- FreeSWITCH-users - FreeSWITCH Mailing Lists

switch loading
by shayne.alone＠gmail.com 14 Mar '13

14 Mar '13

Hi all I had planned to make a max load test on my FS switch: we send in coming calls to opensips and balance them between to FS, then terminate them on one other asterisk (with just MoH app). first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we found the "max-proceeding" limit that control the open sip dialogs... is there any other limit or control witch I need to look forward? or have affect on hi loads? ** I don't think that 1K sessions are really more.... -- Regards, Ali R. Taleghani

3 4

SSL/SRTP VS SSL/ZRTP
by mehroz 13 Mar '13

13 Mar '13

Hi all While configuring FS with secure setup which includes SIP secure over SSL/TLS and RTP with ZRTP, yet i am still unable to get it done. I came across a question i.e, how do we tell FS to do Secure SIP signalling (SSL/TLS) with securing RTP as ZRTP (not SRTP). I am very confused as what comes first! ? If I have enabled ZRTP settings in profile and vars.xml, and if i configured SSL in SIP signalling, how to tell it to not to secure RTP with SRTP, but ZRTP. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp758821… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 6

iNUM available now on Flowroute
by Brian Foster 13 Mar '13

13 Mar '13

Hey all, Heads up that Flowroute is now provisioning iNUM's for your enjoyment. To request them, email support(a)flowroute.com and include how many you want, your account email, your name, and tech prefix. Just got 2 :) -BDF Sent from my iPhone

2 1

FreeTDM originate call
by ashish gautam 13 Mar '13

13 Mar '13

Hi, I want to originated outgoing call to PSTN network throught freetdm-libpri-dahdi stack. What is the syntax for this to originate from command line -- REGARDS ============================================ *Ashish Gautam* (+918802865008)

2 1

SCA wrong from_tag on notify
by Doug Alarm 13 Mar '13

13 Mar '13

I am having an issue with Shared lines. My setup is this Freeswitch (Version 1.3.17+git~20130312T033518Z~7d29a92f55) -> Acme SBC -> Polycom 550 (PolycomSoundPointIP-SPIP_550-UA/4.0.3.7562) . If I point the Polycom phone directly to freeswitch everything works as it should. The issue appears when I route though the Acme SBC.. When I make a call from the Polycom 550, I get a 15sec pause and then the call completes. After much investigation, I have discovered that the issue appears to be that freeswitch sends a notify with a from_tag that is not part of an established dialog and my SBC responds with 481 Missing Dialog. Looking at the sip trace below the phone subscribes -> Freeswitch Accepts(with correct from tag) ->freeswitch then sends a Notify with wrong from_tag. ->SBC responds with 481. The odd part is that that after the phone times out and re-subscribes this time Freeswitch sends a Notify with the same form_tag as the Accept and the call completes and SCA works fine. Below is the siptrace between freeswitch and my SBC Anyone have any ideas what this might be? Thanks Doug ------------------------------------------------------------------------ SUBSCRIBE sip:1234567890@fs.domain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bKpsndha200gk1fu8df4d0.2 From: "1234567890" <sip:1234567890@sip.domain.com>;tag=7F95269B-FA2A50CA To: <sip:1234567890@sip.domain.com> CSeq: 1 SUBSCRIBE Call-ID: 1aa7577-9c569066-3952c105(a)192.168.200.105<mailto:1aa7577-9c569066-3952c105@192.168.200.105> Contact: <sip:1234567890@10.10.10.9:5060;ep=192.168.200.105;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: line-seize Call-Info: <sip:sip.domain.com>;appearance-index=1 User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.3.7562 Accept-Language: en Max-Forwards: 69 Expires: 30 Content-Length: 0 ------------------------------------------------------------------------ send 813 bytes to udp/[10.10.10.9]:5060 at 17:38:11.774056: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bKpsndha200gk1fu8df4d0.2 From: "1234567890" <sip:1234567890@sip.domain.com>;tag=7F95269B-FA2A50CA To: <sip:1234567890@sip.domain.com>;tag=DKrccCoqCvac Call-ID: 1aa7577-9c569066-3952c105(a)192.168.200.105<mailto:1aa7577-9c569066-3952c105@192.168.200.105> CSeq: 1 SUBSCRIBE Contact: <sip:1234567890@10.10.10.10:5060> Expires: 30 User-Agent: FreeSWITCH-mod_sofia/1.3.17+git~20130312T033518Z~7d29a92f55 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=30 Content-Length: 0 ------------------------------------------------------------------------ send 968 bytes to udp/[10.10.10.9]:5060 at 17:38:11.774279: ------------------------------------------------------------------------ NOTIFY sip:1234567890@10.10.10.9:5060;ep=192.168.200.105;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10;rport;branch=z9hG4bKvg7p3cN60g7aQ Max-Forwards: 70 From: <sip:1234567890@sip.domain.com>;tag=939D8tZtU5g7m To: "1234567890" <sip:1234567890@sip.domain.com>;tag=7F95269B-FA2A50CA Call-ID: 1aa7577-9c569066-3952c105(a)192.168.200.105<mailto:1aa7577-9c569066-3952c105@192.168.200.105> CSeq: 41262081 NOTIFY Contact: <sip:1234567890@10.10.10.10:5060> Expires: 30 Call-Info: <sip:sip.domain.com>;appearance-index=1 User-Agent: FreeSWITCH-mod_sofia/1.3.17+git~20130312T033518Z~7d29a92f55 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: line-seize Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=30 Content-Length: 0 ------------------------------------------------------------------------ send 1077 bytes to udp/[10.10.10.9]:5060 at 17:38:11.775249: ------------------------------------------------------------------------ NOTIFY sip:1234567890@10.10.10.9:5060;ep=192.168.200.105;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10;rport;branch=z9hG4bKXS0F5759XSXXj Max-Forwards: 70 From: <sip:1234567890@sip.domain.com>;tag=2OCFR236kQ0l To: "1234567890" <sip:1234567890@sip.domain.com>;tag=882868A3-8F845D52 Call-ID: dab1917f-2f9a96ee-a7f17a0d(a)192.168.200.105<mailto:dab1917f-2f9a96ee-a7f17a0d@192.168.200.105> CSeq: 505025480 NOTIFY Contact: <sip:1234567890@10.10.10.10:5060;transport=udp> Call-Info: <sip:sip.domain.com>;appearance-index=1;appearance-state=seized Call-Info: <sip:sip.domain.com>;appearance-index=*;appearance-state=idle User-Agent: FreeSWITCH-mod_sofia/1.3.17+git~20130312T033518Z~7d29a92f55 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: call-info Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=587 Content-Length: 0 ------------------------------------------------------------------------ recv 344 bytes from udp/[10.10.10.9]:5060 at 17:38:11.777251: ------------------------------------------------------------------------ SIP/2.0 481 Missing Dialog Via: SIP/2.0/UDP 10.10.10.10;received=10.10.10.10;branch=z9hG4bKvg7p3cN60g7aQ;rport=5060 From: <sip:1234567890@sip.domain.com>;tag=939D8tZtU5g7m To: "1234567890" <sip:1234567890@sip.domain.com>;tag=7F95269B-FA2A50CA Call-ID: 1aa7577-9c569066-3952c105(a)192.168.200.105<mailto:1aa7577-9c569066-3952c105@192.168.200.105> CSeq: 41262081 NOTIFY ------------------------------------------------------------------------ recv 505 bytes from udp/[10.10.10.9]:5060 at 17:38:11.802738: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.10;received=10.10.10.10;branch=z9hG4bKXS0F5759XSXXj;rport=5060 From: <sip:1234567890@sip.domain.com>;tag=2OCFR236kQ0l To: "1234567890" <sip:1234567890@sip.domain.com>;tag=882868A3-8F845D52 Call-ID: dab1917f-2f9a96ee-a7f17a0d(a)192.168.200.105<mailto:dab1917f-2f9a96ee-a7f17a0d@192.168.200.105> CSeq: 505025480 NOTIFY Contact: <sip:1234567890@10.10.10.9:5060;transport=udp> Event: call-info User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.3.7562 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------

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Re: [Freeswitch-users] Blocking incoming calls
by Clive Lansink 13 Mar '13

13 Mar '13

Thanks everyone. Perhaps I should have added that we have a separate internal SIP profile that our internal phone extensions register to but the question only related to the public profile through which we receive incoming calls from our VOIP provider. Sounds like the best way to silently and completely block other incoming traffic we don't want is to use a firewall rule so that's what I'll do. Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Alex Lake <alex(a)digitalmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Blocking incoming calls Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Wed, 13 Mar 2013 12:43:29 +0000 Ah! A different IP address. That's handy. Now I'll go away (OK, Avi? ;-P) > mod_sofia supports having different sip 'profiles'. Each one can be > bound to a different IP address or port, and they can have different > options, like different ACL settings, different codec settings, > different NAT settings, etc. > > On 13 March 2013 12:20, Alex Lake <alex(a)digitalmail.com > <mailto:alex@digitalmail.com>> wrote: > > When you say "to a different profile" - you're talking about on > the same box, but a different port? >> They could still be registering, but to a different profile. >> Possibly on an internal network. >> >> On 13 March 2013 11:43, Alex Lake <alex(a)digitalmail.com >> <mailto:alex@digitalmail.com>> wrote: >> >> Ah, so presumably the OP doesn't have (for example) SIP >> handsets registered to his box (presumably that's done on >> port 5060, too) >>> Only if you don't know what IP addresses calls are going to >>> be coming from. In this case, we can probably ask the >>> provider what their IP addresses are and just explicitly >>> allow them. >>> >>> All fail2ban does is check the log files then set up >>> relevant firewall blacklist rules, so for the same job you >>> get slightly more CPU load too. >>> >>> On 13 March 2013 10:28, Alex Lake <alex(a)digitalmail.com >>> <mailto:alex@digitalmail.com>> wrote: >>> >>> Isn't fail2ban the usual solution here? >>> > Hello. I hope someone can quickly see what I want to >>> do and steer me in the right direction. >>> > >>> > I've looked at the documentation for acl.conf.xml and >>> the SIP profile config file external.xml. I want to >>> block incoming calls from all but a single external IP >>> address and I'm sorry I just can't figure out how to do >>> it or even if it can be done. >>> > >>> > We have a SIP trunk service with our VOIP provider. >>> That means we have a static IP address which they use >>> when they forward calls to us. They don't need to >>> register, we just accept their calls but of course they >>> have to be to our destination phone number. That all >>> works and we have been very happy with Freeswitch for I >>> don't know well over a year. >>> > >>> > Recently I became aware that someone is hammering our >>> system trying to make calls. Our provider will only use >>> port 5060 so that does mean our system is sitting on the >>> internet with port 5060 open. Our dial plan works >>> correctly and I can see in the log these calls are going >>> nowhere. But they can be every few seconds and I suspect >>> they might be using a lot of bandwidth just hammering >>> the system. >>> > >>> > We will never receive calls from any other address >>> than the one our VOIP provider will use to call us. So I >>> just want to block SIP traffic from all addresses except >>> theirs. I just want Freeswitch to stay silent when a >>> call comes in on any other address, so there is no >>> evidence that it is there to be attacked. >>> > >>> > I know I can do this with a firewall but I hope I can >>> do it in Freeswitch itself. I am confused about the >>> parameters auth-calls and auth-call and how to apply an >>> access list that would restrict all calls to just one IP >>> address. I did read somewhere in the docs that if you >>> want to block calls you need to use a firewall and maybe >>> that's the answer and so be it. Still I hope I can do it >>> with Freeswitch so I can just apply the right ACL and >>> sort the problem without creating new problems by >>> introducing a firewall. >>> > >>> > Hope you can help. >>> > >>> > >>> > Clive Lansink >>> > Email: Clive(a)Lansink.Co.NZ <mailto:Clive@Lansink.Co.NZ> >>> > Phone: +64 9 520-4242 <tel:%2B64%209%20520-4242> >>> > Mobile: +64 21 663-999 <tel:%2B64%2021%20663-999> >>> > Fax: +64 21 789-150 <tel:%2B64%2021%20789-150> >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting(a)freeswitch.org >>> <mailto:consulting@freeswitch.org> >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> > http://www.cudatel.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users(a)lists.freeswitch.org >>> <mailto: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 >>> > >>> > >>> > ----- >>> > No virus found in this message. >>> > Checked by AVG - www.avg.com <http://www.avg.com> >>> > Version: 2012.0.2240 / Virus Database: 2641/5668 - >>> Release Date: 03/12/13 >>> > >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> <mailto: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 >>> >>> >>> >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>> Managing Director >>> >>> >>> *T <mailto:info@cassidywebservices.co.uk> *03300 100 960 >>> <tel:03300%20100%20960> *F >>> <mailto:info@cassidywebservices.co.uk> *03300 100 961 >>> <tel:03300%20100%20961> >>> *E <mailto:info@cassidywebservices.co.uk> >>> *andrew(a)cassidywebservices.co.uk >>> <mailto:andrew@cassidywebservices.co.uk> >>> *W <mailto:info@cassidywebservices.co.uk> >>> *www.cassidywebservices.co.uk >>> <http://www.cassidywebservices.co.uk> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org <mailto: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 >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com <http://www.avg.com> >>> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release >>> Date: 03/12/13 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> <mailto: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 >> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T <mailto:info@cassidywebservices.co.uk> *03300 100 960 >> <tel:03300%20100%20960> *F <mailto:info@cassidywebservices.co.uk> >> *03300 100 961 <tel:03300%20100%20961> >> *E <mailto:info@cassidywebservices.co.uk> >> *andrew(a)cassidywebservices.co.uk >> <mailto:andrew@cassidywebservices.co.uk> >> *W <mailto:info@cassidywebservices.co.uk> >> *www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto: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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com <http://www.avg.com> >> Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: >> 03/12/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto: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 > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T <mailto:info@cassidywebservices.co.uk> *03300 100 960 *F > <mailto:info@cassidywebservices.co.uk> *03300 100 961 > *E <mailto:info@cassidywebservices.co.uk> > *andrew(a)cassidywebservices.co.uk <mailto:andrew@cassidywebservices.co.uk> > *W <mailto:info@cassidywebservices.co.uk> > *www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.2240 / Virus Database: 2641/5668 - Release Date: 03/12/13 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Change freeswitch.xml.fsxml path
by G 13 Mar '13

13 Mar '13

Hi guys. I'm planning to put FS on a Compact Flash system. I read on docs that ./freeswitch/log/freeswitch.xml.fsxml file is the result of parsing of various xml conf files and that it is the real "engine" file for FS when it is running. I also read to not modify this file while FS is running. Usually I put all logs files in tmpfs volatile ram filesystem to avoid disk space consuming and also limit CF write cycles. Well, the problem is that I also usually delete each night all files from tmpfs log directories because I need to avoid RAM saturation due to log files In this scenario, also freeswitch.xml.fsxml will be deleted each night. This, I understand, is not good ... During compilation phases I found parameters to change the path where FS puts .pid file or db files etc, but nothing about this file. The question is: can anybody suggest a way to change the location of this file? O can anybody ensure me that deleting this file while FS is running is not a big deal? Thanks a lot!!! giuliano

4 3

Blocking incoming calls
by Clive Lansink 13 Mar '13

13 Mar '13

Hello. I hope someone can quickly see what I want to do and steer me in the right direction. I've looked at the documentation for acl.conf.xml and the SIP profile config file external.xml. I want to block incoming calls from all but a single external IP address and I'm sorry I just can't figure out how to do it or even if it can be done. We have a SIP trunk service with our VOIP provider. That means we have a static IP address which they use when they forward calls to us. They don't need to register, we just accept their calls but of course they have to be to our destination phone number. That all works and we have been very happy with Freeswitch for I don't know well over a year. Recently I became aware that someone is hammering our system trying to make calls. Our provider will only use port 5060 so that does mean our system is sitting on the internet with port 5060 open. Our dial plan works correctly and I can see in the log these calls are going nowhere. But they can be every few seconds and I suspect they might be using a lot of bandwidth just hammering the system. We will never receive calls from any other address than the one our VOIP provider will use to call us. So I just want to block SIP traffic from all addresses except theirs. I just want Freeswitch to stay silent when a call comes in on any other address, so there is no evidence that it is there to be attacked. I know I can do this with a firewall but I hope I can do it in Freeswitch itself. I am confused about the parameters auth-calls and auth-call and how to apply an access list that would restrict all calls to just one IP address. I did read somewhere in the docs that if you want to block calls you need to use a firewall and maybe that's the answer and so be it. Still I hope I can do it with Freeswitch so I can just apply the right ACL and sort the problem without creating new problems by introducing a firewall. Hope you can help. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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FreeSWITCH Weekly Conference Call
by Michael Collins 13 Mar '13

13 Mar '13

Hello folks, Today's weekly conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_13 Today we have Areski coming to discuss Newfies dialer. We hope you can make it! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

FS hash limit
by Alex Kostenko 13 Mar '13

13 Mar '13

Hello all, Can somebody explain how can I share call counts between several FS servers using hash limit. Or give me some more info (more then on Wiki) how it's work. I need to control call count on diff servers. Thanks!

4 5