| 00:04.38 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 00:15.44 | *** join/#asterisk Baylink (~jra@cerberus.vicimarketing.com) |
| 00:18.16 | kc8pxy | need some help with an extension, I've been out of this way too long.. i'm trying exten => 100xx,1,Dial(SIP/{$EXTEN} and it's not working.. i'm sure it's a stupid syntax error, but i don't see it int the samples. |
| 00:18.52 | TJNII | exten => _100XX |
| 00:19.09 | WIMPy | and ${} |
| 00:19.17 | TJNII | The extension you currently have is not being pattern matched. |
| 00:19.49 | Baylink-work | Yeah, kc8pxy; what they said. :-) |
| 00:19.59 | Baylink-work | You're missing a closing paren too, but that might have been a typing error. |
| 00:26.01 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-177-17.mia.bellsouth.net) |
| 00:35.53 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
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| 00:42.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:46.06 | sawgood | If have this typeed in from the CLI: set core debug 9 |
| 00:46.20 | sawgood | Does this mean I have SIP debug on and running? |
| 00:46.57 | WIMPy | no. sip set debug ... |
| 00:47.04 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 00:47.11 | sawgood | thank you! |
| 00:48.28 | sawgood | Is there anyway (in which conf file) could I edit (core set verbose 5 and core set debug 5) in ... so each time Asterisk is started, those are the levels set |
| 00:48.40 | sawgood | done automatically without me having to type them in each restart |
| 00:49.10 | sawgood | I thought this might be logger.conf, but I do not know the syntax to make it happen |
| 00:49.22 | p3nguin | asterisk.conf |
| 00:49.54 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:50.10 | p3nguin | verbose = 5 \ debug = 5 |
| 00:50.27 | sawgood | very nice ... I see them commented out |
| 00:50.31 | sawgood | thank you so much! |
| 00:50.34 | sawgood | very cool! |
| 00:51.41 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 00:56.40 | sawgood | What I find so frustrating is this ... you have a SIP end point (which has programmable feature keys), but in the SIP phone GUI ... there are only 2 or 3 choices in the drop down menu to pick from .... none of which are what you want the button to do ... |
| 00:56.58 | sawgood | Why don't the OEM makers of the SIP phones ... add a 'catch-all' line to the drop downs |
| 00:57.15 | sawgood | where one can enter specific syntax for Asterisk to make the PFK do what you need it to do |
| 00:57.48 | sawgood | Its like these PFK buttons are 'wasted' if you are not using them on a hosted "Broadsoft" platform or with the OEM's specific IP PBX model |
| 00:58.13 | WIMPy | Maybe you sould select your hardware more carefully? :-) |
| 00:58.30 | sawgood | WIMPy: yeah, you are probably right ... |
| 00:58.37 | WIMPy | 'd have thought most phones offer what you want. |
| 00:58.53 | sawgood | I want a PFK to act as a MWI light (for another extension) |
| 00:59.13 | sawgood | so, everyone in the office could know when a VM is left in the general voicemail box |
| 00:59.20 | sawgood | start blinking a button on the phone ... |
| 00:59.32 | sawgood | seems rather easy for skilled programmers I'd think |
| 00:59.37 | WIMPy | 2nd identity? |
| 00:59.49 | sawgood | yeah ... using the Grandstream GXP series phones |
| 00:59.59 | WIMPy | Not sure if you can set a hint for VM. |
| 01:00.51 | sawgood | Or ... another point ... all these phones (even Aastra and Snom) claim on the box, and in their documentation that they fully support shared line appearance / bridged line appearance ... |
| 01:01.11 | sawgood | but when you call them, they say, "That feature is only avialable if you are using a Broadsoft platform" |
| 01:01.16 | WIMPy | Yes, horrible. |
| 01:01.20 | WIMPy | :-) |
| 01:01.37 | sawgood | you agree or are you joking with me? |
| 01:01.54 | WIMPy | That really should have died in the 80s. |
| 01:02.34 | sawgood | it is very efficent for a small office with two or three lines ... and four or five workers |
| 01:02.39 | sawgood | Joe, "Pick up line 1" |
| 01:02.53 | sawgood | they are spolied from legacy key systems |
| 01:02.57 | *** join/#asterisk Fu|g0re (~whodis@216.17.67.126) |
| 01:03.16 | WIMPy | Don;t you think a transfer makes much more sense? |
| 01:03.23 | WIMPy | VERY legacy. |
| 01:03.53 | sawgood | WIMPy: well, in some offices ... workers are not always available at the same desk you transfer the call to ... this way they can pick up line 1 from a phone in the wareshouse, etc. |
| 01:04.02 | sawgood | I know call parking could work |
| 01:04.14 | sawgood | but there is no 'flashing' button on phones when you park a call |
| 01:04.18 | WIMPy | So how do they know it's for them then? |
| 01:04.45 | WIMPy | I think you can set hints to parking spaces. |
| 01:05.00 | sawgood | I think you are right |
| 01:05.57 | sawgood | So, I could 'trick' a parked call hint on a phone which supports BLF's I belive |
| 01:06.01 | sawgood | believe |
| 01:06.47 | sawgood | See, the Grandstream phones are so reasonably prices (sub $100 dollar phones) ... when you get to an Aastra or Snom phone ... they are closer to $200 dollars |
| 01:06.57 | sawgood | its hard to sell a $200 dollar phone (times 10) |
| 01:07.33 | WIMPy | likes his Snom a lot more than the others I have seen. |
| 01:07.37 | sawgood | Maybe Aastra or Snom have a lower-priced phone which can support what I am looking for |
| 01:07.50 | sawgood | I like (really like) my Snom 360/370 |
| 01:07.58 | sawgood | it is a great phone ... |
| 01:08.08 | sawgood | I am ordering a white 870 on Monday |
| 01:08.38 | sawgood | The Cisco SPA525G is a good phone too ... (color touch screen) |
| 01:08.42 | WIMPy | I'd like to take a look at the 8xxs, but I think they are not very reasonable. |
| 01:08.45 | sawgood | with Bluetooth support |
| 01:08.59 | WIMPy | 320 should be able to do what the 360 does, exept for the display. |
| 01:09.14 | sawgood | 320 is a good choice too, but the LCD is only 2 lines |
| 01:09.28 | sawgood | I need a phone which can display name, caller's number, and DNIS |
| 01:09.37 | WIMPy | I've got a SPA 962 and I like it so much, it's not even connected any more. |
| 01:10.00 | sawgood | I guess DNIS would not be needed if outside line appearance button support was available |
| 01:10.29 | sawgood | Is it possible on a Snom to have a PFK blink when a call comes in on a SIP trunk? |
| 01:10.43 | sawgood | So the worker will know which number the calling party dialed to reach the extension? |
| 01:11.06 | sawgood | 800 number for example VS. the standard (415) area code business line |
| 01:11.16 | WIMPy | The usual way would be multiple identities, each assigned to a button. |
| 01:11.33 | sawgood | on a Snom 320? |
| 01:11.37 | WIMPy | But doesn't the 320 also diplay the account on the display? |
| 01:12.03 | sawgood | I think so (in front of the caller ID name) |
| 01:12.17 | sawgood | 301/caller ID CNAM |
| 01:12.42 | WIMPy | Otherwise you could alway use dialplan logic to set a prefix to the caller name. |
| 01:13.02 | sawgood | really? |
| 01:13.20 | sawgood | the prefix being the number the customer dialed? |
| 01:13.37 | WIMPy | That's what I use to see via which provider a call came in on. |
| 01:13.38 | sawgood | like the TO: field in the SIP request |
| 01:13.58 | sawgood | SIP INVITE request I mean? |
| 01:14.02 | sawgood | the TO field? |
| 01:14.23 | WIMPy | I use two letter abbreviations, like SG/John Doe whan John Doe called via SipGate. |
| 01:14.34 | sawgood | that seems promising! |
| 01:14.38 | p3nguin | Set(CALLERID(name)=TF-${CALLERID(name)}) |
| 01:14.57 | sawgood | How long of a prefix can it be you think? |
| 01:15.06 | WIMPy | you don;t have to fiddle around with SIP headers, just use teh CALLERID(name) variable. |
| 01:15.16 | WIMPy | Exactely. |
| 01:15.17 | p3nguin | Depends on how many chars the phone's display has. |
| 01:15.32 | WIMPy | Depends on the size of your phones screen. |
| 01:15.45 | sawgood | You will actually get a "/" sign between the two pieces of information? |
| 01:15.58 | p3nguin | If you use a /, you'd get a /. |
| 01:16.02 | WIMPy | If you put it there. |
| 01:16.05 | p3nguin | If you use a -, you'd get a -. |
| 01:16.16 | sawgood | I see your - mark now ... sorry |
| 01:16.24 | sawgood | This goes in extensions.conf? |
| 01:16.35 | p3nguin | Yes, because that's where the dialplan is. |
| 01:16.37 | WIMPy | yes |
| 01:16.40 | sawgood | ha!@ |
| 01:17.16 | sawgood | So, for this to work ... would it go in the 'general' section or the 'local' section |
| 01:17.24 | p3nguin | no |
| 01:17.40 | p3nguin | It goes in dialplan. |
| 01:17.44 | sawgood | Right now, I have a very super simply dialplan ... |
| 01:17.50 | p3nguin | Show me. |
| 01:17.52 | sawgood | it is like 10 or 15 lines |
| 01:17.58 | p3nguin | ~pb |
| 01:17.58 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 01:19.52 | sawgood | http://pastebin.com/rUK1E9eJ |
| 01:20.45 | p3nguin | And which exten are you wanting to modify? |
| 01:21.00 | p3nguin | doesn't see the toll-free number that was mentioned earlier. |
| 01:21.01 | sawgood | either 1000 or 1005 |
| 01:21.12 | sawgood | it is mapped to a standard number |
| 01:21.27 | p3nguin | That's a problem. |
| 01:21.33 | *** join/#asterisk Micc (~Micc@c-98-225-57-96.hsd1.wa.comcast.net) |
| 01:21.56 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 01:22.09 | sawgood | right, but for now I have those two numbers listed |
| 01:22.36 | p3nguin | Let's use them for example purposes. |
| 01:22.37 | sawgood | for practice (just to see something show up on the softphone LCD) |
| 01:23.33 | Micc | Anyone know of a good document on how to setup a polycom park button with asterisk? |
| 01:24.06 | Micc | I got it working with buddies and can pickup parked calls, but the park button itself I have never been able to get working right. |
| 01:24.36 | Micc | On Aastra's I use speeddialxfer to the park extension and that works great. |
| 01:24.44 | p3nguin | sawgood: http://pastebin.com/951FAV7e |
| 01:24.48 | sawgood | ty! |
| 01:25.19 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 01:25.58 | p3nguin | If 6509032154 is your Customer Service number and 4082131414 is your Tech Support number, this is how you could manipulate the caller ID based on what the person called. |
| 01:26.52 | *** join/#asterisk jhirley (~jhirley@adsl-159-227-25.mia.bellsouth.net) |
| 01:27.24 | sawgood | VERY nice! |
| 01:27.28 | sawgood | wow ... amazing! |
| 01:28.45 | p3nguin | But if you have a toll-free number which is routed to 4082131414 BEFORE it gets to Asterisk, then it might be very hard to know when someone called the toll-free number. |
| 01:28.59 | sawgood | right ... |
| 01:29.21 | sawgood | But I have had 13 DID numbers coming into the IP PBX .. this is one way to know which one of the 13 lines was calling |
| 01:29.38 | p3nguin | right |
| 01:29.56 | WIMPy | There might be redirection onformation, if you're lucky, but I don't think you could read it with Asterisk in case yiu get it. |
| 01:31.06 | p3nguin | What is the usage of 13 different numbers? |
| 01:31.22 | sawgood | p3nguin: the customer has 13 numbers for basically 13 different businesses ... |
| 01:31.31 | sawgood | I know its crazy ... but they are in all different area codes at times |
| 01:31.40 | sawgood | They want to know which area code is calling them |
| 01:31.50 | p3nguin | Look at the caller ID. :/ |
| 01:32.04 | p3nguin | It says exactly who's calling. |
| 01:32.08 | WIMPy | 13? I already have 11 at home. |
| 01:32.16 | sawgood | right , but it is possible (619) customer could be calling the (408) number |
| 01:32.34 | sawgood | The (619) customer might not have dialed the (619) number ... |
| 01:32.35 | p3nguin | Does it really make a difference? |
| 01:32.45 | sawgood | to them it is HUGE and a MUST have! |
| 01:33.03 | p3nguin | If it doesn't change anything, I wouldn't do it. |
| 01:33.14 | sawgood | So, let me ask you this ... |
| 01:33.25 | sawgood | Since, I was getting caller ID name and number on my softphone already ... |
| 01:33.50 | sawgood | And since this caller ID name/number setting was not in my extensions.conf file, how did Asterisk know how/what to deliver to the softphone |
| 01:34.02 | p3nguin | It's part of the phone call. |
| 01:34.46 | sawgood | ok ... its included in the SIP INVITE request (I understand that) ... but my point is ... where does Asterisk know what to deliver from the SIP INVITE |
| 01:34.47 | p3nguin | Using Set() in the dialplan, you just change what is already there. |
| 01:34.52 | sawgood | why does it deliver this information? |
| 01:35.39 | sawgood | I did not have any Set commands in this simple dialplan |
| 01:35.59 | sawgood | I was getting CallerID delivered to the softphone with no setting for it in the dialplan |
| 01:36.10 | sawgood | so, I would say this is part of the 'module' for Asterisk (AGI stuff)? |
| 01:36.42 | p3nguin | From: and Contact: maybe? |
| 01:38.06 | p3nguin | I never really gave it any thought how to see caller id info in the sip debug. |
| 01:38.25 | sawgood | I understand Asterisk pulls the info from various SIP INVITE requests ... but what part of Asterisk is telling it to do that? |
| 01:38.48 | sawgood | Its not in my dialplan, so what command is this coming from |
| 01:38.51 | WIMPy | The information from he caller is just passed on to the callee. |
| 01:39.22 | WIMPy | While it executes the dilplan the information is stored in channel variables that can be read and also modified. |
| 01:39.28 | p3nguin | Like I said before, the caller id was already in the call... using Set(), you just alter it. |
| 01:40.07 | sawgood | p3nguin: right ... but before we edited the dialplan, I was still getting incoming caller ID name and number (and there was nothing in the dialplan telling it to do this) |
| 01:40.22 | sawgood | I just was not getting the DNIS information ... now I am |
| 01:40.23 | jhirley | does ffa work on * 1.4.29 over a sip trunk ? |
| 01:40.32 | p3nguin | For the third time, the caller ID info is ALREADY PART OF THE CALL. |
| 01:40.58 | p3nguin | You just modify it in the dialplan. |
| 01:41.17 | p3nguin | Set(CALLERID(name)=CS-${CALLERID(name)}) - This says to set the callerID name to exactly what the callerID name already was, but add CS- on the front of it. |
| 01:42.00 | sawgood | I actually only saw CS on the LCD (not the number + CS) |
| 01:42.03 | p3nguin | You could change it to something totally different if you wanted. |
| 01:42.09 | sawgood | ok |
| 01:42.33 | p3nguin | Set(CALLERID(name)=Jack In The Box) |
| 01:42.48 | p3nguin | Now the display will show that your call is from Jack In The Box. |
| 01:43.46 | p3nguin | If you only had CS- on the display earlier, then your call did not have any callerID name value before. |
| 01:44.38 | sawgood | p3nguin: is there a CALLERID(number) field |
| 01:44.52 | p3nguin | yes |
| 01:45.06 | p3nguin | CALLERID(num), anyway. Not sure if number still works. |
| 01:45.20 | sawgood | I'll try that ... be right back |
| 01:45.58 | p3nguin | Set(CALLERID(num)=408/${CALLERID(num)}) |
| 01:46.17 | sawgood | why the 408? |
| 01:46.39 | p3nguin | Prepend 408 on the front of the caller's number. |
| 01:46.58 | p3nguin | Actually, the / might not be valid in the CID number value. :( |
| 01:47.29 | WIMPy | Numbers are not numbers. |
| 01:47.58 | WIMPy | Might need escaping, but Asterisk should hopefully take of that by itself if neccessary. |
| 01:48.10 | WIMPy | s/of/care of/ |
| 01:48.29 | p3nguin | I'm trying to demonstrate how you can use Set() to change what the existing value is before it reaches the phone. You can put whatever values you want to put there. Doesn't have to be 408. |
| 01:48.31 | sawgood | Set(CALLERID(num)${CALLERID(num)}) |
| 01:48.36 | p3nguin | nope |
| 01:49.01 | sawgood | Set(CALLERID(num)/${CALLERID(num)}) |
| 01:49.06 | p3nguin | nope |
| 01:49.24 | p3nguin | Set(something=value) |
| 01:49.45 | sawgood | Set(CALLERID(num)=408/${CALLERID(num)}) |
| 01:49.51 | p3nguin | CALLERID() is a function, which is writable. The function is your "something" that you're setting. |
| 01:50.32 | p3nguin | ${CALLERID(num)} is the value of the existing CALLERID(num) function. |
| 01:52.54 | sawgood | why can't I just say to display the current Caller ID Number |
| 01:53.07 | p3nguin | You don't HAVE to set callerID. |
| 01:53.21 | p3nguin | If CALLERID is present in the call, the phone should display it. |
| 01:53.34 | sawgood | we might be talking about two different things here |
| 01:53.47 | p3nguin | If the phone doesn't display it, you're either breaking it during asterisk's processing of it, or it doesn't exist. |
| 01:53.53 | sawgood | If I call the number (with no special settings in the dialplan) |
| 01:54.05 | sawgood | I would see this on my softphone LCD screen |
| 01:54.08 | sawgood | caller's name |
| 01:54.11 | sawgood | callers' number |
| 01:54.14 | p3nguin | Okay. |
| 01:54.15 | sawgood | agree? |
| 01:54.24 | p3nguin | That's how it should work, yes. |
| 01:54.26 | sawgood | I want in addition to these two items .... |
| 01:54.49 | sawgood | a third line item which shows the number the calling party dialed to reach me |
| 01:55.18 | p3nguin | And how will that happen, since the phone doesn't have a third line for CID? |
| 01:55.24 | p3nguin | It has name and number. That's two. |
| 01:55.34 | sawgood | my business numbers = 408 555-1212 and/or 415-555-1212 |
| 01:55.44 | sawgood | I want to know which line the customer called (what did they dial) |
| 01:55.52 | sawgood | the 408 number or the (415) number |
| 01:56.00 | p3nguin | I've already covered how to do it. |
| 01:56.07 | p3nguin | Asked and answered. |
| 01:56.18 | sawgood | understood, but what I am saying is .... |
| 01:56.36 | sawgood | I got CS on the screen by itself (without the 408 or 415) number |
| 01:56.44 | sawgood | I also got the caller's name and number |
| 01:57.21 | p3nguin | Perhaps you could go study the book and understand what the values mean. |
| 01:57.51 | WIMPy | Are you just talking about that you see "CS" and want "408" or what? |
| 01:58.17 | sawgood | I would like this: (LCD to show) .... |
| 01:58.32 | sawgood | (408) 515-1212 / caller's name |
| 01:58.37 | sawgood | caller's number |
| 01:58.53 | WIMPy | Yikes! |
| 01:59.07 | sawgood | right now, I get: |
| 01:59.12 | sawgood | CS / caller's name |
| 01:59.15 | sawgood | caller's number |
| 01:59.32 | sawgood | it works ... but does not display the number (just the set = statement) |
| 01:59.40 | p3nguin | no shit |
| 01:59.48 | WIMPy | Now yu should really go and read that example over again and think about what you read. |
| 02:00.00 | sawgood | I understand what you wrote to me ... |
| 02:00.09 | sawgood | 100% understand ... believe me I do |
| 02:00.21 | sawgood | I was just wondering if there was a way to tell Asterisk ... |
| 02:00.22 | WIMPy | does not think so. |
| 02:00.43 | sawgood | put the CallerID number (pull it from the SIP invite) .... |
| 02:00.43 | p3nguin | If the phone has a third line in the CID display, and you know how to write data to it, be my guest. |
| 02:01.15 | sawgood | what is up with this third line ... you just made it work with the two lines |
| 02:01.31 | sawgood | so, instead of the set statement ... simply tell Asterisk to use the Actual caller ID number |
| 02:01.34 | WIMPy | p3nguin: I understand that he just wants another string constant. |
| 02:02.21 | p3nguin | I've enver seen a caller id with three lines. |
| 02:02.35 | sawgood | I have three lines now on a non Asterisk IP PBX |
| 02:02.49 | sawgood | actually four lines (the top line is the date/time) |
| 02:02.59 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
| 02:03.13 | p3nguin | What is in the fourth line? |
| 02:03.25 | sawgood | line 1 = date/time |
| 02:03.31 | sawgood | line 2 = caller's name |
| 02:03.35 | sawgood | line 3 = caller's number |
| 02:03.41 | sawgood | line 4 = the business number |
| 02:03.53 | sawgood | I would like to have this .... |
| 02:04.04 | sawgood | line 1 = the business line / the callers name |
| 02:04.15 | sawgood | line 2 - the caller's number |
| 02:04.24 | p3nguin | Do you have a phone that displays all four of these lines? |
| 02:04.33 | sawgood | yes, I do ... its an Allworx phone |
| 02:05.16 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 02:08.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:12.16 | sawgood | Set(CALLERID(name)=TF-${CALLERID(name)}) |
| 02:12.20 | sawgood | so instead of Set |
| 02:12.27 | sawgood | could I do something like this: |
| 02:12.43 | sawgood | (CALLERID(num)-${CALLERID(name)}) |
| 02:13.03 | sawgood | but of course, have the CALLER ID number be the business LINE ... not the caller ID of the customer |
| 02:15.20 | p3nguin | If there's enough room on the display, that could work. |
| 02:15.32 | sawgood | Is that the right syntax? |
| 02:15.43 | p3nguin | I'll work something out for you in the pastebin. |
| 02:15.50 | sawgood | thank you so much! |
| 02:17.43 | *** join/#asterisk eppigy (~eppigy@c-76-105-72-69.hsd1.ga.comcast.net) |
| 02:22.22 | p3nguin | sawgood: What about something like this? http://pastebin.com/hPLJPKiQ |
| 02:23.45 | p3nguin | The caller ID name line would show name/num, and the called ID num line would show the extension that was called. |
| 02:27.31 | p3nguin | What do you think about this method? |
| 02:27.46 | sawgood | I have two lines showing up on the LCD |
| 02:27.59 | sawgood | line 1 = caller's number / caller's name |
| 02:28.07 | sawgood | line 2 = callers's number (again) |
| 02:28.13 | p3nguin | erm |
| 02:28.22 | sawgood | If I could make line 1 be business line / caller's name |
| 02:28.24 | p3nguin | That's not right. |
| 02:28.44 | p3nguin | That's easy to change. |
| 02:28.47 | sawgood | actually here is what I get (small mistake) |
| 02:28.58 | sawgood | line 1 = caller ID name / caller ID number |
| 02:29.05 | sawgood | line 2 = caller ID number |
| 02:29.43 | p3nguin | http://pastebin.com/SkikZC2c |
| 02:32.04 | sawgood | see, the problem here is the callerID(num) is the 'cell' phone and not the business line |
| 02:32.32 | sawgood | Is there a way to say CallerID(SIP from) / CallerID(name) |
| 02:33.14 | sawgood | or ... CallerID(SIP TO)/CallerID(name) |
| 02:33.20 | sawgood | incase I have it backwards |
| 02:33.31 | p3nguin | I've used ${EXTEN} |
| 02:33.37 | p3nguin | which is the extension that was called. |
| 02:33.49 | p3nguin | Not the number called from. |
| 02:34.37 | p3nguin | in this line... |
| 02:34.39 | p3nguin | exten => 16509032154,1,Set(CALLERID(name)=${EXTEN}/${CALLERID(name)}) |
| 02:34.58 | p3nguin | ${EXTEN} should be 16509032154 |
| 02:35.21 | sawgood | I think it is showing up as the cell phone's number I am calling from |
| 02:35.26 | p3nguin | So the CID display would say 16509032154/Joe Blow on the first line. |
| 02:35.38 | sawgood | cell phone/joe blow |
| 02:35.59 | p3nguin | What number are you dialing on your cell phone? |
| 02:36.27 | sawgood | 16509032154 |
| 02:36.58 | p3nguin | In the CLI, run "dialplan show 16509032154@incoming-custom" and paste the output. |
| 02:39.30 | sawgood | http://pastebin.com/C0aR73xJ |
| 02:39.46 | p3nguin | And there's why it isn't working. |
| 02:39.47 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:40.02 | p3nguin | You didn't save and reload after the last change. |
| 02:42.36 | *** join/#asterisk d00gster (~dt@77.30.47.220) |
| 02:43.07 | sawgood | each time, I do a stop now |
| 02:43.10 | sawgood | then asterisk |
| 02:43.14 | sawgood | then asterisk -r |
| 02:50.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:52.42 | sawgood | I think the answer might be in the (CALLERID)(DNID) statements |
| 02:52.50 | sawgood | I am reading it under show function CALLERID |
| 02:54.33 | p3nguin | Yeah, you shouldn't be stopping and restarting asterisk. |
| 02:54.49 | p3nguin | When you make a change to extensions.conf, you should save it and then run dialplan reload on the CLI. |
| 02:54.49 | sawgood | dialplan reload didn't seem to work |
| 02:54.58 | sawgood | ok |
| 02:55.29 | p3nguin | If it "doesn't work," find out why. |
| 02:55.41 | sawgood | ${DNID} - Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)}) |
| 02:56.05 | p3nguin | If there is any dnid in the call, it might work. |
| 02:56.32 | *** join/#asterisk eppigy (~eppigy@c-76-105-72-69.hsd1.ga.comcast.net) |
| 02:56.35 | sawgood | Is there a command called Get instead of Set |
| 02:56.47 | p3nguin | Verbose() |
| 02:56.49 | sawgood | Get${CALLERID(dnid)}) |
| 02:57.01 | ChannelZ | channel vars are just ${WHATEVER} if using them in your dialplan |
| 02:57.05 | p3nguin | Verbose(${CALLERID(dnid)}) |
| 02:57.29 | sawgood | exten => 16503510134,n,Set(CALLERID(num)=${EXTEN}) |
| 02:58.10 | p3nguin | That would set the caller ID number to the extension that was called. |
| 02:58.24 | p3nguin | Change ${EXTEN} to ${CALLERID(dnid)} |
| 02:58.30 | sawgood | exten => 16503510134,n,Verbose(CALLERID(dnid)}) |
| 02:58.50 | p3nguin | That will output it on the CLI for you. |
| 02:59.09 | p3nguin | but you have to fix the typo first. |
| 02:59.16 | sawgood | so, what would go in extensions.conf? |
| 02:59.26 | p3nguin | What do you want to do now? |
| 02:59.51 | sawgood | exten => 16503510134,n,(CALLERID(dnid)}) |
| 02:59.54 | sawgood | is that right? |
| 02:59.58 | p3nguin | no |
| 03:00.09 | p3nguin | (2157.29) <sawgood> exten => 16503510134,n,Set(CALLERID(num)=${EXTEN}) |
| 03:00.09 | p3nguin | (2158.10) <p3nguin> That would set the caller ID number to the extension that was called. |
| 03:00.13 | p3nguin | (2158.24) <p3nguin> Change ${EXTEN} to ${CALLERID(dnid)} |
| 03:00.31 | sawgood | ok |
| 03:02.33 | sawgood | nothing happened |
| 03:02.41 | sawgood | oh well, time for a break for me .. |
| 03:02.51 | sawgood | I have to walk away for a while ... I'll come back to this later |
| 03:02.57 | sawgood | thanks for your time though! |
| 03:03.03 | p3nguin | Read the book while you're gone. |
| 03:05.00 | ChannelZ | I doubt it |
| 03:06.07 | *** join/#asterisk eppigy (~eppigy@c-76-105-72-69.hsd1.ga.comcast.net) |
| 03:32.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:48.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:06.16 | *** join/#asterisk jmacz (~jmacz@190.27.183.52) |
| 04:19.17 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 04:31.54 | vk4akp | Hey Guys. |
| 04:32.29 | vk4akp | If I have an extension goign to a TDM card. DO I have to pass / set the caller ID from teh original caller some how when teh line is transfered to teh TDM400P? |
| 04:41.08 | ChannelZ | huh? |
| 04:41.18 | vk4akp | OK. |
| 04:41.24 | ChannelZ | what is the path of the call? Vague question. |
| 04:41.37 | vk4akp | When a person dials my box they can sellect 1 to go to the TDM400P line 1 |
| 04:41.46 | vk4akp | BUt it never shows their caller ID . |
| 04:41.48 | ChannelZ | Where is the call coming FROM? |
| 04:41.57 | vk4akp | Do I need to add soemthign to teh extensions to pass the caller ID on to the phone? |
| 04:42.07 | vk4akp | Outside via SIP. |
| 04:42.35 | p3nguin | Caller ID data should be part of the call already. |
| 04:42.49 | vk4akp | OK. So what am I doing wrong. |
| 04:42.50 | ChannelZ | OK.. well you'd have to configure the FXS line to transmit Caller ID to the analog phone |
| 04:42.58 | vk4akp | When teh phone rings it just says CALL. nothing else. |
| 04:43.19 | vk4akp | OK. Where do I configure? (What file) and how please? |
| 04:43.28 | ChannelZ | which I *think* Asterisk/the TDM supports but I'm not sure, the only FXS I use is a fax machine and I don't care about caller ID |
| 04:43.59 | vk4akp | Caller ID is important to me as I have had problems with abuse calls and hackers at eh box. |
| 04:44.06 | ChannelZ | well it'd be something in chan_dahdi.conf I'd imagine |
| 04:44.08 | vk4akp | So I need to see if it is a friend and to answer the phone. |
| 04:44.51 | vk4akp | exten => 1,1,Dial(Zap/1,10,thg) |
| 04:45.06 | vk4akp | THis is what I have in my extensions.conf to pass the call to the phone. |
| 04:45.21 | *** join/#asterisk Milad (~milad@unaffiliated/slackark) |
| 04:45.24 | ChannelZ | oh, zap. Well zapata.conf then. Do you have 'usecallerid' turned on for that channel? |
| 04:45.41 | vk4akp | OK Let me look. |
| 04:46.10 | *** join/#asterisk jhirley (~jhirley@adsl-6-0-248.mia.bellsouth.net) |
| 04:48.21 | vk4akp | http://dpaste.com/198049/ |
| 04:48.28 | vk4akp | HUmm I think you have found the problem maybe. |
| 04:48.40 | vk4akp | I do not understand the config properly. But you can see it there. |
| 04:48.51 | vk4akp | I think it needs anothe rentry for the first TDM line? |
| 04:49.04 | vk4akp | I see the entry for the second TDM line is set (PayPhone). |
| 04:49.56 | ChannelZ | that's just a static CID string, associating a name/number with that channel for outgoing calls |
| 04:50.32 | vk4akp | Yes. But channel 2 has usecallerid=yes. |
| 04:50.44 | vk4akp | Channel 1 does not have any such comments. |
| 04:50.45 | ChannelZ | are you in a non-US country? |
| 04:50.53 | vk4akp | Australia. |
| 04:50.57 | ChannelZ | Yes it does, right up top. |
| 04:51.09 | vk4akp | But this shoudlnt' matter. The TDM is only plugged into a atelephone. Not a PSTN. |
| 04:51.14 | ChannelZ | Parameters flow downwards and apply to channels below them |
| 04:51.50 | vk4akp | So usecallerid=yes at the top is applied to everythign below? |
| 04:52.26 | ChannelZ | yes |
| 04:54.02 | vk4akp | OK. I added the same lines in anyhow jsut in case to see if it helps. |
| 04:54.27 | ChannelZ | I don't really know the answer to your question. It should be able to transmit CID to the analog phone but all I can find is some vague postings that seem to imply it doesn't work on non-US under Zaptel. I don't know if that's really true, or if it was fixed in another version of Zap, or in Dahdi, or if it can be fixed, etc. |
| 04:54.29 | vk4akp | So to reload the file do I just RELOAD from CLI? Or is ther emore needed for the TDM to re-read the file? |
| 04:55.40 | vk4akp | I think the NON-USA thing is more to do with the PSTN. Not Handsets. Handsets understand USA / .EU / .AU etc for Caller ID. |
| 04:57.32 | ChannelZ | So you're using 'bell' |
| 04:57.37 | ChannelZ | CID signalling? |
| 04:58.33 | vk4akp | OK. I even have two phones her eon teh TDM card. One phone is a USA PayPhone (FatBoy) so it does not get the caller ID either. |
| 04:59.00 | vk4akp | Also interesting when I ring out from the TDM card ASterisk is not taking the caller ID as set. It sets Caller ID to `Asterisk'. |
| 04:59.05 | vk4akp | So some more mysteries. |
| 04:59.18 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 05:00.07 | ChannelZ | your quoting might be messing that up, and I'm assuming you mean from the 2nd channel (your 'pay phone') |
| 05:00.29 | ChannelZ | callerid=Foo <number> |
| 05:00.45 | vk4akp | I have set caller ID on both now 9Name / number). Do I need to reload or restart Asterisk for that to take effect? |
| 05:01.05 | ChannelZ | yes |
| 05:03.44 | vk4akp | OK That fixed outgoing Caller-ID (Thanks). Now when I call out fro mteh TDM it sends the set ID. |
| 05:03.58 | vk4akp | But still Incoming, no caller-ID . It just says CALL. |
| 05:04.28 | ChannelZ | What is your 'loadzone' set to in /etc/zaptel.conf |
| 05:05.25 | vk4akp | Ignoring content, Ignoring signalling, Ignoring signalling, [May 23 14:47:40] WARNING[13288]: chan_dahdi.c:4257 dahdi_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 05:05.31 | vk4akp | Some Clues maybe! :) |
| 05:05.56 | ChannelZ | wait, are you using DAHDI or Zaptel?? |
| 05:06.03 | vk4akp | loadzone = au |
| 05:06.03 | vk4akp | defaultzone = au |
| 05:06.26 | vk4akp | Used to be Zaptel. But I believe now the newer Asterisk uses Dahdi. |
| 05:06.55 | ChannelZ | It uses whatever you compiled it with |
| 05:07.47 | ChannelZ | which it sounds to me like you've somehow managed to do both which can't be good |
| 05:08.29 | ChannelZ | You said you're Dial()ing Zap/* so I'm assuming you really are using Zaptel, in which case * shouldn't be complaining about chan_dahdi |
| 05:08.30 | vk4akp | OK |
| 05:09.11 | vk4akp | The system used to run 1.4.?? using Zaptel. Then teh Radio guy updated teh Asterisk to a newer SVN release for the APP_RPT stuff. And I think he changed it over to Dahdi then. |
| 05:09.54 | ChannelZ | so when you just 'fixed' your caller ID on those two channels, what file were you editing? |
| 05:10.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:10.59 | ChannelZ | and you said a little bit ago: "<vk4akp> exten => 1,1,Dial(Zap/1,10,thg)" - which means you're dialing Zaptel |
| 05:11.26 | ChannelZ | And god only knows what drivers you're running if you don't know what package you're even running |
| 05:12.55 | vk4akp | OK. I can answer all thes questions. Just give me time to find the answers. |
| 05:13.02 | vk4akp | The file I edited was the one you said to. |
| 05:13.49 | vk4akp | I think it was /etc/zapata.conf |
| 05:14.07 | vk4akp | Asterisk SVN--r588M built by root @ shazam on a i686 running Linux on 2010-04-01 14:19:13 UTC |
| 05:14.17 | ChannelZ | ok. So the error message you posted above, when did that occur? |
| 05:14.44 | vk4akp | When I tried to ring the extension with the TDM card phone. |
| 05:15.14 | vk4akp | I also notice there is a /etc/asterisk/chan_dahdi.conf file there . |
| 05:15.15 | ChannelZ | ok well it's a wonder that anything is running at all |
| 05:15.44 | ChannelZ | if you type "help dahdi" on the console, do you get a list of various DAHDI commands? |
| 05:15.50 | vk4akp | http://dpaste.com/198056/ |
| 05:16.23 | vk4akp | Yes. Heaps of Dahdi commands listed. |
| 05:16.34 | ChannelZ | And if you type "help zap" on the console you get commands as well? |
| 05:17.02 | vk4akp | help zap |
| 05:17.02 | vk4akp | No such command 'zap'. |
| 05:17.20 | ChannelZ | uhm |
| 05:17.31 | vk4akp | OK I see where this is heading. |
| 05:17.45 | vk4akp | So do I not use Zap in teh extensions file? |
| 05:17.53 | vk4akp | Do I use Dahdi or something? |
| 05:18.15 | ChannelZ | You have some sort of fucked up zombie system there. |
| 05:19.18 | vk4akp | dahdi show channel 1 shows all the info. |
| 05:19.51 | ChannelZ | The point is you shouldn't be running both, and I'm not even sure how your system is working at all.. if you said you edited zapata.conf and it changed your running config, yet you are apparently running DAHDI.. yet you say you are dialing via Zap... |
| 05:20.08 | ChannelZ | Unless your zapata.conf is symlinked to chan_dahdi or something |
| 05:20.38 | vk4akp | No nothing fancy like that. |
| 05:20.59 | vk4akp | Chan_Dahdi seems to be reading info from the /etc/zapata.conf ?? |
| 05:21.13 | ChannelZ | not to my knowledge |
| 05:21.31 | vk4akp | Has to. |
| 05:21.47 | vk4akp | I set the caller ID in there and the CHan_Dahdi knows this caller ID now. It's on my screen. |
| 05:22.14 | ChannelZ | I've never heard of it. Maybe the guy who 'patched' your system did some other fuckery you don't know about |
| 05:22.34 | vk4akp | Don't think so. |
| 05:22.39 | vk4akp | He is a pretty smart guy. |
| 05:22.48 | ChannelZ | in which case I can't help you because what you've told me so far makes to sense, to me. |
| 05:22.54 | vk4akp | He is the guy that wrote APP_RPT |
| 05:23.08 | vk4akp | Ah. But you are doing so well. |
| 05:23.17 | vk4akp | I understand so much more now since you talked to me. |
| 05:23.20 | ChannelZ | The only thing I can offer is a message from 2006: "I encountered the same problem and after a bit of digging discovered that the TDM400 FXS ports apparently only support US caller ID, unlike the FXO port." |
| 05:23.27 | vk4akp | And th emystery is 99% there I think!. :) |
| 05:24.03 | vk4akp | I think you are on teh right track with the Zaptel vs Dahdi thing. |
| 05:24.07 | ChannelZ | The mystery is how your system is running at all. You're positive the Dial() command you posted awhile back is what you're using? |
| 05:24.19 | vk4akp | I think ther eis a config file somewhere to do with DahDi that needs adjusting. |
| 05:24.35 | vk4akp | Yes. I cut that from my extensions.conf |
| 05:24.43 | vk4akp | Could I try aa Dahdi command instead? |
| 05:25.03 | ChannelZ | And when that command runs, your phone rings? |
| 05:25.22 | WIMPy | I didn't follow from the start, but what does 'core show channeltypes' say? |
| 05:26.44 | vk4akp | Zap DAHDI Telephony Driver w/PRI |
| 05:27.08 | vk4akp | Radio USB (CM108) Radio Channel Driver |
| 05:27.17 | WIMPy | ok |
| 05:27.21 | ChannelZ | eh? Maybe you are using some old janky SVN version when Zap was in transition to DAHDI still... wtf |
| 05:27.22 | WIMPy | spookey |
| 05:27.28 | vk4akp | PLus more. |
| 05:27.53 | WIMPy | That's the only thing that springt to my mind as well. |
| 05:28.03 | vk4akp | There are lots of configs for Dahdi that I should look at I think. |
| 05:28.16 | ChannelZ | Is there a reason you're running this system the way it is and not wiped it out and started over? |
| 05:28.30 | vk4akp | But I want to try changing my extension.conf. I need to find the format for the dahdi in there now . |
| 05:28.45 | vk4akp | OMG Wipe it out!> Nooooo!!!!!!!!!!!!!!!! Years of work. |
| 05:28.50 | vk4akp | This system is very special. |
| 05:28.56 | ChannelZ | Yeah it clearly functions well |
| 05:29.09 | ChannelZ | it's just DAHDI/xxx instead of Zap/xxx but that's the least of your problems |
| 05:29.18 | vk4akp | It run's extensions and networks on a number of different AMateur Radio and public Radio networks world wide |
| 05:29.30 | ChannelZ | At the core I think your main issue is possibly a limitation of your zone and the TDM400 and FXS. Trifecta. |
| 05:29.39 | vk4akp | OK I try Dahdi/xxx for the fun of it. :) |
| 05:31.42 | ChannelZ | I'd change your loadzone/defaultzone to us and see what happens for the hell of it based on what i posted above, but beyond that I've no idea |
| 05:32.02 | vk4akp | [May 23 15:15:54] WARNING[14383]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'Dahdi' (cause 66 - Channel not implemented) |
| 05:32.26 | ChannelZ | yeah that's cuz you have a frankensystem |
| 05:32.33 | vk4akp | channel.c:3051 ast_request: No channel type registered for 'Dahdi', : app_dial.c:1242 dial_exec_full: Unable to create channel of type 'Dahdi' (cause 66 - Channel not implemented) |
| 05:33.08 | WIMPy | vk4akp: I'd try a clean install if any possible. |
| 05:33.21 | vk4akp | No not possible. |
| 05:33.25 | ChannelZ | My guess is either your 'smart friend' hacked your setup to use 'Zap' instead of 'DAHDI' because he was too lazy to change all the references to Zap in the dialplan, or you're running some bizzarre interim version of Zaptel/DAHDI |
| 05:33.29 | vk4akp | Very big not possible. |
| 05:33.52 | ChannelZ | well good luck then |
| 05:34.11 | vk4akp | It is a special SVN versio nwith ots of broken Asterisk stuff fixed so the radio (APP_RPT) stuff can work. |
| 05:34.17 | WIMPy | In that case I'd try to replace it. |
| 05:34.29 | vk4akp | Replace what with what? |
| 05:35.03 | WIMPy | A new system. |
| 05:35.32 | WIMPy | To me it sound af if yo#ll ne haunted for as long as it's running. |
| 05:35.53 | WIMPy | f***. really bad typing :-( |
| 05:36.51 | vk4akp | It doesn't matter what revision I run. (I have run many). Sooner or later ther eare always similar issues. IN the end it's always worked out to be a config issue. Not a release issue. But for some reason people always balme the release, the version of Linux, reinstall, or some other big job that doesn't address the issues. |
| 05:36.58 | vk4akp | It will be fixable. I just need to find out how. |
| 05:37.09 | ChannelZ | Or let this POS janky box run on it's own in a corner doing whatever it's supposed to do with this radio thing, and do your 'Real Work' on a different computer & asterisk setup that isn't jacked |
| 05:37.44 | WIMPy | Sounds like a plan. |
| 05:38.03 | vk4akp | LOL. I get that answer too. A lot. LOL. RUn 10 boxes instead of one. No thanks. I'd hate to see your electricity bill. |
| 05:38.16 | vk4akp | But not to worry. |
| 05:38.23 | vk4akp | You have given me some directions to lookinto. |
| 05:38.35 | vk4akp | I now understand there is so cross issue between Zaptel & Dahdi. |
| 05:38.56 | vk4akp | so ==some* |
| 05:39.17 | ChannelZ | Well you're stuck in the past running a hacked-up customized bisexual system, and you're wondering why now certain other things aren't working right. Good luck. |
| 05:39.42 | ChannelZ | There's a cross issue between Zaptel and DAHDI... ON YOUR BOX |
| 05:41.06 | vk4akp | Well all I can say is that this so called hacked up system has run much better then any true Asterisk release. |
| 05:41.32 | vk4akp | THe guys released their own SVN version because ASterisk would fix *Known* issues in teh code. So they had to do ti for them. |
| 05:42.15 | WIMPy | We all do that from time to time. |
| 05:42.24 | WIMPy | That's the way it works. |
| 05:43.07 | vk4akp | Anyhow. I have ideas now where to look. So I will move forward with this. Thanks for your input. |
| 05:43.42 | ChannelZ | I still think your issue is possibly a limitation of the TDM400 |
| 05:44.03 | vk4akp | Humm. Not convinced yet. |
| 05:44.30 | vk4akp | One of the main specs of the card is the ability to pass caller ID. So doubtfull it doesn't do it. Or they would have big problems for false advertising. |
| 05:44.45 | ChannelZ | http://www.voipuser.org/forum_topic_3727.html |
| 05:46.15 | vk4akp | OK |
| 05:46.22 | vk4akp | I can go to zone US that's not a problem. |
| 05:46.24 | vk4akp | Hang ten. |
| 05:48.11 | ChannelZ | There is also this: http://readlist.com/lists/lists.digium.com/asterisk-users/15/75484.html |
| 05:48.54 | ChannelZ | It could be a bug or something in the zaptel drivers fixed at some point in a later release/DAHDI. But we really have no idea what version you're running |
| 05:49.32 | vk4akp | Asterisk SVN--r588M built by root @ shazam on a i686 running Linux on 2010-04-01 14:19:13 UTC |
| 05:50.03 | WIMPy | No. Version of zaptel/dahdi. |
| 05:50.29 | ChannelZ | even that says nothing. What is SVN-r588M? |
| 05:58.46 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 05:59.32 | vk4akp | OK. Where should I look for teh Zaptel/dahdi version |
| 06:00.31 | carrar | version.h |
| 06:00.35 | carrar | in your source dir |
| 06:01.48 | vk4akp | OK I will have to wokr out which one ws used. |
| 06:01.57 | vk4akp | There are a few there still from previous upgrades. |
| 06:02.10 | vk4akp | OUt of interest. Going to US signalling got rid of two fo teh errors. |
| 06:02.15 | vk4akp | Now only getting this . |
| 06:02.18 | vk4akp | [May 23 15:45:14] WARNING[15627]: chan_dahdi.c:4257 dahdi_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 06:03.10 | carrar | so put a wait in there |
| 06:04.18 | carrar | and go upgrade to the latest version of Asterisk: http://www.asterisk.org/downloads |
| 06:04.46 | carrar | & DAHDI complete |
| 06:07.10 | vk4akp | Oh a wait. OK. HUmm. So a Wait(1) before Dial (Zap... I can try. But I can't see that doing a lot. |
| 06:07.35 | vk4akp | And again. I can't run the latest version of ASterisk it won't run the APP_RPT stuff for the radio. |
| 06:07.51 | carrar | Go get the latest APP_RPT |
| 06:08.00 | carrar | or have them upgrade it |
| 06:08.52 | vk4akp | It is the latest. |
| 06:09.00 | vk4akp | They only updated my system a week or so back. |
| 06:10.26 | carrar | Why don't 'They' fix it then? |
| 06:11.00 | vk4akp | Fix what? |
| 06:11.04 | carrar | So they update your system and broke it in the process? |
| 06:11.08 | vk4akp | I will repeat. |
| 06:11.53 | vk4akp | They have to release their own Asterisk release because there are things broken in Asterisk that Asterisk refuses to fix. This is necessary to allow teh APP_RPT and USB_Radio modules to function. |
| 06:12.33 | vk4akp | Oh and the Wait(1) did nothing. |
| 06:12.41 | carrar | Whats broken? |
| 06:12.48 | carrar | in asterisk |
| 06:12.53 | WIMPy | Or it doesn't work *because* some bugs actually were fixed? |
| 06:13.00 | carrar | heh |
| 06:13.26 | vk4akp | YOu would have to talk to teh APP_RPT guys. He didn't go into specifics. But I'm sure if you asked he would. |
| 06:14.23 | carrar | all the issues with 'APP_RPT' in them seem clsoed |
| 06:16.16 | carrar | oh well, stuck in a hard place if you are locked into old broken versions |
| 06:17.42 | carrar | Better off with a ACC 850 repeater controller :) |
| 06:23.17 | vk4akp | I don't think it's broken. Just because there is a config issue with the TDM card. I had similar problems on standard Asterisk releases in the past before I started with the radio stuff. |
| 06:25.02 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 06:25.45 | sawgood | exten => 16509033116,n,Noop($(CALLERID(dnid}) |
| 06:25.56 | sawgood | what does the 'term' Noop 'do' in this statement? |
| 06:26.02 | p3nguin | nothing |
| 06:26.15 | p3nguin | NoOp() does absolutely nothing. It means No Operation. |
| 06:26.31 | p3nguin | And your syntax is still bad. |
| 06:26.38 | carrar | It prints the contents of the variable $(CALLERID(dnid} |
| 06:26.44 | carrar | ) |
| 06:26.47 | sawgood | hmmmm |
| 06:26.56 | sawgood | Why would someone want to use NoOP then? |
| 06:27.11 | p3nguin | It will print in the verbose info. |
| 06:27.20 | p3nguin | It has to run nothing, so it has output. |
| 06:27.21 | carrar | Obviously they want to see what the variable is |
| 06:27.31 | sawgood | ok ... |
| 06:27.37 | sawgood | what was wrong with the statement |
| 06:27.43 | sawgood | exten => 16509033116,n,Noop($(CALLERID(dnid}) |
| 06:27.48 | p3nguin | It's a more obscure form of Verbose(). |
| 06:27.52 | sawgood | oh ok |
| 06:27.52 | carrar | missing a { |
| 06:27.57 | sawgood | thank you |
| 06:27.58 | p3nguin | $(CALLERID(dnid} = fail |
| 06:28.18 | p3nguin | ${CALLERID(dnid)} |
| 06:28.26 | sawgood | oh ... thank you |
| 06:28.40 | sawgood | what does } key do? |
| 06:28.48 | carrar | ~book |
| 06:28.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:29.28 | sawgood | ,1,Set(CALLERID(name)=TECH-${CALLERID(name)}) |
| 06:29.36 | p3nguin | CALLERID(dnid) is the CALLERID() function with the dnid datatype. |
| 06:29.46 | sawgood | In this statement it does not start off with { command before CALLERID |
| 06:29.53 | sawgood | yet, it is working |
| 06:29.58 | p3nguin | ${CALLERID(dnid)} is how you parse the value of it. |
| 06:30.14 | sawgood | oh ... got it |
| 06:30.39 | sawgood | the $ is an AGI command |
| 06:30.48 | p3nguin | Are you asking me or telling me? |
| 06:31.24 | p3nguin | The $ indicates you're using the value of a variable. It has fuckall to do with AGI. |
| 06:38.01 | *** join/#asterisk fofware (fabian@190.7.25.160) |
| 06:39.44 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:47.21 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 06:53.14 | *** join/#asterisk NV` (NV@neo-vortex.net) |
| 06:54.24 | NV` | heyias, I have an asterisk box I'm setting up with a DID, It can make outgoing calls fine, and if i dial 7777 on an extension to simulate an incoming call it works fine (goes to the queue i set up and rings all the extensions i told it to), however if i dial the actual DID from the PSTN, i get number not in service |
| 06:54.57 | NV` | I've run asterisk -r and run sip set debug on, I can see some SIP packets when the call comes in, but for some reason asterisk is dumping it |
| 06:55.00 | NV` | thoughts? |
| 06:56.04 | sawgood | 16509033140,1,Verbose(${SIP_HEADER(TO)}) |
| 06:56.08 | WIMPy | Better turn on verbose and debug and see where the call is (not) going. |
| 06:56.14 | sawgood | I got this 'working' where I see it in the CLI .... |
| 06:56.15 | dzup | you need to see what the incoming strin is, then applay that in the incoming rules |
| 06:56.30 | sawgood | how do I 'change' this so the output comes up on the softphone display instead of the console? |
| 06:58.24 | NV` | WIMPy: i get -- Executing [s@from-sip-external:4] Wait("SIP/<DID>-0952a3f0", "2") in new stack |
| 06:58.50 | NV` | then after 2 seconds, it plays not in service (and shows -- Executing [s@from-sip-external:5] Playback("SIP/<DID>-0952a3f0", "ss-noservice") in new stack) |
| 07:00.20 | WIMPy | So now you know it's going to s in from-sip-external. |
| 07:00.41 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 07:00.42 | NV` | where is that? |
| 07:01.36 | NV` | shouldn't that trigger the any cid / any did inbound route? |
| 07:03.03 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:04.29 | sawgood | 16509033140,1,Verbose(${SIP_HEADER(TO)}) |
| 07:04.54 | sawgood | any ideas on what to 'change' Verbose to ... so it will show up on the LCD of the softphone |
| 07:05.03 | sawgood | I tried 16509033140,1,Set(${SIP_HEADER(TO)}) |
| 07:05.55 | ChannelZ | that does nothing |
| 07:06.09 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 07:06.31 | sawgood | exactly |
| 07:06.40 | ChannelZ | and SIP_HEADER is a read-only function |
| 07:06.46 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 07:07.17 | sawgood | I would like for the TO field to be displayed on the phone on the same line as the caller ID name or number |
| 07:07.21 | sawgood | if possible |
| 07:07.33 | ChannelZ | I feel a cyclic conversation going on |
| 07:07.35 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 07:07.47 | ChannelZ | This was all answered like 5 hours ago wasn't it |
| 07:07.55 | sawgood | some what ... |
| 07:07.58 | sawgood | some answers ... |
| 07:08.15 | sawgood | I can put in a 'fake' name for caller ID num ... |
| 07:08.38 | carrar | impossible! |
| 07:08.47 | ChannelZ | JABOOTY |
| 07:09.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:09.20 | sawgood | I have it working in a 'round about' way ... but not in the exact fashion I would like |
| 07:10.12 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 07:10.27 | sawgood | exten => 16509033143,1,Set(CALLERID(name)=TECH-${CALLERID(name)}) |
| 07:10.32 | sawgood | This works ... |
| 07:11.05 | sawgood | If I wanted to 'change' TECH to the SIP TO field number, it works |
| 07:11.18 | sawgood | I get caller name, caller number and the DNID number |
| 07:12.13 | carrar | Just answer the phone and talk |
| 07:12.22 | ChannelZ | isn't 16509033143 the DID? |
| 07:12.30 | NV` | <PROTECTED> |
| 07:12.41 | sawgood | yes |
| 07:12.45 | sawgood | that is the DID |
| 07:18.02 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 07:19.28 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:20.31 | *** join/#asterisk vk4akp (~Ken@c114-77-251-186.ipswc3.qld.optusnet.com.au) |
| 07:21.50 | *** join/#asterisk proserion (~info@p57B5CAAF.dip.t-dialin.net) |
| 07:22.52 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 07:23.58 | proserion | Hey ppl, I am about to set up an astreix for the first time. What I need to know at first is, what service provider shall i use? Would you suggest sipgate, or is there any better? |
| 07:24.58 | carrar | ~itsp |
| 07:25.00 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 07:29.54 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 07:30.06 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 07:39.37 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
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| 07:50.56 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 07:51.27 | proserion | Is it possible get a German local number with them? (itsp) |
| 07:52.16 | ChannelZ | depends on the itsp |
| 07:59.26 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 07:59.26 | *** mode/#asterisk [+o angler] by ChanServ |
| 08:08.38 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 08:13.15 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 08:14.31 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 08:15.28 | Micc_ | If I get the svn branch 1.6.2 will I be getting the latest release version of 1.6.2? When I do core show version it shows an 1.6.2-r##### |
| 08:16.02 | Micc_ | Asterisk SVN-branch-1.6.2-r265172 |
| 08:16.34 | Micc_ | how can I find out which release that is? Or is it not an official release? Is there a way to get official releases with svn? |
| 08:19.11 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 08:21.19 | kaldemar | Micc_: svn branch is not a release, that's a snapshot of the branch. |
| 08:21.38 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 08:22.04 | kaldemar | Micc_: these are releases: http://svn.digium.com/svn/asterisk/tags/ |
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| 09:17.04 | Micc_ | do I need to reboot after installing a new version of dahdi? or is there a way to reload the kernel modules? |
| 09:20.36 | ChannelZ | stop asterisk, then stop the drivers - usually easy with the init script |
| 09:20.39 | ChannelZ | /etc/init.d/dahdi stop |
| 09:20.54 | ChannelZ | then restart them and start asterisk |
| 09:24.48 | Micc_ | ok |
| 09:25.04 | Micc_ | thats what I thought, but wanted to be sure. |
| 09:26.43 | ChannelZ | Should work fine |
| 09:28.36 | *** join/#asterisk BANSAL (~BANSAL@117.199.116.170) |
| 09:30.02 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 09:30.28 | drmessano | Odd |
| 09:30.45 | Micc_ | uh oh! |
| 09:30.47 | Micc_ | chan_dahdi.c:9815 pri_create_trunkgroup: Failed go get span information on channel 24 (span 1): Inappropriate ioctl for device |
| 09:32.07 | Micc_ | did something change with the config files between earlier dahdi versions? |
| 09:32.37 | drmessano | Apparently I can't get Dahdi to load here, either |
| 09:37.46 | drmessano | What specifically does Asterisk need to have present to compile properly with Dahdi |
| 09:38.35 | Micc_ | I had previous version of dahdi running fine. |
| 09:38.47 | Micc_ | dahdi_cfg -v looks right |
| 09:39.17 | drmessano | Dahdi wasn't running here.. I started dahdi and reinstalled Asterisk, and it seems to be fine now |
| 09:40.05 | drmessano | But that's odd.. Dahdi needs to be running when I configure and/or make? |
| 09:40.57 | Micc_ | what order do I build asterisk/dahdi in? make install dahdi first I thought then asterisk |
| 09:41.34 | Micc_ | I'm scared to reboot the server. I might have to go up to datacenter if it doesn't come back up. |
| 09:41.43 | Micc_ | its an hour drive. |
| 09:42.30 | Micc_ | This is a production server, but its not the main server, just dialout and I hope my main customer a 24 hour nurse hotline will failover to other dialout. |
| 09:42.53 | *** join/#asterisk rare1980_ (rare1980_@115.186.16.201) |
| 09:43.53 | Micc_ | do you think a reboot might help? |
| 09:45.13 | Micc_ | I was running 2.2.1 dahdi before. now trying to use 2.3.0 |
| 09:46.06 | Micc_ | I also upgraded asterisk to 1.6.2.7 at the same time. |
| 09:46.24 | Micc_ | I'll try going back to 1.6.2.6 but I don't see any reason why that would make any difference. |
| 09:47.21 | emora | can someone suggest a quick way to stop a SIP attack from amazonews at ip 174.41.188.185 ? |
| 09:47.37 | emora | it is flodding my system |
| 09:49.31 | emora | I did an iptables -A put its appending the rule to the end of the table |
| 09:51.21 | emora | ok. Got it! I just did iptables -I instead of -A |
| 09:51.37 | emora | . |
| 09:52.02 | emora | Can anyone suggest a tool to detect these attacks and create a rule automatically? |
| 09:53.33 | *** join/#asterisk Trixboxer (~praju@115.113.145.85) |
| 09:53.58 | florz | emora: such a tool would be something between pointless and dangerous |
| 10:00.48 | Micc_ | this is not good. |
| 10:03.18 | Micc_ | what does this mean? /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting |
| 10:04.07 | Micc_ | ok. looks like i'm up again with old dahdi 2.2.1 |
| 10:04.38 | Micc_ | lsmod shows lots of use where it did not before. |
| 10:05.07 | Micc_ | makes me think a reboot was necessary for new version to load properly. |
| 10:07.55 | Micc_ | ok, all is good with 2.3.0 now I think |
| 10:08.07 | Micc_ | dahdi show version still shows no version number. but dahdi_cfg -v shows tools version 2.3 |
| 10:08.53 | Micc_ | well that was fun. |
| 10:14.19 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.227.219.dsl.dyn.forthnet.gr) |
| 10:20.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:23.38 | emora | florz, its obvious your not being attacked |
| 10:29.37 | emora | Just today we have two sites being hit by these http://www.voiptechchat.com/voip/457/amazon-ec2-sip-brute-force-attacks-on-rise/ |
| 10:29.42 | emora | http://www.stuartsheldon.org/blog/2010/04/sip-brute-force-attack-originating-from-amazon-ec2-hosts/ |
| 10:29.48 | emora | http://seclists.org/nanog/2010/Apr/811 |
| 10:30.46 | emora | On one server the CPU utilization was about 60%. |
| 10:31.24 | emora | Lucky for us (and our customers) that its Sunday |
| 10:32.30 | emora | we have tools in place that detect brute force attacks on SSH. But if anyone knows of something that can monitor SIP attacks, that would be very helpful |
| 10:39.46 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 10:40.21 | emora | Found what I was looking for here http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block/ |
| 10:50.56 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
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| 11:25.04 | Jumpie | emora good find on that script |
| 11:25.08 | Jumpie | i bookmarked and will do some testing |
| 11:31.42 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
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| 11:53.52 | florz | emora: does it have any effect on availability? |
| 11:54.19 | *** join/#asterisk eppigy (~eppigy@c-76-105-72-69.hsd1.ga.comcast.net) |
| 11:56.19 | *** join/#asterisk bmg505 (~leon@196-209-99-148-rrba-esr-4.dynamic.isadsl.co.za) |
| 12:03.30 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 12:06.38 | emora | florz: how it affects availability depends on how you use it. There are two parts. One script merely detects IP addresses that are causing failed SIP authentications |
| 12:07.09 | emora | You have to combine this with another script to insert rules into your iptables to block the attack |
| 12:07.39 | p3nguin | Why not use fail2ban and take care of both in one fell swoop? |
| 12:07.55 | florz | emora: I meant the attack |
| 12:09.13 | emora | florz: most certainly |
| 12:09.28 | florz | emora: so, you didn't notice any effects? |
| 12:09.36 | emora | florz: we're beeing hit with tens of thousands of attempts in matter of minutes |
| 12:10.42 | florz | emora: so? |
| 12:10.44 | emora | on an idle system (very low activity on a Sunday) CPU usage has gone over 65% |
| 12:10.57 | emora | what do you mean "so?" |
| 12:11.08 | p3nguin | You've still got 35% left, so what's wrong with that? |
| 12:11.20 | florz | emora: yeah, I totally do understand that - but it all doesn't answer the question of whether it affected availability |
| 12:12.34 | emora | so what do you guys propose? wait for the system to crash before taking action? |
| 12:12.58 | florz | emora: why do you expect the system to crash? |
| 12:13.30 | emora | or should I just go ahead an let them continue until they achieve the goal of the brute force attack? |
| 12:13.59 | florz | emora: so far you are only constructing false dichotomies |
| 12:14.40 | florz | emora: any reason for you to believe that either the system will crash or that you do have weak passwords? |
| 12:14.42 | emora | I'm glad there are people around that just dont care. As long as there is plenty of easy prey they'll stay away from professionally administered systems |
| 12:15.37 | florz | emora: in case you didn't know: having weak passwords is about as bad as adiministration can get |
| 12:15.48 | emora | yep. I agree. |
| 12:16.00 | florz | emora: so, what was your point? |
| 12:16.16 | emora | whats yours? I dont have weak passwords |
| 12:16.17 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:16.43 | florz | emora: ok, then you also don't expect them to "achieve their goal", right? |
| 12:16.44 | p3nguin | Using system resources shouldn't make the system suddenly crash. If it does, it needs more attention anyway. |
| 12:16.48 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 12:17.06 | emora | are you bored? |
| 12:17.19 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 12:17.25 | emora | have fun! |
| 12:17.38 | florz | oh, there goes the best argument of all :-) |
| 12:17.51 | emora | I'm not arguing. You are! |
| 12:17.56 | emora | again. have fun! |
| 12:18.06 | florz | *lol* |
| 12:18.28 | p3nguin | What do you expect from someone that thinks using 65% CPU is going to *gasp* make the system crash? |
| 12:19.19 | florz | well, everyone does err at times ... just that some people are capable of recognizing it when you point it out ;-) |
| 12:19.23 | p3nguin | Surely you wouldn't expect a reasonable and effective argument. |
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| 13:12.40 | qxork | p3nguin florz: the brute force attacks generally send 10,000 reg attempts in less than 60 seconds |
| 13:13.11 | qxork | additionally, if the attack is from a system like amazon, your used bandwidth will jump genereally 6MB |
| 13:13.29 | qxork | s/genereally/generally/ |
| 13:14.16 | qxork | they are not only a brute force attack to gain peers, but also a ddos depending on your system. |
| 13:14.59 | qxork | they have also evolved from being a simple extension attack to more robust dictionary attacks |
| 13:16.20 | florz | qxork: you do know what a ddos is, right? |
| 13:16.46 | qxork | florz: clearly |
| 13:17.44 | qxork | these attacks are now cloud based |
| 13:18.03 | qxork | the power is very impressive |
| 13:21.39 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
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| 13:44.02 | Trixboxer | hi |
| 13:45.05 | Trixboxer | Does installing AsteriskNOW 64bit edition & A Digium 420P pri card runs smooth ? |
| 13:51.04 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
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| 14:10.49 | riddlebox | ha installed pbiaf in a virtual machine to see what its like, and cant even log into the webgui after its all up lol what a pile |
| 14:22.34 | jhirley | what does the documentation say the password should be ? |
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| 14:39.21 | riddlebox | admin/admin |
| 14:40.10 | riddlebox | jhirley: I did find maint/ with the root password works, but when I change the admin password it still doesnt work lol |
| 14:43.44 | jhirley | there are some freepbx packaged distros that use maint/password for the web gui login. |
| 14:47.27 | riddlebox | yeah it seems like it could be a neat distro, but I cannot figure out how to not use a numerical extension number, I like to name my extension, and associate a number to it |
| 14:48.27 | jhirley | all i can say is try all the distros and see which one works best for you. |
| 14:48.44 | [TK]D-Fender | riddlebox: "device & user" module |
| 14:48.49 | riddlebox | ehh I am just playing around, I like editing my conf files |
| 14:49.19 | [TK]D-Fender | riddlebox: then WTF are you doing using a GUI interface? |
| 14:50.13 | riddlebox | just messing around, its all in a virtual machine |
| 14:50.47 | riddlebox | its just a test machine not to go into any production, just want to see what the gui is like |
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| 15:28.09 | kc8pxy | riddlebox: messing with teh config directly, and messing with the gui, intermingled, is about as diagnostic as punching buttons on your stereo, and also fiddling with where teh wires for speakers are plugged in. |
| 15:29.19 | riddlebox | kc8pxy: I am not |
| 15:29.57 | riddlebox | kc8pxy: I have my production machine, all conf files dont touch it, now I am just running a virtual machine to play with freepbx just for kicks |
| 15:37.09 | kc8pxy | riddlebox: ok.. looks like i was misinformed.. i'm actually in the throes of relearning asterisk. i can't seem to get one peer to call another, succesfully |
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| 15:40.03 | kc8pxy | i can get 2 (sip) peers, and they can register, but i can't get it to connect the peers. |
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| 15:43.49 | kc8pxy | lol... it seems i'm wrong about t hat.. but it gets boring when you call yourself(with a 2 line softphone),and whien you put your calling line on hold, to answer the destination line, either line you pick up get MOH played at it :) |
| 15:45.46 | kc8pxy | hmmmmmm |
| 15:47.08 | riddlebox | hehe |
| 15:50.05 | riddlebox | it should be hearing moh, its on hold |
| 15:52.00 | kc8pxy | here is a problem. (i'm sure someone's fixed it) have a number of sip peers. the sip channels they register have username-ish names, and not extension numbers(as I've used in the past) each of the channels has a regexten= for it, and i seem to have forgotten how to call an extension, by number. i've only got the peers dialing by channel name (SIP/channelname) how do i fix this, so i can give them all _100XX extensions? |
| 15:55.03 | [TK]D-Fender | kc8pxy:: Not really happening. There is no way to try and look up who you are trying to call that isn't 10x worse than just hard-coding a single line each |
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| 15:58.01 | Corydon76-dig | kc8pxy: do you have hints enabled? |
| 15:58.43 | [TK]D-Fender | kc8pxy: exten => 10001,1,Dial(SIP/john) |
| 15:59.56 | Corydon76-dig | [TK]D-Fender: I'd suggest pattern match and lookup the peer in a DB |
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| 16:00.44 | [TK]D-Fender | Corydon76-dig: So far his pattern suggests a max of 100 peers... how is this possibly worth it? |
| 16:01.09 | [TK]D-Fender | Corydon76-dig: Likely FAR less.... |
| 16:01.09 | Corydon76-dig | I'd say it's worth it if he even had only 10 peers |
| 16:01.34 | [TK]D-Fender | Corydon76-dig: Having to maintain a database instead of 10 lines of dialplan? Crazy |
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| 16:02.54 | Corydon76-dig | If the DB was Oracle, I'd agree with you. But MySQL and Postgres are very low maintenance |
| 16:03.50 | [TK]D-Fender | Corydon76-dig: VS the added system load... setting up * to even be able to TALK to the DB... then having to add how much extra dialplan just to USE the DB for the lookup? |
| 16:03.51 | Corydon76-dig | I'd never recommend Oracle for use with Asterisk, unless you're trying to integrate with a preexisting setup |
| 16:04.17 | Corydon76-dig | Hah, extra system load? |
| 16:04.23 | ManxPower | It is like rebuilding the engine of a car to be able to go 200Mph and then never drive it faster than 70 Mph |
| 16:04.47 | [TK]D-Fender | Corydon76-dig: Oh.. and of course have to add all the records to it in the first place... just to lok up a peername. |
| 16:04.53 | Corydon76-dig | ManxPower: for a Yugo, yes |
| 16:05.01 | ManxPower | [TK]D-Fender, and write all the dialplan code. |
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| 16:05.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:05.54 | ManxPower | Q: What do you call someone who thinks a dialplan can be simple? A: A n00b |
| 16:05.55 | Corydon76-dig | [TK]D-Fender: I find that building proper structure is worth the initial pain, because you'll never be able to justify adding it later |
| 16:05.59 | coppice | ManxPower: by the time most owners have the money to buy a 200MPH car they are unfit to drive above 70MPH |
| 16:06.41 | Corydon76-dig | and dialplans DO grow, as we put more functionality into the IVR |
| 16:08.56 | Corydon76-dig | It's more analogous to swapping your bicycle for a BMW, then driving it around at only 30 MPH |
| 16:09.19 | Corydon76-dig | It's STILL faster and less effort than a bicycle |
| 16:09.46 | ManxPower | No, it isn't. |
| 16:10.07 | Corydon76-dig | A free BMW, at that |
| 16:10.09 | ManxPower | Maybe swapping a bicycle for a BMW that you personally built by hand. |
| 16:10.59 | [TK]D-Fender | Corydon76-dig: You've just justified adding a DB, making * integrate with it and then lookup code in * dialplan... instead of 10 lines of DIALPLAN. |
| 16:11.50 | Corydon76-dig | [TK]D-Fender: Not because the dialplan is the only thing you would ever write, though. Because dialplans grow organically, and it's better to start with structure than to try to add it later |
| 16:12.41 | [TK]D-Fender | Corydon76-dig: So rather then buying a 4 year old their first tricycle you'd rather just get them a Ferrari right away and get it over with, huh? |
| 16:13.16 | Corydon76-dig | It's like telling me I don't need more than a 10A electrical panel, because my single room house doesn't require it. Right... but I might add another room on later. |
| 16:14.26 | TJNII | Yea, but you're suggestion is on par with wiring a ranch up with 400A industrial 3 phase - in case you need it in the future. |
| 16:14.35 | ManxPower | Corydon76-dig, and it is trivial to replace your 10Z panel with a 200A panel when the time is right |
| 16:15.07 | [TK]D-Fender | Corydon76-dig: You're justifying an entire external databe for 10 lines of dialplan. If he DOUBLES in size, whoopie shit, 10 more lines of dilaplan. When is a DB going to pay off? Its always going to take a shitload more work just to add a record then copy&pasting 1 more line |
| 16:15.24 | Corydon76-dig | This has gone on long enough. I've made my suggestion. The OP is welcome to take it or leave it. |
| 16:15.58 | ManxPower | Corydon76-dig, Normally your suggestions don't seem crazy. 8-) |
| 16:16.12 | Corydon76-dig | It's not crazy |
| 16:16.37 | Corydon76-dig | Excessive, perhaps, but not crazy |
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| 16:21.19 | jblack | They're right, cory. overdesign is crazy |
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| 16:40.20 | kc8pxy | so what i'm hearing, is that it's possible, and works decent, but it's way more work than i probably want to do, if i have only a handful of peers. |
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| 16:44.50 | [TK]D-Fender | kc8pxy: You added "regexten" lines as it is. Thats as many lines as it would have taken in the first place without even needing those |
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| 16:55.27 | kc8pxy | [TK]D-Fender: i'm very noob about asterisk configs. i keep experimenting to try and make ideas work. |
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| 17:56.27 | Baylink1 | p3nguin: You have seen the web page for the German guys who got WinXP to run on a 20MHz Pentium with 32MB of RAM, right? |
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| 18:10.00 | Jumpie | im sure that ran swell |
| 18:10.10 | Jumpie | dunno whats the motivation to even try |
| 18:10.11 | Jumpie | lol |
| 18:17.59 | Baylink1 | Took 31 minutes to boot, ended up at 100% utilization on idle desktop. And, why do people climb Everest again? |
| 18:18.20 | Baylink1 | http://www.winhistory.de/more/386/xpmini_eng.htm |
| 18:28.39 | TJNII | Eh, my favorite was the guy who modified the refrigerator of a dairy display case to get his 2Ghz P4 overclocked up to 8Ghz. |
| 18:37.56 | ChannelZ | People have made RAID arrays out of Zip drives (and floppies!). I guess it proves just because you have an idea, doesn't mean it's a good idea. |
| 18:38.28 | Baylink1 | Heh. I saw a thing about some guy who had a water-cooled loop of about 7 PCs that dumped heat into his swimming pool. |
| 18:39.12 | TJNII | I like that idea |
| 18:39.22 | TJNII | Heat the pool with waste heat. |
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| 18:46.33 | ChannelZ | Now if we could only harness the power of all the hot air in Washington and do something useful with it.. |
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| 18:48.25 | Baylink1 | Well, they're talking about off-shore wind farms; maybe that will help. |
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| 18:51.43 | AJayMN | Anyone know if someone has come up with a IVR trouble ticket taker? |
| 18:52.18 | AJayMN | We are an ISP and would like to log peoples # that call and allow for a trouble ticket either TXT file or email to be sent with the persons IVR responses |
| 18:55.02 | [TK]D-Fender | AJayMN: http://bestpractical.com/rt/ |
| 18:56.52 | AJayMN | mmm ok :) promising.. |
| 18:57.06 | AJayMN | was really looking for simple script lol |
| 18:57.28 | Baylink1 | Yeah, RT doesn't answer his question. :-) |
| 18:57.48 | Baylink1 | Now, you could feed tickets *into* RT from something scripted on an Asterisk. |
| 18:58.31 | AJayMN | During the evening we are not fully staffed. I want to be able to have customer call the support # and be asked is your internt down? If YES it emails me there #. If NO they are put into voicemail box |
| 18:58.44 | ChannelZ | I haven't seen one built on/attached to Asterisk (not that I've been looking) but it seems like something a bit narrow that you'd have to build yourself |
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| 18:59.12 | ChannelZ | What you just descibed should be dead easy with a small external script to send email called via AGI or something |
| 18:59.28 | AJayMN | im sure.. but im no programmer.. lol im a network tech hah |
| 18:59.49 | ManxPower | AJayMN, Then I guess you should hire one. |
| 19:00.03 | ManxPower | Most of Asterisk is "programming" of some sort, even if it's just dialplan |
| 19:00.18 | AJayMN | smart ass :P |
| 19:00.32 | ManxPower | you must be new here |
| 19:01.46 | [TK]D-Fender | AJayMN: Well you were not at all clear about the scope of what you wanted. |
| 19:02.01 | [TK]D-Fender | AJayMN: That isn't a ticket system. There is no follow up, no status, etc |
| 19:02.10 | [TK]D-Fender | AJayMN: There is a "he pressed to, so e-mail me". |
| 19:02.27 | [TK]D-Fender | AJayMN: this is a tiny piece of dialplan. Not even a "script" |
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| 19:03.16 | AJayMN | guess i need to keep hunting through asterisk stuff then |
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| 19:06.52 | ChannelZ | Whose running dahdi_dummy? |
| 19:07.34 | florz | nobody's |
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| 19:16.58 | ChannelZ | Damnation! |
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| 20:46.03 | pwntang | hi there |
| 20:46.36 | pwntang | has anyone had any experience with the Pirelli SIP handset? (the dp-l100 in particular) |
| 20:46.49 | pwntang | might be dp-l10 actually |
| 20:47.10 | ChannelZ | newp |
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| 21:23.48 | devmod | can I get video with h323 on asterisk? |
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| 21:42.50 | emora | devmod: good question. I know asterisk has codes h.261 h.263 and h.264 and it can at least switch h323 calls |
| 21:43.02 | emora | s/codes/codecs/ |
| 21:44.24 | emora | what are you trying to connect? are you using softphones or vc endpoints? |
| 21:54.24 | devmod | emora: yes |
| 21:54.31 | devmod | softphones |
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| 21:54.51 | devmod | I only seem to receive audio, I was wondering if it was a config issue or it wasnt supported |
| 21:56.05 | mdodd | hey guys, I've got a question regarding colors in the CLI output window |
| 21:56.14 | mdodd | I can only get it sometimes, but I want to have it all the time |
| 21:57.01 | mdodd | in reading the man page, I don't see anything about enabling colors (only -f which says it disables them) |
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| 22:11.50 | bio-tty | sdp question welcome? |
| 22:14.19 | hardwire | opens up a portal to 1990 |
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| 22:27.21 | pzn | Hi. My asterisk server musiconhold is bad. the music stops, plays, stops (some times per second)... I don't know how to explain that in english... what can I do to solve this? |
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| 23:00.09 | joako | I am trying to use asterisk phoneprov to provision polycom phones but its not working. How can I see any sort of access log for the asterisk http server? |
| 23:00.28 | joako | I am using verbose 1111111 and see no messages about http access |
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| 23:47.55 | bio-tty | how do i sdp offer a pt with more than one possible sample rates. say 0 pcmu 8000 as well as 16000. is the semantic possible? how do i format the fmtp? |
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