February 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Normal Temporary Failure - Freeswitch Outbout to Dinstar GSM Gateway
by Chhorm Chhatra 09 Feb '19

09 Feb '19

Hello, Currently, I faced an issue in setting up the outbound call from Freeswitch to PSTN via Dinstar GSM Gateway. My question is there any other configuration I would have to follow to make it works? Please kindly give advises. I follow this Freeswitch Dinstar How-To Guide <https://freeswitch.org/confluence/display/FREESWITCH/Dinstar+GSM+gateway+Fr…>. Since the tutorial is not working for the outbound call to PSTN, I followed the DWG_WITH_FREESWITCH.pdf instead. Unfortunately, both of them not working for me with outbound calls. I tried with the following configuration: *1. With Dinstar GSM Gateway IP* <extension name="dwg2008_gateway"> <condition field="destination_number" expression="^0(\d+)$"> <action application="answer"/> <action application="set" data="ringback=${us-ring}"/> <action application="log" data="INFO calling via gateway to 0$1"/> <action application="bridge" data="sofia/internal/0$1(a)192.168.7.137"/> </condition> </extension> *But this is not working, I got NORMAL_TEMPORARY_FAILURE.* Please see the dinstar log named* setup_1.log* *2. With GSM* <extension name="dwg2008_gateway"> <condition field="destination_number" expression="^0(\d+)$"> <action application="answer"/> <action application="set" data="ringback=${us-ring}"/> <action application="log" data="INFO calling via gateway to 0$1"/> <action application="bridge" data="sofia/gateway/gsm/0$1"/> </condition> </extension> *This is not working as well, I got NORMAL_TEMPORARY_FAILURE.* Please see the dinstar log named* setup_2.log* *NB*. The DID from PSTN to Freeswitch via Dinstar GSM gateway is working fine normal audio. I am able to provide the configuration of dinstar gateway as well if you might want to see it. Any solution would be really appreciated. Best regards, Chhorm Chhatra

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No Media in SIP Calling from 3G Mobile Internet to Freeswitch
by Chhorm Chhatra 09 Feb '19

09 Feb '19

Hi, Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet. In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone *but* I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically. *SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com <http://voip.test.com>, SRTP) * *SIP Registration Sucess, Call No Media* My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. Any help would be really appreciated. Best regards, Chhorm Chhatra

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IPv6
by Rick Jarvis 06 Feb '19

06 Feb '19

Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll explain what I mean: VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address -> IPv4 public FreeSWITCH server Not working, and few too many variables to get my head around. I guess whether there is an ALG on the the private router is still an issue as it’s IPv4 at that point. But should the public IPv6 address speak ok to the public IPv4 FreeSWITCH server, or would it help if I give it an IPv6 address too? (Don’t really understand how the 6->4 thing works, it’s voodoo! :) Thanks, R

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DTMF negotiation
by kaiduan xie 04 Feb '19

04 Feb '19

Hi, We are encountering the following DTMF issue, FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back without telephone-event support in 200. However FreeSwitch still sends out DTMF in RFC 2833 event. I think FreeSwitch should send out DTMF inband instead of 2833 event, what is the configuration for this? Many thanks for help, /Kaiduan

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Verto not segregating domains
by Rick Jarvis 04 Feb '19

04 Feb '19

Ok I know this must be me confusing domains / contexts etc, but can’t work out where I’ve gone wrong. Situation: Verto, using xmlcurl from database server for directory. Verto users are user@ with multiple domains to segregate conferences. The xml being returned is below - I *thought* that simply having the <domain name=“whatever”> would keep things separate if the domains were different, but it seems this isn’t the case... <document type="freeswitch/xml"> <section name="directory"> <domain name="domain2"> <groups> <group name="default"> <users> <user id="user"> <params> <param name="password" value="m6h20sj1y1pmysmayst3"/> <param name="vm-password" value="982375982735"/> <param name="jsonrpc-allowed-methods" value="verto"/> <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/> <param name="jsonrpc-allowed-jsapi" value="all"/> <param name="jsonrpc-allowed-fsapi" value="all"/> <param name="verto-context" value="default"/> <param name="verto-dialplan" value="XML"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="user"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="user"/> <variable name="effective_caller_id_number" value="user"/> <variable name="outbound_caller_id_name" value="user"/> <variable name="outbound_caller_id_number" value="user"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> Output from verto status here shows users correctly on different domains, but maybe the prefix is what’s affecting it here? Or some other setting? default-v4::user@domain1 client x.x.x.x:43249 CONN_REG (WSS) default-v4::user@domain2 client y.y.y.y:55987 CONN_REG (WSS)

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TLS traffic visualization HOWTO
by Giovanni Maruzzelli 04 Feb '19

04 Feb '19

# How to TRACE and visualize TLS and non-TLS SIP traffic in real time (thanks to Homer's Lorenzo Mangani for pointing me toward Frida) apt-get install python-pip pip install frida pip install hexdump wget https://raw.githubusercontent.com/google/ssl_logger/master/ssl_logger.py <https://raw.githubusercontent.com/> #first ssh terminal # create fifo pipe, then will send the content from fifo pipe to an sngrep without gui, which will be reading pcap from stdin, and sending eep packets to the other sngrep (third terminal) mkfifo /tmp/pipe cat /tmp/pipe | sngrep -N -q -H udp:127.0.0.1:5077 -I - #second ssh terminal # writes as pcap to fifo pipe what freeswitch writes and reads from ssl lib python ssl_logger_giova.py -pcap /tmp/pipe freeswitch #third ssh terminal # sngrep that receives packets from both the Ethernet device, and the eep packets sent by the other sngrep (eg, the tls packets ssl_logger grabs from freeswitch's ssl lib) sngrep -L udp:127.0.0.1:5077 (you may want to edit ssl_logger.py and change 228 to be 101 - LINKTYPE_IPV4 to be LINKTYPE_RAW ) -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Issues with INVITE without SDP
by igor.potjevlesch＠gmail.com 04 Feb '19

04 Feb '19

Hello ! I'm facing two issues with scenarios with INVITE without SDP. To support this behaviour towards the ingress peer, I activate the following settings: <param name="enable-3pcc" value="true"/> : per peer ; I also tried with "proxy" instead of "true" but I'm not sure to clearly understand the difference. I also put this parameter on my internal interface. Everything is working fine except that this setting bypass the B2BUA behaviour for media. I mean that the IP and port for media are no longer my Freeswitch interfaces but the gateways into the core network that finally handle the calls (but the other parts of SDP are the one expected and configured on the Freeswitch interface). How can I have both support of INVITE without SDP and media proxy active together? The second issue still start with an in-dialog INVITE without SDP. But instead of re-negotiating the SDP, FS sends the SDP as previous in 200OK. Whereas we expect that FS send SDP with device capability. Moreover, I expect to have the o-line version incremented. Regards, Igor.

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Freeswitch As SBC for sip_phones
by Mehdi Shirazi 03 Feb '19

03 Feb '19

Hi I know that Freeswitch is a B2BUA and not proxy but I have seen SBC's based on Freeswitch also used for sipphones and siptrunks. Is it possible to passthru sip registration with Freeswitch now ? Regards

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FIFO notify to all subscribed clients for single re-subscribe request
by mittali jangid 01 Feb '19

01 Feb '19

Hello, We have two polycom phones configured with BLF of fifo-orbit extension number as address. When either of the phone re-register or re-subscribe, freeswitch generates notify to all the subscribed phones. For example we have FIFO orbit extension as 5900. Phone A and B have BLF with address "queue+5900". When phone B sends subscribe to Freeswitch, it responds with 202 followed by notify but also sends notify to phone A. Is it functionality of Freeswitch or some sort of bug. ? As if there are many FIFO addresses configured for a large system with such multiple polycom phones, there will be flooding of notify, at cost of resource utilization of server.

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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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