- FreeSWITCH-users - FreeSWITCH Mailing Lists

ping
by antonio 09 Feb '18

09 Feb '18

hi, just testing... i reply to some mails and i don't see my answer. Also put some question and it seams is not arriving.. -- Saludos / Regards / Cumprimentos Anónio Silva

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T38 re-invite, 488 Not Acceptable Here, Freeswitch 1.6.20
by Michael Avers 08 Feb '18

08 Feb '18

Hello, I understand the T38 behavior changed after 1.6.13. I watched Brian's video from last year, read the docs, but I just don't see what else I would need for a very simple scenario for receiving a fax with the receiving end (ATA) re-inviting T38. PSTN > FS > HT801 ATA > T38 RE-INVITE > FS > PSTN The ATA does re-invite, however FS rejects immediately with 488 Not Acceptable Here. I'm using the following dialplan. I also tried to use export instead of set, but same result. <extension name="faxdid1" continue="false"> <condition field="destination_number" expression="^(4154447777)$"> <action application="set" data="ignore_early_media=true"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="bridge" data="user/6050(a)sip.domain.org"/> </condition> </extension> Is there anything obvious that I'm missing here? FreeSWITCH (Version 1.6.20 -37-987c9b9 64bit) Thanks Mike

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freeswitch odbc in the core and RDS
by David Villasmil 08 Feb '18

08 Feb '18

Hello guys, trying to use RDS as a backend for freeswitch core, but when creating the tables, RDS comes back with: [STATE: HY000 CODE 1118 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.37-log]Row size too large. The maximum row size for the used table type, not counting BLOBs, is 65535. This includes storage overhead, check the manual. You have to change some columns to TEXT or BLOBs for some tables, thus messing up everything Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Stream inbound Freeswitch (RTMP?)
by Scott Howell 07 Feb '18

07 Feb '18

How would I stream audio into Freeswitch for callers to dial in and listen. I suspect RTMP, but I'm uncertain of the configuration, particularly in My FusionBox Setup, although I can configure the underlying freeswitch if necessary. I have an audio/video stream that I already set up, I would like to use the audio portion as a "conference" that users can dial into and listen live. Thank you, -Scott

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Missing git tag for v1.6.20
by Steven Ayre 06 Feb '18

06 Feb '18

I've just noticed the release announcement for v1.6.20 I can see the tarball linked to from the confluence blog, but it looks like that version hasn't been tagged in git checking both stash and my local clone? Can that be corrected for those of us that build from git please?

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is this even possible?
by David Villasmil 06 Feb '18

06 Feb '18

Hello guys, So i have a conference running on a fs... and i want to move it to another fs... is this even possible?? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Long distance calling in the US
by Preston Hagar 06 Feb '18

06 Feb '18

Hello all, We are setting up FreeSwitch to use FreeTDM through a Sangoma A102DE, which is connected to a PRI. We are located in Texas in the US, where we are required to call the full 10 digits (area code and number) for "local" numbers and then dial 1 followed by the full 10 digits for calls that are long distance. Currently, our dialplan is working as expected to make local calls:  <extension name="tdm-outgoing"> <condition field="destination_number" expression="^9?(1{0,1}[2-9]\d\d[2-9]\d{6})$"> <action application="log" data="WARNING outbound call through TDM to $1"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="call_direction=outbound"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="freetdm/1/a/$1"/> </condition> </extension> When we dial a number with a 1 in front of it, the 1 is still included in the connect lines and all the output I can find from the logs or fs_cli. For example: 2018-02-06 12:52:28.376794 [DEBUG] mod_freetdm.c:1343 Connect outbound channel FreeTDM/1:1/17138370311 but we receive an automated message from our local PRI provider saying that we need to dial 1 in front of the number to dial long distance. I called into their support and they claim that "everything is normal" and as a related note, up until about 2 days ago, this PRI was working properly calling long distance on our old Mitel system. I'm a bit of a loss as to where to check next. Any ideas on either what might be pulling out the 1 when delivering it to the PRI or if there is some secret extra code the PRI is expecting for long distance I'm not aware of? Thanks for the help!

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rtp-timer-name / timer issues
by Colin Morelli 06 Feb '18

06 Feb '18

Hey list, I'm running FS on EC2 (I know, I know). Having some issues with random packet loss, which I believe almost certainly I have narrowed down to timer issues and/or network latency/jitter (seems surprising since I'm using c5.xlarge instances). Behavior is that, during a call, brief pauses or notable audio loss will occur. This is on high bandwidth links that are otherwise stable. Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and eventually "auto-flush catching up 1 packet(s)" in rapid succession (usually going through the cycle 4-5 times) before things settle again. Obviously that means a minimum of 4-5 audio packets were dropped within the span of a second which results in considerable audio artifacting. Changing rtp-timer-name to none, which I understand to perform synchronous reads of RTP audio (as opposed to timer-based async reads) makes the audio notably smoother. That said, I'm having a hard time uncovering the consequences of doing this. Obviously I understand that reads will block the RTP thread, but I can't seem to understand the potential ramifications of this. Could anyone help clarify? My other question is: assuming "timer while hot" indicates what I believe it does (that when the timer hit there was >1 packet in the queue to be read), couldn't this issue also just be caused by network jitter, and not necessarily just timer inconsistencies? Thanks in advance. Best, Colin

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[Lumenvox][mod_unimrcp][Issue]
by Bilal Abbasi 06 Feb '18

06 Feb '18

Hi Users, I am using FreeSWITCH mod_unimrcp to detect VM using the lumenvox, Below is my configuration, <action application="set" data="play_and_detect_speech_close_asr=true"/> <action application="play_and_detect_speech" data="${sound_file_string} detect:unimrcp:lumenvox {start-input-timers=false,no-input-timeout=5000,recognition-timeout=5000}builtin:special/amd" /> everything works really good, but under testing i stopped the lumenvox SIP server(called lvmediaserverd server), and when i tried to make a call i was not able to hear the message that needs to be played. I need to know any parameter/configuration that can play the sound file even the lumenvox is down in uni_mrcp(as i don't want to get the service down if lumenvox server is down, max loss i should get is the VM detection that will surely stop working, but i want my messages to be delivered to live users). Regards Abbasi

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mod_soundtouch
by Mike Loiterman 05 Feb '18

05 Feb '18

I’m trying to use mod_sound touch, but the audio doesn’t appear to be processing correctly. The module is loaded, but something isn’t right because all I get on the B Leg is an almost imperceptible bit of static when the A Leg is speaking. Dialplan: <extension name="Soundtouch"> <condition field="destination_number" expression="^(\d+)$"> <action application="soundtouch" data="hook_dtmf -0.3o 0.8r"/> <action application="bridge" data="sofia/gateway/gateway1/$1"/> </condition> </extension> Versions: Linux 3.16.0-5-amd64 #1 SMP Debian 3.16.51-3+deb8u1 (2018-01-08) x86_64 GNU/Linux FreeSWITCH Version 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit) freeswitch-mod-soundtouch 1.6.20~37~987c9b9-1~jessie+1 amd64 I’ve emailed the developer at max(a)evolux.net.br, but the email was just bounced back saying the address doesn’t exist. Seems like this module is dead? Appreciate any input. ------------------------------ Mike Loiterman Cell: 630-302-4944 Email: mike(a)ascendency.net

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