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Trigger playback via network?
by David P 18 Jun '22

18 Jun '22

I would like to trigger playback into a channel, and I'd like to do that using a networked request that indicates the filename. Is there a better way to do it than the following? I'm considering using ESL (in inbound mode). I already have a large dialplan, and rather than reimplement all of it as ESL commands, I thought I could add the uuid_park application into the dialplan at the spot where the inbound channel(s) should be fully configured and active. There would be a separate server running an ESL client, and once it gets an ESL CHANNEL_ANSWER event it would be able to invoke playback until it receives a CHANNEL_DESTROY event. I considered other programmatic ways of doing playback such as mod_python, but they don't address the networked requirement. David

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Disable SOME METHODS in Freeswitch ALLOW header
by Patrick Karton 16 Jun '22

16 Jun '22

Hello, I want to disable/remove some Methods From ALLOW header and freeswitch must generate 405 if it receives a request with unsupported methods. how do we achieve that is freeswitch . Thanks.

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Call center scalability
by Peiffer Eric 13 Jun '22

13 Jun '22

Hi, I'm new in freeswitch and I'm in charge to study the developmnet of a call center using freeswitch. I have read some books and documents about freeswitch scalability, and I have also read the document : "Giovanni_Maruzzelli-OpenSIPS_Summit2016-FreeSWITCH_HA.pdf", here in the link: https://opensips.org/pub/events/2016-05-10_OpenSIPS-Summit_Amsterdam/Giovan… FreeSWITCH High Availability and Scaling - openSIPS<https://opensips.org/pub/events/2016-05-10_OpenSIPS-Summit_Amsterdam/Giovan…> 12/36 OpenSIPS Summit 2016 - Amsterdam gmaruzz(a)OpenTelecom.IT Many FreeSWITCHes as ONE ALL FreeSWITCH Boxes are ACTIVE (and most other boxes are too) – HA Database: Keeps its own State – Distributed FileSystem: Has its own Configuration Writes and Reads Voice Mails – HA Load Balancers and Proxies: Manages NAT Handling (RTP Media and SIP Signaling) opensips.org  In page 32 it is wrote that call center is a special case and the answer is partitioning. Can you tell me a little more about this solution? Regards, Eric

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REFER handling problem
by Thomas de Rooij | CM.com 13 Jun '22

13 Jun '22

Hi all, I'm having trouble with Freeswitch handing a REFER scenario. The following flow is from an actual call where 10.10.10.10 is a Freeswitch machine that should handle the REFER message: 222.22.222.222:5060 10.10.10.10:5060 10.10.10.11:5060 10.10.10.12:5060 ──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬───────── │ INVITE (SDP) │ │ │ │ ──────────────────────────> │ │ │ +0.001396 │ 100 Trying │ │ │ │ <────────────────────────── │ │ │ +0.000845 │ 180 Ringing │ │ │ │ <────────────────────────── │ │ │ +0.003244 │ │ INVITE (SDP) │ │ │ │ ──────────────────────────> │ │ +0.005757 │ │ 100 Trying │ │ │ │ <────────────────────────── │ │ +0.115697 │ │ 200 OK (SDP) │ │ │ │ <────────────────────────── │ │ +0.020056 │ 200 OK (SDP) │ │ │ │ <────────────────────────── │ │ │ +0.095132 │ ACK │ │ │ │ ──────────────────────────> │ │ │ +0.000696 │ │ ACK │ │ │ │ ──────────────────────────> │ │ +1.901472 │ │ REFER │ │ │ │ <────────────────────────── │ │ +0.000528 │ │ 202 Accepted │ │ │ │ ──────────────────────────> │ │ +0.000105 │ │ NOTIFY │ │ │ │ ──────────────────────────> │ │ +0.000198 │ INVITE (SDP) │ │ │ │ <────────────────────────── │ │ │ +0.003230 │ │ 200 OK │ │ │ │ <────────────────────────── │ │ +0.000296 │ │ BYE │ │ │ │ ──────────────────────────> │ │ +0.001659 │ 100 Trying │ │ │ │ ──────────────────────────> │ │ │ +0.019175 │ 200 OK (SDP) │ │ │ │ ──────────────────────────> │ │ │ +0.002738 │ ACK │ │ │ │ <────────────────────────── │ │ │ +0.000287 │ INVITE (SDP) │ │ │ │ <────────────────────────── │ │ │ +0.005444 │ 100 Trying │ │ │ │ ──────────────────────────> │ │ │ +0.021372 │ │ INVITE (SDP) │ │ │ ────────────────────────────────────────────────────────> │ +0.007074 │ │ 100 Trying │ │ │ <──────────────────────────────────────────────────────── │ +0.006919 │ │ 200 OK (SDP) │ │ │ <──────────────────────────────────────────────────────── │ +0.036307 │ 200 OK (SDP) │ │ │ │ ──────────────────────────> │ │ │ +0.005278 │ ACK │ │ │ │ <────────────────────────── │ │ │ +0.179349 │ INVITE │ │ │ │ <────────────────────────── │ │ │ +0.003849 │ 100 Trying │ │ │ │ ──────────────────────────> │ │ │ +0.002510 │ 200 OK (SDP) │ │ │ │ ──────────────────────────> │ │ │ +0.007616 │ │ 200 OK │ │ │ │ <────────────────────────── │ │ +0.265244 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.227210 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +0.772969 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.227103 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +1.772828 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.228480 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +3.770628 │ │ 200 OK (SDP) │ │ │ <──────────────────────────────────────────────────────── │ +0.228179 │ 200 OK (SDP) │ │ │ │ ──────────────────────────> │ │ │ +3.772980 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.227158 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +3.773918 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.225763 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +3.775424 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.224982 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +3.776155 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.223281 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +3.776986 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.223503 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +3.777505 │ │ 200 OK (SDP) │ │ │ <<<────────────────────────────────────────────────────── │ +0.223478 │ 200 OK (SDP) │ │ │ │ ────────────────────────>>> │ │ │ +0.271627 │ │ BYE │ │ │ <──────────────────────────────────────────────────────── │ +0.000958 │ │ 200 OK │ │ │ ────────────────────────────────────────────────────────> │ +0.000117 │ │ ACK │ │ │ ────────────────────────────────────────────────────────> │ +0.020239 │ BYE │ │ │ │ <────────────────────────── │ │ │ +0.008562 │ 200 OK │ │ │ │ ──────────────────────────> │ │ │ +0.000251 │ ACK │ │ │ │ <────────────────────────── │ │ │ As you can see the far end sends a BYE because of an ACK timeout. The thing is, I can see in the Freeswitch debug logs that the 200 OK, from both legs, is received by Freeswitch (logs after the last 200 OK from the A leg): tport.c:3052 tport_deliver() tport_deliver(0x7f17c00043d0): msg 0x7f17c0074950 (840 bytes) from udp/222.22.222.222:5060/sip next=(nil) nta.c:3370 agent_recv_response() nta: received 200 OK for INVITE (52572546) nta.c:3437 agent_recv_response() nta: 200 OK is going to a transaction nta.c:9716 outgoing_duplicate() nta: 200 OK is duplicate response to 52572546 INVITE nta.c:9723 outgoing_duplicate() Via: SIP/2.0/UDP 10.10.10.10 ;branch=z9hG4bKprBXUe0Xyc7eH nta.c:7236 _nta_incoming_timer() nta: timer J fired, terminate 202 response nta.c:5896 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7f17e93b3c10) nta.c:7265 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/9 term, 1/9 free nta.c:1308 agent_timer() nta: timer set next to 20 ms nta.c:7236 _nta_incoming_timer() nta: timer J fired, terminate 200 response nta.c:5896 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7f17e93b3c10) nta.c:7265 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/8 term, 1/8 free nta.c:1308 agent_timer() nta: timer set next to 5 ms nta.c:9221 outgoing_timer_dk() nta: timer D fired, terminate INVITE (52572544) nta.c:8893 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7f17e93b3d00) nta.c:9040 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/2 tout, 1/4 term, 1/6 free nta.c:1308 agent_timer() nta: timer set next to 2 ms nta.c:9102 outgoing_timer_bf() nta: timer F fired, terminating ACK (52572544) nta.c:8893 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7f17e93b3d00) nta.c:9040 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/2 tout, 0/3 term, 1/5 free nta.c:1308 agent_timer() nta: timer set next to 41 ms In the end, after the 200 OK on the BYE transaction, the ACK that should follow the 200 OK for the INVITE transaction is sent. nta.c:6868 incoming_reply() nta: sent 200 OK for BYE (52572561) nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f17c0019a70): removing session usage soa.c:1730 soa_activate() soa_activate(static::0x7f17c000e670, (nil)) called nta.c:2707 nta_tpn_by_url() nta: selecting scheme sip tport.c:3286 tport_tsend() tport_tsend(0x7f17c00043d0) tpn = */10.10.10.12:5060 tport.c:4075 tport_resolve() tport_resolve addrinfo = 10.10.10.12:5060 tport.c:4709 tport_by_addrinfo() tport_by_addrinfo(0x7f17c00043d0): not found by name */10.10.10.12:5060 tport.c:3623 tport_vsend() tport_vsend(0x7f17c00043d0): 498 bytes of 498 to udp/10.10.10.12:5060 tport.c:3521 tport_send_msg() tport_vsend returned 498 2022-06-03 12:38:26.009764 [DEBUG] switch_core_state_machine.c:848 (sofia/Default/2398) Callstate Change ACTIVE -> HANGUP send 498 bytes to udp/[10.10.10.12]:5060 at 12:38:26.027864: ------------------------------------------------------------------------ ACK sip:2398@10.10.10.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10;rport;branch=z9hG4bKQ14NX9g1UNX1c Route: <sip:10.10.10.12;lr=on;ftag=X8gN4t41XD8Dc;did=56a.da01> Max-Forwards: 70 From: "31761234567" <sip:31761234567@10.10.10.10>;tag=X8gN4t41XD8Dc To: <sip:2398@10.10.10.12:5060>;tag=F6g6a1UDjmBXB Call-ID: 57ac2def-8718-4ea9-a70c-74c4acc3ab49 CSeq: 52572545 ACK Contact: <sip:gw+fs@10.10.10.10:5060;transport=udp;gw=fs> Content-Length: 0 nta.c:8390 outgoing_send() nta: sent ACK (52572545) to */10.10.10.12:5060 nta.c:8816 outgoing_free() nta: outgoing_free(0x7f17c0016ef0) nua_stack.c:301 nua_stack_event() nua(0x7f17c0019a70): event r_invite 481 Call/Transaction Does Not Exist nua_session.c:4140 signal_call_state_change() nua(0x7f17c0019a70): call state changed: completing -> terminated nua_stack.c:301 nua_stack_event() nua(0x7f17c0019a70): event i_state 200 Early Session Terminated nua_stack.c:301 nua_stack_event() nua(0x7f17c0019a70): event i_terminated 200 Early Session Terminated soa.c:356 soa_destroy() soa_destroy(static::0x7f17c000e670) called nta.c:4541 nta_leg_destroy() nta_leg_destroy(0x7f17c0021720) nua_stack.c:599 nua_stack_signal() nua(0x7f17c0019a70): recv signal r_destroy nta.c:4541 nta_leg_destroy() nta_leg_destroy((nil)) 2022-06-03 12:38:26.009764 [DEBUG] switch_core_state_machine.c:850 (sofia/Default/2398) State HANGUP From the logs I can't figure out why the ACK is only sent AFTER the BYE and not immediately after the first 200 OK. Any help would be greatly appreciated. Kind regard, Thomas

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Sharing session between 2 freeswitch
by Patrick Karton 12 Jun '22

12 Jun '22

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Re: [Freeswitch-users] debian repos for Raspberry Pi broken?
by Mohammad-Reza Tazari 10 Jun '22

10 Jun '22

Dear Andrey, any news here? Thanks and kind regards, -- Saied Andrey Volk wrote on 21-Dec-2021 21:36: > > Let me see what I can do. May take some time. > > вт, 21 дек. 2021 г. в 23:19, Johan Helsingius <julf at julf.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>: > > > >* Hi, *> >> >* Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), *> >* my /etc/apt/sources.list.d/freeswitch.list has: *> >> >* deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ <http://files.freeswitch.org/repo/deb/rpi/debian-release/> bullseye main *> >* deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ <http://files.freeswitch.org/repo/deb/rpi/debian-release/> bullseye *> >* main *> >> >* but when I try *> >> >* apt-get update && apt-get install -y freeswitch-meta-all *> >> >* I get *> >> >* Unable to locate package freeswitch-meta-all *> >> >* apt-cache search freeswitch only gives me: *> >* libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files *> >* libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries *> >* libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols *> >* sipwitch - secure peer-to-peer VoIP server for the SIP protocol *> >* sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface *> >> >* Any suggestions? *> >> >* Julf *

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Mod_callcenter scalability
by Peiffer Eric 08 Jun '22

08 Jun '22

Hi, I'm new in freeswitch world, My goals is to study a call center system using freeswitch. System must be scalable in order to add new freeswitch instance. To do that I have created 2 freeswitch instances loadbalanced by opensips. The 2 freeswitch instances used the same postgres datasbase in order to shared configuration data. Xml files for directory, dialplan, config, etc .. are also the same. So I have one agent associated to one queue configured in the shared postgres database. When user call the callcenter if the agent is free then the call is offered to this agent. If a second user call occurs (call handled by the other fresswitch instance) this call is queued. So every think is working. Then I do not understand why call center is not scalable. If there is issue with callcenter scalablity please could you tell me what? and what would be the solution? Regards. Eric Peiffer

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force SIP INFO
by David Villasmil 06 Jun '22

06 Jun '22

Anyway to force SIP INFO on the outbound side even if telephone event is present? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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freeswitch logs don't want to write when call setVariable() and Play TTS prompt
by Rajnikant Vanza 06 Jun '22

06 Jun '22

Hi Users, We have one scenario, in which we don't want to write any logs when calling set variable function during active call and playing TTS prompt (Refer highlighted string) 86e386f6-ff4c-4fa7-9b53-52836aa8e150 2022-06-06 06:53:19.530899 98.77% [DEBUG] switch_cpp.cpp:773 CoreSession::*setVariable(customerId, 123654789258)* ----------- 2022-06-06 07:09:06.830902 99.03% [INFO] mrcp_client_connection.c:530 (TTS-6) Send MRCPv2 Data 172.28.230.145:41052 <-> 172.28.199.43:20032 [304 bytes] MRCP/2.0 304 SPEAK 1 Channel-Identifier: 8de38aa0-6053-4b00-9@speechsynth Content-Type: text/plain Speech-Language: en-us Voice-Name: Jackie Vendor-Specific-Parameters: tts-engine=unimrcp:lumenvox;tts-voice=Jackie Content-Length: 62 *<say-as interpret-as="vxml:digits"> 147258369456 </say-as>* 2022-06-06 07:09:06.910901 99.03% [INFO] mrcp_client_connection.c:635 () Receive MRCPv2 Data 172.28.230.145:41052 <-> 172.28.199.43:20032 [125 bytes] Thank you in advance. Best Regards, Rajnikant Vanza

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Extending mod_esf
by Support from NetworkedAudio LLC 02 Jun '22

02 Jun '22

I’m looking to do multicast send and receive as a module, I’ve looked at mod_esf and also using gstreamer but a mod_portaudio style module would really help. Looking through this list, there have been a few mentions, and a mod_rtp_stream that was begun – I have time and money to put into this now, does anyone know of any projects that already add multicast RTP send and receive capabilities to FreeSWITCH rather than starting fresh?

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