Re: [asterisk-users] Trouble getting feature codes to work
Hi, Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear Goodbye when I press ** during a call connected this way in my dial plan: exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) exten = 1,n,Playback(vm-goodbye) The call works fine and the CLI tells me that ** is an active feature: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call When I press ** during a call though, nothing appears in the CLI (verbosity = 4). I do it very quickly so I don't believe timeout is an issue. As DTMF recognition is not the problem (as You told in the other post), You can check two other things: 1) Exclude the timing issue: Are the other 2-character feature codes working? What about testing with a 1-character code setting or with a featuretimeout in the conf-file (I believe the default is very short) 2) If this is a sip-to-sip call, check if asterisk stays in the audio path (you can check it with a network sniffer like tcpdump or wireshark). HTH, have a nice weekend, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting feature codes to work
Thanks for the GREAT tip. Changing to a single feature digit of * for blindxfer worked which led me to changing featuredigittimeout = 2000. Now I can do blindxfer w/ ##. Why I didn't try changing featuredigittimeout long ago is beyond me! *blush* Thanks again. One thing that still doesn't work tho is applicationmap. I have this in features.conf: [applicationmap] testfeature1 = #9,caller,Playback,tt-monkeys testfeature2 = #8,callee,Playback,tt-monkeys and this in the context where the dial takes place: include = featuremap include = applicationmap Any ideas? I'd love to hear tt-monkeys from either side of the call! In the end tho, I'm trying to provide alternate method for hanging up since I don't want to base it on adding the h option to the Dial command. As you know that would hangup on a single *, which is not so good when calling an IVR. Cheers! H On Fri, Jan 22, 2010 at 7:02 AM, Karsten Wemheuer k...@gmx.de wrote: Hi, Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear Goodbye when I press ** during a call connected this way in my dial plan: exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) exten = 1,n,Playback(vm-goodbye) The call works fine and the CLI tells me that ** is an active feature: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call When I press ** during a call though, nothing appears in the CLI (verbosity = 4). I do it very quickly so I don't believe timeout is an issue. As DTMF recognition is not the problem (as You told in the other post), You can check two other things: 1) Exclude the timing issue: Are the other 2-character feature codes working? What about testing with a 1-character code setting or with a featuretimeout in the conf-file (I believe the default is very short) 2) If this is a sip-to-sip call, check if asterisk stays in the audio path (you can check it with a network sniffer like tcpdump or wireshark). HTH, have a nice weekend, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting feature codes to work
hugolivude wrote: I have this in features.conf: [applicationmap] testfeature1 = #9,caller,Playback,tt-monkeys testfeature2 = #8,callee,Playback,tt-monkeys and this in the context where the dial takes place: include = featuremap include = applicationmap You need to re-read the sample features.conf; the categories of features defined there are *not* dialplan contexts, so using include = for them in the dialplan is not going to accomplish anything except to generate a warning message when the dialplan is parsed. Immediately below the [applicationmap] category heading in the sample features.conf it describes exactly how to enable the features you have defined in that category. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble getting feature codes to work
Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear Goodbye when I press ** during a call connected this way in my dial plan: exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) exten = 1,n,Playback(vm-goodbye) The call works fine and the CLI tells me that ** is an active feature: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call When I press ** during a call though, nothing appears in the CLI (verbosity = 4). I do it very quickly so I don't believe timeout is an issue. I'd be grateful for any troubleshooting tips. Thanks, H -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting feature codes to work
At 9:08 PM on 21 Jan 2010, hugolivude wrote: The call works fine and the CLI tells me that ** is an active feature: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call When I press ** during a call though, nothing appears in the CLI (verbosity = 4). I do it very quickly so I don't believe timeout is an issue. I'd be grateful for any troubleshooting tips. Try different values of dtmfmode (rfc2833, inband, info) in sip.conf for the SIP peer that you call in from. Asterisk is probably monitoring the wrong method for DTMF. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting feature codes to work
Thanks for the reply. I'm not convinced it's a DTMF problem anymore because I tried all the options still no luck :-( Also I'm dialing the number from an IVR menu so it is recognizing the '1' that i press from the menu. Any other ideas I could try? I am supposed to put this: include = featuremap in the context containing the Dial command right? Thanks in advance, H On Thu, Jan 21, 2010 at 9:31 PM, C. Chad Wallace cwall...@lodgingcompany.com wrote: At 9:08 PM on 21 Jan 2010, hugolivude wrote: The call works fine and the CLI tells me that ** is an active feature: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call When I press ** during a call though, nothing appears in the CLI (verbosity = 4). I do it very quickly so I don't believe timeout is an issue. I'd be grateful for any troubleshooting tips. Try different values of dtmfmode (rfc2833, inband, info) in sip.conf for the SIP peer that you call in from. Asterisk is probably monitoring the wrong method for DTMF. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
