November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

outbound socket connection erro
by Mumuney Abdlquadri 07 Nov '12

07 Nov '12

Hi All, I am using node-esl to setup an outbound connection. I am on ubuntu. I have a softphone connected to freeswitch from a win7 system on Virtualbox. my dialplan is such: <include> <extension name="abdlquadri.com"> <condition field="destination_number" expression="^(1982)$"> <action application="sleep" data="1000"/> <action application="set" data="name=abdlquadri"/> <action application="log" data="INFO ${name} is online!"/> <action application="socket" data="127.0.0.1:8022 syn full"/> </condition> </extension> </include> When I dial 1982 the output from freeswitch is such: 2012-11-07 12:40:04.403324 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1000(a)10.0.2.2 [df4ad290-28cf-11e2-a3ea-bdbddea31077] 2012-11-07 12:40:04.403324 [INFO] mod_dialplan_xml.c:485 Processing abdlquadri <1000>->1982 in context default 2012-11-07 12:40:05.403318 [INFO] mod_dptools.c:1420 abdlquadri is online! 2012-11-07 12:40:05.403318 [INFO] switch_core_session.c:2137 Sending early media 2012-11-07 12:40:05.423325 [NOTICE] mod_sofia.c:2585 Pre-Answer sofia/internal/1000(a)10.0.2.2! 2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error! 2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:226 sofia/internal/1000(a)10.0.2.2 has executed the last dialplan instruction, hanging up. 2012-11-07 12:40:29.563345 [NOTICE] switch_core_state_machine.c:228 Hangup sofia/internal/1000(a)10.0.2.2 [CS_EXECUTE] [NORMAL_CLEARING] 2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1400 Session 6 (sofia/internal/1000(a)10.0.2.2) Ended 2012-11-07 12:40:29.563345 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1000(a)10.0.2.2 [CS_DESTROY] My node-esl code does not get called. Here is it: var esl = require('modesl'); var conn = new esl.Server({ port:'8022', host:'127.0.0.1' },function(){ conn.on('connection::open', function(){ conn.execute("answer"); conn.getInfo(); console.log('Connection Open'); }); conn.on('connection::ready', function(conne){ conne.execute("answer"); conne.getInfo(); console.log('Connection Ready'); }); conn.on('connection::close', function(){ conn.execute("answer"); conn.getInfo(); console.log('Connection Closed'); }); console.log(conn); }); I guess the problem is this line: 2012-11-07 12:40:29.563345 [ERR] mod_event_socket.c:458 Socket Error! Please is there anything am doing wrong. Thanks. All.

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Re: [Freeswitch-users] Connecting delays too much sometimes
by Phil Quesinberry 07 Nov '12

07 Nov '12

I was wrong about that . just tried again to a different number and it was terminated through CoreTel this time with the same delay. So it's definitely an Alcazar problem. The rep told me that they use FreeSWITCH and Kamilio when we opened an account, perhaps the FS database change has something to do with it. not sure how the integration with Kamilio is set up for them and whether or not it would involve the database. They've been having this problem for close to a couple of weeks now though, I think. - Phil _____________________________________________ From: Phil Quesinberry Sent: Wednesday, November 07, 2012 12:59 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Connecting delays too much sometimes Are you using Alcazar for termination, by chance? They're having an issue with that right now, where calls to certain destinations take around 12 seconds to proceed, while others proceed in a normal fashion. It appears to be occuring when they attempt to terminate the calls through Broadvox. I provided them with some diagnostic info and they said they were looking into the issue. That was a couple of days ago. Phil Quesinberry Q Systems Engineering, Inc. Embedded Systems Development and VoIP Business Telephone Hosting Improve your business telephone services and save money (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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Connecting delays too much sometimes
by Phil Quesinberry 07 Nov '12

07 Nov '12

Are you using Alcazar for termination, by chance? They're having an issue with that right now, where calls to certain destinations take around 12 seconds to proceed, while others proceed in a normal fashion. It appears to be occuring when they attempt to terminate the calls through Broadvox. I provided them with some diagnostic info and they said they were looking into the issue. That was a couple of days ago. Phil Quesinberry Q Systems Engineering, Inc. Embedded Systems Development and VoIP Business Telephone Hosting Improve your business telephone services and save money (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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Weekly News and Notes or something
by Ken Rice 07 Nov '12

07 Nov '12

Hey Guys, Mr Collins is under the weather this week, so I¹m trying to fill in for him. Don¹t forget today is the Weekly FreeSWITCH Community Conference Call at 1PM Easter, 10AM Pacific or 1700 UTC. Join us on the bridge. And of course, notes for this week along with how to connect can be found via the Web at http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_07 Coming today FreeSWITCH 1.2.4, git has already been tagged and source tarballs are available at http://files.freeswitch.org/ And this Friday don¹t forget to join us for the Friday Free For All, 888 will be open, join us unmute and have some fun. If you have a project or topic you feel would be of interest for the FreeSWITCH community in general and would like to do a presentation on the weekly conference call contact me offlist. And as always suggestions for topics on the weekly conference call are always welcomed. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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mod_callcenter missed calls.
by Mimiko 07 Nov '12

07 Nov '12

Hello. I was asked about missed calls increasing when the agent was talking to some client from queue. The strange thing is that mod_callcenter does not attempt to contact an agent which is in state of answering a call ("in a queue call" state) as I understand from log. But despite this fact and the fact that the agent does not get its missing call counter increased in mod_callcenter's database, the phone still somehow get an attempt and increases its internal count of missed call. I don't understand how it is, because if the phone were receiving the second call, it will show on the display as "calling extension..." and if the second parallel call were not answered then it must increase the count. But no second line is shown. Is this a problem of FS trying to contact an active answering agent or it is the phone which messes up? The phone is DPH-150S. -- Mimiko desu.

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Directory entry variables not set on channel when user authenticated by cidr and domains ACL?
by Dmitry Sytchev 07 Nov '12

07 Nov '12

Hi all! I use xml_curl to get subscriber information from database when they authenticate with username/password. In my directory script I specify <params> and <variables> for user entry, and they are set on channel. Now I'm trying to achieve customer authentication by IP address instead of username/password. I have <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> and apply-inbound-acl="domains" at users' profile, and passwordless authentication works fine for users with "cidr" attribute, but FS doesn't set channel variables I return in directory entries in response to 'network-list' request on 'reloadacl', for example. Since my dialplan logic relies on these variables (like customer ID and so on), dialplan scripts don't see these vars on incoming calls from users authenticated by domains ACL and fail their calls :( Is this normal behaviour or I'm missing something? How can I get user variables from my DB set on channel when authenticating call by CIDR attribute? -- Best regards, Dmitry Sytchev, IT Engineer

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SIP client authentication via ActiveDirectory
by shayne.alone＠gmail.com 07 Nov '12

07 Nov '12

Hi all; I'm looking for a way to authentication sip client with AD. here is a simple wiki <http://wiki.freeswitch.org/wiki/Mod_ldap> and i had enabled the module to load. but there is no traffic toward to AD at all! how can i debug or better is this all for LDAP authentication of sip clients? should configure directory to use LDAP API anywhere else? -- Regards, Ali R. Taleghani <a.taleghani(a)aut.ac.ir>

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About "Dial Groups" or "Callgroup intercept"
by sparklezou 07 Nov '12

07 Nov '12

Hi Sir/Madam, I have read the wiki http://wiki.freeswitch.org/wiki/Callgroup_intercept And also implement it on FS. Here I want to know, does FS could implement such features? I know such features are working on some digital phone system. 1. "KAKA" & "GAGA" are in the same "Dial Groups". 2. When someone inside/outside call "KAKA", there will be visible sentens on the phone LCD of "GAGA", "XXX call KAKA"(inside display the name), "12345678 call KAKA" (outside display the number). Is it posible to implement? Thanks in advance! 2012-11-05 sparklezou

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Re: [Freeswitch-users] bypass_media between 2 outbound calls
by Víctor Vladimir Chávez Gallardo 07 Nov '12

07 Nov '12

Hi, i have an spidermonkey script, the script place two outbound calls, but i need to set the RTP directly between the legs (a-b) but i dont know hoy to set the parameter i tried: lega_session.setVariable('bypass_media', 'true'); bridge(lega_session,legb_session); and setting bypass_media also on both sessions before the bridge.. but i dont have the rtp in the correct way, instead my freeswitch it's behind the rtp flow any idea? Víctor Vladimir Chávez Gallardo vivlachaga(a)gmail.com >

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Inbound bypass media not working?
by PhilQ 07 Nov '12

07 Nov '12

FreeSwitch is still proxying media after adding param name="inbound-bypass-media" value="true"/ (with enclosing <>) to sofia.conf.xml. When that didn't work, I added it to internal.xml as well. FS was restarted to ensure the changes were active but there was no change. I even tried setting inbound-no-media. What could I be doing wrong? I realize that not a lot of folks are using bypass-media (afaik) but if this were a bug I'm sure that others would be complaining about it. - Phil -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Inbound-bypass-media-not-work… Sent from the freeswitch-users mailing list archive at Nabble.com.

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