Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
I probably shouldn't be doing this for you, but... http://bugs.digium.com/view.php?id=14431 ;) On Wed, Apr 1, 2009 at 7:41 PM, Alfonso Pinto elhod...@gmail.com wrote: I've searched in google about it and only found a message about the same, Anthony asked for more information and nobody answer. I've tried with an IP phone (aastra 57i) and the same happens. Thank you 2009/4/2 Brian West br...@freeswitch.org: I'm pretty sure this is a bug in Asterisk something to do with dialog matching... I think if you search the archives you'll see about it. /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: Hi guys, I've using asterisk as PSTN gateway. When a call arrives from PSTN, I send the call to freeswitch and this route the call to a SIP gateway. When caller cancels the call (hangups before callee answers), I get this on asterisk CLI: chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call '271c0dad41cc80456b8de2133dc80...@1.1.1.1'. Giving up. I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 This is the sip call flow: u 2009/04/01 21:59:26.402934 2.2.2.2:5060 - 1.1.1.1:5060 INVITE sip:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Contact: sip:99...@2.2.2.2. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29347 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.403717 1.1.1.1:5060 - 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.414810 1.1.1.1:5060 - 2.2.2.2:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm=1.1.1.1, nonce=5df21692-1f08-11de-9d06-83e4a6e70df7, algorithm=MD5, qop=auth. Content-Length: 0. . U 2009/04/01 21:59:26.415395 2.2.2.2:5060 - 1.1.1.1:5060 ACK sip:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1;tag=ceKFmNU84B90c. Contact: sip:99...@2.2.2.2. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:26.415648 2.2.2.2:5060 - 1.1.1.1:5060 INVITE sip:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Contact: sip:99...@2.2.2.2. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username=asterisk02, realm=1.1.1.1, algorithm=MD5, uri=sip:66...@1.1.1.1, nonce=5df21692-1f08-11de-9d06-83e4a6e70df7, response=cb57576192b001f79bd03ebb8bb57d0a, qop=auth, cnonce=47efcad4, nc=0001. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29348 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.416181 1.1.1.1:5060 - 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 103 INVITE. User-Agent:
[Freeswitch-users] FS failover redundancy load balancing
Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover high load. Thanks Ash ___ 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] Skype interaction commands on skypiax
Hi all, background: mod_skypiax is Skype compatible endpoint that allows incoming and outbound calls to/from the Skype network and SkypeOut service. It's seen by FS like other endpoints, so you can originate from sofia, bridge to skypiax, and connect the call to a landline number via SkypeOut service, for eg. skypiax endpoint use a Skype client to interact with the Skype network (see the wiki page for more details http://wiki.freeswitch.org/wiki/Skypiax). The news are: now you can send commands to the skype client related to a skyiax interface, both from the FS command line and programmatically (socket/API/esl/whatever) http://wiki.freeswitch.org/wiki/Skypiax#API_and_CLI_Commands This allow you to use directly the entire power of the Skype API ( https://developer.skype.com/Docs/ApiDoc ), for eg to send chat messages, interact with the buddy list, etc etc. Typing console loglevel 9 at the FS command line allows you to see the Skype API answers from the Skype client instance. So, in short: you bring loglevel to 9 (so you can see the Skype API messages going back and forth), you use sk or skypiax to send Skype API commands to the Skype client instance. This way you can prototype extensions to the current mod_skypiax, that can then be implemented in C directly into the mod_skypiax source code. Please, let me know of extensions you would like to be integrated into the mod_skypiax code ;-). Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ 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 failover redundancy load balancing
Ashley van Gerven pisze: Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? I did not think yet about HA nor LB. I tested how FS handles high load. All my calls are placed in mod_conference. When cpu usage gets it's limits then new calls can be placed but sound quality is getting worst with every next call. When calls are hanged up then sound gets better. I did not test it to see what happens when more and more calls are created. FS has very low memory consumption and I think that CPU is the limit. I did not notice any monitoring of CPU usage by FS, but my installation is limited to only few modules, so maybe I'm missing something. Would love to hear some experiences of deploying FS with failover high My failover is currently made by shell script which every 10 seconds check for working FS and restarts it if it does not work. I use svn trunk so crash happens once a while, but they are successfully fixed by developers. Once there was a problem that conference module was stuck and did not respond to my commands. I made script with netcat which checks once a while for response and restarts if there is none. load. Thanks Ash ___ 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] FS failover redundancy load balancing
Hi Ashley, One easy solution is to use a SIP proxy (opensips/kamailio/...) in front of FS boxes to load balance the charge between boxes. FS already has mechanisms to limit number of calls per boxes ( in switch.conf.xml: max-sessions and sessions-per-second ), that you can couple to load_balancing modules of the sip proxies. Of course you'll have to test to know how many session one box can handle, has it depends a lot on your usage of FS. Don' hesitate to join us on IRC if you want to discuss it ;) Regards, Gled Ashley van Gerven a écrit : Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover high load. Thanks Ash ___ 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] Dialplan for OPTIONS packet
Hi all, Whenever freeswitch recieves INVITE SIP packet, It forwards the packet based on the dial plan. I want to use the same dial plan to forward incoming OPTIONS packet. Please let me know If I need to write my own code for that or is there any such option in our code base. Regards, Sridhar ___ 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] Use rates from lcr in nibblebill module
Hi, I want to use module lcr to find a best route and his rate , then make a call and bill on that rate with nibblebill module. lcr return variable lcr_auto_route that contains [lcr_rate=xxx] variable for new channel. To use nibblebill i need to set nibble_rate = lcr_rate. What is best method to do that? Is there a way to do that with standard tools, without use external scripts? Thanks, Yuriy ___ 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] about freeswitch conference
hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? thanks! andy 2009-04-02 ___ 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] Buzzing when people speak in conference
Thanks for taking the time to help me. Giovanni, I assume you turn comfort noise off by setting it to 0 which I've now done. How can I tell which codecs I'm using in conference and how would I change them. The sound is ok on everything else. Thanks again From: stormin.nor...@hotmail.co.uk To: freeswitch-users@lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. Upgrade to Internet Explorer 8 Optimised for MSN. Download Now _ View your Twitter and Flickr updates from one place – Learn more! http://clk.atdmt.com/UKM/go/137984870/direct/01/___ 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] about freeswitch conference
hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? andy 2009-04-02 ___ 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 failover redundancy load balancing
On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote: I did not think yet about HA nor LB. I tested how FS handles high load. All my calls are placed in mod_conference. When cpu usage gets it's limits then new calls can be placed but sound quality is getting worst with every next call. When calls are hanged up then sound gets better. I did not test it to see what happens when more and more calls are created. FS has very low memory consumption and I think that CPU is the limit. I did not notice any monitoring of CPU usage by FS, but my installation is limited to only few modules, so maybe I'm missing something. Load testing against the conference module is about the worst thing you can do... tossing 100+ people in the same conference isn't going to scale well for load testing because its not something you usually do in a real world scenario. Usually you'll have most of the participants muted. I highly recommend you try doing something like a bridge or a file playback from a ram disk. Would love to hear some experiences of deploying FS with failover high My failover is currently made by shell script which every 10 seconds check for working FS and restarts it if it does not work. I use svn trunk so crash happens once a while, but they are successfully fixed by developers. Once there was a problem that conference module was stuck and did not respond to my commands. I made script with netcat which checks once a while for response and restarts if there is none. load. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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] Use rates from lcr in nibblebill module
Update the to the latest. I've added more channel vars: eg: after doing: action application=lcr data=12148267722 default2/ (not a real number) I get the following: variable_lcr_query_digits: [12148267722] variable_lcr_query_profile: [0] variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677, 12148267, 1214826, 121482, 12148, 1214, 121, 12, 1] variable_lcr_route_1: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gateway/teliax/12148267722] variable_lcr_rate_1: [0.01000] variable_lcr_carrier_1: [teliax] variable_lcr_codec_1: [PCMU] variable_lcr_route_2: [[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/gateway/vitelity/12148267722] variable_lcr_rate_2: [0.01440] variable_lcr_carrier_2: [vitelity] variable_lcr_codec_2: [PCMU] variable_lcr_route_count: [2] variable_lcr_auto_route: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gateway/teliax/12148267722|[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/gateway/vitelity/12148267722] variable_import: [lcr_carrier,lcr_rate] which, I think is what you are asking for. If you know which route you are going to use (eg: 1) then you can get it's rate by using lcr_rate_1. Alternatively, you can use the lcr_auto_route and then once the b-leg connects, query the b-leg variable for lcr_carrier and lcr_rate to see which one was actually used. You really can't use lcr_auto_route and set a single rate since each leg can be rated differently (look at example above). Normally lcr is used for your own rates between you and your carrier. That is independant of the rate table used for your customers. You can use lcr to query both. First use lcr to query your own rates using a different profile. This would return a *single* route if you setup your route table right. Save the rate in a var to be used with nibble bill. Then use lcr with your external rates so you can route according to your own cost with your carrier(s). On Thu, Apr 2, 2009 at 5:22 AM, Yuriy Ivzhenko (WP) yivzhe...@mksat.netwrote: Hi, I want to use module lcr to find a best route and his rate , then make a call and bill on that rate with nibblebill module. lcr return variable lcr_auto_route that contains [lcr_rate=xxx] variable for new channel. To use nibblebill i need to set nibble_rate = lcr_rate. What is best method to do that? Is there a way to do that with standard tools, without use external scripts? Thanks, Yuriy ___ 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 -- -Rupa ___ 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] upper registration in FS?
hi, i wanted to know if there was any way to actually accept all registrations coming towards freeswitch, the normal function is to have all the suerid and passwords configured, but is there a way to accept all registrations coming towards a single ip or domain? Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] new module: mod_memcache
Wow, this is cool. Fantastic work!
I tried this immediately. This is also very useful to share data across
applications.
Here an example how to share data between Freeswitch and a ruby
memcache-client:
On Ruby/Rails I set the namespace e.g. to freeswitch for the same
memcached server in environment.rb
In Freeswitch I add the following line to the dialplan:
action application=set data=ignore=${memcache(set
freeswitch:test 'This is a test')}/ !-- Memcache test --
Take care to prefix your key (here test) with the Ruby namespace
freeswitch:
Now you can receive the data in Ruby in raw mode:
CACHE.get(test,0)
= 'This is a test
The 0 as second parameter is important for the raw mode, otherwise ruby
will try to marshall the result from memcached and fails.
I added this info to the wiki.
Best regards
Peter
Brian West schrieb:
At the very least you didn't say I can't wait to play with it! :P
On Apr 1, 2009, at 7:59 PM, Shelby Ramsey wrote:
Rupa,
This is a big contribution! Thanks! Can't wait to play with this.
SDR
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
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Re: [Freeswitch-users] upper registration in FS?
Hi, You can blindly accept registrations and / or authentication messages with these parameters in a sip profile: param name=accept-blind-reg value=true/ param name=accept-blind-auth value=true/ http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg regards, Leon On Apr 2, 2009, at 3:01 PM, xbipin wrote: hi, i wanted to know if there was any way to actually accept all registrations coming towards freeswitch, the normal function is to have all the suerid and passwords configured, but is there a way to accept all registrations coming towards a single ip or domain? Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] new module: mod_memcache
On Thu, Apr 2, 2009 at 8:08 AM, Peter P GMX prometheus...@gmx.net wrote: Wow, this is cool. Fantastic work! I tried this immediately. This is also very useful to share data across applications. [snip] I added this info to the wiki. Best regards Peter Thanks for the wiki update -- great to see examples of how to actually use it. :) -- -Rupa ___ 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] mod_shout delay in trunk
2009/4/2 Anthony Minessale anthony.miness...@gmail.com Its the buffering and startup of the shout stream taking up the time, HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference. Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. shrug/ I'll go the local stream route for now -- -Rupa ___ 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] upper registration in FS?
I use the access control list acl.conf.xml to configure that.
Put ip/mask into the domain part of this config file, then it accepts
calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24).
list name=domains default=deny
node type=allow domain=$${domain}/
node type=allow cidr=123.123.123.123/32/
node type=allow cidr=124.124.124.1/24/
/list
Best regards
Peter
Leon de Rooij schrieb:
Hi,
You can blindly accept registrations and / or authentication messages
with these parameters in a sip profile:
param name=accept-blind-reg value=true/
param name=accept-blind-auth value=true/
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg
regards,
Leon
On Apr 2, 2009, at 3:01 PM, xbipin wrote:
hi,
i wanted to know if there was any way to actually accept all
registrations
coming towards freeswitch, the normal function is to have all the
suerid and
passwords configured, but is there a way to accept all registrations
coming
towards a single ip or domain?
Regards,
Bipin
--
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Re: [Freeswitch-users] about freeswitch conference
http://wiki.freeswitch.org/wiki/Mod_conference On Apr 2, 2009, at 3:29 AM, bmsword wrote: I want to use another softswitch conference that has been deployed in freeswitch,How should I do? ___ 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] upper registration in FS?
hi,
will the below work if all the registration that is to be accepted come
from different public ip addresses, i mean, clients from all ip ranges
and addresses rather than a single ip
param name=accept-blind-reg value=true/
param name=accept-blind-auth value=true/
Regards,
Bipin
www.xbipin.com
+971-55-9270058
Original Message
Subject: Re: [Freeswitch-users] upper registration in FS?
From: Peter P GMX prometheus...@gmx.net
To: freeswitch-users@lists.freeswitch.org
Date: Thursday, April 02, 2009 6:05:51 PM
I use the access control list acl.conf.xml to configure that.
Put ip/mask into the domain part of this config file, then it accepts
calls from this ip (eg. ip/32.) or from this subnet (e.g. ip/24).
list name=domains default=deny
node type=allow domain=$${domain}/
node type=allow cidr=123.123.123.123/32/
node type=allow cidr=124.124.124.1/24/
/list
Best regards
Peter
Leon de Rooij schrieb:
Hi,
You can blindly accept registrations and / or authentication messages
with these parameters in a sip profile:
param name=accept-blind-reg value=true/
param name=accept-blind-auth value=true/
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#accept-blind-reg
regards,
Leon
On Apr 2, 2009, at 3:01 PM, xbipin wrote:
hi,
i wanted to know if there was any way to actually accept all
registrations
coming towards freeswitch, the normal function is to have all the
suerid and
passwords configured, but is there a way to accept all registrations
coming
towards a single ip or domain?
Regards,
Bipin
--
View this message in context:
http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22847331.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] upper registration in FS?
Bipin Patel wrote: hi, will the below work if all the registration that is to be accepted come from different public ip addresses, i mean, clients from all ip ranges and addresses rather than a single ip yeah, that's kinda why its called blind ... you don't have to know where its coming from, and it doesn't have to be valid... just blindly accepts it -Ray ___ 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] Database schema
Are there documents or wiki page [I've missed during my searches] that detail the records and their types that are stored in the various FS databases; e.g. sofia_reg_profile name.db, core.db ? Regards Richard Lamkin * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN * ___ 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] upper registration in FS?
Turn on Multireg too. /b On Apr 2, 2009, at 9:38 AM, Bipin Patel wrote: hi, will the below work if all the registration that is to be accepted come from different public ip addresses, i mean, clients from all ip ranges and addresses rather than a single ip param name=accept-blind-reg value=true/ param name=accept-blind-auth value=true/ Regards, Bipin www.xbipin.com +971-55-9270058 Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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] Database schema
no, but they all auto-create. You can create a db and set up odbc, start freeswitch, then dump your db schema. Also, please do not send confidential emails to the mailing list. Mike On Apr 2, 2009, at 11:12 AM, Richard Lamkin wrote: Are there documents or wiki page [I’ve missed during my searches] that detail the records and their types that are stored in the various FS databases; e.g. sofia_reg_profile name.db, core.db ? Regards Richard Lamkin ___ 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] upper registration in FS?
hi, this is the scheme im trying to implement. Client --- OpenSBC --- FS route provider gateway Client - Behind NAT and sending calls to OpenSBC on 1.2.3.4:1806 OpenSBC - listening on ip 1.2.3.4:1806 in upper registration mode so all registrations coming to it are sent forward to FS for authentication using ip as 1.2.3.4:5062 (ip and port shown to FS) FS - listening on ip 1.2.3.4:5063 and accepting all registrations from 1.2.3.4:5062 and sending calls to voipswitch the above is the setup, dont ask y is it so messed up bcoz if i dont set it up like above then i wont be able to use voip at all. Now my queation is: 1) how to make FS accept all registrations based on ip as well as port rather than just the ip only using apply-inbound-acl and if in that mode, can it still route calls based on userid and password, meaning, based on the dialplan, can it be designed in such a way that if the registering id and password r follows, then u route call to the following gateway 2) i will be configuring the clients with id and pass in FS now how can i route calls from those ids to different providers? 3) multireg turning on, how to? i have been trying a lot and read the wiki also miltiple times but still cant udnerstand the FS config structure and the mroe i read, the more i get confused as its just too advanced Regards, Bipin www.xbipin.com Original Message Subject: Re: [Freeswitch-users] upper registration in FS? From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Date: Thursday, April 02, 2009 7:21:44 PM Turn on Multireg too. /b On Apr 2, 2009, at 9:38 AM, Bipin Patel wrote: hi, will the below work if all the registration that is to be accepted come from different public ip addresses, i mean, clients from all ip ranges and addresses rather than a single ip param name="accept-blind-reg" value="true"/ param name="accept-blind-auth" value="true"/ Regards, Bipin www.xbipin.com +971-55-9270058 Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com __ NOD32 3983 (20090402) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ 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 __ NOD32 3983 (20090402) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ 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] loopback-b channels stay alive
Hello,
We originate loopback channels and they end up in calling sofia
and transfer the call to a fifo.
If we have a heavy call volume loopback-b channels don't hangup properly.
They stay in core.db.
Unfortunetly we can't reproduce it on test boxes but happens every day.
On this box we had to turn off debug logging, becase we had I/O problems.
The only thing I saw in log that switch_core_session_thread don't call
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, Session %
SWITCH_SIZE_T_FMT (%s) Ended\n,
session-id,
switch_channel_get_name(session-channel));
in these cases.
We have local patches (I don't think they are related) and we are
running FS on virtual machine and we had some problem with that before
so I'm not sure, but I guess it is maybe a lock or mutex problem.
I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know
what to do with it.
FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
export CFLAGS=-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
export MOD_CFLAGS=-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
./configure
gcc -I/DEVEL/freeswitch/src/include
-I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
-ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99
-pedantic -o .libs/freeswitch freeswitch-switch.o -lm
./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt
-lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath
-Wl,/opt/freeswitch//lib
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_read_lock'
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_locate'
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_rwunlock'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1
Could you please tell me how could I test mutexes, rwlocks?
Other option would be to omit loopback channels.
Anthony earlier suggested me to avoid it and call sofia directly
you could make the loopback channel execute the eval app and do the
originate to the sofia channel from the dialplan.
action application=eval data=${originate(sofia/foo/a...@b.com xyz)}/
or make the loopback chan exec a lua or js and fire an originate command and
exit
This way you don't have the loopback a and b leg as well as the sofia chan.
but it doesn't work, because originate api doesn't let us originate inside a
session.
So we still using it.
Thanks in advance,
Tamas
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Re: [Freeswitch-users] loopback-b channels stay alive
Thanks for doing some of the legwork on this. BTW, this thread is probably a
bit too technical for the users list - I recommend sending to the dev list.
:)
-MC
On Thu, Apr 2, 2009 at 9:46 AM, Tamas Cseke cstomi.levl...@gmail.comwrote:
Hello,
We originate loopback channels and they end up in calling sofia
and transfer the call to a fifo.
If we have a heavy call volume loopback-b channels don't hangup properly.
They stay in core.db.
Unfortunetly we can't reproduce it on test boxes but happens every day.
On this box we had to turn off debug logging, becase we had I/O problems.
The only thing I saw in log that switch_core_session_thread don't call
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, Session %
SWITCH_SIZE_T_FMT (%s) Ended\n,
session-id,
switch_channel_get_name(session-channel));
in these cases.
We have local patches (I don't think they are related) and we are
running FS on virtual machine and we had some problem with that before
so I'm not sure, but I guess it is maybe a lock or mutex problem.
I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know
what to do with it.
FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
export CFLAGS=-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
export MOD_CFLAGS=-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
./configure
gcc -I/DEVEL/freeswitch/src/include
-I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
-ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99
-pedantic -o .libs/freeswitch freeswitch-switch.o -lm
./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt
-lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath
-Wl,/opt/freeswitch//lib
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_read_lock'
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_locate'
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_rwunlock'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1
Could you please tell me how could I test mutexes, rwlocks?
Other option would be to omit loopback channels.
Anthony earlier suggested me to avoid it and call sofia directly
you could make the loopback channel execute the eval app and do the
originate to the sofia channel from the dialplan.
action application=eval data=${originate(sofia/foo/a...@b.com xyz)}/
or make the loopback chan exec a lua or js and fire an originate command
and
exit
This way you don't have the loopback a and b leg as well as the sofia
chan.
but it doesn't work, because originate api doesn't let us originate inside
a session.
So we still using it.
Thanks in advance,
Tamas
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Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
hehe I emailed it to him off list :) /b On Apr 2, 2009, at 1:09 AM, Kristian Kielhofner wrote: I probably shouldn't be doing this for you, but... http://bugs.digium.com/view.php?id=14431 ;) Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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] loopback-b channels stay alive
you can't pass it in with -D
you have to actually add
#define SWITCH_DEBUG_RWLOCKS
to the top of switch_core.h
On Thu, Apr 2, 2009 at 11:46 AM, Tamas Cseke cstomi.levl...@gmail.comwrote:
Hello,
We originate loopback channels and they end up in calling sofia
and transfer the call to a fifo.
If we have a heavy call volume loopback-b channels don't hangup properly.
They stay in core.db.
Unfortunetly we can't reproduce it on test boxes but happens every day.
On this box we had to turn off debug logging, becase we had I/O problems.
The only thing I saw in log that switch_core_session_thread don't call
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, Session %
SWITCH_SIZE_T_FMT (%s) Ended\n,
session-id,
switch_channel_get_name(session-channel));
in these cases.
We have local patches (I don't think they are related) and we are
running FS on virtual machine and we had some problem with that before
so I'm not sure, but I guess it is maybe a lock or mutex problem.
I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know
what to do with it.
FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
export CFLAGS=-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
export MOD_CFLAGS=-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS
./configure
gcc -I/DEVEL/freeswitch/src/include
-I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
-ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99
-pedantic -o .libs/freeswitch freeswitch-switch.o -lm
./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt
-lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath
-Wl,/opt/freeswitch//lib
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_read_lock'
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_locate'
./.libs/libfreeswitch.so: undefined reference to
`switch_core_session_rwunlock'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1
Could you please tell me how could I test mutexes, rwlocks?
Other option would be to omit loopback channels.
Anthony earlier suggested me to avoid it and call sofia directly
you could make the loopback channel execute the eval app and do the
originate to the sofia channel from the dialplan.
action application=eval data=${originate(sofia/foo/a...@b.com xyz)}/
or make the loopback chan exec a lua or js and fire an originate command
and
exit
This way you don't have the loopback a and b leg as well as the sofia
chan.
but it doesn't work, because originate api doesn't let us originate inside
a session.
So we still using it.
Thanks in advance,
Tamas
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AIM: anthm
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FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] FS failover redundancy load balancing
Brian West pisze: On Apr 2, 2009, at 4:13 AM, Szymon Olko wrote: I did not think yet about HA nor LB. I tested how FS handles high load. All my calls are placed in mod_conference. When cpu usage gets it's limits then new calls can be placed but sound quality is getting worst with every next call. When calls are hanged up then sound gets better. I did not test it to see what happens when more and more calls are created. FS has very low memory consumption and I think that CPU is the limit. I did not notice any monitoring of CPU usage by FS, but my installation is limited to only few modules, so maybe I'm missing something. Load testing against the conference module is about the worst thing you can do... tossing 100+ people in the same conference isn't going to scale well for load testing because its not something you usually do in a real world scenario. Usually you'll have most of the participants muted. I highly recommend you try doing something like a bridge or a file playback from a ram disk. I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. Would love to hear some experiences of deploying FS with failover high My failover is currently made by shell script which every 10 seconds check for working FS and restarts it if it does not work. I use svn trunk so crash happens once a while, but they are successfully fixed by developers. Once there was a problem that conference module was stuck and did not respond to my commands. I made script with netcat which checks once a while for response and restarts if there is none. load. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ 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] FS failover redundancy load balancing
what kind of hardware? /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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] Handle invite with wrong to:IP
Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonym...@62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id=anonymous attribute and you must configure your device to use the proper domain in it's authentication credentials. I could do that, but this is not clean and I do not have a password for that. How can I workaround this, so that Freeswitch accepts this call? Aliases do not seem to work. Here is a sample message after FS asks for authorization: xx.xx.xxx.xxx is the IP of our Freeswitch 62.65.128.62 is the IP of Netvoip CH I would expect To: sip:0715aaa...@xx.xx.xxx.xxx. instead of To: sip:0715aaa...@62.65.128.62. U 62.65.128.62:5060 - xx.xx.xxx.xxx:5080 INVITE sip:0715aaa...@xx.xx.xxx.xxx:5080 SIP/2.0. Via: SIP/2.0/UDP 62.65.128.62:5060. Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. Max-Forwards: 69. From: sip:0049...@62.65.128.62;tag=8c977d2613672832fd9d03e9. To: sip:0715aaa...@62.65.128.62. Call-ID: 8c977d261329cd80fd9d0...@62.65.128.61. CSeq: 2 INVITE. User-agent: Netstream VoIP Gateway. Contact: sip:0049...@62.65.128.62:5060. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 584. Proxy-Authorization: Digest username=anonymous, realm=62.65.128.62, nonce=a4151ee0-1fbb-11de-b056-494b9de21e06, nc=0001, uri=sip:0715aaa...@62.65.128.62:5060, cnonce=5f109eee, response=62faa6d38b3b12c3626753395a8b507c, algorithm=MD5, qop=auth. . v=0. o=- 225947743692042 1 IN IP4 62.65.128.62. s=-. c=IN IP4 62.65.128.62. t=0 0. m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=no. a=rtpmap:3 GSM/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:99 G726-16/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:97 G726-32/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:105 iLBC/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Best regards Peter ___ 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] FreeSWITCH 1.0.4pre3 is slow to stop
Hi all, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): -- 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external -- Best Regards Lars Sivertsen ___ 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] ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available
Hi, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec. to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): -- 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external -- Best Regards Lars Sivertsen Michael Collins wrote: The FreeSWITCH team would like to let everyone know that the latest version is available. More information can be found here: http://www.freeswitch.org/node/172 By all means download, upgrade, test, and report back! Your feedback helps us make FreeSWITCH even better! -MC ___ 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] FS failover redundancy load balancing
Brian West pisze: what kind of hardware? I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those results, it was over 100 calls that was handle good, I was just curios what will happen. Tomorrow I will make real testes. My production works on 2 core P4 and I have there only 35 agents CPU load is like 7% with 15% small peeks. All phones are sip or analog via sip gateways, PRI is currently still on asterisk which is connected via sip. /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ 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] Handle invite with wrong to:IP
acl uses the remote addr from the socket connection, not anything from the sip message. On Thu, Apr 2, 2009 at 3:07 PM, Peter P GMX prometheus...@gmx.net wrote: Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonym...@62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id=anonymous attribute and you must configure your device to use the proper domain in it's authentication credentials. I could do that, but this is not clean and I do not have a password for that. How can I workaround this, so that Freeswitch accepts this call? Aliases do not seem to work. Here is a sample message after FS asks for authorization: xx.xx.xxx.xxx is the IP of our Freeswitch 62.65.128.62 is the IP of Netvoip CH I would expect To: sip:0715aaa...@xx.xx.xxx.xxx. instead of To: sip:0715aaa...@62.65.128.62 sip%3a0715aaa...@62.65.128.62. U 62.65.128.62:5060 - xx.xx.xxx.xxx:5080 INVITE sip:0715aaa...@xx.xx.xxx.xxx:5080 SIP/2.0. Via: SIP/2.0/UDP 62.65.128.62:5060. Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. Max-Forwards: 69. From: sip:0049...@62.65.128.62sip%3a0049...@62.65.128.62 ;tag=8c977d2613672832fd9d03e9. To: sip:0715aaa...@62.65.128.62 sip%3a0715aaa...@62.65.128.62. Call-ID: 8c977d261329cd80fd9d0...@62.65.128.61. CSeq: 2 INVITE. User-agent: Netstream VoIP Gateway. Contact: sip:0049...@62.65.128.62:5060. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 584. Proxy-Authorization: Digest username=anonymous, realm=62.65.128.62, nonce=a4151ee0-1fbb-11de-b056-494b9de21e06, nc=0001, uri=sip:0715aaa...@62.65.128.62:5060, cnonce=5f109eee, response=62faa6d38b3b12c3626753395a8b507c, algorithm=MD5, qop=auth. . v=0. o=- 225947743692042 1 IN IP4 62.65.128.62. s=-. c=IN IP4 62.65.128.62. t=0 0. m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=no. a=rtpmap:3 GSM/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:99 G726-16/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:97 G726-32/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:105 iLBC/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Best regards Peter ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Handle invite with wrong to:IP
We use the true network ip the invite came from NOT the one in the sip headers. Not very trust worth to do that you think? ;) So if your ACL is correctly setup to 62.65.128.62 it would let them in please verify your ACL is correct... /b On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote: Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonym...@62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id=anonymous attribute and you must configure your device to use the proper domain in it's authentication credentials. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ 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] FreeSWITCH 1.0.4pre3 is slow to stop
wait for pre4 On Thu, Apr 2, 2009 at 3:04 PM, Ceino ceino...@gmail.com wrote: Hi all, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): -- 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external -- Best Regards Lars Sivertsen ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Handle invite with wrong to:IP
My ACL contains:
list name=domains default=deny
node type=allow domain=$${domain}/
node type=allow cidr=62.65.128.62/32/
/list
So this should be fine, right? However it doesn't work.
Best regards
Peter
Brian West schrieb:
We use the true network ip the invite came from NOT the one in the sip
headers. Not very trust worth to do that you think? ;)
So if your ACL is correctly setup to 62.65.128.62 it would let them in
please verify your ACL is correct...
/b
On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote:
Hello,
I am using a SIP account from Netvoip CH. I try to receive inbound call
from this SIP trunk. I discovered that, when they sent an invite, the
IP-Adress of the to: is their own IP address.
There fore ACL doesn't work and FS asks for authorization, which then
fails
I receive the following message on the CLI:
2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth()
Can't find user [anonym...@62.65.128.62 mailto:anonym...@62.65.128.62]
You must define a domain called '62.65.128.62' in your directory and add
a user with the id=anonymous attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/
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Re: [Freeswitch-users] Handle invite with wrong to:IP
look at the debug log and see what happens?
On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX prometheus...@gmx.net wrote:
My ACL contains:
list name=domains default=deny
node type=allow domain=$${domain}/
node type=allow cidr=62.65.128.62/32/
/list
So this should be fine, right? However it doesn't work.
Best regards
Peter
Brian West schrieb:
We use the true network ip the invite came from NOT the one in the sip
headers. Not very trust worth to do that you think? ;)
So if your ACL is correctly setup to 62.65.128.62 it would let them in
please verify your ACL is correct...
/b
On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote:
Hello,
I am using a SIP account from Netvoip CH. I try to receive inbound call
from this SIP trunk. I discovered that, when they sent an invite, the
IP-Adress of the to: is their own IP address.
There fore ACL doesn't work and FS asks for authorization, which then
fails
I receive the following message on the CLI:
2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth()
Can't find user [anonym...@62.65.128.62 mailto:anonym...@62.65.128.62
]
You must define a domain called '62.65.128.62' in your directory and add
a user with the id=anonymous attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] Handle invite with wrong to:IP
I restart FS and initiate an incoming call (trunk is registered at the
SIP provider).
This is what I see on the console:
.
.
.
2009-04-02 23:39:16 [DEBUG] mod_event_socket.c:2224
mod_event_socket_runtime() Socket up listening on 0.0.0.0:8021
2009-04-02 23:39:16 [NOTICE] switch_core.c:981
switch_load_network_lists() Adding xxx.xxx.xxx.xxx/32 (allow) to list strict
2009-04-02 23:39:16 [NOTICE] switch_core.c:981
switch_load_network_lists() Adding xx.xx.xxx.xx/32 (allow) to list domains
2009-04-02 23:39:16 [NOTICE] switch_core.c:981
switch_load_network_lists() Adding 62.65.128.62/32 (allow) to list domains
2009-04-02 23:39:48 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth()
Can't find user [anonym...@62.65.128.62]
You must define a domain called '62.65.128.62' in your directory and add
a user with the id=anonymous attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
Nothing else.
Here is the registration info:
Name Netvoip
Scheme Digest
Realm sip.netvoip.ch
Username 071xxx
Password yes
From
Contact
Exten 071xxx
To sip:071...@sip.netvoip.ch
Proxy sip:sip.netvoip.ch
Context public
Expires 60
Freq 60
Ping 0
PingFreq 0
State REGED
Status UP
CallsIN 0
CallsOUT 0
Best regards
Peter
Anthony Minessale schrieb:
look at the debug log and see what happens?
On Thu, Apr 2, 2009 at 3:34 PM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
My ACL contains:
list name=domains default=deny
node type=allow domain=$${domain}/
node type=allow cidr=62.65.128.62/32
http://62.65.128.62/32/
/list
So this should be fine, right? However it doesn't work.
Best regards
Peter
Brian West schrieb:
We use the true network ip the invite came from NOT the one in
the sip
headers. Not very trust worth to do that you think? ;)
So if your ACL is correctly setup to 62.65.128.62 it would let
them in
please verify your ACL is correct...
/b
On Apr 2, 2009, at 3:07 PM, Peter P GMX wrote:
Hello,
I am using a SIP account from Netvoip CH. I try to receive
inbound call
from this SIP trunk. I discovered that, when they sent an
invite, the
IP-Adress of the to: is their own IP address.
There fore ACL doesn't work and FS asks for authorization,
which then
fails
I receive the following message on the CLI:
2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661
sofia_reg_parse_auth()
Can't find user [anonym...@62.65.128.62
mailto:anonym...@62.65.128.62 mailto:anonym...@62.65.128.62
mailto:anonym...@62.65.128.62]
You must define a domain called '62.65.128.62' in your
directory and add
a user with the id=anonymous attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
mailto:br...@freeswitch.org mailto:br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com
http://www.cluecon.com/
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Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com
mailto:msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
mailto:sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
http://iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
mailto:googletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] FS failover redundancy load balancing
How do you load balance conference calls? Doesn't all the conference members have to be on the same freeswitch server? On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko so...@gcdf.pl wrote: Brian West pisze: what kind of hardware? I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those results, it was over 100 calls that was handle good, I was just curios what will happen. Tomorrow I will make real testes. My production works on 2 core P4 and I have there only 35 agents CPU load is like 7% with 15% small peeks. All phones are sip or analog via sip gateways, PRI is currently still on asterisk which is connected via sip. /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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
