- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Conference Call Today
by Michael Collins 17 Nov '10

17 Nov '10

Hello all, It's Wednesday so that means FreeSWITCH conference call! Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_17 We have a lot of little things to discuss, plus DRK is planning on stopping by to talk more about his fun .NET stuff! Talk to you soon. -MC

1 0

Sending UPDATE to a conference.
by M. Ranganathan 17 Nov '10

17 Nov '10

Hello, I would like to know if FreeSWTICH would support sending an in-Dialog UPDATE to a conference in progress for the purpose of media codec renegotiation while the conference is in progress. I work for Avaya. We are using FreeSWITCH in our Avaya SCS project. Thank you in advance for your help. Ranga -- M. Ranganathan

1 0

freetdm context not defined error
by ovvenkat 17 Nov '10

17 Nov '10

Hi to all, I am using following freetdm configuration file *freetdm.conf.xml* <config_profiles> <profile name="my_pri1"> <param name="outbound-called-ton" value="unknown"/> <param name="outbound-called-npi" value="unknown"/> <param name="outbound-rdnis-ton" value="unknown"/> <param name="outbound-rdnis-npi" value="unknown"/> <param name="outbound-calling-ton" value="unknown"/> <param name="outbound-calling-npi" value="unknown"/> <param name="switchtype" value="euroisdn" /> <param name="interface" value="cpe"/> </profile> </config_profiles> <sangoma_pri_spans> <span name="wp1" cfgprofile="my_pri1"> <param name="mode" value="user"/> <param name="context" value="default"/> <param name="dialplan" value="XML"/> </span> </sangoma_pri_spans> I can do outbound call. when I am trying to handle the incoming call, from PRI line call is not landing, When I check the fs_cli log files, it says that, 2010-11-17 02:10:16.853170 [INFO] mod_dialplan_xml.c:331 Processing <1234012480>->1100 in context 2010-11-17 02:10:16.853170 [WARNING] mod_dialplan_xml.c:361 Context not found 2010-11-17 02:10:16.853170 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2010-11-17 02:10:16.853170 [DEBUG] switch_channel.c:2457 (FreeTDM/1:6/1100) Callstate Change RINGING -> HANGUP 2010-11-17 02:10:16.853170 [NOTICE] switch_core_state_machine.c:143 Hangup FreeTDM/1:6/1100 [CS_ROUTING] [NO_ROUTE_DESTINATION] Its not taking the context which is defined in the above configuration file. If I call 1100 from, sip phone call is landing properly. As of now, I have added my dial plan inside the /usr/local/freeswitch/conf/dialplan/default.xml. Previously it was working fine. After updating latest wanpipe and git version of freeSwitch, It stopped working. Any help would be appreciated. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

4 5

Dialplan: is this the correct way?
by Stephen Wilde 17 Nov '10

17 Nov '10

Hi, I'm trying to implement the dialplan but I'm not sure (due to my poor experience in ftreeswitch) if this is the best way. My necessity is to try sequentially some providers until the call is delivered to destination. I have write my own "mod_routecall" that set some channel variables based on called number and so set the dialstrings for each provider with the correct order of tries. In this way I want that freeswitch make sequentially 5 tries in different providers. It's also present the limit_execute because for each provider I have a maximum number of call that I can send. My Dialplan is: <context name="default"> <extension name="startup"> <condition> <action application="routecall"/> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,DESTINATION_OUT_OF_ORDER,NORMAL_CIRCUIT_CONGESTION,SWITCH_CONGESTION"/> <action application="limit_execute" data="hash outbound ${hash0} ${limit0} bridge ${dialstring0}" /> <action application="limit_execute" data="hash outbound ${hash1} ${limit1} bridge ${dialstring1}" /> <action application="limit_execute" data="hash outbound ${hash2} ${limit2} bridge ${dialstring2}" /> <action application="limit_execute" data="hash outbound ${hash3} ${limit3} bridge ${dialstring3}" /> <action application="limit_execute" data="hash outbound ${hash4} ${limit4} bridge ${dialstring4}" /> </condition> </extension> </context> Is this dialplan correct? There is a best way to do that? Thanks in advance, Stephen

2 2

ftmod_libpri - Channel Selection patch
by lakshmanan ganapathy 17 Nov '10

17 Nov '10

Hi all, When using ftmod_libpri, if a incoming call comes without "Channel Identification" ( in SETUP packet ), then the call didn't get proceeded. I've taken some code snippets from ftmod_isdn and I applied it in ftmod_libpri. I've tested it and in my setup it works fine. I've attached the patch here. Please verify it and found ok commit it. regards, Lakshmanan G.

1 0

Is this a correct SIP message flow?
by Stephen Wilde 17 Nov '10

17 Nov '10

Hi, I have see that Freeswitch sends many "487 Sessuin Ternubatedd-" just after the call is disconnected by the caller. Are all these "487 Session Terminated" or none? To me it seems to be a retransmission, but why happens? Freeswitch waits a response to the 200 OK ? Stephen No. Time Source Destination Protocol Info 1 0.000000 IP-Client IP-Freeswitch SIP/SDP Request: INVITE sip:xxxxxxxxxx@IP-Freeswitch;user=phone, with session description 2 0.000505 IP-Freeswitch IP-Client SIP Status: 100 Trying 3 5.760819 IP-Freeswitch IP-Client SIP/SDP Status: 183 Session Progress, with session description 4 14.250804 IP-Freeswitch IP-Client SIP/SDP Status: 200 OK, with session description 5 14.264365 IP-Client IP-Freeswitch SIP Request: ACK sip:xxxxxxxxxx@IP-Freeswitch:5060 8 194.265481 IP-Client IP-Freeswitch SIP Request: BYE sip:xxxxxxxxxx@IP-Freeswitch:5060 9 194.266483 IP-Freeswitch IP-Client SIP Status: 200 OK 10 194.266538 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 11 194.766725 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 12 195.766983 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 13 197.766163 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 14 201.766806 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 15 205.766738 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 16 209.766724 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 17 213.766809 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 18 217.767843 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 19 221.767900 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated 20 225.767845 IP-Freeswitch IP-Client SIP Status: 487 Session Terminated

2 1

Statuts on FS + ISDN?
by GillesToo 16 Nov '10

16 Nov '10

Hello I'd like to know how well FreeSwitch currently works with ISDN (EuroISDN), whether through PCI cards or some external box. Can I recommend it to customers or is it still work-in-progress? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Statuts-on-FS-ISDN-tp5738534p… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 4

problem registering with sip providers
by dnb＠bluewin.ch 16 Nov '10

16 Nov '10

hello, i'm running FreeSWITCH Version 1.0.head (svn-18355) on vmware behind nat. i have problems registering with sip providers: 2010-11-12 13:20:28.845507 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:20:34.866728 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:20:38.898868 [WARNING] sofia_reg.c:387 iptel Failed Registration, setting retry to 30 seconds. 2010-11-12 13:20:40.902922 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:20:46.953148 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:20:52.974365 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:20:58.998596 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:04.025773 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:09.049973 [NOTICE] sofia_reg.c:342 Registering iptel 2010-11-12 13:21:10.051987 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:16.081210 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:22.106424 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:28.150642 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:34.216910 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:40.281096 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:40.281096 [WARNING] sofia_reg.c:387 iptel Failed Registration, setting retry to 30 seconds. 2010-11-12 13:21:46.297302 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:52.318592 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:58.379775 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:22:04.411977 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:22:10.426183 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:22:11.430227 [NOTICE] sofia_reg.c:342 Registering iptel 2010-11-12 13:22:16.441404 [NOTICE] sofia_reg.c:342 Registering sipcall after a while sipcall still gets registered, im able to receive inbound calls, but i cant get outbound working. iptel doesn't get registered at all, for that i followed the instruction in the FreeSWITCH-106- Book i bought. can someone please give me a hint. thanks

3 2

Freeswitch Long Run
by Ahmed Bhaila 16 Nov '10

16 Nov '10

Hello Has anyone have experience running Freeswitch for longer periods preferably more than 24 hrs with average connected calls around 500 at any given time? I have tried this and the problem I am seeing is that Freeswitch seems to use up all the physical memory after which the system goes unstable and doesnt accept any calls I also see a lot of dropped called after that. I have two test machines which I have started with the recommended ulimit settings (described in the wiki) and using -hp and -waste as the startup params. Both machines are Intel Xeon Quad Core with 8 gigs of RAM and running CentOS. The first machine is using the latest GIT-HEAD while the second machine is using Freeswitch version 1.0.4 (tried with 1.0.6 too but didnt help). Also, this is a production level scenario where I have customers sending in real traffic (signaling + media). Any recommendations and/or suggestions? Thanks Ahmed

3 3

Radius AAA
by Abid Saleem 16 Nov '10

16 Nov '10

Hi All, Can someone provide a clear guideline on how to setup Radius AAA to work with FS. Please send complete installation & configuration instructions. Thanks for help in advance. Regards--------------Abid SaleemSr. Product ManagerTerminus Technologies

4 44