March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

gtalk + oauth
by Sayyed Mohammad Emami Razavi 30 Mar '13

30 Mar '13

We should not use username and password directly in FS! How about OAuth + OpenID? e.g. My customers login with openid and then with token, i want to connect them to their gTalks via [Dingaling]? Any better and good Idea?!

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freeswitch logging to /var/run/freeswitch
by Michel Brabants 29 Mar '13

29 Mar '13

Hello, I have freeswitch running in a redhat-setup and I keep getting logs in /var/run/freeswitch/ from database-queries, ...: PGAPI, ... Where is this controlled? mod_logfile is not controlling this as far as I can tell. Thank you and kind regards, Michel

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Friday Free For All?
by Jeff Leung 29 Mar '13

29 Mar '13

Sure today's a holiday for most people, but Friday Free For All Activate?

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uuid_fileman seek ms
by Tamas.Cseke 29 Mar '13

29 Mar '13

Hello, I'd like to seek a playback by ms. http://wiki.freeswitch.org/wiki/Mod_commands * seek:<+[samples]>|<-[samples]> "Samples are the literally the number of samples in the file to jump forward or backward. In an 8kHz file, 8000 samples would represent one second, in a 16kHz file 16000 samples would be one second, etc." switch_ivr.c samps = step * (codec->implementation->samples_per_second / 1000); wiki says it is done by samples, but I checked the code and I think its in ms. Is it a documentation bug? I also don't understand why the codec sample rate is used It is not a problem, if the file sample rate is different? I'd like to make a seek in conference (http://jira.freeswitch.org/browse/FS-5211) I used the conference sample rate there, but I think its not correct only if the file rate matches the conference rate Could you please explain me how it should work? Thanks in advance, Tamas Cseke

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Adtran
by Blakelund Priddy 29 Mar '13

29 Mar '13

Does anyone here know how to globally set all outbound CID from adtran from Unknown to something someone would answer ha Sent from my iPhone

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Re: [Freeswitch-users] Core.db MSSQL
by Andre 29 Mar '13

29 Mar '13

I'm trying to figure out how to get the Core.db into a Microsoft SQL Server. In Switch.config.xml have <param name="auto-create-schemas" value="true"/> <param name="auto-clear-sql" value="true"/> <param name="core-dbtype" value="MSSQL"/> <param name="core-db-dsn" value="dsn:sqldb::" /> I have a DNS setup called sqldb using ODBC DRIVER 11 FOR SQL SERVER. I don't see anything happening and I don't see a database structure being built. What did I do wrong? Thanks Andre

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VOIP Monitor
by Antonio Teixeira 29 Mar '13

29 Mar '13

Good Afternoon. I have been looking a _Good_ voip monitor solution not just a CDR reader but something to intercept , measure and analyze RTP streams. I have take a look at Solar Winds, Homer and some others, etc does anyone have any input on this matter ? We have all the normal problems of why are calls not been completed , is the signaling from the SBC reaching the media servers , why does extension X can't reach register, etc So you guys have any input free or commercial ? Have a good easter btw. Thanks Antonio

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callee id inbound
by Erwan Hermouet 29 Mar '13

29 Mar '13

Hi all, I use freeswitch as voip server for other ipbx system. My freeswitch server is connected to gateway. Incoming calls works. I have extension, use as voip trunk for other server. Inbound rule redirect to this extension and it's works, but we don't received called number on extension (sip trunk). Caller ---- freeswitch ---- extension (other ipbx) On extension we never receive original caller number, and like that on this ipbx we can't forward inbound rules. Here my public.xml <extension name="xx3278" continue="true"> <condition field="context" expression="public" /> <condition field="destination_number" expression="^xxx278$" > <action application="set" data="dialed_ext=xxx3278"/> <action application="set" data="effective_callee_id_name=xxx3278" /> <action application="export" data="effective_callee_id_number=xxx3278" /> <action application="transfer" data="12345 XML default" /> </condition> </extension> On extension we only receive incomming call from extension 12345, which is my extension on freeswitch. Tks advance for your help

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problems with :_: bridging and call_timeout
by Trever L. Adams 29 Mar '13

29 Mar '13

The following is based on examples in the wiki. If I only have one FreeTDM end point it works. If I only use "," bridging it works, but not all phones ring and a lot of other weirdness. If I use the multithread dialing (:_:), all sorts of problems arise. 1) It no longer detects when someone hangs up. It rings and rings. 2) It no longer goes to voice mail no matter what I do. This seems to be whether I use loopback or not. I only use loop back for the retrieve-vm capability and the example showed it. <condition field="chan_name" expression="^FreeTDM/3:1/">  <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=true" /> <action application="bridge" data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/> <action application="answer"/> <action application="sleep" data="300"/> <action application="hash" data="insert/$${domain}-retrieve_vm/global/${uuid}"/> <action application="set" data="vm-alternate-greet-id=$${FAMILY_PHONE_NUMBER}" />  <action application="voicemail" data="default $${domain} $${FAMILY_PHONE_NUMBER}"/>  </condition> <extension name="retrieve-vm"> <condition field="destination_number" expression="^(\#)$"> <action application="intercept" data="${hash(select/$${domain}-retrieve_vm/global/${uuid}"/> <action application="answer"/> <action application="sleep" data="1000"/> </condition> </extension> This is strange because this is my normal dialplan (disabled when I am trying to get voicemail up) <extension name="direct_inbound"> <condition field="chan_name" expression="^FreeTDM/3:1/"> <action application="bridge" data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/> <action application="hangup"/> </condition> </extension> It works absolutely fine! Is this a bug in call_timeout, or am I missing something. As shown above, I have even tried to ignore early media. Thank you, Trever P.S. I would greatly appreciate help getting this working. It is the last piece I need to use FreeSWITCH exclusively for my folks who are trying to get rid of fax machines, answering machines, etc. as well as get some call screen and Do-Not-Call-List integration. -- "It is error alone which needs the support of government. Truth can stand by itself." -- Thomas Jefferson

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inbound-proxy-media with attended transfer
by Jeff Pyle 28 Mar '13

28 Mar '13

Hello, Is inbound-proxy-media compatible with attended transfers in any way, shape or form? Is there any way to avoid: [ERR] switch_core_io.c:1151 Codec PROXY PASS-THROUGH encoder error! Is there any way to emulate the behavior of a "proxy-media-after-att-xfer" option? I strongly prefer the codec transparency of inbound-proxy-media, but I do need attended transfers, even if the ringback is always with G711. SIP transfers in general are relatively new to me, so I very well may be asking for my cake, eating it, and then wanting the bakery to deliver daily to my office for free. Forgive me if that's the case. - Jeff

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