IRC log for #asterisk on 20110612

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)
00:15.51jc319p3nguin: 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?
00:16.02jc319(and fix the 7960-unable-to-logon issue).
00:16.30jc319s/logon/register
00:17.15jc319how does this thing work, needs whitespaces between ?
00:17.16jc319s/between/betwe3n
00:17.31jc319s/between/betwe3n/
00:17.33jc319aha
00:34.01jc319p3nguin: 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?
00:49.17p3nguinjc319: I changed it to reflect what you've said.  Refresh the pastebin page.
00:50.20p3nguinjc319: The sed replacement works as it should, but you can't make a mistake in your syntax -- you must terminate your expression.
00:51.08p3nguinI 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)
00:57.02fulcanIs this a live channel?
00:57.20WIMPyNo, we are all zombies.
00:59.54virolol
01:01.17WIMPyYou're only allowed to write here if you're dead or at least doomed.
01:02.50jc319p3nguin: Does that updated one mean one extension cannot call another by typing in 3 digits anymore?
01:15.04p3nguinjc319: Phone configurations have nothing to do with how another phone is reached.  That is purely done in dial plan by creating extensions.
01:15.10p3nguinSo to answer your question directly, no it does not mean that.
01:16.35p3nguinJust 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.
01:18.55jc319bonne 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)
01:24.12jc319p3nguin would you mind having a look please? http://pastebin.com/LZ9Rp0ec
01:25.02jc319I started with your basic example but changed some settings when it did not work. and some are commented to test later..
01:34.09p3nguinjc319: You need to fix nat, directmedia, localnet, and externaddr in general.
01:35.44p3nguinGet rid of defaultuser, fromuser, and authname in the phone entries.
01:35.44jc319ok now in all ITSP definitions is nat=no
01:36.23jc319that 3 is gone
01:36.34jc319def/fr users and authname gone.
01:37.24p3nguinremove all the extraneous voipms inbound contexts.  You need only one, and define your phone numbers in it.
01:37.31jc319directmedia >> what about it? I can't find it in the document
01:37.50p3nguinYou still need to fix nat, directmedia, localnet, and externaddr in general.
01:38.47jc319OK all 4 ITSP's have "context=voipms-inbound" now
01:39.04jc319ok now in all ITSP definitions is nat=no. >> what else?
01:39.09p3nguinI would also apply a more sensible outbound context name.
01:39.17p3nguin(2034.09) <p3nguin> jc319: You need to fix nat, directmedia, localnet, and externaddr in general.
01:39.21p3nguin(2037.49) <p3nguin> You still need to fix nat, directmedia, localnet, and externaddr in general.
01:39.24p3nguinGENERAL
01:39.26p3nguingeneral section
01:39.46jc319doh, ok sorry I read 'in general'.. ok
01:41.34p3nguinI guess I should have said "in [general]."
01:43.07jc319shall I add all of these to general section >> http://pastebin.com/25zRjgBE (there's a note here saying those values worked before)
01:43.40p3nguinI would use them all.
01:44.28jc319outbound context name - that's the best I could come up with, any suggestions?
01:44.33p3nguinI 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.
01:44.44p3nguinvoipms-outbound or just outbound
01:45.09p3nguinoutgoing, outgoing_calls, voipms-outgoing
01:45.13jc319voipms-outbound but there are 4 sub-accounts, I need to distinguish them, no?
01:45.13p3nguinanything that makes sense.
01:46.17jc319voipms_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)
01:47.21p3nguinIf 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.
01:48.13jc319ok these are the names then:
01:48.16jc319include => voipms_inbound;
01:48.16jc319include => voipms_abiz-inbound;
01:48.16jc319include => voipms_gbiz-inbound;
01:48.16jc319include => voipms_home-inbound;
01:48.16jc319include => voipms_outbound;
01:48.45jc3191st one is main account, rest are sub-accounts
01:56.45p3nguinThat's not going to work.
01:57.25p3nguinNow you're just arbitrarily including contexts in other contexts.
01:58.05p3nguinYou actually need ONLY ONE inbound context with all of your phone numbers in it.
01:58.13p3nguinNo includes.
01:58.33p3nguinOne inbound context, and all the voipms peer entries have a context set to that inbound context's name.
01:58.42p3nguinThen define all your phone numbers in that context.
01:59.07jc319[phones];
01:59.07jc319include => internal;
01:59.07jc319include => voipms-outbound;
01:59.07jc319& [echo code here]
01:59.23jc319is this all that [phones] should contain?
02:04.33p3nguinprobably
02:05.22jc319Ok 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)
02:07.51p3nguinYou've got extensions matching 11-digit numbers.
02:08.13p3nguinWell, one pattern to match 11-digit numbers, which is suitable.
02:08.41p3nguinBut you need to PROPERLY set setvar=OutPeer= for each phone (line) that you're using.
02:10.03p3nguinAnd your permit lines in every single voipms peer should be permit=78.129.153.20/255.255.255.255 rather than what you have.
02:10.20p3nguinAn ACL is of no use if you apply it incorrectly.
02:14.03jc319ACL fixed now.
02:14.39jc319There's a msg about new syntax of that conditional header rewriting http://pastebin.com/TnP30WuV
02:16.57anumorayohi  all
02:17.32anumorayohaving program with make outgoing and incoming calls
02:18.15anumorayoi get a failed tone for incoming and a all circuits are busy
02:18.50anumorayoi am using an isdn card b410p
02:19.03*** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net)
02:19.33WIMPyDo you have your extensions in the right format? What does the console say when you try to call?
02:20.49anumorayofro incoming call it genrates no out in cli mode
02:21.19WIMPycore set verbose 9
02:21.26WIMPycore set debug 9
02:21.36p3nguinor verbose 4, since 9 doesn't produce more output
02:21.43WIMPyIf you still se nothing,the call doesn't get to Asterisk at all.
02:22.19p3nguinNew syntax, eh?  ExecIf(<expr>?Set(CALLERID(num)=value
02:22.47p3nguinLet me see how that compares to my old syntax.
02:23.11jc319p3nguin: thanks. By the way, I still receive the following if I use the 2nd line button on 7960:
02:23.12jc319[Jun 12 02:21:54] WARNING[2196]: chan_sip.c:13698 check_auth: username mismatch, have <000AB747A418-a>, digest has <000AB747A418-b>
02:23.12jc319[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
02:23.31p3nguinI have ExecIf($[${IF($["${externalCID}" != ""]?1)}],Set,CALLERID(num)=value
02:23.59jc319Related to the above, in sip.conf I have "match_auth_username=yes" (does not seem to help much)
02:24.09WIMPyIf If?
02:24.32anumorayoyes you are correct  i am not hit  the astrisk at all
02:24.41jc319ExecIf($[${IF($["${externalCID}" != ""]?1)}],Set,CALLERID(num)=${externalCID}); <yes this is the old one
02:24.41p3nguinSo let's change the line to:  ExecIf($[${IF($["${externalCID}" != ""]?1)}]?Set(CALLERID(num)=${externalCID}));
02:24.45jc319ok
02:25.29jc319so you basically changed ? to ,
02:25.42p3nguinAnd changed the Set command within the ExecIf().
02:25.51p3nguinIt was Set,CALLERID(num)
02:26.06p3nguinNow they indicate they want Set(CALLERID(num)=value)
02:26.16anumorayois that case of me not using the right channel
02:26.18WIMPyanumorayo: So check your dahdi config or perhaps your cabling. What does 'dahdi show status' give?
02:26.18jc319Oh yes
02:26.26p3nguinIt makes more sense this new way, since that's the syntax for the Set() command anyway.
02:26.55p3nguinIt was ExecIf()'s syntax that required the commas.
02:28.49anumorayowimpy B4XXP (PCI) Card 0 Span 1                OK      0      0      0      CCS AMI  YEL      0 db (CSU)/0-133 feet (DSX-1)
02:29.00anumorayosorry  guy
02:29.40anumorayohttp://pastebin.com/B13vH60E
02:30.09anumorayoWIMPy: check that out
02:30.12WIMPy"OK" is good. Are you sure the number is routed to that inteface?
02:30.38WIMPyYou're using two interfaces only, I assume?
02:30.55anumorayoonly 2
02:31.06jc319p3nguin: 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)
02:31.34anumorayohow do i check i routing to the correct interface
02:31.57WIMPyanumorayo: Ask your telco.
02:32.32WIMPyanumorayo: 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)
02:34.27WIMPyanumorayo: What about calling out?
02:34.51jc319p3nguin: just received an email, they've activated my 2nd DID (local one)
02:35.30anumorayoi can't test that now sorry
02:35.59WIMPyanumorayo: 'channel originate'
02:36.28p3nguinjc319: The change looks right to me.  Does your verbose output indicate that I did it wrong?
02:36.42anumorayosorry
02:38.26anumorayoi am in the UK if that is wjat you are asking
02:39.14WIMPyanumorayo: No, I'd like to find out, what happens if you try to plsce a call from that line.
02:39.59WIMPyYou can use 'channel originate' if you don't have access to a phone.
02:41.29anumorayowhat from cli ?
02:41.40WIMPyyes
02:42.55anumorayoi am getting No such command
02:43.13jc319p3nguin: 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> )
02:43.34jc319p3nguin: http://pastebin.com/z1RBYNKL
02:44.00WIMPyanumorayo: For ISDN connections, I'd recommend a recent version of libpri and Asterisk. For old versions it's only originate.
02:44.31p3nguinjc319: I just thought of something...
02:44.57anumorayoyes i have that command
02:45.41anumorayohowo use it
02:45.53p3nguinjc319: 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});
02:46.10WIMPyanumorayo: It tells you if you use it without parameter.
02:46.36WIMPyanumorayo: e.g. 'originate dahdi/1/yourmobilenumber application sayunixtime'
02:46.42p3nguinYou'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.
02:47.06p3nguinIf that's the problem, adding the 00 will fix it.
02:50.04p3nguin*sigh*  Add the 00, save, "dialplan reload", dial the number.
02:52.25jc319http://pastebin.com/KtUnGaL3
02:52.32jc319I did, the same
02:53.49WIMPy00075...? Surely not.
02:54.34jc319uhmm you're right
02:54.54WIMPyAnd I doubt you only want to call 11-digit numbers.
02:55.05jc319well that was the starting point, and I'm still there
02:56.37jc319OK I found a dialplan here http://forum.voxilla.com/asterisk-support-forum/help-uk-pstn-dial-plan-28903.html
02:56.53jc319The following covers everything (as far as I've used it for the last 2 yrs) in the UK:  0[1278]XXXXXXXXX  [2-8]XXXXX
02:57.30jc319it does not have special services such as 15x (BT repairs etc) but I will think about it later
02:58.21WIMPyI thought UK numbers can have different lengths?
02:59.38WIMPyUnless you don't want to be able to dial anything, you should probably only use _X. for outbound calls.
03:00.18WIMPyAnd make sure your internal extensions are included first.
03:04.22jc319WIMPy: How do I include them, by putting [internal] group above?
03:04.23WIMPyanumorayo: Still listening to your new personal talking clock?
03:06.04anumorayolol  i can't get it too work the command i mean
03:06.35WIMPyCan't find your dialplan any more. Make sure that you include your internal context befor your outgoing context in the phones context.
03:06.42WIMPyanumorayo: What happens?
03:07.16WIMPyjc319: Or maybe better don't use internal extensions that overlap with real phone numbers.
03:09.49p3nguinOr since you know there aren't going to be any phone numbers less than, say, 5 digits, you could use _XXXX. as the pattern.
03:10.15p3nguinThat way your 3-digit internals won't match.
03:11.46WIMPyYou will usually have at least one 3-digit number.
03:12.11p3nguin3-digit phone numbers?  On the real phone network?
03:12.24WIMPyEmergency services.
03:12.32p3nguinI 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${EXTEN}.
03:12.38p3nguinThey have 911 in the UK?
03:12.44WIMPyAnd yes, there are other 3-digit numbers in some places.
03:13.13WIMPy911, 999, 112, 110
03:13.40p3nguinI've never been in the UK to use their phones, so I have no idea what numbers they have.
03:16.48anumorayoyeah it 999
03:18.05anumorayoWIMPy: my channel is  zap/g0
03:18.24WIMPyanumorayo: zap? Thats ancient.
03:18.25jc319I keep getting thin >>   == Everyone is busy/congested at this time (1:1/0/0)
03:18.31jc319can't make any calls now, any ideas why?
03:18.39*** join/#asterisk l2trace99 (~bender@rrcs-71-43-104-238.se.biz.rr.com)
03:19.05jc319http://pastebin.com/A2EWFK64
03:19.06WIMPyanumorayo: Err, no. You did have dahdi, wich replaced zaptel, so it must be dahdi.
03:19.16jc319latest extensions & full busy msg
03:19.30jc319this*
03:20.18WIMPyjc319: That number doesn't make any sense, either. Have you tried the full unfomatted version, i.e. 004475...?
03:20.55anumorayoyes it is
03:21.03anumorayosorry
03:21.37jc319WIMPy: If I call '00447599408968' it adds additional zero so it becomes '000447599408968' and does not work. I think I'll remove that 0$
03:21.56anumorayobut it refer in my gui  has trunk zap/g0
03:22.04WIMPyjc319: Yes, that should work.
03:23.06anumorayoi am using trixbox
03:23.08WIMPyanumorayo: That doesn't sound right, but for help on the gui, you should check it's channel. We're hardli familiar with those.
03:23.49jc319OK I think this one is better, http://pastebin.com/6TvfFZev
03:24.11jc319However I still have this busy signal. Why did it start constantly giving busy now..
03:24.57WIMPyanumorayo: I'd still recommend a current verison of Asterisk, especially for ISDN connectivity.
03:29.07anumorayoman
03:30.18anumorayothis is the most current version of trixbox
03:31.32WIMPyIt seems to be rather outdated.
03:32.34anumorayobased on what
03:33.49WIMPyThe fact that you don't have .channel originate' and it uses zap, which should have gone >2 years ago.
03:34.13WIMPyActually it was replaced 3 years ago.
03:34.53p3nguin(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${EXTEN}.
03:35.25WIMPyYou're missing a CC then.
03:35.42WIMPyMaybe they accept +...
03:36.31p3nguinjc319: I'm ringing your number now.
03:36.36*** join/#asterisk sourcode (~code@ppp-58-8-87-34.revip2.asianet.co.th)
03:37.00p3nguinIf you answered, there was no audio.
03:37.02jc319oups was it you
03:37.13jc319sorry I picked up because I was calling that number too
03:37.40p3nguinI dialed 00447599408968 from my phone, which sends exactly 00447599408968 to voipms.
03:37.57jc319with 0$ or $
03:38.05p3nguinIf you are starting with 04, then you need to use 0${EXTEN}
03:38.17p3nguinBecause you must send 00447599408968
03:38.37p3nguinANd 0${EXTEN} is 0 0447599408968
03:39.00p3nguinDo all of your UK numbers that you dial start with 0?
03:39.37jc319normal  number yes, but what about 999 etc.
03:39.50jc319p3nguin can you have a look at this please - http://pastebin.com/2NZRSJR4
03:39.54p3nguinvoipms has no idea what 999 is.
03:40.08jc319I still get the busy signal all the time, is there something wrong in there
03:41.19p3nguinyes
03:42.04p3nguinIf all normal numbers you dial start with 0, you MUST use 0${EXTEN} in the dial command.
03:42.09p3nguinUntil you do, it's not going to work.
03:42.26jc319yes but I mean I'm typing 0044... full number in int. format
03:42.29jc319it still does not work
03:42.34p3nguinShow me.
03:42.40p3nguincore set verbose 4
03:42.42p3nguinmake the call
03:42.47p3nguinpaste the output.
03:43.13WIMPyIf you dial a national number you need to send it as 0044${EXTEN:1}.
03:43.47p3nguinI expect that he has to use the same rules in voipms that I have to use.  They do not accept THAT format.
03:44.07jc319At the moment without doing ANY patterns I just want to ring using full number 00 - country code - full national number
03:44.10p3nguinThey will accept 00 and 011 as prefixes, but 044 is not going to work.
03:44.11jc319http://pastebin.com/yJ2F3u6n
03:45.00jc319what does 'busy here' mean? Got SIP response 486 "Busy Here" back from 78.129.153.20:5060
03:46.04p3nguinIt means busy.  Show me your voipms_home peer from sip.conf.
03:46.08WIMPyMaybe they bloced your account for too many failed call attempts? I heard some providers do such things.
03:46.29p3nguinNah, I've failed plenty of calls and had no issues.
03:46.42p3nguinI do more than 75% of my calls through voipms.
03:47.07jc319http://pastebin.com/sDaABhpB
03:47.11WIMPySo you have successful ones in between.
03:47.42jc319but I call more than normal, yesterday at 6 AM I had 70+ calls on my mobile (test calls) maybe they're blocking me
03:48.03p3nguinWhat happens if you take out that "fromuser" value?  Just comment it out.
03:48.47p3nguinsave, sip reload, make a call.
03:48.48jc319now they are not busy. this time the 'circuit' is busy.
03:48.49jc319[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'
03:48.49jc319<PROTECTED>
03:49.00p3nguinhmm
03:49.11p3nguinI don't use fromuser for my voipms peer.
03:50.53jc319commented it out, the same. commented out defaultuser= too (just to test) still the same
03:51.28jc319[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'
03:51.28jc319<PROTECTED>
03:51.51p3nguinYou'll need defaultuser because that is what it authenticates against.
03:52.14p3nguinYou recall that setting about match auth user that you added earlier?  Take it out if it's still there.
03:53.11jc319;match_auth_username=yes
03:53.14*** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net)
03:53.16jc319was active, commented out now, testing...
03:53.44jc319the same...
03:55.53p3nguin*shrug*  It sure worked for me.
03:56.16jc319tomorrow I'll start from blank configs
03:56.26jc319thanks for all the help
03:56.31p3nguinThat's the spirit!
03:56.38p3nguinStart from the ground up.
03:56.52jc319I'm on config 13 now (ground up)
03:56.55jc3198 works :D
03:56.59jc319I'll copy that one
03:57.06jc319I think this multi-line Cisco thing is not helping
03:57.22jc319it's not working anyway. I'll use single peer definition for cisco devices
03:57.51jc319It'll be a shame if I can't use multi line buttons without SCCP.
03:58.12jc319I had big dreams about 'em...
03:58.47WIMPyAnything is better than SIP.
03:58.55p3nguinWhen I used SIP, I used two lines: one for regular stuff and one specifically for callcentric.
03:59.39p3nguinTwo completely seperate and unrelated authnames.
04:05.20jc319Before I go to bed, out of curiosity, I copied back the config which worked before
04:05.36jc319It still gives the same msg "Got SIP response 486 "Busy Here" back from 78.129.153.20:5060"
04:05.46jc319it must be an ITSP side issue
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04:58.50*** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein)
05:02.43p3nguinI doubt it's an ITSP problem, since I can call your 044 number and it just works.
05:02.50p3nguinBut what do I know?
05:03.58*** join/#asterisk rhollan_ (~rhollan@173-10-78-121-BusName-Washington.hfc.comcastbusiness.net)
05:04.57rhollan_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.
05:11.00*** join/#asterisk golikwid|mac (~chrislees@207.30.30.130)
05:11.24rhollan_Anyone her have Asterisk process inbound SMS?
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05:31.18*** join/#asterisk golikwid|mac (~chrislees@207.30.30.130)
05:35.22fulcanrhollan_ 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.  :)
05:40.01p3nguin(0018.11)  -!- rhollan ~rhollan@173-10-78-121-BusName-Washington.hfc.comcastbusiness.net has quit [Quit: Leaving]
05:42.16fulcanif 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?
05:43.45p3nguinexten => _[*#0-9].,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}-${EXTEN}.WAV,a);
05:44.04p3nguinexten => _[*#0-9].,n,Goto(some-other-context,${EXTEN},1);
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05:44.31p3nguinThat's the basic concept.
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05:49.12fulcanp3nguin 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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07:02.31p3nguin~thebook
07:02.31infobotAsterisk: 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
07:02.36p3nguinfulcan: ^^^
07:03.53fulcanp3nguin awesome! thank you.
07:04.19fulcanp3nguin question for you real quick. I thought 's' was a catch all?
07:04.29p3nguinIt's not.
07:04.34p3nguins only matches s.
07:04.49p3nguinliterally the 's' extension.
07:05.17p3nguins is most often used where there is no phone number.
07:05.49p3nguinmacros, IVRs, analog lines where you have a real hook
07:08.22fulcanp3nguin 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....  :)
07:09.42p3nguinDefine your phone number... or send calls to the s extension.
07:09.52p3nguinI wouldn't want calls going to the s extension, so I would define my phone numbers.
07:11.21fulcanthe 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)?
07:12.07fulcanI am 100% viop with no hardwire.
07:12.12fulcanvoip
07:12.14fulcan:P
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07:20.00fulcanp3nguin 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?
07:22.24p3nguinI certainly wouldn't.  default really needs to be empty.
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07:23.21fulcanp3nguin if I name the context the same as the number, she won't catch the call?
07:24.41fulcanrequest '7185696588@default' does not exist
07:24.44p3nguinIt also wouldn't make any sense.
07:25.20p3nguinMake a peer definition in sip.conf according to where that call comes from, then assign an APPROPRIATE context.
07:25.43p3nguinAnd I'm going to bed, so I won't answer any more questions for at least eight hours.
07:26.04fulcanno worries my friend. Enjoy your rest.
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12:32.40lesouvage.
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12:33.49Dovidcan anyone help me here with a core dump ?
12:36.07WIMPyDovid: gdb can
12:36.14WIMPy~collectdebug
12:36.14infobotcollectdebug 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
12:37.56DovidWIMPy: I have the back trace. I can't seem to figure out where the issue is
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12:38.14irrootpost it in pastebin
12:38.42irrootwhat is nature of problem and version ??
12:39.08Dovidhttp://pbx.dovid.net/core_dump_06-12-2011-at-0820.log
12:39.24Dovidi upgraded from 1.6.1.20 to 1.6.2.18
12:39.34Dovidand i got that with in an hour
12:39.47Dovidirroot: btw you are my hero for working on T.38
12:40.13irrootlol its not there for 1.6 but have it in production on 1.8 thx
12:40.33Dovidirroot: I know this is for another system. i am trying it on my ss7 box running 1.8
12:40.47irrootcool
12:40.53Dovidis there any tutorial on core dumps so i can learn how to figure out the issue on my own ?
12:41.17irrootlooks like a call picup gone wrong
12:42.30Dovidi am actually upgrading because of the T.38 vulnerabilituy
12:42.44Dovidirroot: gone wronge == ?
12:42.57Dovidbad typing day
12:43.53Dovideh. 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
12:45.36irrootlocal_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
12:46.01Dovidirroot: What do you think ? should I open a ticket about the core dump ? I dont want to get shouted at ;)
12:46.38irroothehe rebuild with dont optomize and see if it happens again
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12:51.07Dovidbakc
12:51.09Dovidback*
12:52.09irrootdovid rebuild with dont optomise the thing is 1.6 is nolonger supported officially
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12:55.41Dovidwhat does the optimize do ?
12:55.53Dovidwhen ever i go to 1.6.2 it craps out
12:56.01Dovidi though 1.6.2.X is stills upported ?
12:56.36Dovidhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
12:56.41Dovidi guess it just passed :(
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13:02.26irrootmake menuconfig
13:02.31irrootthere are some options
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14:04.27fulcan.
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14:12.01fulcanI 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
14:15.41cneb3000-laptophmm, fulcan what version of asterisk are you using?
14:16.32irrootcouple 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'
14:16.52irrootyou should not use | in 1.8 if you are on 1.8
14:16.59cneb3000-laptop^^^ beat me to it
14:17.32fulcanoh geeze, I forgot where to find the asterisk version @. I used gentoo emerge
14:17.33cneb3000-laptopwav|${CALLERID(name)}  should be more like --->  wav,${CALLERID(name)}
14:18.08fulcanI took the pipe out (I actually opened my eyes) and the error is gone and looking for the recorded file now...
14:18.26cneb3000-laptophehe
14:18.29cneb3000-laptoplook within
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14:20.58fulcanAsterisk 1.6.2.17.3 built by root @ li345-191 on a i686 running Linux on 2011-06-12 06:12:19 UTC
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14:21.54irrootuse the source young padowin
14:22.19fulcanwhere does asterisk store the recorded files?
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14:23.01cneb3000-laptopfulcan: if you look in your error it's trying to store the file in /var/spool/asterisk/monitor/. so it's probably in there
14:23.57fulcancneb3000-laptop two files instead of a merged file?
14:24.15irrootsoxmix or sox
14:24.23irrootuse the m option to monitor
14:24.38irrootit will sox mix em when dons
14:24.42irrootdone
14:25.40irroothint you can have a wrapper /usr/bin/soxmix to call sox -m and do any post processing here sign the file perhaps
14:27.04fulcanirroot where do you put the -m switch, in the begining right after extn -> or at the end?
14:27.55irrootMonitor([file_format[:urlbase]][,fname_base[,options]])
14:28.14irrootcore show application monitor
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14:30.27cneb3000-laptopso in your case fulcan it will be something like exten "=> _.*,1,Monitor(wav,${CALLERID(name)},m")
14:30.55cneb3000-laptopsorry, "exten => _.*,1,Monitor(wav,${CALLERID(name)},m)"
14:30.58cneb3000-laptopdrunk fingers
14:31.30cneb3000-laptopbeing playing with perl a lot recently. keep nearly enging every sentance with ;
14:31.33cneb3000-laptop¬_¬
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14:34.21fulcancneb3000-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  ?
14:35.09fulcancneb3000-laptop I have the ,m switch set exactly like that too.
14:35.17cneb3000-laptopdid you reload asterisk?
14:35.42fulcanyup, dialplan reload. let me do a core reload
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14:37.26cneb3000-laptopif that still doesn't work can you pastebin the dialplan again? just to double check. and also pastebin any errors if there are any.
14:39.58fulcanhttp://pastebin.com/y8dyi7Y6 no errors, but same two files and still no index.  :/
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14:41.26cneb3000-laptophmm, i think the joining is acttually done by a third party app as irroot said, called sox
14:41.30cneb3000-laptopmaybe you dont have it installed?
14:42.14fulcancneb3000-laptop emerge sox 'right now  :)'
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14:43.38cneb3000-laptophaha, so it wasnt there? :)
14:46.04fulcancneb3000-laptop nope, I installed her and now I have 1 file!  :)
14:46.11cneb3000-laptopwoohoo
14:46.47fulcancneb3000-laptop only 1 though, after 3 to 4 test calls though.   :(
14:47.13cneb3000-laptopso it stopped working?
14:47.17fulcancneb3000-laptop how do I get her to index file names?
14:47.22fulcanI works!
14:47.27cneb3000-laptopwhat do you mean index file names?
14:47.30fulcandon't get me wrong!
14:47.58cneb3000-laptopyou mean like... 2010-01-01call1.wav 2010-01-1call2.wav?
14:48.01fulcancneb3000-laptop it overwrites the original instead of creating a new one.
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14:48.07fulcanyup!  :)
14:49.42fulcancan this be done??  exten => _.*,1,Monitor(wav,${CALLERID(name)},m)  would change to   exten => _.*,1,Monitor(wav,${CALLERID(name)(systemdate)},m)?
14:49.54cneb3000-laptopit's only doing that because in the filename bit of the monitor command you're saying '$callerid(name)
14:50.06cneb3000-laptop^^^ yes, althouh i'm not sure if systemdate is correct
14:50.12cneb3000-laptop(because i just dont know)
14:50.16cneb3000-laptopbut thats the right idea
14:50.24cneb3000-laptopreason it overwrites the current file, is because its the same file name
14:50.38fulcan<PROTECTED>
14:51.23cneb3000-laptopfulcan look at this...
14:51.30cneb3000-laptophttp://www.voip-info.org/wiki/view/Asterisk+variables
14:51.47cneb3000-laptopnot sure if it's up to date. but it suggests ${DATETIME} will do what you're looking for
14:51.52fulcanon it like white on rice!   :)
14:52.07cneb3000-laptopactually, it says use ${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)})
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14:53.43cneb3000-laptop^^^ confirmed at https://wiki.asterisk.org/wiki/display/AST/Asterisk+standard+channel+variables
14:53.55cneb3000-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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14:54.15cneb3000-laptoplet me know how it goes, i'll be doing something similar myself in the future :)
14:55.13irrootexten => s,n,SET(CLOGID=${IF(${EXISTS(${DDIUNIQUEID})}?${DDIUNIQUEID}:${CDR(linkedid)})})
14:55.14irrootexten => s,n,SET(CALLLEG=${ODBC_LOGCOUNT(${CLOGID})})
14:55.16irrootexten => s,n,SET(ODBC_LOG(${CLOGID},${CALLLEG},${CHANNEL})=${ARG3})
14:55.19irrootexten => s,n,SET(CALLDATE=${IF(${EXISTS(${CALLDATE})}?${CALLDATE}:${STRFTIME(,,%Y-%m-%d)})})
14:55.20irrootexten => s,n,SET(MONITOR_EXEC_ARGS=${ODBC_RTDB(Setup,RecOpt)})
14:55.23irrootexten => s,n,Monitor(wav49,/var/spool/asterisk/monitor/${CALLDATE}/${ARG3}/${CLOGID}-${CALLLEG},mb)
14:55.58irrootthats what i do it tracks trasfers and stores it in directories under monitor by year/exten
14:58.50cneb3000-laptopahh thanks irroot
15:00.10irrootpleasure makes it much faster looking up files under multiple paths
15:01.57irrootwhen there couple 100 / 1000 files it can cause a massive knock on quality and i/o load
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15:02.36cneb3000-laptopknods
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15:02.47cneb3000-laptophmm
15:02.57cneb3000-laptopmind if i ask which variable is tracking the transfers? sorry..
15:03.16irrootcallleg
15:03.17fulcanthis "exten => _.*,1,Monitor(wav,${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????
15:03.44irrootit gets stored in odbc and then added to each leg of the call
15:04.00cneb3000-laptop^ ah i see. nice!
15:04.46irroot(name),${STRFTIME(${E
15:04.46fulcan'MRALL.wav' instaead of SUMRALL.wav
15:04.51irrootcant use a ,
15:04.54irrootuse a -
15:05.09fulcanirroot after name?
15:05.18irrootyip
15:05.34irrootalso missin g }
15:06.02fulcang where?
15:06.11irroot${CALLERID(name)}-${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)},m
15:08.06irrootits possible maybe i have spent too much time in the dialplan
15:09.01fulcanirroot   when I use 'exten => _.*,1,Monitor(wav,${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
15:09.40irroot${CALLERID(name)}-${STRFTIME($,,%d%m%Y-%H:%M:%S)},m
15:09.43irroot${CALLERID(name)}-${STRFTIME(,,%d%m%Y-%H:%M:%S)},m
15:09.55irrooti find the time function a little tricky
15:10.29irrootepoch is default
15:11.19fulcanfirst or second one my friend?
15:11.53irrootsecond
15:12.03cneb3000-laptopirroot: i left a job after being there for 3 years. someone wrote  "BYE sip:[myphonenumber]@[my home address]:5060" on a leaving card
15:12.04irrootfirst has a extra $
15:12.07cneb3000-laptopyou think you have it bad?
15:12.23cneb3000-laptopi thought it was funny
15:12.25irrootlol
15:15.50*** join/#asterisk davlefou (~david@41.227.20.21)
15:16.47fulcanwow 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?????
15:18.36irroot${STRFTIME(,,%Y-%m-%d)}
15:19.02irrootstraight out my DP that only does YYYY-MM-DD
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15:21.02fulcanirroot same filename produces SUMRALL.wav with no date and this is after a core restart.....?    :(
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15:22.43irrootdialplan reload ??
15:23.11fulcanirroot  core restart now
15:25.50irrootor -${CDR(uniqueid)} instead of strftime
15:26.38kaldemarfulcan: what version are you using?
15:27.23kaldemaroh, it was answered already.
15:27.25fulcanAsterisk 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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15:33.19fulcanOk, GREAT news. almost there this "exten => _.*,1,Monitor(wav,-${STRFTIME(,,%Y-%m-%d)}${CALLERID(name)},m)"  produces -> -2011-06-12SUMRALL.wav     :)
15:37.44fulcanand 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...  :)
15:38.22fulcanI have no clue why the date breaks if you put callerid first.
15:38.53fulcanthere is something up with that, but it's not hurting me at all.
15:38.54irrootcool
15:41.21cneb3000-laptopgood to see it working
15:41.36cneb3000-laptopnext up - getting it mixdown the 2 files into stereo? ;)
15:46.54fulcancneb3000-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?
15:47.16cneb3000-laptopso when you listen to .wav file it's silent?
15:47.25fulcanyup
15:47.36cneb3000-laptopis it a 0kb file, can you tell?
15:47.42fulcanall new files produced are the same 44 byte files.
15:47.59fulcanall 44 bytes
15:48.20fulcanbeautiful naming scheme though....  :)
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15:50.06cneb3000-laptophmmm
15:50.11cneb3000-laptopfulcan, turn off the m option
15:50.20cneb3000-laptopthen listen to the two files. and see if you can hear the sound then?
15:50.44cneb3000-laptopthat 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
15:51.15irroot/usr/bin/sox -m <in> <out> <mix>
15:51.27irrootwill mix once tested
15:52.21cneb3000-laptopgood luck fulcan - i'm off to play mario kart with the missus!
15:53.15fulcanirroot now I have 2 44 byte files.....  :/
15:54.01irrootthat is the header only i suspect
15:54.27fulcanirroot yup. at least it is getting that.
15:55.28fulcanirroot I don't see it being permissions at this point. a vox config thing maybe?
15:55.51irrootsox runs when m is used
15:55.52fulcancneb3000-laptop thank you my friend!!!!
15:56.43fulcanirroot shoot that therory. possibly canreinvite=no?
15:56.53fulcanin sip.conf
15:57.36fulcanthat effects the rtp stream, I know that much. but typically only for remote devices.
15:58.01irrootdo a rtp debug and see if it goes through asterisk
15:59.17fulcan<PROTECTED>
15:59.19fulcanSent RTP packet to      63.211.239.14:24332 (type 00, seq 026871, ts 058720, len 000160)
16:00.05fulcanthere went that theory.
16:00.16fulcanI'm batting zero today.
16:02.10fulcanlet 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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16:03.50p3nguinIf you haven't already figured it out, use MixMonitor() if you want a combined file without having to mix the channels manually.
16:06.00fulcanHere is an entire session.  Just a thought, would the demo playing lock up the call recording?    http://pastebin.com/uX2sYGaQ
16:07.56fulcanp3nguin 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.
16:09.11irrootok
16:09.24p3nguinI'd still change it to MixMonitor() anyway.
16:09.29irrootput monitor in the begining
16:09.43fulcanirroot  ?
16:09.46irrootnot in the hangup
16:10.06irrootg [h@7185696288:1] Monitor("SIP/7185696288-00000008", "wav
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16:10.47fulcanirroot I have monitor in the very begining. before wait and even before answer.
16:12.08fulcanthis is my entire extensions.conf   http://pastebin.com/0cK3qp8C   I like them clean.   :)
16:12.53fulcanirroot I am not seeing what you are seeing my friend.  :(
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16:14.19p3nguinBy using the pattern _. to match the extensions rather than what I suggested last night, your Monitor() runs in places it shouldn't.
16:14.40p3nguinIn the case of what irroot said, in the h extension.
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16:15.24p3nguin(0043.44) <p3nguin> exten => _[*#0-9].,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}-${EXTEN}.WAV,a);
16:15.27p3nguin(0044.04) <p3nguin> exten => _[*#0-9].,n,Goto(some-other-context,${EXTEN},1);
16:15.52p3nguinThis is the concept I provided.  Which works.  I recommend using it because it works correctly.
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16:16.52fulcanp3nguin 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.
16:18.42p3nguinUsing 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.
16:19.11p3nguinThen your peers would be assigned context=phones-r rather than just context=phones.
16:19.53p3nguinAssuming you're recording all outbound calls from phones, this seems to have good logic behind it.
16:20.42fulcanp3nguin making it real easy. I am recording absolutely everything that passes through the server. This server has one single job.  :)
16:21.10irroothave a flag in astdb for what to record
16:21.25irrootbased on callerid/channel/....
16:21.32p3nguinSince every call that passes through it should have a phone number, an extension pattern of _. is a HORRIBLE choice.
16:22.22p3nguin_. matches EVERYTHING, including extensions that you should not be matching for recording, such as extensions s, h, i, and t.
16:22.33WIMPyPhone NUMBERS are so last century.
16:22.50irrootlol
16:22.53p3nguinI know, we should always match all extensions, right?
16:22.57fulcanp3nguin 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?
16:23.09p3nguinScroll up.
16:24.16fulcanp3nguin add just that one line instead? your answer appeared more conceptual than 'big bird cookie monster' like what I need.
16:25.03p3nguinI 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.
16:25.39p3nguinThe two lines I gave you are what you need, but you also need more than those two lonely lines to have a working system.
16:25.47fulcanp3nguin one sec.
16:26.35irrootwimpy ipv6 or bust !!!
16:26.37p3nguinThis extension will match any call to phone numbers starting with *, #, or a number, followed by one or more additional characters.
16:26.59WIMPyirroot: Not available here :-(
16:27.17p3nguinIt would match *1.  It would match #611.  It would match 18004444444.
16:27.42irrootyeah i can hook up some via the providers but have not got the infrastructure ATM
16:30.58p3nguinOh, 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.
16:33.52fulcanhere 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.
16:40.13p3nguinWell, you haven't used "identical context names."
16:40.58p3nguinBut you did erroneously duplicate you sip peer names in sip.conf.  You should fix that.
16:42.36p3nguinAre you using dtmfmode inband because you're trying to record tones?
16:42.42fulcanp3nguin the doc I was looking at said teliax wanted 2, one for inbound and another for outbound.
16:42.56p3nguinThey're wrong.
16:43.14p3nguinI've never met an ITSP yet that knows how to correctly configure an end user Asterisk system.
16:43.43fulcanp3nguin 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?
16:43.56p3nguinBut let's pretend for a second that they're right; you still would never duplicate a peer name.
16:44.12irrootwaves at p3nguin lol
16:44.21fulcanp3nguin cheers
16:46.22p3nguinI find it pretty silly (and somewhat sad) that persons or companies operating is that capacity don't know how to configure Asterisk well.
16:47.28irrootp3nguin indeed almost opposed to asterisk sometimes
16:48.24fulcanasterisk is grand daddy. first user auth system built on SER.
16:48.29fulcan:)
16:48.41p3nguinWhat kind of operation are you running where you are in a position to oppose Asterisk for end users?
16:49.18irrootno refering to a local ITSP that tells customer asterisk is bad and dont support it
16:49.33WIMPyThey just don't want to be bothered with configuration issues.
16:49.48irrootlol nope they agents for audiocodes/patton
16:50.11p3nguinIf 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.
16:51.32p3nguinIt 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.
16:51.58p3nguinSomeone who doesn't need to read a book to figure it out should have it done in five or so minutes.
16:52.44p3nguinIf 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.
16:54.15irrootgoing to set up a ITSP shortly for asterisk customer as i haz a gui shiped all it will take is un/pw
16:54.49irrootand ill support with sample as you said above a copy paste sip/iax conf
16:54.57irrootand dialplan example
16:55.08fulcanThis 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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16:57.22irrootno the extension is the name name the context is default
16:57.49irrootneed exten wuzamarine,1,Goto(.....)
16:57.57irrootin the [default] section
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16:58.34fulcan<irroot> need exten wuzamarine,1,Goto(.....)???
16:59.31fulcanexplain the ..... a little better please.
16:59.52irroot[default]
16:59.54irrootexten => wuzamarine,1,Goto(context,exten,1)
17:00.01irrootthe context is the [.....] section
17:00.33irrootthe exten is as above a point in the context the priority is the 1 ....
17:00.45irrootthis is a important concept to grasp
17:00.56irroot~thebook
17:00.56infobotAsterisk: 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
17:01.31cneb3000-AFK^^^ the dial plan basics bit in that IS really helpful
17:01.37cneb3000-AFKand about a 15 minute read
17:01.41cneb3000-AFKgoes back under his rock
17:01.59p3nguinerr... apply good, testing, working values to create good, tested, working samples, rather.
17:02.08p3nguins/testing/tested/
17:08.20fulcancneb3000-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.
17:10.03fulcanBuilt 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.
17:10.39*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
17:11.00fulcanI think that was the last time I really tore into asterisk.
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17:40.50fulcanshe still barks at me about the username not being in my default   [default]  ->  exten => myPhone1,1,GotoIF($["${CALLERID(num)}" = "wuzamarine"]?dial1))      and   [myPhone1] blah...
17:42.23irrootexten => wuzamarine,1,
17:42.36irrootits sending it to the above
17:42.40irrootnot myphon1,1
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17:56.51fulcanhow do you assign a static variable in the context?
17:57.47irrootSET(MYVAR=<VALUE>)
18:04.50p3nguinIf it's not a global variable, that has to be in an extension rather than just a context.
18:05.46p3nguinIf it's a global, then it's simply MyVar=value and it has to go under [globals].
18:06.39fulcanhttp://pastebin.com/8VcnrQk2   :(
18:06.39p3nguinYou can also set variables on the peers and then reference them in dial plan.
18:06.57p3nguinline 18 is wrong.
18:07.20dr0ckor GLOBAL function
18:07.32p3nguinline 30 doesn't have a priority.
18:07.47p3nguinline 40 doesn't have a priority.
18:08.04p3nguinAnd yes I see that they are commented out, but that shows that you intend to use them at some point.
18:08.15fulcanp3nguin if I remove the comment is errors.
18:08.25p3nguinOf course it does.  It's wrong.
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18:09.13p3nguinYou also don't need line 31 and 41.  BackGround() provides the answer when it gets to line 34 and 44.
18:10.38fulcanp3nguin glad to see you see its wrong, what would you suggest to make it right?
18:11.48fulcanthere is no variable in peers that also appears in dialplan.
18:12.41fulcanline 34 and 44, I could care about, not even close to that yet.
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18:17.27fulcan<PROTECTED>
18:17.48p3nguinhttp://pastebin.com/v7CBZbdm
18:18.32fulcanhttp://pastebin.com/tzMcynSH
18:19.21p3nguinAnd I'll rebut again with http://pastebin.com/v7CBZbdm
18:19.29p3nguinI rewrote your dial plan for you.
18:19.34p3nguincorrectly
18:20.26ChannelZwill you come vacuum my house?
18:20.53p3nguinDepends on if you have a Dyson or not.
18:21.14p3nguinThose things really suck.
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18:21.55jc319hello
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18:23.51ChannelZActually I do, and it's awesome
18:24.02fulcan<PROTECTED>
18:24.39fulcanp3nguin That is using yours word for word my friend.
18:25.04p3nguinYep.  Now go fix your sip.conf.
18:25.07ChannelZwhich doesn't have an extension named 'wuzamarine' in default just as it says.
18:25.35p3nguindefault doesn't deserve to have anything in it.  He needs to go fix is peer entry so it matches the device making the call.
18:25.50p3nguinIt's a KNOWN device, not an anonymous device.
18:26.26ChannelZJust pointing out that the error is pretty straightforward.
18:26.33fulcanhttp://pastebin.com/UeFrrtTR   There is NOTHING in sip.conf with wuzamarine in it except for my registration string. thats all.
18:26.33p3nguinAnd in addition to that, no where in your dial plan did you create an extension called 'wuzamarine'.
18:26.58p3nguinSo you can't expect that a call TO wuzamarine will ever succeed.
18:27.15p3nguinMost people can't dial words on their keypads, anyway.
18:27.55fulcanp3nguin in case you didn't notice, I am trying to eliminate that word by even trying to set it to a variable.
18:28.08p3nguinI have no idea what that statement even means.
18:28.55p3nguinUntil you fix your sip.conf to match peers, your dial plan will never work.
18:28.58p3nguinSo fix that first.
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18:30.40fulcanwhere do you see any reference to 'wazamarine' in this sip.conf?  http://pastebin.com/
18:31.05p3nguinWithout even looking, I can tell you that it does not matter.
18:31.20fulcanhttp://pastebin.com/QfqBYeHT
18:31.25p3nguinDoesn't matter.
18:31.43p3nguinYou have to match the peer by creating a correct sip peer definition.
18:31.55fulcanYour telling to "fix my variable that am using in sip.conf that has wasamarine".
18:31.59fulcancorrect?
18:31.59p3nguinThen assign the context that you want calls from that device to go into.
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18:32.16p3nguinNo, I did not say that.
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18:32.26ChannelZis 7185696288 a local device?  A softphone or something?
18:32.40fulcanthen, where in the world is 'wazamarine' coming from?
18:32.49p3nguinhost=xxx.teliax.net
18:33.08p3nguinThe refusal to fix the fucking sip.conf is not gaining you any ground.
18:33.52ChannelZfulcan you actually answered your own question earlier: "There is NOTHING in sip.conf with wuzamarine in it except for my registration string. thats all."
18:34.16p3nguinHere.  I've started fixing it for you.  Now you just have to do the rest.  http://pastebin.com/1zezCFB9
18:34.50fulcanvery 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
18:35.03ChannelZYou have 7185696288 not set to go to any context which is why it's going to 'default'
18:35.23p3nguinAnd that's just one of the issues.
18:35.37ChannelZAnd 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.
18:36.16p3nguinIt needs at the very least a defaultuser and a context.
18:36.20p3nguinfor teliax
18:37.16p3nguinIt'll amuse me if the register statement end with /wuzamarine
18:37.27ChannelZplace yer bets
18:37.42p3nguinWhat kind of odds are you giving?
18:38.17ChannelZI'll give you a cookie.
18:38.18fulcanhandle_request_invite: Call from '' to extension 'wuzamarine' rejected because extension not found in context 'default'
18:38.38ChannelZHey, check it out.  Pretty much the same error
18:38.45fulcanyup!
18:39.26p3nguinIf 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.
18:40.47fulcanp3nguin I had it to the point that you could reach the recording but _. was, well you know.
18:41.48p3nguinYou 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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18:42.21p3nguinUntil you match the peer AND assign the correct context (not default), don't expect it to work.
18:42.35fulcanIt seems to be using my username as caller ID.   :/
18:43.01p3nguinThen you should consider using trustrpid=yes and sendrpid=yes in the peer entry.
18:45.07fulcanp3nguin added it to the teliax peer group but same error.
18:45.29p3nguinHow about that register statement?
18:45.45p3nguinI think you told me caller ID when you meant extension.
18:46.03fulcanp3nguin " Until you match the peer AND assign the correct context" I am using 100% what you sent me.
18:46.19p3nguin(1334.11) <p3nguin> Here.  I've started fixing it for you.  Now you just have to do the rest.
18:46.30p3nguinYou didn't do the rest.
18:46.58p3nguinAnd we're still waiting to see the register statement.
18:47.43p3nguinI'm starting to become bored with this.  Maybe I need to go make some phone calls or something.
18:48.05fulcanexplain this "peer AND assign the correct context" in more detail please.
18:48.22p3nguinThe peer definition.  In sip.conf.  Fix it.
18:49.18p3nguinThe last time I saw it, it didn't have context=anything-useful in it.
18:49.33p3nguinAs channelz pointed out, that's why the calls are going to the wrong (default) context.
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18:51.40fulcanp3nguin 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.
18:54.07p3nguinHere's an example of a good peer definition:  http://pastebin.com/fJgNLGLM
18:54.31p3nguinIt's for VoIP.ms, but Teliax shouldn't be TOO much different.
18:55.11jc319hi p3nguin, how are you?
18:55.19p3nguinirritated
18:55.27jc319:)
18:55.27jc319why
18:55.44jc319I just came in now, it can't be me, right?
18:55.56p3nguinI do most of the work for people trying to help, and they don't accept the work I provide.
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18:57.09jc319I for one appreciate your hard work
18:57.57irroot+1
18:57.59jc319I have a great game plan today: copy & paste voip.ms wiki example and test, when it doesn't work call them. almost error proof.
18:58.38p3nguinI have a better plan.
18:59.01p3nguinSince I use VoIP.ms and I can make calls to your UK DID, configure yours like mine is.
18:59.19p3nguinI'll even configure it FOR YOU just so we can move on.
19:00.11jc319Have to agree that's definitely the most efficient plan
19:00.33jc319Can I provide you extended information then? 2 DIDs now, 1 UK and 1 US
19:00.40fulcanhttp://pastebin.com/sSd1bC05
19:00.46p3nguinAbsolutely.
19:01.16jc319and 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
19:01.30p3nguinThe teliax peer still needs the username/defaultuser added.
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19:02.41p3nguinOh, 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.
19:03.27fulcanand this is with defaultuser=wuzamarine added   Call from '' to extension 'wuzamarine' rejected because extension not found in context 'default'.
19:03.57p3nguinNow set an appropriate context in the teliax peer.
19:04.21p3nguinSetting no context is not appropriate.  Setting it to default is not appropriate.
19:04.40p3nguinYou currently have no context set in it.
19:05.48p3nguinjc319: So... what's your next step?
19:07.59fulcana little farther, asterisk doesn't want to shake the username  ' Call from 'wuzamarine' to extension 'wuzamarine' rejected because extension not found in context 'decisions'"
19:08.32p3nguinYou're making progress.
19:09.31p3nguinNow to understand where extension 'wuzamarine' is coming into the equation.  Are you calling to a DID number that you bought from Teliax?
19:09.58fulcanyes
19:11.00p3nguinIf you added a line below [teliax] that says context=decisions, then we've proved that the peer is finally matching.
19:11.08fulcanthe 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.
19:11.55fulcanyes, matching the peer but NoOp is not getting the CID.
19:12.21fulcanit is getting the username instead.
19:12.41p3nguinCID and extension are two completely different bits of data.
19:12.59p3nguinCID is where the call comes FROM.  Extension is where the call is going TO.
19:13.03fulcanIt need the CID
19:13.11fulcanyup
19:13.27fulcanand it is coming from Teliax.
19:13.40p3nguinRight now, they are sending the call to extension 'wuzamarine'.  This is unrelated to CID.
19:14.03fulcanit is ABSOLUTELY not supposed to.
19:14.44fulcanthere is not now nor will there ever be and extension called 'wuzamarine'.
19:15.00p3nguinIf 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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19:15.37fulcanI have 2 phone lines coming off that 1 single pipe.
19:15.47fulcanthere will be 200
19:15.54p3nguinIn my opinion, calls should be sent to an extension matching the phone number that the caller dialed.
19:16.06p3nguinSo you're going to have 200 phone numbers with Teliax?
19:16.20fulcanthat's fine. just as long as it is not wuzamarine
19:16.23fulcanyup
19:17.02p3nguinUnless 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.
19:17.12p3nguinI know there are Teliax users here.
19:17.17fulcanso, having an extension called wuzamarine is useless.
19:17.33p3nguinIf you're going to have more than one phone number, it sure is.
19:18.02fulcanI know, I have been fighting this same battle all night long. Yours what simply a more educated approach to the same brick wall.
19:18.03p3nguinIt's either a Teliax configuration problem or your peer entry is still missing something.
19:23.45p3nguinSee if insecure=port or insecure=port,invite in the teliax peer make any different.
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19:43.21cneb3000bit of an out there question, but is telephony related..
19:43.41cneb3000i got hold an openmoke neo freerunner phone the other week. i have no use for it.
19:43.49cneb3000by any chance is anyone looking for one on the cheap?
19:44.07p3nguinAre they SIP phones?
19:45.52p3nguinJust looks like a GSM/GMRS phone to me.
19:45.59cneb3000hmm.. not really like that. it's kind of like a sandbox phone. so you would have to develop a SIP app on it.
19:46.08p3nguinerr, GSM/GPRS
19:46.23cneb3000^^^ it has wifi capability.
19:46.54p3nguinThat could be useful.  Maybe.  :)
19:47.26cneb3000well, itll be on ebay soon.. so!
19:47.34cneb3000if its of no use then dont vbother ;)
19:47.46cneb3000collecting dust for me
19:48.13p3nguinLet's just cut to the bottom line.  How much are you trying to get for it?
19:48.24cneb3000£100
19:51.41jc319p3nguin: 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
19:52.02jc319p3nguin: Are you using any sub-accounts at all? If so, do you call outbound with them?
19:52.44p3nguinI only use sub-accounts for my Asterisk systems and I do call outbound.  My main account is basically a placeholder.
19:53.56jc319Precisely what I had on mind.
19:54.16jc319So it is possible... I'll check the web panel to see if there's any limitation or anything
19:54.26jc319on this sub-acct..
19:55.16p3nguinThere is no setting to block calls on my subs.
20:10.26p3nguinjc319: So what's the next step?  I'm ready to move on.
20:15.32jc319Allow International Calls
20:15.32jc319value="0" >Yes - International Calls Enabled
20:15.32jc319value="1"  selected >No - International Calls Disabled
20:15.56jc319Now it's clear why the line has been BUSY
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20:16.15jc319why can they not say FORBIDDEN......
20:20.32jc319ok 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)
20:25.56p3nguinIf you're willing to spend a little money on it, it all can be done within a day.
20:26.28p3nguinFree support is, well, free.  And often it's slow.
20:28.25jc319Yes 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
20:29.12p3nguinTrixbox sucks.
20:29.47jc319I 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
20:30.09p3nguinI'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.
20:31.04jc319I 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.
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20:44.59jc319Is there a good free Windows SIP softphone with multiple lines? (also for MAC OS X?)
20:45.13p3nguinI like zoiper.
20:51.35[sr]jc319: 3CX
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20:58.13[sr]p3nguin: cool, zoiper has IAX!
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20:59.48p3nguinYep, zoiper classic is my favorite for Windows machines.
21:00.41[sr]i was searching for a IAX win client for a few weeks
21:01.11p3nguinI 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
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21:09.25[sr]p3nguin: thanks for the tip anyway :)
21:11.10jc319if 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
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21:30.14p3nguinjc319: 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.
21:32.24p3nguinThe only reason I use IAX2 is for bandwidth savings during multiple concurrent calls.  That's done with trunking, which SIP does not support.
21:39.47JerJerheh - until the first iax2 worm hits
21:40.52sxperthighly unlikely, but who knows ;)
21:41.24JerJeri / we already found one 'issue'
21:43.02JerJers/already/previously/
21:43.22WiretapWork_how far away is 1.8.5?
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21:46.56p3nguinAs they say, as soon as it's ready it will be released.
21:56.22WiretapWork_lol
21:56.23WiretapWork_nice timeline
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22:18.15p3nguinI've never understood why people expect developers to predict or guess when they'll be done working on something.
22:19.35p3nguinWhen it's done, it's done.  And not before that.
22:21.46WiretapWork_as a developer, I usually give some indication of an expected release date
22:21.55p3nguinThat's just silly.
22:22.06WiretapWork_while that goalpost can move regarding issues, it gives people some kind of idea when to prepare for a new release
22:22.34WiretapWork_meaning that they can have their systems ready to accept said release on time
22:22.37p3nguinUnless 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.
22:22.54WiretapWork_you can't predict it exactly
22:23.13p3nguinBecause you can't predict it, don't try to predict it.
22:23.28WiretapWork_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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22:53.27jc319Yeah I that SIP 'trunk' is not a single 'wide' channel or anything, they should call what they're selling (SIP trunk) a SIP uplink
22:54.22jc319p3nguin: 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...
22:55.28p3nguinMany 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.
22:55.56p3nguinBut I use IAX2 for the trunking even though I can't control VoIP.ms's side.  I don't have any problems with it.
22:57.18jc319They 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?
22:59.04p3nguinThere have been reports that IAX2 sometimes has problems between dissimilar Asterisk versions.  That might have something to do with it.
23:00.33p3nguinI 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.
23:01.43jc319OK 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...
23:02.08WiretapWork_IAX2 requires a reliable timing source
23:02.15WiretapWork_or so I've heard
23:03.16jc319Do 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)
23:03.30jc319hardware based timing source?
23:07.16p3nguinOne 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.
23:08.02WiretapWork_jc319, I config and ship 7912s to my agents (they work from home)
23:08.08p3nguinYou could use a 7900 series phone or you could use one of the newer 500 series phones.
23:08.11WiretapWork_the setup is trivial
23:08.31WiretapWork_the XML-config based 79xx dont' really appreciate being away from their TFTP server
23:08.35p3nguinOr you could go with a different brand completely: many people love the Polycoms.
23:08.55WiretapWork_(And require SIP-TCP which is not properly supported in 1.8.4)
23:09.40p3nguinProbably 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.
23:09.48jc319I think I have 8 or so 7910s or 7912? would both work? I'll check the version now
23:09.50WiretapWork_yep
23:10.09WiretapWork_the 7912, 7940 and 7960 are the three phones cisco will 'officially' allow you to use with non CUCM gear
23:10.12p3nguinThe boot-up process is lengthened significantly in the absense of the tftpd, but they eventually boot from "memory."
23:10.20WiretapWork_the 7910 is SCCP only if I'm not mistaken
23:10.45jc319model is 7905 would this work? (I also have 7940 in worst case I can send that if it'll work)
23:10.58WiretapWork_if I'm not mistaken the 7905 may work
23:11.08WiretapWork_you'll have to see if cisco have released a non-cucm firmware
23:11.12WiretapWork_or if its cucm-only
23:11.27WiretapWork_the non-cucm firmware is the one that is easiest to 'ship-n-go'
23:11.39jc319Cisco IP Phone 7905G/7912G (SIP) Release Notes for Firmware Release 1.2.0
23:11.49WiretapWork_sweet
23:11.54WiretapWork_looks like its just a 'dumb' 7912
23:12.15WiretapWork_I _Really_ need to finish my writeups
23:12.31WiretapWork_since my article on 79xx Unified phones is ranked 5th on google
23:12.35WiretapWork_but is incomplete
23:12.52jc319great, 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)
23:13.42jc319On what site is your article?
23:16.02WiretapWork_wiretap.net.nz
23:16.08WiretapWork_and no
23:16.11*** join/#asterisk golikwid|mac (~chrislees@207.30.30.130)
23:16.20WiretapWork_the 7912,7940 and 7960 can be configured on the phone screen or via web interface
23:16.22WiretapWork_TFTP is optional
23:16.34WiretapWork_I use them in places where a TFTP server is unavailable
23:17.31jc319Exactly, 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).
23:17.40WiretapWork_yep
23:17.50WiretapWork_power up the 7905, go into the menu and configure it
23:18.00WiretapWork_you'll need to **# to unlock the config
23:18.18WiretapWork_make sure your PBX has an external hostname
23:18.22WiretapWork_that matches internal
23:18.29WiretapWork_so that you can test the config before you ship
23:22.22jc319Will 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
23:23.38WiretapWork_depends on how many channels your local can support
23:23.48WiretapWork_I trunk 5 phones off an ADSL line
23:23.51WiretapWork_works fine
23:27.31*** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202)
23:27.56p3nguinAsynchronous Digital Subscriber Line line?
23:28.37WiretapWork_yes, but at least its not a SHVADSL line as someone once told me they had :P
23:29.59p3nguinI'll file that right below "Personal Identification Number number."
23:30.12p3nguinbut before "Automated Teller Machine machine."
23:30.35jc319I 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
23:30.54jc319don't*
23:30.57*** join/#asterisk micols (~0x2AA7F64@rlogin.dk)
23:32.25WiretapWork_jc319, learn to 'swap out'
23:32.55WiretapWork_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)
23:49.46linuxgeckoit 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?
23:50.37WiretapWork_forgotten to load any modules
23:52.52linuxgeckoWiretapWork_: sounds right,  but iv'e forgotten where to.how to do that in the config
23:54.50linuxgeckoWiretapWork_:  i'm VERY rusty,  and just need RTFM pointers. sorry for needing `stupid` help.
23:55.32linuxgeckos/and just need/and MAY just need/
23:56.09linuxgeckowow... :)   never seen someone actually make a bot that does that :)
23:57.06WiretapWork_on phone

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