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Can uPNP be turned off?
by Mario 13 Oct '10

13 Oct '10

I am using ext... autonat:1.2.3.4 in sip internal and external as per wiki. The sofia status is ok for the profiles bu the nat_map shows the wrong address because the router is sending the wrong wan IP in the pnp request. Since I have autonat:1.2.3.4 coded is there a way to force FS not to use uPNP so it does not request nat info from the router? I need it on in the router for other things. Thanks. Mario

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BUG BUG -- mod_spidermonkey or mod_spidermonkey or mod_curl --- BUG BUG???
by Shamun toha md 13 Oct '10

13 Oct '10

Hello, I am almost 2 week on the same script and same problem. It gets OK, and then automatically hangup the call, instead of moving/transfer the call to somewhere else. **In fedora 12, the same script same setup is working 100% accurate. Then i installed it to CentOS 5.5, there it doesnt work. What is the problem, please advise kindly. Should work: =========== Here you can see i am receiving 'OK' via curl 2010-10-13 00:39:04.570826 [INFO] conference.js:45 OK Normally it should now execute this: function my_callback(string,arg) { console_log("info", string); session.sayPhrase("valid", menuselection, "en"); session.setVariable("myroom", menuselection); session.execute("conference" , menuselection, "en"); return true; } Debug: ======= 2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Enter your room number.] (en:en) 2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:2160 OPEN TTS cepstral 2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:2169 Raw Codec Activated 2010-10-13 00:38:46.947636 [DEBUG] switch_ivr_play_say.c:1878 Speaking text: Enter your room number. 2010-10-13 00:38:58.046751 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 3:800 2010-10-13 00:38:58.046751 [DEBUG] conference.js:126 digit: 3 2010-10-13 00:38:58.626757 [DEBUG] switch_ivr_play_say.c:2050 done speaking text 2010-10-13 00:38:58.726760 [DEBUG] conference.js:140 Prompt done=[DialByNumberMenu] Collected 1 digits [3] 2010-10-13 00:38:58.907732 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800 2010-10-13 00:38:59.767758 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 3:800 2010-10-13 00:39:00.546774 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800 2010-10-13 00:39:01.986791 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 9:800 2010-10-13 00:39:02.727801 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 8:800 2010-10-13 00:39:03.626796 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800 2010-10-13 00:39:04.366833 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:800 2010-10-13 00:39:04.366833 [DEBUG] mod_spidermonkey_curl.c:182 Running: method: [POST] url: [http://portal.x86_64.com/conference/query] data: [room=32329821&security=___whyyoudothis___] cred=[] cb: [yes] 2010-10-13 00:39:04.570826 [INFO] conference.js:45 OK 2010-10-13 00:39:04.576826 [DEBUG] switch_ivr_play_say.c:244 Handle execute:[sleep(500)] (en:en) EXECUTE sofia/internal/1002(a)3.x86_64.com sleep(500) 2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Room number is invalid, Please try again.] (en:en) 2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:2160 OPEN TTS cepstral 2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:2169 Raw Codec Activated 2010-10-13 00:39:05.207837 [DEBUG] switch_ivr_play_say.c:1878 Speaking text: Room number is invalid, Please try again. 2010-10-13 00:39:18.606958 [DEBUG] switch_ivr_play_say.c:2050 done speaking text 2010-10-13 00:39:18.707961 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1002(a)3.x86_64.com has executed the last dialplan instruction, hanging up. 2010-10-13 00:39:18.707961 [DEBUG] switch_channel.c:2357 (sofia/internal/ 1002(a)3.x86_64.com) Callstate Change ACTIVE -> HANGUP 2010-10-13 00:39:18.707961 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1002(a)3.x86_64.com [CS_EXECUTE] [NORMAL_CLEARING] 2010-10-13 00:39:18.707961 [DEBUG] switch_channel.c:2373 Send signal sofia/internal/1002(a)3.x86_64.com [KILL] 2010-10-13 00:39:18.707961 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/1002(a)3.x86_64.com [BREAK] 2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1002(a)3.x86_64.com) State EXECUTE going to sleep 2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1002(a)3.x86_64.com) Running State Change CS_HANGUP 2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1002(a)3.x86_64.com) State HANGUP 2010-10-13 00:39:18.707961 [DEBUG] mod_sofia.c:457 Channel sofia/internal/ 1002(a)3.x86_64.com hanging up, cause: NORMAL_CLEARING 2010-10-13 00:39:18.723960 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/internal/1002(a)3.x86_64.com 2010-10-13 00:39:18.723960 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1002(a)3.x86_64.com Standard HANGUP, cause: NORMAL_CLEARING 2010-10-13 00:39:18.723960 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1002(a)3.x86_64.com) State HANGUP going to sleep 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1002(a)3.x86_64.com) State Change CS_HANGUP -> CS_REPORTING 2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/1002(a)3.x86_64.com [BREAK] 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1002(a)3.x86_64.com) Running State Change CS_REPORTING 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/1002(a)3.x86_64.com) State REPORTING 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1002(a)3.x86_64.com Standard REPORTING, cause: NORMAL_CLEARING 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/1002(a)3.x86_64.com) State REPORTING going to sleep 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1002(a)3.x86_64.com) State Change CS_REPORTING -> CS_DESTROY 2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1047 Send signal sofia/internal/1002(a)3.x86_64.com [BREAK] 2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1210 Session 10 (sofia/internal/1002(a)3.x86_64.com) Locked, Waiting on external entities 2010-10-13 00:39:18.724961 [NOTICE] switch_core_session.c:1228 Session 10 (sofia/internal/1002(a)3.x86_64.com) Ended 2010-10-13 00:39:18.724961 [NOTICE] switch_core_session.c:1230 Close Channel sofia/internal/1002(a)3.x86_64.com [CS_DESTROY] 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/1002(a)3.x86_64.com) Callstate Change HANGUP -> DOWN 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/1002(a)3.x86_64.com) Running State Change CS_DESTROY 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1002(a)3.x86_64.com) State DESTROY 2010-10-13 00:39:18.725963 [DEBUG] mod_sofia.c:362 sofia/internal/ 1002(a)3.x86_64.com SOFIA DESTROY 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1002(a)3.x86_64.com Standard DESTROY 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1002(a)3.x86_64.com) State DESTROY going to sleep

2 2

Use Freeswitch
by micho fr 13 Oct '10

13 Oct '10

Dear All, I'm planning to use the freeswitch in my network but I have some doubt regarding some features and need your help please before taking my decision... I need to use the freeswitch as a registrar server + Sip Proxy...I need to know the below: 1- Does freeswitch supports encrypted SIP inside UDP 2- Does freeswitch supports TCP 3- How much TCP connections can handle? Regards

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Re: [Freeswitch-users] Can uPNP be turned off?
by JRichey 13 Oct '10

13 Oct '10

Starting FreeSwitch with "-nonat" (e.g. /usr/local/freeswitch/bin/freeswitch -nc -nonat) will disable UPnP. -Justin -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Mario Sent: Tuesday, October 12, 2010 5:23 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Can uPNP be turned off? I am using ext... autonat:1.2.3.4 in sip internal and external as per wiki. The sofia status is ok for the profiles bu the nat_map shows the wrong address because the router is sending the wrong wan IP in the pnp request. Since I have autonat:1.2.3.4 coded is there a way to force FS not to use uPNP so it does not request nat info from the router? I need it on in the router for other things. Thanks. Mario _______________________________________________ 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

2 1

skip/exit mod_fifo when no agents?
by Jeroen C. van Gelderen 13 Oct '10

13 Oct '10

Hi, Is there a proper way to drop callers out of mod_fifo when no agents are logged in? When no agents are logged in I'd like to proceed in the dialplan escalating the call to a backup number and/or backup fifo. Almost like I need a nowait option for callers. Cheers, -Slim -- Jeroen C. "Slim" van Gelderen

3 2

tone_detect vs. ignore_early_media (per leg?)
by Jeroen C. van Gelderen 13 Oct '10

13 Oct '10

Hi Guys, I have a legacy Panasonic PBX which does not support CPC/Disconnect Supervision. Calls from this PBX are sent to Freeswitch by way of DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal with far-end hang-ups. This works fine:  <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/> Some calls from the Panasonic PBX are put in a FIFO and from there they are sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones in a round-robin fashion. The GSM bridge requires me to use "ignore_early_media=true" otherwise the caller will receive messages like "the number you are calling does not answer". When I set "ignore_early_media=true" the FIFO correctly keeps hunting for a GSM phone that is actually answered and will ignore phones that are busy, no-answer or turned off. This too works fine. The problem occurs when the two are combined as follows: Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge If I enable ignore_early_media then tone_detect doesn't work UNTIL one of the GSMs is answered. This is a problem when none of the GSMs are answered and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO will keep hunting until it times out. If I don't use "ignore_early_media=true" then tone_detect works fine but then every telco message gets mistaken for an answered call and the hunting stops early. I tried changing this example line from my fifo.conf: <member timeout="30" simo="1" lag="5"> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701(a)192.1 68.3.11:5060 </member> to <member timeout="30" simo="1" lag="5"> {member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701@192. 168.3.11:5060 </member> in a vain attempt to ignore early media on the SIP leg only. This doesn't seem to do anything however. Can anyone clue me in on what I'm missing? I've snipped the relevant configuration bits below. I have the feeling I'm missing something obvious. Cheers, -Slim ----8<----8<----8<----8<----8<---- <configuration name="fifo.conf" description="FIFO Configuration"> <settings> <param name="delete-all-outbound-member-on-startup" value="false"/> </settings> <fifos> <fifo name="cool_fifo@$${domain}" importance="0"> <member timeout="30" simo="1" lag="5"> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701(a)192.1 68.3.11:5060</member> <member timeout="30" simo="1" lag="5"> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693(a)192.1 68.3.11:5060</member> <member timeout="30" simo="1" lag="5"> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698(a)192.1 68.3.11:5060</member> </fifo> </fifos> </configuration> ----8<----8<----8<----8<----8<---- <include> <context name="pbx_test"> <extension name="test1"> <condition field="${freetdm_span_name}" expression="^span_fxo_test$"> <action application="answer"/>  <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/> <action application="playback" data="ivr/ivr-hold_connect_call.wav"/> <action application="fifo" data="cool_fifo@$${domain} in"/> </condition> </extension> </context> </include> ----8<----8<----8<----8<----8<---- Cheers, -Slim -- Jeroen C. "Slim" van Gelderen

4 8

Problems with fs_cli on OS X 10.6.4
by Peter Schrock 13 Oct '10

13 Oct '10

Okay, so I managed to get FS working the other day and I even managed to test a call and test my voicemail. All seemed to be working smoothly until, because of the rain here, my power went out and I had to reboot my computer. I logged in through the terminal, set up FS in background went to fs_cli and I get this error message: [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error] I managed to figure out that in the file "fs_cli.c" on line 1206 you find the code for displaying this error message. The problem is that I don't know why this error message is occurring. Does anyone have any helpful hints as to what I should look at to resolve this problem? I even tried going to the git tree and make current, but that gave me problems that forced me to turn off mod_spandsp and mod_skyopen in the modules.conf, which I had running earlier. Any thoughts? PeterS PS I am not sure if this is of any help, but in addition to the error line above, it also posted this info: Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [profile] -?,-h --help Usage Information -H, --host=hostname Host to connect -P, --port=port Port to connect (1 - 65535) -u, --user=user@domain user@domain -p, --password=password Password -x, --execute=command Execute Command and Exit -l, --loglevel=command Log Level -q, --quiet Disable logging -r, --retry Retry connection on failure -R, --reconnect Reconnect if disconnected -d, --debug=level Debug Level (0 - 7)

7 15

Routing DIDs to Dynamically Registered SIP User
by David Allen 13 Oct '10

13 Oct '10

Hi, I'm trying to send multiple Direct Indial Numbers down to a dynamically registered SIP User. I need to ensure that both the To and Target URI contain the direct Indial number. I'm able to modify the SIP TO Header of a call that is sent to them like below: <action application="bridge" data="sofia/external/56778977%${domain}^61390009000"/> which sends the request as: ------------------------------------------------------------------------ INVITE sip:56778977@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.2.200> But in order to maintain compatability with a number of PBX's/VoIP devices on the market, I need to be able to send the invite to the dynamically registered SIP user, however have it set the Target URI and To as the same contact number like below: ------------------------------------------------------------------------ INVITE sip:61390009000@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.22.2:5061> Is there a way to do this setting via variables? I can't seem to find any details for it. Thanks David

6 12

Routing DIDs to Dynamically Registered SIP User
by David Allen 12 Oct '10

12 Oct '10

Hi, I'm trying to send multiple Direct Indial Numbers down to a dynamically registered SIP User. I need to ensure that both the To and Target URI contain the direct Indial number. I'm able to modify the SIP TO Header of a call that is sent to them like below: <action application="bridge" data="sofia/external/56778977%${domain}^61390009000"/> which sends the request as: ------------------------------------------------------------------------ INVITE sip:56778977@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.2.200> But in order to maintain compatability with a number of PBX's/VoIP devices on the market, I need to be able to send the invite to the dynamically registered SIP user, however have it set the Target URI and To as the same contact number like below: ------------------------------------------------------------------------ INVITE sip:61390009000@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.22.2:5061> Is there a way to do this setting via variables? I can't seem to find any details for it. Thanks David

3 3

64 bit CPU and 32 bit OS, is it good for freeswitch
by awais nazir 12 Oct '10

12 Oct '10

Hi I am newbie using and appreciating this remarkable software. I have heard that CENTOS 5.3 is best for freeswitch deployment but I am using 64 bit CPU and 32 bit OS. Can it give any problem? Earliest response will be appreciated. Thanks and Best Regards. Awais Nazeer

5 4