| 00:00.06 | nick125_lappy | anyone here know of a change log (or something kind of condensed) that shows the major changes between 1.2.x and 1.4.0? |
| 00:00.22 | Corydon-w | Scalability should probably be handled in the database itself, in this case |
| 00:00.31 | blitzrage | fair enough |
| 00:00.39 | Corydon-w | Remember, the database is optimized to return those results quickly |
| 00:00.48 | Corydon-w | (or should be, anyway) |
| 00:01.36 | Corydon-w | nick125_lappy: UPGRADE.txt |
| 00:03.14 | *** join/#asterisk wiljacket (n=wilson@cpe-76-173-243-4.socal.res.rr.com) |
| 00:09.23 | variable_office | would it be less expensive to get a ds1 at a colo or purchase 24 voip channels from a reseller/ |
| 00:09.26 | variable_office | ? |
| 00:11.03 | x86 | variable_office: if you can offset the cost, get a PRI |
| 00:11.49 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:11.51 | variable_office | is a pri in colo going to be ALOT cheaper than to my office? ie. do they charge loop at a colo? |
| 00:12.37 | nick125_lappy | variable_office: one issue you might want to worry about is if you are trying to get voip to your office, you'll need bandwidth at the colo, then, bandwidth at your office |
| 00:13.19 | variable_office | ya, but if i can cut out a $400/month loop it may offset the colo fee? |
| 00:13.43 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 00:13.58 | nick125_lappy | But, you would still have to get decent bandwidth at your office, I'm just saying, its something to think about. |
| 00:14.04 | variable_office | and copper here is unreliable at best, i have nxt1 just to keep bandwidth flowing, to my knowledge there is no equivalent of nxt1 for voice? |
| 00:14.55 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 00:15.12 | monsted | your intarweb connection is unreliable or the PRI is? |
| 00:15.48 | variable_office | anything over copper is, so both, i dont have a pri now, but i fear what would happen when the line cuts out. |
| 00:16.23 | variable_office | thats why i have multiple bonded t1s, so if one goes down at least the whole network doesnt go down |
| 00:17.02 | monsted | that's gotta suck |
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| 00:17.27 | variable_office | yep, thats why i curse every att truck i see |
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| 00:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:18.19 | variable_office | att is the worst company in existance imo; they are right up there with the gov |
| 00:18.25 | monsted | well, you could do multiple PRIs, but that might be a little expensive ;) |
| 00:19.11 | variable_office | if one pri went down would the calls fail over to the remaining line/ |
| 00:19.13 | variable_office | ? |
| 00:19.38 | monsted | if you get it set up correctly, yes |
| 00:20.39 | nick125_lappy | Meh, I'm getting confused on how I should setup my home asterisk system |
| 00:20.48 | nick125_lappy | So many features to add, so little time! |
| 00:21.03 | monsted | we do redundant PRIs to two data centers, each with redundant voice gateways, redundant call managers, redundant firewalls, routers, MPLS connections and so on :) |
| 00:21.25 | variable_office | good deal |
| 00:21.41 | monsted | ofcourse most customers only want to pay for one call manager, one PRI, one fibre and one firewall, but we *can* :) |
| 00:22.58 | monsted | and when some twit digs through our fibers, it almost doesn't break ;) |
| 00:22.58 | nick125_lappy | Did I ever say how stupid I think fedex is? They are shipping my package from SF, CA to Memphis, TN, then, its going to go back to albuquerque, nm |
| 00:22.58 | variable_office | monsted any ideas on why dtmf wont work in any way, be it rfc2833, inband, info, past asterisk on a outgoing sip line? |
| 00:22.58 | monsted | variable_office: none at all |
| 00:23.09 | monsted | it's not just your upstream provider that strips it off? |
| 00:23.19 | variable_office | ya, me neither |
| 00:23.31 | variable_office | dont know, i have no other upstream to try |
| 00:23.40 | JT | variable_office: what codec? |
| 00:24.14 | variable_office | ulaw |
| 00:24.52 | JT | hrm |
| 00:25.00 | JT | probably an issue with your upstream |
| 00:25.05 | *** join/#asterisk CrashHD (i=CrashHD@67.182.167.222) |
| 00:27.27 | variable_office | ya, and its driving me crazy because i cant confirm if its them or not because i have no second upstream to try |
| 00:28.18 | CrashHD | variable_office you get my message earlier? |
| 00:28.21 | CrashHD | about logger.conf? |
| 00:28.27 | JT | so setup another asterisk box or something, or try a free sip provider |
| 00:28.30 | variable_office | nope, i didnt see it |
| 00:28.33 | JT | or some random person with sip |
| 00:28.44 | CrashHD | in your logger.conf add dtmf to console => |
| 00:28.49 | CrashHD | should give you some dtmf debugging info |
| 00:30.22 | variable_office | CrashHD cool thanks, well asterisk is definatly receiving dtmf then |
| 00:31.52 | JT | setup an FWD account or something |
| 00:32.02 | JT | but i'd say it's your provider |
| 00:33.44 | CrashHD | after you upgrade....that would be my guess as well |
| 00:34.14 | JT | upgrade from what? |
| 00:34.24 | CrashHD | he's running 1.2.9.1 last I checked |
| 00:34.26 | variable_office | this evening then ill begin to work on trying that |
| 00:34.35 | variable_office | but i also ran 1.4 and it did the same |
| 00:34.51 | JT | if it's production, you should avoid 1.4 |
| 00:35.06 | variable_office | its not, but i wont be running 1.4 anyways |
| 00:35.18 | variable_office | i just tried it to see if it fixed my dtmf issue |
| 00:36.08 | JT | if you set it to rfc2833, you should see it on the SIP port outgoing if you packet snif |
| 00:36.09 | CrashHD | JT, 1.4 is pretty stable |
| 00:36.11 | CrashHD | any issues you have? |
| 00:36.21 | variable_office | bbl |
| 00:36.34 | file | rfc2833 sends over RTP stream |
| 00:36.42 | JT | CrashHD: even so, general wisdom is to not use it in production |
| 00:36.53 | CrashHD | wisdom is relative I suppose |
| 00:36.55 | Hmmhesays | god I hate tommy tutone |
| 00:36.58 | JT | file: ah really? |
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| 00:40.28 | file | JT: yes really |
| 00:40.49 | JT | is it INFO that sends it over SIP then? |
| 00:41.14 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 00:41.41 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 00:41.52 | *** part/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:41.55 | robin_sz | so, mISDN gurus? |
| 00:41.58 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:42.36 | JT | heh, what's happening with misdn? |
| 00:44.24 | *** join/#asterisk dhpeterson (n=dhpeters@203.192.146.185) |
| 00:48.10 | Hmmhesays | ahh my directory mod works |
| 00:48.12 | Hmmhesays | thats cool |
| 00:51.13 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:52.11 | *** join/#asterisk Strom_M (n=strom@dsl092-221-173.lax1.dsl.speakeasy.net) |
| 00:53.09 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:57.46 | blitzrage | does ChanIsAvail() return all available channels, or just the first one? |
| 00:58.46 | *** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk) |
| 01:01.15 | JT | robin_sz: what's the misdn problem? |
| 01:01.51 | flenders | JT: mate, remember we talked about switching to ISDN yesterday? |
| 01:02.02 | JT | yeah |
| 01:02.09 | flenders | JT: what sort of card would I need? a TE110P? |
| 01:02.54 | JT | for a PRI, yes |
| 01:03.40 | JT | but for a bri, you will need either one of the many cheap single port HFC cards, or a multiport card from junghanns.net or berosnet, or the new bri card from digium that is pretty new |
| 01:03.56 | brian | whats a bri |
| 01:04.05 | JT | basic rate interface |
| 01:04.16 | brian | and what is it |
| 01:04.17 | brian | :P |
| 01:04.18 | JT | 2B + 1D (D @ 16kbit/s) |
| 01:04.29 | JT | digital telephony circuit |
| 01:04.33 | flenders | what's the difference between PRI and BRI? |
| 01:04.42 | brian | Yeah, that's what I meant! |
| 01:04.43 | brian | That one! |
| 01:04.44 | flenders | can you get both over here? |
| 01:04.58 | JT | PRI can have up to 30 channels in the E1 variant, 23 in the PRI T1 variant |
| 01:05.02 | JT | BRI has 2 channels |
| 01:05.03 | JT | yes |
| 01:05.16 | JT | we use E1 pris in .au, as well as BRIs |
| 01:05.20 | flenders | JT: is that what telstra sells as ISDN2? |
| 01:05.25 | JT | telstra calls them ISDN30s and ISDN2s |
| 01:05.26 | JT | yes |
| 01:05.47 | JT | the PRI is available in fractional forms here of an ISDN10 and ISDN20, too |
| 01:05.50 | brian | Sounds like you need lots of BRI to do anything useful. |
| 01:05.53 | JT | with 10 and 20 channels, respectively |
| 01:06.05 | JT | brian: rubbish, it's a lot more useful than POTS |
| 01:06.14 | JT | not everyone needs that many lines |
| 01:06.19 | JT | it's like a small PRI |
| 01:06.34 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 01:06.36 | brian | how many channels does PRI have |
| 01:06.44 | JT | and it puts 2 lines over the copper that previously held one line for POTS |
| 01:06.54 | CrashHD | 23+1 |
| 01:06.57 | JT | scroll up, i already mentioned it |
| 01:07.00 | *** join/#asterisk azidenth (n=aby_azid@60.50.220.139) |
| 01:07.13 | JT | 30 channels for PRI E1, 23 channels for PRI T1 |
| 01:07.34 | bkruse_home | b chans |
| 01:07.48 | JT | b channels/usable traffic channels :) |
| 01:08.17 | JT | also a 64kbit/s D channel for signalling on each, of course |
| 01:08.19 | flenders | JT: but I can use that TE110P for BRI as well, can't I? |
| 01:08.25 | JT | flenders: no. |
| 01:08.30 | flenders | just in case we end up needing more lines in the future |
| 01:08.45 | JT | it's only good for E1/T1/J1 |
| 01:08.55 | flenders | ok |
| 01:09.15 | JT | if you can justify 10 lines, than definately go for it |
| 01:09.21 | JT | if not, bri it is |
| 01:09.34 | *** join/#asterisk morex (i=morex@host86-133-6-195.range86-133.btcentralplus.com) |
| 01:09.43 | flenders | are these the ones? http://www.junghanns.net/en/ISDNmole_produkt.html |
| 01:09.47 | *** join/#asterisk Mad|Cow (n=madcow@74.92.109.205) |
| 01:10.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:10.30 | JT | no |
| 01:10.35 | JT | quadbri/octobri |
| 01:10.51 | JT | isdnmole is an isdn interception/monitoring device |
| 01:11.01 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 01:11.01 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:12.02 | perd | is there a way to stop asterisk from trying to create a native bridge on sip channels? |
| 01:12.39 | [hC] | perd: you mean, make it stop bridging RTP between clients? |
| 01:12.43 | perd | yeah |
| 01:12.50 | [hC] | perd: reinvite=never in sip.conf |
| 01:12.53 | [hC] | per user |
| 01:12.56 | [hC] | wait |
| 01:12.57 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 01:12.59 | [hC] | that might not be the variable |
| 01:12.59 | [hC] | sec. |
| 01:13.01 | perd | no kidding! damn i wondered what that meant |
| 01:13.03 | perd | hah oh |
| 01:13.10 | perd | canreinvite=no |
| 01:13.14 | [hC] | canreinvite=no |
| 01:13.14 | [hC] | yeah |
| 01:13.19 | perd | i guess that makes sense |
| 01:13.22 | perd | cool thanks man |
| 01:13.24 | [hC] | reinvite is the command that is issued to bridge the RTP sessions |
| 01:13.27 | [hC] | no problem. |
| 01:13.32 | JT | flenders: just to give you an idea, a fractional 10 channel PRI is $305/mo |
| 01:13.43 | perd | now lets see if it fixes my problem... :) |
| 01:13.48 | perd | bingo |
| 01:13.53 | perd | kick ass dude! |
| 01:14.14 | flenders | JT: not much |
| 01:14.25 | CrashHD | here full pri's are about 480 |
| 01:14.26 | flenders | if you think you would pay about 30 each PSTN line |
| 01:14.36 | `Sean | ~cdr |
| 01:14.37 | jbot | i guess cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 01:14.41 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 01:14.47 | [hC] | perd: i usually turn reinvite's off as well as native transfers on iax. it caused a lot of one way audio issues for me with nat. |
| 01:14.53 | JT | CrashHD: full 30chan pri is AUD$885/mo here |
| 01:15.05 | [hC] | perd: or missing audio chunks at the start of a call, or the inverse.. hearing someone for 2-3 seconds, then a click, then one way audio. |
| 01:15.11 | JT | which is probably around USD$600 |
| 01:16.02 | perd | ahh yeah, good to know |
| 01:16.48 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 01:17.19 | JT | flenders: it depends if you go business line plus of businessline complete, but it's around $70 or $80/mo for a BRI |
| 01:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:18.29 | wunderkin | more frequently now too i think |
| 01:18.44 | Qwell | why not? |
| 01:19.18 | wunderkin | sure |
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| 01:20.59 | flenders | JT: would this work? http://www.voipshop.com.au/product_info.php?cPath=30_71&products_id=115 |
| 01:22.08 | JT | yes but you can buy those for closer to $60 overseas, flenders |
| 01:22.26 | flenders | wow! |
| 01:22.30 | flenders | that's cheap |
| 01:22.34 | *** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 01:22.55 | JT | not sure about the fritzcard brand itself, but definately for something like the billion or asus one |
| 01:23.06 | JT | most are just hologne HFC clones |
| 01:23.10 | JT | cologne HFC |
| 01:24.28 | JT | flenders: on the software side, you probably need to look into a modification to asterisk known as bristuff |
| 01:24.39 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 01:24.40 | JT | the only other main option is misdn |
| 01:24.44 | JT | and it's highly alpha |
| 01:24.58 | Sweeper | hmmm |
| 01:25.29 | flenders | is it just a module? |
| 01:25.41 | Sweeper | I want to set up * to automagically call people. can I do this with realtime, or am I gonna have to stick to putting files into /tmp/asterisk/calls/ or whatever? |
| 01:26.02 | JT | flenders: no, it's a patchset or a "spoon" |
| 01:26.15 | JT | patches asterisk, libpri and zaptel |
| 01:26.35 | flenders | I guess it's not much of a hassle to do it then |
| 01:26.40 | flenders | just recompile |
| 01:26.55 | JT | Sweeper: you can use either the manager interface or .call files in /var/spool/asterisk/outgoing/ |
| 01:27.22 | Sweeper | "maanager interface" sounds ominous D: |
| 01:27.51 | JT | yeah it's a port you get programs to connect to and issue commands |
| 01:28.06 | Sweeper | oh, that's not too bad |
| 01:28.19 | Sweeper | I was thinking along the lines of some weird GUI shit |
| 01:28.36 | JT | you can get weird GUI shit that uses the manager interface :) |
| 01:28.59 | JT | flenders: how many lines do you need? |
| 01:29.01 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 01:29.28 | flenders | JT: 6 |
| 01:29.52 | JT | 3 BRIs then |
| 01:30.08 | JT | do you have an existing system in place? |
| 01:30.14 | flenders | I would need a quadBRI then |
| 01:30.21 | flenders | not on the new place |
| 01:30.47 | JT | so i assume you want to use SIP hardphones or something? |
| 01:31.37 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 01:31.42 | flenders | yeah, we're getting those SPA941s |
| 01:31.54 | JT | ah ok |
| 01:32.02 | JT | quadbri then i guess |
| 01:32.10 | *** join/#asterisk tengulre (n=tengulre@61.185.220.249) |
| 01:32.19 | Sweeper | fucking w00t |
| 01:32.25 | Sweeper | RAMI <3 |
| 01:33.01 | JT | expect to pay up to $1300 for the quadbri, after all import costs |
| 01:33.06 | Dr-Linux|home | what's the suggested opensource program for asterisk 2 identical servers redundancy? |
| 01:33.09 | JT | at a guess |
| 01:33.16 | JT | i have an octobri here |
| 01:34.13 | nick125_lappy | noooo, where did iax2 debug go? |
| 01:34.38 | flenders | octo would give you 16 lines, right? |
| 01:35.51 | perd | arg, i'm so close to getting this CCM_+asterisk crap working |
| 01:35.59 | perd | just gotta figure out how to make a call from asterisk -> ccm |
| 01:36.02 | perd | oh baby oh baby |
| 01:36.29 | nick125_lappy | Ugh, 24 hours or so for a FWD IAX2 account |
| 01:36.29 | blitzrage | yah... I had to talk dirty to the CCM to get it working too |
| 01:36.48 | perd | blitz how did you do it? |
| 01:37.02 | perd | i can make calls to the outside world but for some reason the voip router wont let me make calls to extensions :/ |
| 01:37.15 | JT | nick125_lappy: didn't take me 24 hours |
| 01:37.18 | azidenth | good day everyone...in realtime asterisk wat is (err 1045)? |
| 01:37.24 | JT | you sign up, then change the settings to IAX |
| 01:37.31 | JT | it didn't take that long for me. |
| 01:37.34 | azidenth | im trying to connect asterisk to mysql |
| 01:37.48 | Nugget | Well there's your problem. ;) |
| 01:38.13 | blitzrage | perd: depends what version... I wrote a doc for a CCM a while ago... wonder if I still have it |
| 01:40.25 | perd | it's like 3.3 |
| 01:40.26 | perd | old rap |
| 01:41.00 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 01:41.08 | blitzrage | yah... doesn't look like I did anything special |
| 01:41.37 | Nugget | It's tricky to rock a voip to rock a voip that's right on time it's tricky. |
| 01:42.02 | blitzrage | just exten => _X.,1,Dial(SIP/callman01/${EXTEN}) it looks like |
| 01:42.08 | perd | hrm |
| 01:42.16 | Qwell | rock the voip. Don't rock the voip Nugget |
| 01:42.17 | perd | and callman01 is pointing to your voip router? |
| 01:42.17 | blitzrage | didn't have to setup anything in sip.conf for that |
| 01:42.24 | blitzrage | pointing to a CCM |
| 01:42.36 | perd | hrm |
| 01:42.44 | perd | one of those windows boxes == the ccm? |
| 01:42.57 | blitzrage | looks like CCM 4.x |
| 01:43.01 | perd | ah |
| 01:43.08 | perd | damnit |
| 01:43.10 | blitzrage | had to look through the document to find it |
| 01:43.13 | perd | i SWEAR i had this working both ways |
| 01:43.15 | perd | earlier today |
| 01:43.25 | blitzrage | probably did, then changed something |
| 01:43.34 | perd | yeah unfortunately my mind is not well today :/ |
| 01:43.45 | perd | i was bad last night. very bad. |
| 01:43.45 | perd | hehe |
| 01:44.21 | blitzrage | heh |
| 01:44.29 | Nugget | Sharif didn't like it -- rock the voip call - rock the voip call. |
| 01:44.40 | perd | you so crazy |
| 01:44.41 | Sweeper | hmm |
| 01:44.50 | blitzrage | nerds |
| 01:45.02 | Sweeper | looks like .call files might be my best bet |
| 01:45.14 | Nugget | I'm on IRC and playing WoW at the same time. I think the nerdity is exponential. |
| 01:45.23 | blitzrage | Sweeper: for what? |
| 01:45.32 | blitzrage | Nugget: yep -- it sure is |
| 01:45.41 | Sweeper | blitzrage: automating call origination |
| 01:45.49 | Qwell | pfft, until you start dreaming about WoW... |
| 01:45.54 | perd | if you wrote an asterisk->wow interface i'd be impressed with your geekiness |
| 01:46.02 | Sweeper | I'm allowed 25 simultaneous calls on my IAX trunk |
| 01:46.10 | Nugget | that's an excellent idea. |
| 01:46.11 | perd | using TTS and RA |
| 01:46.13 | Sweeper | can't figure out a way to do things simply with RAMI |
| 01:46.32 | perd | err RA = VR |
| 01:46.34 | blitzrage | yah... callfiles are certainly easier than manager until you get it working |
| 01:47.46 | Sweeper | I think the manager interface would be counterproductive when trying to doo 100's of calls at once :/ |
| 01:48.15 | file | try multitasking with THAT many conversations! |
| 01:48.53 | blitzrage | ok, time to go do something OTHER than work on Asterisk |
| 01:49.02 | file | blitzrage: ummm work on Asterisk! |
| 01:49.04 | blitzrage | GO LEAFS GO! |
| 01:49.11 | blitzrage | file: no... I'm going to intoxicate myself :) |
| 01:49.21 | file | blitzrage: good plan |
| 01:49.26 | file | blitzrage: go with the blessings of Telcomjoshvoxmart |
| 01:49.29 | blitzrage | w00t! |
| 01:49.36 | blitzrage | I need a Telcomjoshvoxmart shirt |
| 01:49.54 | Qwell | HA |
| 01:50.09 | file | Telcomjoshvoxmart Shirt complete with GPS tracking device! |
| 01:50.53 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 01:50.54 | Sweeper | where's a good reference for call files? |
| 01:51.58 | Sweeper | ~call file |
| 01:52.04 | jbot | ACTION looks around and then screams out file as loudly as possible |
| 01:52.11 | file | yesssssss? |
| 01:52.18 | Sweeper | XD |
| 01:52.21 | Sweeper | now that's funny |
| 01:52.31 | Sweeper | maybe... |
| 01:52.34 | Sweeper | ~call\ file |
| 01:52.41 | Sweeper | ~call_file |
| 01:52.50 | perd | mmmm beer |
| 01:53.05 | Qwell | That was particularly funny |
| 01:54.20 | nick125_lappy | Hmm..this is going to be interesting.. |
| 01:54.51 | Sweeper | oh, fucking awsome |
| 01:55.03 | Sweeper | I can haxxor the modified date into da FUTURE |
| 01:55.03 | nick125_lappy | Yay, this works! |
| 01:55.21 | perd | did you just learn about touch? |
| 01:55.35 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:55.59 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:56.27 | nick125_lappy | Hmm....I'm trying to figure out the best way to do this...I have two (might increase to three) SIP devices that I want to be called when an incoming call comes in, what would be the best way to do that? Queue? |
| 01:56.50 | droops | so you want them all called at once, or in some order |
| 01:57.03 | nick125_lappy | all at once |
| 01:57.16 | droops | use a comma in your dial statement |
| 01:57.18 | nick125_lappy | some devices might not always be connected though |
| 01:57.31 | nick125_lappy | so, I don't want it to error if the device isn't "there" |
| 01:57.47 | droops | it shouldnt error in the call |
| 01:57.55 | droops | you will see it in the console |
| 01:59.23 | Corydon76-home | Evening, droops |
| 01:59.37 | droops | oh i think its an ampersand, not a comma |
| 01:59.41 | droops | nick125_lappy you still there |
| 01:59.44 | droops | and hey Corydon76-home |
| 02:00.17 | droops | http://www.voip-info.org/wiki-Asterisk+cmd+dial |
| 02:01.19 | treat | i have registerd my asterisk as a sip client (register => user:pass:@sip.com/1001) and my exstension is exten => 1001,1,Dial(SIP/1004&SIP/treat) |
| 02:01.29 | treat | but is does not work |
| 02:01.31 | treat | it |
| 02:01.41 | nick125_lappy | Hmm...I wonder why my incoming calls from FWD aren't showing up in sip debug.. |
| 02:01.58 | hardwire | how well does asterisk work on opensolaris? |
| 02:02.46 | hardwire | http://www.voip-info.org/wiki/index.php?page=Asterisk+Solaris+Support |
| 02:02.47 | hardwire | heh |
| 02:02.50 | *** join/#asterisk tga (n=tga@unaffiliated/tga) |
| 02:02.56 | tga | greetings |
| 02:03.20 | nick125_lappy | I wonder if they are still sending calls over IAX.. |
| 02:03.47 | JT | so you were on iax? why change |
| 02:04.56 | nick125_lappy | I wonder if the "Call me" function on FWD is broken |
| 02:05.56 | nick125_lappy | If I call myself, it *seems* to work |
| 02:06.02 | nick125_lappy | Yet, i'm not sure since I'm calling myself |
| 02:07.50 | Dr-Linux|home | what's the suggested opensource program for asterisk 2 identical servers redundancy? |
| 02:08.21 | JT | "program"? what sort of redundancy? |
| 02:08.24 | tga | hey folks, what's a good POTS provider out there with flat rates for the US and Canada? |
| 02:11.29 | blitzrage | Dr-Linux: DUNDi |
| 02:12.04 | blitzrage | Dr-Linux: I like combining it with pgcluster and ODBC connections to Asterisk. |
| 02:12.12 | blitzrage | func_odbc makes the system really powerful |
| 02:14.06 | Dr-Linux|home | blitzrage: DUNDi is a program? |
| 02:16.06 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 02:17.02 | blitzrage | Dr-Linux: http://leifmadsen.com/papers/dundi-intro.pdf |
| 02:17.45 | tzanger | have any of you guys seen two * servers connected via SIP have connectivity but simply be unable to establish a call? |
| 02:17.52 | Dr-Linux|home | blitzrage: ok thanks |
| 02:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:18.26 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 02:19.22 | tzanger | I've seen this numerous times between my home and office * servers. If the office server drops off the 'net, the home* box *cannot* connect to it |
| 02:20.08 | JT | while the office box is offline? sounds understandable |
| 02:20.11 | tzanger | office* has a qualify to home* and sees it (now reachable, 81ms or so) |
| 02:20.29 | tzanger | home* does not have qualify to office*, but restarting it, it says that the registration packet needs to be retransmitted |
| 02:20.38 | tzanger | even if I *stop* home* and restart asterisk |
| 02:20.40 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:20.45 | tzanger | even if I kill the ineternet ocnnection and restart it |
| 02:20.50 | tzanger | (home* is also the pppoe endpoint) |
| 02:20.50 | JT | if they are both static IPs, there's no need to use register |
| 02:21.02 | tzanger | generally I have to *reboot* home* to get it working again |
| 02:21.19 | tzanger | JT: I understand that. I use register to get the voicemail notifications |
| 02:21.26 | tzanger | I can try turning off the registration |
| 02:21.27 | JT | i see |
| 02:21.39 | tzanger | but it just seems odd that the boxes cannot talk once the link is back up |
| 02:21.48 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 02:21.53 | JT | maybe you need a shorter registration timeout? *shrug* |
| 02:21.55 | tzanger | and even odder that I seemto have to completely reboot home* to get it working again |
| 02:22.06 | tzanger | JT: hmm, never thought of that |
| 02:22.09 | tzanger | gotta go look up that parameter |
| 02:22.20 | tzanger | but still, stopping and restarting * should clear all of that |
| 02:22.28 | JT | it should |
| 02:22.35 | JT | maybe there are other issues at play |
| 02:23.46 | blitzrage | yah... that is a might strange situation |
| 02:23.50 | blitzrage | mighty* |
| 02:24.36 | Dr-Linux|home | blitzrage: that looks nice, but what about SIP, it's talking about IAX |
| 02:25.16 | *** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 02:26.15 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:26.59 | blitzrage | substitute the signalling protocol -- it doesn't matter |
| 02:33.46 | sevard | that's a program well worth the stabbing. |
| 02:33.49 | robl^ | stab it real good |
| 02:34.01 | nick125_lappy | there, it finally works |
| 02:34.24 | nick125_lappy | I least get hold music! that's a start |
| 02:34.32 | sevard | it was probably chewing up 300mb of ram and paging 700mb, or you know, normal xlite operation while not in a call. |
| 02:34.54 | nick125_lappy | I guess I have to do with what i've got |
| 02:35.53 | nick125_lappy | All other SIP clients don't work correctly for some reason |
| 02:35.53 | nick125_lappy | (For Linux) |
| 02:35.53 | nick125_lappy | Though, my PAP2T-NA will be in tomorrow |
| 02:35.53 | nick125_lappy | ...providing Fedex doesn't screw up like usual |
| 02:36.09 | nick125_lappy | Wewt, sound is working on my asterisk box! yay |
| 02:36.10 | robl^ | nick125_lappy: it will end up in Murmansk |
| 02:36.26 | nick125_lappy | robl^: probably |
| 02:36.40 | De_Mon | how can I kick all users out of a conference from the CLI? |
| 02:36.52 | nick125_lappy | Fedex was two days late with the gears of war shipment to the gamestop.... |
| 02:37.04 | nick125_lappy | I don't know of a way to kick them all out at once, but.. |
| 02:37.09 | nick125_lappy | meetme kick <conf> <user> |
| 02:37.24 | De_Mon | im sorry, not cli dialplan |
| 02:38.00 | De_Mon | I think hteres a meetmeadmin() but thats still the same issue |
| 02:38.25 | De_Mon | oh im a tard for not looking at the docs first |
| 02:38.44 | De_Mon | meetmeadmin will kick all users out where meetme from the CLI cant |
| 02:38.50 | De_Mon | thats kinda dumb |
| 02:39.11 | De_Mon | I should patch meetme cli command to do everything meetmeadmin does (kick all) |
| 02:39.36 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 02:40.16 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:44.09 | nick125_lappy | Dangit, I can't figure out how to get blind transfers and attended transfers to work |
| 02:45.09 | x86 | enable it in features.conf, then add it to the dialplan in extensions.conf |
| 02:45.13 | x86 | easy cheesy :) |
| 02:45.28 | x86 | nick125_lappy: google for "blind transfers site:voip-info.org" |
| 02:47.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:47.23 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 02:47.23 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:47.56 | nick125_lappy | Why do I have a feeling that xlite isn't sending the DTMF.. |
| 02:49.05 | x86 | i'm guessing it's more likely a bad Dial string ;) |
| 02:49.08 | tengulre | anybody using eicon card for asterisk? |
| 02:49.23 | nick125_lappy | What's a "proper" dial string for features.conf things? |
| 02:50.21 | x86 | nick125_lappy: voip-info.org has a very nice wiki page on the Dial command |
| 02:50.31 | x86 | nick125_lappy: check out the flags |
| 02:50.39 | x86 | apparantly :) |
| 02:51.34 | nick125_lappy | I guess getting the flags right would help |
| 02:51.46 | x86 | yarr matey |
| 02:51.47 | x86 | :) |
| 02:52.16 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 02:52.59 | nick125_lappy | Ugh, xlite is horrible |
| 02:53.09 | nick125_lappy | 30 seconds after hitting the red button, its still connected.. |
| 02:53.28 | jtexter3 | I have a question about sip headers when making calls across servers. If I initiate a call on Box A, and use SIPAddHeader to add Alert-Info, call Box B, and have that initiate a SIP call to a Polycom phone, will the header get passed all the way along the chain? |
| 02:56.05 | tzanger | hmm |
| 02:56.11 | tzanger | all this time and my sip jb was off |
| 02:56.21 | tzanger | turned on the adaptive jb, let's hope it works :-) |
| 02:56.51 | tzanger | sip show settings fucking rocks |
| 02:57.03 | *** join/#asterisk CrashHD (i=CrashHD@67.182.167.222) |
| 02:58.04 | [hC] | tzanger: 1.4 or 1.2? |
| 02:58.11 | tzanger | svn trunk |
| 02:58.28 | [hC] | Ahh |
| 02:58.36 | [hC] | Let me know how it works for ya (the jb) |
| 02:58.51 | tzanger | if it works, I won't hear about it... if I catch hell tomorrow, I'll let you know :-) |
| 02:59.55 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:02.39 | nick125_lappy | Okay, I'm starting to hate kphone too.. |
| 03:04.16 | nick125_lappy | DTMF in kphone is not working, and, yes, I have dtmfmode set to inband |
| 03:04.25 | nick125_lappy | and I'm using g711u as the codec |
| 03:04.40 | JT | tried rfc2833 or info? |
| 03:04.50 | Sweeper | anyone here use voxee? |
| 03:05.07 | nick125_lappy | JT: I've heard you are supposed to use inband for kphone |
| 03:05.41 | JT | really? kphone must be shit then if it doesn't support out of band dtmf |
| 03:05.41 | tzanger | nick125_lappy: look at the kphone code and see what it is doing |
| 03:05.48 | *** join/#asterisk olsen_ (n=Diego@200.61.236.33) |
| 03:06.00 | tzanger | is it generating a DTMF waveform or is it sending an SDP (I think that's what it's called) packet? |
| 03:06.16 | nick125_lappy | It doesn't suport out of band dtmf last time I checked |
| 03:06.36 | JT | that's really crap |
| 03:06.36 | tzanger | I can't believe they actually generate teh dtmf waveform |
| 03:06.56 | tzanger | or better, just packet capture and see if you get SDP.. use ethereal to decode it |
| 03:07.07 | Sweeper | Jan 16 23:05:24 NOTICE[34846]: chan_iax2.c:2821 auto_congest: Auto-congesting call due to slow response <-- ok, what's wrong here? I'm pretty sure I've successfully registered with the IAX peer.... |
| 03:07.11 | nick125_lappy | I just want a SIP client that works. |
| 03:07.21 | tzanger | if it's nothing but regular audio RTP (maybe rfc2833 dtmf is special RTP, I can't remember) then yeah it's inband |
| 03:07.27 | nick125_lappy | Least long enough until my PAP2T-NA gets in tomorrow |
| 03:07.31 | tzanger | Sweeper: it's not getting a response |
| 03:07.59 | JT | do you actually *need* sip? |
| 03:08.16 | Sweeper | hmmm |
| 03:08.24 | Sweeper | now I get no authority found! |
| 03:08.26 | Sweeper | yay :P |
| 03:08.33 | Globetrotter | hI gUYS, Ii get this error after the call ends.. I am trying to store my CDR's in mssql.. can someone please tell me what i am doing wrong?? cdr_odbc: Error SQLConnect -1 |
| 03:17.20 | nick125_lappy | Dangit, I had call transfering working once, but, I can't reproduce the issue |
| 03:17.26 | nick125_lappy | er, the fix |
| 03:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:19.42 | Mad|Cow | If I have a SIP device registered to Asterisk, can I force it to re-register? |
| 03:20.04 | Mad|Cow | I have a phantom SIP device and I want to "kick it" :-) |
| 03:24.54 | Sweeper | Jan 16 23:25:26 NOTICE[34884]: chan_iax2.c:2821 auto_congest: Auto-congesting call due to slow response <-- how can I set this to be less picky? |
| 03:28.10 | *** join/#asterisk Techie-Micheal (n=techie-m@phpbb/support/techie-micheal) |
| 03:30.37 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:32.42 | Techie-Micheal | What would cause Asterisk to respond Unauthorized when a softphone is trying to Register with it? I have one server that works, one server that doesn't. Same config, but one that works is on my local network and the one that doesn't is on a public IP. |
| 03:34.24 | nick125_lappy | Hmm, this is interesting: If I park a call, then, pick it up from another extension, I can't park the call again.. |
| 03:35.38 | JT | Techie-Micheal: probably only set to allow certain ip ranges in sip.conf |
| 03:37.33 | Techie-Micheal | JT: I thought about that, but this is a sample config with make samples and the user is set to host=dynamic |
| 03:39.19 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
| 03:40.50 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
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| 03:41.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:43.50 | *** join/#asterisk Thazza (n=no@229.9.233.220.exetel.com.au) |
| 03:43.51 | Techie-Micheal | Oh, and nat=yes is also set for the user in question. |
| 03:44.07 | Techie-Micheal | Because I'm behind nat right now. :P |
| 03:45.03 | nick125_lappy | Anyone here know of somewhere that will terminate tollfree calls for free? |
| 03:45.14 | nick125_lappy | (I'm having a hard time getting DTMF working on trxtel) |
| 03:48.25 | Sweeper | I guess it's possible that voxee is just plain flaky :v |
| 03:49.42 | nick125_lappy | Sweeper: having issues with voxee? (I'm looking at adding them to my provider "blend", but, I won't if their service is crap) |
| 03:50.36 | Sweeper | nick125_lappy: well, I really don't know atm if the service is crap, or if my network connection is flaky, or if I'm being net-neutral'd, or if my config is bad |
| 03:50.52 | nick125_lappy | Sounds like fun! |
| 03:51.40 | AtomicStack | i'm struggling getting a SIP trunk to work between two boxes... from what i can see from sip show users/sip show peers, both ends are configured correctly, but the terminating box is currently rejecting the authenticated invite with a 403 forbidden. |
| 03:52.04 | AtomicStack | is the MD5 digest sent as authorisation just the md5secret + the nonce? |
| 03:52.51 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 03:53.05 | Techie-Micheal | AtomicStack: I'm by no means pretend to know much of anything about Asterisk, but as far as I can see, yes, the secret plus the nonce. |
| 03:53.10 | teknoprep | is the polycom noticiably better then the linksys spa-9xx series speakerphone ? |
| 03:53.14 | Techie-Micheal | s/I'm/I/; |
| 03:54.57 | nick125_lappy | I can't remember how to prepend a digit to a dial string...(for example, if I dial 55555555555, and, I need to send *5555555555 to my provider) |
| 03:55.12 | teknoprep | *+ ? |
| 03:55.22 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 03:58.28 | JT | Dial/blah/*${EXTEN} |
| 03:58.32 | JT | err |
| 03:58.38 | *** join/#asterisk azidenth (n=aby_azid@60.50.220.139) |
| 03:58.40 | JT | let's try that again |
| 03:59.12 | *** part/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 03:59.17 | JT | Dial(blah/1/*${EXTEN}) |
| 03:59.28 | JT | it's that simple |
| 04:00.27 | Sweeper | nick125_lappy: turns out it was voxee |
| 04:00.33 | Sweeper | I suggest les.net :o |
| 04:00.34 | nick125_lappy | JT: Yay, it works |
| 04:00.55 | nick125_lappy | And DTMF over FWD works. |
| 04:01.06 | nick125_lappy | Yay. |
| 04:01.18 | nick125_lappy | Sweeper: How reliable are they? |
| 04:01.44 | nick125_lappy | Hmm...$0.015 is a tiny bit more then I'm looking to pay per minute for termination |
| 04:01.58 | wunderkin | ... wow |
| 04:02.06 | nick125_lappy | (to US) |
| 04:02.10 | Sweeper | o.O |
| 04:02.32 | Sweeper | what I want to know is, who are you getting for less than that, that isn't voxee |
| 04:02.57 | nick125_lappy | vitelity is 1.37c per minute to US |
| 04:03.09 | danp | is there a trac-timeline-like list of asterisk svn changesets? |
| 04:03.40 | wunderkin | you're splitting hairs for residential use |
| 04:03.48 | Techie-Micheal | JT: Any other ideas? I'm sure I'm screwing something up, but I haven't a clue what or how. |
| 04:05.09 | wunderkin | for 2500 min in a mo, the difference is 3.25 |
| 04:05.42 | wunderkin | you'll want 1 or 2 backups |
| 04:05.53 | nick125_lappy | wunderkin: Hmm.. |
| 04:06.07 | azidenth | Kigh: thanks for the info on realtime yesterday |
| 04:09.12 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:10.21 | *** join/#asterisk LoneShadow (n=duh@59.92.196.119) |
| 04:15.15 | JT | Techie-Micheal: eh, dunno?, is localnetworks or whatever set in sip.conf? |
| 04:15.22 | Mad|Cow | I'm trying to connect a softphone behind a firewall to my Asterisk server on the public inet. It wont connect... I have NAT=yes etc. in my sip.conf file. My tcpdumps show the phone trying to register but sip debuging for the IP in asterisk dont show anything. Any ideas? |
| 04:15.30 | nick125_lappy | Damnit! I hate xlite! Call transfering only works half the time |
| 04:16.04 | Techie-Micheal | JT: No, it isn't. |
| 04:16.13 | CunningPike | Mad|Cow: Try a wireshark pcap to see what's going on |
| 04:16.45 | Mad|Cow | CunningPike: I have... all i see are REGISTER requests |
| 04:17.05 | Mad|Cow | CunningPike: but I dont see anything in the sip debug in the asterisk console |
| 04:17.16 | AtomicStack | Mad|Cow: could be binding to the wrong IP/interface, are you getting any ICMP responses sent back out? or are the invites just going into a black hole? |
| 04:17.17 | Techie-Micheal | Mad|Cow: I have the _exact_ same problem. |
| 04:17.49 | JT | Mad|Cow: where are you packet dumping from? |
| 04:17.53 | Mad|Cow | Techie-Micheal: I take it you havent been able to resolve it? |
| 04:17.53 | CunningPike | Mad|Cow: Are you sniffing on the Asterisk side of the firewall? |
| 04:17.59 | Techie-Micheal | Mad|Cow: nope. |
| 04:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:18.10 | Mad|Cow | JT: from my Asterisk server |
| 04:18.16 | JT | hmm |
| 04:18.18 | nick125_lappy | Hmm...Dangit, I need to pass the Tt flag when I pick up parked calls (so, if need be, I can park them again) |
| 04:18.45 | JT | check with netstat or lsof that asterisk is actually bound to the necessary ip |
| 04:18.53 | Techie-Micheal | Mad|Cow: I have an asterisk server on the same network I'm on, and it works using the sample config files, but using the same setup you are, it doesn't work. I get asterisk saying unauthorized in sip set debug and my softphone reports a 408 after a while. |
| 04:18.57 | Mad|Cow | AtomicStack: I see ICMP responces going out but they timeout |
| 04:19.33 | JT | Techie-Micheal: it's not the same problem, Mad|Cow is not seeign anything in sip debug |
| 04:19.37 | Mad|Cow | techie-micheal: Yep... sounds like what I got going :-) |
| 04:20.14 | Mad|Cow | JT: Any ideas? |
| 04:20.56 | JT | like i just said |
| 04:20.59 | JT | < JT> check with netstat or lsof that asterisk is actually bound to the |
| 04:20.59 | JT | <PROTECTED> |
| 04:21.49 | Mad|Cow | jt: Well... its gets even stranger. If I run the softphone from infront of the firewall it connects without a problem. |
| 04:22.09 | Mad|Cow | JT: I have tried from several different networks (behind firewalls) all with the same result |
| 04:22.23 | JT | well there's nothing strange there, disabling the firewall is always a good diagnostic step :) |
| 04:23.04 | Mad|Cow | JT: tell me about it. It allways works in front of the firewall.. just not behind. |
| 04:23.53 | JT | you should probably pastebin your full sip.conf, minus your actual passwords |
| 04:23.53 | Techie-Micheal | Mad|Cow: Do you have nat=yes set on the user in sip.conf? |
| 04:23.59 | nick125_lappy | Hmm, anyone here know of a way to put a call on hold though some kind of keypress (that will put hold music, and, allow me to hang the extension up)? |
| 04:24.18 | Mad|Cow | Techie-Micheal: I do. No luck |
| 04:24.37 | wunderkin | you need localnet and externip set |
| 04:24.46 | JT | you're making it very hard for people to assist without pastebinning your sip.conf |
| 04:24.53 | wunderkin | and to the proper values, i have no idea what you tried setting it to |
| 04:25.07 | Techie-Micheal | wunderkin: My understanding is that you only need to do that if both the client and the server are behind a nat and you want the server to connect outside the nat |
| 04:25.26 | *** join/#asterisk Soms (n=soms@wikimedia/Somitho) |
| 04:25.33 | Mad|Cow | JT: Let me find it... I put it up earlier today |
| 04:25.35 | wunderkin | oh, is asterisk behind nat? |
| 04:25.38 | JT | Techie-Micheal: theoretical or practical? |
| 04:26.17 | Techie-Micheal | wunderkin: No, not in this case. |
| 04:26.21 | Techie-Micheal | JT: hm? |
| 04:26.38 | JT | your understanding of it... |
| 04:27.12 | Techie-Micheal | sip.conf :P |
| 04:27.32 | Mad|Cow | JT: http://www.pastebin.ca/318928 <-- After going thru this with someone else today, i finnaly gave in and they gave me this config (which did the same thing as what i described) |
| 04:27.47 | JT | Techie-Micheal: ? |
| 04:27.57 | Techie-Micheal | From what I've read of the docs, externip needs to be set only if your asterisk server is behind a nat. |
| 04:28.18 | *** part/#asterisk Soms (n=soms@wikimedia/Somitho) |
| 04:28.18 | Techie-Micheal | I'm new to Asterisk/SIP, so I could be wrong. :P |
| 04:29.21 | JT | don't trust the documentation unless you've proven it right :) |
| 04:29.56 | wunderkin | it is |
| 04:31.10 | JT | what is |
| 04:31.32 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 04:31.37 | wunderkin | externip is if asterisk is behind nat |
| 04:31.43 | wunderkin | and localnet |
| 04:31.55 | JT | Mad|Cow: set insecure=very |
| 04:32.10 | rpm | insecure=very is 1.2.x, you use insecure=port,invite now |
| 04:32.39 | JT | people should still be using 1.2.x for production, unless something has changed recently |
| 04:32.52 | Mad|Cow | jt: I think I tried that but I'll give it another shot tomorrow. Any other ideas? |
| 04:33.12 | JT | does it qualify? |
| 04:33.20 | rpm | if you are new to asterisk, im sure people are using 1.4 |
| 04:33.50 | Mad|Cow | JT: No (becuase its behind a firewall) |
| 04:34.03 | Mad|Cow | JT: That doesnt allow ICMP |
| 04:34.11 | wunderkin | kjaskfadsf |
| 04:34.16 | wunderkin | we went through that already |
| 04:34.25 | wunderkin | and i think you are using firewall = router |
| 04:34.27 | JT | rpm: i see 1.4.0 has just been released |
| 04:34.32 | JT | i wouldn't use a .0 release |
| 04:34.58 | JT | Mad|Cow: i don't think qualify uses ICMP |
| 04:35.04 | wunderkin | no |
| 04:35.05 | JT | although i'm not 100% |
| 04:36.19 | JT | Mad|Cow: what asterisk release are you using? |
| 04:36.27 | Mad|Cow | JT: 1.2 |
| 04:36.55 | JT | you can set qualify = no for testing purposes |
| 04:36.59 | Mad|Cow | JT: 1.2.12.1 to be exact |
| 04:37.26 | Mad|Cow | JT: I have tried both ways... neither seem to affect the outcome ;-) |
| 04:37.26 | JT | ok |
| 04:37.40 | JT | yeah, it's just an unnecessary potential hickup |
| 04:37.56 | JT | you can set it to yes when everything works |
| 04:38.13 | wunderkin | so whats happening? its not registering? |
| 04:38.43 | Mad|Cow | wunderkin: yeah... it just keeps trying to register |
| 04:38.47 | blitzrage | .0 is not for trusting |
| 04:39.12 | Mad|Cow | If someone is behind a firewall... I would be happy to set you up with an extention and you can try to connect |
| 04:39.13 | wunderkin | and do you see the registration attempts on asterisk |
| 04:39.44 | Mad|Cow | wunderkin: no. Thats whats so strange. but tcpdumps show it trying |
| 04:40.00 | Mad|Cow | wunderkin: (to REGISTER) |
| 04:40.05 | wunderkin | on the client or server |
| 04:40.13 | Mad|Cow | wunderkin: on the server |
| 04:40.27 | JT | nothint with sip debug on? |
| 04:40.27 | Mad|Cow | wunderkin: the client eventually times out with a 408 responce code |
| 04:41.29 | Mad|Cow | JT: nothing that I can see... I have other clients so it get tough to see the traffic |
| 04:41.43 | Mad|Cow | JT: I have to end up using sip debug IP to filter |
| 04:42.58 | Mad|Cow | wunderkin / JT: If either of you are behind a NAT'd firewall, I would be curious to see if you could connect |
| 04:43.26 | Mad|Cow | 23:24:40.175225 IP 136.141.2.76.33556 > 74.92.109.201.5060: SIP, length: 592 |
| 04:43.26 | Mad|Cow | E .l....n......LJ\m......X@.REGISTER sip:voip.foxtailsolutions.com SIP/2.0 |
| 04:43.26 | Mad|Cow | Via: SIP/2.0/UDP 136.141.2.76:7749;branch=z9hG4bK-d87543-58551941d101d747-1--d87543-;rport |
| 04:43.26 | Mad|Cow | Max-Forwards: 70 |
| 04:43.26 | Mad|Cow | Contact: <sip:3070s@136.141.2.76:7749;rinstance=8d143a7c0f594e0f> |
| 04:43.27 | Mad|Cow | To: "Carissa Hirt"<sip:3070s@voip.foxtailsolutions.com> |
| 04:43.29 | Mad|Cow | From: "Carissa Hirt"<sip:3070s@voip.foxtailsolutions.com>;tag=9f491861 |
| 04:43.31 | Mad|Cow | Call-ID: MTdhOTQ0Yjk4NDU2YzcxODhlNmYxNjlhZGI5MmJjMjI. |
| 04:43.33 | Mad|Cow | CSeq: 1 REGISTER |
| 04:43.35 | Mad|Cow | Expires: 3600 |
| 04:43.37 | Mad|Cow | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO |
| 04:43.39 | Mad|Cow | User-Agent: X-Lite release 1006e stamp 34025 |
| 04:43.41 | Mad|Cow | Content-Length: 0 |
| 04:43.43 | Mad|Cow | That was from my last tcpdump of the night |
| 04:43.53 | danp | hey, that's great |
| 04:43.56 | Mad|Cow | I just keep seeing REGISTER |
| 04:45.09 | Mad|Cow | danp: sorry :-) |
| 04:45.46 | wunderkin | well 3069s is what you have on pb |
| 04:46.23 | Mad|Cow | wunderkin: I know... I changed it for my dialplan |
| 04:46.24 | blitzrage | Mad|Cow: pastebin.ca |
| 04:47.04 | Mad|Cow | wunderkin: I just didnt want to confuse anyone into thinking I was using the wrong username/password |
| 04:49.48 | wunderkin | well we are not talking about a dialplan, just sip.conf and the registration.. so is it 3070s or 3069s |
| 04:50.01 | Techie-Micheal | Mad|Cow: That's similar to what my sip debug shows. |
| 04:50.15 | nick125_lappy | Is there a way to append the Tt flags to an incoming call (or, when I pick up a parked call)? |
| 04:51.09 | Mad|Cow | wunderkin: understood. The dump above was for another extention that the user has (3070s) |
| 04:51.10 | nick125_lappy | I need to be able to park incoming calls as well as repark calls |
| 04:51.35 | Mad|Cow | wunderkin: s = softphone ; my lame excuse for a nameing convention ;-) |
| 04:52.24 | Mad|Cow | Techie-Micheal: Set me up with a user/extenion on your * and i'll try to connect with my soft client |
| 04:53.30 | wunderkin | it isnt helpful to show something that isnt what we are looking for |
| 04:53.37 | Techie-Micheal | Mad|Cow: Sure, can you give me about 10 minutes? I'm on my laptop right now and the root password to the server is upstairs on my PC. |
| 04:53.52 | Mad|Cow | Techie-Micheal: np |
| 04:54.28 | Mad|Cow | wunderkin: ok... give me minute and i'll get my complete, up to date sip.conf online |
| 04:54.32 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 04:59.09 | *** join/#asterisk Techie-Micheal (n=Techie-M@phpbb/support/techie-micheal) |
| 04:59.16 | Mad|Cow | wunderkin: http://www.pastebin.ca/319256 |
| 04:59.34 | Techie-Micheal | Mad|Cow: One sec, let me get you a test user. |
| 04:59.59 | Mad|Cow | Techie-Micheal: NP. |
| 05:00.47 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:00.48 | Mad|Cow | Techie-Micheal: I'll be comming from 74.92.109.205; tcpdump -A -n -s 1500 port 5060 and host 74.92.109.205 |
| 05:00.55 | Techie-Micheal | Mad|Cow: Incoming PM with the details |
| 05:00.57 | Techie-Micheal | Okie, thanks. |
| 05:00.58 | Mad|Cow | Techie-Micheal: If you want to watch the traffic |
| 05:01.51 | variable_office | CrashHD well i tested it to a second asterisk box, and the second box sees the dtmf, which means it HAS to be the upstream! |
| 05:02.13 | Mad|Cow | Techie-Micheal: I have to be quite though.. my wife is asleep... so if I connect, i'm not going to be able to chat (hope thats ok) |
| 05:02.35 | Techie-Micheal | Certainly. :) Not like I can connect to it anyway. :P |
| 05:02.45 | Mad|Cow | lol |
| 05:04.18 | Techie-Micheal | Hrm. Getting anywhere with it? I'm not seeing any traffic from you (or anybody :P) |
| 05:04.25 | Mad|Cow | techie-micheal: incomming now |
| 05:04.33 | Techie-Micheal | Yup, there it is. |
| 05:04.46 | Techie-Micheal | 401 unauthorized. |
| 05:04.58 | Mad|Cow | techie-micheal: 408 timeout |
| 05:04.59 | Techie-Micheal | CSeq: 1 REGISTER |
| 05:05.10 | Techie-Micheal | Then a scheduling destruction of the sip. |
| 05:05.12 | Mad|Cow | Techie-Micheal: can you send PM me the tcpdump |
| 05:05.27 | Techie-Micheal | Yeah, one sec. |
| 05:06.37 | Mad|Cow | Techie-Micheal: I assume you have NAT=yes etc. etc. in your sip.conf for my user? |
| 05:06.44 | Techie-Micheal | Yes. |
| 05:07.49 | Mad|Cow | Techie-Micheal: Can you pb your sip.conf? Curious how it compaires to mine |
| 05:08.08 | Techie-Micheal | Sure. |
| 05:08.11 | Mad|Cow | Techie-Micheal: for your reference: http://www.pastebin.ca/319256 |
| 05:44.36 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 05:44.36 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:44.36 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:44.36 | nick125_lappy | Ugh, this is fun. Vitelity doesn't have any DIDs in my ratecenter (they have some in another rate center, same area code though) |
| 05:44.37 | nick125_lappy | I'm wondering, would I be okay to order from a different ratecenter..hmm |
| 05:44.37 | *** join/#asterisk einfo (n=Sparrow@200.26.135.129) |
| 05:44.37 | einfo | hola |
| 05:44.38 | treat | i had troubles registering sip clients but then i realised i was using the wrong realm |
| 05:44.38 | Techie-Micheal | treat: How do you mean wrong realm? |
| 05:44.38 | Techie-Micheal | Because Mad|Cow and I are having trouble registering sip clients. |
| 05:44.38 | Techie-Micheal | We are getting 401 replies from asterisk. |
| 05:44.38 | *** join/#asterisk rc-1 (n=rc-1@ip68-229-102-1.hr.hr.cox.net) |
| 05:44.39 | treat | whats that forbidden? |
| 05:44.39 | Techie-Micheal | Unauthorized. |
| 05:44.39 | Techie-Micheal | I think forbidden is something else. |
| 05:44.39 | flenders | hey, there's something weird with my zap channels... if I place an incoming call, and when it the auto-attendant answers, if I hangup, it doesn't hangup the call, it waits for the timeout. |
| 05:44.39 | treat | well i changed realm in [general] to my hostname |
| 05:44.39 | flenders | on incoming sip calls, asterisk hangs up the call as soon as I hang up on the calling phone |
| 05:44.39 | Techie-Micheal | treat: Hm. |
| 05:44.39 | nick125_lappy | Okay, here's what I need to do: I need to get a call pattern to match this: *563*<ARG1>*<ARG2>*, and, I need to be able to access ARG1 and ARG2 though the dialplan |
| 05:59.47 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 06:01.28 | *** join/#asterisk hoobastooba (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 06:03.05 | hoobastooba | i get a very faint echo on calls over iax2. I lace a call from cell that goes to a server at my office via zap (which does not have echo when i terminate the call there to a phone) I send that call then over iax to my home server then to zap phone. no echo unless using iax from office. |
| 06:03.24 | hoobastooba | i have tried with nufone and there is no echo. |
| 06:13.26 | *** join/#asterisk dongc (n=dongc@203.117.206.249) |
| 06:16.05 | *** join/#asterisk [F] (n=f@pool-72-83-105-204.washdc.fios.verizon.net) |
| 06:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:22.04 | *** join/#asterisk ardaei (n=ardaei@70-59-21-112.hlrn.qwest.net) |
| 06:22.29 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 06:22.39 | ardaei | hello |
| 06:22.51 | *** part/#asterisk hoobastooba (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 06:23.29 | ardaei | anyone awake here? |
| 06:32.06 | *** join/#asterisk ardaei (n=ald@mail.bci-group.com) |
| 06:34.31 | tzafrir_laptop | yes |
| 06:35.17 | perd | can i get a hell yeah |
| 06:35.33 | JT | nick125_lappy: don't quite get your pattern problem |
| 06:36.16 | nick125_lappy | I *finally* figured it out.. |
| 06:36.24 | nick125_lappy | it's a really ugly hack, but, it works |
| 06:36.35 | JT | explain what you need to do |
| 06:36.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:36.51 | *** join/#asterisk judas_iscariote (n=cristian@200.7.26.201) |
| 06:37.22 | *** part/#asterisk judas_iscariote (n=cristian@200.7.26.201) |
| 06:38.21 | nick125_lappy | I needed to write a small dial "code" to change caller ID (for example, sometimes when I call into my ISP, they check caller ID, and, it has to be my home number otherwise they will want to call me back, etc) |
| 06:38.41 | JT | ok, so what's hard about that? |
| 06:38.52 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:38.53 | Qwell | What'd you do, hack app_dial.c? |
| 06:38.56 | nick125_lappy | here's what I eventually did: |
| 06:38.57 | nick125_lappy | exten => _*585*XXXXXXXXXXX*1NXXNXXXXXX*,1,Set(CALLERID(num)=${EXTEN:5:11}) |
| 06:39.02 | nick125_lappy | exten => _*585*XXXXXXXXXXX*1NXXNXXXXXX*,2,Dial(SIP/${EXTEN:17:11}@vitel-outbound,30,Tt) |
| 06:39.08 | Qwell | umm |
| 06:39.11 | Qwell | why? |
| 06:39.16 | JT | this is looking non-optimal |
| 06:39.22 | Chris-NB | morning |
| 06:39.26 | Qwell | JT: THAT'S an understatement |
| 06:39.31 | nick125_lappy | I know it is |
| 06:39.46 | dongc | Hi, i'm having problem with my TE405P. The span LED never turn RED or green after zaptel loaded. Is it normal or abnormal? |
| 06:41.00 | JT | nick125_lappy: why would you do that? |
| 06:41.09 | JT | i'm still trying to work out exactly what it does |
| 06:41.51 | nick125_lappy | I can dial a number, let's say....*585*11234567890*15055551212*, and it will set the callerid to 11234567890 and call 15055551212 |
| 06:41.53 | perd | dong not sure about that model but my TE110P pulses orange with zaptel loaded and no connection |
| 06:42.53 | dongc | perd: may i know what is your OS and zaptel version? |
| 06:42.57 | JT | nick125_lappy: does the callerid you need to set always change? |
| 06:43.09 | nick125_lappy | JT: It would just be nice to be able to change it easely |
| 06:43.13 | perd | centos 4, zaptel 1.2.something |
| 06:43.40 | perd | 1.2.12 |
| 06:43.49 | nick125_lappy | haha |
| 06:44.10 | perd | get a voipjet account :P |
| 06:44.10 | JT | i'm sure your telcos will fix their systems one day :P |
| 06:44.13 | *** join/#asterisk shinux__ (n=shinux@196.220.24.241) |
| 06:44.15 | nick125_lappy | when I was telling my friend my plan for my asterisk system, he was like "why don't you get vonage?" |
| 06:44.39 | JT | perd: don't know of any australian telco that allows you to set callerid to a number you don't own |
| 06:45.04 | perd | ahh |
| 06:45.18 | perd | sucky |
| 06:45.32 | nick125_lappy | 16-Jan-2007 18:30:00 Arrived at FedEx Ramp ALBUQUERQUE NM US \o/ |
| 06:45.38 | JT | nick125_lappy: you could have it prompt you for caller id first, then the actualy number |
| 06:45.39 | dongc | perd: i'm using centos 4, tried with both zaptel 1.2.12 and 1.4.0, both giving me same problem. :( |
| 06:45.50 | EyeCue | :D |
| 06:45.52 | perd | well i'm using a diff card so who knows |
| 06:45.56 | perd | do you get a /proc/zaptel/0 ? |
| 06:46.02 | perd | what's dmesg show |
| 06:46.02 | dongc | yes |
| 06:46.05 | nick125_lappy | JT: That would require me to do things like record prompts and such.. |
| 06:46.12 | nick125_lappy | and, I don't have a microphone |
| 06:46.12 | perd | what is in /proc/zaptel/0 ? |
| 06:46.22 | JT | maybe, maybe not |
| 06:46.27 | JT | umm, you have a telephone |
| 06:46.28 | JT | i assume |
| 06:46.30 | perd | oh, i guess it's 1, not 0 |
| 06:46.30 | *** join/#asterisk _luckie_ (n=luckie@ip24-252-90-141.no.no.cox.net) |
| 06:46.36 | nick125_lappy | (I used to, but, I don't know where it is) |
| 06:46.38 | perd | Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF RED |
| 06:46.42 | perd | that's what mine shows |
| 06:46.45 | JT | then how do you use asterisk? |
| 06:46.57 | nick125_lappy | JT: Softphone -_- |
| 06:47.04 | dongc | http://pastebin.ca/319326 |
| 06:47.25 | perd | yeah looks good |
| 06:47.29 | JT | that uses a microphone |
| 06:47.37 | nick125_lappy | JT: Well.... |
| 06:48.03 | dongc | perd: i tried to plug in a loopback connector, but LED still doesn't change to green. |
| 06:48.04 | perd | does asterisk not work with it? |
| 06:48.06 | nick125_lappy | I didn't do any talking though the softphone.. |
| 06:48.09 | JT | anyway, you don't need to record prompts necessarily, and you could set it to have a preset range of callerids to set |
| 06:48.13 | perd | oh you never get green? hmm |
| 06:48.18 | perd | maybe there's a physical problem |
| 06:48.21 | dongc | not even RED |
| 06:48.26 | perd | do you get errors in dmesg? |
| 06:49.01 | perd | what about /proc/interrupts |
| 06:49.12 | JT | dongc: umm, are the kernel drivers even loaded? |
| 06:49.28 | perd | kernel drivers are loaded if he has /proc/zaptel/1 |
| 06:50.11 | dongc | perd: i got this http://pastebin.ca/319331 |
| 06:50.29 | dongc | can't really understand what it means... |
| 06:50.57 | dongc | i reied modprobe zaptel and modprobe wct4xxp, both end whitout error. |
| 06:51.06 | dongc | *tried |
| 06:51.10 | tzafrir_laptop | perd, if zaptel is loaded, you have /proc/zaptel |
| 06:51.35 | perd | right |
| 06:51.41 | perd | i wasnt disputing that |
| 06:51.41 | tzafrir_laptop | the module may be loaded, not identify hardware and thus not generate a span (and thus a file under /prc/zaptel) |
| 06:51.50 | JT | dongc: what comes up in zttool? |
| 06:51.57 | JT | or zttest, get any percentages? |
| 06:52.29 | JT | dongc: pastebin /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 06:52.30 | dongc | zttool shows RED |
| 06:52.34 | *** join/#asterisk reber (i=reber@gateway/tor/x-0e62b98efb75671c) |
| 06:53.44 | JT | dongc: also, what is plugged into the card? |
| 06:54.33 | dongc | http://pastebin.ca/319334 |
| 06:55.25 | dongc | Previously i've plugged and E1 to the card (from my Excel switch). Found that D-channel not up from the Excel. |
| 06:55.48 | dongc | Now, i bring back to office and tried to test using a loopback connector. |
| 06:56.09 | JT | you have group=1 twice |
| 06:56.30 | JT | and signalling is unnecessary in the first instance under [channels] |
| 06:56.46 | JT | specify the signalling seperately for each group like you otherwise are |
| 06:57.02 | JT | i note you also have it alternating pri_net, pri_cpe, pri_net, pri_cpe |
| 06:58.22 | dongc | i tried varoius configuration hopefully can make the LED turn RED or Green.. :) |
| 06:58.41 | JT | are the onboard jumpers set to E1 not T1? |
| 06:58.53 | dongc | jumper set to E1. |
| 06:59.09 | JT | describe this loopback cable |
| 06:59.22 | dongc | 1 -->4, 2 -->5 |
| 07:00.16 | JT | on the one jack? |
| 07:00.28 | dongc | yes |
| 07:00.48 | JT | tried making a crossover cable? |
| 07:00.51 | dongc | fyi, the LED on all spans blinking when the kernel not yet loaded (server restarting) |
| 07:01.03 | JT | going between the net and cpe ports |
| 07:01.05 | JT | hmm |
| 07:01.08 | *** join/#asterisk susinths (n=susinths@trollhoin.ifi.uio.no) |
| 07:01.10 | dongc | once the kernel loaded, all LED off. |
| 07:01.27 | dongc | i mean the zaptel loaded..LED never light up. |
| 07:01.42 | JT | i assume you have asterisk running |
| 07:01.47 | dongc | yes. |
| 07:01.53 | JT | ports are all DOWN? |
| 07:02.30 | dongc | zap show status -- all RED. |
| 07:02.58 | JT | does it show all of the channels from all spans? |
| 07:03.02 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:03.09 | dongc | http://pastebin.ca/319337 |
| 07:03.33 | *** join/#asterisk quigleymd (n=Matt@c-71-62-143-171.hsd1.va.comcast.net) |
| 07:03.41 | JT | what happens when you issue the zttest command/ |
| 07:04.15 | perd | bells and whistles sound.. wtf? |
| 07:04.36 | JT | ? |
| 07:04.37 | dongc | btw, do i need to press ctrl-c after the zttest? |
| 07:04.39 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:04.46 | JT | dongc: to stop it, yes |
| 07:04.49 | JT | did any percentages appear? |
| 07:05.19 | dongc | http://pastebin.ca/319338 |
| 07:05.43 | JT | ok looks like you have problems with your drivers |
| 07:06.10 | JT | does the driver come up in lsmod? |
| 07:06.24 | dongc | yes |
| 07:06.56 | dongc | http://pastebin.ca/319339 |
| 07:08.23 | *** join/#asterisk azidenth (n=aby_azid@60.50.220.139) |
| 07:08.46 | dongc | any suggestion on what zaptel version should i use? |
| 07:10.02 | perd | 1.2.12 works here.. what kernel version and gcc version do you have? |
| 07:10.41 | dongc | 2.6.9-34.0.2.EL and gcc-3.4.6-3 |
| 07:10.54 | JT | more importantly, what voltage is the PCI slot you have the card in, dongc ? |
| 07:11.03 | dongc | PCI slot is 5V |
| 07:11.07 | JT | ok |
| 07:11.40 | dongc | do we need to do anything diff. for the 5V PCI card? |
| 07:12.24 | JT | not to my knowledge |
| 07:12.31 | JT | not that i've ever used one of those cards |
| 07:13.02 | dongc | i tried zaptel 1.4.0. having same problem.. and this drive me mad.. |
| 07:13.47 | JT | cat /proc/interrupts |
| 07:14.20 | dongc | http://pastebin.ca/319344 |
| 07:17.42 | dongc | Is zaptel have better support in kernel 2.4 compare to 2.6? |
| 07:17.49 | JT | no |
| 07:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:18.08 | JT | dongc: lsmod? |
| 07:18.15 | perd | you got a lemon, dude! |
| 07:18.37 | dongc | http://pastebin.ca/319351 |
| 07:19.07 | *** join/#asterisk reber (i=reber@gateway/tor/x-545636cd4b46611f) |
| 07:19.08 | perd | damn is that a mylex controller you have? |
| 07:19.13 | perd | old school shit |
| 07:19.22 | perd | that's from the 90s |
| 07:19.24 | dongc | u mean the RAID controller? |
| 07:19.26 | perd | yeah |
| 07:19.55 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 07:20.16 | dongc | haha...it is a supermicro P4 server.. |
| 07:20.26 | perd | wow |
| 07:21.10 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:22.36 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 07:22.58 | dongc | btw, what is the minimum requirement to compile libpri, zaptel and asterisk? only gcc and openssl-devel? |
| 07:23.15 | Hmmhesays | i need some food |
| 07:23.56 | JT | dongc: so the loopback test in zttool doesn't work? |
| 07:24.30 | dongc | once i pressed loopback, it hung. and LED still didn't turn on. |
| 07:24.53 | JT | hrm |
| 07:25.02 | Hmmhesays | i want to watch some stargate damnit |
| 07:25.04 | Hmmhesays | NOW |
| 07:25.09 | perd | new eps out? |
| 07:25.17 | Hmmhesays | no i'm only on season 7 |
| 07:25.25 | perd | oh |
| 07:25.35 | perd | none on newsgroups? |
| 07:25.45 | Hmmhesays | downloading 5 minutes left |
| 07:25.47 | dongc | JT: i mean the screen shows Looping up span x..., please wait.. |
| 07:26.13 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 07:26.14 | JT | hmm ok, i've never used the loopback function |
| 07:26.24 | Hmmhesays | i only have a 3mbit connection |
| 07:26.36 | Hmmhesays | i can't have instant gratification |
| 07:27.00 | perd | yeah that sucks |
| 07:27.06 | perd | you live on an island or something? |
| 07:27.16 | Hmmhesays | yeah an Island called Fargo |
| 07:27.23 | perd | oh man |
| 07:27.26 | perd | dig a hole |
| 07:27.27 | perd | get out |
| 07:27.36 | Hmmhesays | no drivers license |
| 07:27.46 | perd | you dont drive in a hole |
| 07:27.48 | perd | you crawl |
| 07:27.49 | Hmmhesays | not for another 5 months |
| 07:28.20 | Hmmhesays | god damn state took it away |
| 07:28.26 | Hmmhesays | BASTARDS |
| 07:28.32 | perd | dont drive like an asshole. |
| 07:28.42 | perd | haha! yeah i said it |
| 07:28.49 | Hmmhesays | need I remind you... fargo... |
| 07:28.56 | Hmmhesays | the only thing there is to do here is drink |
| 07:29.14 | *** join/#asterisk alamantia (n=Anthony@65.4.1.65) |
| 07:29.16 | Hmmhesays | here after 2 drinks you are over the legal limit |
| 07:29.50 | Hmmhesays | do the freaking math |
| 07:30.02 | Hmmhesays | someone in office was looking to make money |
| 07:30.19 | JT | dongc: i can try logging in if you want, but to do any real testing it will need to be plugged into something useful (in theory, a loopback cable between a pri_cpe and pri_net port should work, although i've never done that on a pri |
| 07:30.45 | Hmmhesays | lets see, 120k people in fargo, 3/4 of them drink... dui = 700 bucks for the state |
| 07:31.04 | Hmmhesays | what 700x90k |
| 07:31.11 | Hmmhesays | derrrrr..... |
| 07:31.17 | *** join/#asterisk apardo (n=apardo@87.217.145.10) |
| 07:31.32 | *** join/#asterisk denzs (n=denzs@mail.henkel-kg.com) |
| 07:31.40 | Hmmhesays | 42,000,000 |
| 07:32.26 | Hmmhesays | yeah... anyway, my rant is done |
| 07:32.33 | denzs | hi, can anybody telle me if it is possible to control the states of the LEDs from a snom360 remotely, maybe by a php script triggered by an action url? |
| 07:32.45 | Hmmhesays | yes |
| 07:32.47 | Hmmhesays | sipsack |
| 07:33.15 | denzs | thank you :) |
| 07:33.32 | Hmmhesays | don't thank me yet, you'll need more research for sure |
| 07:33.52 | Hmmhesays | sipsack is a command line tool in linux that can generate sip messages |
| 07:35.01 | Hmmhesays | alright my stargate episode has arrived |
| 07:35.23 | denzs | i see ... i hoped it would be easier but thanks for the hint |
| 07:36.07 | flenders | hey, would this work? exten => 85,1,GotoIf($[${CALLERIDNUM} = <MY_PHONE_HERE> ]?10:20) |
| 07:36.29 | perd | sure |
| 07:36.47 | perd | dpeending on what you filled <MY_PHONE_HERE> with |
| 07:37.09 | Corydon76-home | not exactly the safest expression, or even the most up to date |
| 07:37.34 | Corydon76-home | $["${CALLERID(num)}" = "12345"] |
| 07:37.41 | Corydon76-home | That's a bit safer |
| 07:38.01 | flenders | oh I need the "" |
| 07:39.48 | flenders | Corydon-w: it works now, thanks mate |
| 07:40.07 | flenders | Corydon76-home: thanks mate |
| 07:40.35 | Corydon76-home | The quotes make the expression valid even when callerid is blank |
| 07:41.35 | potsboy | hi all, is it possible to dial out from a conference room.. the reason is i want to be able to bridge a audio agi into the conference? |
| 07:46.03 | CrashHD | eeek |
| 07:46.05 | CrashHD | trillian on irc |
| 07:46.06 | CrashHD | ewww |
| 07:46.14 | mitcheloc | shh, i'm guilty as well |
| 07:46.24 | CrashHD | lol |
| 07:46.25 | CrashHD | eek x 2 |
| 07:46.47 | CrashHD | whatever floats your boat I suppose |
| 07:47.28 | CrashHD | as long as it doesn't rock mine |
| 07:50.37 | sevard | don't rock the boat |
| 07:50.39 | sevard | rock the boat baby |
| 07:51.05 | JT | or sink yours |
| 07:54.57 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491673F.dip.t-dialin.net) |
| 07:55.19 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:59.24 | *** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it) |
| 08:00.34 | flenders | does the dialplan execute anything after the hangup? |
| 08:00.53 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:01.06 | flenders | I mean, can I have priority 6,Hangup and 7,SomethingElse? |
| 08:02.17 | nibbler_de | i don't think so |
| 08:02.29 | [hC] | anyone ever experienced an issue with asterisk (maybe most importantly, a linksys ATA) where it seems to randomly pick up DTMF "#" during a call when the person doesnt hit it? |
| 08:02.44 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 08:02.45 | [hC] | I talked ot people quite often where this happens to them, and of course my asterisk is configured to use # for 'transfer' |
| 08:02.49 | nibbler_de | flenders: depending on what you plan to do you could exec something before hanging up |
| 08:03.25 | flenders | nibbler_de: just want to create a call file, for call-back |
| 08:03.48 | *** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net) |
| 08:04.04 | shepimport | hey guys - anyone know how to set a preference on trunk per ext... vs dialing prefix? |
| 08:04.12 | nibbler_de | +of |
| 08:05.01 | perd | someone is going to make a trixbox comment |
| 08:05.04 | perd | or freepbx comment |
| 08:05.05 | Bhaal | [11:18:10] * wunderkin wonders why Bhaal does that |
| 08:05.05 | Bhaal | [11:18:30] <wunderkin> more frequently now too i think |
| 08:05.08 | perd | and i'm goign to eat food |
| 08:05.09 | nibbler_de | flenders: then just do it before the hangup so you can even give an audible indication of the process that informs the customer of the option :) |
| 08:05.37 | Bhaal | Just so you guys know, I do that to detect the ONJOIN spambots which occasionally come in here and a number of other channels... |
| 08:06.16 | Bhaal | By 'that' wunderkin is talking about me parting/joining the channel every hour... |
| 08:06.44 | nibbler_de | Bhaal: what did you do? |
| 08:06.50 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:06.50 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:06.54 | Bhaal | That... |
| 08:07.07 | nibbler_de | mh |
| 08:08.35 | nibbler_de | i see. well... it's the lesser evil i guess :) then again i don't join that often but keep idling for days. also freenode offers good spamprotection already (only registered users can /msg orso) |
| 08:11.01 | Bhaal | nibbler_de: And the bots are becoming more sophistocated... |
| 08:11.53 | JT | umm |
| 08:12.12 | *** join/#asterisk shinux__ (n=shinux@196.220.27.122) |
| 08:12.42 | JT | flenders: a guru can correct me if i'm wrong, but i think certain commands like Dial() end dialplan execution if they were successfully answered, then hung up by a user |
| 08:12.53 | JT | so often it doesn't even get to Hangup() |
| 08:12.59 | JT | depends on dialplan structure |
| 08:16.31 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:21.36 | flenders | JT: I want to create .call files after it was hungup, but I guess all i need to do is create them before hanging up |
| 08:22.20 | flenders | JT: that is for callback, filtering my callerid |
| 08:23.49 | flenders | when I get called back, it asks me for a pin using Authenticate(), and if successful, it lets me dial out |
| 08:24.18 | JT | so pretty much exactly what you're trying to do is callback? |
| 08:24.33 | flenders | yes |
| 08:24.37 | JT | call asterisk, it detects callerid, hangs up, calls you back, asks for auth, gives you dialtone |
| 08:24.46 | flenders | yeah |
| 08:24.47 | JT | it's pretty hard to avoid using AGI |
| 08:25.14 | flenders | why is that? |
| 08:25.53 | JT | here's one example of callback that uses a .php script, but it gives you the general idea even if you don't want to use php http://www.junghanns.net/en/callback.html |
| 08:25.58 | JT | you need to make a call file |
| 08:26.54 | JT | not easily done self-contained within dialplan |
| 08:26.54 | flenders | I don't need anything fancy, just need to be able to make international calls from my mobile, while on the road |
| 08:26.55 | JT | sure |
| 08:27.12 | JT | you need to make a callfile, or tell asterisk to make a call using the manager interface, it's pretty much unavoidable |
| 08:27.42 | flenders | yeah, I'll make the call file using System() |
| 08:28.06 | flenders | just a very simple shell script |
| 08:28.39 | JT | well it needs to wait until the current call is finished |
| 08:29.10 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 08:29.14 | zoa | Yo Yo |
| 08:29.15 | flenders | if it hangs up right after creating the call file, should be fine, no? |
| 08:29.45 | *** join/#asterisk apardo (n=apardo@87.217.145.10) |
| 08:30.06 | JT | give the network a few second to reset the call status of your mobile |
| 08:30.30 | zoa | does anybody have the email addy for damin ? |
| 08:30.32 | flenders | ok |
| 08:31.02 | flenders | what happens if on System() I run something on the background with '&', does it work |
| 08:31.03 | flenders | ? |
| 08:31.10 | JT | no idea |
| 08:31.15 | flenders | a simple sleep 5, would do it then |
| 08:31.20 | JT | agi isn't actually hard |
| 08:31.36 | JT | but let me know if system() works |
| 08:31.55 | yansolo90 | does anyody have send/received fax with asterisk 1.4 ? |
| 08:32.06 | flenders | will test it in a bit |
| 08:32.10 | JT | make sure you cp or mv the callfile into the spool directory, of course, do not create the callfile from scratch in the spool directory |
| 08:36.58 | flenders | well, sleep 10 ; echo blahblah doesn't work, it only executes the first command |
| 08:37.12 | flenders | let me see if it runs a shell script on the background |
| 08:40.14 | flenders | haha, it runs shell scripts on the background |
| 08:40.20 | flenders | with '&' |
| 08:40.32 | flenders | exten => 85,11,System(/tmp/testing.sh&) |
| 08:41.42 | flenders | next thing will be to create the call file by sending an sms |
| 08:41.42 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:42.07 | JT | how will that work? |
| 08:42.22 | flenders | through e-mail and procmail |
| 08:42.26 | JT | ah ok |
| 08:42.53 | flenders | 20 cents, and I can make international calls using voip rates |
| 08:43.11 | JT | heh |
| 08:43.27 | JT | free calls to mobiles? |
| 08:43.59 | flenders | not free, but cheap |
| 08:44.11 | JT | damn |
| 08:44.22 | JT | here calls to mobiles are the most expensive thing |
| 08:44.28 | JT | way cheaper to call overseas |
| 08:44.42 | sevard | i dumped like 4 dollars in to a payphone just now |
| 08:44.49 | flenders | where are you? |
| 08:44.55 | sevard | i'm on vacation in SF |
| 08:45.14 | JT | .au |
| 08:45.17 | sevard | for a flipping 5 minute phone call to a suburb one hour north |
| 08:45.34 | flenders | JT: yeah, I know what you mean |
| 08:46.03 | flenders | I think from landlines, they're over 30 cents a minute, aren't they? |
| 08:46.23 | JT | around 30-40cpm traditionally from landlines |
| 08:46.38 | flenders | well, on engin they're 27, I think |
| 08:47.14 | JT | yes, 27 for a standard plan, 23 or 20c/min on the higher volume plans |
| 08:49.12 | JT | flenders: you in .au? |
| 08:49.19 | flenders | JT: yeah, sydney |
| 08:49.30 | JT | cool |
| 08:49.43 | flenders | pretty close to the city, actually |
| 08:49.45 | JT | so what was the verdict on the callback thing? |
| 08:50.17 | flenders | shell script is easier, I think, as System() commands can run on the background. :o) |
| 08:50.33 | JT | ah yes, i was talking to you earlier about telstra, silly me |
| 08:50.38 | JT | but does it work for callback? |
| 08:51.11 | flenders | yeah, it creates a call file, and mv it to the outgoing directory |
| 08:51.23 | JT | does the call come? |
| 08:51.38 | flenders | it did this morning |
| 08:51.48 | JT | hrm? |
| 08:51.49 | flenders | placing them manually |
| 08:51.54 | JT | ah |
| 08:51.59 | JT | so you haven't tried fully |
| 08:52.13 | flenders | I'm just about to finish the shell script that will create them automatically |
| 08:52.19 | flenders | not yet |
| 08:52.32 | flenders | will do in a bit |
| 08:52.59 | JT | if it's only to one number, you can have a static .call file around |
| 08:53.05 | JT | waiting to be cped |
| 08:53.28 | flenders | yeah |
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| 08:54.20 | tzafrir | for jeremy_g: |
| 08:54.23 | tzafrir | ~seen oej |
| 08:55.08 | jbot | oej <n=olle@apollo.webway.se> was last seen on IRC in channel #asterisk, 18d 9h 35m 1s ago, saying: 'mountainm2k: Debug messages are *NOT* error messages. Turn them off please.'. |
| 08:55.35 | JT | flenders: i'm looking for a voip provider that has .au calls to mobiles for under 20c/min, it's a tall order |
| 08:55.54 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 08:56.40 | flenders | JT: it works! |
| 08:56.57 | JT | sweet |
| 08:57.10 | flenders | JT: i remember one that had mobile calls for 23, I think |
| 08:57.21 | flenders | but the quality wasn't very good |
| 08:57.22 | JT | engin has them for down to 20c |
| 08:57.32 | flenders | but that's on a business plan |
| 08:57.33 | JT | but it's $50/mo |
| 08:57.41 | JT | 450 included local/national calls |
| 08:57.55 | flenders | I don't think make 450 local calls on a month |
| 08:57.56 | flenders | :D |
| 08:58.14 | flenders | *think WE make* |
| 08:58.20 | JT | i tried to sign up for one with 19.5c/min, but i couldn't even make it top up my account (prepay only), to try out mobile calls |
| 08:58.42 | flenders | wow, that is cheap |
| 08:58.46 | JT | what asterisk hardware do you have now? |
| 08:59.00 | flenders | an AMD 2200+ I think |
| 08:59.12 | JT | what card though |
| 08:59.29 | flenders | the digium TDM400P |
| 09:00.05 | flenders | sorry TDM04B |
| 09:00.06 | JT | are you sticking with that, or considering isdn |
| 09:00.17 | JT | same thing, tdm04b is a tdm400p config |
| 09:00.30 | flenders | ahh, ok |
| 09:00.42 | flenders | at the moment, we're sticking with what we have |
| 09:00.47 | flenders | we got 2 of those |
| 09:00.53 | JT | hmm |
| 09:00.57 | flenders | with 4FXO each |
| 09:01.11 | flenders | I guess they're pretty easy to sell |
| 09:01.19 | JT | just keep in mind disconnect supervision is a bitch with analogue |
| 09:01.24 | JT | and you can't set callerid |
| 09:01.29 | JT | heh, probably |
| 09:01.47 | flenders | even though telstra says you can |
| 09:01.49 | flenders | :D |
| 09:02.06 | flenders | I had problems with disconnect thingy already |
| 09:02.21 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:02.34 | flenders | on incoming sip calls, asterisk hangs up the call as soon as I hang up on the calling phone |
| 09:02.57 | flenders | hey, there's something weird with my zap channels... if I place an incoming call, and when it the auto-attendant answers, if I hangup, it doesn't hangup the call, it waits for the timeout. |
| 09:03.02 | flenders | I posted that before |
| 09:03.22 | flenders | is that what you're talking about? |
| 09:04.04 | JT | yes, a computer cannot accurately detect an analogue line hanging up without help |
| 09:04.14 | JT | ask telstra to enable disconnect supervision on the lines |
| 09:04.27 | flenders | oh, you can do that? |
| 09:04.47 | flenders | I found this URL on google: http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html |
| 09:05.07 | JT | they call it Reverse On Idle Condition |
| 09:05.22 | JT | they reverse line polarity when the far-end caller hangs up |
| 09:05.33 | flenders | great! |
| 09:05.34 | JT | zaptel can detect it |
| 09:05.55 | JT | it's free iirc |
| 09:06.02 | flenders | I'm pretty sure the fuckers at the business center won't have a clue about what I'm talking about |
| 09:06.36 | JT | http://www.voip-info.org/wiki/view/Australia+Asterisk+Details |
| 09:06.52 | JT | you must always get bad luck heh |
| 09:07.06 | JT | or maybe people with isdn always get better support :P |
| 09:08.11 | flenders | that's probably it... hopefully we'll get ISDN soon |
| 09:09.53 | flenders | wow! 8:10, no wonder I'm starving |
| 09:10.03 | flenders | have a good one mate |
| 09:11.31 | JT | you too |
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| 09:45.07 | yansolo90 | hi, does anyody have send/received fax with asterisk 1.4 ? |
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| 09:50.31 | Aces1Up | is anyone here familiar with the bandtel service? |
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| 10:02.37 | angryuser | good day |
| 10:03.55 | zoa | YoYo |
| 10:05.19 | angryuser | i have some misdn problems, i have installed aster 1.4 misdn was detected, so i have downloaded misdn-init for b410p card make & make install , my cars is detected; but when i do /etc/init.d/misdn-init start, i got FATEL module mISDNcapi not found |
| 10:05.39 | angryuser | *FATAL |
| 10:06.07 | angryuser | i did kernel patches, dont really know wht to do |
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| 10:20.53 | angryuser | i have searched i dont have module hfcmulti under /proc/modules/ but mISDN have reported no problems during install |
| 10:21.31 | angryuser | maybe i have installed wrong modules? |
| 10:25.00 | SheriF_SpacE | i'm using asterisk 1.4 + asterisk-gui and i'm getting this error app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) i have a TDM400P card what is wrong ? |
| 10:28.12 | JT | angryuser: misdn has two parts |
| 10:28.18 | angryuser | gm |
| 10:28.21 | angryuser | and? |
| 10:28.35 | JT | they both need to be installed |
| 10:28.36 | angryuser | i forgot something? |
| 10:28.45 | angryuser | i have installes M |
| 10:28.46 | JT | i can't tell |
| 10:29.02 | angryuser | misdn and mISDNuser, |
| 10:29.06 | JT | yes |
| 10:30.16 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:30.54 | angryuser | i installed them manually, then, i have tryed to do make b410p in zaptel repository, no errors upon install, but still module mISDN_core and couple others, not founs |
| 10:31.09 | angryuser | *found |
| 10:33.38 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 10:36.30 | *** join/#asterisk viperdude (n=viperdud@195.74.96.120) |
| 10:36.39 | Aces1Up | got a quick question, can anyone suggest a low budget qos router and switch combination? 24-port switch? |
| 10:37.38 | *** join/#asterisk reber (i=reber@gateway/tor/x-e757d1d235cbb1ca) |
| 10:44.02 | angryuser | any help for my issue please, misdn is compiling without errors but no modules in /proc |
| 10:46.03 | angryuser | so modprobe gives nothing, aster starting with chan_misdn.so loaded |
| 10:48.04 | *** join/#asterisk s1gny|wrk (n=s1gny@p54914D1A.dip.t-dialin.net) |
| 10:48.19 | *** part/#asterisk s1gny|wrk (n=s1gny@p54914D1A.dip.t-dialin.net) |
| 10:53.11 | L|NUX | i got this error in sip debug |
| 10:53.12 | L|NUX | Invalid information element contents |
| 10:53.41 | L|NUX | i got did from china telecom and trying to bind it into my machine but its giving me following error "Invalid information element contents" |
| 10:55.08 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 10:55.46 | viperdude | is anyone else getting timeout on www.digium.com ? |
| 10:56.08 | viperdude | www.digium.com even lol |
| 10:56.09 | penguinFunk | yep |
| 10:56.26 | viperdude | hmm site must be down |
| 10:56.38 | penguinFunk | dead |
| 10:57.26 | viperdude | i guess its to early in the morning for those guys to sort it |
| 10:58.32 | Aces1Up | can you guys suggest a good low budget qos router to be used in a sip trunking envoiornment over a broadband link? |
| 10:59.18 | viperdude | I like Zyxel Prestige range with the built in VoIP IAD |
| 11:00.02 | viperdude | NAT'd stuff behind it works like a dream |
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| 11:03.12 | Aces1Up | viper... hrmm. |
| 11:03.50 | Aces1Up | well.. i'm running this over a cable broadband, do you have any suggestions for a standard qos voip router? |
| 11:04.18 | viperdude | sorry most of the stuff i do is DSL or leased line |
| 11:10.55 | JT | angryuser: if you get it working, or you progress further with your misdn problem, please let me know, i'm interested |
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| 11:18.34 | angryuser | JT:kk |
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| 11:29.56 | *** join/#asterisk mavior (n=Miranda@88-149-160-22.f5.ngi.it) |
| 11:30.52 | mavior | hello everybody...can please somebody help me with a sip 404 strange error ? (strange behaviour because i'm logged in with xlite in my ast server, but i can't get incoming calls on my xlite) |
| 11:32.22 | stoffell | mavior try pasting your CLI output on pastebin or something.. |
| 11:32.27 | mavior | ok |
| 11:34.39 | mavior | is there such a command to outpout to a file the entire cli ast debug messages? |
| 11:36.23 | stoffell | mavior; if logging is enabled you might find it in /var/log/asterisk/full .. or any other location :) |
| 11:37.11 | mavior | i have to just do a "sip set debug", isn't it? |
| 11:37.36 | stoffell | if you really want to log all sip messages, yes.. |
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| 11:39.29 | mavior | ok...wait |
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| 11:41.14 | angryuser | after zaptel 'make b410p' and zaptel install(no errors) when i do make in aster 1.4 i have this http://www.pastebin.ca/319460 |
| 11:42.39 | angryuser | mISDN is killing me |
| 11:44.01 | *** join/#asterisk apardo (n=apardo@87.217.145.213) |
| 11:46.27 | mavior | doing an "ls" in my /var/log/asterisk , report this items cdr-csv cdr-custom event_log messages queue_log, so probably i have not full log option activated (messages does not report all the sip debug)... |
| 11:47.16 | stoffell | mavior; ok, check your logger.conf in your /etc/asterisk, you can enable it there.. |
| 11:47.46 | mavior | roger :) |
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| 12:18.49 | mavior | stoffell: i have to go right now,brb in 2 hours, if you are here i would like to show you my debug |
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| 12:34.31 | Chris-NB | hi |
| 12:35.20 | Chris-NB | how can I reach that one of my speed-dial buttons on my cisco 7960 dials a number? what is the name of the parameter to define this in the cnf file? |
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| 12:37.47 | *** join/#asterisk snitt (i=nem@kicsit.addikt.hu) |
| 12:37.53 | snitt | o/ |
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| 13:08.42 | DrukenLPY | wholly shit, where did the month go? |
| 13:08.47 | DrukenLPY | it's the 17th already? |
| 13:08.49 | monsted | that way |
| 13:14.13 | DrukenLPY | i need to find a payphone booth |
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| 13:28.07 | santibiotico | hi |
| 13:28.54 | DrukenLPY | god damn, check out kijiji for my city, someone has a phonebooth for sale that lives around the corner from me |
| 13:29.33 | cpm | buy it |
| 13:29.44 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 13:31.35 | simplexio | hi. is there way run applications from cli ? |
| 13:32.06 | simplexio | i know manager interface but its seems to pain just run one applocation command from there |
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| 13:33.35 | DrukenLPY | cpm: yeah i jumped on it, but no phone number :( |
| 13:33.45 | DrukenLPY | so i had to email the guy |
| 13:34.43 | cpm | I am trying to find a cheap brit phone booth |
| 13:35.37 | cpm | want to outfit it with a 'crank' phone and a TS-blahblah touchtone field phone adapter, and hook it to my asterisk system, and just have a cash-drop slot |
| 13:37.19 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 13:37.21 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
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| 13:41.18 | DrukenLPY | cpm: a cash drop slot? |
| 13:41.25 | puzzled | hi |
| 13:42.00 | DrukenLPY | i have a payphone already, and programmed it not to take any money, allows for free local calls, and give 5 mins for LD calls... |
| 13:42.08 | DrukenLPY | just need the booth for it :) |
| 13:42.45 | DrukenLPY | but those brit booths, my god, 1200 lbs at least, made of cast iron, and glass |
| 13:43.23 | coppice | the original booths, with many small panes of glass? |
| 13:43.31 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com) |
| 13:43.38 | DrukenLPY | yeah |
| 13:43.57 | DrukenLPY | some guy had a bunch of them in brampton a few years ago, was selling them for a grand a peice |
| 13:44.33 | coppice | the old Dr Who style police phone booth might be more fun :-) |
| 13:44.43 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:44.45 | jeffik | Anybody know what is going on with Digium? site is down, bug tracker down too |
| 13:45.00 | DrukenLPY | well, the guy around the corner apparently has a superman type booth |
| 13:45.12 | *** join/#asterisk bkw__ (n=brian@82.152.100.241) |
| 13:46.29 | DrukenLPY | which would go well with my 80's style payphone |
| 13:52.01 | santibiotico | to use email notification in voicemail, i suppose i need a smtp server working in the asterisk box... |
| 13:52.09 | santibiotico | if so, which would you recommend me? |
| 13:52.52 | DrukenLPY | sendmail/qmail/postfix they all work |
| 13:53.27 | angryuser | i have compilled zaptel 1.4 on mandriva 2007 -no errors, but when i do modprobe zaptel i got error, zaptel not found |
| 13:53.55 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 13:53.57 | angryuser | i ahve changed Subversion in makefile, but no changes |
| 13:54.33 | angryuser | EXTRAVERSION in src/linux ;à |
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| 14:01.12 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 14:02.26 | Corydon76-home | angryuser: you need to change both /usr/src/linux/Makefile as well as /usr/src/linux/include/linux/version.h |
| 14:02.46 | angryuser | Corydon76-home: i did, no effect |
| 14:02.49 | Corydon76-home | angryuser: Strip "custom" off the end |
| 14:03.08 | De_Mon | is there a cli way to see descriptions of cli only commands like 'meetme' |
| 14:03.17 | Corydon76-home | angryuser: what's the output of 'uname -a'? |
| 14:03.59 | De_Mon | I found a desc on voip-info but just typing meetme doesnt tell you everything |
| 14:03.59 | De_Mon | err 'help meetme' |
| 14:03.59 | De_Mon | meetme (un)lock|(un)mute|kick|list <confno> <usernumber> |
| 14:04.16 | De_Mon | implies confno and usernum are required for all commands -- but its not |
| 14:05.02 | angryuser | Corydon76-home:2.6.17-5mdv |
| 14:05.59 | Corydon76-home | angryuser: okay, so you need to make the EXTRAVERSION=-5mdv |
| 14:06.49 | angryuser | Corydon76-home: i did that, same in version.h and after make clean &make linux26 |
| 14:06.57 | angryuser | no changes:( |
| 14:07.04 | Corydon76-home | angryuser: make install |
| 14:07.33 | angryuser | Corydon76-home: and make install of course;) |
| 14:07.52 | Corydon76-home | Check where it installed it: ls -l /lib/modules |
| 14:08.10 | Corydon76-home | or: find /lib/modules -name zaptel\* |
| 14:08.58 | cpm | DrukenLPY, yeah, a cash drop slot, so you could pay for the call if you wanted to. But you don't have to. kinda like a donate button |
| 14:09.24 | angryuser | Corydon76-home: gm gm it is still in 'custom' under lib/modles |
| 14:09.24 | Corydon76-home | angryuser: what's the output of that command? |
| 14:09.37 | angryuser | Corydon76-home::very strange |
| 14:09.39 | Corydon76-home | angryuser: you changed the Makefile? |
| 14:09.51 | angryuser | Corydon76-home: yes i am sure of it |
| 14:10.12 | angryuser | Corydon76-home: i will delete modules and try to cean reinstall |
| 14:10.17 | zoa | hey ho |
| 14:10.20 | angryuser | *clean |
| 14:10.22 | zoa | gooooooooooooooooooood afternoon |
| 14:10.31 | Corydon76-home | angryuser: I'm running that exact version |
| 14:10.38 | Corydon76-home | angryuser: and it's working here |
| 14:10.53 | Corydon76-home | angryuser: those two changes are all I did |
| 14:11.09 | angryuser | Corydon76-home:ik thank you il try again |
| 14:11.20 | Corydon76-home | angryuser: oh, and make sure that your kernel source version matches your running kernel |
| 14:11.42 | angryuser | it is, installed by default |
| 14:11.50 | Corydon76-home | If you upgraded from an earlier version, chances are, it upgraded your kernel source without upgrading your kernel |
| 14:12.33 | angryuser | Corydon76-home:it was a clean install |
| 14:12.47 | Corydon76-home | Okay |
| 14:13.07 | Corydon76-home | This one wasn't, and I had to do a bit of cleanup because of it |
| 14:14.07 | Corydon76-home | angryuser: grep -R mdvcustom /usr/src/linux |
| 14:15.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.21 | angryuser | kernel version 2.6.17-5mdvcustom |
| 14:16.11 | Corydon76-home | angryuser: out of what file? |
| 14:16.56 | Corydon76-home | angryuser: the alternative is simply to restore that text in the two source files and just install a custom kernel |
| 14:17.02 | angryuser | Corydon76-home: from /usr/src/linux/.config |
| 14:17.26 | angryuser | and /usr/src/linux/.kernelrelease |
| 14:17.41 | Corydon76-home | angryuser: what's the output of 'uname -r' ? |
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| 14:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:18.12 | angryuser | 2.6.17-5mdv |
| 14:18.56 | Corydon76-home | Well, I've given you pretty much everything I know how to do |
| 14:19.29 | Corydon76-home | What's the output of: find /lib/modules -name zaptel\* |
| 14:19.33 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:19.45 | *** join/#asterisk andyshack (n=andyshac@reels.iinet.net.au) |
| 14:20.28 | robl^ | Mooooo! Morning, everyone |
| 14:20.45 | angryuser | the output is /lib/modules/2.6.17-5mdv/misc/zaptel.ko |
| 14:20.57 | Corydon76-home | angryuser: okay! that's the right place |
| 14:21.05 | andyshack | evening. just posted this in asterisknow although i think most people are asleep. ive instlled the same asterisknow on two seperate compaq servers, one with higher specs than the other, although the high spec machine takes at least 10 seconds to render the gui over the lcoal lan where as the low spec machine flys. any ideas ? |
| 14:21.08 | Corydon76-home | angryuser: try 'modprobe zaptel' now |
| 14:21.30 | angryuser | Corydon76-home: gm not found;) |
| 14:21.36 | Corydon76-home | angryuser: reboot |
| 14:22.06 | angryuser | Corydon76-home: ..in process |
| 14:22.12 | *** join/#asterisk ManxPower (n=manxpowe@92.sub-75-202-2.myvzw.com) |
| 14:22.19 | robl^ | morning ManxPower |
| 14:22.33 | ManxPower | mornings bad. |
| 14:22.47 | robl^ | only until coffee #3 |
| 14:22.52 | Corydon76-home | angryuser: is this an x86 32-bit install? |
| 14:23.20 | angryuser | Corydon76-home:i586 |
| 14:23.34 | Corydon76-home | angryuser: so it's not x86_64 ? |
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| 14:24.06 | angryuser | Corydon76-home:it is i686 |
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| 14:26.00 | Corydon76-home | angryuser: are you typing the exact command: modprobe zaptel |
| 14:26.06 | angryuser | Corydon76-home: thank you for help, when i will resolve it, i wil let you know |
| 14:26.16 | Corydon76-home | or what are you exactly typing? |
| 14:26.29 | angryuser | Corydon76-home: yes i am sure of it;) |
| 14:26.52 | Corydon76-home | And you're root? |
| 14:27.09 | angryuser | i have FATAL module zaptel not found |
| 14:27.32 | Corydon76-home | That doesn't make any sense |
| 14:27.40 | Corydon76-home | depmod -a |
| 14:27.57 | angryuser | Corydon76-home: sec |
| 14:28.26 | angryuser | Corydon76-home: before il try to clean reinstall of /librpi/zap/aster |
| 14:28.26 | Corydon76-home | Err, depmod -A |
| 14:28.41 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:28.46 | angryuser | to be sure that ther is no mistake |
| 14:30.39 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:34.20 | angryuser | Corydon76-home: ok i have news, zaptel installs to /lib/modules/2.6.17-5mdvcustom ;( |
| 14:36.14 | *** join/#asterisk _matt (n=matt@82-33-68-44.cable.ubr01.trow.blueyonder.co.uk) |
| 14:36.25 | Corydon76-home | angryuser: dunno |
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| 14:44.58 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:46.46 | yansolo90 | hi, does anyody have send/received fax with asterisk 1.4 ? |
| 14:50.47 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
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| 14:59.12 | robl^ | hrmm.. just updated to SVN 1.4 branch. it ain't working right.. getting " -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.101" every second |
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| 15:06.34 | *** mode/#asterisk [+o mog] by ChanServ |
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| 15:15.14 | yansolo90 | hello, need some help with MWI on cisco phone 79XX |
| 15:15.20 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 15:15.43 | yansolo90 | a guy said : MWI failing to activate - BAD NOTIFY. I sniffed the asterisk MWI SIP messages and did the same for callmanager 5 SIP messages. Compared the two and determined the problem can be fixed in the SEP<mac>.cnf.xml file. My config file is based on the example provided on this wiki page. In the line section, replace the contact XML parameter 7b452e87-4496-4762-e11f-b26751a1884b with the extension/SIP ID configured in ast |
| 15:15.43 | yansolo90 | erisk. |
| 15:15.44 | jtexter3 | Is it possible to use the InstantMessage feature of the Polycom phones with Asterisk? |
| 15:17.18 | yansolo90 | but i don't know what "extension/SIP ID configured in asterisk" is ...? |
| 15:17.52 | *** join/#asterisk thomi (n=thomi@host81-149-152-221.in-addr.btopenworld.com) |
| 15:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:18.10 | yansolo90 | the wiki page : http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx |
| 15:20.20 | mercestes | yansolo90: The sip username. |
| 15:21.42 | jtexter3 | Damn, looks like it doesn't |
| 15:22.04 | mercestes | jtexter3: I think katty had somethign working with Yahoo. |
| 15:23.05 | jtexter3 | mercestes: Okay, thanks |
| 15:25.07 | *** join/#asterisk drone1 (n=kova@tech.quentris.be) |
| 15:25.55 | penguinFunk | Hey guys, just installed the TDM2400p wildcard, got zaptel + zapata configured. got the wctdm24xxp module loaded for zaptel. I've also got asterisk installed and condfigured. I'm having problems that would seem like there is a physical hardware misconfiguration somewhere... |
| 15:26.00 | Crescendo | Can I use the Asterisk logo in print? |
| 15:26.06 | Crescendo | legally. |
| 15:26.41 | tzanger | Crescendo: I think you need to contact digium and ask for permission, but that's all there is to it |
| 15:26.56 | penguinFunk | i noticed that our modules came installed (2 FXO modules... each providing 4 FXO ports) on position 5 + 6 on the board |
| 15:26.59 | Gido-E | Crescendo i would ad a note, asterisk logo bla... |
| 15:27.07 | tzanger | Crescendo: they will likely want to see what you want to print and make sure you're not saying that you're working for them or basically misrepresenting them |
| 15:27.21 | penguinFunk | that would mean that ports 17-24 are the 8 we need |
| 15:27.42 | penguinFunk | so ive configured zaptel.conf and zapata.conf to use these channels |
| 15:27.57 | penguinFunk | bbut the problem arises when i want to reference these in extenstions.conf |
| 15:28.10 | penguinFunk | Zap/1 = FXO port 17 ? |
| 15:28.53 | penguinFunk | when i use a voip phone to dial 9 for outside line it doesnt get as far as the PSTN |
| 15:30.35 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 15:31.02 | bhrobinson | I am having some issues with inbound calling through NAT. Is there anyone out there that can help me? |
| 15:31.46 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:34.52 | penguinFunk | how can i make sure the mappings are corret? |
| 15:34.55 | penguinFunk | correct even* |
| 15:35.55 | penguinFunk | hardware [17-24], zaptel.conf [17-24], zapata.conf [17-24] extensions.conf [Zap/? - Zap/?] ?? |
| 15:36.04 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 15:36.09 | angryuser | whatever my changes in makefile(extreversion) in sources of linux, it still installs zaptel under /lib/modules/2.6.17-5mdvcustom instead of /lib/modules/2.6.17-5mdv |
| 15:36.10 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 15:36.14 | TheCops | Hi |
| 15:36.31 | backblue | hi, anyone experienced linksys problems, that the phone continues to ring, even when it's busy on the other side? |
| 15:36.56 | *** join/#asterisk antlers (n=antlers@ip70-189-187-211.lv.lv.cox.net) |
| 15:37.14 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 15:37.37 | TheCops | I'm using ** as transfer (feature.conf) and the number 7 for parking an extension. Currently, when I'm parking an incomming call all work great, when I'm taking back the call, and trying to put back into the parking slot, I can't compose the **. The remote caller hear the DTMF. Someone know why^ |
| 15:38.40 | phearless | ahhhhh!!!! |
| 15:38.52 | phearless | I am trying to make the transfert to work, too |
| 15:40.07 | TheCops | phearless, a second transfer on the same line^ |
| 15:40.08 | TheCops | ? |
| 15:40.16 | phearless | no |
| 15:40.20 | phearless | just a normal one |
| 15:40.27 | phearless | on another voip phone |
| 15:40.50 | TheCops | put tT in your Dial command, and add your string into feature.conf for the blind transfer |
| 15:42.35 | phearless | tT ? |
| 15:43.22 | phearless | I use a Linksys SPA942 |
| 15:43.51 | TheCops | in Asterisk |
| 15:43.58 | TheCops | in your Dial cmd for your incomming or outgoing call |
| 15:43.59 | TheCops | put tT |
| 15:44.11 | TheCops | phearless, http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 15:44.36 | wunderkin | do you know what t and T do? |
| 15:44.46 | TheCops | yeah |
| 15:44.54 | TheCops | permit the caller or callee to transfer call |
| 15:45.20 | TheCops | I don't know his scenario to tell him what tT to take :) |
| 15:45.35 | wunderkin | not or.. and.... but .. ok |
| 15:46.35 | phearless | ;FOR LOCAL CALLS |
| 15:46.35 | phearless | exten => _2XX,1,Dial(SIP/${EXTEN},10,tT) ;SIP, local, for numbers like 2XX, 10s dial |
| 15:46.35 | phearless | exten => _2XX,2,VoiceMail(${EXTEN}@default) ;after the 10s, voicemail |
| 15:46.50 | TheCops | or or and, same thing |
| 15:47.19 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 15:47.49 | wunderkin | a nubb wont know the difference... and here he goes.. |
| 15:48.46 | wunderkin | phearless, do a show application dial before you blindly put that in there |
| 15:49.42 | TheCops | lol |
| 15:49.52 | TheCops | You telling me that I'm a nubb^ |
| 15:50.15 | TheCops | not a nubb, a french person. |
| 15:50.16 | wunderkin | no |
| 15:50.17 | *** join/#asterisk apardo (n=apardo@87.218.190.232) |
| 15:50.18 | TheCops | :) |
| 15:54.08 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 15:54.39 | x86 | phearless: you like the SPA942? |
| 15:54.47 | ctooley | I'm having some issues with using Macros. It seems that a hangup will get caught inside a macro but only a couple of steps are executed |
| 15:57.36 | backblue | phearless: do you have a 942? |
| 16:01.14 | x86 | backblue: he said he did :) |
| 16:01.37 | x86 | 09:43 <phearless> I use a Linksys SPA942 |
| 16:01.57 | ardaei | anyone is using TDM2400 here? |
| 16:02.41 | TheCops | I'm using ** as transfer (feature.conf) and the number 7 for parking an extension. Currently, when I'm parking an incomming call all work great, when I'm taking back the call, and trying to put back into the parking slot, I can't compose the **. The remote caller hear the DTMF.The transfer failed to parking. Someone know why? |
| 16:05.08 | ardaei | if i rememeber correctly, there was some issues for tranfering calls and parking |
| 16:05.13 | ardaei | what ver u r using |
| 16:07.33 | backblue | x86: and you? |
| 16:07.48 | x86 | backblue: i use Polycom IP601's |
| 16:08.01 | x86 | backblue: but i'm looking for a mid-grade phone for lower-budget projects |
| 16:08.16 | mercestes | x86: Polycom IP501, Polycom IP301 |
| 16:08.31 | x86 | mercestes: IP301 doesnt even have full duplex speakerphone |
| 16:08.32 | _DAW | Aastra 480i |
| 16:08.54 | mercestes | Invest heavily in Grandstream. By grandstream phones, grandstream stocks, grandstream ATA's. Grandstream is the *FUTURE*. It is the one phone, to rule them all! |
| 16:09.02 | robl^ | polycom 430s for lower end ;-) |
| 16:09.04 | mercestes | yea, ok, I'm lying again. |
| 16:09.09 | *** join/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 16:09.15 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 16:09.32 | x86 | robl^: looking for something with a little lower price point :) |
| 16:09.32 | robl^ | mercestes: have you been putting vodka in your cornflakes again |
| 16:09.37 | x86 | robl^: haha |
| 16:09.54 | robl^ | x86: Polycom 430s are like $130 |
| 16:09.58 | backblue | x86: last implementation we have done we have used 3 types of sip phones, polycom 601/401, f1000g, and spa-942. |
| 16:10.05 | phearless | <x86> phearless: you like the SPA942? |
| 16:10.07 | phearless | yes it's good |
| 16:10.13 | backblue | polycom's only have one thing i need to solve |
| 16:10.19 | x86 | robl^: so are SPA942's are like $109, my cost :) |
| 16:10.26 | mercestes | Vodka and milk. It's a white russian cornflake breakfast. |
| 16:10.28 | backblue | spa942, when they get busy, the continue ringing... stupidly. |
| 16:10.36 | backblue | s/the/they |
| 16:10.53 | backblue | f1000g, lost the registration with dlink ap's |
| 16:11.10 | x86 | backblue: do they have a "reject" button where you can reject inbound calls when you're on the phone? |
| 16:11.13 | backblue | they all have troubles, but the polycom are the better ones, so the price! :) |
| 16:11.22 | backblue | x86: which ones? |
| 16:11.30 | x86 | SPA942 |
| 16:11.33 | *** join/#asterisk Op3r (i=Op3r@203.82.37.211) |
| 16:11.43 | backblue | i think yes. |
| 16:12.21 | backblue | i dont have touched almost the spa-942 ones, because it was not provided by us. |
| 16:12.21 | mercestes | backblue: Have you tried disabling call waiting and setting "calls per line" to 1 and then only adding *1* line to the SPA942? |
| 16:12.24 | backblue | mercestes: no, why should i need to do that? |
| 16:12.50 | backblue | mercestes: the problem it's, when i dial from the spa-942, not when i dial *TO* spa-942. |
| 16:12.57 | Op3r | any one knows a cheap but good VOIP provider for call centers? |
| 16:13.41 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:15.01 | backblue | Op3r: cheap != quality. |
| 16:15.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:15.06 | Op3r | yep I know |
| 16:15.07 | Op3r | :( |
| 16:15.17 | backblue | and you should look for providers, near too you. |
| 16:15.32 | *** join/#asterisk reber (i=reber@gateway/tor/x-c63d8a64a17e0c92) |
| 16:16.23 | mercestes | backblue: oh, so the SPA942 is providing it's own ringback? |
| 16:16.40 | ManxPower | Uh, all SIP devices provide their own ringback |
| 16:17.50 | *** join/#asterisk bkw_ (n=brian@82.152.100.241) |
| 16:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:19.54 | Sweeper | arg. I'm getting automagical hangups as soon as I try to dial out on my pri :/ |
| 16:20.43 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 16:21.20 | drone1 | anybody with Queue experience |
| 16:22.00 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
| 16:22.15 | drone1 | I'm calling my queue with one member |
| 16:22.18 | tutt9876 | hi |
| 16:22.20 | PupenoR | drone1: I'd said just ask your question. |
| 16:22.23 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:22.29 | drone1 | this member sends a "603 decline" |
| 16:22.30 | Katty | mew, hi, etc. |
| 16:22.44 | drone1 | although he's free and ready to answer |
| 16:23.07 | tutt9876 | with AGI Request I get a max size of 20 characters with accountcode: can I supersize this variable? |
| 16:23.26 | *** part/#asterisk rycan (i=ryan@nat/digium/x-3ba3fc57ae44fc08) |
| 16:23.47 | *** join/#asterisk Marty-OTT (n=marty@host-209-50-87-3.dyn.295.ca) |
| 16:23.49 | tutt9876 | is AGI can be changed ? |
| 16:24.08 | tutt9876 | any idea? |
| 16:24.24 | Marty-OTT | Morning.. I decided to test this Mediatrix box... can anyone tell me how I can configure Asterisk to "Accept" a voip gateway to register to it? |
| 16:24.37 | Marty-OTT | Or is it just like any old peer phone? |
| 16:24.54 | tutt9876 | silence of desert |
| 16:24.59 | *** join/#asterisk xtr (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 16:25.03 | docelmo | tutt9876 yes do you know C? |
| 16:25.04 | De_Mon | Oof voip-info says meetme kick <conf> kicks everyone from a conference and it doesnt |
| 16:25.33 | tutt9876 | No without C compilation? |
| 16:25.52 | docelmo | tutt9876 you will need to change the channel structure for accountcode |
| 16:26.07 | docelmo | or cdr structure.. I forgot the exact location but I know it needs changed there |
| 16:26.14 | tutt9876 | docelmo: channel structure ? |
| 16:26.25 | docelmo | wanna pay me? :) |
| 16:27.14 | tutt9876 | is it in a .conf file? |
| 16:27.30 | mercestes | ManxPower: oh yea.... |
| 16:27.32 | docelmo | Marty-OTT you need to go check the wiki or google first |
| 16:27.36 | naitram | anyone know; if i use Dial in a macro, will the h extension in the macro work so that if the caller hangs up the h extension will take over in the macro |
| 16:27.45 | docelmo | tutt9876 nope.. a .h file if I remember correctly |
| 16:28.00 | mercestes | drone1: Fire whoever is sitting at that desk. |
| 16:28.07 | ManxPower | drone1: Decline can be caused by a codec mismatch, do not disturb being set, or reject button being pressed |
| 16:28.12 | tutt9876 | docelmo: so you mean without recompilation, it's not possible? |
| 16:28.18 | docelmo | nope |
| 16:28.29 | docelmo | this is why I asked if you knew C |
| 16:29.09 | tutt9876 | ok: I will change some others thins in my PBX: thanks for you advcie |
| 16:29.22 | tutt9876 | things, Seeu |
| 16:29.26 | *** part/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 16:29.53 | ManxPower | Marty-OTT: most of those SIP gateways do not support registration so they must be on a static IP address (The ONLY THING REGISTRATION DOES IS NOTIFY THE SERVER AS TO WHAT IP ADDRESS THE DEVICE IS ON IT DOES NOTHING ELSE) |
| 16:30.49 | Marty-OTT | ok.. |
| 16:30.59 | Marty-OTT | Well, the SIP gateaway has a static ip right now so that's cool |
| 16:31.06 | *** join/#asterisk s1gny|wrk (n=s1gny@p54914D1A.dip.t-dialin.net) |
| 16:31.24 | *** part/#asterisk s1gny|wrk (n=s1gny@p54914D1A.dip.t-dialin.net) |
| 16:31.32 | Marty-OTT | So then, I guess I have to tell the dialplan in ASterisk to route calls for these particular numbers to the gateway ten |
| 16:31.34 | Marty-OTT | then |
| 16:37.40 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:38.00 | naitram | this code http://pastebin.ca/319640 is an attempt to use the h extension in a macro, any idea why this will not work |
| 16:39.00 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 16:42.16 | mercestes | naitram: 0.0 Can you replace all that with a h,1,NoOp(Yay, I made it here.) maybe? |
| 16:44.18 | myiagy | i'm thinking exten => h has to be in the context that called the macro and not inside the macro itself maybe? |
| 16:44.36 | drone1 | thanks for your input guys, but it's none of the above :-) |
| 16:45.15 | drone1 | I have a more precise question: can I use a var in a queue member definition? |
| 16:45.20 | naitram | mercestes: im not sure, will that be executed within the macro context when the caller hangs up? |
| 16:45.32 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 16:45.33 | mercestes | drone1: In queues.conf? or in extensions.conf? |
| 16:45.45 | drone1 | queues.conf |
| 16:45.54 | mercestes | naitram: As sure as whatever yoru doing will be executed. |
| 16:46.07 | backblue | anyone have a linksys spa942 running there, that cant do a litle test with me? |
| 16:46.15 | naitram | myiagy: yeah, i was hoping to find out a definitive answer |
| 16:46.15 | mercestes | drone1: no. Unless I've been severely misled. |
| 16:46.34 | bhrobinson | I have 941 phones... |
| 16:46.38 | myiagy | naitram well, you could just try, and see if it is a definitive answer :) |
| 16:46.47 | mercestes | and he cant do a little test with you, sorry. |
| 16:47.31 | *** join/#asterisk svenna_ (n=svenna@p548D01B3.dip0.t-ipconnect.de) |
| 16:49.15 | naitram | mercestes: what exactly should be done? replace the exten => h with ? |
| 16:49.45 | mercestes | niatram: I wouldn't replace the exten => h. |
| 16:50.36 | mercestes | I would replace the ,1,System(blah.php omg, args. zOMG more args. OMG, is this written in Klingon? What are you doing to me? HELP! Rape! Could it be these mass of PHP scripts and arguemetns could be wrong?) with NoOp(Yay, I made it here!) |
| 16:51.09 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 16:51.19 | mercestes | See what I'm getting at? |
| 16:51.32 | naitram | mercestes: yeah, just for debug |
| 16:51.38 | mercestes | precisely. |
| 16:52.26 | naitram | mercestes: ok will try it. Thanks all |
| 16:52.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:53.43 | angryuser | anybody knows how te respawn prepertly misdn-init? i did rt:3:respawn:/etc/init.d/misdn-init start (in inittab) but i have some messages like rt respawning too quickly disabling for 5 minutes |
| 16:56.28 | *** join/#asterisk anthonyl (n=Anthony@65.4.1.65) |
| 17:00.40 | *** join/#asterisk CANO-1982 (i=alejandr@190.48.68.151) |
| 17:03.46 | jeremy_g | angryuser: #linux |
| 17:03.59 | jeremy_g | angryuser:i used to get such messages with getty |
| 17:04.50 | mercestes | How hard is the dcap exam? |
| 17:05.01 | L|NUX | i have [farrukh] context which is defined in sip.conf as well as iax.conf |
| 17:05.16 | L|NUX | in extensions.conf i have this value |
| 17:05.23 | L|NUX | exten => s,1,Dial(SIP/${EXTEN}@gtalk,60,tr) |
| 17:05.26 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 17:05.31 | L|NUX | when some one call using SIP it works |
| 17:05.32 | teknoprep | hi all |
| 17:05.39 | L|NUX | but when some one try to call from IAX |
| 17:05.43 | L|NUX | it will not work |
| 17:05.46 | teknoprep | g.729a ... how much inet bandwidth does that codec use? |
| 17:05.47 | L|NUX | either from sip |
| 17:06.49 | L|NUX | i am dialing like this from IAX guest@localhost/farrukh |
| 17:06.59 | *** join/#asterisk agarguest (n=agargues@CPE0015e985d544-CM00111ade4822.cpe.net.cable.rogers.com) |
| 17:07.05 | agarguest | hello |
| 17:07.16 | agarguest | i need some help :) |
| 17:07.22 | BSDTech | sorry |
| 17:07.22 | L|NUX | any one help ? |
| 17:07.32 | BSDTech | this help center is closed for lunch |
| 17:07.37 | backblue | teknoprep: in sip 23k |
| 17:07.38 | agarguest | noooo |
| 17:07.41 | agarguest | :) |
| 17:07.41 | backblue | i think |
| 17:07.49 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 17:07.50 | agarguest | what's for lunch? |
| 17:07.55 | agarguest | sounds like a good idea. |
| 17:08.02 | BSDTech | PLease take a number and come back in 1 hour |
| 17:08.03 | agarguest | i'll be back after my luch. |
| 17:08.20 | agarguest | i'll take the number 42 |
| 17:08.36 | BSDTech | Tuna on wheat |
| 17:08.52 | BSDTech | baked chips and great tea smothie |
| 17:08.56 | agarguest | with fresh crispy lettuce |
| 17:09.09 | BSDTech | yes and ripe red tamatos |
| 17:09.21 | BSDTech | light mayo |
| 17:09.26 | agarguest | but how do i call and order, when my asterisk box won't dial out? :) |
| 17:09.29 | nortex | mercestes, How long have you used Asterisk? |
| 17:09.42 | teknoprep | backblue, is it more on IAX ? |
| 17:10.08 | BSDTech | get a better provider |
| 17:10.12 | BSDTech | BV sucks |
| 17:10.22 | BSDTech | they seem to be down 90% of the time |
| 17:10.46 | teknoprep | http://www.netlogic.net/products-services/voip-wholesale.php been using these guys for larger accounts |
| 17:12.07 | nortex | mercestes, The dCAP is tough, but for some one with experince in Asterisk and an understanding of voip protocols and teleophany it is passable. |
| 17:12.08 | BSDTech | Account Minimum: $500.00 Per Month |
| 17:12.15 | BSDTech | wholy crap |
| 17:12.24 | teknoprep | they have a 100minimum |
| 17:12.28 | mercestes | nortex: Hrm. |
| 17:12.33 | teknoprep | but like i said if you need a large account with 50 trunks |
| 17:12.35 | teknoprep | its worth it |
| 17:12.42 | BSDTech | yeah |
| 17:12.48 | teknoprep | http://www.netlogic.net/products-services/voip-choice.php |
| 17:12.51 | mercestes | nortex: Been doing * for a few years...just wonderin gif the bootcamp was a requirement or if I had a chance just taking the exam. |
| 17:12.58 | teknoprep | BSDTech, check that link |
| 17:13.22 | BSDTech | looking |
| 17:13.39 | teknoprep | for small 4 channel numbers i use Voicepulse |
| 17:13.51 | teknoprep | they have a great deal for SOHO |
| 17:14.37 | backblue | can asterisk have stun suport? there is some patch or something? |
| 17:14.55 | nortex | mercestes, It used to be that you could just take the exam, I took it last year after a boot camp, but it might be worth proding Digium for an answer on that. They have been changing the program a bit since I took the test. |
| 17:14.56 | BSDTech | so you payy 100 + 2c a min |
| 17:15.02 | BSDTech | man |
| 17:15.09 | BSDTech | thats hars |
| 17:15.18 | BSDTech | harsh for home users |
| 17:15.20 | mercestes | nortex: Sweet |
| 17:15.26 | BSDTech | but not fo busniesses |
| 17:15.31 | Sweeper | that's fucking spendy |
| 17:15.36 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 17:15.38 | Sweeper | 2c a minute? yar |
| 17:15.49 | Sweeper | 1.1-1.5 is about normal |
| 17:15.56 | BSDTech | VoIP Choice - Business Quality Phone Service |
| 17:16.02 | Qwell[] | Sweeper: for a bargain basement provider, sure |
| 17:16.04 | BSDTech | its busniess grade |
| 17:16.17 | mercestes | nortex: I'll be sure to check up on that. Thanks. |
| 17:16.21 | Sweeper | mmmm |
| 17:16.22 | Qwell[] | 2c/min is right on target, unless you have a million minutes a month or something |
| 17:16.26 | nortex | mercestes, No problem |
| 17:16.34 | Sweeper | yea, not a huge diff |
| 17:17.47 | fourcheeze | thinking about recruiting someone to learn asterisk (uk based) any idea what would be a good starting set of qualifications/experience? |
| 17:17.53 | fourcheeze | and what's a good salary to pay? |
| 17:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:18.08 | Qwell[] | fourcheeze: for somebody you're going to train? |
| 17:18.24 | Qwell[] | hire an intern and pay them $7/hour... it won't make a big difference if they don't already know it |
| 17:18.34 | Qwell[] | The results will be the same |
| 17:18.57 | Sweeper | poor interns :( |
| 17:19.00 | mercestes | yea, but then they'd dump you and move to the US and you'd have to start all over. You should hire me and move me to the UK. |
| 17:19.17 | fourcheeze | define "intern" |
| 17:19.17 | Sweeper | hmmmm |
| 17:19.23 | fourcheeze | I don't think we have them over here |
| 17:19.29 | Qwell[] | fourcheeze: somebody just out of college |
| 17:19.32 | fourcheeze | ahh ok |
| 17:19.39 | mercestes | fourcheeze: Someone going to school and working "in their field" for their degree. |
| 17:19.44 | Qwell[] | fourcheeze: alternatively... |
| 17:19.45 | DrukenLPY | fourcheeze: low level, desperate for a job, fresh out of college |
| 17:19.46 | Sweeper | the guys who can't get jobs, since the dropouts already took them ;) |
| 17:19.51 | fourcheeze | hehe |
| 17:19.55 | Qwell[] | hire a consultant, pay them more, and have it working from the start |
| 17:20.07 | Qwell[] | end up paying less overall |
| 17:20.13 | fourcheeze | well I have a reasonable knowledge of * by now, but I have too many other bases to cover |
| 17:20.20 | fourcheeze | so I need someone to train ideally |
| 17:20.22 | DrukenLPY | Qwell: you aren't phishing now are you?? |
| 17:20.34 | Qwell[] | DrukenLPY: I don't do consulting |
| 17:20.35 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:20.37 | *** join/#asterisk ellisdee (n=ellisdee@cpe-72-181-171-187.houston.res.rr.com) |
| 17:20.47 | fourcheeze | ok, well that's another approach |
| 17:20.48 | DrukenLPY | thought you did for some reason.... |
| 17:20.52 | Qwell[] | DrukenLPY: I did :p |
| 17:20.57 | DrukenLPY | oh well pfft |
| 17:20.58 | Qwell[] | did != do |
| 17:21.05 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:21.05 | Sweeper | I'm a fan of the training idea |
| 17:21.10 | fourcheeze | how much should I pay a consultant for (a) development and (b) ongoing support? |
| 17:21.20 | Qwell[] | fourcheeze: whatever they charge, heh |
| 17:21.31 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 17:21.32 | Qwell[] | it depends on the consultant, on how good they (think they) are |
| 17:22.31 | DrukenLPY | hire the one that openly admits that shit happens, but they do their best to prevent an overflow of it :) |
| 17:22.39 | zoa | hehe lol |
| 17:23.01 | zoa | check how much they ask, then come to me and check if i can do better :p |
| 17:23.19 | mercestes | I should put my name on the list....lol |
| 17:23.42 | *** join/#asterisk Sephen (n=Sephen@c-69-245-182-37.hsd1.in.comcast.net) |
| 17:24.04 | Sephen | Is the System() command broken? I can't get it to do anything now that I've upgraded to 1.4.0 |
| 17:24.39 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:26.58 | fourcheeze | maybe I'll come back and invite tenders |
| 17:27.08 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:27.15 | fourcheeze | so when you say college leaver, you mean graduate? |
| 17:27.23 | fourcheeze | computer science or other? |
| 17:27.39 | Sweeper | don't discriminate against dropouts now ;) |
| 17:28.26 | fourcheeze | scraped through a physics degree |
| 17:28.31 | mercestes | fourcheeze: Or band. definately computer science, or band. or systems security. Or information technology. something along those lines. (dont' forget band) |
| 17:28.44 | fourcheeze | what's band? |
| 17:29.28 | Sweeper | band is where you play instruments |
| 17:29.33 | fourcheeze | yeah that's what I thought |
| 17:29.40 | Sweeper | but I don't know if there's such a thing as band degree |
| 17:29.44 | Sweeper | maybe music degree |
| 17:29.44 | fourcheeze | dropped out to follow a rock n roll career |
| 17:29.53 | *** part/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 17:30.54 | mercestes | I guess he went to go find himself a music major. |
| 17:31.59 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 17:34.58 | *** join/#asterisk shinux__ (n=shinux@196.220.29.93) |
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| 17:41.39 | Dr-Linux | i'm frequently below on CLI: |
| 17:41.40 | Dr-Linux | - Remote UNIX connection disconnected |
| 17:41.40 | Dr-Linux | <PROTECTED> |
| 17:41.53 | Dr-Linux | is it related to manager interface or what? |
| 17:44.13 | ManxPower | Dr-Linux: no, that happens when asterisk -r happens |
| 17:45.24 | Dr-Linux | ManxPower: yes, but that happends with my only server |
| 17:45.34 | Dr-Linux | ManxPower: what should i do to stop this |
| 17:45.43 | Dr-Linux | it's apearing very frequently |
| 17:46.35 | Dr-Linux | ManxPower: one connection/disconnected in 1 sec |
| 17:46.59 | mercestes | Dr-Linux: do you use FOP? |
| 17:47.14 | Dr-Linux | mercestes: no i don't use |
| 17:47.32 | L|NUX | will this rule work if i use characters instead of number |
| 17:47.35 | Dr-Linux | mercestes: i suspect it's happening with AMI |
| 17:47.46 | L|NUX | s,1,Dial(SIP/${EXTEN}@test) |
| 17:47.47 | ManxPower | Dr-Linux: you stop it by not having a script that uses "asterisk -rx" |
| 17:47.49 | mercestes | Dr-Linux: Hrm. I think FOP has a script that does that. There are some other scripts that do it as well. I used to see that too, you can safely ignore it. |
| 17:47.50 | Dr-Linux | mercestes: what's manager channel name? |
| 17:48.13 | ManxPower | mercestes: FOP uses manager interface, that is only for the remote unix socket connections |
| 17:48.35 | Dr-Linux | ManxPower: what's the manager.conf channel name? |
| 17:48.46 | ManxPower | Dr-Linux: channel name? |
| 17:48.51 | Dr-Linux | sorry, module name |
| 17:48.55 | Dr-Linux | i gonna unload it |
| 17:49.00 | ManxPower | I don't think it has one. |
| 17:49.06 | ManxPower | you just disable it in manager.conf |
| 17:49.10 | ManxPower | but that will not fix your issue. |
| 17:49.11 | Dr-Linux | aww that's why i coudn't find :P |
| 17:49.20 | Dr-Linux | hhm.. |
| 17:49.22 | Dr-Linux | okey |
| 17:49.33 | L|NUX | can some one help me |
| 17:49.37 | L|NUX | with my query please |
| 17:49.38 | L|NUX | :) |
| 17:49.39 | ManxPower | Something or someone is runnning asterisk |
| 17:49.53 | mercestes | L|NUX: It should. |
| 17:49.58 | ManxPower | L|NUX: that won't work if you use either numbers OR letters! |
| 17:50.00 | L|NUX | but its not working |
| 17:50.21 | L|NUX | ManxPower : humm any way to make it work with both ? |
| 17:50.24 | L|NUX | number+letters |
| 17:50.25 | ManxPower | extension "s" is run when there is NO dialed digits received. |
| 17:50.32 | ManxPower | About the only time that is the case is when a call comes in on an analog FXO port. |
| 17:50.36 | Dr-Linux | ManxPower: fixed |
| 17:50.37 | Dr-Linux | :) |
| 17:51.09 | ManxPower | L|NUX: exten => _.,1,whatever would work, but then that extension will be called twice (because it matches extension h as well) |
| 17:52.05 | *** join/#asterisk anthonyl (i=Anthony@nat/digium/x-9271eb134ce499d0) |
| 17:52.12 | L|NUX | so i will add h,1,hangup |
| 17:52.13 | L|NUX | :) |
| 17:52.21 | ManxPower | it will also match extensions t, a, T, o, etc |
| 17:53.04 | Dr-Linux | ManxPower: i'm going to setup new asterisk server on SUN server using RHEL 4, i need your suggestioni regarding should i go for 1.4 or 1.2.13 ? |
| 17:53.15 | Dr-Linux | ManxPower: it will be a production server |
| 17:53.25 | ManxPower | Dr-Linux: 1.4 has not been out long enough. |
| 17:53.58 | Dr-Linux | ManxPower: so you would suggest 1.2.13 for now. Is that correct? |
| 17:54.10 | ManxPower | No. I would suggest 1.2.14 |
| 17:54.20 | Dr-Linux | aww ok |
| 17:55.13 | Sweeper | any way to disable notices, but keep verbosity on the cli? |
| 17:55.40 | ManxPower | Sweeper: logger.conf |
| 17:55.43 | Sweeper | NOTICE[15544]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 <-- I get lots and lots of these, but the pri works and shit |
| 17:55.44 | *** join/#asterisk UlbabraB (n=salama@81.72.43.241) |
| 17:56.18 | ManxPower | Sweeper: that will become a significant issue. Don't expect faxes to work and expect audio quality to suffer, and maybe even disconnected calls. |
| 17:56.33 | ManxPower | Sweeper: That notice means that DATA WAS LOST. |
| 17:56.35 | Dr-Linux | ManxPower: DUNDi is fine to synchronise the 2 servers dialplan/users etc? |
| 17:56.38 | Sweeper | damnit D: |
| 17:56.47 | L|NUX | ManxPower : so its means there is no solution |
| 17:56.48 | L|NUX | :( |
| 17:57.12 | Sweeper | only thing I can find that causes this issue is hardware-releated :/ |
| 17:57.31 | x86 | `Sean: hey man |
| 17:57.33 | backblue | does asterisk have limit to the number, of presence notification a user can have? |
| 17:57.43 | Sweeper | ManxPower: are there configuration things to fix this issue, or does it just mean "lol get new mobo"? |
| 17:57.52 | ManxPower | Sweeper: The problem is caused by 1 of 2 things. 1) errors on the T-1 from the telco or 2) some device or driver is locking interrupts for long enough to lose data |
| 17:58.11 | *** join/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 17:58.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:58.58 | Sweeper | ManxPower: how do I troubleshoot interrupts? |
| 17:59.15 | Juggie | Sweeper: with great difficulty. |
| 17:59.44 | Joel1978 | What would cause one way audio loss for a person calling into my * box but for me calling out i get full audio |
| 17:59.48 | Sweeper | only thing I can see in /proc/interrupts is a "timer" device at irq0 |
| 17:59.50 | ManxPower | Sweeper: cat /proc/interrupts If the card driver is sharing an interrupt with something else, that is a good thing to fix first. |
| 18:00.00 | Sweeper | card has its own interrup |
| 18:00.22 | Sweeper | *only thing I can see that has lots of interrupts |
| 18:00.30 | ManxPower | Sweeper: disable onboard lan, disable onboard raid, disable onboard sata, don't run in graphics mode. |
| 18:00.45 | ManxPower | Sweeper: Zaptel cards generatre 1,000 interrupts per second |
| 18:00.58 | Sweeper | I mean besides that, obviously |
| 18:01.07 | L|NUX | ManxPower : that work but when i pick the phone it will hangup |
| 18:01.09 | Sweeper | mmmm |
| 18:01.13 | Sweeper | yay disable :/ |
| 18:01.19 | BSDTech | everyone is fired asterisk is a failure |
| 18:01.21 | ManxPower | Frequently the cause of the interrupt latency/jitter is devices on the system that take over the PCI bus to "improve performance" |
| 18:01.35 | BSDTech | time to put it out to pasture |
| 18:02.03 | ManxPower | Sweeper: The same issues with high end PC audio devices. |
| 18:03.39 | L|NUX | ManxPower : can you help me bro |
| 18:04.16 | naitram | anyone have any ideas if the h extension should work in macros as shown here http://pastebin.ca/319640 . I am using an older asterisk version 1.0.4 |
| 18:04.35 | Sweeper | ManxPower: mkay. |
| 18:04.48 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 18:05.19 | Sweeper | mmm |
| 18:05.25 | Sweeper | well, it's not the usb drivers |
| 18:05.32 | BSDTech | how do you turn up the aduio on a zap channel? |
| 18:06.05 | ManxPower | Sweeper: most commonly it is the onboard lan or SATA controller |
| 18:06.13 | ManxPower | BSDTech: rxgain / txgain |
| 18:06.22 | ManxPower | no you can't do it on the fly from a phone |
| 18:06.29 | BSDTech | okk so rxgain=2 |
| 18:07.05 | BSDTech | in the zapata.conf correct |
| 18:07.11 | ManxPower | naitram: try it an see |
| 18:07.29 | ManxPower | BSDTech: I believe so. the sample config files will show you |
| 18:09.56 | naitram | ManxPower: have tried it, doesn't seam to work. Thought someone might know if its a bug in older version or by design. Dont want to upgrade / become unstable trying to get something that does not exist in newer version |
| 18:10.34 | ManxPower | naitram: I doubt it has changed in later verisons |
| 18:10.46 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 18:13.17 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:13.29 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 18:14.08 | EmleyMoor | Can anyone recommend for me a good, economical phone (preferably SIP or IAX) that takes Power over Ethernet? |
| 18:14.21 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 18:14.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:15.17 | BSDTech | ok that worked |
| 18:15.27 | BSDTech | had to turn it upto 4.0 |
| 18:15.32 | BSDTech | but its better |
| 18:16.00 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 18:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:18.23 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 18:18.25 | TheCops | Hi |
| 18:19.49 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:20.05 | jmls | anyone gotten slimserver .flac to work with asterisk 1.4 ? |
| 18:20.18 | jmls | or is it going to have to be another adventure in asterisk land ? |
| 18:20.32 | simplexio | flac ? |
| 18:20.44 | simplexio | do you mean .flac voice files ? |
| 18:20.52 | Charles[NS] | yop |
| 18:21.05 | jmls | I mean flac lossless compression |
| 18:21.06 | jmls | http://flac.sourceforge.net/ |
| 18:21.48 | jmls | I've ripped all my cd's to .flac as I can then turn them back into .wav with no loss |
| 18:22.06 | *** join/#asterisk bkw__ (n=brian@82.152.100.241) |
| 18:22.08 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 18:22.08 | simplexio | i ment just that, why you would want to use flac with asterisk ? |
| 18:22.21 | jmls | my slimdevices play 'em just dandy and I was wondering if I could stream a playlist for music on hold etc |
| 18:22.54 | jmls | iow make asterisk play moh from the slimserver |
| 18:23.06 | Charles[NS] | someone have a good misdn.conf with extension.conf ? |
| 18:23.15 | jmls | I know you can do it if you use .mp3 ... |
| 18:23.21 | simplexio | probably bad idea if you have many users in asterisk box |
| 18:23.24 | Sweeper | awsome. either I broke the pri, or bell is fucking with it again |
| 18:23.25 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 18:23.25 | Charles[NS] | phone works in local but not external |
| 18:23.25 | CunningPike | EmleyMoor: A lot of people like their Polycoms (us included) - they have a range of models at different price points |
| 18:23.37 | TheCops | I have echo between 2 SIP phone, how can I solve it ? This is pretty bizzard, this is 2 Polycom IP601 |
| 18:23.41 | CunningPike | EmleyMoor: IP430 might work for you |
| 18:24.33 | Charles[NS] | Jan 17 19:18:50 WARNING[2573] chan_misdn.c: Could not create channel on port:-1 with extensions:0141163490 |
| 18:24.36 | Charles[NS] | Jan 17 19:18:50 NOTICE[2573] app_dial.c: Unable to create channel of type 'mISDN' (cause 0 - Unknown) |
| 18:24.38 | ManxPower | TheCops: It is impossible to have echo between 2 sip phones unless the phones are bad or poorly designed or the room is echoy |
| 18:24.39 | EmleyMoor | Why does ekiga join voicemail slightly later than moziax or a Zap phone? |
| 18:24.51 | CunningPike | TheCops: It's acoustic coupling in the phone - you can try lowering the volume on the phone |
| 18:25.12 | simplexio | jmls: but fast look into musiconhold.conf and it seems to give option to define your player by self |
| 18:25.32 | jmls | it's the format that I am worried about - I can do it with .mp3 |
| 18:25.50 | CunningPike | TheCops: What we also found is that the IP601 is very picky about its audio config -the settings for that phone have changed quite a bit between firmware updates, so make sure that the sip.cfg is the one that came with your most recent firmware |
| 18:25.57 | simplexio | jmls: do you any cli-linux flac players ? |
| 18:26.11 | TheCops | ho |
| 18:26.16 | TheCops | CunningPike, good point |
| 18:26.18 | TheCops | very good point |
| 18:26.19 | TheCops | thank you |
| 18:26.24 | EmleyMoor | Hmmm... IP430 is 140 |
| 18:26.40 | CunningPike | EmleyMoor: You won't get a decent PoE phone for much less....... |
| 18:27.01 | CunningPike | EmleyMoor: Another alternative is Linksys |
| 18:27.06 | EmleyMoor | OK - just out of interest, how do you rate the Grandstream GXP-2000? |
| 18:27.07 | robl^ | th3e polycom phones have GREAT sound quality.. compared to most others I have used |
| 18:27.07 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 18:27.09 | perd | h323 people, which tree do you find works best? h323, oh323 or ooh323? |
| 18:27.25 | CunningPike | EmleyMoor: It's crap |
| 18:27.29 | CunningPike | EmleyMoor: IMHO |
| 18:27.31 | CunningPike | :) |
| 18:27.42 | simplexio | jmls: flac123 Description: flac-tools provides flac123 a console app for playing FLAC audio files. |
| 18:29.08 | jmls | simplexio: hmm, thanks. will look into that |
| 18:29.08 | simplexio | jmls: just replace mpg123 with flac123, maybe works maybe not |
| 18:29.09 | ManxPower | EmleyMoor: Gransdtream has been producing IP phones for 5 years or so. Their hardware may have improved, but their firmware STILL sucks. You'll have to go thru many versions to find one that works for you. |
| 18:29.09 | EmleyMoor | Fair enough |
| 18:29.09 | ManxPower | Their hardware improved, but it still sucks compared to most other phones |
| 18:29.09 | robl^ | EmleyMoor: Grandstream is likely the cheapest of all the VoIP phones. Cheapest in price and in quality |
| 18:29.22 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:29.31 | EmleyMoor | How are Atcom? |
| 18:29.54 | [TK]D-Fender | EmleyMoor : GrandSuck is to be avoided with extreme prejudice. |
| 18:30.22 | ManxPower | EmleyMoor: last I head the ATACOM phones are just rebranded cheap phones from china. |
| 18:30.29 | _DAW | Do DNS SRV lookups work correctly now in 1.4? |
| 18:30.32 | [TK]D-Fender | EmleyMoor : Atcom is every bit as bad. YGWYPF |
| 18:30.48 | EmleyMoor | ipDialog? |
| 18:31.01 | [TK]D-Fender | EmleyMoor : Overpriced, and low on functionality. |
| 18:31.05 | ManxPower | EmleyMoor: If you use an unknown SIP phone nobody will be able to help you with it either. |
| 18:31.24 | CunningPike | EmleyMoor: Polycom, Linksys, Cisco. Do yourself a favor and stick with those |
| 18:31.46 | EmleyMoor | Linksys has the cheapest PoE phone out of those three |
| 18:31.50 | robl^ | EmleyMoor: Polycom, Snom, Aastra,Cisco are about the best support here,... |
| 18:31.57 | [TK]D-Fender | EmleyMoor : What you should be aiming for in order of quality/functionality (in order) : Polycom (any), Aastra 480i, Cisco (7940+), Linksys. |
| 18:31.59 | EmleyMoor | (the 942) |
| 18:32.11 | EmleyMoor | How about Swissvoice? |
| 18:32.11 | fetcher | Is there a better quality phone than ATCOM that supports IAX? |
| 18:32.16 | [TK]D-Fender | EmleyMoor : Where are you located? |
| 18:32.21 | EmleyMoor | UK |
| 18:32.34 | ManxPower | fetcher: ALL the IAX hardphones are rebranded cheap china phones with bad firmware |
| 18:32.34 | [TK]D-Fender | EmleyMoor : Then aim for at least Linksys. |
| 18:32.51 | [TK]D-Fender | IAX was not meant for en-points. Look at what is stands for... |
| 18:32.52 | *** join/#asterisk reber (i=reber@gateway/tor/x-439e9f15e66a2a22) |
| 18:33.00 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 18:33.19 | fetcher | ManxPower: drats. I was hoping for something reasonable for home telecommuter extensions, to avoid NAT hassles... |
| 18:33.28 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:33.43 | fetcher | maybe just setting up a STUN server will work well enough |
| 18:34.20 | _Sam-- | i will still defend my grandstream phones |
| 18:34.28 | EmleyMoor | Aastra's 9133i is slightly cheaper |
| 18:34.29 | [TK]D-Fender | fetcher : most of the time STUN is unnecessary. |
| 18:34.30 | _Sam-- | they work perfectly for me, in my enviroment |
| 18:34.35 | _Sam-- | gxp2000s |
| 18:34.45 | robl^ | fetcher: I have both polycom and Snom phones behind NATs and they work well.. no STUN |
| 18:34.51 | _Sam-- | i dont think you can find more bang for the buck...but thats just my opinion. |
| 18:34.51 | [TK]D-Fender | _Sam-- : What other phones have you used? |
| 18:35.18 | EmleyMoor | What are the pros/cons of the Polycom 301? |
| 18:35.18 | terrapen | hey d-fender, have you ever written an AGI that balances callers across two asterisk "queue" servers? |
| 18:35.32 | fetcher | robl^, Sam: never any issues with one-way audio? |
| 18:35.32 | terrapen | you mentioned this yesterday |
| 18:35.36 | terrapen | it's a great idea, i think |
| 18:35.52 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 18:35.59 | terrapen | but it seems like it might not work so well when call levels are low |
| 18:36.05 | terrapen | unless the AGI can take that into account |
| 18:36.06 | [TK]D-Fender | EmleyMoor : Pro : all the quality of every other model up. Con (where applicable) PoE requires extra expense on the special cable needed, no Speakerphone. Thts it. |
| 18:36.10 | _Sam-- | [TK]D-Fender : none. i have never had the need. we have about 20 gxps here. |
| 18:36.27 | _Sam-- | fetcher : nope. i found firmware that worked for us, and i have never changed. |
| 18:36.29 | robl^ | EmleyMoor: 301s don't do PoE without an optional calbe. And don't have a speaker phone. The 430 is only $15 more nad has PoE, full duplex speaker and better display |
| 18:36.41 | terrapen | the 430 is crap |
| 18:36.45 | terrapen | we had major problems with them |
| 18:36.50 | [TK]D-Fender | terrapen : I was the one who suggested this concept to you the other day. No I haven't done it, but it sounds pretty easy. Jsut open an AMI interface, and cheack the queu saize and flip it out. |
| 18:36.50 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 18:36.51 | Charles[NS] | anyone have the BN8S0 ? |
| 18:36.52 | terrapen | sent them back and got 501s |
| 18:37.23 | [TK]D-Fender | terrapen : Only 2 people here ever seem to have had problems with Polycom. You and oh whats his name..... |
| 18:37.31 | terrapen | the display on the 430s is garbage |
| 18:37.43 | _Sam-- | [AV]bani back in the old days had polycom hassles |
| 18:37.44 | terrapen | and we had problems with them rebooting constantly |
| 18:37.46 | [TK]D-Fender | terrapen : And if you don't need PoE, the 501 si wortha few bcks more anyways. |
| 18:37.48 | EmleyMoor | How is the Cisco 7912G? Can it take the standard PoE? |
| 18:37.53 | terrapen | i love polycom |
| 18:37.55 | [TK]D-Fender | terrapen : I don't find.... |
| 18:38.07 | [TK]D-Fender | EmleyMoor : No, Cisco PoE only. bad choice... |
| 18:38.09 | terrapen | i just don't like the 430's :) |
| 18:38.19 | [TK]D-Fender | terrapen : i've got a few and one here at home. |
| 18:38.31 | terrapen | then again, when I got our 430s, they were very new and the firmware was new |
| 18:38.42 | terrapen | they may have fixed the problems |
| 18:38.58 | fetcher | how do SNOM phones stack up? |
| 18:39.07 | [TK]D-Fender | terrapen : yeah I had screen alignment on boot issues, and a little flicker, but have since been well adjusted. |
| 18:39.08 | terrapen | we've never had a single problem with our IP501s...love those phones |
| 18:39.31 | [TK]D-Fender | fetcher : Snom is still buggy, shitty LCD usability, and only passably audio quality. Other can elaborate. |
| 18:39.49 | [TK]D-Fender | terrapen : Agreed. Very solid model. It is my primary at home. |
| 18:39.58 | fetcher | hmm, and not that much cheaper than Polycom either |
| 18:39.59 | terrapen | not terribly impressed with the SNOM and the ring tones are obnoxious |
| 18:40.32 | robl^ | Snom firmware is hit or miss. decent hardware quality. Easy to set up. a little flakey but better than most of the other cheap brands |
| 18:40.32 | TheCops | [TK]D-Fender, I have snom here, about 8, and all working proprely, good audio, good feature, there's more feature than Polycom. But yeah, they like to be more cheap than Polycom for Audio |
| 18:40.34 | EmleyMoor | Presumably, given a choice of protocols, SIP is the one to use these days? |
| 18:40.36 | _Sam-- | i havent shopped for phones for about 12 months -- but do any phones have bluetooth built in yet? |
| 18:40.38 | terrapen | the Linksys phones would be awesome (the 942s) if only they would do FTP-based configuration |
| 18:40.42 | terrapen | err provisioning |
| 18:41.08 | _Sam-- | or, why not a wifi desktop phone |
| 18:41.40 | [TK]D-Fender | terrapen : I found the Linksys to bee too light and shifted around a lot (crappy rubber feet too), and the stand shoddy. Also their scrren readbility/functionality left much to be desired. |
| 18:41.58 | [TK]D-Fender | terrapen : Nothing Cisco-owned does Presence either which really blows. |
| 18:41.59 | robl^ | [TK]D-Fender: Have you seen an issue where a sidecar will randomly reboot while the IP 601 phone is in use? |
| 18:42.28 | [TK]D-Fender | robl^ : Through the 3 installation I worked at that use then, no, nor have I ever heard it elsewhere. |
| 18:42.54 | [TK]D-Fender | _Sam-- : WiFi phones suck. All of them. HARD. Some only a little less than others. |
| 18:43.03 | terrapen | d-fender, the 942 has the backlit LCD which is sweet |
| 18:43.11 | terrapen | it's actually very very readale |
| 18:43.11 | _Sam-- | but why would a phone, on a desk, need , say 100 megabits of bandwidth |
| 18:43.16 | _Sam-- | why COULDNT it work well wifi |
| 18:43.31 | xheliox | I like my WiFi phone. |
| 18:43.33 | [TK]D-Fender | terrapen : that is a bonus, but thats the point where I advise more for the Aastra 480i. It is considerable more functional than the Linksys. |
| 18:43.39 | Nugget | http://www.defectiveyeti.com/archives/001841.html <-- router |
| 18:43.41 | _Sam-- | and why ISNT there bluetooth in desktop phones-- am i missing something, or it seems an obvious need |
| 18:43.46 | terrapen | d-fender: but last time I checked, they could not be provisioned by FTP/TFTP which totally blows |
| 18:44.04 | [TK]D-Fender | terrapen : Linksys LCD uses bad abreviations, with crappy pixel density. |
| 18:44.25 | robl^ | aastra can be configed by tftp. |
| 18:44.36 | terrapen | nothing beats the Polycom IP501 and IP601 :) |
| 18:45.00 | EmleyMoor | Assuming a standard * installation in Debian, which codecs are supported? |
| 18:45.07 | terrapen | has anybody written a good asset manager for IP phone systems? |
| 18:45.17 | terrapen | automated provisioning and tracking of phones, etc |
| 18:45.58 | [TK]D-Fender | terrapen : Depends on a certain point of view. While yes, the quality and functionality is VERY solid, TECHNICALLY the 480i is for single registration use better than an IP 501. |
| 18:46.06 | terrapen | robl, i was referring to the Linksys phones |
| 18:46.19 | robl^ | terrapen: I am working on such a system for polycom phones. a GUI app that does all the confg files and manages everything from a central database |
| 18:46.33 | Charles[NS] | quelqu'un de francophone ici ? |
| 18:46.46 | [TK]D-Fender | terrapen : 480i adds state-based programmable keys that CAN do in-call DTMF (great for calling features), and you can have a few pages of them to go through. Better for presenc as well (again compared to a 501). |
| 18:47.02 | _Sam-- | !seen [av]bani |
| 18:47.21 | [TK]D-Fender | terrapen : however the 601 wins out on massive identities and shuffling LOTS of call, and w/ sidecards beats the 480i for presence. |
| 18:47.26 | terrapen | d-fender, that definitely is attractive |
| 18:47.43 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:48.05 | *** join/#asterisk amdtech (i=adaniel@nat/digium/x-22c8b9f01833cde2) |
| 18:48.06 | [TK]D-Fender | terrapen : but for typical use the 480i is very appealing. The audio quality IS a little weaker, and I can tell the difference (I am rather discerning) |
| 18:48.30 | terrapen | what is presence, anyway? |
| 18:48.48 | [TK]D-Fender | terrapen : Choosing a GOOD mid-range phone takes some very careful consideration with regards to how they are going to be used / deployed. |
| 18:48.53 | xheliox | s/discerning/anal |
| 18:49.04 | [TK]D-Fender | terrapen : You know, the little indicators that someone you are watching is on the phone... |
| 18:49.17 | terrapen | ohhhhh ok yeah |
| 18:49.23 | [TK]D-Fender | xheliox : No, I'm a musician, sound is my canvas :) |
| 18:49.31 | EmleyMoor | I turned off ringing tone in ekiga and now I can get to voicemail just as it declares how many messages I have... with moziax, I get the very beginning though |
| 18:49.39 | terrapen | yeah, we don't do much of that....we're just a big inbound call center and a corporate office |
| 18:49.51 | [TK]D-Fender | EmleyMoor : Add a Wait(2) in front. |
| 18:49.53 | terrapen | i guess our secretary could use it, for monitoring the owners' phones |
| 18:50.03 | Charles[NS] | quelqu'un de francophone ici ? |
| 18:50.16 | [TK]D-Fender | terrapen : Exactly. |
| 18:50.19 | EmleyMoor | Is this common to all SIP phones or is it just ekiga? |
| 18:50.35 | Charles[NS] | i need help with my BN8S0 |
| 18:50.45 | [TK]D-Fender | Charles[NS] : C'est quoi un francophone? Il-ressemble a une telephone? |
| 18:50.57 | [TK]D-Fender | :) |
| 18:51.03 | terrapen | i wish there was some way for two servers running call queues to send callers to send callers to each other when the queues are unbalanced |
| 18:51.20 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 18:51.41 | [TK]D-Fender | terrapen : like I said, make an AGI that will dump queue stats from both sides and transfer the calla ccordingly. a relatively simple thing to do. |
| 18:52.12 | Charles[NS] | lol :) |
| 18:53.00 | terrapen | so, imagine two servers with 100 agents each... and 120 callers on each server... if, all of the sudden, one server drops to 90 callers, I want to send at least 10 of the callers from the more-loaded server to the less-loaded server |
| 18:53.09 | terrapen | so the queue servers could shuffle callers |
| 18:53.29 | [TK]D-Fender | terrapen : Ah, STEALING people already in queue.... that sounds God-aweful difficult :) |
| 18:53.33 | terrapen | it's easy to get them load balanced with the AGI, but that doesn't really account for changing conditions |
| 18:53.38 | terrapen | yeah, it's gnarly, no doubt |
| 18:55.25 | [TK]D-Fender | terrapen : I think you could do it using AMI as well, but their priority arriving in the new queue would be detrimental. |
| 18:55.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-13199.bb.online.no) |
| 18:55.48 | [TK]D-Fender | FOP offered the ability to hijack calls, so I'm pretty sure its possbile, though somewhat daunting. |
| 18:56.44 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 18:57.04 | terrapen | is there a way to have an agent logged into queues on two different servers but prevent that agent from recieving calls from server B if they are handling a call from server A? |
| 18:57.30 | robin_sz | backblue, so this mISDN stuff |
| 18:57.57 | terrapen | being able to do that would essentially give us a shared queue |
| 18:58.12 | [TK]D-Fender | terrapen : 2 places at once? Not with any sanity. I can't picture it personally.... |
| 18:58.25 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:58.31 | robin_sz | happen to know if anyone is working on fixing the decoding of dtmf ?? |
| 18:58.46 | *** join/#asterisk Lurchtoke (i=Lurchtok@adsl-75-37-74-230.dsl.frs2ca.sbcglobal.net) |
| 18:58.54 | Lurchtoke | hello everyone |
| 18:59.12 | Lurchtoke | I had a question I needed help with |
| 19:00.15 | backblue | robin_sz: ? what? |
| 19:00.38 | Lurchtoke | bear with me since I am a total noob.....but I am moving my business and I am moving my sip to a new location with a new block of IP's |
| 19:00.46 | Lurchtoke | err sips |
| 19:01.34 | Lurchtoke | my phone menu is working fine....but all I need to do is move my sips to the new ips....is there a faq section I can browse to help me do that?? |
| 19:02.08 | *** join/#asterisk ^sandro^ (n=lskdjfls@internal--router.canaca.com) |
| 19:02.10 | ^sandro^ | hello |
| 19:02.14 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 19:02.18 | ^sandro^ | can someone help me with zaptel |
| 19:02.28 | [TK]D-Fender | Lurchtoke : "sips"? |
| 19:02.29 | ^sandro^ | i have 6 PRis and today i added another |
| 19:02.46 | Lurchtoke | sip adapters |
| 19:02.52 | ^sandro^ | the new PRI that i got is an addition to my first PRI ( group 1 ) ... it has no d channel as it runs off the first one |
| 19:03.07 | ^sandro^ | so my /etc/zaptel.conf says : |
| 19:03.17 | ^sandro^ | span=7,0,0,esf,b8zs # Toronto Number 2 |
| 19:03.17 | ^sandro^ | bchan=145-168 |
| 19:03.21 | [TK]D-Fender | Lurchtoke : Typically endpoint are DHCP and only need to know the IP of the server they are registering too.... whats to configure beyond that? |
| 19:03.50 | ^sandro^ | and my zapata.conf says ... |
| 19:03.51 | ^sandro^ | group=1 ; <-------- Toronto |
| 19:03.51 | ^sandro^ | context=from-pstn |
| 19:03.51 | ^sandro^ | channel=>1-23 |
| 19:03.51 | ^sandro^ | ;channel=>145-168 |
| 19:03.55 | ^sandro^ | can anyone help? |
| 19:03.58 | ^sandro^ | it wont start when i have it on |
| 19:03.59 | ^sandro^ | any ideas? |
| 19:04.23 | [TK]D-Fender | ^sandro^ : Don't span in channel, user pastebin.ca |
| 19:04.44 | [TK]D-Fender | ^sandro^ : and please include your ENTIRE zapata.conf |
| 19:04.45 | Strom_C | s/span/spam/ |
| 19:04.50 | Strom_C | s/user/use/ |
| 19:04.52 | *** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
| 19:05.00 | *** join/#asterisk masked (i=masked@shell.iinet.net.au) [NETSPLIT VICTIM] |
| 19:05.02 | [TK]D-Fender | Sush Storm! ;) |
| 19:05.11 | Strom_C | OK, fendar |
| 19:05.19 | a1fa | sushi storm |
| 19:05.24 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) [NETSPLIT VICTIM] |
| 19:05.27 | [TK]D-Fender | :O |
| 19:05.39 | rudholm | Storm Carlson - he's a mutant! |
| 19:05.40 | robl^ | s/shuffel/shuffle/ |
| 19:05.42 | [TK]D-Fender | a1fa : I knew there was something fishy going on in here! |
| 19:06.01 | [TK]D-Fender | robl^ : In case you missed it, that was a PUN :) |
| 19:06.22 | Strom_C | rudholm: that would explain my crazy phone-jockey abilities |
| 19:06.31 | robl^ | ;-) |
| 19:06.38 | a1fa | d-fender > * |
| 19:06.50 | a1fa | dfednah is my god! |
| 19:06.52 | [TK]D-Fender | My spelling is impeccable, my typing leaves something to be desired... especially as I have a cold/flu/other-random-bug and am home... |
| 19:06.58 | ^sandro^ | sorry ok |
| 19:07.00 | ^sandro^ | i posted under http://pastebin.ca/319738 |
| 19:07.05 | ^sandro^ | can someone have a look and let me know |
| 19:07.35 | ^sandro^ | i dont knwo what im missing here |
| 19:07.43 | ^sandro^ | i thought it to be pretty straight forward |
| 19:08.05 | [TK]D-Fender | ^sandro^ : so what.. your 7th PRI has no D-Chan? |
| 19:08.26 | ^sandro^ | no |
| 19:08.39 | ^sandro^ | its just to B channels .. it uses the D channel from the first toronto pri |
| 19:08.44 | [TK]D-Fender | ^sandro^Not a PRI w/o one you know... |
| 19:09.01 | fetcher | probably set up for NFAS |
| 19:09.07 | ^sandro^ | ya thats right |
| 19:09.07 | [TK]D-Fender | ^sandro^ : then you are missing a "spanmap" |
| 19:09.10 | ^sandro^ | how do i set it there |
| 19:09.19 | ^sandro^ | ive never done that one before :( |
| 19:09.23 | ^sandro^ | its new to me |
| 19:09.30 | [TK]D-Fender | ^sandro^ : Go look that up, as well as NFAS on the Wiki. |
| 19:09.32 | [TK]D-Fender | ~wikis |
| 19:09.42 | jbot | [wikis] http://www.voip-info.org |
| 19:11.36 | ^sandro^ | ok going to read up on it |
| 19:11.46 | ^sandro^ | can anyone paste what its supposed to look like? |
| 19:12.04 | robin_sz | backblue, you seemed have some knowledge of the mISDN stuff when I spoke to you last, wondered if you'd seen any updates to the DTMF detection ... it seems abit poor at the mpment |
| 19:12.14 | [TK]D-Fender | ^sandro^ : go READ |
| 19:12.42 | terrapen | so, assuming 60 man hours/week, do you think 10 weeks is a little too ambitious of an estimate for building an asterisk-based call center for 300 agents? |
| 19:13.10 | brettnem | hey, has anyone ever used NVlineDetect? |
| 19:13.16 | terrapen | im trying to convince the executives to let my department build our PBX with asterisk instead of a stupid avaya |
| 19:13.16 | [TK]D-Fender | terrapen : 2 hours per agent? eek. |
| 19:13.29 | Sweeper | arg. how would I go about nuking a NIC so it doesn't generate interrupts? |
| 19:13.52 | robin_sz | terrapen, depends what you mean by "build" |
| 19:13.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:14.08 | terrapen | design, purchase equipment, build, test, deploy |
| 19:14.10 | [TK]D-Fender | terrapen : Queues are a weak spot with *, reporting is shoddy, and lacks summarization capabilitys, call calssification, and the other stuff CS managers might demand |
| 19:14.26 | robin_sz | terrapen, does deploy mean "insall cables under the floors" ? |
| 19:14.30 | terrapen | we would be purchasing QueueMetrics |
| 19:14.35 | terrapen | robin: no. the building is wired |
| 19:14.37 | [TK]D-Fender | terrapen : even then.... |
| 19:14.56 | terrapen | d-fender, are Asterisk queues not robust enough? |
| 19:15.00 | robin_sz | terrapen, so all you have to do is configure the software and dump the phones on the desks? |
| 19:15.08 | terrapen | Am I setting myself up for trouble? :) |
| 19:15.09 | [TK]D-Fender | terrapen : Still no categorization, and cascading / queue reprioritization. |
| 19:15.21 | [TK]D-Fender | terrapen : No, they really aren't... by themselves... |
| 19:15.33 | [TK]D-Fender | terrapen : Check these guys out : www.aheeva.com |
| 19:15.41 | terrapen | robin, and build the media gateways...or buy them |
| 19:16.13 | robin_sz | ah yes |
| 19:16.17 | robin_sz | the media gateways |
| 19:16.19 | backblue | robin_sz: i dont have any kind of problem with dtmf, only can be tricky with the single Bri cards. |
| 19:16.47 | robin_sz | backblue, thats my problem ... it seems to only work about 75% of the time on my HFC cards |
| 19:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:19.28 | terrapen | why would i want to go with aheeva? |
| 19:19.36 | terrapen | do they have something that standard * does not? |
| 19:20.08 | [TK]D-Fender | terrapen : yes, they have extrenssively hacked the raw stuff,a nd have their own SIP client to add specific features that are missing. |
| 19:20.18 | terrapen | interesting |
| 19:20.59 | backblue | robin_sz: dtmftreshold=50 |
| 19:21.03 | backblue | decrease it. |
| 19:21.09 | backblue | and keep trying. |
| 19:21.18 | backblue | cya * |
| 19:22.51 | terrapen | aheeva looks interesting |
| 19:23.05 | terrapen | i'll have to call them after lunch |
| 19:23.49 | zoa | terrapen |
| 19:23.55 | zoa | we have a clustered solution for the queues |
| 19:23.59 | zoa | on top of asterisk |
| 19:24.10 | zoa | so queues that span multiple servers |
| 19:24.35 | terrapen | really? |
| 19:24.40 | terrapen | can you e-mail me some info? |
| 19:24.47 | zoa | sec im on the phone |
| 19:29.44 | robin_sz | sigh ... decrease dtmfthreshold indeed. |
| 19:29.52 | ^sandro^ | guys.. i set the trunkgroup and spanmap... but do i put that at the top... do i put in channel = whatever - whatever still under the group 1? |
| 19:32.52 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 19:35.06 | ^sandro^ | anyone? |
| 19:36.35 | [TK]D-Fender | ^sandro^ : TRY it. |
| 19:36.56 | [TK]D-Fender | ^sandro^ : few people use this functionality, and fewer still come here regularly. |
| 19:37.24 | Sweeper | use the "TRY" functionality? |
| 19:38.17 | ^sandro^ | i did .. |
| 19:38.22 | ^sandro^ | i put it in wont start asterisk |
| 19:38.35 | ^sandro^ | still gives me in the logs chan_zap.c: Unable to register channel '145-168 |
| 19:39.09 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:40.05 | blitzrage | hey all. I'm trying to place multiple calls to multiple phones which I need all to answer and be dumped into a MeetMe. I've tried the Local channel and also sending Originates from the manager, but it seems I'm only ever able to get one phone at a time to answer. What am I missing? |
| 19:40.32 | mercestes | blitzrage: Polycoms? |
| 19:40.38 | blitzrage | The phones will actually all Auto-Answer using the Polycom config file, but that is not really the issue |
| 19:40.45 | blitzrage | yes, all Polycom's |
| 19:41.06 | mercestes | Yea, Use the Alert_into ring answer and use asterisk cmd page() |
| 19:41.06 | [TK]D-Fender | blitzrage : "show application page" |
| 19:41.43 | blitzrage | holy fuck was I ever making that more complicated than necessary |
| 19:41.48 | blitzrage | [TK]D-Fender: thanks |
| 19:41.55 | mercestes | .. |
| 19:41.55 | Sweeper | oh wtf |
| 19:42.02 | Sweeper | so that interrupt problem I was having? |
| 19:42.13 | blitzrage | mercestes: I already know how to use the Ring Answer :) |
| 19:42.17 | blitzrage | that wasn't the issue |
| 19:42.19 | Sweeper | turns out ztcfg wasn't really reloading the config for my card |
| 19:42.21 | mercestes | .... |
| 19:42.28 | mercestes | <mercestes> Yea, Use the Alert_into ring answer and use asterisk cmd page() |
| 19:42.29 | Sweeper | and it was actually the t1 timing |
| 19:42.32 | mercestes | I so said page first. |
| 19:42.44 | tzanger | mercestes: heh page is pretty awesome |
| 19:43.03 | mercestes | but it's ok, I know how you *really* feel now. |
| 19:43.10 | blitzrage | mercestes: oops... I didn't notice the last part :) |
| 19:43.16 | blitzrage | thanks mercestes :) |
| 19:43.23 | Sweeper | that's two bugs for me |
| 19:43.24 | mercestes | uh huh. |
| 19:43.32 | mercestes | ;) |
| 19:43.38 | blitzrage | I can't believe how effing complicated I was making it :) |
| 19:43.51 | blitzrage | you should see my dialplan -- its nuts :D |
| 19:44.19 | blitzrage | although would be nice to be able to do the same thing as page, but have the choice as to where the calls were dumped in |
| 19:44.23 | blitzrage | I'll figure that part out later |
| 19:44.43 | [TK]D-Fender | blitzrage : What do you mean "dumped" in? |
| 19:45.00 | [TK]D-Fender | blitzrage : in that case use .call files that use the auto-answer as well. |
| 19:45.04 | ^sandro^ | damn im confused :( |
| 19:45.15 | ^sandro^ | can ayone message me and help me out with the NFAS stuff |
| 19:45.28 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 19:45.31 | ^sandro^ | im a bit confused on how to get the B channels working with my B channel on another PRI |
| 19:45.43 | blitzrage | [TK]D-Fender: call files suck though |
| 19:45.45 | ^sandro^ | ive tried a few different things but i can't seem to get it going |
| 19:45.59 | blitzrage | didn't get it working with the manager, but I'll figure it out later |
| 19:46.05 | [TK]D-Fender | blitzrage : Well if you want them calling in elsewhere, the challenge is concurrency . |
| 19:46.12 | blitzrage | [TK]D-Fender: this i realize |
| 19:46.19 | [TK]D-Fender | blitzrage : Sure, AMI Originate will work just fine too. |
| 19:46.32 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 19:46.39 | blitzrage | [TK]D-Fender: I couldn't get more than one Originate at a time for some reason |
| 19:46.46 | blitzrage | but like I said, I'm not going to fight that part anymore |
| 19:46.46 | [TK]D-Fender | blitzrage : Its all up to your eventual target to handle the incoming flood of calls. |
| 19:47.00 | [TK]D-Fender | blitzrage : What are you hoping to do with them if not a Page effectively? |
| 19:47.18 | blitzrage | actually.. nevermind.. Application would work for that |
| 19:47.26 | blitzrage | thanks guys... back to work! |
| 19:47.54 | *** part/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
| 19:48.12 | [TK]D-Fender | *whee* ! |
| 19:49.43 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 19:50.09 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 19:50.23 | [TK]D-Fender | Holy McFuck ...... SUV hits deer..... utter carnage.... http://www.cartype.com/page.cfm?id=668&alph=ALL&dec=ALL |
| 19:50.29 | [TK]D-Fender | *shudder* |
| 19:50.53 | *** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net) |
| 19:51.25 | Strom_C | wow |
| 19:51.25 | [TK]D-Fender | Wow, that was just psoted, then PULLED from Digg. |
| 19:51.33 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 19:51.51 | mercestes | LOL |
| 19:51.54 | mercestes | I missed it |
| 19:52.08 | [TK]D-Fender | mercestes : hence the survivability of the link I made |
| 19:52.18 | mercestes | oh, now it works. |
| 19:52.24 | mercestes | it was "page could not be displayed" first |
| 19:52.52 | _DAW | Wow, I wonder how much of that blood is the drivers? |
| 19:52.56 | mercestes | zomg... |
| 19:52.59 | mercestes | zomg....ROFL |
| 19:53.04 | mercestes | LMAO |
| 19:53.20 | mercestes | I want *one* more picture. |
| 19:53.30 | mercestes | the look on the driver's face about 2 seconds after impact. |
| 19:54.37 | [TK]D-Fender | mercestes : Very fortunate it was a doe, and not a buck. Those tend to completely annihilate acrs, and kill passengers.... |
| 19:55.23 | mercestes | Yea. |
| 19:55.41 | ^sandro^ | sorry guys i know this must be anoying but anyone ever see this error |
| 19:55.41 | ^sandro^ | Jan 17 19:54:46 ERROR[13013] chan_zap.c: Unable to reconfigure channel '145-168' |
| 19:55.41 | ^sandro^ | Jan 17 19:54:46 WARNING[13013] chan_zap.c: Reload of chan_zap.so is unsuccessful! |
| 19:56.01 | mercestes | ^sandro^: no, never. |
| 19:56.05 | ^sandro^ | maybe i will post everything on the pasteit.ca |
| 19:56.11 | ^sandro^ | ya i guess that was silly :( |
| 19:56.43 | mercestes | I've seen similar errors tho....but the channel numbers are different. |
| 19:56.58 | mercestes | it's a configuration error. |
| 19:57.12 | ^sandro^ | can someone have a look http://pastebin.ca/319796 and tell me what im doing wrong |
| 19:57.19 | ^sandro^ | at least guide me in the right direction |
| 19:58.05 | mercestes | ^sandro^: LOL http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA |
| 19:58.06 | perd | has anyone here used a phone connected to asterisk to dial an extension on a CCM system? i can call from CCM to asterisk but i'm having no luck calling from asterisk to ccm |
| 19:58.10 | mercestes | yea yea yea, ok, I'm looking now. |
| 19:58.24 | perd | i have ccm set up to dial asterisk via an h323 trunk (i have v3.3 so no sip trunk!) |
| 19:58.52 | ^sandro^ | u seriuos? :P |
| 19:59.16 | ^sandro^ | i thought someoen knows this stuff can just give me a hint |
| 19:59.28 | ^sandro^ | im a little confused about it i have done PRi's but never NFAS |
| 19:59.48 | mercestes | ^sandro^: Where is the dchan for span 7? |
| 19:59.48 | ^sandro^ | pretty please.... i would like to get this finished this afternoon and learn this stuff not get someone else to do it for me :( |
| 19:59.55 | ^sandro^ | i just need to be told what im doing wrong that's all |
| 20:00.22 | Strom_C | ^sandro^: you can't reconfigure signalling on a running asterisk system. |
| 20:00.37 | Strom_C | you need to shut down asterisk, run ztcfg, then start asterisk again |
| 20:02.30 | *** join/#asterisk romanoffi (n=robg@ip78.131.realtime.net) |
| 20:03.44 | tzanger | shit man deer parts in the rear of the vehicle... how fast were they going? |
| 20:04.22 | ^sandro^ | mercestes it has no |
| 20:04.23 | mikefoo | Hey I need upload a 50Meg file somewhere for a user tod ownload, any free ones out there that will allow 50megs? |
| 20:04.41 | mercestes | That doesn't freak me out.....because when i was young...my dad hit a cow..doing 120+. In a cadillace |
| 20:04.41 | ^sandro^ | StromC .. if it turn it off it should start with those settings |
| 20:05.01 | ^sandro^ | because honestly .. i have tried and it wont start until i comment out the channels 145-168 |
| 20:05.53 | Strom_C | did you run ztcfg? |
| 20:06.02 | [TK]D-Fender | mercestes : D-Chan is 24 (PRI #1's D-chan) |
| 20:07.58 | ^sandro^ | not yet.. waiting for less channels to be in use |
| 20:08.04 | Strom_C | .... |
| 20:08.05 | ^sandro^ | a lot of people talking on them |
| 20:08.06 | ^sandro^ | :) |
| 20:08.11 | Strom_C | you're doing this ona PRODUCTION SYSTEM?? |
| 20:08.24 | ^sandro^ | ya so i have to wait .. well not full production no |
| 20:08.28 | Strom_C | *thwap* |
| 20:08.35 | *** part/#asterisk andyshack (n=andyshac@reels.iinet.net.au) |
| 20:08.35 | ^sandro^ | its only our internal systems |
| 20:08.45 | ^sandro^ | but some of the staff is talking .. im waiting for them to finish |
| 20:09.13 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 20:09.28 | [TK]D-Fender | wow, Beyonce's "Irreplaceable" ... total ho :/ |
| 20:09.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:10.01 | Hmmhesays | [TK]D-Fender: my mod to app directory worked |
| 20:10.24 | Hmmhesays | now I need to figure out how to tell it to accept the call right away instead of reading off the entire name |
| 20:10.24 | [TK]D-Fender | Hmmhesays : Cool, going to post it to the tracker? |
| 20:10.30 | perd | damnit asteriskguru is still broken |
| 20:10.38 | Hmmhesays | i have no idea if I did it the way they would want |
| 20:10.49 | [TK]D-Fender | Hmmhesays : Easy check I guess... you just need to make sure there is only 1 valid choice. |
| 20:11.10 | [TK]D-Fender | Hmmhesays : Thats not the point, the fact is you DID it. A ccept first, tweak second. |
| 20:11.13 | Hmmhesays | well when it starts reading off an entire name, after so many characters you know if it is the one you want |
| 20:11.31 | [TK]D-Fender | Hmmhesays : you men its like a playback VS Background? |
| 20:11.49 | robl^ | its better if someone justr records their name instead of spelling it out. ;-) |
| 20:12.05 | Hmmhesays | [TK]D-Fender: when you dial app directory and it comes back after you've dial the 1st 3 digits, it reads off the entire name of the person |
| 20:12.06 | [TK]D-Fender | robl^ : Very true. |
| 20:12.36 | [TK]D-Fender | Hmmhesays : then your 1D10T users need to record their name for the directory. |
| 20:12.45 | [TK]D-Fender | :D |
| 20:12.58 | robl^ | users that do not setup their mailbox should be castrated by dull rusty cheese grater |
| 20:13.17 | Hmmhesays | [TK]D-Fender: not using app directory for that |
| 20:13.19 | [TK]D-Fender | robl^ : Nah.... a lead spork :) |
| 20:13.22 | Sweeper | mmmm |
| 20:13.25 | Hmmhesays | i'm building an only contacts book |
| 20:13.28 | Hmmhesays | *online |
| 20:13.37 | Sweeper | I get timing issues for both 0 and 1 options :/ |
| 20:13.58 | Sweeper | is it possible NOBODY is giving nice timing? |
| 20:14.02 | [TK]D-Fender | Hmmhesays : You should ahve a provision to record the name and hack it in then, or have an "operator" monitor new additions and do it. |
| 20:14.21 | [TK]D-Fender | Sweeper : Pastebin it. |
| 20:14.25 | Hmmhesays | or just have a option for the user to press a digit to accept |
| 20:14.45 | [TK]D-Fender | Hmmhesays : You ahve one already, jsut that you can't stop mid-stream, right? |
| 20:15.07 | Hmmhesays | correct |
| 20:15.07 | [TK]D-Fender | Hmmhesays : Another easy patch 'd bet. |
| 20:15.49 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:16.11 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:17.06 | mercestes | robl^: OMG..ow. |
| 20:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:18.11 | Sweeper | [TK]D-Fender: http://pastebin.com/861529 |
| 20:18.15 | romanoffi | Hi, I am an asterisk newbie, using a vmware version of AsteriskNow 1.4.0 beta 2, with a polycom 501 sip phone, and can't get the phone to register. The log says WARNING[4489] chan_sip.c: username mismatch, have <102>, digest has <robg>. Anyone have some hints on how to track this down? |
| 20:18.19 | ^sandro^ | ok i ran the ztcfg |
| 20:18.22 | ^sandro^ | it said.. |
| 20:19.07 | [TK]D-Fender | romanoffi : Shouldne' be messing with Digest, the user should match th context name "[]", not use "username". |
| 20:19.12 | ^sandro^ | http://pastebin.ca/319823 |
| 20:20.12 | ^sandro^ | any idea? |
| 20:20.17 | romanoffi | [TK]D-Fender: is the digest coming from the phone? |
| 20:20.17 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 20:20.31 | [TK]D-Fender | Sweeper : linking to a PBX? |
| 20:20.38 | Sweeper | [TK]D-Fender: sadly, yes D: |
| 20:20.40 | [TK]D-Fender | romanoffi : Just DON'T, ok? |
| 20:21.22 | [TK]D-Fender | Sweeper : Ok, pastebin (.ca please), "cat/proc/interrupts". I'd bet its sharing and screwing with your timing. And being NET, it should BE the timing source (0) |
| 20:21.30 | romanoffi | [TK]D-Fender: ok, but not sure how to act on your advice |
| 20:21.55 | [TK]D-Fender | romanoffi : REMOVE it entirely. |
| 20:22.22 | [TK]D-Fender | romanoffi : And use pastebin.ca to show your sip config. |
| 20:23.02 | romanoffi | [TK]D-Fender: ok, will do |
| 20:23.16 | Sweeper | [TK]D-Fender: http://pastebin.com/861537 |
| 20:23.40 | Strom_C | ^sandro^: that's not output from ztcfg |
| 20:23.46 | *** part/#asterisk CANO-1982 (i=alejandr@190.48.68.151) |
| 20:24.17 | Sweeper | oops, forgot .ca |
| 20:27.15 | ^sandro^ | yes i know.. i wasn't sure what the deal was there |
| 20:27.33 | ^sandro^ | thats someting in the spanmap line |
| 20:27.41 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 20:28.06 | *** join/#asterisk backblue (n=moo@87-196-3-227.net.novis.pt) |
| 20:29.03 | *** join/#asterisk guilherme_jorge (n=guilherm@200-170-201-134.core01.spo.ifx.net.br) |
| 20:29.11 | ^sandro^ | Srom_C > http://pastebin.ca/319829 |
| 20:29.17 | ^sandro^ | Strom_C* |
| 20:30.01 | Katty | </random> |
| 20:30.07 | mercestes | Congratz, katty |
| 20:30.19 | romanoffi | [TK]D-Fender: http://pastebin.ca/319831 for sip.conf |
| 20:30.21 | Katty | heh |
| 20:30.23 | Sweeper | is that wicca? |
| 20:30.42 | Hmmhesays | D&D |
| 20:30.42 | Katty | why does everyone think everything has to do with wicca? |
| 20:30.51 | Hmmhesays | or some equivalent |
| 20:30.59 | mercestes | Sweeper: Satanism. |
| 20:31.19 | [TK]D-Fender | Katty : Mew |
| 20:31.22 | Katty | [TK]D-Fender: mew. |
| 20:31.22 | tzafrir_laptop | ^sandro^, if you have just one span, I don't think you need a spanmap |
| 20:31.23 | ^sandro^ | i think there is someting wrong with my spanmap line |
| 20:31.48 | ^sandro^ | tzafrir what do you mean |
| 20:31.54 | ^sandro^ | well i have two cards |
| 20:31.57 | perd | ~gatekeeper |
| 20:32.07 | jbot | it has been said that gatekeeper is A component of the ITU H.323 "umbrella" of standards defining real-time multimedia communications and conferencing for packet-based networks. The gatekeeper is the central control entity that performs management functions in a Voice and Fax over IP network and for multimedia applications such as video conferencing. Gatekeepers ... |
| 20:32.07 | ^sandro^ | and 4 ports each |
| 20:32.07 | Hmmhesays | sunday hopefully I get my Wii |
| 20:32.07 | tzafrir_laptop | ^sandro^, but you do need a definition of the span and the channels (or bchan/dcha-s) |
| 20:32.07 | ^sandro^ | 7 are in use.. |
| 20:32.17 | ^sandro^ | tzafrir i get that.. but i have other channels and b channels in use |
| 20:32.22 | ^sandro^ | each port is a unique PRI |
| 20:32.30 | ^sandro^ | but span 7 ( card 2 port 3 ) |
| 20:32.38 | tzafrir_laptop | ^sandro^, sorry, my mistake: the definition of the channels in the section [channels] |
| 20:32.54 | ^sandro^ | has no d channels.. only b channels .. it has to use the D channel from Span 1 ( D channel 24 ) |
| 20:33.30 | [TK]D-Fender | romanoffi : where the hell is your phone's definition? |
| 20:33.34 | perd | can anyone recommend a gatekeeper daemon? |
| 20:33.48 | guilherme_jorge | Hello all, I'll sell voip for home clients, and any of the things that I think is does some tests in a workstation of my client. These tests include calculate jitter, latency and some tests in SIP ports. Do you know any software that does it? |
| 20:34.23 | [TK]D-Fender | perd : Key Master :) |
| 20:35.24 | Sweeper | [TK]D-Fender: any ideas? :/ |
| 20:35.25 | perd | how about some docs about what exactly a gatekeeper does? :) |
| 20:35.25 | ^sandro^ | tzafrir can you tell me what im doing wrong there.. ? |
| 20:35.25 | [TK]D-Fender | Sweeper : I'd first start thinkg that order MATTERS., and move the spanmap up. |
| 20:35.58 | EmleyMoor | What's a good webcam for use with Linux softphones? |
| 20:36.08 | EmleyMoor | (preferably a current one) |
| 20:36.13 | tzafrir_laptop | ^sandro^, and anyway, you mentioned the output of ztcfg: does a plain 'ztcfg' run suceessfully? |
| 20:37.06 | *** join/#asterisk calvinhp (n=calvinhp@rrcs-24-172-172-92.central.biz.rr.com) |
| 20:37.34 | EmleyMoor | How can I stop # being turned into %23? |
| 20:37.37 | ^sandro^ | yes i think it does.. and it configures the 168 channels |
| 20:37.48 | calvinhp | anyone using Cisco 7900 hardware with the 8.6 SIP firmware and asterisk? |
| 20:38.44 | EmleyMoor | (whenever I try to dial a # on a SIP softphone, it comes through as %23 - seen with ekiga and SJphone) |
| 20:38.50 | EmleyMoor | (on moziax, # works |
| 20:38.52 | EmleyMoor | ) |
| 20:38.58 | Sweeper | [TK]D-Fender: huh? the spanmap is at the top of zaptel.conf :/ |
| 20:40.19 | [TK]D-Fender | EmleyMoor : "pedantic=yes" in [general] |
| 20:40.27 | EmleyMoor | Ah... |
| 20:41.31 | EmleyMoor | That worked - thanks |
| 20:41.40 | EmleyMoor | Won't often need it |
| 20:41.42 | [TK]D-Fender | EmleyMoor : quite welcome |
| 20:44.13 | *** join/#asterisk gordonjcp (n=gordonjc@cpc1-broo2-0-0-cust991.renf.cable.ntl.com) |
| 20:44.31 | bagpuss_thecat | morning gordonjcp |
| 20:44.36 | Qwell[] | gordonjcp: :D |
| 20:45.15 | bagpuss_thecat | is anyone able to explain these warnings that repeat themselves every 30 seconds? http://pastebin.co.uk/9067 |
| 20:45.27 | gordonjcp | Qwell[]: hello |
| 20:45.30 | Hmmhesays | hmm I'm not sure how I would make directory return while it is reading back the users name |
| 20:45.32 | gordonjcp | long time no see |
| 20:45.41 | gordonjcp | hey bagpuss_thecat |
| 20:45.47 | Qwell[] | gordonjcp: I was talking about you (and your playtones stuff) the other day |
| 20:45.51 | bagpuss_thecat | externip is set correctly, localnet is correct, and it only happens when my Sipura 2k is registered |
| 20:45.58 | Qwell[] | gordonjcp: and I realized, you never did see what I did with it, did you? |
| 20:46.05 | gordonjcp | uhm, possibly not... |
| 20:46.18 | gordonjcp | dear god, what monster have I unleashed? |
| 20:46.28 | Qwell[] | http://uuoc.com/1683 also 1684 and 1685 :p |
| 20:47.25 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:47.28 | gordonjcp | right, time to start working on getting an Asterisk server going again then ;-) |
| 20:48.05 | Dr-Linux|home | anybody is using DUNDi? |
| 20:48.11 | *** join/#asterisk kirberich (n=robert@i53871022.versanet.de) |
| 20:48.21 | kirberich | good evening |
| 20:48.55 | Dr-Linux|home | hhm.. |
| 20:49.03 | Dr-Linux|home | what i'll say if here is midnight |
| 20:50.15 | kirberich | i'm trying to set up a first asterisk test-server here and i appear to be a little stupid. the question is: what do i have to do to get asterisk to know that i'm going to call from a softphone over sip? |
| 20:50.23 | kirberich | i can't get the softphone to connect to the server |
| 20:51.26 | Strom_C | edit sip.conf |
| 20:51.27 | Dr-Linux|home | kirberich: what softphone? is your softphone registered with your asterisk server? |
| 20:52.26 | kirberich | Strom_C, that much i did |
| 20:52.54 | Dr-Linux|home | kirberich: what softphone you are using? |
| 20:52.55 | kirberich | Dr-Linux, xten, i entered username and password in both client and server, and i entered the servers ip adress in xten |
| 20:53.10 | Strom_C | kirberich: and then you reloaded the sip rules in asterisk? |
| 20:53.33 | kirberich | i restarted asterisk |
| 20:53.53 | [TK]D-Fender | kirberich : what is your specific problem? |
| 20:53.59 | Dr-Linux|home | kirberich: pastebin your sip.conf users settings |
| 20:54.21 | kirberich | [TK]D-Fender, xten says "awaiting proxy login information" and does nothing more |
| 20:55.00 | [TK]D-Fender | kirberich : Sounds like your SIP user isn't set up right. please continue to paste your sip.conf in pastebin.ca |
| 20:55.13 | kirberich | ok, hold on |
| 20:56.35 | kirberich | http://pastebin.ca/319854 it's essentially just the sample sip.conf from the book though |
| 20:56.37 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:56.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:57.22 | [TK]D-Fender | kirberich : looks simple enough, though yoy should specify codecs, etc... |
| 20:57.43 | [TK]D-Fender | kirberich : where is your * server relative to your soft-phone? |
| 20:58.11 | kirberich | on the same computer, i also tried it from a different computer on the same network though |
| 20:58.26 | kirberich | ok i believe my problem is that i don't really know how to start the server |
| 20:58.38 | kirberich | i just now started it with the safe_asterisk script and i get this: |
| 20:58.50 | [TK]D-Fender | kirberich Try it from the other one, having enable SIP debugging "sip debug on" |
| 20:58.52 | kirberich | Asterisk died with code 127. |
| 20:59.16 | [TK]D-Fender | kirberich : Kill off all instances of *, and start it manually for testing like "asterisk -gvvvvvvc |
| 21:01.28 | kirberich | [TK]D-Fender, where am i supposed to enter "sip debug on"? |
| 21:01.38 | [TK]D-Fender | kirberich : in the * CLI. |
| 21:02.01 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 21:02.02 | kirberich | it gives me a usage message |
| 21:02.03 | [TK]D-Fender | kirberich L hoever your CURRENT problem is that * doesn't even LOAD right. Its crashing |
| 21:02.43 | [TK]D-Fender | kirberich : that should have been "sip debug |
| 21:02.52 | kirberich | ah, k |
| 21:03.21 | kirberich | and now i just try and connect with xten again? |
| 21:03.29 | [TK]D-Fender | sure |
| 21:03.34 | kirberich | alright, brb |
| 21:03.40 | [TK]D-Fender | and pastebin all of the output |
| 21:06.09 | bagpuss_thecat | gordonjcp: ping |
| 21:09.05 | BSDTech | ls |
| 21:10.06 | kirberich | [TK]D-Fender, does asterisk write that output to a file somewhere? cause it's a lot a lot of output |
| 21:11.16 | kirberich | anyway, here's some of the output, maybe that's enough already |
| 21:11.17 | kirberich | http://pastebin.ca/319868 |
| 21:13.19 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 21:13.25 | Hmmhesays | app_directory doesn't seem to like extconfig |
| 21:14.11 | [TK]D-Fender | kirberich : I need ti all... |
| 21:14.23 | kirberich | alright ;) |
| 21:15.28 | *** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:16.01 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
| 21:17.03 | Hmmhesays | bah |
| 21:17.26 | perd | i cant believe ccm doesnt have a damn sip daemon |
| 21:17.35 | perd | and i cant connect via h323 for some reason |
| 21:17.36 | perd | oi! |
| 21:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:18.26 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 21:18.28 | kirberich | actually [TK]D-Fender, the problem just mysteriously disappeared |
| 21:18.37 | kirberich | it works now :) |
| 21:18.55 | kirberich | i must have done somthing stupid, but i really don't know what |
| 21:19.13 | kirberich | well thanks a lot! |
| 21:20.23 | Dr-Linux|home | anybody is using DUNDi? |
| 21:20.26 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 21:21.20 | nortex | Dr-Linux, Yup, works great once you get it setup. |
| 21:21.22 | hoobastooba | i am looking for info on creating an SIP hotline phone. One that when you pick up the handset it dials a predefined number. I see how to do it with zap... but do not see any other options. is it only possible via zap? |
| 21:23.08 | [TK]D-Fender | hoobastooba : depends on the phone. Some can be configured to do that. |
| 21:23.25 | hoobastooba | could i do it with a polycom 301? |
| 21:23.59 | [TK]D-Fender | hoobastooba : never tried. |
| 21:24.00 | Dr-Linux|home | nortex: for what purpose you use DUNDi? |
| 21:24.00 | gordonjcp | you can do it with an Avaya 4602 |
| 21:24.00 | gordonjcp | bagpuss_thecat: pong? |
| 21:24.04 | Dr-Linux|home | nortex: as it's not clear to me |
| 21:24.44 | Sweeper | [TK]D-Fender: http://pastebin.ca/319885 <-- shouldn't this show up "moo cow" as caller id when calling aclark? |
| 21:25.16 | Strom_C | Sweeper: no |
| 21:25.28 | Strom_C | Set(CALLERID(all)="Moo Cow"<1234>) |
| 21:25.35 | [TK]D-Fender | Sweeper : clearly should ahve the # IN the quotes,and you should break it up. |
| 21:25.43 | Dr-Linux|home | nortex: if my servers are conntected to each other via IAX trunk and i can do dial from other server if local PSTN doesn't response, what's the difference? |
| 21:25.45 | ManxPower | Sweeper: It might if you REMOVED THE QUOTES |
| 21:25.49 | bagpuss_thecat | gordonjcp: your asterisk server up and running? |
| 21:26.00 | ManxPower | Some phone CHOKE on Quotes |
| 21:26.11 | gordonjcp | bagpuss_thecat: not at the moment |
| 21:26.23 | gordonjcp | well, maybe it is, I don't know |
| 21:26.27 | Sweeper | Strom's example dinna work |
| 21:26.36 | *** join/#asterisk rc-1 (n=rc-1@ip68-229-102-1.hr.hr.cox.net) |
| 21:26.52 | bagpuss_thecat | gordonjcp: lol |
| 21:27.09 | gordonjcp | 2 tics |
| 21:27.27 | gordonjcp | seems to be... |
| 21:27.34 | gordonjcp | let me hook the phone up |
| 21:27.41 | [TK]D-Fender | Sweeper : do it in 2 steps. Set(CALLERID(name)=moo cow) |
| 21:27.41 | Dr-Linux|home | DUNDi we only use to share the extensions? what if primary asterisk is down? i mean the local one? |
| 21:27.42 | *** join/#asterisk amdtech (i=adaniel@nat/digium/x-404855a2947fc2c8) |
| 21:27.55 | [TK]D-Fender | Sweeper : Set(CALLERID(num)=1234) |
| 21:28.56 | *** join/#asterisk shepimport (n=FreePBX4@h194.189.31.71.ip.alltel.net) |
| 21:29.37 | Sweeper | ok, that works |
| 21:29.56 | Sweeper | interestingly, it works from sip -> * -> pbx, but not vice versa :/ |
| 21:29.58 | shepimport | Hey guys - has anyone delt with any QoS appliances that handle ingress traffic as well as egress?> |
| 21:30.04 | Sweeper | sip = x-lite |
| 21:30.35 | Dr-Linux|home | [TK]D-Fender: what about if we put a variable instead of 1234? |
| 21:30.49 | gordonjcp | bagpuss_thecat: phone is booting |
| 21:31.02 | gordonjcp | now to navigate the stinking morass that is linux softphones |
| 21:31.15 | [TK]D-Fender | Dr-Linux : same thing. its just text substitution. |
| 21:32.06 | bagpuss_thecat | gordonjcp: it was just so I could steal your configs :-) |
| 21:32.22 | gordonjcp | bagpuss_thecat: ah |
| 21:32.26 | gordonjcp | tried wheelybin? |
| 21:32.29 | ManxPower | Does anyone know of a company that, given a credit card number, can issue a calling card and pin on the spot, via a web interface? |
| 21:32.44 | Dr-Linux|home | would it be like Set(CALLERID(num)=${myvariable}) ? |
| 21:32.54 | Dr-Linux|home | err |
| 21:33.09 | mog | ManxPower, it wouldnt be that |
| 21:33.10 | mog | hard |
| 21:33.24 | mog | could be down with some agi and php or ruby in a few hours |
| 21:33.27 | [TK]D-Fender | Dr-Linux : sure, why not... |
| 21:33.36 | ManxPower | I do not require international calling, USA only. |
| 21:33.40 | [TK]D-Fender | Dr-Linux : Depening of course on its CONTENTS. |
| 21:33.44 | ManxPower | mog: I don't want to deal with it at all |
| 21:33.51 | mog | ahh |
| 21:34.05 | wunderkin | credit card merchant accounts = big pita |
| 21:34.17 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 21:34.38 | Dr-Linux|home | i see |
| 21:34.56 | robin_sz | hear the dtmf you mISDN software you. |
| 21:35.21 | *** join/#asterisk Manax (n=toertel-@server.samasher.com) |
| 21:35.37 | *** join/#asterisk karmatronic (n=karmatro@84.77.148.117) |
| 21:36.47 | Dr-Linux|home | [TK]D-Fender: i read some about DUNDi, but i'm little confused why i need DUNDi if i'm uisng IAX trunk between two servers |
| 21:37.09 | Dr-Linux|home | [TK]D-Fender: maybe i didn't understand DUNDi as it works |
| 21:39.28 | ManxPower | Dr-Linux: ALL DUNDi does is help you route calls based on the dialed number. |
| 21:39.33 | [TK]D-Fender | Dr-Linux : Clearly NOT. Go back to reading. |
| 21:39.41 | ManxPower | If your call routing between servers is not complex then you should not use DUNDi |
| 21:39.50 | [TK]D-Fender | Dr-Linux : Odds are if you don't think you need it, you probably DON'T. |
| 21:40.12 | ManxPower | *grumble* The ONLY CLEC serving my CO is IDC Deltacom |
| 21:41.53 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:42.22 | [TK]D-Fender | ManxPower : so much for "C". |
| 21:42.46 | wunderkin | a is cool! |
| 21:43.18 | ManxPower | [TK]D-Fender: It's a small CO |
| 21:44.25 | Dr-Linux|home | ManxPower: actually i'm looking for asterisk servers redundancy solution, someone suggested me DUNDi |
| 21:45.39 | brettnem | ew |
| 21:45.59 | brettnem | if you need redundancy, look for another product. :) |
| 21:46.04 | ManxPower | Dr-Linux: DUNDi was not designed for redundancy, but you might be able to use it to help you create a redundant system. |
| 21:46.11 | brettnem | you don't really need asterisk anyway, right? :) |
| 21:46.31 | brettnem | DUNDi really only shines when you have extensions.conf files changing constantly |
| 21:46.34 | ManxPower | Many people don't realize that Asterisk is NOT really a PBX. |
| 21:46.43 | Dr-Linux|home | brettnem: what opensource product is there for redundancy? :) |
| 21:46.45 | ManxPower | It is a telephony toolkit that allows you to BUILD a PBX |
| 21:46.45 | robin_sz | if you need redundancy, insult your boss! :) |
| 21:47.02 | brettnem | SER can do some nice redundancy tricks |
| 21:47.21 | BSDTech | openser is nice |
| 21:47.23 | bkruse | brettnem: uh, ya |
| 21:47.25 | BSDTech | when you need it |
| 21:47.26 | bkruse | BSDTech: uh, ya |
| 21:47.27 | bkruse | lol |
| 21:47.35 | Dr-Linux|home | :S |
| 21:47.53 | brettnem | openser is really nice |
| 21:48.02 | brettnem | have you tried the perl module.. wow.. it's incredible. :) |
| 21:48.09 | Dr-Linux|home | what you guys think about "www.linux-ha.org" ? |
| 21:48.16 | brettnem | blah |
| 21:48.38 | brettnem | I don't think the HA stuff is going to do what you want with asterisk |
| 21:48.51 | brettnem | the shared memory stuff and all, it just won't be what you'd expect when you are done. |
| 21:48.52 | perd | ha+LVS will though |
| 21:49.05 | perd | oh shared memory stuff? craziness |
| 21:49.48 | JT | robin_sz: what hardware are you using misdn with? |
| 21:49.49 | Dr-Linux|home | brettnem: well, i just want, like i have 2 identical servers, if one goes down other should takeover the things |
| 21:50.01 | brettnem | so just rsync the two together |
| 21:50.08 | brettnem | then use UCARP |
| 21:50.40 | brettnem | maybe you can write a script that sends an OPTIONS and looks for a ping and triggers UCARP when the OPTIONS test fails.. That's what I've done |
| 21:50.49 | Dr-Linux|home | brettnem: yes, i can resync them together, but .. hhm.. what's UCARD? is it a hardware? |
| 21:50.56 | brettnem | I used http://argus.tcp4me.com to detect the SIP failure |
| 21:51.09 | brettnem | no UCARP is software.. IP heartbeat |
| 21:51.24 | brettnem | I'll like you |
| 21:51.26 | brettnem | oops |
| 21:51.26 | brettnem | haha |
| 21:51.28 | brettnem | link you |
| 21:51.37 | brettnem | ~ucarp |
| 21:51.40 | Dr-Linux|home | brettnem: can you give me a link |
| 21:51.54 | robin_sz | JT: cologne/HFC card |
| 21:51.58 | Dr-Linux|home | jbot wakeup |
| 21:52.05 | brettnem | jbot: ucarp information is at http://www.ucarp.org/project/ucarp |
| 21:52.06 | jbot | okay, brettnem |
| 21:52.11 | brettnem | ~ucarp |
| 21:52.12 | gordonjcp | bagpuss_thecat: ping? |
| 21:52.12 | JT | is there any reason you must use misdn? |
| 21:52.40 | brettnem | worked nicely for me.. very fast. |
| 21:52.59 | brettnem | I used a cronjob to sync up conf files, audio files and /var/lib stuff (ugly, but works) |
| 21:53.14 | Dr-Linux|home | brettnem: how it takes hosts decision? :S |
| 21:53.17 | brettnem | but really, asterisk is a total dog in redundant applications and you'll never really be happy. :) |
| 21:53.23 | brettnem | well |
| 21:53.29 | JT | robin_sz: |
| 21:53.33 | brettnem | one can be elected as the "primary" |
| 21:53.38 | Dr-Linux|home | brettnem: yes that what i think |
| 21:53.53 | robin_sz | JT, ? |
| 21:53.55 | JT | Dr-Linux|home: it also really depends what the asterisk system atually does |
| 21:54.00 | brettnem | and then they share an encrypted heartbeat and make a decision to takeover the main ip when the heartbeat fails |
| 21:54.01 | JT | robin_sz: < JT> is there any reason you must use misdn? |
| 21:54.14 | Dr-Linux|home | brettnem: maybe ISDNgaurd hardware will good option |
| 21:54.19 | brettnem | yeah, which is why I send an OPTIONS to the server to determine if it's dead or nto |
| 21:54.28 | robin_sz | JT, to connect my * server to my isdn line? |
| 21:54.33 | brettnem | Dr-Linux: you'll need that if you need to protect trunks |
| 21:54.42 | JT | robin_sz: well there are other options |
| 21:55.27 | *** join/#asterisk data23 (n=data@92.b6.3845.static.theplanet.com) |
| 21:55.29 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:55.30 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:55.49 | Dr-Linux|home | brettnem: hhm.. as my asterisk servers are trunked with each others |
| 21:55.55 | brettnem | oh.. blah.. these SIT tones will be the end of me |
| 21:56.03 | brettnem | trunked how? |
| 21:56.18 | Dr-Linux|home | brettnem: my telco is happy to provide me redundant PRI's |
| 21:56.20 | brettnem | anyone know how to detect SIT tones in SIP early media? :) I'll pay you to do it. :) |
| 21:56.31 | JT | Dr-Linux|home: describe your asterisk server setup |
| 21:56.33 | brettnem | Dr-Linux: so sounds like you are good then |
| 21:56.40 | JT | what does it terminate, what does it do? |
| 21:57.05 | wunderkin | it terminates people |
| 21:57.09 | brettnem | Why would they send me SIT tones, it's rediculous. |
| 21:57.24 | Dr-Linux|home | but PRI is not a only things, there is asterisk/SIP/IAX stuff also MySQL DB etc |
| 21:57.44 | JT | be more specific, we can't read mines |
| 21:57.53 | JT | what phsical and ip connections does it terminate? |
| 21:57.57 | JT | physical |
| 21:57.58 | Strom_C | ah, mine-reading |
| 21:58.01 | Strom_C | don't lose your head |
| 21:58.10 | JT | gar, it's early :) |
| 21:58.11 | Strom_C | </horrible horrible joke> |
| 21:58.22 | robin_sz | JT, and those other options are? |
| 21:58.44 | JT | robin_sz: bristuff, which i would recommend, and visdn to a lesser extent |
| 21:58.47 | brettnem | zoa, you there? |
| 21:58.54 | JT | i found misdn completely unusable |
| 21:59.10 | JT | tonnes of bugs, and it is extremely clunky |
| 21:59.16 | *** part/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:59.27 | Dr-Linux|home | JT: i'm concerend with only three things Asterisk service, MySQL DB and PRI |
| 21:59.50 | robin_sz | JT, I tried the xorcom rapid / bristuffed install ... couldnt get it to work with my cologne/hfc card |
| 22:00.08 | Dr-Linux|home | JT: if any of them goes down, the system should switch to other identical server seamlesly |
| 22:00.13 | JT | Dr-Linux|home: ok, so the PRI you should consider either an L1 failover device, or the redfone PRI to TDMoE convertor |
| 22:00.21 | brettnem | bah redfone |
| 22:00.34 | JT | brettnem: don't like it? |
| 22:00.46 | brettnem | why not just use TDMOE in ast? |
| 22:00.53 | JT | err |
| 22:00.59 | data23 | hi, i wonder if someone can point me in the right direction... I have an Asterisk box connected to a BT Meridian PBX via a homemade PRI, if i route a call to the Asterisk box for IVR treatment and it decides it needs to dial out to the meridian again, how can i get it to 'hand-off' the call? It currently uses 2 channels, one for inbound, one for outbound. |
| 22:01.03 | Dr-Linux|home | JT: well, PRI issue is resolved, as my telco is happy to give me redundant pri's |
| 22:01.09 | JT | because how do you get your PRI into TDMoE, brettnem ? |
| 22:01.11 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 22:01.12 | brettnem | I haven't looked at it in a year so I may not be giving it a fair assesment |
| 22:01.14 | Dr-Linux|home | JT: but about other 2 things |
| 22:01.28 | brettnem | JT: Asterisk supports that capability.. what do you think redfone is doing |
| 22:01.48 | JT | brettnem: i know, so how do you convert your PRI into TDMoE, brettnem ? |
| 22:02.06 | gordonjcp | hm |
| 22:02.13 | JT | the whole point of it is to allow redundant servers to use the same PRI |
| 22:02.16 | gordonjcp | how do I recover a forgotten iaxtel password? |
| 22:02.18 | JT | accessing the PRI via TDMoE |
| 22:02.41 | bagpuss_thecat | gordonjcp: pong |
| 22:02.52 | bagpuss_thecat | gordonjcp: we'll need to get an IAX link set up soon |
| 22:02.52 | gordonjcp | bagpuss_thecat: fancy getting a trunk going? |
| 22:02.56 | gordonjcp | heh |
| 22:02.56 | bagpuss_thecat | lol |
| 22:03.04 | bagpuss_thecat | not tonight though |
| 22:03.18 | gordonjcp | aww |
| 22:03.24 | bagpuss_thecat | well, ok then |
| 22:03.36 | bagpuss_thecat | make it quick though, I should've been sleeping ages ago with this flu... |
| 22:04.03 | bagpuss_thecat | I have iax udp forwarded on my router |
| 22:04.29 | JT | Dr-Linux|home: redundant pris sounds needlessly expensive and complex |
| 22:05.12 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 22:05.18 | Manax | Can anyone point me in the right direction regarding setting up a PRI to VoIP gateway? I've got digium cards, that are working fine under *, as is the VOIP conn. I think it's just a matter of doing the right thing with extensions.conf... |
| 22:05.41 | Dr-Linux|home | JT: i'm already using 2 for each server, and each server has 2 T1 cards(2 port) |
| 22:05.53 | JT | i see |
| 22:06.18 | Dr-Linux|home | JT: and my first PRI is connected to the first card's first port and 2nd is to 2nd card's first port |
| 22:06.45 | Dr-Linux|home | JT: this way if card or PRI anyone goes down, other will take over the things |
| 22:07.00 | JT | sure, sounds like it will work |
| 22:07.08 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 22:07.09 | JT | obviously active calls will be lost |
| 22:07.26 | Dr-Linux|home | JT: my zaptel configuration is also something different |
| 22:07.32 | Techie-Micheal | JT: Incidentally, mad|cow and my problem seem to be the exact same thing. |
| 22:07.47 | Dr-Linux|home | JT: yes correct |
| 22:07.55 | JT | Techie-Micheal: fixed it? |
| 22:08.08 | Dr-Linux|home | JT: but now things are asterisk and MySQL DB |
| 22:08.19 | Techie-Micheal | No, but we think we know what it is, and I'm about to test it right now. |
| 22:08.36 | JT | Techie-Micheal: that's cool, let me know how it goes |
| 22:09.19 | Techie-Micheal | Sure. What's really odd is that if I take out the secret for a sip.conf entry, asterisk replies with a 200 instead of a 401, but the softphone still doesn't work. |
| 22:13.48 | kirberich | ok i got a lot further since my last problem, but now i have the problem that i can't call a sip softphone |
| 22:13.57 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-137.mtco.com) |
| 22:14.11 | kirberich | the message i get is Jan 17 23:12:45 NOTICE[12096]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 22:15.05 | JT | is it connected to the Internet? |
| 22:15.28 | JT | or wherever the sip server is |
| 22:16.01 | kirberich | no, server and client are on the same machine. with two machines on the same network i get the same result |
| 22:16.35 | [TK]D-Fender | kirberich : use 2 spereate machine. next, pastebin the output of the failed call w/ SIP debug. |
| 22:17.00 | Dr-Linux|home | brettnem: ucarp.org has no documention or anything except download |
| 22:17.12 | robin_sz | JT, what do you think might work then for ISDN? I just need bri-stuffed? doesn't that also need zaphfc to work? I had that installed and I apparently had bristuffed installed too, the configs looked right and zttool showed the card, in an un-alarmed state etc, but dialling in produced nothing. not a suasage. not the faienst glimmer of a event on the CLI, no one seemed to know anyting about zaphfc, so I went with mISDN as it seemed to work out of |
| 22:17.13 | robin_sz | the box, except for this DTMF annoyance |
| 22:17.23 | kirberich | i haven't yet found a good way to do that though, i get several pages of terminal output which i can't just copy and paste |
| 22:17.30 | robin_sz | oh, and it crashes occasionally, once a month or so |
| 22:17.50 | [TK]D-Fender | kirberich : why not? |
| 22:17.58 | JT | robin_sz: zaphfc is just the driver, bristuff includes that |
| 22:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:18.24 | kirberich | well at least in xterm i don't seem to be able to select more than one screen full of text |
| 22:19.09 | robin_sz | JT, well, as I said, I never saw any activity in the CLI when trying to dial in, it looked to all intents and purposes, dead. |
| 22:19.11 | JT | robin_sz: hrm, i assume you set the line signalling correctly? |
| 22:19.19 | JT | anything in bri intense debug? |
| 22:19.39 | robin_sz | in what? |
| 22:19.48 | [TK]D-Fender | kirberich : What does your non-server system run? |
| 22:20.05 | kirberich | what os you mean? |
| 22:21.35 | [TK]D-Fender | yes |
| 22:21.46 | kirberich | gentoo |
| 22:21.53 | Sweeper | :v |
| 22:22.07 | JT | robin_sz: signalling in /etc/asterisk/zapata.conf, bri intense debug <span> in console |
| 22:22.21 | [TK]D-Fender | kirberich : well you should havea choice to use a terminal that will let you select more... |
| 22:22.42 | JT | robin_sz: that's another thing, the debugging in misdn is horrible in comparison to bristuff imho |
| 22:23.14 | robin_sz | JT I had: |
| 22:23.16 | robin_sz | switchtype = euroisdn |
| 22:23.16 | robin_sz | signalling = bri_cpe |
| 22:24.18 | JT | was your line configured by your telco as point to point? |
| 22:24.43 | robin_sz | ? |
| 22:24.46 | robin_sz | dunno. |
| 22:24.59 | JT | it's sort of important |
| 22:25.05 | JT | bris can be p2p or p2mp |
| 22:25.15 | JT | the correct signalling needs to be set |
| 22:25.31 | robin_sz | in misdn I have: te_ptmp=1 |
| 22:25.40 | JT | bri_cpe configures bristuff to use point 2 point signalling |
| 22:25.47 | JT | ah, well that's the other setting |
| 22:25.49 | Dr-Linux|home | what's the difference btween sipura 2000 and 2100 except extra PC port? :S |
| 22:25.53 | JT | that's point to multipoint |
| 22:26.00 | robin_sz | riiiight .. |
| 22:26.02 | JT | maybe your line is p2mp (a lot are) |
| 22:26.10 | Sweeper | mmm |
| 22:26.16 | robin_sz | and in bristuffed I should use what as a setting then? |
| 22:26.25 | Sweeper | anyone have some example traditional pbx quotes lying around? |
| 22:26.33 | JT | robin_sz: what country is your line in? |
| 22:26.36 | robin_sz | UK |
| 22:26.45 | JT | robin_sz: bri_cpe_ptmp |
| 22:27.39 | JT | either your telco can tell you if your line is p2p or p2mp, or you can with an oscilloscope |
| 22:27.44 | JT | a p2mp line will usually shutdown line sync after a period of inactivity |
| 22:27.49 | *** join/#asterisk nick125_ (n=ndevito@atarack/staff/nick125) |
| 22:27.51 | JT | as in no carrier |
| 22:27.59 | nick125_ | hiya guys |
| 22:28.53 | robin_sz | JT hmmm ... OK, I might try and test this ... |
| 22:29.01 | nick125_ | they better deliver my pap2t-na before 4:30pm |
| 22:29.15 | JT | robin_sz: the most reliable method is to just ask them :) |
| 22:29.22 | mercestes | nick125_: If they don't...you shoudl egg the delivery driver. |
| 22:29.24 | djflux | it's already 5:29 pm ET |
| 22:29.42 | djflux | so they have either 1 minute, 1:01 or 2:01 |
| 22:30.01 | nick125_ | its 3:30pm here |
| 22:30.14 | nick125_ | or somewhere around 3:30pm |
| 22:31.02 | nick125_ | brb |
| 22:31.03 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 22:32.03 | JT | robin_sz: if it's a home bri, it's more likely to be a p2mp |
| 22:32.19 | lenne_dk | can I in the dialplan figure out if somebody on ext A is talking to ext B? |
| 22:32.29 | cpm | who in the heck has a PRI at home? |
| 22:32.59 | robin_sz | JT: for some reason I'm now getting "unknown symbol in module" when insmoding zaphfc ... sigh |
| 22:33.07 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:33.09 | lenne_dk | not Pri, Bri, cpm |
| 22:33.10 | robin_sz | cpm, what makes you say that? |
| 22:33.17 | cpm | ahh, k |
| 22:33.19 | calvinhp | Are there issues running the 8.6 SIP firmware for Cisco 7900 phones with Asterisk? |
| 22:33.20 | cpm | sorry, misread |
| 22:33.23 | kirberich | [TK]D-Fender, i might have to continue with this tomorrow, so for now thanks again |
| 22:34.36 | lenne_dk | I'm supposed to get 8MB adsl, if the ISP can get their fingers out... |
| 22:34.41 | JT | sp0n9e: bri |
| 22:34.54 | JT | gar |
| 22:34.57 | Sweeper | lenne_dk: if the ISP is stretching, you're not gonna get 8mb |
| 22:34.57 | JT | cpm i meant |
| 22:35.07 | mercestes | I want an 8mb adsl. |
| 22:35.09 | Sweeper | DSL speed decreases as distance increases D: |
| 22:35.09 | robin_sz | JT, ahh, poo. it seems I still have an older version of zaphfc hanging aroudn from before I installed * with misdn |
| 22:35.12 | DonX | depends on the type of dsl |
| 22:35.27 | DonX | And is this Euro or America? |
| 22:35.32 | JT | cpm: an actual T1 from telco, or just a t1 going to a channel bank? |
| 22:35.51 | lenne_dk | Euro |
| 22:35.57 | cpm | JT, well the channel bank T, makes it 2. yes, an actual old-school T1 |
| 22:36.04 | JT | mind you, those people are usually from america ;) |
| 22:36.05 | DonX | I used to have a T-1 when I was a sys admin for a telco |
| 22:36.06 | mercestes | JT: Isn't that just the case? |
| 22:36.10 | danp | i had a T1 to my house |
| 22:36.15 | *** join/#asterisk FUF (i=fuf@fangorn.starshadow.com) |
| 22:36.20 | JT | mercestes: i'm in neither of those parts of the world |
| 22:36.41 | [TK]D-Fender | JT : Keep saying that while annexing plans are in motion :D |
| 22:36.51 | JT | cpm: data or voice? |
| 22:36.57 | cpm | JT, data |
| 22:37.01 | lenne_dk | If the sip-phone doesn't support call-waitning beeps, can * inject them? |
| 22:37.03 | DonX | At one point I was setup with BGP from my ADSL provider and through work |
| 22:37.13 | cpm | it's still the only way to get any kind of bandwidth in my neighborhood |
| 22:37.20 | DonX | those were the good ole' days |
| 22:37.47 | JT | cpm: doesn't sound cheap |
| 22:37.53 | cpm | JT, it ain't. |
| 22:37.59 | cpm | 27 mile loop |
| 22:38.01 | JT | hmm |
| 22:38.07 | robin_sz | JT, it all comes back now ... I installed mISDN using the beronet installer, after building a * box from the xorcom/rapid installer. I built it that way because previous attempts to install bristuffed failed for $randomReason |
| 22:38.14 | DonX | JT: My old T1 was 80$/mo per leg...So 150$ |
| 22:38.21 | DonX | err 160 |
| 22:38.33 | DonX | ouch |
| 22:38.38 | DonX | that mileage will getyou |
| 22:39.13 | cpm | JT, my *only* other option was 'business class dsl' at $210 per 128/256 bump |
| 22:39.19 | JT | robin_sz: well if you get the latest bristuff, bristuff-0.3.0-PRE-1w.tar.gz |
| 22:39.20 | cpm | so, in that case, it's cheap |
| 22:39.40 | lenne_dk | Getting 100MB fibre at work... But let's see where it ends. No good if it's not connected to the internet :-) |
| 22:39.41 | JT | robin_sz: it has a script inside that downloads asterisk, zaptel and libpri, and patches them, and compiles them |
| 22:40.20 | DonX | Texas? |
| 22:40.25 | DonX | ;-) |
| 22:40.27 | JT | robin_sz: i've had a lot of success with this version, mind you i'm using the qozap driver though, not zaphfc |
| 22:40.31 | cpm | Davis, WV |
| 22:40.35 | DonX | ouch |
| 22:40.39 | robin_sz | JT: without damaging my current system? last time I tried that it wanted me to compile as root, which is of course madness |
| 22:40.51 | JT | wow, i don't even know what state WV is :P |
| 22:40.59 | EmleyMoor | West Virginia? |
| 22:41.19 | lenne_dk | WolksVagen? No, that's VW |
| 22:41.19 | cpm | http://en.wikipedia.org/wiki/Davis,_West_Virginia |
| 22:41.52 | JT | robin_sz: damaging? umm ut will overrite any asterisk binaries you have |
| 22:41.57 | DonX | cpm took up all of the bandwidth in that town |
| 22:42.00 | JT | and i don't know if it wants root or not |
| 22:42.04 | DonX | with his t1 |
| 22:42.10 | JT | but it compiles and installs in one step, so probably |
| 22:42.31 | JT | i'm sure you can issue all the dozens of commands manually under different privilidges if you wanted to |
| 22:42.31 | cpm | DonX, I *am* the bandwidth in town, |
| 22:42.39 | wunderkin | i guess thats how you stop kids from viewing porn at school and the library |
| 22:42.52 | DonX | You should be ashamed...Just sitting here on IRC has left your town without any telecommunications...They're all waiting for you to /idle |
| 22:42.53 | DonX | lol |
| 22:42.55 | DonX | cool |
| 22:43.34 | robin_sz | JT, yeah, i think I'll pass on that ... i'd need to test it from a seprate dir before I installed it |
| 22:43.53 | cpm | I went live with my WISP just over 2 years ago, just one month before the cable company started offering their 'broadband' solution. And chopped me off at the knees |
| 22:43.54 | JT | test it? |
| 22:43.59 | DonX | I can see it now...Billy Bob: "What's that? Mama's in the hosptial?" *click* "NOOOOOOOO! FUCKING IRCCCCCCCCC!!!!!!!!!!!!!!!!!!!!" |
| 22:44.10 | cpm | lol! |
| 22:44.13 | DonX | ouch |
| 22:44.15 | robin_sz | yeah, you know, try it and see that it works? |
| 22:44.39 | JT | robin_sz: well it's your choice, but i've found that it works a lot better than misdn |
| 22:44.40 | cpm | I signed with AT&T for 3 years to get my pricing. So, 10 months to go |
| 22:44.55 | JT | robin_sz: if you save the config files, is it that hard to get back to where you were with misdn? |
| 22:45.27 | DonX | A buddy of mine that I used to work with moved into a dark spot of Dallas about 5 years ago so he took on a t1 at 650/mo |
| 22:45.31 | robin_sz | mmm ... I suppose i can run 'make install from the current source |
| 22:46.02 | DonX | I couldn't help but rub in the fact that my company was only paying 150 for mine whenever I saw him |
| 22:46.12 | JT | robin_sz: it's a headache i know |
| 22:46.14 | robin_sz | i guess I can run the download and compile scripts ok |
| 22:46.20 | JT | but bri support generally sucks under asterisk |
| 22:46.23 | DonX | and I still had a DSL line that I was paying out the term on |
| 22:46.49 | robin_sz | JT, yeah .. I mean, who has bri huh? just err, most of europe ;) |
| 22:46.59 | JT | i've tried a few incarnations of both misdn and bristuff, only the latest bristuff hasn't given me trouble (mostly related to the qozap multiport bri driver) |
| 22:47.01 | robin_sz | well, expcet the UK obviously |
| 22:47.04 | DonX | at one point I had a 1.5/768 adsl line with a local rinky dink ISP and then a T1 from my work |
| 22:47.04 | JT | robin_sz: heh yeah |
| 22:47.14 | JT | robin_sz: and australia |
| 22:47.29 | robin_sz | ive always wanted to visit australia |
| 22:47.29 | JT | and most of the first world that isn't north america |
| 22:47.43 | *** join/#asterisk Asterman (n=newkinet@shell4.sea5.speakeasy.net) |
| 22:47.45 | robin_sz | sadly, I dont have a criminal record |
| 22:47.58 | JT | heh, i guess australia might be a nice place to visit |
| 22:48.11 | JT | man, what is it with you brits pre-occupation with that? :P |
| 22:48.33 | JT | stealing a loaf of bread because your family is starving, pretty hardcore |
| 22:48.51 | robin_sz | I dont think that would be sufficient |
| 22:48.56 | JT | it was |
| 22:48.58 | robin_sz | it has to be at least a shep |
| 22:49.01 | robin_sz | sheep |
| 22:49.03 | robin_sz | realy? |
| 22:49.03 | Asterman | question for anyone, I've seen people using DB's to hold their config (like mysql/postgres/etc.) instead of the regular /etc/asterisk text files. Apart from the usual pluses that are gained from using a DB, are there any direct asterisk pros and cons to using a DB to hold config? |
| 22:49.09 | robin_sz | oh ok |
| 22:49.20 | JT | people were sent here for very petty crimes |
| 22:49.26 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 22:49.27 | JT | usually trying to support themselves |
| 22:49.43 | JT | so britain pretty much sent out some of the poorest people |
| 22:50.23 | *** join/#asterisk mavior (n=Miranda@88-149-161-207.f5.ngi.it) |
| 22:51.10 | robin_sz | JT, I think the humour has escaped you ... |
| 22:51.32 | JT | maybe |
| 22:51.41 | JT | we apply the sheep joke to new zealanders |
| 22:51.53 | JT | so when people try to bag us out over it, we just think "huh?" |
| 22:52.22 | *** part/#asterisk Manax (n=toertel-@server.samasher.com) |
| 22:52.24 | robin_sz | JT, old joke .. "englishman goes to Oz, immigration officer says "do you have a criminal record" english bloke replies "no sorrym, I dont, I didn;t reallise it was still a requirement" .. |
| 22:52.40 | JT | heh |
| 22:53.09 | robin_sz | we apply sheep jokes to the welsh, same sort of thing |
| 22:53.26 | EmleyMoor | New Zealanders don't half talk funny |
| 22:53.31 | JT | new zealand has 3 million people, 9 million sheep |
| 22:53.35 | robin_sz | and ive heard the other jokes aussies tell about kiwis |
| 22:53.41 | JT | they talk full funny |
| 22:53.56 | robin_sz | rubbishm, NZ and Aus have the same accent |
| 22:54.04 | robin_sz | we can never tell them apart |
| 22:54.10 | JT | err, they're very different |
| 22:54.13 | EmleyMoor | robin_sz: Really? Sex! |
| 22:54.19 | JT | that's like saying that US and canadian accents are the same |
| 22:54.26 | JT | or texan and new yorker accents are the same |
| 22:54.28 | JT | it's not so |
| 22:54.32 | robin_sz | they are arent they |
| 22:54.38 | robin_sz | hmm |
| 22:54.44 | robin_sz | tex and nyc maybe not |
| 22:55.03 | JT | new zealanders change the sound of the letter "i" compared to australians, most notably |
| 22:55.04 | robin_sz | and canada is part of the US anyway, so that makes sense |
| 22:55.21 | EmleyMoor | Yes, in New Zealand, sex is a number |
| 22:55.22 | JT | they pronounce it closer to an "e" |
| 22:56.11 | wunderkin | sex is six and six is sex? |
| 22:56.23 | robin_sz | in NZ yes |
| 22:56.24 | Hmmhesays | anyone here using asterisk 1.4 app_directory with odbc? |
| 22:56.32 | robin_sz | but theres a reason for that ;) |
| 22:56.45 | JT | Nugget: ask for "fush and chups" that will get them pissed off |
| 22:56.46 | EmleyMoor | You should hear what they call the Queen's husband |
| 22:56.47 | Corydon-w | Hmmhesays: no, but what's wrong? |
| 22:57.16 | CunningPike | ~lart robin_sz |
| 22:57.23 | EmleyMoor | "Prence Phar Lap" |
| 22:57.32 | JT | uneducated australians make fun of new zealanders' accents with the "i" as a "u" which isn't really accurate at all |
| 22:57.38 | JT | as it's closer to an "e" |
| 22:57.41 | JT | but anyway |
| 22:57.52 | robin_sz | there are educated australians too? |
| 22:57.59 | JT | yes |
| 22:58.07 | robin_sz | blimey ;) |
| 22:58.22 | JT | there are non-chav brits, too? |
| 22:58.27 | robin_sz | few |
| 22:58.39 | gordonjcp | oi |
| 22:58.54 | robin_sz | you can't blame me! .. tis you lot that send out the Castlemaine 4X adverts not I |
| 22:58.56 | JT | gordonjcp: how's that? |
| 22:58.57 | gordonjcp | in fact, I'm anti-chav |
| 22:59.05 | JT | heh |
| 22:59.07 | gordonjcp | people become less chav simply by being around me |
| 22:59.07 | EmleyMoor | My dad went to France once and met up with another school party who's coach driver was from New Zealand. Someone asked him what part of Australia he was from... he answered "I'm not from Australia - if I was, I'd have one of those silly hats with corks round it." |
| 22:59.15 | gordonjcp | EmleyMoor: lol |
| 22:59.59 | JT | new zealand almost became the sevent state of australia |
| 23:00.03 | JT | seventh |
| 23:00.09 | Hmmhesays | Corydon-w: it seems my app_directory is not reading from my database it is only looking in voicemail.conf |
| 23:00.10 | JT | it was in the 1901 refurendum |
| 23:00.16 | JT | but was closely defeated |
| 23:00.20 | gordonjcp | hey incidentally, where is the area related to 631-337-xxxx and what is it like? |
| 23:00.30 | gordonjcp | Suffolk County I understand |
| 23:00.33 | robin_sz | JT, im afraid our image of ausstralian culture comes soley from: home and away, skippy, dame edna everidge and the 4X adverts |
| 23:00.42 | gordonjcp | and Rolf Harris |
| 23:00.53 | robin_sz | JT, oh and jenny agutter in walkabout |
| 23:01.01 | JT | all of the home and away shooting locations are within 30km of me :) |
| 23:01.02 | Corydon-w | Hmmhesays: did you define the voicemail option in 'make menuselect' ? |
| 23:01.15 | JT | dunno who jenny is |
| 23:01.17 | Hmmhesays | haha what option is that? |
| 23:01.26 | robin_sz | gordonjcp, tie my kangaroo down sport! |
| 23:01.33 | gordonjcp | wubba wubba wubba wubba |
| 23:01.36 | JT | skippy, that hasn't been on tv in years |
| 23:01.47 | JT | rofl harris |
| 23:01.52 | Corydon-w | Hmmhesays: do you mean odbc storage or realtime? |
| 23:01.54 | robin_sz | JT, what you trying to tell us skip? |
| 23:02.36 | Hmmhesays | my odbc stuff is working for the dialplan and func_odbc, I need app_directory to use it also though |
| 23:02.49 | Corydon-w | Hmmhesays: for storage or realtime? |
| 23:02.51 | JT | robin_sz: it's a pity you only get soap trash and tragic icons |
| 23:02.58 | Hmmhesays | Corydon-w: realtime |
| 23:03.15 | Corydon-w | Hmmhesays: did you define the "voicemail" entry in extconfig.conf ? |
| 23:03.26 | robin_sz | JT, Eh? skippy was my hero! . or rather the kids dad was, I always wanted a house with a helicopter! |
| 23:03.37 | Hmmhesays | Corydon-w: yes I did, and app_voicemail works fine with that, but app_directory doesn't |
| 23:03.41 | JT | i seriously don't remember much of it |
| 23:03.48 | JT | but houses and helicopters are cool |
| 23:04.26 | robin_sz | "tu tu tu .. tu tu" |
| 23:04.31 | Hmmhesays | when I call app_directory and enter the first three digits of the name I have defined it returns "no entries found" |
| 23:04.33 | Corydon-w | Hmmhesays: pastebin an example |
| 23:04.49 | robin_sz | whats that skip, someone has broken their leg in the old creek? |
| 23:04.58 | Hmmhesays | Corydon-w: what do you want to see exactly? |
| 23:05.33 | EmleyMoor | We don't get a lot of NZ TV over here... Going Straight occasionally, and Motorway Patrol, but that's it |
| 23:05.43 | Corydon-w | Hmmhesays: Everything. How it's called from the dialplan, what's in the database, what you're entering |
| 23:05.56 | EmleyMoor | We get quite a lot of Australian, though |
| 23:06.20 | robin_sz | JT, having worked in TV, I met people, including a guy who worked on the skippy series, apparently the only reason the damn animal stayed still often was becasue they used steel pegs and leatehr straps to nail its tail to the floor while they filmed :) |
| 23:06.48 | robin_sz | id like to believe thats true, for the humour value anyway |
| 23:07.16 | JT | lol |
| 23:07.29 | JT | i'm surprised how many law enforcement serieses come out of new zealand |
| 23:07.42 | JT | Motorway Patrol, Border Patrol, Police Ten Seven |
| 23:07.49 | Hmmhesays | http://www.pastebin.ca/319991 |
| 23:07.59 | robin_sz | JT, never heard of any of them, sorry |
| 23:08.04 | JT | heh |
| 23:08.06 | EmleyMoor | Motorway Patrol is the only one of those we get |
| 23:08.20 | JT | police ten seven is only on cable here |
| 23:08.24 | JT | on CI |
| 23:08.48 | Hmmhesays | in my table voicemail I have context mailbox password and fullname defined |
| 23:09.08 | Corydon-w | Hmmhesays: http://pastebin.ca |
| 23:09.20 | Hmmhesays | http://www.pastebin.ca/319991 |
| 23:09.43 | *** join/#asterisk _paulos_ (n=_PauloS@mail.eletrodireto.com.br) |
| 23:09.46 | Hmmhesays | you want to see my db entry? |
| 23:09.50 | JT | nz police always look so laid back and relaxed, but they don't seem to issue much discretion "yeah, i'm just going to have to issue you with an infringement notice" they're always issuing fines, even for very minor stuff |
| 23:09.56 | _paulos_ | Hi people... |
| 23:10.11 | Corydon-w | Hmmhesays: yes |
| 23:10.21 | robin_sz | JT, well, everyone shoudl have a hobby |
| 23:10.25 | _paulos_ | Does unicall work with * 1.4 ??? |
| 23:11.34 | _paulos_ | Or do I have to keep * 1.2 if I use Unicall? |
| 23:11.57 | Hmmhesays | http://www.pastebin.ca/319993 |
| 23:12.06 | Corydon-w | You'd have to ask the developer of unicall |
| 23:12.24 | robin_sz | hmmm .. to compile zaptel, I need the 2.6 kernel source, or just the headers? |
| 23:12.27 | lenne_dk | Where is priv-callerintros supposed to be? |
| 23:12.37 | lenne_dk | <PROTECTED> |
| 23:12.42 | JT | robin_sz: i think some of the best satirical comedy comes out of australia, but then again, you'd expect me to think that :) |
| 23:12.52 | Hmmhesays | everything in the db is ok |
| 23:13.06 | _paulos_ | So let I ask if is there anyone using * 1.4 with MFCR2 signaling... |
| 23:13.21 | Hmmhesays | on the cli shouldn't there be some kind of output stating asterisk is using odbc to try and get the directory entries? |
| 23:13.29 | *** join/#asterisk jmacz (n=jmacz@201.244.173.61) |
| 23:13.52 | robin_sz | JT, well, comedy is a natioanl thing, other nations rarely find it funny. for example ... america. brits just dont get US comedy |
| 23:14.02 | _paulos_ | ~seen coppice |
| 23:14.22 | jbot | coppice <n=chatzill@129.168.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 9h 29m 49s ago, saying: 'the old Dr Who style police phone booth might be more fun :-)'. |
| 23:14.23 | JT | no-one gets us comedy |
| 23:14.23 | JT | except for south park and family guy |
| 23:14.23 | JT | everything else is unfunny |
| 23:14.23 | robin_sz | family guy? |
| 23:14.25 | robin_sz | not heard of that one |
| 23:14.28 | JT | ... |
| 23:14.31 | JT | under a rock? |
| 23:14.40 | karmatronic | JT, i love seinfeld |
| 23:14.46 | robin_sz | nor that one |
| 23:14.52 | JT | so do a lot of australians, i can't stand that shit |
| 23:15.03 | JT | american sitcoms are pretty lame |
| 23:15.05 | JunK-Y | hiya Hmmhesays |
| 23:15.24 | robin_sz | ther was a couple of good ones |
| 23:15.25 | JT | they do have a few good satirical comedys though |
| 23:15.26 | lenne_dk | Too offtopic, signing off to spank gf :-) |
| 23:15.36 | robin_sz | "married with children" .. al bundy! |
| 23:15.46 | JT | american dad is ok, but is more like the poor cousin of family guy |
| 23:15.49 | robin_sz | lenne_dk, post jpegs |
| 23:16.02 | lenne_dk | where? |
| 23:16.13 | Hmmhesays | hey JunK-Y |
| 23:16.15 | Hmmhesays | how are ya? |
| 23:16.19 | robin_sz | flickr? |
| 23:16.20 | lenne_dk | pb? |
| 23:16.23 | mercestes | lenne_dk: To my email address...or photobucket. |
| 23:16.28 | JunK-Y | not too bad, just a bit tired, ya? |
| 23:16.42 | mercestes | lenne_dk: and would a few mpegs hurt you? |
| 23:16.43 | Hmmhesays | Corydon-w: i restarted asterisk and got a different cli output now |
| 23:16.50 | JT | robin_sz: family guy can be downloaded or bought, i'm guessing you don't get much on tv :) |
| 23:16.52 | mercestes | maybe a live webcast. |
| 23:17.01 | robin_sz | JT, and "sledghammer" was funny too |
| 23:17.04 | Corydon-w | Hmmhesays: okay, what'd you get? |
| 23:17.12 | robin_sz | JT, I junked the TCs 8 years ago :) |
| 23:17.16 | robin_sz | TVs |
| 23:17.26 | lenne_dk | Perhaps asterisk can be set up as a video server? |
| 23:17.26 | JT | robin_sz: there's a big following of british comedy in .au |
| 23:17.37 | mercestes | British comedy sucks. |
| 23:17.38 | robin_sz | JT, yeah? probaby ex pats .. |
| 23:17.48 | EmleyMoor | British comedy is very good |
| 23:17.50 | robin_sz | JT, wait, australia IS expats! |
| 23:17.57 | mercestes | It's all high brow, nose in the air, "look at me, I'ms o special, I'm a Brit. ha ha! woo...Worship the Brits." |
| 23:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:18.02 | mercestes | it's so unfunny |
| 23:18.02 | Hmmhesays | http://www.pastebin.ca/319998 |
| 23:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:18.13 | EmleyMoor | You're watching the wrong kind |
| 23:18.15 | JT | robin_sz: if you ignore all the massive other sources of immigration in australia, sure |
| 23:18.20 | robin_sz | mercestes, what? even Pyhton? |
| 23:18.21 | Corydon-w | Hmmhesays: I see the problem |
| 23:18.22 | mercestes | If I hear jerry builder one more time I'll hang myself. |
| 23:18.29 | Corydon-w | Hmmhesays: column hidefromdir=yes |
| 23:18.43 | Hmmhesays | haha great |
| 23:18.44 | _paulos_ | But you need a IQ above 50 to get British comedy |
| 23:18.49 | mercestes | robin_sz: lol. Nah, I'm lying again, trying to get attention. Everyone was bashing US comedy...I was just waiting for an opportunity. |
| 23:18.57 | mercestes | robin_sz: Python is da bomb! :D |
| 23:19.05 | Hmmhesays | Corydon-w: how trivial would it be to make app_dir look up a different table? |
| 23:19.06 | mercestes | .......and US comedy does suck a little.. :( |
| 23:19.12 | robin_sz | a little? |
| 23:19.18 | mercestes | ...not *all* of it sucks. |
| 23:19.23 | Corydon-w | Hmmhesays: just change extconfig.conf |
| 23:19.25 | JT | i think the US is progressing though |
| 23:19.29 | mercestes | just...the TV part. |
| 23:19.29 | JT | the problem was the lack of satire |
| 23:19.40 | robin_sz | they dont get sattire at all in the US |
| 23:19.43 | mercestes | we do have Robin Williams. ..and Chris Dane, ...and Carlos Mencia... |
| 23:19.52 | mercestes | well...we sorta have Carlos Mencia...he lives in the US atleast. |
| 23:20.00 | mercestes | JT: US is too stupid for satire. |
| 23:20.05 | JT | robin_sz: clearly some people do now, with the existence and popularity of south park and family guy |
| 23:20.08 | Hmmhesays | Corydon-w: thanks much |
| 23:20.15 | robin_sz | JT, unless it has a sign saying "joke starts here" they dont reallise its supposed to be funny |
| 23:20.19 | mercestes | ...and irony..... |
| 23:20.35 | Corydon-w | mercestes: Every Chevy Chase movie is satire. Not good satire, but satire nonetheless |
| 23:20.36 | JT | i think that's changing with the younger generation there, imho |
| 23:20.53 | mercestes | Corydon-w: *flat stare* |
| 23:21.09 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 23:21.10 | robin_sz | JT, but south park can be taken on two levels .. one a literal level, which i suspect is how the US see it and a second satirical level |
| 23:21.14 | JT | isn't chevy chase more sitcom than satire |
| 23:21.19 | mercestes | Corydon-w: Even the three amigos? |
| 23:21.20 | *** join/#asterisk ToyMan (n=Stuart@user-12ldv5d.cable.mindspring.com) |
| 23:21.46 | Corydon-w | mercestes: yeah, even that one |
| 23:21.46 | mercestes | wow. |
| 23:21.46 | mercestes | I love that movie. |
| 23:21.49 | JT | robin_sz: you'd have to be pretty thick to be able to not notice a lot of the seething satire in south park |
| 23:21.59 | mercestes | Girl: "We can go for a walk and you can kiss me on the veranda." Chevy Chase: "Lips would be fine." |
| 23:22.05 | _paulos_ | its like comparing US and European comics |
| 23:22.28 | robin_sz | german comedy is still based on Benny Hill |
| 23:23.07 | robin_sz | JT, ummm ... yes, you've not been to the US have you ;) |
| 23:23.17 | JT | robin_sz: clearly not |
| 23:23.52 | JT | if you've seen the moview red dawn, and then don't get the reference in the south park episode grey dawn, then yeah... issues |
| 23:26.39 | Corydon-w | Plus Colbert is pretty much all satire |
| 23:26.45 | mercestes | i'm outie...byes |
| 23:26.50 | Corydon-w | John Stewart, too |
| 23:26.52 | flenders | morning |
| 23:26.55 | JT | who |
| 23:27.11 | JT | hi flenders |
| 23:27.11 | EmleyMoor | John Stewart, to me at least, is that bloke from the Kingston Trio |
| 23:27.14 | Corydon-w | JT: Stephen Colbert does "The Colbert Report" |
| 23:27.20 | JT | never seen it |
| 23:27.31 | FuriousGeorge | has anyone ever had an issue where they will attempt to answer an incoming call (zap) on a sip channel only to find that the call continues to ring on every channel, but you are incapable of answering? |
| 23:27.38 | flenders | JT: g'day mate :D |
| 23:27.44 | Corydon-w | The entire show, he masquerades as a conservative, getting so many things completely wrong that it's funny |
| 23:28.01 | FuriousGeorge | i havent seen that since i started rebooting asterisk daily, but it reared its ugly head again |
| 23:28.03 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:28.06 | JT | ah ok |
| 23:28.06 | flenders | hey, can I do a 'or' statement on a GotoIf()? |
| 23:28.28 | Corydon-w | flenders: $[cond1 | cond2] |
| 23:28.36 | JT | we have shows that have segments making fun of O'Riley's The Factor |
| 23:28.46 | JT | that show is an international embarrassment to the US |
| 23:28.51 | _paulos_ | flenders : $[ (x) | (y )]? |
| 23:29.01 | flenders | Corydon-w: thanks |
| 23:29.05 | flenders | _paulos_: thanks |
| 23:29.12 | FuriousGeorge | rebooting asterisk solves the issue, but its totally unacceptable, and ive now seen it across every version of 1.2.X that ive tried and two different hardware platforms |
| 23:29.32 | *** join/#asterisk topping (n=topping@207.47.6.185.static.nextweb.net) |
| 23:29.34 | Corydon-w | JT: what's funny is that the majority of teenagers get their knowledge of national news from The Daily Report, which isn't a real news show... it's a comedy show |
| 23:29.50 | JT | sure they do |
| 23:29.54 | Corydon-w | Majority of teens who get news from the TV, that is |
| 23:30.22 | JT | has there actually been a study on it? |
| 23:30.53 | Corydon-w | Yep |
| 23:31.17 | Corydon-w | The study made national news when it came out |
| 23:31.21 | *** join/#asterisk Supaplex (n=supaplex@166-70-62-205.ip.xmission.com) |
| 23:31.44 | JT | hmm but there's lies, damn lies and statistics |
| 23:31.58 | JT | fox news, there's another embarrassment |
| 23:31.58 | JT | we get that on cable |
| 23:32.03 | JT | it's like a comedy channel |
| 23:32.15 | Supaplex | and there's JT, he's perfect. |
| 23:32.48 | Corydon-w | http://newsinfo.iu.edu/news/page/normal/4159.html |
| 23:32.55 | JT | Supaplex: you could determine that after only just entering the channel? |
| 23:33.07 | Supaplex | sarcasm |
| 23:33.14 | Supaplex | :-D |
| 23:33.20 | JT | heh |
| 23:33.37 | JT | it's all good (as long as you don't like fox news as a news channel) |
| 23:34.03 | Supaplex | I don't care for most of the media. waste of air time. |
| 23:34.08 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 23:34.09 | JT | true |
| 23:34.18 | Supaplex | unless you have tivo. just hit the hilights. cut the crap. :) |
| 23:34.42 | *** join/#asterisk pavera (n=tomc@c-67-177-60-26.hsd1.ut.comcast.net) |
| 23:34.50 | pavera | hello? |
| 23:34.50 | Supaplex | but yes, I'm generally NOT interested in one view or opinion in those matters. |
| 23:34.57 | JT | news around the world in 20 second |
| 23:34.57 | JT | s |
| 23:34.59 | wunderkin | Corydon-w: you are combining the names of the daily show and cobert report :D |
| 23:35.13 | Corydon-w | wunderkin: eh? |
| 23:35.14 | map7 | is there a way to alert people that there are parked calls, but keep those calls parked whilst doing so? |
| 23:35.17 | Supaplex | pavera: the channel you have reached is now disconnected and no longer in service, please contact your operator for futher assistance. ;) |
| 23:35.30 | wunderkin | they might as well be the same though, jon stewart is a producer of cobert report too |
| 23:35.32 | pavera | thanks :) |
| 23:35.34 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 23:35.54 | *** join/#asterisk Manfish (n=themanfi@82-68-173-121.dsl.in-addr.zen.co.uk) |
| 23:35.55 | pavera | ok, so I've been having this problem for probably 6 months but it has become much worse in the last week |
| 23:36.02 | Supaplex | sounds personal |
| 23:36.04 | flenders | JT: I assume you have foxtel, how crap are the movies on the last few weeks? |
| 23:36.18 | pavera | it has to do with call parking I think, but |
| 23:36.29 | JT | flenders: i was mostly watching it at a friends place, i rarely watch it by myself |
| 23:36.36 | JT | but the moviews didn't seem too good |
| 23:36.38 | flenders | JT: thank god I have a mythtv at home, with all latest releases |
| 23:36.44 | pavera | used to be once or twice a week the system would freeze up |
| 23:36.45 | JT | nice |
| 23:36.49 | pavera | well it wouldn't freeze |
| 23:36.51 | Manfish | any ideas why my MOH jumps to a new track each time i put the same user on hol or a queue announcement is made? |
| 23:36.56 | pavera | existing calls would work |
| 23:37.00 | pavera | and you could place new calls |
| 23:37.12 | pavera | but if you tried to park a call it would just disconnect the call |
| 23:37.44 | *** part/#asterisk romanoffi (n=robg@ip78.131.realtime.net) |
| 23:37.58 | pavera | now I am getting channel.c: avoided initial deadlock warnings |
| 23:38.24 | pavera | and my PRI starts spitting out ring requested on channel 0/1 already in use |
| 23:38.49 | pavera | and the reason I'm coming here is because it is now happening every 30 minutes to 1 hour |
| 23:39.09 | pavera | and because it is stomping on the PRI it won't allow calls in or out |
| 23:39.13 | flenders | is there any dual BRI card out there? |
| 23:39.16 | Supaplex | is AgentCallbackLogin the only means to allow staff to login for receiving calls in one or more queues? |
| 23:39.20 | nick125_lappy | anyone here know why my PAP2 would give me fits over this: |
| 23:39.35 | pavera | no queues |
| 23:39.42 | pavera | just a receptionist who parks calls |
| 23:39.47 | nick125_lappy | I try to dial this number: *585*15050000000*1505XXXXXXX* |
| 23:40.02 | nick125_lappy | it just gives me a busy tone, doesn't even try to send it to the asterisk box |
| 23:40.47 | pavera | any ideas? |
| 23:40.54 | pavera | especially on the error messages |
| 23:41.15 | pavera | they seam pretty generic and I have no clue how to start digging deeper into where/what is causing the problem |
| 23:41.27 | JT | flenders: i think avm might make one |
| 23:41.32 | JT | flenders: not really worth it |
| 23:41.54 | flenders | JT: is that the fritz card? |
| 23:41.57 | JT | dual bri, may aswell buy two hfc cards |
| 23:42.03 | JT | flenders: not the one you showed me |
| 23:42.19 | flenders | 2 cards wouldn't be a problem |
| 23:42.29 | flenders | but I couldn't find one over here |
| 23:43.15 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:43.30 | JT | why buy them over here? |
| 23:43.36 | flenders | dunno |
| 23:43.44 | flenders | couldn |
| 23:43.45 | nick125_lappy | I guess the PAP2 doesn't like the insanely long number I'm dialing |
| 23:44.02 | flenders | couldn't find them overseas either... well, not too sure what to look for |
| 23:44.34 | pavera | ok, well is there a better place to ask for help? I've searched all over on google, I find quite a few people having these error messages but invariably no answer as to what they mean, what might be causing the problem, and no clue where to turn next. |
| 23:45.58 | *** part/#asterisk pavera (n=tomc@c-67-177-60-26.hsd1.ut.comcast.net) |
| 23:48.43 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:50.52 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:51.46 | nick125_lappy | Would someone here be able to help me with my PAP2T-NA? |
| 23:53.41 | flenders | nick125_lappy: mate, I had a similar problem on a WRT54GP2, and it was the dialplan that was wrong |
| 23:53.57 | wunderkin | it has a phone dialplan too |
| 23:54.26 | flenders | I meant the dialplan on the PAP2 |
| 23:54.42 | JT | flenders: you can buy single port isdn cards for as low as EUR10 from germany on ebay |
| 23:55.18 | flenders | JT: wow! |
| 23:55.22 | flenders | that's cheap |
| 23:55.33 | angler | Supaplex, AddQueueMember |
| 23:56.32 | JT | yeah, my only advice would be to make sure they are HFC cologne chipset |
| 23:56.46 | JT | otherwise there's limited software you can work with |
| 23:57.06 | flenders | should I search ISDN BRI? |
| 23:57.12 | JT | just isdn |
| 23:59.22 | FuriousGeorge | o i c... apparently this type of thing is called a deadlock... do you guys think upgrading to 1.4 will lead to less deadlocks? i thought restarting asterisk daily would solve the problem, but it just helps. ive been reading up on voip info about it, in short the advice is to "submit bug report" |