January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

USER_NOT_REGISTERED
by Jim Archer 27 Jan '09

27 Jan '09

Sorry for all these simple questions, but I'm having a registration issue. I have two Polycom 501 phones and have configured both to FreeSwitch. One is extension 1000 and the other is extension 1001. I can call 1000 from 1001, but not the reverse. If I try to call 1001, I go straight to voice mail and the console tells me that 1001 is not registered: 2009-01-26 19:11:24 [ERR] switch_ivr_originate.c:1391 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] Why would it be able to make calls if it is not registered?

2 1

Trouble with default config - Can't call default extensions
by Jim Archer 26 Jan '09

26 Jan '09

Hi All... I installed FS trunk a few days ago on a Debian Etch AMD64 machine. I configured two Polycom 501 phones to talk to it. From each phone, I can dial extension 9999 to get the MOH and 5000 to get the sample IVR. But, I can not call one phone from the other. One is extension 1000 and the other is 1001. I checked and the sample XML files are there in /usr/local/freeswitch/conf/directory/default It seems that as soon as I dial the second digit, the phone decides it is done waiting for me to dial any more, so it tried to connect me to extension 10. Here is what FS spits on to the console: 2009-01-26 18:10:35 [NOTICE] switch_channel.c:566 switch_channel_set_name() New Channel sofia/internal/1001(a)72.46.5.252 [898cf87c-ebfe-11dd-a8c8-b7a3de0bf1ff] 2009-01-26 18:10:35 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FS->10 in context default 2009-01-26 18:10:35 [NOTICE] switch_ivr.c:1253 switch_ivr_session_transfer() Transfer sofia/internal/1001(a)72.46.5.252 to enum[10@default] 2009-01-26 18:10:36 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-01-26 18:10:36 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/internal/1001(a)72.46.5.252 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 25 (sofia/internal/1001(a)72.46.5.252) Ended 2009-01-26 18:10:36 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/1001(a)72.46.5.252 [CS_HANGUP] Any suggestions would be appreciated, thanks! Jim

3 3

Call Groups
by Laurent Fabre 26 Jan '09

26 Jan '09

Hi, I'm having trouble with call groups. They are declared in the directory.xml as mentioned by the documentation : <group name="Sales" id="2006> <users> <user id="1000" type="pointer" /> <user id="1003" type="pointer" /> <user id="1004" type="pointer" /> <user id="1005" type="pointer" /> </users> </group> And included in the dialplan : <extension name="GRP-Sales 2006"> <condition field="destination_number" expression="2006"> <action application="bridge" data="group/Sales(a)sip.kirranet.com"/> </condition> </extension> The endpoints are registered, the dialplan match but I get a cause: NO_ROUTE_DESTINATION I obviously screwed up somewhere. Anybody got an hint where? :) Regards, LF

2 1

Inbound calls
by whudson05 26 Jan '09

26 Jan '09

Hi, I'm experiencing a very specific problem to do with inbound calls through a gateway. When either end of the phone call hangs up then the other end will remain connected as if still in the conversation. If anyone has any idea why this would occur it would be greatly appreciated. Thanks -- View this message in context: http://www.nabble.com/Inbound-calls-tp18694165p18694165.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 4

Inbound calls
by Ron McCarthy 26 Jan '09

26 Jan '09

Hi, If I allow the IPs in a ACL the context always ges changed, is their a way you can have inbound calls be auth'ed via a gateway and then sent to a context. Do I use the directory in that case and not treat it as a external gateway, it would be considered internal then? Thanks!

2 1

Re: [Freeswitch-users] My fs_cli stopped working, can you help me please?
by Milena 26 Jan '09

26 Jan '09

Good morning Ok, here is what i get from the console, do you know what can i do to fix it? thank you very much -bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7 [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] libs/esl/fs_cli.c:573 main() profile default does not exist using builtin profile [DEBUG] libs/esl/fs_cli.c:597 main() Using profile internal [127.0.0.1] [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE Content-Type: [auth/request] [DEBUG] libs/esl/src/esl.c:853 esl_send() SEND auth ClueCon [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] libs/esl/src/esl.c:700 esl_recv_event() RECV HEADER [Reply-Text] = [-ERR invalid] [DEBUG] libs/esl/src/esl.c:833 esl_recv_event() RECV MESSAGE Content-Type: [command/reply] Reply-Text: [-ERR invalid] [ERROR] libs/esl/fs_cli.c:610 main() Error Connecting [Connection Error] -bash-3.2# Anthony Minessale wrote: > > Try running > > fs_cli -d 7 > > Maybe the debug log will shed some light. > > >

3 4

Conference javascript and hanuphooks giving me headaches
by Sias Mey 26 Jan '09

26 Jan '09

Hi, Im trying to originate calls from a conference and use javascript to watch out for hangup events so I can use the data in the session to flesh out some database info. However it seems that Im having some strangeness. It might just be my code. So I include that. I run FreeSwitch Version 1.0.trunk (11226) Dialplan: <extension name="js_logtodb"> <condition field="destination_number" expression="^confout-(10\d{2})$"> <action application="javascript" data="confout.js"/> </condition> </extension> confout.js: is attached I use API calls to pull one user into a conference. Then I use more api calls to do a conference dial to loopback/confout-1001 This should run the js and then bridge extension 1001 into the same conference. (I have hardcoded the additional extension for testing). I dont know if there is another way to get a conference dial to run a javascript file for information logging, but I am open to enlightenment. Oh im using conference dial because that provides clear audible progress to the other conference memebers as to what is actually happening with the new call. Any help would be greatly apreciated, Thanks in advance. Sias

3 8

Announcing the FreeSWITCH Technology Preview VMWare Appliance.
by Ken Rice 26 Jan '09

26 Jan '09

Hey guys, I'm not trying to start 1 a day releases, Things just happened to fall that way... FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken

15 21

auto dialing question ...
by Shelby Ramsey 26 Jan '09

26 Jan '09

Sorry for the double post ... actually hit send too early ... OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful. Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this: -- Remote machines query central DB for numbers to call based on certain configs -- Use AMI to generate the call -- If call gets answered, extension info queried via rta (central db again) The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... So ... conceptually I'm trying to accomplish the same thing ... Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome (not to xml cdr ... and the "proxying" of media) ... Would like to do something like: -- originate request (looks simple enough) -- on answer XML_CURL posts info But for the life of me I can't figure out how to translate this into the xml response ... [campaign] exten => 100,1,ANSWER() exten => 100,n,WAIT(2) exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) exten => 100,n,WAITEXTEN(10) exten => 100,n,HANGUP() exten => 1,1,PLAYBACK(goodbye) .... and so on ... I've looked at the ivr.conf stuff but it's all static and all of this has to be manageable via a web interface .... meaning dumping into a DB and returning an XML response seems reasonable ... but trying to stick or modify static text files from the web interface is too much text parsing and bad things will happen ... Any thoughts or pointing me in the right direction would be appreciated. Shelby

3 8

Pylons example on the wiki
by can_man＠gmx.de 26 Jan '09

26 Jan '09

Hello, I just want to say that I have uploaded a Pylons wiki page with examples in svn. Pylons is a python web framework, which I think can help people to get started easily with xml_curl. If there is demand to extend the examples I am happy to do at. Suggestions are always welcome. You can normally find me on irc: phm_it Cheers, Phil -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

2 1