June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound calls from GSM to extension that called person on gsm
by Bipin Patel 05 Jun '16

05 Jun '16

I'm having one client who has a special requirement, i have setup FS for them using static XML files and a GSM dongle, they are able to call other extensions and gateways as well as dial out of the GSM line but they wanted incoming calls from GSM to ring the extension that called the person using the GSM line. Meaning extension 1008 calls a number which FS routes out of the GSM line so the called party sees the GSM number always and suppose if the called party is busy and doesn't answer or rejects call then when later he dials back in I want FS to know which extension had called that number so incoming calls from the GSM would ring that extension only. I know I can do this dynamically using XML curl and CDR etc but is there a simple way to achieve this using static XML files only as this FS setup is very small and very basic

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mod_gsmopen wont compile on rpi3
by Bipin Patel 05 Jun '16

05 Jun '16

sorry if this seems like a stupid question but im a newbie to linux, i compiled FS 1.6 on rpi3, later i wanted to add the mod_gsmopen module so i enabled that in modules and recompiled but it never got compiled so then i tried compiling it standalone using the below commands but make clean and make install keep failing with message no rule to make target clean/install apt-get install gsm-utils apt-get install libgsmme-dev apt-get install usb-modeswitch-data usb-modeswitch cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build make DEBUG=0 GPIB=0 make DEBUG=0 GPIB=0 install ldconfig cd /usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/ make clean make install -- Regards, Bipin ------------------------------------------------------------------------

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mod_gsmopen: are USB dongles the only option?
by Stanislav Sinyagin 05 Jun '16

05 Jun '16

does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and is able to deliver voice signal that can be passed to mod_gsmopen? The only one that has any kind of voice signal that I found, is Huawei MU609 and the like. But they deliver PCM voice as a raw signal on mini-PCIe pins which are marked as "reserved" in the interface specifications, so normally the boards don't have any wiring for those pins. I just don't like the idea of an USB dongle, and there are many hardware platforms with mini-PCIe interfaces. They work perfectly for mobile data, but I couldn't find any option for voice.

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Domain and Context
by William Colls 05 Jun '16

05 Jun '16

This is really a very basic question, but it is confusing me. If I want to have my users in a particular domain, do I need to create the domain entry in the directory, and then the various contexts within the domain, or the other way around? e.g /usr/local/freeswitch/conf/directory/domain/context or /usr/local/freeswitch/conf/directory/context/domain Thanks for your time. William.

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SwitchPi
by William Scott 04 Jun '16

04 Jun '16

Is the SwitchPi project and blog still happening?

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Bridge Call and Join Conference
by Colin Morelli 04 Jun '16

04 Jun '16

So, I'm trying to setup certain calls such that they enter into a conference asap. I have the process working using bridging conferences right now, which is okay, but bridging conferences forces the dialing leg of the call to go to an answered state immediately, while the bridge leg is still ringing. So, while everything seems to work correctly, it messes with some of my reconciliation/reporting process with another database that are fed off of channel state events. Essentially it looks like every call was answered. So I was curious to know if there's a way to make a call that creates a bridged leg, and then as soon as the leg is answered immediately joins both legs into a conference. Is the only way to do it to listen for the channel_bridge events and then transfer the call myself? Best, Colin

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Low cost voip phone ?
by Shawn Wheeler 04 Jun '16

04 Jun '16

I am looking for recommendations for a low cost voip handset. I have been using grandstream but can't seem to get past the delay when making a call. Thanks in advance Sent from my Verizon, Samsung Galaxy smartphone

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Dialplan Intercept/Eavesdrop Clarification
by Colin Morelli 04 Jun '16

04 Jun '16

Hey all, I'm sure I'm missing something but I think I'm looking for a middle ground between eavesdrop and intercept. Essentially I have a call running on channel A. New channel B is created. I want channel B to take over all media of channel A, and hangup channel A. Channel B should be put in the exact spot that channel A was in at the time of the interception. Theoretically, if I understand correctly, this is what intercept with -bleg should do, but the call I'm intercepting may not be bridged to another leg. It maybe in a conference, it may be running through the dialplan, etc. I just want this new channel to continue in the same place as the current. Is there a way I can get that behavior? Thanks in advance. Best, Colin

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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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