July 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recover queues after crash
by TTT 11 Jul '22

11 Jul '22

I have call recovery enabled, which is properly restoring active calls after a crash. But what about queues? Are they recovered too? Where is the information about call queues kept? Thanks!

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Recording calls causes FS generated RTP packets with different timestamp on missed packets
by Shaun Stokes 11 Jul '22

11 Jul '22

Hi All, We're having issues when recording calls, using either record_session or uuid_record. FreeSWITCH is generating RTP packets to fill in the gaps when there's missed RTP packets (packet loss), but the FreeSWITCH generated RTP packets use a different timestamp. This causes jitter and periods of silence on active calls. The only solution seems to be enabling RTP rewrite timestamps, but this causes audio delay\sync issues. [cid:cfdedbcf-5048-4f07-8fe1-1c761ffdfee6] [cid:768e2f51-801b-4411-aade-a77cecad6d45] We've tried with and with-out the following, but there's no change. * bridge_generate_comfort_noise * send_silence_when_idle * suppress_cng * suppress-cng (SIP profile param) Has anyone come across this before, and is there a solution? Thanks, Shaun

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xml_cdr problem using sip_from_user
by Jason Holden 04 Jul '22

04 Jul '22

All, Currently using fs 1.8.56 on a demo unit. I’ve noticed that on some CDRS the sip_from_user shows the dialed number that the extension placed when it should show the extension number. This is a speratic issue and I can’t narrow down why this may be happening. I am using a php script to post my CDRS. Does anyone have any suggestions? I do have set to record a / b legs of the call and to prefix the a leg. A snippit from the php script is below. $var = $cdr->variables; $sql = sprintf("INSERT INTO cdrClients (site_code,npa,call_direction,cac_call_type_id,call_uuid,direction,user,domain,from_user,from_host,to_user,to_host,req_user,req_host,bridge_channel,sip_term_status,sip_term_cause,sip_hangup_disposition,bridge_hangup_cause,hangup_cause,hangup_cause_q850,start_stamp,answer_stamp,end_stamp,duration,billsec) VALUES('%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s','%s');", $var->site,$var->NPA,$var->call_direction,$var->cac_call_type_id,$var->call_uuid,$var->direction,$var->user_name,$var->domain_name,$var->sip_from_user,$var->sip_from_host,$var->sip_to_user,$var->sip_to_host,$var->sip_req_user,$var->sip_req_host,rawurldecode($var->bridge_channel),$var->sip_term_status,$var->sip_term_cause,$var->sip_hangup_disposition,$var->bridge_hangup_cause,$var->hangup_cause,$var->hangup_cause_q850,rawurldecode($var->start_stamp),rawurldecode($var->answer_stamp),rawurldecode($var->end_stamp),$var->duration,$var->billsec); Jason Holden Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.com <http://www.thinksimplicity.com>

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Trigger playback via network request
by David P 02 Jul '22

02 Jul '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.

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Haven't received a msg in weeks
by David P 01 Jul '22

01 Jul '22

Testing after fixing subscription weeks ago.

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