to/from conferenced backend SIP/RTP systems
by Mundkowsky, Robert 13 Feb '18

13 Feb '18

We are wondering if FreeSWITCH supports sending and receiving text messages between Verto/webRTC to/from backend SIP/RTP systems that conferenced together? 1) We want to use this to send commands from the web browser to the backend systems, for example: a. "hey backend system, please turn on the ASR to start listening." 2) We also wonder if FreeSWITCH supports sending commands from the backend to the frontend (web browser), for example: a. "hey avatar, here are some visemes to tell you how you should move your lips, arms, ... at the same time as playing this audio" If both Verto/SIP or webRTC/RTP are supported, is either more ideal? For instance, if the text/commands are over RTC then maybe we can use one clock for all RTP streams and/or RTP sessions and use the RTP timestamps to synchronize the commands with audio/video. Or maybe Verto/SIP is better for commands. I see there are various RFCs for text over RTP and text over RTCDataChannel in webRTC, but not sure if FreeSWITCH supports any of these. I also see FreeSWITCH mod_verto support text messages, but not sure if FreeSWITCH sends these to/from SIP (e.g. maybe via SIMPLE SIP) or via RTCDataChannel? 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. ________________________________

2 2

Originate with Dialplan
by Sonnie 13 Feb '18

13 Feb '18

I have a somewhat newbie question I have a current scenerio were a user registers with freeswitch and makes a call out - Works Perfect! <include> <user id="1000">  <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> <variable name="cgr_reqtype" value="*prepaid"/> </variables> </user></include> and then in the default dialplan, i have this   <extension name="CGRateS_Auth"> <condition field="${cgr_notify}" expression="^$">  <action application="park" /> </condition> </extension> Now, i would love to send a call out with the originate command I've tried this - originate {origination_caller_id_number='123456789',ignore_early_media=true,originate_timeout=45}sofia/gateway/provider_one/7774262712 &playback(/usr/share/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello.wav) default CGRates_Auth That sends the call out and then hangs it up(expectedly) Is there a way i can replicate my working scenerio(maybe pass in those channel variable) but with the originate command? I'm naively thinking, maybe sending the call to a dialplan first before sending it to a gateway? Please pardon me if the answers are obvious.

2 1

Re: [Freeswitch-users] FS fails to restart when TLS endpoints are registered (Solaris 11.3)
by GBC Networks Oy 13 Feb '18

13 Feb '18

Thanks Alexey for your help in diagnosing the problem. I dug further into the TCP stack and found out we are having 2 separate restart problems. The first problem is not that the socket is failing to be released, but apparently Sofia is trying to open the port without the SO_REUSEADDR option, which means FS must wait for all connection to close first. So as long as there are any TCP connections to those ports in the TIME_WAIT state, FS will not start Sofia. I tested with netcat and that program can bind to the port regardless of the state of old connections, so I suspect this is a compile error. The second problem is that the TLS and TCP registrations in sofia_reg_internal.db cause FS to throw the assertion error “Assertion failed: orq->orq_queue, file nta.c, line 8101”. UDP registrations are fine. Does FS try to continue these registrations assuming that the connection is still up or reopen a connection to the same port? The assert line is found in nta.c’s outgoing_create function. It seems that the outgoing message queue is uninitialized, so the assertion error is likely more a symptom of some other problem, instead of being the cause of the problem. How should I go about debugging the outgoing message queue? Thanks in advance, Mikko ---------- Edelleenlähetetty viesti ---------- > From: Alexey Sibyakin <alex(a)freeswitch.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 13 Feb 2018 17:23:06 +0900 > Subject: Re: [Freeswitch-users] FS fails to restart when TLS endpoints are > registered (Solaris 11.3) > It looks like Sofia is not releasing ip on Solaris. If so it's a bug and > it should be reported to Sofia's bug tracker or Solaris or both. If you can > reproduce this problem on supported OS (Debian 8 at this moment) you can > open an issue on FreeSWITCH Jira. > > Consider to read this first: https://freeswitch.org/ > confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA > > On Tue, Feb 13, 2018 at 2:30 AM, GBC Networks Oy <info(a)gbc.fi> wrote: > >> We're running FS on Solaris 11.3 and it is working perfectly, except that >> we're unable to restart FS without deleting the TLS registrations from >> sofia_reg_internal.db. >> Our FS version is FreeSWITCH Version 1.9.0+git~20171010T231940Z~5940b04d4e~64bit >> (git 5940b04 2017-10-10 23:19:40Z 64bit) >> >> Here are the symptoms: >> 1. Service stops FS with fs_cli -x "fsctl shutdown now" >> 2. FS stops and the service is restarted >> 3. FS starts up and fs_cli is working, but ” sofia status” is empty >> The console shows: >> freeswitch@fs> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3221 Error >> Creating SIP UA for profile: internal (sip:mod_sofia@123.123.123.123 >> :5060;maddr=123.123.123.123;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 >> SEC) >> >> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3231 Error Creating SIP UA for >> profile: internal (sip:mod_sofia@123.123.123.123 >> :5060;maddr=123.123.123.123;transport=udp,tcp) >> >> >> >> The likely causes for this are: >> >> >> 1) Another application is already listening on the specified >> address. >> >> 2) The IP the profile is attempting to bind to is not local to this >> system. >> >> Note: the IP is correct and nothing is listening on the port. >> >> 4. Next we stop FS and restart >> This time the console shows: >> 2018-02-12 13:33:51.132676 [INFO] switch_core_sqldb.c:1693 sofia:internal >> Starting SQL thread. >> 2018-02-12 13:33:51.132770 [DEBUG] sofia.c:3373 Starting thread for >> internal >> 2018-02-12 13:33:51.132777 [DEBUG] sofia.c:3020 Launching worker thread >> for internal >> 2018-02-12 13:33:51.138255 [NOTICE] sofia_reg.c:3398 Added gateway >> '5e4ac47f-6ce4-462d-9247-565711fe3d4e' to profile 'internal' >> 2018-02-12 13:33:51.138405 [NOTICE] sofia_reg.c:3398 Added gateway >> 'b2563fad-0051-4067-abc3-6a431e05481e' to profile 'internal' >> 2018-02-12 13:33:51.247731 [DEBUG] freeswitch_lua.cpp:372 DBH handle >> 125b300 Connected. >> 2018-02-12 13:33:51.254128 [DEBUG] freeswitch_lua.cpp:401 DBH handle >> 125b300 released. >> Assertion failed: orq->orq_queue, file nta.c, line 8101 >> >> 5. sqlite3 sofia_reg_internal.db "DELETE from sip_registrations WHERE >> status LIKE '%TLS%'" >> 6. FS starts normally >> >> >> Here is a registration (non-TLS) that doesn’t break sofia: >> # sqlite3 -header sofia_reg_internal.db "SELECT * from sip_registrations" >> call_id|sip_user|sip_host|presence_hosts|contact|status|ping >> _status|ping_count|ping_time|force_ping|rpid|expires|ping_ >> expires|user_agent|server_user|server_host|profile_name| >> hostname|network_ip|network_port|sip_username|sip_realm| >> mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host >> 313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS" >> <sip:USERNAME@REMOTE-IP:2054>|Registered(UDP)|Reachable|3|39 >> 824|1|unknown|1518455533|1518453577|snom320/8.7.5.35|USERNAM >> E||internal|fs|REMOTE-IP|2054|USERNAME|fs.domain.com|USERNAME| >> fs.domain.com||fs|fs.domain.com >> >> Here is a registration (TLS) that breaks the restart: >> # sqlite3 -header sofia_reg_internal.db "SELECT * from >> sip_registrations" >> call_id|sip_user|sip_host|presence_hosts|contact|status|ping >> _status|ping_count|ping_time|force_ping|rpid|expires|ping_ >> expires|user_agent|server_user|server_host|profile_name| >> hostname|network_ip|network_port|sip_username|sip_realm| >> mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host >> 313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS" >> <sip:USERNAME@192.168.6.236:2170;transport=tls;fs_nat=yes;fs >> _path=sip%3AUSERNAME%40REMOTE-IP%3A2170%3Btransport%3Dtls>| >> Registered(TLS-NAT)|Reachable|0||1|unknown|1518457302|1518453664|snom320/ >> 8.7.5.35|USERNAME||internal|fs|REMOTE-IP|2170|USERNAME|fs.domain.com >> |USERNAME|fs.domain.com||fs|fs.domain.com >> >> >> Best regards, >> Mikko >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > > > -- > > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> > Email: alex(a)freeswitch.com > Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> > Need commercial support? Contact sales(a)freeswitch.com for details. > > _______________________________________________ > 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 > http://www.freeswitch.org > >

1 0

test
by Ken Rice 13 Feb '18

13 Feb '18

Test 1234

6 6

chatplan: control sip messages
by antonio 13 Feb '18

13 Feb '18

Hi, Is it possible to block sip messages between sip clients? I try to set the action stop in chatplan but it doesn't do anything, the message is sent between the two endpoints. What i want is to be able to control witch accounts can send messages and limited destinations, i also want to able receive the messages in a common direction, like, chat(a)domain.local, without having the error the warning message "sofia_presence.c:225 Can't find registered user".. Thanks. -- Saludos / Regards / Cumprimentos Anónio Silva

1 1