| 00:00.19 | coldsteal | i have no idea how |
| 00:00.45 | shido6 | its quite simple, really |
| 00:01.03 | shido6 | if priority 1 times out or is busy you use priority 103 |
| 00:01.22 | shido6 | well no, if its busy |
| 00:01.31 | shido6 | if it times out it goes to the next priority |
| 00:02.01 | *** join/#asterisk davidcsi (n=davidcsi@80.27.42.156) |
| 00:02.12 | coldsteal | i have this for ext 20 |
| 00:02.13 | coldsteal | exten => 20,1,dial(SIP/20,30) \n exten => 20,n,busy() \n exten => 20,n,congestion() |
| 00:02.17 | *** join/#asterisk msetim (i=msetim@200-140-230-235.ctame705.dsl.brasiltelecom.net.br) |
| 00:02.28 | msetim | night |
| 00:02.45 | shido6 | yeah |
| 00:03.13 | shido6 | so what do u wanna do if its busy? |
| 00:03.17 | coldsteal | so i want it if i dont anser or if its busy to dial my cell phone |
| 00:04.19 | [TK]D-Fender | coldsteal, exten => 20,2,dial(Zap/2/5551212) |
| 00:04.33 | davidcsi | question: i have NO PEER REGISTERED. When a call comes in it goes to the default context. I have a peer in sip.conf with host=IP, that i send to conext whatever. Asterisk sends calls coming in from that IP to the context just fine. Queston is: Where does asterisk gets the IP Source? from the TCP/IP layer or from the SIPURI????? |
| 00:05.09 | coldsteal | [TK]D-Fender: whats Zap/2/5551212 |
| 00:05.23 | JT | another technology and number, coldsteal |
| 00:05.38 | keith4_ | zaptel, 2nd channel |
| 00:05.50 | *** join/#asterisk nath0099 (n=James@82-34-167-18.cable.ubr02.maid.blueyonder.co.uk) |
| 00:05.58 | [TK]D-Fender | coldsteal, just a sample that the only reason you get to priority 2 is because you didn't answer the first dial. |
| 00:07.55 | davidcsi | anyone? |
| 00:08.02 | AdamB0122 | anyway. |
| 00:08.16 | AdamB0122 | thanks guys, I'm actually getting to head home at a relativly decent hour tonight |
| 00:09.43 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 00:10.00 | EricL | Is there an equivilent way to do a SipAddHeader(foo: bar) in a .call file? |
| 00:11.51 | davidcsi | question: i have NO PEER REGISTERED. When a call comes in it goes to the default context. I have a peer in sip.conf with host=IP, that i send to conext whatever. Asterisk sends calls coming in from that IP to the context just fine. Queston is: Where does asterisk gets the IP Source? from the TCP/IP layer or from the SIPURI????? |
| 00:14.40 | *** join/#asterisk elriah (n=e@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:15.11 | elriah | Hi all. Any suggestions on how to address dtmf issues? I'm getting keypresses that should be, for example, "1234" as "123444444" in a lot of cases. |
| 00:15.29 | shido6 | turn the sound down or whip out relaxdtmf in sip.conf |
| 00:16.00 | elriah | Oops, asterisk 1.2. Is relaxdtmf in 1.2? |
| 00:16.27 | davidcsi | elriah: is it sip, iax or zap? what's on the other side? |
| 00:16.39 | elriah | All sip. |
| 00:16.47 | elriah | Using ulaw. |
| 00:16.54 | elriah | In 1.2.18 |
| 00:17.17 | davidcsi | what's on the other side? |
| 00:17.32 | elriah | Another asterisk box. |
| 00:18.07 | elriah | Seemds to only be an issue with calls from ye-ole telco ... Doesn't seem to be an issue with IP only calls. |
| 00:18.34 | elriah | But that's just an assumption based on what we've seen. It doesn't happen all the time. |
| 00:19.41 | davidcsi | have you tried dtmf=info? |
| 00:20.05 | Strom_M | dtmfmode |
| 00:20.07 | Strom_M | not dtmf= |
| 00:20.40 | Strom_M | elriah: if you're doing sip-to-sip on asterisk, dtmfmode should be rfc2833 |
| 00:20.44 | elriah | Nope, I've always read that rfc2833 is the best? |
| 00:20.50 | davidcsi | thats it |
| 00:20.59 | davidcsi | i solce it once with info |
| 00:21.08 | davidcsi | s/c/s/ |
| 00:21.20 | davidcsi | s/c/v/ :D |
| 00:21.29 | Strom_M | elriah: the calls from ye olde telco are coming into your asterisk box via...PRI? |
| 00:21.32 | davidcsi | solved it |
| 00:21.34 | davidcsi | jeez |
| 00:22.18 | elriah | Strom_M: I assume such, the other side is Vitelity. |
| 00:22.32 | Strom_M | how are they bringing calls into you? |
| 00:23.06 | davidcsi | so you're RECEIVING bad dtmf... |
| 00:23.23 | elriah | Possible... |
| 00:23.52 | Strom_M | elriah: what method is vitelity using to deliver calls to you? |
| 00:23.55 | Strom_M | iax? sip? |
| 00:24.29 | elriah | Strom_M: Sip. |
| 00:24.50 | davidcsi | i've done that hundreds of times between asterisk boxes... no problems... ever... try dtmfmode=info... |
| 00:25.28 | elriah | Ok, dtmfmode=info doesn't work, doesn't forward any dtmf that way. Inband is out of the question. So Vitelity has a dtmf issue. |
| 00:25.57 | davidcsi | they MUST configure dtmfmode=info as well |
| 00:25.58 | Strom_M | elriah: yeah, i'd blame 'em |
| 00:26.08 | Strom_M | davidcsi: info blows as a dtmfmode |
| 00:26.10 | davidcsi | ditto |
| 00:26.12 | Strom_M | rfc2833 is sexier |
| 00:26.32 | davidcsi | you want it to be sexy or to work? ;) |
| 00:26.46 | elriah | That's what I thought, I was hoping there was some pixie dust I could sprinkle on it .. oh well, thanks, all!!! |
| 00:26.47 | Strom_M | sexier as in "works more efficiently" :) |
| 00:27.26 | elriah | Is one codec better than others with dtmf? |
| 00:27.40 | Strom_M | elriah: with rfc2833 codec doesnt matter |
| 00:27.43 | JT | elriah: dtmf only works on G.711, nothing else. |
| 00:27.48 | Strom_M | otherwise, use g711 |
| 00:27.51 | JT | for inband |
| 00:28.17 | davidcsi | anyone ever got it to really work with cisco? |
| 00:28.30 | JT | ciscos working, hah |
| 00:28.48 | davidcsi | that's what i thought. |
| 00:29.02 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 00:30.10 | elriah | For clarification, with rfc2833, the DTMF info is sent directly in the sip packet, right? Not inband in the RTP stream? So if my asterisk box is getting "123444444" when it should be getting "1234", and it's a sip-to-sip connection, it has to be on the sending end? Is this correct? |
| 00:30.23 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 00:30.25 | JT | rfc2833 is RTP |
| 00:30.31 | JT | info is SIP |
| 00:30.48 | elriah | So is it send as actual tones in the audio stream? |
| 00:30.52 | davidcsi | that's why i said: "try info" |
| 00:31.03 | elriah | davidcsi: yea... I'm seing that... |
| 00:31.05 | JT | elriah: obviously not, that's what g.711 inband is for |
| 00:31.24 | JT | elriah: it's sending RTP packets that tell it to make a dtmf noise, and for how long |
| 00:31.40 | davidcsi | yep |
| 00:31.53 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-18-1-84.nycmny.east.verizon.net) |
| 00:32.19 | davidcsi | so, is there anyone out there who can answer my question? |
| 00:32.47 | elriah | Got it. So it's still quite probable it's all on the sending end ... |
| 00:34.41 | davidcsi | but, if the sending end it itself RECEIVING wrong... forget it... just a thought... you should ask how the sending end is getting its dtmf |
| 00:35.11 | elriah | I'm just looking for a "solid" way to blame them so their support can't give me the royal blow-back ... |
| 00:37.58 | davidcsi | make an ethereal trace |
| 00:39.02 | *** join/#asterisk ManxPower (n=manxpowe@032-493-418.area7.spcsdns.net) |
| 00:40.20 | elriah | Oh dear, I think I'd rather go to the dentist. We process like 10,000 calls a day. |
| 00:40.26 | elriah | lol, that would suck digging through that data. |
| 00:40.54 | JT | you can restrict it to an ip |
| 00:41.35 | davidcsi | filter it by ipī |
| 00:41.55 | elriah | It's all from Vitelity ... Filtering by IP wouldn't help ... |
| 00:41.58 | *** join/#asterisk hyphen (n=hyphen@c-71-224-214-148.hsd1.pa.comcast.net) |
| 00:42.07 | elriah | Well, we have about 2000 that come from les.net |
| 00:44.25 | elriah | We basically have 2 OpenSER boxes (for internal phone registration handling and internal call setup), 4 asterisk boxes, and two MySQL back-ends. |
| 00:44.34 | *** join/#asterisk heh_v_water (n=heh_v_wa@207-225-3-205.hlna.qwest.net) |
| 00:44.41 | elriah | We're using realtime-static plus flat-file for some configs. |
| 00:45.03 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 00:45.47 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-a5c502138331f32a) |
| 00:45.49 | elriah | Gotta run, thanks for the help. |
| 00:45.53 | davidcsi | tshark -f "host [ip]" | grep "RTP EVENT Payload" |
| 00:46.20 | Shoeb | [TK]D-Fender! I'm back. |
| 00:46.34 | Shoeb | Got our guy at the callcentre to go in and fix it! |
| 00:46.45 | Shoeb | At the datacentre, lol, what am I saying |
| 00:47.07 | Shoeb | Oh, no! Looks like you're not here. :( |
| 00:47.58 | Shoeb | Ok question to others, what are the prerequisites of creating .call files and moving them to the appropriate directory to make asterisk make the call? |
| 00:48.02 | *** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net) |
| 00:48.41 | Shoeb | Currently, a web php script when started creates the .call files and tries to move it to the the directory but encountering permission problems. The user that creates it is apache. |
| 00:48.48 | Shoeb | Any help in that regard? |
| 00:50.17 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:54.38 | ManxPower | <PROTECTED> |
| 00:54.58 | *** part/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 00:55.08 | JT | nic |
| 00:55.10 | JT | e |
| 00:55.35 | ManxPower | Shoeb: the file must be readable by Asterisk. I recommend creating the file in a different directory on the same filesystem, then moving it to the correct place. Make the ctime of the file in the future if you don't want asterisk to process it right away |
| 00:55.47 | ManxPower | JT: They are already like 3 months behind schedule. |
| 00:56.36 | Shoeb | ManxPower: It's not about that. Asterisk is running as root. The .call files are being made by user apache. Do you know what I mean now? |
| 00:57.07 | ManxPower | Shoeb: Is Asterisk not processing the files? |
| 00:58.22 | Shoeb | It's not letting us move the callfile to the spool dir. |
| 00:58.47 | Shoeb | Permission problems. |
| 00:58.58 | Shoeb | Not permitted to "write" to the outgoing director. |
| 00:59.19 | ManxPower | Shoeb: that is not an Asterisk issue. |
| 00:59.33 | ManxPower | the apache user must have write access to that directory |
| 00:59.50 | Shoeb | Ahh! |
| 00:59.52 | Shoeb | Hmm, I'm not the best when it comes to that. |
| 01:00.11 | Shoeb | Do you mind pointing me to a resource that can show me how to give "apache" write acccess to that folder? |
| 01:00.42 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:00.42 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:01.49 | Shoeb | that 'directory', sorry, heh |
| 01:02.26 | ManxPower | Shoeb: it is a basic linux admin thing. you'll have major other problems if you don't have someone that knows how to admin linux. The easiest thing to do is chown root.apache /var/spool/asterisk/outgoing |
| 01:02.32 | ManxPower | or whatever group your apache user is in |
| 01:02.42 | Shoeb | It's apache |
| 01:02.47 | Shoeb | The group.. |
| 01:05.55 | Shoeb | ManxPower: I do have some people for that, but I'm also trying to learn it so these guys can't take me for a fool. |
| 01:07.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:09.43 | Shoeb | Hmm. |
| 01:09.48 | Shoeb | Did that, and it's still not letting. |
| 01:10.40 | [TK]D-Fender | Shoeb, Ditch that method entirely and use AMI Originate instead. |
| 01:10.41 | [TK]D-Fender | ~ami |
| 01:10.42 | jbot | ami is probably the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 01:10.46 | [TK]D-Fender | ~wikis |
| 01:10.47 | jbot | i guess wikis is http://www.voip-info.org |
| 01:10.59 | Shoeb | Hmm, manager API, huh. |
| 01:11.20 | Shoeb | But that needs to be done manually, right? |
| 01:15.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:19.20 | Shoeb | [TK]D-Fender: I'm going to get us bck on track tmrw mornin. :) |
| 01:19.24 | Shoeb | You rock btw. |
| 01:22.53 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 01:23.05 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-182-96.hsd1.or.comcast.net) |
| 01:27.55 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 01:28.56 | fujin | hi |
| 01:29.06 | fujin | could anyone tell me if agents require a password? |
| 01:29.40 | fujin | or if it can be blank |
| 01:30.18 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:34.06 | mmlj4 | agents? |
| 01:34.36 | fujin | yes |
| 01:34.37 | fujin | agents |
| 01:35.00 | fujin | you know, the agents that queues have? |
| 01:36.13 | rbd | hi, in my extensions.conf for a given extension, I'd like to set a channel variable from a custom sip header only if the call is coming in on a SIP channel, if not, it should be blank. Will SIP_HEADER() work like this (e.g. not error out for non-sip channels, but just return blank values)? |
| 01:37.59 | rbd | or, do I need some logic there to determine if a SIP channel is in use and go from there (this is because I have some extensions where calls can come in on SIP or Local channel types) |
| 01:37.59 | [TK]D-Fender | rbd, You can test to see if its a SIP channel anyways.... |
| 01:38.51 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 01:39.00 | rbd | hmm, it looks like ${CHANNEL(channeltype)} is the way to get that info |
| 01:39.50 | [TK]D-Fender | rbd, One of a few |
| 01:40.25 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 01:40.33 | rbd | yeah, I see that now. thanks |
| 01:45.54 | ManxPower | I don't suppose anyone knows of a good place to order DOUBLE sided BLACK AND WHITE business cards online for delivery to the USA? |
| 01:46.38 | JT | it's amazing how many variables there are these days to order business cards |
| 01:47.08 | ManxPower | JT: they all want me to pick a logo or a color logo. |
| 01:47.25 | JT | do you have a choice of laser or offset printing? |
| 01:47.28 | ManxPower | they will be "bar cards", not business cards. |
| 01:47.43 | JT | what's a bar card? |
| 01:48.01 | JT | it's a lot easier to do business cards if you already have all the artwork |
| 01:48.04 | ManxPower | JT: give it to someone you meet at a bar and hope they call you to have sex |
| 01:48.28 | JT | oh |
| 01:48.32 | ManxPower | or hell, give one to someone you had a one night stand with in hopes they call again |
| 01:48.35 | JT | i have a much better solution for that |
| 01:48.48 | ManxPower | The correct term in *my* world is "trick card". |
| 01:49.05 | ManxPower | it beats writing a phone number or e-mail address on a matchbook |
| 01:49.12 | JT | ManxPower: moo.com |
| 01:49.13 | *** part/#asterisk ManxPower (n=manxpowe@032-493-418.area7.spcsdns.net) |
| 01:49.20 | *** join/#asterisk ManxPower (n=manxpowe@032-493-418.area7.spcsdns.net) |
| 01:49.23 | JT | ManxPower: moo.com |
| 01:49.35 | ManxPower | thanks |
| 01:49.42 | JT | ManxPower: they're about half the height of a normal business card |
| 01:49.45 | JT | can be full colour |
| 01:49.58 | JT | can be all the same, or every single one can have a different picture |
| 01:50.09 | JT | seen them in person, they have quite a nice semigloss finish |
| 01:50.19 | JT | shipped from the uk |
| 01:50.55 | ManxPower | I don't WANT a picture. |
| 01:51.09 | *** join/#asterisk aao_pwner (n=obnautic@c-67-160-182-96.hsd1.or.comcast.net) |
| 01:51.28 | ManxPower | JT: www.fnords.org is my web site. Take a look at it to get an idea what my idea of "graphic design" is. |
| 01:51.30 | blitzrage | I don't want to meet your mom! |
| 01:51.31 | JT | doesn't have to be of YOU ;) or a person even |
| 01:51.49 | JT | heh |
| 01:52.08 | JT | i think they make it easy to do templated layouts |
| 01:52.13 | JT | that said i've never ordered |
| 01:52.30 | ManxPower | I'll go to a bricks and mortar printer. |
| 01:52.43 | JT | shrug |
| 01:52.56 | JT | don't know what was so bad about moo cards, heh |
| 01:53.37 | ManxPower | JT: I kept expecting an animated pink My Pretty Pony to dance across the screen |
| 01:53.39 | [TK]D-Fender | blitzrage, I JUST WANT |
| 01:53.45 | blitzrage | ! ! ! |
| 01:53.51 | [TK]D-Fender | zomg |
| 01:53.57 | JT | ManxPower: not sure if that would be Web 2.0 compliant |
| 01:54.35 | [TK]D-Fender | My printers tend to be made out of plastic and metal primarily.... |
| 01:55.05 | [TK]D-Fender | ManxPower, I suspect you meant to say "My Little Pony" |
| 01:55.12 | coldsteal | how can i say what ext its transfering to? |
| 01:55.40 | coldsteal | like dynamicly |
| 01:55.49 | [TK]D-Fender | coldsteal, Prease attempt to avoid using nondescript pronouns like "it" without prodiving context... |
| 01:56.10 | ManxPower | [TK]D-Fender: no. My Pretty Pony is a childrens toy, for girls, maybe age 6 |
| 01:56.37 | [TK]D-Fender | ManxPower, Same for my reference, circa 80's fad/cartoon, etc |
| 01:56.50 | [TK]D-Fender | ManxPower, by Hasbro |
| 01:56.51 | coldsteal | [TK]D-Fender: okay well how do i have * say something like "te person at ext# could not be reached" |
| 01:56.59 | coldsteal | *the |
| 01:57.03 | ManxPower | [TK]D-Fender: Yes, they are related products, as I just discovered |
| 01:57.12 | [TK]D-Fender | coldsteal, "Playback(soundfilethatsayswhatyouwant) |
| 01:57.35 | coldsteal | [TK]D-Fender: so i cant have the ext # be dynamic |
| 01:57.45 | [TK]D-Fender | coldsteal, you can record something yourself with Record". |
| 01:57.51 | ManxPower | http://en.wikipedia.org/wiki/Image:MyLittlePony-RunawayRainbow.jpg |
| 01:58.12 | [TK]D-Fender | coldsteal, here : "Playback(thepersonatext)" |
| 01:58.33 | [TK]D-Fender | coldsteal, here : "SayDigits(${varwiththeext})" |
| 01:58.43 | [TK]D-Fender | coldsteal, here : "Playback(isnotavailable)" |
| 01:58.52 | *** join/#asterisk saftsack (n=saftsack@pD9E07124.dip.t-dialin.net) |
| 01:59.08 | [TK]D-Fender | ManxPower, indeed as I suspected. Have you seen "Transformers" yet? |
| 01:59.25 | ManxPower | [TK]D-Fender: nope. |
| 01:59.32 | [TK]D-Fender | coldsteal, You'd say it in 3 steps |
| 01:59.36 | [TK]D-Fender | ManxPower, Seen the ads? |
| 01:59.37 | ManxPower | The BF wants to. I'd rather see Underdog |
| 02:00.03 | coldsteal | isnotavailable is a file that i record right |
| 02:00.31 | coldsteal | * wont actually read what i put in it |
| 02:00.48 | [TK]D-Fender | coldsteal, Correct. you would use the Record app to make those |
| 02:01.32 | [TK]D-Fender | coldsteal, * also comes with a lot of useful bits. This exact combo I believe is used in the VM subsystem, so you could probably do it from stock recordings all done by Allison |
| 02:01.47 | ManxPower | sounds.txt lists the text of all the sounds in asterisk |
| 02:02.08 | coldsteal | but it will say SayDigits(${ext})? |
| 02:02.35 | ManxPower | coldsteal: not unless you set EXT to something. Most people just use ${EXTEN} |
| 02:02.53 | [TK]D-Fender | coldsteal, that is an app that rill read whatever digits you want out. I gave you a sample where it would read based on the contexts of a variable |
| 02:03.10 | [TK]D-Fender | coldsteal, You can do this direct for example SayDigits(12345) |
| 02:03.53 | [TK]D-Fender | coldsteal, My sample was entirely fictitious jsut to give you an idea without implying where said data originated. |
| 02:04.04 | coldsteal | ok |
| 02:04.30 | BSD_Tech | tk I will look for you tomarrow |
| 02:05.47 | [TK]D-Fender | BSD_Tech, k |
| 02:06.00 | coldsteal | i installed * from apt so how do i check if i have the default sound recordings |
| 02:06.05 | [TK]D-Fender | BSD_Tech, setup a SIP entry for the passthrough and I'll help with the scripting then |
| 02:06.23 | [TK]D-Fender | coldsteal, check out /var/lib/asterisk/sounds |
| 02:06.47 | [TK]D-Fender | coldsteal, thats the default place anyways |
| 02:06.53 | coldsteal | cool i have it |
| 02:09.41 | *** join/#asterisk ukris (n=ukris@aa20060807547d355914.userreverse.dion.ne.jp) |
| 02:13.13 | *** join/#asterisk GothAlice (n=amcgrego@190.140.153.199) |
| 02:14.15 | GothAlice | I have three phones connecting to Asterisk; two as the same SIP user. On the third I try calling the shared SIP account and only one phone rings. It appears to be the last phone to register takes over the SIP mapping. How can I have one account on multiple phones? |
| 02:14.55 | [TK]D-Fender | GothAlice, you CAN'T |
| 02:15.05 | GothAlice | Well, that blows chunks. |
| 02:15.09 | [TK]D-Fender | GothAlice, Each phone has to register as a different account |
| 02:15.17 | GothAlice | The only way to do this is to have a ring group, eh? |
| 02:15.33 | [TK]D-Fender | GothAlice, Inappropriate term, but essentially, yes |
| 02:16.17 | GothAlice | (freePBX calls them ring groups in the menu.) |
| 02:16.25 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 02:16.35 | [TK]D-Fender | GothAlice, hence "inappropriate". |
| 02:16.40 | GothAlice | XD |
| 02:16.41 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:16.52 | JT | most rerms that freepbx use are innapropriate ;) |
| 02:16.56 | JT | terms |
| 02:16.59 | GothAlice | How can I have a person's presence follow them from phone to phone, then, and provide a single extension to reach them wherever they are? |
| 02:17.04 | [TK]D-Fender | GothAlice, FreePBX bastardizes proper telecom terminology throughout its interface |
| 02:17.28 | [TK]D-Fender | GothAlice, You can do this with a whole pile of dialplan logic. |
| 02:17.44 | GothAlice | ... but not if it's managed by freePBX? ;^) |
| 02:17.47 | [TK]D-Fender | GothAlice, Quite an amount of work... |
| 02:18.12 | [TK]D-Fender | GothAlice, words can barely express how much help you AREN'T going to get with that here.... |
| 02:20.09 | GothAlice | (I know how to patch custom stuff into freePBX. I understand that this room is for asterisk alone.) My original idea was to have a bunch of muti-line phones each with an account based on its physical location (office, kitchen, bedroom, etc.) then have "user" accounts which are shared among them as appropriate (john at the office and bedroom, linda at the bedroom, etc.) and still allow X-Lite softclients, wi-fi phones and other goodies. |
| 02:20.28 | GothAlice | Are there any example configs? |
| 02:20.55 | JT | i suggest takeing a look at the book, GothAlice |
| 02:20.57 | JT | ~thebook |
| 02:20.57 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:21.01 | GothAlice | I could probably do it with a queue, where the user registers with their own queue on each phone they touch? |
| 02:21.03 | JT | and avoiding freepbx |
| 02:21.09 | *** join/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 02:21.59 | *** join/#asterisk nath0099 (n=James@77-96-249-177.cable.ubr02.maid.blueyonder.co.uk) |
| 02:23.02 | *** join/#asterisk jefforulez (n=jeffo@pool-71-187-66-159.nwrknj.fios.verizon.net) |
| 02:23.19 | [TK]D-Fender | GothAlice, No, there are no "samples" really, because this actually entails a fair amout of work storing the identity info, login info, dailaplan checking, custom dial macros (kiss FreePBX GOODBY because of this), tec |
| 02:23.45 | [TK]D-Fender | GothAlice, problem is that your CID wouldn't follow. |
| 02:23.52 | [TK]D-Fender | goth and queues.... FUGLY |
| 02:24.08 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-18-1-84.nycmny.east.verizon.net) |
| 02:24.27 | JT | ? |
| 02:24.41 | GothAlice | Outbound CID wouldn't, but inbound would. That may be sufficient for me. (Esp. as I can prefix the inbound CID with the appropriate name...) |
| 02:25.21 | [TK]D-Fender | GothAlice, think twice about that.... when you call out you get the PHONE'S CID, not the person "logged in" |
| 02:25.28 | JT | well that bento box was excellent |
| 02:25.42 | [TK]D-Fender | JT ? |
| 02:25.52 | [TK]D-Fender | JT : Sushi? |
| 02:26.03 | JT | bento box is like a japanese combo box |
| 02:26.05 | JT | some sushi |
| 02:26.07 | jerlique | How should a bri-SIP gateway be configured in *? |
| 02:26.23 | GothAlice | The phone CID, when dialing outside the local, will be set to a reasonable dial-back DID with a generic name. Internally it's good to get directly back to the phone which originated the call you're returning. Hmmm. |
| 02:26.26 | JT | also had tempura calimari, kegetable, fried port, some salad, some potato stuff |
| 02:26.35 | JT | jerlique: as a sip friend... |
| 02:26.38 | [TK]D-Fender | JT : We have a brand here for prepared lunch/snack sushi platters byt he name "Bento Nouveau" |
| 02:26.43 | JT | ah |
| 02:27.00 | JT | [TK]D-Fender: this is made fresh by a japanese at the store :) |
| 02:27.07 | JT | a japanese chef |
| 02:27.09 | JT | even |
| 02:27.52 | GothAlice | ^_^ Sushi Express here kicks ass, though some of the combinations are strange, and all of them include cream cheese. Plus its made by panamanians who can't speak a word of Japanese, which ruins the effect. |
| 02:28.34 | JT | i hear that it's hard to get much authentic multicultural/ethnic food in the us |
| 02:28.58 | jerlique | JT: Ok, this is what I have done. What would be the reason that * can process DMTF from telephones, but it cannot accept DTMF from the sip gateway? |
| 02:29.20 | JT | jerlique: wrong dtmf mode |
| 02:29.31 | jerlique | (The sip debug shows the message coming through, but * says unauthorised.) What modes are there? |
| 02:29.45 | JT | well, are calls working? |
| 02:29.56 | jerlique | Yes. |
| 02:29.59 | jerlique | bothin in and out |
| 02:30.07 | JT | what dtmfmode is set in sip.conf? |
| 02:30.11 | jerlique | info |
| 02:30.13 | JT | what is set in the gateway? |
| 02:30.26 | JT | you should usually use rfc2833 by default |
| 02:30.35 | Daejeo1 | JT: hello :) |
| 02:30.38 | jerlique | info is set in the gw |
| 02:30.40 | JT | hi |
| 02:30.50 | JT | set them both to rfc2833 |
| 02:30.58 | jerlique | ok let me try... |
| 02:32.26 | *** join/#asterisk ChrisTSIS (n=killa666@24.182.21.208) |
| 02:33.52 | ChrisTSIS | Is there any way to kill this channel w/o restarting asterisk? Soft hangup isn't dropping it: Zap/16-1 s@macro-hangupcall:1 Up (None) |
| 02:33.52 | `Sean | if there |
| 02:34.07 | `Sean | anyway to get asterisk to not ask for recording of name when u join a conf |
| 02:34.09 | `Sean | or play any msgs |
| 02:34.12 | [TK]D-Fender | ChrisTSIS, show us |
| 02:34.27 | jerlique | JT: that still didnt work. |
| 02:34.47 | ChrisTSIS | [TK]D-Fender: Show you what? |
| 02:35.04 | `Sean | anyone?? is there a way to get asterisk when a user calls to shove them straight into conf |
| 02:35.13 | `Sean | without asking them to record a username etc or anything like that |
| 02:35.18 | blitzrage | sure... |
| 02:35.21 | blitzrage | don't use those options |
| 02:35.32 | blitzrage | by default it won't ask for that stuff |
| 02:36.04 | *** join/#asterisk Shizuo (n=pato@200-171-49-211.dsl.telesp.net.br) |
| 02:36.05 | Shizuo | ccesario -> HOMO |
| 02:36.28 | `Sean | blitzrage it does want me to paste, my meetme.conf |
| 02:36.30 | `Sean | it for some reason does |
| 02:36.42 | blitzrage | ~pb |
| 02:36.43 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:36.43 | [TK]D-Fender | ChrisTSIS, Show us how you're attempting to kill the channel and the outcome. |
| 02:37.03 | JT | jerlique: maybe the gateway is not detecting the DTMF |
| 02:37.07 | *** part/#asterisk Shizuo (n=pato@200-171-49-211.dsl.telesp.net.br) |
| 02:37.23 | blitzrage | `Sean: why not just do: MeetMe(123|d) ? |
| 02:37.35 | `Sean | in console? |
| 02:37.39 | blitzrage | no... |
| 02:37.40 | blitzrage | in the dialplan |
| 02:37.53 | blitzrage | where you control the route of the call |
| 02:38.08 | `Sean | [conf1] |
| 02:38.08 | `Sean | exten => s,1,MeetMe(6180,i,3355) |
| 02:38.11 | `Sean | thats what i have |
| 02:38.13 | blitzrage | ignore that |
| 02:38.24 | `Sean | is it because of the I? |
| 02:38.34 | blitzrage | 'i' -- announce user join/leave with review |
| 02:38.52 | blitzrage | *CLI> show application meetme |
| 02:38.55 | blitzrage | it's magical |
| 02:39.05 | jerlique | JT: we can use DTMF going out to an external site, eg sipphone -> * - >SIPGW --->PSTN DTMF here, and furthermore, * is receiving the fact that I am punching in DTMF numbers, it just rejects them with |
| 02:39.07 | blitzrage | and the 3355 is going to ask for a pin |
| 02:39.49 | `Sean | 3355 is te pin... |
| 02:39.52 | JT | jerlique: what are you punching number in to? |
| 02:39.54 | `Sean | s/te/the/ |
| 02:40.10 | `Sean | i only announces user/join/leave |
| 02:40.19 | `Sean | even if i remove that |
| 02:40.25 | `Sean | it will still ask me to record a username |
| 02:40.27 | jerlique | my cell phone. The test I am doing is cell phone-> PSTN-> SIPGW -> * |
| 02:40.29 | `Sean | and review it |
| 02:40.37 | ChrisTSIS | [TK]D-Fender: http://pastebin.com/d406573e5 |
| 02:40.47 | *** join/#asterisk Shizuo (n=pato@200-171-49-211.dsl.telesp.net.br) |
| 02:40.54 | Shizuo | JI said the channel is irrelevant |
| 02:40.56 | Shizuo | JT |
| 02:40.57 | *** part/#asterisk GothAlice (n=amcgrego@190.140.153.199) |
| 02:41.01 | Shizuo | JT said the channel is irrelevant |
| 02:41.02 | *** part/#asterisk Shizuo (n=pato@200-171-49-211.dsl.telesp.net.br) |
| 02:41.06 | JT | Shizuo: you're an idiot, fuck off |
| 02:41.11 | `Sean | http://pastebin.ca/633998 |
| 02:41.15 | `Sean | blitzrage, http://pastebin.ca/633998 |
| 02:41.22 | `Sean | look at all the things it does by deual playing those files |
| 02:41.23 | coldsteal | when i try to Backdround(/path/to/file.gsm) i get Unable to open No such file or directory |
| 02:42.02 | shido6 | grrr |
| 02:42.12 | coldsteal | o i guess i dont put .gsm |
| 02:42.19 | shido6 | stick the file in /var/lib/asterisk/sounds |
| 02:42.20 | JT | coldsteal: you don't |
| 02:42.24 | shido6 | and u dont have to do all that |
| 02:42.34 | shido6 | take the suffix off |
| 02:42.40 | [TK]D-Fender | ChrisTSIS, Umm... yikes |
| 02:43.02 | shido6 | bleh.gsm would be referred to as bleh , so u get exten => s,1,Playback,bleh |
| 02:43.27 | ChrisTSIS | [TK]D-Fender: hence the question... otherwise I have to dump a bunch of active channels and agents logged into queues |
| 02:43.42 | *** join/#asterisk MdeP (n=mdep@103-84-22-190.adsl.tie.cl) |
| 02:44.11 | [TK]D-Fender | ChrisTSIS, I suspect you cleanest salvations is "restart gracefully" |
| 02:44.34 | [TK]D-Fender | ChrisTSIS, or reloading chan_zap.so when no zap channels are in use |
| 02:44.48 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 02:45.44 | ChrisTSIS | [TK]D-Fender: My fear is it won't restart gracefully because of the locked open channel |
| 02:45.52 | [TK]D-Fender | ChrisTSIS, Oh, it WILL.... |
| 02:46.14 | [TK]D-Fender | ChrisTSIS, * has little trouble STOPPING :) Heck it may even do it without you requesting it to! |
| 02:46.15 | [TK]D-Fender | ;) |
| 02:46.35 | jgoddess | hehe |
| 02:46.42 | ChrisTSIS | True, but normally with gracefully it waits until channels are all dropped |
| 02:46.54 | ChrisTSIS | oh well, I was just hoping to not have to stay up until 12am to do it |
| 02:48.49 | [TK]D-Fender | ChrisTSIS, "restart now" .... ummmm guys..... something funny er... happened with the server.... not sure what, but I think everything's OK now.... |
| 02:49.19 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 02:52.39 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:56.38 | *** join/#asterisk CrazyTux (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 03:00.33 | *** part/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 03:02.48 | JT | jerlique: outgoing DTMF has nothing to do with incoming DTMF detection |
| 03:03.03 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 03:03.22 | *** join/#asterisk joshr (n=joshr@65.103.116.63) |
| 03:06.32 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 03:12.09 | *** join/#asterisk Shizuo (n=pato@200-171-49-211.dsl.telesp.net.br) |
| 03:12.12 | Shizuo | JT = HOMO |
| 03:12.13 | *** part/#asterisk Shizuo (n=pato@200-171-49-211.dsl.telesp.net.br) |
| 03:13.58 | JT | what a twit |
| 03:19.43 | coldsteal | how do i have an ext as # |
| 03:20.10 | blitzrage | what do you mean? |
| 03:20.32 | blitzrage | please elaborate with your question so we may understand what you are trying to do |
| 03:20.44 | coldsteal | well for my IVR i have "press # for dir index" |
| 03:21.01 | coldsteal | but i get an error of Invalid extension '#', but no rule 'i' in context 'incoming' |
| 03:21.07 | [TK]D-Fender | coldsteal, "exten => #,1,NoOp(yippy-kay-yay-mo.....) |
| 03:21.28 | [TK]D-Fender | coldsteal, Yes, it's THAT easy... |
| 03:21.47 | coldsteal | what does NoOp mean> |
| 03:21.48 | coldsteal | ? |
| 03:21.57 | [TK]D-Fender | coldsteal, Its jsut an app, like any other. |
| 03:22.11 | [TK]D-Fender | coldsteal, You can do whatever you want. |
| 03:22.23 | coldsteal | i was f=trying goto |
| 03:22.26 | coldsteal | *trying |
| 03:22.33 | [TK]D-Fender | coldsteal, Please review the chapter on dialplan patterns..... this is serious 101 stuff... |
| 03:22.36 | [TK]D-Fender | ~book |
| 03:22.36 | 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 |
| 03:22.39 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 03:23.11 | coldsteal | okay thanks |
| 03:24.03 | honeybeebuzz | on hardware phone, I need some suggestion about what is the good start to buy? |
| 03:24.27 | [TK]D-Fender | honeybeebuzz, What kind of hardware phone? |
| 03:25.14 | honeybeebuzz | a basic, home purpose asterisk/trixbox voip phone |
| 03:25.47 | [TK]D-Fender | honeybeebuzz, Polycom IP 320. |
| 03:26.05 | JT | trixbox, naughty |
| 03:26.29 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 03:26.37 | honeybeebuzz | okey... how you compare with GrandStream GXP-2000, in term of functionlity. |
| 03:26.45 | [TK]D-Fender | honeybeebuzz, ... |
| 03:26.46 | [TK]D-Fender | ~gs |
| 03:26.47 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 03:26.48 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 03:26.55 | JT | grandstreams are piles of rubbish |
| 03:27.26 | honeybeebuzz | hmm... I never seen them in my life this far :) |
| 03:27.35 | JT | that's a good thing |
| 03:27.51 | Sci_05 | JT greandstreams are not that good.....they are worse then rubbish |
| 03:27.54 | [TK]D-Fender | honeybeebuzz, GS is known for flakey firmware, shoddy construction, poor audio quality, etc. |
| 03:27.57 | honeybeebuzz | okey ... you think that Polycom IP 320 is a good start? |
| 03:28.20 | [TK]D-Fender | honeybeebuzz, super solid phone. |
| 03:28.38 | [TK]D-Fender | honeybeebuzz, Linksys are "acceptable" as well |
| 03:28.38 | JT | Sci_05: it's amusing, there was a guy here last night saying how good they were (GXP-2000s) |
| 03:29.10 | [TK]D-Fender | JT : Yes, I remember him :) Doesn't hold under scrutiny though... |
| 03:29.18 | JT | hehe |
| 03:29.50 | JT | [TK]D-Fender: "but i've worked at the biggest voip phone store ever and have sold 4734737883 phones!!11" |
| 03:30.32 | wunderkin | oneoneoneone |
| 03:30.37 | [TK]D-Fender | JT : And that could be entirely accurate. Chumps buy the cheapest things they can get their hands on.... |
| 03:31.02 | JT | [TK]D-Fender: he claimed to have used polycom and the audio was no different on handset |
| 03:31.04 | [TK]D-Fender | JT : I've been researching like MAD for a high VALUE & QUALITY ultra-zoom digital camera.... |
| 03:31.15 | JT | [TK]D-Fender: slr? |
| 03:31.20 | [TK]D-Fender | JT : I believe he said "not that bad" |
| 03:32.09 | [TK]D-Fender | JT : Just showing my approach as different with regard to purchases. My range has started as low as 300$, and my current model of choice is $409 currently. A price I'm willing to pay... |
| 03:32.36 | [TK]D-Fender | JT : though I'm still looking if there is something better within range. |
| 03:32.37 | JT | [TK]D-Fender: ok, not a DSLR then :) |
| 03:32.45 | honeybeebuzz | http://www.cdw.ca/shop/products/default.aspx?EDC=1210343 is the one you guys refered? |
| 03:32.50 | [TK]D-Fender | JT : No, a point& shoot 10x+ zoom. |
| 03:32.51 | JT | actually, you probably could |
| 03:33.11 | JT | can't remember what a D40 with 18-55mm lens kit is in USD |
| 03:33.19 | JT | Nikon D40 |
| 03:33.22 | [TK]D-Fender | honeybeebuzz, You can get it for $87.50 USD retail. make sure your purchase SCALES accordingly. |
| 03:33.23 | JT | pretty compact slr |
| 03:33.38 | Nugget | I just recently bought a DSLR. It's nice having a good quality camera if I want one, but I still like my point-n-click better because it fits in my pocket and I always have it with me |
| 03:33.38 | [TK]D-Fender | JT : yes, the D40 is nice, as is the D80.. but I'm not a photographer. |
| 03:33.49 | JT | you don't have to be... |
| 03:33.56 | JT | auto mode works fine |
| 03:34.10 | [TK]D-Fender | JT : I jsut want a GOOD point & shoot with big-zoom & OIS |
| 03:34.17 | Nugget | pictures are so much more compelling when you can play with focal length, though. |
| 03:34.27 | [TK]D-Fender | JT : Oh.... and the camera isn't worth THAT much to me :) |
| 03:34.28 | Nugget | the "everything's in perfect" focus of a point-n-click looks really cheap |
| 03:34.39 | JT | big zoom, the zoom on most compacts is pretty crappy |
| 03:34.41 | honeybeebuzz | what is SCALES? |
| 03:35.06 | [TK]D-Fender | honeybeebuzz, 87.50$ USD != 166.69$ CAD |
| 03:35.12 | [TK]D-Fender | honeybeebuzz, www.xe.com |
| 03:35.18 | [TK]D-Fender | honeybeebuzz, CDW = SHIT. |
| 03:35.26 | Nugget | http://macnugget.org/photos/2007c2s/DSC_0169 <-- you just can't do that with the small zoom cameras. |
| 03:35.39 | JT | Nugget: what camera do you have? |
| 03:35.54 | [TK]D-Fender | Nugget, kills my budget however. |
| 03:36.08 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:36.17 | JT | [TK]D-Fender: was that 400CAD before? |
| 03:36.35 | [TK]D-Fender | JT : yes. |
| 03:36.40 | JT | ah ok |
| 03:36.48 | JT | a D40 is around 600CAD |
| 03:36.57 | Nugget | I bought a D80, mostly because my business partner has one and he's spent a scary amount of money on fancy lenses. |
| 03:37.01 | [TK]D-Fender | JT :this is what I'm loking at http://www.dpreview.com/reviews/panasonicfz8/ |
| 03:37.02 | Nugget | so I wanted to be able to borrow his lenses |
| 03:37.04 | [TK]D-Fender | JT : Where? |
| 03:37.19 | JT | mind you, they could be body only |
| 03:37.21 | JT | haven't checked |
| 03:37.22 | JT | http://www.shopbot.ca/p-36191.html |
| 03:37.25 | [TK]D-Fender | Nugget, Yes, that offsets the base cost allright ;) |
| 03:37.57 | JT | Nugget: the fancy lenses work with much less fancy cameras ;) |
| 03:38.20 | JT | i'd prefer a D200 to a D80, but D200s are still hell expensive |
| 03:38.57 | *** join/#asterisk nvicf (n=nvicf@201.250.181.27) |
| 03:38.58 | Nugget | I'm happy enough with the D80. The additional benefit of a D200 would be lost on me, I think. |
| 03:39.33 | JT | i'm quite happy with my D70s too, there's actually some specs that are lower on the D80, like flash sync |
| 03:39.44 | JT | i'd appreciate the extra 2 fps on a D200 |
| 03:39.45 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.211.11.revip2.asianet.co.th) |
| 03:40.05 | JT | i'm worried about how long my D70s, hence considering a new body one day |
| 03:40.24 | JT | i think mine is at around 35000 shutter activations |
| 03:40.33 | JT | it's MTBF is 100000 shutter operations |
| 03:40.53 | JT | "how long my D70s will last", that should say |
| 03:41.01 | *** part/#asterisk stridernzl (n=neville@125-237-98-1.jetstream.xtra.co.nz) |
| 03:41.22 | Nugget | I expect I'll want something new long before I wear out this one. |
| 03:41.46 | JT | i shoot 1500 frames on a busy day |
| 03:41.50 | Nugget | wow |
| 03:42.07 | JT | busy meaning some event with lots of things happening :) |
| 03:42.25 | JT | when i was in japan i went through 5000 shots in 5 days |
| 03:42.36 | JT | being there was an event in itself ;) |
| 03:43.26 | [TK]D-Fender | D40 is still pricey around here. |
| 03:43.44 | [TK]D-Fender | I just doubt I'll get my moneys worth based on my skills and expected usage. |
| 03:43.48 | JT | hmm |
| 03:43.56 | [TK]D-Fender | 2 rather important factors |
| 03:43.58 | JT | [TK]D-Fender: did you look at that url, were they body only or what? |
| 03:44.47 | [TK]D-Fender | JT : Some, but these places are all out of town and shipping will likely add up. Also it comes with a base lens and thats where the cost really starts to add up |
| 03:45.32 | JT | only if you're unhappy with that lense |
| 03:45.44 | JT | and most people not into photography are happy with the kit lens |
| 03:45.47 | honeybeebuzz | I found IP301 ~ 140 CAD and IP320 ~114 CAD on canadianvoipstore.ca |
| 03:46.18 | Nugget | I love japan. :) |
| 03:46.43 | JT | honeybeebuzz: the 320 is BETTER than the 301 |
| 03:46.46 | Nugget | http://macnugget.org/photos/wallpaper/bluebridge and http://macnugget.org/albums/wallpaper/tokyostreets.thumb.jpg |
| 03:46.51 | JT | 301 has no speakerphone, for starters |
| 03:46.54 | Nugget | er http://macnugget.org/photos/wallpaper/tokyostreets |
| 03:47.03 | Nugget | although both with just a little canon s400 |
| 03:47.26 | honeybeebuzz | then I should get advntage of prince vs functionaliy be haing 320.... |
| 03:47.30 | honeybeebuzz | thanks folks |
| 03:47.41 | [TK]D-Fender | honeybeebuzz, canadianvoistore = voipsupply, which if that includes duty, etc, isn't too bad.... |
| 03:47.48 | Nugget | 320 is precisely 19 better than a 301! |
| 03:48.28 | [TK]D-Fender | Nugget, Your math skills are unparalleled (thats only because you never took TRIG!) |
| 03:48.31 | honeybeebuzz | good judgement |
| 03:48.51 | Nugget | when buying electronics just buy the largest model number you can afford. it's universal. |
| 03:48.55 | [TK]D-Fender | honeybeebuzz, keep in mind you'll need a power brick for that. |
| 03:49.17 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:50.33 | Nugget | it bugs me that two bulbs are burnt out in http://macnugget.org/photos/wallpaper/bluebridge |
| 03:50.47 | Nugget | I fixed it in photoshop but haven't made a new wallpaper from the fixed image |
| 03:51.20 | tengulre | hi,all |
| 03:51.52 | Nugget | complete with google earth kmz. |
| 03:52.27 | *** join/#asterisk bmg505 (n=leon@196.209.179.90) |
| 03:52.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:58.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:58.08 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:58.34 | honeybeebuzz | JT: do I need PoE Adapter, perhaps a very confusing question of the day |
| 03:59.17 | [TK]D-Fender | honeybeebuzz, That, or but the stanrd wall-power brick for it |
| 03:59.38 | coldsteal | im getting an error http://rafb.net/p/XNAiNy28.html |
| 03:59.42 | coldsteal | i put it there |
| 03:59.55 | [TK]D-Fender | honeybeebuzz, I would personally suggest a PoE injector if the price is only a little bit more. This can be recycled for other phones & uses afterwards. Makes for good seperate resale value. |
| 04:00.26 | [TK]D-Fender | coldsteal, You must start with priority **1**, not 6. |
| 04:00.34 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 04:00.47 | [TK]D-Fender | coldsteal, exten => 20,3,goto(antonni,20,1) <- this is no good either. |
| 04:01.17 | [TK]D-Fender | coldsteal, You are failing to understand the basics of dialplan logice. I highly recommend you give chapter 5 a good read again.... |
| 04:01.18 | [TK]D-Fender | ~book |
| 04:01.19 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:02.05 | [TK]D-Fender | Darn, every DSLR solution I'm looking at effectively starts at double the cost of the Panasonic I was looking at. |
| 04:02.11 | blitzrage | I heard the authors are dicks though |
| 04:02.19 | blitzrage | :) |
| 04:02.37 | Corydon76-home | Real big dicks. :-P |
| 04:02.52 | blitzrage | :-O |
| 04:03.26 | [TK]D-Fender | nah.. its SOFT-COVER. Clearly limp & non-threatening ;) |
| 04:03.40 | Corydon76-home | rofl |
| 04:03.44 | honeybeebuzz | okey, then this vendor is giving an option to buy or not ablout PoE for ~30 CAD... |
| 04:03.44 | [TK]D-Fender | pwned |
| 04:03.49 | *** join/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 04:03.56 | [TK]D-Fender | honeybeebuzz, link it |
| 04:04.03 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:04.20 | vutamhoan | I want to have some info about digium card, but don't know where to post |
| 04:04.31 | honeybeebuzz | http://www.canadianvoipstore.com/product_info.php?manufacturers_id=32&products_id=2907 |
| 04:05.05 | vutamhoan | So sorry everybody first: In the digium card, is there any info of it's origin? |
| 04:05.08 | honeybeebuzz | which digium card you want to talk about? |
| 04:05.45 | vutamhoan | I want to buy TE207P, but any line like "Made in USA" on it? |
| 04:06.07 | Corydon76-home | All Digium cards are made in the USA |
| 04:06.27 | vutamhoan | I know, but my customer ask it must has CO |
| 04:06.33 | Corydon76-home | It's part of the company pride |
| 04:06.51 | Corydon76-home | I don't think I've seen that text on the board |
| 04:07.12 | vutamhoan | (certificate of origin) - but Digium do not provide, so I need somethink to prove that become from USA |
| 04:07.16 | Nugget | There's a disclaimer, though... Digium telephone cards are made possible, in part, by contributions from Canadia. |
| 04:07.59 | Corydon76-home | Well, the components come from other places... but the cards are assembled in the USA |
| 04:08.13 | *** part/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 04:08.29 | *** join/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 04:08.55 | Corydon76-home | vutamhoan: call Digium sales in the morning. I'm sure they can work something up |
| 04:09.00 | kiscokid | vutamhoan: ask Digium to write a letter |
| 04:09.13 | vutamhoan | Yes, thank you very much |
| 04:09.35 | [TK]D-Fender | honeybeebuzz, 30$ is pretty god, I'd go for it. |
| 04:09.37 | [TK]D-Fender | good* |
| 04:10.18 | honeybeebuzz | keul.... so I need it besides its price.. aside, this would be my only voip phone |
| 04:10.49 | [TK]D-Fender | honeybeebuzz, lemme looks for a sec |
| 04:11.20 | [TK]D-Fender | honeybeebuzz, Where is this going to be used? Any thoughts of buying more phones in the future? |
| 04:12.16 | *** join/#asterisk Stridernzl (n=neville@125-239-161-21.jetstream.xtra.co.nz) |
| 04:12.40 | blitzrage | [TK]D-Fender: lol |
| 04:14.02 | honeybeebuzz | it will be used at home with digium card... |
| 04:17.09 | honeybeebuzz | I am not sure, but this would be my start for voip experimentation |
| 04:17.16 | [TK]D-Fender | honeybeebuzz, just looking for a full-featured basic phone? |
| 04:18.09 | [TK]D-Fender | honeybeebuzz, because sometimes if this is "your" master phone I might suggest you spend a little bit more for something nicer, but this depends on your tastes |
| 04:18.11 | honeybeebuzz | yes |
| 04:19.13 | [TK]D-Fender | honeybeebuzz, http://www.canadianvoipstore.com/product_info.php?cPath=95_106&products_id=758 |
| 04:19.19 | *** part/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:21.18 | honeybeebuzz | I will need some more reading... this seems to be 3x lines business phone... for home, here usually one line comes in, a bit less featured would be not nice? |
| 04:21.56 | JT | lines are a foreign concept in voip |
| 04:22.02 | JT | more like line appearances |
| 04:22.11 | JT | you can configure them in a number of different ways |
| 04:25.26 | honeybeebuzz | ah... i mixed up phone's physical lines |
| 04:25.39 | *** join/#asterisk kimosabe (n=kimosabe@189.175.37.162) |
| 04:26.07 | kimosabe | how can i force my sipura to disconect becuse it sseems to stay conected even when i finish a call |
| 04:26.10 | [TK]D-Fender | honeybeebuzz, in all honesty, ANY polycom is plenty for the job. |
| 04:26.19 | Maliuta | JT: yeah, and different hard phones deal with the "line" thing in different ways, the cisco 7940 I have can use the same SIP peer info for the 2 "line" buttons it has on it. Others require different sip peer stuff for each "line" |
| 04:26.39 | [TK]D-Fender | honeybeebuzz, Its a question of haveing a bigger scree to enjoy the visul display, play with the XHTML microbrowser, etc. |
| 04:26.49 | kimosabe | voip state says conected but the call has been finished |
| 04:27.34 | [TK]D-Fender | honeybeebuzz, the IP 501 comes with a power brick, and has a 2-port switch so you can plug it in-line with your PC if you don't have a Switch handy. |
| 04:28.26 | [TK]D-Fender | honeybeebuzz, value varies depending on how, where, and why you deploy it. |
| 04:28.47 | honeybeebuzz | I thanks all for good suggesions... hopefully whatever I go with is compatible with trixbox |
| 04:29.16 | JT | argh |
| 04:29.19 | [TK]D-Fender | honeybeebuzz, For those planning on PoE, the IP 320 is a KILLER. Onces you add the cost of powering it, you then consider if you need to use it in-line... oops, that requires an IP 330 instead, adds more cost, etc. |
| 04:29.26 | JT | you should try and avoid trixbox |
| 04:29.34 | [TK]D-Fender | honeybeebuzz, EVERY SIP phone at that store is. |
| 04:30.12 | [TK]D-Fender | honeybeebuzz, I'm trying to suggest something of quality that will fit the budget as close as possible and optimise its value for the way you'll use it |
| 04:31.02 | honeybeebuzz | and that is 501 with asterisk |
| 04:31.24 | [TK]D-Fender | honeybeebuzz, ? |
| 04:31.31 | honeybeebuzz | ! |
| 04:32.06 | [TK]D-Fender | honeybeebuzz, Try rephrasing that into something comprehensible please.... |
| 04:32.29 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:32.50 | honeybeebuzz | I mean that IP501 is good when combined with Asterisk... |
| 04:32.59 | [TK]D-Fender | honeybeebuzz, All of them are. |
| 04:33.21 | *** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net) |
| 04:33.39 | Stridernzl | [TK]D-Fender: did you get to it? re forwarding? |
| 04:33.57 | [TK]D-Fender | honeybeebuzz, Polycom is a GOOD quality phone regardless of the model. Total cost & style of deployment will sway which model might bes suit you. |
| 04:34.09 | [TK]D-Fender | Stridernzl, incomplete but will finish soon. |
| 04:34.46 | honeybeebuzz | I initially setup asterisk box, later switch to trixbox to smell the differences in term of deployments.... but I am sure that I can do asterisk as well... |
| 04:35.02 | JT | ~trixbox |
| 04:35.03 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 04:35.14 | Stridernzl | [TK]D-Fender: ? today ? ..... we are @ end of day .. but if you doing soon I'll hang around and watch / test ? listen |
| 04:35.16 | JT | trixbox makes nasty dialplans |
| 04:35.20 | JT | ~zeeek |
| 04:35.21 | jbot | zeeek is, like, someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 04:35.39 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:35.44 | pigpen | Trixbox...heh...that's a good one. |
| 04:35.58 | [TK]D-Fender | honeybeebuzz, Trixbox uses * at the core. they ALL work with it |
| 04:36.04 | MrTelephone | how does openser with nat support work? good? |
| 04:36.19 | JT | MrTelephone: isn't there a more relevant channel? ;) |
| 04:36.34 | MrTelephone | well i just want to compare with asterisks no reinvite |
| 04:36.48 | MrTelephone | is there "nat support" really just passing rtp? |
| 04:36.51 | JT | gar "asterisk" |
| 04:37.05 | honeybeebuzz | okey... I better do asterisk onwards... since I am new, better be good start |
| 04:37.18 | JT | why don't you ask them |
| 04:37.29 | JT | MrTelephone: openser is a sip proxy, not rtp |
| 04:37.36 | MrTelephone | im trying to do call routing, media gateway with 2 t1s all on one box.. i have to break it up a little im having too many issues :-/ |
| 04:37.40 | JT | MrTelephone: that said, it does have an rtp proxu module available |
| 04:37.55 | MrTelephone | sounds cool |
| 04:38.07 | JT | MrTelephone: is the media gateway external or asterisk? |
| 04:38.28 | MrTelephone | an asterisk box that does simple sip to zap (t1) should last a while without restarting?? |
| 04:38.32 | MrTelephone | its asterisk |
| 04:38.47 | tzafrir_laptop | Sure |
| 04:38.59 | MrTelephone | my single asterisk box handles mgcp clients, sip clients, 1 adit 600 channel bank, 1 telco pri |
| 04:38.59 | JT | if there's no bugs, sure |
| 04:39.29 | MrTelephone | after a week if i type "stop now" someone it takes like 20 seconds for it to shutdown |
| 04:39.46 | pigpen | So are there any sip wifi phones that are worth a dam yet? I haven't looked for a while. |
| 04:39.59 | MrTelephone | pigpen, people are saying no to that |
| 04:40.10 | MrTelephone | cisco did a presentation of their 7920 series and they work great in the office |
| 04:40.18 | pigpen | yeah..same story as before. |
| 04:40.19 | kimosabe | how can i force sipura to hang up |
| 04:40.19 | JT | pigpen: the technology is not worth a damn |
| 04:40.22 | MrTelephone | people say they are shit out in the field |
| 04:40.51 | MrTelephone | asterisk slow to shutdown when executing "stop now"? i guess i'll google that |
| 04:40.55 | pigpen | I have been using long range 900Mhz phones... |
| 04:41.04 | pigpen | kinda sucks, but they work. |
| 04:41.12 | [TK]D-Fender | MrTelephone, Just try doings something useful with *, its stop immediately! |
| 04:41.15 | MrTelephone | is it because asterisk used so much heap because of bad mgcp code |
| 04:41.36 | MrTelephone | I don't know what you mean fender |
| 04:41.46 | JT | it was a joke |
| 04:42.04 | MrTelephone | I had to make sure :P |
| 04:42.40 | kimosabe | is there somthing i can set in sipura to force it to hang up when done the line stays enabled |
| 04:43.50 | pigpen | So has anyone gotten the polycom "buddy watch" to work with a hint that refers to a db value? |
| 04:44.46 | MrTelephone | sounds interesting pigpen |
| 04:44.46 | pigpen | ie: Custom Device State. |
| 04:44.54 | pigpen | For some reason, my customers (mostly the ones with PHD's) keep forwarding the dam phone to odd places. |
| 04:45.08 | pigpen | So I have them trigger it via a dialplan entry. |
| 04:45.20 | pigpen | but, they would like a cute "blinking light" telling them it is on. |
| 04:45.33 | pigpen | I have all of it, but the blinking light part. |
| 04:45.50 | pigpen | http://www.asterisk.org/node/48325 |
| 04:46.05 | pigpen | ^^^info on it..but I guess I am missing the idea to get it to work. |
| 04:46.36 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:47.11 | kimosabe | how can i force reboot in spa 3000 fxo stayed conected |
| 04:47.59 | JT | kimosabe: i'm sure disconnecting the power then reapplying it will do the trick |
| 04:48.08 | [T]ank | need some help with an error i am getting here are the details: http://pastebin.ca/634096 |
| 04:48.09 | MrTelephone | thats some real cool stuff pigpen |
| 04:48.16 | [TK]D-Fender | pigpen, You need to install trunk or manually patch your install. |
| 04:48.31 | [TK]D-Fender | pigpen, And even then, FORGET "flashing". ON/OFF is all you're going to get. |
| 04:48.34 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 04:48.40 | pigpen | ah, it is not in the main? |
| 04:48.48 | raidenz | hello |
| 04:48.53 | pigpen | On is fine. Flashing is for pussys anyway. |
| 04:48.54 | [TK]D-Fender | pigpen, no, and it won't be until 1.6 |
| 04:48.55 | pigpen | :) |
| 04:48.59 | pigpen | ah. |
| 04:49.18 | kimosabe | come on now some one give me a hand please |
| 04:49.19 | pigpen | Works for me...I can wait. I have better things to do! |
| 04:49.20 | raidenz | What happens if you put a Digium TEXXX 3.3 volt card into a 5v PCI slot? |
| 04:49.32 | pigpen | kimosabe, clap, clap, clap. |
| 04:49.51 | andrewg_fm | grrr. bloody ddos kiddies |
| 04:49.54 | raidenz | So basically putting a Quad card into an older systems |
| 04:49.57 | [TK]D-Fender | raidenz, load chan_combustion.so |
| 04:50.14 | raidenz | TKD-Fender: Does it fry the card? |
| 04:50.34 | [TK]D-Fender | raidenz, IIRC... it won't even FIX. |
| 04:50.34 | pigpen | Shit, a 3.3v card won't fit will it? |
| 04:50.36 | [TK]D-Fender | FIT* |
| 04:50.42 | kimosabe | pigpen i mean my spa 3000 is in use i need it to reebot |
| 04:50.43 | raidenz | or just the card won't work but the card won't be fried |
| 04:51.08 | JT | raidenz: it won't fit, get the correct card. |
| 04:51.09 | *** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net) |
| 04:51.10 | pigpen | kimosabe, yeah..sorry, I haven't used one for awhile. Pull the plug works well. |
| 04:51.28 | pigpen | raidenz, use a hack saw. |
| 04:51.30 | [TK]D-Fender | raidenz, but the RIGHT card. Translation : Sangoma A10[x]d <- |
| 04:51.37 | MrTelephone | sludge hammer |
| 04:51.42 | MrTelephone | my a102d isn't perfect |
| 04:51.43 | JT | kimosabe: i already told you to power cycle |
| 04:51.52 | [TK]D-Fender | MrTelephone, because sludge is so darned tough! |
| 04:51.58 | JT | kimosabe: now read the damn SPA3000 instruction manual |
| 04:52.07 | MrTelephone | it will bend your fender |
| 04:52.19 | pigpen | So...how far off is 1.6 (yes, I will duck) |
| 04:52.34 | kimosabe | jt i have and by the way the spa 3000 is 1500 miles away it not as easy as removing the electrical cord |
| 04:52.57 | JT | kimosabe: then check the instructions on how to remote reboot it |
| 04:53.19 | pigpen | kimosabe, well you didn't say that in the beginning. |
| 04:53.29 | pigpen | yeah..the manual tells you how. I forgot. |
| 04:53.32 | pigpen | forget. |
| 04:53.42 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 04:53.56 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:53.56 | kimosabe | you can reboot via the admin/reboot but i cant reboot becuase it says voi/ip state conected |
| 04:54.22 | JT | sorry but it seems no-one knows, kimosabe |
| 04:54.30 | JT | and this isn't much of an asterisk issue |
| 04:55.34 | pigpen | kimosabe, http://www.sipura.com/support/spa3000faq/Section_3.html |
| 04:55.54 | pigpen | See item 3 to force hangup. |
| 04:56.10 | pigpen | hopefully it applies. |
| 04:58.02 | Maliuta | JT: what do you make of http://www.voip-info.org/wiki/view/pennytel ??? something about pennytel reject asterisk as a sip agent? |
| 04:58.51 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 04:58.57 | kimosabe | pigpen thanks man |
| 04:59.05 | JT | Maliuta: a load of utter rubbish |
| 04:59.05 | pigpen | kimosabe, good luck. |
| 04:59.08 | [TK]D-Fender | Maliuta, I take that as "change your damn UA and get on wih life" |
| 04:59.15 | JT | Maliuta: i am editing it now, as that is wrong. |
| 04:59.39 | Maliuta | [TK]D-Fender: since you don't deal with them your opinion is superfluous |
| 05:00.23 | JT | Maliuta: there is no need to be rude |
| 05:00.51 | [TK]D-Fender | Maliuta, I suspect you will find few people here do, and in the longer run, that page alone tells you what you need to do and it certainly isn't difficult or terribly unreasonable. |
| 05:00.58 | pigpen | Shit. I have to look that word up. |
| 05:01.18 | [TK]D-Fender | Maliuta, So feel free to shoot us all down in whatever order you please. |
| 05:02.25 | pigpen | Me first. |
| 05:03.23 | pigpen | Well, the bed is calling. Night all. |
| 05:03.31 | JT | [TK]D-Fender: i use that ITSP and the wiki is WRONG |
| 05:03.37 | JT | just fixed it |
| 05:04.08 | [TK]D-Fender | JT : ok, fine, sure! |
| 05:06.55 | vutamhoan | [TK]D-Fender, may I ask you a question |
| 05:06.56 | *** part/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 05:07.01 | *** join/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 05:07.02 | [TK]D-Fender | ~ask |
| 05:07.02 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 05:07.24 | [TK]D-Fender | vutamhoan, and stop bouncing every time you speak, its getting obnoxious |
| 05:07.29 | vutamhoan | In the digium card has the line "Made in USA" or something like that? |
| 05:07.51 | [TK]D-Fender | vutamhoan, Don't know. Go ask them directly yourself. |
| 05:08.01 | [TK]D-Fender | vutamhoan, And why is it so important? |
| 05:08.25 | vutamhoan | My customer want to know where the card come from |
| 05:08.48 | Qwell | vutamhoan: Call Digium sales tomorrow. |
| 05:09.04 | Qwell | Nobody here is going to be able to certify that for you. |
| 05:09.09 | vutamhoan | Yes, because of this is urgent |
| 05:09.26 | fujin | OH REALLY? |
| 05:09.37 | fujin | Lack of planning on your part != emergency on ours |
| 05:09.45 | MrTelephone | hahaha |
| 05:09.46 | JT | what's more important, where the card came from, or how soon you get it? |
| 05:09.46 | MrTelephone | bouncing |
| 05:10.23 | vutamhoan | I have to go to my partner right now for checking.. - thanks for your support |
| 05:10.56 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 05:12.10 | [TK]D-Fender | Qwell[], Your cards have to be certified Terror-Free, Dolphin-Safe, and made of 100% recycled walnuts or they just won't make the grade! |
| 05:12.29 | JT | don't forget RoHS |
| 05:12.38 | JT | and no cryptographic exports |
| 05:12.53 | Strom_M | and also, usually, digium cards have the word "Digium" on them |
| 05:12.55 | Strom_M | just a thought |
| 05:14.28 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 05:16.51 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 05:16.55 | lsodi | greetings, I have macro: exten => s,1,GotoIf(${DB_EXISTS(CFU/${ARG1})}?2:4) and database shows *CLI> database get CFU 6837677 |
| 05:16.57 | lsodi | Value: 6962210 |
| 05:17.29 | lsodi | but asterisk produces -- Executing [6837677@macro-stdexten:1] Goto("Zap/1-1", "internal|6837677|1") in new stack |
| 05:17.42 | lsodi | what could be wrong? |
| 05:19.42 | Strom_M | you didn't read the documentation for DB_EXISTS is what's wrong |
| 05:21.13 | Strom_M | or your code is borked |
| 05:21.13 | [TK]D-Fender | lsodi, please note those 2 lines do NOT match |
| 05:21.48 | [TK]D-Fender | lsodi, please pastebin the WHOLE macor, and the WHOLE calls CLI output, and not clearly mismatched bits |
| 05:22.07 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 05:25.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:28.21 | lsodi | D-Fender> http://pastebin.com/d4574de79 |
| 05:29.19 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:31.02 | [TK]D-Fender | lsodi, WTF is "include => incoming" doing INSIDE a macro? because thats clearly what is getting executed instead.... |
| 05:31.31 | [TK]D-Fender | lsodi, lines 6-10 aren't being called at all. |
| 05:31.40 | [TK]D-Fender | lsodi, go caffeinate! |
| 05:32.15 | [TK]D-Fender | -- Executing [6837677@macro-stdexten:1] Goto("Zap/2-1", "internal|6837677|1") in new stack <- this is a GOTO. |
| 05:32.26 | [TK]D-Fender | exten => s,1,GotoIf(${DB_EXISTS(CFU/${ARG1})}?2:4) <- this is a GOTOIF |
| 05:32.46 | [TK]D-Fender | . <- this is * without CAFFEINE. See how small it is! |
| 05:33.08 | [TK]D-Fender | it wants to grow up an be a BIG *! |
| 05:36.15 | [TK]D-Fender | ok, thats it for me tonight.... time to hit the sack, later all |
| 05:37.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 05:57.20 | mosty | if i wanted to forward callerid data from an iax client when i send the call onwards, should i just leave the callerid function alone? |
| 05:58.18 | denon | the callerid moves with the call |
| 05:58.29 | denon | you dont need to "send" it |
| 06:00.12 | mosty | well for example a call comes in from an iax client, and then in turn i forward the call on via PRI, then would i have to set (not send) the callerid before doing the dial? |
| 06:01.21 | mosty | er, note that the IAX client's callerid is not necessarily the same as that of the PRI line i dial via |
| 06:03.10 | enioreh | hi ppl |
| 06:06.59 | *** join/#asterisk Amerkl (n=Mskes@ool-182edcd3.dyn.optonline.net) |
| 06:09.31 | nvicf | hi enio |
| 06:09.44 | Amerkl | I'm looking for the simplest VOIP solution that will allow me to do the following: Call up a user using my voip service, speak with them, and then if required divert them to an extension with an IVR menu. isnt asterisk too complex for my simple usage? |
| 06:10.41 | JT | Amerkl: no, it's not too complex. IVRs are not basic |
| 06:10.43 | mosty | amerkl: what provides the IVR? |
| 06:11.31 | Amerkl | everything should be on 1 computer |
| 06:11.33 | Amerkl | software |
| 06:11.51 | Amerkl | i have heard asterisk has ivr capabilities, but asterisk just seems to complex for my use |
| 06:12.24 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:12.26 | Amerkl | isn't there a simpler solution? |
| 06:12.46 | mosty | amerkl: i don't know of anything simpler for creating an IVR |
| 06:13.50 | Amerkl | there wont be a softphone or a voip service with basic ivr capabilities yeah? |
| 06:13.52 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 06:14.05 | JT | Amerkl: no, asterisk is already simple enough |
| 06:14.20 | JT | Amerkl: i'm sure some providers will happily provide you with an outsourced IVR for a fee |
| 06:14.21 | mosty | amerkl: simple phones don't do IVR's (that I know of) |
| 06:16.07 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:19.35 | Amerkl | alright, thanks guys : ) |
| 06:25.45 | *** join/#asterisk minkus (n=minkus@pool-71-182-32-236.clrkwv.east.verizon.net) |
| 06:26.39 | tengulre | which website provide flash video? |
| 06:27.19 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 06:30.45 | Amerkl | guys, will IVR work on a conference call? |
| 06:31.03 | JT | Amerkl: what do you mean? |
| 06:31.42 | Amerkl | it works through tone yeah? so suppose |
| 06:31.58 | Amerkl | User \ / IVR |
| 06:31.58 | Amerkl | <PROTECTED> |
| 06:32.14 | JT | please explain more clearly |
| 06:32.31 | Amerkl | basically I make a conference with the User and the IVR system, allowing the user to interact with the IVR after I have spoken with him, and while he is interacting with the IVR i am silent |
| 06:32.47 | Amerkl | since an IVR basically reads the sound made by the digit pressed, it should work yeah? |
| 06:33.08 | JT | i don't think so |
| 06:33.21 | JT | it would be a call, not a conference, to the user first |
| 06:33.36 | JT | don't know if you can hand them off to an ivr with you still listening |
| 06:34.47 | Amerkl | to the IVR i will appear the user, its no different |
| 06:34.54 | Amerkl | *i'll appear to be the user |
| 06:35.20 | Amerkl | From the IVRs perspective, its connected to me, it doesnt know about the user |
| 06:35.24 | JT | it is different |
| 06:35.24 | Amerkl | so it shouldnt be any different |
| 06:35.29 | JT | if you have 2 users connected |
| 06:35.30 | Amerkl | why? |
| 06:35.35 | JT | try to learn a bit more about asterisk |
| 06:35.44 | Amerkl | ahh I' |
| 06:35.44 | JT | because IVRs are usually for one user |
| 06:35.53 | Amerkl | I'm talking about softphone with a hosted ivr here |
| 06:36.11 | JT | even so |
| 06:37.56 | Amerkl | as far as the IVR knows, tehre is one user |
| 06:38.39 | JT | i suppose you could do conferencing at your softphone |
| 06:38.54 | Amerkl | yes |
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| 06:52.46 | creativx | oh damn tiredness |
| 06:52.47 | creativx | morning JT |
| 06:52.56 | creativx | or mid day. or whatever it may be down under, i never remember |
| 06:53.03 | JT | hello creativx |
| 06:53.09 | JT | it's late afternoon |
| 06:53.19 | JT | but it's always morning somewhere in the world |
| 06:53.27 | snuff-work | almost time to knock off :P go 5pm |
| 06:54.21 | creativx | envy! i just got into the office |
| 06:54.29 | creativx | with some good 3 hrs sleep before that |
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| 06:56.08 | Aurs | 1.4.8 was pretty short lived? :) |
| 06:56.55 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:56.58 | Chris-NB | hi |
| 06:57.10 | Chris-NB | anyone knows a good solution for a switchboard? |
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| 07:01.40 | mosty | Chris-NB, what do you need a switchboard for? |
| 07:03.02 | Chris-NB | mosty, to answer calls and distribute to the right persons or give information |
| 07:03.08 | Chris-NB | mosty, in a company |
| 07:03.31 | mosty | how many internal extensions? |
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| 07:06.40 | *** join/#asterisk vanlink (n=vanlink@122.161.27.126) |
| 07:07.06 | vanlink | i am unable to send dtmf tones to my voice gateway through sip phone,any help? |
| 07:08.03 | Chris-NB | 240 |
| 07:09.29 | vanlink | does anybody even talk anything here? |
| 07:09.50 | andrewg_fm | this channel is quite active, actually |
| 07:10.11 | vanlink | i dont see any msgs except yours |
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| 07:10.17 | creativx | imagine that |
| 07:10.46 | vanlink | can anyone help me with my issue? |
| 07:10.54 | vanlink | trouble with sip phone |
| 07:11.31 | creativx | in-band, out of ban? |
| 07:11.32 | creativx | d |
| 07:11.46 | vanlink | hmm inband |
| 07:12.17 | Chris-NB | vanlink, same settings in asterisk and the sip phone? |
| 07:12.28 | vanlink | yes |
| 07:12.36 | vanlink | let me explain in detail |
| 07:12.39 | Chris-NB | what phone? |
| 07:12.54 | vanlink | i am using vanlink voice gateway,which has fxs and fxo ports on it |
| 07:12.55 | Chris-NB | mosty, you see why I'm asking for a switchboard? |
| 07:13.08 | vanlink | i am using xlite sip phone |
| 07:13.43 | vanlink | so i am able to make calls to my fxo port using phones connected to fxs and able to get to pbx |
| 07:14.00 | vanlink | but when i try to do that with a sip phone,i am unable to do it |
| 07:14.19 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-182-96.hsd1.wa.comcast.net) |
| 07:14.20 | vanlink | sip phone to FXS phone is working too |
| 07:14.45 | vanlink | but not SIP to FXO => PBX |
| 07:15.27 | vanlink | any help? |
| 07:16.28 | Chris-NB | sry, no experience with fxo/fxs |
| 07:16.47 | vanlink | hmm,ok |
| 07:16.53 | vanlink | thanks anyways |
| 07:17.04 | vanlink | anyone else have any idea? |
| 07:17.26 | creativx | sorry no idea here either, im all-ip |
| 07:17.51 | vanlink | ok |
| 07:18.01 | JT | sounds like a gateway issue |
| 07:18.48 | vanlink | im not sure,may be a gateway issue,but i am able to get to pbx using phone connected to FXS of gateway |
| 07:19.16 | JT | i thought this was a dtmf issue |
| 07:19.55 | vanlink | it is a dtmf issue,as when i dial from sip phone,the gateway is not taking my dtmf tones |
| 07:20.23 | vanlink | this is my verbose msgs |
| 07:20.26 | vanlink | Called 108@192.168.0.104 |
| 07:20.26 | vanlink | <PROTECTED> |
| 07:20.26 | vanlink | <PROTECTED> |
| 07:20.26 | vanlink | <PROTECTED> |
| 07:20.26 | vanlink | exten => 108,1,Dial(Sip/108@192.168.0.104) |
| 07:20.26 | vanlink | exten => 108,2,Senddtmf(ww9109#) |
| 07:20.32 | JT | no PASTING here |
| 07:20.35 | JT | ~pb |
| 07:20.35 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 07:20.42 | vanlink | ok,sorry |
| 07:20.58 | JT | does the gateway have the same dtmfmode as asterisk? |
| 07:21.48 | vanlink | here is my pastebin http://pastebin.com/pastebin.php?dl=d36aa6c0a |
| 07:22.07 | vanlink | JT: i am not sure about it |
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| 07:22.26 | vanlink | there is no documentation with gateway which tells which dtmf mode it is using |
| 07:22.39 | JT | well you need to configure the damn thing |
| 07:23.09 | vanlink | but am i using the dial command in asterisk correctly or not? |
| 07:23.32 | JT | not ideal, no |
| 07:23.42 | JT | you should setup a sip friend in sip.conf |
| 07:23.55 | JT | and do Dial(SIP/friend/number) |
| 07:24.12 | vanlink | my fxo port is registered as friend |
| 07:24.35 | vanlink | but if i do Dial(SIP/108@192.168.0.104/109) then it shows |
| 07:24.45 | vanlink | no address 104/109 |
| 07:25.29 | JT | yes i didn't tell you to do that. |
| 07:25.38 | JT | give it a name in sip.conf |
| 07:26.11 | vanlink | but then it wont register |
| 07:26.39 | *** join/#asterisk saftsack (n=oliver@p54A7E013.dip.t-dialin.net) |
| 07:26.40 | vanlink | see this is how i register my fxo port |
| 07:27.04 | JT | what are you talking about? registrations do NOT happen in the dialplan |
| 07:27.08 | vanlink | register => 108@192.168.0.104 |
| 07:27.12 | JT | read the book |
| 07:27.13 | JT | ~thebook |
| 07:27.14 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:27.21 | vanlink | no i am talking about the sip.conf file |
| 07:27.24 | JT | ok |
| 07:27.40 | JT | and why can't you add a friend entry for the gateway? |
| 07:28.06 | vanlink | coz if i give it anyother name other then 108 it wont get registerd |
| 07:28.12 | vanlink | when i do sip show registry |
| 07:28.38 | vanlink | after register command,i do |
| 07:28.43 | JT | i didn't say to change the register command |
| 07:28.53 | vanlink | [108] |
| 07:28.54 | vanlink | type=friend |
| 07:28.54 | vanlink | username=108 |
| 07:29.15 | vanlink | i understand,but this is the only way it works |
| 07:29.31 | JT | can you please put a space after your commas, thanks :P |
| 07:29.39 | vanlink | sorry |
| 07:30.15 | vanlink | if possible for you, can u please give me a sample sip.conf file ? |
| 07:30.30 | JT | there is a sample that comes with asterisk |
| 07:30.37 | vanlink | no |
| 07:30.45 | vanlink | i mean,the way u are talking about |
| 07:30.53 | vanlink | sorry again for space |
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| 07:34.14 | vanlink | no? |
| 07:36.08 | vanlink | well, thanks anyways |
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| 07:40.22 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 07:40.24 | foo | In /etc/asterisk/sip.conf, is nat=yes an option? |
| 07:40.34 | vanlink | yes |
| 07:40.43 | vanlink | foo:it is |
| 07:40.53 | foo | okay, and I do that if my asterisk box is behind nat? and I manually have port forwarding configured, right? |
| 07:41.08 | vanlink | foo:right |
| 07:41.11 | foo | And I can stick that after [general], I imagine |
| 07:41.23 | foo | Right? |
| 07:41.50 | vanlink | foo:yes, and u can give assign it to every peer/user too |
| 07:42.52 | vanlink | by writing nat=yes to peer/user's context |
| 07:42.54 | foo | vanlink: Thank you |
| 07:49.24 | juuva | anyone knows working softphones for windows mobile 5? |
| 07:49.25 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:50.58 | *** join/#asterisk Uzzi (n=andrea@host192-169-dynamic.60-82-r.retail.telecomitalia.it) |
| 07:51.01 | Uzzi | hi |
| 07:51.39 | Uzzi | Does Asterisk work with Communication controller: Conexant HCF 56k Data/Fax/Voice/Spkp device? |
| 07:51.58 | JT | foo: no no no, nat=yes is only if you have far end clients behind nat |
| 07:52.10 | JT | if you are connecting to a sip server you do NOT need nat=yes |
| 07:52.22 | JT | if you are connecting to a sip server you do NOT need port forwarding |
| 07:53.53 | foo | JT: ah |
| 07:54.37 | JT | you only need port forwarding if you are acting as sip server and you are behind nat and have clients out on the Internet |
| 07:54.49 | foo | JT: hmm, well, the network with my asterisk install (I have asterisk+SIP setup with trixbox) is behind a linux router. I have ports 5060, 5061, and 10000-20000 forwarded to the asterisk box. |
| 07:55.18 | JT | 5061 is pointless, why did you forward that |
| 07:56.05 | foo | JT: hm, I would have a client out on the Internet, my palm treo, and I don't know if my install also acts as a SIP server or not. I mean, in trixbox I have 2 trunks configured to the way vitelity, my SIP provider, said to - and it works. I forward 5061 because vitelity said so, hm, someone else said it is pointless too, though. *shrug* |
| 07:56.28 | JT | it is acting as server then |
| 07:56.36 | JT | eww, "trunks" |
| 07:56.39 | JT | ~trixbox |
| 07:56.39 | jbot | [trixbox] a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 07:57.38 | creativx | heh |
| 07:57.46 | creativx | seems like SIP "trunk" is here to stay |
| 07:57.54 | JT | lies |
| 08:00.28 | foo | JT: hm, trunks are bad? Are they not part of asterisk and something else or something? I'm curious to learn, excuse me if I'm in the wrong place, please |
| 08:01.37 | JT | trunks is the wrong name for a sip connection |
| 08:02.36 | foo | JT: oh, I see |
| 08:02.39 | vanlink | hey thanks |
| 08:02.41 | vanlink | it worked |
| 08:02.45 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 08:02.58 | vanlink | it was configuration issue with my voice gateway only |
| 08:03.00 | vanlink | thanks JT |
| 08:03.38 | foo | JT: What are they called? I'm still learning the VoIP lingo. |
| 08:04.21 | JT | vanlink: no problem |
| 08:04.26 | vanlink | :) |
| 08:04.32 | vanlink | bye guys |
| 08:04.34 | *** part/#asterisk vanlink (n=vanlink@122.161.27.126) |
| 08:04.46 | JT | foo: sip connections, calls |
| 08:04.50 | foo | JT: gotcha, I see |
| 08:05.01 | foo | Anyone in here used a softphone on a palm treo before? |
| 08:05.47 | *** join/#asterisk tuzhila (n=kvirc@84.47.128.99) |
| 08:06.21 | tuzhila | hi all |
| 08:08.46 | Chris-NB | anyone realized a switchboard with asterisk? |
| 08:09.48 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:10.50 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:11.05 | Aurs | "-g Remove resource limit on core size, thus forcing Asterisk to dump core in the unlikely event of a segmentation fault or abort signal." |
| 08:11.05 | JT | Chris-NB: you'd be better off with some form of address book i'd think |
| 08:11.11 | Aurs | ha ha, very funny man page |
| 08:11.33 | Aurs | :P |
| 08:12.01 | creativx | Chris-NB: yes |
| 08:12.09 | creativx | although with 15% of the extensions that you have |
| 08:12.34 | Chris-NB | JT, switchboard for incoming calls. what benefit should bring a address book? |
| 08:12.43 | Chris-NB | creativx, how have you done that? |
| 08:14.01 | creativx | Chris-NB: a combination of lots of things. a service application that talks to the AMI and broadcasts udp packets to the clients (everybody has access to the "switchboard" in our CRM), another script that does CID lookups and decides where to route the calls |
| 08:14.12 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:15.17 | Chris-NB | creativx, have you written the software? |
| 08:15.26 | creativx | yes. in.... visual basic |
| 08:15.28 | Chris-NB | creativx, it's an automatic switchboard? |
| 08:15.30 | creativx | quick, shudder everyone |
| 08:15.39 | Chris-NB | creativx, VB? : ) |
| 08:15.39 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 08:15.50 | coldsteal | if my voip service (broadvoice) doesnt use iax and i buy a iax ata thing for my phone can i conect my ata to my * server and still call out ? |
| 08:16.09 | Chris-NB | creativx, no one is answering incoming calls and 'manually' distributing it? |
| 08:16.17 | creativx | Chris-NB: we can decide.. the idea was to have the possibility for x number of agents taking incoming calls and route them or service them, or turn it all around and let the incoming calls reach the most likely person they are trying to reach on the first try |
| 08:16.43 | creativx | so during the summer we have 2-3 persons taking the incoming calls manually |
| 08:16.57 | creativx | when our employees are too busy to talk |
| 08:16.59 | Chris-NB | creativx, ok. sounds interesting |
| 08:17.25 | Chris-NB | creativx, but that software is only usable with your crm? |
| 08:17.28 | creativx | and then in the fall when things have normalized the idea is to let each employee have the responsibility of answering incoming calls |
| 08:17.34 | creativx | yes this is custom written for our use Chris-NB |
| 08:17.42 | Chris-NB | i c |
| 08:17.58 | creativx | but once you figure out the AMI things arent that bad :> |
| 08:18.14 | Chris-NB | so it's interesting, but I can't benefit from it? : ) |
| 08:18.24 | creativx | not other than conceptually :) |
| 08:19.33 | Chris-NB | k |
| 08:19.52 | *** join/#asterisk af_ (n=getsmart@81-174-46-138.dynamic.ngi.it) |
| 08:20.13 | creativx | it all depends.. what kind of business you do etc |
| 08:20.18 | creativx | and what applications you use in-house |
| 08:20.53 | creativx | then you can start hacking into asterisk and see the great added value it brings along to effeciency etc |
| 08:23.43 | JT | Chris-NB: sorry, why do you need 473487334 buttons for incoming calls? |
| 08:25.55 | *** join/#asterisk Dirk- (n=a@82-33-155-212.cable.ubr04.wiga.blueyonder.co.uk) |
| 08:26.14 | Chris-NB | JT, what do you mean with that? |
| 08:26.16 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-182-96.hsd1.wa.comcast.net) |
| 08:26.57 | JT | Chris-NB: you asked me what good an address book is, i'm asking you what good ten thousand buttons is |
| 08:27.07 | JT | Chris-NB: you haven't made it at all clear what you want |
| 08:28.54 | Chris-NB | JT, I need 'something' for a switchboard where calls are routet, someone picks up the phone, places call on hold, calls someone else and connect those two calls or tells the 1st caller that the 2nd isn't available or something else. |
| 08:29.20 | Chris-NB | JT, I can't see where a address book can solve this? |
| 08:29.54 | JT | oh |
| 08:30.03 | JT | you mean like queues and call transfers, right |
| 08:30.07 | JT | very basic |
| 08:30.22 | JT | you don't need a big switchboard for that |
| 08:31.56 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:33.49 | creativx | sigh.. i love it when things just break randomly |
| 08:41.18 | Chris-NB | JT, It should be a bit bigger. If a call is transfered and nobody picked up the other end, the call should be routet back to the switchboard/queue and the person answering the call again should know that the call was transfered and the other person didn't pick up |
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| 08:44.36 | JT | Chris-NB: that's all doable in dialplan logic |
| 08:45.14 | Chris-NB | JT, how can I tell the person that this call already passed the switchboard before? |
| 08:45.32 | creativx | Chris-NB: even better, why try to transfer a call if the person is unavailable? presence is the keyword here :-) |
| 08:45.57 | Chris-NB | creativx, but for that I ned a SER or OpenSER which I haven't got |
| 08:46.18 | JT | no. |
| 08:46.22 | Chris-NB | creativx, oh, ok. with blf it'd also reachable |
| 08:46.26 | JT | you don't need a sip proxy at akk |
| 08:46.51 | JT | all |
| 08:46.51 | Chris-NB | JT, with blf it would be possible. right? |
| 08:48.39 | JT | you don't even need blf |
| 08:48.51 | JT | it's all very acheivable in dialplan logic, and many people do it |
| 08:48.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:49.16 | Chris-NB | JT, how can I 'see' if a person is un/available? |
| 08:49.17 | JT | use of astdb or an sql db, or just variables to hold bits of data if you're itterating to different extensions |
| 08:49.36 | JT | blf, or try and call them |
| 08:49.49 | Chris-NB | JT, ok. set something in astdb if a call starts. an check that field |
| 08:50.34 | JT | sounds prone to problems |
| 08:50.48 | JT | logging extensions being in use in astdb |
| 08:50.54 | JT | you don't need to go that complex |
| 08:51.06 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:51.11 | Chris-NB | what would be better? |
| 08:53.07 | JT | i've suggested some ideas already |
| 08:53.13 | JT | there's quite a lot of options |
| 08:53.20 | JT | implementing is up to you |
| 08:56.38 | JT | ~thebook |
| 08:56.41 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:56.48 | Dirk- | is there a way to read some queue info (like how many callers are in a queue) in any other way than having a script read the output of "queue show" ? |
| 08:56.51 | JT | should give you a better idea of how things can be done |
| 08:57.11 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 08:58.38 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
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| 09:00.09 | creativx | Dirk-: show manager command QueueStatus |
| 09:04.04 | Dirk- | that doesnt seem to provide a useful output? |
| 09:05.31 | Dirk- | I was actually hoping to be able to read it out of the sql tables or something |
| 09:06.01 | JT | the sql tables of what? |
| 09:07.31 | Dirk- | Sorry, unclear. From wherever asterisk stores its config's. Would a value like the amount of calls waiting in a queue be stores in an accessible database somewhere on the asterisk server by default, or would that information only be available by interfacing with the management interface |
| 09:07.50 | JT | no. |
| 09:07.55 | JT | AMI is there for a reason |
| 09:08.02 | Dirk- | That answers that then :) |
| 09:09.27 | creativx | yup |
| 09:09.41 | creativx | hence my previous reference to QueueStatus ;) |
| 09:12.48 | Dirk- | I can do what I need by parsing the output of 'queue show', so its ok, but QueueStatus is confusing me, when I type 'show manager command QueueStatus' I get three lines in response showing "Action: QueueStatus | Synopsis: Queue Status | Privilege: none" is there something I should understand from this? |
| 09:13.08 | creativx | yes |
| 09:13.16 | creativx | you connect to the AMI on port 5038 |
| 09:13.19 | creativx | then login |
| 09:13.38 | creativx | then you send Action: QueueStatus\r\nQueue: queuename\r\n |
| 09:13.44 | creativx | and then parse whatever you get in return |
| 09:16.46 | Dirk- | hmm, I didnt know about this. I'll do some research - thanks a lot |
| 09:17.10 | creativx | ~ami |
| 09:17.11 | jbot | well, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 09:17.21 | creativx | the AMI is powerful |
| 09:17.22 | coldsteal | whats a good way to tell if MeetMe() is working? |
| 09:20.05 | creativx | show modules like |
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| 09:36.38 | coldsteal | i cant get meetme to work this is my error and configs http://rafb.net/p/sazn6f30.html |
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| 09:58.32 | *** join/#asterisk skitfish (n=skitfish@83-244-153-55.cust-83.exponential-e.net) |
| 09:59.10 | coldsteal | i cant get meetme to work this is my error and configs http://rafb.net/p/sazn6f30.html |
| 09:59.24 | skitfish | hey guys, I'd like a little help setting asterisk up to auto dial out and deliver a prerecorded message |
| 10:02.39 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 10:03.26 | *** part/#asterisk slima (i=slima@unaffiliated/slima) |
| 10:03.40 | *** join/#asterisk dreamind (n=dreamind@p54A79649.dip0.t-ipconnect.de) |
| 10:03.43 | dreamind | Hi folks |
| 10:04.12 | dreamind | can anybody help me how to restrict the applicationmap features only to local sip phones and not calls comming in through the zaptel device? |
| 10:05.54 | skitfish | can anyone point me to documentation on setting asterisk up to automatically dial out and deliver a prerecorded message? |
| 10:06.47 | r0d3nt | no voip spamming/telemarketer/political spammer. |
| 10:08.08 | dreamind | skitfish: look for call files and create a context in your extensions.conf for playing back a soundfile |
| 10:08.35 | dreamind | I personally use that for automatic messages when some server fails to work. |
| 10:08.36 | skitfish | thanks, I've done that (I followed http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message) |
| 10:08.45 | skitfish | yes that's exactly what I'd like to achieve! |
| 10:09.13 | dreamind | :) |
| 10:09.15 | skitfish | thanks, r0d3nt, for the reminder about telemarketing, etc, but I'm not using asterisk for commercial gain |
| 10:09.33 | dreamind | I personally use Festival() to play back my messages |
| 10:09.42 | dreamind | because I didn't want to record every single message ;) |
| 10:09.48 | skitfish | I wouldn't sink so low as to come here and ask for help on spamming people |
| 10:10.23 | dreamind | hm, nobody an idea on DYNAMIC_FEATURES? - I just would like to restrict it to either the caller or the callee |
| 10:10.39 | dreamind | and the features.conf documentation isn't that good on this point :( |
| 10:11.26 | coldsteal | dreamind: what r u trying? |
| 10:12.26 | dreamind | I'm trying to implement n-way conferences |
| 10:12.47 | dreamind | after: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 10:12.50 | dreamind | but for asterisk 1.2 |
| 10:13.01 | dreamind | just now I'm testing agi and co |
| 10:13.22 | dreamind | and I'm also using macros being called through the applicationmap |
| 10:13.30 | skitfish | Can someone give me a quick recommendation on what I need to deliver alerts to staff on the POTN besides an internet-connected laptop with asterisk installed on it, and an extensions.conf with an appropriate set of contexts in it? |
| 10:13.38 | dreamind | but I would like to restrict the usage of the applicationmap only to local sip clients |
| 10:13.53 | dreamind | currently also zap channels can use features defined in the applicationmap. |
| 10:16.33 | dreamind | coldsteal: hm, did that explanation clear things? |
| 10:19.23 | *** join/#asterisk Tili (n=tili@87.16.221.87.dynamic.jazztel.es) |
| 10:19.56 | coldsteal | yeah |
| 10:20.37 | skitfish | has anyone here set asterisk up to auto dial out and deliver a pre-recorded message? |
| 10:20.37 | skitfish | I'll be placing call files in /var/spool/asterisk/outgoing/ when some part of the network fails |
| 10:21.05 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:22.18 | dreamind | coldsteal: and any idea? :/ |
| 10:22.27 | *** part/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-a5c502138331f32a) |
| 10:22.34 | coldsteal | nope |
| 10:22.35 | coldsteal | lol |
| 10:23.54 | HaMYaI | why would " pcntl_signal(SIGHUP, SIG_IGN);" work on one script but not another |
| 10:24.19 | HaMYaI | both running on the same box |
| 10:24.37 | *** join/#asterisk boogieman (n=boogiema@203.143.22.146) |
| 10:26.15 | boogieman | hello all, i'm planning to setup a SW PBX.. however I am bit confused with the needed telephony hardware cards ... can you all help/advice me.. |
| 10:27.11 | boogieman | I have 4 PSTN lines; and 1 of them is used for faxes... |
| 10:27.53 | boogieman | my requirement is that when someone from the PSTN (outside) calls me, the phones should just ring |
| 10:28.03 | JT | it would save a lot of hassle if you kept the fax line seperate to asterisk |
| 10:28.04 | boogieman | and i should be able to take PSTN calls as well |
| 10:28.32 | boogieman | pls. tell me the HW requirement first for this setup |
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| 10:28.54 | boogieman | for the asterisk, I have a decent machine with a NIC available |
| 10:29.01 | JT | how many extensions? |
| 10:29.15 | boogieman | setting up the VoIP network is ok (i went through the tutorials) |
| 10:29.26 | boogieman | could be around 100 |
| 10:29.44 | boogieman | thinking of giving all the uses an extension |
| 10:29.45 | JT | 100 extensions for 3-4 lines? |
| 10:30.23 | boogieman | em...extension = each user |
| 10:30.38 | boogieman | they can talk to each other in office |
| 10:30.47 | JT | how many phone handsets/extensions? |
| 10:31.02 | boogieman | 3 |
| 10:31.09 | boogieman | rather 4 |
| 10:31.13 | boogieman | with the FAX |
| 10:31.22 | JT | that's not 100 extensions |
| 10:31.26 | dreamind | coldsteal: |
| 10:31.27 | dreamind | ups |
| 10:31.28 | JT | keep fax seperate |
| 10:31.29 | dreamind | coldsteal: :( |
| 10:32.07 | boogieman | ok; i'll keep that line separate ... i saw that FAX handling is bit different in a VoIP network |
| 10:32.27 | boogieman | JT: pls. tell me about the FXO hardware I should buy |
| 10:32.59 | JT | skitfish: what is POTN? |
| 10:33.26 | boogieman | most of the sights i visited; digium and sangoma talk about separate FXO/FXS modules .. i think |
| 10:33.36 | JT | dreamind: you have a UPS? |
| 10:33.41 | dreamind | yes |
| 10:33.47 | JT | boogieman: you have 3 FXO, get something with 3 FXO capability |
| 10:34.00 | JT | dreamind: was trying to work out the context |
| 10:34.13 | dreamind | JT: hehe ok ^^ |
| 10:34.14 | skitfish | plain old telephone network |
| 10:34.21 | skitfish | sorry, it's probably a bogus acronym |
| 10:34.55 | JT | PSTN |
| 10:34.57 | JT | POTS |
| 10:35.01 | skitfish | yup |
| 10:35.59 | *** join/#asterisk oej_ (n=olle@soll4-125.cust.blixtvik.net) |
| 10:36.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:36.26 | boogieman | JT: will the Rhino R8FXX-EC Modular PCI Plug-In Card do? its rather cheap than than the quad FXO card ? |
| 10:36.41 | JT | no idea |
| 10:36.45 | JT | no-one uses rhino |
| 10:38.06 | boogieman | JT: any particular reason why its not used ? |
| 10:39.00 | creativx | heh oops.. so funny to detect config errors by accident |
| 10:39.00 | JT | err, because it's not |
| 10:39.09 | JT | it's just a minor brand |
| 10:39.14 | JT | nothing spectacular |
| 10:39.49 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 10:40.29 | boogieman | JT: how about the Sangoma A200? |
| 10:41.08 | JT | should be fine |
| 10:43.54 | boogieman | JT: this is the problem; the A200 (and other HW) talks about 2 modules; FXO and FXS; do i need to have the FXS model ? or must i have 2 FXO modules to utilize the quad RJ11 ports ? |
| 10:44.11 | boogieman | model=modules |
| 10:45.13 | JT | boogieman: no. you need 3 FXO ports. |
| 10:46.25 | skitfish | so, dialing out to the PSTN to notify staff of problems on the network via a pre-recorded message, example configs anyone? |
| 10:46.51 | coldsteal | whhere do i get the default recordings |
| 10:46.51 | coldsteal | can i download them again |
| 10:46.52 | coldsteal | im missing allot |
| 10:46.58 | coldsteal | like vm-rec-name |
| 10:47.28 | boogieman | JT: if you look at the A200 demo; they talk about 2 ports per module; its like FXO/FXO, FXO/FXS... FXS,FXS .. but each module is like binded to just 2 RJ11 ports... that's why i'm asking whether we need to buy 2 FXO modules with the A200 |
| 10:48.15 | JT | boogieman: then yes |
| 10:48.16 | boogieman | JT: I already sent them an email; they are yet to respond.... couldn't wait till then...:D |
| 10:49.48 | JT | what do you need to email them for? |
| 10:54.42 | boogieman | JT: pricing and this FXO/FXS issue |
| 10:55.00 | boogieman | JT: do they have product prices online ? |
| 10:55.01 | JT | they don't sell direct |
| 10:55.14 | JT | see the whole "resellers/distributors" section of their site.... |
| 10:56.34 | boogieman | JT: what i need is to know the price (even a rough figure), but their sites just point to distributor web addresses and contacts |
| 10:56.46 | JT | well go to them |
| 10:56.55 | JT | you must buy it from a retailler |
| 10:59.01 | boogieman | JT: i'm in Sri Lanka, the Perl of the Indian Ocean :) ... i'll have to call them and get the price |
| 10:59.05 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 10:59.15 | boogieman | JT: no resellers in LK |
| 10:59.29 | JT | not my problem |
| 10:59.33 | JT | search the Internet |
| 10:59.51 | JT | plenty of american online stores sell it, maybe that will give you an idea of the price |
| 11:02.29 | boogieman | JT: thanks |
| 11:06.07 | *** join/#asterisk Stormfr (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 11:11.09 | Stormfr | I try to get hangupcause with AGI, but with IAX channel, i don't have any data return. any idea where could be the problem ? no problem with a sip call |
| 11:17.11 | coldsteal | okay i still cant get meetme() to work i did get a little farther tho |
| 11:17.13 | coldsteal | http://rafb.net/p/j4bRWX70.html |
| 11:17.44 | coldsteal | that what shoes up in asterisk -vvvvvvvvvvr |
| 11:17.57 | coldsteal | dam thats to meny Vs |
| 11:18.40 | boogieman | JT: what is the use of a FXS port in a VOIP network such as what i'm planning to setup ? |
| 11:19.05 | JT | boogieman: i thought you wanted to connect to your PSTN lines, my mistake |
| 11:19.22 | coldsteal | it also doesnt see my vm-rec-name.gsm thats is /var/lib/asterisk/sounds/vm-rec-name.gsm |
| 11:20.51 | boogieman | JT: you thought correct; i want to connect to my 3 PSTN lines (minus the FAX line) .. i'm just asking about the FXS port and its would be use |
| 11:21.11 | JT | i did not say to get an FXS port |
| 11:21.29 | boogieman | JT: no you didn't; you asked me to get 3 FXO ports |
| 11:21.48 | boogieman | JT: apart from that; i'm pondering what sort of use a FXS port would do ? |
| 11:22.04 | DrukenLPY | FXS is for connecting analog phones to your system... but i personally reccomend ata's |
| 11:22.28 | *** join/#asterisk friedrich| (n=friedric@e177251158.adsl.alicedsl.de) |
| 11:22.56 | boogieman | DrukenLPY: meaning; analog phones to my VOIP phone system ? |
| 11:22.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:23.26 | DrukenLPY | yes |
| 11:24.41 | boogieman | DrukenLPY: ata's imply IP Phones ? |
| 11:24.58 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:24.58 | JT | No. |
| 11:24.58 | DrukenLPY | analog telephone adaptor, it's a device |
| 11:25.29 | skitfish | can anyone help with my setup? |
| 11:26.05 | DrukenLPY | skitfish: ask questions, if we have answers we'll give them |
| 11:26.27 | skitfish | ok |
| 11:27.11 | boogieman | DrukenLPY: analog phone <--> ATA <--> VOIP Network <--> PBX <--- FXO card ---> PSTN ... is my understanding correct ? |
| 11:27.29 | coldsteal | does meetme() have to have a pryority of 1? |
| 11:27.32 | DrukenLPY | looks right... |
| 11:28.11 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 11:28.24 | JT | coldsteal: are you drunk? |
| 11:28.50 | coldsteal | why? |
| 11:28.54 | coldsteal | i cant spell |
| 11:29.02 | coldsteal | lol i dont drink either |
| 11:29.18 | skitfish | Ideally I would like an example setup for a system which auto dials out on the POTS and delivers a prerecorded message, when a call file is placed in the relevant directory |
| 11:29.19 | deegan | Is there a way to use GotoIfTime so that it works like an if-if-else statement? I have to GotoIfTime and if it's not anyone of thoose i want it to play a message. |
| 11:29.30 | deegan | s/to/two. |
| 11:30.04 | JT | coldsteal: yeah, your typing atm... |
| 11:30.15 | DrukenLPY | skitfishL well, don't count on me helping ya... i don't help asshole telemarketers :) |
| 11:30.24 | JT | skitfish: take a look at the book and work out how to do it yourself |
| 11:30.26 | JT | ~thebook |
| 11:30.27 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:32.37 | skitfish | if telemarketing is calling the mobiles of engineers with a prerecorded message to let them know when the network is down so they can go fix it then yes, I'm a telemarketer |
| 11:33.56 | JT | skitfish: it's quite easy, just check out the book and read the sample.call file |
| 11:34.08 | skitfish | ok thanks, but what about hardware requirements? |
| 11:34.40 | JT | what about them? same as for making any other calls |
| 11:34.52 | JT | you will need hardware if you're connecting to physical lines |
| 11:35.18 | HaMYaI | can't we "reload module res_agi.so" from cli? |
| 11:36.30 | *** join/#asterisk lukassky (n=lukass@212.145.121.103) |
| 11:36.38 | lukassky | hi everybody |
| 11:37.00 | DrukenLPY | skitfish: if the network is down... how do you expect to make calls? |
| 11:37.22 | creativx | magic airwaves |
| 11:37.23 | creativx | ofcourse |
| 11:37.25 | JT | DrukenLPY: physical lines :) |
| 11:37.37 | DrukenLPY | JT: he said cell phones :) |
| 11:37.51 | JT | yes |
| 11:37.58 | JT | calling mobiles on the other end |
| 11:38.12 | JT | that does not preclude you from using physical lines to make the call |
| 11:38.46 | DrukenLPY | agreed |
| 11:39.25 | skitfish | yes, true, but it's also perfectly possible to have a host with more than one network adapter installed in it |
| 11:39.25 | skitfish | one could be connected to a network which is susceptible to going 'down' and the other could be use to send out 'network down' notifications on a second network |
| 11:39.51 | creativx | also called "redundancy" |
| 11:39.54 | creativx | how about a power outage |
| 11:39.55 | creativx | :) |
| 11:40.06 | DrukenLPY | creativx: hehe |
| 11:41.40 | skitfish | ok you got me ;P |
| 11:43.18 | creativx | ofcourse you get a gsm module on a separate powersupply |
| 11:43.38 | creativx | .. or you could have external monitoring |
| 11:43.44 | creativx | the opportunities are endless :P |
| 11:44.09 | DrukenLPY | ya get a gridtie system with a hybrid system behind it with solar, wind and diesel gennies :) |
| 11:49.06 | skitfish | thanks guys |
| 11:50.32 | DrukenLPY | wtf is up with the trucks in toronto... god damn... 3 major roll overs just this week... AND THIS IS ONLY THURSDAY!!!! |
| 11:51.27 | skitfish | any fatalities? |
| 11:52.00 | DrukenLPY | i don't think so... |
| 11:54.31 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 11:55.02 | creativx | mm thursday |
| 11:55.10 | creativx | damn its thursday already. mini friday! sweet shit |
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| 12:11.06 | JT | creativx: it's friday in 2 hours ;) |
| 12:11.15 | creativx | bastard :) |
| 12:12.54 | tuzhila | JT, where are you live? |
| 12:13.10 | Maliuta | JT: 45 mins |
| 12:13.25 | Maliuta | not like I have to be back at work until monday anyhow |
| 12:14.35 | JT | tuzhila: sydney, australia |
| 12:14.48 | tuzhila | oh, cool! |
| 12:14.55 | tuzhila | i want to visit australia |
| 12:15.04 | JT | Maliuta: 45mins, how do you figure? |
| 12:15.17 | Maliuta | it's 22:15 |
| 12:15.31 | JT | 1h 45mins |
| 12:15.35 | JT | not 45mins |
| 12:15.55 | Maliuta | sydney blows goats, and melbourne is not far behind ... I'm moving back to Brisneyland |
| 12:16.08 | JT | eww brisbane sucks balls :P |
| 12:16.17 | JT | Maliuta: so still think it's 45mins? :P |
| 12:16.17 | Maliuta | JT: yeah, yeah. I blame the scotch for losing an hour :P |
| 12:16.49 | Maliuta | went to a bar called "Spleen" after dinner, had a Morangie |
| 12:16.59 | Maliuta | brisbane rocks |
| 12:17.10 | Maliuta | far betterer than hellbourne |
| 12:17.18 | JT | melbourne is far better |
| 12:17.22 | JT | brisbane has awful weather |
| 12:17.26 | Maliuta | not really |
| 12:17.28 | JT | it's like a country town |
| 12:17.30 | JT | yes, really |
| 12:17.37 | DrukenLPY | piss on all of aussie... :) |
| 12:17.41 | Maliuta | no, not really |
| 12:17.53 | JT | brisbane is way too hot and sunny |
| 12:18.05 | Maliuta | been in hellbourne for almost 2 years now, more miserable than ever |
| 12:18.12 | skitfish | the Aussie in my office says Brisbane is getting better |
| 12:18.17 | Maliuta | you call _this_ weather? |
| 12:18.18 | creativx | i know some aussies in brisbane |
| 12:18.23 | creativx | they are mostly crazy in the head |
| 12:18.24 | skitfish | but he's with us here in London so we don't listen to him |
| 12:18.31 | creativx | and one in melbourne who is a total nutter |
| 12:18.31 | JT | i visit melbourne every year once or twice |
| 12:18.31 | Maliuta | better than it used to be |
| 12:18.36 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:18.48 | JT | brisbane is an utter yawn |
| 12:18.52 | JT | too many bogans too |
| 12:18.53 | Maliuta | nice to visit, crap to live in ... just like canberra |
| 12:19.08 | Maliuta | not if you go to the right places |
| 12:19.12 | JT | melbourne is a much better place to live in than canberra |
| 12:19.17 | Maliuta | there are more bogons in melbourne |
| 12:19.24 | JT | haha |
| 12:19.29 | waKKu | u need came to brazil :D |
| 12:19.39 | Maliuta | it's bogon central down here |
| 12:19.47 | JT | ok i think we've found a queenslander in denial of queensland's boganess |
| 12:19.55 | Maliuta | waKKu: only to Ipenema :P |
| 12:20.19 | Maliuta | I know where the QLD bogons are .... heck my brother is one |
| 12:20.27 | waKKu | Maliuta nah.. Ipanema is just a hoax.. u need to know the island from Santa Catarina (Florianopolis) :) |
| 12:20.28 | Maliuta | I just also know how to avoid them |
| 12:20.33 | waKKu | where i live, now :) |
| 12:21.02 | JT | Maliuta: and avoiding the sun? |
| 12:21.20 | waKKu | if u like surfing, sandboarding, best parties and best beaches - here is the place :) |
| 12:22.30 | JT | i don't think i'll like it too much, waKKu |
| 12:22.36 | lirakis | morning |
| 12:22.43 | JT | sounds like almost everything i want to avoid :P |
| 12:23.01 | waKKu | JT hehe.. well.. here have some lan houses too :D |
| 12:23.07 | Maliuta | JT: you need to go live in antarctica for a bit :P |
| 12:23.18 | JT | Maliuta: too extreme weather conditions |
| 12:23.47 | Maliuta | JT: so maybe london then, somewhere with miserable weather anyhow |
| 12:23.59 | skitfish | yeah London has miserable weather right now |
| 12:24.02 | skitfish | I can vouch for that |
| 12:24.11 | skitfish | Grey skies, generally depressing |
| 12:24.20 | creativx | whee we have sun.. and grey skies |
| 12:24.30 | JT | i like overcast sky |
| 12:24.33 | creativx | how can it be depressing, we have already established that its thursday and soon to be friday :) |
| 12:24.37 | JT | doesn't have to be grey |
| 12:24.40 | skitfish | for you maybe |
| 12:24.45 | JT | just shielded from evil sun |
| 12:24.48 | skitfish | 10 hours 35 mins to go for me |
| 12:24.57 | skitfish | (but then I'm off to Sweden on Saturday :)) |
| 12:25.52 | Maliuta | JT: I'm a goth, and I thought _I_ was bad for depression, you sound worse |
| 12:26.16 | skitfish | damn day star |
| 12:26.26 | enioreh | goth ? |
| 12:26.55 | enioreh | like those ppl dressed in black and red, having sex on the dancefloor |
| 12:27.06 | waKKu | lol |
| 12:27.09 | Maliuta | no red |
| 12:27.14 | Maliuta | goth > emo |
| 12:27.18 | enioreh | you don't like blood ? |
| 12:27.19 | skitfish | sex on the dancefloor?? I know what goths are, but I've never seem them doing that! |
| 12:27.20 | enioreh | so sad :/ |
| 12:27.35 | enioreh | skitfish: you haven't been to so right place so |
| 12:27.36 | Maliuta | I take my sex where I can get it, dancefloor, park .... |
| 12:27.45 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 12:27.46 | skitfish | lol |
| 12:27.56 | waKKu | emo? http://www.youtube.com/watch?v=w-1IW7AkrSs |
| 12:27.56 | creativx | sex on dancefloor aka chlamydia pit |
| 12:28.18 | JT | Maliuta: depression? |
| 12:28.26 | JT | Maliuta: i'm not depressed |
| 12:28.33 | Maliuta | waKKu: seen it |
| 12:28.39 | waKKu | hehehe |
| 12:28.39 | enioreh | JT: all depressed ppl says that |
| 12:28.59 | JT | enioreh: sorry? |
| 12:29.01 | Maliuta | waKKu: http://www.youtube.com/watch?v=LkDhh1pfG-4 |
| 12:29.09 | JT | enioreh: what evidence do you have that i'm depressed |
| 12:29.10 | JT | ? |
| 12:29.29 | waKKu | let me c |
| 12:29.36 | enioreh | you says that you are not. that's the proof you are. |
| 12:29.44 | JT | enioreh: you are an extremely rude, extremely ignorant individual |
| 12:29.57 | coppice | JT: are you a rational human being, who looks squarely at the world around? |
| 12:30.18 | JT | enioreh: you've made a medical assessment over irc from the little time you've spend around here and a few lines of my chat? |
| 12:30.22 | JT | coppice: i'd like to think so |
| 12:30.26 | enioreh | note that the same works for junky, gay and terrorist |
| 12:30.26 | DrukenLPY | that youtube is soo wrong |
| 12:30.34 | coppice | see. you are depressed |
| 12:30.40 | JT | i see |
| 12:30.42 | JT | clearly |
| 12:30.44 | tuzhila | 8495.... is it vietnam? |
| 12:31.11 | JT | enioreh: also please learn to use proper english |
| 12:31.24 | JT | enioreh: i hear intentional poor spelling is a sign of depression |
| 12:31.31 | pj_ | pwned |
| 12:31.47 | enioreh | JT: you lie. |
| 12:31.54 | creativx | i would rather classify you as a coconut crazy JT. |
| 12:32.10 | creativx | performed over the irc interweb net, free of charge. |
| 12:32.13 | JT | creativx: at least that's not depressed! |
| 12:32.44 | lukassky | er..it's that #asterisk o #fightin'-club? |
| 12:33.02 | JT | so apparently liking cooler climates and avoiding skin cancer makes one depressed ;) |
| 12:33.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:33.29 | DrukenLPY | move to canada :) |
| 12:33.37 | JT | maybe |
| 12:33.46 | JT | your winters seem extreme |
| 12:33.48 | Maliuta | JT: I have nothing to fear from cancer. I already hold the record for highest circulating Leukaemic count in an adult :P |
| 12:33.56 | JT | Maliuta: nice |
| 12:34.00 | Maliuta | cancer is my bitch :P |
| 12:34.06 | DrukenLPY | but we have a temp rang of 60 degrees celcius |
| 12:34.12 | DrukenLPY | range |
| 12:34.25 | Maliuta | I keep it in the closet, and pull it out for dinner parties :) |
| 12:36.07 | JT | Maliuta: so do you like being "so sad" (not my words) |
| 12:36.41 | waKKu | Maliuta damn long and annoying video.. but some cool :) |
| 12:37.42 | Maliuta | waKKu: "goth chick. that always happens; you think 'thats a hot goth chick, I'm gonna get me some' and it turns out to be sissy boy" |
| 12:37.59 | Maliuta | JT: you learn to cope :P |
| 12:38.13 | JT | Maliuta: gee i dunno how you manage </sarcasm> |
| 12:39.03 | Maliuta | JT: I take 30mg of morphine a day :P |
| 12:40.15 | [TK]D-Fender | Maliuta: "I wish my lawn was EMO... because it would cut itself" |
| 12:40.33 | JT | "i wish i had a new emo joke" |
| 12:40.44 | [TK]D-Fender | http://icanhascheezburger.com/2007/01/16/go-cry-emo-kid/ |
| 12:41.07 | Uzzi | does asterisk work with Conexand HCF modem? |
| 12:42.38 | [TK]D-Fender | Uzzi: No. |
| 12:42.39 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 12:42.47 | Uzzi | sigh |
| 12:42.54 | [TK]D-Fender | Uzzi: Go to the WIKI and read the hardware compatability list. |
| 12:42.57 | [TK]D-Fender | ~wikis |
| 12:42.58 | jbot | from memory, wikis is http://www.voip-info.org |
| 12:43.46 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 12:49.18 | *** join/#asterisk saftsack (n=saftsack@pD9E058A6.dip.t-dialin.net) |
| 12:50.51 | Uzzi | :( |
| 12:51.14 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 12:51.19 | santibiotico | hi |
| 12:51.43 | santibiotico | i'm tryng to use the multilingual app |
| 12:52.01 | santibiotico | when i use Set(Language("es")) |
| 12:52.04 | santibiotico | sorry |
| 12:52.10 | santibiotico | when i use Set(Language()="es") |
| 12:52.46 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
| 12:53.13 | santibiotico | i get the following error |
| 12:53.13 | santibiotico | Jul 26 14:52:10 ERROR[2893]: pbx.c:1417 ast_func_write: Function Language not registered |
| 12:53.19 | santibiotico | any help? |
| 12:53.23 | Corydon76-home | Set(LANGUAGE()=es) |
| 12:53.28 | Corydon76-home | all caps |
| 12:53.34 | santibiotico | i've tried it too |
| 12:54.05 | *** part/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 12:54.18 | Corydon76-home | Function names are cAsE-sEnSiTiVe |
| 12:54.41 | santibiotico | i've tried it with all caps |
| 12:54.44 | santibiotico | and the same error |
| 12:54.47 | [TK]D-Fender | santibiotico: Show us |
| 12:54.50 | Corydon76-home | What version? |
| 12:55.12 | Corydon76-home | It's about to become CHANNEL(language) |
| 12:55.27 | santibiotico | Jul 26 14:54:15 ERROR[2902]: pbx.c:1417 ast_func_write: Function LANGUAGE not registered |
| 12:56.07 | santibiotico | 1.2.14 |
| 12:56.53 | [TK]D-Fender | santibiotico: Not 1-lines, full CLI output including the version info when you start it up. |
| 12:57.00 | [TK]D-Fender | ~pb |
| 12:57.01 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:57.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 12:57.12 | Corydon76-home | santibiotico: "load func_language.so" |
| 12:57.34 | [TK]D-Fender | santibiotico: And your dialplan as well please. |
| 12:58.35 | EricL | Is there an equivilent way to do a SipAddHeader(foo: bar) in a .call file? |
| 12:59.13 | Corydon76-home | EricL: no |
| 12:59.21 | Corydon76-home | Well, kind of |
| 13:00.06 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 13:00.23 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 13:00.24 | EricL | Corydon76-work: I create a bunch of .call files from an AGI script and I need the added SIP header to get them to work properly. |
| 13:00.53 | Corydon76-home | EricL: SetVar: __SIPADDHEADER01=foo |
| 13:01.21 | Corydon76-home | but that's not guaranteed to work in future versions |
| 13:01.47 | [TK]D-Fender | EricL: Pick a channel type that lets you set stuff like this before you actually do your dial. There is only 1 option for this. Think on it. HARD. It really ISN'T. |
| 13:01.49 | Corydon76-home | If you need it to work reliably, then you should insert it into your dialplan |
| 13:01.51 | EricL | Is someone planning on adding a way to add SIP headers to .call files? |
| 13:02.06 | Corydon76-home | EricL: No, we are not |
| 13:02.27 | EricL | I have it inserted into my dialplan, but it doesn't seem to work with the .call files. |
| 13:02.49 | *** join/#asterisk morex (n=m@91.84.56.12) |
| 13:02.52 | Corydon76-home | EricL: then use a Local channel |
| 13:02.55 | morex | Hi there |
| 13:03.08 | [TK]D-Fender | Corydon76-home: There you g, just blurting it out again! |
| 13:03.11 | morex | We keep getting unexplained Yellow alarms on our E1 ISDN |
| 13:03.13 | [TK]D-Fender | go* |
| 13:03.14 | morex | With this error: |
| 13:03.26 | morex | [Jul 26 13:33:49] ERROR[17867] chan_zap.c: Write to 153 failed: Unknown error 500 |
| 13:03.26 | morex | [Jul 26 13:33:49] ERROR[17867] chan_zap.c: Short write: 0/15 (Unknown error 500) |
| 13:03.32 | morex | Anyone seen anything like this before? |
| 13:03.36 | [TK]D-Fender | Corydon76-home: I was attempting go get him to THINK! |
| 13:03.44 | EricL | I am still learning *, I guess I need to go figure out the difference between a local and non-local channel. |
| 13:03.46 | morex | * is hanging up when the alarm clears, and our customer is pissed... |
| 13:03.47 | Corydon76-home | morex: yellow alarm means that the OTHER side saw red |
| 13:04.01 | Corydon76-home | morex: it's ONLY an indication of the remote status |
| 13:04.22 | morex | Is there any way we can just ignore the yellow alarms, and not hang up the channel when it clears? |
| 13:04.33 | Corydon76-home | No, you cannot |
| 13:04.34 | [TK]D-Fender | EricL: SIP is a non-Local channel, so is Zap, IAX2, and every other device. |
| 13:04.46 | morex | 'Cos they're threatening to uninstall us... |
| 13:05.03 | Corydon76-home | EricL: no, Local, as in the Local driver |
| 13:05.18 | morex | OK Corydon thanks for the info |
| 13:05.31 | *** join/#asterisk hyphen (n=hyphen@dsl081-022-034.phl1.dsl.speakeasy.net) |
| 13:05.37 | EricL | Corydon76-work: Ok, there is no need for me to waste your guys time on this. I don't know what the local driver is, so I have some reading to do. |
| 13:05.38 | Corydon76-home | morex: bitch to your provider |
| 13:05.56 | Corydon76-home | EricL: "show channeltypes" |
| 13:05.56 | morex | Actually we're connecting to their PBX directly |
| 13:06.09 | morex | And the Yellows always come on the connection to the PBX |
| 13:06.20 | Corydon76-home | morex: their PRI card is probably dying |
| 13:06.41 | morex | It's brand new. |
| 13:06.44 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 13:06.45 | Corydon76-home | morex: are they syncing to you? |
| 13:06.48 | *** join/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 13:06.52 | Rienzilla | yeeha |
| 13:06.57 | Rienzilla | meetme works ... finally |
| 13:06.58 | Corydon76-home | Failure to sync could also cause weird issues |
| 13:07.05 | morex | Um, how would I check that? |
| 13:07.31 | Corydon76-home | morex: zaptel.conf. Are you generating signalling (1), or are you taking it off the line (0) ? |
| 13:07.39 | Corydon76-home | Err, switch that |
| 13:07.40 | morex | Checking... |
| 13:07.47 | Corydon76-home | generating is 0, taking it is 1. |
| 13:08.36 | MindTheGap | hello all... I set up this old Dell Poweredge 600SC server w a digium TE110. drivers load ok, zttool reports green, call comes in on zap, gets routed fine, i hear voice both ways on sip clients but as soon as the call finishes the server crashes, wont ping or anything, it just locks up, no info, anything... i read it may be something with onboard video w shared mem or gigabit ethernet. anyone had this problem? what should i check first? |
| 13:08.57 | [TK]D-Fender | EricL: chan_local is like a wrapper that has one end of the call as dialplan. Inside of there you could do your actual DIAL (without doing an Answer first) after issuing your header to add. |
| 13:09.03 | Corydon76-home | minkus: crashes or locks up? |
| 13:09.21 | morex | We've got four ports |
| 13:09.29 | morex | The first two connected to the network, timing 1 and 2 |
| 13:09.35 | Corydon76-home | MindTheGap: crash or lockup are completely two DIFFERENT things |
| 13:09.43 | MindTheGap | locks up... |
| 13:09.44 | morex | The second two connected to the pbx, timing 3 and 4 |
| 13:10.12 | MindTheGap | i think... it just freezes, stops pinging but wont reboot or anything... |
| 13:10.23 | Corydon76-home | morex: okay, the two connected to the pbx should both be 0 |
| 13:10.23 | MindTheGap | it wont log anything either... |
| 13:10.34 | morex | OK thanks Corydon |
| 13:10.39 | morex | I'll give it a try tonight. |
| 13:10.43 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 13:11.04 | Corydon76-home | MindTheGap: then yes, it's definitely a hardware problem |
| 13:11.11 | MindTheGap | wathing the cli, asterisk hangs um the call, writes to CDR than nothing... |
| 13:11.19 | Corydon76-home | MindTheGap: call Digium support |
| 13:12.17 | EricL | [TK]D-Fender: So your saying that the .call files should use Local/XXX@internal/ and this will send it back into the dialplan where I can properly use the SipAddHeader command to do the dial? |
| 13:13.12 | [TK]D-Fender | EricL: EXACTLY. |
| 13:13.37 | EricL | Damn, you guys thought of everything. |
| 13:14.14 | HaMYaI | Corydon76-home: hi, I need your quick advice regarding SIGHUP and Dial with the "g" option |
| 13:15.02 | HaMYaI | Corydon76-home: my agi script ignores the SIGHUP but since 1.2.20 the Dial will still exit |
| 13:15.34 | HaMYaI | Corydon76-home: I needed to place the Dial with "g" option to get it work |
| 13:19.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 13:31.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:37.27 | Sci_05 | was the system command replaced by a different command in the 1.4? I noticed its not in the core show function section any more |
| 13:38.22 | [TK]D-Fender | Sci_05: it : Show us :) |
| 13:38.57 | Rienzilla | good morning |
| 13:39.51 | Sci_05 | [TK]D-Fender: 3 guys walk into a bar.......the 4th one ducks |
| 13:40.15 | [TK]D-Fender | Sci_05: c'mon.... pastebin it up.... |
| 13:41.04 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:41.37 | Sci_05 | [TK]D-Fender: here is what I am trying to do http://pastebin.com/m74b3900c |
| 13:42.19 | [TK]D-Fender | Sci_05: Show me whree it tells you the System doesn't exist.... |
| 13:42.29 | [TK]D-Fender | Sci_05: Not 10 things that DON'T |
| 13:43.39 | Sci_05 | if I do a core show function ? SYstem doesn't show up http://pastebin.ca/634523 |
| 13:44.25 | [TK]D-Fender | Sci_05: And do you know why that is? |
| 13:44.38 | Sci_05 | nope thats why I am asking |
| 13:44.40 | [TK]D-Fender | Sci_05: Becase it isn't a **FUNCTION** |
| 13:45.04 | Sci_05 | ok fill me in then :) |
| 13:45.08 | Stormfr | I try to get hangupcause with AGI, but with IAX channel, i don't have any data return. any idea where could be the problem ? no problem with a sip call |
| 13:45.13 | [TK]D-Fender | Sci_05: is is an APPLICATION <--------------- |
| 13:45.35 | *** join/#asterisk j-goddess (n=humblein@phrank.aus.us.siteprotect.com) |
| 13:45.36 | [TK]D-Fender | Sci_05: One, which in your case isn't even being CALLED in the dialplan. |
| 13:46.20 | Strom_M | welcome to the "Someone in Quebec hasn't had their morning coffee yet" show |
| 13:46.30 | Sci_05 | lol |
| 13:47.15 | EricL | If I have the .call file use Local/XXX@internal2/n, then can I have internal2 drop the user into a MeetMe after Dial or will it just Dial and wait for the talking to occur? |
| 13:47.23 | Sci_05 | do you know why its not being called in my dialplan? its driving me nuts |
| 13:48.12 | HaMYaI | anyone knows where to look for the change log for Dial command? |
| 13:48.28 | [TK]D-Fender | EricL: a call file has 2 sides. when the one started by Channel:" Answers, it is then dumped into the dialplan where specified. |
| 13:48.51 | [TK]D-Fender | Strom_M: Sure I have, nothing is slipping by me :) |
| 13:49.38 | HaMYaI | for 1.2 branch |
| 13:49.43 | [TK]D-Fender | EricL: your first side should just do your header add, then actually DIAL the party. the rest of your call-file is as it was before to dump into meetme |
| 13:50.10 | DarKnesS_WolF | i hav a very strange problem .... a GSM device pluged in FXO port in asterisk was working and now when i try to dial using it when the mobile answers i got hangup from teh zaptel channel any idea how to debug that ? |
| 13:50.26 | EricL | [TK]D-Fender: Should the MeetMe command be in the dialing context or the context that called the AGI? |
| 13:50.48 | [TK]D-Fender | EricL: meetme should be right where you had it the first time. |
| 13:50.55 | HaMYaI | okay found it |
| 13:51.02 | [TK]D-Fender | EricL: and noone mentioned AGI |
| 13:51.45 | EricL | [TK]D-Fender: I mentioned AGI, all my call files are being generated via an AGI script. |
| 13:51.53 | Shoeb | Hello [TK]D-Fender, problem #1 of the day.. what is this: Jul 26 09:51:42 NOTICE[14402]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 13:52.27 | Shoeb | Meanwhile, Xlite shows the error: |
| 13:52.40 | Shoeb | Call failed: Service unavailable |
| 13:52.47 | [TK]D-Fender | EricL: Ah, well remember you dial "A" then drop them into your dialplan. the AGI has nothing to do with that call-out once the file is created. |
| 13:53.21 | [TK]D-Fender | Shoeb: PASTEBIN isyour friend. , and so is "sip debug". Togther they are AWESOME. |
| 13:53.30 | Shoeb | lol ok |
| 13:54.04 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 13:54.11 | EricL | [TK]D-Fender: Right, but the context of that call once its answered is determined by the .call file or is continued from where the Local/XXX@<other_context>/n sent it? |
| 13:54.53 | hyphen | anyone have any luck with sip behind an openBSD gateway/firewall? |
| 13:55.11 | Shoeb | [TK]D-Fender: http://pastebin.ca/634529 |
| 13:55.47 | [TK]D-Fender | EricL: once your local channel is answered. you can have that channel do all sorts of NON call related stuff, and just bomb out if you wanted. |
| 13:56.18 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-233.net.novis.pt) |
| 13:56.59 | MrMister2 | Does anyone know if it's possible to reset a sip phone from the CLI? |
| 13:57.14 | HaMYaI | anyone know what's the relationship between SIGHUP, DeadAGI and Dial() with 'g' option? |
| 13:57.41 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 13:57.50 | [TK]D-Fender | Shoeb: And your sip.conf please.... |
| 13:57.59 | syzygyBSD | wow, already up to 1.2.23? |
| 13:57.59 | [TK]D-Fender | MrMister2: Depends on the phone. |
| 13:58.00 | HaMYaI | If I already use DeadAGI(), why do I still need to ignore SIGHUP? |
| 13:58.02 | MrMister2 | I have a extension that is in ExtensionState 16 which core show hints shows as being on hold and would like to reset it from the CLI. |
| 13:58.11 | [TK]D-Fender | syzygyBSD: Don't jinx it ;) |
| 13:58.29 | syzygyBSD | how long until the 1.2 releases are discontinued? |
| 13:58.50 | [TK]D-Fender | syzygyBSD: not for some time, though it will only receive BUGFIXES very shortly... |
| 13:59.12 | MrMister2 | [TK]D-Fender: It's a standard analog phone connected to a FXS port of a draytek router that registers on asterisk as a standard SIP extension |
| 13:59.13 | syzygyBSD | right, probably security patches for a long time too |
| 13:59.34 | [TK]D-Fender | MrMister2: then go read the manual for your draytek router |
| 13:59.50 | MrMister2 | [TK]D-Fender: aparently it's a bug that's been solved in Asterisk. http://bugs.digium.com/view.php?id=10165 |
| 14:00.05 | Shoeb | [TK]D-Fender: http://pastebin.ca/634534 |
| 14:00.25 | MrMister2 | [TK]D-Fender: My question was if there was any way to send the equivalent of a SIP reset peer or similar |
| 14:00.26 | ccesario | HaMYaI, try use AGI |
| 14:00.41 | [TK]D-Fender | MrMister2: I've already answered you. |
| 14:00.42 | ccesario | with 'g' in Dial command |
| 14:00.56 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-107-175-93.ny325.east.verizon.net) |
| 14:01.43 | *** join/#asterisk maz1977 (n=maz@ip153.metafora.mi.it) |
| 14:02.34 | [TK]D-Fender | Shoeb: and "sip show peers" <- |
| 14:02.45 | Dirk- | the Extension State 16 issue is NOT fixed |
| 14:02.46 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:02.47 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:02.53 | MrMister2 | [TK]D-Fender: I understood your answer. But that wasn't the question. I understand that in this specific case I must reset the extension at the router and not at the CLI. Since you said and I quote "it depends on the phone" I would like to know what that means in terms of CLI commands. |
| 14:02.59 | maz1977 | hi all. I'm trying to test asterisk in my officie. I have some question about analog device |
| 14:04.22 | [TK]D-Fender | MrMister2: You asked how to reset a remote device and as I mentioned it depend on the DEVICE. Your Draytek may not HAVE a way to be remotely reset nor is there any STANDARD for doing so. |
| 14:04.28 | Shoeb | [TK]D-Fender: http://pastebin.ca/634541 |
| 14:04.44 | [TK]D-Fender | Shoeb: see this? 200/200 (Unspecified) D N 0 UNKNOWN |
| 14:04.51 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 14:04.53 | [TK]D-Fender | Shoeb: * doesn't know how to GET to that phone. |
| 14:05.01 | [TK]D-Fender | Shoeb: It isn't registered. |
| 14:05.05 | Shoeb | [TK]D-Fender: I see 200 is not registered. But yesterday when 200 wasn't, I called it, and it said "Not available", not a misleading answer like "Service unavailable" |
| 14:05.28 | waKKu | maybe if u set qualify=yes it can help u |
| 14:05.31 | waKKu | in this peer |
| 14:05.35 | maz1977 | it's possible to configure asterisk to answer to a external sip call and route to an analog modem? |
| 14:05.47 | [TK]D-Fender | Shoeb: may depend on other circumstances. |
| 14:06.08 | Shoeb | [TK]D-Fender: Yesterday's event or today's event? |
| 14:06.10 | MrMister2 | [TK]D-Fender: Thank you. Since I did know of a way to do it I asked here since there might be a CLI command that I didn't know. Apparently that isn't the case. Again, thank you for the answer, that's what I was looking for, i.e. not possible to do it from CLI. |
| 14:06.13 | [TK]D-Fender | Shoeb: And indeed you have NOT set up the user correctly for NAT |
| 14:06.48 | [TK]D-Fender | MrMister2: I didn't say it wasn't, I said there is no STANDARD way. Hence you need to read its MANUAL. |
| 14:06.57 | *** topic/#asterisk by blitzrage -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- Flame suites required... |
| 14:07.02 | [TK]D-Fender | MrMister2: I know a way to reset a Polycom remotely from CLI jsut fine. |
| 14:07.23 | [TK]D-Fender | blitzrage: "suits" ;) |
| 14:07.25 | Optic | power cycle the port on the PoE switch |
| 14:07.25 | Optic | har har har |
| 14:07.33 | blitzrage | oops~! |
| 14:07.36 | [TK]D-Fender | Optic: Amongst others ;) |
| 14:07.41 | blitzrage | that's what I live in :) |
| 14:07.44 | *** topic/#asterisk by blitzrage -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- Flame suits required... |
| 14:07.58 | blitzrage | that's what happens when you type after being up for 7 mins |
| 14:07.58 | MrMister2 | [TK]D-Fender: really? just out of curiosity (yes, I know it doesn't apply in this case, just trying to learn) how would you do it with the Polycom? |
| 14:08.13 | Shoeb | [TK]D-Fender: whoa!! It was there!! lol, lemme check again |
| 14:08.49 | [TK]D-Fender | MrMister2: Go check the WIKI, its well listed, and there is a config file for it that comes with *'s samples. |
| 14:08.50 | DrukenLPY | AHHH!!!!!!!!!!!! |
| 14:09.35 | [TK]D-Fender | "Jeff's nuts roasting on an open fire........" |
| 14:09.37 | MrMister2 | [TK]D-Fender: Ah, k. That'll do. Again thanks for the answer. A explained negative is always good instead of just "can't be done" :) |
| 14:09.50 | Shoeb | [TK]D-Fender: Sorry, showed you the wrong sip.conf... |
| 14:10.13 | [TK]D-Fender | MrMister2: Because I never said it coultn'y :) I said this was a giant "DEPENDS". |
| 14:10.24 | [TK]D-Fender | Shoeb: Either way, the phone isn't up. |
| 14:10.41 | j-goddess | at this point the phone should go back to bed |
| 14:10.44 | j-goddess | it is too early |
| 14:10.46 | [TK]D-Fender | Shoeb: that may or may not have anything to do with your configs (it would right now) |
| 14:10.56 | Shoeb | Gotcha. |
| 14:11.04 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:12.14 | maz1977 | I have a easy question, but I find no answer . Could I use a analog modem as out call device? |
| 14:13.29 | Shoeb | [TK]D-Fender: Xlite error: "Call failed: Not found" and here's the CLI http://pastebin.ca/634554 |
| 14:13.42 | blitzrage | maz1977: no |
| 14:13.43 | HaMYaI | hi, anyone has a problem since 1.2.20 that agi script (using DeadAGI) doesn't ignore SIGHUP during the Dial() command? |
| 14:14.03 | blitzrage | HaMYaI: I'm surprised it did that at all since that was *just* added to turnk |
| 14:14.25 | blitzrage | and are you using DeadAGI() in the right scenario? |
| 14:14.30 | HaMYaI | I mean EXEC DIAL from agi script |
| 14:14.33 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 14:14.56 | [TK]D-Fender | <PROTECTED> |
| 14:14.58 | [TK]D-Fender | <PROTECTED> |
| 14:14.59 | *** join/#asterisk CuriosCat (i=stian@ninja.noc.host.net) |
| 14:14.59 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 14:15.07 | [TK]D-Fender | Shoeb: More dialplan screwups. |
| 14:15.12 | Shoeb | [TK]D-Fender: http://pastebin.ca/634555 :( |
| 14:15.16 | [TK]D-Fender | Shoeb: You should have masteered this one already |
| 14:15.19 | HaMYaI | blitzrage; I use DeadAGI to process the whole call, even after hangup |
| 14:15.21 | Shoeb | I read more about it last night!! I swear |
| 14:15.38 | blitzrage | HaMYaI: that should be right... but why are you doing another Dial() from there? |
| 14:15.46 | Shoeb | [TK]D-Fender: I made the sure the context outgoing is going to be well suited for that.. :S |
| 14:15.52 | HaMYaI | blitzrage; so, I will need to detect ANSWEREDTIME as well |
| 14:16.20 | [TK]D-Fender | Shoeb: Look at you context and realize that there isn't a way to process "4162202220" in there. This is BLATANLTY obvious |
| 14:16.21 | pigpen | So what is everyone using to give the suits call detail reports? AsteriskStat is nice..but well.... |
| 14:16.21 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.137) |
| 14:16.44 | maz1977 | blitzrage: asterisk can only do outgoing call in ip? |
| 14:16.50 | Shoeb | [TK]D-Fender: I know there isn't a way to do so in users context, but in outgoing context it should work though! :( |
| 14:17.07 | [TK]D-Fender | Shoeb: Your phones aren't USING it <--------------------- |
| 14:17.08 | ccesario | HaMYaI, try use AGI wit 'g' param in Dial command |
| 14:17.14 | blitzrage | maz1977: it can do it via hardware if you buy the right hardware. You can't just use any modem -- it doesn't work like that. |
| 14:17.24 | [TK]D-Fender | Shoeb: just becuase you named it [outgoing] doesn't mean ANYTHING. |
| 14:17.32 | MrMister2 | [TK]D-Fender: mmmm..... I'm probably doing something wrong on the phones... I now have a X-Lite softphone that shows Hold with a core show hints even after shutting down the softphone and running again. I can however make calls to itself and other extensions. weird. must be something messed up on my side.... |
| 14:17.36 | HaMYaI | blitzrage; I do just one Dial from within the agi script but since 1.2.20 the call will just hangup after EXEC DIAL is completed |
| 14:17.37 | Shoeb | [TK]D-Fender: If I can set context to users AND outgoing.. it should be possible, right? But how do I do it? |
| 14:17.40 | maz1977 | blitzrage: and with a isdn pci card? |
| 14:17.49 | [TK]D-Fender | Shoeb: you CANNOT. |
| 14:17.53 | blitzrage | maz1977: if it is supported by the drivers |
| 14:17.59 | HaMYaI | ccesario: I know but this used to work before |
| 14:18.13 | [TK]D-Fender | Shoeb: You CAN however make a 3rd context that INCLUDES the other 2 giving you access to the contents of BOTH. |
| 14:18.19 | Shoeb | [TK]D-Fender: Then should I move the outgoing dialplan to into users? |
| 14:18.27 | Shoeb | ah |
| 14:18.35 | [TK]D-Fender | Shoeb: you COULD, but I suggest the way above. |
| 14:18.45 | [TK]D-Fender | Shoeb: Go read chapter 5 all over AGAIN.... |
| 14:18.47 | HaMYaI | ccesario: the 'g' has heen for several months I guess |
| 14:18.50 | [TK]D-Fender | Shoeb: ... |
| 14:18.52 | [TK]D-Fender | ~osmosis |
| 14:18.53 | jbot | it has been said that osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 14:18.54 | Shoeb | [TK]D-Fender: Ok, trying. Chapter 5 from the book? Ok. |
| 14:18.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:19.05 | Shoeb | LOLOL |
| 14:19.11 | Shoeb | You're hilarious! |
| 14:19.16 | maz1977 | blitzrage: could I show you my scenario and you tell me if it's possible to realize? |
| 14:19.18 | blitzrage | no, jbot is hilarious |
| 14:19.23 | [TK]D-Fender | Shoeb: I'd like to think so :) |
| 14:19.27 | Shoeb | Well, someone fed it! |
| 14:19.31 | blitzrage | maz1977: you can show the channel... I gotta do some work |
| 14:19.31 | [TK]D-Fender | ;) |
| 14:19.42 | Rienzilla | hmm |
| 14:19.48 | [TK]D-Fender | blitzrage: Yes, that is MINE :) |
| 14:19.55 | blitzrage | I don't believe it |
| 14:19.57 | Shoeb | See! |
| 14:19.58 | Shoeb | haha |
| 14:19.58 | Rienzilla | are the people going to be annoyed if I spam my asterisk trouble? :-) |
| 14:20.01 | [TK]D-Fender | blitzrage: I keep jbot well fed indeed |
| 14:20.15 | Shoeb | [TK]D-Fender: So, if I give ext 100 the context outgoing, it _should_ work, right? |
| 14:20.18 | blitzrage | AAAAAAAND... I don't want to meet your mom, because I just want.... |
| 14:20.19 | [TK]D-Fender | Rienzilla: Yes, spam=bad. |
| 14:20.27 | HaMYaI | blitzrage: the EXEC DIAL with 'g' option seems to solve the problem but why? |
| 14:20.35 | [TK]D-Fender | Shoeb: it would then lose the ability to dial 200. |
| 14:20.45 | blitzrage | HaMYaI: no idea... I don't do an Dial()'s from my DeadAGI()... |
| 14:20.52 | [TK]D-Fender | blitzrage: ! ! ! |
| 14:20.53 | Shoeb | [TK]D-Fender: I understand, I'm just doing that for testing. But it should work, right? |
| 14:20.54 | blitzrage | and I don't remember what the 'g' option is :) |
| 14:21.00 | [TK]D-Fender | Shoeb: Yes |
| 14:21.22 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 14:21.25 | Shoeb | Rienzilla: Paste it on pastebin.ca |
| 14:21.26 | Optic | hello |
| 14:21.31 | Rienzilla | s/spam/ask :) How would I go about debugging bad audio quality? Some clients on my pbx have excellent quality, but one has a lot of jitter and can barely hear what we're aying |
| 14:22.28 | pigpen | [TK]D-Fender, looks like they are ganging up on you. |
| 14:22.37 | [TK]D-Fender | Rienzilla: DETAILS about the exact hardware being used and network topology would be helpful.... |
| 14:22.43 | HaMYaI | blitzrage; man, g - Proceed with dialplan execution at the current extension if the destination channel hangs up. |
| 14:22.50 | Rienzilla | ok |
| 14:24.02 | Rienzilla | my asterisk server is a server located in a datacentre a couple of hops away (100mbit fsx connection). We have several snom360's connected to the pbx via a VPN, and some softphones connecting from the big bad internet. |
| 14:24.07 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:24.21 | Rienzilla | all phones, and all-but-one softphone seem to work fine |
| 14:24.31 | coolbeans | Anyone using mysql static configs and asterisk 1.2.x? I can't get my voicemail passwords to update in the db ... |
| 14:24.49 | Rienzilla | de softpohones are on a dsl line (8 down 1 up) |
| 14:24.57 | Rienzilla | behind nat, most of the time |
| 14:25.00 | Shoeb | [TK]D-Fender: Not working :( .. doing a pastebin |
| 14:25.49 | [TK]D-Fender | Rienzilla: So you have 1 bad soft-phone? |
| 14:25.54 | Rienzilla | yes |
| 14:26.20 | maz1977 | I have a SIP telephone number . I need to filter the incoming call and then switch to my PBX or to cellular number |
| 14:26.22 | [TK]D-Fender | Rienzilla: Could be a shitty sound-card, headset, or internet connection. Take your pick. |
| 14:26.23 | Shoeb | [TK]D-Fender: http://pastebin.ca/634567 |
| 14:26.37 | Rienzilla | well, he can use other voip programs fine |
| 14:26.53 | Rienzilla | (non-sip) |
| 14:27.14 | Rienzilla | so I guess that would pretty much rule out soundcard and audio pheripherals |
| 14:28.24 | *** join/#asterisk jj56 (n=jflo@a80-127-56-82.adsl.xs4all.nl) |
| 14:29.51 | [TK]D-Fender | Shoeb: You have no entry [outbound] in your sip.conf |
| 14:30.07 | [TK]D-Fender | Rienzilla: Could jsut be jitter. |
| 14:30.27 | Shoeb | [TK]D-Fender: True, I have the carrier's information. |
| 14:30.43 | [TK]D-Fender | Shoeb: Whatever the hell that means :) |
| 14:31.05 | Shoeb | I mean I don't have outbound in there. Should I? |
| 14:31.19 | [TK]D-Fender | Shoeb: exten=>_NXXNXXXXXX,n,Dial(sip/1${EXTEN}@outbound) <- where there hell is [outbound] in your sip.conf? |
| 14:31.27 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:31.28 | [TK]D-Fender | Shoeb: It has no account to use! |
| 14:31.33 | Shoeb | AH! |
| 14:32.06 | Shoeb | So I go back to the dialplan and tell it what to use! |
| 14:32.07 | [TK]D-Fender | Shoeb: Stop thinking these names you are giving your sip devices and contexts mean ANYTHING. There are NO MAGIC NAMES. |
| 14:32.15 | [TK]D-Fender | Shoeb: ....... |
| 14:32.25 | [TK]D-Fender | Shoeb: its your job to tell it EVERYTHING. |
| 14:33.13 | DrukenLPY | [TK]D-Fender: you looked at asterisknow ? |
| 14:33.24 | [TK]D-Fender | DrukenLPY: From a safe distance, yes :) |
| 14:33.25 | Shoeb | Gotcha |
| 14:33.41 | DrukenLPY | know where incoming calls are handled ? |
| 14:34.04 | Shoeb | [TK]D-Fender: What can I say, you seriously rock. |
| 14:34.32 | [TK]D-Fender | DrukenLPY: No, I safely escaped it by that point :) |
| 14:34.38 | DrukenLPY | ahh |
| 14:34.55 | Shoeb | For some reason you remind me of jennifer aniston from Friends, with the stress on some words, hehe |
| 14:35.12 | pigpen | DrukenLPY, Same here. I had it loaded for 5 minutes before I started looking for my gentoo live cd. |
| 14:35.41 | Shoeb | pigpen: Especially for us beginners, *NOW isn't ready yet. Still in beta, let the experts handle it. |
| 14:36.54 | pigpen | IMHO, beginners should start with only asterisk. No web interface. If you start with a pretty gui, what will you do when something goes wrong. |
| 14:37.15 | Shoeb | Petty question, how do I "switch off" sip debug |
| 14:37.35 | [TK]D-Fender | Shoeb: "sip no debug" |
| 14:37.39 | pigpen | I can probably search my irc logs and find where [TK]D-Fender told me this in my "newbe" years. |
| 14:37.44 | Shoeb | Thanks! |
| 14:38.09 | Shoeb | pigpen: You're totally correct. That's what I'm doing. |
| 14:38.29 | [TK]D-Fender | pigpen: You'll WHINE about it obviously.. then blame *, anything except claim responsibility. |
| 14:38.43 | Shoeb | lol |
| 14:38.47 | [TK]D-Fender | pigpen: Its like the steps to dealing with someones death :) |
| 14:40.40 | [TK]D-Fender | Every time someone comes in here whining about why ASTERISK doesn't work, and they are using Trixbox, GOD KILLS A KITTEN |
| 14:41.24 | creativx | no |
| 14:41.26 | creativx | god kills a lolcat |
| 14:41.39 | creativx | and the reason asterisk doesnt work is because of the faulty LOLCODE module |
| 14:42.28 | [TK]D-Fender | IM IN UR DIE-LPAN R00TING UR CALLZ! |
| 14:42.42 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:42.49 | *** join/#asterisk flart (n=flart@atommuell.mum.jku.at) |
| 14:42.55 | flart | hi |
| 14:43.35 | pigpen | [TK]D-Fender, why is my system not working. it just stopped on it's own. I did nothing! :P |
| 14:44.05 | [TK]D-Fender | pigpen: aH... A SIN OF omission |
| 14:44.16 | [TK]D-Fender | darn caps inversion ;) |
| 14:44.54 | pigpen | Yeah, my typing gets better with the lack of coffee. |
| 14:45.07 | pigpen | Dude, do you ever sleep? |
| 14:45.13 | Shoeb | [TK]D-Fender: Have you heard of anyone using a cheap laptop to be a home asterisk server? I so want to do that. But can't do it with no digium cards. :( |
| 14:45.31 | flart | i configured asterisk to work with my isdn-card with chan_capi and everything seems to be alright. but when i call my ntba i can't reach asterisk. any hints? |
| 14:46.06 | Shoeb | flart: I'm a newb too, but it sounds like dialplan. |
| 14:46.09 | [TK]D-Fender | Shoeb: Sure you can. Who says PCI cards are the only infaces out there? |
| 14:46.25 | Shoeb | [TK]D-Fender: I tried looking for some out there, couldn't find any. :-S |
| 14:46.27 | [TK]D-Fender | Shoeb: lol... nice try :) |
| 14:46.29 | *** join/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 14:46.33 | Shoeb | hehe |
| 14:46.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:46.39 | [TK]D-Fender | Shoeb: Haven't tried hard enough. |
| 14:46.59 | [TK]D-Fender | Shoeb: what are you looking to interface to? |
| 14:47.13 | Shoeb | The same FXO cards stuff, basically my home landline given by Bell. |
| 14:47.16 | flart | Shoeb: thought so too, but shouldn't i see something on the asterisk console? |
| 14:47.43 | Shoeb | I wanna be able to get that on this old ibm satellite. |
| 14:47.58 | [TK]D-Fender | Shoeb: Linksys SPA-3102 <------------ |
| 14:48.07 | Shoeb | flart: Your verbosity level is probably too low, and sip debug needs to be enabled. |
| 14:48.17 | Shoeb | flart: I think, but disclaimer, I'm a newb too, lol |
| 14:48.24 | Shoeb | [TK]D-Fender: Lemme look |
| 14:48.26 | [TK]D-Fender | Shoeb: Ah.. the blind leading the blind... welcome to #asterisk |
| 14:48.30 | Shoeb | HASHAHAHAHAHAHA |
| 14:48.35 | Shoeb | hahahahahaha |
| 14:48.40 | Rienzilla | ? |
| 14:48.49 | coolbeans | Does anyone know if updating voicemail passwords works in a mysql static realtime config? |
| 14:48.50 | flart | ;) |
| 14:49.01 | pigpen | coolbeans, yes. |
| 14:49.04 | [TK]D-Fender | flart: PASTEBIN is your friend. |
| 14:49.06 | [TK]D-Fender | ~pb |
| 14:49.07 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:49.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 14:49.17 | coolbeans | pigpen: What's the secret? I can't get mine to update. |
| 14:49.24 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:49.57 | [TK]D-Fender | flart: show your configs, devices status info, CLI output at high verbose, etc.... |
| 14:50.23 | pigpen | coolbeans, personally I use Postgresql |
| 14:50.40 | coolbeans | Did you have to do anything special to get it to work? I'm running 1.2.18 ... |
| 14:50.41 | pigpen | but the mysql driver was way before the psql. |
| 14:51.05 | pigpen | It should be supported. |
| 14:51.18 | pigpen | are the permisions setup correctly on the db? |
| 14:52.02 | coolbeans | Yep, perms are fine. I'm about to turn on mysql logging and see. It's just an extremely high traffic set of servers and I was hoping to find some magic answer. |
| 14:52.05 | Shoeb | [TK]D-Fender: Oh, so basically, all I need this little interface in the middle and then I connect the laptop normal via ethernet and the phone line goes directly in there? So, the phone line goes directly in there. Where will I plug my phones in then? |
| 14:52.06 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:52.12 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:52.15 | pigpen | Also, there are many good docs for setting it up for mysql. |
| 14:52.26 | pigpen | There are pretty much no good docs to set it up for postgresql. |
| 14:52.28 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:52.40 | [TK]D-Fender | Shoeb: The SPA-3102 lets you take in 1 LINE, and 1 PHONE. |
| 14:53.04 | Shoeb | ah |
| 14:53.05 | [TK]D-Fender | Shoeb: If you want to use more analog phones, get additional SPA-2102's (2 ports each) |
| 14:53.21 | Shoeb | Gotcha. |
| 14:53.30 | pigpen | coolbeans, make sure your extconfig.conf, res_mysql.conf are setup right. |
| 14:53.33 | coolbeans | pigpen: Yea, it's setup per the docs. |
| 14:53.33 | Shoeb | I think I'm going to finish the book now! And then jump on this. |
| 14:53.43 | coolbeans | It works fine except for updating vmail passwords. |
| 14:53.49 | coolbeans | hrm... |
| 14:53.49 | flart | ok, pastebin'd my extensions.conf and capi.conf: http://paste.debian.net/33461 |
| 14:54.09 | Shoeb | Thanks [TK]D-Fender, I always kept thinking PCI cards are the way. And I thought I couldn't use my dusty old laptop that has usb and eth. |
| 14:54.16 | pigpen | coolbeans, well, turn on max debugging on mysql and try it. |
| 14:54.24 | pigpen | you should be getting a hint from them. |
| 14:54.32 | pigpen | debug the crap out of asterisk at the same time. |
| 14:54.43 | pigpen | your issue should show up. |
| 14:54.46 | [TK]D-Fender | Shoeb: PCI based FXS is ASS<----------- |
| 14:55.07 | Shoeb | May you live long. I was thinking the same thing, for some reason. |
| 14:55.14 | coolbeans | Yea, that's my dreaded next step. Thanks! |
| 14:55.29 | pigpen | coolbeans, debug away young lad. |
| 14:55.57 | coolbeans | (sigh) |
| 14:56.42 | [TK]D-Fender | flart: exten = 123,1,Dial(SIP/${EXTEN},5) <- I suspect that CAPI does not send its calls to "123", but rather the DID that was dialed.... |
| 14:56.48 | pigpen | Ah, it isn't that bad. Think of it like a Safari for geeks. |
| 14:57.13 | [TK]D-Fender | pigpen: Meaning I can shoot them? ;) |
| 14:57.28 | pigpen | [TK]D-Fender, yes. I have extra ammo if needed. |
| 14:57.34 | flart | [TK]D-Fender: it's the DID i wanted to use for testing |
| 14:57.36 | [TK]D-Fender | pigpen: Yee-haw |
| 14:57.37 | tzafrir | [TK]D-Fender, should I agree? |
| 14:57.52 | [TK]D-Fender | tzafrir: to what? |
| 14:57.58 | tzafrir | nm |
| 14:58.28 | Shoeb | [TK]D-Fender: So, in your experience, I have two RJ11 jacks in the house and they both have analog phones connected to them. And I have this laptop that's connected to the internet that I would like to see as an asterisk server as well. Which one is the BEST of those adapters that could work well? |
| 14:58.47 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 14:58.49 | *** join/#asterisk karrot-x (n=karrotx@dsl092-174-226.wdc2.dsl.speakeasy.net) |
| 14:59.07 | [TK]D-Fender | Shoeb: Do you want those jacks to act independant of each other? |
| 14:59.34 | Shoeb | [TK]D-Fender: Nope, because they both give me the same phone line. Just one phone number, so I don't think I'd need that. |
| 14:59.35 | [TK]D-Fender | Shoeb: Or just * enable you incoming line and treat all phones as the same phone (like it feels with bell) |
| 14:59.44 | Shoeb | Yup. |
| 14:59.58 | [TK]D-Fender | Shoeb: You can have each phone be independant so they can do their own thing. |
| 15:00.16 | Shoeb | Oh? You mean like have different phone numbers and stuff? |
| 15:00.27 | [TK]D-Fender | Shoeb: No, seperate devices. |
| 15:00.31 | Shoeb | ah |
| 15:00.58 | [TK]D-Fender | Shoeb: When calls come in you can choose which phones to ring, and one could be on a call, the other on a different call, checking VM, etc... |
| 15:01.05 | Shoeb | Nah, not necessary. Just that if we're home, and both the phones ring.. and one is picked up (like how it is now) it should work. That's all. |
| 15:01.23 | [TK]D-Fender | Shoeb: Then a single SPA-3102 will do the job. |
| 15:01.25 | Shoeb | Aaaah, that would be good. Except I'd want both the phones to ring. One is in the bedroom and the other in the living. |
| 15:01.43 | Shoeb | Yeah? Nice. |
| 15:02.12 | [TK]D-Fender | Shoeb: physically disconnect your home wiring from the demarc point, plug the Bell side that into the SPA's FXO port, and connect the rest of your place to the FXS port |
| 15:02.46 | [TK]D-Fender | Shoeb: then all of your phones will share that one part. |
| 15:02.49 | [TK]D-Fender | port* |
| 15:03.10 | coolbeans | pigpen: I'm not even getting a SQL insert generated in the log when I change my asterisk voicemail password... hrm... |
| 15:03.12 | Shoeb | Yeah, see. That's not possible because I live in a building, and it's one of the few bldgs in town that has some extremely different wiring with Bell, and ofcourse I couldn't get to the demarc. |
| 15:03.14 | [TK]D-Fender | Shoeb: And you don't need the concept of "internal extensions" like a normal PBX. |
| 15:03.16 | coolbeans | or update |
| 15:03.26 | *** join/#asterisk wunderkin (n=wunderki@ip68-2-62-143.ph.ph.cox.net) |
| 15:03.35 | [TK]D-Fender | Shoeb: Ok, grab a staple-gun and run your own wire :) |
| 15:03.40 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 15:03.41 | Shoeb | lol |
| 15:03.52 | [TK]D-Fender | Katty: Mew. |
| 15:04.11 | Shoeb | Could be a possibility. For now, I just want to take care of this one line that comes in, and both the phones. |
| 15:04.21 | pigpen | coolbeans, yeah...so your issues is either with asterisk or permissions to the db. |
| 15:04.24 | [TK]D-Fender | Shoeb: That'll do it. |
| 15:04.31 | Shoeb | Gotcha, thanks man. |
| 15:05.34 | coolbeans | Well, no insert in mysql.log, so it's not making it that far.. hrm.. |
| 15:06.03 | coolbeans | pigpen: You're using static in postgres? |
| 15:06.48 | *** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com) |
| 15:06.48 | flart | [TK]D-Fender: if 'capi show channels' shows mit 2 isdn channels, am i right that asterisk is aware of the isdn-card? |
| 15:06.48 | flart | s/mit// |
| 15:07.04 | flart | oh, nice bot ;) |
| 15:07.05 | [TK]D-Fender | flart: I would assume so having very little ISDN BRI experience. Try and enable channel debug. |
| 15:08.08 | *** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net) |
| 15:08.13 | flart | [TK]D-Fender: is enabled. i just don't get any message about a incomming call in the asterisk console (verbosity is on 5) |
| 15:08.39 | [TK]D-Fender | flart: Ok, I'll leave you in more capable hands then.... |
| 15:08.47 | flart | ;) |
| 15:09.07 | pigpen | coolbeans, depends. |
| 15:09.19 | flart | maybe i should get some isdn-phone and try if the ntba work correctly |
| 15:09.28 | flart | *works |
| 15:09.35 | pigpen | coolbeans, I have pretty much done it all in some sort using realtime and postgres. |
| 15:10.19 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 15:10.29 | flart | [TK]D-Fender: thx anyway |
| 15:10.32 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:11.41 | coolbeans | I wonder if this is a 1.2.18 bug... I guess I need to go through app_voicemai.c |
| 15:12.30 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:12.31 | pigpen | coolbeans, getting the first realtime to work is the hardest. |
| 15:12.40 | pigpen | once you have it, the others come much easier. |
| 15:13.09 | pigpen | Personally, I am only planning to use it for voicemail and dialplan objects. |
| 15:13.29 | coolbeans | I'm just using it for voicemail and sip |
| 15:13.32 | pigpen | Otherwise it bitches too much with postgres (at least it does with 1.4.2) |
| 15:14.17 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:14.28 | *** join/#asterisk NirS (n=Nir@87.68.232.33.adsl.012.net.il) |
| 15:14.46 | coolbeans | Ok, there's only one update in app_voicemail.c: ast_update_realtime("voicemail", "uniqueid", vmu->uniqueid, "password", password, NULL); |
| 15:14.58 | coolbeans | So it's in the ast_update_realtime driver and this uniqueid column that doesn't exist. |
| 15:16.19 | coolbeans | But here's the select statement when app_voicemail.so reloads: SELECT category, var_name, var_val, cat_metric FROM voicemail WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id |
| 15:16.22 | coolbeans | No mention of uniqueid. |
| 15:16.23 | coolbeans | hrm.... |
| 15:16.28 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 15:17.14 | coolbeans | Dare I change app_voicemail.c's column to just 'id'? |
| 15:17.59 | Corydon76-work | coolbeans: you probably should not, no. |
| 15:18.34 | coolbeans | Corydon76-home: Well, it ain't working now.. what's the harm? |
| 15:18.58 | *** join/#asterisk sasch (n=info@host117-234-static.4-79-b.business.telecomitalia.it) |
| 15:19.11 | Corydon76-work | Oh, you mean change the source? If your unique column is called id, then yes, you'll need to change the source. |
| 15:19.25 | Corydon76-work | However, we recommend that people use the database schema as written |
| 15:20.05 | Corydon76-work | See contrib/scripts/vmdb.sql |
| 15:20.43 | *** join/#asterisk NirS_ (n=Nir@87.68.171.192) |
| 15:21.10 | wunderkin | isn't id reserved in sql? |
| 15:21.20 | Corydon76-work | It is not reserved, no |
| 15:21.42 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 15:21.52 | Corydon76-work | It's a common practice, though |
| 15:22.37 | BSD_Tech | hmmm |
| 15:22.39 | coolbeans | Corydon76-work: Hey, I missed everything you just said, we had a brown-out and my workstation rebooted. If it's not too much trouble, could you /msg me what you said as it may be helpful. |
| 15:22.46 | BSD_Tech | where is that Rabbit |
| 15:22.58 | Corydon76-work | coolbeans: we recommend that people use the database schema as written |
| 15:23.05 | Corydon76-work | coolbeans: See contrib/scripts/vmdb.sql |
| 15:23.09 | [TK]D-Fender | load chan_omni_present.so |
| 15:23.26 | BSD_Tech | ? |
| 15:23.34 | BSD_Tech | never hear that module |
| 15:24.28 | coolbeans | Corydon76-work: I'm using static realtime. |
| 15:24.34 | BSD_Tech | I have the app_Corydon76_joke module |
| 15:24.46 | Corydon76-work | coolbeans: then it's irrelevant to you |
| 15:24.51 | BSD_Tech | but it fails to load |
| 15:24.58 | Corydon76-work | coolbeans: that column is only for dynamic realtime |
| 15:25.28 | coolbeans | Corydon76-work: Right, but my issue is getting voicemail passwords to update in static realtime with mysql, asterisk 1.2.18, the update statement isn't making it to mysql so I've been digging through app_voicemail.c and app_realtime.c |
| 15:25.42 | Corydon76-work | coolbeans: we don't support that |
| 15:25.47 | BSD_Tech | app_[TK]D-Fender.so also keeps crashing my system at load time, |
| 15:25.51 | coolbeans | Ahh, well then there lies the issue. |
| 15:25.53 | coolbeans | lol |
| 15:26.17 | coolbeans | Can I use a combination of realtime and static tables? |
| 15:26.27 | Corydon76-work | You can, yes |
| 15:26.28 | coolbeans | I need to keep my extensions.conf and its includes flat file. |
| 15:26.34 | coolbeans | Ok, let me go research that route. |
| 15:26.48 | coolbeans | Corydon76-work: Thanks. At least now I've confirmed that it's not intended to work. |
| 15:26.52 | Corydon76-work | dynamic realtime takes precedence over lines in the config file |
| 15:26.56 | HaMYaI | Corydon76-work: When do I need Dial() with 'g' option? |
| 15:27.06 | coolbeans | pigpen: You still here? |
| 15:27.14 | Corydon76-work | HaMYaI: why are you addressing me? |
| 15:27.42 | pigpen | coolbeans, one sec..on phone. |
| 15:27.53 | HaMYaI | Corydon76-work: okie, there's explanation |
| 15:27.54 | *** join/#asterisk javb (n=javb@190.80.235.113) |
| 15:28.12 | coolbeans | pigpen: just see Corydon76-work's statement, voicemail password updats don't work in 1.2.x with static realtime, essentially, never intended to. |
| 15:28.33 | coolbeans | Is 1.4.x ready for high volume production? |
| 15:28.36 | HaMYaI | Corydon76-work: frankly speaking, I have been following the bug tracker #10245 and saw that you were involved with that |
| 15:28.48 | javb | when a call is comming from a zap channel the voicemail plays in "en" (i have it in es), but when the call is comming from inside, it plays ok.. (everything is set to es), any ideas? |
| 15:28.56 | Corydon76-work | coolbeans: consider that Digium has moved their production PBX to 1.4 |
| 15:29.03 | HaMYaI | Corydon76-work: about removing DeadAGI() from further release |
| 15:29.07 | Corydon76-work | coolbeans: is that good enough recommendation? |
| 15:29.30 | Corydon76-work | HaMYaI: yes, but that's only in trunk |
| 15:29.39 | coolbeans | Corydon76-work: thanks. |
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| 15:29.59 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:30.00 | [TK]D-Fender | coolbeans: translation : Only if you have a staff of over 20 programmers handy to fix any problems that arise ;) |
| 15:30.34 | pigpen | yeah...I have only done it on 1.4 |
| 15:30.39 | coolbeans | [TK]D-Fender: yea, that's what I read it to mean. We're processing like 10k calls per day most days. |
| 15:30.54 | coolbeans | pigpen: So it's likely just not in 1.2.x |
| 15:30.59 | *** join/#asterisk skyphyr (n=alanj@host81-151-250-11.range81-151.btcentralplus.com) |
| 15:31.08 | Corydon76-work | Also realize that 1.2 will stop getting bugfixes and get security-only fixes as of NEXT Wednesday |
| 15:31.10 | pigpen | No..it is there...but not as "polished" |
| 15:31.24 | [TK]D-Fender | javb: PASTEBIN everything so we can see |
| 15:31.28 | HaMYaI | Corydon76-work: yeah, I just knew that today after talking to Juggie but I still have problem with my DeadAGI() not working after 1.2.20+ |
| 15:31.39 | coolbeans | Corydon76-work: This 1.2.18 has worked flawlessly for us with this one exception ... |
| 15:31.45 | skyphyr | hi all - can anyone recommend a DSL connection in soho london worth having. We've got just standard BT business here and it's all over the place and not reliable enough for even a single IAX channel |
| 15:32.30 | HaMYaI | Corydon76-work: EXEC DIAL from agi script seems to behave differently since then |
| 15:32.53 | Corydon76-work | HaMYaI: correct. EXEC now returns the same as the underlying app |
| 15:33.11 | javb | PASTEBIN ---> |
| 15:33.21 | javb | ? |
| 15:33.34 | HaMYaI | Corydon76-work: so, it's been changed since 1.2.20? |
| 15:33.37 | Shoeb | [TK]D-Fender: Can I msg you please? |
| 15:34.03 | javb | [TK]D-Fender : PASTEBIN --> Pase bin ? |
| 15:34.15 | Corydon76-work | HaMYaI: are you using DeadAGI for a live channel? |
| 15:34.48 | HaMYaI | Corydon76-work: after EXEC DIAL (wihtout 'g') completed it just doesn't want to continue |
| 15:34.51 | [TK]D-Fender | Shoeb: for now... |
| 15:34.56 | [TK]D-Fender | ~pb |
| 15:34.57 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:35.01 | [TK]D-Fender | javb: ^^^^^^^^^^^^^^^^^^^^^^ |
| 15:35.04 | HaMYaI | Corydon76-work: yes, for the live channel |
| 15:35.25 | Corydon76-work | HaMYaI: do NOT use DeadAGI for a live channel. Things don't work correctly when you do that |
| 15:35.32 | HaMYaI | Corydon76-work: with ignoring SIGHUP |
| 15:36.03 | Corydon76-work | Correct. You ignore the SIGHUP if you want to do things beyond the hangup (like cleanup connections) |
| 15:36.20 | HaMYaI | Corydon76-work: but it works fine for 1.2.X before 1.2.20, that's what I'm wondering |
| 15:36.32 | Corydon76-work | HaMYaI: right, because we fixed a bug |
| 15:37.10 | HaMYaI | Corydon76-work: ohh, mine script used to run correctly with a buggie version then -( |
| 15:37.27 | HaMYaI | s/mine/my |
| 15:38.24 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-1ffadf826bbd337e) |
| 15:40.16 | Dan0maN_Work | all, i'm looking to set up a test box to evaluate * for my company. can anyone suggest a variety of phones for me to order for the test? i am completely new to voip, so any help with features, recommended supported protocols, etc would be a great help |
| 15:41.28 | coolbeans | Dan0maN_Work: On phones, Polycom 5xx and better seem to work the best for most people. |
| 15:41.42 | [TK]D-Fender | Dan0maN_Work: Polycom IP 320 |
| 15:41.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 15:42.07 | coolbeans | [TK]D-Fender: Did they add a speakerphone to the 320 vs 301? |
| 15:42.30 | Katty | [TK]D-Fender: herro. |
| 15:42.38 | Corydon76-work | Yes, there's a speakephone in the 320 and above |
| 15:42.55 | Dan0maN_Work | thanks. and thanks to others who respond in advance |
| 15:43.08 | Corydon76-work | Katty: no, only the 601 |
| 15:43.15 | Katty | :< |
| 15:43.20 | [TK]D-Fender | coolbeans: Speakerphone, PoE, etc. |
| 15:43.25 | coolbeans | [TK]D-Fender: cool |
| 15:43.33 | [TK]D-Fender | coolbeans: at $87.50 USD its a complete category-killer |
| 15:43.35 | Corydon76-work | Katty: it requires the minibrowser functionality |
| 15:43.43 | Katty | whyfor? |
| 15:43.48 | Katty | it displays a webpage in the background? |
| 15:43.58 | Corydon76-work | Katty: yes, actually |
| 15:44.01 | Katty | hot |
| 15:44.11 | [TK]D-Fender | Katty: Yes it supports custom logo's AND supports the micro-browser. these 2 aspects are INDEPENDANT |
| 15:44.12 | Katty | animated? :> |
| 15:44.13 | Corydon76-work | Grayscale webpage, but yes |
| 15:44.14 | Katty | color? :> |
| 15:44.21 | Katty | :< |
| 15:44.23 | [TK]D-Fender | Katty: B&W. |
| 15:44.24 | Katty | gifs? |
| 15:44.34 | Strom_M | i didn't know pendants had anything to do with being independent |
| 15:44.39 | Corydon76-work | IIRC, it permits 16 shades of gray |
| 15:44.46 | Katty | what do people use the microbrowser for? |
| 15:44.51 | Katty | google searches? |
| 15:44.55 | [TK]D-Fender | Katty: Nope. |
| 15:45.01 | [TK]D-Fender | Katty: internal custom stuff. |
| 15:45.05 | Katty | ooooh |
| 15:45.11 | [TK]D-Fender | Katty: its XHTM and a very limited set |
| 15:45.33 | [TK]D-Fender | Katty: I use mine for live queue-stats, company directory, etc. |
| 15:45.37 | Katty | support info for our clients. |
| 15:46.36 | Katty | that's still pretty neat (= |
| 15:46.41 | [TK]D-Fender | Katty: on idle you can set an MB refresh rate if you want to push something on-screen. |
| 15:47.07 | Katty | scrolling marquee? |
| 15:47.31 | [TK]D-Fender | Katty: No. |
| 15:47.34 | Katty | :< |
| 15:47.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 15:47.40 | Katty | i just want it all, don't i. |
| 15:47.48 | SwK | the 550 has the microbrowser and a backlight too |
| 15:47.49 | Katty | never happy. |
| 15:47.51 | [TK]D-Fender | Katty: With a cherry on top ;) |
| 15:48.00 | Katty | what a typical girl. shame on me. |
| 15:48.03 | [TK]D-Fender | SwK: IP 550 = way overpriced. |
| 15:48.56 | [TK]D-Fender | SwK: it does not make it onto my "suggest" list anywhere. |
| 15:49.16 | Katty | 601s support side bar thingies |
| 15:49.18 | Katty | that's pretty neat. |
| 15:49.30 | BSD_Tech | 601 and 650 |
| 15:49.38 | BSD_Tech | support the side car |
| 15:49.43 | [TK]D-Fender | Katty: indeed, but the IP 650 is displacing it as the model to support that. |
| 15:49.51 | Katty | :< |
| 15:49.57 | Katty | but 650s don't support the logo? |
| 15:50.07 | Katty | <PROTECTED> |
| 15:50.13 | BSD_Tech | it uses the xml directory file to propgate it |
| 15:50.13 | SwK | [TK]D-Fender, i like the 550... bought one for home... its cheaper then the 650 and all the 650 really gives you as far as I can tell is sidecar capability and 2 extra appearance buttons |
| 15:50.14 | [TK]D-Fender | Katty: the 650 is acceptably more expensive and offers a few more features and longer support cycle. |
| 15:50.32 | [TK]D-Fender | Katty: Every phone excep the IP30X's support logo. |
| 15:50.38 | Katty | oh. |
| 15:50.38 | BSD_Tech | and g722 digital codec |
| 15:50.54 | SwK | [TK]D-Fender, the 4X0's have the microbrowser too? |
| 15:51.06 | [TK]D-Fender | SwK: +USB expansion. And the price difference really increases its resale value. |
| 15:51.13 | SwK | yeah |
| 15:51.13 | BSD_Tech | in your tftpboot dir make a dir poilycom |
| 15:51.19 | [TK]D-Fender | SwK: I refuse to spend the price of a 601 on a dead-end |
| 15:51.29 | [TK]D-Fender | SwK: Yes |
| 15:51.36 | SwK | [TK]D-Fender, is there any real info what the usb expansion can do? |
| 15:51.49 | BSD_Tech | it does not yet work |
| 15:52.06 | [TK]D-Fender | SwK: Only the 500, 30X fails to support the MB |
| 15:52.11 | BSD_Tech | they are still testing firmware to activate it |
| 15:52.37 | BSD_Tech | it will have nany functions to it |
| 15:52.39 | SwK | yeah... it could end up like the IR port on the sidekick... hardware was there but it was never put into a released firmware |
| 15:53.08 | BSD_Tech | not its in the works |
| 15:53.10 | [TK]D-Fender | SwK: Still I wouldn't pay for a 500 being a dead-end. |
| 15:53.27 | BSD_Tech | they say late 4th quarter they should have it working |
| 15:53.45 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 15:53.58 | penguinFunk | hey guys, anyone familiar with isdn30? |
| 15:54.19 | penguinFunk | i have noticed that in order to make calls you have to leave out the preceeding 0 |
| 15:54.51 | penguinFunk | so to dial 01554 723 345 you need to dial 1554 723 345 |
| 15:54.59 | penguinFunk | because of the way isdn30 works |
| 15:55.04 | penguinFunk | but what about international calls? |
| 15:56.57 | [TK]D-Fender | BSD_Tech: Have you tried it yet? |
| 15:57.04 | penguinFunk | i have tried leaving out one 0, both 0's leaving the number completely intact |
| 15:57.06 | penguinFunk | nothing |
| 15:57.32 | penguinFunk | cant get any sense out of BT either |
| 15:57.41 | BSD_Tech | TK ? |
| 15:57.55 | BSD_Tech | Tried what ? |
| 15:58.08 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:58.13 | penguinFunk | anyone here got isdn30 in the UK with international calls working? |
| 15:58.25 | BSD_Tech | looking for map |
| 15:58.56 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 16:01.04 | [TK]D-Fender | BSD_Tech: MB on a 430 |
| 16:01.21 | BSD_Tech | I dont have a 430 |
| 16:01.24 | *** part/#asterisk Uzzi (n=andrea@host192-169-dynamic.60-82-r.retail.telecomitalia.it) |
| 16:01.34 | BSD_Tech | I have a 501 601 and 650 |
| 16:01.50 | BSD_Tech | 430 will be next paycheck |
| 16:02.37 | [TK]D-Fender | BSD_Tech: all personal purchases on your bill? |
| 16:03.09 | Dan0maN_Work | so far, just polycom phones. any other recommendations to try out? i've been given auth to purchase up to 5 phones, but i doubt they want to exceed $100USD per |
| 16:03.27 | [TK]D-Fender | Dan0maN_Work: Nope, Polycom > All |
| 16:04.08 | [TK]D-Fender | Dan0maN_Work: $87.50 @ http://www.telephonydepot.com/product_p/105-058-320.htm |
| 16:05.23 | [TK]D-Fender | Dan0maN_Work: And get yourself an inexpensive PoE switch to power them. |
| 16:06.17 | Dan0maN_Work | ok. i've always appreciated polycom anyway. just making sure i cover the bases ;) |
| 16:06.38 | Innatech | Dan0maN_Work: I have 430s and 501s. They're very nice. If you absolutely must go cheap Snom has improved the 300. The newer firmwares work reliably, and they have a decent look and feel. (Athough for the extra ~$40, I'd much rather have a Poly 430. ) |
| 16:07.42 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 16:07.58 | [TK]D-Fender | Innatech: IP320 costs less than the Snom and is of better quality. As I've claseed it, the IP 320 is a complete category-killer |
| 16:08.48 | Strom_M | Polycom: Kills Phones Dead. |
| 16:09.01 | Strom_M | <3 the 320 |
| 16:09.20 | Innatech | I haven't seen the in person 320 yet. It does look nice. I tend to think that the 430 is the killer entry level config. I might even like it better than the 501, for the less clueful users. |
| 16:09.49 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.1.159.revip2.asianet.co.th) |
| 16:09.57 | Strom_M | Innatech: for what it's worth, I love my 430, but I'm playing with the 320 this week and OMG I WANT ONE |
| 16:10.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 16:10.07 | Innatech | eent-ter-est-ing. |
| 16:10.11 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:10.20 | Innatech | ./mouthpinkie |
| 16:10.27 | BSD_Tech | I like the new look but kind reminds me of merlin phones |
| 16:10.45 | Strom_M | BSD_Tech: heh, i hadnt considered that |
| 16:11.11 | Innatech | I just ripped a merlin switch off a wall yesterday. I think I heard the mutterings of Cthulu when I touched the main module. |
| 16:11.36 | Strom_M | i gave a merlin 410 to a friend a while back |
| 16:12.03 | Innatech | couldn't have been a very good friend. ;P |
| 16:13.08 | BSD_Tech | I have all the merlin ringtones |
| 16:13.35 | BSD_Tech | for polycom |
| 16:13.52 | Strom_M | Innatech: no, he and i give each other old phone crap all the time |
| 16:13.52 | BSD_Tech | if you ever want to torture a client |
| 16:13.57 | Strom_M | BSD_Tech: hahahahaha |
| 16:14.03 | Strom_M | i'll take a copy please :D |
| 16:14.30 | Innatech | Strom_M: ah. Yeah, old stuff can be fun. |
| 16:14.52 | Strom_M | i've got a garage full of WECo 1A2 |
| 16:15.22 | Innatech | doesn't that have mechanical components? |
| 16:16.19 | pigpen | Strom_M, do you know if the realtime postgres driver supports more than 1 database? |
| 16:17.08 | pigpen | as you know, the postgres driver is very undocumented. |
| 16:17.26 | Strom_M | Innatech: relays as far as the eye can see |
| 16:17.32 | Strom_M | pigpen: i know nothing about that driver |
| 16:17.52 | pigpen | k. thanks. Few do. |
| 16:18.05 | Innatech | Strom_M: crazy. But I bet it's cool to play with..... |
| 16:19.45 | BSD_Tech | I have 5 nortel 3 merlin 3 mitel and 1 nec systems in boxes in the basement |
| 16:19.47 | [TK]D-Fender | Strom_M: The IP 430 is nice in a way as well. Dual-powered out of the box, extra soft-key, more hard-buttons.... |
| 16:20.07 | [TK]D-Fender | Strom_M: and the switch |
| 16:20.24 | pigpen | yeah...they are nice. I have one in my kitchen. |
| 16:20.26 | [TK]D-Fender | Strom_M: But for new corporate PoE deployments the saving for an IP 320 is BIG |
| 16:20.33 | Innatech | I really like the 430, and it gets good reactions. I just looked at the 320, though, and it does have nice specs. It definitely looks like its squarely aimed at eating Snom's lunch. |
| 16:20.53 | [TK]D-Fender | Innatech: Snom = complete waste in north america. |
| 16:21.10 | [TK]D-Fender | And the IP320/330 supports DUAL headset connection styles |
| 16:21.19 | Innatech | yeah. They don't really bother me, but that 320 does look way better. |
| 16:21.36 | pigpen | Hmm..I think I have a snom in my tool box in my truck. |
| 16:21.50 | BSD_Tech | Storm Enjoy torturing people |
| 16:22.24 | BSD_Tech | oldschool vs newschool |
| 16:22.25 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:23.44 | MrMister2 | I've just been asked what hardware would be best to connect 3 ISDN lines to a Asterisk server. Anyone can recommend (only used TDM400 so far) good hardware? I understand it might not necessarily be cheap :) |
| 16:23.50 | coppice | dual headset? for people with two heads? :-\ |
| 16:23.52 | BSD_Tech | I am going to gold leaf my polycom |
| 16:24.04 | BSD_Tech | headset and handset |
| 16:24.54 | pj_ | f0rk the pic |
| 16:25.25 | MrMister2 | 3 ISDN lines = 3 NTBA, not 1 NTBA with 3 phone numbers :) |
| 16:25.27 | BSD_Tech | [TK]D-Fender, still looking to bridge conf |
| 16:26.38 | [TK]D-Fender | BSD_Tech: channel : Local/otherpbx@contextwiththisextentodialtheotherside |
| 16:27.15 | [TK]D-Fender | BSD_Tech: exten : conference |
| 16:27.27 | [TK]D-Fender | BSD_Tech:context : contextwithoutourmettmeinit |
| 16:27.52 | [TK]D-Fender | BSD_Tech: call file calls up the remote PBX and joins the meetme. upon being answered bridges in its own. |
| 16:29.57 | MrMister2 | Iīve seen recommendations of Digium's B410P and Eicon 4BRI. Anyone who installed either of them have any advice? I'm more interested in Asterisk support and no problems first and card price second. |
| 16:30.50 | [T]ank | i am getting errors when receiving calls from my sip provider... he says also that I am sending my local ip address out with the sip headers... here is my info any help would be appreciated: 206.71.78.173 |
| 16:31.13 | BSD_Tech | try setting canreinvite = no |
| 16:31.45 | BSD_Tech | on the trunk |
| 16:32.01 | blitzrage | [T]ank: sounds like you need to setup externip and localnet |
| 16:32.11 | BSD_Tech | yeah that also |
| 16:32.56 | blitzrage | MrMister2: you also get support from Digium when you buy hardware from them (if that makes any difference :)) |
| 16:34.18 | MrMister2 | blitzrage: only purchased TDM400 with analog modules so far from a German supplier since it was the cheapest and have had no problems with them |
| 16:34.56 | [TK]D-Fender | [T]ank: .... |
| 16:34.58 | [TK]D-Fender | ~sipnat |
| 16:34.59 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:35.00 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 16:35.04 | *** part/#asterisk Gled|work (n=gled@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
| 16:35.11 | blitzrage | MrMister2: good news! :) |
| 16:35.13 | [T]ank | thank you... will look |
| 16:35.30 | MrMister2 | What I wanted was for someone who had experience installing ISDN cards to recommend a card that wouldn't give _TOO MANY_ problems :) |
| 16:35.40 | MrMister2 | price is a second consideration in this case |
| 16:36.41 | BSD_Tech | the digium card should just work |
| 16:37.16 | BSD_Tech | but I am a sangoma man myself |
| 16:37.27 | BSD_Tech | Sangoma has great bsd support |
| 16:37.50 | mvanbaak | yeah |
| 16:38.00 | mvanbaak | unless you have this ADSL nic v2 |
| 16:38.28 | BSD_Tech | not played with the adsl cards |
| 16:39.23 | mvanbaak | I have an S518 |
| 16:39.28 | MrMister2 | BSD_Tech: mmmm.... So in your experience you say that either digium or sangoma should be just a plug in experience? |
| 16:39.59 | BSD_Tech | sangome you just install thier driver and run the setup and it does all the work |
| 16:40.11 | BSD_Tech | and digium cards should work right out of the box |
| 16:40.15 | MrMister2 | open the server, plug it in, do a modprobe and install driver and go? |
| 16:40.36 | mvanbaak | open server, plug it in, install driver, do a modprobe |
| 16:40.53 | mvanbaak | sangoma driver will patch and recompile your zaptel driver |
| 16:40.54 | *** join/#asterisk JoelSolanki (i=Joel@220.224.77.102) |
| 16:41.04 | [T]ank | ok... the issue shows up on the providers side with ethereal... I am sending my local ip instead of my external ip when i get a call: Contact: <sip:866459XXXX@10.10.5.7> |
| 16:41.11 | *** join/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com) |
| 16:41.27 | [T]ank | how do I make that show my external ip instead... I have externip set. |
| 16:41.35 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 16:41.42 | mvanbaak | [T]ank: did you set the localnet as well ? |
| 16:41.49 | [T]ank | yeah |
| 16:42.10 | blitzrage | and localnet is matching your local subnet? |
| 16:42.16 | [T]ank | yeah |
| 16:42.27 | blitzrage | because Asterisk will substitute any local IP it sees that matches the localnet with the externip |
| 16:42.34 | blitzrage | if it's not, you have something setup wrong, or you didn't reload chan_sip.so |
| 16:42.47 | MrMister2 | mvanbaak and BSD_Tech: Thank you for your advice and opinion. I'll most likelly go with Digium because of support and ease of installation. |
| 16:43.01 | blitzrage | although I think Contact doesn't change |
| 16:43.25 | blitzrage | if I remember correctly |
| 16:43.28 | blitzrage | (which I may not :)) |
| 16:44.01 | mvanbaak | I think you are right blitzrage |
| 16:44.41 | blitzrage | mvanbaak: very cool :) |
| 16:45.05 | zpertee | I have a telephone switch box that my main lines and all of my extensions plug into. My questions is if I first plug my main lines into an asterisk box and then from there plug them into the switch will I then have my whole system converted to asterisk? |
| 16:45.16 | Shoeb | Hello. I have one carrier, but 3-4 different SIP softphones. Extension 10, 20, 30 and 40. Also have 4 DIDs, I know how to route inbound calls on each DID to different extensions. BUT, using the same outbound trunk, how can I make each extension to show the a different caller id? |
| 16:45.31 | Shoeb | As if it were 4 different phones with 4 different phone numbers. |
| 16:46.16 | mvanbaak | Shoeb: you use the CALLERID(num) dialplan function for that |
| 16:46.29 | mvanbaak | together with some sourcematching |
| 16:46.31 | mvanbaak | like this: |
| 16:47.08 | mvanbaak | exten => _90XXXX/10,1,Set(CALLERID(num)=${DID10}) |
| 16:47.31 | mvanbaak | exten => _90XXXX/20,1,Set(CALLERID(num)=${DID20}) |
| 16:47.46 | mvanbaak | whe _90XXXX is your outbound rules extension match |
| 16:47.56 | Shoeb | ok... |
| 16:48.32 | blitzrage | also of note, if you're using the 'n' priority and not setting that at the start of the priority level you can use the 's' priority for 'same' |
| 16:48.41 | blitzrage | exten => _90XXX,1,NoOp() |
| 16:48.53 | blitzrage | exten => _90XXX,n,NoOp() |
| 16:49.12 | blitzrage | exten => _90XXX/10,s,Set(CALLERID(num)=foo) |
| 16:49.22 | blitzrage | exten => _90XXX/20,s,... |
| 16:49.23 | blitzrage | etc... |
| 16:49.26 | Strom_M | blitzrage: ooh, i did not know that |
| 16:49.28 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 16:49.34 | blitzrage | Strom_M: I just found that out very recently :) |
| 16:51.34 | mvanbaak | OEH ! |
| 16:51.37 | mvanbaak | nice one blitzrage |
| 16:52.13 | *** join/#asterisk `paul (n=aldee@124.107.13.212) |
| 16:52.37 | tzanger | interesting |
| 16:52.51 | tzanger | I tend to not trust that though and use GotoIf($[${CALLERID(num)... |
| 16:53.08 | Strom_M | http://www.flickr.com/photos/stromcarlson/906164579/ <--- polycom ip320 + f/1.4 50mm lens + nikon d70 == teh droolphoto |
| 16:53.35 | tzanger | you shouldn't |
| 16:53.46 | tzanger | IM IN UR FRIDGE EATIN UR FOODZ |
| 16:53.48 | pj_ | (don't trust him) |
| 16:53.53 | pj_ | (er do) |
| 16:53.54 | `paul | is there a tool (web interface) to monitor the number of calls (answered or not answered)? |
| 16:53.59 | blitzrage | hrmmm... I should post a picture of all the phones on my desk :) |
| 16:54.02 | tzanger | Strom_M: nice |
| 16:54.15 | tzanger | f/1.4 is a narrow depth of field? |
| 16:54.17 | Strom_M | tzanger: thanks :D |
| 16:54.19 | Strom_M | oh yes |
| 16:54.22 | Strom_M | uber-narrow |
| 16:54.31 | Strom_M | only like one row of touchtone buttons is actually in focus |
| 16:54.33 | tzanger | considering the bottom buttons aren't in focus, I'd say so |
| 16:54.45 | tzanger | maybe a little too narrow for that shot :-) |
| 16:54.54 | Strom_M | heh, i wanted it like that |
| 16:55.37 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 16:56.18 | Shoeb | Thanks mvanbaak |
| 16:57.30 | mvanbaak | Shoeb: ur welcome |
| 16:58.04 | sweeper | `paul: look into cdr web frontends, there are a few |
| 16:59.17 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:00.18 | *** join/#asterisk Strom_C (n=strom@h72-2-22-215.bigpipeinc.com) |
| 17:03.16 | [TK]D-Fender | Strom_M: I'm looking at a Nikon D40 w/ AF-S 18-55mm Lens right now...... |
| 17:03.55 | mcab | Strom_M: Nice pic! Well done |
| 17:04.20 | [TK]D-Fender | Strom_M: thinking I may want another kit that includes a 55-200 lens as well... |
| 17:04.24 | tzanger | I want to get a good lens for my camera (powershot something, can't remember now, heh) |
| 17:04.35 | tzanger | I can adapt it for standard lenses but really I think I just need a better camera |
| 17:04.36 | blitzrage | booo.... I can't import photo's now that I'm running FC7 |
| 17:04.38 | [TK]D-Fender | Strom_M: But debating my learning curve & expected usage. |
| 17:05.12 | blitzrage | [TK]D-Fender: start telling girls you're a photographer :) |
| 17:05.16 | hypa7ia | someone should edit ~cisco to include the great firewall of china :) |
| 17:05.58 | mcab | [TK]D-Fender: taking pictures is easy, taking *good* ones, well... :-) |
| 17:06.31 | hypa7ia | oops |
| 17:06.38 | [TK]D-Fender | mcab: Yeah.... I know. my other option right now is the Panasonic DMC-FZ8K |
| 17:06.54 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 17:07.00 | blitzrage | hypa7ia: y0 y0! |
| 17:07.05 | [TK]D-Fender | mcab: Got good reviews & is point & shoot with 12x zoom for half the price. |
| 17:07.27 | mcab | [TK]D-Fender: I think my in-laws have that one. It's a great camera |
| 17:08.23 | mcab | [TK]D-Fender: but, I'm still an SLR guy at heart :-D |
| 17:08.27 | [TK]D-Fender | mcab: thing is that I am losing the "economy reflex", and I keep thinking longer term./... |
| 17:09.21 | [TK]D-Fender | mcab: And I am trying to grow out of my "Buyer's Remorse" complex. |
| 17:09.30 | mcab | heh |
| 17:09.55 | mcab | buying my D70 was utterly insane for a number of reasons, but I haven't regretted it |
| 17:10.21 | mcab | I only regret that I'm not a better photographer, but that means I need more practice |
| 17:16.34 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 17:18.16 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:19.16 | *** join/#asterisk sysreq (n=sysreq@modemcable213.72-59-74.mc.videotron.ca) |
| 17:19.24 | [T]ank | doing a sip debug i am getting "From: "asterisk" <sip:asterisk@10.10.5.7>;tag=as796a0c23" I need it to send my external ip... everything i try will not change it. |
| 17:20.45 | *** join/#asterisk CyberPony (n=CyberPon@66-194-25-11.static.twtelecom.net) |
| 17:20.56 | coppice | [TK]D-Fender: get some buyer's remorse therapy. they have it on E-Bay |
| 17:21.25 | [TK]D-Fender | coppice: "Satisfaction Garanteed!" |
| 17:21.46 | [TK]D-Fender | [T]ank: PASTEBIN is your friend..... |
| 17:21.48 | [TK]D-Fender | ~pb |
| 17:21.49 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:23.25 | [T]ank | <PROTECTED> |
| 17:24.08 | [T]ank | http://pastebin.ca/634767 |
| 17:26.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:28.18 | Katty | mmm, ice cream |
| 17:28.53 | Katty | thats yummy too! |
| 17:29.12 | Katty | they have these little 100 calorie mini bars at teh store |
| 17:29.27 | blitzrage | so I only need to eat 10 of them :)_ |
| 17:29.31 | Katty | :P |
| 17:29.40 | cpm | get all your days nutrition in one sitting! |
| 17:29.41 | Katty | they're sugar free (= |
| 17:29.50 | Katty | which is good. |
| 17:29.57 | Nugget | http://www.defectiveyeti.com/archives/002177.html <-- ha ha |
| 17:29.59 | [TK]D-Fender | [T]ank: a lot of that belongs under [general] , not your ENTRY |
| 17:30.05 | blitzrage | "It's a good thing" (tm) |
| 17:30.08 | Katty | hi Nugget! |
| 17:30.15 | [TK]D-Fender | [T]ank: read the guide again |
| 17:30.16 | Nugget | :D |
| 17:30.26 | Katty | i put myself on a diet. |
| 17:30.45 | Katty | a seefood diet. |
| 17:30.48 | Katty | if i see it, i eat it :P |
| 17:31.01 | blitzrage | I'm on the same one! :) |
| 17:31.03 | karleeto | someone please HELP! i've got a job that i'm supposed to be starting on that involves 3 different physical locations, with a private VPN in between each location so that they can all talk to each other, one location is 192.186.1.. I've done jobs with 2 locations and an IAX trunk in between, but i'm having trouble visualizing how I would go about doing this, is there anyone who would be willing to chat with me a minute and give me some ideas? |
| 17:31.12 | Katty | it must be a major fad. |
| 17:31.18 | Katty | actually, i kicked myself down to 900 a day |
| 17:31.26 | karleeto | err 192.168.1.*, 2nd is 192.168.2., and the 3rd is 192.168.3. |
| 17:31.27 | Katty | instead of like... 1400 |
| 17:31.41 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:32.02 | karleeto | i really just dont know what approach i should take in order to have the 3 locations asterisk boxes to all be able to dial each others phones |
| 17:32.05 | Katty | i'm going to give myself a birthday present of loosing 20lbs :> |
| 17:32.38 | karleeto | could anyone give me an idea of how i'm supposed to set this up? |
| 17:33.14 | karleeto | should i use trunks or should there be a 4th box that is the master box that routes calls to the respective machine, or what? |
| 17:33.38 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:33.41 | Qwell[] | karleeto: dundi |
| 17:33.48 | Katty | so... can you play different music on hold stuffs... based on context |
| 17:33.54 | karleeto | i've done lots of asterisk systems with 20 or more phones, etc, but never one where there are multiple locations |
| 17:33.56 | Katty | like zaptrg, zapebc, zapthisotherbusiness |
| 17:34.06 | Katty | and have 3 different sets of musics |
| 17:34.10 | Katty | or averts |
| 17:34.10 | flujan | hi guys, I seeking a fax solution to my pabx.. I am searching about Hylafax and iaxmodem and Asterfax.... |
| 17:34.17 | karleeto | Qwell[]: so, if i research on dundi, i should be able to figure out what to do? |
| 17:34.25 | flujan | Which do you recommend to a production environment? |
| 17:34.30 | Strom_C | [TK]D-Fender: just from personal experience, I bought my D70 thinking it was going to be overkill, and i've taken more photos in the last three months than I have in probably the last five years |
| 17:34.41 | Qwell[] | flujan: dedicated analog line connected directly to the fax machine |
| 17:34.43 | flujan | which is easier to install and configure? |
| 17:34.49 | [T]ank | [TK]D-Fender: thank you... moving externip and localnet to the global section corrected the issue. thank you. |
| 17:34.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:35.10 | flujan | Qwell[]: for sure, but when this behavior is not a option what should I do? |
| 17:35.10 | denon | flujan: what Qwell said, or a dedicated fax server with fax modems or multiport fax cards |
| 17:35.21 | denon | doing it right is always an option |
| 17:35.29 | denon | and you will *not* be happy with any type of IP based fax solution |
| 17:35.41 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 17:35.45 | denon | which I'm guessing is where you're going with all this |
| 17:35.54 | flujan | what about thet fax over ip, and the T.38 protocol? |
| 17:36.00 | denon | it's a myth |
| 17:36.23 | karleeto | flujan: ip based fax solutions suck.. if it must be that way, you could use callwave, port your number to them and they email you the faxes |
| 17:36.28 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 17:36.56 | [TK]D-Fender | [T]ank: np |
| 17:37.24 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 17:37.29 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:37.31 | [TK]D-Fender | Strom_C: Considering how few that is for me personally... not sure I could jsutify it :) |
| 17:37.49 | coppice | T.37 is the sane fax over IP protocol, but nobody wants to use it |
| 17:38.00 | denon | nod |
| 17:38.03 | denon | well, we do :) |
| 17:38.10 | denon | on as5400s though |
| 17:38.41 | [TK]D-Fender | Strom_C: problem is that I keep thinking "what could I do with all the $ I'd save (300$-500$) difference", and will I really get a good value for the difference in products. |
| 17:38.55 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 17:39.21 | taqua2008 | what type of 5400 chassis do you use? |
| 17:39.29 | Strom_C | [TK]D-Fender: I haven't regretted a single cent of what I've spent |
| 17:39.49 | Strom_C | it's gone from "I took some snapshots" to "WHOA these are amazing photos" |
| 17:39.51 | denon | taqua2008: I dont deal with them personally anymore - just a bunch of PRIs into 5300s and 5400s |
| 17:40.00 | denon | not sure the specifics of what they have in there though |
| 17:40.26 | flujan | bah, email rox... I dunno why some people still uses fax. |
| 17:40.28 | flujan | :( |
| 17:40.28 | denon | extremely reliable fax rig though |
| 17:40.42 | taqua2008 | denon: Ok thanks |
| 17:41.05 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:41.25 | coppice | I think if you started shooting a few fax users, the others would fall into line and send e-mails |
| 17:41.48 | denon | haha |
| 17:42.17 | denon | coppice: funny, I was just discussing with Qwell what could be done about trying to convince people that faxing in realtime over ip was a bad idea |
| 17:42.29 | denon | hadn't considered that as an option, though |
| 17:42.34 | irule | why is there a long silence in letters/*.gsm? the directory application takes ages to spell names :s |
| 17:43.18 | coppice | while courts give credence to faxes they will continue |
| 17:43.26 | [TK]D-Fender | irule: w h a t a r e y o u t a l k i n g a b o u t ? |
| 17:43.46 | denon | coppice: well, it is a really handy device |
| 17:43.48 | denon | I dont use it .. |
| 17:44.01 | denon | but to just jack it into a phone line wherever you are, shove a piece of paper in, and walk away |
| 17:44.19 | denon | no isp subscription, no getting an IP.. no finding a local provider number of wifi |
| 17:44.23 | denon | s/of/or/ |
| 17:44.49 | karleeto | DUNDi is an opensource free thing, right? |
| 17:45.36 | irule | hi there [TK]D-Fender, the files in /var/lib/asterisk/sounds/letters have the letters spelled each with a silence after them, I just dont know why they have been recorded like that, I should just remove the silence from the sound files |
| 17:46.00 | [TK]D-Fender | irule: Go grab the HQ versions and get editing :) |
| 17:47.38 | irule | [TK]D-Fender HQ? even the mexican sounds I found somewhere on the net have the silence included! lol |
| 17:48.04 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:48.16 | [TK]D-Fender | irule: Oh yes, and "mexican sounds I found somewhere on the net" sounds like a quality source indeed! |
| 17:49.11 | wunderkin | they need a little silence or they will just be talking really fast |
| 17:49.18 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 17:49.23 | jmls | hey guys |
| 17:49.49 | irule | [TK]D-Fender well, actually I dont know who tried to convince allison smith that she could be understood in HER spanish lol |
| 17:49.54 | jmls | is there anyway of setting a variable in another channel from the dialplan ? |
| 17:50.24 | jmls | I know I can use the astdb, but would rather not |
| 17:50.30 | [TK]D-Fender | jmls: Noth that I can think of. |
| 17:50.44 | jmls | [TK]D-Fender: Boo. Hiss. |
| 17:50.46 | jmls | ;) |
| 17:50.57 | irule | jmls there is always mysql |
| 17:51.40 | jmls | I'd rather not have to read/write a db record and then have to remember to delete it when I'm done |
| 17:52.02 | jmls | at least channel variables are cleaned up for you when the channel ends. |
| 17:52.26 | [TK]D-Fender | jmls: Yes, we call this tactic AGI <- |
| 17:52.37 | jmls | is there anyway of reading a variable from another channel from the dialplan ;) |
| 17:53.04 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
| 17:53.28 | *** part/#asterisk Shoeb (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 17:54.14 | *** join/#asterisk sysreq (n=sysreq@197.219-ppp.3menatwork.com) |
| 17:54.17 | [TK]D-Fender | jmls: Same answer.... |
| 17:54.48 | wunderkin | you don't need agi for that.. use ImportVar |
| 17:54.50 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 17:54.59 | jmls | importvar ?? |
| 17:55.19 | *** join/#asterisk pusanggala (n=a@58.69.243.203) |
| 17:56.02 | [TK]D-Fender | ImportVar |
| 17:56.11 | jmls | oooohhhh. |
| 17:56.31 | jmls | ah hah. |
| 17:56.59 | jmls | oh - there's no ExportVar :( |
| 17:57.14 | jmls | nevermind. ImportVar may well suit my needs |
| 17:57.55 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 17:58.21 | ramindia | how can i store the recordings on other server ? |
| 17:59.50 | [TK]D-Fender | ramindia: nfs, smb, etc, take your pick. |
| 18:00.02 | mvanbaak | rsync |
| 18:00.19 | ramindia | rsync ? on realtime ? |
| 18:00.48 | ramindia | using nfs..i can store directly on other server right? how about performance.. |
| 18:01.05 | mvanbaak | uhhuh |
| 18:01.33 | ramindia | mvanbaak: rsync .. want to move recordings to other server |
| 18:01.49 | ramindia | but iam looking realtime recording store on other server |
| 18:01.54 | mvanbaak | then no rsync |
| 18:02.04 | mvanbaak | use some remote mount for that |
| 18:02.38 | karleeto | !! is there anyone in here willing to have a private conversation about DUNDi with me for a few minutes? |
| 18:02.46 | ramindia | how performance like compare to 15K rpm over NFS ethernet mount drive.. |
| 18:03.25 | mvanbaak | ramindia: depends on your needs |
| 18:03.35 | mvanbaak | if you only want to store voicemail on nfs it's ok I think |
| 18:03.56 | ramindia | no voicemails..................RECORDINGS |
| 18:03.59 | mvanbaak | unless you nfs mount it on a 33k6 line and want to record 10.000 simultanious calls |
| 18:04.20 | ramindia | talking about 200cals |
| 18:05.01 | mvanbaak | ah |
| 18:05.04 | mvanbaak | hhmm |
| 18:05.12 | mvanbaak | you'll need a good backlink for that |
| 18:05.22 | mvanbaak | but 100mbit to a netapp will be enough for that |
| 18:05.52 | Daviey | Hi, what 2 port PRI card do people recommend? Is the echo cancellation module worthwhille? |
| 18:05.54 | ramindia | netapp will be expensive..how about P4 kind of box ? |
| 18:06.29 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:07.07 | *** join/#asterisk bintut (n=bintut@cm38.gamma176.maxonline.com.sg) |
| 18:07.11 | bintut | hello all.. |
| 18:08.18 | [TK]D-Fender | Daviey: Sangoma A102d |
| 18:08.24 | bintut | quick question: how can i send (attach) voicemails to a specific gmail hosted e-mail address and delete afterwards? |
| 18:08.49 | [TK]D-Fender | bintut: its all in the sample voicemail.conf.... |
| 18:08.50 | Daviey | [TK]D-Fender: what about echo cancellation, is it worthwhille? |
| 18:08.57 | [TK]D-Fender | Daviey: ESSENTIAL |
| 18:09.13 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:09.22 | Strom_C | Daviey: digium te2xxp w/echo can module |
| 18:09.45 | *** join/#asterisk iBuMp- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:09.56 | Daviey | [TK]D-Fender: That seems more expensive than digium hw, is it better? |
| 18:10.12 | [TK]D-Fender | Daviey: Price is virtually identical, and yes. |
| 18:10.48 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:10.50 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 18:12.10 | [TK]D-Fender | Daviey: www.telephondepot.com , Sangoma seems even cheaper than the digium |
| 18:12.46 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:13.02 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:13.25 | Daviey | [TK]D-Fender: I'm UK based.. been quoted ÂĢ719.50 for te212p |
| 18:13.44 | Daviey | (if i place large order on handsets) |
| 18:14.21 | bintut | [TK]D-Fender: i'm not sure on my configs at http://paste.debian.net/33469 |
| 18:14.49 | *** join/#asterisk jaiger (n=jaiger@fire.innovationsw.com) |
| 18:15.04 | Yourname` | Hi, so I have two sessions I see in the console, one from me, and one from my friend. How can I kill the other connection? I tried kill pts/4 |
| 18:15.09 | Yourname` | And it didn't work |
| 18:15.26 | bintut | [TK]D-Fender: i got a voicemail voice prompt when no one answers but the .wav file is not sent to my e-mail.. |
| 18:15.53 | [TK]D-Fender | bintut: I prefer setting those options on the VM box line, and I don't use exim so I can't comment ont he format.... |
| 18:16.09 | bintut | [TK]D-Fender: i must admit that i don't have a mail server here. my e-mail address is being hosted by gmail. and, my exim is not even running as a daemon on my debian etch. |
| 18:16.23 | [TK]D-Fender | bintut: That may be the problem... |
| 18:16.46 | bintut | [TK]D-Fender: which one? |
| 18:16.56 | [TK]D-Fender | bintut: lack of server daemon running |
| 18:17.35 | bintut | [TK]D-Fender: but, can't i just send an e-mail without running a daemon? |
| 18:17.42 | Yourname` | Oops, wrong window. |
| 18:17.53 | [TK]D-Fender | bintut: I'm not sure on the fine points personally... |
| 18:17.56 | bintut | [TK]D-Fender: i mean, like just sending an e-mail using the "mail" command? |
| 18:18.00 | [TK]D-Fender | bintut: go check your mail queue |
| 18:18.23 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-182-96.hsd1.wa.comcast.net) |
| 18:20.44 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:21.54 | *** join/#asterisk _DAW_ (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:22.28 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:24.11 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:24.55 | Innatech | bintut: mail just shoves things into sendmail. |
| 18:25.34 | Innatech | bintut: from "man mail" : "all mail goes through sendmail." |
| 18:27.47 | Innatech | If you want to get all crazy, you might be able to convince asterisk to send mail through pine or something. Probably more troublesome that it's worth. |
| 18:28.04 | Innatech | *than |
| 18:29.45 | bintut | Innatech: but, does it need to run a daemon? |
| 18:30.02 | Innatech | pine? |
| 18:30.14 | bintut | Innatech: i'm running debian etch here and sendmail is just a symbolic link to exim which i don't run |
| 18:30.29 | Innatech | ah. |
| 18:30.34 | bintut | Innatech: i don't have pine either. i have the mail command though. |
| 18:30.54 | Innatech | Pine is a simple SMTP client. I can promise you it's in the Etch repos. |
| 18:31.00 | bintut | Innatech: i mean, exim is not running |
| 18:31.18 | bintut | Innatech: yeah, i know pine is in the repository |
| 18:31.20 | Innatech | Pine is a client. It shouldn't need a local daemon. |
| 18:31.34 | bintut | Innatech: but, can't i just use the mail command here? |
| 18:32.01 | bintut | Innatech: it's not possible to use the mail command directly from the voicemail.conf ? |
| 18:32.05 | Innatech | If you look at man mail, you will notice that it mentions that it sends everything through sendmail. |
| 18:32.27 | Nugget | There's a difference between having a mail transport agent (sendmail, exim, postfix, etc...) and having one listening for remote connections. |
| 18:32.50 | Nugget | even if you don't have exim listening for remote connections it can still perform deliveries |
| 18:32.50 | Innatech | I think he wants to avoid running one at all, unless I missed the mark. |
| 18:33.06 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:33.40 | bintut | Innatech: yes, i don't want to run an MTA because i already have gmail account already |
| 18:33.43 | Nugget | "running" is vague in this context. |
| 18:33.59 | Nugget | You need an MTA on a unix box or all sorts of things stop working right. |
| 18:34.06 | Nugget | you can't get crontab emails, for example |
| 18:34.12 | bintut | Nugget: i mean, i don't want to have a MTA daemon running on my server |
| 18:34.18 | Innatech | Yeah, unless you know what you're doing, you want to run an MTA. |
| 18:34.23 | Nugget | that's fine. "running" isn't neccessary |
| 18:34.36 | Innatech | "have" an MTA. Whatever. |
| 18:34.38 | Nugget | exim can still deliver those mails just fine even if there's not a "running" daemon |
| 18:34.40 | bintut | Nugget: i don't have any mta running on my server and it works fine for me |
| 18:34.53 | Nugget | "running" is too vague a word to use in this context. |
| 18:35.12 | bintut | ok |
| 18:35.29 | bintut | Nugget: so, what shall i do then? |
| 18:35.56 | Nugget | don't mess with the mta, continue to use it for delivery. |
| 18:36.08 | Innatech | ^ sage advice. |
| 18:36.20 | Nugget | asterisk, crontab, and a host of other processes on your asterisk server are all depending on the mta being there |
| 18:36.36 | Nugget | that has absolutely nothing at all to do with your gmail account. |
| 18:36.47 | Nugget | they aren't even remotely related |
| 18:37.07 | [TK]D-Fender | .... telnet |
| 18:37.08 | Nugget | telnet is eeeeeeevil! |
| 18:37.10 | [TK]D-Fender | ;) |
| 18:37.42 | BSD_Tech | ssh is your frien |
| 18:37.43 | BSD_Tech | d |
| 18:38.09 | Innatech | telnet to port 25 for max lewlz. EHLO thar! |
| 18:39.57 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 18:40.10 | Nugget | I know a guy who has rcpt.to as his vanity domain. |
| 18:41.35 | *** join/#asterisk anthm (n=anthm@m015f36d0.tmodns.net) |
| 18:41.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:42.11 | Innatech | heh. |
| 18:43.33 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:43.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:43.54 | bintut | hhmmm |
| 18:45.59 | bintut | how about using nail? |
| 18:47.58 | _mm_ | ~book |
| 18:47.59 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:54.24 | *** join/#asterisk Avalone (n=Avalone_@217.118.82.47) |
| 19:00.44 | Innatech | bintut: I suppose that's one way to make sure people get their messages. Make sure you've got good workers comp coverage. |
| 19:04.55 | bintut | Innatech: i'm sorry. what do you mean? |
| 19:05.34 | Innatech | "<bintut> how about using nail?" ba-dum-dum. Ching. |
| 19:06.22 | *** join/#asterisk Axet (n=john@smirnoff.nurvnet.org) |
| 19:06.50 | Axet | Hi all |
| 19:07.16 | Axet | I'm new to asterisk and intend using it with Cisco 7961G phones but was wondering if there are any firmwares that are to avoid or that have an excellent reputation ? |
| 19:08.44 | Axet | anyone ? :) |
| 19:09.29 | Strom_C | have you already purchased the phones? |
| 19:09.34 | *** join/#asterisk toombaloomba (n=hola@do.you.like.my.frippers.com) |
| 19:09.42 | Axet | I have managed to get one for free |
| 19:09.46 | Axet | to test it out |
| 19:09.49 | Qwell[] | send it to me, and buy Polycoms |
| 19:09.52 | Strom_C | i'd recommend you buy polycoms |
| 19:09.59 | [TK]D-Fender | I'll third that... |
| 19:10.04 | Axet | polycoms ? |
| 19:10.08 | Qwell[] | ~polycom |
| 19:10.08 | jbot | well, polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 19:10.08 | Axet | i'll look them up now* |
| 19:10.13 | coolbeans | No NAT with 79x1's |
| 19:10.29 | Strom_C | coolbeans: no, it's usable behind NAT |
| 19:10.31 | Qwell[] | coolbeans: NAT works fine with them on chan_skinny ;) |
| 19:10.34 | Axet | coolbeans : whats the problem with that ? |
| 19:11.28 | coolbeans | Axet: It doesn't work? |
| 19:11.44 | Axet | why do the phones need nat to reach the asterisk server ? |
| 19:12.02 | Axet | if the asterisk server is on the same network what's the use for nat ? |
| 19:12.15 | putnopvut | Axet: I don't think that's what he means. |
| 19:12.35 | putnopvut | I think he means that the phones won't work behind NAT. |
| 19:12.51 | toombaloomba | hello, anyone know of a way to change your voice using asterisk or a softphone? |
| 19:12.57 | [TK]D-Fender | Axet: Cisco costs more, and offers less. Best forgotten about. |
| 19:12.57 | putnopvut | Or rather don't traverse NAT's properly. |
| 19:13.11 | Strom_C | toombaloomba: a helium balloon |
| 19:13.21 | Axet | putnopvut : I don't understand why the cisco phones would need to go through nat |
| 19:13.25 | toombaloomba | LOL not a bad idea Strom_C |
| 19:13.32 | coolbeans | Axet: Because some of us use NAT? |
| 19:13.33 | [TK]D-Fender | toombaloomba: underwear 2 sizes too small. |
| 19:13.48 | Innatech | toombaloomba: a swift kick in the 'nads. |
| 19:13.52 | Strom_C | [TK]D-Fender: cf "Take On Me" |
| 19:14.02 | coolbeans | Axet: Because one of the biggest value propositions of VoIP is the ability to have a disperate phone system? |
| 19:14.07 | [TK]D-Fender | Strom_C: cf? |
| 19:14.12 | putnopvut | Axet: when you make an outgoing call, it's likely you're going to have to traverse a NAT, and so if those phones don't traverse NAT's properly, then you can't make outgoing calls. |
| 19:14.15 | Strom_C | compare |
| 19:14.24 | Axet | coolbeans : ah ok but I intend using an asterisk server on my lan |
| 19:14.25 | [TK]D-Fender | Strom_C: ...? |
| 19:14.37 | Strom_C | [TK]D-Fender: you are familiar with the song "Take On Me" right? |
| 19:14.38 | Axet | coolbeans : not a distant one |
| 19:14.45 | coolbeans | Axet: Oh, sorry. I thought you were one of those guys knocking NAT and using VoIP over the internet. My bad, my appologies. |
| 19:14.48 | [TK]D-Fender | Strom_C: A-Ha, yes |
| 19:15.02 | Innatech | toombaloomba: in all seriousness, if you want to use a voicechanger, use a POTS phone, a voice changer box, and an ATA. It'll be easier than trying to decode, process and reencode on the fly if that's even possible. |
| 19:15.02 | Axet | coolbeans : nah but thanks for warning me about cisco ip phones :) |
| 19:15.03 | [TK]D-Fender | Strom_C: Not sure about the meaning of your usage of it |
| 19:15.17 | Strom_C | tight underwear -> high voice -> high note -> "Take on Me" |
| 19:15.32 | [TK]D-Fender | Strom_C: ok, fine, sure :) |
| 19:15.48 | Strom_C | next time i'll explain it in horrible horrible French! |
| 19:16.01 | Axet | back to my first question ... are there any firmwares I should avoid ? which one should I use ? |
| 19:16.04 | coolbeans | Axet: Polycom phone just work very well with Asterisk. We actually have Cisco 7941's in our main office using a VPN to our datacenter for voice. Works well, but the Polycoms are just easier to use and more readily support the features of Asterisk. |
| 19:16.13 | toombaloomba | Innatech yea i figured as much just thought I'd ask, I think someone mentioned they saw a softphone that did it, but may have been lying |
| 19:16.13 | coolbeans | s/phone/phones |
| 19:16.31 | bintut | gtg now.. thanks all.. |
| 19:16.37 | Axet | coolbeans : well if I ever come to buying ip phones I'll follow up on your advice |
| 19:16.42 | Nivex | A request for the digium folk: can a readonly DAV configuration be put on http://ftp.digium.com/ so I can use cadaver to grab the latest updates as they come out? |
| 19:16.47 | Innatech | toombaloomba: well, with a softphone you might be able to use jacktools or similar to process the audio before it gets to the softphone. |
| 19:16.49 | Axet | coolbeans : but right now all I have are cisco phones :p |
| 19:16.55 | [TK]D-Fender | Strom_C: I just missed the glue to associate that all together is all... |
| 19:16.58 | *** join/#asterisk janinge (n=janinge@211.80-202-239.nextgentel.com) |
| 19:17.23 | blitzrage | Nivex: probably something you'll have to email to someone at Digium |
| 19:17.27 | toombaloomba | Innatech i just saw Asterisk+Realtime+Voice+Pitch+Changer on the wiki hmmm |
| 19:17.37 | Innatech | haven't seen that. |
| 19:17.47 | toombaloomba | its beta |
| 19:17.52 | Katty | weeeeeeeeeeee!!! |
| 19:17.59 | wothinn | If it just changes pitch, it can probably be trivially undone. |
| 19:18.00 | blitzrage | !!!eeeeeeeeeeeeew |
| 19:18.05 | Katty | oh hush. |
| 19:18.07 | Katty | actually! |
| 19:18.08 | wothinn | Depends what you want it for whether or not that matters. |
| 19:18.09 | Katty | i have query |
| 19:18.11 | Innatech | If you want to do it in software, I'd use jacktools and a plugin host. |
| 19:18.12 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 19:18.15 | blitzrage | !yllautca |
| 19:18.31 | Katty | can i have different music on hold musics/adverts for my different zapgroups? (g1,g2,g3,etc) |
| 19:18.37 | Strom_C | Katty: yes |
| 19:18.39 | toombaloomba | hmm I think most sound cards come with some way of messing with the voice, might work too |
| 19:18.41 | Katty | hottt |
| 19:18.57 | Katty | is it just [zapwhatevergroup] in musiconhold.conf? |
| 19:19.01 | Katty | and then you put your stuff underneath it |
| 19:19.12 | dlynes_laptop | Are there any good softphones out there besides Eyebeam/xten? |
| 19:19.19 | Katty | i like sjphone |
| 19:19.19 | dlynes_laptop | That run on Windows? |
| 19:19.22 | Katty | and iaxcomm |
| 19:19.27 | Innatech | idefisk |
| 19:19.38 | Katty | of course iaxcomm doesn't do sip, i dont think |
| 19:19.39 | Innatech | <3 idefisk forever. |
| 19:19.40 | Katty | and sjphone does sip |
| 19:19.40 | dlynes_laptop | sjphone, iaxcomm, and idefisk all run on windows? |
| 19:19.48 | Katty | sjphone and iaxcomm do |
| 19:19.52 | dlynes_laptop | And they don't look like crap? |
| 19:19.54 | Katty | never used idefisk, so i dunno if it runs on windows |
| 19:20.00 | Katty | image google them |
| 19:20.04 | dlynes_laptop | Ok |
| 19:20.05 | [TK]D-Fender | dlynes_laptop: idefisk does |
| 19:20.08 | dlynes_laptop | Thanks, guys |
| 19:20.16 | dlynes_laptop | [TK]D-Fender: idefisk looks like crap? |
| 19:20.23 | dlynes_laptop | And can they all do g729? |
| 19:20.25 | Innatech | no, it runs on windows. Looks very nice. |
| 19:20.35 | [TK]D-Fender | crap so-so, G729, IIRC, yes |
| 19:20.48 | Innatech | I think they changed the name tho. |
| 19:21.30 | Innatech | aww...apparently my softphone aesthetics are lacking. :( |
| 19:21.53 | Katty | software designers should hire interior decorators. |
| 19:21.57 | Katty | or professional website designers. |
| 19:22.04 | Katty | who hire interior decorators. |
| 19:22.14 | Katty | is it bad i've gone to lowes for color schemes for websites? |
| 19:22.28 | Innatech | As long as you stick to the websafe isle. |
| 19:22.31 | Katty | the paint section is awesome for color schemes :> |
| 19:23.58 | Innatech | "Yes, I'd like a gallon bucket of #C8C8C8 semigloss, please. I'm refinishing my CSS this weekend." |
| 19:24.06 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 19:24.39 | [TK]D-Fender | Innatech: C4 would finish things much faster.... ballistically speaking... |
| 19:25.37 | Innatech | heh. |
| 19:25.49 | Innatech | "I don't know....do you have a more explosive shade of grey?" |
| 19:26.08 | dlynes_laptop | $20G's for annual license for sjphone? |
| 19:26.13 | dlynes_laptop | Somebody's dreaming! |
| 19:26.26 | [TK]D-Fender | Innatech: "Plastique Pewter" |
| 19:26.41 | Innatech | Good for strained dinner parties with the in-laws. |
| 19:26.48 | [TK]D-Fender | <- Iron Chef of the Anarchist's Cook-book |
| 19:26.52 | Innatech | <light coffe cup, fling, duck> |
| 19:26.57 | dlynes_laptop | Wow...and doesn't even do g.729 or g.723 for that price, either |
| 19:28.07 | *** join/#asterisk guillote_GNU (n=guillote@host111.190-30-66.telecom.net.ar) |
| 19:28.52 | redax | how can I choose the codec when I do Dial(SIP/ext@remotehost) ? |
| 19:29.51 | Strom_C | redax: in sip.conf |
| 19:33.44 | zpertee | where is a good place to buy commercial grade equipment |
| 19:33.47 | Jingles | also, make sure your SIP device can support that codec. |
| 19:35.36 | redax | aaa... wait a bit... sip.conf on the remotehost ? |
| 19:35.43 | *** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net) |
| 19:36.44 | redax | but if sip.conf has several codecs allowed, how to choose g729 for example? |
| 19:36.52 | Strom_C | disallow=all |
| 19:36.55 | Strom_C | allow=g729 |
| 19:37.33 | *** join/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 19:38.09 | redax | Strom_C: basicly, I wanted to ask is there a way to Set() some channelvar before the Dial() to `prefer' some codec at the client side |
| 19:38.59 | Strom_C | no; you have to do that in the channel driver configuration file |
| 19:39.09 | BSD_Tech | earthlink true voice support sucks |
| 19:39.11 | *** join/#asterisk TedNJ37 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 19:39.13 | BSD_Tech | man |
| 19:39.19 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 19:39.21 | TedNJ37 | Can someone help me please? I am running Trixbox. How can I determine the order in which the box plays the files setup for Music On Hold? It is not alphabetically. I don't have Music on Hold set up to play files randomly yet, it is not playing them in alphabetical order. |
| 19:39.46 | MindTheGap | suppose i have this: "2121 => 100" in GLOBALS. can I use this? " exten => 2121,1,Goto(context|$"${ARG1}"|1) ". |
| 19:40.10 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 19:40.16 | Shaun2222 | anybody know if a P4 3.0 can handle 15 concurrent calls using the g711 codec? |
| 19:40.21 | Katty | help, i've lost my voicemail directory |
| 19:40.25 | Shaun2222 | and or the g729 codec? |
| 19:41.11 | redax | Strom_C: I see, thanks. |
| 19:41.19 | MindTheGap | sorry, exten => 2121,1,Goto(context|$"${EXTEN}"|1) ". |
| 19:41.45 | Strom_C | Shaun2222: yes |
| 19:42.34 | [TK]D-Fender | Shaun2222: tons more than that |
| 19:42.51 | Katty | [TK]D-Fender: remind me, again, where voicemail is located. |
| 19:43.06 | [TK]D-Fender | Katty: /var/spool/asterisk/voicemail |
| 19:43.09 | Hmmhesays | friendly neighbors |
| 19:43.20 | Shaun2222 | can you guestimate how many you think it could handle? |
| 19:43.27 | Katty | oh yeah |
| 19:43.28 | Katty | spool |
| 19:43.29 | *** part/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 19:43.29 | Katty | not lib |
| 19:44.09 | [TK]D-Fender | Katty: there is usually a symlink under /var/lib/asterisk/sounds but I really don't advocate the "fake" way |
| 19:44.15 | karleeto | is there anyone in here willing to have a private conversation about DUNDi with me for a few minutes? |
| 19:44.40 | [TK]D-Fender | karleeto: Virtually nobody cares about Dundi |
| 19:44.51 | karleeto | [TK]D-Fender: why is that? |
| 19:44.51 | *** join/#asterisk x1fa47 (n=address@226.Red-80-26-39.staticIP.rima-tde.net) |
| 19:45.09 | [TK]D-Fender | karleeto: Ask yourself the reverse. Its * specific. |
| 19:45.20 | karleeto | so |
| 19:45.37 | karleeto | this is the * chat room is it not? |
| 19:45.49 | Hmmhesays | i don't know a single person that uses dundi |
| 19:46.03 | [TK]D-Fender | karleeto: True, and you here little about chan_mgcp here either. |
| 19:46.08 | [TK]D-Fender | hear* |
| 19:46.12 | sevard | no do i |
| 19:46.17 | [TK]D-Fender | ditto |
| 19:46.19 | karleeto | well, i just have this 3 location setup i'l like to do, i just dont know how i should configure it |
| 19:46.31 | karleeto | i've never done a multiple * box setup before |
| 19:46.33 | [TK]D-Fender | karleeto: Just a basic IAX link would do. |
| 19:46.46 | [TK]D-Fender | karleeto: lookup "asterisk dual servers" on the WIKI |
| 19:46.48 | [TK]D-Fender | ~wikis |
| 19:46.49 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 19:47.14 | [TK]D-Fender | karleeto: You keep asking about the means without stating the NEED. You were thinking backwards. |
| 19:47.15 | karleeto | [TK]D-Fender: and someone on the 100 extentions could forward a call to someone on the 200 extentions or forward to their voicemail? |
| 19:47.27 | [TK]D-Fender | karleeto: Sure. |
| 19:47.33 | [TK]D-Fender | karleeto: Get reading. |
| 19:47.40 | karleeto | [TK]D-Fender: alright |
| 19:49.58 | TedNJ37 | Can someone help me please? I am running Trixbox. How can I determine the order in which the box plays the files setup for Music On Hold? It is not alphabetically. I don't have Music on Hold set up to play files randomly yet, it is not playing them in alphabetical order. |
| 19:52.45 | neverblue | looking for any VOIP providers, pm me please |
| 19:52.51 | sevard | TedNJ37: #trixbox |
| 19:53.00 | TedNJ37 | They are not responding at all sevard. |
| 19:53.05 | TedNJ37 | I'll keep trying there. |
| 19:53.06 | TedNJ37 | Thanks. |
| 19:58.49 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 19:59.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:00.27 | Hmmhesays | [TK]D-Fender: can I pm you? |
| 20:00.52 | lirakis | when should i use the Answer() funciton.. and when shouldnt i? |
| 20:01.00 | *** join/#asterisk bkruse_home (i=kruz@nat/digium/x-058aae855be6cd66) |
| 20:01.01 | [TK]D-Fender | Hmmhesays: sure |
| 20:01.37 | lirakis | I seem to get strange results when I answer .. and then try to bridge .. |
| 20:02.09 | lirakis | and im not sure if im supposed to be using Answer in that setting |
| 20:03.26 | Shaun2222 | [TK]D-Fender / Strom_C : Any of you guys know or have setup a HA env? Right now i have sip phones in remote locations. Asterisk server gets the calls, i have queues setup, to hold the calls until sombody picks up. I want to have this system be more redundant though and have another aksterisk server in a remote location. I'm not sure how that would work with the queues or the phones... I also need the phones to connect to both servers |
| 20:03.55 | Shaun2222 | or if one asterisk could not make calls out it should route the call to the other... |
| 20:04.39 | lirakis | Shaun2222: you need a load balancing proxy |
| 20:04.44 | lirakis | Shaun2222: like openSER |
| 20:04.59 | Shaun2222 | thats still a single point of failure |
| 20:06.12 | lirakis | it is.. i suppose you could have.. two servers in round robin DNS |
| 20:06.25 | lirakis | .. that would probably wreak havok with registrations |
| 20:06.59 | Shaun2222 | ya i want to say i heard somthing about SRV records to do a HA type setup, but i dont really know if that was right. |
| 20:08.23 | *** join/#asterisk dirk- (n=root@82-33-155-212.cable.ubr04.wiga.blueyonder.co.uk) |
| 20:08.33 | coolbeans | Hey, in app_voicemail.c's msgXXXX.txt file, the origtime, is that a UNIX epoc time stamp? |
| 20:08.38 | lirakis | Shaun2222: .. well i am just now setting up a call center with 2 servers in seperate geographic locations... i am using openser as a hotcut failover.. in the case of total IP failure.. i have T1 links to an Alcatel DEX |
| 20:12.43 | Shaun2222 | lirakis: i assume your either letting others connect using asterisk or just dailing out to there phone/office/cell phones? |
| 20:13.20 | *** part/#asterisk TedNJ37 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 20:15.17 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 20:16.22 | lirakis | Shaun2222: .. well its a call center.. so weve got 800 DIDS that come in TDM and we convert to SIP... those go into the asterisk for customer service calls. There are also sales campaigns there.. so they go out SIP to a Nextone... then the TDM stuff is just failover.. like if the media gateway fails .. we stop getting SIP 800 DIDS... so then the Alcatel takes them in TDM and sends PRI to asterisk. Also the Alcatel has outbound configured. |
| 20:20.13 | kombi | I'm having this strange issue and maybe someone can help: I execute System() with a php shell script from the dialplan. The script fires an originate statement over manager and connects two extensions, one starts meetme, the other ices. I works like 5 times, then doesn't 4 times, works again etc. I have checked all logs but can't find anything.. |
| 20:21.52 | Katty | herro. |
| 20:22.09 | kombi | Katty, what's for dinner? |
| 20:22.15 | Katty | how do i make the directory ask for first name or last name, rather than just last name? |
| 20:22.22 | Katty | kombi: tuna casserole and pecanless pie |
| 20:22.28 | kombi | yum.. |
| 20:22.33 | Dan0maN_Work | mmmm. pie. |
| 20:22.40 | Katty | pie good. |
| 20:23.09 | Nivex | when come back bring pie |
| 20:23.15 | Katty | :> |
| 20:28.10 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 20:28.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:28.38 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 20:29.03 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:29.19 | blitzrage | of the poon-tang flavour perhaps even |
| 20:29.37 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 20:29.53 | *** join/#asterisk pepse (n=pepse@71-223-123-64.phnx.qwest.net) |
| 20:30.04 | pepse | hi guys |
| 20:30.44 | pepse | i'm trying to route 1NXXNXXXXXX numbers to one provider and then 18XXNXXXXXX to another, how can I accomplish this? |
| 20:31.19 | pepse | seems like whether I put the extens of the tollfree numbers before or after the 1NXX one, they are completely ignored |
| 20:31.22 | blitzrage | create separate pattern matches and Dial(SIP/provider_one/${EXTEN}) and Dial(SIP/provider_two/${EXTEN}) |
| 20:31.39 | blitzrage | the order does not matter |
| 20:31.45 | blitzrage | more specific matches matter |
| 20:32.05 | blitzrage | the order you put stuff in the dialplan (extensions.conf) doesn't matter because it all gets parsed and sorted based on the internal rules |
| 20:32.41 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:33.31 | pepse | hm, ok. so do i need to like, make extens for 12xx, 13xx, etc? |
| 20:33.36 | blitzrage | oej: go on vacation already would ya?! |
| 20:33.58 | blitzrage | pepse: depends what you're trying to do... remember to pattern match by starting with _ (underscore) |
| 20:34.04 | pepse | cause if I have the _1XXXXXXXXXX one, it's the only one that works |
| 20:34.17 | zpertee | has anyone tried zapmicro hardware? |
| 20:34.39 | blitzrage | pepse: you probably want to pastebin your dialplan and explain what you're trying to do |
| 20:34.46 | blitzrage | probably something trivial you're missing |
| 20:35.53 | pepse | two lines should explain my deal.. |
| 20:35.59 | pepse | exten => _1800NXXXXXX,1,Dial(IAX2/etc..) |
| 20:36.36 | pepse | exten => _1XXXXXXXXXX,2,Dial(SIP/etc..) |
| 20:36.50 | pepse | obviously that 2 is because the 1 sets callerid |
| 20:37.30 | pepse | the _1800 is ignored |
| 20:38.44 | dirk- | what about adding a series of 11xxx 12xxx 13xxx etx to force it? nine rules instead of 2? |
| 20:38.58 | pepse | dirk-: yeah that's what i was askin if i need to do earlier |
| 20:39.08 | dirk- | ah, sorry |
| 20:39.28 | pepse | it's not a bad idea, i'm just wondering if there should be a better way |
| 20:39.29 | dirk- | given the way the matching works, Id think its the only way |
| 20:39.59 | pepse | so you can't make a more specific match take precedence over a more broad match? |
| 20:40.11 | Strom_C | specific should always take precedence |
| 20:40.20 | Strom_C | pepse: pastebin your whole dialplan |
| 20:41.14 | BSD_Tech | never never use earthlink they bounce you around and give you the run around when you try to get a supervisor |
| 20:41.44 | pepse | Strom_C: hm, well i have my dialplan chopped up in seperate files |
| 20:41.53 | Qwell[] | BSD_Tech: like every other company |
| 20:42.16 | Strom_C | pepse: well then pastebin the relevant portion of your dialplan |
| 20:42.39 | pepse | Strom_C: but that was what i just pasted :) those two lines are the relevant portion, i think anyway.. |
| 20:42.45 | pepse | other than the includes |
| 20:42.46 | Strom_C | uh no |
| 20:42.50 | *** join/#asterisk ESCulapio_ (n=elvyn@66.44.88.200.l.sta.codetel.net.do) |
| 20:42.57 | Strom_C | that's a summary |
| 20:43.09 | Strom_C | pastebin the complete code, please |
| 20:43.41 | Strom_C | damn, these brownie muffin things from Swiss Chalet are yuuuuuuuuuuummmmmmmmy |
| 20:43.58 | Katty | anyone know how to make the directory ask for first AND last name, rather than just last name? |
| 20:44.10 | Strom_C | Katty: you can make it ask for either or |
| 20:44.13 | Strom_C | but not both |
| 20:44.16 | Katty | Strom_C: okay. |
| 20:44.17 | Strom_C | IIRC |
| 20:45.28 | pepse | Strom_C: http://pastebin.ca/635074 |
| 20:45.34 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 20:45.44 | dirk- | i've not tried it but the current beta of freepbx seems to offer that function |
| 20:45.55 | dirk- | if it works you could examine what they are doing |
| 20:45.56 | Katty | how do i get to directory stuff from the cli? |
| 20:46.02 | Katty | i don't see anything about 'directory' listed under help. |
| 20:46.12 | *** join/#asterisk MoutaPT (n=Blink@a213-22-40-195.cpe.netcabo.pt) |
| 20:46.15 | Strom_C | Katty: shoe application directory |
| 20:46.24 | ESCulapio_ | |
| 20:46.25 | ESCulapio_ | I have a problem with agi in bash and plicaciÃģn GET DATA |
| 20:46.46 | Strom_C | pepse: "most specific match" only applies within a single context |
| 20:46.56 | pepse | doh |
| 20:47.02 | Katty | k, so it's option f |
| 20:47.09 | MoutaPT | hi does any one can help me how to set CallerID anonymous using TE210P on a per user request, I mean i just want to set anonymous callerID for specfici users... |
| 20:47.25 | Katty | 'f - allow the caller to enter the first name of a user in the directory instead of using the last name' |
| 20:47.29 | Katty | but.. |
| 20:47.31 | ESCulapio_ | I have a problem with agi in bash and plicaciÃģn GET DATA |
| 20:47.32 | Katty | where do i set option f? |
| 20:47.36 | Strom_C | MoutaPT: show application SetCallerPres |
| 20:47.44 | pepse | I'll just copy the trunktollfree context into the les one |
| 20:47.49 | Strom_C | Katty: in your dialplan when you call Directory() |
| 20:47.49 | MoutaPT | thanks Strom_C |
| 20:47.56 | MoutaPT | I will have a look |
| 20:47.58 | Katty | ooo |
| 20:48.08 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:48.22 | Strom_C | [TK]D-Fender: help, being in canada is making me irascible |
| 20:49.13 | Katty | so my line reads: exten => 2,1,Directory(downstairs|downstairs), so i need to make it: exten => 2,1,Directory(downstairs|downstairs|f)? |
| 20:50.03 | [TK]D-Fender | Strom_C, contrary to your usage, "irascible" is not going to become the new "copacetic" :) |
| 20:50.26 | [TK]D-Fender | Strom_C, and a top rule of effective writing is never use a big word when a smaller one will do :) |
| 20:50.46 | kombi | which pastebin? (my beloved pastebin.ca is down..) |
| 20:51.04 | Katty | weee!! it worked. thanks Strom |
| 20:51.10 | kombi | ..not: http://pastebin.ca/635078 |
| 20:51.20 | kombi | why does it suck? |
| 20:52.02 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 20:52.39 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:53.07 | Katty | it sucks because it needs a hug |
| 20:53.09 | Katty | and possibly chocolate. |
| 20:53.19 | Katty | and if those don't fix it, i'd recommend a visit to the mall and new shoes. |
| 20:53.22 | Strom_C | it just needs to be given a little love. and sexual favors. |
| 20:53.27 | Katty | yes, and sex. |
| 20:53.38 | Katty | sex fixes a lot of things! |
| 20:53.44 | kombi | lol.. |
| 20:53.58 | Katty | especially if it's a male :P |
| 20:54.12 | kombi | which line shall i put the sex in? |
| 20:54.34 | kombi | kombi wonders whether his code is male or female.. |
| 20:54.50 | *** part/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com) |
| 20:55.34 | Katty | is it direct and to the point, or subtle with hints of deceipt? |
| 20:55.42 | Katty | deceit. |
| 20:55.44 | *** join/#asterisk digimania (n=none@24-119-242-84.cpe.cableone.net) |
| 20:55.45 | Katty | ...i can't spell. |
| 20:55.52 | sevard | i.... noticed. |
| 20:56.01 | Katty | >.< |
| 20:56.02 | kombi | Katty: got the point, I'd say the former |
| 20:56.05 | Katty | k |
| 20:56.20 | kombi | does that make it male or female? |
| 20:57.04 | kombi | damn, is anyone actually looking at the thing? |
| 20:59.36 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:59.48 | kombi | evidently not.. sigh.. you guys leave me hangin' |
| 20:59.59 | Katty | blah |
| 21:00.18 | Katty | wait, what's the url again? |
| 21:00.28 | kombi | http://pastebin.ca/635078 |
| 21:00.35 | Katty | thanks |
| 21:01.33 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 21:02.54 | Katty | oh php stuff |
| 21:02.59 | Katty | i dont' knwo anything about php stuff |
| 21:04.21 | kombi | thanks for looking though |
| 21:06.31 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 21:06.44 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 21:10.05 | Hmmhesays | depends on what you're looking for |
| 21:11.02 | MoutaPT | Strom_C I have 100 DDIs from my E1 circuit do you know what should I use on argument of SetCallerPres( ) to make aonymous outbound calls? |
| 21:11.21 | MoutaPT | prohib_failed_screen? |
| 21:11.31 | MoutaPT | or just prohib |
| 21:11.31 | MoutaPT | ? |
| 21:11.42 | MoutaPT | i'm not in the office to test it |
| 21:14.49 | *** join/#asterisk stridernzl (n=neville@125-237-98-1.jetstream.xtra.co.nz) |
| 21:24.38 | Strom_C | prohib_not_screened would be my best guess |
| 21:25.56 | *** join/#asterisk Chuji (n=brian@mail.point3media.com) |
| 21:26.04 | *** join/#asterisk kotique[male] (n=v@host-86-106-210-75.moldtelecom.md) |
| 21:26.48 | kombi | this drives me bananas, it works seven times in a row, then doesn't 3 times, then works once, then doesn't, then does... hrrmpf |
| 21:26.56 | kotique[male] | hey guys. does cisco's IOS have SIP registrar support ? I want to register my SIP phone to cisco running in dynamips :-) |
| 21:28.59 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:30.50 | Hmmhesays | I wish I could find a sip phone that supported BLA well |
| 21:30.51 | Hmmhesays | and was cheap |
| 21:30.56 | Hmmhesays | these spa-942s don't |
| 21:31.26 | [hC] | Aastra might be ok |
| 21:33.39 | Strom_C | Hmmhesays: polycom IP320 :) |
| 21:35.05 | [hC] | Any of you guys tried the SLA stuff in asterisk 1.4? |
| 21:35.10 | *** join/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 21:35.38 | *** join/#asterisk galeras (n=galeras@201.244.199.31) |
| 21:36.38 | x86 | [hC]: SLA stuff? |
| 21:36.44 | [hC] | Shared line appearances |
| 21:36.52 | x86 | BLA? |
| 21:37.01 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 21:37.05 | [hC] | same thing |
| 21:37.06 | [hC] | bridged/shared |
| 21:37.10 | x86 | ah i see |
| 21:37.20 | x86 | asterisk finally supports that? |
| 21:37.42 | Qwell[] | always has |
| 21:37.46 | [hC] | 1.4 does supposedly, but ive never tried it |
| 21:37.47 | Hmmhesays | do all of the polycoms support the bla? |
| 21:37.50 | [hC] | and i dont know how well it works |
| 21:37.51 | Qwell[] | or do you mean the Shared? |
| 21:37.52 | [hC] | and a customer wants it. |
| 21:37.56 | Hmmhesays | hint extensions have been around since 1.0 |
| 21:37.58 | Hmmhesays | I believe |
| 21:38.00 | [hC] | shared. key system emulation |
| 21:38.13 | [hC] | BLA/BLF are not the same, are they? |
| 21:38.15 | phessler_ | if I have a file include several others, and each of them have [global] defined, are they merged, or clobbered? |
| 21:38.19 | Hmmhesays | no they aren't |
| 21:38.26 | [hC] | bla =bridged line appearances, blf (hints) = busy lamp field |
| 21:38.39 | Hmmhesays | yeah they are the same |
| 21:38.43 | Hmmhesays | bla and sla are not the same |
| 21:38.47 | [hC] | oh |
| 21:38.57 | Hmmhesays | Do all the poly's support bla via the hint extension? |
| 21:39.12 | phessler_ | eg: extensions.conf includes both extensions_a.conf, extentions_b.conf; which both have globals (not defining the same variables, of course) |
| 21:39.35 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:39.41 | Hmmhesays | anyone? |
| 21:39.56 | Hmmhesays | I've read the aastra 480i's do |
| 21:40.10 | [hC] | aastras do, polycoms do, ciscos do |
| 21:40.21 | [hC] | if all you want is hints, and indicators for whos on the phone |
| 21:40.37 | Hmmhesays | I do |
| 21:40.40 | Hmmhesays | cisco's do? |
| 21:40.46 | [hC] | 79x1 series yes |
| 21:40.50 | Hmmhesays | I have a 7940 here |
| 21:40.53 | Hmmhesays | damnit |
| 21:40.54 | Hmmhesays | lol |
| 21:40.56 | [hC] | youd need a 7941. |
| 21:41.01 | [hC] | or a 7914 sidecar |
| 21:41.14 | [hC] | and i think the 7914 only attaches to the 7960 |
| 21:41.18 | [TK]D-Fender | 7914 only works in SCCP |
| 21:41.24 | Strom_C | pooooooooooooolllllllllllllllyyyyyyyyyyyyyyyyyyyyccccccccccccccooooooooooooooooooommmmmmmmmmmmmmmm |
| 21:41.30 | x86 | polycom++ |
| 21:41.32 | [hC] | anyways, has anyone played with shared line appearances in 1.4? I wanna know if its worth trying |
| 21:41.33 | [TK]D-Fender | Hmmhesays, if its presence you want, Polycom or Aastra |
| 21:41.45 | Hmmhesays | [TK]D-Fender: yes presence is what I want |
| 21:41.47 | [TK]D-Fender | [hC], No. |
| 21:41.48 | Hmmhesays | any polycom? |
| 21:41.55 | [hC] | [TK]D-Fender: no you havent, or not its not worth trying. |
| 21:41.55 | *** part/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 21:41.59 | [TK]D-Fender | Hmmhesays, depends how many phones yuo want to watch |
| 21:42.08 | [TK]D-Fender | [hC], YES :) |
| 21:42.14 | [hC] | [TK]D-Fender: I HATE YOU!!! |
| 21:42.15 | [hC] | hahaha |
| 21:42.22 | Strom_C | ah, Polycom, where "poly" means "many" and "com" means "those disgusting Soviet bastards" |
| 21:42.44 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 21:42.59 | Hmmhesays | [TK]D-Fender: 5 or 6 |
| 21:43.02 | Dan0maN_Work | so you're saying polycom is the way to go?!? ;) |
| 21:43.07 | Katty | mew? |
| 21:43.09 | [TK]D-Fender | Hmmhesays, then any will do |
| 21:43.33 | Hmmhesays | how many does a 501 support? |
| 21:43.37 | Katty | 3 |
| 21:43.45 | [TK]D-Fender | Hmmhesays, at least 8 |
| 21:43.50 | Katty | 3 lines :P |
| 21:44.01 | [TK]D-Fender | Katty, perhaps ;) |
| 21:44.04 | Hmmhesays | Katty: I'm looking for bla/presence indicators |
| 21:44.16 | Hmmhesays | [TK]D-Fender: you can make it subscribe to 8 different extensions? |
| 21:44.36 | Katty | ^_- |
| 21:44.39 | Katty | 3 |
| 21:45.06 | Katty | there's only 3 lines |
| 21:45.16 | Katty | with 2 server enteries per line |
| 21:45.18 | Dan0maN_Work | looks like to swapped back to this window just in time to join this one. my pres, who's driving me to test asterisk out, suggested i look at the aastra 480i. i'm also ordering the polycom 330 and 430. anyone know of any problems with the 480i? |
| 21:45.51 | Dan0maN_Work | i think he just liked the looks ;) |
| 21:46.03 | Strom_C | Dan0maN_Work: I own a 480i and a polycom. The 480i is gathering dust in the drawer and the Polycom is happily in service. |
| 21:46.11 | Dan0maN_Work | lol |
| 21:46.20 | Dan0maN_Work | lack of the features? problems? |
| 21:46.22 | Hmmhesays | Katty: I don't think that has to do with subscribing to hint extensions |
| 21:46.37 | [TK]D-Fender | Hmmhesays, yes |
| 21:46.37 | DrukenLPY | Strom_C: want to pass on the 480i?? :) i wouldn't mind trying it out |
| 21:46.39 | Katty | k'then, i'll just hush up (= |
| 21:47.17 | Hmmhesays | [TK]D-Fender: can you tell me where on the web interface I configure the phone to subscribe to a hint extension? |
| 21:47.35 | [TK]D-Fender | Hmmhesays, first, you should know better than to touch that at all. |
| 21:47.43 | Hmmhesays | [TK]D-Fender: yes yes |
| 21:47.46 | Hmmhesays | but I want to test it |
| 21:47.56 | [TK]D-Fender | Hmmhesays, Second you need to enable Presence in your provisioning. You then just add them to your directory & enable Buddy Watch |
| 21:48.10 | Strom_C | DrukenLPY: it's useful for those rare occasions where a client is actually using them and I need to lab something up |
| 21:48.22 | Hmmhesays | via tftp right? |
| 21:48.23 | Strom_C | but other than that, I really do prefer my polycom |
| 21:48.42 | [TK]D-Fender | Hmmhesays, FTP is my preference |
| 21:48.51 | Dan0maN_Work | Strom_C: lack of the features? problems? |
| 21:49.04 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:49.09 | Hmmhesays | is there an example of this on the wiki? |
| 21:49.16 | DrukenLPY | Strom_C: ahh.. :) i use the aastra 9112i i like them.... |
| 21:49.23 | Strom_C | Dan0maN_Work: it just feels like a cheaper phone than the polycom |
| 21:49.32 | Dan0maN_Work | that's what i wanted to hear |
| 21:49.33 | Dan0maN_Work | thanks |
| 21:49.40 | Strom_C | now, granted, analog-wise, the Aastra 9147CW is an awesome phone |
| 21:50.10 | [TK]D-Fender | Hmmhesays, yup |
| 21:50.35 | DrukenLPY | Strom_C: 9147CW or 9714CW ? |
| 21:50.39 | [TK]D-Fender | aastra = waste. |
| 21:50.42 | Strom_C | 9417CW |
| 21:50.50 | Strom_C | er |
| 21:50.50 | Hmmhesays | I know some people that would differ |
| 21:50.54 | DrukenLPY | er, yeah that's it |
| 21:51.00 | Strom_C | yeah |
| 21:51.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:51.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:51.17 | DrukenLPY | i like the 9112i cause it looks and feels like the 9417 |
| 21:51.33 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:51.48 | Katty | should i see transformers, harry potter, or pirates tonight? |
| 21:51.53 | Dan0maN_Work | other phone that was suggested was grandstream gxp-200 for a cheaper model, but it looks to be about as much as the polycom. |
| 21:51.59 | Strom_C | Katty: rent Brazil |
| 21:52.01 | Dan0maN_Work | transformers |
| 21:52.02 | DrukenLPY | and well, if my 62 year old man can figure out how to use it... anyone can |
| 21:52.09 | Strom_C | Dan0maN_Work: RUN AWAY |
| 21:52.12 | Katty | Strom_C: we're going to the theator. |
| 21:52.16 | Strom_C | run far far far away from grandstream |
| 21:52.20 | Strom_C | Katty: oh |
| 21:52.22 | Strom_C | well then |
| 21:52.23 | Dan0maN_Work | roger that |
| 21:52.23 | DrukenLPY | stay as far away from grandstream as possible |
| 21:52.30 | Qwell[] | ~gs |
| 21:52.31 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:52.32 | *** join/#asterisk swatchy (n=chatzill@dslb-088-076-061-155.pools.arcor-ip.net) |
| 21:52.40 | Strom_C | the best tool for working on a grandstream is a sledgehammer |
| 21:52.58 | DrukenLPY | 2 tonne bessy |
| 21:53.00 | Dan0maN_Work | again, thanks for the input |
| 21:53.25 | lesouvage | Strom_C: what about the Snom 320, reday for sledgehammer too? |
| 21:53.33 | Strom_C | ehhhhhhhhhhhh, snom is so-so |
| 21:53.38 | Strom_C | but i dont like the feel |
| 21:53.52 | Hmmhesays | is there any sidecar type addon for poly's? |
| 21:53.57 | Strom_C | Hmmhesays: yes |
| 21:54.07 | Strom_C | now, if there was a Snom phone called the Carlson, then I'd snap it up |
| 21:54.10 | Strom_C | otherwise, no :) |
| 21:54.52 | Hmmhesays | it says in the wiki polys can't watch more than 7 buddies |
| 21:55.00 | Strom_C | that's old info |
| 21:55.16 | Strom_C | the wiki also has things that are NEW NEW NEW as of asterisk 1.0.4 |
| 21:55.18 | lesouvage | Strom_C: what phone would you shoose if it has to support POE and has to have 2 ethernet ports that can be configured on different subnets? |
| 21:55.30 | Strom_C | lesouvage: different subnets? |
| 21:55.49 | Strom_C | *shrug* |
| 21:55.55 | Strom_C | the ip330 might do it |
| 21:56.12 | Dan0maN_Work | it did say it supported 802.1p/Q |
| 21:56.16 | Dan0maN_Work | the ip330 that is |
| 21:56.25 | Dan0maN_Work | one of my tests i'm going to do |
| 21:56.37 | lesouvage | Strom_C: yes different subnets, so the voice doesn't interfere with the data. |
| 21:57.06 | Katty | looks like i'mma go see transfomers |
| 21:57.09 | Katty | later gaters (= |
| 21:57.46 | *** join/#asterisk tako-san (n=Tako-san@24.108.162.254) |
| 21:58.37 | digimania | is it possible to ftp a text file to my asterisk box and have flite (or some tts prog) convert it to a voicemail? If so, is there a tutorial somewhere? |
| 22:03.32 | Hmmhesays | something with a color display would be cool |
| 22:04.14 | pigpen | Hmmhesays, fyi, I have several watching over 30 |
| 22:05.23 | Hmmhesays | pigpen: the 601's with teh sidecar? |
| 22:05.40 | pigpen | 601's with several side cars. |
| 22:05.45 | Hmmhesays | cool |
| 22:05.49 | Hmmhesays | thats exactly what I need |
| 22:05.53 | pigpen | each handles 14 |
| 22:06.17 | pigpen | the phone itself with do 5 (with a single registration) |
| 22:07.27 | j-goddess | meep |
| 22:07.31 | sevard | moop |
| 22:08.05 | blitzrage | miip |
| 22:09.10 | sevard | annnnd we're back. |
| 22:09.45 | pigpen | Hmmhesays, beware, buddy watch is a bit different in Asterisk 1.4. |
| 22:11.41 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:12.23 | DrukenLPY | wuts buddy watch ? |
| 22:12.49 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 22:12.55 | saint_ | hi all |
| 22:13.18 | Hmmhesays | pigpen, how so? |
| 22:15.05 | saint_ | anyone here has an asterisk connected to an Alcatel PBX by any chance ? |
| 22:17.42 | *** part/#asterisk galeras (n=galeras@201.244.199.31) |
| 22:21.20 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:21.33 | sevard | http://www.davidshrigley.com/images/photo_pics/notice.jpg |
| 22:21.36 | sevard | :| |
| 22:21.39 | syzygyBSD | what does NOP status mean on PRI? |
| 22:24.29 | riddlebox | does asterisk/ and voip in general work well with dsl? |
| 22:24.45 | *** join/#asterisk HomeyG (n=r@74-34-2-187.dsl1-merch.roc.ny.frontiernet.net) |
| 22:24.46 | HomeyG | HI |
| 22:24.51 | HomeyG | what is this |
| 22:24.56 | HomeyG | how can it benefit me |
| 22:25.15 | sevard | HomeyG: space age telephone system. |
| 22:25.22 | sevard | HomeyG: it can do everything except fry you bacon |
| 22:25.26 | HomeyG | hmm |
| 22:25.29 | sevard | well, it could probably fry you bacon. |
| 22:25.32 | HomeyG | how can it work on my BSD box |
| 22:25.44 | HomeyG | and what do I need for it to work |
| 22:25.57 | sevard | http://www.oreillynet.com/pub/a/network/2005/09/30/what-is-asterisk.html |
| 22:26.25 | sevard | also |
| 22:26.27 | sevard | ~thebook |
| 22:26.28 | 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 |
| 22:27.52 | HomeyG | so |
| 22:27.54 | HomeyG | how can it benefit me |
| 22:28.56 | syzygyBSD | it cleans your bathroom HomeyG |
| 22:28.59 | sevard | reduced costs, greater flexibility, and it expands your mind in a similar method to LSD |
| 22:29.03 | syzygyBSD | its really swell |
| 22:29.18 | sevard | syzygyBSD: it's |
| 22:29.19 | HomeyG | seropis;y |
| 22:29.21 | HomeyG | seriously |
| 22:29.22 | HomeyG | heh |
| 22:29.24 | sevard | seriously. |
| 22:29.25 | HomeyG | its a virtual PBX |
| 22:29.30 | HomeyG | what happens If dont use virtual ip |
| 22:29.32 | HomeyG | voiop |
| 22:29.41 | sevard | you may use analog or digital lines. |
| 22:29.55 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 22:29.57 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:30.12 | sevard | HomeyG: are you considering the use of this in a business setting or are you just a curious hacker? |
| 22:30.56 | Hmmhesays | it sure is good at giving hummers |
| 22:32.30 | HomeyG | in a businses setting |
| 22:32.55 | Hmmhesays | Hire someone to help you |
| 22:33.03 | HomeyG | wHY |
| 22:33.15 | Hmmhesays | So you get a second job |
| 22:33.17 | Hmmhesays | lol |
| 22:33.29 | HomeyG | how can I set this up |
| 22:33.33 | HomeyG | on my FREEBSD machine |
| 22:33.39 | HomeyG | and how would it work with my existing analog lines? |
| 22:36.28 | geoff_k | i'd get 2 peices of string and label them Dot & Dash respectivly, possibly have hi and low pitched bells on the end do some kind of morse codethen use vistas voice recognition to decode it on the otherside, itsonly any use for internal stuff though, cross continent is a bit of a mission |
| 22:38.18 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 22:40.31 | Hmmhesays | HomeyG did you hang out in #fark? |
| 22:41.25 | HomeyG | YES! |
| 22:41.33 | Hmmhesays | hey mang how goes it? |
| 22:42.05 | HomeyG | not bad |
| 22:42.05 | HomeyG | ytou |
| 22:42.06 | HomeyG | you |
| 22:42.12 | Hmmhesays | same |
| 22:42.14 | Hmmhesays | doing the voip thing |
| 22:42.26 | *** join/#asterisk pusanggala (n=a@58.69.243.203) |
| 22:42.33 | Hmmhesays | course i've been hanging out here as long as #fark |
| 22:44.26 | dlynes_laptop | HomeyG: Last time I used it on FreeBSD, analog didn't work very well, if at all |
| 22:44.39 | dlynes_laptop | HomeyG: But, there's been significant development on the FreeBSD drivers since then |
| 22:44.42 | sevard | dlynes! |
| 22:44.49 | dlynes_laptop | HomeyG: So, there's a good chance it's stable on FreeBSD by now |
| 22:44.55 | dlynes_laptop | sevard: hey bitch...how's it going? |
| 22:45.02 | sevard | not bad, wazzzup dawg |
| 22:45.06 | Hmmhesays | finally these dynamic tables are done |
| 22:45.12 | Hmmhesays | my super awesome address book is on its way |
| 22:46.09 | dlynes_laptop | sevard: Just super busy lately |
| 22:46.13 | *** join/#asterisk ManxPower (n=manxpowe@015-844-184.area5.spcsdns.net) |
| 22:46.16 | dlynes_laptop | sevard: getting married probably in december |
| 22:46.24 | sevard | Yeah, I've been pretty busy lately as well, with your mom. |
| 22:46.32 | sevard | dlynes_laptop: am I *#&%ing best man, or what? |
| 22:47.06 | dlynes_laptop | sevard: not a chance, with language like that :p |
| 22:47.16 | sevard | lame. |
| 22:47.36 | sevard | at least put up an open bar and invite Hmmhesays and me. |
| 22:48.00 | sevard | we'll throw you a pretty good bachelor party |
| 22:48.55 | J4k3 | haha |
| 22:50.57 | syzygyBSD | what does NOP status mean for a PRI? |
| 22:51.24 | Nugget | I don't know, but it probably means the same thing it meant when you asked 30 minutes ago. |
| 22:51.36 | syzygyBSD | hmm, I don't know.. it might change |
| 22:51.50 | syzygyBSD | it is telecom and all |
| 22:52.53 | Nugget | heh |
| 22:53.38 | syzygyBSD | I just have never seen a NOP, I know green red and yellow.. they are all colors, that makes sense, but what color is NOP? |
| 22:54.38 | syzygyBSD | ahh, Not-OPerational |
| 22:56.22 | Innatech | Not Our Problem. |
| 22:56.24 | Innatech | heh. |
| 22:56.29 | syzygyBSD | LOL |
| 22:56.48 | syzygyBSD | hmm, so it is the person that is "upgrading" the system |
| 23:00.01 | *** join/#asterisk rhombus (i=user239@74.12.124.179) |
| 23:00.14 | rhombus | hello |
| 23:00.54 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 23:01.15 | *** join/#asterisk Grapsus (n=IceChat7@135.224.100-84.rev.gaoland.net) |
| 23:01.21 | Grapsus | Hello ! |
| 23:01.31 | syzygyBSD | Hello? |
| 23:02.32 | Grapsus | I'm new to Asterisk (it's really impressive) and I have few questions aboit res_mysql |
| 23:03.53 | russellb | use odbc, it's better supported by the dev team. |
| 23:04.01 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 23:04.15 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 23:04.27 | *** part/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 23:04.50 | Grapsus | I'm trying to fully replace sip.conf with a mysql database, it works with all my users, but how do I enter a line like "register =>" in the table ? there's no "register" field |
| 23:05.10 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.142) |
| 23:07.57 | Grapsus | (so it's possible to connect mysql through odbc ?) |
| 23:08.28 | Nugget | That's sort of what the "O" signifies. |
| 23:08.49 | Nugget | with odbc you're free to use any database, even a shitty one like mysql. ;) |
| 23:09.22 | Grapsus | ok, and how can I include register lines to my database ? |
| 23:09.42 | Nugget | I have no idea, sorry. |
| 23:09.44 | Grapsus | (no matter if it's not relatime, I just want all in mt database) |
| 23:12.03 | ManxPower | Using databases with Asterisk for config stuff is mostly an Enterprise Issue |
| 23:12.08 | ManxPower | or ITSP, of course |
| 23:13.10 | Grapsus | Yes, in fact I'm wrtinig a PHP interface for configuration |
| 23:13.33 | Grapsus | I've managed to put all in the database, excepted there "register => ..." lines |
| 23:15.54 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 23:16.40 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 23:17.24 | Grapsus | found it http://www.voip-info.org/wiki/view/Asterisk+sip+conf+from+mysql ! |
| 23:17.57 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:18.38 | jameswf | do the telemarketer torture soundfiles still exist anywhere |
| 23:19.13 | JT | syzygyBSD: you from the us? |
| 23:22.44 | syzygyBSD | for now |
| 23:23.46 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 23:25.11 | tzafrir_laptop | jameswf, do you refer to the tt-* sounds? |
| 23:25.14 | JT | syzygyBSD: did you ever live in australia? |
| 23:25.21 | syzygyBSD | NZ |
| 23:25.27 | JT | hrm |
| 23:25.30 | syzygyBSD | going back there soon |
| 23:25.39 | JT | just wondering if i know you from elsewhere on irc |
| 23:25.41 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-40-204.dsl.tul2ok.sbcglobal.net) |
| 23:25.49 | syzygyBSD | possibly |
| 23:26.11 | syzygyBSD | had this nick for 6 years or so |
| 23:27.05 | *** join/#asterisk hi365_m (i=HydraIRC@cablep-219-62-26.cablep.bezeqint.net) |
| 23:27.19 | hi365_m | does txgain/rxgain also take a precentage? i.e. txgain=15% ? |
| 23:31.05 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:34.26 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:35.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:35.15 | *** join/#asterisk Mad|Cow (n=madcowl@74.95.181.237) |
| 23:41.46 | tzafrir_laptop | hi365_m, it's a floating-point number |
| 23:41.55 | *** join/#asterisk sysreq (n=sysreq@209.44.112.102) |
| 23:42.04 | tzafrir_laptop | 0.15 should work |
| 23:42.38 | hi365_m | tzafrir_laptop: but wont that be much less than 15 db? (i.e. wont that mean .15 db) |
| 23:43.03 | tzafrir_laptop | right. 0.15db. |
| 23:43.24 | tzafrir_laptop | set it to 15 for 15db (a huge value) |
| 23:43.30 | tzafrir_laptop | huge gain? |
| 23:44.01 | hi365_m | tzafrir_laptop: duno. got these gsm-fxo things from orange and they really suck |
| 23:44.05 | tzafrir_laptop | Why would you need such values? |
| 23:44.14 | hi365_m | ^^ |
| 23:44.21 | tzafrir_laptop | hi365_m, which fxo adapter? |
| 23:44.41 | hi365_m | cellulink? im not 100% sure |
| 23:44.59 | tzafrir_laptop | this is what you connect to asterisk? |
| 23:45.03 | JT | syzygyBSD: ever use austnet? |
| 23:45.24 | hi365_m | tzafrir_laptop: from my cell |
| 23:45.40 | hi365_m | i.e. cell-cell/fxo-fxo/asterisk |
| 23:45.53 | tzafrir_laptop | huh? |
| 23:45.58 | tzafrir_laptop | fxo-fxo? |
| 23:46.09 | tzafrir_laptop | what is fxo/asterisk ? |
| 23:46.21 | hi365_m | its a gsm to fxo thing. used to add a sim card to your pbx |
| 23:46.41 | hi365_m | <tzafrir_laptop> what is fxo/asterisk ? - the fxo card in the asterisk server |
| 23:46.58 | tzafrir_laptop | it provides you an FXO interface? so you cannot connect a standard phone to it? |
| 23:47.11 | rhombus | any north american BRI users? |
| 23:47.14 | rhombus | here, I mean? |
| 23:47.34 | ManxPower | rhombus: I don't know if there are any north american BRI users *anywhere* |
| 23:47.44 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 23:48.00 | tzafrir_laptop | hi365_m, what fxo card on asterisk? |
| 23:48.00 | ManxPower | Generally cell adapters provide and FXS interface |
| 23:48.03 | ManxPower | ~fxofxs |
| 23:48.04 | jbot | rumour has it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 23:48.19 | rhombus | ManxPower: what about users who would use a BRI if they had hardware? |
| 23:48.49 | ManxPower | rhombus: As I understand it, they are out of luck |
| 23:48.54 | hi365_m | tzafrir_laptop: just the oposite. because its provides you with an fxo port (i.e. a "line") you cna connect a phone to it |
| 23:48.55 | tzafrir_laptop | rhombus, hardware is cheap: HFC-s or AVM single-port cards are easy to get |
| 23:49.05 | ManxPower | the Digium BRI card might support North American ISDN. I don't know one way or the other. |
| 23:49.05 | hi365_m | tzafrir_laptop: a200 |
| 23:49.13 | hi365_m | tzafrir_laptop: sangoma a200 |
| 23:49.21 | ManxPower | none of the other BRI cards that I am aware of support North American BRI. |
| 23:49.29 | rhombus | The Digium BRI does not |
| 23:49.49 | rhombus | well, it does not support NAm BRI |
| 23:49.53 | rhombus | here's the thing |
| 23:50.01 | ManxPower | Seems rather silly since Digium IS a North American company |
| 23:50.09 | tzafrir_laptop | I don't know the A200. But at least on Digium card (and on ours) you can set the FXO gain also on the analog->digital conversion |
| 23:50.28 | JT | tzafrir_laptop: no, the HFC-s does NOT work with nationalisdn/etc |
| 23:50.28 | rhombus | ManxPower: well, Digium suffers from the same prejudice that many other people seem to -- "there's no BRI in North America" |
| 23:50.44 | andresmujica | Hi, anyones knows if there's already a calendar/agenda management IVR system with asterisk? i mean, i call to the IVR and the system creates an appointment automagically?? |
| 23:50.50 | ManxPower | rhombus: Well there isn't! 8-) |
| 23:50.50 | tzafrir_laptop | I suspect this is a bit more reliable than boosting gain to the sigital stream. Though there's still a limit to what you can get |
| 23:50.55 | JT | tzafrir_laptop: the digium B410P does NOT work with nationalisdn/etc |
| 23:50.56 | rhombus | the truth is that ALL of these cards would work if somebody would provide a stack for them |
| 23:51.04 | rhombus | ManxPower: oh, but there IS |
| 23:51.04 | tzafrir_laptop | JT, this is slightly incorrect |
| 23:51.11 | JT | tzafrir_laptop: not with current drivers |
| 23:51.13 | ManxPower | rhombus: ISDN in North America is way over priced (except for in TN) |
| 23:51.19 | JT | no-one has tested it, anyway, tzafrir_laptop |
| 23:51.21 | hi365_m | tzafrir_laptop: what do you mean by the analog-> digital conversion? what is the setting? |
| 23:51.22 | tzafrir_laptop | working with that is more a matter of ISDN stack |
| 23:51.27 | rhombus | ManxPower: what are they charging in your corner of the woods? |
| 23:51.35 | JT | tzafrir_laptop: right, so for practical purposes it doesn't work |
| 23:51.39 | ManxPower | rhombus: I know. I used to run an ISP back before DSL. |
| 23:51.49 | ManxPower | rhombus: 2B+D would run about $100/month |
| 23:51.50 | tzafrir_laptop | If Digium's card works with it, then mISDN supports it. mISDN also has a driver for HFC-s |
| 23:52.05 | rhombus | Yeah, but we're talking about voice BRI... it makes no sense for data |
| 23:52.34 | JT | tzafrir_laptop: digium's card is not reported to work with it |
| 23:52.34 | rhombus | ManxPower: $100/month is very competitive -- at least, it is in these parts. Do you know what I pay for a single analog line? |
| 23:52.51 | *** join/#asterisk gzero (n=gzero@81.175.82.2) [NETSPLIT VICTIM] |
| 23:52.55 | ManxPower | rhombus: BRI is like $20/month in much of Europe. |
| 23:53.14 | tzafrir_laptop | so let's go back to the basics: anybody tried it with zapbri? |
| 23:53.21 | tzafrir_laptop | zaptel does support national |
| 23:53.25 | rhombus | ManxPower: I'll reserve judgement on that until I've seen the evidence first hand |
| 23:53.26 | JT | BRI is around USD$50 in Australia |
| 23:53.26 | tzafrir_laptop | chan_zap, that is |
| 23:53.36 | rhombus | JT: 2B+D? |
| 23:53.39 | JT | yes |
| 23:53.54 | JT | it's $63 AUD for the cheapest rental |
| 23:53.54 | tzafrir_laptop | HFC-s will work just as well with zap/bri (bristuff) |
| 23:53.59 | rhombus | Well, that's because they're selling it to residential customers (also in Europe) |
| 23:54.14 | rhombus | but in North America it's interesting for a business |
| 23:54.16 | ManxPower | rhombus: the thing is, with the the state of BRI support for Asterisk for north america, it just is not worth the weeks it takes to make it work |
| 23:54.18 | JT | rhombus: hardly any residential BRI customers in .au, mostly business |
| 23:54.23 | ManxPower | unless your time is free. |
| 23:54.38 | rhombus | ManxPower: Point taken -- but I'm actually trying... well |
| 23:54.43 | *** part/#asterisk hi365_m (i=HydraIRC@cablep-219-62-26.cablep.bezeqint.net) |
| 23:54.54 | rhombus | There's a rumour that somebody is taking pre-orders for the Sangoma A500 for NAm |
| 23:55.02 | JT | so it's around AUD$33/mo/channel |
| 23:55.13 | JT | i'll wait till A500 drivers are actually released |
| 23:55.29 | JT | but bri is not that competitive compared to Optus PRI service |
| 23:55.33 | JT | AUD$20/mo/ch |
| 23:55.38 | JT | USD$15 or so |
| 23:55.43 | rhombus | JT: You don't need to wait, you guys use EuroISDN in Oz, don't you? |
| 23:55.57 | [hC] | anyone know how i might get around this caller id issue: I want to suppress all caller id when passing calls to my pri, but every time i try to send "no caller id" my PRI overrides it and sticks in the pilot number |
| 23:56.04 | JT | rhombus: we do, but A500 drivers i don't think have even been released in beta |
| 23:56.20 | JT | it's a new card |
| 23:56.23 | JT | june announce |
| 23:56.23 | rhombus | JT: I have it on good authority that they are in the pipeline and only weeks away |
| 23:56.28 | ManxPower | [hC]: "show application setcallingpres" |
| 23:56.37 | JT | rhombus: yes, apparently they'll be using chan_woomera |
| 23:56.48 | JT | not sure how well that will work |
| 23:57.05 | rhombus | JT: that won't be the only channel driver it supports |
| 23:57.07 | ManxPower | rhombus: is this the same authority that said Sangoma was releasing a DS3 card with Zaptel drivers? |
| 23:57.14 | JT | rhombus: beta drivers aren't very good for production |
| 23:57.18 | rhombus | ManxPower: no, it is not |
| 23:57.18 | JT | rhombus: are you sure? |
| 23:57.31 | rhombus | JT: If I say any more I will be shot. |
| 23:57.59 | JT | rhombus: is there a possibility of a magic mexican with a z involved? |
| 23:58.18 | rhombus | JT: I don't follow :) |
| 23:58.24 | [hC] | ManxPower: is that a 1.4 thing? |
| 23:58.35 | ManxPower | [hC]: no. |
| 23:58.38 | tzafrir_laptop | rhombus, if you want to experiment, just get a simple HFC-s card. Try zapbri |
| 23:58.40 | [hC] | oh, callerpres. not callingpres |
| 23:58.55 | rhombus | tzafrir_laptop: I'm not a coder, and I'm not looking to experiment |
| 23:59.18 | JT | rhombus: are you being a smartarse, or you seriously don't know the mexican i'm refering to? :) |
| 23:59.20 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-afaccf8ff28f5b7c) |
| 23:59.44 | rhombus | JT: Zapata? |
| 23:59.52 | ManxPower | JT: I think most people don't know who Zapata is. |
