September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error loading ODBC
by preetha 29 Sep '08

29 Sep '08

Hi, when i try to run a sample odbc code from freeswitch console like freeswitch@localhost> jsrun odbc.js I found the following error: API CALL [jsrun(odbc.js)] output: OK freeswitch@localhost> 2008-09-25 19:44:35 [ERR] mod_spidermonkey.c:3303 js_api_use() Error loading ODBC 2008-09-25 19:44:35 [ERR] odbc.js:11 mod_spidermonkey() ReferenceError: ODBC is not defined code is: use("ODBC"); var people = {}; people["authur"] = 30; people["ford"] = 42; people["trillian"] = 29; people["zaphod"] = 304; people["marvin"] = 1000001; db = new ODBC("freeswitchdb", "freeswitch", "1234"); db.connect(); db.exec("create table demo (name varchar(255), age varchar(80));"); for (var i in people) { sql = "insert into demo values('" + i + "', " + people[i] + ");"; console_log("debug", sql + "\n"); db.exec(sql); } sql = "select * from demo;"; var line = "==============================\n"; console_log("notice", line); db.exec(sql); console_log("debug", "row count: " + db.numRows() + "\n"); while(db.nextRow()) { var row = db.getData(); for (var i in row) { console_log("debug", "[" + i + "] = [" + row[i] + "]\n"); } console_log("debug", "\n"); } console_log("notice", line); what should i specify in DSN.here i gave freeswitchdb which is a database name.please assist me to solve this problem Thanks -- View this message in context: http://www.nabble.com/Error-loading-ODBC-tp19665456p19665456.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Bridge call directly to extension
by Gopal krishnan 29 Sep '08

29 Sep '08

Hi, Is there a possible way as like in Asterisk where the agents will login in queue, so that the established call will be directly transferred to the extension. Is there any module for that? Any help would be appreciated. Thanks -- Thank you with regards, Gopal,

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Load test - performance not even matching Asterisk
by Jon Bruel 28 Sep '08

28 Sep '08

I have made some load test, where an Asterisk server was controlled to make a high number of calls to a FreeSWITCH, which was registered on Asterisk. Each had its own server. The calls made to the FreeSWITCH were answered and echoed. The CPU load on the FreeSWITCH server approached 100 % at around 100 calls. Similar test with an Asterisk server would show 2-3 times less CPU load. Looping back to the Asterisk, thus avoiding a possible CPU-loading application (echo), did not show a significantly different result. The test has been done on a CentOS 4 and a FC8 with similar result. Any hints about how to increase the capacity to the level of several times the level of Asterisk - as reported by others? Are we possibly dealing with some hardware issues related to the NIC and its drivers. /Jon

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Multiple DID number with freeswitch
by Xtpljava 28 Sep '08

28 Sep '08

Hi, Can we configure multiple DID number with freeswitch? Thanks and regards, Vinod nigade Sent from my iPod

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 28 Sep '08

28 Sep '08

The protocol is SIP. The profiles relevant for the incoming call test with echo are here: Dialplan: *********  <include> <context name="public"> <extension name="From_Asterisk"> <condition field="destination_number" expression="^(K00003333100004444\d{3})$"> <action application="answer"/> <action application="echo"/> </condition> </extension> </context> </include> Sip Profiles: *************  <profile name="external">  <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="1"/> <param name="comfort-noise" value="false"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5080"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> (comment: using PCMA) <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>  </settings> </profile> Gateway (one of them): ********************** <include> </gateway> <gateway name="100004444300"> <param name="username" value="100004444300"/> <param name="realm" value="10.3.1.21"/> <param name="from-user" value="100004444300"/> <param name="from-domain" value="10.3.1.21"/> <param name="password" value="XXXXXX"/> <param name="extension" value="K00003333100004444300"/> <param name="proxy" value="10.3.1.21"/> <param name="expire-seconds" value="600"/> <param name="caller-id-in-from" value="false"/> </gateway> </include>

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Re gistering remote extensions.
by henkoegema 27 Sep '08

27 Sep '08

I want to register a telephone 2010 (a remote extension) to my FS server. My server has dynamic ip address, registered with DynDNS. (dyndns ip address is: xxxx.ftpaccess.cc) In the remote telephone I programmed: a. stun server=stun.freeswitch.org b. sip port=5090 (i replaced all the 5060 with 5090 at several places) c. rtp port=17000 d. domain= xxxx.ftpaccess.cc (my dyndns) The remote phone is also on a dynamic ip address. I followed the instructions on http://wiki.freeswitch.org/wiki/Example_Offsite_phones Step 1. I created an extension 2010 in /..../directory/default.xml ================================================= <include> user id="2010" mailbox="2010"> params> param name="password" value="2010"/> param name="vm-password" value="2010"/> /params> variables> variable name="accountcode" value="2010"/> variable name="user_context" value="default"/> variable name="effective_caller_id_name" value="Terje"/> variable name="effective_caller_id_number" value="2010"/> /variables> /user> </include> Setp 2: I created a new profile. called "doublenat" profile set to port 5090 in /..../sip_profiles/doublenat.xml ================================================================================== <profile name="doublenat"> gateways> X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> /gateways> settings> param name="debug" value="0"/> param name="sip-trace" value="no"/> param name="rfc2833-pt" value="101"/> param name="sip-port" value="5090"/> param name="dialplan" value="XML"/> param name="context" value="public"/> param name="dtmf-duration" value="100"/> param name="codec-prefs" value="$${outbound_codec_prefs}"/> param name="use-rtp-timer" value="true"/> param name="hold-music" value="$${hold_music}"/> param name="rtp-timer-name" value="soft"/> param name="manage-presence" value="false"/> param name="aggressive-nat-detection" value="true"/> param name="apply-nat-acl" value="rfc1918"/> 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"/> /settings> </profile> Step 3: Inside the /..../conf/dialplan/public.xml, I put ========================================= <extension name="public_extensions"> <condition field="destination_number" expression="^(20[01][0-9])$"> <action application="bridge" data="sofia/doublenat/$1%$${domain} "/> </condition> </extension> However the phone can't register. In the FS CLI window I get: 2008-09-24 20:10:39 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [2010(a)xxxx.ftpaccess.cc] You must define a domain called 'henkoegema.ftpaccess.cc' in your directory and add a user with the id="2010" attribute and you must configure your device to use the proper domain in it's authentication credentials. %-| This line repeats and repeats and repeats. Evenso the warning is clear, I'm not sure what exactly I should do. (and where) :confused: Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19655521.ht… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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IVR greet_long
by Jim Flowers 27 Sep '08

27 Sep '08

I'm getting there but still don't understand some things: 1. There are variables in the XML files for greet-long and greet-short phrases. I understand the concept but not the execution. When is the one or the other used and how is this controlled? I don't see anything in the ivr code except variables to hold the information. 2. I also see some useful looking stuff in the src scripts directory that don't show up in /usr/local/freeswitch (validate-ivr.pl or fconsole.pl) and won't run from src because of missing perl libraries. I think I must have to execute some installation commands that I have missed but no targets in the Makefiles. Any pointers appreciated and thanks. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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DB error in 9677
by Alex Kinch 27 Sep '08

27 Sep '08

Getting 2008-09-27 13:03:31 [ERR] switch_core_sqldb.c:131 switch_core_db_persistant_execute() SQL ERR [table sip_dialogs has no column named profile_name] when making calls via IAX on the latest trunk (9677). Doesn't seem to be causing too much trouble, but thought I'd flag it. Alex

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Re: [Freeswitch-users] IAX outbound trunk and distortion
by kinchyuk＠googlemail.com 27 Sep '08

27 Sep '08

Sure, will send it over when I get back. Also appear to have a dialplan issue where a call on sip matches but an iax trunk doesn't, but I'll put that on a separate email. Alex -- I sent this from my 3 mobile -- -original message- Subject: Re: [Freeswitch-users] IAX outbound trunk and distortion From: "Brian West" <brian(a)freeswitch.org> Date: 26/09/2008 10:06 pm if the recorded wav exhibits the issue sure. /b On Sep 26, 2008, at 3:39 PM, kinchyuk(a)googlemail.com wrote: > No load as no other calls. Just one channel active. Sip to the same > provider has worked previously. > > I recorded a test call on xlite if you want a copy of the wav? > > Cheers, > Alex > -- I sent this from my 3 mobile --

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Re: [Freeswitch-users] IAX outbound trunk and distortion
by Brian West 27 Sep '08

27 Sep '08

I'm gonna guess you're using the alaw codec and the far end is asterisk? /b On Sep 27, 2008, at 4:17 AM, Alex Kinch wrote: > WAV file of recording attached. > > <eyeBeamRecording_080926_195223.wav> > > > Alex > > On 26 Sep 2008, at 21:40, Brian West wrote: > >> if the recorded wav exhibits the issue sure. >> >> /b >> >> On Sep 26, 2008, at 3:39 PM, kinchyuk(a)googlemail.com wrote: >> >>> No load as no other calls. Just one channel active. Sip to the >>> same provider has worked previously. >>> >>> I recorded a test call on xlite if you want a copy of the wav? >>> >>> Cheers, >>> Alex >>> -- I sent this from my 3 mobile -- >> >

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