June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Blog post about SIP
by Kristian Kielhofner 18 Jun '12

18 Jun '12

FreeSWITCHers, As some of you may have noticed I frequently comment on SIP-related FreeSWITCH posts. Long story short I decided to finally put some of these thoughts and experiences down in writing. What I ended up with is a 21 page (and counting) document describing various SIP header fields, SDP information, RTP issues, DTMF issues, NAT traversal technologies, interop headaches, etc. I wrote a little intro and linked to the document if any of you would like to check it out (link at the end of post): http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html Of course I'd appreciate any feedback you may have. What do you disagree with? Find a typo? Have burning SIP/RTP/SDP/T.38/codec questions you'd like to see me address? Should I get into more implementation (FreeSWITCH) specific issues? Let me know. Thanks! -- Kristian Kielhofner

9 10

Conference Caller Announce
by Bobthebeat 18 Jun '12

18 Jun '12

Hi Folks, brand new Freeswitch user here. I have modified the example shown on the mod_conference (http://wiki.freeswitch.org/wiki/Mod_conference#FAQ) wiki for my own deployment. I am attempting to have a user record his name, then have this recording announced on joining the conference, but I am struggling. Here is my dialplan entry:- <extension name="main_number_11" continue="true"> <condition field="destination_number" expression="^xxxxxxxxxxx$"> <action application="export" data="hold_music=silence"/> <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="conference/conf-welcome.wav"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-record_name1.wav"/> <action application="playback" data="tone_stream://%(1000,0,500)"/> <action application="record" data="${namefile} 1"/> <action application="playback" data="ivr/ivr-call_being_transferred.wav"/> <action application="set" data="res=${sched_api +1 none conference $1-${domain_name}@conference_2 play file_string://${namefile}!conference/conf-has_joined.wav}"/> <action application="transfer" data="$1 XML default"/> </condition> </extension> The call connects and I am prompted to record, then I am notified of the imminent transfer, but it bombs out here. Sophia output reads Command conference($1-192.168.3.178@conference_2 play file_string:///tmp/b34c9679-9fdb-4934-a1d9-dd4b646c88e2-name.wav!conference/conf-has_joined.wav): Conference $1-192.168.3.178@conference_2 not found I can dial into the conference without issues when I disregard the 'Set' expression and just dial the conference directly. Can anybody point me in the right direction? Thank you very much, B -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-Caller-Announce-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

5 8

Freeswitch [DEBUG] libdingaling.c:1242 TLS NOT SUPPORTED IN THIS BUILD!
by Mariya Zheleva 18 Jun '12

18 Jun '12

Hello, I am trying to configure freeswitch for outbound calls and SMS through google voice using mod_dingaling. I have succeeded with the SMS configuration but I can't get call to work. It seems the problem is that TLS can't be successfully built upon freeswitch installation. I followed the instructions here: http://wiki.freeswitch.org/wiki/Mod_dingaling I added the flag "-lgnutls" to gcc in relink_command in the file file src/mod/endpoints/mod_dingaling/mod_dingaling.la, however I still get *[DEBUG] libdingaling.c:1242 TLS NOT SUPPORTED IN THIS BUILD!* in the freeswitch.log This is the output when I make mod_dingaling-install. * * *openbts@ubuntu:/usr/local/src/freeswitch$ sudo make mod_dingaling-install* * /bin/bash /usr/local/src/freeswitch/quiet_libtool --mode=install /usr/bin/install -c libfreeswitch.la '/usr/local/freeswitch/lib'* *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0* *quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })* *quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })* *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai /usr/local/freeswitch/lib/libfreeswitch.la* *quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a /usr/local/freeswitch/lib/libfreeswitch.a* *quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a * *quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a* *quiet_libtool: finish: PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/X11R6/bin:/sbin" ldconfig -n /usr/local/freeswitch/lib* *----------------------------------------------------------------------* *Libraries have been installed in:* * /usr/local/freeswitch/lib* * * *If you ever happen to want to link against installed libraries* *in a given directory, LIBDIR, you must either use libtool, and* *specify the full pathname of the library, or use the `-LLIBDIR'* *flag during linking and do at least one of the following:* * - add LIBDIR to the `LD_LIBRARY_PATH' environment variable* * during execution* * - add LIBDIR to the `LD_RUN_PATH' environment variable* * during linking* * - use the `-Wl,-rpath -Wl,LIBDIR' linker flag* * - have your system administrator add LIBDIR to `/etc/ld.so.conf'* * * *See any operating system documentation about shared libraries for* *more information, such as the ld(1) and ld.so(8) manual pages.* *----------------------------------------------------------------------* * * *making install mod_dingaling* Any suggestions as to how to build TLS are highly appreciated. Thanks, mariya~

2 2

BYE authentication problem
by Carlo Dimaggio 18 Jun '12

18 Jun '12

Hello, my FS has a trunk with an openser: FS -> openser (sip.test.com - 1.2.3.4) -> cisco gateway I have configured a gateway with this sip proxy / realm and I have a problem with BYE. When I call from PSTN (cisco gateway) to FS, the cisco device set a realm "1.2.3.4" and openser ask for authentication, while FS tell: 2012-06-14 14:23:44.018150 [ERR] sofia_reg.c:2160 Cannot locate any authentication credentials to complete an authentication request for realm '"1.2.3.4"' I know that FS asks that realm configured in the gateway section (sip.test.com) should match the realm in the challenge packet from the other device (that instead is 1.2.3.4), but the Reverse DNS of 1.2.3.4 is sip.test.com (and viceversa). Should FS respond to the challenge packet? How can I solve this problem? Best Regards,

1 1

Using * as "terminate and cancel"
by Alex Lake 18 Jun '12

18 Jun '12

Is there a way to get playAndGetDigits to allow the user to enter a (telephone) number, but if they decide part way through that they messed up and they want to abort, they can press * - and then (without checking validation) the lua script knows that they pressed * rather than the more normal # (which means "I've finished - now go do it!").

1 0

Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
by awais nazir 17 Jun '12

17 Jun '12

Actually , some user defined variables like huting attempt ,customer and vendor names I define in dialplan. The solution to make join with uuid is bit complicated and seem to require bit efforts like pushing in to mysql and then make join. Even then the variables only appears according to successful or last bridge attempt. On Sun, Jun 17, 2012 at 10:15 PM, Avi Marcus <avi(a)avimarcus.net> wrote: Oh that's what you meant.If it's in leg A, why do you expect it to be in Leg B CDR? That said, there's an option to let you do that: http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr but I've never tried it. -Avi On Sun, Jun 17, 2012 at 7:27 PM, awais nazir <awais-nazeer(a)hotmail.com> wrote: I will attempt get reply again, the channel variable being set in dialplan are seen in A leg cdr but not in B leg cdr, can somebody help ? From: awais-nazeer(a)hotmail.com To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful Date: Sat, 16 Jun 2012 23:07:14 +0500 Hello Thanks enabling leg B is certainly giving expected results but it's now showing up few variables including some user defined variables defined in the dialplan being executed. This problem did not exist in default enabled leg A. On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi(a)avimarcus.net> wrote: By default, the modules for logging only log the A leg of the call. You can turn on b-leg logging in whichever module you are using for call logs. Do note you'll have to reconstruct the call then, based on which is the A-leg and which is B leg, and using bridge_uuid, signal_bond or last_bridge_to of the B leg (whichever is set) to see which A leg it came from. (Also note on enterprise originates: they are reported as leg A in xml_cdr and need to be tied using the variable: ent_originate_aleg_uuid) This kind of "advanced" CDR handling should probably be wikified... -Avi On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer(a)hotmail.com> wrote: > Hi > > I am trying to use following context to get failover gateways working > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/local_profile/1111(a)example1.company.com" /> > <action application="bridge" > data="sofia/local_profile/1111(a)example2.company.com" /> > > if call on first gateway fails , call connection is attempted on second > gateway but FS is writing only one CDR.How can it write CDR on every bridge > attempt. > > BR > --waisee _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com

1 1

(no subject)
by awais nazir 17 Jun '12

17 Jun '12

Actually , some user defined variables like huting attempt ,customer and vendor names I define in dialplan. The solution to make join with uuid is bit complicated and seem to require bit efforts like pushing in to mysql and then make join. Even then the variables only appears according to successful or last bridge attempt. On Sun, Jun 17, 2012 at 10:15 PM, Avi Marcus <avi(a)avimarcus.net> wrote: Oh that's what you meant.If it's in leg A, why do you expect it to be in Leg B CDR? That said, there's an option to let you do that: http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr but I've never tried it. -Avi On Sun, Jun 17, 2012 at 7:27 PM, awais nazir <awais-nazeer(a)hotmail.com> wrote: I will attempt get reply again, the channel variable being set in dialplan are seen in A leg cdr but not in B leg cdr, can somebody help ? From: awais-nazeer(a)hotmail.com To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful Date: Sat, 16 Jun 2012 23:07:14 +0500 Hello Thanks enabling leg B is certainly giving expected results but it's now showing up few variables including some user defined variables defined in the dialplan being executed. This problem did not exist in default enabled leg A. On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi(a)avimarcus.net> wrote: By default, the modules for logging only log the A leg of the call. You can turn on b-leg logging in whichever module you are using for call logs. Do note you'll have to reconstruct the call then, based on which is the A-leg and which is B leg, and using bridge_uuid, signal_bond or last_bridge_to of the B leg (whichever is set) to see which A leg it came from. (Also note on enterprise originates: they are reported as leg A in xml_cdr and need to be tied using the variable: ent_originate_aleg_uuid) This kind of "advanced" CDR handling should probably be wikified... -Avi On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer(a)hotmail.com> wrote: > Hi > > I am trying to use following context to get failover gateways working > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/local_profile/1111(a)example1.company.com" /> > <action application="bridge" > data="sofia/local_profile/1111(a)example2.company.com" /> > > if call on first gateway fails , call connection is attempted on second > gateway but FS is writing only one CDR.How can it write CDR on every bridge > attempt. > > BR > --waisee _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

1 0

Re: [Freeswitch-users] freeswitch to write cdr after every bridge attempt (failed or successful
by awais nazir 17 Jun '12

17 Jun '12

Hello Thanks enabling leg B is certainly giving expected results but it's now showing up few variables including some user defined variables defined in the dialplan being executed. This problem did not exist in default enabled leg A. On Fri, Jun 15, 2012 at 6:51 PM, Avi Marcus <avi(a)avimarcus.net> wrote: By default, the modules for logging only log the A leg of the call. You can turn on b-leg logging in whichever module you are using for call logs. Do note you'll have to reconstruct the call then, based on which is the A-leg and which is B leg, and using bridge_uuid, signal_bond or last_bridge_to of the B leg (whichever is set) to see which A leg it came from. (Also note on enterprise originates: they are reported as leg A in xml_cdr and need to be tied using the variable: ent_originate_aleg_uuid) This kind of "advanced" CDR handling should probably be wikified... -Avi On Fri, Jun 15, 2012 at 4:17 PM, awais nazir <awais-nazeer(a)hotmail.com> wrote: > Hi > > I am trying to use following context to get failover gateways working > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/local_profile/1111(a)example1.company.com" /> > <action application="bridge" > data="sofia/local_profile/1111(a)example2.company.com" /> > > if call on first gateway fails , call connection is attempted on second > gateway but FS is writing only one CDR.How can it write CDR on every bridge > attempt. > > BR > --waisee

2 2

Support for NetEQ in FS.
by Venkatesh 17 Jun '12

17 Jun '12

Hi ! I wanted to know if there are plans to integrate NetEQ in FreeSwitch. I read a few postings where there were some proposals indicating on these lines, but wanted to see if anybody is working on the same? Regards, Venkatesh

1 0

No dial-tone on Siemens C610
by ocset 17 Jun '12

17 Jun '12

Hi I have deployed 3 cordless Siemens C610 phones on FS and was very surprised that there is no dial-tone when a user makes a call. The users hears a sound similar to an engage signal while FS makes the connection, followed by the sound of the phone ringing once the connection is established. (hope that makes sense). Siemens reckons that is the way their phones behave. Is there a way for FS to simulate a dial-tone while the user waits for the other parties phone to start ringing? Thanks O

2 3