December 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Adding user location with ESL
by Mehdi Shirazi 06 Dec '19

06 Dec '19

Hi Is there a way to add a sip user location info to freeswitch with ESL or fscli (similar to ul_add in opensips) ? Regards M.Shirazi

1 0

Unable to compile mod_v8
by Shahnawaj Khan 05 Dec '19

05 Dec '19

Hi, I need to test freeswitch asr sample <extension name="pizza_demo"> <condition field="destination_number" expression="^(9711614454)$"/> <condition field="${module_exists(mod_v8)}" expression="true"/> <condition field="${module_exists(mod_pocketsphinx)}" expression="true"> <action application="javascript" data="ps_pizza.js"/> </condition> </extension> mod_v8 is required to run the javascript. It is not there by default, So i tried to compile it but facing the issue as. ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or directory #include <v8.h> Please let me know if anyone have faced the same issue and its resolution. Thanks & Regards, shahnawaj

4 5

Recording RECORD_APPEND=true Prefix
by Moshe Rosenberg 05 Dec '19

05 Dec '19

Hi right now i had send my dial-plan to RECORD_APPEND=true and its working fine how ever it appends the recording to the end of the file, i would really like to append the recording to the begging of the file is there away to do this on freeswitch ? -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data <http://www.dataday.consulting/>phone.cloud Moshe(a)dataphone.cloud

3 2

mod_watson available for debian 10
by António Silva 04 Dec '19

04 Dec '19

Hi, Is mod_watson (https://freeswitch.org/confluence/display/FREESWITCH/mod_watson) available for debian buster? Can it be activated on compile version like g729? Thanks. -- Saludos / Regards / Cumprimentos António Silva

1 0

Audiocode MP-114 FXS_FXO device setup
by Rajil Saraswat 04 Dec '19

04 Dec '19

Hello, My trusted Linksys SPA3102 died and I am trying to replace it with an Audiocodes MP-114 device which has 2 FXS ports and 2 FXO ports. My first plan was to get the FXS port working. I specified my FreeSwitch PBX details in VOIP>SIP Definitions>Proxy & Registration and specified my number in VOIP>GW and IP to IP>Hunt Group>Endpoint Phone Manager. With this i was able to successfully register an extension '204' on the pbx. However, if i make a call from a softphone to this number the calls do not come through, and the ATA logs says 4d:11h:17m:15s ( lgr_TrnkGrp)(9672 ) !! [ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 204 4d:11h:17m:15s ( lgr_psbrdif)(9673 ) !! [ERROR] MotherBoard::GetEndPoint- Can't find EndPoint for Dest:204 Source:mod_sofia SourceIp:ac10020a 4d:11h:17m:15s ( lgr_call)(9674 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 204 Any idea where do i define the 'Endpoint' for number 204? Thanks, Rajil The .ini file for ATA is as follows ;************** ;** Ini File ** ;************** ;Board: MP-114 FXS_FXO ;Board Type: 56 ;Serial Number: 4929340 ;Slot Number: 1 ;Software Version: 6.60A.355.004 ;DSP Software Version: 204IM3=> 660.15 ;Board IP Address: 172.16.2.4 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 172.16.2.1 ;Ram size: 32M Flash size: 8M ;Num of DSP Cores: 1 Num DSP Channels: 4 ;Profile: NONE ;License Key limits aren't active full features capabilities are available !; ;---------------------------------------------- [SYSTEM Params] SyslogServerIP = 10.1.1.89 ;NTPServerIP_abs is hidden but has non-default value NTPServerUTCOffset = 19800 ;VpFileLastUpdateTime is hidden but has non-default value NTPServerIP = '172.16.2.1' [BSP Params] PCMLawSelect = 3 RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 [Analog Params] PolarityReversalType = 1 MinFlashHookTime = 100 [ControlProtocols Params] AdminStateLockControl = 0 [MGCP Params] [MEGACO Params] EP_Num_0 = 0 EP_Num_1 = 1 EP_Num_2 = 1 EP_Num_3 = 0 EP_Num_4 = 0 [Voice Engine Params] CallProgressTonesFilename = 'usa_tones_13.dat' FaxTransportMode = 0 V22ModemTransportType = 0 V23ModemTransportType = 0 V32ModemTransportType = 0 V34ModemTransportType = 0 RFC2833TxPayloadType = 101 [WEB Params] LogoWidth = '208' HTTPSCipherString = 'RC4:EXP' [SIP Params] MAXDIGITS = 3 ISREGISTERNEEDED = 1 AUTHENTICATIONMODE = 0 GWDEBUGLEVEL = 5 ;ISPRACKREQUIRED is hidden but has non-default value ENABLEEARLYMEDIA = 1 PROXYNAME = 'BLR' REGISTRARIP = '172.16.2.10' SIPGATEWAYNAME = 'pbx.blah.com' USERNAME = '204' ;PASSWORD is hidden but has non-default value ;SHOULDREGISTER is hidden but has non-default value SUBSCRIPTIONMODE = 0 REGISTRARNAME = 'pbx.blah.com' REGISTRARTRANSPORTTYPE = 0 MSLDAPPRIMARYKEY = 'telephoneNumber' [IPsec Params] [SNMP Params] [ DspTemplates ] ; ; *** TABLE DspTemplates *** ; This table contains hidden elements and will not be exposed. ; This table exists on board and will be saved during restarts. ; [ \DspTemplates ] [ PREFIX ] FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort, PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID, PREFIX_DestSRD, PREFIX_CostGroup, PREFIX_ForkingGroup, PREFIX_CallSetupRulesSetId; PREFIX 0 = "10", "10.1.10.10", "*", 0, 255, 0, -1, "", -1, "", -1, -1, -1, "", -1, -1; PREFIX 1 = "20", "10.1.10.11", "*", 0, 255, 0, -1, "", -1, "", -1, -1, -1, "", -1, -1; [ \PREFIX ] [ TrunkGroup ] FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId, TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel, TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId, TrunkGroup_LastTrunkId, TrunkGroup_Module; TrunkGroup 0 = 1, 255, 1, 1, "204", 1, 255, 255; [ \TrunkGroup ] [ ProxyIp ] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; ProxyIp 0 = "172.16.2.10", 0, 0; [ \ProxyIp ] [ TxDtmfOption ] FORMAT TxDtmfOption_Index = TxDtmfOption_Type; TxDtmfOption 0 = 4; [ \TxDtmfOption ] [ TrunkGroupSettings ] FORMAT TrunkGroupSettings_Index = TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode, TrunkGroupSettings_GatewayName, TrunkGroupSettings_ContactUser, TrunkGroupSettings_ServingIPGroup, TrunkGroupSettings_DedicatedConnectionMode, TrunkGroupSettings_MWIInterrogationType, TrunkGroupSettings_TrunkGroupName; TrunkGroupSettings 0 = 1, 0, 0, "", "", -1, 0, 255, ""; [ \TrunkGroupSettings ] [ EnableCallerId ] FORMAT EnableCallerId_Index = EnableCallerId_IsEnabled, EnableCallerId_Port, EnableCallerId_PortType; EnableCallerId 0 = 1, 1, "FXS"; [ \EnableCallerId ] [ ProxySet ] FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap, ProxySet_SRD, ProxySet_ClassificationInput, ProxySet_ProxyRedundancyMode, ProxySet_KeepAliveFailureResp, ProxySet_HomingSuccessDetectionRetries; ProxySet 0 = 0, 60, 0, 0, 0, 0, -1, "", 1; [ \ProxySet ] [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = "g711Alaw64k", 20, 255, -1, 0; CodersGroup0 1 = "g711Ulaw64k", 20, 255, -1, 0; [ \CodersGroup0 ] [ RoutingRuleGroups ] FORMAT RoutingRuleGroups_Index = RoutingRuleGroups_LCREnable, RoutingRuleGroups_LCRAverageCallLength, RoutingRuleGroups_LCRDefaultCost; RoutingRuleGroups 0 = 0, 1, 1; [ \RoutingRuleGroups ] [ ResourcePriorityNetworkDomains ] FORMAT ResourcePriorityNetworkDomains_Index = ResourcePriorityNetworkDomains_Name, ResourcePriorityNetworkDomains_Ip2TelInterworking; ResourcePriorityNetworkDomains 1 = "dsn", 1; ResourcePriorityNetworkDomains 2 = "dod", 1; ResourcePriorityNetworkDomains 3 = "drsn", 1; ResourcePriorityNetworkDomains 5 = "uc", 1; ResourcePriorityNetworkDomains 7 = "cuc", 1; [ \ResourcePriorityNetworkDomains ]

1 0

Outbound calls to remote gateway on dynamic IP
by LinMin 03 Dec '19

03 Dec '19

Hi folks, I have a Freeswitch server with a PSTN gateway logged in as a normal user, which gives access to the gateway on a dynamic IP address (trying to avoid static IP and extra DNS settings in this situation). Calls coming in from the gateway to FS work fine, but trying to route calls to the gateway hit a snag, because instead of sending calls to: <dialed_number>@gateway_IP , they go to: <FS_gateway_user_account>@gateway_IP^<dialed_number> which results in the gateway attempting to dial it's own user ID via the PSTN line. I've tried using this option in the dialplan extension to work around it: <action application="bridge" data="${sofia_contact(internal/Remote-GW-1)}^$1" /> which results in the above situation. Manually inserting the remote IP works: <action application="bridge" data="sofia/internal/$1(a)172.27.2.120" /> .... however, I'm hoping there's a way that I can fish and insert the IP address on the fly, or otherwise tell the gateway to dial the correct number. Cheers, Francis

1 0

bash script with variables
by thomas peterseil 02 Dec '19

02 Dec '19

hello freeswitch-list, i would like to start from the dialplan a bash script and the freeswitch is handing over one variable to the script. in the cli i see the correct start of the script: [NOTICE] mod_dptools.c:2120 Executing command: /root/scripts/test1.sh 3333 the script is very simple: #!/bin/bash number=$1 mutt -s "Call $number" thomas.peterseil(a)mine-project.eu < /root/mailtexte/registrierung.txt echo "$number" > test1.txt i get the email with the variable in the subject, that works fine, but i can´t see the 3333 in the test1.txt file. when i start the script from the command line with ./test1.sh 3333 all is working fine. can somebody give me a hint why it doesn´t work from the dialplan. thanks a lot and have a nice day! best regards, thomas

4 3

call transfer
by Mustafa Ali Kahraman 01 Dec '19

01 Dec '19

hi, i want to transfer calls from one provider to another one. i've done like this. <condition field="network_addr" expression="^xx\.yy\.zz\.tt$"> <action application="set" data="ringback=...."/> <action application="pre_answer" data=""/> <action application="bridge" data="{...}sofia/external/${destination_number}@$${OTHER_PROVIDER}"/> <action application="answer" data=""/> <action application="hangup" data=""/> </condition> *Questions*: 1- is this the right way? 2- when i do like this, do RTP packages go over my server? 3- what should i do, just to transfer signalling, not to transfer media over my server? thanks. -- -tafali- www.tafali.net

4 3