June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Set Codec per Endpoint
by JuanMa 09 Jun '09

09 Jun '09

Hello, Is it posible to set a codec-pref per Endpoint instead to set it in sip-profiles? <param name="codec-prefs" value="GSM"/> In my directory: <user id="2000" mailbox="2000"> <params> <param name="password" value="2000"/> <param name="vm-password" value="2000"/> <param name="codec-prefs" value="GSM"/> </params> <variables> <variable name="accountcode" value="2000"/> <variable name="user_context" value="default"/> <variable name="sip-force-contact" value="NDLB-connectile- dysfunction"/> </variables> I do this but FS do codec negotiation with all codecs in the profile internal. What i am doing wrong? Thx in advance

2 1

bug in channel event other leg parameters?
by zigurds 09 Jun '09

09 Jun '09

Hi, all. While expecting channel events that are fired in call lifetime, have encountered that for lot of channel events Other-Leg-Username points to user that belongs to channel event for what is fired. For example, let's say I'm calling from 101 to 102, when receiving events CHANNEL_UNBRIDGE, CHANNEL_EXECUTE_COMPLETE, CHANNEL_HANGUP, if channel name is Channel-Name: sofia/Test/101%40XXXX, I'm getting also Other-Leg-Username: 101. I was expected that there should be 102 or I'm misunderstanding something? Some other information in Other-Leg too releates to user 101. By the way, does there exist more detailed information about call events flow and what event attributes mean. Seems that http://wiki.freeswitch.org/wiki/Event_list is very uncomplete... Thanks, Zigurds -- View this message in context: http://www.nabble.com/bug-in-channel-event-other-leg-parameters--tp23947374… Sent from the Freeswitch-users mailing list archive at Nabble.com.

1 0

Error making FS configure
by Lars Zeb 09 Jun '09

09 Jun '09

I tried to make FS current with the following command, which got errors in the ./configure step. Is it OK to proceed to make install? make clean && svn up && ./configure configure: creating ./config.status config.status: creating Makefile config.status: creating libedit.pc config.status: creating src/Makefile config.status: creating doc/Makefile config.status: creating examples/Makefile config.status: creating config.h config.status: executing depfiles commands configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre Thanks, Lars

3 2

broken compilation on windows?
by Giovanni Maruzzelli 09 Jun '09

09 Jun '09

Hi all, I cannot compile on Windows the current svn, 13722. The first error it gives is: freeswitch\libs\pcre\pcre_internal.h(368) : fatal error C1189: #error : LINK_SIZE must be either 2, 3, or 4 then it fails 81 projects (42 succeeded), because no freeswitchcore.lib (obviously) I tried both the Freeswitch.2008.sln and the freeswitch.express.2008.sln, I'm using VC Express 2008. -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

1 0

mod vmd and lua - Solved
by Nik Middleton 09 Jun '09

09 Jun '09

I finally got around to looking at why mod vmd didn't appear to run when using LUA. Turned out that the example in the wiki was wrong. It should have been session:execute("vmd","start"); And not session:execute("vmd"); I've updated the wiki Regards,

2 1

ClueCon 2009 Information - Roommates
by Michael Collins 09 Jun '09

09 Jun '09

FYI, We've had several people inquire about finding someone with whom to share a room for ClueCon. Please keep in mind that we do have a minimum number of rooms we need to book with the Wyndham. However, we would hate for someone not to attend simply because they could not afford a hotel room. To that end I would like to ask for a volunteer from the community to be the go-to person for helping roommates to get connected. Perhaps we could get more total rooms booked by helping those with rooming needs and who might not otherwise be able to come to ClueCon this year. Please email me off list if you are able to help. Thanks! -Michael S Collins http://www.cluecon.com

1 0

Re: [Freeswitch-users] Inbound using FS
by Peter Olsson 09 Jun '09

09 Jun '09

Have you configured the sip profile to use the acl list you have created (Inbound_Test)? /Peter ----- Ursprungligt meddelande ----- Från: Rex_Alex <rex.alex345(a)yahoo.com> Skickat: den 8 juni 2009 17:40 Till: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Inbound using FS Hi Brian, I am using audiocode box with IP 2.2.2.2 and FS with IP 1.1.1.1 Addions made are, acl.conf.xml <list name="Inbound_Test" default="allow"> <node type="allow" cidr="2.2.2.2/32"/> </list> freeswitch.xml <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> sip_profiles/external.xml under <settings> tag public.xml <extension name="Inbound"> <condition field="destination_number" expression="^(123456)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> default.xml <extension name="Inbound"> <condition field="destination_number" expression="^123456$"> <action application="ring_ready" /> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/internal/1007%1.1.1.1"/> </condition> </extension> Still, Inbound is not hitting my FS console itself. Please assist where am I going wrong? Thanks, Rex Brian West wrote: > > > On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote: > >> >> Hello, >> >> My public.xml configration is: >> >> <extension name="inboundsip"> >> <condition field="destination_number" expression="^123456$">  >> <action application="transfer" data="$1 XML default"/> >> </condition> >> </extension> > > $1 will not exist in this case because your regular expression doesn't > capture anything. So replace $1 with your target number or use > ^(123456)$ > > >> >> My default.xml configration is: >> >> <extension name="Inbound"> >> <condition field="destination_number" expression="^*****$"> >> <action application="ring_ready" /> >> <action application="set" data="call_timeout=20"/> >> <action application="bridge" data="sofia/sip/1001%freeswitchip"/> >> </condition> >> </extension> > > Can you elaborate how you're registering with your provider? > > >> >> >> When I am trying to call 123456 from my mobile no. Not able to see any >> logging in FS console. Please assist where I am going wrong? Or do I >> require >> any extra modules to be installed? >> >> Thanks, >> Rex > > Brian West > brian(a)freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > 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 > > -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3043665.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ 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 !DSPAM:4a2d30e032931222027793!

3 10

Re: [Freeswitch-users] DTMF Problems
by Rudolf Denert 09 Jun '09

09 Jun '09

Hello, I can give the all-clear! It was my mistake ( ...and it was a silly one :-/ ) I had to applications that interfere each other. They are:   I don't know why I skip that in my dialplan!!! http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Yes! A small mistake with a huge effect. But thanks for all you help. :-) Greetz ----- Ursprüngliche Mail ----- Von: "Brian West" <brian(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Montag, 8. Juni 2009 17:12:35 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] DTMF Problems I wrote that to demonstrate that exact situation but you still can't tell if they are inband or info :P /b On Jun 8, 2009, at 10:08 AM, Kristian Kielhofner wrote: Rudolf, I believe there is a snippet in the sample XML dialplan to detect the lack of telephone-event in the SDP and activate inband detection. You could use that for inspiration. Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ 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

Reject call without answering
by Klaus Teller 09 Jun '09

09 Jun '09

Hi, I'm still looking for a way to reject a call without answering. I've tried various things without solution. >From the socket interface i tried: SendMsg 9015430e-82cf-418c-bf4c-f3ac6e85caf2 call-command: execute execute-app-name: respond execute-app-arg: 503 >From Javascript, i tried each of the followings: session.hangup(); session.reject(); session.execute("respond", data="503"); In the two first cases as well as in the socket interface case, it seemed as if Freeswitch sent a kill signal to the VoiP provider. But that isn't enough apparently to cancel the call. In the third case, i get the message: Session is not active. Any other suggestion? Thanks, Klaus. -- GMX FreeDSL mit DSL 6.000 Flatrate und Telefonanschluss nur 17,95 Euro/mtl.! http://dslspecial.gmx.de/freedsl-aktionspreis/?ac=OM.AD.PD003K11308T4569a

1 0

Re: [Freeswitch-users] Problem with attendant transfer
by Durk de Beer 09 Jun '09

09 Jun '09

Hallo Ray Siemens type Gigaset DE380 IPR, Cisco type 7965 and 7960. Also tested Grandstream type 2010 and Linksys type 921 no problem with these phones. I've downloaded a new firmware for the Siemens but wasn't able to test it jet. Durk > Durk de Beer wrote: >> >> Hello all, >> >> I have observed an issue on using Freeswitch and some SIP-phones. Ok >> the problem is this. Some phones >> > which phones? > -Ray

1 0