August 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

First steps and outbound calls
by John Tuxies 10 Aug '20

10 Aug '20

I am looking for outbound calls and i am stuck with it on how to place outgoing calls. In the /sip_profiles/external there is an xml with the sip trunk provider_A .xml and i can see it registered 👍. The provider_A has a range of DIDs 00302109065730-9 In the /dialplan/public i have created one xml for each DID to route to particular extension 👍 . When someone calls one of the DIDs it goes to the assigned extension. I would like to place outgoing calls and when each extension places an outgoing call to show its DID, as in 1000 as 2109065730...1009 as 2109065739. All other extensions without DID, to show the default number (2109065730) I would like to have a prefix of 0 for the outgoing calls at the moment i would like to allow only the following numbering plans 2XXXXXXXXX (10digits) national landline 69XXXXXXXX (10digits) national mobile 800XXXXXXX (10digits) national Freephones 801XXXXXXX (10digits) national local charge 1XX emergency calls 1XXX Special services 1XXXX Special services Thanks in advance

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Handle media of all calls via port 443?
by David P 09 Aug '20

09 Aug '20

Can FS v1.10 be configured to handle SRTP of all calls via port 443? This would allow us to stop using coturn, which is often too slow to provide a TLS relay candidate before browsers finish gathering ice candidates.

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Dynamic WAN IP
by John Tuxies 08 Aug '20

08 Aug '20

I am doing my first steps with Freewswitch and i have a few questions regarding the config. I have a DSL connection at home with dynamic WAN IP and the router has the No-IP script to resolve a domain eg atux.ddns.net how can i configure it in the Freeswitch, please? Also how do i configure a STUN server, since i would like to register some remote extensions on Freeswitch, for testing. How do i configure a STUN server and if you have a decent STUN server to suggest, please.

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Weird behaviour/unavailability of a B gateway
by Igor Potjevlesh 06 Aug '20

06 Aug '20

Hello, I'm facing an issue with Freeswitch. Suddenly, and on an random basis time interval, calls are no more forwarded to the B gateway. I see then that Freeswitch did an attempt to start sending the call to leg B but finally terminate on 503. And a tshark doesn't shows that an INVITE is sent to the B gateway. The result of a "sofia status" on the concerned gateway showing that the gateway is considered as Up. Freeswitch is running on version 1.10.1. B gateway is reach through a DNS URI. Would it be possible that one time failure (such as 1x503 from B gateway and/or a failure in the DNS lookup of the URI) could result in this behaviour? Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

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registration with DID range
by John Tuxies 06 Aug '20

06 Aug '20

for lab purposes i need to emulate ITSP connectivity with multiple DIDs (range of numbers). For example in FS create 4 extensions 1000-1003. Each extension will forward a particular range of numbers: -1000: +441144630000-99 -1001: +441144630100-99 -1002: +441144630200-99 -1003: +441144630300-99 Then different PBXs will get one the numbers 100x and register it as sip trunk eg 1000@FS_LAN_IP. I would like to ask for some help on how to configure it, please.

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Issue with SDP - Call hangs up right after SIP 200 OK
by Marc Bernard 05 Aug '20

05 Aug '20

Hello People, Ive been investigating an issue with SDP when sending a call to a Panasonic KX-TGP600 Phone, the call end as soon as the Panasonic send its SIP OK I could not see what is wrong with the SDP and why I get "INVITE: error parsing SDP answer in 200 OK" resulting in "988 Incomplete offer/answer" I also find it weird that there is no output for this error when Sofia debug is not enabled. It took me a while before I thought of trying to enable more debugging. There is also the "not found by name */1.1.217.127:1029" Part which I am not sure is related/relevant in this. -- Here is the console log portion that I thought was relevant: 2020-07-17 15:44:33.495660 [DEBUG] sofia.c:7325 Channel sofia/private/+15145557030(a)2.2.127.218 <mailto:sofia/private/+15145557030@2.2.127.218> entering state [early][180] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-07-17 15:44:33.495660 [NOTICE] switch_ivr_originate.c:556 Ring Ready sofia/private/+15145557030(a)2.2.127.218 <mailto:sofia/private/+15145557030@2.2.127.218> ! tport.c:2759 tport_wakeup_pri() tport_wakeup_pri(0x7f0e440044a0): events IN tport.c:2882 tport_recv_event() tport_recv_event(0x7f0e440044a0) tport_type_udp.c:324 tport_recv_dgram() tport(0x7f0e440044a0): zero length packet tport.c:2759 tport_wakeup_pri() tport_wakeup_pri(0x7f0e440044a0): events IN tport.c:2882 tport_recv_event() tport_recv_event(0x7f0e440044a0) tport.c:3223 tport_recv_iovec() tport_recv_iovec(0x7f0e440044a0) msg 0x7f0e44011e30 from (udp/1.1.131.37:5060) has 845 bytes, veclen = 1 tport.c:3041 tport_deliver() tport_deliver(0x7f0e440044a0): msg 0x7f0e44011e30 (845 bytes) from udp/1.1.217.127:5060/sip next=(nil) nta.c:3304 agent_recv_response() nta: received 200 OK for INVITE (22953743) nta.c:3371 agent_recv_response() nta: 200 OK is going to a transaction tport.c:4240 tport_release() tport_release(0x7f0e440044a0): 0x7f0e4400bca0 by 0x7f0e44010c40 with 0x7f0e44011e30 nua_dialog.c:564 nua_dialog_usage_set_refresh_range() nua(): refresh session after 55 seconds (in [55..65]) soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f0e4401a9b0, (nil), 0x7f0e4401267a, 259) called nua_session.c:970 nua_session_client_response() nua(0x7f0e48048c30): INVITE: error parsing SDP answer in 200 OK (200) nua_stack.c:271 nua_stack_event() nua(0x7f0e48048c30): event r_invite 200 OK nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip tport.c:3275 tport_tsend() tport_tsend(0x7f0e440044a0) tpn = */1.1.217.127:1029 tport.c:4064 tport_resolve() tport_resolve addrinfo = 1.1.217.127:1029 tport.c:4698 tport_by_addrinfo() tport_by_addrinfo(0x7f0e440044a0): not found by name */1.1.217.127:1029 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering tport.c:3612 tport_vsend() tport_vsend(0x7f0e440044a0): 391 bytes of 391 to udp/1.1.217.127:1029 tport.c:3510 tport_send_msg() tport_vsend returned 391 nta.c:8315 outgoing_send() nta: sent ACK (22953743) to */1.1.217.127:1029 nua_stack.c:271 nua_stack_event() nua(0x7f0e48048c30): event i_media_error 988 Incomplete offer/answer nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:1784 soa_terminate() soa_terminate(static::0x7f0e4401a9b0) called nua_stack.c:359 nua_application_event() nua: nua_application_event: entering soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f0e4401a9b0) called nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip tport.c:3275 tport_tsend() tport_tsend(0x7f0e440044a0) tpn = */1.1.217.127:1029 tport.c:4064 tport_resolve() tport_resolve addrinfo = 1.1.217.127:1029 tport.c:4698 tport_by_addrinfo() tport_by_addrinfo(0x7f0e440044a0): not found by name */1.1.217.127:1029 tport.c:3612 tport_vsend() tport_vsend(0x7f0e440044a0): 616 bytes of 616 to udp/1.1.217.127:1029 tport.c:3510 tport_send_msg() tport_vsend returned 616 nta.c:8315 outgoing_send() nta: sent BYE (22953744) to */1.1.217.127:1029 tport.c:4178 tport_pend() tport_pend(0x7f0e440044a0): pending 0x7f0e4400d880 for udp/1.1.131.37:5060 (already 0) nta.c:1348 set_timeout() nta: timer shortened to 1000 ms nua_session.c:4140 signal_call_state_change() nua(0x7f0e48048c30): call state changed: proceeding -> terminating, received answer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f0e4401a9b0, [0x7f0e705a0778], [0x7f0e705a0780], [(nil)]) called soa.c:616 soa_get_params() soa_get_params(static::0x7f0e4401a9b0, ...) called nua_stack.c:271 nua_stack_event() nua(0x7f0e48048c30): event i_state 200 OK nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-07-17 15:44:35.795689 [DEBUG] sofia.c:7325 Channel sofia/public/golfvalneigette3 entering state [terminating][200] -- And here is the SIP OK from the Panasonic: <mailto:&iTEq@8GæÙ¸ª%25ÄÄ]äáSIP/2.0> &iTEq(a)xn--8g a%]sip/2-p7a1babz73al06a.0 200 OK Via: SIP/2.0/UDP 1.1.131.37;rport=5060;branch=z9hG4bKttaHKmXNgBtSQ;received=1.1.131.37 Call-ID: cbecb6ef-4300-1239-dfa0-000c29124868 From: "Debug Proxy" <sip:+15145552353@1.1.131.37>;tag=rDQF79t14QmKr To: <sip:9919605@dev-proxy.xxx.com>;tag=1661639224 CSeq: 22952030 INVITE Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER Supported: 100rel,precondition Contact: <sip:Account1@192.168.1.193:5060> Require: replaces Content-Type: application/sdp Server: Panasonic-KX-TGP600/11.110 (4c364e38b998) Content-Length: 263 v=0 o=- 1595011648 1595011648 IN IP4 192.168.1.193 s=- c=IN IP4 192.168.1.193 t=0 0 m=audio 16176 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 m=video 0 RTP/AVP 34 0 16 178 64 0 16 178 0 0 -- I have a PCAP file for the whole call should it be needed. Any help resolving this issue would be greatly appreciated. Kindly,

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mod_http_cahce doesn't delete old media files using default-max-age
by Yuriy Nasida 05 Aug '20

05 Aug '20

Hi guys, Do you know why mod_http_cahce doesn't delete old media files? FS only removes internal cache data but downloaded media files still here after default-max-age timeout. I have to reload this module to get audio files removed. Thanks.

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Internal profile stops responding
by Stefan Kainz 04 Aug '20

04 Aug '20

Hello, i have come by a very strange problem. The internal sofia-profile just stops responding after a few days on Port 5060. Restarting the profile does not help, but restarting freeswitch itself does. Strangely enough, the external profile ( on the same ip, but different port of course ) keeps working. No Problems there. A few things i have checked while it was happening the last time: - Profile watchdog is disabled - netstat -tlupn shows that freeswitch is still listening on Port 5060 - There is nothing useful in the logs, especially no error/crit message - I can see sip-traffic arriving on Port 5060 with tcpdump, but it is not answered - turning on sofia loglevel all 9 does not show anything - It is not one or two specific ip's that get blocked, but all traffic I also checked the firewall / fail2ban, but no problem there either. After all, a restart of freeswitch solves the problem, with no changes to the firewall. It happens on pretty regular basis ( every one or two days ), but there is no time-pattern in between the occurrences. Im using debian 10, with FreeSWITCH Version 1.10.3-release~64bit (-release 64bit). Has anybody come across something similiar? regards,

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Retrieval of Caller-ID
by Chandramouli P 03 Aug '20

03 Aug '20

Hello, Please find my below environment: Operating System: Debian (v. 9.13) FreeSWITCH Version: 1.10.3 (64bit) Softphones that I am using: MicroSIP, 3CX, Grandstream I hosted our server in AWS and was able to make and receive calls successfully. Along with "valet_park_in" and "valet_park_out", also enabled the "CID Lookup" module. I am facing two issues in Valet parking and below is the scenario: User A calls User B. User B transfers User A call to *5900. Now User B is free, and User A is in valet parking at 5901 extension. Here, It is announcing the parked extension i.e. 5901 to User A, instead of User B. It should announce to User B. Right? *How to solve this issue?* Now, User C calls *5901 and is able to speak to User A. In this scenario, User C should be able to see the caller ID of User A. *How to get the caller ID of User A and display and display to User C?* Any help would be appreciated. Thank you. Best Regards, Chandramouli.

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Re: [Freeswitch-users] Can a module tap media?
by David P 02 Aug '20

02 Aug '20

Where can I find guidance about creating a media bug for audio+video? Adapt mod_cv? I'd like the inbound streams to be handled by a scriptable env like python. Or could streams be exposed to mod_python? ---------- Forwarded message ---------- From: Brian West <brian(a)freeswitch.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Sun, 26 Jul 2020 13:02:47 -0500 Subject: Re: [Freeswitch-users] Can a module tap media? Yes, you'd need to write your own media bug for that! On Sun, Jul 26, 2020 at 12:43 PM David P <davidswalkabout(a)gmail.com> wrote: Can a module like mod_lua continually monitor the audio and video of a call leg? I can't find any examples.

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