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problem with mod_xml_odbc
by Woody Dickson 14 Nov '09

14 Nov '09

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> [root@localhost autoload_configs]# isql myodbc -v +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> How can I fix this problem? thanks, woody

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FreeSWITCH Now Supports Broadvoice BV16, BV32 Voice Codecs
by Dome Charoenyost 14 Nov '09

14 Nov '09

http://freeswitch.org/node/217 Very fast develop :) one reason why i love FS. another is good performance Dome C.

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Freeswitch configure error using --srcdir option
by Robert Hadley 14 Nov '09

14 Nov '09

Hello All On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the -srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. It doesn't work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. The steps I am doing are: 1. Building as root 2. Unzip freeswitch-1.0.4-tar.gz in /opt 3. cd into /opt/freeswitch-1.0.4 4. mkdir Debug 5. cd Debug 6. ../configure -srcdir=".." CFLAGS="-g -ggdb -O2" 7. After several seconds of configuring I get: === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp [root@roberth-c53 Debug]# The file that's executing is this: [root@roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu #! /bin/sh ./configure "$@" --disable-shared --with-pic Please tell me if I understood the -srcdir option correctly and if there is a way to do build in a different directory. Thanks, Robert

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Using play_and_get_digits (or IVR) without a final delay
by Robert L Mathews 14 Nov '09

14 Nov '09

Hi, I'm a new FreeSWITCH convert from asterisk. It's great software; thanks for making it. I'm trying to play a sound file while listening for possible digits dialed (although in most cases callers will not be dialing anything). If callers do start dialing an extension while the sound file is playing, I want them to be able to dial it slowly without any problems. So the inter-digit timeout should be, say, 2 seconds. However, if people don't start dialing anything while the sound file is playing, I don't want any delay at the end of it. I've tried play_and_get_digits with a 2000 timeout -- but that causes a 2 second "dead air" pause at the end of the sound file if callers don't dial anything. I've also tried using a trivial IVR menu to simulate this, but it has the same problem. Interestingly, it doesn't *look* like IVRs should have the problem, because they allow both "inter-digit-timeout" and "timeout" to be specified separately -- but at least in FreeSWITCH 1.0.4, a short "timeout" value always interrupts the dialing of digits, even if "inter-digit-timeout" is much longer. Is there any way to play a sound file from the dial plan with a long inter-digit delay, but without any final delay if no digits are dialed? Thanks for your time! -- Robert L Mathews, Tiger Technologies

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suggestions for hardware.
by Orien Love 13 Nov '09

13 Nov '09

First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!! I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions. Here is my plan, please let me know if it does not make sense, or if I am going about it System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server. System Design. Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ??? here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available? I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format? Thanks Orien

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How To Disable MD5 Authentication?
by Jerry Richards 13 Nov '09

13 Nov '09

How can I disable MD5 Authentication upon registration? Best Regards, Jerry

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SIP trunk without authentication
by Sergey Kobzar 13 Nov '09

13 Nov '09

Hello. I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP. I've searched through FS documentation and didn't find clear answer. Could you help me or maybe give a link to a doc which can help? Thanks. -- Sergey

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Re: [Freeswitch-users] suggestions for hardware.
by Orien Love 13 Nov '09

13 Nov '09

Thank you Dana and Michael for your replies, I am getting a spa3000 in the mail soon so I can try it out and see if it will work for my needs, I am going to implement a automatic attendant thanks to the information provided. Since I have not had any replies about the atom board I am guessing that nobody has used one, Could somebody tell me what is a good CPU speed / Memory / FSB be? I really do not have a large budget and cannot afford to buy something that will not work. Thanks again Orien

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Configuring freeswitch with voicepulse
by Paul Thirumalai 13 Nov '09

13 Nov '09

Hello All I am trying to configure freeswitch so that it sends outgoing calls to the PSTN through voicepulse My configuration is as follows. I created a file $PREFIX/conf/sip_profiles/external/voicepulse.xml <include>  <gateway name="voicepulse"> <param name="username" value="my-user"/> <param name="realm" value="sjc-primary.voicepulse.com"/> <param name="password" value="my-password"/> <param name="proxy" value="sjc-primary.voicepulse.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> <gateway name="voicepulse-backup"> <param name="username" value="my-user"/> <param name="realm" value="sjc-backup.voicepulse.com"/> <param name="password" value="my-password"/> <param name="extension" value="1000"/> <param name="proxy" value="sjc-backup.voicepulse.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include> I also have a dial plan defined as follows <extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>   <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> When I dial an external number using extension 1000 I get the following message on the CLI ] freeswitch@ubuntu> 2009-11-10 00:35:44.365614 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1000(a)74.207.249.79[e4301180-cdba-11de-a864-8927fe94a9f0] 2009-11-10 00:35:44.366623 [INFO] mod_dialplan_xml.c:315 Processing Paul->5555555555 in context default 2009-11-10 00:35:44.368645 [NOTICE] switch_channel.c:602 New Channel sofia/external/5555555555 [e43092f4-cdba-11de-a864-8927fe94a9f0] 2009-11-10 00:35:47.59221 [NOTICE] sofia_glue.c:2698 Pre-Answer sofia/external/5555555555! 2009-11-10 00:35:47.59221 [INFO] switch_ivr_originate.c:2017 Sending early media 2009-11-10 00:35:47.60524 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1257800805 1257800806 IN IP4 74.207.249.79 s=FreeSWITCH c=IN IP4 74.207.249.79 t=0 0 m=audio 30542 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 a=sendrecv 2009-11-10 00:35:47.60524 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1000(a)74.207.249.79! 2009-11-10 00:35:51.449542 [NOTICE] sofia.c:3849 Hangup sofia/external/5555555555 [CS_EXCHANGE_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-11-10 00:35:51.452539 [NOTICE] switch_ivr_bridge.c:419 Hangup sofia/internal/1000(a)74.207.249.79 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/internal/1000(a)74.207.249.79) Ended 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1000(a)74.207.249.79 [CS_DESTROY] 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/5555555555) Ended 2009-11-10 00:35:51.454125 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/5555555555 [CS_DESTROY] I am really new to VOIP and having a hard time with this. I am really not sure how to proceed. Any help would be really appreciated. Thanks Paul

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mod event socket
by srinivasula reddy 12 Nov '09

12 Nov '09

HI all, i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 port, when i am trying to connect freeswtich it it taking 20 seconds to get response from FS, can i able to reduce tcp response time? thanks Srinivasula Reddy K

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