- sr-users - kamailio.org

Loadbalancer/dispatcher with dialog/cnxcc
by Oliver Roth 23 Apr '14

23 Apr '14

Hi all Following situation 1 dispatcher/loadbalancer getting all the inbound traffic and sending it to 3 different gateway (round robin). The loadbalancer has no (or even very few) business logic. Just "in" - split to different gateway - "out" 3 sip gateway doing all the business logic like auth, modifications on header, different checks, ... These 3 gateways use the same database (cluster) with routing tables and so on. A call gets terminated to a carrier through these sip gateway. Now I would like to implement call limiting (no of calls) by using either only dialog module or cnxcc module based on "source ip" or later on "cli". My problem: A call from one source ip can be sent to the (3) different sip gateway - so not all calls are processed by the same sip gateway. How can I ensure, that only a certain number of calls are allowed - even if they are split up on the 3 different gateway? Or do I need to implement this kind of business logic on the dispatcher/loadbalancer? That would not make much sense, because this is just a "stupid" machine... Thanks for helping Oli

1 0

Pick domain name of a user from location list of siremis
by aawaise 23 Apr '14

23 Apr '14

Hello all, I want to pick the domain name of a particular online user from location list of siremis and then perform some action on this basis. How can it be picked ?? Secondly how can the user name of called user be picked from the INVITE packet received by the server ?? Any help will be appreciated. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Pick-domain-name-of-a-user-from-loc… Sent from the Users mailing list archive at Nabble.com.

1 0

Drouting
by Keith 22 Apr '14

22 Apr '14

Hi, I am trying to use the drouting module and was wondering if there is a wildcard character for the dr_groups table for the user? My situation is that I want to look at any user from a certain IP range (i.e. the carrier). Any ideas? Thanks Keith

2 1

Issues with Kamailio Asterisk Integracion
by William Baylón Huerta 22 Apr '14

22 Apr '14

Hi, i see this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb, I've done step by step as described in the tutorial, but i have a problem, when A extension dial to B extension, B extension doesn't ringing... I send my sip.conf and kamailio.cfg Addittional, asterisk and kamailio are installed on same server, which have one private IP (192.168.50.217) and public IP 200.41.110.94 -- *Kevin W. Baylón Huerta* *Departamento Sistemas - Sitatel.* *Teléfono**: (51) 17073501 Anexo 208* *Celular: (51) 989715598*

2 1

SIREMIS - SQL-based CDR rating engine
by Dea Putera 22 Apr '14

22 Apr '14

Hi, I'm looking for any documentation on SIREMIS especially related to the rating engine. Where can I find it ? Or at least which module(s) are related to the rating engine. Any help is greatly appreciated. Thanks. -- -Dea

2 1

Caller ID number being passed to Gateway
by Joel E White 21 Apr '14

21 Apr '14

I have been playing with this for a couple days. What I have pulls from the usr_preferences DB and "Should" insert CID Name and CID Number What I am ending up with is that the name portion is being sent, not the number. Is there a certain format that I should add the DID in usr_preferences? Also when I changed the AVP type in the DB from a 0 to a 1 the value for CID DID changes... can someone help me with correcting this behavior? avp_db_load("$from","$avp(s:callerid-name)"); avp_db_load("$from","$avp(s:caller-did)"); uac_replace_from("$avp(s:callerid-name)","$avp(s:callerid-did)"); xlog("SIP From Header returned $hdr(From)\n"); xlog("AVP CallerID-Name returned $avp(s:callerid-name)\n"); xlog("AVP CallerID-Ext returned $avp(s:callerid-did)\n"); Thank you, Joel --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

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tls.so, no module matching <tls> found on FreeBSD 10
by Yoann Gini 21 Apr '14

21 Apr '14

Hello, It’s my first message here so here is a little introduction about me. I’m Yoann gini, a french system administrator specialized in Apple product for enterprise. I also manage some other UNIX sever (essentially BSD based) and some windows when customers pay me a lot… I’m trying to install Kamailio on by personal FreeBSD 10 server. My intend is discover products functionalities and see what can be done whit it. Maybe I can made a package for OS X Server and deploy it along with e-mail and xmpp services when my customers need efficient communication system. So, at this time, I’m playing with various options and I’ve some problems. The one who lead me here concern TLS module. I’m unable to start the service with TLS module enabled, digging into the debug log, I’ve catch those lines: 0(35943) DEBUG: <core> [cfg.y:1635]: yyparse(): loading module tls.so 0(35943) DEBUG: <core> [sr_module.c:591]: load_module(): load_module: trying to load </usr/local/lib64/kamailio/modules/tls.so> 0(35943) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... 0(35943) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <tls> found It seems that the module is loaded, but impossible to configure… Here is the output of grep -i tls kamailio.cfg ### #!define WITH_TLS # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS #!ifdef WITH_TLS listen=tls:<ipv4>:5061 listen=tls:<ipv6>:5061 enable_tls=yes #!ifdef WITH_TLS loadmodule "tls.so" #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") And the content of /usr/local/etc/kamailio/tls.cfg [server:default] method = TLSv1 verify_certificate = yes require_certificate = no private_key = /etc/ssl/keys/my_real_ssl_cert.key certificate = /etc/ssl/certs/my_real_ssl_cert.crt ca_list = /etc/ssl/cert.pem #crl = /usr/local/etc/kamailio/crl.pem [client:default] verify_certificate = no require_certificate = no After some search, I’ve found this message on the list http://lists.sip-router.org/pipermail/sr-users/2013-August/079439.html It seems that a bug on OpenSSL side cause some trouble with TLS module… Is it still true ? How do you manage your TLS configuration ? I’m using OpenSSL 1.0.1g with kamailio 4.1.2. Best regards, Yoann.

2 2

Pick Domain name from Location Table
by aawaise 21 Apr '14

21 Apr '14

Hi All, When my kamailio server receives an "invite" packet to make a call. I want the kamailio server to make a decision about call routing. Whether to route call to other kamailio server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. I have two tasks here. One is to pick the domain of user from "Location Table" of Siremis. And secondly, route this call to other server on basis of the string I get from "Domain" of "Location Table". Need help regarding changes in kamailio.cfg for this. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Pick-Domain-name-from-Location-Tabl… Sent from the Users mailing list archive at Nabble.com.

2 1

PCSCF configuration towards pcrf
by Dees 19 Apr '14

19 Apr '14

Hi, Very new to kamailio, following instruction have copied the examples file for pcscf and ran kamaillo. I have installed ims modules, but I don't see to see any connection initiation towards to PCRF? Any clues on what I am doing wrong. Thanks. Dees root@kamalio:/etc/bind# more /etc/hosts 127.0.0.1 localhost 10.209.96.85 pcscf.kamailio-ims.org kamalio #10.209.96.92 pcrf.kamailio-ims.org root@kamalio:/etc/bind# ping pcrf.kamailio-ims.org PING pcrf.kamailio-ims.org (10.209.96.92) 56(84) bytes of data. 64 bytes from 10.209.96.92: icmp_req=1 ttl=64 time=0.392 ms

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event_route[tm:branch-failure] question
by Juha Heinanen 18 Apr '14

18 Apr '14

in branch route i execute: t_on_branch_failure("contact"); xlog("L_INFO", "Routing $rm <$ru> to contact\n"); and in branch_failure route i have: event_route [tm:branch-failure:contact] { if (t_check_status("486")) { xlog("L_INFO", "Got 486 response to <$ru>\n"); }; and i get to syslog: Apr 13 21:44:07 siika /usr/sbin/sip-proxy[17475]: INFO: Routing INVITE <sip:test-0x11107b0@192.98.102.30:5064> to contact Apr 13 21:44:11 siika /usr/sbin/sip-proxy[17437]: INFO: Got 486 response to <sip:test@test.tutpro.com> why has $ru in branch_failure route changed from $ru of the branch to original request uri (aor)? i would expect to see in $ru of branch_failure route the $ru of the branch. is this a bug? if not, is there any means in branch_failure route to get access to the branch that failed. -- juha

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kamailio db
by Slava Bendersky 18 Apr '14

18 Apr '14

Hello Everyone, Is possible run kamailio without database ? I got ldap working and extension is registered, but can't place calls between two extensions. Getting 404 all the time. Slava.

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invite always contains "s"
by Vasco Mendes 18 Apr '14

18 Apr '14

Hi, I created an incoming trunk between iptel and my asterisk. Whatever the account is on iptel I always have in the from of received INVITE < sip :s@mypublicip :5060 >. < s > is not a recognized DID on my asterisk. Why do I receive this < s > in the from ?? Many thanks for your help, Vasco --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 1

Editings in kamailio.cfg for routing decision
by aawaise 18 Apr '14

18 Apr '14

Hi All, When a user sends an "invite" packet to make a call. I want to make a decision about call routing. Whether to route call to other server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. What changes are to be made in kamailio.cfg ?? Actually in my environment my user can connect to any of the servers. I want to know to which server it is connected and then make the call routing decision. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Editings-in-kamailio-cfg-for-routin… Sent from the Users mailing list archive at Nabble.com.

1 0

Custom fr timer for dialog's OPTIONS keep alives
by Vassilis Radis 17 Apr '14

17 Apr '14

Hello, I would like to try customizing the fr_timer for the OPTIONS keep-alive msgs transactions that the dialog module sends. I tried to call t_set_fr() function from inside the dialog modules's code (in dlg_req_within.c , in dlg_send_ka() function after the call to d_tmb. t_request_within(&uac_r); But it seems that this tm function is not available from inside other module's code. Is the above is the correct place to try and do it, and how can I programmatically ,inside a module's code, adjust fr_timer? On a side note, i followed the call flow from dlg_send_ka() down to the t_uac_prepare() function of tm module, where the fr_timer is set. In there, there is this comment+code: /* init timers hack, new_cell->fr_timer and new_cell->fr_inv_timer * must be set, or else the fr will happen immediately * we can't call init_new_t() because we don't have a sip msg * => we'll ignore t_set_fr() or avp timer value and will use directly the * module params fr_inv_timer and fr_timer -- andrei */ new_cell->fr_timeout=cfg_get(tm, tm_cfg, fr_timeout); new_cell->fr_inv_timeout=cfg_get(tm, tm_cfg, fr_inv_timeout); new_cell->end_of_life=get_ticks_raw()+lifetime; Does this mean that it cant be done in a straighforward way? Any advice on how I can achieve the customized fr_timers for those messages? Thanks.

2 2

Basic question regarding Routing
by aawaise 17 Apr '14

17 Apr '14

Hello, I want to ask some basic questions. Anyone please guide me to links with answers. I am sure such basics must be explained online. 1. What is exactly achieved by "route(RELAY)" ?? Routing calls from local users to other domains ?? 2. If user 1 is caller and user 2 is being called. Which pseudo variable will be used to catch value of called user i.e user 2 and his domain ?? I want to pick domain name from location table in database and not from the authorization module. Thanks, Cheers, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Basic-question-regarding-Routing-tp… Sent from the Users mailing list archive at Nabble.com.

2 1

Routing decision based on domain entry in Location Table
by aawaise 17 Apr '14

17 Apr '14

Hello, I have deployed two kamailio servers, both sharing the same database. I want my kamailio servers to make a decision on the routing of incoming call by seeing the domain of "called user" from location table of online users. If the "called user" is connected to same server, the call should be routed to user directly. In other case if the user is connected to other server, the call should be forwarded to other user for further processing. Will someone please guide me how can this be achieved ?? P.S : I am using kamailio-3.1.1. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Routing-decision-based-on-domain-en… Sent from the Users mailing list archive at Nabble.com.

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rls multipart body does not contain watcherinfo+xml
by Mihai Marin 16 Apr '14

16 Apr '14

Hello Sirs, I'm trying to accomplish a rls subscribe but I have some miss-understandings regarding the multipart publish response that should come. I'm using the integrated xcap from the kamailio with all the xml files added. My problem is that event if I request a multipart containing application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml I only get the first two. The request contains the following headers: Event: presence Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml The response only sends the first two: --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:bob@93.187.138.214.681064155> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri="sip:bob@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="4" fullState="false"><resource uri="sip:alice@93.187.138.214"><instance id="Scf8UhwQ" state="active" cid="1397508890.sip:alice@93.187.138.214.1552580371"/></resource></list> --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:alice@93.187.138.214.1552580371> Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8" standalone="no"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:alice@93.187.138.214"> <dm:person id="p6850"> <rpid:activities/> <rpid:status-icon>http://93.187.138.214:5060/xcap-root/oma_status-icon/users/sip:alice@93.187…</rpid:status-icon> </dm:person> <tuple id="t7673"> <status> <basic>open</basic> </status> <contact>sip:alice@93.187.138.214</contact> <note>Online</note> </tuple> </presence> --4kSPpRGyJQct4DIhyGnyobqO-- What do I missing? I need the body part with the watchers info in oder to see the state of the watcher (active, pending) - to accomplish the authorization for users. Thank you in advance for you help. Best regards, Mihai M

2 3

Register request forwarding from kamailio to asterisk
by betul.cerit＠netgsm.com.tr 16 Apr '14

16 Apr '14

Hi to all We integrated kamailio and asterisk as explained here http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb When kamailio receives a register request it authenticates and forwards the request to asterisk.Why do we need to do this?What happens if we dont pass registration requests to asterisk? Thanks in Advance Betül

2 2

Security
by Keith 16 Apr '14

16 Apr '14

Hi All, I am looking to apply some security on invites so I only accept from IP addresses in the address table or authenticated users. I have the address table bit working but I am struggling with the authenticated users bit. Wondering if anyone has done it? I was thinking of using the location table in some way as those users have authenticated? Any ideas would be appreciated. Keith

2 3

uac_auth problem
by neumann 16 Apr '14

16 Apr '14

Hi all sorry for my english! Iam trying to authenticate as uac on 3rd party server(Asterisk). Kamailio get 401, send next invite with DIGEST but Asterisk send 401 again((( Asterisk don’t write in log what auth is failed. 115.600045 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1093 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.601818 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.602470 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx 115.604310 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1267 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.605537 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.606431 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx My aster peer: [kamailio] type=peer defaultuser=kamailio secret=1234 host=dynamic disallow=all allow=alaw nat=no qualify=yes My kamailio config: modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") route[DISPATCH_OUT] { if(!ds_select_dst( 4, 8)){ send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); t_relay(); exit; } failure_route[RTF_DISPATCH_OUT] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(s:uac_user) = "kamailio"; $avp(s:uac_pass) = "1234"; $avp(s:uac_realm) = "asterisk"; if (isflagset(7)) { t_reply("513","Authentication failed"); xlog("L_INFO", "[%ci]: Remote authentication failed\n"); break; } if (uac_auth()) { xlog("L_INFO", "401/407 message recived"); # mark that auth was performed setflag(7); # trigger again the failure route t_on_failure("RTF_DISPATCH_OUT"); # repeat the request with auth response this time append_branch(); #t_relay_to_udp("94.25.100.198","5060"); t_relay(); break; } else { xlog("L_INFO", "uac_auth filed!!!!!!!!!!!"); } t_reply("514","Unknow authentication peer"); xlog("L_INFO", "[%ci]: Unknown authentication peer.\n"); break; } # next DST - only for 500 or local timeout if (t_check_status("500") or t_check_status("480") or (t_branch_timeout() and !t_branch_replied())){ ds_mark_dst("p"); if(ds_next_dst()){ t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); route(RELAY); exit; } } } ———————————— Timofeev Dmitry VoIP Engineer Linux, Asterisk, Freeswitch, Cisco solutions Skype: itsroot icq: 227227933

2 1

client provisioning.
by Slava Bendersky 15 Apr '14

15 Apr '14

Hello Everyone, Is possible do phone provisioning with kamailio ? Slava.

3 8

PERMISSIONS module issue
by Samuel Ware 15 Apr '14

15 Apr '14

I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show displays the new address; however kamcmd permissions.addressDump does not neither does kamcmd permissions.subnetDump. The messages from this new address return a false to the !allow_source_address("1”) command in my routing logic. I am wondering if this is a bug or I am doing something wrong. I am on the most recent GIT version to the best of my knowledge. Sam

6 6

SDP body rewritten issue at RTPproxy
by Wingsravi R 15 Apr '14

15 Apr '14

Dear All, I am working with Kamailio (4.1.2) and RTPproxy (1.2.1) servers and Jitsi as SIP clients. I have a problem in Audio/Video calling with Kamailio+RTPproxy server combination that Video is not streaming to other side through RTPproxy(also no audio both sides), I am suspecting something is rewritten in SDP body at RTPproxy, concerning to audio/video parameters (I am unable to rectify what is causing problem there) . Please find the Tcpdump based SIP capture of video call with this Kamailio+RTPproxy set-up. And with only kamailio server running (without RTPproxy instance running), audio/video call is fine on both sides. (find SIP capture for the same call). Also find my kamailio.cfg file for any reference. Please anybody help me in rectifying problems at RTPproxy server. Awaiting somebody's comments on this issue. Regards, Ravi

3 2

Re: [SR-Users] SDP c= IP doesn't get rewritten
by Ciprus, Daniel 14 Apr '14

14 Apr '14

Yes I do Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Daniel-Constantin Mierla Date:04/11/2014 6:13 AM (GMT-05:00) To: "Ciprus, Daniel" ,"Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] SDP c= IP doesn't get rewritten Hello, are you using a rtp relay (e.g., rtpproxy)? Cheers, Daniel On 11/04/14 02:46, Daniel Ciprus wrote: All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

4 5

LUA - UAC package implementation
by Daniel Ciprus 11 Apr '14

11 Apr '14

Hi, How much of UAC package is implemented in LUA embedded module ? According to wiki seems like only "sr.uac.replace_from" is implemented. My goal is to "emulate" 3rd party registration by using UAC and LUA meaning that I will have to recreate register, tweak existing attributes and send it over to application server to process. thanks Dan -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

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