February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Group confusions
by Lars Zeb 07 Feb '11

07 Feb '11

I went to two individual extensions and registered them for group 01 by dialing 8101. I then tested that group by dialing 8201 - both extensions rang. I then went to the cli and entered the command: group delete:01:1013@192.168.10.29 where 1013 is one of the two extensions I had joined into group 01 earlier and the url is the domain and fs address. However, when I dialed 8201 after entering this cli command, both extensions rang. I had expected that only one would ring. I am confused. Are these two different types of "groups"? Thanks, Lars

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Console Shutdown always crashes
by Daniel Wells 07 Feb '11

07 Feb '11

Any idea why the console always crashes on shutdown (compiled on Windows). These are the last few logs to the console: [CONSOLE] switch_loadable_module.c:1405 mod_siren has no shutdown routine [NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 32000hz 20ms [NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 16000hz 20ms [NOTICE] switch_loadable_module.c:518 Deleting Codec SPEEX 99 Speex 8000hz 20ms [CONSOLE] switch_loadable_module.c:1405 mod_speex has no shutdown routine I thought that it must have been because there was no shutdown routine for mod_speex but I noticed mod_siren doesn't either and it passes it ok. Sincerely, Daniel Wells IT Systems Engineer

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mod_dinglaing errors and google voice
by Daniel Wells 07 Feb '11

07 Feb '11

So I finally have what appears to be a working version of mod_dingaling on my windows box. I say working only because the console starts and I see evidence in the logs that freeswitch is communicating with my google account though mod_dingaling (I see presence information about some of my contacts). However I am not sure my dialplan is working (just a guess since this my first experience with freeswitch-or any PBX for that matter). I have been trying to follow the instructions as found here: http://wiki.freeswitch.org/wiki/Google_Voice I have a client connected using the default user 1000. I created a dingaling profile called gv1000.xml (I am using the second example of the dialplan on the above site and inferred that the client profile needed to renamed). My profile is identical to the one in the example with the exception of the username, password and "exten" (which is set to 1000). I put a copy of the dial plan in the "default" directory. But I am getting an error when trying to make outbound calls. Does this mean I have done something wrong, or is google blocking this now? freeswitch@dw-laptop> 2011-02-07 10:31:44.361514 [WARNING] sofia_reg.c:1247 SIP auth challenge (INVITE) on sofia profile 'internal' for [18XXXXXXXXX(a)192.168.2.97] from ip 192.168.2.97 2011-02-07 10:31:44.477521 [NOTICE] switch_channel.c:811 New Channel sofia/internal/1000(a)192.168.2.97 [cfda05d4-2f61-4ab0-8ca6-f759cfac53b7] 2011-02-07 10:31:44.689533 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->18XXXXXXXXX in context default 2011-02-07 10:31:44.689533 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/1000(a)192.168.2.97! 2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:697 Ring Ready sofia/internal/1000(a)192.168.2.97! 2011-02-07 10:31:44.689533 [NOTICE] mod_dingaling.c:721 Close Channel N/A [CS_NEW] 2011-02-07 10:31:44.689533 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER] 2011-02-07 10:31:44.689533 [INFO] mod_dptools.c:2621 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:2684 Hangup sofia/internal/1000(a)192.168.2.97 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/1000(a)192.168.2.97) Ended 2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000(a)192.168.2.97 [CS_DESTROY] Sincerely, Daniel Wells

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conferenceproblems after execute_application
by Thomas Hoellriegel 07 Feb '11

07 Feb '11

Hi all, i have a working freeswitch git on ubuntu lucid 64bit. I setup a conference. I add these line in my call-control-section: <control action="execute_application" digits="3" data="playback /fssounds/helpconference.wav"/> I press 3 the file will be played correctly. After the playback i go back to the conference. I talk to others in the conference. after the "execute_application" command i have many delays 3 or 5 or more seconds. My voice is very clearly. the people mean i.m a voicecomputer-). I must exit the conference and i start again. my voice is normal. is execute_application a problem? can you help me please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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mod_fsk not detecting call waiting caller id
by Anirudha Shimpi 07 Feb '11

07 Feb '11

Waiting for digit 'D' did not yield any results. Any other ideas? I am very much interested in getting call waiting caller id to work with Freeswitch and analog lines. Any suggestions would be very helpful. If there are any modifications that can be done to mod_fsk to get it to work, or writing a new module for this purpose, I am ready to help in any fashion. Thanks

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Re: [Freeswitch-users] Grandstream Freeze on Originate command
by rex.alex 07 Feb '11

07 Feb '11

Hello, This solution worked for me as well. Thank you so much. Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Origina… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Using mod_opal with MSN Messenger
by mazilo 07 Feb '11

07 Feb '11

Has anyone managed to configure mod_opal to register to MSN Messenger server to place/receive calls to/from any MSN Messenger users? I took a look at the conf/autoload_configs/opal.conf.xml file and don't know what to fill the gk-address, gk-identifier, and gk-interface. Anyone? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Using-mod-opal-with-MSN-Messe… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Lua Session creation is blocking?
by Stephen Wilde 07 Feb '11

07 Feb '11

I have a lua script that originate a new call creating a session as: outbound_session = freeswitch.Session(dialstring, session); The problem I have is that session creation sometimes is blocking and sometimes is not blocking. To me it seems that is related to the RTP so the session creation is blocked until early media is present or until the called answer the call. Is this true or I have miss something? There is a way to do a non blocking session creation? Stephen

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blocking 183 w/o sdp
by Sam 07 Feb '11

07 Feb '11

Hi, how can i ignore 183 without sdp, what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>" the FS sends 180 to the leg a. Here i want to block the 183 with SDP just like router as b2bua and send nothing to leg a, and when actual 183 with sdp comes it should send . Its because, providers are sending false signaling by sending 183 without sdp,and it hampers while @ production, Although by cisco sbc i have done this but i want to do it by FS, Take a scenario, when call is send 183 without sdp for 10 secs and then followed by 183 with sdp ( actual signal), but when some one dials invalid number it rings for 10 secs and then gives SIP cause 404, which is bad from the providers. So this is the reason i want to block it. Most of the providers do this, the way out is blocking. I have got an advice from Tihomir to do "execute_on_ring and parse your 180 / 183 messages in search of SDP, once you get 183 without SDP do not send it back to leg a and send signal only when you got 183 with sdp or 180 " And this could be valid call flow. This happens in many cases where the provider is having nextone as a sbc and that too tier 1 ! Regards Sam

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mod_h323 error
by Gustavo Espeche 07 Feb '11

07 Feb '11

Hello we are having a error with mod_h323 2011-02-06 22:04:06.999851 [ERR] mod_h323.cpp:2150 AUDIO RTP REPORTS ERROR: [Missing local port] when this error appear not more calls connect. someone have this error with mod_h323. regards. Gustavo Espeche www.easyipcall.com

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