- FreeSWITCH-users - FreeSWITCH Mailing Lists

opensips & Freeswitch calling issue
by smrdoshi 31 May '13

31 May '13

Hi, I have follow below steps to implement load balancing of FS servers using opensips. http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSW… Here opensips is forwarding registration request to FS servers. But I need users to register only in opensips and when they do calls, FS should work as media server. So in simple, opensips should work as user registration server and Freeswitch should work as media server. I did some modifications in opensips.cfg to do registration in opensips only. Here is the modifications, Removed below code from script if (!add_path_received()) { xlog("L_ERR", "$ci|log|unable to add path"); sl_send_reply("503", "Internal path befuddlement"); #remove the association between the call-id and the media server (if one) #but leave the contact user and server to support transfers cache_remove("local", "$ci"); xlog("L_INFO", "$ci|end|cleaned up call id from cache"); exit; } Added code in place of above code if (!zz_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if ( proto==TCP || 0 ) setflag(7); if (!save("location")) sl_reply_error(); exit; Using above code registration is working fine. But now the problem is with DID number calls. I have two extensions registered in opensips 12345 and 123456. I have DID number 0123456789 in FS. >From Freeswitch I have pointed DID to 123456 extension using sofia/profile/123456(a)192.168.1.10. Here, 192.168.1.10 is opensips ip 192.168.1.31 is fs ip But when i am calling that DID number, I am getting error in opensips like, http://pastebin.com/AhuDP6hC Freeswitch is also showing error like "SIP/2.0 503 Maximum Calls In Progress" I am trying trial & error and searching for solution but not getting anything. Please guys help me to fix the issue. Thanks in Advance. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 285
by Jeffkingsley Kigg 31 May '13

31 May '13

<action application="limit" data="db outgoing gw1 10" />  <action application="limit" data="hash fraud_protection calls_max 1 !NORMAL_TEMPORARY_FAILURE"/>  <action application="limit" data="hash fraud_protection call_per_day 100/86400 !NORMAL_TEMPORARY_FAILURE"/> ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, May 31, 2013 2:21 AM Subject: FreeSWITCH-users Digest, Vol 83, Issue 285 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Book! (Michael Collins) 2. Re: Gateway Call Limits (Michael Collins) w00t! -MC On Thu, May 30, 2013 at 4:03 PM, Nathan Neulinger <nneul(a)mst.edu> wrote: Just showed up today, and looks great! > >-- Nathan > >------------------------------------------------------------ >Nathan Neulinger nneul(a)mst.edu >Missouri S&T Information Technology (573) 612-1412 >System Administrator - Architect > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org Talk to Areski. He built it and I'm sure he would love to know about any limitations or bugs that you may have found. They have a git-hub repo w/ the requisite issue tracker: https://github.com/Star2Billing/newfies-dialer/issues -MC On Wed, May 29, 2013 at 11:00 PM, Alex Ynema <alex(a)opensystems.net.au> wrote: Okay so it's a fault with newfies rather than a problem with Freeswitch. Any suggestions on where I should look to work out what exactly is happening/not happening with those Event Sessions not being cleared out from the process. > > > > >Alex Ynema | IT Consultant >alex(a)opensystems.net.au > > >Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >Office: +61 8 9427 2500 >Mobile: +61 404 796 894 > > > >IT Consultant for Open Systems Support > >www.opensystems.net.au > > >On 30 May 2013 10:25, Brian Foster <bdfoster(a)davri.com> wrote: > >Newfies Dialer uses the ESL. See http://www.newfies-dialer.org/documentation/how-it-works/ >>- BDF >>On May 29, 2013 10:22 PM, "Alex Ynema" <alex(a)opensystems.net.au> wrote: >> >>What do you mean by ESL Program. This system is running a newfies-dialer frontend with a pretty stock install of Freeswitch. >>> >>> >>> >>> >>>Alex Ynema | IT Consultant >>>alex(a)opensystems.net.au >>> >>> >>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>Office: +61 8 9427 2500 >>>Mobile: +61 404 796 894 >>> >>> >>> >>>IT Consultant for Open Systems Support >>> >>>www.opensystems.net.au >>> >>> >>>On 29 May 2013 23:22, Michael Collins <msc(a)freeswitch.org> wrote: >>> >>>It sounds like you have an ESL program that is not consuming events quickly enough and the event queue is filling up. >>>>-MC >>>> >>>> >>>> >>>> >>>>On Wed, May 29, 2013 at 12:17 AM, Alex Ynema <alex(a)opensystems.net.au> wrote: >>>> >>>>so setting the limit at 150 was fine but as soon as I set it to 200 I've now hit a problem. >>>>>Freeswitch has slowly grown to 372 sessions and getting lots of these errors in the cli >>>>> >>>>> >>>>> >>>>>2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any more events..... >>>>> >>>>> >>>>>UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, 606 microseconds >>>>>FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready >>>>>1500 session(s) since startup >>>>>372 session(s) - 0 out of max 10 per sec >>>>>10000 session(s) max >>>>>min idle cpu 0.00/100.00 >>>>>Current Stack Size/Max 240K/8192K >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>Alex Ynema | IT Consultant >>>>>alex(a)opensystems.net.au >>>>> >>>>> >>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>Office: +61 8 9427 2500 >>>>>Mobile: +61 404 796 894 >>>>> >>>>> >>>>> >>>>>IT Consultant for Open Systems Support >>>>> >>>>>www.opensystems.net.au >>>>> >>>>> >>>>>On 29 May 2013 15:01, jay binks <jaybinks(a)gmail.com> wrote: >>>>> >>>>>http://wiki.freeswitch.org/wiki/Limit >>>>>> >>>>>> >>>>>> >>>>>><action application="limit" data="hash ${sip_received_ip} ${destination_number} ${calls_per_second}/1" /> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>On 29 May 2013 16:52, Muhammad Naseer Bhatti <nbhatti(a)gmail.com> wrote: >>>>>> >>>>>> >>>>>>>Sorry for the thread hijack, but on the other hand, is it possible to limit the number of outgoing CPS? Don't seem to see that either in the wiki or a way to make it work. >>>>>>> >>>>>>> >>>>>>>-- >>>>>>>Thanks, >>>>>>>Muhammad Naseer Bhatti >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>Alex Ynema wrote: >>>>>>>Cheers Avi I've now changed that to hash as I don't need it to be persistent. >>>>>>>>What should I see in the clie to confirm this is working without attempting 150+ calls >>>>>>>> >>>>>>>> >>>>>>>>Basically I've added this to my default.xml >>>>>>>> >>>>>>>> >>>>>>>> <action application="set" data="destnum=${destination_number}" /> >>>>>>>> <action application="bridge" data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >>>>>>>> >>>>>>>> >>>>>>>> <extension name="zetta-cisco-1"> >>>>>>>> <condition field="destination_number" expression="zetta-cisco-1"> >>>>>>>> <action application="limit" data="hash outgoing zetta-cisco-1 150" /> >>>>>>>> <action application="bridge" data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>>>> </condition> >>>>>>>> </extension> >>>>>>>> <extension name="zetta-cisco-2"> >>>>>>>> <condition field="destination_number" expression="zetta-cisco-2"> >>>>>>>> <action application="limit" data="hash outgoing zetta-cisco-2 150" /> >>>>>>>> <action application="bridge" data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>>>> </condition> >>>>>>>> </extension> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>Alex Ynema | IT Consultant >>>>>>>>alex(a)opensystems.net.au >>>>>>>> >>>>>>>> >>>>>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>Office: +61 8 9427 2500 >>>>>>>>Mobile: +61 404 796 894 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>IT Consultant for Open Systems Support >>>>>>>> >>>>>>>>www.opensystems.net.au >>>>>>>> >>>>>>>> >>>>>>>>On 29 May 2013 14:13, Avi Marcus <avi(a)avimarcus.net> wrote: >>>>>>>> >>>>>>>>... just note that's stored in a database (db) not ram (hash) so if you don't need to share it / have persistence, just store it in ram. >>>>>>>>> >>>>>>>>> >>>>>>>>>-Avi >>>>>>>>> >>>>>>>>>On Wed, May 29, 2013 at 9:07 AM, Alex Ynema <alex(a)opensystems.net.au> wrote: >>>>>>>>> >>>>>>>>>I've implemented this in default.xml hoping to limit each of my two gateways to 150. >>>>>>>>>>Based on what's in http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-u… so hopefully that works. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> <extension name="zetta-cisco-1"> >>>>>>>>>> <condition field="destination_number" expression="zetta-cisco-1"> >>>>>>>>>> <action application="limit" data="db outgoing zetta-cisco-1 150" /> >>>>>>>>>> <action application="bridge" data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>>>>>> </condition> >>>>>>>>>> </extension> >>>>>>>>>> <extension name="zetta-cisco-2"> >>>>>>>>>> <condition field="destination_number" expression="zetta-cisco-2"> >>>>>>>>>> <action application="limit" data="db outgoing zetta-cisco-2 150" /> >>>>>>>>>> <action application="bridge" data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>>>>>> </condition> >>>>>>>>>> </extension> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>Alex Ynema | IT Consultant >>>>>>>>>>alex(a)opensystems.net.au >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>>>Office: +61 8 9427 2500 >>>>>>>>>>Mobile: +61 404 796 894 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>IT Consultant for Open Systems Support >>>>>>>>>> >>>>>>>>>>www.opensystems.net.au >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>On 29 May 2013 13:59, Alex Ynema <alex(a)opensystems.net.au> wrote: >>>>>>>>>> >>>>>>>>>>Hi, >>>>>>>>>>>I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. >>>>>>>>>>>What's the best way to handle it for an outgoing only system. >>>>>>>>>>>I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>Alex Ynema | IT Consultant >>>>>>>>>>>alex(a)opensystems.net.au >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>>>>Office: +61 8 9427 2500 >>>>>>>>>>>Mobile: +61 404 796 894 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>IT Consultant for Open Systems Support >>>>>>>>>>> >>>>>>>>>>>www.opensystems.net.au >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>_________________________________________________________________________ >>>>>>>>>>Professional FreeSWITCH Consulting Services: >>>>>>>>>>consulting(a)freeswitch.org >>>>>>>>>>http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>http://www.cudatel.com >>>>>>>>>> >>>>>>>>>>Official FreeSWITCH Sites >>>>>>>>>>http://www.freeswitch.org >>>>>>>>>>http://wiki.freeswitch.org >>>>>>>>>>http://www.cluecon.com >>>>>>>>>> >>>>>>>>>>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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>>_________________________________________________________________________ >>>>>>>>>Professional FreeSWITCH Consulting Services: >>>>>>>>>consulting(a)freeswitch.org >>>>>>>>>http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>http://www.cudatel.com >>>>>>>>> >>>>>>>>>Official FreeSWITCH Sites >>>>>>>>>http://www.freeswitch.org >>>>>>>>>http://wiki.freeswitch.org >>>>>>>>>http://www.cluecon.com >>>>>>>>> >>>>>>>>>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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 >>>>>>>_________________________________________________________________________ >>>>>>>Professional FreeSWITCH Consulting Services: >>>>>>>consulting(a)freeswitch.org >>>>>>>http://www.freeswitchsolutions.com >>>>>>> >>>>>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>http://www.cudatel.com >>>>>>> >>>>>>>Official FreeSWITCH Sites >>>>>>>http://www.freeswitch.org >>>>>>>http://wiki.freeswitch.org >>>>>>>http://www.cluecon.com >>>>>>> >>>>>>>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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>>-- >>>>>>Sincerely >>>>>> >>>>>>Jay >>>>>>_________________________________________________________________________ >>>>>>Professional FreeSWITCH Consulting Services: >>>>>>consulting(a)freeswitch.org >>>>>>http://www.freeswitchsolutions.com >>>>>> >>>>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>http://www.cudatel.com >>>>>> >>>>>>Official FreeSWITCH Sites >>>>>>http://www.freeswitch.org >>>>>>http://wiki.freeswitch.org >>>>>>http://www.cluecon.com >>>>>> >>>>>>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 >>>>>> >>>>>> >>>>> >>>>>_________________________________________________________________________ >>>>>Professional FreeSWITCH Consulting Services: >>>>>consulting(a)freeswitch.org >>>>>http://www.freeswitchsolutions.com >>>>> >>>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>http://www.cudatel.com >>>>> >>>>>Official FreeSWITCH Sites >>>>>http://www.freeswitch.org >>>>>http://wiki.freeswitch.org >>>>>http://www.cluecon.com >>>>> >>>>>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 >>>>> >>>>> >>>> >>>> >>>>-- >>>>Michael S Collins >>>>Twitter: @mercutioviz >>>>http://www.FreeSWITCH.org >>>>http://www.ClueCon.com >>>>http://www.OSTAG.org >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting(a)freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>http://www.cudatel.com >>>> >>>>Official FreeSWITCH Sites >>>>http://www.freeswitch.org >>>>http://wiki.freeswitch.org >>>>http://www.cluecon.com >>>> >>>>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 >>>> >>>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting(a)freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>http://www.cudatel.com >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>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 >>> >>> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _______________________________________________ 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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Incoming call and Fax Problems
by mkmk 31 May '13

31 May '13

Hi@all,a little Problem with some callers (5% of all calls).The last Number of the extension comes very late (15-30 seconds later ).The most callers Hang up.logfilehttp://pastebin.freeswitch.org/21003If I send a Fax the following text in the log.http://pastebin.freeswitch.org/21004Can you help me.mkmk -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Incoming-call-and-Fax-Problem… Sent from the freeswitch-users mailing list archive at Nabble.com.

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no ESL event for SIP SIMPLE message if destination number prefixed by +
by Anton VG 31 May '13

31 May '13

Hi! Just was playing with SIP SIMPLE ESL messages, actually all works fine if message sent to 1234567890(a)freeswitch.host. BUT if message is sent to +1234567890(a)freeswitch.host (+ prefix!)- no event fired :( I can see the PCAP the message accepted by FS the same way. So + somehow forces fs to not fire the message. Did anyone encountered such behavior, so should I fill the bug report on JIRA? Regards, Anton

3 4

Book!
by Nathan Neulinger 30 May '13

30 May '13

Just showed up today, and looks great! -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

2 1

inherit_codec not working with renegotiate-codec-on-reinvite
by Jeff Pyle 30 May '13

30 May '13

Hello, This is on version 1.2.10 from the apt repository. I have a functioning configuration with late negotiation and inherit_codec. If I enable renegotiate-codec-on-reinvite in the relevant sofia profiles, the A-leg no longer receives the codec of the B-leg on the initial INVITE setup. Is this expected? I see relevant discussion in FS-3739 <http://jira.freeswitch.org/browse/FS-3739> but I'm having trouble wrapping my head around whether it relates to this situation. - Jeff

1 0

Struggling Here
by Mike Hendrie 30 May '13

30 May '13

Cannot get the SIP phone to ring... Where am I supposed to add the route? In the dialplan, but what am I doing wrong? 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No Route, Aborting Registrations: ================================================================================================= Call-ID: 3624a118-f19b051d-a4d9fed2(a)10.2.1.50 User: 1000(a)gothamcity.xom Contact: "user" <sip:1000@10.2.1.50:5060 ;transport=tcp;fs_nat=yes;fs_path=sip%3A1000%4010.2.1.50%3A33228%3Btransport%3Dtcp> Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 Status: Registered(TCP-NAT)(unknown) EXP(2013-05-30 08:13:09) EXPSECS(179) Host: GothamCity-00 IP: 10.2.1.50 Port: 33228 Auth-User: 1000 Auth-Realm: gothamcity.xom MWI-Account: 1000(a)gothamcity.xom Total items returned: 1 ================================================================================================= /dialplan/public/GothamCity-00.xom.xml <include> <extension name="vitel-inbound"> <condition field="destination_number" expression="^1?(2624481175)$"> <action application="transfer" data="1000 GothamCity.xom.xml default"/> </condition> </extension> </include> =========================== fs_cli when calling the number: EXECUTE sofia/external/7777777777(a)66.66.66.66 transfer(1000 GothamCity.xom.xml default) 2013-05-30 08:04:03.407264 [DEBUG] switch_ivr.c:1815 (sofia/external/ 7777777777(a)66.66.66.66) State Change CS_EXECUTE -> CS_ROUTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777(a)66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:860 Send signal sofia/external/7777777777(a)66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [NOTICE] switch_ivr.c:1822 Transfer sofia/external/7777777777(a)66.66.66.66 to GothamCity.xom.xml[1000@default] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:477 (sofia/external/7777777777(a)66.66.66.66) State EXECUTE going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 (sofia/external/7777777777(a)66.66.66.66) Running State Change CS_ROUTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 (sofia/external/7777777777(a)66.66.66.66) State ROUTING 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:137 sofia/external/ 7777777777(a)66.66.66.66 SOFIA ROUTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:117 sofia/external/7777777777(a)66.66.66.66 Standard ROUTING 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No Route, Aborting 2013-05-30 08:04:03.407264 [NOTICE] switch_core_state_machine.c:193 Hangup sofia/external/7777777777(a)66.66.66.66 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2013-05-30 08:04:03.407264 [DEBUG] switch_channel.c:3096 Send signal sofia/external/7777777777(a)66.66.66.66 [KILL] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777(a)66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 (sofia/external/7777777777(a)66.66.66.66) State ROUTING going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 (sofia/external/7777777777(a)66.66.66.66) Running State Change CS_HANGUP 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 (sofia/external/7777777777(a)66.66.66.66) State HANGUP 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:463 Channel sofia/external/ 7777777777(a)66.66.66.66 hanging up, cause: NO_ROUTE_DESTINATION 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:597 Responding to INVITE with: 404 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:48 sofia/external/7777777777(a)66.66.66.66 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 (sofia/external/7777777777(a)66.66.66.66) State HANGUP going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:689 (sofia/external/7777777777(a)66.66.66.66) Callstate Change RINGING -> HANGUP 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:446 (sofia/external/7777777777(a)66.66.66.66) State Change CS_HANGUP -> CS_REPORTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777(a)66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 (sofia/external/7777777777(a)66.66.66.66) Running State Change CS_REPORTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 (sofia/external/7777777777(a)66.66.66.66) State REPORTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:92 sofia/external/7777777777(a)66.66.66.66 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 (sofia/external/7777777777(a)66.66.66.66) State REPORTING going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:440 (sofia/external/7777777777(a)66.66.66.66) State Change CS_REPORTING -> CS_DESTROY 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777(a)66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1548 Session 1 (sofia/external/7777777777(a)66.66.66.66) Locked, Waiting on external entities 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1566 Session 1 (sofia/external/7777777777(a)66.66.66.66) Ended 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1570 Close Channel sofia/external/7777777777(a)66.66.66.66 [CS_DESTROY] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:565 (sofia/external/7777777777(a)66.66.66.66) Callstate Change HANGUP -> DOWN 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:568 (sofia/external/7777777777(a)66.66.66.66) Running State Change CS_DESTROY 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:578 (sofia/external/7777777777(a)66.66.66.66) State DESTROY 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:373 sofia/external/ 7777777777(a)66.66.66.66 SOFIA DESTROY

2 2

How can I cancel a play_and_get_digits?
by D D 30 May '13

30 May '13

Hi, I am using an play_and_get_digits in an ESL program. Based on an external event, I need to cancel the play_and_get_digits operation. If I send a "break", it only seems to stop the audio from playing, but the play_and_get_digits operation continues, and later the response event comes. How can I stop the entire play_and_get_digits operation from ESL? Thanks, David

4 4

INCOMPATIBLE_DESTINATION for SPA942
by Ivan C Myrvold 30 May '13

30 May '13

When I call from my Bria sip phone, or incoming from my SIP provider (they use Asterisk), I always get INCOMPATIBLE_DESTINATION. See a trace here: http://pastebin.freeswitch.org/21005 This is to a Linksys SPA942. If I call between two SPA942 phones, the call gets through. Do anyone have any clue to why I get INCOMPATIBLE_DESTINATION? Ivan

4 3

OT: sonetel incopatible with fusionbpx/freeswitch???
by Luis Daniel Lucio Quiroz 30 May '13

30 May '13

DOes anybody has sucessfully use sonetel to do SIP forward to another port different than 5060. It seems that what ever string configuration: mydomain:5080 number@mydomain:5080 sip:number@mydomain:5080 the To field is sent okay with correct port, but TCPdump capture shows me that they are hitting port 5060/udp Just wondering if someone has sucessfully configure it

2 1