January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Caller-ID variables and another Caller-ID question
by Steven Schoch 26 Jan '13

26 Jan '13

There are several caller-ID variables defined in the wiki, and I'm trying to understand the difference: caller_id_number, caller_id_name, origination_caller_id_number, origination_caller_id_name, effective_caller_id_number, effective_caller_id_name, outbound_caller_id_number, outbound_caller_id_name, If I understand correctly, caller_id_number and caller_id_name are read-only variables that contain the caller-ID from an incoming call. I don't understand origination_caller_id_number. It says it sets it for LEG A, but I don't understand LEG A. What's the difference between origination_caller_id_number and effective_caller_id_number? As near as I can tell, the outbound_caller_id_number variable is not used by any module, but is mainly used to send information from a directory entry to an outbound dialplan. Do I have that right? In the dialplan, it seems the right way is to do this: <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> Does this require me to set outbound_caller_id_name in *all* my directory entries? What if it's missing in one. Can I have it use a default? The next question is about the outgoing Caller-ID Name. From the SIP trace, I see the name is set both in the "From:" line and the "Remote-Party-ID:" line, in quotes (""). However, the name does not appear on the caller-ID box of the POTS line. It seems the phone company (AT&T) is getting the name from a CNAM lookup. Is this something I can change, or is it just not sent in the path between Flowroute and AT&T? -- Steve

3 2

Executing multiple applications in parallel
by JP 26 Jan '13

26 Jan '13

I want to do the following... 1. Play a long music file from a dialplan. 2. While the music is playing, I want to launch a Lua script to do some data dip. 3. Once the data is fetched and Lua exits, I want to interrupt the music and continue in the dialplan How do I do this. Any sample code or pointers would be greatly appreciated. Thanks

5 8

Re: [Freeswitch-users] PSTN via Google Talk
by Johan Brannlund 26 Jan '13

26 Jan '13

Sanjay Soni wrote: > Have you specified audio codec in autoload_config/dingaling.conf.xml > as <param name="codec-prefs" value="PCMU"/> ? I thought I had, but a look revealed that I hadn't. I put that in, and now everything is working perfectly. Many thanks! - Johan

1 0

Passing credentials when invoking "curl"
by JP 26 Jan '13

26 Jan '13

I am trying to use curl from a lua script to get data from a webservice and need to pass user/password information. Can someone help me with this? Here is what I am doing, which does not seem to work. api = freeswitch.API() agent_status_url = "-u userid:password http://mywebservice.com:8080/status/agent/test" response = api:execute("curl", agent_status_url) freeswitch.consoleLog("INFO", response) Thanks

4 8

Freeswitch TLS certificate control
by Burak Öztürk 25 Jan '13

25 Jan '13

Hi, I m a new FS user. I am trying to use FS with TLS as a SIP server. My client application is CSipSimple. According to my tests, FS do not need to verify itself to client or do not verify clients' certificates. Maybe there are some problems at the client side but i thought it is not. But I m also open for new, fresh ideas about FS-CSipSimple configuretions. I read wiki and some mail-group posts. There is a paramater "tls-verify-policy". If I didn't understand wrong, I must set it with "all" value in sofia.conf.xml. I m not sure that if I set it true or wrong. Also not sure about the syntax of sofia.conf.xml file. Can anyone tell me what is the correct syntax for sofia config file for adding a new parameter. Or if I am wrong, please, tell me the correct solution. Thank you for all. Best regards, Burak

2 1

Cannot save Voicemail message
by Emrah 25 Jan '13

25 Jan '13

Hi all, I've noticed a strange problem lately, I can't press 3 to save a VM. I use mod_shout, my VMs are in MP3 format. Log abstract: 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@16000hz 1 channels 20ms 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF 3:1120 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-you_have.wav] (en:en) freeswitch@internal> version FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Deleting works fine. Any idea? Thanks and all the best, Emrah

2 5

Voicemail and forwarding MWI
by Johan Brannlund 25 Jan '13

25 Jan '13

Hello. I recently installed Freeswitch for use as a home PBX. I was planning on using my VoIP provider's voicemail system, but I'm not sure if that's possible. In particular, I don't know how to get MWI working in such a configuration. Is there a way for Freeswitch to forward the SIP NOTIFY MWI messages from my provider to a certain extension? In other words, I am trying to do exactly the same thing that Gabriel Burca asked about a year ago (see http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078703… ). Any help would be much appreciated.

3 2

Friday Free For All
by Ken Rice 25 Jan '13

25 Jan '13

Hey Guys, Its Friday, you know what that means... Stop by the bridge and see whats going on... Officially it doesn¹t start til like 3P eastern, but its a free for all so we can start any time... Asking yourself how can I get in on this? Well sip:888@conference.freeswitch.org will get you there or use one of our handy PSTN access numbers USA +1-919-386-9900 Spain +34-91-290-12-71 UK +44-3300-100-295 Germany +49-228-9293-9009 Australia +61-7-3188-7519 Israel +972-2-372-0394 Canada +1-438-800-0531 France +33-975-181-606 Netherlands +31-858-880-387 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

2 1

Patton FXO configuration on Freeswitch
by Alessio 25 Jan '13

25 Jan '13

Hi all! I am trying to configure a Patton FXO Gateway in my Freeswitch, but I don't know how and where i have to write configuration file. Anyone can help me? Thanks, Alessio

1 0

ESL Perl Fork
by Usama Zaidi 25 Jan '13

25 Jan '13

Hi Guys, So I'm kinda stuck with this non trivial issue few might ever encounter; Please have a look at the script here http://pastebin.freeswitch.org/20508 Issue is that when the child returns it closes the esl connection and I don't receive any further events on the script, a hack would be to check the connection state each time I loop in the main thread and connect if state = 0. Ideally I would like to prevent the child to mess up the connection. Thnaks. -- I'd love to change the world, but they wont gimme the source code to it

1 0