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[ERR] switch_ivr_originate.c:224 sofia/sipinterface_1/ Error Playing File!
by John.Nowlin＠tresta.com 21 Dec '16

21 Dec '16

FreeSWITCH Version 1.4.26~64bit ( 64bit) We receive this error, "[ERR] switch_ivr_originate.c:224 sofia/sipinterface_1/ Error Playing File!" whenever the leg receiving the call hangs up before the extant media can start playing. After looking through switch_ivr_originate.c, I tried setting the global variables playback_timeout_sec=10 and playback_timeout_as_success=true but did not see a change in behavior. status = switch_ivr_read(collect->session, (uint32_t)len, (uint32_t)len, collect->file, NULL, buf, sizeof(buf), collect->confirm_timeout, NULL, 0); if (status != SWITCH_STATUS_SUCCESS && status != SWITCH_STATUS_BREAK && status != SWITCH_STATUS_TOO_SMALL) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(collect->session), SWITCH_LOG_ERROR, "%s Error Playing File!\n", switch_channel_get_name(channel)); switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER); } Is there some other setting that can inform switch_ivr_originate to not worry about having not played the media after the recipient has hung up? john nowlin Senior Software Developer Tresta office: 850.383.7756 mobile: 850.294.5635 www.tresta.com - A Better Phone Experience

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enable-3pcc=proxy behavior
by Sharath Kumar 20 Dec '16

20 Dec '16

Hello, I am using FS as an endpoint. I need some clarification the below scenario - When I receive an INV w/o SDP and enable-3pcc=proxy, the FS sends out a 18x and then when I answer the call it indicates in the logs that it is sending a 200 OK i.e[3PCC-PROXY, Sent a 200 OK, waiting for ACK] but the call fails with "SIP/2.0 500 No session set by user". Should I be setting the local-sdp somewhere from my application? I know this is not a very common usage of FS. I would've tried this with the FS client to confirm the behavior but I don't think it supports 3pcc=pxoxy only 3pcc=true. Thank you, Shaks

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Problem using FSClient and FreeSWITCH
by Paul Mateer 20 Dec '16

20 Dec '16

I’m doing a little experimenting with FreeSWITCH and I've encountered a problem which hopefully someone can help with. I have a system running a generic build/install of FreeSWITCH and another with the FSClient running. I am trying to dial some of the numbers that should respond, such as 9664 or 9198, but when I try the call connects to the server but is then disconnected at the client end. Debugging the client I've found that there are two threads executing the audio_bridge_thread function. Both have the same two channels but they're reversed between threads. The problem seems to be that one channel has no media_handle assigned so when switch_core_session_write_frame is called to write the sound data that should be played by the client, the function returns a failure code and the client hangs up. Does anyone have any idea why one channel would not have had a media handle assigned – I can call the same numbers from X-Lite on the same client PC and the call connects successfully and the audio stream is played ok.

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Bridged calls randomly interrupted by MOH
by Adam Raszynski 20 Dec '16

20 Dec '16

Hi all I have some strange problem with my FS setup. During bridged call sometimes callers hear MOH music for a moment (instead of each other) and than call is continued. It happens completely randomly and without additional action (ie. no DTMF). It happens both on stable 1.6.x version and on legacy system based on 1.2.x I use call queue position announcement hack described in wiki - executing in dialplan <action application="set" data="x=${expand(luarun fifo_with_position.lua)}"/> and <action application="fifo" data="my_fifo_name in tone_stream://%(1000,100,440,480) local_stream://moh/8000"/> - lua script that runs in loop, uuid_broadcasts position in queue, uuid_bridges callers when agent answers call Is this problem with random call interruption by MOH related to any known bug? or maybe there is currently some standard method of announce position in queue without background execution hack?

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gsmopen with E1762 ?
by Jeff Chua 20 Dec '16

20 Dec '16

Before I go out to purchase the E1762 for gsmopen, just want to check to see if there is a better alternative? Is there any update to the list from https://wiki.freeswitch.org/wiki/Gsmopen ? Thanks, Jeff

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To get rid of P-Asserted-Identity ?
by David Ponzone 20 Dec '16

20 Dec '16

Hello all, when FS receives an inbound call with a PAID header, it uses this number to fill the caller_id_number channel variable. I can’t manage to alter this behavior for inbound legs. At least, I’d like to tell FS to ignore the PAID because it’s wrong, and to use the SIP From for the caller_id_number. Ideas ? Am I missing something obvious ? David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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mod_callcenter and agents disconnected
by Victor Medina 20 Dec '16

20 Dec '16

Hello guys! I have been working with mod callcenter today and noted something. If logged as an agent and shutdown the softphone, FS shows the extension when off, but keep the agent as Available, is there any way to change agent status when a registration dies? freeswitch@internal> callcenter_config agent list name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time operador127@soporte|single_box||callback|[call_timeout=10]user/127@ ${domain_name}|Available|Waiting|20|10|10|40|0|0|0|0|1482186051|0|0|0|0 operador128@soporte|single_box||callback|[call_timeout=10]user/128@ ${domain_name}|Available|Waiting|20|10|10|40|0|0|0|0|1482186045|0|0|0|0 operador126@soporte|single_box||callback|[call_timeout=10]user/126@${domain_name}|Logged Out|Waiting|20|10|10|40|0|0|0|0|1482185645|0|0|0|0 operador129@soporte|single_box||callback|[call_timeout=10]user/129@${domain_name}|Logged Out|Waiting|20|10|10|40|0|0|0|0|1482185645|0|0|0|0 operador148@soporte|single_box||callback|[call_timeout=10]user/148@${domain_name}|Logged Out|Waiting|20|10|10|40|0|0|0|0|1482185645|0|0|0|0 operador149@soporte|single_box||callback|[call_timeout=10]user/149@${domain_name}|Logged Out|Waiting|20|10|10|40|0|0|0|0|1482185645|0|0|0|0 operador999@soporte|single_box||callback|[call_timeout=10]user/999@${domain_name}|Logged Out|Waiting|5|10|10|40|0|0|0|0|1482185645|0|0|0|0 +OK Sin mas a que hacer referencia, Victor Medina

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oubound proxy
by Slava Bendersky 20 Dec '16

20 Dec '16

Hello Everyone, I have issue where freeswitch use received part and not uri of proxy for saved extension contact. This cause send call to wrong direction instead of to edge proxy. freeswitch(a)casip00.myorg.prod> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 137861_rel44NDdhMTljNzNmYTJmZTY3OTVlNmQzYmIwNmYxMzI5NzI User: 45241(a)domain.net Contact: "volga629" <sip:45241@client_public_ip:45950;rinstance=5d29fcadab1edf67;transport=udp;fs_path=sip%3A10.18.130.27%3Blr%3Breceived%3Dsip%3Aclient_public_ip%3A45950%3Btransport%3Dtls> Agent: Bria Android 3.7.5 build 92486 Status: Registered(UDP)(unknown) EXP(2016-12-18 09:43:24) EXPSECS(745) Ping-Status: Reachable Ping-Time: 0.00 Host: casip01.myorg.prod IP: 10.18.130.27 Port: 5060 Auth-User: 45241 Auth-Realm: domain.net MWI-Account: 45241(a)domain.net Total items returned: 2 ================================================================================================= freeswitch(a)casip00.myorg.prod> here are failing call log. https://paste.fedoraproject.org/508859/48206989/ Any help thank you. volga629

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Reloading mod_verto certificates
by Tom Parrott 20 Dec '16

20 Dec '16

Hi, With the rise of LetsEncrypt and short lived automated certificates being allocated, I wondered what is the best approach for reloading a Verto encryption certificate without taking down the Freeswitch server or verto module? Thanks Tom

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How do I get dialplans from database?
by david william 20 Dec '16

20 Dec '16

Can Freeswitch get dialplans from database without using *mod_xml_curl? I just played FusionPBX for days and I got some impression that it just uses a pure database for configuration and dialplans. Am I right? I like to save users and dialplans in database. What is the best approach for it. I read through two books about Freeswitch. * *Thanks,* *David*

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