MC. temporarily, i had to create an pagd extension in place of the ivr, but this caused issues with a bda i was using so i really need an ivr. what is the method to change or clear a realm before entering an IVR? thanks. ________________________________ From: "freeswitch-users-request@lists.freeswitch.org" <freeswitch-users-request@lists.freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, December 28, 2011 6:10 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 181 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Nat Setup Help (Ryan How) 2. Re: freeswitch versions/releases (Ken Rice) 3. FreeSWITCH Conference Call Today (Michael Collins) 4. Re: freeswitch versions/releases (curriegrad2004) 5. Re: bind digit action (Michael Collins) 6. Re: Nat Setup Help (curriegrad2004) 7. Help on RTMP (RuiQiang Huang) Hi, I've been playing with freeswitch and having a bit of trouble getting devices to register from outside my network. I want devices to be able to use a single IP and connect whether they are inside or outside of the network. Network setup is a NAT router / gateway with a static IP. I've tried a few different setups as follows: Server inside the NAT. Works well, but external devices can't register. Tried port forwarding 5060-5091 & 16384-32768 but can't get devices to authenticate properly, I'm thinking maybe the ACL or the "domain" is causing issues but I don't know how to debug this). Also I want devices to be able to roam from internal to external, so it needs to use the external IP address all the time, but NAT reflection and VOIP just don't seem to work at all (It works for mail server and web server, so I don't know why not for VOIP). Server on gateway and multihomed. Again the internal / external IP issue... but it seems to work better than behind the NAT. Have a big of trouble with "domains", clients cannot authenticate because they don't exist on a different domain or something, I don't really understand it or know how to debug other than trial error, usually fixing 1 thing breaks another thing. Server on gateway and bound just to the external IP (I think!). Everything works as I want, except internal clients hear no sound, I don't know why, they can register and call fine. I think it must be a NAT issue? I Wouldn't think NAT would be involved here... Has anyone got a good way or pointers to how I should be trying to set this up?, The only way I've got it working so far is to VPN in and use it over that, but that adds a lot of overhead and makes it less reliable... This stuff does my head in :). Thanks! Ryan FreeSWITCH hasn't had an official release in quite some time... In fact it is highly recommended that you check out a current copy from git, build it and test it to make sure it works for your specific environment. There are plans to change this in the near future K On 12/28/11 6:54 AM, "Arnuld Uttre (Phonologies)" <arnuld@phonologies.com> wrote:
At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch and came to know that 1.0.7 was never an offical release. Even files.freeswitch.org does not have it. Last offical release was 1.0.6 and as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git for now which is extremely stable.
I just want to conform that 1.0.7 was not official release. And is there any document on versioning/release system of freeswitch ?
Hello all! Today on the FreeSWITCH conference call we will be discussing the event socket and how to generate calls. Please join us! The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_28 Talk to you soon! -Michael1.0.7 was really a snapshot from the git tree :P On Wed, Dec 28, 2011 at 8:10 AM, Ken Rice <krice@freeswitch.org> wrote:
FreeSWITCH hasn't had an official release in quite some time... In fact it is highly recommended that you check out a current copy from git, build it and test it to make sure it works for your specific environment.
There are plans to change this in the near future
K
On 12/28/11 6:54 AM, "Arnuld Uttre (Phonologies)" <arnuld@phonologies.com> wrote:
At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch and came to know that 1.0.7 was never an offical release. Even files.freeswitch.org does not have it. Last offical release was 1.0.6 and as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git for now which is extremely stable.
I just want to conform that 1.0.7 was not official release. And is there any document on versioning/release system of freeswitch ?
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maybe you could just change the digit action realm prior to entering the IVR? -MC On Sat, Dec 24, 2011 at 5:39 AM, Rodney <notlikeme75@yahoo.com> wrote:
Michael,
I got BDA to work by manually configuring the dialplan, seems fusionpbx was not doing it right when i added the actions. but now that I have it working, my issue is using 0 as a transfer back to main menu. is there a way to clear digit action on an IVR menu?
scenario
i have 9 static conference rooms
bind digits are
0 transfer back to ivr room list 6 to move back in room list 7 to move up the room list
to make this work, i clear bind digit on every conf
room extension. the problem is my ivr has option 0 to go back to previous menu, but its not seen once you go into the room list bindings. I know making the IVR binding a different number would work, but I would like to keep with the consistent 0 to go back to previous menu. if there was a method to clear bind digit in the IVR, that would be great. I have tried before the ivr answer and after, with no luck.
ie. pressing 0 , should go to room menu, and 0 again should go to main menu, but as of now, pressing 0 is binded to room menu only :(
another good thing would be, i could set an expression with boundaries so i would only need 1 extension or 1 conference control like a low boundary of 501 and a high boundary of 509. so if someone presses 6 it would move down the list variable {6=current conf # - 1 but if in room 1, go to room 9, and if in room 9 pressing 7 would start over at 1}
i accomplished this the long way with 9 separate extensions using bind digits to move up or down the list. i had to manually add a transfer for the next extension. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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That's why there's ext-rtp-ip and ext-sip-ip varibles in the sofia conf file! You're getting close, however :P On Wed, Dec 28, 2011 at 7:58 AM, Ryan How <rhow@exemail.com.au> wrote:
Hi,
I've been playing with freeswitch and having a bit of trouble getting devices to register from outside my network. I want devices to be able to use a single IP and connect whether they are inside or outside of the network. Network setup is a NAT router / gateway with a static IP. I've tried a few different setups as follows:
Server inside the NAT. Works well, but external devices can't register. Tried port forwarding 5060-5091 & 16384-32768 but can't get devices to authenticate properly, I'm thinking maybe the ACL or the "domain" is causing issues but I don't know how to debug this). Also I want devices to be able to roam from internal to external, so it needs to use the external IP address all the time, but NAT reflection and VOIP just don't seem to work at all (It works for mail server and web server, so I don't know why not for VOIP).
Server on gateway and multihomed. Again the internal / external IP issue... but it seems to work better than behind the NAT. Have a big of trouble with "domains", clients cannot authenticate because they don't exist on a different domain or something, I don't really understand it or know how to debug other than trial error, usually fixing 1 thing breaks another thing.
Server on gateway and bound just to the external IP (I think!). Everything works as I want, except internal clients hear no sound, I don't know why, they can register and call fine. I think it must be a NAT issue? I Wouldn't think NAT would be involved here...
Has anyone got a good way or pointers to how I should be trying to set this up?, The only way I've got it working so far is to VPN in and use it over that, but that adds a lot of overhead and makes it less reliable...
This stuff does my head in :).
Thanks! Ryan
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Hi, I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, mod_rtmp.la and put it under freeswitch/mod/. My freeswitch version is FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) My rtmp.conf.xml is <configuration name="rtmp.conf" description="RTMP Endpoint"> <profiles> <profile name="default"> <settings> <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" /> <!-- If this is set to true, no unauthenticated inbound calls will be allowed --> <param name="auth-calls" value="true" /> <!-- How much time should the clients buffer the media stream (miliseconds) --> <param name="buffer-len" value="50" /> <!-- Sets the maximum size of outbound RTMP chunks --> <param name="chunksize" value="512" /> </settings> </profile> </profiles> </configuration> However, I didn't see freeswitch listening on 1935 port and in fs_cli when I run freeswitch > rtmp status It's showing unknown command. My question is, is there anything else needed to get RTMP working? How to know which modules are loaded? Thanks. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org