[Freeswitch-users] Dec 20 Global Freeswitch All Free SW HW Culture meeting - BerkeleyTIP
Hi FreeSwitchers, Anthony Anthony - Thanks for letting me post the monthly announcement here. :) FSers: We are working toward moving to FS from Asterisk. We welcome you to join the BTIP Global VOIP bimonthly meetings, if you like, help us get the FS sw running on our server. :) = A great December Solstice to you yours. :) JOIN the Global All Free SW, HW, Culture meeting via VOIP Dec 20 Sunday, 12N-3PM (Pacific = UTC-8) = 3P-6P Eastern = 8P-11P UTC [Jan 2009 meetings: 2nd, 17th - mark your calendar] http://sites.google.com/site/berkeleytip/schedule == WATCH some VIDEOS: Mark Shuttleworth Interview - 10.04 Lucid Larynx Learning from Code History , Andreas Zeller Why does my program fail? Your version history might have the answer. Audio Hardware Enablement Session, UbuntuDevelopersSummit in Dallas Distributed Development, UDS in Dallas Splunk, Jeremy Thurgood CLUG Upstart, Stefano Rivera CLUG Interfacing with the real world, Mark Ter Morshuizen, Marc Welz CLUG Accelerating Graphics; Camp KDE 2009 http://sites.google.com/site/berkeleytip/talk-videos == Join the MAILING LIST tell us which videos you will watch why: http://groups.google.com/group/BerkTIPGlobal == JOIN the meeting via IRC VOIP: Come discuss any everything, work on your individual or group projects. HOT TOPICS: Ub or KUb 9.10?, Ubuntu 10.04 plans, Android, Python3000 in 2010? Start on the #berkeleytip irc.freenode.net channel, we'll help you get your VOIP system up working. For VOIP SW, connection info, see: http://sites.google.com/site/berkeleytip/remote-attendance Berkeley meeting LOCATION: Watch the website mail list for latest details, perhaps at the Berkeley Public Library, or a cafe, due to Free Speech Cafe closed for winter break. http://sites.google.com/site/berkeleytip/ == OPPORTUNITIES to VOLUNTEER or learn new JOB SKILLS for 2010: Help set up our: Mailing list, FreeSwitch VOIP server, website http://sites.google.com/site/berkeleytip/opportunities Inquire discuss at the meeting. == For Forwarding - You are invited to forward this announcement wherever it would be appreciated. ___ 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
[Freeswitch-users] No audio after Remote SDP:
Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2...@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? 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
Re: [Freeswitch-users] How to debug TLS handshake errors?
I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __Yehavi: 2009/12/17 Yehavi Bourvine yehavi.bourv...@gmail.com I am trying Audiocodes and Vegastream ATAs, and work with either the manufacturer or the local representative here. On SNOM I managed to make it work, and will try Polycom soon (once I manage to grab one unit from our users...). Thanks, __yehavi: 2009/12/17 Brian West br...@freeswitch.org Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: You could try ssldump: http://www.rtfm.com/ssldump/ ___ 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 ___ 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
[Freeswitch-users] fs_cli connection error
Trying to setup a new config in the pfSense 1.2.3 final package and when I try to connect to the console I get an auth error? # ./fs_cli -H 10.0.0.1 [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Authentication Error] I tried to search for docs to indicate where one might set the password for this (it never used to have one) but I could only see docs suggesting to provide one, not set one. There is no .fs_cli_conf anywhere. Socketstat shows it listening on 8021... Thanks! jlc ___ 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
[Freeswitch-users] Interfacing to RabbitMQ
Hi, I'll appreciate if someone who has a practice in interfacing FreeSWITCH to RabbitMQ or suggestions could share it to me. Regards, -- afshin ___ 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
Re: [Freeswitch-users] fs_cli connection error
The password is set in conf/autoload_configs/event_socket.conf.xml -MC Sent from my iPhone On Dec 20, 2009, at 7:58 AM, Joseph L. Casale jcas...@activenetwerx.com wrote: Trying to setup a new config in the pfSense 1.2.3 final package and when I try to connect to the console I get an auth error? # ./fs_cli -H 10.0.0.1 [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Authentication Error] I tried to search for docs to indicate where one might set the password for this (it never used to have one) but I could only see docs suggesting to provide one, not set one. There is no .fs_cli_conf anywhere. Socketstat shows it listening on 8021... Thanks! jlc ___ 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 ___ 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
Re: [Freeswitch-users] Interfacing to RabbitMQ
On Sun, Dec 20, 2009 at 07:38:27PM +0330, afshin afzali wrote: Hi, I'll appreciate if someone who has a practice in interfacing FreeSWITCH to RabbitMQ or suggestions could share it to me. You could try to use mod_erlang_event and the erlang rabbitmq client (in native message passing mode). I've never worked with rabbitMQ however, I just know a little about it. Andrew ___ 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
[Freeswitch-users] Authenticating end points by IP
People, Please do excuse me if this is a FAQ. I've so far not worked out a way to implement IP authentication effectively. I have a number of gateways/end points/...etc. hitting the switch without registration and originating calls to a number of upstreams I have configured. So far, everything is smooth and FREESwitch is legendary in every way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon machine running CentOS and CPU usage is barely a blip. My problem is authentication. I need a method to authenticate end points based only on their IP addresses. I am currently filtering access through the firewalls, but I would really like to delegate this task to FS in prep for an SBC setup I'm working on. If I remove the firewall filters, then anyone is able to get in. I could't workout precisely how ACLs work in FS from the WiKi and the documentation, and I haven't been able to make sense of how Digest functions either. Can anyone shed some light on those two areas ? I would like to really get to the bottom of this and update the WiKi pages with the working setup once done. Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS ___ 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
Re: [Freeswitch-users] Authenticating end points by IP
Check out: http://wiki.freeswitch.org/wiki/ACL#Users It'll automatically add users with a cidr= attribute to the ACL list. This way you can set channel variables in the users and use them through your dialplan, all authenticated by ip address. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 20-Dec-09, at 3:19 PM, Ahmed Naji wrote: People, Please do excuse me if this is a FAQ. I've so far not worked out a way to implement IP authentication effectively. I have a number of gateways/end points/...etc. hitting the switch without registration and originating calls to a number of upstreams I have configured. So far, everything is smooth and FREESwitch is legendary in every way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon machine running CentOS and CPU usage is barely a blip. My problem is authentication. I need a method to authenticate end points based only on their IP addresses. I am currently filtering access through the firewalls, but I would really like to delegate this task to FS in prep for an SBC setup I'm working on. If I remove the firewall filters, then anyone is able to get in. I could't workout precisely how ACLs work in FS from the WiKi and the documentation, and I haven't been able to make sense of how Digest functions either. Can anyone shed some light on those two areas ? I would like to really get to the bottom of this and update the WiKi pages with the working setup once done. Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS ___ 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 ___ 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
Re: [Freeswitch-users] ACLs through proxy
Then it would appear that my original suggestion to use mod_xml_curl
would be best for now and you may need to offer a bounty for this
feature as others have suggested. Based on the sofia related snippets
presented--I would assume it would be trivial to implement since most of
the functionality is already there it just needs to be enhanced for your
purpose. It would also be extremely easy to do this in OpenSIPS as well
(using blacklists or avpops).
Just so that I understand your dilemna, you want to reject an incoming
REGISTER associated with a specific user unless it comes from a fixed
location and if it does, you want to simply challenge it as usual to
prevent toll fraud?
I have found that its best to mitigate an attack at ingress before it
even makes it to critical infrastructure (media gateways,
application/media servers, etc.).
-metik
Bill W. wrote:
Hey Metik,
Yes. Well, actually, I can have the cidr in two places in the directory.
user cidr=190.218.97.83/32 id=testphone01
params
param name=auth-acl value=190.218.97.83/32/param
From what I understand the cidr= parmeter is used in conjunction with
the apply-inbound-acl parameter in the sofia profile to just allow
someone to make calls from a certain IP without authenticating.
And from what I understand the auth-acl= parameter is used to restrict a
user to a particular cidr, but the user has to authenticate as well.
*The second feature is the one I want to use.* I want to force users to
authenticate, but only allow that authentication from a particular cidr
as an added measure against toll fraud.
And this appears to be causing the issue. Because once I specify the
auth-acl parameter in the directory, sofia-reg enforces that acl. And
unfortunately it's using the IP of the proxy, not of the user-agent.
I looked in sofia.c and found this comment:
/*
* if network_ip is a proxy allowed to send calls, check for auth
* ip header and see if it matches against the inbound acl
*/
And this coincides with my testing.
I have param name=apply-proxy-acl value=ip_of_proxy/ in my
profile. I have my proxy sending the X-AUTH-IP header (verified with
tcpdump). And yet the REGISTER is still being denied.
So it appears that the apply-proxy-acl is set up to work with the
apply-inbound-acl ( to allow users from an IP without authenticating)
But that hasn't been carried over to sofia_reg.c, which appears to
simply check the IP of who FreeSWITCH is talking to against the auth-acl
cidr specified in the directory. (Line 1926)
So I guess the question is, is my analysis correct?
Thoughts anyone?
Thanks,
Bill
Metik wrote:
Bill,
I think you would add this to the user profile in the directory. The
brian.xml example (located in ${confdir}/directory/) provided with the
default/sample configuration files demonstrates how to to do this by
introducing a cidr attribute to the the user element.
Example:
user id=7105551212 cidr=127.0.0.0/8//
params
param name=password value=opensaysme/
param name=vm-password value=14916/
/params
variables
variable name=user_context value=default/
/variables
/user
http://wiki.freeswitch.org/wiki/Acl; contains some great info
(including a relevant example).
-metik
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Re: [Freeswitch-users] No audio after Remote SDP:
You'll need to fix your device to know its IP and it should stop doing that. /b On Dec 20, 2009, at 5:58 AM, Mark Campbell-Smith wrote: Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2...@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? 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
Re: [Freeswitch-users] No audio after Remote SDP:
DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt. I have a similar setup (and problem) - the wiki documentation refers to it as "double nat". Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). According to the documentation, and my limited experience, you have at least 2 options for NATted endpoints. However, I am unable to make the second work. 1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/ to the directory xml file that describes your spa3102 endpoint Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. Anyway, hope this helps you with your SPA3102. Mark Campbell-Smith wrote: Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2...@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? 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 ___ 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
Re: [Freeswitch-users] No audio after Remote SDP:
On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt. Welcome to the community. I have a similar setup (and problem) - the wiki documentation refers to it as double nat. Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). Your endpoint need only insert rport and FreeSWITCH will do the right thing. 1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add variable name=sip-force-contact value=NDLB-connectile-dysfunction/ to the directory xml file that describes your spa3102 endpoint The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar. Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. Anyway, hope this helps you with your SPA3102. Bottom line is enable rport and use stun on the SPA and it'll just work. /b ___ 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
Re: [Freeswitch-users] How to debug TLS handshake errors?
You have to watch it with TLS. Make sure your distro didn't mess up your SSL libs due to the recent vulnerability found. I havn't tested with my polycom in a few weeks but it was working on my Polycom after I uploaded the ca cert and marked it as trusted/used on the phone. /b On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __Yehavi: ___ 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
Re: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports?
The funny part is... it won't matter. Their are times when people post questions or issues and its well into debugging the issue before we realize oh, you're on windows?. For the most part the windows installer is one of the most popular files on our website. /b On Dec 19, 2009, at 10:18 PM, Jason White wrote: Gabriel Gunderson g...@gundy.org wrote: Funny that you assume his desktop is running Windows (maybe it is). I would have guessed that the average person on this list doesn't run Windows on the desktop. But, what do I know? Some of us on the list have never run Windows on anything. It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone via a USB head set, and also handling my Snom 320 SIP phone. ___ 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
Re: [Freeswitch-users] No audio after Remote SDP:
Thanks Brian and Gad, I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102. These are the options that I have set (according to the 3102 manual). • Handle VIA received: yes • Handle VIA rport: yes • Insert VIA received: yes • Insert VIA rport: yes • Substitute VIA Addr: yes • Send Resp To Src Port: yes • STUN Enable: Choose yes. • STUN Server: stun.freeswitch.org I assume that is all is needed? On Mon, Dec 21, 2009 at 9:36 AM, Brian West br...@freeswitch.org wrote: On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt. Welcome to the community. I have a similar setup (and problem) - the wiki documentation refers to it as double nat. Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). Your endpoint need only insert rport and FreeSWITCH will do the right thing. 1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add variable name=sip-force-contact value=NDLB-connectile-dysfunction/ to the directory xml file that describes your spa3102 endpoint The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar. Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. Anyway, hope this helps you with your SPA3102. Bottom line is enable rport and use stun on the SPA and it'll just work. /b ___ 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 ___ 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
[Freeswitch-users] Setting Restrictions on Default Dialplan
Hi Sir,
How can I allow international calling in the dialing plan but for
select accounts only?
For example i want to restrict 855 to call this ip address
182.138.252.12 using the default configuration.. Does this command should be
put in the default.xml or in the default folder and the filename is
00_restict.xml?
extension name=dialsample
condition field=destination_number expression=^(855)$\
action application=set
data=effective_caller_name=${effective_caller_id_name}/
action application=set
data=effective_caller_number=${effective_caller_id_number}/
action application=set data=hang_up_after_bridge=true/
action application=bridge data=sofia/default/$...@182.138.252.12/
/condition
/extension
When i tried this command both of them nothing happen 855 can call
182.138.252.12 i want it to restrict this account for not calling
182.138.252.12..
Please help..
Thanks,
Edmar
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Re: [Freeswitch-users] Setting Restrictions on Default Dialplan
On Sun, Dec 20, 2009 at 6:39 PM, Edmar Cruz darklio...@yahoo.com wrote:
Hi Sir,
How can I allow international calling in the dialing plan but for
select accounts only?
For example i want to restrict 855 to call this ip address
182.138.252.12 using the default configuration.. Does this command should
be
put in the default.xml or in the default folder and the filename is
00_restict.xml?
extension name=dialsample
condition field=destination_number expression=^(855)$\
action application=set
data=effective_caller_name=${effective_caller_id_name}/
action application=set
data=effective_caller_number=${effective_caller_id_number}/
action application=set data=hang_up_after_bridge=true/
action application=bridge data=sofia/default/$...@182.138.252.12/
/condition
/extension
When i tried this command both of them nothing happen 855 can call
182.138.252.12 i want it to restrict this account for not calling
182.138.252.12..
Please help..
This functionality already exists in the default dialplan and sample
directory entries, assuming that you are using authorization. First off,
look in 1000.xml (or any of the other sample user files) for this variable
declaration:
variable name=toll_allow value=domestic,international,local/
For any user whom you wish to restrict to local or domestic calling only
just remove the 'international' from the list:
variable name=toll_allow value=domestic,local/
Now when that user registers and makes calls he/she won't have
'international' in the ${toll_allow} channel variable. Something like this
in your dialplan could handle both cases:
extension name=international dialing
condition field=${toll_allow} expression=international
anti-action application=playfile
data=misc/you-are-not-authorized.wav/
anti-action application=hangup/
/condition
condition field=destination_number expression=^(\d+)$ !-- use
whatever value works for you --
action application=bridge data=sofia/internal/$...@whatever/
/condition
Now that I've typed all that, I should go back and ask: are you using digest
authorization? Or are you using an ACL to let your callers in? Anyway,
hopefully the above example will give you some ideas.
-MC
Thanks,
Edmar
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[Freeswitch-users] Setting Restrictions on Default Dialplan
Hi Sir,
How can I allow international calling in the dialing plan but for
select accounts only?
For example i want to restrict 855 to call this ip address
182.138.252.12 using the default configuration.. Does this command should be
put in the default.xml or in the default folder and the filename is
00_restict.xml?
extension name=dialsample
condition field=destination_number expression=^(855)$\
action application=set
data=effective_caller_name=${effective_caller_id_name}/
action application=set
data=effective_caller_number=${effective_caller_id_number}/
action application=set data=hang_up_after_bridge=true/
action application=bridge data=sofia/default/$...@182.138.252.12/
/condition
/extension
When i tried this command both of them nothing happen 855 can call
182.138.252.12 i want it to restrict this account for not calling
182.138.252.12..
Please help..
Thanks,
Edmar
--
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Re: [Freeswitch-users] Setting Restrictions on Default Dialplan
Not actually for now... I commented first the ACL restrictions...
Edmar Cruz wrote:
Hi Sir,
How can I allow international calling in the dialing plan but for
select accounts only?
For example i want to restrict 855 to call this ip address
182.138.252.12 using the default configuration.. Does this command should
be put in the default.xml or in the default folder and the filename is
00_restict.xml?
extension name=dialsample
condition field=destination_number expression=^(855)$\
action application=set
data=effective_caller_name=${effective_caller_id_name}/
action application=set
data=effective_caller_number=${effective_caller_id_number}/
action application=set data=hang_up_after_bridge=true/
action application=bridge data=sofia/default/$...@182.138.252.12/
/condition
/extension
When i tried this command both of them nothing happen 855 can call
182.138.252.12 i want it to restrict this account for not calling
182.138.252.12..
Please help..
Thanks,
Edmar
--
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Re: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database
Hi,
I got it working.
Can somebody explain me this error:
2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR:
[Missing local host]. Also I am confused about heartbeat rate. Is
enable_heartbeat_events=5 setting the heartbeat to 5?
Thank you in advance,
Amarakeerthi S wrote:
Dear Sir,
I have successfully installed freeSWITCH and it works fine in passthrough
mode. I installed nibblebill and it deduct money from the accounts
database
and it works fine. but I have two problems.
1. Calls can be initiated even though there is a minus value in accounts
database
2. Calls doesn't hangup when it goes to minus values.
Any answers are greatly appreciated.
This is my dialplan:
action application=nibblebill data=flush/
extension name=hangup
condition field=destination_number expression=^(hangup)$
action application=playback data=no_more_funds.wav/
action application=hangup/
/condition
/extension
extension name=Omega_Out
condition field=caller_id_number expression=^(\d{4})$/
condition field=destination_number expression=^(\d{11})$
action application=set data=nibble_rate=0.0448/
action application=set data=nibble_account=${accountcode}/
action application=set data=bypass_media=true/
action application=bridge
data={absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1/
/condition
/extension
This is the configuration file;
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- See http://wiki.freeswitch.org/index.php?title=Mod_nibblebill for
help with these options --
!-- Information for connecting to your database --
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
!-- The column name for the unique ID identifying the account --
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. --
!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. --
!-- If a call goes beyond a certain dollar amount, flag or terminate
it
--
/settings
/configuration
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Re: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database
what did you have to change, to get this working ?
Jay
On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S senaka...@gmail.com wrote:
Hi,
I got it working.
Can somebody explain me this error:
2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR:
[Missing local host]. Also I am confused about heartbeat rate. Is
enable_heartbeat_events=5 setting the heartbeat to 5?
Thank you in advance,
Amarakeerthi S wrote:
Dear Sir,
I have successfully installed freeSWITCH and it works fine in passthrough
mode. I installed nibblebill and it deduct money from the accounts
database
and it works fine. but I have two problems.
1. Calls can be initiated even though there is a minus value in accounts
database
2. Calls doesn't hangup when it goes to minus values.
Any answers are greatly appreciated.
This is my dialplan:
action application=nibblebill data=flush/
extension name=hangup
condition field=destination_number expression=^(hangup)$
action application=playback data=no_more_funds.wav/
action application=hangup/
/condition
/extension
extension name=Omega_Out
condition field=caller_id_number expression=^(\d{4})$/
condition field=destination_number expression=^(\d{11})$
action application=set data=nibble_rate=0.0448/
action application=set data=nibble_account=${accountcode}/
action application=set data=bypass_media=true/
action application=bridge
data={absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1/
/condition
/extension
This is the configuration file;
configuration name=nibblebill.conf description=Nibble Billing
settings
!-- See http://wiki.freeswitch.org/index.php?title=Mod_nibblebillfor
help with these options --
!-- Information for connecting to your database --
!-- The database table where your CASH column is located --
!-- The column name where we store the value of the account --
!-- The column name for the unique ID identifying the account --
!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) --
!-- By default, warn a caller when their balance is at $5.00. You
can
set this to a negative number. --
!-- By default, terminate a caller when their balance hits $0.00.
You
can set this to a negative number. --
!-- If a call goes beyond a certain dollar amount, flag or terminate
it
--
/settings
/configuration
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--
Sincerely
Jay
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Re: [Freeswitch-users] Setting Restrictions on Default Dialplan
Where should I write this line
extension name=international dialing
condition field=${toll_allow} expression=international
anti-action application=playfile
data=misc/you-are-not-authorized.wav/
anti-action application=hangup/
/condition
condition field=destination_number expression=^(\d+)$ !-- use
whatever value works for you --
action application=bridge data=sofia/internal/$...@whatever/
/condition
on the default.xml or in the default category?
Edmar Cruz wrote:
Hi Sir,
How can I allow international calling in the dialing plan but for
select accounts only?
For example i want to restrict 855 to call this ip address
182.138.252.12 using the default configuration.. Does this command should
be put in the default.xml or in the default folder and the filename is
00_restict.xml?
extension name=dialsample
condition field=destination_number expression=^(855)$\
action application=set
data=effective_caller_name=${effective_caller_id_name}/
action application=set
data=effective_caller_number=${effective_caller_id_number}/
action application=set data=hang_up_after_bridge=true/
action application=bridge data=sofia/default/$...@182.138.252.12/
/condition
/extension
When i tried this command both of them nothing happen 855 can call
182.138.252.12 i want it to restrict this account for not calling
182.138.252.12..
Please help..
Thanks,
Edmar
--
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Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)
Hi its good to hear any compare document between Vicidial and this project Ram On Fri, Dec 18, 2009 at 7:16 PM, Andrew Thompson and...@hijacked.us wrote: I've been asked to provide some screenshots, so here's some of the agent/supervisor interface: http://eagle.bsd.st/~andrew/cpxshots/ Hopefully the image names are self-explanatory. In the ringing picture, that URL pop is a configurable URL that can be used to integrate with a CRM, in my case our own CRM - spicecsm. The URL supports interpolation for variables like callerid, clientid, call type, etc. The supervisor view is a little hard to describe via static images, but you're able to drag and drop agents into another profile (empty profiles are hidden when not dragging an agent), drag agents onto an agent to send them the call, and there's also various right click menus available. Oh, and I forgot to mention this before; this system is in 'live testing' and the goal is to do a final deployment sometime in January. Andrew ___ 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 ___ 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
