April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

New LinkedIn Group
by Ken Rice 07 Apr '08

07 Apr '08

Hey Guys I have created a linked in Group for FreeSwitch... If you are a freeswitch user or developer and want to find other people on LinkedIn feel free to use the following link to join the group... http://www.linkedin.com/e/gis/84302/3EC65A7CD494 Ken http://www.cnam4voip.com/ krice(a)rmktek.com

1 0

freeswitch as a SIP Bridge
by Tim Meade 07 Apr '08

07 Apr '08

Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work!

7 15

Route outbound-only calls through XML dialplan?
by tuhl＠ix.netcom.com 07 Apr '08

07 Apr '08

I've been using FS for almost a year now as a call-blaster type of device (originating SIP calls and playing wav/mp3 when user answers). In the past I was simply sending all calls to a single SIP switch, but now I need to think about routing to several different SIP switches, based on NPA-NXX for example. The XML dialplan would seem perfect for this, but that's only triggered by an incoming call, right? I am not sure how to cause 'outbound-only' calls to go through it (since Freeswitch is originating the call, it only has 1 leg, if that makes sense). Currently I am originating via XML-RPC from a Perl script, using a command like 'originate <sip address> &playback(wav file)'. The final caveat is that we not use javascript to originate the call because I am still having problems with garbage collection during capacity testing (the box is pushing out 20cps, 30-second calls, 600 simultaneous calls). Thanks, Tom =============== tuhl(a)ix.netcom.com

2 1

DTMF detect dialplan example
by jonathan augenstine 04 Apr '08

04 Apr '08

Can someone provide an example of how to collect DTMF in the dialplan? My search through docs and google has not clarified the steps to accomplish this task. I need to play a prompt and collect DTMF, with the ability to cut-through the prompt to start collecting digits. Jonathan

2 1

Jingle To Sip Traffic
by Ali Jawad 04 Apr '08

04 Apr '08

Hi All We do have a jingle client and we do want to use that client to send traffic to sip devices, I have read in the FAQ that this is possible, has anyone done it ? Any advantages/disadvantages ? Any special issues I should be aware of ? Any good howto's ? Thx

2 1

My FS + Dingaling Experience With Qs
by Ali Jawad 04 Apr '08

04 Apr '08

Hi All I am trying to test the jingle features of FS. I have compiled and setup FS on debian all went well and I made a call using softphones from 1000 to 1001. After that I read http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working I did the changes and rebuilt I edited my config files my ./conf/jingle_profiles/client.xml looks as follows <include>  <x-profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="thisisme(a)gmail.com/talk"/> <param name="password" value="thisIsMyPwd"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/>  <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to join my conference"/> <param name="exten" value="1000"/> <param name="ext-rtp-ip" value="stun:stun.xten.com"/>  <param name="sasl" value="plain"/>    <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="888"/>    <param name="vad" value="both"/>  </x-profile> </include> As you can see above I used <param name="exten" value="1000"/> after that I logged in with my SIP softphone "Eyebeam" and tried to make a call to gmail account that did not work..but I was able to make a conference call to conf+888(a)conference.freeswitch.org Please note that my clients and server are both running in the same private LAN, I can put the server on a public IP if necessary. So what must I do/change/add/check to able to make calls to my gmail accounts (or from my gmail account to my sip phone) as mentioned in the FAQ Q: What? Did you say it can talk to GoogleTalk? Yes in March of 2006 I developed my own XMPP telephony signaling library that is capable of communicating with Google's GoogleTalk. With a single Jabber account you can receive endless simultaneous calls from GoogleTalk clients and gateway those calls to IVR or another voice protocol like SIP or H.323. When FreeSWITCH is on both ends of the call you can bypass NAT and send extended data such as Caller ID and DNIS.

2 1

Added close method to spidermonkey_odbc
by Chris Danielson 04 Apr '08

04 Apr '08

I found that some of my long running JavaScript scripts were holding connections open for way too long. I debugged and valgrind'd this change in order to make sure that we do not have any memory leaks. I'd appreciate a second opinion. http://jira.freeswitch.org/browse/MODLANG-55 Kind Regards, Chris Danielson Index: mod_spidermonkey_odbc.c =================================================================== --- mod_spidermonkey_odbc.c (revision 8009) +++ mod_spidermonkey_odbc.c (working copy) @@ -79,15 +79,15 @@ static void destroy_odbc_obj(odbc_obj_t ** objp) { odbc_obj_t *obj = *objp; - + if (obj == NULL) return; if (obj->stmt) { SQLFreeHandle(SQL_HANDLE_STMT, obj->stmt); } - if (obj->handle) { - switch_odbc_handle_destroy(&obj->handle); - } - switch_safe_free(obj->colbuf); - switch_safe_free(obj->code); + if (obj->handle) { + switch_odbc_handle_destroy(&obj->handle); + } + switch_safe_free(obj->colbuf); + switch_safe_free(obj->code); switch_safe_free(obj); } @@ -156,11 +156,12 @@ static void odbc_destroy(JSContext * cx, JSObject * obj) { - odbc_obj_t *odbc_obj = (odbc_obj_t *) JS_GetPrivate(cx, obj); - + if (obj == NULL) return; + odbc_obj_t *odbc_obj = (odbc_obj_t *) JS_GetPrivate(cx, obj); if (odbc_obj) { destroy_odbc_obj(&odbc_obj); - } + JS_SetPrivate(cx, obj, NULL); + } } static JSBool odbc_connect(JSContext * cx, JSObject * obj, uintN argc, jsval *argv, jsval *rval) @@ -405,6 +406,12 @@ } +static JSBool odbc_close(JSContext * cx, JSObject * obj, uintN argc, jsval *argv, jsval *rval) +{ + odbc_destroy(cx, obj); + return JS_TRUE; +} + enum odbc_tinyid { odbc_NAME }; @@ -417,6 +424,7 @@ {"numRows", odbc_num_rows, 1}, {"nextRow", odbc_next_row, 1}, {"getData", odbc_get_data, 1}, + {"close", odbc_close, 1}, {0} };

1 0

mod_radius_cdr error
by Santiago Gimenez Ocano 03 Apr '08

03 Apr '08

Hello all, I'm trying to use mod_radius_cdr in order to do accounting. I have follwed the wiki tutorial, the module is loading ok but when I try to make a call I got this error and the call is not been make. 2008-04-03 17:29:06 [ERR] mod_radius_cdr.c:172 my_on_ring() [mod_radius_cdr] Failed setting Acct-Status-Type: Start In addition, when I hung up I get this error. 2008-04-03 17:36:54 [ERR] mod_radius_cdr.c:417 my_on_hangup() failed adding Acct-Session-ID: 1dd9dc8c-8677-4581-8b1d-85d718a47dc6 There is something bad with the configuration? How can I solve this issue? Thanks in advance. Santiago _________________________________________________________________ Do more with your photos with Windows Live Photo Gallery. http://www.windowslive.com/share.html?ocid=TXT_TAGLM_Wave2_photos_022008

1 0

Call Not Found Error
by Ali Jawad 03 Apr '08

03 Apr '08

Hi All I am able to register my two x-lite clients to freeswitch using the accounts 1000 and 1001. However when I make calls to the account itself or from one account to another I get an error 404 Call not found. The same happens when I try the 9999 number as mentioned in the wiki. When I also have a lot of the following error in my logs Server IP: 192.168.0.229 My IP : 192.168.0.149 I don't know where these IPs came from a=alt:2 2 : mwIxNtCK CQevGxDD 192.168.111.1 3430 a=alt:3 1 : ntW8FIVH RJp+fyhg 192.168.238.1 3430 Here is the log for a complete call thx. http://pastebin.ca/969100

2 1

How to call a person on the same network.
by Ritesh Singh 03 Apr '08

03 Apr '08

Hi All, My aim is to call a person who's ip address is x.x.x.x. My ip address is y.y.y.y. What i tried was: Run the freeswitch on m/c having ips x.x.x.x and y.y.y.y. In both the freeswitch instance, i have loaded the mod_portaudio and mod_spidermonkey. Now i used commands from x.x.x.x freeswitch console: 1) pa call sip:SIP_URI@y.y.y.y. 2) pa call sip:y.y.y.y. And in both the cases i was unsuccessfull in calling y.y.y.y from x.x.x.x. Basically i was thinking that extension SIP_URI in default.xml will work with ip addresses only. In my scenario, x.x.x.x and y.y.y.y are on the same network. Please suggest. Thanks and Regards

2 1