July 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Park command terminates script
by Francesco Facco de Lagarda 19 Jul '18

19 Jul '18

I have a simple js script that answers to an extension and calls operator on ext 1000: session.answer(); // code that asks for user pin, checks credit, plays error messages, etc., hangsup on error // if it gets here, all ok var sessOp = new Session("user/1000"); if (sessOp.ready()) { bridge(session, sessOp); while(session.ready()) { } } session.hangup(); sessOp.hangup(); SO FAR SO SIMPLE! Now from ESL (I'm currently testing issuing commands on fs_cli), I need to: 1. Customer chats with op 2. Park call 3. Start a new session with an external number 4. Bridge with operator 5. Park operator-external number 6. Bridge customer-external number THE PROBLEM As SOON as I park the customer-operator call at point(2) with uuid_transfer <customer sessionid> -both park inline the js script considers the call terminated and hangs up.. what can I put in the js loop (currently while(session.ready() {} ) To keep the script running until the customer finally hangs up??? Could I use uuid_hold to achieve the same? I tried but when I hold the customer's leg, it's the OPERATOR that gets the music, I tried to originate call in fs_cli and then bridge to customer, but silence on both ends.

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Loop audio file with uuid_displace
by Ciprian Dosoftei 19 Jul '18

19 Jul '18

Hello everyone, I am looking for a way to execute uuid_displace in a fashion which would loop the file indefinitely, til uuid_displace stop is invoked. The confluence page does not advertise such options: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_comma… However, upon digging deeper in the switch_ivr_displace_session function (switch_ivr_async.php), it appears the last parameter of the command (flags) is parsed somewhat different than suggested in the confluence page, namely it would set bug's user data loop member to a non-zero value if the 'l' character is found in the flags argument: if (flags && strchr(flags, 'l')) { dh->loop++; } Subsequently, the displacement callback (read and write) do account for the user data loop member when they are being invoked (i.e. evaluated in if (dh->loop) statements), which suggest the looping feature/flag should be supported. Nonetheless, I could not get it working by specifying 'l' as uuid_displace fifth parameter. Any advice is appreciated! See you all next week in Chicago! -Ciprian -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender.

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compile with gcc 8
by Juan Pablo L 18 Jul '18

18 Jul '18

Hello, i am trying to compile master branch using the gcc version 8.1.1 20180502 but it gives all sort of error related to strncpy and other string library functions such as format errors or string truncation issues i was replacing the strncpy with memcpy but i reached a point ... any advice how to compile using that compiler on linux fedora 28 64bits ? thank you!!

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limit reset when transferring to same context - wiki wrong?
by Royce Mitchell III 18 Jul '18

18 Jul '18

According to this page: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit "Note that a limit is active within a given context - if you transfer an inbound call from the public dialplan to an extension in the default dialplan - any limit you just set in the public dialplan will be reset." Either I am misunderstanding something, or this statement may not be 100% accurate for version FreeSWITCH Version 1.6.20+git~20180507T185611Z~43a9feb7f8~64bit (git 43a9feb 2018-05-07 18:56:11Z 64bit) The following is an excerpt from the logs showing a call in the public context getting transferred to another destination in the public context and the limit is getting reset: EXECUTE sofia/external/CENSORED(a)CE.NS.OR.ED limit(hash inbound IHCAV 2 didlimit$1) 2018-07-17 10:30:15.452366 [DEBUG] switch_limit.c:126 incr called: inbound_IHCAV max:2, interval:0 2018-07-17 10:30:15.452366 [DEBUG] mod_hash.c:196 Usage for inbound_IHCAV is now 1/2 EXECUTE sofia/external/CENSORED(a)CE.NS.OR.ED transfer(freedom2 xml public) 2018-07-17 10:30:15.452366 [DEBUG] switch_ivr.c:2165 (sofia/external/ CENSORED(a)CE.NS.OR.ED) State Change CS_EXECUTE -> CS_ROUTING 2018-07-17 10:30:15.452366 [DEBUG] mod_hash.c:297 Usage for inbound_IHCAV is now 0 2018-07-17 10:30:15.452366 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/CENSORED(a)CE.NS.OR.ED to xml[freedom2@public] Am I missing something? Is there a way to keep the call in the limit count when transferring it to the same context? Royce Mitchell, IT Consultant ITAS Solutions royce3(a)itas-solutions.com

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verto and adapter.js
by David P 17 Jul '18

17 Jul '18

What is the FreeSwitch team's stance on whether to use adapter.js in verto? Recent speakers in the weekly call, like Chad Hart, mention that adapter.js is the current best option for WebRTC. Also, I see adapter.js in the 'vendor' folder in the FS repo: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/vert… however, it does not seem to play any role with the other files there, files which are used in "demo" and "video_demo".

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No video when calling into conference with Cisco Jabber
by Ari Sonesh 17 Jul '18

17 Jul '18

We are getting audio only but no video when calling into conference with Cisco Jabber. Other SIP video clients (e.g Jabber) calling into the conference are OK. Sip negotiations are fine, SDP is fine and all parties agreeing on H.264 codec. The log (at debug level) doesn't show any relevant errors or warnings. Any ideas what can be wrong and/or how to troubleshoot the problem? Thank you *Ari *

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continue_on_fail triggered by NORMAL_CLEARING
by Royce Mitchell III 17 Jul '18

17 Jul '18

We have a freeswitch box acting as a DR backup to the live system (called "freedom2" because that is what our vendor calls it). We are using continue_on_fail to catch conditions when the live system is down in order to forward it to a DR fifo. Occasionally (rarely) I'm getting a BYE back from the live system and the call is getting sent to DR and I can't figure out why this is happening. Would someone please take a glance at this and help me understand if I'm doing this wrong or if FreeSWITCH is misbehaving here? We are running FreeSWITCH Version 1.6.20+git~20180507T185611Z~43a9feb7f8~64bit (git 43a9feb 2018-05-07 18:56:11Z 64bit) /proc/version: Linux version 3.16.0-6-amd64 (debian-kernel(a)lists.debian.org) (gcc version 4.9.2 (Debian 4.9.2-10+deb8u1) ) #1 SMP Debian 3.16.56-1+deb8u1 (2018-05-08) /etc/debian_version: 8.10 Here is the dialplan in question: <extension name='freedom2 (default)' continue='false' priority='99'> <condition field='destination_number' expression='^\+?1?(\d+)$'> <action application="log" data="ALERT INBOUND ${sip_gateway} -> $1 -> freedom2"/> <action application="limit" data="hash inbound freedom2 38 freedomlimit$1"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER"/> <action application="bridge" data="sofia/internal/$1@$${freedom2_trunk}"/> <action application="log" data="ALERT FREEDOM2 CALL FAILURE, SENDING $1 TO q1_dr_mode"/> <action application="set" data="route=q1_dr_mode"/> <action application="jitterbuffer" data="60"/> <action application="set" data="effective_caller_id_name=${custom_caller_name}"/> <action application="set" data="effective_caller_id_number=${custom_caller_id}"/> <action application="set" data="origination_caller_id_name=${custom_caller_name}"/> <action application="set" data="origination_caller_id_number=${custom_caller_id}"/> <action application="answer"/> <action application="playback" data="/dr/caller_preannounce.wav"/> <action application="set" data="fifo_chime_list=/dr/caller_preannounce.wav"/> <action application="set" data="fifo_chime_freq=30"/> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="q1 in"/> </condition> </extension> Here is an example of a call that got sent to the live system, even though the call appears to have ended in NORMAL_CLEARING. I started with the BYE packet back from the live system and cut it off at the point that the log application executes on the line after the bridge. I attempted to replace instances of public IP and phone #'s with placeholders. recv 361 bytes from udp/[$${freedom2_trunk}]:5060 at 11:50:43.514704: ------------------------------------------------------------------------ BYE sip:mod_sofia@$${freeswitch_ip}:5060 SIP/2.0 Via: SIP/2.0/UDP $${freedom2_trunk}:5060;branch=z9hG4bK707449280;rport From: <sip:${DID}@$${freedom2_trunk}>;tag=934960508 To: "${ANI}" <sip:${ANI}@$${freeswitch_ip}>;tag=FS9pD3eK574Sj Call-ID: 42db32d8-0220-1237-a8a1-02217f6f1b40 CSeq: 125455605 BYE Max-Forwards: 70 User-Agent: eConn 1.03 Content-Length: 0 ------------------------------------------------------------------------ tport.c:3023 tport_deliver() tport_deliver(0x7f6608004290): msg 0x7f6608061fc0 (361 bytes) from udp/$${freedom2_trunk}:5060/sip next=(nil) nta.c:2880 agent_recv_request() nta: received BYE sip:mod_sofia@$${freeswitch_ip}:5060 SIP/2.0 (CSeq 125455605) nta.c:3248 agent_aliases() nta: canonizing sip:mod_sofia@$${freeswitch_ip}:5060 with contact nta.c:3060 agent_recv_request() nta: BYE (125455605) going to existing leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:271 nua_stack_event() nua(0x7f65fc01dcd0): event i_bye 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [NOTICE] sofia.c:1012 Hangup sofia/internal/${DID}@$${freedom2_trunk} [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7f65fc01dcd0): sent signal r_respond nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:529 nua_signal() nua(0x7f65fc01dcd0): sent signal r_destroy nua_stack.c:573 nua_stack_signal() nua(0x7f65fc01dcd0): recv signal r_respond 200 OK nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f6608074730, ...) called tport.c:3257 tport_tsend() tport_tsend(0x7f6608004290) tpn = UDP/$${freedom2_trunk}:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = $${freedom2_trunk}:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f6608004290): not found by name UDP/$${freedom2_trunk}:5060 tport.c:3594 tport_vsend() tport_vsend(0x7f6608004290): 517 bytes of 517 to udp/$${freedom2_trunk}:5060 tport.c:3492 tport_send_msg() tport_vsend returned 517 send 517 bytes to udp/[$${freedom2_trunk}]:5060 at 11:50:43.534548: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP $${freedom2_trunk}:5060;branch=z9hG4bK707449280;rport=5060 From: <sip:${DID}@$${freedom2_trunk}>;tag=934960508 To: "${ANI}" <sip:${ANI}@$${freeswitch_ip}>;tag=FS9pD3eK574Sj Call-ID: 42db32d8-0220-1237-a8a1-02217f6f1b40 CSeq: 125455605 BYE User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180507T185611Z~43a9feb7f8~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ nta.c:6791 incoming_reply() nta: sent 200 OK for BYE (125455605) nta.c:1350 set_timeout() nta: timer set to 32000 ms nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f65fc01dcd0): removing session usage nua_session.c:4139 signal_call_state_change() nua(0x7f65fc01dcd0): call state changed: ready -> terminated nua_stack.c:271 nua_stack_event() nua(0x7f65fc01dcd0): event i_state 200 Session Terminated nua_stack.c:271 nua_stack_event() nua(0x7f65fc01dcd0): event i_terminated 200 Session Terminated soa.c:356 soa_destroy() soa_destroy(static::0x7f6608074730) called nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7f6608030530) nua_stack.c:569 nua_stack_signal() nua(0x7f65fc01dcd0): recv signal r_destroy nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil)) 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/${DID}@ $${freedom2_trunk}] 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/${DID}@$${freedom2_trunk}) State EXCHANGE_MEDIA going to sleep 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/${DID}@$${freedom2_trunk}) Running State Change CS_HANGUP (Cur 2 Tot 455) 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/${DID}@$${freedom2_trunk}) Callstate Change ACTIVE -> HANGUP 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/${DID}@$${freedom2_trunk}) State HANGUP 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] mod_sofia.c:438 Channel sofia/internal/${DID}@$${freedom2_trunk} hanging up, cause: NORMAL_CLEARING 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:60 sofia/internal/${DID}@$${freedom2_trunk} Standard HANGUP, cause: NORMAL_CLEARING 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/${DID}@$${freedom2_trunk}) State HANGUP going to sleep 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/${DID}@$${freedom2_trunk}) State Change CS_HANGUP -> CS_REPORTING 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/${DID}@$${freedom2_trunk}) Running State Change CS_REPORTING (Cur 2 Tot 455) 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/${DID}@$${freedom2_trunk}) State REPORTING 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:174 sofia/internal/${DID}@$${freedom2_trunk} Standard REPORTING, cause: NORMAL_CLEARING 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/${DID}@$${freedom2_trunk}) State REPORTING going to sleep 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/${DID}@$${freedom2_trunk}) State Change CS_REPORTING -> CS_DESTROY 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_session.c:1665 Session 455 (sofia/internal/${DID}@$${freedom2_trunk}) Locked, Waiting on external entities dea5af71-7a9d-4da0-b783-048dc5344a3d 2018-07-14 11:50:43.524096 [DEBUG] switch_ivr_bridge.c:706 sofia/internal/${DID}@$${freedom2_trunk} ending bridge by request from write function dea5af71-7a9d-4da0-b783-048dc5344a3d 2018-07-14 11:50:43.524096 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/external/${ANI}@ ${carrier_ip}] 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [NOTICE] switch_core_session.c:1683 Session 455 (sofia/internal/${DID}@$${freedom2_trunk}) Ended 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/${DID}@$${freedom2_trunk} [CS_DESTROY] dea5af71-7a9d-4da0-b783-048dc5344a3d EXECUTE sofia/external/${ANI}@${carrier_ip} log(ALERT FREEDOM2 CALL FAILURE, SENDING ${DID} TO q1_dr_mode) 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/${DID}@$${freedom2_trunk}) Running State Change CS_DESTROY (Cur 1 Tot 455) 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/${DID}@$${freedom2_trunk}) State DESTROY 3a6ef51f-e02e-42d3-ab91-34fcc7e5bd60 2018-07-14 11:50:43.524096 [DEBUG] mod_sofia.c:343 sofia/internal/${DID}@$${freedom2_trunk} SOFIA DESTROY dea5af71-7a9d-4da0-b783-048dc5344a3d 2018-07-14 11:50:43.524096 [ALERT] mod_dptools.c:1742 FREEDOM2 CALL FAILURE, SENDING ${DID} TO q1_dr_mode Royce Mitchell, IT Consultant ITAS Solutions royce3(a)itas-solutions.com

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FreeSWITCH Conference
by andretodd＠verizon.net 17 Jul '18

17 Jul '18

Hi is there a way to bridge a conference to another FreeSWITCH conference? Example FreeSWITCH Conference "A" has 2 people Then OpenSips puts the next person who calls onto FreeSWITCH Conference "B". now we need everyone together. I'm looking for a way to make the conference scalable to many boxes using OpenSips as the Load balancer and RTP Proxy. Any examples of what the dial string would look like would be very helpful Thanks Andre

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m=audio 0 RTP/AVP 19
by Victor Bogatyryev 17 Jul '18

17 Jul '18

Hello, I apologize for my bad English. Three years ago there was a message http://lists.freeswitch.org/pipermail/freeswitch-users/2015-June/113829.html and here I am faced with the same problem. A stupid provider sends crypto in RTP/AVP and to ensure processing this request, I inserted in the profile the parameter NDLB-allow-crypto-in-avp=true But Freeswitch answers to INVITE inserting in OK this line: m=audio 0 RTP/AVP 19 and the provider understands this as "no audio" and does not send ACK and does not send rtp. How to deal with this I do not know. Regards. Victor Bogatyryev

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Conference audio issues when Verto user sends video to audio-only conference
by Chad Phillips 13 Jul '18

13 Jul '18

I’ve got an audio-only conference configured, and some Verto clients (iOS to be specific) are also sending video to that conference. I had hoped that the video would simply be discarded, but instead I get this error, and audio doesn’t seem to be sent/received properly: [WARNING] switch_core_media.c:13944 verto.rtc/3521 has no video codec The only solution I’ve found is to configure the conference in question for video as well, but that’s not ideal. Wondering if this is a bug to report to Jira, or possibly there’s some config thing I can do to fix it? Because of iOS limitations, I’m not seeing how to have that client send audio only in my case.

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