July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] H264 transcodin
by Yufei Tao 28 Jul '12

28 Jul '12

Thanks everyone for the responses! If I understand it correctly, if I installed ffmpeg on itself separately from FS, I could write a module for FS, in which I just call the ffmpeg program by running a command line. This way would it be classified as "not combine them into a larger work", thus free from license incompatibility problem? Not sure if that'll work for real-time transcoding of x264 though? Thanks very much for you opinions! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 1

10 Days till ClueCon!
by Anthony Minessale 28 Jul '12

28 Jul '12

Countdown has started! Anyone who is going who has not registered, do it now so we can expedite the process for you and get the most accurate food totals. Anyone who is not going, Drat.... You still have 10 days to change your mind. Keeping in the tradition, BKW is always willing to sell your employer on the idea! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

1 0

What's better Unix ro Windows? LOL
by Sean Devoy 27 Jul '12

27 Jul '12

I really hope this does not blow up into an ideological jihad, but I am curious what people think and if they have any evidence to support their claims. I am in fact a long time windows developer (anyone remember windows 2.2 - that was hard shit). I have chosen what may be a unique approach to FS "Configuration, Command and Control". I started using FS by building it on Centos 5.n. Fighting, scratching and clawing it into a working multi-tenant switch. Looking back, everything I needed to do was in the email tree, I just didn't know what to ask. I have now moved FS on to a "production" VPS server for $30 a month with amazing success. See it here: http://www.synapseglobal.com/voip_services.php It comes prebuilt with Centos and FS latest build compiled and ready to go. That's not why I am writing though. Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at the base configuration. I learned about IPTABLES and FAIL2BAN and like them very much. However, I still work better/faster/surer in my Windows environment. So, I have taken what some might think is the worst possible approach: Configure and Control my Centos FS Server from my own ASP.NET Web Application (hosted elsewhere). My approach is to use the socket interface to send commands and use programmatic SFTP to the SHH shell for XML file exchange. I am about 85% done with version 1.0 and very pleased with it. I hope to have customers be able to login and modify their own configurations (call routes, IVRs, extensions mapping to devices line keys, Cisco spa504g provisioning, etc). Other device provisioning is in the pipe, but we have all 504Gs here and the provisioning code has been a tremendous help. Anyway, that is how I got to the odd work configuration, now I would like a discussion: My belief is that the "slim profile" of Centos will allow FS to handle greater load on a given hardware profile than could be handled by FS on Windows with the same hardware. I would like to AVOID the issues of security for this discussion, I firmly believe that you will provide better security on the platform that you understand the best. Let's just talk RAM, MIPS, NICs and FS performance and other issues I might be missing. Should a Windows guy go with FS on Windows or do you really get more bang for your buck in a Unix environment? I look forward to reading your thoughts. Sean

15 22

HTTP Request from within the core
by Gerald Weber 27 Jul '12

27 Jul '12

Hi, are there any ready-to-use core commands (to use in a c module ) to make http requests ? regards, gerald weber

5 7

Problems with multi-FS ODBC cluster
by Johannes Jakob 27 Jul '12

27 Jul '12

Hi, I'm operating a cluster of three FS boxes "behind" an OpenSIPs balancer/proxy that's not in the media path, but proxies everything SIP related in- and outbound. The central brain is a mysql master/slave setup and I'm using a combination of NFS and rsync to keep configs and voicemail synced. I want to balance registrations and calls to all three boxes, but because of the following issues, currently only one FS is active, the others on hot standby. The things that are still troubling me are: 1) Group intercept (*8) isn't working 2) Presence subscriptions aren't working 3) mod_callcenter is misbehaving badly - nonworking. I was thinking, that with everything moved to the database, the single boxes should be aware of callstates and uuids on the other boxes, like they know about which user is registered to which FS. Now I'm not sure if I've forgotten something, or if a multie-active setup would be a trade-off between redundancy and loss of features? All FreeSWITCHes are up-to-date, odbc-dsn has been set for all important modules: /usr/local/freeswitch/conf/autoload_configs$ grep dsn * | grep -v '!--' callcenter.conf.xml: <param name="odbc-dsn" value="freeswitch"/> db.conf.xml: <param name="odbc-dsn" value="freeswitch"/> switch.conf.xml: <param name="core-db-dsn" value="freeswitch" /> voicemail.conf.xml: <param name="odbc-dsn" value="freeswitch"/> and for both profiles: $ grep dsn /usr/local/freeswitch/conf/sip_profiles/* | grep -v '!--' /usr/local/freeswitch/conf/sip_profiles/external.xml: <param name="odbc-dsn" value="freeswitch"/> /usr/local/freeswitch/conf/sip_profiles/internal.xml: <param name="odbc-dsn" value="freeswitch"/> track-calls is also set for both of them, even though I guess this isn't relevant for my problems as long as I don't use recovery? $ grep track /usr/local/freeswitch/conf/sip_profiles/* | grep -v '!--' /usr/local/freeswitch/conf/sip_profiles/external.xml: <param name="track-calls" value="true"/> /usr/local/freeswitch/conf/sip_profiles/internal.xml: <param name="track-calls" value="true"/> As for group intercept, the log says (yes, I stripped most of it, you all know the default dialplan I guess ;) --- Dialplan: sofia/internal/1001(a)sip.example.org Regex (PASS) [group-intercept] destination_number(*8) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1001(a)sip.example.org Action answer() Dialplan: sofia/internal/1001(a)sip.example.org Action set(intercept_unanswered_only=true) Dialplan: sofia/internal/1001(a)sip.example.org Action intercept(${hash(select/${domain_name}-last_dial/${call_group})}) Dialplan: sofia/internal/1001(a)sip.example.org Action sleep(2000) EXECUTE sofia/internal/1001(a)sip.example.org set(call_direction=local) EXECUTE sofia/internal/1001(a)sip.example.org answer() 2012-07-27 22:31:35.693292 [NOTICE] mod_dptools.c:1148 Channel [sofia/internal/1001(a)sip.example.org] has been answered EXECUTE sofia/internal/1001(a)sip.example.org set(intercept_unanswered_only=true) EXECUTE sofia/internal/1001(a)sip.example.org intercept() 2012-07-27 22:31:35.693292 [ERR] mod_dptools.c:714 Usage: [-bleg] <uuid> --- in extension local_extension, the needed information gets stored in this hash: <action application="hash" data="insert/${domain_name}-last_dial/${called_party_call_group}/${uuid}" /> and this is working just fine, when enabling only one of the FS boxes. Did I miss something to share the content of hash's memory? What could I have missed, that subscriptions aren't working cross FS? (sip_subscriptions is filled pretty good, but BLF isn't working anyway) Any ideas if mod_callcenter should be working at all? I'm not quite sure how to interpret this note in the config:  does that mean, mod_callcenter doesn't support multi FS setups or does it mean, that XML config isn't supported for multi FS environments, but it should be working fine with odbc? All of it's config exists in the database (callcenter_config works and confirms the config on all of the FS boxes), but it doesn't work since the moment more than one FS box accessed the database... Well, I know this is quite a beast of an email, but I'd really appreciate any pointers in the right direction! Thanks!

1 0

Re: [Freeswitch-users] AutoNAT - Local Networks not excluded
by Phil Quesinberry 27 Jul '12

27 Jul '12

With that in mind - in my working configuration with phones both on the local LAN with FS as well as remote natted networks, I have: ./sip_profiles/internal.xml: <param name="apply-nat-acl" value="nat.auto"/> (All extensions are registered to the internal profile) In most cases, it was necessary to have FS rewrite the contact IP and port for remote extensions. - Phil I found someone talking about similar problems. I read that "... some lines of code in sofia_reg.c if (is_nat && profile->local_network && switch_check_network_list_ip(network_ip, profile->local_network)) { if (profile->debug) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "IP %s is on local network, not seting NAT mode.\n", network_ip); } is_nat = NULL; } " So I think there is the possibility to set which are local networks. Also I found in my sip_profiles for nat-mode contain (as it should be) the ext- IP declaration: <param name="ext-sip-ip" value="auto-nat"/> <param name="ext-rtp-ip" value="auto-nat"/> but not the line <param name="apply-nat-acl" value="rfc1918"/> May be this line solve my problem. I'll try as soon as possible, but all my FS servers ara actually in production environments :( If someone has the chance to test it successfully, please report it. g _____________________________________________ From: Phil Quesinberry Sent: Friday, July 27, 2012 10:31 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: re: AutoNAT - Local Networks not excluded One other thing comes to mind. A lot of routers (especially SOHO routers) have ALG functionality that can break the SIP signaling, even when the ALG functionality is supposedly turned off. You can usually get around this by changing the SIP port to something other than 5060. If the phones connect via TLS (usually on port 5061) then this shouldn't be a problem, as they can't mess with the encrypted traffic. - Phil _____________________________________________ From: Phil Quesinberry Sent: Friday, July 27, 2012 10:24 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: re: AutoNAT - Local Networks not excluded G, Are you registering your phones to the internal sip profile? Do you have anything like aggressive NAT detection enabled for that profile? For the extensions, are you rewriting the contact IP/port (is NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for sip-force-contact)? Do a 'show registrations' from the fs_cli as well as a 'sofia status profile internal reg' and post the results here (you may want to partially obscure any external IP addresses shown before posting) to give us more of an idea of what's going on. - Phil _____________________________________________ From: Phil Quesinberry Sent: Tuesday, July 24, 2012 3:19 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: re: AutoNAT - Local Networks not excluded Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users% 5D%20%20AutoNAT%20-%20Local%20Networks%20not%20excluded&In-Reply-To=> Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

3 2

ssl and sips and srtp configure issue.
by dolphincn 27 Jul '12

27 Jul '12

there have some problem when sip client and sips client call each other. 1, auto hangup in one minute. 2, sip client call sips client ,sips client can not pickup the call sometimes, and sips client with srtp mode can not pick up the call or sip client only hear hold music. 3, sips client using srtp could not use irv navigation and bind digit action, like press number 0 will not transfer to operator, no response. used softs: freeswitch x64 1.2rc for windows. 3cxphone client for windows ssl configure: var.xml internal.xml other used default setting. sip client configured sip server is local ip address (local_ip_v4 like 100.100.100.20). sips client configured sip server is external ip address (fs.share.com) ca files: conf/ssl/agent.pem conf/ssl/cafile.pem client installed ca file is cafile.pem. firewall open 5060-5061 for tcp/udp, 16384-32768 for udp, other closed. ssl work well, call 9664 can hear is secured call. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ssl-and-sips-and-srtp-configu… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 1

Re: [Freeswitch-users] AutoNAT - Local Networks not excluded
by Phil Quesinberry 27 Jul '12

27 Jul '12

One other thing comes to mind. A lot of routers (especially SOHO routers) have ALG functionality that can break the SIP signaling, even when the ALG functionality is supposedly turned off. You can usually get around this by changing the SIP port to something other than 5060. If the phones connect via TLS (usually on port 5061) then this shouldn't be a problem, as they can't mess with the encrypted traffic. - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq@qsystemsengineering.com] Sent: Friday, July 27, 2012 10:24 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: re: AutoNAT - Local Networks not excluded G, Are you registering your phones to the internal sip profile? Do you have anything like aggressive NAT detection enabled for that profile? For the extensions, are you rewriting the contact IP/port (is NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for sip-force-contact)? Do a 'show registrations' from the fs_cli as well as a 'sofia status profile internal reg' and post the results here (you may want to partially obscure any external IP addresses shown before posting) to give us more of an idea of what's going on. - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq@qsystemsengineering.com] Sent: Tuesday, July 24, 2012 3:19 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: re: AutoNAT - Local Networks not excluded Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users% 5D%20%20AutoNAT%20-%20Local%20Networks%20not%20excluded&In-Reply-To=> Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

1 0

Re: [Freeswitch-users] AutoNAT - Local Networks not excluded
by Phil Quesinberry 27 Jul '12

27 Jul '12

G, Are you registering your phones to the internal sip profile? Do you have anything like aggressive NAT detection enabled for that profile? For the extensions, are you rewriting the contact IP/port (is NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for sip-force-contact)? Do a 'show registrations' from the fs_cli as well as a 'sofia status profile internal reg' and post the results here (you may want to partially obscure any external IP addresses shown before posting) to give us more of an idea of what's going on. - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq@qsystemsengineering.com] Sent: Tuesday, July 24, 2012 3:19 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: re: AutoNAT - Local Networks not excluded Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users% 5D%20%20AutoNAT%20-%20Local%20Networks%20not%20excluded&In-Reply-To=> Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

1 0

AutoNAT - Local Networks not excluded
by VirteX 27 Jul '12

27 Jul '12

Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p3420… Sent from the Freeswitch-users mailing list archive at Nabble.com.

2 1