February 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch As SBC for sip_phones
by Mehdi Shirazi 19 Feb '19

19 Feb '19

Hi Thanks for suggestions, If we want to comply strictly with rfc 5853 in our SBC so we should have a B2BUA also int it. What is your recommendation? 1-Using Kamailio/Opensips for forwarding REGISTER/SUBSCRIBE/NOTIFY to external Registrar and using a Freeswitch for media+B2BUA ? 2-Using Kamailio/Opensips + rtpengine and just using Freeswitch as B2BUA in media bypass mode Regards. M.shirazi

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Hold BUG
by Nathan Stratton 19 Feb '19

19 Feb '19

Larry calls Bob Bob puts Larry on hold this sends UPDATE to FreeSWITCH with setup:actpass media attribute 200 OK from FreeSWITCH responds correctly with setup:active everything works correctly Larry calls Bob Larry puts Bob on hold this send UPDATE to FreeSWITCH with setup:actpass media attribute 200 OK from FreeSWITCH responds incorrectly with setup:actpass this causes chrome to kill the call. The client is lastest JsSIP ><> nathan stratton

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Called id name is not displayed - Cisco 7912G devices
by Panagiotis Voutskidis 18 Feb '19

18 Feb '19

I run a freeswitch pbx with about 10 cisco 7912G (with SIP firmware) and 2 linksys devices. When I call a local device from a cisco ip phone, we don't see the called device id name, during ringing and talking. The only thing displayed is the dialed number. Linksys devices display called user name as expected. Is it a problem with the specific device and firmware or there is something I could do to fix it? Is there anyone else using cisco 7912G phones that can help with the situation? Regards, Panos

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How to find mistake in FS config?
by Yu Boot 15 Feb '19

15 Feb '19

I changed some things in internal.xml and now I got this: # fs_cli -x reloadxml +OK [[error near line 10374]: missing >] Searching in dialplan.xml at line 13074 gives nothing. How to find where's mistake?

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missing allison/digits/40.wav audio file
by Brandon Youngdale 14 Feb '19

14 Feb '19

Hi, Is there a reason there is no '40.wav' for the allison recordings? I noticed on my install when I got an error that it was not in the digits directory, and when I looked it is not in the repo either. https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/en/… allison/digits/48000 Thank you, -- Brandon Youngdale

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NORMAL_TEMPORARY_FAILURE Freeswitch Outbound via Gateway to PSTN
by Chhorm Chhatra 14 Feb '19

14 Feb '19

Hi, Currently, I faced the issue of getting NORMAL_TEMPORARY_FAILURE when FreeSWITCH is originating outbound calls to the PSTN via Dinstar GSM gateway device. I follow the guides from FreeSWITCH confluence and the inbound call is working fine. My Dinstar Gateway device version is UC2000-VE Business. Any help would be appreciated. Best regards, Chhorm Chhatra

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Clean expired SIP registrations of FS internal database
by Jose David Jurado Alonso 13 Feb '19

13 Feb '19

Hi, I have seen that in the FS internal database the account SIP registrations are stored during a lot of time and it seems that they are not deleted never. Is there any mode or parameter to automatically delete these database records after X time, by expired time or number of records for example? To be better explained, I refer to the SIP profile records that appear when the command is executed: fs_cli> sofia status profile external reg Thank a lot. Regards, José David Jurado Alonso.

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Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION
by Paul Muaddib 13 Feb '19

13 Feb '19

vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU"/> My phones, all Grandstream GXP2140, accept PCMA and PCMU I don't have trouble with other people calling me Am Mi., 22. Aug. 2018 um 15:21 Uhr schrieb Thorsten Göllner < tg-maillistings(a)level5.de>: > Only PCMA@8000 is offered. Check device if codec is activated. > > Am 21.08.2018 um 15:06 schrieb Paul Muaddib: > > Hi, > > I have a rarely error where I get this failure message in the log file. I > am bit wondering because this is suposed to be a normal voice call and not > a fax. > > Best regards, > Paul. > > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [INFO] > switch_ivr_originate.c:1215 Sending early media > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [ERR] > mod_sofia.c:2343 CODEC NEGOTIATION ERROR. SDP: > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 v=0^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 o=- 20012711 20012711 IN IP4 > 217.0.15.67^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 s=on transit^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 c=IN IP4 X.X.X.X^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 t=0 0^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=audio 13274 RTP/AVP 8 118 126 101^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:8 PCMA/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:118 PCMA/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:126 CLEARMODE/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:101 telephone-event/8000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=fmtp:101 0-15^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=gpmd:118 vbd=yes^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=ptime:20^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=image 11232 udptl t38^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxVersion:0^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38MaxBitRate:14400^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxBuffer:3000^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxDatagram:500^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 > a=T38FaxRateManagement:transferredTCF^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxUdpEC:t38UDPRedundancy^M > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 > fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [NOTICE] > switch_channel.c:3515 Hangup sofia/external/+ > 49XXXXXXXXX(a)sip-trunk.telekom.de [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales@freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://clu… > > FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > >

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Count active calls on individual gateways
by Kieran Hackshall 13 Feb '19

13 Feb '19

Looking to get a count of active calls on a particular gateway for outbound calls.. Running "show calls" and "show channels" I cannot see anything that would allow me to determine what gateway is being used. Is there another way to do this? Cheers, Kieran

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streamFile speed deteriorates the voice quality badly
by Aqs Younas 12 Feb '19

12 Feb '19

Greetings list. I am using streamFile in lua script to control speed/volume of the audio file. But i see when i fast ward the audio file(set the speed to +1) it extremely deteriorates the voice quality. Does anybody know some parameters/suggestions that might prevent the voice quality from becoming bad? Best Regards, Aqs

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