September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sharing presence between FS boxes
by Chris B. Ware 04 Jun '16

04 Jun '16

Hi, On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358). If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B? By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_" as I read on wiki. Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared between boxes, and even presence works sometimes, but is not stable. Here my internal sip profiles config: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="configuration" description="Various Configuration"> <configuration name="sofia.conf" description="Sofia Endpoint"> <profiles> <profile name="internal"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="default"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="nat.auto"/> <param name="apply-inbound-acl" value="domains"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="$${recordings_dir}"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="$${internal_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${internal_tls_port}"/> <param name="tls-cert-dir" value="$${internal_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="presence-hosts" value="_DISABLED_"/> <param name="odbc-dsn" value="systemDB:freeswitch:pazzword"/> <param name="challenge-realm" value="auto_from"/> <param name="outbound-proxy" value="sip::5060"/> </settings> </profile> </profiles> </configuration> </section> </document> Can somebody help?

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Using freeswitch as Generic SIP SBC/RtpProxy without predefined endpoints
by Taisto Qvist 14 May '16

14 May '16

Hi, I am an OpenSIPS/Kamailio user that has spent the whole day trying to figure out which software is most suitable to build an SBC that only acts as a full b2bua with rtp-relay, with an undefined number of destinations(entire domains with lots of unknown proxies) on each side. I've been looking at yate, freeswitch, SEMS, but I am overwhelmed by the gazillions of configuration-files that seem to be needed. I have absolutely NO need of pbx/user functionality, just a B2BUA and rtp-proxy capabilities(with full codec-transparency), with the addition that I must be able to add some custom logic to decide the DNS or IP(v6) address of the next-hop sip-node. All examples I've found on freeswitch SBC config, has predefined gateways/trunks/proxies as "predefined next-hops", and I want the SBC to be able to send anywhere, depending on custom application logic. I'd stick with kamailio/rtp-proxy and the standard topology-hiding module, but unfortunately I need the full topology hiding of a b2bua. Is this easily achievable with Freeswitch? Any help with be immensely appreciated! Regards Taisto Qvist

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Mastering FreeSWITCH Book
by Anthony Minessale 09 Apr '16

09 Apr '16

Hi, We've been asked by packt to produce a 3rd book called "Mastering FreeSWITCH" Would anyone be interested in working on the authoring of that book and get their name in print? We need help with creating the content as well as formatting the text into the word templates they require etc. Please contact consulting(a)freeswitch.org if you are interested. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900

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Is MySQL OK to use?
by Alex Lake 06 Apr '16

06 Apr '16

I find these "Error in my_thread_global_end()" messages somewhat annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice (other than "don't use MySQL") for how to install it better? Might it be something to do with thread-safe libraries (I'm using libmyodbc_r.so at the moment)

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Re: [Freeswitch-users] Event socket and Python
by Johan Brannlund 17 Jan '16

17 Jan '16

Sami Montour wrote: > Sorry I misunderstood your post about testing with a closed port. > Anyway, I ran your script (port 8041) but didn't get a connection to > FS, different from what you received. Hmm, that's interesting. I guess either I happened to check out a buggy version of FS, there's something strange with my configuration or there's some ARM-specific issue (I run FS on a Raspberry Pi). I'll have to do some more troubleshooting. Thank you for checking! - Johan

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outbound proxy and port
by Chris B. Ware 05 Jan '16

05 Jan '16

Hi, I need to send registrations and calls from external profile to a si provider, passing through an outbound proxy, port 5080. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151"/> Registrations pass through the OB Proxy, but using, of course, standard 5060 port. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151:5080"/> Param seems to be not understood and registrations go directly to the sip provider. How can I set outbound proxy port? Thanks

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So you wanna setup your own CA for WSS/SSL/TLS?
by Brian West 02 Jul '15

02 Jul '15

Someone should probably turn this into a nice how-to: Here is how I did it. wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz tar zxfv ssl.ca-0.1.tar.gz cd ssl.ca-0.1/ perl -i -pe 's/md5/sha1/g' *.sh perl -i -pe 's/2048/2048/g' *.sh ./new-root-ca.sh ./new-server-cert.sh self.bkw.org ./sign-server-cert.sh self.bkw.org cat self.bkw.org.crt self.bkw.org.key > /usr/local/freeswitch/certs/wss.pem Setup Apache: default-ssl: SSLCertificateFile /usr/local/freeswitch/certs/wss.pem SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem Setup Sofia TLS: cat self.bkw.org.crt self.bkw.org.key > /usr/local/freeswitch/certs/agent.pem cat ca.crt > /usr/local/freeswitch/certs/cafile.pem vars.xml: <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> Restart FreeSWITCH. Now make sure your system has ca.crt imported so it will trust your new found hotness. TEST: openssl s_client -connect self.bkw.org:443 openssl s_client -connect self.bkw.org:8082 Depending on what you've setup you'll see: subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web Server/CN= self.bkw.org/emailAddress=brian@bkw.org issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang Bang/OU=Certification Services Division/CN=WBB Root CA/emailAddress= brian(a)bkw.org Or there abouts. -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Receive chat
by Sameer Khan 05 May '15

05 May '15

Hi, Is there any command which can receive chat from sip channel so I can put that received message in to a variable.

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eliminating fax header line
by Bruce Lefko 26 Mar '15

26 Mar '15

I'd like to completely eliminate the header line sent at the top of each fax page using spandsp. i've set fax-header to a null string, but that doesn't seem to be removing it, and it defaults back to 'Spandsp fax header'. This seems to be the default in the source code in 1.2. What do I have to do to eliminate the header line? Do I have to make changes to freeswitch source? Thanks!

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Real-time billing application for the FreeSWITCH (mod_lua, mod_perl or ESL)
by Sami Montour 04 Mar '15

04 Mar '15

Hello Everyone, I am a long time user of OpenSIPS and have recently been tasked with developing a call control/real-time billing application for the FreeSWITCH. I would like to develop the application in either C/C++, Perl or lua (totally new to lua) in this order. I have used ESL to develop some basic functionality of the application using Perl. For instance, when the call comes in, I pass over control from FreeSWITCH to the call control application using outbound TCP socket. Then the call control application listens for CHANNEL_ANSWER event to start billing process and CHANNEL_HANGUP event to stop billing. The application does some database work as well to retrieve and check user status and balance and all that stuff. My main concern is with load testing because of the TCP connection between the FreeSWITCH and call control application and also TCP connection to database to retrieve subscriber info. The basic functionality seems to work fine but haven't tested it with high call volume, which I am about to do. The last couple of days I have been experimenting with mod_lua but can't find a method under "Session" to get information related to CHANNEL_ANSWER and CHANNEL_HANGUP events. The "session:geVariable" does not have variables related to session answer time or hangup time, something similar to "$e->getHeader("Caller-Channel-Answered-Time")" in ESL. I must be missing something very simple here. Again, my experience with mod_lua is only a couple of days old. I am new Freeswitch and thought I would seek the advice of all the experts and people with more experience with Freeswitch. For this type of application, am I better off with mod_lua, or mod_perl, or shall I stick with ESL using Perl? How about ESL with C/C++? Will I gain some performance gain there? Any feedback would be appreciated. Thanks. Sam

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