May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dialplan action sched_hangup
by Jim Burke 20 May '09

20 May '09

Hi All, Can someone clarify the usage of the 'sched_hangup' application in mod_dptools. In the wiki it suggests the timer starts when the call is answered, however testing proves this not to be the case and the call is released while in the ringing state if the timer is short enough. Or should it be used in conjuction with channel variable 'execute_on_answer'. Thus forcing setting of the timer at this point of the call/ If you need the debug or any other info please let me know. Thanks, Jim

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unable to recieve audio on endpoints
by ravi hum 20 May '09

20 May '09

I have installed Freeswitch and configured to use SIP Proxy Server for making calls. I tested Originate and Bridge commands and both sides of the call are getting connected. Now the problem is :- --I am not recieving any voice/audio on either endpoint of the call ( both endpoints are mobile number) --firewall is off on the server *call request is send from FreeSwitch to SIP Proxy server ( FreeSwitch --->> SIP Proxy Server) please let me know how to solve this issue. -- View this message in context: http://www.nabble.com/unable-to-recieve-audio-on-endpoints-tp23629268p23629… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Unable to successfully bridge calls to an "external" user
by David Robinson 20 May '09

20 May '09

> My suspicion is that the RTP traffic isn't traversing the NAT > properly. You > may have to configure the routers at both ends to forward the RTP > packets to > the correct destinations. There is a good discussion of NAT on the > wiki. Situation: FS (10.0.0.12) -> DMZ (124.254.81.250) -> Internet -> NAT (203.206.171.118) -> Softphone (10.0.0.2) The problem is there's so much discussion of NAT that I'm not sure where to start. OK the problem is that I can't control the "external" user's router so I need a solution that works by only fixing the FS end. I've put my FS in the DMZ, but of course it's still got a local LAN IP address. Is there something I can configure to make FS realise that it _doesn't_ need to use NAT ? Whenever my softphones register to FS they register as UDP-NAT. Can I prevent that and make them register as regular UDP ? It would seem like they don't need to be in NAT mode since FS is in a DMZ, or do they ? I tried setting inbound-late-negotiation in my external (is this right ?) SIP profile and added proxy_media to my extension configurations in the dialplans, but this made no difference. It's possible that I haven't done this in the right spot or something. The other thing that looks promising is on http://wiki.freeswitch.org/wiki/External_profile which gives an example of a softphone registering to a NAT'd FS from outside on the internet (Switch with External Softphone example) which suggests I create a new external profile on a different port. I've done this and the user's softphone can register fine, but when he makes calls we still get no audio, presumably from lack of RTP data. I then tried adding in values for rtp-ip, sip-ip, ext-rtp-ip and ext-sip- ip on the new external profile to see if that made any difference but it didn't. Step 6 of the example says "reference the caller from your FreeSWITCH system as: sofia/external5090/<caller extension>@x.x.x.x: 5090". I'm not sure what that means. Do I have to change something else to make it "reference" the caller by that external profile ? I figured it must be at least using that external profile because the phone is successfully registering on port 5090, but I'm not sure if I have to do something different to route incoming calls from the main external profile to the new 5090 one. I'm just not sure which NAT-related solution I'm supposed to be using. The External_profile wiki page example for the external softphone seems to fit my situation but didn't solve anything. The proxy_media solution seemed promising but had no real effect. It seems to me that the solution has something to do with having FS know that it's in a DMZ and that it doesn't need to do any NAT traversal, thereby making it think it's got a live internet IP and therefore only the external user would be using NAT traversal. I hope someone can give me some insight into which particular NAT- related solution I need because there seems to be dozens of ways to deal with this problem and I can't figure out which applies.

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Logging 503's or other errors
by Ron McCarthy 20 May '09

20 May '09

Hi list, Ive been trying to find a way to log 503's, 480's and other SIP response codes. If we have continue_on_fail=true and have multiple gateways for the call to go out, if the 1st,2nd or whatever gateways fail can we log it somehow? We'd like to know if a carrier is having issues or not letting us send calls for some reason, from what I can tell I only show one CDR get written and that's at the end of the call, so it says nothing about the gateways we tried to send a call before and if they failed. Any ideals on how to do this? Im using the XML CURL dialplan if that matter. Any ideals how this could be setup so we can keep track of what is going on? Thanks

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Testimonials
by Maxim Tsvetov 20 May '09

20 May '09

Hello, Our company want to use Freeswitch and now we testing this solution in the lab. If someone already using Freeswitch as office pbx, ivr or any other commercial purposes could you please share you experience (and if it is not a secret - company name)? Regards, Maxim Tsvetov -- View this message in context: http://www.nabble.com/Testimonials-tp23611131p23611131.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Freeswitch hosting
by Erez Simon 19 May '09

19 May '09

Hi, I want to get a Freeswitch instance running on a hosted service, such as the Amazon EC2 cloud. Can anyone help in recommending Freeswitch hosting alternatives in terms of price, scale, support, quality, configurability, etc.? much obliged, Erez

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As seen on DIGG
by Anthony Minessale 19 May '09

19 May '09

please find it in your hearts to digg this story. http://digg.com/search?s=skype+switch -- 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%3Aanthony.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%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Load test issue
by Niall Crosby 19 May '09

19 May '09

Hi, Am running SIPP on a pretty fresh default install of FS and am only getting 3 calls processed per second when put under load. I see others are getting in the region of 100 calls per second, so am guessing my config has something suspect? And not just a 'tuning' issue as the difference is huge. CPU usage is <3% and hard disk sounds very active (however a 10ms operation every 200ms would sound busy to a human). So puzzles me why it is taking so long. My scenario is the SIPP provided UAC pointed to extension 3000 (the default set up conference with FS). Want to create a 100+ participant conference where 3 calls per second will not suffice. ;-( thanks, Niall. System Info: Installed from default Windows MSI installer. 6GB Ram Windows 7 64 Bit Intel Core i7 920 CPU

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New ZDNet Article About FreeSWITCH
by Michael Collins 19 May '09

19 May '09

Gang, Please visit the main FreeSWITCH site and check out the links<http://www.freeswitch.org/node/185>to the new ZDNet story. Please spread the word with all of the link sharing sites that you have. Thanks! Michael S Collins http://www.cluecon.com

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Dialplan bridge action
by Larry Marshall 19 May '09

19 May '09

What are the differences between these two actions? <action application="bridge" data="USER/1001@$${domain}"/> <action application="bridge" data="sofia/internal/1001%$${domain}"/> Thanks, Lars

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