July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Multiple (3) NICs
by Тарас Гірник 16 Jul '10

16 Jul '10

Hi, guys. Sorry for posting suppose such a frequently asked question, but anyway it's still problem for me to handle Freeswitch on 3 different subnetworks. In fact i was able to run, but with one discfuntion - on one of the instances there was no sound in one direction. I know you say this problem is related to NAT and so on, but let me ask anyway. It should mentioned that first i was run Asterisk and it was fine with that setup. Another important thing is Freeswitch runs under guest OS Debian/Lenny in OpenVZ environment (kernel 2.6.18 with ovz patch) Network: venet0:0 192.168.1.128/24 venet0:1 public_ip + default route here eth2: 172.20.252.6/30 + route 172.20.252.0/24 via 172.20.252.5 (complete interface derived to guest OS, nothing to deal with HN) My peers is 192.168.1.1 (asterisk runs here, under my administration) 192.168.1.204 (gsm gateway, under my administration) x.x.x.x (remote peer under my administration) 172.20.252.2 (voip providers proxy, no authentication here, not under my administration) I successfully installed profiles belong to first 3 ones and everything was ok. I substituted external-sip external-rtp and others with my public ip. The internal ones - with my internal. The problem was with the last one. I have tried to force Freeswitch run on that one but have no luck. Finally i copied profile "internal.xml" to "internal-2.xml" (actually i tried the same with external.xml) in "sip_profiles" dir and change it to parse files from "sip_profiles/internal-2" dir. I have no idea is this right way, but anyway it works and Freeswitch start to listen on 172.20.252.6. I also used inbound-acl and proxy-acl to handle with no authorization. Freeswitch was able to handle incoming calls on 172.20.252.6 and even it was sound in both directions. But if you try to add a dialplan which will use 172.20.252.2 as an outbound proxy you will see that call is handled correctly but no voice there. I examined outgoing packets send by my phone to Freeswitch and from Freeswitch to 172.20.252.2 and found no rtp was transmitted over eth2 (from freeswitch to proxy) only from my phone to freeswitch. There is no FW on guest OS, i tried to stop in on HN but nothing changed. Packets come to venet0 ok, but then for some reasons are not transmitted to eth2. I'm sure that is something with nat-acl as far as 172.16/16 is also private space addresses, but i'm not sure. Please, help me with that and if you know the right way to handle such 3 subnets config correctly, tell me. Freeswitch was build from the git. Corresponding conf from Asterisk, works [172.20.252.2] type=friend context=*** canreinvite=no host=172.20.252.2 disallow=all allow=alaw videosupport=no outboundproxy=172.20.252.2 fromuser=*** from freeswitch (no audio during outgoing calls) http://www.privatepaste.com/bebc7e8988 sip_profiles/internal-2.xml and http://www.privatepaste.com/4df33fc7a5 sip_profiles/internal-2/int. xml

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pocketsphinx questions
by Seven Du 16 Jul '10

16 Jul '10

Hi, I'm trying pocketsphinx, here are some questions: I'm almost on git head. I followed http://wiki.freeswitch.org/wiki/Mod_pocketsphinx , and can run the pizza test. The recognize not as accurate as expected when I'm using zoiper. Portaudio sounds better.(my background is a little noisy). I also read http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts , lua example, but cannot make it work. 1) Cannot find Makefile when make in grammer/ dir. Goole found it did exists but for some reason deleted, why? and no mklm. 2) the lua example doesn't need a grammer ? It only have a .dic and .lm 3) I didn't see Speech-Type related events on event socket, found a var asr_fire_events in source, but the js example has nothing with this var. Do we need this? Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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.wav/.mp3 recording problems - lag
by Bogdan Patrascoiu 16 Jul '10

16 Jul '10

Hello, I recently set up a FS system for one call-center. One of the request was to enable call recording for inbound calls (it's just an inbound callcenter (20 lines) no outbound calls). At first i went with .mp3 enconding but i soon found out that in a "longer" conversation 8-9 minutes the two channels would get very much lagged.T The caller's, leg A, questions would be far ahead of the callees', leg B, answers. I though that this was caused by the .mp3 encoding so i switched back to plain old .wav. The problem is that the lag is still there and comes to "life" totally random. Has anyone ever encountered this before ? Thanks, Bogdan

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suppress ORIGINATOR_CANCEL errors
by Tim St. Pierre 16 Jul '10

16 Jul '10

Hello, Whenever calls are placed from a SIP endpoint, and that user cancels the call before it's answered, I get an error on the console: [ERR] switch_ivr_originate.c:2529 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] This error is correct, the call failed because someone bailed on it, but I don't really need to know this every time, at least I don't think it justifies logging an error. Is there a way to tell FS not to view an originator_cancel as an error, but to consider it "normal"? Any suggestions appreciated. Thanks! -Tim

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No Answer to Hold Request After Simultaneous-Ring Call Answered
by Jerry Richards 15 Jul '10

15 Jul '10

I have a configuration that simultaneously rings multiple extensions using the following XML statement: <action application="bridge" data="user/${dialed_extension}@${domain_name},sofia/internal/1071%${domain_n ame},sofia/internal/1072%${domain_name}"/> However, after a phone other than the last phone (in the list) answers an inbound call, it cannot successfully hold the call. The INVITE only receives 100 Trying from FS and nothing more. See Line #2041 of the following post: http://pastebin.freeswitch.org/13470 The surrounding XML configuration tags are here: http://pastebin.freeswitch.org/13468 Thanks and Best Regards, Jerry

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no dtmf call back during file_string conjunction
by xuyan yang 15 Jul '10

15 Jul '10

hi all, I write a small IVR in javascript to react on user's 1 key input session.streamFile("a.wav!b.wav",dtmf_handle); when user press a key at the end of a.wav and the beginning of b.wav, there is no dtmf call back. if another key is pressed after that, the dtmf_handle is call with a event containing 2 digits. Is this a bug?

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FreeTDM Dialplan help w/ TDM410p
by Joseph L. Casale 15 Jul '10

15 Jul '10

I was using <condition field="source" expression="mod_openzap"> to catch my incoming zap fxo line, but now w/ FreeTDM changing mod_openzap to mod_freetdm as I expected no longer causes a match. Outgoing calls work, but looking at the log while an incoming call is made, it appears fs isn't even seeing it, nothing appears in the log. Any idea how to troubleshoot this? Thanks, jlc freetdm.conf [span zt FreeTDM] number => 1 fxo-channel => 1:1 freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="1"/> </settings> <analog_spans> <span id="1"> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="enable-callerid" value="true"/> </span> </analog_spans> </configuration> conf/dialplan/public/01_freetdm_in.xml <include> <extension name="freetdm_in"> <condition field="source" expression="mod_freetdm"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="answer"/> <action application="transfer" data="2000 XML default"/> </condition> </extension> </include>

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parallel execution in dialplan
by Achim Stamm 15 Jul '10

15 Jul '10

Hello! I have following problem: I want that 2 applications are executed parallel from my dialplan. Actually the dialplan is executed step by step, that means that application "bridge" is started and phone 1011 is ringing. But application "test" will not start at the same time. This is a snipit from my dialplan: ----------------------------------- <extension name="TEST-Anrufer"> <condition field="destination_number" expression="70"> <action application="info"/> <action application="bridge" data="user/1011(a)192.168.1.30"/> <action application="test" data="EProcessCall,EXT50,70"/> </condition> </extension> ---------------------------------- Is there a solution that both applications are executed parallel and start at same time ? regards Achim Stamm -- Achim Stamm, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 04323897052 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis

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Display update support on Cisco SPA 509g?
by Steven Ward 15 Jul '10

15 Jul '10

Hello list, Does anyone know if the Cisco SPA 509g supports display updates? Is there any good way to check this? I run a uuid_display <uuid> <display> on a channel that the SPA is on, but I'm not even seeing freeswitch send an UPDATE to the phone. This works on my Aastras and Polycoms, so I'm wondering if there is any way to get display updates on the SPA, or if it's simply unsupported by the phone. Thanks! - Steve

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Re: [Freeswitch-users] Variable in Bridge string
by Anthony Minessale 15 Jul '10

15 Jul '10

Its a security feature to prevent injection. I think I can get it to work, try latest GIT otherwise create an new object in your lua script and run distributor in the script. then build the string. or use session:execute("set", "var=val") On Thu, Jul 15, 2010 at 7:20 AM, Ahmed Ajmal <ahmed.ajmal(a)breezecom.ae>wrote: > Hi > > > > I have a strange problem creating a bridge string from LUA. The LUA script > returns a bridge string which looks like this: > > > > sofia/gateway/${distributor(my_dist)}/1773XXXXXXX > > > > I pass this bridge string to the bridge app in the dialplan: > > <action application="bridge" data="${bridge_string}"/> > > > > On executing the dialplan I get the following error: > > *Invalid data* (${bridge_string} *contains a variable* > > * * > > *I have tried executing the bridge app from within the lua script it works > but I would prefer to have the dialplan process it. I am on Freeswitch > 1.0.6.* > > * * > > *Thanks* > > *Ahmed* > > > > > > > > > > _______________________________________________ > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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