February 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Enterprise-number (New Confirm) (Assigned)
by Brian West 07 Feb '07

07 Feb '07

Here ya go guys. /b > > IANA has assigned the following Private Enterprise > Number (PEN) to FreeSWITCH: > > 27880 > > Please see the following for your assignment in the registry: > > http://www.iana.org/assignments/enterprise-numbers

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Website Management... Call for help.
by Brian West 06 Feb '07

06 Feb '07

We are now looking for persons to assist in maintenance of the FreeSWITCH website. If anyone is interested in assisting the project in this manner please contact me off list. Thanks, Brian

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Bug Tracker and Source Viewer
by Brian West 04 Feb '07

04 Feb '07

A new website should be rolled out by monday but till then lets go over a few things. http://jira.freeswitch.org for bug tracking http://fisheye.freeswitch.org for source viewer Use these for bugs and source viewing. On another note... We are trying to work on a hardware abstraction layer for TDM hardware for both digital and analog lines. Can anyone wishing to be involved in this please join #freeswitch on IRC and we can outline an API/IOCTL list that we would like to see to interface with hardware from our software layer. Thanks, Brian West

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Threaded IVR/Voicemail?
by michael papet 03 Feb '07

03 Feb '07

Hi, I got things working with openser by having freeswitch register as an sip user with openser. Freeswitch is an sip client that handles voicemail. The problem with this is multiple users simultaneously using voicemail is not possible. The Freeswitch user is busy with a single caller. If I attempt to just forward the call to freeswitch's ip address:5060 (without freeswitch registering to openser) freeswitch does not respond at all. I assume I need to do something in sofia.conf.xml, but I'm not sure. Any advice, even if I'm missing the obvious, would be very helpful. Thanks. MP

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Re: [Freeswitch-users] Freeswitch Newbie Questions
by Anthony Minessale 03 Feb '07

03 Feb '07

You have a parse error in your file. Try installing the 100% clean default configs by removing all the files in /usr/local/freeswitch/conf and re-issuing make installall Also note there is a freeswitch bug tracker http://jira.freeswitch.org Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: michael papet <mpapet(a)gmail.com> To: freeswitch-users <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, February 2, 2007 3:04:41 PM Subject: [Freeswitch-users] Freeswitch Newbie Questions Okay, so I've figured out one or two things but have stumbled across a minor issue. I've got the following as my test plan:  <context name="default"> <extension name="playsound1" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="caller_id_name" expression="mpapet"> <action application="answer"/> <action application="playback" data="/var/sounds/congrats.wav"/> </condition> </extension> 2007-02-02 13:00:36 [NOTICE] switch_core.c:2802 switch_core_standard_on_execute() Execute playback(/var/sounds/congrats.wav) 2007-02-02 13:00:36 [DEBUG] mod_sndfile.c:138 sndfile_file_open() Opening File [/var/sounds/congrats.wav] 8000hz 2007-02-02 13:00:36 [DEBUG] switch_ivr.c:1125 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2007-02-02 13:00:36 [DEBUG] mod_sofia.c:2698 sip_i_state() Channel sofia/zed/mpapet(a)voip.katigo.mooo.com entering state [completed] It answers and starts playing the file according to the console but I hear no sound. Do I need some kind of other statement in there? Or, is it a NAT issue. Or? What does this error message mean? [DEBUG] mod_sofia.c:4585 event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a FYI: If I remove freeswitch_combined.xml, the file default_context.xml does not get read/included from freeswitch.xml. Something like "cannot find default context" kept appearing in the console despite the presences of default_context.xml. My quick and dirty solution was to put the default context inside freeswitch.xml. I verified this in the freeswitch console with the error message. After the change, call works great. As an FYI for future users, I'm running openser and freeswitch on the same machine using virtual ip addresses. You need to modify sofia.conf.xml <param name="sip-ip" value="your.freeswitch.virtual-ip.address"/> to get it to register freeswitch to your sip server. _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

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Freeswitch Newbie Questions
by michael papet 02 Feb '07

02 Feb '07

Okay, so I've figured out one or two things but have stumbled across a minor issue. I've got the following as my test plan:  <context name="default"> <extension name="playsound1" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="caller_id_name" expression="mpapet"> <action application="answer"/> <action application="playback" data="/var/sounds/congrats.wav"/> </condition> </extension> 2007-02-02 13:00:36 [NOTICE] switch_core.c:2802 switch_core_standard_on_execute() Execute playback(/var/sounds/congrats.wav) 2007-02-02 13:00:36 [DEBUG] mod_sndfile.c:138 sndfile_file_open() Opening File [/var/sounds/congrats.wav] 8000hz 2007-02-02 13:00:36 [DEBUG] switch_ivr.c:1125 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2007-02-02 13:00:36 [DEBUG] mod_sofia.c:2698 sip_i_state() Channel sofia/zed/mpapet(a)voip.katigo.mooo.com entering state [completed] It answers and starts playing the file according to the console but I hear no sound. Do I need some kind of other statement in there? Or, is it a NAT issue. Or? What does this error message mean? [DEBUG] mod_sofia.c:4585 event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a FYI: If I remove freeswitch_combined.xml, the file default_context.xml does not get read/included from freeswitch.xml. Something like "cannot find default context" kept appearing in the console despite the presences of default_context.xml. My quick and dirty solution was to put the default context inside freeswitch.xml. I verified this in the freeswitch console with the error message. After the change, call works great. As an FYI for future users, I'm running openser and freeswitch on the same machine using virtual ip addresses. You need to modify sofia.conf.xml<param name="sip-ip" value=" your.freeswitch.virtual-ip.address"/> to get it to register freeswitch to your sip server.

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pleas cancle the mail list for me ,thanks
by pengcz＠126.com 01 Feb '07

01 Feb '07

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