December 2015 - 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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Re: [Freeswitch-users] Presence with shared database
by Adam Kuśmirek 04 Jun '16

04 Jun '16

Hello, Yes we enabled manage-presence. According to documentation dbname is used to share presence between profiles. What we want to achieve is to share presence between hosts. We investigated some presence source code and found that there are sql queries that ask for specific FS host. We modified SELECT statements and UPDATES of version parameter, so they have no host in WHERE clause. After mod_sofia recompilation everything works fine. It seems that the only way to share presence between hosts is to set the same hostname in switch.xml for every FS node, but we cannot do this because we need host information for other services. I wonder why FS checks whether the subscription was made on a host that sets up the connection. Maybe this is a bug in the concept and it should be removed. Regards Adam Kusmirek ---------- Wiadomość przekazana dalej ---------- >> > From: "Ítalo Rossi" <italorossib(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 7 Aug 2015 22:42:37 -0500 > Subject: Re: [Freeswitch-users] Presence with shared database > Hard to say without more info, where's your configs? > > Did you enabled the manage-presence in your profile? and set the dbname? > > >

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FreeSWITCH using outbound proxy but to bypass media
by Muhammad Naseer Bhatti 28 May '16

28 May '16

I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It’s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created) The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and bypass_media_after_bridge but it’s not working. Is this even doable? Thanks. -- Muhammad Naseer Bhatti

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Fax....How I hate you.
by Blake Priddy 24 May '16

24 May '16

Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use faxing.. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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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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Unable to conmplie mod_shout
by sunil more 04 Apr '16

04 Apr '16

Hello All,I am trying to compile freeswitch latest commit on Ubuntu Server and I am getting stuck while compiling mod_shout. It asked me for adding dependency libmpg123-dev. I have this lib package already installed on my machine. However i still get stuck over this error. Please help me.Operating system:Distributor ID: UbuntuDescription: Ubuntu 14.04.3 LTSRelease: 14.04Codename: trusty Error:making all mod_shoutmake[4]: Entering directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout'Makefile:895: *** You must install libmpg123-dev to build mod_shout. Stop.make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout'make[3]: *** [mod_shout-all] Error 1make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'make[2]: *** [all-recursive] Error 1make[2]: Leaving directory `/usr/local/src/freeswitch/src'make[1]: *** [all-recursive] Error 1make[1]: Leaving directory `/usr/local/src/freeswitch'make: *** [all] Error 2 Regards,Sunil More

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FS, SS7 and Dialogic IMG2020 in the UK
by Gavin Henry 15 Mar '16

15 Mar '16

Hi all, We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom regulated SS7 interconnects: http://www.dialogic.com/en/products/gateways/img/img2020.aspx Any issues known with FS behind these? Thanks, Gavin. -- http://www.surevoip.co.uk

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FS behind NAT
by Tao Zhang 14 Mar '16

14 Mar '16

Hi all, I have been tracking down why my FS verto video conf demo setup not working. Now I believe the reason is my FS server is behind a NAT. My previous posts may be misleading now, so I am creating a new thread. Some evidence I saw: 1) In the SDP dialog, I saw my FS server's internal IP got used, rather than its public IP, for example Dialog 6c62f631-469c-a2f0-e484-6e9cfcda7eedANSWER SDP v=0 o=FreeSWITCH 1444254094 1444254095 IN IP4 10.240.15.244 s=FreeSWITCH c=IN IP4 10.240.15.244 <============= t=0 0 2) Also, in wireshark , I saw my client is trying to send STUN packets to the FS server's internal IP, 450 18.252726000 192.168.1.7 10.240.15.244 STUN 158 Binding Request user: mlvLvmDzCphyQYz5:N3MZoB99aGiAZotr So now my questions are Will the verto video conf demo work when the FS server is behind NAT? If so, could someone help me with the necessary config changes and etc.? Thanks so much! Tao

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