March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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 >

4 3

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.

3 2

To do telephony functions from web page
by Rex_Alex 02 Mar '09

02 Mar '09

Hi All, I am trying to do the telephonic functions(like dial, hangup, conference and etc.) from a webpage (for a customization) rather than doing it from a soft phone. What would be the optimal way of doing it? Please suggest. Thanks, Rex. -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405… Sent from the freeswitch-users mailing list archive at Nabble.com.

5 7

Problems loading mod_spidermonkey_curl
by David Dan 01 Mar '09

01 Mar '09

I'm getting the following error when I try to load the mod_spidermonkey_curl module. I didn't get any errors when I compiled it. I also tried --without-libcurl but I got the same result. Any help would be appreciated. freeswitch@internal> load mod_spidermonkey_curl -ERR [module load file routine returned an error] 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** freeswitch@internal> version FreeSWITCH Version 1.0.3 (exported)

2 2