[Freeswitch-users] park on hook
Hi, Is there anyway to detect when a channel is park in a way that is similar to hangup-hook or answer-hook? I would like to detect that inside a custom mod, without using the event mechanism? woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] problem with mod_xml_odbc
Hi, I am having problem trying to use mod_xml_odbc using freeswitch-1.0.5pre. Here is the error I am getting: 2009-11-15 00:17:23.571293 [INFO] mod_xml_odbc.c:647 XML ODBC module loading... 2009-11-15 00:17:23.571354 [NOTICE] mod_xml_odbc.c:563 Binding XML Search Function [directory] 2009-11-15 00:17:23.572299 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-11-15 00:17:23.572361 [CRIT] mod_xml_odbc.c:617 Cannot Open ODBC Database! 2009-11-15 00:17:23.572397 [ERR] mod_xml_odbc.c:650 Unable to load xml_odbc config file 2009-11-15 00:17:23.572424 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **Module load routine returned an error** In my config, I have: settings param name=binding value=directory/ param name=odbc-dsn value=myodbc:root:123456/ param name=debug value=true/ param name=keep-files-around value=true/ /settings [r...@localhost autoload_configs]# isql myodbc -v +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL How can I fix this problem? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] check sip client availability
Hi, Is there any API to tell freeswitch to send a SIP OPTION message to check the availability of a SIP client? thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] unable to configure Digium TDM400P
Hi, I am trying to setup a Digium TDM400P following the instruction on the wiki. It is a 1 fxo and 1 fxs card, so I tried loadzone=in defaultzone=in fxsks=2 fxoks=1 and loadzone=in defaultzone=in fxsks=1 fxoks=2 None works. Does anyone know how it should be configured? Here is what I get by following the wiki. [r...@localhost zaptel]# ztcfg -vv Zaptel Version: SVN-branch-1.4-r4629M Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) 2 channels to configure. Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart [r...@localhost zaptel]# ztcfg -vv Zaptel Version: SVN-branch-1.4-r4629M Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels to configure. Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart [r...@localhost zaptel]# lspci 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] unable to configure Digium TDM400P
Hi Michael Is the ztcfg output supposed to say something like 2 channels configured? I have not set up openzap yet because I don't know if the openzap.config is good or not. Can you give me any suggestion if the ztcfg output I am getting is proper or not? thx, woody -MC On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson woodydick...@gmail.comwrote: Hi, I am trying to setup a Digium TDM400P following the instruction on the wiki. It is a 1 fxo and 1 fxs card, so I tried loadzone=in defaultzone=in fxsks=2 fxoks=1 and loadzone=in defaultzone=in fxsks=1 fxoks=2 None works. Does anyone know how it should be configured? Here is what I get by following the wiki. [r...@localhost zaptel]# ztcfg -vv Zaptel Version: SVN-branch-1.4-r4629M Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) 2 channels to configure. Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart [r...@localhost zaptel]# ztcfg -vv Zaptel Version: SVN-branch-1.4-r4629M Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels to configure. Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart [r...@localhost zaptel]# lspci 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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] problem with compiling freeswith
Hi, Is this just me who is having this problem? I can't compile the latest freeswitch source code and here is the error: checking for gcc option to accept ANSI C... none needed checking for style of include used by make... GNU checking dependency style of gcc... gcc3 checking whether gcc and cc understand -c and -o together... yes ./configure: line 3377: syntax error near unexpected token `echo' ./configure: line 3377: `echo $as_me:$LINENO: checking for a BSD-compatible install 5' configure: error: /bin/sh './configure.gnu' failed for libs/tiff-3.8.2 [r...@localhost freeswitch-snapshot]# Does anyone know why? woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] overriding conference preference
Hi, Is there anyway of using curl without having to setup a standalone http service? Is it possible to generate curl xml using scripts? woody On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris m...@jerris.com wrote: On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote: Is is possible to override any of the setting specified in the conference profile? Just the flags you can pass per user such as pin and mute What I want to do is to have a default profile, and be able to modify certain fields if necessary in the dialplan. Alternatively, I would prefer to have a dynamic profile setting for the conference to obtain those parameters from odbc. you can do this with mod_xml_curl Mike ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] overriding conference preference
Hi, Is is possible to override any of the setting specified in the conference profile? What I want to do is to have a default profile, and be able to modify certain fields if necessary in the dialplan. Alternatively, I would prefer to have a dynamic profile setting for the conference to obtain those parameters from odbc. Is it possible? woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] problem with performance testing
Hi, I tried to performance test freeswitch with media proxy thur fs. With 400 cps, I start to see 2000 channels remaining in Freeswitch, and then no read codec error starts to pop up. With only 1875 channels, how come freeswitch is complaining about no read codec? Also, I am using media_proxy = true, whey should it need a codec anyway? freeswi...@mycom.com 2009-09-14 20:59:16.777675 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. freeswi...@mycom.com show channels count API CALL [show(channels count)] output: 1875 total. freeswi...@mycom.com 2009-09-14 21:00:36.212767 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of these zombies and how can I fix it? uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 20:53:55,1252932835,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 20:53:57,1252932837,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 20:54:02,1252932842,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 20:54:05,1252932845,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 3af84a6c-02f2-44ae-8a4b-1c7940522005,outbound,2009-09-14 20:54:06,1252932846,sofia/external ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] possible sofia_contact bug
Thanks Brian, you are correct. My problem is solved. Thank you so much. On Sun, Sep 13, 2009 at 9:20 AM, Brian West br...@freeswitch.org wrote: Also I'm going to suspect you have removed the domain aliases from the profile. If you have then you can't just do sofia_contact u...@domain... You must do sofia_contact profile/u...@domain since your hint for the domain is no longer on the profile. /b On Sep 12, 2009, at 10:25 AM, Anthony Minessale wrote: connect to sqlite directly with the sqlite3 binary and dump the record for that registration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] voicemail problem
Hi, While trying to record some sounds with the voicemail app, I keep getting message saying my record is below the minimal length even I was actually still speaking. Is it not detecting my voice? How can I configure it so that freeswitch's vm app can detect my speech? Thanks, woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] possible sofia_contact bug
Hi, I am pretty sure it should be u...@domain as I have used it before. Does anyone know why sofia_contact does not return correct result? thx, woody 2009/9/12 João Mesquita jmesqu...@freeswitch.org Just thinking out loud. Wouldn't be sofia_contact 180...@192.168.1.163 ? jmesquita On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson woodydick...@gmail.comwrote: Hi, I am having a strange problem here. sofia status shows that the user is registered, but sofia_contact says the user is not registered. Does anyone know why this is happening? freeswi...@localhost.localdomain sofia status profile internal reg 180004 API CALL [sofia(status profile internal reg 180004)] output: Registrations: = Call-ID:530339592782-1484696326...@192.168.1.163 User: 180...@192.168.1.102 Contact:180004 sip:180...@192.168.1.163:9000 Agent: Voip Phone 1.0 Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) Host: localhost.localdomain IP: 192.168.1.163 Port: 9000 Auth-User: 180004 Auth-Realm: 192.168.1.102 = freeswi...@localhost.localdomain sofia_contact 180...@192.168.1.102 API CALL [sofia_contact(180...@192.168.1.102)] output: error/user_not_registered freeswi...@localhost.localdomain freeswi...@localhost.localdomain sofia_contact user/180004 API CALL [sofia_contact(user/180004)] output: error/facility_not_subscribed ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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] possible sofia_contact bug
Hi, I am having a strange problem here. sofia status shows that the user is registered, but sofia_contact says the user is not registered. Does anyone know why this is happening? freeswi...@localhost.localdomain sofia status profile internal reg 180004 API CALL [sofia(status profile internal reg 180004)] output: Registrations: = Call-ID:530339592782-1484696326...@192.168.1.163 User: 180...@192.168.1.102 Contact:180004 sip:180...@192.168.1.163:9000 Agent: Voip Phone 1.0 Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) Host: localhost.localdomain IP: 192.168.1.163 Port: 9000 Auth-User: 180004 Auth-Realm: 192.168.1.102 = freeswi...@localhost.localdomain sofia_contact 180...@192.168.1.102 API CALL [sofia_contact(180...@192.168.1.102)] output: error/user_not_registered freeswi...@localhost.localdomain freeswi...@localhost.localdomain sofia_contact user/180004 API CALL [sofia_contact(user/180004)] output: error/facility_not_subscribed ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] auto expiration
Hi, I would like to set up freeswitch to automatically expire a user registration if either NOTIFY or REGISTER is not received within certain time frame. Does anyone know how to do that? Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] mod_limit and memcache
Hello, I read something that talks about using memcache for mod_limit before. Is it something that is available now? If I have multiple instances of freeswitch that need to share the same limit status, it there any existing solution? If no existing solution is available, what is the best way to go about modifying mod_limit to accomplish limiting for multiple freeswitch servers together? Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] zombie channels
Hi I checked and there is no looping in cdr. Also, only a very small percentage of the channels become zombie. What could cause fs to not releasing the channels? Also, it seems to happen on under high traffic. Could the fact that FS does not receive BYE or BYE timing out on the uac side may cause this problem? Thanks, woody On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, CS_REPORTING is the state in which cdrs are written, if the channel gets stuck in that state, the cdr module you are using is probably hanging somewhere. Use the freeswitch-gcore script in your source tree's scripts directory to generate a bug report for hanging channels. should be like.. cd /usr/src/freeswitch # or whatever your source tree is bash ./scripts/freeswitch-gcore bugreport.txt then submit it on http://jira.freeswitch.org/ so we can look at it. As you wish, you can also hop on #freeswitch / irc.freenode.net and have someone look into it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 21-Aug-09, at 5:54 AM, Woody Dickson wrote: Hi, I am running 1.0.4 right now using latest trunk. After a high traffic session, I do show channels, I would find a bunch of CS_HIBERNATE channels that don't get removed after all the traffic is gone. Does anyone know what is the case of thoes CS_HIBERNATE'd channels? How can I set a timeout for those channels to be removed? Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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] zombie channels
Hi, I am running 1.0.4 right now using latest trunk. After a high traffic session, I do show channels, I would find a bunch of CS_HIBERNATE channels that don't get removed after all the traffic is gone. Does anyone know what is the case of thoes CS_HIBERNATE'd channels? How can I set a timeout for those channels to be removed? Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] zombie channels
Hello, Yes, I am using cdr, so I guess CS_REPORTING could be a problem. I tried running the core, but I am getting some errors: ./freeswitch-gcore /usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced command file: ptrace: No such process. gcore: failed to create /usr/local/freeswitch/log/freeswitch.gcore.16240 What is the proper way of using freeswitch-gcore? Thanks, Woody On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, CS_REPORTING is the state in which cdrs are written, if the channel gets stuck in that state, the cdr module you are using is probably hanging somewhere. Use the freeswitch-gcore script in your source tree's scripts directory to generate a bug report for hanging channels. should be like.. cd /usr/src/freeswitch # or whatever your source tree is bash ./scripts/freeswitch-gcore bugreport.txt then submit it on http://jira.freeswitch.org/ so we can look at it. As you wish, you can also hop on #freeswitch / irc.freenode.net and have someone look into it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 21-Aug-09, at 5:54 AM, Woody Dickson wrote: Hi, I am running 1.0.4 right now using latest trunk. After a high traffic session, I do show channels, I would find a bunch of CS_HIBERNATE channels that don't get removed after all the traffic is gone. Does anyone know what is the case of thoes CS_HIBERNATE'd channels? How can I set a timeout for those channels to be removed? Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to set different action for different cause code
Hi,
I have my dialplan to do some simple routing. What I need to do is when
certain hangup code is received, route advance to the next or the route
after next based on the hangup code received.
So, I have:
condition
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=false/
action application=bridge data=sofia/internal/a...@123.3.3./
action application=bridge data=sofia/internal/b...@123.3.3./
action application=bridge data=sofia/internal/c...@123.3.3./
action application=bridge data=sofia/internal/d...@123.3.3./
/condition
What I want to do is that based on the error code received from ( bridge to
aaa), it can route to either bbb or ccc depends on the hangup code received
from aaa.
So there anyway to do it in the dialplan without any scripting?
thannks,
woody
On Mon, Aug 17, 2009 at 11:59 PM, Michael Collins m...@freeswitch.orgwrote:
On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson woodydick...@gmail.comwrote:
Hello,
I find hangup_hook, but I would like to define different actions for
different hangup codes. Is there anyway to do that?
I can think of at least two ways you could do this: one that uses only the
dialplan and one that uses a script. If you don't mind using a scripting
language then you can make it very clean:
action application=set data=api_hangup_hook=luarun chancleanup.lua
${hangup_cause}/
Then have your Lua script handle all the if-then-else or case stuff.
Question: are you trying to transfer the a-leg to some other destination if
the b-leg hangup is a specific cause, or are you just doing some external
cleanup stuff? Just curious...
-MC
Woody
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[Freeswitch-users] how to set different action for different cause code
Hello, I find hangup_hook, but I would like to define different actions for different hangup codes. Is there anyway to do that? Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Question about using switch_caller_extension_add_application
Hi, I want to implement a module where freeSWITCH would try to bridge to an extension and if the bridging operation fails, my module can use the hangup code to determine the next cause of action. With switch_caller_extension_add_application(session, extension, bridge, sofia/gateway/mygw/1232323);, if there is an error ( 503 received for instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the module's APP) and go on to the next action. Is there anyway to control it so that freeSWITCH would remain to be within the module's APP funtion and continue executing the code after switch_call_extension_add_application, when let's say a 4XX or 5XX or CANCEL ( from originator) is received? I have tried it and found that if the bridging is successful, freeSWITCH would continue executing the code after switch_caller_extension_add_application, but if an error is received, then it would just move on to the next action. Does anyone know how to deal with this problem? Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question about using switch_caller_extension_add_application
Hi, In my module, I will collect a list of available failover route that I can use to failover to whenever a particular error is received. However, these available routes has different condition and the condition changes every half a minute. Therefore, I need to catch the hangup cause after bridge, and then figure out the next workable available route based on the latest condition setting. It seems like this is only prossible to be done within a C module. Any suggestion will be greatly appreciated. Woody On Thu, Aug 6, 2009 at 8:36 AM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, You can set the continue_on_fail variable to true (or to the hangup causes you want it to ignore) and it'll keep executing whats queued. For receive_message, unless you hook the session thats being created as a B-leg, you won't get anything relevant. Also set hangup_after_bridge=true if you want to stop failing over when it worked. Im curious, what are you coding? you can transfer the call in the dialplan without having to do all this manual queuing in C, thats why the routing state and dialplan modules exist. If you need to pull data from somewhere you can fill in channel variables that you can reference in the dialplan. /*! \brief Transfer an existing session to another location \param session the session to transfer \param extension the new extension \param dialplan the new dialplan (OPTIONAL, may be NULL) \param context the new context (OPTIONAL, may be NULL) */ SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_ switch_core_session_t *session, const char *extension, const char *dialplan, const char *context); Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson: Hi, The problem is that I need freeswitch to continue executing the code after switch_status_t channel_receive_message even when it gets error SIP code from the destination. Is that possible? I know if I set up another action after my module in the dialplan.xml, I can catch that. But I would like the code to execute within the route that I have. Is that doable? Woody On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene mrene_li...@avgs.ca wrote: The hangup cause will be in the originate_disposition channel variable on the A-leg. sip_term_status will contain the sip code and proto_specific_hangup_cause will contain sip:code. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca Am 5-Aug-09 um 11:23 AM schrieb João Mesquita: My guess is that you will receive a message here: switch_status_t channel_receive_message(switch_core_session_t *session, switch_core_session_message_t *msg) The problem here is that you don't have the exact SIP code but there is a clear relationship between the codes and the messages you receive on the channel, so I am guessing that is all the same. Hope this helps. jmesquita On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson woodydick...@gmail.comwrote: Hi, I want to implement a module where freeSWITCH would try to bridge to an extension and if the bridging operation fails, my module can use the hangup code to determine the next cause of action. With switch_caller_extension_add_application(session, extension, bridge, sofia/gateway/mygw/1232323);, if there is an error ( 503 received for instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the module's APP) and go on to the next action. Is there anyway to control it so that freeSWITCH would remain to be within the module's APP funtion and continue executing the code after switch_call_extension_add_application, when let's say a 4XX or 5XX or CANCEL ( from originator) is received? I have tried it and found that if the bridging is successful, freeSWITCH would continue executing the code after switch_caller_extension_add_application, but if an error is received, then it would just move on to the next action. Does anyone know how to deal with this problem? Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http
[Freeswitch-users] Can FreeSWITCH send and receive SIP MESSAGE
Hi, I would like to use freeswitch as a gateway for sending and receiving short message. Does Freeswitch have the capability to send and recevie SIP MESSAGE? How can I set it up? I can't find any document on how to use Freeswitch for text message. Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Set problem in dialplan
Hello,
I am getting a strange problem in my dialplan.
After doing SET, I want to use it in the next condition field. But then
the value is not being set properly.
Could someone please tell me what is wrong?
Thanks,
Woody
Here is the dialplan:
context name=conf-execution
extension name=get-pin continue=true
condition field=${destination_number} expression=^(.*)$ break=never
action application=set data=conference_id=111/
action application=set data=is_moderator=true/
action application=info/
/condition
/extension
extension name=conf
condition field=${is_moderator} expression=^true$ break=never
action application=conference data=${conference_...@default
+flags{Moderator}+1234/
/condition
condition field=${is_moderator} expression=^false$ break=never
action application=conference data=${conference_...@default/
/condition
condition field=${is_moderator} expression=^$ break=always
action application=playback data=/var/app/prompt/wav/bye.wav/
action application=hangup/
/condition
/extension
/context
Here is the FS log.
Dialplan: sofia/internal/1...@192.168.1.101 parsing
[conf-execution-get-pin] continue=true
Dialplan: sofia/internal/1...@192.168.1.101 Regex (PASS) [get-pin]
${destination_number}(117) =~ /^(.*)$/ break=never
Dialplan: sofia/internal/1...@192.168.1.101 Action set(conference_id=111)
Dialplan: sofia/internal/1...@192.168.1.101 Action set(is_moderator=true)
Dialplan: sofia/internal/1...@192.168.1.101 Action info()
Dialplan: sofia/internal/1...@192.168.1.101 parsing [conf-execution-conf]
continue=false
Dialplan: sofia/internal/1...@192.168.1.101 Regex (FAIL) [conf]
${is_moderator}() =~ /^true$/ break=never
Dialplan: sofia/internal/1...@192.168.1.101 Regex (FAIL) [conf]
${is_moderator}() =~ /^false$/ break=never
Dialplan: sofia/internal/1...@192.168.1.101 Regex (PASS) [conf]
${is_moderator}() =~ /^$/ break=always
Dialplan: sofia/internal/1...@192.168.1.101 Action
playback(/var/app/prompt/wav/bye.wav)
Dialplan: sofia/internal/1...@192.168.1.101 Action hangup()
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[Freeswitch-users] More than one profile
Hello In the wiki, it is suggested that more than one profile should be used if libsofia is the bottleneck. When using multiple profiles to handle incoming call and each profile having an unique port, what is the best way to redirect and distribute incoming traffic? Is there any mod in Freeswitch that can do that? For instance, I may have internal1( port 5070) and internal2 (port 5080). External incoming traffic hitting 5060 will need to be distributed to one of the two profiles. How should that be done? Regards, Woody ___ Freeswitch-users mailing list Freeswitch-users@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] dynamically add ip to an ACL
Hi, Is it possible to dynamically add entries to an ACL without having to go through the xml file? Can it be done via command line or api? Thanks, Woody ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No-media problem with opensips-freeswitch setup
Hi
My external.xml is just the default configuration:
param name=debug value=0/
param name=sip-trace value=no/
param name=rfc2833-pt value=101/
param name=sip-port value=$${external_sip_port}/
param name=dialplan value=XML/
param name=context value=public/
param name=dtmf-duration value=100/
param name=codec-prefs value=$${outbound_codec_prefs}/
param name=hold-music value=$${hold_music}/
param name=use-rtp-timer value=true/
param name=rtp-timer-name value=soft/
param name=manage-presence value=false/
param name=inbound-codec-negotiation value=generous/
param name=nonce-ttl value=60/
param name=auth-calls value=false/
param name=rtp-timeout-sec value=1800/
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=rtp-timeout-sec value=300/
param name=rtp-hold-timeout-sec value=1800/
In my opensips.cfg, all the nated traffic is sent to the
external_sip_ip and external_rtp_port.
Is there anything I should add or change to enable media for device behind nat?
Regards,
Woody
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[Freeswitch-users] No-media problem with opensips-freeswitch setup
Hi, I tried to configure opensips as sip proxy and sip registrars and freeswitch as B2BUA. Everything works until I start to connect sip clients that are behind ADSL. Both freeswitch and opensips are on public IP and I am using external profile as well. Does anyone have experience in setting up opensips and freeswitch together and can share the configuration with me? Thank you very much in advance for any help. Regards, Woody ___ Freeswitch-users mailing list Freeswitch-users@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] Seeking opinion on shared disk space
Hi, In my deployment scenario, I plan to have two redundant freeswitch servers running on two different boxes. Two key features I am leveraging on freeswitch are voicemail and call recording and playback., and as a result of that, a shared storage for playback of the recorded wav files is needed. When the user traffic is high, I am affraid that NFS or even GFS can't scale well. On the other hand, a real SAN hardware with optical-fabric is too expensive for us. We are therefore considering using iSCSI SAN to build a cheap SAN for that purpose. Does anyone have experience setting up a shared storage between multiple freeswitch servers and can share some inputs with me? Thanks for all your help. Woody ___ Freeswitch-users mailing list Freeswitch-users@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] Lua script directory
Hi, Is it possible to change the directory where freeswitch looks for .lua scripts? I would like to place the lua scripts in the shared drive so multiple freeswitch can refer to it. Thanks, Woody ___ Freeswitch-users mailing list Freeswitch-users@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] Question about wrapping libfreeswitch
Hi,
I am sorry again for sending another email to the group again. I am working
on embedding libfreeswitch to provide better monitoring. The first thing I
attempt to do is to run the sample code provided in the website:
#include switch.h
int main(int argc, char **argv)
{
switch_core_flag_t flags = SCF_USE_SQL;
int nc=0; /* this is for 'no console' mode, FALSE console is there, TRUE
it isnt */
const char **err = NULL; /* error value for return from freeswitch
initialization */
#define LOGFILE freeswitch.log
static char *lfile = LOGFILE; /* if NULL no logfile is generated */
switch_core_init_and_modload(*lfile,flags,err);
switch_core_runtime_loop(nc);
switch_core_destroy();
return (0); /* per C89 spec */
}
But this code gives me segmentation fault when executing it. This piece of
code is supposed to start up freeswitch and run it is a loop. Does anyone
see what is wrong with it? Does anyone have any working example that I can
refer to?
Thanks,
Woody
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[Freeswitch-users] libfreeswitch question
Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling: [EMAIL PROTECTED] fs]# gcc switchnode.c -I/usr/local/freeswitch/include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread switchnode.c: In function 'main': switchnode.c:11: warning: passing argument 1 of 'switch_core_init_and_modload' makes integer from pointer without a cast switchnode.c:11: warning: passing argument 3 of 'switch_core_init_and_modload' from incompatible pointer type /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `clock_gettime' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `uuid_generate' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r' collect2: ld returned 1 exit status [EMAIL PROTECTED] fs]# Does anyone know which library is missing? Thanks, Woody ___ Freeswitch-users mailing list Freeswitch-users@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] Freeswitch hangs up after 30 s when using Record-Route
Hi I am using Openser as the sip proxy in front of freeswitch. When using Record-Route, Freeswitch hangs of every call after 30 s. 277.32.22.33:5060 is the public ip of openser and 192.168.1.101:5800 is freeswitch's external profile port. Both openser and freeswitch are within the same box. In the console, I am getting: 2008-11-25 01:31:24 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/[EMAIL PROTECTED] entering state [terminating] [EMAIL PROTECTED] [EMAIL PROTECTED] 2008-11-25 01:31:24 [DEBUG] sofia.c:2318 sofia_handle_sip_i_state() Channel sofia/external/[EMAIL PROTECTED] state [terminated] Here is the sip trace: U 192.168.1.101:5800 - 277.32.22.33:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.101 ;branch=z9hG4bK022f.a4e32a57.0;received=277.32.22.33. Via: SIP/2.0/UDP 192.168.1.102:12334;received=121.15.98.134 ;branch=z9hG4bK-d87543-a5439229f1204a4e-1--d87543-;rport=14392. Record-Route: sip:192.168.1.101;lr=on;ftag=c947d86b. From: 1000 sip:[EMAIL PROTECTED];tag=c947d86b. To: 0 sip:[EMAIL PROTECTED];tag=4Uve20t8p31Ba. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 2 INVITE. Contact: sip:[EMAIL PROTECTED]:5800;transport=udp. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk. Session-Expires: 120;refresher=uas. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 268. . v=0. o=FreeSWITCH 6527595211019529703 806853432324137362 IN IP4 277.32.22.33. s=FreeSWITCH. c=IN IP4 277.32.22.33. t=0 0. m=audio 11046 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 277.32.22.33:5800 - 192.168.1.101:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.101 ;branch=z9hG4bK022f.a4e32a57.0;received=277.32.22.33. Via: SIP/2.0/UDP 192.168.1.102:12334;received=121.15.98.134 ;branch=z9hG4bK-d87543-a5439229f1204a4e-1--d87543-;rport=14392. Record-Route: sip:192.168.1.101;lr=on;ftag=c947d86b. From: 1000 sip:[EMAIL PROTECTED];tag=c947d86b. To: 0 sip:[EMAIL PROTECTED];tag=4Uve20t8p31Ba. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 2 INVITE. Contact: sip:[EMAIL PROTECTED]:5800;transport=udp. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk. Session-Expires: 120;refresher=uas. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 268. . v=0. o=FreeSWITCH 6527595211019529703 806853432324137362 IN IP4 277.32.22.33. s=FreeSWITCH. c=IN IP4 277.32.22.33. t=0 0. m=audio 11046 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 192.168.1.101:5800 - 277.32.22.33:5060 BYE sip:[EMAIL PROTECTED]:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: sip:192.168.1.101;lr=on;ftag=c947d86b. Max-Forwards: 70. From: 0 sip:[EMAIL PROTECTED];tag=4Uve20t8p31Ba. To: 1000 sip:[EMAIL PROTECTED];tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: sip:[EMAIL PROTECTED]:5800;transport=udp. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text=ACK Timeout. Content-Length: 0. . U 277.32.22.33:5800 - 192.168.1.101:5060 BYE sip:[EMAIL PROTECTED]:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: sip:192.168.1.101;lr=on;ftag=c947d86b. Max-Forwards: 70. From: 0 sip:[EMAIL PROTECTED];tag=4Uve20t8p31Ba. To: 1000 sip:[EMAIL PROTECTED];tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: sip:[EMAIL PROTECTED]:5800;transport=udp. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text=ACK Timeout. Content-Length: 0. . U 192.168.1.101:5800 - 277.32.22.33:5060 BYE sip:[EMAIL PROTECTED]:14392 SIP/2.0. Via: SIP/2.0/UDP 277.32.22.33:5800;rport;branch=z9hG4bK10ttgjpr9KeQg. Route: sip:192.168.1.101;lr=on;ftag=c947d86b. Max-Forwards: 70. From: 0 sip:[EMAIL PROTECTED];tag=4Uve20t8p31Ba. To: 1000 sip:[EMAIL PROTECTED];tag=c947d86b. Call-ID: NmZkYzU5MGYzZjBhZGM5YjQ0MjhiNzlmMDc5NzNjNjI.. CSeq: 107655293 BYE. Contact: sip:[EMAIL PROTECTED]:5800;transport=udp. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10454M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported:
[Freeswitch-users] playback_terminator for phrase
Hi, I have my dialplan set as follows: action application=set data=playback_terminators=# / action application=phrase data=vm_count,4:new:9:old / But the playback terminators does not work when the phrase is being played. Is the the wrong way of specifying the terminator key for phrase? Thanks, Woody ___ Freeswitch-users mailing list Freeswitch-users@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] Question about FIFO event
Hi, I am trying to write an event listener that can record the time when a FIFO consumer rejoins the queue after the caller hangs up. Tracing through all the event traffic, I notice that there is FIFO-Action= consumer_start, consumer_stop, and consumer_pop, but there isn't one that indicates that the consumer rejoins the queue after the caller hangs up. Is there anyway I can detect this particular event? Thanks for your help. Woody ___ Freeswitch-users mailing list Freeswitch-users@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] avoiding moh for meta_app
Hi,
I tried to avoid Freeswitch from playing moh when the meta_app is executed
by setting hold_music=silence:
condition field=destination_number expression=^(.*)$
action application=set data=hold_music=silence /
action application=set data=call_timeout=120 /
action application=set data=hangup_after_bridge=true /
action application=set data=language=zh /
action application=set data=ringback=$${us-ring} /
action application=bind_meta_app data=1 a s execute_extension::record
XML features/
action application=bind_meta_app data=2 a a
execute_extension::stoprecord XML features/
action application=bind_meta_app data=3 a s execute_extension::att_xfer
XML features/
action application=bridge data=sofia/gateway/openser/1000 /
action application=phrase data=user_not_reg /
However, when #1 is pressed, Freeswitch still attempts to play moh:
92.168.1.101 Processing meta digit '1' [execute_extension::record XML
features]
2008-11-07 05:18:30 [DEBUG] switch_core_session.c:616
switch_core_session_queue_private_event()
Kill sofia/internal/[EMAIL PROTECTED] [BREAK]
2008-11-07 05:18:30 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Kill
sofia/internal/[EMAIL PROTECTED]
2008-11-07 05:18:30 [DEBUG] switch_core_session.c:616
switch_core_session_queue_private_event() Kill sofia/external/1000 [BREAK]
2008-11-07 05:18:30 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message()
Kill sofia/external/1000 [BREAK]
2008-11-07 05:18:30 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event()
sofia/external/1000 Command Execute playback(local_stream://moh)
2008-11-07 05:18:30 [DEBUG] mod_local_stream.c:320 local_stream_file_open()
Opening Stream [moh/8000] 8000hz
How can I fix this problem?
Thanks alot in advance for your help.
Woody
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Re: [Freeswitch-users] Multiple actions for bind_meta_data app
Hi Anthony, Thanks for your reply. It solves the problem. Just one more thing. Is it possible to have Freeswitch not to play moh when *1 is pressed? In the following log, the moh is always played after *1 is pressed and before the execution of the new extension: 2008-10-26 18:21:22 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf() sofia/internal/[EMAIL PROTECTED] Processing meta digit '1' [execute_extension::record XML features] 2008-10-26 18:21:22 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/internal/ [EMAIL PROTECTED] [BREAK] 2008-10-26 18:21:22 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-26 18:21:22 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/external/1000 [BREAK] 2008-10-26 18:21:22 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/1000 [BREAK] 2008-10-26 18:21:22 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/external/1000 Command Execute playback(local_stream://moh) 2008-10-26 18:21:22 [DEBUG] mod_local_stream.c:320 local_stream_file_open() Opening Stream [moh/8000] 8000hz 2008-10-26 18:21:22 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 30ms 2008-10-26 18:21:22 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/1000 [BREAK] 2008-10-26 18:21:22 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/internal/[EMAIL PROTECTED] Command Execute execute_extension(record XML features) Thanks for your great help. Woody On Sat, Oct 25, 2008 at 10:33 PM, Anthony Minessale [EMAIL PROTECTED] wrote: are you using record or record_session? The latter is a background record where the former is a foreground record. ___ Freeswitch-users mailing list Freeswitch-users@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] Multiple actions for bind_meta_data app
Hi,
I would like to know if the following scenario is possible with Freeswitch:
A connection is established between two endpoints ( soft phones). On one
end, someone presses *1, a BEEP that indicates the beginning of recording is
heard, and then Freeswitch starts to record. I can get recording to work
with the following line from wiki:
action application=bind_meta_app data=2 a s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
In order to generate the BEEP tone, I would need the bin_meta_app to execute
two actions, one to gnerate the BEEP and the other to recrod.
How can I change the dialplan to do that?
Thanks in advance for all your help.
Woody
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Re: [Freeswitch-users] Multiple actions for bind_meta_data app
Hi,
Thank you very much for your prompt response. I actually did try to put the
read app in a separate extension, but what happened was that when *1 was
pressed, Freeswitch would play MOH and both both soft phone couldn't hear
each other anymore. After MOH is played, Freeswitch just continues to place
it forever. What I need is for the record to take place at the background
without interrupting the existing call. That part can be accomplished by
using
action application=bind_meta_app data=2 a s
record_session::$${base_dir}/
recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
In addition to that, the party that presses *1 would hear a BEEP sound,
indicating the beginning of the recording.
I think the problem may due to a misconfigured dialplan, but I can't figure
out what's wrong.
Here is my dialplan:
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=dialplan description=Redirect to agent dialplan
context name=public
extension name=test9
condition field=destination_number expression=^(.*)$
action application=bind_meta_app data=3 a a execute_extension::transfer
XML features/
action application=bind_meta_app data=1 a a execute_extension::record
XML features/
action application=bind_meta_app data=2 a a
stop_record_session::/mnt/app/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
action application=bind_meta_app data=4 a s execute_extension::att_xfer
XML features/
action application=bridge data=sofia/gateway/mygateway/1000/
action application=hangup /
/condition
/extension
/context
context name=features
extension name=features
condition field=destination_number expression=^att_xfer
break=on-true
action application=read data=1 10 'tone_stream://%(1,0,350,440)'
digits 3 #/
action application=set data=action=att-xfer /
action application=execute_extension data=att${digits} XML
public/
/condition
condition field=destination_number expression=^record break=on-true
action application=record
data=/mnt/app/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
/condition
/extension
/context
/section
/document
The Freeswitch log shows that it plays a MOH after *1 is pressed, which is
not what I want:
2008-10-25 17:05:13 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
sofia/internal/[EMAIL PROTECTED] Processing meta digit '1'
[execute_extension::record XML features]
2008-10-25 17:05:13 [DEBUG] switch_core_session.c:616
switch_core_session_queue_private_event() Kill sofia/internal/
[EMAIL PROTECTED] [BREAK]
2008-10-25 17:05:13 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Kill
sofia/internal/[EMAIL PROTECTED]
2008-10-25 17:05:13 [DEBUG] switch_core_session.c:616
switch_core_session_queue_private_event() Kill sofia/external/1000 [BREAK]
2008-10-25 17:05:13 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Kill sofia/external/1000 [BREAK]
2008-10-25 17:05:13 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event()
sofia/external/1000 Command Execute playback(local_stream://moh)
2008-10-25 17:05:13 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event()
sofia/internal/[EMAIL PROTECTED] Command Execute execute_extension(record
XML features)
2008-10-25 17:05:13 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing
5454-record in context features
2008-10-25 17:05:13 [DEBUG] mod_local_stream.c:320 local_stream_file_open()
Opening Stream [moh/8000] 8000hz
2008-10-25 17:05:14 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file()
Codec Activated [EMAIL PROTECTED] 1 channels 30ms
2008-10-25 17:05:14 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Kill sofia/external/1000 [BREAK]
2008-10-25 17:05:14 [CONSOLE] mod_xml_curl.c:206 xml_url_fetch() XML
response is in /tmp/4384f1d4-b856-4817-8ac7-d6f9fb3dd866.tmp.xml
2008-10-25 17:05:14 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex:
[features] destination_number(record) =~ /^att_xfer/
2008-10-25 17:05:14 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex
mismatch
2008-10-25 17:05:14 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex:
[features] destination_number(record) =~ /^record/
2008-10-25 17:05:14 [NOTICE] switch_core_session.c:1219
switch_core_session_execute_exten() Execute
record(/mnt/app/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Thank you so much for all your help.
Woody
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Re: [Freeswitch-users] Setting the moh
Hi Brian, Thanks for your help. It works now, but I have another problem. The MOH that is being heard by the other side does not sound right. It sounded like the bit rate is not right or something. Freeswitch did send out Audio but xlite can't properly translate. Any idea what this may be due to? There are three rates specified ( 8000, 16000, and 32000), so how do I select which one to use? Here is the log: 2008-10-21 18:06:49 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf() sofia/internal/[EMAIL PROTECTED] Processing meta digit '4' [execute_extension::att_xfer XML features] 2008-10-21 18:06:49 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/internal/ [EMAIL PROTECTED] [BREAK] 2008-10-21 18:06:49 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-21 18:06:49 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/external/5003 [BREAK] 2008-10-21 18:06:49 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/5003 [BREAK] 2008-10-21 18:06:49 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/external/5003 Command Execute playback(local_stream://moh) 2008-10-21 18:06:49 [DEBUG] mod_local_stream.c:320 local_stream_file_open() Opening Stream [moh/8000] 8000hz 2008-10-21 18:06:49 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 30ms 2008-10-21 18:06:49 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/5003 [BREAK] 2008-10-21 18:06:49 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/internal/[EMAIL PROTECTED] Command Execute execute_extension(att_xfer XML features) Thanks, Woody ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Setting the moh
Hi Brian, What do you think could be the reason for the bad MOH sound quality? I am just thinking that it could be some setting issue. Thanks, Woody On Tue, Oct 21, 2008 at 10:22 AM, Brian West [EMAIL PROTECTED] wrote: in the defaults the rate is selected by the channel rate. You have no need to run a 16k hold music on an 8k channel. /b On Oct 20, 2008, at 9:15 PM, Woody Dickson wrote: There are three rates specified ( 8000, 16000, and 32000), so how do I select which one to use? ___ Freeswitch-users mailing list Freeswitch-users@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@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] Unable to build FS with odbc support
Hi, I tried to run ./configure --enable-core-odbc-support but got the following error during make: Compiling src/switch_pcm.c ... Compiling libs/libteletone/src/libteletone_detect.c ... Compiling libs/libteletone/src/libteletone_generate.c ... Compiling src/switch_odbc.c ... In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or directory ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or directory In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:66: error: expected declaration specifiers or '...' before 'SQLHSTMT' ./src/include/switch_odbc.h:69: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c:39: error: expected specifier-qualifier-list before 'SQLHENV' src/switch_odbc.c: In function 'switch_odbc_handle_new': src/switch_odbc.c:72: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:72: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:72: error: (Each undeclared identifier is reported only once src/switch_odbc.c:72: error: for each function it appears in.) src/switch_odbc.c:73: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': src/switch_odbc.c:92: error: 'switch_odbc_handle_t' has no member named 'state' cc1: warnings being treated as errors src/switch_odbc.c:93: warning: implicit declaration of function 'SQLDisconnect' src/switch_odbc.c:93: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:101: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_connect': src/switch_odbc.c:109: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:109: error: expected ';' before 'err' src/switch_odbc.c:112: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:112: error: expected ';' before 'valueLength' src/switch_odbc.c:115: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:115: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:116: warning: implicit declaration of function 'SQLAllocHandle' src/switch_odbc.c:116: error: 'SQL_HANDLE_ENV' undeclared (first use in this function) src/switch_odbc.c:116: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:118: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:118: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:123: warning: implicit declaration of function 'SQLSetEnvAttr' src/switch_odbc.c:123: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:123: error: 'SQL_ATTR_ODBC_VERSION' undeclared (first use in this function) src/switch_odbc.c:123: error: 'SQL_OV_ODBC3' undeclared (first use in this function) src/switch_odbc.c:127: warning: implicit declaration of function 'SQLFreeHandle' src/switch_odbc.c:127: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:131: error: 'SQL_HANDLE_DBC' undeclared (first use in this function) src/switch_odbc.c:131: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:131: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:135: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:138: warning: implicit declaration of function 'SQLSetConnectAttr' src/switch_odbc.c:138: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:138: error: 'SQL_LOGIN_TIMEOUT' undeclared (first use in this function) src/switch_odbc.c:138: error: 'SQLPOINTER' undeclared (first use in this function) src/switch_odbc.c:138: error: expected expression before ')' token src/switch_odbc.c:140: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c:148: warning: implicit declaration of function 'SQLConnect' src/switch_odbc.c:148: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:148: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:148: error: expected expression before ')' token src/switch_odbc.c:150: error: expected ';' before 'outstr' src/switch_odbc.c:151: error: expected ';' before 'outstrlen' src/switch_odbc.c:153: warning: implicit declaration of function 'SQLDriverConnect' src/switch_odbc.c:153: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:153: error: expected expression before ')' token src/switch_odbc.c:159: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c:163: warning: implicit declaration of function 'SQLGetDiagRec' src/switch_odbc.c:163: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:163: error: 'err' undeclared (first use in this function) src/switch_odbc.c:166:
Re: [Freeswitch-users] Storing voicemail in DB
Hi The error that I am getting is : 2008-08-28 19:04:41 [CRIT] switch_odbc.c:248 db_is_up() The sql server is not responding for DSN fsdb [STATE: 24000 CODE 0 ERROR: [unixODBC][MySQL][ODBC 3.51 Driver][mysqld-5.0.32-Debian_7etch6]Invalid cursor state I am sure fsdb is correct as I can do the following: isql fsdb fs password +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ Any idea why the DNS fsdb does not work within Freeswitch? Thanks, Woody On Wed, Aug 27, 2008 at 10:23 PM, Anthony Minessale [EMAIL PROTECTED] wrote: are you mixing modules.conf modules.conf.xml modules.conf in the dir you type make from MUST have mod_voicemail enabled. /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml is where you can comment out mod_voicemail so you can load it manually from the CLI once FS is started and see if there are any errors. be sure to press f8 first if you are upgrading from previous versions and having trouble, you may want to mv /usr/local/freeswitch /usr/local/freeswitch.bak and then do make current vm-sync from the build root. On Wed, Aug 27, 2008 at 3:56 AM, Woody Dickson [EMAIL PROTECTED]wrote: Hi, When typing in load mod_voicemail , I am getting the following error: 2008-08-28 01:05:42 [CRIT] switch_loadable_module.c:757 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_voicemail .so **/usr/local/freeswitch/mod/mod_voicemail .so: cannot open shared object file: No such file or directory** If I start Freeswitch with mod_voicemail loaded, I don't get that error. But the error that I did get when starting Freeswitch is the following: 2008-08-28 01:09:29 [NOTICE] switch_loadable_module.c:353 switch_loadable_module_process() Adding Chat interface 'sip' 2008-08-28 01:09:29 [NOTICE] switch_loadable_module.c:393 switch_loadable_module_process() Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1] 2008-08-28 01:09:30 [ERR] switch_core_sqldb.c:95 switch_core_db_persistant_execute_trans() SQL ERR [database is locked] I don't know where this error comes come. It seems to happen after I upgrade to the latest build. I still don't know how to fix it and what kind of impact it brings to the system. Is this the reason why Freeswitch does not store the voicemail detail in DB? Btw, I did configure odbc with ./configure --enable-core-odbc-support I would appreciate any help in getting my voicemail setting to work. Thanks, Woody On Mon, Aug 25, 2008 at 11:17 PM, Anthony Minessale [EMAIL PROTECTED] wrote: comment the line in your modules.conf.xml that loads mod_voicemail then start freeswitch then set the loglevel to debug by pressing f8 then enter load mod_voicemail into the cli the resulting text should clarify the problem. On Mon, Aug 25, 2008 at 9:27 AM, Brian West [EMAIL PROTECTED]wrote: Did you configure with odbc? ./configure --enable-core-odbc-support /b On Aug 25, 2008, at 3:36 AM, Woody Dickson wrote: Hi, I am new to Freeswitch. After playing with it for a couple of weeks, I have to say Freeswitch is GREAT! Right now I am kind of stuck on getting voicemail info to be stored in mysql. I have defined the following in voicemail.conf.xml: param name=odbc-dsn value=fsdb:fs:password/ And I am sure my odbc is ok, as I have validated it with isql: /usr/src/freeswitch-snapshot# isql fsdb fs password +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ Is there anything wrong with my setting? Tks, Woody ___ Freeswitch-users mailing list Freeswitch-users@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:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC
