December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

create temporary code for 1 on 1 conference
by Rodney 30 Dec '11

30 Dec '11

Is there a method i can use to generate a 3 digit number that syncs to that crazy long uuid number that stays valid until that caller hangs up? I would like to create this code and play the code before menu options then play it on demand from an extension. I would also like a method to exchange this code with another current caller to allow them to speak in a 1 on 1 conference together. how do i put these two individuals into a uuid bridge without using an operator to do so? I would like them to exchange codes verbally and press an option that asks for the other persons code and "match" them into a conf.

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BDA on IVR
by Rodney 30 Dec '11

30 Dec '11

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(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)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(a)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(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)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(a)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(a)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(a)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 ? >> >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)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 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org 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(a)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(a)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 > >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > 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(a)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(a)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 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org 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" />  <param name="auth-calls" value="true" />  <param name="buffer-len" value="50" />  <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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Asterisk and Freeswitch on the same machine!
by Anita Hall 29 Dec '11

29 Dec '11

Hi friends Since my efforts to find an open source solution for SS7 on freeswitch did not meet any success, we had to resort to a cheap trick! chan_ss7 <-> Asterisk <-> FreeSWITCH where * and FS are on the same machine. Since we wanted to do only 120 calls per machine, it has worked out pretty well. Doing more than 150 calls is meeting a CPU bottleneck (surprisingly from FS side!) One interesting thing, we noted was that FS was taking considerably more CPU cycles than Asterisk (sometimes thrice as much!) when the calls were sent from Asterisk to FS. The dialplan on Asterisk was playing gsm audio (demo context) while FS dialplan was just sleeping! I believe, this is surprising? When doing SIP calls from FS to Asterisk, the CPU loads are comparable. I was under the impression that FS take less CPU cycles? My team member is writing a report with all performance stats, shall share when it is done. Bye, Anita.

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Using xmpp to control conference
by fieldpeak 29 Dec '11

29 Dec '11

Dear friends, Could you any one give any tips or an example how to use xmpp protocol to control FS to realize conference. i see the wiki, there is few docs for mod_xmpp_event ( http://wiki.freeswitch.org/wiki/Mod_xmpp_event) also i've considered using mod_socket_event to control conferece, while i prefer using xmpp since i've deployed xmpp server... Thanks! -- Regards, Charles

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Error "Channels not ready" when bridge call in Lua
by dma 29 Dec '11

29 Dec '11

Dear support, I have a Lua script that accepts inbound call, creates outbound call, and then bridges the 2 sessions. Its code is simply like this: -------------------- session:answer() session2 = freeswitch.Session(callstring) if (session2:ready()) then freeswitch.bridge(session, session2) end -------------------- However, when bridge() is called, the call immediately hangs up. The following is reported: 2011-12-13 18:08:00.674769 [ERR] switch_cpp.cpp:1247 Channels not ready 2011-12-13 18:08:00.674769 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup By searching through the forum, I find a previous report of the same error with Lua script. However no solution is provided for that ticket. Please kind advise how to fix this problem. Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Nat Setup Help
by Ryan How 29 Dec '11

29 Dec '11

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

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bind digit action
by Rodney 28 Dec '11

28 Dec '11

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.

2 1

freeswitch versions/releases
by Arnuld Uttre (Phonologies) 28 Dec '11

28 Dec '11

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 ? -- Arnuld Uttre Systems Software Engineer arnuld(a)Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992

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FreeSWITCH Conference Call Today
by Michael Collins 28 Dec '11

28 Dec '11

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! -Michael

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Monitor performance of modules
by Muhammad Naseer Bhatti 28 Dec '11

28 Dec '11

Hello, Is it possible to monitor performance of modules loaded by FS? For example, mod_nibble or say mod_distributor. Is there a way to know how much time the module took from the time when called by FS till the module released data back to FS? Thanks.

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