- FreeSWITCH-users - FreeSWITCH Mailing Lists

Gateway with no registration
by Cliff Wells 09 Feb '11

09 Feb '11

I'd like to setup a gateway with no registration (other end allows by IP). <include> <gateway name="outbound"> <param name="username" value="none"/> <param name="password" value="none"/> <param name="proxy" value="x.x.x.x"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> As I understand it, the username and password shouldn't matter since registration is challenge/auth. What happens is I get a 403 when FreeSWITCH attempts to register (before any calls are placed), so FreeSWITCH has the gateway marked as FAIL_WAIT. Is this proof the other end is requiring registration? Or is FS expecting a challenge, getting none and marking the gateway down? Most of the solutions I've seen for this type of setup involve using a dialplan rather than a gateway, but my dialplans are served via mod_xml_curl and expect a functional gateway named "outbound" on the FS box. I'd rather not clutter up the xml dialplans with a bunch of special cases, so a named gateway is preferred. The problem certainly may lie on the other end (I don't have control of that system), but I'm not sure how to verify that. Regards, Cliff

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RTP keep alive
by Sam 09 Feb '11

09 Feb '11

Hello, We have a situation wherein we need to keep alive RTP ,is there any parameter to do that, because when someone is on long conversation and not talking fro brief duration the call disconnects . Regards Sam

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Re-order or Invalid
by Peder 09 Feb '11

09 Feb '11

What is the correct way to send a re-order or invalid message/tone to a caller? We have a couple hundred DIDs and quite a few of them are not used. If someone calls into one of those, we just want to send a re-order or busy tone. My preference would be to NOT answer the call as I would prefer not to be charged for the call if I can avoid it. Peder

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FreeSWITCH Conference Call Today
by Michael Collins 09 Feb '11

09 Feb '11

Hey gang, We decided that we'd have an open call today. No real agenda items, but we're happy to hang out and maybe answer a few questions. -Michael

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ext-rtp-ip value behind NAT
by mazilo 09 Feb '11

09 Feb '11

With FreeSWITCH Version 1.0.head (git-2944364 2011-02-04 16-53-38 -0600) on my Seagate DockStar, I configured my jingle profile using the following two different options: 1. 2. With option #1, no incoming call and caller got intercepted by GV voicemail. fs_cli dumped this http://pastebin.com/8sJjUZx1 message as seen on line# 26 and here is the http://pastebin.com/gnb8dXmW output of dl_debug on. With option #2, the call comes in. When I picked up to answer the call, no audio and caller still heard the ringing tone until the call got intercepted by GV voicemail. fs_cli dumped this http://pastebin.com/fJ1s14tX message and the http://pastebin.com/mHBFECs3 output of dl_debug on. If anyone here has any idea how to fix this, I sure would appreciate that. Thanks. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ext-rtp-ip-value-behind-NAT-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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G729
by Sam 09 Feb '11

09 Feb '11

Hi, Which type of G.729 codec does freeswitch uses in passthrough mode ? Is it g729a or g729b or g729ab . Regards Sam

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Skype conference call
by Dmitry Bely 09 Feb '11

09 Feb '11

Can I participate in a Skype conference call via FreeSWITCH/Skypopen? Is there any howto on that? - Dmitry Bely

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Can ESL work from remote computer?
by Malay Thakershi 09 Feb '11

09 Feb '11

Hello, I am trying to connect to FS server via ESL from another computer on the LAN. ------------- eslConnection = new ESLconnection("10.25.20.202", "8021", "ClueCon"); if (eslConnection.Connected() != ESL_SUCCESS) { return "Fail"; } ------------- But I keep getting not connected. I checked firewall. I tried addressing FS computer by name/IP. Do I need to change anything in the FS config? Please share your views. Malay

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setting/getting variable in callcenter
by Eli Hayun 09 Feb '11

09 Feb '11

Hi I am using mod_callcenter and have a question 1) How do I get the value from a callcenter variable from a php or lua script? suppose that I want to change the MOH of a queue: I see that the variable "cc_moh_override" is the one that I have to change, but how do I do that? I dont have the session (i am in a php /lua script, that run outside FS and calling to it with ESL api) 2) I am using ESL call to get the list of agents but I want to do more then just see the list of agents. I see that when the phone is ringing in an agent phone the status changed to "Receiving" but without the uid or the caller phone number. How do I get this information (I prefer to get this without using an event) Thanks -- Eli Hayun Hebrew University Jerusalem

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Jira status emails seems to be down since some time?
by Peter Olsson 09 Feb '11

09 Feb '11

I think this has been reported before, but it seems like Jira status emails are not being sent anymore - and I suspect this might cause the core devs to miss out some of the updates :) Just wanted to let you know.. /Peter

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