September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

NAT issue with SBC
by ThanhTruong 13 Sep '13

13 Sep '13

Hi all, I have configure freeswitch as SBC server with one asterisk server. I am getting nat issue when make call from FS to FS. my case as: client --- FS ----- Asterisk server ----- FS -----client. my configure in public.xml like: <extension name="asteirsk"> <condition field="destination_number" expression="(\d+)$" require-nested="true"> <condition field="network_addr" expression="^192\.168\.1\.123$" break="never"> <action application="set" data="hangup_after_bridge=true"/> <action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${destination_number}@${domain_name}"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="export" data="sip_h_X-ROUTE=LOOKUP"/> <anti-action application="set" data="bypass_media=true"/> <anti-action application="bridge" data="sofia/internal/${destination_number}(a)192.168.1.123"/> </condition> </condition> </extension> Call is process ok, But I see that when I answer call in destination, asterisk server alway send 200OK to my FS, FS return ACk, but asterisk resend 200OK. if any one can help, thank all alot. I am getting this stage to long and cant find any solution here. Thanks, ThanhTruong

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X-FS-Support Header Removal
by Ira Tessler 12 Sep '13

12 Sep '13

One of my carriers is asking me to remove the following header from the Invite: X-FS-Support: update_display,send_info I already am setting ignore_display_updates=true. I am using FS 1.2.9. Is there a way to remove this header? Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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Certificate Name Mismatch for TLS
by Vincent Xia 12 Sep '13

12 Sep '13

im trying to encrypt the voice call over FS in terms of SIP and RTP stream, what i do is simply follow instructions in http://wiki.freeswitch.org/wiki/SIP_TLS, the FS is running on Windows, so i generated TLS certificates as told by http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32), the softphone is eyebeam 1.5.14 build 40820 and i get the error "Certificate Name Mismatch". The FS wiki says "If you get the error, "Certificate Name Mismatch", then the server's certificate name and the name of the domain you put in eyebeam don't match.", but i wonder what's "the server's certificate name"? "the name of the domain you put in eyebeam" is my FS ip address(192.168.0.100), how do i generate TLS certificates accordingly? i tried to have commonName = 192.168.0.100 but no luck, could someone help?

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Updating TLS version?
by Privus 007 12 Sep '13

12 Sep '13

Hi, Given the recent revelations regarding the state of the net's (in)security, and as a longtime user of TLS with SRTP in FS, I think it would be prudent to update the TLS version and ciphers that FS can negotiate. Currently FS only supports TLS 1.0 (correct me if I'm wrong), which has some worrying security issues<http://security.stackexchange.com/questions/32817/why-dont-major-browsers-c…>. Is it possible to upgrade to TLS 1.1 (or even better, 1.2)? I'd like to use AES in the 256 key length instead of the standard 128. Anyone know if this is on the agenda and/or where I can have a closer look at the cipher order presented by FS? Thanks

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uuid_transfer help
by Usama Zaidi 12 Sep '13

12 Sep '13

Hey thanks guys, I sorted out the issue, basically what was happening was that I was calling a lua script in the dialplan that was calling my perl ESL script and putting the user into conference, and the last line on that script was session:hangup() :) so I just changed that to return 0, and everything works great now, on the dtmf even it transfers the channel as expected without hanging up.

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ACK+SDP, answered after hold, missing media
by Miha 12 Sep '13

12 Sep '13

Hi, I have one question regarding media issue. A calls B, B answers and transfer to C. A is put on hold but after C answers A still hears music on hold and C does not here anything. A is registered in FS. After A is put on hold, C send new INVITE with sdp c=0.0.0.0 and after C aswers new SDP is send in ACK with new ip in c but FS still does not send media to it. I have also tried with <param name="enable-3pcc" value="true"/> as I found this on google for this issue but no luck. Here is a sip trace from FS. http://pastebin.freeswitch.org/21398 tnx miha

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Problem with G729a codec
by smrdoshi 12 Sep '13

12 Sep '13

Hi Guys, We are testing calls in FS with SPA942 and PAP2. When we are trying call with G729a we are getting below error in fs. We tried with other codecs and in that case calls are working fine. == 2013-09-12 00:09:59.004336 [ERR] mod_sofia.c:2114 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 25222 25222 IN IP4 192.168.5.4 s=- c=IN IP4 192.168.5.4 t=0 0 m=audio 16468 RTP/AVP 18 101 a=rtpmap:18 G729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] == I tried to make calls work by changing some of the option in sip profile like enabled late negotiation, disable_transcoding=true etc but no success. Can anybody please take a look and help me to figure out the issue? Thanks in Advance. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-G729a-codec-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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media problem first seconds
by Miha 12 Sep '13

12 Sep '13

Hi, I am experiancing wird behaviour. Sometime happens that media is missing for first few second and that everything is ok. Can I prevent this in any way? tnx! miha

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Re: [Freeswitch-users] originate not creating xml_cdr record
by Lloyd Aloysius 11 Sep '13

11 Sep '13

Originate will create a b-leg record. I guess in your xml_cdr.conf file b_leg set to false(That is by default). change to true and reload the configuration. Lloyd On Wed, Sep 11, 2013 at 3:29 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming(a)simplicitymedialtd.co.uk> wrote: > Oh damn, sorry I was thinking of mod_xml_curl! > > In that case, try this instead; > http://wiki.freeswitch.org/wiki/Mod_xml_cdr > <param name="err-log-dir" value="/tmp"/> > > Enable that param, and check to see if the request ends up in that > directory. > > You can also enable 'log-http-and-disk' which will ensure the request is > always written to disk. > > You can also enable access logs on your web server to check if the request > is received, or even just run "nc -l 80" on the web server and wait for the > connection to be received. > > The main point of the above is to pinpoint where the fault is happening > (if the entry is pushed to disk, then it means the problem is with your web > server.. if the entry is not pushed to disk, then there's a problem > somewhere in FS, whether it be in config or a bug). > > Sorry for the confusion > > Cal > > > > On Wed, Sep 11, 2013 at 7:55 PM, Matt Broad <matt(a)inveroak.com> wrote: > >> Hi Cal, >> >> I have tried the method that you described but I get the error "-ERR >> xml_cdr debug_on Command not found!" >> could this be a typo? I have tried googling but with no success. >> >> thanks >> Matt >> >> >> On 11/09/2013 19:26, Matt Broad wrote: >> >> Thanks for the quick reply Cal. >> >> I will give that a try and see what I can see. I will report back with >> my findings :) >> >> thanks >> Matt >> >> >> On 11/09/2013 15:55, Cal Leeming [Simplicity Media Ltd] wrote: >> >> Matt, >> >> As far as I know, the API originate method should also trigger xml_cdr >> to post back to your server. >> >> Can you please run "console loglevel debug" and "xml_cdr debug_on", >> attempt to use api originate, then see if anything shows up in console? >> (that command will enable xml_cdr debugging). >> >> If this still does not work, please capture all the logs and explain >> the test procedure you went through and send it all in a JIRA ticket. >> >> Hope this helps >> >> Cal >> >> >> On Wed, Sep 11, 2013 at 3:32 PM, Matt Broad <matt(a)inveroak.com> wrote: >> >>> Hi, >>> >>> I have setup xml_cdr and it is working great, both aleg and bleg calls >>> are logged. >>> I am using the api originate method, but this does not create an xml_cdr >>> record (though a record is logged in /log/cdr-csv/master/csv). Is this >>> correct? >>> >>> >>> >>> thanks >>> Matt >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.… >> >> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… >> >> >> >> >> _________________________________________________________________________ >> 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 >> >> > > _________________________________________________________________________ > 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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uuid_transfer help
by Usama Zaidi 11 Sep '13

11 Sep '13

Hey MC, thanks for responding, here's the debug level output, and it makes no sense either. http://pastebin.freeswitch.org/21397 -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, September 11, 2013 10:38 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 87, Issue 80 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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