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Freeswitch restarting profiles
by Jonas Gauffin 17 Oct '08

17 Oct '08

Hello Today Freeswitch restarted the profiles, and I don't know why. 2008-10-08 09:44:49 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] 2008-10-08 09:44:49 [DEBUG] sofia.c:634 sofia_profile_thread_run() Write lock outbound 2008-10-08 09:44:49 [NOTICE] sofia.c:641 sofia_profile_thread_run() waiting for worker thread 2008-10-08 09:44:49 [DEBUG] sofia.c:684 sofia_profile_thread_run() Write unlock outbound 2008-10-08 09:44:49 [WARNING] sofia.c:1478 config_sofia() Invald IP 0.0.0.0replaced with XXXXXXXXXXX 2008-10-08 09:44:49 [WARNING] sofia.c:1487 config_sofia() Invald IP 0.0.0.0replaced with XXXXXXXXXXX 2008-10-08 09:44:49 [NOTICE] sofia.c:1871 config_sofia() Started Profile outbound [sofia_reg_outbound] 2008-10-08 09:44:49 [DEBUG] sofia.c:511 sofia_profile_thread_run() Creating agent for outbound 2008-10-08 09:44:49 [ERR] sofia.c:541 sofia_profile_thread_run() Error Creating SIP UA for profile: outbound 2008-10-08 09:44:49 [DEBUG] sofia.c:634 sofia_profile_thread_run() Write lock nat 2008-10-08 09:44:49 [NOTICE] sofia.c:641 sofia_profile_thread_run() waiting for worker thread 2008-10-08 09:44:49 [DEBUG] sofia.c:684 sofia_profile_thread_run() Write unlock nat 2008-10-08 09:44:49 [NOTICE] sofia.c:1871 config_sofia() Started Profile nat [sofia_reg_nat] 2008-10-08 09:44:49 [DEBUG] sofia.c:511 sofia_profile_thread_run() Creating agent for nat 2008-10-08 09:44:49 [ERR] sofia.c:541 sofia_profile_thread_run() Error Creating SIP UA for profile: nat 2008-10-08 09:44:50 [DEBUG] sofia.c:634 sofia_profile_thread_run() Write lock default 2008-10-08 09:44:50 [NOTICE] sofia.c:641 sofia_profile_thread_run() waiting for worker thread 2008-10-08 09:44:50 [DEBUG] switch_odbc.c:95 switch_odbc_handle_disconnect() Disconnected 0 from [gate] 2008-10-08 09:44:50 [DEBUG] sofia.c:684 sofia_profile_thread_run() Write unlock default 2008-10-08 09:44:50 [NOTICE] sofia.c:1861 config_sofia() Adding Alias [XXXXXXXXXXX] for profile [default] 2008-10-08 09:44:50 [NOTICE] sofia.c:1861 config_sofia() Adding Alias [ mydomain.com] for profile [default] 2008-10-08 09:44:50 [NOTICE] sofia.c:1871 config_sofia() Started Profile default [sofia_reg_default] 2008-10-08 09:44:50 [DEBUG] sofia.c:511 sofia_profile_thread_run() Creating agent for default 2008-10-08 09:44:50 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect() Connecting gate 2008-10-08 09:44:50 [DEBUG] switch_odbc.c:182 switch_odbc_handle_connect() Connected to [gate] 2008-10-08 09:44:50 [INFO] sofia_glue.c:2751 sofia_glue_init_sql() Connected ODBC DSN: gate 2008-10-08 09:44:50 [ERR] sofia.c:541 sofia_profile_thread_run() Error Creating SIP UA for profile: default 2008-10-08 09:53:29 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] And since FS couldn't start the default profile again, everything stopped working. Anyone knows why this happened? //Jonas

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Auto Changing port
by Thiago Maia 16 Oct '08

16 Oct '08

Hi I'm using openser+FS, FS as B2BUA, so, the call goes to openser and then to FS. The problem is, when customer is behind nat, some times work and some times not. When freeswitch show this message, work: 2008-10-17 09:50:25 [INFO] switch_rtp.c:1289 rtp_common_read() Auto Changing port from x.x.x.x:1122 to x.x.x.x:1244 So, I can make 10 calls from the same voip, in sequence, and sometimes work and sometimes not, all time that work I see this message. There are any way to force FS to do that all time? Thanks Thiago

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Help on call transfer
by Rajagopal, Sridhar (Sridhar) 16 Oct '08

16 Oct '08

Hi, I am unable to do call transfer when media is in bypass mode. Please let me know how it can be done. Also why is it implemented in such a way that call transfer works only when media flows through freeswitch. Appreciate your help Regards Sridhar

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Can't see any Sofia messages
by Gavin Henry 16 Oct '08

16 Oct '08

Hi All, I'm trying to get a SIP forwarded call to do something with FS, i.e. go into a conference. I can't even see anything getting rejected: sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia@87.X.X.X:5060 RUNNING (0) external profile sip:mod_sofia@87.X.X.X:5080 RUNNING (0) nat profile sip:mod_sofia@87.X.X.X:5070 RUNNING (0) default alias internal ALIASED pbx.XXXXX alias internal ALIASED outbound alias external ALIASED ================================================================================================= 3 profiles 3 aliases The sip request is coming fine, no firewall issues. pbx.XXXXX :/usr/local/freeswitch/conf# tcpdump -i eth0 -n -s0 -v udp port 5060 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 14:08:34.464662 IP (tos 0x0, ttl 58, id 37907, offset 0, flags [DF], proto: UDP (17), length: 890) 193.111.200.132.5060 > 87.X.X.X.5060: SIP, length: 862 INVITE sip:0XXXXXX@87.X.X.X SIP/2.0 Via: SIP/2.0/UDP 193.111.200.132:5060;branch=z9hG4bK2344219b;rport From: "0XXX" <sip:0XXX@193.111.200.132>;tag=as6f63bcf8 To: <sip:0XXX@87.X.X.X> Contact: <sip:0XXX4@193.111.200.132> Call-ID: 4a5d70d13c3e6f1b5d7c5791318c02cd(a)193.111.200.132 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Oct 2008 13:14:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 317 v=0 o=root 20381 20381 IN IP4 193.111.200.132 s=session c=IN IP4 193.111.200.132 t=0 0 m=audio 14998 RTP/AVP 8 0 18 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - I've switched on all debugging with: cat set_debug.sh #!/bin/bash export SOFIA_DEBUG=9 export NUA_DEBUG=9 export SOA_DEBUG=9 export NEA_DEBUG=9 export IPTSEC_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=9 export TPORT_DUMP=/tmp/tport_sip.log export SU_DEBUG=9 Set: sofia loglevel 9 console loglevel 9 Calls will only ever come in via 193.111.201.114 and I have an ACL for it: 2008-10-16 13:35:28 [NOTICE] switch_core.c:886 switch_load_network_lists() Adding 193.111.201.114/255.255.255.0 (allow) to list strict I've come from Asterisk and I'm used to seeing a lot more info. What am I missing? I haven't added anything for the conference to the dialplan or anything, but I just want to see *something* showing the inbound call. Thanks.

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Getting terminator value from PlayAndGetDigit
by Keith Wood 16 Oct '08

16 Oct '08

Hi, I am looking for ways to extract the actual "terminator" value being pressed by user in the case where user can press # or * to signal the end of DTMF input. By extracting this "terminator" value, I would be able to perform action accordingly. For instance, if * is pressed, it means the user is aware that he/she has entered the wrong number and would like to retry. Could someone tell me if it is something possible within Freeswitch? Thanks, Keith

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failover on landline vs cellular
by Ted 16 Oct '08

16 Oct '08

I'm having a problem with failover. I have the call_timeout set to 15 seconds which results in a landline ringing 4 rings, and INSTANTLY after the 4th ring fails over to the failover destination. However, if the number being dialed is a cell phone I've had one of two things happen: it will either ring and ring until voicemail, or what seems to happening now (I'm not sure what setting caused this) it will rings twice and there will be dead air and eventually the calling side will time out and fail. timeout=15 desk phone: 4 rings -> failover sprint mobile: 5 rings -> voicemail verizon mobile: 5 rings -> voicemail <section name="dialplan" description=""> <context name="public"> <extension name="did"> <condition field="destination_number" expression="^($did)$">  <action application="set" data="effective_caller_id_name=Bob Smith"/> <action application="set" data="effective_caller_id_number=8005551234"/>    <action application="set" data="ignore_early_media=true"/>  <action application="set" data="continue_on_fail=NO_ANSWER"/> <action application="set" data="continue_on_fail=true"/>   <action application="set" data="call_timeout=15"/> <action application="bridge" data="sofia/vitelity/<number to try>@<carrier>"/>  <action application="bridge" data="sofia/vitelity/$0@<failover dest>"/> </condition> </extension> </context> </section> call to desk: 2008-10-15 15:47:13 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/<caller>@<carrier> [1022c19a-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:47:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing <caller>->4073921320@public 2008-10-15 15:47:13 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/4078023342@<carrier> [1026f4ae-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:47:14 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/vitelity/4078023342@<carrier>! 2008-10-15 15:47:14 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/vitelity/<caller>@<carrier>! 2008-10-15 15:47:14 [NOTICE] switch_ivr_originate.c:1148 switch_ivr_originate() Ring Ready sofia/vitelity/<caller>@<carrier>! 2008-10-15 15:47:28 [NOTICE] switch_ivr_originate.c:1277 switch_ivr_originate() Hangup sofia/vitelity/4078023342@<carrier> [CS_CONSUME_MEDIA] [NO_ANSWER] 2008-10-15 15:47:28 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NO_ANSWER 2008-10-15 15:47:28 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 181 (sofia/vitelity/4078023342@<carrier>) Ended 2008-10-15 15:47:28 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/4078023342@<carrier> [CS_HANGUP] 2008-10-15 15:47:28 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/4073921320@<failover dest> [18d07116-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:47:28 [NOTICE] sofia.c:2456 sofia_handle_sip_i_state() Channel [sofia/vitelity/4073921320@<failover dest>] has been answered 2008-10-15 15:47:28 [NOTICE] sofia.c:2469 sofia_handle_sip_i_state() Channel [sofia/vitelity/<caller>@<carrier>] has been answered 2008-10-15 15:47:45 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/vitelity/<caller>@<carrier> [CS_HIBERNATE] [NORMAL_CLEARING] 2008-10-15 15:47:45 [NOTICE] switch_ivr_bridge.c:586 signal_bridge_on_hangup() Hangup sofia/vitelity/4073921320@<failover dest> [CS_HIBERNATE] [NORMAL_CLEARING] 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 180 (sofia/vitelity/<caller>@<carrier>) Ended 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/<caller>@<carrier> [CS_HANGUP] 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 182 (sofia/vitelity/4073921320@<failover dest>) Ended 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/4073921320@<failover dest> [CS_HANGUP] call to cell phone: 2008-10-15 15:51:09 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/<caller>@<carrier> [9cce820a-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:51:09 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing <caller>->4073921320@public 2008-10-15 15:51:09 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/3216956446@<carrier> [9cd274fa-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:51:13 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/vitelity/3216956446@<carrier>! 2008-10-15 15:51:13 [NOTICE] sofia.c:2188 sofia_handle_sip_i_state() Pre-Answer sofia/vitelity/3216956446@<carrier>! 2008-10-15 15:51:13 [INFO] mod_sofia.c:1072 sofia_receive_message() Asked to send early media by sofia/vitelity/<caller>@<carrier> 2008-10-15 15:51:13 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/vitelity/<caller>@<carrier>! 2008-10-15 15:51:13 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer sofia/vitelity/<caller>@<carrier>! 2008-10-15 15:51:24 [NOTICE] switch_ivr_originate.c:1277 switch_ivr_originate() Hangup sofia/vitelity/3216956446@<carrier> [CS_CONSUME_MEDIA] [NO_ANSWER] 2008-10-15 15:51:24 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NO_ANSWER 2008-10-15 15:51:24 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 187 (sofia/vitelity/3216956446@<carrier>) Ended 2008-10-15 15:51:24 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/3216956446@<carrier> [CS_HANGUP] 2008-10-15 15:52:14 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/vitelity/<caller>@<carrier> [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-10-15 15:52:25 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-10-15 15:52:25 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 186 (sofia/vitelity/<caller>@<carrier>) Ended 2008-10-15 15:52:25 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/<caller>@<carrier> [CS_HANGUP] I'm not sure what to make of the differences between these logs. Any ideas? Ted (irc:fedya)

1 0

How to get DISA working ?
by henkoegema 16 Oct '08

16 Oct '08

-- View this message in context: http://www.nabble.com/How-to-get-DISA-working---tp20014047p20014047.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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capture DTMF
by Noah Silverman 16 Oct '08

16 Oct '08

Hi, I'm looking for a way to capture DTMF in the dialplan. My goal is to have a way for a user to enter an account number, pin code, etc. I've seen some examples in javascript and/or perl, but would really like to avoid using an "outside language". Does FS have any kind of function to capture the DTMF. (The format for the javascript function looks great if there was something similar in FS. It allows you to set the number of digits, timeout, etc..) Thanks, -N

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Re: [Freeswitch-users] ODBC through JS (To all interested:)
by Gayatri Kulkarni 16 Oct '08

16 Oct '08

To all those who are interested: Freeswitch uses *.odbc.ini* file to connect to the database and not the {prefix for unixODBC}/etc/odbc.ini *so make sure that one is right before you use("ODBC")* in your applications; otherwise even if you can successfully connect to your DB with the isql command, you wont be able to use it through your FS app. It's good to find out things on my own :) (time consuming though!) -- Regards, Gayatri Kulkarni On Mon, Oct 13, 2008 at 11:21 AM, Gayatri Kulkarni <xtpl.gayatri(a)gmail.com>wrote: > Like we have to add odbc-dsn as a parm name in the voicemail.xml for voice > mail, > do we ave to do add it somewhere for running it in javascript too? > > like i said before, i can successfully connect with same username, > password, DSN through isql utility > why does fs not find the DSN name? > > > -- > Regards, > Gayatri Kulkarni >

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HTTP methods
by Gayatri Kulkarni 16 Oct '08

16 Oct '08

Hi Guys, The XmlRpc is implemented from the web port - 8080 Can we also specify which HTTP methods FS implements - like GET, POST, PUT, DELETE, etc ?? in xml_rpc.conf.xml or some place ....? Is there a way to implement these? -- Regards, Gayatri Kulkarni

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