October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Couple Quick Questions
by Eric Liedtke 21 Oct '08

21 Oct '08

First incredibly nice work. I've been beating on a freeswitch box in an SBC type config with a hardware call generator and it's holding up extremely well. I just have 2 questions at the moment. 1) Is there any place to get call statitics for reporting/trending puproses, specifically around complettion rates of gateways/endpoints. If it already exists is it switch wide or can I collect them per endpoint? Apart from parsing the cdrs files of course. If not, any pointers as to where it should live in the code ? I haven't dug into the source much yet, which is the only reason I ask. 2) I think this may have already been answered but I couldn't find the previous email I thought I saw. The calls per second limit that exists today is only for a switch wide config and is not configurable, at this point, on a per endpoint basis right ? Thanks for you time -e

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H323 Support
by Anthony Minessale 21 Oct '08

21 Oct '08

We get people on and off asking for H323 support and a few kind donors have contributed to the cause at http://www.chipin.com/contribute/id/a37fb42ecb0c4d1c If we can even raise a portion of the funding we are pretty close to getting at least a working mod_opal module we can add to tree. (I'm working with the author of opal but it's in our spare time and he needs to eat too) If anyone is interested in helping it along please contact me offlist. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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OpenZap and TDM400p problems
by Sias Mey 21 Oct '08

21 Oct '08

Hi, Im sure this has been beaten to death somewhere... but I have no idea where. I have spent most of the day trying to get this sorted out but now its just confounding me. OK ... I have a trixswitch 0.4 server that I want to add a tdm400p to. I install all of the requirements (gcc, make, kernel-devel) to actually compile zaptel and managed to get the card show the following [root@trixswitch1 conf]# ztcfg -vvv Zaptel Version: 1.4.12.1 Echo Canceller: MG2 Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. So all seems fine up to here. However starting freeswitch yields 2008-10-21 16:19:32 [ERR] zap_io.c:2082 process_module_config() Cannot open wanpipe.conf 2008-10-21 16:19:32 [ERR] zap_io.c:2082 process_module_config() Cannot open zt.conf 2008-10-21 16:19:32 [ERR] zap_io.c:2155 zap_global_init() No modules configured! 2008-10-21 16:19:32 [ERR] mod_openzap.c:1861 mod_openzap_load() Error loading OpenZAP 2008-10-21 16:19:32 [CRIT] switch_loadable_module.c:734 switch_loadable_module_load_file() Error Loading module /opt/freeswitch/mod/mod_openzap.so **Module load routine returned an error** I have been messing arround with the conf/openzap.conf and conf/autoload_configs/openzap.conf.xml but that message is the only thing I can get it to show. I have checked the permissions on those files and moved them arround in between the config dirs... but still no luck. Any ideas? I know im missing something silly, I just dont know what?. Thanks, Sias

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USER_NOT_REGISTERED (?)
by henkoegema 21 Oct '08

21 Oct '08

I have one remote extension 2013 registered (?) to my FS server: Call-ID: NGQwNDQyMmY1ZTFjN2I1NDY4MGRlOGRiN2UzM2JhY2Q. User: 2013(a)oegema.com Contact: "Simon" <sip:2013@193.173.175.235:5060;rinstance=f4ba663fd28da726> Agent: Zoiper for Windows rev.599 Status: Registered(UDP)(unknown) EXP(2008-10-18 22:01:48) Host: ubuntu I can call FROM 2013 to all my local extensions, but not TO 2013 2008-10-18 22:16:54 [INFO] switch_ivr_async.c:1536 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-10-18 22:16:54 [ERR] switch_ivr_originate.c:1011 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2008-10-18 22:16:54 [ERR] switch_ivr_originate.c:1011 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2008-10-18 22:16:54 [INFO] mod_dptools.c:1839 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED What must I do to call extensions in the domain 'oegema.com' ? Thanks Henk -- View this message in context: http://www.nabble.com/USER_NOT_REGISTERED-%28-%29-tp20050486p20050486.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH as pure SIP proxy
by Arturo Díaz Almagro 21 Oct '08

21 Oct '08

Hi all, I am new in the FreeSWITCH world and I am not been able to discover how FS can acts as a pure SIP forwarder, sorry. My problem is that I need a kind of B2BUA "in the middle" of the signaling path between my SIP phones and my SIP Proxy. The purpose of this is to get additional features in my VoIP system like CAC, MoH, etc, etc. The first problem I get is I need to forward all REGISTER methods: PHONE -> FS -> SIP PROXY, but WITHOUT changing the Contact header field just adding a Via. I started with Asterisk due to I have quite experience with it but Asterisk is not able unless you hack chan_sip. FS claims to be a softswitch with SIP router capabilities so, it should be able to do that task. At this stage I am quite confused about the configuration of FS as a SIP proxy and today, I am not completely sure if that is possible to do with FS. Could anyone help me? Is that possible? If that is the case, how could I do it? Thanks a lot. -- Arturo Díaz Contact me on FWD: 870436 Skype: arturo.diaz.almagro

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Newbie: Avaya SES <>Freeswitch 407 Proxy Authentication error
by Gerry Hull 21 Oct '08

21 Oct '08

Here's my configuration: <include> <gateway name="avaya"> <param name="username" value="3824"/> <param name="password" value="password"/> <param name="proxy" value="10.0.2.154"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry_seconds" value="30"/> </gateway> </include> - sofia shows successfully registered with SES (SIP Enablement Services) When I call 3824 from another softphone registered with SES, sofia issues a 407 Proxy Authentication error and the call does not go through. I presume their is something wrong in my configuration, but I cannot figure it out. Any ideas? Gerry

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ODBC through JS
by Gayatri Kulkarni 21 Oct '08

21 Oct '08

Hi all, I want to connect to a remote database using javascript. When i searched the WIki I got this page: http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#SpiderMonkey_ODBC I did the steps what are mentioned there, namely: 1. First download and install unixodbc. See below<http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc>if you are struggling with this step. Make sure to install headers, libraries and source for the unixodbc library set. 2. In the FreeSWITCH source directory, uncomment the languages/mod_spidermonkey_odbc line within the build/modules.conf file. 3. Compile core odbc support into freeswitch. ./configure --enable-core-odbc-support make all install The ./configure doesnt give any error nor does the build When i restart freeswitch, i can see this: 2008-10-06 19:01:43 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared object file: No such file or directory** Will my script run inspite of this? What am i not doing - something i should do more? -- Regards, Gayatri Kulkarni

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regarding conference (basic questions)
by Gayatri Kulkarni 21 Oct '08

21 Oct '08

Hi guys, I read through mod_conference page on wiki but i am still not clear about how do i initiate a conference after i do the configuration conference <conf-name> dial user1 ? and then how do i add the other users? can i add two+ users simultaneously? as in say, conference <conf-name> dial user1 user2 ... ?? I tried that but it failed Is there a 'min-num-of-users' requirement, as in, a conference can happen between at least 2/3 users, etc ? how do i exit from a conference using command line? for that matter how can i use any of the conference features using FS console? can i use those features for a UUID? Thanks for your help -- Regards, Gayatri Kulkarni

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sdp header rewrite
by Gabriel Kuri 21 Oct '08

21 Oct '08

I'm having an issue with the linksys spa devices when enabling inbound proxy media mode (inbound-proxy-media=true) and late negotiation (inbound-late-negotiation=true) in the sofia profile. The spa immediately sends a BYE when the call is answered by the called party. For whatever reason, it works fine between two linksys devices directly connected to FS, but when the call goes out to the PSTN via the SIP provider, the spa isn't happy and sends a BYE. After comparing the raw SIP packets on the wire (tcpdump) and between enabling/disabling proxy-media mode and late negotiation, the only difference I notice is the port in the m= line of the SDP header. According to the freeswitch log, the rtp port would be rewritten to 28044 in the sdp header of the SIP packet sent to the spa device. But on the wire, the port is rewritten to 0, which I'm guessing is why the spa isn't happy and sending a BYE. Here's the excerpt from the freeswitch log showing FS rewriting the port to 28044 for the packet going to the spa device. [DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp() sofia/internal/<phone_number_removed>@mydomain.net Patched SDP --- v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 24174 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 +++ v=0 o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX s=sip call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 28044 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 However, the packet on the wire reveals FS rewriting the port to 0 ... v=0. o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY. s=sip call. c=IN IP4 YY.YY.YY.YY. t=0 0. m=audio 0 RTP/AVP 96 101. a=rtpmap:96 G729/8000. a=fmtp:96 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. Is this a bug or is there some other problem? Thanks for the help, Gabe

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Setting the moh
by Woody Dickson 21 Oct '08

21 Oct '08

Hi, I am getting the following error when executing the meta_app (*4). The error is complaining about moh not found. Does anyone know where I can define the moh? If I want to define different moh for each user, is it doable? HANGE_MEDIA 2008-10-21 04:09:14 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Kill sofia/external/5003 [BREAK] ===Received event=CHANNEL_ANSWER ===Received event=CODEC 2008-10-21 04:09:14 [DEBUG] sofia.c:2233 sofia_handle_sip_i_state() Channel sofia/internal/5002(a)192.168.1.101 entering state [completed] ===Received event=CHANNEL_ANSWER ===Received event=CODEC ===Received event=CHANNEL_BRIDGE handle_start_media===================================== 2008-10-21 04:09:14 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/5003) State CONSUME_MEDIA going to sleep 2008-10-21 04:09:14 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/5003 Running State Change CS_EXCHANGE_MEDIA 2008-10-21 04:09:14 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/external/5003) State EXCHANGE_MEDIA 2008-10-21 04:09:14 [DEBUG] mod_sofia.c:373 sofia_on_exchange_media() SOFIA LOOPBACK 2008-10-21 04:09:14 [DEBUG] sofia.c:2233 sofia_handle_sip_i_state() Channel sofia/internal/5002(a)192.168.1.101 entering state [ready] ===Received event=CHANNEL_STATE 2008-10-21 04:09:16 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf() sofia/internal/5002(a)192.168.1.101 Processing meta digit '4' [execute_extension::att_xfer XML features] 2008-10-21 04:09:16 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/internal/ 5002(a)192.168.1.101 [BREAK] 2008-10-21 04:09:16 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/5002(a)192.168.1.101[BREAK] 2008-10-21 04:09:16 [DEBUG] switch_core_session.c:616 switch_core_session_queue_private_event() Kill sofia/external/5003 [BREAK] 2008-10-21 04:09:16 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/5003 [BREAK] 2008-10-21 04:09:16 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/external/5003 Command Execute playback(local_stream://moh) 2008-10-21 04:09:16 [ERR] mod_local_stream.c:301 local_stream_file_open() unknown source moh 2008-10-21 04:09:16 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/external/5003 [BREAK] 2008-10-21 04:09:16 [DEBUG] switch_ivr_bridge.c:222 audio_bridge_thread() Kill sofia/internal/5002(a)192.168.1.101 [BREAK] ===Received event=CHANNEL_EXECUTE ===Received event=CODEC ===Received event=CODEC ===Received event=CHANNEL_EXECUTE_COMPLETE ===Received event=PRIVATE_COMMAND 2008-10-21 04:09:26 [DEBUG] switch_ivr.c:382 switch_ivr_parse_event() sofia/internal/5002(a)192.168.1.101 Command Execute execute_extension(att_xfer XML features) 2008-10-21 04:09:26 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 5002->att_xfer in context features ===Received event=CHANNEL_EXECUTE 2008-10-21 04:09:26 [CONSOLE] mod_xml_curl.c:206 xml_url_fetch() XML response is in /tmp/8a94486a-6749-4e1a-94b9-6c337b4e324b.tmp.xml 2008-10-21 04:09:26 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [features] destination_number(att_xfer) =~ /^att_xfer/ 2008-10-21 04:09:26 [NOTICE] switch_core_session.c:1219 switch_core_session_execute_exten() Execute read(1 10 'tone_stream://%(10000,0,350,440)' digits 30000 #) 2008-10-21 04:09:26 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 30ms 2008-10-21 04:09:26 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/5002(a)192.16 y Thanks alot. Woody

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