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seeking script to test FS loads (can be PAID work)
by Brian B 09 Aug '08

09 Aug '08

We're wanting to test our very particular FS conference configuration, 1st on a small VPS slice and then on a dedicated server. To do that, we have written our requirements for a testing script, to generate outbound SIP calls. Is anyone interested in getting involved? This could be paid work, or for the thrill of pushing the pedal to the metal. Ping me and I'll send you details. -Brian 415 308425 8 freeswitch(a)brian-burt.com

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dialplan of multi profile
by Lee JJ 09 Aug '08

09 Aug '08

Dear All Guru : 1. I am curious what's the relation between alias "name", "type" ? Does it affect the dialplan routing call ? How to know more inside of this ? [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1526->1528@default ^^^^^^^^^ From the FAQ: Q: How do I assign endpoints to contexts with different sets of extensions Here are the different possible approaches: - Use 1 profile per context you want to route to (each one needs a distinct ip:port) [root@localhost dialplan]# netstat -aup Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp 0 0 210.24y.y.y:sip *:* 3748/freeswitch udp 0 0 210.24x.x.x:sip *:* 3748/freeswitch udp 0 0 210.24y.y.y:5070 *:* 3748/freeswitch udp 0 0 210.24x.x.x:5070 *:* 3748/freeswitch udp 0 0 210.24x.x.x:5080 *:* 3748/freeswitch Sofia status still the same Name Type Data State ================================================================================================= internal profile sip:mod_sofia@210.24x.x.x:5060 RUNNING (0) external profile sip:mod_sofia@210.24x.x.x:5080 RUNNING (0) nat profile sip:mod_sofia@210.24x.x.x:5070 RUNNING (0) default alias internal ALIASED 210.24x.x.x alias internal ALIASED inter2 profile sip:mod_sofia@210.24y.y.y:5060 RUNNING (0) 210.24y.y.y alias inter2 ALIASED outbound alias external ALIASED ================================================================================================= 2. <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/<user/$%7Bdialed_ext%7D@$$%7Bdomain%7D%22/>> --- this syntax working <anti-action application="bridge" data=" sofia/profilename/${dialed_ext}@$${domain}"/<sofia/profilename/$%7Bdialed_ext%7D@$$%7Bdomain%7D%22/>> ---- while this not ! What is the syntax difference ? FYI: setting files sip profiles use the default config , and modify the correspoding parameter profile name , context , domain [root@localhost sip_profiles]# ls -F ./ ../ external/ external.xml inter2.xml internal/ internal.xml nat/ nat2.xml nat.xml  <profile name="inter2" domain="210.243.126.72">  <aliases> <alias name="210.243.126.72"/> <alias name="inter2"/> </aliases> [root@localhost dialplan]# ls -F ./ default.xml default.xml.old extensions/ inter2.xml ../ default.xml.simpl-test features.xml public.xml the most simple dialplan setting [root@localhost dialplan]# more default.xml <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default">  <extension name="Local_Extension"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <anti-action application="set" data="call_timeout=30"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="bridge" data=" user/${dialed_ext}@$${domain}"/ <user/$%7Bdialed_ext%7D@$$%7Bdomain%7D%22/>>  <anti-action application="answer"/> </condition> </extension> </context> </include> ------------------------------ *From:* Jengjr Lee ( 李正之 ) *Sent:* Thursday, August 07, 2008 6:23 PM *To:* 'freeswitch-users(a)lists.freeswitch.org' *Subject:* dialplan of two profile users 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

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Re: [Freeswitch-users] Where to find freeswitch.jar
by Adeel Ansari 08 Aug '08

08 Aug '08

Hi folks, I am looking to download freeswitch.jar, but couldn't find it till now. May be its hidden somewhere, or my mind is quite exhausted right now. Please help. Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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web voice mail interface / web-vm.tpl
by John Wehle 08 Aug '08

08 Aug '08

The short version (to answer my own question) is: a) Enable mod_xml_rpc. b) Enable mod_shout. c) Change file-extension in voicemail.conf.xml to mp3. d) The URL is: http://my_ip:8080/api/voicemail/web -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Getting user-agent string at time of call
by Ashutosh 07 Aug '08

07 Aug '08

Hi, I m doing user dialplan processing through mod_xml_curl . I understand that FS passes some pre-defined params to the mod_xml_curl handler script like dnis,dnid,callerid, but i wonder if i can also get the user-agent string of the user who is trying to make a call. This way, i want to be able to dynamically create the codec string to be used for the call according to the UA being used. (eg: AMR codec for nokia handsets etc. BTW, is AMR support avaialble yet in FS ?) Thanks very much! -ashutosh

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Documentation suggestion
by Robert Dyck 07 Aug '08

07 Aug '08

I have recently become aware of Freeswitch and I see much potential. Obviously there are huge gaps in the documentation. There are a few example setups and they are great but they suggest we are seeing only the tip of the iceberg. I believe what is needed is a list of all applications and channel variables, their purpose and the module which must be loaded to implement them if not in the core. Then the new user can begin to customize the examples. If this information already exists, kindly point me in the right direction. Thank you

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Anonymous checkout
by mayamatakeshi 07 Aug '08

07 Aug '08

Hello, I'm trying to update my sources, but now http://svn.freeswitch.org is asking for usr/pass. It never did this before. Is this on purpose? Should we register for them? Regards, Takeshi

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Re: [Freeswitch-users] Directory and UserAgent Registration
by Кривушин Михаил 07 Aug '08

07 Aug '08

May be <param name="accept-blind-reg" value="true"/> in sofia profile must be false? > > On Wed, Aug 6, 2008 at 4:42 PM, mayamatakeshi <mayamatakeshi(a)gmail.com> wrote: > > 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. > > Correction: > If a xml file for the user exists, then FS enforces the password. > But if there is no xml file, then the password can be anything. > > 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 > > _______________________________________________ > 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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"Stun failed" and DESTINATION_OUT_OF_ORDER correlated
by Erol Akarsu 07 Aug '08

07 Aug '08

Hi, I found that there is a correlation between DESTINATION_OUT_OF_ORDER and "Stun failed" error specified here: 2008-08-07 08:54:50 [ERR] sofia_glue.c:450 sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org:3478 [Remote Address Error!] My internet connection is up and running. Why the connection drops in the middle of load test? Is it because network card is not able to handle too many network requests and drop? I highly appreciate if you can point me to some help. Regards Erol Akarsu ----- Original Message ---- From: Erol Akarsu <eakarsu_33(a)yahoo.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, August 6, 2008 6:11:45 PM Subject: load test of FS and receiving DESTINATION_OUT_OF_ORDER 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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Re: [Freeswitch-users] Directory and UserAgent Registration
by mayamatakeshi 07 Aug '08

07 Aug '08

Ray, I've just updated to rev 9232 and the problem is gone. I could not find the issue at FS-JIRA (looked into FreeSwitch-Core and Sofia-Sip). Well, if someone has really corrected this issue, thank you. On Thu, Aug 7, 2008 at 5:43 AM, Raymond Chandler <intralanman(a)freeswitch.org > wrote: > Could you please post a bug report of this on jira.freeswitch.org? This > is definitely not expected behavior, so if this is happening it should be > fixed. > > -Ray > > mayamatakeshi wrote: > > > > On Wed, Aug 6, 2008 at 4:42 PM, mayamatakeshi <mayamatakeshi(a)gmail.com>wrote: > >> 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. >> > > Correction: > If a xml file for the user exists, then FS enforces the password. > But if there is no xml file, then the password can be anything. > > >> 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 >> >> > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > > > -- > Regards, > Raymond Chandler > Senior Systems Engineer > Zingo Telecom, Inc. > +1.434.738.2173 offc > +1.434.315.4132 cell > raymondc(a)zingotel.com > >

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