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calling all native speakers of languages other than english. We need your help, and this will be easy.
by Michael Jerris 20 Jun '16

20 Jun '16

Just committed into tree is internationalization of Verto Communicator, so far, just English and Italian. If you speak another language, we need your help to translate these into other languages. The file that needs to be translated is: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/vert… <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/vert…> If you can assist with this, Please make a new file named locale-language.json (where "language" is the 2 letter code for your language) and translate all the text in quotes to the right thing for your language and send them back our way. Lets see how quickly we can crowd source this Thanks!

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How to playback background call process
by Tanim 20 Jun '16

20 Jun '16

Hi, I want to playback file background call process. So that in playing IVR time, call will process to send to gateways. Currently my dialplan like this: <action application="playback" data="/IVR/FILE" /> <action application="bridge" data="sofia/gateway/gw1/123456" /> In this way before call processing IVR is playing then start to process the call. But I want both in same time. Can you please help me, how can I do it. Thanks in advance. Tanim

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What field type supports microsecond epochs in SQLite CDR?
by vfclists 18 Jun '16

18 Jun '16

What field type in SQLite should be used to store the micro second epoch fields like answer_uepoch. Should they be bigint or double? -- Frank Church ======================= http://devblog.brahmancreations.com

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Anyone doing SMS for their end users?
by Kevin Long 18 Jun '16

18 Jun '16

I am curious if anyone on this list is doing SMS for their SIP end users, and if so, how. Which SIP providers are you using, how are you receiving and sending SMS to the outside world, and what SIP or chat client software are you using to facilitate this? Thank you, Kevin Long

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popup web api
by FERNANDO VILLARROEL 18 Jun '16

18 Jun '16

Dear All. I need your help in order to do the follow: If i have a softphone registered on FS, so i need that when the peer (softphone) answered a inbound call in the browser display a form popup. Somebody could help about how i cand do or what module i can use for do (mod_esl, mod_xml_curl....) or show me some example. I appreciate some idea or example please. Regards.

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ANN RoCKSwitch 0.993 with multiple inbound destinations
by Richard Mace 18 Jun '16

18 Jun '16

Hi all, Just a quick note to say that a new version of RoCKSwitch is available that now supports multiple inbound destinations. Would be really grateful for any feedback. Richard www.rocksoftware.co.uk

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CHANNEL_BRIDGE event is not coming when bridging two calls which are already in answered state
by Naveen Tamanam 17 Jun '16

17 Jun '16

Hi , CHANNEL_BRIDGE event is not coming when bridging two calls which are already in answered state. Am I missing any thing? Why EVENT CHANNEL_BRIDGE is not coming when bridging two calls which are in answered state. -- Thanks & Regards, Naveen Tamanam

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issue with uuid_set_media_stats showing zero packet counts
by Dave Horton 17 Jun '16

17 Jun '16

I am trying to get media stats on my call, using the following dialplan: <extension name="announcement"> <condition field="${sip_req_user}" expression="^annc$"/> <condition field="${sip_req_params}" expression="^play=(.*)$"> <action application="answer"/> <action application="set" data="api_result=${uuid_set_media_stats(${uuid})}"/> <action application="sleep" data="1000"/> <action application="playback" data="http_cache://http://wcs-icr.s3.amazonaws.com/$1"/> <action application="set" data="api_result=${uuid_dump(${uuid})"/> <action application="log" data="incoming bytes ${rtp_audio_in_raw_bytes}"/> <action application="log" data="incoming bytes ${rtp_audio_out_raw_bytes}"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> However, the data that is dumped shows packet counts of zero. The prompt is playing, so we are certainly sending packets out. Any ideas? Here is some of the output from the uuid_dump: variable_endpoint_disposition: ANSWER variable_rtp_audio_in_raw_bytes: 0 variable_rtp_audio_in_media_bytes: 0 variable_rtp_audio_in_packet_count: 0 variable_rtp_audio_in_media_packet_count: 0 variable_rtp_audio_in_skip_packet_count: 0 variable_rtp_audio_in_jitter_packet_count: 0 variable_rtp_audio_in_dtmf_packet_count: 0 variable_rtp_audio_in_cng_packet_count: 0 variable_rtp_audio_in_flush_packet_count: 0 variable_rtp_audio_in_largest_jb_size: 0 variable_rtp_audio_in_jitter_min_variance: 0.00 variable_rtp_audio_in_jitter_max_variance: 0.00 variable_rtp_audio_in_jitter_loss_rate: 0.00 variable_rtp_audio_in_jitter_burst_rate: 0.00 variable_rtp_audio_in_mean_interval: 0.00 variable_rtp_audio_in_flaw_total: 0 variable_rtp_audio_in_quality_percentage: 100.00 variable_rtp_audio_in_mos: 4.50 variable_rtp_audio_out_raw_bytes: 0 variable_rtp_audio_out_media_bytes: 0 variable_rtp_audio_out_packet_count: 0 variable_rtp_audio_out_media_packet_count: 0 variable_rtp_audio_out_skip_packet_count: 0 variable_rtp_audio_out_dtmf_packet_count: 0 variable_rtp_audio_out_cng_packet_count: 0 variable_rtp_audio_rtcp_packet_count: 0 variable_rtp_audio_rtcp_octet_count: 0 EXECUTE sofia/drachtio_mrf/5083084809(a)68.64.80.16 log(incoming bytes 0) 2016-06-17 10:59:28.757502 [DEBUG] mod_dptools.c:1713 bytes 0 EXECUTE sofia/drachtio_mrf/5083084809(a)68.64.80.16 log(incoming bytes 0) 2016-06-17 10:59:28.757502 [DEBUG] mod_dptools.c:1713 bytes 0

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3GPP Enhanced Voice Services (EVS) codec
by Colton Conor 17 Jun '16

17 Jun '16

Any chance that freeswitch will support the 3GPP Enhanced Voice Services (EVS) codec? This codec seems to be that the codec LTE carriers are using for enhance HD voice. According to this paper, this codec is even better than OPUS: http://networks.nokia.com/sites/default/files/document/nokia_3gpp_evs_codec…

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Freeswitch changes payload type of codec
by John Nash 17 Jun '16

17 Jun '16

I am facing same issue described in this old post http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/042823.html I am using media_bypass mode. If any more information like traces etc needed I can provide.

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