| 00:00.00 | Rob3Rt | [TK]D-Fender, of course i am following the instructions, issue is my provider only supports g729, they say you might not be able to hear the caller if you use ulaw |
| 00:00.17 | Rob3Rt | im allowing all 3, but theres some other issue, and im not an aster pro yet |
| 00:08.55 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 00:09.39 | Gnewt | I'd like to set up line 2 for extension 6000 to set the caller ID to the same as the number being dialed before dialing out |
| 00:10.04 | Gnewt | eg if I dial 5552021010, it sets the caller ID to 5552021010 and then places the call |
| 00:10.22 | Gnewt | Is there a way to do that? |
| 00:17.56 | Rob3Rt | fine |
| 00:18.31 | Rob3Rt | question on g729, if i receive a call and make a call at the same time over g729, how many channels or licenses do i need |
| 00:24.21 | *** join/#asterisk s14ck (n=s14ck@190-76-118-151.dyn.movilnet.com.ve) |
| 00:32.54 | seanbright | Rob3Rt: 2 |
| 00:33.18 | seanbright | every concurrent channel needs a license |
| 00:33.31 | seanbright | if you are going to have 10 g729 calls going on at the same time, you need 10 licenses. |
| 00:34.21 | Rob3Rt | thanks :) |
| 00:34.24 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 00:34.25 | Rob3Rt | ive got 4 licenses |
| 00:34.38 | Rob3Rt | but i cant hear the other party |
| 00:34.46 | seanbright | sounds like nat issue |
| 00:34.51 | Rob3Rt | im going out of my mind, this stuff isnt so hard surely |
| 00:34.58 | Rob3Rt | ive go 5060 cleared and pointing to aster |
| 00:35.06 | seanbright | there is an article out there |
| 00:35.08 | Rob3Rt | no firewalls etc. |
| 00:35.09 | seanbright | ~sipnat |
| 00:35.10 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:35.10 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 00:35.13 | seanbright | that one |
| 00:35.20 | Rob3Rt | ya read them both this morning and last nite |
| 00:35.23 | seanbright | ah |
| 00:35.34 | seanbright | i gotta run. watching the home run derby. |
| 00:35.57 | Rob3Rt | might be rtp |
| 00:36.02 | Rob3Rt | npz see ya later :) |
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| 01:10.49 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:12.01 | Rob3Rt | OK |
| 01:12.06 | Rob3Rt | CANT HEAR THE INCOMING VOICE |
| 01:12.13 | Rob3Rt | everything else is fine |
| 01:12.16 | Rob3Rt | outgoing calls |
| 01:12.18 | Rob3Rt | soz caps |
| 01:12.56 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
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| 01:16.47 | *** part/#asterisk sjobeck (n=Adium@137.118.193.9) |
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| 01:22.46 | {Sean} | is there anyway to set the SIP to header when calling a peer? |
| 01:22.46 | {Sean} | or user |
| 01:23.53 | *** join/#asterisk voip_troll (n=voip_tro@96.51.229.227) |
| 01:25.17 | voip_troll | What is the best way to configure asterisk as a fax server for multiple users? (Sales team each needs their own DID for faxing, but as the team grows, multiple instances of faxgetty probably isn't the most efficient method) |
| 01:25.51 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 01:26.48 | coppice | if you use app_fax or iaxmodem + hylafax the calling and called numbers are both available at the end of the call |
| 01:28.36 | Gnewt | Can I set line 2 to have a different dialplan than line 1? |
| 01:29.14 | *** join/#asterisk a2_ (n=a2@174-21-90-172.tukw.qwest.net) |
| 01:29.26 | a2_ | Anyone have a 1800 DID i can use that will connect me for free? |
| 01:30.09 | voip_troll | coppice: Do you know where the app_fax documentation can be found? So far Google has not been playing nicely:( |
| 01:30.09 | Gnewt | You're asking for a completely free 1800 DID? |
| 01:30.28 | a2_ | Im asking for a 1800 DID that is setup so i can call it, give it a number and it will connect me |
| 01:30.36 | a2_ | Im trying to get around payphone toll fees |
| 01:30.39 | Gnewt | Ah. |
| 01:30.40 | seanbright | {Sean}: core show application SIPAddHeader |
| 01:30.44 | Gnewt | Set up your own ;) |
| 01:30.58 | Gnewt | I don't actually know how much 1800 DIDs cost |
| 01:30.59 | a2_ | I dont have the resources :( |
| 01:31.02 | Gnewt | what're the prices like? |
| 01:31.14 | a2_ | idk, i hear they are cheap but im an amatuer |
| 01:31.21 | MikeJ | toll free from payphones is more expensive |
| 01:31.29 | MikeJ | they get charged a premium |
| 01:31.33 | a2_ | ok |
| 01:31.38 | a2_ | You guys might be able to answer this then |
| 01:32.02 | a2_ | I need to call a landline phone from a payphone. I just need the payphones caller id to showup on the landlines caller id, for free |
| 01:32.04 | a2_ | any suggestions? |
| 01:32.18 | a2_ | I was thinking of just depositing 50 cents and calling and not answering, and then calling the payphone back |
| 01:32.45 | Gnewt | Do you just need the payphone's number? |
| 01:32.49 | Gnewt | What is the projected goal? |
| 01:32.50 | a2_ | Kind of |
| 01:32.55 | voip_troll | a2_: Most payphones don't allow inbound dialing.... |
| 01:33.12 | a2_ | the ones where i live all have #'s and accept inbound |
| 01:33.15 | voip_troll | Hollywood has lied to you. |
| 01:33.29 | voip_troll | wow |
| 01:33.58 | {Sean} | can you replace the To header usisg SIPAddHeader |
| 01:34.26 | {Sean} | didn't see your msg Sean |
| 01:34.43 | {Sean} | slaps MikeJ with a large trout |
| 01:34.53 | MikeJ | :( |
| 01:35.34 | {Sean} | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPAddHeader |
| 01:35.34 | {Sean} | You can not replace SIP headers with this function, only add new ones. |
| 01:35.42 | {Sean} | says Voipinfo |
| 01:36.19 | Gnewt | a2_: What, in the end, is your expected outcome? |
| 01:36.24 | Gnewt | Do you need the payphone's #? |
| 01:36.28 | Gnewt | Do you need free calls? |
| 01:37.06 | a2_ | Gnewt: I need to get a payphone number to show up on a landline number without depositing any money, the landline phone is local to the payphone |
| 01:37.47 | a2_ | I would imagine that would be easy with the right knowledge |
| 01:40.41 | Gnewt | Hm. |
| 01:41.12 | Gnewt | You get your money back if the other end doesn't answer right? |
| 01:41.51 | a2_ | lol yea |
| 01:41.53 | a2_ | thats what i was thinking |
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| 01:42.10 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 01:42.17 | a2_ | I was just curious if there was a 1800 number i could dial or something that would show the payphones number without depositing anything |
| 01:42.28 | a2_ | i guess not |
| 01:43.01 | Gnewt | There may be |
| 01:43.03 | Gnewt | but I don't know of one |
| 01:43.05 | Gnewt | sorry man |
| 01:43.29 | a2_ | thanks for the help Gnewt |
| 01:44.01 | Gnewt | np |
| 01:44.03 | Gnewt | Also |
| 01:44.05 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:44.14 | Gnewt | I just looked at the Flowroute inbound rates page |
| 01:44.27 | Gnewt | You get charged 95c every time someone calls your toll-free from a payphone >_> |
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| 01:45.08 | a2_ | ok lol, thansk |
| 01:56.07 | iflux | is there anything like SIPStripHeader? |
| 01:57.23 | Rob3Rt | BAH |
| 01:57.36 | Rob3Rt | eveythings sorted cept when I call into the pbx two lines ring on the sipphone |
| 01:57.39 | Rob3Rt | ROFLLOLZ |
| 02:00.39 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 02:00.47 | ruben23 | hi |
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| 02:22.14 | voip_troll | Anyone configured hylafax and iaxmodem to support multiple DIDs with a single hylafax/iaxmodem instance? |
| 02:24.18 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 02:32.10 | DarthPointer | a2_ ; the payphone is probably from one of hte major LECs (ILECs); most LECs do implement a set of testing #'s for techs to use while out in the field; they do things like speak the # back to you, call you back, etc. Who's the LEC (or CLEC) |
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| 02:43.02 | Jumpie | anybody know when i do a sip show peers, all my aastra are like 30-40ms response, but my polycom (only have 1) is always under 8 |
| 02:43.09 | Jumpie | is this just an issue with the aastra's nic? |
| 02:43.22 | Jumpie | and its consistant |
| 02:44.45 | iflux | is it possible to use SIPAddHeader to overwrite a header that would have been there? |
| 02:45.41 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 02:47.01 | [TK]D-Fender | voip_troll: Single IAXmodem = single IAX modem. 1 identity. All you can pass on to HylaFax is the CallerID for routing the e-mail |
| 02:47.14 | [TK]D-Fender | iflux: No, you can only add headers |
| 02:48.09 | *** part/#asterisk {Sean} (n=sean@freeswitch/developer/Sean) |
| 02:48.23 | [TK]D-Fender | Jumpie: that is not a ping time, that is a SIP stack response time to an Options packet. The phone can choose to prioritize that lower and the score will look worse. It is not a generally valid test of latency |
| 02:48.37 | iflux | fender: but you should be able to accept a call, then dial another call and set whatever headers you want, correct? |
| 02:50.07 | [TK]D-Fender | iflux: Wrong order. Receive a call. Add headers. Dial. That Dialed call will have the added headers |
| 02:50.28 | *** join/#asterisk andres833 (n=andres83@190.156.210.248) |
| 02:50.59 | iflux | fender: yeah exactly what I mean.. |
| 02:51.24 | iflux | and the 2nd call could just be between the pbx and any extension, right? |
| 02:52.04 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 02:52.06 | [TK]D-Fender | iflux: * and a SIP device |
| 02:52.33 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 02:52.50 | iflux | ok.. thanks |
| 02:56.40 | *** join/#asterisk c4t3l (n=c4t3l@c-76-31-57-251.hsd1.tx.comcast.net) |
| 02:56.45 | c4t3l | howdy gang |
| 02:58.19 | c4t3l | n00b question here, but I'll ask anywayz... I've set up agents and a call queue in 1.4 and everytime my agent logs in I get hold music. When I hang up the receiver, the agent gets logged out. |
| 02:58.33 | c4t3l | i havn't played with * much since 1.2.16-ish |
| 02:58.42 | Jumpie | fender..ah ok makes sense..especially since its in line with all aastras show the same, so it shows i guess its a config difference |
| 03:00.00 | [TK]D-Fender | iflux: No way to strip the existing header AFAIK |
| 03:00.08 | c4t3l | i dont remember this being the default behavior |
| 03:01.12 | [TK]D-Fender | c4t3l: depends what your agent is doing by way of "logging in" |
| 03:02.12 | c4t3l | hmm. you mean what I'm passing via agentlogin app? |
| 03:03.07 | iflux | d-fender: hmmph :) |
| 03:03.13 | c4t3l | exten => *28,1,AgentLogin(${CALLERID(num)}) is what I'm sending |
| 03:03.26 | [TK]D-Fender | c4t3l: That cofirms it period. AgentLogin forces yuo to stay on that call and SIT THERE |
| 03:03.45 | c4t3l | which app should I be using? |
| 03:03.46 | [TK]D-Fender | c4t3l: You are "logged in" until you hangup |
| 03:03.53 | c4t3l | ok |
| 03:03.55 | [TK]D-Fender | c4t3l: Depends what you want to do |
| 03:04.41 | c4t3l | I want 3 sip phones to log into a queue and have calls routed to them in round-robin fashion |
| 03:05.14 | c4t3l | I dont need them to have to be "off-hook" the entire time tho |
| 03:05.44 | [TK]D-Fender | c4t3l: for 1.4 (forget this for 1.6+) "core show application AgentCallbackLogin" |
| 03:05.56 | eppigy | 8[] |
| 03:06.20 | c4t3l | [TK]D-Fender: you have never steered me wrong. thanks man |
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| 03:19.52 | Rob3Rt | hey so if u include a different dial plan |
| 03:20.02 | Rob3Rt | and that plan has the same extension as the current dialplan |
| 03:20.10 | Rob3Rt | will it call the extension twice ? |
| 03:20.25 | b14ck | nope |
| 03:20.36 | b14ck | it'll add onto it |
| 03:20.39 | Rob3Rt | let me show u something |
| 03:20.40 | Rob3Rt | hmm. |
| 03:21.31 | Rob3Rt | http://pastebin.com/m23d1e3a |
| 03:21.34 | Rob3Rt | intreresting. |
| 03:22.47 | b14ck | why are you surprised? |
| 03:23.17 | b14ck | never have outbound contexts where inbound contexts should be |
| 03:23.22 | b14ck | you leave yourself open for haxing D: |
| 03:23.42 | [TK]D-Fender | Rob3Rt: You have 2 priority 1's |
| 03:23.58 | [TK]D-Fender | BAD |
| 03:24.20 | Rob3Rt | HHMMM |
| 03:24.32 | Rob3Rt | oh |
| 03:24.35 | Rob3Rt | kool |
| 03:25.06 | Rob3Rt | i took out the outbound context and i only get the call on the correct (single) line now ;) |
| 03:25.06 | Rob3Rt | :)* |
| 03:25.14 | Rob3Rt | oh lol priority 1 i see |
| 03:25.23 | Rob3Rt | i changed that but mustnt have saved. |
| 03:25.42 | Rob3Rt | [TK]D-Fender ty too for all the help lately |
| 03:25.52 | Rob3Rt | turns out speedtouch blocked stuff internally. |
| 03:26.28 | Rob3Rt | so how owuld you guys set up that specific plan ? |
| 03:26.41 | Rob3Rt | if im gonna do this i want to do it the bet way |
| 03:26.42 | [TK]D-Fender | Rob3Rt: Yeah, its called NAT... the reason that that guide was written <- |
| 03:27.26 | Rob3Rt | nah nat was fine, internally the speedie blocks SIP 5060 even if you add the nat rule, you need to remove the rule |
| 03:38.07 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:42.50 | *** join/#asterisk fister (n=fister@91.197.11.243) |
| 03:45.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 03:59.40 | voip_troll | [TK]D-Fender: Would it be correct to assume I should define the same number of iaxmodems as I have channels for inbound faxing? |
| 04:06.34 | Rob3Rt | lolnick |
| 04:08.59 | [TK]D-Fender | voip_troll: How many channels are dedicated to it? |
| 04:10.47 | mmlj4 | realistically speaking, how much would a non-* conference bridge cost? this would hook into an existing avaya switch? ballpark figure? |
| 04:11.04 | mmlj4 | I've googled, but don't really know what I'm looking at |
| 04:13.01 | [TK]D-Fender | mmlj4: Novel idea : Call a reseller of one |
| 04:13.05 | voip_troll | [TK]D-Fender: 23 (full PRI) |
| 04:13.10 | *** join/#asterisk empiric (n=empiric@124.29.242.138) |
| 04:14.04 | [TK]D-Fender | voip_troll: Well if you expect taht kind of concurrency, then that answers itself |
| 04:14.29 | voip_troll | heh... ok :) |
| 04:15.06 | voip_troll | Other question... do you know if there's really any benefit to using hylafax/iaxmodem as opposed to app_rxfax/app_txfax? |
| 04:15.54 | [TK]D-Fender | voip_troll: IAXModem is more reliable and Hylafax more configurable and usable for real outgoing |
| 04:16.35 | voip_troll | ah, can I still use the built-ins for getting the file location of an inbound fax? |
| 04:18.05 | [TK]D-Fender | ...huh? |
| 04:19.38 | empiric | guys when i dial any no it says http://pastebin.com/m3b16da8 |
| 04:19.41 | empiric | any idea |
| 04:20.51 | [TK]D-Fender | voip_troll: empiric Says what? |
| 04:20.56 | [TK]D-Fender | empiric Says what? |
| 04:25.48 | empiric | what |
| 04:25.56 | empiric | Fender any idea? |
| 04:27.47 | [TK]D-Fender | empiric: Idea about what? You haven't decribe what is "bad" about what you have shown us. |
| 04:28.04 | [TK]D-Fender | empiric: described* |
| 04:28.36 | kb3ien | i've found hylafax buggy i use efax with my iaxmodem. |
| 04:35.43 | empiric | ok wait |
| 04:36.51 | voip_troll | [TK]D-Fender: I'm trying to figure out how to get the name of the file that the fax is stored in during/after transmission, so I can trigger additional logic. |
| 04:37.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:37.34 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 04:37.53 | [TK]D-Fender | voip_troll: well that has nothing to do with * with HylaFAX doing it |
| 04:38.21 | empiric | here is my dail plan |
| 04:38.22 | empiric | http://pastebin.com/m503ae31 |
| 04:38.49 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:41.48 | empiric | Fender when i dial why it says Spawn extension (default, 03122201455, 3) exited non-zero on 'SIP/205-0825ce40' |
| 04:41.51 | empiric | and hung up |
| 04:41.55 | empiric | it wont call |
| 04:42.36 | [TK]D-Fender | empiric: Won't call what? http://pastebin.com/m3b16da8 <-- I see it placing a call |
| 04:42.49 | [TK]D-Fender | empiric: You seem to have a real issue describing WHAT IS BAD. |
| 04:43.06 | empiric | yes true |
| 04:43.21 | empiric | it seems calling my on other end i wont recieve call |
| 04:43.34 | [TK]D-Fender | empiric: Passes back progress which is presumably "ringing" and then gets answered. |
| 04:43.40 | empiric | and after 25 sec its says exited non zero |
| 04:44.28 | [TK]D-Fender | empiric: Says it answered. Maybe you should give us some sort of useful description about what you are calling exactly. MAYBE even place another callw ith SIP DEBUG enabled so we can see the comunication attempt |
| 04:45.18 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 04:46.26 | empiric | it was all working last night |
| 04:47.21 | [TK]D-Fender | empiric: You have still not described anything useful about what is wrong. You haven't shown SIP debug for the call. And you haven't even described what you are doing. |
| 04:48.34 | empiric | wait doing |
| 04:50.20 | empiric | fender ill let u kn later sorry |
| 04:50.23 | empiric | i have to leave |
| 04:50.58 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
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| 04:55.46 | Rob3Rt | i dont understand |
| 04:55.58 | Rob3Rt | if i ring into my DID i get two calls on my sipphone |
| 04:56.07 | Rob3Rt | what a freakin mess who coded this sh1t? |
| 04:57.33 | velxundussa | Hey peoples, i'm completly new to Asterisk, i've begun reading some docs and got this channel, i'd just like to know if asterisk's the good thing for what I wish to do: i'd like to call up my PC with a cell-phone and get the call routed somewhere else with some voip. Is asterisk able to do that? Thanks for the answer :) |
| 04:58.11 | Rob3Rt | yeah but youll be charged for both calls u know that right ? |
| 04:58.23 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-0d029278facbc0f7) |
| 04:58.40 | velxundussa | yup, it's for avoiding some extra with long-distances call :) |
| 04:58.56 | Rob3Rt | i thought mobile was more expensive lol |
| 04:59.01 | Rob3Rt | but whatever works |
| 04:59.41 | velxundussa | Kay :) (i've got 5 unlimited numbers on my mobile.. so if i get my asterisk box on them, all my call'll be unlimited :-) ) |
| 05:00.02 | velxundussa | thanks for the answer! =D |
| 05:00.30 | [TK]D-Fender | velxundussa: You can do whatever you want with calls in to your system |
| 05:02.22 | Sargun | # 'T' ? set talker detection (sent to manager interface and meetme list) |
| 05:02.33 | Sargun | What does that do? |
| 05:02.38 | Sargun | (meetme) |
| 05:03.14 | [TK]D-Fender | Sargun: Means that it sends out messages when it detects a caller speaking |
| 05:03.24 | Sargun | Oh, neat. |
| 05:04.33 | Sargun | Hm, I'm implementing a Omegle for phones, I'm curious as to how asterisk scales, or to handle a lot of the complexity outside of asterisk |
| 05:04.33 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 05:08.02 | [TK]D-Fender | Sargun: meaning? |
| 05:08.14 | Sargun | [TK]D-Fender, do you know what omegle is? |
| 05:08.34 | [TK]D-Fender | Sargun: No |
| 05:08.45 | Sargun | It's a way to chat with strangers over the internet |
| 05:08.47 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 05:08.58 | Sargun | I basically want to do the same thing for phones |
| 05:09.42 | Sargun | The way I'm thinking of implementing is by having people call in, have an extension that just Wait()s. Then I have an AMI program which manages each call |
| 05:09.54 | Sargun | and when two calls are free, it creates a conference, and then the users join |
| 05:10.04 | Sargun | basically, as much sits outside of asterisk as possible |
| 05:10.17 | Sargun | I'm wondering if it might be wise to put more into the dial plan |
| 05:10.43 | *** join/#asterisk AlmightyOatmeal (n=jamie@66-190-59-171.dhcp.mdsn.wi.charter.com) |
| 05:11.00 | AlmightyOatmeal | do polycom phones require a constant tftp connection like cisco phones do? |
| 05:12.39 | [TK]D-Fender | AlmightyOatmeal: No |
| 05:12.56 | AlmightyOatmeal | thank god |
| 05:13.23 | AlmightyOatmeal | any recomendations on a particular polycom with basic functionality? one or two lines, CID, hold, transfer, etc? |
| 05:13.33 | *** join/#asterisk maxagaz (n=maxagaz@124.205.74.34) |
| 05:13.51 | AlmightyOatmeal | i don't need a huge conference setup or video |
| 05:15.58 | [TK]D-Fender | AlmightyOatmeal: Any will do. IP 321/331 |
| 05:16.15 | AlmightyOatmeal | nice |
| 05:16.17 | AlmightyOatmeal | :) |
| 05:27.44 | Sargun | hm |
| 05:27.56 | AlmightyOatmeal | hmrmrmrm |
| 05:28.33 | *** join/#asterisk bmg505 (n=leon@196-209-8-30-rndf-esr-5.dynamic.isadsl.co.za) |
| 05:29.33 | *** join/#asterisk TrentCreek (n=kvirc@adsl-99-20-31-130.dsl.hrlntx.sbcglobal.net) |
| 05:30.53 | TrentCreek | anyone know of a ATA to USB to connect to * without the need for drivers for a computer that may be locked down? |
| 05:31.36 | [TK]D-Fender | TrentCreek: No such thing |
| 05:31.41 | AlmightyOatmeal | i've never heard of such a device |
| 05:31.56 | [TK]D-Fender | And this is a conversation I've had before. |
| 05:32.02 | TrentCreek | well, one can wish |
| 05:32.08 | AlmightyOatmeal | TrentCreek: make one |
| 05:32.29 | [TK]D-Fender | TrentCreek: Or invent it yourself. Frankly noone has seen a market enough to waste dev time making it |
| 05:32.39 | TrentCreek | problem is is needs to share the IP socket |
| 05:32.57 | [TK]D-Fender | TrentCreek: Oh yeah, like there wouldn't be problems ;) |
| 05:33.20 | [TK]D-Fender | TrentCreek: Like WTF does it become a packet interface with NO DRIVERS. and the have the right to route.... |
| 05:33.42 | TrentCreek | I had a relative that was using a company owned laptop and restricted from putting softwar eon it. Wanted to use it to cal home. No doing |
| 05:34.20 | [TK]D-Fender | TrentCreek: Can he run an app? |
| 05:34.39 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 05:34.41 | TrentCreek | run, yes, but not install |
| 05:34.45 | [TK]D-Fender | TrentCreek: Then have him run Zoiper and get a hedset and he's done |
| 05:34.55 | *** join/#asterisk soman (n=somnath@115.119.41.98) |
| 05:35.39 | TrentCreek | Okay, I never knew that one did nto need install. I used another one, and it really sucks on sound quality |
| 05:37.58 | TrentCreek | [TK]D-Fender: It seems to have a install package. Is there another version, or does it auto-detect? |
| 05:38.59 | [TK]D-Fender | look around. I know it doesn't require installation |
| 05:39.20 | [TK]D-Fender | Checkout time. LAter all |
| 05:39.47 | *** join/#asterisk muthah (n=muthah@41.212.27.10) |
| 05:39.49 | soman | Hi, I would like to test basic call setup from VOIP to PSTN, and i have ordered TE122P T1/E1 card from digium. And I have an ISDN phone. Can i connect ISDN phone to TE122p and test the same. |
| 05:44.31 | Rob3Rt | rofl |
| 05:44.40 | Rob3Rt | woulda investigated that prior to purchase |
| 05:44.43 | Rob3Rt | sorry i cant |
| 05:45.49 | soman | Rob3Rt: means will it not work if i connect TE122p to ISDN phone |
| 05:46.17 | Rob3Rt | i dont know, does it have any isdn/bri ports/sockets/trunks ? |
| 05:46.53 | soman | My ISDN phone has WAN interface, |
| 05:47.19 | soman | and TE122p is with PRI interface |
| 05:47.38 | soman | Can you please suggest me, how can i proceed. |
| 05:48.40 | soman | Can i connect T1/E1 card to my ISDN phone |
| 05:49.50 | Rob3Rt | i cant no, i havent used ISDN ... EVER |
| 05:51.01 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 05:51.02 | soman | Rob3Rt, My requirement is that i would like to develop supplementary applications over PSTN->VOIP and vice versa. |
| 05:51.21 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 05:52.09 | soman | Rob3Rt, i am following the link http://www.cesnet.cz/doc/techzpravy/2006/asterisk-ss7/ , and would like to use chan_ss7 solution for that. |
| 05:52.17 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 05:53.01 | soman | Rob3Rt, instead of using two TE110P cards on two machines, I wanted to use one card and at the other end ISDN phone. |
| 05:54.28 | soman | Rob3Rt, Can you tell me how can i proceed. |
| 05:54.44 | soman | Rob3Rt, I appreciate any help on this. |
| 05:56.36 | soman | Can i use TE122p T1/E1 card to connect with the ISDN phone which has WAN interface |
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| 06:12.39 | *** part/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 06:14.10 | Rob3Rt | <Rob3Rt> i cant no, i havent used ISDN ... EVER |
| 06:28.09 | *** join/#asterisk xrmx__ (n=rm@host129-254-dynamic.2-87-r.retail.telecomitalia.it) |
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| 06:56.46 | Rob3Rt | Jul 14 16:54:52 WARNING[2772] chan_sip.c: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 06:56.51 | Rob3Rt | anyone know what gives? |
| 06:57.07 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 06:57.09 | Rob3Rt | That when I call the pizza place and the call cant be heard, IM HUNGRY !!! |
| 07:02.32 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-27-9.w90-56.abo.wanadoo.fr) |
| 07:04.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:07.16 | merlin8282 | If I have understood it right, mISDN and DAHDI are both a driver for HFC4/8S hardware, and can't be used at the same time. Am I right ? |
| 07:07.24 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
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| 07:21.40 | lftsy | Rob3Rt: Hello, I'm interested in your question, I has the same warning lines yesterday |
| 07:22.26 | lftsy | Rob3Rt: But I suppose it was when I was sending g729 to be transcoded when the transcoding card was fully occupied |
| 07:33.44 | WindowsUser | merlin8282: cant use two drivers for one device, but i dont know anything about your specific hardware :) |
| 07:40.47 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 07:43.54 | merlin8282 | WindowsUser: ok, thanks. |
| 07:45.38 | merlin8282 | In fact I just tried this: adding hisax, isdn, hfc4s8s_l1, slhc and crc_ccitt to /etc/modprobe.d/blacklist (like on my production asterisk 1.4.22 server) and rebooted, but it keeps unsuccessful, the driver seems to crash // |
| 07:45.50 | merlin8282 | I've a junghanns QuadBRI ISDN PCI. |
| 07:47.12 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 07:49.54 | tzafrir_laptop | merlin8282, what do you want to use? misdn? zaptel? dahdi? |
| 07:51.16 | merlin8282 | tzafrir_laptop: at least something that works ^^ |
| 07:51.29 | merlin8282 | What are the big differences ? |
| 07:51.36 | merlin8282 | I'm using * 1.6.1.1 |
| 07:52.02 | tzafrir_laptop | merlin8282, I'm not familiar with misdn |
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| 08:03.06 | Rob3Rt | i still have absolutley nfi why, |
| 08:03.39 | Rob3Rt | when i dialin to my stupid pbx(exetel voip DID), it rings 3cx via 1000 (great), and then calls the sipphone twice. |
| 08:06.16 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 08:11.37 | merlin8282 | Ok, so I think I would use DAHDI. |
| 08:11.56 | merlin8282 | Does it include the qozap driver ? |
| 08:14.20 | tzafrir_laptop | what version of dahdi-linux ? |
| 08:15.34 | merlin8282 | dahdi-linux-2.2.0.1 |
| 08:25.02 | supa_disko | guys: on my internal firewall, I've let rtp/sip from networks to any, and any back |
| 08:25.19 | supa_disko | however I don't get incomding audio, anything I should be looking at? |
| 08:26.04 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 08:26.42 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 08:33.06 | tzafrir_laptop | merlin8282, that version still does not include support for that card (it exists as a patch, and was commited into dahdi a bit after that) |
| 08:34.05 | merlin8282 | Oh ok. So I should use the svn version of dahdi ? |
| 08:34.37 | tzafrir_laptop | hmm, I believe it should work |
| 08:37.55 | merlin8282 | ok thanks, i'll try this. |
| 08:39.33 | Rob3Rt | when i dialin to my stupid pbx(exetel voip DID), it rings 3cx via 1000 (great), and then calls the sipphone twice. |
| 08:39.54 | Rob3Rt | what can i look for to fix? |
| 08:40.03 | supa_disko | go go exetel |
| 08:40.12 | supa_disko | complete @#$@#ers if you ask me |
| 08:41.12 | Rob3Rt | why ? |
| 08:41.28 | Rob3Rt | exetel aint the problem |
| 08:41.36 | Rob3Rt | and thier irc channel is cool |
| 08:41.38 | Rob3Rt | but |
| 08:41.53 | Rob3Rt | my F*&^*&^(* sipphone getting called twice IS a problem. |
| 08:44.38 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:45.09 | tzafrir_laptop | notes that if you remove that 3cx that problem will go away |
| 08:45.52 | Rob3Rt | not removing it |
| 08:45.58 | Rob3Rt | http://pastebin.com/m5766c1f9 |
| 08:46.05 | Rob3Rt | extension plan |
| 08:46.12 | *** join/#asterisk ctp (n=ctp@brsg-d9bee6a0.pool.mediaWays.net) |
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| 09:35.07 | a2__ | Whats the best per minute DID service? |
| 09:35.40 | a2__ | I found a service thats offering $1 a pop per DID and like $0.6c-$0.30c per minute..is that the cheapest? Im pretty sure you can get them for cheaper |
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| 10:21.41 | maxagaz | hi |
| 10:23.17 | maxagaz | when i call an external number using an SIP phone, and that they ask me to choose for a language, the digits don't work, can someone help me to solve this problem ? |
| 10:23.24 | *** join/#asterisk sack (n=sack@29.Red-81-33-112.dynamicIP.rima-tde.net) |
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| 10:34.42 | *** join/#asterisk ernetas (n=Ernestas@78-60-212-86.static.zebra.lt) |
| 10:34.45 | ernetas | Hi guys. |
| 10:34.50 | ernetas | I'm using ArchLinux. |
| 10:34.54 | ernetas | And just installed Asterisk. |
| 10:35.15 | ernetas | And after I start /etc/rc.d/asterisk - there's nothing changeing in nmap -sS localhost output. |
| 10:35.19 | ernetas | Any ideas why? |
| 10:39.08 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 10:42.28 | tzafrir_laptop | ernetas, for starters, 'netstat -lntup' is probably more handy |
| 10:42.48 | tzafrir_laptop | and then again, start with the logs under /var/log/asterisk |
| 10:43.11 | ernetas | Oh. I've totally forgot about those... :D |
| 10:43.37 | ernetas | Ghm... |
| 10:43.40 | ernetas | They're empty. |
| 10:43.56 | ernetas | Only the queue_log contains some lines of this: 1247501428|NONE|NONE|NONE|QUEUESTART| |
| 10:44.08 | ernetas | And netstat doesn't see any asterisk. |
| 10:44.28 | ernetas | Although, ps aux | grep asterisk does see: |
| 10:44.34 | ernetas | asterisk 14201 0.0 0.2 4152 1168 ? Ssl 13:41 0:00 /usr/sbin/asterisk -G asterisk -U asterisk |
| 10:45.43 | ernetas | Any ideas what's wrong? :) |
| 10:46.31 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 10:47.13 | tzafrir_laptop | the process started, but failed to bind to ports for whatever reason? |
| 10:47.56 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:48.04 | *** part/#asterisk Any (n=Any@elxsi.de) |
| 10:51.02 | maxagaz | could someone help me to formulate better my question ? |
| 10:54.16 | *** join/#asterisk J4zen (n=j4zen@a82-95-153-17.adsl.xs4all.nl) |
| 10:55.14 | tzafrir_laptop | ernetas, does asterisk run as its own user? if so: does it have write permissions to /var/log/asterisk ? |
| 10:56.23 | ernetas | tzafrir_laptop: how do I check if it's being run as its own user? :) Ain't that shown in ps aux? |
| 10:56.48 | ernetas | Yeah, it does have writing permissions. |
| 10:56.50 | J4zen | Hi there, i'd like to replace my incoming Callerid(name) by the output that an external PHP script will give me. The script accepts a parameter $cid and will echo a name if it's found in our database. How would i do so? Can i do "exten => _X.,1,Set(CALLERID(name=System(curl [url]?cid=${CALLERID(num)}))) |
| 10:57.00 | J4zen | I dont think thats the proper way of doing it |
| 10:57.31 | tzafrir_laptop | endemic, it does run as user 'asterisk' (from the output of ps) |
| 10:58.56 | J4zen | In short, i'd like to put the output of CURL into CALLERID(name) |
| 10:59.33 | tzafrir_laptop | J4zen, there's a function called SHELL() in later versions of Asterisk |
| 10:59.52 | tzafrir_laptop | hmm.... sorry. CURL is a function of its own right |
| 11:00.03 | J4zen | it is? |
| 11:00.17 | J4zen | i was refering to the unix curl |
| 11:00.35 | J4zen | ah, thanks tzafrir_laptop. I had no idea |
| 11:00.37 | J4zen | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+curl |
| 11:00.45 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 11:00.50 | mvanbaak | heya all |
| 11:00.55 | tzafrir_laptop | hi |
| 11:01.12 | mvanbaak | anyone know about a spanish or mexican ITSP ? |
| 11:13.04 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:14.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 11:18.03 | J4zen | Does anyone see anything wrong with this syntax: |
| 11:18.03 | J4zen | exten => _X.,1,Set(CALLERID(name)=${CURL(https://mydomain.tld/setcid?password=mypassword&cid=${CALLERID(num))}}) |
| 11:18.25 | J4zen | it should put the output of CURL into variable CALLERID(name) |
| 11:21.41 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:23.41 | mvanbaak | J4zen: make sure you have a valid cert on mydomain.tld or import the cert into your local keyring |
| 11:23.54 | J4zen | its a valid cert |
| 11:24.04 | mvanbaak | I have seen a lot of stuff fall apart when moving from testing to production because of wrong ssl certs |
| 11:24.11 | mvanbaak | ah, then I said nothing |
| 11:24.13 | J4zen | :) |
| 11:26.38 | J4zen | Is there anyone that knows how to put the output of an external PHP script into a local asterisk variable? |
| 11:29.13 | creativx | using set() and curl() .. |
| 11:29.53 | creativx | exten => s,n,set(url=https://test.com/app.php) |
| 11:30.07 | creativx | exten => s,n,set(qstr=?param1=1) |
| 11:30.37 | creativx | exten => s,n,set(foo=${CURL(${url}${qstr})}) |
| 11:30.52 | creativx | URI split to 2 vars for readability.. |
| 11:31.28 | J4zen | and that should work equally well for set(callerid(name)${CURL(${url}${qstr})}) ? |
| 11:31.36 | J4zen | callerid(name)= * |
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| 11:41.51 | creativx | yeah |
| 11:42.20 | maxagaz | still nobody to help me for my problem ? |
| 11:43.04 | *** join/#asterisk tompaw (n=tompaw@slave20.tesserakt.eu) |
| 11:43.09 | tompaw | Hi. |
| 11:43.20 | tompaw | Is there a way for Asterisk to send a custom SIP error message? |
| 11:43.29 | tompaw | like SIP(error_code, error_text) |
| 11:45.35 | tompaw | --or-- |
| 11:45.54 | tompaw | is there a way to send 503 / Service unavailable? |
| 11:46.25 | kaii | tompaw: you can do this with hangup causes .. Hangup(17) for example is "normal call clearing" |
| 11:46.57 | kaii | (or was it busy?) have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+Hangup |
| 11:48.23 | kaii | maxagaz: possibly your SIP phone and the SIP peer setting in asterisk do not match with the right DMTF settings |
| 11:48.42 | *** join/#asterisk stix (n=stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 11:49.37 | stix | Hi guys. I am placing calls from telnet via AMI. Can anyone tell me if I can use a command to record the calls to a wav? |
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| 11:52.47 | tompaw | kaii: checking Hangup, thx |
| 11:53.07 | kaii | stix: see manager command "Monitor" |
| 11:53.19 | stix | thanks :) |
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| 12:17.05 | ThoMe | hello |
| 12:17.39 | ThoMe | is it posible with AEL when I use "if" with if (${BLA} = 'ups' && <<the AND ? |
| 12:18.15 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 12:20.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 12:20.47 | [TK]D-Fender | ThoMe: I'm pretty certain it is |
| 12:22.00 | ThoMe | [TK]D-Fender: ok.. i try it. have a asterisk book and is not written in my book |
| 12:22.04 | ThoMe | [TK]D-Fender: i think, the best> test .-) |
| 12:22.13 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:22.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:22.36 | [TK]D-Fender | ThoMe: I think limiting yourself to 1 book and ignoring all the basic docs * even comes with is a big mistake |
| 12:22.54 | ThoMe | jo |
| 12:27.14 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 12:27.23 | J4zen | creativx: Thanks, works like a charm |
| 12:29.08 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 12:29.26 | *** join/#asterisk ingenius (n=alektro@host38.190-138-197.telecom.net.ar) |
| 12:32.13 | ThoMe | [TK]D-Fender: is it posible with asterisk, |
| 12:32.27 | ThoMe | when i try to dial a ext.. play music? |
| 12:32.58 | ThoMe | like dial(sip/me); but with music.. and when connected then music stop |
| 12:33.11 | kaii | Dial(sip/me,,m) |
| 12:33.21 | stix | What is the command "show channels" replaced with in 1.6.x ? |
| 12:33.21 | kaii | option "m" = "m"usic on hold |
| 12:33.27 | ThoMe | ajo.. ok :-) |
| 12:33.33 | guax | stix, core show channels |
| 12:33.36 | [TK]D-Fender | ThoMe: "core show application" <--- read the instructions first |
| 12:33.38 | stix | oki |
| 12:33.46 | ThoMe | kaii: kaii and the music on hold, i can set this? i have a default music... |
| 12:33.56 | guax | stix, as well in 1.4, but it still works for compactibility |
| 12:33.56 | ThoMe | kaii: but i can change it for this call? |
| 12:34.07 | kaii | ThoMe: read instructions ... Dial(SIP/me,,m(class)) |
| 12:34.53 | ThoMe | ok |
| 12:35.52 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:38.06 | stix | How can I monitor channel SIP/100 when the channel is called something like: SIP/100-0954e8d0? and 0954e8d0 seems to be a new random number on the next call. |
| 12:39.22 | ThoMe | on my asterisk console i have message like |
| 12:39.23 | ThoMe | Really destroying SIP dialog '0648b98d0792870f481db915047047e3@127.0.0.1' Method: REGISTER |
| 12:39.26 | ThoMe | or |
| 12:39.26 | ThoMe | [Jul 14 14:38:16] NOTICE[17949]: chan_sip.c:15769 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 53066 |
| 12:39.32 | ThoMe | is it posible set verbose none? |
| 12:39.39 | ThoMe | have set verbose 0 and sip no debug |
| 12:39.45 | [TK]D-Fender | ThoMe: Of course |
| 12:40.05 | ThoMe | i try core set debug |
| 12:40.11 | [TK]D-Fender | ThoMe: "set debug 0 |
| 12:40.14 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:40.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:40.15 | ThoMe | ah ok |
| 12:40.19 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 12:40.19 | ThoMe | [TK]D-Fender: mh. ok. :-) |
| 12:42.47 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 12:46.01 | *** join/#asterisk takashi_85 (n=ahmed@unaffiliated/takashi-85/x-0528375) |
| 12:46.04 | *** part/#asterisk takashi_85 (n=ahmed@unaffiliated/takashi-85/x-0528375) |
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| 12:51.07 | stix | How can I monitor channel SIP/100 when the channel is called something like: SIP/100-0954e8d0? and 0954e8d0 seems to be a new random number on the next call. |
| 12:53.13 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 12:53.29 | [TK]D-Fender | stix: monitor how? |
| 12:54.27 | *** join/#asterisk mazpe (n=mazpe@c-98-203-52-170.hsd1.fl.comcast.net) |
| 12:54.46 | stix | [TK]D-Fender: I have a php-script which sends commands via telnet to the AMI. It should send the "Monitor"-command but my script doesn't know this random peer-id which is suffixed on the channel-name |
| 12:55.09 | [TK]D-Fender | stix: Well use AMI to get a LIST of the channels |
| 12:55.28 | Rob3Rt | [TK]D-Fender, my aster-bible. Sir, what other directive are imperative to sip.conf and extensions.conf, id like to clear the default junk out of my configs so i see better. |
| 12:55.36 | stix | I guess I could do that |
| 12:56.12 | [TK]D-Fender | Rob3Rt: only [general] and [globals] are "standard" |
| 12:56.18 | *** join/#asterisk ctp (n=ctp@brsg-d9bee6a0.pool.mediaWays.net) |
| 12:56.33 | Rob3Rt | Thank you - looking into it :) |
| 12:58.07 | mazpe | I keep getting the following error "rtp.c:1010 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: XXX.XXX.XXX.XXX" |
| 12:58.18 | mazpe | the client ip is the ip of my voip provider. |
| 12:58.27 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 12:58.47 | mazpe | it seems to come on when i first press a number on the IVR |
| 12:58.50 | Rob3Rt | [TK]D-Fender, Globals is in sip.conf too ? or is that just in Extensions.conf ? |
| 12:59.19 | [TK]D-Fender | Rob3Rt: extensions.conf. in sip.conf, only [general] is standard and required |
| 12:59.29 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 12:59.45 | Rob3Rt | Great thanks man. |
| 13:01.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:03.49 | stix | [TK]D-Fender: how would you return the channel-name of SIP/100? If I use CoreShowChannels, it shows all of them. |
| 13:04.44 | [TK]D-Fender | stix: If it returns a whole bunch, are you telling me you can't parse out the names for the one that starts with SIP/100 ? |
| 13:05.21 | stix | sure I can do it by grep'ing - but maybe there was an asterisk-command which could do it |
| 13:07.37 | Rob3Rt | [TK]D-Fender, started afresh, i now have 'sip show peers' showing as one extension (good) and one inbound and one outbound extension via my voip provider, is this correct? |
| 13:08.06 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 13:08.59 | [TK]D-Fender | Rob3Rt: I don't know... is it? |
| 13:10.16 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:12.22 | Rob3Rt | Probably not.. |
| 13:12.23 | Rob3Rt | p |
| 13:12.49 | Rob3Rt | got this since i cleared the conf, Unable to find a codec translation path from g729 to slin |
| 13:12.54 | Rob3Rt | on incoming calls |
| 13:12.56 | Rob3Rt | i hate life. |
| 13:12.57 | Rob3Rt | :p |
| 13:15.30 | mvanbaak | Rob3Rt: looks like you dont have the g729 codec installed |
| 13:15.47 | [TK]D-Fender | Rob3Rt: I'm largely doubting that you purchased G729 licenses and set them up |
| 13:16.24 | Rob3Rt | works as pass thru doesnt it |
| 13:16.48 | [TK]D-Fender | Rob3Rt: More specifically I'm betting you did not bother to specify your codecs under [general] and for each of your peers |
| 13:16.50 | mazpe | also.. what causes the following error: chan_sip.c:2686 ast_sip_ouraddrfor: stun failed |
| 13:16.59 | [TK]D-Fender | Rob3Rt: Whatever you're doing isn't passthrough |
| 13:17.05 | Rob3Rt | yeah i did specify all three codecs under peers |
| 13:17.39 | [TK]D-Fender | Rob3Rt: I'm not going to validate what I can't see |
| 13:17.42 | Rob3Rt | [TK]D-Fender, it is passthrough, when the asterisk answers incoming calls for an internal pbx, and passes them on |
| 13:17.44 | Rob3Rt | yeah ok |
| 13:18.05 | [TK]D-Fender | Rob3Rt: yeah... and I trust you did it right... suuuurrrreee |
| 13:18.07 | Rob3Rt | im on vista running win32 in console mode, but it wont let me right click and mark for paste, I jsut cant win lol. |
| 13:18.28 | Rob3Rt | it was working fine just i was getting the sipphone called twice for some reason |
| 13:18.34 | [TK]D-Fender | Rob3Rt: You've PB'd this stuff before. |
| 13:18.40 | Rob3Rt | brb configuring stuff |
| 13:19.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:21.16 | [TK]D-Fender | "Everything is right, except that it needs configuring" |
| 13:25.11 | Rob3Rt | Err no |
| 13:25.18 | Rob3Rt | started from scratch. |
| 13:25.21 | Rob3Rt | its working now . |
| 13:25.34 | Rob3Rt | few things to sort through but from-scratch is much better. |
| 13:27.04 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 13:29.54 | *** join/#asterisk iflux (i=iconicfl@www.kevinlynn.com) |
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| 13:37.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:38.46 | jaytee | anyone know of a website where I can input my area code and get a list of all nxx exchange numbers for that area code? |
| 13:39.55 | iflux | jaytee: there's one off cohutta.com |
| 13:40.08 | stix | Is the sip-peer-id, displayed after the channel-name, always 8 characters long? |
| 13:40.57 | iflux | http://www.cohutta.com/npanxx.php |
| 13:41.31 | [TK]D-Fender | stix: If youre referring to the "-" + 8 digit hex, is jsut a random sequence soo that the channel name is "unique" (or more). |
| 13:42.37 | iflux | fender: I've almost got this stuff I've been working on fixed now.. just a couple more gotoif's and some parsing tricks and I'm done.. |
| 13:43.50 | iflux | then I just gotta ask my work if I can release it and hope they don't claim ownership :( |
| 13:45.12 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:46.31 | stix | [TK]D-Fender: yes that is what I mean - called peer-id on voip-info.org. As long as it is always 8 characters I am happy. |
| 13:46.54 | [TK]D-Fender | stix: Parse backwards until a "-" |
| 13:47.59 | stix | Which means that it is not always 8? |
| 13:48.07 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 13:48.21 | [TK]D-Fender | stix: Which means it certainly should be... for SIP. not Zap, etc |
| 13:48.32 | [TK]D-Fender | stix: So use a more generalized method |
| 13:49.10 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:49.12 | stix | it is always SIP, so I can match it with /SIP/100-[0-9a-f]{8} |
| 13:49.38 | shazaum | 100/100 127.0.0.1 D 5061 OK (1372 ms) <----- :D |
| 13:50.00 | iflux | so I have this annoying neighbor's grandson that's 10.. his folks drop him off at his grandparents for a week in the summer and they just tell him to go do something outside but he's not allowed to go anywhere out of the neighborhood. He keeps coming over to my house and knocking on the door wanting to see if our nanny and my daughter (who's 3) want to go outside. I've told him not to knock on my door or ring my doorbell or go in my garage (caught him |
| 13:51.24 | iflux | I should go put a computer outside.. hook it up to the net.. and let the kid vegetate all day.. |
| 13:52.15 | KavanS | iflux, I'd hang out with him and teach him how to be a man |
| 13:52.36 | KavanS | one of the coolest things you get to do as an adult is mold the future generations... |
| 13:53.58 | iflux | kavan: that's true.. but my idea of being a man and his father's seem to be very different. |
| 13:54.31 | KavanS | iflux, yeah you can't do much about that |
| 13:54.48 | Katty | shivers. |
| 13:54.56 | leifmadsen | shudders |
| 13:54.59 | iflux | yesterday the kid was over at my house talking about how his father doesn't help his mom out whatsoever with money and they had to move out of their house because his dad wouldn't pay for it anymore |
| 13:55.09 | Katty | leifmadsen: the building is 70F :< |
| 13:55.21 | leifmadsen | Katty: mine is about 74F and I'm just wearing shorts |
| 13:55.26 | Katty | leifmadsen: >.< |
| 13:55.30 | Katty | we need blankets. |
| 13:55.41 | leifmadsen | I generate a lot of heat |
| 13:55.53 | Katty | leifmadsen: can i borrow you? |
| 13:55.53 | leifmadsen | I'm almost always hot |
| 13:56.00 | iflux | kavan: so I had to explain to him that if his dad did help his mom pay for that house he probably couldn't afford a place of his own and that's not fair to expect his dad to be homeless |
| 13:56.03 | leifmadsen | sure! I'm like a heating blanket |
| 13:56.17 | iflux | you could tell his mom has been filling his head with stupid crap |
| 13:56.34 | leifmadsen | people suck |
| 13:56.36 | leifmadsen | that is all. |
| 13:56.44 | iflux | well said leifmadsen |
| 13:57.26 | iflux | on another note.. I need a thermostat that I can get onto my network in some way so I can tell what time of day someone changed the temperature |
| 13:57.38 | [TK]D-Fender | leifmadsen: I BLAME YOU! Yesterday we lost power, got hit was torrential rainfall (45 deg incline), and HAIL! |
| 13:57.46 | iflux | I have a suspicion that our nanny keeps turning the temp down to 70 degrees but I can't prove it right now |
| 13:57.52 | leifmadsen | [TK]D-Fender: mwahahaha... I did it on purpose |
| 13:57.56 | leifmadsen | just to screw with you |
| 13:58.04 | iflux | fender: how big was the hail? |
| 13:58.05 | [TK]D-Fender | KNEW it... |
| 13:58.23 | leifmadsen | iflux: a plastic cover with a lock should solve that problem |
| 13:58.37 | [TK]D-Fender | iflux: dime-sized at largest thankfully |
| 13:58.39 | leifmadsen | can't wait for his tee-time at the virtual golf in his condo tonight |
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| 13:58.59 | Katty | nanny? |
| 13:59.04 | Katty | why on earth do you have a nanny. |
| 13:59.15 | file | [TK]D-Fender: that rain was fun. |
| 13:59.22 | iflux | leifmadsen: true.. but I want a fancy networked thermostat and those covers have zero wife acceptance factor |
| 13:59.26 | Katty | [TK]D-Fender: file: http://www.youtube.com/watch?v=dpxzvUtrHOE |
| 13:59.35 | iflux | katty: 2 parents that work and need a flexible schedule |
| 13:59.39 | leifmadsen | iflux: hehe... true :) plus the networked one is way cooler |
| 13:59.42 | [TK]D-Fender | leifmadsen: I got fubar'd on one of those. Stupid Pebble Beach with a huge crevasse they won't let you shout AROUND. I swear the specks at the bottom weren't sand, they were the 30 stroke wasted FILLING IT |
| 13:59.46 | iflux | leif: exactly |
| 13:59.53 | file | Katty: can't watch, in a presentation |
| 13:59.54 | Katty | iflux: your child is going to turn out like your nanny, and not like you. |
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| 14:00.17 | Katty | iflux: i would put your flexibilities and priorities to favor your child. |
| 14:00.28 | Katty | file: watch it later (= |
| 14:00.39 | Katty | iflux: you will never get that time back. |
| 14:00.41 | leifmadsen | so... asterisk rocks! |
| 14:00.50 | [TK]D-Fender | Katty: http://www.youtube.com/watch?v=p6Hy5HW1y6Y |
| 14:01.02 | iflux | katty: that's wonderful to say but unfortunately we live in the real world where we bought a house at the peak of the market and are now upside down by 200k |
| 14:01.13 | Katty | [TK]D-Fender: aye, i saw that several months ago :> |
| 14:01.26 | Katty | iflux: adjust. |
| 14:01.30 | iflux | hell.. it wasn't even more house than we could afford.. but it unfortunately means we have to keep working |
| 14:01.36 | Katty | iflux: it is your CHILD |
| 14:01.38 | [TK]D-Fender | Katty: I know, I sent it to you ;) |
| 14:01.42 | Katty | [TK]D-Fender: oh ;) |
| 14:02.05 | iflux | katty: true.. and she gets to see us for more time each day than she spends with her nanny.. |
| 14:02.33 | Katty | iflux: how old is your child? |
| 14:02.53 | iflux | katty: 3 |
| 14:03.03 | Katty | iflux: why don't you ask her if she's happy. |
| 14:03.15 | leifmadsen | Katty: enough please |
| 14:03.36 | iflux | thanks leif.. |
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| 14:05.28 | Katty | leifmadsen: as you wish. |
| 14:05.42 | Katty | but i am still disgruntled. |
| 14:06.07 | Katty | and i am filing an official report with star fleet. |
| 14:06.10 | leifmadsen | regardless,this is not the forum to discuss how to raise children. |
| 14:06.15 | leifmadsen | heh |
| 14:07.58 | [TK]D-Fender | "Beam me up Scotty.. there's no intelligent life down here" |
| 14:08.13 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 14:08.22 | coppice | there's none up there, either |
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| 14:11.23 | iflux | is there any way to do pcre regex's in dialplans? |
| 14:11.42 | iflux | cause that'd really be handy and spare me 4-5 lines |
| 14:11.51 | Katty | hi Deeewayne! |
| 14:12.13 | Deeewayne | Katty, good morning! |
| 14:12.28 | leifmadsen | iflux: REGEX() |
| 14:12.55 | [TK]D-Fender | iflux: "core show function REGEX" |
| 14:12.58 | iflux | leif: but regex just allows you to check if a regex matches |
| 14:13.07 | [TK]D-Fender | iflux: then no. |
| 14:13.10 | iflux | it doesn't allow you to select text out of a regex and return it |
| 14:13.19 | leifmadsen | I don't think you can do that... |
| 14:13.32 | iflux | yeah I don't think so either but it sure would be handy |
| 14:13.39 | leifmadsen | potentially :) |
| 14:13.55 | leifmadsen | you could create an AGI() to pass the text and return it in a variable |
| 14:14.11 | leifmadsen | then you could pass the values to your perl script, and return the result |
| 14:14.46 | iflux | leif: yeah.. I just didn't want to introduce an agi into this.. |
| 14:17.00 | eppigy | hello |
| 14:17.02 | eppigy | I am dave |
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| 14:23.47 | [TK]D-Fender | iflux: Feel free to write out your own app/function then :) |
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| 14:31.15 | iflux | fender: I can do it without that.. i was just wondering if it existed.. |
| 14:31.39 | iflux | maybe sometime I'll see about hooking up the pcre libs |
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| 14:49.10 | kaii | iflux, [TK]D-Fender: an interface to real pcre (not only match) would indeed be handy |
| 14:49.19 | *** part/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 14:50.00 | [TK]D-Fender | kaii: Ok, I'm sold. I support your decision to volunteer to code it ;) |
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| 14:56.48 | Kobaz | how do i tell if a call is on hold, with something like show channels... i don't see any info whatsoever in the show channels or show channel output, to indicate on-hold |
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| 14:59.42 | kaii | [TK]D-Fender: will do as soon as i updated our codebase from 1.2-patched-from-bristuff to trunk :-P |
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| 15:05.04 | maverick | getting this on log : interface.c: Junk at the beginning of frame |
| 15:05.18 | maverick | does anybody knows the meaning ? |
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| 15:12.04 | *** join/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net) |
| 15:13.58 | rudeboy_xix | hi!, how to auto answer a channel, im using manager api, and whenever i run a command 'originate', i have to answer the call first before the number is being dialed. is there a way to automatically answer the call, then dial the number? |
| 15:14.45 | [TK]D-Fender | rudeboy_xix: What if you never answer? |
| 15:15.25 | Alfio | hehehehehe good one |
| 15:15.31 | rudeboy_xix | it will not continuew |
| 15:16.02 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:16.31 | kombi | from *1.6.1 CALLERID(num) seems to drop leading zeroes. Couldn't find anything neither on voip-info nor guru. Would someone know of a fix/switch/workaround? |
| 15:16.36 | ariel_ | Hello everyone |
| 15:16.47 | [TK]D-Fender | rudeboy_xix: How do you figure.. the other side has been called. They ansewr before you do. What happens until then? |
| 15:17.10 | [TK]D-Fender | kombi: That functions does not drop digits |
| 15:17.19 | [TK]D-Fender | kombi: PAStebIN. |
| 15:17.24 | rudeboy_xix | im calling my mobile number, thats y i know |
| 15:17.26 | Kobaz | hmm |
| 15:17.36 | Kobaz | there needs to be a sip show channels consise |
| 15:17.39 | [TK]D-Fender | rudeboy_xix: Doesn't answer my second question |
| 15:17.40 | Kobaz | concise |
| 15:17.59 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 15:18.08 | rudeboy_xix | no, i have to answer first before my mobile will ring |
| 15:18.31 | *** join/#asterisk ian__ (n=ian@96.252.127.11) |
| 15:18.40 | kombi | fender: hmm, it's only one line that resolves caller numbers agains a database, worked until I put on 1.6.1 yesterday: exten => _XXXXXXXX,3,AGI(get_callername.php,${CALLERID(num)}) |
| 15:18.49 | [TK]D-Fender | rudeboy_xix: I'm not getting what it you want here... Originate calls your "Channel" first, then dumps them into the dialplan. |
| 15:18.58 | [TK]D-Fender | kombi: PASTEBIN <------- |
| 15:19.16 | kombi | ok... |
| 15:19.22 | rudeboy_xix | hmmmmm |
| 15:19.26 | rudeboy_xix | yah |
| 15:19.44 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
| 15:20.40 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:23.45 | ian__ | hi everyone, Im looking to install Asterisk in my office. It will be used on Polycom 320 sip phones and I need to have the ability to get calls from outside into my office, then route the calls to different phones and have the ability to do things such as conference calling |
| 15:24.11 | ian__ | can someone recommend a good OS to use for these purposes ? |
| 15:24.28 | [TK]D-Fender | ian__: Any you can install * and its dependencies on. |
| 15:24.33 | [TK]D-Fender | ian__: And manage of course. |
| 15:25.00 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 15:25.08 | ian__ | right, I am right now looking at a freshly installed CentOS 5.3 terminal |
| 15:25.16 | ian__ | will that be sufficient? |
| 15:25.20 | [TK]D-Fender | ian__: As far as getting support from others, CentOS is probably the best choice, with debian, Slackware, FC, etc also having a decent following. |
| 15:25.30 | [TK]D-Fender | ian__: Certainly |
| 15:25.37 | *** join/#asterisk errotan (n=errotan@62.201.122.97) |
| 15:26.05 | ian__ | ok cool, is there any place that you can direct me that would help me install */make sure I have the correct dependencies? |
| 15:26.33 | bmoraca | digium's website? asterisk the future of telephony book? |
| 15:28.17 | [TK]D-Fender | ian__: Go on the WIKI and search for CentOS and you'll see some decent guides. |
| 15:28.19 | [TK]D-Fender | ~wikis |
| 15:28.19 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:28.56 | ian__ | Very cool, thank you guys |
| 15:28.57 | *** join/#asterisk astswan (n=hanamich@wana-171-237-12-196.wanamaroc.com) |
| 15:29.17 | astswan | hi |
| 15:30.04 | astswan | <PROTECTED> |
| 15:30.39 | astswan | senddtmf action or redirect action in an extension who answer ? |
| 15:30.45 | kombi | fender: http://pastebin.se/198502 |
| 15:32.01 | [TK]D-Fender | kombi: ....... are you that slow? |
| 15:32.27 | [TK]D-Fender | kombi: what good does that PB do? |
| 15:32.38 | [TK]D-Fender | kombi: You give a story, and no backup |
| 15:33.06 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:33.13 | astswan | No one has done this trick ??? |
| 15:34.19 | [TK]D-Fender | astswan: AMI Redirect <- |
| 15:34.54 | astswan | [TK]D-Fender: yes |
| 15:35.05 | astswan | [TK]D-Fender : i try this |
| 15:35.24 | astswan | but i can t find the good channel |
| 15:35.40 | astswan | incomming calls are from a queue |
| 15:35.52 | maverick | getting this on log : interface.c: Junk at the beginning of frame |
| 15:35.54 | maverick | does anybody knows the meaning ? |
| 15:36.49 | [TK]D-Fender | maverick: Have you considered actually PASTEBINNING the complete failure from just before, through the end so we have a sense of CONTEXT? |
| 15:36.50 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 15:37.20 | [TK]D-Fender | astswan: I guess you'd better find your target before you worry about redirecting it. |
| 15:37.34 | [TK]D-Fender | astswan: Go look at the active channels. |
| 15:39.08 | kombi | fender: with all due respect I am busy here... However, if I error_logged the content of CALLERID(num) and prooved to you that it does not contain leading zeros anymore, what would you say? |
| 15:39.27 | astswan | [TK]D-Fender: SIP/666-099dc398 666@default:1 Ringing AppDial((Outgoing Line)) |
| 15:39.27 | astswan | Agent/666 112@default:1 Down AppQueue((Outgoing Line)) |
| 15:39.27 | astswan | Local/666@default-65 666@default:1 Ring Dial(SIP/666) |
| 15:39.27 | astswan | Local/666@default-65 666@default:1 Down (None) |
| 15:39.27 | astswan | SIP/200-b7a09038 112@default:1 Ring Queue(5017|tHh|||100) |
| 15:39.27 | astswan | 5 active channels |
| 15:39.29 | astswan | 2 active calls |
| 15:39.34 | bmoraca | wow |
| 15:39.37 | bmoraca | ~pb |
| 15:39.38 | infobot | [~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 , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 15:39.40 | astswan | whre is the actice channel ?? |
| 15:40.19 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:40.44 | [TK]D-Fender | astswan: PASTEBIN, do not spam in here |
| 15:40.45 | [TK]D-Fender | ~pb |
| 15:40.46 | infobot | [~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 , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 15:41.03 | [TK]D-Fender | astswan: Local/666@default-65 <- channel..... |
| 15:41.11 | astswan | sorry |
| 15:41.17 | [TK]D-Fender | astswan: and that's only PART of the channel name... its getting cut off. |
| 15:41.26 | [TK]D-Fender | astswan: "core show channels concise" |
| 15:41.40 | Katty | core show katty's lunch |
| 15:41.43 | astswan | i know this |
| 15:43.05 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 15:43.47 | ariel_ | Katty: your already thinking of lunch? Wow, |
| 15:44.23 | *** join/#asterisk flashnet (n=fdfdsf@201-213-121-79.net.prima.net.ar) |
| 15:47.46 | Katty | ariel_: it's nearly lunchtime here. |
| 15:48.21 | ariel_ | Katty: figured that, I am on the west coast today so it's not even 9 am yet |
| 15:48.29 | Katty | ah |
| 15:48.47 | eppigy | d: |
| 15:48.49 | eppigy | D: |
| 15:48.58 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:49.10 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 15:49.39 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 15:50.45 | *** join/#asterisk serph (n=serph@CPE001f5b006e5e-CM001371144daa.cpe.net.cable.rogers.com) |
| 15:51.19 | ariel_ | StarBucks in Downtown Vancouver is expensive, there small cup of coffee is over 2 dollars... |
| 15:52.28 | Alfio | D: |
| 15:52.35 | Katty | that's starbucks for you |
| 15:53.37 | leifmadsen | Asterisk-Addons 1.4.9-rc1, 1.6.0.3-rc2, 1.6.1.1-rc2, and 1.6.2.0-rc1 are now available for your testing pleasure! Please see http://www.asterisk.org/node/48607 for more information. Thank you for your continued support of Asterisk! |
| 15:54.11 | leifmadsen | Asterisk 1.4.26-rc6 is now available for your testing pleasure! Please see http://www.asterisk.org/node/48608 for more information. |
| 15:56.21 | ariel_ | argh going to see if I can vpn into my asterisk system this hotel is blocking sip channels.... |
| 15:58.07 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 16:01.28 | kombi | http://pastebin.se/198503 |
| 16:03.51 | *** join/#asterisk mphill_ (n=mphill@174.37.19.92) |
| 16:05.42 | [TK]D-Fender | kombi: Still not any brighter.... |
| 16:06.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:09.27 | maverick | getting this (http://pastebin.com/m7e16450e ) on logs |
| 16:09.29 | maverick | does anybody knows the meaning ? |
| 16:10.26 | kombi | fender: how's this http://pastebin.se/198504 |
| 16:12.45 | bmoraca | kombi: why don't you just use a PHP function to pad the input with as many zeros as to make it the length that you want? |
| 16:12.59 | bmoraca | it's like 2 lines of code |
| 16:13.31 | kombi | bmoraca: that's correct, but how do I guess the amount of zeros taken away? |
| 16:14.00 | kombi | two for international calls, one for domestic.. |
| 16:14.04 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:14.28 | bmoraca | uhm...you designed the database. aren't all of the callerid numbers the same length in it? |
| 16:15.37 | kombi | nope.. two leading zeros for international, one for domestic. As I said, one could just change every record no problem. I would just like some hint that it will stay this way from now on.. |
| 16:16.39 | maverick | getting this (http://pastebin.com/m7e16450e ) on logs |
| 16:16.40 | maverick | does anybody knows the meaning ? |
| 16:17.06 | bmoraca | kombi: i'm not aware that leading zeros were ever a "feature". sounds more like your provider changes the way they send you callerid info. |
| 16:18.29 | kombi | bmoraca: that is of course possible, but very unlikely. The change occured right after I installed * 1.6.1. How do those numbers arrive in your *? |
| 16:20.11 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:23.20 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:23.25 | kombi | I'll check with the telco anyway, but I guess changing the data in the db is the way to go here. |
| 16:26.37 | [TK]D-Fender | kombi: the function works. It does not modify anything. the CHANNEL DRIVER sends it in. |
| 16:26.54 | [TK]D-Fender | kombi: And you are not debunnging your DAHDI/CAPI call. |
| 16:26.58 | [TK]D-Fender | debugging* |
| 16:27.19 | *** join/#asterisk dovid (n=annon@tony09-121-90.inter.net.il) |
| 16:27.27 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 16:27.27 | [TK]D-Fender | maverick: that log is worthless |
| 16:27.37 | [TK]D-Fender | maverick: We don't see what caused it. |
| 16:27.50 | kombi | ahh.... that might be it! changed from misdn to dahdi. Who would have guessed that... Thanks fender! Always a pleasure |
| 16:29.02 | maverick | Fender: but are the possible causes of this ? |
| 16:29.43 | [TK]D-Fender | maverick: Go google, and start guessing. because thats all you'll be able to do. |
| 16:30.02 | [TK]D-Fender | maverick: I see several MP3 references which might mean a bad MP3 used as MoH, etc |
| 16:30.08 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:30.12 | [TK]D-Fender | maverick: But then again yuo don't ahve a live sample to show us |
| 16:31.57 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-167-45.telkomadsl.co.za) |
| 16:32.06 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 16:33.19 | jaytee | maverick? what'd your mother not like you as a child or something? :-) |
| 16:34.08 | Rob3Rt | hey dont be mean |
| 16:34.17 | Rob3Rt | its not his fault his mother didnt like him |
| 16:34.33 | Rob3Rt | oh wait |
| 16:34.39 | Rob3Rt | retracted. |
| 16:35.56 | *** join/#asterisk NERvOus (n=nervous@host75-56-dynamic.60-82-r.retail.telecomitalia.it) |
| 16:36.16 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 16:37.08 | NERvOus | hi, I recently upgraded from asterisk 1.2 (sic!) to 1.6.1.1 and now I have a problem with calls coming from my DID. I always get: [Jul 14 18:29:54] NOTICE[29639]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '7067054022' rejected because extension not found. |
| 16:37.30 | dovid | NERvOus: looks more like a configuration issue |
| 16:37.32 | NERvOus | however, the DID has the "did-private" context and the did-private context does have the 7067054022 extension |
| 16:37.40 | NERvOus | the same configuration was working on asterisk 1.2 |
| 16:37.46 | [TK]D-Fender | NERvOus: Clearly there is no exten to match that # in the context the call is landing on |
| 16:37.48 | jaytee | i was just kidding....it's a line from the movie Top Gun |
| 16:37.55 | rob0 | did you read the release notes? Syntax and behaviors have changed. |
| 16:38.01 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 16:38.17 | dovid | rob0: If it was syntax he wouldnt get a 404 error |
| 16:38.27 | dovid | most likely some kind of include |
| 16:38.28 | [TK]D-Fender | yeah, because there were only 2 major versions BETWEEn them... |
| 16:38.44 | [TK]D-Fender | dovid: Syntax can cause a 404. |
| 16:38.45 | NERvOus | this is an excerpt from my config: |
| 16:38.50 | [TK]D-Fender | NERvOus: PASTEBIN~! |
| 16:38.55 | [TK]D-Fender | NERvOus: Do not spma. |
| 16:38.58 | [TK]D-Fender | SPAM* |
| 16:39.02 | [TK]D-Fender | ~pb |
| 16:39.03 | infobot | [~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 , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 16:40.06 | dovid | [TK]D-Fender: Like what ? Exten => 7067054022..... has not |
| 16:40.20 | [TK]D-Fender | dovid: "Has not" what? |
| 16:40.35 | dovid | changed. |
| 16:40.42 | dovid | unless he has more like includes |
| 16:40.52 | [TK]D-Fender | dovid: I don't SEE anything. Guess how much I trust. |
| 16:40.58 | dovid | lol |
| 16:41.08 | [TK]D-Fender | dovid: Give you a hint, sounds like "zero" |
| 16:41.30 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 16:41.32 | dovid | [TK]D-Fender: I am bad at hints. u should know that by now |
| 16:42.45 | NERvOus | http://pastebin.com/m10b5f86f |
| 16:43.15 | *** join/#asterisk Ex_peter2 (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 16:43.24 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 16:44.18 | [TK]D-Fender | NERvOus: ever considered looking at the SIP debug for your failed attempts? |
| 16:44.18 | dovid | broadvice is sending the call to 7067054022@your_IP |
| 16:44.39 | dovid | all you have in did-private is the s extension |
| 16:44.45 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.207.68) |
| 16:44.54 | [TK]D-Fender | dovid: FAIL |
| 16:45.04 | [TK]D-Fender | dovid: Look aagin |
| 16:45.04 | NERvOus | dovid: no, I have 7067054022 too |
| 16:45.06 | [TK]D-Fender | Aagain* |
| 16:45.13 | [TK]D-Fender | NERvOus: ever considered looking at the SIP debug for your failed attempts? <---------- |
| 16:45.18 | NERvOus | [TK]D-Fender: checking |
| 16:45.21 | dovid | ops. i missed that |
| 16:46.59 | dovid | can you post a SIP debug ? |
| 16:47.14 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 16:47.25 | [TK]D-Fender | dovid: Mised my asking him TWicE for it, and then his :checking" too? |
| 16:47.45 | dovid | ok. i quit |
| 16:48.09 | [TK]D-Fender | dovid: Go caffeinate! |
| 16:48.17 | dovid | im outa red bull |
| 16:48.43 | NERvOus | andrew*CLI> sip show registry |
| 16:48.43 | NERvOus | Host dnsmgr Username Refresh State Reg.Time |
| 16:48.43 | NERvOus | sip.broadvoice.com:5060 N 7067054022@s 23 Request Sent Tue, 14 Jul 2009 18:48:03 |
| 16:48.43 | NERvOus | p |
| 16:48.46 | NERvOus | I just noticed this |
| 16:48.47 | *** join/#asterisk jkroon (n=jkroon@dsl-240-167-45.telkomadsl.co.za) |
| 16:48.53 | NERvOus | it looks like it's not registered yet |
| 16:49.08 | NERvOus | in fact, I see a lot of SIP REGISTER packets, in sip debug |
| 16:49.09 | dovid | well the call is coming to you |
| 16:49.11 | Qwell | dovid: redbull... good idea |
| 16:49.25 | dovid | Qwell: I hear the digium staff lives off of it |
| 16:49.56 | [TK]D-Fender | NERvOus: Know what I don't see? SIP debug... |
| 16:49.58 | seanbright | does the attended transfer feature built in to asterisk allow for a 'conference' before the transferer hangs up? |
| 16:50.14 | seanbright | i guess you would call it a 'warm transfer' |
| 16:50.41 | dovid | seanbright: wouldn't that be called an attended transfer ? |
| 16:51.14 | seanbright | customer (A) calls company (B) |
| 16:51.17 | [TK]D-Fender | Attended transfers don't confrernce. |
| 16:51.22 | seanbright | ... |
| 16:51.23 | seanbright | perfect |
| 16:51.24 | seanbright | thanks |
| 16:51.45 | [TK]D-Fender | seanbright: And no, I'm prety sure *'s handling does not allow this. Polycom's handling can |
| 16:52.06 | seanbright | i can hack something together with chanspy/meetme/whatever |
| 16:52.20 | seanbright | just wanted to know if i was going to need to do something that wasn't ootb |
| 16:52.58 | [TK]D-Fender | seanbright: Just depends whose box ;) |
| 16:53.14 | seanbright | that almost sounds dirty |
| 16:53.52 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:20be:156:722f:c6db) |
| 16:54.12 | NERvOus | http://pastebin.com/d573d8ae2 |
| 16:54.21 | [TK]D-Fender | spins up some Corey Hart |
| 16:54.29 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.207.68) |
| 16:55.00 | NERvOus | btw sip show registry now show "Registered" in the status column |
| 16:55.08 | [TK]D-Fender | NERvOus: No matching peer for '+39333749177' from '147.135.0.128:5060' <------- |
| 16:55.26 | NERvOus | that's the caller's id |
| 16:55.34 | [TK]D-Fender | NERvOus: Looking for 7067054022 in default (domain 94.228.131.69) SIP/2.0 404 Not Found |
| 16:55.43 | NERvOus | ah ok |
| 16:55.45 | [TK]D-Fender | NERvOus: * can't ID the systems ending the call |
| 16:55.57 | [TK]D-Fender | NERvOus: So its landing in the context under [general] |
| 16:55.58 | NERvOus | so it's looking in the 'default' context? |
| 16:56.12 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:56.13 | NERvOus | understood |
| 16:56.54 | [TK]D-Fender | NERvOus: nat=yes <- Should be "no" for your peer |
| 16:57.11 | [TK]D-Fender | NERvOus: And instead of "friend" should be "peer" as well |
| 16:57.14 | NERvOus | trying |
| 16:57.44 | [TK]D-Fender | NERvOus: ALSO... insecure=very < - s/b port,invite |
| 16:59.11 | dovid | the issue is that sip.broadvoice.com goes to 147.135.32.221 and as TK pointed out the call is coming from 94.228.131.69 |
| 16:59.15 | dovid | oops |
| 16:59.19 | dovid | 147.135.0.128 |
| 16:59.59 | NERvOus | http://pastebin.com/m2f682dd2 <-- this my current config |
| 17:00.04 | NERvOus | reloading, and retrying |
| 17:00.49 | NERvOus | same error, re-enabling sip debug |
| 17:01.14 | dovid | one sec. i am writing something for u |
| 17:02.05 | dovid | try this: http://pastebin.com/mccac418 |
| 17:03.17 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 17:03.20 | dovid | you can take out user=phone from sip.broadvoice.com2 |
| 17:04.23 | NERvOus | http://pastebin.com/d677a65c5 (updated sip debug, similar to previous one) |
| 17:04.27 | NERvOus | dovid: ok |
| 17:05.30 | *** join/#asterisk zarloc (n=zarloc@host87-188-dynamic.211-62-r.retail.telecomitalia.it) |
| 17:05.51 | *** part/#asterisk zarloc (n=zarloc@host87-188-dynamic.211-62-r.retail.telecomitalia.it) |
| 17:07.08 | NERvOus | works! |
| 17:07.17 | NERvOus | dovid: thank you so much :) |
| 17:07.23 | NERvOus | [TK]D-Fender: thank you for your help too |
| 17:07.32 | dovid | NERvOus: do you understand what the issue was ? |
| 17:07.40 | dovid | TK: Seee. I can think some times ;) |
| 17:07.51 | NERvOus | asterisk is expecting broadvoice to send packets from a certain ip |
| 17:07.57 | NERvOus | but broadvoice is sending packets from another ip |
| 17:08.02 | dovid | ok |
| 17:08.03 | NERvOus | and * doesn't know how to treat them? |
| 17:08.18 | dovid | well no because all you have for broadvice is the one domain |
| 17:08.21 | NERvOus | so we tell asterisk that that certain ip belongs to bv |
| 17:08.37 | dovid | if any other IP comes to your server it goes to the default context in sip.conf |
| 17:08.41 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 17:09.01 | dovid | when you added the second IP it knew to go to the did-private context |
| 17:09.08 | NERvOus | gotcha |
| 17:09.11 | NERvOus | makes sense |
| 17:09.52 | NERvOus | I got the config entry directly from broadvoice, guess they should update their website ;) |
| 17:11.08 | dovid | NERvOus: The issue is more by them. why arent they sending you the call from the same IP that you registerd to. that is "standard" with most carriers (atleast from what I have seen) unless its complex and they are not dialing via the proxy |
| 17:13.23 | beek | What should a carrier expect on an NI2 PRI for 911? Just the digits, 911 or 0000000911? |
| 17:13.42 | [TK]D-Fender | beek: 911 |
| 17:13.51 | beek | [TK]D-Fender: That's too easy! Thanks |
| 17:14.16 | dovid | i would just test it. you want to make sure either way that they have the right address |
| 17:14.36 | beek | dovid: Absolutely will. |
| 17:14.37 | beek | Thanks |
| 17:14.52 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:15.00 | dovid | [TK]D-Fender: what is the sagnifigance of NI2 ? |
| 17:15.38 | [TK]D-Fender | its a signaling standard. |
| 17:15.52 | dovid | ok. |
| 17:16.00 | dovid | is that the US standard ? |
| 17:16.06 | dovid | i know. google |
| 17:23.51 | leifmadsen | dovid: there are several "standards" depending on who you connect to |
| 17:24.08 | dovid | oh ok. like euroisdn etc. ? |
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| 17:30.52 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:31.56 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 17:33.20 | buttons840 | Corydon76-dig, (or anyone) you helped me much with the call spool some time ago, i was wondering if you knew much about the AMI Originate command? when used with async: yes it creates a originateresponse, but i cannot find documentation on what the response numbers mean, i've keep receiving response 3 and 5 and don't know their meanings. do you know of any documentation on this? |
| 17:34.17 | [TK]D-Fender | buttons840: I strongly suspect these are the status codes as defiened for channels. IE 3 = rining IIRC |
| 17:34.32 | *** join/#asterisk |Rain| (i=rain@ev.il.net) |
| 17:34.43 | oglynn | trying to use the record_out parameter on an extension in sip.conf do i need to have some particular flags on the Dial in the extensions.conf? |
| 17:34.59 | |Rain| | is GOSUB_RETVAL a stack? |
| 17:36.11 | [TK]D-Fender | oglynn: Dial option required for on-demand, or Monitor call before |
| 17:36.18 | buttons840 | where can i learn more about these channel codes [TK]D-Fender ? |
| 17:36.37 | buttons840 | i've read the book, but don't remember seeing anything about this |
| 17:36.49 | [TK]D-Fender | oglynn: That value you mentioned does not exist |
| 17:37.09 | |Rain| | oh, no it's not, I'm just a dumbass. \o/ if you're using a macro for logging.... |
| 17:37.16 | [TK]D-Fender | buttons840: Its either in the docs int he tarball, or decodable in source. I know I passed by it once before, but couldn't say where |
| 17:37.18 | dwery | If someone wants to buy the C470 please note that doing call transfers is absolutely ugly with it! |
| 17:37.28 | *** part/#asterisk |Rain| (i=rain@ev.il.net) |
| 17:38.09 | oglynn | [TK]D-Fender If I want to record the outbound calls of only a couple of extensions (always) how is that best acheived? |
| 17:38.32 | Katty | hi |
| 17:40.08 | [TK]D-Fender | oglynn: this is all dialplan.... |
| 17:41.38 | Katty | some gotoifs based on callerid information would probably work. |
| 17:42.08 | [TK]D-Fender | Katty: I'd SetVar them.... |
| 17:42.14 | Katty | that'd also work. |
| 17:42.21 | dovid | i would just break up the contexts so you dont have a string of them |
| 17:42.24 | [TK]D-Fender | Katty: Makes it more trackable to the peer |
| 17:42.40 | [TK]D-Fender | dovid: Code duplication FTL! |
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| 17:44.30 | oglynn | Katty/[TK]D-Fender. I understand the GotoIf option how would i use Set/SetVar for just a couple of extensions outbound. I am guessing I would Set(recordthiscall=1) or some such but how to only do this for the 2/3 exts outbound is not clear to me |
| 17:44.57 | [TK]D-Fender | oglynn: Setvar <- sip.conf peer parameter |
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| 17:47.28 | oglynn | <PROTECTED> |
| 17:48.07 | chrisb | anyone from xmission here? |
| 17:48.19 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:48.57 | Katty | never heard of it. |
| 17:50.05 | *** join/#asterisk ThatKidKel (n=Kelvin@c-24-98-147-214.hsd1.ga.comcast.net) |
| 17:50.18 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 17:50.25 | Corydon76-dig | chrisb: try thehar |
| 17:50.31 | ThatKidKel | what's the current status of Faxing with Asterisk? |
| 17:50.42 | KavanS | ThatKidKel, use hylafax |
| 17:50.50 | KavanS | rxfax and txfax caused me headaches... |
| 17:50.57 | KavanS | hylafax is quite reliable for production in my use |
| 17:51.07 | Corydon76-dig | ThatKidKel: or use the commercial solution |
| 17:51.19 | Corydon76-dig | FaxForAsterisk |
| 17:51.26 | KavanS | that too :) support digium! :) |
| 17:51.27 | dovid | what is better about the commercial solution ? |
| 17:51.38 | KavanS | dovid, it's more reliable |
| 17:51.56 | dovid | in what sense ? |
| 17:52.06 | Corydon76-dig | mega mega tested |
| 17:52.12 | dovid | ok |
| 17:52.18 | Alfio | dovid and if you have problems you only need a call |
| 17:52.19 | dovid | is there support for 1.4.X ? |
| 17:52.39 | dovid | i use straight voip/ulaw which works for me but i wouldnt mind paying so i can complain ;) |
| 17:52.44 | Alfio | * 1.4 just passthrought of t38 |
| 17:52.46 | Qwell | dovid: yes, but no T.38 |
| 17:53.04 | Alfio | 1.6 fully supported |
| 17:53.19 | ThatKidKel | ok |
| 17:53.28 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 17:53.47 | Alfio | if you want to test you can get one free license from digium |
| 17:53.51 | Alfio | for one channel |
| 17:53.54 | dovid | Qell: would help if I had jitter ? |
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| 18:02.38 | *** join/#asterisk empiric (n=empiric@116.71.52.243) |
| 18:13.36 | empiric | guys i want to set a conference room |
| 18:13.40 | empiric | how i do it |
| 18:14.11 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 18:14.44 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 18:14.51 | teknoprep | hi all.. i have a polycom ip 650 |
| 18:15.02 | [TK]D-Fender | empiric: "core show application meetme" |
| 18:15.09 | chrisb | Corydon76-dig: thanks |
| 18:15.33 | empiric | what i have to do in meetme.conf |
| 18:15.48 | [TK]D-Fender | empiric: Go read the instructions. |
| 18:16.02 | [TK]D-Fender | empiric: This should also be int he boko and on the WIKI |
| 18:16.07 | [TK]D-Fender | book* |
| 18:16.11 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:16.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:19.20 | teknoprep | nvm fixed it |
| 18:19.40 | [TK]D-Fender | teknoprep: Funny, we never knew you had a problem with it... |
| 18:19.56 | teknoprep | yeah i know but i joined the channel saying... hi all.. i have a polycom ip 650 |
| 18:20.01 | teknoprep | sounds stupid unless i finish it |
| 18:20.07 | teknoprep | or boastfull |
| 18:20.24 | elguero | Can anyone tell me if Asterisk 1.4 supports B-channel service messaging? I have been troubleshooting a problem with our PRI lines and the switch technician is trying to tell me that this needs to be turned on |
| 18:21.06 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 18:21.16 | Deeewayne | elguero, no it doesn't |
| 18:21.30 | elguero | doh |
| 18:21.54 | elguero | he mentioned that it is supposed to be the standard for NI2 |
| 18:22.16 | elguero | Deeewayne: any idea if that is being worked on at present? |
| 18:22.30 | Deeewayne | its in trunk and possibly in the latest 1.6.x branch |
| 18:23.09 | bmoraca | elguero: what's the issue you're having? |
| 18:23.15 | elguero | Deeewayne: oh really!... I have no problem running trunk... we are having a major issue with the increase in call volume and this is what the switch technician is trying to point to |
| 18:23.36 | elguero | bmoraca: I am having the D-channel restart, resulting in dropped calls |
| 18:23.46 | Deeewayne | elguero, don't forget to upgrade libpri as well |
| 18:24.00 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 18:24.10 | elguero | Deeewayne: right, I have the latest release but I will probably need the trunk version of libpri, right? |
| 18:24.36 | Deeewayne | I don't think there is a trunk version of libpri. I think all libpri development is in the 1.4 branch |
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| 18:25.01 | elguero | bmoraca: I am seeing in the intense debugging that they send a SABME, we respond with UA, and then it does this for a little bit until the t200 timer expires resulting in a PRI restart |
| 18:25.06 | empiric | Fender |
| 18:25.18 | elguero | Deeewayne: I think you are right... thanks for the help |
| 18:26.15 | empiric | i Fender in meetme.conf i add conf => 1000 |
| 18:26.20 | bmoraca | elguero: i can safely say that i've never had that issue, and I can't count the number of Asterisk installs I've done on NI2 PRIs...don't you just love telco fingerpointing? |
| 18:26.26 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 18:26.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:26.33 | empiric | when i dial 1000 it give me hold music |
| 18:26.43 | empiric | is this ok? |
| 18:27.24 | elguero | bmoraca: right, we have had this setup running for a good while now.... in fact I hadn't upgraded in about a year since we have had other projects going.... so when this all started I upgraded one of the machines... but this was running perfect for a good while... I don't know why all of a sudden it would cause problems |
| 18:28.25 | [TK]D-Fender | empiric: No idea... how should I know what you configured "1000" to do in your dialplan? |
| 18:28.36 | elguero | bmoraca: the tech is trying to tell me that it is due to the increase in volume... we have 4 pri lines and they hunt to each other... he is trying to say that we need this b channel service messaging turned on... he says that it just probably built up to this and started to have problems now with the increase in call volume |
| 18:28.40 | empiric | no no |
| 18:28.48 | empiric | i want to setup conferenceing |
| 18:28.51 | [TK]D-Fender | empiric: though it is common for MeetMe to provide MoH while there is only 1 caller in the room |
| 18:29.09 | empiric | in meetme i add 1000 so that when user dials it connects to conference |
| 18:29.29 | empiric | when another user comes in then |
| 18:29.32 | empiric | same MOH |
| 18:29.55 | elguero | bmoraca: I am going to check into 1.6 and trunk and see when it was added and give this a try... we have been dealing with this for over 3 weeks and it is starting to impact our service a great deal... thanks for the help |
| 18:29.59 | bmoraca | elguero: that doesn't make a whole lot of sense...course, i once had to replace an entire PBX in order to satisfy a telco that the issue was not on our end (ended up being an upstream OC3 that was having issues) |
| 18:30.43 | [TK]D-Fender | empiric: I don't see you showing me anything. |
| 18:31.05 | elguero | bmoraca: I have two pbx on site, and tried switching the lines around to try and prove to them that it wasn't the equipment... that is why he is narrowing it down to this b-channel servicing |
| 18:31.05 | empiric | should i show u logs |
| 18:31.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:31.47 | [TK]D-Fender | empiric: CLI output and configs |
| 18:32.03 | empiric | ok |
| 18:32.54 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:33.26 | empiric | here |
| 18:33.27 | empiric | http://pastebin.com/m794105c2 |
| 18:36.56 | [TK]D-Fender | empiric: that is a QUEUE, not a CONFERENCE (MeetMe) |
| 18:38.13 | empiric | so what should i do |
| 18:38.24 | empiric | (Meetme) |
| 18:38.55 | [TK]D-Fender | empiric: call MEETME, not QUEUE. |
| 18:39.39 | empiric | i add this line |
| 18:39.40 | empiric | exten => 1000,1,Meetme(1000) |
| 18:39.50 | empiric | should wor |
| 18:39.51 | empiric | work |
| 18:40.39 | Alfio | empiric use a "d" to used dynamic |
| 18:41.18 | empiric | Alfio where i use d |
| 18:41.32 | Alfio | exten => 1000,1,Meetme(1000,d) |
| 18:44.25 | empiric | what abt my dial plan |
| 18:44.28 | empiric | is that ok |
| 18:44.38 | empiric | should i disable testq |
| 18:45.09 | empiric | Executing [1000@default:1] Queue("SIP/205-0825d028", "testq") in new stac |
| 18:45.18 | empiric | its going in queue |
| 18:45.25 | empiric | not in meetme |
| 18:45.49 | [TK]D-Fender | empiric: Why do you ahve that exten in there with that number still? |
| 18:46.07 | empiric | where |
| 18:46.14 | jkroon | dialplan reload perhaps? |
| 18:46.18 | [TK]D-Fender | empiric: IN YOUR DIALPLAN! |
| 18:46.34 | empiric | i did reload |
| 18:46.41 | empiric | sorry i wont understand |
| 18:47.34 | [TK]D-Fender | empiric: Its calling that because it is in your dialplan. |
| 18:48.09 | bmoraca | empiric: line 21 of your pastebin...you already have an extension 1000, and it's not going to a meetme |
| 18:48.39 | empiric | [Jul 12 17:37:30] WARNING[6278]: pbx.c:3082 pbx_extension_helper: No application 'Meetme' for extension (default, 1000, 1) |
| 18:48.39 | empiric | <PROTECTED> |
| 18:48.46 | empiric | yes i remove that |
| 18:48.50 | empiric | still same issue |
| 18:49.15 | *** part/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 18:49.49 | [TK]D-Fender | empiric: Because meetme never got compiled because it was missing Zaptel/DAHDI support <- |
| 18:50.00 | empiric | oh |
| 18:50.02 | empiric | then |
| 18:50.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:50.08 | empiric | how i compile |
| 18:50.42 | [TK]D-Fender | empiric: Go install Zaptel/DAHDI, then recompile * from scratch |
| 18:51.48 | Alfio | :) |
| 18:52.44 | empiric | how i install Zaptel.DAHDI |
| 18:53.07 | KavanS | empiric, not sure your platform....but this covers this for centos: http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation |
| 18:53.15 | empiric | its debain |
| 18:53.16 | [TK]D-Fender | empiric: Go download it and follow the instructions. |
| 18:53.30 | empiric | my astersk in 1.6.0.5 |
| 18:53.34 | KavanS | empiric, there's a good howto for debian on howtoforge....walks you through compilation/dependency install |
| 18:53.53 | KavanS | empiric, listen to [TK]D-Fender he has good advice... |
| 18:55.26 | tzafrir_laptop | there are also debs for Debian :-) |
| 18:55.44 | empiric | hey should i downlaod DHADI linux or DHADI tools |
| 18:55.58 | Alfio | empiric both |
| 18:56.31 | empiric | does it works for asterik 1.6.0.5 |
| 18:56.32 | Corydon76-dig | or just dahdi-linux-complete |
| 18:56.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:56.53 | Corydon76-dig | since -complete contains both |
| 18:56.59 | Alfio | http://www.debian-resources.org/node/129 |
| 18:57.15 | Alfio | look an install guide for debian |
| 19:01.14 | bmoraca | off to go find an aram sandwich...woo! |
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| 19:26.36 | Kobaz | hmm |
| 19:26.42 | Kobaz | the polycom 321 and 331 are out |
| 19:27.12 | Qwell | Kobaz: what changed? |
| 19:27.45 | [TK]D-Fender | Qwell: Only more memory to support custom apps, etc |
| 19:27.59 | Qwell | makes sense |
| 19:28.00 | [TK]D-Fender | Qwell: much like the 300/500/600 _. +1 |
| 19:28.05 | Qwell | same price? |
| 19:28.14 | Qwell | (I would sure hope so..) |
| 19:28.15 | [TK]D-Fender | Qwell: Logically, currently a few $ here or there |
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| 19:32.47 | iflux | re fender |
| 19:36.39 | Nugget | A SQL query walks up to two tables in a restaurant and says: "Mind if I join you?" |
| 19:38.52 | iflux | plays the trombone for Nugget |
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| 19:41.12 | cribob | I am trying to use a call file to place an automated outbound call via a SIP trunk to a PBX which in turn is connected to the PSTN. The problem I have is that I need to outpulse an auth code once the PBX dials the call but before the connect is signaled on the PRI and as a result befor the call is completed via SIP and before the call is delivered to the context and extension specified in the call file. Can anyone help with this? |
| 19:45.19 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 19:45.35 | [TK]D-Fender | cribob: dial a local channel to do your dialout via SIP and use D() <- |
| 19:46.01 | sah-work | hello. |
| 19:46.10 | sah-work | i am on my new system. yah |
| 19:46.41 | sah-work | is there an easyway to set the outbound callerid to be the sip phone number vs having to cfg it on every user |
| 19:47.02 | [TK]D-Fender | sah-work: Set it in the dialplan. |
| 19:47.47 | sah-work | k |
| 19:48.24 | leifmadsen | cribob: use 'w' (or multiples) to add a pause before dialing |
| 19:48.30 | leifmadsen | D(www1234) for example |
| 19:48.58 | leifmadsen | sah-work: core show function CALLERID |
| 19:49.24 | [TK]D-Fender | cribob: for your Channel: use a LOCAL CHANNEL, not SIP and do the dial in your dialplan using the D() option to pass DTMF for the auth code |
| 19:52.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:54.36 | cribob | D-Fender: how do I avoid blocking the futher execution of my dialplan (delivery of a notification message and some other processing) when usign the dial commend in my dialplan? |
| 19:56.02 | cribob | D-Fender: disregard that last question. I think I've got it. Thanks. |
| 19:56.24 | [TK]D-Fender | cribob: Good... i was about to reach for my ClueBat (tm) :d |
| 19:57.25 | cribob | D-Fender: Sorry. Had a bad case of tunnel vision there for a few min. The ClueBat (tm) might have actually helped. |
| 19:59.12 | [TK]D-Fender | Spare the clueBat (tm), spoil the newb... |
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| 20:00.45 | Katty | i think my vendor is drunk. |
| 20:00.46 | Katty | or twinked |
| 20:00.47 | Katty | or something |
| 20:00.55 | Katty | he doesn't seem right :< |
| 20:01.42 | [TK]D-Fender | Katty: Could be a Mormon ;) |
| 20:03.34 | Katty | in north carolina?! |
| 20:03.38 | [TK]D-Fender | fetches an asbestos suit |
| 20:04.00 | [TK]D-Fender | Katty: Just like Great Whites, they show up where you leaast expect them! |
| 20:05.19 | Kobaz | ohnoses |
| 20:05.26 | Kobaz | my patch has been rejected :( |
| 20:05.40 | [TK]D-Fender | Kobaz: http://tinyurl.com/bh66yf |
| 20:05.44 | [TK]D-Fender | Katty: http://tinyurl.com/bh66yf |
| 20:05.46 | [TK]D-Fender | :D |
| 20:06.05 | Kobaz | https://issues.asterisk.org/view.php?id=15503 |
| 20:06.28 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ea:62:a931:c61) |
| 20:07.58 | elguero | Deeewayne: On the b channel service messaging, the CHANGES file says: "Added service message support for 4ESS/5ESS switches", the telco says they don't support 5ess; I was trying to look at the code, if I turn service messaging on even though switchtype is ni2, do you think asterisk will try try to use service messages? If you don't know off the top of your head, that is fine... I don't mind experimenting this evening... I just don't want to upgrad |
| 20:08.04 | elguero | e to only find out that it isn't going to work with NI2... thanks |
| 20:09.14 | Katty | [TK]D-Fender: good use of the photoshop extract filter |
| 20:10.02 | Deeewayne | elguero, you would be the first to my knowledge to try service messages w/ NI2, although that doesn't mean someone hasn't already tried it. |
| 20:11.32 | elguero | Deeewayne: I was looking at the bug tracker and it looked like it was only tested on those other switches... I figured that is why it said it only supported those switches... I guess I can give it a try and see what happens... the telco switch tech was trying to tell me that it was part of ni2 standard and that I have to turn that on... so I guess I will try it and maybe report back |
| 20:12.32 | Deeewayne | elguero, I'd be interested in hearing about your results |
| 20:12.45 | elguero | Deeewayne: it looks like there is nothing limiting service messages to just 4ess/5ess, so I will give it a whirl |
| 20:13.05 | elguero | Deeewayne: Sure, I will be glad to share the results |
| 20:13.10 | elguero | Deeewayne: thanks for the help |
| 20:13.17 | Deeewayne | elguero, if you have a problem, show me a pri trace |
| 20:13.26 | elguero | Deeewayne: okay... will do |
| 20:13.35 | elguero | Deeewayne: intense debug? |
| 20:13.38 | Deeewayne | yeah |
| 20:13.46 | elguero | Deeewayne: okay, great |
| 20:14.18 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:17.30 | *** join/#asterisk ingenius (n=alektro@host38.190-138-197.telecom.net.ar) |
| 20:20.16 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 20:20.28 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:21.40 | Maxxed | hey'a fellas, im trying to get the Message Center on a polycom IP330 to dial my voicemail context. Right now, it just dials the extension that the phone is. Is there a way to change what the phone dials when the Message Center is selected? |
| 20:22.33 | Maxxed | phone is registerd as 2600, when a voice mail is left, i can navigate the phone menue to the Message Center, when I select it, it just dials 2600 |
| 20:22.49 | Maxxed | i guess i could do some kinda stuff with the dialplan, but i much rather do it on the phones |
| 20:23.08 | Maxxed | anyone know off hand? i havent had much luck googling.. i may be asking the wrong question |
| 20:23.31 | eppigy | hello |
| 20:23.33 | eppigy | I am dave |
| 20:23.47 | Maxxed | guess i can do a cmd_gotoif :/ |
| 20:25.28 | [TK]D-Fender | Maxxed: Look at the "contact" tag under mwi in your provisioning |
| 20:26.06 | [TK]D-Fender | Maxxed: It should be "contact" instead of "registration" and the contact updated... |
| 20:26.09 | [TK]D-Fender | Maxxed: and go read the Admin guide. |
| 20:26.12 | Maxxed | [TK]D-Fender: ah hah! |
| 20:26.33 | Maxxed | [TK]D-Fender: thanks! yeah i was looking thru the guide but i guess i wasnt reading it :p |
| 20:27.11 | kn0x | anyone familair with SIPP |
| 20:27.18 | kn0x | i can't get it to send RTP |
| 20:27.28 | kn0x | so asterisk won't actually send any media |
| 20:28.39 | eppigy | kn0x: are you testing it in a local LAN envoriment with no blocked port? |
| 20:28.44 | eppigy | *ports |
| 20:29.21 | Maxxed | [TK]D-Fender: where does one update the contact info? |
| 20:29.32 | kn0x | yes, but I'm not sure if im using the right syntx |
| 20:29.45 | Maxxed | il hit the manul if you dont know off hand |
| 20:29.58 | [TK]D-Fender | Maxxed: I gave you enough keywords for a FIND. |
| 20:30.08 | Maxxed | [TK]D-Fender: haha, right on buddy ;) |
| 20:30.15 | [TK]D-Fender | Checkout time, BBIAB |
| 20:30.44 | eppigy | there he goes |
| 20:30.50 | eppigy | one of gods own creatures |
| 20:30.57 | kn0x | <PROTECTED> |
| 20:31.12 | Kobaz | anyone know of a digit map tester script |
| 20:31.31 | Kobaz | i hate having to reboot the polycoms i have to try out a digit map, it takes like 2 minutes to boot each time |
| 20:32.06 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 20:32.30 | timeshell | Happy Twosday! |
| 20:34.06 | *** join/#asterisk dni (n=dniz0r@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 20:34.49 | cribob | [TK]D-Fender: I tried having my call file go to a local channel in which I'm usign the dial command with the D option with my auth code in (), but the auth code is not outpulsed because the servicing telco is trying to collect the digits before they signal the call as connected and as such the call SIP leg also occurs pre-connect and the D() option does not seem to process untill connect. Is there a way to modify this behavior? |
| 20:35.37 | kn0x | eppigy: what is the correct sipp syntax? |
| 20:36.23 | cribob | All: I trying to place an outbound call via a call file and need to dial a DTMF auth code during a preconnect state. Any suggestions? |
| 20:39.54 | dovid | have it go to a context where you use senddtmf |
| 20:40.15 | dni | hell all,.. I;m getting some odd error,. from two particular PHONES i get one way audio dialing another number thats behind 3com V3000 switch,.. these two particular phones are th eonly ones having the issue,. can someone provide some insight as to waht it may be,. here is a global debug |
| 20:40.16 | dni | http://pastebin.com/m1d37b871 |
| 20:41.05 | Katty | anyone know what the -av switches are for rsync off the top of their head? |
| 20:41.38 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 20:43.04 | cribob | david: The problem is that the auth code collection is happening in a preconenct state. As such I can figure a way to get call control into the dialplan in such a way to use SendDTMF. Am I misising somehting? |
| 20:43.32 | kn0x | is SIPP not supposed to receive any RTP unless it sebnds? |
| 20:45.11 | leifmadsen | asterisk needs to receive RTP in order to know where to send it back |
| 20:45.46 | leifmadsen | at least that's how I understand it, and is the reason for no-way audio in some NAT cases |
| 20:50.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:50.35 | jaytee | Katty, a is archive and v is verbose |
| 20:51.31 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 20:51.35 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:52.24 | rob0 | Both are like cp's a and v; rsync tries to stay close to cp(1) and ssh(1) options. |
| 20:54.16 | [8none1] | Is there a reason to use 1.6.0 over 1.6.1? Is 1.6.1 the latest stable 1.6 release? |
| 20:55.35 | [TK]D-Fender | [8none1]: No. 1.6.0 and 1.6.1 are different branches like 1.4 to 1.6.0 |
| 20:56.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:56.08 | [8none1] | Where can I find what the differences are? |
| 20:56.18 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 20:56.55 | cribob | [TK]D-Fender: I tried having my call file go to a local channel in which I'm usign the dial command with the D option with my auth code in (), but the auth code is not outpulsed because the servicing telco is trying to collect the digits before they signal the call as connected and as such the call SIP leg also occurs pre-connect and the D() option does not seem to process untill connect. Is there a way to modify this behavior? |
| 20:56.59 | [TK]D-Fender | [8none1]: upgrade.txt <- |
| 20:57.10 | [8none1] | thx, I'll check it out |
| 20:57.30 | [TK]D-Fender | cribob: If its not actually answered I don't see how to do it... |
| 20:59.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:02.52 | jaytee | quittin time, bbl |
| 21:10.04 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 21:18.11 | leifmadsen | ~asteriskversioning |
| 21:18.12 | infobot | asteriskversioning is, like, Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 21:18.22 | leifmadsen | [8none1]: see above |
| 21:20.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:21.32 | dwery | Hi. I would like to limit the calls to a sip device using a variable configured in its entry in sip.conf as the maximum number of accepted calls . however in the dialplan I get, obviously, the variable of the calling device . Is there a way to handle this situaztion? |
| 21:22.23 | [TK]D-Fender | dwery: What "situation"? You don't seem to be DOING anything with your variable so far... |
| 21:23.49 | dwery | [TK]D-Fender: I'm trying this: exten => _2XX,n,GotoIf($[${GROUP_COUNT(OUTBOUND_GROUP@${EXTEN})}>${MAXCALLS}]?busy) |
| 21:23.49 | dwery | that MAXCALLS should be related to the called device |
| 21:23.49 | [TK]D-Fender | dwery: Ok.... and? |
| 21:24.01 | dwery | [TK]D-Fender: I'd like to configure MAXCALL in sip.conf |
| 21:24.15 | [TK]D-Fender | dwery: You have our permission. |
| 21:24.19 | dwery | :D |
| 21:24.21 | [TK]D-Fender | dwery: Go for it |
| 21:24.38 | dwery | howeverm I get the MAXCALL of the alling device instead of the one of the called device |
| 21:26.20 | [TK]D-Fender | dwery: Well I guess you should reconsider where your count information is stored & retreived from |
| 21:27.01 | dwery | [TK]D-Fender: well, it was handy to have it in sip.conf, just like the deprecated call-limit |
| 21:27.21 | [TK]D-Fender | dwery: Well that clearly won't do, so time for "plan B" |
| 21:27.33 | dwery | [TK]D-Fender: what could be a suitable place? |
| 21:28.01 | [TK]D-Fender | dwery: AstDB. Dialplan Global, etc |
| 21:28.30 | [TK]D-Fender | s/Global/constant/ |
| 21:29.15 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 21:29.20 | dwery | I'll have to check astdb. the variable is different based on the type of the called phones. On more simper phones I want to limit to one call |
| 21:30.24 | WindowsUser | dont want them to refuse thier own calls? |
| 21:32.02 | dwery | WindowsUser: I have two handsets connected on the C470IP dect base, I one handset receives two calls the second handset will get none. So I have to limit to on call per device |
| 21:35.46 | *** part/#asterisk mphill_ (n=mphill@174.37.19.92) |
| 21:36.55 | *** join/#asterisk bwat (n=IceChat7@atlantis.kanobe.com) |
| 21:38.45 | [TK]D-Fender | dwery: Glorious piece of crap :) |
| 21:38.55 | dwery | [TK]D-Fender: yeah :( |
| 21:40.17 | *** join/#asterisk loos (n=luiz@189-18-123-189.dsl.telesp.net.br) |
| 21:40.22 | *** part/#asterisk loos (n=luiz@189-18-123-189.dsl.telesp.net.br) |
| 21:41.36 | WindowsUser | ah, my spa3102 doesn't seem to support a second call |
| 21:44.46 | dwery | the C470 is almost crap but dirt cheap and the only one within th ebudget |
| 21:47.47 | Katty | byebye |
| 21:47.53 | Katty | it's time to go home. |
| 21:49.05 | beek | good night Katty |
| 21:51.24 | WindowsUser | I was seriously thinking of an IP DECT phone until i saw that all the cordless phones for sale are like 3-5 handsets and "DECT 6.0" |
| 21:51.44 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 21:51.53 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 21:52.17 | dwery | WindowsUser: you might try Siemens OptiPoint series, they should be more professional |
| 21:54.01 | *** join/#asterisk noisewaterphd (n=noisewat@c-67-182-233-99.hsd1.ut.comcast.net) |
| 21:56.27 | dwery | [TK]D-Fender: Is DEVICE_STATE(SIP/xxx) known to work on 1.6.1.0 ? It seems I'm always getting NOT_INUSE |
| 21:57.08 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:58.54 | *** join/#asterisk lucasb (n=bussey@s154-5-252-231.bc.hsia.telus.net) |
| 21:59.30 | Maxxed | mmm.. i cant seam to get the key.IP_330.10.function.prim = âMessagesâ soft button to work right on a IP330.. |
| 21:59.49 | Maxxed | iv got it set just like the admin guide says, yet it dont do anything |
| 22:00.00 | Maxxed | erum.. does the line2 have to be registered or something? |
| 22:00.32 | Maxxed | il give it a shot for giggles |
| 22:00.43 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:01.57 | [TK]D-Fender | dwery: "core show application chanisavail" <-- |
| 22:02.25 | [TK]D-Fender | dwery: Yes, this is the answer, if it doesn't work, keep beating yourself over the ehad with it till ti does or I return from martial arts :) |
| 22:02.30 | [TK]D-Fender | CHOP CHOP peopl! |
| 22:02.33 | [TK]D-Fender | +e |
| 22:02.35 | [TK]D-Fender | BBIAb |
| 22:02.43 | dwery | [TK]D-Fender: chanisavail will do just fine ;) |
| 22:03.12 | dwery | I hate practising kung fu while doing * dialplans ;) |
| 22:03.55 | Kobaz | axeterisk |
| 22:04.33 | [TK]D-Fender | dwery: Tenshin Shoden Katori Shinto <- look it up |
| 22:05.07 | jdnWEST | hmmmm, as long as no nunchuks are involved, those should be illigal... |
| 22:05.47 | dwery | [TK]D-Fender: Japanese! seems nice. I practice Hung gar |
| 22:07.58 | dwery | well, I'll call it a day. Good night and ty |
| 22:12.45 | *** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 22:14.34 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 22:15.13 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 22:21.34 | *** join/#asterisk molnarp (n=molnarp@nx7400.ohsh.u-szeged.hu) |
| 22:22.20 | molnarp | Hi! Could anyone please help me setting up a TDM410P card with Asterisk? |
| 22:23.33 | molnarp | When I try to dial out through the Zap line, I get: |
| 22:23.38 | molnarp | Executing [06702457022@voip-univ-local-dist-mobil:1] Dial("SIP/203-00bdd720", "ZAP/g1/w0w06702457022|30") in new stack |
| 22:23.39 | molnarp | [Jul 15 00:14:16] WARNING[8975]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) |
| 22:23.39 | molnarp | <PROTECTED> |
| 22:23.40 | molnarp | <PROTECTED> |
| 22:24.21 | jameswf | ~pb |
| 22:24.22 | infobot | [~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 , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 22:24.25 | Maxxed | damnit this messages button is pissing me off! |
| 22:24.35 | Maxxed | i have to be missing something simple.. |
| 22:24.44 | molnarp | At the Zap incoming dialplan, I have set: Answer(), then Echo(), but if calling from outside, I can't hear anything |
| 22:24.54 | Maxxed | beh, il just jack with it another day when im thinking stright |
| 22:25.29 | molnarp | also, if I hang up the phone, Echo() remains running forever |
| 22:25.49 | molnarp | uh, sorry for flooding guys |
| 22:26.59 | molnarp | my zap channel seems to be up: http://molnarp.pastebin.com/m609f1c02 |
| 22:27.09 | *** join/#asterisk codestr0m (n=cbergstr@unaffiliated/codestr0m) |
| 22:27.32 | *** join/#asterisk jtodd (i=tpjnhgbc@ns.fox-den.com) |
| 22:27.33 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:28.19 | molnarp | zapata.conf I have: http://molnarp.pastebin.com/mcf36f3f |
| 22:28.20 | codestr0m | is there a definitive guide to why sip sucks? I'm looking for some opinions which are backed by facts that would help shed some light on this |
| 22:28.56 | jameswf | ~sip |
| 22:28.57 | infobot | i guess sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 22:29.16 | *** part/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 22:29.31 | jameswf | codestr0m: there are no bad protocals only idiot users who don't understand them |
| 22:30.18 | codestr0m | ok .thanks guy |
| 22:30.21 | codestr0m | guys* |
| 22:30.23 | *** part/#asterisk codestr0m (n=cbergstr@unaffiliated/codestr0m) |
| 22:31.14 | jameswf | guys? did he just call me fat WTF? |
| 22:31.53 | WindowsUser | does it matter? |
| 22:32.50 | Rob3Rt | jameswf, you're just _larger_ than most ppl. God loves u. |
| 22:33.26 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:33.36 | WindowsUser | im still not sure if being chubby will be an advantage or disadvantage for the apocolapyse |
| 22:37.57 | *** join/#asterisk [netman] (n=netman@158.Red-88-24-108.staticIP.rima-tde.net) |
| 22:42.34 | *** join/#asterisk ntbourey (n=laveur@c-76-110-22-251.hsd1.fl.comcast.net) |
| 22:42.39 | ntbourey | Hey Everyone |
| 22:43.33 | *** join/#asterisk Jumpie (n=zz@c-76-100-241-4.hsd1.md.comcast.net) |
| 22:46.22 | ntbourey | Can any one help me resolve an issue getting asterisk to start via /etc/init.d/asterisk start? |
| 22:48.22 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 22:48.35 | jameswf | ntbourey: probably |
| 22:48.50 | bmoraca | jameswf: i don't think so |
| 22:49.02 | ntbourey | I am getting a complaint about: "Cannot find specified TTY" |
| 22:49.04 | jameswf | bmoraca: Im an optimist |
| 22:49.33 | jameswf | ntbourey: probably cant find the specified tty |
| 22:49.44 | ntbourey | Yeah I figured as much how do I fix it? |
| 22:49.58 | jameswf | perhaps asterisk.conf |
| 22:50.05 | *** part/#asterisk molnarp (n=molnarp@nx7400.ohsh.u-szeged.hu) |
| 22:50.15 | ntbourey | Naw its in /usr/sbin/safe_asterisk |
| 22:50.44 | jameswf | well if you know why you asking? |
| 22:51.47 | ntbourey | Because I'm not sure what the heck it means I hacked the script and made it work with what I think is how the box is configured |
| 22:52.06 | ntbourey | but then it complains even more with Input/Output errors |
| 22:52.07 | jameswf | ntbourey: look what i found http://tinyurl.com/ml4sjb |
| 22:52.26 | ntbourey | heh |
| 22:52.31 | ntbourey | I've already been there |
| 22:52.48 | ntbourey | I wouldn't be asking if I hadn't tried that |
| 22:52.52 | jameswf | can you start asterisk outside the init script |
| 22:52.55 | ntbourey | Yes |
| 22:53.34 | jameswf | why not write your own init script or pull the one from trunk |
| 22:53.51 | ntbourey | That is the one from trunk |
| 22:54.04 | ntbourey | Its a fresh install of 1.6.1.1 |
| 22:54.16 | ntbourey | I downloaded it today and compiled it myself |
| 22:54.46 | leifmadsen | try 1.6.1 branch from SVN to see if it was a problem in that release |
| 22:55.04 | jameswf | ~sarcasm |
| 22:55.05 | infobot | Oh a sarcasm detector, that's a *really* useful invention! |
| 22:55.43 | ntbourey | Heh |
| 22:56.02 | *** join/#asterisk ctp (n=quassel@brsg-d9bee6a0.pool.mediaWays.net) |
| 22:58.11 | ntbourey | Any other thoughts |
| 22:58.50 | jameswf | lot's none related to your issue... perhaps a pastebin of the output... |
| 22:59.01 | leifmadsen | I was being serious... for some reason I think I heard something about that, which may be fixed now. No idea, but it sounded familiar, and 1.6.1.1 is pretty old now. |
| 22:59.12 | leifmadsen | heads off to try out some virtual golf |
| 23:00.21 | ntbourey | mpg123: no process killed |
| 23:00.22 | ntbourey | Asterisk ended with exit status 1 |
| 23:00.22 | ntbourey | Asterisk died with code 1. |
| 23:00.22 | ntbourey | Automatically restarting Asterisk. |
| 23:00.30 | ntbourey | http://pastebin.com/m74910d55 |
| 23:00.33 | ntbourey | Grr |
| 23:03.16 | Kobaz | you deaded it |
| 23:03.24 | ntbourey | What? |
| 23:05.44 | ntbourey | and I have the same problem in 1.6.0.10 |
| 23:20.28 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
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