00:00.43 | flynux | ok, i'll have a look, thanks |
00:03.38 | Dr-Linux | CunningPike: ever you tried sphinx? |
00:03.59 | CunningPike | Dr-Linux: No |
00:04.10 | Dr-Linux | ok |
00:04.27 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
00:04.54 | Dr-Linux | CunningPike: actually i'm trying to install sphinx voice recognition with asterisk .. but not sure if anyone already done. |
00:05.36 | CunningPike | Dr-Linux: Did you search the wiki? |
00:05.55 | Dr-Linux | RoyK[uk]: thanks |
00:05.57 | Un1x | •Dr-Linux• fucker i asked u for help on msn |
00:05.59 | Un1x | and u didn't help :S |
00:06.02 | Un1x | damn gayvision |
00:06.23 | Dr-Linux | CunningPike: i didn't found any help |
00:06.43 | CunningPike | Un1x: See below |
00:06.48 | CunningPike | ~thebook |
00:06.53 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
00:06.54 | Un1x | ? |
00:07.02 | CunningPike | ~thewiki |
00:07.04 | jbot | methinks thewiki is at http://www.voip-info.org/wiki-Asterisk |
00:07.05 | Un1x | this aint, for sipra lol |
00:07.20 | CunningPike | Un1x: Sorry! Wrong person lol |
00:07.26 | Un1x | oh ok |
00:07.30 | Un1x | Cunning pike |
00:07.31 | CunningPike | Un1x: Do you have the admin guide? |
00:07.34 | Un1x | tell me soem voip providers |
00:07.35 | Un1x | i can get |
00:07.52 | CunningPike | Un1x: I found that very useful for the SPA-3000 |
00:07.58 | Un1x | no only the litrle booklet it came with has 4 pages 3 of wich are how to plug in wires lol and 1 of wich just tells u how to find out, wich ip address is being used etc |
00:08.07 | CunningPike | Un1x: As for ITSPs, I have no idea - we use a PRI |
00:08.24 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
00:08.25 | CunningPike | Un1x: You can get the admin guide from the Sipura web site - it's well worth a read |
00:08.32 | Un1x | no Cunningpike i need a provider for my Spa-3000 |
00:08.34 | Un1x | u dont know one? |
00:09.00 | CunningPike | Un1x: Um - your SPA-3000 connects asterisk to a POTS line,,,,,,, |
00:09.15 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
00:09.17 | Un1x | no |
00:09.19 | Un1x | just to the net |
00:09.21 | Un1x | no Asterisk |
00:12.28 | Un1x | man |
00:12.31 | Un1x | crap no help here |
00:12.46 | *** join/#asterisk test34- (n=test34@unaffiliated/test34) |
00:13.42 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
00:16.36 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
00:17.25 | flynux | CunningPike: ok, found out, there seems to be jumpers on the TE4XXP cards but setting is bad, fortunately it's overridable by software they give the solution there: http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html |
00:18.42 | CunningPike | OK - great |
00:18.55 | Un1x | CunningPike you dont know any SIP providers? |
00:19.08 | CunningPike | Un1x: No - we do direct to the PSTN |
00:19.47 | knarfly | Un1x: try myvoice.splitinfinity.com |
00:20.00 | Un1x | knarfly thanks |
00:20.06 | Un1x | anyone use it the spa-300 tho :P? |
00:22.42 | knarfly | Un1x: tell them user #49 sent you...they have DID's and their tech support is very * friendly. |
00:23.04 | Un1x | knarfly i wanna know that they suppora SIP meaning, Spa-3000 stuff.. |
00:23.08 | Un1x | well nvm i got my answer :P |
00:23.19 | Un1x | ok thanks knarfly wanna help me a bit on configuring the spa as well? |
00:23.50 | hads | Un1x: Try a Sipura support forum. |
00:23.58 | Un1x | ok |
00:24.01 | knarfly | Un1x: I'll try...I'm still pretty new at this but I got several systems working so far |
00:24.34 | knarfly | Un1x: Yes http://myvoice.splitinfinity.com does sip |
00:24.42 | Nivex | <PROTECTED> |
00:24.42 | Nivex | ` |
00:24.56 | Un1x | okay thanks :) |
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00:28.12 | Un1x | knarfly how long do spitfinity take to get ur account setup |
00:28.14 | Un1x | so u can use it? |
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00:30.10 | knarfly | Un1x: instant |
00:30.20 | knarfly | Un1x: instantly that is |
00:30.25 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
00:31.05 | knarfly | Un1x: I was up in minutes...if you want DID you'll need to catch them during normal biz hours...they're in San Diego, CA |
00:32.04 | Un1x | why do they take soo long for the DID? |
00:33.53 | knarfly | Un1x: Can't say but they do have a new interface that looks like you might get DID without their help...I haven't tried it but their outgoing stuff works clear as a bell. |
00:34.50 | knarfly | Un1x: I have a toll-free DID with them. I just called them during regular biz hours and they set it up. |
00:35.45 | knarfly | Un1x: They have 24/7 tech support that's been a big help. |
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00:45.24 | Un1x | knarfly |
00:45.27 | Un1x | u there |
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00:50.41 | Un1x | somone help man |
00:50.46 | Un1x | shit man |
00:50.52 | Un1x | knarfly coz of u i got charged 80$ :| |
00:51.05 | Un1x | from spitfinity but now i dont know how to use the info they gave me with, Spa-3000 |
00:51.05 | Un1x | :S |
00:53.08 | Un1x | shit thanks to knarfly im fucked now lol |
00:53.31 | riddlebox | can someone help me with this error? http://pastebin.ca/87497 |
00:53.52 | Strom_C | Un1x: what did knarfly tell you to do? |
00:54.01 | Un1x | he told me use spitfinity |
00:54.05 | Un1x | i got a account there |
00:54.24 | Un1x | they charged me 80$ lol wichi dont care about |
00:54.25 | Un1x | but now |
00:54.25 | Un1x | i dont know how to set it up with Spa-3000 |
00:54.25 | Un1x | L:S |
00:54.25 | Strom_C | does "lol" |
00:54.30 | Strom_C | actually appear on your credit card bill? |
00:55.32 | Un1x | what? |
00:55.39 | Un1x | ofc man if they charged me 80$ obviously |
00:55.45 | Un1x | thatsd 1 day of work gone to them :S |
00:55.51 | Strom_C | well you said they charged you "$80 lol" |
00:56.37 | *** join/#asterisk setti (n=sock@160-77-112.adsl.terra.cl) |
00:57.03 | Un1x | oh no |
00:57.14 | Un1x | Strom_c comon man you use spitfinity with Sipra? |
00:57.23 | Strom_C | I've never heard of spitfinity |
00:57.46 | Strom_C | but I would personally be wary of any telco with "spit" in the name |
00:58.13 | Un1x | lol |
00:58.21 | Un1x | http://splitinfinity.com/aboutus_contactinfo.html |
00:58.22 | Un1x | split sorry |
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01:01.59 | knarfly | Un1x: sorry I missed that...was in another room... let's try it again. what was your question? |
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01:02.16 | Un1x | knarfly |
01:02.24 | Un1x | i need help setting up the Spitfinity with Spa-3000 |
01:02.24 | Un1x | :S |
01:02.28 | Un1x | i dont know where what goes |
01:05.09 | *** join/#asterisk Avochelm (n=damien@203.122.248.254) |
01:06.47 | Avochelm | does anybody know how asterisk generates a uniqueid for a call? i need to be able to generate my own uniqueids. |
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01:07.52 | Docelm0 | haha ya unix timestamp |
01:08.11 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
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01:12.42 | Avochelm | ah, it is too... but there's the funny decimal point on the end |
01:13.04 | wunderkin | ummm what are you really trying to do |
01:13.16 | Avochelm | something awfull |
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01:32.16 | livinded | I know that asterisk has support for streaming a conference to icecast, but is there any support or a 3rd party script for shoutcast? |
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01:42.07 | knarfly | Avochelm: you just want to spoof your callerid? |
01:44.45 | livinded | are the police/other law enforcement starting to crack down on spoofing, i heard a few months back it was being looked into? |
01:44.56 | livinded | at least within america? |
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01:48.45 | Avochelm | knarfly, i'm copying a mysql table of call records into another table (already populated) and want to generate new uniqueid for them to prevent clashes. |
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01:51.36 | Un1x | fuck man |
01:51.38 | Un1x | fucking splitfinity sucks |
01:51.42 | Un1x | there tech support sucks too |
01:52.04 | mog | what you need un1x |
01:52.16 | Un1x | man i been trying to get help to setup my spa-3000 with them |
01:52.21 | Un1x | and they say hello |
01:52.24 | Un1x | on msn and elave for 2 hours |
01:52.31 | Un1x | :S |
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01:52.49 | russellb | ~striplsd |
01:52.55 | russellb | ~striplastdigit |
01:52.56 | jbot | striplastdigit is probably ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits. |
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01:58.35 | skraelings001 | hi |
02:00.53 | riddlebox | can someone help me with this error? http://pastebin.ca/87497 |
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02:09.13 | dlynes_laptop | Un1x: have you tried the sipura user group forums on voxilla? |
02:09.41 | dlynes_laptop | Un1x: there's a number of really good threads there, detailing how to set up a sipura 3000 to work with asterisk |
02:09.56 | Un1x | im not setting it up with asterisk |
02:10.55 | dlynes_laptop | Un1x: oh |
02:11.12 | dlynes_laptop | Un1x: what exactly are you trying to do with it, then? |
02:11.41 | dlynes_laptop | riddlebox: looks like someone wrote an asterisk module, and never finished writing it |
02:11.49 | dlynes_laptop | riddlebox: or it was written for a really old version of asterisk |
02:14.51 | riddlebox | <PROTECTED> |
02:15.10 | *** join/#asterisk MikeJ__ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
02:15.56 | dlynes_laptop | riddlebox: ummm....res_smdi doesn't come with asterisk, afaik |
02:17.43 | dlynes_laptop | riddlebox: put a noload => res_smdi.so in your modules.conf file |
02:18.11 | file | meep |
02:18.11 | riddlebox | dlynes_laptop, I deleted the /usr/lib/asterisk/modules folder, and I am going to compile from source |
02:18.12 | dlynes_laptop | i have no idea what ubuntu dapper is |
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02:18.39 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
02:18.52 | livinded | dlynes_laptop: its the next release of ubuntu |
02:19.03 | dlynes_laptop | the next release after the current one? |
02:19.13 | livinded | yes, dapper is currently in beta |
02:19.20 | dlynes_laptop | cool...so what's the current one? :P |
02:19.22 | riddlebox | what |
02:19.28 | riddlebox | dapper is the current release |
02:19.30 | livinded | breazy beaver or something |
02:19.41 | livinded | when was dapper released? |
02:19.42 | dlynes_laptop | livinded: that sounds kinky |
02:19.57 | dlynes_laptop | a breazy beaver |
02:20.00 | riddlebox | last month |
02:20.10 | dlynes_laptop | i thought that's when a girl was wearing a really short skirt |
02:20.33 | livinded | dlynes_laptop: its not really breazy beaver, its breazy something else, i always call it breazy beaver becuase i can't remember the real animal and i thought it sounded better |
02:20.34 | riddlebox | ok how do I make absolute sure that asterisk is removed from my system? |
02:20.54 | Un1x | lol sounds like you used someon elses system as a PBx.... |
02:21.02 | Un1x | and wherent allowed to and now trying to clean ur tracks :P |
02:21.25 | dlynes_laptop | for dirfile in `find / -name asterisk`; do rm -rf $dirfile; done |
02:21.37 | riddlebox | no, I have had so many problems with this machine |
02:22.05 | livinded | dlynes_laptop: its breazy badger |
02:22.15 | dlynes_laptop | i like breezy beaver better |
02:22.23 | livinded | me too |
02:22.32 | dlynes_laptop | so it's going to stay breezy beaver |
02:22.41 | dlynes_laptop | who ever heard of a breezy badger, anyways? |
02:22.49 | livinded | exactly |
02:23.01 | livinded | but i was wrong, dapper drake was released |
02:23.10 | riddlebox | just for fun I did apt-get install asterisk on my laptop, it worked fine, which is also dapper, but on my main machine it doesnt work |
02:23.25 | livinded | riddlebox: never ever use an asterisk package |
02:23.30 | dlynes_laptop | riddlebox: maybe you're grabbing an unstable package |
02:23.48 | dlynes_laptop | riddlebox: i always compile from source |
02:23.56 | dlynes_laptop | riddlebox: then i know exactly what i'm getting myself into |
02:24.23 | livinded | dlynes_laptop: its not like its even hard, you wget 4 files, untar, and typ make && make install |
02:24.48 | livinded | assuming you already got the dependencies |
02:24.54 | dlynes_laptop | livinded: ummm...it's a bit more than that..I use sangoma hardware :) |
02:24.55 | riddlebox | thats the thing, I did apt-get build-dep asterisk, to get every package I needed to, compile from source but yet there are errors |
02:25.09 | livinded | sangoma? |
02:25.17 | dlynes_laptop | yeah |
02:25.21 | dlynes_laptop | www.sangoma.com |
02:25.35 | dlynes_laptop | they make some really good pri and tdm cards |
02:25.55 | skraelings001 | how can i avoid peers from being unreachable or too lagged? |
02:25.59 | livinded | are they cheaper than digium? |
02:26.11 | dlynes_laptop | livinded: retail, around the same price |
02:26.24 | dlynes_laptop | livinded: however, sangoma doesn't screw over their distributors like digium does |
02:26.39 | dlynes_laptop | livinded: so you can get the cards cheaper wholesale than you can retail |
02:26.41 | livinded | :( i can't justify paying for a tdm card for a pbx that wont be connected to my punchdown block when i don't have a stable job |
02:27.06 | hads | dlynes_laptop: What do you mean by screwing over the distributors? |
02:27.35 | dlynes_laptop | dlynes_laptop: i've heard digium's reseller network is all fubar because they sell the same price or cheaper than their distributors |
02:27.44 | riddlebox | no I get this, after doing what dlynes_laptop said to remove asterisk, and then recompiling it, http://pastebin.ca/87614 |
02:27.50 | dlynes_laptop | i.e. they compete with their middlemen |
02:28.05 | livinded | wow that sucks |
02:28.35 | hads | Ah, yeah I guess they do compete a bit in the US, but it doesn't apply to me as I'm too far away. |
02:28.53 | dlynes_laptop | hads: ah |
02:29.04 | dlynes_laptop | riddlebox: you didn't install it properly |
02:29.10 | livinded | i'm just waiting for my digium screwdriver to come |
02:29.16 | dlynes_laptop | riddlebox: read the docs on how to install it properly |
02:29.19 | livinded | those are the coolest things ever! |
02:29.20 | hads | The stuff they sell is "retail boxed" and all the distributed stuff is OEM packaged. Not that it's a mojor difference. |
02:29.27 | dlynes_laptop | hads: if you're an end consumer, you notice no difference |
02:29.29 | riddlebox | dlynes_laptop, I untarred make sudo make install |
02:29.39 | dlynes_laptop | riddlebox: yeah, iow, you didn't install it properly |
02:29.50 | hads | dlynes_laptop: I sell gear :) |
02:29.55 | dlynes_laptop | riddlebox: you forgot make samples |
02:30.11 | livinded | dlynes_laptop: you don't need samples |
02:30.14 | riddlebox | why do I need make samples? |
02:30.22 | dlynes_laptop | livinded: you do if you deleted all your config files |
02:30.28 | livinded | riddlebox: do you want to make your own configs |
02:30.41 | riddlebox | yes |
02:30.52 | riddlebox | I have all of my .conf files backed up |
02:30.57 | dlynes_laptop | riddlebox: type ls -al /etc/asterisk |
02:30.58 | livinded | then don't make samples, but you need config files to start asterisk |
02:31.04 | *** part/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
02:31.22 | riddlebox | ahh that may be it then |
02:31.24 | dlynes_laptop | hads: i suppose screwing over is too strong of a term, though |
02:31.50 | hads | Probably, I understand where you are coming from though. |
02:31.53 | dlynes_laptop | hads: their tech support is pretty good |
02:32.08 | dlynes_laptop | hads: but I have not had very good experience with digium hardware |
02:32.12 | livinded | unfortunatly i've never got to deal with their tech support |
02:32.15 | hads | TBH, I've never talked to Digium direct. |
02:32.24 | riddlebox | that was it |
02:32.25 | dlynes_laptop | hads: you shouldn't need specialized boxes in order to run their hardware |
02:32.34 | dlynes_laptop | hads: any off-the-shelf server should work |
02:32.48 | livinded | dlynes_laptop: since when do you need specialized hardware to use a digium tdm card? |
02:32.57 | livinded | aren't they normal pci cards? |
02:32.59 | dlynes_laptop | livinded: who knows...never tried |
02:33.07 | dlynes_laptop | livinded: i got tired of all the bs with their pri cards |
02:33.10 | hads | dlynes_laptop: Huh? I know :) I sell the gear and install it. |
02:33.12 | riddlebox | wohoo I am sooo happy now |
02:33.32 | livinded | riddlebox: now go setup a miliwat and listen to it |
02:33.39 | dlynes_laptop | livinded: and so when it came time to buy tdm cards, I went straight to sangoma instead |
02:34.03 | livinded | dlynes_laptop: i'll keep that in mind when i can afford a card, i need to buy a new box first though |
02:34.06 | riddlebox | livinded, miliwat? |
02:34.19 | dlynes_laptop | livinded: besides...you cannot get hwec on a 4 port tdm card from digium |
02:34.27 | dlynes_laptop | livinded: you can from sangoma...even two port |
02:34.30 | livinded | hwec? |
02:34.32 | hads | Yeah, that's a little annoying. |
02:34.35 | dlynes_laptop | hardware echo canceller |
02:34.39 | livinded | oh |
02:34.40 | Luke-Jr | iConnectHere sucks |
02:34.48 | hads | Sangoma is nice for that. |
02:34.49 | dlynes_laptop | and sangoma's are all carrier grade from Octasic |
02:34.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
02:35.01 | livinded | ooooo awsome wont eat up my cpu using asterisk |
02:35.05 | dlynes_laptop | for me that's a double bonus |
02:35.12 | dlynes_laptop | sangoma and octasic are both Canadian companies |
02:35.27 | dlynes_laptop | I'd prefer to support Canadian business, as I'm also Canadian |
02:35.50 | hads | Fair enough. |
02:36.13 | dlynes_laptop | I'd like to support Digium because they help finance the development of asterisk too |
02:39.58 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
02:39.58 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released with a critical security fix for chan_iax2, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
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02:40.04 | livinded | i think the server is broke, thats the same one |
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02:40.08 | Un1x | i wish networks would get motre money and then theyd buy new servers and wouldn't have that much load |
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02:40.16 | dlynes_laptop | Un1x: in my case, it was my server chopping my connection |
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02:40.18 | Un1x | my network never used to split |
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02:40.32 | Un1x | same here i think |
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02:40.54 | livinded | my network never splits, but we only have maybe 120 at any given time and 4 servers |
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02:41.12 | dlynes_laptop | yeah...but in this case, the server went completely down; it wasn't a netsplit |
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02:41.35 | dlynes_laptop | it was chat.freenode.net |
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02:42.56 | livinded | i need an "i <3 netsplits" sticker for my laptop |
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02:43.18 | *** join/#asterisk RageMax (n=max@c-24-3-181-140.hsd1.pa.comcast.net) |
02:43.43 | RageMax | you guys know the site to get a free DID? |
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02:43.52 | livinded | RageMax: ipkall |
02:43.53 | RageMax | I can't seem to find the link |
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02:44.09 | RageMax | I think it was freedid or something similar |
02:44.16 | livinded | theres that one too |
02:44.36 | brut- | question: it is possible to host the asterisk box behind NAT and make it work..., right? :< |
02:45.06 | livinded | brut-:sure, just forward the ports |
02:45.09 | RageMax | brut-: yeah, if you use the IAX protocol, SIP gets kind of messy |
02:45.41 | brut- | aye, ok..., then I've got something mis-configured... thanks. :) |
02:45.42 | brut- | I was just hoping i wasn't trying to do something that's known to not work |
02:45.42 | livinded | i've never had a problem with sip behind my nat but there are problems with it |
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02:46.49 | RageMax | livinded: is there one that works with IAX? |
02:47.22 | livinded | RageMax: a free did service? not that i know of. |
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02:53.05 | dlynes_laptop | aol offers free dids, too |
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02:56.11 | dlynes_laptop | sipgate.co.uk, sipgate.de also offer free didds |
03:02.41 | danp | how much is a 4-port FXO sangoma card with echo cancellation? |
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03:04.17 | RageMax | is there one with "free did" in its name? |
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03:05.19 | dlynes_laptop | danp: retail? |
03:05.35 | rob0 | www.ipkall.com ? |
03:05.55 | dlynes_laptop | yeah.."ip kall" looks a lot like "free did" :) |
03:06.04 | RageMax | danp: those types of cards are usually around $500 |
03:06.12 | rob0 | someone mentioned ipkall up there |
03:06.26 | dlynes_laptop | RageMax: ummm...that's retail, and that's without the hwec |
03:07.31 | rob0 | ipkall is a great service. In fact I think I might hang out a virtual shingle to do business in the Seattle area. :) |
03:14.17 | danp | so they're about the same as a similar digium card? |
03:14.45 | hads | danp: Where are you located? |
03:15.00 | danp | arizona, US |
03:15.06 | RageMax | hrm |
03:15.12 | dlynes_laptop | danp: yeah, except digium doesn't have hwec on their tdm cards until you get into the tdm2400p |
03:15.39 | RageMax | rather than using FWD as the sip proxy for ipkall, can you just use a dyndns address directly to your asterisk box? |
03:15.58 | hads | I'm sure someone in here will be able to point you to some online stores in the US, that's the easiest way to get prices. |
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03:16.15 | danp | i was just looking for a ballpark |
03:16.56 | hads | danp: http://www.google.co.nz/search?hl=en&ie=UTF-8&oe=UTF-8&q=sangoma+a200+price |
03:17.02 | RageMax | like I said, ballpark is about $500 |
03:17.03 | RageMax | US |
03:17.05 | danp | cool |
03:18.39 | rob0 | RageMax: that's what I do, 'cept it's my own DNS. |
03:19.07 | RageMax | rob0: so is it setup like a normal sip extension? |
03:19.35 | rob0 | Yes, there's a HOWTO page on the Wiki. I followed it, it works. |
03:20.11 | RageMax | the voip wiki? |
03:20.27 | RageMax | voip-info |
03:20.51 | rob0 | right, I think that's what's normally known as "the" wiki here. :) |
03:21.26 | RageMax | figured |
03:21.41 | RageMax | I've been using it for almost 4 years, when asterisk was in early beta ;) |
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03:23.33 | danp | i remembered that we get our stuff through voipsupply.com...that card's about $660 |
03:23.41 | danp | 4 port FXO with echo cancellation |
03:23.55 | danp | in case anyone was curious :P |
03:25.07 | wunderkin | froogle.google.com |
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03:35.16 | shmaltz | danp, what card was 4 port fxo with echo can? |
03:35.56 | shmaltz | for around 660? |
03:36.14 | danp | the sangoma a20002d |
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03:36.45 | SarahEmm | hihi |
03:37.12 | shmaltz | danp, where are you using it? |
03:37.22 | danp | i'm not, i was just curious about it |
03:37.30 | danp | we use 4-port digium cards at work |
03:37.53 | shmaltz | well, the TDM400P from digium is something I stopped using a while ago |
03:38.05 | danp | yeah, TDM400P is what we use |
03:38.08 | shmaltz | I now use T1 card with channel banks |
03:38.33 | danp | we don't have many analog lines...we deploy mainly small offices |
03:38.37 | danp | bbiab |
03:39.14 | shmaltz | http://www.voipsupply.com/product_info.php?products_id=1339 |
03:39.26 | RageMax | you'll probably need the T1 sooner or later |
03:39.32 | shmaltz | so what do you have in small offices? |
03:40.17 | RageMax | you guys remmeber that really cheap win/fax modem that supposedly worked with asterisk with a small hack |
03:40.48 | SarahEmm | RageMax: yep. i use one here |
03:40.49 | SarahEmm | x100p clone |
03:41.07 | RageMax | is that pretty much the only one that works? |
03:41.22 | SarahEmm | err |
03:41.26 | SarahEmm | there's a bunch that work |
03:41.35 | SarahEmm | a couple intel chipsets, an ambient chipset |
03:41.58 | SarahEmm | http://www.voip-info.org/wiki/view/X100P+clone |
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03:44.31 | SarahEmm | RageMax: a lot of people have issues with x100p's with echo and other issues tho |
03:44.37 | SarahEmm | just a warning if you care about audio quality... |
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03:45.31 | file | yay relaxing |
03:45.40 | russellb | yay |
03:45.44 | *** mode/#asterisk [+o file] by russellb |
03:46.00 | file | russellb is totally like, 800km away from me |
03:46.16 | russellb | i.e. 10 feet |
03:46.26 | hads | must have a long poking device |
03:46.33 | file | :D |
03:46.55 | file | russellb: I can't tell if that's a guy or a girl, despite knowing it's a guy :( |
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03:48.04 | wunderkin | your poker device? |
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03:48.32 | file | russellb just tried to poke me |
03:48.32 | russellb | ow ... pain .... |
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03:48.47 | wunderkin | oh, russellb's poker.. |
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03:58.27 | Luke-Jr | so anyone know a good origination provider? |
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04:01.26 | mitcheloc | Luke-Jr, i'm looking for one right now as well, i need 9K minutes and it seems like nobody has targeted that sub-bulk amount =/ |
04:01.56 | mitcheloc | rm -rf /file |
04:01.59 | mitcheloc | ;) |
04:02.46 | hads | Yay, weekend time! |
04:03.33 | file | russellb: this TV is so very nice |
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04:07.52 | russellb | file: indeed |
04:08.03 | russellb | we have invaded the house of kpfleming! |
04:08.48 | file | zomg |
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04:10.21 | CunningPike | wb [TK]D-Fender |
04:10.53 | mitcheloc | haha yea he has a nice tv, are you guys watching coblert report? |
04:11.05 | file | yeah, he's gone out... just me and russellb |
04:11.18 | mitcheloc | nice, loud crickets eh? |
04:11.30 | file | quite |
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04:12.05 | mitcheloc | :) |
04:13.46 | trelane | file, quick run up the phone bill! |
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04:14.11 | [TK]D-Fender | I went *b00m* |
04:14.43 | trelane | file, call the pope! |
04:15.34 | file | no phone! |
04:15.40 | trelane | no phone? |
04:15.41 | trelane | how!? |
04:15.46 | SarahEmm | err. |
04:15.52 | file | crazy things called cellphones |
04:15.55 | SarahEmm | aren't you at a house with a bunch of VoIP people? |
04:15.55 | trelane | call the dalai lama! |
04:15.57 | SarahEmm | there's no phone? |
04:16.01 | trelane | SarahEmm, no kidding |
04:16.07 | trelane | file, gank his cellphone and call HHDL |
04:16.09 | file | no analog house phone ^_^ |
04:16.18 | trelane | voip house phone? |
04:16.22 | file | what kind of cellphone would it be if he didn't take it with him? |
04:16.24 | trelane | he's gotta have some snoms or some cisco's or even polycoms? |
04:16.41 | SarahEmm | heh :) |
04:16.49 | trelane | grandstream? |
04:16.50 | mitcheloc | file, there is no phone, look over to your left, on the round coffee table, i'm pretty sure there is |
04:16.53 | trelane | 3com? |
04:16.59 | file | it's not hooked up |
04:17.12 | mitcheloc | ah, well nevermind then |
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04:20.06 | file | sleeeeeeepy |
04:20.51 | russellb | mhm |
04:20.57 | russellb | file: this is a silly tv show |
04:21.07 | SarahEmm | heh :) |
04:21.08 | SarahEmm | oops |
04:21.52 | mitcheloc | irc d-o-r-k-s, you are in the same room! |
04:23.59 | russellb | ...or are we |
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04:32.46 | CunningPike | We're all in the same room......... |
04:32.57 | Qwell[laptop] | or are we? |
04:33.02 | CunningPike | lol |
04:33.05 | SarahEmm | lol |
04:33.16 | file | Qwell[laptop]: we're all just 800km apart |
04:33.27 | Qwell[laptop] | file: exactly |
04:33.48 | Qwell[laptop] | CunningPike: all the cool people (and file) were in the same room earlier |
04:33.59 | file | nub |
04:34.03 | Qwell[laptop] | :D |
04:34.05 | mitcheloc | dorks... |
04:34.17 | Qwell[laptop] | mitcheloc: You should've stayed an extra week :p |
04:34.23 | CunningPike | mitcheloc is just jealous....... |
04:34.32 | Qwell[laptop] | nice pseudo-meeting you, btw, heh |
04:34.41 | Qwell[laptop] | for all of 30 seconds... |
04:34.44 | mitcheloc | i know! damn |
04:35.00 | Qwell[laptop] | I'm thinking "wtf...that isn't mog..." |
04:35.25 | mitcheloc | yep, the security guard gave me a hard time, haha, i could totally see that in your face, you were like wtf why is this guy talking to me... |
04:35.36 | Qwell[laptop] | heh, why'd he give you a hard time? |
04:35.51 | file | Qwell[laptop]: because you're a known terrorist and he was talking to you... |
04:35.52 | mitcheloc | i was ignoring him while i was talking to you, and he was trying to check my boarding pass |
04:35.53 | file | or at least known nub |
04:35.55 | Qwell[laptop] | file: ahh |
04:36.01 | mitcheloc | that too |
04:36.02 | Qwell[laptop] | mitcheloc: heh |
04:36.17 | mitcheloc | you all will be at astericon in boston anyway right? |
04:36.19 | Qwell[laptop] | mitcheloc: definitely interesting timing... |
04:36.41 | mitcheloc | ** dallas |
04:36.43 | Qwell[laptop] | my flight was 20 mins late getting in too |
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04:37.08 | file | mitcheloc: dunno |
04:37.31 | mitcheloc | oh yea nice timing for sure, good guess on my part too as to picking you out |
04:37.53 | Qwell[laptop] | mitcheloc: not hard. ;) |
04:38.10 | Qwell[laptop] | There was probably something obvious that tipped you off :P |
04:38.35 | mitcheloc | well when he said ugly, i didn't know you would be *that ugly* so yes it was easy =P |
04:38.38 | Qwell[laptop] | heh |
04:38.44 | mitcheloc | (just kidding) |
04:38.54 | file | ha |
04:40.54 | Qwell[laptop] | Kevin gave you a bed? |
04:40.58 | Qwell[laptop] | lame |
04:41.05 | mitcheloc | Qwell[laptop], where are you staying? |
04:41.11 | Qwell[laptop] | mitcheloc: residence inn |
04:41.20 | mitcheloc | ouch, lame |
04:41.22 | file | very lame |
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04:41.41 | Qwell[laptop] | nah, it's a nice place, I think |
04:41.58 | mitcheloc | but you don't get to use the nice tv at kps |
04:42.03 | Qwell[laptop] | ahh |
04:42.10 | file | Qwell[laptop]: you haven't been here, have you? |
04:42.16 | Qwell[laptop] | nope |
04:42.21 | mitcheloc | somebody was there a week or two ago, and told me to try the waffle house, which i did, so i recommend you all to try it out, good food :) |
04:42.23 | file | eep |
04:42.28 | Qwell[laptop] | mitcheloc: already did |
04:42.31 | file | we did the waffle house |
04:42.32 | mitcheloc | nice! |
04:42.44 | file | i bit my tongue while eating my lunch there >.< |
04:42.45 | mitcheloc | omg did you all see kram's new car? |
04:42.53 | mitcheloc | we got stuck in the rain in it!!!!!!!! |
04:42.58 | mitcheloc | i have pictures too, i'll develop them tomorrow |
04:43.01 | file | ha |
04:43.15 | mitcheloc | it wasn't funny! |
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04:47.30 | Luke-Jr | mitcheloc: I just need something :/ |
04:47.49 | Luke-Jr | mitcheloc: any non-bulk seems to be trash |
04:48.08 | Luke-Jr | even iConnectHere, which seemed to be a well-established company |
04:48.54 | Luke-Jr | they just randomly forgot my DID # sometime recently |
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04:56.15 | battt | Hello, I am a bit confused. according to all the doccumentation I found, in order to hook asterisk up to a voip service I have to use either t1 or ISDN. Isn't there a way I can use my plain jane dsl connection? |
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05:06.15 | CunningPike | battt: Yes, if you subscribe to a service provider that connects to the PSTN, or do not wish to use the PSTN at all |
05:06.52 | copantl | any body use a2billing? |
05:06.56 | battt | what kind of hardware would I use? surely not a wildcard or t1 card, would I just use an extra nic? |
05:07.38 | CunningPike | battt: Provided what I said applies, you can use your existing NIC |
05:08.46 | battt | ah, okay, thanks for clearing that up for me. the asterisk book made me think I had to use a t1 or an ISDN connection, no ifs ands or buts. |
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05:12.03 | copantl | any body use a2billing? |
05:12.14 | CunningPike | battt: No, only for connecting to the PSTN - but you will probably want to connect to the PSTN at some point....... |
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05:53.53 | twisla | melerisme roxor. |
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05:59.40 | mosty | i have some DID's being routed to be via iax2, in my * console i see "Rejected connect attempt from <IP>, request '<DID>@incoming_voip' does not exist, but my incoming_voip context does exist. what could be wrong? |
05:59.53 | mosty | routed to me, rather |
06:00.34 | drray | I've only dealt with PRI did's |
06:00.50 | drray | but don't you need a channe= > xxxxxxxx for it? |
06:00.54 | drray | er, channel |
06:01.23 | drray | for example, only did's that I sepcify get into my dialplan the rest are rejected |
06:01.45 | mosty | where do you specify that? |
06:01.52 | drray | extyensions.conf |
06:01.57 | drray | minus the y |
06:02.17 | mosty | well i have an incoming_voip context, which simply sends all calls to incoming,s,1 |
06:02.36 | drray | exten => 1074,1,Background(/etc/asterisk/main-menu) |
06:02.51 | dlynes_laptop | mosty: [incoming_voip]\nexten => <DID>,1,dothis() |
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06:02.59 | dlynes_laptop | mosty: do you have something like that? |
06:03.02 | drray | 1074 being the last 4 digits |
06:03.08 | drray | of the DID |
06:03.15 | drray | listen to dlynes.. |
06:03.22 | dlynes_laptop | mosty: i.e. you have to specify the extension in that context |
06:03.30 | dlynes_laptop | mosty: it doesn't come in on extension 's' |
06:03.30 | mosty | dlynes_laptop: no, but i have exten => s,1,Goto(mainmenu,s,1) |
06:03.34 | mosty | oh ok |
06:04.01 | mosty | i was sure this worked the other day. i will specify the did and see what happens |
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06:05.01 | mosty | that worked- thankyou both |
06:05.13 | dlynes_laptop | np |
06:05.34 | drray | thank dlyne more:) |
06:05.37 | dlynes_laptop | mosty: you might even be able to do exten => _X.,1,... |
06:05.57 | dlynes_laptop | mosty: but I wouldn't use that approach myself, in case someone's just trying to find a way into your system to spam your customers |
06:06.08 | mosty | dlynes_laptop: that's what i did. i would do _. if only * wouldn't warn me not to |
06:06.09 | drray | isn't that insecure |
06:06.10 | drray | ? |
06:06.10 | dlynes_laptop | mosty: i.e. the voip version of spammers |
06:06.57 | mosty | dlynes_laptop: well i have the iax2 ports firewalled to only let a specific voip provider send me calls |
06:07.09 | dlynes_laptop | ah |
06:07.15 | drray | sounds like you are done then |
06:07.20 | mosty | yes |
06:07.33 | mitcheloc | mosty good idea, what if they change ips though? |
06:08.50 | mosty | mitcheloc: a friend is the admin for the provider, he will let me know if that happens |
06:09.40 | drray | chances are your users will let you know :) |
06:10.15 | mitcheloc | well, it'd be nice if there was a more reliable way to set that up |
06:10.35 | mitcheloc | dns would be more accurate? but it's not as reliable |
06:11.02 | dlynes_laptop | mitcheloc: well, i'm sure his voip provider is on a dynamic ip |
06:12.23 | drray | you could allocate the subnet |
06:14.22 | pdtmobile | anybody in here done much with FastAGI? |
06:15.50 | _Guhit | I'm trying to get asterisk setup and everything is installed and started. I can call in an I get the demo, but I can't seem to figure out how to disable the demo and start adding my own dialplan. I'm running FreeBSD and installed it via the ports. |
06:16.05 | pdtmobile | I have a very simple setup currently a hello world of FastAGI if you will... AGI(agi://127.0.0.1) |
06:16.26 | pdtmobile | the server never sees a connect but asterisk is saying everything is A OK |
06:17.03 | pdtmobile | if I telnet to the port the server acknowledges i connected, that doesn't happen when I call into the dialplan I just get a AGI Script agi://127.0.0.1 completed, returning 0 |
06:17.03 | Nugget | telnet is eeeeeeevil! |
06:17.19 | pdtmobile | i am gonna have to start saying netcat in here |
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06:20.16 | dlynes_laptop | telnet |
06:20.16 | dlynes_laptop | rules |
06:20.17 | dlynes_laptop | nugget |
06:20.17 | dlynes_laptop | drools |
06:21.59 | CunningPike | _Guhit: You need to modify extensions.conf to create your own dialplan |
06:22.36 | _Guhit | CunningPike: yeah, I just found the include => demo bit |
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06:30.25 | dlynes_laptop | CunningPike: have you gotten around to writing any agi scripts yet? |
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06:30.48 | CunningPike | dlynes_laptop: No - not yet. Are you? |
06:30.53 | dlynes_laptop | thinking about it |
06:31.01 | dlynes_laptop | was just curious how easy it was |
06:31.22 | CunningPike | dlynes_laptop: I attended a presentation at Astricon last year and it didn't seem hard |
06:31.32 | dlynes_laptop | ah |
06:34.40 | CunningPike | I'm smashing my head against SugarCRM atm |
06:35.53 | mitcheloc | don't use it? |
06:36.20 | CunningPike | mitcheloc: My head, or SugarCRM? |
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06:38.10 | CunningPike | dlynes_laptop: Got FOP working - it's quite cool |
06:38.12 | mitcheloc | SugarCRM |
06:38.35 | CunningPike | mitcheloc: It's not that bad - I'm importing data from their previous system, which is a PITA |
06:38.44 | dlynes_laptop | CunningPike: ah, you did? |
06:38.55 | dlynes_laptop | I'm having issues with it, myself |
06:39.10 | dlynes_laptop | got the layout working fine, but the status and stuff like that doesn't seem to be working |
06:39.21 | CunningPike | dlynes_laptop: Really? I've only got a barebones installation running, but it works just fine |
06:39.25 | dlynes_laptop | and the drag 'n drop seems to be not dragging and dropping the way it's supposed to |
06:39.43 | dlynes_laptop | CunningPike: can you transfer calls to voicemail, and monitor the status of extensions and lines? |
06:40.00 | CunningPike | dlynes_laptop: Yes - I haven't got that working yet - none of clicks and drags work, but the status works like lightning |
06:40.16 | mitcheloc | CunningPike, regarding fop, are you actually pleased with it as an interface? in so much as you accept it as being the best out there (and yet not as good as it could be?) |
06:40.31 | loopt | hi! somebody can tell me what is this warning and how can i fix it? |
06:40.32 | loopt | WARNING[5359]: channel.c:2221 ast_write: Thread 1092921696 Blocking 'mISDN/2-1', already blocked by thread 1095051616 in procedure ast_waitfor_nandfds |
06:41.22 | CunningPike | mitcheloc: It is, ahem, adequate - and certainly the best I've found. I tried HUDLite, but can't get it working and anyway its configuration looks way complicated |
06:41.26 | drray | FOP is not open |
06:41.36 | mitcheloc | drray: not open? |
06:41.45 | CunningPike | drray: Correct - not OSS |
06:41.55 | drray | you can't modify the flash component |
06:42.09 | CunningPike | HUDLite is a binary also...... |
06:42.13 | mitcheloc | CunningPike, i thought it was... it's just flash, tough to use |
06:42.21 | drray | plus I have too many lines for FOP to work with |
06:42.32 | drray | and flash/firefox does not play nice |
06:42.53 | mitcheloc | well if you just need your personal queue you can use snap ;) |
06:43.04 | CunningPike | What I would like to do with it is have different pages for different departments - not sure if that's possible without having multiple instances |
06:43.18 | mitcheloc | but yes i'm curious as to feedback along the lines for software like FOP, like what would you improve? |
06:43.23 | hads|home | from the FOP FAQ: "You will find the perl source to compile the swf under the ming-source directory" |
06:43.25 | dlynes_laptop | CunningPike: yeah, it's possible |
06:43.36 | dlynes_laptop | CunningPike: fop comes iwht the perl code to generate the flash applet |
06:43.44 | mitcheloc | so it is open source,nothing wrong with that then? but flash code is cryptic heh |
06:43.49 | dlynes_laptop | CunningPike: you can modify the flash applet perl generation code |
06:43.55 | CunningPike | dlynes_laptop: Ugh |
06:43.59 | dlynes_laptop | CunningPike: i think it needs to be run under windows though |
06:44.09 | CunningPike | dlynes_laptop: What does? |
06:44.17 | mitcheloc | the perl generator |
06:44.24 | CunningPike | mitcheloc: Ah, I see |
06:44.25 | dlynes_laptop | CunningPike: the perl swf generator |
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06:44.55 | dlynes_laptop | CunningPike: because you probably need adobe flash studio or whatever it's called installed |
06:45.04 | CunningPike | Well, it's certainly the best of what's out there....... |
06:45.06 | dlynes_laptop | CunningPike: so that it has the creation routines |
06:45.08 | spooky7 | good morning asterisk users |
06:45.26 | dlynes_laptop | CunningPike: however, the latest version does dhtml, too |
06:45.32 | drray | I wonder if a SNMP setup would work for that |
06:45.53 | dlynes_laptop | drray: yeah, but if cp's anything like me, he needs something a secretary can handle, not a system administrator |
06:45.54 | CunningPike | dlynes_laptop: I couldn't get that to display - might be my browser |
06:46.00 | spooky7 | need some help with asterisk and druid . Can anyone help me ? |
06:46.09 | CunningPike | dlynes_laptop: Precisely ;) |
06:46.10 | dlynes_laptop | CunningPike: are you using a non-standards compliant browser like internet explorer? |
06:46.20 | CunningPike | dlynes_laptop: Nope - Omniweb |
06:46.25 | dlynes_laptop | CunningPike: try firefox |
06:46.29 | dlynes_laptop | CunningPike: it displays it just fine |
06:46.30 | mitcheloc | dlynes_laptop, ie set the standard, everyone else just didn't want to follow it! |
06:46.45 | dlynes_laptop | mitcheloc: standards are documented |
06:46.46 | CunningPike | dlynes_laptop: I'll try it - thanks |
06:46.53 | dlynes_laptop | mitcheloc: half the crap ie does isn't documented |
06:47.18 | mitcheloc | who put the standards body in charge anyway? |
06:47.25 | mitcheloc | have you looked at msdn? |
06:47.38 | dlynes_laptop | but if that half of it actually was documented, spammers and virus writers would have even more free reign on windows machines |
06:47.44 | mitcheloc | there is also a lot not documented in FF, but the excuse is that it's open source so you can go look |
06:47.53 | hads|home | deja vu, didn't this conversation happen yesterday? |
06:47.59 | dlynes_laptop | hads: which convo? |
06:48.13 | hads|home | IE vs FF etc. |
06:48.31 | mitcheloc | hads|home, i'm just playing devil's advocate :) |
06:48.41 | CunningPike | I have to say that I'll take the IE documentation in MSDN over the pitiful Mozilla documentation any day |
06:49.16 | hads|home | :) Fair enough, I guess people voucing for IE would be the minority in an open source channel :) |
06:49.27 | mitcheloc | personally i think microsoft has been doing a really good job of cleaning up their act lately |
06:49.56 | CunningPike | But - big caveat - when I code web stuff, I always seem to have to put 'if browser==IE' stuff in my code. I also have to put a fair amount of 'if browser==firefox' too |
06:50.30 | dlynes_laptop | CunningPike: i don't put any of that stuff in |
06:50.35 | spooky7 | need some help with asterisk and druid . Can anyone help me ? |
06:50.41 | dlynes_laptop | CunningPike: i let a web jockey handle all that headachy stuff :) |
06:50.55 | dlynes_laptop | spooky7: isn't druid a disk volume manager for redhat? |
06:51.02 | CunningPike | dlynes_laptop: I am a web jockey :D |
06:51.15 | dlynes_laptop | heh |
06:51.20 | spooky7 | no it is a web interface for asterisk |
06:51.26 | x86 | hmm doesnt asterisk-sounds contain all the month names spoken out? |
06:51.28 | dlynes_laptop | ah...never heard of it |
06:51.43 | dlynes_laptop | x86: probably /usr/lib/asterisk/sounds/months/january.gsm or something |
06:52.02 | mitcheloc | if you guys are trying web interfaces try out phone call from their svn repo, it's pretty sweet |
06:52.04 | x86 | i have a months.gsm, but no months/ |
06:52.18 | dlynes_laptop | erm /var/lib i mean |
06:52.21 | dlynes_laptop | guess not though |
06:52.24 | hads|home | spooky7: Druid's a commercial thing? |
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06:52.37 | spooky7 | yes i have bought it |
06:53.01 | hads|home | Do you get support with it? |
06:53.02 | mitcheloc | spooky7, why did you pick it over freepbx or phonecall or any other one? |
06:53.32 | dlynes_laptop | mitcheloc: because it wasn't free? |
06:53.49 | spooky7 | I like the web interface of druid |
06:53.52 | mitcheloc | dlynes_laptop, if i was going for non-free i'd have taken a serious look at scopserv |
06:54.01 | dlynes_laptop | but seriously...I think i've seen another commercial offering out there that was tonnes better than freepbx |
06:54.04 | spooky7 | I have paid fro druid |
06:54.20 | mitcheloc | dlynes_laptop, scopserv i think, and phonecall is pretty good |
06:54.32 | dlynes_laptop | yeah...scopserv i think was the one i looked at |
06:55.07 | dlynes_laptop | yeah...it was scopserv |
06:55.41 | dlynes_laptop | I'm guessing scopserv is the offering packaged with Allworx' PBX? |
06:55.42 | mitcheloc | yep theirs looked very professional |
06:56.03 | dlynes_laptop | cause i see Allworx' PBX pictured on scopserv's website |
06:57.09 | x86 | dlynes_laptop: nope |
06:57.22 | x86 | dlynes_laptop: niether asterisk-sounds or asterisk-sounds-extra contain month names |
06:57.28 | spooky7 | so can anyone help me ? |
06:57.29 | x86 | dlynes_laptop: know where i might find them? |
06:57.35 | CunningPike | We looked at a few GUIs, but didn't like any of them |
06:57.48 | CunningPike | I find the conf files much easier to deal with |
06:57.49 | dlynes_laptop | no idea...maybe talk to digium to get them to get allison to record them for you? |
06:58.07 | CunningPike | x86: Talk to Allison directly |
06:58.10 | CunningPike | ~thevoice |
06:58.15 | dlynes_laptop | CunningPike: yeah, but eventually i have to look for a gui config tool or a telnet config tool or something |
06:58.32 | hads|home | I heard telnet was eeeeeeevil. |
06:58.33 | dlynes_laptop | CunningPike: she won't talk to you directly, apparently...you pay for her through digium |
06:58.37 | CunningPike | dlynes_laptop: I'm building those scripts I was telling you about - for our help desk |
06:58.50 | drray | it's cheaper to just buy voice credits from digium |
06:58.50 | dlynes_laptop | which scripts were those again? |
06:59.19 | x86 | CunningPike: eh, too much money ;) |
06:59.29 | CunningPike | dlynes_laptop: Some shell scripts that allow them to add a new phone, reset passwords, change names etc |
06:59.45 | dlynes_laptop | ah...like cgi scripts that they can activate from a web page? |
06:59.56 | CunningPike | dlynes_laptop: In a kind of ./configure style from a CLI |
07:00.08 | CunningPike | dlynes_laptop: Text based menu |
07:00.16 | dlynes_laptop | ah...and secretaries have enough gray matter to figure stuff like that out? |
07:00.28 | CunningPike | dlynes_laptop: Not secretaries - help desk |
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07:00.57 | dlynes_laptop | ok...help desk people then |
07:01.04 | dlynes_laptop | same gene pool i think |
07:01.07 | mitcheloc | i think that phone call has that |
07:01.49 | CunningPike | dlynes_laptop: lol - sometimes I wonder |
07:02.21 | dlynes_laptop | CunningPike: ever try talking to shaw tech support? |
07:03.00 | CunningPike | dlynes_laptop: Did once - needed a lobotomy afterwards |
07:03.13 | CunningPike | dlynes_laptop: They are a waste of God's good air |
07:03.17 | dlynes_laptop | heh |
07:03.35 | dlynes_laptop | they really put the tech in tech support, don't they? :) |
07:03.44 | dlynes_laptop | right up there with the sales people at Can Computers |
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07:03.56 | dlynes_laptop | erm actually |
07:04.14 | dlynes_laptop | the sales people at can computers make shaw tech support guys look like brain surgeons |
07:04.19 | CunningPike | dlynes_laptop: Their role in Shaw is to prevent anyone from actually talking to anyone in there with clue |
07:04.35 | dlynes_laptop | oh...you're running Linux? |
07:04.50 | dlynes_laptop | Well, that must be why it's not working; Linux is not a supported platform! |
07:05.14 | dlynes_laptop | or there's the other ones that don't even know what Linux is :) |
07:05.16 | drray | linux is for sucks and squares |
07:05.30 | Nugget | Linux is poo. |
07:05.31 | CunningPike | "You have a Mac? Oh, we don't support Macs. I don't think they have the Internet anyway" |
07:05.35 | CunningPike | Verbatim |
07:05.38 | dlynes_laptop | hahahahahah |
07:05.45 | dlynes_laptop | now that's hilarious :) |
07:07.09 | CunningPike | I nearly had a stroke |
07:07.53 | x86 | http://cafe.bevocal.com/libraries/audio/female1/en_us/datetime/ |
07:07.55 | x86 | w00t :) |
07:09.19 | mitcheloc | mac? internet? |
07:09.19 | CunningPike | mitcheloc: Oh, yes - since 1984, actually ;) |
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07:09.26 | mitcheloc | well even if they do windows is better |
07:09.28 | dlynes_laptop | CunningPike: or 602-555-1212 is even funnier |
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07:09.47 | dlynes_laptop | CunningPike: You're calling from Canada? Which state is that in? |
07:09.56 | CunningPike | mitcheloc: You're entitled to your opinion........ |
07:10.09 | mitcheloc | =P |
07:10.10 | CunningPike | mitcheloc: Even if you are deluded |
07:10.12 | CunningPike | ;) |
07:10.35 | CunningPike | dlynes_laptop: That's Canada, Texas, ma'am |
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07:14.37 | CunningPike | Yay - all done! |
07:15.21 | SimoAmi | hi there |
07:16.00 | SimoAmi | little quite in here |
07:16.02 | SimoAmi | ;) |
07:16.51 | *** join/#asterisk ApEtc (i=apetc@ip70-162-201-182.ph.ph.cox.net) |
07:16.55 | SimoAmi | what digium card to use for a pri or fractional pri line |
07:17.39 | dlynes_laptop | te110p, or te105p(?) |
07:17.46 | dlynes_laptop | basically a single port pri card |
07:19.18 | SimoAmi | ok, thanks |
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07:29.00 | cy3o3 | word |
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07:53.18 | chazman | hello |
07:54.03 | chazman | is anyone able to assist me here with Asterisk? |
07:57.37 | chazman | lol is anyone here gunna talk? |
08:01.22 | SimoAmi | hi again. Is a ISDN PRI line a flat rate or per usage ? |
08:03.38 | pnlarsson | ISDN PRI is just a way to get you a connection, then you can get all kind of deals with the telco. But normally a montly cost + usage |
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08:05.12 | SimoAmi | how much would verizon charge for a pri in the new york area ? |
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08:06.19 | pnlarsson | give them a call... |
08:08.36 | SimoAmi | ok, how many simultaneous calls one can receive with a fractional pri |
08:09.46 | hads | 47 |
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08:11.12 | SimoAmi | wow 47! |
08:11.50 | chazman | Does anyone know where I can go to get the cards I need for my machine to allow me to connect my incoming phone line and handsets to the Asterisk server? |
08:12.45 | SimoAmi | in your opinion, does it make sense to order a fractional pri line to satisfy that the client can dial and receive up to 6 calls (incoming, outgoing or mixed) |
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08:13.57 | SimoAmi | chazman: http://www.digiumcards.com, http://www.voipsupply.com |
08:14.44 | chazman | I am needing cheap ones, if there are any |
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08:18.43 | SimoAmi | chazman: get a Linksys SPA-3102 NA 1FXS / 1FXO Analog VoIP Gateway for $89.95 |
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08:19.01 | SimoAmi | link: http://www.voipsupply.com/product_info.php?products_id=1646 |
08:19.11 | drray | can you use that with asterisk? |
08:19.17 | chazman | Ok is it possible to connect more than one handset to it? and can they be a standard household phone? |
08:20.04 | SimoAmi | it's for one handset only, unless you use ip phones |
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08:20.43 | drray | it's not locked into vonage or some other provider? |
08:22.21 | SimoAmi | nope you can use it with asterisk |
08:22.43 | SimoAmi | it's mentioned in the Asterisk guide here: |
08:22.45 | SimoAmi | http://nerdvittles.com/index.php?p=123 |
08:23.17 | SimoAmi | note that this is a new replacement for the popular SPA-3000 |
08:24.31 | drray | I saw |
08:24.48 | drray | maybe i can start dumping IAxy |
08:24.49 | drray | s |
08:25.05 | chazman | can I put a splitter on it and use mroe than one? |
08:25.30 | chazman | or have it automatically wire my house by connecting it to an existing phone line? |
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08:28.11 | chazman | also, I was looking for a PCI card? |
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08:29.21 | SimoAmi | you could use a splitter, however it is recommended to use self powered phone terminals |
08:29.43 | MrChimpy | good morning telemonkeys |
08:30.20 | chazman | Ok. I need support for one phone line and 5 handsets. do I need special phones like the Nortel phones to work with Asterisk, or can a standard phone found in the Average house work? |
08:30.50 | SimoAmi | you have 2 choices |
08:31.36 | chazman | What are those choices? |
08:31.53 | SimoAmi | etheir invest in ip phones and wire them to an ethernet network (they're for less than $100 each) |
08:32.19 | drray | you can use a splitter, but only one phone will work at a time, and yuo have to watch out for ring equivalency rating |
08:32.21 | SimoAmi | asterisk will communicate with them through tcp/ip |
08:32.55 | chazman | ok I am trying to use my current phones that I have. They are not Digital phones, but Panasonic cordless phones that the average joe would get |
08:33.09 | SimoAmi | the other way is to get 2 tdm400 pci cards with appropriate modules in them |
08:33.46 | drray | chazman - that should work, assuming the phones don't take too much power |
08:34.02 | chazman | But how would I program them with their own extensions? |
08:34.15 | drray | to do that, you'd need a tdm card |
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08:34.21 | drray | or one magic box per phone |
08:34.32 | chazman | I have a server room, which is where all my stuff is, and I want to have easy communication with all rooms in my gome |
08:34.32 | drray | or a tcp/ip voip phone |
08:34.33 | chazman | home |
08:35.20 | chazman | Ok I looked at a TDM card, i found one for under 80 |
08:35.28 | chazman | http://www.voipsupply.com/product_info.php?products_id=290&searchid=38413 |
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08:35.51 | chazman | would I just install that onto my existing fxo or fxs card? |
08:36.18 | chazman | the card I was looking at getting is: http://www.voipsupply.com/product_info.php?products_id=1388 |
08:36.24 | drray | that's a module, for the tdm400 |
08:36.26 | chazman | That is, once Ive got the money and a job |
08:36.27 | SimoAmi | that's a module that you plug in a tdm pci card |
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08:36.41 | SimoAmi | so you need to buy the card first |
08:36.46 | SimoAmi | this is what you need |
08:36.47 | SimoAmi | http://www.voipsupply.com/product_info.php?products_id=290&searchid=38413 |
08:36.53 | SimoAmi | oops wait |
08:37.00 | SimoAmi | Digium TDM40B - (4) FXS VoIP SIP IAX H.323 Asterisk $342.90 |
08:37.00 | SimoAmi | + |
08:37.00 | SimoAmi | Digium TDM11B - (1) FXS & (1) FXO VoIP SIP IAX H.323 Asterisk |
08:37.15 | chazman | Ok what if I ordered the Nortel phones that I was looking at? They are digital i think |
08:37.59 | SimoAmi | that should work but you stil need to connect the analog phone line itself |
08:38.00 | chazman | http://products.nortel.com/go/product_content.jsp?segId=0&catId=null&parId=0&prod_id=47365&locale=en-US |
08:38.07 | chazman | that is the kind that I would get if I had to |
08:40.00 | drray | how much per phone? |
08:40.18 | chazman | MSRP 139.99 US |
08:40.18 | SimoAmi | doesn't look like an ip phone to me |
08:40.29 | chazman | They have those exact phones at my school |
08:40.35 | drray | it's an analog |
08:41.02 | chazman | ok, but I take it a school doesnt have a virtual PBX, and a physical unit that does all the work |
08:41.50 | chazman | http://www.voipsupply.com/product_info.php?cPath=95_106&products_id=1057 |
08:41.53 | chazman | what about that? |
08:42.26 | SimoAmi | if you wanna connect 5 phones as you stated earlier, and you want intercom feature, then you need a pbx |
08:42.49 | *** part/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it) |
08:42.51 | chazman | Is Asterisk not a PBX? |
08:43.02 | chazman | I have a machine running Asterisk@home |
08:43.09 | SimoAmi | well that's what I'm coming to |
08:43.37 | chazman | so then do I need an actual PBX unit? will Asterisk not work as a PBX system? |
08:44.04 | SimoAmi | so yes, look for cheap ip phones, it could be a good option for you |
08:44.16 | SimoAmi | yes |
08:44.21 | drray | asterisk can work as a pbx |
08:44.26 | SimoAmi | yes |
08:44.43 | SimoAmi | you need a pbx and asterisk is a pbx |
08:44.44 | chazman | Ok well http://www.voipsupply.com/product_info.php?cPath=95_106&products_id=1057 appears to be an IP phone. Is Asterisk a IP PBX or can it support normal phones as well? |
08:45.06 | SimoAmi | it does both |
08:45.19 | chazman | Ok so for that Nortel phone, how would I use those? |
08:45.21 | drray | I run 180+ normal phones in a hotel with asterisk |
08:45.34 | drray | for that nortel phone you'd need FXS ports |
08:45.53 | SimoAmi | to connect normal phones you need an adapter (they call it a FXS) |
08:46.14 | drray | it can either be in the function of a PCI card, or a MTA |
08:46.27 | drray | but if you are going to buy a MTA you might be better off with a good IP phone |
08:46.36 | chazman | Ok so if I get only an FXS PCI card, can I just use that to establish a small internal PBX network? At the moment I do not want to extend it to support incoming calls and outgoing calls |
08:47.00 | drray | yes, you'd need 1 fcs port per extension in asterisk |
08:47.03 | drray | fxs |
08:47.15 | chazman | Ok, and do I program the actual FXS port and not the phone? |
08:47.22 | drray | correct |
08:47.29 | SimoAmi | ok, I'm going to bed |
08:47.31 | drray | you make the fxs port behave how you want |
08:47.36 | drray | (Within reason) |
08:47.51 | SimoAmi | goodnight guys |
08:47.58 | drray | night |
08:47.58 | chazman | Well thats wierd. How would you program a FXS port? I will have some extensions here, eventually making an auto attendant answer incoming calls, |
08:48.36 | drray | if you use the fxs pci cards, you put a line in zapata.conf and zaptel.conf for it |
08:48.53 | drray | then you put a line in extensions.conf point to it say Zap/01 |
08:49.24 | chazman | also, I have a small problem with Asterisk. I test everything by using a piece of software called Express Talk. The music keeps cutting out on me. Also, can Asterisk do the programming for me? I am using Asterisk@home so theres a GUI interface |
08:53.10 | chazman | Would that do all the editing for me since thats also how I setup extensions? |
08:55.56 | drray | I don't use asterisk@home |
08:56.22 | mitcheloc | yes you do! |
08:56.34 | drray | I do no! |
08:56.37 | drray | shut your mouth! |
08:56.40 | drray | lies! |
08:57.22 | chazman | Oh, well it gives me everything I need and installs everything to start me off, including FreePBX, Some Flash thing, SugarCRM, ARI, etx |
08:57.25 | chazman | etc(*) |
08:59.26 | chazman | It adds the extensions for me, so I assume it will do all the config for me |
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09:01.12 | chazman | I just went into the config, and I can add what is called a Trunk |
09:01.23 | chazman | The default one was called Trunk ZAP/g0 |
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09:02.33 | chazman | I dont know if that is what I would do or not |
09:04.50 | chazman | http://www.freepbx.org/trac |
09:04.57 | chazman | That is the freePBX Web Config program |
09:08.13 | tzafrir_laptop | chazman, try #freepbx |
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09:09.29 | tzafrir_laptop | anyway, are you sure you're not confusing FXS and FXO? |
09:09.40 | tzafrir_laptop | ~fxsfxo |
09:09.42 | jbot | methinks fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
09:11.46 | chazman | You use a trunk to carry a call (or any number of calls) to a VSP or a device that cares about what number you send to it (eg, another Asterisk/FreePBX Machine). There are 5 types of trunks supported |
09:11.52 | chazman | Is that what I need to do then? |
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09:12.30 | tparcina | hi channel! |
09:12.46 | tzafrir_laptop | chazman, how many phones do you need? How far are the phones from the server room? |
09:13.53 | chazman | Ok, my server room is on the top floor (2 floors). One of the phones will be 5 feet away. Another will be on the other side of the room, no more than 25 feet. Another will be 50 feet away, and then another 75 feet away. Then I will have one downstairs, approx. 125 ft total |
09:14.08 | chazman | i mean on the other side of the wall, not room |
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09:15.24 | chazman | here is something I found about FreePBX |
09:15.25 | chazman | Zap trunks consist of physical hardware in your machine that uses the Zapata interface. This is configured in /etc/zaptel.conf and /etc/asterisk/zapata.conf. Documentation on these files is available on the voip-info wiki. |
09:16.54 | tzafrir_laptop | chazman, ask FreePBX questions in the channel #freepbx |
09:17.04 | tzafrir_laptop | try: /j #freepbx |
09:17.06 | chazman | nobody is responding in there |
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09:18.12 | tzafrir_laptop | I must say that I don't really understand what you're stating and what you're asking |
09:20.49 | chazman | In the topic bar it says I can get help with FreePBX here too! |
09:20.55 | chazman | oh nvm |
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09:23.48 | qdk | chazman: so you read the _entire_ topic now? |
09:26.00 | chazman | lol yeah |
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09:44.05 | tparcina | any interesting DaPrivateeriscusion ghenryoing on hadsere? |
09:44.32 | tparcina | any interesting discusion going on here? |
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09:49.34 | drray | "No matter how hot she is, someone, somewhere, is sick of her shit" |
09:50.02 | tparcina | you are talking about Paris? ;) |
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09:51.16 | Blake0ps | I want to setup a PRI at a business. If I installed the TE110P (single span T1) in the Asterisk box, would I need any other hardware besides phones (they will be SIP phones)? |
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09:52.07 | tparcina | Blake0ps, no you wan't |
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09:54.39 | kmilitzer | Hi everyone ... I have a slightly OT question. Does anyone of you know of a "clustered" Filesystem where I can store my Voicemail on, that will continue working if one Node goes down. And I don't want DRDB ... ;) |
09:55.05 | darviria | can anyone help me with a problem of sip phones not getting any audio? and no i'm not using nat |
10:03.22 | MrChimpy | why not drbd? |
10:03.48 | MrChimpy | there's some oracle thingy that's a clustered fs that they've open sourced IIRC |
10:07.07 | tzafrir_laptop | gfs? |
10:07.28 | tzafrir_laptop | not anything by oracle. More by RH |
10:08.51 | MrChimpy | i must be hallucinating ocfs then |
10:09.43 | MrChimpy | not sure if it's FOSS. not looked at it at all really, but it's what oracle punt for doing linux clusters. |
10:10.13 | MrChimpy | http://oss.oracle.com/projects/ocfs/ |
10:10.33 | MrChimpy | I think i'd trust oracle shit more than redhat shit |
10:10.38 | MrChimpy | but it's a narrow margin. |
10:13.16 | Blake0ps | thanks tparcina |
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10:19.18 | benjk | anybody here using AstLinux? |
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10:29.23 | bikola_c | hey, guys, was wondering how is it possible to make outgoin calls asterisk box but recieve incoming on my SIP phone |
10:29.37 | Blake0ps | bleh, I just wrote a 500 word email to a client who is about to seal the deal with me on this PBX |
10:30.51 | bikola_c | yeah, anyone lol |
10:31.07 | bikola_c | pretty quiet, for a packed room |
10:37.49 | RoyK[uk] | benjk: wtf is astlinux? |
10:38.56 | stoffell_h | benjk: i would love to use it, but it's not bristuffed, and i'm to lazy to do it all myself :p |
10:39.03 | stoffell_h | RoyK[uk]: put .org behind it ;) |
10:39.11 | stoffell_h | and then surf it :) |
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10:56.57 | benjk | stoffel_h, that's precisely what I want to do with AstLinux, add BRIstuff |
10:57.10 | benjk | but the damn thing is borken |
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10:57.29 | benjk | I had to set up a fake name server just so it would boot |
10:57.48 | benjk | cause it wasn't finding it's NTP server |
10:57.56 | knarfly | bikola_c: I don't know you can do that...the phone will want to register with the sip proxy or the * box....why not bring your sip traffic in thru * too? |
10:57.57 | benjk | all sorts of nonsense |
10:58.13 | benjk | and the main configuration file where you configure the thing |
10:58.19 | benjk | well, that file is volatile |
10:58.28 | benjk | reboot and all your edits disappear |
10:58.45 | benjk | and of course there is zero documentation |
10:59.00 | daysmen3 | can anyone provide some pointers regarding the setup of asterisk for 20 users using sip and pstn lines. |
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10:59.36 | knarfly | daysmen3: what you wanna know |
10:59.48 | daysmen3 | my question is regarding the asterisk server and additional components necessary to run asterisk well - |
11:00.28 | knarfly | daysmen3: 20 users will need a little more horsepower...what's under the hood? |
11:01.10 | knarfly | daysmen3: 20 users is no problem though |
11:01.11 | daysmen3 | for instance what tool would you use for monitoring asterisk or what tool would you use to look at call stats etc etc - when of the free guis at there |
11:02.04 | benjk | daysmen3: www.voip-info.org |
11:02.14 | knarfly | daysmen3: the free gui's I've seen out there are not much to write home about....the commercial ones offer more if your in beez-ness and need to track calls |
11:02.28 | MrChimpy | write yer own, you big girly-man |
11:02.31 | MrChimpy | :) |
11:02.49 | daysmen3 | wish i could but programming is not my strong point - how about astbill |
11:03.56 | drray | why not just spend some time learning "awk" |
11:04.06 | daysmen3 | ok monitoring would not be an issue and scripting for automatic recovery would be would be easy |
11:04.07 | Dr-Linux|work | what does it mean? Jul 14 02:31:03 NOTICE[24011]: chan_sip.c:6275 check_auth: stale nonce received from '4073<sip:4073@70.89.66.122>' |
11:04.43 | MrChimpy | means a pederast was trying to use your sip server for grooming kids |
11:04.57 | daysmen3 | i know awk a little im just wondering what the best setup would be - |
11:07.17 | daysmen3 | knarfly: -do you allow users to customise the dialplan or do you do that yourself |
11:10.44 | MrChimpy | what users? |
11:11.07 | MrChimpy | as in user users? the gimps on the end of the phone? |
11:11.43 | MrChimpy | i'd expect most people to run away screaming on sight of a dialplan. the ones that don't do that immediately will invariably break it for you. |
11:13.15 | daysmen3 | i know yu wouldnt let them touch extensions.conf ;-) however would you provide a webgui to a sys admin at the company to do it |
11:13.40 | daysmen3 | or get them to contract you out at mega bucks to do it yourself - |
11:13.51 | MrChimpy | ah. well, it's possible. |
11:14.00 | MrChimpy | depends how kind you're feeling :) |
11:14.31 | MrChimpy | but if you're doing that stuff it's probably best using macros and DB and not altering extensions.conf directly |
11:14.54 | MrChimpy | though I haven't done all that stuff. my asterisk systems aren't PBX type things, they're IVR |
11:15.02 | daysmen3 | ok im new to asterisk and im simply gaging best practices |
11:15.18 | MrChimpy | got the o'reilly book? |
11:15.40 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
11:15.45 | daysmen3 | yep downloaded the pdf?? |
11:15.52 | MrChimpy | it's worth a read. it's good. |
11:15.54 | daysmen3 | a few months ago |
11:15.58 | daysmen3 | OK |
11:16.03 | MrChimpy | i like my hardcopy version. it's well thumbed. |
11:16.03 | daysmen3 | ill go back over it |
11:16.44 | daysmen3 | im get my teeth into again - cool |
11:16.50 | MrChimpy | if it's a simple install asterisk@home or one of those things might do and give you a gui with it |
11:17.23 | daysmen3 | would love to do it manually incase i get into some problems that needs debugging |
11:18.56 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
11:22.21 | daysmen3 | MrChimpy: youve been great - what do you think of Asterisk2Billing - anyone |
11:23.16 | *** join/#asterisk cvv (n=cvv@212.8.35.34) |
11:33.32 | *** join/#asterisk cvv (n=cvv@212.8.35.34) |
11:39.23 | cvv | why there may be unidirection connection in local network? |
11:39.31 | cvv | protocol - sip |
11:39.42 | cvv | codec - alaw |
11:39.53 | cvv | (G711a) |
11:40.18 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
11:40.22 | cvv | sip-client: X-Lite |
11:40.26 | CMike | hi all |
11:41.08 | cvv | Hi! |
11:50.19 | tparcina | forwarding of all incoming calls, has anybody done it? |
11:51.37 | tparcina | i'm planing to done it puting data into asterisk internal DB, so extension that needs to be forwarded will have something like this in * DB - forward/ext_no : forwart_to_ext_no |
11:52.12 | af_ | I have I have a tdm400, connected to a legacy pbx with 2 fxo. hangup on thos lines is not detected. what could be? |
11:52.37 | drray | the line type does not support it? |
11:52.48 | af_ | voice is just fine |
11:52.58 | af_ | does not support hangup? |
11:53.05 | drray | sure, but the tdm can't tell the line has let go |
11:53.07 | tparcina | now, i'm havin trouble puting this data indo asterisk DB |
11:53.13 | af_ | I think is very basic requirement..... |
11:53.21 | drray | what type of lines does the PBX have? |
11:53.40 | af_ | well I did put fxs_ks signaling |
11:53.43 | drray | it could be that it is not set right |
11:53.53 | af_ | fxs_ls? |
11:53.57 | drray | I don't know |
11:54.04 | af_ | how could I detect? |
11:54.08 | drray | you need to find out what your PBX offers |
11:54.22 | af_ | I have jus the line no doc |
11:54.33 | drray | you have the internet |
11:55.06 | drray | what type/brand/model pbx are you hooking up to? |
11:55.08 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
11:55.11 | af_ | no idea |
11:55.23 | drray | maybe you should find out? |
11:55.54 | af_ | I can't even enter in the pbx room |
11:55.58 | drray | I mean, you could randomly change signalling types until it worked |
11:56.05 | drray | assuming that is the problem |
11:56.10 | af_ | oh yeah. I now how to do that |
11:56.25 | af_ | I think not any is good for this kind of link |
12:01.02 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
12:02.00 | af_ | http://kb.digium.com/entry/1/30/ |
12:02.08 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
12:06.42 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
12:11.41 | e-ddie | anyone got any ideas of how to make our * server talk the same language, both when calling to the phones directly, and when calling to the queues? |
12:12.00 | e-ddie | like, the language is set to the same language in both iax and sip.conf |
12:12.10 | e-ddie | and it only happens while in queue |
12:12.22 | e-ddie | or when you call the queue nr's |
12:13.15 | E-bola | Jul 14 16:11:18 WARNING[6363]: pbx.c:4796 ast_add_extension2: Unable to register extension 'jonas', priority 1 in 'incoming', already in use |
12:13.21 | E-bola | is this normal when reloading? |
12:13.42 | E-bola | i have 1 extension defined, if i type reload without changing anything it warns me that it already exists? |
12:14.48 | file | is priority 1 already in use in your config... |
12:15.33 | *** join/#asterisk Tili (n=liil@cm109.gamma248.maxonline.com.sg) |
12:16.23 | E-bola | for that extension? |
12:16.35 | [TK]D-Fender | E-bola : pastebin the context if you can't see it for yourself. * doesn't make that sorta stuff up.... |
12:16.49 | E-bola | well im reading the asterix book made by the community |
12:16.55 | E-bola | and the first example of a dialplan dont work for me |
12:17.07 | E-bola | i made new sip.conf and new extensions.conf |
12:18.19 | *** join/#asterisk memic (n=memic@timeoutd.org) |
12:18.33 | [TK]D-Fender | E-bola : It wasn't make by "the community", but rather by a select FEW who happen to be ACTIVE in it. |
12:19.05 | E-bola | http://pastebin.ca/88008 |
12:19.19 | MrChimpy | if everyone wrote it it'd be shakespeare written by infinite monkeys |
12:19.24 | MrChimpy | but no use for asterisk |
12:19.25 | [TK]D-Fender | E-bola : Please observe lines 8 & 9 |
12:19.36 | [TK]D-Fender | E-bola : BOTH PRIORITY 1 |
12:19.43 | [TK]D-Fender | E-bola : Didn't take 2 seconds |
12:19.47 | E-bola | doh lol |
12:19.57 | E-bola | it was so short i asumed i hadnt made errors hehe |
12:20.07 | [TK]D-Fender | Trout.. its not just for breakfast anymore! |
12:20.08 | E-bola | i read u can just use n instaid of sequential numbering? |
12:20.11 | E-bola | is that bad habbit? |
12:20.43 | [TK]D-Fender | E-bola : Depends... it lets people who don't pay attention enough as it is take even MORE for granted :/ |
12:21.07 | [TK]D-Fender | E-bola : I don't use "n", and won't for the forseeable future |
12:22.03 | E-bola | k |
12:22.30 | E-bola | hmm i got my softphones registered and the 3 line extension as pasted but correct to extension s, and with correct priorities |
12:22.41 | E-bola | i thought i could then just dial any number on my softphone and hear the audio file? |
12:22.42 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
12:23.26 | E-bola | the example is for zap channels, im using sip so maybe im missing something? |
12:25.12 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
12:25.12 | *** mode/#asterisk [+o anthm] by ChanServ |
12:25.33 | [TK]D-Fender | E-bola : SIP always dials an explicit #. If you were to LITERALLY put "s" in your dial line it should work. if you want a catch-all for NUMBERS, you should use something like _X. |
12:25.50 | [TK]D-Fender | E-bola : Go read the book again about dial patterns! |
12:26.28 | memic | anybody can tell my why ChanIsAvail doenst check if an iax2 peer is reachable? |
12:27.03 | E-bola | thanks [TK]D-fender: _X worked |
12:29.36 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
12:33.20 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
12:33.36 | [TK]D-Fender | E-bola : np. Keep in mind the book is a guide to the concepts of * and not to be hand typed line for line. |
12:34.09 | [TK]D-Fender | E-bola. read up on the different line techs and the methods you would use to identify and route calls coming from them. |
12:38.37 | E-bola | well i found out i had to read something heavy |
12:38.52 | E-bola | cuz after spending 3 hours looking at stuff it was sill overwhelming |
12:39.20 | E-bola | im quote supprised at how complex * is, its more complex than both apache, samba and nagios to get an overview of |
12:39.35 | E-bola | maybe it would have helped if i knew the least about telephony hehe |
12:39.40 | mut | ^^^ |
12:39.54 | mut | holy crap i am pooped |
12:40.00 | mut | i finally got to move into my house last night |
12:40.09 | mut | so i was moving from 2 til 11pm last night |
12:40.20 | *** join/#asterisk ariel_ (n=Ariel@70.46.87.158) |
12:40.23 | mut | didn't get much sleept cause my legs kept cramping up and woke up at 5 for work |
12:41.55 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
12:42.07 | *** join/#asterisk abatista (n=Ariel@70.46.87.154) |
12:43.38 | ariel_ | morning everyone |
12:45.46 | tparcina | morining ariel_ |
12:45.50 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
12:46.04 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.212) |
12:46.12 | Kernel_core | hi all |
12:46.14 | tparcina | can anybody explain this to me - http://pastebin.ca/88028 - why this doesn't get into * db? |
12:46.50 | *** join/#asterisk abatista (n=Ariel@70.46.87.154) |
12:47.16 | Kernel_core | which version of SPANDSP ( soft-switch.org ) Fax solution , is compatible with Asterisk 1.2.9 ?! |
12:47.35 | Luke-Jr | so anyone know a good origination provider? |
12:47.59 | Kernel_core | Luke-Jr: I know |
12:48.33 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
12:48.38 | Luke-Jr | Kernel_core: who? |
12:48.50 | tparcina | anybody knows the reason why some data doesn't get into asterisk db? |
12:48.54 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
12:49.42 | cvv | evening ariel_ |
12:51.05 | ariel_ | damm network is giving me shit today... |
12:51.26 | tzafrir_laptop | Kernel_core, latest 0.2 |
12:51.35 | tzafrir_laptop | 0.2.26pre |
12:51.45 | tzafrir_laptop | 0.0.2, that is |
12:52.23 | Kernel_core | tzafrir: I just compiled spandsp-0.0.3pre22.tgz and copied apps_rxfax.c and apps_txfax.c to asterisk |
12:52.31 | Kernel_core | when I compile I get this error : |
12:52.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:54.13 | Luke-Jr | so anyone know a good origination provider in the US? |
12:54.35 | tparcina | can I put variable to asterisk DB? I mean Set(DB(forward/${CALLERID(number)})=${FORWARD}) |
12:55.01 | Kernel_core | tzafrir: I get this error http://pastebin.ca/index.php |
12:55.18 | tparcina | this ${CALLERID(number)} i know it works for sure, but is my problem because of second variable? - =${FORWARD} |
12:56.06 | Qwell[laptop] | tparcina: will work fine |
12:56.38 | ariel_ | Luke-Jr, that is a loaded question. But I use Voicepulse.com and Race.com as my primary providers for customers. |
12:57.16 | Kernel_core | tzafrir_laptop : I get this error http://pastebin.ca/88035 ops |
12:58.09 | tparcina | Qwell: well i have some problem and i don't know whay it doesnt work. |
12:58.32 | memic | anybody knows why ChanIsAvail(IAX2/ast-int:xxxx@ast-rgbg|js) is not working? this gives me "IAX2/ast-int:test1234@ast-rgbg|js") for the channel to check |
12:58.43 | Qwell[laptop] | tparcina: set ${DB(blah)} |
12:59.13 | memic | but should be IAX2/ast-int:test1234@ast-rgbg only |
12:59.15 | Qwell[laptop] | wait, no :p |
12:59.16 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
12:59.39 | memic | the |js is the option for chanisavail |
13:00.52 | memic | ideas? |
13:01.34 | Qwell[laptop] | memic: Show the exact line you're using |
13:01.49 | memic | exten => _9X.,2,ChanIsAvail(IAX2/ast-int:test1234@ast-rgbg|js) |
13:02.39 | memic | but |js is not recognized as option for chanisavail |
13:02.42 | Qwell[laptop] | And the line in the CLI says? |
13:03.30 | memic | -- Executing ChanIsAvail("IAX2/memic@memic/3", "IAX2/ast-int:test1234@ast-rgbg|js") in new stack |
13:03.38 | Qwell[laptop] | So what is the problem? |
13:03.49 | memic | Jul 14 15:03:06 WARNING[12477]: chan_iax2.c:2215 create_addr: No such host: ast-rgbg|js |
13:03.51 | memic | thats |
13:04.08 | Qwell[laptop] | That's a little better |
13:04.14 | memic | host should not contain options for chanisavail |
13:04.23 | Qwell[laptop] | That's interesting |
13:04.46 | memic | yea? %) |
13:05.04 | Qwell[laptop] | what version of *? |
13:05.04 | Luke-Jr | ariel_: do they offer origination? |
13:05.12 | Luke-Jr | (Race, that is) |
13:05.16 | ariel_ | yes |
13:05.23 | Dr-Linux|work | i have installed Sphinx3 |
13:05.30 | memic | Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by root@asterisk on a i686 running Linux |
13:05.34 | Dr-Linux|work | anybody knows Sphinx? |
13:05.52 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
13:05.57 | Qwell[laptop] | memic: and does `show application chanisavail` show s and/or j as valid options in 1.0? |
13:06.02 | Qwell[laptop] | I'm guessing no |
13:06.12 | *** join/#asterisk m4rkl4r (n=markp@66.129.95.30) |
13:06.14 | memic | hu you are asking question :P |
13:06.22 | memic | mh |
13:06.38 | memic | you could be right, i should upgrade * |
13:06.45 | Qwell[laptop] | Yes you should |
13:06.51 | Dr-Linux|work | Qwell[laptop]: ever you use Sphinx voice recognition system? |
13:07.02 | Qwell[laptop] | Dr-Linux|work: yes, but...no, I can't help with it |
13:07.06 | Luke-Jr | ariel_: any idea where I can find info on it? their main page only seems to have retail services |
13:07.24 | *** mode/#asterisk [+o Qwell[laptop]] by russellb |
13:07.28 | Qwell[laptop] | ! |
13:07.44 | russellb | :) |
13:07.44 | ariel_ | give them a call. Ask for Carlos the also have a var user section. |
13:07.44 | Qwell[laptop] | russellb: I have to disconnect in a minute though :p |
13:07.45 | Dr-Linux|work | Qwell[laptop]: i just wanna know, if sphinx3 works |
13:07.58 | russellb | Qwell[laptop]: lame |
13:08.07 | file | works is relative |
13:08.10 | *** mode/#asterisk [+o Corydon-w] by russellb |
13:08.19 | Luke-Jr | ariel_: var user? O.o |
13:08.38 | ariel_ | they call it partners |
13:08.48 | Dr-Linux|work | file: why i can't find any document for Sphinx? |
13:09.06 | Dr-Linux|work | hhm.. |
13:09.15 | Dr-Linux|work | looks like i asked wrong question :S |
13:09.16 | file | Dr-Linux|work: because nobody has written one? maybe they gave up? |
13:09.35 | *** join/#asterisk franekstein (n=coast@coasta.ca) |
13:09.47 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.92.Dial1.SanJose1.Level3.net) |
13:10.00 | [TK]D-Fender | tparcina : "db" is a function and is CASE-SENSITIVE. It must be in lower-case. That is your problem. |
13:10.05 | Dr-Linux|work | file: file i just wanna make sure if it works just fine with asteirsk, then i can do some effort. but if it is not good, then i should not waste my time. |
13:10.26 | Qwell[laptop] | [tkYou smoking crack? :p |
13:10.45 | file | it's not a commercial product... so it won't be *that* good |
13:10.50 | file | but apparently it works semi-ok |
13:10.50 | Qwell[laptop] | tparcina: I question his sanity...DB is upper case |
13:11.07 | file | and I do believe there is a guide out there somewhere |
13:11.17 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.92.Dial1.SanJose1.Level3.net) |
13:11.59 | franekstein | hey guys I have just done some major version jumping :( and having some voicemail issues are there any updated voicemail docs ? |
13:12.47 | franekstein | I am still using static config will go realtime at a later date |
13:12.47 | *** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de) |
13:13.16 | franekstein | but for now just need to make my existing entries work again |
13:15.33 | franekstein | I have tried may different combonations such as exten => XXXXXXXXX,3,Voicemail,uXXXXXXXX Voicemail(XXXXXXXX@context) |
13:15.43 | franekstein | is there a new proper format |
13:16.16 | *** join/#asterisk CleanerX (n=nix@p54A38C2B.dip0.t-ipconnect.de) |
13:18.28 | russellb | franekstein: you probably need to add searchcontexts=yes in the [general] section of voicemail.conf |
13:18.46 | Qwell[laptop] | or use the proper context.. |
13:18.52 | russellb | Qwell[laptop]: or that |
13:18.58 | russellb | i was going for the quick solution. |
13:19.02 | Qwell[laptop] | heh |
13:19.28 | *** join/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it) |
13:19.54 | franekstein | is searchcontexts=yes new ? |
13:20.47 | *** join/#asterisk Laerte (n=bho@217.221.36.10) |
13:21.27 | franekstein | well ading searchcontexts=yes worked |
13:21.29 | *** part/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it) |
13:21.34 | franekstein | adding even |
13:23.17 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
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13:25.24 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:28.10 | *** join/#asterisk jetaway2009 (n=asd@60.50.25.201) |
13:29.49 | jetaway2009 | hihi |
13:30.08 | *** join/#asterisk clive- (n=pirch@dsl-145-58-35.telkomadsl.co.za) |
13:30.20 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
13:30.48 | clive- | does anyone have any pointers for me how to create a symbolic link to a directory (as opposed to a file) |
13:31.02 | *** join/#asterisk veepster_ (n=veepster@67.130.38.2) |
13:32.31 | CleanerX | clive-, exactly the same |
13:33.38 | clive- | Cleaner hi, I keep creating the directory which doesnt opoint to the required directory, but to one level below |
13:34.10 | CleanerX | ln -s /home/sourcedir /home/targetdir |
13:34.20 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
13:34.32 | clive- | This is what I typed: ln -s /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/ ./linux |
13:34.35 | CleanerX | ls -l targetdir give you contents of source dir |
13:34.49 | CleanerX | +s |
13:37.15 | clive- | bingo,,,thanks Cleaner |
13:38.00 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
13:38.00 | *** mode/#asterisk [+o kram] by ChanServ |
13:40.14 | clive- | hi Kram, greeting from south africa |
13:41.00 | clive- | X-gen...hey, you tsotsi |
13:41.16 | X-Gen | clive-, i think kram is a bot, never seen IT speak in here |
13:41.20 | X-Gen | see |
13:41.53 | clive- | kram is the man, he is alive and kicking:) |
13:42.17 | *** join/#asterisk justlee7 (n=jirc@65.68.38.57) |
13:42.28 | *** join/#asterisk oej (n=oej@38.115.133.12) |
13:42.45 | justlee7 | does anyone here know the current address of Asterisk's CVS server? |
13:43.15 | clive- | justlee does CVS still work ? |
13:43.52 | nortex | justlee7, I think it has been replaced by the svn server |
13:44.07 | justlee7 | ah, that's why i get a host unknown |
13:44.17 | justlee7 | bah |
13:44.21 | justlee7 | ok thanks guys |
13:44.51 | Assid | woohoooo |
13:44.56 | Assid | wassup peeps |
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13:45.27 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
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13:48.53 | *** join/#asterisk JffMRIII (n=JffMRIII@c-67-167-202-60.hsd1.il.comcast.net) |
13:49.07 | JffMRIII | hello all |
13:49.22 | JffMRIII | can anyone help with a cisco upgrade 7960 |
13:49.44 | JffMRIII | currently on default firmware 5.0 (3.0) |
13:52.29 | Luke-Jr | advise: avoid iConnectHere at all costs |
13:52.57 | Assid | Luke-Jr: why whats up |
13:53.53 | Assid | what happened with them? |
13:54.09 | Luke-Jr | Assid: well, first a few months ago their VoIP was broken and would only forward my calls |
13:54.26 | Luke-Jr | now they've deleted my number (without telling me at all) because I "haven't paid since April" |
13:54.33 | Assid | you had an incoming line with them? |
13:54.38 | Luke-Jr | yet my bank statement shows the charges and all |
13:54.39 | Luke-Jr | yes |
13:54.44 | Assid | hrmm |
13:54.54 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
13:55.00 | Luke-Jr | oh, and it appears their website no longer functions without Internet Explorer |
13:55.12 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
13:55.12 | Luke-Jr | *and* their chat support doesn't even work with XP |
13:55.19 | Assid | hhahaaha.. well.. in their defence .. they still love ie |
13:55.20 | Luke-Jr | so I had to find a neighbor running 98 |
13:55.23 | ph|ber | anyone run into problems installing a phone on astgui?? |
13:55.27 | Assid | you kidding me |
13:55.34 | Assid | you need 98 to run their chat? |
13:55.59 | [TK]D-Fender | Luke-Jr : You forgot to mention that its also on on Tuesday nights it its raining ;) |
13:56.01 | Assid | okay. you win.. not toucing them.. |
13:56.19 | Assid | [TK]D-Fender ? |
13:56.20 | Qwell[laptop] | Maybe it's just you |
13:56.28 | Assid | yeah maybe they dont like you |
13:56.35 | Luke-Jr | Assid: yep |
13:57.00 | Luke-Jr | now their support guy is trying to say $3 of outgoing calls is paid for by two $10 charges over 2 months |
13:57.02 | Assid | [TK]D-Fender: told them about the TDM.. will let you know what they say |
13:57.49 | Assid | hrmm.. anyones sipdiscount accepting dtmf? |
13:58.48 | Assid | err.. anyone got an ivr i can test this on? |
13:59.31 | nortex | JffMRIII, Can you explain the problem you are having while upgrading? |
13:59.46 | *** join/#asterisk RoyK[uk] (n=roy@83.105.70.179) |
14:00.35 | *** part/#asterisk tparcina (n=tparcina@lns02-1906.dsl.iskon.hr) |
14:00.43 | *** join/#asterisk snowy_owl (i=0@200.218.196.2) |
14:00.57 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
14:01.26 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
14:01.44 | GerbilWrk | Can anyone recomment the best way to rotate the Master.csv file? |
14:01.58 | Assid | crontab |
14:02.10 | snowy_owl | hi fellows. Im using grandstreams devices (HT 486) to make calls, OpenSER and Asterisk like a rtp proxy. Im receiving this message: chan_sip.c:2530 sip_write: Asked to transmit frame type 256, while native formats is 1 (read/write = 256/256) |
14:02.15 | Assid | [TK]D-Fender: got a number i could check ivr on? |
14:02.23 | snowy_owl | Is this a problem? |
14:02.34 | Assid | snowy_owl: codec issues from what i can see |
14:02.55 | Assid | 256 is 729 |
14:03.00 | snowy_owl | im using 729 |
14:03.09 | Assid | right |
14:03.18 | Assid | but your input is on 723 |
14:03.55 | Assid | either edit your device to use 729 or .. transcode |
14:04.55 | snowy_owl | hummm... so, the HT is using 723 and asterisk is 'translating' it to 729 to send to carrier |
14:05.16 | Assid | i guess |
14:05.25 | Assid | whatever your input is .. is on 723 |
14:05.27 | *** join/#asterisk DuRaZNo (n=durazno@201.230.129.212) |
14:05.32 | DuRaZNo | slds |
14:05.40 | Assid | slds?!?!?! |
14:05.48 | Assid | so long dumb suckers? |
14:06.32 | nortex | super long dial signal ??? |
14:06.37 | DuRaZNo | sorry |
14:06.53 | snowy_owl | i'll see that |
14:06.57 | snowy_owl | thanks Assid |
14:07.03 | DuRaZNo | i speak spanish and 'slds' means something like say hi |
14:07.14 | DuRaZNo | heh :D |
14:07.27 | nortex | Well then slds to you too. |
14:07.28 | *** join/#asterisk MikeJ__ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
14:07.30 | DuRaZNo | so, i think my english is not very good |
14:07.40 | DuRaZNo | haha hi there |
14:08.33 | *** join/#asterisk trig (n=jb@xob.neospire.net) |
14:08.58 | RoyK[uk] | afternoon |
14:09.26 | mog | morning |
14:09.27 | E-bola | ul 14 18:07:13 WARNING[6802]: pbx.c:1700 pbx_extension_helper: No application 'MeetMe' for extension (internal, 600, 1) |
14:09.27 | E-bola | <PROTECTED> |
14:09.32 | E-bola | why wont meetme work? |
14:09.52 | ph|ber | E-bola: an id 10 t problem? |
14:10.12 | E-bola | mmm sory whats that? |
14:10.24 | E-bola | i simply put conf => 600 in meetme.conf |
14:10.36 | E-bola | and made an extension to point to it |
14:10.39 | Qwell[laptop] | E-bola: Do you have zaptel installed? |
14:10.46 | drray | or ztdummy |
14:10.47 | E-bola | nope, purely sip |
14:10.54 | Qwell[laptop] | E-bola: You need zap |
14:10.57 | E-bola | ahh ok |
14:11.43 | *** part/#asterisk trig (n=jb@xob.neospire.net) |
14:11.49 | hmmhesays | nice name |
14:12.33 | nortex | I have a asterisk server setup as a gateway to my PR?I and then IAX trunks to my call servers. Does anyone know of an easy way to combine the CDR of the asterisk servers so I can show what caller went out what ZAP channel? |
14:12.49 | E-bola | can i get conference functionality without using the zaptel stuff? |
14:12.55 | [TK]D-Fender | ph|ber : My users have a lot of id10 t problems here too.... |
14:12.57 | nortex | id 10 t, I have not seen that in years. Very true though. |
14:13.01 | E-bola | i'd prefer not to fiddle with kernel modules atm if i can help it |
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14:13.10 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:15.04 | [TK]D-Fender | E-bola : You don't need to mess with the kernel. |
14:15.48 | [TK]D-Fender | E-bola : If you're on 2.6 you can just use the RTC timer and you're set. If not, you will need a UCHI USB interface. |
14:16.26 | E-bola | i am on 2.6 |
14:16.45 | E-bola | got a page describing how to use the rtc timer, or ? |
14:17.29 | fndude | My provider tells me to use two accounts on asterisk I must use 'from user' I tried to add 'fromuser=mynumber' to the sip.conf of the trunk, still a pw error, could somebody help me understand where this authentication is going wrong? |
14:18.14 | DuRaZNo | I know there are hardware voip solutions offered by Cisco, Avaya, Panasonic, etc... Do you know how much could I pay for those solutions? |
14:18.23 | Qwell[laptop] | DuRaZNo: a lot |
14:18.55 | [TK]D-Fender | E-bola : no need, it should choose it automatically. just download zaptel, enable ZTDUMMY in it, compile & install it, then recompile *. |
14:19.02 | *** join/#asterisk Skarmeth (n=Skarmeth@201008240231.user.veloxzone.com.br) |
14:19.04 | Skarmeth | hi all |
14:19.23 | DuRaZNo | well, i know they are expensive... |
14:19.27 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
14:19.50 | E-bola | tk: thanks |
14:19.51 | DuRaZNo | but I don't if that means $1000, $10000 or how much |
14:20.32 | Skarmeth | anybody using polycom soundpoint ip 301 and sip 1.6.6, bootrom 3.1.3 ? I have updated two phones to last sip and bootrom version, and the phones show's a icon (line icon) as a phone off-hook and like a RJ45 plug, I cant make or receive calls |
14:20.59 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
14:20.59 | Skarmeth | just updating the bootrom, works, but if I update sip software, this problem comes |
14:21.47 | nortex | DuRaZNo, When we got ready to deploy our first sites system the Cisco equivilant was around 56,000 compared to the 26,000 asterisk setup, phones and all. |
14:22.26 | *** join/#asterisk viler (i=1000@200.114.70.228) |
14:22.28 | E-bola | When ur register attemps with a voip provider times out whats the most likely reason? |
14:22.43 | E-bola | internet works, i can resolve their hostname and ive doublechecked hostname and password |
14:23.17 | DuRaZNo | 56000? omg |
14:23.46 | afrosheen | yeah Cisco charges the big bucks |
14:23.50 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
14:24.07 | mut | cisco is usually rock solid tho |
14:24.09 | DuRaZNo | and they use propietary codecs and protocols right? |
14:24.09 | nortex | DuRaZNo, And honestly, Asterisk packed more features. |
14:24.28 | [TK]D-Fender | Skarmeth : Shat ver were you on before? |
14:24.46 | E-bola | I have an odd request... Do anybody have a place i can register with to test my system? |
14:24.49 | nortex | Cisco is switching to SIP in their lastest Call Managers and the phones already support it. |
14:25.01 | E-bola | The VOIP provider i signed up with times out, so im wondering if the problem is on my part |
14:25.07 | DuRaZNo | I'm reading about hardware solutions, and a lot of people say that they are limited, and I would need to pay more to get more specific services |
14:25.16 | afrosheen | that's true |
14:25.24 | nortex | DuRaZNo, Yup. |
14:25.38 | mut | anyone know if the t3 cards will be * supported anytime soon? |
14:26.03 | nortex | E-bola, You might try http://www.iaxtel.com/ |
14:27.59 | nortex | DuRaZNo, An example of that is we ask cisco about voicemail to email, pretty simple in my mind since the voicemail is on a server. The cost to add that feature for a 100 users doubled cost of the entire voicemail server. server |
14:27.59 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
14:28.44 | Assid | iaxtel? |
14:28.49 | Assid | hrmm |
14:28.57 | Assid | which one was it which had issues? iaxtel or teliax? |
14:29.15 | DuRaZNo | i'm writing a lot of information abou software and hardware solutions to open my boss's eyes |
14:29.33 | DuRaZNo | thanks everybody |
14:29.33 | nortex | iaxtel is the digium test network 700 and toll free access only. |
14:29.56 | nortex | DuRaZNo, No problem and good luck :) |
14:31.55 | *** join/#asterisk jbroome (n=jbroome@204.16.138.8) |
14:32.17 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
14:32.35 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
14:33.46 | Skarmeth | [TK]D-Fender, It was 1.6.3 (SIP) and 3.1.3 (BootROM) |
14:34.04 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
14:34.08 | Skarmeth | with SIP 1.6.6 I can't receive or make calls |
14:34.31 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
14:34.59 | Assid | Skarmeth: does your CLI show your call being placed? |
14:35.02 | *** join/#asterisk daysmen3 (n=primus@host86-141-242-160.range86-141.btcentralplus.com) |
14:35.46 | E-bola | nortex: great tip, just signed up |
14:36.28 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
14:38.50 | E-bola | Do you have something similar with sip? |
14:39.03 | E-bola | just somewhere i can register with my asterix server to test it? |
14:39.08 | *** join/#asterisk Egonis (n=chultay@207.245.14.10) |
14:39.32 | Egonis | I am trying to setup a multiple context zapata.conf where channels 1-3 are context1, and channel 4 is context2. How do I do this? is there a sample? I can't find one |
14:40.25 | [TK]D-Fender | Skarmeth : I'm betting your config files go mangled. |
14:41.14 | cvv | bye-bye |
14:41.18 | *** part/#asterisk cvv (n=cvv@212.8.35.34) |
14:43.15 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
14:43.21 | [TK]D-Fender | Egonis : context = context1 |
14:43.30 | [TK]D-Fender | Egonis : channels => 1-3 |
14:43.32 | [TK]D-Fender | Egonis : context = context2 |
14:43.39 | [TK]D-Fender | Egonis : channels => 4 |
14:44.09 | Egonis | ah, thank you! And that goes in zapata.conf? |
14:46.20 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
14:47.57 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
14:48.07 | [TK]D-Fender | Egonis : Correct. when done in that order the other channels inherit all the characteristcs that remain unchanged (as in ALL of them) and the only thing we override is the context |
14:48.43 | E-bola | How can i verify a sip proxy is running? |
14:48.55 | E-bola | remotely i mean, i cant connect to my voip providers sip proxy |
14:49.10 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
14:51.51 | *** join/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
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14:54.51 | dwrecktion | Question: I'm on an office network that already has asterisk installed and configured and working fine. but i'm trying to set up another asterisk test system that I can eventually move into another location. Anybody have any recommendations/tips/suggestions on making this work as far as configuration? |
14:55.31 | *** join/#asterisk Gr1ncheux_ (n=devine@AStDenis-105-1-22-82.w81-248.abo.wanadoo.fr) |
14:56.20 | afrosheen | <PROTECTED> |
14:56.29 | nortex | E-bola, Have you checked you firewall/router to make sure nothing is blocking the request? |
14:56.37 | Luke-Jr | Looking for origination: US, Kansas City DIDs, respectable pricing (per minute or otherwise), *non* Flash website, preferably LNP |
14:58.58 | afrosheen | Luke-Jr: commpartners.us |
14:59.18 | afrosheen | or txlink.net |
14:59.19 | dwrecktion | I'm not cloning a system. I'm designing a new system, but for now it is on the physical network |
14:59.19 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
14:59.37 | afrosheen | dwrecktion: as long as nothing is trying to register to it, it will be invisible |
15:00.01 | Luke-Jr | afrosheen: thanks |
15:00.20 | E-bola | nortex: the host is directly on the internet |
15:00.30 | E-bola | the router is passthough |
15:00.35 | afrosheen | Luke-Jr: np..commpartners is preferred, they have insanely cheap intl. rates |
15:00.46 | E-bola | i just need some sort of tool to verify a given host runs a sip proxy |
15:00.53 | E-bola | i found a nagios plugin and that times out too |
15:01.35 | dwrecktion | so I can put another Asterisk server and SIP phones on the same network as the existing Asterisk setup and have them work independently? |
15:02.06 | trelane_ | <3 handset |
15:04.55 | dwrecktion | afrosheen: so I can put another Asterisk server and SIP phones on the same network as the existing Asterisk setup and have them work independently? |
15:05.42 | Luke-Jr | afrosheen: unfortunately, CommPartners doesn't seem to cover KC |
15:05.58 | jbalcomb | dwrecktion: absolutely |
15:07.06 | dwrecktion | jbalcomb: and can you summarize how? |
15:07.39 | *** join/#asterisk kiddy (n=achu@59.93.32.89) |
15:07.53 | kiddy | How can I connect two asterisk server's ? |
15:08.12 | Luke-Jr | IAX2 |
15:08.28 | jbalcomb | dwrecktion: did you set up the first system? |
15:08.57 | dwrecktion | i didn't, but i feel like i've got a fairly decent understanding of how its set up |
15:09.15 | kiddy | Luke-Jr : can you pls give me the url where I can find the configuration of interconnecting two servers ? |
15:09.34 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
15:09.46 | jbalcomb | dwrecktion: give your secondary asterisk server a different IP and configure the phones with that IP as thier SIP server. done. |
15:10.02 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
15:10.51 | jbalcomb | dwrecktion: you might consider using a different private class bock for clarity and/or VLANs for traffic segmentation |
15:11.14 | *** join/#asterisk blaylock (n=sfv100@68-69-102-120.chvlva.adelphia.net) |
15:11.20 | dwrecktion | jbalcomb: does SIP server = IP Gateway? |
15:12.10 | jbalcomb | dwrecktion: i would think IP gateway would be your router. where do you see this term? |
15:12.17 | blaylock | anyone using the new TE412P or TE407P cards? |
15:12.20 | *** join/#asterisk jetaway2009 (n=asd@218.111.10.56) |
15:12.49 | blaylock | well maybe not new really |
15:13.08 | jetaway2009 | hui |
15:13.10 | dwrecktion | jbalcomb: this is on the phone. i've trying to figure out how to configure the phones to connect to the SIP server |
15:13.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
15:13.38 | afrosheen | Luke-Jr: I just checked commpartners website and I have a whole list of KC area codes |
15:14.37 | Luke-Jr | afrosheen: weird... I searched :/ |
15:14.45 | Luke-Jr | and go "No records were found" |
15:15.37 | afrosheen | I clicked service area then the link for kansas |
15:16.27 | Luke-Jr | Where is that? |
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15:16.56 | *** join/#asterisk SexyKen (n=Ken@c-71-202-149-39.hsd1.ca.comcast.net) |
15:17.19 | mut | anyone know if the t3 cards will be * supported anytime soon? |
15:17.34 | SexyKen | Hey guys -- currently my Asterisk setup has some Voicemail configs to actually allow for over the phone voicemail - but some of them delete the file from the server and have it sent via e-mail. |
15:17.46 | SexyKen | I'm wondering if there is anyway to get it sent in MP3 format instead of WAV |
15:21.04 | *** join/#asterisk TeePOG (n=temp@dsl-145-178-200.telkomadsl.co.za) |
15:21.18 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
15:21.32 | *** join/#asterisk Eggplant (i=No@dsl-216-155-214-162.cascadeaccess.com) |
15:21.41 | *** join/#asterisk Precion (n=crhodes@adsl-75-7-75-29.dsl.milwwi.sbcglobal.net) |
15:21.49 | TeePOG | hi guys |
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15:22.04 | Luke-Jr | SexyKen: MP3 isn't ideal for voice |
15:22.19 | SexyKen | MP3 isn't ideal for voicemail? |
15:22.27 | Luke-Jr | MP3 is designed for music, not voice |
15:22.57 | Luke-Jr | and even then, Vorbis is better ;) |
15:23.09 | SexyKen | Right but in my situation, MP3 would be useful.; |
15:23.15 | SexyKen | So that still doesn't answer the question I had. |
15:23.18 | Luke-Jr | Why not use speex? :) |
15:23.20 | SexyKen | Or did I miss your answer? |
15:23.32 | Luke-Jr | MP3 encoding would probably be a patent violation, so unlikely |
15:23.50 | Qwell[laptop] | among other things |
15:23.54 | SexyKen | Explain |
15:24.08 | Qwell[laptop] | SexyKen: MP3 is a patented tech... |
15:24.19 | Qwell[laptop] | You need to have licenses in order to do so (legally) |
15:24.49 | SexyKen | Oh goodness -- I better call someone and tell them tha 80% of the internet is illegally encoding MP3's!!! |
15:24.59 | Qwell[laptop] | SexyKen: Yes, this is quite true |
15:25.03 | *** join/#asterisk Blaze312 (i=Blaze312@24-247-183-114.dhcp.aldl.mi.charter.com) |
15:25.10 | Blaze312 | just a quick question |
15:25.44 | Luke-Jr | lol |
15:25.51 | Blaze312 | im a total newb |
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15:25.55 | jetaway2009 | rhrhr |
15:25.58 | jetaway2009 | hhehe |
15:25.59 | jetaway2009 | true |
15:26.12 | E-bola | grrr dammit |
15:26.13 | Blaze312 | just wondering if avaya ip phones work with asterisk |
15:26.17 | E-bola | i finally found a free sip provider |
15:26.29 | E-bola | and it works right away, so it turns out it WAS my payed for voip provider who |
15:26.32 | E-bola | 's fucked up |
15:26.52 | Luke-Jr | o.o |
15:28.35 | Blaze312 | should any IP phone work with asterisk or is it limited to certain ones? |
15:28.47 | SexyKen | SIP/IAX phones should work. |
15:28.54 | SexyKen | Go look it up you jackass. |
15:29.36 | Blaze312 | i looked on the site but didnt see a list of supported phones or anything |
15:29.46 | Blaze312 | this particular phone is a SIP phone |
15:30.18 | jbroome | then you're fine |
15:30.22 | Blaze312 | im very new to PBX and just learning linux as well |
15:30.32 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
15:30.34 | Blaze312 | just thought asking here would be the fastest way to get my answer |
15:30.45 | Blaze312 | im looking to set this up when i move in a month |
15:31.27 | SexyKen | I'd be careful if I were you. |
15:31.46 | *** join/#asterisk mmarker (n=mmarker@216.220.209.239) |
15:31.50 | RoyK[uk] | hm.. anyone that knows how i can debug asterisk while it's running? it is hanging with some 'active' calls which are not active at all |
15:32.12 | RoyK[uk] | this is with some custom patches. i beleive those patches are to blame... |
15:32.20 | rob0 | Anyone in UK? I am wondering about 0870 numbers. I take it that those cost a premium to call? |
15:32.20 | [TK]D-Fender | SexyKen : SHUP YUO. |
15:32.46 | SexyKen | SHUP YUO? |
15:33.57 | rob0 | Blaze312: the learning curve can be steep in places and it goes a VERY long way. In general IRC is not a good substitute for learning the basics. Good luck. |
15:35.04 | mut | hey tk |
15:35.21 | mut | know if sangoma plans to make drivers for their t3 card for *? |
15:35.30 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
15:36.22 | Blaze312 | rob0 |
15:36.31 | Blaze312 | thanks for the info. i know what you mean |
15:36.39 | Blaze312 | i was just coming here to get a couple of quick answers |
15:37.01 | Blaze312 | ill probably spend time actually learning by looking through the documentation and tutorials |
15:37.41 | MrChimpy | hmm. |
15:38.01 | MrChimpy | i need to be able to run lots of concurrent perl AGIs |
15:38.13 | MrChimpy | is there anything like FastCGI? |
15:38.33 | rob0 | Blaze312: the Wiki has a lot of good stuff. |
15:38.50 | [TK]D-Fender | mut : Already have drivers for linux so * can use that card.... |
15:39.04 | mut | channelized? |
15:40.33 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
15:43.00 | *** join/#asterisk matkix01 (n=null@71-209-37-209.bois.qwest.net) |
15:43.02 | matkix01 | Hey all |
15:43.40 | Strom_C | good morning! |
15:44.34 | mut | [TK]D-Fender: 2 A104D's in one system will work well right? |
15:45.44 | *** part/#asterisk matkix01 (n=null@71-209-37-209.bois.qwest.net) |
15:45.50 | *** join/#asterisk chorlick (n=Chris@63.81.26.126) |
15:46.09 | *** join/#asterisk piper69 (n=piper69@69.155.81.24) |
15:46.12 | [TK]D-Fender | mut : Sure, but why not just buy an A108d? |
15:46.22 | mut | because they're new |
15:46.30 | [TK]D-Fender | mut : Channelized? Now you're being PICKY! ;) |
15:46.43 | [TK]D-Fender | mut : Works gread for SIP though ;) |
15:46.44 | mut | well |
15:46.54 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
15:47.03 | [TK]D-Fender | mut : Translation : No idea when/if it will be channelized anytime soon |
15:47.04 | mut | i'm looking to make a class 4/5 switch |
15:47.10 | mut | instead of 100k on a new one |
15:47.26 | mut | i don't need anything like that right now |
15:47.44 | mut | [TK]D-Fender: didn't know they had an a108d |
15:48.06 | *** join/#asterisk sandra78 (n=aerae@200.106.67.49) |
15:48.10 | sandra78 | hi pls |
15:48.15 | sandra78 | help!! |
15:48.30 | [TK]D-Fender | mut : perfect solution. Though you CAN add 2 A104d without any real problems usually. |
15:49.02 | sandra78 | i have create a context exten => s,1,dial(zap/g0/${DNID}#,40,r) |
15:49.25 | sandra78 | the SIP incomming nummber its paas fine with SIP |
15:49.34 | sandra78 | but with IAX i get a empy number |
15:51.07 | sandra78 | in cli console with sip i get zap/g0/639604000#|40|r but in iax i get zap/g0/639604000#|40|r |
15:51.13 | mut | wonder what kinda cpu i'de need to do simple switching |
15:51.24 | mut | wouldn't think much |
15:51.41 | [TK]D-Fender | mut : not much |
15:53.42 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
15:54.16 | sandra78 | ?? |
15:54.36 | Strom_C | sandra78: those are the same string |
15:54.49 | *** part/#asterisk JffMRIII (n=JffMRIII@c-67-167-202-60.hsd1.il.comcast.net) |
15:54.49 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
15:55.53 | sandra78 | no, incomming number don't pass when i use iax |
15:56.26 | Strom_C | well, you pasted the same string for both examples |
15:56.31 | sandra78 | with sip i guess dial the channel with the DNID number |
15:56.43 | sandra78 | ohhh yes sorry |
15:57.04 | sandra78 | i get this with iax Executing Dial("IAX2/1000-3", "zap/g0/#|40|r") in new stack |
15:57.11 | Strom_C | what's on the other end of your IAX trunk? |
15:57.25 | sandra78 | with sip i get this fine Executing Dial("SIP/77701-69f4", "zap/g0/439565345#|40|r") in new stack |
15:57.36 | sandra78 | another asterisk box |
15:57.52 | Strom_C | what's the dial string on that asterisk box? |
15:58.11 | pdtmobile | this channel doesn't happen to get archived anywhere does it? |
15:59.15 | sandra78 | the dial string it's fine i get the number filtered |
15:59.20 | sandra78 | <PROTECTED> |
15:59.57 | Strom_C | pastebin the dialplan and iax.conf on the box you're having trouble with |
16:00.01 | sandra78 | i'm calling from a internal sip extension and get out with a iax trunk |
16:00.05 | Strom_C | pastebin.ca |
16:00.39 | sandra78 | i have the last asterisk release |
16:00.47 | sandra78 | 1.2.9.1 |
16:00.50 | Strom_C | pastebin the dialplan and iax.conf on the box you're having trouble with |
16:03.08 | sandra78 | what it's this¿? |
16:03.20 | sandra78 | pastebin.ca?? |
16:03.29 | Strom_C | ~pb |
16:03.30 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
16:03.40 | Dr-Linux|work | file |
16:03.45 | Dr-Linux|work | guys |
16:03.57 | Dr-Linux|work | this problem is killing me since last day .. |
16:03.58 | Dr-Linux|work | Jul 14 08:48:01 NOTICE[12346]: rtp.c:564 ast_rtp_read: Unknown RTP codec 104 received |
16:04.10 | Dr-Linux|work | other end can't hear me |
16:04.15 | Dr-Linux|work | what could be wrong? |
16:04.29 | Strom_C | you screwed up your SIP configuration ;) |
16:05.04 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
16:05.43 | Dr-Linux|work | Strom_C: what do you mean? |
16:05.58 | Dr-Linux|work | Strom_C: do you understand what's wrong with my codecs? |
16:06.13 | Strom_C | what codec are you trying to use |
16:07.00 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
16:07.12 | jetaway2009 | could someone pls tell me how to extend T1 cables |
16:07.24 | [TK]D-Fender | Strom_C : Looks like video |
16:07.41 | [TK]D-Fender | jetaway2009 : std CAT5E extension. |
16:07.42 | jetaway2009 | any device to enable the cable for a long distance |
16:07.45 | *** join/#asterisk dandan (i=dandan@pacanka.com) |
16:07.51 | dandan | hey all |
16:07.52 | dandan | :) |
16:07.53 | Strom_C | jetaway2009: define "long distance" |
16:07.55 | Sonderblade | is it possible to code your whole dialplan in agi? |
16:08.03 | [TK]D-Fender | jetaway2009 : For long distances you need a repeater and will have to change your LBO |
16:08.04 | Dr-Linux|work | Strom_C: i tried all codecs one by one |
16:08.08 | Strom_C | Sonderblade: if you're totally insane, yes |
16:08.22 | dandan | I already have a sangoma card and now I need to put in a x100p for overhead paging, how do I cnfigure zapata.conf for multiple cards? |
16:08.25 | jetaway2009 | i read t1 cable has limitation on the length. |
16:08.32 | Sonderblade | Strom_C: why insane? |
16:08.35 | Strom_C | jetaway2009: hence the need for a repeater |
16:08.47 | dandan | jetway: yes it does which you compensate on the hardware |
16:08.48 | Strom_C | Sonderblade: why the hell would you want to do the whole thing in AGI? |
16:08.51 | iCEBrkr | jetaway2009: stretch them! :D |
16:08.55 | mut | PRAISE THE LORD FOR BACKUPS! |
16:08.59 | Dr-Linux|work | Strom_C: on priority bases, like g729, g723, ulaw , alaw and ilbc |
16:09.06 | Sonderblade | Strom_C: because asterisk's dialplan language sucks |
16:09.09 | jetaway2009 | how much for the repeater |
16:09.20 | jetaway2009 | could u recommendation specific brand |
16:09.25 | Strom_C | adtran |
16:09.30 | [TK]D-Fender | Sonderblade : Works fine for 99% of needs. What are you having trouble implementing in it? |
16:09.45 | jetaway2009 | is repeater=channel bank? |
16:09.48 | Strom_C | no |
16:09.51 | Strom_C | repeater = repeater |
16:09.52 | [TK]D-Fender | jetaway2009 : No idea. You'll have to call a place that sells them |
16:09.55 | dandan | I already have a sangoma card and now I need to put in a x100p for overhead paging, how do I cnfigure zapata.conf for multiple cards? |
16:10.02 | Sonderblade | [TK]D-Fender: MOST pbx:es works fine for 99% of needs |
16:10.09 | Strom_C | dandan: don't ask the same question twice in two minutes |
16:10.12 | [TK]D-Fender | jetaway2009 : Channel bank CAN be a repeater, but no, you typically buy a little box to repeat the signal on. |
16:10.32 | [TK]D-Fender | Sonderblade : So what exactly are you trying to do that make you want to use AGI instead? |
16:10.35 | jetaway2009 | will i able to google with term repeater and t1 |
16:10.35 | dandan | strom: I didn't even hear no idea... go google or anything |
16:10.42 | jetaway2009 | is it the correct term |
16:10.52 | Strom_C | dandan: if no one wants to help, no one will say anything |
16:10.58 | Strom_C | dandan: welcome to #asterisk |
16:11.03 | Sonderblade | [TK]D-Fender: i want to log each incoming and outgoing call in a db |
16:11.11 | Strom_C | Sonderblade: that's easy |
16:11.18 | Dr-Linux|work | Strom_C: no clue? |
16:11.19 | Strom_C | Sonderblade: asterisk already does that in call detail records |
16:11.23 | dandan | strom: heh |
16:11.30 | Strom_C | Dr-Linux|work: i dont feel like debugging it right now |
16:11.42 | jetaway2009 | whats the different between cat5 and T1 cable |
16:11.45 | jetaway2009 | look the same |
16:11.51 | jetaway2009 | with exact 8 pin |
16:11.55 | Strom_C | jetaway2009: T1 can run over cat5 |
16:12.05 | Strom_C | jetaway2009: ethernet and T1 use different pairs |
16:12.17 | Sonderblade | Strom_C: its more advanced than that, i want to log each state change for each sip device connected to the asterisk |
16:12.17 | Strom_C | ethernet uses pairs 2 and 3, T1 uses pairs 1 and 3 |
16:12.47 | jetaway2009 | but the cable are the same in size ..but different pairs...is it true |
16:12.57 | Strom_C | jetaway2009: that's what I just said |
16:13.21 | Strom_C | Sonderblade: what do you mean? you want to log each keypress? |
16:13.47 | Sonderblade | Strom_C: each device can AFAIK be either InUse, Ringing, Unavailable, Idle each time a device changes its state i want to log the change |
16:13.53 | jetaway2009 | yes...just rechecking my understand... |
16:13.58 | Strom_C | Sonderblade: /why/? |
16:14.00 | jetaway2009 | reconfirming |
16:14.03 | jetaway2009 | :) |
16:14.26 | jetaway2009 | how is T1 able to accomodate alot extension with only 8 pin.. |
16:14.41 | Strom_C | jetaway2009: it only uses four wires |
16:14.42 | Sonderblade | Strom_C: to monitor how the pbx is used |
16:14.50 | Strom_C | jetaway2009: look up time division multiplexing |
16:15.09 | Strom_C | jetaway2009: it's this really amazing new stuff that's only been around since the 1950s |
16:15.26 | SplasPood | Does a module for asterisk exist that can do direct XMLRPC calls to a remote service? (similar to CURL()..) |
16:15.38 | Strom_C | Sonderblade: I still don't get what the end purpose of doing so is |
16:15.38 | Sonderblade | Strom_C: which extension does the most calls? which extension gets the most calls? etc, you need advanced logging to answer such questions |
16:15.47 | jetaway2009 | oh..thanks...a will search for that term...u seem to havee alll the answerss... |
16:15.56 | Strom_C | Sonderblade: you can figure all that out by parsing the call detail records |
16:15.56 | jetaway2009 | for everythin...amazing.. |
16:16.08 | Strom_C | Sonderblade: have you /looked/ at the call detail records asterisk generates? |
16:16.41 | Dr-Linux|work | Strom_C: wtf, again stopping/restarting asterisk resolved my problem :S |
16:16.43 | Dr-Linux|work | what's this |
16:17.05 | jetaway2009 | what if the effect if i run T1 over cat5 |
16:17.14 | Strom_C | jetaway2009: T1 runs over cat5 cable |
16:17.15 | Sonderblade | Strom_C: i had no idea it logged how much a device was ringing |
16:17.33 | Sonderblade | Strom_C: nor did i know that it could log attemts to reach a device when that device is unavailable |
16:17.37 | Strom_C | Sonderblade: the asterisk CDRs are fairly detailed |
16:17.40 | sandra78 | i have this http://pastebin.ca/88168 |
16:17.43 | sandra78 | issue |
16:17.49 | jetaway2009 | so...there is no such thing as T1 cable...so when i shop around ..just buy cat5 cable |
16:18.06 | jetaway2009 | then why in the shop classified as t1 and cat5 cables.. |
16:18.09 | Sonderblade | Strom_C: or that it can log calls that end with a busy tone etc.. |
16:18.10 | Strom_C | jetaway2009: yes, the only thing you need to worry about is getting a T1-specific crossover cable |
16:18.21 | Strom_C | Sonderblade: yes, that's in the CDR |
16:18.38 | RoyK[uk] | jetaway2009: you can use anything for T1 cabling. it's not like it's really high speed |
16:19.14 | jetaway2009 | oh..thanks...strom C...u had cleared a lot of doubts over the cabling,,bugging me for some time |
16:19.26 | jetaway2009 | <RoyK[uk]> : thanks men |
16:19.27 | Strom_C | thats why I'm Strom Carlson |
16:19.38 | Strom_C | telephone deity |
16:19.45 | RoyK[uk] | jetaway2009: i just use a normal cat 5 |
16:19.47 | Sonderblade | Strom_C: and i need to display the data in realtime |
16:19.48 | jetaway2009 | :) |
16:19.48 | Strom_C | [fanfare] |
16:19.49 | RoyK[uk] | just another colour |
16:19.58 | RoyK[uk] | ~strom_c |
16:20.16 | jbot | i heard strom_c is just some nub |
16:20.16 | Strom_C | Sonderblade: so you use the asterisk mysql cdr addon |
16:20.17 | *** join/#asterisk kc5cqm (i=mwilliam@2002:a55f:d1d:0:0:0:0:1) |
16:20.25 | jetaway2009 | oh..i been looking the whole day..the different between them..oculd have just asked here earlier |
16:20.31 | kc5cqm | howdy |
16:20.55 | Strom_C | Sonderblade: or you write an AMI program that monitors the status of all channels in use |
16:21.07 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
16:21.32 | *** join/#asterisk hess\n (n=hess@2084452.cps.virtua.com.br) |
16:21.37 | Tall-guy | Any "eyebeam" users hanging about? |
16:21.59 | dandan | I already have a sangoma card and now I need to put in a x100p for overhead paging, how do I cnfigure zapata.conf for multiple cards? |
16:22.11 | kc5cqm | is there a way to transfer to another extension without putting the other party on hold? Something similar to an analog-equivilent of having both phones off hook at the same time? |
16:22.25 | Sonderblade | Strom_C: yes, except that polling is very inefficient |
16:22.26 | Strom_C | dandan: it's really easy, but i recommend you use something less abysmally bad than an x100p ;) |
16:22.33 | kc5cqm | I want to switch between my cordless and corded phone without the other party knowing. |
16:22.44 | dandan | strom: my valcomm unit has an fxs interface |
16:23.00 | *** join/#asterisk matheusbh (n=pankz@200150009221.corp.wayinternet.com.br) |
16:23.03 | dandan | i just need to configure x100p on * to once in a while talk to the shop floor :) |
16:23.07 | kc5cqm | I'm using an x100p...works fine for home use |
16:23.20 | Qwell[laptop] | dandan: You'll need an fxs for that |
16:23.22 | jetaway2009 | pls suggest a solution..in a 8 floor building..i plan to install asterisk server on the 5 floor.and intend to share with my firends with analog channel bank..the issue is the cabling.how to supply the phone cable to 1st floor without line derioation ...as the cable has limitation in distance..isthe repeater the best solution? |
16:23.24 | kc5cqm | actually have it plugged into my packet8 sip adapter |
16:23.26 | matheusbh | hello all. somebody knows where i can found a tutorial of how i can use a modem to be a FXO? |
16:23.42 | dandan | Qwell[laptop]: huh? yesterday I connected a regular phone since valcomm generates a dialtone |
16:23.47 | Qwell[laptop] | matheusbh: You don't |
16:23.48 | dandan | and was able to talk to them |
16:23.51 | sandra78 | pls help!! |
16:24.00 | Strom_C | Sonderblade: well then perhaps you should talk to the guys in #asterisk-dev and see if there's a way you can hook into asterisk more efficiently, because AGI isn't going to save you |
16:24.01 | matheusbh | pls help me too! |
16:24.06 | sandra78 | http://pastebin.ca/88168 i have issues with iax incomming DNID numbers |
16:24.08 | Strom_C | oh will you all pls shut up |
16:24.16 | Tall-guy | jetaway: I have a 5km connection to my telco, doesn't degrade THAT bad :) |
16:24.43 | dandan | ~books |
16:24.48 | dandan | ~docs |
16:24.50 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
16:24.53 | Strom_C | ~book |
16:24.54 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
16:25.03 | dandan | tx |
16:25.04 | Strom_C | sandra78: i'm looking at your pastebin now |
16:25.09 | dandan | maybe there will be something |
16:25.11 | sandra78 | http://pastebin.ca/88168 i have issues with iax incomming DNID numbers with sip works fine with iax not |
16:25.20 | kc5cqm | anyone here intereface a digium analog card to a 2-way radio? |
16:25.27 | Sonderblade | Strom_C: yes, but first maybe you should back up why you think writing the dialplan in FastAGI is insane? |
16:25.27 | Strom_C | sandra78: please stop whining so I can help you |
16:25.30 | kc5cqm | I know there's support for it. |
16:25.55 | sandra78 | thanks strom_c :* |
16:25.55 | MrChimpy | whuh? there is a fastagi? |
16:26.00 | kc5cqm | sandra78, you allowing UDP traffic on the iax port? |
16:26.20 | MrChimpy | gosh there is too |
16:26.24 | MrChimpy | just what I need! |
16:26.27 | Strom_C | Sonderblade: because the dialplan is not a channel driver |
16:26.30 | sandra78 | yeah i have ringing in the extensions |
16:26.47 | Strom_C | Sonderblade: and by writing the whole thing in AGI, you're not gaining the functionality you're looking for |
16:26.48 | jetaway2009 | Mr.Strom.. |
16:26.50 | sandra78 | 4569 port |
16:26.56 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
16:26.56 | Strom_C | jetaway2009: what |
16:27.18 | jetaway2009 | pls suggest a solution..in a 8 floor building..i plan to install asterisk server on the 5 floor.and intend to share with my firends with analog channel bank..the issue is the cabling.how to supply the phone cable to 1st floor without line derioation ...as the cable has limitation in distance..isthe repeater the best solution? |
16:27.24 | sandra78 | i have trying to get the iax DNID number but i can't get it |
16:27.35 | Strom_C | man, you guys whine a lot |
16:27.35 | jetaway2009 | need ur expert opinion .. |
16:27.59 | Strom_C | jetaway2009: at those distances, you're fine with regular cat5 |
16:28.05 | dandan | jetaway2009: supply it digitally and install channel bank on 8 floor |
16:28.07 | CunningPike | kc5cqm: I haven't heard of anyone else doing it, but I met "Dude" and his sidekick at Astricon last year and they had a working setup |
16:28.09 | dandan | it is not that far |
16:28.12 | Tall-guy | jetaway: really dude, it's not that far.... |
16:28.24 | Tall-guy | 10 floors X 10 feet per floor is only 100feet!!! |
16:28.29 | Strom_C | jetaway2009: if you really want to get fancy, do fiber to the floors and then demux to analog on each floor |
16:28.30 | sandra78 | <kc5cqm> yeah i have 4569 port fordward to my asterisk box |
16:28.48 | CunningPike | kc5cqm: I'm trying to dig out his contact details....... |
16:28.50 | dandan | port 4569 tcp? udp? |
16:28.57 | kc5cqm | CunningPike, interesting |
16:29.01 | Sonderblade | Strom_C: im gaining a more convenient way to implement the functionality |
16:29.03 | Strom_C | sandra78: you didnt pastebin what I asked for |
16:29.08 | Strom_C | Sonderblade: good luck with that then |
16:29.25 | kc5cqm | sandra78, you might not need to explicitly forward that port...it's UDP based and should behave on its own |
16:29.39 | Strom_C | Sonderblade: I suggest you go to #asterisk-dev |
16:30.39 | sandra78 | <Strom_C> http://pastebin.ca/88172 |
16:30.50 | *** join/#asterisk file (n=file@neutrino.joshua-colp.com) |
16:31.34 | Strom_C | sandra78: no, thats not what I asked for |
16:31.36 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
16:31.49 | jetaway2009 | Strom_c..u are one men show... |
16:31.52 | Strom_C | sandra78: I asked for the iax.conf AND the extensions.conf of the terminating asterisk box |
16:31.56 | jetaway2009 | helping all at same time.. |
16:31.57 | jetaway2009 | amazing |
16:31.59 | sandra78 | what did you asked? you asked me by my iax.conf |
16:32.09 | *** join/#asterisk pengyong (n=lala@218.93.68.246) |
16:32.09 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
16:32.17 | Strom_C | jetaway2009: in sixteen minutes, I also do magic tricks |
16:32.42 | CunningPike | kc5cqm: Here we go - http://www.zapatatelephony.org/app_rpt.html |
16:32.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
16:32.46 | jetaway2009 | great to meeet a person like u |
16:32.51 | jetaway2009 | multitaskingg... |
16:33.06 | sandra78 | ok |
16:33.26 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-164-167.dsl.irvnca.pacbell.net) |
16:33.30 | MGSsancho | bam! |
16:33.55 | kc5cqm | thanks CunningPike |
16:34.18 | Tall-guy | jetaway: wait till you see "Revolution" when he's on a crack-induced typing spree |
16:34.29 | CunningPike | kc5cqm: You're welcome de VA7IRL ;) |
16:34.36 | jetaway2009 | :) |
16:35.16 | Waverly360 | yo |
16:35.16 | jbalcomb | when I put my phone on DND is unavail.gsm the file it plays when someone calls? |
16:35.17 | kc5cqm | ;-) |
16:35.49 | [TK]D-Fender | jbalcomb : depends on the phone. On many you can set what SIP status it will return. |
16:36.28 | CunningPike | jbalcomb: Provided you have set your dialplan and VoiceMail() options correctly, yes. Some phones don't distinguish between busy and no answer, but most do |
16:36.34 | jbalcomb | [TK]D-Fender: hrmm.. we are doing DND server side.. |
16:37.06 | CunningPike | jbalcomb: Then it's up to your dialplan to call the appropriate VoiceMail() options |
16:37.30 | jbalcomb | "DBGet(dnd=dnd/SIP/${ARG3})" and then "GotoIf($["X${dnd}" = "X0800"]?9:6)" |
16:37.47 | *** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
16:38.07 | jbalcomb | Is that saying goto priority 9 if it is in DND but goto 6 if its not? |
16:38.21 | CunningPike | jbalcomb: Probably.......... :D |
16:38.55 | jbalcomb | so then "s,9,Voicemail(su${ARG2})" is whats getting played.... |
16:39.59 | jbalcomb | so it sends them to the Voicemail function with status 'su' which is 'something unavailable? |
16:40.19 | Waverly360 | So I'm getting a tiny bit irritated with my Polycom phones... |
16:41.07 | Strom_C | Waverly360: leave it t [TK]D-Fender to solve all your polycom problems |
16:41.12 | Strom_C | :) |
16:41.17 | Waverly360 | heh...he couldn't last time :P |
16:41.23 | Strom_C | what?! |
16:41.23 | Strom_C | gasp |
16:41.29 | Strom_C | I, for one, am shocked |
16:41.32 | Waverly360 | hah |
16:41.35 | Waverly360 | Well, he was helpful |
16:41.36 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
16:42.03 | CunningPike | jbalcomb: 'u' is the unavailable greeting - change that to 'b' if you want the busy greeting instead |
16:42.04 | Strom_C | but yea, I've done installs with polycom phones, and I'm sticking with cisco |
16:42.05 | Waverly360 | I'm trying to get the MyStat options working on my phones |
16:42.17 | CunningPike | Waverly360: Won't happen |
16:42.35 | jbalcomb | are do the MyStat options do? |
16:42.37 | Waverly360 | CunningPike: I called Polycom tech support, and they said that the options should.. |
16:42.39 | jbalcomb | s/are/what/ |
16:42.43 | [TK]D-Fender | jbalcomb : Not familiar with that method if its supposed to be something more that just a value YOU set. Phone isn't supposed to communicate DND except as a response to a call. |
16:42.47 | CunningPike | Waverly360: On Asterisk?? |
16:43.11 | Waverly360 | CunningPike: Not on asterisk persay, but on other pbxs |
16:43.25 | jbalcomb | [TK]D-Fender: yeah, we are setting the dnd variable in the asterisk DB so that we can check it in our dialplan |
16:43.31 | CunningPike | jbalcomb: They allow you to set 'Out to Lunch' on your phone and have other phones display that status for your line |
16:43.35 | Waverly360 | CunningPike: But that means the MyStat options are supposed to send something to the PBX |
16:44.01 | Waverly360 | CunningPike: I just can't get my phones to send anything..and I can't figure out how trick the phones into thinking asterisk supports it. |
16:44.22 | CunningPike | Waverly360: Haven't we had this conversation before? :D |
16:44.23 | Waverly360 | CunningPike: I'm wondering if maybe there's an option I can set in the phone.cfg or sip.cfg file that'll turn that feature on in the phone.. |
16:44.23 | jbalcomb | [TK]D-Fender there is some hassle going on right now with are queues setup. they just realized after 9 months that no one set the unavailable message on are rebates dept. call queues. |
16:44.34 | Waverly360 | CunningPike: Yes..but I haven't given up on it yet ;) |
16:44.44 | CunningPike | Waverly360: Good for you! |
16:44.49 | [TK]D-Fender | jbalcomb : SMRT |
16:44.51 | jbalcomb | CunningPike ah, that is totally cool. too bad they dont work. :/ |
16:44.54 | Waverly360 | CunningPike: I want to get the phones sending stuff to asterisk, so that I can possibly write a patch for the next version of asterisk. |
16:45.15 | CunningPike | Waverly360: That would be great - you'd make a lot of people very happy |
16:45.19 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
16:45.35 | CunningPike | jbalcomb: You get basic status - Online, Offline and Busy |
16:45.35 | jbalcomb | [TK]D-Fender yeah, so of course i now have four managers, a supervisor, and some helper bee chewing on my shoes about getting it fixed Right Now(tm) |
16:45.53 | Waverly360 | CunningPike: It won't happen unless I can figure out how the polycom phones work. Polycom tech support was less than helpful. |
16:46.01 | terrapen | jbalcomb, where do you work |
16:46.04 | CunningPike | Waverly360: You surprise me :| |
16:46.08 | Strom_C | it's Cresl1n! |
16:46.18 | Strom_C | will you sign my ISDN book? |
16:46.32 | [TK]D-Fender | jbalcomb : thats a step up from my typical Yesterday (tm) |
16:46.32 | jbalcomb | terrapen: Any Company, Inc. ? ;) IMT in Brunswick Ohio |
16:46.40 | Waverly360 | CunningPike: ? |
16:46.42 | terrapen | ah |
16:46.43 | *** part/#asterisk matheusbh (n=pankz@200150009221.corp.wayinternet.com.br) |
16:47.10 | CunningPike | Waverly360: I've never found their Tier 1 folks particularly useful |
16:47.10 | terrapen | my DTMF recognition problem is STILL happening |
16:47.17 | terrapen | upgraded the sangoma drivers and everything |
16:47.22 | *** join/#asterisk Venust1 (n=Emiliano@69-12-128-128.dsl.static.sonic.net) |
16:47.32 | terrapen | last thing to try is to upgrade polycom firmware |
16:47.47 | Waverly360 | CunningPike: That was my first time talking to them about anything. They were pretty much just BSing me..I'd like to talk to someone higher up, but I'm not exactly sure how to go about it. |
16:47.55 | Cresl1n | Strom_C: that sounds a little kinky |
16:48.04 | Strom_C | Cresl1n: you have no idea |
16:48.19 | Strom_C | ;) |
16:48.20 | CunningPike | Waverly360: How many sets do you plan to have? |
16:48.22 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
16:48.42 | Waverly360 | CunningPike: Sets? How many phones? |
16:48.43 | terrapen | the worst thing about this problem is that I have no idea what to blame it on...Polycom? Sangoma? Asterisk itself? Configuration error? |
16:48.58 | mut | anyone used a GR-303? |
16:49.02 | terrapen | and even worse, the problem occurs about 75% of the time |
16:49.08 | Venust1 | hi |
16:49.12 | terrapen | i hate sometimes-it-works-sometimes-it-doesn't problems |
16:49.19 | terrapen | invariably, the hardest to debug |
16:49.35 | CunningPike | Waverly360: Yes - sorry - POTS speak still creeps through sometimes :) |
16:49.37 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
16:50.01 | Waverly360 | CunningPike: Well..any ideas where I can get more info on the inner workings of these polycoms? |
16:50.04 | Venust1 | sometimes when I'm workin out in my shop I can't hear my SIP phone ringing |
16:50.25 | Strom_C | Venust1: get a louder ringer |
16:50.26 | terrapen | waverly, what are you trying to do |
16:50.38 | [TK]D-Fender | terrapen : I specialize on both nd never had problems with either. |
16:50.43 | CunningPike | Waverly360: We've found a chap called Stephen Sprunk quite helpful - he's one of the head tech honchos |
16:50.47 | Venust1 | Strom_C, since it's not an AC POTS line how do I do that? |
16:50.58 | terrapen | d-fender, have you ever seen call parking break? |
16:51.07 | Strom_C | Venust1: get a separate ringer box on an ATA and then ring both at the same time |
16:51.19 | terrapen | dfender, as in, the agent dials the code to park the call but instead of parking the call, the caller hears the DTMF? |
16:51.28 | Waverly360 | terrapen: I want to MyStat functionality on my 501 and 601s to actually do something. The options are there, but when I sniff the network traffic coming off of the phone, changing my status to busy, or away doesn't actually send anything to the pbx. |
16:51.35 | Venust1 | Strom_C, that's what I was thinking, just wanted to see if there was a phone already made to do what I want |
16:51.45 | Venust1 | google doesn't show anything |
16:51.45 | terrapen | dfender, but sometimes they don't hear the DTMF and the call is parked properly. i can't explain it |
16:51.53 | Strom_C | Venust1: what kind of sip phone? |
16:52.05 | Waverly360 | terrapen: once I get the phones sending info across the wire, I can start working on asterisk support for those options. |
16:52.07 | Venust1 | Strom_C, Grandstream GXP2000 |
16:52.20 | Waverly360 | CunningPike: Any chance you could give me a way to get in touch with him? |
16:52.30 | Strom_C | Venust1: the cisco phones have loud ringers :) |
16:52.31 | terrapen | waverly, mystat? like, DND? I haven't noticed this on my polycoms |
16:52.54 | Venust1 | Strom_C, I want a ringer light too |
16:52.57 | Waverly360 | terrapen: You have to enable presence in asterisk for the options to show up. Once you do that, and reboot the phone, the options magically appear ;) |
16:52.59 | terrapen | waverly, how are you sniffing? |
16:53.17 | Venust1 | Strom_C, I'm thinking ATA with an add on Flash/Ringer box |
16:53.23 | Waverly360 | terrapen: I connected my laptop and polycom phone to a hub, and ran ethereal on my laptop |
16:53.27 | Strom_C | Venust1: cisco has it |
16:53.41 | Strom_C | Venust1: or you can do the ringer box |
16:53.47 | MstlyHrmls | Waverly360: what version of polycom software are you using? |
16:53.48 | Venust1 | Strom_C, aren't Cisco phones extremely expensive? |
16:53.54 | Strom_C | $250? |
16:54.03 | terrapen | a real hub, eh? not a switch |
16:55.01 | MstlyHrmls | Waverly360: do you have a capture of the signalling from boot-up and registration? |
16:56.21 | jbalcomb | does the temporary greeting get played first and foremost regardless of the status of you phone? |
16:56.40 | Waverly360 | MstlyHrmls: I'm using SIP version 1.6.6.0036 |
16:57.01 | [TK]D-Fender | terrapen : Never tried call parking actually. |
16:57.05 | Waverly360 | MstlyHrmls: I didn't get a capture of the boot-up, but I can re-set everything backup and do that if you think it will help. |
16:57.14 | Venust1 | Strom_C, what model Cisco phone? |
16:57.25 | *** join/#asterisk kpettit (n=keith@adsl-70-241-120-196.dsl.hstntx.swbell.net) |
16:57.28 | Strom_C | Venust1: 7940 was 250 for one unit last I looked |
16:57.29 | MstlyHrmls | Waverly360: I'm just looking at the Admin guide for 1.6 |
16:57.36 | kpettit | can you allow people to dial options while on hold? |
16:57.36 | CunningPike | jbalcomb: Yes - it overrides everything |
16:57.53 | MstlyHrmls | Waverly360: it reads to me like the phone will only send the presence info if another device SUBSCRIBEs to it... |
16:57.54 | [TK]D-Fender | Venust1 : Not extremely, just more that they're worth VS Polycom |
16:57.54 | terrapen | d-fender, really? you should try it some time, it's quite cool. when it works. :) |
16:58.03 | kpettit | I'd like them to be able to "press 1 to leave a message" or other options while there waiting on hold |
16:58.17 | terrapen | what component of asterisk detects the DTMF during a call? |
16:58.35 | Strom_C | terrapen: the channel driver, IIRC |
16:58.48 | jbalcomb | CunningPike thanks |
16:59.14 | CunningPike | jbalcomb: np |
16:59.20 | terrapen | so, since the agents are using Polycom SIP phones, I should do some SIP debugging |
16:59.39 | MstlyHrmls | Waverly360: I've gotta muck around with a Polycom and another system later today, I can see if I can play around with the presence stuff then as well |
16:59.39 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
16:59.50 | kc5cqm | hey, what are dta310's going for these days, used? |
16:59.55 | kc5cqm | I've got 5 of them I want to sell. |
17:00.05 | Strom_C | kc5cqm: I'll do a dance for one |
17:00.14 | kc5cqm | they're a pain in the arse |
17:00.14 | *** join/#asterisk TeePOG (n=temp@dsl-145-178-200.telkomadsl.co.za) |
17:00.27 | fholmes | IN the [Demo] section of the extensions.conf file the Starting script uses exten => s,n,Answer. What does the n do? Should there be a number there or not? |
17:00.31 | terrapen | when the agents dial the park-call code, the caller hears a clicking or popping noise....except occasionally, they hear the agent's DTMF....except for sometimes they hear nothing and the call is parked properly. It's the strangest thing I've ever encountered with * |
17:00.35 | mut | anyone here into telcom ever used a Telcordia GR-303 |
17:00.48 | Strom_C | mut: isnt GR-303 a specification |
17:00.51 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
17:01.11 | rob0 | fholmes: not completely sure, I think it means "next", just makes it easier to insert/remove priorities. |
17:01.17 | terrapen | fholmes, the wiki would probably help you a lot |
17:01.21 | terrapen | esp, the extensions.conf page |
17:01.23 | Strom_C | the n priority means "next" |
17:01.27 | mut | yea |
17:01.36 | rob0 | fholmes: you can use labels to jump to a specific "n". |
17:01.44 | Strom_C | mut: i forget, what does GR-303 specify? |
17:01.54 | fholmes | rob0: So where to the labels go? |
17:02.00 | rob0 | terrapen: I guess I missed that on the Wiki. |
17:02.03 | mut | generic dlc requirements |
17:02.35 | mut | and some 'next gen' stuff i guess |
17:02.38 | rob0 | fholmes: there is an example in the sample extensions.conf |
17:02.44 | CunningPike | fholmes: exten => 1234,n(label),DoStuff() |
17:03.09 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
17:03.12 | mut | i gotta find out exactly what switch is there.. |
17:03.26 | Strom_C | mut: gimme an NPA-NXX and I can look it up |
17:03.36 | *** join/#asterisk docE (n=docelmo@66.237.242.41.ptr.us.xo.net) |
17:03.40 | mut | 989-507 |
17:03.40 | fholmes | Cool. Thanks guys. |
17:04.00 | docE | whadup |
17:04.21 | carl0s- | I had a chat with a guy today, an seller on eBay who's selling some voip fxo gateways. He said he's given up because he was "terminating gsm lines" with them and the gsm providers (Vodafone etc.) kept discovering this and blocking his simcards. He said he was using 600minutes/day. What exactly does he mean? Is this something naughty that you aren't supposed to do? |
17:04.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
17:05.31 | Strom_C | mut: odd, it doesnt show a switchtype |
17:05.36 | mut | :P |
17:05.54 | terrapen | carlos, he was probably using one of those cellphone->POTS converters and then plugging those into his FXO gateways |
17:05.59 | mut | i'll just run down to the CO and look at it next week or something |
17:06.13 | Strom_C | mut: DMS-something-or-other would be a safe bet |
17:06.14 | terrapen | using them to make advertisement calls to other cell phone users |
17:06.24 | carl0s- | terrapen: yes, he was. but for what purpose? To route calls from his voip service onto the mobile networks? |
17:06.28 | terrapen | since in-network calls are typically free |
17:06.45 | terrapen | possibly...but more likely, he was doing some kind of spamvertisement |
17:06.56 | carl0s- | terrapen: I wish they were! (here in the UK). They are cheaper than network-network calls. |
17:07.08 | carl0s- | terrapen: hmm yes spamming did cross my mind. |
17:07.21 | terrapen | there is a guy in turkey that uses asterisk to make political advertisement calls over GSM gateways |
17:07.28 | terrapen | like thousands of them each minute |
17:07.50 | carl0s- | blimey. he must have a big cellphone bill! |
17:08.07 | terrapen | carlos, well, here in the states, same-provider calls are typically free |
17:08.35 | mut | its an access navigator m1o |
17:08.47 | carl0s- | that's outrageous. I knew you had free local calls on your POTS lines. I didn't know you had free same-network calls. |
17:09.49 | carl0s- | same-network calls cost 10p/minute here. That's about 18c/min to you. |
17:10.51 | terrapen | yeah, we can talk for free to anyone who uses the same provider. and if we call between 2100-0600, we can call anyone in the US for free |
17:10.52 | carl0s- | or $11/hour |
17:11.02 | carl0s- | life isn't fair. |
17:11.10 | terrapen | come on over :) |
17:11.14 | carl0s- | :D |
17:11.18 | carl0s- | lol |
17:11.40 | carl0s- | I might struggle now. I received a small conviction for buying goods which had been stolen. |
17:11.58 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
17:13.15 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
17:13.26 | terrapen | "See these goods--they've never seen daylight, moonlight, Israelite, fanny-by-the-gas-light..." |
17:13.46 | carl0s- | :o |
17:14.17 | [TK]D-Fender | </eyesclosed> |
17:14.55 | carl0s- | So I've finally given up on the Motorola based X100P and have a TDM01B arriving on Monday. |
17:15.10 | Strom_C | carl0s-: YAY |
17:15.15 | carl0s- | :D |
17:15.31 | [TK]D-Fender | carl0s- : I give you till wednesday TOPS.... |
17:15.54 | carl0s- | [TK]D-Fender: what, before I come on here asking "How do I..?" or until I've got everything working properly? |
17:16.19 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
17:16.21 | *** join/#asterisk dacleric (n=dacleric@p54821F8C.dip0.t-ipconnect.de) |
17:18.09 | [TK]D-Fender | carl0s- : No, that will be monday NIGHT. Wednesday you will be bald, and retuning it SCREAMING :) |
17:18.21 | carl0s- | :( |
17:18.22 | carl0s- | ;) |
17:18.34 | Strom_C | [TK]D-Fender: why dont you bag on digium some more, please |
17:18.45 | Strom_C | :) |
17:19.01 | carl0s- | The place I ordered it from actually took my call through one, and it sounded great. Actually, it was an openvox board they were using but much the same I should think |
17:19.26 | [TK]D-Fender | Strom_C : I never placed BLAME, just his mental state at the end! |
17:19.31 | Strom_C | :) |
17:20.02 | [TK]D-Fender | Strom_C : Don't put words in my mouth... its full enough with foot as it is ;) |
17:20.08 | Strom_C | haha |
17:20.25 | carl0s- | I've already been up 'til 4am the last two nights, and in front of the computer all through the daytime. I look pale and feel a bit zombified. |
17:20.39 | Strom_C | ok, ill double check to see if chan_mouth is busy before executing a dial next time |
17:21.03 | jhiver | anybody knows when these new transcoding boards will be out, and at what price? |
17:21.14 | mut | well lets hope * can do gr-303 stuff well |
17:21.16 | Strom_C | jhiver: they will be out soon |
17:21.26 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
17:21.56 | jhiver | and also, how many concurrent channels they will support? I supposed they will be sized to a 4 E1 / T1 board |
17:22.11 | Strom_C | jhiver: 120 channels |
17:22.22 | jhiver | that's it then :) |
17:22.35 | jhiver | and what about the price? any info on this? |
17:22.35 | Strom_C | disclaimer: specifications may change before final release |
17:22.41 | Strom_C | i haven' |
17:22.49 | Strom_C | i haven't yet heard prices being thrown around |
17:23.20 | Strom_C | however, they'll include transcoding licenses for 120 channels for g729 and g723 IIRC, so figure that it wont be cheap |
17:23.23 | jhiver | with onboard echo cancel and hardware transcoding, we might finally have some pretty good quality |
17:23.49 | Strom_C | wait, did you just say transcoding and good sound quality in the same sentence? |
17:24.12 | jhiver | well it's better than CPU transcoding and no echo cancel :) |
17:24.25 | Strom_C | best sound quality == all ulaw, no transcoding :) |
17:24.38 | jhiver | I transcode g729 to ulaw on my audiocodes gw and it sounds great |
17:25.05 | jhiver | I prefer to do g729 all the way usually |
17:25.06 | Strom_C | g729 still sounds like cellphone calls though |
17:25.18 | jhiver | no, cell is clearly worse |
17:25.19 | Strom_C | you dont get the same clarity that you do with ulaw |
17:25.23 | jhiver | of course |
17:25.44 | Strom_C | hell, I'd do 48khz 16-bit telephony if I could |
17:25.54 | jhiver | but then you pay the price in terms of bandwith |
17:26.00 | Strom_C | well sure |
17:26.16 | Strom_C | but bandwidth is worth paying for if it means better-quality phone calls |
17:26.40 | jhiver | well, it depends what you are doing really |
17:27.02 | jhiver | when you work in the termination business, it's all g729 and g723.1 |
17:27.10 | jhiver | so having this board should be very good |
17:27.51 | Strom_C | i would run from a carrier that transcoded my wireline calls to g.729 :) |
17:28.19 | jhiver | then you can run from a _lot_ of telcos |
17:28.33 | jhiver | even the 'non voip' ones |
17:28.40 | Strom_C | trust me, I do :) |
17:28.51 | [TK]D-Fender | Run Forrest, RUN! |
17:29.03 | jhiver | because a lot of them buy from other carriers that interconnect with TDM with them, but then go all VoIP behind the scenes |
17:29.38 | Strom_C | yeah, but you can do voip without touching g.729 |
17:29.54 | Strom_C | the transport and the coding are not intrinsically linked |
17:30.02 | jhiver | well yes and no |
17:30.22 | jhiver | sometimes you have a route that is only H323 / G723.1 and you need to sell as SIP/g729 |
17:30.34 | jhiver | and so it's not like you have a choice |
17:31.09 | jhiver | to sell a route better, you need to accept as many codecs and protocols as possible |
17:31.16 | Strom_C | well, yes |
17:31.20 | jhiver | and hopefully this new card will let me do just that |
17:31.23 | Strom_C | *nod* |
17:31.43 | jhiver | BTW, I'm having quite a lot of headaches with H323 |
17:31.54 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
17:31.56 | Strom_C | the H stands for Headache |
17:31.59 | Corydon-w | Note that Digium itself isn't licensing the G.723.1 codec. It's the chip manufacturer that has done that. |
17:32.14 | jhiver | it's a complete bitch to compile, and when it works it only sort of half works |
17:32.31 | jhiver | I wonder if * business edition has proper support for H323 |
17:32.38 | jhiver | probably not |
17:32.39 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
17:32.48 | mut | i don;t think you can get single 723 licenses |
17:33.00 | mut | it's bulk, very large bulk |
17:33.14 | jhiver | apparently the way to do H323 is to use Yate but the docs are pretty harsh |
17:33.23 | jhiver | and there is no howto either :( |
17:36.11 | StromComm | I AM TEH NEXT VONAGE |
17:36.27 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
17:40.43 | drray | I am the next AOL |
17:42.17 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
17:42.40 | Nivex | I am the next Tastee Freeze! </random> |
17:42.52 | Strom_C | pfft, tastee freeze |
17:43.04 | drray | testes freeze! |
17:43.08 | jhiver | actually I have a little trick question for you guys |
17:43.09 | shmaltz | a clients system that is running externhost in sip.conf went down today because of no internet and therefore no dns. What program can I run on the Asterisk box that will act as a DNS proxy so that it doesn't happen again? |
17:43.25 | jhiver | I *think* my ISP is discriminating / limiting VoIP traffic |
17:43.48 | drray | shmaltz - just run a dns server? |
17:43.53 | jhiver | I wonder if there are tools I could use to prove this (sending different types of traffic to a given box and measuring the results) |
17:44.07 | jhiver | any ideas? |
17:44.08 | drray | jhiver - why not just tunnel the traffic |
17:44.09 | shmaltz | drray, I don't want a full dns server, I just want a silly dns proxy |
17:44.17 | mut | my isp doesn't give a crap what i do cause i pay them enormous amount of monies |
17:44.22 | drray | a dns server is a proxy |
17:44.25 | shmaltz | jhiver, whos your provider? |
17:44.39 | jhiver | mut, I _do_ pay them outrageaous amounts of money |
17:44.49 | shmaltz | drray, you are right but I want one that all it can do is proxy |
17:44.49 | jhiver | orange business services (i.e. france telecom) |
17:45.18 | jhiver | I am supposed to have a 4 Megs guaranteed connection |
17:45.28 | jhiver | I use only 1.3 Megs peak |
17:45.43 | mut | i have 2 ds3's |
17:45.50 | jhiver | and I still have packet loss at times of the day when pinging one of their routers |
17:46.24 | jhiver | and that kind of pisses me off since the connection costs something like 2k€ / month |
17:46.29 | fholmes | ~wiki |
17:46.43 | fholmes | ~wiki extensions.conf |
17:46.47 | drray | freshmeat returns DNRD a proxy dns server, along with 200 other hits |
17:47.20 | shmaltz | jhiver, coplain |
17:47.20 | jhiver | so I was wondering what tool you would use to prove that there is an issue... |
17:47.24 | shmaltz | complain* |
17:47.26 | jhiver | I do! |
17:47.43 | jhiver | I have loggued all ping times and packet loss and produced pretty graphs |
17:47.52 | jhiver | they still say there's no problem |
17:48.05 | Strom_C | jhiver: complain to the telecom regulation agencies |
17:48.08 | jhiver | when it's clear that there is something going very wrong |
17:48.21 | *** join/#asterisk iax (i=mbrooks@hijacked.us) |
17:48.32 | jhiver | still, I was wondering if there is another tool than "ping" I could use |
17:48.47 | Strom_C | mtr? |
17:48.57 | jhiver | mtr? what's that? |
17:49.13 | Strom_C | really really fancy traceroute |
17:49.31 | jhiver | cool |
17:49.52 | jhiver | cheers |
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17:55.56 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
17:56.54 | jhiver | thanks lad, nice tool |
17:57.04 | jhiver | I'll be able to spot where the problem is now |
17:58.56 | *** join/#asterisk Un1x (i=Un1x@CPE0040ca94518b-CM00137116f37a.cpe.net.cable.rogers.com) |
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18:00.26 | fndude | Hi , I'm trying to get call forwarding working with asterisk and my grandstream phone. It says in the logs its passing the call to local@default, but then my provider is giving me back a 603 error, any suggestions? |
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18:03.11 | trelane_ | this is madly offtopic but I figure someone probably knows, does anyone have an idea what Nextel's current FCC licenses are (operating frequencies)? |
18:09.05 | Skarmeth | Waverly360, hi, are you in toubles with Polycom SIP 1.6.6 ? I'm too |
18:09.51 | Skarmeth | I just updated some units here to test this new software and I can't make or receive calls with this new sip |
18:12.21 | Strom_C | exit |
18:12.22 | Strom_C | er |
18:12.26 | Strom_C | gah |
18:15.38 | *** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca) |
18:18.25 | *** join/#asterisk xbmodder_newlapp (i=nobody@atarack/staff/xbmodder) |
18:18.32 | xbmodder_newlapp | How long until chan_skype? |
18:19.31 | denon | never, hopefully |
18:19.39 | xbmodder_newlapp | why do you say that? |
18:19.50 | *** join/#asterisk evisu (i=hIRC@bzq-88-153-134-199.red.bezeqint.net) |
18:20.08 | denon | dunno, skype is kinda the OSX of the computer world |
18:20.22 | gandhijee | are the TDM400's 3.3V or 5V? |
18:20.31 | xbmodder_newlapp | It is? |
18:20.37 | xbmodder_newlapp | Well: |
18:20.42 | xbmodder_newlapp | ATA with skype |
18:21.01 | JakBeatZ | Folks, having an IAX2 registration problem. I keep getting registration refused. Configs and debugs at http://pastebin.ca/88258 Anyone have any ideas? |
18:21.02 | xbmodder_newlapp | integrate skype into asterisk would mean I can talk to all my non-techy friends |
18:21.23 | denon | or you could just dial out pstn at like 1c/min :) |
18:22.28 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
18:23.11 | xbmodder_newlapp | JakBeatZ, slight erorr in the configs |
18:23.50 | xbmodder_newlapp | JakBeatZ, next time post the config files/errors seperately? |
18:23.51 | xbmodder_newlapp | ok? |
18:24.22 | JakBeatZ | sure, sorry. |
18:25.30 | xbmodder_newlapp | well my reccomendation is: |
18:25.35 | xbmodder_newlapp | set one peer |
18:25.39 | xbmodder_newlapp | set one user? |
18:25.43 | xbmodder_newlapp | and vice-versa |
18:25.46 | xbmodder_newlapp | friends are complex |
18:26.00 | xbmodder_newlapp | and I have too bad of a headache to remember how to set that up |
18:26.29 | JakBeatZ | I see.. ok.. So instead of friend - friend do peer - user.. |
18:26.43 | xbmodder_newlapp | JakBeatZ, er |
18:26.47 | xbmodder_newlapp | Wanna talk in PM? |
18:26.52 | JakBeatZ | Sure, yes. |
18:31.11 | Waverly360 | Skarmeth: I'm having some issues, but I don't think they're related to your problems... |
18:31.15 | *** join/#asterisk jarrod (i=jarrod@juniperyour.net) |
18:31.37 | jarrod | hey is there a way to group individual sip accounts into some type of group in order to specify an call-limit |
18:31.55 | jarrod | and asterisk checks the group limit rather than individual |
18:31.56 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
18:32.11 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
18:32.44 | Dr-Linux|work | Jul 14 11:11:46 NOTICE[18865]: rtp.c:564 ast_rtp_read: Unknown RTP codec 104 received |
18:32.51 | Dr-Linux|work | anybody know what's wrong? |
18:33.14 | stoffell_h | Dr-Linux|work: did you try setting your asterisk to force the sip client to ulaw/alaw ? |
18:33.32 | Waverly360 | MstlyHrmls: I need to figure out how to make asterisk subscribe to the status options in the phone..any ideas where I should start? |
18:33.58 | Dr-Linux|work | stoffell_h: well, my codecs prority is something like: |
18:34.01 | Dr-Linux|work | g729 |
18:34.07 | Dr-Linux|work | g723 |
18:34.12 | Dr-Linux|work | ilbc |
18:34.15 | Dr-Linux|work | ulaw |
18:34.17 | Dr-Linux|work | alaw |
18:34.20 | Dr-Linux|work | gsm |
18:34.38 | stoffell_h | Dr-Linux|work: to see if it changes, try setting only 1 codec (example: the well known u/alaw)... to see if it changes.. |
18:34.56 | MstlyHrmls | Waverly360: unfortunately, no. I know Polycoms, but I'm not great with asterisk yet... :-7 |
18:35.03 | Dr-Linux|work | stoffell_h: i can't that |
18:35.34 | stoffell_h | for testing purposes, anything goes ;) |
18:35.53 | Dr-Linux|work | stoffell_h: i'm having this problem with only one node like outside caller >> Mulititech VOIP GW (g729) >> asterisk >> softphone |
18:36.14 | Dr-Linux|work | so if i remove g723 nothing will work |
18:36.39 | stoffell_h | hm, try "sip debug" and prepare for lots of debug messages then.. then post to mailing list ? |
18:37.01 | Waverly360 | MstlyHrmls: Hrm... There has to be a way I can trick my phone into thinking someone subscribed to that feature... |
18:37.36 | *** join/#asterisk rva (n=rafa@200.210.51.130) |
18:37.41 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
18:37.46 | MstlyHrmls | Waverly360: if you could find something that would send it the right SUBSCRIBE message, but I'm not sure what event you have to subscribe to |
18:37.55 | rva | has anyone ever managed to make sip can reinvite work with asterisk? |
18:39.04 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
18:39.08 | Waverly360 | MstlyHrmls: I guess I could go through the Admin manual some more...not sure if I'll find what I'm looking for there. |
18:44.07 | *** part/#asterisk mtaht4 (n=m@71.198.23.124) |
18:44.10 | *** join/#asterisk bmg505 (n=leon@c1-244-13.rndf.isadsl.co.za) |
18:44.36 | Corydon-w | rva: I wasn't aware that it was ever broken |
18:44.43 | *** part/#asterisk rva (n=rafa@200.210.51.130) |
18:47.06 | jarrod | hey is there a way to group individual sip accounts into some type of group in order to specify an call-limit so asterisk checks the group and not the individual |
18:48.42 | *** join/#asterisk matheusbh (n=pankz@200150009221.corp.wayinternet.com.br) |
18:48.46 | Mercestes | jarrod We do something like that with a mysql database using an AGI script with .php. |
18:49.01 | matheusbh | Hi All... Somebody know how to use one Pap2na ATA like a FXO port? |
18:49.07 | *** join/#asterisk veepster_ (n=veepster@67.130.38.2) |
18:49.10 | jarrod | im looking at checkgroup() or something and app_groupcount |
18:49.12 | jarrod | or something |
18:49.45 | matheusbh | it´s possible to do that? |
18:50.16 | Mercestes | possible. That's just the working model I have. |
18:51.06 | matheusbh | but using like a fxo? |
18:51.48 | matheusbh | Mercestes: Are u using a pap2na like a fxo port? |
18:52.16 | Mercestes | matheusbh: We use it to convert to analog. If by FXO port you mean "auto answer" or "not provide dial tone" then no. |
18:52.27 | sandra78 | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg143374.html this is my issue |
18:52.38 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
18:53.04 | alexns | yo |
18:53.41 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
18:54.30 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
18:55.02 | alexns | so what are you guys using to remotly monitor your clients asterisk systems ? |
18:55.13 | Waverly360 | SSH :P |
18:55.17 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-58.cpe.netcabo.pt) |
18:55.24 | alexns | hehe |
18:55.30 | Tall-guy | I use the phone, when it doesn't answer, my asterisk box is down :) |
18:55.35 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
18:55.45 | alexns | i mean graphically through the managers interface... and im talking about your clients |
18:56.24 | alexns | so the average idiot can know whats up |
18:56.35 | Tall-guy | oh, well you're assuming we're average....HUGE mistake :) |
18:56.47 | alexns | above average in here |
18:57.01 | alexns | but can't pay an expert to monitor these systems |
18:57.26 | alexns | just need the average tech to be able to see if providers are down and such |
18:57.41 | alexns | i used to have a program but cant remember what it was called |
18:57.49 | *** join/#asterisk enjay- (n=enjay@71.216.165.97) |
18:58.09 | jarrod | are there any presence utilities that are multiple client aware as in an isp platform? |
18:58.27 | enjay- | If I wanted to use a FAX machine on my asterisk server to sound out fax would I use a "HandyTone" type solution for analog->voip conversion? |
18:58.56 | enjay- | sound/send |
19:01.27 | Waverly360 | Ok, I have a request for anyone using a Polycom 501/601 with a NON asterisk PBX... |
19:02.46 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
19:03.24 | jarrod | how do i disable the music on hold all together so it doesnt even try to use it |
19:03.37 | Tall-guy | um, comment it out of the .conf file? |
19:03.49 | jarrod | it still tries to use it |
19:03.59 | enjay- | reload your config after commenting out |
19:04.22 | jarrod | i did a reload res_musiconhold.so |
19:05.31 | enjay- | just do a reload of all config files (i.e. reload) see if that resolves it.. |
19:06.49 | Waverly360 | MstlyHrmls: fyi, I just grabbed a packet capture of my polycom 501 after it's rebooted. |
19:07.58 | *** part/#asterisk fulgas (n=fulgas@a81-84-116-58.cpe.netcabo.pt) |
19:09.11 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
19:10.40 | [TK]D-Fender | Waverly360 : What is this question just offhand... |
19:10.46 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
19:11.52 | Waverly360 | [TK]D-Fender: I'm still trying to figure out how to make the statuses work with asterisk :P |
19:13.10 | [TK]D-Fender | Waverly360 : Did a sip debug on it? |
19:13.57 | [TK]D-Fender | Waverly360 : it may depend on * ASKING for "102 OPTIONS" which it doesn't IIRC. I believe * just tracks its OWN activity and reports it to phones that poll it. |
19:14.07 | daysmen3 | which do you prefer using ael or extensions |
19:14.17 | daysmen3 | and why?? |
19:14.52 | Waverly360 | [TK]D-Fender: I haven't done a sip debug on it...still learning the intricacies of using the CRI. (If that's even where it's done) |
19:15.36 | Waverly360 | [TK]D-Fender: hmm...I'd really like to see how other PBXs ask for those options... |
19:17.44 | Waverly360 | [TK]D-Fender: Crap...how do you turn SIP debugging off once it's on? |
19:17.52 | enjay- | sip no debug |
19:17.54 | [TK]D-Fender | Waverly360 "sip no debug" |
19:18.11 | Waverly360 | [TK]D-Fender: Ahh... thanks :) |
19:18.19 | [TK]D-Fender | Waverly360 : Use "sip debug peer [peerentry]" to only debug 1 phone to see how it communicates |
19:19.23 | Waverly360 | [TK]D-Fender: oh good..that makes things much easier.. :) |
19:19.30 | *** join/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net) |
19:19.41 | sharp | anybody know any good wireless hardphones? |
19:20.02 | [TK]D-Fender | sharp : They all suck ATM |
19:20.30 | sharp | then any good wired hardphones? |
19:20.42 | sharp | preferrably cheap? |
19:21.13 | Waverly360 | [TK]D-Fender: Is it possible to log the stuff from the CRI to a file temporarily? Or am I stuck copying and pasting? |
19:23.04 | [TK]D-Fender | sharp : Polycom > ALL |
19:23.05 | CunningPike | alexns: We're using Intermapper with Nagios plugins to monitor our Asterisk servers |
19:23.21 | [TK]D-Fender | Waverly360 : Don't know the details, but I think its in logger.conf |
19:23.39 | sharp | [TK]D-Fender, thanks for the suggestion |
19:23.40 | Waverly360 | [TK]D-Fender: K... Appreciated :) |
19:28.34 | Waverly360 | [TK]D-Fender: I need to get my hands on some polycom source code :P |
19:31.17 | [TK]D-Fender | Waverly360 : No, time to start reverse engineering :) |
19:31.52 | Waverly360 | [TK]D-Fender: heh... I suppose you're right...just start sending random things to the phone and see what happens...lol |
19:33.55 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
19:34.02 | [TK]D-Fender | Waverly360 : exactly! |
19:34.06 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@38.115.133.12) |
19:34.31 | [TK]D-Fender | Waverly360 : start debug, set a stat, change to another, then back. Try to create messages in a pattern that will tell you if its a random packet or a reaction. |
19:36.45 | xbmodder_newlapp | Anyone here used voxee? |
19:37.22 | fholmes | So if I have an extension 1200 and in my dialplan I want someone who presses 1 from the menu and I have exten => 1,1,Dial(1200) is that going to work properly? |
19:37.41 | Qwell[laptop] | fholmes: Local/1200 might work better |
19:38.00 | fholmes | Cool thanks Qwell |
19:38.38 | fholmes | Should I put an answer first for extension 1? |
19:38.45 | Qwell[laptop] | Don't need o |
19:38.46 | Qwell[laptop] | to |
19:38.59 | [TK]D-Fender | fholmes : strike that. Avoice creating additional Local channels and do something like "Goto(contextwithmyexteninit,theexten,1)" |
19:39.10 | [TK]D-Fender | avoid* |
19:39.50 | fholmes | So I should have a second context for the call queue for my tax sales team. |
19:40.22 | fholmes | Or if it is in the same contex then I just do Goto(,1200,1)? |
19:40.30 | fholmes | context.. |
19:42.13 | [TK]D-Fender | fholmes : it SHOULDN'T be in the same context. |
19:43.54 | fholmes | I don't want to include that in the main context either. It should only be called with the goto command. |
19:45.04 | *** join/#asterisk SanketMedhi (n=sanket@221.135.150.187) |
19:45.21 | fholmes | So what about the agent extensions? What context should they go into? The default, their own context or the Tax-Queue context? |
19:48.17 | SanketMedhi | hello, I am getting this error when I run asterisk -vvvvvvvvddddddddd ..... |
19:48.20 | SanketMedhi | Jul 15 01:16:55 WARNING[7947]: cdr_addon_mysql.c:295 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf |
19:48.49 | drray | realtime messed up? |
19:48.53 | SanketMedhi | I am trying to set up mysql Realtime, but I am not using cdr |
19:48.57 | SanketMedhi | sort of |
19:49.22 | SanketMedhi | any idea? |
19:50.44 | SanketMedhi | drray: does a warning make a different? I am anyways not using CDR, and it says Mysql has regd successfully |
19:51.35 | drray | I don't know |
19:51.52 | SanketMedhi | ok |
19:52.37 | SanketMedhi | anybody else? |
19:54.13 | JonR800 | 15:49 < SanketMedhi> any idea? |
19:54.17 | JonR800 | sorry |
19:54.19 | JonR800 | it means nothing |
19:54.31 | JonR800 | it means it couldn't load the config so it won't load the module, no big deal. |
19:54.37 | SanketMedhi | ok cool |
20:01.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
20:01.01 | *** mode/#asterisk [+o anthm] by ChanServ |
20:02.15 | SanketMedhi | how do I check if Realtime has actually parsed the DB? |
20:03.21 | carl0s- | groovy. |
20:03.29 | Alric | realtime load ? |
20:03.47 | SanketMedhi | You must supply a family name, a column to match on, and a value to match to. |
20:03.51 | SanketMedhi | I get that |
20:03.54 | tzafrir_laptop | check the actual config? |
20:04.02 | Alric | Yeah... |
20:04.04 | SanketMedhi | extconfig? |
20:04.08 | Alric | So then you supply that information? |
20:04.15 | SanketMedhi | in extconfig? |
20:04.17 | carl0s- | We don't get free same-network mobile calls over here (UK), but I just found out Orange will let me rent another handset on my business tarrif, and have free calls between the two mobiles on my contract. So for an extra £17.50/month I could route all my inbound SIP calls to my mobile.. |
20:04.18 | SanketMedhi | I have |
20:04.20 | Alric | like realtime load sip_users username 100 |
20:04.32 | SanketMedhi | what should that do? |
20:04.35 | Alric | Thats probably sipusers |
20:04.43 | Alric | Show whatever username 100 has |
20:04.54 | Alric | in the sip config. What are you trying to load from realtime? |
20:05.02 | *** join/#asterisk Primer (n=vi@sh.nu) |
20:05.05 | *** join/#asterisk chksolutions (n=damian@201.232.77.205) |
20:05.23 | Primer | So now that the skype protocol's been cracked, can we expect to see a chan_skype soon? :) |
20:05.36 | SanketMedhi | Alric: I have only created one table called sip_buddies and I am loading users from that DB |
20:05.47 | chksolutions | hello, I really need help in a topic |
20:06.10 | Alric | Okay. |
20:06.16 | Alric | So try to pull some data from it? |
20:06.19 | carl0s- | Hey even better.. I could route all my own mobile calls though my free mobile-to-mobile plan, via my Asterisk box and pay landline/SIP rates! That £17.50 could pay for itself! |
20:06.25 | Alric | Think of it like a select statement from SQL. |
20:06.39 | SanketMedhi | Alric: should I be having a username field in the table for that command to work? I have a name column and other |
20:06.50 | *** join/#asterisk s0lid (n=s0lid@61.28.161.132) |
20:06.52 | chksolutions | we've installed about 4 tdm400 cards in diferent computers, and all of them have noise, like a tick |
20:07.02 | Alric | Try matching by name then. |
20:07.04 | chksolutions | how can I resolve this issue |
20:07.16 | Alric | It just wants a column to match by. The code doesn't care what column it happens to be. |
20:07.45 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
20:08.13 | SanketMedhi | Alric: I have mapped sip_users and sip_peers to the same table , when I tried that command with these names it worked! :) Thanks |
20:08.25 | Alric | Cool |
20:08.31 | Alric | Welcome to Realtime. |
20:09.04 | SanketMedhi | Thanx :) |
20:09.36 | SanketMedhi | Alric: is there any other resource than voip-info to help me with Realtime? |
20:09.44 | Alric | Probably. |
20:09.59 | Alric | Google might be of assistance. I've only ever used voip-info though. |
20:10.16 | SanketMedhi | ok |
20:13.07 | chksolutions | I need help |
20:13.10 | chksolutions | please |
20:14.00 | *** join/#asterisk N3WWN (n=N3WWN@ns1.futuretek.cx) |
20:14.40 | N3WWN | Hi Guys! Anyone know why asterisk would hang up immediately when a call comes in (or out) of my PRI with a TE110 card? |
20:15.39 | Alric | Does it throw an error, or give any kind of information that would help point in a direction? |
20:15.51 | *** join/#asterisk MikeJ__ (n=vircuser@38.115.133.12) |
20:16.01 | chksolutions | This issue used to happend to me until my provider change the modem |
20:16.04 | N3WWN | Not that I can see.... |
20:16.19 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
20:16.20 | N3WWN | I see the Accepting call msg, the Answer("Zap/1-1" etc msg |
20:16.23 | Alric | What verbosity are you running at? |
20:16.26 | drray | hey bkw |
20:16.27 | *** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net) |
20:16.32 | Alric | And what logging is enabled? |
20:16.34 | N3WWN | verbosity 17 ;) |
20:17.04 | Alric | ... more paranoid than I am... |
20:17.12 | tzafrir_laptop | N3WWN, to what context do the call go? |
20:17.21 | N3WWN | I then see "Channel 1/1, span 1 got hangup" and a busy |
20:17.27 | tzafrir_laptop | you can see that in 'zap show channels' |
20:17.28 | N3WWN | default context |
20:17.42 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
20:17.51 | N3WWN | all default |
20:18.21 | tzafrir_laptop | What do you have in the context 'default'? 'show dialplan deault' . pastebin it if it is long |
20:18.23 | N3WWN | I usually use verb 4, but I"m trying to find out why the calls hangup immediately |
20:19.13 | N3WWN | http://pastebin.ca/88334 |
20:21.50 | N3WWN | I'm just using MusicOnHold to test cuz I didn't know if jumping to the Winbeam context was causing the hangup |
20:22.21 | *** join/#asterisk Eggplant (i=No@dsl-216-155-214-162.cascadeaccess.com) |
20:25.28 | *** part/#asterisk chksolutions (n=damian@201.232.77.205) |
20:25.35 | N3WWN | any thoughts? |
20:26.47 | *** part/#asterisk Primer (n=vi@sh.nu) |
20:28.47 | N3WWN | http://pastebin.ca/88343 now includes the output from /var/log/asterisk/debug |
20:31.56 | wrmem | N3WWN: What are you expecing it to do? You don't put the call on hold, so it's not going to play music, it will fall through to the next priority which doesn't exist (hence hangup) |
20:32.09 | N3WWN | even changing the exten so it calls a SIP phone causes the hangup |
20:32.19 | N3WWN | <PROTECTED> |
20:32.19 | N3WWN | <PROTECTED> |
20:32.19 | N3WWN | <PROTECTED> |
20:32.20 | N3WWN | <PROTECTED> |
20:32.20 | N3WWN | <PROTECTED> |
20:32.20 | N3WWN | <PROTECTED> |
20:32.22 | N3WWN | <PROTECTED> |
20:32.35 | N3WWN | The SIP phones work fine... they can call each other, call voicemail, etc |
20:32.36 | Alric | You can send an incoming call straight to MusicOnHold()... |
20:32.59 | wrmem | Try "Answer(), then SayDigits(12345) and see what happens |
20:33.23 | N3WWN | Alric, you're right, I use exten 8601 (legacy from the old PBX) to play music through the speaker |
20:33.27 | N3WWN | wrmem, I"ll try |
20:34.08 | N3WWN | <PROTECTED> |
20:34.08 | N3WWN | <PROTECTED> |
20:34.08 | N3WWN | <PROTECTED> |
20:34.08 | N3WWN | <PROTECTED> |
20:34.08 | N3WWN | Jul 14 16:35:09 WARNING[10100]: file.c:587 ast_readaudio_callback: Failed to write frame |
20:34.09 | N3WWN | <PROTECTED> |
20:34.11 | N3WWN | <PROTECTED> |
20:34.13 | N3WWN | <PROTECTED> |
20:34.35 | Alric | Oh yum. Good luck with that one :) |
20:35.04 | N3WWN | tks ;) |
20:35.45 | wrmem | Incoming on a PRI? Hmm. Has the PRI been turned up? |
20:36.07 | SanketMedhi | N3WWN: http://pastebin.ca |
20:36.08 | N3WWN | Yes, that's how the calls are being sent to the * server |
20:36.23 | N3WWN | SanketMedhi, what do you want me to post there? |
20:36.37 | N3WWN | I have some info at http://pastebin.ca/88343 |
20:36.40 | SanketMedhi | N3WWN: next time, post code there |
20:36.52 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
20:37.10 | wrmem | I've had PRI's not provisioned completely that wouldn't complete the call, but would have signalling (D up'ed, all B's down) |
20:38.14 | Corydon-w | It's not possible for the B's to be down... they are clear channel signalled |
20:38.41 | wrmem | Corydon-w: Ok. I'm not using the term exactly correct... |
20:38.59 | wrmem | (marked as Out-of-service) |
20:39.29 | Corydon-w | Some telcos implement something known as B-channel maintenance protocol, which is not supported by libpri. The telco needs to turn that off and restart the PRI before it will work with Asterisk |
20:39.40 | N3WWN | How about this... I know zttool doesn't show the sync source right, but should it show total/conf/act channels right? |
20:39.46 | xbmodder_newlapp | When, in an AGI script I "EXEC DIAL SIP/19252029415@plainvoip-out|120r |
20:39.46 | xbmodder_newlapp | " it no longer reads from the AGI script until I hang up, is there a way to still work with the channel? |
20:40.29 | N3WWN | http://pastebin.ca/88353 |
20:40.49 | N3WWN | Oh, that got ugly :( |
20:40.56 | Corydon-w | xbmodder_newlapp: not after it's bridged, no |
20:41.03 | xbmodder_newlapp | damn |
20:41.42 | xbmodder_newlapp | What about EAGI? |
20:41.56 | xbmodder_newlapp | How do I write directly to the audio stream, with PCM data? |
20:42.30 | syzygybsd | xbmodder_newlapp: create a local channel |
20:42.46 | xbmodder_newlapp | syzygybsd, eh? |
20:42.47 | N3WWN | essentially, all the TxA/B/C/D and RxA/B/C/D lines are filled with dashes (-) and Total/Conf/Act is 24/24/0 |
20:43.14 | xbmodder_newlapp | syzygybsd, how? |
20:43.45 | JakBeatZ | Folks, box A is * 1.2.0, box B is * 1.2.9.1. I have a SIP peer configured between the two boxes. On box A, I get Jul 14 16:43:08 WARNING[5275]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0xb397b0f8 (len 483) to boxBip:-1 returned 0: Invalid argument Anyone seen anything like this before?! |
20:44.11 | syzygybsd | heh, sorry, little information, but the way I do it is spawn a local extension, (dial(local/something@othercontext)) then bridge them together or put them in the same meetme |
20:44.48 | syzygybsd | agi won't hang because the local channel is spawned immediatly |
20:44.59 | xbmodder_newlapp | so have a dynamically created meetme? |
20:45.03 | xbmodder_newlapp | and do it that way |
20:45.09 | carl0s- | Has anybody seen a miniPCI GPRS/GSM card? And.. next question.. would one work as a zap channel in Asterisk? I'm planning on using a WRAP embedded board, which has dual ethernet and dual miniPCI, but no way to hookup POTS which is what most of the GSM gateways output. |
20:45.19 | syzygybsd | sure |
20:45.40 | syzygybsd | you might just be able to bridge the calls without a meetme, but I have always done it with one |
20:46.11 | xbmodder_newlapp | what about EAGI? |
20:46.20 | Corydon-w | carl0s-: theoretically, yes. However, when was the last time you wrote a kernel driver? |
20:46.28 | syzygybsd | what about it? |
20:46.56 | carl0s- | Corydon-w: never. So it hasn't been done already then? I guess I'm going to be limited to a SIP GSM gateway then. There aren'y many/any of those on eBay.. |
20:47.14 | syzygybsd | the dial command doesn't return until it is hungup. Doesn't matter where it is called from |
20:47.43 | carl0s- | am I correct in thinking there is a bluetooth channel driver which will utilise a cellphone as an FXO? |
20:48.22 | syzygybsd | carl0s-: http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels |
20:48.27 | syzygybsd | first result in google... |
20:48.48 | carl0s- | syzygybsd: bugger. no call functionality. |
20:49.48 | syzygybsd | well, this could just be me, but bluetooth doesn't support calls from another device over it |
20:50.21 | carl0s- | syzygybsd: I was thinking the bluetooth code would utilise the cellphone in the same way a car handsfree kit does. |
20:50.29 | syzygybsd | data service yes.. but when I look at the services provided by my phone connected by bluetooth I dont' see call... |
20:50.52 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
20:51.16 | syzygybsd | carl0s-: there is a huge differnce between sending mic/speaker information and sending the whole call |
20:51.55 | syzygybsd | your car handsfree doesn't have a keypad on it does it? |
20:51.56 | cytrak | hi I have Asterisk 1.2.7.1 and I'm now trying to setup the manager.conf file like I did before but for some reason it won't work |
20:52.13 | syzygybsd | lol.. k, can you tell us what doesn't work? |
20:52.18 | carl0s- | syzygybsd: No it doesn't. I see where you're coming from. It can only tell the cellphone to initialise a call from its existing phonebook. |
20:52.25 | cytrak | I try to telnet to iP:5038 I get into it but then it just hangs |
20:52.25 | Nugget | telnet is eeeeeeevil! |
20:52.42 | cytrak | I get no prompt for action, login, secret |
20:52.43 | syzygybsd | I hate that auto response |
20:53.02 | syzygybsd | cytrak: you never get a prompt for those |
20:53.04 | carl0s- | I though it might be an auto response.. a bit quick really. telnet. |
20:53.21 | carl0s- | bah |
20:53.29 | syzygybsd | all you should get is something like "Asterisk call manager version 1.0" |
20:53.43 | syzygybsd | I don't know what the exact message is.. |
20:54.21 | syzygybsd | ya, there is a minimum time before he will respond to "telnet" again |
20:54.38 | carl0s- | syzygybsd: I figured that :D |
20:54.47 | syzygybsd | I want to know what it is |
20:54.57 | syzygybsd | more then 2 minutes... |
20:55.05 | carl0s- | three minutes I reckon |
20:55.06 | cytrak | syzygybsd: ok I also tried to type them in |
20:55.12 | syzygybsd | I would guess 5 |
20:55.22 | syzygybsd | cytrak what did you type in |
20:55.53 | syzygybsd | cuz I am guessing you havn't copied from voip-info and tried that |
20:56.47 | *** join/#asterisk ivanfm (n=ivanfm@201.52.162.52) |
20:57.17 | cytrak | syzygybsd: Action:login Username:asterisk Secret:geo |
20:57.23 | cytrak | it just hangs |
20:57.35 | tzafrir_laptop | cytrak, is a connection established? Is this a linux telnet? |
20:57.46 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
20:57.49 | cytrak | I do remeber getting the promts on an older version |
20:57.53 | cytrak | yes |
20:58.02 | cytrak | it connects |
20:58.03 | syzygybsd | k, did you get the call manager line, and did you do 2 returns after it? |
20:58.14 | carl0s- | syzygybsd: http://cgi.ebay.co.uk/VoIP-GSM-Gateway_W0QQitemZ330005452870QQihZ014QQcategoryZ61840QQrdZ1QQcmdZViewItem |
20:58.21 | cytrak | that's what I forgot |
20:58.45 | cytrak | thanks |
20:59.21 | syzygybsd | never tried hitting enter again? Whenever it looks like a console is hanging I hit enter |
20:59.23 | cytrak | that worked now I got find out why my Asterisk-IM plugin for wildfire doesn't work |
21:01.04 | syzygybsd | carl0s-: that isn't what you think it is |
21:01.07 | russellb | file ... you're literally 2 feet from me |
21:01.25 | file | too lazy |
21:01.29 | carl0s- | syzygybsd: How so? |
21:01.55 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
21:02.09 | alexns | anyone have any luck with 1.2.9 and g729 ? |
21:02.14 | syzygybsd | I am thinking you want that to dial up to your phone provider (Cingular, T-mobile, whatever) that will allow an unlocked phone to connect to it.. so your cell phone can call yoru asterisk box... |
21:02.49 | hads|home | wow, I went out, got drunk, came home and slept and you guys are still poking :) |
21:03.01 | syzygybsd | still poking? |
21:03.07 | file | different location though I suppose |
21:03.10 | syzygybsd | mmm.. drunk.. good idea |
21:03.49 | syzygybsd | boss bringss the beers in 1 hour |
21:03.59 | carl0s- | syzygybsd: I don't understand 100%, but basically I have free calls between my two phones on my Orange contract. So I want to place my calls by calling up my second number (which will be that box - it'll have my SIM in it), and getting through to my Asterisk box where I dial a 9 or whatever for an outside line and place my calls at non-cell-rates. Also I would have my incoming sipgate.co.uk calls coming out to me via that also (follow-me). |
21:04.40 | hads|home | It's 0904 here, I can't get drunk again till I'm finished being hungover. |
21:05.07 | file | you better run, you better take cover |
21:05.09 | alexns | YO anyone have any luck with 1.2.9 and g729 ? |
21:05.12 | carl0s- | hads|home: I'm just on my first Reassuringly Expensive drink. |
21:05.37 | syzygybsd | carl0s-: doesn't orange have it so you can add landlines or other lines you can call for free too? |
21:05.51 | *** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net) |
21:06.29 | syzygybsd | hads|home: if you never stop being drunk you won't ever get hungover |
21:06.55 | carl0s- | syzygybsd: I think they have a deal where you can have them take over your office landlines and have cheap calls between all three - every cell and landline in the office. So anyway, I presume that box does what I was hoping? Basically runs my cellular SIM card like an FXO, except using SIP instead of analogue? |
21:06.58 | syzygybsd | alexns: I have used that, but call quality goes to shit after 5-20 calls |
21:07.28 | syzygybsd | carl0s-: no, that box doesnt' do what you want |
21:07.44 | hads|home | True, but my liver hurts. |
21:08.18 | carl0s- | syzygybsd: grr. What does that box do then that's different? I thought being a 'SIP gateway' it would be able to pass all incoming GSM calls over the Asterisk, and also place outgoing calls originating from Asterisk? What is that box meant for then? |
21:08.20 | syzygybsd | that box can make you your own service provide/phone company. |
21:08.39 | syzygybsd | k, let me explain it this way... |
21:08.44 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
21:09.02 | syzygybsd | what you want: Cingular -> phone/box whatever -> asterisk |
21:09.20 | syzygybsd | what that box does: phone -> that box -> asterisk |
21:09.27 | syzygybsd | no cingular anywhere |
21:09.31 | carl0s- | cingular? |
21:09.52 | syzygybsd | doesnt' matter, just an example of a phone comany, do3esn't matter who |
21:09.54 | carl0s- | oh |
21:09.59 | *** join/#asterisk oej (n=oej@38.115.133.12) |
21:10.27 | syzygybsd | had them for a while, they are gsm too.. mentioned orange.. I was guessing |
21:10.38 | alexns | syzygybsd: hmm i use it quite a bit but now since upgrade to latest version 729 codec doesn't work |
21:10.45 | carl0s- | I don't understand. You're saying that box doesn't do Cellphone? What's the point of the Antenna and the fact that it's called a 'gsm' gateway? |
21:10.57 | carl0s- | I don't want to use a handset with it. I just want to take the SIM card and place in into that box. |
21:11.19 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
21:11.40 | syzygybsd | carl0s-: think of it this way.. that is a base station not a wifi card |
21:11.59 | syzygybsd | hmm.. bad example.. base stations can connect to eachother... |
21:12.02 | kpettit | in ParkAndAnnount can you use a goto type statement rather than a dial in the notify section? |
21:12.15 | carl0s- | syzygybsd: hmm. Are you saying that in effect, that box can't connect to a GSM provider, but instead it makes you "your own GSM company" type thing? |
21:12.22 | syzygybsd | yes |
21:12.25 | carl0s- | crikey |
21:12.27 | carl0s- | that's useless. |
21:12.27 | kpettit | I love the ParkandAnnounce feature but it's killing me that I can only announce in a "DIAL" type fashion. I'm trying to page polycom phones with the annouce |
21:12.31 | carl0s- | Are you sure of that? |
21:12.33 | syzygybsd | that is what I was telling you |
21:13.12 | syzygybsd | go read the description, i looked into one of those to save me minutes while I am in the office, but then I realize why not just use a normal sip phone that doesn't cost $5,000 |
21:13.35 | carl0s- | dude, that's $5,000 taiwan whatsits. It's about $150 |
21:13.55 | syzygybsd | oh, didn't look at that |
21:13.58 | carl0s- | :D |
21:14.12 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
21:14.44 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
21:14.59 | syzygybsd | carl0s-: look at the pretty pictures they have on that page |
21:15.00 | FuriousGeorge | ? |
21:15.17 | *** join/#asterisk aster22 (n=aster@202.177.165.236) |
21:15.20 | carl0s- | I'll have to message the seller. I usually ignore taiwanese auctions but this looks genuine from the feedback and the company seems real. The description is typically poor and difficult to understand though. |
21:15.25 | aster22 | hello guys |
21:15.35 | aster22 | i need some help |
21:15.38 | aster22 | :) |
21:15.40 | syzygybsd | the pictures explain what it does well |
21:15.54 | FuriousGeorge | i think i found a bug in asterisk whereby if i call my own analog lines via my other analog lines, eventually the line that made the call stays "off the hook" and that fxo becomes permanently "in use" |
21:16.02 | FuriousGeorge | until i restart asterisk |
21:16.07 | aster22 | <PROTECTED> |
21:16.12 | aster22 | on 100 mbps connection |
21:16.22 | carl0s- | syzygybsd: I see what you're saying with the pictures and the phones using it as a base station, but they might just have missed out the GSM-provider cloud from that picture. hmm. A GSM base station of that size would be useless. They need to be everywhere to work and they'd be a hell of a lot more expensive than that. |
21:16.38 | FuriousGeorge | eventually all the fxo are in use and we start getting warnings as to that when we make local calls |
21:16.45 | syzygybsd | lol.... no... that isn't what that does |
21:16.45 | FuriousGeorge | and incomming calls stop working |
21:16.53 | syzygybsd | but go ahead and message if you want |
21:16.53 | aster22 | ntil i restart asterisk |
21:16.53 | aster22 | <aster22> is a 3200+ athlon 64 with 1 gb ram enough to handle around 50 calls and 100 extensions?? |
21:16.53 | aster22 | <aster22> on 100 mbps connection |
21:17.20 | aster22 | ?? |
21:17.25 | [TK]D-Fender | aster22 : internal LAN w/ SIP on G.711 and 2x PRI? |
21:17.28 | carl0s- | syzygybsd: I'll see what they say. The price is even cheaper than the FXO->GSM gateways, which certainly do acheive what I want. But I want to avoid needing analog ports on the Asterisk side. |
21:17.30 | syzygybsd | aster22: are you connecting via sip anywhere else? |
21:17.40 | syzygybsd | or just internal usage... |
21:17.43 | aster22 | no hadware all sip softphones |
21:17.53 | aster22 | both side g711 |
21:17.55 | *** join/#asterisk RoyK[uk] (n=roy@83.105.70.179) |
21:18.09 | syzygybsd | k... are you connecting to the pstn at all? |
21:18.17 | carl0s- | Nikki is out of the big brother house, just in case anybody cares :D |
21:18.18 | aster22 | like VOIP provider--> my dedicated server-->my office far away |
21:18.26 | aster22 | yes through a voip provider |
21:18.33 | syzygybsd | k, what is the bandwidth between offices? |
21:18.34 | [TK]D-Fender | aster22 : no issue then |
21:18.52 | syzygybsd | and to the voip provider? |
21:18.52 | syzygybsd | and how many calls to them at the same time |
21:18.55 | aster22 | dedicated server will have 100 mbps uplink |
21:19.07 | syzygybsd | wow.. ya.. no issues with anything then |
21:19.19 | aster22 | and offices will be kept like each pc will get 128 kbps up/down |
21:19.24 | syzygybsd | I was thinking that was just internal network... |
21:19.46 | aster22 | oki thx i just wanted to make sure b4r i buy server :) |
21:19.59 | aster22 | besides can i disable call transfer on certain extensions ?? |
21:20.01 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
21:20.16 | syzygybsd | is it bad that i find the ad on this page so funny? http://www.google.com/search?num=50&hs=wZU&hl=en&lr=&safe=off&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&q=dead+babies&btnG=Search |
21:20.24 | sevard | Can anyone take a look at this please and tell me what I'm missing? It's a simple while look in a dialplan. Results after the jump: http://pastebin.ca/88390 |
21:20.43 | syzygybsd | sevard: can you describe what you want to happen? |
21:20.47 | sevard | syzygybsd: url reconstruction blows, tinyurl.com :) |
21:20.57 | syzygybsd | oops.. sorry |
21:21.03 | file | operator operator! |
21:21.04 | carl0s- | syzygybsd: no adverts for me at all. |
21:21.06 | sevard | syzygybsd: yes, i constructed a while loop to play an audio file 3 times |
21:21.16 | syzygybsd | carl0s-: no sponsored links? |
21:21.16 | sevard | syzygybsd: at the pastebin i have thee diaplan and the results. |
21:21.21 | carl0s- | syzygybsd: no :( |
21:21.29 | syzygybsd | hmm... |
21:21.52 | carl0s- | Must be an EU censoring thing |
21:22.02 | syzygybsd | hopefully... |
21:22.11 | aster22 | i am pretty n00b to asterisk and use frepbx for config and somewhat changing files myself ... |
21:22.25 | syzygybsd | basically the ad is |
21:22.25 | aster22 | i want to know abt then settings given while maing extension |
21:22.27 | aster22 | like dtmf = |
21:22.28 | aster22 | nat = |
21:22.30 | aster22 | and all |
21:22.36 | syzygybsd | "Dead Babies at Amazon.com" |
21:22.38 | aster22 | any good coumentation on that |
21:22.45 | aster22 | :O whats that |
21:22.46 | aster22 | lol |
21:22.49 | sevard | aster22: http://www.voip-info.org |
21:23.30 | aster22 | k thx |
21:23.45 | sevard | syzygybsd: can you take a look at the pb? |
21:24.55 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
21:25.02 | syzygybsd | sevard: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+While |
21:25.09 | syzygybsd | $[] |
21:25.29 | mitcheloc | asterisk just went fatal on me heh, -- http://pastebin.ca/88393 any ideas anyone? i've never seen that before... |
21:26.33 | carl0s- | syzygybsd: The MT-350 (slightly different model to that on eBay) seems to do what I want from the text on the site. What do you think? Look at the second paragraph "When you dial out from your MT-350..." http://www.portech.com.tw/eweb/index1.htm |
21:26.53 | carl0s- | syzygybsd: sorry, damn frames. http://www.portech.com.tw/eweb/mt/mt350.htm |
21:27.38 | *** part/#asterisk SanketMedhi (n=sanket@221.135.150.187) |
21:27.39 | sevard | syzygybsd: i'm apparently missing some silly asterisk syntax? |
21:27.51 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-145-173.dyn.embarqhsd.net) |
21:28.03 | carl0s- | syzygybsd: bah. it's not SIP. It's FXO/FXS. |
21:28.19 | file | shows should be more... what's the word? |
21:28.39 | syzygybsd | it isn't really silly once you understand it, basically it is saying to evaluate whether what is inside that is true or not |
21:30.02 | sevard | syzygybsd: that makes sense. |
21:30.18 | carl0s- | syzygybsd: hmm. the MT-370 block diagram shows a SIM card. I think it might just do what I want. I'm downloading the user-manual now. http://www.portech.com.tw/eweb/MV370/mv370.htm |
21:30.37 | syzygybsd | ya, sim cards are a good indication.. |
21:30.37 | enjay- | Can flash operator panel integrate with do not disturb? |
21:30.48 | syzygybsd | enjay-: yes |
21:30.49 | carl0s- | syzygybsd: yup. |
21:31.03 | enjay- | syzygybsd; where can I find data on that? |
21:31.07 | syzygybsd | carl0s-: http://www.voip-info.org/wiki/view/GSM |
21:31.17 | *** part/#asterisk m4rkl4r (n=markp@66.129.95.30) |
21:31.44 | syzygybsd | look under pci adapters |
21:32.48 | carl0s- | syzygybsd: I looked at the Junghanns stuff but I want to be able to use this with an embedded WRAP ethernet/mini-pci only solution. The 2N thing will do also but I suspect it's pricey. |
21:32.57 | syzygybsd | enjay-: voip-info |
21:33.10 | syzygybsd | I don't really know what you want so that is as specific as I can get |
21:33.40 | nortex | enjay-, Check asternic.org and in the current release readup on the astdb conifg. |
21:33.46 | syzygybsd | enjay-: I wouldn't guess any more then $200 but I could be wrong |
21:34.34 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
21:34.34 | *** mode/#asterisk [+o anthm] by ChanServ |
21:34.41 | carl0s- | oh yes. The manual for MT-370 shows : "Network registration: The telecom carrier with which the SIM card is registered". I think we're onto a winner! |
21:34.59 | enjay- | thanks. |
21:34.59 | syzygybsd | ya, but how much does it cost? |
21:35.11 | aster22 | iax2 better or sip if pingtime is more than 300 ms ?? between server and extension |
21:35.31 | carl0s- | syzygybsd: £92 + delivery |
21:35.50 | syzygybsd | wow, ping time > 300ms... wouldn't really suggest voip |
21:35.54 | syzygybsd | I* |
21:36.05 | nortex | aster22, Neither :) |
21:36.12 | aster22 | lol |
21:36.14 | aster22 | why?? |
21:36.43 | syzygybsd | I don't really like the delay in my conversations, hearing what they said .3 or .4 seconds after they said it |
21:36.43 | aster22 | actually i do call in US from my softphone through internet voip providers |
21:36.47 | nortex | the latency would make it about as good as a cell phone in a bunker. |
21:36.52 | aster22 | and quality is pretty good there |
21:36.52 | CunningPike | aster22: Bcse i wnt wrk vy wl, if at all |
21:36.59 | nortex | lol |
21:37.16 | syzygybsd | thank you CunningPike |
21:37.27 | carl0s- | I get 37ms to sipgate.co.uk. That's adequate I suppose. |
21:38.02 | syzygybsd | i get 1ms to my voip server |
21:38.16 | aster22 | wow |
21:38.50 | syzygybsd | that is going through 3 switches, before it hits the gateway router |
21:38.54 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
21:39.07 | nortex | Correct me if I'm wrong, but you want something sub 100ms |
21:39.19 | CunningPike | We have about 70ms on average in 'sip show peers' |
21:39.21 | syzygybsd | then it goes to the ISP... then their ISP.. then back in through a couple.. |
21:39.57 | syzygybsd | I am suprised it is 1 ms personally.. with that many networking interfaces I would think at least 10 ms.. |
21:40.00 | FuriousGeorge | my snom 360 users are telling me that if they talk fast the other party tells them they break up. when they call me i hear it, but im not sure it has to do with the rate of speach |
21:40.19 | FuriousGeorge | now that i think about it, maybe it has to do with the switch built into the phone that i'm using |
21:40.38 | syzygybsd | hmm.. all of my sip show peers is unmonitored |
21:40.38 | aster22 | but i get more than 300 ms to my voip provider |
21:40.45 | aster22 | and quality is still very bearable |
21:40.48 | aster22 | n gsm |
21:40.50 | aster22 | on* |
21:41.00 | carl0s- | syzygybsd: mine show unmonitored too. I was just wondering how to change that. |
21:41.08 | syzygybsd | FuriousGeorge: what codec are you using? |
21:41.17 | aster22 | gsm or g711 |
21:41.24 | aster22 | on 256 k conn |
21:41.30 | FuriousGeorge | syzygybsd: ulaw |
21:41.33 | TrixVox | Anyone here using VoicePulse? |
21:41.42 | jbroome | i am |
21:41.59 | aster22 | damn no good voip providers in india :(:( |
21:42.32 | aster22 | which app do u use normally for post paid billing ?? |
21:42.37 | aster22 | on asterisk servers |
21:42.38 | syzygybsd | well, international calls will always have a bit higher of a delay.. so if you think it is acceptable then that is your choice |
21:42.41 | aster22 | for small scale |
21:42.51 | TrixVox | Their "Rates" page now shows 0.019 as their highest rate instead of 0.024... When did that happen!? Some of my calls are still showing as 0.024, even though most are less than 0.01... |
21:43.00 | syzygybsd | I dont' use post paid billing... make my company handle all the cost |
21:43.10 | aster22 | yes thats ok |
21:43.19 | aster22 | suppose if i am reselling from higher providers |
21:43.29 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
21:43.32 | aster22 | ofcourse i cna get billing done via sql records and excel |
21:43.35 | jbroome | TrixVox: they had us change our outgoing server the first week of june, when we did that we dropped down to the cheaper outging rate |
21:43.37 | carl0s- | I wonder what "Payment terms:EXW" means. |
21:43.39 | aster22 | but anything lil automated |
21:43.42 | syzygybsd | TrixVox: is that for "new subscribers only" |
21:43.52 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
21:44.23 | De_Mon | I've got a main-menu that plays some options. The caller can choose one of those options, which goes to submenus, or dial an employees extension directly... |
21:45.19 | TrixVox | Even the wiki still shows "0.5¢ to 2.4¢/min outgoing US long distance"... I think they just dropped their rates from 2.4 to 1.9c |
21:45.26 | De_Mon | the timeout extension is used to repeat the menu options, and I need a way to disconnect people that don't dial any extensions |
21:45.31 | *** mode/#asterisk [+o Corydon76-home] by Corydon-w |
21:45.57 | TrixVox | jbroome: Are most of your calls less than 1¢ with VoicePulse? |
21:46.37 | De_Mon | If I set the abs timeout, I'd have to turn it off if they called any of the local extensions, and turning off in that many places is not a plesent thought |
21:47.03 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
21:48.27 | jbroome | TrixVox: i believe so, i don't handle the billing, just the * upkeep |
21:48.35 | syzygybsd | De_Mon: that is what timeout is for... you can also have a counter set.. so if counter > 5 hangup |
21:49.23 | TrixVox | Ahh, I see... I don't know why people say they're "expensive" |
21:49.59 | jbroome | me either. call quality is great, haven't had any kind of problems either |
21:50.02 | De_Mon | syzygybsd I'd *like* to use the timeout, but turning it off if they leave that menu leaves a *lot* of placs to turn it off |
21:51.36 | TrixVox | Yeah, I didn't realize the audio quality difference... I think Asterisk users have gotten used to 'cell phone' quality due to their own bad hardware and/or these smaller providers with echo/dtmf issues. |
21:52.11 | tzafrir_laptop | OT: anybody knows if there is any provider except google of voip->pstn service that supports Jingle clients? |
21:54.15 | syzygybsd | tzafrir_laptop: how is that OT? |
21:54.31 | tzafrir_laptop | It's not about Asterisk |
21:54.41 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
21:55.21 | syzygybsd | 3/4 of the stuff in here is about general VOIP |
21:57.28 | feld_ | I have 5 incoming analog phone lines |
21:58.01 | feld_ | I want it to ring multiple phones. By using the DIAL command, if the first person picks up a line and another call comes in it gets sent straight to voicemail. |
21:58.18 | feld_ | Is the Queue with members the correct solution? |
21:58.50 | syzygybsd | ya, that sounds like it would work for what you want |
21:59.49 | feld_ | okay, so am I right in saying that I have to configure these sip phones as agents, set it to "ringall", and give all these members the same priority? |
22:01.30 | *** join/#asterisk TheBlack (n=sirius@p54BB63BC.dip.t-dialin.net) |
22:01.38 | TheBlack | hi^^ |
22:01.57 | syzygybsd | beers are out, i am afk |
22:02.17 | TheBlack | i'm searching a howto to install asterisk with visdn as trunk ... |
22:02.21 | TheBlack | can anyone help me ?^^ |
22:04.35 | *** mode/#asterisk [+o file] by ChanServ |
22:04.54 | Qwell[laptop] | still a nub |
22:06.52 | RoyK[uk] | fucking israelis..... |
22:08.07 | De_Mon | ^_o |
22:08.20 | De_Mon | are they interupting your VOIP service? |
22:08.25 | jbroome | they not doing voip right? |
22:09.04 | De_Mon | show function GOTOIF doesn't give me anything on the CLI, isn't it a function? |
22:09.45 | Qwell[laptop] | De_Mon: I don't think so |
22:09.58 | *** part/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
22:09.59 | hads|home | De_Mon: show application |
22:10.00 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
22:10.05 | *** mode/#asterisk [+o Qwell[laptop]] by ChanServ |
22:10.09 | Qwell[laptop] | :D |
22:10.29 | De_Mon | ah, thats better |
22:14.11 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
22:16.50 | *** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net) |
22:24.08 | docelmo | HAY! Anyone in here ever load balanced asterisk w/o SER? Like by using a F5 or Cisco L4 Switch? |
22:24.25 | benjk | anybody here using AstLinux? |
22:24.38 | docelmo | hehe.. What do you wanna know? |
22:25.06 | benjk | I want to know how to modify configuration files on the disk (not on the ram disk) |
22:25.09 | carl0s- | that's it. Daddy's bought a Portech MV-370. YAY. |
22:25.15 | docelmo | I personally know the developer.. He puked all over my car when he drank WAY TOO much |
22:25.21 | benjk | because everytime I reboot all the changes are gone |
22:25.28 | Qwell[laptop] | docelmo: That boy likes to drink... |
22:25.33 | Qwell[laptop] | heavily... |
22:25.45 | file | Qwell[laptop]: more then you? |
22:25.49 | Qwell[laptop] | file: much more |
22:25.51 | docelmo | yes.. I know.. I have a drunken pic of him to prove it |
22:25.57 | file | scary |
22:26.00 | Qwell[laptop] | file: very |
22:26.03 | benjk | Well, that's another issue, the AstLinux bugtracker doesn't seem to work, created an account but didn't get any email with the password |
22:26.27 | benjk | and there appears to be no documentation whatsoever |
22:26.46 | docelmo | kick kris in the balls |
22:27.32 | benjk | I already had to set up a fake DNS so the system would boot because it went into an endless loop trying to find its NTP server |
22:27.35 | docelmo | when he was at my house a couple months ago he passed out just before the sprinklers went off he got soaked.. |
22:27.51 | Damin | I've got Pictures of Kristian swigging vodka in our Hotel room at the Luxor.. :) |
22:27.58 | Damin | That was a WILD night.. :) |
22:27.58 | docelmo | haha |
22:28.09 | docelmo | I sent you the pic of him passed out at my house didnt I? |
22:28.14 | RoyK[uk] | ~lart benjk for fun |
22:28.14 | Damin | Yeah.. that's great.. |
22:29.09 | benjk | RoyK, so you know how to make AstLinux work then? |
22:29.42 | *** join/#asterisk gcarrillog (n=gcarrill@dsl-201-128-97-89.prod-infinitum.com.mx) |
22:29.45 | gcarrillog | hi |
22:30.02 | gcarrillog | sorry my english isnt good |
22:30.04 | RoyK[uk] | benjk: i just use asterisk. sorry. |
22:30.17 | gcarrillog | i have some questions about asterisk |
22:30.31 | aster22 | what? |
22:30.34 | benjk | yeah, I would like to do that, but I can't install any system on that damn CF card |
22:30.57 | benjk | got a 2GB card, cant get any Linux installed |
22:31.07 | carl0s- | benjk: AstLinux doesn't work right so far then? :) You know I'll be pestering you for the solutions to these problems you're encountering once my WRAP board arrives. |
22:31.17 | docelmo | Hay G whats your experience with load balancing a SIP device with a F5 or L4 switch? |
22:31.30 | gcarrillog | i was had my server asterisk behind of a router |
22:31.41 | gcarrillog | with DMZ enabled |
22:31.57 | benjk | no probs with the WRAP board because you don't have to do anything other than copying the WRAP image onto the CF |
22:32.02 | gcarrillog | but i need know which ports i need for SIP authentication |
22:32.11 | gcarrillog | please help me |
22:32.21 | benjk | but if you want to use PC and isntall from ISO CD, that's where things get tricky |
22:32.28 | carl0s- | benjk: ah I see. |
22:32.32 | enjay- | 5060 |
22:32.49 | gcarrillog | enjay- 5060 dont works |
22:32.57 | enjay- | 5060 is the port |
22:33.04 | gcarrillog | my softclient says request timeouth |
22:33.11 | enjay- | got a firewall? |
22:33.18 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
22:33.27 | enjay- | got NAT behind the firewall? do you have the port forwarded to the server etc etc etc? |
22:33.41 | hads|home | ooo |
22:34.00 | gcarrillog | enjay- yea but the port 5060 its routing to asterisk server |
22:34.27 | gcarrillog | enjay- only works if i enabled DMZ to server |
22:35.23 | *** part/#asterisk mog (n=mogorman@gateway.digium.com) |
22:35.26 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
22:35.26 | *** mode/#asterisk [+o mog] by ChanServ |
22:35.51 | enjay- | uh |
22:35.55 | *** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com) |
22:36.00 | enjay- | whats your border device? |
22:36.34 | gcarrillog | border? |
22:36.36 | gcarrillog | router? |
22:36.37 | enjay- | Like a PIX firewall, a Netgear Wireless AP/router? |
22:36.40 | Luke-Jr | gcarrillog: it's UDP, not TCP |
22:36.50 | gcarrillog | Luke-Jr :O ok |
22:36.52 | enjay- | yup |
22:36.52 | fndude | my call forwarding from my grandstream phone keeps getting '603' from my provider, can somebody tell me what would cause this? |
22:36.54 | gcarrillog | i will try |
22:36.58 | enjay- | sorry thought that was obvious :D |
22:37.00 | gcarrillog | my router its a 2wire |
22:37.08 | Luke-Jr | gcarrillog: unfortunately, Asterisk doesn't support TCP SIP |
22:37.29 | rob0 | 2wire ... ugh. |
22:37.47 | Luke-Jr | enjay-: SIP can be either |
22:38.01 | enjay- | Luke-Jr, as you said not with asterisk.. |
22:38.20 | Luke-Jr | enjay-: he might not have known ;) |
22:38.22 | enjay- | and Im assuming he's working with an asterisk system.. |
22:38.26 | enjay- | yea absolutely, my bad.. |
22:38.40 | Luke-Jr | and it's not really a bad assumption to think Asterisk can handle TCP SIP |
22:38.45 | Luke-Jr | since TCP makes sense for SIP |
22:38.50 | gcarrillog | :D |
22:38.54 | gcarrillog | that works |
22:38.54 | gcarrillog | :D |
22:39.01 | gcarrillog | enabling UDP |
22:39.02 | gcarrillog | :D |
22:39.04 | gcarrillog | thanks |
22:39.53 | *** join/#asterisk MACscr (n=MACscr@66.73.154.70) |
22:40.03 | enjay- | kinda makes sense.. I dont see it as entirely necessary to have an established connection.. I would think a datastream would be more efficient, more problematic yet more efficient.. |
22:40.17 | E-bola | Do any1 know of a click-to-call browser solution for windows? For none firefox users |
22:40.30 | Luke-Jr | enjay-: UDP is always more efficient =p |
22:40.37 | MACscr | Any phone techs in here that work a lot with avaya equipment? |
22:40.39 | Luke-Jr | enjay-: it's not reliable tho |
22:40.49 | enjay- | when dealing with text :D |
22:41.08 | hads|home | E-bola: http://www.snapanumber.com/ possibly. |
22:41.17 | E-bola | thanks hads |
22:41.21 | enjay- | Luke-Jr familiar with FOP? |
22:41.34 | Luke-Jr | FOP? never heard of it |
22:41.40 | enjay- | flash operator panel? |
22:41.46 | Luke-Jr | ... |
22:41.47 | mitcheloc | whoo hoo! |
22:41.57 | Luke-Jr | if you mean Macromedia |
22:41.59 | Luke-Jr | Flash can die and burn |
22:42.09 | mitcheloc | E-bola, we are working on IE right now, maybe less then 2 weeks away |
22:42.10 | Luke-Jr | along with companies who abuse it |
22:42.16 | E-bola | ohhh sweet |
22:42.21 | enjay- | haha |
22:42.36 | Luke-Jr | where 'abuse' is defined as anothing other than animation |
22:42.52 | enjay- | Im talking more specifically teh "Operators Panel, Receptionist Panel, etc" called Flash Operators Panel "http://www.asternic.org" |
22:42.57 | E-bola | whats difference between pro and free version micheloc? |
22:43.06 | enjay- | but I'll take that as a no.. |
22:43.15 | E-bola | ahh nm |
22:43.16 | E-bola | found it |
22:43.45 | mitcheloc | E-bola, one helps feed my 6 children, the other doesnt ;) |
22:44.08 | E-bola | hehe k |
22:44.16 | drray | 6 children |
22:44.24 | feld_ | ok guys I have an agents/queue setup and working, only I have one gripe.... It will ring in from the analog phone lines and then get picked up by extension 5000 which rings, waits 2 seconds, and puts them in the Queue. |
22:44.48 | feld_ | The problem is that the instant they're in the call group they're on music on hold |
22:45.09 | feld_ | I would prefer there to ONLY be music on hold if all agents are busy; I'd rather have them hear it ringing. |
22:45.37 | E-bola | mitcheloc: i dont see any license info |
22:45.43 | E-bola | is the free version freeware? |
22:45.53 | feld_ | *instant they're in the call queue, not call group |
22:46.32 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
22:47.08 | groogs | Goto(context,s/5551234,1) fails -- is there a way to make that work? I have exten=>s/5551234,1,Answer..... for doing CID-matching |
22:49.02 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
22:49.21 | fndude | when asterisk trys to forward calls from my grandstream, my service provider (telasip) denies the request, any ideas on what could be causing this? |
22:50.38 | mitcheloc | E-Bola: it runs as pro thens switches to basic, if you have more questions you can pm me ;) |
22:51.29 | E-bola | hehe alright |
22:51.39 | E-bola | i guess i gotta fix my asterisk installation first though :) |
22:57.22 | *** join/#asterisk adorah (n=Administ@87.69.72.228) |
23:01.24 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
23:10.03 | xbmodder_newlapp | Has anyone here messed with Asterisk EAGI? |
23:15.51 | *** join/#asterisk adker (n=adker@70-100-230-148.br1.glv.ny.frontiernet.net) |
23:16.47 | *** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com) |
23:17.24 | n9urk | hi all, Is it possible to have music on hold playing while an extension is dialed? |
23:17.52 | n9urk | instead of playing the ringing |
23:18.53 | xbmodder_newlapp | probably |
23:19.30 | n9urk | which function do you think might do it? (or which func. would you look at first) |
23:20.00 | CunningPike | n9urk: Look at the options for the Dial() command - I think one of them does that |
23:20.20 | n9urk | Thanks |
23:20.36 | xbmodder_newlapp | m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. |
23:20.59 | xbmodder_newlapp | that'll be 19.95, do you want frys with that? |
23:21.18 | n9urk | Yes, SuperSize me as well please |
23:21.30 | CunningPike | Hey - where's my cut? |
23:21.39 | xbmodder_newlapp | Would you like to with credit card, check, or paypal? |
23:21.42 | xbmodder_newlapp | CunningPike, ... |
23:21.42 | n9urk | I would like a milkshake |
23:21.45 | xbmodder_newlapp | your a cook |
23:21.51 | n9urk | IOU in blood |
23:22.06 | rob0 | mmmmmm milkshake |
23:22.11 | CunningPike | :) |
23:23.13 | n9urk | mmmmmmmmmmmmm milkshake with sprinkles! |
23:23.20 | xbmodder_newlapp | n9urk, which vein would you prefer? |
23:23.27 | xbmodder_newlapp | jugular is my favorite |
23:23.30 | xbmodder_newlapp | quick & dirt |
23:23.31 | xbmodder_newlapp | y |
23:23.37 | n9urk | left ventrical |
23:23.42 | xbmodder_newlapp | hm |
23:23.45 | xbmodder_newlapp | thats a little hard |
23:24.52 | rob0 | Vein? No way, arterial blood is better tasting and better for you. (And much faster for the victim.) |
23:25.15 | rob0 | Especially in a milkshake. :) |
23:25.16 | xbmodder_newlapp | still, its a pain |
23:27.09 | xbmodder_newlapp | I mean peircing it |
23:31.25 | rob0 | Not far past that jugular, to get to the carotid artery. |
23:31.45 | rob0 | oh ... you mean the wall resistance |
23:32.13 | rob0 | Anything worth doing is worth extra effort. :) |
23:32.49 | xbmodder_newlapp | would you prefer mouth to organ, or using some sort of siphon. |
23:33.13 | n9urk | I always like it mouth to organ |
23:35.30 | *** join/#asterisk knarfly (n=bdavis@c-69-180-98-189.hsd1.fl.comcast.net) |
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23:46.04 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
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23:56.23 | PakiPenguin | what is the meaning of this Forcing Marker bit, because SSRC has changed |
23:59.59 | aster22 | anyone know a good wholesale voip provider for corporate purposes |