July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] [Telegraph-users] Telegraph for FreeSWITCH
by EdPimentl 18 Jul '08

18 Jul '08

I think the Telegraph is a very good and see a bright future for it. Though as already voiced in this thread, it can really benefit from improve documentation. My interest is certainly FS and Telegraph... I would be willing to post a bounty to get this started. -E http://tagr.mobi http://mobiquity.ws

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trunk fails on FreeBSD
by Adrian Gschwend 18 Jul '08

18 Jul '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi group, I can't compile latest trunk on FreeBSD 7.0-STABLE, bootstrap works, configure fails here: === configuring in libs/libsndfile (/usr/home/ktk/freeswitch.trunk/libs/libsndfile) configure: running /bin/sh ./configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' --cache-file=/dev/null --srcdir=. checking build system type... i386-unknown-freebsd7.0 checking host system type... i386-unknown-freebsd7.0 checking target system type... i386-unknown-freebsd7.0 ./configure.lineno: 1997: Syntax error: word unexpected (expecting ") I was already discussing that two nights ago on IRC. Does that work for anyone on FreeBSD with the latest trunk code? My BSD has the following autoconf installed: autoconf-2.61_2 autoconf-2.62 I can't take 1.0.0 as this fails in another module :) cu Adrian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfezQqpMUYrZbQBERAt9pAJ9CUKvAmqVaUc9j7wz+F7hlYfleCgCgq8eh lSiYDlJmjIUcK6McFILmLw4= =CEVv -----END PGP SIGNATURE-----

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how to start xml-rpc
by Peter P GMX 17 Jul '08

17 Jul '08

Hello, I have the standard Ubuntu installation for Hardy and freeswitch mostly works fine so far. Now I test the xmlrpc functionality in order to try the new Telegraph project which is based on freeswitch now ( http://code.google.com/p/telegraph <http://code.google.com/p/telegraph/wiki/GettingStarted>/). Firstly I tried the ruby code snipped on the freeswitch website. I tried with the external IP, with 127.0.0.1 and localhost, no chance. Finally, with nmap I found out that actually no process is listening at port 8080 althoung in /opt/freeswitch/conf/autoload_configs/xml_rpc.conf.xml this port is defined. What must I do in order to start xmlrpc for freeswitch? Best regards Peter

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Freeswitch and Twinkle and ZRTP
by Peter P GMX 17 Jul '08

17 Jul '08

Hello, did anybody get Twinkle with ZRTPworking? I tried this with 2 Twinkle clients with both zrtp enabled. Twinkle sends a=zrtp in the sdp message but this dows not arrive at the other Twinkle instance after the message passed Freeswitch. Any help is welcome.

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Dialplan: record_session - silence after bridge
by Birgit Arkesteijn 17 Jul '08

17 Jul '08

Hi all, I'm running FreeSwitch Version 1.0.trunk (498:8901) on Suse 10.0 (x86_64). I'm trying to record a session and closely followed the steps in: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session The recording file is generated OK, however it only record the audio UP TO the bridge action. After the bridging it records only silence. Here is the snippet of my dialplan: <extension name="westhawk" number="8002096143"> <application app_name="record_session" app_data="/usr/local/freeswitch/recordings/2008-07-17-14-12-20_8002096143_8187760092.wav"></application> <application app_name="bridge" app_data="sofia/gateway/westhawk/8187760092"></application> </extension> The odd thing is, when I do the same in javascript, it does record the whole session, including the bridging. Here is the snippet of the javascript: session.execute("record_session", recordfile) session.execute("bridge", bridge); Any idea what I need to change to get my dialplan working? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Performance
by Krešimir Tonković 17 Jul '08

17 Jul '08

Finally I made the time try Anthony's recommendation for performance enhancement: ------------------------- ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited comment out the manage-presence option in your sip profile to make the sip a lot less resource intensive ------------------------- (This is quoted from a previous mail on this list). With this, my laptop now takes 2.5 times the original CPS! Thanks Anthony! :-) I notice that my hard drive still trashes heavily while doing high-cps tests, and I suspect that is what holds freeswitch down. Any ideas how to reduce that further? -- Krešimir Tonković

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RTP Bind error
by Krešimir Tonković 17 Jul '08

17 Jul '08

Hi! I'm getting these errors on freeswitch: 2008-07-10 09:30:29 [ERR] sofia_glue.c:1779 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-07-10 09:30:29 [NOTICE] sofia_glue.c:1780 sofia_glue_activate_rtp() Hangup sofia/internal/xxxxxxx(a)192.168.1.10 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] Any idea what is going on? -- Krešimir Tonković

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Routing calls based on 'Referred-By'
by Matt Darnell 17 Jul '08

17 Jul '08

Aloha, I receive this snipped packet: INVITE sip:4479139@64.75.xxx.xxx:5060;transport=udp SIP/2.0 To: <sip:4479139@meta.xxx.com> Referred-By: <sip:4473452@meta.xxx.com> Diversion: <sip:4473452@meta.xxx.com>;reason=no-answer I would like the dial plan to route based on the number in the Refered-By or Diversion field. I have 10 DID's from a carrier and when the forward to me, they all have the same number in the invite, they say I should be able to route based on the Diversion field. Is there a way to get the data in the dial plan? Thanks for your help, Matt

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[Fwd: Openser + FreeSwitch Integration.]
by Aadilkhan Maniyar 17 Jul '08

17 Jul '08

Hi All, I would like to try and integrate OpenSER and FS. In this setup OpenSER would be the proxy and the Registrar. The registrations and subscriber info are stored in MySQL database. I would like to use FS as a B2BUA, so that it maintains all the dialog information, different call legs etc. (Something similar to Openser and Asterisk integration) Now my queries: 1. How can I proceed to achieve the above? 2. How can I share the MySQL database so that FS can access the registrations stored by OpenSER. 3. I understand that FS stores its data in SQLite database files. How can I make FS talk to MySQL database. Also I have tried integrating mod_spidermonkey_odbc (http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc) But still the registrations are being stored in the SQLite file and not in the MySQL database. Any help regarding this would be appreciated. Thanks in Advance. Regards, Aadil

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mod_socket_event two-way audio example code?
by tim＠marocon.com 17 Jul '08

17 Jul '08

On the FreeSWITCH wiki page entitled "How does FreeSWITCH compare to Asterisk?" (http://www.freeswitch.org/node/117) there is a pair of sentences that reads "The in-tree mod_event_socket provides a TCP connection on which events can be consumed as well as log data. In addition call control commands may be sent over this interface as well as bi-directional audio flow.". Does anyone have example code that demonstrates this functionality? I've looked through the code and I'm just too new to FreeSWITCH to be able to understand how to approach this. I'll take anything I can get (any language, OS, etc). Thanks, Tim Mattison

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