| 00:00.10 | [TK]D-Fender | Jameno123, constantly |
| 00:00.26 | Jameno123 | what is his nick? |
| 00:01.01 | snuff-home | ruied_, cdr is the only thing the internal pgsql is used for.. rest is odbc sips/extensions etc.. |
| 00:02.51 | [TK]D-Fender | Jameno123, "file" |
| 00:03.41 | Jameno123 | ah!... :) shoulda known :( |
| 00:05.13 | Jameno123 | trying to figure out how to properly handle remote agents :( 2 servers linked via IAX2; queue on 1, sip phone on the other-- i return CONGESTION if the phone isnt available to accept another call, but as of 1.4.4(and now 1.4.6) that seems to cause the agents to be logged out. |
| 00:05.35 | Jameno123 | reverted to digging through channels/chan_agent.c to try and figure it out :( but not making much progress |
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| 00:37.23 | Hmmhesays | what is the svn address for the 1.4 gui? |
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| 01:30.55 | l0verb0y | hey hows it going |
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| 01:55.56 | elguille | hell |
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| 01:56.05 | elguille | o |
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| 02:20.42 | Strom_M | is this where i get halp connecting my asterisk to the interlol? |
| 02:21.15 | ManxPower | Strom_M: Do you have the correct tube adapter for the Internpol? |
| 02:21.48 | Strom_M | i think i have an isdn pots tube and i need a sip iax voip tube |
| 02:22.07 | Strom_M | but halp can't you fix it for me |
| 02:22.23 | anonymouz666 | haha |
| 02:22.33 | ManxPower | Exactly. Yo can download one from http://www.voip-inform.net/ but I think they are down right now. |
| 02:22.38 | Strom_M | oh noes |
| 02:22.47 | anonymouz666 | this is what I call 'nothing to do' |
| 02:22.49 | anonymouz666 | :D |
| 02:24.31 | rtcg | where can I get an example of how to setup a group of trunk lines so that outbound calls can rotate through a prioritized list of available trunks?? |
| 02:24.53 | Strom_M | rtcg: what kind of trunks are they? |
| 02:25.06 | rtcg | sip, iax and X101P trunks. |
| 02:25.07 | ManxPower | rtcg: You are being technology mtopyic. |
| 02:25.17 | ManxPower | rtcg: What you want to do is not simple. |
| 02:25.42 | Strom_M | like i said earlier: how can i connect my asterisk to the interlol? |
| 02:25.43 | rtcg | I was hoping to do a macro...rather than have each outbound context contain duplicate trunk routings. |
| 02:25.52 | rtcg | what is interlol??? |
| 02:25.53 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 02:26.14 | rtcg | is that a humorous mispelign of something? |
| 02:26.15 | Strom_M | a cow. a trampoline. together they fight crime. |
| 02:27.24 | kn0x | hmm |
| 02:28.02 | rtcg | OK...well, I guess I'll hard code the FXO ports and try to 'macro-ize' the SIP trunks. |
| 02:28.08 | ManxPower | rtcg: create a global variable to hold all the possible destinations then iterate thru them in a macro |
| 02:28.22 | rtcg | ok.. I think I understand that. |
| 02:28.40 | ManxPower | You'll have an issue parsing SIP destinations because they do not follow the format of all the other voip technologies. |
| 02:29.09 | rtcg | but I could separate things into multiple macros... |
| 02:29.18 | ManxPower | rtcg: would you like to see how I handled iterating thru multiple destinations? |
| 02:29.39 | rtcg | break the sip trunks into their own macro. Yes, ManxPower, I would *LOVE* to see that. |
| 02:29.52 | ManxPower | http://www.fnords.org/~eric/macros.inc |
| 02:29.56 | Hmmhesays | what the hell package has g++ in fedora? |
| 02:30.19 | rtcg | scorchedearth3d? |
| 02:30.23 | ManxPower | Hmmhesays: gcc-c++ |
| 02:30.24 | Hmmhesays | yum search g++ doesn't return any matches |
| 02:30.43 | ManxPower | do a search for "c++" |
| 02:30.52 | ManxPower | or just gcc |
| 02:31.25 | Hmmhesays | thanks ManxPower |
| 02:33.58 | ManxPower | pay special attention to DIAL_DEST |
| 02:34.18 | anonymouz666 | the damn mysql fetch can't be overwritten |
| 02:35.55 | ManxPower | It is set as part of the dialplan the call follows before it gets to the macro. exten => 1234.1.Set(DIAL_DEST[1]=SIP/123345) or exten => 1234,n,Set(DIAL_DEST[2]=Zap/G1) |
| 02:35.57 | rtcg | Wow, ManxPower ...I'm swimming in this.. this is....WAY beyond what I had for a dialplan. |
| 02:36.26 | JT | no such thing as a sip trunk ;) |
| 02:36.28 | ManxPower | rtcg: Just pay attention to the DIAL_DEST stuff |
| 02:36.41 | rtcg | I'm paying attention to the best of my ability. |
| 02:36.51 | rtcg | JT: why no such thing as a sip trunk...what do you call it then? |
| 02:36.52 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 02:37.03 | ManxPower | The macro is a mess, I admit that. It is a mix of 1.0 and 1.2 stuff |
| 02:37.06 | JT | sip |
| 02:37.15 | rtcg | and I am trying to go 1.4.... |
| 02:37.51 | rtcg | I had a working 1.2, but I was using deprecated 'stuff'...and I figured if I'm going to convert to using sip 'trunks'..then I'd start fresh and clean up the dialplan. |
| 02:38.37 | JT | sip connections then |
| 02:38.57 | rtcg | sip connection as an interface to the PTSN. |
| 02:39.06 | rtcg | PSTN / PTSN? |
| 02:39.07 | ManxPower | Try not to use the term "sip trunks". It makes you look like a FreePBX user. |
| 02:39.12 | Strom_M | PSTN |
| 02:39.51 | rtcg | or a PremierVoice subscriber... http://www.premiervoice.net/Services.cfm?service=SIPTrunking&page=Pricing |
| 02:40.33 | JT | there are lots of people mistakingly using the "sip trunking" term |
| 02:40.48 | rtcg | point taken, JT. |
| 02:42.07 | rtcg | ManxPower: IT seems like CFU_DEST is more what I should be looking at.. |
| 02:42.11 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 02:42.32 | ManxPower | That stuff prolly isn't working. We don't use that feature anymore. |
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| 02:44.01 | JT | rtcg: good good :) |
| 02:44.25 | rtcg | DId you really???? |
| 02:44.42 | ManxPower | THANK YOU! |
| 02:45.14 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 02:45.17 | JT | yeah, sip trunking redirects to sip telephony |
| 02:45.27 | JT | which is what the article was moved to |
| 02:45.28 | ManxPower | Do you get to keep the copyright of stuff you put in places like voip-info.org Wiki? |
| 02:45.47 | JT | ManxPower: not sure about voip-info, it's sort of like anarchy |
| 02:45.50 | asterisknerds | <PROTECTED> |
| 02:45.50 | JT | but wikipedia, yes |
| 02:46.08 | ManxPower | that is good |
| 02:46.30 | rtcg | oh no! It's asterisknerds! I see the redirect to sip technology! :) |
| 02:49.30 | ManxPower | TEMP MOVED? |
| 02:50.16 | snuff-home | aka cfwd all on a cisco.. its a CDR issue |
| 02:50.37 | ManxPower | Ah. |
| 02:50.55 | snuff-home | but if u put in enough forkcdrs etc and have a smart db u can charge properly when u have multiple clients |
| 02:51.04 | ManxPower | Yes. RDNIS is set when the Ciscos do CFWD |
| 02:51.55 | ManxPower | you can catch that and use a Goto into a pattern match somewhere to make the CDRS look right. |
| 02:53.21 | snuff-home | yes but it should be easier to catch.. aka if u have a incoming call on external line.. then a get a 302 redir.. then new cdr is from external num to another external num (worst case) and the only thing really correct is the context it went into |
| 02:54.07 | ManxPower | Give me a min to whip up something that might show what I mean. |
| 02:54.26 | snuff-home | i dont mind.. i've already got my solution |
| 02:54.29 | snuff-home | :) |
| 02:57.44 | ManxPower | http://pastebin.ca/609828 |
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| 03:04.34 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:04.58 | tengulre | Hi,all |
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| 03:30.23 | rtcg | If 's' is the 'start' context.... what is 'n' ?? Or better yet... where in this veritable see of asterisk documentation can one find the definitive guide to asterisk dialplan configuration? |
| 03:31.00 | ManxPower | "s" means "no destination given" |
| 03:31.23 | ManxPower | "n" priority is a 1.2 thing to help in wirting dialplan stuff |
| 03:31.39 | rtcg | but it's in the 1.4 sample extensions.conf file?? |
| 03:31.48 | rtcg | ohoh ohohoh |
| 03:31.50 | ManxPower | It should be. |
| 03:32.19 | rtcg | where is the documentation that describes 's' ?? I know I had found it once-upon-a-time.... |
| 03:32.21 | JT | well |
| 03:32.24 | JT | ~thebook |
| 03:32.25 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:33.09 | rtcg | Thanks again, JT. :) |
| 03:33.41 | ManxPower | "s" is where a call goes to when Asterisk does not know the destination of the call. Examples of this would be calls coming in on FXO ports (no DID), a call going into a macro, or a call coming in from an ITSP that is too stupid or evil to provide the dialed number. |
| 03:33.56 | rtcg | that is indeed where I had seen the concepts made clear.. |
| 03:34.15 | ManxPower | Also from when FXS port goes off hook and immediate=yes is set for that channel |
| 03:34.59 | rtcg | So, ManxPower, that is why the call coming in on my sip 'connection' didn't get processed by the context into which I sent it..because the sip provider was sending a destination along with the call! :) cool |
| 03:35.20 | ManxPower | Most providers do send the destination along with the call. |
| 03:35.29 | ManxPower | "s" is NOT a wildcard |
| 03:35.59 | snuff-home | _. is bad wildcard :) |
| 03:36.30 | ManxPower | Except for in a very few situations. |
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| 03:37.39 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
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| 03:48.02 | shmaltz | if I put my digium card in loop back using zttool, what should I see in asterisk console? |
| 03:48.36 | JT | it has nothing to do with asterisk |
| 03:51.16 | [TK]D-Fender | ~stdextens |
| 03:51.17 | jbot | "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 03:51.22 | [TK]D-Fender | There we go. |
| 03:51.26 | shmaltz | JT, so I shouldn't see anything in asterisk console? |
| 03:51.39 | JT | maybe errors, i dunno |
| 03:51.46 | shmaltz | also how do I get libnewt? |
| 03:51.47 | JT | but zttool is zaptel, not asterisk |
| 03:52.10 | shmaltz | JT, so it shouldn't change from red to yellow or green because of loop back? |
| 03:52.40 | JT | well loopback, you certainly can't use the pri for normal use :) |
| 03:52.47 | *** join/#asterisk bmg505 (n=leon@196.209.181.218) |
| 03:54.33 | shmaltz | JT, I understand, the problem is digium card is not coming up, all spans are red, so I'm trying to see if I can get them in green thru loopback |
| 03:55.07 | JT | i don't usually bother with loopback, never needed to |
| 03:55.14 | JT | but you'll need a loopback connector |
| 03:56.12 | *** join/#asterisk livesN[box] (n=chadkous@rrcs-24-123-233-204.central.biz.rr.com) |
| 03:56.20 | livesN[box] | anyone awake ? |
| 03:59.22 | kiscokid | shmaltz: did this card ever work? |
| 03:59.27 | ManxPower | [TK]D-Fender: Thank you. |
| 03:59.39 | shmaltz | kiscokid, yes it did |
| 04:00.10 | ManxPower | shmaltz: if you just plug a loop back into the port without doing anything else you should see the port go green |
| 04:00.13 | [TK]D-Fender | ManxPower, np, saves a lot of repetition. and FYI, refer people to ~sipnat when needed. I blogged a complete description. |
| 04:00.26 | shmaltz | ManxPower, I know that, but I aint there |
| 04:00.31 | ManxPower | ~sipnat |
| 04:00.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://aocomputing.net/wordpress/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:00.33 | shmaltz | no access till tommorrow |
| 04:00.47 | JT | shmaltz: so what's wrong with it? |
| 04:00.55 | ManxPower | then wait until tomorrow |
| 04:01.05 | ManxPower | If you have a red alarm nothing you can do will fix it. |
| 04:01.11 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 04:01.12 | shmaltz | JT, doesn't come up, it says RED when I do zap show status |
| 04:03.20 | shmaltz | CLI> zap show status |
| 04:03.22 | shmaltz | Description Alarms IRQ bpviol CRC4 |
| 04:03.23 | shmaltz | T4XXP (PCI) Card 0 Span 1 RED 0 0 0 |
| 04:03.25 | shmaltz | T4XXP (PCI) Card 0 Span 2 RED 0 0 0 |
| 04:03.27 | shmaltz | T4XXP (PCI) Card 0 Span 3 RED 0 0 0 |
| 04:03.28 | shmaltz | T4XXP (PCI) Card 0 Span 4 RED 0 0 0 |
| 04:03.29 | shmaltz | oh no, sorry |
| 04:03.31 | shmaltz | didn't realize |
| 04:03.32 | shmaltz | :( |
| 04:04.22 | ManxPower | OH GOD NO! THE FLOOD! NO! |
| 04:04.41 | ManxPower | shmaltz: you can also plug a T-1 crossover cable between any two spans |
| 04:04.59 | livesN[box] | anyone know of a flash (or javascript) sip or iax phone that i could embed into a webpage and use to call out ? |
| 04:05.01 | shmaltz | ManxPower, no I cannot since I don't have access :P |
| 04:05.09 | [TK]D-Fender | livesN[box], Go check the WIKI |
| 04:05.12 | [TK]D-Fender | ~wikis |
| 04:05.12 | jbot | it has been said that wikis is http://www.voip-info.org |
| 04:06.05 | livesN[box] | any kind of hint? I've been looking through the wiki for the last 10-15 minutes with no real success.... |
| 04:06.35 | ManxPower | shmaltz: As I said, you need physical access to the box. |
| 04:07.38 | livesN[box] | hmm.. actually might have finally stumbled upon something |
| 04:10.24 | livesN[box] | well nevermind.. |
| 04:11.08 | JT | livesN[box]: they cost money, and they're not advisable to use, anyway |
| 04:11.12 | Corydon76-home | Heh, a phone in javascript? |
| 04:11.18 | livesN[box] | wait a sec.. found another one.. |
| 04:11.20 | JT | ;) |
| 04:11.20 | livesN[box] | SipLinks |
| 04:11.38 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 04:12.46 | livesN[box] | bummer.. it's not there |
| 04:14.04 | JT | livesN[box]: they're a waste of time anyway |
| 04:14.10 | livesN[box] | why? |
| 04:14.20 | livesN[box] | I have a demo that I'd love for people to be able to listen to over the web. |
| 04:14.24 | livesN[box] | and interact with.. |
| 04:14.34 | kiscokid | interact how? |
| 04:14.36 | livesN[box] | don't need really anything fancy |
| 04:14.43 | livesN[box] | like "press 1 for this, press 2 for this other hting" |
| 04:14.45 | livesN[box] | kind of thing |
| 04:14.45 | JT | do you really want to support softphones running in web browers on random peoples' PCs? |
| 04:14.56 | JT | softphones are annoying enough |
| 04:14.58 | livesN[box] | it's aa marketing thing.. |
| 04:15.05 | JT | random web-initiated ones even more annoying |
| 04:15.06 | livesN[box] | people won't be making real calls. |
| 04:15.09 | JT | that doesn't make it any easier |
| 04:15.12 | kiscokid | I have a need for that kind of thing too |
| 04:15.29 | JT | well i suspect you want it to work |
| 04:15.47 | Corydon76-home | livesN[box]: do you absolutely need two-way or do you just need people to be able to listen? |
| 04:15.54 | kiscokid | I want to get people to start at a web page and enter into a MeetMe session |
| 04:16.13 | livesN[box] | if all I needed was for them to listen I'd be able to just play an audio file. |
| 04:16.27 | livesN[box] | I gotta give them the ability to interactive with the ivr |
| 04:16.34 | Corydon76-home | Well, you can also live stream a meetme |
| 04:16.34 | livesN[box] | no audio needs to come back really. |
| 04:16.46 | kiscokid | livesN: did you find any others? |
| 04:16.59 | livesN[box] | found a couple of active-x clients that I haven't gotten too far into et.. |
| 04:17.00 | livesN[box] | yet |
| 04:17.04 | Corydon76-home | livesN[box]: you can simulate that, without implementing a softphone |
| 04:17.08 | JT | what's the bet these web pages will be really annoying anyway ;) |
| 04:17.14 | ManxPower | What you want to do is not a trivial thing |
| 04:17.33 | livesN[box] | Corydon76-home, yeah I know... just thought if there was a quick and easy flash-based sip phone could save me some time. |
| 04:17.35 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 04:17.40 | Corydon76-home | In fact, Javascript would be a possible way to do that |
| 04:17.56 | ManxPower | I hate cell companies |
| 04:18.10 | ManxPower | I'm trying to get It's a Small World as a ringtone. |
| 04:18.40 | livesN[box] | Corydon76-home, yeah -- I realize I could take all of the prompts from the IVR and script it all through Javascript... just trying to save some time |
| 04:19.22 | [TK]D-Fender | ManxPower, Thats a bonus of my phone's ability to use MP3's for them... |
| 04:19.26 | ManxPower | search the mailinglist archives. |
| 04:19.36 | ManxPower | [TK]D-Fender: I got a cheap prepay 2 years ago. |
| 04:20.18 | [TK]D-Fender | ManxPower, I shelled out 100$ for mine (Motorola E815). Was worth it. Got a 1 GB card (free), and use its camera regularly and mp3's on occasion. |
| 04:20.38 | [TK]D-Fender | ManxPower, Now I'm looking for a phone with a GOOD camera w/ zoom lens |
| 04:20.49 | JT | mp3s ;) |
| 04:20.54 | ManxPower | I'm locked to verizon because of my location. Mostly I've not been impressed with any of the phones, there's always a catch. |
| 04:20.54 | JT | haha good camera... phone |
| 04:20.58 | [TK]D-Fender | ManxPower, OpenMoko lacks a camera right now.... |
| 04:21.05 | [TK]D-Fender | JT : VERY possible.... |
| 04:21.26 | ManxPower | I don't mind the lack of a camera, the lack of WiFi is what keeps me from considering it, |
| 04:21.27 | JT | i'd debate how good any camera that small is |
| 04:21.47 | ManxPower | At this point it does not give me $400 in value |
| 04:22.42 | ManxPower | Also the GSM only, while totally understand why, keeps it from being a viable phone for me. |
| 04:23.17 | ManxPower | But if it had WiFi VoIP it could work WELL for me |
| 04:23.19 | [TK]D-Fender | ManxPower, Indeed... MAJOR bummer |
| 04:23.37 | ManxPower | Oh, I think the whole world should be GSM and most of it is. |
| 04:23.46 | [TK]D-Fender | ManxPower, I'd shell out more for it to have both. |
| 04:24.20 | JT | i think gsm should be obliterated |
| 04:24.24 | JT | and it's going that way |
| 04:24.26 | JT | with 3g |
| 04:24.33 | JT | gsm is a crap technology |
| 04:24.47 | JT | for anywhere but densly populated small countries |
| 04:24.57 | ManxPower | I don't care exactly WHAT tech is used as long as there is ONE standard (for volume discounts) and does not have carrier lock in. |
| 04:25.07 | ManxPower | GSM is mostly that right now. |
| 04:25.34 | ManxPower | Did you know that I can't even convert my prepay verizon phone into a contract plan without getting a new phone? |
| 04:25.53 | JT | what can i say, america is f*cked ;) |
| 04:28.32 | ManxPower | When I was in Europe, I bought a used GSM phone from a woman I met on the plane flight and used that phone for the month I was there, changing carriers every country I went to. |
| 04:29.23 | JT | yeah, in australia, it's a fairly free market for phones too |
| 04:30.03 | kiscokid | Its so messed up that the iPhone is GSM but locked to AT&T |
| 04:31.28 | JT | iYawn |
| 04:31.40 | JT | another Apple product released, another iYawn |
| 04:31.57 | kiscokid | Good integration of media with a phone |
| 04:32.28 | JT | still sounds quite boring |
| 04:32.31 | JT | media and a phone |
| 04:32.46 | [TK]D-Fender | JT : http://www.youtube.com/watch?v=rw2nkoGLhrE |
| 04:32.55 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:33.02 | JT | i'm at work... |
| 04:42.37 | J4k3 | well |
| 04:42.44 | J4k3 | first you get a cheap LCD picture frame |
| 04:42.51 | J4k3 | attach a shitty digitizer that will fail in a few months |
| 04:43.09 | J4k3 | build a cheezy circuit board with a GSM dev kit radio attached to it |
| 04:43.19 | J4k3 | bury the battery as deep and hard to repair as you can inside |
| 04:43.31 | J4k3 | and close it up in a somewhat fragile chassis |
| 04:43.49 | J4k3 | and you have the Apple iPhone. |
| 04:44.10 | JT | yep |
| 04:44.19 | Qwell | You forgot port osx |
| 04:44.25 | Qwell | or, rather "osx" |
| 04:44.28 | J4k3 | pft |
| 04:44.38 | J4k3 | #1 - osx isn't anything to brag about |
| 04:44.47 | Qwell | didn't say it was |
| 04:44.48 | J4k3 | #2 - the iphone is locked down hard enough that the OS doesn't matter. |
| 04:45.15 | Qwell | it runs osx for the sake of saying it runs osx |
| 04:45.22 | J4k3 | yeah, except it runs "osx" |
| 04:45.30 | Qwell | it runs "osx" for the sake of saying it runs osx |
| 04:45.35 | J4k3 | which is "whatever pile of crap Apple decided to call its operating system on this product" |
| 04:45.39 | JT | osxce |
| 04:45.47 | J4k3 | gimmie a wm5 phone |
| 04:45.49 | J4k3 | kthx |
| 04:45.49 | [TK]D-Fender | Well it's apparently not that fragile, and they are supposed to open up 3rd party apps last I heard |
| 04:45.52 | Qwell | that's too close to goatse for comfort |
| 04:46.16 | JT | prnounced "oh sexy" but with a sarcastic tone in the voice |
| 04:46.27 | J4k3 | goatse, apple customer... its all about the same physical condition. |
| 04:47.16 | J4k3 | I'm in a pretty shitty position, I may have to cancel (and pay a contract cancelation) on verizon due to how worthless their roaming partner Alltel is. |
| 04:47.27 | J4k3 | and that half the area I stomp in, Alltel is the only CDMA carrier. |
| 04:47.42 | Qwell | not really Verizons problem... |
| 04:47.48 | Qwell | unfortunately, that'll be their reason for denial |
| 04:47.53 | J4k3 | yeah, but its verizon's problem when I cancel service because their roaming partner sucks. |
| 04:47.59 | J4k3 | I can pay $175 at any time |
| 04:48.02 | J4k3 | and my contract is over. |
| 04:48.05 | Qwell | oh, right, you said and pay it |
| 04:48.08 | Qwell | yeah, that sucks |
| 04:48.11 | J4k3 | and considering I pay about that per month for service, its not really anything to cry over. |
| 04:48.55 | J4k3 | I'll happily go a month with a 4 watt AM CB in my car for "safety communication" knowing the next month I won't have a worthless-half-the-time cellular phone. |
| 04:49.19 | J4k3 | at least I have half a prayer of being heard with 4 watts at 27 mhz :P |
| 04:49.53 | ManxPower | Prolly hear you 1/2 way around the world |
| 04:50.06 | J4k3 | yep... assuming the local noise isn't that painful. |
| 04:50.20 | J4k3 | CB goes through nasty spells, at least in the southern US |
| 04:50.36 | ManxPower | How does 27Mhz interact with the ionosphere |
| 04:50.41 | J4k3 | in the early 90s trying to use a CB in Houston was a waste of time. All you heard was noise from south america and mexico. |
| 04:51.28 | J4k3 | it reacts like 10 meter ham, pretty much |
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| 04:51.57 | J4k3 | sometimes its wide open, most of the time its somewhat closed (which is a good thing) |
| 04:52.07 | JT | ssb 27MHz would be superior |
| 04:52.10 | J4k3 | it is |
| 04:52.12 | rkeels | Hey what's up all |
| 04:52.28 | J4k3 | unluckily most ssb cb rigs are either crap, or expensive |
| 04:52.28 | JT | do your CBs do SSB too? |
| 04:52.29 | ManxPower | ssb? |
| 04:52.34 | JT | single side band |
| 04:52.35 | J4k3 | nah, I can't afford a good SSB rig. |
| 04:52.37 | rkeels | Anyone know of anyone other providers relative to iaxtel.com |
| 04:52.50 | JT | ManxPower: am without the pointless waste |
| 04:52.52 | J4k3 | that'd take a couple months of cellular bills to pay for :) |
| 04:53.10 | J4k3 | yeah, normal CBs transmit (iirc) 4 watts dead carrier and 1 watt of modulation |
| 04:53.20 | JT | heh, all the 27MHz CB rigs sold here in Australia in the last decade have AM and SSB |
| 04:53.23 | J4k3 | the carrier is there to let your cheap-ass CB actually 'center up' and hear my 1 watt of modulation with good clarity |
| 04:53.27 | JT | well AM modulation is at least 50 |
| 04:53.30 | J4k3 | SSB requires precision everything to get a decent conversation |
| 04:53.33 | JT | well AM modulation is at least 50% wasted power |
| 04:53.59 | JT | carrier, and repeated data on each side of the carrier |
| 04:54.04 | J4k3 | hrm... I need some aussie CBs... fcc be damned! j/k |
| 04:54.22 | J4k3 | I thought .au went to UHF for CB? |
| 04:54.22 | JT | we have both |
| 04:54.29 | JT | 40chs 27MHz cb |
| 04:54.40 | JT | 40chs 477MHz FM cb |
| 04:55.02 | J4k3 | ahh, we have "frs" (uhf, 7(iirc) channels unlicensed), "murs" (vhf 5 channel, repeaters allowed), and 27 mhz CB, 40 channel |
| 04:55.10 | ManxPower | All the RF stuff that I work with is broadcast TV, and CATV over coax. |
| 04:55.21 | JT | you forgot gmrs |
| 04:55.25 | JT | small licence fee |
| 04:55.28 | J4k3 | gmrs is a $80/5 year license. |
| 04:55.44 | JT | we have repeaters on UHF CB too |
| 04:55.51 | JT | ch 1-8 are repeater channels |
| 04:55.52 | J4k3 | its pretty good for flat terrain with few trees, the folks I know with gmrs repeaters around here don't seem to like them |
| 04:56.00 | J4k3 | I say lay on the power, they act chicken-y about it. |
| 04:56.01 | JT | 31-38 are their inputs |
| 04:56.07 | ManxPower | I've learned to lust after the 700Mhz band |
| 04:56.21 | JT | and there's pretty much no enforcement of power limits |
| 04:56.37 | J4k3 | JT: pretty much the same in the US until you stomp something licensed. |
| 04:56.43 | JT | so on major metro UHF CB repeaters, you sometimes need over 50Watts of power to get hear on the repeater |
| 04:57.06 | J4k3 | ick |
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| 04:57.12 | JT | most die hard metro CBers run 100-200W Amps from their home base |
| 04:57.14 | FuriousGeorge | hey all |
| 04:57.16 | JT | 25W+ from car |
| 04:57.17 | J4k3 | that was the big "implosion" of 27 watt CB in the USA |
| 04:57.29 | J4k3 | amps got cheap, everybody started running them |
| 04:57.41 | JT | most people use ex-repeater Motrola Power Amps or similar |
| 04:57.48 | J4k3 | and everybody got tired of hearing the same people spread across 5 channels up and down, halfway across the nation |
| 04:57.50 | JT | oh, they're still not cheap |
| 04:57.52 | J4k3 | and gave up |
| 04:57.56 | JT | unless you know where to look :) |
| 04:58.09 | JT | yeah well UHF FM doesn't propogate like that |
| 04:58.16 | J4k3 | ah, 27 amps are cheap here. I'm sure they work like cheap amps ;) |
| 04:58.19 | J4k3 | yeah, luckily. |
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| 04:58.41 | JT | i have a pile of 100W VHF Hi/UHF amps sitting at homr |
| 04:58.42 | JT | home |
| 04:58.48 | JT | 50W UHF amp in the car ;) |
| 04:58.56 | J4k3 | hehe nice |
| 04:59.36 | J4k3 | radio mobile says I'd likely be happy with legal MURS gear for business applications, off the top of my home tower. |
| 04:59.37 | JT | but to be honest, for simplex short range stuff, i use the commerical 800MHz band |
| 04:59.53 | J4k3 | hehe |
| 04:59.54 | JT | great quality, no intereference |
| 05:00.11 | J4k3 | around here 800 is dead except for cellular |
| 05:00.13 | JT | and almost no hobbyists own 800meg gear, which is a bonus ;) |
| 05:00.32 | J4k3 | well, the hospitals have like 6 narrow channels around the cell stuff. |
| 05:00.55 | JT | for long range metro, i use commerical uhf repeater if i can |
| 05:00.55 | J4k3 | hehe yeah. my big issue is the newer the gear, the more often you need expensive/rare programming cables. |
| 05:01.03 | JT | hah |
| 05:01.12 | J4k3 | at least in the UA |
| 05:01.13 | J4k3 | er USA |
| 05:01.14 | JT | rarity is only a problem in unusual brands |
| 05:01.27 | JT | programming anything motorola/kenwood is no issue |
| 05:01.32 | J4k3 | yeah, I've noticed all the moto stuff seems to be online |
| 05:01.47 | JT | or available at a price <_< |
| 05:01.59 | J4k3 | yeah |
| 05:02.09 | J4k3 | crystals got too damned expensive |
| 05:02.14 | J4k3 | I thought $5/rock was pricey back in the day |
| 05:02.20 | J4k3 | $20 is just over the top |
| 05:02.25 | JT | pretty much the only thing that is holy grail with motorola is writing flashcode |
| 05:02.36 | JT | that's still seriously rare/expensive |
| 05:02.46 | Corydon76-home | Have MURS radios come down in price yet? |
| 05:03.46 | Corydon76-home | I've been considering getting a MURS license for a charity I help run, but the radios are too damn expensive still for the quantity we'd need |
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| 05:04.07 | J4k3 | MURS is unlicensed, at least in the USA |
| 05:04.11 | JT | how expensive can the radios be? |
| 05:04.28 | Corydon76-home | $200 apiece, last I checked |
| 05:04.44 | J4k3 | yeah |
| 05:05.40 | JT | can you use commercial radios? |
| 05:06.01 | J4k3 | I believe, if they were originally sold for the MURS channel set |
| 05:06.03 | Corydon76-home | Only for 2 of the 5 frequencies |
| 05:06.12 | J4k3 | yeah |
| 05:09.27 | JT | what do you mean? |
| 05:09.43 | JT | if the radio covers the frequency range as part of the design specifications? |
| 05:10.12 | Corydon76-home | only 2 of the 5 frequencies are in the business band |
| 05:11.05 | Corydon76-home | the other 3 are in the 151MHz range |
| 05:11.27 | Corydon76-home | (as opposed to 154MHz for business band) |
| 05:11.30 | JT | i thought commercial radios generally covered a fairly wide range, but okay :) |
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| 05:15.39 | J4k3 | anybody that charges $50+ for shipping should be taken out in the street and shot. |
| 05:15.44 | J4k3 | for a portable radio |
| 05:16.36 | Corydon76-home | JT: I'm used to radios that are frequency-limited, so you can't use frequencies for which you aren't licensed |
| 05:16.49 | Corydon76-home | i.e. ham |
| 05:16.59 | JT | J4k3: hong kong? |
| 05:17.06 | J4k3 | yes. |
| 05:17.18 | JT | J4k3: well obviously you have to combine both charges :P |
| 05:17.47 | JT | Corydon76-home: i'm used to programming radios with whatever the VCO can handle ;) |
| 05:17.56 | Corydon76-home | Heh |
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| 05:23.47 | J4k3 | JT: yeah, unluckily ebay plays into their game and doesn't make it easy |
| 05:23.55 | J4k3 | you have to dig through all this azn riceboy radio shit. |
| 05:24.10 | J4k3 | DOES THIS HANDIE TALKIE COME WITH A FART PIPE AND SPOILER? |
| 05:24.24 | J4k3 | god knows they all come with obnoxious blue LEDs! |
| 05:27.37 | JT | heh |
| 05:27.48 | JT | come on, they're definitely not the only offenders |
| 05:28.01 | JT | *cough* Uniden 396T |
| 05:33.02 | JT | J4k3: where do you buy your routerboards from? |
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| 06:07.43 | J4k3 | JT: I was getting them from titan, but I'm not so sure if I'll buy any more from them. |
| 06:08.56 | JT | if you order from them direct, $50 fee is the order is under $1000, what a rort |
| 06:09.14 | J4k3 | yeah, and they charge you retail. |
| 06:09.22 | JT | the only place in australia i can find with prices on line, the prices are ridiculous |
| 06:09.39 | JT | $59 for a 133 or whatever? |
| 06:09.49 | J4k3 | 133c3 |
| 06:09.55 | J4k3 | the 133 is 90-100 |
| 06:10.05 | J4k3 | 133c3 = 1 slot |
| 06:11.12 | JT | $89 for the 133 on their site |
| 06:11.16 | J4k3 | yeah |
| 06:11.26 | J4k3 | brb, rebooting |
| 06:11.37 | J4k3 | XP is acting like the piece of crap it is, I'm hoping it defrag'd away |
| 06:11.47 | *** join/#asterisk Strom_M (n=strom@12.175.45.219) |
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| 06:13.17 | J4k3 | whee, no more excessive la. |
| 06:13.18 | J4k3 | er lag. |
| 06:13.28 | JT | hegh |
| 06:13.48 | JT | it's looking alltogether too hard or uneconomical to go with routerboards |
| 06:13.52 | J4k3 | yeah |
| 06:14.32 | J4k3 | I wonder how fast the cheap-o 175 mhz mips atheros soc's move data |
| 06:14.44 | J4k3 | a lot of them go for peanuts here in the US |
| 06:14.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:14.51 | JT | hrm |
| 06:14.58 | J4k3 | like the dlink di-624 |
| 06:15.03 | JT | not sure which ones exactly you refer to |
| 06:15.16 | JT | gumstix are nice, i just wish they had more real world interfacing |
| 06:15.23 | J4k3 | the routers that claim "108 mbit" capability |
| 06:15.27 | J4k3 | yeah, these are pretty well closed |
| 06:15.33 | J4k3 | only gpio you might find is attached to LEDs |
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| 07:10.36 | jameswf | lol |
| 07:11.01 | jameswf | rodents in the server room |
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| 08:21.56 | Swat2 | Anyone able to tell me what the correct syntax is for trunking 2 asterisk servers via SIP, can find heaps on IAX2 but nothing on SIP :( |
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| 08:31.23 | kiwi`MouThon | lo vry body |
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| 08:42.20 | dikdust | hi, I'm using asterisk 1.4.6 and I would like to know if using ivr is possible to bypass it if I receive a fax call and route the call to the fax machine |
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| 09:07.37 | IgorG | hello, guys. I have little quick question, not only about asterisk. Is any standarts exists on telephone vertical codes? |
| 09:08.07 | Gh0sty | vertical codes? |
| 09:08.34 | IgorG | yes |
| 09:09.09 | IgorG | codes that used for additional features activation |
| 09:09.18 | IgorG | or deactivation |
| 09:09.44 | Habbie | i don't understand the question |
| 09:10.35 | IgorG | hmm, in other words |
| 09:10.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:11.43 | IgorG | is any document that desribes that, for example, code *68 must be used for activation of unconditional call forward, and *66 for call forward on busy |
| 09:11.50 | Habbie | ah like that |
| 09:11.57 | Habbie | my best guess would be isdn specs really |
| 09:12.15 | Habbie | as it indeed seems that these codes are pretty standardised between telcos and isdn hardware |
| 09:12.40 | *** join/#asterisk gardo (n=gardo@121.97.176.180) |
| 09:12.45 | madcap | http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 09:13.18 | Habbie | why are they called vertical? |
| 09:13.32 | IgorG | thanks |
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| 09:17.40 | madcap | because you use to dial two verticals to activate them on old rotary phones. |
| 09:19.07 | Habbie | ah |
| 09:34.17 | JT | service codes have nothing to do with isdn specs |
| 09:34.42 | JT | your american service codes are completely different to our australian ones, for example :) |
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| 09:40.01 | Habbie | JT, i just noticed that indeed :) |
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| 09:49.44 | *** join/#asterisk dickyjoe (n=richardl@124.149.102.183) |
| 09:50.16 | dickyjoe | hello |
| 10:00.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:01.22 | J4k3 | psk31 over ssb cb for sms-like activity using PDAs... now thats a sick idea. |
| 10:02.45 | *** join/#asterisk VijayG (i=VijayG@220.224.38.29) |
| 10:04.39 | VijayG | Hello |
| 10:05.04 | VijayG | I need to create a dialplan in asterisk, with a limit that none of the call should go above 30 seconds |
| 10:05.07 | VijayG | how can i do that? |
| 10:05.23 | VijayG | or any other application in which i can get this kind of functionality |
| 10:05.30 | snuff-home | set(TIMEOUT(absolute)=30) |
| 10:05.39 | snuff-home | before calling dial |
| 10:05.47 | VijayG | ok |
| 10:05.52 | snuff-home | but that includes ringing.. |
| 10:06.03 | VijayG | but will this disconnect the call after 30 seconds |
| 10:06.06 | snuff-home | yep |
| 10:06.09 | VijayG | ya, thats fine with me |
| 10:06.10 | VijayG | ok |
| 10:06.23 | VijayG | in extentions.conf, before calling dial, i should put this, right? |
| 10:06.28 | snuff-home | yep |
| 10:06.31 | VijayG | ok |
| 10:06.33 | VijayG | let me try |
| 10:06.56 | VijayG | and this timeout can be as low as 5 seconds also, right? |
| 10:07.13 | VijayG | set(TIMEOUT(absolute)=5) |
| 10:07.22 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:07.35 | snuff-home | yes.. |
| 10:08.44 | VijayG | thanks |
| 10:10.03 | Chris-NB | hi |
| 10:10.17 | Chris-NB | is it possible to limit parallel calls on a iax trunk? |
| 10:28.39 | *** join/#asterisk Dovid (n=Dovid@79.178.24.155) |
| 10:31.32 | dickyjoe | Hi all. Whats the easyist way to allow connections to an asterisk server which is hiding behind a NAT firewall |
| 10:36.01 | JT | port forwarding |
| 10:36.02 | SktyNick | Get rid of the NAT firewall :-) |
| 10:42.36 | dickyjoe | do you just need to port forward UDP 5060 from the outside of the to the firewall or are there extra ports? These are SIP clients I am talking about |
| 10:45.25 | SktyNick | So your Asterisk box is behind the NAT, or your clients are behind the NAT? |
| 10:45.44 | dickyjoe | asterisk box and possibly clients also |
| 10:45.46 | SktyNick | And if your trying to double NAT (Ie. Asterisk is behind one NAT and your clients are behind another) you may aswell give up now |
| 10:48.06 | JT | SktyNick: wrong |
| 10:48.27 | JT | all you need to do is port forward on the gateway to the asterisk box, set externip= and localnet= in sip.conf |
| 10:50.26 | *** join/#asterisk jkif1 (n=jkiffmey@unaffiliated/vorondil) |
| 10:52.16 | JT | forward udp 5060 and udp 10000 to 20000 |
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| 10:52.35 | *** mode/#asterisk [+o Deeewayne] by irc.freenode.net |
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| 10:57.57 | lilalinux | My bristuff is working now with 2 hfc cards (1st TE, 2nd NT) but I get various messages: "== Primary D-Channel on span 1 down" and "chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway!" |
| 10:58.47 | lilalinux | sometimes the caller gets "The calling party is not available", and sometimes it gets digital noise |
| 11:02.42 | deegan | lilalinux: We had that problem too, swapped the card around on different PCI-slots, tried different machines (all from dell) but still the same problem. the sollution was using the card in the previous PBX and basicly having that as one huge zap-router. |
| 11:05.35 | lilalinux | deegan: you mean the problem is that 2 hfcs are too much for one machine? |
| 11:06.53 | lilalinux | where's the difference between your previous PBX and the new one? |
| 11:07.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:08.12 | deegan | lilalinux: The machines that spewed out errors where all intel machines and the one that it works in is a amd with nforce chipset. |
| 11:08.34 | deegan | lilalinux: we didnt have 2 cards though, we only had 1. |
| 11:10.15 | lilalinux | is your AMD 64bit? |
| 11:10.57 | JT | lilalinux: so did you get NT mode to work at all? |
| 11:10.58 | deegan | No it's a sempron or possibly an athlon. |
| 11:11.21 | deegan | 32-bit to answer your question. |
| 11:13.59 | *** join/#asterisk ai-a[wrk] (n=joe@megan.healthnet.co.uk) |
| 11:20.34 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 11:21.09 | kombi | rtc lost some interrupts <-- how to fix that? |
| 11:21.32 | *** join/#asterisk mightnare (n=mike@p5124-ipad03motosinmat.mie.ocn.ne.jp) |
| 11:21.58 | my007ms | what is the good why to make asterisk send to CRM the call id and the call status ? |
| 11:22.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 11:22.08 | my007ms | s/why/way |
| 11:22.37 | ai-a[wrk] | my007ms: asterisk can interact with mysql quite easy. |
| 11:22.53 | ai-a[wrk] | also, you can run perl / scripts on calls. |
| 11:23.15 | my007ms | is that can keep the CRM update for what is happen |
| 11:23.24 | ai-a[wrk] | you have to plug it together. |
| 11:23.34 | ai-a[wrk] | asterisk or me are not going to do it. |
| 11:23.36 | pj_ | I'm using the Queue application and I noticed that when agents are already on the line, asterisk still "try" in rrobin algorithm... Then the agent can take it on its second line, however I'd rather have it call the agents that are _really_ free. any clue ? |
| 11:24.04 | ai-a[wrk] | pj_: good question.. |
| 11:24.30 | *** join/#asterisk dikdust (n=dikdust@2001:1418:1c7:0:211:11ff:fed3:ecc7) |
| 11:24.32 | pj_ | I tried setting the "call limit" to 1, but it doesn't work "as intended" - the agent can't transfer anymore for instance (xcept blind xfer) |
| 11:24.34 | ai-a[wrk] | you can force the line it uses.. but then if they are using #2 the #1 will ring. |
| 11:24.40 | my007ms | ai-a[wrk], not me also :) we have developers here who ask me to only send him this data in sql/http/... etc any thing |
| 11:24.53 | ai-a[wrk] | so do it.. |
| 11:25.07 | my007ms | i can not do my part :) who i get this data |
| 11:25.09 | my007ms | to send |
| 11:25.12 | ai-a[wrk] | update foo set status = whatever where ext = whichever |
| 11:25.35 | ai-a[wrk] | they can do what they please with that table then. |
| 11:25.59 | ai-a[wrk] | my007ms: you a zap / isdn / asterisk expert ? |
| 11:26.01 | my007ms | ai-a[wrk], and URL/doc take abut this |
| 11:26.05 | pj_ | ai-a[wrk]: basically I have a phone that support multiline and I want to emulate the behavior of a "standard" by registering several lines and subscribing them all to a queue... Then incoming calls would fall on the first "free" line |
| 11:26.26 | pj_ | except asterisk consider a line is "free" when there's already someone talking :/ |
| 11:26.33 | ai-a[wrk] | pj_: im no expert, came here for help myself ;) |
| 11:26.51 | ai-a[wrk] | but its a problem we will have later i can imagine. |
| 11:26.54 | my007ms | ai-a[wrk], i know how to use asterisk well but don't know how to make it work with other apps |
| 11:27.02 | pj_ | ai-a[wrk]: Yeah but you're answering, it is good enough ;) What's your problem ? |
| 11:27.11 | pj_ | I'm no expert either but you never know ;) |
| 11:27.16 | my007ms | hehehe |
| 11:27.20 | ai-a[wrk] | my007ms: installed a Sangoma A101D-Esco Cancellation ISDN Card on Asterisk system, seems to be installed and detected by wanrouter hwprobe, zapscan says "Scanning for zaptel devices...OK", using Asterisk-Gui, how do i use this card to make calls? when i dial 9X says Service 9X unavailable (asterisk output - http://pastebin.ca/610272 ) |
| 11:27.29 | ai-a[wrk] | *echo! |
| 11:28.31 | my007ms | ai-a[wrk], i can not find zap any where in ur post |
| 11:28.47 | ai-a[wrk] | nope. thats the * output. |
| 11:29.01 | my007ms | ai-a[wrk], Call rejected by 213.249.208.85 this was from iax not zap |
| 11:29.17 | ai-a[wrk] | how do i set up asterisk to use the zap for 9xxx when i dial 90111 from my phones ? |
| 11:29.56 | mvanbaak | exten => _90XXX,1,Dial(Zap/g1/${EXTEN}) |
| 11:30.25 | ai-a[wrk] | using the *-gui ;) |
| 11:30.49 | ai-a[wrk] | have exten=_9X!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1}) |
| 11:30.51 | pj_ | ai-a[wrk]: your card doesn't seem to be working |
| 11:30.57 | my007ms | ai-a[wrk], if u will asterisk GUI then why not use trixbox or asterisk now !! |
| 11:31.08 | ai-a[wrk] | my007ms: im using asterisk-now |
| 11:31.38 | pj_ | because ztcfg should not tell you "failed" I guess |
| 11:31.54 | pj_ | if you "zap show channels" does it show ? |
| 11:32.26 | ai-a[wrk] | <PROTECTED> |
| 11:32.30 | ai-a[wrk] | thats all it shows. |
| 11:32.41 | pj_ | then your card is just not recognized |
| 11:32.45 | ai-a[wrk] | zapscan says Scanning for zaptel devices...OK |
| 11:32.52 | pj_ | it's lying to you ;) |
| 11:32.52 | *** join/#asterisk UVSoft (n=UVSoft@motorola154-31.ip.PeterStar.net) |
| 11:32.56 | ai-a[wrk] | lol |
| 11:33.00 | pj_ | you should see your 32 channels |
| 11:33.10 | ai-a[wrk] | i see. |
| 11:33.11 | pj_ | 31 sorry |
| 11:33.14 | *** join/#asterisk powerkill (n=powerkil@84.205.154.247) |
| 11:33.22 | pj_ | "ZT_SPANCONFIG failed on span 1: No such device or address (6)" is bad bad |
| 11:33.32 | pj_ | gotta be afk, bbl |
| 11:33.54 | ai-a[wrk] | wait, thats wrong pastebin. |
| 11:34.24 | ai-a[wrk] | grrr. thats old one, its last week.. its the reply at the top. |
| 11:34.33 | ai-a[wrk] | ignore the main pastebin post. |
| 11:35.12 | ai-a[wrk] | was only the ast-output to be seen - http://pastebin.ca/610288 |
| 11:37.06 | UVSoft | hi, there! im writing a driver for an fxs device, and there's a problem with callerid: the phone, connected to the device, doesn't display it. so the question is do i need to implement something in my driver to allow cid? or cid doesn't depend on drivers and the problem is in the asterisk/zapata.conf'iguration (http://pastebin.ru/59731)? thanks |
| 11:43.20 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
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| 11:49.29 | ai-a[wrk] | Do i need my ISDN card connected to an ISDN Line to get zaptel / asterisk to detect the channels ? |
| 11:50.09 | my007ms | ai-a[wrk], are u PRI ? |
| 11:50.40 | ai-a[wrk] | yep. |
| 11:51.50 | ai-a[wrk] | its a Sangoma, using Wanpipe to emulate zaptel i guessing.. |
| 11:52.02 | HaMYaI | coppice: does unicall-0.0.5pre1 work for asterisk 1.4.x? |
| 11:52.11 | coppice | no |
| 11:53.24 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 11:53.49 | *** join/#asterisk dickyjoe (n=richardl@124.149.102.183) |
| 11:56.05 | Uatec_ | hey, i'm using mix monitor to record my calls, but for some reason ${uniqueid} and ${CDR(uniqueid)} are returning an empty string, so i can't record any of my calls now, they just hang up cos the filename is empty... |
| 11:56.08 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 11:56.13 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 11:56.25 | Uatec_ | exten => _9.,2,MixMonitor(${CDR(uniqueid)}.wav} |
| 11:57.11 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-40-34.lns3.syd7.internode.on.net) |
| 11:57.44 | Uatec_ | why is that? |
| 11:57.49 | Uatec_ | oooh |
| 11:57.51 | Uatec_ | a stray } |
| 11:58.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:58.59 | HaMYaI | coppice: I have "chan_unicall" loaded on my 1.4.0 box but when I tried using Dial(Unicall/...) I got "chan_unicall.c:2599 handle_uc_event: Unicall/32 event Protocol failure" |
| 11:59.29 | HaMYaI | coppice: which part should I look into? |
| 11:59.43 | coppice | I have not provided any support for asterisk 1.4. I have no idea what happens if you try building my software with it |
| 11:59.59 | HaMYaI | coppice: ok |
| 12:00.29 | HaMYaI | coppice: which version should work with 1.2.x? |
| 12:01.04 | coppice | 0.0.3pre11 |
| 12:02.40 | HaMYaI | coppice: ok, I better change to 1.2.x and test it |
| 12:03.18 | lilalinux | JT: NT-Mode is not the problem, I had this working some months ago with 1 card, the main problem seems to be TE mode (respectively 2 hfc cards in one machine) |
| 12:03.39 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 12:03.52 | ifnotwhynot | hi there |
| 12:03.53 | lilalinux | I'm using vzaphfc btw. |
| 12:06.51 | *** join/#asterisk Op3r (n=op3r@125.212.125.250) |
| 12:07.04 | ifnotwhynot | can someone help me my boss is killing me i got asterisk to work with voicemail ivr and autoattendant thanks to mark and the asterisk channel now they want me to get a call centre working, If some one can help me I need so know how do i tie in a snom phone with a web page where the user enters information into a webpage whiles talking to a client, the call must be recorded + i need to save the information on the wepage into a database any |
| 12:07.27 | JT | lilalinux: you using the right mode? ptp/ptmp? |
| 12:07.31 | Op3r | ifnotwhynot, sugarcrm with asterisk works. |
| 12:07.43 | JT | lilalinux: and how's the zap timing? |
| 12:07.50 | ifnotwhynot | thx Op3r will have a look |
| 12:08.41 | Op3r | ifnotwhynot, check out trixbox too, it fits the bill |
| 12:09.19 | ifnotwhynot | i kinda know my way around asterisk source do i need to learn trixbox aswell? |
| 12:10.20 | JT | ifnotwhynot: if you use trixbox, you'll get very little support here |
| 12:10.28 | Op3r | thats one thing though |
| 12:10.39 | Op3r | but anyway |
| 12:10.42 | JT | a good thing |
| 12:10.46 | JT | trixbox is a nasty mess |
| 12:14.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:17.05 | *** join/#asterisk viraptor (n=viraptor@87.194.164.154) |
| 12:17.12 | viraptor | hi |
| 12:17.28 | viraptor | anyone can help me with IAX configuration? |
| 12:18.49 | *** join/#asterisk jm|laptop (n=jm|home@sentry.flags.co.uk) |
| 12:20.58 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
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| 12:22.17 | *** join/#asterisk dijungal (n=kdaniel@72-255-39-116.client.stsn.net) |
| 12:23.25 | dijungal | hello... how can i trasfer incoming ZAP calls to a SIP channel? |
| 12:24.30 | viraptor | CALL in extensions.conf probably, but i'm not sure |
| 12:24.51 | dijungal | hmmm |
| 12:24.54 | viraptor | Dial - sorry :) |
| 12:26.02 | dijungal | thought so |
| 12:26.29 | viraptor | exten => whatever,x,Dial(SIP...... |
| 12:26.45 | viraptor | just filter out that ZAP traffic before :) |
| 12:27.34 | dijungal | what do u mean filter out the ZAP traffic? |
| 12:28.08 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:29.01 | viraptor | make sure only those calls you want go to that extension |
| 12:32.43 | dijungal | hmmm |
| 12:34.19 | *** join/#asterisk linex (n=Krooks@202.184.116.210) |
| 12:34.21 | linex | hello |
| 12:34.45 | linex | Whats the lastest version of the book The Future of Telephony ? |
| 12:35.29 | dijungal | ll |
| 12:35.31 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:35.32 | dijungal | lol |
| 12:35.36 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:35.54 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:35.55 | dijungal | hey anyone in here in that Alabama Digium Asterisk training starting today? |
| 12:36.09 | linex | !asterisk |
| 12:36.30 | linex | Where can I download the book. They keep pasting it in here |
| 12:37.38 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:38.44 | [TK]D-Fender | ~book |
| 12:38.46 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:38.53 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:38.59 | linex | thanks |
| 12:40.53 | linex | ~book |
| 12:40.53 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
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| 12:51.30 | dominic1 | any misdn users here? |
| 12:52.37 | tzafrir_laptop | dominic1, I'm not. But ask your question anyway |
| 12:53.13 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:53.34 | dominic1 | I am looking for a solution of my little problem: chanisavail(misdn/g:TEPorts) always gives me the reply misdn/0 is my avail channel |
| 12:54.03 | dominic1 | but misdn/0 can never exist, cause the first port in misdn is always misdn/1 |
| 12:56.55 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
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| 13:00.12 | ifnotwhynot | where can i find someone to help me with setting up a callcentre using vicidial anyone please |
| 13:00.58 | Polis_ttt | ifnotwhynot: try #vicidial or take a look in the forum |
| 13:01.17 | [TK]D-Fender | ifnotwhynot: lol.... doubt too many people will do all that for free, check out the consultants list on the WIKI.... |
| 13:01.25 | ifnotwhynot | thx Polis_ttt |
| 13:01.30 | [TK]D-Fender | And this is not the vicidial support channel.... |
| 13:01.55 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 13:02.14 | Dr-Linux | i'm very fedup by seeing always loop for: |
| 13:02.19 | Dr-Linux | <PROTECTED> |
| 13:02.20 | Dr-Linux | <PROTECTED> |
| 13:02.20 | Dr-Linux | <PROTECTED> |
| 13:02.23 | Dr-Linux | on consol |
| 13:02.29 | Dr-Linux | how can i stop this? |
| 13:02.31 | Polis_ttt | [TK]D-Fender: i think he knows that, but vicidial uses asterisk, so that's why he started to search here :) |
| 13:02.56 | [TK]D-Fender | Dr-Linux: Find out whats connecting and STOP IT. |
| 13:04.03 | [TK]D-Fender | Dr-Linux: thats AMI firing off, and if you want it suppressed without canning the process, get CODING. |
| 13:05.05 | Dr-Linux | [TK]D-Fender: i commented out the manager configuration but still getting |
| 13:05.32 | [TK]D-Fender | Dr-Linux: Could be another process logging in at a shell prompt. |
| 13:05.50 | [TK]D-Fender | "asterisk -rx somethingoranother" |
| 13:06.00 | [TK]D-Fender | Dr-Linux: you should know all of this already... |
| 13:06.06 | Dr-Linux | lemme see |
| 13:07.33 | *** join/#asterisk zpertee (n=root@oh-69-34-21-229.sta.embarqhsd.net) |
| 13:09.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.27 | k31th | anyone know of a decent billing systeh, i just want to be able to import CDR CSV file into it and work out whos made what calls... |
| 13:11.33 | k31th | and the prices. |
| 13:12.25 | tzafrir_laptop | k31th, your spreadsheet? |
| 13:12.34 | [TK]D-Fender | k31th: these is a list on the WIKI, or go write one yourself. virtually nobody who does billing uses CSV..... |
| 13:12.50 | k31th | so how would it be done then. |
| 13:12.56 | k31th | wats the best way. |
| 13:13.03 | tzafrir_laptop | anybody wrote some spreadsheet macros to do this? |
| 13:13.31 | [TK]D-Fender | k31th: DB driven clearly. |
| 13:14.01 | [TK]D-Fender | k31th: Go to the WIKI and download everything you can get your hands on for this and just TRY THEM. |
| 13:14.30 | k31th | ok |
| 13:14.32 | k31th | will do |
| 13:17.23 | *** join/#asterisk Strom_M (n=strom@12.175.45.201) |
| 13:17.29 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 13:18.25 | pnlarsson | Whats wrong with the asterisk-user mailing list? last msg from 4 july. |
| 13:19.01 | one-of-the-idiot | Hello. "zaptel Disabled echo canceller because of tone (tx) on channel 1" in dmesg/syslog. What could cause this? |
| 13:19.03 | pnlarsson | dev and commit are doing ok |
| 13:23.57 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:23.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:27.13 | [TK]D-Fender | one-of-the-idiot: It received a fax or modem tone. |
| 13:28.01 | JT | or thinks it did ;) |
| 13:28.09 | lilalinux | JT: maybe not, I'll try different signallings tonight when I'm alone |
| 13:28.38 | lilalinux | JT: what do you mean by zap timing? I'll paste my configs somewhere ... |
| 13:29.22 | JT | ask the telco if it is p2p or p2mp |
| 13:29.30 | JT | lilalinux: run zttest for a while, look for what it normally scores, and what the lowest score is |
| 13:29.34 | JT | ignore the totals |
| 13:29.46 | JT | just read the numbers as they appear |
| 13:30.28 | lilalinux | k |
| 13:32.46 | lilalinux | http://pastebin.ca/610413 |
| 13:33.35 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:33.47 | one-of-the-idiot | [TK]D-Fender: Aah, yes. The events in dmesg match incoming/outgoinf faxes in CLI history. Thanks. |
| 13:34.17 | JT | lilalinux: umm yeah that's not really zttest results |
| 13:34.45 | lilalinux | JT: :) of course not, I can do this only tonight |
| 13:35.16 | lilalinux | lilalinux: staff is killing me if I break the phone again during the day |
| 13:35.26 | JT | zttest can run at the same time |
| 13:35.29 | JT | in fact it should. |
| 13:35.42 | lilalinux | As soon as I load the driver, the line is broken |
| 13:35.56 | lilalinux | or can this be done without? |
| 13:36.41 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:36.58 | lilalinux | Does anybody know, if "Arcor" (a german telco) uses p2p or p2mp? |
| 13:38.44 | JT | lilalinux: err what, are you using bristuff or not? |
| 13:38.55 | JT | it's business hours, ring them already |
| 13:40.00 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:40.23 | lilalinux | JT: yes, bristuff, and you won't get someone with technical skills at the phone (I live in germany :-/) |
| 13:40.50 | JT | lilalinux: i don't usually speak to a tech either, but the telco system says |
| 13:41.08 | JT | lilalinux: then why can you not run zttest if bristuff is running? |
| 13:41.33 | lilalinux | oh "running at the moment", no it's not I thought you meant in general bristuff or classic |
| 13:41.53 | JT | ok i have no idea what you're doing |
| 13:41.59 | JT | good luck :)PP |
| 13:42.20 | *** join/#asterisk Avero (n=Avero@216.186.253.120) |
| 13:42.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:42.59 | Avero | Does anyone know what "PROGRESS with cause code 127 received" received off a PRI means? |
| 13:43.19 | Mercestes | Google knows |
| 13:44.24 | Avero | I Googled and couldn't find an explanation, unless I missed it... |
| 13:44.39 | JT | "isdn cause code 127" |
| 13:44.57 | Mercestes | http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 13:45.29 | Avero | Thanks! |
| 13:45.35 | Mercestes | http://www.cisco.com/en/US/docs/ios/11_3/debug/command/reference/disdn.html |
| 13:45.44 | Mercestes | google: pri cause codes |
| 13:45.58 | JT | you can find all the cause code lists by googling for isdn cause code or q.931 cause code |
| 13:46.29 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 13:46.32 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-47-67.pskn.east.verizon.net) |
| 13:47.46 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:48.05 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 13:49.03 | *** join/#asterisk oej (n=olle@static-195.84.115.62.addr.tdcsong.se) |
| 13:51.28 | *** join/#asterisk mtoups (n=mtoups@HENSON.ISR.CS.CMU.EDU) |
| 13:53.11 | lilalinux | JT: ok it's ptmp |
| 13:54.32 | lilalinux | Yesterday I got a cause code, that wasn't listed, it was 26 iirc |
| 13:56.20 | mtoups | hi, so, I upgraded my asterisk 1.4.4 to 1.4.5 and since then, asterisk frequently takes up 100% cpu |
| 13:56.24 | lilalinux | hm, maybe it was a dec->hex problem |
| 13:56.42 | mtoups | if i restart it, the cpu load goes down for a while but eventually comes back up to 100% even if not in use |
| 13:57.03 | *** join/#asterisk censor (n=you@p54BE5802.dip.t-dialin.net) |
| 13:57.18 | censor | hi all |
| 13:57.48 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 13:58.01 | [TK]D-Fender | mtoups: Then upgrade to 1.4.6 ans stop whining about 1.4.5 ;) |
| 13:58.49 | mtoups | [TK]D-Fender: oh, heh, i didn't notice it was released |
| 13:59.02 | [TK]D-Fender | mtoups: Read the BIG PRINT.... |
| 13:59.42 | *** join/#asterisk syco (n=mike@176.163-243-81.adsl-dyn.isp.belgacom.be) |
| 14:00.27 | mtoups | [TK]D-Fender: ok, got it |
| 14:01.35 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:01.35 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:02.05 | *** join/#asterisk zuesman (n=pb0036@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 14:03.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:06.14 | irule | http://pastebin.ca/610468 on debian etch, 1.4 was working ok, I compiled 1.2.19 and then trying to run it, it wont, with this error message, any ideas? thanks |
| 14:06.19 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 14:07.02 | Juggie | irule, rm -rf /usr/lib/asterisk/modules |
| 14:07.13 | Juggie | then make install again on your 1.2.19 source directory |
| 14:07.54 | irule | thanks |
| 14:08.16 | Juggie | asterisk 1.2 is trying to load 1.4 modules |
| 14:08.19 | Juggie | kaboom :) |
| 14:08.53 | [TK]D-Fender | irule: AKA wipe our your modules forlder before switching versions.... |
| 14:08.58 | dominic1 | following problem: My sip hardphone isn't able to generate conferencecalls for more than 3 persons, so I need a solution with app_meetme and dynamic conferences. The procedure looks like the following. I call all the other parties and put them on hold, after calling the last party, I press a keycombination and asterisk will generate a conference and put all my connected channels to the conference. I somebody is calling me while I am in the conference, I ca |
| 14:09.04 | dominic1 | will that be possible? |
| 14:09.34 | irule | there is no 'make uninstall' :s |
| 14:09.56 | Juggie | dominic1, call each person one at a time and transfer them into a conference |
| 14:10.04 | Juggie | setup the conference so it plays moh until the admin joins. |
| 14:10.51 | [TK]D-Fender | irule: No. Just go and wipe out that folders contents by HAND. |
| 14:11.07 | dominic1 | no, that's uncomfortable, I want to create quick conferences for 10 or more people |
| 14:11.35 | dominic1 | is there a command to check which parties are on hold an connected to my phone? |
| 14:12.00 | dominic1 | so I think it will be possible to transfer all people which are currently on hold to my phone |
| 14:13.27 | *** join/#asterisk ai-a[wrk] (n=joe@megan.healthnet.co.uk) |
| 14:14.07 | [TK]D-Fender | dominic1: No. |
| 14:14.26 | [TK]D-Fender | dominic1: your phone is not some super-asterisk-conferencing engine |
| 14:14.51 | dominic1 | why not? |
| 14:15.02 | dominic1 | my phone not, but asterisk perhaps... |
| 14:16.31 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 14:17.02 | [TK]D-Fender | dominic1: its YOUR job to transfer them to the conference. |
| 14:17.11 | [TK]D-Fender | dominic1: its a 1 at a time thing |
| 14:17.28 | irule | dominic1 I think you are just wording what happens in asterisk differently from what is possible, what is possible, is to send them all into a meetme() conference, I dont know if you can clearly see who they are with the flash operator panel, then then, instead of transfering them all to your line, you simply join that conference, and voila! you are all together in the conference! :) |
| 14:17.38 | [TK]D-Fender | dominic1: You receive call, you transfer it to the conference. |
| 14:18.01 | Juggie | i'm not sure if the # of people in a conference on a set is a sip restriction or a restriction of the set |
| 14:19.02 | *** join/#asterisk Cardoe (n=cardoe@gentoo/developer/Cardoe) |
| 14:19.46 | pj_ | I'm using the Queue application and I noticed that when agents are already on the line, asterisk still "try" in rrobin algorithm... Then the agent can take it on its second line, however I'd rather have it call the agents that are _really_ free. any clue ? |
| 14:19.54 | Cardoe | Looking to disconnect my incoming Bellsouth from the wall and plug a PAP2 or SPA-2102 into the wall and use it in all the jacks in my place? Anyone have any experience doing so? |
| 14:20.45 | *** join/#asterisk perf3kt (i=perf3kt@149.166.33.199) |
| 14:22.14 | [TK]D-Fender | Cardoe: yup, thats what places like Vonage have you do |
| 14:22.52 | dominic1 | how will the transfer look like? |
| 14:22.55 | Cardoe | [TK]D-Fender: well guess I'm not the first person with the idea then... Glad I'm not crazy. :-D |
| 14:23.02 | dominic1 | Just the transfer command in the features.conf? |
| 14:23.06 | Cardoe | [TK]D-Fender: know if a SPA-2100 or a PAP2 would be a better approach? |
| 14:23.22 | Juggie | dominic1, the transfer function on your sip set. |
| 14:23.24 | [TK]D-Fender | dominic1: You don't know how to even transfer a call with your phone?! |
| 14:23.51 | dominic1 | with my phone of course |
| 14:23.52 | uwe | does any one know what "acl.c: 255.255.255.0,0.0.0.0/0.0.0.0 is not a valid netmask" means ? and why i might get it in the logs ? |
| 14:23.56 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 14:23.56 | [TK]D-Fender | Cardoe: Any will do, but if you haven't bought one yey I'd suggest the SPA-2102 |
| 14:25.05 | *** part/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net) |
| 14:25.41 | irule | isnt there an ata with a single ethernet cable and 20 phone jacks or something? |
| 14:25.57 | *** join/#asterisk dijungal (i=kdaniel@nat/digium/x-bc6aa5996e5daefd) |
| 14:26.14 | [TK]D-Fender | irule: thats a mass-port SIP gateway |
| 14:26.24 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 14:26.24 | irule | oohhh :) |
| 14:26.34 | dominic1 | but I want a one touch conference for every user and with my phone I need three steps for every user, I can not imagine, that it should not be possible to press a key and the user will get transferred to a new generated or existing conferenceroom |
| 14:26.58 | Cardoe | [TK]D-Fender: great. thanks. I'm shopping around for prices on both. I'll aim towards the SPA-2102 then |
| 14:27.01 | [TK]D-Fender | dominic1: There IS not "miracle" button. Get over it. |
| 14:27.11 | dominic1 | okay |
| 14:27.16 | dominic1 | I understand, thank you |
| 14:27.17 | dominic1 | :-) |
| 14:27.27 | [TK]D-Fender | Cardoe: 2102 = current and offers router functioanlity Even if you don't need it NOW it can pay off later |
| 14:27.38 | pifiu | guys im selling a WIP300 if anyone cares |
| 14:27.39 | pifiu | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&ih=017&sspagename=STRK%3AMESE%3AIT&viewitem=&item=270141927708&rd=1&rd=1 |
| 14:28.57 | irule | dominic1 actually, you can just sit back and start getting creative, you have variables and stuff to play with, if there is no miracle button, you may create your special miracle button :) |
| 14:29.22 | dominic1 | interesting: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro |
| 14:29.56 | perf3kt | ho functional is a ip based cell phoen if you don't live in a wifi city |
| 14:29.58 | irule | dominic1 there you go, now try to port it to a current version, I failed lol |
| 14:30.22 | dominic1 | okay... thank you |
| 14:30.33 | Cardoe | [TK]D-Fender: cool. thanks for the advice. |
| 14:31.04 | [TK]D-Fender | Cardoe: better on resale and longer supported. Not like it costs more or anything. (PAP2 = low end, avoid) |
| 14:31.43 | Op3r | is it advisable to put munin or any other server monitoring tools on an Asterisk production servers? |
| 14:33.54 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-65848532d7643b08) |
| 14:36.03 | *** join/#asterisk ron_gage (n=ron@216.234.109.90) |
| 14:36.37 | De_Mon | I've got a dynamic queue that rings several phones. Sometimes with I pickup one queue member answers, the others members keep rining... |
| 14:37.00 | De_Mon | s/with I pickup/when/ |
| 14:37.12 | De_Mon | arg /me stabs jbot in the eye with an asterisk |
| 14:37.22 | ron_gage | does anyone know if the TE1xx T1 card can be used to do protocol analysis of a T1 line? |
| 14:38.13 | De_Mon | I haven't been able to reproduce the problem on purpose, it happens ~1/20 pickups... |
| 14:38.39 | De_Mon | any ideas whats causing it and how to stop it from happening? |
| 14:39.05 | *** part/#asterisk dijungal (i=kdaniel@nat/digium/x-bc6aa5996e5daefd) |
| 14:42.20 | *** part/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 14:42.49 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 14:43.59 | ron_gage | De_Mon - what type of phones are these? |
| 14:44.07 | *** join/#asterisk kiwi`MouThon (n=Tomy@ADijon-256-1-102-29.w86-204.abo.wanadoo.fr) |
| 14:44.11 | kiwi`MouThon | http://saint-tomy.miniville.fr/ i don't ud nothing! |
| 14:44.53 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
| 14:44.55 | MrChicken | Hello |
| 14:45.07 | MrChicken | I am trying to build spandsp libraries for asterisk 1.4.5 |
| 14:45.13 | MrChicken | I have compiled spandsp correctly |
| 14:45.36 | MrChicken | asterisk compiles correctly as well |
| 14:45.56 | MrChicken | however no app_rxfax.so and app_txfax.so get generated |
| 14:46.04 | MrChicken | can anybody help me out? |
| 14:46.22 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:46.33 | ron_gage | MrChicken, did you add the spandsp stuff to the modules config? |
| 14:46.54 | MrChicken | yes |
| 14:46.57 | MrChicken | of course |
| 14:47.17 | ron_gage | just checking - when you start asterisk, does it show the module loading without errors? |
| 14:47.27 | ron_gage | asterisk -vvvc |
| 14:47.37 | MrChicken | no |
| 14:47.43 | MrChicken | the module does not get generated |
| 14:48.22 | MrChicken | (only the .c files on the apps/ dir once you "make" asterisk) |
| 14:48.34 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 14:48.40 | ron_gage | well, that covers the basics anyhow. Sorry I can't help further at this time. I have a "fax" system but I can't get to it from here |
| 14:48.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:49.35 | MrChicken | its strange |
| 14:49.53 | ron_gage | checking some things right now - hold a sec |
| 14:49.59 | MrChicken | ok |
| 14:51.22 | MrChicken | plz lemme know when ure back :) |
| 14:51.28 | ron_gage | back |
| 14:51.42 | MrChicken | its strange |
| 14:51.53 | *** join/#asterisk daguz (n=leo@208-1-63-34.celito.net) |
| 14:52.09 | MrChicken | does the fax thingy require openssl or something like that? |
| 14:52.11 | ron_gage | did you look at http://www.voip-info.org/wiki-Asterisk+fax#SpanDSPSendingandReceivingFaxeswithAster |
| 14:53.14 | MrChicken | actually I was looking at a howto in voipphreak |
| 14:54.12 | ron_gage | let me know if that helps |
| 14:55.44 | MrChicken | well that one was not much help... theres a link to asterisk guru |
| 14:58.02 | *** join/#asterisk eliter (n=jbartek@66.179.79.69) |
| 14:58.30 | *** join/#asterisk currach (n=currach@89.16.90.180) |
| 14:58.39 | ron_gage | Looks like voip-info has more on asterisk and spandsp... http://www.voip-info.org/wiki/view/Asterisk+spandsp |
| 14:58.51 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 15:03.36 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-82-36.w86-205.abo.wanadoo.fr) |
| 15:03.43 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 15:03.44 | DragoraN | hi |
| 15:03.45 | yannj_fr | hello everybody |
| 15:04.10 | DragoraN | is Linksys WIP330 able to perform VPN connection before trying to connect to his SIP server? |
| 15:05.21 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:07.45 | *** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net) |
| 15:11.22 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:12.23 | *** join/#asterisk toombaloomba (n=hola@216.14.11.254) |
| 15:12.38 | kombi | in a dialplan, how do you execute a command on hangup? |
| 15:12.49 | MrChicken | Hello |
| 15:13.10 | MrChicken | I am trying to compile fax support for asterisk (spandsp) |
| 15:13.23 | MrChicken | when I make menuselect |
| 15:13.29 | MrChicken | and then choose applications |
| 15:13.31 | lilalinux | JT: zttest doesn't output anything |
| 15:13.39 | Nugget | MrChicken: when you |
| 15:13.41 | Nugget | talk like |
| 15:13.42 | Nugget | this |
| 15:13.43 | Nugget | it sounds |
| 15:13.44 | Nugget | like |
| 15:13.47 | Nugget | you have |
| 15:13.49 | Nugget | asthsma |
| 15:14.25 | toombaloomba | hello, Q> i need to re-ip an asterisk server and users on it, how can I make asterisk reply using a different source IP when there are multiple IPs on the server so I can use both IPs while migrating? |
| 15:15.24 | MrChicken | (sorry... ) I find XXX in front of app_rxfax and app_txfax, what does this mean? Does it mean it cant find spandsp libraries or what? |
| 15:15.26 | [TK]D-Fender | Reply with a different IP, LOL.... seriously not happening.... |
| 15:15.35 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 15:15.37 | toombaloomba | lol crap :( |
| 15:15.47 | toombaloomba | coz im having REGISTER come in on one IP and then the server replies from another and naturally it doesnt work |
| 15:15.57 | Nugget | MrChicken: yes, you're missing a dependancy. menuselect should tell you what that application needs at the bottom |
| 15:16.02 | kombi | do we have "onHangup"? |
| 15:16.05 | MrChicken | SpanDSP |
| 15:16.11 | MrChicken | but I have spandsp installed (!!!) |
| 15:16.29 | Nugget | ./configure again and see if autoconf is successfully locating spandsp |
| 15:16.29 | [TK]D-Fender | jsut have it bind to both so it can use either |
| 15:16.49 | MrChicken | Nugget ... I think it isnt |
| 15:17.05 | [TK]D-Fender | ldconfig <----------- |
| 15:17.13 | MrChicken | I already did! |
| 15:18.06 | irule | http://pastebin.ca/610565 may someone please give me a simple hint on what is mpg123 looking for? |
| 15:18.08 | MrChicken | when I do ldconfig -v | grep spandsp I get libspandsp.so.0 -> libspandsp.so.0.0.2 |
| 15:18.27 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:19.05 | [TK]D-Fender | kombi: Go re-read the chapter on Asterisk Standard Extensions. |
| 15:20.15 | *** join/#asterisk version5 (i=version5@nat/ibm/x-376f9f58f2e538bb) |
| 15:20.20 | [TK]D-Fender | irule: Have you tried the NORMAL comile method? ./configure ; make clean ; make ; make install ? |
| 15:20.33 | irule | good point, thanks |
| 15:20.51 | irule | -su: ./configure: No such file or directory |
| 15:20.52 | version5 | hey guys, is it possible to set up a call (conference call perhaps) in such a way that the server will call both participants as opposed to one of the people calling the other? |
| 15:21.19 | irule | [TK]D-Fender README says to just 'make' and 'make linux' |
| 15:21.44 | irule | version5 .call file? |
| 15:22.26 | MrChicken | (sorry... ) I find XXX in front of app_rxfax and app_txfax. but I have spandsp installed (!!!) and ldconfig reports it in place |
| 15:22.36 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:22.57 | Mercestes | A bit OT: Can anyone tell me anything about the 3Com "power of zero" promtion they are running that's advertised *everywhere*? |
| 15:23.13 | kombi | Fender: hmm, sort of did, got MeetMe(stuff) followed by MeetMeAdmin(stuff|K) but it does no killing |
| 15:24.39 | *** join/#asterisk Redback (n=Redback@82.152.56.113) |
| 15:25.29 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:25.52 | [TK]D-Fender | kombi: Show me. |
| 15:26.41 | version5 | irule: for these call files do i just write the file, move it into the directory and asterisk will take it from there? |
| 15:26.54 | [TK]D-Fender | version5: Go to the WIKI and READ. |
| 15:27.07 | [TK]D-Fender | version5: this is also documented in THE BOOK |
| 15:27.08 | [TK]D-Fender | ~book |
| 15:27.09 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:27.10 | [TK]D-Fender | ~wikis |
| 15:27.11 | jbot | well, wikis is http://www.voip-info.org |
| 15:28.19 | irule | version5 yes, read the wiki, just a tip though, you may need to use AGI interfaces to use an external script that wil echo"call-file-contents.agi">file.call and asterisk will dial the moment the file is created :) |
| 15:28.39 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:28.54 | kombi | Fender: Never mind, it just occurs to me that the room is started one line above meetme by system(), so the caller has no rights to killing. |
| 15:30.15 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 15:30.57 | kombi | Fender: http://pastebin.ca/610587 |
| 15:31.39 | stimpie | Iam sending calls to an openser gateway, this is logged in the cdr as a sip call the the proxy. How can I log the IP where the media is going? |
| 15:32.30 | [TK]D-Fender | kombi: Yeah well if you hangup you can bet it won't get killed... |
| 15:32.40 | [TK]D-Fender | kombi: Go re-read the chapter on Asterisk Standard Extensions. <-------------- |
| 15:32.57 | kombi | Fender: hmm, http://pastebin.ca/610591 |
| 15:33.19 | [TK]D-Fender | kombi: Go re-read the chapter on Asterisk Standard Extensions. <------------------------------- |
| 15:33.50 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 15:34.01 | kombi | ok, ok.. you mean Dialplan basics? can't find it in there.. |
| 15:34.34 | [TK]D-Fender | kombi: You aren't looking very hard.... |
| 15:35.02 | waverly360 | [TK]D-Fender: c'mon now..be nice :) |
| 15:35.07 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 15:35.11 | kombi | I'll look harder now..page 79 "Extensions".. |
| 15:35.30 | joetester | Can a PRI be missing DTMFs? Is that in the realm of the possible? |
| 15:35.43 | joetester | From the telcos side? |
| 15:36.35 | waverly360 | joetester: What seems to be the problem? |
| 15:37.21 | [TK]D-Fender | kombi: Don't forget the WIKI... |
| 15:37.54 | joetester | waverly360: I seem to be having problems getting the dtmfs... happens as soon as I start playing back a sound... |
| 15:38.30 | MrChicken | Hi I'm trying to compile spandsp libraries for asterisk 1.4.5. I've patched all the files, downloaded the .c files and placed them correctly |
| 15:38.42 | waverly360 | joetester: so you get dtmf tones before you try to play a sound? |
| 15:38.48 | MrChicken | however when I try to ./configure ... I cant see any reference to spandsp |
| 15:38.50 | *** join/#asterisk heh_v_water (n=heh_v_wa@70-57-205-130.hlna.qwest.net) |
| 15:39.07 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-ec9b05e359b7a0bd) |
| 15:39.11 | kombi | Fender: there is something...;) |
| 15:40.27 | joetester | waverly360: The setup is as such: PRI -> Asterisk 1 -> IAX2 -> Asterisk 2. Asterisk 1 is just a "gateway" and "Asterisk 2" is a PBX. I am monitoring the packets that come out of Asterisk 1 into Asterisk 2 over IAX on an incoming call on the PRI |
| 15:41.25 | joetester | waverly360: If I press the keypad on the phone BEFORE actually starting to playback, the packets are sent (DTMF_B, DTMF_E for each DTMF) |
| 15:42.58 | joetester | waverly360: As soon as the playback begins, then the DTMF_B and DTMF_E packets are no longer sent by Asterisk 1, I have no idea what the hell is going on. |
| 15:43.48 | waverly360 | joetester: I've never done anything like that before..but sounds like the playback might be triggering asterisk 1 to stop sending somehow |
| 15:44.00 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:44.08 | kombi | Fender: what I don't get though: can you execute the next line before the previous one has finished? (like I would like to do with that System() after Meetme()) |
| 15:44.09 | joetester | waverly360: From where I stand, it looks like the playback is influencing the way Asterisk 1 is able to get the dtmfs |
| 15:44.10 | waverly360 | joetester: I really don't know though..there are too many variables to consider |
| 15:45.33 | joetester | waverly360: Maddening isn't it, it's been days and I have no idea what is going on. Can the guy on the telco side monitor what is going on on the PRI? Like can he check while I dial digits during the playback and see if he's getting them or not? |
| 15:45.37 | *** part/#asterisk version5 (i=version5@nat/ibm/x-376f9f58f2e538bb) |
| 15:45.42 | lilalinux | JT: could it be, that zttest doesn't work with vzaphfc? |
| 15:45.47 | waverly360 | joetester: Well, whenever I have dtmf problems, which I have a lot, I make sure to set relaxdtmf=yes in zapata.conf..whether that'll help you out, I have no idea. |
| 15:46.11 | joetester | waverly360: Does that apply to PRIs? |
| 15:46.11 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 15:46.14 | *** join/#asterisk joe (n=nnnnnnnn@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 15:46.49 | waverly360 | joetester: I seem to remember it making some kind of a difference for me. |
| 15:48.19 | *** join/#asterisk corpse2007 (n=thelords@202.79.50.239) |
| 15:48.30 | *** part/#asterisk syco (n=mike@176.163-243-81.adsl-dyn.isp.belgacom.be) |
| 15:48.44 | corpse2007 | what the hell is open source? |
| 15:48.56 | joetester | waverly360: So it does! It does seem to work a lot better! |
| 15:49.06 | waverly360 | joetester: Sweet! :) |
| 15:49.48 | *** part/#asterisk corpse2007 (n=thelords@202.79.50.239) |
| 15:50.04 | joetester | waverly360: If I could kiss you I would! |
| 15:50.22 | waverly360 | joetester: I'm not sure what the deal is with that setting, but I've always had to set it to yes to make dtmf reliable. |
| 15:50.36 | waverly360 | joetester: Hah. I'm good :P |
| 15:51.02 | joetester | waverly360: I don't understand it either... I thought that only worked on analog interfaces :S |
| 15:51.09 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 15:51.25 | *** part/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 15:51.32 | waverly360 | joetester: I seem to remember that's what they said on voip-info, but apparently it does work for PRIs as well. |
| 15:52.03 | waverly360 | joetester: At any rate, glad I could help. |
| 15:52.54 | joetester | waverly360: It does work because I couldn't even type dtmfs fast before, it missed them all. I also disabled echo cancellation in zapata.conf since I have a TE212P which seems to have it in hardware? |
| 15:53.36 | lilalinux | shoud tail -f /dev/zap/pseudo return anything? |
| 15:53.36 | waverly360 | joetester: no idea. I've never used that device before. |
| 15:53.38 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:53.51 | waverly360 | CunningPike: howdy stranger |
| 15:54.08 | CunningPike | Hey, waverly360 - long time |
| 15:54.33 | waverly360 | CunningPike: indeed |
| 15:55.30 | phearless | hey folks |
| 15:55.43 | phearless | how could I run Diaplan commands via the Asterisk Manager ? |
| 15:56.56 | colde | I'm trying to do a call via a .call file. It calls the desired phone alright, but as soon as the phone is answered it hangs up on the call, any idea why? |
| 15:57.48 | [TK]D-Fender | colde: Show us the call file & your dialpln and MAYBE.... |
| 15:57.50 | [TK]D-Fender | ~pb |
| 15:57.51 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 15:57.59 | [TK]D-Fender | colde: We're not PSYCHIC you know... |
| 15:58.03 | waverly360 | [TK]D-Fender: lmao |
| 15:58.06 | [TK]D-Fender | colde: and PASTEBIN it <- |
| 15:58.19 | waverly360 | [TK]D-Fender: our resident pastebin nazi :P |
| 15:58.42 | [TK]D-Fender | waverly360: I'm almost a RTFM nazi too... work in progress :D |
| 15:58.56 | [TK]D-Fender | waverly360: but I've been blogging & JBOT training lately. |
| 15:59.06 | waverly360 | [TK]D-Fender: I should've guessed as much. |
| 15:59.51 | waverly360 | speaking of RTFM, does anyone know where I can get documentation on the AudioCodes MP-114 device? They don't have it publicly available on their website. |
| 16:00.15 | lilalinux | colde: here you can find a working example: http://www.lilalinux.net/e-trolley/page_8690/index.html |
| 16:00.29 | lilalinux | JT: after a reboot I get these results of zttest: Best: 100.000000 -- Worst: 99.987793 -- Average: 99.991226 |
| 16:00.32 | colde | [TK]D-Fender: good point ;) |
| 16:00.58 | *** join/#asterisk sopo2k4 (n=jam@host81-152-232-54.range81-152.btcentralplus.com) |
| 16:01.03 | colde | hmm, it seems like it tries to connect the call to the "default" context |
| 16:01.05 | sopo2k4 | hi, anyone able to help? |
| 16:01.14 | colde | Eventhough i specify context: incomming in the call file |
| 16:01.57 | lilalinux | colde: maybe the wrong extensions? |
| 16:02.01 | lilalinux | or priority? |
| 16:02.01 | sopo2k4 | ive got my pbx setup, works and all that for USA numbers' however im trying to get it to dial Internationally and i get the following error: No such context/extension. |
| 16:02.23 | sopo2k4 | any idea's? |
| 16:02.32 | colde | lilalinux: no, that works, it does try extensions s with priority 1, however, the context is wrong |
| 16:02.49 | colde | in cli it says it tries to dial s,1 in default context |
| 16:03.00 | colde | hah, wrong spelling |
| 16:03.03 | colde | fixed it :d |
| 16:03.27 | phearless | how could I run Diaplan commands via the Asterisk Manager ? |
| 16:03.27 | sopo2k4 | anyone? |
| 16:03.47 | CunningPike | waverly360: Specifically that model, or all models -we got the docs for ours on their site |
| 16:05.23 | sopo2k4 | *CLI> Jul 9 07:57:48 NOTICE[4547]: chan_iax2.c:5791 update_registry: Restricting registration for peer 'wu' to 60 seconds (requested 1200) |
| 16:05.23 | sopo2k4 | Jul 9 07:58:27 NOTICE[4547]: chan_iax2.c:5791 update_registry: Restricting registration for peer 'wu' to 60 seconds (requested 1200) |
| 16:05.23 | sopo2k4 | Jul 9 07:58:37 NOTICE[4547]: chan_iax2.c:7346 socket_read: Rejected connect attempt from 81.152.232.54, request '011447734533888@outgoing' does not exist |
| 16:05.28 | sopo2k4 | anyone able to help me fix this error? |
| 16:05.42 | Qwell[] | it isn't an error |
| 16:05.45 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:06.04 | sopo2k4 | .... any more information would be greatly appreciated... |
| 16:06.40 | Qwell[] | oh, didn't see that last line. It means you're dialing something that doesn't exist. |
| 16:06.56 | Juggie | Qwell, are you planning on commiting that moh patch? |
| 16:07.03 | Qwell[] | if somebody tells me it works :p |
| 16:07.05 | ai-a[wrk] | HELP - Asterisk has lost its beautiful color in my terminal window. whats happened? hard to follow calls in white on black only display. |
| 16:07.16 | Juggie | Qwell, oh yah, i'll get blitzrage on it. |
| 16:07.21 | sopo2k4 | it works if i call to usa, but to any other country i get that..... |
| 16:07.39 | Qwell[] | sopo2k4: Do you have anything in your dialplan to handle 011? |
| 16:08.23 | sopo2k4 | exten => _X.,2,Dial(IAX2/x is part of the dial plan |
| 16:08.55 | lilalinux | JT: with ptmp (and a reboot!) everything is working. thx for your patience |
| 16:08.56 | sopo2k4 | <Qwell[]> sopo2k4: Do you have anything in your dialplan to handle 011? |
| 16:08.56 | sopo2k4 | * zeeesh has quit IRC |
| 16:08.56 | sopo2k4 | <sopo2k4> exten => _X.,2,Dial(IAX2/x is part of the dial plan |
| 16:09.12 | sopo2k4 | sorry, didnt mean to paste that. |
| 16:09.22 | sopo2k4 | have i gotta add the 011 to the _x? |
| 16:09.32 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:10.47 | tzafrir_laptop | ai-a, you're not running it from the console. There's a nice little patch that gives asterisk back its colours even then |
| 16:11.05 | waverly360 | CunningPike: Well, all models would be fine. I went to their site, and they require me to register for them. I registered, but they haven't gotten back with me yet. |
| 16:11.08 | ai-a[wrk] | used to have color about 30 minutes ago. |
| 16:11.11 | sopo2k4 | could you paste me an example of what i would be adding to my dialplan to handle 011? |
| 16:11.17 | ai-a[wrk] | just restarted asterisk.. nothing special.. |
| 16:12.22 | tzafrir_laptop | ai-a, if not, then from what terminal are you running? |
| 16:12.32 | phearless | guys, how could I run Diaplan commands via the Asterisk Manager ? |
| 16:12.45 | ai-a[wrk] | tzafrir: its putty.. ls shows color.. but asterisk has none now. |
| 16:12.47 | phearless | i am sure that [TK]D-Fender knows the answer :) |
| 16:13.04 | tzafrir_laptop | ai-a, echo $TERM |
| 16:13.11 | ai-a[wrk] | xterm |
| 16:15.26 | CunningPike | waverly360: http://www.audiocodes.com/asp/DisplayFoldersFiles2.asp?FolderID=6 |
| 16:16.15 | waverly360 | CunningPike: Was that on their website and I just missed it? |
| 16:17.02 | tzafrir_laptop | ai-a, ps auxww | grep asterisk # and now tell us what are the command-line parameters of asterisk |
| 16:18.26 | ai-a[wrk] | no parameters on the service one. |
| 16:18.30 | ai-a[wrk] | root 6375 0.0 1.7 274032 18468 ? Ssl 16:48 0:01 asteriskroot 7352 0.0 0.9 44312 9272 pts/0 S+ 17:03 0:00 rasterisk rvvvvvvvvvvvv |
| 16:18.46 | ai-a[wrk] | i could reboot the service ;) |
| 16:18.55 | CunningPike | waverly360: I think so........ Support >> Public Documentation Downloads >> MediaPack Series |
| 16:19.13 | ai-a[wrk] | tzafrir: service restart fixed it :) |
| 16:19.32 | ai-a[wrk] | has /usr/sbin/asterisk -f -vvvg -c now |
| 16:19.51 | ai-a[wrk] | dodgy sangoma asterisk restart |
| 16:19.52 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 16:21.09 | waverly360 | CunningPike: man I'm a n00b. Thanks :) |
| 16:21.24 | sopo2k4 | damn this, cant get it to dial internationally :@ |
| 16:21.37 | CunningPike | waverly360: No problem. I have the same problem with the margarine in the fridge |
| 16:21.47 | Mercestes | waverly360, Don't feel bad. |
| 16:21.49 | Mercestes | ~mercestes |
| 16:21.50 | jbot | mercestes is definitely a total nub |
| 16:21.50 | waverly360 | CunningPike: lmao |
| 16:22.01 | waverly360 | Mercestes: hah hah. |
| 16:22.16 | waverly360 | Mercestes: but you're a nub..is that worse than a n00b? |
| 16:22.21 | [TK]D-Fender | phearless: mostly.......like? |
| 16:22.30 | Mercestes | Probably worse. |
| 16:22.37 | Mercestes | and I'm a total nub, instead of just mostly a n00b. |
| 16:22.47 | Mercestes | definitely. |
| 16:22.50 | phearless | [TK]D-Fender: what do you mean with "mostly like?" ? |
| 16:22.57 | waverly360 | Are any of you guys very familiar with AGI? |
| 16:23.09 | Mercestes | phpagi |
| 16:23.34 | [TK]D-Fender | phearless: Be SPECIFIC about what you want to do in case your idea or approach is entirely inappropriate. |
| 16:23.56 | phearless | I want to use ChannelRedirect in a ruby script |
| 16:24.14 | phearless | the ruby script uses "asterisk manager" |
| 16:24.23 | sopo2k4 | has anyone got a fully working asterisk conf for voicepulse working with international dialing? |
| 16:24.29 | sopo2k4 | their able to let me use? |
| 16:24.47 | sopo2k4 | obviously with my own l/p.... |
| 16:25.21 | waverly360 | Well, here's the problem I'm having. If I call the Dial command from asterisk, and simply use extensions.conf to dial a number that's busy, I actually get a busy signal. |
| 16:25.41 | waverly360 | If I try to dial the same number, using the same dialing options from my agi script, I just get dead silence. |
| 16:25.42 | [TK]D-Fender | phearless: There is an AMI function to do that already... |
| 16:26.06 | phearless | [TK]D-Fender : which one ? |
| 16:26.43 | waverly360 | What's so different about using the dial command from AGI than straight from extensions.conf? |
| 16:26.48 | irule | is there anything to know when I compile asterisk on a pc without any zaptel hardwarem, and then the hardware is added? must I recompile asterisk by then? or can I prepare it so that it will work without recompiling? |
| 16:26.54 | phearless | [TK]D-Fender : not "Redirect" |
| 16:27.16 | waverly360 | irule: you shouldn't have to recompile asterisk |
| 16:27.38 | [TK]D-Fender | phearless: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect |
| 16:27.45 | [TK]D-Fender | phearless: Why not? |
| 16:27.49 | irule | is ztdummy no longer required by 1.2.20? |
| 16:28.14 | phearless | [TK]D-Fender : I tried to use Redirect and I can't make it work |
| 16:28.27 | phearless | [TK]D-Fender : there are very very few example with Redirect |
| 16:28.28 | [TK]D-Fender | irule: * doesn't require ztdummy, never did. |
| 16:28.41 | phearless | [TK]D-Fender : that's why i want to try channelredirect |
| 16:28.47 | [TK]D-Fender | phearless: YOUR failure is another matter. lets try to fix the PROPER way to do it... |
| 16:28.53 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 16:28.55 | casix | hello |
| 16:28.57 | waverly360 | lol |
| 16:28.59 | [TK]D-Fender | phearless: the other way is damn messy |
| 16:29.10 | casix | I've had a asterisk crash |
| 16:29.17 | phearless | [TK]D-Fender : what is the other way ? |
| 16:29.31 | casix | i'm debugging de core but i don't know what I have to put to the bug info |
| 16:29.32 | tzafrir_laptop | irule, zaptel hardware is not required for asterisk building |
| 16:29.41 | *** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it) |
| 16:29.44 | [TK]D-Fender | phearless: Trying to use the dialplan app like you were asking. That is BAD. Go back to the pure AMI way and we'll find out where you go wrong |
| 16:29.51 | [TK]D-Fender | phearless: because it WORKS. |
| 16:30.07 | casix | because there are 24 different threads |
| 16:30.26 | phearless | ok [TK]D-Fender :) |
| 16:30.35 | casix | I think that the bad threat is one that have this: #3 0x00002aaab0d0ff93 in acf_odbc_read (chan=0xa19480, cmd=<value optimized out>, s=0x1 <Address 0x1 out of bounds>, buf=0x405953c0 "", len=4096) at func_odbc.c:252 |
| 16:30.39 | phearless | [TK]D-Fender: i will try to debug my probleme... |
| 16:30.40 | casix | it is possible? |
| 16:31.16 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:32.01 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-d18cffdb85017939) |
| 16:32.11 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 16:32.11 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:36.38 | sopo2k4 | could someone paste me an extension for voicepulse to dial internationally, number format, 011 + country code + number |
| 16:36.42 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 16:40.38 | Cardoe | sopo2k4: _011.,1,Dial(whatever) |
| 16:41.59 | irule | how can I ignore i on an empty [error-message-contecxt]? I just want to playfile(error-message) and make sure it is played with no interruption and then move on :) |
| 16:43.19 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 16:43.28 | *** join/#asterisk krdian_ (i=krdian@killer.radom.net) |
| 16:43.49 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:44.28 | sopo2k4 | Cardoe |
| 16:44.32 | sopo2k4 | Jul 9 08:36:10 NOTICE[14093]: chan_iax2.c:7346 socket_read: Rejected connect attempt from 81.152.232.54, request '011441189481789@outgoing' does not exist |
| 16:44.38 | sopo2k4 | any idea's? |
| 16:45.39 | [TK]D-Fender | sopo2k4: Check your dialplan, its telling you to your face exactly what is missing. |
| 16:45.42 | file | you do not have dialplan logic present in the outgoing context to allow that extension to be dialed |
| 16:46.45 | sopo2k4 | to a super noob like me that doesnt really help much :s |
| 16:46.59 | sopo2k4 | ill keep playing with it tho, eventually bound to get there :| |
| 16:47.12 | file | exten => _011X.,1,Dial(blah) |
| 16:47.16 | toombaloomba | anyone know if its possible to change the TFTP IP configured on a cisco 79x0 phone remotely? either via the config file itself or via telnet? |
| 16:47.16 | Nugget | telnet is eeeeeeevil! |
| 16:47.39 | toombaloomba | lol |
| 16:47.46 | sopo2k4 | ok rite |
| 16:47.49 | sopo2k4 | can anyone see the problem |
| 16:47.50 | sopo2k4 | [outgoing] |
| 16:47.50 | sopo2k4 | exten => _1NXXNXXXXXX,1,setcallerid(x1) |
| 16:47.50 | sopo2k4 | exten => _011.,2,Dial(IAX2/hkxQI64:vSnxdad11@connect02.voicepulse.com/${EXTEN}) |
| 16:47.50 | sopo2k4 | ;exten => _X.,2,Dial,IAX2/bsxy@NuFone/${EXTEN} |
| 16:47.50 | sopo2k4 | exten => _1NXXNXXXXXX,3,Congestion() |
| 16:47.52 | sopo2k4 | exten => _1NXXNXXXXXX,103,Busy() |
| 16:48.58 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 16:48.59 | irule | sopo2k4 so the password is vSnxdad11? |
| 16:49.29 | sopo2k4 | if that what floats your boat... |
| 16:49.38 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:49.44 | waverly360 | Ok..here's the CLI output of my AGI dialing problem. http://pastebin.ca/610742 |
| 16:49.57 | file | sopo2k4: you do not have a priority 1 for that extension... so it'll never work |
| 16:50.09 | waverly360 | Mercestes: you're familiar with agi some right? Could you look at this and see what the problem is? |
| 16:50.58 | [TK]D-Fender | sopo2k4: You have no priority #1 in there and next time, don't paste your PASSWORDS |
| 16:51.13 | sopo2k4 | forgot... sorry. |
| 16:51.14 | sopo2k4 | lol |
| 16:51.41 | file | there is a priority 1, but that only gets matched for numbers following the standard 1<blah> match... not for stuff starting with 011 |
| 16:51.41 | sopo2k4 | been trying to get this working for 3 days str8 now.... |
| 16:51.51 | waverly360 | How about this, has anyone here written an agi script that handles their entire dialplan? |
| 16:52.04 | NOT_guru | toombaloomba : isn't your tftp setting getting set through DHCP options? |
| 16:52.17 | NOT_guru | err thats how I do mine is all |
| 16:52.19 | [TK]D-Fender | Op3r: Funny like the rest of his stuff? |
| 16:53.28 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 16:53.45 | Op3r | [TK]D-Fender, just this one cos It got me a girlfriend |
| 16:54.00 | [TK]D-Fender | Op3r: ooooh |
| 16:54.15 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 16:54.21 | [TK]D-Fender | Op3r: "Celebrity" and "I'm Gonna Miss Her" <- Hilarious |
| 16:54.37 | Op3r | [TK]D-Fender, but yeah, I like the his music, its just that this is the one I listen frequently and also that one about trading his wife for fishing |
| 16:54.51 | Op3r | oh yeah thats it Im gonna miss her |
| 16:54.58 | [TK]D-Fender | :D |
| 16:55.00 | [TK]D-Fender | yup |
| 16:55.41 | Op3r | I dont normally listen to country music but due to this Im starting to like it :D |
| 16:56.29 | Taadow | We are a company of aproximately 30 staff looking for a voip provider reputable (preferably w/ switches at or near Vancouver, BC, Canada). We're looking for a provider offering top notch voice quality, not really concerned w/ price. Don't spose anyone can offer any suggestions? Avg about 40,000 minutes monthly. |
| 16:56.42 | [TK]D-Fender | Op3r: Ditto.... jsut because he's so damn funny... go watch "Celebrity", you'll LOVE it... |
| 16:56.51 | sopo2k4 | ok, this works for regular calls, to make it work for 011 + 44 + 1189481789 - i change what? |
| 16:56.51 | sopo2k4 | [outgoing] |
| 16:56.52 | sopo2k4 | exten => _1NXXNXXXXXX,1,setcallerid(2032855911) |
| 16:56.52 | sopo2k4 | exten => _X.,2,Dial(IAX2/user:pass@connect02.voicepulse.com/${EXTEN}) |
| 16:56.52 | sopo2k4 | ;exten => _X.,2,Dial,IAX2/xhy@NuFone/${EXTEN} |
| 16:56.52 | sopo2k4 | exten => _1NXXNXXXXXX,3,Congestion() |
| 16:56.53 | sopo2k4 | exten => _1NXXNXXXXXX,103,Busy() |
| 16:56.54 | Op3r | [TK]D-Fender, searching at youtube now :D |
| 16:57.03 | DEac- | it's possible to forward ports to asterisk? my openwrt nat-router should forward the ports to the asterisk-machine. is this possible, that a can talk with peoples from the internet? |
| 16:57.23 | [TK]D-Fender | sopo2k4: you do NOT mix priorities like that. |
| 16:57.25 | Op3r | sopo2k4, just create another that accepts 011. |
| 16:57.46 | [TK]D-Fender | sopo2k4: You have failed to grasp the very basics of extens & priorities and desperately need to read .... THE BOOK |
| 16:57.47 | [TK]D-Fender | ~book |
| 16:57.48 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:58.47 | irule | http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip or here :) |
| 16:59.01 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 17:00.13 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 17:00.35 | *** join/#asterisk [[blah]asdf (n=ckwall@63.149.122.93) |
| 17:01.39 | *** join/#asterisk HaMYaI (i=HaMYaI@125-25-196-183.adsl.totbb.net) |
| 17:02.21 | Op3r | [TK]D-Fender, man william shatner and the guy from seinfield is in that video celebrity |
| 17:03.48 | [[blah]asdf | anyone interested in a TE405P Quad T1 card? DIgium is asking $1495. I would do it for substantially less. It has only been used for about 1 month. I went SIP and dont need it anymore. |
| 17:05.16 | *** join/#asterisk friedrich| (n=friedric@e177241143.adsl.alicedsl.de) |
| 17:05.46 | [[blah]asdf | I would take 50% just to recoup some cost out of it. |
| 17:06.36 | *** join/#asterisk osiris250 (i=r8x4umvm@bsd02.evansengineering.net) |
| 17:08.49 | Nugget | wow, moved from PRI to SIP. I just went the opposite direction and couldn't be happier about it |
| 17:09.13 | [[blah]asdf | well... I have 3 DS3s full of voice. This was in addition to that. |
| 17:09.24 | [[blah]asdf | Needed a backup solution to my TMD |
| 17:09.28 | [TK]D-Fender | Op3r: Entirely worthwhile, wasn't it? :) |
| 17:09.50 | Op3r | [TK]D-Fender, yep |
| 17:11.28 | [TK]D-Fender | [[blah]asdf: EBAY <--- |
| 17:11.48 | [[blah]asdf | yeah.... tried once there. |
| 17:11.49 | *** join/#asterisk sweeper (i=sweeper@softcheese.net) |
| 17:12.16 | sweeper | oh, when building zaptel, how do I specify where my kernel headers live? |
| 17:13.24 | *** join/#asterisk friedrich| (n=friedric@e177241143.adsl.alicedsl.de) |
| 17:13.46 | HaMYaI | I have " exten => _[13]X.,1,Set(GRP=g${EXTEN:0:1})" in my context but when I dialed 1000 it just didn't go to that exten |
| 17:14.14 | *** join/#asterisk gardo (n=gardo@121.97.176.180) |
| 17:14.52 | HaMYaI | it went to "exten => _X.,1,NoOP(${EXTEN})" instead |
| 17:15.17 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:15.30 | HaMYaI | it used to work in 1.4.x but I just downgrade to 1.2.x |
| 17:19.04 | [TK]D-Fender | HaMYaI: _X. = DUMB, you should not be using that in the same context that way |
| 17:19.13 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:19.27 | *** join/#asterisk sci_05 (n=peter@205-170-75-162.dia.static.qwest.net) |
| 17:21.23 | sci_05 | anyone ever setup a te207p with a dss circuit (no d channel, old style pri)? |
| 17:24.01 | [TK]D-Fender | sci_05: very low odds. You may want to ask Digium support on that one. |
| 17:26.27 | *** join/#asterisk tako-san (n=Tako-san@24.108.162.254) |
| 17:26.28 | tzafrir | sweeper, you normally don't need to specify where your kernel headers live |
| 17:26.36 | tzafrir | ls -l /lib/modules/`uname -r`/.config |
| 17:26.56 | tzafrir | sweeper, if something is there, then your kernel headers are most likely in place |
| 17:27.09 | sweeper | tzafrir: ah. well, it turns out I just didn't have the right ones installed, but it looked like it was looking for /include/linux/autoconf.h |
| 17:27.23 | *** join/#asterisk sci_05 (n=peter@205-170-75-162.dia.static.qwest.net) |
| 17:27.24 | sweeper | so I thought I might need to specify manually :) |
| 17:27.33 | tzafrir | sweeper, I think you're wrong |
| 17:27.36 | sci_05 | ok I will give them a shot [TK]D-Fender, thanks |
| 17:27.38 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 17:27.52 | BSD_Tech | brains halted |
| 17:27.58 | tzafrir | when building zaptel you should not use /usr/include/linux |
| 17:28.03 | sweeper | nonon |
| 17:28.10 | sweeper | /include/linux/autoconf.h <-- exactly thing |
| 17:28.13 | tzafrir | <PROTECTED> |
| 17:28.13 | sweeper | *that |
| 17:28.19 | sweeper | it was saying "not found" |
| 17:28.23 | sweeper | for that exact path |
| 17:28.30 | sweeper | which I thought was odd |
| 17:28.44 | sweeper | wow, asterisk compiled in 3 minutes |
| 17:28.53 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 17:28.53 | DEac- | how i can call sipphone, which asterisk doesn't know? |
| 17:28.55 | Uatec | Hi there |
| 17:29.03 | Uatec | can anyone tell me where the voices for the voicemail are kept? |
| 17:29.12 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:29.24 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:29.40 | tzafrir | sweeper, autoconf.h is actually an obsoleete file. If a module was looking for it and failed to find it, it should be fixed |
| 17:29.52 | jameswf | \ /var/spool/asterisk |
| 17:30.06 | sweeper | tzafrir: hmm |
| 17:30.18 | sweeper | I'll run make on zaptel again, see if it still looks for it |
| 17:30.25 | Uatec | jameswf, i mean the voice, not the messages |
| 17:30.44 | tzafrir | sweeper, zaptel looks at /lib/modules/`uname -r`/build by default |
| 17:31.25 | tzafrir | my mistake earlier. It should be: ls -l /lib/modules/`uname -r`/build/.config |
| 17:31.47 | tzafrir | if you built your own kernel, it should be valid |
| 17:32.15 | tzafrir | If you have a distro kernel, then a decent kernel headers package should provide that link |
| 17:32.32 | sweeper | mkay |
| 17:32.40 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:32.41 | tzafrir | The RH "kernel-headers" package is kernel headers for userspace programs. This is not what you need |
| 17:32.59 | tzafrir | in redhats you need kernel*-devel |
| 17:33.05 | sweeper | yep |
| 17:33.12 | sweeper | that's what I installed |
| 17:33.28 | sweeper | the problem was I didn't install the smp devel package |
| 17:33.45 | sweeper | but it struck me as odd that it would look for /include/ |
| 17:34.15 | tzafrir | sweeper, http://svn.digium.com/svn/zaptel/branches/1.4/README |
| 17:34.24 | tzafrir | comments are welcomed |
| 17:36.56 | *** part/#asterisk [[blah]asdf (n=ckwall@63.149.122.93) |
| 17:37.32 | HaMYaI | anyone running chan_unicall? |
| 17:37.45 | Uatec | how can i use g723 sound files in asterisk? |
| 17:40.38 | [TK]D-Fender | Uatec: call from a phone using that codec and you can play them back. |
| 17:41.13 | sopo2k4 | whats the most universal codec? |
| 17:41.28 | [TK]D-Fender | sopo2k4: G.711 |
| 17:41.29 | sweeper | g711 |
| 17:41.36 | sopo2k4 | ty. |
| 17:41.44 | sopo2k4 | ill make sure to use that one :) |
| 17:41.46 | [TK]D-Fender | sopo2k4: Ulaw for north America, ALW in most other places, |
| 17:41.55 | sopo2k4 | europe? |
| 17:41.58 | sopo2k4 | alw |
| 17:41.59 | [TK]D-Fender | sopo2k4: yup |
| 17:42.02 | [TK]D-Fender | ALAW* |
| 17:42.03 | sopo2k4 | ok ty |
| 17:42.08 | [TK]D-Fender | ALAW = G.711a |
| 17:42.17 | sopo2k4 | ok, cheers. :P |
| 17:42.26 | sopo2k4 | nearly fixd my problem :D |
| 17:42.47 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 17:44.18 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-87.ph.ph.cox.net) |
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| 17:46.11 | *** join/#asterisk gardo (n=gardo@121.97.176.180) |
| 17:46.41 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:51.17 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 17:51.20 | *** join/#asterisk tako-san (n=Tako-san@24.108.162.254) |
| 17:55.16 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 17:56.14 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-214.usadatanet.com) |
| 17:57.39 | Uatec | Conclusion. My work pc is not capable of running HL2DM |
| 17:59.10 | *** join/#asterisk enjay5150 (n=yea@74.202.4.2) |
| 18:02.16 | *** join/#asterisk VijayG (i=VijayG@202.131.145.247) |
| 18:03.33 | *** join/#asterisk Gershwin (n=fake@74.94.101.141) |
| 18:10.47 | MrChicken | HEllo |
| 18:10.58 | MrChicken | I have some SIP extensions in a queue (ATAs) |
| 18:11.30 | MrChicken | and from time to time I get this message |
| 18:11.31 | MrChicken | the device state of this member is still 'not in use' when it probably should not be |
| 18:18.32 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 18:18.40 | enjay5150 | brb |
| 18:19.24 | MrChicken | <PROTECTED> |
| 18:19.34 | Assid | anyone seen bkw ? |
| 18:19.44 | Mercestes | ~seen bkwruse |
| 18:20.06 | jbot | Mercestes: i haven't seen 'bkwruse' |
| 18:20.06 | Mercestes | ~seen bkw_ruse |
| 18:20.07 | jbot | Mercestes: i haven't seen 'bkw_ruse' |
| 18:20.07 | Mercestes | .. |
| 18:20.07 | Mercestes | ~seen bkw |
| 18:20.08 | jbot | bkw <n=bkw@tor.lindesign.se> was last seen on IRC in channel #debian, 6d 12h 18m 1s ago, saying: 'I trying to migrate a host to a virtual machine. I've copied all the files and mounted / to /target at the virtual machine. But I have problems writing bootloader with grub-install. I mean I cannot chroot target ; grub-install /dev/sda since sda doesn't ... |
| 18:20.12 | [TK]D-Fender | Mercestes: You are mixing people up.. |
| 18:20.21 | Mercestes | [TK]D-Fender, Yea, so I see. |
| 18:20.33 | *** join/#asterisk MindTheGap (n=iote@c9503fb4.bhz.virtua.com.br) |
| 18:20.35 | Mercestes | I see "BK" and I just think "ruse" can't help it. |
| 18:20.41 | Mercestes | ....really changed my feelings towards Burger King. |
| 18:20.47 | [TK]D-Fender | Mercestes: bkruse = Bryan Kruse. bkw = Brian K. West. |
| 18:21.01 | Mercestes | ..wow, that's good. |
| 18:21.05 | Mercestes | What does Mercestes stand for? |
| 18:21.05 | bkruse | woah |
| 18:21.19 | [TK]D-Fender | bkruse: Correct, no? |
| 18:21.21 | bkruse | bkruse = brandon kruse |
| 18:21.29 | [TK]D-Fender | bkruse: Apologies :) |
| 18:21.33 | [TK]D-Fender | and there you have it! |
| 18:21.34 | bkruse | close enough :] |
| 18:21.41 | file | Mr. Cake Guy |
| 18:21.48 | Mercestes | .. |
| 18:21.48 | Mercestes | :( |
| 18:22.01 | Mercestes | I don't even like cake. |
| 18:22.01 | [TK]D-Fender | bkruse: Like I say in pooll.... missed by less that 9 feet ;) |
| 18:22.06 | Mercestes | I'll take the sodomy please. |
| 18:22.11 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 18:22.18 | [TK]D-Fender | Mercestes: So you ARE looking for BKW! ;) |
| 18:22.24 | Mercestes | lmao |
| 18:22.24 | bkruse | [TK]D-Fender: yep, i would say about 3 inches |
| 18:22.35 | [TK]D-Fender | I am SO bad.... |
| 18:22.35 | bkruse | Mercestes: whats it about? ;[ |
| 18:22.52 | Mercestes | what's what about? |
| 18:23.00 | sopo2k4 | any ideas why i cant hear the other end of the line? |
| 18:23.03 | bkruse | whats "it" about? |
| 18:23.09 | Mercestes | what it? |
| 18:23.18 | Mercestes | sopo2k4, They are ignoring you. |
| 18:23.20 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:23.23 | sopo2k4 | apart from that |
| 18:23.26 | Mercestes | sopo2k4, Either that, or you have a NAT/Firewall issue. |
| 18:23.27 | sopo2k4 | i cant ignore myself :P |
| 18:23.39 | naitram | what is the recommended version of for linux? was using 1.4.0 and none of the sip dtmf signals worked. Used 1.4.4 and worked good. Update to 1.4.6 and works but audio is crap during record. what is best stable |
| 18:23.46 | waKKu | good afternoon folks |
| 18:23.53 | [TK]D-Fender | sopo2k4: considering you have shown us NOTHING and described very little.... no, we CAN'T help you. |
| 18:24.02 | sopo2k4 | what is there too show? |
| 18:24.04 | sopo2k4 | :S |
| 18:24.11 | Mercestes | a sip debug for one. |
| 18:24.12 | [TK]D-Fender | naitram: 2.6 is highly recommended |
| 18:24.15 | sopo2k4 | ok |
| 18:24.35 | Mercestes | 2.6? |
| 18:24.48 | Mercestes | did.....I miss something? |
| 18:24.55 | waKKu | folks.. someone knows how can I solve this problem intercommunicating 2 asterisks via IAX: [Jul 9 15:23:33] NOTICE[3344]: chan_iax2.c:6980 socket_process: Rejected connect attempt from 200.2.2.2, who was trying to reach '880@' ??? |
| 18:25.05 | [TK]D-Fender | Mercestes: Linux 2.6 |
| 18:25.11 | waKKu | both asterisk's r registered |
| 18:25.21 | Mercestes | I think he meant asterisk versions. |
| 18:25.35 | MindTheGap | does anyone ave any clue on asterisk 1.2 calls qualiti being degrade after some minutes? he have an old asterisk 1.2 system wich presents this behaviour. Local SIP calls work flawlessly for as long as we want them. Incomming and outgoing calls from Zap will start to degrade somewhere between 3 to 5 minutes and will require the user to dial again. Not all of the calls though... |
| 18:25.37 | BSD_Tech | mornig |
| 18:25.41 | waKKu | Mercestes was it for me ? |
| 18:25.43 | Mercestes | he clearly said "what is the recommended version of * for linux" |
| 18:25.45 | file | waKKu: registration simply tells the machine what IP address and port to send the call to, it does not tell about authentication... which seems to be your problem, it is not being authenticated as a user and isn't being directed to the right context |
| 18:25.48 | waKKu | oh.. sorry |
| 18:25.50 | naitram | Mercestes: yes, what asterisk version? |
| 18:25.53 | Op3r | does anyone heard any linux based sip phone that supports g729? |
| 18:26.04 | BSD_Tech | has anyone written dial plan to pull the local traffic report and play it |
| 18:26.04 | [TK]D-Fender | Mercestes: I asnwered the question he ASKED. |
| 18:26.15 | Mercestes | He asked what is the recommended version of * for linux. |
| 18:26.15 | anonymouz666 | [TK]D-Fender: what would you use to integrate two ast boxes? iax2 or sip? |
| 18:26.19 | Mercestes | so no you didn't. |
| 18:26.32 | waKKu | file hm... have some idea about where i need look ? |
| 18:26.58 | file | waKKu: well, if you could provide the iax.conf configuration sections for both sides, plus the Dial lines involved, then yes - I could get an idea (minus passwords of course) |
| 18:27.00 | waKKu | i had tried do a call ${TRUNK}/${EXTEN}@context .. but doesnt work |
| 18:27.01 | [TK]D-Fender | anonymouz666: What protocol do YOU think you should use for an Inter Asterisk connection? |
| 18:27.07 | Mercestes | naitram, Give 1.4.5 a try |
| 18:27.10 | naitram | [TK]D-Fender: ok, I am using Debian Linux 2.6.18, what is the recommended version of asterisk |
| 18:27.22 | waKKu | file one minute.. I have it done :) |
| 18:27.31 | [TK]D-Fender | naitram: Most would recommend 1.2 series * for production. |
| 18:27.48 | [TK]D-Fender | Mercestes: Got probelsm with 1.4.6? |
| 18:27.51 | Mercestes | No one with a green dot would recommend 1.2 for production. |
| 18:27.56 | anonymouz666 | [TK]D-Fender: SIP |
| 18:27.56 | [TK]D-Fender | Mercestes: Or are you just slow? :0 |
| 18:27.57 | Mercestes | [TK]D-Fender, he does. I haven't tried it yet. |
| 18:28.09 | Mercestes | He specified poor recording audio in 1.4.6 so I suggested 1.4.5 |
| 18:28.18 | [TK]D-Fender | anonymouz666: Fine, go for it that way then |
| 18:28.34 | Mercestes | Wow, your off today, fender? No wheaties? |
| 18:28.36 | anonymouz666 | IAX2 |
| 18:28.42 | anonymouz666 | I don't know |
| 18:28.48 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-e00e3bc7aee585c3) |
| 18:28.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:28.49 | anonymouz666 | I am just asking because I never used IAX2 |
| 18:29.08 | generalhan | Mercestes: what do you mean "no one with a green dot" ? |
| 18:29.29 | brian | in xchat channel operators are designated by a green dot |
| 18:29.39 | generalhan | ah ! i see |
| 18:29.40 | Mercestes | generalhan, Oh, I have xchat, I have a green dot. @ likely in your client. |
| 18:29.44 | [TK]D-Fender | anonymouz666: how about you just TRY and find out. |
| 18:30.16 | generalhan | Mercestes: ok, Green Dot, what would you recommend for production ? |
| 18:30.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:30.30 | anonymouz666 | [TK]D-Fender: Learning with others experience is just fine |
| 18:30.35 | waKKu | file http://pastebin.ca/610951 can u see ? |
| 18:30.51 | [TK]D-Fender | anonymouz666: thats what co-dependence is for.... |
| 18:31.01 | Op3r | one more try then Ill stop asking this. does anyone heard any linux based sip phone that supports g729? |
| 18:31.18 | generalhan | im still using 1.2.10, but i am moving to a new, better, server and was planning on going for the most recent 1.4 release at the time of the move |
| 18:31.26 | file | waKKu: use a separate user and peer |
| 18:31.52 | waKKu | file hm.. one to receive calls and other to make calls ? |
| 18:32.08 | file | waKKu: user gets matched for incoming calls, peer is used for outgoing calls... let me write up an example |
| 18:32.26 | [TK]D-Fender | waKKu: and specify your CONTEXT in your dial statement |
| 18:32.32 | waKKu | ok.. :) |
| 18:32.41 | waKKu | [TK]D-Fender yeah.. i had tried it too.. |
| 18:32.44 | [TK]D-Fender | waKKu: Shouldn't have to split your "friend" |
| 18:32.58 | Mercestes | generalhan, Meh, depends on which bugs you want. I use 1.2.13 but I have a voicemail forwarding issue. It's pretty stable tho. Otherwise, I am pretty sure 1.4.5 or 1.4.6 is pretty good. But, I haven't personally run it in production so I'm only guessing. |
| 18:33.28 | waKKu | [TK]D-Fender but only specifying @context I got same problem with auth |
| 18:33.38 | generalhan | Mercestes: well the recording issue you were mentioning someone else having is scary to me ... we LIVE off of our recordings here. |
| 18:33.57 | [TK]D-Fender | waKKu: SHOW us the attempt. |
| 18:34.23 | waKKu | oka ;) |
| 18:34.35 | Op3r | generalhan, been using asterisk 1.2.18 on production. It is being used by vicidial, but to the point. I never had any issues with recordings using it. |
| 18:34.37 | file | waKKu: http://pastebin.ca/610963 |
| 18:34.47 | *** part/#asterisk enjay5150 (n=yea@74.202.4.2) |
| 18:34.49 | file | there, my standard... most basic... examples for connecting two machines using IAX2 |
| 18:34.56 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-176-175.inter.net.il) |
| 18:34.59 | naitram | generalhan: I am the one with the recording issue. I am very very inexperienced so don't rely on much of what I say:} |
| 18:35.25 | generalhan | naitram: well what kind of issues ? i applogize if this is redundant, i didnt catch your issues before |
| 18:36.24 | VijayG | Hello, i need a dialplan using which my call should get disconnected automatically after a minute it has been connected |
| 18:36.34 | waKKu | [TK]D-Fender [Jul 9 15:35:15] WARNING[3339]: chan_iax2.c:7175 socket_process: Call rejected by 200.2.2.2: No authority found |
| 18:36.46 | file | VijayG: Set(TIMEOUT(absolute)=60) |
| 18:36.47 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 18:36.49 | VijayG | i am using set(TIMEOUT(absolute)=60) |
| 18:36.51 | VijayG | ya |
| 18:36.54 | waKKu | Jul 9 15:35:53 NOTICE[9083]: chan_iax2.c:6775 socket_read: Rejected connect attempt from 201.1.1.1, who was trying to reach '405@interno' |
| 18:36.58 | Op3r | VijayG, or check with your voip provider? |
| 18:37.00 | waKKu | [TK]D-Fender ^^ |
| 18:37.03 | VijayG | but this includes, dialing time also |
| 18:37.04 | waKKu | file checking |
| 18:37.09 | [TK]D-Fender | waKKu: never jsut paste a useless message like that alone. Always pastebin the ENTIRE call with iax debug enabled |
| 18:37.16 | file | VijayG: then you would want to use the Dial options |
| 18:37.25 | waKKu | ok.. good to know |
| 18:37.27 | Op3r | VijayG, oh sorry havent seen your question clearly |
| 18:37.28 | VijayG | whats that option/ |
| 18:37.29 | file | VijayG: type show application Dial |
| 18:37.31 | VijayG | ok |
| 18:37.35 | mocker | Anyone recommend any ACD reporting software addons? |
| 18:37.38 | naitram | generalhan: well, I am trying to do one touch recording via SIP channels. First I could not get 1.4.0 to even catch the dtmf signals. Nothing worked. Then 1.4.4 worked but would only accept the first char (ie.. 1 worked but *1 wouldn't). 1.4.6 will accept the two digits but the called party gets terrible motor boating sounds once recording starts |
| 18:37.40 | file | VijayG: specifically the L option |
| 18:37.41 | mocker | or ACD addons in general? |
| 18:38.30 | generalhan | naitram: is this an issue only with "one touch" recording ? |
| 18:39.44 | naitram | generalhan: the dtmf catching , yes, Don't know about using Monitor(...) in my dial script directly. Suppose I could check but, since don't intend to use it thay way.... |
| 18:40.06 | generalhan | naitram: thats what i was wondering ... i only use the Monito * |
| 18:40.26 | generalhan | Monitor() cause all calls are recorded here |
| 18:40.34 | naitram | generalhan: yeah, that worked fine from the dial scripts with 1.4.0. Probably the others, too |
| 18:40.35 | generalhan | oh well .. i will just have to test it i guess ! |
| 18:40.54 | generalhan | good to know ... i think i will go with a 1.4 release for the new machine ! |
| 18:41.42 | naitram | generalhan: do you mix with soxmix using the Monitor(..,m) flag |
| 18:41.55 | generalhan | yes |
| 18:42.08 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 18:42.16 | generalhan | naitram: but something happened to my system about a month ago and nothing works right anymore. |
| 18:42.33 | generalhan | so i have to manually run a soxmix script each night to make it happen ! |
| 18:43.33 | *** join/#asterisk gardo (n=gardo@121.97.197.207) |
| 18:43.44 | naitram | generalhan: oh, might try the 1.4.0 release first then. Right now on the 1.4.6 I cant get the soxmix right either. It did work on the 1.4.4 |
| 18:43.49 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 18:43.57 | kombi | I hate to it but I havn't gotten any further on this.. how do you kill a conference on hangup? |
| 18:44.02 | bkruse | hey techie |
| 18:44.07 | kombi | *say.. |
| 18:44.13 | techie | hello brandon |
| 18:44.25 | Qwell[] | kombi: the conf should die when the last user leaves |
| 18:45.06 | kombi | Qwell: that's right, only I'd like it to die regardless |
| 18:45.45 | Qwell[] | when who hangs up? |
| 18:45.48 | kombi | or need even, maybe fire of an agi script on hangup, but that seems so not elegant |
| 18:46.15 | Qwell[] | isn't there already an option to kill a conf when the admin leaves? |
| 18:46.42 | kombi | Qwell: well that's the thing, it is started by a script and then joined by the caller who invoked the script |
| 18:46.46 | yonahw | anybody ever hear of a default webserver password for snom phones other than admin/admin? |
| 18:47.05 | waKKu | file man.. very thanks ;) |
| 18:47.15 | Hmmhesays | file |
| 18:47.18 | file | found |
| 18:47.22 | waKKu | ur example using vitoria@flripa and floripa@vitoria solve my problem man :D |
| 18:49.20 | file | Strom_M: you don't have to have a reason |
| 18:49.36 | naitram | Op3r: look at pjsip.org this api works pretty good and does not support the codec directly but you are supposed to be able to use free g729 codec with some changes. Search for pjsip g729 and read the thread |
| 18:50.57 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 18:51.08 | Hmmhesays | ok my centos kernel-headers packages doesn't seem to have any headers in it |
| 18:51.55 | macTijn | heh |
| 18:52.09 | macTijn | rpm -ql kernel-headers |
| 18:52.21 | Op3r | Hmmhesays, yuminstall kernel-devel? |
| 18:56.25 | Hmmhesays | yeah I got it |
| 18:56.27 | Hmmhesays | danke |
| 18:57.34 | *** join/#asterisk Andretii (i=Andretii@adsl-75-22-21-45.dsl.chcgil.sbcglobal.net) |
| 18:58.34 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 18:59.02 | Andretii | anyone knows how to set the RXgain for an individual channel? |
| 18:59.39 | [TK]D-Fender | Andretii: Yeah.... rxgain=2 |
| 18:59.46 | [TK]D-Fender | Andretii: channel=>3 |
| 18:59.50 | [TK]D-Fender | *yay* |
| 19:00.14 | Andretii | just specifying the channel like that will od? |
| 19:00.16 | Andretii | do* |
| 19:00.42 | Hmmhesays | hmm modprobe ztdummy not found |
| 19:00.43 | Hmmhesays | gar |
| 19:01.24 | [TK]D-Fender | Andretii: well it'll set it for that channel and any below it. |
| 19:01.42 | [TK]D-Fender | Andretii: So you'd want to set it back right after |
| 19:02.17 | Andretii | [TK]D-Fender i want to leave it on higher for a fax line |
| 19:02.32 | [TK]D-Fender | Andretii: tahts fine |
| 19:03.09 | Andretii | [TK]D-Fender so my fax line is in channel 8 i will do rxgain=8 channel=8 |
| 19:03.12 | mocker | Hmm, QueueMetrics looks fairly nice. |
| 19:03.29 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 19:03.30 | [TK]D-Fender | Andretii: that is a high gain....... |
| 19:03.48 | Andretii | [TK]D-Fender do you know the file to edit? zapata.conf will it be? |
| 19:03.59 | [TK]D-Fender | Andretii: Yes. |
| 19:04.34 | Mercestes | ~phones |
| 19:04.34 | jbot | somebody said phones was http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:04.37 | Andretii | [TK]D-Fender yea i know is high but it worked like that fine |
| 19:04.53 | [TK]D-Fender | Andretii: Sure, whatever works I guess... |
| 19:06.09 | [TK]D-Fender | I am going to completely redo [av]bani's phones list.... |
| 19:11.17 | Andretii | [TK]D-Fender i see what was my problem again, if i have the rxgain=5 and channel=8 what rxgain will get the channels 1-7 i need them with 2 |
| 19:11.35 | Andretii | or with teh default |
| 19:11.58 | [TK]D-Fender | Andretii: Set your gains for the first few channels. Then define them. Change the gain, set THAT channel. Change back after if more channels to define. |
| 19:12.53 | MrChicken | <PROTECTED> |
| 19:13.52 | [TK]D-Fender | MrChicken: "probably" = useless comment. Pastebin CLI output proving your systems state and your configs and maybe we'll be able to help you. |
| 19:14.40 | MrChicken | oki gimme a min |
| 19:14.56 | Hmmhesays | ok what the hell |
| 19:15.17 | Hmmhesays | after I make install zaptel I can't modprobe it |
| 19:15.31 | *** join/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 19:15.56 | bancus | Hey. Does anyone know if T1 cards like the TE220 can be used to do both voice and data, or will a separate adapter be needed to run data? |
| 19:16.57 | Andretii | [TK]D-Fender can you check my zapata and give me an idea of teh breakdown of the channels since i only see one group |
| 19:17.41 | [TK]D-Fender | bancus: Yes they can. Go read the WIKI on them |
| 19:17.50 | bancus | Oh? |
| 19:17.57 | bancus | I tried googling but didn't find anything. |
| 19:19.33 | *** join/#asterisk PHPadam (n=ok@82.166.209.166) |
| 19:19.41 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 19:19.53 | [TK]D-Fender | bancus: My 10 second search : http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration |
| 19:19.56 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 19:20.08 | bancus | What were the search terms? |
| 19:20.08 | [TK]D-Fender | bancus: Try HARDER. And NO, I've never done this personally. |
| 19:20.19 | bancus | Out of curiosityy. |
| 19:20.24 | agile | hey any of you use festival through AGI ? |
| 19:20.27 | bancus | Clearly your google-fu is better than mine. |
| 19:20.27 | [TK]D-Fender | <- #asterisk 's resident Google / WIKI proxy..... |
| 19:20.35 | [TK]D-Fender | bancus: "T1 data" <- |
| 19:20.35 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 19:20.38 | agile | trying to figure out how to properly execute it |
| 19:20.41 | bancus | damn |
| 19:20.44 | kombi | How do you trigger an event on hangup? DeadAGI? |
| 19:20.45 | PHPadam | hi, im a newbie to asterisk, is there a special equipment that i need other than 1pc(linux) ? |
| 19:20.53 | Optic | hihi, does anyone have the latest polycom sip software? :) |
| 19:20.59 | kombi | PHPadam: nope |
| 19:21.01 | docelmo | anyone in here having any problems with 1.2.20 and Polycom 601 with SIP2.1.0? I have set it up to register but when it comes in to register its not registering.. |
| 19:21.04 | [TK]D-Fender | PHPadam: Depends what hardware you want to use with it |
| 19:21.11 | Mercestes | gentoo-jutsu |
| 19:21.25 | Mercestes | hehe |
| 19:21.36 | bancus | [TK]D-Fender: Thanks. |
| 19:21.37 | PHPadam | i want to have multiple phones in my fathers business, but i wonder how i plug those phones to the pc, and which phones do i need? |
| 19:21.46 | Hmmhesays | hmmm undefined reference to `tasklet_kill' |
| 19:21.52 | [TK]D-Fender | Mercestes: What was that about bringing a kinfe to a gun-fight? And you want to come empty handed? :D |
| 19:22.20 | [TK]D-Fender | PHPadam: For the lines : Sangoma A200d , for phones, depends on your needs/wants/budget. |
| 19:22.22 | docelmo | PHPadam I like the polycom if you can afford them.. if not then use grandstream gxp2000's |
| 19:22.26 | [TK]D-Fender | PHPadam: What kind of call volume? |
| 19:22.42 | [TK]D-Fender | PHPadam: SIP hard phones agreaable for you? Any special needs? |
| 19:22.51 | Hmmhesays | http://www.pastebin.ca/611061 <--- there is my ztdummy error |
| 19:22.54 | [TK]D-Fender | ~gs |
| 19:22.55 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:22.57 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:23.25 | docelmo | I have a 2000 and 3000 I kinda like them.. |
| 19:23.42 | PHPadam | call volume? hmm, 30 phones, people call here and there, not a busy line |
| 19:23.54 | PHPadam | [TK]D-Fender, SIP ? |
| 19:24.06 | MrChicken | http://www.pastebin.ca/611066 |
| 19:24.20 | kombi | Hmmhesays: you might want to recompile everything in proper order |
| 19:24.26 | MrChicken | <PROTECTED> |
| 19:24.31 | [TK]D-Fender | PHPadam: SIP = VoIP protocol. Basically hardware phones that plug into your network to talk to your * server |
| 19:24.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:24.58 | *** join/#asterisk |yonahw| (n=yonahw@IGLD-83-130-176-175.inter.net.il) |
| 19:24.58 | [TK]D-Fender | PHPadam: http://www.telephonydepot.com/Polycom_s/25.htm |
| 19:25.09 | Hmmhesays | is libpri required if you are using ztdummy? |
| 19:25.12 | [TK]D-Fender | PHPadam: IP 320 = plenty for most users needs |
| 19:25.16 | [TK]D-Fender | Hmmhesays: no |
| 19:25.23 | kombi | Jmmhesays: yip |
| 19:25.25 | Hmmhesays | so just zaptel |
| 19:25.28 | Hmmhesays | then asterisk |
| 19:25.29 | |yonahw| | did anyone answer my question about snom's default password I had trouble with my internet connection |
| 19:25.32 | [TK]D-Fender | Hmmhesays: Correct |
| 19:25.48 | ai-a[wrk] | |yonahw|: whats the problem ? |
| 19:25.53 | PHPadam | so, do i plug the server to my internet wan? how does it dial to phones? |
| 19:26.01 | PHPadam | do i need like a skype account? |
| 19:26.13 | *** part/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 19:26.25 | ai-a[wrk] | |yonahw|: i thought snom phones dont have a pw as default... |
| 19:26.39 | Hmmhesays | are usb ports required for ztdummy with teh 2.6 kernel? |
| 19:26.51 | [TK]D-Fender | MrChicken: Very poor pastebin. No channel dump, no sip peer dump. |
| 19:26.51 | kombi | caller hangs up -> execute command.. how? |
| 19:27.00 | kombi | Hmmhesays: I'd say no |
| 19:27.12 | [TK]D-Fender | MrChicken: No QUEUE STATUS dump.... |
| 19:27.39 | [TK]D-Fender | kombi: Get off your ass, go to the WIKi and read up on "Asterisk Standard Extensions". |
| 19:27.43 | NOT_guru | question : does the Zaptel driver 1.2.18 require the spinlock fix still? |
| 19:27.53 | |yonahw| | ai-a[wrk]: it would seem that way from the documentation but when i go to the ipo address it requests a password |
| 19:28.18 | [TK]D-Fender | PHPadam: No need for anything outside you local LAN |
| 19:28.20 | ai-a[wrk] | |yonahw|: reset the phone to factory settings. |
| 19:28.37 | |yonahw| | ai-a[wrk]:good idea, I should have thought of that |
| 19:28.39 | |yonahw| | thanks |
| 19:28.42 | [TK]D-Fender | PHPadam: You will connect your lines intot he A200d in your server, and your phones will talk to you * server via your local LAN. |
| 19:28.46 | PHPadam | [TK]D-Fender, i dont get it, how does it go from the internet to the phone lines? with phone numbers etc. |
| 19:28.49 | kombi | Fender, I read that thing inside out, whyn't you just tipp me off to somewhere? |
| 19:28.51 | [TK]D-Fender | PHPadam: nothing "internet" about this. |
| 19:28.55 | ai-a[wrk] | |yonahw|: not sure HOW haha |
| 19:29.01 | kombi | PHPadam: use sip provider |
| 19:29.03 | Hmmhesays | ztdummy.c:59:26: error: linux/module.h: No such file or directory |
| 19:29.10 | [TK]D-Fender | PHPadam: hardware card in your server to interface with the lines. |
| 19:29.18 | Andretii | [TK]D-Fender can you check my zapata and give me an idea of teh breakdown of the channels since i only see one group? http://pastebin.ca/611044 |
| 19:29.19 | Hmmhesays | ahh I see, it can't find module.h, but it is in my /usr/src/linux/include/linux |
| 19:29.30 | [TK]D-Fender | kombi: Link me to the page you claim to have read <- |
| 19:29.43 | docelmo | sigh.. forgot the damn nat=yes statement |
| 19:29.50 | [TK]D-Fender | Andretii: ..... |
| 19:29.52 | [TK]D-Fender | ~freepbx |
| 19:29.53 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:29.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:30.05 | PHPadam | so basicly my asterisk server is supposed to be connected to the phone line? |
| 19:30.12 | [TK]D-Fender | PHPadam: Yup. |
| 19:30.20 | [TK]D-Fender | PHPadam: Via the PCI card I listed for you |
| 19:30.29 | kombi | Fender: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions <- that one? |
| 19:30.50 | kombi | it just does not tell me shit.. |
| 19:31.01 | PHPadam | i dont get it, wasnt it supposed to behave like skype, which reduces my expenses by using voip? |
| 19:31.03 | *** join/#asterisk sopo2k4 (n=jam@host81-152-232-54.range81-152.btcentralplus.com) |
| 19:31.18 | [TK]D-Fender | kombi: the you are ^%#@$# blind, its sitting right in there |
| 19:31.18 | sopo2k4 | anyone able to paste me part of their extensions file that sets the CID set inside the voip application? |
| 19:31.31 | MrChicken | [TK]D-Fender <-- http://www.pastebin.ca/611087 QUEUE dump |
| 19:31.31 | kombi | hrmpf.. |
| 19:31.43 | NOT_guru | yes phpadam you can use services like viatalk ( I only mention them as thats who I use ) and not need to connect your PBX to a phone line |
| 19:31.51 | [TK]D-Fender | kombi: And you have failed to even text search it for the obvious key-word... |
| 19:31.58 | kombi | PHPadam: I you use a sip provider, it will take care of PSTN to SIP |
| 19:32.09 | kombi | hrmpf... |
| 19:32.09 | |yonahw| | ai-a[wrk]: thanks for the notion solved the problem as im sure you knew it would |
| 19:32.13 | Hmmhesays | so where is it looking for linux/module.h? |
| 19:32.25 | NOT_guru | I do also have a tdm 4XX card but thats a later addition |
| 19:32.40 | kombi | I look at man pages all day, they fade in front of my eyes.. |
| 19:32.57 | [TK]D-Fender | PHPadam: http://www.telephonydepot.com/product_p/105-052-a200brme.htm <- this wit enough modules to account for your lines |
| 19:33.04 | kombi | Fender, please, why don't you, for once, tell me... lol... |
| 19:33.22 | [TK]D-Fender | kombi: FFS look for the damn word "HANGUP" in there. |
| 19:33.30 | kombi | l... |
| 19:34.00 | [TK]D-Fender | PHPadam: You said you wanted to use your EXISTING lines. |
| 19:34.26 | [TK]D-Fender | PHPadam: You need to be very clear between what you have NOW, and what you WANT TO USE in your new scenario. |
| 19:34.53 | Hmmhesays | ok this is driving me nuts |
| 19:35.06 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 19:35.21 | [TK]D-Fender | kombi: well? found it yet?! |
| 19:35.37 | PHPadam | [TK]D-Fender, sorry im a newbie, i wanna convert their current normal phones to use voip to reduce costs on monthly usage |
| 19:35.40 | kombi | yip.. |
| 19:35.44 | kombi | still.. |
| 19:36.01 | kombi | where does damn letter h go, christ.. |
| 19:36.01 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 19:36.09 | [TK]D-Fender | PHPadam: Ah, have you picked out a provider to port your land lines #? |
| 19:36.25 | PHPadam | nop, i have no idea who can do that, im from israel |
| 19:36.30 | [TK]D-Fender | kombi: its a frggen EXTEN. You need to relearn the most basic bits of * all over... |
| 19:36.41 | [TK]D-Fender | PHPadam: Ok, thats the first thing then. |
| 19:36.56 | [TK]D-Fender | PHPadam: Go find out what providers will give you the service you want in the area you want. |
| 19:37.00 | kombi | jeez.. |
| 19:37.17 | PHPadam | [TK]D-Fender, which equipment do i need inorder to be a provider myself ? |
| 19:37.18 | Mercestes | [TK]D-Fender, how do I assign priorties to my commands in extensions.conf? if I want one command to run before the other one, how do I make sure that happens? |
| 19:37.19 | Mercestes | >.> |
| 19:38.17 | Hmmhesays | can someone tell me the full path that zaptel is looking for linux/module.h? |
| 19:38.21 | kombi | PHPadam: why would you want to be that? |
| 19:38.41 | PHPadam | kombi, if such service isnt available in israel.. it can be an oppertunity |
| 19:38.54 | [TK]D-Fender | PHPadam: technically that card alone could be all you need, but that'd be nuts... |
| 19:39.04 | [TK]D-Fender | PHPadam: learn to walk before worrying about FLYING. |
| 19:39.17 | kombi | good one.. |
| 19:39.26 | PHPadam | [TK]D-Fender, im curious first |
| 19:39.32 | [TK]D-Fender | PHPadam: Especially since you didn't even know what SIP is. |
| 19:39.43 | [TK]D-Fender | PHPadam: Sure thing pussycat.... |
| 19:39.47 | PHPadam | im not really gonna do it, im just curious |
| 19:40.10 | PHPadam | ill need to find a provider |
| 19:40.21 | PHPadam | can i use a provider from the usa or do i need on that is local ? |
| 19:40.24 | [TK]D-Fender | PHPadam: Go download *. Start playing around with it with just soft-phones. Then feel free to get some "ideas" :) |
| 19:40.36 | Mercestes | PHPadam, I'm pretty sure it involves a lofty payment to your local government for the appropriate ...."licenses" we call them in America. |
| 19:40.47 | mocker | ~softphone |
| 19:40.48 | jbot | something that should be drug out into the street and shot |
| 19:41.01 | mocker | Damn, looking for recommendations on Windows softphones. |
| 19:41.09 | Mercestes | Then you will need to setup your switch, and then when you have #'s ported to you you will have to go through the proper channels to advertise your ANI to your switch location. |
| 19:41.13 | mocker | The last time I used any I just went w/ eyeBeawm. |
| 19:41.15 | [TK]D-Fender | mocker: idefisk |
| 19:41.15 | PHPadam | what are softphones? |
| 19:41.17 | kombi | mocker: x-lite, idefisk.. |
| 19:41.18 | tako-san | mocker: SIP or IAX? |
| 19:41.21 | mocker | SIP |
| 19:41.30 | mocker | thanks |
| 19:41.38 | Mercestes | Probably the *easiest* way to do that is to buy a regular commercial switch instead of trying to hook * up and have everyone connect to you that way |
| 19:41.41 | [TK]D-Fender | PHPadam: Stop now. Go download THE BOOK, and get busy! |
| 19:41.43 | [TK]D-Fender | ~book |
| 19:41.43 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:41.45 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 |
| 19:42.00 | PHPadam | which book? where? |
| 19:42.06 | Mercestes | ...oh gods. |
| 19:42.08 | kombi | sigh.. |
| 19:42.09 | PHPadam | oh |
| 19:42.23 | PHPadam | 10x |
| 19:43.01 | PHPadam | sorry to piss you off guys |
| 19:43.01 | Mercestes | That should be on bash.org |
| 19:43.01 | kombi | you didn't.. |
| 19:43.01 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 19:43.05 | mocker | idefisk looks nice. |
| 19:43.10 | mocker | except for the skin ;) |
| 19:43.29 | [TK]D-Fender | mocker: FUGLY I know... but it has native transfer, and supports SIP/IAX2 |
| 19:43.48 | NOT_guru | FUGLY thats my CSS name |
| 19:43.50 | NOT_guru | LOL |
| 19:43.51 | mocker | And click to dial, reading of URLs sent from Asterisk. |
| 19:43.52 | [TK]D-Fender | mocker: I was presuming you meant the best FREE soft-phone for Windows.... |
| 19:43.56 | NOT_guru | sorry offtopic I shush now |
| 19:44.05 | mocker | Free is always good. |
| 19:44.16 | mocker | But if there is a *great* for pay, I'm not opposed. |
| 19:45.46 | PHPadam | LOL http://bash.org/?99060 |
| 19:45.46 | [TK]D-Fender | mocker: What do you execpt / need? |
| 19:45.50 | [TK]D-Fender | expect |
| 19:45.59 | mocker | [TK]D-Fender: Easy for users to understand. |
| 19:46.11 | mocker | Not having to throw YaaCID in for screen pops would be nice too. |
| 19:46.22 | mocker | Click to dial is nice. |
| 19:46.24 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:46.37 | [TK]D-Fender | mocker: you mean for REGULAR use?!?! ICK! |
| 19:46.40 | mocker | Ahh, the *must* feature would be a seperate ring / audio device.. |
| 19:47.13 | [TK]D-Fender | mocker: eyebeam does it all I think./ |
| 19:50.10 | PHPadam | this is the funniest ever, oh god - http://bash.org/?287414 |
| 19:50.14 | PHPadam | ok cya guys bye |
| 19:52.58 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:53.54 | kombi | exten => h,n,DoFreakinStuff() does not do the freakin stuff after caller hung up, I'll bite my finger off next and get a job digging ditches.. |
| 19:54.29 | tzanger | kombi: what is it being told to do, exactly |
| 19:54.30 | [TK]D-Fender | kombi: pastebin your attempt to use it.... |
| 19:54.34 | tzanger | if it's an AGI it won't work |
| 19:54.45 | [TK]D-Fender | tzanger: Let him incriminate himself! |
| 19:54.51 | [TK]D-Fender | tzanger: SHUSH! |
| 19:54.51 | tzanger | [TK]D-Fender: :-) |
| 19:54.58 | kombi | ..about to.. |
| 19:55.04 | tzanger | aren't you supposed to be parlaying vous francaise? :-) |
| 19:55.14 | tzanger | (good cop, bon cop... good flick) |
| 19:55.23 | [TK]D-Fender | tzanger: va t'ens mon ostie! |
| 19:55.45 | tzanger | ostie? |
| 19:56.26 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 19:56.34 | kombi | http://pastebin.ca/611131 <- treat him gentle.. |
| 19:56.48 | [TK]D-Fender | tzanger: va t'ens mon ostie trou-de-cul! |
| 19:57.13 | [TK]D-Fender | kombi: Now you have to relearn PRIORITIES. |
| 19:57.26 | [TK]D-Fender | tzanger: Its for the best, really ;) |
| 19:57.37 | tzanger | heh |
| 19:57.40 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 19:57.46 | pifiu | helloooo! |
| 19:58.09 | kombi | Fender: I now it sounds far fetched, but could elaborate on that just a teeny weeny bit? |
| 19:58.23 | kombi | first answer, then meet, then kick, no? |
| 19:58.59 | [TK]D-Fender | kombi: "h,n", <- there is no griggen "n" because that EXTEN (it is its own!) doesn't have a step ***1 *** |
| 19:59.22 | bkruse | free your phone. |
| 19:59.27 | kombi | oh, I just left that out in the bin there.. |
| 20:00.14 | kombi | http://pastebin.ca/611140 -> there |
| 20:00.28 | kombi | (Bob Ross couldn't have said it better ) |
| 20:00.43 | [TK]D-Fender | kombi: You are NOT getting it... exten => h,n,MeetMeAdmin(100|K) <- where is "h1,"?! |
| 20:00.45 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:00.51 | *** join/#asterisk jungleexplorer (n=kvirc@dsl54006045.pool.t-online.hu) |
| 20:01.19 | mountainm2k | Is there Realtime for meetme? I'm not getting it to work as I would think it should... |
| 20:01.25 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 20:01.37 | kombi | aiyyy.. |
| 20:01.38 | jungleexplorer | hello everyone |
| 20:02.21 | jungleexplorer | it might be a bit off topic... |
| 20:02.40 | Strom_M | jungleexplorer: no, pepsi is not an asterisk protocol |
| 20:02.45 | jungleexplorer | did anybody tried to make cti connection to an alcatel 4400? |
| 20:02.48 | ai-a[wrk] | okay, i have a ISDN line plugged into my super smart Sangoma A101D card (wv. Echo Cancellation) - and have no idea what is going on ;) (asterisk output - http://pastebin.ca/611144 ) any ideas why zaptel is failing? |
| 20:02.50 | *** join/#asterisk ManxPower (n=manxpowe@88.sub-70-218-253.myvzw.com) |
| 20:03.17 | kombi | not me marbles... |
| 20:03.28 | jungleexplorer | ... from linux maybe... |
| 20:03.42 | kombi | oh christ, I am glad nobody sees me.. |
| 20:03.43 | *** join/#asterisk dandan (n=dandan@yarde-GW.customer.alter.net) |
| 20:03.45 | dandan | ~books |
| 20:03.56 | dandan | jbot: books |
| 20:03.59 | dandan | :) |
| 20:04.01 | dandan | come on! |
| 20:04.09 | Strom_M | ~book |
| 20:04.10 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:04.12 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:04.14 | Strom_M | I WIN |
| 20:04.22 | dandan | lol |
| 20:04.25 | dandan | hey guys :) |
| 20:04.36 | [TK]D-Fender | Strom_M: My client seems to think I won ;) |
| 20:04.45 | dandan | 16:04 < dandan> come on! |
| 20:04.46 | dandan | 16:04 < Strom_M> ~book |
| 20:04.46 | dandan | 16:04 < [TK]D-Fender> ~book |
| 20:04.47 | dandan | 16:04 < jbot> from memory |
| 20:04.48 | [TK]D-Fender | Strom_M: although jbot seems to swing your way |
| 20:04.50 | dandan | sorry :) |
| 20:04.53 | MrChicken | <PROTECTED> |
| 20:05.01 | [TK]D-Fender | I'll get over the shame, don't worry ;) |
| 20:05.04 | Strom_M | jbot always swings my way |
| 20:05.05 | Strom_M | ;) |
| 20:05.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 20:05.41 | dandan | btw. does oreilly have any other * books? |
| 20:05.52 | dandan | afair - they do... |
| 20:07.22 | *** join/#asterisk op3r (n=op3r@125.212.125.250) |
| 20:07.29 | op3r | any one experienced this? rtp.c:576 ast_rtp_read: Unknown RTP codec 126 received |
| 20:08.21 | kombi | and the thing is working... oh my god.. if you had vision you'd see someone trying to whip his own bud.. |
| 20:08.23 | [TK]D-Fender | op3r: http://www.asteriskguru.com/tutorials/unknown_codec_received.html |
| 20:08.51 | op3r | [TK]D-Fender: good thing or a bad thing? |
| 20:09.14 | [TK]D-Fender | op3r: READ |
| 20:10.34 | op3r | [TK]D-Fender: yep Im reading but its not there. Anyhow its just that it annoys me when seeing that but all in all I dont have any problems with dialling or anything except that stuff pops out on the cli :( |
| 20:11.16 | generalhan | is anyone here running 1.4 on a 64bit distro ? im just looking for some feedback as to whether its better, worse, the same ? |
| 20:11.45 | op3r | generalhan: ur using it on production environment? |
| 20:12.00 | generalhan | op3r: yes |
| 20:12.29 | op3r | generalhan: brave |
| 20:12.30 | [TK]D-Fender | op3r: Looking like a G.76 varient. |
| 20:12.36 | [TK]D-Fender | op3r: Disable in your client |
| 20:12.41 | dandan | oh one more thing: do you guys use TDMoE? is it a viable option? |
| 20:12.55 | dandan | (need to split multiple Ts nicely, for redundancy) |
| 20:12.57 | [TK]D-Fender | dandan: option for WHAT? |
| 20:12.59 | generalhan | op3r: "brave" ? |
| 20:13.09 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 20:13.20 | dandan | well... |
| 20:13.29 | [TK]D-Fender | dandan: Depends if you trust the device DOING it, as it centralizes a point of failure. |
| 20:13.41 | dandan | http://www.red-fone.com/Products/ |
| 20:13.42 | dandan | this |
| 20:13.44 | [TK]D-Fender | dandan: And keep in mind nobody gives a rats ass about TMDoE |
| 20:13.55 | [TK]D-Fender | dandan: a NON-HEWC crap box?! LOL |
| 20:13.56 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 20:13.59 | [TK]D-Fender | HWEC* |
| 20:14.11 | dandan | They are supposed to come up with one soon |
| 20:14.11 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 20:14.14 | dandan | waiting for it |
| 20:14.17 | dandan | got an alternative? |
| 20:14.23 | [TK]D-Fender | dandan: All I can say is : yuck. |
| 20:14.35 | [TK]D-Fender | dandan: How many ports, and what do you really neeed? |
| 20:15.07 | dandan | well... I got 10-12 branches, all with Ts (at least two, fractional/PRI) and need to do some failover |
| 20:15.13 | dandan | if the server decides to die |
| 20:15.27 | dandan | need to route all the traffic to the other box |
| 20:15.35 | dandan | that I have in Standby |
| 20:15.42 | dandan | would like to have that done automatically |
| 20:16.19 | mtoups | so, i am running 1.4.6 now and i still have the 'asterisk' process using 100% CPU while doing apparently nothing. (this originally started happening after a 1.4.4 -> 1.4.5 upgrade) |
| 20:16.41 | dandan | mt: I am still on 1.4.4 can't help you :/ |
| 20:21.27 | russellb | mtoups: would you be willing to let someone log in to see what is causing that to happen? |
| 20:25.43 | mtoups | russellb: possibly, i would sanitize the configs and such first, but we could try that |
| 20:28.00 | mtoups | russellb: if there are simple things i could check first let me know |
| 20:28.10 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 20:29.46 | russellb | well, you could try to get a backtrace yourself first |
| 20:30.01 | russellb | first build without optimizations ... make menuselect -> Compiler Flags -> DONT_OPTIMIZE |
| 20:30.04 | russellb | make clean ; make ; make install |
| 20:30.24 | russellb | then, when it is in that state, use the contrib/scripts/ast_grab_core script to get a backtrace |
| 20:30.25 | holiday_42 | ntoups: is it something simple like synchronouse loggin instead of async? |
| 20:30.40 | russellb | huh? |
| 20:30.49 | mtoups | holiday_42: would this change by doing an upgrade? |
| 20:31.02 | mtoups | originally i was going between debian's 1.4.4 and 1.4.5 packages |
| 20:31.08 | mtoups | but i have built 1.4.6 from source now |
| 20:31.09 | holiday_42 | ntoups: sorry, thought you said it was doing it prior and upgrade did not help |
| 20:31.22 | mtoups | russellb: i will rebuild as you suggest |
| 20:31.37 | holiday_42 | ntoups: i re-read.. nevermind |
| 20:31.42 | mtoups | holiday_42: no problem |
| 20:34.33 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 20:35.43 | af_ | why use 1.4 instead 1.2? |
| 20:36.19 | Qwell[] | Because 1.2 won't have any new releases after less than a month from now. |
| 20:36.36 | af_ | stability is my major concern |
| 20:36.46 | wunderkin | o rly? |
| 20:36.54 | Qwell[] | then people had better start reporting bugs against 1.4 if they want them fixed. |
| 20:37.22 | *** join/#asterisk xlyz (n=xz@host-84-223-114-7.cust-adsl.tiscali.it) |
| 20:37.24 | af_ | there is any good reason to switch to 1.4 for a production system? |
| 20:38.12 | *** part/#asterisk xlyz (n=xz@host-84-223-114-7.cust-adsl.tiscali.it) |
| 20:38.16 | *** join/#asterisk xlyz (n=xz@host-84-223-114-7.cust-adsl.tiscali.it) |
| 20:39.06 | *** part/#asterisk xlyz (n=xz@host-84-223-114-7.cust-adsl.tiscali.it) |
| 20:39.20 | bkruse | because 1.2 wont be supported soon? |
| 20:39.42 | af_ | production system=anything works |
| 20:39.58 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:40.02 | af_ | wonderign if any major improvment |
| 20:41.03 | waKKu | folks.. where do I configure a timeout for extensions checking ? example: I press 889, asterisk waits 10 seconds to begin a ring :/ ... I know that using # I can solve it, but 10 seconds is very long time |
| 20:41.21 | Juggie | dont configure overlaping extensions |
| 20:41.26 | dandan | lol, I found some CVS from 04 today |
| 20:41.28 | dandan | it works |
| 20:41.32 | dandan | never upgraded |
| 20:41.46 | waKKu | Juggie i dont have overlap :/ |
| 20:42.00 | waKKu | its happen with zap channel too |
| 20:42.09 | Juggie | its your dialplan |
| 20:42.15 | Juggie | its not setup properly |
| 20:42.19 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 20:42.20 | Juggie | poste it to www.pastebin.ca |
| 20:42.21 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 20:42.25 | Juggie | *paste |
| 20:42.33 | dandan | *CLI> show version |
| 20:42.34 | dandan | Asterisk CVS-04/27/04-18:17:59 built by root@rowing on a i686 running Linux |
| 20:42.37 | dandan | lol :) |
| 20:42.41 | purplet | hi all. I've got a music on hold thingy... When I dial out over a zapline, sometimes when the other end puts me on hold, Asterisk starts it's own moh! Is that something I can control? |
| 20:44.22 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 20:44.58 | Defraz | Question: What kind of packet size or how does the qualify work. I am tyrhing to figureout where the numbers it gives back are coming from? |
| 20:45.00 | waKKu | Juggie http://pastebin.ca/611239 :) |
| 20:45.55 | anonymouz666 | 200 active calls |
| 20:45.56 | anonymouz666 | Verbosity is at least 3 |
| 20:46.03 | Juggie | waKKu, whats the context of the phone having the problem |
| 20:46.53 | waKKu | Juggie all context.. :/ |
| 20:47.06 | waKKu | when doing internal and external calls |
| 20:47.31 | Juggie | right, its because your dialplan isnt properly seperated |
| 20:47.40 | Juggie | you are mixing outgoing and incoming calls |
| 20:47.53 | Juggie | you do have overlaping, you have exten => _XXXXXXXX |
| 20:48.08 | Juggie | so asterisk cant assume 889 is finished, its wwaiting to see if it gets 4 more chars for that extension |
| 20:48.12 | Assid | hrmm.. im thinking of possibly getting nagios to autodial to me and let me know if the service goes down |
| 20:48.35 | holiday_42 | defraz, sip options message, i think |
| 20:48.39 | waKKu | Juggie hm.. have some simple solution to it ? |
| 20:49.00 | Juggie | waKKu, you are using extensions so you are going to pick a number to dial external |
| 20:49.07 | Juggie | commonly 8 or 9 |
| 20:49.51 | Juggie | but your dialplan isnt really setup properly at all |
| 20:50.16 | Juggie | you need to carefully seperate calls going out (either external or to another local extension) |
| 20:50.20 | Juggie | vs incoming calls from the pstn |
| 20:50.20 | waKKu | any help is appreciate :D |
| 20:50.43 | waKKu | hm.. some hint ? |
| 20:51.11 | waKKu | remove that "include => ramais" from default ? |
| 20:51.55 | Juggie | do you send and receive calls via the pstn? and then also send/receive via iax? |
| 20:52.44 | waKKu | no.. i have a single line... |
| 20:52.54 | Juggie | can it be outgoing or incoming? |
| 20:52.59 | Juggie | or is it just for outgoing calls |
| 20:53.10 | waKKu | can receive incoming . |
| 20:53.19 | waKKu | both |
| 20:53.27 | Juggie | so what you really need to do is create a context for each direction |
| 20:53.34 | Juggie | eg, [pstn-in] |
| 20:53.38 | Juggie | and [pstn-out] |
| 20:53.44 | Juggie | then make one for iax [iax-in] |
| 20:53.49 | Juggie | and [iax-out] |
| 20:54.34 | waKKu | ok.. but, how do I eliminate overlaps ? all that happen on outgoing calls |
| 20:54.39 | waKKu | or i'm wrong ? |
| 20:55.13 | Juggie | your dialplan needs to be organized... |
| 20:55.35 | Juggie | i dont really have time to redo the entire thing, but you need to split it by direction and by technology |
| 20:55.47 | Juggie | then build your pieces and include them in |
| 20:56.20 | Juggie | take a look @ www.voip-info.org |
| 20:57.14 | *** join/#asterisk PDani (n=pdani@IP-178-85.tvnetwork.hu) |
| 20:57.14 | PDani | hi |
| 20:59.36 | dandan | later |
| 21:00.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:02.09 | waKKu | Juggie thanks for all ... i'll try sth |
| 21:02.54 | waKKu | well.. howhever... have no option to reduce this "overlaping timeout" ? |
| 21:04.01 | PDani | i know not much about asterisk. I have a callcenter, with some voip streams connecting to the outside pstn network through an asterisk server. is it possible to serve the stream to third party clients, i mean, like a stream-server with encoded streams, one for each active line. and how can i tell the source ip of the voip connection i actually listen to through the asterisk-served stream? |
| 21:04.22 | Assid | err.. anyone here have a polycom voice station 100 |
| 21:05.12 | [TK]D-Fender | Assid, Looks like the lowest model produced |
| 21:05.17 | Assid | yeah |
| 21:05.24 | Assid | i need to find out how to get the ethernet ip |
| 21:05.48 | [TK]D-Fender | Analog PBX or public switched telephone <- |
| 21:05.57 | [TK]D-Fender | Where does it say anyting about IP? |
| 21:05.57 | *** join/#asterisk tako-san (n=Tako-san@S010600179a5211fe.gv.shawcable.net) |
| 21:06.03 | Assid | they plugged it in remotely and i cant figure out the dhcp ip |
| 21:06.10 | Assid | how do i configure it ? |
| 21:06.20 | _DAW | you dont |
| 21:06.23 | [TK]D-Fender | Assid, it lokos like an ANALOG PHONE, not a VOIP PHONE. |
| 21:06.44 | mocker | That seems like it should have been done.. ;) |
| 21:07.11 | purplet | hi all. I've got a music on hold thingy... When I dial out over a zapline, sometimes when the other end puts me on hold, Asterisk starts it's own moh! Is that something I can control? |
| 21:07.12 | [TK]D-Fender | mocker, fear not... your wheel will clear be rounder... |
| 21:07.30 | PDani | any comments? |
| 21:07.34 | Assid | [TK]D-Fender: nah.. its a voip one |
| 21:07.43 | [TK]D-Fender | Assid, Link me to it. |
| 21:07.52 | Assid | http://www.polycom.com/common/documents/support/sales_marketing/products/voice/voicestation_100_datasheet.pdf |
| 21:08.14 | mocker | [TK]D-Fender: I have no idea what you just said there. |
| 21:08.34 | [TK]D-Fender | Network interface |
| 21:08.34 | [TK]D-Fender | • Analog PBX or public switched telephone |
| 21:08.42 | [TK]D-Fender | Assid, right off your silly sheet! |
| 21:09.04 | _DAW | Assid: Thats an analog unit, I have an older one connected via a SPA2000 and it works nicely. |
| 21:09.06 | [TK]D-Fender | Assid, Where does it say SIP / VOIP / Etherenet ANYWHERE on there? |
| 21:09.13 | Assid | oh damn |
| 21:09.25 | [TK]D-Fender | Assid, Put. Down. The. Crack. Pipe! |
| 21:09.29 | Assid | hahaha |
| 21:09.30 | [TK]D-Fender | (c) JerJer |
| 21:09.38 | Assid | so what kind of conference phone is it ? |
| 21:09.46 | [TK]D-Fender | Assid, ANALOG! |
| 21:10.12 | [TK]D-Fender | Assid, stop now and get some sleep, I've told you like 3 times, and the data sheet does it itself. |
| 21:10.32 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 21:10.33 | NOT_guru | I think I asked this earlier but don't remember getting a response... does the zaptel 1.2.18 driver need the spinlock fix like the 1.2.17.1 driver did? |
| 21:10.50 | [TK]D-Fender | NOT_guru, that wasn't a ZAPTEL problem. |
| 21:10.52 | Assid | so its liek a spa2002 ? |
| 21:10.59 | NOT_guru | oh heh me be silly |
| 21:11.02 | NOT_guru | thanks you |
| 21:11.05 | [TK]D-Fender | Assid, No, its just a friggen stupid ANALOG PHONE. |
| 21:11.08 | Assid | oh |
| 21:11.14 | Assid | what a waste |
| 21:11.25 | Assid | they keep telling me its got an ethernet jack |
| 21:11.28 | [TK]D-Fender | Assid, No, your ability to read 2 simple words and process them is. |
| 21:11.37 | [TK]D-Fender | Assid, they = retards. |
| 21:11.41 | Assid | yeah |
| 21:11.51 | [TK]D-Fender | Assid, RJ9, RJ11, RJ45, RJ48... all the same right? |
| 21:12.12 | Assid | hehe.. sorry mate.. |
| 21:12.13 | [TK]D-Fender | Assid, read the damn sheet you even linked me and THINK |
| 21:12.27 | [TK]D-Fender | *gasp& |
| 21:12.49 | Assid | err.. is the 501's speaker that bad.. they keep complaining it sucks |
| 21:13.02 | Qwell[] | No, polycom speakers are very good |
| 21:13.29 | Assid | lemme tryu adn change some settings |
| 21:14.04 | Assid | err.. the mic he says sounds funny.. i keep telling them they have to tell me if they have issues.. for me to do anything |
| 21:14.59 | [TK]D-Fender | Assid, IP501 = great |
| 21:15.15 | Assid | yeah they kept the 501's for themselves and sent me a 301 |
| 21:15.16 | [TK]D-Fender | Assid, what is on the OTHER side of the call? |
| 21:15.36 | Assid | regular pstn most of the time.. or a cell phone |
| 21:15.43 | [TK]D-Fender | I'd still sooner use an IP 301 at my office than the Aastra 57i CT I have there now. |
| 21:15.53 | [TK]D-Fender | Assid, taht is NOT the right answer. |
| 21:16.07 | Assid | " Assid, what is on the OTHER side of the call?" ? humans? |
| 21:16.33 | [TK]D-Fender | Assid, what friggen CARD. And then what is the person on the OTHER END using? |
| 21:16.39 | *** part/#asterisk illsci (n=illsci@evil.hack3rs.org) |
| 21:16.48 | [TK]D-Fender | YES THE CARD MATTERS |
| 21:16.59 | Assid | oh.. all over sip.. outgoing is through asterlink.. incoming voicepulse. |
| 21:17.09 | [TK]D-Fender | Why is it people can't describe stuff in a simple path? |
| 21:17.29 | [TK]D-Fender | Assid, And on the the far end, what phone? |
| 21:17.52 | Assid | no clue.. |
| 21:17.59 | Jameno123 | Assid, PHONE->CARD->SERVER->PROVIDER->PROVIDER->SERVER->CARD->PHONE |
| 21:18.07 | Jameno123 | explain what each step is! |
| 21:18.29 | Assid | ip501->asterisk->asterlink->pstn |
| 21:18.32 | [TK]D-Fender | Assid, And keep in mind that you are going from SIP to analog, back to SIP introducing all sorts of echo possiblities and then transcode loss & latency. |
| 21:18.40 | [TK]D-Fender | Assid, why the hell don't them jsut peer up! |
| 21:19.18 | Assid | i try to peer as many people as i can.. like the other office they call quite often.. but not everyone on voip.. and even less on asterisk/sip |
| 21:20.02 | tzafrir_laptop | does the torisa module require ISA or EISA? |
| 21:20.37 | *** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl) |
| 21:20.49 | [TK]D-Fender | Assid, You mean... except this exact scenario that you just finished giving me.... |
| 21:21.16 | tzafrir_laptop | hmm... my kernel has no CONFIG_EISA and still that module builds OK |
| 21:21.21 | Assid | yeah.. most people they call are on pstn. and hence this scenario is the most frequent |
| 21:22.14 | [TK]D-Fender | Assid, And your path is more like : IP 501 > * > (internet, what codec?) > Asterlink > PSTN > VoicePulse (what codec?) > * > what phone? |
| 21:22.14 | nohop | hey... i was trying to download a zaptel driver, cause i've ordered a x100p card... but cvs.digium.com 's dns is broken (Host cvs.digium.com not found: 3(NXDOMAIN)), any alternatives ? |
| 21:22.32 | [TK]D-Fender | nohop, CVS has been dead for YEARS |
| 21:22.37 | [TK]D-Fender | nohop, SVN <- |
| 21:22.41 | [TK]D-Fender | nohop, www.asterisk.org |
| 21:22.43 | Assid | err.. no voicepulse.. pstn.. after that i dont know what they use |
| 21:22.44 | nohop | ahhh |
| 21:22.50 | [TK]D-Fender | nohop, and I suggest you use an FTP release, not SVN. |
| 21:22.52 | Assid | IP 501 > * > ulaw> Asterlink > PSTN |
| 21:23.00 | Assid | i dont know what the opposite end uses |
| 21:23.02 | [TK]D-Fender | Assid, yay! |
| 21:23.32 | nohop | the svn release is 'too' bleeding-edge ? :) |
| 21:23.32 | Assid | can be voip.. can be gsm |
| 21:23.35 | [TK]D-Fender | Assid, translation : You are clueless. Tell them if they want help, you need FACTS |
| 21:24.13 | Assid | voice.gain.tx. in the configs should alllow me to increase the volume gain right ? |
| 21:24.16 | Assid | for polycoms |
| 21:24.52 | fujin | Is there anything I can do with the asterisk console to force my peers to re-register? |
| 21:25.06 | fujin | I've updated sip.conf with some new options, but the phones are only set to re-register every 3600 seconds I think |
| 21:26.02 | Assid | hrmm i think i do need that sleep |
| 21:26.18 | nohop | ahhh, thanks... that was pretty damn easy via the asterisk site :) |
| 21:27.27 | [TK]D-Fender | Assid, You should avoid messing with base gains.... |
| 21:28.54 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 21:30.13 | *** join/#asterisk [[blah]asdf (n=ckwall@63.149.122.93) |
| 21:30.18 | *** part/#asterisk [[blah]asdf (n=ckwall@63.149.122.93) |
| 21:30.53 | holiday_42 | OT:does voipjet take a day or two to start working? i'm not near my * box (so trying iax soft phone with no luck) |
| 21:32.00 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:34.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:35.52 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 21:42.26 | sandorp | I'm running asterisk 1.4.5 on a 2.6GHz celeron with 512MB RAM; I'm using TDM800P to connect to analog phone lines; the remote caller sounds OK if they use very short sentences and they pause between words; however, they break up really bad when trying to have a normal conversation; is my system underpowered? do I need to configure something to fix this? I am using x-lite on my side |
| 21:44.20 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 21:44.29 | [TK]D-Fender | sandorp, describe the full path of each end of the call. |
| 21:44.44 | [TK]D-Fender | sandorp, include all phone and tdm interface models. |
| 21:44.58 | [TK]D-Fender | sandorp, And consider upgrading to 1.4.6 and verify your zaptel version. |
| 21:45.12 | [TK]D-Fender | sandorp, And finally, include your zapata.conf |
| 21:45.22 | *** join/#asterisk fiber0pti (i=fiber0pt@216.31.101.41) |
| 21:46.02 | sopo2k4 | [TK]D-Fender, do you earn alot being a cosultant for this sort of stuff? like is there a big market? |
| 21:46.43 | Mercestes | sopo2k4, A moderate market. |
| 21:46.59 | Mercestes | diversification helps however. Phone systems in general is a pretty huge market... |
| 21:47.16 | *** join/#asterisk lwh (n=lwh192@66.212.165.127.tor.pathcom.com) |
| 21:47.20 | [TK]D-Fender | sopo2k4, I earn enough to make me happy. |
| 21:47.59 | sopo2k4 | yeah, but i mean presumebly all those customer service number's with the automated voices n things are powered by asterisk? |
| 21:48.13 | [TK]D-Fender | sopo2k4, no, they are not. |
| 21:48.20 | [hC] | anyone here using a aastra w/ a CT phone attached? Im having some strange call waiting behavior |
| 21:48.25 | sopo2k4 | how are they done? |
| 21:48.25 | [TK]D-Fender | sopo2k4, there are hundreds of PBX's out there for this. |
| 21:49.10 | [TK]D-Fender | sopo2k4, take your pick. Toshiba, 3com,panasonic, nortel, Avaya, Cisco, and so on. |
| 21:49.26 | sopo2k4 | ic |
| 21:49.34 | sopo2k4 | well open source is always better. |
| 21:49.36 | [TK]D-Fender | sopo2k4, and the innumerable masses just using dialogic boards and PC's |
| 21:49.47 | [TK]D-Fender | sopo2k4, so You say. |
| 21:50.04 | sopo2k4 | yup |
| 21:50.06 | [TK]D-Fender | </sarcasm> |
| 21:50.52 | [TK]D-Fender | sopo2k4, is OpenOffice better than MS Office? No. |
| 21:51.15 | [TK]D-Fender | sopo2k4, is MySQL better than Oracle? Probably not. |
| 21:51.30 | sopo2k4 | being able to add features and fix things is a huge benefit |
| 21:51.32 | sopo2k4 | and mysql is |
| 21:51.35 | sopo2k4 | imho |
| 21:51.46 | x86 | ugh |
| 21:51.53 | x86 | nub alert ;) |
| 21:51.58 | sopo2k4 | _|_ |
| 21:52.15 | sopo2k4 | everyone to their own. |
| 21:52.55 | [TK]D-Fender | Humble huh? |
| 21:53.14 | sopo2k4 | wshats with the sarcasm? |
| 21:53.20 | [TK]D-Fender | sopo2k4, some may consider the NEED to add so much to catch up as being a LOSS rather than a "feature" |
| 21:53.25 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 21:53.40 | [TK]D-Fender | sopo2k4, Sthe sarcasm is for the blatantly open-ended "OSS = better" |
| 21:54.03 | [TK]D-Fender | sopo2k4, while OSS may be a better approach, it does not mean its a better PRODUCT. |
| 21:54.03 | sandorp | [TK]D-Fender: http://pastebin.ca/611362 |
| 21:54.11 | sopo2k4 | i never said that.... |
| 21:54.21 | [TK]D-Fender | "<sopo2k4> well open source is always better" |
| 21:54.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 21:54.40 | [TK]D-Fender | I must be blind today |
| 21:55.28 | sandorp | [TK]D-Fender: did I give you everything you asked for? |
| 21:55.53 | sopo2k4 | personaly i reckon it is "better" to have the source code aswell as the end product, i didnt mean the products are ALWAYS a better end product. |
| 21:56.38 | [TK]D-Fender | sandorp, permanenty remove all that commented crap and repaste. |
| 21:56.54 | [TK]D-Fender | sandorp, include "dmesg" as well |
| 21:56.59 | sandorp | ok, gimme a sec |
| 21:57.04 | sopo2k4 | [TK]D-Fender, Linux or Windows? |
| 21:57.28 | [TK]D-Fender | sopo2k4, try phrasing that in the form of a complete question and maybe I'll answer :) |
| 21:57.32 | mocker | Plan9 |
| 21:57.34 | mocker | er. |
| 21:57.38 | sopo2k4 | which do you use/prefer? |
| 21:57.47 | sopo2k4 | and think is a better product altogether. |
| 21:58.07 | [TK]D-Fender | sopo2k4, depends what for. |
| 21:58.22 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:58.22 | sopo2k4 | i dont know what you do? |
| 21:58.26 | sopo2k4 | you tell me. |
| 21:59.04 | [TK]D-Fender | sopo2k4, Lets say for gaming I'd probably say that no Linux setup can rival an at all sane WinXP configuration. |
| 21:59.29 | [TK]D-Fender | sopo2k4, For servers, it would depends exactly what server apps you need. |
| 21:59.44 | sopo2k4 | SSH woops remote desktop. |
| 22:00.00 | sopo2k4 | although u can probably get something similar or winxp tho. |
| 22:00.07 | sopo2k4 | for* |
| 22:00.28 | [TK]D-Fender | sopo2k4, if I were to say "flexible choice of midrange SQL server", FTP & HTTP servers, then I'd say a AMP stack on OpenBSD would probably be the strongest. |
| 22:01.08 | sopo2k4 | same. |
| 22:01.08 | [TK]D-Fender | sopo2k4, RDP = graphical. SSH = text, unless you're tunneling "X" |
| 22:01.39 | sopo2k4 | yup |
| 22:01.45 | [TK]D-Fender | I have a philosophical preference for PostgreSQL as well. |
| 22:01.56 | [TK]D-Fender | everything is a matter of perspective. |
| 22:02.55 | sopo2k4 | i've never had the pleasure to play with postgresql. |
| 22:02.56 | [TK]D-Fender | for servers I prefer *NIX because its shell background and ltoolkits lend themselves to heavily scripted integration. The fact of being low-cost (free in most cases) is a bonus. |
| 22:03.24 | sopo2k4 | i reckon its probably more reliable too. |
| 22:03.29 | sandorp | [TK]D-Fender: do you want the entire dmesg output or just asterisk related stuff? |
| 22:03.53 | [TK]D-Fender | sopo2k4, in the end to me SQL is SQL, and PG just seems to respect the rules better and has certain respectable functionality. MySQL performs faster, but for me faster != better. |
| 22:04.02 | [TK]D-Fender | sandorp, jsut dump it all |
| 22:04.09 | sopo2k4 | ;p |
| 22:05.31 | [TK]D-Fender | MySQL is a great choice for a lot of stuff too, sets up damn easy on multiple platforms, has robust Windows tools for those kind of admins and native support with interesting tools. |
| 22:05.46 | sandorp | [TK]D-Fender: http://pastebin.ca/611392 |
| 22:05.47 | [TK]D-Fender | MS-SQL has the same except ($) attached to all of them ;) |
| 22:06.01 | [TK]D-Fender | mocker, Gimme root, I'm MUCH faster ;) |
| 22:06.09 | sopo2k4 | lol;p |
| 22:06.14 | mocker | Oh, I've already messed it up. |
| 22:06.21 | mocker | Just waiting for the restart. :) |
| 22:06.35 | [TK]D-Fender | mocker, I "totaled" my server with 1 stupid command once.... |
| 22:06.46 | sandorp | rm -rf / ;) |
| 22:06.57 | [TK]D-Fender | sandorp, Worse.... CHOWN <- |
| 22:07.00 | mocker | [TK]D-Fender: First thing I did when I started this job.. rsnapshot backups of all servers I'm responsible for. ;) |
| 22:07.13 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:07.18 | mocker | Currently converting this server to ODBC voicemail. |
| 22:07.30 | mocker | So recompile asterisk for ODBC support, create ODBC connector, etc.. |
| 22:07.31 | sandorp | that's funny, I use rsnapshot as well |
| 22:07.40 | mocker | I think I have everything, but never know until the restart. |
| 22:10.06 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
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| 22:11.50 | Supaplex | how do I run asterisk in openvz with realtime priority? |
| 22:12.12 | [TK]D-Fender | sandorp, "cat /proc/interrupts " please |
| 22:13.12 | macTijn | <PROTECTED> |
| 22:13.20 | macTijn | ho. |
| 22:13.34 | mocker | macTijn: +++ATH0 |
| 22:13.35 | mocker | ;) |
| 22:13.42 | macTijn | yeah |
| 22:13.47 | macTijn | something like that :) |
| 22:13.50 | macTijn | wifi died here :) |
| 22:14.07 | sandorp | [TK]D-Fender: http://pastebin.ca/611407 |
| 22:14.43 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:15.20 | [TK]D-Fender | sandorp, Ok, everything looks clean. Your gains look psychotically high from a personal POV. I suspect that may be a facteor with your EC in there. Chance EC routines and test. Disable entirely if you must for one of them. |
| 22:15.57 | sandorp | I moved those values up/down but it did not seem to make a difference |
| 22:17.01 | sandorp | [TK]D-Fender: what do you mean by "change EC routines and test" |
| 22:17.27 | [TK]D-Fender | sandorp, that statement couldn't be much more clear. |
| 22:17.40 | [TK]D-Fender | sandorp, Swith off of HPEC and try another routine. |
| 22:17.57 | [TK]D-Fender | sandorp, See if that has an impact. Then try another, and without EC at all. |
| 22:17.58 | *** join/#asterisk enjay5150 (n=yea@74.202.4.2) |
| 22:18.36 | [TK]D-Fender | sandorp, Because at least your base setup looks fine. All thats left is EC + Gain, unless your card is jsut a flaming POS. |
| 22:19.01 | [TK]D-Fender | sandorp, flakey unit, who knows.... |
| 22:19.26 | sandorp | ok, I'm trying with ??gain = 0.0 |
| 22:20.05 | sandorp | I installed the HPEC because I was hearing a lot of static on the line; HPEC seemed to clear it up a bit |
| 22:24.59 | *** join/#asterisk knowlogik (i=knowlogi@unaffiliated/knowlogik) |
| 22:25.38 | knowlogik | can anybody help me forward an incoming call to asterisk to an outbound cellphone? do I use the Dial cmd or something else |
| 22:25.56 | shido6 | you can use the Dial command |
| 22:26.06 | shido6 | do you have a PSTN line or VoIP account? |
| 22:26.14 | knowlogik | VoIP |
| 22:26.31 | [TK]D-Fender | knowlogik, If you can dial out, you should already know what to do.... |
| 22:26.37 | knowlogik | I tried and it makes progress and answers but I do not hear |
| 22:28.43 | knowlogik | I'm just using DID_NUMBER,1,Dial(SIP/sip-gw/<number>) |
| 22:28.55 | knowlogik | is that right? |
| 22:30.27 | De_Mon | knowlogik it didn't work? |
| 22:30.33 | knowlogik | no |
| 22:30.43 | De_Mon | what happened? |
| 22:31.33 | knowlogik | it shows that it called the <number> but doesn't actually connect |
| 22:31.48 | sandorp | [TK]D-Fender: setting ??gain to 0.0 seemed to improve things somewhat; the remote caller still broke up every now and then but not on every word, like before; removing HPEC caused a really bad echo ... my voice kept echoing back and the remote caller said I sounded static-y |
| 22:32.10 | [TK]D-Fender | sandorp, ok, use MG2 or something else then |
| 22:32.37 | [TK]D-Fender | sandorp, gain + lack of EC = BAD echo. |
| 22:33.03 | De_Mon | knowlogik pastebin the verbose output of that call and lets see what asterisk really said |
| 22:36.21 | sandorp | [TK]D-Fender: my gain is currently set to 0.0; I'm reading the manual and it looks like I have to recompile zaptel to use a different echo cancellation method; is that correct? |
| 22:36.25 | [TK]D-Fender | sandorp, typically yes. |
| 22:36.45 | sandorp | bummer, I was hoping it was a config option |
| 22:38.43 | [TK]D-Fender | sandorp, Sorry..... |
| 22:39.22 | sandorp | [TK]D-Fender: MG2 appears to be the EC method compiled into my zaptel |
| 22:39.44 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:40.22 | [TK]D-Fender | sandorp, Then there may be an easier way to turn off HPEC and fall back to MG2 |
| 22:40.27 | sandorp | I moved the digium directory so that it would not load their driver; is that enough to force a switch to MG2 or do I need to recompile without the digium files present |
| 22:44.50 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) [NETSPLIT VICTIM] |
| 22:45.17 | [TK]D-Fender | sandorp, try and find out |
| 22:47.16 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:50.50 | knowlogik | De_Mon I can answer the call, but no audio is passed |
| 22:50.50 | knowlogik | weird. |
| 22:51.37 | knowlogik | thanks for the help |
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| 22:57.16 | *** topic/#asterisk by irc.freenode.net -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.21, 1.4.7 and Libpri 1.2.5, 1.4.1 releases (July 9, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 22:57.32 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.21, 1.4.7 and Libpri 1.2.5, 1.4.1 released (July 9, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 22:58.48 | [TK]D-Fender | yay, our weekly upgrade is here! |
| 22:59.03 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 22:59.05 | [TK]D-Fender | I want me new fuscia screen of death! |
| 22:59.16 | [TK]D-Fender | or perhaps sea-foam green! |
| 22:59.20 | [TK]D-Fender | go retro! |
| 22:59.55 | mountainm2k | SFSOD? |
| 23:00.13 | mountainm2k | SFGSOD? Just doesn' have the same ring as "BSOD" |
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| 23:04.02 | sandorp | it looks like you have to recompile to use a different EC; I don't get much of an echo using MG2 but the remote callers voice seems to cut out a little at the beginning of each word |
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| 23:04.27 | [TK]D-Fender | sandorp, work your gains back down |
| 23:04.36 | sandorp | they are at 0.0 right now |
| 23:04.54 | [TK]D-Fender | sandorp, ok, then slow step them around. This is VERY hit-or-miss |
| 23:04.55 | MACscr | Will asterisk run a couple sip channels ok on a dual p3 800 w/512mb ram> |
| 23:06.42 | [TK]D-Fender | MACscr, more than fine |
| 23:07.38 | MACscr | Thats what i thought, just wanted to make sure |
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| 23:22.22 | *** join/#asterisk sci_05 (n=peter@205-170-75-162.dia.static.qwest.net) |
| 23:22.31 | sci_05 | i got that dss circuit up finally |
| 23:22.49 | JT | T1? |
| 23:23.30 | sci_05 | yes its a t1, just a very basic one |
| 23:23.36 | sci_05 | no real d or b channels |
| 23:23.44 | JT | so it's a CAS T1 |
| 23:23.59 | sci_05 | no it was a regular t1 |
| 23:24.25 | sci_05 | its was a esf |
| 23:24.37 | JT | err |
| 23:24.44 | JT | it has to signal calls some how |
| 23:24.58 | JT | it's either PRI, CAS, E&M, what else...? :) |
| 23:25.07 | sci_05 | it was an e&m |
| 23:25.29 | mountainm2k | can meetme.conf come from realtime? |
| 23:25.38 | JT | ok then |
| 23:25.39 | sci_05 | all tho eveytime I talked to the tecks they said it was a dss theres no other name for it |
| 23:26.11 | JT | but not really any such thing as it being just a "regular T1", what's regular? T1 is just the line coding |
| 23:26.16 | JT | heh |
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| 23:26.23 | JT | so you couldn't get a pri? |
| 23:26.29 | sci_05 | nope |
| 23:26.39 | JT | that sucks |
| 23:26.43 | sci_05 | its in an area in colorado where quest is king and thats all that is available |
| 23:26.51 | sci_05 | yes it did, but I got it |
| 23:26.55 | sci_05 | now my problem is when I dail out, it hangs for about 10-20 sec before it completes a call |
| 23:27.08 | JT | it's times like this i'm glad i don't live in america ;) |
| 23:27.53 | mountainm2k | sci_05: what area of Colorado? |
| 23:28.06 | sci_05 | I get the consol saying "Calling G1/number" but it just hangs there, debug says notta durring that time |
| 23:28.18 | sci_05 | right now I am over in montros colorado |
| 23:28.56 | mountainm2k | sci_05: Hmmm, yeah, probably no CLEC's there... I'm just outside Denver. |
| 23:29.08 | mountainm2k | sci_05: It's possible you're going through an old CO switch |
| 23:29.16 | sci_05 | ya thats about 4 hours away, grand junction is about a 1+ away frm here |
| 23:29.24 | sci_05 | got I am glad I dont live here, I would go nuts |
| 23:29.43 | sci_05 | could it be timing on the circuit? |
| 23:29.44 | neverblue | any VOIP providers in the channel? |
| 23:29.49 | sci_05 | yes |
| 23:30.01 | sci_05 | neverblue, whats up |
| 23:30.05 | JT | i'm sure there are |
| 23:30.07 | neverblue | pm? |
| 23:30.11 | sci_05 | go for it |
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| 23:30.50 | Corydon76-work | sci_05: in the area of Grand Valley Telecom? |
| 23:30.52 | sci_05 | I did find out that faxing over a local lan will work |
| 23:31.45 | sci_05 | Corydon76-home no idea, I am not from around here. I am from chicago land area |
| 23:32.06 | Corydon76-work | sci_05: GVT is based in Grand Junction. They're one of the CLECs |
| 23:32.16 | sci_05 | ahhh |
| 23:32.17 | sci_05 | ok |
| 23:32.21 | sci_05 | didn't now that |
| 23:32.28 | sci_05 | probably not in this area tho |
| 23:32.32 | Corydon76-work | Guess what they use for their core call routing. ;-) |
| 23:33.29 | JT | ccm |
| 23:33.44 | Corydon76-work | JT: you fail |
| 23:34.07 | sci_05 | does it begun with a A |
| 23:34.21 | JT | an A |
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| 23:35.10 | Corydon76-work | It does, although their billing is different |
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| 23:37.54 | mocker | Huh, so odbc voicemail still uses the /var/spool/asterisk/voicemail/ filesystem.. |
| 23:37.59 | Hmmhesays | alright compiling zaptel is driving me insane |
| 23:38.38 | Corydon76-work | mocker: correct |
| 23:38.56 | Corydon76-work | mocker: but for temporary storage, mostly |
| 23:38.56 | Hmmhesays | has anyone compiled zaptel successfully on centos -5? |
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| 23:39.35 | tzafrir_laptop | Hmmhesays, please pastebin your errors |
| 23:40.01 | mountainm2k | ~pastebin |
| 23:40.02 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 23:40.10 | tako-san | when someone calls in to my IVR they cannot direct dial an extension immediately or they get the message "that is not a valid extension". If they wait a few more seconds however they can direct dial without any problems. is this a limitation of Asterisk or is there someway to adjust this? |
| 23:40.26 | Hmmhesays | http://www.pastebin.ca/611533 |
| 23:41.14 | ai-a | tako-san: you can expect the dtmf to follow though... hence the p (pause) tone. |
| 23:41.22 | tzafrir_laptop | Please rune 'make' rather than 'make ztdummy' |
| 23:41.36 | tzafrir_laptop | It tried using the kernel 2.4 build rules |
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| 23:41.42 | Hmmhesays | tzafrir_laptop: I have |
| 23:42.19 | tzafrir_laptop | Hmmhesays, so pastebin the output from 'make' |
| 23:43.07 | JT | tako-san: freepbx/trix? |
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| 23:43.26 | tako-san | JT: freepbx |
| 23:43.28 | tako-san | ai-a: thank you |
| 23:43.49 | tako-san | JT: oops, yes |
| 23:43.50 | ai-a | tako-san: japanese ? |
| 23:43.51 | arcanine | hi |
| 23:44.01 | tako-san | ai-a: not exactly |
| 23:44.07 | tako-san | ai-a: my wife is |
| 23:44.14 | ai-a | okay, san being 3, and tako ;) |
| 23:44.24 | ai-a | Calgary |
| 23:44.32 | JT | tako-san: yeah, i could tell you weren't running straight asterisk with that question :) |
| 23:44.35 | ai-a | wow, i almost dated a girlf from there . |
| 23:45.12 | ai-a | Indigo Moon Massage Company in Calgary. |
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| 23:45.44 | JT | eh |
| 23:45.52 | JT | that sounds dodgy |
| 23:46.03 | Hmmhesays | after I make && make install I get the module not found error when I modprobe |
| 23:46.05 | ai-a | well she said she was great as massages :) |
| 23:46.21 | ai-a | Hmmhesays: which modules ? you know updatedb and locate -i ? |
| 23:46.41 | ai-a | make should move them to the right folder, but depending on dist, it can make a wopsy. |
| 23:46.45 | ai-a | *whopse. |
| 23:47.01 | tzafrir_laptop | Hmmhesays, ls /lib/modules/`uname -r`/build/.config |
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| 23:47.54 | tzafrir_laptop | Hmmhesays, did make run successfully? |
| 23:48.08 | tako-san | ai-a: You are in Calgary? |
| 23:48.20 | Hmmhesays | yeah I see what is going on here |
| 23:48.53 | mocker | Does anyone know if there is a web voicemail frontend that supports odbc voicemail storage? |
| 23:49.45 | tako-san | JT: So is the answer basically "no I cannot change the way the direct dial works in the IVR"? |
| 23:50.08 | anonymouz666 | what the hell is ISDN Facility? |
| 23:50.16 | JT | tako-san: the answer is, i have no idea, please use asterisk, or at least share the relevant dialplan lines |
| 23:50.24 | JT | anonymouz666: in what context? |
| 23:50.39 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-40-34.lns3.syd7.internode.on.net) |
| 23:50.39 | Hmmhesays | tzafrir_laptop whats with the .config file i'm supposed to be looking for |
| 23:50.50 | tako-san | JT: Understood |
| 23:51.16 | tzafrir_laptop | Hmmhesays, I wanted to see if you get an error. And if not: what is your kernel release |
| 23:55.05 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 23:55.09 | Hmmhesays | 2.6.18-8.el5 |
| 23:55.16 | ai-a | tako-san: nope, im in Oxford, UK. |
| 23:55.19 | anonymouz666 | JT: http://www.pastebin.ca/611557 |
| 23:55.29 | anonymouz666 | I got hangup on all calls |
| 23:55.39 | anonymouz666 | I am using callfiles to originate |
| 23:56.42 | JT | anonymouz666: i don't see "isdn facility" in there |
| 23:56.43 | anonymouz666 | I just can't place no calls at all |
| 23:57.08 | anonymouz666 | look at the first line |
| 23:57.31 | JT | you really have no pastebinned a full call |
| 23:57.32 | JT | not |
| 23:58.40 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |