- FreeSWITCH-users - FreeSWITCH Mailing Lists

09 Dec '08

My dialplan is pretty simple. I have a single trunk with a vonage softphone DID (1303... we'll call it main) and a "virtual" DID (1816...) which rings the softphone DID. All incoming calls show up as from softphone DID but the sip_to_user holds the actual number dialed so I can enter the dialplan properly. I have 2 extensions in my directory/extensions, one for each of the DID's. The extensions check sip_to_user for match and that works great. I match on ([0,1]?)(<10 digit did>) and it enters the dialplans correctly, plays the right music for each DID while the dial is occuring, so all that works. The bridge to user/$2@$${domain} also works fine. The continue_on_fail is set properly so on no answer call_timeout hits (at 25 secs), and goes to voicemail... works also for both numbers. transfer to voicemail is as follows <action application="answer"/> <action application="voicemail" data="default $${domain} $2"/> which should be pulling $2 from the condition check shown above, which it does, cuz the bridge works... When I call in on main DID, I get "leave a message for 1303..." The Main DID.. When I call in on virtual, I get "leave a message for 1303..." The Main DID rather than the 1816.... How can I get voicemail to use the correct DID. HELP!! :D -- View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20791453.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

9 28

mod_xml_ldap
by Helmut Kuper 09 Dec '08

09 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried compile FS with mod_xml_ldap with trunk of yesterday. During compiling it can't find http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked there and found that the filename on freeswitch.org side has changed to http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkk+RHwACgkQ4tZeNddg3dyp7wCeN7fvIj4OSH1rsuglD46qtS36 iR8AnjypwB2XT/rAYr61yyMXJ+iUY4/d =iJw/ -----END PGP SIGNATURE-----

3 2

Read app ignores custom variable when called via socket interface
by Jan Kubr 09 Dec '08

09 Dec '08

Hi, I checked out the current trunk (rev 10641) and found out that the read app ignores the varname parameter, it always puts the result in the DTMF-Digit variable. I'm calling it via the socket interface: sendmsg call-command: execute execute-app-name: read execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 # event-lock:true In the XML dialplan it works fine: <action application="read" data="1 1 sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/> <action application="phrase" data="spell,${res}"/> I have been using the socket call above successfully in the 1.0.1 release. Any ideas? Thanks, Jan Kubr

2 4

No audio after transfer
by John Rutherford 08 Dec '08

08 Dec '08

I'm trying to get an attended transfer work with freeSWITCH, but it's not quite working. I have Microsoft Speech Server on one side and Televantage on the other. MSS is originating a call, which freeSWITCH is bridging to Televantage. That calls connects just fine. Then, MSS sends a re-INVITE to Televantage to put the call on hold. This works too. Then, MSS originates another call to freeSWITCH, which is again bridged to Televantage. This works fine too. Then, MSS sends a REFER to freeSWITCH to do the transfer. The transfer should be complete, but there is no audio between the two calls-just silence. I have looked at pcaps and the freeSWITCH logs, but I'm not seeing anything obviously wrong. After the REFER, I can see audio for both calls going between freeSWITCH and Televantage, so it seems that the only thing missing is freeSWITCH routing the audio from one call to the other call and vice-versa. Any help would be greatly appreciated. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Thanks! John

2 2

Two nice to have features in fs
by Dennis 08 Dec '08

08 Dec '08

hi, we are using socket outbound and found out, that there are two (perhaps) small things, we would like to see in fs. 1.) if we try to playback a soundfile, which does not exist, we do not get an error or something. in the cli of fs we can see the error, but we do not get anything about it over the socket. we get an execute event for the playback and directly after it an execute-complete. wouldn't it be nicer, if one could get a notice about an error, like: execute-complete-error or a variable, which is set with an error, if an error occours? 2.) the session heartbeat event is an absolutely great feature in fs. but sometimes we would like to get the session heartbeat every 5 seconds. at the moment the allowed minimum is every 10 seconds. a smaller setting, like every 5 seconds, will result in a session heartbeat of every 60 seconds. to help us out, we edit the switch_core_session.c at line 899 and change the "seconds < 10" to "seconds < 5". because one has to set "enable_heartbeat_events=5" manually, i do not think, that there is a risk, that others, who do not want the heartbeat to come that often, will be negatively affected by this change. might it be possible, to do the same changes to the default code? thanks dennis

3 5

rootkit?
by Martin Joseph 08 Dec '08

08 Dec '08

What is the rootkit item that appears to be added to the SVN of trunk? Thanks, Marty

3 2

conference configured to call automatically the attended does not work
by Carole O. 08 Dec '08

08 Dec '08

Hello, I have got some problems for the configuration of a simple conference which should be established by calling an extension and automatically inviting 2 people. Actually, this is based on the default configuration of Freeswitch (extension 0911). I have changed it a little: <extension name="conference_test"> <condition field="destination_number" expression="^0911$">  <action application="set" data="conference_auto_outcall_caller_id_name=telephoneX" /> <action application="set" data="conference_auto_outcall_id_number=0911" /> <action application="set" data="conference_auto_outcall_timeout=60" /> <action application="set" data="conference_auto_outcall_flags=none" />  <action application="conference_set_auto_outcall" data="sofia/internal/1010@$${domain}" /> <action application="conference_set_auto_outcall" data="sofia/internal/1002@$${domain}" /> <action application="conference" data="conferenceTest@default" /> </condition> </extension> I have attached a file with the console errors. There are some errors (moh errors) but since these were also present for room conference and it did not prevent it for working, I guess this is not the fundamental reason for the previous problem. I have an additional question. I have installed freeswitch from opensuse.org, there is a simple "one-click installation" but I am not sure this was a good idea, it seems to be light isn't? Thanks for your help, Carole http://www.nabble.com/file/p20856465/error_console.txt error_console.txt -- View this message in context: http://www.nabble.com/conference-configured-to-call-automatically-the-atten… Sent from the Freeswitch-users mailing list archive at Nabble.com.

1 1

Autoanswer
by Jonas Gauffin 08 Dec '08

08 Dec '08

Hello I send an API command through the event socket that looks like this (the first two variables are used by our server): api originate {gate_user_id=44,gate_site_id=1,sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true,origination_caller_id_name='Jonas Gauffin',origination_caller_id_number=+4623666XXXX,sip_auto_answer=true}sofia/localdomain/u1000044% 192.168.1.112 '&execute_extension(8901 XML)' The command works just as it should. The problem is that the auto-answer variables seems to stick to the b-leg (execute extensions), which means that both calls gets auto answered. What I want is that only the first call gets answered. (A logged in user presses the "call" icon in our webdirectory, which makes freeswitch use his phone to call 8901) //Jonas

2 3

Re: [Freeswitch-users] Handling directory of sound files
by Michael S Collins 07 Dec '08

07 Dec '08

Check out mod_localstream on the wiki and see if that sounds like what you need. I'm still learning it all myself but I believe that's where you should start. Please report back with any questions and we will take it from there! -MC On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <faisalmaqsoodi(a)yahoo.com> wrote: > Its not without music on hold completely. Say, e.g, moh is being > played but when i press 1 it should start playing files contained in > a specific directory sequentially or randomly. I havent got any > solution to this problem yet. Can anyone plz guide me to some > documentation or anything else regarding this matter. > > > > > > > > > > > > > > > > > Faisal > > _______________________________________________ > 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

3 2

VoIP Product/Service and Man/Woman of the year 2008
by UV 07 Dec '08

07 Dec '08

Biz-news.com <http://www.linkedin.com/redirect?url=http%3A%2F%2FBiz-news%2Ecom&urlhash=oN Bl&_t=disc_detail_link> is conducting a survey to discover the leading VoIP man/woman of the year 2008, and the best VoIP product or service also for the year 2008 . The winner will be selected based on the opinions of professionals and technology enthusiasts in the industry. A) best VoIP product or service of 2008 form: http://voip.biz-news.com/forms/py2008 <http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff orms%2Fpy2008&urlhash=dspi&_t=disc_detail_link> B) VoIP Man/Woman of the year 2008 form at: http://voip.biz-news.com/forms/my2008 <http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff orms%2Fmy2008&urlhash=eDrg&_t=disc_detail_link> The results will be published early 2009 and share the raw data with the community.

1 0