- FreeSWITCH-users - FreeSWITCH Mailing Lists

Qt portaudio interface
by Mathieu Rene 04 Mar '09

04 Mar '09

Hi all, Anyone interested in contributing to a Qt interface in order to make a decent softphone using FS please reply to this thread. (also give your availability so we can have a conference call to decide stuff) Thanks, Math

2 1

First time test set up FreeSwitch and SPA3102/SPA3000
by Aplayful Idiot 03 Mar '09

03 Mar '09

I have no background in telephony but probably need to use a PBX. FreeSwitch was recommended by a casual contact so I would like to start first by setting up a small test. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and it works ok but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as <variable name="effective_caller_id_number" value="line1"/>. Looking back over the FS wiki, I'm now wondering if the SPA should of been set-up as a gateway but reading it is confusing at least to me. Sometimes I think the analogue-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? api

2 1

Detecting the origin of voice activity using VAD
by Cameron Sorlie 03 Mar '09

03 Mar '09

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hide caller_id_number, transfer and then change caller_id_number
by Peter P GMX 02 Mar '09

02 Mar '09

Hello, I have the following problem while providing callback (mod_eventsocket is used): 1) I want to call a certain destination number A with a suppressed caller_id_number (this works fine with some vars in the origination string) 2) The destination number A picks up the phone and enters a target number B by DTMF 3) freeswitch then forwards the call to target number B by DTMF and I want to show the number A. I do this with uuid_setvar. The problem is that it still shows unknown. This is all with SIP. uuid_setvar however worked if I did not set the caller_id_number to unknown. Per default this is then "00000000000" and can then be changed with uuid_setvar to the number of A. But if I set caller_id_number to unknown I can no longer change it to A. Any hint? Best regards Peter

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Howto config early dial
by Sergio Alecha 02 Mar '09

02 Mar '09

In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use "early dial" Grandstream telephones. How do Freeswitch in? thank you very much.

1 0

Sangoma USB FXO U100 + Asus notebook?
by Fred 02 Mar '09

02 Mar '09

Thanks guys for the feedback. So, the OpenZap driver isn't ready for production yet?

2 1

First time setting up FreeSwitch and SPA3102 / SPA3000
by Aplayful Idiot 02 Mar '09

02 Mar '09

Hi. I have little background in telephony and need to use a PBX but would like to start first with a small test set-up. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and this works to a point but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as <variable name="effective_caller_id_number" value="line1"/>. Looking back over the FS wiki, I'm now wondering if the SPA was registered or set-up in FS correctly but reading the documentation is confusing me a bit. Sometimes I think the analogue-phone-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? Thanks. api

1 0

Orginate: getting status of call fail
by Nik Middleton 02 Mar '09

02 Mar '09

Hi Guys, I've been running a test script written in lua which now works very well thanks to Anthony's fix to stream file. Right now I'm using an event socket to initiate the call and passing the name of the script along with originate thus: $dialstring = "originate {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/ Mygw/phonenum '&lua(helloworld.lua )'"; $result = $obj ->bgapi_command($dialstring); The script gets fired (it would appear) on answer. However, if the number is invalid , timed out or was busy, I'm not sure the script gets executed or am I wrong? I want to be able to fire an event back on what happed to the call in the event that it failed for whatever reason. I know I can simply call the originate and pass the number as an argument and execute the dial within the script but I'm led to believe that's not very efficient, or am I completely wrong? Looking for the most FS friendly way here Regards,

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Running freeswitch on powerpc
by Rajagopal, Sridhar (Sridhar) 02 Mar '09

02 Mar '09

Hi all, I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code Regards Sridhar > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On > Behalf Of freeswitch-users-request(a)lists.freeswitch.org > Sent: Monday, February 02, 2009 9:12 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 32, Issue 17 > > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more > specific than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Call Variable not available when call hangup (shehzad p) > 2. Re: How do I set my FS internal ip address to a "static" > value. (clif(a)eugeneweb.com) > 3. Re: Call Variable not available when call hangup > (Anthony Minessale) > 4. Re: How do I set my FS internal ip address to a "static" > value. (Brian West) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) > From: shehzad p <pmhshz(a)gmail.com> > Subject: Re: [Freeswitch-users] Call Variable not available when call > hangup > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <21791503.post(a)talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > > one question is that when javascript is being called from > dial plan, I get the session object already available, It is > for A leg of channel, So when javascript is called after > Bridge how can I get the session object for B leg also? > > > Anthony Minessale-2 wrote: > > > > the leg you are running the script on is not hungup, the > other leg of the > > call is. > > > > If it was hungup you would not be executing the script. > > > > Asterisk and the h ext and the whole dead-agi thing are all > poor design > > showing it's teeth. > > We do not support anything like it. > > > > > > You can however try this: (see the link below) > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks > -giving-me-headaches-p21614840.html > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz(a)gmail.com> wrote: > > > >> > >> Is there any settings that when call hangup control can be > transferred to > >> another context and these CDR values can be accessible > there? (just like > >> in > >> Asterisk, h extension) > >> > >> shehzad p wrote: > >> > > >> > Hi all, > >> > > >> > I need to process some CDR variables in Dialplan, like > call duration, > >> > Answered time etc. > >> > but when I place info application after bridge, it is > not listing them > >> > properly as below: > >> > =========================================== > >> > Caller-Channel-Created-Time: [1233573341672157] > >> > Caller-Channel-Answered-Time: [1233573342712939] > >> > Caller-Channel-Hangup-Time: [0] > >> > ========================================== > >> > Here Hangup time is 0, So how can I find actual values? > >> > > >> > --I know that we can use xml_cdr or cdr_csv, but my > current need is to > >> get > >> > those values from dialplan itself so that can be passed to some > >> script... > >> > > >> > > >> > thanks, > >> > msp > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21789152.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > <MSN%3Aanthony_minessale(a)hotmail.com> > > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho > ny.minessale(a)gmail.com> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > <sip%3A888(a)conference.freeswitch.org> > > iax:guest@conference.freeswitch.org/888 > > > googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon > f%2B888(a)conference.freeswitch.org> > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21791503.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 2 > Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) > From: clif(a)eugeneweb.com > Subject: Re: [Freeswitch-users] How do I set my FS internal ip address > to a "static" value. > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <Pine.LNX.4.62.0902011323230.2342@redwall> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > Hi Gang, > > I've been struggleing with this also. Actually I can get it > to bind to my > address, the problem is it randomly drops my calls. :-( > > I have a FS running on a box with a static IP and I can start > a call between > two extensions and it will go for hours. Then I add anther > interface say eth0:0 > with a new static IP and reconfigure my phones and FS to use > that, and the > calls drop after about 15-20 mins. Though it's pretty random. > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > freeswitch-1.0.1. > Here is my /etc/network/interfaces: > > # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) > > # The loopback interface > auto lo > iface lo inet loopback > > # The first network card - this entry was created during the Debian > installation > auto eth0 eth0:0 > iface eth0 inet dhcp > iface eth0:0 inet static > address 192.168.0.249 > netmask 255.255.255.0 > gateway 192.168.0.254 > > The only change I made to the FS config is in Vars.xml. I > added this line close > to the top: > > <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/> > > Here is the console log of the call being dropped: > > freeswitch@archive> sofia status > API CALL [sofia(status)] output: > Name Type > Data > State > ============================================================== > =================================== > external profile > sip:mod_sofia@67.171.158.226:5080 > RUNNING (0) > internal profile > sip:mod_sofia@192.168.0.249:5060 > RUNNING (2) > nat profile > sip:mod_sofia@67.171.158.226:5070 > RUNNING (0) > default alias > internal > ALIASED > outbound alias > external > ALIASED > 192.168.0.249 alias > internal > ALIASED > ============================================================== > =================================== > 3 profiles 3 aliases > > freeswitch@archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > sofia_glue_restart_all_profiles() Reload XML [Success] > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() > ENUM Reloaded > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 > sofia_read_frame() Hangup > sofia/internal/1003@192.168.0.53:5060;user=phone;transport=udp > ;fs_nat=yes > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > switch_ivr_multi_threaded_bridge() Hangup > sofia/internal/1001(a)192.168.0.249 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 6 > (sofia/internal/1003@192.168.0.53:5060;user=phone;transport=ud > p;fs_nat=yes) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/1003@192.168.0.53:5060;user=phone;transport=udp > ;fs_nat=yes > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 5 > (sofia/internal/1001(a)192.168.0.249) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/1001(a)192.168.0.249 > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [192.168.0.249] for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias [default] > for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() > Started Profile > internal [sofia_reg_internal] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [outbound] for profile [external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() > Started Profile > external [sofia_reg_external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() > Started Profile nat > [sofia_reg_nat] > sofia status > API CALL [sofia(status)] output: > Name Type > Data > State > ============================================================== > =================================== > external profile > sip:mod_sofia@67.171.158.226:5080 > RUNNING (0) > internal profile > sip:mod_sofia@192.168.0.249:5060 > RUNNING (0) > outbound alias > external > ALIASED > 192.168.0.249 alias > internal > ALIASED > nat profile > sip:mod_sofia@67.171.158.226:5070 > RUNNING (0) > default alias > internal > ALIASED > ============================================================== > =================================== > 3 profiles 3 aliases > > There is an older thread that says one should set > <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/> > but in this (later) thread is says only Jingleling usese that > variable. > ie. see: > http://www.mail-archive.com/freeswitch-users@lists.freeswitch. > org/msg00695.html > http://www.mail-archive.com/freeswitch-users@lists.freeswitch. > org/msg07345.html > > So what do you think causes this? What is the correct way? ;-) > > > Thanks, > Clif > > > > > ------------------------------ > > Message: 3 > Date: Mon, 2 Feb 2009 09:41:05 -0600 > From: Anthony Minessale <anthony.minessale(a)gmail.com> > Subject: Re: [Freeswitch-users] Call Variable not available when call > hangup > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <191c3a030902020741k779e2488o38ca578a3b40e9ad(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > you can't that's why i said it was a horrible approach. > That's also why i posted you the instructions on the only > elegant solution > to your problem. > > > On Mon, Feb 2, 2009 at 9:21 AM, shehzad p <pmhshz(a)gmail.com> wrote: > > > > > > > one question is that when javascript is being called from > dial plan, I get > > the session object already available, It is for A leg of channel, > > So when javascript is called after Bridge how can I get the > session object > > for B leg also? > > > > > > Anthony Minessale-2 wrote: > > > > > > the leg you are running the script on is not hungup, the > other leg of the > > > call is. > > > > > > If it was hungup you would not be executing the script. > > > > > > Asterisk and the h ext and the whole dead-agi thing are > all poor design > > > showing it's teeth. > > > We do not support anything like it. > > > > > > > > > You can however try this: (see the link below) > > > > > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks > -giving-me-headaches-p21614840.html > > > > > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p > <pmhshz(a)gmail.com> wrote: > > > > > >> > > >> Is there any settings that when call hangup control can > be transferred > > to > > >> another context and these CDR values can be accessible > there? (just like > > >> in > > >> Asterisk, h extension) > > >> > > >> shehzad p wrote: > > >> > > > >> > Hi all, > > >> > > > >> > I need to process some CDR variables in Dialplan, like > call duration, > > >> > Answered time etc. > > >> > but when I place info application after bridge, it is > not listing them > > >> > properly as below: > > >> > =========================================== > > >> > Caller-Channel-Created-Time: [1233573341672157] > > >> > Caller-Channel-Answered-Time: [1233573342712939] > > >> > Caller-Channel-Hangup-Time: [0] > > >> > ========================================== > > >> > Here Hangup time is 0, So how can I find actual values? > > >> > > > >> > --I know that we can use xml_cdr or cdr_csv, but my > current need is to > > >> get > > >> > those values from dialplan itself so that can be passed to some > > >> script... > > >> > > > >> > > > >> > thanks, > > >> > msp > > >> > > > >> > > >> -- > > >> View this message in context: > > >> > > > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21789152.html > > >> Sent from the Freeswitch-users mailing list archive at > Nabble.com. > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users(a)lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale@hotmail.com > <MSN%3Aanthony_minessale(a)hotmail.com>< > > > MSN%3Aanthony_minessale(a)hotmail.com<MSN%253Aanthony_minessale@ > hotmail.com> > > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho > ny.minessale(a)gmail.com> > > > <PAYPAL%3Aanthony.minessale(a)gmail.com<PAYPAL%253Aanthony.mines > sale(a)gmail.com> > > > > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888@conference.freeswitch.org > <sip%3A888(a)conference.freeswitch.org>< > > > sip%3A888(a)conference.freeswitch.org<sip%253A888(a)conference.fre > eswitch.org> > > > > > > iax:guest@conference.freeswitch.org/888 > > > > googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon > f%2B888(a)conference.freeswitch.org> > > > <googletalk%3Aconf%2B888(a)conference.freeswitch.org<googletalk% > 253Aconf%252B888(a)conference.freeswitch.org> > > > > > > pstn:213-799-1400 > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > > http://www.freeswitch.org > > > > > > > > > > -- > > View this message in context: > > > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21791503.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho > ny.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > <sip%3A888(a)conference.freeswitch.org> > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon > f%2B888(a)conference.freeswitch.org> > pstn:213-799-1400 >

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Sangoma USB FXO U100 + Asus notebook?
by Fred 02 Mar '09

02 Mar '09

Hello As an easy way to show a Freeswitch server to prospects, I'm thinking of buying an Asus notebook along with a Sangom USB FXO gateway. www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port If someone's been using those two thingies, I'm curious to know if they happily run Freeswitch, or if I should look for some other hardware? Thank you.

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