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FS dies after some minutes
by Klaus Hochlehnert 23 Nov '09

23 Nov '09

Hi, I did a new installation with the trunk from Saturday (21. Nov.) and it always dies with a core after 5-10 minutes. It happened several times. After that I did a new installation of 1.0.4 and this runs without problems on the same host. I'm using Ubuntu 8.04 Server with all patches. Anyone else experiencing this problem? Thanks, Klaus Here's the bt: #0 0x00007f7aa2eb22fc in sofia_reg_nonce_callback (pArg=0x40f3ca50, argc=<value optimized out>, argv=0x7f7a9c006758, columnNames=<value optimized out>) at ../../../../src/include/switch_utils.h:78 #1 0x00007f7aa91f4a12 in sqlite3_exec (db=0x7f7a9c00a6a0, zSql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", xCallback=0x7f7aa2eb22d0 <sofia_reg_nonce_callback>, pArg=0x40f3ca50, pzErrMsg=0x40f3c680) at ./src/legacy.c:95 #2 0x00007f7aa917b98d in switch_core_db_exec (db=0x7f7a9c00a6a0, sql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", callback=0x7f7aa2eb22d0 <sofia_reg_nonce_callback>, data=0x40f3ca50, errmsg=0x40f3c6e8) at src/switch_core_db.c:93 #3 0x00007f7aa2e985b1 in sofia_glue_execute_sql_callback (profile=0x72e940, mutex=0x0, sql=0x7f7a9c006cd0 "select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8'", callback=0x7f7aa2eb22d0 <sofia_reg_nonce_callback>, pdata=0x40f3ca50) at sofia_glue.c:4297 #4 0x00007f7aa2ead8ec in sofia_reg_parse_auth (profile=0x72e940, authorization=0x7f7a9c078ad0, sip=0x7f7a9c0695d8, regstr=0x7f7aa2fe7137 "REGISTER", np=0x40f3d940 "b7ed6efa-d801-11de-a716-67cbb4a551f8", nplen=128, ip=0x40f3d840 "10.134.38.59", v_event=0x40f3d930, exptime=3600, regtype=REG_REGISTER, to_user=0x7f7a9c0dd18e "29", auth_params=0x40f3cd60, reg_count=0x40f3cd58) at sofia_reg.c:1704 #5 0x00007f7aa2eb004a in sofia_reg_handle_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sip=0x7f7a9c0695d8, regtype=REG_REGISTER, key=0x40f3d940 "b7ed6efa-d801-11de-a716-67cbb4a551f8", keylen=0, v_event=0x40f3d930, is_nat=0x0) at sofia_reg.c:888 #6 0x00007f7aa2eb2f1c in sofia_reg_handle_sip_i_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=<value optimized out>, sip=0x7f7a9c0695d8, tags=<value optimized out>) at sofia_reg.c:1362 #7 0x00007f7aa2e9371c in sofia_event_callback (event=<value optimized out>, status=100, phrase=0x7f7a9c071700 "Trying", nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=0x0, sip=0x7f7a9c0695d8, tags=0x7f7a9c0716f0) at sofia.c:672 #8 0x00007f7aa2f1119e in nua_application_event (dummy=0x0, sumsg=0x40f3dd10, ee=0x7f7a9c0716c8) at nua_stack.c:393 #9 0x00007f7aa2f7e2c1 in su_base_port_execute_msgs (queue=0x0) at su_base_port.c:280 #10 0x00007f7aa2f7e039 in su_base_port_getmsgs (self=0x721320) at su_base_port.c:202 #11 0x00007f7aa2f7e5dc in su_base_port_step (self=0x721320, tout=0) at su_base_port.c:473 #12 0x00007f7aa2f7b68e in su_port_step (self=0x721320, tout=1000) at su_port.h:340 #13 0x00007f7aa2f7b656 in su_root_step (self=0x723320, tout=1000) at su_root.c:858 #14 0x00007f7aa2e8d40a in sofia_profile_thread_run (thread=<value optimized out>, obj=<value optimized out>) at sofia.c:1194 #15 0x00007f7aa88b63f7 in start_thread () from /lib/libpthread.so.0 #16 0x00007f7aa7e20b4d in clone () from /lib/libc.so.6 #17 0x0000000000000000 in ?? ()

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Re: [Freeswitch-users] mod_flite sound profiles
by Michael Jerris 23 Nov '09

23 Nov '09

Sounds like they don't want your business that much. You can try using mrcp with them , not sure if they have that released on their side or not. I think the build integration for mrcp client just went into the windows build earlier today. To be honest we used to have a pretty good relationship with them but we have had basically no response at all to any technical problems we have had with them in quite some time, so maybe they have decided to move on and not work with open source any more. It would appear so from their actions at least. Mike On Nov 23, 2009, at 1:41 PM, Malay Thakershi wrote: > Thank you for your responses. > > I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. > > I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. > > Malay Thakershi > > From: Brian West [mailto:brian@freeswitch.org] > Sent: Monday, November 23, 2009 12:14 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_flite sound profiles > > You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. > > /b

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Using odbc in FS core
by Mike Tkachuk 23 Nov '09

23 Nov '09

Hello Folks, I'm interesting in completely moving away from sqlite and use postgresql everywhere including core ( switch_core.c ) All other applications can use odbc without issues (sofia, limit, fifo etc), but as I see in core only sqlite3 supported. I correctly set 'core-db-dsn' parameter, but looks like the problem that latest psqlodbc_08_04_0100 don't support multiple statements in one request that is often used in switch_core_sqldb.c: > sql = switch_mprintf( > "update channels set uuid='%q' where uuid='%q' and hostname='%q';" > "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" > "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", > switch_event_get_header_nil(event, "unique-id"), > ... SKIP ... So, does anyone have any clue how to us postgresql in the FS core? Thanks. -- Mike Tkachuk

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Question about rtp-timeout-sec variable
by Maciej Aniserowicz 23 Nov '09

23 Nov '09

Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? Regards, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Question-about-rtp-timeout-sec-variable-tp4050650p4050… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Adding headers to INFO messages for Advice of Charge on SNOM
by Keith Laaks 23 Nov '09

23 Nov '09

Hi, I have tried maintaining charging information on a SNOM 300's display using 'display' - but found that the phone has some timer, whereby every 60 seconds it wipes out whatever happens to be on the display at that time and replaces is with the dialled number. So not a viable option as it impacts usability. Really annoying when the display was just updated with valuable information for the user and a split second later it gets replaced. [If somebody knows how to disable this behaviour - please do tell...] I see that SNOM supports a number of features for Advice of Charge. >From their Wiki: http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP Example of an SIP-Info Message: ----------------------------------------------------- INFO sip:bla@snom.com SIP/2.0 From: <biller(a)snom.com>;tag=5354n3 To: <ua(a)snom.com>;tag=33rfh3 CSeq: 23423 INFO Call-ID: 3452tw43dt354dm03 AOC: charging;state=active; charging-info=currency; currency=EUR; amount=2000; multiplier=0.001 Content-Length: 0 ----------------------------------------------------- So the question - Is there some method available today to add these additional 'new' headers to an INFO message I can send out to these phones? If not, I guess it's a matter of looking at enhancing the "case SWITCH_MESSAGE_INDICATE_DISPLAY" section in mod_sofia.c ? Best Regards Keith

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Git
by William Suffill 23 Nov '09

23 Nov '09

Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. -- W

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Re: [Freeswitch-users] mod_flite sound profiles
by Brian West 23 Nov '09

23 Nov '09

You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: > I am not using Linux. I am using Windows 2008 server. > > Malay Thakershi

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Re: [Freeswitch-users] mod_flite sound profiles
by Brian West 23 Nov '09

23 Nov '09

If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: > Ok. I understand that. > > It would be great if someone can help me figure out: > 1. Why mod_flite is not changing to the female voice even > though I tried switching all 4 profiles it provides? > 2. I would be alright for purchasing Cepstral for its quality. > But FS doesn’t come with it compiled I guess (it says swift.dll > required when I enabled it in the config file). I asked Cepstral > support but they say I have to purchase their SDK (no trial > available) even though I just need it to compile it with FS. I > understand I will be purchase the voices but how can I get Cepstral > DLLs without purchasing the SDK. > > Thank you for help. > > > Malay Thakershi

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NAT problem
by Jonas Gauffin 23 Nov '09

23 Nov '09

Hello I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: API CALL [sofia(status profile external)] output: ======================================================== Name external Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false API CALL [sofia(status profile default)] output: ======================================================== Name default Domain Name N/A Alias Of internal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED false STUN-AUTO-DISABLE false Sample phone registration: Call-ID: Xmbw9PyQ5Q6L2MnQ(a)192.168.1.121 User: u1000009@default Contact: "u1000009" <sip:u1000009@192.168.1.121:6094> Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u1000009 Auth-Realm: default MWI-Account: u1000009@default Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: ------------------------------------------------------------------------ INVITE sip:0706930XXX@sipgw2.XXXXX.se <sip%3A0706930XXX(a)sipgw2.XXXXX.se>SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: "KundtjÃ¤nst Arne" <sip:0500650XXX@85.89.XX.XX>;tag=B7pve7F6eeH7c To: <sip:0706930821@sipgw2.XXXXX.se <sip%3A0706930821(a)sipgw2.XXXXX.se>> Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: <sip:mod_sofia@192.168.1.110:5060> Call-Info: <answer-after=400> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: "KundtjÃ¤nst Arne" <sip:0500650XXX@85.89.XX.XX >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Many thanks, Jonas

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IP0010 SIP Phone
by David V. Fansler 23 Nov '09

23 Nov '09

After the help of a couple of people from this list, I now have FreeSWITCH running - yeah! I have installed X-Lite on a couple of computers and they dial each other, play music on hold, etc. I have not yet connected to the outside world. I purchased an IP-0010 phone off eBay ($20 including shipping - docs at http://www.vanaccess.com/news/news_images/2007131_73_User%20Manual%20-%20IP…) I cannot get this phone to work with the system. It gets an IP address, time/date, and a dial tone. After many tries with the http congifuration tool, I got the phone "configured" with the address of the SIP server, and a SIP User ID. When you dial an extension the FreeSWITCH window shows the following: sofica.c3844 Hanugup sofia/internal/101(a)192.168.1.165 [CS_NEW] [INCOMPATIBLE_DESTINATION] switch_core_session.c1139 Session 20 (sofia/internal/101(a)192.165.1.65) Ended switch_core_session.c1141 Close Channel sofia/internal/1001(a)192.168.1.165 [CS_DESTROY] Has anyone else tried this phone, or does anyone have suggestions I could try. I have looked through the website but have not found anything to help. Thanks, David David V. Fansler S/V Annabelle www.dv-fansler.com dfansler(a)dv-fansler.com

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