| 00:00.32 | MichealC | Qwell[]: Any help with software these cards will work on.. |
| 00:00.33 | wunderkin | rkeels: normal |
| 00:00.40 | Qwell[] | MichealC: latest zaptel |
| 00:01.02 | MichealC | Qwell[]: cheers, shall try that. |
| 00:01.04 | Qwell[] | latest 1.2 zaptel, that is...I think |
| 00:01.18 | doolph | anyone know how to translate sip error codes? |
| 00:01.22 | Qwell[] | or zaptel from svn branch 1.4 |
| 00:01.49 | bkruse | doolph: sip rfc? usually asterisk will give you some info with the error |
| 00:01.57 | bkruse | then go to the line in chan_sip.c and its usually commented |
| 00:02.03 | bkruse | VERY useful |
| 00:02.18 | doolph | but can I use it for only 1 provider? |
| 00:02.24 | doolph | guess no |
| 00:02.58 | doolph | my main provider is kinda stupid, it sends me 403 instead of 503 |
| 00:03.02 | DrukenLPY | bkruse: you mean the developers actually comment? :) |
| 00:03.38 | bkruse | DrukenLPY: sometimes if your lucky |
| 00:03.57 | bkruse | i have to admit though, i comment on stuff that isnt necessary, when im bored, but when im deep in coding i never comment......kinda of the opposite, but all well |
| 00:04.09 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:04.17 | DrukenLPY | :) |
| 00:04.20 | LeddyHM | so, no suggestions on how to get the correct time on this polycom 501? |
| 00:04.42 | DrukenLPY | i know the feeling, if your bored, you want to comment, if your deep in the code, you just want to get a working model out |
| 00:05.13 | DrukenLPY | one of those fuck it, i'll comment it later |
| 00:05.33 | *** part/#asterisk stubert (i=stu@techtools.actusa.net) |
| 00:06.49 | j0 | anyone else have problems with the gxp2000/asterisk dropping calls when using the mute button? |
| 00:06.59 | j0 | i've even tried the newest beta firmware, same problem |
| 00:07.11 | Qwell[] | j0: turn off VAD |
| 00:07.20 | Qwell[] | /silence suppression |
| 00:07.24 | j0 | Qwell: it's off for sure |
| 00:07.27 | Ac1dcrawl | I'm compiling the zaptel drivers, but chan_zap.so is not being created, any ideas? |
| 00:07.44 | Qwell[] | Ac1dcrawl: re-run configure |
| 00:08.02 | Ac1dcrawl | on zaptel or asterisk? |
| 00:08.05 | Qwell[] | asterisk |
| 00:08.11 | Qwell[] | then run make install again |
| 00:08.22 | j0 | Qwell: when i hit mute, all outbound traffic stops, then give it 5min or so and the call gets dropped |
| 00:08.33 | Qwell[] | j0: yes, because you have VAD on |
| 00:08.42 | Qwell[] | unless the grandstream is just horribly broken |
| 00:08.50 | Qwell[] | (which it is - but I digress) |
| 00:08.57 | j0 | Qwell: says "silence supression" no.. |
| 00:09.14 | Ac1dcrawl | Qwell, still nothing |
| 00:09.22 | Qwell[] | Ac1dcrawl: is it enabled in make menuselect? |
| 00:09.30 | Qwell[] | and are you compiling zaptel 1.4 and asterisk 1.4? |
| 00:09.45 | Ac1dcrawl | hmm |
| 00:09.53 | Ac1dcrawl | let me se |
| 00:09.59 | angryuser | good night everybody |
| 00:12.08 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 00:12.15 | j0 | bloody hell i hate these grandstreams |
| 00:12.41 | j0 | does the polycom 501 have any nifty echo cancellation too? |
| 00:13.24 | Ac1dcrawl | Qwell, yes they are both 1.4 |
| 00:15.23 | blitzrage | evening everyone. I have a bunch of Polycom phones that pull their configuration via HTTP. However, I want to move those phones to pulling their configuration from another server. Is there any options in the .cfg files to tell the Polycom where to pull its configuration from the next time it boots? |
| 00:15.45 | blitzrage | seems silly to have to go to each phone manually and change the server IP |
| 00:15.54 | angryuser | can somebody tell me if i qualify my sip peers later or sooner they becme unreachable? and if i reboot(not stop gracefully) they become reachable for some time?(1-2 hrs) |
| 00:16.11 | ltdwk | j0: use snom =p |
| 00:16.43 | *** join/#asterisk coppice (n=chatzill@67.206.17.210.dyn.pacific.net.hk) |
| 00:16.52 | angryuser | dor got to put "why" ;) |
| 00:17.10 | j0 | ltdwk: why snom instead? |
| 00:17.35 | angryuser | no ideas? i have asterisk 1.41 |
| 00:17.52 | ltdwk | j0: because they're awesome |
| 00:18.07 | j0 | but they're ugly... oooh, web browser |
| 00:18.07 | ltdwk | i use them everywhere |
| 00:18.22 | ltdwk | ugly? who gives a flying frack |
| 00:18.33 | ltdwk | they run linux, they're stable and easy to deploy |
| 00:18.57 | j0 | how fast a processor in them? |
| 00:19.11 | blitzrage | 1GHz |
| 00:19.19 | blitzrage | w/ 512MB of RAM |
| 00:19.26 | j0 | what a deal :> |
| 00:19.29 | angryuser | no no 3GHz watercooled |
| 00:19.44 | ltdwk | liquid nitrogen cooled OC'd to 5 GHz |
| 00:19.47 | JT | quad core :) |
| 00:19.56 | blitzrage | obviously |
| 00:19.58 | JT | liquid helium, beat that |
| 00:19.58 | j0 | hows the speakerphone compared to polycoms? |
| 00:20.08 | blitzrage | liquid hydrogen |
| 00:20.21 | ltdwk | i find the speakerphone ample |
| 00:20.25 | angryuser | Absolute zero, atoms freeze |
| 00:20.25 | JT | ~phones |
| 00:20.26 | jbot | phones is probably http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX |
| 00:20.39 | *** part/#asterisk fmueller (n=user@p548F7327.dip.t-dialin.net) |
| 00:21.20 | ltdwk | that quality/suffestability list is ... evil |
| 00:21.29 | JT | whay? |
| 00:21.30 | JT | why |
| 00:21.38 | JT | i wouldn't touch cisco myself |
| 00:21.45 | JT | but apart from that, seems reasonable |
| 00:21.54 | *** join/#asterisk zpertee (n=chatzill@cpe-65-25-121-5.neo.res.rr.com) |
| 00:22.20 | ltdwk | Personally I rate the snom very high yet it doesnt appear on that list |
| 00:22.20 | tuan_modulis | polycom has higher quility than astra? |
| 00:22.30 | JT | yes |
| 00:22.35 | tuan_modulis | gotcha |
| 00:22.47 | JT | ltdwk: [tk]d-fender rates snom as very low |
| 00:22.51 | blitzrage | jbot: no, phones is probably http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 00:22.52 | jbot | okay, blitzrage |
| 00:22.54 | JT | stability and voice quality |
| 00:22.58 | j0 | ltdwk: do you primarily use the 320 or 360? |
| 00:23.05 | ltdwk | primarily 190's |
| 00:23.09 | Qwell[] | blitzrage: is probably? :D |
| 00:23.14 | ltdwk | the quality is excellent |
| 00:23.15 | Qwell[] | should remove that - he'll add his own |
| 00:23.22 | blitzrage | oh yah :) |
| 00:23.32 | blitzrage | jbot: no, phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 00:23.33 | jbot | blitzrage: okay |
| 00:23.40 | blitzrage | Qwell[]: cp/pst error :) |
| 00:23.43 | Qwell[] | indeed |
| 00:24.07 | JT | ltdwk: seriously, if you think snoms are excellent, you musn't have used better phones |
| 00:24.16 | blitzrage | Snom's are decent phones |
| 00:24.18 | j0 | hmm.. didn't even mention the 190 on their website |
| 00:24.26 | ltdwk | the 190 is an old model |
| 00:24.27 | blitzrage | I just don't like the handset on them |
| 00:24.29 | JT | 190s are olld phones |
| 00:24.43 | blitzrage | same reason I don't liek the Mitel 5220, although it's a pretty decent phone |
| 00:24.51 | ltdwk | At the time we bought them, they were the best you could get. |
| 00:25.02 | JT | maybe in the snom range |
| 00:25.09 | j0 | wish i could my pocket pc phone as a viable voip phone in the office |
| 00:25.26 | ltdwk | Everything I read at the time we bought them rated them at the higher end of hardphones |
| 00:25.28 | j0 | now i have to buy another headset for my desk phone |
| 00:26.59 | ltdwk | I use g.711 codec on them, and i find the quality to be on par if not better than the old analog handsets |
| 00:28.03 | ltdwk | which were customnet/spectrum lines |
| 00:28.37 | *** join/#asterisk krunk- (n=krunk-@unaffiliated/krunk-) |
| 00:30.40 | tuan_modulis | is there anything out there that combines asterisk and network monitoring? |
| 00:30.43 | j0 | so, the aastra 480i or polycom 501. |
| 00:31.25 | tuan_modulis | i'd like a system to phone me whenever something goes down... or at least, someone else |
| 00:31.36 | *** part/#asterisk zpertee (n=chatzill@cpe-65-25-121-5.neo.res.rr.com) |
| 00:32.35 | *** join/#asterisk apardo (n=apardo@87.217.145.59) |
| 00:32.37 | tuan_modulis | if not, it's a pretty nice side project |
| 00:32.39 | JT | ltdwk: well that's not hard, analogue is not very good quality |
| 00:33.08 | j0 | tuan_modulis: why not just get emails on your cell phone? |
| 00:33.16 | ltdwk | JT: I don't really know what you expect for a phone, but personally they provided all the quality that was needed |
| 00:33.25 | tuan_modulis | emails dont really wake u up at night |
| 00:33.46 | tuan_modulis | but that's what we're currently using |
| 00:33.51 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:34.01 | JT | tuan_modulis: look at nagios |
| 00:34.24 | ltdwk | JT: The failure rate out of about 50 phones has been about max 3 phones |
| 00:34.39 | tuan_modulis | nagios, actually, I'm supposed to install that tomorrow as an assignment |
| 00:34.57 | JT | ltdwk: well comparing apples to apples, there's a big difference between ip phones, from utter shit, like grandstream, to high end link polycom and aastra (and questionably cisco) |
| 00:35.18 | ltdwk | JT: what's this "big difference" |
| 00:35.33 | ltdwk | JT: Bear in mind I've never used a grandstream, polycom or aastra |
| 00:35.37 | JT | stability, voice quality, features, support, buold quality |
| 00:35.42 | JT | build |
| 00:35.51 | JT | grandstreams are just rubbish |
| 00:35.57 | JT | you don't want to try them :) |
| 00:35.59 | JT | ~gs |
| 00:36.00 | jbot | hmm... gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 00:36.09 | j0 | heh.. don't use a grandstream.. i just bought one to play with until i decided i wanted to switch everything to voip |
| 00:36.34 | ltdwk | all those things you mention come to mind when I think of the snom handsets I used in that deployment |
| 00:37.13 | j0 | tuan_modulis: i believe there's an agi module or something that can create an outbound call with minimal work |
| 00:37.22 | ltdwk | there's only one thing that irks me about them, they don't do proper automatic DST adjustment |
| 00:40.14 | wunderkin | can someone with a high usage polycom sip 2.1.0 phone send me their config files to use as an example? ive tried following polycom's directions, using manx's phone config as a model (his are 1.6), i've had a few different problems, used to have reboots on ip430s and they were swapped for ip501s and they are ok, but we have a problem with the keys too... on the new and old phones.. |
| 00:41.15 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 00:41.38 | ltdwk | high end eh? :P |
| 00:42.44 | wunderkin | the key problems seem to be cpu related... i have registrations set to 60 sec because of failover, and 30 sec nat pings... not sure what else i have that would be unique that would cause problems |
| 00:43.27 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 00:43.33 | wunderkin | they think the key problems were better on 1.6.7... which did not have the nat ping thing |
| 00:44.49 | wunderkin | and it all seemed to start within the past 2 months, hmm... and i dont think we were doing 60 sec regs to start either since they didnt have failover that that point, but cant remember for how long |
| 00:45.38 | markit | hi, asterisk 1.4.x from svn, was talking (long lasting conversation) with a friend through an ata and mISDN, at a certain moment the ata and other sip phone re-registered, I had the conversation dropped, and at the bottom of the registration messages I got a strange "nice: soxmix: No such file or directory", any clue? |
| 00:47.23 | tuan_modulis | here's a problem... my phone system is going to do credit card transactions, but asterisk makes all sorts of logs (namely the "full" file) |
| 00:47.30 | tuan_modulis | is there a way to disable that? |
| 00:48.19 | tuan_modulis | can't have anyone simply read log |
| 00:48.45 | JT | you have other issues if anyone can "simple read" your sysadmin logs |
| 00:48.50 | JT | simply |
| 00:48.58 | tuan_modulis | well, only i can do it |
| 00:49.00 | tuan_modulis | but still |
| 00:49.03 | tuan_modulis | I'm not supposed to |
| 00:49.22 | tuan_modulis | and we're in beta, so no problem yet |
| 00:49.23 | JT | don't use full log unless you need to then |
| 00:49.56 | tuan_modulis | wait second, full log is only generated when i use asterisk CLI? |
| 00:50.02 | tuan_modulis | wait a second* |
| 00:50.06 | ltdwk | edit logger.conf |
| 00:50.11 | tuan_modulis | icic! |
| 00:50.13 | tuan_modulis | thx |
| 00:50.37 | JT | lots of things add shit to full log, but you can switch it off |
| 00:51.00 | wunderkin | how about this, has anyone ever changed the registration timeout on a polycom? |
| 00:52.03 | wunderkin | that would be sad if a registration used too much horsepower though |
| 00:53.24 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:55.45 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-130-113.vic.bigpond.net.au) |
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| 01:05.14 | tuan_modulis | good night all, thanks for your help |
| 01:05.41 | *** join/#asterisk orkid (i=orkid@bas1-barrie18-1242376399.dsl.bell.ca) |
| 01:05.49 | *** join/#asterisk p4r14h (n=j0sh@69.92.145.178) |
| 01:06.34 | *** join/#asterisk Strom_M (n=strom@65.14.229.26) |
| 01:07.05 | p4r14h | anyone tied in an old panasonic PBX for use with asterisk? |
| 01:07.55 | wunderkin | strommmm |
| 01:08.20 | Strom_M | hi |
| 01:08.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:08.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:09.46 | wunderkin | have you ever changed the registration expiration on a polycom? i'm wondering if that is my problem .. it is set to 60 and i have a 30 sec nat ping thing... they say it is better on 1.6.7 which doesn't have the nat thing and i think the timing may match where we started using 60 sec registration because of failover.. hmm ? |
| 01:10.23 | wunderkin | seems to be a cpu related thing and thats all i can think of :D |
| 01:10.35 | Strom_M | yeah, I think I changed it to 60, but I changed the registration timeout on asterisk, not on the phone |
| 01:10.41 | wunderkin | yeah |
| 01:10.47 | wunderkin | darn |
| 01:14.04 | LeddyHM | can't fix the screen flashing |
| 01:14.13 | LeddyHM | polycom 501 |
| 01:16.10 | blitzrage | need to give it a valid NTP server |
| 01:16.24 | LeddyHM | Yeah I've tried multiple |
| 01:16.39 | wunderkin | the clock is flashing? |
| 01:16.43 | LeddyHM | yes |
| 01:16.51 | wunderkin | aha 12:00 flasher :D |
| 01:17.02 | LeddyHM | 6pm actually |
| 01:17.12 | wunderkin | sometimes it takes a little bit before it goes.. time.nist.gov? firewall? |
| 01:17.20 | blitzrage | what time is it at Billy's house? twelve o'clock! twelve o'clock! twelve o'clock! |
| 01:17.23 | LeddyHM | yeah I used the ip of time.nist.gov |
| 01:17.28 | wunderkin | blitz gets it |
| 01:18.00 | LeddyHM | I'm able to get to nist from other machines on this network |
| 01:18.51 | markit | anyone using debian unstable? I've seen that "soxmix" is no more there (was in sox package) |
| 01:18.53 | wunderkin | make sure you set it to override dhcp for the time stuff too |
| 01:18.55 | LeddyHM | and I am at home connecting to asterisk remotely, so I know it's not any other tcpip issues |
| 01:19.04 | LeddyHM | I set static ip |
| 01:19.43 | wunderkin | or are you getting it from dhcp? :D |
| 01:19.51 | LeddyHM | no |
| 01:20.10 | wunderkin | set the config to override dhcp.. there are a couple of places in sntp |
| 01:20.15 | LeddyHM | wth |
| 01:20.32 | LeddyHM | status network shows wrong sntp addy |
| 01:20.33 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 01:20.41 | wunderkin | yeah i thought so |
| 01:21.16 | wunderkin | Strom_M, is that on 2.x? are you setting a nat timeout too? |
| 01:21.29 | Strom_M | this is on 2.0.3 IIRC |
| 01:21.45 | Strom_M | i dont think i did anything other than change the maxexpirey in sip.cfg |
| 01:21.47 | Strom_M | er |
| 01:21.47 | LeddyHM | where is that set? I can't seem to find it |
| 01:21.50 | Strom_M | sip.conf |
| 01:21.52 | wunderkin | seems silly that it would be a problem but not sure what else to try :D |
| 01:23.00 | wunderkin | i guess that would have been set to no before also with the bare config... ugh |
| 01:24.54 | Ac1dcrawl | ever since I installed asterisk 1.4 I haven't been getting a /var/log/asterisk/full log, is something wrong? |
| 01:25.07 | Qwell | Ac1dcrawl: check logger.conf |
| 01:26.46 | *** join/#asterisk toyowheelin (n=greg@69-10-215-242.rainierconnect.com) |
| 01:28.11 | *** part/#asterisk Dane1 (n=DaneM@adsl-75-25-36-35.dsl.chi2ca.sbcglobal.net) |
| 01:28.53 | LeddyHM | found a setting in sip.cfg, created a new one with time.nist.gov for remote users |
| 01:28.57 | LeddyHM | rebooting noew |
| 01:30.37 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 01:31.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:32.54 | LeddyHM | whoot |
| 01:33.03 | LeddyHM | you guys rock |
| 01:34.44 | wunderkin | i havent been able to reproduce my little buffer overflow thing on the ip430 yet .. only on the 501... but it takes a few times ... |
| 01:36.26 | *** join/#asterisk supjigatr (n=syslod@152.53.17.2) |
| 01:36.54 | *** join/#asterisk mjun007 (n=mjun007@221.221.156.206) |
| 01:37.07 | supjigatr | Anyone seen polycoms in a office just start rebooting randomly during the day. We have several office but it seems to only be in a single office. |
| 01:37.51 | LeddyHM | ours don't exhibit that behaviour |
| 01:38.05 | supjigatr | Its just one office and been going on for about a month. |
| 01:38.06 | [hC] | anyone here using asterisk on a soekris net4801? |
| 01:38.13 | supjigatr | We changed everything out. |
| 01:38.16 | [hC] | specifically, having any audio quality issues with it |
| 01:38.32 | [hC] | using g729 passthru mode |
| 01:39.09 | wunderkin | supjigatr, we've had some problems with ip430s but haven't been able to narrow it down all the way, other than the 501s are ok :P have you checked the power source? |
| 01:40.31 | supjigatr | wunderkin: We acutally bought all new triplites to make sure they had UPS power all the time. Devices were powered but they reboot. |
| 01:40.51 | mcab | supjigatr/wunderkin: anything in the <mac>-app.log files? |
| 01:41.04 | supjigatr | mcab: just that it rebooted and logged a new file. |
| 01:41.08 | supjigatr | No errors |
| 01:41.28 | mcab | supjigatr: using TFTP for bootserver? |
| 01:41.29 | wunderkin | supjigatr, i mean.. are you using a power brick or poe? |
| 01:41.52 | wunderkin | mcab, for my reboots? yes.. assert in dsp, usually |
| 01:42.10 | mcab | heh, that doesn't sound good... |
| 01:42.30 | wunderkin | happened with 1.6.7 up to 2.1.0... we got them swapped for 501s and they are ok, but they are still blaming me and wont swap the other phones |
| 01:43.51 | supjigatr | TFTP |
| 01:43.56 | supjigatr | and we are using bricks |
| 01:44.04 | wunderkin | supjigatr, phone model? |
| 01:44.09 | supjigatr | 500 |
| 01:44.14 | wunderkin | nfi |
| 01:44.26 | supjigatr | Did a version greater than 2.1.0 work? |
| 01:45.09 | wunderkin | my key problem i replicated by switching to the client server instead of mine, the difference.... 60 sec reg instead of 3600.. uh ? .... |
| 01:45.29 | wunderkin | um, there isn't anything released greater than 2.1.0... and that spans quite a long time of firmwares there ;D |
| 01:45.52 | wunderkin | are the phones that f'n slow? |
| 01:46.00 | supjigatr | Yea its a major problem and I bet its in the firmware cause before we upgraded it worked great. |
| 01:46.15 | mcab | supjigatr: you're going to lose any useful logging then, with TFTP :-7 |
| 01:46.16 | wunderkin | supjigatr, what's a problem? i've had many :D |
| 01:46.27 | wunderkin | oh.. |
| 01:46.32 | wunderkin | the rebooting |
| 01:46.52 | supjigatr | mcab: Does FTP do any better or is there a better option? |
| 01:47.48 | mcab | FTP is probably best; you can even configure the LOG_DIRECTORY parameter in the <mac>.cfg file with an ftp:// URL, and the phone will upload there, if you don't want to move all your provisioning to FTP |
| 01:48.05 | mcab | my choice is anything but TFTP :-D |
| 01:48.07 | wunderkin | kjsdakfjsdfkdaf polycom |
| 01:48.57 | supjigatr | Is there a way to changed from TFTP to FTP without going to each phone? |
| 01:49.16 | wunderkin | supjigatr, you're not using dhcp options? |
| 01:51.28 | supjigatr | wunderkin: No we hardset the phones. |
| 01:51.37 | wunderkin | supjigatr, me too, sucks to be us |
| 01:51.40 | wunderkin | stupid router |
| 01:51.43 | supjigatr | haha |
| 01:53.22 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 01:53.22 | Idle | how do I set debugging? |
| 01:53.22 | *** join/#asterisk etfonhome_ (n=Administ@74-140-213-69.dhcp.insightbb.com) |
| 01:53.22 | Idle | on the console, with -r |
| 01:55.41 | wunderkin | set debug? |
| 01:55.51 | Idle | says unknown... :S |
| 01:56.15 | wunderkin | 1.4 is different.. |
| 01:56.38 | JT | core set debug |
| 01:56.50 | Idle | << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] |
| 01:56.52 | Idle | hm |
| 01:58.34 | Idle | [Mar 8 18:36:48] WARNING[4062]: app_voicemail.c:2916 leave_voicemail: No entry in voicemail config file for 'su8521' |
| 01:58.41 | Idle | so, did the syntax of 'voicemail' change? |
| 01:59.13 | wunderkin | yes.. |
| 01:59.14 | ltdwk | su etc now go to the second option of Voicemail() |
| 01:59.27 | ltdwk | Voicemail(8521|su) or similar |
| 01:59.39 | Idle | ah |
| 01:59.39 | toyowheelin | so dose anyone know if I get lingo or vonage can I just setup a asterisk server in my home and point it to one of those accounts presuming that a vonage or lingo account is just sip |
| 02:00.00 | ltdwk | however, the old format still works for me,at least just b and u do |
| 02:01.02 | *** join/#asterisk interworx (n=weechat@203.220.28.66) |
| 02:01.13 | Idle | damnit... how do you reload extensions... |
| 02:01.22 | Qwell | extensions reload |
| 02:01.30 | Idle | nothing |
| 02:01.36 | Idle | its not found |
| 02:01.37 | Idle | cvs |
| 02:02.01 | ltdwk | ^core ? |
| 02:02.07 | Qwell | dialplan reload |
| 02:02.17 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:02.25 | wunderkin | core dump |
| 02:02.44 | wunderkin | :D |
| 02:02.56 | ltdwk | dumping ze core |
| 02:03.42 | toyowheelin | so umm yay or nay |
| 02:03.45 | toyowheelin | :D |
| 02:03.50 | Kumbang | hi guys, does anyone know where can i buy Adit 600 channel bank? |
| 02:03.58 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-57.mtco.com) |
| 02:04.47 | wunderkin | toyowheelin, what is your reasoning for getting lingo or vonage? |
| 02:05.34 | toyowheelin | well it was just a theory type question |
| 02:05.36 | toyowheelin | really |
| 02:08.03 | interworx | anyone else having trouble connecting to ftp.digium.com ? |
| 02:08.15 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 02:13.53 | interworx | I want to register the G729 codecs we just bought but cannot connect to ftp.digium.com :( |
| 02:14.17 | *** join/#asterisk thoughtpolice (n=austin@ip68-98-250-69.lu.dl.cox.net) |
| 02:14.33 | wunderkin | i can... it is ftp2.digium.com though |
| 02:14.43 | interworx | ty i'll try that |
| 02:15.26 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 02:15.34 | wunderkin | we must be overloading the phone with having 60 sec reg and 30 sec nat, but even with nat ping off, it still happens :D |
| 02:15.36 | wunderkin | fennnderrr |
| 02:15.57 | [TK]D-Fender | y0 |
| 02:16.09 | wunderkin | ive been playing around... the key stuff seems to be cpu related and turning off nat and changing reg to 3600 sec it is ok :P |
| 02:16.23 | wunderkin | i need 60 sec reg tho :( |
| 02:17.03 | [TK]D-Fender | wunderkin: Why so frequest? |
| 02:17.08 | wunderkin | because of failover |
| 02:17.12 | [TK]D-Fender | frequent* |
| 02:17.28 | [TK]D-Fender | wunderkin: What are you talking about? They fail plenty already! ;) |
| 02:17.36 | wunderkin | i know!@#!1111111 |
| 02:17.53 | [TK]D-Fender | wunderkin: You're worried about * going down? |
| 02:18.37 | wunderkin | no, their internet connection, 2 offices with PtP T1 going to colo + Asterisk, they have a secondary internet connection at each office... so if the t1 goes down, and it is not a problem at the colo, then they still can access the * box :P |
| 02:19.52 | wunderkin | strom 'thinks' he has 60 sec reg on 2.0.3 but i guess he did not check :D |
| 02:20.01 | [TK]D-Fender | wunderkin: Sounds like you should setup SER and a registration proxy on the inside |
| 02:20.06 | [TK]D-Fender | as a* |
| 02:21.07 | wunderkin | would that require a pc at each office? |
| 02:22.10 | [TK]D-Fender | wunderkin: yup |
| 02:22.18 | wunderkin | then no ;0 lol |
| 02:22.29 | wunderkin | they are already over budget and they still need a hardware echo can card |
| 02:22.35 | [TK]D-Fender | wunderkin: Let them eat cake :) |
| 02:22.40 | wunderkin | ramen? |
| 02:22.50 | wunderkin | ramen cake! |
| 02:22.58 | [TK]D-Fender | No.... Ramen is REALISTIC ;) |
| 02:23.35 | [TK]D-Fender | Hence it loses the historical impact my stab was meant to convey |
| 02:23.53 | wunderkin | i guess so |
| 02:24.11 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 02:24.22 | wunderkin | that would probably help with reinvites locally? but not really necessary right now |
| 02:25.38 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:25.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:25.46 | [TK]D-Fender | <PROTECTED> |
| 02:25.55 | *** join/#asterisk mmartinn- (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 02:26.00 | [TK]D-Fender | wunderkin: It gets really involving at that point. Then again.. they sound anal enough for it |
| 02:26.42 | wunderkin | well, they have plenty of b/w to spare right now so it doesn't really matter :P the offices are so small it won't be necessary.. |
| 02:27.03 | mmartinn- | Does anyone know if there's a nice pre-done list of US local/ld dialplan contexts; I found a link to one at 0xDECAFBAD but it isn't there anymore. |
| 02:27.03 | ltdwk | why do they need hardware ec? |
| 02:27.17 | [TK]D-Fender | wunderkin: Actually what would probably be easier is to have a daemon run on a box there that would do the path test FOR the phones, and on failure "sip-check-config" them with sipsak (based on subnet scan, and force them to rebot. |
| 02:27.38 | [TK]D-Fender | wunderkin: This is the lowest load, dirtiest trick that would probably do the job. |
| 02:28.11 | wunderkin | ltdwk, they have echo that software echo cans wont help, except for hpec which we have a problem with and would only be a temporary fix... |
| 02:28.19 | wunderkin | hmm.. |
| 02:28.23 | [TK]D-Fender | ltdwk: Easy answer might sound like "Because zaptels SUCk loads" in his case? |
| 02:28.33 | wunderkin | h3h3 :D |
| 02:28.42 | ltdwk | must be a big echo |
| 02:28.44 | wunderkin | and polycom? |
| 02:28.46 | wunderkin | hahahahaha |
| 02:28.48 | wunderkin | :D~ |
| 02:28.54 | Nugget | http://www.youtube.com/watch?v=Fow7iUaKrq4 |
| 02:29.19 | wunderkin | ltdwk, yes... unfortunately |
| 02:30.53 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:30.58 | [TK]D-Fender | <wunderkin> :D~ <- drooling idiot smiley? ;) |
| 02:31.38 | wunderkin | z0mg yes! |
| 02:32.28 | *** join/#asterisk apardo (n=apardo@87.217.145.176) |
| 02:32.59 | wunderkin | but again.. a local box... |
| 02:33.19 | [TK]D-Fender | wunderkin: They have no linux on-site? |
| 02:33.29 | wunderkin | linux? what's that? |
| 02:33.48 | [TK]D-Fender | wunderkin: What the WTR54G you'll prep for them will run to do this for you ;) |
| 02:34.11 | wunderkin | :P right |
| 02:34.11 | [TK]D-Fender | wunderkin: $50 fix |
| 02:34.27 | SkramX | anyone using cepstral? |
| 02:34.28 | wunderkin | i'll see.. |
| 02:34.57 | [TK]D-Fender | wunderkin: Now don't say I'm not being creative here with nigh-miraculous economics...... |
| 02:35.04 | wunderkin | hehe |
| 02:35.32 | interworx | d |
| 02:35.32 | ltdwk | i got lucky with my EC, 256ms with MG fixed mine |
| 02:36.19 | wunderkin | mg2 was worse for us than kb1, on 1.2, i have not bothered trying it on 1.4... every call was bad... that does not work for them... tired of hearing them bitch :P |
| 02:36.49 | ltdwk | mark and kb1 both sucked (this was on an e1) |
| 02:36.57 | wunderkin | and this is all a work around for polycoms not handling a 60 second registration timeout? :P~ |
| 02:37.06 | [TK]D-Fender | wunderkin: What you should really be cursing is the inability to throw money at the problem.... |
| 02:37.55 | ltdwk | Get BGP! sounds like better failover |
| 02:38.11 | wunderkin | yeah this has all been a real bitch from the start, the ethernet connection was wrong, duplex mixmatch, m/b incompatabilities, SCREENSAVER on citrix using all of their b/w askfaskdfjakf .. bad phones? |
| 02:41.03 | wunderkin | hey we could use a real dhcp server too! feature that! |
| 02:43.01 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 02:44.09 | *** join/#asterisk normast (n=normast@bas1-toronto01-1177656953.dsl.bell.ca) |
| 02:44.13 | normast | Hi |
| 02:44.39 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 02:51.20 | *** join/#asterisk Fr0zen_ (i=Fr0zen_@unaffiliated/fr0zen/x-000001) |
| 02:52.56 | wunderkin | the buffalo whr-g54s have more memory (than the newer linksys wrt54g) have |
| 02:53.19 | wunderkin | i have the hp model and i've had to reset it sometimes but that may be my own f'n up |
| 02:57.47 | Fr0zen_ | does anyone here run asterisk on freebsd? |
| 02:57.55 | Nugget | I do. I don't recommend it. |
| 03:00.14 | ltdwk | hehe |
| 03:00.36 | ltdwk | do you not recommend it because there are issues relating to threading etc or because the hardware support is bad? |
| 03:00.47 | Nugget | I hate Linux just as much as the next guy, but for Asterisk I think it's the path of least resistance. |
| 03:01.17 | ltdwk | another BSD bigot... they're everywhere! |
| 03:01.19 | Nugget | zaptel on freebsd is poor and even avoiding zaptel you'll always have trouble getting help. people will just shrug and tell you it probably works just fine in linux. |
| 03:01.32 | Fr0zen_ | nugget, why not? |
| 03:01.44 | Fr0zen_ | i dont run any hardware cards |
| 03:01.44 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:01.51 | Fr0zen_ | i connect to my voip providers sip lines |
| 03:02.14 | JT | ltdwk: to be honest he really hasn't acted as a bigot |
| 03:02.19 | Nugget | If you knew asterisk well enough to be happy running it on linux you wouldn't have had to ask that question. :) |
| 03:02.27 | Nugget | er, on freebsd. |
| 03:02.33 | ltdwk | JT: maybe, but he does admit he hates linux =] |
| 03:02.47 | Nugget | choosing to run asterisk on freebsd means you're choosing to blaze your own bumpy trail. |
| 03:03.01 | ltdwk | some people like it rough |
| 03:03.26 | Fr0zen_ | im asking why it's ruff |
| 03:03.29 | Fr0zen_ | im a bsd guy |
| 03:03.36 | Fr0zen_ | i had trixbox istalled for 20 min |
| 03:03.43 | Fr0zen_ | then reformated to freebsd |
| 03:04.04 | ltdwk | How's FleaBSD with latency these days? |
| 03:05.53 | Nugget | my freebsd asterisk box runs ok, but it's not as solid as my linux asterisk boxes. I do have timing and jitter issues that I suspect stem from the "non native" platform. |
| 03:06.04 | Nugget | and I've completely given up on zaptel, which means no meetme() |
| 03:06.19 | Fr0zen_ | im not up for debating linux vs freebsd. I use freebsd, you dont like it, fine, but i do.. |
| 03:06.23 | Nugget | although app_conference will make that less of a headache than it has been historically. |
| 03:06.28 | Nugget | who said I don't like freebsd? |
| 03:06.29 | Fr0zen_ | it's what i started using at first, and it's what i still use. |
| 03:06.32 | Nugget | I fucking love freebsd. |
| 03:06.37 | Fr0zen_ | not you, ltdwk |
| 03:06.40 | Nugget | ah |
| 03:06.45 | Ac1dcrawl | So I upgraded to asterisk 1.4 and my logging is crap, I have everything setup to log to full in the logger.conf. Any idea why I'm not getting detailed logging? |
| 03:06.47 | ltdwk | i'm not debating anything |
| 03:06.56 | Fr0zen_ | i dont hate linux, i just never used it. |
| 03:06.59 | Fr0zen_ | just freebsd |
| 03:07.12 | [TK]D-Fender | Same shit, different smell. get over it. ALL OF YOU |
| 03:07.13 | ltdwk | just asking questions |
| 03:07.23 | JT | Ac1dcrawl: show us the config line |
| 03:07.26 | Fr0zen_ | i already asked it.. you were too busy talking shit about bsd biggots. |
| 03:07.28 | JT | for full |
| 03:07.42 | Ac1dcrawl | full => notice,warning,error,debug,verbose |
| 03:07.46 | Nugget | I use slackware which is the least linuxy linux I've encountered. a nice minimal install of slackware isn't too bad, and I just use it as a bootloader for asterisk. |
| 03:07.46 | Fr0zen_ | anyway, nugget. Why isn't it a good idea to run asterisk on bsd/ |
| 03:07.56 | Nugget | read what I wrote above. I answered you. |
| 03:08.08 | Fr0zen_ | ahh |
| 03:08.15 | Fr0zen_ | damn, i hope I don't have issues. :( |
| 03:08.20 | Fr0zen_ | i might just go back to trixbox. |
| 03:08.21 | JT | Ac1dcrawl: looks okay |
| 03:08.24 | Fr0zen_ | but it's blaoted with a lot i don't need. |
| 03:08.25 | Ac1dcrawl | I have to admit this is a development version from svn, but still, I don't think they changed the logging |
| 03:08.39 | JT | ~trixbox |
| 03:08.45 | jbot | i guess trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 03:09.20 | *** join/#asterisk Stridernzl (n=neville@222-152-248-128.jetstream.xtra.co.nz) |
| 03:09.31 | Fr0zen_ | isn't trixbox a diff flavor of asterisk? |
| 03:09.50 | ltdwk | it's a standardised version |
| 03:09.56 | ltdwk | less fluctuation |
| 03:10.54 | ltdwk | some of what trixbox does is very tied in to specific parts of the codebase they use, so it takes them time to catch up |
| 03:11.11 | JT | it's not standardised |
| 03:11.16 | JT | it's centos + freepbx |
| 03:11.34 | JT | freepbx is asterisk + a horrible gui and a couple of other things |
| 03:11.46 | ltdwk | i was referring to the freepbx part |
| 03:11.50 | JT | which makes poor dialplans which are difficult to understand |
| 03:11.57 | JT | which is why no-one here will support it |
| 03:12.57 | Fr0zen_ | ah ok |
| 03:14.21 | JT | it makes easy things easy, and hard things impossible |
| 03:14.24 | JT | almost |
| 03:14.43 | Fr0zen_ | i see |
| 03:14.46 | ltdwk | If you know your way around asterisk, you're better off not using it |
| 03:14.50 | Fr0zen_ | gotcha |
| 03:15.10 | JT | if you understand text based configuration files, you're smart enough to user normal asterisk |
| 03:15.17 | JT | s/user/use/ |
| 03:15.52 | JT | ~thebook |
| 03:15.54 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:15.54 | Fr0zen_ | ;) |
| 03:15.58 | JT | Fr0zen_: the book is a good starting place |
| 03:16.07 | JT | the wiki is an invaluable reference too |
| 03:16.26 | Fr0zen_ | ya, im not much of a reader. |
| 03:16.32 | Fr0zen_ | I just need asteriskt to work with 1 extension |
| 03:16.50 | ltdwk | That doesn't sound particularly complex |
| 03:17.10 | Fr0zen_ | yea, i had it working on trixbox. |
| 03:17.25 | JT | then use the book as a reference |
| 03:17.27 | Fr0zen_ | shouldn't be impossible with asterisk. |
| 03:17.32 | Fr0zen_ | wiki should be sufficent. |
| 03:18.11 | JT | ok, seems you know best :P |
| 03:19.10 | Fr0zen_ | no, i'm just saying. I don't like books and I only need 1 extension working. I'm not trying to master asterisk or setup large networks so I think the wiki will be sufficent. |
| 03:19.27 | Fr0zen_ | not denying your advice. |
| 03:19.59 | JT | yeah, well the book and wiki can both be taken as a reference only |
| 03:20.12 | JT | no-one is forcing you to read cover to cover ;) |
| 03:22.58 | Fr0zen_ | does asterisk need zaptel? |
| 03:23.05 | Fr0zen_ | i thought zaptel is only for hardware and such |
| 03:23.10 | gambolputty | not nececelery |
| 03:23.32 | gambolputty | or better yet, not neceSaraLee |
| 03:23.50 | ltdwk | it's good to have a zaptel timing source |
| 03:23.56 | JT | hardware, meetme, iax2 trunking |
| 03:23.58 | ltdwk | or is that not needed anymore |
| 03:24.01 | JT | possibly MoH too |
| 03:25.10 | ltdwk | i remember back a while ago I was instructed to use a module that drew timing from the USB controller in my box if I had one |
| 03:25.10 | *** join/#asterisk Rick999 (n=rpulido_@adsl-074-164-111-083.sip.bct.bellsouth.net) |
| 03:25.11 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:25.22 | JT | ztdummy |
| 03:25.31 | ltdwk | ...if I didn't have any zaptel hardware, anyway |
| 03:25.34 | JT | most things don't need it |
| 03:35.34 | Rick999 | anyone: I know this is not the asterisknow channel .. noone is there. Does asterisknow works with tdm400p card? |
| 03:39.55 | fetcher | how is the Avaya 4610SW ? Anyone tried it? |
| 03:44.10 | flenders | fetcher, I saw a demo of it a couple of months ago |
| 03:44.51 | flenders | it was pretty cool, but the demo was with avaya's phone system, so all the buttons/features worked straight out of the box |
| 03:45.29 | ltdwk | i used one very similar at my old job |
| 03:45.58 | ltdwk | the quality was very good and I seldom had any problems with it |
| 03:46.33 | ltdwk | plus it has infrared! technology expose |
| 03:46.51 | ltdwk | s/has/had |
| 03:47.11 | fetcher | flenders: did they say whether those soft-buttons to either side of its LCD all be used for BLF/presence monitoring? |
| 03:49.17 | fetcher | s/all/could all/ |
| 03:49.17 | *** join/#asterisk TedNJ37 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 03:49.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:49.18 | TedNJ37 | Please, help. I have just re-installed my PBX service on another computer, everything works now but we can not hear each other. There is no transmission of sound at all. |
| 03:49.19 | fetcher | TedNJ37: firewall rules blocking RTP? RTP needs to have a large range of UDP ports open |
| 03:49.19 | JerJer | nat or codec |
| 03:49.19 | key2 | If I call from SIP in video to Asterisk, how can I record the video ? |
| 03:50.08 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 03:51.09 | TedNJ37 | The range of ports for RTP is large, from 10000 to 20000 |
| 03:51.21 | TedNJ37 | The ports have been re-directed to the new box. |
| 03:51.41 | TedNJ37 | The strange thing is that I can't hear the other person's voice, they can't hear me either but I can hear the voicemail. |
| 03:56.27 | JacksLivr | my voicemail volume is very low. some people in newsgroups suggest changing the levels for the zap cards, but talking to someone on the phone is fine. How can i make the VM louder? |
| 03:56.32 | JacksLivr | thanks |
| 03:59.56 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:02.06 | ltdwk | you can also set the receive gain |
| 04:02.24 | ltdwk | specifically for voicemail |
| 04:03.20 | ltdwk | ; volgain=0.0 ; Emails bearing the voicemail may arrive in a volume too quiet |
| 04:04.03 | ltdwk | if your problem is present even when the mails are checked via VoiceMailMan(), then yes you'll need to look at your rxgain parameter in zapata.conf |
| 04:05.55 | JacksLivr | it is low when checked on the phone too |
| 04:06.19 | JacksLivr | if i turn up the gain on the zap chann wont it make my volume on conversations louder? |
| 04:06.52 | ltdwk | maybe, maybe not |
| 04:07.13 | ltdwk | in reality yes, but what the user perceives may not change much |
| 04:07.23 | ltdwk | that's been my experience anyway |
| 04:07.54 | JacksLivr | hmmm, thanks |
| 04:08.10 | JacksLivr | just weird how it sounds perfect when talking and low when VM |
| 04:08.14 | TedNJ37 | Can someone help me please? I can't hear the person on the other side of the line, I am funning SIP Phones and the ports are forwarded correctly. |
| 04:08.32 | ltdwk | that'll be because the handset volume turned is up, most likely |
| 04:08.34 | TedNJ37 | The NAT is set to YE |
| 04:08.36 | TedNJ37 | *yes |
| 04:25.33 | *** join/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 04:27.24 | AJaymn | I want to image a complete Asterisk box to a new clean system over the internet.. is that possible? |
| 04:28.02 | Nugget | It's sort of beyond the scope of this channel, AJaymn. |
| 04:28.19 | AJaymn | just wondering if anyones done it.. |
| 04:28.39 | Nugget | Sure, there are plenty of people who have techniques for remotely imaging or installing Linux machines. |
| 04:28.47 | Nugget | doesn't have a thing at all to do with Asterisk, though. |
| 04:29.19 | *** join/#asterisk Strom_M (n=strom@m065e36d0.tmodns.net) |
| 04:29.24 | AJaymn | it was just a question Nugget dont have to be an ass... |
| 04:30.42 | *** part/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 04:40.30 | Nugget | Sorry, I'm cold and sick. |
| 04:40.35 | Nugget | Didn't mean to take it out on you. |
| 04:43.27 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 04:43.49 | genz | anybody know about this ".hpec_x86_32.o.cmd" that the hpec build is looking for? |
| 04:45.20 | *** join/#asterisk eald_home (n=eald@189.157.105.23) |
| 04:46.19 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:48.53 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:49.04 | *** join/#asterisk Swabby- (n=dp@74-137-58-68.dhcp.insightbb.com) |
| 04:49.11 | Swabby- | I got a question for the asterisk xperts |
| 04:49.18 | Swabby- | got 2 phone lines along with 1 fax line.. |
| 04:49.33 | Swabby- | I'm going to have 8 phones..but we're going to use ethernet phones |
| 04:49.39 | Swabby- | what kinda card do i need to get to get it working? |
| 04:49.48 | Swabby- | just a card for the analong phonelines right? |
| 04:50.17 | genz | Swabby: yes |
| 04:51.34 | Swabby- | So i need like at least 3 FXO ports right? |
| 04:52.42 | genz | Swabby: Yes |
| 04:52.57 | Swabby- | gotcha |
| 04:53.03 | *** join/#asterisk daveburr (i=Miranda@146.sub-70-193-86.myvzw.com) |
| 04:53.04 | Swabby- | and i can use ANY ethernet phones i want pretty much huh |
| 04:54.00 | JT | yeah, most sip phones |
| 04:54.03 | JT | ~phones |
| 04:54.10 | jbot | extra, extra, read all about it, phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 04:54.25 | Swabby- | ok.. one last question.. |
| 04:55.12 | Swabby- | are there any alternatives to buying an actual card..like am i going to pay about 300-400 for a card that supports 3 analog lines? |
| 04:55.13 | JT | not for quality |
| 04:55.23 | JT | you can go with an ATA like a linksys, but it may not be the same standard |
| 04:55.56 | *** join/#asterisk znoG (n=gs@OL132-95.fibertel.com.ar) |
| 04:56.22 | *** part/#asterisk daveburr (i=Miranda@146.sub-70-193-86.myvzw.com) |
| 04:57.46 | Swabby- | so like DGM-TDM2401B maybe? |
| 04:58.32 | Qwell | if you only need a couple lines, a 2400 is a bit excessive |
| 04:58.34 | JT | TDM400P with the correct modules would do |
| 04:58.39 | Qwell | might consider looking at the 400 or 800 |
| 04:58.50 | Qwell | depending on whether you plan to grow or not |
| 04:59.46 | Swabby- | probably no growth... |
| 05:00.08 | Swabby- | I can't find TDM400P on voipsupply.com is there another provider that sells it? |
| 05:00.09 | Qwell | 400 would probably be a good choice then, if you only need 3 lines |
| 05:00.17 | Qwell | digium sells direct :p |
| 05:00.19 | Swabby- | nice |
| 05:00.32 | Swabby- | going there now1 |
| 05:00.33 | Qwell | oh, disclaimer - I work for digium |
| 05:01.26 | interworx | how many people blow up their FXO modules? |
| 05:01.46 | interworx | add one to the list :( |
| 05:01.46 | Swabby- | I think it's really cool how digium has partnered with opensource software.. |
| 05:02.07 | Qwell | Swabby-: other way around. we started out with open source software, then we expanded to hardware :) |
| 05:04.26 | Swabby- | nice |
| 05:04.34 | Swabby- | it's still awesome |
| 05:04.53 | genz | *will someday meet somebody who'll talk about HPEC with me |
| 05:07.59 | wunderkin | genz, i didn't have any problems getting it compiled..following the directions |
| 05:08.07 | *** join/#asterisk Strom_M (n=strom@66.0.239.106) |
| 05:08.07 | genz | you're using 1.2 |
| 05:09.24 | wunderkin | it only works on 1.2.13-1.2.15 or 1.4.1 |
| 05:09.53 | genz | wunderkin: i'm using 1.4, 1.4.1 isn't out yet |
| 05:10.05 | Strom_M | it isnt? |
| 05:10.11 | Strom_M | whats that link on asterisk.org then? |
| 05:10.29 | hads | /topic too |
| 05:10.31 | genz | wunderkin: latest svn has merged in the hpec part from 1.2, but its not complete |
| 05:10.39 | wunderkin | ..? |
| 05:10.54 | wunderkin | well digium said it was ok.. but what do the l1 people know :P |
| 05:11.06 | genz | wunderkin: i'd file a bug, but mantis will only let me add a bug for product 1.2.14 |
| 05:11.35 | genz | wunderkin: actually, people in digium say its not working but will soon. i was hoping to catch one of them again. |
| 05:11.40 | wunderkin | thats no reason to stop... but they dont want hardware/pay software on mantis |
| 05:11.45 | *** join/#asterisk entelechy (i=user@mail.beanproducts.com) |
| 05:12.05 | genz | wunderkin: right, but a bug report isn't useful if they're thinking its for the wrong version |
| 05:12.09 | flenders | hey, I have a question about the Page() command |
| 05:12.20 | flenders | where do I add the 'd' option? |
| 05:12.30 | flenders | Page(SIP/08,d)?? |
| 05:12.45 | genz | wunderkin: and btw, their website says 1.4.0 is the newest, but what does their website know:) (http://www.asterisk.org/downloads) |
| 05:13.09 | hads | flenders: what does 'show application page' say? |
| 05:13.52 | flenders | hads: Page(Technology/Resource&Technology2/Resource2[|options]) |
| 05:14.07 | flenders | I never used '|' for the options before |
| 05:14.27 | hads | That's what a comma gets converted to anyway I think. |
| 05:14.31 | flenders | as far as my short memory goes, I always used a ',' |
| 05:14.37 | hads | Either way, they both do the same thing |
| 05:14.42 | Qwell | hads: correct |
| 05:14.44 | flenders | good to knoe |
| 05:14.47 | CunningPike | genz: I think you need to empty your cache |
| 05:14.47 | flenders | know |
| 05:15.19 | genz | CunningPike: for the asterisk downloads page? |
| 05:15.48 | CunningPike | genz: Yup - 1.4.1 is there - in peach and white |
| 05:15.53 | flenders | one other question, when I get a call in, and people "dial 9 for assistance", it Dials all phones (all sip accounts), does Page() with 'd' work the same way? |
| 05:16.02 | CunningPike | Can't look at that page too long |
| 05:16.05 | flenders | do whoever picks up first answers the call? |
| 05:17.00 | Strom_M | CunningPike: give it a rest already :) |
| 05:17.06 | Strom_M | or just use lynx |
| 05:17.37 | genz | CunningPike: Zaptel 1.4? No its not, if it was it'd be in here - http://ftp.digium.com/pub/zaptel/releases/ |
| 05:17.42 | *** join/#asterisk entelechy (n=chatzill@mail.beanproducts.com) |
| 05:18.06 | entelechy | hi |
| 05:19.22 | CunningPike | genz: My bad - I thought you meant asterisk |
| 05:19.47 | genz | CunningPike: Quite alright. That's half the hard part of getting help on this is proving to people I'm not an idiot |
| 05:20.01 | CunningPike | genz: :) |
| 05:20.10 | CunningPike | Strom_M: Hey - how's it going? |
| 05:20.46 | flenders | any ideas? |
| 05:20.54 | flenders | any love? |
| 05:20.58 | flenders | :o) |
| 05:21.04 | Strom_M | hey CunningPike |
| 05:21.06 | Strom_M | it's going well |
| 05:21.13 | Strom_M | just arrived in alabama |
| 05:21.18 | Strom_M | tired as hell |
| 05:21.45 | CunningPike | Strom_M: Good, good - visiting the mother ship? |
| 05:23.19 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:23.20 | Qwell | CunningPike: abduction |
| 05:23.29 | CunningPike | Qwell: Heh |
| 05:24.00 | Strom_M | yeah, ive got to get the digium chip in my head reflashed... |
| 05:24.30 | CunningPike | Strom_M: Make sure you get the new Octastic chip while you're at it |
| 05:24.46 | CunningPike | Strom_M: I noticed some echo when you were here |
| 05:24.51 | Strom_M | hahaah |
| 05:25.10 | CunningPike | :D |
| 05:25.34 | wunderkin | must have been from Strom_C |
| 05:25.52 | CunningPike | strom.c, you mean? |
| 05:26.05 | CunningPike | chan_strom.c :) |
| 05:26.47 | CunningPike | Here till the end of the week, folks, thank you very much! |
| 05:27.38 | Strom_M | hahhaah |
| 05:28.03 | JT | "greetings mr storm" "argh jitter" |
| 05:28.46 | Strom_M | storm? |
| 05:28.48 | Strom_M | gah |
| 05:28.54 | Strom_M | thats jitter :) |
| 05:29.31 | CunningPike | Latency |
| 05:29.45 | Strom_M | there we go :) |
| 05:31.17 | wunderkin | :P |
| 05:31.38 | JT | hehe |
| 05:32.18 | CunningPike | Well, it's after midnight in Space City - no wonder you're tired |
| 05:32.28 | Qwell | that's Rocket City |
| 05:32.44 | Strom_M | and its only half past eleven |
| 05:33.42 | CunningPike | Space, Rocket, whatever :) |
| 05:33.53 | CunningPike | CST? |
| 05:34.01 | CunningPike | Oh - I thought it was EST |
| 05:34.20 | Strom_M | we're only CST here for a short while longer |
| 05:34.36 | Qwell | then it's CDT :p |
| 05:34.36 | CunningPike | Our Exchange calendars are all fscked up at work - apparently versions 1, 2, and 3 of the MS patch are borked |
| 05:34.53 | Strom_M | exactly |
| 05:35.09 | CunningPike | People complain about not having enough ours in the day - well, Microsoft have a solution for that! |
| 05:35.18 | Strom_M | i always cringe when people use "xST" to indicate a physical time zone |
| 05:35.20 | CunningPike | s/ours/hours/ |
| 05:35.28 | Qwell | Strom_M: eh? |
| 05:35.28 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 05:35.30 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 05:35.32 | Strom_M | especially during the summer |
| 05:35.41 | Qwell | oh |
| 05:43.57 | CunningPike | http://support.microsoft.com/kb/933146/ - for a good laugh |
| 05:45.08 | JT | all timezones hould be specified in UTC offsets :D |
| 05:45.11 | JT | should |
| 05:45.43 | CunningPike | JT: I honestly thought that's how MS Exchange did them - I had no idea how fskced up it was |
| 05:45.51 | JT | heh |
| 05:46.52 | CunningPike | :S |
| 05:48.45 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 05:50.39 | Nugget | man that's heinous. |
| 05:51.32 | CunningPike | The consultant that's in our place atm has been working at a huge site for 3 months - they have 16,000 mailboxes, and a script has to be run on every single one of them |
| 05:52.03 | CunningPike | And people keep buying it |
| 05:52.32 | JT | i'd love to be the consultant |
| 05:52.55 | CunningPike | My boss came to me worrying about our Polycom phones - "did them before Christmas" |
| 05:52.56 | CunningPike | Oh |
| 05:53.02 | CunningPike | What about Asterisk? |
| 05:53.20 | CunningPike | Just updated tzdata and copied the new timezone file over |
| 05:53.22 | CunningPike | Oh |
| 05:54.25 | CunningPike | Better get back to that $10K MS Exchange install then........ |
| 05:54.33 | CunningPike | :) |
| 05:55.06 | CunningPike | JT: Me too |
| 05:56.13 | ltdwk | Is there an equivalent of "sip notify" that can be used in dialplans? |
| 05:59.06 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:59.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 06:12.51 | CunningPike | ltdwk: Take a look at SipAddHeader() |
| 06:14.44 | ltdwk | thanks, but i'm not actually making a call though, i don't think that will help |
| 06:20.18 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 06:21.04 | flenders | is there a place I can lookup what all SIP headers do? |
| 06:24.59 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 06:26.06 | russellb | the SIP RFCs :) |
| 06:26.48 | russellb | but find a nice chair, because there is over 2000 pages of them |
| 06:26.48 | russellb | SIP is terrible. |
| 06:26.55 | russellb | "too many cooks in the kitchen" sort of thing, maybe ... |
| 06:29.36 | wunderkin | stupid question but if i require say 128ms for soft echo can to sort of work, does that mean a hardware echo can will need higher? or does it depend upon their mechanism |
| 06:30.13 | *** join/#asterisk Op3r (n=op3r@121.97.193.111) |
| 06:31.12 | russellb | not necessarily ... |
| 06:31.38 | Op3r | whats the name of the feature that u can actually call ur asterisk box and will give u another dialtone? |
| 06:31.40 | wunderkin | ok thats good |
| 06:31.45 | Qwell | disa |
| 06:31.46 | russellb | the tail length should be the same regardless of software/hardware echo can |
| 06:31.52 | russellb | it's just ... how good is it at fixing it |
| 06:32.05 | russellb | the tail length is basically the audio history buffer size ... |
| 06:32.20 | Qwell | is tail length == taps? |
| 06:32.22 | russellb | how much time in the past it remembers |
| 06:32.27 | russellb | yeah, call it what you want |
| 06:32.30 | russellb | i'm being pretty general here |
| 06:32.49 | Qwell | I always hear them interchanged..was never really sure what "taps" were |
| 06:33.07 | wunderkin | hmm k |
| 06:33.36 | Op3r | ? |
| 06:33.41 | Op3r | ahhh |
| 06:33.42 | Op3r | yeah |
| 06:33.43 | Op3r | DISA |
| 06:33.47 | Op3r | i was thinking about dundi |
| 06:33.48 | Op3r | :( |
| 06:34.31 | russellb | though dundi is cool, too :)( |
| 06:34.32 | russellb | dundi >>> disa. |
| 06:34.51 | russellb | eeeep |
| 06:34.56 | file | russellb: bet you didn't expect that! |
| 06:34.57 | russellb | zimbra is making me sad right now |
| 06:35.16 | file | zimbra doesn't make you sad all the time? |
| 06:35.29 | russellb | i generally don't use it directly ... |
| 06:36.52 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:37.55 | russellb | i am ..... cleaning my room |
| 06:38.05 | file | eep |
| 06:38.19 | *** join/#asterisk lorinc (n=ang@pool-7161.adsl.interware.hu) |
| 06:48.00 | *** join/#asterisk cod3hax0r (n=codehaxo@124.6.153.226) |
| 06:48.32 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:50.30 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 06:52.57 | cod3hax0r | how do i route inbound calls from my 1 of my FXO ports to a US #? |
| 06:57.36 | *** join/#asterisk antlers (n=antlers@ip68-224-230-141.lv.lv.cox.net) |
| 07:01.12 | DrCron | um, so a call in automatically connects back out? |
| 07:01.31 | DrCron | cod3hax0r, just throw in a dial command iirc |
| 07:02.02 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:04.35 | DrCron | is there anyone here? |
| 07:05.21 | russellb | nobody but us chickens |
| 07:05.58 | DrCron | do you mind popping into #openbsd and asking if an op would either unban me or msg me regarding the ban? |
| 07:06.43 | russellb | um, yes :) |
| 07:07.11 | russellb | i am not affiliated with openbsd, nor do I intend on getting in the middle of any issues there |
| 07:09.01 | DrCron | after an admin posted a link to a conspiracy site |
| 07:09.18 | DrCron | stupid me |
| 07:11.17 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:13.19 | russellb | politics and IRC don't mix usually :) |
| 07:15.06 | Nugget | That certainly sounds on-topic for a channel about openbsd. |
| 07:16.38 | DrCron | well, i was silly, i agree |
| 07:16.45 | DrCron | i should have stayed quite |
| 07:16.48 | DrCron | quiet even |
| 07:18.44 | DrCron | anyways, if someone would be so kind as to mention that i learned my lesson and will stay quiet, it would be much appreciated(sp?) |
| 07:22.42 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-211-16.socal.res.rr.com) |
| 07:32.25 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 07:32.38 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:33.08 | *** join/#asterisk AfricanSlik (n=slik@196.3.63.252) |
| 07:33.44 | AfricanSlik | hello guys greetings to you all |
| 07:35.20 | mendol | good morning |
| 07:36.11 | AfricanSlik | i am new |
| 07:36.18 | AfricanSlik | here |
| 07:36.41 | mendol | another question from me, i made sip trunk, made very simple context for it and still got SIP/2.0 403 Forbidden |
| 07:37.48 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:38.54 | mendol | SIP/2.0 603 Declined (no dialog) |
| 07:39.19 | *** join/#asterisk unimprtnt (i=bk81@208.98.20.163) |
| 07:43.27 | Fr0zen_ | how do i add another extension? |
| 07:52.57 | *** part/#asterisk unimprtnt (i=bk81@208.98.20.163) |
| 07:53.11 | Kumbang | hello guys, does Atlas 550 work with * ? |
| 07:53.29 | *** join/#asterisk phpboy (n=shane@196.211.17.202) |
| 07:53.46 | *** join/#asterisk rgsteele (n=chatzill@nat-pool.agora-net.com) |
| 07:55.47 | phpboy | hey all I'm trying to get asterisk to store voice recordings in a diff dir but it doesn't seem to want to reflect the changes. savecallsin=/usr/asterisk-monitor in agents.conf |
| 07:59.09 | SuperLag | DrCron: message the op that banned you |
| 08:12.04 | *** join/#asterisk Assid (n=assid@59.183.74.245) |
| 08:12.09 | Assid | heya |
| 08:12.27 | russellb | SuperLag: there was definitely some lag in that response. |
| 08:12.27 | Assid | since asterisk has 2 trees which would be preferred 1.2 or 1.4 now ? |
| 08:12.45 | russellb | Assid: it depends |
| 08:13.11 | russellb | choose between something that has existed for over a year so has more proven stability |
| 08:13.18 | russellb | or something with a lot more features :) |
| 08:13.29 | sevard | features smeatures |
| 08:14.06 | russellb | or if you want to be really evil, use the 1.0 branch |
| 08:14.24 | sevard | s/evil/cool |
| 08:14.44 | russellb | it is so cool that it's not even maintained anymore |
| 08:14.45 | Assid | well the main reason i am upgrading is cause im getting dtmf issues recwntly with providers |
| 08:14.53 | Assid | mainly voicepulse |
| 08:15.13 | russellb | iax2 or sip? |
| 08:16.01 | DrCron | cod3hax0r wants to forward incoming calls on one interface to another number through a viop provider. thats a fairly simple thing to do right? |
| 08:16.18 | cod3hax0r | thanks drcron |
| 08:16.22 | cod3hax0r | i figured it out |
| 08:16.29 | cod3hax0r | thanks to the DISA module of trixbox |
| 08:16.39 | russellb | of trixbox? |
| 08:16.40 | *** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it) |
| 08:16.41 | sevard | speaking of "forwarding" there isn't another method to forward calls other than to use minutes, is there? |
| 08:16.47 | russellb | the disa module has nothing to do with trixbox |
| 08:17.35 | sevard | if a call comes into my voip did and i dial() my cellphone, one is still using voip minutes, right? |
| 08:17.37 | DrCron | and if all you want to do is forward you dont want to use disa |
| 08:18.15 | DrCron | if you dont need the ability to dial other numbers, disa is a security risk (iirc) |
| 08:19.10 | Assid | russellb : sip |
| 08:19.34 | russellb | disa is only a security risk if you configure it in a bad way |
| 08:19.58 | Assid | russellb: i have dtmfmode=rfc2833 |
| 08:19.58 | russellb | in the same way that a poorly configured dialplan is a security risk |
| 08:19.58 | russellb | you just don't want to give the world access to make outbound calls through your pbx |
| 08:20.27 | russellb | then you'll get a bill for 100 thousand minutes one month |
| 08:20.39 | russellb | Assid: are the problems with dtmf incoming to your box or outgoing to the provider? |
| 08:20.50 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 08:21.11 | DrCron | russellb, but if you can avoid using disa without too much pain you should, right? |
| 08:21.11 | Assid | incoming |
| 08:22.18 | Assid | i am trying to exclude all possibilities. i do have a ticket open with vp, and they are looking at it from their end. But i want to do whats possible on my end to make sure its working as well |
| 08:22.24 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 08:23.34 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 08:24.50 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 08:26.36 | Assid | russellb? |
| 08:26.49 | russellb | DrCron: it's perfectly fine to use it |
| 08:26.49 | russellb | you just have to make sure that if DISA is publicly accessible in any way, that people can't make outbound calls that cost you money |
| 08:26.49 | russellb | Assid: the first thing to do is to capture an "rtp debug" of the problem occuring, so that you can see exactly what it is |
| 08:26.49 | russellb | whether you're missing some events ... or they're in the wrong order ... |
| 08:26.50 | russellb | however, we have made significant DTMF handling improvements in 1.4, so it is probably worth trying. |
| 08:26.51 | russellb | i know this sounds geeky ... but DTMF in 1.4 is pretty cool |
| 08:27.38 | russellb | if you have a call between a SIP phone using rfc2833, and a zap FXO channel ... you can hold down a button on your SIP phone and the length of the digit as you hold it down is preserved through asterisk and will be heard on the zap side |
| 08:27.57 | russellb | before, asterisk had no concept or care about the length of a digit ... |
| 08:28.52 | russellb | Assid: i may have missed messages from you if you said anything ... my net is in and out |
| 08:29.19 | Assid | actually i did try calling in back into the asterisk box.. even my own call wasnt recognised properly by dtmf. i dialled 201 but it didnt reach i think went as 21 (console wasnt on that moment) |
| 08:29.40 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 08:29.57 | Assid | it happens specially when the numbers are pressed rather quickly |
| 08:29.58 | russellb | well try it with rtp debug on |
| 08:30.26 | Assid | you still think i should move to 1.4 ? |
| 08:30.34 | Assid | i am using 1.2.10 |
| 08:32.05 | AfricanSlik | hey guys i am trying to install asterisk on my server |
| 08:32.14 | AfricanSlik | i am trying this cvs command |
| 08:32.18 | AfricanSlik | not working |
| 08:32.55 | AfricanSlik | -bash: cvs: command not found |
| 08:33.35 | russellb | I'm going to have to get some sleep soon ... |
| 08:33.40 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 08:33.50 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-211-16.socal.res.rr.com) |
| 08:33.52 | russellb | but I'm on all day while I'm at work if you would like me to glance at some rtp debug for you |
| 08:34.58 | Assid | russellb: so far havent got a chance for it to dial wrong extension |
| 08:35.12 | Assid | as of this second its working correctly |
| 08:36.01 | russellb | of course it works correctly when you actually want it to fail :) |
| 08:36.35 | Assid | cant get it to fail @!!# :( |
| 08:36.38 | Assid | wtf |
| 08:36.46 | Assid | should i just load 1.4 ? |
| 08:36.56 | Assid | and maybe move to inband instead of rc2833 ? |
| 08:37.39 | russellb | it's up to you. |
| 08:38.49 | Assid | what would you do if you were me |
| 08:39.01 | Assid | these guys complain ervery day 3 times a day to do sometghing about it |
| 08:39.14 | gfraysse | <PROTECTED> |
| 08:39.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:41.48 | russellb | if I were you, I would start by isolating the cause of the problem. |
| 08:42.02 | russellb | which would be to catch rtp debug of the problem occurring ... |
| 08:42.18 | russellb | if the digits aren't even making it to your box, then asterisk can't do anything about it |
| 08:42.48 | Assid | im gonna try upgrading.. and then if it still dont work.. time to isolate the problem |
| 08:42.56 | Assid | already doing that |
| 08:43.02 | Assid | gmme a few mins |
| 08:43.06 | Assid | will try again |
| 08:43.17 | russellb | sounds good |
| 08:43.32 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 08:46.48 | russellb | i don't know how big this setup is ... but in general, you should read over UPGRADE.txt before switching to 1.4 |
| 08:46.48 | russellb | that will tell you about all known and intentional changes in behavior and such |
| 08:48.03 | phpboy | hey all I'm trying to get asterisk to store voice recordings in a diff dir but it doesn't seem to want to reflect the changes. savecallsin=/usr/asterisk-monitor in agents.conf |
| 08:48.28 | Assid | will want to turn on the rfc2833compensate option. Without this option your DTMF reception may act poorly. |
| 08:48.37 | Assid | rfc2833compensate? |
| 08:49.06 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 08:49.24 | *** join/#asterisk amit1 (n=aa@202.79.37.177) |
| 08:49.48 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 08:50.44 | badcfe | is the termcap-compat the one to install when i get "termcap support not found" for an 1.4.1 build |
| 08:51.26 | Assid | hehe.. i like the new ascii asterisk at the end |
| 08:51.58 | Assid | do i have to get rid of /usr/lib/asterisk .. or something? |
| 08:52.18 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 08:57.44 | russellb | badcfe: i would install ncurses-dev |
| 08:57.54 | russellb | Assid: i made that ascii art :-p |
| 08:57.54 | badcfe | russellb: thank you |
| 08:58.05 | russellb | Assid: and yeah, rm -f /usr/lib/asterisk/modules/* |
| 08:59.15 | Assid | [Mar 9 03:58:56] WARNING[6020]: rtp.c:883 ast_rtcp_read: RTCP Read too short |
| 08:59.24 | Assid | russellb: nice |
| 09:00.31 | russellb | Assid: you can ... probably ignore that |
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| 09:02.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:03.30 | Assid | okay tried 3 calls.. every call since i added rfc2833compensate option.. just either doesnt answer to my calls or answers very late |
| 09:04.41 | Assid | hey russellb: you wanna make 2 test calls for me ? |
| 09:04.55 | Assid | rather can you ? |
| 09:05.04 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 09:05.14 | russellb | i don't really have a way to make any calls at the moment ... |
| 09:06.37 | Assid | okay another thing i am noticing thats happening is the ivr starts cutting (audio cutting).. and everytime that happens.. i see more of these WARNING[6121]: rtp.c:883 ast_rtcp_read: RTCP Read too short |
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| 09:08.56 | russellb | maybe it is incorrectly thinking valid rtp is rtcp ... |
| 09:09.07 | russellb | i'm out for the night ... good luck for now .. |
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| 09:18.24 | phpboy | hey all I'm trying to get asterisk to store voice recordings in a diff dir but it doesn't seem to want to reflect the changes. savecallsin=/usr/asterisk-monitor in agents.conf |
| 09:24.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:25.09 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-16638.otenet.gr) |
| 09:25.15 | nasls_lsa | hello people ! |
| 09:26.46 | nasls_lsa | I put : register => user:passwd@sip.i-call.gr/800 and then after reload I do sip show registry , and appears registed .. now , how do I config my extensions.conf to make calls through that ? ! |
| 09:26.53 | *** join/#asterisk Aurs (n=Aurs@c6870BF51.dhcp.bluecom.no) |
| 09:27.10 | creativx | nasls_lsa: try the wiki |
| 09:27.19 | nasls_lsa | yea yea .. I tried |
| 09:27.34 | nasls_lsa | after 2 days experiments I as here :) |
| 09:28.32 | creativx | hehe |
| 09:28.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:28.47 | creativx | set up an extension that matches your sip number |
| 09:28.54 | creativx | in the correct extension |
| 09:29.43 | nasls_lsa | may I do a second question too ... I did the resister => .... |
| 09:30.01 | nasls_lsa | is that the only thing that I need in sip.conf to register in a SIP provider ? |
| 09:30.23 | nasls_lsa | or I have to write too [icall] type=peer ..... ? |
| 09:30.33 | *** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 09:30.50 | nasls_lsa | I do : exten => _9.,1,Dial(SIP/800/${EXTEN:1}) |
| 09:30.55 | nasls_lsa | in my extensions.conf gile ? |
| 09:33.46 | creativx | damnit being hungover really helps understanding the manager interface output. |
| 09:37.27 | mendol | "Dial failed due to CHANUNAVAIL" |
| 09:37.38 | mendol | anybody help please :-/ |
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| 10:20.49 | phpboy | hey all I'm trying to get asterisk to store voice recordings in a diff dir but it doesn't seem to want to reflect the changes. savecallsin=/usr/asterisk-monitor in agents.conf |
| 10:23.35 | AfricanSlik | people in here are sleeping |
| 10:43.24 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:45.32 | tzafrir_laptop | phpboy, not sure what the real bug is, but if you can't find a proper solution, symlink your way around it |
| 10:46.00 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 10:46.21 | fourcheeze | hi, anyone using voispeed with asterisk? |
| 10:46.44 | fourcheeze | I have a customer who wants to connect to our asterisk with it so that we can provide incoming calls |
| 10:47.09 | *** join/#asterisk mgbowman (n=xyklopzi@86.122.8.28) |
| 10:47.26 | mgbowman | has anyone here managed to get app_rxfax or app_txfax to compile? |
| 10:47.30 | fourcheeze | he wants N incoming numbers to go to 1 sip account - what's the best way for us to mark them so that he can tell them apart? |
| 10:47.49 | mgbowman | i am at my wits end with this one |
| 10:48.37 | mgbowman | Yeah I know it's a hairy subject |
| 10:48.48 | mgbowman | i'm thinking of just paying for a service |
| 10:49.16 | mgbowman | anyone? |
| 10:51.09 | fourcheeze | is there a generic solution for identifying incoming lines on sip devices? |
| 10:51.28 | fourcheeze | currently when I do this with a sip phone I kludge the callerid |
| 10:57.24 | mendol | i need quick help, how can I add another SIP/xxx to exten => s,1,Dial(SIP/yyyy) |
| 10:58.33 | mgbowman | exten => s,1,Dial(SIP/yyy&SIP/xxx) |
| 10:58.43 | mgbowman | that will ring both xxx & yyy until one answers |
| 10:58.48 | mendol | ahh |
| 10:58.50 | mgbowman | which ever one answers first |
| 10:58.51 | mendol | great thats my answer |
| 10:58.54 | mgbowman | "wins" |
| 10:58.59 | mendol | thanks a lot mg :-) |
| 10:59.03 | *** join/#asterisk Inez (i=faceoff@devel4.net) |
| 11:00.16 | mgbowman | no problem |
| 11:00.20 | mgbowman | now help me :-) |
| 11:00.29 | mendol | hehe wish i could |
| 11:01.41 | mgbowman | think I might try OpenPBX |
| 11:01.46 | mgbowman | i really need this support :( |
| 11:06.45 | mendol | yeh i got same problem |
| 11:06.51 | mendol | i fixed incoming calls with sip trunk |
| 11:07.04 | mendol | but cant make outbound calls |
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| 11:19.58 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
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| 11:37.12 | kippi | I am getting this error, I have a green light on the card |
| 11:37.14 | kippi | <PROTECTED> |
| 11:39.50 | kippi | anyideas how to stop this? |
| 11:42.20 | vlrk | does astersik-1.4.1 need autoconf greater than 2.60 ? |
| 11:44.50 | *** join/#asterisk solar_ant (n=solar@122.164.136.128) |
| 11:46.52 | mendol | hm for incoming calls type=peer? |
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| 11:51.46 | vlrk | if i do a txfax with asterisk new version will that go with a rtp transmission or will that choose t38 ? as for as i know asterisk does not support t38 am i right |
| 11:51.52 | vlrk | asterisk-1.4.1 ( i mean new version ) |
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| 12:06.18 | r0d3nt | <SecNews> Title: Vonage loses phone patent dispute |
| 12:06.18 | r0d3nt | <SecNews> Link: http://news.bbc.co.uk/go/rss/-/2/hi/business/6433525.stm |
| 12:06.18 | r0d3nt | <SecNews> Description: Internet phone company Vonage loses a patent case that could threaten its business. |
| 12:06.30 | r0d3nt | i think we're fucked. |
| 12:09.25 | coppice | those verizon patents are pretty broad |
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| 12:19.27 | mendol | what is "s-CHANUNAVAIL|1 error? |
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| 12:54.26 | *** join/#asterisk Nirs (n=Nirs@p11811120.orange.net.il) |
| 12:54.54 | Nirs | Hi all, anybody home? |
| 12:55.21 | Nirs | How geeky cam u get, i'm ircing on a nokia e61 |
| 12:56.37 | *** part/#asterisk Nirs (n=Nirs@p11811120.orange.net.il) |
| 12:58.09 | *** join/#asterisk Nirs (n=Nirs@p11811120.orange.net.il) |
| 12:59.42 | penguinFunk | damn |
| 12:59.48 | penguinFunk | i tried ircing on my nokia e61 |
| 12:59.49 | Nirs | Hello? |
| 12:59.54 | penguinFunk | what client you using ? |
| 13:00.08 | Nirs | Jmircj |
| 13:00.11 | penguinFunk | mine wouldnt connect |
| 13:00.22 | penguinFunk | think the mobile network restrict irc |
| 13:00.26 | penguinFunk | what network you with ? |
| 13:00.29 | penguinFunk | im with 3 |
| 13:00.31 | Nirs | Well it wasn't easy but it works |
| 13:00.55 | Nirs | 'm with orange' but i'm in israel |
| 13:01.21 | *** join/#asterisk active_si (n=AcTiVe@84-255-238-137.static.dsl.t-2.net) |
| 13:01.51 | Nirs | i think this is the ultimate geeky thing i've ever done |
| 13:02.21 | *** join/#asterisk marc\cba (n=marc@cpc1-whit2-0-0-cust972.cdif.cable.ntl.com) |
| 13:03.09 | Nirs | I sitting next to the pool, in a hotel, my wife is next to me reading a book, and i'm ircing on my mobile - i think i've acheived uber-geekness |
| 13:03.16 | *** join/#asterisk step_quasar (n=step_qua@191-91-235-201.fibertel.com.ar) |
| 13:03.24 | step_quasar | Hi, need the package asterisk-1.4 for debian, exists? somebody knows where to obtain it? |
| 13:03.36 | marc\cba | Nirs- vacation or work? |
| 13:03.49 | Nirs | Vacation |
| 13:03.55 | marc\cba | nice |
| 13:03.56 | coppice | Nirs: pool, hotel and wife score negative geek points |
| 13:03.57 | *** join/#asterisk str_ (n=str@251.9.39-62.rev.gaoland.net) |
| 13:04.13 | str_ | hi folks. Do you have a pointer about this security vulnerability fixed in latest asterisk release ? |
| 13:04.51 | Nirs | Thanks copp, i feel better now |
| 13:05.41 | DrukenLPY | r0d3nt: yeah... i'd say at least vonage is fucked.... |
| 13:06.44 | coppice | vonage is just the first phase |
| 13:07.11 | DrukenLPY | i want to know what "patents" they broke.... |
| 13:07.48 | coppice | someone listed them on slashdot (yeah, amazing, actual info on slashdot :-) ) |
| 13:07.58 | step_quasar | Hi, I need the package asterisk-1.4 for debian, exists? somebody knows where to obtain it ? |
| 13:08.30 | DrukenLPY | woah.. wait a min... slashdot and information? that's just wrong..... |
| 13:10.51 | Nirs | Say, anyine using a2billing? |
| 13:15.28 | Nirs | Ok, diffrerent question. I need a normal ssh for the e61, any suggestions? |
| 13:18.02 | *** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar) |
| 13:18.09 | AlfaScorpii | morning people! |
| 13:18.55 | AlfaScorpii | does any one know what is the best sip softphone for linux ? im using Ekiga |
| 13:19.50 | step_quasar | twinkle |
| 13:19.57 | step_quasar | softpohone twinkle |
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| 13:21.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:22.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:22.42 | step_quasar | Hi, need the package asterisk-1.4 for debian, exists? somebody knows where to obtain it? |
| 13:22.56 | e-ddie | google |
| 13:23.06 | *** part/#asterisk inspired (n=mikael@85.221.7.59) |
| 13:23.11 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:26.34 | Nirs | ok, this is now too geeky, even for a married guy. Ircing and sshing on the mobile while on vacation. |
| 13:27.38 | macTijn | no it isn't |
| 13:28.12 | macTijn | although the hotel where I was had linux workstations, so I didn't really need it |
| 13:28.47 | coppice | Nirs: if you relate to this http://www.xkcd.com/c230.html you have serious geeky problems (personally, I think he's confused, because he should know a Hamiltonian cycle is just a special instance of a Hamiltonian path) |
| 13:28.48 | AlfaScorpii | step_quasar: tks |
| 13:29.50 | AlfaScorpii | nop |
| 13:30.05 | AlfaScorpii | i need something like eyebeam, but for Linux |
| 13:30.16 | AlfaScorpii | something that looks like a real phone |
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| 13:41.44 | step_quasar | AlfaScorpii :como va ? |
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| 14:13.48 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 14:14.40 | badcfe | do one need a g729 license in order to use the generic jitterbuffer on sip to sip? |
| 14:23.43 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:24.11 | blitzrage | morning all! What is the command at the CLI to show jitterbuffer stats. I think it is netstat, but I can't seem to find what it's under |
| 14:27.11 | [TK]D-Fender | blitzrage: "iax2 show netstats" |
| 14:27.16 | *** join/#asterisk TaiSHi (n=TaiSHi@zion.dattaweb.com) |
| 14:27.36 | TaiSHi | Hell everyone, I register with my SIP provider, but I dont seem to receive any call |
| 14:27.39 | blitzrage | [TK]D-Fender: hrmmm... that seems like it'd be wrong for 1.4 where jb isn't only under IAX2 |
| 14:27.46 | TaiSHi | I alsodont receive messages in * console |
| 14:30.15 | [TK]D-Fender | blitzrage: Thats all I can find... |
| 14:30.21 | new2345 | hello |
| 14:30.26 | blitzrage | [TK]D-Fender: me too -- I think I just found a bug |
| 14:31.14 | [TK]D-Fender | blitzrage: Hey, since you're here, is there a new release of THE BOOK available somewhere? I heard something about a 2007 revision but see no mention or proof of it on astriskdocs |
| 14:31.42 | blitzrage | [TK]D-Fender: we're still writing it :) |
| 14:32.01 | blitzrage | [TK]D-Fender: http://www.oreilly.com/catalog/covers/9780596514051_lrg.jpg |
| 14:32.39 | [TK]D-Fender | blitzrage: Beyond the typical kind of answeer for this question, do you have any personal expectations for when it would likely be released? |
| 14:33.07 | creativx | sweet book |
| 14:33.39 | blitzrage | [TK]D-Fender: well... we were supposed to have the rough draft done for March 1st, but there is sooooo much stuff we want to get into the book, so I'm thinking you'll see an online copy around May/June, printed copy around July/August |
| 14:34.13 | [TK]D-Fender | blitzrage: Ok, exact kind of range I was hoping for :) thanks |
| 14:34.48 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 14:35.07 | blitzrage | If anything, I want to have printed copies by AstriCon :) |
| 14:35.32 | markit | hi, yesterday during a conversation all my sip ata/phones re-registered against asterisk, and, of course, the conversation dropped... is it normal? something I have to set in sip.conf to prevent this? * 1.4.x svn |
| 14:35.42 | file | blitzrage: yay |
| 14:35.51 | blitzrage | [TK]D-Fender: if you talk to O'Reilly, I think it is possible to buy a copy now, and read the book as it's being written (PDFs are generated nightly from SVN) |
| 14:36.14 | blitzrage | markit: SIP != RTP... so re-registration shouldn't break that... |
| 14:36.52 | blitzrage | [TK]D-Fender: I don't necessarily mean you should buy a copy -- I'm just saying I think that's how their working it :) |
| 14:37.38 | markit | blitzrage: well, CLI yous showed me the registrations of sip phones/ata when the conversation broke... do you want the pastebin of the screen at that moment? |
| 14:37.57 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:38.19 | blitzrage | markit: I don't, but maybe someone in here does |
| 14:38.22 | blitzrage | <-- buuuuusy |
| 14:38.39 | creativx | tis friday |
| 14:38.41 | creativx | tis weekend!! |
| 14:39.02 | blitzrage | when you're a consultant there is no such thing as weekends |
| 14:39.19 | blitzrage | no such thing as days of the week, you simply work on the premise of yesterday, today, tomorrow, and next week |
| 14:39.29 | creativx | well |
| 14:39.35 | creativx | for me its monday, monday, monday, monday, friday, saturday, sunday |
| 14:39.43 | blitzrage | heh |
| 14:39.52 | creativx | although there might be the occasional wednesday and thursday if it involves beer |
| 14:39.56 | markit | http://www.pastebin.ca/387637 |
| 14:40.32 | badcfe | when i did noop(${ACCOUNTCODE} D=${DNID} C=${CALLERID} T=${TIMESTAMP}) in * 1.2 i saw the data, in 1.4 its all blank .. what should i change it to? |
| 14:40.37 | markit | or could be a mISDN issue and I youst did not watch CLI until conversation dropped... |
| 14:41.56 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 14:42.34 | TaiSHi | Hell everyone, I register with my SIP provider, but I dont seem to receive any call (also, I don't see messages in console) |
| 14:42.35 | supjigatr | Anyone have the option lines that work with polycom, ftp, and linux dhcpd handy. Google gives lots of examples. |
| 14:42.37 | JunK-Y | badcfe: read the variables doc, they were deprecated since 1.2 |
| 14:43.00 | [TK]D-Fender | badcfe: You should ready all the docs for whats deprecated.... upgrade.txt, check the WIKI, the changelogs on asterisk.org, etc.... |
| 14:43.02 | mgbowman | «« Leaving »» Reason~[out of the office]~ « Ë×Çü®§îöñ » |
| 14:46.38 | badcfe | JunK-Y, [TK]D-Fender: thanks. i see i will need to use "functions" like CALLERID(dnid) and so |
| 14:46.53 | JunK-Y | for more info: core show function CALLERID in ur CLI |
| 14:47.06 | JunK-Y | be careful, function are UPPERCASE. |
| 14:47.37 | badcfe | JunK-Y: and do you happen to know what to get a timestamp resembling the on in 1.2s $TIMESTAMP ? |
| 14:47.39 | [TK]D-Fender | badcfe: Correct. Most of that was deprecated in 1.2 and should have been changed. 104 completely REMOVED support for the old way.... you need to keep up more on these sort of changes. |
| 14:48.20 | badcfe | s/104/1.4 no, [TK]D-Fender meant that |
| 14:48.38 | badcfe | my joke with jbot didnt work this time 8-( |
| 14:49.08 | JunK-Y | badcfe: core show function STRFTIME |
| 14:49.48 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:52.31 | *** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net) |
| 14:56.38 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 14:57.23 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 14:57.39 | TaiSHi | How do I get verbosity to higher levels ? |
| 14:58.17 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:58.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:58.26 | naitram | TaiSHi: -rvvvvvvvvvvv sets it to how many v's you put |
| 14:58.59 | creativx | set verbosity twenty tousand |
| 14:59.49 | naitram | anyone know how to do automatic ring down. IE... pick up handset and automatically dial a certain number? |
| 15:00.01 | badcfe | JunK-Y: im sorry to hang on with this inquisitional questioning. what i look for actually is a variable that identifies the call leg -- that will be the same before the dial and even in the cdr. It is the one given as "UniqueID:" in the manager cdr event. |
| 15:00.35 | [TK]D-Fender | badcfe: ....."${UNIQUEID} |
| 15:01.04 | badcfe | [TK]D-Fender: i feel stupid. thank you |
| 15:01.11 | creativx | tracking uniqueid is fun |
| 15:01.38 | creativx | i once tried following an incoming call based on its uniqueid over the ami |
| 15:01.42 | [TK]D-Fender | creativx: Imagine how much moreso were they NOT ;) |
| 15:01.55 | creativx | imagine no uniqueids |
| 15:01.57 | creativx | just random id's |
| 15:01.59 | JunK-Y | uniqueid isnt specify the call leg, its specify the call. |
| 15:02.20 | [TK]D-Fender | ${CHANNEL} |
| 15:02.23 | [TK]D-Fender | ^^^^^^^^ |
| 15:02.43 | *** join/#asterisk Deeewayn_ (n=dwayne@c-68-62-166-235.hsd1.al.comcast.net) |
| 15:03.44 | creativx | damn now iam ready for weeeeeeeekend |
| 15:03.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:05.06 | badcfe | in asterisk 1.0 this uniqueid gets its last digit cut off when above xxx.99999 -- this caused me a "puzzle" once. |
| 15:06.01 | creativx | hehe |
| 15:06.06 | creativx | can imagine that |
| 15:06.40 | badcfe | creativx: and i was awfully sick (of alcohol) and all the clients panicked because of the realtime system screwing up |
| 15:07.02 | creativx | haha..scary |
| 15:07.08 | creativx | i havent done one single productive thing today |
| 15:07.15 | badcfe | creativx: but hey -- what doesnt kill us makes us stronger 8-) |
| 15:07.18 | creativx | and i was supposed to continue on the asterisk implementation |
| 15:07.24 | creativx | hell yes |
| 15:07.29 | creativx | or atleast hungover :) |
| 15:08.11 | naitram | anyone have any idea how to do automatic ringdown? |
| 15:08.34 | creativx | naitram, you cant make the phone do anything when you pick up the handset |
| 15:08.48 | af_ | what is zaptel transcode? |
| 15:08.53 | creativx | if its a sip phone you can make asterisk set up a call to it with answer-after=1 |
| 15:08.54 | creativx | or 0 |
| 15:09.12 | creativx | but im not sure if thats what you are after |
| 15:10.24 | af_ | is it need some specific hardware? |
| 15:10.32 | naitram | creativx: it is a sip phone, how would you do it with a sip |
| 15:11.01 | creativx | naitram: im not sure what a ringdown is.. and what you are trying to do |
| 15:11.25 | creativx | i only know how to do a click-2-dial with the sip phone.. eg click a link and originate a call without needing to pick up the sip phone |
| 15:11.32 | creativx | which is gg when you have a headset |
| 15:12.49 | naitram | creativx: I want a phone that will dial 1 extension when the handset is picked up. Imagine that a phone is set up as an intercom device. Essentially, whenever you pick up the phone, I want to dial the other end of the intercom without having to enter an extension |
| 15:13.35 | creativx | yeah i understand |
| 15:13.42 | creativx | then you would have to know somehow that the handset is picked up |
| 15:14.05 | creativx | and afaik theres no notification of that... hints wont work for that i think |
| 15:14.19 | naitram | I had hoped that asterisk could detect a handset pickup and execute scripts based on that, sounds like it cant? |
| 15:14.36 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 15:15.08 | creativx | the phone never notifies that its picked up |
| 15:15.10 | creativx | bbl.. weekend |
| 15:16.59 | [TK]D-Fender | naitram: Certain SIP phones allow you to do this. You can also do it with "immediate=yes" on Zaptel FXS channels |
| 15:17.36 | *** join/#asterisk step_quasar (n=step_qua@191-91-235-201.fibertel.com.ar) |
| 15:20.25 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:23.02 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 15:27.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:27.23 | new2345 | im having basic new user issues, can someone help me out? |
| 15:28.31 | *** join/#asterisk RoyK (n=roy@cEE71BF51.dhcp.bluecom.no) |
| 15:29.53 | new2345 | anyone?...i am unable to get audio one way |
| 15:30.08 | new2345 | on SIP behind nat |
| 15:30.25 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:33.11 | new2345 | anyone? |
| 15:34.05 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:35.05 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-57.mtco.com) |
| 15:35.50 | tzafrir_laptop | new2345, we can't help you without knowing your problem first |
| 15:36.05 | tzafrir_laptop | one way audio: |
| 15:36.17 | tzafrir_laptop | is there some NAT in the middle? |
| 15:36.26 | [TK]D-Fender | new2345>on SIP behind nat |
| 15:36.30 | [TK]D-Fender | ^ |
| 15:37.23 | TaiSHi | How do I set default codec for * server? |
| 15:37.29 | badcfe | for migration from 1.2 to 1.4, i dont find a replacement for the ${ANSWEREDTIME} variable |
| 15:37.33 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 15:37.43 | giesen | is there a way to check your voicemail from your own voicemail prompt |
| 15:37.48 | giesen | or to configure asterisk to do it |
| 15:38.01 | giesen | havent been able to find anything yet |
| 15:38.07 | giesen | ie you dial yourself |
| 15:38.15 | giesen | you start to hear your greeting |
| 15:38.17 | giesen | and you push a key |
| 15:38.23 | giesen | to get a password prompt |
| 15:39.33 | new2345 | i am getting 2 warnings one is that i dont have the "insecure" variable set for the provider, and the other is that the remote host can't NOTIFY (yes I am using VTWhite from behind a WRT router with AsteriskNow beta in the DMZ) |
| 15:40.06 | Merlin83b | So, erm, where do I get libcapi from? |
| 15:40.13 | new2345 | i have made some progress (allowed codecs was missing, now i get at least outbound audio) |
| 15:41.28 | badcfe | ah i just found the wonderfull CDR function. wich answers my need |
| 15:41.36 | [TK]D-Fender | giesen: look at the "a" standard extension on the WIKI |
| 15:41.44 | giesen | thansk |
| 15:41.50 | tzafrir_laptop | jbot, sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:41.53 | jbot | tzafrir_laptop: okay |
| 15:42.25 | [TK]D-Fender | Merlin83b: Depends on your distro learly... |
| 15:42.28 | [TK]D-Fender | clearly* |
| 15:42.34 | Merlin83b | AsteriskNOW b4. |
| 15:42.51 | Merlin83b | I can't even find the place to grab the source from. |
| 15:42.58 | Merlin83b | I must be missing something very obvious. |
| 15:45.50 | giesen | [TK]D-Fender: there's nothing actually on the wiki about it |
| 15:46.43 | *** join/#asterisk dswillia (n=swilliam@199.3.247.34) |
| 15:46.44 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 15:47.15 | dswillia | what are peoples thoughts on Sangoma's Cards? |
| 15:47.35 | giesen | my impression is they're generally superior to digium cards |
| 15:47.49 | dswillia | I would prefer digiums, however, the New Dell Servers we buy do not offer PCI ports |
| 15:48.02 | cpm | dswillia, I've heard nothing but good about'em |
| 15:48.37 | [TK]D-Fender | giesen: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 15:48.43 | [TK]D-Fender | giesen: Top line practically |
| 15:48.52 | giesen | yeah I know |
| 15:48.54 | giesen | but that's all there is |
| 15:48.56 | dswillia | I need pci express cards and digium just doesnt have em |
| 15:49.04 | giesen | no explanation |
| 15:49.05 | giesen | nothing |
| 15:49.07 | giesen | click it |
| 15:49.10 | giesen | it leads to a blank page |
| 15:49.41 | giesen | hmm |
| 15:49.48 | giesen | I think I may be able to work it |
| 15:49.54 | giesen | I was thinking it was sometbing else |
| 15:50.06 | giesen | not an extension but a setting like mMtT |
| 15:50.12 | giesen | on the dial command |
| 15:51.01 | TaiSHi | What codec do you suggest for using on a LAN envirovement ? |
| 15:51.04 | [TK]D-Fender | giesen: If you are listening to the greeting, you hit "*" and if "exten => a,1,askjhkjsdhjksd" exists in the current context, if will ext at which point you can do what you want |
| 15:51.13 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:51.17 | [TK]D-Fender | TaiSHi: G.711 naturally |
| 15:51.20 | Dr-Linux | hi guys |
| 15:51.28 | *** join/#asterisk dual-man (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 15:51.46 | TaiSHi | And how do I set * to use that codec? |
| 15:51.55 | TaiSHi | (between soft phones internally) |
| 15:52.08 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 15:52.23 | [TK]D-Fender | TaiSHi: Depends for which protocol. You really need to read the basic on setting up channels... go downlaod and readTHE BOOK |
| 15:52.24 | Dr-Linux | anybody refer me link where i can get help for my simple ivr, i.e. caller should get "Goodmorning" message when he/she calls in morning ... same goodafternoon and good evening? |
| 15:52.25 | [TK]D-Fender | ~book |
| 15:52.27 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:52.36 | dual-man | lets say i have an include command in a dialplan and on certian conditions the dialplan passes into a macro, in which i don't want the include data avialable how can i cancel it out? |
| 15:52.39 | [[blah]asfd | has anyone had the problem with polycom phones where for no reason after a reboot... all of the phones have a duplicate ip address? |
| 15:52.46 | [TK]D-Fender | Dr-Linux: THE BOOK.... as linked |
| 15:53.02 | Qwell[] | Dr-Linux: hey |
| 15:53.12 | Dr-Linux | :S |
| 15:53.17 | Qwell[] | Dr-Linux: http://bugs.digium.com/view.php?id=9245 :) |
| 15:53.19 | Dr-Linux | Qwell[]: hi there :) |
| 15:53.22 | Qwell[] | test it out please |
| 15:53.32 | Dr-Linux | lemme see thank you |
| 15:54.50 | Dr-Linux | wowww |
| 15:55.01 | Dr-Linux | Qwell: you did that? |
| 15:55.12 | Qwell[] | nope, Slimey did |
| 15:55.18 | [[blah]asfd | i am having problems where if I reboot my polycom phone it comes up with a duplicate ip address.... but in dhcp tables, there is nothing listed. |
| 15:55.57 | dual-man | i have a features.conf file which i include into a dialplan, but when an extension is busy, the dialplan passes into a different macro, and i don't want the features available there, how can i do this? |
| 15:55.57 | Dr-Linux | cool |
| 15:56.00 | Dr-Linux | is it tested? worked? |
| 15:56.08 | Qwell[] | he tested it |
| 15:56.09 | *** join/#asterisk svenna_ (n=svenna@p548D2F47.dip0.t-ipconnect.de) |
| 15:56.51 | Dr-Linux | good good |
| 15:56.58 | Dr-Linux | i'll test it as well :) |
| 15:57.49 | *** join/#asterisk ppyy (i=ppyy@218.93.153.191) |
| 15:58.18 | brettnem | [[blah]asfd you probably have another DCHP server on your network somewhere. |
| 15:59.29 | giesen | [TK]D-Fender: thanks I managed to make it work |
| 15:59.36 | giesen | not quite the granularity I wanted |
| 15:59.54 | [TK]D-Fender | giesen: Its actually as easy as it says. add the exten, press * go there, do stuff :) |
| 15:59.58 | giesen | yeah |
| 16:00.02 | [TK]D-Fender | giesen: How so? |
| 16:00.06 | giesen | except tha applies for the whole context |
| 16:00.19 | giesen | I'd rather have it as a standard priority than a standard extension +) |
| 16:00.39 | Dr-Linux | [TK]D-Fender: can i find that in the book what i'm looking for? |
| 16:00.39 | giesen | it applies to all extensions in that context |
| 16:00.42 | [TK]D-Fender | giesen: That means that you can make it do WHATEVER though. You'd be smart to use Macro's for things like this so as to encapulate it |
| 16:01.03 | giesen | haha I have orders not to spend more than 10 minutes on this |
| 16:01.09 | Dr-Linux | anybody refer me link where i can get help for my simple ivr, i.e. caller should get "Goodmorning" message when he/she calls in morning ... same goodafternoon and good evening? |
| 16:01.10 | giesen | it's for my boss' brother in law |
| 16:01.15 | giesen | and he doesnt like him very much =) |
| 16:01.29 | giesen | so writing a macro is out of the question |
| 16:01.41 | giesen | either way it works well enough |
| 16:01.46 | giesen | thank you for the help |
| 16:01.53 | [TK]D-Fender | Dr-Linux: "show application gotoiftime" <- |
| 16:02.19 | [TK]D-Fender | Dr-Linux: How on earth can you not have gone through "show applications" to see what dialplan commands there were this? |
| 16:02.43 | *** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net) |
| 16:02.43 | Dr-Linux | [TK]D-Fender: i'm sorry sir |
| 16:03.01 | [TK]D-Fender | Dr-Linux: I'm just a bit shocked. You've been working with * for how long now? |
| 16:03.06 | Dr-Linux | but i gone through, actually i was looking for a WIKI's example |
| 16:03.39 | Dr-Linux | [TK]D-Fender: hhmm... maybe i'm dumb? |
| 16:03.41 | [TK]D-Fender | Dr-Linux: You should be more than comfortable goin to the wiki and loking up the BIG links like "full list of applications" , "asterisk variables", "asterisk functions", etc... |
| 16:03.59 | [[blah]asfd | i am trying to find the key stroke combo for hard resetting the configs on the polycom 301 phones... but i can only find reference to how to do it from the menus. can anyone tell me the button combo to reset a phone that wont boot up all the way? |
| 16:04.02 | Dr-Linux | yeah |
| 16:04.07 | [TK]D-Fender | Dr-Linux: I'd sooner believe LAZY, but if you want to pin that label on yourself, thats your peragative :) |
| 16:04.38 | *** join/#asterisk Avochelm (n=damien@CPE-144-136-166-42.sa.bigpond.net.au) |
| 16:04.42 | Dr-Linux | heh :) but i'm the ONE where i live |
| 16:04.47 | *** join/#asterisk Urgleflogue (n=plamen@87-126-143-181.btc-net.bg) |
| 16:05.41 | reber | Are /etc/asterisk/ 1.2.14 configuration files compatible with * 1.4.0 ? |
| 16:06.10 | Juggie | yes |
| 16:06.26 | reber | i think especially of sip.conf extensions.conf meet*.conf |
| 16:06.35 | Juggie | yes |
| 16:06.36 | reber | Juggie, perfect |
| 16:10.07 | [TK]D-Fender | reber: Depends |
| 16:10.30 | [TK]D-Fender | reber: Extensions.conf might not be good if you're using deprecated commands, variables, etc in there.... |
| 16:11.06 | [TK]D-Fender | reber: If you want a better opinion on its upgrade-readiness, pastebin the whole thing. |
| 16:12.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:12.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:17.53 | Hmmhesays | you could always go the easy way and just test it |
| 16:18.34 | file | it's Hmmhesays! |
| 16:18.50 | Qwell[] | file: Does he really say hmm? |
| 16:18.56 | file | Qwell[]: no |
| 16:19.00 | file | Qwell[]: he says hrm |
| 16:19.07 | Qwell[] | very interesting |
| 16:19.15 | file | Qwell[]: very |
| 16:27.02 | *** join/#asterisk Ac1dcrawl (n=cow@64.31.169.118) |
| 16:27.05 | *** join/#asterisk douglas_om (n=dom@201.67.216.4) |
| 16:27.52 | Ac1dcrawl | so my telco is providing me with a T1, and they want to know if I want the T1 64k or 56k, what would be the most common that asterisk supports? |
| 16:27.56 | Hmmhesays | hello file |
| 16:28.09 | Hmmhesays | I'm watching a guitarworld dvd |
| 16:28.10 | Ac1dcrawl | or is it a T1 card thing? |
| 16:28.49 | douglas_om | Greetings! I want something like my own skypeout. What I really want is to make phone calls using a remote computer with a modem. I wonder if asterisk can be used for that. |
| 16:29.17 | brettnem | sure it can |
| 16:29.26 | douglas_om | Great! |
| 16:30.11 | douglas_om | I need to make phone calls to a city but I have a computer there with internet connection. So, I could connect to this computer by internet and use it to make the phone calls for me. |
| 16:30.19 | brettnem | sure |
| 16:30.25 | douglas_om | Excelent! :-) |
| 16:30.32 | *** join/#asterisk putnopvut (n=putnopvu@c-76-18-109-29.hsd1.al.comcast.net) |
| 16:30.32 | brettnem | piece of cake |
| 16:30.46 | douglas_om | Thanks... Gonna check on the website how to do that, then. |
| 16:30.58 | reber | [TK]D-Fender, actually i consider upgrading to 1.4.0 because all works great, but i'm facin problems with conference calls (meeting rooms). Is there a list of deprecated stuff in configuration files to see if it's worth to upgrade ? |
| 16:31.05 | douglas_om | That will help me save lots of skype credits... hehe |
| 16:31.18 | brettnem | what's skype?? :) |
| 16:31.21 | douglas_om | hehehe |
| 16:31.27 | douglas_om | Just one more thing... |
| 16:31.48 | [TK]D-Fender | reber: Maybe you should see if your problem can be fixed in 1.2 first rather that make a big upgrade and introduce REAL problems. |
| 16:31.53 | douglas_om | That remote computer has to be linux/freebsd.... Not Windows, right? |
| 16:32.22 | brettnem | really, don't do it on windows.. you can, but you're likely to have a ton of problems and 0 support |
| 16:32.35 | brettnem | in reality, I don't think you need an entire computer |
| 16:32.39 | brettnem | hmm.. must be using windows |
| 16:32.40 | Qwell[] | brettnem: not 0 problems and a ton of support? |
| 16:32.52 | [TK]D-Fender | douglas_om: If you want to make your life easy, jsut install a Linksys SPA-3102 at that sight and run * where YOU are. |
| 16:32.54 | Qwell[] | nevermind, I like yours better :p |
| 16:33.03 | brettnem | are you going to support his install of asterisk on windows? :) |
| 16:33.12 | Qwell[] | brettnem: absolutely |
| 16:33.17 | Qwell[] | $250/h, 4h min |
| 16:33.18 | *** join/#asterisk douglas_om (n=dom@201.67.216.4) |
| 16:33.25 | douglas_om | Ugh... Xgl hanged. |
| 16:33.25 | brettnem | haha Qwell |
| 16:33.29 | Qwell[] | ;) |
| 16:33.33 | brettnem | darn that Xgl |
| 16:33.33 | Qwell[] | douglas_om: yeah, it does that |
| 16:34.01 | douglas_om | That just happened with you too? |
| 16:34.08 | brettnem | yeah, so as [TK]D-Fender so astutely said while you were out, If you want to make your life easy, jsut install a Linksys SPA-3102 at that sight and run * where YOU are. |
| 16:34.11 | Qwell[] | douglas_om: no, I stopped using it, heh |
| 16:34.14 | douglas_om | ahha |
| 16:34.27 | douglas_om | I think I'll have to stop using it too... :-( |
| 16:34.32 | douglas_om | So fancy... so cool... so instable |
| 16:34.44 | Qwell[] | unstable* |
| 16:34.47 | douglas_om | oh sorry |
| 16:43.16 | *** join/#asterisk lerat (n=dnormand@bas2-montreal19-1178031979.dsl.bell.ca) |
| 16:43.34 | douglas_om | Do you know how to make new windows appear in the front? They are all appearing in the background. Hate that. |
| 16:43.38 | douglas_om | Using Beryl |
| 16:43.51 | lerat | does anybody know if there is any ASTERISK GUI CREATOR out there????? |
| 16:44.03 | Qwell[] | lerat: a gui creator? |
| 16:44.13 | Qwell[] | like, something that creates guis, or a person who does so? |
| 16:44.21 | lerat | a software |
| 16:44.42 | [TK]D-Fender | lerat: Try a Linux distro, they have all sort of great progeamming tools typically ;) |
| 16:44.55 | Qwell[] | vi? |
| 16:45.01 | lerat | thanks |
| 16:45.10 | *** join/#asterisk Fr0zen_ (i=Fr0zen_@unaffiliated/fr0zen/x-000001) |
| 16:45.11 | [TK]D-Fender | Qwell :That'd be a great tool... one of many! |
| 16:45.14 | Fr0zen_ | Mar 9 09:33:24 NOTICE[31705] chan_sip.c: Peer 'viatalk' is now REACHABLE! (12ms / 2000ms) |
| 16:45.16 | Fr0zen_ | Mar 9 09:36:26 NOTICE[31705] chan_sip.c: Peer 'viatalk' is now TOO LAGGED! (2013ms / 2000ms) |
| 16:45.18 | Fr0zen_ | Mar 9 09:36:36 NOTICE[31705] chan_sip.c: Peer 'viatalk' is now REACHABLE! (35ms / 2000ms) |
| 16:45.20 | Fr0zen_ | what can be causing those lag spikes? |
| 16:45.20 | lerat | well i m not so familiar with VI |
| 16:45.20 | BrianR___ | The polycom 2.1.0 firmware has a working microbrowser with tables on the 430, 501, and 601... |
| 16:45.33 | BrianR___ | Is anyone using the polycom microbrowser for things like a phone directory? |
| 16:45.34 | Fr0zen_ | i can ping that address and it never hicups, but asterisk shows lag spikes |
| 16:45.38 | [TK]D-Fender | BrianR___: Indeed it does |
| 16:45.58 | [TK]D-Fender | BrianR___: And Yes, I've used it for directories before |
| 16:46.02 | BrianR___ | [TK]D-Fender: I can't figure out a URL scheme which will make the phone dial a number. |
| 16:46.09 | *** join/#asterisk test34- (n=test34@unaffiliated/test34) |
| 16:46.35 | BrianR___ | <a href="tel:somephonenumber">some link text</a> doesn't seem to work. |
| 16:46.39 | [TK]D-Fender | BrianR___: Thats because there is no magic tag to make the PHONE dial. You need to think a little more outside the box. |
| 16:46.54 | Qwell[] | [TK]D-Fender: Cisco has one ;) |
| 16:47.03 | Qwell[] | I implemented voicemail in xml :D |
| 16:47.10 | BrianR___ | [TK]D-Fender: My other thought was to kludge it with a cgi that calls the phone back or something, but that seems a little nasty :( |
| 16:47.24 | BrianR___ | Qwell[]: The cisco xml thing is way better than the polycom one... |
| 16:47.24 | [TK]D-Fender | Qwell[]: Sure, but its the other "bonus" features they come with that make that value rather moot ;) |
| 16:47.34 | Qwell[] | such as? |
| 16:47.47 | BrianR___ | Of course there's other stuff that sucks on the cisco phones... |
| 16:47.48 | Qwell[] | what can a polycom do that my cisco can't? Besides make calls |
| 16:47.52 | [TK]D-Fender | BrianR___: Quick PHP can do the job easy. |
| 16:47.59 | BrianR___ | The 601's speakerphone is way better than the 79xx speakerphone. |
| 16:48.06 | Qwell[] | BrianR___: no it isn't |
| 16:48.11 | Qwell[] | it's the same tech |
| 16:48.37 | BrianR___ | [TK]D-Fender: Stuffing a call file in the /var/spool/asterisk/outgoing |
| 16:48.59 | [TK]D-Fender | Qwell[]: Presence, larger line/call handling, massive provisioning methods, etc... |
| 16:49.11 | BrianR___ | Qwell[]: Maybe it's a subjective thing, but the 79xx seems to use accoustic echo suppression - the 601 seems to have real AEC. |
| 16:49.30 | [TK]D-Fender | Qwell[]: The fact Ciisco's SIP implementationis about as stable as BC Vesuvius ;) |
| 16:49.35 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:50.04 | [TK]D-Fender | BrianR___: Most accounts I've heard say that Cisco's audio quality is pretty much on par. |
| 16:50.11 | *** join/#asterisk boch (n=fran@190.48.237.246) |
| 16:50.11 | BrianR___ | [TK]D-Fender: If I use an autoanswer alertinfo.... |
| 16:50.41 | BrianR___ | [TK]D-Fender: I've got both a 7940 and a 601 on my desk... We trialed the 79xx and the 601 and went with the 601... |
| 16:50.43 | [TK]D-Fender | BrianR___: z0mg, careful, you're starting to THINK! Get an extinguisher fast before combustion ensues! |
| 16:51.27 | BrianR___ | [TK]D-Fender: Seems a little hackish, but... |
| 16:51.37 | BrianR___ | That was our scheme for doing click-to-dial around here too. |
| 16:51.57 | [TK]D-Fender | BrianR___: tip : AMI Originate. super easy.... |
| 16:52.13 | [TK]D-Fender | BrianR___: And so much better than call files |
| 16:52.28 | boch | duds, do i have to clear resultsets when querying with odbc? cause i need to do querys before clearing lastest resultsets and mysql-server is complaining: [Mar 9 13:26:56] WARNING[15663]: app_addon_sql_mysql.c:268 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: Commands out of sync; you can't run this command now |
| 16:52.33 | *** part/#asterisk lerat (n=dnormand@bas2-montreal19-1178031979.dsl.bell.ca) |
| 16:52.44 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 16:53.27 | rhombus | Callers trying to make outgoing calls on a Zap channel are getting "congestion" and a fast busy signal. |
| 16:53.43 | rhombus | at first I thought it was because the line was off-hook, but the reason given in the CLI output is "Unknown" |
| 16:53.45 | rhombus | so now I'm not sure |
| 16:54.00 | rhombus | The CLI output for an example is http://pastebin.ca/387757 |
| 16:54.16 | rhombus | most calls go through; this only happens occasionally |
| 16:54.38 | rhombus | the two channels are part of an Zap group |
| 16:55.20 | [TK]D-Fender | -- Started music on hold, class 'default', on channel 'Zap/1-1' |
| 16:55.32 | [TK]D-Fender | rhombus: Messages in there sure seem to tell us that its busy... |
| 16:56.15 | tzafrir_laptop | hmmm, someone "hung up" Zap/1 but instead flashed it? |
| 16:57.02 | boch | anyone querying a DB from the dialplan ? |
| 16:57.04 | rhombus | tzafrir: well, I'm not sure -- can a SIP extension even do that? |
| 16:57.32 | rhombus | <[TK]D-Fender> who started the music on hold? I think that was just him trying to clear the line |
| 16:57.38 | rhombus | let me look at it again |
| 16:58.23 | *** join/#asterisk Glasswlkr (n=me@209.217.101.66) |
| 16:58.26 | rhombus | <[TK]D-Fender>: Why would it say "cause 0 - Unknown" if the line is open? |
| 16:58.30 | [TK]D-Fender | rhombus: Its all pretty clear the line was busy from before the start of your pastebin. . "show channels" is your friend. |
| 16:58.39 | rhombus | let me go and get the missing stuff |
| 17:00.24 | rhombus | okay, it's here now: http://pastebin.ca/387765 |
| 17:00.51 | rhombus | the problem is that 200 is supposed to get a Zap group when dialing out, but it looks like it doesn't even try the group here |
| 17:00.51 | tzanger | in 1.4.x, why are my voicemail attachments coming in as plain text instead of wave files? It's gotta be something stupid I'm missing |
| 17:01.53 | rhombus | <[TK]D-Fender>: you can see that 201 has Zap/1, so why is 200 even trying Zap/1? It's supposed to try the group, and there is an open channel in the group |
| 17:02.26 | [TK]D-Fender | rhombus: Your dial is not choosing a GROUP, its choosing a CHANNEL |
| 17:02.39 | rhombus | okay, I'll look at the dialplan again |
| 17:02.55 | [TK]D-Fender | rhombus: Its more than clear right in your pastebin... |
| 17:03.17 | rhombus | oh, because you can see the actual dial command |
| 17:03.21 | rhombus | typo again |
| 17:03.21 | rhombus | thanks |
| 17:03.48 | rhombus | i really appreciate all this help, and when the pressure is off and I get more familiar with this, my questions will become less stupid |
| 17:03.50 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:03.54 | rhombus | nonetheless I appreciate your patience |
| 17:04.21 | [TK]D-Fender | :) |
| 17:04.50 | [TK]D-Fender | rhombus: Not stupid.... the few characters involved in the error the hard it may be to find :) |
| 17:04.57 | [TK]D-Fender | fewer* |
| 17:05.31 | [TK]D-Fender | rhombus: And the fact that you told us you thought you were dialing a GROUP only AFTER your pastebin would naturally throw us off... |
| 17:05.31 | rhombus | <[TK]D-Fender>: the unfamiliarity makes me nervous and so I have a tendency to scan rapidly instead of being relaxed and systematic about it |
| 17:05.51 | rhombus | well, you still found it |
| 17:06.06 | [TK]D-Fender | rhombus: On no... do nit-pick the hell out of each app line being called and ask yourself "is this EXACTLY as I want it called?" |
| 17:06.07 | rhombus | next time I'll say it before I post the pastebin :) |
| 17:06.23 | rhombus | <[TK]D-Fender>: yes, that's excellent advice |
| 17:06.39 | [TK]D-Fender | rhombus: And everyone overlooks little stuff here and there.. at least it didn't take more than 10s to pin it down :) |
| 17:07.15 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 17:07.34 | *** join/#asterisk Assid (n=assid@59.183.60.248) |
| 17:07.37 | Assid | heya |
| 17:08.05 | Assid | i just upgraded to 1.4.1 but now .. calleridname shows as asterisk .. and when i try to redial.. it dials asterisk |
| 17:08.13 | Assid | i got the calleridnum to work fine tho |
| 17:10.19 | rhombus | When parking calls, I see phones give a SIP 500 error, which is a bit unnerving, but from what I can see it doesn't affect the call experience. Should I be worried about this? |
| 17:10.26 | rhombus | There's a pastebin: http://pastebin.ca/387777 |
| 17:10.41 | [TK]D-Fender | rhombus: Nope, random Polycom spewing... largely unimportant. |
| 17:12.01 | Assid | anyone know what to do for this? i set caller(ani) and callerid(num) and callerid(name) |
| 17:12.09 | Assid | i still get "asterisk" as the name |
| 17:13.30 | *** join/#asterisk RoyK (n=roy@cEE71BF51.dhcp.bluecom.no) |
| 17:15.08 | *** join/#asterisk RoyK (n=roy@cEE71BF51.dhcp.bluecom.no) |
| 17:15.11 | markit | how have current datetime in the DDMMYYYY-HH:MM:SS format? the suggestion for EPOCH in the wiki does not work (${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}) |
| 17:15.29 | Glasswlkr | Hey, anyone know how to configure the feature keys on the Polycom Soundpoint IP 501? |
| 17:15.35 | [TK]D-Fender | Assid: Happens occasionally on analog CID channels |
| 17:15.44 | [TK]D-Fender | Glasswlkr: Which? |
| 17:15.47 | Assid | asterisk@ip ? |
| 17:16.01 | Fr0zen_ | do you need the defaultip for each "friend" aka phone on the network? |
| 17:16.13 | [TK]D-Fender | Fr0zen_: Nope |
| 17:16.17 | Assid | [TK]D-Fender: how do i fix it.. this only started with 1.4 |
| 17:16.22 | [TK]D-Fender | Fr0zen_: Let phones register like normal. |
| 17:16.33 | [TK]D-Fender | Assid: No clue, I'm still avoiding 1.4 |
| 17:16.43 | *** part/#asterisk putnopvut (n=putnopvu@c-76-18-109-29.hsd1.al.comcast.net) |
| 17:16.46 | Glasswlkr | [TK]D-Fender: I need to change the voicemail button to dial *97 for asterisk, and want to remap a few of the other buttons. |
| 17:16.56 | Fr0zen_ | thx tkd |
| 17:17.23 | [TK]D-Fender | Glasswlkr: Go download the SIP Administrators Guide for your firmware revision off Polycom's site. Its all detailed in there |
| 17:17.26 | Glasswlkr | I can see in the docs it says use key.IP_500.XX.func.prim to change it (but what number is each button?) and it says I can map it to the pre-programmed features, but can I make a button dial a feature code? |
| 17:17.44 | [TK]D-Fender | Glasswlkr: That is NOT where you set this |
| 17:18.07 | Glasswlkr | [TK]D-Fender: ok well in the admin guide for my firmware that is the only reference to the keys I can find. |
| 17:18.10 | [TK]D-Fender | Glasswlkr: Look near the bottom, or try a text search |
| 17:18.49 | Assid | [TK]D-Fender you sticking to latest 1.2.x tree ? |
| 17:18.54 | Glasswlkr | ok looking again... will let you know if I find it |
| 17:19.10 | Fr0zen_ | are the actual extension numbers and such all stored in extensions.conf? Sorry to ask, but i'm used to trixbox and I just migrated over to pure asterisk. ;) |
| 17:19.59 | [TK]D-Fender | Glasswlkr: Look HARDER - 4.6.2.5.1 |
| 17:20.14 | Ac1dcrawl | I'm getting an error: Unable to create channel of type 'ZAP' |
| 17:20.25 | Ac1dcrawl | My span is OK, so I know it's not that |
| 17:20.27 | Ac1dcrawl | any idea's? |
| 17:20.56 | [TK]D-Fender | Ac1dcrawl: 1 we don't trust your assessment that "its ok", and 2 you didn't SHOW us anything.... |
| 17:21.12 | [TK]D-Fender | Ac1dcrawl: That error message alone is regrettably meaningless |
| 17:21.27 | Glasswlkr | [TK]D-Fender: Oh yeah, for the message button only (and I also tried that allready and it doesn't work, it changes the number it dials but it still goes to the "message" window rather than one-touch message access that's what prompted me to want to change the button) |
| 17:21.38 | Glasswlkr | I allready set the bypassinstantmessage to 1 |
| 17:21.47 | [TK]D-Fender | Ac1dcrawl: pastebin a whole whack of backup for your servers current stats, some CLI output fo a failed call, etc |
| 17:21.47 | Glasswlkr | and there is only one extension registered on the phone |
| 17:21.48 | Assid | [TK]D-Fender: care to help me look on the docs for this? i am setting all 3 of them .. callerid number is being set correctly |
| 17:22.00 | Glasswlkr | Anyway, but that also doesn't help me map any other feature keys to different options. |
| 17:22.05 | Assid | or maybe can i have polycom call the callerid number instead of the asterisk@ip ? |
| 17:22.20 | [TK]D-Fender | Glasswlkr: What you want is under the section I pointed you to. |
| 17:22.20 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 17:22.30 | Assid | [TK]D-Fender: why does polycom ignore the calleridnumber and still dial the asterisk@ip ? |
| 17:22.40 | new2345 | i still can't seem to get this thing working, i have set the nat settings, but still i get an error that the remote host is trying to respond to a NOTIFY with a local address |
| 17:22.44 | TheCompWiz | quick question... when building a dial plan... can you have 2 exten=> with the same id? (i.e. 1) |
| 17:23.02 | [TK]D-Fender | Assid: Polycem gets what Asterisk GIVES it. Polycom doesn't make this stuff up.... |
| 17:23.25 | [TK]D-Fender | TheCompWiz: Sure... jsut expect the 2nd to actually WORK (if you're lucky) |
| 17:23.41 | Glasswlkr | Dude, I have this set in my config, and it is NOT working: <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="*97" /> is that not exactly what the section you referred me to says? |
| 17:23.56 | Ac1dcrawl | Well, when I go into zttools the span says OK, it's not alarmed |
| 17:23.59 | TheCompWiz | ok... is there a way to pre-ceed an exten? |
| 17:24.08 | Glasswlkr | If it is something I am doing wrong, then perhaps I can correct it, I just don't know what I am doing wrong |
| 17:24.11 | [TK]D-Fender | Glasswlkr: Yes, thats fine so far. |
| 17:24.23 | Glasswlkr | or if there is somewhere else I need to change something |
| 17:24.28 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 17:24.33 | [TK]D-Fender | Glasswlkr: Now I'd suggest you do BypassInstantmessage, and that 1 other setting much further up.... |
| 17:24.50 | Glasswlkr | <PROTECTED> |
| 17:24.54 | [TK]D-Fender | Glasswlkr: I'll let you fish for it for a few..... |
| 17:25.02 | Glasswlkr | one sec checking for the "other" setting you mentioned :) |
| 17:25.06 | TheCompWiz | in freepbx... they setup several include blah-custom before any of the context is defined... and they always start at "1" ... and I need to preceed one. |
| 17:25.10 | [TK]D-Fender | 1 more clearly related setting at least half-way up |
| 17:25.18 | [TK]D-Fender | ~freepbx |
| 17:25.20 | jbot | extra, extra, read all about it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:25.22 | TheCompWiz | so, I was hoping in the "custom" context.. I could do something before a call. |
| 17:25.36 | [TK]D-Fender | TheCompWiz: Too late, they already own your soul. |
| 17:25.41 | TheCompWiz | lol |
| 17:25.41 | [[blah]asfd | when I am placing outbound calls from one of my servers, the call goes through, and the cell phone i call can hear the person who dialed from asterisk, but they cannot hear me. They do not hear any ringing either. Inbound calls to those phones work just fine, outbound do not: Here is my pri intense debug: http://pastebin.ca/387798 |
| 17:25.51 | [[blah]asfd | can anyone possibly help me understand my issue? |
| 17:25.54 | [TK]D-Fender | ChanSmash(hope/all) |
| 17:26.00 | TheCompWiz | LOL |
| 17:26.20 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@static-71-116-124-200.snfcca.dsl-w.verizon.net) |
| 17:26.20 | [[blah]asfd | I upgraded my t1 drivers (sangoma) So I had to reconfigure the card... I am sure that is what has caused this. |
| 17:26.21 | rhombus | I'm looking for a VOIP provider for western Canada with low latency to Calgary and which is not totally amateur |
| 17:26.38 | jeremy_g | can i configure asterisk to register with itself |
| 17:26.41 | rhombus | I want latency under 50ms, so Ontario doesn't cut it |
| 17:26.46 | jeremy_g | and route call to itself |
| 17:27.12 | rhombus | does anybody have personal experience with one and that isn't also WORKING for said provider? |
| 17:27.21 | [[blah]asfd | rhombus: why would you want to do that? |
| 17:27.28 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:27.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:27.28 | [[blah]asfd | sorry... wrong person |
| 17:27.35 | jeremy_g | <PROTECTED> |
| 17:27.38 | file | rhombus: I've never used them but les.net is one I believe |
| 17:27.39 | [[blah]asfd | jeremy_g:that was for you |
| 17:27.39 | jeremy_g | :p |
| 17:28.04 | [[blah]asfd | jeremy_g: what are you trying to do? |
| 17:28.08 | Glasswlkr | Found it :) lol... the user prefrences for onetouchdial wasn't set... Totally missed that one |
| 17:28.09 | Glasswlkr | thanks! |
| 17:28.12 | file | rhombus: in Winnipeg, so shouldn't be too bad depending on routing |
| 17:28.24 | jeremy_g | [[blah]asfd:um not rhombus |
| 17:28.53 | [[blah]asfd | yeah.. I corrected myself |
| 17:28.58 | [[blah]asfd | what are you trying to do? |
| 17:28.59 | rhombus | les.net has no POP in winnipeg |
| 17:29.00 | markit | anyone with a STRFTIME format that makes me have the same as the old $TIMESTAMP variable? |
| 17:29.15 | rhombus | they are based in winnipeg but the POP is in the US and the latency is not great |
| 17:29.16 | Glasswlkr | Now onto my second issue, is there anywhere I can find what button is which "number" for remapping feature keys? (my users have mentioned a few remappings they want, like to dial a specific feature code from a key, such as the "services" key which does nothing anyway lol |
| 17:30.46 | [[blah]asfd | jeremy_g: guess you figured it out... good luck |
| 17:30.59 | [TK]D-Fender | Glasswlkr: I was hoping you'd find it on your own :) |
| 17:31.20 | [TK]D-Fender | Glasswlkr: So that aside is there anything else you really feel you'd need to "reprogram"? |
| 17:31.29 | Glasswlkr | well as I said I found the section on the key mapping, that I can do, but is the only way to know which key is hard-bound to which number just trial and error? |
| 17:31.56 | [TK]D-Fender | Glasswlkr: No, there is a full key-map as the beginning of the guide |
| 17:32.01 | Glasswlkr | oh |
| 17:32.02 | [TK]D-Fender | Glasswlkr: with pictures per-model |
| 17:32.06 | Glasswlkr | shit I totally missed that lmao |
| 17:32.14 | [TK]D-Fender | Glasswlkr: You really ought to read it more ;) |
| 17:32.51 | TheCompWiz | lemme get this right... if you use an "include" .. it's always processed "AFTER" everything else? is that right? |
| 17:33.42 | phillipk | Is there a way for me to tell if the PBX my Asterisk box is attached to is sending Qsig data? |
| 17:33.44 | Assid | [TK]D-Fender upgrade to 1.4.1 please |
| 17:33.48 | [TK]D-Fender | TheCompWiz: I believe its all in order of occurance |
| 17:33.57 | [TK]D-Fender | Assid: But my systems WORK! |
| 17:34.39 | Assid | hehe |
| 17:35.12 | TheCompWiz | [TK]D-Fender... just reading what is on voip-info... that's not true. what it says... is that all stuff in the context is processed first.. the sub-contexts in order of which they were included. |
| 17:35.47 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 17:35.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:36.32 | l2cache | Has anyone heard of any solutions other than ultramonkey with failover to cluster-loadbalance/highavailability Asterisk? |
| 17:36.52 | markit | any obvious reaso whi this will not work: exten => 996,n,NoOp,DateTime_prova222: ${STRFTIME(${EPOCH},GMT+1,%C%y%m%d%H%M)} while if I assign the STRFTIME stuff to a variable, and then display it with NoOp, is ok? |
| 17:37.29 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:38.54 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 17:40.35 | jeremy_g | [[blah]asfd:no i went to kitchen, my * box registers with another server. now that another server is gone and in order not to break my setup (a test bed) i must still register with it but itself |
| 17:42.24 | Assid | hrmm anyone here on 1.4.1? |
| 17:42.33 | bkruse | i am |
| 17:42.36 | l2cache | yes |
| 17:42.58 | new2345 | seems like there is no good reference that solves my problem, it is probably a single conf setting somewhere that is keeping my system from working, the call recipient on pstn can hear me, but I can't hear them... can anyone help me out...i have read the references about using NAT, still i am missing something |
| 17:43.20 | Assid | bkruse: you got caller id to be set correctly when you get incoming call? |
| 17:43.33 | Assid | my polycoms show the name correctly .. but number justr doersnt work |
| 17:43.37 | bkruse | in what?? sip.conf ? |
| 17:43.48 | l2cache | it has to do with reinvites |
| 17:43.51 | bkruse | Assid: sounds like a syntax mistake |
| 17:43.56 | bkruse | whats ur CID line plz |
| 17:44.01 | TheCompWiz | Assid ... probably an issue with your provider sending wrong CID info. |
| 17:44.11 | bkruse | TheCompWiz: he says from a polycom phone. |
| 17:44.21 | TheCompWiz | I have 30 polycom phones... no issues here. |
| 17:44.22 | bkruse | well, maybe not |
| 17:44.26 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 17:44.26 | bkruse | same |
| 17:44.55 | supjigatr | Anyone switch polys to DHCP option 66 and see boot server error even thou it seems to be working ? |
| 17:45.04 | Assid | bkruse: trying ti from 1 asterisk box to another as well.. callerid number just doesnt show |
| 17:45.08 | Assid | only callerid name |
| 17:45.39 | new2345 | so, anyone have another good reference or can someone help me out...google doesn't seem to know much about asterisk |
| 17:46.10 | Assid | i keep getting asterisk@ip |
| 17:46.15 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:46.31 | Assid | am trying on my soft phone.. on polycom.. it comes up as 'asterisk' |
| 17:47.00 | l2cache | I manage over 550 polycoms, and its running smooth |
| 17:47.06 | Assid | is there a change in sip.conf or something that i am overlooking? |
| 17:47.15 | TheCompWiz | supjigatr... use full URL for boot server... i.e. tftp://myasteriskbox/ |
| 17:48.10 | l2cache | new2345: in your sip.conf do you have reinvites enabled? |
| 17:48.38 | bkruse | canreinvite=yes |
| 17:49.50 | new2345 | i have tried yes and nonat |
| 17:50.53 | l2cache | change the canreinvite=yes to no so the call must stay in the server path, its usually nat that causes one-way audio like that |
| 17:50.53 | supjigatr | TheCompWiz: I am using ftp:poly:poly@0.0.0.0 |
| 17:50.53 | l2cache | and that is on your carriers sip.conf entry? |
| 17:50.53 | l2cache | not the extension |
| 17:50.54 | supjigatr | TheCompWiz: It works no real errors in the log but the display indicated can't contact. |
| 17:50.54 | new2345 | do i need to restart after that change? |
| 17:50.56 | l2cache | no just do a sip reload |
| 17:51.12 | l2cache | asterisk -rx 'sip reload' at your linux command line |
| 17:51.31 | TheCompWiz | supjigatr... are you seriously trying to ftp into 0.0.0.0? |
| 17:51.34 | Glasswlkr | Ok now that the phone config stuff is out of the way, a more complex problem (from my viewpoint). I have multiple sites, and we will be deploying more sites soon, so we thought 3 digit extensions internally at each site, with a single digit prefix to identify the site. This way if user dials 105 they get extension 105, but if they dial 1105 they get 105 at site 1 and 2105 gets 105 at site 2. Is this possible using asterisk? and if so what should I consider |
| 17:51.49 | bkruse | TheCompWiz: haha |
| 17:52.03 | new2345 | thats it...thanks |
| 17:52.09 | l2cache | it works now? |
| 17:52.41 | new2345 | lol...probably was just that i didn't reload sip after making the change before |
| 17:52.53 | l2cache | now worries :) have a good one |
| 17:53.25 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqvl.cable.mindspring.com) |
| 17:53.38 | *** join/#asterisk RoyK (n=roy@cEE71BF51.dhcp.bluecom.no) |
| 17:54.06 | [TK]D-Fender | Glasswlkr: Very doable and a great reason to use * |
| 17:54.29 | supjigatr | TheCompWiz: No just showing the format. |
| 17:54.33 | Qwell[] | Glasswlkr: bad to start an exten with 1... and what happens when you get more than 8-9 sites? |
| 17:54.54 | TheCompWiz | should be ftp:// not ftp:user:pass@server |
| 17:54.57 | Glasswlkr | well the problem is we are migrating from a legacy pbx which had all extensions numbered 101 102 103 and so on |
| 17:55.16 | *** join/#asterisk jkimball4 (n=jerrid@ip24-252-32-248.om.om.cox.net) |
| 17:55.39 | Glasswlkr | which is a pain, and management is insisting we keep the old extensions active so we don't loose customer calls from customers who don't know the new extensions (and because the staff likely won't like having all their extensions changed) |
| 17:55.43 | *** join/#asterisk friedrich| (n=friedric@e177240122.adsl.alicedsl.de) |
| 17:56.02 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 17:56.11 | supjigatr | ftp://polycom:****@192.168.60.252 |
| 17:56.27 | Glasswlkr | Qwell: well I can go with a 2 digit site-prefix |
| 17:56.36 | TheCompWiz | supjigatr... works? or dosn't work? |
| 17:56.44 | supjigatr | TheCompWiz: What is that error telling me? I can see it downloading the file and yes the phone works. |
| 17:56.46 | Qwell[] | Glasswlkr: why not just start with 4 digit extensions, and not even have a site id? |
| 17:57.01 | Glasswlkr | backwards compatibility |
| 17:57.03 | supjigatr | TheCompWiz: But I still get the annoying error. |
| 17:57.04 | TheCompWiz | supjigatr... you havn't told me about any error. |
| 17:57.12 | TheCompWiz | what is the error text? |
| 17:57.21 | jkimball4 | What is the current name for the variable returned by GetVar for a channel's callerid? |
| 17:57.24 | Glasswlkr | clients and staff want their existing extensions to continue to work on the new system |
| 17:57.39 | Glasswlkr | but they ALSO want the ability to scale... so the site-prefix was my solution to that |
| 17:58.24 | Glasswlkr | more accurately staff don't want to change extensions, and management is concerned with clients who direct dial salesteam by their current extensions, they just want those extensions to keep working to avoid customer headaches |
| 17:58.27 | TheCompWiz | Glasswlkr.. you can setup several options to do that. setup an outbound route that matches 1XXX & directs outbound path to be pbx at site 1... 2xxx to site 2... etc... and on the other end strip off the preceeding digit. |
| 17:59.03 | supjigatr | TheCompWiz: Cannont contact bootserver on the poly display. |
| 17:59.28 | TheCompWiz | supjigatr... typically that means the FTP failed for whatever reason. |
| 17:59.53 | TheCompWiz | i.e. sip.cfg is not present, boot files are missing, invalid username/password... long list of reasons. |
| 17:59.57 | supjigatr | TheCompWiz: Should it be in the app-log file? |
| 17:59.57 | Glasswlkr | supjigatr: what do the logs on your ftp server say? can you tail the logs to confirm the connection and what is happening on the serverside |
| 18:00.00 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 18:00.09 | supjigatr | Yep. |
| 18:00.12 | TheCompWiz | supjigatr not for boot stuff no. |
| 18:00.20 | TheCompWiz | should be in the boot-log |
| 18:00.59 | supjigatr | No boot log is left. |
| 18:01.16 | TheCompWiz | supjigatr... then your boot rom could not connect to the ftp server. |
| 18:01.34 | Glasswlkr | supjigatr: what about your ftp daemon logs? that should tell you if the connection was attempted and why it failed |
| 18:01.37 | TheCompWiz | hence... the error |
| 18:01.39 | Glasswlkr | (on the boot server itself) |
| 18:02.03 | supjigatr | I don't see any errors. It tries first to download the bootrom.ld but its the same version. |
| 18:02.09 | Corydon-w | TheCompWiz: there's a huge advantage to using FTP, though... timestamps |
| 18:02.13 | supjigatr | I don't see any FTP errors. |
| 18:02.15 | jkimball4 | AMI GetVar variable for callerid is? |
| 18:02.34 | TheCompWiz | Corydon-w... how is it any different? my tftp logs timestamp reboots & such... |
| 18:02.48 | TheCompWiz | supjigatr... where are you looking for errors? |
| 18:02.56 | Glasswlkr | the only advantage to ftp/http/https is encryption and/or authentication |
| 18:03.00 | supjigatr | On the FTP server. Ethereal |
| 18:03.03 | Corydon-w | TheCompWiz: if the timestamp has not changed, Polycom doesn't bother with redownloading the file |
| 18:03.42 | TheCompWiz | supjigatr... with ethereal... are you running it ON the boot server? |
| 18:03.57 | Corydon-w | That's a major advantage, because you don't waste time or bandwidth redownloading something that you already have on the phone filesystem |
| 18:04.01 | wwalker | is comfort noise (RFC 3389) supported in 1.4.1? |
| 18:04.17 | badcfe | the ${CDR(billsec)} gives me 0 for a call with comm. i do it in a exten => h,1,UserEvent( |
| 18:04.31 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 18:04.49 | Glasswlkr | Corydon76-home: I can see time... but bandwidth? :) these aren't big files, and on modern infrastructure that shouldn't be an issue. But yes, time is helpful when the phone shaves like a minute off it's boot time :) |
| 18:04.57 | *** part/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 18:04.59 | Corydon-w | badcfe: that's because the call is not yet concluded |
| 18:05.23 | Glasswlkr | but still that doesn't apply to all phones, if you have a mixed vendor environment you are much better off going with tftp for simple compatibility |
| 18:05.31 | sandorp | I am looking into implementing asterisk; I just finished installing trixbox and have configured 2 extensions -- 6000 and 6001; I have installed x-lite on 2 PCs on the same LAN; they both appear to be logged into the asterisk machine (no more 40x errors); how do I get the 2 extensions to call each other? did I miss the docs somewhere? |
| 18:05.31 | wwalker | I get horrible audio because my provider has RFC 3389 turned on and I'm at 1.2.14. is there any way for me to disable it in asterisk? I only see references to disabling it in the client (and my "client" is the PSTN) |
| 18:05.32 | Corydon-w | Glasswlkr: Nothing says that the server from which Polycom downloads its configs has to be on the LAN |
| 18:05.32 | badcfe | Corydon-w: do you know how i could either conclude it or get the billsec till then otherhow |
| 18:05.36 | supjigatr | TheCompWiz: Yep |
| 18:05.59 | Corydon-w | badcfe: PostCDR |
| 18:06.33 | Corydon-w | Glasswlkr: I have an installation where the configuration server is on the other side of a direct T1 |
| 18:06.47 | badcfe | Corydon-w: what kind of thing is that? |
| 18:06.56 | Corydon-w | badcfe: application |
| 18:07.29 | codefreeze | sandorp: Let's see. In sip.conf, you specified a context for those two phones, hopefully the same context, right! |
| 18:07.44 | codefreeze | sandorp: i meant right? |
| 18:08.02 | badcfe | Corydon-w: core show application PostCDR --> Your application(s) is (are) not registered |
| 18:08.18 | [TK]D-Fender | sandorp: .... |
| 18:08.24 | [TK]D-Fender | ~trixbox |
| 18:08.28 | jbot | trixbox is, like, unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 18:09.01 | sandorp | I actually didn't edit an sip.conf |
| 18:09.10 | sandorp | unless the web GUI did that for me |
| 18:09.52 | Corydon-w | badcfe: well, you could always do the calculation yourself |
| 18:09.54 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:10.08 | codefreeze | sandorp: ah, you'll have to follow it there, but basically, hopefully, the principle is the same. Whatever context your sip.conf entries use, that's the one with the extensions. You'll have to figure it out in the trixbox world. |
| 18:10.09 | sandorp | is sip.conf on the asterisk machine or on the client PCs? |
| 18:10.35 | codefreeze | sandorp: its on the asterisk machine. |
| 18:10.40 | sandorp | ok |
| 18:11.04 | [TK]D-Fender | sandorp: Your problem is a lack of understanding on how to use FreePBX (which is a key part of Trixbox. This is NOT supported here. Please try in their support channel : #freepbx |
| 18:11.11 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 18:11.30 | sandorp | ok, I will try #freepbx |
| 18:11.37 | *** join/#asterisk kraypius (n=kumar@72.171.136.205) |
| 18:11.39 | sandorp | thanks for the guidance |
| 18:12.13 | badcfe | Corydon-w: how? |
| 18:12.27 | Glasswlkr | Where does asterisk store voicemail recordings? |
| 18:12.40 | badcfe | Corydon-w: can i get that postcdr application registered? is it in a module? |
| 18:12.56 | Glasswlkr | I want to map all recordings to another drive, which I have mounted as /storage. So I want to symlink in all the recording directories from there |
| 18:13.07 | badcfe | Corydon-w: or is there some other variables that i could use to calculate the actual comm time? |
| 18:13.15 | Corydon-w | badcfe: get the EPOCH and subtract CDR(start) |
| 18:13.19 | kraypius | I have never installed asterisk before and im about to attempt it on my linux server. do I have to install asterisk and freepbx as root? |
| 18:13.21 | Glasswlkr | (I allready know about /var/lib/asterisk/sounds/custom which is for system recordings such as IVR but where does voicemail go?) |
| 18:13.43 | badcfe | Corydon-w: ill try that substraction then.. |
| 18:13.48 | [TK]D-Fender | Glasswlkr: typically /var/spool/asterisk/voicemail |
| 18:14.19 | badcfe | Corydon-w: by the way, does this CDR(billsec) contain something other than 0 in any dialplan case? |
| 18:14.19 | *** part/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 18:14.29 | Corydon-w | badcfe: nope |
| 18:14.31 | [TK]D-Fender | ~freepbx |
| 18:14.33 | jbot | extra, extra, read all about it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:19.57 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 18:21.53 | anonymouz666 | dammit |
| 18:22.17 | anonymouz666 | I have a wct4xxp loading automatically when I restart the system... but I have only zaptel and ztdummy calls |
| 18:22.27 | anonymouz666 | who is loading this than damn module? |
| 18:23.30 | mafkees | udev ? |
| 18:23.34 | mafkees | hotplug |
| 18:24.18 | anonymouz666 | there is zaptel.rules inside udev |
| 18:24.26 | anonymouz666 | what if i remove this file |
| 18:24.56 | mafkees | things will break ? |
| 18:25.01 | mafkees | no idea, never tried it |
| 18:25.34 | badcfe | Corydon-w: doesnt work either. CDR(start) gives the moment when the incoming call was received, not when the outgoing call was asnwered. |
| 18:25.54 | badcfe | s/asnered/answered/ |
| 18:26.03 | Corydon-w | Use CDR(answer) then |
| 18:26.27 | badcfe | oh |
| 18:28.11 | anonymouz666 | still load |
| 18:28.16 | anonymouz666 | grrr |
| 18:28.16 | anonymouz666 | heheh |
| 18:28.17 | anonymouz666 | :D |
| 18:28.35 | *** part/#asterisk SkramX (n=mark@HERCULES.sentiensystems.net) |
| 18:30.21 | dual-man | do i need mailserver installed on the asterisk box to have it send voicemail as email? |
| 18:33.04 | BrianR___ | [TK]D-Fender: Only problem with using ami originate is that the calls show up as "From:" in on the phone, in the wrong call history, and redial does the wrong thing. |
| 18:34.11 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 18:34.39 | mafkees | dual-man: no, the sendmail binary will be enough |
| 18:35.13 | [TK]D-Fender | BrianR___: Exacly like it should! ;) |
| 18:35.27 | [TK]D-Fender | BrianR___: Imperfect solutions for an imperfect world... |
| 18:35.34 | BrianR___ | [TK]D-Fender: true... |
| 18:35.56 | BrianR___ | [TK]D-Fender: I wonder if there's an API hidden somewhere in the phone itself for implementing click-to-dial |
| 18:36.12 | [TK]D-Fender | BrianR___: Nothing documented.... |
| 18:36.25 | tzafrir_laptop | anonymouz666, gerp zap /etc/udev/rules.d/* |
| 18:36.39 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 18:36.46 | BrianR___ | [TK]D-Fender: My one big gripe about the polycoms - not enough documentation :( |
| 18:36.50 | tzafrir_laptop | FC6 adds its own zaptel udev rules . You don't really need the ines installed by zaptel |
| 18:36.59 | jkimball4 | What is the AMI variable for callerid that used with GetVar? |
| 18:37.23 | [TK]D-Fender | BrianR___: No, right now I'd put this in the "If you can't find it document, thats because it doesn't EXIST" category... |
| 18:37.48 | Corydon-w | CALLERID(all) |
| 18:38.35 | anonymouz666 | tzafrir_laptop: 50-udev.rules:KERNEL=="zap[0-9]*" |
| 18:38.59 | anonymouz666 | I comment all these lines and the module wct4xxp still loads at startup |
| 18:39.06 | anonymouz666 | I don't know who is calling this module |
| 18:39.12 | tzafrir_laptop | ah, this is unrelated |
| 18:39.12 | bkruse | jkimball4: looks in docs/ |
| 18:39.25 | bkruse | of your asterisk source, names all the vars. |
| 18:39.44 | tzafrir_laptop | the module is loaded by automatic "hotplugging" at startup. Look for modprobe in /etc/rc.sysinit . |
| 18:39.49 | anonymouz666 | tzafrir: I have only modprobe zaptel and modprobe ztdummy... but wct4xxp loads automatically |
| 18:39.54 | tzafrir_laptop | The zaptel modules are part of the "others" |
| 18:40.11 | bkruse | grep -r "wct4xxp" * |
| 18:40.29 | tzafrir_laptop | If you want to disable the modprobe of a specific module, just blacklist it |
| 18:40.44 | tzafrir_laptop | put the line: blacklist wct4xxp |
| 18:40.54 | tzafrir_laptop | in /etc/modprobe.d/somefile |
| 18:41.15 | anonymouz666 | thanks! |
| 18:41.47 | tzafrir_laptop | and you don't really need a special modprobe of zaptel . 'modprobe ztdummy' loads zaptel |
| 18:42.01 | [TK]D-Fender | anonymouz666: Go kill off the KO |
| 18:42.11 | tzafrir_laptop | If it's not, you probaly have a broken modprobe configuration that runs ztcfg needlessly |
| 18:43.24 | tzafrir_laptop | but why do you want to use ztdummy when you have an actual hardware card? |
| 18:43.43 | tzafrir_laptop | you can get timing from the card even if it's not configured in Asterisk |
| 18:43.52 | *** join/#asterisk Assid (n=assid@59.183.35.202) |
| 18:43.57 | Assid | finally |
| 18:44.54 | anonymouz666 | it isnt necessary ? |
| 18:45.13 | dual-man | mafkees: is there any other special thing i need to do? |
| 18:45.30 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:45.38 | mafkees | dual-man: depends on the setup |
| 18:45.48 | tzafrir_laptop | anonymouz666, if zttest shows that you have a valid timing source, it is good enough |
| 18:45.51 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 18:46.19 | heison | [TK]D-Fender: for office use, would u recommend IP-601 or IP-501? |
| 18:46.30 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:46.51 | anonymouz666 | tzafrir_laptop: thanks again |
| 18:47.13 | [TK]D-Fender | heison: Technically the IP 601 is a beter phone, but the IP 501 is more than enough for just about anybody short of a receptionist |
| 18:47.13 | *** join/#asterisk lorinc (n=ang@pool-7161.adsl.interware.hu) |
| 18:47.25 | [TK]D-Fender | heison: Do you have PoE there? |
| 18:47.59 | dual-man | mafkees: what part of the setup |
| 18:48.02 | Assid | okay instead of caller id number .. asterisk is sending asterisk@ip to the phone |
| 18:48.07 | dual-man | it's just an asterisk box with voicemail |
| 18:48.34 | mafkees | dual-man: wether the network allows any machine to send mail to the big-bad-internet |
| 18:48.37 | heison | [TK]D-Fender: currently we don't |
| 18:48.50 | heison | [TK]D-Fender: would IP601 / IP501 work with injectors? |
| 18:48.53 | mafkees | in my networks, every machine has to be configured to use the local mailrelay box |
| 18:48.56 | [TK]D-Fender | heison: That in mind, yeah, go with the IP 501 and save the $ |
| 18:49.19 | dual-man | ok, have a local mail server, must i bounce the mail off it? |
| 18:49.21 | BrianR___ | The 601's speakerphone is much better than the 501's.. |
| 18:49.26 | [TK]D-Fender | heison: WEre you already running PoE or imminently planning to I'd suggest the IP 430 instead. |
| 18:49.29 | heison | [TK]D-Fender: so i take it the 601 only works with PoE? |
| 18:49.35 | [TK]D-Fender | BrianR___: Not so much I find.... |
| 18:49.38 | mafkees | dual-man: it's not a must, but it's better |
| 18:50.04 | BrianR___ | Aparently the microbrowser pukes on href= arguments that contain &'s... |
| 18:50.05 | [TK]D-Fender | heison: 601, 430 = both, 301/501 = either, but requres a speical cable at added cost |
| 18:50.13 | dual-man | but that is just a sendmail configuration, not asterisk right? |
| 18:50.16 | Assid | dont you need an poe injector as well? |
| 18:50.24 | mafkees | BrianR___: of course, & shoulb be & |
| 18:50.30 | BrianR___ | the 301/501 can be ordered with the PoE cable _instead_ of the wall wart.. |
| 18:50.31 | *** join/#asterisk gatuno (n=gatuno@82.158.212.230) |
| 18:50.43 | Qwell[] | "PoE cable"? heh |
| 18:51.02 | heison | [TK]D-Fender: special cable? |
| 18:51.28 | [TK]D-Fender | heison: Special cable with 802.3af negociation IC inline with it. |
| 18:51.43 | BrianR___ | Qwell[]: The PoE DC->DC converter stuff is in a lump on the cable. They sold it in twothree different versions, one for IEEE PoE, one for Cisco, and one with a barrel connector for a wall wart. |
| 18:51.50 | [TK]D-Fender | heison: kludgy solution |
| 18:51.52 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 18:51.54 | Qwell[] | a DC-DC converter? what? |
| 18:52.18 | heison | can i use a PoE injector with any of the polycom's? |
| 18:52.32 | MrTelephone | how do you set the callerid and number on outgoing calls on a pri? I tried using exten => s,1,set(CALLERID(name)=??) but s only starts on incoming calls? |
| 18:52.51 | BrianR___ | Qwell[]: The cable converts the PoE voltage to the 12v that the phone expects. |
| 18:53.08 | Qwell[] | why didn't they just put that...in...the phone? |
| 18:53.23 | Qwell[] | ciscos work ;) |
| 18:53.25 | BrianR___ | Trying to keep costs down on the low end models... |
| 18:53.34 | BrianR___ | The 301 is under $100 in bulk. |
| 18:53.55 | BrianR___ | The 430, IMO, is a better compromise. |
| 18:53.59 | Assid | how much is a injector anwyasy |
| 18:54.23 | BrianR___ | The PoE stuff is built in on the 430 and the 301 has no microbrowser. |
| 18:54.33 | dual-man | is asterisk compatible with postifx? |
| 18:54.36 | dual-man | postfix? |
| 18:54.46 | Qwell[] | dual-man: is postfix compatible with sendmail? |
| 18:54.53 | florz | dual-man: No, you must use postfisk |
| 18:54.55 | heison | Assid: 3com PoE injector is around $25 |
| 18:55.36 | dual-man | is postfix compatible with sendmail? |
| 18:56.08 | Assid | not bad.. how many phones can that support ? |
| 18:56.13 | dual-man | i've never used anything other that postfix |
| 18:56.26 | heison | Assid: one per injector |
| 18:56.31 | Assid | i see |
| 18:56.38 | Assid | rather keep power cable :| |
| 18:57.21 | heison | Assid: i use them at home coz i don't want a UPS beeping in my bedroom in the middle of the night |
| 18:57.29 | *** join/#asterisk solar_ant (n=solar@122.164.144.85) |
| 18:57.34 | solar_ant | hey all |
| 18:57.41 | Assid | hrmm makes sense |
| 18:57.54 | Assid | you dont have non ups power sockets? |
| 18:58.20 | heison | of course i do, but then i'd loose phone service without power |
| 19:00.19 | bkruse | mine UPS, with my 400 watt speakers, lasts about 13 seconds when im rocking out |
| 19:00.40 | Assid | hehee |
| 19:00.54 | Assid | brb.. my dsl doesnt let me login to this sip service |
| 19:01.00 | Assid | gotta reboot it |
| 19:02.19 | bkruse | dsl not letting you login to a sip service? |
| 19:02.26 | bkruse | why does that sound so wierd and user errorish to me. |
| 19:02.34 | [TK]D-Fender | heison: You don't want to use injectors.... |
| 19:02.39 | tzafrir_laptop | just about any decent MTA is compatible enough with sendmail for Asterisk |
| 19:02.43 | bkruse | [TK]D-Fender: AGREED! |
| 19:03.21 | heison | [TK]D-Fender: you mean not with the polycom's? |
| 19:03.59 | [TK]D-Fender | heison: You are thinking BACKWARDS. |
| 19:04.11 | heison | what do u mean? |
| 19:04.15 | [TK]D-Fender | kjla;sdhjksdhkajlhjdsahjsdlajhhjsdasda |
| 19:04.19 | [TK]D-Fender | UGH |
| 19:04.25 | bkruse | lol |
| 19:04.32 | bkruse | ~lart heison |
| 19:04.48 | *** join/#asterisk Bouke (n=bouke@b-haarsma.demon.nl) |
| 19:04.50 | heison | i don't see any other way to provision a phone without putting a UPS in a bedroom - what do you suggest? |
| 19:04.56 | Bouke | hi all |
| 19:05.02 | BrianR___ | Even a soho poe switch is better than using injectors... |
| 19:05.17 | [TK]D-Fender | heison: Listen, the 301/501 don't do PoE natively, you need to pay extra for that special cable (30$ or so). By using a solitary injector you are waste even MORE money to basically get it plugged in through a wall-wart again! |
| 19:05.28 | *** join/#asterisk Assid (n=assid@59.183.52.39) |
| 19:05.31 | Assid | back |
| 19:05.33 | Assid | again |
| 19:06.09 | [TK]D-Fender | heison: And what the heck does "provisioning" have to do with "UPS"?! |
| 19:06.26 | Bouke | I've got a server with Sipcat on it; it uses Asterisk under the bonit. However, I'm having a problem with a trunk I've recently added. The trunk supports in- and outbound calls, but I only got inbound calls to work. Outbound calls won't work :(. |
| 19:06.27 | heison | BrianR___: which soho poe switch would u recommend? i need 12 ports |
| 19:06.29 | [TK]D-Fender | heison: Take a break and let the crack filter out of your system :) |
| 19:06.41 | Bouke | I'm new to Asterisk, so I might be doing something wrong I guess :P. |
| 19:06.42 | [TK]D-Fender | heison: D-Link DES-1526 |
| 19:06.48 | BrianR___ | Of course you can't put the Polycom cord lump in the wiring closet - the voltage between the lump and the phone is too low and any additional length of cable between them will make the phone unreliable. |
| 19:07.13 | heison | [TK]D-Fender: all phones need to be UPS powered |
| 19:07.36 | BrianR___ | If you don't already own 301/501's, buy 430's instead. |
| 19:07.37 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 19:07.51 | [TK]D-Fender | heison: And get a PoE Switch |
| 19:07.53 | BrianR___ | The 430 is good enough for almost every 501 application and not much more expensive than the 301. |
| 19:07.57 | [TK]D-Fender | heison: Like linked |
| 19:08.07 | Assid | i shoulda got the 430 then |
| 19:08.13 | [TK]D-Fender | The IP 430 is just great for your average user |
| 19:08.27 | JoNate | Hey guys, do I need to install or compile anything additional from asterisk to get a meetme confrence working? |
| 19:08.37 | heison | BrianR___: i don't have any polycom yet... only Cisco's phones with Cisco 3548 and a bunch of injectors |
| 19:08.44 | [TK]D-Fender | JoNate: Zaptel. |
| 19:08.45 | Assid | JoNate: you need zaptel |
| 19:08.49 | JoNate | okaly dokaly |
| 19:08.53 | JoNate | thanks! |
| 19:09.02 | Bouke | ^^ could somebody help me with my problem? |
| 19:09.17 | JoNate | But there isn't an actual MeetMe app right? I mean it's in asterisk already right? |
| 19:09.32 | BrianR___ | Also, don't introduce a totally different type of phone without good cause. It'll just make provisioning a headache. |
| 19:09.57 | *** join/#asterisk apardo (n=apardo@87.217.145.129) |
| 19:10.34 | [TK]D-Fender | JoNate: Correct. |
| 19:11.02 | [TK]D-Fender | JoNate: There IS an App and it IS included in *, just not compiled in unless Zaptel is built first |
| 19:11.22 | heison | BrianR___: i was looking at the Polycom's because [TK]D-Fender keeps telling how good they are - and i also want backlight which the Cisco's don't have |
| 19:11.37 | [TK]D-Fender | BrianR___: Hardly a nightmare, more like 1 extra minute spend on the template. |
| 19:11.59 | [TK]D-Fender | heison: The only Polycom with a backlight is the IP 650 right now |
| 19:12.07 | JoNate | D-Fender: Thanks! I've been trying to learn * for about a month now and I'm still lost! |
| 19:12.22 | BrianR___ | only the ip650 and ip4000 have a backlight... |
| 19:12.51 | [TK]D-Fender | heison: If you insist on a backlight and PoE, get Aastra 480i's instead. |
| 19:13.45 | heison | gee... i'll need to look at another brand? sigh |
| 19:14.41 | [TK]D-Fender | heison: There is no "perfect" phone without paying through the nose (which typically violates "perfect" for most) |
| 19:15.03 | [TK]D-Fender | heison: If you want it all, prepare to pay, or aim for a different middle-ground |
| 19:15.27 | heison | i'm going to stick with Cisco + lamp :) |
| 19:15.51 | heison | until i can find a 1000Mbps PoE switch for $400 |
| 19:17.54 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 19:19.58 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 19:20.19 | Dr-Linux | i don't know the current password of this cisco 7960 phone, how can i reset the password? |
| 19:20.44 | [TK]D-Fender | heison: Why exactly do you ned gbit + poe? |
| 19:20.46 | ManxPower | Dr-Linux: the cisco web site has info on doing a factory reset on all their products |
| 19:20.58 | toombaloomba | Dr-Linux: what version of firmware? |
| 19:21.00 | Qwell[] | I thought poe+gbit wasn't possible? |
| 19:21.06 | Qwell[] | since both used all 8 wires |
| 19:21.10 | Dr-Linux | sip 7.4 |
| 19:21.11 | [TK]D-Fender | heison: and if you don't need backlight, then our suggestion for the IP 430 stands |
| 19:21.27 | *** part/#asterisk Bouke (n=bouke@b-haarsma.demon.nl) |
| 19:21.29 | Dr-Linux | toombaloomba: 7.4 |
| 19:21.29 | [TK]D-Fender | Qwell[]: the highest-end ciscos can do it. |
| 19:21.32 | kraypius | I ran the configure file and its telling me error: termcap support not found... but termcap IS installed |
| 19:21.33 | heison | [TK]D-Fender: thx man, i'll order one and try |
| 19:21.36 | Qwell[] | eh, how? |
| 19:21.42 | [TK]D-Fender | Qwell[]: No doubt only with certain Cisco switches |
| 19:21.46 | Qwell[] | some cdp funkiness? |
| 19:22.04 | Dr-Linux | ManxPower: yeah, i got that, but that would have much easy if i get to change the current password |
| 19:22.05 | heison | Qwell: depending on which pairs is supplying voltage |
| 19:22.09 | Qwell[] | I know - they have a cap that stores juice |
| 19:22.16 | [TK]D-Fender | Qwell[]: Being Cisco you know I don't have any SERIOUS details ;) |
| 19:22.20 | Qwell[] | when a gbit packet comes in, they remove juice :P |
| 19:22.33 | toombaloomba | Dr-Linux: unpluig from power, hold pound, plug power back in, when asked for enter key sequence enter 123456789*0# then it will ask u if u want to save config or not |
| 19:22.37 | kraypius | anyone know why i might be having this problem? |
| 19:22.41 | toombaloomba | Dr-Linux that will reset it to factory defaults, blank config |
| 19:23.02 | [TK]D-Fender | Qwell[]: Cisco's are no longer AC or DC, but now per your claims OC (Occasional Current) ;) |
| 19:23.08 | *** join/#asterisk Maroderr (n=drago@fanatici.net) |
| 19:23.11 | Qwell[] | ;) |
| 19:23.13 | Maroderr | hello |
| 19:23.19 | Qwell[] | AGBIT |
| 19:23.24 | Qwell[] | Alternating gbit |
| 19:23.24 | Maroderr | i have a question |
| 19:23.39 | *** join/#asterisk Malph (n=chatzill@66-231-0-194.hosts.sdnet.net) |
| 19:23.41 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 19:23.44 | PupenoR | Hello. |
| 19:23.50 | Dr-Linux | toombaloomba: yeah, i got that on a web, but SIP firmware will be still there? or ... |
| 19:24.03 | Maroderr | how i can know the who is the called number in s extension |
| 19:24.04 | [TK]D-Fender | Qwell[]: BARF : Bricked And Returned Frequently <- |
| 19:24.09 | Maroderr | ? |
| 19:24.35 | [TK]D-Fender | Maroderr: "s" is the exten, and there is no more known about "who" was called. |
| 19:24.52 | [TK]D-Fender | Maroderr: Go read up on your "standard extensions" on the WIKI |
| 19:24.53 | [TK]D-Fender | ~wikis |
| 19:25.03 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 19:25.12 | Maroderr | [TK]D-Fender i read ... |
| 19:25.19 | PupenoR | If I have a IAX record of type user, I'll be able to accept connections with the specified user and secret. If I have a IAX record of type peer, then my asterisk will connect to other, right? |
| 19:25.27 | Maroderr | ok see |
| 19:25.53 | Maroderr | i have around 100 numbers in one context |
| 19:26.20 | Maroderr | but i i dont want to make 100 exten for eache number |
| 19:27.02 | Maroderr | and i try exten => s,1,set(foo=${CALLERID(dnid)}) |
| 19:27.17 | Maroderr | exten => s,n,dial(sip/foo) |
| 19:27.25 | Maroderr | understand me ? |
| 19:28.04 | [TK]D-Fender | Maroderr: Calls don't just land on "s" you know.... |
| 19:28.17 | [TK]D-Fender | Maroderr: If its searching for a number you need to use a pattern match |
| 19:28.44 | Maroderr | [TK]D-Fender but i can't get dnid number ... |
| 19:28.55 | [TK]D-Fender | Maroderr: What are the calls coming in on? |
| 19:29.14 | Maroderr | from h323 provider |
| 19:29.45 | toombaloomba | Dr-Linux yea sip firmware will be there it will just reset all settings but firmware stays the same |
| 19:29.56 | BrianR___ | heison: No need for gig on your phone lan anyway... |
| 19:31.13 | [TK]D-Fender | Maroderr: Sorry, can't help you there, but if the calls are landing on "S" thats because they aren't sending in the targeted DID |
| 19:31.19 | Maroderr | Start. Used primarily for dialplans that enter a context with no other extension information. Think of a non DID phone line, call comes in, and we may only know that the line is ringing and nothing else. |
| 19:31.20 | Maroderr | shit |
| 19:31.24 | Maroderr | from voip-info |
| 19:31.25 | Dr-Linux | toombaloomba: should i save config or not? |
| 19:31.50 | Maroderr | [TK]D-Fender have any other solution ? |
| 19:31.52 | [TK]D-Fender | Maroderr: Yup.. thats the BIG PRINT version... |
| 19:32.07 | [TK]D-Fender | Maroderr: Have your provider dial in with an exten matching hte DID |
| 19:32.18 | [TK]D-Fender | Maroderr: Otherwise its 100 #'s that all lead to the same place. |
| 19:32.29 | Maroderr | blah |
| 19:34.08 | Maroderr | noop(${CHANNLE}) output is : NoOp("H323/ip$217.x.x.x:4114/20787", "H323/ip$217.x.x.x:4114/20787") |
| 19:35.32 | toombaloomba | Dr-Linux that will save network config, if you want to, but theres no point thats easy to enter again |
| 19:35.51 | [TK]D-Fender | Maroderr: Consider a better provider |
| 19:36.20 | Maroderr | :)) |
| 19:38.48 | *** join/#asterisk bkw__ (i=brian@adsl-70-143-50-36.dsl.tul2ok.sbcglobal.net) |
| 19:42.09 | heison | BrianR___: no, but i don't want to maintain 2 switches... which is what i have now |
| 19:42.50 | heison | 3548 + a cheap ass gig switch |
| 19:43.03 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-071-077-048-198.nc.res.rr.com) |
| 19:45.56 | MrTelephone | if you put a wildcard extension in like _X. will asterisk look at exten 503 if you dial 503 or will it goto _X.? |
| 19:46.07 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:46.18 | [TK]D-Fender | MrTelephone: Go try |
| 19:46.46 | MrTelephone | well I'm trying to set callerid on outgoing but I need to have a set callerid run on every outgoing call except local extensions... |
| 19:47.22 | *** join/#asterisk bulle (n=bulle@c-db2971d5.015-48-626c671.cust.bredbandsbolaget.se) |
| 19:47.46 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 19:47.51 | MrTelephone | exten => _NXXXXXX,n,Set(CALLERID(num)=${IF( ${REGEX("^50[0-9]" ${CALLERID(num)})} ?2295555:${CALLERID(num)} )}) |
| 19:47.55 | MrTelephone | using something like that |
| 19:48.33 | MrTelephone | oh well hopefully I can figure it out |
| 19:49.33 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 19:54.41 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 19:57.16 | Maroderr | [TK]D-Fender i found solution :) |
| 20:00.15 | MrTelephone | in the asterisk dialplan can you set callerid and then have the call redialed as if your calling from the phone again? that way asterisk will search all the extenions in the dialplan after having the new callerid? |
| 20:01.23 | MrTelephone | this is difficult geez |
| 20:01.29 | ManxPower | Of the destination matches _NXXXXXX then the call CANNOT be going to an extension, right? |
| 20:01.52 | ManxPower | so just set the callerid for all exten's that are for outgoing calls. |
| 20:03.58 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 20:04.01 | MrTelephone | right |
| 20:04.23 | ManxPower | so whats the problem |
| 20:04.52 | MrTelephone | because i have a conext called [sipcustomers] with extensions like 2290000 and 2301111 |
| 20:05.33 | MrTelephone | and if the outbound call matches _NXXXXXX then how do I get it to goto 2290000 if someone calls that |
| 20:05.45 | MrTelephone | without having setcallerid on every unique extension |
| 20:06.22 | MrTelephone | my brain logic is not functioning properly |
| 20:06.29 | ManxPower | Your phone extensions are all 7-digits long? |
| 20:06.57 | ManxPower | why not just callerid= in sip.conf? |
| 20:06.59 | MrTelephone | up here we only use 7digits for local calls at this moment |
| 20:07.25 | ManxPower | Well THAT was poor planning. using 7 digits for extensions and 7 digits for local calls |
| 20:07.27 | MrTelephone | because Im trying to use the asterisk box as the office pbx at the same time as a sip provider |
| 20:07.45 | MrTelephone | my extensions are 501-510 |
| 20:07.57 | MrTelephone | but some 7 digits go out over zap and some are sip |
| 20:08.16 | ManxPower | Your design makes things horrribly complicated. |
| 20:08.27 | ManxPower | I can't really help you fix a badly designed system. |
| 20:08.37 | MrTelephone | yeah I think its complicated because I set the customers SIP number to the same as the DID they are using |
| 20:08.44 | MrTelephone | i'll have to change that |
| 20:08.57 | ManxPower | and do your company extensions all have a DID? |
| 20:09.04 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 20:09.06 | MrTelephone | one single DID |
| 20:09.27 | *** join/#asterisk sherif (n=sherif@unaffiliated/sherif) |
| 20:09.43 | MrTelephone | but I want internal calls to have personal extensions but outgoing calls to the sip/zap customers to have the did callerid and not our office extensions |
| 20:10.33 | MrTelephone | It seems hard to do |
| 20:11.01 | MrTelephone | the thing is if I could just say _X.,1,Macro(outgoing calls) |
| 20:12.10 | MrTelephone | I guess I could then have it dial the number.. but I want it to dial using another context's extenensions |
| 20:12.18 | MrTelephone | im confusing myself and everyone else |
| 20:12.19 | MrTelephone | haha |
| 20:12.42 | MrTelephone | must sleep on it or something |
| 20:14.41 | badcfe | quit "sleepin" |
| 20:14.49 | Vm4Ever | 16 0 24528 11m 4956 S 61 1.2 156:22.40 asterisk |
| 20:15.01 | Vm4Ever | anyreason its using high loads.. 0.50 .. 5 calls |
| 20:15.07 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 20:15.09 | mafkees | transcoding ? |
| 20:15.10 | *** part/#asterisk rene- (n=rene-@200.34.66.137) |
| 20:15.36 | Vm4Ever | hmm yeah |
| 20:15.49 | MrTelephone | if there was a way to run a command without actually needing to dial an extension that would be spectacular |
| 20:15.49 | mafkees | there you are :) |
| 20:16.18 | mafkees | MrTelephone: what you want to do ? |
| 20:17.03 | *** part/#asterisk test34- (n=test34@unaffiliated/test34) |
| 20:17.06 | Vm4Ever | well2 g729 trasncodes on 5 calls total should not do that |
| 20:17.12 | Vm4Ever | touhg asterisk could handle 100 calls |
| 20:17.16 | Vm4Ever | at this rate it wont |
| 20:17.28 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 20:18.21 | Vm4Ever | could the g729 be faulty i assume yes |
| 20:18.40 | mafkees | g729 transcoding is cpu hungry |
| 20:18.50 | Vm4Ever | now 1 calls g729 anf 0.60 load |
| 20:19.09 | Vm4Ever | 24496 11m 4956 S 72 1.2 157:52.60 asterisk |
| 20:19.13 | Vm4Ever | 73% cpu hmm |
| 20:19.35 | ManxPower | Vm4Ever: what are you running this on a Pentium 100Mhz? |
| 20:20.35 | Vm4Ever | DUAL Xeons 3.6 2 gig ram |
| 20:20.46 | Vm4Ever | theres few agis |
| 20:20.55 | ManxPower | MrTelephone: Hint: The solution to your problem is by using contexts |
| 20:21.32 | Vm4Ever | anyway to know waht is causing htis ? like if its trx or AGI? |
| 20:22.08 | ManxPower | AGIs could do it, expecially if you are launching them often |
| 20:22.12 | MrTelephone | If there is a _X. match then if I jump to another context it will then try and rematch? |
| 20:22.16 | Vm4Ever | yeah ok that it.. |
| 20:22.35 | mafkees | MrTelephone: yeah |
| 20:22.36 | Vm4Ever | s fast agi could be less heavy maybe |
| 20:22.40 | ManxPower | MrTelephone: yes |
| 20:22.45 | heison | does any know what may be the cause of this? Mar 9 15:20:12 WARNING[4398]: chan_sip.c:6722 get_rdnis: Huh? Not an RDNIS SIP header (tel:4163481500)? |
| 20:23.20 | heison | the call path is from a Nortel switch -- T1/NI2 --> Audiocodes M1000 --> Asterisk |
| 20:24.25 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 20:24.34 | ManxPower | heison: why are you not just doing Nortel switch -- T1/NI2 --> Asterisk |
| 20:25.19 | Vm4Ever | ah |
| 20:25.25 | Vm4Ever | hmm ok i think php agi is nasty |
| 20:25.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:26.33 | heison | ManxPower: coz i don't have any PRI interface on Asterisk |
| 20:27.04 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 20:28.17 | mafkees | Vm4Ever: depend what you are doing with it. but yeah, php can be nasty on cpu |
| 20:28.58 | Vm4Ever | ill rewrte it in C and see if that helps.. if not ill disbale it.. then ill see some more.. also got a macro that runs on inbound.. but hard to say WHAT is taking cpu |
| 20:32.15 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 20:39.48 | MrTelephone | manxpower, goto won't just jump to a context |
| 20:40.29 | mafkees | MrTelephone: goto(context|exten|priority) |
| 20:41.10 | MrTelephone | but then you have to jump to a specific exten and won't let asterisk pick the closest one |
| 20:41.13 | MrTelephone | :( |
| 20:41.23 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:41.35 | MrTelephone | _X.,1,SETcallerid info |
| 20:41.45 | MrTelephone | _X.,2,goto(context) |
| 20:41.50 | MrTelephone | [outgoing context] |
| 20:41.54 | mafkees | that wont work |
| 20:41.58 | MrTelephone | I know I tried |
| 20:42.02 | MrTelephone | damn |
| 20:42.09 | mafkees | you need context and exten and priority |
| 20:42.58 | MrTelephone | anyways i'll try again later |
| 20:43.04 | ManxPower | so perhaps you need Goto(outgoing-context,${EXTEN},1) |
| 20:43.27 | MrTelephone | possible |
| 20:43.35 | MrTelephone | I didn't think of the var part |
| 20:43.38 | MrTelephone | right on |
| 20:43.44 | MrTelephone | that might work? |
| 20:44.07 | mafkees | yeah |
| 20:44.11 | mafkees | I use it all the time |
| 20:50.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:52.39 | JacksLivr | is fwd up? i can't get registered |
| 20:56.29 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:59.54 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 21:00.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:03.01 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 21:03.03 | Mahmoud | hello guys |
| 21:03.15 | Mahmoud | any one noticed a bug in asterisk, when going on-hold on SIP channels? |
| 21:03.37 | Mahmoud | when the called party click on "on hold" key on his phone, the calling party hears music (fine) |
| 21:03.54 | Mahmoud | but once the called party removes "on hold", they are able to talk, but again, the music is being played while talking :/ |
| 21:04.14 | *** join/#asterisk vlt|home (n=daniel@dslb-088-073-244-030.pools.arcor-ip.net) |
| 21:05.35 | vlt|home | Hello. How can I UNregister from a sip server? I removed the "register => " line from sip.conf and reloaded. But when I call the number The call is still announced. Any idea? |
| 21:08.34 | *** part/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:09.05 | *** join/#asterisk eald (n=eald@189.157.105.23) |
| 21:10.15 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:10.39 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 21:11.03 | rhombus | yo homies! Can the "call park" feature on the Polycom Soundpoint phones be made to work with Asterisk call parking? |
| 21:11.26 | thekidrio | rhombus, you were a question on a horrible show lastnight, are you smarter then a 4th grader |
| 21:11.34 | *** join/#asterisk waberforce (n=dtt_vde@129-30-111-208-in-addr-arpa.omnispring.net) |
| 21:11.38 | thekidrio | an adult had no idea how many sides a rhombus had |
| 21:11.44 | thekidrio | made my stomach hurt to see that |
| 21:12.55 | rhombus | That is why I exist |
| 21:13.02 | rhombus | to increase awareness of the rhombus |
| 21:13.09 | rhombus | natures most ignored and maligned polygon |
| 21:13.26 | thekidrio | haha |
| 21:13.35 | Corydon-w | I thought the obtuse triangle was the most ignored |
| 21:13.35 | thekidrio | all my kites were rhombuses |
| 21:13.40 | thekidrio | rhombi? |
| 21:13.44 | rhombus | rhombi. |
| 21:13.44 | thekidrio | what is plural of rhombus |
| 21:13.49 | thekidrio | heh |
| 21:14.06 | thekidrio | rhombuses sounds funnier though |
| 21:14.09 | rhombus | I have long believed that the they should have made the Pentagon the Rhombus instead |
| 21:14.13 | JacksLivr | can anyone register to fwd? |
| 21:14.25 | thekidrio | JacksLivr: yes i was able to a week or so ago |
| 21:14.34 | thekidrio | it have not tried since though |
| 21:14.49 | wwalker | I get horrible audio because my provider has RFC 3389 turned on and I'm at 1.2.14. is there any way for me to disable it in asterisk? I only see references to disabling it in the client (and my "client" is the VoIP provider) |
| 21:14.53 | rhombus | Can you imagine Rummy standing in front of a shield that says "THE RHOMBUS WASHINGTON DC" |
| 21:15.03 | thekidrio | anyone have any good articles on asterisk security? |
| 21:15.20 | rhombus | anyway: can the Polycom call park feature be made to work with Asterisk call parking? Has anybody done it? |
| 21:15.23 | thekidrio | i want a building named the truncated dodecahedral |
| 21:15.49 | rhombus | but that's a three-dimensional polygon, it would be enormous |
| 21:16.02 | mafkees | MafCastle |
| 21:16.03 | mafkees | ;) |
| 21:17.32 | *** join/#asterisk rvhi3 (n=as@66.175.65.82) |
| 21:17.56 | rvhi3 | anyone uses T1 failover switch? |
| 21:18.02 | *** join/#asterisk ManxPower (n=manxpowe@226.sub-70-222-51.myvzw.com) |
| 21:18.11 | rvhi3 | i looked at http://www.voip-info.org/wiki/view/Failover+switches |
| 21:18.26 | rvhi3 | in t1 failover there are a few vendors, anyone has any experience with them? |
| 21:21.56 | *** join/#asterisk Schreiber1337 (i=d8a9b0b6@gateway/web/cgi-irc/ircatwork.com/x-3cf02b7449984ea9) |
| 21:28.02 | Mahmoud | hmmm |
| 21:28.11 | Mahmoud | when I press keys with SIP, is it sent over RTP? or over SIP? |
| 21:28.21 | Qwell[] | Mahmoud: depends what dtmfmode you're using |
| 21:28.38 | Mahmoud | i see |
| 21:29.32 | Mahmoud | hmm it's rfc2833 |
| 21:29.53 | Qwell[] | then it's over rtp, but not in the audio |
| 21:30.43 | Mahmoud | hmmm |
| 21:30.52 | Mahmoud | i found a way to by pass my ISP's deep inspectoin |
| 21:30.59 | Mahmoud | they block SIP based on packet sig |
| 21:31.13 | Mahmoud | i changed the "SIP/2.0" in the header, into "SXP/2.0" |
| 21:31.37 | Mahmoud | i modified chan_sip.c |
| 21:31.39 | Mahmoud | and it works |
| 21:31.46 | Mahmoud | but I failed to send numbers in menus |
| 21:31.58 | Mahmoud | "to check your voice mail, press 1" |
| 21:32.02 | Mahmoud | when I press 1, it's not sent |
| 21:37.04 | Mahmoud | eyebeam doesn't support "dtmfmode=info" ? |
| 21:39.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:42.21 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-184-73.ny325.east.verizon.net) |
| 21:42.28 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:49.11 | Ac1dcrawl | how do I loopback an interface? |
| 21:49.19 | Qwell[] | Ac1dcrawl: what type of interface? |
| 21:51.40 | *** join/#asterisk heison (n=heison@209.167.5.1) |
| 21:56.24 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 21:58.44 | *** join/#asterisk chema (n=chema@198.Red-88-7-6.staticIP.rima-tde.net) |
| 22:00.14 | data23 | hmm |
| 22:02.45 | JacksLivr | would someone be willing to test their fwd account for me? If you can't get through, then I will stop trying. |
| 22:02.50 | JacksLivr | im using iax |
| 22:03.07 | JacksLivr | IAX2/192.246.69.186:4569-2 is circuit-busy |
| 22:03.11 | JacksLivr | is what i am getting |
| 22:04.02 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 22:04.13 | Shaun2222 | ANybody used telepasific for t1/pri? |
| 22:07.59 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 22:09.10 | chrisknight | If I have *Now, & viatalk, do I have to forward any ports? If so, which ones & TCP or UDP? Thanks... |
| 22:10.29 | chrisknight | I read 5060 & 5061 (one for each channel) & 69. Then 10000-20000... is that right? |
| 22:10.36 | [TK]D-Fender | chrisknight: 5060, 10000-20000 all UDP |
| 22:11.08 | chrisknight | ok, does 5061 get opened for my 2nd channel? Not sure how this thing works. |
| 22:11.54 | [TK]D-Fender | chrisknight: No, * should be the only SIP device talking to the outside so only a single SIP port. |
| 22:13.09 | chrisknight | ok... So both channels use 5060, gotcha. Having a hard time setting this up... I can cal extension to extension though... |
| 22:14.09 | [TK]D-Fender | ~sipnat |
| 22:14.11 | jbot | sipnat is probably for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:14.20 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 22:14.51 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 22:15.00 | chrisknight | thanks |
| 22:18.27 | suma | asterisknow software is a linux distribution for i386 pc ? or a software to have a gui for asterisk ? |
| 22:18.45 | Qwell[] | both |
| 22:19.02 | suma | can i have it as an gui for my existing asterisk ? |
| 22:20.15 | thekidrio | what asterisk do you have? |
| 22:20.19 | thekidrio | i think it requires 1.4 |
| 22:20.21 | suma | 1.4 |
| 22:20.23 | thekidrio | not 100% on that |
| 22:20.26 | thekidrio | yeah you can install gui |
| 22:20.31 | thekidrio | aussie voip has intructions i am sure |
| 22:26.44 | chrisknight | Has anyone here set up viatalk with *Now... I'm having a hell of a time... |
| 22:31.24 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:31.27 | [TK]D-Fender | chrisknight: Perhaps you could actually describe the problem.... |
| 22:32.50 | mut | Its broken |
| 22:32.54 | mut | i mean gosh, what more do ya need |
| 22:33.27 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 22:33.40 | *** join/#asterisk jjshoe (n=jjshoe@adsl-75-14-241-209.dsl.irvnca.sbcglobal.net) |
| 22:33.45 | chrisknight | The problem is that im so new to this, I dont even know where to start. I am NOT new to linux. I have ext. configed in *Now... I can call extention to extention. I just dont know how to set up my new sip trunks, & auto attendant. (this is for my home) |
| 22:34.01 | jjshoe | has anyone setup voicemail such that you can press a character while leaving a voicemail to switch to checking that voicemail? |
| 22:34.31 | mut | jjshoe, you can press # while the recording is playing |
| 22:34.36 | mut | and it'll skip to the login |
| 22:34.40 | chrisknight | I did the custom Service Provider option in *Now |
| 22:34.43 | [TK]D-Fender | chrisknight: Well if you're starting from scratch, start here : |
| 22:34.45 | [TK]D-Fender | ~book |
| 22:34.46 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:34.46 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 22:34.51 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-94.lsanca.dsl-w.verizon.net) |
| 22:35.01 | [TK]D-Fender | chrisknight: Then from there you can move on to : |
| 22:35.03 | [TK]D-Fender | ~wikis |
| 22:35.04 | jbot | wikis is probably http://www.voip-info.org |
| 22:35.16 | dahunter3 | Can a digium TE110P take a strictly analog input? |
| 22:35.33 | [TK]D-Fender | dahunter3: No. It is a strictly DIGITAL card. |
| 22:36.23 | dahunter3 | [TK]D-Fender: Well, that's why I'm having problems ! Thanks :) |
| 22:36.41 | chrisknight | Thats general info... I need details. Maybe I should call tech support at viatalk |
| 22:37.36 | [TK]D-Fender | chrisknight: You've staed rather clearly that you are completely new to * and don't really even know the basics. thats the point. If you are expecting a hand-held solution, typically that'll mean hiring a consultant. |
| 22:37.50 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 22:38.27 | [TK]D-Fender | chrisknight: And you could always ask them, but * is rarely supported by these companies, and rarer still are those that could advise users of that specific distro & GUI package. |
| 22:39.23 | jjshoe | mut what version of asterisk? |
| 22:39.25 | jjshoe | mut 1.2 or 1.4? |
| 22:39.58 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 22:40.16 | chrisknight | The main reason I went with viatalk is because they answered "yes" when I asked if they supported *. If that means just pointing me to: |
| 22:40.17 | chrisknight | http://support.viatalk.com/index.php?_a=knowledgebase&_j=questiondetails&_i=123&nav=+%26gt%3B+%3Ca+href%3D%27index.php%3F_a%3Dknowledgebase%26_j%3Dsubcat%26_i%3D42%27%3EInstallation+Guides%2FTech%3C%2Fa%3E+%26gt%3B+%3Ca+href%3D%27index.php%3F_a%3Dknowledgebase%26_j%3Dsubcat%26_i%3D60%27%3EAsterisk+Setup+And+Configuration%3C%2Fa%3E |
| 22:40.22 | chrisknight | ill probably be pissed |
| 22:41.43 | chrisknight | I have changed these files already... still having trouble... Ill try again I suppose. |
| 22:41.56 | [TK]D-Fender | chrisknight: Actaully thats a pretty good little guide |
| 22:42.37 | mut | jjshoe: sorry it's * not # |
| 22:42.48 | [TK]D-Fender | chrisknight: However you can't jsut cut & paste it. You need to understand the contexts being used by YOUR setup and where to insert the dialout options in your dial-plan and where you should be sending incoming calls to. |
| 22:42.54 | chrisknight | hmmm... Maybe I fat fingered something... I backed up those files and have already rm'ed the ones I was messing with... Ill copy over the backups and try again |
| 22:42.56 | mut | Also. during the prompt if the caller presses: |
| 22:42.56 | mut | <PROTECTED> |
| 22:43.04 | [TK]D-Fender | chrisknight: this is what a TYPICAL * would be able to make useful. |
| 22:43.20 | mut | in your dial plan ya just make an exten => a,1,Voicemailmain(${EXTEN}) |
| 22:43.23 | [TK]D-Fender | chrisknight: again, you can't jsut cut & paste verbatim.... |
| 22:43.23 | mut | under that context |
| 22:43.51 | chrisknight | I see... |
| 22:44.07 | chrisknight | Do you all use * or *Now? |
| 22:44.15 | CrashHD | mut: a,1,Voicemailmain(${EXTEN}) would go to voicemailbox * |
| 22:44.22 | jjshoe | mut link to that documntation? |
| 22:44.25 | CrashHD | a,1,Voicemailmain(${MACRO_EXTEN}) |
| 22:44.42 | [TK]D-Fender | chrisknight: The is #asterisk. There is a seperate channel for GUI specific stuff ( #asterisk-gui ) |
| 22:44.59 | chrisknight | hmmm... ok. thanks... |
| 22:45.02 | mut | yea |
| 22:45.03 | mut | what he said |
| 22:45.11 | [TK]D-Fender | chrisknight: However once the GUI is up, it doesn't do EVERYTHING for you, and you DO have to still learn how to work * independantly |
| 22:45.17 | jjshoe | do you have a link to where you found that? |
| 22:46.05 | chrisknight | Im sure ill catch on... thanks |
| 22:46.57 | mut | should be in the wiki |
| 22:47.04 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:47.40 | jjshoe | mut yeah, searching for a turns up the exact document |
| 22:47.49 | jjshoe | I know a, and o, are valid options |
| 22:48.15 | jjshoe | unfortunatly they arn't easy to search on. |
| 22:52.24 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 22:52.41 | [TK]D-Fender | *boing* |
| 22:54.35 | Malph | I just used grahamstownscholwiki to configure my asterisk server and I think i want to find the person who put the databse queries onthe site and shake the holy bejeezus out of him. |
| 22:57.32 | chrisknight | Is codec ulaw & g711u the same? |
| 22:57.58 | jjshoe | yes |
| 22:58.23 | chrisknight | ok, thanks |
| 22:59.06 | chrisknight | Do I just need to set the codec in the phones? ...or in * as well? |
| 23:00.18 | jjshoe | most tend to auto-negotiate |
| 23:00.25 | jjshoe | but it's always wise to set it in both places |
| 23:00.35 | chrisknight | ok |
| 23:02.41 | JacksLivr | when i call my * server blocking my callerid, it shows up as my home number when it gets to the NoOp(${CALLERID}) line. the only place anywhere i have my callerid set to this is on an fxs port that an analog phone hangs off of. this phone is not being touched in this scenario. |
| 23:03.02 | JacksLivr | when i dont block my callerid it shows up correctly |
| 23:03.48 | rhombus | My DID provider expects me to be on a static IP address. They provide SIP and IAX channels. Is a static IP really necessary, or should they support client-side registration? |
| 23:04.11 | jjshoe | rhombus necessary for their service apparently. |
| 23:04.18 | jjshoe | rhombus find someone else if you don't like their requirements |
| 23:04.28 | boch | features.conf says not to use any dialplan flow related command in [applicationmap] section, but wiki says "Note: You can use the Goto() application to jump anywhere into the extensions conf...". which should i trust ? |
| 23:05.02 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:05.20 | rhombus | jjshoe: i'm asking a technical question -- is there any reason why they couldn't allow client registration? |
| 23:07.48 | jjshoe | rhombus yes, their setup |
| 23:10.41 | rhombus | thanks for you help, jjshoe |
| 23:10.43 | rhombus | anyone else? |
| 23:10.53 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 23:10.58 | tdi | hi all |
| 23:11.29 | tdi | i am seeking for way to add asking for name in conferences |
| 23:11.51 | tdi | a voice ask to record the name, and later says 'name joined the conference' |
| 23:12.19 | *** join/#asterisk mrbnet (n=mrbnet@corpmail1.mrbnetworks.com) |
| 23:12.43 | mrbnet | What is the diff between 1.4 and 1.2? |
| 23:12.44 | JacksLivr | i think you use ,i, in the meetme.conf |
| 23:12.48 | jjshoe | tdi app_conference has that |
| 23:12.56 | jjshoe | <3 app_conference |
| 23:13.55 | tdi | and reserving also ? |
| 23:14.01 | tdi | because thet meetme2 has |
| 23:14.23 | russellb | meetme has that. |
| 23:14.31 | russellb | recording names, that is |
| 23:14.41 | tdi | how is that ? |
| 23:15.06 | russellb | the 'i' or 'I' option to MeetMe |
| 23:15.42 | tdi | small i aha announce user with rewiev |
| 23:15.43 | tdi | yes? |
| 23:15.51 | tdi | but it does not have reservations |
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| 23:45.52 | Shaun2222 | whats the advantage to use a PRI T1 vs's say just having a bunch of phone lines? is there one? |
| 23:46.27 | russellb | not having 24 pairs of copper coming in? |
| 23:46.45 | russellb | more reliable ... cheaper ... |
| 23:47.02 | Shaun2222 | uhh... sounds to me like redundancy! |
| 23:47.04 | Shaun2222 | :) |
| 23:47.06 | russellb | send and receive more information ... |
| 23:48.52 | Shaun2222 | so other than having extra pairs which in reality sounds more ideal because at least if for some reason a few pairs get damaged i have the others their is nothing else? |
| 23:49.11 | boch | what is spawn extensions? can i catch them ? |
| 23:50.51 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 23:51.06 | mmlj4 | Shaun2222: no, it's not about redundancy |
| 23:52.07 | [TK]D-Fender | Shaun2222: PRI si a digital connection that gives you the option for DID's (multiple #'s that can land on your inbound channels), call progress, etc |
| 23:52.17 | mmlj4 | a T1 is equal to 24 analog phone lines... those 24 lines are multiplexed onto 2 pairs of wires, but no, redundancy plays no part in it |
| 23:53.34 | [TK]D-Fender | Also usuallly lets you set your outbound callerid, and more |
| 23:53.47 | *** join/#asterisk Schreiber1337 (i=d8a9b0b6@gateway/web/cgi-irc/ircatwork.com/x-3735b365f5c1409c) |
| 23:55.14 | thekidrio | anyone here messed with hacked modems as a replacement for an fxo? |
| 23:55.49 | thekidrio | seems like a fun way to use up all these old modems |
| 23:56.07 | jjshoe | thekidrio vocp specializes just in that |
| 23:56.10 | jjshoe | thekidrio uses vgetty |
| 23:56.17 | Schreiber1337 | Having a lot of problem with transfers in 1.4 and I don't want to us an unrelease version. What is a good stable release 1.2.1? |
| 23:56.56 | thekidrio | hrmm jjshoe so it sounds like its atleast worth weekend project status |
