June 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] session.cause and session.causecode
by Francesco Facco de Lagarda 17 Jun '18

17 Jun '18

This is what the log says: 2018-06-17 11:18:41.186422 [ALERT] stuff.js:74 *********************** DIALING 2018-06-17 11:18:41.186422 [NOTICE] switch_channel.c:1104 New Channel sofia/external/065551234 [411ac79d-e8e0-4f77-b51f-7da24a71ea32] 2018-06-17 11:18:45.606424 [NOTICE] sofia.c:8273 Hangup sofia/external/065551234 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] 2018-06-17 11:18:45.626430 [NOTICE] switch_core_session.c:1683 Session 915 (sofia/external/065551234) Ended 2018-06-17 11:18:45.626430 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/065551234 [CS_DESTROY] 2018-06-17 11:18:45.626430 [ALERT] stuff.js:74 *********************** WAITING FOR SESSION READY .. 2018-06-17 11:18:45.626430 [ALERT] stuff.js:74 *********************** DISCONNECT CAUSE :false:undefined So this is what I want: 2018-06-17 11:18:45.606424 [NOTICE] sofia.c:8273 Hangup sofia/external/065551234 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] So why do I get: False: undefined from console_log("notice", "************************* disconnect cause:" + sessOut.cause + ":" + sessOut.causeCode); From: Francesco Facco de Lagarda <francesco(a)delagarda.com> Sent: domenica 17 giugno 2018 11:12 To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Subject: session.cause and session.causecode 'Morning all I can't get session.cause and session.causecode to work. All I get is false for the code and blank for the cause! Im running FS 1.6.20 This is a simplified version of my JS script if (session.ready()) { session.answer(); var dialedNum = "3332094333"; console_log("notice", "*********************** DIALING"); var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060"); console_log("notice", "*********************** WAITING FOR SESSION READY .."); if (sessOut.ready()) { console_log("notice", "*********************** SESSION READY"); bridge(session, sessOut); } else { console_log("notice", "************************* disconnect cause:" + sessOut.cause + ":" + sessOut.causeCode); } sessOut.hangup(); session.hangup(); console_log("notice", "*********************** CALL ENDED"); }

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wiki and org sites
by David Villasmil 14 Jun '18

14 Jun '18

Hello Guys, I can't connect to wiki.freeswitch.org, is the web down? I just get redirected to freeswitch's (new?) commercial web page (freeswitch.com) thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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number occupied
by Francesco Facco de Lagarda 14 Jun '18

14 Jun '18

I'm scripting in javascript How can I tell if the numer I called was occupied / busy? session.answer(); var sessOut = new Session(trunk + dialedNum); if (session.ready()) { /// all ok } // WHAT IF dialed num was busy or occupied ??

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Channel Hold Info
by mayamatakeshi 14 Jun '18

14 Jun '18

Hello, I have a requirement to generate reports of for how long calls were left on hold. My original plan was to monitor CHANNEL_HOLD/CHANNEL_UNHOLD events and update a channel variable preserving hold history. Then, I happened to find: https://freeswitch.org/confluence/display/FREESWITCH/CDR#CDR-Howlongwasthec… and realized there are already channel variables for this kind of purpose like hold_events, hold_accum_seconds and even hold-record. However, testing we recent commit: FreeSWITCH Version 1.9.0+git~20180516T213111Z~cfd99a03d4~64bit (git cfd99a0 2018-05-16 21:31:11Z 64bit) I verified all this data from XML CDR is related to the side that put the call on hold: Leg1) Side that put the other on hold: <hold_stamp>2018-06-14%2008%3A23%3A18</hold_stamp> <hold_events>%7B%7B1528932198592029,1528932199869251%7D%7D</hold_events> <last_hold_epoch>1528932198</last_hold_epoch> <last_hold_uepoch>1528932198592027</last_hold_uepoch> <hold_accum_seconds>1</hold_accum_seconds> <hold_accum_usec>1277224</hold_accum_usec> <hold_accum_ms>1277</hold_accum_ms> <hold-record> <hold on="1528932198592029" off="1528932199869251" bridged-to="1adf353f-a4e0-4e16-8953-fbd8beccca99"></hold> </hold-record> <last_hold_time>1528932198592027</last_hold_time> <hold_accum_time>1277224</hold_accum_time> Leg2) Side put on hold: <last_hold_epoch>0</last_hold_epoch> <last_hold_uepoch>0</last_hold_uepoch> <hold_accum_seconds>0</hold_accum_seconds> <hold_accum_usec>0</hold_accum_usec> <hold_accum_ms>0</hold_accum_ms> <last_hold_time>0</last_hold_time> <hold_accum_time>0</hold_accum_time> I am curious about this. Wouldn't it be better to have this kind of hold info at the side that was put on hold? Was this missed or done on purpose? I mean, we can with extra processing of hold-record of all Leg1s to resolve hold history for Leg2 however having this info at the Leg2 itself would be much simpler (and so I am considering in using the original CHANNEL_HOLD/CHANNEL_UNHOLD approach).

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Port numbers
by Rick Jarvis 14 Jun '18

14 Jun '18

Ok kind of FreeSWITCH 101 here I know, but something I’ve never understood completely: A provider wants to send us calls as number@our-ip-address - fine, but it’s hitting us on port 5060 (whereas external is set up as 5080). So can’t use external, have to use internal profile, right? So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL I think? But how do I handle call routing for inbound calls from provider’s IPs, vs my registered users dialling out? If anyone can simplify this for me that would be great :) Thanks __ R

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When 1.6.21 ?
by Mario 13 Jun '18

13 Jun '18

Can anyone answer when 1.6.21 will be available? Just wondering since I haven’t been able to test FS production for macOS wiki since PCRE was updated (FS-11061). Also wanted to start testing on macOS Mojave to see how that goes. Thanks, Mario G

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Initiate SIP session with MRCPv2 server
by Rob Capo 13 Jun '18

13 Jun '18

Hello, I have an MRCPv2 client that I'd like to use in an ASR application which is connected to FreeSWITCH via eventsocket (application written in Go). I'd like to use this client for MRCP signaling because it has the fine-grained control that I need, and the API is easier to use than the unimrcp eventsocket API. However, I'd like to have FreeSWITCH take care of answering the incoming call from the user. I don't care quite as much about RTP and SIP signaling to/from the ASR, but would like some control over it if possible (e.g. being able to see the RTP data for recording segments of audio that are important to me). There are a few ways I can think of solving this, some hackier than others. Currently, what I'm doing is probably the hackiest: 1. answer the call with FS 2. Initiate MRCPv2 session with ASR using my client 3. get the call's local media port over ES 4. pcap that port, and proxy the media to the ASR manually Perhaps a better way would be: 1. answer the call with FS 2. initiate MRCPv2 session with ASR using my client 3. tell FS to proxy audio to the ASR's RTP port from SDP (maybe passing through a proxy UDP socket that I open) But I'm not sure if this is possible. Another way that might be more approachable: 1. answer the call with FS 2. bridge in the ASR using FS 3. get the SDP / MRCPv2 port from FS 4. initiate TCP connection to MRCPv2 port and use my client The disadvantage would be that I no longer have visibility / control over the SIP / RTP paths, but I could always use pcap as before if the need arises (the situations where I need this control are most likely going to be debugging / non-production purposes). Does anyone have better recommendations or details on how I could achieve the second and / or third methods? Thanks, Rob

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Access to release 1.8
by Kevin Olbrich 13 Jun '18

13 Jun '18

Hi! In Jan 2017 release 1.8 was described as "With the release date of 1.8 fast approaching" or "The result of all of this you ask? FreeSWITCH 1.8, and after that just think we'll be at FreeSWITCH 2.0!". As of now, there is only 1.6 available for public access. Freeswitch 1.6 is working fine but is 1.8 still in the works? Source: http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124534.… Kind regards, Kevin

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Wrong release cause text, bug?
by David Villasmil 09 Jun '18

09 Jun '18

Hello guys, When hanging up a call with a specific release, FS sends the specific code, i.e.: if i release a call like so: <action inline="true" application="set" data="release_code=483 EXCHANGE_ROUTING_ERROR"/> The BYE message is as follows: SIP/2.0 483 EXCHANGE_ROUTING_ERROR Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK0y1H2X8899veB Max-Forwards: 69 From: "david.villasmil" <sip:david.villasmil@1.2.3.4>;tag=t35r9gatZFcjj To: <sip:9999999@1.2.3.4:5080>;tag=pB7mFpUrc30gB Call-ID: e2da7a79-e537-1236-5691-02e1e654f872 CSeq: 123866386 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR" Content-Length: 0 X-FS-Display-Name: 9999999 X-FS-Display-Number: sip:9999999@1.2.3.4 Remote-Party-ID: "9999999" <sip:9999999@1.2.3.4 >;party=calling;privacy=off;screen=no Another example: <action inline="true" application="set" data="release_code=410 NUMBER_CHANGED"/> The BYE message is SOMETIMES as follows: SIP/2.0 410 NUMBER_CHANGED Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKZN8r02Q5c16UF Max-Forwards: 69 From: "david.villasmil" <sip:david.villasmil@1.2.3.4>;tag=Q8Se4ZQF8m9SF To: <sip:9999999@1.2.3.4:5080>;tag=mSm3B0SHjHmBm Call-ID: dfbb5497-e537-1236-5691-02e1e654f872 CSeq: 123866383 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=22;text="NUMBER_CHANGED" Content-Length: 0 X-FS-Display-Name: 9999999 X-FS-Display-Number: sip:9999999@1.2.3.4 Remote-Party-ID: "9999999" <sip:9999999@54.84.149.30 >;party=calling;privacy=off;screen=no And other times as follows: SIP/2.0 410 Gone Via: SIP/2.0/UDP 1.2.3.4:32200 ;branch=z9hG4bK-d8754z-86e4295fca2efa0a-1---d8754z-;rport=32200 Max-Forwards: 69 From: <sip:david.villasmil@1.2.3.4;transport=UDP>;tag=42169b65 To: <sip:9999999@1.2.3.4;transport=UDP>;tag=pZ0N246BBcK7K Call-ID: YmU5YjEwODk2ZmZhODJjNDgzNTNlM2FiZWVmMjI2MDg. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "9999999" <sip:9999999@1.2.3.4 >;party=calling;privacy=off;screen=no But I've noticed that when i do: <action inline="true" application="set" data="release_code=488 BEARERCAPABILITY_NOTIMPL"/> The BYE message is as follows: SIP/2.0 488 BEARERCAPABILITY_NOTIMPL Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK3SDv7eUK14Z6D Max-Forwards: 69 From: "david.villasmil" <sip:david.villasmil@1.2.3.4>;tag=F3am0c2594H7a To: <sip:9999999@1.2.3.4:5080>;tag=6XDmXaZj5BQ8S Call-ID: aeb6624e-e539-1236-5691-02e1e654f872 CSeq: 123866772 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 X-FS-Display-Name: 9999999 X-FS-Display-Number: sip:9999999@1.2.3.4 Remote-Party-ID: "9999999" <sip:9999999@54.84.149.30 >;party=calling;privacy=off;screen=no So I guess, my question is why sometimes the "text" is set properly, and sometimes not?? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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SIP over TLS configuration problem
by fabio 08 Jun '18

08 Jun '18

Hi all I'm a Freeswitch newbie and I'm trying to setup SIP over TLS in my FS version 1.5.15. As first step I have configured a SIP Gateway that successfully registers to a dedicated SIP Registrar/Proxy (opensips) using SIP over UDP. With this configuration I can successfully place outbound and inbound calls without any problem. Everything works as a charm. Further I have tried to switch to SIP over TLS and I followed the steps described in https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS. I have installed the agent.pem and cafile.pem generated by opensips (my SIP Registrar) and I configured FS to use them. After restart the sofia gateway profile can successfully register to the SIP Registrar by SIP over TLS. Further I can successfully place outbound call (from internal channel through the SIP gateway). It sounds great! Unfortunately FS fails to handle inbound calls (SIP INVITE from an external SIP UA registered to the same SIP Registrar to the SIP UA extension of the FS SIP gateway). I have tried to trace all the logs I can. Here below some traces from the FS console when an inbound INVITE is received: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new secondary tport 0x1398c0 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x1398c0): new connection from tls/10.3.10.110:38632/sips tport_tls.c:919 tls_connect() tls_connect(0x1398c0): events NEGOTIATING tport_tls.c:1008 tls_connect() tls_connect(0x1398c0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2090 tport_close() tport_close(0x1398c0): tls/10.3.10.110:38632/sips tport.c:2263 tport_set_secondary_timer() tport(0x1398c0): set timer at 0 ms because zap In order to simplify the test I have also tried to connect to the 5061 TLS port by a simple openssl command from a linux shell of the SIP Registrar box: openssl s_client -connect 10.11.4.103:5061 -tls1_2 CONNECTED(00000003) 3074304200:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure:s3_pkt.c:1256:SSL alert number 40 3074304200:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake failure:s3_pkt.c:596: --- no peer certificate available --- No client certificate CA names sent --- SSL handshake has read 7 bytes and written 0 bytes --- New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE SSL-Session: Protocol : TLSv1.2 Cipher : 0000 Session-ID: Session-ID-ctx: Master-Key: Key-Arg : None PSK identity: None PSK identity hint: None SRP username: None Start Time: 1528361426 Timeout : 7200 (sec) Verify return code: 0 (ok) --- In the FS console I read the same traces received in the previous test with the inbound call. tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new secondary tport 0x248210 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x248210): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x248210): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x248210): new connection from tls/10.11.4.103:33168/sips tport_tls.c:919 tls_connect() tls_connect(0x248210): events NEGOTIATING tport_tls.c:1008 tls_connect() tls_connect(0x248210): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2090 tport_close() tport_close(0x248210): tls/10.11.4.103:33168/sips tport.c:2263 tport_set_secondary_timer() tport(0x248210): set timer at 0 ms because zap I have attached also a wireshark capture of the inbound call. In this capture the SIP Registrar has IP 10.3.10.110. The FS device is 10.11.4.103. The Client Hello is sent by the SIP Registrar, but the FS device replies with an "Alert: Level: fatal, Description: handshake failure (40). I guess that there is some misconfiguration related to the TLS version or proposed ciphers or any certifcates but I cannot understand what. For comparison I have tried to run the same openssl command from FS to the external SIP Registrar (outbound). openssl s_client -connect 10.3.10.110:5061 -tls1_2 CONNECTED(00000003) depth=1 CN = Your_NAME, ST = Your_STATE, C = CO, emailAddress = YOUR_EMAIL, O = YOUR_ORG_NAME verify error:num=19:self signed certificate in certificate chain verify return:0 --- Certificate chain 0 s:/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of Large Organization/CN=somename.somewhere.com/emailAddress=root@somename.somewhere… i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME 1 s:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME --- Server certificate -----BEGIN CERTIFICATE----- MIIC6TCCAdGgAwIBAgIBATANBgkqhkiG9w0BAQUFADBpMRIwEAYDVQQDFAlZb3Vy X05BTUUxEzARBgNVBAgUCllvdXJfU1RBVEUxCzAJBgNVBAYTAkNPMRkwFwYJKoZI hvcNAQkBFgpZT1VSX0VNQUlMMRYwFAYDVQQKFA1ZT1VSX09SR19OQU1FMB4XDTE4 MDUwODEyMzcyM1oXDTE5MDUwODEyMzcyM1owgb8xCzAJBgNVBAYTAlhZMRMwEQYD VQQIEwpTb21lIFN0YXRlMSMwIQYDVQQKExpNeSBMYXJnZSBPcmdhbml6YXRpb24g TmFtZTEpMCcGA1UECxMgTXkgU3VidW5pdCBvZiBMYXJnZSBPcmdhbml6YXRpb24x HzAdBgNVBAMTFnNvbWVuYW1lLnNvbWV3aGVyZS5jb20xKjAoBgkqhkiG9w0BCQEW G3Jvb3RAc29tZW5hbWUuc29tZXdoZXJlLmNvbTBcMA0GCSqGSIb3DQEBAQUAA0sA MEgCQQDL7uikSc1kVIvw5rhyQzk2dSJcmJ6EJ1LSmtAoafZH8bqfZ25cDQZQGi05 YcuxGR0vSaW7xPnyhaWCLQlxQFx7AgMBAAGjDTALMAkGA1UdEwQCMAAwDQYJKoZI hvcNAQEFBQADggEBAHv4WzGdYhoEyHZmBQTVjdEKOVBMnNoOqum79uzWtSzSjG4E pP/9c331uT7fBZ/Z7XNhIV+PbDZXorLgUhwwT7zxYURNnV52Of2SWRmWtPBrgEX1 +8S0IMtJFfJta8FAfTTaNqLpRDaiTQs3em1Maxls15cTyRQzMIjIJnY4eRrh5CNM YV/+kg/lpKAe0awiMu96cxpnMdz9h33g7RedBnh9wDi6k7pfYtvlC6o4snZO01AN 8qRiQf54OPvKcVeseJFBPWLhdYns6g+/SXhq1Lek2us93ZpuKgIaBtzkyDm2+SFa QXF9f0a+UuEdPvrtvMjAijcDwcaXq0r2f2MA++M= -----END CERTIFICATE----- subject=/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of Large Organization/CN=somename.somewhere.com/emailAddress=root@somename.somewhere… issuer=/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME --- No client certificate CA names sent --- SSL handshake has read 1979 bytes and written 337 bytes --- New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384 Server public key is 512 bit Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE SSL-Session: Protocol : TLSv1.2 Cipher : AES256-GCM-SHA384 Session-ID: EA8B17008E58F1D04CD1CEA53103CF477AA9DE0DC80A4FF4F0DD4814031E4C15 Session-ID-ctx: Master-Key: D28ED5C21D288944D2277AF86FE82A9BF3BEDABAA14DBCD5AE32B190EF0A0CA6AB99719E751E6DD4FECAA9DD1307A3C0 Key-Arg : None PSK identity: None PSK identity hint: None SRP username: None TLS session ticket lifetime hint: 300 (seconds) TLS session ticket: 0000 - 2f c5 82 ea bf 8b 66 49-bc bc ee 48 1a fb 8e 6c /.....fI...H...l 0010 - de 42 d9 e0 6e 36 40 78-06 cc 68 c6 74 6d 6e aa .B..n6@x..h.tmn. 0020 - b6 53 8a ed b2 8d 5a c4-02 e1 88 8b d2 a9 56 5f .S....Z.......V_ 0030 - ee c6 b9 14 55 da 37 df-8f aa af 81 b4 22 4e be ....U.7......"N. 0040 - 9c c5 87 d6 46 22 47 03-4a 88 dd 1e 9d 05 81 09 ....F"G.J....... 0050 - c3 8b 9f 44 29 90 4d 93-c9 f5 41 e2 4d 72 1b de ...D).M...A.Mr.. 0060 - 8d c2 15 ab 49 ad da 26-0e 72 a9 01 02 3e 89 33 ....I..&.r...>.3 0070 - 6e 6c 2f 20 1c 15 06 7a-8d c5 a6 6e ee 46 d2 76 nl/ ...z...n.F.v 0080 - 63 c1 89 1e 9b 3c a1 10-d0 78 31 9e e6 8e 86 ab c....<...x1..... 0090 - ff bc 3a 4c ab 3d 33 8f-e9 56 c5 f1 45 46 73 41 ..:L.=3..V..EFsA Start Time: 1528361487 Timeout : 7200 (sec) Verify return code: 19 (self signed certificate in certificate chain) --- closed In this case the command seems to have been successfully executed. I remark that the outbound TLS transactions seems to be working fine also from FS (SIP Registrar, SIP INVITE in outbound don't have any problem). If required I can provide also the FS configuration files (vars.xml, sofia.conf.xml, etc etc). Any help will be greatly appreciated. Thanks in advance Best regards fabio

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