00:00.29 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
00:09.52 | *** join/#asterisk mer_ge (~martin@chello062178156192.9.14.tuwien.teleweb.at) |
00:15.51 | jc319 | p3nguin: If I do this http://pastebin.com/Y2nSPFbT line 4 should be MAC-a / MAC-b or it does not register to Asterisk. Is there a setting in sip.conf that can change this behaviour? |
00:16.02 | jc319 | (and fix the 7960-unable-to-logon issue). |
00:16.30 | jc319 | s/logon/register |
00:17.15 | jc319 | how does this thing work, needs whitespaces between ? |
00:17.16 | jc319 | s/between/betwe3n |
00:17.31 | jc319 | s/between/betwe3n/ |
00:17.33 | jc319 | aha |
00:34.01 | jc319 | p3nguin: Several peer definitions for each 7960 does not seem to work again, it only registers to Asterisk if line name is MAC-a (does not register with extension such as 201). Any ideas? |
00:49.17 | p3nguin | jc319: I changed it to reflect what you've said. Refresh the pastebin page. |
00:50.20 | p3nguin | jc319: The sed replacement works as it should, but you can't make a mistake in your syntax -- you must terminate your expression. |
00:51.08 | p3nguin | I may have made a mistake preparing that paste for you. Remember, it has been a long time since I used SIP on my Cisco phones. |
00:56.10 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
00:56.47 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
00:57.02 | fulcan | Is this a live channel? |
00:57.20 | WIMPy | No, we are all zombies. |
00:59.54 | viro | lol |
01:01.17 | WIMPy | You're only allowed to write here if you're dead or at least doomed. |
01:02.50 | jc319 | p3nguin: Does that updated one mean one extension cannot call another by typing in 3 digits anymore? |
01:15.04 | p3nguin | jc319: Phone configurations have nothing to do with how another phone is reached. That is purely done in dial plan by creating extensions. |
01:15.10 | p3nguin | So to answer your question directly, no it does not mean that. |
01:16.35 | p3nguin | Just for the record, I did buy those southwest chicken taquitos I said I was probably going to get, but I've decided to have nachos for supper. |
01:18.55 | jc319 | bonne appetit. I think I'll pastebin all configs. It's 2AM here I am starting to lose it again |
01:21.56 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
01:24.12 | jc319 | p3nguin would you mind having a look please? http://pastebin.com/LZ9Rp0ec |
01:25.02 | jc319 | I started with your basic example but changed some settings when it did not work. and some are commented to test later.. |
01:34.09 | p3nguin | jc319: You need to fix nat, directmedia, localnet, and externaddr in general. |
01:35.44 | p3nguin | Get rid of defaultuser, fromuser, and authname in the phone entries. |
01:35.44 | jc319 | ok now in all ITSP definitions is nat=no |
01:36.23 | jc319 | that 3 is gone |
01:36.34 | jc319 | def/fr users and authname gone. |
01:37.24 | p3nguin | remove all the extraneous voipms inbound contexts. You need only one, and define your phone numbers in it. |
01:37.31 | jc319 | directmedia >> what about it? I can't find it in the document |
01:37.50 | p3nguin | You still need to fix nat, directmedia, localnet, and externaddr in general. |
01:38.47 | jc319 | OK all 4 ITSP's have "context=voipms-inbound" now |
01:39.04 | jc319 | ok now in all ITSP definitions is nat=no. >> what else? |
01:39.09 | p3nguin | I would also apply a more sensible outbound context name. |
01:39.17 | p3nguin | (2034.09) <p3nguin> jc319: You need to fix nat, directmedia, localnet, and externaddr in general. |
01:39.21 | p3nguin | (2037.49) <p3nguin> You still need to fix nat, directmedia, localnet, and externaddr in general. |
01:39.24 | p3nguin | GENERAL |
01:39.26 | p3nguin | general section |
01:39.46 | jc319 | doh, ok sorry I read 'in general'.. ok |
01:41.34 | p3nguin | I guess I should have said "in [general]." |
01:43.07 | jc319 | shall I add all of these to general section >> http://pastebin.com/25zRjgBE (there's a note here saying those values worked before) |
01:43.40 | p3nguin | I would use them all. |
01:44.28 | jc319 | outbound context name - that's the best I could come up with, any suggestions? |
01:44.33 | p3nguin | I sometimes define values which are actually default values if you don't define them, but the nat stuff needs to be defined explicitly with your real values. |
01:44.44 | p3nguin | voipms-outbound or just outbound |
01:45.09 | p3nguin | outgoing, outgoing_calls, voipms-outgoing |
01:45.13 | jc319 | voipms-outbound but there are 4 sub-accounts, I need to distinguish them, no? |
01:45.13 | p3nguin | anything that makes sense. |
01:46.17 | jc319 | voipms_abiz: voipms provider & A's business line // voipms_gbiz: voipms provider & G's business line does it make sense now? |
01:46.53 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
01:47.21 | p3nguin | If you're going to have a separate context for outbound calls for every phone, I don't know what makes sense to you. If you're going to have one outbound context and use the variable solution that I suggested, I would call it voipms-outbound. |
01:48.13 | jc319 | ok these are the names then: |
01:48.16 | jc319 | include => voipms_inbound; |
01:48.16 | jc319 | include => voipms_abiz-inbound; |
01:48.16 | jc319 | include => voipms_gbiz-inbound; |
01:48.16 | jc319 | include => voipms_home-inbound; |
01:48.16 | jc319 | include => voipms_outbound; |
01:48.45 | jc319 | 1st one is main account, rest are sub-accounts |
01:56.45 | p3nguin | That's not going to work. |
01:57.25 | p3nguin | Now you're just arbitrarily including contexts in other contexts. |
01:58.05 | p3nguin | You actually need ONLY ONE inbound context with all of your phone numbers in it. |
01:58.13 | p3nguin | No includes. |
01:58.33 | p3nguin | One inbound context, and all the voipms peer entries have a context set to that inbound context's name. |
01:58.42 | p3nguin | Then define all your phone numbers in that context. |
01:59.07 | jc319 | [phones]; |
01:59.07 | jc319 | include => internal; |
01:59.07 | jc319 | include => voipms-outbound; |
01:59.07 | jc319 | & [echo code here] |
01:59.23 | jc319 | is this all that [phones] should contain? |
02:04.33 | p3nguin | probably |
02:05.22 | jc319 | Ok phones are registered now (both line buttons show registered icon). I think I need a fix in extensions.conf because cant call 004475xxxx mobile |
02:07.37 | *** join/#asterisk anumorayo (c2cbd7fe@gateway/web/freenode/ip.194.203.215.254) |
02:07.51 | p3nguin | You've got extensions matching 11-digit numbers. |
02:08.13 | p3nguin | Well, one pattern to match 11-digit numbers, which is suitable. |
02:08.41 | p3nguin | But you need to PROPERLY set setvar=OutPeer= for each phone (line) that you're using. |
02:10.03 | p3nguin | And your permit lines in every single voipms peer should be permit=78.129.153.20/255.255.255.255 rather than what you have. |
02:10.20 | p3nguin | An ACL is of no use if you apply it incorrectly. |
02:14.03 | jc319 | ACL fixed now. |
02:14.39 | jc319 | There's a msg about new syntax of that conditional header rewriting http://pastebin.com/TnP30WuV |
02:16.57 | anumorayo | hi all |
02:17.32 | anumorayo | having program with make outgoing and incoming calls |
02:18.15 | anumorayo | i get a failed tone for incoming and a all circuits are busy |
02:18.50 | anumorayo | i am using an isdn card b410p |
02:19.03 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
02:19.33 | WIMPy | Do you have your extensions in the right format? What does the console say when you try to call? |
02:20.49 | anumorayo | fro incoming call it genrates no out in cli mode |
02:21.19 | WIMPy | core set verbose 9 |
02:21.26 | WIMPy | core set debug 9 |
02:21.36 | p3nguin | or verbose 4, since 9 doesn't produce more output |
02:21.43 | WIMPy | If you still se nothing,the call doesn't get to Asterisk at all. |
02:22.19 | p3nguin | New syntax, eh? ExecIf(<expr>?Set(CALLERID(num)=value |
02:22.47 | p3nguin | Let me see how that compares to my old syntax. |
02:23.11 | jc319 | p3nguin: thanks. By the way, I still receive the following if I use the 2nd line button on 7960: |
02:23.12 | jc319 | [Jun 12 02:21:54] WARNING[2196]: chan_sip.c:13698 check_auth: username mismatch, have <000AB747A418-a>, digest has <000AB747A418-b> |
02:23.12 | jc319 | [Jun 12 02:21:54] NOTICE[2196]: chan_sip.c:21515 handle_request_invite: Failed to authenticate device "A <004420 8441 3867>" <sip:000AB747A418-b@10.9.8.4>;tag=000ab747a41800257071a152-04517b33 |
02:23.31 | p3nguin | I have ExecIf($[${IF($["${externalCID}" != ""]?1)}],Set,CALLERID(num)=value |
02:23.59 | jc319 | Related to the above, in sip.conf I have "match_auth_username=yes" (does not seem to help much) |
02:24.09 | WIMPy | If If? |
02:24.32 | anumorayo | yes you are correct i am not hit the astrisk at all |
02:24.41 | jc319 | ExecIf($[${IF($["${externalCID}" != ""]?1)}],Set,CALLERID(num)=${externalCID}); <yes this is the old one |
02:24.41 | p3nguin | So let's change the line to: ExecIf($[${IF($["${externalCID}" != ""]?1)}]?Set(CALLERID(num)=${externalCID})); |
02:24.45 | jc319 | ok |
02:25.29 | jc319 | so you basically changed ? to , |
02:25.42 | p3nguin | And changed the Set command within the ExecIf(). |
02:25.51 | p3nguin | It was Set,CALLERID(num) |
02:26.06 | p3nguin | Now they indicate they want Set(CALLERID(num)=value) |
02:26.16 | anumorayo | is that case of me not using the right channel |
02:26.18 | WIMPy | anumorayo: So check your dahdi config or perhaps your cabling. What does 'dahdi show status' give? |
02:26.18 | jc319 | Oh yes |
02:26.26 | p3nguin | It makes more sense this new way, since that's the syntax for the Set() command anyway. |
02:26.55 | p3nguin | It was ExecIf()'s syntax that required the commas. |
02:28.49 | anumorayo | wimpy B4XXP (PCI) Card 0 Span 1 OK 0 0 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) |
02:29.00 | anumorayo | sorry guy |
02:29.40 | anumorayo | http://pastebin.com/B13vH60E |
02:30.09 | anumorayo | WIMPy: check that out |
02:30.12 | WIMPy | "OK" is good. Are you sure the number is routed to that inteface? |
02:30.38 | WIMPy | You're using two interfaces only, I assume? |
02:30.55 | anumorayo | only 2 |
02:31.06 | jc319 | p3nguin: is this the modification you requested or did I get it wrong? >> http://pastebin.com/4BYNgwUg (you can see the old line commented out and the new one) |
02:31.34 | anumorayo | how do i check i routing to the correct interface |
02:31.57 | WIMPy | anumorayo: Ask your telco. |
02:32.32 | WIMPy | anumorayo: You can also use 'pri set debug...', but I doubt there's anything to see. |
02:33.13 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
02:34.27 | WIMPy | anumorayo: What about calling out? |
02:34.51 | jc319 | p3nguin: just received an email, they've activated my 2nd DID (local one) |
02:35.30 | anumorayo | i can't test that now sorry |
02:35.59 | WIMPy | anumorayo: 'channel originate' |
02:36.28 | p3nguin | jc319: The change looks right to me. Does your verbose output indicate that I did it wrong? |
02:36.42 | anumorayo | sorry |
02:38.26 | anumorayo | i am in the UK if that is wjat you are asking |
02:39.14 | WIMPy | anumorayo: No, I'd like to find out, what happens if you try to plsce a call from that line. |
02:39.59 | WIMPy | You can use 'channel originate' if you don't have access to a phone. |
02:41.29 | anumorayo | what from cli ? |
02:41.40 | WIMPy | yes |
02:42.55 | anumorayo | i am getting No such command |
02:43.13 | jc319 | p3nguin: I think the programmatic rewrite works. However it does not ring other party. (everyone is busy). Also I still cannot use the 2nd line button, is there anything else to try to make it work? (error >>> username mismatch, have <000AB747A418-a>, digest has <000AB747A418-b> ) |
02:43.34 | jc319 | p3nguin: http://pastebin.com/z1RBYNKL |
02:44.00 | WIMPy | anumorayo: For ISDN connections, I'd recommend a recent version of libpri and Asterisk. For old versions it's only originate. |
02:44.31 | p3nguin | jc319: I just thought of something... |
02:44.57 | anumorayo | yes i have that command |
02:45.41 | anumorayo | howo use it |
02:45.53 | p3nguin | jc319: For me, in the US, I cannot dial 07599408968 and make it dial that number. I have to prefix anything outside the US with 00 or 011. Let's change your outbound extension to be this: Dial(SIP/${OutPeer}/00${EXTEN}); |
02:46.10 | WIMPy | anumorayo: It tells you if you use it without parameter. |
02:46.36 | WIMPy | anumorayo: e.g. 'originate dahdi/1/yourmobilenumber application sayunixtime' |
02:46.42 | p3nguin | You'll notice you'll still dial the same numbers on the phones, but what will be dialed to voipms will be 00+the number you dialed on the phone. |
02:47.06 | p3nguin | If that's the problem, adding the 00 will fix it. |
02:50.04 | p3nguin | *sigh* Add the 00, save, "dialplan reload", dial the number. |
02:52.25 | jc319 | http://pastebin.com/KtUnGaL3 |
02:52.32 | jc319 | I did, the same |
02:53.49 | WIMPy | 00075...? Surely not. |
02:54.34 | jc319 | uhmm you're right |
02:54.54 | WIMPy | And I doubt you only want to call 11-digit numbers. |
02:55.05 | jc319 | well that was the starting point, and I'm still there |
02:56.37 | jc319 | OK I found a dialplan here http://forum.voxilla.com/asterisk-support-forum/help-uk-pstn-dial-plan-28903.html |
02:56.53 | jc319 | The following covers everything (as far as I've used it for the last 2 yrs) in the UK: 0[1278]XXXXXXXXX [2-8]XXXXX |
02:57.30 | jc319 | it does not have special services such as 15x (BT repairs etc) but I will think about it later |
02:58.21 | WIMPy | I thought UK numbers can have different lengths? |
02:59.38 | WIMPy | Unless you don't want to be able to dial anything, you should probably only use _X. for outbound calls. |
03:00.18 | WIMPy | And make sure your internal extensions are included first. |
03:04.22 | jc319 | WIMPy: How do I include them, by putting [internal] group above? |
03:04.23 | WIMPy | anumorayo: Still listening to your new personal talking clock? |
03:06.04 | anumorayo | lol i can't get it too work the command i mean |
03:06.35 | WIMPy | Can't find your dialplan any more. Make sure that you include your internal context befor your outgoing context in the phones context. |
03:06.42 | WIMPy | anumorayo: What happens? |
03:07.16 | WIMPy | jc319: Or maybe better don't use internal extensions that overlap with real phone numbers. |
03:09.49 | p3nguin | Or since you know there aren't going to be any phone numbers less than, say, 5 digits, you could use _XXXX. as the pattern. |
03:10.15 | p3nguin | That way your 3-digit internals won't match. |
03:11.46 | WIMPy | You will usually have at least one 3-digit number. |
03:12.11 | p3nguin | 3-digit phone numbers? On the real phone network? |
03:12.24 | WIMPy | Emergency services. |
03:12.32 | p3nguin | I still have the feeling that VoIP.ms is going to require these "international" numbers to be prefixed with 00 or 011. If that leading 0 on your number isn't part of the number, then use 0${EXTEN}. |
03:12.38 | p3nguin | They have 911 in the UK? |
03:12.44 | WIMPy | And yes, there are other 3-digit numbers in some places. |
03:13.13 | WIMPy | 911, 999, 112, 110 |
03:13.40 | p3nguin | I've never been in the UK to use their phones, so I have no idea what numbers they have. |
03:16.48 | anumorayo | yeah it 999 |
03:18.05 | anumorayo | WIMPy: my channel is zap/g0 |
03:18.24 | WIMPy | anumorayo: zap? Thats ancient. |
03:18.25 | jc319 | I keep getting thin >> == Everyone is busy/congested at this time (1:1/0/0) |
03:18.31 | jc319 | can't make any calls now, any ideas why? |
03:18.39 | *** join/#asterisk l2trace99 (~bender@rrcs-71-43-104-238.se.biz.rr.com) |
03:19.05 | jc319 | http://pastebin.com/A2EWFK64 |
03:19.06 | WIMPy | anumorayo: Err, no. You did have dahdi, wich replaced zaptel, so it must be dahdi. |
03:19.16 | jc319 | latest extensions & full busy msg |
03:19.30 | jc319 | this* |
03:20.18 | WIMPy | jc319: That number doesn't make any sense, either. Have you tried the full unfomatted version, i.e. 004475...? |
03:20.55 | anumorayo | yes it is |
03:21.03 | anumorayo | sorry |
03:21.37 | jc319 | WIMPy: If I call '00447599408968' it adds additional zero so it becomes '000447599408968' and does not work. I think I'll remove that 0$ |
03:21.56 | anumorayo | but it refer in my gui has trunk zap/g0 |
03:22.04 | WIMPy | jc319: Yes, that should work. |
03:23.06 | anumorayo | i am using trixbox |
03:23.08 | WIMPy | anumorayo: That doesn't sound right, but for help on the gui, you should check it's channel. We're hardli familiar with those. |
03:23.49 | jc319 | OK I think this one is better, http://pastebin.com/6TvfFZev |
03:24.11 | jc319 | However I still have this busy signal. Why did it start constantly giving busy now.. |
03:24.57 | WIMPy | anumorayo: I'd still recommend a current verison of Asterisk, especially for ISDN connectivity. |
03:29.07 | anumorayo | man |
03:30.18 | anumorayo | this is the most current version of trixbox |
03:31.32 | WIMPy | It seems to be rather outdated. |
03:32.34 | anumorayo | based on what |
03:33.49 | WIMPy | The fact that you don't have .channel originate' and it uses zap, which should have gone >2 years ago. |
03:34.13 | WIMPy | Actually it was replaced 3 years ago. |
03:34.53 | p3nguin | (2212.32) <p3nguin> I still have the feeling that VoIP.ms is going to require these "international" numbers to be prefixed with 00 or 011. If that leading 0 on your number isn't part of the number, then use 0${EXTEN}. |
03:35.25 | WIMPy | You're missing a CC then. |
03:35.42 | WIMPy | Maybe they accept +... |
03:36.31 | p3nguin | jc319: I'm ringing your number now. |
03:36.36 | *** join/#asterisk sourcode (~code@ppp-58-8-87-34.revip2.asianet.co.th) |
03:37.00 | p3nguin | If you answered, there was no audio. |
03:37.02 | jc319 | oups was it you |
03:37.13 | jc319 | sorry I picked up because I was calling that number too |
03:37.40 | p3nguin | I dialed 00447599408968 from my phone, which sends exactly 00447599408968 to voipms. |
03:37.57 | jc319 | with 0$ or $ |
03:38.05 | p3nguin | If you are starting with 04, then you need to use 0${EXTEN} |
03:38.17 | p3nguin | Because you must send 00447599408968 |
03:38.37 | p3nguin | ANd 0${EXTEN} is 0 0447599408968 |
03:39.00 | p3nguin | Do all of your UK numbers that you dial start with 0? |
03:39.37 | jc319 | normal number yes, but what about 999 etc. |
03:39.50 | jc319 | p3nguin can you have a look at this please - http://pastebin.com/2NZRSJR4 |
03:39.54 | p3nguin | voipms has no idea what 999 is. |
03:40.08 | jc319 | I still get the busy signal all the time, is there something wrong in there |
03:41.19 | p3nguin | yes |
03:42.04 | p3nguin | If all normal numbers you dial start with 0, you MUST use 0${EXTEN} in the dial command. |
03:42.09 | p3nguin | Until you do, it's not going to work. |
03:42.26 | jc319 | yes but I mean I'm typing 0044... full number in int. format |
03:42.29 | jc319 | it still does not work |
03:42.34 | p3nguin | Show me. |
03:42.40 | p3nguin | core set verbose 4 |
03:42.42 | p3nguin | make the call |
03:42.47 | p3nguin | paste the output. |
03:43.13 | WIMPy | If you dial a national number you need to send it as 0044${EXTEN:1}. |
03:43.47 | p3nguin | I expect that he has to use the same rules in voipms that I have to use. They do not accept THAT format. |
03:44.07 | jc319 | At the moment without doing ANY patterns I just want to ring using full number 00 - country code - full national number |
03:44.10 | p3nguin | They will accept 00 and 011 as prefixes, but 044 is not going to work. |
03:44.11 | jc319 | http://pastebin.com/yJ2F3u6n |
03:45.00 | jc319 | what does 'busy here' mean? Got SIP response 486 "Busy Here" back from 78.129.153.20:5060 |
03:46.04 | p3nguin | It means busy. Show me your voipms_home peer from sip.conf. |
03:46.08 | WIMPy | Maybe they bloced your account for too many failed call attempts? I heard some providers do such things. |
03:46.29 | p3nguin | Nah, I've failed plenty of calls and had no issues. |
03:46.42 | p3nguin | I do more than 75% of my calls through voipms. |
03:47.07 | jc319 | http://pastebin.com/sDaABhpB |
03:47.11 | WIMPy | So you have successful ones in between. |
03:47.42 | jc319 | but I call more than normal, yesterday at 6 AM I had 70+ calls on my mobile (test calls) maybe they're blocking me |
03:48.03 | p3nguin | What happens if you take out that "fromuser" value? Just comment it out. |
03:48.47 | p3nguin | save, sip reload, make a call. |
03:48.48 | jc319 | now they are not busy. this time the 'circuit' is busy. |
03:48.49 | jc319 | [Jun 12 03:48:26] WARNING[2196]: chan_sip.c:19344 handle_response_invite: Received response: "Forbidden" from '"A <00447915010547>" <sip:00447915010547@78.86.169.203>;tag=as4c5d96e8' |
03:48.49 | jc319 | <PROTECTED> |
03:49.00 | p3nguin | hmm |
03:49.11 | p3nguin | I don't use fromuser for my voipms peer. |
03:50.53 | jc319 | commented it out, the same. commented out defaultuser= too (just to test) still the same |
03:51.28 | jc319 | [Jun 12 03:51:08] WARNING[2196]: chan_sip.c:19344 handle_response_invite: Received response: "Forbidden" from '"212" <sip:00447599408968@78.86.169.203>;tag=as4402b786' |
03:51.28 | jc319 | <PROTECTED> |
03:51.51 | p3nguin | You'll need defaultuser because that is what it authenticates against. |
03:52.14 | p3nguin | You recall that setting about match auth user that you added earlier? Take it out if it's still there. |
03:53.11 | jc319 | ;match_auth_username=yes |
03:53.14 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
03:53.16 | jc319 | was active, commented out now, testing... |
03:53.44 | jc319 | the same... |
03:55.53 | p3nguin | *shrug* It sure worked for me. |
03:56.16 | jc319 | tomorrow I'll start from blank configs |
03:56.26 | jc319 | thanks for all the help |
03:56.31 | p3nguin | That's the spirit! |
03:56.38 | p3nguin | Start from the ground up. |
03:56.52 | jc319 | I'm on config 13 now (ground up) |
03:56.55 | jc319 | 8 works :D |
03:56.59 | jc319 | I'll copy that one |
03:57.06 | jc319 | I think this multi-line Cisco thing is not helping |
03:57.22 | jc319 | it's not working anyway. I'll use single peer definition for cisco devices |
03:57.51 | jc319 | It'll be a shame if I can't use multi line buttons without SCCP. |
03:58.12 | jc319 | I had big dreams about 'em... |
03:58.47 | WIMPy | Anything is better than SIP. |
03:58.55 | p3nguin | When I used SIP, I used two lines: one for regular stuff and one specifically for callcentric. |
03:59.39 | p3nguin | Two completely seperate and unrelated authnames. |
04:05.20 | jc319 | Before I go to bed, out of curiosity, I copied back the config which worked before |
04:05.36 | jc319 | It still gives the same msg "Got SIP response 486 "Busy Here" back from 78.129.153.20:5060" |
04:05.46 | jc319 | it must be an ITSP side issue |
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05:02.43 | p3nguin | I doubt it's an ITSP problem, since I can call your 044 number and it just works. |
05:02.50 | p3nguin | But what do I know? |
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05:04.57 | rhollan_ | How are [context] sections in sip.conf matched? I have incoming SMS over SIP using custom SIP headers and I want to process them. but I can't match on the incoming provider to select any context other than default. |
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05:11.24 | rhollan_ | Anyone her have Asterisk process inbound SMS? |
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05:35.22 | fulcan | rhollan_ i think default catches anything it doesn't understand. sip.conf doesn't exactly 'match' to extensions.conf per se. sip.conf is your door and lock. extension.conf is your map/chart in the hallway on your left when you enter. :) |
05:40.01 | p3nguin | (0018.11) -!- rhollan ~rhollan@173-10-78-121-BusName-Washington.hfc.comcastbusiness.net has quit [Quit: Leaving] |
05:42.16 | fulcan | if i am looking to setup asterisk simply record calls and nothing more than a static forward to an outside line, where would i find the most simple and straight forward 'howto' on this? |
05:43.45 | p3nguin | exten => _[*#0-9].,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}-${EXTEN}.WAV,a); |
05:44.04 | p3nguin | exten => _[*#0-9].,n,Goto(some-other-context,${EXTEN},1); |
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05:44.31 | p3nguin | That's the basic concept. |
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05:49.12 | fulcan | p3nguin I haven't played with asterisk in years my friend. I took one look at the new extensions.conf and decided I needed a refresher course. I used to be talented in asterisk but I am horribly rusty right now. Where could you send me for some good bedtime reading material on dialplans and recording? |
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07:02.31 | p3nguin | ~thebook |
07:02.31 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
07:02.36 | p3nguin | fulcan: ^^^ |
07:03.53 | fulcan | p3nguin awesome! thank you. |
07:04.19 | fulcan | p3nguin question for you real quick. I thought 's' was a catch all? |
07:04.29 | p3nguin | It's not. |
07:04.34 | p3nguin | s only matches s. |
07:04.49 | p3nguin | literally the 's' extension. |
07:05.17 | p3nguin | s is most often used where there is no phone number. |
07:05.49 | p3nguin | macros, IVRs, analog lines where you have a real hook |
07:08.22 | fulcan | p3nguin yup, I am rusty. Rejected connect attempt from 63.211.239.14, request '7185696288@default' does not exist "But", I was trying to use 's' :) I can put a static map in extension naming the number is she will connect, but just hang there. This fine :) Just playing, new server and an old friend.... :) |
07:09.42 | p3nguin | Define your phone number... or send calls to the s extension. |
07:09.52 | p3nguin | I wouldn't want calls going to the s extension, so I would define my phone numbers. |
07:11.21 | fulcan | the end result I am hoping to do is just a static maping/forward and record the call (I am assuming canreinvite=yes is still key to this)? |
07:12.07 | fulcan | I am 100% viop with no hardwire. |
07:12.12 | fulcan | voip |
07:12.14 | fulcan | :P |
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07:20.00 | fulcan | p3nguin she barked at me untill I put the line into [default]. Went straight to the demo as long as [default]. This should hurt anything if I cram all of my numbers into a default context would it? there will be a few of them? |
07:22.24 | p3nguin | I certainly wouldn't. default really needs to be empty. |
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07:23.21 | fulcan | p3nguin if I name the context the same as the number, she won't catch the call? |
07:24.41 | fulcan | request '7185696588@default' does not exist |
07:24.44 | p3nguin | It also wouldn't make any sense. |
07:25.20 | p3nguin | Make a peer definition in sip.conf according to where that call comes from, then assign an APPROPRIATE context. |
07:25.43 | p3nguin | And I'm going to bed, so I won't answer any more questions for at least eight hours. |
07:26.04 | fulcan | no worries my friend. Enjoy your rest. |
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10:45.27 | jc319 | hello |
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12:32.40 | lesouvage | . |
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12:33.49 | Dovid | can anyone help me here with a core dump ? |
12:36.07 | WIMPy | Dovid: gdb can |
12:36.14 | WIMPy | ~collectdebug |
12:36.14 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
12:37.56 | Dovid | WIMPy: I have the back trace. I can't seem to figure out where the issue is |
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12:38.14 | irroot | post it in pastebin |
12:38.42 | irroot | what is nature of problem and version ?? |
12:39.08 | Dovid | http://pbx.dovid.net/core_dump_06-12-2011-at-0820.log |
12:39.24 | Dovid | i upgraded from 1.6.1.20 to 1.6.2.18 |
12:39.34 | Dovid | and i got that with in an hour |
12:39.47 | Dovid | irroot: btw you are my hero for working on T.38 |
12:40.13 | irroot | lol its not there for 1.6 but have it in production on 1.8 thx |
12:40.33 | Dovid | irroot: I know this is for another system. i am trying it on my ss7 box running 1.8 |
12:40.47 | irroot | cool |
12:40.53 | Dovid | is there any tutorial on core dumps so i can learn how to figure out the issue on my own ? |
12:41.17 | irroot | looks like a call picup gone wrong |
12:42.30 | Dovid | i am actually upgrading because of the T.38 vulnerabilituy |
12:42.44 | Dovid | irroot: gone wronge == ? |
12:42.57 | Dovid | bad typing day |
12:43.53 | Dovid | eh. I think i am just going to patch my 1.6.1 with: http://downloads.asterisk.org/pub/security/AST-2011-002-1.6.1.diff |
12:45.36 | irroot | local_fixup is called when a call is masqueraded in this case it causes a segfault you have it built with optomizations rather use dont_optomize for debug purposes |
12:46.01 | Dovid | irroot: What do you think ? should I open a ticket about the core dump ? I dont want to get shouted at ;) |
12:46.38 | irroot | hehe rebuild with dont optomize and see if it happens again |
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12:51.07 | Dovid | bakc |
12:51.09 | Dovid | back* |
12:52.09 | irroot | dovid rebuild with dont optomise the thing is 1.6 is nolonger supported officially |
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12:55.41 | Dovid | what does the optimize do ? |
12:55.53 | Dovid | when ever i go to 1.6.2 it craps out |
12:56.01 | Dovid | i though 1.6.2.X is stills upported ? |
12:56.36 | Dovid | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
12:56.41 | Dovid | i guess it just passed :( |
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13:02.26 | irroot | make menuconfig |
13:02.31 | irroot | there are some options |
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14:04.27 | fulcan | . |
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14:12.01 | fulcan | I am getting the error of " Cannot change monitor filename of channel SIP/7185696288-00000004" I am thinking permissions but checking here first before screwing something else up. :) http://pastebin.com/1SSbmJz5 |
14:15.41 | cneb3000-laptop | hmm, fulcan what version of asterisk are you using? |
14:16.32 | irroot | couple things the filename is not unique so look at that if it exists already and then as above with cneb3000 [Jun 12 20:07:42] WARNING[31044]: file.c:1165 ast_writefile: No such format 'wav|SUMRALL' |
14:16.52 | irroot | you should not use | in 1.8 if you are on 1.8 |
14:16.59 | cneb3000-laptop | ^^^ beat me to it |
14:17.32 | fulcan | oh geeze, I forgot where to find the asterisk version @. I used gentoo emerge |
14:17.33 | cneb3000-laptop | wav|${CALLERID(name)} should be more like ---> wav,${CALLERID(name)} |
14:18.08 | fulcan | I took the pipe out (I actually opened my eyes) and the error is gone and looking for the recorded file now... |
14:18.26 | cneb3000-laptop | hehe |
14:18.29 | cneb3000-laptop | look within |
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14:20.58 | fulcan | Asterisk 1.6.2.17.3 built by root @ li345-191 on a i686 running Linux on 2011-06-12 06:12:19 UTC |
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14:21.54 | irroot | use the source young padowin |
14:22.19 | fulcan | where does asterisk store the recorded files? |
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14:23.01 | cneb3000-laptop | fulcan: if you look in your error it's trying to store the file in /var/spool/asterisk/monitor/. so it's probably in there |
14:23.57 | fulcan | cneb3000-laptop two files instead of a merged file? |
14:24.15 | irroot | soxmix or sox |
14:24.23 | irroot | use the m option to monitor |
14:24.38 | irroot | it will sox mix em when dons |
14:24.42 | irroot | done |
14:25.40 | irroot | hint you can have a wrapper /usr/bin/soxmix to call sox -m and do any post processing here sign the file perhaps |
14:27.04 | fulcan | irroot where do you put the -m switch, in the begining right after extn -> or at the end? |
14:27.55 | irroot | Monitor([file_format[:urlbase]][,fname_base[,options]]) |
14:28.14 | irroot | core show application monitor |
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14:30.27 | cneb3000-laptop | so in your case fulcan it will be something like exten "=> _.*,1,Monitor(wav,${CALLERID(name)},m") |
14:30.55 | cneb3000-laptop | sorry, "exten => _.*,1,Monitor(wav,${CALLERID(name)},m)" |
14:30.58 | cneb3000-laptop | drunk fingers |
14:31.30 | cneb3000-laptop | being playing with perl a lot recently. keep nearly enging every sentance with ; |
14:31.33 | cneb3000-laptop | ¬_¬ |
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14:34.21 | fulcan | cneb3000-laptop asterisk creates these files "SUMRALL-in.wav SUMRALL-out.wav" (and no new one like 01_SUMRALL-out.wav 02_SUMRALL-out.wav). If I delete the files, asterisk creates two new ones with the exact same name. no merge and no indexing of the file names ? |
14:35.09 | fulcan | cneb3000-laptop I have the ,m switch set exactly like that too. |
14:35.17 | cneb3000-laptop | did you reload asterisk? |
14:35.42 | fulcan | yup, dialplan reload. let me do a core reload |
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14:37.26 | cneb3000-laptop | if that still doesn't work can you pastebin the dialplan again? just to double check. and also pastebin any errors if there are any. |
14:39.58 | fulcan | http://pastebin.com/y8dyi7Y6 no errors, but same two files and still no index. :/ |
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14:41.26 | cneb3000-laptop | hmm, i think the joining is acttually done by a third party app as irroot said, called sox |
14:41.30 | cneb3000-laptop | maybe you dont have it installed? |
14:42.14 | fulcan | cneb3000-laptop emerge sox 'right now :)' |
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14:43.38 | cneb3000-laptop | haha, so it wasnt there? :) |
14:46.04 | fulcan | cneb3000-laptop nope, I installed her and now I have 1 file! :) |
14:46.11 | cneb3000-laptop | woohoo |
14:46.47 | fulcan | cneb3000-laptop only 1 though, after 3 to 4 test calls though. :( |
14:47.13 | cneb3000-laptop | so it stopped working? |
14:47.17 | fulcan | cneb3000-laptop how do I get her to index file names? |
14:47.22 | fulcan | I works! |
14:47.27 | cneb3000-laptop | what do you mean index file names? |
14:47.30 | fulcan | don't get me wrong! |
14:47.58 | cneb3000-laptop | you mean like... 2010-01-01call1.wav 2010-01-1call2.wav? |
14:48.01 | fulcan | cneb3000-laptop it overwrites the original instead of creating a new one. |
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14:48.07 | fulcan | yup! :) |
14:49.42 | fulcan | can this be done?? exten => _.*,1,Monitor(wav,${CALLERID(name)},m) would change to exten => _.*,1,Monitor(wav,${CALLERID(name)(systemdate)},m)? |
14:49.54 | cneb3000-laptop | it's only doing that because in the filename bit of the monitor command you're saying '$callerid(name) |
14:50.06 | cneb3000-laptop | ^^^ yes, althouh i'm not sure if systemdate is correct |
14:50.12 | cneb3000-laptop | (because i just dont know) |
14:50.16 | cneb3000-laptop | but thats the right idea |
14:50.24 | cneb3000-laptop | reason it overwrites the current file, is because its the same file name |
14:50.38 | fulcan | <PROTECTED> |
14:51.23 | cneb3000-laptop | fulcan look at this... |
14:51.30 | cneb3000-laptop | http://www.voip-info.org/wiki/view/Asterisk+variables |
14:51.47 | cneb3000-laptop | not sure if it's up to date. but it suggests ${DATETIME} will do what you're looking for |
14:51.52 | fulcan | on it like white on rice! :) |
14:52.07 | cneb3000-laptop | actually, it says use ${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}) |
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14:53.43 | cneb3000-laptop | ^^^ confirmed at https://wiki.asterisk.org/wiki/display/AST/Asterisk+standard+channel+variables |
14:53.55 | cneb3000-laptop | "${DATETIME} * - Current date time in the format: DDMMYYYY-HH:MM:SS (Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})" |
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14:54.15 | cneb3000-laptop | let me know how it goes, i'll be doing something similar myself in the future :) |
14:55.13 | irroot | exten => s,n,SET(CLOGID=${IF(${EXISTS(${DDIUNIQUEID})}?${DDIUNIQUEID}:${CDR(linkedid)})}) |
14:55.14 | irroot | exten => s,n,SET(CALLLEG=${ODBC_LOGCOUNT(${CLOGID})}) |
14:55.16 | irroot | exten => s,n,SET(ODBC_LOG(${CLOGID},${CALLLEG},${CHANNEL})=${ARG3}) |
14:55.19 | irroot | exten => s,n,SET(CALLDATE=${IF(${EXISTS(${CALLDATE})}?${CALLDATE}:${STRFTIME(,,%Y-%m-%d)})}) |
14:55.20 | irroot | exten => s,n,SET(MONITOR_EXEC_ARGS=${ODBC_RTDB(Setup,RecOpt)}) |
14:55.23 | irroot | exten => s,n,Monitor(wav49,/var/spool/asterisk/monitor/${CALLDATE}/${ARG3}/${CLOGID}-${CALLLEG},mb) |
14:55.58 | irroot | thats what i do it tracks trasfers and stores it in directories under monitor by year/exten |
14:58.50 | cneb3000-laptop | ahh thanks irroot |
15:00.10 | irroot | pleasure makes it much faster looking up files under multiple paths |
15:01.57 | irroot | when there couple 100 / 1000 files it can cause a massive knock on quality and i/o load |
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15:02.36 | cneb3000-laptop | knods |
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15:02.47 | cneb3000-laptop | hmm |
15:02.57 | cneb3000-laptop | mind if i ask which variable is tracking the transfers? sorry.. |
15:03.16 | irroot | callleg |
15:03.17 | fulcan | this "exten => _.*,1,Monitor(wav,${CALLERID(name),${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}},m)" does nothing more than chop the first 2 letters off of the callerid/filename???? |
15:03.44 | irroot | it gets stored in odbc and then added to each leg of the call |
15:04.00 | cneb3000-laptop | ^ ah i see. nice! |
15:04.46 | irroot | (name),${STRFTIME(${E |
15:04.46 | fulcan | 'MRALL.wav' instaead of SUMRALL.wav |
15:04.51 | irroot | cant use a , |
15:04.54 | irroot | use a - |
15:05.09 | fulcan | irroot after name? |
15:05.18 | irroot | yip |
15:05.34 | irroot | also missin g } |
15:06.02 | fulcan | g where? |
15:06.11 | irroot | ${CALLERID(name)}-${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)},m |
15:08.06 | irroot | its possible maybe i have spent too much time in the dialplan |
15:09.01 | fulcan | irroot when I use 'exten => _.*,1,Monitor(wav,${CALLERID(name)}-${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)},m)' it no longer chops the name up, but it doen not append the date. Filename still SUMRALL.wav |
15:09.40 | irroot | ${CALLERID(name)}-${STRFTIME($,,%d%m%Y-%H:%M:%S)},m |
15:09.43 | irroot | ${CALLERID(name)}-${STRFTIME(,,%d%m%Y-%H:%M:%S)},m |
15:09.55 | irroot | i find the time function a little tricky |
15:10.29 | irroot | epoch is default |
15:11.19 | fulcan | first or second one my friend? |
15:11.53 | irroot | second |
15:12.03 | cneb3000-laptop | irroot: i left a job after being there for 3 years. someone wrote "BYE sip:[myphonenumber]@[my home address]:5060" on a leaving card |
15:12.04 | irroot | first has a extra $ |
15:12.07 | cneb3000-laptop | you think you have it bad? |
15:12.23 | cneb3000-laptop | i thought it was funny |
15:12.25 | irroot | lol |
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15:16.47 | fulcan | wow crazy, still no date using "exten => _.*,1,Monitor(wav,${CALLERID(name)}-${STRFTIME($,,%d%m%Y-%H:%M:%S)},m)" and I tried it without the $ before the ,, same thing????? |
15:18.36 | irroot | ${STRFTIME(,,%Y-%m-%d)} |
15:19.02 | irroot | straight out my DP that only does YYYY-MM-DD |
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15:21.02 | fulcan | irroot same filename produces SUMRALL.wav with no date and this is after a core restart.....? :( |
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15:22.43 | irroot | dialplan reload ?? |
15:23.11 | fulcan | irroot core restart now |
15:25.50 | irroot | or -${CDR(uniqueid)} instead of strftime |
15:26.38 | kaldemar | fulcan: what version are you using? |
15:27.23 | kaldemar | oh, it was answered already. |
15:27.25 | fulcan | Asterisk 1.6.2.17.3 built by root @ li345-191 on a i686 running Linux on 2011-06-12 06:12:19 |
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15:33.19 | fulcan | Ok, GREAT news. almost there this "exten => _.*,1,Monitor(wav,-${STRFTIME(,,%Y-%m-%d)}${CALLERID(name)},m)" produces -> -2011-06-12SUMRALL.wav :) |
15:37.44 | fulcan | and THIS "exten => _.*,1,Monitor(wav,${STRFTIME(,,%Y-%m-%d-%h-%m-%s)}${CALLERID(name)},m)" solves the issue entirely and is somewhat of a screwup in reverse. By doing it backward with the date first will make sorting easier. Thank you so much for your help... :) |
15:38.22 | fulcan | I have no clue why the date breaks if you put callerid first. |
15:38.53 | fulcan | there is something up with that, but it's not hurting me at all. |
15:38.54 | irroot | cool |
15:41.21 | cneb3000-laptop | good to see it working |
15:41.36 | cneb3000-laptop | next up - getting it mixdown the 2 files into stereo? ;) |
15:46.54 | fulcan | cneb3000-laptop your never gonna believe this one. Got the file naming straight, when to check the file produce, its a blank .wav file :( I cannot not confirm if it ever worked though. Vox is installed. Any thoughts? |
15:47.16 | cneb3000-laptop | so when you listen to .wav file it's silent? |
15:47.25 | fulcan | yup |
15:47.36 | cneb3000-laptop | is it a 0kb file, can you tell? |
15:47.42 | fulcan | all new files produced are the same 44 byte files. |
15:47.59 | fulcan | all 44 bytes |
15:48.20 | fulcan | beautiful naming scheme though.... :) |
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15:50.06 | cneb3000-laptop | hmmm |
15:50.11 | cneb3000-laptop | fulcan, turn off the m option |
15:50.20 | cneb3000-laptop | then listen to the two files. and see if you can hear the sound then? |
15:50.44 | cneb3000-laptop | that will tell you whether it's sox combining the files (maybe we need to configure Sox somehow?) or it's never recording media at all |
15:51.15 | irroot | /usr/bin/sox -m <in> <out> <mix> |
15:51.27 | irroot | will mix once tested |
15:52.21 | cneb3000-laptop | good luck fulcan - i'm off to play mario kart with the missus! |
15:53.15 | fulcan | irroot now I have 2 44 byte files..... :/ |
15:54.01 | irroot | that is the header only i suspect |
15:54.27 | fulcan | irroot yup. at least it is getting that. |
15:55.28 | fulcan | irroot I don't see it being permissions at this point. a vox config thing maybe? |
15:55.51 | irroot | sox runs when m is used |
15:55.52 | fulcan | cneb3000-laptop thank you my friend!!!! |
15:56.43 | fulcan | irroot shoot that therory. possibly canreinvite=no? |
15:56.53 | fulcan | in sip.conf |
15:57.36 | fulcan | that effects the rtp stream, I know that much. but typically only for remote devices. |
15:58.01 | irroot | do a rtp debug and see if it goes through asterisk |
15:59.17 | fulcan | <PROTECTED> |
15:59.19 | fulcan | Sent RTP packet to 63.211.239.14:24332 (type 00, seq 026871, ts 058720, len 000160) |
16:00.05 | fulcan | there went that theory. |
16:00.16 | fulcan | I'm batting zero today. |
16:02.10 | fulcan | let me max out -vvvv and see if anything comes from that. I will pastebin and entire session from reload to call. |
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16:03.50 | p3nguin | If you haven't already figured it out, use MixMonitor() if you want a combined file without having to mix the channels manually. |
16:06.00 | fulcan | Here is an entire session. Just a thought, would the demo playing lock up the call recording? http://pastebin.com/uX2sYGaQ |
16:07.56 | fulcan | p3nguin We were able to get vox or sox or one of those ox's to preform the merge. We have that part disabled for debugging purposes. |
16:09.11 | irroot | ok |
16:09.24 | p3nguin | I'd still change it to MixMonitor() anyway. |
16:09.29 | irroot | put monitor in the begining |
16:09.43 | fulcan | irroot ? |
16:09.46 | irroot | not in the hangup |
16:10.06 | irroot | g [h@7185696288:1] Monitor("SIP/7185696288-00000008", "wav |
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16:10.47 | fulcan | irroot I have monitor in the very begining. before wait and even before answer. |
16:12.08 | fulcan | this is my entire extensions.conf http://pastebin.com/0cK3qp8C I like them clean. :) |
16:12.53 | fulcan | irroot I am not seeing what you are seeing my friend. :( |
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16:14.19 | p3nguin | By using the pattern _. to match the extensions rather than what I suggested last night, your Monitor() runs in places it shouldn't. |
16:14.40 | p3nguin | In the case of what irroot said, in the h extension. |
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16:15.24 | p3nguin | (0043.44) <p3nguin> exten => _[*#0-9].,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}-${EXTEN}.WAV,a); |
16:15.27 | p3nguin | (0044.04) <p3nguin> exten => _[*#0-9].,n,Goto(some-other-context,${EXTEN},1); |
16:15.52 | p3nguin | This is the concept I provided. Which works. I recommend using it because it works correctly. |
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16:16.52 | fulcan | p3nguin the peers were set up last night tho the best of my abilaties because of your recommendations my friend. shows up in 'show sip peers' now too. |
16:18.42 | p3nguin | Using my concept for recording calls, you could place those two lines in a context called, maybe, phones-r (for phones with recording). Then the some-other-context that I refer to would be the normal phones context where dialing would occur without being subjected to recording. |
16:19.11 | p3nguin | Then your peers would be assigned context=phones-r rather than just context=phones. |
16:19.53 | p3nguin | Assuming you're recording all outbound calls from phones, this seems to have good logic behind it. |
16:20.42 | fulcan | p3nguin making it real easy. I am recording absolutely everything that passes through the server. This server has one single job. :) |
16:21.10 | irroot | have a flag in astdb for what to record |
16:21.25 | irroot | based on callerid/channel/.... |
16:21.32 | p3nguin | Since every call that passes through it should have a phone number, an extension pattern of _. is a HORRIBLE choice. |
16:22.22 | p3nguin | _. matches EVERYTHING, including extensions that you should not be matching for recording, such as extensions s, h, i, and t. |
16:22.33 | WIMPy | Phone NUMBERS are so last century. |
16:22.50 | irroot | lol |
16:22.53 | p3nguin | I know, we should always match all extensions, right? |
16:22.57 | fulcan | p3nguin grabed that one of the internet from the asterisk cookbook. no real educated decision on the _. choice. just trying to get her to work. What the best, easiest and cleanest solution? |
16:23.09 | p3nguin | Scroll up. |
16:24.16 | fulcan | p3nguin add just that one line instead? your answer appeared more conceptual than 'big bird cookie monster' like what I need. |
16:25.03 | p3nguin | I used the word concept because you'll still need to add the contexts and assign contexts to phones,and you might want to adjust the file naming convention. |
16:25.39 | p3nguin | The two lines I gave you are what you need, but you also need more than those two lonely lines to have a working system. |
16:25.47 | fulcan | p3nguin one sec. |
16:26.35 | irroot | wimpy ipv6 or bust !!! |
16:26.37 | p3nguin | This extension will match any call to phone numbers starting with *, #, or a number, followed by one or more additional characters. |
16:26.59 | WIMPy | irroot: Not available here :-( |
16:27.17 | p3nguin | It would match *1. It would match #611. It would match 18004444444. |
16:27.42 | irroot | yeah i can hook up some via the providers but have not got the infrastructure ATM |
16:30.58 | p3nguin | Oh, and that's not going to be adding one line, as you suggested. You'll be adding three lines minimum: one for the [context] and the two extension lines I gave you. |
16:33.52 | fulcan | here is my complete dialplan, maybe you can give me a better plan than this. I had to do one or two silly things to get her to work like identical context names for both inbound and outbound but nice and simple. my sip.conf is OTB and the info you find here -> http://pastebin.com/tDdP6nk7 at the very bottom and I completely hosed extensions.conf to a clean file before working with her. |
16:40.13 | p3nguin | Well, you haven't used "identical context names." |
16:40.58 | p3nguin | But you did erroneously duplicate you sip peer names in sip.conf. You should fix that. |
16:42.36 | p3nguin | Are you using dtmfmode inband because you're trying to record tones? |
16:42.42 | fulcan | p3nguin the doc I was looking at said teliax wanted 2, one for inbound and another for outbound. |
16:42.56 | p3nguin | They're wrong. |
16:43.14 | p3nguin | I've never met an ITSP yet that knows how to correctly configure an end user Asterisk system. |
16:43.43 | fulcan | p3nguin I am assuming it takes the first one and ignores the second one making it easy to delete number 2 and I should be fine? |
16:43.56 | p3nguin | But let's pretend for a second that they're right; you still would never duplicate a peer name. |
16:44.12 | irroot | waves at p3nguin lol |
16:44.21 | fulcan | p3nguin cheers |
16:46.22 | p3nguin | I find it pretty silly (and somewhat sad) that persons or companies operating is that capacity don't know how to configure Asterisk well. |
16:47.28 | irroot | p3nguin indeed almost opposed to asterisk sometimes |
16:48.24 | fulcan | asterisk is grand daddy. first user auth system built on SER. |
16:48.29 | fulcan | :) |
16:48.41 | p3nguin | What kind of operation are you running where you are in a position to oppose Asterisk for end users? |
16:49.18 | irroot | no refering to a local ITSP that tells customer asterisk is bad and dont support it |
16:49.33 | WIMPy | They just don't want to be bothered with configuration issues. |
16:49.48 | irroot | lol nope they agents for audiocodes/patton |
16:50.11 | p3nguin | If they don't want to be bothered, they should invest a few minutes now to provide GOOD, WORKING samples to the customers so they can save hours later. |
16:51.32 | p3nguin | It should take under 30 minutes (this time includes reading The Book to see how to get it done) to configure a peer for a given ITSP and have it working. |
16:51.58 | p3nguin | Someone who doesn't need to read a book to figure it out should have it done in five or so minutes. |
16:52.44 | p3nguin | If you're the tech guy at the ITSP, you should be familiar with most of the options, so you just have to apply good, tested, working samples. |
16:54.15 | irroot | going to set up a ITSP shortly for asterisk customer as i haz a gui shiped all it will take is un/pw |
16:54.49 | irroot | and ill support with sample as you said above a copy paste sip/iax conf |
16:54.57 | irroot | and dialplan example |
16:55.08 | fulcan | This is what I was fighting last night. "NOTICE[10906]: chan_sip.c:20443 handle_request_invite: Call from '7185696288' to extension 'wuzamarine' rejected because extension not found in context 'default'" Teliax appears to want a context named after my username. Hence the reason I ended up with two peers with the same name, trying to figure out why asterisk was trying to force my username down it's gullet..??? |
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16:57.22 | irroot | no the extension is the name name the context is default |
16:57.49 | irroot | need exten wuzamarine,1,Goto(.....) |
16:57.57 | irroot | in the [default] section |
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16:58.34 | fulcan | <irroot> need exten wuzamarine,1,Goto(.....)??? |
16:59.31 | fulcan | explain the ..... a little better please. |
16:59.52 | irroot | [default] |
16:59.54 | irroot | exten => wuzamarine,1,Goto(context,exten,1) |
17:00.01 | irroot | the context is the [.....] section |
17:00.33 | irroot | the exten is as above a point in the context the priority is the 1 .... |
17:00.45 | irroot | this is a important concept to grasp |
17:00.56 | irroot | ~thebook |
17:00.56 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
17:01.31 | cneb3000-AFK | ^^^ the dial plan basics bit in that IS really helpful |
17:01.37 | cneb3000-AFK | and about a 15 minute read |
17:01.41 | cneb3000-AFK | goes back under his rock |
17:01.59 | p3nguin | err... apply good, testing, working values to create good, tested, working samples, rather. |
17:02.08 | p3nguin | s/testing/tested/ |
17:08.20 | fulcan | cneb3000-AFK I read the first asterisk book put out by a company in Atlanta I met through Digium back in '02. It's been years and I didn't play with the dial plan to terribly much because I typically use her to perform really simple task but work her like a Bi@*ch. I still have the first VoIP war fighter training simulation system for the US Army Sgt Major Academy and I don't think she has ever been rebooted. That was in '03. |
17:10.03 | fulcan | Built on asterisk and the city of el paso did an artical on it a year or two ago, just named it so they are still using the heck out of it. |
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17:11.00 | fulcan | I think that was the last time I really tore into asterisk. |
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17:40.50 | fulcan | she still barks at me about the username not being in my default [default] -> exten => myPhone1,1,GotoIF($["${CALLERID(num)}" = "wuzamarine"]?dial1)) and [myPhone1] blah... |
17:42.23 | irroot | exten => wuzamarine,1, |
17:42.36 | irroot | its sending it to the above |
17:42.40 | irroot | not myphon1,1 |
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17:56.51 | fulcan | how do you assign a static variable in the context? |
17:57.47 | irroot | SET(MYVAR=<VALUE>) |
18:04.50 | p3nguin | If it's not a global variable, that has to be in an extension rather than just a context. |
18:05.46 | p3nguin | If it's a global, then it's simply MyVar=value and it has to go under [globals]. |
18:06.39 | fulcan | http://pastebin.com/8VcnrQk2 :( |
18:06.39 | p3nguin | You can also set variables on the peers and then reference them in dial plan. |
18:06.57 | p3nguin | line 18 is wrong. |
18:07.20 | dr0ck | or GLOBAL function |
18:07.32 | p3nguin | line 30 doesn't have a priority. |
18:07.47 | p3nguin | line 40 doesn't have a priority. |
18:08.04 | p3nguin | And yes I see that they are commented out, but that shows that you intend to use them at some point. |
18:08.15 | fulcan | p3nguin if I remove the comment is errors. |
18:08.25 | p3nguin | Of course it does. It's wrong. |
18:08.36 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
18:09.13 | p3nguin | You also don't need line 31 and 41. BackGround() provides the answer when it gets to line 34 and 44. |
18:10.38 | fulcan | p3nguin glad to see you see its wrong, what would you suggest to make it right? |
18:11.48 | fulcan | there is no variable in peers that also appears in dialplan. |
18:12.41 | fulcan | line 34 and 44, I could care about, not even close to that yet. |
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18:17.27 | fulcan | <PROTECTED> |
18:17.48 | p3nguin | http://pastebin.com/v7CBZbdm |
18:18.32 | fulcan | http://pastebin.com/tzMcynSH |
18:19.21 | p3nguin | And I'll rebut again with http://pastebin.com/v7CBZbdm |
18:19.29 | p3nguin | I rewrote your dial plan for you. |
18:19.34 | p3nguin | correctly |
18:20.26 | ChannelZ | will you come vacuum my house? |
18:20.53 | p3nguin | Depends on if you have a Dyson or not. |
18:21.14 | p3nguin | Those things really suck. |
18:21.53 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
18:21.55 | jc319 | hello |
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18:23.51 | ChannelZ | Actually I do, and it's awesome |
18:24.02 | fulcan | <PROTECTED> |
18:24.39 | fulcan | p3nguin That is using yours word for word my friend. |
18:25.04 | p3nguin | Yep. Now go fix your sip.conf. |
18:25.07 | ChannelZ | which doesn't have an extension named 'wuzamarine' in default just as it says. |
18:25.35 | p3nguin | default doesn't deserve to have anything in it. He needs to go fix is peer entry so it matches the device making the call. |
18:25.50 | p3nguin | It's a KNOWN device, not an anonymous device. |
18:26.26 | ChannelZ | Just pointing out that the error is pretty straightforward. |
18:26.33 | fulcan | http://pastebin.com/UeFrrtTR There is NOTHING in sip.conf with wuzamarine in it except for my registration string. thats all. |
18:26.33 | p3nguin | And in addition to that, no where in your dial plan did you create an extension called 'wuzamarine'. |
18:26.58 | p3nguin | So you can't expect that a call TO wuzamarine will ever succeed. |
18:27.15 | p3nguin | Most people can't dial words on their keypads, anyway. |
18:27.55 | fulcan | p3nguin in case you didn't notice, I am trying to eliminate that word by even trying to set it to a variable. |
18:28.08 | p3nguin | I have no idea what that statement even means. |
18:28.55 | p3nguin | Until you fix your sip.conf to match peers, your dial plan will never work. |
18:28.58 | p3nguin | So fix that first. |
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18:30.40 | fulcan | where do you see any reference to 'wazamarine' in this sip.conf? http://pastebin.com/ |
18:31.05 | p3nguin | Without even looking, I can tell you that it does not matter. |
18:31.20 | fulcan | http://pastebin.com/QfqBYeHT |
18:31.25 | p3nguin | Doesn't matter. |
18:31.43 | p3nguin | You have to match the peer by creating a correct sip peer definition. |
18:31.55 | fulcan | Your telling to "fix my variable that am using in sip.conf that has wasamarine". |
18:31.59 | fulcan | correct? |
18:31.59 | p3nguin | Then assign the context that you want calls from that device to go into. |
18:32.10 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
18:32.16 | p3nguin | No, I did not say that. |
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18:32.26 | ChannelZ | is 7185696288 a local device? A softphone or something? |
18:32.40 | fulcan | then, where in the world is 'wazamarine' coming from? |
18:32.49 | p3nguin | host=xxx.teliax.net |
18:33.08 | p3nguin | The refusal to fix the fucking sip.conf is not gaining you any ground. |
18:33.52 | ChannelZ | fulcan you actually answered your own question earlier: "There is NOTHING in sip.conf with wuzamarine in it except for my registration string. thats all." |
18:34.16 | p3nguin | Here. I've started fixing it for you. Now you just have to do the rest. http://pastebin.com/1zezCFB9 |
18:34.50 | fulcan | very good, now there NO reason for me, or asterisk or anyone for that who should care about that username after authentication. you know, little thing we forget about on purpose. why is it even looking for a context |
18:35.03 | ChannelZ | You have 7185696288 not set to go to any context which is why it's going to 'default' |
18:35.23 | p3nguin | And that's just one of the issues. |
18:35.37 | ChannelZ | And we haven't seen your registration line (or I haven't) and/or maybe Teliax just sends calls to your username as an extension always, I have no idea. |
18:36.16 | p3nguin | It needs at the very least a defaultuser and a context. |
18:36.20 | p3nguin | for teliax |
18:37.16 | p3nguin | It'll amuse me if the register statement end with /wuzamarine |
18:37.27 | ChannelZ | place yer bets |
18:37.42 | p3nguin | What kind of odds are you giving? |
18:38.17 | ChannelZ | I'll give you a cookie. |
18:38.18 | fulcan | handle_request_invite: Call from '' to extension 'wuzamarine' rejected because extension not found in context 'default' |
18:38.38 | ChannelZ | Hey, check it out. Pretty much the same error |
18:38.45 | fulcan | yup! |
18:39.26 | p3nguin | If I had a working example of a teliax config, I'd write the whole damn entry myself just so we can move on to the next issue. |
18:40.47 | fulcan | p3nguin I had it to the point that you could reach the recording but _. was, well you know. |
18:41.48 | p3nguin | You had it in default context, which was wrong, and you were using _. which was also not good. This is why we start at the beginning of the problem to fix it rather than continuing in the middle. |
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18:42.21 | p3nguin | Until you match the peer AND assign the correct context (not default), don't expect it to work. |
18:42.35 | fulcan | It seems to be using my username as caller ID. :/ |
18:43.01 | p3nguin | Then you should consider using trustrpid=yes and sendrpid=yes in the peer entry. |
18:45.07 | fulcan | p3nguin added it to the teliax peer group but same error. |
18:45.29 | p3nguin | How about that register statement? |
18:45.45 | p3nguin | I think you told me caller ID when you meant extension. |
18:46.03 | fulcan | p3nguin " Until you match the peer AND assign the correct context" I am using 100% what you sent me. |
18:46.19 | p3nguin | (1334.11) <p3nguin> Here. I've started fixing it for you. Now you just have to do the rest. |
18:46.30 | p3nguin | You didn't do the rest. |
18:46.58 | p3nguin | And we're still waiting to see the register statement. |
18:47.43 | p3nguin | I'm starting to become bored with this. Maybe I need to go make some phone calls or something. |
18:48.05 | fulcan | explain this "peer AND assign the correct context" in more detail please. |
18:48.22 | p3nguin | The peer definition. In sip.conf. Fix it. |
18:49.18 | p3nguin | The last time I saw it, it didn't have context=anything-useful in it. |
18:49.33 | p3nguin | As channelz pointed out, that's why the calls are going to the wrong (default) context. |
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18:51.40 | fulcan | p3nguin how can I get asterisk the strip the username before it even hits extensions.conf because even setting the context in sip.conf yields the same error. |
18:54.07 | p3nguin | Here's an example of a good peer definition: http://pastebin.com/fJgNLGLM |
18:54.31 | p3nguin | It's for VoIP.ms, but Teliax shouldn't be TOO much different. |
18:55.11 | jc319 | hi p3nguin, how are you? |
18:55.19 | p3nguin | irritated |
18:55.27 | jc319 | :) |
18:55.27 | jc319 | why |
18:55.44 | jc319 | I just came in now, it can't be me, right? |
18:55.56 | p3nguin | I do most of the work for people trying to help, and they don't accept the work I provide. |
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18:57.09 | jc319 | I for one appreciate your hard work |
18:57.57 | irroot | +1 |
18:57.59 | jc319 | I have a great game plan today: copy & paste voip.ms wiki example and test, when it doesn't work call them. almost error proof. |
18:58.38 | p3nguin | I have a better plan. |
18:59.01 | p3nguin | Since I use VoIP.ms and I can make calls to your UK DID, configure yours like mine is. |
18:59.19 | p3nguin | I'll even configure it FOR YOU just so we can move on. |
19:00.11 | jc319 | Have to agree that's definitely the most efficient plan |
19:00.33 | jc319 | Can I provide you extended information then? 2 DIDs now, 1 UK and 1 US |
19:00.40 | fulcan | http://pastebin.com/sSd1bC05 |
19:00.46 | p3nguin | Absolutely. |
19:01.16 | jc319 | and I figured why it is $1 for you while $4.5 for me. I went with the 3500 minutes inclusive plan (perhaps unnecessarily). I'll have a look now and see if I can switch at the end of first month |
19:01.30 | p3nguin | The teliax peer still needs the username/defaultuser added. |
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19:02.41 | p3nguin | Oh, I use the pay-per-minute service because I don't take that many inbound calls on my toll-charge number and my toll-free number is pay-per-minute regardless. |
19:03.27 | fulcan | and this is with defaultuser=wuzamarine added Call from '' to extension 'wuzamarine' rejected because extension not found in context 'default'. |
19:03.57 | p3nguin | Now set an appropriate context in the teliax peer. |
19:04.21 | p3nguin | Setting no context is not appropriate. Setting it to default is not appropriate. |
19:04.40 | p3nguin | You currently have no context set in it. |
19:05.48 | p3nguin | jc319: So... what's your next step? |
19:07.59 | fulcan | a little farther, asterisk doesn't want to shake the username ' Call from 'wuzamarine' to extension 'wuzamarine' rejected because extension not found in context 'decisions'" |
19:08.32 | p3nguin | You're making progress. |
19:09.31 | p3nguin | Now to understand where extension 'wuzamarine' is coming into the equation. Are you calling to a DID number that you bought from Teliax? |
19:09.58 | fulcan | yes |
19:11.00 | p3nguin | If you added a line below [teliax] that says context=decisions, then we've proved that the peer is finally matching. |
19:11.08 | fulcan | the server is nothing more than a gateway. capture incoming calls and then forward to an outside line (we are no where near this part though. |
19:11.55 | fulcan | yes, matching the peer but NoOp is not getting the CID. |
19:12.21 | fulcan | it is getting the username instead. |
19:12.41 | p3nguin | CID and extension are two completely different bits of data. |
19:12.59 | p3nguin | CID is where the call comes FROM. Extension is where the call is going TO. |
19:13.03 | fulcan | It need the CID |
19:13.11 | fulcan | yup |
19:13.27 | fulcan | and it is coming from Teliax. |
19:13.40 | p3nguin | Right now, they are sending the call to extension 'wuzamarine'. This is unrelated to CID. |
19:14.03 | fulcan | it is ABSOLUTELY not supposed to. |
19:14.44 | fulcan | there is not now nor will there ever be and extension called 'wuzamarine'. |
19:15.00 | p3nguin | If they can't or won't send calls to your phone number, and if you only have one single phone number, we can create the 'wuzamarine' extension. I don't want to do that, but we may have to until you can get Teliax to explain what they are doing. |
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19:15.37 | fulcan | I have 2 phone lines coming off that 1 single pipe. |
19:15.47 | fulcan | there will be 200 |
19:15.54 | p3nguin | In my opinion, calls should be sent to an extension matching the phone number that the caller dialed. |
19:16.06 | p3nguin | So you're going to have 200 phone numbers with Teliax? |
19:16.20 | fulcan | that's fine. just as long as it is not wuzamarine |
19:16.23 | fulcan | yup |
19:17.02 | p3nguin | Unless some other Teliax user pops in here and clues me in on why the call goes to your username as the extension rather than your phone number, you will probably have to ask Teliax about it. |
19:17.12 | p3nguin | I know there are Teliax users here. |
19:17.17 | fulcan | so, having an extension called wuzamarine is useless. |
19:17.33 | p3nguin | If you're going to have more than one phone number, it sure is. |
19:18.02 | fulcan | I know, I have been fighting this same battle all night long. Yours what simply a more educated approach to the same brick wall. |
19:18.03 | p3nguin | It's either a Teliax configuration problem or your peer entry is still missing something. |
19:23.45 | p3nguin | See if insecure=port or insecure=port,invite in the teliax peer make any different. |
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19:43.21 | cneb3000 | bit of an out there question, but is telephony related.. |
19:43.41 | cneb3000 | i got hold an openmoke neo freerunner phone the other week. i have no use for it. |
19:43.49 | cneb3000 | by any chance is anyone looking for one on the cheap? |
19:44.07 | p3nguin | Are they SIP phones? |
19:45.52 | p3nguin | Just looks like a GSM/GMRS phone to me. |
19:45.59 | cneb3000 | hmm.. not really like that. it's kind of like a sandbox phone. so you would have to develop a SIP app on it. |
19:46.08 | p3nguin | err, GSM/GPRS |
19:46.23 | cneb3000 | ^^^ it has wifi capability. |
19:46.54 | p3nguin | That could be useful. Maybe. :) |
19:47.26 | cneb3000 | well, itll be on ebay soon.. so! |
19:47.34 | cneb3000 | if its of no use then dont vbother ;) |
19:47.46 | cneb3000 | collecting dust for me |
19:48.13 | p3nguin | Let's just cut to the bottom line. How much are you trying to get for it? |
19:48.24 | cneb3000 | £100 |
19:51.41 | jc319 | p3nguin: I discovered why I lost outbound call... Apparently this happened when I switched outbounds from main account to sub-accounts. As of now sub-accounts cannot make calls on my system, I use exactly the same config just add "_home" sub-account suffix and the corresponding password |
19:52.02 | jc319 | p3nguin: Are you using any sub-accounts at all? If so, do you call outbound with them? |
19:52.44 | p3nguin | I only use sub-accounts for my Asterisk systems and I do call outbound. My main account is basically a placeholder. |
19:53.56 | jc319 | Precisely what I had on mind. |
19:54.16 | jc319 | So it is possible... I'll check the web panel to see if there's any limitation or anything |
19:54.26 | jc319 | on this sub-acct.. |
19:55.16 | p3nguin | There is no setting to block calls on my subs. |
20:10.26 | p3nguin | jc319: So what's the next step? I'm ready to move on. |
20:15.32 | jc319 | Allow International Calls |
20:15.32 | jc319 | value="0" >Yes - International Calls Enabled |
20:15.32 | jc319 | value="1" selected >No - International Calls Disabled |
20:15.56 | jc319 | Now it's clear why the line has been BUSY |
20:16.10 | *** join/#asterisk wonderworld (~ww@port-92-201-214-139.dynamic.qsc.de) |
20:16.15 | jc319 | why can they not say FORBIDDEN...... |
20:20.32 | jc319 | ok so I'll load up the old config which you build for me (wit outpeer and etc) and see all works as I want (or actually need to find out what I want). once the basics are ok next in my list is to finally put the 2nd line buttons to use. (long term goals are voicemail, voicemail-to-email, auto attendant but I doubt I can finish them in a month with my current pogress speed) |
20:25.56 | p3nguin | If you're willing to spend a little money on it, it all can be done within a day. |
20:26.28 | p3nguin | Free support is, well, free. And often it's slow. |
20:28.25 | jc319 | Yes I'm sure you or another expert can do this all in a few hours. I guess even without much knowledge & with my speed, still can be done within a day by cheating and using a packaged distro like trixbox :D |
20:29.12 | p3nguin | Trixbox sucks. |
20:29.47 | jc319 | I actually enjoy the learning process, weird but it's a fact. BTW my 7960 sometimes goes crazy (e.g. right now), the line icon is cycling 5 times/second |
20:30.09 | p3nguin | I'm not sure if there is a single person here who could provide you with support for Trixbox if you made the mistake of going that route. |
20:31.04 | jc319 | I want to see what's going on behind the scenes so not considering using trixbox. I did try to see what's it about though. |
20:37.03 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
20:42.26 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
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20:44.59 | jc319 | Is there a good free Windows SIP softphone with multiple lines? (also for MAC OS X?) |
20:45.13 | p3nguin | I like zoiper. |
20:51.35 | [sr] | jc319: 3CX |
20:54.49 | *** join/#asterisk jcook_5xdata (~jcook_5xd@173.162.32.1) |
20:58.13 | [sr] | p3nguin: cool, zoiper has IAX! |
20:59.07 | *** join/#asterisk wonderworld (~ww@port-92-201-214-139.dynamic.qsc.de) |
20:59.48 | p3nguin | Yep, zoiper classic is my favorite for Windows machines. |
21:00.41 | [sr] | i was searching for a IAX win client for a few weeks |
21:01.11 | p3nguin | I guess I never saw you ask here. |
21:02.26 | [sr] | never did, only on google in fact |
21:02.47 | [sr] | i kinda forgot the matter for some time, just checked it now for curiousity |
21:06.51 | *** join/#asterisk ruied (~ruied@pa4-84-91-140-68.netvisao.pt) |
21:09.25 | [sr] | p3nguin: thanks for the tip anyway :) |
21:11.10 | jc319 | if my local Asterisk talks to ITSP using IAX, my deskphone/softphones can still use SIP to talk to my Asterisk, there's no connection between two link channels, right? |
21:19.51 | [sr] | bed time for me |
21:19.53 | [sr] | see ya |
21:24.04 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
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21:30.14 | p3nguin | jc319: That's basically correct. Since Asterisk is a B2BUA, it bridges dissimilar channel technologies together. I use IAX2 between my Asterisk systems and VoIP.ms, but I use SIP and SCCP phones. |
21:32.24 | p3nguin | The only reason I use IAX2 is for bandwidth savings during multiple concurrent calls. That's done with trunking, which SIP does not support. |
21:39.47 | JerJer | heh - until the first iax2 worm hits |
21:40.52 | sxpert | highly unlikely, but who knows ;) |
21:41.24 | JerJer | i / we already found one 'issue' |
21:43.02 | JerJer | s/already/previously/ |
21:43.22 | WiretapWork_ | how far away is 1.8.5? |
21:45.43 | *** join/#asterisk cneb3000 (~cneb30000@02da1dc1.bb.sky.com) |
21:46.56 | p3nguin | As they say, as soon as it's ready it will be released. |
21:56.22 | WiretapWork_ | lol |
21:56.23 | WiretapWork_ | nice timeline |
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22:17.51 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
22:18.15 | p3nguin | I've never understood why people expect developers to predict or guess when they'll be done working on something. |
22:19.35 | p3nguin | When it's done, it's done. And not before that. |
22:21.46 | WiretapWork_ | as a developer, I usually give some indication of an expected release date |
22:21.55 | p3nguin | That's just silly. |
22:22.06 | WiretapWork_ | while that goalpost can move regarding issues, it gives people some kind of idea when to prepare for a new release |
22:22.34 | WiretapWork_ | meaning that they can have their systems ready to accept said release on time |
22:22.37 | p3nguin | Unless you're psychic, there's no way you can predict when everything will fall into place, be done with testing, pass reviews, and then be released. |
22:22.54 | WiretapWork_ | you can't predict it exactly |
22:23.13 | p3nguin | Because you can't predict it, don't try to predict it. |
22:23.28 | WiretapWork_ | but you can give some kind of indication based on past experience, the rate of present progress and the number of tasks remaining |
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22:53.27 | jc319 | Yeah I that SIP 'trunk' is not a single 'wide' channel or anything, they should call what they're selling (SIP trunk) a SIP uplink |
22:54.22 | jc319 | p3nguin: In the light of your explanation above, I think it makes perfect sense to switch to IAX at some point, but I will keep SIP just a bit longer... |
22:55.28 | p3nguin | Many people feel that you should use SIP between your system and any system which you do not have control over. Similarly, use IAX2 between any of your own systems. |
22:55.56 | p3nguin | But I use IAX2 for the trunking even though I can't control VoIP.ms's side. I don't have any problems with it. |
22:57.18 | jc319 | They prefer IAX2 because it makes sense with bandwith saving, why do they prefer SIP in uncontrolled territory? Is it about compatibility? Calls are more likely to 'connect' or what? |
22:59.04 | p3nguin | There have been reports that IAX2 sometimes has problems between dissimilar Asterisk versions. That might have something to do with it. |
23:00.33 | p3nguin | I think maybe the SIP "standard" might have more tolerance or something. I'm just not sure. I use IAX2 whenever I have a chance, regardless if I have control over both ends. |
23:01.43 | jc319 | OK will do the same. Read Mark Spencer's mail (from 2004) SIP has some advantages at the bottom of the email but IAX seems much better overall... |
23:02.08 | WiretapWork_ | IAX2 requires a reliable timing source |
23:02.15 | WiretapWork_ | or so I've heard |
23:03.16 | jc319 | Do you use any simple plug-n-play hardphones for any users? Once I have the system up & running I would like to send one to my parents, it must be very easy to set up before shipping and easy to support. I have seen some Cisco boxes relatively cheap but would like to know if you have any experience to benefit from rather than trying several devices until I find a good one? |
23:03.23 | *** join/#asterisk cmendes0101 (~nn@pool-173-51-199-161.lsanca.fios.verizon.net) |
23:03.30 | jc319 | hardware based timing source? |
23:07.16 | p3nguin | One HUGE reason to use IAX2 is NAT traversal where SIP runs into a problem. If you're behind a NAT that SIP can't get through, in almost all cases IAX2 will succeed. |
23:08.02 | WiretapWork_ | jc319, I config and ship 7912s to my agents (they work from home) |
23:08.08 | p3nguin | You could use a 7900 series phone or you could use one of the newer 500 series phones. |
23:08.11 | WiretapWork_ | the setup is trivial |
23:08.31 | WiretapWork_ | the XML-config based 79xx dont' really appreciate being away from their TFTP server |
23:08.35 | p3nguin | Or you could go with a different brand completely: many people love the Polycoms. |
23:08.55 | WiretapWork_ | (And require SIP-TCP which is not properly supported in 1.8.4) |
23:09.40 | p3nguin | Probably the 7912, and for sure the 7940 and 7960 run SIP just fine without the tftpd as long as you've loaded from the tftpd at least once. |
23:09.48 | jc319 | I think I have 8 or so 7910s or 7912? would both work? I'll check the version now |
23:09.50 | WiretapWork_ | yep |
23:10.09 | WiretapWork_ | the 7912, 7940 and 7960 are the three phones cisco will 'officially' allow you to use with non CUCM gear |
23:10.12 | p3nguin | The boot-up process is lengthened significantly in the absense of the tftpd, but they eventually boot from "memory." |
23:10.20 | WiretapWork_ | the 7910 is SCCP only if I'm not mistaken |
23:10.45 | jc319 | model is 7905 would this work? (I also have 7940 in worst case I can send that if it'll work) |
23:10.58 | WiretapWork_ | if I'm not mistaken the 7905 may work |
23:11.08 | WiretapWork_ | you'll have to see if cisco have released a non-cucm firmware |
23:11.12 | WiretapWork_ | or if its cucm-only |
23:11.27 | WiretapWork_ | the non-cucm firmware is the one that is easiest to 'ship-n-go' |
23:11.39 | jc319 | Cisco IP Phone 7905G/7912G (SIP) Release Notes for Firmware Release 1.2.0 |
23:11.49 | WiretapWork_ | sweet |
23:11.54 | WiretapWork_ | looks like its just a 'dumb' 7912 |
23:12.15 | WiretapWork_ | I _Really_ need to finish my writeups |
23:12.31 | WiretapWork_ | since my article on 79xx Unified phones is ranked 5th on google |
23:12.35 | WiretapWork_ | but is incomplete |
23:12.52 | jc319 | great, I thought this would require a server to boot from etc. So I just enter the SIP details using TFTP once and it'll be the permanent config right? (and will work until sip details change) |
23:13.42 | jc319 | On what site is your article? |
23:16.02 | WiretapWork_ | wiretap.net.nz |
23:16.08 | WiretapWork_ | and no |
23:16.11 | *** join/#asterisk golikwid|mac (~chrislees@207.30.30.130) |
23:16.20 | WiretapWork_ | the 7912,7940 and 7960 can be configured on the phone screen or via web interface |
23:16.22 | WiretapWork_ | TFTP is optional |
23:16.34 | WiretapWork_ | I use them in places where a TFTP server is unavailable |
23:17.31 | jc319 | Exactly, they can provide only an RJ-45 jack on home broadband router. would be great if I can add voip to that setup (with 7905 in this case). |
23:17.40 | WiretapWork_ | yep |
23:17.50 | WiretapWork_ | power up the 7905, go into the menu and configure it |
23:18.00 | WiretapWork_ | you'll need to **# to unlock the config |
23:18.18 | WiretapWork_ | make sure your PBX has an external hostname |
23:18.22 | WiretapWork_ | that matches internal |
23:18.29 | WiretapWork_ | so that you can test the config before you ship |
23:22.22 | jc319 | Will do thanks, I'll also test on some other site before I ship. I'll have to think about whether using my in-house Asterisk or ITSP for 7905s uplink though. Speed & availability on ITSP would be much better obviously but I can provide more functionality locally hmm |
23:23.38 | WiretapWork_ | depends on how many channels your local can support |
23:23.48 | WiretapWork_ | I trunk 5 phones off an ADSL line |
23:23.51 | WiretapWork_ | works fine |
23:27.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
23:27.56 | p3nguin | Asynchronous Digital Subscriber Line line? |
23:28.37 | WiretapWork_ | yes, but at least its not a SHVADSL line as someone once told me they had :P |
23:29.59 | p3nguin | I'll file that right below "Personal Identification Number number." |
23:30.12 | p3nguin | but before "Automated Teller Machine machine." |
23:30.35 | jc319 | I think speed would be OK but reliability is my main concern, I have the habit of fixing working things, and they work better in the end - now that's a good thing. However some other parties claim the down time (days/weeks/months 'don'n worth it') of course they are wrong but still it is a valid concern (for them) :D |
23:30.54 | jc319 | don't* |
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23:32.25 | WiretapWork_ | jc319, learn to 'swap out' |
23:32.55 | WiretapWork_ | p3nguin, Symmetric Highspeed Very-highspeed Asymmetrical Digital Subscriber Line Line |
23:38.42 | *** join/#asterisk linuxgecko (~playgroun@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
23:49.46 | linuxgecko | it seems that i have forgoten too many basics to do anything more than run an asterisk daemon. i can `asterisk -r` into it, but it's not seeming to read/run my dialplan in extensions.conf, nor is it handling sip.conf, gtalk,conf, or jaber.conf. what is the most likely thing I've done horribly wrong? |
23:50.37 | WiretapWork_ | forgotten to load any modules |
23:52.52 | linuxgecko | WiretapWork_: sounds right, but iv'e forgotten where to.how to do that in the config |
23:54.50 | linuxgecko | WiretapWork_: i'm VERY rusty, and just need RTFM pointers. sorry for needing `stupid` help. |
23:55.32 | linuxgecko | s/and just need/and MAY just need/ |
23:56.09 | linuxgecko | wow... :) never seen someone actually make a bot that does that :) |
23:57.06 | WiretapWork_ | on phone |