- FreeSWITCH-users - FreeSWITCH Mailing Lists

double invite
by budi wibowo 27 Mar '12

27 Mar '12

dear all i use FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) and connect to some gateway over sip. during test i found - 2 Invite from FS then receive 1 Trying from other gw - FS send cancel and other gw return 481 - FS send ACK , and got return BYE is it common happen? or any config in FS i can use to fine tune the setting? attached is detail pcap file for consideration TIA budi wibowo

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How to configure Bandwidth.com
by Farooq Hussain 27 Mar '12

27 Mar '12

Hello Everyone, Please let me know how would we configure bandwith.com SIP turnk. Please help us. -- Thanks Farooq Hussain

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Re: [Freeswitch-users] Strange "timer_test" result
by Peter Olsson 27 Mar '12

27 Mar '12

You could try to enable the "old" 1 ms-timer. It will be a little less efficient (a little more CPU load), and I'm not sure it helps, but it's worth a try. In switch.conf.xml (under autoload_configs), add this: <param name="1ms-timer" value="true"/> - within the settings-tags. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Stephen Wilde Skickat: den 26 mars 2012 16:20 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Strange "timer_test" result Thank you Peter for your reply! I have already tried the upgrade to latest git, now I'm on FreeSWITCH Version 1.1.beta1 (git-c31a799 2012-03-24 14-11-49 -0700) and the result is the same. I have tried with few load (during night) and in this case the timer_test is perfect. Any suggestion? On Mon, Mar 26, 2012 at 4:08 PM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Yes, most likely. It's very strange though, that a machine like this gives so poor timing results. The first thing I would do is to upgrade to latest GIT HEAD. A patch was commited about two weeks ago, that does the calculation for timer_test more properly (so time for logging is not calculated). Have you tried to do the same test with no load at all? /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Stephen Wilde Skickat: den 26 mars 2012 15:52 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Strange "timer_test" result Can this issue affect the voice quality? On Fri, Mar 23, 2012 at 7:46 PM, Stephen Wilde <wstephen80(a)gmail.com<mailto:wstephen80@gmail.com>> wrote: Real hardware, a dedicate server with 2 Xeon X5670 (a total 24 core each one with 12Mb cache at 2.93GHz) that is running at 20% - 25% of load. OS is CentOS 5.7 64bit and FS is (git-0626c89 2012-02-29 14-45-39 -0600) On Fri, Mar 23, 2012 at 6:33 PM, Brian Foster <bdfoster(a)endigotech.com<mailto:bdfoster@endigotech.com>> wrote: Is this on virtualized or real hardware? -BDF On Fri, Mar 23, 2012 at 1:31 PM, Stephen Wilde <wstephen80(a)gmail.com<mailto:wstephen80@gmail.com>> wrote: I have run a "timer_test" in a dedicated FS server and I see strange result: it's normal? Stephen freeswitch@internal> timer_test 20 40 Avg: 19.866ms Total Time: 795.880ms 2012-03-23 18:25:54.157822 [CONSOLE] mod_commands.c:549 Timer Test: samplecount after init: 1 2012-03-23 18:25:54.178818 [CONSOLE] mod_commands.c:554 Timer Test: samplecount after first step: 2 2012-03-23 18:25:54.199833 [CONSOLE] mod_commands.c:563 Timer Test: 1 sleep 20 19568 2012-03-23 18:25:54.231890 [CONSOLE] mod_commands.c:563 Timer Test: 2 sleep 20 38231 2012-03-23 18:25:54.252816 [CONSOLE] mod_commands.c:563 Timer Test: 3 sleep 20 18847 2012-03-23 18:25:54.262818 [CONSOLE] mod_commands.c:563 Timer Test: 4 sleep 20 13982 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 5 sleep 20 34793 2012-03-23 18:25:54.305813 [CONSOLE] mod_commands.c:563 Timer Test: 6 sleep 20 2 2012-03-23 18:25:54.326811 [CONSOLE] mod_commands.c:563 Timer Test: 7 sleep 20 23166 2012-03-23 18:25:54.347807 [CONSOLE] mod_commands.c:563 Timer Test: 8 sleep 20 16957 2012-03-23 18:25:54.357811 [CONSOLE] mod_commands.c:563 Timer Test: 9 sleep 20 17643 2012-03-23 18:25:54.378828 [CONSOLE] mod_commands.c:563 Timer Test: 10 sleep 20 18786 2012-03-23 18:25:54.399856 [CONSOLE] mod_commands.c:563 Timer Test: 11 sleep 20 25100 2012-03-23 18:25:54.420855 [CONSOLE] mod_commands.c:563 Timer Test: 12 sleep 20 18552 2012-03-23 18:25:54.441855 [CONSOLE] mod_commands.c:563 Timer Test: 13 sleep 20 18815 2012-03-23 18:25:54.462798 [CONSOLE] mod_commands.c:563 Timer Test: 14 sleep 20 19464 2012-03-23 18:25:54.484300 [CONSOLE] mod_commands.c:563 Timer Test: 15 sleep 20 22012 2012-03-23 18:25:54.494804 [CONSOLE] mod_commands.c:563 Timer Test: 16 sleep 20 13980 2012-03-23 18:25:54.515793 [CONSOLE] mod_commands.c:563 Timer Test: 17 sleep 20 19065 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 18 sleep 20 39585 2012-03-23 18:25:54.556792 [CONSOLE] mod_commands.c:563 Timer Test: 19 sleep 20 2 2012-03-23 18:25:54.577790 [CONSOLE] mod_commands.c:563 Timer Test: 20 sleep 20 26255 2012-03-23 18:25:54.598790 [CONSOLE] mod_commands.c:563 Timer Test: 21 sleep 20 17872 2012-03-23 18:25:54.630794 [CONSOLE] mod_commands.c:563 Timer Test: 22 sleep 20 32191 2012-03-23 18:25:54.651790 [CONSOLE] mod_commands.c:563 Timer Test: 23 sleep 20 22634 2012-03-23 18:25:54.672788 [CONSOLE] mod_commands.c:563 Timer Test: 24 sleep 20 15483 2012-03-23 18:25:54.693783 [CONSOLE] mod_commands.c:563 Timer Test: 25 sleep 20 22813 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 26 sleep 20 17099 2012-03-23 18:25:54.714783 [CONSOLE] mod_commands.c:563 Timer Test: 27 sleep 20 1 2012-03-23 18:25:54.734832 [CONSOLE] mod_commands.c:563 Timer Test: 28 sleep 20 29108 2012-03-23 18:25:54.755836 [CONSOLE] mod_commands.c:563 Timer Test: 29 sleep 20 11492 2012-03-23 18:25:54.776830 [CONSOLE] mod_commands.c:563 Timer Test: 30 sleep 20 20855 2012-03-23 18:25:54.808851 [CONSOLE] mod_commands.c:563 Timer Test: 31 sleep 20 32579 2012-03-23 18:25:54.818833 [CONSOLE] mod_commands.c:563 Timer Test: 32 sleep 20 18173 2012-03-23 18:25:54.850828 [CONSOLE] mod_commands.c:563 Timer Test: 33 sleep 20 22666 2012-03-23 18:25:54.871855 [CONSOLE] mod_commands.c:563 Timer Test: 34 sleep 20 23792 2012-03-23 18:25:54.892823 [CONSOLE] mod_commands.c:563 Timer Test: 35 sleep 20 26158 2012-03-23 18:25:54.913823 [CONSOLE] mod_commands.c:563 Timer Test: 36 sleep 20 13080 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 37 sleep 20 24609 2012-03-23 18:25:54.934844 [CONSOLE] mod_commands.c:563 Timer Test: 38 sleep 20 1 2012-03-23 18:25:54.954766 [CONSOLE] mod_commands.c:563 Timer Test: 39 sleep 20 19413 2012-03-23 18:25:54.975764 [CONSOLE] mod_commands.c:563 Timer Test: 40 sleep 20 19820 freeswitch@internal> status UP 0 years, 0 days, 16 hours, 10 minutes, 25 seconds, 539 milliseconds, 696 microseconds FreeSWITCH is ready 955611 session(s) since startup 2192 session(s) 0/50 6000 session(s) max min idle cpu 0.00/74.00 _________________________________________________________________________ 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 -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com<mailto:bdfoster@endigotech.com> Phone: 317-800-7876<tel:317-800-7876> Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ 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 _________________________________________________________________________ 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 !DSPAM:4f7079c632761636018988!

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P-Asserted-Identity
by Miha 27 Mar '12

27 Mar '12

Hi, I was looking in wireshark why some of the other providers are rejecting some of my calls. Some of our client are connected to Freeswitch via cabel-modem with voip support. I found that the problem appears when in sip header sip.P-Asserted-Identity is send and that FS send it to the SBC. The other softswith whitch is not FS and is working properly (this part), also get sip.P-Asserted-Identity in sip header but does not send it to sbc, it change it to P-Charge-info. Is it possbile to change that behaviour on FS so that other providers will not rejecting our calls? Thanks! Miha

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The Best Way to Send Fax on Sangoma?
by Anita Hall 27 Mar '12

27 Mar '12

Hi What should be the optimal settings for HWEC related parameters in wanpipe1.conf to be able to receive Fax over T.30 on a Sangoma A108DE Card ? In particular, should I enable echo ? If yes, to what parameter ? What other values like RXGAIN, TXGAIN should I tune ? TDMV_HW_DTMF = YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal regards, Anita

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Re: [Freeswitch-users] setting caller ID on a call to Asterisk
by Peter Olsson 27 Mar '12

27 Mar '12

Try to set caller-id-in-from to true on the gateway config. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Vik Killa Skickat: den 27 mars 2012 14:08 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] setting caller ID on a call to Asterisk When I send a call from FreeSWITCH to Asterisk, I cannot get Asterisk to properly display the caller ID. I have my FS set to register to Asterisk as a Gateway like sip:00019924@testdomain.com. Asterisk always seems to use the '00019924' in FROM: 00019924(a)testdomain.com as the callerID. I tried setting the 'sip-force-user' directory variable but that did not work. _________________________________________________________________________ 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 !DSPAM:4f71abfb32761977921347!

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Re: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10
by Peter Olsson 27 Mar '12

27 Mar '12

Please report to Jira, and make sure to attach the full build log - this doesn't show the actual error. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Joshua Nankin Skickat: den 27 mars 2012 03:52 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Trouble building Freeswitch on Ubuntu 11.10 I've followed all the steps at http://wiki.freeswitch.org/wiki/Quick_Start. I've seen similar issues in previous threads, and I've installed gawk and run make with -j1, but I'm still not able to do a clean build: quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Full log at http://joshnankin.com/freeswitch.log !DSPAM:4f7158e332761051920756!

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Trouble building Freeswitch on Ubuntu 11.10
by Joshua Nankin 27 Mar '12

27 Mar '12

I've followed all the steps at http://wiki.freeswitch.org/wiki/Quick_Start. I've seen similar issues in previous threads, and I've installed gawk and run make with -j1, but I'm still not able to do a clean build: quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Full log at http://joshnankin.com/freeswitch.log

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setting caller ID on a call to Asterisk
by Vik Killa 27 Mar '12

27 Mar '12

When I send a call from FreeSWITCH to Asterisk, I cannot get Asterisk to properly display the caller ID. I have my FS set to register to Asterisk as a Gateway like sip:00019924@testdomain.com. Asterisk always seems to use the '00019924' in FROM: 00019924(a)testdomain.com as the callerID. I tried setting the 'sip-force-user' directory variable but that did not work.

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Large delay/latency when bridging SIP calls
by Tor Petterson 27 Mar '12

27 Mar '12

Hi I'm having problems with bridging sip calls in Freeswitch after the revision from February 27 "only flush on break when its a blocking situation part 1". I am using Freeswitch in a phone center application where I first call an agent, then call a lead and then bridge the two calls. If I use a version after Feb 27. it takes about 5 seconds after bridging before any sound gets through. Here is an anonymized version of my sofia profile: <profile name="external">   <gateways> <gateway name="provider">  <param name="username" value="user"/>  <param name="realm" value="provider IP"/>    <param name="from-domain" value="FS host IP"/>  <param name="password" value="Password"/>  <param name="extension" value="test"/>        <param name="register" value="false"/>              </gateway> </gateways> <aliases>  </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/>   <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/>   <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/>    <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/>          <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/>  <param name="rtp-ip" value="FS host IP"/> <param name="sip-ip" value="FS host IP"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>  <param name="rtcp-audio-interval-msec" value="5000"/>  <param name="tls" value="$${external_ssl_enable}"/>  <param name="tls-only" value="false"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="$${external_tls_port}"/>  <param name="tls-cert-dir" value="$${external_ssl_dir}"/>  <param name="tls-passphrase" value=""/>  <param name="tls-verify-date" value="true"/>   <param name="tls-verify-policy" value="none"/>  <param name="tls-verify-depth" value="2"/>  <param name="tls-verify-in-subjects" value=""/>  <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile> -- Tor Petterson tpe(a)actimizer.com Tobaksvejen 25, 2. tv. - 2860 Søborg Telephone: +45 39 55 05 32 www.actimizer.com

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