- FreeSWITCH-users - FreeSWITCH Mailing Lists

dialplan of two profile users
by Jengjr Lee ( 李正之 ) 07 Aug '08

07 Aug '08

Dear all : I would like to create 2 profiles which represent 2 company. There will be the same extension no in these 2 profiles. For example , one is "default" , another is "inter2". I am not familiar with the dialplan file , command, logic , control ; don't know how to control the call from these 2 profile extensions. Or is there any more info explain it deeply ? thanks Name Type Data State ================================================================================================= internal profile sip:mod_sofia@210.x.x.x:5060 RUNNING (0) external profile sip:mod_sofia@210.x.x.x:5080 RUNNING (0) nat profile sip:mod_sofia@210.x.x.x:5070 RUNNING (0) default alias internal ALIASED 210.x.x.x alias internal ALIASED inter2 profile sip:mod_sofia@210.y.y.y:5060 RUNNING (0) 210.y.y.y alias inter2 ALIASED outbound alias external ALIASED ================================================================================================= 4 profiles 4 aliases freeswitch@localhost> sofia_contact 1526(a)210.x.x.x API CALL [sofia_contact(1526(a)210.x.x.x)] output: sofia/internal/1526@219.8.z.z:15421 DISCLAIMER: Sample Disclaimer added in a VBScript.

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Event on bad destination?
by Gerry Hull 07 Aug '08

07 Aug '08

I'm using mod_eventsocket. If I perform the API command originate sofia/internal/5344 &park() and 5344 is not a valid destination, I receive the following: 2008-08-06 16:04:30 [WARNING] mod_sofia.c:1890 sofia_outgoing_channel() Cannot l ocate registered user 5344(a)10.0.6.63 2008-08-06 16:04:30 [NOTICE] mod_sofia.c:1975 sofia_outgoing_channel() Close Cha nnel N/A [CS_NEW] 2008-08-06 16:04:30 [ERR] switch_ivr_originate.c:912 switch_ivr_originate() Cann ot create outgoing channel of type [sofia] cause: [NO_ROUTE_DESTINATION] API CALL [originate(sofia/internal/5344 &park())] output: -ERR NO_ROUTE_DESTINATION in the log. However, How can I receive an event as to trap this issue? I don't see one I can subscribe to to get this information. Thanks, Gerry

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conference with background sound
by t 06 Aug '08

06 Aug '08

i like to setup conferences with some ambient music in the background. is there a background sound parameter? when i use the perpetual sound with local_stream its fine. but all the users have unmute themselve first. how can i to things with a user after he entered a conference? in the dialplan? via script? a more general question is how to play no-sound for certain events. if there is something like: <param name="kicked-sound" value="conference/conf-kicked.wav"/> can i set it to something like: <param name="kicked-sound" value="nosound"/> ?? t.

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load test of FS and receiving DESTINATION_OUT_OF_ORDER
by Erol Akarsu 06 Aug '08

06 Aug '08

I am making a load test on FS. I am using a FS box that calls an extension of remote FS box. I am sending originate calls in background command at once and receiving responses through mod_socket interface. Almost 570 out of 1000 calls are having DESTINATION_OUT_OF_ORDER error. I am sending this originate command from my load test client. originate {campaign=kampanya_1,origination_caller_id_name=eakarsu,origination_caller_id_number=8044445555,ignore_early_media=true,originate_timeout=360,call_timeout=360,progress_timeout=360,hangup_after_bridge=true,continue_on_fail=false}sofia/gateway/fs_local_copy_hp/2000(a)192.168.1.7 1238 all these error cases has 2008-08-06 17:59:26 [DEBUG] switch_ivr_originate.c:1251 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] I appreciate if you can help, Regards Erol Akarsu

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load test of FS and receiving DESTINATION_OUT_OF_ORDER
by Erol Akarsu 06 Aug '08

06 Aug '08

I am making a load test on FS. I am using a FS box that calls an extension of remote FS box. I am sending originate calls in background command at once and receiving responses through mod_socket interface. Almost 570 out of 1000 calls are having DESTINATION_OUT_OF_ORDER error. I am sending this originate command from my load test client. originate {campaign=kampanya_1,origination_caller_id_name=eakarsu,origination_caller_id_number=8044445555,ignore_early_media=true,originate_timeout=360,call_timeout=360,progress_timeout=360,hangup_after_bridge=true,continue_on_fail=false}sofia/gateway/fs_local_copy_hp/2000(a)192.168.1.7 1238 all these error cases has 2008-08-06 17:59:26 [DEBUG] switch_ivr_originate.c:1251 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] I appreciate if you can help, Regards Erol Akarsu ----- Original Message ---- From: Michael Collins <msc(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, August 6, 2008 4:32:18 PM Subject: Re: [Freeswitch-users] Limiting call length As a token of your appreciation you could add this to the Rosetta Stone page... :D BTW, any time someone says, "Here's how we do it in Asterisk" and then someone else says "Well in FS you can do that like this..." it would be great to add that to the Rosetta Stone page. That page is kinda thin and community TLC will fatten it up nicely! Thanks, MC On Wed, Aug 6, 2008 at 1:27 PM, Wasim Baig <wasim(a)convergence.pk> wrote: On Thu, Aug 7, 2008 at 2:10 AM, Wasim Baig <wasim(a)convergence.pk> wrote: In asterisk we've been using L(x[:y][:z]) option to Dial. to follow up, use sched_hangup for the x: see http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_sched_hangup and for :y and :z use sched_broadcast see http://wiki.freeswitch.org/wiki/FreeSwitch_Scheduler_API did I ever say thank you, tony? well .. thank you, dear ... -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Limiting call length
by Wasim Baig 06 Aug '08

06 Aug '08

We're working on the billing solution for fs (astpp) fastpp? And need some input on what is the best way to limit call length. In asterisk we've been using L(x[:y][:z]) option to Dial. Whats the recommended route for this in fs? Does something like this exist? If not, can I add this to the "if I were at cluecon, i'd wish" list ... and what would the bounty be? -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070

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Freeswitch and SIPConnect technical recommendations
by Robert Dyck 06 Aug '08

06 Aug '08

I am trying to determine the capabilities of Freeswitch. I have also read through the technical recommendations for SIPConnect from the SIP forum. The reference architecture at the enterprise end shows IP-PBX <-> SIP proxy <-> firewall <-> internet. Using Freeswitch is the proxy necessary or is FS capable of doing the fixups and the TLS negotiation? The following requirements come to mind. TLS connection to a service provider no private addresses, implies local NAT traversal canonical addresses only manipulation of To and From headers support asserted identity

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Any Softphone that works with TLS/SRTP and Freeswitch
by Peter P GMX 06 Aug '08

06 Aug '08

Has anybody successfulkly tested a Softphone that works with TLS/SRTP and Freeswitch? - I tried Minisip (I think it works with MIKEY) - no success - I tried Zoiper Biz - it did not like to connect via TLS (any hints?) - more ?? Best regards Peter

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Cluecon live stream
by Simon Shaw 06 Aug '08

06 Aug '08

As one of the unlucky people not able to attend Cluecon, are any of the sessions going to be streamed?

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Directory and UserAgent Registration
by mayamatakeshi 06 Aug '08

06 Aug '08

Hello, I'm a little confused with FreeSWITCH behavior. At http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide we can read: "The directory section is used to add accounts for all users that should be able to register in the pbx by using User Agents (SIP Phones)." So I suppose I should provide one xml file for each user. But it seems FreeSWITCH lets anyone to register with it, ignoring username/password, checking only the domain. I mean, I can set anything as username/password in my sip phone and as long as the domain is correct, FreeSWITCH will let me register. Is this a bug or am I missing something in my configuration files? I'm testing with the most recent revision 9225, but I can see the same happens with previous versions. Regards, takeshi

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