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mod_radius_cdr questions and thoughts
by Apostolos Pantsiopoulos 28 Jan '09

28 Jan '09

I am trying to implement a radius based solution using FS. I have seen that the mod_radius_cdr module is actively maintained. so I have a few questions/remarks : 1) When I place a call and my radius server is down, the call blocks forever instead of just radius_timeout * radius_retries seconds (I have declared only one server). I would expect that FS would stop trying to send an Acc Start packet after some time and get on with the call. 2) I have also noticed that FS sends only 1 packet (I waited for a minute) instead of 3 (default in the config) since the first (and second) attempt failed. If my server was up (the port was responding) but it returned a req. failed answer would the above time-out be valid? 3) When I tried to load the dictionary.freeswitch to my freeradius server, it complained : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[33]: dict_addattr: Duplicate attribute name NAS-Port-Id" How can I overcome this? When I "instructed" my freeradius to only load the freesitch dictionary I got this : "Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary.freeswitch[257]: unknown option "Freeswitch"" How can I use the dictionary with the freeradius? (or other radius servers?) 4) The radius attributes included in the current requests are a) hard-coded, b) limited in number. I think many of us would like to use more attributes. Or even better define what to include (and what to put in them) using a config file (the same maybe?) 5) Does the module send accounting packets only for the a-leg of a call or for both legs? (Maybe that could be configurable too). If anyone is interested in the above questions/remarks please post a reply. I would really like to know how many of the mailing list users are also interested in FS radius support and your opinions on the matter. Regards, -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Method getVariable cause error on FS 1.0.2 in javascript
by shehzad p 28 Jan '09

28 Jan '09

Hi all, I am using the javascript for call termination, and after call complete, I use getVariable mehtod to get some values like PDD. Below is my javascript that perform this: ============================================================ dialstr="sofia/outbound/1111(a)xxx.xxx.xx.xx"; newsession.originate(session, dialstr); if(session.ready() && newsession.ready()) bridge(session, newsession); if(newsession.getVariable("progressmsec") != false) //in FS 1.0.2 and SVN version SCRIPT TERMINATES on above line pdd = newsession.getVariable("progressmsec"); else pdd = 0; .................<other code> ========================================================== Above script works fine in FS 1.0.1. Now I am testing FS 1.0.2, It also works fine when call is being ANSWERED, PDD is correctly collected, BUT when Call is not routed successfully due to some network problem, or User No-Answer / Busy, then below error occurs: ------------------------------------------------------------------------------------------------------------------------------------------------------- 2009-01-19 19:40:43 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! -------------------------------------------------------------------------------------------------------------------------------------------------------- And after this error, script DOES NOT continue execution and TERMINATES. I also tested SVN version (revision 11266), but same error occurs there too. What should be the cause here, so this behavior of FS is changed from 1.0.1 to 1.0.2. Thanks. msp -- View this message in context: http://www.nabble.com/Method-getVariable-cause-error-on-FS-1.0.2-in-javascr… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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javascript setHangupHook .. looking for a working example
by Stephen Crosby 27 Jan '09

27 Jan '09

I'm running rev 11131, and I cannot make these hangup events work in javascript... does anyone have a working example? I tried this script from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook --Stephen

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Call out to gtalk, asterisk and skype
by EdPimentl 27 Jan '09

27 Jan '09

Hello Everyone, Has anyone done a scalable deployment of calling to Skype/Gtalk/Asterisk Say many 3-25/50 users with various endpoint (skype/gtalk/asterisk)platforms. http://blog.tmcnet.com/blog/tom-keating/asterisk/skype-for-asterisk-launche… http://www.mhspot.com/siptheeskype_skype_trunk_howto.html Thanks in advance, -E

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[CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock
by Chav Paskov 27 Jan '09

27 Jan '09

Hi everybody, i got a strange message today: [CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock i'm running 1.0.trunk (10803) as a VM on ESXi i have a similar setup of 1.0.1 as VM on ESXi but i've never seen this message. Does anybody have an idea what is wrong here? All tips / hints appreciated. Best Regards Chav

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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?

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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

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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

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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.

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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!

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