July 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto bridge
by Gregor Nanger 04 Jul '18

04 Jul '18

We are saving verto registrations in our DB and have ip and port of every registration. Can we make bridge to verto endpoint directly with ip and port? 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

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Error copying SIP packets to homer
by David Villasmil 04 Jul '18

04 Jul '18

Hello guys, We have several Fs (1.6.20) configured to send SIP packets over to homer, but we're not seeing all packets being sent over, and we see the following errors on the CLI: error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: No such file or directory error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: No such file or directory error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: No such file or directory error: tport_logging: capture socket error: Bad file descriptor error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: No such file or directory error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: No such file or directory error: tport_logging: capture socket error: No such file or directory error: tport_logging: capture socket error: Bad file descriptor error: tport_logging: capture socket error: Success error: tport_logging: capture socket error: Bad file descriptor Has this happened to anyone? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Limiting call length
by Rick Jarvis 03 Jul '18

03 Jul '18

Hey all Looking for a way to limit all calls to a maximum of 10 minutes, when originating a call using the event socket. I’ve tried using sched_hangup like this: originate {origination_caller_id_number=01234567890,execute_on_answer=sched_hangup +600 alloted_timeout}sofia/. . . . But no dice… R

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FireFox 61 + Freeswitch 1.6.20
by Geoff Mina 03 Jul '18

03 Jul '18

Is anyone else having issues with WebRTC and the latest release of FireFox? We are showing one way media - with media flowing properly TO the browser, but no media coming back from the browser. Everything in the logs looks OK - no errors. We are using SIP.js as the library of choice for signaling. We are showing SRTP packets flowing bi-directionally (using tcpdump), but since they are DTLS encrypted, I am unable to see exactly what is in each packet to determine if there is media being sent from the browser or they are just empty packets. Any ideas and/or other folks seeing the same issue - would love to hear your thoughts. Thanks, Geoff

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Memory leak with MIME
by Mody, Darshan (Darshan) 03 Jul '18

03 Jul '18

Hi We are observing memory leak when we have MIME data in the incoming invite. The resident memory usage increases constantly. When we tried the traffic without MIME and only SDP the usage stays constant. Attaching the freeswitch logs for the same. Thanks Darshan

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Failed Registration [908]
by Paul Muaddib 03 Jul '18

03 Jul '18

Hi, can someone please look at it and tell my what I am doing wrong? NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration [908], setting retry to 30 seconds. My setup: ######## I have a static IP address for my router. My router does not have UPnP or NAT-PMP. Phones and Freeswitch server are on the same local network. The local network is behind NAT PHONE -> FS -> NAT -> Public Internet I only want to connect to the PSTN via my sip gateway provider. According to this manual I did the following setting: https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal vars.xml ###### <X-PRE-PROCESS cmd="set" data="external_rtp_ip=X.X.X.X“/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=X.X.X.X"/> internal.xml ######### <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml ########## <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> ( What is a little bit strange though is that nat_map status is not working [ERR] mod_commands.c:751 nat_map API called while NAT not initialized ) Is the reason why it does not work the port specification in the Contact field (sngrep output)? Contact: <sip:gw+sip-trunk.telekom.de@X.X.X.X:5080;transport=tcp;gw= sip-trunk.telekom.de> I do not want to setup port forwarding in the firewall. The connection should remain open via a keep alive signal. sngrep output ########### 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060 REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0 Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De Max-Forwards: 70 From: <sip:username@reg.sip-trunk.telekom.de>;tag=9aldsj8n67a To: <sip:username@reg.sip-trunk.telekom.de> Call-ID: CSeq: 111844514 REGISTER Contact: <sip:gw+sip-trunk.telekom.de@X.X.X.X:5080;transport=tcp;gw= sip-trunk.telekom.de> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 Provider settings: ############# output proxy: reg.sip-trunk.telekom.de registrar: sip-trunk.telekom.de sip_profiles/external/telekom_voip.xml ############################## <include> <gateway name="sip-trunk.telekom.de"> <param name="username" value=„XXXXXX“/> <param name="password" value=„XXXXXX“/> <param name="realm" value="sip-trunk.telekom.de"/> <param name="proxy" value="reg.sip-trunk.telekom.de"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> </gateway> </include> Firewall settings ############# # block all traffic in all directions block log # loopback interface is not filtered set skip on lo0 voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}" voip_udp_client_out = "{domain, ntp, sip, 3478}" match out on $wan_if from any nat-to ($wan_if) pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out pass on $voip_if inet proto udp from any to port $voip_udp_client_out pass out on $wan_if inet proto tcp from any to port $tcp_client_out pass out on $wan_if inet proto udp from any to port $udp_client_out Thank you for helping :) Regards, Paul

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Video call recording and mobile push notifications
by Jeremy Renner 03 Jul '18

03 Jul '18

Hi, I would like to setup up my SIP server / PBX for my business, likes broadsoft, now we have some candidates: 1. Open source solution: - Asterisk PBX, - Freeswitch PBX - Kamailio - OpenSIPS 2. Business solution: - Brekeke PBX(https://www.brekeke.com - Vodia PBX(https://www.vodia.com) - 3CX PBX(https://www.3cx.com) - PortSIP PBX(https://www.portsip.com/portsip-pbx) *Below features are mandatory for our project:* - Video call recording (For the finance industry, the video recording is necessary) - Push notifications for mobile app - Multi-tenant support - Both Linux and Windows support (at 1st stage, we would like to run it on Windows server and migrate it to Linux server in the future if users increased), the Linux support is required, the Windows support is preferred. We have some questions: 1. Does the the Freeswitch can works as the Broadsoft ? 2. If yes, does the Freeswitch support push notifications and video recording ? 3. Does the Freeswitch can works for Multi-tenant ? 4. If Freeswtich doesn't support push notifications, does there has any 3rd plugin supported ? So far according to our research, with the business solution: - The Vodia PBX, PortSIP PBX and brrekeke all are support Multi-tenant, the 3CX is not. - The 3CX and PortSIP support push notifications, - The PortSIP also provide client SDK, with 3CX we only see the 3CX provide client apps, does 3CX has client SDK provided ? - It's seems all these PBX are support video recording ? - The PortSIP PBX and 3CX both support Linux. Please help me to make the decision, base on your experiences, which one (open source or business solution) is good to us ? I'm really new to VoIP... Thanks in advance. Best regards,

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problem loading fs_curl (contrib)
by Blackhold 03 Jul '18

03 Jul '18

Hi, I'm adding mod fs_curl from intralanman[1]. I have freeswitch installed from git[2]. I followed this manual[3] to install fs_curl. It seems that pgsql connection is ok, but when configuring users to DB, list_clients doesn't shows users from database (I also tried to load FS without users file configuration). Looking at logs it appears that: 2018-07-02 19:20:49.250075 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object file: No such file or directory** Oviously, I could not install freeswitch-mod-xml-curl package, so I simply downloaded the .deb package[4] and extracted file mod_xml_curl.so to /usr/local/freeswitch/mod/ directory. Then it appeared an other error that I can't lead 2018-07-02 21:16:47.207969 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **libssl.so.1.0.0: cannot open shared object file: No such file or directory** I tried to create symlinks calling libssl.so.1.0.0 but it seems it have some problem to read it. The libraries that I have are libssl.so.1.1 and libssl.so.1.0.2 and are located at /usr/lib/x86_64-linux-gnu the error continues, then I tried to install libssl that contains libssl.so.1.0.0[5] and finally get mod_xml-curl activates, then appeared and other error: 2018-07-02 21:52:23.411646 [ERR] mod_xml_curl.c:315 Received HTTP error 500 trying to fetch http://localhost/fs_curl/ data: [hostname=XXX&section=directory&tag_name=&key_name=&key_value=&Event-Name=REQUEST_PARAMS&Core-UUID=XXX&FreeSWITCH-Hostname=XXX&FreeSWITCH-Switchname=XXX&FreeSWITCH-IPv4= X <http://10.90.226.83/>XX &FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2018-07-02%2021%3A52%3A23&Event-Date-GMT=Mon,%2002%20Jul%202018%2019%3A52%3A23%20GMT&Event-Date-Timestamp=1530561143405060&Event-Calling-File=sofia.c&Event-Calling-Function=launch_sofia_worker_thread&Event-Calling-Line-Number=3048&Event-Sequence=35&purpose=gateways&profile=external] The apache saids me that got this error: [Mon Jul 02 21:55:23.817938 2018] [:error] [pid 27216] [client 10.228.192.219:38874] PHP Fatal error: Uncaught Error: Call to undefined function ereg_replace() in /var/www/html/fs_curl/index.php:13\nStack trace:\n#0 {main}\n thrown in /var/www/html/fs_curl/index.php on line 13 I fixed with preg_replace() and gives this error: [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] PHP Warning: preg_replace(): No ending delimiter '^' found in /var/www/html/fs_curl/index.php on line 13 Changing ereg for preg, it only gives a warning and freeswitch don't throw any errors* [Mon Jul 02 22:05:53.353522 2018] [:error] [pid 27218] [client ::1:32960] PHP Warning: preg_replace(): No ending delimiter '^' found in /var/www/html/fs_curl/index.php on line 13 * yep it do, but on trying to create table schemas to db for example: switch_pgsql.c:415 Query (create index sr_call_id on sip_registrations (call_id)) returned PGRES_FATAL_ERROR Finally I executed fs_cli and listed users, but it appears empty. In database I have created 5 users. freeswitch@freeswitch> list_users userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number +OK # IN DB freeswitch_db=# select * from directory_domains; id | domain_name ----+-------------------- 4 | 10.XXX 1 | crustaci.mydomain.com (2 rows) freeswitch_db=# select * from directory; id | username | domain_id | cache ----+----------+-----------+-------- 1 | 1010 | 1 | 300000 2 | 1011 | 1 | 300000 3 | 20000 | 4 | 0 4 | 20001 | 4 | 0 5 | 20002 | 4 | 0 (4 rows) freeswitch_db=# select * from directory_params; id | directory_id | param_name | param_value ----+--------------+------------+------------- 1 | 3 | password | XXX 2 | 4 | password | XXX 3 | 4 | password | XXX (2 rows) Some ideas for why it is not working? Thanks you much! [1] https://freeswitch.org/stash/projects/FS/repos/freeswitch-contrib/browse/in… [2] http://blackhold.nusepas.com/2018/05/26/freeswitch-1-9-en-debian-8-psql/ [3] https://saevolgo.blogspot.com/2012/07/freeswitch-with-sip-users-in-mysql-mo… [4] http://files.freeswitch.org/repo/deb/debian-unstable/pool/main/f/freeswitch… [5] https://packages.debian.org/wheezy/amd64/libssl1.0.0/download - 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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Re: [Freeswitch-users] Failed Registration with status DNS Error [503]
by Henning Heedfeld 02 Jul '18

02 Jul '18

Hi, to get the „Deutsche Telekom SIP Trunk“ working, configure your gateway like this: <gateway name="telekom"> <param name="realm" value="sip-trunk.telekom.de"/> <param name="register-proxy" value="reg.sip-trunk.telekom.de"/> <param name="outbound-proxy" value="reg.sip-trunk.telekom.de"/> <param name="register" value="true"/> <param name="from-user" value="+49xxxxxxxx"/> <param name="username" value="<Zugangsnummer>"/> <param name="password" value="<Passwort>“/> <param name="extension" value="auto_to_user"/> </gateway> There is no need to set any tcp params, the DNS SRV record does the job. Good luck, Henning > Am 29.06.2018 um 19:38 schrieb Paul Muaddib <paul.muaddib83(a)gmail.com>: > > Hi, > > how do I fix this problem: > Failed Registration with status DNS Error [503]? > > I already search the net but couldn't find a proper answer. The only thing that I found was something about DNS SRV Resource Record but this was fixed in freeswitch a long time ago. > > My Setup: > FreeSWITCH Version 1.6.19 > Debian 7 wheezy > Freeswitch(Debian) -> Router(OpenBSD) -> Internet > > Failed Registration with status DNS Error [503] > 2018-06-29 19:27:21.732868 [NOTICE] sofia_reg.c:448 Registering telekom_voip > nua.c:622 nua_register() nua: nua_register: entering > nua_stack.c:529 nua_signal() nua(0x8c8c0c8): sent signal r_register > nua_stack.c:569 nua_stack_signal() nua(0x8c8c0c8): recv signal r_register > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:403 soa_set_params() soa_set_params(static::0xb5c1a298, ...) called > nua_dialog.c:338 nua_dialog_usage_add() nua(0x8c8c0c8): adding register usage > nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb6279478) > nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip > sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "sip-trunk.telekom.de.") returned 1 entries > nta.c:10398 outgoing_query_naptr() nta: for "sip-trunk.telekom.de" query "sip-trunk.telekom.de" NAPTR (cached) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._udp.sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._udp.sip-trunk.telekom.de.") returned 1 entries > nta.c:10598 outgoing_query_srv() nta: for "sip-trunk.telekom.de" query "_sip._udp.sip-trunk.telekom.de" SRV (cached) > sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "sip-trunk.telekom.de.") called > sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "sip-trunk.telekom.de.") returned 1 entries > nta.c:10803 outgoing_query_a() nta: for "sip-trunk.telekom.de" query "sip-trunk.telekom.de" A (cached) > nta.c:1350 set_timeout() nta: timer set to 32000 ms > nta.c:1348 set_timeout() nta: timer shortened to 5000 ms > nua_stack.c:271 nua_stack_event() nua(0x8c8c0c8): event r_register 503 DNS Error > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x8c8c0c8): removing register usage > nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb6279478) > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2018-06-29 19:27:21.732868 [ERR] sofia_reg.c:2447 telekom_voip Failed Registration with status DNS Error [503]. failure #34 > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2018-06-29 19:27:22.772957 [WARNING] sofia_reg.c:505 telekom_voip Failed Registration [503], setting retry to 30 seconds. > nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER (124810556) > nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0xb60911e0) > nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free > nta.c:1289 agent_timer() nta: timer not set > > sofia_dig --tcp reg.sip-trunk.telekom.de > Preference Weight Transport Port Address > ================================================================================ > 1 1.000 tcp 5060 217.0.26.165 > 2 1.000 tcp 5060 217.0.26.163 > 3 1.000 tcp 5060 217.0.26.133 > > Gateway settings > <include> > <gateway name="telekom_voip"> > <param name="username" value=„XXXXXXXX“/> > <param name="password" value=„XXXXXXXX“/> > > <param name="proxy" value="reg.sip-trunk.telekom.de"/> > <param name="register-proxy" value="sip-trunk.telekom.de"/> > <param name="register" value="true"/> > <param name="register-transport" value="tcp"/> > <param name="expire-seconds" value="3600"/> > > </gateway> > </include> > > Regards, > Paul

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Seperate "new session" and "dial" in JS
by Francesco Facco de Lagarda 02 Jul '18

02 Jul '18

I am trying to make a call progess monitor I need to bind together the answering of incoming session The call to the outgoing session. What I am doing is basically trying to set a user defined variable "t3Id" to both sessions so I "marry then together" from the ESL At present I am doing (just a simplified extract of the code) var t3Id = "##randomly generated##"; session.answer(); session.setVariable("t3Id", t3Id); sessOut=new Session(##my dial string##) sessOut.setVariable("t3Id", t3Id); if(sessout.ready()) { bridge(session, sessOut); } session.hangup(); sessOut.hangup(); Unfortunately I noticed that IF I use sip services (I'm using Messagenet) the outgoing call never gets the variable set! Is it possible to do something like sessOut=new Session() sessOut.setVariable("t3Id", t3Id) sessOut.dial(##my dial string##) ??? I cant find any other parameter common to BOTH sessions to marry the two together!!!

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