June 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_verto
by Vishal Pai 11 Jul '18

11 Jul '18

Hi All Can we record the video conferencing using mod_verto. If yes how it is possible and and what would it’s file format. Thanks Vishal Pai

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RECORD_STEREO should not make both channels the same
by David P 11 Jul '18

11 Jul '18

We'd like to record the two legs of our calls separately (both 16kHz). I followed https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO and added <action application="set" data="RECORD_STEREO=true"/> to our dialplan. I now get two channels instead of one in our mp4's but the data in the two channels is identical, containing speech from both legs of the call. This is contrary to what the confluence page says to expect. Btw, I wanted to include our FS version here, and I tried to get it by running the cli and using /help, but it's not shown there.

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Any feature access code of disconnect B-leg
by Ketan Kothari 11 Jul '18

11 Jul '18

Hello All, I have configured one IVR using that IVR customer can i dial destnation number. The flow is like that. Customer Dial DID number ---> Freeswitch --> Its will ask to dial destination number using DTMF --> [USA]Route to Destination A-leg = Customer ----> Freeswitch B leg = Freeswitch ---> Destination[USA] Again generate B- leg B leg = Freeswitch ---> Destination[INDIA] In this case customer are talking with [USA]destination number and now urgently he want to dial one more destination number[INDIA] and want disconnect[USA] So is there any access code or suggestion using that i can achieve above call flow ?

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About Verto
by 王聡 09 Jul '18

09 Jul '18

Hey all, I had tried to build a WebRTC video call service via mod_verto and Verto Comunicator. Now I could make calls from (to) an Android device with Chrome, but it didn’t work on iOS with Safari. Is there any point to be modified to run on Safari? Regards.

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Setting Headers in mod_xml_curl and mod_curl
by Chhorm Chhatra 09 Jul '18

09 Jul '18

Thank you for making freeswitch awesome! Currently, i face an issue regarding setting headers in mod_xml_curl as per the security requirements to the remote server. I would like to know how to include custom headers in mod_xml_curl configuration. If you do not mind, please kindly let me know if there is any possibility to do so. Thank you so much. Best Regard, Chhatra Chhorm

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Manipulate the sip: string in the INVITE header on an outgoing Bridge
by Joseph Waite 04 Jul '18

04 Jul '18

Hi Guys Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box. We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this. Is there anyway to manipulate the INVITE to be INVITE sip:DID@ <sip:DID@> instead of what is currently INVITE sip:user@ <sip:user@> Im hoping there is something like sip_invite_to_uri, however google is failing me. Regards

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FireFox 61 + Freeswitch 1.6.20
by Geoff Mina 03 Jul '18

03 Jul '18

Is anyone else having issues with WebRTC and the latest release of FireFox? We are showing one way media - with media flowing properly TO the browser, but no media coming back from the browser. Everything in the logs looks OK - no errors. We are using SIP.js as the library of choice for signaling. We are showing SRTP packets flowing bi-directionally (using tcpdump), but since they are DTLS encrypted, I am unable to see exactly what is in each packet to determine if there is media being sent from the browser or they are just empty packets. Any ideas and/or other folks seeing the same issue - would love to hear your thoughts. Thanks, Geoff

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Memory leak with MIME
by Mody, Darshan (Darshan) 03 Jul '18

03 Jul '18

Hi We are observing memory leak when we have MIME data in the incoming invite. The resident memory usage increases constantly. When we tried the traffic without MIME and only SDP the usage stays constant. Attaching the freeswitch logs for the same. Thanks Darshan

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Failed Registration [908]
by Paul Muaddib 03 Jul '18

03 Jul '18

Hi, can someone please look at it and tell my what I am doing wrong? NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration [908], setting retry to 30 seconds. My setup: ######## I have a static IP address for my router. My router does not have UPnP or NAT-PMP. Phones and Freeswitch server are on the same local network. The local network is behind NAT PHONE -> FS -> NAT -> Public Internet I only want to connect to the PSTN via my sip gateway provider. According to this manual I did the following setting: https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal vars.xml ###### <X-PRE-PROCESS cmd="set" data="external_rtp_ip=X.X.X.X“/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=X.X.X.X"/> internal.xml ######### <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml ########## <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> ( What is a little bit strange though is that nat_map status is not working [ERR] mod_commands.c:751 nat_map API called while NAT not initialized ) Is the reason why it does not work the port specification in the Contact field (sngrep output)? Contact: <sip:gw+sip-trunk.telekom.de@X.X.X.X:5080;transport=tcp;gw= sip-trunk.telekom.de> I do not want to setup port forwarding in the firewall. The connection should remain open via a keep alive signal. sngrep output ########### 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060 REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0 Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De Max-Forwards: 70 From: <sip:username@reg.sip-trunk.telekom.de>;tag=9aldsj8n67a To: <sip:username@reg.sip-trunk.telekom.de> Call-ID: CSeq: 111844514 REGISTER Contact: <sip:gw+sip-trunk.telekom.de@X.X.X.X:5080;transport=tcp;gw= sip-trunk.telekom.de> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 Provider settings: ############# output proxy: reg.sip-trunk.telekom.de registrar: sip-trunk.telekom.de sip_profiles/external/telekom_voip.xml ############################## <include> <gateway name="sip-trunk.telekom.de"> <param name="username" value=„XXXXXX“/> <param name="password" value=„XXXXXX“/> <param name="realm" value="sip-trunk.telekom.de"/> <param name="proxy" value="reg.sip-trunk.telekom.de"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> </gateway> </include> Firewall settings ############# # block all traffic in all directions block log # loopback interface is not filtered set skip on lo0 voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}" voip_udp_client_out = "{domain, ntp, sip, 3478}" match out on $wan_if from any nat-to ($wan_if) pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out pass on $voip_if inet proto udp from any to port $voip_udp_client_out pass out on $wan_if inet proto tcp from any to port $tcp_client_out pass out on $wan_if inet proto udp from any to port $udp_client_out Thank you for helping :) Regards, Paul

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freeswitch internal profile hang freequently
by sukitha jayasinghe 02 Jul '18

02 Jul '18

Dear All, I have a freeswitch (1.6.20) server in production with db configured to postgresql through pgpool2. This system hangs time to time and has become a major issue. It happen only to the internal profile where our UACs connects through webrtc. After the incident profile stop responding to cli commands, only solution is restarting the entire application. I managed to get the back trace which indicate system waiting forever for profile database mutex. Further, It seems thread #19 is waiting for result from pgsql server cause this issue. Please find the back trace link below, https://pastebin.com/Uqe8HjZn Has anyone experienced this issue before, what is the solution for this. Best Regards, Sukitha.

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