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Anybody want to go in on some Spanish GM voices recordings?
by Chad Phillips 01 Feb '19

01 Feb '19

My company is just getting ready to do a round of Spanish voice prompts through GM voices. They have excellent talent, and charge reasonable rates per word/phrase, but their setup fees are a bit on the steep side. Anyone else with this same need want to join in our order to defer some of the setup costs? I imagine we can pay the per word/phrase costs individually, and split the setup fee. Also, FreeSWITCH core team: are you interested in having some Spanish prompts done? My company would cover the cost of some, if there weren't too many, as a service to the community... Chad

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Debian 9 FreeSWITCH 1.8 Sangoma Card A102
by Alaadin Abdurrahman 01 Feb '19

01 Feb '19

Dears, I have an issue when i try to compile and make freeswitch 1.8 with mod_freetdm to use sangoma card. please note that i followed every step in the sangoma wiki https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH and i still get this error message cc1: all warnings being treated as errors Makefile:1209: recipe for target 'ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo' failed make[2]: *** [ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo] Error 1 make[2]: Leaving directory '/usr/src/freeswitch/libs/freetdm' Makefile:15: recipe for target '../libfreetdm.la' failed make[1]: *** [../libfreetdm.la] Error 2 make[1]: Leaving directory '/usr/src/freeswitch/libs/freetdm/mod_freetdm' ../../../build/modmake.rules:89: recipe for target 'all' failed make: *** [all] Error 1 root@debian:/usr/src/freeswitch/libs/freetdm/mod_freetdm# best regards

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Sofia stops responding after a few days
by Bipin Patel 01 Feb '19

01 Feb '19

hi, I have 2 instances of FS running on a single windows box, first instance uses normal sip UDP profiles mainly used for routing calls to carriers and its running as a service since a few months without any issues. The second instance runs a sip TLS profile and accepts inbound registrations and forwards calls in sip UDP ahead, it uses xml_curl for directory users but the problem is every 2-3 days the inbound TLS profile stops responding to registrations, and when that happens i cant even stop and restart the service, have to kill it and start again. FS_CLI works but doesnt show any error, at first i though it could be the xml_curl causing the issue but later realized it never sends any requests when sofia stops responding. i have been banging my head from the past week or so but not able to find the cause, could any1 help in guiding me what to check when this happens so can find the root cause, im using commercial certs for TLS and when its running there r no issues other than sofia stops responding every few days, it happens at random times. -- Regards, Bipin ------------------------------------------------------------------------

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ASR Recognizer channel error
by Mansi Gupta 30 Jan '19

30 Jan '19

Hello Everyone, I'm having an issue working with mod_unimrcp.I am in the process of configuring a speech recognition IVR with the help of mod_unimrcp in freeswitch 1.6.This seems to work fine however, i keep getting the error of RECOGNIZER CHANNEL: [ERR] mod_unimrcp.c:1913 (ASR-12) RECOGNIZER channel error! Here are few relevant lines of the log: 2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:265 () Signal Message to [MRCP Client] [0x7fed34016200;1;0] 2019-01-30 04:28:20.072090 [DEBUG] apt_poller_task.c:251 () Wait for Messages [voxeo-prophecy8.0-mrcp1] 2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:337 () Process Message [MRCP Client] [0x7fed34016200;1;0] 2019-01-30 04:28:20.072090 [ERR] mod_unimrcp.c:1913 (ASR-13) RECOGNIZER channel error! 2019-01-30 04:28:20.072090 [DEBUG] mod_unimrcp.c:1577 (ASR-13) CLOSED ==> ERROR 2019-01-30 04:28:20.091506 [DEBUG] apt_consumer_task.c:141 () Wait for Messages [MRCP Client] 2019-01-30 04:28:20.091506 [DEBUG] mod_unimrcp.c:1061 (ASR-13) Terminating MRCP session 2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:265 () Signal Message to [MRCP Client] [0x7fed300afe10;4;0] 2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:337 () Process Message [MRCP Client] [0x7fed300afe10;4;0] 2019-01-30 04:28:20.091506 [INFO] mrcp_client_session.c:387 (ASR-13) Receive App Request ASR-13 <new> [1] 2019-01-30 04:28:20.091506 [DEBUG] mrcp_client_session.c:1283 (ASR-13) Dispatch App Request ASR-13 <new> [1] Any ideas will be appreciated. Thanks -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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zRTP + OPUS one-way audio issue
by Andrea Mazzeo 30 Jan '19

30 Jan '19

Hi, I'm facing an one-way audio issue with zRTP while using OPUS. zRTP is set in passthru mode. Called party can hear Calling party but not vice versa. Full log: https://pastebin.freeswitch.org/view/639adfe9 This issue is happening *only* with dynamic payloads codecs. Everything is working fine if I use G722 or G729. Scenario Leg-A (207) -> FS -> Leg-B (213) Checking FS logs, I see Leg-A's SDP: a=rtpmap:103 opus/48000/2 Leg-B's SDP: a=rtpmap:102 opus/48000/2 Actually having different payloads should not be an issue. I opened a case to Acrobits, they said the issue is in the actual RTP traffic sent to Leg-A from FS: >From the client log Leg-A Sending RTP packet #200 192.168.128.158:59176 > 31.102.111.134:28560, len=116, really=116, data=80679E55DF67E34A291167D899830CC03C124F353DA3C069574410EC0333125F This is using 103, correct as agreed on this side of the call Received RTP packet #200 31.102.111.134:28560 > 192.168.128.158:59176, len=90, data=80663D3B22CC8CB414490864A230F49B6EFF42E7BCA9EB64B864374B1377965C This is using 102, which is wrong for this side of the call. Seems that after negotiated payload 102 with Leg-B, FS is trying to use it on Leg-A, where it should be used 102 instead. Any idea? FreeSWITCH Version 1.8.1-2-4f54cff~64bit (-2-4f54cff 64bit) OS.: Debian 8.11 Linux pbx-186 3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64 GNU/Linux Thank you, Andrea Mazzeo

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Unimrcp recognizer channel error
by Mansi Gupta 30 Jan '19

30 Jan '19

Hello Everyone, I'm having an issue working with mod_unimrcp . I keep getting the error of RECOGNIZER CHANNEL ERROR!. Anyone having idea how to solve the issue...

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ASR recognizer channel error.
by Mansi Gupta 30 Jan '19

30 Jan '19

Hello everyone, I'm in the process of configuring a speech recognition IVR using freeswitch 1.6 with the help of mod_unimrcp.This seems to be work fine,however i keep getting this error: [ERR] mod_unimrcp.c:1913 (ASR-14) RECOGNIZER channel error! Here are few relevant lines of log: 2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:265 () Signal Message to [MRCP Client] [0x7fed3404b980;1;0] 2019-01-30 04:30:27.312074 [DEBUG] apt_poller_task.c:251 () Wait for Messages [voxeo-prophecy8.0-mrcp1] 2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:337 () Process Message [MRCP Client] [0x7fed3404b980;1;0] 2019-01-30 04:30:27.312074 [INFO] mrcp_client_session.c:151 (ASR-14) Receive Answer ASR-14 <new> [c:0 a:0 v:0] Status 0 2019-01-30 04:30:27.312074 [INFO] mrcp_client_session.c:455 (ASR-14) Raise App Response ASR-14 <new> [2] FAILURE [2] *2019-01-30 04:30:27.312074 [ERR] mod_unimrcp.c:1913 (ASR-14) RECOGNIZER channel error!* 2019-01-30 04:30:27.312074 [DEBUG] mod_unimrcp.c:1577 (ASR-14) CLOSED ==> ERROR 2019-01-30 04:30:27.312074 [DEBUG] apt_consumer_task.c:141 () Wait for Messages [MRCP Client] 2019-01-30 04:30:27.312074 [DEBUG] mod_unimrcp.c:1061 (ASR-14) Terminating MRCP session 2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:265 () Signal Message to [MRCP Client] [0x7fed30023ae0;4;0] 2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:337 () Process Message [MRCP Client] [0x7fed30023ae0;4;0] 2019-01-30 04:30:27.312074 [INFO] mrcp_client_session.c:387 (ASR-14) Receive App Request ASR-14 <new> [1] Any opinions will be appreciated. Thanks. -- Mansi Gupta

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when using goup_confirm_key=exec, the ringback tone is not heard.
by Julien Terrasson 28 Jan '19

28 Jan '19

Hi everyone, I'm working with a dialplan where i need the called party confirmation before establishing the call. So i'm using group_confirm_key=exec (because, not only i need to play a file to the caller, but also to the called party, so a little bit of lua scripting is needed). When doing this, the ringback tone is not heard anymore by the caller (when 180 Rining is received, nothing happen). This doesn't happen with then standard usage of group_confirm_key and group_confirm_file, for example group_confirm_key=#. group_confirm_file=hello.wav Any clue on how the RBT can be heard ? Best regards, J.Terrasson

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Re: [Freeswitch-users] when using goup_confirm_key=exec, the ringback tone is not heard. (Brian West)
by Julien Terrasson 25 Jan '19

25 Jan '19

Thank for the reply Brian but i found a workaround : I put A_party is in hold just before B_party is bridged. => "A_party" will listen music_on_hold when "B_party" is ringing. That way A_party knows that his call is still being processed (eventhough no RBT can be heard) Julien On Fri, Jan 18, 2019 at 1:00 PM < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: when using goup_confirm_key=exec, the ringback tone is > not heard. (Brian West) > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 17 Jan 2019 14:10:28 -0600 > Subject: Re: [Freeswitch-users] when using goup_confirm_key=exec, the > ringback tone is not heard. > Work up a test case and file a JIRA. > > /b > > > On Thu, Jan 17, 2019 at 9:45 AM Julien Terrasson < > julien.terrasson(a)gmail.com> wrote: > >> Hi everyone, >> >> I'm working with a dialplan where i need the called party confirmation >> before establishing the call. >> >> So i'm using group_confirm_key=exec (because, not only i need to play a >> file to the caller, but also to the called party, so a little bit of lua >> scripting is needed). >> When doing this, the ringback tone is not heard anymore by the caller >> (when 180 Rining is received, nothing happen). >> >> This doesn't happen with then standard usage of group_confirm_key and >> group_confirm_file, for example group_confirm_key=#. >> group_confirm_file=hello.wav >> >> Any clue on how the RBT can be heard ? >> >> Best regards, >> >> J.Terrasson >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales(a)freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales(a)freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> > > Email: brian(a)freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> > > [image: https://www.facebook.com/signalwireinc?src=email] > <https://www.facebook.com/freeswitch> [image: > https://twitter.com/freeswitch] <https://twitter.com/freeswitch> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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change ring- back after 180 from one of multiple destinations?
by Keith 23 Jan '19

23 Jan '19

Hi all. I'm trying to design a dialplan (in python) with some audible feedback for users. It occurred to me that it would be nice if I use instant_ringback and then could change the ring-back once one of multiple destinations sends a 180. I know that I can do it where if the remote sends 183 and ring-back, it will "take-over", but in this case, my callee won't generate ringback. An alternative would be to do it the other way around, I bridge to two destinations and one destination 1 hangs up with SUBSCRIBER_ABSENT for example, I could change the ring-back. I cannot however find anyway to do this in documentation or hints on forums etc. I supposed the basic question here is, Is there anyway to know what is happening, or indeed prepare for such events before calling bridge? Many thanks!! Keith.

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