- FreeSWITCH-users - FreeSWITCH Mailing Lists

Linksys/Cisco SPA400 (ATA 4 Line FXO) Now Documented in Wiki
by Karl Vesterling 29 Nov '08

29 Nov '08

Folks; I've just taken the time to document the Sipura, err, Linksys, errr Cisco SPA400 4 line FXO Analog Telephone Adapter in the Wiki. http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo If anyone uses these ATA's and has questions about it let me know and I'll see if I can answer them in the Wiki. Of course, if you know something about it that I haven't documented, by all means, document it in the Wiki. ;-) **Note: MikeJ seems to think that I'm overlooking something pertaining to the registration of the SPA400 without a password. Evidently this pertains to accepting blind auth, but I'd need to know more about that before knowing how to piece it in. Best Regards, Karl J. Vesterling kjv(a)ken-ton.com 202-448-3009 x0

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Answering machine detection
by Keith Wood 29 Nov '08

29 Nov '08

Hi, Does anyone know if it is possible to config freeswitch to do answering machine detection? I see that there is a tone_detect command. Is it possible to use it to achieve a certain level of answering machine detection? Thanks, Keith

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where do variables like local_ip_v4 get their values?
by Carole O. 28 Nov '08

28 Nov '08

Hello, I have just installed Freeswitch on OpenSUSE 10.3. I was wondering where the variable indicated with $$ get their values. I get some problems with the variables local_ip_v4 and domain which are both initialized with the loopback address 127.0.0.1. Thereof, no registration is possible on the ip address of the PC (192.168.1.200) since the port 5060 is closed. Freeswitch is apparently waiting on the port 5060 of the ip address 127.0.0.1 which is open. How does it come that the loopback address is chosen? I try the following: add in vars.xml $${local_ip_v4}=192.168.1.200. After this freeswitch is listening at the port 5060 of 192.168.1.200. So it seems to work, even if it does not answer my first question (where does the default value of local_ip_v4 come?). Moreover, I got another problem. The registrations are now possible but I observe that the default domain is still bound to the ip address 127.0.0.1. So I try to add in vars.xml $${domain}=192.168.1.200, which did not work. The domain is still bound to 127.0.0.1. Why???? I thank you in advance for your help. Best regards, Carole -- View this message in context: http://www.nabble.com/where-do-variables-like-local_ip_v4-get-their-values-… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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send_dtmf: Understanding question...
by Dennis 28 Nov '08

28 Nov '08

hi, i wonder, if or what i do not understand how to do send_dtmf in the right way. for example i want to send dtmf tones to my mobile mailbox the enter the menu and do some changes to the settings. but whatever i try, it does not work. i tried: sendmsg send_dtmf outbound_uuid 123# sendmsg playback outbound_uuid tone_stream://123# sendmsg gentones outbound_uuid 123# api uuid_send_dtmf outbound_uuid 123# [bleg] i am using socket outbound. if i test from one phone to another phone, i will not get any events about incoming dtmf-tones. so i myself could not even react on sent dtmf tone. if i press a digit on the phones, i get events about that. fs generates the tones: i can see them in the console and i can hear them. but the other side does not seem to receive them - at least not right. i always get the message, that the login was wrong. do i have to uncomment something in sofia conf?

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MOH problems
by Helmut Kuper 28 Nov '08

28 Nov '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, we have two different versions of FS trunk running. On both we are testing MOH. Version "FreeSWITCH Version 1.0.trunk (10138M)" works fine, but version "FreeSWITCH Version 1.0.trunk (10370M)" seems to stream only silence. At least we find out that the call is established and RTP is transmitted in both directions but there is no music. We did today a "svn update" and then a "make sure". Since then no moh anymore. We used snom and xlite phones. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkkcRKoACgkQ4tZeNddg3dxZ4QCfQDZQQNdmy9ndI+XDPKMv87kl I60AniKRLiGx9ai4RCjFiwhMD2x1JUZL =Ur2k -----END PGP SIGNATURE-----

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xcap server that works with freeswitch
by matrim 28 Nov '08

28 Nov '08

Hi, Is anyone aware of an XCAP server that works with freeswitch? If there is none available now does anyone know if one is in development? XCAP is and xml document format communicated via http that is used in SIP\SIMPLE for resource management (e.g. buddy lists). -- View this message in context: http://www.nabble.com/xcap-server-that-works-with-freeswitch-tp20725369p207… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Connection to a QuesCom gateway without password
by Jon Bruel 27 Nov '08

27 Nov '08

I have tried connecting to a SIP GSM gateway made by the French company, QuesCom. It does not handle the authentication process properly, and the connection cannot be setup because of this. I have tried to omit the password parameter but apparently FS does not accept a setup without password. Is there any way to circumvent this? /Jon

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fallback url in xml_cdr ?
by Tim Panton 27 Nov '08

27 Nov '08

I'm building a cluster with a couple of freeswitch nodes. (I'll try and write it up when I'm done). I'm pleased with the way that freeSWITCH lets me set a fallback url for xml_curl getting the dialplan which is used when the 'primary' config engine isn't working. I am now looking for an equivalent fallback for xml_cdr. Is there a way to give a secondary url in xml_cdr.conf.xml ? Thanks. Tim.

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strftime(%u)
by henkoegema 27 Nov '08

27 Nov '08

strftime(%u) never gives me any output in an application ! While strftime(%w) does. (output 6 (saterday) ) Is this a bug or could there be any other reason? Just as a test, I tried: <action application="record_session" data="$${base_dir}/recordings/${strftime(%u)}_to_${destination_number}_from_${caller_id_number}.wav"/> output: _to_2000_from_2003.wav ------------------------------------------------------------------------------------------------------------------------------------- <action application="record_session" data="$${base_dir}/recordings/${strftime(%w)}_to_${destination_number}_from_${caller_id_number}.wav"/> output: 6_to_2000_from_2003.wav -- View this message in context: http://www.nabble.com/strftime%28-u%29-tp20635116p20635116.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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False call running so long..
by shehzad p 27 Nov '08

27 Nov '08

I faced a strange behavior with my freeswitch... Freeswitch setup is as follows: Any call comes to freeswitch are sent to javascript, for routing based on some logic. After call completes, the same script insert the CDR values into mysql database. All is working fine for quite long time, But when I was looking in freeswitch.log file for particular call, I found one CDR entry with big difference in time between the call terminated and time of its CDR inserted into database. The part of freeswitch.log is as follows: #############################################freeswitch.log##################################################### 2008-11-23 17:40:26 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/internal/1234567890(a)xxx.xxx.xxx.xxx Execute javascript(softswitch/main.js) ............... 2008-11-23 17:40:26 [DEBUG] sofia.c:197 sofia_event_callback() event [nua_i_state] status [487][Request Terminated] session: sofia/internal/1234567890(a)xxx.xxx.xxx.xxx 2008-11-23 17:40:26 [DEBUG] sofia.c:2129 sofia_handle_sip_i_state() Channel sofia/internal/1234567890(a)xxx.xxx.xxx.xxx entering state [terminated] 2008-11-23 17:40:26 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1234567890(a)xxx.xxx.xxx.xxx [CS_EXECUTE] [ORIGINATOR_CANCEL] ...... ....... ..... 2008-11-23 23:14:20 [DEBUG] switch_ivr_bridge.c:353 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/1234567890(a)xxx.xxx.xxx.xxx] ....... 2008-11-23 23:14:20 [INFO] main.js:150 console_log() CDR Inserted 2008-11-23 23:14:20 [INFO] main.js:151 console_log() ***** start date : 2008-11-23 17:40:26 2008-11-23 23:14:20 [INFO] main.js:152 console_log() ***** caller : 1234567890 ################################################################################################################ This means that time difference between the call terminated and its CDR inserted is about 5.30 hours, which is too late. During this time it was also listing out in 'show calls' and 'show channels'. Any clue for such delay.??? thanks in advance... msp -- View this message in context: http://www.nabble.com/False-call-running-so-long..-tp20678102p20678102.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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