August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

What should be the dial plan for gateway calls?
by Adeel Ansari 11 Sep '08

11 Sep '08

Hi all, I have tried a few possibilities to match the dial plan for the command below, but none is working. *originate sofia/gateway/gizmo1/6098989898 &bridge(sofia/gateway/gizmo9/0116054545454) * Below is what I have tried. === <condition field="source" expression="mod_sofia"> and <condition field="destination_number" expression="^(gizmo[1-10])$"> === Thanks for your support. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
by Anthony Minessale 09 Sep '08

09 Sep '08

nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote: > Just to warn you we do not use mod_iax all that much. It was used as a > tool to test the core of FreeSWITCH in the early stages of development > and we just kinda left it hanging around since it does at least work. > We had to add a bunch of code to the stock iaxclient lib to make it > threadsafe etc because the lib is designed to handle 1 or 2 calls and we > are using it in a b2bua type env where hundreds of concurrent calls may > be possible. That said, since we more or less forked the code to avoid > generating support issues for the other soft-phones that use iaxclient, > you may want to check if there were any bugs in the real version of the > library that match your issue. > > http://sourceforge.net/projects/iaxclient > > > We've had little to no demand for IAX2 so we have not focused very much > on it. We do have strategies for creating a more robust and perhaps even > an "improved" implementation of the IAX2 protocol. Alas, it's pretty > much purely an academic exercise because there is little reason to > implement it. > > > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Birgit Arkesteijn <birgit(a)westhawk.co.uk> > To: Freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 10, 2008 10:52:40 AM > Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable > > Hi, > > We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64). > > I'm experiencing problems with DTMF: > The IAX2 DTMF frames aren't always transferred into RTP packets. > Sometimes none of the frames are translated, sometimes only the first > one is omitted. All the IAX2 DTMF packets are ack-ed. > > Our setup: > PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) > -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) > -> SIP to SystemX (sofia/default/123(a)66.212.134.192) > > > I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug > that could explain this, but > failed, so I submitted this as a bug, see > http://jira.freeswitch.org/browse/FSCORE-109 > > Has anyone else experienced the same or a similar problem? > > Thanks, Birgit > -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk> _______________________________________________ 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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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GUI
by Ilan Perez 03 Sep '08

03 Sep '08

Anyone developing a GUI for freeswitch Ilan Perez Webmaster 0432 326 017

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INVITE no SDP
by hemant kumar 03 Sep '08

03 Sep '08

Hi All, I have registered freeswitch to an external gateway as a softphone, and it has registered nicely. Now, i had given extension 1000 to freeswitch. When i try to call from external gateway to freeswitch, it always gives Temporarily Unavailable. However, if i register a phone with freeswitch i am able to dial that extension. Can someone please help me out in this ? Thanks, Hemant

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Re: [Freeswitch-users] How to originate a session from outside
by Boris Krivonog 02 Sep '08

02 Sep '08

The easiest way would be using XML-RPC (http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC) with Apache XML-RPC (http://ws.apache.org/xmlrpc/client.html) libraries the code would look something like: XmlRpcClientConfigImpl config = new XmlRpcClientConfigImpl(); config.setServerURL( new URL("http://<freeswitchip>:8080/RPC2") ); config.setBasicPassword("freeswitch"); config.setBasicUserName("works"); XmlRpcClient client = new XmlRpcClient(); client.setConfig(config); Object[] params = new Object[]{ "originate", "sofia/gateway/gizmo2/01160176xxxxxx &park()" }; client.execute( "freeswitch.api", params ); Since the code above is written "from my memory", take it for illustrative purpose only, however you should get the idea. If you need more control over the call(s), consider using http://wiki.freeswitch.org/wiki/Event_Socket. Hope this helps, Boris On Fri, Aug 22, 2008 at 12:24 PM, Boris Krivonog <boris.krivonog(a)gmail.com> wrote: > > The easiest way would be using XML-RPC (http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC) with Apache XML-RPC (http://ws.apache.org/xmlrpc/client.html) libraries the code would look something like: > > XmlRpcClientConfigImpl config = new XmlRpcClientConfigImpl(); > config.setServerURL( new URL("http://<freeswitchip>:8080/RPC2") ); > XmlRpcClient client = new XmlRpcClient(); > client.setConfig(config); > Object[] params = new Object[]{ "originate", "sofia/gateway/gizmo2/01160176xxxxxx &park()" }; > client.execute( "freeswitch.api", params ); > > Since the code above is written "from my memory", > > > > Adeel Ansari wrote: > > Hi All, > > I had success in bridging a session, using my Java Program as a hook, and manage to connect one mobile phone to another. It goes like below. > > === > --> originate sofia/gateway/gizmo2/01160176xxxxxx &java(net.digitalapps.freeswitch.CallTest) > === > > In this case, freeswitch call my Java program. Now, My intension is to not use console at all, instead use my java program to originate a session in freeswitch and then bridge the session. So, call freeswitch from my program, not the other way around. > > Is it possible? Need suggestions in any case. I mean if I need to come up with my own client. Actually, I was looking JAIN-SIP to achieve this. But then thought of asking your suggestions. With JAIN I might go like this, place a call, hit the dial plan invoke my Java Hook Program and bridge the call. But I am afraid that it might connect my client with the mobile phone, not one phone to another. > > Thanks. > > -- > Best, > Adeel Ansari > > http://www.linkedin.com/in/adeelansari > > ________________________________ > _______________________________________________ > 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 >

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IVR menu ending with #
by Sheeju Alex 02 Sep '08

02 Sep '08

Hi All, I have created a IVR menu where in the digits is ended with #, could anyone point me why a menu with ending digits (e.g. 12345#) doesn't work? <entry action="menu-exec-app" digits="/^(\d{5})#$/" param="transfer $1 XML default"/> Does Freeswitch IVR doesn't support # in digits? Thanks, Sheeju

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Re: [Freeswitch-users] TLS and SRTP between 2 Freeswitch servers
by Peter P GMX 02 Sep '08

02 Sep '08

Hello Brian, any idea, when we can expect a fix for that? Best regards Peter Brian West schrieb: > I just realized this is a gateway call... that will have to be fixed. > > /b > > On Aug 27, 2008, at 3:24 PM, Peter P GMX wrote: > >> Hello Brian, >> >> I tried to set ;transport=tls on the sofia string and it showed me an >> invalid gateway >> >> So I checked the gateway: server1 is not registered on server2 >> freeswitch@freeswitch> sofia status gateway freeswitch2 >> API CALL [sofia(status gateway freeswitch2)] output: >> Invalid Gateway! >> >> When I set the register-transport parameter back from tls to udp on >> server1 I get >> freeswitch@freeswitch> sofia status gateway freeswitch2 >> API CALL [sofia(status gateway freeswitch2)] output: >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> Name freeswitch2 >> Scheme Digest >> Realm xxx.xxx.xxx.56 >> Username 1007 >> Password yes >> From <sip:1007@xxx.xxx.xxx.56;transport=udp> >> Contact <sip:1007@xxx.xxx.xxx.55;transport=udp> >> To sip:1007@xxx.xxx.xxx.56 >> Proxy sip:xxx.xxx.xxx.56 >> Context default >> Expires 3600 >> Freq 3600 >> Ping 0 >> PingFreq 0 >> State REGED >> Status UP >> ============================== >> I ngrepped the traffic between server1 and server2 and could see that >> there was traffic on port 5060. >> >> Switching back to TLS it didn't work again. >> <param name="register-transport" value="tls"/> >> I ngrepped the traffic between server1 and server2 on all ports and >> could see that there was absolutely no traffic between them. >> >> So, as soon as I enter tls instead of upd, the traffic stops. >> Any hint? >> >> Best regards >> Peter >> >> >> >> Brian West schrieb: >>> You Append ;transport=tls on the sofia string which is much easier. >>> >>> /b >>> >>> >>> >>> On Aug 26, 2008, at 4:51 PM, Peter P GMX wrote: >>> >>> >>>> I have tried to set this up, but I need some help to get TLS to work >>>> >>>> What does work: >>>> I setup 2 freeswitch servers with IP xxx.xxx.xxx.55 and xxx.xxx.xxx. >>>> 56. >>>> They are connected via UDP/Port 5060 (SIP) >>>> I have 2 snom phones connected to the servers via TLS (1002@ >>>> xxx.xxx.xxx.55 and 1003@ xxx.xxx.xxx.56) >>>> Server1 registers to Server2 as UA 1007 >>>> I can dial "8001" on 1002@server and reach 1003@server2 and make a >>>> call >>>> >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> _______________________________________________ >> 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 > > Brian West > sip:brian@freeswitch.org > > >

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Country Codes + Area codes
by Seysan 01 Sep '08

01 Sep '08

Hello, Does anyone have or Can help me find a Complete list of All the Countries in the world with the Countrycode? Also, is there a list with cities breakdown o at least important cities of each country plus their codes? I hoping an Excel or CSV file, but if this list exist it doesn't matter what format, I will be able to use it. thank you in advance, Seysan

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HA clustering solution?
by Tom Warren 01 Sep '08

01 Sep '08

Hi everyone, this is my first post. I'd like to know if there is any plan for an HA solution for FreeSWITCH. I know the 'heartbeat' trick, but is there or will there be any way to mirror the configuration changes and even call state between two boxes in a sort of clustered arrangement? Thanks, Tom

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TLS newbie question
by xbipin 01 Sep '08

01 Sep '08

my question is that in the freeswitch ssl folder i have 2 files agent.pem (contains RSA PRIVATE KEY and CERTIFICATE) cacert.pem (contains CERTIFICATE) now for the device connecting to to freeswitch, i use the linksys spa3102 ATA and it supports TLS and there r 2 fields in it: Mini Certificate SRTP Private Key now i want to be able to use TLS as well as SRTP so what do i put in those 2 fields, i mean from the 2 files in freeswitch folder, what portion should i use for the device or is it that in TLS, freeswitch gives the key to the device automatically? in the device properties it shows: Client Certificate: installed -- View this message in context: http://www.nabble.com/TLS-newbie-question-tp19232000p19232000.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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