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Problem with missing / duplicated DTMF
by Matt Stockton 02 Feb '11

02 Feb '11

I have having trouble with both missing and duplicated DTMF in Freeswitch. Here are the steps of how I am using it: 1. Leg A - I am calling out from my Freeswitch instance (through iCall), and I am calling an iCall number that is also connected to the same Freeswitch instance. 2. Leg B - The above call is routed through iCall and then answered by the same Freeswitch instance. 3. On Leg B, I play a file and attempt to get DTMF in a lua script. Here is the code of interest: callPin = session:playAndGetDigits(1, 10, 4, 30000, "#","/tmp/cw_17.wav", "", "\\d+"); 4. On Leg A, I send DTMF information in a lua script. Here is the code of interest. I initiate a delay between each digit: local newPin = ""; for i = 1, string.len(pin) do newPin = newPin .. string.sub(pin, i, i) .. "W"; end session:execute("send_dtmf", newPin .. "#@200"); ** Note that there is a session:sleep on Leg A before I send the DTMF to make sure i don't send it too early ** The problem is that the recognized DTMF on Leg B is wrong about 30% of the time. For example, if Leg A enters: 22063083, Leg B will get the DTMF digits 222063083. This is an example of duplication, but I have also experienced missing DTMF codes (and an occasional wrong code completely) I have messed with a bunch of DTMF settings in hopes of fixing this issue, but I cannot seem to find something that is reliable 100% of the time. _____________________________ Here are the DTMF settings I have looked at / messed with. I've tried various values for the dtmf-duration in the config (and in the send_dtmf command above)  <param name="dtmf-duration" value="500"/> <param name="dtmf-type" value="rfc2833"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="inbound-codec-negotiation" value="greedy"/>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU"/> ________________________ I have run fs_cli with event logging and the DTMF events that Freeswitch gets do correlate to the wrong value (e.g. the duplication / missing digits is noticable in the Freeswitch events as well). Also, I am not running any dtmf-related applications on the session before I give control to the lua scripts (e.g. not running start_dtmf) Has anyone experienced this type of issue? Or know what I can do to resolve it? My next step was going to be trying this against another provider besides iCall, but I figured I would see if anyone has encountered a similar problem before. Any help is appreciated. Thanks

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Modify Registration String to ITSP SipStation
by Brad Mina 01 Feb '11

01 Feb '11

I'm having trouble registering a SipStation account. They require username:password@trunk1.freepbx.com<username%3Apassword(a)trunk1.freepbx.com>to be sent to the registrar. I've done a bit of searching but was unable to find any specific details on how to do this. My config is as follows: <include> <gateway name="SipStation"> <param name="username" value="myUsername"/> <param name="password" value="somePassword"/> <param name="realm" value="trunk1.freepbx.com"/> <param name="from-user" value="TriTelComm"/> <param name="proxy" value="trunk1.freepbx.com"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="retry-seconds" value="60"/> <param name="context" value="public"/> </gateway> </include>

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H323 Setup
by Tim King 01 Feb '11

01 Feb '11

> > I am following http://wiki.freeswitch.org/wiki/Mod_h323 and trying to get > H.323 setup. I installed using buildopal.sh script in the build directory.I > am not finding h323.conf.xml anywhere except the sample in > /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/h323.conf.xml. Do > I need to create this file and if so where should it be? > Thanks Tim

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Load mod_xml_rpc via cli?
by Addison Martin 01 Feb '11

01 Feb '11

I seem to be able to load mod_xml_rpc at runtime, but netstat -a isn't showing port 8080 as listening. Is it possible to do this at runtime, or is a restart of the switch required? I'm buidling a click2dial greasemonkey script, and the xml_rpc api seems the way to do it without a bunch of client side requirements. Regards, Nik

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freeswitch.com is returned to us!
by Anthony Minessale 01 Feb '11

01 Feb '11

http://www.freeswitch.com/ Thank you everyone who helped with this! -- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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(no subject)
by Saji Honey 01 Feb '11

01 Feb '11

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Developer Dinner Coming Up - Help Feed The FreeSWITCH Team!
by Michael Collins 01 Feb '11

01 Feb '11

Well, it's that time again. The FreeSWITCH developers are gathering at a secret location in Wisconsin next week and the community is invited to participate in this year's "buy the devs dinner" event. How to do it: Go to freeswitch.org and click on the Paypal link. Toss a few bucks in the hat and send along your bon apetit message. If for some reason you cannot use Paypal but still wish to participate then by all means call Brian West on his SIP phone: 2000(a)bkw.org. (Hint: you can dial 9191 if you have a default FreeSWITCH config.) Brian is well-prepared to take your money. :) To those who have donated recently we say, "Thank you!" If you wish to donate again specifically for this occasion you are more than welcome to do so. In any case, thank you in advance for your generosity. You are great community members and the FreeSWITCH team is absolutely glad to have you with us! -Michael Collins

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minimum UDP packet size
by Madovsky 01 Feb '11

01 Feb '11

is it useful to set minimum UDP pakcet to the MTU rate like 1500 and avoid any fragmentation ? Thanks

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port reg request pbm
by Sam 01 Feb '11

01 Feb '11

hello, The device is sending request to my server ip x.y.z.v from port 35666 a.b.c.e:35666 -> x.y.z.v :5060 Register but FS is not sending back to the same port on the device x.y.z.v:5060 -> a.b.c.e : 5060 401 unauthorized how this could be solved Regards Sam

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execute_on_answer=send_dtmf 1
by mazilo 01 Feb '11

01 Feb '11

I have the following simple google_in dialplan (conf/dialplan/public/094_gtalk_in.xml) meant to handle incoming calls for my Google Voice DID by mod_dingaling, but it doesn't seem like it ever gets executed. As such, I always have to manually press 1 to accept the incoming call on my GV DID. If I answer the call after the 1st ring, then I hear no GV call presentation announcer asking me to press 1 to accept the call (see the http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-answering-prema… remarks by Anthony Menessale if you want to know why). By the time I realize this, the incoming call has been intercepted by the GV voicemail and pressing 1 will have no use. I am hoping anyone here who has managed to configure his/her FS which automatically sends dtmf 1 as outlined in http://wiki.freeswitch.org/wiki/Google_Voice here will be able to help me out. <extension name="google_in" continue="true"> <condition field="caller_id_name" expression="^(Google Voice)$"> <action application="log" data="Google Voice Call Incoming" /> <action application="set" data="execute_on_answer=send_dtmf 1"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/execute-on-answer-send-dtmf-1… Sent from the freeswitch-users mailing list archive at Nabble.com.

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