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Conceptual/design question - per gateway/DTMF & PSQL
by Mike Taylor 09 Jun '11

09 Jun '11

Hi, I hope that I've taught myself enough over the last few weeks that this isn't a blindingly stupid question I am providing services to multiple customers on FS 1.0.6 I have a basic setup; (which I have inherited, hence the basic questions); AS5350--------<external>----- FS-----<internal>------customer GW(s) (No Registration) (No Registration) All of the GW info is pulled from a PSWL database. Carriers, carrier_id, lcr table etc. The bit I am missing is; If all of my GW config is in the PSQL database, and that's all accessed by mod_lcr before routing the call, how do I allow for different DTMF types, codecs etc, PER CUSTOMER GW ? Am I stuck with actually defining the gateways statically under /etc/freeswitch/conf/sip_profiles/ (or I see I can doit in the dialplan, but that won't achieve what I need) I was hoping to be able to pull the DTMF (and other misc info) from the database, but the more I look into it, the less likely it is that this will work I think. (mod_lcr is already at the routing stage, can't modify DTMF at that point AFAIK) Like I said, hope the question isn't so obvious that it's crazy Regards, Mike

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Setup help Conference, 3 way call, FIFO, MOH, record to Shoutcast
by Rev Michael Carbone 09 Jun '11

09 Jun '11

I have things setup and working and am using fusionpbx and am totally new to all this. I am setting things up to have an online call in talk show. What I want to be able to do is this: Have me and a guest on and the call broadcasting to a shoutcast/icecast server (which I have already) and want to be able to broadcast this as MOH, how can I also have people in a FIFO cue listening in on the show and be able to bring them in one at a time. Broken down: 2 of us on a call broadcasting to shoutcast other people call in and are put in cue with conversation as MOH bring people on air one at a time while others are still muted I'm sure this can be done, but how? Please remember I'm new to all this so instructions would be helpful. Thank you, Michael

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Problem receiving calls from E1 to TE122p
by Javier Galaz Jeria 08 Jun '11

08 Jun '11

Hi, I'm using FS git, on Ubuntu 11.04, with Digium TE122p card (from now on srv1), and FS git, on Ubuntu Server 11.04 with Sangoma A101 card (from now on srv2), both currently configured by vendor's scripts. When I make a call from srv1 to srv2, everything goes well, I can hear both ways no problem, however when I try it the other way around (srv2->srv1) the softphone registered to srv2 gets "User Not Available", hardphone registered to srv1 gets INVITE and just after that CANCEL. The FS log: [1] Looking at the logs I can see that after the RINGING goes to EARLY and tries to play ringback tone, but it can't write and it can't read from channel because it's not opened. So it hangs up. [lines 520-528] I've searched the wiki, asked on IRC, but can't solve my problem, anyone who has configured TE122p could give some pointers? This are my config files: -----TE122p----- /etc/dahdi/system.conf loadzone = es defaultzone = es span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 /usr/local/freeswitch/conf/freetdm.conf [general] cpu_monitor => no cpu_monitoring_interval => 1000 cpu_set_alarm_threshold => 80 cpu_reset_alarm_threshold => 70 cpu_alarm_action => warn [span zt TE122p] trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 /usr/local/freswitch/conf/freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="sip_headers" value="true"/> </settings> <libpri_spans> <span name="TE122p"> <param name="node" value="network"/> <param name="switch" value="euroisdn"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> </configuration> ------A101----- /etc/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 2 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = NCRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 430 LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode TDMV_HW_DTMF = YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 /usr/local/freeswitch/conf/freetdm.conf [span wanpipe wp1] trunk_type => e1 group=1 b-channel => 1:1-15 b-channel => 1:17-31 d-channel => 1:16 /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/>    </settings> <config_profiles> <profile name="my_pri_te_1"> <param name="switchtype" value="euroisdn" /> <param name="interface" value="pri_cpe"/> </profile> </config_profiles> <sangoma_pri_spans> <span name="wp1" cfgprofile="my_pri_te_1"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </sangoma_pri_spans> </configuration> How can I fix this? Any help would be greatly appreciated. If you need more info just tell me. [1] http://pastebin.freeswitch.org/16397 Best Regards -j-

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Passing a map to session.setVariable()
by Yungwei Chen 08 Jun '11

08 Jun '11

Hi, In javascript, is passing a map or array to session.setVariable() supported? Thanks.

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LuaSQL, Curl, mod_skypopen, mod_dingaling....Can't get 'em to work!
by Mint.Nick 08 Jun '11

08 Jun '11

Hi all, I've noticed there isn't much traffic on this forum anymore. I'm having issues with the things listed in the subj. Is there anyone out there who might be willing to help a guy figure one or more of these out? Thanks -Nick -- View this message in context: http://old.nabble.com/LuaSQL%2C-Curl%2C-mod_skypopen%2C-mod_dingaling....Ca… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH Conference Call Today
by Michael Collins 08 Jun '11

08 Jun '11

Hello gang! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_08 We have a special guest from George Washington University today: Robert Daniel. We are looking forward to hearing from him about how FreeSWITCH is being used in his telecommunications classes. -MC

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An extension with a javascript action can handle multiple concurrent calls?
by Yungwei Chen 08 Jun '11

08 Jun '11

Hi, I'm wondering if the following extension can handle multiple concurrent calls. Thanks. <extension name="test"> <condition field="destination_number" expression="^9912$"> <action application="answer"/> <action application="javascript" data="test.js"/> </condition> </extension>

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domain
by Sam 08 Jun '11

08 Jun '11

Hello, I have a user registered by domain (1001(a)xyz.com) and when the invite comes to my FS server it should recognize the domain and route the calls accordingly to the group <domain name="xyz.com"> ... </domain> . The call would be entering via the acl to public.xml. Here i want to route the call according to domain names, how will i do that. Any suggestions... Regards Sam

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Freeswitch dialplan help
by awais nazir 08 Jun '11

08 Jun '11

Hi I want to achieve a simple scenario , after attempting a bridge in my dialplan if I get cause code 480, it should be converted to 503 to originator. <action application="bridge" data="sofia/${Dialstring}"/> <!--If ${hangup_cause} == "480" then ${hangup_cause} =503 > Can somebody help here.

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FS does not uses the IP/PORT resolved using STUN.
by DEVARAHALLI, PRAKASH (PRAKASH) 08 Jun '11

08 Jun '11

I configured FS to use the STUN server to find the public IP address used by the NAT. FS sends the request to STUN and receives the reply from the STUN Server , But it does not uses this info ( IP and port obtained from STUN) neither while negotiation ( in SDP) nor in the RTP. My testbed: ----------- < FreeSwitch > ---- < NAT > ---+----<NAT>--- < SIP Phone Client > | | [ STUN Server] File conf/sip_profiles/internal.xml has below configuration. <param name="ext-rtp-ip" value="stun:120.120.120.120"/> ; This is the local STUN server. <param name="ext-sip-ip" value="stun:120.120.120.120"/> <param name="stun-auto-disable" value="false"/> <param name="rtp-ip" value="auto"/> Thanks Prakash

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