| 00:01.22 | [TK]D-Fender | silvertip257: Your CLI output looks like you've been chopping stuff off. |
| 00:01.42 | teknoprep | [TK]D-Fender, do you like aastra phones ? |
| 00:01.49 | silvertip257 | i'm not chopping anything out - nothing omitted at all |
| 00:02.19 | *** join/#asterisk moy (n=moy@189.169.82.128) |
| 00:02.32 | [TK]D-Fender | teknoprep: meh |
| 00:02.40 | teknoprep | [TK]D-Fender, really that bad ? |
| 00:02.51 | [TK]D-Fender | teknoprep: only if I was desperate for massive presence |
| 00:03.02 | teknoprep | [TK]D-Fender, oh |
| 00:03.11 | teknoprep | [TK]D-Fender, yeah 180 lines on the 57i |
| 00:03.21 | [TK]D-Fender | teknoprep: I hated my 57i CT |
| 00:03.21 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 00:03.31 | teknoprep | [TK]D-Fender, but how is the call quality ... and why did you hate it ? |
| 00:04.15 | [TK]D-Fender | teknoprep: rubbery shit buttons, low angle of visibilty of the screen, shitty use of basic calling, lack of DECT independance, handset has NOT weight, tinny speakerphone, etc |
| 00:04.28 | [TK]D-Fender | teknoprep: Meant well, missed the mark |
| 00:04.37 | [TK]D-Fender | teknoprep: It got my hopes up |
| 00:04.50 | teknoprep | [TK]D-Fender, which phones do you prefer ? polycom ? |
| 00:05.03 | j0 | teknoprep: yes.. ;) |
| 00:05.08 | [TK]D-Fender | teknoprep: Yup |
| 00:05.21 | teknoprep | [TK]D-Fender, yeah polycom's are nice i just don't like there displays |
| 00:05.25 | jaytee | I love my 330 |
| 00:05.41 | teknoprep | [TK]D-Fender, i like how the 57i screen buttons are very customizable |
| 00:05.53 | [TK]D-Fender | teknoprep: Only downside to polycom is that the backlit ones are unreasonably more expensive |
| 00:05.55 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:06.16 | [TK]D-Fender | teknoprep: Indeed if nothing else Aastra use of dynamic soft-keys is unparalleled |
| 00:06.19 | teknoprep | [TK]D-Fender, yeah i have an ip650 for our office's receptionist with the Backlit Exp module |
| 00:06.38 | teknoprep | [TK]D-Fender, how is the call quality |
| 00:06.46 | teknoprep | [TK]D-Fender, on the aastra |
| 00:07.15 | [TK]D-Fender | teknoprep: OK enough, but nothign touches Polycom to date. |
| 00:07.41 | teknoprep | [TK]D-Fender, not speakerphone tho.. no one uses speakerphones where i put phones in |
| 00:07.49 | teknoprep | [TK]D-Fender, and if they did i would buy them a polycom |
| 00:08.01 | [TK]D-Fender | teknoprep: I'm talking handset quality here... |
| 00:08.10 | teknoprep | [TK]D-Fender, hmm that sucks |
| 00:08.15 | [TK]D-Fender | teknoprep: the Aastra is OK, but too light, and sounds a bit hollow |
| 00:08.34 | teknoprep | [TK]D-Fender, audiophiles in the VoIP industry |
| 00:08.36 | [TK]D-Fender | teknoprep: I HATE the handset. For basic use I'd rather have my bedside IP 301. |
| 00:08.40 | teknoprep | [TK]D-Fender, isn't that an oxymoron ? |
| 00:08.52 | [TK]D-Fender | teknoprep: Considering how far there is to fall? NO :p |
| 00:08.57 | teknoprep | [TK]D-Fender, i always did LOVE the weight of a cisco phone |
| 00:09.32 | [TK]D-Fender | teknoprep: Cisco has the best feel, Polycom the most stable and usable interface & audio. Aastra the most dynamic functionailty |
| 00:09.39 | silvertip257 | [TK]D-Fender: I haven't omitted anything from what I pasted there - verbose/debug=10000 ... |
| 00:10.05 | teknoprep | [TK]D-Fender, well i am going to have to try these aastra phones out... i can not find a site yet... that has aastra being a bad phone |
| 00:10.16 | teknoprep | [TK]D-Fender, they aren't grandstream audio bad are they ? |
| 00:10.24 | Qwell | nothing is gs audio bad |
| 00:10.29 | teknoprep | lol |
| 00:10.32 | teknoprep | so true |
| 00:10.40 | Qwell | tin can + string > gs audio |
| 00:10.48 | teknoprep | haha |
| 00:10.51 | *** part/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 00:10.51 | [TK]D-Fender | silvertip257: Sorry, you have invite for your outbound attempt. Your output is fraudulent or broken. |
| 00:10.56 | teknoprep | tin can + string = analouge |
| 00:11.10 | [TK]D-Fender | Qwell: GS GS bad. Tautology++ |
| 00:11.14 | [TK]D-Fender | = |
| 00:11.28 | marc7 | is a significant delay introduced if a call is daisy chained through multiple asterisk servers rather than having it just pass through one? |
| 00:11.29 | Qwell | EVEN Grandstream isn't that bad |
| 00:11.31 | Qwell | err, wait |
| 00:11.56 | jaytee | rofl |
| 00:12.06 | Qwell | marc7: are the Asterisk servers separated by any large bodies of water? |
| 00:12.22 | marc7 | Qwell: nope, same datacenter. |
| 00:12.24 | Qwell | like, do you have to go through Australia in order to make a call from California to Texas? |
| 00:12.39 | Qwell | then, no, not significant |
| 00:12.47 | Qwell | unless you're talking like 20 servers |
| 00:13.02 | teknoprep | even on 20 servers you wouldn't have that big of a transport delay |
| 00:13.18 | teknoprep | as long as you don't transcode anything |
| 00:13.40 | marc7 | I'm just putting this down on paper, imagining that calls from a Polycom will hit an internet-facing call gateway that will handle the G729 translation... then pass over IAX to an application server (voicemail, conference bridge, whatever), then over to a second call gateway which has physical circuits with a carrier |
| 00:14.06 | *** part/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 00:14.29 | marc7 | so IP Phone <-> {Internet} <-> Internet Gateway <-> Application Server <-> Telephony Gateway <-> {PSTN} <-> Plain Old Telephone |
| 00:15.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:15.38 | marc7 | and having the RTP stream pass through all three servers in the middle --- providing they're all in close contact and are fairly capable machines --- shouldn't introduce significant delay, and generally isn't a terrible idea |
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| 00:23.34 | x86 | marc7: don't see why you'd need three servers like that, unless perhaps you were running SER on the outter-most one |
| 00:24.31 | marc7 | x86: OpenSER is going to be directing calls to the Gateways, and will be a process running on a separate set of servers |
| 00:24.51 | x86 | jesus christ are you trying to start a Vonage killer? :) |
| 00:25.28 | x86 | must be a startup company with a lot of VC to burn through |
| 00:25.51 | marc7 | heh, this setup *will* be a little overkill |
| 00:26.51 | marc7 | just trying to build a modular redundant system |
| 00:28.38 | x86 | nothing in that sounds redundant ;) |
| 00:28.40 | marc7 | god, that sounds like I've just picked up buzzwords off a promotional article. "make it scalable for the aim of synergizing" |
| 00:29.08 | x86 | piecemeal'ed out, sure... but not redundant |
| 00:29.15 | marc7 | the aim is fault-tolerancy... one node could go down and the least amount of calls affected, and the role can be picked up by any other number of servers. |
| 00:29.42 | marc7 | i'm missing words in my sentences, (always proofread to see if you any words out), but that's the general idea. |
| 00:30.02 | Qwell | proofread your subdialogs too |
| 00:30.19 | marc7 | it was intentional that time ;-) |
| 00:30.24 | Qwell | sure ;) |
| 00:30.57 | x86 | hah |
| 00:31.03 | x86 | Qwell ftw |
| 00:31.12 | x86 | qwell++ |
| 00:31.15 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 00:31.19 | x86 | do we have a karma bot here? that'd be rad |
| 00:31.21 | Qwell | jbot: karma Qwell |
| 00:31.21 | jbot | qwell has karma of 9 |
| 00:31.24 | marc7 | I know that traditional telcos rely on a small number of prohibitively expensive call switches to handle the bulk of their work, so I'm just trying to gauge if it's easy enough to distribute the load across smaller servers without impeding overall quality |
| 00:31.26 | ibnolqaiyem | what is Zapata? |
| 00:31.26 | Qwell | yes, yes we do |
| 00:31.34 | x86 | jbot: karma x86 |
| 00:31.34 | jbot | x86 has karma of -2 |
| 00:31.48 | x86 | (was seeing if it was random or not hehe) |
| 00:32.11 | x86 | ouch, people hate me :( |
| 00:32.37 | [TK]D-Fender | ~karma |
| 00:32.37 | jbot | [tk]d-fender has karma of 10 |
| 00:32.42 | [TK]D-Fender | OMG |
| 00:32.44 | Qwell | pfft |
| 00:32.51 | [TK]D-Fender | ~karmakarma |
| 00:32.51 | jbot | Karma Chameleon! |
| 00:32.53 | [TK]D-Fender | :D |
| 00:32.56 | jaytee | ~karma |
| 00:32.56 | jbot | jaytee has neutral karma |
| 00:33.09 | jaytee | damn, I'm neutered |
| 00:33.20 | [TK]D-Fender | Qwell: My karma ran over your dogma ;) |
| 00:33.27 | Qwell | ~dogma |
| 00:33.28 | jbot | extra, extra, read all about it, dogma is a mediocre KMFDM album or called a linux high |
| 00:34.54 | x86 | ibnolqaiyem: http://www.zapatatelephony.org/ |
| 00:37.43 | drfreeze | Ok, part way there to getting call pickup working |
| 00:38.25 | drfreeze | the *8 works, but have to do *8 <<dial |
| 00:43.12 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 00:43.23 | x86 | damn, zapatatelephony.org is WAY outdated |
| 00:44.29 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:44.31 | coppice | it hasn't been touched since the early days of the tormenta 2 card |
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| 00:53.18 | TJNII | Can iax debug be set to a specific IP like sip debug? I'm not seeing the option here... |
| 00:55.06 | *** join/#asterisk swiftkick (n=Miranda@mail.beanproducts.com) |
| 00:55.46 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-270fa81871a30642) |
| 00:56.28 | swiftkick | HI, question about extension pattern matching on DID calls. Pattern matching only seems to work properly when I *omit* the _ ..... |
| 00:56.38 | swiftkick | and I am not really sure why. |
| 00:58.21 | swiftkick | asterisk-gui enforces placing a _ infront of its pattern matching interface for "Incoming Call Rules". shouldn't a fixed phone number such as 3124445555 match correctly as a pattern also, e.g. _3124445555 ? |
| 00:58.57 | swiftkick | e.g. exten = _3124445555,1,Goto(voicemenu-custom-1|s|1) doesnt seem to work |
| 00:59.10 | Maliuta | if you want gui support go to #asterisk-gui |
| 00:59.13 | swiftkick | whereas exten = 312444XXXX ,1,Goto(voicemenu-custom-1|s|1) *DOES* work |
| 00:59.25 | swiftkick | this isnt about asterisk-gui. this is about the behavior of a standard asterisk macro |
| 00:59.58 | Maliuta | well "exten = " will get you nowhere to start with |
| 01:00.30 | swiftkick | nobody has ever explicated a meaningful difference between = and => ; as far as I have been able to discern they are synonyms |
| 01:00.41 | swiftkick | i'd love to learn otherwise if such is the case. |
| 01:01.06 | jaytee | have you tried exten => _3124445555 |
| 01:01.09 | swiftkick | besides which the parts of extensions.conf that asterisk-gui writes, it uses = pretty much exclusively |
| 01:01.29 | swiftkick | no i would be happy to try it |
| 01:01.47 | [TK]D-Fender | swiftkick: pastebin your dialplan and the CLI output of a failed call |
| 01:01.49 | [TK]D-Fender | ~pb |
| 01:01.50 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:01.51 | [TK]D-Fender | ^^^^^^^^^^^^ |
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| 01:02.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:04.55 | Katty | hai russell |
| 01:06.51 | russellb | hi2u |
| 01:08.04 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
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| 01:12.12 | *** join/#asterisk thing1 (n=Dwayne@64.42.227.97) |
| 01:12.57 | thing1 | hi i'm trying to place a call from cisco call manager to my asterisk box and it gives me Failed to authenticate user phonenumber@callmanagerip |
| 01:13.10 | *** join/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 01:13.13 | swiftkick | [TK]D-Fender: pastebin not necessary. this is only like my nth time under the hood with extensions.conf where (200 < n < 500). |
| 01:13.16 | thing1 | can anyone help |
| 01:13.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 01:13.49 | thing1 | hi i'm trying to place a call from cisco call manager to my asterisk box and it gives me Failed to authenticate user phonenumber@callmanagerip |
| 01:13.55 | swiftkick | [TK]D-Fender: pastebin not necessary. this is only like my nth time under the hood with extensions.conf where (200 < n < 500). |
| 01:14.01 | Maliuta | thing1: you said that already |
| 01:14.06 | thing1 | soory |
| 01:14.29 | Maliuta | thing1: do you have any more details? |
| 01:14.44 | thing1 | what would u like? |
| 01:14.53 | [TK]D-Fender | swiftkick: Don't quite get your analogy. |
| 01:14.56 | thing1 | i know the sip trunk is up because it talks to asterusj |
| 01:15.05 | swiftkick | the offending line in extensions.conf is exten => _.7739996638,1,Goto(ringrroups-custom-2,s,1) the CLI output of the failed call is: [Jul 23 20:07:27] NOTICE[4621]: chan_sip.c:13879 handle_request_invite: Call from '' to extension '7739996638' rejected because extension not found. |
| 01:15.12 | Maliuta | thing1: well is the * box sending the error? |
| 01:15.29 | Maliuta | thing1: have you tried a sip debug on the * cli? |
| 01:15.34 | thing1 | i see the auth error in cli when i try and place a call to asterisk through call manger |
| 01:15.38 | thing1 | yes |
| 01:16.10 | swiftkick | [TK]D-Fender: the error is the same with or without the leading "." |
| 01:16.31 | Maliuta | thing1: are you sure the sip trunk is configured properly? sounds like something is not being passed along properly |
| 01:16.36 | swiftkick | [TK]D-Fender: however, lose the leading _ and either syntax performs as I would expect (!!!!) |
| 01:17.09 | [TK]D-Fender | swiftkick: if you want it debugged, show the probelm. |
| 01:17.10 | thing1 | http://pastebin.com/m13c7e122 |
| 01:17.46 | *** join/#asterisk tengulre (n=tengulre@125.69.124.131) |
| 01:17.52 | swiftkick | [TK]D-Fender: the CLI output of the failed call is: [Jul 23 20:07:27] NOTICE[4621]: chan_sip.c:13879 handle_request_invite: Call from '' to extension '7739996638' rejected because extension not found. |
| 01:18.06 | swiftkick | [TK]D-Fender: The offending line in extensions.conf is exten => _.7739996638,1,Goto(ringrroups-custom-2,s,1) |
| 01:18.09 | swiftkick | that is the problem. |
| 01:18.15 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:18.20 | [TK]D-Fender | swiftkick: Well certainly you should not have a "." in front. |
| 01:18.25 | thing1 | [Jul 23 20:16:14] NOTICE[23946]: chan_sip.c:13200 handle_request_invite: Failed to authenticate user <sip:2042677134@10.2.0.2 |
| 01:18.28 | Maliuta | _.7739996638 won't match 7739996638 |
| 01:18.34 | [TK]D-Fender | swiftkick: and "_" would not matter |
| 01:19.10 | Maliuta | thing1: no user and secret? |
| 01:19.40 | thing1 | ok, what should it be i tried with user=user1 pass=1234 from user=user1 |
| 01:20.04 | swiftkick | [TK]D-Fender: Well, neither form works. _7739996638 does not match but 77739996638 *does* |
| 01:20.13 | thing1 | where should it auth to? there is nothing in call manager? |
| 01:20.15 | swiftkick | er... one 7 too many but you get the idea |
| 01:21.04 | [TK]D-Fender | swiftkick: pastebin a precise non-funcfiotnal set without the "." |
| 01:21.08 | swiftkick | also I have a case where I have multiple DID's 7739995XXX . specifying 7739995XXX works. Specifying _7739995XXX *doesnt*. ??? |
| 01:22.05 | swiftkick | 'k gimme a sec to sanitize this stuff. brb |
| 01:22.40 | swiftkick | but meanwhile |
| 01:22.47 | swiftkick | just to satisfy my curiosity |
| 01:22.53 | swiftkick | because i am getting more and more involved with our asterisk system here |
| 01:23.14 | swiftkick | what, if any, is the difference between = and => ? is there a URL that can explain this to me? |
| 01:23.20 | [TK]D-Fender | swiftkick: none. |
| 01:23.27 | swiftkick | thanks. thats what i thought. |
| 01:24.12 | [TK]D-Fender | swiftkick: Certain things were sampled with one VS the other and for visual clarity you'll see them mixed. No real need however |
| 01:24.16 | tzanger | my parasite is immune to the vaccine |
| 01:24.23 | tzanger | but it seems to be unable to spread to the rest of the world |
| 01:24.38 | jaytee | tzanger, they said the same thing about Karl Rove |
| 01:24.42 | [TK]D-Fender | tzanger: Yes, last game I got stopped in Argentina. Pissed me right off. |
| 01:25.07 | Maliuta | thing1: is there a good reason for the "fromdomain"? and the "nat=yes"? |
| 01:25.57 | swiftkick | [TK]D-Fender: thank you. asterisk-gui writes ='s . i realize this IS essentially a question about why asterisk-gui insists on inserting a _ in its incoming call (DID) rules, but I've tried asking questions there before and the silence there is often deafening. |
| 01:26.07 | Maliuta | thing1: also since it's only a partial sip.conf I have NFI what is in your [general] context |
| 01:26.18 | tzanger | [TK]D-Fender: heh, I have infected everything but a handful of countries |
| 01:26.28 | tzanger | takes too damn long :-( |
| 01:26.32 | [TK]D-Fender | swiftkick: "_" denotes a pattern. |
| 01:26.38 | jaytee | [TK]D-Fender, the game you played last nite where you wiped out the species, was that in Realistic or Relaxed? |
| 01:26.43 | swiftkick | [TK]D-Fender yep ive gotten that far |
| 01:26.56 | [TK]D-Fender | swiftkick: It doesn't mean you have to use pattern chars in what follows, but it does mean that they are not necessarily literal. |
| 01:27.02 | tzanger | I've never played relaxed |
| 01:27.17 | swiftkick | [TK]D-Fender: does a "pattern" require at least one "nonliteral" character? actually that doesnt seem to be the issue |
| 01:27.19 | [TK]D-Fender | tzanger: relaxed. 2 world-kills |
| 01:27.36 | tzanger | I'll have to try relaxed |
| 01:27.39 | Maliuta | thing1: and have you read http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration? |
| 01:27.47 | swiftkick | e.g. a leading _ should have no effect on exten => _9999999999,1,x(x) |
| 01:28.04 | tzanger | swiftkick: cirrect |
| 01:28.33 | swiftkick | so (just to be ultra pedantic about this) exten => _9999999999,1,x(x) and exten => 9999999999,1,x(x) are supposed to be synonymous. |
| 01:28.55 | swiftkick | essentially? |
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| 01:29.11 | tzanger | swiftkick: _ si only for pattern matching |
| 01:29.19 | tzanger | which makes me wonder why it's there to begin with |
| 01:30.00 | swiftkick | it has to do with how asterisk-gui writes its DID rules. |
| 01:30.05 | tzanger | no |
| 01:30.15 | tzanger | it's how it was designed |
| 01:30.34 | tzanger | I'm just wondering out loud why you have to alert asterisk to the fact that wildcards will be used |
| 01:31.16 | [TK]D-Fender | tzanger: exten => xavier,1,NoOp(because you can't dial a nifty name with X in it otherwise) |
| 01:31.38 | tzanger | heh |
| 01:31.41 | [intra]lanman | would you rather it used regex all the time? |
| 01:31.45 | [TK]D-Fender | load chan_obvious.so |
| 01:31.46 | tzanger | contrived, but true |
| 01:32.24 | tzanger | exten _n1nc0mp00p,1,... |
| 01:32.25 | [intra]lanman | i personally am pretty fond of regex, but i guess they're not for everyone |
| 01:33.22 | TJNII | Learning curve is a bit steep for regex, imho. |
| 01:34.13 | [TK]D-Fender | I agree with Regex as a basis personally |
| 01:34.37 | [intra]lanman | hmmm, that seems weird to me... you can match ^anything_here$ and not necessarily use patterns |
| 01:34.59 | [intra]lanman | that doesn't seem so steep |
| 01:35.04 | russellb | you can match an X, you just put it in brackets |
| 01:35.33 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:35.39 | russellb | exten => _ro[X]0rXXX,1,NoOp |
| 01:35.41 | silvertip257 | What search terms would I need to find out how to forward all inbound calls to my * to one extension? I'm drowning after seeing so much varied config on the web. |
| 01:35.46 | russellb | that will match r0x0r following 3 digits |
| 01:35.54 | [intra]lanman | can you match 300[0-4] and 300[5-9] in another exten? |
| 01:35.58 | russellb | er, roX0r |
| 01:35.59 | russellb | whatever. |
| 01:36.07 | tzanger | heh |
| 01:36.16 | russellb | [intra]lanman: yeah |
| 01:36.28 | bkw__ | TJNII: regex is easy to learn |
| 01:36.41 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582590.dsl.bell.ca) |
| 01:36.52 | bkw__ | er are easy. damn i'm tired |
| 01:38.05 | [TK]D-Fender | silvertip257: Time to start at the BASICS. |
| 01:38.08 | [TK]D-Fender | ~book |
| 01:38.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:38.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 01:38.25 | silvertip257 | :-) |
| 01:38.33 | silvertip257 | yeah that thing .. |
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| 01:43.41 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 01:47.04 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:50.31 | thing1 | Maliuta: it doesn't say nothing about recieving calls |
| 01:51.42 | [TK]D-Fender | thing1: And you shouldn't not use double-negatives in here neither |
| 01:52.31 | thing1 | aint that quaint |
| 01:52.51 | *** part/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 01:53.42 | Maliuta | thing1: again I am making assumptions about your network layout and the rest of your sip.conf http://pastebin.com/d30b264ef |
| 01:54.07 | thing1 | sip show peers says its reachable |
| 01:54.27 | thing1 | so i'm thinking that part is ok |
| 01:54.48 | Maliuta | thing1: the nat=yes will screw with the SIP packets |
| 01:54.58 | Maliuta | as they are sent from * |
| 01:55.01 | thing1 | i put it off |
| 01:55.04 | thing1 | still that same |
| 01:55.07 | *** part/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 01:55.32 | [TK]D-Fender | I'm thinking that pastebin is by itself worthless. |
| 01:55.36 | Maliuta | thing1: can you show me more of the sip.conf? and tell me what version of callmanager? |
| 01:55.40 | thing1 | why is it using the callerid of the number calling for auth |
| 01:55.56 | thing1 | it's call manager express, i'm not sure of the exact version |
| 01:56.16 | Maliuta | thing1: because of something in the general context in your sip.conf? |
| 01:56.30 | [TK]D-Fender | thing1: "username=blah" |
| 01:56.45 | Maliuta | we have no idea, we havn't seen the rest of the file |
| 01:56.50 | thing1 | general contect is from internal |
| 01:57.00 | thing1 | do u want to c sip conf? |
| 01:57.30 | [TK]D-Fender | thing1: If you expect help, you need to show a failed call with SIP DEBUG, along with your sip config. |
| 01:57.42 | thing1 | http://pastebin.com/m359fa7f1 |
| 01:58.58 | [TK]D-Fender | thing1: Were is that 10. subnet? |
| 01:59.03 | [TK]D-Fender | where* |
| 01:59.05 | thing1 | http://pastebin.com/pastebin.php |
| 01:59.15 | [TK]D-Fender | thing1: Bad link |
| 01:59.15 | thing1 | its the subnet the call manager is on |
| 01:59.21 | thing1 | my network is 192.168.0.0 |
| 01:59.25 | [TK]D-Fender | thing1: how does * get to it? |
| 01:59.34 | thing1 | http://pastebin.com/m6268f6e2 |
| 01:59.44 | thing1 | through a smoothwall box |
| 01:59.52 | thing1 | nat |
| 02:00.00 | [TK]D-Fender | thing1: networking details, brand names don't help/ |
| 02:00.02 | [TK]D-Fender | NAT |
| 02:00.05 | [TK]D-Fender | ok. |
| 02:00.07 | [TK]D-Fender | not good. |
| 02:00.17 | [TK]D-Fender | perhaps workable |
| 02:00.33 | thing1 | i don't know how else i would do it |
| 02:00.50 | thing1 | my * needs to by in 192.168.0.0 |
| 02:01.26 | [TK]D-Fender | Found no matching peer or user for '10.2.0.2:57417' |
| 02:02.01 | thing1 | what line is that |
| 02:02.02 | thing1 | ? |
| 02:02.13 | thing1 | got it |
| 02:02.33 | thing1 | y is it saying that |
| 02:02.36 | Maliuta | thing1: and you have a nat between your localnet and the 10.2.0.0 network? |
| 02:02.46 | thing1 | yes |
| 02:02.55 | [TK]D-Fender | thing1: your phones should all be "host=dynamic" |
| 02:03.30 | thing1 | they are except for the sip trunk one |
| 02:03.47 | thing1 | o |
| 02:03.57 | Maliuta | thing1: so the path is CM -> NAT -> * ? |
| 02:04.11 | thing1 | yes |
| 02:04.51 | Maliuta | thing1: and you can't set a username and pass on the call manager? |
| 02:04.58 | [TK]D-Fender | thing1: You seem to have failed auth for the call coming from CM |
| 02:05.06 | thing1 | apperantly not |
| 02:05.21 | thing1 | why is it using the high port |
| 02:05.27 | Maliuta | thing1: you can't user allowguest=no then |
| 02:05.57 | Maliuta | thing1: if you read appendix A of the book it will tell you this explicitly |
| 02:07.28 | *** join/#asterisk eboncomm (n=pmccaffr@students.nott.liberty.ask4.co.uk) |
| 02:07.32 | eboncomm | greetings all |
| 02:07.54 | thing1 | i guess i must have missed that |
| 02:08.07 | Maliuta | thing1: so you _could_ try just putting "allowguest=yes" in the [PRI-Trunk] peer definition |
| 02:08.21 | eboncomm | i seem to be having a strange issue on a new asterisk box i put together with 5 Cisco 7940s flashed with the latest SIP image, it seems the server is not detecting the hangup of the cisco phones on a iax voip call |
| 02:08.27 | eboncomm | does anyone know what might cause that? |
| 02:08.43 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 02:09.06 | Maliuta | eboncomm: I just flashed mine to 8.8 and have no issues |
| 02:09.25 | Maliuta | eboncomm: could be something in the sip.conf or iax.conf |
| 02:09.34 | eboncomm | i have these phones working correctly on another system |
| 02:09.42 | Maliuta | eboncomm: or your dialplan |
| 02:09.48 | eboncomm | do you have any idea where i might check? |
| 02:09.56 | eboncomm | i have the HangUp() command at the end of the call |
| 02:10.01 | eboncomm | and Answer() at the beggining |
| 02:10.10 | Maliuta | what options are you passing to the dial? |
| 02:10.30 | Maliuta | is the other end holding the line open? |
| 02:10.34 | eboncomm | Dial(SIP/1001,20) |
| 02:10.43 | eboncomm | yes, the line wont close until i hangup on the outside line side |
| 02:11.17 | eboncomm | the phone "thinks" it hungup the line, but asterisk isnt receiving the command appareantly, the CLI does nothing |
| 02:12.10 | Maliuta | eboncomm: hang on, are you going client->IAX->*->SIP->cisco? |
| 02:12.30 | eboncomm | yes |
| 02:12.45 | thing1 | well i'm still getting the same error |
| 02:12.54 | eboncomm | client = my cell phone or my other voip line |
| 02:13.06 | Maliuta | so show me the sip.conf, iax.conf and extensions.conf |
| 02:13.19 | eboncomm | should i post it directly in here? |
| 02:13.26 | Maliuta | ~pastebin |
| 02:13.27 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:13.43 | eboncomm | ok thank you, bare with me |
| 02:13.44 | Maliuta | thing1: what does the new sip.conf look like? |
| 02:14.43 | eboncomm | my sip.conf |
| 02:14.44 | eboncomm | http://pastebin.com/d11e71349 |
| 02:15.03 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-8-115.nyc.res.rr.com) |
| 02:16.20 | eboncomm | my iax.conf http://pastebin.com/d535b12a5 |
| 02:16.24 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 02:17.29 | eboncomm | my extensions.conf http://pastebin.com/d47e504f3 |
| 02:21.23 | Maliuta | eboncomm: try adding ",g" onto the end of the Dial() in the incoming context |
| 02:21.41 | eboncomm | mmk, ill do that now |
| 02:22.14 | *** join/#asterisk guilherme-jorge (n=guilherm@189.100.240.34) |
| 02:22.32 | eboncomm | didnt change it |
| 02:22.39 | Maliuta | eboncomm: to understand why go to Appendix B of the book and look at the explanation of Dial() |
| 02:22.52 | Maliuta | eboncomm: did you reload the diaplan? |
| 02:23.10 | eboncomm | yes |
| 02:23.42 | eboncomm | it appears that asterisk isnt getting the fact the phone has hungup |
| 02:24.14 | Maliuta | what does sip show channels give you? during the call and after the hangup? |
| 02:24.45 | thing1 | http://pastebin.com/m1f759bfa |
| 02:24.49 | thing1 | Maliuta |
| 02:27.37 | mosty | is there a way to get chan_woomera to do software echo cancellation? or would that have to be done on the woomera server/backend? |
| 02:28.33 | eboncomm | cli output http://pastebin.com/d47463049 |
| 02:30.18 | Maliuta | eboncomm: I was more interested in "sip show channels" during the call and after the cisco hangs up |
| 02:30.54 | eboncomm | oooo ok, i apologize, one moment |
| 02:32.50 | Maliuta | thing1: what does the pri trunk look like at the moment? |
| 02:33.01 | Maliuta | thing1: and how does it fit into your dialplan? |
| 02:33.14 | eboncomm | http://pastebin.com/d643f2644 |
| 02:33.34 | eboncomm | Maliuta: was that for me? |
| 02:33.57 | thing1 | i have a digital receptionist that is currently answered from-external calls, that is what would pickup if the call would go through |
| 02:35.24 | *** join/#asterisk ManxPower (n=manxpowe@60.sub-75-250-16.myvzw.com) |
| 02:36.05 | Maliuta | eboncomm: "sip show channels" not "sip show peers" ... 2 very different bits of information |
| 02:36.35 | Maliuta | thing1: I am thinking it can't find an extension to go into in the dialplan |
| 02:36.42 | eboncomm | again, my apologies, please hold, i very much appreciate your patience |
| 02:37.39 | Maliuta | thing1: and that debug you are showing me indicates no nat on the outgoing packets (meaning they won't look right to the call manager) |
| 02:38.22 | pkunkra | I'm trying to find a good hard phone that looks like a cellular flip phone. something like this one. http://www.dlink.com/products/?pid=485 |
| 02:38.34 | pkunkra | but i read some bad reviews about the dlink one. |
| 02:38.38 | [TK]D-Fender | pkunkra: You won't. |
| 02:38.44 | pkunkra | have any recommendations? |
| 02:38.45 | pkunkra | hmmm |
| 02:39.25 | *** join/#asterisk ManxPower (n=manxpowe@149.sub-75-200-181.myvzw.com) |
| 02:39.27 | Maliuta | pkunkra: two tincans and some string |
| 02:39.35 | pkunkra | you seem pretty sure it.. what makes you sure? |
| 02:39.49 | pkunkra | s/sure it/sure about it/ |
| 02:39.52 | eboncomm | here goes number 3... http://pastebin.com/d313cf96a |
| 02:40.03 | [TK]D-Fender | pkunkra: because : |
| 02:40.04 | pkunkra | wow. grammar correcting bot. |
| 02:40.07 | [TK]D-Fender | ~wifivoip |
| 02:40.08 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 02:40.30 | pkunkra | oh |
| 02:40.34 | pkunkra | good point. |
| 02:41.11 | pkunkra | crap |
| 02:42.01 | pkunkra | i was hoping to to find a decent phone i could have lying around on my work desk to answer home calls with. |
| 02:42.11 | pkunkra | ... without someone asking... "what is that?" |
| 02:42.37 | pkunkra | it looks weird to have a cordless sitting on your desk at work |
| 02:43.13 | [TK]D-Fender | pkunkra: Send home & work calls to your work desk phone then |
| 02:43.57 | pkunkra | tk, i need a way to distinguish between the home phone call and the work phone call. besides, my work phone is a cisco ip phone. |
| 02:44.12 | [TK]D-Fender | pkunkra: ... and? |
| 02:44.40 | pkunkra | unless i reload it (which i can do perfectly well)... i can't have it connect both to call manager vis H.323 and via SIP to my asterisk server. |
| 02:44.50 | pkunkra | vis -> via |
| 02:45.17 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 02:45.17 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:45.17 | Maliuta | eboncomm: what are on the ips .105 and .106? other handsets in that Dial()? |
| 02:45.20 | [TK]D-Fender | pkunkra: Well you just had to go and complicate it, now didn't you? |
| 02:45.31 | pkunkra | i also don't want to have to resend it back out over the POTS lines. |
| 02:45.34 | pkunkra | hahaha |
| 02:46.10 | eboncomm | yes, i have a total of 3 phones currently hooked up, the dial plan is set for 5 phones (the other two i dont have yet) |
| 02:46.30 | pkunkra | if i did send it back out over the pots lines, i'd probably hack up the caller id to distinguish between work calls and home calls. |
| 02:46.40 | pkunkra | like change the area code to 999 or something. |
| 02:47.07 | ManxPower | pkunkra: Call Manager uses SCCP aka Skinny |
| 02:47.33 | pkunkra | manx, oh. forgot about that. I thought it was H.323. |
| 02:47.52 | pkunkra | i've never admin'ed callmanager before. |
| 02:50.33 | Maliuta | eboncomm: have you tried setting canreinvite to no? |
| 02:50.59 | Maliuta | thing1: ICMP(echo) |
| 02:51.09 | Maliuta | is almost out of time |
| 02:51.21 | Maliuta | have to eat lunch and go see doctors soon |
| 02:51.28 | eboncomm | no i have not, i will try that now |
| 02:52.45 | Maliuta | eboncomm: looks a little odd that once one SIP device hangs up it is trying to do something with others |
| 02:53.06 | eboncomm | i agree, ill change that now and give it a shot |
| 02:54.38 | eboncomm | no it appears not to have fixed it :/ |
| 02:54.43 | eboncomm | i know u r running out of time |
| 02:54.51 | eboncomm | i do really appreciate all the help you have given me |
| 02:55.05 | eboncomm | could this be somehow related to my zapata files even though im using voip? |
| 02:57.12 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 02:57.54 | Maliuta | wouldn't think so |
| 02:58.09 | Maliuta | I would keep looking at stuff like options to Dial() |
| 02:58.12 | thing1 | Maliuta: now i get the phone carrier saying the number you have reached is not in servive |
| 02:58.15 | thing1 | service |
| 02:58.47 | Maliuta | thing1: I have to run, sorry. If you are still around in a few hours I can have another look |
| 02:59.00 | thing1 | ok thanks anyeay |
| 02:59.06 | eboncomm | ok, i will look into that, again, i appreciate your help, have a good day |
| 03:00.56 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-104-82-57.pskn.east.verizon.net) |
| 03:01.33 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 03:02.10 | guilherme-jorge | hello all, I've some doubts to record sounds in asterisk. I've a server running asterisk 1.4.17 and all of my extensions are Polycom phones that supports g729,ulaw and alaw. Everything running on this PBX supports g729 codec (outgoing and incoming providers, IP phones), and Im gonna record some sounds to make a IVR system. Can I configure all users (phones, providers) in this pbx disabling all codecs and enabling just g729? If yes, have I t |
| 03:02.11 | guilherme-jorge | o record sounds using g729? |
| 03:02.50 | guilherme-jorge | Does it make sense? |
| 03:02.51 | guilherme-jorge | :) |
| 03:03.02 | drfreeze | Anyone have core dump problems when trying to park a call? |
| 03:03.03 | drfreeze | http://pastie.textmate.org/private/9m2wda0jjog4wdkiaybma |
| 03:05.10 | [TK]D-Fender | <PROTECTED> |
| 03:05.25 | [TK]D-Fender | guilherme-jorge: And you will not be able to use MeetMe, or Page. |
| 03:06.04 | Qwell | or monitor, etc, etc |
| 03:06.08 | drfreeze | When * isn't crashing on parking calls, it is hanging up on the caller right after they are parked and before the parker hangs up |
| 03:06.11 | drfreeze | http://pastie.textmate.org/private/yfjssaqi5ksd8ysvpuq |
| 03:06.15 | Qwell | well, maybe monitor. probably not though |
| 03:07.04 | [TK]D-Fender | Qwell: Monitor should just fork the strem IIRC, and of course would also have to be in G.729, but I'd bet you can't MIX it easily |
| 03:07.24 | Qwell | yeah, mix wouldn't work |
| 03:07.36 | Qwell | but I think monitor might be slin for some reason |
| 03:07.57 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 03:08.32 | [TK]D-Fender | Qwell: Could be. Mine is just a pure extrapolation. |
| 03:08.40 | Qwell | so's mine :p |
| 03:08.50 | Qwell | I can't think of any reason it would be slin though |
| 03:09.56 | guilherme-jorge | <[TK]D-Fender> Does it represent some advantage or disadvantage? |
| 03:10.01 | [TK]D-Fender | Qwell: Lowest common denominator. Chanspy would be a big doubt item too... |
| 03:10.10 | Qwell | right |
| 03:10.20 | Qwell | but again - I can't think of a reason why |
| 03:10.25 | [TK]D-Fender | guilherme-jorge: When I tell you there are a pile of things you ca't do, those are what we would call "idsadvantages" |
| 03:10.35 | [TK]D-Fender | guilherme-jorge: Only difference is how much you care about the,/ |
| 03:10.56 | [TK]D-Fender | Qwell: Chanspy because of "whisper" which for sure mixes |
| 03:11.03 | [TK]D-Fender | them* |
| 03:12.11 | Qwell | yeah, whisper would have to mix - unless it just blocked audio while the whisperer was talking |
| 03:12.16 | Qwell | oh, but it couldn't do that |
| 03:12.23 | Qwell | have to be slin for talker detection |
| 03:12.36 | drfreeze | Ok, I can now do call pickup, transfer and 10% park a call |
| 03:12.56 | inv_arp | m00 |
| 03:13.02 | drfreeze | for parking, I get a park extension, but the caller gets hung up on after about 3 seconds aftre parking |
| 03:14.57 | file | is a ulaw guy living in a signed linear world |
| 03:14.58 | [TK]D-Fender | Qwell: Ah so many ways to fail. |
| 03:15.14 | Qwell | file: get out - we only support alaw here |
| 03:15.20 | Qwell | <3 |
| 03:15.31 | [TK]D-Fender | IAM the law! |
| 03:15.53 | [TK]D-Fender | serves up his best Sly Stallone impression |
| 03:18.01 | drfreeze | [TK]D-Fender: hey lawman, any ideas why call parking is being flaky? |
| 03:18.31 | [TK]D-Fender | drfreeze: pastebin |
| 03:19.18 | drfreeze | [TK]D-Fender: 2 pastes above |
| 03:19.21 | MikeJ | Qwell: you are SUCH a ulaw |
| 03:19.34 | drfreeze | http://pastie.textmate.org/private/yfjssaqi5ksd8ysvpuq |
| 03:19.39 | drfreeze | http://pastie.textmate.org/private/9m2wda0jjog4wdkiaybma |
| 03:19.59 | MikeJ | you let pasties in here.. is that a little indecent? |
| 03:20.53 | Qwell | MikeJ: what're you trying to say? |
| 03:21.22 | MikeJ | hmm.. about pasties or ulaw? :P |
| 03:21.39 | MikeJ | i am trying to say to NEVER buy ram from http://MacRamDirect.com |
| 03:21.53 | MikeJ | you'll end up with a pretty silver kernel panic box |
| 03:22.02 | [TK]D-Fender | drfreeze: Neato. No Clue |
| 03:22.32 | file | MikeJ: hammers fix everything |
| 03:22.39 | MikeJ | heh |
| 03:22.42 | drfreeze | [TK]D-Fender: thnks anyway |
| 03:23.00 | [TK]D-Fender | file: Break it down... Hammer time! |
| 03:26.57 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.121) |
| 03:26.59 | Qwell | MikeJ: I was referring to the the ulaw, but I've got nothing |
| 03:27.10 | MikeJ | heh |
| 03:27.34 | Qwell | btw, I hate codecs. |
| 03:27.44 | Qwell | all of them. |
| 03:28.09 | Qwell | maybe not the 16khz ones... dunno |
| 03:28.23 | Qwell | sick of people calling me "ma'am" on the phone :p |
| 03:28.49 | MikeJ | heh |
| 03:28.59 | MikeJ | uwb slin all the way! |
| 03:29.04 | Qwell | slin too |
| 03:29.12 | MikeJ | uwb? |
| 03:29.20 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 03:29.21 | Qwell | 48khz? |
| 03:29.28 | MikeJ | 32 |
| 03:29.32 | Qwell | either way |
| 03:29.33 | MikeJ | but sure.. why bot |
| 03:29.34 | MikeJ | not |
| 03:29.54 | Qwell | maybe I actually sound like that - who knows |
| 03:30.21 | file | you don't |
| 03:30.35 | MikeJ | Qwell: where you live these days.. you move down south? |
| 03:30.40 | Qwell | yeo |
| 03:30.42 | Qwell | yep* |
| 03:31.01 | [TK]D-Fender | Wow, kiss the USD$ bai-bai : http://digg.com/world_news/YOU_JUST_GOT_SCREWED_and_Nobody_Will_tell_you_URGENT |
| 03:31.03 | MikeJ | file.. you finally give in and move down to AL? |
| 03:31.05 | Qwell | haven't been back Californee-way in like 18 months |
| 03:31.10 | file | MikeJ: nay |
| 03:31.15 | MikeJ | he |
| 03:31.40 | [intra]lanman | weren't impressed by the new building? |
| 03:31.57 | Qwell | file: but surely you know what I'm talking about, re phone? |
| 03:32.02 | file | Qwell: yes |
| 03:32.30 | file | Qwell: that's why my Sears catalog says Mrs. Joshua Colp... |
| 03:32.44 | file | same thing happens to meeee |
| 03:33.20 | Qwell | I need to setup some jack stuff, and just always call through my Asterisk box |
| 03:33.26 | Qwell | do some pitch change stuff :p |
| 03:34.57 | MikeJ | yeah.. but file really does sound like a girl.. Qwell doesn't :P |
| 03:35.05 | file | gasps |
| 03:35.07 | *** join/#asterisk ManxPower (n=manxpowe@149.sub-75-200-181.myvzw.com) |
| 03:35.08 | MikeJ | heh |
| 03:35.12 | Qwell | no he doesn't :p |
| 03:35.16 | MikeJ | the soundtouch stuff work with newer asterisk.. |
| 03:35.25 | MikeJ | Qwell: i see how it is.. |
| 03:35.27 | MikeJ | ? |
| 03:35.30 | MikeJ | err. |
| 03:35.44 | MikeJ | 2 thoughts at once.. intermixed.. didn't come out right. |
| 03:35.45 | MikeJ | nm |
| 03:36.03 | MikeJ | [intra]lanman: file likes canada |
| 03:36.06 | Qwell | does it support jack? somebody had written an Asterisk app at one point, linking against it directly |
| 03:36.23 | MikeJ | yeah.. justin tunny wrote it iirc |
| 03:36.31 | MikeJ | not sure if its been updated from 1.2 |
| 03:36.32 | file | MikeJ: and US hates me |
| 03:36.34 | Qwell | mmm, sounds familiar |
| 03:36.48 | MikeJ | digium can't sponsor your in? |
| 03:36.52 | MikeJ | you in |
| 03:36.55 | Qwell | there are a bunch of apps that support jack that can do that type of thing though |
| 03:36.58 | file | MikeJ: not exactly, no |
| 03:37.06 | MikeJ | lame |
| 03:37.12 | MikeJ | come visit.. |
| 03:37.24 | MikeJ | oh.. I'm in toronto next week.. |
| 03:37.43 | file | oh? what'cha doing in that neck of the woods |
| 03:37.49 | file | please don't invade |
| 03:37.49 | MikeJ | hangin |
| 03:37.56 | Qwell | Aren't you like right by TO? |
| 03:38.03 | Qwell | Chicago? no? |
| 03:38.44 | MikeJ | detroit |
| 03:38.48 | MikeJ | right between the 2 |
| 03:38.48 | drfreeze | Well, I have confirmed that parking a call restart *. The reason the user gets hung up on is that * restarts |
| 03:38.49 | Qwell | close enough |
| 03:39.08 | Qwell | what is it, a 2 hour drive to TO? |
| 03:39.25 | Qwell | err, I don't know where detroit is :D |
| 03:39.50 | drfreeze | Anyone know about the call parking bomb ? |
| 03:39.55 | Qwell | wow, that is really close to the border |
| 03:41.06 | MikeJ | 4.5 |
| 03:41.30 | Qwell | drfreeze: what is a "call parking bomb"? |
| 03:44.04 | [intra]lanman | drfreeze: i know it doesn't happen on when i park calls :-D |
| 03:44.44 | MikeJ | it's like a ddos to call parking? |
| 03:45.59 | pkunkra | ~wifivoip |
| 03:46.00 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
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| 03:52.00 | fuzzy | does anyone know if the vtech ip8300 will work as a voip client for asterisk? |
| 03:57.53 | mosty | fuzzy, does it support SIP? |
| 03:59.32 | fuzzy | it doesn't look like it, only yahoo voice |
| 03:59.45 | mosty | then i wouldn't count on it |
| 04:00.19 | fuzzy | joyt |
| 04:00.25 | fuzzy | goes back ot best buy |
| 04:00.46 | mosty | buy a sip phone |
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| 04:28.35 | wwalker | with comedian mail, if I get dropped into the "leave a message" part is there a button to get into the "manage my mail box part? |
| 04:33.49 | [TK]D-Fender | <PROTECTED> |
| 04:34.06 | [TK]D-Fender | wwalker: Go read up on the "a" and "o" Asterisk Standard Extensions. |
| 04:34.34 | [TK]D-Fender | wwalker: This should enlighten you as to how you would go about doing what you ask. |
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| 04:46.33 | swiftkick | <PROTECTED> |
| 04:46.33 | swiftkick | "72.54.40.206 is GOOD: 26 queries in 1.8 seconds from 26 ports with std dev 17702.60" |
| 04:46.47 | swiftkick | oops sorry wrong channel |
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| 05:02.02 | wwalker | [TK]D-Fender: thank you, will do |
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| 06:01.53 | drfreeze | Qwell: the 'bomb' is that when I '# 800' to park a call, asterisk core dumps about three seconds after returning the parked extension |
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| 06:13.32 | dominic1 | what is faster astdb or mysql? |
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| 06:31.29 | MadTBone | hi asterisk gurus... I'm looking for a video codec to help get moderate quality video out of a 128k uplink. any suggestions? |
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| 06:44.07 | codestr0m | dominic1: you're asking a very ambiguous question.. mysql can have different backends.. some are faster and some are more reliable.. best way is test for your usage and use case |
| 06:44.37 | codestr0m | MadTBone: try google.. and I may be wrong, but 128k uplink (cellular uplink?) isn't much afaik |
| 06:46.33 | MadTBone | yeah...128k isn't much at all... it's actually a DSL line for an older family member who want's to keep her monthly bills to a minimum |
| 06:48.24 | Strom_M | MadTBone: video? on 128k?? |
| 06:50.08 | Strom_M | MadTBone: looks like H.264 supports bitrates as low as 64kbps |
| 06:50.18 | Strom_M | don't know if that's even remotely decent quality though |
| 06:51.48 | MadTBone | Strom_M: yeah...I don't expect much from the connection....at least she's got 768k down...so she'll be able to receive at least decent video... |
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| 07:24.33 | codestr0m | there wouldn't happen to be a gtalk echo test would there? |
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| 07:30.14 | mrnick | the localnet prevents me from reaching my voiptermination |
| 07:30.22 | mrnick | anybody got an idea how to solve this? |
| 07:30.25 | mrnick | thank you |
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| 07:53.54 | Dr-Linux|home | hi guys |
| 07:54.00 | mrnick | hi |
| 07:55.08 | Dr-Linux|home | my clients are PSTN callers, i want to offer long distance service, what opensource software should i try? |
| 07:55.40 | Dr-Linux|home | calling party and called party both are PSTN users |
| 07:55.54 | Dr-Linux|home | i don't think A2billing can help? |
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| 08:09.54 | Dr-Linux|home | anyone answer my question? |
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| 08:11.05 | codestr0m | Dr-Linux|home: depending on what sort of call plans you are going to offer you may have to highly customize your own |
| 08:12.33 | Dr-Linux|home | codestr0m: but what software is close to help |
| 08:13.03 | Dr-Linux|home | as i said .. callers are PSTN users and called party will be also PSTN |
| 08:13.34 | codestr0m | Dr-Linux|home: dear lazy web.. help me start my own itsp in a box... not be rude, but you'll have to google and test them yourself.. feel free to report back what you find |
| 08:13.54 | codestr0m | it really depends on your calling plans |
| 08:14.01 | codestr0m | billing options can vary sooo widely |
| 08:14.46 | Dr-Linux|home | just want to |
| 08:15.42 | Dr-Linux|home | know .. because that will odd if i install each software and check it, so that will be better if someone suggest me right direction |
| 08:16.18 | codestr0m | ok. don't believe me.. listen to others.. ask again when people are awake :) |
| 08:16.53 | codestr0m | frankly. you have to know what you want before you can ask others |
| 08:16.59 | Dr-Linux|home | codestr0m: thanks friend |
| 08:17.45 | Dr-Linux|home | when the awake then i'll sleep |
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| 08:25.08 | mrnick | the localnet= parameter prevents * from reaching my DID and voiptermination, how can i solve this? |
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| 09:15.42 | bboschman | Hi |
| 09:17.05 | bboschman | when I do asterisk -rx "sip show peers" I get 297 sip peers - when I run on a remote host ssh root@sipserver 'sterisk -rx "sip show peers"' I only get 20 sip peers |
| 09:17.17 | bboschman | any idea why? |
| 09:18.41 | mrnick | the localnet= parameter prevents * from reaching my DID and voiptermination, how can i solve this? |
| 09:23.27 | Keypad | Any one pro with the GUI ? |
| 09:23.33 | Keypad | It seams every one died |
| 09:41.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:42.33 | bboschman | I got the problem ... |
| 09:42.42 | bboschman | wrong /etc/hosts ;) |
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| 09:49.59 | DarKnesS_WolF | guys i want to add a country in the indecations.conf i got the rings and so on but i don't know how i should load it .. and should it be includeded into zonedata.c in zaptel driver ? or what and if this is the case so why we do have indecations.conf ? |
| 09:55.02 | swiftkick | so, can someone explain to me what the '' in this notice means: chan_sip.c:13879 handle_request_invite: Call from '' to extension '3129876543' rejected because extension not found. |
| 09:55.33 | SwK | theres not a user part for the from field in the sip invite |
| 09:56.23 | swiftkick | SwK: thank you |
| 09:57.40 | swiftkick | "user part" = entry in users.conf ? |
| 09:57.43 | SwK | no |
| 09:58.11 | SwK | in the actual invite the starts that call... theres a from header that contains the originators URI |
| 09:58.34 | tzafrir_laptop | swiftkick, the confusing part is that the sip parsing of users.conf creates only a peer entry . not a user entry |
| 09:58.41 | SwK | that uri most likely doesnt contain a user part ie: its just sip:host instead of sip:user@host |
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| 09:59.02 | swiftkick | i am really trying to figure out why that DID context doesnt work |
| 09:59.05 | tzafrir_laptop | (and also the equivalent of a register=> line, if needed) |
| 09:59.51 | swiftkick | with this entry in the incoming context: exten = 3129876543,1,Goto(ringroups-custom-3,s,1) |
| 09:59.59 | oej | switftkick: It does say "extension not found". Doesn't that give you a hint of where to look? Another hint: extensions.conf |
| 10:00.12 | SwK | swiftkick, if thats the actual line its not correct |
| 10:00.26 | swiftkick | SwK: what is in error? |
| 10:00.29 | SwK | swiftkick, its exten => (note the > sign_ |
| 10:00.39 | swiftkick | the two forms are synonymous |
| 10:00.45 | swiftkick | according to everything ive read |
| 10:00.50 | SwK | they are not |
| 10:00.54 | tzafrir_laptop | they are |
| 10:00.54 | swiftkick | reference please? |
| 10:01.01 | SwK | since when |
| 10:01.02 | DigitalIrony | in some places they are |
| 10:01.03 | swiftkick | showing they are not? |
| 10:01.14 | DigitalIrony | but not usually for an exten |
| 10:01.18 | swiftkick | <- doesnt have too many hairs left to pull out at this point |
| 10:01.26 | mrnick | the localnet= parameter prevents * from reaching my DID and voiptermination, how can i solve this? |
| 10:01.26 | SwK | it will toss an error in the dialplan |
| 10:01.30 | DigitalIrony | its considered bad habit not to use > |
| 10:01.42 | tzafrir_laptop | the test is simple: 'dialplan show' or 'dialplan show <context>' will show the dialplan that was actually parsed |
| 10:01.54 | SwK | set your verbosity and/or debug level high enuff you should see errors |
| 10:02.00 | SwK | on reloading the dialplan |
| 10:02.39 | tzafrir_laptop | actually, remove verbosity |
| 10:02.42 | swiftkick | tzafrir_laptop: I am getting this from dialplan show on that DID context |
| 10:02.48 | tzafrir_laptop | and you'll actually see errors |
| 10:03.04 | tzafrir_laptop | set verbosity to 1 , maybe? |
| 10:03.06 | DigitalIrony | thats not true |
| 10:03.18 | swiftkick | [ Context 'DID_trunk_2' created by 'pbx_config' ] |
| 10:03.18 | swiftkick | <PROTECTED> |
| 10:03.18 | swiftkick | <PROTECTED> |
| 10:03.18 | swiftkick | <PROTECTED> |
| 10:03.19 | DigitalIrony | you have to make sure verbose is set to log |
| 10:03.22 | tzafrir_laptop | with 3 you see too much noise and errors are lost |
| 10:03.41 | DigitalIrony | <PROTECTED> |
| 10:04.00 | SwK | *YAWN* |
| 10:04.10 | swiftkick | why wouldnt the above work ??? |
| 10:04.25 | DigitalIrony | Iunno....I do tech support and I always use 3.... |
| 10:04.45 | DigitalIrony | well |
| 10:04.50 | tzafrir_laptop | maybe you have a small enough dialplan |
| 10:04.53 | DigitalIrony | that looks bad |
| 10:05.07 | swiftkick | yet incoming calls on that trunk trying to reach 3129876543 give |
| 10:05.07 | swiftkick | NOTICE[939]: chan_sip.c:13879 handle_request_invite: Call from '' to extension '3129876543' rejected because extension not found. |
| 10:05.10 | DigitalIrony | don't use a . |
| 10:05.33 | tzafrir_laptop | swiftkick, in what context was it? |
| 10:05.40 | DigitalIrony | and take the ' off the number |
| 10:05.45 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:05.58 | DigitalIrony | oh thats from CLI..... |
| 10:06.01 | dominic1 | how can I check the connection state to a odbc database within asterisk? |
| 10:06.18 | swiftkick | tzafrir_laptop: should be the DID_trunk_2 context |
| 10:06.36 | swiftkick | that is what is defined for that provider in users.conf |
| 10:06.45 | tzafrir_laptop | <PROTECTED> |
| 10:06.47 | swiftkick | (provider = trunk, i guess?) |
| 10:06.59 | DarKnesS_WolF | ~centos52bug |
| 10:07.00 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 10:07.00 | tzafrir_laptop | What about '3129876543' => 1 |
| 10:07.00 | tzafrir_laptop | ? |
| 10:07.23 | swiftkick | i thought it matched s first? |
| 10:07.41 | DigitalIrony | swiftkick s is only for no exten dialed |
| 10:08.49 | swiftkick | hmmm |
| 10:08.55 | DigitalIrony | but usually |
| 10:09.09 | DigitalIrony | it matches by closest first |
| 10:09.26 | DigitalIrony | if there is an exact number in the dialplan it will match that before a wildcard |
| 10:09.34 | swiftkick | i think the problem is the '' part of the message |
| 10:09.39 | swiftkick | it is not arriving to the right context |
| 10:10.14 | SwK | swiftkick, look to see what context its sending the call to when it comes in |
| 10:10.28 | DigitalIrony | SwK what is set in your zapata.conf? |
| 10:10.31 | swiftkick | whats the easiest way to do that without 10000 pages of debug? |
| 10:10.34 | swiftkick | SwK |
| 10:10.51 | DigitalIrony | or swiftkick whoever |
| 10:10.55 | swiftkick | watches SIP debug headers scroll by |
| 10:10.57 | swiftkick | im not using zapata.conf |
| 10:11.03 | DigitalIrony | huh? |
| 10:11.08 | SwK | what does zapata.conf have to do with anything |
| 10:11.09 | DigitalIrony | what kind of card is it? |
| 10:11.13 | swiftkick | no card |
| 10:11.13 | SwK | its SIP |
| 10:11.17 | DigitalIrony | oh |
| 10:11.32 | DigitalIrony | then in your sip.conf |
| 10:11.35 | SwK | I forget the easiest way... i just turn on sip debug and a few other thigns all the time |
| 10:11.46 | swiftkick | heh |
| 10:11.50 | DigitalIrony | are you sure the context is set correctly |
| 10:12.00 | SwK | I'm used to watching tons of shit scroll by heh... mix it with tee out to a file and grep and you can find anything |
| 10:12.04 | swiftkick | sip set debug would be easier if i shut off hte other 20 phones on the network :) |
| 10:12.17 | SwK | sip debug ip |
| 10:12.18 | SwK | :P |
| 10:12.25 | DarKnesS_WolF | cool asterisk 1.6 will suports TLS |
| 10:12.33 | DigitalIrony | I do the same |
| 10:12.44 | *** join/#asterisk PakiPenguin_ (n=junaid@linuxpakistan/admin/pakipenguin) |
| 10:13.49 | swiftkick | well get this |
| 10:13.52 | oej | swiftkick: All extensions start with priority 1 |
| 10:13.55 | swiftkick | doing set sip debug for the ip of the provider |
| 10:14.00 | SwK | DigitalIrony, are you in HSV |
| 10:14.02 | swiftkick | didnt show anything when i dialed that DID from my cell |
| 10:14.13 | swiftkick | except the same NOTICE message |
| 10:14.42 | swiftkick | which is strange |
| 10:15.14 | swiftkick | <PROTECTED> |
| 10:15.27 | oej | DarKness_wo: It's only experimental support of TCP and TLS. Don't expect it to work properly in all cases. |
| 10:16.04 | swiftkick | so how do i figure out what SIP channel the NOTICE message is coming from? |
| 10:16.17 | swiftkick | e.g. call coming from '' ??? |
| 10:17.18 | SwK | sip debug ip ip.of.the.phone |
| 10:17.23 | swiftkick | its a *provider* |
| 10:17.27 | swiftkick | there is no phone |
| 10:17.32 | SwK | ok of the provider |
| 10:17.46 | swiftkick | its not showing a sip packet before it gives the message :-( |
| 10:17.53 | SwK | its probably not matching soemthing correct and its going to the default context |
| 10:18.23 | SwK | if that outputs too much crap you can always do soething like ngrep -Wbyline -q 3129876543 |
| 10:18.30 | ibnolqaiyem | i need all zaptel.conf & zapata.conf options? |
| 10:18.50 | swiftkick | can you > from the CLI ? |
| 10:19.12 | swiftkick | how do you redirect from inside the cli? :-) |
| 10:19.24 | swiftkick | eg posix --help > file 2>&1 & |
| 10:20.42 | swiftkick | ok the SIP header |
| 10:21.29 | swiftkick | shows To:"CIDNAME ."<sip:3129876543@provider.url.net> |
| 10:21.46 | swiftkick | SwK: what part of the header isnt matching that makes the '' blank in the Notice? |
| 10:21.57 | swiftkick | you said "User" ? |
| 10:22.13 | SwK | the from field look at it |
| 10:22.44 | swiftkick | hmmmmmm! |
| 10:22.47 | swiftkick | thanks one sec |
| 10:23.06 | SwK | thats why you get the '' in the notice |
| 10:23.22 | swiftkick | it is From"CIDNAME "<sip:3121234567@@different.provider.url.net> |
| 10:23.31 | swiftkick | er -@ |
| 10:23.51 | swiftkick | but that provider url is not defined anywhere, hmmm |
| 10:24.08 | swiftkick | different.provider.url.net in the From: message - what is that matching against? users.conf ? |
| 10:24.53 | SwK | users.conf i dunno... i never use it |
| 10:24.57 | *** join/#asterisk udzinari (n=david@6-164.cdn.ge) |
| 10:25.05 | swiftkick | I am not sure I understand |
| 10:25.11 | SwK | i still just define stuff in sip.conf cause i know how that works |
| 10:25.12 | Keypad | can some one help me with connecting my phoneline up to Asterisk GUI ? |
| 10:25.19 | Keypad | I tryed asking in the GUI channel |
| 10:25.20 | swiftkick | what is the From field not matching in the SIP header ??? |
| 10:25.23 | Keypad | but they died |
| 10:25.24 | tzafrir_laptop | swiftkick, you can't really . Use the logs |
| 10:26.19 | tzafrir_laptop | swiftkick, or, to redirect output of some commands: look for the script astcli in the contrib of trunk |
| 10:26.35 | tzafrir_laptop | takes some extra setup, though |
| 10:26.50 | swiftkick | it appears to not be matching because of the url in the header is different, but that url resolves to the same IP as the one defined in users.conf |
| 10:34.56 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 10:35.03 | fatcop | what can I do about this Warning: rtp.c: Unable to set TOS to 184 |
| 10:35.08 | *** join/#asterisk ToTo (n=ToTo@209.8.41.38) |
| 10:35.13 | fatcop | tried what it said at bottom of this thread. no help: http://www.trixbox.org/forums/trixbox-forums/help/rtp-c-unable-set-tos-184 |
| 10:36.46 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:38.57 | swiftkick | :ok |
| 10:38.57 | swiftkick | like |
| 10:39.00 | swiftkick | here is what is REALLY annoying |
| 10:39.10 | swiftkick | -> /etc/init.d/asterisk restart |
| 10:39.14 | swiftkick | and now *** IT WORKS *** . |
| 10:39.20 | swiftkick | WTF? ???? ? ?? ? |
| 10:39.38 | Vec | Is ael going to be the defacto asterisk config language ? |
| 10:43.07 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 10:43.49 | swiftkick | *bounces head off desk a few times* |
| 10:44.02 | swiftkick | now, with no changes to extensions.conf whatsoever, it works. |
| 10:44.19 | SwK | go figure.. maybe somethig didnt reload correctly |
| 10:44.40 | swiftkick | what does "reload" from the CLI *not* do that a cold restart of asterisk via the init.d script, does do? |
| 10:45.02 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:45.12 | tzafrir_laptop | many things |
| 10:45.14 | SwK | reload doesnt drop all your calls ;P |
| 10:45.36 | swiftkick | i noticed this behavior a few iterations back |
| 10:45.46 | swiftkick | that just after restarting asterisk, it seemed to function for (1) call |
| 10:45.53 | swiftkick | then upon reloading the config from the CLI, the error reappeared |
| 10:46.02 | swiftkick | but now, it is remaining functional. (!?) |
| 10:46.13 | swiftkick | its intermittancies like this that can really drive ya nuts, ya know? |
| 10:46.50 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:46.52 | swiftkick | this is a stock asterisk-gui install, very simple, modified to accomodate multiple inbound trunks |
| 10:47.01 | swiftkick | e.g. multi-tenant |
| 10:47.26 | swiftkick | and one trunk serves like 20 phone numbers |
| 10:47.46 | swiftkick | hmmmm this is very strange |
| 10:48.17 | swiftkick | but it works, despite only taking hours and ending inconclusively in greater confusion than i started. :) |
| 10:48.57 | swiftkick | as usual, i appreciate the feedback from everyone who has taken the time to reply, you guys are certainly expert |
| 10:50.44 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:51.47 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:51.58 | dominic1 | if anybody is using func_devstate: http://www.voip-info.org/wiki/index.php?page_id=4398&tk=8f117fe29290a25f8e4b&comments_page=1 |
| 10:53.43 | *** join/#asterisk tyldis (n=vidar@tyldum.com) |
| 11:00.55 | *** join/#asterisk blackhole (n=blackhol@unaffiliated/blackhole) |
| 11:01.36 | blackhole | Hi, If i want to deploy asterisk as VOIP server what i need. What i want is to deploy asterisk and a2billing for making national and international calls. Do i need any T1 or E1 line etc? |
| 11:02.38 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 11:03.37 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:04.10 | ibnolqaiyem | DarKnesS_WolF, http://www.voip-info.org/wiki/index.php?page=Asterisk+config+indications.conf |
| 11:09.18 | MaliutaLap | blackhole: maybe, depends on what option you take. You'll also need things like a computer, a source of electricity(lemons and copper work, or so I hear) network cables, oxygen .... the list is quite long |
| 11:09.33 | *** join/#asterisk metastable (n=superfly@212.71.24.170.res.static.edpnet.net) |
| 11:10.34 | metastable | hey all |
| 11:10.40 | blackhole | MaliutaLap, he he but what i want to know can i make asterisk work for VOIP calls. If i don't have T1 or E1 line is it possible to make VOIP calls as a2billing do? |
| 11:10.46 | metastable | i've been rtfm and didn't find an answer |
| 11:11.06 | metastable | my customer has 2 isdn lines, will be being the B410P |
| 11:11.36 | metastable | the question is this: can he use voip phones, that transfer over the isdn connection ? |
| 11:11.56 | metastable | i'm guessing this is like shooting the cat out of the water, but just trying to make sure :) |
| 11:12.22 | MaliutaLap | blackhole: a) that's not what you asked. b) are you running data only over the E/T1? (not VoIP if you don't) c) we don't support a2billanything |
| 11:13.40 | MaliutaLap | metastable: yes, read the book. it's just using channels on the B410P rather than the TMD400P in the example |
| 11:14.17 | metastable | maliutalap: thank you |
| 11:14.36 | MaliutaLap | metastable: BTW ... ~thebook |
| 11:14.43 | blackhole | MaliutaLap, I am just a novoice in asterisk. All i see is one of my friend company has something called a2billing which uses asterisk server. They can login to website and make a callback by typing his phone number and number he wishes to call. It would place a call to his number and then he will hear a voice saying your call is being connected and he will be connected to the other number.. |
| 11:15.02 | MaliutaLap | ~thebook |
| 11:15.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 11:15.16 | blackhole | I read those books |
| 11:15.22 | blackhole | but i can't find my answer.. |
| 11:15.25 | blackhole | thats why i am here |
| 11:15.34 | blackhole | MaliutaLap, What i want to know is what i need to set up almost same kind of setup.. |
| 11:15.44 | metastable | maliutalap: thanks ! |
| 11:15.44 | MaliutaLap | blackhole: we do not support a2billing. you want to use it go elsewhere |
| 11:15.49 | metastable | and good job on asterisk, guys |
| 11:15.58 | metastable | this is a great step forward :) |
| 11:16.36 | blackhole | MaliutaLap, Okay its fine if you don't support a2billing but you can give me idea what i need to have same kind of setup. Do i need some phone line or etc? |
| 11:16.43 | mrnick | anybody knows why my localnet= stops my voiptermination to be reachable? |
| 11:17.13 | MaliutaLap | blackhole: do you just want to do VoIP (* can do more than just VoIP) |
| 11:17.35 | MaliutaLap | mrnick: it's likely to be more than just a localnet declaration |
| 11:17.55 | MaliutaLap | mrnick: probably got something to do with a nat delcaration |
| 11:18.14 | MaliutaLap | mrnick: without INFORMATION we can only guess |
| 11:18.23 | mrnick | once i disable it they become reachable, but that is no proof of course |
| 11:18.35 | blackhole | MaliutaLap, Say i have some script where i provide my number and number i wish to call. The call would be placed, i will get a call 1st and then the other user.. |
| 11:18.47 | blackhole | MaliutaLap, This is what i only want from asterisk as of now.. |
| 11:18.57 | MaliutaLap | blackhole: depends on how you configure * |
| 11:19.16 | mrnick | i tried to pastbin some debug here yesterday but could find the informative debug |
| 11:19.17 | MaliutaLap | blackhole: so learn to either write dialplans or AGI's |
| 11:19.20 | blackhole | MaliutaLap, Do i need any phone line or something! |
| 11:19.42 | MaliutaLap | blackhole: maybe, you aren't telling us much about the situation |
| 11:20.23 | MaliutaLap | blackhole: I think you need to go do some researh in to general PBX concetps |
| 11:20.31 | blackhole | MaliutaLap, I have my mobile number and say my friend mobile number. I want that i should get a call on my mobile saying your call is being connected and then it should get connected to my friend mobile number |
| 11:20.35 | MaliutaLap | s/concetps/concepts/ |
| 11:21.14 | MaliutaLap | blackhole: it's possible, you'd need to have an AGI and a properly conf'd diaplan |
| 11:21.20 | SwK | if anyone is in chicago Aug 5 - 7 ccheck out ClueCon ( www.cluecon.com ) Mr John Todd of Digium will be there giving a talk, and several other projects and companies will be there too like Craig Southeren of OpenH323 and OPAL fame </spam> (time for sleep now) |
| 11:21.43 | *** join/#asterisk oilinki (n=oil@ppp-124-120-7-108.revip2.asianet.co.th) |
| 11:21.44 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:21.50 | MaliutaLap | SwK: I think you need to hit some people with a cluecon |
| 11:22.04 | blackhole | MaliutaLap, and not anything like some phoneline? |
| 11:22.10 | SwK | MaliutaLap, or a clue-bat |
| 11:23.49 | MaliutaLap | SwK: or tt-monkeys |
| 11:24.08 | MaliutaLap | blackhole: that depends on HOW YOU WANT TO DO IT |
| 11:24.11 | mrnick | http://pastebin.com/m14cc007e |
| 11:24.32 | mrnick | there's the problem, somewhere in there... |
| 11:24.39 | mrnick | it's my sip.conf |
| 11:25.35 | MaliutaLap | mrnick: nat=yes |
| 11:26.18 | mrnick | i'm sorry, which one? |
| 11:26.19 | MaliutaLap | mrnick: type=friend |
| 11:26.28 | *** join/#asterisk BBHoss_ (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 11:26.29 | mrnick | the general? or the voipcheap |
| 11:26.39 | dominic1 | anybody here with knowledge of iaxmodem and hylafax? |
| 11:27.35 | SwK | it works |
| 11:27.51 | SwK | dominic1, what seems to be the problem with it |
| 11:28.21 | MaliutaLap | mrnick: http://pastebin.com/d7059e59a |
| 11:28.29 | dominic1 | if I get inbound faxes the first time the iaxmodems are always busy |
| 11:28.51 | dominic1 | on the second or the third recall everything is okay |
| 11:29.04 | dominic1 | is asterisk answering to quick |
| 11:29.04 | dominic1 | ? |
| 11:29.09 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-dcae35ccbd96ca07) |
| 11:29.11 | dominic1 | should I add a wait? |
| 11:29.27 | SwK | asterisk shouldnt answer the calls iaxmodem should |
| 11:29.50 | MaliutaLap | generally you put a wait after an answer so packets can catch up |
| 11:30.04 | SwK | if you have an answer() in your dialplan before sending the calls over to iaxmodem take it out |
| 11:30.15 | tyldis | I've got a problem with Asterisk on a server which has 2 IP-addressess on the same interface with different subnets. Default GW is on same subnet as IP#1 which makes Asterisk always use IP#1 even though the request is coming om IP#2. This setup seems to work nicely with other daemons, like OpenSSH |
| 11:30.23 | MikeJ | stabs sip |
| 11:30.26 | *** join/#asterisk udzinari (n=david@6-164.cdn.ge) |
| 11:30.29 | SwK | hqh |
| 11:32.02 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 11:32.22 | dominic1 | I added an answer, nothin changed. The entries for the iax peers are in the datease and registered. |
| 11:32.36 | MaliutaLap | has had enough of people that need to be attacked by the M.M.O.D.(tm) for once 36hr period with no sleep |
| 11:32.57 | dominic1 | okay, I war wrong, the modems are always unregistering. |
| 11:32.59 | MaliutaLap | MMOD(tm) == Minion Monkeys of Doooooooom(tm) |
| 11:33.06 | dominic1 | was |
| 11:35.24 | styelz | domnic1: check the iaxmodem logs |
| 11:37.11 | dominic1 | no errors found in the logs |
| 11:37.29 | dominic1 | it isn't telling me about the registration to asterisk. Can I adjust the loglevel? |
| 11:38.19 | mrnick | <MaliutaLap> it's not working, but it still does if i disable the localnet= |
| 11:38.30 | dominic1 | should I set the communication to static, without registration the modems? |
| 11:38.52 | mrnick | <PROTECTED> |
| 11:39.17 | styelz | dominic1: try core set debug 4 . or something |
| 11:39.22 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 11:39.29 | styelz | hey joobie |
| 11:39.46 | joobie | y0 styelz |
| 11:39.47 | joobie | sup |
| 11:40.26 | mrnick | http://pastebin.com/d7d5b780c |
| 11:40.57 | mrnick | i hope this isn't to confusing |
| 11:43.51 | *** part/#asterisk pkunkra (n=chris@cpe-74-73-8-115.nyc.res.rr.com) |
| 11:46.40 | mrnick | anybody knows how to solve my localnet/nat problem |
| 11:48.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:50.54 | bboschman | Hi |
| 11:51.59 | bboschman | anyone has an example for apache active/active and only the IP does a failover (if apache has died / or maintainance) |
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| 11:58.09 | mrnick | good news everybody, I found the solution! |
| 11:58.40 | mrnick | i just had to specify localnet for all my connections wheter they or in or out of my nat |
| 11:59.04 | mrnick | thanks <MaliutaLap> and others for your help |
| 11:59.08 | styelz | professor farnsworth: great |
| 12:00.17 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
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| 12:01.07 | DarKnesS_WolF | ibnolqaiyem: i already know it and i already added the country i wants this is not the answer for my question but thx anyway |
| 12:01.31 | DarKnesS_WolF | SwK: sorry dude had some power problems |
| 12:05.25 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:05.50 | mrnick | does anybody know of a working tutorial about transfer i've read voip-info asteriskguru, the whole google index :-) but couldn't find one that works for me |
| 12:06.15 | [TK]D-Fender | mrn"transfer". What "transfer"? |
| 12:06.21 | DarKnesS_WolF | mrnick: what u want to do exactly |
| 12:06.23 | [TK]D-Fender | mrnick: rather |
| 12:06.26 | DarKnesS_WolF | ah [TK]D-Fender still up :-)? |
| 12:06.43 | *** join/#asterisk freddyk (n=freddy@host131-140-dynamic.42-79-r.retail.telecomitalia.it) |
| 12:06.44 | [TK]D-Fender | DarKnesS_WolF: JSUT up actually. |
| 12:06.52 | [TK]D-Fender | just* |
| 12:06.54 | [TK]D-Fender | asdhfdajdhaklfdg |
| 12:07.01 | DarKnesS_WolF | lol |
| 12:07.10 | DarKnesS_WolF | [TK]D-Fender: i can see :-) it is like 7:10 there |
| 12:07.24 | [TK]D-Fender | DarKnesS_WolF: 8:10am actually |
| 12:08.09 | DarKnesS_WolF | mmmm [TK]D-Fender i don't know i get the US clock time from x86 chatty machine :-) |
| 12:08.42 | DarKnesS_WolF | [TK]D-Fender: tried TLS with 1.6 yet ? |
| 12:08.48 | *** join/#asterisk angryuser (n=sldf@78.115.10.40) |
| 12:08.50 | [TK]D-Fender | DarKnesS_WolF: There are several timezones on this continent yuo know.. and I'm not in the US :) |
| 12:08.54 | [TK]D-Fender | DarKnesS_WolF: nope. |
| 12:09.12 | DarKnesS_WolF | [TK]D-Fender: really ? i thought ur in US .. mmm canada ? |
| 12:09.22 | mrnick | well i'm dial a pstn number and after some talk i want to connect the call to an ivr on my asterisk system |
| 12:09.24 | [TK]D-Fender | DarKnesS_WolF: #2 gets it... |
| 12:09.40 | [TK]D-Fender | mrnOk, what phone are you using? |
| 12:09.42 | mrnick | so i figured transfer is the best way |
| 12:10.02 | mrnick | but my softphone isn't responding to # |
| 12:10.09 | mrnick | or #700 or #1 |
| 12:10.15 | [TK]D-Fender | mrnick: what PHONE? |
| 12:10.49 | DarKnesS_WolF | seems to be a softphone |
| 12:10.56 | mrnick | I try it again but more technically, i don't have the right vocabulary though |
| 12:10.57 | DarKnesS_WolF | mrnick: dose it support transfer ? |
| 12:11.13 | [TK]D-Fender | DarKnesS_WolF: Yes, I got that. I want to here which one. |
| 12:11.14 | mrnick | can't i just press # |
| 12:11.24 | mrnick | isn't that what it says in the tutorials? |
| 12:11.29 | [TK]D-Fender | mrnick: What softphone are you using? |
| 12:11.44 | mrnick | ekiga xlite openwengo |
| 12:11.49 | [TK]D-Fender | mrnick: "core show application dial" <- Go read Dial's instruction THOROUGHLY |
| 12:11.53 | mrnick | depends from where i try it |
| 12:12.40 | mrnick | thx i'll start the reading |
| 12:12.49 | angryuser | TK , Tcl/TK source forge project ? |
| 12:15.42 | DarKnesS_WolF | mrnick: ur talking about blind transfer do u have res_features.so loaded ? |
| 12:15.51 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 12:15.53 | DarKnesS_WolF | and make sure u have blindtransfer enabled in features.conf |
| 12:18.08 | mrnick | i'll check it in a minute my remote pc is temporarily down |
| 12:21.34 | dominic1 | I have the problem that the reregister of my iaxmodems is not working |
| 12:21.56 | phix | pwned |
| 12:22.12 | phix | you may as well throw your corps off a cliff |
| 12:22.21 | dominic1 | did alreasy start debugging, but I get no errors |
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| 12:27.54 | DarKnesS_WolF | dominic1: iaxmodems registration should be easy u create accounts in ur iax.conf and then u do teh accounts in teh iaxmode.conf or wahtever the modem name will be and then u run iaxmode |
| 12:29.20 | freddyk | anyone knows who is actually developing chan_skinny ??? |
| 12:30.14 | codestr0m | freddyk: you can't find someone's name in the changelog for the source? |
| 12:30.22 | mrnick | load => res_features.so is that what i have to ad to modules.conf |
| 12:30.23 | mrnick | ? |
| 12:30.46 | codestr0m | also I'm pretty sure there's more than one chan_skinny project/effort so make sure you're referring to one that works currently |
| 12:31.04 | [TK]D-Fender | mrnick: If you started from the sample configs you shouldn't have to do anything. |
| 12:31.20 | [TK]D-Fender | mrnick: Thats one of the advisable ones to use "as-is" |
| 12:31.48 | mrnick | it was still in the sample config |
| 12:32.00 | mrnick | so i'll remove it again and restart * |
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| 12:36.46 | styelz | dominic1: i remember having a similar issue whith iaxmodem. then realised i had not edited tty's to include faxgetty for ttyIAX1 |
| 12:38.22 | mrnick | it used to say transfer followed by something like "unable to find ..." |
| 12:38.34 | mrnick | but now it says nothing when i try to transfer |
| 12:38.59 | [TK]D-Fender | mrnick: PASTEBIN the CLI outpt of your call from beginning to end at verbose 10 |
| 12:42.15 | mrnick | http://pastebin.com/d50504049 |
| 12:42.27 | mrnick | i hope that's what you are looking for |
| 12:43.31 | [TK]D-Fender | NOTHING? |
| 12:43.53 | tzanger | [TK]D-Fender: well I hope your happy. :-) |
| 12:45.48 | mrnick | http://pastebin.com/d79248707 |
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| 12:48.12 | dominic1 | I found out what exactly my problem is: (started iaxmodem in foregroud). After starting iaxmodem I get the registration, then I get the message Taking receiver off-hook. |
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| 12:48.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:48.18 | [TK]D-Fender | mrnick: Make sure you have set the right "dtmfmode for your phone |
| 12:48.32 | [TK]D-Fender | mrnShould be "rfc2833" in most cases |
| 12:48.38 | mrnick | that's a new concept for me, i'll have to read about it |
| 12:48.46 | dominic1 | after set I have to dial a few times, then I get a register from iaxmodem, then I am able to dial again |
| 12:50.03 | dominic1 | I have to dial exactly 2 times |
| 12:50.05 | [TK]D-Fender | mrnick: this is for your sip peer. do "dtmfmode=rfc2833" and do a "reload" and try again |
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| 12:58.54 | ibnolqaiyem | is it bad to replace tdm card with an ATA linksys? |
| 12:59.10 | mrnick | doesn't work |
| 12:59.55 | mrnick | i've got to go, thanks for the help! |
| 13:02.24 | [TK]D-Fender | ibnolqaiyem: for FXS I advise ATA's over PCI solutions. |
| 13:04.00 | tzafrir_laptop | what card is it? FXS or FXO? |
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| 13:05.23 | [TK]D-Fender | tzafrir_laptop: Last we spoke he was looking for FXS and he hadn't bought anything yet |
| 13:06.13 | tzafrir_laptop | well, if you have a card, why not use it? |
| 13:06.56 | tzafrir_laptop | There's also the difference between what you have and what you might have in a week. Or maybe. |
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| 13:08.17 | dominic1 | okay, geht iaxmodem only to work if I set host and port statically |
| 13:08.54 | dominic1 | the I get error that the user is not dynamic, but the connects seem to be okay and it works |
| 13:09.04 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 13:09.38 | dominic1 | any ideas what I can do against Peer 'iaxmodem2' is not dynamic without setting it dynamic? |
| 13:10.18 | [TK]D-Fender | dominic1: Fixed port, dynamic host. |
| 13:10.19 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 13:12.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:13.05 | dominic1 | I am not able to specify the port, the database is overwritten if I set host=dynamic in my realtimedatabase |
| 13:14.12 | mikeshank | [TK]D-Fender: I understand if you don't want to deal with me after I drove you nuts yesterday. I put MaliutaLap thru the ringer also, he walked me thru getting the nat stuff set up correctly. So, back to my original issue which is the calling phone gives no ringtone, here's my sip.conf, extensions.conf and sip debug of a call http://pastebin.com/d30efa42a |
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| 13:18.15 | [TK]D-Fender | mikeshank: set "nat=no" for your BV peer, and "canreinvite=no" is every one. |
| 13:18.40 | tzafrir_laptop | dominic1, I think you need to set nat=no as well |
| 13:18.43 | [TK]D-Fender | mikeshank: And do "set versboe 10" |
| 13:18.47 | [TK]D-Fender | verbose* |
| 13:19.52 | tzafrir_laptop | With nat=yes, the port of the registration is used for sending, IIRC |
| 13:20.26 | tzafrir_laptop | (and fankly, I'm not sure IRC) |
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| 13:27.34 | mikeshank | [TK]D-Fender: ok, made those changes http://pastebin.com/d4439c589 |
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| 13:29.13 | [TK]D-Fender | mikeshank: exten => ${BUSINESS_NUMBER},2,Dial(${SAM},,rR) <- remover the ",,Rr" |
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| 13:33.45 | [TK]D-Fender | mikeshank: And you should replace that with ",20" or sanother reasonable timeout. You probably shouldn't be ringing that device indevinitely. |
| 13:33.51 | [TK]D-Fender | indefinitely* |
| 13:37.26 | mikeshank | [TK]D-Fender: ok, http://pastebin.com/d42cc868d |
| 13:38.20 | [TK]D-Fender | mikeshank: So the BV caller still doesn't hear a ring while SAM is ringing? |
| 13:38.36 | mikeshank | [TK]D-Fender: that's correct |
| 13:38.50 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:39.22 | [TK]D-Fender | mikeshank: Ah... common possbility. Do you have an indications.conf ? If not, copy it over from the samples folder in your source tarball |
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| 13:41.17 | mikeshank | [TK]D-Fender: yes, i have it open |
| 13:41.44 | [TK]D-Fender | is it in your /etc/asterisk folder? |
| 13:41.49 | mikeshank | yes |
| 13:43.45 | [TK]D-Fender | mikeshank: right after answer, try adding a Playback(silence/1) and see if that helps |
| 13:46.18 | mikeshank | [TK]D-Fender: yes, that worked |
| 13:47.09 | [TK]D-Fender | mikeshank: Great |
| 13:47.48 | mikeshank | [TK]D-Fender: yes, thanks. whats the reason for the Playback(silence/1) |
| 13:49.06 | [TK]D-Fender | mikeshank: I'm guessing that as you did an "answer" the call should not report "ringing", but audio was not separately initiated so the Dial coming right away didn't have anywhere to go |
| 13:49.32 | [TK]D-Fender | (something like that) |
| 13:50.28 | mikeshank | [TK]D-Fender: again thanks alot |
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| 14:25.47 | hunmonk | is anybody aware of problems with Realtime voicemail and ODBC voicemail storage working together? using a voicemail box in voicemail.conf work fine, but when i move it to Realtime voicemail, it's not storage the message in the database.... |
| 14:29.37 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 14:30.12 | Dr-Linux|home | anybody is using A2billing? |
| 14:32.01 | MaliutaLap | not supported here. move along |
| 14:32.11 | MaliutaLap | ~topic |
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| 14:38.41 | *** join/#asterisk dr_gogeta86 (n=dr_goget@81-208-88-100.ip.fastwebnet.it) |
| 14:38.53 | dr_gogeta86 | hi to all |
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| 14:40.00 | MaliutaLap | there is no help here, read the book, read the wiki t'as'all |
| 14:40.27 | [TK]D-Fender | MaliutaLap: unload chan_pessimist.so |
| 14:41.00 | MaliutaLap | [TK]D-Fender: only if you can load chan_sleep.so for me |
| 14:41.44 | Dr-Linux|home | MaliutaLap: what not supported here? |
| 14:42.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:43.08 | MaliutaLap | [TK]D-Fender: and those 2 things would fix 40% of what we see, 30% would be fixed by people getting a clue about general tech stuff (networking etc.), 15% by people READING THE TOPIC and the other 15% are genuine |
| 14:43.40 | MaliutaLap | Dr-Linux|home: READ THE FECKING TOPIC ... does this _look_ like #a2billing? |
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| 14:50.18 | ManxPower | At least it's not a GUI |
| 14:50.21 | tclineks | any advice on the best python agi implementation |
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| 14:50.59 | *** join/#asterisk zxd (n=zapw@213.31.43.2) |
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| 14:51.13 | zxd | how do i check the latencies of all the connected sip servers? |
| 14:51.35 | ManxPower | zxd: there are many kinds of latency |
| 14:51.56 | zxd | [Jul 24 17:50:57] WARNING[1202]: chan_sip.c:2923 create_addr: No such host: voicepulse |
| 14:51.56 | zxd | [Jul 24 17:50:57] WARNING[1202]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 14:52.01 | zxd | this dosen't look good |
| 14:52.09 | zxd | ManxPower, what kinds? |
| 14:52.09 | ManxPower | I assume you mean "latency for the device to respond to a SIP OPTIONS packet", if that is the case then qualify= will help. |
| 14:52.22 | ManxPower | zxd: you do not have a latency problem, you have a config problem. |
| 14:52.40 | Dr-Linux|home | MaliutaLap: what is fecking? well A2billing is something that works with Asterisk .. otherwise that's is nothing |
| 14:52.40 | ManxPower | It thinks you are supposed to be connecting to the machine "voicepulse" and we all know that's not a valid DNS name. |
| 14:53.07 | jeyk | This may not be the right channel to ask this... but are there services out there that provide local US phone numbers for my Asterisk PBX? What are they called / what should I google for? |
| 14:53.11 | ManxPower | That is the ACTUAL Dial line as shown in the CLI that causes that error. JUST ONE LINE |
| 14:53.20 | ManxPower | ~itsp |
| 14:53.21 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 14:53.24 | zxd | ManxPower, still how do i list all the servers asterisk is connected to ? |
| 14:53.39 | ManxPower | zxd: registered too, peers, or users? |
| 14:54.13 | ManxPower | "sip show registry" shows systems Asterisk is registered to, but that has NOTHING to do with outgoing calls from Asterisk. "sip show peers" will show remote systems Asterisk will connect to. |
| 14:54.41 | ManxPower | zxd: now, for the 2nd time, paste just the one dial line from the CLI that generates that error. I won't ask again. |
| 14:54.59 | [TK]D-Fender | zxd: pastebin your failed attempt CLI output at verbose 10, and include your sip.conf masking only passwords |
| 14:55.03 | Dr-Linux|home | I'm looking for a a opensource billing software that works with asterisk. My plan is, a caller dial in through PSTN and enter a desired number and and system fwd him to his desired number, now i want to billing for this |
| 14:55.23 | jeyk | ManxPower: ok, thanks |
| 14:55.25 | ManxPower | Dr-Linux|home: you won't find much help on that here. |
| 14:55.26 | [TK]D-Fender | Dr-Linux|home: So whats your actual question? |
| 14:55.47 | jeyk | ~itsplist-us |
| 14:55.47 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 14:55.49 | Dr-Linux|home | therefore i went through A2billing but i'm not sure if that fits in my case |
| 14:56.51 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:56.59 | Dr-Linux|home | the calling and called both are PSTN users |
| 14:57.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:57.26 | [TK]D-Fender | Dr-Linux|home: You should know better. what device the call comes in through has no impact |
| 14:57.42 | Dr-Linux|home | so my question is if someone is using A2billing already he can tell me if this program can help me or not |
| 14:58.04 | ManxPower | Dr-Linux|home: the A2 billing people didn't have any clue about their own software? |
| 14:58.35 | [TK]D-Fender | Dr-Linux|home: Its a ^#%$ blling add-on, WTF do you think its for? Yes, hundred of people use it. Go visit their site, read their docs, download and TRY IT |
| 14:59.06 | ManxPower | I guess zxd didn't really want any help. |
| 14:59.19 | [TK]D-Fender | ManxPower: just slow. |
| 14:59.28 | Dr-Linux|home | ManxPower: I tried to find on the web, but i suspect A2billing only helps if the customers/callers SIP or IAX but my callers are come throught PSTN/PRI |
| 14:59.29 | zxd | ManxPower, wait |
| 14:59.32 | zxd | ManxPower, i wasn't here |
| 14:59.36 | ManxPower | [TK]D-Fender: almost as bad 8-| |
| 14:59.44 | ManxPower | zxd: why were you not here? |
| 14:59.53 | zxd | ManxPower, i know nothing about asterisk syntax , the admin that was responsible for asterisk is abroad |
| 14:59.58 | zxd | and i getting this errors |
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| 15:00.20 | ManxPower | zxd: your next line should contain the information I asked for, |
| 15:00.30 | zxd | and users say they are saying on the other end customers are hearing poor sound quality or not hearing at all |
| 15:00.43 | zxd | sec |
| 15:00.46 | [TK]D-Fender | zxd: that is now a completely different problem |
| 15:00.52 | zxd | yea |
| 15:00.54 | ManxPower | zxd: No! STOP!!! We are working on one specific problem. Stop bringing up other problems that confuse the issue. |
| 15:01.15 | zxd | so i prefer working on the second problem instead |
| 15:01.17 | ManxPower | zxd: you will NOT be able to just learn to fix asterisk problems in a day or two. |
| 15:01.19 | Dr-Linux|home | [TK]D-Fender: yeah there are many, that's why i'm trying to find someone here who uses A2billing |
| 15:01.32 | zxd | ManxPower, you don't have to tell me that |
| 15:01.37 | ManxPower | zxd: best of luck. I cannot help you further. |
| 15:01.38 | [TK]D-Fender | Dr-Linux|home: Just get off your ass and TRY IT. |
| 15:01.41 | zxd | ManxPower, but maybe there are a few things i can check |
| 15:02.10 | ManxPower | maybe [TK]D-Fender or someone else with more patience today can help you. |
| 15:02.34 | [TK]D-Fender | ManxPower: Not sure how to prioritize taht last remark :) |
| 15:02.45 | *** join/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
| 15:03.00 | nny_2 | is REFER Failed in SIP SHOW CHANNELS anything to be concerned about? |
| 15:03.25 | [TK]D-Fender | nny_2: Reinvite failure |
| 15:03.36 | ManxPower | [TK]D-Fender: he pissed me off by switching questions mid-stream. |
| 15:03.36 | [TK]D-Fender | nny_2: Watch out for NAT's |
| 15:03.56 | nny_2 | Hmm, all on the same network, i think this is the sidecar's subscribe |
| 15:03.59 | ManxPower | [TK]D-Fender: So I sent him to you 8-) |
| 15:04.13 | nny_2 | it is persistent and all other channels seem to move with the traffic flow in the network |
| 15:04.14 | [TK]D-Fender | ManxPower: I understand. Its jsut a question of whether I'm another source of help with more patience, or just another source of help and that everyone ELSE is more patient :) |
| 15:04.42 | [TK]D-Fender | nny_2: PB up some detail if you'd like a closer look. |
| 15:04.48 | ManxPower | [TK]D-Fender: the patience remark was sarcasm. *tease* |
| 15:05.07 | [TK]D-Fender | ManxPower: Just wasn't sure of its direction.... and still not any more directed either |
| 15:05.17 | nny_2 | hmm I can do some sip debugging, i'll see if anything interesting pops up for that peer |
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| 15:08.24 | nny_2 | is there a way to debug by CALL ID? |
| 15:08.46 | nny_2 | under the sip show channels, that is a field |
| 15:10.23 | ManxPower | you can do it by peer and by ip |
| 15:10.37 | [TK]D-Fender | nny_2: No, because its generated. you can't see it coming |
| 15:10.54 | [TK]D-Fender | nny_2: You said its from a specific phoen, so debug the peer |
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| 15:22.51 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 15:23.53 | saunterer | I've been recieving calls on all my phones twice every hour, the caller id says "asterisk"... can anyone give me a starting point to finding a solution to this problem? |
| 15:24.23 | [TK]D-Fender | saunterer: What is the call arriving on? Does it show that ALL the time from that source? |
| 15:25.27 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 15:25.49 | Qwell | just won £891,934.00 |
| 15:26.46 | many | gimme 1000 prettypls. |
| 15:30.04 | redax | hi, |
| 15:30.18 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 15:32.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:33.51 | tzanger | hmm |
| 15:33.58 | tzanger | is it possibel to turn OFF the zaptel timer test when asterisk starts? |
| 15:36.00 | saunterer | Fender, next time it calls, I'll check |
| 15:38.09 | pbxfan | \bye |
| 15:38.11 | pbxfan | oops |
| 15:40.38 | dominic1 | anybody using iaxmodem with realtime? |
| 15:40.59 | ManxPower | saunterer: the problem is with your analog lines, you should be able to use WaitForRing or look in the mailinglist archives |
| 15:41.01 | ManxPower | ~mailinglist |
| 15:41.01 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 15:41.11 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 15:41.55 | angler | saunterer, when you pick up the phone is it an actual person? |
| 15:43.11 | saunterer | no |
| 15:43.41 | ManxPower | Telco or line noise is triggering asterisk to think a call is coming in. |
| 15:43.51 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:43.57 | redax | can I make a ringback to transferer at no answer, if the transferer started to use attended transfer, but he wont wait until the extension answers and he hit the <Transfer> button again on her SIP phone... |
| 15:43.57 | saunterer | it happens on specific times though |
| 15:44.05 | Qwell | $20 on cronjob |
| 15:44.10 | saunterer | er at specific times |
| 15:44.20 | redax | will that look like a blindtransfer? (ie. do I have BLINDTRANSFER variable set?) |
| 15:44.37 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:44.49 | ManxPower | redax: depends on the phone, usually it would be a blnd transfer |
| 15:45.10 | ManxPower | saunterer: That does not change my advice |
| 15:45.15 | redax | it's a linksys spa9xx ... actually 922 |
| 15:45.20 | angler | saunterer, these calls are coming over a POTS line? |
| 15:45.32 | ManxPower | redax: try it and see |
| 15:45.59 | redax | ManxPower: I'm afraid I don't have BLINDTRANSFER variable set in that case... just when user hit the blindtransfer button... |
| 15:46.13 | redax | ManxPower: can I workaround this somehow? |
| 15:46.18 | ManxPower | redax: no. |
| 15:46.43 | ManxPower | Yell, yes you can. Switch to Polycom. |
| 15:46.50 | ManxPower | even then you might not be able to. |
| 15:46.50 | redax | ManxPower: there's something about Transfer context or whatever.. |
| 15:47.14 | ManxPower | redax: you cannot magically convert a supervised transfer into a blind transfer within Asterisk |
| 15:47.16 | redax | ManxPower: :) we have 5 polycom ip550 and 5 spa922 ;-( |
| 15:47.38 | ManxPower | redax: then if you can do it in a Polycom it would be in the Admin Guide. |
| 15:48.01 | ManxPower | Most people that want to do this are confused. |
| 15:48.02 | saunterer | @angler, we have DSL and the line is split: it goes into the DSL and also our asterisk server |
| 15:48.11 | Qwell | eww |
| 15:48.23 | ManxPower | saunterer: Well best of luck with that. I cannot help you further. |
| 15:48.31 | redax | ManxPower: anyhow... am I know at hangup time it was either an attended transfer, or a normal call ? |
| 15:48.35 | ManxPower | Qwell: it's a pretty common setup. |
| 15:48.40 | Qwell | scary |
| 15:49.03 | ManxPower | redax: you would have to figure it out based in the CDRs after the fact. |
| 15:49.45 | ManxPower | redax: but the real question is "Why do you care?" |
| 15:53.39 | angler | saunterer, what PCI card are you using to bring the line into the Asterisk system? Also what version of Zaptel? |
| 15:54.09 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 16:00.52 | redax | ManxPower: I want to ring back a transfered call to the trensferer after X seconds of ring.... |
| 16:01.08 | redax | ManxPower: in case of blindtransfer I can do it.. |
| 16:01.35 | redax | ManxPower: when user using the <lame_way/> blindtransfer, I can't |
| 16:06.18 | neurosys | Hmm. I'm using Set(CALLERID(all)=test <3050001234>) but the name continues to read "Out of Area", but the number does go through. Any thoughts |
| 16:06.23 | [TK]D-Fender | redax: Set a channel variable copying that channel name at the start of the call. It should still be around when the hand-off happens |
| 16:06.37 | ManxPower | neurosys: "goes thru" to WHERE? |
| 16:06.54 | [TK]D-Fender | neurosys: Maybe they don't let you set the name |
| 16:06.56 | neurosys | ManxPower: to the outside telephone caller ID |
| 16:07.08 | neurosys | [TK]D-Fender: Normal POTS allows it |
| 16:07.08 | ManxPower | neurosys: you can't set the callerid name on the PSTN |
| 16:07.16 | ManxPower | neurosys: NO IT DOES NOT |
| 16:07.18 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 16:07.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:07.45 | neurosys | ManxPower: Only the #? |
| 16:07.47 | ManxPower | Telcos ignore the CLID Name and replace it with whatever name is listed for that NUMBER in the telco databases |
| 16:08.09 | ManxPower | And you can only send CID Number when you have a PRI (or ss7) link to the telco. |
| 16:08.15 | *** join/#asterisk tobias (n=tobias@cpe-069-134-204-037.nc.res.rr.com) |
| 16:08.26 | neurosys | ManxPower: well.. the number portion works now. |
| 16:08.37 | ManxPower | there you go |
| 16:08.41 | coppice | there are places where an analogue line can send CLI :-) |
| 16:08.42 | neurosys | ManxPower: thru an ITSP |
| 16:08.56 | ManxPower | an ITSP would be using PRI or SS7 |
| 16:09.47 | neurosys | ManxPower: so it will run the # thru their DB and if it finds it it will display the related name to that Number? |
| 16:09.51 | *** join/#asterisk oej (n=olle@81-224-166-188-o1036.telia.com) |
| 16:11.16 | [TK]D-Fender | neurosys: ITSP != POTS |
| 16:11.24 | ManxPower | neurosys: the terminating telco does that |
| 16:11.25 | neurosys | [TK]D-Fender: I know :) |
| 16:11.25 | [TK]D-Fender | neurosys: ITSP = PSTN |
| 16:14.02 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:17.59 | hunmonk | i've got a problem i'm unable to solve, and i'm willing to pay somebody to help me solve it. if anybody is interested, lemme know and we can take it to pm |
| 16:18.17 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:19.02 | neurosys | hmm the telco still registers it as out of area. |
| 16:19.08 | *** join/#asterisk moy (n=moy@nat/ibm/x-e7e0c5cfd4fe857f) |
| 16:19.44 | [TK]D-Fender | neurosys: Perhaps you should look at the number you're sending... |
| 16:19.51 | [TK]D-Fender | neurosys: Not exactly vlid. |
| 16:20.24 | neurosys | [TK]D-Fender: Im sending the businesses main number. When i call standard.. it displays the calledid name and # |
| 16:20.35 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:20.43 | neurosys | [TK]D-Fender: If i mimic the number thru asterisk .. the number passes... but name says out of area |
| 16:20.57 | [TK]D-Fender | neurosys: Less talk, more pastebin |
| 16:21.07 | neurosys | [TK]D-Fender: :) |
| 16:21.28 | freddyk | has anyone heard about Cisco firmware localization for SIP ? |
| 16:21.52 | neurosys | [TK]D-Fender: How bout I spare you for a bit... Ill go eat lunch.. come back and play more.. and give you an update |
| 16:22.31 | [TK]D-Fender | neurosys: Not sparing, merely postponing. Come back when you actually did all the legwork on this. |
| 16:22.52 | [TK]D-Fender | neurosys: You should most likely be slapped for having wasted all this time without evidence in hand. |
| 16:23.06 | neurosys | [TK]D-Fender: Perhaps ... |
| 16:24.15 | hunmonk | maybe i should be more specific: i'll pay $50 for somebody to help me get Realtime voicemail working with ODBC voicemail storage, $100 if it takes more than 30 minutes |
| 16:26.42 | ManxPower | hunmonk: Realtime is complicated |
| 16:27.02 | ManxPower | hunmonk: you know Digium does paid support, right? |
| 16:27.07 | *** join/#asterisk ZeroLux (n=none@96-25-30-144.ral.clearwire-dns.net) |
| 16:28.00 | hunmonk | ManxPower: well, i've got realtime extensions working. i've got odbc voicemail storage working with regular voicemail.conf. i just can't get the odbc vm to work w/ realtime vm |
| 16:28.24 | hunmonk | ManxPower: it's weird. as soon as i reconfig to realtime voicemail, messages won't save to the db |
| 16:28.59 | hunmonk | ManxPower: asterisk can find the realtime VM, i get the beep, it writes the files. just no db storage love when i hang up :( |
| 16:29.31 | hunmonk | perhaps i should call digium |
| 16:30.27 | ManxPower | hunmonk: I have never in my life used realtime, so talking to me about it is a waste of everyone's time. |
| 16:30.36 | hunmonk | shuts up |
| 16:35.12 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:37.23 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
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| 16:51.06 | *** mode/#asterisk [+o angler] by ChanServ |
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| 16:51.40 | neurosys | nickserv psypete |
| 16:51.45 | neurosys | oops |
| 16:52.43 | [TK]D-Fender | h4x |
| 16:55.12 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:55.27 | Katty | herro |
| 16:58.13 | [TK]D-Fender | Katty: Mew. |
| 16:58.20 | Katty | [TK]D-Fender: mew. |
| 16:59.18 | Katty | [TK]D-Fender: how're you? |
| 17:00.38 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 17:00.41 | [TK]D-Fender | Katty: Same old, same old. Watching the summer evaporate while the precipitation refuses to. |
| 17:00.59 | Katty | [TK]D-Fender: i see, i see. |
| 17:01.16 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 17:02.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:03.59 | *** join/#asterisk hardwire (n=hardwire@rdbk-2608.mtaonline.net) |
| 17:04.06 | hardwire | anybody have a few t100p's I can borrow? |
| 17:08.35 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 17:09.42 | tzafrir_laptop | What happens if a blind transfer is transferred to a busy extension? |
| 17:10.14 | [TK]D-Fender | tzafrir_laptop: the same as any other call to the same |
| 17:10.38 | tzafrir_laptop | Can I do anything before the call is disconnected? |
| 17:11.09 | tzafrir_laptop | Where exactly in the dialplan am I? |
| 17:11.12 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 17:13.41 | [TK]D-Fender | tzafrir_laptop: its just a call like any other. You're exactly where you are in the dialplan as any other call to that exten. |
| 17:15.16 | hardwire | tzafrir_laptop: you could use some AGI to handle it |
| 17:15.41 | tzafrir_laptop | what will an AGI do to help that a simple dialplan can't? |
| 17:15.45 | hardwire | or maybe some macros |
| 17:16.05 | hardwire | whatever you want. |
| 17:16.13 | hardwire | depends on the complexity of the situation :) |
| 17:16.14 | _ShrikE | tzafrir_laptop: you could lookup devstate after the dial and goto accordingly. |
| 17:16.24 | hardwire | what kind of "intercept" do you want? callback? |
| 17:17.41 | *** join/#asterisk mrnick (n=ubugo@88.197.232.204) |
| 17:17.58 | tzafrir_laptop | so if a transfer failes, I'm after the dial? |
| 17:19.42 | [TK]D-Fender | tzafrir_laptop: If a transfer fails on blind (usually only 404), its in dead-air |
| 17:20.43 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 17:21.06 | hardwire | tzafrir_laptop: that's one way to handle it |
| 17:21.11 | tzafrir_laptop | I'm talking about the asterisk feature, not the SIP transfer. How different are the two? |
| 17:21.34 | hardwire | blind transfer to a macro extension + actual extension adds you as the invalid, busy, or whatever priority. |
| 17:21.41 | [TK]D-Fender | tzafrir_laptop: *'s has control, more channel history, etc |
| 17:21.56 | [TK]D-Fender | tzafrir_laptop: Exactly what that can offer I can't say, but for sure * has more options. |
| 17:22.03 | jaytee | has anyone ever run into issues with Polycom phones reporting duplicate IP addresses when plugged into an unmanaged Linksys switch? |
| 17:22.51 | hardwire | jaytee: probably because there is a duplicate IP issue on your network, elsewhere. |
| 17:22.56 | Dovid | lol |
| 17:23.16 | hardwire | jaytee: find the windows computer or phone somewhere else that is reporting a duplicate IP. :) |
| 17:23.17 | Dovid | get the ip of the phone, pull it out and ping it |
| 17:23.25 | hardwire | jaytee: exactly |
| 17:23.47 | hardwire | if that happens to me I unplug the NIC on whatever is having issues then from another machine ping the IP then look it up in the arp tables |
| 17:24.02 | hardwire | then do a MAC lookup on the vendor |
| 17:24.08 | nny_2 | not exactly on topic, but anyone here a guru with SSH tunneling? |
| 17:24.13 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 17:24.14 | hardwire | then I eventually find the person who set their machine to static, and smack them on the head. |
| 17:24.14 | ThoMe | hihoi :-) |
| 17:24.20 | hardwire | nny_2: I am |
| 17:24.23 | hardwire | msg me for fun and profit |
| 17:24.40 | jaytee | nope, when both phones are unplugged and I ping either address I get no response |
| 17:24.59 | nny_2 | hardwire: I am using puTTY and have a tunnel setup in my session as L3000 192.168.100.1 |
| 17:25.12 | hardwire | nny_2: message me |
| 17:25.19 | nny_2 | hardwire: ok |
| 17:25.25 | hardwire | I promise I won't charge for the first 2 minutes. |
| 17:27.31 | jplank | lI dont get endusers - I have a customer who has their own asterisk, I'm bringing him in 8 lines, I told him that since he has his own asterisk that we can just use SIP trunks instead of converting my lines into analog and then plugging them into his asterisk, he keeps telling me no, that it works better converting it to analog first :/ |
| 17:27.48 | *** join/#asterisk `Sean (i=Un1x@CPE001d451b875f-CM00111ade88b6.cpe.net.cable.rogers.com) |
| 17:28.03 | `Sean | Hey Does anyone have any known soloutiions to fax over IP? |
| 17:28.19 | [TK]D-Fender | jplank: Yup, he's a dumb-fuck |
| 17:28.27 | jplank | the best is |
| 17:28.34 | jplank | he only has 2 four port cards |
| 17:28.39 | jplank | and he wanted 12 lines |
| 17:28.47 | jplank | so hes like, ok i'll buy another 4 port card |
| 17:29.25 | Dovid | 'Sean: solution = ? T.38 ? over ulaw ? In ? Out ? |
| 17:29.34 | jplank | first he asked if I can multiplex two lines over one analog pair |
| 17:30.05 | [TK]D-Fender | jplank: This person should have no say in determining a telephony solution |
| 17:30.26 | jplank | i agree |
| 17:31.23 | jplank | the best is he obviously knows how to setup a SIP trunk because he has another office in india, with a asterisk, thats connected over a SIP trunk |
| 17:31.44 | jplank | in retrospect, if he has to asterisk's connected over a SIP trunk, he probably has no clue what he's doing |
| 17:32.06 | `Sean | Dovid, i heard T38 is still unstable has the situation changed? |
| 17:32.11 | `Sean | has steve done any updates to it |
| 17:32.42 | `Sean | cisco has T37 wonder how stable that is |
| 17:32.46 | `Sean | ~t37 |
| 17:32.48 | `Sean | ~t38 |
| 17:32.48 | jbot | from memory, t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 17:33.30 | jplank | personally, my company just says no faxing over VOIP - period. Too unreliable. |
| 17:33.45 | jplank | customers ask us all the time |
| 17:34.30 | jplank | too many headaches involved |
| 17:35.12 | `Sean | hrmp indeed |
| 17:35.22 | `Sean | but its better then buying a dedicated fax line from your carier :) |
| 17:36.19 | jplank | not if you want to be able to send faxes :) |
| 17:36.52 | jplank | unless a 75% success rate on g711 is good enough |
| 17:37.02 | jplank | if you even use g711 |
| 17:37.21 | jplank | if your using g729 try like a 5% success rate, and those faxes usually still wind up blank |
| 17:37.24 | `Sean | There must be commercial products that have a higher success rate or a commercial codec for faxing |
| 17:37.36 | jplank | not a reliable one |
| 17:37.50 | Juggie | t.38 is the spec |
| 17:37.53 | `Sean | sigh :( |
| 17:38.07 | jplank | trust me, if faxing over VOIP was reliable, we'd do it |
| 17:38.31 | `Sean | ya but if your only using your fax for personal reasons voip is a good way to go in my opinion |
| 17:38.37 | `Sean | since your faxing barely 10 pages a month |
| 17:39.13 | hardwire | so yeh |
| 17:39.15 | hardwire | faxing is fun |
| 17:39.22 | hardwire | hylafax is funner |
| 17:39.27 | hardwire | and emailing is great fun. |
| 17:39.33 | hardwire | I'd stick to emailing |
| 17:39.49 | hardwire | the more you force people to use the internet to do internet like things, the better |
| 17:40.07 | `Sean | thats just fax to email |
| 17:40.28 | hardwire | is on the bandwagon where forcing everybody to use email will eventually mean less fax crap to support. |
| 17:40.33 | hardwire | universally. |
| 17:40.35 | jaytee | I've got one fax line going PRI --> Asterisk ----> SIP T.38 ------> Handytone 286 ATA ----- fax machine and I've not had any serious problems with it. |
| 17:40.58 | hardwire | which t.38 implementation? |
| 17:40.59 | `Sean | jaytee define serious lol? it means youve had problems so what kind of problems? |
| 17:40.59 | Dovid | 'Sean: some1 wrote a T38 patch for asterisk 1.4.X. very unstable at the moment |
| 17:41.01 | jaytee | but I consider myself lucky compared to all the chit-chat I've seen in here on the topic. |
| 17:41.24 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 17:41.33 | hardwire | jaytee: so you have a sip gateway next to your asterisk machine that supports T.38? |
| 17:41.53 | jaytee | `Sean, I had a couple of problems back when it was going through a TDM card but it was a cheap X100P clone card. Since I moved everything to PRI I've not had any issues. |
| 17:42.21 | `Sean | Jaytee, ive heard sangoma cards are excellent but dont know how excellent LOL |
| 17:42.29 | jaytee | hardwire, no gateway, just a Handytone 286 SIP/FXS ATA adapter. |
| 17:42.38 | hardwire | going to asterisk? |
| 17:43.05 | hardwire | if so, how are you implementing T.38? |
| 17:43.05 | jaytee | I'm using Digium's TE212P card for 2 T-1s and it's been running since May 15th without a hiccup. |
| 17:43.25 | jaytee | hardwire, over SIP |
| 17:43.31 | hardwire | so.. no T.38? |
| 17:44.07 | `Sean | ya but thats with a PRI, how about faxing using a T38 passthrough carrier |
| 17:44.23 | hardwire | ? |
| 17:46.42 | jaytee | hardwire, I don't understand your question. I setup the line in my sip.conf and set t38pt_udptl=yes in sip.conf and enabled t.38 in the Handytone and it all worked. |
| 17:47.09 | *** join/#asterisk trelane (i=trelane@lan.trelane.net) |
| 17:47.17 | hardwire | jaytee: it's not using T.38 |
| 17:47.36 | jaytee | really? what's it using then? |
| 17:47.42 | hardwire | ULAW :) |
| 17:48.37 | jaytee | so why do I have to have the t.38 settings set like that in order for it to work right? |
| 17:48.53 | jaytee | both in sip.conf and on the ATA adapter? |
| 17:48.57 | hardwire | maybe so it can say it doesn't support T.38 termination? |
| 17:49.01 | hardwire | correctly |
| 17:49.13 | hardwire | instead of just assuming T.38 works |
| 17:49.15 | `Sean | jaytee if you wheren't using PRI and had to use a voip carrier that supported t38 passthrough then ud have problems i think |
| 17:49.46 | jaytee | so if t.38 doesn't work then why is even in there? |
| 17:49.54 | *** part/#asterisk mrnick (n=ubugo@88.197.232.204) |
| 17:50.23 | hardwire | jaytee: dunno if the logs say anything, but I'm guessing the handytone is being told "not a t.38 termination point" and it instead uses the default SIP codec and expects a regular session for the voice channel |
| 17:50.33 | [TK]D-Fender | jaytee: *'s T.38 in 1.4 is passthrough only, and the problem is a quesiotn of compliance amongst vendors |
| 17:50.41 | [TK]D-Fender | jaytee: Ask coppice about that :) |
| 17:50.44 | hardwire | if you check the channel list while a fax is going through, I'm guessing it will show you that the handytone is conencted to your system VIA ulaw. |
| 17:50.53 | hardwire | sip show channel |
| 17:50.55 | hardwire | stuff like that |
| 17:52.28 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
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| 17:56.32 | jaytee | I'll have to try that when I actually get some time |
| 17:56.42 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:56.55 | jaytee | but right now I've got to figure out why an unmanaged Linksys skunks DHCP with Polycom phones. |
| 17:57.36 | jaytee | what's weird is if I move a phone to another subnet that's in it's own VLAN I don't get the duplicate IP problem. |
| 17:59.52 | hardwire | what's your dhcp server say about this? |
| 17:59.59 | jaytee | ok, I'm not certain of it but I think I've got it narrowed down to something in the Linksys firmware. I've got a Linksys SD205 that's on my 104 vlan and it works fine. I tried using a SD208 on my primary VLAN and any Polycoms give the duplicate address message yet none of the addresses are in use. |
| 18:00.04 | *** part/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 18:00.47 | jaytee | and I just replaced the SD208 with an older SD205 and that works ok but I ran into the same problem with an SD216 last week. |
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| 18:15.24 | neverbart | hi. i'm getting the following: [Jul 24 19:14:53] ERROR[3202]: cdr_addon_mysql.c:467 my_load_module: Failed to connect to mysql database asterisk on localhost. |
| 18:15.24 | neverbart | <PROTECTED> |
| 18:15.24 | neverbart | <PROTECTED> |
| 18:16.13 | *** join/#asterisk FriedCPU (n=friedcpu@fluxbuntu/user/FriedCPU) |
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| 18:31.47 | neverbart | i shall take the immense silence as people not knowing : |
| 18:31.48 | neverbart | :) |
| 18:32.21 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 18:33.24 | [TK]D-Fender | neverbart: PASTEBIN EVERYTHING. |
| 18:35.29 | ibnolqaiyem | is asterisk has good configuration options for linksy SPA3102 ata? |
| 18:35.32 | neverbart | [TK]D-Fender - http://beta.pastebin.cz/show/6994 |
| 18:36.22 | [TK]D-Fender | ibnolqaiyem: ...HUH? |
| 18:36.50 | ibnolqaiyem | [TK]D-Fender, what happened? |
| 18:37.17 | [TK]D-Fender | ibnolqaiyem: What happened is that the words are english... but nobody can understand what you're asking. |
| 18:38.01 | [TK]D-Fender | never... MORE. Backup that the sock file is actually in that location in your distro (#1 risk). Then back it up with a local login. |
| 18:38.05 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 18:38.10 | *** part/#asterisk omeow (n=omega@co63471-b.olden1.ov.home.nl) |
| 18:39.02 | neverbart | [TK]D-Fender... wah? |
| 18:39.03 | ibnolqaiyem | [TK]D-Fender, what is problem with my question? |
| 18:39.25 | neverbart | it's difficult to understand, ibnolqaiyem :) |
| 18:39.40 | ibnolqaiyem | ok i try again |
| 18:40.39 | [TK]D-Fender | neverbart: it should be trying the local socket based on that file in your config. the LOCATION is usually suspect as it varies depending on how it was packaged |
| 18:40.53 | [TK]D-Fender | neverbart: sock=/var/lib/mysql/mysql.sock <--- go look for it |
| 18:40.56 | neverbart | ah - i checked when i created the config. that is the file path |
| 18:40.58 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-187.hsi.ish.de) |
| 18:41.07 | ibnolqaiyem | is asterisk can work fine with linksys spa3102? if there are some ata better than link sys ,can you tell me? |
| 18:41.09 | [TK]D-Fender | neverbart: Ok, then go check the rest and share. |
| 18:41.16 | neverbart | it should work fine, ibnolqaiyem |
| 18:41.18 | neverbart | file /var/lib/mysql/mysql.sock |
| 18:41.18 | neverbart | /var/lib/mysql/mysql.sock: socket |
| 18:41.28 | [TK]D-Fender | ibnolqaiyem: What do you want to do exactly? |
| 18:41.35 | neverbart | and it was created from the repo, [TK]D-Fender |
| 18:41.46 | [TK]D-Fender | neverbart: Ok, as long as its there. |
| 18:41.52 | [TK]D-Fender | never go look at the rest. |
| 18:42.05 | neverbart | look at the rest of what? |
| 18:42.29 | neverbart | i can locally login using the mysql prompt |
| 18:42.33 | neverbart | using the same credentials |
| 18:43.32 | ibnolqaiyem | [TK]D-Fender, run asterisk with linksys spa3102 and analog phone |
| 18:43.49 | ibnolqaiyem | are these package good? |
| 18:43.49 | neverbart | i have an ata and analogue phone and it works fine, ibnolqaiyem |
| 18:44.17 | [TK]D-Fender | ibnolqaiyem: Only reason to pick the SPA-3102 is if you specifically want to use an analog LINE. If you have need of an analog LINE, and only want to use an analog PHONE, then pick another model. |
| 18:44.32 | angryuser | has anyone created a sip client vith videosupport for mobile devices? like nokia series or all wm6 smartphones3G |
| 18:44.51 | angryuser | it would be such a nice feature |
| 18:44.58 | [TK]D-Fender | neverbart: please pastebin it, drilling through to the table |
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| 18:46.00 | ibnolqaiyem | [TK]D-Fender, can explain more ,pls? |
| 18:46.07 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 18:46.17 | [TK]D-Fender | ibnolqaiyem: Do you have an analog LINE? |
| 18:47.17 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 18:47.22 | neverbart | http://beta.pastebin.cz/show/6995 [TK]D-Fender |
| 18:48.54 | [TK]D-Fender | neverbart: is * running as root? |
| 18:49.16 | neverbart | [TK]D-Fender yes |
| 18:50.28 | tclineks | any advice on the best python agi implementation? I haven't done twisted and am leaning towards trying to avoid it. |
| 18:51.35 | [TK]D-Fender | never I'm out of guesses ATM |
| 18:52.15 | neverbart | me too, [TK]D-Fender :( |
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| 19:00.14 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
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| 19:08.40 | Ritzerisk | anyone know a good Sip provider |
| 19:10.47 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 19:11.51 | [TK]D-Fender | itsplist-us |
| 19:11.53 | [TK]D-Fender | ~itsplist-us |
| 19:11.54 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:14.34 | Ritzerisk | how bout for a shanghai location i need to probally just get a sipuar 2102 and just convert it to analog to our system.... in vegas to hit a mitel phone in Shanghai.. ill check if they have shanghai numbers.... |
| 19:14.51 | Ritzerisk | thanks |
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| 19:21.03 | neverbart | well, thanks for trying to help [TK]D-Fender, i'm going to try and fight it out :) see ya |
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| 19:29.17 | *** join/#asterisk guilherme-jorge (n=guilherm@200.213.210.2) |
| 19:29.34 | guilherme-jorge | hello all, Is there some tool to convert gsm files to g729 format? |
| 19:31.45 | Strom_M | guilherme-jorge: you're going to want to start with wav files, not gsm files...or else the quality will be truly abysmal instead of merely bad |
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| 19:41.51 | [TK]D-Fender | guilherme-jorge: For converting your own, there is a tool on Digium's site. |
| 19:42.09 | [TK]D-Fender | guilherme-jorge: All stock * sounds are also available in G.729 native encoded. |
| 19:42.52 | fogo | what do I need to do to troubleshoot a T1 span that is Provisioned, Down, Active? |
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| 20:00.36 | jameswf-home | heh http://www.100factsabout.com/James/Finstrom |
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| 20:02.24 | Katty | hai |
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| 20:08.33 | tzafrir_laptop | jameswf-home, some other interesting facts to you: http://elcuco.blogli.co.il/archives/166 |
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| 20:10.45 | carrar | w00t! |
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| 20:22.04 | ibnolqaiyem | [TK]D-Fender, are you mean the line coming from pstn? |
| 20:22.27 | [TK]D-Fender | ibnolqaiyem: only 2 hours later... YES |
| 20:22.43 | ibnolqaiyem | [TK]D-Fender, i am sorry for that |
| 20:22.52 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 20:22.52 | [TK]D-Fender | ibnolqaiyem: and the ANSWER? |
| 20:23.32 | ibnolqaiyem | [TK]D-Fender, are you mean about analog line , the line coming from pstn? |
| 20:23.40 | [TK]D-Fender | YES |
| 20:23.52 | ibnolqaiyem | yes , i have one |
| 20:24.03 | jameswf-home | is lost |
| 20:24.05 | [TK]D-Fender | ibnolqaiyem: And would you like to let * use it? |
| 20:24.15 | ibnolqaiyem | yes |
| 20:24.27 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-187.hsi.ish.de) |
| 20:24.27 | [TK]D-Fender | ibnolqaiyem: Fine, then the SPA-3102 is a fine choice. |
| 20:24.43 | [TK]D-Fender | hands jameswf-home a "You are here" sign |
| 20:24.44 | ibnolqaiyem | [TK]D-Fender, thank you very much |
| 20:25.19 | jameswf-home | I have an issue.... wherever I go there I am, it is almost like I am stalking myself |
| 20:26.30 | [TK]D-Fender | ok, checkout time. Heading home. Later all |
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| 21:00.04 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:00.20 | jaytee | ok, [TK]D-Fender, that was like a 13 minute commute. do you live like a block from your work? |
| 21:00.55 | [TK]D-Fender | 20 mins on bike :) |
| 21:01.23 | jaytee | then you must have been pedalling extra fast today |
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| 21:01.44 | *** mode/#asterisk [+o angler] by ChanServ |
| 21:02.06 | jaytee | quittin time, be back later |
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| 21:04.11 | hardwire | so.. nobody wants to loan me there expensive hardware eh? |
| 21:04.12 | hardwire | FINE! |
| 21:10.05 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 21:19.48 | Deeewayne | ~thebook |
| 21:19.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
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| 21:23.27 | *** join/#asterisk ZeroLux (n=none@96-25-30-144.ral.clearwire-dns.net) |
| 21:23.27 | hardwire | anybody doing passive PRI audio monitoring? |
| 21:23.33 | *** join/#asterisk NickKoji (n=nicholas@168-215-215-195.static.twtelecom.net) |
| 21:25.04 | NickKoji | Hello, I am farely new to asterisk and was wondering what website you would recommend to begin reading. I am trying to understand the basics of asterisk so tutorial websites would help the most but any links is appreciated. I will also be posting the list of addresses i compile onto any support forums that i register on |
| 21:25.10 | putnopvut | ~book |
| 21:25.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:25.31 | putnopvut | NickKoji: ^^^ that is very helpful for beginning Asterisk. |
| 21:25.55 | NickKoji | thank you |
| 21:26.14 | putnopvut | Also |
| 21:26.15 | putnopvut | ~wiki |
| 21:26.25 | putnopvut | crap |
| 21:26.58 | putnopvut | http://voip-info.org/wiki/view/Asterisk |
| 21:27.12 | putnopvut | That can be helpful too, although some of the information on that is out-of-date. |
| 21:31.14 | [TK]D-Fender | ~wikis |
| 21:31.15 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
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| 21:47.33 | esaym | howdy, I have an already compiled source that I need to make clean. is "make clean" all that I need to do? |
| 21:47.34 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 21:48.20 | russellb | esaym: or "make distclean" if you want to remove _everything_ |
| 21:48.52 | esaym | ah, distclean, that is what I was looking for |
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| 21:54.40 | ManxPower | russellb: does make distclean also kill whatever cache file it is that makes Asterisk ./configure think zaptel is never installed if it was ever run without it installed? |
| 21:54.55 | Qwell | ManxPower: yes |
| 21:55.02 | ManxPower | Qwell: cool |
| 21:55.16 | Qwell | (but so does re-running configure) |
| 21:55.40 | ManxPower | Qwell: I can't PERSONALLY dispute that, but it doesn't seem that way based on the reports here. |
| 21:55.46 | NickKoji | is there a irc channel for asterisknow |
| 21:55.53 | ManxPower | "I installed Zaptel and make menuconfig still shows zaptel greyed out!" sort of things |
| 21:55.55 | Qwell | NickKoji: see topic |
| 21:56.00 | Qwell | ManxPower: "You're doing it wrong." |
| 21:56.11 | ManxPower | Qwell: I'll tell them that. |
| 21:56.12 | Qwell | ManxPower: I can and will personally dispute those claims. :) |
| 21:56.28 | ManxPower | Qwell: it's all 1.4'isms anyway. |
| 21:56.30 | NickKoji | thank you |
| 21:56.52 | ManxPower | NickKoji: thank you for asking and not assuming this is the correct channel (it's not) |
| 21:57.22 | NickKoji | is there a massive difference? |
| 21:57.48 | ManxPower | NickKoji: Yes. I'll type the long answer next. |
| 21:57.48 | NickKoji | better yet is there an article comparing to two |
| 21:57.56 | NickKoji | yeah i figured it was long ha |
| 21:58.39 | ManxPower | NickKoji: Asterisk isn't really a PBX. It's a TOOLKIT that lets you build a PBX. The GUIs for Asterisk basically all build their own custom config files in ways no HUMAN would do it, because a person would find it very hard to figure out. |
| 21:58.42 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 21:58.59 | ManxPower | ~freepbx |
| 21:59.00 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:59.01 | ManxPower | ~trixbox |
| 21:59.02 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 21:59.04 | exothermc | I'm trying to install some polycom phones behind pfsense, and it isn't playing nicely |
| 21:59.16 | ManxPower | That Asterisk GUI is the least evil of all the GUIs in this regard. |
| 21:59.42 | ManxPower | exothermc: Provisioning issues or audio/signalling issues? |
| 21:59.48 | exothermc | I have setup siproxd on pfsense which is suppose to help with those issues, but I'm not sure how to set the polycoms to register. |
| 21:59.58 | exothermc | ManxPower: signaling |
| 22:00.11 | NickKoji | what would be the best system to install if i am creating an autidialer that calls out numbers and returns us the values of things like callerid and stores them to a DB |
| 22:00.40 | ManxPower | exothermc: from a NETWORKING sense, registration packets will come from the IP address of the phone using a random source port number and be going to the IP of Asterisk port 5060. All this is UDP, NOT TCP. |
| 22:01.08 | ManxPower | Asterisk should respond from it's IP address/port 5060 to the IP of the phone with a destination port of whatever the phone originally used. |
| 22:01.14 | exothermc | ManxPower: ya that is the issue all the phones register using a source port of 5060 |
| 22:01.19 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 22:01.31 | ManxPower | exothermc: The phones are welcome to do so and many do, but NEVER count on that. |
| 22:01.43 | WhiteWolf | if everything is behind the same nat you shouldn't have problems |
| 22:02.09 | WhiteWolf | if the phones are behind pfsense and asterisk is somewhere else... set pfsense to use 1:1 nat static mapping |
| 22:02.18 | exothermc | http://www.pfsense.org/index.php?option=com_content&task=view&id=40&Itemid=43 as you can see from there sip is an issue. |
| 22:02.18 | WhiteWolf | that way it doesn't attempt to rewrite the port |
| 22:02.25 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-187.hsi.ish.de) |
| 22:02.47 | ManxPower | exothermc: can you just turn off any SIP packet fixup in the pfsense firewall? |
| 22:03.01 | ManxPower | most devices seem to screw it up when they try to do special stuff for SIP NAT |
| 22:03.34 | exothermc | ManxPower: Well I could be wrong but doesn't it say that pfsense doesn't rewrite? |
| 22:03.40 | ManxPower | oh, I guess the sipproxyd would be that thing |
| 22:04.17 | ManxPower | exothermc: I noticed it did not change the source port, but that may or may not mean it changes the data portion of the packet |
| 22:04.43 | ManxPower | I'm one of those mythical people that has never had a NAT issue with SIP that I could not solve in under 30 mins. |
| 22:05.39 | exothermc | ManxPower: Really this is the first device I have run up against that has cause me to scratch my head. |
| 22:06.09 | ManxPower | But I also never try doing that with connections with dynamic IP address, nothing but Cisco router or Linksys router, Asterisk only, SIPura only. |
| 22:07.50 | ManxPower | The secret to most success with Asterisk is design it well, keep it simple, use good hardware. |
| 22:16.15 | *** join/#asterisk hi365_m (n=hi365@213.151.61.251) |
| 22:19.33 | *** join/#asterisk cybrside (n=cybrside@rrcs-71-41-145-210.sw.biz.rr.com) |
| 22:20.07 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 22:21.26 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 22:22.30 | C4colo | is it possible to do a string replace from within the dialplan? |
| 22:22.43 | C4colo | do I need to use the System() application? |
| 22:23.27 | C4colo | I want to take a variable from the asterisk internal database and replace any spaces with escaped spaces, such as "Joe Smith" -> "Joe\ Smith" |
| 22:24.31 | *** join/#asterisk Katty (n=angela@adsl-209-30-144-78.dsl.stlsmo.swbell.net) |
| 22:24.37 | Katty | herro. |
| 22:25.16 | *** join/#asterisk rolandf (n=roland@124.31-241-81.adsl-dyn.isp.belgacom.be) |
| 22:25.26 | C4colo | I could just save it to the database with three \\\'s wherever there are spaces, such as "Joe\\\ Smith" |
| 22:26.02 | C4colo | but that requires the addition to the database to be rigorous, and provides no fault tolerance |
| 22:29.02 | *** part/#asterisk rolandf (n=roland@124.31-241-81.adsl-dyn.isp.belgacom.be) |
| 22:29.44 | *** join/#asterisk mmarcost (n=mmarcos@151.193.220.28) |
| 22:31.51 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 22:32.42 | Katty | so quiet. |
| 22:32.47 | C4colo | yea |
| 22:32.59 | Strom_C | let's combine superstring theory with rodney king |
| 22:33.09 | *** part/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 22:33.13 | Katty | i like superstring theory |
| 22:33.18 | Katty | i wish i was made of superstrings |
| 22:33.53 | ManxPower | Wouldn't rodney king be a superstar rather than a superstring? |
| 22:34.51 | Katty | technically i would be made of superstrings |
| 22:34.53 | ManxPower | C4colo: doc/channelvariables.txt and "core show functions" Function names are CASE SENSITIVE. |
| 22:34.58 | Katty | but closed string loops. |
| 22:35.04 | Katty | i'd rather be openly stringy. ^_- |
| 22:35.05 | Katty | and loopy. |
| 22:35.06 | Katty | yeah. |
| 22:35.22 | ManxPower | Katty: Aren't you already a bit loopy? |
| 22:35.48 | Katty | ManxPower: a bit ;) |
| 22:35.53 | Katty | ManxPower: when i'm in the mood to be. |
| 22:36.11 | Katty | ManxPower: scatter brained is probably a better description. |
| 22:38.06 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:39.58 | C4colo | ManxPower: I see concatenation, substrings, string length, etc, but no find/replace functions for manipulating strings |
| 22:40.25 | seanbright | C4colo: prepare to be SPOKEN to in UPPERCASE words :) |
| 22:40.26 | C4colo | do I need to run a System( perl <regex crap here> ) |
| 22:41.02 | C4colo | oh I'm always ready for that when I come here |
| 22:41.09 | *** join/#asterisk Dan0maN_Wor1 (n=dschuh@64.149.174.136) |
| 22:41.40 | ManxPower | a perl AGI would be better. |
| 22:41.57 | ManxPower | write one that lets you give it a regex and it sets a channel variable with the result? |
| 22:42.48 | C4colo | well hell, if I'm going to write an AGI I'll just grab the bash script that converts the tiff to pdf and emails it into the agi and hand execution off to that |
| 22:43.14 | *** join/#asterisk mosty (i=foobar@60-241-198-194.static.tpgi.com.au) |
| 22:43.22 | ManxPower | Even better |
| 22:43.30 | ManxPower | That's what I did. |
| 22:43.46 | C4colo | this is old code I'm working with, about 5 years old, needs updating but I was looking for a simple, quick answer that would let this server hobble along until we move to the new system |
| 22:44.11 | C4colo | for now I think I'll suggest we just continue putting underscores instead of spaces into the database |
| 22:44.19 | ManxPower | an agi should take less than an hour for something simple as perl regex, even for a newbie |
| 22:44.31 | C4colo | as I will be writing a new fax email script for the new server |
| 22:44.39 | *** part/#asterisk Dan0maN_Wor1 (n=dschuh@64.149.174.136) |
| 22:44.43 | ManxPower | you want a copy of mine? |
| 22:44.45 | C4colo | I"m not worried about the perl regex |
| 22:44.59 | C4colo | sure, I would greatly appreciate it |
| 22:45.23 | ManxPower | It won't meet your needs, but you can use it as an example of one way to do it. Give me a few mins. |
| 22:45.41 | C4colo | thanks |
| 22:46.03 | C4colo | honestly it is probably better than the shoddy shell script currently sending the faxes |
| 22:46.04 | *** join/#asterisk hadronzoo (n=user@cpe-069-134-096-136.nc.res.rr.com) |
| 22:46.14 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:46.52 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 22:47.00 | teknoprep | have you guys seen what counterpath is going to be offering |
| 22:47.01 | teknoprep | omfg its nice |
| 22:47.08 | C4colo | huh? |
| 22:47.19 | teknoprep | the mobile suite |
| 22:47.27 | teknoprep | i really like it |
| 22:47.28 | C4colo | oh? |
| 22:47.44 | teknoprep | better than that unified crap MS offers |
| 22:47.47 | C4colo | doesn't counterpath litterally mean "wrong way"? |
| 22:48.17 | hadronzoo | Does anyone have an opinion on the Cisco 7900 series phones? I'm considering getting two to work with my asterisk box. |
| 22:48.25 | teknoprep | 7970 is great |
| 22:48.34 | C4colo | never used them |
| 22:48.42 | teknoprep | every other cisco 79XX phone sucks becuase of its lack of sip functions |
| 22:48.57 | C4colo | get a real phone, a GXP-2000 or something |
| 22:48.58 | teknoprep | as a cheap refurb phone if you are going to hang it on a wall in a manufacturing area |
| 22:49.02 | hadronzoo | teknoprep: can you elaborate? |
| 22:49.04 | C4colo | ducks |
| 22:49.06 | teknoprep | they will never break |
| 22:49.13 | teknoprep | i love polycom phones |
| 22:49.21 | teknoprep | i am going to be trying out an aastra 57i |
| 22:49.29 | teknoprep | 7970 phones are very cool |
| 22:49.38 | C4colo | I'm ordering a few of the 5i series to play with |
| 22:49.41 | teknoprep | you can do alot with that phone.. but in all actuality its just a nice desk peice |
| 22:49.47 | dlynes | teknoprep: the 57i's aren't too shabby |
| 22:49.47 | C4colo | I think I'm going to use them on a 48 seat install I'm doing |
| 22:49.52 | hadronzoo | C4colo: What's the price difference? |
| 22:49.59 | C4colo | between? |
| 22:50.08 | teknoprep | hadronzoo, polycom are in the 250-350 range for a good 550 or 650 |
| 22:50.12 | ManxPower | http://www.fnords.org/~eric/fax2email.txt |
| 22:50.17 | dlynes | teknoprep: unlike the 91xxi series, they boosted up the memory and the cpu with the 5xi series |
| 22:50.21 | teknoprep | hadronzoo, cisco 7971 is very expensi ve |
| 22:50.22 | hadronzoo | C4colo: the 7970 and the GXP-2000 |
| 22:50.27 | C4colo | I was joking |
| 22:50.32 | C4colo | hince the ducking |
| 22:50.37 | teknoprep | hadronzoo, you don't want a GXP-2000... |
| 22:50.49 | hadronzoo | C4colo: Ah, sorry I missed it |
| 22:50.51 | C4colo | the grandstream GXP-2000 is a $80 phone that is worth about $20 |
| 22:51.00 | teknoprep | hadronzoo, its probably tied in the bracket of worst sip phones ever made |
| 22:51.15 | hadronzoo | Gotcha |
| 22:51.20 | dlynes | teknoprep: even worse than the budgetone 100/102? |
| 22:51.26 | teknoprep | i would rather put used cisco phones from ebay on ppl's desk than grandshit |
| 22:51.30 | C4colo | the new firmware upgraded it from "worst sip phone ever" to "one of the worst phones ever" |
| 22:52.28 | hadronzoo | Thus, the, "Do not consider Grandstream phones. Ever." |
| 22:52.29 | *** join/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 22:52.33 | C4colo | unfortunately the company I work for has settled on the GXP-2000 as their budget phone, I never touched one before, and wouldn't have, and now know why |
| 22:52.41 | C4colo | go with a $80 aastra or snom instead |
| 22:52.53 | ManxPower | For Polycom people there is also http://www.fnords.org/~eric/polycom-config-examples |
| 22:52.56 | *** join/#asterisk vee8 (n=ffff@c-98-217-184-103.hsd1.ma.comcast.net) |
| 22:53.08 | teknoprep | dlynes, do you like the 57i ? |
| 22:53.13 | teknoprep | dlynes, how is the call quality ? |
| 22:53.16 | dlynes | teknoprep: for the most part, yes |
| 22:53.17 | hadronzoo | You would recommend the Polycom as being a better phone and better deal than the Cisco? |
| 22:53.26 | teknoprep | hadronzoo, for SIP.. yes |
| 22:53.26 | dlynes | teknoprep: the call quality is just fine...nothing wrong with it |
| 22:53.34 | C4colo | ManxPower: thank you, this looks good, I'm sure it will be a big step up from what they were using |
| 22:53.35 | teknoprep | dlynes, what don't you like about it ? |
| 22:53.44 | dlynes | teknoprep: the only issues I've had with it, is when I've got major BLF's configured for the phone |
| 22:53.51 | ManxPower | C4colo: That script does not come with support. |
| 22:53.56 | dlynes | teknoprep: if the phone gets overloaded with blf, it'll lock up |
| 22:54.06 | teknoprep | i don't use BLF |
| 22:54.09 | dlynes | teknoprep: but other than that, it's a good phone |
| 22:54.30 | dlynes | teknoprep: really easy to configure using tftp, too |
| 22:54.34 | hadronzoo | How important is SIP support? |
| 22:54.39 | teknoprep | hadronzoo, lol |
| 22:54.42 | hadronzoo | (newbie here) |
| 22:54.49 | C4colo | ManxPower: of course |
| 22:55.00 | teknoprep | hadronzoo, honestly ... cisco is NOT the phone for you if you are a newb |
| 22:55.09 | teknoprep | hadronzoo, i would suggest a polycom phone |
| 22:55.23 | teknoprep | hadronzoo, there isn't a TON of support for cisco and SIP implementations |
| 22:55.26 | hadronzoo | I'm just trying to connect two phones in an office to an asterisk box to use a gateway |
| 22:55.28 | dlynes | hadronzoo: sip is not that important, unless you're going after office users |
| 22:55.39 | teknoprep | hadronzoo, then buy some polycom ip550's |
| 22:55.41 | dlynes | hadronzoo: and/or multiple phone lines |
| 22:55.42 | ManxPower | hadronzoo: Imagine your self in the middle of a desert. You can see nothing anywhere. That is your community support for any protocol other than SIP for phones. |
| 22:55.42 | teknoprep | hadronzoo, you will be happy |
| 22:55.43 | hadronzoo | dlynes: can you elaborate? |
| 22:56.21 | dlynes | hadronzoo: if you're only planning on using 1 or 2 lines, you might actually find it simpler to get a linksys spa2102, and hooking up both lines to a two line phone |
| 22:56.35 | *** part/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 22:56.43 | C4colo | nah, you can get support for SCCP/Skinny ... from Cisco... on their Call Manager... with a support contract |
| 22:56.47 | dlynes | hadronzoo: and then just forget asterisk altogether, unless you need voicemail that's not provided by your ITSP |
| 22:57.28 | hadronzoo | ManxPower: So, why would the cisco 7960 series not have SIP support? |
| 22:57.28 | hadronzoo | dlynes: I want to setup an IVR and multiple lines. I'm not a stranger to linux, but this is my first asterisk project |
| 22:57.40 | hadronzoo | dlynes: but I want to learn more |
| 22:57.45 | teknoprep | hadronzoo, it does have sip support |
| 22:57.50 | teknoprep | hadronzoo, it's just not very good |
| 22:57.56 | dlynes | hadronzoo: ah...then you probably want sip phones |
| 22:58.11 | *** join/#asterisk ZeroLux (n=none@cpe-071-077-034-161.nc.res.rr.com) |
| 22:58.15 | dlynes | hadronzoo: if you don't have a huge budget, i'd suggest going with polycoms or aastra phones |
| 22:58.23 | ManxPower | hadronzoo: because Cisco wants you to use the phones only with the Call Manager playform |
| 22:58.24 | dlynes | hadronzoo: they're both user-friendly phones, too |
| 22:58.31 | teknoprep | i would suggest for a test environment buying refurbished phones |
| 22:58.48 | teknoprep | ManxPower, call manager is actually moving to SIP now.. |
| 22:58.56 | teknoprep | ManxPower, so the new phones are easier to setup with asterisk |
| 22:59.27 | C4colo | aastras are asterisk friendly, especially the newer 5i series ... as the aastra pbx system is built on asterisk 1.4 |
| 22:59.27 | ManxPower | that will last just as long as Call Manager revenue does not drop. |
| 22:59.50 | C4colo | so most of the features work with asterisk, even blf and blf-pickup |
| 23:00.14 | C4colo | you have to set the pickup feature code thingie on the phone to ** |
| 23:00.31 | C4colo | and then you can pick up a ringing extension on a BLF button by hitting the flashing button |
| 23:00.40 | C4colo | stuff like that |
| 23:00.45 | teknoprep | lol |
| 23:00.48 | teknoprep | i hate BLF |
| 23:00.57 | C4colo | I'm ordering a number of 5i series to play with stuff like that |
| 23:01.03 | teknoprep | i prefer using CTi's to do that |
| 23:01.04 | C4colo | ... love the BLF ... your customers do |
| 23:01.11 | hadronzoo | What's the price range on the 5i series? |
| 23:01.14 | C4colo | well some of them anyway I would assume |
| 23:01.15 | teknoprep | just drag and drop a call from a window to your extension |
| 23:01.19 | GhOnDiE | very usefull blf |
| 23:01.24 | JT | hadronzoo: telephonydepot.com |
| 23:01.24 | teknoprep | i like isymphony |
| 23:01.30 | *** join/#asterisk ZeroLux (n=none@cpe-071-077-034-161.nc.res.rr.com) |
| 23:01.32 | hadronzoo | JT: thanks |
| 23:01.45 | C4colo | I just downloaded isymphony, haven't got around to configuring it yet |
| 23:02.00 | *** join/#asterisk ZeroLux (n=none@cpe-071-077-034-161.nc.res.rr.com) |
| 23:02.23 | teknoprep | its great |
| 23:02.37 | teknoprep | and since i am a freepbx luser |
| 23:02.39 | teknoprep | its easy to setup |
| 23:02.53 | C4colo | my god you are brave |
| 23:02.57 | teknoprep | yup |
| 23:03.00 | teknoprep | f' em |
| 23:03.01 | C4colo | saying that in here? |
| 23:03.08 | ManxPower | Heretic! |
| 23:03.10 | teknoprep | lol |
| 23:03.14 | C4colo | lol |
| 23:03.25 | teknoprep | i do some custom stuff man.. but most of it is so quick and easy to setup |
| 23:03.26 | hadronzoo | So, you like the Aastra 55i? |
| 23:03.29 | teknoprep | within freepbx |
| 23:03.36 | teknoprep | 55i is ugly |
| 23:03.40 | teknoprep | get the 57i |
| 23:03.49 | ManxPower | screams "My eyes! My eyes!" and goes AFK. |
| 23:03.50 | C4colo | yea or the 57iCT |
| 23:03.56 | hadronzoo | teknoprep: I'll look it up |
| 23:04.00 | C4colo | those cordless handsets are nice |
| 23:04.03 | teknoprep | they are ? |
| 23:04.05 | teknoprep | skrew that |
| 23:04.13 | teknoprep | http://www.counterpath.com/mobility-suite.html |
| 23:04.14 | teknoprep | check that out |
| 23:04.22 | C4colo | I went outside a metal building about half a block away with multiple metal sheds around me and it juts barely started breaking up |
| 23:04.23 | teknoprep | for better wireless communications |
| 23:04.59 | C4colo | aastra has some DECT access points and cordless handsets, but they have some goofy requirements for who can buy and resell them |
| 23:05.07 | C4colo | I want to get my hands on a set of those and really play |
| 23:05.09 | hadronzoo | teknoprep: Yeah, the 57i looks much better |
| 23:05.20 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 23:06.06 | teknoprep | C4colo, did you check out that link ? |
| 23:06.13 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:06.16 | teknoprep | C4colo, just use your cell phone |
| 23:06.21 | hadronzoo | So, one can access essentially 9 lines concurrently with the 57i? |
| 23:06.22 | teknoprep | C4colo, why use anything else |
| 23:06.44 | teknoprep | hadronzoo, no real reason to access 9 lines at once unless you are a receptionist |
| 23:06.58 | MatBoy | can I run a * on port 5060 and 5070 at the same time ? |
| 23:07.34 | ManxPower | MatBoy: I don't think so. |
| 23:07.38 | hadronzoo | teknoprep: But, you can switch between multiple lines, right? |
| 23:07.41 | MatBoy | ManxPower: mhh issue |
| 23:07.54 | teknoprep | hadronzoo, yes |
| 23:08.39 | C4colo | teknoprep: the cool thing about those 5i series is that they have everything from a single-line for the waiting room or break room to something the receptionist can use with sidecars to monitor every extension in the building, and they all work together |
| 23:08.40 | hadronzoo | Thanks so much for pointing me in the right direction. I really appreciate the assistance! |
| 23:09.01 | C4colo | there are two models with cordless handsets too |
| 23:09.22 | mosty | is it possible to use the sangoma A500 with zaptel instead of chan_woomera? |
| 23:09.59 | hadronzoo | The wireless 57i CT looks really cool |
| 23:10.05 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 23:10.21 | C4colo | don't get the 480iCT |
| 23:10.28 | C4colo | it is the older firmware/hardware |
| 23:10.35 | legis | Are linksys pap2 good to test a asterisk setup? |
| 23:10.43 | C4colo | sure, they are cheap |
| 23:10.48 | C4colo | and plentiful |
| 23:10.56 | C4colo | and there are lots of config things out there |
| 23:11.12 | hadronzoo | C4colo: right, but I should have good luck with the 57i or the 57i CT with Asterisk, right? |
| 23:11.12 | legis | cool, thx |
| 23:11.16 | C4colo | things = howtos |
| 23:11.26 | C4colo | yes |
| 23:11.31 | hadronzoo | C4colo: thanks |
| 23:12.10 | C4colo | also if you get to the point where you are selling around $10,000 of product for them they have a reseller incentives program |
| 23:12.26 | C4colo | cache in rebate things for phones or something |
| 23:12.31 | C4colo | oh, that's quarterly |
| 23:12.43 | *** join/#asterisk elfurud (n=Mike@CPE001ee5559754-CM0011ae908bc4.cpe.net.cable.rogers.com) |
| 23:12.45 | GhOnDiE | blimey thats alot quarterly |
| 23:12.47 | jameswf | ping tzafrir_laptop |
| 23:12.50 | elfurud | howdy folks |
| 23:12.55 | C4colo | yea, that's what I said when I applied |
| 23:12.59 | tzafrir_laptop | pong |
| 23:13.07 | elfurud | anyone know a provider of Canadian DID that supports multiple channels per DID? |
| 23:13.16 | C4colo | les.net |
| 23:13.25 | hi365_m | was thinking tzafrir_laptop woulb be asleepe durring these hours |
| 23:13.51 | elfurud | I've used Les.net in the past and haven't been overly impressed with quality |
| 23:13.51 | C4colo | they have 2chan unlimited inbound or $0.011 CAD per minute for multiple channels |
| 23:13.59 | elfurud | I've used Unlimitel also |
| 23:14.05 | tzafrir_laptop | thinks hi365 should be asleep during these hours |
| 23:14.20 | elfurud | I need more than 2 channels |
| 23:14.29 | C4colo | I use les.net for one of my primary DIDs and have found it to studder from time to time |
| 23:14.49 | elfurud | yah, the quality isn't great all the time. |
| 23:14.50 | C4colo | stutter? |
| 23:14.54 | C4colo | however that is spelt |
| 23:14.59 | elfurud | Unlimitel is good |
| 23:15.11 | jameswf | tzafrir_laptop: since ztdummy seems to be getting dumped for the crash-you have no carde wtf do you have asterisk maybe the zaptel init should unload zaptel on no hardware rather than load ztdummy... |
| 23:15.15 | hardwire | unlimitel! |
| 23:15.18 | elfurud | they have an option for 10 channels per DID for $90 a month, but that includes 10,000 minutes |
| 23:15.18 | elfurud | heh |
| 23:15.20 | hardwire | what an awesome name |
| 23:15.32 | jameswf | s/asterisk/zaptel |
| 23:15.41 | mosty | tzafrir: can you point me at some directions for using a sangoma A500 with zaptel instead of chan_woomera? |
| 23:16.09 | tzafrir_laptop | jameswf, sorry, I don't follow. What exactly is the problem? |
| 23:16.37 | C4colo | 10 channels for $90 being $9/mo per channel? |
| 23:16.40 | C4colo | that's not bad actually |
| 23:16.51 | elfurud | yah, thats reasonable |
| 23:16.56 | elfurud | but I might need even more! |
| 23:17.03 | elfurud | running a small call center/office |
| 23:17.06 | C4colo | they don't offer more? |
| 23:17.12 | C4colo | what if you paid $180 for 20 chan? |
| 23:17.28 | tzafrir_laptop | mostly, not exactly sure. I think you should use either bristuff or astterisk 1.6 . You don't need the zaptel parts of the bristuff patch |
| 23:17.30 | elfurud | yah, not sure if that is an option. I told them I needed 15 channels, they came back with this quote for 10/$90 |
| 23:17.51 | tzafrir_laptop | jameswf, what crash? |
| 23:18.15 | C4colo | then reply and say "i am lead to believe that the price for 15 would be $135 then right?" |
| 23:18.34 | C4colo | and would come with 1500 minutes |
| 23:18.38 | mosty | tzafrir: do you know which parts of bristuff? i am trying to avoid using the entire thing. perhaps it's 122-chan_zap-BRI-euroisdn.diff ? |
| 23:18.42 | C4colo | er 15,000 |
| 23:18.45 | C4colo | however many, |
| 23:19.32 | jameswf | trying to find the error hang on |
| 23:20.28 | Katty | wibbles |
| 23:21.24 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:22.15 | jameswf | zaptel_timer_error in main/asterisk.c |
| 23:25.21 | tzafrir_laptop | jameswf, ztdummy is loaded by default with the init.d script |
| 23:25.48 | tzafrir_laptop | That error is most likely to hit cards that don't bother providing timing all the time |
| 23:26.00 | jaytee | what the hell is wibbling? how does one wibble? |
| 23:26.06 | tzafrir_laptop | In this case it has exposed a bug in the driver |
| 23:26.23 | tzafrir_laptop | mosty, right, that patch |
| 23:26.47 | tzafrir_laptop | and the libpri patch |
| 23:27.00 | hardwire | any way to turn off fax detect from Dial()? |
| 23:27.16 | tzafrir_laptop | it's in the channel driver |
| 23:27.26 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:29.19 | tzafrir_laptop | jameswf, are you talking about a system with zaptel hardware? |
| 23:30.06 | tzafrir_laptop | This seems to be a problem with e.g. zaphfc |
| 23:31.06 | vee8 | hi...my asterisk suddenly failed incoming callers get a busy signal...im using a TDM11B...is it possible my FXO module has failed?...but dmesg sees it as expected "Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules)" |
| 23:44.39 | jameswf | tzafrir_laptop: nm it didnt associate in my head until I read the code asterisk.c Revision 112689 |
| 23:45.50 | jameswf | tzafrir_laptop: asterisk now crashes on a timing error rather than let you troubleshoot the silence |
| 23:45.53 | JT | tzafrir_laptop: does the sangoma A500 definitely work with zaptel? |
| 23:46.31 | tzafrir_laptop | JT, people here reported using it. I'm not from Sangoma :-) |
| 23:46.53 | JT | hrm fair enough |
| 23:47.07 | JT | what zap driver needs to be used? |
| 23:48.20 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 23:48.38 | edwin_quijada | There is a progresive dialer project for asterisk |
| 23:48.38 | edwin_quijada | ? |
| 23:49.26 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 23:55.49 | jaytee | edwin_quijada, do you mean a predictive dialer? |
| 23:56.07 | *** join/#asterisk mactimes (n=mactimes@unaffiliated/mactimes) |