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More Nibblebill questions
by Andrew Cassidy 02 Dec '13

02 Dec '13

So this is starting to get complicated and it looks like I might have to write some lua to do this... So, say I want to bridge a freephone number to both a landline and a mobile in parallel. All 3 potential legs need different call rates, only 2 will be used when the call is picked up, of course, and I ideally need the CDR and cost for each leg separate. Here's an idea of what I have so far: <action application="set" data="nibble_rate=0.01" /> <action application="bridge" data="[nibble_rate=0.01]sofia/gateway/x/landline,[nibble_rate=0.06]sofia/gateway/x/mobile" /> This doesn't work, and neither does setting the variables nolocal. Can this be done in this way? I can probably work around this by exporting the incoming rate as nolocal and the outgoing rate on the incoming channel. Confusing but might actually work. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F <info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

2 6

reinstall module
by Joli Martinez 02 Dec '13

02 Dec '13

Hello I have the dignaling module is damaged. I need to reinstall it. How can I recompiile it again? thanks,

2 1

Originate and g729 codec license
by Raj Saini 02 Dec '13

02 Dec '13

Hi, I have a question regarding usage of g729 codec for one of my application. My application dial outs using originate command. My SIP trunk providers supports g729. Do I need a g729 license? Thanks, Raj

3 4

issue on FreeSWITCH 1.4 webrtc for windows
by Vincent Xia 02 Dec '13

02 Dec '13

*has anybody ever tested freeswitch 1.4 with webrtc support under windows platform?* *i tried to build and run fs 1.4 in windows 7 and windows server 2008 as the webrtc server but no success, i simply uncommented <param name="ws-binding" value=":5066"/> and had the jssip client connect to it, but nothing happened on the fs console, the log from chrome said it's "Connecting to WebSocket URI * *ws://fs_host_ip:5066 jssip-0.2.1.js:619", * *i did netstat -ano and saw freeswitch is already listening to 5066 and chrome is connecting to is* *TCP local_ip:59687 fs_host_ip:5066 ESTABLISHED 12316* *(fs uses 5066, chrome uses 12316)* *so did i miss something to have freeswitch up for webrtc in windows?* *ps:the firewall is off and the same issue happens with tryit.jssip.net <http://tryit.jssip.net>.* *Regards,*

1 1

Adjust Contact port in SUBSCRIBE request with TLS
by Jon Bruel 02 Dec '13

02 Dec '13

I have a problem with the Contact header in a SIP SUBSCRIBE request. The situation is the following: FS is setup as a presence client using TLS as the transport layer. The server is to send presence information about a large number of mobile telephones. The FS sends a SUBSCRIBE to the server, on a newly setup SSL connection. The SUBSCRIBE has in its Contact address 5061 as the port to be used for NOTIFYs to be send by the server. I would like to use another port, namely the port used by the server in the new setup connection. It that possible and how? This would allow the server to respond on an established TLS connection. I know there is another solution the problem involving the setup of a new secure connection initiated by the server, but I would prefer to change the port in our SUBSCRIBE request. I should mention that we do not use registration towards the server. Thanks /Jon

1 0

FreeTDM and ring_ready
by Siri MM 02 Dec '13

02 Dec '13

Hi, I have been using Sangoma A102 card with wanpipe-3.5.28 and FS 1.2.8. At a couple of different customer sites, common issue reported is, when a call is made from external world (mobile/fixed line), although the destination extension rings, no ringback is heard at the caller end. Extension dialplan is to bridge to the desired extension, something like: <action application="bridge" data="user/3001" /> On capturing the isdn traces, I see that packet exchanges involve SETUP, CALL PROCEEDING, and ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band") After experimenting a bit, what I find is, ringback is heard if I were to modify my dialplan as follows: <action application="ring_ready" /> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="user/3001" /> This time, ISDN traces have the following: SETUP, CALL PROCEEDING, ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band"), and PROGRESS (with a progress indicator saying "In-band information or an appropriate pattern is now available") My question is, while working with FreeTDM, do I need to always set ringback explicitly? Could somebody please help me understand this correctly? Thanks!

2 4

Disastery Plan
by Nikita Tarasov 02 Dec '13

02 Dec '13

Hi guys. We have loadbalancer(opensips) and a huge amount of incoming calls going to ONE sip server, which is unscalable and priopretary. I need to create some kind of disastery plan for this server service is down or on high load. The main idea of this plan is, when disastery happends, to balance all incoming calls to freeswitch and keep them with music, while main server is down. On recovery calls should be forwarded to main server with the same destination number as they was. I have tryed to make it with mod_callcenter. It was done by placing calls in queue and when main server is on and re-registered on freeswitch, calls was forwarded. But without destination number. And I can't copy destination number to forwarded calls. Can you help me?

3 6

freeswitch stopped working
by Joli Martinez 30 Nov '13

30 Nov '13

Hello, My freeswitch server does not want to start anymore. When I try to start it from the init.d script It say "freeswitch dead but subsys locked". When I go to "/usr/local/freeswitch/bin" and start it that way it does start but I am not able to dial anyone internal I get "switch_core_session.c:2616 Invalid Application transfer" The last thing I remember doing was last week I installed a python module but did not restart it until now. I also uncommened out the python module in modules.conf.xml and same result. Please help Thanks,

2 1

(no subject)
by Vincent Xia 30 Nov '13

30 Nov '13

has anybody ever tested freeswitch 1.4 with webrtc support under windows platform? i tried to build and run fs 1.4 in windows 7 and windows server 2008 as the webrtc server but no success, i simply uncommented <param name="ws-binding" value=":5066"/> and had the jssip client connect to it, but nothing happened on the fs console, the log from chrome said it's "Connecting to WebSocket URI ws://fs_host_ip:5066 jssip-0.2.1.js:619", i did netstat -ano and saw freeswitch is already listening to 5066 and chrome is connecting to is TCP local_ip:59687 fs_host_ip:5066 ESTABLISHED 12316 (fs uses 5066, chrome uses 12316) so did i miss something to have freeswitch up for webrtc in windows? ps:the firewall is off and the same issue happens with tryit.jssip.net. Regards,

1 0

Adjust Contact port in SUBSCRIBE request with TLS
by Jon Bruel 30 Nov '13

30 Nov '13

1 0