Re: [Freeswitch-users] Creating Default Accounts on Directory
Please check your dialplan to match the new extension. You are looking for dialplan/default.xml extension Local_Extension. Check the cond destination_number, it should give you a good hint. Regards, JM On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz darklio...@yahoo.com wrote: Hi Sir, I want to create a new xml file on the default directory of freeswitch where 1000.xml is located, sample i created 9387821.xml and copy the contents of the 1000.xml. The problem is when I used the account 9387821.xml and call 1000.xml it doesn't work the message in freeswitch it always CS_DESTROY... Please help me this with issue thanks... Edmar -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- João Mesquita FreeSWITCH™ Solutions t: +1 (646) 4959927 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Link between Use-context and dialplan
inline
JM
On Sun, Dec 13, 2009 at 1:40 AM, Otis ab...@greatiam.com wrote:
Sorry
I posted this earlier but did not do the due diligence and sent it with
so much typo them meaning does not come out:
In a nutshell I would like to know :
1. How FS would know which dialplan to use for an extension with user
context other than default.
The SIP profile that the call comes in has a context. All calls that do not
have users associated (not authenticated) or users that do not have the
user_context var set will use that context.
If the user has the user_context var set, it will use the specified one.
2. If a file file has to be created does the name matter
No.
3. Where should that file be located.
${FSROOT}/conf/dialplan/*
I *strongly *suggest you to read the default configs and the wiki.
Thanks.
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Re: [Freeswitch-users] Java ESL
Can't we just swig it to Java? JM On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby niall.cro...@gmail.comwrote: Hi, I am about to start writing a Java Event Socket Library as I can't find one already written thats available. 1 - Is there one already out there? 2 - If not, any pointers as to what design I should follow? Which of the current ESL's is the best modal to follow? Thanks, Niall. -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- João Mesquita FreeSWITCH™ Solutions t: +1 (646) 4959927 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer
That is more dependent on the endpoint than on the switch itself. I guess you can always use mod_limit to come up with some crazy key to identify one endpoint or the other but still it seems overly complicated for something that is not supposed to be working this way. You can also park the call instead of transferring, can't ya? JM On Wed, Dec 9, 2009 at 3:13 PM, Peter P GMX prometheus...@gmx.net wrote: Hello, in our dialplan we have enabled multiple-registrations, so 2 phones can register on a single directory entry. param name=multiple-registrations value=true/ Both phones are registered, both phones can be called and each phone can call the other phone. However in an attended_transfer mode calls cannot be transferred to the other phone with the same number. Attended_transfer in this case is needed when you take a call on your main SIP phone and and then want to transfer it to your mobile DECT/SIP phone, because you may have to check something in another room. I did a SIP trace and see the following: * A invites B(phone 1) = ok * B(phone 1) places call on hold = ok * B(phone 1) dials number B(phone 2 DECT) on second line * Freeswitch send Invite to B(phone 1) = ok * Freeswitch send Invite to B(phone 2 DECT) * B(phone 2 DECT) sends Ringing to Freeswitch = ok * B(phone 1) sends Busy to Freeswitch * B(phone 1) displays Busy and hangs up the second line Is there any way to overcome this? Is there a way to ignore the Busy from phone 1 when phone 2 answers Ringing? Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- João Mesquita FreeSWITCH™ Solutions t: +1 (646) 4959927 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] compilation error of skypiax_protocol.c
Maybe, just maybe isse that make target to reconf libtiff? Regards, JM On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang jingwei.y...@gmail.com wrote: I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o at_interpreter.c: In function ‘command_search’: at_interpreter.c:5299: error: ‘COMMAND_TRIE_LEN’ undeclared (first use in this function) at_interpreter.c:5299: error: (Each undeclared identifier is reported only once at_interpreter.c:5299: error: for each function it appears in.) at_interpreter.c:5308: error: ‘command_trie’ undeclared (first use in this function) at_interpreter.c: In function ‘at_interpreter’: at_interpreter.c:5424: error: ‘at_commands’ undeclared (first use in this function) make[8]: *** [at_interpreter.lo] Error 1 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang jingwei.y...@gmail.comwrote: No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, That one is on your side. If you changed/updated system libs it might be worth doing another ./configure Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory make[8]: *** [at_interpreter_dictionary.h] Error 127 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 Do you have idea about this one? Thanks! On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.cawrote: Consider it fixed. Committed revision 15765. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... *cc1: warnings being treated as errors* skypiax_protocol.c: In function ‘X11_errors_handler’: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_send_message’: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_do_skypeapi_thread_func’: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
Re: [Freeswitch-users] CDR records
What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions. JM On Tue, Dec 1, 2009 at 3:31 PM, Michael Collins m...@freeswitch.org wrote: On Sun, Nov 29, 2009 at 10:06 AM, Puskás Zsolt erro...@gmail.com wrote: Hi Guys! I'm using the latest svn (15711) with the default xml config. Only modified cdr_csv.conf.xml the line param name=legs value=a/ to param name=legs value=ab/ Here is what i do: 1. 1000 calls 1001 (1001 answers the call) 2. 1001 do blind transfer to 1002 (using *1) 3. 1001 hangs up 4. 1002 answers the call 5. 1002 and 1000 hangs up 3 cdr records are generated (simplified): from,to,start,duration 1000 1001 2009-11-29 15:21:53 53 50 1000 1002 2009-11-29 15:21:53 79 76 1000 1002 2009-11-29 15:22:46 26 23 As you can see the second cdr is incorrect because 1000 doesn't speak with 1002 for 76 second. Is this a normal ? Is it possible to make only 2 record ? You may want to turn on mod_xml_curl and look at XML CDRs, comparing them to the corresponding CSV files. That should help you figure out why the values in the CSV are what they are. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fwd: passive recording
These guys can on E1, not T1. They are not compatible with FS just yet, but we are working on it. Let me know off-list if you are interested. JM On Wed, Nov 25, 2009 at 3:29 PM, Imthiyaz Ahmed imthiy...@gmail.com wrote: I mean to tap tx and rx of a PRI line using sangoma tap and record the call information and actual calls without distrubing the existing line . freeswitch will work in passive mode like trunk side call recorder. Thanks Imthiyaz On Wed, Nov 25, 2009 at 10:43 PM, Anthony Minessale anthony.miness...@gmail.com wrote: What do you mean by passive encoding? On Wed, Nov 25, 2009 at 11:00 AM, Imthiyaz Ahmed imthiy...@gmail.com wrote: hi is it possibe to enable passive recording in sangoma tdm interface in feeswich. pls advice Best Regards G.Imthiyaz Ahmed -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS mod_SQL
This is the final answer: http://wiki.freeswitch.org/wiki/Mod_xml_curl JM On Sun, Nov 15, 2009 at 1:39 PM, Samuel Mukoti samuelmuk...@gmail.comwrote: Hi, I'm a newbie to FS, and I wanted to implement a setup where I provision the sip endpoints though a SQL database like mysql and also manage call routing too? Is this possible since I understand FS uses XML config files. Best regards Sam ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Displaying caller ID on LED?
If you donate one to the FsGui project, I can make it happen for you. Contact me off list if you are interested. Regards, JM On Wed, Nov 11, 2009 at 1:01 AM, Mitch Capper mitch.cap...@gmail.comwrote: I did something like this recently.From the dial plan it is easy to execute an external application on an incoming call with the caller's info. At that point if you can just push it down to the LCD panel all the better, but if your FS server is remote, and has no direct access to the client to render the caller ID, you will have to setup a fake push to get instant responses. You can do this through apache, or a simple tcp server but the idea being the client connects up to the server, and the server blocks until an incoming call comes in, it then responds to the client, and you have the caller id fairly instantly showing up. You could also use the event socket, heck even maybe use the event socket remotely if you wanted to, and then avoid some of the server side complexity too. ~Mitch On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 codecompl...@free.fr wrote: ... or alternatively, on one of those USB digital picture frames? www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B87BHC -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] PortAudio needs work on Mac OS X 10.6
Or write one for Mac specifically since PA is fine for all the rest (I think)? JM On Mon, Nov 9, 2009 at 2:50 PM, Anthony Minessale anthony.miness...@gmail.com wrote: maybe we should write a new audio abstraction lib =D On Mon, Nov 9, 2009 at 2:07 AM, Bruce Fletcher br...@nani.ca wrote: The patch from the PortAudio site does get the library to build, but it still fails with the same assertion when I try to play MOH. The patch I'm talking about is this one: http://www.portaudio.com/trac/changeset/1418 If the same build problem applies to other 64 bit systems, it might be a good idea to incorporate this patch. It looks clean and reasonable to me, at least. I've managed to work around the problem entirely by building FreeSWITCH for i386, but I'll go ask the PortAudio folks what the status is of their 64 bit support. They are clearly assuming 32 bit long integers in some places, which is hopefully on a to-fix list somewhere. Thanks, - Bruce On 2009-11-08, at 12:25 PM, Michael Jerris wrote: If you can figure out a clean way for us to do this with proper ifdefs in tree in a way that will not break others that would be the most preferred. Mike On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: OK, I'll ignore that MacPorts patch for now and try to find a better approach. I'll look into this further tonight, but this morning I found a more recent promising patch on the PortAudio site: http://www.portaudio.com/trac/changeset/1418 It seems to push some data types to 32 bit regardless of platform, which might work better than the MacPorts approach of migrating some data structures to 64 bit. At any rate, this patch being on the PortAudio site suggests it might be a more approved fix. I'll keep plugging at this in my free time and report any significant progress back to the list. Thanks, - Bruce ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Cordless VOIP Phones
I have Siemens A58IP and Snom M3. Both work very well with pros and cons. Nonetheless, both lack HD JM On Mon, Nov 9, 2009 at 4:44 PM, Dave Stevenson steve...@primrosebank.netwrote: Hi, has anyone any good results to share with using cordless phones for VOIP with FreeSwitch ? I have seen a few around that appear to operate with wireless networks and make SIP connections to VOIP PBXs. I have seen various models from Engenius, Prestige, DORO and Siemens as well as Snom. regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Cordless VOIP Phones
Beat me with a dead cat all you want but I rather the snom m3 than the Siemens A580IP Siemens has very low volume which makes its call quality suck despite of being ergonomic and all... That gigaset application sucks and the base station is slow as hell... Maybe I have a bad unit? The snom m3 has its downsides, but all and all, I am happy with the phone if you consider its price tag here in South America where a Polycom can easily cost over 200USD the cheapest unit. Regards, JM On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale anthony.miness...@gmail.com wrote: asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3 On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale jcas...@activenetwerx.com wrote: The Snom M3 is one of the ones that I was looking at - I would be interested in the Pro's Cons ? Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad... I have Aastra 480i CT's which work well. jlc ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Extension: No audio
It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise,
no go
Have you changed the ext-sip-ip too?
Regards,
JM
On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi again,
Actually, changing the param name=ext-rtp-ip value=auto-nat/ to
param name=ext-rtp-ip value=$${external_sip_ip}/ means that I
now see the IP address in the INVITE message:
v=0
o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 124.xxx.xxx.xxx
t=0 0
m=audio 21234 RTP/AVP 0 2 9 8 101 13
Why would this be? I thought auto-nat was meant to solve these issues?
However, I still do not see the TRYING or RINGING messages ideas
appreciated.
Thanks!
On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
OK.. thanks Mike.
I assume I am using the Internal profile. I have defined user 2000
in the 'directory' using a context called family: switch_ivr.c:1367
Transfer sofia/internal/1...@192.168.1.120 to xml[2...@family]
This is an extract from sofia:
sofia status profile internal
=
Nameinternal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
DialplanXML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.xxx
URL sip:mod_so...@192.168.1.120:5060
BIND-URLsip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEGfalse
PROXY-MEDIA false
AGGRESSIVENAT true
STUN-ENABLEDtrue
STUN-AUTO-DISABLE false
CALLS-IN100
FAILED-CALLS-IN 25
CALLS-OUT 38
FAILED-CALLS-OUT31
Registrations:
=
Call-ID:68534bba9b461...@58.169.138.53
User: 2...@192.168.1.120
Contact:user sip:2...@58.xxx.xxx.xxx:5060
Agent: dunno
Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)
Host: freeswitch
IP: 58.xxx.xxx.xxx
Port: 5060
Auth-User: 2000
Auth-Realm: markcs.dyndns.org
MWI-Account:2...@192.168.1.120
The internal.xml file has a lot in it, but I guess these are the
important things for this profile:
param name=ext-rtp-ip value=auto-nat/
param name=ext-sip-ip value=auto-nat/
param name=sip-port value=$${internal_sip_port}/
param name=rtp-ip value=auto/
I will try to change auto-nat to be $${external_sip_ip}
One question though: Any idea why I never see the TRYING or RINGING
messages? Are tehse related to the RTP IP address or not? Without
these I assume something is incorrect and I do not hear ringback
Thanks!
On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris m...@jerris.com wrote:
Your packet traces would disagree with the statements below. It is
sending your internal address in rtp, so its not set correctly on
whatever profile your using to call out,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my
domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com (which registers with
an external IP address), then I get audio. In sofia, I see my IP
addresses:
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
==
Nameinternal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
DialplanXML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.120
Ext-RTP-IP 124.xxx.xxx.xxx
SIP-IP 192.168.1.120
Ext-SIP-IP 124.xxx.xxx.x
URL sip:mod_so...@192.168.1.120:5060
BIND-URLsip:mod_so...@192.168.1.120:5060
HOLD-MUSIC silence
OUTBOUND-PROXY N/A
CODECS G726-32,G722,PCMU,PCMA
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO
Re: [Freeswitch-users] SPA3102 FreeSwitch HowTo
Look at this line on the freeswitch.fsxml and it will tell you exactly where the problem is. Beware that nested comments are not allowed in XML. -- JM On Wed, Nov 4, 2009 at 9:59 PM, Dave Stevenson steve...@primrosebank.netwrote: I am trying to follow the configuration give in the SPA3102 FreeSwitch HowTo. When I create the 00_spa3102.xml file, FreeSwitch won't load. If I rename the file (to, say .txt) then rename it to an xml once FreeSwitch is up and do a reloadxml command, I get an error flagged :- +OK [[error near line 3379]: missing ] I'm pretty sure the error must be in the 00_spa3102.xml file, but I can't see where the error might be - it looks identical to that on the Wiki page ? regards Dave ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Many CS_REPORTING state Zombie session
Dome, just to explain what Brian is saying: Doing billing inline in this case means that the session thread (what you see in show channels). If, for whatever reason, something goes wrong on the DB connection or task for the billing, this session thread will be stuck with it leaving it dangling around the system. This is what most likely is happening to you. The right way to do it is to let the session thread go (which mod_cdr_csv does) and then process the billing. None of the methods you have described do that. You might consider post-processing your CDR/billing information to avoid coming up with this kind of problem that are very hard to predict. Hope that helps, JM PS: Nonetheless, I still think it is valid to get a core dump like Rupa metioned. On Sat, Oct 31, 2009 at 2:44 PM, Brian West br...@freeswitch.org wrote: You should never do billing inline with the session thread is all I'm saying. /b On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote: I use odbc_query for retrive balance and get LCR from my billing DB. and use nibble_bill Dome C. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Many CS_REPORTING state Zombie session
No, mod_nibblebill definetely needs to be enhanced but it is not the problem and it can be used with high load traffic. The one I am not sure about is odbc_query since it was not developed for that. Do what Rupa said, please. Regards, JM On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org wrote: I think once you get the backtrace like rupa said we can see that maybe odbc_query is really hanging or something similar. /b On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote: 2009/10/31 Brian West br...@freeswitch.org: You should never do billing inline with the session thread is all I'm saying. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_nibblebill and memory problem
Why don't you get us more information for debugging? We could use some vg output, maybe? JM On Mon, Oct 26, 2009 at 9:24 AM, Dome Charoenyost d...@tel.co.th wrote: Dear All, I'm running mod_nibblebill for my prepaid solution. I still have problem with memory. I have 4 GB RAM and runing debian squeeze 64 bit and 200 calls concurrent Last time nibblebill running with 1 min heartbeat. when i check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr. and then FS crash. Now i change hearbeat to 0 it's mean nibble update balance when end of call. but everything are same FS start from 2% and growth to 20% in 2 days. When i unload nibblebill FS running fine. My question is when concurrent calls drop to 1-2 calls why FS (I think nibblebill) still use memory ? something wrong in nibblebill ? BG Dome C. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting to FS CLI...just hangs..
Hangs for how long? Are you sure you are not just waiting on a timeout? JM On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: It just hangs….and I CTRL-C out of it. [r...@ss]# ./fs_cli -H 127.0.0.1 ^C [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error] Freeswitch process is running: [r...@ss bin]# ps -ef|grep free root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch root 8952 31039 0 12:42 pts/200:00:00 grep free ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Brazilians (Off-Topic)
We do. :-) João Mesquita On Fri, Oct 16, 2009 at 11:02 AM, Itamar Reis Peixoto ita...@ispbrasil.com.br wrote: I think no. On Fri, Oct 16, 2009 at 10:48 AM, Pedro Prado pedropr...@msn.com wrote: Hi, Do you have a group of Brazilians here? Thanks, Pedro Prado -- Itamar Reis Peixoto e-mail/msn: ita...@ispbrasil.com.br sip: ita...@ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Brazilians (Off-Topic)
Just a heads up, I have talked to Jeremias from Khomp today and he is setting up the wiki. I will personally be adding contents to the that wiki if it ever picks up. Regards, jm On Sat, Oct 17, 2009 at 12:52 AM, Jason White ja...@jasonjgw.net wrote: Diego Viola diego.vi...@gmail.com wrote: I'm with Moises and with the other people supporting this initiative. I'm not Brazilian, but they should be able to do whatever they want, after all, that's how open source works, if you can do it go ahead and do it. Correct. We have enough of them as far as English-language fora are concerned; other languages are a different question altogether, though. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] adding new extension
Never _EVER_ change the freeswitch.xml.fsxml file directly. If you need something dynamic you would have to implement directory using mod_xml_curl, otherwise, you change the files on the conf/ tree and do a reloadxml on the CLI. JM On Wed, Oct 14, 2009 at 8:58 AM, srinivasula reddy srinivas.ksvre...@gmail.com wrote: Hi, can any please let me know, how to add new extension(eg 1000.xml) dynamically while running freeswitch? while running freeswitch i have created new xml(eg: 1500.xml) file and i have changed freeswitch.xml.fsxml still not working, Thanks Srinivasula Reddy K ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference call
Look at eavesdrop on the wiki. JM 2009/10/14 Nikita Belov nbe...@abisoft.spb.ru HI all, I want to configure FS to make special conference call between three users (A, B, C). In this conference C will hear A and B, but A will hear only B. Can I make it using FS API commands? Does anybody know which approach is better to use? ___ Thanks, Nikita ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Update: Valet Parking
I would say that the parking meter is a good idea and it is the default behavior of parking on legacy PBXs. Since we always do _more_, what do you think about having the option to transfer to any extension instead of just the one that transfered the call? Regards, jm On Mon, Oct 12, 2009 at 4:20 PM, russell.mosem...@cune.org wrote: Michael Collins m...@freeswitch.org said: On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi mayamatake...@gmail.comwrote: On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins m...@freeswitch.orgwrote: FYI, The FreeSWITCH devs have added valet parking! Check it out: http://www.freeswitch.org/node/207 Let us know what you think. Very nice. But I think a valet_unpark app is missing. If the intention of the person sent to the valet lot is to retrieve a call there, the person can assume the call was already retrieved by someone else or that the caller hung up if he/she hears MOH. But it would be nicer to have a valet_unpark app to fail and let the dialplan play a message. I understand what you are saying. I'm not sure I agree, but we'll kick the idea around when we have a few minutes and let you know what we decide. -MC If you do decide to implement something, I would encourage that it be flexible so that when the parking meter runs out :-), it could either play a message or forward the call to an extension (default to the extension that parked it). -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Questions regarding to mod_nibble
I am testing the latest version of nibblebill so let me see if I can help you with your questions. jm On Fri, Oct 9, 2009 at 12:39 AM, Ahmed Munir ahmedmunir...@gmail.comwrote: I want to ask three questions related to mod_nibble bills, as I'm listing down below; 1- Can we select/use dynamic tables for billing using nibble bill? What do you mean for dynamic tables? Like LCR does that you can specify your own SQL statement to be executed? If that's what you are asking, no, but it would be a nice todo. 2- Can we define more than two tables and attributes in nibblebill.conf.xml? What else do you want to define and how do you imagine it to behave? 3- As Nibble bill is use to deduct amount of user account, Can we deduct minutes instead of cash? Because my case is, if a user buy a package and I only want to deducts his/her minutes. How we can resolve it by nibble bill? / What other way we can resolve it? When we say cash on the column, we are really saying just a number that is being deducted, that's it. If you deduct 1 every 60 seconds, you will have your cash converted to minutes, won't you? Kindly advise soon. -- Regards, Ahmed Munir ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones
Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.comwrote: If you have time to take a look, I could put a trace in the pastebin? Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Thursday, October 01, 2009 10:29 AM *To:* 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry -- *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com] *Sent:* Thursday, October 01, 2009 9:36 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: 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.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Port question
You shouldn't have to do nothing since we have NAT detection. Port 5080 (external SIP profile) is just for unauthenticated clients. jmesquita On Sun, Sep 27, 2009 at 3:00 AM, Henk Maaijen postb...@postbus.info wrote: Hello All, I am a total newbie with FS. I use pbxiaf but would really like to try out FS. I have installed FS And everything seems to be working. My problem is that i have some family scattered round the globe who are all logging in to to my piaf. I am not able to reconfigure those phones ( Some SIP and some IAX2 ). Would it be enough to redirect port 5060 in my router to port 5080? Or would it be possible to still use port 5060. ( I don't want to be too long offline for tests ) -- Best regards, Henk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server
Only 3 init scripts available on trunk today (${SVNROOT}/build) are for
archlinux, redhat or suse.
We would love to have more for other distros.
Regards,
jmesquita
On Sun, Sep 27, 2009 at 10:42 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Hi All,
I am trying to setup FreeSwitch on a Ubuntu Server.
Where can I find the start up(boot time) script for FreeSwitch on a Ubuntu
Server?
Thank you .
Lloyd
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Re: [Freeswitch-users] Bind extention to a different Dialplan andcdr php?
Frank, That kind of logic needs to be performed at your application, if I am not wrong. I will do some testing here, but I think that mod_xml_cURL sends purpose and other information regarding the type of data it is requesting so you can decide on your application exactly what to do. Regards, jmesquita On Sun, Sep 20, 2009 at 1:10 PM, Frank @ Impact fr...@impactfax.com wrote: Currently all incoming calls to my FS to all extensions are sent off by curl to a particular php script called dialplan.php. I would like to have certain extensions that are called to have their xml dialplan built by a curl to a different php script, say dialplan2.php. Is there a way to have certain extensions get their dialplan by calling a different php script other than the default? -Original Message- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Tuesday, September 08, 2009 3:37 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Bind extention to a different Dialplan andcdr php? On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact fr...@impactfax.com wrote: Is there a way to bind a particular extension to a different dialplan php and a different cdr php script than the default one? Could you re-phrase this question with a bit more detail? Thanks. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Any FreeSWITCH training courses out there?
You could always talk to consult...@freeswitch.org. They can help you with that! ;) And, this is probably best to be sent to the -biz, isn't it? (really asking, not being ironic) jmesquita On Sun, Sep 20, 2009 at 4:47 PM, Gavin Henry gavin.he...@gmail.com wrote: Hi all, Is there anyone out there doing beginner courses or conversion courses from an Asterisk mindset? Cheers. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skill-based ACD
Andrew, I am sorry for forgetting about you. This is exactly why asked if you were you on IRC the other day... Can you tell me if this is going to stay open source when production ready? jmesquita On 9/18/09, Andrew Thompson and...@hijacked.us wrote: On Thu, Sep 17, 2009 at 11:20:22AM -0700, Michael Collins wrote: I was curious about this myself. Even if someone has built a non-free skills-based ACD using FS I'd like to know about it. -MC I guess nobody paid any attention to my Cluecon presentation... :( http://wiki.opencsm.org/wiki/index.php/Spice_Telephony is a skill-based ACD that uses FS for its voice components. I havent pimped it here in quite a while but here's some of its major features * Skill based routing * Priority Queues (instead of just FIFO) * Multiple call types (voice, voicemail and email are currently supported, instant message support (via libpurple) is prototyped) * Outbound call support (no autodialer though) * Distributed system so you can aggregate multiple FS instances/locations into one big 'virtual' callcenter * Web-based agent and administrative interface There's quite a bit more, but that's the overview. The project is finally approaching a 1.0 after over a year of development - I hope to deploy it in production sometime around the end of this year or the beginning of 2010 (replacing my previous custom asterisk solution). You can grab the code at http://git.opencsm.org/index.cgi/spice-telephony/ (you can browse or git clone that URL). All you should need to run it is a modern erlang release (R12B5 or newer) and ruby/rake to run the build. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_nibblebill
Guys, I have been testing mod_nibblebill lately and there are 2 params that I could not make work. !-- If a call goes beyond a certain dollar amount, flag or terminate it -- param name=percall_max_amt value=1/ param name=percall_action value=hangup/ Looking at code, I could not find a single line that would actually test those. Is this confirmed to be implemented? If not, this should be removed from the configs so it won't get ppl lured. Regards, jmesquita ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skill-based ACD
I would be very interested in getting my poor programming skills into getting some decent real skill based routing working and shut those Avaya bastards up. Functional model? Get it to me and I will try to make it happen as time lets me. jmesquita On Thu, Sep 17, 2009 at 7:17 PM, Anthony Minessale anthony.miness...@gmail.com wrote: I can tell you from years of painful experience, don't use asterisk for queues. see http://www.freeswitch.org/node/117 You don't have to use FS, but please don't let the asterisk siren lure you to the rocks. mod_fifo is like a tool with basic functions you can exploit however you wish, it does not try to do high level features because those are best left in external logic. mod_fifo has priorities which means each individual fifo is really an array of 10 fifos when you set the priority you are choosing which index in the array to insert the caller. when an agent belongs to a queue he drills down the array from 0-9 so you could for instance put everyone in 5 by default and put more important people in 0 so they always go to the front when you assign an agent to take calls off hook you can set a fifo_pop_order variable that tells you which array indexes to service and in what order. so if you pretend slot 1 is for general problems and slot 2 is for hard problems you can put one agent in 1,2 and a more stupid agent in just 1 *shrug* On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen christian.jen...@teligence.net wrote: This would be a fantastic addition – my company is currently looking to Asterisk as a potential candidate for this if FS can’t do it. I want FS to win of course J *Christian Jensen* Software Development Manager Back Office -- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, September 17, 2009 11:20 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] skill-based ACD On Sun, Sep 13, 2009 at 8:01 AM, mark morreny markmorr...@gmail.com wrote: Hello Has any tried setting up an ACD based on skillset? The current out-of-box version of fifo does not seem to support acd based on agent skillset. Does anyone have any experience in doing it with some external scripting using lua or javascript? I am interested in hearing how others may have done it as I am trying to implement one myself. thx, mark I was curious about this myself. Even if someone has built a non-free skills-based ACD using FS I'd like to know about it. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] session record does not for very short calls
I think you need to upgrade your version before we even take a look at that... You are so far behind trunk right now and lots of things have been changed since then. Not sure if this would solve your problem but not a lot of ppl will look at your problem when you run this version. jmesquita On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact fr...@impactfax.com wrote: FreeSWITCH Version 1.0.trunk (12790M) I have this in my DP action application=set data=RECORD_ANSWER_REQ=true/ action application=set data=RECORD_STEREO=true/ action application=record_session data=/mnt/rd/file.wav/ works fine as long as the call is long enough. But if the call is only, say, 3-4 seconds long (or something very short like that), then the wav file is never created with the audio in it. Is there a work around for this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference performance
I would be really interested to replay your test on Linux. Would you be willing to provide me all the details and relevant files so I can reproduce the test with a Linux box here? If yes, contact me offlist and we can work together on this. Regards, jmesquita On Wed, Sep 16, 2009 at 2:56 PM, Роберт Тверитнер siniy...@gmail.comwrote: Hi guys! I've tested FreeSWITCH conference module performance trying to figure out maximum number of simultaneous calls my FS box can serve. It took all 100% of CPU with only 50 calls (in average depending on conference rate) and leaking stream handle messages started appearing. The environment I was testing in: OS - Windows Server 2007 SP1 64 Bit CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz RAM - 2 GB FreeSwitch version 1.0.4 (14460) I've written a test program that used to originate calls once in 5 seconds from the other box. These calls were routed to particular conference room I was testing. I had a number of rooms with different rate (8000-32000) and interval (20,30) settings and with perpetual-sound turned on steraming music continiously. I've switched off all unnecessary modules, but left logging on in order to trace what was happening later. Client test softphone used respective speex codec according to conference room rate. This is a dialplan I used: extension name=test_conference condition field=destination_number expression=^(800020)$ break=on-true action application=conference data=$...@default20/ /condition condition field=destination_number expression=^(800030)$ break=on-true action application=conference data=$...@default30/ /condition condition field=destination_number expression=^(1600020)$ break=on-true action application=conference data=$...@wideband20/ /condition condition field=destination_number expression=^(1600030)$ break=on-true action application=conference data=$...@wideband30/ /condition condition field=destination_number expression=^(3200020)$ break=on-true action application=conference data=$...@ultrawideband20/ /condition condition field=destination_number expression=^(3200030)$ break=on-true action application=conference data=$...@ultrawideband30/ /condition /extension My questions are: Do you know any way I can increase my FS conference capacity? What do I have to tune in FS or in my box? Best regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] possible sofia_contact bug
Just thinking out loud. Wouldn't be sofia_contact 180...@192.168.1.163 ? jmesquita On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson woodydick...@gmail.comwrote: Hi, I am having a strange problem here. sofia status shows that the user is registered, but sofia_contact says the user is not registered. Does anyone know why this is happening? freeswi...@localhost.localdomain sofia status profile internal reg 180004 API CALL [sofia(status profile internal reg 180004)] output: Registrations: = Call-ID:530339592782-1484696326...@192.168.1.163 User: 180...@192.168.1.102 Contact:180004 sip:180...@192.168.1.163:9000 Agent: Voip Phone 1.0 Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) Host: localhost.localdomain IP: 192.168.1.163 Port: 9000 Auth-User: 180004 Auth-Realm: 192.168.1.102 = freeswi...@localhost.localdomain sofia_contact 180...@192.168.1.102 API CALL [sofia_contact(180...@192.168.1.102)] output: error/user_not_registered freeswi...@localhost.localdomain freeswi...@localhost.localdomain sofia_contact user/180004 API CALL [sofia_contact(user/180004)] output: error/facility_not_subscribed ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Chat redirect
I am anxious to provide my first real patch into FreeSWITCH and since this looked like a good candidate, I looked at the code for a little while and I have a few thoughts about the subject. FreeSWITCH (mod_sofia) does not route chat messages to endpoints who are not reachable (obviously). If you look at the API, the mod_sofia won't even take the message if endpoint is not registered and will respond with Cannot find user. So, basically, to implement what you are looking for, you need to have hooks set upon message receival (from mod_sofia point of view). mod_sofia only sends events on ESL when message has been sent to the destination endpoint. The way I see, there are 2 options here. The quick way and the hard (not so hard) way. The quick way is to just fire an event when registered user is not found and it will depende on something external to replay the message when user is offline. The longer way is to make the core queue offline messages and deliver them when user register. What I would like to hear from the core dudes is, which one is wanted? None is a good answer too. Regards, jmesquita On Fri, Sep 11, 2009 at 9:16 PM, Michael Jerris m...@jerris.com wrote: This would require changes to the c code in mod_sofia. If you have a patch to change this behavior (probably should address configuration and authentication as well as this could be a denial of service path) you can post it to http://jira.freeswitch.org. Mike On Sep 6, 2009, at 6:32 AM, Juan Backson wrote: Hi Brian, From the event socket, there is no message received when a MESSAGE is sent from one sip user to another. If both users are registered, I can send message between them. But if the receiving party is not registered, I want to be able to store it. However, there is no way to intercept this MESSAGE. Is there anyway to solve this problem. thx, jb On Sun, Sep 6, 2009 at 2:36 AM, Brian West br...@freeswitch.org wrote: Not automatically. But you could use the event socket to get the message and forward it via ESL. /b On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: If so, how can it be done? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Implementing h extension in FS
Try to explain a little bit better what add_cdr does right here. Unlike
Asterisk, FreeSWITCH do have lots on information on CDR and it feels like
you are trying to do things on the wrong place.
If you want to understand where I am going with this, take a look at this
example XML CDR that can be posted by FreeSWITCH to a webserver at the end
of a call: http://wiki.freeswitch.org/wiki/Example_XML_cdr
Also you might want to check this referece here:
http://wiki.freeswitch.org/wiki/Mod_xml_cdr
jmesquita
On Thu, Sep 10, 2009 at 2:44 AM, Josh Rivers j...@radianttiger.com wrote:
You should be able to handle hangups in one of two ways:1) Register a
hangup handler in your script or dialplan. This will execute a script on the
hangup of the call.
2) Use the Event Socket Layer(ESL) to listen to hangup events and then
perform your actions there.
You can find more about these options in the wiki.
On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
HI,
I'm newbie in FS, I want to know how to implement h extension of asterisk
to FS. As I listed down below;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
My other question is, which application/function/class is use in mod_perl
to check the channel status? i.e. busy, answer,hangup,ringing,etc.
Kindly advice me soon.
--
Regards,
Ahmed Munir
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Re: [Freeswitch-users] filter in fs_cli
No can do. There are better tools to do that. tshark, wireshark and all other variants can do that for you. jmesquita On Thu, Sep 10, 2009 at 3:29 AM, Dome Charoenyost d...@tel.co.th wrote: How to use filter with sofia trace on ? Like Asterisk we can debug sip by sip set debug ip xx.xx.xx.xx. BG Dome C. 2009/9/10 Michael Collins m...@freeswitch.org: On Wed, Sep 9, 2009 at 1:36 PM, Michael Collins m...@freeswitch.org wrote: On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost d...@tel.co.th wrote: Dear All, I'm looking for document,example for /filter command. where to get it ? This is a handy way to add filters to what you see on the fs_cli. Event sockets allow for filters and the /filter command lets you add them to your fs_cli session. Check this page for specifics: http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC Also, I forgot to mention that this is used in conjunction with the /event' command. Open fs_cli and execute these commands: /log 0 /event plain all At this point you will get no log messages and just events. Now you can filter them as needed. Example: /filter Event-Name CHANNEL_EXECUTE Have fun! -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Checking Busy Status
No expert in perl but you are looking for hangup_cause variable. Check how to get channel variables from a session in perl and you are set. jmesquita On Mon, Sep 7, 2009 at 2:47 AM, Ahmed Munir ahmedmunir...@gmail.com wrote: Hi, Thanks for quick reply. I want to know how can I apply USER_BUSY in perl? Like for hangup I'm calling it from function Freeswitch::CoreSession i.e. $session-hangup(); Do I have to call it as listed below; $session-USER_BUSY(); or there other way around in perl? Kindly do let me know. -- Forwarded message -- From: Mathieu Rene mrene_li...@avgs.ca To: freeswitch-users@lists.freeswitch.org Date: Sun, 6 Sep 2009 22:01:58 -0700 Subject: Re: [Freeswitch-users] Checking Busy Status The hangup cause will be USER_BUSY. You can hop on #freeswitch if you need some explanations. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 6-Sep-09, at 9:58 PM, Ahmed Munir wrote: Hi, How can I check the busy status in FS? I've searched all the wiki pages i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but couldn't find checking busy status. I've written a perl script but couldn't complete it because theres no any function or class regrding busy status. Kindly let me know how can I check the busy status in mod_perl and also in dialplan tools as well. -- Regards, Ahmed Munir ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Ahmed Munir ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ESL C questions
Remko, I wrote the documentation that is on docs.freeswitch.org Take a look there, it is far from being complete but it might help. jmesquita On Mon, Sep 7, 2009 at 10:26 AM, Remko Kloosterman r.klooster...@mtel.nlwrote: Hi there, I wonder, is ESL documentation available for C or does someone have something in draft? I’m trying to write an outbound socket application for some generic IVR features. I didn’t find exactly that on the wiki except TODO J. The perl/ruby/javascript pages help a bit and the libs/esl source code provides examples that seem useful for trial and error, but I’d rather understand a bit more first. Right now I have a socket server that forks a process, answers a call, generates beep and plays voice. How can I retrieve digits? Place an outbound call and bridge both legs or retrieve a cause if the call failed? Send/receive SIP INFO? Disconnect the call with some cause code? And all that (and some more) in C. Any help or pointers is appreciated. Thanks, Remko ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] restart when convenient
Look at the fsctl api on the wiki. It has what you need. jmesquita On 9/4/09, Christian Löschenkohl christian.loeschenk...@xpirio.com wrote: hello i'm looking for a possibility to restart freeswitch like it is possible with asterisk. for me i tried to created a script that looks for open channels and if no channel is open it restarts freeswitch with the init script (not the most efficient way). i think i would be great if we would have a buildin function for this, i think such command would help with maintenance and not only for me. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] remote endpoints
Check the password dialog. It will tell you what the username/password is. post the logs for a call as well, please. Regards, jmesquita On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer e.schmidba...@gmail.comwrote: = Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_so...@192.168.0.125:5080 BIND-URLsip:mod_so...@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS c...@48000h,c...@32000h,sp...@32000h@20i,sp...@16000h@20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT9 Registrations: = Call-ID:MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1...@server1.altpressonline.com Contact:1000 sip:1...@69.204.30.67:16006 ;rinstance=50cbd6140e1ab991 Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com = sorry, i don't know how to login to pastebin On Mon, Aug 31, 2009 at 1:42 PM, Michael Collinsm...@freeswitch.org wrote: Could you give a few more details? For example, could you pastebin the output of sofia status profile external? -MC On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer e.schmidba...@gmail.com wrote: I am unable to call a user outside of our local area network. the user is registered on the external profile but there is no way to call the phone. does anyone have any suggestions how to do this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] remote endpoints
1. 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel sofia/external/anonym...@anonymous.invalid entering state [terminated][ 487] The far end seems to be replying with 487 - Request Terminated... Nothing wrong on FS, seems to be a problem with your endpoints. Can you enable a sip trace? jmesquita On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer e.schmidba...@gmail.comwrote: thanks...heres the pastebin: http://pastebin.freeswitch.org/10171 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Check the password dialog. It will tell you what the username/password is. post the logs for a call as well, please. Regards, jmesquita On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer e.schmidba...@gmail.com wrote: = Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_so...@192.168.0.125:5080 BIND-URLsip:mod_so...@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS c...@48000h,c...@32000h,sp...@32000h@20i,sp...@16000h@20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT9 Registrations: = Call-ID:MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1...@server1.altpressonline.com Contact:1000 sip:1...@69.204.30.67:16006;rinstance=50cbd6140e1ab991 Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com = sorry, i don't know how to login to pastebin On Mon, Aug 31, 2009 at 1:42 PM, Michael Collinsm...@freeswitch.org wrote: Could you give a few more details? For example, could you pastebin the output of sofia status profile external? -MC On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer e.schmidba...@gmail.com wrote: I am unable to call a user outside of our local area network. the user is registered on the external profile but there is no way to call the phone. does anyone have any suggestions how to do this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] remote endpoints
Problem is definetly on far end. If you look at the siptrace, you have the following sequence: 1. Asterisk calls in 2. FreeSWITCH replies with a Trying(100) to complete call right away and proceeds to dialplan 3. FreeSWITCH invites (calls) 7 times the final destination that never responds. 4. Asterisk sends a CANCEL message In all that, your final endpoint never responds to any message. Are you sure you can reach it? jmesquita On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer e.schmidba...@gmail.comwrote: here's the sip trace: http://pastebin.freeswitch.org/10172 2009/8/31 João Mesquita jmesqu...@freeswitch.org: 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel sofia/external/anonym...@anonymous.invalid entering state [terminated][487] The far end seems to be replying with 487 - Request Terminated... Nothing wrong on FS, seems to be a problem with your endpoints. Can you enable a sip trace? jmesquita On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer e.schmidba...@gmail.com wrote: thanks...heres the pastebin: http://pastebin.freeswitch.org/10171 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Check the password dialog. It will tell you what the username/password is. post the logs for a call as well, please. Regards, jmesquita On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer e.schmidba...@gmail.com wrote: = Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_so...@192.168.0.125:5080 BIND-URLsip:mod_so...@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS c...@48000h,c...@32000h,sp...@32000h@20i,sp...@16000h @20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT9 Registrations: = Call-ID:MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1...@server1.altpressonline.com Contact:1000 sip:1...@69.204.30.67:16006;rinstance=50cbd6140e1ab991 Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com = sorry, i don't know how to login to pastebin On Mon, Aug 31, 2009 at 1:42 PM, Michael Collinsm...@freeswitch.org wrote: Could you give a few more details? For example, could you pastebin the output of sofia status profile external? -MC On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer e.schmidba...@gmail.com wrote: I am unable to call a user outside of our local area network. the user is registered on the external profile but there is no way to call the phone. does anyone have any suggestions how to do this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users
Re: [Freeswitch-users] Authorizations when using DNS SRV bug?
Bkw, I would recommend charging a fee from callcentric for the consultancy. This consultant thing can get you going someday! LOL Jmesquita funny joke On 8/28/09, Carlos S. Antunes c...@nowthor.com wrote: Brian, You've been vindicated. Callcentric is now advertising zero weighted SRV records! :) I've re-enabled SRV lookups for the Callcentric profile and will monitor to see if I get any errors. Carlos Brian West wrote: Or as I have argued today they should fix their SRV records to be zero weighted. /b On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote: You can bypass the srv records if you like by passing a :port with the hostname where you use it in freeswitch. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question about presence
Open on endpoint modules. We will relocate if needed. jmesquita On Thu, Aug 27, 2009 at 8:06 AM, Dennis oderm...@googlemail.com wrote: sorry, but i do not know i which category i have to set this problem. could you help me with that? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Couple of questions
Hey there, FsGui uses ESL a lot and I had to go through the code to document
it so here is a few hints inline ...
Don't hesitate to keep the questions coming. I will fill in whenever I can.
jmesquita
On Sun, Aug 23, 2009 at 8:50 PM, Brian West br...@freeswitch.org wrote:
On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:
Hi,
I don't see how I can read some responses to command using esl.
I.E. esl_send_recv(handle, api show calls count\n\n);
and
printf(Header Test %s\n, esl_event_get_header(event, API-
Command));
printf(Body Test %s\n, esl_event_get_body(event));
the header details are returned.
The body is null.
Body is null on every event that does not use headers to output information.
A good example would be console logs. I haven't seen too many default events
besides log that have body besides a application custom events.
I'm not too sure about using ESL in C, I have used it pretty much
exclusively in perl.
Also, I can originate a call and set the account code for it, but
how do
I get a list of calls with their account codes?
originate {account_code=1234}sofia/profile/tar...@ip
You can get the list of the channels via show channels or bridged
calls with show calls
From there you have the UUID's you can call uuid_dump on them to get
all the variables.
Do I get a list of calls then go through them one by one and get the
variables for those calls by uuid?
All ESL does is output events to socket and expose the API commands. It does
not maintain any kind of list of calls or anything like that so it is up to
you to maintain that yourself if you don't want to parse API output every
time.
You could do this or setup a listener to get the events as they happen
and keep the info you need.
Does anyone have any documentation for the esl api?
http://docs.freeswitch.org/ (this should help, its under files list
see esl.h)
I need to work a little bit more on that documentation as well.. I saw a few
conflicts with the core documentation too. Will get there once I have some
more time left.
Even if I could read some comments from a usage of it would be useful.
I just find it interesting you're doing this with C.
--
Cheers,
Matt Riddell
Director
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Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH
Way to go moy! On Fri, Aug 21, 2009 at 7:59 PM, Michael Collins m...@freeswitch.org wrote: On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva moises.si...@gmail.comwrote: So, I finally took some days to put up OpenR2 working with OpenZAP, which means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has support for. Including Mexico, Brazil, Argentina and others. The stack has been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most countries that users may be interested in, support for new variants will be added on-demand only (in any case users can always tweak the advanced configuration file to create their own variants as a last resort). I created a wiki page to illustrate the basic setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2Now is time for testing. I just did minimal testing on my development environment, no serious testing, and I know that some stuff is not working at this point (I had some issues with variable length DNIS and ANI) which should be fixed soon. If anyone around happens to have an R2 link and wants to test R2 support in OpenZAP, I can give them a hand with the configuration and any issues you may find. You can find me on IRC at #freeswitch, #freeswitch-dev and #openzap as moy. You rock, dude! -MC -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod Touch Users
No one said a thing but I really feel like initiatives like this should be cheered. Thank you for this app and thank you for making it free as well. On 8/14/09, Chris Danielson ch...@maxpowersoft.com wrote: Announcing the release of FreeSWITCH Console in the Apple Application Store. The application is FREE and allows you to connect to a FreeSWITCH event socket layer module that is bound to an external interface. Great for development purposes and general remote debugging. Blog announcement: http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/ iTunes Store Link: http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221mt=8 http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221mt=8 Kind Regards, Chris Danielson ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ClueCon2009 Torrents
I am interested and would also seed to the community On 8/15/09, Gabriel Gunderson g...@gundy.org wrote: On Sat, Aug 15, 2009 at 12:13 PM, Pederpe...@networkoblivion.com wrote: If you want the torrents, email me off list. Why off list? Isn't the point of torrents to have more people sharing in the load? Gabe ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question about an ESL function
Recv will lock the calling thread until it gets an event or gets disconnected while recvtimed will return controll when event received or timer expires. Whatever comes fisrt. On 8/15/09, Jean-Marc Hyppolite hyppolit...@yahoo.com wrote: Hello, I would like to know the purpose of the ESL function named recvEventTimed. Thank you for your help. __ Be smarter than spam. See how smart SpamGuard is at giving junk email the boot with the All-new Yahoo! Mail. Click on Options in Mail and switch to New Mail today or register for free at http://mail.yahoo.ca -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ClueCon2009 Torrents
I am already seeding from here. jmesquita On Sat, Aug 15, 2009 at 7:34 PM, Peder pe...@networkoblivion.com wrote: I don’t have access to do that or I would. That’s why I offered to email them to whoever wants them.I did send them to Brian earlier, but he must have some sort of life outside of FreeSWITCH because he hasn’t put them there yet. ;-) *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Diego Viola *Sent:* Saturday, August 15, 2009 4:47 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] ClueCon2009 Torrents Upload the torrent files in http://files.freeswitch.org ;) On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks jaybi...@gmail.com wrote: I'd also seed such a torrent. Please send the link :) On 16/08/2009, at 6:34, João Mesquita jmesqu...@gmail.com wrote: I am interested and would also seed to the community On 8/15/09, Gabriel Gunderson g...@gundy.org wrote: On Sat, Aug 15, 2009 at 12:13 PM, Pederpe...@networkoblivion.com wrote: If you want the torrents, email me off list. Why off list? Isn't the point of torrents to have more people sharing in the load? Gabe ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org -- Sent from my mobile device ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] VoiceMail transcription
I am sorry for the ignorance on the matter, but how does google voice does? Do they also have humans? jmesquita On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman kirk.bate...@gmail.comwrote: I'm still interested in getting pocketsphinx to attempt speech recognition on an audio file. To be honest, most of the problem is that at 8Khz (mobile phone call rate), speech detection is NOT very accurate, at 16Khz it IS significantly better. I'm planning to have a play with the speechtools module and mod_pocketsphinx etc to try and get an audio file parsed, spare time permitting. Will let the list know if I get anywhere. Regards Kirk Bateman 2009/8/11 David Knell d...@3c.co.uk Hi Pete, I'm afraid that the answer's still the same: use a human. Here's an article describing the state of the art: http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/ - the links to previous stories at the bottom provide good background. --Dave I apologize, I should have been more clear. We will be using humans to scan the translated results. But we are looking for a system to perform the first pass on the audio to hopefully help the human type less. Although the question has been raised if it's faster to have a human just transcribe the whole thing, or fix up what the computer spit out. If you have any insights on this, that would be great. -pete Original Message Subject: Re: [Freeswitch-users] VoiceMail transcription From: David Knell d...@3c.co.uk Date: Mon, August 10, 2009 11:51 am To: freeswitch-users@lists.freeswitch.org Good evening Pete, The only way to do this is, I'm afraid, to use a human. We use Amazon's Mechanical Turk to good effect. Cheers -- Dave Good morning all, I realize this is slightly off the FS topic, but I am wondering if anyone out there has experience with software packages designed for the transcription of voicemails to text. I've used pocketsphinx with FS to handle IVR menus, but now have the task of figuring out how to convert recorded phone conversations (voicemails mostly) to text. This does not have to be a real-time process, I can store the audio files and process them over time. This would need to be a software (preferable open source) solution. ASPs like VoiceCloud would not work for this application. Thanks for any help -pete ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Spanish Prompts
Mike, the gender thing will eventually have to change code, I guess. I have not yet looked at the say code, so I am just imagining here. On Tue, Aug 11, 2009 at 4:45 PM, Michael Jerris m...@jerris.com wrote: again, this issue should be addressed when you do a sound set for that dialect, we are attempting to keep the c code common for all dialects within a language, we will see if this works unless anyone can point to a place this will not work. Mike On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote: samuel wrote: I'm also for different spanish localization if it's not too complicated. It was also for me the first time I see signo de número for pound ;) I just idly noticed this - so just a comment from a Brit who iteracts quite a bit with Americans. The # symbol in the UK is not called pound because (I don't know if this will come out on your screens correctly) we use £ for our currency. I would refer to the # symbol as hash or just possibly the number symbol. -- Alan Chandler http://www.chandlerfamily.org.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Spanish Prompts
Oops, I thought you were saying different languages. Sorry about that. jmesquita On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins m...@freeswitch.org wrote: 2009/8/11 João Mesquita jmesqu...@gmail.com Mike, the gender thing will eventually have to change code, I guess. I have not yet looked at the say code, so I am just imagining here. Are there gender differences between dialects of the same language? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Spanish Prompts
Hey! I have recorded a couple of samples and I will patch whatever is needed to support portuguese(Brazil) and spanish on say. Don't worry, I am on top of it. jmesquita On Mon, Aug 10, 2009 at 8:04 PM, Michael Collins m...@freeswitch.org wrote: Be sure to hop on IRC and speak with jmesquita because he's been working on this also. It would be very good to have Spanish-speaking users review the Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking programmers to assist with the mod_say application for Spanish - there are things that you have to do in Spanish that you don't have to do in English. Thanks, MC On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea lfur...@gmail.com wrote: On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for edition . Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for FS playback. Here's a guide that has been put together for reference on what to record. http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml Regards, On Fri, Aug 7, 2009 at 9:21 AM, bakko asannu...@gmail.com wrote: I'd like to begin record spanish prompts for FS. Do you know any software/hardware to make it? Thank you BR ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Softphone control
Stay tuned on fsgui. It will get there really soon. jmesquita On Fri, Aug 7, 2009 at 3:50 PM, Raffaele P. Guidi raffaele.p.gu...@gmail.com wrote: Maybe Artem is interested in CTI (computer telephony integration) - click2dial, opening a url (or statrting a program) on incoming call...? On Fri, Aug 7, 2009 at 17:00, Kevin Green ke...@johnnyvoip.com wrote: From what I am aware you can't use FreeSWITCH to control a softphone directly though you can make it do things that will have a similar end result. You could set eyeBeam to auto-answer calls if you want them to answer right away or orginiate a call that is auto-answered but not bridge the call until a user on the eyeBeam presses a digit or a socket control tells it to connect the two ends. You can also use FreeSWITCH to place the line on hold using event sockets, this will place it on hold in the server and not directly like placing it on hold in eyeBeam (i.e. the hold button in eyeBeam likely wont show it as being on hold). Beyond that if you want to directly control the clients you would need to look at getting an API access into the eyeBeam client. I hope this will help. Regards, Kevin Green On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev ryde...@googlemail.comwrote: No, I don't want to make softphone from FreeSwitch I have FS and several users with eyeBeam softphones. I need to control those eyeBeams You can run FreeSWITCH as a softphone and control it. http://wiki.freeswitch.org/wiki/Freeswitch_softphone 2009/8/7 Artem Vasiliev ryder86 at googlemail.com Hi I have FreeSwitch and external application, which communicates to it via event socket - listens for events for certain number and gives some commands. Is it possible for this application to control client softphones, for example, make them answer or hold, using the event socket or other FreeSwitch capabilities? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change the contact when fs sending REGISTER??
Add the following line to the gw definition: param name=extension-in-contact value=true/ jmesquita On Wed, Aug 5, 2009 at 7:26 AM, Brad Tuan brad.t...@gmail.com wrote: As title ,I know how to do when sending INVITE but how to do it when fs sending REGISTER?? For example , when gateway registering , the contact is gw+a...@xxx.xxx.xxx.xxx , how to change it to *a...@xxx.xxx.xxx.xxx??* ** *Please help* ** ** ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question about using switch_caller_extension_add_application
My guess is that you will receive a message here: switch_status_t channel_receive_message(switch_core_session_t *session, switch_core_session_message_t *msg) The problem here is that you don't have the exact SIP code but there is a clear relationship between the codes and the messages you receive on the channel, so I am guessing that is all the same. Hope this helps. jmesquita On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson woodydick...@gmail.comwrote: Hi, I want to implement a module where freeSWITCH would try to bridge to an extension and if the bridging operation fails, my module can use the hangup code to determine the next cause of action. With switch_caller_extension_add_application(session, extension, bridge, sofia/gateway/mygw/1232323);, if there is an error ( 503 received for instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the module's APP) and go on to the next action. Is there anyway to control it so that freeSWITCH would remain to be within the module's APP funtion and continue executing the code after switch_call_extension_add_application, when let's say a 4XX or 5XX or CANCEL ( from originator) is received? I have tried it and found that if the bridging is successful, freeSWITCH would continue executing the code after switch_caller_extension_add_application, but if an error is received, then it would just move on to the next action. Does anyone know how to deal with this problem? Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Monitoring On-Hold/Off-Hold
I only see one way out of this. If you manage presence, an event like the following is sent: Event-Name: PRESENCE_IN Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f FreeSWITCH-Hostname: cl-t146-421cl FreeSWITCH-IPv4: XX FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-08-05%2013%3A42%3A24 Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT Event-Date-Timestamp: 1249494144628132 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_presence Event-Calling-Line-Number: 472 Channel-State: CS_HIBERNATE Channel-State-Number: 8 Channel-Name: X Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered Caller-Username: 1000 Caller-Dialplan: XML Caller-Caller-ID-Name: Mesquita Caller-Caller-ID-Number: 1000 Caller-Network-Addr: X Caller-Destination-Number: 1005 Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f Caller-Source: mod_sofia Caller-Context: X Caller-Channel-Name: X Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1249494132128119 Caller-Channel-Created-Time: 1249494132128119 Caller-Channel-Answered-Time: 1249494139500129 Caller-Channel-Progress-Time: 1249494132368119 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Other-Leg-Username: 1000 Other-Leg-Dialplan: XML Other-Leg-Caller-ID-Name: Joao%20Mesquita Other-Leg-Caller-ID-Number: 1000 Other-Leg-Network-Addr: 190.2.41.65 Other-Leg-Destination-Number: sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559 Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f Other-Leg-Source: mod_sofia Other-Leg-Context: X Other-Leg-Channel-Name: XX Other-Leg-Screen-Bit: true Other-Leg-Privacy-Hide-Name: false Other-Leg-Privacy-Hide-Number: false proto: src/switch_channel.c login: src/switch_channel.c from: XX rpid: unknown status: hold event_type: presence alt_event_type: dialog event_count: 3 Content-Length: 543 Content-Type: text/event-plain Other than that, I think it can be patched. I will take a look at it. Guys, should this be patched on the state machine itself or on the mod_sofia channel_receive_message? jmesquita On Wed, Aug 5, 2009 at 1:35 PM, mayamatakeshi mayamatake...@gmail.comwrote: Hello, I'm using mod_event_socket to monitor FS. I'm using events plain ALL' and I get lots of channel events. But curiously, when some channel puts the call on-hold/off-hold, I don't get any notification. Is it possible to get these events? Am I missing some setting? regards, takeshi ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story
If this is good for me to hear, I would imagine to the core team. Despite of this not being a group support meeting, I have to say that: Thank you for sharing, Seven. jmesquita On Tue, Aug 4, 2009 at 4:17 AM, Seven Du dujinf...@gmail.com wrote: Hello All - In the spirit of ClueCon (which we are missing this year, but hopefully not next), we wanted to document our FreeSWITCH Story. We've posted it to the wiki( http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and it is copied below. Thank you all and enjoy a good conference! Seven Du (seven) Jonathan Palley (jpalley_idapted) Idapted Ltd. *How FreeSWITCH has created hundreds of job opportunities and changed lives. * We want to share our experience working with FreeSWITCH. FreeSWITCH has been a key enabler of our business. We hope this story can be a small way to say a very big THANK YOU ALL. Changing lives is an over-used cliche, but in this case, FreeSWITCH has really allowed us to do just that. What We Do: We are not a telephony business; we are an educational technology and service business. In Asia (China, in our case) students must pass English examinations to study or work abroad and gain new experiences. However, there is limited access to native English speakers and the access students can gain is typically very expensive. At the same time, in the U.S., there are many professionals looking for work-at-home opportunities - people who need jobs and would create great teachers. Through our technology and content we empower these people to be effective English teachers. Does it work? Yes. The majority of our students are getting test scores that many failed for years to get. Just hours ago one student called one of our sales agents crying with joy. And for our teachers, they are now working in an industry that was previously unavailable to those living in the U.S. http://www.idapted.com Why FreeSWITCH Enables This: FreeSWITCH has been a key enabler of our business. Recording calls, controlling routing, integrating with various web-based interfaces, enabling multiple endpoints - these are all key features of what we must do. Most importantly, setting up various servers and routes to mitigate cross-Pacific and country-specific network challenges is key. Doing what we are doing with commercial solutions would have made the business unworkable. Our Experiences with FreeSWITCH: We started using FreeSWITCH as our VoIP Platform in April 2008, after receiving unsatisfactory results with other open source solutions. It took one day of reading through the FreeSWITCH source code to know, this is it. This is the VoIP platform we build our business on. It took a few days of working with the extremely competent and focused community to re-affirm this commitment. Our Setup: Our teachers use a custom software that integrates a VoIP client with our web based platform. Students connect to our teachers on-demand. Simply put, on a web-based comet interface the student enters a phone number (or a skype name or a gtalk account) and our platform bridges the best available trainer and the student. At the same time a web-based interface is being updated. The challenge for us is the connection between teachers and students over a cross-continent network. For example, we experienced problems earlier this year when a Asis-Pacific communication fiber broken... So, we've learned to setup multi servers in multiple datacenters for redundancy. We run multi instances of FreeSWITCH so we can always use the cutting edge and mitigate the effects of bugs. A main, stable FreeSWITCH(FS) instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to create different conf dirs (/usr/local/freeswitch, /usr/local/skype, /usr/local/gtalk etc). This allows us to run the same code base over different configurations, and call skype and gtalk accounts just like a normal PSTN gateway (sofia/gateway/pstn/ or sofia/gateway/skype/ or sofia/gateway/gtalk/ ). More important, if one FS (say FS-skype) behaves abnormally or crashes, we can easily change to another FS-skype server (we run other servers located in various places in China and HK for redundancy). FS --| |---PSTN gateways |--- FS-skype |--- FS-gtalk |--- FS-skype2 |--- more ... COMMUNITY: The community's commitment cannot be undervalued. The insightful, modular design of FreeSWITCH allows anyone to contribute, whereever their skills lie. It also allows us to easily make modifications to the underlying code to suit our specific use-cases We want to highlight a few key people and modules in the FS ecosystem: mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers clients. PSTN is zero-conf for the user and mitigates troubles with the end users
Re: [Freeswitch-users] FsGUI
Thank you very much for your support. Brian, how can I put MacOSX dmg and linux binaries on files.freeswitch.org? jmesquita On Sun, Jul 19, 2009 at 5:23 PM, Brian West br...@freeswitch.org wrote: Its sycned to files. now. /b On Jul 19, 2009, at 2:46 PM, Carlos Talbot wrote: FYI, for those interested I've built an FsGui MSI file compiled for Windows via VS 2008 the QT SDK library. It includes 2 necessary QT dlls. Future builds of the MSI for Freeswitch will include this. Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi (until I can sync it up to files.freeswitch.org) Carlos 2009/7/18 João Mesquita jmesqu...@gmail.com Added to the wiki: http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu jmesquita On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX prometheus...@gmx.netwrote: Thanks, I have found the sources in contrib/jmesquita/fsgui Any recommendatioins how to compile it under Linux? Best regards Peter João Mesquita schrieb: Dear folks, Even tho it might be premature, I would like to already spread the word. Check out FsGUI and feel free give feedback if this is a wanted tool and what direction it should take. Beware that the code is still contrib code and might now be yet mature for production use. http://wiki.freeswitch.org/wiki/Fsgui Thanks, João Mesquita ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FsGUI
Added to the wiki: http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu jmesquita On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX prometheus...@gmx.net wrote: Thanks, I have found the sources in contrib/jmesquita/fsgui Any recommendatioins how to compile it under Linux? Best regards Peter João Mesquita schrieb: Dear folks, Even tho it might be premature, I would like to already spread the word. Check out FsGUI and feel free give feedback if this is a wanted tool and what direction it should take. Beware that the code is still contrib code and might now be yet mature for production use. http://wiki.freeswitch.org/wiki/Fsgui Thanks, João Mesquita ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FsGUI
Dear folks, Even tho it might be premature, I would like to already spread the word. Check out FsGUI and feel free give feedback if this is a wanted tool and what direction it should take. Beware that the code is still contrib code and might now be yet mature for production use. http://wiki.freeswitch.org/wiki/Fsgui Thanks, João Mesquita ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Language Handling: call for assistance
Guys, I don't know if I really get the problem here. I mean, I do get that the 2+2 model does not work not even for where I live. I really hate the fact that all spanish south american dialects (some within the same country) are put in the same bag as it wouldn't matter to ppl so I am with you Steve on this one to find an alternative to the 2+2 model. So, in summary, what I am asking is: What would be the problem with mod_say_es_ar_ba for Porteño dialect spoken in Buenos Aires, Argentina besides the verbosity of it and the limited amount of levels we have? Do we know any country that has a sub-dialect from a dialect? jmesquita PS: Please, forgive me if I totally misunderstood it. Afterall, I do have I high fever. On Thu, Jul 2, 2009 at 8:58 PM, Michael Jerris m...@jerris.com wrote: On Jul 2, 2009, at 7:50 PM, Steve Underwood wrote: If by the usual way you mean the standard 2 + 2 letter codes we are used to on computers, that just doesn't work. As I said before, those are for written languages, not spoken languages. There are no standard codes for many spoken languages. For example, the standard codes for Chinese are zh_cn for mainland China, zh_tw for Taiwan, zh_hk for Hong Kong. However, in GuangDong you will probably want to offer Cantonese as well as Mandarin voice prompts, so you will want a zh_gd, or something, which you won't find among the standard 2 + 2 letter codes. That's why the SSML people had a hard time coming up with a language scheme, and SSML 1.0 didn't even reference one. The more you look around the world, the most complex the issue of language variants becomes. If you don't face that at the beginning it just gets messier later on. Steve Do we know that the language model at least always pairs with the first 2 letter code? So zh_* we can use mod_say_zh for? or do we need to address different language rules for different dialects as well? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CTI
I am interested to know more about this. Are you using ESL to then translate CSTA calls to FS? Wouldn't this be great to be added as an FS module as an alternative to ESL? It would enable lots of existing CTI applications to work with FS. jmesquita On Mon, Jun 29, 2009 at 12:20 PM, Brian West br...@freeswitch.org wrote: Nice are you the project leader? /b On Jun 28, 2009, at 8:36 PM, szentesik wrote: Currently working on some CSTA support (http://cstainside.sourceforge.net/ ). The MakeCall, DeliveredEvent, ClearConnection, TransferCall/SingleStepTransfer things required for the features above are on the list, the AnswerCall implementation is open (I'm not sure whether the FreeSWITCH is able to answer calls for any of the SIP clients available). ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CTI
I would strongly suggest that. At least for the mod itself. That way, we can all contribute with it and keep it always compatible with the lib. jmesquita On Mon, Jun 29, 2009 at 5:38 PM, Brian West br...@freeswitch.org wrote: Are you interested in hosting any of it in our tree? /b On Jun 29, 2009, at 3:34 PM, szentesik wrote: Yes. It will also use to bring FS with a CTI application I'm a lead developer of closer, decided to make the integration open source/open standard based. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Confused with event content lengths
If I am not mistaken, you are always safe reading the amount data expressed on Content-Length since it is calculated based on the total message length before it is sent out of FS. From a protocol point of view, it would indeed be much better to rely on something such as Content-Length then \n\n termination string. As I get to know more and more the core developers, I doubt they would rely on the latter. Hope it helps... jmesquita On Sun, Jun 28, 2009 at 3:49 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, I’m trying to parse events in C++ for an outbound socket. The docs are a little contradictory, so I wonder if someone could help me out. As I understand it and event is terminated with double LF’s (\n\n) However if there is a Content-Length header the wiki very confusingly says ‘Content-Length is the length of the event beginning *AFTER* the very next LF only line (\n) and *inclusive* the trailing LF/LF pair (\n\n)’ BUT the example says it’s after the \n\n in the header!! Which is it? In addition, it also looks like the event body is also terminated by a \n\n. If this is the case, why do I care about content length value, can’t I simply read until I get the termination sequence? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Myevents in outbound socket
You should definitely look at ESL, dude. Take a look at
${SVNROOT}/libs/esl/. There is a esl_oop inside that might give you a go.
Beware that this is only an interface for SWIG, but might be useful to you
if you extend it.
Later,
jmesquita
On Sun, Jun 28, 2009 at 8:14 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Nope.
Can’t find much on the Wiki on how to interface with ESL using C++. I want
to control the outbound socket from a windows 2003 server only because
that’s what I’m familiar with. Is there some portable C++ or C code?
--
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Brian West
*Sent:* 29 June 2009 00:00
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Myevents in outbound socket
Are you using ESL?
/b
On Jun 28, 2009, at 5:55 PM, Nik Middleton wrote:
Hi Guys,
I’ve almost got my c++ outbound socket control prog running, however even
though the filter works, it would be truly great to just subscribe to
myevents as even with the filter in place I get lots of channel Execute and
complete events which I don’t really need. Problem is, is that mod_VMD
isn’t included in those events, even though it is channel specific. Is
there any chance that this will be included? If not, can someone point me
to where myevents is defined and I’ll have a go at it myself.
Regards,
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Re: [Freeswitch-users] How to originate gtalk calls
try load mod_dingaling. If that does not work, get to the source dir, edit modules.conf, uncomment mod_dingaling, make make install Dont forget to load the mod once FS is up again.. jmesquita On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang jingwei.y...@gmail.comwrote: Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch *originate dingaling/gmail.com/user...@gmail.com echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* Please kindly let me know what the correct originate string is. Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sofia external profile: external IP problem
What I would guess is the the STUN lookup failed. Do you have anything on
this box that would prevent FS from doing DNS lookup?
jmesquita
On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon g...@i.ph wrote:
the default setting is auto-nat.
i changed ext-sip-ip=$${external_sip_ip} and
ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun:
stun.freeswitch.org. result: same problem
i tried your suggestion. still the same problem.
On Sun, Jun 21, 2009 at 1:45 PM, Jason White ja...@jasonjgw.net wrote:
Nandy Dagondon g...@i.ph wrote:
hi,
i tested the latest SVN build (13884) using the sample configuration
files
... no modifications whatsoever. but in sofia external profile, the IP
address is my internal address instead of my external IP address.
did i miss something here?
Try setting ext-sip-ip and ext-rtp-ip in the external profile to
stun:stun.freeswitch.org
This can alternatively be set using global variables in vars.xml in the
supplied configuration.
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Re: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ?
Right now, I am working on a board that will soon support all those features but it isn't compatible to FreeSWITCH just yet. Other then that, there was thread here before discussing PorTech GSM gateways. They might be able to help. If you are interested in using other platform with the Khomp boards, I can provide you a contact. Just get in touch with me offlist. Thanks, jmesquita On Thu, Jun 18, 2009 at 7:41 PM, Ing. Edwin Villarreal evi...@chipoly.comwrote: Hello, I am planning to build a plataform to sell content, pictures, tones, MMS, etc. Do you know wich GSM 3G boards should work? Anyone has done this? *Greetings!* *Edwin* ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Which GSM gateway to buy?
Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP tricks 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV enno.egb...@web.de wrote: Hi, look at www.kuhnt.com. It´s a german page. There you can find Kontakt where you can ask for special requirements. NOx Diego Viola wrote: Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Which GSM gateway to buy?
Pricewise, is it worth it? jmesquita On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr jan.k...@gmail.com wrote: We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html since you can use SMTP/POP3 to manage SMS. Jan 2009/6/17 João Mesquita jmesqu...@gmail.com Guys, I was looking at the advantages and disadvantages of having a GSM gateway vs. a GSM board. The conclusions I get are: Board pros 1. Boards are able to get/send SMS without SIP tricks 2. You don't have to make a SIP call to check if channel is available and don't rely o SIP messages to get channel status 3. FS will be able to check for signal level on the board and fire events on pre-defined thresholds. Gateway pros 1. I think of is the a GW can be used by more then one server, therefore, can have failover. 2. A GW is more scalable It would be nice if you, that have already used GSM GWs in production, could comment on this. Thanks, jmesquita On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV enno.egb...@web.de wrote: Hi, look at www.kuhnt.com. It´s a german page. There you can find Kontakt where you can ask for special requirements. NOx Diego Viola wrote: Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Which GSM gateway to buy?
Get Khomp GSM cars! Ihihihih They will soon be compatible with FreeSWITCH. Laterz, jmesquita On Tue, Jun 16, 2009 at 6:48 PM, p...@privateconnect.com wrote: I did a fair amount of research into GSM gateways about 8 months ago. I should first ask what are you looking to do with the gateway? -pete Original Message Subject: [Freeswitch-users] Which GSM gateway to buy? From: Diego Viola diego.vi...@gmail.com Date: Tue, June 16, 2009 2:39 pm To: freeswitch-users@lists.freeswitch.org Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MadBoss Conferences Examples - bug?
Look at the newly implemented wait-mod conference flag on mod_conference. This is: http://wiki.freeswitch.org/wiki/Mod_conference#.3Cprofiles.3E under parameters-conference-flags jmesquita On Tue, Jun 16, 2009 at 10:22 PM, Ing. Edwin Villarreal evi...@chipoly.comwrote: Hello friends. I’ve been playing with the mad boss examples. There is an issue I’d like to see: For example in MadBoss3: The first leg added to conference is the loopback/… Then you can add more users by conference_set_auto_outcall function. The problem I see is that: 1) Loopback music is still in the background of conference. 2) When everyone hang up, the conference is still active, because the user (music) is still inside the room. How can music be stoped once meeting is going to start? *Edwin* ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Unregister extension?
Lars, don't get me wrong but you have been asking questions that are all answered on the wiki: http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoints Might be a good idea to value the work of lots of ppl who have been documenting by actually using the documentation, no? Sorry if that sounds a bit harsh. jmesquita On Fri, Jun 12, 2009 at 10:33 AM, Lars Zeb larc...@yahoo.com wrote: How can I unregister a softphone’s registration? I have a SNOM 320 for which I had defined two extensions, 1010 and 1001. I changed the second one to 1000. Now when I do ‘sofia status profile internal’ all three show up. How do I get rid of the 1001 extension? I shutdown and restarted FS but that didn’t do it. I have defined 1001 on a Polycom phone. I hope that the SNOM 1001 is blocking the Polycom at that same extension and that is the reason the Polycom is not showing. Thanks, Lars Registrations: = Call-ID:3c267015ab6b-bd6gioq5ytor User: 1...@192.168.10.29 Contact:1010 sip:1...@192.168.10.104:2048;line=dg4k4xql Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:24) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1010 Auth-Realm: 192.168.10.29 Call-ID:3c267015afa6-6v0sw4o3qei3 User: 1...@192.168.10.29 Contact:1001 sip:1...@192.168.10.104:2048;line=co52ym3a Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:02:25) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1001 Auth-Realm: 192.168.10.29 Call-ID:OGUxMzExZmNjMTQxOTk4ZDM1ZmJkODNhOTFhZDMwM2Y. User: 1...@192.168.10.29 Contact:1019 sip:1...@192.168.10.11:19040 ;rinstance=5394acb4dfa00c0a Agent: Bria Professional release 2.4.3 stamp 50906 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:10:28) Host: fs IP: 192.168.10.11 Port: 19040 Auth-User: 1019 Auth-Realm: 192.168.10.29 Call-ID:3c270d667ff5-47fq2p6n1ou1 User: 1...@192.168.10.29 Contact:1000 sip:1...@192.168.10.104:2048;line=gzdwwjqr Agent: snom320/7.3.14 Status: Registered(UDP)(unknown) EXP(2009-06-12 08:16:35) Host: fs IP: 192.168.10.104 Port: 2048 Auth-User: 1000 Auth-Realm: 192.168.10.29 = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Status Event
Nik, I am a noobie and all, but most API responses can come as xml just by adding as xml at the end of the call. jmesquita On Thu, Jun 11, 2009 at 6:57 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Not sure where enhancement requests should be posted, but here it is anyway I would dearly love to be able to send a status event that returns an event style output that provides machine readable output rather than the wordy human readable response. (I hate parsing) Is there such an event already? Regards ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Rotating log files not working
Just for the record, always update do latest trunk when testing and provide revision number (version command). Later, jmesquita On Sun, May 31, 2009 at 2:35 PM, Lars Zeb larc...@yahoo.com wrote: I am trying to rotate the logs, specifically the cdr ones. But the existing extension and Master csv files are not rotated; they remain untouched. I issue the command ‘kill –s HUP pid’ (pid of freeswitch). The fs console says 2009-05-31 10:25:58 [NOTICE] mod_logfile_c:157 mod_logfile_rotate() New log started. The conf/autoload_configs/cdr-csv.conf.xml shows: configuration name=cdr_csv.conf description=CDR CSV Format settings param name=default-template value=sql/ param name=rotate-on-hup value=true/ param name=legs value=a/ /settings … What am I doing wrong here? Thanks, Lars ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Conference users hear MOH until leader enters?
I could not get this working on current trunk. Can you post your configuration on conference module and the dialplan example? Thanks, jmesquita On Thu, May 28, 2009 at 12:56 PM, Michael Collins m...@freeswitch.orgwrote: On Wed, May 27, 2009 at 8:00 PM, j3flight jcro...@gmail.com wrote: Wiki Tax paid... That was my first contribution to the freeswitch wiki! MC, you're welcome to have a look over it and see if i made things clear enough. Feel free to edit. Nicely done! Thanks for taking the time to create a wiki user and jump in with both feet. BTW, as you gain more practical experience with this wait-mod/moderator feature please feel free to come back and add any useful tidbits to the wiki. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS PABX experiences?
I use it on a 12 extension office. Works like a charm. Specially because I host it on a cheap dedicated server (iWEB). The only thing I would say is to be careful not to loose focus on your primary business and start developing your own GUI for the pbx. I have seen that happen with lots of companies. They eventually fail. Mesquita On Wed, May 27, 2009 at 8:53 PM, Nandy Dagondon g...@i.ph wrote: IMHO, you have tons of features w/ FS. i've setup FS on a low-power consumption Intel D945GCLF2 motherboard (Atom dual-core CPU) ideal for 24/7 operation on a 10-seat contact center w/ default conversation recording. no problem. another cool feature. you can route the call based on the Caller ID. so u hv to consider the selection of the telco (FXO) gateway. one advantage over key system - you can turn PCs into extension phones using free softphones. just use USB phones instead of headsets. re maintenance, just provide remote access to the FS box. in my home FS, i create dialplan to reboot or shutdown my FS. it helps when problems occur (not encountered so far). -nandy === LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Thu, May 28, 2009 at 6:51 AM, Neale Banks ne...@lowendale.com.auwrote: Hi, We're considering deploying FS instead of a traditional PABX/Key-System in a small office environment (i.e. primarily non-technical users, 15-20 handsets). Anyone have any experiences (good/bad/whatever) in this sort of scenario? Thanks, Neale. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Conference users hear MOH until leader enters?
Quoting Mr. Anthony Minessale:
thThe easiest way would be the new feature I added to 13442
in the conference profile add
param name=conference-flags value=wait-mod/
to your profile
and in your dialplan
action application=set data=conference_member_flags=*moderator*/
action application=conference data=1...@wideband/
or
action application=conference data=1...@wideband+flags{*moderator*}/
Don't forget the wishlist and donate button on the main site
On Tue, May 26, 2009 at 10:20 AM, j3flight jcro...@gmail.com wrote:
I'm attempting to replicate the behavior of an Asterisk system with
FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to
be supported (easily).
Ok, so I've setup my dialplan so that when a specific extension is hit, it
calls out to some javascript which acts like an IVR to handle the
conference
setup. (Similar to this:
http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own
improvements.) Anyway, the conferences are stored permanently in a
database, but I want them protected by their owner so they can only be
used when that conference owner dials. If other users have entered the
conference prior to the owner, they should hear music-on-hold until the
leader enters.
This is easy in Asterisk because you can pop someone into MeetMe with
different flags. So, in my IVR, I prompt for the conference number
(known
to all) and then the password (known only to the owner/leader). If the
proper password is entered, the user is sent to conference XYZ with the
leader flag set. If no password is entered, the user goes to conference
XYZ, without the leader flag. If anyone enters before the leader, they're
told by MeetMe that the conference will begin when the leader arrives and
MeetMe provides MOH until that time.
Help! This is an absolute deal-breaker for my install... How can I do
this
in FreeSwitch?
Thanks...
--
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Re: [Freeswitch-users] Compact, fanless appliance?
Guys, I was trying to do the same to the Pika Warp. Too bad their support and development enviroment sucks. The best I could do (without too much effort) was run FS without the analog support. www.pika.com jmesquita On Apr 25, 2009, at 11:30 AM, Carlos Talbot wrote: Dave, it's not hard. I suppose I should add this to the wiki: http://forum.openwrt.org/viewtopic.php?pid=83701#p83701 Carlos On Thu, Apr 23, 2009 at 3:02 PM, David Knell d...@3c.co.uk wrote: You might want to take a look at this: http://www.amazon.com/IEEE802-11N-Wireless-Broadband-MZK-W04NU-Designed/dp/B000YDS0YG - twice as much everything as the NSLU2, and is supposed to run OpenWRT just fine. I've one sat in front of me right now, although I've not yet plugged it in - have to work out how to take it apart first ;-) --Dave BTW, at 85€, the Linksys NSLU2 looks like a bargain: http://en.wikipedia.org/wiki/NSLU2 Has someone successfully ran Freeswitch on this to handle a couple of simultaneous SIP conversations? What about the more expensive but very tiny Gumstix? http://en.wikipedia.org/wiki/Gumstix ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asterisk Fast AGI like support from FreeSwitch. Also mod_python + Django doc
On Apr 21, 2009, at 4:13 PM, technologyinspired wrote: Hello users, I am new to FreeSwitch. I have used Asterisk a lot but now want to use FreeSwitch. Presently I write complex Asterisk applications using Fast AGI in Python. How could I achieve the same from FreeSwitch? ESL is the answer to your question. Take a look at the wiki. There is a Python binding for it as well. The main reason is that the server has heavy call load and I want to shift that load on two systems one the FreeSwitch and another the application server. Ok, I don't see a problem there ... Could you also give a pointer to such an example or documentation where it says how to run FreeSwitch applications in client server mode (Fast AGI mode in Asterisk). Is there any good documentation on mod_python + Django support in FreeSwitch? Carefull with Django + FS. You could run over the problem that, since FS is multi domain, you might want to have a SaaS sometime and Django does not connect to multiple DBs with its native ORM. What kind of pointers are you looking for? Thanks, Regards, Vin ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Jmesquita ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Ideas for my presentation
Diego, which country? JMesquita On Apr 18, 2009, at 10:51 PM, Jason White wrote: Diego Viola diego.vi...@gmail.com wrote: Let me know if you have some nice ideas for my presentation, I already got some by myself, but more are always welcome :). You could demonstrate the flexibility of the dial plans, in particular the use of regular expressions and the dial plan syntax to achieve desired results. There are plenty of good examples in the default dial plan, too. The use of wide-band codecs also sets FreeSWITCH apart from alternatives: if you have a USB head set, and a reasonably good Internet connection, you can have fun with Celt-encoded phone calls and conferences with your FreeSWITCH-using friends! I like the dual-stack support for IPv4 and IPv6, whereby NAT goes away when connecting to systems that have IPv6 access. This would be worth mentioning. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Adding Spanish support to say
I know spanish and I would translate it no problem. MC, get in touch with me off-list so we can handle that. I can also translate to portuguese-brazil. jmesquita On Apr 14, 2009, at 2:37 PM, Michael Collins wrote: KK, Do you have someone who knows Spanish and who can translate? If not I will whip up some volunteers from the FS community. Thanks, MC On Tue, Apr 14, 2009 at 10:01 AM, Brian West br...@freeswitch.org wrote: Nobody has written the es language files. Those would need to be written. /b On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote: Replying to myself... I forgot to indicate my version! I am running trunk rev 12862 on CentOS 5 x86_64. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call For Help: Janitor Projects
I am sorry, but I really have to comment this one. Why the fuck do we need to have sooo much politics on an open source project? Janitor, non-janitor, developer, non-developer, girl or boy, we are all trying to get this thing better, aren't we? So leave your fucking ego out of the question and get your ass doing something that will actually get this project somewhere like we all instead of trying to get yourself called something. You want the president title? Get it and start working. Tony is the master dude in this place because, like he said, he wrote most of the 300,000 line of code. That simple. The title core developers team (sounds great, doesn't it?) are because they do CORE! Wanna be called core developer, DO CORE! Anyway, my suggestion is, want something done? DO IT. Don't know how? Study! Don't want to know how ... buy Avaya or whatever. They will charge for your laziness. Sorry for the bad language. Mesquita On Apr 1, 2009, at 10:59 AM, Raymond Chandler wrote: seven wrote: I know that. And I'd like to read code. Developers written great code and also plenty of comments(which is documentation) in code. However, there are sth. don't need to comment in code but should be available on wiki. E.g. I followed the svn commit log, and found sip_auth_username and sip_auth_password added, so I documented to the wiki. That's the right attitude to have... now if there were more people doing that and less people complaining like little school girls, we could actually reach the next level in Open-Sourcetopia. -Ray ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Deployment information and use cases
Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add- on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra- building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at each site still answer all outside calls. If not answered, the call fails over to an IVR. Should we ever decide to do so, we are now perfectly positioned to have all inbound calls to the
Re: [Freeswitch-users] More troubles with SQLAlchemy and mod_python
I have to ask to _please_ do add more info on the wiki about that cos python is a language that has been growing a lot on commercial and open source worlds. I have a personal interest on that as well since I intend to do some things with the mod_python module as well. Thanks, Mesquita On Jan 30, 2009, at 12:34 AM, Brian West wrote: Can you do some examples and documentation on the wiki about what you're doing to maybe help others? /b On Jan 29, 2009, at 8:27 PM, Brian Deacon wrote: TA-DA! My python can now not only import the sqlalchemy module, but the code I had before that was actually doing some database interaction is working now. Thank you very much for all the help. It is much appreciated. It sounds like the contents on the wiki with the modules.conf.xml or the LD workarounds are superceded now. Should I change the comments on there to reflect that it should now be fixed? Brian ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_g729
Steve, As we speak I am actually negotiating with one of those companies to make a mod for their cards. Khomp has a very nice product and they are exporting to the rest of latin america now. Thanks, Mesquita On Jan 27, 2009, at 12:46 AM, Steve Underwood wrote: Hi Abdul, Abdul Hakeem wrote: Is Brazil a 3rd world country ? The last I hear Brazil was building aeroplanes, has it's own space and nuclear program and a GNP UK would be envious of. Cheers, AH What relevance does that have to the current discussion? Brazil is a country with large trade barriers, which skews the cost of hardware from the world market considerably. It is also pretty advanced, technically, and has a local base of electronics manufacturers. That considerably affects the economic tradeoffs in the use computers, telecoms, and other technology in Brazil. If something can be manufactured (or at least pass through final assembly) in Brazil, it will generally be much cheaper than something imported. That means some people find the use of locally made intelligent E1 cards is cheaper than the use of dumb cards from Digium or Sangoma. Regards, Steve ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH
Andrew, if you are interested, you could check out http://www.pjsip.org These guys have build a great lib that runs multiplatform on top o PA as well and are _REALLY_ small footprint. Perfect for a client, right? Thanks, Mesquita On Jan 16, 2009, at 9:35 PM, Andrew Thompson wrote: I'd like to announce the first beta release of a cross-platform ruby/ tk GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's not particularly fancy, but I needed a cross platform softphone with good voice quality that was debuggable and didn't have a ton of features to confuse the users. I couldn't find one so we built one. I've got some sparse documentation up at: http://opencsm.org/wiki/index.php/Spice_SoftPhone And you can download it from http://opencsm.org/download . It's under the MPL and I've been cleared to re-licence my other FreeSWITCH related projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris and OSX (it used to work on linux, I assume it still does). Comments/complaints/bugreports welcome. It's definitely still got some rough spots (I don't think it'll run without a controlling terminal, for example), but we're going to be polishing it up and hopefully putting it in production here in the next few weeks to replace a very buggy closed-source phone we've had to endure far too long. Please download it if you're interested, the download count helps us continue working on this kind of stuff :) Andrew - opencsm.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Would like to pickup a call that is on hold on another extension
Wouldnt that be call parking?? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park I have been told that would be better o use mod_fifo instead... It would be nice if someone would post something on mod_fifo wiki page about how to do fancy call parking with mod_fifo (even tho it might be pretty easy). Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Would like to pickup a call that is on hold on another extension
Well, sorry. That would be better, wouldnt it? http://wiki.freeswitch.org/wiki/Mod_fifo#Park_Time_Out_Example Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] XML lib curl - what is the best practice for directory binding?
Take a look at the wiki for this module. I have been updating it
constantly and there are a lot of new information there.
http://wiki.freeswitch.org/wiki/Mod_xml_curl
Regards,
Mesquita
On Jan 5, 2009, at 1:16 PM, can_...@gmx.de wrote:
Hello,
I have been looking into the xml curl directory binding and I would
like to update the wiki with the accepted best practice. I have
listed the HTTP POST request I am getting and how I respond. If
there is a better way please let me know and I will update the wiki
accordingly. Btw, what I have done works - so no bug hunting this
time ;-)
I will make a pylons webserver available in the next few days,
starting with dialplan and directory support.
Thank you,
Phil
At boot:
HTTP POST request 1
[('hostname', u'voip'), ('section', u'directory'), ('tag_name',
u''), ('key_name', u''), ('key_value', u'')]
my response:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory description=arbitrary stuff here
/section
/document
I have left the response empty as I want to provide the users at
runtime.
---
At boot:
HTTP POST request 2
[('hostname', u'voip'), ('section', u'directory'), ('tag_name',
u''), ('key_name', u''), ('key_value', u'')]
my response:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory description=arbitrary stuff here
/section
/document
---
At boot:
HTTP POST request 3
[('hostname', u'voip'), ('section', u'directory'), ('tag_name',
u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'),
('domain', u'192.168.178.22'), ('purpose', u'network-list')]
my response:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory description=arbitrary stuff here
/section
/document
What is meant by network list here? If all the users should be
loaded at boot time, is this the request which should get a response
with the complete list?
--
During runtime following this action:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=dialplan description=RE Dial Plan For FreeSwitch
context name=public
extension name=test1
condition field=destination_number expression=^(1)$
action application=voicemail data=default $${domain} 315/
/condition
/extension
/context
/section
/document
HTTP POST request:
('hostname', u'voip'), ('section', u'directory'), ('tag_name',
u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'),
('mailbox', u'315'), ('key', u'id'), ('user', u'315'), ('domain',
u'192.168.178.22'), ('ip', u'217.10.79.9')
my response:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory description=arbitrary stuff here
domain name=192.168.178.22 //change to your domain
groups
group name=default
users
user id=315 mailbox=315
params
param name=password value=1234/
param name=vm-password value=/
/params
variables
variable name=accountcode value=315/
variable name=user_context value=default/
variable name=vm_extension value=315/
variable name=max_calls value=1/
variable name=fail_over value=415/
variable name=cringback value=us-ring/
/variables
/user
/users
/group
/groups
/domain
/section
/document
--
Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit
allen: http://www.gmx.net/de/go/multimessenger
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Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client Available!
Thank you Jason, I was just going thru the code when I got your email. Saved me up some time. ;) Mesquita On Dec 30, 2008, at 2:00 AM, Jason White wrote: By the way, the command to exit fs_cli is /exit (or /bye or /quit). Commands starting with / are handled internally by the process_command() function of the CLI, instead of being treated as FreeSWITCH API commands. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
