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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones
by Jerry Richards 05 Oct '09

05 Oct '09

By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the "manage-presence" parameter to "true" in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry

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New to freeswitch and have a few questions
by Orien Love 05 Oct '09

05 Oct '09

Hello Everybody, I am new to freeswitch, so forgive me if I ask stupid questions. I am planning a test setup consisting of: 1 - Pfsense router with the freeswitch package installed. 1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones. 1 - LINKSYS SPA3000 to connect to my existing land line and phones. 2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs The first question I have, Are the IP601 phones supported? The wiki lists 320, 431, 501, 550, 650 but not the 601. Second, is there a place that helps a person new to the IP phone world learn what is needed to set up a PBX using freeswitch at a small office? Finally is my test setup a good one? is there something I am missing or that I need to get the learning process started, I have found in the past, with a little information and a test system, I can learn what I am doing by breaking and fixing the test bed. Thanks for your time Orien

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Gateways, Limits, & Routes
by Kevin Green 05 Oct '09

05 Oct '09

It seems many people are looking for ways to control gateways, resiliency of termination, and limit on connections easily in FS. Here are some of the thoughts I had, and I would like to hear what others think of this. In tradition phone hardware you would define lines, put them into a group, and then assign a route to go through that group. For resiliency you could group multiple routes together into a route list, if the first route failed, was all busy, or unavailable it would start to try the second route. If you go out a secondary route you can also play a warning tone to indicate this might be going out a more expensive connection. For example you may have an IP link between two boxes, then fail back to TDM if the IP links go down, the TDM would be more expensive so you would want to warn business users so they don't spend hours on the phone. Inbound and outbound calls could both go on the lines so when a call comes in or goes out a route it takes up one slot. My thought is, why don't we create something similar to this that will allow us to handle a lot of these cases without complex dialplans. We could create routes that are assigned to gateways and limit the number of incoming, outgoing, and total connections that can be on that route. We could also specify what cases we would consider as failure to move onto the next route if we are in a route list. Route lists would then be similar to using multiple limits, and failover, but in this case it would simply be a list. The following is an example of how I think this could be prgrammed. <routes> <route id="route1" gateway="provider1" max-inbound="5" max-outbound="5" max-total="5" countinue_on_fail="NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> <route id="route2" gateway="provider2" max-inbound="0" max-outbound="5" max-total="5" countinue_on_fail="NORMAL_TEMPORARY_FAILURE"/> <route id="route3" gateway="provider3" max-inbound="5" max-outbound="5" max-total="10" countinue_on_fail="NO_ROUTE_DESTINATION"/> </routes> <route_lists> <route_list id="list1"> <route id="route1"/> <route id="route2" warning_tone="expensive.wav"/> <route id="route3" warning_tone="really_expensive.wav"/> </route_list> </route_litsts> Then in your dialplan you would simply put: <extension name="outbound_rout"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="route/route2/$1"/> </condition> </extension> OR <extension name="outbound_route"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="route_list/list1/$1"/> </condition> </extension> The module would need to track not only outbound calls, but also inbound calls that come in through the specified gateways. This would help track cases where you have say only two channels that can be used for both incoming and outgoing calls. I'm not 100% sure if this is all feasable or if it would be that much of an improvement compaired to what is already there so I put it out to all of you for feedback. Regards, Kevin Green JohnnyVoIP 350 Legget Drive Kanata, ON, Canada K2K 2W7 Phone: 613 271 5993 Ext 1203 Fax: 613 271 9810

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configure FS: config.status: error: cannot find input file: Makefile.in
by Tihomir Culjaga 05 Oct '09

05 Oct '09

hello, i just got the last trunk and tried to compile it on one of my development machines... Well configure fails on tiff-3.8.2 where it is unable to find Makefile.in ... Can someone advice? checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes checking dynamic linker characteristics... GNU/Linux ld.so checking how to hardcode library paths into programs... immediate checking for OpenGL Utility library... no checking for GLUT library... no configure: creating ./config.status config.status: error: cannot find input file: Makefile.in tculjaga@subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l total 2224 -rw-r--r-- 1 tculjaga tculjaga 23741 2009-10-02 13:19 acinclude.m4 -rw-r--r-- 1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4 -rwxr-xr-x 1 tculjaga tculjaga 121 2009-10-02 13:19 autogen.sh -rw-r--r-- 1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 config -rw-r--r-- 1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log -rwxr-xr-x 1 tculjaga tculjaga 73065 2009-10-02 14:00 config.status -rwxr-xr-x 1 tculjaga tculjaga 740145 2009-10-02 13:28 configure -rw-r--r-- 1 tculjaga tculjaga 20492 2009-10-02 13:19 configure.ac -rwxr-xr-x 1 tculjaga tculjaga 56 2009-10-02 13:19 configure.gnu -rwxr-xr-x 1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno drwxr-xr-x 16 tculjaga tculjaga 4096 2009-10-02 13:19 contrib -rw-r--r-- 1 tculjaga tculjaga 1146 2009-10-02 13:19 COPYRIGHT -rw-r--r-- 1 tculjaga tculjaga 1570 2009-10-02 13:19 HOWTO-RELEASE drwxr-xr-x 5 tculjaga tculjaga 4096 2009-10-02 13:19 html drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 libtiff drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 m4 -rw-r--r-- 1 tculjaga tculjaga 1908 2009-10-02 13:19 Makefile.am -rw-r--r-- 1 tculjaga tculjaga 1724 2009-10-02 13:19 Makefile.vc drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 man -rw-r--r-- 1 tculjaga tculjaga 6270 2009-10-02 13:19 nmake.opt drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 port -rw-r--r-- 1 tculjaga tculjaga 2363 2009-10-02 13:19 README -rw-r--r-- 1 tculjaga tculjaga 9 2009-10-02 13:19 RELEASE-DATE -rw-r--r-- 1 tculjaga tculjaga 5893 2009-10-02 13:19 SConstruct drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 test -rw-r--r-- 1 tculjaga tculjaga 433 2009-10-02 13:19 TODO drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 tools -rw-r--r-- 1 tculjaga tculjaga 6 2009-10-02 13:19 VERSION tculjaga@subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ tculjaga@subZero:~/freeswitch-trunk/libs/tiff-3.8.2$

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Fail over
by srinivasula reddy 05 Oct '09

05 Oct '09

Hi, can any know how to implement fail over with freeswitch, please help me Regards -- Srinivasula Reddy K

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oz dump Saying error
by lakshmanan ganapathy 05 Oct '09

05 Oct '09

HI all,

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register problem
by Frank Carmickle 04 Oct '09

04 Oct '09

Can someone point out what is wrong here. Thanks. Siptrace at http://carmickle.com/fs.txt <include> <gateway name="voiptalk.org"> <param name="username" value="81100068"/> <param name="password" value="my_pass"/> <param name="from-user" value="81100068"/> <param name="from-domain" value="voiptalk.org"/> <param name="proxy" value="voiptalk.org"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="1000"/> <param name="contact-params" value="domain_name=$${domain}"/> <param name="context" value="public"/> </gateway> </include> --FC

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internal & external ip addresses of freeswitch
by Timur Irmatov 04 Oct '09

04 Oct '09

Hi. We have a local network 192.168.1.0/24, where all the users are. Out FreeSWITCH server is connected to this network, and has ip address 192.168.1.242. Through different network card it is connected to external gateway, and has address 172.16.12.11 in this network. I set up a test client with softphone. When incoming call is deliviered to this client, call is set up normally, but client can't hang it up. It sends BYE to external address - 172.16.12.11 - which is not reachable from the client. It seems this address is coming from Contact: field in INVITE that FreeSWITCH sends: U 192.168.1.242:5060 -> 192.168.1.34:37169 INVITE sip:100@192.168.1.34:37169 SIP/2.0. Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF. Max-Forwards: 70. From: "FreeSWITCH" <sip:0000000000@172.16.12.11>;tag=v817pS9c6v6Fe. To: <sip:100@192.168.1.34:37169>. Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5. CSeq: 121117089 INVITE. Contact: <sip:mod_sofia@172.16.12.11:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 267. Remote-Party-ID: "FreeSWITCH" <sip:0000000000@172.16.12.11>;party=calling;screen=yes;privacy=off. What should I tweak in freeswitch to change this behaviour? -- Timur Irmatov, xmpp:irmatov@jabber.ru

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Problem with subscription expire
by Helmut Kuper 04 Oct '09

04 Oct '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, it seems exired subsciptions are never cleared in FS. A look into sofia_presence.c confirms explains this /* negative in exptime means keep bumping up sub time to avoid a snafu where every device has it's own rules about subscriptions that somehow barely resemble the RFC not that I blame them because the RFC MAY be amibiguous and SHOULD be deleted. So to avoid the problem we keep resetting the expiration date of the subscription so it never expires. Eybeam completely ignores this option and most other subscription-state: directives from rfc3265 and still expires. Polycom is happy to keep upping the subscription expiry back to the original time on each new notify. The rest ... who knows...? */ For some reasons subscriptions created by Snom phones are filling up the sip_subscriptions table over time. This leads to some kind of DOS by FS against the subscribing phone ... The subscribtions are differentiate by call-id. This can be explained by RFC 3842 chapter 3.6 where expired subscriptions must be renewed with a NEW call-id. Because there is no hint about unsubscribing the old subscription I guess the clean up process has to be done by FS. Any way to get FS to do this job? Since there is no creation date or expire value which represents the expire as a timestamp I have no way to clean up the table manually via sql and cronjob - except cleaning the whole table ... A further (but background) question is, why do the subscriptions expire in snom phones at all ... regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKxG3q4tZeNddg3dwRArNEAJ9fjHLox1tt038ze0liUG0ki+wrfgCgsz09 pO+XUioXrBKJ/ozUOy1ZqeA= =nZaf -----END PGP SIGNATURE-----

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Outbound INVITE rejected with 480 Temp Unavail, Reason "MANDATORY_IE_MISSING"
by Jerry Richards 04 Oct '09

04 Oct '09

Hello All, I have an internal extension that needs to send an INVITE without SDP body (Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable with reason "MANDATORY_IE_MISSING". Would anyone know what I need to do to enable this? Best Regards, Jerry

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