| 00:00.00 | tbs_ | yesh |
| 00:00.10 | [TK]D-Fender | Guess you didn't have much spare time if it took you 4 weeks :) |
| 00:00.20 | tbs_ | [TK]D-Fender: right :) |
| 00:00.58 | [TK]D-Fender | I do practically "drop-in" intalls most of the time. The real effor tends to lie in the phone configs + custom menus |
| 00:01.31 | tbs_ | well, yeah -- but we're fooling around with all kinds of features, trying to learn-by-doing |
| 00:02.33 | Drew__ | sounds familiar ;) |
| 00:03.16 | harryvv | sixtel been down part of today. |
| 00:03.25 | rayvd | Some ham. |
| 00:04.58 | _Sam-- | is there a way to hang up a call that is in a queue from cli? |
| 00:05.52 | *** join/#asterisk Tamarisk (n=adrian@user-6887.lns5-c11.dsl.pol.co.uk) |
| 00:07.51 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 00:07.56 | _Sam-- | soft hangup ...i see |
| 00:08.10 | _Sam-- | poor bastard. |
| 00:08.37 | Juggie | theres no queue command to boot someone from the queue? |
| 00:08.38 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 00:08.49 | _Sam-- | boot them to where? |
| 00:09.17 | Juggie | ah, no theres not. |
| 00:09.25 | Juggie | only show queue and show queues |
| 00:09.27 | Juggie | no control |
| 00:11.35 | WillySilly | how do i install the webmin module? |
| 00:12.05 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 00:12.27 | *** part/#asterisk Tamarisk (n=adrian@user-6887.lns5-c11.dsl.pol.co.uk) |
| 00:12.28 | [TK]D-Fender | ...what webmin module? |
| 00:12.56 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-215.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 00:13.16 | WillySilly | http://ftp.digium.com/pub/asterisk/webmin/ |
| 00:13.20 | *** join/#asterisk sprnova (n=oregonal@www.nemirovskyfamily.com) |
| 00:14.29 | [TK]D-Fender | Have you LOOKED at the file? |
| 00:14.45 | Juggie | heh |
| 00:14.46 | WillySilly | yeah |
| 00:14.48 | Juggie | i woudnt touch that thing |
| 00:14.49 | Juggie | ever |
| 00:14.50 | _Sam-- | wow people pay money for the thirdlane webmin thing? |
| 00:14.58 | [av]bani | yep |
| 00:15.03 | _Sam-- | insane |
| 00:15.09 | [av]bani | people even pay money for pre packaged amp |
| 00:15.20 | [av]bani | and you will pay me money for gxp autoprovisioner :) |
| 00:15.21 | _Sam-- | how much for the thirdlane? |
| 00:15.24 | _Sam-- | lol |
| 00:15.30 | hertell | is ser (SIP express router) something that is still usable? |
| 00:15.57 | sprnova | hello.. did there anything specially I have to do to enable the ztdummy driver in asterisk? Running 1.0.7 on Debian 31r2 and hearing "metalic" type audio when accessing voicemail. |
| 00:15.58 | hertell | found over here: http://developer.berlios.de/projects/ser/ |
| 00:16.37 | _Sam-- | sprnova : did you "modprobe ztdummy" before you started * |
| 00:16.54 | _Sam-- | hertell : i think alot of people use ser still, and openser? im no expert there |
| 00:17.25 | sprnova | _Samm--.. no I didn't.. new to Linux.. was trying asterisk under BSD and was told to switch to Linux for this very problem. I will give that a try. |
| 00:17.47 | _Sam-- | sprnova : you also may not get much support until you try a newer version of * |
| 00:17.59 | _Sam-- | there are newer debian packages |
| 00:18.12 | sprnova | _Samm--.. no problem.. I will try whatever version works. |
| 00:18.22 | sprnova | what's recommended? |
| 00:18.28 | hertell | _Sam: do you know if that is something that should be concidered, or can asterisk overcome potential NAT-problems? |
| 00:18.35 | _Sam-- | i ran the debian 1.2.1 package fine |
| 00:18.49 | _Sam-- | hertell: there are alot of workarounds. |
| 00:18.55 | _Sam-- | most people dont need ser |
| 00:19.05 | hertell | _Sam: ok |
| 00:19.34 | [TK]D-Fender | SER is good for large installations and those needing redundancy |
| 00:19.35 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:19.39 | Ariel_ | hello everyone |
| 00:19.44 | hertell | are the workaround * specific, or does it require more tweaking of my firewall? |
| 00:19.48 | [av]bani | ~ser |
| 00:19.53 | jbot | somebody said ser was Sip Express Router - see http://www.iptel.org/ser/ |
| 00:19.59 | _Sam-- | the workarounds are alot of times phone specific. |
| 00:20.32 | _Sam-- | i dont know your current firewall situation in relation to where your * is...but you can tell us |
| 00:20.36 | _Sam-- | and maybe others will have more input |
| 00:20.47 | hertell | ok. btw. is echo something that has to do with nat, or is there any other way in getting rid of it? |
| 00:21.03 | _Sam-- | how do you connect to regular the regular phone network? |
| 00:21.08 | hertell | eg. skhype has minimal echo.. |
| 00:21.10 | _Sam-- | sorry cant ttalk. |
| 00:21.14 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 00:21.29 | _Sam-- | how do you get your calls out to the regular phone network? |
| 00:21.33 | _Sam-- | pstn |
| 00:21.54 | _Sam-- | analog pots lines? pri? remote provider? |
| 00:21.55 | hertell | yep. But I have not yet tested that route |
| 00:22.03 | _Sam-- | so how do you hear the echo? |
| 00:22.08 | hertell | i got it working a few hours ago |
| 00:22.18 | sprnova | _Sam--: when I tell it to modprobe ztdummy it says not found.. I did a find and couldn't find it on my system.. I guess it is a seperate debian package? |
| 00:22.33 | _Sam-- | yep. its called zaptel |
| 00:22.42 | sprnova | _Sam--.. thanks |
| 00:22.45 | _Sam-- | 1) apt-get update 2) apt-cache search zatpel |
| 00:22.53 | _Sam-- | 3) apt-get install zaptel-something |
| 00:23.01 | hertell | i hear the echo when I was talking yesterday with my friend (like my voice as a light backgroundnoice behind my friend) |
| 00:23.14 | _Sam-- | how did you call your friend |
| 00:23.24 | hertell | ohh, over fwd |
| 00:23.41 | _Sam-- | what type of device were you calling from |
| 00:23.52 | hertell | spa3k->asterisk->fwd->linphone |
| 00:24.04 | _Sam-- | could have been anything |
| 00:24.14 | hertell | ;-) |
| 00:24.34 | _Sam-- | how far away (ms) are you from fwd? |
| 00:24.41 | _Sam-- | <ive never used fwd> |
| 00:24.51 | hertell | 130 |
| 00:24.58 | _Sam-- | could be part of it |
| 00:25.01 | _Sam-- | who knows |
| 00:25.25 | sprnova | _Sam--.. should I get libzap1 too? |
| 00:25.26 | _Sam-- | i dont have an spa3k..but i THINK they are supposed to be pretty good about echo |
| 00:25.36 | hertell | yeah, i will skip that route too now whey i figured out how to call directly to eg. my box.. :-) |
| 00:25.49 | sprnova | I have a spa3k.. echo on the FXO is common |
| 00:26.00 | sprnova | no echo on FXS |
| 00:26.35 | hertell | ok |
| 00:26.36 | sprnova | to remove echo I was able to adjust the SPA to PSTN gain |
| 00:26.57 | hertell | where do you do that? |
| 00:27.04 | hertell | in spa-admin or * |
| 00:27.29 | [TK]D-Fender | SPA |
| 00:27.35 | _Sam-- | sprnova : i dont think you need libzap1 to do ztdummy. |
| 00:27.41 | _Sam-- | <famous last words> |
| 00:27.49 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 00:27.49 | _Sam-- | its easy enough to get if you do. |
| 00:27.50 | sprnova | in the PSTN tab.. towards the bottom.. might be under the FXS Internalization section |
| 00:28.15 | hertell | [TK]D-Fender: i btw got my setup working now :-) I finally figured out the logic behind iax.conf, sip.conf and extensions.conf :-) |
| 00:30.19 | hertell | sprnova: my valus are @ 0.. did you change do much changes to those values? |
| 00:30.42 | hertell | did you do much changes to... |
| 00:30.47 | sprnova | _Sam--.. ok.. I installed zaptel and libzap1.. still not able to find a ztdummy module.. where should I find it.. I should know but I just switched to Linux from Bsd today. ;-) |
| 00:31.21 | sprnova | hertell.. try turning SPA to PSTN 4 |
| 00:31.33 | hertell | sprnova: what's wrong with bsd? |
| 00:31.35 | sprnova | also make sure the Adaptive echo cancelation is turned on. |
| 00:31.40 | hertell | ok. I'll try that |
| 00:31.44 | _Sam-- | you could try 'updatedb' first, then 'locate ztdummy' |
| 00:32.21 | sprnova | _Sam--.. yup.. did a cd / ; find / -name ztdummy -print... will try the updatedb then locate |
| 00:32.27 | _Sam-- | should be somewhere in /lib/modules/yourkernelhere |
| 00:33.00 | hertell | sprnova: do you have the kernelheaders installed? |
| 00:33.15 | hertell | uhh.. nevermind ;-) |
| 00:33.34 | hertell | thought you where looking in /usr/src... |
| 00:33.40 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 00:33.44 | sprnova | _Sam--.. still don't have it..hmm. |
| 00:33.50 | _Sam-- | wondering if the regular debian zaptel didnt come with ztdummy..hold. |
| 00:34.14 | _Sam-- | check this page: |
| 00:34.15 | _Sam-- | http://rapid.dotsrc.org/unstable/ |
| 00:34.26 | _Sam-- | what kernel are you running |
| 00:34.35 | sprnova | 2.4.27-2 |
| 00:34.46 | _Sam-- | damn dude its the 21st century |
| 00:35.02 | _Sam-- | you got a problem with 2.6? :) |
| 00:35.05 | sprnova | just download Debian stable.. that is came with it. ;-) |
| 00:35.13 | sprnova | _Sam--.. whatever works. |
| 00:35.21 | *** part/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 00:35.33 | _Sam-- | tzafrir_laptop : you there? |
| 00:35.47 | _Sam-- | sprnova : tzafrir maintains that debian package |
| 00:35.50 | _Sam-- | <zaptel> |
| 00:35.54 | *** join/#asterisk MoutaPT (i=MoutaPT@85.139.183.206) |
| 00:36.03 | sprnova | _Sam.. so should I reinstall a different debian release? |
| 00:36.18 | MoutaPT | Hello, I've been installing asterFax in my *@home 2.5 |
| 00:36.24 | _Sam-- | what type of machine is this for? |
| 00:36.27 | MoutaPT | and i get conflict with port25 |
| 00:36.31 | _Sam-- | i mean, for screwing around and learning? |
| 00:36.34 | _Sam-- | or for setting up for something? |
| 00:36.36 | sprnova | Celeron 800mhz |
| 00:36.39 | MoutaPT | seems to be with SendMAil |
| 00:37.08 | _Sam-- | sprnova : is it just a personal box, or? |
| 00:37.11 | MoutaPT | Any one knows if i should keep sendMail and AsterFAx in same machine? |
| 00:37.19 | sprnova | _Sam--.. setting up a reliable PBX for family. |
| 00:37.41 | _Sam-- | i think it makes sense to install the 2.6 kernel |
| 00:38.12 | _Sam-- | i dont know how you'd upgrade what you have to 2.6....ive never done it that way. |
| 00:38.15 | sprnova | _Sam--.. should I do that from my current kernel or do I need to reinstall via CDROM? |
| 00:38.22 | sprnova | ok |
| 00:38.25 | _Sam-- | im sure there is a way |
| 00:38.34 | _Sam-- | like 'apt-get upgrade"...but that may break more stuff than its worth |
| 00:38.54 | hertell | sprnova: you can upgrade the kernel by installing the 2.6 with apt.. |
| 00:38.55 | *** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net) |
| 00:39.13 | MoutaPT | Any one here got AsterFax running with SendMail in asterisk@home ? ( igot conflict port:25) |
| 00:39.14 | _Sam-- | that wont mess up other things that are compiled against/for the 2.4 kernel? |
| 00:39.20 | hertell | sprnova: what do you need the ztdummy module for? |
| 00:39.51 | buZz | could someone tell me what they 'monitor' in the corner of their eye (maybe not literal) for their asterisk server? |
| 00:39.53 | hertell | MoutaPT: I have no experience about AsterFax.. |
| 00:40.07 | buZz | i'm currently just doing online sip peers and active channels & calls |
| 00:40.28 | sprnova | hertell.. I guess I need it for timing.. I am getting metalic sounding audio and dropped packets. |
| 00:40.35 | sprnova | between SPA and asterisk |
| 00:40.42 | MoutaPT | thanks hertell |
| 00:40.43 | hertell | hmm.. |
| 00:40.48 | _Sam-- | the only thing ztdummy is help timing on MOH |
| 00:40.49 | hertell | is it over nat? |
| 00:40.51 | _Sam-- | and conferences/meetme |
| 00:40.54 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 00:40.56 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
| 00:41.00 | _Sam-- | it wont help regular calls |
| 00:41.07 | MoutaPT | talking with me hertell? |
| 00:41.30 | sprnova | hertell... voicemail between asterisk and SPA is direct on same lan. |
| 00:41.43 | sprnova | hertell.. SPA has nat disabled. |
| 00:41.57 | hertell | MoutaPT: well, i just told you that i can't help you.. |
| 00:42.08 | MoutaPT | hertell: ok |
| 00:42.22 | sprnova | so is the thought that ztdummy wont help me? |
| 00:42.24 | MRH2 | anyone know what is up with polycom firmware looks like they released 1.6.4 recently and then just updates thebootrom and sip to 1.6.5 a few days after - i don;t see any release notes though |
| 00:42.32 | hertell | sprnova: strange.. I run also a spa on the same lan, but with no probs.. |
| 00:42.43 | hertell | just the female-voice suxx ,-) |
| 00:42.48 | sprnova | hertel.. what asterisk version and kernel? |
| 00:43.05 | sprnova | I am using 1.0.7. |
| 00:43.30 | hertell | i run 1.0.9 (i removed debians stock-version and installed the 1.0.9 version so that I got the Wengo-patch working) |
| 00:43.40 | hertell | but I had no problem with 1.0.7 |
| 00:44.24 | sprnova | the problem is even worse between SJPhone as Asterisk .. also on same lan |
| 00:44.40 | hertell | i had problems to get a parallel installation working on debian, so I removed the stock-version |
| 00:44.45 | sprnova | between SJPhone AND Asterisk |
| 00:45.40 | Mavvie | set verbose should be settable per console. |
| 00:46.02 | Mavvie | I hate it when I can't read what I'm trying to debug.... |
| 00:46.10 | hertell | on what platform are you running sjphone? |
| 00:46.36 | hertell | Mavvie: eg: set verbose 10 |
| 00:46.54 | sprnova | hertell.. did you mess with any of the RTP settings in the SPA? running sjphone on XP. via 802.11b.. and no.. I don't have any 2.4Ghz cordless phones. |
| 00:47.06 | Mavvie | hertell: normally I have it on 3, which gives me enough information to see what's happening. |
| 00:47.22 | Mavvie | hertell: but when I try to see what the settings of a channel are..... |
| 00:47.28 | Mavvie | scroll scroll scroll |
| 00:47.39 | hertell | sprnova: no, it's more or less default.. |
| 00:48.06 | hertell | Mavvie: yep. that suxx.. I would rather have it in a file and search for stuff directly |
| 00:48.47 | sprnova | hertell.. when you do a a "sip show peer" what kind of ping times do you show to your SPA? I am seeing about 10ms. |
| 00:49.42 | hertell | how do you enable the ping-values? |
| 00:50.05 | hertell | i have the status-column"unmonitored".. |
| 00:50.31 | austinnichols101 | qualify = yes |
| 00:50.50 | Mavvie | hertell: qualify=yes |
| 00:51.08 | hertell | in sip.conf? |
| 00:51.10 | Mavvie | in the sip-configuration |
| 00:51.15 | Mavvie | [foo] |
| 00:51.17 | Mavvie | qualify=yes |
| 00:51.18 | sprnova | ok... maybe it one of my Ethernet switches along the way.. I am going through at least 4 |
| 00:51.27 | sprnova | darn things |
| 00:51.41 | hertell | ok |
| 00:52.01 | sprnova | I really should try a point to point connections to rule things out. |
| 00:52.18 | austinnichols101 | sprnova: watch out for asymetric routing... |
| 00:52.29 | fugitivo | anyone using rxfax and txfax version 0.0.2pre25 with asterisk-1.2.4? |
| 00:52.54 | sprnova | austinnichols101.. not going throught NAT.. is there another way to be bitten by this? |
| 00:53.50 | hertell | sprnova: line1 has 24ms, and PSTN 36ms |
| 00:53.59 | sprnova | i am hoping it is a simple as a broken ethernet switch someplace.. the noice comes and goes almost periodically. |
| 00:55.15 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 00:55.32 | austinnichols101 | sprnova: not talking about nat. You just need to make sure that the packets take the same route to and from the endpoints. |
| 00:55.45 | hertell | sprnova: now when i dropped all but the spa qualify = yes, they ping dropped to 7 and 14 ms |
| 00:55.51 | austinnichols101 | if you take one path in one direction and one path in the return direction then you can have a big issue |
| 00:56.01 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 00:56.17 | austinnichols101 | typically you'll run spanning tree on the switches in a case where you have multiple links |
| 00:56.26 | sprnova | when I watch the ethernet link light it goes on solid when the voice is good.. just before the stutter/metalic noice the light gots off for a few sec then back on |
| 00:56.27 | hertell | sprnova: dont know if thiw would help: localnet = 192.168.1.0/255.255.255.0 |
| 00:56.34 | Telamon | When a page says a codec uses X kbps, ie, GSM uses 13 kbps, is that bits or bytes? And can we go back in time and kick the idiot who decided to pick two B words for data sizing in the nuts? |
| 00:57.02 | hertell | btw. what codec-order do you have in your sip.conf? |
| 00:57.12 | *** part/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net) |
| 00:57.22 | sprnova | astinnichols101.. I just have cheap ethernet switches.. not intelligent ones.. can this still happen? |
| 00:58.05 | Telamon | sprnova: It sound like a bad network cable, if the link light is not stable. |
| 00:58.10 | austinnichols101 | sprnova: sure. look at how they're cabled together. Do you have multiple links between any two switches that form a loop? |
| 00:58.26 | sprnova | hertel.. using ulaw and alaw codec.. disallow everything else. |
| 00:58.37 | *** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net) |
| 00:58.50 | sprnova | austinnichols101.. I do have a loop |
| 00:59.07 | austinnichols101 | sprnova: why? |
| 00:59.22 | hertell | sprnova: ok.. you might concider speex.. |
| 00:59.41 | hertell | (not sure if it comes with stock-debian *.. |
| 00:59.50 | *** part/#asterisk ChaotY2k (n=juergen@port-195-158-180-70.dynamic.qsc.de) |
| 00:59.56 | sprnova | austinnichols101... trying to be a little to fancy in routing my company HW vpn onto my local lan. |
| 01:00.07 | sprnova | stupid stuff.. I guess I should know better. |
| 01:00.35 | hertell | (check with find /usr/libls -la /usr/lib/asterisk/modules/ |grep cod |
| 01:00.48 | hertell | ls -la /usr/lib/asterisk/modules/ |grep cod |
| 01:01.08 | sprnova | what does speex do? |
| 01:01.28 | hertell | http://www.speex.org/comparison.html |
| 01:01.36 | hertell | opensource speech-codec.. |
| 01:01.48 | sprnova | hertel.. ok.. thanks |
| 01:01.51 | hertell | changes bitrate dynamically.. |
| 01:02.26 | hertell | not sure how good it is (not tested yet) |
| 01:02.35 | hertell | sounds anyway as a good option.. |
| 01:02.57 | hertell | sprnova: did you have that codec installed? |
| 01:03.06 | sprnova | ok... so I am thinking I have some ethernet switches to checkout. thanks austinnichols101 for the pointer. |
| 01:03.49 | sprnova | hertel.. yes its here. |
| 01:03.57 | hertell | ok.. |
| 01:04.11 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 01:04.19 | sprnova | hertell.. did you check your echo cancelling switches on the SPA3k? |
| 01:04.20 | hertell | cause when i compiled * by hand, it was missing (had to install it separately for debian..) |
| 01:05.25 | hertell | and recompile |
| 01:05.41 | hertell | sprnova: no, i have not changed those settings |
| 01:06.12 | sprnova | I have my SPA to PSTN gain cranked up.. to 7.. PSTN to SPA Gain at 0. |
| 01:06.34 | hertell | btw. on my line1-tab all are set to yes (Echo Canc Enable: etc) |
| 01:06.58 | sprnova | there are another set in the PSTN tab |
| 01:07.07 | sprnova | under Audio Configuration section |
| 01:07.18 | hertell | i will test those during daylight (it's 03.07 in the middle of the night over here ;-) |
| 01:07.47 | sprnova | I have then all one.. Echo Cac Enable, Echo Canc Adapt Enable, Echo Supp Enable |
| 01:07.54 | sprnova | all those are ON |
| 01:08.04 | hertell | yep. mine too. |
| 01:08.29 | sprnova | the echo seems to be worse when both the gains to and from SPA are the same. |
| 01:08.53 | hertell | i've have to test that next time i'm in my office.. |
| 01:08.56 | sprnova | having them at different levels helped |
| 01:09.30 | hertell | ok. now it's time to get some sleep.. |
| 01:09.53 | sprnova | good night hertell. |
| 01:09.56 | hertell | (6h til i have to wake up.. ;-) |
| 01:10.15 | hertell | sprnova: good luck with your *..! :-) |
| 01:10.21 | sprnova | thanks |
| 01:10.41 | hertell | i've got finally myne working thatnks to [TK]D-Fender .-) |
| 01:11.03 | hertell | next project is bakground music.. etc :-) |
| 01:11.12 | sprnova | fun stuff |
| 01:11.27 | [TK]D-Fender | I'm surprised your up this late. |
| 01:11.38 | [TK]D-Fender | its like 2am for you isn't it? |
| 01:11.46 | hertell | [TK]D-Fender: 3.. |
| 01:11.48 | hertell | ;-) |
| 01:11.52 | [TK]D-Fender | oops |
| 01:11.57 | [TK]D-Fender | SLEEP DAMMIT! |
| 01:12.05 | hertell | LOL! |
| 01:12.29 | hertell | I got so exited when all this wonderstuff is working :-) |
| 01:12.44 | hertell | [TK]D-Fender: Thanks again for your help! |
| 01:17.15 | sprnova | asternnichols101... ethernet switch it is.. damn thing.. loop loop loop |
| 01:17.26 | sprnova | that was the problem. |
| 01:17.40 | sprnova | slap my hand. |
| 01:17.50 | sprnova | bad sprnova bad sprnova |
| 01:18.05 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 01:24.05 | [av]bani | slap mah fro! |
| 01:25.39 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 01:29.26 | *** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net) |
| 01:29.39 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 01:30.08 | ManxPower | WOW! fxotune REALLY works! |
| 01:30.25 | [av]bani | o.o |
| 01:31.00 | ManxPower | Until now I've always been able to remove echo on analog lines without needing FXO tune. |
| 01:31.22 | ManxPower | Maybe having an 11 mile loop might have something to do about it. |
| 01:31.25 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-113-60.dsl.tulsok.swbell.net) |
| 01:31.56 | ManxPower | I suspect the loop itelf is not actually 11 miles long, but the CO my line is connected to is 11 miles away. |
| 01:34.12 | [av]bani | its probably longer |
| 01:34.26 | *** join/#asterisk anonymouz666 (n=lynx@allende.redetaho.com.br) |
| 01:34.37 | [av]bani | loop length only seems to affect impedance though, the real delay comes if you get switched between LECs |
| 01:34.42 | [av]bani | or go out to cellphones |
| 01:34.51 | SibRphrek | this is weird |
| 01:34.57 | SibRphrek | my internal extensions work |
| 01:35.01 | ManxPower | [av]bani, normal echo can be removed with the Asterisk EC, this echo was not. |
| 01:35.05 | SibRphrek | i can call every extension |
| 01:35.10 | ManxPower | In fact I got echo after the dial, but before the call connected. |
| 01:35.12 | SibRphrek | i can call out from _my_ extension |
| 01:35.26 | SibRphrek | but if you dial my outside number which points to my extension the caller get's a 404 |
| 01:35.45 | [av]bani | ManxPower: thats not suprising, since the path characteristics often change radically once the other end gets connected |
| 01:36.14 | [av]bani | ManxPower: after dial but before connect, you're just talking to the CO switch. when connected, then you get the full path |
| 01:36.32 | ManxPower | *nod* |
| 01:36.40 | ManxPower | i.e. the only analog is my loop at that point |
| 01:36.44 | [av]bani | i get same thing here really |
| 01:36.47 | Abydos313 | afternoon everyone |
| 01:36.58 | [av]bani | i have an spa3k and i can hear it racheting the EC while the far end rings |
| 01:37.00 | ManxPower | I'm just happy that fxotune works. |
| 01:37.11 | [av]bani | i guess the new EC is far better than the old ones |
| 01:37.18 | [av]bani | but HW EC is still the way to go |
| 01:38.15 | [av]bani | still seems hard to beat throwing a tellabs EC in |
| 01:38.29 | [av]bani | but then, tellabs has been making EC hardware for ILECs for ~20 years |
| 01:42.41 | ManxPower | I've never had significant issues with echo on analog loops before. |
| 01:43.06 | ManxPower | I've had no luck with Tellabs on our PRIs. Can't figure out why. |
| 01:44.11 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 01:44.12 | SibRphrek | http://pastebin.com/558834 |
| 01:44.16 | SibRphrek | why would all of a sudden this happen |
| 01:44.20 | SibRphrek | http://pastebin.com/558834 |
| 01:47.14 | Ariel_ | means it did not find the server or device |
| 01:47.24 | SibRphrek | but why all of a sudden would it start |
| 01:47.31 | SibRphrek | it was working earlier today |
| 01:47.32 | Ariel_ | network issues |
| 01:47.35 | SibRphrek | and then just poof stopped working |
| 01:47.54 | SibRphrek | Ariel_: i would agree, but i can make calls out |
| 01:48.15 | SibRphrek | the setup is werid |
| 01:48.18 | *** join/#asterisk santiago (n=afsdf@63.245.86.219) |
| 01:48.20 | *** join/#asterisk PaulHuynh (n=paulhuyn@c-68-44-237-105.hsd1.de.comcast.net) |
| 01:48.21 | SibRphrek | we have 1 asterisk server talking to antoerh |
| 01:48.33 | SibRphrek | so the number is being passed from one to the other |
| 01:48.37 | SibRphrek | and i can't figure out where it's dying |
| 01:48.44 | PaulHuynh | Sam are u there? |
| 01:48.57 | PaulHuynh | i need another SC number |
| 01:49.37 | PaulHuynh | do anyone have any recommend for DID |
| 01:49.45 | PaulHuynh | for south carolina |
| 01:51.14 | PaulHuynh | Ariel_ |
| 01:51.18 | PaulHuynh | are u there? |
| 01:51.31 | Ariel_ | yes are you? |
| 01:52.32 | Ariel_ | I only have an did from connection at voicepulse. They have been good for me for the pass 2 years. but it's 11 dollars now. |
| 01:52.46 | PaulHuynh | damn |
| 01:52.52 | SibRphrek | argh! |
| 01:52.54 | SibRphrek | wtf!!!! |
| 01:53.20 | Ariel_ | it's unlimited inbound and I am able to get 4 calls in at one time via iax2 |
| 01:53.31 | PaulHuynh | oh ok |
| 01:53.37 | PaulHuynh | So that not bad |
| 01:53.47 | PaulHuynh | but i want to use SIP |
| 01:54.07 | PaulHuynh | well that i'm must familar with |
| 01:54.14 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-67-128.cybersurf.com) |
| 01:55.12 | austinnichols101 | PaulHuynh: telasip |
| 01:55.42 | Ariel_ | PaulHuynh, they have sip as well. But I needed the to only open inbound on one port. |
| 01:56.05 | austinnichols101 | PaulHuynh: or Didx.org / virtualphoneline.com |
| 01:56.11 | PaulHuynh | I need more than one 2-4 would be idead |
| 01:56.47 | Ariel_ | didx has some good numbers but it's on a sip unsecure ip to ip address setup which I don't like |
| 01:57.04 | Abydos313 | how much do they charge you for 4 inbound? |
| 01:57.08 | *** part/#asterisk santiago (n=afsdf@63.245.86.219) |
| 01:57.44 | austinnichols101 | I've seen them for around 5 bucks a month with a bunch of minutes attached to them (didx.org). virtualphoneline.com has the same numbers but is around 7 bucks. |
| 01:58.12 | austinnichols101 | telasip is 14.95/month with unlimited us outdialing |
| 01:58.15 | Abydos313 | so that would mean you could handle 4 outgoing at the same time too? |
| 01:58.35 | Abydos313 | austinnichols101 what about incomming |
| 01:58.39 | Abydos313 | and a number |
| 01:58.50 | austinnichols101 | I believe incoming is 1:1 |
| 01:58.59 | austinnichols101 | not 100% sure on that one |
| 01:59.02 | Ariel_ | voicepulse outbound charges by the minute. not by the channel. inbound via connections is 4 channels unlimite minutes |
| 01:59.14 | Abydos313 | ok |
| 01:59.31 | Ariel_ | so 11 dollars for 4 channels unlimite is not bad in this case. |
| 01:59.34 | Abydos313 | i'd personally like to setup 2 in/out on a dsl line |
| 01:59.45 | Abydos313 | that price sounds good |
| 02:00.15 | *** part/#asterisk Drew__ (n=foo@zux221-186-224.adsl.green.ch) |
| 02:00.25 | linlin | ok, so ive got some friends connecting to my asterisk machine through iax2, using the idefisk softphone |
| 02:00.30 | PaulHuynh | Ariel |
| 02:00.34 | linlin | they cal make calls to eachothers extentions |
| 02:00.36 | *** join/#asterisk XnoN (n=xnon@200.8.30.23) |
| 02:00.42 | XnoN | hello |
| 02:00.52 | XnoN | anybody speak spanish here? |
| 02:00.54 | PaulHuynh | do you have to buy a basic plan to get did from voicepulse |
| 02:00.56 | linlin | but when they goto make an outgoing call, outside of the machine, it will ring and connect the other phone, but neither end can hear anything |
| 02:01.16 | XnoN | alguien que hable español en este channel?¿ |
| 02:01.31 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 02:01.36 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 02:01.38 | Abydos313 | linlin your running sip and your asterisk server is behind nat? |
| 02:01.42 | wellng | hi all |
| 02:01.42 | Abydos313 | i have the same issue |
| 02:01.50 | *** part/#asterisk welles (n=welles@61.150.43.113) |
| 02:02.40 | linlin | no, they are connecting using iax2 |
| 02:02.48 | linlin | but it is behind a nat, i have forwarded 1 port though |
| 02:02.56 | Abydos313 | nevermind, iax2 doesn't have that issue |
| 02:03.04 | Abydos313 | so i read |
| 02:03.13 | linlin | yeah iax2 is supposed to work well through nat |
| 02:03.17 | wellng | why meetme list cmd can not work? when i run meetme list it show me the usage. but before it show me all the active conference |
| 02:03.28 | Ariel_ | PaulHuynh, no just from this site: http://connect.voicepulse.com/ |
| 02:03.30 | linlin | but its wierd, internal asterisk calls work, but calls to the pstn dont |
| 02:04.06 | *** join/#asterisk krischnoff (n=chatzill@209.77.205.12) |
| 02:04.40 | wellng | tzafrir, can u help me? why meetme list cmd can not work? when i run meetme list it show me the usage. but before it show me all the active conference. |
| 02:04.48 | Abydos313 | maybe something is wrong with your dailplan |
| 02:04.51 | linlin | any ideas anyone? |
| 02:05.30 | Ariel_ | linlin, what is the error your getting when you try a pstn call |
| 02:07.13 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 02:08.16 | wellng | what's wrong? |
| 02:09.57 | *** join/#asterisk voip470 (n=A_mail@pool-71-246-11-20.phlapa.fios.verizon.net) |
| 02:09.59 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 02:11.50 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 02:12.23 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com) |
| 02:12.46 | wellng | zoa, r u there? |
| 02:14.41 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 02:19.56 | *** join/#asterisk ReD-MaN (i=redman@207.210.19.76) |
| 02:20.20 | PaulHuynh | and i do this with asterisk |
| 02:20.51 | PaulHuynh | fax-linksyspap2-asterisk-sipprovider(sixtel) |
| 02:21.04 | PaulHuynh | what codec would i have to use |
| 02:22.53 | Ariel_ | ulaw |
| 02:23.43 | PaulHuynh | OK anything else i should do for the pap2 setting |
| 02:23.58 | PaulHuynh | or just like i would do for a sip phone |
| 02:24.53 | PaulHuynh | i also need to do this (please let me know if it possible) |
| 02:27.21 | PaulHuynh | two pap2=4 different ext-nat-asterisk |
| 02:27.48 | PaulHuynh | do i need to set anything special or just nat=yes in the extension conf? |
| 02:28.02 | *** join/#asterisk johnsu01 (n=user@fsf/staff/johnsu01) |
| 02:28.25 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 02:31.52 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 02:37.32 | XnoN | alguien que hable español |
| 02:37.33 | XnoN | ? |
| 02:37.47 | De_Mon | !spanish |
| 02:37.57 | De_Mon | nope |
| 02:38.19 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 02:38.30 | De_Mon | !es |
| 02:38.42 | PaulHuynh | ariel |
| 02:38.48 | PaulHuynh | are u there help |
| 02:38.59 | PaulHuynh | please |
| 02:39.25 | XnoN | ok |
| 02:39.26 | XnoN | sorry |
| 02:39.37 | XnoN | i need so help with asterisk |
| 02:39.51 | XnoN | i would like to take full information about this |
| 02:40.04 | XnoN | my from is venezuela |
| 02:40.22 | XnoN | i need to make a asterisk server un my local area network |
| 02:40.41 | Qwell | Your from is Venezuela? |
| 02:40.46 | XnoN | but i dont know much about asterisk |
| 02:40.51 | XnoN | yes it is |
| 02:41.01 | _Sam-- | babelfish.altavista.com |
| 02:41.02 | XnoN | Im a venezuelan boy friend! |
| 02:41.08 | XnoN | jejejeje ok! |
| 02:41.36 | De_Mon | http://www.asterisk-es.org/ ? |
| 02:41.42 | De_Mon | voip-info.org |
| 02:42.04 | _Sam-- | Qwell: after seeing a few more GUIs i know why vi is your best friend |
| 02:42.11 | _Sam-- | its crazy that people pay for some of that crap |
| 02:42.13 | XnoN | woao jejeje |
| 02:42.20 | De_Mon | vi? vim is better |
| 02:42.21 | XnoN | thanx so much De_Mon |
| 02:42.30 | De_Mon | XnoN :) |
| 02:43.00 | justinu | jejejeje |
| 02:43.06 | PaulHuynh | can i do this |
| 02:43.09 | justinu | reminds me of that robot on battlestar galatica |
| 02:43.12 | PaulHuynh | two pap2=4 different ext-nat-asterisk |
| 02:43.15 | _Sam-- | lol |
| 02:43.18 | PaulHuynh | do i need to set anything special or just nat=yes in the extension conf? |
| 02:43.55 | XnoN | take a nice night |
| 02:44.05 | [TK]D-Fender | PaulHuynh : You need to set one of EXTERNIP or EXTERNHOST, and LOCALNET in [general] in sip.conf |
| 02:44.28 | PaulHuynh | what do you mean? |
| 02:44.41 | [TK]D-Fender | Your * is behind NAT, correct? |
| 02:44.53 | PaulHuynh | NO |
| 02:45.02 | PaulHuynh | asterisk is in public |
| 02:45.16 | PaulHuynh | two pap2 will be on a same NAT |
| 02:45.25 | PaulHuynh | connect to outside asterisk |
| 02:45.32 | [TK]D-Fender | Ok, then you should only need to use "qualify=yes" and "nat=yes" for those phones and you should be OK. Though I would suggest you set a different UDP port for each of them. |
| 02:45.53 | PaulHuynh | oh ok |
| 02:46.20 | [av]bani | \o/ |
| 02:46.32 | _Sam-- | <PROTECTED> |
| 02:46.39 | [TK]D-Fender | 69? |
| 02:46.43 | _Sam-- | haha |
| 02:47.44 | _Sam-- | [av]bani: all of the phones for your ISP run over remote gateway? |
| 02:48.22 | _Sam-- | or rather, all of your phone service to the outside world is implemented over a remote gw? |
| 02:48.25 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 02:48.39 | sprnova | _Sam--... I figured out where ztdummy was hidding... it is not built by default.. so I built it from source.. any idea what is wrong here when I run modprobe. |
| 02:48.55 | _Sam-- | what is the error |
| 02:48.58 | sprnova | asterisk:/usr/src/modules/zaptel# !modpr |
| 02:48.58 | sprnova | modprobe ztdummy |
| 02:48.58 | sprnova | /lib/modules/2.4.27-2-386/zaptel/zaptel.o: /lib/modules/2.4.27-2-386/zaptel/zaptel.o: unresolved symbol devfs_unregister_R65aaa37a |
| 02:49.26 | sprnova | it goes on like this |
| 02:49.38 | _Sam-- | i cant tell ya...but i can tell ya if you get the .deb package from the site i showed you, it will work |
| 02:49.48 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 02:50.03 | sprnova | I got the .deb source package and rebuilt |
| 02:50.04 | welles | hi _Sam-- |
| 02:50.11 | _Sam-- | http://rapid.dotsrc.org/unstable/ |
| 02:50.17 | sprnova | oh |
| 02:50.20 | _Sam-- | there is a package just for your kernel already |
| 02:50.23 | sprnova | sorry about that. |
| 02:50.26 | _Sam-- | there is nothing wrong with wanting to compile it |
| 02:50.27 | sprnova | cool |
| 02:50.28 | _Sam-- | it SHOULD work |
| 02:50.34 | _Sam-- | but that WILL work :) |
| 02:50.50 | _Sam-- | also i still feel you should consider upgrading your asterisk version. |
| 02:51.12 | PaulHuynh | where can i get china did? |
| 02:51.26 | PaulHuynh | any help would be great |
| 02:51.29 | _Sam-- | i was just reading on the asterisk-biz list its illegal to resell china DID |
| 02:51.46 | PaulHuynh | what? |
| 02:51.59 | PaulHuynh | there go that deal out of the windows |
| 02:52.17 | PaulHuynh | can we bring US did to china via internet in that case? |
| 02:52.30 | _Sam-- | paste coming <sorry pastebin advocaters> |
| 02:52.31 | _Sam-- | >> Has anyone ever gotten good quality DID's from China? |
| 02:52.31 | _Sam-- | >> Is it illegal for SIP accounts to be sold outside of China? |
| 02:52.31 | _Sam-- | > |
| 02:52.31 | _Sam-- | > I read on this list a long time ago someone saying that it is illegal |
| 02:52.31 | _Sam-- | > to |
| 02:52.33 | _Sam-- | > offer DID's from China. From time to time someone sells DIDs, which are |
| 02:52.35 | _Sam-- | > |
| 02:52.37 | _Sam-- | > vanished after a while - some kind of disposable accounts. |
| 02:52.45 | PaulHuynh | also can we somehow use ssl with asterisk to secure the call |
| 02:53.07 | welles | _Sam--, meetme list cmd on asterisk can not work . |
| 02:54.00 | _Sam-- | welles: im sorry i cant be much help there, dont use it |
| 02:54.18 | puppet | _sam--: why didnt u use pastebin >< |
| 02:54.30 | _Sam-- | not enough action here to worry about it |
| 02:54.35 | _Sam-- | it didnt disrupt any conversations |
| 02:54.48 | puppet | well true but its still annoying when people do get back ;P |
| 02:55.19 | _Sam-- | if that is the most annoying thing you deal with all day, i think you've had a good day, hopefully :) |
| 02:57.19 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 02:58.09 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 02:58.28 | Telamon | What does the format_g726 module do? |
| 02:59.12 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 02:59.22 | sprnova | _Samm--... I figured it out.. I am such a Linux newbie.. I compiled the source linked to the 686 kernel source instead of 386 (the one I am running) |
| 03:00.06 | _Sam-- | always helps to symlink the right kernel! at least you are figuring it out |
| 03:00.11 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 03:00.12 | sprnova | _Sam--... I went to that site you told me.. lots of cool stuff up there.. but didn't find the right module version.. so I went back to my source code again. |
| 03:00.30 | _Sam-- | i saw the modules for your kernel there eaerlier but whatever works is fine by me |
| 03:01.00 | sprnova | it had the kernel version but I couldn't match the asterisk version (1.0.7) |
| 03:01.12 | sprnova | earliest was 1.0.9 |
| 03:01.32 | _Sam-- | what is uname -sr say? |
| 03:01.38 | _Sam-- | s/is/does/ |
| 03:01.51 | _Sam-- | i guess you dont need the s :) |
| 03:01.58 | sprnova | Linux 2.4.27-2-386 |
| 03:02.48 | _Sam-- | if you cant get the compiled one to work, i think the other zaptel modules will work fine. |
| 03:02.56 | sprnova | is there a startup file to insure ztdummy is run at next reboot? sorry. |
| 03:02.59 | _Sam-- | but in either event, i dont think it is going to solve your complaint |
| 03:03.06 | sprnova | it compiled and loaded ok |
| 03:03.49 | X-Rob | Oh man 'The IT Crowd' is fucking hilarious. |
| 03:04.13 | _Sam-- | i usually make a 'local' file in /etc/init.d that has some modprobes in there for booting |
| 03:04.18 | _Sam-- | i dont know how other people do it |
| 03:04.20 | X-Rob | sprnova, there's a zaptel.init file you can copy to /etc/init.d |
| 03:04.39 | sprnova | thanks gentlemen |
| 03:04.49 | X-Rob | as it's written, create an /etc/sysconfig/zaptel script and put in it 'TELEPHONY=yes' and 'MODULES=ztdummy' |
| 03:05.00 | X-Rob | (or, whatever other modules you're using) |
| 03:05.17 | Telamon | sprnova: What Linux distro are you using? |
| 03:05.18 | _Sam-- | then you still have to run update-rc.d |
| 03:06.01 | _Sam-- | rob how about we fight about phones instead? :) |
| 03:06.04 | sprnova | Telamon.. Debian 31r2 |
| 03:06.04 | X-Rob | However, being that he's uding 2.4.27, his timing's going to suck. Someone convince him to upgrade to a kernel released this century. |
| 03:06.17 | _Sam-- | i tried to convince him, rob |
| 03:06.35 | sprnova | X-Rob... ok.. ok... |
| 03:06.37 | Telamon | X-Rob: Yeah, but different distros do things in different ways. For instance, Gentoo puts all modules in /etc/modules.autoload.d/kernet-2.<whatever> |
| 03:06.57 | X-Rob | Telamon, really. Why do I care? |
| 03:06.58 | sprnova | just got Linux installed today. |
| 03:07.12 | X-Rob | sprnova, seriously. Download Asterisk@Home. |
| 03:07.12 | _Sam-- | robin_z: you almost echoed my identical words..."2.4? how about running something from the 21st century" |
| 03:07.19 | _Sam-- | damn nick completion |
| 03:07.19 | sprnova | its like pulling hair when you are used to *BSD |
| 03:07.30 | Telamon | X-Rob: *You* don't, but sprnova might. If he's not using a RedHat distro, it might not use the sysconfig dir. |
| 03:07.32 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 03:07.47 | _Sam-- | <i odnt have problems with 2.4 kernels...but for a new install...> |
| 03:07.50 | X-Rob | Telamon, that zaptel.init looks in /etc/sysconfig/zaptel. |
| 03:07.57 | X-Rob | That's why I said put it there. |
| 03:07.57 | _Sam-- | i still have 2 out 5 boxes on 2.4 |
| 03:08.49 | _Sam-- | Telamon: XratedRob knows all. |
| 03:09.42 | X-Rob | It's actually X for X-Chat. I used to have 'Rob', then I used to fire up X-Rob on linux. Then I never used 'Rob' for ages, and it was stolen. |
| 03:09.48 | Telamon | Heh, my apologies for doubting you, X-Rob. :) |
| 03:09.51 | _Sam-- | he has built in polygraphic abilities...he knew when i was lying, and he is 8000 miles away :) |
| 03:10.36 | X-Rob | _Sam--, nfi what you're talking about, but I'm guessing you said something blatantly wrong, and I was in a bad mood. |
| 03:10.46 | _Sam-- | nah, you and i went at it over the GXPs :) |
| 03:10.50 | X-Rob | Aaaah yeah |
| 03:11.04 | X-Rob | They're not bad now at all. |
| 03:11.10 | X-Rob | .2.9 is going _very_ nicely. |
| 03:11.15 | Telamon | Okay, here's a test for you: Why doesn't g726 work with my GXP-2000? 1.0.1.13 firmware. |
| 03:11.17 | X-Rob | still has spastic screen on the original batch |
| 03:11.18 | _Sam-- | they're not good, but they're just not as bad |
| 03:11.38 | _Sam-- | robin_z: how about you let me try the 2.9 and let you know how it works here? :) |
| 03:11.43 | _Sam-- | dammit, nick completion |
| 03:12.04 | Telamon | They are very good for the price. Of course they cost half of what any other VOIP business phone costs, so that might not be saying much... :) |
| 03:12.40 | X-Rob | Telamon, Is that like a riddle? I don't know, why DOESN'T g729 work with your gxp? |
| 03:12.43 | X-Rob | I say I say I say |
| 03:12.46 | _Sam-- | hahah |
| 03:12.58 | _Sam-- | Telamon: why dont you try the latest firmware |
| 03:13.01 | _Sam-- | see if it fixes your prob |
| 03:13.21 | _Sam-- | in my opinion, its a lot better than the .13 you're on |
| 03:13.24 | X-Rob | Ooh, he said 726. Ikky. |
| 03:13.57 | _Sam-- | X-Rob: where'd you get 29 |
| 03:13.58 | _Sam-- | ? |
| 03:14.02 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 03:14.20 | _Sam-- | 1.0.2.9 that is |
| 03:14.30 | Telamon | X-Rob: Heh, no, legitimate question. uLAW and GSM work fine, but with g726 I get a weird dialtone (staticky) and no audio when calling the PBX. |
| 03:14.38 | _Sam-- | i thought maybe the german place had it online |
| 03:14.38 | X-Rob | _Sam--, from Richard Huang@grandstream |
| 03:14.59 | X-Rob | Telamon, it's probably broken. 726 is useless. Here's a hint: |
| 03:15.04 | X-Rob | http://kvin.lv/pub/Linux/Asterisk/ |
| 03:15.06 | Telamon | I'm a little leary of going to a 1.0.2 since you can't go back to 1.0.1 after the upgrade. |
| 03:15.19 | _Sam-- | there is nothing in the newer firmware that would make you want to go back. |
| 03:15.24 | X-Rob | You want the 'icc' version, as they're faster. |
| 03:15.26 | _Sam-- | as long as you have a decent mac address |
| 03:15.44 | PaulHuynh | so which codec do you have to paid digium licensed? |
| 03:15.53 | X-Rob | Of course, that's only legal to use if you live in a country without software patents. |
| 03:15.54 | _Sam-- | i run the new firmware in 2 production environments...total of about 40 gxps |
| 03:16.04 | Telamon | "Decent mac address"? That referring to phone revision or something? |
| 03:16.12 | _Sam-- | that is referring to the mac address |
| 03:16.19 | _Sam-- | you know, macintosh |
| 03:16.21 | X-Rob | Telamon, the first 5000 gxp's used slightly different hardware. |
| 03:16.33 | X-Rob | This causes the screen to go spastic with the 1.2 firmware |
| 03:16.36 | _Sam-- | if you go to the gxp web status page.... |
| 03:16.39 | _Sam-- | and click on the phone status page |
| 03:16.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:16.42 | _Sam-- | you see the mac address |
| 03:17.04 | _Sam-- | that wasnt well worded |
| 03:17.12 | _Sam-- | if you log into your gxp 2000 via the web interface... |
| 03:17.14 | X-Rob | 000b8203a0e1 |
| 03:17.15 | _Sam-- | and click on the phone status page. |
| 03:17.18 | Telamon | X-Rob: Yeah, that's the problem. I live in Canada, I don't think we can use 729 without paying license fees. I'm looking for something better than GSM voice quality, but with some compression. 64kB/sec is a little high for phone if you aren't on a LAN. |
| 03:17.19 | X-Rob | ^^^ bad mac |
| 03:17.20 | _Sam-- | you can see your mac address. |
| 03:17.34 | X-Rob | Telamon, well pay the $10 then. It's worth it |
| 03:17.35 | _Sam-- | ulaw is 83kbps |
| 03:17.40 | _Sam-- | according to qwell. |
| 03:17.44 | _Sam-- | 64 + overhead |
| 03:17.51 | X-Rob | It's actually best to pay the $10, don't bother with the digium codec, and use the icc one on that page. It's better8) |
| 03:18.02 | *** join/#asterisk fdask (i=fdask@CPE0013d479c929-CM0011e6edd218.cpe.net.cable.rogers.com) |
| 03:18.14 | X-Rob | or speak to trixter on here, he's got a new licencing scheme happening |
| 03:18.36 | _Sam-- | rob does 2.9 fix anything that was b0rked in 8? |
| 03:18.42 | _Sam-- | or? give me some clues |
| 03:18.49 | X-Rob | My screen stays less broken for longer |
| 03:18.54 | Telamon | X-Rob: The licensing fee is only $10 for the whole server? Hell, might as well... |
| 03:18.54 | _Sam-- | lol |
| 03:18.59 | X-Rob | Telamon, $10 per channel |
| 03:19.12 | welles | _Sam--, i want two asterisk work together. they can now .when a call to asterisk A then asterisk A call Asterisk B. the call will continute 40 seconds and then hangup. why? |
| 03:19.16 | X-Rob | _Sam--, 2.8 is fine for most people, I've just been bitching about the screens 8) |
| 03:19.26 | Telamon | Ah, that's a bit more expensive... Still not bad though. |
| 03:20.27 | _Sam-- | welles: i have no idea.....is it exactly 40 seconds every single time? |
| 03:20.39 | welles | yes |
| 03:20.45 | _Sam-- | is it a timeout rule somewhere? |
| 03:20.56 | welles | i use mp3player app |
| 03:21.39 | _Sam-- | X-Rob: you use hints/blf on the 7 buttons on the right side of the GXP? |
| 03:21.47 | X-Rob | _Sam--, no, not all of 'em |
| 03:21.51 | X-Rob | I use snoms for hints |
| 03:22.02 | _Sam-- | i had problems with the gxp using hints on * |
| 03:22.11 | _Sam-- | * segfaulted |
| 03:22.11 | X-Rob | in fact, I haven't got any BLF's configured on the GXP's at the moment |
| 03:22.17 | X-Rob | Ooh really. That's not good. |
| 03:22.17 | _Sam-- | like to see if you could duplicate sometime |
| 03:22.23 | X-Rob | Will try now |
| 03:22.26 | _Sam-- | what * you have? |
| 03:22.31 | X-Rob | I'm using 1.2-svn |
| 03:22.37 | X-Rob | zaptel-trunk, libpri-trunk |
| 03:22.38 | _Sam-- | it only happens when there are many gxps doing stuuff, and having the gxps show the lights |
| 03:22.44 | _Sam-- | you cant breakt it with one single gxp |
| 03:23.05 | [av]bani | sam breaks gxps |
| 03:23.08 | _Sam-- | i had 10 extensions setup in asterisk to hint |
| 03:23.14 | X-Rob | ah. Well I've got a machine with 600 subscriptions with the snom-pickup patch, and that only segfaults due to bugs in the snom code |
| 03:23.15 | [av]bani | for fun and profit |
| 03:23.21 | _Sam-- | and the gxps (10) were setup to use BLF |
| 03:23.44 | _Sam-- | [av]bani : rob has 1.0.2.9! |
| 03:23.51 | X-Rob | When it crashes next, give me a backtrace |
| 03:23.53 | _Sam-- | but apparently it doesnt fix much |
| 03:24.01 | _Sam-- | i have the cores |
| 03:24.08 | _Sam-- | it crashed twice in one day, and i havent tried it since |
| 03:24.12 | [av]bani | _Sam--: gimme |
| 03:24.16 | _Sam-- | after i took the hints out / turned blf off... |
| 03:24.21 | _Sam-- | havent crashed since |
| 03:24.24 | X-Rob | So you haven't recompiled asterisk yet? |
| 03:24.35 | _Sam-- | i am running 1.2.4 release version |
| 03:24.45 | [av]bani | actually, i'd like to see BLA support for gxp/* |
| 03:24.56 | [av]bani | so our gxp's would work more like traditional pbx |
| 03:25.00 | X-Rob | gdb /tmp/core.asd /usr/sbin/asterisk |
| 03:25.09 | X-Rob | [av]bani, you mean like a key system |
| 03:25.15 | X-Rob | it won't happen for a while. |
| 03:25.23 | _Sam-- | hold on, i have some pastebins of the output from before from gdb |
| 03:25.34 | [av]bani | no, it's apparently on the list for next * release |
| 03:25.50 | [av]bani | since polycom suffers from issues related to buddy lists |
| 03:25.52 | X-Rob | BLA? Busy LINE Activity? |
| 03:25.59 | [av]bani | and BLA is the only way to work around it |
| 03:26.04 | _Sam-- | here's some of the gdb output: |
| 03:26.06 | _Sam-- | http://sam.pastebin.com/545854 |
| 03:26.55 | [av]bani | i'm sure one might be able to fake it using Pickup() and some scripting |
| 03:27.03 | Telamon | It would be nice if BLF didn't trigger when a phone went offline (ie, unplugged) but only when in use. |
| 03:27.06 | X-Rob | What do you mean by BLA? |
| 03:27.20 | [av]bani | X-Rob: http://lists.digium.com/pipermail/asterisk-users/2006-February/146983.html |
| 03:27.51 | *** join/#asterisk Cresl1n (n=matt@146.229.191.68) |
| 03:27.52 | X-Rob | Ah, fair enough |
| 03:27.56 | X-Rob | Crappy polycom problem 8) |
| 03:28.03 | [av]bani | well, its a generic issue |
| 03:28.11 | X-Rob | Not really, BLF works fine for everyone else |
| 03:28.20 | X-Rob | call pickup, ringing/inuse, works fine. |
| 03:28.30 | [av]bani | sorta, it doesnt let you do shared lines which is what one really wants |
| 03:28.53 | [av]bani | line status great, but i want to punch the line button and pick up that extension |
| 03:29.01 | X-Rob | Which, when you look at how there's no conecpt of a line _anywhere_ in asterisk, is gunna be damn hard to do. |
| 03:29.06 | X-Rob | Hang on |
| 03:29.13 | X-Rob | punch the _line_ button and pickup the _extension_? |
| 03:29.18 | X-Rob | how the hell are you planning on doing that? |
| 03:29.24 | [av]bani | quit being pedantic, you know what i mean |
| 03:29.33 | _Sam-- | pedantic = y? |
| 03:29.36 | X-Rob | push the extension button on the GXP and it dials **xtn, use Pickup for that |
| 03:29.50 | X-Rob | for snoms, they just do a reinvite and it just works |
| 03:30.07 | Telamon | pednaic = overly literal. |
| 03:30.27 | X-Rob | No, it's different. You want shared lines (eg, 'Joe, call on line 3') or extension status? |
| 03:30.39 | X-Rob | and extension pickup |
| 03:30.39 | [av]bani | reinvite fine, what about joining an active line |
| 03:30.53 | X-Rob | [av]bani, barge-in? |
| 03:30.59 | [av]bani | yes |
| 03:31.09 | X-Rob | That's harder than you think |
| 03:31.12 | [av]bani | the current 'solution' would be to throw everything into meetme rooms |
| 03:31.22 | Telamon | X-Rob: I think he wants the incoming call to go to a queue, the line lights monitor that queue and allow you to take the call. |
| 03:31.26 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 03:31.27 | X-Rob | There's 14000 lines of code to do conferencing. |
| 03:31.37 | X-Rob | but 'a lot' |
| 03:31.57 | websae | what do i add in my dial plan in order to be able to dial another person extension for example, i am 100, and want to dial 102...anyone :) |
| 03:31.58 | [av]bani | X-Rob: shared lines, as in extension 6800 exists on 12 phones |
| 03:32.00 | X-Rob | barge-in is conferencing, it's not the same as just splicing two wires in to the conversaion as you would with an analog pbx |
| 03:32.19 | [av]bani | the polycoms do it i guess |
| 03:32.32 | [av]bani | * does not yet |
| 03:32.39 | X-Rob | [av]bani, 6800,1,Dial(SIP/6801&SIP/6802&SIP6803)... but I see what you mean |
| 03:32.51 | Darwin35 | poloy want a hand job swqak |
| 03:33.03 | Telamon | websae: <exten>,1,Dial(SIP/<username>) |
| 03:33.03 | [av]bani | i think snom supports it also |
| 03:33.06 | X-Rob | websae, 'exten => 102,1,Dial(SIP/102)' |
| 03:33.12 | [av]bani | gxp "plans to" |
| 03:33.14 | websae | thanks |
| 03:33.26 | Egonis | I just re-compiled my kernel w/ CRC_blah = y, which has resolved some symbols.. however modprobing zaptel results in: Unknown symbol class_simple_device_add class_simple_destroy class_simple_device_remove class_simple_create -- I am using udev, coldplug, hotplug, and kernel-2.6.15-r1 |
| 03:33.48 | [av]bani | wonder what the 'ext' is for on the back of the gxp2k... sidecar? |
| 03:33.55 | X-Rob | [av]bani, yup |
| 03:34.02 | X-Rob | it's a 9600 serial link |
| 03:34.19 | [av]bani | same as the snom then |
| 03:34.22 | X-Rob | yup |
| 03:34.28 | _Sam-- | thats an svideo cable |
| 03:34.38 | _Sam-- | in case you want to watch xml on your monitor |
| 03:34.42 | X-Rob | although I think the snoms use a slower speed. |
| 03:34.42 | [av]bani | sam yea! HBO ! |
| 03:34.46 | _Sam-- | hahah |
| 03:35.00 | [av]bani | combined with xml support = megapr0n ph0ne |
| 03:35.42 | _Sam-- | [av]bani: the only way to show queue status on the gxp would be with xml minibrow? |
| 03:35.58 | johnsu01 | I'm just starting to get asterisk set up, trying to use kphone. When I try to make the test call my dialing 500, I lose audio after dialing 5. But from the console, it looks like the call is still going through. |
| 03:36.08 | johnsu01 | s/my/by/ |
| 03:36.20 | johnsu01 | oh. what a bot. |
| 03:36.53 | [av]bani | yah, dunno what we'd do without it |
| 03:37.18 | justinu | spend time with the wife? |
| 03:37.19 | justinu | lol |
| 03:37.24 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-cf1330eddd772505) |
| 03:37.26 | _Sam-- | hmmm that reminds me |
| 03:37.30 | johnsu01 | I don't need the NAT settings on when both kphone and the asterisk server are on the same computer, do I? |
| 03:37.52 | gaupe | nope |
| 03:38.19 | gaupe | anybody know how many characters the Snom 360 can show as callerid? |
| 03:38.29 | X-Rob | not enough 8) |
| 03:38.37 | gaupe | hehe |
| 03:39.17 | X-Rob | I set it to Number Only, otherwise it tends to go 004[02077155 0402077]155 |
| 03:39.20 | _Sam-- | X-Rob: can you tell your grandstream pals to check out our wiki page and all the bugs: http://www.voip-info.org/tiki-index.php?page=GXP-2000 |
| 03:39.25 | _Sam-- | bani found each and every bug himself |
| 03:39.27 | X-Rob | _Sam--, they have been |
| 03:39.50 | [av]bani | hopefully they fixed the fecking speakerphone/handset volume |
| 03:39.54 | X-Rob | They're not interested in open source tho |
| 03:40.03 | X-Rob | I've been asking them that since they _released_ the gxp |
| 03:40.20 | [av]bani | they also seem to have dropped ilbc |
| 03:40.23 | X-Rob | but, being that I was the one who promoted the pa1688 project, they prolly think of me as a bit biased. |
| 03:40.38 | X-Rob | Anyway |
| 03:40.42 | _Sam-- | how did the linksys wrt54 become open source? |
| 03:40.49 | [av]bani | _Sam--: the law required it |
| 03:40.52 | X-Rob | _Sam--, it always was linux based. |
| 03:40.59 | X-Rob | Anyway. I'm doing some AMP stuff. |
| 03:41.57 | [av]bani | well, grandstream purchased an echo can so that might make it hard to opensores it |
| 03:42.26 | [av]bani | though they could just hardcode it into the flash and then the opensores would just link when it loads |
| 03:42.26 | _Sam-- | there would be no financial benefit to them to do so. |
| 03:42.39 | [av]bani | well, it would mean us end users could fix the fecking bugs |
| 03:42.46 | [av]bani | since grandstream cant :/ |
| 03:42.47 | _Sam-- | that doesnt make them more money |
| 03:43.13 | _Sam-- | and if it were open sourced, then in theory, people could take similar code and try to make similar phones? |
| 03:43.13 | [av]bani | grandstream doesnt sell software |
| 03:43.25 | [av]bani | they already can, source or not |
| 03:43.34 | [av]bani | grandstream uses a reference design |
| 03:43.39 | gaupe | they have probably signed NDAs for the DSPs and such |
| 03:43.47 | [av]bani | nope |
| 03:43.53 | [av]bani | it all off the shelf, anyone can buy it |
| 03:44.04 | gaupe | ok |
| 03:44.15 | _Sam-- | i dont think there is any benefit to them open sourcing it...but maybe if it were, more people would buy the phones. |
| 03:44.18 | gaupe | well, buy it and build it :) |
| 03:44.23 | _Sam-- | the benefit is to the users |
| 03:44.28 | _Sam-- | which obviously come second |
| 03:45.05 | _Sam-- | as a company, in my own opinion, they dont gain anything if they were to release it. |
| 03:45.12 | [av]bani | well, i guess you would have to ask aredfox why they open sourced pa168, since it was obviously a dumb decision |
| 03:45.18 | _Sam-- | they would ulimately gain bttter firmware |
| 03:45.25 | *** join/#asterisk apardo (n=apardo@87.218.45.191) |
| 03:45.48 | [av]bani | well, end users contributed stuff to aredfox for pa1688 |
| 03:46.04 | [av]bani | localized IVR recordings, ilbc, bugfixes, etc |
| 03:46.13 | _Sam-- | the pa1688 is a dsp or what is it? |
| 03:46.40 | [av]bani | its an integrated hardware platform for phones built around an 8051 (8bit microcontroller) design |
| 03:47.12 | _Sam-- | how come it looks like no decent phones use that platform? |
| 03:47.32 | SkramX | in a dial command, i want to go to extension 999.. then have it wait, and insert the correct password (dtmf tones) for me,... can I do this? |
| 03:48.03 | X-Rob | SkramX, 'show application dial' on the asterisk console |
| 03:48.06 | [av]bani | http://www.soyogroup.com/products/proddesc.php?id=307 |
| 03:48.13 | [av]bani | shrug.. doesnt look any shitier than gxp :) |
| 03:48.21 | SkramX | X-Rob: right.. any particular commands or whay |
| 03:48.21 | Telamon | SkramX: http://www.asteriskguru.com/tutorials/authenticate.html |
| 03:48.26 | SkramX | Telamon: okay.. |
| 03:48.47 | SkramX | btu see, i want the dial command or i want asterisk to insert dtmf while a call is in progress |
| 03:48.53 | _Sam-- | that thing doesnt look near as fucntional as gxp |
| 03:48.57 | _Sam-- | not as many buttons |
| 03:49.22 | _Sam-- | in fact, i crap on that soyo phone! |
| 03:49.32 | [av]bani | a phone.. for me to poop on? |
| 03:49.35 | _Sam-- | lol |
| 03:49.47 | _Sam-- | it looks like something that belongs in my moms house in her kitchen or something |
| 03:50.22 | websae | where's a good place to buy DIDs? |
| 03:50.26 | websae | anyone have any suggestions? |
| 03:50.35 | _Sam-- | dude you just figured out how to dial an extension |
| 03:50.38 | websae | I tried didx, but never got an email to confirm my account |
| 03:50.56 | _Sam-- | the chances of you successfully setting up a did are not great. |
| 03:50.57 | Telamon | Hmm, not sure I know what you mean. You want when someone dials extention 999 it dials out and enters a password? |
| 03:51.06 | _Sam-- | if you could just now setup a way to dial extension 102 |
| 03:51.11 | [av]bani | _Sam--: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a0080159de2.html |
| 03:51.30 | websae | lol |
| 03:51.44 | [av]bani | _Sam--: isnt that just the bestest phone evar? |
| 03:51.44 | _Sam-- | websae : i think you can do it, just busting your balls. |
| 03:52.06 | [av]bani | http://www.reseaudirect.com/catalog/images/Cisco%20IpPhone%207902G%20G.jpg |
| 03:52.12 | websae | where is a good place to buy DIDs? |
| 03:52.16 | [av]bani | and its CISCO!!1! |
| 03:52.22 | websae | that's all i am asking |
| 03:53.02 | _Sam-- | i need to get a high end phone for my own desk |
| 03:53.11 | Telamon | websae: Checkout http://voip-info.org/wiki/view/DID it lists a bunch of places. I get mine from our phone company, so I don't have any opionions on the matter. |
| 03:53.13 | [av]bani | whats high end mean |
| 03:53.17 | [av]bani | stilts? |
| 03:53.38 | _Sam-- | i HAVE to have an xml minibrowser :) |
| 03:53.47 | [av]bani | whats your budget? |
| 03:53.53 | _Sam-- | unlim. |
| 03:53.58 | _Sam-- | its for my own desk |
| 03:54.01 | [av]bani | $385? |
| 03:54.05 | _Sam-- | sure |
| 03:54.06 | websae | do you have a PRI telamon? |
| 03:54.10 | [av]bani | cisco 7970g it is |
| 03:54.14 | Telamon | websae: Yes. |
| 03:54.22 | websae | where are you located? |
| 03:54.38 | Telamon | websae: Prince Edward Island. Eastern end of Canada. |
| 03:54.40 | _Sam-- | what do you do with the pretty color screen? |
| 03:54.44 | _Sam-- | display a jpg of your wife? |
| 03:54.45 | tronix | _Sam--: if it's unlimited, go for the 7985. ;) |
| 03:55.15 | [av]bani | _Sam--: i thought you would know by now |
| 03:55.17 | tronix | though more seriously, 7970G sounds pretty nice. I like my 7960G but sure would be nice to have nicer contrast with color |
| 03:55.22 | [av]bani | _Sam--: comeon, you can figure it out |
| 03:55.33 | [av]bani | starts with p, ends with n and has a 0 in it |
| 03:55.37 | _Sam-- | my light bulb is not buring at this hour |
| 03:55.55 | [av]bani | tronix: backlight |
| 03:56.06 | [av]bani | i would get a 7960g if it had a backlight |
| 03:56.20 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 03:56.22 | _Sam-- | that screen on the 7970 is a touch screen? |
| 03:56.26 | [av]bani | yes |
| 03:56.33 | _Sam-- | how do you program the on screen items? |
| 03:56.41 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
| 03:56.45 | [av]bani | http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a00801c9638.html |
| 03:56.48 | [av]bani | xml |
| 03:57.00 | [av]bani | touch screen pr0n! |
| 03:57.21 | [TK]D-Fender | Plug & play! |
| 03:58.36 | johnsu01 | It seems like it's probably some kind of dtmf setting, but I've followed the dtmf settings on the wiki for kphone... |
| 03:59.05 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 03:59.40 | _Sam-- | [av]bani : that phones does SIP off the shelf? |
| 04:01.23 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 04:02.08 | orlock | Does anybody know what the grandstream's ext port is for? |
| 04:02.12 | orlock | is it serial? |
| 04:02.25 | _Sam-- | it hooks up to a dvd player to record calls |
| 04:02.33 | _Sam-- | svideo |
| 04:02.40 | _Sam-- | <just kidding> |
| 04:03.33 | tuxinator_linux | Does _Sam-- stand for Smart Ass Man? |
| 04:03.45 | tuxinator_linux | ;-) |
| 04:03.46 | [av]bani | _Sam--: no, 7970g is sccp only |
| 04:03.55 | [av]bani | _Sam--: but Qwell swears by it |
| 04:04.06 | tuxinator_linux | Yes he does |
| 04:04.18 | [av]bani | so if you get a 7970g then you get 24/7 support from Qwell |
| 04:04.23 | _Sam-- | lol |
| 04:04.31 | tuxinator_linux | Asterisk is suppose to play pretty well with sccp now |
| 04:04.36 | _Sam-- | sorry to be a dumbass, but how do yo do get SIP , or how do you make linux speak sccp? |
| 04:04.59 | tuxinator_linux | you need a smartnet agreement with Cisco either way |
| 04:05.12 | tuxinator_linux | chan_sccp for asterisk |
| 04:05.18 | _Sam-- | i see it now |
| 04:05.23 | tuxinator_linux | SIP image from cisco for phone |
| 04:05.38 | _Sam-- | if you used chan_sccp, you would have sccp.conf? |
| 04:05.45 | tuxinator_linux | I ordered my smartnet a while ago, still waiting for access |
| 04:06.12 | tuxinator_linux | I think I will be trying both, SIP and SCCP |
| 04:06.14 | [av]bani | there is no sip for 7970 |
| 04:06.20 | tuxinator_linux | nope |
| 04:06.32 | [av]bani | you dont need smartnet |
| 04:06.46 | tuxinator_linux | [av]bani: Do tell |
| 04:07.20 | tuxinator_linux | well, I suppose you don't need it if you're doing chan_sccp |
| 04:07.28 | [av]bani | tuxinator_linux: and since there is no sip for 7970... |
| 04:07.50 | tuxinator_linux | but receiving updats is kinda nice, not required though |
| 04:08.07 | [av]bani | yeah, maybe a couple years ago it might ahve been an issue |
| 04:08.14 | _Sam-- | thanks for all the info...it put me right to sleep....night. |
| 04:08.23 | tuxinator_linux | night |
| 04:10.08 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 04:13.43 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 04:13.46 | Nugget | what's the state of chan_sccp these days? is it viable? |
| 04:13.56 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
| 04:13.56 | [av]bani | i guess its usable |
| 04:15.18 | orlock | Hmmm |
| 04:15.18 | *** join/#asterisk blackgecko (n=blackgec@201.144.217.221) |
| 04:15.19 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 04:15.38 | orlock | i'm trying to connect to upstream sipserver with a grandstream, and its not registereing.. any suggestions? |
| 04:16.27 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 04:16.47 | Egonis | When loading zaptel, I get the message 'unable to open master device '/dev/zap/ctl' although the module is loading |
| 04:17.20 | mogorman | you probably dont have udev rules correct |
| 04:17.55 | Egonis | mogorman: How do I check that? :) |
| 04:18.37 | X-Rob | or /etc/zaptel.conf isn't set up right. |
| 04:18.39 | mogorman | well is there /dev/zapctl or /dev/zap/ctl |
| 04:18.46 | Egonis | mogorman: nope |
| 04:19.01 | Qwell | Nugget: it works really well |
| 04:19.02 | Egonis | X-Rob, mogorman: /dev/udev/rules.d/10-zaptel.rules exists |
| 04:19.12 | blackgecko | anyone has experience using a2billing ??? |
| 04:19.32 | mogorman | what card do you have? |
| 04:19.59 | *** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au) |
| 04:20.17 | Egonis | mogorman: Sangoma A200, haven't started w/ that just yet tho |
| 04:20.52 | mogorman | oohhh.... |
| 04:20.57 | emergion | Hello all, Just wondering how many calls can a DID Simultaneously handle a VoipSP said pretty much unlimited but wouldnt it give an engaged signale when a call is in process |
| 04:20.59 | mogorman | well i imagine you might have some wanpipe issue |
| 04:21.10 | Egonis | mogorman: I haven't loaded that yet... so probably |
| 04:21.14 | mogorman | but /me only really knows digium stuff |
| 04:21.19 | Egonis | mogorman: just trying to get zaptel loaded properly first |
| 04:21.31 | mogorman | well you can modprobe zaptel |
| 04:21.56 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:22.01 | Egonis | mogorman: did that, it's the init.d/zaptel which complains about no /dev/zap/blah stuff, but the module loads fine, just doesn't create any devices, at all |
| 04:22.34 | mogorman | well its probably not loading properly |
| 04:22.38 | mogorman | dmesg will tell you why |
| 04:22.45 | Egonis | mogorman: no complaints in dmesg at all about it |
| 04:22.55 | MikeJ[Laptop] | hello Mr. O... |
| 04:23.00 | mogorman | well id be happy to take a look for ya |
| 04:23.05 | mogorman | hello Mr. J |
| 04:23.20 | JunK-Y | hey mogorman ! |
| 04:23.31 | mogorman | and Mr. Y |
| 04:23.43 | [av]bani | a200, oooo |
| 04:23.46 | [av]bani | shiny |
| 04:23.47 | JunK-Y | DrunK-Y! |
| 04:24.00 | MikeJ[Laptop] | heh |
| 04:24.02 | Egonis | mogorman: it simply says -- Zapata Telephony Interface Registered on Major 196, nothing more |
| 04:24.55 | mogorman | hmm well it should be there then |
| 04:25.03 | Egonis | my point exactly.. :) |
| 04:25.19 | mogorman | well i can look at it for you |
| 04:25.21 | Egonis | maybe I'll try re-emerging udev, just for kicks |
| 04:26.31 | MikeJ[Laptop] | Egonis, string and cans? |
| 04:26.36 | mogorman | reverse ssh? |
| 04:26.56 | Egonis | I suppose |
| 04:27.00 | Egonis | MikeJ[Laptop]: Not a bad idea |
| 04:27.02 | mogorman | lol |
| 04:27.20 | mogorman | you are silly, sorry i cant help you |
| 04:27.27 | Egonis | mogorman: I just upgraded udev, do I need to tell udev.conf that I have a 10-zaptel.rules file? |
| 04:27.38 | mogorman | you just need the rules |
| 04:27.39 | Egonis | yes, yes I am |
| 04:27.42 | mogorman | and it should make it |
| 04:27.52 | mogorman | and it shouldnt matter |
| 04:27.52 | Egonis | mogorman: So having the file makes it aware of the rules, right? |
| 04:28.00 | mogorman | as gentoo has had it in their rules for a few months now |
| 04:28.07 | mogorman | at least the digium cards |
| 04:28.12 | mogorman | so zaptel should be there too |
| 04:28.36 | Egonis | okay, time for a reboot to refresh udev, and other broken toys |
| 04:28.38 | Egonis | brb |
| 04:28.59 | *** join/#asterisk L|NUX (n=linux@202.5.145.57) |
| 04:32.04 | X-Rob | he could have just restarted udevd |
| 04:32.38 | mogorman | i know that |
| 04:32.39 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 04:32.42 | mogorman | and you know that |
| 04:32.45 | Egonis | no dice.. :( |
| 04:32.46 | mogorman | but he did not know that |
| 04:33.27 | Egonis | well, the zaptel module is loaded! :) but no devices in /dev.... should I disable hotplug? |
| 04:33.38 | mogorman | no it should make em |
| 04:33.44 | mogorman | you could manually make dev nodes yourself |
| 04:33.47 | mogorman | but that is lame |
| 04:33.50 | X-Rob | not with udev |
| 04:33.53 | X-Rob | udev does it all |
| 04:33.56 | mogorman | yes you can |
| 04:34.03 | mogorman | you can make dev nodes amongst the udev tree |
| 04:34.05 | mogorman | its not correct |
| 04:34.09 | mogorman | but you can do it |
| 04:34.19 | [av]bani | anyone used the zoom 5801 yet? |
| 04:34.41 | Egonis | X-Rob: I'm using udev |
| 04:34.58 | Egonis | X-Rob: but it's being gay |
| 04:35.32 | mogorman | how can udev be gay???? |
| 04:35.55 | Egonis | I would love to make the dev nodes just to shut it up, but I also have my sangoma card to get working (wanrouter) -- udev is being gay because it's wearing a scarf |
| 04:36.22 | mogorman | ???? |
| 04:36.25 | Egonis | lol |
| 04:36.36 | Egonis | I have no idea.. :) |
| 04:38.26 | Egonis | so what do you suggest? manually create dev nodes? how do I do so/ |
| 04:38.36 | mogorman | i would not suggest that |
| 04:38.56 | Egonis | so where should I start with troubleshooting udev/coldplug/hotplug |
| 04:38.57 | Egonis | lol |
| 04:39.00 | Egonis | yeah, that too |
| 04:39.13 | Egonis | please excuse my nub-ness |
| 04:39.23 | Egonis | but learnin this will help me show others |
| 04:39.30 | Egonis | learnin(g) |
| 04:40.10 | mogorman | well the thing is |
| 04:40.15 | mogorman | there is not much to screw up |
| 04:40.24 | mogorman | it should just work TM |
| 04:40.43 | Egonis | which is what really gets me |
| 04:42.02 | Egonis | I'm running a fresh as pie gentoo install, I've installed gentoo like 50 times on 50 different servers... this is my first time setting up a box w/ a sangoma card... but I haven't even started with that part yet |
| 04:43.18 | mogorman | ahh canadia |
| 04:46.05 | mogorman | you dont refer to your home land as canadia? |
| 04:46.06 | [TK]D-Fender | Egonis : Did you follow the zaptel compile reqs for it? |
| 04:48.14 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:49.08 | mogorman | well he can load it [TK]D-Fender |
| 04:49.11 | mogorman | so its built |
| 04:50.24 | [TK]D-Fender | Not sure if it implies he did the POST WanPipe recompile.... |
| 04:50.37 | mogorman | no he hasnt |
| 04:50.43 | mogorman | but he is just trying to load zaptel |
| 04:50.44 | *** join/#asterisk sonicGB- (n=Miranda@138.25.71.101) |
| 04:51.29 | [TK]D-Fender | I'm wondering if its being "incomplete" from the process might interfere with it loading... |
| 04:51.32 | [TK]D-Fender | just a though |
| 04:52.08 | *** join/#asterisk BingoPajama (n=zcw100@68-233-16-90.stcgpa.adelphia.net) |
| 04:52.30 | mogorman | shouldnt |
| 04:53.26 | sonicGB- | I have hw_ip_phone <-----SIP----> asterisk(1) <-----IAX2-----> asterisk(2) working fine. I want to dial a number on hw_ip_phone and have asterisk(1) pass the call to asterisk(2) (that much works) and have asterisk(2) Dial(SIP/username@host,,r). asterisk(2) says "Unable to create channel of type 'SIP' (cause 3 - No route to destination)". I've googled/faq'ed, but no clear answer as to what I'm doi |
| 04:53.26 | sonicGB- | ng wrong. Any suggestions please? |
| 04:53.37 | BingoPajama | Anyone know were I can get a copy of the cisco 7912 firmware? |
| 04:54.29 | mogorman | dont you buy it from cisco? |
| 04:55.34 | BingoPajama | You'd think that cisco would make it easy to take your money but they don't. |
| 04:55.56 | sonicGB- | It's been a long time since I played with anything Cisco, but IIRC, for the routers/switches at least, there's a 'tac' web site that has firmware for everything for download. Catch is that you need to authenticate to get on to that site, and that you have to get from Cisco (which I always did via a third party reseller) |
| 04:55.57 | Egonis | [TK]D-Fender: yes |
| 04:56.18 | Egonis | [TK]D-Fender: nope, didn't touch wanpipe yet... I want zaptel working 100% first |
| 04:56.52 | Egonis | I figure that if the devnodes aren't created w/ zaptel, why would they be w/ wanpipe? |
| 04:57.51 | justinu | um, zaptel won't work without wanpipe |
| 04:57.58 | mogorman | ????? |
| 04:58.07 | mogorman | justinu that is the dumbest thing i have heard |
| 04:58.09 | mogorman | tonight |
| 04:58.09 | [TK]D-Fender | Egonis : You need to follow the procedure for it. Zaptel first, then Wanpipe, THEN ZAptel AGAIN |
| 04:58.10 | justinu | wanpipe is the foundation layer that creates the zaptel look alike |
| 04:58.17 | justinu | it's an abstraction layer ;) |
| 04:58.26 | mogorman | it doesnt matter, he is just trying to load zaptel |
| 04:58.30 | mogorman | not anything else |
| 04:58.36 | Egonis | [TK]D-Fender: emerge zaptel a second time, you mean? |
| 04:59.10 | Egonis | [TK]D-Fender: I haven't even touched wanpipe yet.. like, I didn't untar the wanpipe .tgz |
| 04:59.39 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
| 04:59.41 | masked | hi |
| 04:59.46 | [TK]D-Fender | not emerge... recompile. You should do it from source. Wanpipe patches Zaptel in order to function. |
| 05:00.04 | masked | has anyone here used telstra call waiting via a fxo? |
| 05:00.23 | BingoPajama | Ya, I asked my university departments admin. Maybe he'll get it to me maybe he won't. |
| 05:00.30 | masked | X-Rob maybe? |
| 05:00.43 | justinu | dude, you need to get the winpipe stuff installed |
| 05:00.54 | justinu | it needs the zaptel source, then it patches it |
| 05:00.57 | justinu | then it installs it |
| 05:01.04 | justinu | it's like automagic |
| 05:01.10 | BingoPajama | Anyone have any luck building chan_bluetooth? |
| 05:01.11 | [TK]D-Fender | yup |
| 05:01.30 | mogorman | must resist urge to troll |
| 05:01.53 | justinu | it's actually a very slick setup |
| 05:02.03 | justinu | considering their products work with other open source pbx softwares ;) |
| 05:02.09 | Egonis | just a thought, I will try -devfs26 in package.use, just for kicks |
| 05:02.31 | justinu | egonis, you can continue down that line as long as you want... without wanpipe, it won't help you. |
| 05:02.33 | Egonis | do I need bri? |
| 05:03.02 | Egonis | justinu: Yes, however... if it doesn't create device nodes for zaptel... like, it doesn't have the ability -- where will I be when I try to do wanpipe? |
| 05:03.17 | justinu | wanpipe is a layer under zaptel |
| 05:03.19 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 05:03.58 | X-Rob | masked, it's a pain in the arse, sending a flash.. I've never done it, but it is possible. |
| 05:04.10 | Egonis | FYI: Setting -devfs26 in package.use breaks zaptel... it now says: Unknown symbol -- class_simple_device_add, etc |
| 05:04.34 | mogorman | most people Egonis build everything asterisk related straight from source |
| 05:04.43 | mogorman | dont use gentoo debian redhat packages |
| 05:04.50 | *** join/#asterisk xyklopz (n=xyklopx@216-91-89-21.biltmorecomm.com) |
| 05:04.51 | Egonis | mogorman: maybe I should go that route |
| 05:05.03 | X-Rob | kkkkkkk |
| 05:05.09 | X-Rob | wups, vi'ing in the wrong window |
| 05:05.31 | xyklopz | can someone clarify something for me ... if I have a call coming in via a SIP softphone, the SIP channel is used by I don't quite understand the relationship to the contexts ... |
| 05:05.56 | *** part/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 05:06.06 | xyklopz | where do I define which context is for say local extensions versus calls which must be proxied to a 3rd-party voip provider |
| 05:06.23 | IronHelix | you make the call from the softfone |
| 05:06.29 | IronHelix | the call goes in whatever context the phone is in |
| 05:06.34 | IronHelix | from there its matched by what you dial |
| 05:07.08 | xyklopz | "whatever context the phone is in" => meaning sip vs h.323 vs iax? |
| 05:07.17 | IronHelix | no |
| 05:07.19 | IronHelix | sip is a protocol |
| 05:07.23 | IronHelix | a signalling method |
| 05:07.24 | xyklopz | right |
| 05:07.26 | xyklopz | i know |
| 05:07.28 | xyklopz | clear text |
| 05:07.29 | xyklopz | INVITE |
| 05:07.31 | IronHelix | as is 323 and iax |
| 05:07.31 | xyklopz | REGISTER |
| 05:07.32 | xyklopz | etc |
| 05:07.37 | IronHelix | exactly, along with RTP for audio |
| 05:07.50 | IronHelix | in sip.conf, there is for each phone a context= |
| 05:07.53 | xyklopz | I'm actually researching a method of active RTP injections |
| 05:08.05 | IronHelix | that tells you waht [context] in extensions.conf that phone is in |
| 05:08.19 | IronHelix | a context is a group of exten's (things you can dial) as well as any other included contexts |
| 05:08.29 | IronHelix | extens can include patterns and other such things |
| 05:08.37 | sprnova | having problems setting up Music on hold.. I get a bunch of weird sounds.. I think it is playing but it sounds all wierd.. how can I tell if the zfdummy driver is being used?? I loaded it via modprobe and restarted asterisk. |
| 05:09.21 | IronHelix | ie exten => _1NXXNXXXXXX,1,Dial(SIP/myVOIPprovider/${EXTEN}) would match any 11 digit (american long distance) number and dial it out on your sip trunk |
| 05:10.06 | xyklopz | I don't quite understand the concept of a "trunk" |
| 05:10.21 | IronHelix | trunk is a channel that can carry more than one call at once |
| 05:10.31 | IronHelix | in this case i mean it as what you connect to the pstn with |
| 05:10.49 | IronHelix | in the above example, your account with myVOIPprovider is you 'sip trunk' |
| 05:10.56 | xyklopz | i c |
| 05:11.25 | xyklopz | how do you associate incoming trunk calls then? |
| 05:11.35 | xyklopz | do they use the default context |
| 05:11.45 | xyklopz | or is that specified when you configure the trunk |
| 05:12.08 | IronHelix | you can use a dial prefix to connect to another voip provider, for example: exten => _91NXXNXXXXXX,1,Dial(SIP/myOTHERbetterVOIPprovider/${EXTEN:1}) exten:1 strips off the 0 |
| 05:12.25 | IronHelix | asterisk does not see much of any difference between your softphone and your voip provider |
| 05:12.35 | IronHelix | they are both sip channels to * |
| 05:13.02 | IronHelix | so the context thing still applies- you just want the incoming calls in a different context |
| 05:13.24 | IronHelix | does that make sense? |
| 05:13.43 | IronHelix | you can use the exten 's' for where no pattern can be matched |
| 05:13.45 | IronHelix | example: |
| 05:13.57 | IronHelix | exten => s,1,Playback(welcome) |
| 05:14.05 | IronHelix | exten => s,2,Dial(SIP/1234) |
| 05:14.12 | IronHelix | exten => s,3,Voicemail(1234) |
| 05:14.33 | IronHelix | toss that in a context, and associate a sip channel that might get an incoming call to that context |
| 05:15.08 | IronHelix | whenever a call comes in from that account/provider/etc, it will start by playing a welcome message, then try to send the call to SIP/1234 (defined in sip.conf) then go to voicmeail |
| 05:15.20 | IronHelix | minor typo- it should actually be Dial(SIP/1234,20) |
| 05:15.27 | IronHelix | the ,20 says to only ring 20sec |
| 05:15.33 | IronHelix | sorry if im overloading or not helping :\ |
| 05:17.07 | xyklopz | but okay, how do I say dial * from a softphone |
| 05:17.22 | xyklopz | w/o dialing an extension... or am I looking at it wrong? |
| 05:17.27 | xyklopz | I can't just type the IP ... |
| 05:17.41 | xyklopz | do I setup an extension = my phone #? |
| 05:17.54 | IronHelix | yeah you are... assuming the softphone is registered to * and stuff, it has no need to dial * because whatever it dials is through * |
| 05:17.54 | xyklopz | NXXXXXXXXX |
| 05:18.16 | IronHelix | how bout this |
| 05:18.18 | xyklopz | but that's how you would get it to be able to dial * directly |
| 05:18.32 | IronHelix | but what does * do when you dial it? |
| 05:18.42 | IronHelix | thats what the context/extension thing is |
| 05:19.06 | IronHelix | asterisk wont on its own answer your call and wait to be told what to do |
| 05:19.16 | IronHelix | there has to already be a path for your call in place |
| 05:19.18 | IronHelix | how bout this |
| 05:19.23 | IronHelix | tell me exactly what you want to setup |
| 05:19.27 | IronHelix | and i'll tell you what you need to do |
| 05:20.42 | xyklopz | I have a few softphones (future wifi phones) that I want to setup with individual extensions & voicemail (etc. etc.). after I get this configured and I understand it better, I'd like to move my exisitng # to vonage and then configure * as a client and do the turnk forwarding as we mentioned before |
| 05:21.04 | IronHelix | pretty common setup |
| 05:21.07 | IronHelix | one first gotcha |
| 05:21.12 | xyklopz | Yeah ... nothing to complex |
| 05:21.12 | IronHelix | DONT USE VONAGE! |
| 05:21.17 | IronHelix | ~vonage |
| 05:21.18 | jbot | rumour has it, vonage is a bunch of monkeys |
| 05:21.22 | xyklopz | they only reason ... keep my # |
| 05:21.29 | xyklopz | i don't know about others that allow this |
| 05:21.43 | IronHelix | vonage will not work as a SIP trunk under any circumstances. |
| 05:21.55 | IronHelix | vonage will send you a device called an ATA |
| 05:21.56 | IronHelix | ~ata |
| 05:21.58 | jbot | i heard ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 05:22.10 | IronHelix | the ATA is locked to their network and your account |
| 05:22.17 | IronHelix | and the only way to access your account si through the ata |
| 05:22.32 | xyklopz | okay ... I have a sipphone account |
| 05:22.43 | xyklopz | and they are only $35/yr for a # |
| 05:22.50 | IronHelix | its fine for ppl that dont use asterisk- they plug a phone into the box and they're done, but for asteirsk vonage is a bad way to go. |
| 05:22.50 | xyklopz | and 2c/min US |
| 05:23.01 | IronHelix | thats not a bad rate. There are many such providers |
| 05:23.12 | IronHelix | also companies like vonage but that allow byod (bring your own device) |
| 05:23.16 | xyklopz | i haven't seen anything comparable to the $35/year |
| 05:23.18 | IronHelix | asterisk counts as 'your own device' |
| 05:23.27 | xyklopz | right |
| 05:23.48 | IronHelix | keep in mind tho the 35/year counts only for the number, you start paying when you make calls on it |
| 05:23.51 | xyklopz | so I'm staring my configs fron scratch |
| 05:23.57 | xyklopz | huge configs scare me |
| 05:23.57 | IronHelix | it really depends on how many minutes you use |
| 05:24.06 | xyklopz | not incoming |
| 05:24.17 | xyklopz | only outgoing @ 2c |
| 05:24.19 | IronHelix | you can safely delete a good portion of the stuff in the default configs |
| 05:24.22 | IronHelix | yeah i mean outgoing |
| 05:24.29 | xyklopz | i barely use it |
| 05:24.31 | IronHelix | if you dont make a lot of calls then thats a sweet deal for you |
| 05:24.35 | xyklopz | to call out ... mostly in |
| 05:24.37 | xyklopz | :-) |
| 05:24.38 | IronHelix | hehe |
| 05:24.43 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:24.43 | xyklopz | so it's perfect |
| 05:24.46 | IronHelix | also if you want a number to play around with |
| 05:24.58 | IronHelix | www.ipkall.com will give you a free DID (phone number), only 360 area code tho |
| 05:25.15 | IronHelix | and many other providers can port your number, try teliax broadvoice quantumvoice |
| 05:25.20 | IronHelix | all support byod |
| 05:25.22 | xyklopz | so what is necessary in my sip.conf |
| 05:25.26 | IronHelix | i use QV, they have good service |
| 05:25.27 | IronHelix | anyway |
| 05:25.29 | IronHelix | steps you need |
| 05:25.38 | xyklopz | I have ... |
| 05:25.55 | sprnova | I want to use ztdummy.. but when I do a lsmod zaptel shows up as being used by ztdummy and ztdummy is (unused) is this normal? |
| 05:26.06 | IronHelix | thats fine |
| 05:26.14 | sprnova | thx |
| 05:26.18 | IronHelix | 1. create each SIP provider a sip.conf entry, use sip show registry to make sure they register ok |
| 05:26.34 | IronHelix | 2. create sip.conf entries for each of your softphones that you want to use |
| 05:26.36 | IronHelix | ~softphone |
| 05:26.38 | jbot | something that should be drug out into the street and shot |
| 05:26.57 | IronHelix | so consider buying hardphones if you have ethernet- i can recommend some if you want. |
| 05:27.12 | IronHelix | once you have the sip entries |
| 05:27.19 | xyklopz | wait wait |
| 05:27.23 | xyklopz | "SIP provider" ...? |
| 05:27.30 | IronHelix | ie sipphone.com etc |
| 05:27.32 | IronHelix | if you have any |
| 05:27.45 | IronHelix | sipphone has a default config for sip.conf |
| 05:27.59 | xyklopz | in sip.conf? |
| 05:28.03 | xyklopz | the sample |
| 05:28.23 | IronHelix | you can dump all the default configs at the end of the thing |
| 05:28.31 | IronHelix | just keep the top thing witht he variables and stuff |
| 05:31.04 | IronHelix | http://www.voip-info.org/wiki/view/SIPphone has a thing of the stuff to put |
| 05:31.39 | xyklopz | in the [authentication], do I have to add entiries for the REGISTER messages, I was getting errors before on that Username/auth name mismatch |
| 05:31.55 | xyklopz | aka <user>:<secret>@mydomain.com ?? |
| 05:32.49 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 05:32.56 | IronHelix | well in the [general] section at the top you'll need to do something like register => yoursipphonenumber:yourpass@sipphone |
| 05:33.02 | IronHelix | then put [sipphone] |
| 05:33.06 | IronHelix | and add the stuff like it has there |
| 05:33.18 | IronHelix | what should be [general] and [sipphone] are just underlined on that page |
| 05:34.56 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 05:37.11 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 05:37.27 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 05:37.49 | jeebusroxors | hey there, can anyone recommend a web admin interface? Im pulling my hair out with voiceone... |
| 05:38.18 | [av]bani | ahahaha. retarded workaround for grandstream limitation.... |
| 05:38.39 | IronHelix | ? |
| 05:38.58 | [av]bani | more autoprovisioning silliness from grandstream |
| 05:39.07 | IronHelix | how so |
| 05:39.19 | [av]bani | but i have a workaround :) |
| 05:39.25 | krischnoff | jeebus: what is voiceone? |
| 05:39.48 | IronHelix | do tell... |
| 05:39.53 | jeebusroxors | krischnoff; its a web admin interface.... |
| 05:40.16 | krischnoff | for a * GUI check out thirdlane.com |
| 05:40.20 | IronHelix | ( i made the mistake of rolling 5 of them at a small site, after screwing with their cfg generator for a while i decided it would be faster to set them up manually ) |
| 05:40.22 | *** join/#asterisk YoMama (n=tchen@c-68-61-101-36.hsd1.mi.comcast.net) |
| 05:40.40 | krischnoff | it's also a company, that's why I was confused |
| 05:40.53 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
| 05:40.59 | xbmodder_lappy | how can I see active channels |
| 05:41.00 | xbmodder_lappy | ? |
| 05:41.02 | jeebusroxors | krischnoff; thanks, |
| 05:41.04 | IronHelix | show channels |
| 05:41.04 | YoMama | anyone here have a GXP-2000? |
| 05:41.06 | IronHelix | yeah |
| 05:41.12 | krischnoff | PBX Manager is a webmin plugin |
| 05:41.17 | YoMama | hey IronHelix..ltns |
| 05:41.19 | krischnoff | from thirdlane.com |
| 05:41.23 | IronHelix | im staring at the garbled upside down display right now :D |
| 05:41.30 | IronHelix | sup yomama |
| 05:41.31 | IronHelix | hwos life |
| 05:41.44 | YoMama | IronHelix: eh...alright...haven't had time to monkey with * as much as I used to |
| 05:41.51 | YoMama | IronHelix: were u saying u had a GXP-2000? |
| 05:41.55 | IronHelix | yeah i do |
| 05:42.04 | YoMama | IronHelix: did u upgrade it to 1.0.2.8? |
| 05:42.10 | IronHelix | yuppers |
| 05:42.26 | YoMama | IronHelix: k...why are the files named gxp2000a and boot55a? |
| 05:42.34 | IronHelix | new bootloader |
| 05:42.36 | YoMama | instead of gxp2000.bin and boot55.bin? |
| 05:42.42 | IronHelix | maybe new firmware format or something |
| 05:42.50 | YoMama | well, my 1.0.1.13 phone doesn't look for the a |
| 05:42.56 | YoMama | did u haveta rename the files? |
| 05:43.06 | jeebusroxors | krischnoff; got anything free? ;) |
| 05:43.26 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 05:43.30 | [av]bani | the beta firmware uses *a.bin suffix |
| 05:43.36 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 05:43.49 | IronHelix | yomama- you gotta get 1.0.2.6 first |
| 05:43.54 | [av]bani | but you need the newer boot55.bin and gxp2000.bin also, so it knows to get a.bin |
| 05:43.59 | IronHelix | yeah |
| 05:44.13 | IronHelix | it has the last of the non-a.bin ones which tell it to look for the a.bin ones |
| 05:44.13 | Dr-Linux | :S |
| 05:44.32 | IronHelix | flashing 1.0.2.6 will make your phone reboot 2-3-infinity times |
| 05:44.37 | krischnoff | hey, you get what you pay for... ;) |
| 05:44.52 | YoMama | IronHelix: so...i can't go straight from 1.0.1.13 to 1.0.2.8 |
| 05:45.03 | IronHelix | you can |
| 05:45.10 | IronHelix | just get the non-a.bin's from the .6 release |
| 05:45.13 | jeebusroxors | krischnoff; i know but im trying to put together something for a distro ;) |
| 05:45.18 | IronHelix | combine them with the a.bin's from the .8 release |
| 05:45.25 | IronHelix | and put those 4 files on your tftp |
| 05:45.27 | YoMama | IronHelix: gotcha |
| 05:45.29 | YoMama | thanks |
| 05:45.29 | IronHelix | that said |
| 05:45.33 | IronHelix | if you have an older phone |
| 05:45.40 | IronHelix | use the first one of the 1.0.2 series |
| 05:45.55 | IronHelix | there the phone display only goes blank instead of garbling/wrapping/flipping/flippingout |
| 05:46.05 | YoMama | IronHelix: what do u mean the first one |
| 05:47.43 | IronHelix | older phone? |
| 05:47.48 | IronHelix | certain MAC addresses |
| 05:47.55 | krischnoff | jeebus: there are the well known ones, AMP etc. Haven't used them myself though. Try to find the free O'Reilly Asterisk book, I recall that it reviews some GUI's |
| 05:48.02 | YoMama | IronHelix: i'm pretty sure i have an older one..i ordered it soon after it came out |
| 05:48.04 | IronHelix | ~book |
| 05:48.06 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:48.19 | IronHelix | yeah |
| 05:48.32 | IronHelix | i find mine lasts for about two hours tops with .8 before the display starts to screw up |
| 05:48.33 | IronHelix | also |
| 05:48.45 | IronHelix | be very very careful as there is no way to downgrade to 1.0.1.13 |
| 05:48.50 | IronHelix | or any 1.0.1 series |
| 05:48.56 | YoMama | eh...now i'm wondering if i should even bother upgrading |
| 05:48.58 | Mavvie | hmm... wonder if the congestion problem I experience now is related to the time-problem of asterisk 1.2.3 that friday afternoon. |
| 05:49.02 | YoMama | sounds like i'll mess it up more than fix anything |
| 05:49.12 | Mavvie | after all, it always starts at 15:00 or 16:00. |
| 05:49.43 | IronHelix | i'd suggest unless you want to beta test it (its very beta) or you want one of the features in the changelog stick with .13 |
| 05:49.46 | IronHelix | esp if you have older hw |
| 05:50.17 | YoMama | IronHelix: good to know..k..screw it...it works just fine how it is |
| 05:50.30 | IronHelix | my MAC is 00.0b.82.etc |
| 05:50.59 | IronHelix | http://www.voip-info.org/users/610/2610/images/420/medium.jpg <- this could happen to you! :D |
| 05:51.37 | IronHelix | or perhaps this http://www.voip-info.org/users/557/15557/images/435/upsidedown-wraped.jpg |
| 05:51.50 | justinu | heh |
| 05:51.52 | IronHelix | (they still have a few bugs in the 1.0.2 series :) |
| 05:51.53 | YoMama | mine is 00.0b.82.... |
| 05:52.37 | YoMama | IronHelix: that's not so nice |
| 05:52.42 | YoMama | screw it...it ain't worth it |
| 05:52.46 | YoMama | thanks for the warning |
| 05:52.52 | IronHelix | no problem :D |
| 05:52.59 | IronHelix | also check out the wiki gxp2000 page |
| 05:53.13 | YoMama | yeah..that's what i've been reading |
| 05:53.14 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:53.27 | IronHelix | lot of updates last few weeks as gs has been in active development |
| 05:53.28 | YoMama | i don't see any fabulous features in the 1.0.2.x series that make me want to jump up and down |
| 05:53.54 | X-Rob | Well, except for sidetone, agc tha tworks, echo can on speaker phone, blf, paging, |
| 05:54.07 | MikeJ[Laptop] | 1.0.2.x ???? |
| 05:54.07 | X-Rob | just 'little' things. |
| 05:54.09 | IronHelix | those were around in 1.0.1.13 |
| 05:54.28 | litage | would someone mind giving me an example FWD phone number? i just want to know its format |
| 05:54.29 | IronHelix | dunno about sidetone or agc |
| 05:54.31 | MikeJ[Laptop] | oh... nm |
| 05:54.34 | X-Rob | sidetone and agc are _significantly_ broken in .1.13, resulting in feedback in a noisy environment. |
| 05:54.37 | IronHelix | litage- its 5-7 digits |
| 05:54.51 | X-Rob | litage, 47876 is mine |
| 05:54.54 | YoMama | what's sidetown? |
| 05:54.54 | litage | IronHelix: does it have any other particular pattern? |
| 05:54.57 | YoMama | sidetone |
| 05:54.57 | litage | thanks X-Rob |
| 05:54.58 | IronHelix | non-useless speakerphone, paging and blf were around in .13 |
| 05:55.08 | X-Rob | YoMama, when you speak in the mouthpiece, youhear your own voice in your ear. |
| 05:55.17 | YoMama | X-Rob: oh yeah... |
| 05:55.18 | IronHelix | its a comfort thing |
| 05:55.27 | IronHelix | phone with no sidetone feels 'dead' |
| 05:55.34 | X-Rob | and users always shout |
| 05:55.44 | YoMama | the speakerphone isn't loud enough |
| 05:55.47 | X-Rob | deafening the person on the other end. |
| 05:55.50 | YoMama | people can rarely hear me well when i'm on it |
| 05:55.57 | YoMama | they say i sound like i'm on the other side of the room |
| 05:56.01 | IronHelix | are you? |
| 05:56.05 | YoMama | hahaha..no |
| 05:56.12 | [av]bani | gxp has no sidetone... |
| 05:56.13 | YoMama | about 2 feet away |
| 05:56.18 | X-Rob | [av]bani, yes it does. |
| 05:56.20 | IronHelix | im about 3' away from mine and its worked pretty well for a while |
| 05:56.24 | [av]bani | X-Rob: where? |
| 05:56.30 | X-Rob | in the handset? |
| 05:56.34 | X-Rob | where else would you put it? |
| 05:56.34 | [av]bani | mine has zilch |
| 05:56.35 | [av]bani | zero |
| 05:56.37 | [av]bani | nada |
| 05:56.46 | X-Rob | use .2 then |
| 05:56.47 | [av]bani | dead as a doornail |
| 05:56.57 | [av]bani | its never had any ... |
| 05:57.01 | X-Rob | .1 is the suxx0rz. |
| 05:57.02 | YoMama | IronHelix: it's a cheap phone...too bad there's no way i'd sell it to a customer |
| 05:57.05 | [av]bani | 1.0.1.9 through 1.0.2.8 |
| 05:57.10 | YoMama | it's a bit too chintzy |
| 05:57.14 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:57.15 | [av]bani | ive been waiting for them ot add it |
| 05:57.17 | X-Rob | [av]bani, sidetone is _definately_ there. |
| 05:57.22 | X-Rob | like, I just tested it then. |
| 05:57.23 | X-Rob | it's there. |
| 05:57.27 | YoMama | sidetone works on mine |
| 05:57.28 | [av]bani | X-Rob: well, i dont get any sidetone on any of them. |
| 05:57.29 | YoMama | just tested it |
| 05:57.43 | IronHelix | agreed- i sold them to one dude who must have said 'cheap, i want it cheap' 20 times while i was talking to him |
| 05:57.44 | X-Rob | [av]bani, you're on drugs. |
| 05:57.44 | IronHelix | PITA |
| 05:57.51 | IronHelix | i couldnt get the autoprovisioner to work |
| 05:57.59 | YoMama | i just wish i could get rid of my analog line echo...i even tried the echo preload and it didn't work |
| 05:58.04 | IronHelix | their config file generator utility kept asking for a license key |
| 05:58.12 | [av]bani | X-Rob: well, thats what grandstream told me when i discovered the http upgrade looping bug |
| 05:58.35 | Mavvie | chan_sip.c: In function â: |
| 05:58.36 | Mavvie | chan_sip.c:11066: warning: unused variable â |
| 05:58.40 | Mavvie | wonder why FC4 does do that. |
| 05:58.40 | YoMama | IronHelix: even the snoms are cheaper than most PBX handsets |
| 05:58.45 | X-Rob | [av]bani, I agree with them. The phones have sidetone. |
| 05:59.13 | X-Rob | you're on drugs |
| 05:59.21 | IronHelix | i can tell you as of 1.0.2.8 they have side tone (i just picked up my phone, hit 3 (kill dialtone) and blew into the mic |
| 05:59.22 | YoMama | mmmm...drugs |
| 05:59.48 | YoMama | mine's at 1.0.1.13 and it's got sidetown |
| 05:59.51 | YoMama | sidetone |
| 06:00.07 | X-Rob | YoMama, your sidetone can give you feedback |
| 06:00.10 | X-Rob | you'll enjoy that |
| 06:00.17 | [av]bani | hmm, yes it has sidetone but its very low... its in a noisy environ |
| 06:00.22 | YoMama | X-Rob: i've used the phone extensively...so far..it's been alright |
| 06:00.24 | [av]bani | i can hear the sidetone in my snom 360 and polycom |
| 06:00.32 | [av]bani | but the grandstream i get very faint |
| 06:00.41 | IronHelix | yeah thats another thing |
| 06:00.47 | IronHelix | either of you had the handset die on a gxp? |
| 06:00.53 | X-Rob | Yes, I agree, the grandstreams sidetone _is_ quieter than the snoms |
| 06:00.59 | YoMama | IronHelix: not mine..still working |
| 06:01.02 | X-Rob | but the snoms have better acoustic baffling in the handset |
| 06:01.03 | [av]bani | its quieter than everything, i can barely hear it |
| 06:01.10 | X-Rob | so they can crank the sidetone up more. |
| 06:01.13 | YoMama | IronHelix: why? do i have that to look forward to? |
| 06:01.17 | IronHelix | lol |
| 06:01.18 | IronHelix | dunno |
| 06:01.25 | IronHelix | one died the other day |
| 06:01.30 | IronHelix | gotta get them a new one |
| 06:01.36 | YoMama | IronHelix: if this piece of shit breaks..i won't be buying another |
| 06:01.44 | IronHelix | bani- if i email GS asking for a replacement will i get a response? |
| 06:01.54 | [av]bani | IronHelix: ? |
| 06:02.01 | IronHelix | gxp handset died |
| 06:02.13 | [av]bani | yay? |
| 06:02.14 | IronHelix | you've dealt directly with them right? |
| 06:02.19 | [av]bani | no |
| 06:02.20 | IronHelix | oh |
| 06:02.25 | IronHelix | nm then |
| 06:02.26 | IronHelix | heh |
| 06:02.35 | IronHelix | i muust be thinking of thetatag |
| 06:02.41 | [av]bani | i went round in circles with them over the http upgrade looping bug |
| 06:02.43 | IronHelix | that did the alpha testing |
| 06:02.56 | [av]bani | never dealt with repairs |
| 06:03.15 | YoMama | hmm..atacomm has the polycom IP 301 for $115 |
| 06:03.28 | YoMama | polycom speakerphones rock |
| 06:03.32 | IronHelix | not bad price... |
| 06:03.33 | IronHelix | hmmm |
| 06:03.37 | [av]bani | YoMama: 301 is monitor only |
| 06:03.43 | IronHelix | i just wish polycom would fix their firmware policy (its broken) |
| 06:03.44 | YoMama | oh |
| 06:04.01 | [av]bani | IronHelix: just become a polycom reseller. problem solved |
| 06:04.23 | *** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 06:04.37 | IronHelix | heh |
| 06:04.40 | IronHelix | thats one way to fix |
| 06:04.43 | IronHelix | wonder how hard that is... |
| 06:04.46 | YoMama | what's their firmware policy? |
| 06:04.51 | [av]bani | YoMama: "foad" |
| 06:04.59 | YoMama | foad? |
| 06:04.59 | [av]bani | pretty much sums it |
| 06:05.06 | IronHelix | F*** off and die |
| 06:05.14 | xbmodder_lappy | I keep getting this error: |
| 06:05.14 | xbmodder_lappy | exten => YOURNUMBER,1,Answer() |
| 06:05.15 | xbmodder_lappy | exten => YOURNUMBER,1,DIAL(SIP/user,20) |
| 06:05.16 | xbmodder_lappy | ack |
| 06:05.18 | xbmodder_lappy | wrong thing |
| 06:05.22 | YoMama | lovely... |
| 06:05.23 | xbmodder_lappy | Feb 16 22:04:49 NOTICE[7684]: chan_sip.c:10294 handle_request_invite: Failed to authenticate user "DHILLON AJAY" <sip:9252097318@207.174.111.12>;tag=as1cfbcb8d |
| 06:05.25 | xbmodder_lappy | that one |
| 06:05.27 | IronHelix | aka they wont give firmware to anybody except polycom registered partners and resellers, ever |
| 06:05.28 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 06:05.45 | YoMama | IronHelix: that's lame |
| 06:05.56 | [av]bani | YoMama: one of many complaints about polycom |
| 06:06.03 | [av]bani | probably in the top 3 |
| 06:06.12 | YoMama | IronHelix: so u tell the people u bought it from that they have to make the firmware updates available to you |
| 06:06.44 | YoMama | oh my god..this is scary..grandstream is making video phones now??? |
| 06:06.44 | [av]bani | YoMama: good luck if you buy a used polycom from ebay. |
| 06:07.06 | IronHelix | yeah its lame... i try to avoid them because of that (philosophical reasons) |
| 06:07.12 | sprnova | got MOH working.. guess mpg123 did not like my complicated 192kb VBR files. |
| 06:07.13 | xbmodder_lappy | YoMama, yeah |
| 06:07.16 | xbmodder_lappy | and there good! |
| 06:07.29 | IronHelix | yeah whats with that- the gxv3000 |
| 06:07.38 | YoMama | xbmodder_lappy: nothing i've bought from GS would be defined as "good"...cheap maybe..but not good |
| 06:07.39 | [av]bani | looks like its a rebadged clone |
| 06:07.51 | [av]bani | i found a phone exactly the same with slightly different styling |
| 06:07.57 | [av]bani | looks like grandstream just rebadged something |
| 06:08.37 | YoMama | so the snom 320 is a nice phone? |
| 06:08.45 | YoMama | i've only heard good things..haven't used one in person |
| 06:08.53 | [av]bani | if you can stand the firmware, sure |
| 06:09.00 | YoMama | what's wrong with the firmware? |
| 06:09.09 | [av]bani | its buggy, and the ui is terrible |
| 06:09.19 | [av]bani | snom has no idea how to design a phone ui |
| 06:09.19 | YoMama | for shit's sake..doesn't anyone have a stable, nice, inexpensive SIP phone? |
| 06:09.31 | [av]bani | inexpensive, stable, nice choose 2 |
| 06:09.36 | YoMama | avban: neither does grandstream |
| 06:09.43 | [av]bani | YoMama: grandstream's ui is better than snom's |
| 06:09.47 | YoMama | that's scary |
| 06:09.50 | [av]bani | yep |
| 06:10.01 | YoMama | i'm spoiled...before this..i was using a full blown cisco call manager solution |
| 06:10.05 | [av]bani | grandstream speaks better english than snom too |
| 06:10.27 | YoMama | frightening |
| 06:10.43 | [av]bani | every phone has its warts, even cisco |
| 06:11.17 | [av]bani | at least with the gxp i dont feel like i got ripped off |
| 06:11.48 | YoMama | true...but what phones do u use in a business environment then? |
| 06:11.59 | YoMama | they all seem...flaky...and crappy |
| 06:12.22 | IronHelix | im setting up a few aastra's later this week |
| 06:12.23 | xbmodder_lappy | I don't see how my family can talk on the phone for 60 frickin' hours! |
| 06:12.23 | [av]bani | depends on what warts you can tolerate |
| 06:12.25 | IronHelix | we'll see how that goes |
| 06:12.37 | [av]bani | IronHelix: i hear aastra, at least the 9133i, has echo issues |
| 06:12.39 | IronHelix | new firmware just released looks to clear up most of the warts |
| 06:12.44 | IronHelix | including that one |
| 06:12.47 | [av]bani | yay? |
| 06:13.03 | [av]bani | aastra seems the most progressive vendor, they test everything with * |
| 06:13.04 | YoMama | aastra? |
| 06:13.07 | kuku5 | YoMama: cisco call manager? so you used cisco 79xx phones? |
| 06:13.14 | YoMama | kuku5: yes |
| 06:13.21 | [av]bani | YoMama: www.aastra.com |
| 06:13.27 | *** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
| 06:13.28 | YoMama | kuku5: i know more about Cisco's CM than *.. |
| 06:13.28 | kuku5 | YoMama: how many do you have under you ? |
| 06:13.31 | IronHelix | yeah aastra telecom, used to be partnered wtih nortel or they did something with nortel as i recall |
| 06:13.40 | YoMama | kuku5: under me? |
| 06:13.43 | [av]bani | aastra used to supply phones for nortel i think |
| 06:13.43 | kuku5 | that you mange |
| 06:13.49 | [av]bani | theyve been making analogue phones for decades |
| 06:13.59 | YoMama | kuku5: depends on how many projects i got going at once..i run a professional services division |
| 06:14.03 | [av]bani | you can find jillions of aastra analogue phones on ebay |
| 06:14.07 | YoMama | kuku5: anywhere from 15-60 |
| 06:14.10 | IronHelix | now they partner with sayson, lots of adsi stuff |
| 06:14.14 | IronHelix | yeah |
| 06:14.15 | [av]bani | adsi :< |
| 06:14.19 | [av]bani | eeeeeevil |
| 06:14.30 | kuku5 | YoMama: I have 3 different clients using * and 7940/7960 's, each with like 20-30 phones |
| 06:14.34 | YoMama | IronHelix: where do u get them? |
| 06:14.40 | IronHelix | i like voipsupply |
| 06:14.46 | YoMama | kuku5: using SCCP or SIP? |
| 06:14.53 | IronHelix | prices are decent and they have yet to royally screw anything up |
| 06:14.57 | IronHelix | voipsupply.com |
| 06:15.01 | kuku5 | YoMama: for some reason, within the last 4 weeks, at each client, the secretary'sphone (7960) just reboots, or hangs, or stops working, randomly |
| 06:15.05 | kuku5 | sip |
| 06:15.08 | [av]bani | voipsupply are ok, they arent the cheapest |
| 06:15.12 | [av]bani | tho their sales people are .. odd |
| 06:15.19 | IronHelix | how so? never talked to them |
| 06:15.36 | kuku5 | i have a voipsupply guy on my aim - so its quick to get orders |
| 06:15.56 | kuku5 | YoMama: did you see that happening with any of your phones? |
| 06:16.01 | YoMama | kuku5: so are u using them with SIP or SCCP? |
| 06:16.03 | YoMama | kuku5: never |
| 06:16.06 | kuku5 | sip |
| 06:16.09 | YoMama | kuku5: they were as solid as a rock |
| 06:16.17 | YoMama | kuku5: try SCCP...i hear it's more stable |
| 06:16.21 | kuku5 | its messed up - |
| 06:16.22 | [av]bani | maybe we just had bad luck, but they had trouble remembering that we declined their replacement and config contracts |
| 06:16.29 | kuku5 | i tried 7.3, 7.4, 7.5 |
| 06:16.31 | Qwell | I don't know about more stable... |
| 06:16.31 | YoMama | kuku5: and i hear the sound quality is better |
| 06:16.32 | kuku5 | but it wasnt like this |
| 06:16.35 | Qwell | sccp is fun though |
| 06:16.53 | [av]bani | the ciscos were designed for sccp. so of course they will work better with sccp |
| 06:17.03 | YoMama | is there a cheaper company than voipsupply..yeah..i agree..sounds like eveyrone's really happy with them, but their pricing isn't the best |
| 06:17.19 | [av]bani | well, other pricing is usually only like $5 less |
| 06:17.23 | YoMama | kuku5: wait until your clients start asking u to write XML applications |
| 06:17.34 | Qwell | XML apps are really fun |
| 06:17.42 | [av]bani | atacomm for example.. a little less, but their shipping prices are outrageous |
| 06:17.54 | *** join/#asterisk _Thor (i=CS@dialup-4.250.138.230.Dial1.Weehawken1.Level3.net) |
| 06:17.56 | [av]bani | like $40 to ship a single spa3000 |
| 06:18.08 | Qwell | atacomm is run by an assclown |
| 06:18.16 | _Thor | hello everyone |
| 06:18.26 | YoMama | IronHelix: have u tried the speakerphone on the Aastra? |
| 06:18.39 | IronHelix | not yet although i've heard its pretty good |
| 06:19.07 | _Thor | Qwell: is it only one guy answering the phones at atacomm? |
| 06:19.07 | [av]bani | aastra also seem pretty responsive on firmware issues, making releases regularly |
| 06:19.08 | YoMama | hmm...$140 for the 9112i...too bad it's only one line |
| 06:19.24 | IronHelix | they're all sitting in boxes excpet for the manager of the place who couldnt resist the temptation to tear his 480i-ct box apart and play with the little cordless thing |
| 06:19.30 | [av]bani | ~phones |
| 06:19.32 | jbot | i guess phones is at http://bani.anime.net/phones/ |
| 06:19.38 | Qwell | _Thor: dunno |
| 06:19.41 | IronHelix | 160 for the 9133... got a bunch of those |
| 06:19.46 | [av]bani | polycom 501 is not bad, if you do not need xml |
| 06:19.54 | _Thor | it looks like it |
| 06:20.02 | IronHelix | holy crap bani that rocks |
| 06:20.14 | [av]bani | shame polycom has yet to discover this mysterious device known as "backlight" |
| 06:20.37 | [av]bani | maybe theyre vampires or something, need to avoid exposure |
| 06:21.05 | IronHelix | hahahaha |
| 06:21.36 | IronHelix | same with the linksys/sipura/cisco SPA-841/9xx... |
| 06:21.40 | YoMama | nice comparison chart |
| 06:21.44 | justinu | do you really talk on the phone in the dark ? |
| 06:21.59 | YoMama | justinu: phone sex |
| 06:22.00 | Qwell | ^ and if so, do you really need a light? |
| 06:22.06 | YoMama | hahahaha |
| 06:22.16 | [av]bani | IronHelix: sipura support has basically dried up since they were bought out. |
| 06:22.21 | [av]bani | IronHelix: and the 942... what a joke. |
| 06:22.31 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 06:22.38 | jeebusroxors | where can i get a cool linux shirt? haha |
| 06:22.39 | IronHelix | its too bad :( |
| 06:22.46 | Qwell | jeebusroxors: cafepress |
| 06:22.47 | IronHelix | same thing with linksys and the wrt routers |
| 06:22.50 | IronHelix | all vxworks now |
| 06:22.55 | justinu | i did like my old western electric bar phone with the backlit keypad tho |
| 06:22.58 | justinu | that was a nice phone |
| 06:23.12 | YoMama | IronHelix: so u think these Aastra phones are pretty rock solid? what about the interface? |
| 06:23.25 | [av]bani | YoMama: http://www.o2m8.com/modules.php?name=News&file=article&sid=25 |
| 06:23.34 | IronHelix | havent even powered the thing yet... they are mostly in boxes except one i had to bolt to the wall |
| 06:23.37 | justinu | aastra phone seems solid |
| 06:23.46 | IronHelix | the one i did open (a 9133) felt pretty solid build-wise |
| 06:23.56 | justinu | i have the 480i |
| 06:24.01 | [av]bani | the only complaint ive seen so far was echo... nobody complained about usability or construction |
| 06:24.03 | IronHelix | more solid than the gs gxp2000 |
| 06:24.08 | IronHelix | brb |
| 06:24.20 | [av]bani | cardboard is more solid than the gxp2000 |
| 06:25.04 | justinu | it's not that bad |
| 06:25.05 | YoMama | avban1: neat |
| 06:25.06 | [av]bani | wow justinu has a 480i? |
| 06:25.10 | justinu | yep |
| 06:25.12 | [av]bani | you are full of suprises |
| 06:25.16 | [av]bani | gxp, 480i, and snom |
| 06:25.21 | justinu | ip601 |
| 06:25.26 | justinu | ip501 |
| 06:25.38 | Qwell | what, no cisco? |
| 06:25.40 | justinu | nope |
| 06:25.44 | justinu | i need one for my zoo |
| 06:25.50 | justinu | that 7970 |
| 06:25.51 | justinu | :P |
| 06:25.54 | Qwell | zoo? |
| 06:26.00 | justinu | phone petting zoo |
| 06:26.05 | Qwell | I see |
| 06:26.08 | justinu | for the clients |
| 06:26.19 | [av]bani | justinu: you agree, snom's ui is terrible? |
| 06:26.26 | justinu | yes |
| 06:26.42 | justinu | maybe slightly better than the gxp w/ 1.x firmware |
| 06:26.58 | [av]bani | i wonder how much i can redesign it with xml |
| 06:27.17 | YoMama | this phone's pricey though |
| 06:27.28 | YoMama | but i guess u get what u pay for... |
| 06:27.35 | YoMama | is the 480i pretty rock solid? |
| 06:27.40 | justinu | yes |
| 06:27.50 | [av]bani | its been described as something you could club someone with |
| 06:27.53 | justinu | caveat emptor: it's PoE only |
| 06:28.07 | [av]bani | yea, thats a real suprise to people... no wallwart at all |
| 06:28.12 | justinu | yep |
| 06:28.14 | YoMama | which can be fixed with a $20 dongle |
| 06:28.19 | justinu | just a friendly tip |
| 06:28.20 | [av]bani | YoMama: its bizarre though |
| 06:28.39 | YoMama | avban: how come? if you're doing it in a business setting..not using PoE is lame |
| 06:28.59 | [av]bani | YoMama: lots of people arent setup for poe.. imagine them buying a pile of 480i's and going 'wtf' |
| 06:29.10 | [av]bani | afaik its the only voip phone which is _only_ poe |
| 06:29.10 | YoMama | avban: oops..hehehe |
| 06:29.30 | [av]bani | i was 'wtf' when i read the specs... i was looking for power supply for a while and didnt see one |
| 06:29.37 | [av]bani | took me a while to realize its poe only... |
| 06:29.41 | YoMama | avban: Cisco is lame for PoE...i only recently found out that their polarity is opposite the standard |
| 06:29.48 | IronHelix | the 480i ct comes with a wall wart... |
| 06:29.53 | [av]bani | YoMama: more recent ciscos are 802.3af |
| 06:30.02 | YoMama | avban: yeah? they fixed it? |
| 06:30.06 | [av]bani | YoMama: and many vendors support 802.3af _and_ cisco poe |
| 06:30.12 | IronHelix | hmmm |
| 06:30.18 | IronHelix | phone petting zoo |
| 06:30.20 | [av]bani | YoMama: ~phones |
| 06:30.21 | IronHelix | thats not a bad idea... |
| 06:30.23 | IronHelix | *schemes* |
| 06:30.36 | YoMama | avban: i'm just saying that leave it up to Cisco to do something opposite everyone else |
| 06:30.37 | [av]bani | there is a poe column |
| 06:30.40 | _Thor | help, how do you place an unregister sip call with a prefix? |
| 06:30.51 | [av]bani | YoMama: to be fair, they implemented poe before the standard was ratified. |
| 06:31.02 | IronHelix | put the pattern exten in the default (guest) context |
| 06:31.07 | YoMama | avban: hehe..ok ok..just ragging on them a little |
| 06:31.07 | [av]bani | YoMama: to their discredit, they kept using their mangled version after it was ratified |
| 06:31.32 | YoMama | avban: it's hubris..they think they run the world |
| 06:31.40 | [av]bani | they do, mostly |
| 06:31.41 | freat | a couple months ago I picked up a cisco catalyst switch... had to make this special cable to connect it to a serial port... |
| 06:32.00 | YoMama | freat: dude..it's been like that for ages |
| 06:32.01 | [av]bani | freat: yes, thats cisco serial in general. welcome to 1994 |
| 06:32.07 | _Thor | I am doing: dial(sip/xxxx${EXTEN}@domain.com) |
| 06:32.18 | freat | yeah I know... I'm just saying, it's nothing new for cisco to do that crap |
| 06:32.23 | freat | everything different |
| 06:32.23 | _Thor | whereas xxxx is the prefix |
| 06:32.24 | YoMama | freat: light blue cable...i got 400 of them laying around somewhere..gimme $15 and your address and i'll send you 10 of 'em |
| 06:32.43 | freat | that was the point though... I wasn't about to spend more on the cable than on the switch hehe |
| 06:32.43 | [av]bani | freat: i dont mind that as much as i mind their list price for cisco db9 being $350 |
| 06:32.56 | kuku5 | anyone tried dell's poe switch ? |
| 06:32.57 | [av]bani | freat: or $700 for a 5 foot v35 cable |
| 06:33.10 | YoMama | haha..yeah..the old cables were expensive as hell |
| 06:33.28 | YoMama | freat: u know what i do? so i don't haveta carry around a special cable? |
| 06:33.36 | *** join/#asterisk |||sLaSh||| (i=th3_gam3@203.215.100.96) |
| 06:33.45 | freat | YoMama: no I don't know |
| 06:33.48 | |||sLaSh||| | hello where can i get cisco sip version 6 P0S3-06-0-00 |
| 06:33.51 | YoMama | freat: get one of those pin kits for a DB9..and pin your own thing so u can use a regular straight-through |
| 06:33.55 | Qwell | |||sLaSh|||: cisco |
| 06:34.06 | YoMama | freat: that way you only carry around a little dongle rather than two cables..one ethernet and one cisco |
| 06:34.16 | YoMama | freat: cost ya $3 |
| 06:34.20 | Himeko | i just build my own cable |
| 06:34.22 | YoMama | freat: and about 10 minutes of your time |
| 06:34.30 | [av]bani | YoMama: better yet, rj45 and then rj45->db9 connectors |
| 06:34.39 | YoMama | avban: that's what i just said |
| 06:34.50 | freat | yeah I used a multimeter and some documentation online to make a cable |
| 06:35.06 | YoMama | avban: but i didn't explain it so well |
| 06:35.08 | freat | ripped apart a serial cable |
| 06:35.12 | [av]bani | :) |
| 06:35.44 | [av]bani | YoMama: kentrox has wacky serial too. so i have 3 sets of connectors around. rs232, cisco, and kentrox |
| 06:35.45 | YoMama | avban: i used to carry around one for each of the companies..one for bay..one for cisco..one to make it null modem...etc etc etc |
| 06:36.18 | [av]bani | oh yeah, APC has wacky serial too! |
| 06:36.22 | YoMama | avban: and that one company that used to dominate the ISDN router space...what were they again? |
| 06:36.30 | YoMama | uhh..starts with a P |
| 06:37.09 | YoMama | avban: yeah..but they're gracious enough to give u a serial cable for their equipment |
| 06:37.29 | [av]bani | fortuantely they moved to USB, something they cant fuckup :) |
| 06:38.44 | YoMama | these aastra 480i phoens are pretty badass |
| 06:38.59 | YoMama | anyone have the CT with the cordless? is the cordless any good? |
| 06:39.05 | [av]bani | cept for the display being character based :/ |
| 06:39.07 | IronHelix | im installing one this week |
| 06:39.17 | linlin | whats an approiate dialing rule for a number in the UK that looks like this: +44 (0) 870 011 2988 |
| 06:39.18 | IronHelix | i'll let you guys know how it goes |
| 06:39.24 | YoMama | IronHelix: the CT? with the cordless? |
| 06:39.28 | IronHelix | yeah |
| 06:39.43 | YoMama | linlin: u don't need the 0 unless you're in the UK |
| 06:39.50 | linlin | ok |
| 06:40.00 | YoMama | linlin: it's just (your international dialing prefix) 44 870 011 2988 |
| 06:40.01 | linlin | im in usa but im using a european sip provider |
| 06:40.08 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 06:40.16 | YoMama | linlin: doesn't matter unless you're in the UK |
| 06:40.24 | YoMama | which is why it's in paratheses |
| 06:40.34 | linlin | so... 1NXXXXXXXXXXXX |
| 06:40.44 | YoMama | uhh |
| 06:40.48 | YoMama | 1? |
| 06:40.53 | YoMama | if you're in teh us...don't you use 011? |
| 06:40.56 | [av]bani | IronHelix: 480i or 9133i? |
| 06:41.11 | Qwell | It's a UK provider...not an international call |
| 06:41.18 | IronHelix | 9133's except the boss there who demanded a 480i-ct when i explained that you can check the weather on it |
| 06:41.19 | linlin | you tel me, sorry really new at international dialing |
| 06:41.27 | IronHelix | aka he wanted the baddest most awesome phone in the office |
| 06:41.28 | Qwell | linlin: Where is your provider located? |
| 06:41.29 | *** join/#asterisk pengyong (n=lala@222.188.130.101) |
| 06:41.33 | Qwell | and how do they connect to the pstn? |
| 06:41.39 | IronHelix | they dont yet |
| 06:41.40 | [av]bani | haha |
| 06:41.41 | IronHelix | new business |
| 06:41.42 | linlin | europe somewhere, no idea |
| 06:41.44 | IronHelix | gonna be voip |
| 06:41.46 | linlin | its SIPDiscount |
| 06:41.57 | Qwell | support@sipdiscount.com |
| 06:41.57 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
| 06:42.11 | YoMama | IronHelix: great..now you're stuck writing an XML app :-P |
| 06:42.15 | [av]bani | 480i ct is just 480i with support for wireless handset right? |
| 06:42.24 | IronHelix | yeah |
| 06:42.33 | IronHelix | yomama- they already have the weather one |
| 06:42.33 | Qwell | 480i==aastra? |
| 06:42.36 | IronHelix | yea |
| 06:42.38 | Qwell | Why not get a 7970? :p |
| 06:42.48 | IronHelix | cuz aastra <3's asterisk |
| 06:42.52 | YoMama | avban: the 480i supports a wireless headset? |
| 06:42.57 | [av]bani | heh yeah, cant say that about cisco |
| 06:43.11 | IronHelix | 480i-ct comes with a cordless handset that looks like the linksys wip300 |
| 06:43.13 | IronHelix | brb |
| 06:43.19 | Qwell | cordless handset? |
| 06:43.23 | Qwell | or headset? |
| 06:43.26 | IronHelix | hand |
| 06:43.30 | Qwell | why? |
| 06:43.41 | [av]bani | http://www.aastra.com/enterpriseip/pro_243.asp |
| 06:43.46 | [av]bani | because |
| 06:43.51 | YoMama | avban: how? u can get a wireless headset with cisco's phone... |
| 06:44.08 | [av]bani | YoMama: handset not headset |
| 06:44.16 | welles | hi all. i need the confirm . in asterisk 1.2.4 the cmd 'meetme list 1000' only return the user numbers of conference room 1000? |
| 06:44.30 | YoMama | avban: oh |
| 06:44.41 | YoMama | you know what'd be sweet...if someone made these phones bluetooth compatible |
| 06:44.43 | [av]bani | Qwell: i've just been informed chan_sccp and 7970 doesnt work with asterisk 1.2, only 1.0 ... |
| 06:44.56 | YoMama | that way..you could use your existing headset with your business phone |
| 06:44.56 | [av]bani | Qwell: does one need a different chan_sccp for 1.2 ? |
| 06:45.16 | brookshire | Qwell: hi! |
| 06:45.27 | Qwell | brookshire: omg hi! |
| 06:45.33 | justinu | like omg |
| 06:45.35 | Qwell | [av]bani: no, chan_sccp works on anything |
| 06:45.36 | justinu | barf out |
| 06:45.42 | [av]bani | YoMama: there's bluetooth support for * |
| 06:45.43 | Qwell | 1.0, 1.2, svn, openpbx (allegedly) |
| 06:46.10 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 06:46.28 | [av]bani | Qwell: whats this chan_sccp2 business then? |
| 06:46.35 | Qwell | chan-sccp.berlios.de |
| 06:46.37 | Qwell | get that one |
| 06:46.47 | Qwell | this one is actually maintained/used |
| 06:46.48 | YoMama | avban: i meant the phones |
| 06:47.11 | welles | any one will answer my question? i do need the info of meetme list <conferno>. |
| 06:47.39 | YoMama | welles: huh? |
| 06:48.32 | [av]bani | ah the berlios one is chan_sccp2 |
| 06:48.43 | [av]bani | or at least referred to as such |
| 06:48.51 | welles | YoMama, cmd 'meetme list <conferno> ' in asterisk 1.2.4 only return the numbers of user in conference .i am right/ |
| 06:48.52 | Qwell | It's THE chan_sccp, IMO |
| 06:48.53 | welles | ？ |
| 06:49.00 | [av]bani | theres three sccp drivers :/ |
| 06:49.01 | YoMama | IronHelix: so the 480i CT comes wiht a power brick? |
| 06:49.05 | [av]bani | what a mess |
| 06:49.07 | Qwell | [av]bani: the others suck |
| 06:49.09 | IronHelix | yeah |
| 06:49.38 | YoMama | IronHelix: when i get my bonus..i'm gonna haveta buy one of these suckers :) |
| 06:49.52 | YoMama | on a totally off-topic..anyone here got a smartphone? |
| 06:49.59 | Qwell | my 7970 is smart |
| 06:50.04 | YoMama | welles: lemme check |
| 06:50.10 | [av]bani | YoMama: if you want really top of the line phone, you can get a cisco 7970 for $385 |
| 06:50.17 | welles | YoMama, ok |
| 06:50.35 | YoMama | avban: nah...i don't wanna spend over $200 |
| 06:50.40 | YoMama | avban: this is for home shit... |
| 06:50.50 | YoMama | although the 480i looks like it'd make a good office phone |
| 06:50.53 | *** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net) |
| 06:51.31 | [av]bani | depends on your requirements |
| 06:51.53 | welles | YoMama, i remember this cmd will list the detail of users in the conference, at least in asterisk 1.0.9 it is true. |
| 06:52.15 | YoMama | welles: mine's got the detail |
| 06:52.30 | YoMama | it shows extension, the user's name, the channel, etc. |
| 06:52.33 | YoMama | then has a total count |
| 06:52.42 | welles | YoMama, then what's wrong my asterisk? |
| 06:53.03 | YoMama | welles: umm..i dunno...maybe your verbosity level is set too low? that's a total guess |
| 06:53.25 | sprnova | anyone know who MOH decides which song to play in the /usr/share/asterisk/mohmp3?? I have 3 songs.. it always plays the second one (alpabetically) |
| 06:53.33 | welles | YoMama, my asterisk only return one line. i set verbose to 10 .no change |
| 06:54.08 | YoMama | sprnova: do a ps and look at the command line of mpg123 |
| 06:54.18 | YoMama | sprnova: it might not be properly loading the first mp3 |
| 06:54.59 | [av]bani | http://movies.apple.com/movies/universal/miami_vice/miami_vice-tsr1_h640w.mov <- ONOES |
| 06:55.10 | welles | YoMama, is there any other reason ? |
| 06:55.48 | YoMama | welles: sorry..i'm not familiar enough with the code to tell u if there's a reason why there's no detail in your meetme list |
| 06:56.24 | welles | YoMama, thanks ,any way. very strange |
| 06:56.40 | sprnova | YoMama.. I have two mpg123 processes running wierd. |
| 06:56.57 | YoMama | sprnova: i've noticed that mpg123 doesn't die sometimes |
| 06:57.04 | YoMama | when u kill asterisk |
| 06:57.38 | sprnova | sounds like a bad first file.. I hear "thud" then it starts paying the second one.. LOL |
| 06:57.54 | YoMama | sprnova: there ya go |
| 06:58.41 | sprnova | only diff I can see is when I do a "file" command the one that does not play reports IDTag 2.3.0 the other two are 2.2.0 |
| 06:58.54 | YoMama | so who sells flash-based rackmountable PCs that u can run * on? |
| 07:00.01 | ManxPower | YoMama, That would depend on your needs. |
| 07:00.29 | YoMama | ManxPower: put the OS and asterisk on flash...voicemail and everything bulky on a HD |
| 07:00.59 | ManxPower | YoMama, interfaces to the PSTN and phones? |
| 07:01.17 | YoMama | ManxPower: it should have two PCI slots |
| 07:01.37 | ManxPower | Best of luck. |
| 07:01.43 | YoMama | one for some FXO/FXS ports..and one for dual-span T1 card |
| 07:01.54 | YoMama | what...are free PCI slots a problem? |
| 07:02.02 | ManxPower | Things like Sokeris (sp!) tend to be pretty low power. |
| 07:02.13 | ManxPower | but ulaw only, no transcoding might work. |
| 07:02.53 | masked | has anyone here used telstra call waiting via a fxo? |
| 07:03.11 | masked | u bout X-Rob ? |
| 07:03.19 | YoMama | ManxPower: 1U machines usually only have 1 PCI slot :( |
| 07:03.56 | ManxPower | YoMama, You could prolly do a standard 1U if you can find one with 2 slots and use a CF IDE thingy |
| 07:04.50 | YoMama | ManxPower: yeah..just thinking about what it'd take to build a pretty good * box for business |
| 07:05.11 | YoMama | PRI plus some analog for backup/emergency |
| 07:05.24 | linlin | how can i manually force the hang up of a phone through asterisk -vvvr ? |
| 07:05.31 | YoMama | means you haveta get a T1 card and a FXO card |
| 07:05.49 | YoMama | that's two slots right there |
| 07:07.34 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:08.37 | *** join/#asterisk Fedoracore6 (n=dsd@219.95.15.117) |
| 07:12.17 | Fedoracore6 | hello i try using ivr |
| 07:12.27 | Fedoracore6 | but my amp didint work |
| 07:14.08 | masked | did you plug an instrument in? |
| 07:15.17 | Fedoracore6 | what mean instrument |
| 07:15.26 | Fedoracore6 | i press *77 |
| 07:15.32 | Fedoracore6 | to record my voice |
| 07:15.36 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:16.22 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 07:16.24 | *** join/#asterisk smurfix (n=smurf@debian/developer/smurf) |
| 07:16.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:17.15 | Fedoracore6 | masked its ... i must install to the addons version 1.2.1 |
| 07:17.31 | Fedoracore6 | and sound version 1.2.1 |
| 07:17.46 | Fedoracore6 | bacause i try to built touch tones registraion system |
| 07:19.30 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:21.17 | YoMama | u still there Iron? |
| 07:21.22 | IronHelix | yeah just on phone |
| 07:21.23 | IronHelix | stand by |
| 07:21.30 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:22.38 | welles | meetme list cmd can not work .anyone know what's wrong? |
| 07:24.21 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:25.00 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 07:27.07 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:27.46 | welles | cmd 'meetme list <conferno>' can not return the detail of the conference' |
| 07:29.10 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 07:29.30 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:30.43 | IronHelix | yomama- jabra makes a headset plug to bluetooth adapter, * also supoprts bluetooth. welles- meetme list 1000 will list users in conf 1000 (i think). yomama- yeah the 480i from what i've seen rocks, if this deployment goes well i'll probably pick one up, also FWIW the 9133 looks nice when wall mounted. |
| 07:31.01 | IronHelix | linlin- do show channels, then soft hangup (channelname), ie soft hangup SIP/1234-abcd |
| 07:31.17 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:31.23 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 07:31.35 | YoMama | IronHelix: the 9133i apparently also has XML capabilities..that whole series uses the same firmware as the 480i |
| 07:31.54 | IronHelix | i dunno that it can do much with xml on a 2 line display |
| 07:31.58 | IronHelix | the display is kinda small |
| 07:32.06 | welles | anyone has some ideas? |
| 07:32.14 | YoMama | IronHelix: yeah...but u can proabably do basic directory lookups and stuff |
| 07:32.34 | YoMama | IronHelix: i wonder what u can access thru the XML interface |
| 07:33.01 | IronHelix | if it does have xml then theoretically at least the skys the limit |
| 07:33.03 | IronHelix | ldap even maybe |
| 07:33.24 | YoMama | IronHelix: as long as it supports push XML and query fields..then it'll do anything |
| 07:33.33 | YoMama | well, and allow protected access to some of its features |
| 07:35.05 | YoMama | reading the API guide now |
| 07:35.17 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 07:35.35 | IronHelix | i skimmed that doc once |
| 07:36.42 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 07:37.06 | IronHelix | you can (as i recall) def. set password fields |
| 07:37.23 | YoMama | IronHelix: seems pretty complete |
| 07:37.56 | IronHelix | yeah extremely helpful |
| 07:38.06 | Qwell | kernel-module-ntfs-2.6.15-1.1831_FC4.stk16-2.1.25-0.rr.10.4.i686.rpm |
| 07:38.11 | Qwell | I freaking LOVE redhat |
| 07:40.48 | YoMama | IronHelix: cool...u can push phone #'s for it to dial |
| 07:40.55 | YoMama | the only probolem is...the only auth method is IP |
| 07:41.05 | johnsu01 | ok, so I've got local extensions working, and I can make an iax test call, but I can't make local calls (using Junction)... |
| 07:43.52 | IronHelix | hey has polycom changed their firmware policy at all? i just went to their site and i get a form that looks like its going to let me register to download FW... |
| 07:44.14 | YoMama | hehe |
| 07:44.17 | YoMama | that'd be interesting |
| 07:44.20 | MstlyHrmls | you can get the older firmware now |
| 07:44.27 | MstlyHrmls | but not the up-to-date stuff |
| 07:44.47 | IronHelix | i still cannot understand what they hoep to gain with that |
| 07:45.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:45.10 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 07:45.30 | YoMama | IronHelix: NOTE: At this time, end-user customers can only download previous software. Please work directly with the Polycom Certified VoIP Reseller you purchased the products from to obtain the most current and appropriate software. |
| 07:45.33 | YoMama | from their website |
| 07:45.38 | YoMama | idiots |
| 07:45.42 | IronHelix | wtf |
| 07:45.45 | IronHelix | i want to smack them |
| 07:45.46 | IronHelix | so i do |
| 07:45.56 | IronHelix | by not buying their products *smirk* |
| 07:46.31 | YoMama | IronHelix: they're trying to treat their resellers like value-added partners |
| 07:46.45 | YoMama | but the problem is that if they're going to do that..then they shouldn't let companies like voipsupply sell them |
| 07:46.54 | YoMama | it should only be to integrators |
| 07:46.56 | YoMama | so they're idiots |
| 07:47.02 | IronHelix | yeah |
| 07:47.07 | IronHelix | i find many companies have this problem |
| 07:47.15 | YoMama | yeah..stupid |
| 07:47.23 | IronHelix | they do not know / understand (or at least act like they dont know/understand) who their customer actually is |
| 07:47.23 | YoMama | Cisco doesn't even have this policy |
| 07:47.41 | YoMama | u can get smartnet on their phones for $20/year or something like that |
| 07:47.46 | IronHelix | Cisco charges for firmware- but they know you are their customer and they are happy to bill you like $8 to sell it to you |
| 07:47.52 | YoMama | yeah |
| 07:47.57 | YoMama | smartnet is cheap for the phones |
| 07:48.13 | IronHelix | whereas polycom needs to decide if their customers are the VARs or the customers |
| 07:48.16 | YoMama | these aastra phones look pretty fancy schmancy |
| 07:48.20 | IronHelix | i say its the customers, but thats me cuz im a customer :) |
| 07:48.21 | YoMama | IronHelix: exactly |
| 07:48.35 | YoMama | IronHelix: what's a good phone for the receptionist? |
| 07:48.42 | YoMama | for let's say..an office with 30 people |
| 07:48.54 | YoMama | any phones with like 30 BLFs? |
| 07:49.02 | IronHelix | try a snom 360, it supports expansion sidecars with i think 48blf's each |
| 07:49.19 | IronHelix | you can link up to 2 sidecars onto a 360 |
| 07:49.29 | YoMama | i wonder if the cisco 7960 with two sidecars works |
| 07:49.34 | trixter | Mmmm.. ribs |
| 07:49.37 | YoMama | a cisco 7960 will take two sidecars |
| 07:49.41 | IronHelix | <3 ribs |
| 07:50.07 | YoMama | IronHelix: have u set up the 360 with a sidecar before? |
| 07:50.18 | IronHelix | no |
| 07:50.20 | IronHelix | brb |
| 07:50.39 | trixter | the sidecar must be nice, as you ride around town your dog can ride in the sidecar |
| 07:50.45 | trixter | get some wind in his face |
| 07:52.12 | trixter | its cold and stuff |
| 07:52.20 | IronHelix | lol |
| 07:52.37 | YoMama | ha |
| 07:52.38 | jaike | there was a discussion yesterday about canreinvite able to lower system utilization. how? |
| 07:52.55 | *** join/#asterisk Sajid_Khan (n=human@203.145.159.37) |
| 07:53.19 | trixter | if you directly connect the rtp endpoints asterisk doesnt have irqs from the network traffic, it doesnt have cpu overhead for processing rtp streams, it doesnt do codec translation, etc |
| 07:53.41 | trixter | there are a lot of reasons why it lowers utilization on that specific server but there are some dangers depending on environment |
| 07:54.38 | jaike | dangers? like natted networks? |
| 07:54.56 | trixter | if you want to ensure that you get billing info (CDR) the media streams have to go through asterisk because if someone doesnt send end of call back to asterisk (say both endpoints just unplug their device instead of hanging up - especially becuase they know its a free call that way) or whatever |
| 07:55.27 | trixter | that generally isnt as big of a problem because normally net->net is free and its harder for them to just unplug the pstn, but its something to consider |
| 07:56.00 | jaike | oh ok..so it can mess up CDR |
| 07:56.11 | trixter | yeah but generally that isnt as easy in real world situations |
| 07:56.15 | trixter | its just 'possible' |
| 07:56.51 | trixter | if you have to do wiretapping you cant if you directly connect them to someone else.. so like CALEA requirements in america - if you dont have the RTP data you cant record it |
| 07:57.23 | YoMama | ok...bedtime |
| 07:57.25 | YoMama | goodnight y'all |
| 07:57.29 | YoMama | thanks for the help |
| 07:58.17 | trixter | if you run a calling card system where someone can for example press *# to make another call without hanging up and calling back you must use sip-info for dtmf, which may not be supported on the remote end |
| 07:59.52 | *** join/#asterisk |||sLaSh||| (i=th3_gam3@203.215.100.96) |
| 08:01.49 | |||sLaSh||| | hi im using 7960 and asterisk@home, i got the message tftp file not found, "system is unavailable" |
| 08:03.08 | jaike | trixter: you lost me there :). googling canreinvite |
| 08:03.35 | trixter | um there was an american idol? and I missed the chance to skew the voting results? |
| 08:03.41 | trixter | I have 120k votes to cast! |
| 08:04.11 | trixter | I gotta find someone to watch it for me so they can tell me who was worst |
| 08:05.43 | IronHelix | lol |
| 08:06.46 | *** join/#asterisk oej (n=oej@62.97.243.70) |
| 08:11.17 | zoa | hey ho olle |
| 08:11.41 | zoa | any news on the exams |
| 08:16.09 | *** join/#asterisk Fedoracore6 (n=dsd@219.95.15.117) |
| 08:20.00 | *** join/#asterisk Gunnar (n=gunnar@62.97.243.70) |
| 08:21.56 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-244.claranet.co.uk) |
| 08:22.20 | [av]bani | hmm anyone used avaya phones ? |
| 08:24.48 | L|NUX | hello every one |
| 08:25.17 | L|NUX | can any one tell me is this possible to run asterisk on multiple ports ? |
| 08:25.32 | trixter | how do you mean? |
| 08:25.34 | trixter | specifically |
| 08:25.35 | L|NUX | like i have provider for both 8891 and 5060 how can i bind them in my asterisk ? |
| 08:25.45 | jaike | am able to get passthrough to work for g729 between my phone and our provider with asterisk in betwen, but if i use put Monitor in the dialplan, i see 2 decoder licenses used. Asterisk has to decode both in and out channels? |
| 08:25.47 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:26.04 | trixter | you need a license to encode or decode |
| 08:26.07 | trixter | not just decode |
| 08:26.07 | *** join/#asterisk welles (n=welles@61.150.43.113) |
| 08:26.24 | trixter | the way it works is that it will decode from you, then reencode to the remote site |
| 08:26.28 | jaike | 0/2 encoders/decoders of 30 licensed channels are currently in use |
| 08:26.37 | trixter | it could be made such that it only needs 1 license but asterisk doesnt work that way |
| 08:26.41 | jaike | without monitor, its 0/0 |
| 08:26.58 | trixter | ok then its decoding both streams |
| 08:27.05 | trixter | as such it requires 2 licenses one for each endpoint |
| 08:27.17 | trixter | either way, it still could be done to only use 1 license |
| 08:27.20 | trixter | but it doesnt work that way |
| 08:27.24 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:27.50 | welles | trixter, cmd 'meetme list <conferno>' can not return the detail info in my asterisk. can u help me? |
| 08:28.13 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 08:28.24 | welles | trixter, i really troubled by it. |
| 08:28.36 | jaike | welles: are you using 1.2.4? it works fine with us |
| 08:32.51 | *** join/#asterisk CaRb0n^ (i=Genocide@203.81.238.51) |
| 08:34.34 | shido6 | anyone diabetic? |
| 08:35.02 | jaike | www.voip-info.org currently down? |
| 08:36.34 | [av]bani | i get to it fine |
| 08:37.41 | *** join/#asterisk wellng (n=welles@222.90.175.97) |
| 08:37.52 | jaike | hmmm |
| 08:38.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:42.14 | johnsu01 | if I have this: "exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@jnctn)" in my extensions.conf, and an entry for [jnctn] in my sip.conf, why might asterisk still be treating a number like 15555551212 as a local extension? |
| 08:42.51 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 08:43.00 | justinu | ipv6? |
| 08:44.14 | johnsu01 | hm? don't think so. |
| 08:44.55 | shido6 | errr |
| 08:44.59 | shido6 | sip debug. |
| 08:45.11 | shido6 | what context is it looking in |
| 08:48.26 | wellng | hi jaike iam welles i come back |
| 08:48.28 | johnsu01 | shido6: It's looking in default. |
| 08:49.17 | wellng | jaike, let 's continute to talk my issue .cmd meetme list <conferno> ok? |
| 08:49.18 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 08:49.38 | jaike | wellng: meetme command works fine in 1.2.4, or even in 1.2.0 |
| 08:49.59 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:50.43 | wellng | jaike, yes. i know. but now i don't know why my asterisk can return the detail but only one line . |
| 08:51.07 | [av]bani | http://www.intertex.se/products/page.asp?iPageID=146 |
| 08:51.07 | [av]bani | o.o |
| 08:51.30 | justinu | (00:42:51) florz [n=florz@2001:1a50:503c:0:0:0:0:1] entered the room. |
| 08:51.30 | justinu | (00:43:01) justinu: ipv6? |
| 08:51.37 | justinu | that's an ipv6 address |
| 08:51.41 | johnsu01 | oh, right :) |
| 08:51.59 | wellng | jaike, yes. i know. but now i don't know why my asterisk can not reurn the detail but only one line . |
| 08:53.20 | jaike | that i cant answer |
| 08:53.36 | johnsu01 | ah, got it |
| 08:54.39 | jaike | this is what i get when when i issue the meetme command |
| 08:54.51 | jaike | User #: 01 400 400 Channel: SIP/400-f697 (unmonitored) |
| 08:54.52 | jaike | 1 users in that conference. |
| 08:55.25 | wellng | jaiger, yes. this is also what i expect .but mine is only the last line |
| 08:57.04 | wellng | jaike, , yes. this is also what i expect .but mine is only the last line |
| 08:57.11 | jaike | recompile? hehe |
| 08:58.00 | wellng | maybe can not work.because my asterisk version was 1.2.1 and update to 1.2.4 .it still can not work |
| 09:01.05 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 09:02.30 | *** join/#asterisk walhala (n=walhala@stardust.noc.frontier.fr) |
| 09:02.31 | walhala | does anyone know how to setting up sms with * ? I use trunk version actually |
| 09:06.02 | *** join/#asterisk sergeus|w (n=sergeus@ippe-245.ippe.ru) |
| 09:10.01 | [av]bani | http://www.voip-info.org/wiki/view/Pingtel+Hardphone |
| 09:10.08 | [av]bani | good lord. ugliest phone ever? |
| 09:10.17 | [av]bani | looks like something out of the jetsons |
| 09:12.49 | trixter | maybe it is |
| 09:14.00 | trixter | the freeswitch hardphone doesnt sound too bad ... hardphone that can act as either your phone, your pbx, both, color lcd touchscreen, runs linux, playing with the internals is encouraged, projected price $300 |
| 09:16.10 | [av]bani | ? |
| 09:17.33 | liran_ | would anyone care to test my billing program? it's a config file, and cgi script. it reads everything from master.csv |
| 09:17.51 | trixter | how does it do billing? |
| 09:18.35 | *** join/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br) |
| 09:19.29 | [av]bani | paypal! |
| 09:19.41 | trixter | e164.org earned $1.50 today.. didnt know there were so many people using enum, its not a lot of cash, but its less than 1 day, it also suprised me cause I figured they would only do $5/month |
| 09:20.00 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 09:20.51 | trixter | [av]bani: the freeswitch people are designing a hardphone, with the goal that its open enough that you can do whatever you want with it. it will run freeswitch in linux, although you can swap that out if you want ... it sounds really cool, and for some people that would be the only device they need.. for just email 640x480 lcd touchscreen is fine |
| 09:21.09 | trixter | CF slot too so that you can store your data and take it with you |
| 09:21.19 | liran_ | trixter: with a configuration file. it looks up rates you define for the phone number field in master.csv |
| 09:21.36 | trixter | liran_: does it support custom rate tables per customer? |
| 09:21.49 | liran_ | trixter, yeah, its very dynamic |
| 09:22.28 | liran_ | trixter, i've got a sf project page up but i havent uploaded any files yet cause i want to test it somewhere first. |
| 09:22.28 | alexandrekeller | I think there's a bug on app_queue; can anybody help me ?! |
| 09:24.22 | trixter | I might look it at, where can I get it? what language is it written in? |
| 09:24.26 | zoa | trixter: the phone is probably going to be very expensive then :) |
| 09:24.30 | trixter | what platforms does it work on? |
| 09:24.43 | trixter | zoa: projected price is $300 |
| 09:24.54 | trixter | which isnt bad for what you get considering it can be your whole pbx if you wanted |
| 09:25.17 | zoa | i dont think they will be able to make it for that price |
| 09:25.23 | trixter | why not? |
| 09:25.41 | liran_ | trixter, ill email it to you. its in perl and requires ofcourse apache with cgi support and it needs a perl module. |
| 09:25.41 | zoa | they will need to have to pay for a big production run |
| 09:26.17 | trixter | they seem to think that they can after talking to people who will be building it |
| 09:26.31 | liran_ | trixter, if you see fit, i'd like to customize straight down to your needs and have some sort of restricted access just to edit the code and test it. |
| 09:26.43 | trixter | oh well if its just a cgi I am not that interested ... |
| 09:26.51 | liran_ | how come? |
| 09:27.17 | trixter | because I am not that interested in a web based solution |
| 09:27.19 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 09:27.26 | liran_ | ahh ok |
| 09:27.27 | cjk | hi, any cdr master here? |
| 09:27.31 | trixter | I guyess I will look at it, but prolly wont use it for real.. trixter@0xdecafbad.com |
| 09:27.47 | zoa | it would be cool though |
| 09:27.54 | cjk | if i do a forkcdr, any command following this modifiying cdr data should only be applied for the new cdr? correect? |
| 09:27.56 | liran_ | trixter, ok. would it be possible for me to test it on your box? |
| 09:28.10 | trixter | it also probably looks at the src field, if so then it certainly wont work for what I am doing |
| 09:28.19 | trixter | I can test it for you on my box |
| 09:30.35 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 09:31.02 | wellng | hi all, my asterisk ' cmd 'meetme list <conferno>' can not work.it can not return the detail of a conference room. anyone has ideas? |
| 09:31.34 | trixter | what details specifically are you looking for? |
| 09:32.22 | wellng | trixter, i want control the conference using agi |
| 09:32.48 | trixter | that doesnt answer the question |
| 09:35.00 | wellng | trixter, for example .User #: 01 11234 welles Channel: IAX2/11234 (unmonitored) what i want get is 11234 |
| 09:36.18 | trixter | what is your verbosity set to? |
| 09:36.29 | wellng | trixter, 10 |
| 09:36.46 | wellng | trixter, any value will not work |
| 09:37.34 | alexandrekeller | hi all |
| 09:37.55 | [av]bani | trixter: url for the freeswitch hardphone? |
| 09:37.59 | alexandrekeller | I'm looking for someone who can help me with app_queue ?! |
| 09:38.08 | alexandrekeller | anybody ?! |
| 09:38.32 | wellng | trixter, more stange is that it will still occur i reboot the machine or recompile the asterisk |
| 09:43.59 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 09:45.49 | [av]bani | ~phones |
| 09:45.50 | jbot | phones is probably at http://bani.anime.net/phones/ |
| 09:45.54 | [av]bani | list is getting long :o |
| 09:47.17 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:47.43 | fishboy1669 | hi |
| 09:47.54 | fishboy1669 | is royk here today? |
| 09:48.18 | zoa | he just joined |
| 09:48.27 | fishboy1669 | cooleo |
| 09:49.53 | RoyK | fishboy1669: tried the mrtg plugin as well? |
| 09:49.54 | trixter | [av]bani: afaik they dont have anything but conversations right now. I just happen to have talked to them about it |
| 09:50.05 | trixter | you are free to do the same ... |
| 09:51.13 | trixter | wellng: I am almost thinking that you have a mad module for meetme and that you may want to look at reinstalling that module and making sure that it gets overwritten with the current install. that may not be it, but there is certainly something not right |
| 09:51.13 | [av]bani | heh. seems to me they could just take a reference design off the shelf (like snom and grandstream do) and build something around it |
| 09:51.20 | trixter | I dont use meetme enough to know what the problem is |
| 09:52.55 | trixter | from what they have said they are building from the ground up to get the features they want |
| 09:53.01 | trixter | they also said the case alone is like $25 :( |
| 09:53.19 | [av]bani | unless you buy in lot 10,000, cases are usually expensive |
| 09:53.20 | *** part/#asterisk Aragone (n=arathorn@puma.mxtelecom.com) |
| 09:53.46 | trixter | well it was something about it being custom made rather than an off the shlef case they can shove something into |
| 09:54.04 | [av]bani | seems silly, unless they want to do something with the case nobody has ever done before |
| 09:54.49 | trixter | well 1. it has to be a deskphone case, 2. it needs the lcd, I am unsure of the other properties that are required but at the very least those are design constraints |
| 09:54.56 | trixter | most deskphones dont have a 640x480 lcd |
| 09:55.01 | [av]bani | custom made cases are $$$ just for the design, because they have to be made for the molding process |
| 09:55.21 | [av]bani | if you get an off the shelf case, thats already been taken care of |
| 09:55.27 | trixter | that is aparently built into the cost from what I gathered |
| 09:55.31 | trixter | which results in $25/case |
| 09:55.32 | [av]bani | 640x480 seems over the top to me |
| 09:55.40 | [av]bani | if youre doing a videophone, sure |
| 09:55.42 | trixter | well its more than a simple phone |
| 09:55.59 | trixter | the system runs linux, you can do email, sms, im, etc all off the phone |
| 09:56.09 | trixter | for many people there wont be a great need for an actual computer |
| 09:56.10 | [av]bani | nothing special there, i could do that with a zaurus |
| 09:56.25 | trixter | you wont have the cpu power off the zaurus |
| 09:56.32 | [av]bani | for what? |
| 09:56.37 | trixter | scroll up |
| 09:56.38 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:56.48 | [av]bani | what, you need cpu for im? email? |
| 09:57.03 | trixter | I was refereing to the at least 2 times I siad that the phone could be your pbx |
| 09:57.18 | trixter | as in it runs freeswitch you can have *other* things connect to it, it can do voicemail for those *other* phones, etc |
| 09:57.21 | trixter | codec translations |
| 09:57.22 | [av]bani | never tried, but i could probably put asterisk on zaurus |
| 09:57.37 | trixter | and how many channels can you get working? |
| 09:57.39 | [av]bani | its beefier than my wrt54gs, which runs zaurus |
| 09:57.43 | [av]bani | which runs * |
| 09:57.55 | trixter | how about recording to voicemail at the same itme you are processing multiple other channels with codec translations? |
| 09:57.58 | [av]bani | shrug, anything embedded is going to be about the same |
| 09:58.07 | trixter | if you say so |
| 09:58.08 | [av]bani | unless you go x86 variant |
| 09:58.10 | [av]bani | or ppc |
| 09:58.37 | [av]bani | arm, mips, youre not going to be able to push them very far |
| 09:58.43 | trixter | regardless they are doing it and doing it differently than you said, such as lower price, etc.. they have talked to hardware designers/manufacturers about this specific phone as well |
| 09:58.47 | trixter | they arent talking about arm |
| 09:58.56 | trixter | I never said they were building a zaurus |
| 09:59.01 | trixter | I said it had more cpu than a zaurus |
| 09:59.13 | [av]bani | well, if you want anything approaching something for pbx, you want ppc then |
| 09:59.23 | [av]bani | spendy... |
| 09:59.32 | trixter | I hope the release more info on it soon, especially pictures |
| 09:59.43 | [av]bani | ppc is not cheap to embed |
| 09:59.46 | [av]bani | sadly |
| 09:59.53 | trixter | the concept is nice, and the fact that they arent trying to make it all secret and closed is also nice |
| 10:00.00 | trixter | its not a ppc |
| 10:00.04 | trixter | but anyway |
| 10:00.28 | [av]bani | well, mips and arm dont have real processing power for much... |
| 10:00.31 | trixter | I would like to see some initial design pictures of the case |
| 10:01.39 | [av]bani | i assume they rent going to sell it for any profit |
| 10:01.41 | *** join/#asterisk cng^ (n=cg@217.23.169.4) |
| 10:01.43 | wellng | trixter, yes. you are right. i rm a module . they work fine |
| 10:02.10 | trixter | as I understand it htey also want to put zeroconf in it so that when you plug it in to your network other zeroconf enabled devices will just work, granted you will have to enter a username and password, but you wont have to configure everything else |
| 10:02.39 | trixter | [av]bani: I dont know how much profit they are planning on making with it, I do know they arent profiting off the software side of things |
| 10:02.43 | *** part/#asterisk cng^ (n=cg@217.23.169.4) |
| 10:02.54 | [av]bani | if they want to build something for themselves, great they can knock themselves out |
| 10:03.14 | trixter | why would they have a suggested sales price if that were the case? |
| 10:03.17 | [av]bani | but i have seen soooo many people make 'open hardware' which failed because they made some retarded design decision nobody wanted or which was just not practical |
| 10:03.18 | wellng | trixter, thanks |
| 10:03.40 | [av]bani | 300 sounds just about what it would cost for small production run of something |
| 10:03.58 | trixter | I fail to see why you want to argue with me when I have made it clear its someone else that is doing this, and the fact that they have presented, so I feel its time to say again, please let it go |
| 10:04.04 | trixter | wellng: np, glad it worked out for you.. |
| 10:07.01 | alexandrekeller | is this a business channel ?! |
| 10:09.58 | trixter | this is an asterisk channel, so basically anything that is compatible with asterisk is fair game |
| 10:11.36 | trixter | wow someone is selling for $89 a section of the CFR which you can get for free.. makes you wonder how some companies can even get customers |
| 10:15.44 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 10:18.22 | [av]bani | does editline still need to be exorcised from asterisk? |
| 10:18.58 | pif | hi, any user of the cisco 7920 (wi-phone)? |
| 10:19.08 | [av]bani | thats an expensive phone. |
| 10:19.21 | pif | i know :) |
| 10:19.57 | [av]bani | youre probably the only one here who has one. :/ |
| 10:20.17 | pif | and it's crap to boot |
| 10:20.31 | pif | cisco is a bunch of theives |
| 10:23.45 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:24.13 | [av]bani | no, theyre just often out of touch with their customer base |
| 10:24.30 | [av]bani | sometimes it takes them a couple models before they get a technology right |
| 10:24.49 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 10:25.04 | [av]bani | though lately they seem to be really stupid, making their support pages IE-only |
| 10:25.08 | pif | cisco is a profit-driven company, not a technology-driven company |
| 10:25.48 | [av]bani | err.. all companies are profit driven, or they wouldnt be companies :) |
| 10:26.38 | [av]bani | all the wifi phones on the market seem to suck, its suprising the expensive cisco one does too |
| 10:27.08 | [av]bani | i might expect that of the zyxel |
| 10:27.11 | pif | what I mean is: they'll not stop at taking ugly technical shortcuts to stuff their customers (ans lose some in the way) |
| 10:27.16 | [av]bani | but cisco is suprising |
| 10:27.48 | [av]bani | hmm.. i dont know if i ever saw cisco take a technical shortcut. they just like proprietary stuff |
| 10:27.52 | pif | whereas a tech-driven co will not stoop to rebranding a chinese made piece of crap and mark it up 5X |
| 10:28.06 | [av]bani | fortunately the market they are in is not favorable to proprietary |
| 10:28.13 | [av]bani | so they can only do so much damage :) |
| 10:28.47 | pif | the suits like cisco now |
| 10:28.52 | *** join/#asterisk Assid (n=assid@203.115.64.11) |
| 10:28.53 | [av]bani | they always did |
| 10:29.00 | pif | 'cause it's 'safe' |
| 10:29.11 | [av]bani | it was always 'safe', and for a long time was the only game in town really |
| 10:29.26 | johnsu01 | I can't get audio on my sip calls. I think I probably need some of the nat settings, since there is a firewall between my asterisk and Junction, but I can't seem to get them right. |
| 10:29.46 | [av]bani | johnsu01: nat=yes qualify=yes in sip.conf |
| 10:30.40 | [av]bani | pif: what cisco is good at is obsoleting your investment very quickly, making you re-buy the same stuff over and over. |
| 10:30.50 | [av]bani | pif: something microsoft is only just getting good at |
| 10:31.09 | pif | a pattern there |
| 10:31.33 | [av]bani | of course tech stuff becomes worthless insanely fast in general... |
| 10:31.52 | pif | free software is having remarkably little impact on these attitudes yey |
| 10:31.54 | pif | yet |
| 10:31.55 | [av]bani | $2000pc is worth $100 in 4 years :/ |
| 10:32.45 | [av]bani | what? free software kicked sgi's and dec's ass |
| 10:32.51 | [av]bani | and it's pounding sun |
| 10:33.13 | [av]bani | sun is having to do a major attitude readjustment thanks to free software |
| 10:33.41 | [av]bani | and.. juniper is competing with cisco, and they build their routers around bsd... |
| 10:34.18 | johnsu01 | [av]bani: still no luck. |
| 10:34.29 | [av]bani | give asterisk a few more years and im sure it will start pressuring cisco's callmanager |
| 10:34.37 | [av]bani | johnsu01: :/ |
| 10:36.28 | johnsu01 | gah, now I've got some audio.. |
| 10:36.43 | *** join/#asterisk X-Gen (n=x-gen@dsl-146-124-49.telkomadsl.co.za) |
| 10:36.54 | johnsu01 | I ring my own cell phone, and when I send the call to voicemail, the call drops. |
| 10:37.01 | johnsu01 | But at least I hear the ringing :) |
| 10:38.13 | *** join/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br) |
| 10:38.42 | alexandrekeller | I really need some help on app_queue....anybody please ?! |
| 10:38.59 | johnsu01 | very strange, I hear ringing when I call some numbers, but not others. |
| 10:39.53 | *** part/#asterisk X-Gen (n=x-gen@dsl-146-124-49.telkomadsl.co.za) |
| 10:40.03 | jaike | alexandre: just post your question |
| 10:41.21 | alexandrekeller | I have 5 agentes logged in. But only one call at a time rings on them, when a second call enters the queue it' kept on hold until the first call is answered |
| 10:41.28 | alexandrekeller | is it a bug or what ?! |
| 10:43.56 | jaike | asterisk version? |
| 10:44.25 | alexandrekeller | 1.2.4 |
| 10:44.36 | alexandrekeller | but it happens too on 1.0.10 |
| 10:45.11 | jaike | ive never really noticed that |
| 10:45.36 | alexandrekeller | me neither until yesterday |
| 10:45.37 | johnsu01 | [av]bani: So, I can hear ringing now, but as soon as the call is picked up by voicemail, I lose the audio. The call stays connected, though. |
| 10:45.38 | *** join/#asterisk Falle (i=falstaf@213.141.80.88) |
| 10:46.20 | jaike | alexandre: lemme test that |
| 10:46.27 | alexandrekeller | ok |
| 10:47.03 | [av]bani | johnsu01: ringing is via sip signaling, audio is via rtp. so your rtp is being blocked |
| 10:47.30 | johnsu01 | hm. I have the rtp.conf ports set according to Junction's numbers, and I have my firewall forwarding that range of ports as well... |
| 10:48.11 | [av]bani | origination from junction to your * ? |
| 10:48.59 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 10:49.03 | ramtha | hi |
| 10:49.23 | ramtha | 2 tep wilcdards (quad span) |
| 10:49.29 | ramtha | one is working correct |
| 10:49.36 | ramtha | the second seems not so |
| 10:49.41 | [av]bani | if your * is behind a nat firewall, you'll probably need to set externip= in sip.conf |
| 10:49.49 | ramtha | in /proc/zap/ i only have 5 spans |
| 10:50.19 | ramtha | i think i must have 8 |
| 10:50.24 | jaike | alexandre: same here..i guess its supposed to work that way |
| 10:50.34 | jaike | FIFO |
| 10:50.35 | ramtha | zttool did not display the second card |
| 10:50.39 | ramtha | what is wrong here |
| 10:50.39 | alexandrekeller | really ?! |
| 10:50.45 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 10:50.54 | jaike | try strategy=ringall :) |
| 10:51.40 | johnsu01 | [av]bani: I've added externip, but same result. |
| 10:51.55 | alexandrekeller | oh sh.....I don't believe..... |
| 10:52.01 | alexandrekeller | well, thanks mate |
| 10:52.08 | johnsu01 | Will that take effect on reload, or do I need to restart? I'll try restarting as well. |
| 10:53.03 | [av]bani | johnsu01: sip reload should be enough |
| 10:53.05 | *** join/#asterisk SupZ (n=icechat5@200-158-166-207.dsl.telesp.net.br) |
| 10:53.42 | [av]bani | johnsu01: you should see if you can packet dump on your firewall, see if rtp packets are arriving and being blocked |
| 10:56.25 | sternn | ls |
| 10:57.10 | [av]bani | mv |
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| 10:58.10 | johnsu01 | [av]bani: they rtp packets are getting through. I can see them via ethereal on the machine behind the firewall. |
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| 11:01.02 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 11:03.02 | trixter | http://www.trxtel.com/index.php?page=Tollfree_Termination make money sending voip traffic, what a bargain |
| 11:03.37 | [av]bani | johnsu01: so they are probably not being translated properly |
| 11:08.37 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 11:09.28 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 11:09.41 | stoffell | goodday all |
| 11:11.02 | johnsu01 | [av]bani: I'll have to work on it more later I guess. At least I've got the calls working with iax. |
| 11:11.08 | johnsu01 | [av]bani: thanks for the tips. |
| 11:13.15 | sternn | ftp ftp.digium.com |
| 11:13.25 | sternn | doh! Wrong window again. |
| 11:15.26 | *** join/#asterisk aymeric (n=ablazy@62.36.227.220) |
| 11:16.15 | stoffell | lol |
| 11:19.51 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 11:21.57 | *** join/#asterisk DataCompBoy (n=datacomp@217.8.236.1) |
| 11:22.06 | DataCompBoy | Hi all :) |
| 11:22.27 | DataCompBoy | Sorry, may be I have missed anything... But: is pre-compiled .deb's for ztdummy present? |
| 11:22.45 | fourcheeze | DataCompBoy: not last time I checked |
| 11:22.59 | fourcheeze | (about 2 days ago) |
| 11:23.07 | DataCompBoy | what about asterisk 1.2.4 ? |
| 11:23.16 | stoffell | you could make one yourself with checkinstall or something |
| 11:23.16 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 11:23.19 | fourcheeze | yeah, there's one of those I think |
| 11:23.49 | fourcheeze | DataCompBoy: it's easy to build the zaptel modules if you have the kernel headeres |
| 11:24.29 | DataCompBoy | I know... but I have trying to stay with "less hands" on that box :D |
| 11:25.55 | DataCompBoy | o, asterisk I see there: http://ftp-master.debian.org/new.html |
| 11:28.37 | fourcheeze | you can easily make a kernel package though |
| 11:28.46 | fourcheeze | only takes a couple of minutes |
| 11:29.03 | aymeric | hi, anyone using the channel chan_h233 in production environment ? |
| 11:29.17 | fourcheeze | DataCompBoy: I do it on one box and then reuse the package on other boxen |
| 11:29.49 | DataCompBoy | fourcheeze: i'm about "easy upgrade" :) |
| 11:36.44 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:37.42 | fourcheeze | DataCompBoy: how often do you upgrade kernels? |
| 11:38.08 | trixter | I do it daily |
| 11:41.33 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 11:41.59 | DataCompBoy | not often.... |
| 11:42.17 | nextime | ther's no app_sql_postgres in latest svn trunk? |
| 11:42.50 | fourcheeze | DataCompBoy: so its a 2 minute job not very often |
| 11:42.58 | DataCompBoy | :DD |
| 11:43.00 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 11:43.04 | DataCompBoy | about once a month |
| 11:43.23 | DataCompBoy | but in production -- doesn't touch anything |
| 11:43.23 | DataCompBoy | :) |
| 11:43.34 | DataCompBoy | for now I have upgraded from 2.6.12 to 2.6.15 for ztdummy |
| 11:43.38 | Mavvie | is there any listener-thread reserved space to store thread specific data? |
| 11:48.59 | DataCompBoy | hmmm... where is packages, that listed at http://ftp-master.debian.org/new.html ?? |
| 11:53.05 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 11:55.41 | hertell_sleeping | Is there a way in how to create an extension that would return me the PSTN dialtone on a spa-3000? |
| 11:56.22 | *** part/#asterisk DataCompBoy (n=datacomp@217.8.236.1) |
| 11:57.50 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 11:58.13 | Hertell | what i mean that if i hit 0 (without the trailing #-key) asterisk would pick up the line to my PSTN |
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| 12:10.52 | *** join/#asterisk enemy^x (n=eqwrweqr@morpheus.dataguard.no) |
| 12:11.24 | enemy^x | why is the 0 removed when dialing out? Is that in zapata.conf? |
| 12:12.20 | Hertell | enemy^x: what do you mean? |
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| 12:19.18 | nextime | is there something like app MySQL() but for postgres? |
| 12:19.39 | nextime | ( maybe as an external addons? ) |
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| 12:37.30 | Inkubot | hi |
| 12:37.39 | Inkubot | i've got a simple questions about a linksys PAP2 |
| 12:37.49 | Inkubot | how can i enable that both FXS act as one ? |
| 12:38.17 | I-MOD | just put a phone line splitter on one port and use that |
| 12:38.38 | Inkubot | i can make call with both but when i recieve just one ring |
| 12:38.47 | Inkubot | ok. |
| 12:40.31 | I-MOD | or do you want it to ring both phones whenever a call comes in, but still be able to make two separate calls at the same time? |
| 12:40.58 | I-MOD | cause i dunno about that one |
| 12:41.05 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:43.56 | Inkubot | nah.. both phones ring when a call comes in, and only one way call out. |
| 12:44.09 | {zombie} | Inkubot: configure asterisk to dial both - eg Dial(SIP/101&SIP/102) |
| 12:44.12 | Inkubot | thats what i wan't.. and its only to know how to do it.. |
| 12:44.18 | Inkubot | just that. |
| 12:44.25 | trixter | wouldnt each port have to have its own account? and when a call comes in you dial both extensions? |
| 12:44.36 | trixter | yeah like zombie said |
| 12:44.37 | iDunno | yup |
| 12:44.48 | Inkubot | {zombie} thanks.. but this sip server it is not an Asterisk :\ |
| 12:44.57 | {zombie} | Inkubot: well this is #asterisk |
| 12:45.03 | Inkubot | i know.. |
| 12:45.06 | [swb] | anyone had a problem where you are using n(label) and Goto(label) and it doesnt work? |
| 12:45.14 | {zombie} | so if you ask questions you get asterisk specific answers.. sorry :) |
| 12:45.18 | [swb] | Its just "exiting non zero" and hanging up |
| 12:45.23 | Inkubot | but you always know how to do this things.. |
| 12:45.31 | {zombie} | I guess the answer is to install an asterisk box and use that to talk to your sip provider |
| 12:45.36 | Inkubot | {zombie} it is not a problem that :D |
| 12:45.39 | {zombie} | and hang the PAP2 off the asterisk box |
| 12:45.46 | Inkubot | yeps |
| 12:46.55 | Inkubot | all i want to know is how to do this.. |
| 12:47.08 | Inkubot | :) |
| 12:47.51 | {zombie} | start by reading the asterisk docs, then installing asterisk |
| 12:47.59 | {zombie} | ~docs |
| 12:48.01 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:48.24 | Inkubot | i know how to do that :P |
| 12:48.33 | {zombie} | ask specific questions then |
| 12:48.43 | Inkubot | that is why i came here to ask.. |
| 12:48.56 | Inkubot | i know that here are answer for that kind of stuff |
| 12:49.06 | [swb] | Hello, anyone with a problem of using n(label) and then a Goto(label) and asterisk just hangsup without doing the goto? |
| 12:49.09 | {zombie} | I gave you the answer to that |
| 12:49.15 | Mavvie | include/asterisk/logger.h asterisk.c cli.c logger.c |
| 12:49.17 | {zombie} | you just need to log the PAP2 both ports into asterisk |
| 12:49.19 | Inkubot | yeps.. and i apreciate :D |
| 12:49.22 | Mavvie | woops. no enter. |
| 12:49.24 | {zombie} | and register asterisk with your sip provider |
| 12:49.28 | [swb] | this is the wierdest problem I have seen |
| 12:49.45 | Inkubot | {zombie} ok |
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| 13:11.49 | Mavvie | new toy! http://bugs.digium.com/view.php?id=6524 |
| 13:13.00 | trixter | personally I think each console should be able to set its verbosity level, there is afaik no way, even with that patch to turn down verbosity levels |
| 13:13.33 | crusher | I agree |
| 13:13.50 | Mavvie | trixter: the verbose logging in asterisk is a very tricky bussiness. |
| 13:13.53 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 13:14.01 | trixter | but it doesnt have to be |
| 13:14.16 | trixter | the cli stuff is tricky far more than it has to be, have you looked at the code for the cli? |
| 13:14.24 | Mavvie | it's not until the lowest levels that it finds the remote consoles. |
| 13:14.40 | trixter | further why is there a client and daemon in the same program? I dont want firefox embedded into apache |
| 13:14.58 | mutilator | it's not a client |
| 13:14.58 | trixter | I failed to see why it was acceptable to have both into one program such as it is with asterisk |
| 13:15.15 | trixter | it is, when you run asterisk -r it makes a connection over a stream pipe as a client |
| 13:15.24 | trixter | yet that client has all the code for the daemon, which largely it doesnt need |
| 13:16.00 | trixter | it doesnt need to know how to load modules because when in -r mode it wont do that, it doesnt need to know how to load channel drivers because again, with -r it wont do that |
| 13:16.29 | Mavvie | some well placed #ifdefs could generate an "asterisk-client" program. |
| 13:17.00 | trixter | and technically anything that speaks the correct protocol can open that stream pipe and talk to asterisk, so someone could write a master program that talks to both the stream pipe and the manager interface allowing for more robust client programs |
| 13:17.22 | trixter | well it would take more than a few #ifdefs and they would have to be VERY well placed, such that it eliminates much of the code that goes into the asterisk binary |
| 13:17.30 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:19.11 | [swb] | has anyone had a problem using Goto and priority labels, it works fine all over my dialplan but in one app it just hangs up and refuses to perform the goto, I cant figure out why as it looks the same as all the places it is working |
| 13:19.16 | [swb] | I get 2006-02-17 13:17:13 WARNING[12005] pbx.c: Priority 'test' must be a number > 0, or valid label in the logs |
| 13:19.30 | BugKham | I have some questions about dialplan |
| 13:23.02 | BugKham | we always jump to priority + 101 in case of error, is that right? |
| 13:23.22 | BugKham | 1 -> 102 -> 203 -> 304 ... |
| 13:24.55 | *** join/#asterisk ChrisUK (n=chris@82.108.126.170) |
| 13:26.00 | ChrisUK | Hello :) |
| 13:26.45 | fugitivo | anyone using rxfax spandsp-0.0.2 with asterisk 1.2.4? |
| 13:27.30 | fugitivo | (spandsp-0.0.2pre25) |
| 13:27.53 | ChrisUK | Anyone else got echo problems on SIP to SIP calls on a local LAN? :S |
| 13:28.05 | ChrisUK | with the Grandstream GXP2000 |
| 13:28.30 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:28.54 | trixter | no I do that all the time and have no problems |
| 13:28.59 | trixter | with a gxp2000 specifically |
| 13:29.09 | [swb] | ChrisUK, yes before I upgraded the firmware |
| 13:29.21 | ChrisUK | what version of firmware you using now ? |
| 13:29.35 | [swb] | I got fed up with the phone and sent it back, using SNOM 320 instead |
| 13:29.48 | trixter | I was running 1.0.1.9 upgraded to 1.0.2.3 had problems with neither |
| 13:29.58 | ChrisUK | lol ok |
| 13:30.09 | ChrisUK | well im sort of stuck ive got 50 of them |
| 13:30.14 | [swb] | hehe |
| 13:30.22 | [swb] | I hear they are alot better on the later version of the firmware |
| 13:30.30 | [swb] | and that the customer service from Grandstream is good |
| 13:30.33 | trixter | 1.0.2.3 isnt bad and has soem features that I felt were missing |
| 13:30.34 | [swb] | have you tried contacting them? |
| 13:30.39 | ChrisUK | No not yet |
| 13:30.47 | trixter | like telling you about missed calls and quickly letting you view those calls and call those people back |
| 13:31.16 | [swb] | so anyone had problems Using Goto and Labeled priorities in the dialplan |
| 13:31.19 | [swb] | its driving me up the wall |
| 13:31.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:31.32 | [swb] | I have used them all over with no probs but suddenly I just cant get one example to work |
| 13:31.39 | [swb] | the pbx thinks its an invalid label |
| 13:31.49 | *** join/#asterisk JohnJacob (n=m00p@pool-71-246-132-74.aubnin.fios.verizon.net) |
| 13:32.16 | [swb] | I have the most simple example possible now and it still wont work |
| 13:32.22 | [swb] | the goto just exits and hangsup |
| 13:34.34 | [TK]D-Fender | [swb] : Pastebin it.... |
| 13:34.41 | [TK]D-Fender | ~pb |
| 13:34.47 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:35.45 | [swb] | how much should I paste bin? I cant see it being a problem with the two lines as they are THE most simple, must be something else upsetting it |
| 13:35.47 | [swb] | pasteing now |
| 13:36.06 | fugitivo | wow, irc is great with a console with the size of the screen |
| 13:37.23 | [swb] | [TK]D-Fender, http://pastebin.com/559466 |
| 13:39.08 | *** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 13:41.06 | docelm0 | fugitivo, huh? |
| 13:41.45 | fugitivo | docelm0: i was using a console of 10 lines for irc, now i use the hole screen |
| 13:42.09 | fugitivo | whole |
| 13:43.02 | trixter | it amazes me how many people dont clean their caller id before placing a call to a random provider |
| 13:43.24 | crusher | why would you? |
| 13:43.43 | crusher | the provider overrides it anyhow |
| 13:43.48 | trixter | not all |
| 13:43.55 | crusher | my providers do |
| 13:44.09 | trixter | I only do if someone doesnt send me something valid, many people are sending their extension numbers which doesnt quite work |
| 13:44.32 | mutilator | what;re the old t1 connector dealys calleD? |
| 13:44.34 | Beirdo | heh |
| 13:44.36 | trixter | but I also only provide free access to tollfrees where people can send their own caller id which I convert to ani |
| 13:44.51 | Beirdo | made me laugh the first time I saw callerid of "1006" on my cellphone |
| 13:44.59 | trixter | mutilator: a rj48x jack? or something else? |
| 13:45.03 | crusher | hehe |
| 13:45.14 | mutilator | i thought they were DB something |
| 13:45.20 | crusher | Well, I used to put the external number in callerid, but cleaned it out my extensions.conf yesterday |
| 13:45.24 | Beirdo | "oops, missed a spot overriding my callerid" |
| 13:45.28 | mutilator | it's a serial style thing |
| 13:45.31 | trixter | mutilator: well I am not sure which connector you are talking about specifically |
| 13:45.39 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:45.52 | trixter | like v.35? |
| 13:46.01 | trixter | like what a csu/dsu would plug into? |
| 13:46.11 | trixter | those didnt go away they just became integrated |
| 13:46.22 | mutilator | nop |
| 13:46.37 | trixter | I am not quite sure what you are talking about then |
| 13:46.59 | mutilator | db15 |
| 13:47.24 | trixter | that is just a 15 pin connector ... didnt realize that was a t1 thing specifically |
| 13:47.36 | trixter | nor would I have ever guessed that it was t1 specifically.. |
| 13:47.38 | crusher | My inbound voip provider actually sets callerid "asterisk"<asterisk> when an anonymous caller rings us. |
| 13:47.39 | mutilator | may not be |
| 13:47.49 | trixter | crusher: asterisk does that by default |
| 13:47.53 | crusher | hehe |
| 13:47.59 | mutilator | i just know i've only dealt with it on t's |
| 13:48.08 | crusher | okay, obviously they use asterisk behind their sipexpress |
| 13:48.29 | trixter | are you sure its not your box that is doing that by default? |
| 13:48.42 | trixter | mine does only because I am too lazy to change it |
| 13:49.08 | crusher | where would it be set then? while handling the extension it's already set to that |
| 13:49.19 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
| 13:49.32 | crusher | I shall test it this evening by connecting a sip phone directly |
| 13:49.51 | mutilator | found what i was lookin or |
| 13:49.53 | mutilator | for |
| 13:49.54 | mutilator | http://www.trianglecables.com/db15f-rj45.html |
| 13:49.56 | trixter | not that it matters that much, but it would let you know ... |
| 13:50.16 | crusher | alright |
| 13:50.17 | mutilator | er i need male |
| 13:50.40 | mutilator | there it is |
| 13:50.52 | mutilator | eh sry typin outloud |
| 13:50.54 | fjean | Hello ! Anybody uses ASTSS and understands the "adjusted time" formula ? adjtime = (((answertime - inc) + inc - 1) / inc) * inc |
| 13:50.58 | fugitivo | ~seen coppice |
| 13:51.07 | jbot | coppice <n=chatzill@199.193.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 5d 5h 19m 34s ago, saying: 'but it is the one people complain about the most :-)'. |
| 13:51.07 | mutilator | i think i had too much coffee |
| 13:51.27 | fjean | sorry not ASTSS but ASTCC |
| 13:52.43 | fjean | Should I expect multiples of 6 as a result, if not, why ? :- ) |
| 13:53.20 | fjean | here is the code: eval { $adjtime = int((($answeredtime - $numdata->{includedseconds}) + $increment - 1) / $increment) * $increment; |
| 13:53.31 | [TK]D-Fender | [swb] : Ok... I'm stumped.... |
| 13:55.46 | trixter | [TK]D-Fender: did whomever stump you at least buy you dinner first? |
| 13:56.00 | fugitivo | [TK]D-Fender: are you using rxfax? |
| 13:56.09 | X-Rob | fjean, isn't that saying answertime = answertime? |
| 13:58.44 | X-Rob | No, it's saying $adjtime = $answerdtime - $numdata->{inc..} + increment - 1 |
| 14:00.22 | fjean | in my case: adjtime = int(((50 - 30) + 6 - 1) / 6 * 6; |
| 14:01.01 | fjean | 50=talk time and 30=included seconds |
| 14:03.03 | fjean | adjtime = 25 ... |
| 14:03.25 | X-Rob | No, it's not / 6 * 6 |
| 14:03.44 | X-Rob | It's ) / 6 ) * 6) |
| 14:03.57 | X-Rob | if it was / 6 * 6, it would actually be / 36 |
| 14:04.22 | fjean | mmmmm |
| 14:04.32 | X-Rob | but being that it's / 6 ) * 6), it's useless. |
| 14:04.34 | X-Rob | you can take it off |
| 14:04.48 | X-Rob | so you're doing 50-30 + 6 - 1 |
| 14:05.04 | X-Rob | eg, 25. |
| 14:05.32 | fjean | mmmm,cool |
| 14:06.30 | fjean | let me see... |
| 14:06.55 | X-Rob | However, if they're casting int's in strange places, that's not quite right. |
| 14:07.17 | *** join/#asterisk PakiPenguin (n=bah@linuxpakistan/admin/pakipenguin) |
| 14:07.19 | PakiPenguin | hello everyone |
| 14:07.53 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 14:08.08 | X-Rob | if it's (int (50-30+6-1)/6)*6, that's not the same, as the 25/6 will be rounded to 4, and then multipled back to 24, not 25. |
| 14:08.08 | PakiPenguin | i'd like to ask if its better to get the Digium TDM2422B PCI Card with echo cancelation or without echo cancelation ? I'd use this card in a dual xeon machine , that would do nothing else then handling these calls |
| 14:08.15 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:08.31 | mikefoo | hey guys in general, if I have a working system, would there be any significant reason I would want to upgrade at any point in time? |
| 14:08.38 | X-Rob | PakiPenguin, get Echo Cancellation hardware if you can afford it. Always. |
| 14:08.40 | mikefoo | other the security issues and such. |
| 14:08.46 | mikefoo | s/the/then |
| 14:09.07 | skeffling | hi,is there anywork being done to have a xfersound feature for call transfers on sip channels? |
| 14:09.19 | crusher | mikefoo: no, or maybe new features you'd like |
| 14:09.33 | iCEBrkr | yo yo yo |
| 14:09.46 | [TK]D-Fender | fugitivo : Yes I run SpanDSP |
| 14:10.22 | PakiPenguin | X-Rob, if i cannot , can the cpu make up for echo cancelation? |
| 14:11.10 | [TK]D-Fender | PakiPenguin : As far as EC on the TDM2400 goes, if you follow the mailing lists it seems that it doesn't seem to offer much (yet) and is moderately buggy.... not sure if you're better off just stay in SW. # of channels may have an impact... |
| 14:11.30 | zoa | hey ho! |
| 14:11.31 | X-Rob | PakiPenguin, not anywhere near as well. |
| 14:11.34 | fugitivo | [TK]D-Fender: what version of spandsp and asterisk? |
| 14:11.39 | zoa | morning, you fool! its almost evening!!!! |
| 14:11.40 | zoa | :p |
| 14:12.18 | PakiPenguin | X-Rob, would i be better off using the TDM 4X series? |
| 14:12.19 | [TK]D-Fender | fugitivo : 1.0.9 CVS and not sure of SpanDSP |
| 14:12.20 | asteriskmonkey | its 6am always somewhere |
| 14:12.42 | [TK]D-Fender | I'm upgrading EVERYTHING tonight. PRI card F/W, drivers, *, Zaptel, GUI, the works.... |
| 14:13.23 | asteriskmonkey | [TK]D-Fender : beware youll have to recomile your spandsp for 1.2 the 1.0.9 one wont work |
| 14:13.30 | [TK]D-Fender | PakiPenguin : Who many FXO/FXS ports do you need? |
| 14:13.37 | fugitivo | [TK]D-Fender: ok, because i upgraded spandsp and i'm getting errors with rxfax now |
| 14:13.53 | [TK]D-Fender | asteriskmonkey : Well aware, and my vendor is the one managing the source revisions :) I'm just doing the hardware half/ |
| 14:14.11 | asteriskmonkey | nice |
| 14:14.18 | PakiPenguin | [TK]D-Fender, 6FXO and 6 FXS |
| 14:15.47 | [TK]D-Fender | PakiPenguin : Well TDM400 is clearly NOT the way for you to go. Your options are TDM2400, A200, or T1 + Channel-bank. |
| 14:16.12 | [TK]D-Fender | Why FXS on the card? |
| 14:16.28 | PakiPenguin | hmms one card solution :) |
| 14:16.33 | fugitivo | [TK]D-Fender: why not a sip gateway for fxs? |
| 14:17.03 | fugitivo | an audiocodes with 8fxs |
| 14:17.19 | PakiPenguin | yes , i can think about that too |
| 14:17.49 | X-Rob | PakiPenguin, that's hugely expensive. Do what you can afford to do. |
| 14:18.15 | *** join/#asterisk bas123 (n=bas@www3.datarack.nl) |
| 14:18.18 | bas123 | hellp |
| 14:18.23 | bas123 | hello, that is |
| 14:18.26 | bas123 | :) |
| 14:18.27 | [TK]D-Fender | PakiPenguin : For FXS you could save a bundle and use ATA's. its only 3... |
| 14:18.30 | fugitivo | i don't think the audiocodes is more expensive than the same solution with the tdm2400 |
| 14:18.32 | *** join/#asterisk saftsack (n=saftsack@p54A7DF79.dip.t-dialin.net) |
| 14:18.46 | bas123 | anybody using polycom Soundpoint 600 phones? |
| 14:18.50 | PakiPenguin | yes , i was talking about the atas |
| 14:19.02 | [TK]D-Fender | PakiPenguin : the A200 solution would cost less than the TDM2400 for your needs while still being expanable. |
| 14:19.06 | [TK]D-Fender | bas123 : Yes |
| 14:19.12 | fugitivo | the problem with the atas is that you'll have 3 more devices to admin |
| 14:19.17 | bas123 | @TKD-fender |
| 14:19.25 | *** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129) |
| 14:19.28 | bas123 | have you ever had problems update the bootrom? |
| 14:19.50 | [TK]D-Fender | fugitivo : True, but a fraction of the cost but that balances as well against the increased functionality they offer over TDM withing *. Simplifies setup and reliability |
| 14:19.57 | [TK]D-Fender | bas123 : Never a problem with them. |
| 14:19.59 | bas123 | I am doing a new install at a customer site at the moment, and half of them refuse to load the new bootrom |
| 14:20.20 | [TK]D-Fender | bas123: You should leave the BR at 2.6.1 if I were you.... |
| 14:20.26 | bas123 | "error updating bootROM" |
| 14:20.28 | [TK]D-Fender | just upgrade SIP. |
| 14:20.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.13 | PakiPenguin | hmms yeah |
| 14:21.32 | bas123 | [TK]D-Fender: we have multiple installs and like to keep them on the same version. But your answer makes sense |
| 14:21.44 | PakiPenguin | brb a sec |
| 14:21.51 | [TK]D-Fender | bas123 : How do you provision them currently? |
| 14:22.22 | bas123 | [TK]D-Fender: with scopserv |
| 14:22.36 | asteriskmonkey | remember you always get what you pay for though ata's vs proper fxs gateways can be night and day in terms of quailty sometimes |
| 14:23.02 | asteriskmonkey | like compare the iaxy to a audiocodes fxs gateway.. |
| 14:23.28 | [TK]D-Fender | bas123 : I meant what protocol (ftp/tftp (what I'm betting on for your case), http) |
| 14:23.35 | mikefoo | we have a few audiocode fxs/fxo gateways here |
| 14:23.43 | bas123 | [TK]D-Fender: sorry, tftp it is |
| 14:24.44 | [TK]D-Fender | bas123 : Leave them at 2.6.1 if you can and just up the SIP revision then. The new BR is not needed for your scenario. I'm running 1.5.3. here personally and am quite happy although I have every version handy from 1.4.3 up. |
| 14:25.28 | [TK]D-Fender | including the BR. I just got 1.6.5 yesterday and am following the changelogs on them regularly. As soon as they offer something I really need or fix a bug I've encountered here, THEN I'll switch |
| 14:25.29 | bas123 | [TK]D-Fender: The phones are loading the new SIP version now, looks good. |
| 14:25.56 | [TK]D-Fender | I'll have to take a look and see how ScopServ's provisioning has come along.... |
| 14:26.26 | bas123 | [TK]D-Fender: I've found the guys at scopserv very help helpful, but then again, we're reselling it too :) |
| 14:26.45 | [TK]D-Fender | bas123 ..... I think we may have scrossed paths before.... Whats your real first name? |
| 14:27.06 | X-Rob | or sab |
| 14:28.02 | [TK]D-Fender | <- Andrew Oulton (Montreal, QC). I've been on a conference call with someone on your side before with Denis Trepanier over echo/fax issues..... |
| 14:28.20 | [TK]D-Fender | About 2/3 months ago |
| 14:28.41 | iCEBrkr | [TK]D-Fender: you're a hoser??!!? |
| 14:29.08 | fourcheeze | should a user / peer specify how it wants dtmf? |
| 14:29.12 | *** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net) |
| 14:29.12 | ManxPower | iCEBrkr, perhaps you mean Canuck. Hosers are from Indianan |
| 14:29.16 | fourcheeze | I mean can there be any negotiaition |
| 14:29.24 | fourcheeze | or is it just up to everyone to be using the same? |
| 14:29.32 | *** join/#asterisk bas1234 (n=bas@www3.datarack.nl) |
| 14:29.33 | iCEBrkr | ManxPower: Apparently you haven't watched Strange Brew, eh? |
| 14:29.51 | bas1234 | [TK]D-Fender: sorry, my collegue tightened the firewall while troubleshooting VPN issues |
| 14:29.55 | ManxPower | fourcheeze, you didn't get an answer to your question on asterisk-dev about rfc2833 DTMF and LEvel3? |
| 14:29.56 | bas1234 | the bastard :) |
| 14:29.57 | ManxPower | iCEBrkr, nope. |
| 14:30.06 | NewSole | stupid question anyone have experinance with sipra and asterisk |
| 14:30.10 | iCEBrkr | ManxPower: It's a classic 80's movie |
| 14:30.27 | fourcheeze | ManxPower: we didn't go with L3 :-) |
| 14:30.38 | [TK]D-Fender | iCEBrkr : Hoser? |
| 14:30.39 | NewSole | we we call the sipra device it sends back busy |
| 14:30.57 | ManxPower | NewSole, SIPa work fine. |
| 14:31.05 | *** join/#asterisk GMsoft (n=gmsoft@gentoo/developer/gmsoft) |
| 14:31.07 | iCEBrkr | [TK]D-Fender: Yeeeesh! You haven't seen Strange Brew, either?! |
| 14:31.13 | iCEBrkr | Young'ns. |
| 14:31.16 | GMsoft | hi everybody |
| 14:31.31 | NewSole | why am I getting busy back |
| 14:31.52 | [TK]D-Fender | iCEBrkr : I'm older than you :) |
| 14:31.54 | iCEBrkr | NewSole: You can set verbose 9 at the CLI and watch the console |
| 14:32.05 | iCEBrkr | [TK]D-Fender: Maybe :P |
| 14:32.13 | NewSole | kind of hard |
| 14:32.14 | fourcheeze | ManxPower: althouhg L3 inthe UK claim to use rfc2833 |
| 14:32.20 | [TK]D-Fender | 30 <- |
| 14:32.22 | iCEBrkr | NewSole: Why's taht? |
| 14:32.25 | NewSole | we have over 100 calls going at once |
| 14:32.26 | iCEBrkr | [TK]D-Fender: 31 yo |
| 14:32.27 | [TK]D-Fender | iCEBrkr : Why start now? |
| 14:32.30 | [TK]D-Fender | LIES! |
| 14:32.31 | iCEBrkr | lol |
| 14:32.36 | iCEBrkr | [TK]D-Fender: 1974 |
| 14:32.42 | GMsoft | is there a documentation regarding IAX bandwitdh calculation ? |
| 14:32.50 | iCEBrkr | [TK]D-Fender: and I'll be 32 this sept. |
| 14:32.54 | [TK]D-Fender | ok, I'll bet you've got months at best on me... I've got MILAGE! |
| 14:33.01 | iCEBrkr | lol |
| 14:33.12 | NewSole | I was just wondering if someone has sample settings |
| 14:33.14 | iCEBrkr | [TK]D-Fender: Rode hard and hung up wet? |
| 14:33.28 | [TK]D-Fender | iCEBrkr : Think I saw that movie ;) |
| 14:33.47 | iCEBrkr | haha |
| 14:34.05 | iCEBrkr | NewSole: I'm thinking there's something wrong with your sip.conf |
| 14:34.31 | NewSole | thats what I was wondering... its nat based |
| 14:35.03 | iCEBrkr | I wonder were my good friend sevard is? |
| 14:35.24 | iCEBrkr | >= ) |
| 14:36.32 | NewSole | iCEBrkr... anyone can help with device config |
| 14:36.39 | zoa | im here but i am disguised as zoa |
| 14:36.48 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 14:36.52 | zoa | please give me your credit card number now |
| 14:36.57 | zoa | i forgot to write it down before |
| 14:36.58 | [TK]D-Fender | bas1234 : Yeah, looks like ScopServ has improved their provisioning of the SPIP's lately. How well does it handle multiple line keys/reg? It LOOKS like it'll create a clash if you pick the same ext more than once.. |
| 14:37.02 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 14:37.05 | iCEBrkr | NewSole: I think you're going to have to watch the console |
| 14:37.26 | iCEBrkr | zoa: You need my paypal and ebay account info too? |
| 14:37.38 | iCEBrkr | zoa: Just to verify my info is correct.. |
| 14:37.41 | mikefoo | iCEBrkr: sup sup |
| 14:37.44 | iCEBrkr | mikefoo: yo |
| 14:37.50 | NewSole | <PROTECTED> |
| 14:37.50 | NewSole | <PROTECTED> |
| 14:37.50 | NewSole | <PROTECTED> |
| 14:37.50 | NewSole | <PROTECTED> |
| 14:37.51 | NewSole | <PROTECTED> |
| 14:37.53 | NewSole | <PROTECTED> |
| 14:37.54 | iCEBrkr | NewSole: dude. |
| 14:37.56 | Winkie | I'm trying to set up a queueing system, but i am a little confused, i'd like more than one call to ring in at once, because we have say 30 agents and there are always 6 or 7 calls going on at the time |
| 14:37.57 | iCEBrkr | NewSole: ~pb |
| 14:38.00 | Winkie | any ideas? |
| 14:38.00 | iCEBrkr | ~pb |
| 14:38.01 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:38.45 | NewSole | http://pastebin.com/559554 |
| 14:38.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 14:38.51 | Winkie | also leastrecent doesn't seem to work, it repeatedly times out and restarts the queue, or so it seems, am i doing something stupid? |
| 14:39.23 | Winkie | or would using agents help instead of members? |
| 14:40.35 | [TK]D-Fender | Winkie : pastebin your queues.conf and the extensions.conf context that calls it |
| 14:40.50 | [TK]D-Fender | Winkie : and what was the last thing you changed before it stopped working? |
| 14:40.51 | Winkie | [TK]D-Fender: well at the moment there's virtually nothing there |
| 14:40.55 | Winkie | and it's not stopped working |
| 14:41.00 | [TK]D-Fender | Winkie : Show anyways |
| 14:41.02 | NewSole | iCEBrkr... any idea |
| 14:41.03 | Winkie | no worries |
| 14:41.05 | Winkie | one min |
| 14:41.25 | remiss | <PROTECTED> |
| 14:41.31 | remiss | why is that? |
| 14:41.32 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:41.48 | *** join/#asterisk klerer (n=klerer@ool-44c72037.dyn.optonline.net) |
| 14:42.05 | iCEBrkr | NewSole: 'sip show peers' and see if it's registered |
| 14:42.09 | [TK]D-Fender | ~pb |
| 14:42.11 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:42.12 | NewSole | it is |
| 14:42.40 | ManxPower | "seeding peer" seems to indicate that the peer might not be registered. |
| 14:43.27 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:43.55 | Winkie | [TK]D-Fender: http://pastebin.ca/41941 |
| 14:44.07 | ManxPower | NewSole, well what is the value of DIALSTATUS after the Dial? |
| 14:44.48 | Winkie | [TK]D-Fender: effectively we need queues to push as many calls to phones as possible without waiting for the phones to pick up |
| 14:45.01 | Winkie | as we handle far too many calls to make a single call ringing at a time feasable |
| 14:45.43 | Winkie | this is obviously just a test and a proof of concept but it seems weird that asterisk queues must be only one at a time |
| 14:46.03 | [TK]D-Fender | Winkie : not sure how to increase the number of simultaneous ringing calls.... does it abort out of queue early? |
| 14:46.23 | Winkie | [TK]D-Fender: not as far as i'm aware, it times out, and re-rings the same phone instead of alternating |
| 14:46.31 | Winkie | let me quickly check |
| 14:47.10 | [TK]D-Fender | I think because you are calling the queue with 5 seconds the first time with a timeout of 10 it never gets to 10 and then re-rings the same phone.... |
| 14:47.52 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfljo.dialup.mindspring.com) |
| 14:47.53 | Winkie | no it definately makes it past the first queue |
| 14:48.27 | Winkie | http://pastebin.ca/41942 |
| 14:48.34 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:48.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:50.22 | *** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net) |
| 14:50.48 | paulhuynh | i need little help today can someone help me!!! |
| 14:50.56 | Winkie | just ask |
| 14:51.13 | [TK]D-Fender | Actually it may keep trying to beat down that same guy if he's listed as available... you need to get them to pause or log off.... |
| 14:51.15 | paulhuynh | My asterisk@home iso2.4 is not taking call from from my sip provider |
| 14:51.27 | wunderkin | Winkie, make sure you are using a recent 1.2 release branch and see if it has the autofill option, use it |
| 14:51.53 | Winkie | wunderkin: 1.2.4, autofill in queues.conf and i assume that's for multiple rings? |
| 14:51.57 | Winkie | multiple incoming rings* |
| 14:52.00 | wunderkin | im not sure if it was put into 1.2 or not but if not you may be able to patch it |
| 14:52.21 | Winkie | i can use CVS if neccasary, it's a test system, but obviously stability will be important when it goes live |
| 14:52.48 | wunderkin | that helps fix some of the queue quirks but i dont remember what all |
| 14:52.55 | Winkie | also [TK]D-Fender you reckon it's just a lack of agents? I mean leastrecent does say it will call the 'interface' least recently dialled, which wouldn't be linksys |
| 14:52.57 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 14:53.21 | Winkie | paulhuynh: who is your sip provider / what does sip show registry produce (use pastebin) |
| 14:53.33 | Winkie | wunderkin: i will google my way to finding out, ta |
| 14:53.34 | [TK]D-Fender | Winkie : I believe its not the least recently CALLED, its the least recently ANSWERED/. |
| 14:53.44 | paulhuynh | winkie i use cvcternimation |
| 14:53.53 | [TK]D-Fender | paulhuynh ..... |
| 14:53.53 | NewSole | the DIALSTATUS is CONGESTION |
| 14:53.54 | [TK]D-Fender | ~amp |
| 14:53.56 | jbot | from memory, amp is NOT supported here! people using it should join #amportal |
| 14:53.57 | burton | hello, anyone has working setup with two wifi AP + asterisk + SIP wifi phones ? how about hand over roam ? |
| 14:54.40 | Winkie | [TK]D-Fender: ah, from voip-info it says 'leastrecent: ring interface which was least recently called by this queue' but that may be inaccurate |
| 14:55.04 | [TK]D-Fender | I'd bet on it... |
| 14:55.06 | wunderkin | paulhuynh, oh yeah? have you used them yet, what do you think? i was thinking about using them, they are just a few racks down from me.. i don't have an account yet so i can't really help setting it up yet |
| 14:55.20 | [TK]D-Fender | minor difference that makes all the difference... |
| 14:56.28 | Assid | heya tkd |
| 14:56.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:56.31 | Assid | whats happening |
| 14:56.48 | *** join/#asterisk Teeli (i=Tili@202-133-67-129-dialup.sat.net.pk) |
| 14:56.51 | [TK]D-Fender | Assid : just working on stuff... tonight is my works big * upgrade. |
| 14:56.53 | Winkie | [TK]D-Fender: indeed, also autofill and autopause, which seem to be the two interesting ones were committed in revision 8060 4 weeks ago, but i have no idea how to find out if that was included in 1.2.4 |
| 14:57.05 | Assid | really? what u doing in it? |
| 14:57.15 | *** join/#asterisk littlejohn (n=little@host130-254.pool8263.interbusiness.it) |
| 14:57.17 | [TK]D-Fender | auto-pause? Pauses static members for not answering (like auto-logoff?) |
| 14:57.20 | wunderkin | Winkie, i would just use the latest release branch |
| 14:57.23 | Winkie | [TK]D-Fender: indeed |
| 14:57.32 | paulhuynh | here are my pastebin for the sip reg |
| 14:57.33 | paulhuynh | http://pastebin.ca/41945 |
| 14:57.34 | Assid | didnt get you. how does that help |
| 14:57.35 | Winkie | wunderkin: it looks like i have to, what's asterisk's release schedule like? |
| 14:57.36 | [TK]D-Fender | Winkie : definately good news... I run a call center here as well.. |
| 14:58.09 | Assid | nice... i always wanted to do atleast 1 solution for a call center |
| 14:58.11 | Winkie | paulhuynh: i don't know your provider, do they require sip registration or what? |
| 14:58.28 | Assid | even if its around 20 users... just wanted to play with that |
| 14:58.32 | Winkie | [TK]D-Fender: fun isn't it? we run about 30 agents, 6000 calls a day or so |
| 14:58.38 | wunderkin | Winkie, not sure, it may be on the website, i think the next release is scheduled in the summer sometime |
| 14:58.39 | mikefoo | Can ayone recommened a voip provider I can terminate on that offers concurrent calls? |
| 14:58.40 | [TK]D-Fender | Assid : I'm going from 1.0.9 to 1.2.4 and upgrading PRI f/w drivers, NIC drivers, and more. |
| 14:58.41 | mikefoo | in the US |
| 14:58.42 | Winkie | well i'd say nearer 8000, our phone system is holding us back |
| 14:58.46 | [TK]D-Fender | Winkie : WOW.... |
| 14:59.01 | Winkie | [TK]D-Fender: how many do you handle? |
| 14:59.14 | Winkie | we're looking at setting up 4 locations total with asterisk and interlinking them with leased lines + IAX trunking |
| 14:59.16 | Winkie | which will be fun |
| 14:59.23 | paulhuynh | mikefoo try sixtel.net or cvctermination |
| 14:59.33 | mikefoo | paulhuynh: thanks. |
| 14:59.46 | paulhuynh | winkie this they do |
| 15:00.09 | Winkie | paulhuynh: what does 'sip show registry' tell you? |
| 15:00.27 | paulhuynh | that they are register |
| 15:00.28 | [TK]D-Fender | Winkie : 4 agents, harly more than 3 in Q at any time... |
| 15:00.34 | Assid | is there any perfomance hit when you have sip clients and iax interconnects? |
| 15:00.36 | [TK]D-Fender | (backloged) |
| 15:00.47 | Winkie | [TK]D-Fender: heh, we barely make 3 calls waiting unless it's really busy |
| 15:00.55 | paulhuynh | i contact my provider but they tell me my asterisk box is reject their call that was forward to us |
| 15:01.01 | Winkie | it goes from 1 guy alone on a sunday night to about 13-14 logged on at the moment |
| 15:01.12 | Winkie | paulhuynh: pb your dialplan |
| 15:01.17 | paulhuynh | i can make call to pstn from my asterisk to though their network |
| 15:01.20 | Winkie | or at least, the from-pstn context |
| 15:01.37 | [TK]D-Fender | Winkie : He's running A@H, don't ask that!!! |
| 15:01.46 | Winkie | oh right |
| 15:01.51 | Winkie | i know nothing of this rubbish :) |
| 15:01.57 | wunderkin | paulhuynh, i would look at a sip debug, you probably don't have your dialplan correct |
| 15:01.59 | [TK]D-Fender | paulhuynh : Pastbin the FAILED CALL from CLI only please. |
| 15:02.31 | wunderkin | there also aren't any codecs specified there.. |
| 15:02.41 | paulhuynh | ok can you walk me through the cli command |
| 15:03.06 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfljo.dialup.mindspring.com) |
| 15:03.44 | PakiPenguin | okay i am back |
| 15:04.07 | Winkie | [TK]D-Fender: the interesting thing is getting decent quality voice to canada |
| 15:04.11 | PakiPenguin | [TK]D-Fender, what card do you suggest for incoming pstn ( FXO ) ? 6 lines , i am going to use ata for extensions |
| 15:04.19 | Assid | hey is there any perfomance hit when all your phones use sip.. and iax to interconnect.. or is it the same as sip providers instead of iax since the sip layer is the only one invoked |
| 15:04.20 | Winkie | i can't wait to see how much our provider's going to charge |
| 15:04.47 | Winkie | Assid: SIP and IAX are essentially framing, performance hits occur with transcoding afaik |
| 15:04.55 | Winkie | so as long as you're using the same codecs, you shouldn't have much of a problem |
| 15:05.17 | *** join/#asterisk hugo-v6 (n=alterego@ns1.bundesunixminister.de) |
| 15:05.20 | hugo-v6 | hiho |
| 15:05.24 | Assid | some one told me that there is a hit even in the carrier method (iax/sip) i was like no way.. transcoding would be the only thing |
| 15:05.45 | Assid | but then i started reading about SER.. where the sip layer just interconnects the sip clients |
| 15:05.55 | Assid | so i started thinking maybe there was something to it |
| 15:05.57 | Winkie | there will be *some* performance difference but i'll emphasise it because i don't think we're talking serious differences |
| 15:05.57 | fjean | x-rob: would you know what "included seconds" means ? is it meaning the customer will get this number of seconds free ? |
| 15:06.05 | *** join/#asterisk Cresl1n (n=matt@146.229.178.19) |
| 15:06.25 | Winkie | also if anyone has experience in transcoding from an E1 to ulaw, let me know the stats :) |
| 15:06.46 | Winkie | anyway i'm off for a ciggy, thanks [TK]D-Fender and wunderkin, i'll CVS it later and try |
| 15:06.49 | Assid | E1 to ulaw? |
| 15:07.01 | Assid | Winkie: CVS is dead.. gotta use SVN |
| 15:07.03 | Winkie | yes, as in ISDN PRI > SIP ulaw |
| 15:07.04 | [TK]D-Fender | PakiPenguin : A200 w/ HWEC |
| 15:07.07 | Winkie | well CVS, SVN |
| 15:07.10 | Winkie | they're all TLAs to me |
| 15:07.12 | Winkie | 8) |
| 15:07.13 | Winkie | ciao |
| 15:07.14 | PakiPenguin | cool , i was looking into that |
| 15:07.15 | PakiPenguin | :) |
| 15:07.19 | ManxPower | Winkie, E-1 uses alaw |
| 15:07.21 | PakiPenguin | thanks , i'll be back with more questions hehe |
| 15:07.26 | ManxPower | that's all yu really need to know |
| 15:07.32 | Winkie | ManxPower: oh really, well that will certainly simplify speccing the servers |
| 15:07.37 | Winkie | no dual x2s needed here then |
| 15:08.04 | fjean | hi guys, anybody knows ASTCC and it's associated DB fields well ? |
| 15:08.09 | paulhuynh | this is what i got when do a sip show registry |
| 15:08.09 | ManxPower | alaw is the telco standard for USA/Canada (and maybe a few other countries), alaw is what the rest of the world uses. |
| 15:08.10 | paulhuynh | http://pastebin.ca/41946 |
| 15:08.17 | ManxPower | So you want your SIP clients to use alaw. |
| 15:08.40 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-128-113-60.dsl.tulsok.swbell.net) |
| 15:08.49 | Assid | hrmm.. is alaw any 'clearer' than ulaw? |
| 15:09.01 | ManxPower | Assid, I don't think so, just slightly different |
| 15:09.09 | Assid | in framing? |
| 15:09.17 | Ahrimanes | both uncompressed, ulaw mainly used in the us and alaw in europe |
| 15:09.26 | ManxPower | Assid, I would have to read the codec specs to say for sure. |
| 15:09.27 | Assid | hrmm |
| 15:09.36 | Assid | well.. ulaw works.. i aint touching it |
| 15:09.56 | ManxPower | Assid, you should usually generally just use the codec that your location uses. |
| 15:10.20 | fugitivo | does the t110p work on a pcix slot? |
| 15:10.21 | viperdude | hi has anyone here provided a operator console for a solution? |
| 15:10.29 | Assid | well.. mostlye looking at US anyways.. so really just following that standards |
| 15:10.33 | viperdude | i am looking for hardware not FOP |
| 15:10.58 | Assid | fugitivo: i thought pci-x is for the gfx card only |
| 15:11.55 | Assid | i need to learn more |
| 15:12.09 | wunderkin | fugitivo: yes |
| 15:12.19 | fugitivo | hmm, why would a motherboard server come with 2 slots for gfx cards? :) |
| 15:12.28 | fugitivo | wunderkin: great, thanks |
| 15:12.44 | Assid | pci-express and agp ? |
| 15:12.46 | Assid | lol |
| 15:12.52 | Ahrimanes | pci-x is for all sorts of pci cards |
| 15:13.04 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:13.18 | Assid | man.. that teaches me for not touching hardware in a year |
| 15:13.26 | Assid | except for old ones |
| 15:14.00 | [TK]D-Fender | viperdude : as in to see who's on the phone and to do speed-dials? |
| 15:14.06 | PakiPenguin | a question , would i need anything else , like a rj-11 connector panel or anything with the sangoma a200 with 6 FXO ports? |
| 15:14.52 | [TK]D-Fender | PakiPenguin : A200 uses RJ12 (handset style) and comes with adapter extensions. What you'll need from there if anything depends on what you've got. |
| 15:14.58 | viperdude | [TK]D-Fender: yes, but also to be able to send text to to tell the operator certain info.. eg if a camped call gets returned the extension they were camped on |
| 15:15.03 | [TK]D-Fender | if you're on punchdown you'll need to adapte from that |
| 15:15.04 | NewSole | ok anyone here good with SER |
| 15:15.09 | Assid | brb |
| 15:15.48 | [TK]D-Fender | viperdude : For the BLF stuff, Snom is the only current option (Polycom will be viable hopefully as of * 1.4) as for parking indicators, not sure.... |
| 15:16.00 | viperdude | BLF? |
| 15:16.11 | paulhuynh | is anyone here use sixtel for did? |
| 15:16.15 | PakiPenguin | [TK]D-Fender, i have standard pots lines ( with rj-11s at their ends ) , do i need something else then? |
| 15:16.16 | [TK]D-Fender | Busy Lamp Field : (in use indicator / speed-dial) |
| 15:16.30 | [TK]D-Fender | PakiPenguin : nope, all good out of the box... |
| 15:16.35 | NewSole | ok anyone here good with SER want a job... pvt me |
| 15:16.39 | viperdude | ok that works currently on Snom with * 1.2.1? |
| 15:17.05 | [TK]D-Fender | viperdude : yup, and Snom is cheapr than the rest once you add on the sidecar. |
| 15:17.18 | [TK]D-Fender | And thats a LOT of buttons... |
| 15:17.20 | Assid | snom cheaper than polycom ? |
| 15:17.22 | PakiPenguin | cool |
| 15:17.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:17.51 | viperdude | does anyone know if Cisco 7940 support SIP MESSAGE? |
| 15:17.57 | [TK]D-Fender | Assid : Slightly. but which one I'd choose depends on the application. Typically I'd pick Polycom over everything else EXCEPT high-density receptionist. |
| 15:18.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:18.13 | Assid | i see |
| 15:18.18 | [TK]D-Fender | (best of breed selection) |
| 15:18.31 | Assid | thanks for the insight |
| 15:18.36 | Assid | will keep it in mind |
| 15:18.45 | Assid | any particular model? |
| 15:18.57 | jaiger | [TK]D-Fender, have you looked at the add-on panel for the IP600? |
| 15:19.00 | Ahrimanes | [TK]D-Fender: any integration done for function buttons on polycom? |
| 15:19.01 | [TK]D-Fender | Cisco is a great phone as well, but not for the money, and the higher end stuff supporting the sidecar doesn't do SIP yet. So Polycom will likely beat them out for a while yet... then things may become interesting. |
| 15:19.03 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 15:19.23 | viperdude | ok its just I have a cisco 7940 on my desk |
| 15:19.31 | [TK]D-Fender | Ahrimanes : They're working on the ACD login/out/pause right now. As for function buttons what else did you have in mind? |
| 15:19.44 | [TK]D-Fender | jaiger : I have 2 on our 601 |
| 15:19.46 | Ahrimanes | [TK]D-Fender: ACD ? |
| 15:19.53 | [TK]D-Fender | Ahrimanes : Queues' |
| 15:20.04 | jaiger | [TK]D-Fender, and how does it work? |
| 15:20.09 | Ahrimanes | [TK]D-Fender: ah, nice.. that was most of what i had in mind.. call pickup would be nice too |
| 15:20.29 | Assid | hey tkd how big is this call center you run? |
| 15:20.47 | jaiger | I was thinking of buying one to play with |
| 15:20.53 | [TK]D-Fender | jaiger : It works up until the 7-buddy watch bug.... |
| 15:21.14 | Ahrimanes | [TK]D-Fender: know how far along the ACD stuff is with the polycoms? |
| 15:21.23 | jaiger | [TK]D-Fender, so what would you recommend as an alternative? |
| 15:21.28 | paulhuynh | may i ask ACD? |
| 15:21.40 | [TK]D-Fender | Ahrimanes : You can program that as a speed-dial on a line-key, as the Messages button or on one of the 2 other "loose" buttons on the 501 |
| 15:21.57 | [TK]D-Fender | Ahrimanes : so far it looks like this summer (* 1.4) |
| 15:22.17 | Ahrimanes | [TK]D-Fender: ok |
| 15:22.23 | [TK]D-Fender | Assid : 4 agents |
| 15:22.53 | nextime | anyone with chan_ooh323 with latest svn trunk? |
| 15:23.16 | [TK]D-Fender | pickup can be done as I just wrote above, OR once * supports SIP-B BLA draft, shared line appearances will make it look seamless. |
| 15:24.35 | Ahrimanes | [TK]D-Fender: yeah i've done it with speed dials on the snom190 function buttons.. but would like to leave a button lit if the queue is active or something like that |
| 15:25.32 | NewSole | ok anyone here good with SER and want a job... pvt me |
| 15:25.39 | Winkie | ah man it's goddamn freezing outside, anyway an additional question, what's the easiest way to initiate a call on a gxp2000 from the manager interface? is it quite simple? |
| 15:26.03 | viperdude | Winkie: "Action: originate" |
| 15:26.22 | freat | [TK]D-Fender: you know if Polycom has any plans on encryption support? |
| 15:26.22 | [TK]D-Fender | Ahrimanes : Then use BRISTUFF and devstate with a script that periodically checks the queue to toggle the light. |
| 15:26.35 | Winkie | viperdude: i mean an outbound call to a zaptel channel but originating from a free line on a gxp2000 |
| 15:26.43 | Winkie | or perhaps that's what you meant too :) |
| 15:26.44 | freat | we've got a bunch of ip500s... in health care... HIPAA... |
| 15:26.51 | [TK]D-Fender | freat : Likely... it supports HTTPS and SFTP for provisioning so SRTP can't be too far behind. |
| 15:27.00 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-37.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:27.00 | paulhuynh | i'm on asterisk 1.2.1 |
| 15:27.15 | paulhuynh | could that be a problem for my did? |
| 15:27.16 | viperdude | you mean a grandstream GXP-2000 |
| 15:27.30 | Winkie | i do viperdude |
| 15:27.44 | Assid | hey anyone know how much voicepulse charges for adding more chanlimits for incoming? |
| 15:27.54 | Winkie | also i'll brb, i have to set up a windows share, laffo |
| 15:28.13 | viperdude | so yes you do a 'Action: originate' on the manager ..see wiki for details, we use this for our click to dial on our intranet |
| 15:28.27 | paulhuynh | anyone? |
| 15:28.35 | paulhuynh | help please!!! |
| 15:28.37 | paulhuynh | i'm on asterisk 1.2.1 |
| 15:28.39 | paulhuynh | could that be a problem for my did? |
| 15:28.53 | viperdude | whats the problem paul? |
| 15:29.14 | Ahrimanes | [TK]D-Fender: i tried devstate, but was never able to turn off the lights.. |
| 15:29.28 | [TK]D-Fender | only ON? |
| 15:29.47 | Ahrimanes | yep |
| 15:29.50 | paulhuynh | DID from two of my carrier is not making it to asterisk |
| 15:29.53 | paulhuynh | box |
| 15:30.09 | wunderkin | paulhuynh, you need to do a sip debug |
| 15:30.11 | paulhuynh | carrier said that my asterisk is drop the call from them |
| 15:30.17 | [TK]D-Fender | paulhuynh : is your * behind a NAT? If so did you configure it accordingly and forward the needed ports? You also never pastebin'd that failed call like requested. |
| 15:30.21 | wunderkin | i always make them a peer and do sip debug peer blah |
| 15:30.33 | viperdude | paulhuynh: sip debug is your friend |
| 15:30.46 | paulhuynh | ok |
| 15:30.58 | paulhuynh | i got it but it flashing through alot of stuff |
| 15:31.13 | paulhuynh | how can i just show sip debug for the incoming call |
| 15:31.30 | Ahrimanes | [TK]D-Fender: have you had devstate working turning leds on/off? |
| 15:31.41 | NewSole | ok anyone here good with SER and want a job... pvt me (auto timer) |
| 15:31.43 | [TK]D-Fender | Ahrimanes : Haven't tried yet... soon... |
| 15:31.52 | viperdude | sip debug peer <exten> |
| 15:32.52 | Ahrimanes | [TK]D-Fender: please do let me know if it succeeds |
| 15:32.58 | Winkie | viperdude: yeah that seems to be what i need, the question is if i originate a call, how does it interact with the gxp, does it just appear as a second line, or does it ring on it? and how would i transfer one line to another via the manager interface? |
| 15:33.24 | Winkie | also i need to turn LEDs on and off on the gxps too :( |
| 15:33.32 | viperdude | i have written a win32 app that shows the status of the exten's, busy, ringing, DND etc, sounds like that is a good start |
| 15:33.44 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:34.11 | viperdude | Winkie: if you originate then the gxp will ring... when the handset is picked up then it dials out the number required |
| 15:34.29 | Winkie | viperdude: excellent, and i assume if they're already on a call it will ring on a second line? |
| 15:34.36 | viperdude | yes |
| 15:34.37 | Winkie | i'll test it as soon as i have a chance but i'm swamped with other crap at the moment |
| 15:34.48 | Winkie | ok, and is there a way to easily transfer the two without using the actual phone? |
| 15:34.49 | paulhuynh | i got it |
| 15:34.51 | Winkie | (long story) |
| 15:34.55 | paulhuynh | here is my pastebin |
| 15:35.02 | viperdude | if you want to transfer a call via the manager then use "Action: Redirect" again see the wiki |
| 15:35.03 | paulhuynh | please let me know what i can do to fix it |
| 15:35.14 | Winkie | viperdude: the asterisk wiki or a channel specific one? |
| 15:35.14 | nextime | it seem that ooh323c don't compile with latest * svn trunk |
| 15:35.16 | nextime | anyone can confirm this issue? |
| 15:35.24 | paulhuynh | http://pastebin.ca/41951 |
| 15:35.29 | viperdude | voip-info.org/wiki |
| 15:35.46 | Winkie | ah yes, i'm already checking out Originate on there :) |
| 15:36.18 | viperdude | Winkie: the manager API allows you to do some nifty stuff |
| 15:36.31 | Winkie | viperdude: i just use it for a crappy call bot at the moment on our other * setup :) |
| 15:36.40 | Winkie | 15:37.43 <@confbot> Conference meatwhore: Total connections 1 |
| 15:36.41 | NewSole | ok anyone here good with SER and want a job... pvt me (auto timer) |
| 15:36.47 | remiss | SIP/2.0 407 Proxy Authentication Required <-- umpf... what does this mean? |
| 15:36.50 | Winkie | what the hell is SER anyway? |
| 15:36.54 | viperdude | lol |
| 15:37.04 | areski | nestar, I am good but dont want a job |
| 15:37.05 | paulhuynh | SER is sip express router |
| 15:37.05 | Ahrimanes | remiss: it's rather self-explanatory? |
| 15:37.06 | austinnichols101 | SER = sip express router |
| 15:37.09 | Winkie | ah of course |
| 15:37.12 | Ahrimanes | areski: you suck ;) |
| 15:37.16 | areski | hahah |
| 15:37.19 | areski | sorry |
| 15:37.29 | Ahrimanes | areski: go do some php :) |
| 15:37.34 | remiss | Ahrimanes: yes.. i was just wondering what settings control proxy-authentication in sip.conf :-/ |
| 15:37.49 | areski | Ahrimanes, oki.... go back to my work |
| 15:38.01 | Ahrimanes | areski: hehe.. anything big going on these days? |
| 15:38.04 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 15:38.05 | kippi | hi |
| 15:38.26 | kippi | what is the digium IP to analog box? |
| 15:38.28 | areski | Ahrimanes, lot of progress man :) |
| 15:38.38 | klerer | Hi, I’m having an experience with moving my iaxpeeers to mysql (Realtime) – the performance seems to go to hell and I get congestion. Has anyone seen similar? |
| 15:38.51 | areski | Ahrimanes, I guess, next week will be start of a new age |
| 15:39.04 | Ahrimanes | areski: how? hehe |
| 15:39.53 | areski | Ahrimanes, I had speed-dial features yesterday :) funny stuff for the user |
| 15:39.58 | paulhuynh | did anyone see my problem? |
| 15:39.58 | *** part/#asterisk fjean (n=fjean@201.29.122.10) |
| 15:40.06 | Ahrimanes | areski: ah nice :) |
| 15:41.06 | wasim | kippi: digium make pci cards for analog FXO/FXS and digital e1 interfaces which when plugged into an * box work wonders for your phone bill |
| 15:41.38 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:41.41 | NewSole | ok anyone here good with SER and want a job... pvt me (auto timer) |
| 15:41.53 | paulhuynh | newsole |
| 15:41.54 | Winkie | if you have an E1 interface and are worried about your phone bill there's a problem :) |
| 15:41.59 | [TK]D-Fender | NewSole : Turn that stupid timer off and put your request on the Wiki |
| 15:42.00 | paulhuynh | contact this people |
| 15:42.04 | paulhuynh | mike@idv.net |
| 15:42.11 | paulhuynh | he is expert on SER |
| 15:42.17 | wasim | e1s generate big bills :( |
| 15:42.26 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:42.40 | remiss | SIP/2.0 404 Not Found |
| 15:42.47 | remiss | alright.. that one i don't understand.. |
| 15:43.00 | Winkie | remiss: you dialling a client? |
| 15:43.03 | paulhuynh | so no taker on my problem |
| 15:43.13 | remiss | Winkie: incoming call from my sip-provider |
| 15:43.24 | Winkie | remiss: sounds like you need a specific extension number? |
| 15:43.34 | remiss | oh... |
| 15:43.41 | Winkie | sipgate requires it, last i used anyway |
| 15:43.51 | remiss | and how do i set that? |
| 15:43.52 | _Sam-- | anyone have any suggestions what to tweak for working on echo in meetme using digium card? |
| 15:44.02 | _Sam-- | its all SIP clients, no PSTN |
| 15:44.05 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 15:44.17 | remiss | i set context=my-number because of that, but that obviously isn't it then... |
| 15:44.29 | Winkie | remiss: no, you'll need an incoming context of some sort |
| 15:44.41 | Winkie | then either a s exten, or an exten with the sip number your provider assigned to you |
| 15:44.50 | Winkie | i really can't help because i have barely a clue how to get sipgate working |
| 15:45.34 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 15:45.52 | viperdude | Winkie: if its sipgate in the UK THEY dont have a clue either, my account is forever timing out on registration |
| 15:46.38 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 15:46.41 | NewSole | ok anyone here good with SER and want a job... pvt me (auto timer) |
| 15:46.58 | Winkie | viperdude: haha oh dear yes, they're hardly an 'enterprise' provider |
| 15:47.23 | Winkie | although i need to find a canadian company that can handle ~3000 calls a day to ~400 numbers |
| 15:47.23 | viperdude | yeah won't even let you have two sip accounts on the same user account |
| 15:47.28 | Winkie | incoming, that is |
| 15:48.25 | remiss | w00t :D |
| 15:48.33 | remiss | put up an extension |
| 15:48.40 | remiss | with my number... |
| 15:48.49 | Winkie | that working now? |
| 15:48.53 | remiss | yeah |
| 15:48.55 | Winkie | excellent |
| 15:49.17 | remiss | but why won't it work with just 'context=number' and 's' as extension? |
| 15:50.16 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 15:51.12 | Winkie | remiss: i'll paraphrase into asterisk |
| 15:51.16 | Winkie | they don't just dial SIP/remiss |
| 15:51.21 | Winkie | they dial SIP/whatever@remiss |
| 15:51.41 | NewSole | ok anyone here good with SER and want a job... pvt me (auto timer) |
| 15:52.00 | Winkie | man that auto timer needs to be auto slower >:( |
| 15:52.03 | remiss | Winkie: oh. ok |
| 15:52.10 | austinnichols101 | somebody kick newsole |
| 15:53.26 | paulhuynh | anyone plase help me and take a look @ my sip debug? |
| 15:53.34 | Abydos313 | he must not pay well since no one is biting on his job offer..heh |
| 15:55.49 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 15:56.22 | NewSole | acauly no one has msg me.... just need a ser server setup.... for 250$ |
| 15:56.30 | iDunno | a real unixgeek wouldn't have that nick. |
| 15:56.30 | Winkie | remiss: if that makes sense :) |
| 15:56.39 | ManxPower | Maybe it's just that most of believe that SER is frequently the problem, not the answer. |
| 15:56.44 | Winkie | he's obviously not a real unixgeek as he doesn't have rdns :) |
| 15:56.53 | iDunno | that's also true ;) |
| 15:56.57 | Winkie | says me |
| 15:57.05 | Winkie | 15:58.18 [freenode] -!- Winkie [n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com] |
| 15:57.07 | Winkie | lol |
| 15:57.10 | remiss | Winkie: it does.. but i don't feel like it somehow :p |
| 15:57.16 | iDunno | he's also using OS X, so, well ;) |
| 15:57.23 | Winkie | OSUX |
| 15:57.33 | NewSole | ManxPower... what we need is a proxy for faxes thus we are looking at ser |
| 15:57.37 | Nugget | OS X > * |
| 15:57.53 | Nugget | (* in the "all" sense, not "asterisk" sense. I hate that ambiguity in here :) |
| 15:58.06 | remiss | i don't like os x anymore.... |
| 15:58.11 | iDunno | Nugget: no, OS X has a broken BSD userspace - it is *not* greater than anything, ever. it's broken. inherently. |
| 15:58.19 | Winkie | apparantly you also don't understand the meaning of > and < :) |
| 15:58.20 | Nugget | broken how? |
| 15:58.23 | ManxPower | I use "Asterisk" to mean "Asterisk", and * to mean wildcard |
| 15:58.27 | Winkie | iDunno: well it's better than windows, and os/2 :) |
| 15:58.36 | Winkie | i use * when i'm goddamn lazy |
| 15:58.42 | iDunno | Winkie: oh, I don't know, OS/2 has it's good points ;) |
| 15:58.48 | caio1982 | ManxPower: good one |
| 15:59.20 | Winkie | iDunno: it has a funky name |
| 15:59.26 | Winkie | i'm struggling to think of much else |
| 16:01.13 | Nugget | I loved OS/2. I think it's a shame it didn't stay competetive. |
| 16:01.26 | Winkie | viperdude: you mentioned redirect, it's documentation is a little short, what the hell is 'extrachannel' all about? |
| 16:01.28 | Abydos313 | still in service for some phones Nugget |
| 16:01.35 | Nugget | ATMs too, I'm sure. |
| 16:01.37 | ManxPower | http://store.voxilla.ca/product.php?productid=16159&cat=0&page=1 I didn't know Polycom Soundpoint supported iLBC. |
| 16:01.39 | Abydos313 | yeah |
| 16:01.44 | Winkie | yep, ATMs historically run on os/2, but have started being windows |
| 16:01.48 | Winkie | which is fun when they bluescreen |
| 16:02.00 | Abydos313 | os/2 is rock solid |
| 16:02.20 | Winkie | yeah but it's easy to be rock solid when it has no functionality :) |
| 16:02.29 | Abydos313 | true |
| 16:02.33 | austinnichols101 | did anyone happen to see the Httpanties on thinkgeek. What a riot! http://www.thinkgeek.com/tshirts/ladies/6792/ |
| 16:02.37 | iDunno | we were using OS/2 v1 (which was still MS) back in the good ol' days, it was remote booting win 3.1 workstations! |
| 16:02.50 | Abydos313 | really? wow |
| 16:03.19 | iDunno | :) |
| 16:03.21 | *** join/#asterisk lorinc (n=ang@caracas-2964.adsl.interware.hu) |
| 16:05.47 | *** join/#asterisk tomtom_ (n=tom@83.217.70.163) |
| 16:05.51 | tomtom_ | joe |
| 16:05.58 | *** join/#asterisk salviadud (n=ralfalfa@201.137.161.198) |
| 16:06.12 | salviadud | good mornin' |
| 16:06.13 | Winkie | iDunno: ouch, luckily i'm not that old |
| 16:06.26 | iDunno | Winkie: dude - I'm only 24 ;) |
| 16:06.32 | tronix | heh I remember booting Warp (OS/2 3.0) on a roomie's 486/33 w/4MB... it swapped like hell, was unusable, and had 5 min long boots |
| 16:06.36 | iDunno | this was while I was still at school *grin* :) |
| 16:06.50 | tronix | Warp on the other roomie's 486dx2/66 w/8 MB..booted in 40sec and was usable. |
| 16:07.06 | Winkie | iDunno: i'm 21, and the nearest i've dealt with os/2 is some legacy stuff at my old place |
| 16:07.08 | tronix | pretty nice OS, protected memory and stuff (back when Windows didn't have it, pre-NT) |
| 16:07.20 | tronix | too bad it didn't take off. |
| 16:07.23 | Winkie | a nice text novell console too |
| 16:07.35 | wunderkin | paulhuynh, the call is being sent to the s exten, use that |
| 16:08.10 | tronix | Winkie: heh... I very vaguely remember Windows 1.0... text mode. 2.0 wasn't much better. 3.0 was a big day, nice GUI and stuff. |
| 16:08.22 | Winkie | tronix: i found some windows 1 beta disks at my uncle's farm once |
| 16:08.26 | iDunno | Novell has some "interesting" ideas, they were way ahead of their time - unfortunately it sucked management wise ;) |
| 16:08.26 | Winkie | now that was an interesting experience |
| 16:08.57 | iDunno | and now everyone's going on about how great Active Directory is... ffs, Novell had that *years* ago ;) |
| 16:09.04 | tronix | indeed. |
| 16:09.14 | Winkie | active directory more like ldap am i rite? |
| 16:09.45 | iDunno | yeah - it is infact ldap... |
| 16:09.53 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 16:09.54 | iDunno | but with a slightly broken implementation. |
| 16:10.00 | Winkie | indeed |
| 16:10.12 | Winkie | ldap is deprecated anyway :) |
| 16:10.17 | *** join/#asterisk enemy^x (n=eqwrweqr@morpheus.dataguard.no) |
| 16:10.39 | iDunno | erm - it's not ;) |
| 16:10.40 | eKo1 | quick ?: as I understand it, every call requires two channels but when I enter 'show channels', I see an odd number of channels. Why is this? |
| 16:10.52 | iDunno | (we're using LDAP at work - and network booting debian machines *grin*) |
| 16:11.01 | Winkie | eKo1: it depends, not every call requires 2 channels |
| 16:11.02 | iDunno | NIS should be deprecated, though. |
| 16:11.48 | viperdude | eKo1: every BRIDGED call requires 2 channels |
| 16:11.48 | Winkie | iDunno: PXE? |
| 16:11.48 | iCEBrkr | wunderkin: I still have OS/2 Warp. It kicked ass. :) |
| 16:11.48 | iDunno | Winkie: yup :) |
| 16:11.49 | eKo1 | viperdude: I see... |
| 16:11.49 | iDunno | Winkie: and root on NFS |
| 16:11.49 | eKo1 | hmm...all these calls should be bridged |
| 16:11.57 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 16:12.48 | iCEBrkr | I've been wanting to build an OS/2 machine again. Make it mirror all the functionality of my linux box |
| 16:14.10 | wunderkin | i wish that os/2 succeeded, need another option to windows (for lusers) |
| 16:14.18 | iCEBrkr | wunderkin: Yea, me too. |
| 16:14.22 | hensema | iCEBrkr: and your other hobbys include swimming with sharks and eating nails? |
| 16:14.28 | iCEBrkr | hensema: hehe |
| 16:14.43 | *** join/#asterisk T-Squared (n=T-Square@hidden.serreyn.com) |
| 16:14.51 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 16:14.54 | iCEBrkr | Not sure if it still exists, but Apache ran on OS/2 |
| 16:15.00 | enemy^x | TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8300' |
| 16:15.07 | enemy^x | why is the 0 stripped? the number was 08300 |
| 16:15.13 | enemy^x | is it within zapata.conf? |
| 16:15.20 | Winkie | iDunno: we use PXE w/etherboot to bootstrap the boxes, and home will be on NFS sooner or later |
| 16:15.30 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 16:15.31 | Winkie | enemy^x: is this incoming? |
| 16:15.44 | Winkie | iDunno: you have any experience with setting up HA NFS? |
| 16:15.52 | enemy^x | winkie: outgoing. |
| 16:16.07 | Winkie | enemy^x: it's possible you need to pause, or so i've heard |
| 16:16.15 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 16:16.42 | T-Squared | anyone have any experience with AMP? when doing outgoing ZAP channels, I am not seeing rollover to the other ZAP trunks for outgoing calls, the 2nd person just gets a busy signal |
| 16:16.52 | iCEBrkr | ~amp |
| 16:16.53 | jbot | it has been said that amp is NOT supported here! people using it should join #amportal |
| 16:16.58 | Winkie | bah beat me to it |
| 16:17.01 | iCEBrkr | :) |
| 16:17.03 | enemy^x | Winkie:nationalprefix=0 does this have something to do with it? |
| 16:17.11 | Winkie | someone wanna tell me what AMP actually is? |
| 16:17.18 | Winkie | enemy^x: I couldn't say i'm afraid |
| 16:17.29 | iDunno | Winkie: nah :) |
| 16:17.30 | wunderkin | enemy^x, probably |
| 16:17.32 | iCEBrkr | Winkie: It's some 'control panel' for Asterisk. |
| 16:17.37 | Winkie | ah right |
| 16:17.38 | T-Squared | I would but it says that the channel does not exist (so not like I didn't try) |
| 16:17.40 | austinnichols101 | winkie: brower-based admin for astrisk. |
| 16:17.44 | Winkie | pff control panels are for people who can't use a CLI |
| 16:17.47 | iCEBrkr | Winkie: I attempted to install it once, but it was bloated and required all this lame dependencies |
| 16:18.03 | Winkie | iDunno: i need to set up ideally some sort of striped and mirrored NFS service |
| 16:18.18 | Winkie | IE we have 2x200 gig in each server, and i want say 500 gig total space available :) |
| 16:18.23 | *** part/#asterisk T-Squared (n=T-Square@hidden.serreyn.com) |
| 16:18.32 | Winkie | i'm thinking raid 0 each server and then use drbd to duplicate it |
| 16:18.39 | Winkie | but as for the network side, god knows |
| 16:18.48 | iCEBrkr | Winkie: It's just that the learning curve for Asterisk is 3-fold. The person needs to understand Linux, Asterisk itself and basic telephoney stuff. |
| 16:19.02 | *** join/#asterisk coppice (n=chatzill@90.201.17.210.dyn.pacific.net.hk) |
| 16:19.06 | enemy^x | it was nationalprefix |
| 16:19.12 | austinnichols101 | iCEBrkr: well said |
| 16:19.36 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 16:19.55 | Winkie | iCEBrkr: that's true enough but i've started out from virtually nothing into setting up an E1 connected asterisk that handles 8000 calls a day, although i haven't got it working yet :) |
| 16:19.56 | iCEBrkr | ...on top of learning all that people just want it to WORK, cuz it's 1) cool, 2) can save them money. |
| 16:20.00 | Winkie | although i know linux :) |
| 16:20.22 | Winkie | well if they want it to just work they should have bought something embedded ;) |
| 16:20.23 | iCEBrkr | It's just too much stuff to learn at once.. It's all interconnected in some way or another |
| 16:20.37 | redder86 | I've got a channel coming in to Asterisk via PSTN (TDM400) and being delivered over IAX2 to the peer. If I make a recording using the Monitor() app in Asterisk, and if I also make a recording of the same call on the IAX2 peer, and if I then compare the two recordings ... I see that they are identical in timings and frequencies, but the amplitude of the waves (the volume) is noticeably reduced on the IAX2 peer-side. As I've authored the IAX2 pee |
| 16:20.41 | *** join/#asterisk Drew__ (n=foo@zux221-186-224.adsl.green.ch) |
| 16:22.20 | Winkie | redder86: wrapped i'm afraid |
| 16:22.24 | ManxPower | redder86, Maybe Monitor on the system with the Zap interface gets the audio before the gains are applied. |
| 16:22.27 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:22.46 | redder86 | Winkie: wrapped? |
| 16:22.56 | redder86 | MAnxPower: which gains affect IAX2 ? |
| 16:23.01 | Winkie | redder86: amplitude of the waves (the volume) is noticeably reduced on the IAX2 peer-side. As I've |
| 16:23.01 | Winkie | <PROTECTED> |
| 16:23.01 | Winkie | 16:22.11 -!- Drew__ [n=foo@zux221-186-224.adsl.green.ch] has joined #asterisk |
| 16:23.05 | Winkie | woah that's some bad pasting |
| 16:23.09 | Winkie | i apologise |
| 16:23.12 | iCEBrkr | Sure sure. |
| 16:23.27 | redder86 | As I've authored the IAX2 peer (iaxmodem) I know that the peer is not doing anything with incoming audio to trigger this. So, could anyone explain what could be causing the volume difference between each end of the IAX2 channel? |
| 16:23.59 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 16:24.02 | salviadud | squirrels man... |
| 16:24.15 | salviadud | i mean, i dunno |
| 16:24.15 | Winkie | redder86: ManxPower's idea is interesting, but it's rxgain in zapata.conf i assume he's referring to |
| 16:24.21 | iCEBrkr | BADGER BADGER BADGER BADGER BADGER BADGER BADGER |
| 16:24.21 | Winkie | or zaptel.conf, i dunno :( |
| 16:24.40 | ManxPower | redder86, rdgain/txgain is only for Zap |
| 16:24.57 | redder86 | Winkie, ManxPower: in any case, gains in Zap are at 0 |
| 16:25.44 | Winkie | redder86: it's an interesting problem, the answer would be that i don't know. How significant is this difference? |
| 16:25.55 | redder86 | I'd say that the volume is reduced at about 75%. I'm getting reports of this, and as it doesn't happen to all iaxmodem users I'd guess that it's something wrong with these fellows' Asterisk configs. |
| 16:26.54 | redder86 | It's a significant-enough decrease in volume that spandsp in iaxmodem won't even detect the audio as fax audio. |
| 16:26.59 | *** join/#asterisk stoffell (n=stoffell@d51A4D720.access.telenet.be) |
| 16:27.29 | Winkie | uh, chan_zap does that |
| 16:27.31 | Winkie | iirc |
| 16:27.37 | Winkie | perhaps iaxmodem has something i don't know about |
| 16:28.30 | redder86 | iaxmodem just takes the audio samples that it receives on iax2 and passes them to spandsp. The recording is made by just saving it to disk between those two steps. |
| 16:29.14 | remiss | uh.. i don't get it |
| 16:29.33 | Winkie | i'm confused as to how it could happen, have you set up a test peer and tried locally? |
| 16:29.52 | remiss | Failed to authenticate on INVITE to '"Unknown" <sip:Unknown@10.45.0.3>;tag=as7ecbf807' <-- alright.. i don't want it to do that? |
| 16:30.01 | redder86 | I've no problem with faxing and such with iaxmodem locally, so no I can't reproduce the problem here. |
| 16:30.06 | remiss | doh.. i got it.. never mind.. |
| 16:30.07 | *** join/#asterisk RoyK (n=roy@ti211310a080-4532.bb.online.no) |
| 16:30.17 | Winkie | redder86: do you have a copy of their asterisk configs? |
| 16:30.31 | redder86 | they've sent me some of them |
| 16:30.37 | redder86 | which ones would be suspect? |
| 16:31.00 | Winkie | redder86: well iax.conf and extensions.conf i'd guess, but i haven't a CLUE |
| 16:31.02 | Winkie | to be honest |
| 16:31.34 | *** join/#asterisk RV-Dioxide (i=appleboy@ip68-231-211-153.oc.oc.cox.net) |
| 16:31.49 | remiss | alright.. i don't get it.. |
| 16:33.06 | redder86 | okay, thanks anyway guys |
| 16:33.08 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 16:37.00 | *** join/#asterisk SibRwork (i=hidden-u@fw-int.transbeam.com) |
| 16:37.20 | SibRwork | calling from sip exten to sip exten, but not getting any audio to pass through...what could be the issue |
| 16:37.23 | SibRwork | ? |
| 16:38.45 | ManxPower | SibRphrek, sounds like a calssic NAT and/or firewall problem |
| 16:40.46 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 16:41.05 | SibRwork | ManxPower: my sip.conf has nat=yes |
| 16:41.14 | SibRwork | for the internal extentions |
| 16:41.23 | Winkie | SibRwork: is there NAT between them? |
| 16:41.40 | SibRwork | yeah |
| 16:41.51 | SibRwork | we're all on the inside of a network |
| 16:41.58 | SibRwork | 192.168.*.* |
| 16:42.05 | Winkie | and this is internal to internal? |
| 16:42.18 | SibRwork | int extension to int extension yes |
| 16:42.21 | ManxPower | SibRphrek, NAT is MUCH more complicated than nat=yes if you have Asterisk behind NAT and the SIP devices are not behind NAT or behind a different NAT |
| 16:42.36 | Winkie | SibRwork: why would you have NAT on an internal 192.168.0.0/24? |
| 16:42.55 | SibRwork | asterisk is external - in a colocation, the sip phones are internal |
| 16:43.50 | Winkie | ah |
| 16:43.58 | Winkie | set up a VPN? |
| 16:44.13 | SibRwork | no |
| 16:44.16 | SibRwork | can't do that right now |
| 16:44.35 | Winkie | well given multiple internal phones it's just silly not to |
| 16:45.31 | SibRwork | i'm just trying to run a test |
| 16:45.35 | SibRwork | b/c we are hooking up a new provider |
| 16:45.41 | SibRwork | and we are rushing to get something working |
| 16:45.47 | SibRwork | even tho we have other systems that work |
| 16:45.49 | SibRwork | long story |
| 16:46.16 | Winkie | ah, well there's probably a guide to NAT and STUN or whatever you need to use on voip-info.org |
| 16:47.00 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 16:49.01 | ManxPower | SibRphrek, if asterisk is on a public IP the SIP devices are behind NAT, it should work if you have nat=yes, but do NOT have localnet= or externip= in sip.conf and make sure there is no firewall active |
| 16:49.06 | *** join/#asterisk w32 (n=123@adsl-70-224-74-227.dsl.sbndin.ameritech.net) |
| 16:49.14 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 16:49.16 | oogle | download MP3, destroy the music industry. download Asterisk, destroy AT&T |
| 16:49.17 | ManxPower | also make sure you DO NOT HAVE allow=all use disallow=all and then allow=thesinglecodecyouwant |
| 16:49.36 | ManxPower | oogle, your ToDo list? |
| 16:49.42 | oogle | yes |
| 16:50.02 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 16:50.07 | ManxPower | My ToDo list just has "Become Emperor of the World" |
| 16:50.07 | Winkie | I have no ill feeling towards AT&T |
| 16:50.18 | Winkie | but if you want to destroy Your Communications, go right ahead |
| 16:50.27 | Winkie | our account manager has been ignorng my texts and my workmate's emails for 3 days |
| 16:50.29 | ManxPower | AT&T has been pretty good to my clients |
| 16:50.30 | Winkie | SERVICE! |
| 16:50.37 | ms345 | can someone suggest a good yet not too pricey VoIP phone for a receptionist? |
| 16:50.49 | Winkie | grandstream budgettone? |
| 16:50.54 | ManxPower | ms345, depends on what you need. |
| 16:50.59 | Winkie | name your features |
| 16:51.12 | ManxPower | The BudgeTone doesn't even support callerid name |
| 16:51.32 | ms345 | lots of buttons and blinky lights - seriously - probably doesn't matter |
| 16:51.45 | ManxPower | ms345, There are no cheap and good phones. |
| 16:51.46 | Winkie | gxp2000 has more blinky lights |
| 16:51.47 | Mimmus | 5 minutes to setup an Asterisk trunk between two sites for which company pays many many money every year... not bad... I'm pretty satisfied |
| 16:51.48 | ms345 | they only have CIDNUM now |
| 16:51.51 | Winkie | ManxPower: that's entirely not true |
| 16:51.52 | ManxPower | Try Polycom or Cisco |
| 16:51.59 | Winkie | the gxp2000 is the cheapest of our review set and it's the second best |
| 16:52.10 | ManxPower | ms345, no, the BT101 does not even have a display that is able to support letters |
| 16:52.12 | Winkie | the sipura one is midrange and still the best :) |
| 16:52.12 | ms345 | cool - I'll check the gxp2000 out |
| 16:52.27 | oogle | does anyone know a good low-cpu-cost text to speech generator that i can stream in to asterisk besides festival? |
| 16:52.45 | Mimmus | Winkie: is sipura now Linksys or Cisco? |
| 16:52.47 | brettnem | do you guys find that a lot of providers don't give you CNAM? |
| 16:52.49 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 16:52.49 | ms345 | any polycom or cisco model numbers to look at? |
| 16:52.54 | ManxPower | Winkie, For 3 years GS has been releasing somewhat crappy hardware with some of the most bug ridden firmware in the industry. Sorry, but I can't trust them. |
| 16:53.03 | oogle | i can always write the integration myself |
| 16:53.08 | Winkie | Mimmus: i think they're linksys' and cisco own them, or something |
| 16:53.09 | oogle | as long as it has an api |
| 16:53.14 | Winkie | ManxPower: no that's aastra :) |
| 16:53.23 | Winkie | but yes i will agree there are issues |
| 16:53.25 | IronHelix | GS releases products then 1-3 years later the firmware comes to usable state :) |
| 16:53.25 | brettnem | ms345 have you found that many providers don't provider caller id name? |
| 16:53.30 | Winkie | headphone port speaker disconnection / firmware issues etc |
| 16:53.41 | ManxPower | Winkie, AAstra has not even had a SIP phone for a year |
| 16:53.50 | IronHelix | ...? |
| 16:53.54 | Winkie | ManxPower: i have the 480i and the other one and they're appauling |
| 16:54.06 | Winkie | extremely expensive, don't work at all in the case of the 480i and the build quality is terrible |
| 16:54.11 | IronHelix | 480i, 480i-ct, 9133i, 9122i... |
| 16:54.15 | Winkie | the volume control on the 9133i is SO bad |
| 16:54.16 | ms345 | no - I get CIDNAME from BellSouth and ITC, my two PRI providers |
| 16:54.21 | ManxPower | brettnem, the only providers that I know of that don't do CIDNAME are VoIP ones and only a moron would use a VoIP ITSP for production use. |
| 16:54.23 | Winkie | it's just a bit of plastic |
| 16:54.38 | Mimmus | my boss says that 100Euro/phone limit is mandatory |
| 16:54.43 | ManxPower | Winkie, Oh, there are many horrble IP phones out there, but most of them are fairly new. |
| 16:54.45 | brettnem | ManxPower: interesting.. ;) |
| 16:54.46 | Winkie | right i'm off to get burger king i think |
| 16:55.03 | Winkie | Mimmus: you won't get much for that |
| 16:55.21 | brettnem | ManxPower: I'm trying to figure out how to sell a CNAM dipping service.. I have the program together... don't know if there is interest |
| 16:55.22 | ManxPower | 100 Euro is, what, $140 |
| 16:55.33 | Winkie | 16:57.16 <@Shaniqua> 100 eur = 119.04 usd |
| 16:55.34 | ManxPower | brettnem, there are a couple of them out there. |
| 16:55.51 | ManxPower | Wow! The exchange rate is getting slightly better. |
| 16:56.03 | Winkie | 16:57.45 <@Shaniqua> 100 usd = 56.75 gbp |
| 16:56.08 | Winkie | not by much |
| 16:56.08 | Mimmus | ehm... saying all truth: limit is lower... |
| 16:56.09 | IronHelix | brett- i think theres interest, if the price is right |
| 16:56.31 | Winkie | Mimmus: nobody produces a phone at that price i'm afraid |
| 16:56.35 | Winkie | so unless you want to use software |
| 16:56.41 | Mimmus | I'm forced to buy GrandStream or 'made in china/taiwan' |
| 16:56.46 | Winkie | well if they do produce a phone at that price i haven't seen it and it probably sucks |
| 16:56.59 | IronHelix | 120 would buy you a linksys i think, maybe not one of the better ones tho |
| 16:57.10 | Winkie | perhaps the old 841? |
| 16:57.27 | Mimmus | Winkie: GS 101: 65E, GS GXP-2000: 90E |
| 16:57.32 | IronHelix | yeah you're pretty much stuck to a sipura 841 or a grandstream 2000 |
| 16:57.38 | Winkie | Mimmus: that's pretty cheap |
| 16:57.55 | Winkie | costs me 109 eur for a gxp2000 :( |
| 16:57.56 | Winkie | bastards |
| 16:57.58 | Winkie | anyway i'm going! |
| 16:57.59 | Winkie | bye |
| 16:58.03 | IronHelix | before you go |
| 16:58.06 | Mimmus | I'm pushing GXP2000. Can I afford the risk to buy 50 of these or not? |
| 16:58.08 | IronHelix | DONT BUY A GS 101 |
| 16:58.12 | IronHelix | buy one |
| 16:58.15 | IronHelix | see if you like it |
| 16:58.17 | IronHelix | if you do buy 50 |
| 16:58.23 | Qwell | 49 |
| 16:58.36 | Winkie | Mimmus: if you plan to use headsets, make sure you get the latest revision and check it out |
| 16:58.43 | Winkie | because there is a problem with the headset port on ze back |
| 16:58.48 | Winkie | anyway i really am gone now |
| 16:59.24 | Mimmus | IronHelix: ok, thanks! I buyed a GS101 but there are problems, especially because there is not an alphanumeric display and microphone is low |
| 16:59.32 | Mimmus | Winkie: bye, thanks |
| 16:59.42 | IronHelix | yeah the GS phones look and feel like toys |
| 16:59.47 | IronHelix | thus the nickname 'barbietone' |
| 17:00.18 | _Sam-- | Mimmus : there are only a handful of reasonably happy GXP2000 owners |
| 17:00.19 | Mimmus | IronHelix: tried also an ATCOM AT320 (PA1688S chip), not so bad but some problems too |
| 17:00.23 | _Sam-- | i happen to be one of them |
| 17:00.32 | IronHelix | me too |
| 17:00.33 | _Sam-- | but many dissatisfied customers as well. |
| 17:00.36 | IronHelix | the gxp is a good phone |
| 17:00.38 | IronHelix | just not the 100 |
| 17:00.50 | zoa | hey ho sam |
| 17:00.58 | zoa | we have a lot of happy gxp users |
| 17:01.01 | _Sam-- | hey there joach |
| 17:01.15 | zoa | sold quite a big amount of those already |
| 17:01.17 | _Sam-- | most people here do nothing but talk down the gxp2000 |
| 17:01.20 | zoa | hey ho! |
| 17:01.28 | zoa | ive seen some issues but very few |
| 17:01.32 | _Sam-- | it has a lot of probelms actually. |
| 17:01.40 | _Sam-- | i wouldnt put 50 in one place, not with qualify = yes anyway :) |
| 17:01.45 | IronHelix | hey sam/zoa, have you found the gxp has a high failure rate (esp the handset)? also is GS support responsive with replacement parts? |
| 17:01.49 | Mimmus | I'm not looking for a lot of features but decent sound quality, alphanumeric LCD, MWI, call-transfer and a few others... |
| 17:02.09 | zoa | IronHelix: we rma'd some |
| 17:02.09 | zoa | compared to snom a lot |
| 17:02.13 | zoa | compared to the bt101 very few |
| 17:02.17 | _Sam-- | i have 40 gxp2000s in different places |
| 17:02.21 | _Sam-- | i have had none fail yet |
| 17:02.27 | _Sam-- | but they are all bug filled |
| 17:02.35 | zoa | true |
| 17:02.35 | _Sam-- | if i turn on BLF, * segfaults |
| 17:02.40 | _Sam-- | if i turn on qualify = yes, moh breaks |
| 17:02.41 | zoa | depends what you want to use it for, yes |
| 17:02.41 | salviadud | i need some help... |
| 17:02.43 | salviadud | http://pastebin.ca/41959 |
| 17:02.58 | salviadud | i am not sure what to add to my extensions.conf file |
| 17:03.00 | Mimmus | why not "qualify = yes"? |
| 17:03.02 | _Sam-- | they are fine if you want to check for new firmware every day |
| 17:03.07 | zoa | :) |
| 17:03.20 | IronHelix | salviadud- iax sends context as well as exten |
| 17:03.21 | _Sam-- | read the GXP2000 wiki page |
| 17:03.38 | _Sam-- | http://www.voip-info.org/wiki/view/GXP-2000 |
| 17:03.40 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 17:04.01 | _Sam-- | i think your customers/users/whoever could be happy with 50 of them...but its only a 60/40 chance. |
| 17:04.03 | salviadud | so, i should create a context called {fwdnumber}-fwd-incoming??? |
| 17:04.06 | _Sam-- | just ask justinu |
| 17:04.08 | IronHelix | its looking for the FWD number in [fwd-incoming] |
| 17:04.18 | IronHelix | no make [fwd-incoming] and put an extension s in that |
| 17:04.38 | _Sam-- | i used to be the biggest gxp supporter here...used to be that is. |
| 17:04.39 | salviadud | thanx man! :D |
| 17:04.51 | _Sam-- | i like them...but i am seeing why some people wouldnt |
| 17:04.55 | Mimmus | _Sam--: I read. Sic. |
| 17:06.03 | _Sam-- | Mimmus: what type of environment is the installation? like, a call center, a regular office, what? |
| 17:06.13 | *** join/#asterisk starwarez (n=starware@201.135.68.20) |
| 17:06.22 | starwarez | hi all |
| 17:06.58 | zoa | i would recommend snom or so for a customer |
| 17:07.11 | Mimmus | people think that Asterisk is a cheap change... then you suggesto 250$ phones... and my boss dies! |
| 17:07.30 | zoa | give him a channel bank |
| 17:07.30 | Mimmus | _Sam--: an office with 60 people |
| 17:07.32 | wasim | Mimmus: buy $50 pa168 phones |
| 17:07.52 | _Sam-- | channel banks cost as much per port as a gxp |
| 17:07.53 | wasim | Mimmus: or buy $85 4 port fxs ata |
| 17:07.54 | _Sam-- | i would rather have a gxp |
| 17:07.59 | austinnichols101 | it's cheap if you compare it to a Lucent G3 |
| 17:08.09 | Mimmus | wasim: I have three AT320 but I'm currently unable to find them on the italian market |
| 17:08.26 | _Sam-- | you are talking 60 PHONES>..your boss if he can afford 60 desks, should be able to afford a phone system. |
| 17:08.27 | wasim | Mimmus: atcom ship direct as well |
| 17:08.33 | _Sam-- | i mean, its not like you are a small shop |
| 17:08.43 | _Sam-- | spend now, or you will end up just spending more later when you're unhappy |
| 17:09.11 | _Sam-- | <personal opinion of course> |
| 17:09.15 | Mimmus | wasim: I don't like ATAs: I have no CallerID display on commonly used analog phone sin Italy |
| 17:09.33 | wasim | Mimmus: you should, we get clid just fine on cheap $5 phones |
| 17:09.36 | *** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net) |
| 17:10.03 | Mimmus | _Sam--: I'm trying to explain that currently maintenance of our legacy PBXs is 25,000Euro/year... |
| 17:10.06 | paulhuynh | i'm using asterisk w/ pap2 what codec can i use that used least amount of bandwidth |
| 17:10.18 | paulhuynh | and that i don't have to pay the licensed from digium |
| 17:10.23 | wasim | 60*5=300 + 60/4*85 + a small * box |
| 17:10.36 | Mimmus | wasim: clid in Italy is a luxury! |
| 17:10.40 | wasim | thats not much, enough for a cheap ass boss |
| 17:13.41 | _Sam-- | Mimmus: maybe not every desktop needs the same phone |
| 17:13.58 | *** join/#asterisk gkchicago (n=gkchicag@66.9.120.18) |
| 17:14.50 | Mimmus | _Sam--: this is true even if thius could be mor emanageable |
| 17:15.05 | gkchicago | Does anyone know why `exten => s,202,MixMonitor(/tmp/recording-%d.wav)` isn't putting an incrementing number in the filename? |
| 17:15.16 | Mimmus | _Sam--: provisioning, configuring, training... |
| 17:15.16 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 17:15.35 | paulhuynh | any idea on the codec setting |
| 17:15.40 | paulhuynh | i'm using asterisk w/ pap2 what codec can i use that used least amount of bandwidth |
| 17:15.43 | paulhuynh | and that i don't have to pay the licensed from digium |
| 17:15.49 | Mimmus | but too risky... better differentiating... |
| 17:16.20 | Mimmus | I hope to buy some GXP-2000s and some AT320 |
| 17:17.06 | *** join/#asterisk RoyK (n=roy@ti211310a080-4532.bb.online.no) |
| 17:18.46 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 17:18.52 | Mimmus | bye and thanks, I will break you in the future again... |
| 17:19.07 | Winkie | for applications where you could be sending over a thousand manager commands to an Asterisk server from different client machines daily you will almost definitely have at least one crash/deadlock happen per day. |
| 17:19.07 | websae | I keep getting dropped calls, how can one debug this? |
| 17:19.10 | Winkie | is this still true? |
| 17:19.27 | websae | i would greatly appreciate if anyone knows how to debug dropped calls.. |
| 17:19.35 | Winkie | websae: sip? |
| 17:19.40 | websae | yes |
| 17:19.48 | Winkie | sip debug? :) |
| 17:20.05 | websae | is there a way to throw sip debug in a file? |
| 17:20.18 | websae | i guess that's what i am trying to figure out |
| 17:21.24 | Winkie | not that i'm aware of, possibly through the manager API but don't quote me on that |
| 17:21.32 | NewSole | anyone good with ser |
| 17:21.46 | Winkie | NewSole: you've been asking all day, no, and stop it |
| 17:22.08 | websae | so sib debug...no way to get that to a file? |
| 17:25.27 | websae | hi |
| 17:26.16 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:27.53 | DaPrivateer | trying to compile on a new box and getting undefined reference to `__h_error' any ideas? |
| 17:28.00 | DaPrivateer | (freebsd 5.4-stable) |
| 17:29.07 | wasim | www.gentoo.org |
| 17:29.14 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 17:29.16 | salviadud | guys, i got FWD working |
| 17:29.17 | salviadud | but |
| 17:29.26 | salviadud | i can't dial toll free numbers yet... |
| 17:29.40 | salviadud | something to do with the dialplan? |
| 17:29.49 | mikefoo | is it legal to originate a call from a terminating voip provider and have a outbound display number of say, a home phone, any not any did the voip provider assigned. |
| 17:30.28 | Winkie | mikefoo: it depends on your country, i'd be surprised if your voip provider's telecomms provider allows that |
| 17:30.51 | mikefoo | Ahh they might not allow a custom outbound number to go through? |
| 17:30.59 | mikefoo | im in the US btw |
| 17:31.33 | Winkie | mikefoo: they almost 100% won't |
| 17:31.39 | Winkie | and i don't know US laws i'm afraid |
| 17:32.09 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 17:32.23 | Flauto | hi all |
| 17:33.44 | Flauto | it is quiet here |
| 17:33.45 | *** join/#asterisk Assid (n=assid@203.115.64.11) |
| 17:33.46 | Assid | heya |
| 17:33.57 | Flauto | hi assid |
| 17:34.08 | salviadud | i need some help |
| 17:34.09 | salviadud | http://pastebin.ca/41963 |
| 17:34.17 | Assid | umm.. can anyone suggest a provider which lets you port a number in for DID.. and lets add more incoming channels |
| 17:34.22 | Assid | voicepulse has a hard limit of 4 |
| 17:34.23 | salviadud | you guys have fwd with toll free numbers enabled? |
| 17:34.27 | PupenoL | what character set/encoding does the manager of asterisk use/receive ? |
| 17:34.55 | *** join/#asterisk _dusty (n=Dusty@64.89.118.139) |
| 17:34.57 | Flauto | assid, look into broadvoice |
| 17:34.59 | gkchicago | In the record app you can put "%d" in the filename to add an incrementing number.. is there a similar function for the MixMonitor app? |
| 17:35.05 | Assid | Flauto: heard bad reviews of BV |
| 17:35.09 | Flauto | as i understand, incoming is not limited |
| 17:35.23 | Flauto | i have used it for more than a half year now |
| 17:35.32 | Flauto | i had problems last year in may and june |
| 17:35.36 | Flauto | other than that |
| 17:35.41 | Flauto | it has been okay |
| 17:36.05 | Flauto | once in a while, one of the 5 or 6 proxies would screw up for a little bit |
| 17:36.08 | Flauto | that is about it |
| 17:36.17 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:36.19 | Assid | what about quality |
| 17:36.40 | Assid | voicepulse used to give good quality |
| 17:37.09 | mikefoo | Assid: how many concurrent calls can you make with voicepulse? |
| 17:37.19 | Assid | not make.. incoming |
| 17:37.24 | Assid | incomng has a 4 channel limit |
| 17:37.31 | mikefoo | outbound? |
| 17:37.35 | mikefoo | sdame? |
| 17:37.37 | mikefoo | same* |
| 17:37.39 | Assid | nah |
| 17:37.41 | Assid | pay per use |
| 17:37.41 | Assid | so.. |
| 17:37.56 | mikefoo | unlimited..? |
| 17:37.56 | mikefoo | in theory. |
| 17:38.07 | Assid | yeah |
| 17:38.09 | Flauto | the call quality is not bad depending on where you are calling |
| 17:38.10 | Flauto | hehe |
| 17:38.11 | Assid | so long as you have the bandwith |
| 17:38.24 | Assid | Flauto: shouldnt be that big of a difference |
| 17:38.27 | freat | am I doing the right thing, asking users to please not use those swivel jack things (that go inline on handsets to keep it from tangling up)? |
| 17:38.36 | Flauto | the funny thing is |
| 17:38.38 | Flauto | i call china a lot |
| 17:38.44 | mikefoo | Assid: yah I will be on a 10mbit connection, need a decent provider to terminate for me. |
| 17:38.47 | Flauto | most of the major cities are good |
| 17:38.55 | Winkie | freat: what's the problem with them? |
| 17:39.03 | Flauto | but some smaller ones would come cross to have problems sometimes |
| 17:39.09 | Assid | outbound ive seen pretty decent feedback for VP and teliax |
| 17:39.22 | Flauto | i would think that they land in the major cities only |
| 17:39.26 | remiss | Failed to authenticate on INVITE to '"Unknown" <sip:Unknown@84.48.68.129>;tag=as274d9983' <-- on outbound calls i get this.. what do i need to do to change "Unknown"? |
| 17:39.31 | Assid | voipjet is in there.. but THESE guys apparetly prefer teliax over voipjet |
| 17:39.56 | Assid | remiss: change the username and caller id/ name registered on the sip device |
| 17:40.14 | mikefoo | Assid: how about on outbound caller id display, can you cutomize? |
| 17:40.15 | freat | Winkie: seems like they are causing problems with their calls. Report is that their speakerphone works great (Polycom phones) but that their handsets are not as good. The people with the complaints all bought some cheap kind of swivel thingie for their phone cords. |
| 17:40.25 | remiss | Assid: no way to set it in asterisk? |
| 17:40.33 | Assid | mikefoo: yes.. all of them let you do so |
| 17:41.07 | mikefoo | Assid: yah there is a way just making sure they allow it. |
| 17:41.15 | Winkie | freat: ah, well if they're cheap they'll probably use a copper strip or something similar to provide the connection, and i imagine it oxidises easily |
| 17:41.24 | mikefoo | Assid: is there any legal issues I should worry about with that. |
| 17:41.33 | freat | Winkie: are there good ones out there? |
| 17:41.44 | Assid | if im not mistaken yuo are SUPPOSED to provide your caller id |
| 17:41.47 | Assid | however.. |
| 17:41.48 | freat | Winkie: I would like to get one and test it myself |
| 17:41.51 | Winkie | freat: i wouldn't know, all i can say is my mum's had one on her phone for the last 4 years and it sounded pretty good :) |
| 17:41.58 | freat | ahh ok |
| 17:42.02 | Assid | if they realy wanna.. they can always trace the call back to you by seeing the company which is terminating |
| 17:42.08 | Assid | and reading the call records off that |
| 17:42.34 | Assid | freat: polycoms shouldnt be giving oyou those problems |
| 17:42.37 | Assid | sipura.. yes |
| 17:42.48 | mikefoo | Assid: yah but I will be the outsource for appoints of different companies, can I legally display the companies I dial out for, their caller id? |
| 17:42.51 | Assid | but then sipura you just upgrade the firmware |
| 17:42.57 | freat | Assid: yeah exactly. Polycom quality is awesome |
| 17:43.07 | freat | Assid: got one right here on my desk (IP500) |
| 17:43.13 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 17:43.14 | Assid | i really wish i can get a poly phone |
| 17:43.15 | SpaceBass | hey folks |
| 17:43.27 | Assid | the guy was supposed to send me one.. but unfortunately couldnt make it |
| 17:43.28 | Assid | so.. |
| 17:43.33 | Assid | im stuck with soft phone |
| 17:43.47 | Assid | they are using poly 501 |
| 17:44.12 | Assid | i was supposed to get a phone as a thank you for setting up voip for him in a real short amt of time |
| 17:44.23 | Assid | not to mention.. i asked him for it |
| 17:44.24 | Assid | hehe |
| 17:44.43 | freat | Assid: keep harassing them then, if that was part of the deal. |
| 17:44.46 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
| 17:44.54 | kpettit | Any ways to handle echo on IAXY's devices? I have a confrence bridge with 6 or so Digium IAXY's connected to it |
| 17:44.55 | Assid | wasnt "REALLY" part of the deal |
| 17:44.57 | websae | anyone know a good way to record sip debug to afile? |
| 17:45.02 | Assid | he said yeah cool.. im sending it to ya |
| 17:45.10 | Assid | i dont REALLY REALLY need it |
| 17:45.16 | kpettit | after some time I get horrid echo feedback. And if they all hang up and call back in the previous echo is still there |
| 17:45.17 | Assid | but yeah.. would be nice |
| 17:45.18 | stoffell | websae, yes, putty to the box, and let your putty capture the session.. then you have a file |
| 17:45.24 | Assid | softphones arent the best thing to play with |
| 17:45.34 | kpettit | If I yell "Hello" it will start to echo, and everybody hangs up, I call back in and it's still echo'ing |
| 17:45.36 | Assid | Winkie: you ever used any inbound provider |
| 17:45.50 | Winkie | Assid: sipgate |
| 17:45.59 | Assid | you in UK ? |
| 17:46.05 | Winkie | aye |
| 17:46.09 | Flauto | assid, the good think about bv is that you get a flat monthly fee |
| 17:46.09 | SpaceBass | does digium still sell a student/hobbiest (IE Low Cost!) version of their fxo/fxs board? |
| 17:46.09 | Assid | thought so |
| 17:46.10 | Assid | hehe |
| 17:46.15 | SpaceBass | my clone x100p sucks a-hole |
| 17:46.27 | Assid | Flauto: i do need quality.. reviews of BV isnt that great |
| 17:46.51 | Assid | SpaceBass: how do you find it |
| 17:46.57 | kpettit | SpaceBass, clone x100p ?? |
| 17:47.02 | Winkie | Assid: why what's up? |
| 17:47.23 | Assid | wanna give these folks a good provider |
| 17:47.32 | Assid | they willl need around 6-8 channels.. and scalable |
| 17:47.41 | SpaceBass | so-so...used it for incoming calls mostly for a while but not using it for all LD calls for work and home... quality is better than my analog lines by far, but I do get some "break-up" as the call progresses into more than 1 hour 30 mins |
| 17:47.51 | Winkie | ah, not got any experience of an incoming provider, really i need a top end one in canada at some point |
| 17:47.52 | Assid | voicepulse isnt offering anything on that.. and well.. frankly i never tried any other provider except nufone for incoming |
| 17:47.56 | SpaceBass | And I put in a request to port my number to BV 2 months ago and they still havent done it |
| 17:48.01 | SpaceBass | kpettit, yet...x100p clone |
| 17:48.10 | kpettit | http://www.x100p.com/products_2.htm like this device? |
| 17:48.18 | kpettit | or there PCI card/ |
| 17:48.34 | remiss | :( |
| 17:48.37 | kpettit | I just bought two of there IAX FX boxes this morning. But I hate using PayPal to buy crap |
| 17:48.43 | kpettit | where did you get yours? |
| 17:48.55 | SpaceBass | kpettit, no...like this: http://cgi.ebay.com/X100P-Compatible-Card-Asterisk_W0QQitemZ5866415725QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 17:48.58 | remiss | why do i keep getting proxy authentication required on outbound calls? |
| 17:49.32 | *** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 17:49.35 | Winkie | because you haven't authenticated with your proxy? |
| 17:50.10 | kpettit | SpaceBass, ahh, I hven't used those yet. I was excited about there FX one so I can find a replacement for the Digium IAXY's. The really bug me |
| 17:50.31 | remiss | Winkie: probably :-/ |
| 17:50.37 | SpaceBass | is the IAXy done? |
| 17:50.39 | Winkie | remiss: you got a register statement in sip.conf? |
| 17:50.56 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 17:51.03 | SpaceBass | kpettit, unless your stuck on IAX, the linksys PAP2 works well |
| 17:51.12 | Winkie | or the appropriate entries in the outbound definition? |
| 17:51.30 | *** join/#asterisk sdali (n=FrankM@68-189-43-111.dhcp.rdng.ca.charter.com) |
| 17:51.42 | SpaceBass | kpettit, that x100p clone worked GREAT in my PII 350mhz box...dropped them in my 1.7ghz box with 512mb ram and I get BAD echo |
| 17:51.46 | remiss | Winkie: yes.. and it's registered.. incoming works ok too |
| 17:51.53 | kpettit | we have alot of firewall issues with SIP, IAX seems to do the trick for the most part, but the boxes are unreliable |
| 17:52.00 | remiss | Winkie: the outbound definition? |
| 17:52.04 | kpettit | know of any clones for the FX box? |
| 17:52.07 | wasim | so, whats the longest iax2 call ever? |
| 17:52.13 | Flauto | spacebass, dont' you have echo problem with x100p |
| 17:52.19 | Qwell[] | clones of a cheap device? |
| 17:52.21 | Winkie | remiss: pastebin your sip config? |
| 17:52.30 | remiss | k |
| 17:52.32 | Winkie | wasim: i dunno if there's a world championship but i've left one on for a couple hours before 8) |
| 17:52.34 | SpaceBass | Flauto, yeah...thats what I just said |
| 17:52.40 | Assid | remiss: which provier |
| 17:52.55 | wasim | Winkie: i'm sure its in the days, if not weeks |
| 17:52.55 | mikefoo | Assid: sorry went to get some food.. |
| 17:53.01 | wasim | Winkie: if not more ... |
| 17:53.16 | SpaceBass | Flauto, to the point my wife is threatening to move out and get her own POTS phone service....now I'm forced to chose...asterisk...wife...asterisk...wife...(fyi, wife will win) |
| 17:53.38 | kpettit | I have major echo problems in my confrence bridge with the IAXY's |
| 17:53.52 | kpettit | wondering if it's device, asterisk or both |
| 17:53.56 | Winkie | wasim: i'm sure it probably is :) |
| 17:54.05 | SpaceBass | mine is like hearing yourself back through Kurt Cobain's guitar amp |
| 17:54.18 | sdali | Newbie duh type question - Why does the Voicemail application not recognize when a caller has pressed "#"? I'm expecting it to finish recording the msg, and then fall through. |
| 17:54.25 | Flauto | hehe, spacebass |
| 17:54.26 | Winkie | SpaceBass: nicely overdriven? |
| 17:54.27 | kpettit | Any way to get rid of that echo?? |
| 17:54.28 | Flauto | sorry to hear that |
| 17:54.45 | *** join/#asterisk Givur (n=mail@Gb95f.g.pppool.de) |
| 17:54.47 | Givur | Hi all |
| 17:54.49 | sdali | Here's 3 lines from extensions.conf |
| 17:54.53 | SpaceBass | Winkie, I've played with the RX and TX a lot |
| 17:54.56 | remiss | Winkie: http://pastebin.com/559852 |
| 17:55.02 | sdali | exten => 3,1,Voicemail(3456) |
| 17:55.02 | sdali | exten => #,1,Playback(thank-you-for-calling) |
| 17:55.02 | sdali | exten => #,n,Hangup |
| 17:55.04 | remiss | Assid: televoip.. norwegian.. |
| 17:55.07 | kpettit | I've got audiobuffers=32 set in my meetme.conf but that dosen't seem to help |
| 17:55.40 | kpettit | SpaceBass, you talking fax? |
| 17:55.48 | Winkie | remiss: shouldn't your type be friend? |
| 17:55.53 | Flauto | space, i have the same echo problem or other party can not hear me |
| 17:55.57 | SpaceBass | kpettit, fax is another issue I have...but no I was tlking voice |
| 17:56.03 | austinnichols101 | is there a way to do a factory reset of an SPA3000 from the web interface? I need to reset one at a remote site... |
| 17:56.04 | Flauto | either one, x100p does not work well |
| 17:56.06 | Winkie | but regardless, i'm going home and i can't see much else wrong, username/secret is fine from what i can see |
| 17:56.06 | kpettit | ah. |
| 17:56.06 | *** join/#asterisk ToTo (n=ToTo@host97-136.pool875.interbusiness.it) |
| 17:56.15 | kpettit | Working on fax now. Got it working really nice with iaxmodem and hylafax |
| 17:56.22 | austinnichols101 | manual says nothing |
| 17:56.26 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 17:57.02 | Flauto | austin, i guess no |
| 17:57.06 | remiss | W`gon: read something about that... i don't think it will work with this provder... |
| 17:57.10 | SpaceBass | I had fax working really well using dring until I upraded to AAH 2.x... |
| 17:57.24 | remiss | and it doesn't fix the problem either :-/ |
| 17:58.02 | kpettit | SpaceBass, I had about 90% with spandsp and rx/tx_fax. But I'm about 100% with iaxmodem and hylafax |
| 17:58.06 | Primer | Anyone know how a provider charges for forwarded calls to PSTN when voip is unavailable? |
| 17:58.16 | kpettit | plus you get all the cool stuff that normally comes with hylafax, makes it nice |
| 17:58.52 | stoffell | kpettit, did you use an existing howto? |
| 17:59.05 | kpettit | no, i'm making one though |
| 17:59.10 | Primer | we're looking to move our support 800 number to voip, but we need to ensure that it's always available. Seems some providers will route calls over the normal phone network if asterisk can't be reached over the internet |
| 17:59.18 | kpettit | It's kind of gentoo specific sense that's what I use, but it should work for about anybody |
| 17:59.31 | kpettit | I'll have a good wiki on it, here in a day or two. Just about have it done |
| 17:59.42 | stoffell | kpettit, ah, i see, great info is always appreciated, be it on the voip-info, or linked from there.. tnx |
| 17:59.52 | kpettit | Installing it on a few more system to make sure it works like I think. I've got this same setup on 5 machines now. |
| 17:59.56 | kpettit | one with 200 fax did's |
| 18:00.07 | stoffell | hehe :) |
| 18:00.15 | remiss | Proxy-Authenticate: Digest realm="84.48.68.129", nonce="43f610347d676648aa833a782f4dc569a3a530d3" <-- can this possibly be right? |
| 18:00.36 | Flauto | is there anyone would help me with a2billing? |
| 18:00.40 | kpettit | The guy that wrote iaxmodem has been helping us. It's been really cool, guess he's also the maintainer to hylafax |
| 18:01.07 | stoffell | ah, fun, guess that'll work out then ;) |
| 18:01.09 | kpettit | Contracting him for a week is sure alot cheaper than buying one of those $5000 fax boards. |
| 18:02.15 | sdali | Anyone know why the Voicemail() doesn't stop recording when a caller hits #? |
| 18:02.18 | stoffell | great! |
| 18:02.27 | Qwell[] | sdali: Bad dtmfmode? |
| 18:02.44 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 18:02.45 | stoffell | haven't tried iaxmodem yet, but will also have look in int now I know you're using it big-scale:) |
| 18:02.51 | sdali | Nope. Because it "hears" the correct mailbox to use. |
| 18:03.22 | sdali | Qwell[]: Nope. Because it "hears" the correct mailbox to use. |
| 18:04.08 | Qwell[] | sdali: Don't you call it with Voicemail(12345), you said? |
| 18:04.54 | sdali | Qwell[]: exten => 3,1,Voicemail(3456) |
| 18:04.55 | sdali | exten => #,1,Playback(thank-you-for-calling) |
| 18:04.55 | sdali | exten => #,n,Hangup |
| 18:05.09 | Qwell[] | yeah...so...check your dtmfmode |
| 18:05.49 | sdali | dtmfmode=rfc2833 |
| 18:05.57 | Qwell[] | and on the device? |
| 18:06.01 | Qwell[] | or whatever |
| 18:06.20 | sdali | I know it's hearing the dtmf digits correctly because a message does go to the correct box |
| 18:06.46 | Qwell[] | no, because when you pass in digits to Voicemail...that isn't dtmf |
| 18:07.14 | sdali | Is my basic understanding bad? I'm expecting it to har the #, then playback the thank you messge. |
| 18:07.31 | Qwell[] | No, because your dtmfmode is likely wrong, so it can't interpret any digits |
| 18:07.39 | areski | Flauto, I can help u a little bit if u have straight question :) |
| 18:07.55 | sdali | But getting to the correct mailbox requires correctly interpreting the dtmf tone I selected to get to that particular mailbox. |
| 18:08.15 | Qwell[] | sdali: you don't use dtmf to pick a mailbox, if you pass one in |
| 18:09.04 | xachen | If I do an extensions reload when there are calls going on, will that drop my calls? |
| 18:09.18 | Qwell[] | xachen: No, but things could get b0rked |
| 18:09.18 | sdali | Qwell[]: In the above code from my ext.conf, you see that they have to hit 3 to get to the vm box. |
| 18:09.38 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8) |
| 18:09.44 | Qwell[] | Thank you |
| 18:10.08 | *** join/#asterisk justinu (n=justin@72.18.13.34) |
| 18:10.14 | salviadud | hey! im trying to get toll free numbers, still no cigar http://pastebin.ca/41968 |
| 18:10.23 | salviadud | you guys got any ideas? |
| 18:10.24 | sdali | so it seems to me that the dtmf is being interpreted correctly |
| 18:10.39 | SpaceBass | sdali, you running asterisk@home? |
| 18:10.53 | sdali | nope, just straight asterisk 1.2.4 |
| 18:10.56 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:11.15 | SpaceBass | have # anywhere in your dialplan being used as a transfer? |
| 18:11.36 | sdali | Nope |
| 18:11.44 | sdali | Maybe I should verify that |
| 18:12.04 | SpaceBass | does the CLI show anything when you press # in voicemail? |
| 18:12.10 | SpaceBass | (also is # supposed to do anything in VM?) |
| 18:12.28 | Qwell[] | # should say you're done recording |
| 18:12.29 | Dr-Linux | hi _Sam-- |
| 18:12.39 | sdali | CLI doesn't hear it - or show anything |
| 18:13.01 | harryvv | hi spaceballs |
| 18:13.02 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 18:13.14 | harryvv | who here has delt with unlimitel? |
| 18:13.17 | SpaceBass | Harrrry, how are ya :) |
| 18:13.34 | afrosheen | has anyone had problems with DTMF via Zaptel suddenly not working very well? |
| 18:13.36 | harryvv | sixtel is having issues. |
| 18:13.48 | SpaceBass | afrosheen, not working at all, or just intermittent? |
| 18:13.58 | harryvv | it rings once on there then then my * rolls over to the local long distance carrier. |
| 18:14.03 | afrosheen | SpaceBass: intermittent, our incoming calls aren't always hitting their target extensions |
| 18:14.14 | harryvv | I need to put a play file name saying its out. |
| 18:14.32 | SpaceBass | afrosheen, hummmm what hardware? |
| 18:14.40 | afrosheen | SpaceBass: a pair of tdm400's |
| 18:15.14 | afrosheen | I tried rearranging the modules because I had a bad one last year but the problem persists |
| 18:15.24 | *** join/#asterisk LoonaTick (i=LoonaTic@office.contrabandict.nl) |
| 18:15.36 | LoonaTick | hi |
| 18:15.43 | starwarez | newbie question, Im stuck with sip clients behind nat, i read the documentation but i still have problems, using a public stun server... my question is: is posible to make a call between 2 cliens behind the same nat via an remote non-nat asrterisk server? |
| 18:15.58 | justinu | only if you set reinvite=no |
| 18:16.01 | sdali | The CLI says: Executing VoiceMail("SIP/gs2-ebec", "1234") , recording the msg, then stopped after a silence of 10 seconds, despite the fact that I've hit # several times |
| 18:16.22 | LoonaTick | anyone know where I can find settings for zapata.conf to install my telephone line? My telephone company didn't deliver them (yet). I have a Dutch KPN ISDN30 line (PRI) |
| 18:16.34 | afrosheen | starwarez: we solved that with our firewall by adding a second NIC to our * server and having one connection inside the firewall and one outside |
| 18:16.56 | justinu | LoonaTick: http://www.digium.com/asterisk_handbook/zapata.conf.pdf |
| 18:16.56 | LoonaTick | (is there perhaps a list with defaults available) |
| 18:17.05 | LoonaTick | thanks, will look at that one:) |
| 18:17.35 | starwarez | i tried nat=yes on the asterisk´s client config is not working |
| 18:17.57 | afrosheen | starwarez: that works for single nat but not double nat |
| 18:17.59 | justinu | nat=yes, canreinvite=no |
| 18:18.22 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 18:18.25 | afrosheen | starwarez: i.e. your server is firewalled/nat'ed and the client is behind a router or whatever, that's double nat and is a nightmare |
| 18:18.31 | jontow | awww.. wind took out the power in the CO and i lost a 225 day uptime on my voicemail machine :( :( |
| 18:18.34 | justinu | he said his asterisk wasn't natted |
| 18:18.42 | justinu | at least thats what I gather |
| 18:18.45 | warthawg | my story on asterisk on openwrt is up on newsforge |
| 18:18.53 | warthawg | http://mobile.newsforge.com/article.pl?sid=06/02/09/1727256&tid=104&tid=132 |
| 18:19.00 | LoonaTick | justinu: Thanks, but that doesn't contain any default values to get this specific line to work.. was hoping someone here maybe had the same situation. |
| 18:19.02 | starwarez | no, no double nat here... * is not nated |
| 18:19.06 | SpaceBass | anyone have a recomendation for a french DID provider? |
| 18:19.06 | warthawg | it's a wonderful story |
| 18:19.13 | justinu | LoonaTick: euroisdn? |
| 18:19.19 | LoonaTick | justinu: yes |
| 18:19.22 | afrosheen | oh if it's single nat then you don't have to worry about a stun server, just set nat=yes in your config on the server |
| 18:19.27 | justinu | d channel should be on timeslot 16 |
| 18:19.32 | justinu | switchtype => euroisdn |
| 18:19.44 | Dr-Linux | hi justinu |
| 18:19.54 | LoonaTick | justinu: Thanks, and the signalling? |
| 18:19.55 | Dr-Linux | i made something very good, but i a bit problem :) |
| 18:20.03 | afrosheen | Dr-Linux: how did it taste |
| 18:20.09 | justinu | pri_cpe |
| 18:20.23 | *** join/#asterisk saftsack (n=saftsack@p54A7FEB2.dip.t-dialin.net) |
| 18:20.30 | starwarez | i have no audio, i enabled sip debug but i cant find the error :S |
| 18:20.41 | LoonaTick | thanks :) |
| 18:20.57 | LoonaTick | justinu: Once the dc plugs the line back in i'll let you know if it works :) |
| 18:20.57 | afrosheen | starwarez: well if you don't do nat=yes on a per-device basis then you should get one-way audio at least |
| 18:21.08 | *** join/#asterisk PakiPenguin (n=bah@linuxpakistan/admin/pakipenguin) |
| 18:21.08 | justinu | LoonaTick: you also need zaptel.conf configured |
| 18:21.10 | Dr-Linux | afrosheen: you will know when you suck it ;) |
| 18:21.22 | afrosheen | Dr-Linux: well you said you bit a problem |
| 18:21.23 | afrosheen | ;) |
| 18:21.31 | justinu | LoonaTick: you should prorbably use span=1,0,0,ccs,hdb3,crc |
| 18:21.35 | justinu | Dr-Linux: what's up dude? |
| 18:21.40 | Dr-Linux | hehe |
| 18:21.51 | starwarez | so, is possible to habe calls with 2 phones behind the SAME nat router? |
| 18:21.58 | justinu | yep |
| 18:22.07 | afrosheen | starwarez: yeah |
| 18:22.21 | LoonaTick | justinu: That's in the zaptel config I assume? |
| 18:22.26 | justinu | yes |
| 18:22.37 | justinu | there's some other things that need to go in there too |
| 18:22.40 | justinu | the channel definitions |
| 18:23.04 | justinu | LoonaTick: http://www.digium.com/downloads/configuring_zaptel.pdf |
| 18:23.15 | afrosheen | channels, grouping, incoming/outgoing volume boost, etc. etc. |
| 18:23.23 | warthawg | a question about agi: does asterisk kill the process when it returns? |
| 18:23.29 | [av]bani | \o> |
| 18:23.30 | LoonaTick | i think i have that one correct, bchan on 1-15,17-31 and dchan on 16 |
| 18:23.30 | [av]bani | <o/ |
| 18:23.38 | justinu | LoonaTick: sounds good |
| 18:23.47 | PakiPenguin | What atas do you guys suggest for connecting ~6 extensions? |
| 18:23.54 | LoonaTick | but I have span=1,1,0,css,hdb3,crc4 |
| 18:24.01 | starwarez | i will remove the nat=yes to check if i have sound... :) |
| 18:24.04 | Dr-Linux | justinu: i setup this >> http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP |
| 18:24.10 | justinu | LoonaTick: it should be "ccs", not "css", imo |
| 18:24.23 | LoonaTick | ccs, my typo, sorry |
| 18:24.36 | Dr-Linux | everything is very fine .. but it doesn't copy created file to /var/spool/asterisk/outgoing/ dir |
| 18:24.54 | justinu | LoonaTick: ah yes, your span = 1,1,0 is correct |
| 18:24.57 | wunderkin | LoonaTick, you need 1,1 if you are taking timing from the line |
| 18:25.06 | wunderkin | i thought there were 2 d chans on an e1 pri |
| 18:25.10 | justinu | no |
| 18:25.27 | afrosheen | starwarez: remove? no, you want it in there..there's no danger with it being there if it isn't needed but not the other way around |
| 18:25.29 | justinu | you can ask for one, but it's kinda pointless |
| 18:25.43 | wunderkin | i have nfi, i just thought i heard it mentioned |
| 18:25.53 | justinu | backup d-chans are useful on NFAS groups only really |
| 18:26.09 | wunderkin | thats what i figured |
| 18:26.15 | LoonaTick | wunderkin: I'm sorry, what does this timing mean? |
| 18:26.35 | justinu | you have to sync your e1 tx source with the rx from the telco |
| 18:26.38 | LoonaTick | i see something with the distance of the other side |
| 18:26.47 | justinu | or else you'll get snap crackle and pop on the line |
| 18:26.53 | LoonaTick | ii see |
| 18:27.19 | LoonaTick | the telco delivered a hdsl modem, from there there's a line to the E1 interface |
| 18:27.40 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 18:27.48 | justinu | yep |
| 18:28.06 | LoonaTick | will it only affect the quality or can it cause the line to not work? |
| 18:28.14 | justinu | generally it will still work |
| 18:28.33 | justinu | but it could be slipping badly enough to cause your d-channel to stay out of service |
| 18:28.37 | Dr-Linux | anyone is using this wakeup call: |
| 18:28.46 | justinu | Dr-Linux: wake up! |
| 18:28.46 | Dr-Linux | <PROTECTED> |
| 18:29.09 | starwarez | here is my lab install :) (lan)--net=192.168.1.x--|nat router|--net=9.9.9.x--|router|---net=1.1.1.x--(*) |
| 18:29.30 | LoonaTick | justinu: Might be a really stupid question, but can you tell me the difference between B and D channels? |
| 18:29.41 | W`gon | starwarez: assuming you don't own 9.9.9.x and 1.1.1.x perhaps you should use a real subnet? :) |
| 18:29.46 | warthawg | according to cnn, the whippet lost at the airport was "carried away by monkeys" |
| 18:29.50 | justinu | b = bearer |
| 18:29.53 | Winkie | LoonaTick: technical or actual? |
| 18:29.56 | PakiPenguin | guys , can someone help me out with fxs atas? i need 6 ports ... any suggestions with atas or some other solution? |
| 18:30.01 | LoonaTick | Winkie: actual |
| 18:30.03 | SpaceBass | arrruuggg this stupid frenchy website won't process my credit card and give me my damn sip account |
| 18:30.04 | justinu | bearer channels bear the audio of the call |
| 18:30.10 | Winkie | LoonaTick: b carries signal, d carries signalling |
| 18:30.11 | justinu | d = data (signalling) all call setup info |
| 18:30.25 | LoonaTick | aah that really explains thanks :) |
| 18:31.28 | justinu | isdn can do freaky stuff like ask for a call to be setup on 4 b-channels at once |
| 18:31.36 | justinu | for wideband video conference, for example |
| 18:31.47 | justinu | they don't have to be carrying audio |
| 18:32.03 | LoonaTick | ah |
| 18:32.25 | justinu | i think there are very few people using features like that though |
| 18:32.37 | Abydos313 | what ports do i need to open for sip users to get sound? i have 5060-5082 udp and 10000-20000 udp forwarded to * server |
| 18:32.52 | LoonaTick | quite old numbers, 64kbps (i think?) per channels "64k should be enough for anyone" :) |
| 18:33.00 | Dr-Linux | justinu: i'm awaking :P |
| 18:33.06 | justinu | 64kbps actually has a purpose |
| 18:33.14 | justinu | it's due to the nyquist frequency |
| 18:33.22 | SpaceBass | Abydos313, that should be it |
| 18:33.31 | justinu | you need 8000 8bit samples/sec to get 0-4000hz frequency responce |
| 18:33.40 | SpaceBass | assuming 10000-20000 is what is defined in your rtp.conf.. (and you can GREATLY narrow that down) |
| 18:34.01 | Abydos313 | then what else can i check? when i connect vpn sound works great, but thru firewall..nada but dialtone |
| 18:34.33 | LoonaTick | i see |
| 18:34.46 | SpaceBass | Abydos313, missed the issue at hand...assuming remote phone cannot connect?...do you have net=yes on that exten? |
| 18:34.56 | justinu | Dr-Linux: hehe |
| 18:35.41 | LoonaTick | that's quite beyond my current education :( |
| 18:35.48 | sivana | is there a background cdr process that kp wrote? Anyone know what I'm talking about? |
| 18:36.25 | justinu | LoonaTick: nyquist theory simple says "you need double the sampling rate for the frequency range you want to sample" |
| 18:36.32 | justinu | (heavily paraphrased) |
| 18:36.36 | Abydos313 | SpaceBass yes |
| 18:36.43 | Dr-Linux | justinu: i'm looking for _Sam-- ;) |
| 18:36.43 | justinu | so if you want 4000hz, you need 8000 samples/sec |
| 18:36.48 | SpaceBass | Abydos313, any audio? like one-way? |
| 18:36.50 | justinu | Dr-Linux: i'm surprised he's not here! |
| 18:36.56 | Abydos313 | nope |
| 18:37.01 | Dr-Linux | justinu: he doesn't start ... but he gets start then never stops :P |
| 18:37.21 | SpaceBass | Abydos313, you might want to try opening TCP ports too...crazy as that sounds, I've had success when the UDP stuff fails...maybe how the firewall processes the requrest or something |
| 18:37.30 | Abydos313 | ok |
| 18:37.40 | Abydos313 | it's a sonicwall.. so maybe it's alittle diff |
| 18:37.53 | LoonaTick | ah that explains a little, thanks :) |
| 18:38.09 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 18:38.17 | SpaceBass | Abydos313, I was helping some poor guy via IM that had the exact same problem with a sonicwall...never figured out what it was |
| 18:38.21 | Abydos313 | which tcp ports? the 5060-5082? |
| 18:38.30 | SpaceBass | VPN helped him, but it changed the MTU and caussed jitter |
| 18:38.37 | SpaceBass | Abydos313, should only need 5060 i think |
| 18:38.40 | Abydos313 | that sucks |
| 18:39.05 | Abydos313 | i could try the dmz but i'm testing on my work machine with vmware :) |
| 18:40.07 | SpaceBass | Abydos313, ahhh it could be that! |
| 18:40.13 | SpaceBass | using bridged or host networking? |
| 18:40.44 | Abydos313 | the default.. the vmware machine has it's own ip etc |
| 18:40.45 | justinu | LoonaTick: 8000samples/sec at 8bits/sample = 64kbps |
| 18:40.52 | Abydos313 | i think that is bridged |
| 18:41.00 | justinu | LoonaTick: and that concludes your basic telephony lesson today |
| 18:41.06 | Abydos313 | haha |
| 18:41.16 | LoonaTick | hehe indeed, thanks |
| 18:41.33 | SpaceBass | arrruuuggg Im starting to think Asterisk hasn't made it to France yet...i just need a damn sip provider and paris DID |
| 18:42.01 | justinu | i think that voxbone is offering paris DIDs |
| 18:42.14 | SpaceBass | eurika! |
| 18:42.17 | SpaceBass | thanks justinu |
| 18:42.27 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 18:42.45 | justinu | np |
| 18:43.20 | *** join/#asterisk Eggplant (i=No@64.233.107.198) |
| 18:43.43 | SpaceBass | justinu, what do you know about voxbone? I dont see pricing anywhere |
| 18:44.26 | justinu | it's not overly expensive |
| 18:44.34 | justinu | 5 bucks a month/did? something like that |
| 18:44.36 | justinu | it's prepaid also |
| 18:45.11 | Eggplant | anyone use astlinux? The listed default login is not working, did they change it between the livecd adn live cf versions? |
| 18:45.45 | starwarez | What is the best sip client for "behind nat client" and "non-nat asterisk" installation |
| 18:47.33 | Juggie | starwarez, the best sip client is common sense. |
| 18:47.38 | jbalcomb | Any good resources out there dealing with SIPAddHeader? (please don't <tilde>docs) |
| 18:47.41 | Juggie | understanding nat can make even the worst configuratino work fine. |
| 18:47.55 | Juggie | i have *<-nat->internet<-nat->clients |
| 18:47.58 | Juggie | and they work perfect |
| 18:49.27 | starwarez | Juggie:I cant find the way to debug the problem... :( |
| 18:50.25 | starwarez | tricky configurations on the sip clients are needed too? |
| 18:50.29 | Dr-Linux | justinu: i was on the phone with a US girl |
| 18:50.56 | LoonaTick | justinu: I have the new configurations; got the first asterisk message, but don't know if it is good or bad: |
| 18:50.56 | LoonaTick | Feb 17 19:50:09 NOTICE[21264]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 18:50.56 | LoonaTick | Feb 17 19:50:10 NOTICE[21264]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 |
| 18:51.07 | justinu | bad |
| 18:51.08 | shido6 | vmware and asterisk, Abydos313 ? |
| 18:51.09 | LoonaTick | I assume bad though |
| 18:51.21 | justinu | HDLC errors indicate your d-channel is not stable |
| 18:51.29 | justinu | Dr-Linux: again? |
| 18:51.35 | Juggie | starwarez, no, no special config is needed on sip clients |
| 18:51.50 | Juggie | do all the work on * and the sip clients wil work easily :) |
| 18:51.59 | LoonaTick | justinu: Ok, does this mean there's a problem at my telco's (or the modem's) side, or at my configuration? |
| 18:52.03 | Dr-Linux | justinu: yes she is in chat, and she wanted to talk to me and she was crying on the phone :S |
| 18:52.08 | Juggie | starwarez, explain your setup and yuor problem. |
| 18:52.19 | justinu | LoonaTick: unknown, you will have to use loopbacks to isolate the problem. |
| 18:52.20 | Juggie | and do it concisely as possible please. |
| 18:52.49 | LoonaTick | great, thanks |
| 18:52.54 | starwarez | Juggie: sure... lemme post it.. |
| 18:53.11 | LoonaTick | (for example I have to put a crossover cable in and let them call each other?) |
| 18:53.45 | justinu | not cross over |
| 18:53.50 | justinu | that's for hooking up ast back to back |
| 18:53.58 | justinu | you need a loopback plug |
| 18:54.08 | justinu | or ask the telco if they can loop up your CSU |
| 18:54.23 | LoonaTick | thanks! |
| 18:55.49 | *** join/#asterisk JCC_ (n=john@207.41.92.131) |
| 18:56.03 | *** join/#asterisk L|NUX (i=linux@203.101.160.218) |
| 18:57.06 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:57.55 | SpaceBass | arrruuggg....this echo is unbarable |
| 18:57.57 | LoonaTick | justinu, correct that the HDSL modem is the DSU, and the card the CSU |
| 18:58.15 | SpaceBass | my father just called and accused me of being drunk b/c I was sluring and studdering b/c all i could hear was myself |
| 18:58.15 | jbalcomb | Any good resources out there dealing with SIPAddHeader as standard functions such as Call-Info, Alert-Info, etc? |
| 18:58.51 | justinu | LoonaTick: HDSL modem is called NIU |
| 18:59.02 | justinu | your E1 card is called CSU/DSU |
| 18:59.08 | jbalcomb | Network Interface Unit |
| 18:59.08 | Qwell[] | Anybody happen to know how I can access feature codes on a nortel meridian, if my phone doesn't have a feature button? heh |
| 18:59.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:59.21 | justinu | but no DSU really, because you're only running channelize voice |
| 18:59.50 | LoonaTick | ah ok |
| 19:00.17 | LoonaTick | and the loopback cable, is that a special cable or a regular cable with some setup? |
| 19:00.17 | Abydos313 | s |
| 19:00.18 | justinu | it's usually a small plug |
| 19:00.21 | justinu | it takes pins 1,2 and connects them to 4,5 |
| 19:00.30 | justinu | you can make your own cable that does it as well |
| 19:00.39 | LoonaTick | ah i think one of those is connected with a plastic thing to the HDSL modem |
| 19:01.05 | justinu | could be, they often come with adtran CSU equipment |
| 19:01.32 | LoonaTick | and with the loopback in instead of the cable to the NIU, it should not give those signal errors anymore? Should it give any other message? |
| 19:01.50 | justinu | your d-channel will not come up to itself |
| 19:02.07 | justinu | what you need to do is plug the looback in to your end of the E1, instead of your E1 card |
| 19:02.21 | justinu | then ask your telco to run BERT patterns on the line to verify the line quality |
| 19:02.51 | starwarez | Juggie/people:here is my setup http://starwarez.no-ip.org/asterisko.txt i think is lohn to paste it here :) |
| 19:02.51 | LoonaTick | ah thank you |
| 19:03.59 | LoonaTick | my telco did some testing with that small plug (labelled: PTT ISDN30 test), said there was no data coming over the line or something like that, so I think the problem is on their side then |
| 19:04.22 | LoonaTick | but when the test block was plugged out he could see that the line was up upto the hdsl modem |
| 19:04.38 | Juggie | starwarez, that doesnt tell me where asterisk is |
| 19:04.45 | Juggie | ah |
| 19:04.46 | Juggie | nm i see |
| 19:05.03 | starwarez | * |
| 19:05.05 | starwarez | :) |
| 19:05.06 | Juggie | yah |
| 19:05.10 | Juggie | so then * has a routeable ip |
| 19:05.30 | Juggie | so the clients are the only ones that dont have routeable ip's. |
| 19:05.37 | Juggie | did you set nat=yes for the clients |
| 19:05.49 | Juggie | so that asterisk uses the ip from the tcpip packet rather then what the client reports |
| 19:06.09 | Juggie | that really should be all you have to do. |
| 19:06.30 | Juggie | does your client hear audio but * not hear the client? |
| 19:06.35 | Juggie | is one direction working? but the other not? |
| 19:06.38 | sdali | Qwell[]: I found the problem - dialing into the FXO port of a Grandstream HT488, and forwarding that call to a VOIP extension, the * and # dtmf digits are not being handled properly. Grandstream is aware of this and have told me to wait for the next firmware upgrade. |
| 19:06.40 | Juggie | whats the situation |
| 19:06.40 | LoonaTick | justinu: Oh one config option I left on the default: rxwink=300. Can this matter? |
| 19:06.44 | starwarez | yes, nat=yes didnt did the job |
| 19:06.59 | Juggie | starwarez, didnt do the job? |
| 19:07.12 | Juggie | starwarez, does any audio path work? |
| 19:07.13 | Qwell[] | sdali: grandstream makes crap... |
| 19:07.19 | sdali | I see. :) |
| 19:07.19 | Qwell[] | sdali: I'd suggest getting another device |
| 19:07.27 | sdali | Thanks. |
| 19:07.32 | starwarez | no sound from any client |
| 19:07.54 | Juggie | can the clients hear *? |
| 19:08.12 | starwarez | yes, the retension music can be heared |
| 19:08.18 | justinu | LoonaTick: winking isn't used on PRI, so it will likely not do anything |
| 19:08.31 | Juggie | starwarez, place a call and do a rtp debug |
| 19:08.36 | LoonaTick | ah ok, thanks |
| 19:08.37 | Juggie | so you can see whats going on with the rtp traffic |
| 19:08.40 | starwarez | ok |
| 19:08.57 | salviadud | i got it workin'!!!! |
| 19:08.59 | kuku5 | anyone having issues with ther cisco phnes? |
| 19:09.00 | salviadud | piweiwiwiewe |
| 19:09.04 | kuku5 | 7960/40 |
| 19:09.16 | LoonaTick | justinu: If my telco tests the line with the loopback block plugged in, I should see data coming in and out, right? |
| 19:10.24 | *** join/#asterisk nain (n=nain@202.125.143.66) |
| 19:11.05 | nain | hi |
| 19:12.34 | justinu | no... you're plugging the line into the loopback plug, which loops their tx signal back to them |
| 19:12.39 | justinu | so you get nothing |
| 19:12.46 | justinu | you shouldn't have anything plugged into your E1 card |
| 19:13.22 | LoonaTick | justinu: I mean the traffic lights on the HDSL modem btw, your statement still applies then? |
| 19:13.52 | *** join/#asterisk Xen^ (i=linux@203.101.164.215) |
| 19:15.31 | DaPrivateer | If anyone can help. I'm on FreeBSD and when compiling I keep getting utils.o: In function `gethostbyname_r': |
| 19:15.31 | DaPrivateer | utils.o(.text+0x3eb): undefined reference to `__h_error' |
| 19:15.54 | justinu | yes, but those lights are probably always going to be lit |
| 19:16.26 | LoonaTick | ok |
| 19:16.47 | LoonaTick | by the way, I have it on span=1,1,0,ccs,hdb3,crc4 now |
| 19:16.54 | LoonaTick | and I don't get the error anymore |
| 19:17.24 | LoonaTick | is that good, or just a stupid idea? |
| 19:17.36 | GerbilWrk | is it possible to setup with a queue, to auto not send the calls to an agent if it's a certain time, or to only send calls to a specific agent at certain times without the agents logging in and out? |
| 19:18.16 | GerbilWrk | I have 7 agents that always need to be in the queue, and one that only needs in it a few hours a day |
| 19:19.56 | starwarez | Juggie here is it: http://starwarez.no-ip.org/asterisko2.txt |
| 19:21.00 | nain | h |
| 19:21.02 | nain | hi |
| 19:21.19 | nain | hello is every body |
| 19:24.07 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 19:24.37 | *** join/#asterisk gammacoder (n=chatzill@207.67.51.233) |
| 19:24.37 | LoonaTick | justinu: and by the way, even though the line still gives a network error |
| 19:25.00 | warthawg | yes |
| 19:25.07 | warthawg | yes |
| 19:25.23 | justinu | LoonaTick: network error? |
| 19:26.00 | LoonaTick | justinu: that's when i call the phone number |
| 19:26.30 | justinu | LoonaTick: pri show status? |
| 19:26.45 | justinu | er pri show span 1 |
| 19:27.10 | LoonaTick | Primary D-channel: 16 |
| 19:27.10 | LoonaTick | Status: Provisioned, Down, Active |
| 19:27.29 | justinu | so that's why |
| 19:27.31 | justinu | no d-channel |
| 19:28.02 | nain | Can any body now any Predictive Dialer for Asterisk ? |
| 19:28.14 | starwarez | Juggie: were you able to see the sip log? |
| 19:28.52 | LoonaTick | justinu: Ah thanks, didn't know that command, does this mean the problem is at the telco's side? |
| 19:28.57 | stoffell | nain: try here, http://www.voip-info.org/wiki/view/Predictive+dialer |
| 19:29.38 | iCEBrkr | Werd. |
| 19:30.21 | LoonaTick | justinu: (or that the zaptel.conf is configured wrong?) |
| 19:31.05 | nain | Stoffell: there is gnudialler for linux that is open source i am looking for windows based dialer which can be configured with asterisk |
| 19:31.38 | justinu | LoonaTick: unknown |
| 19:31.43 | stoffell | nain, there is one on that page, but don't expect it to be open source.. |
| 19:31.45 | justinu | your zaptel.conf is probably ok |
| 19:32.16 | nain | i think that is sinedialer ... and that is commercial can't afford |
| 19:32.33 | LoonaTick | ok, thank you *very* much, i will call the telco and check out their opinions :) |
| 19:33.13 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:33.49 | warthawg | you guys sound like a bunch of phreaks |
| 19:34.06 | Abydos313 | phreaking went out years ago :) |
| 19:34.06 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 19:34.07 | LoonaTick | warthawg: Thanks for sharing |
| 19:34.14 | warthawg | heh |
| 19:34.30 | nain | Any Open Source Predictive Dialer for Windows ?????? |
| 19:34.31 | warthawg | i can't understand 3/4 of what is said here, it's all telephony talk |
| 19:34.52 | Abydos313 | you need two books. voip for dummies and the asterisk book..haha |
| 19:35.02 | warthawg | i got the asterisk book |
| 19:35.09 | warthawg | i guess i need to the voip one :) |
| 19:35.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:35.15 | Abydos313 | i just bought it myself two days ago. i want to learn |
| 19:35.39 | Abydos313 | i actually was unable to find a voip for dummies at borders |
| 19:38.11 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 19:38.24 | starwarez | buds... here is the log: http://starwarez.no-ip.org/aster.txt |
| 19:40.32 | GerbilWrk | has anyone found a way to imporve QoS with Teliax? |
| 19:40.49 | iCEBrkr | nain: You can buy my dialer :P |
| 19:41.03 | starwarez | http://starwarez.no-ip.org/asterisko2.txt------> full log here |
| 19:41.09 | oogle | has anyone ever dealth with voip reach? |
| 19:41.14 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 19:41.15 | iCEBrkr | starwarez: here man use this |
| 19:41.16 | iCEBrkr | ~pb |
| 19:41.17 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:41.31 | *** join/#asterisk ^Accie^ (n=chrake@void.leapfrog.se) |
| 19:42.22 | ^Accie^ | anyone who'd like to help a newbie... umm... managed to set up asterisk and can dial the demo from my snom190 but I can't seem to register it with asterisk even though I'm using the same username/password stated in sip.conf |
| 19:42.40 | ^Accie^ | sip debug keeps telling me 'registration failed' |
| 19:43.14 | remiss | w00t |
| 19:44.16 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 19:44.45 | starwarez | what means "SIP/2.0 488 Not acceptable here"???? |
| 19:45.11 | justinu | codec problem |
| 19:46.06 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:46.23 | starwarez | justinu: without nat it works fine that phone... |
| 19:48.57 | justinu | generally 488 is telling you that the two UAs could not find a compatible codec |
| 19:49.28 | _Sam-- | trixter...you stole someone elses idea |
| 19:49.42 | _Sam-- | there was a guy here already doing the free toll free terminations and paying people |
| 19:49.45 | _Sam-- | i forget who it was |
| 19:50.39 | _Sam-- | you thief you |
| 19:52.05 | nain | Can any one know about windows Based Open source Predictive Dialer |
| 19:52.33 | starwarez | activating nat on clients on sip .conf i get: the following message: |
| 19:52.35 | starwarez | <PROTECTED> |
| 19:54.03 | _Sam-- | nain: nobody knows because one isnt there |
| 19:54.24 | _Sam-- | i could be wrong. |
| 19:54.35 | justinu | starwarez: then your phone isn't sending the right SDP message |
| 19:54.45 | justinu | pastebin the entire invite |
| 19:55.30 | starwarez | justinu: you mean the sip debug? |
| 19:55.39 | [av]bani | \o/ |
| 19:56.09 | *** join/#asterisk backblue (n=moo@87-196-46-49.net.novis.pt) |
| 19:56.12 | _Sam-- | nain: class 'default', on IAX2/66.250.69.13:4569-3 |
| 19:56.14 | _Sam-- | er |
| 19:56.19 | _Sam-- | nain: http://www.voip-info.org/wiki/index.php?page=Predictive%20dialer |
| 19:56.19 | justinu | starwarez: yes |
| 19:56.50 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 19:57.09 | nain | Sam---: Sorry i have looked for it but it has no windows based Free Dialer |
| 19:58.21 | remiss | bah... i had it working and i managed to remove the configuration :( |
| 19:58.32 | starwarez | justinu: here is it |
| 19:58.44 | starwarez | sorry, http://starwarez.no-ip.org/lastcall.txt |
| 20:01.03 | justinu | starwarez: your phone is whacked... it sends the sdp on the first invite, but not the 2nd |
| 20:01.41 | starwarez | justinu: bad configuration? |
| 20:01.42 | trixter | _Sam-- trixter...you stole someone elses idea -- I started investigating this back in 1997 :P |
| 20:01.51 | justinu | starwarez: possible |
| 20:04.36 | starwarez | justinu: the weird thig is: if i remove the nat from the routers, works like a charm |
| 20:05.18 | Renacor | is there a way for me to send a call to VM after it's been sitting in a queue for 10 min? |
| 20:05.35 | Qwell[] | Renacor: Sure, put a timeout on the queue, and have the next line be voicemail |
| 20:07.09 | _Sam-- | trixter: took you long enough to implement! |
| 20:07.15 | justinu | starwarez: perhaps without nat asterisk doesn't send the proxy auth required? |
| 20:07.17 | _Sam-- | im just busting your balls..i hope you know |
| 20:07.27 | _Sam-- | but there is someone on this channel doing the same thing already |
| 20:07.52 | [av]bani | _Sam-- got any phones other than gxp2k? |
| 20:08.16 | robin_z | ahh, at last! ... updated software for my GXP 2000 |
| 20:08.16 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 20:08.17 | _Sam-- | at this place the only other phone i have here is a utstarcomm wifi phone |
| 20:08.19 | mikefoo | hey guys if I was to make a dialplan, for when someone dials in, hits a certain extension it connects to someone, its still going through my pbx yes? is there a way I can conenct him to another number and disconnect from my pbx? |
| 20:08.38 | robin_z | IronHelix: you there? |
| 20:08.43 | justinu | bani: i'm here with the snom today |
| 20:08.49 | justinu | still want me to listen to the tones? |
| 20:08.49 | [av]bani | justinu: yay! er, boo! |
| 20:08.50 | websae | anyone know of a good DID provider? I tried signing up for didx.org---but never was sent the confirmation email to complete registration---any other suggestions anyone ? thanks |
| 20:08.56 | [av]bani | justinu: yes |
| 20:08.59 | justinu | k, i'll let you know |
| 20:09.07 | [av]bani | make an extension which goes 1,Congestion() |
| 20:09.07 | _Sam-- | robin_z : where is the new firmware? |
| 20:09.18 | [av]bani | and one which goes 1,Playtones(dial) n,Wait(60) |
| 20:09.34 | robin_z | _Sam--: sorry, I was using sarcasm as a low form of wit. |
| 20:09.42 | _Sam-- | someone has the new firmware already |
| 20:09.43 | _Sam-- | X-rob does |
| 20:09.46 | _Sam-- | but its not public |
| 20:09.48 | robin_z | not me ... |
| 20:09.51 | robin_z | oh good. |
| 20:09.52 | _Sam-- | and it fixes , somewhat, your display. |
| 20:09.53 | [av]bani | and one which goes 1,Playtones(busy) |
| 20:10.01 | _Sam-- | he said "it takes longer before the display breaks" |
| 20:10.59 | websae | anyone have a good DID provider? |
| 20:10.59 | robin_z | well, thats an improvement already |
| 20:10.59 | _Sam-- | maybe they will release something soon, who knows |
| 20:10.59 | _Sam-- | he said he has 1.0.2.9 |
| 20:11.06 | robin_z | grr |
| 20:11.07 | afrosheen | websae: commpartners is good |
| 20:11.13 | [av]bani | justinu: the snom native congestion sound is really bizarre. playtones(busy) sounds exactly like what i get from the PSTN |
| 20:11.21 | robin_z | anyway ... I solved my SIP login problem .... |
| 20:11.41 | robin_z | the remote objected to "astersik PBX" as the user agent ... |
| 20:11.50 | robin_z | I was setting it in the sip conf for that host ... |
| 20:12.13 | robin_z | but, it was handing off to a sip proxy thing .. so I had to set it in [general] or the [sip-proxy-out] |
| 20:12.56 | justinu | i'll check it out |
| 20:12.59 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 20:13.17 | *** join/#asterisk mischko (n=Scott@cvo-cr1-200-242.peak.org) |
| 20:13.30 | Abydos313 | would you guys recommend latest tar.gz or cvs setup for home users |
| 20:13.49 | justinu | tarball |
| 20:14.04 | Abydos313 | any reason? |
| 20:14.12 | justinu | unless you're participating in development, you don't want to run the code out of SVN |
| 20:14.20 | justinu | it could be unstable |
| 20:14.25 | justinu | in flux, etc. |
| 20:14.26 | Abydos313 | ok thx |
| 20:14.32 | Renacor | Qwell: Do you have an example of this? |
| 20:14.37 | Abydos313 | even if you sync with stable? |
| 20:15.54 | justinu | you can check out a stable branch of svn i suppose |
| 20:15.59 | justinu | but why not just get the tarball? |
| 20:16.03 | mischko | Small company about 5-15 phones. Only 2 incoming lines currently. We're thinking of upgrading to a VOIP phone system in-house with Askerisk/Linux to be our PBX and add a couple lines. Later we want to go to straight VOIP but only want to buy VOIP phones one time. Suggestions? |
| 20:16.38 | [av]bani | do more research. |
| 20:16.41 | Abydos313 | ok. i did do the latest tarballs, just curious. thanks for input |
| 20:16.48 | justinu | np |
| 20:18.29 | stoffell | mischko, try a few pones, buy 1 of each you think suits the needs, ... |
| 20:18.46 | stoffell | phones :) |
| 20:19.15 | mischko | Are there any appliance solutions that I should be looking at rather than running my own Asterisk? |
| 20:19.27 | [av]bani | should be? depends on your budget. |
| 20:19.29 | _Sam-- | you could use a hosted PBX and just have your phones connect to it |
| 20:19.36 | _Sam-- | then you wouldnt need your own * |
| 20:19.49 | [av]bani | yeah, use SamSoft(tm) Hosted PBX Solutions(tm) |
| 20:20.01 | stoffell | mischko, depends what area you are, and what you want to spend, and what you know of linux :) |
| 20:20.02 | mischko | Vonnage does not offer us the ability to keep our existing phone number. |
| 20:20.03 | mikefoo | hey guys if I was to make a dialplan, for when someone dials in, hits a certain extension it connects to someone, its still going through my pbx yes? is there a way I can conenct him to another number and disconnect from my pbx? |
| 20:20.19 | [av]bani | mischko: if you arent a linux guru, then * is not for you |
| 20:20.37 | mischko | I'm quite familiar with linux but not phone systems. I' |
| 20:20.46 | mischko | ve run servers on Linux no problem. |
| 20:21.05 | [av]bani | can you compile applications from source, apply patches, edit config files? |
| 20:21.10 | [av]bani | handle iptables? |
| 20:21.12 | mischko | yes. |
| 20:21.14 | *** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net) |
| 20:21.30 | [av]bani | then the only limit is your budget |
| 20:21.31 | mikefoo | mischko: vonage doesn't do lnp's? |
| 20:21.33 | stoffell | mischko, then you could try a test system with asterisk@home, and see how it goes |
| 20:21.47 | [av]bani | your budget will dictate what you can and cannot do |
| 20:21.55 | mischko | Vonnage doesn't have a local presence in this part of Oregon so we couldn't keep our local phone number. |
| 20:22.11 | mischko | ??asterisk@home |
| 20:22.14 | [av]bani | you havent stated your requirements though, so we dont know what to recommend |
| 20:22.18 | austinnichols101 | asterisk@home is a good way to kick the tires without a huge investment in time |
| 20:22.33 | *** join/#asterisk cassio (n=cassio@c91133b9.rjo.virtua.com.br) |
| 20:22.44 | austinnichols101 | and [av]bani is right - budget is the major deciding factor on phones |
| 20:22.56 | _Sam-- | what is wrong with hosted pbx is you are talking about 5 phones or so |
| 20:23.00 | stoffell | mischko, search on google, it's a good way to get to learn the basics of telephony and asterisk |
| 20:23.03 | _Sam-- | other than, you dont have to worry about learning * |
| 20:23.14 | mischko | hosted pbx would be fine if we can keep our local phone number. |
| 20:23.23 | [av]bani | nobody serves corvallis |
| 20:23.37 | Qwell[] | Why go to voip? |
| 20:23.44 | cassio | guys, I have 5 broadvoice lines, but I am getting this error on 2 lines, the configs on sip.conf are the same, why is this happening? http://pastebin.com/560087 |
| 20:23.44 | [av]bani | cuz its neat0 |
| 20:23.45 | Qwell[] | Just get some analog cards, and use the lines you have |
| 20:23.46 | mischko | Local ISP will be adding VOIP in about 6 months. |
| 20:24.04 | austinnichols101 | Qwell: cheap calls |
| 20:24.08 | justinu | 31337 |
| 20:24.13 | mischko | We don't like our existing Panasonic phone system and people call in getting busy signals. |
| 20:24.22 | mischko | We can't add new lines to this system. |
| 20:24.26 | mischko | So we're looking at options. |
| 20:24.29 | Qwell[] | So move to *, and get more lines... |
| 20:24.32 | Qwell[] | Don't need voip |
| 20:25.07 | JCC_ | For what its worth, I set up an A@H box, turned off long dist calling, added voxee and a Sipura 3000 on an older 900MHz Pent III - total cost about $125 (the box was in the closet doing nothing) |
| 20:25.11 | [av]bani | hmm.. jared likes his 7960 better thant he 7970... |
| 20:25.16 | mischko | Qwell, hadn't thought of that. |
| 20:25.27 | austinnichols101 | I really like the 7960s |
| 20:25.41 | Qwell[] | [av]bani: eh? |
| 20:25.43 | stoffell | mischko, indeed, like qwell says, you could do that, and in the future try voip phones, you can mix it all |
| 20:25.46 | Qwell[] | 7970 is FAR better than a 7960 |
| 20:25.49 | austinnichols101 | but they're $300-$350 each |
| 20:26.02 | [av]bani | Qwell[]: jared mauch. says he likes the 7960 better... weird |
| 20:26.10 | austinnichols101 | Qwell: yeah, I have phone 7970-envy right now |
| 20:26.11 | Qwell[] | he's dumb then |
| 20:26.18 | [av]bani | he has problems with sccp |
| 20:26.20 | [av]bani | might be why |
| 20:26.21 | Qwell[] | 7970 does everything the 7960 does...in color |
| 20:26.27 | Qwell[] | Then it's his fault, not the phones |
| 20:26.33 | Qwell[] | tell him I'll buy his 7970 for $100 |
| 20:26.56 | austinnichols101 | I'll trade him for a 7960 :) |
| 20:27.18 | austinnichols101 | ~seen opsys |
| 20:27.30 | jbot | opsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 4d 14h 46m 28s ago, saying: 'betaboi" true'. |
| 20:27.30 | [av]bani | the avaya phones look nice |
| 20:27.31 | mischko | What phones work best with *? |
| 20:27.44 | [av]bani | avaya 4630sw... |
| 20:27.48 | austinnichols101 | somebody was on here yesterday raving about the mitels |
| 20:27.48 | mischko | I know it's supposed to be almost hardware agnostic but ? |
| 20:27.59 | [av]bani | mischko: phones which speak SIP |
| 20:27.59 | Qwell[] | austinnichols101: probably Juggie |
| 20:28.21 | mikefoo | hey guys if I was to make a dialplan, for when someone dials in, hits a certain extension it connects to someone, its still going through my pbx yes? is there a way I can conenct him to another number and disconnect from my pbx? |
| 20:28.46 | austinnichols101 | Qwell: sounds right. |
| 20:28.50 | mikefoo | what I am trying to accomplish is connecting two people and me getting of the way. |
| 20:28.57 | austinnichols101 | Last thing I bought from Mitel was a dialer |
| 20:29.06 | _Sam-- | has anyone heard of ESI phones? i was at a doctors office and was checking them out, they seem nice, but have no idea of they are |
| 20:29.15 | [av]bani | never heard of them |
| 20:29.24 | _Sam-- | search google esi phones |
| 20:29.27 | _Sam-- | first result |
| 20:29.34 | [av]bani | http://search.ebay.com/search/search.dll?sofocus=unknown&sbrftog=1&catref=C6&fstype=1&from=R10&satitle=esi+phone&sacat=-1%26catref%3DC6&bs=Search&fsop=1%26fsoo%3D1&coaction=compare&copagenum=1&coentrypage=search&sargn=-1%26saslc%3D2&sadis=200&fpos=97526&ftrt=1&ftrv=1&saprclo=&saprchi= ? |
| 20:30.12 | _Sam-- | i think maybe their phone are only usable with their own servers |
| 20:30.14 | austinnichols101 | anyone tried the allworx phones? |
| 20:30.30 | [av]bani | they dont mention sip at all |
| 20:30.45 | austinnichols101 | found a nice comparison chart from them (the cisco retail shows way off): http://www.allworx.com/XQ/ASP/p.3406/QX/pdfs/PhoneComparisonChart.pdf |
| 20:30.46 | _Sam-- | this one says IP: pbx? |
| 20:30.46 | _Sam-- | <austinnichols101> Qwell: sounds right. |
| 20:30.47 | _Sam-- | er |
| 20:30.53 | _Sam-- | http://www.esi-estech.com/products/systems/phones/remote/ |
| 20:31.02 | mischko | Anything on here: http://www.iptel.org/info/products/sipphones.php that I should definitely stay away from in hard phones? |
| 20:31.03 | [av]bani | they talk about integrating esi phones with microsft outlook. |
| 20:31.13 | Qwell[] | mischko: grandstream |
| 20:31.24 | austinnichols101 | sam: sounds right about what? |
| 20:31.38 | _Sam-- | that was a mess up copy/paste |
| 20:31.41 | _Sam-- | somehow that was in my buffer |
| 20:32.03 | stoffell | a good-looking phone: thomson st2030, but doesn't support MWI and BLF yet in SIP image.. |
| 20:32.06 | [av]bani | _Sam--: looks like no sip |
| 20:32.14 | [av]bani | _Sam--: looks proprietary, like mitel |
| 20:32.27 | _Sam-- | figured...it was a really nice office system they had. |
| 20:32.28 | mischko | So avoid mitel? |
| 20:32.45 | [av]bani | mischko: mitel makes some SIP phones, but mitel is pretty expensive for what you get |
| 20:32.50 | Qwell[] | mischko: mitel is supposed to be really good |
| 20:33.06 | [av]bani | mischko: most mitel phones are either analogue or speak proprietary mitel protocol |
| 20:33.31 | afrosheen | is it safe to mix zaptel versions, i.e. the latest zaptel 1.2.4 with asterisk 1.2.0? |
| 20:33.33 | Juggie | wrong. |
| 20:33.33 | stoffell | mischko, try ThomsonST2030, Aastra, Polycom, .. |
| 20:33.36 | mischko | "proprietary protocol" means somethings it doesn't do SIP? |
| 20:33.37 | Juggie | mitel makes all sip phones now. |
| 20:33.41 | [av]bani | ~phones |
| 20:33.43 | jbot | it has been said that phones is at http://bani.anime.net/phones/ |
| 20:33.47 | Juggie | all their phones support minet/sip |
| 20:33.53 | Juggie | anything 5215+ |
| 20:34.17 | Juggie | 5215/5220 require the dual mode model (dual firmware) |
| 20:34.18 | _Sam-- | he's right |
| 20:34.21 | _Sam-- | "Support for SIP and MiNET Protocols " |
| 20:34.21 | [av]bani | Juggie: mitel is really expensive for what you get. cisco and others are better bargain |
| 20:34.22 | Juggie | and everything else supports sip nateivly |
| 20:34.28 | Juggie | av, cisco sucks |
| 20:34.31 | mischko | [av]bani, ok. |
| 20:34.34 | Juggie | the phones are impossible to manage |
| 20:34.50 | [av]bani | Juggie: mitel design their phones to be integrated into their pbx systems. friend fo mine is a mitel reseller |
| 20:34.51 | mischko | Who "doesn't suck" and is reasonably priced? |
| 20:35.00 | Juggie | i mean common, editing like 5 files to update the firmware? |
| 20:35.00 | oogle | don't use the 12-button mitels |
| 20:35.00 | Qwell[] | mischko: polycom |
| 20:35.11 | [av]bani | Juggie: polycom is the same way |
| 20:35.11 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 20:35.12 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 20:35.21 | Juggie | av, i have 3 mitel phones next to me |
| 20:35.25 | Juggie | one is minet |
| 20:35.29 | Juggie | two are on asterisk |
| 20:35.30 | mischko | [av]bani, "same way" = "hard to manage" |
| 20:35.33 | Juggie | and they work perfect |
| 20:35.35 | oogle | the 24-button mitel is pretty good, and allows dhcp-tftp provisioning |
| 20:35.35 | mischko | ? |
| 20:35.41 | [av]bani | mischko: yes |
| 20:35.41 | Qwell[] | Juggie: When are you gonna send me some of the "other stuff" sitting next to you? :p |
| 20:35.49 | mischko | [av]bani, who's easy to manage? |
| 20:35.52 | mikefoo | is there a reason why I shouldn't use asterisk@home on a production level? |
| 20:36.07 | Qwell[] | mikefoo: Because it's complete junk, primarilly |
| 20:36.14 | Juggie | heh. |
| 20:36.17 | [av]bani | mischko: linksys, snom, grandstream |
| 20:36.17 | austinnichols101 | omg: the question! |
| 20:36.20 | oogle | what's wrong with asterisk@home? |
| 20:36.21 | Juggie | thats the big thing yes :) |
| 20:36.22 | [av]bani | mischko: not that i would recommend any of those |
| 20:36.27 | oogle | i've never used it myself, j/w |
| 20:36.40 | mikefoo | hah, ahh ok didn't realize that, thought it was just a dumbed down version. |
| 20:36.49 | Juggie | mitel phones support http for config files/firmware |
| 20:36.54 | mischko | [av]bani, what would you recommend? |
| 20:36.55 | Juggie | which removes havnig to run tftp shit |
| 20:36.56 | afrosheen | from what I've seen on the more expensive snom's, they have decent management built in |
| 20:36.58 | stoffell | [av]bani; you're right, but 'what' would you recommend? |
| 20:37.01 | [av]bani | mischko: what are your requirements and budget? |
| 20:37.14 | afrosheen | but we're 100% polycom here |
| 20:37.14 | [av]bani | stoffell: depends. define your budget and requirements |
| 20:37.15 | Juggie | WHICH also, ets you use apache rewrite to dynamically create your config files |
| 20:37.18 | Juggie | which like php or something |
| 20:37.24 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 20:37.26 | SplasPood | What are people generally using /w 1.2.X to hand LCR? Generally custom development, or is there something off the shelf that's populaR? |
| 20:37.26 | SpaceBass | hey again |
| 20:37.27 | [av]bani | afrosheen: snom have nice webadmin, but the phone UI is poo |
| 20:37.30 | stoffell | [av]bani, when that's not an issue.. (budget) |
| 20:37.32 | SplasPood | s/hand/handle |
| 20:37.34 | mischko | [av]bani, requirements as in features? i.e. call waiting or forwarding? |
| 20:37.42 | _Sam-- | Juggie : are you making fun of [av]bani's php provisioner? :) |
| 20:37.44 | Renacor | so to send a call to vm after it's been sitting in a queue for a 2 minutes, I would have to put timeout=2 in the queues.conf for that queue and then a priority after the call is answered in the extensions.conf for that queue? |
| 20:37.57 | [av]bani | mischko: # of lines, touchscreen, backlight, speakerphone, etc |
| 20:37.57 | Juggie | no its a good thing |
| 20:37.58 | Qwell[] | more like 120 |
| 20:38.06 | Juggie | who wants to have a million flat files if one php script can generate them all |
| 20:38.07 | SpaceBass | hey folks |
| 20:38.10 | mikefoo | its seconds not minutes |
| 20:38.14 | oogle | hello SpaceBass |
| 20:38.14 | SpaceBass | need some assistance configuring Voxbone as a trunk |
| 20:38.15 | Juggie | that being said, i have a php tftp server too someone wrote. |
| 20:38.20 | [av]bani | Juggie: i have a php script to do that for snom/sipura/grandstream |
| 20:38.29 | oogle | people write the wackiest stuff in php |
| 20:38.31 | [av]bani | Juggie: and i will be adding polycom when i get time |
| 20:38.34 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:38.43 | Juggie | av, the one i have is was written by nuxi |
| 20:38.52 | [av]bani | Juggie: i wrote mine myself |
| 20:38.54 | Juggie | you wrote a tftp php server? |
| 20:38.59 | SplasPood | Juggie: I'd actually be interested in seeing that tftp server.. |
| 20:39.02 | harryvv | skype was just discussed on cnn right now. Seems skype calls are 128 bit encryption. |
| 20:39.07 | [av]bani | no, i wrote a php autoprovisioner for snom/sipura/grandstream |
| 20:39.09 | austinnichols101 | the aastra is pretty good for a $125-$150 phone |
| 20:39.24 | stoffell | [av]bani; a phone that meets 'BLF/MWI/distinctive ring through SIP header' and min. 2 lines, what is a good one? |
| 20:39.24 | austinnichols101 | 9112 or 9133 |
| 20:39.32 | Renacor | anybody got a good example on how to send a call to vm after it's been sitting in a queue for 2 min? |
| 20:39.32 | harryvv | aastra are nice looking design |
| 20:39.36 | Qwell[] | stoffell: cisco |
| 20:39.41 | [av]bani | take a fresh out of the box factory reset phone, plug it in and it autoconfigures totally, including creating sip extensions for astersik and reloading them |
| 20:39.50 | austinnichols101 | harryw: and they're heavy enough that you can club an intruder with the handset |
| 20:39.55 | [av]bani | stoffell: snom or cisco |
| 20:39.56 | harryvv | Renacor u running a call center? |
| 20:40.02 | stoffell | Qwell, the 7970? |
| 20:40.12 | Juggie | http://eder.us/projects/ <- php tftp server |
| 20:40.13 | Qwell[] | stoffell: 7940 and 7960 also support those |
| 20:40.16 | stoffell | [av]bani, snom much better then polycom |
| 20:40.18 | Qwell[] | I don't know about the rest |
| 20:40.19 | Renacor | harryvv: yep |
| 20:40.20 | [av]bani | i dont use tftp at all |
| 20:40.25 | stoffell | ok Qwell, tnx |
| 20:40.30 | harryvv | austinnichols101 :) |
| 20:40.31 | [av]bani | stoffell: for blf/mwi/distinctive ring, yes.. |
| 20:40.45 | stoffell | ok, tnx very much.. |
| 20:40.46 | jontow | renacor; set the maximum in-queue time in queues.conf to 2mins |
| 20:40.58 | Juggie | its a shame res_php was never finished |
| 20:41.02 | SplasPood | Juggie: danke |
| 20:41.12 | SplasPood | so no one here does least cost routing? |
| 20:41.34 | [av]bani | SuitcaseAvail() |
| 20:41.44 | Mavvie | for zap-only interfaces of course. |
| 20:41.54 | SplasPood | stoffell: Polycom. |
| 20:41.58 | mikefoo | if I wanted to connect two parties, as in one person calls in to a dial plan, hits a prompts, calls another, I want to do this then disconnect myself from the loop, this possible? |
| 20:42.12 | [av]bani | SplasPood: stoffell might not like the blf issue |
| 20:42.27 | stoffell | SplasPood, do they also support distinctive ring via sip-header? |
| 20:42.41 | stoffell | i love issues ;) |
| 20:42.42 | SplasPood | stoffell: yes |
| 20:42.54 | SplasPood | [av]bani: what issue in particular? |
| 20:42.55 | [av]bani | stoffell: snom supports arbitrary wav URL in the sip ring header. so you can have litterally millions of distinctive rings |
| 20:43.10 | [av]bani | SplasPood: max 7 buddy limit. kind of takes the piss out of the sidecars :) |
| 20:43.13 | stoffell | nice |
| 20:43.28 | SplasPood | [av]: Limit? There isn't any limit that I've encountered... |
| 20:43.36 | [av]bani | SplasPood: you dont have a sidecar then |
| 20:43.39 | Renacor | jontow: how do you set that? |
| 20:43.40 | SplasPood | IP601 + Attendent Console |
| 20:43.51 | jontow | look at the example queues.conf |
| 20:43.56 | Renacor | jontow: thanks |
| 20:43.58 | jontow | you can either do it that way, or in the Queue() application |
| 20:43.59 | [av]bani | SplasPood: you can only blf 7 extensions max on polycom, its a hard firmware limit. |
| 20:44.08 | websae | any wholesale providers in here? |
| 20:44.09 | [av]bani | SplasPood: its a limit a _lot_ of people complain about |
| 20:44.25 | SplasPood | [av]: Wait I'm confused.. They sell an add-on for the 601 that DEF has more than 7 entries... |
| 20:44.26 | Juggie | look at the mitel 5235 |
| 20:44.40 | [av]bani | SplasPood: its done via a different mechanism than * uses |
| 20:44.56 | Juggie | http://www.mitel.com/DocController?documentId=14886&c=9512&sc=9517 |
| 20:45.00 | [av]bani | SplasPood: snom, cisco, everyone else uses the mechanism * currently supports. polycom does not. |
| 20:45.14 | SplasPood | [av]bani: hints ? |
| 20:45.15 | _Sam-- | there was a just thing in the asterisk-users about it...bani is right |
| 20:45.23 | _Sam-- | it makes the sidecar thing basically useless |
| 20:45.27 | [av]bani | SplasPood: its partly polycom's fault and partly * doesnt support the sip extension polycom wants |
| 20:45.29 | Juggie | http://www.mitel.com/resources/5235SilverBezelforPCv24.swf |
| 20:45.33 | *** join/#asterisk DrData (n=michael@p54B24BCE.dip.t-dialin.net) |
| 20:45.42 | [av]bani | SplasPood: but theres no good reason for polycom to arbitrarily limit buddy watch to 7 |
| 20:45.53 | MstlyHrmls | [av]bani: dumb question, what mechanism does everyone else use with * for blf? |
| 20:45.58 | [av]bani | SplasPood: because it limits max blf with current * to 7 |
| 20:45.59 | SplasPood | [av]bani: why isn't this on the wiki? |
| 20:46.03 | DrData | is it possible to register asterisk AND a softphone to e.g. sipgate? |
| 20:46.06 | [av]bani | SplasPood: it is afaik |
| 20:46.18 | Qwell[] | DrData: No. Register the softphone to *, then * to the itsp |
| 20:46.19 | Juggie | Qwell, http://www.mitel.com/resources/5235SilverBezelforPCv24.swf |
| 20:46.23 | [av]bani | SplasPood: go ahead and try to blf > 7 lines with ip601 and *. you cant do it. |
| 20:46.25 | stoffell | to me it's new and interesting info :) |
| 20:46.31 | _Sam-- | DrData : you cant have two devices register with the same credentials to the same place |
| 20:46.43 | SplasPood | [av]: don't have one handy to test with.. I'd be all over it if I did. |
| 20:46.52 | Qwell[] | _Sam--: You can |
| 20:46.53 | remiss | w00t, w00t, w00t :D |
| 20:46.58 | _Sam-- | well you can...but only one will work |
| 20:46.58 | remiss | i got it working :) |
| 20:47.02 | SplasPood | [av]bani: can you help me find documentation about this? |
| 20:47.02 | [av]bani | SplasPood: its a common complaint htough, you can find threads on voxilla, asterisk-users and other places about it |
| 20:47.07 | DrData | _Sam--, credentials == user/pw? |
| 20:47.11 | [av]bani | SplasPood: about as retarded as polycom's firmware policy |
| 20:47.28 | _Sam-- | DrData : i will let the real experts help you |
| 20:47.30 | _Sam-- | i am just a novice |
| 20:47.48 | austinnichols101 | [av]bani: limits *should* be zero, exactly one or infinite. 7 is a stupid limit (should be infinite) |
| 20:47.55 | SplasPood | [av]bani: I'm having a lot of trouble finding any mention of this limitation |
| 20:48.09 | [av]bani | austinnichols101: "should" and "polycom" ... heh |
| 20:48.27 | Qwell[] | Juggie: Not loading |
| 20:48.40 | austinnichols101 | [av]bani: that's really one of my pet peeves in programming... |
| 20:48.45 | [av]bani | SplasPood: http://lists.digium.com/pipermail/asterisk-users/2006-February/146983.html |
| 20:48.46 | austinnichols101 | pissed me off |
| 20:49.07 | [av]bani | austinnichols101: blame management |
| 20:49.22 | MstlyHrmls | austinnichols101: the problem is "infinite" and embedded systems don't play well together. I agree that 7 is a stupid limitation, but "infinite" is unreasonable as well |
| 20:49.26 | mischko | [av]bani, we're using these Panasonic cordless units now. We don't use the answering service. Something similar would be great. http://www2.panasonic.com/webapp/wcs/stores/servlet/vModelDetail?storeId=15001&catalogId=13401&itemId=62849&cacheProgram=11002&cachePartner=7000000000000005702&surfModel=KX-TG2740S&catGroupId=25041&surfCategory=Expandable%20Systems&displayTab=O |
| 20:49.28 | austinnichols101 | I found it (Zero-One-Infinity Rule): http://www.catb.org/jargon/html/Z/Zero-One-Infinity-Rule.html |
| 20:50.00 | [av]bani | MstlyHrmls: "as much as the hardware will support" |
| 20:50.11 | [av]bani | MstlyHrmls: usually its more work to put _in_ a limit, than to leave it open |
| 20:50.27 | MstlyHrmls | [av]bani: right, which should be more than 7 but less than infinity :-) |
| 20:50.35 | stoffell | hm, I can live with limit of 7, if one needs more, one can use snom.. but blf/mwi AND dist.ring in a polycom, sounds great |
| 20:50.37 | *** join/#asterisk xyklopz (n=xyklopzi@216-91-89-21.biltmorecomm.com) |
| 20:50.45 | SpaceBass | anyone using voxbone? |
| 20:51.07 | [av]bani | stoffell: configuring polycom is poo though |
| 20:51.07 | austinnichols101 | MstlyHrmls: but the limit should be the hardware itself, not the program |
| 20:51.09 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 20:51.10 | SpaceBass | I dont totally get the concept of the URL stuff...they want to forward my DID to an exten@myIP ... |
| 20:51.24 | Qwell[] | austinnichols101: Would you rather it not work at all, or limit it to a reasonable number? |
| 20:51.29 | [av]bani | at least it is now, wait till i get my autoprovisioners working for polycom ;) |
| 20:51.31 | Qwell[] | "address space and memory permitting" |
| 20:51.38 | SplasPood | [av]bani: ugh.. I never tested to that many, I suppose... |
| 20:51.48 | [av]bani | SplasPood: kind of makes the sidecar pointless with * |
| 20:51.52 | MstlyHrmls | austinnichols101: yes, but you don't want non-Call features interfering with the phones primary purpose: to make calls |
| 20:51.54 | austinnichols101 | Qwell: right - a.s and memory permitting |
| 20:51.54 | stoffell | [av]bani, I've heard. using GXP-2000, Cisco and Thomson ST2030 at the moment |
| 20:52.09 | Qwell[] | MstlyHrmls: indeed |
| 20:52.14 | SplasPood | [av]bani: what alternatives exist, if any? |
| 20:52.21 | stoffell | too bad aastra doesn't have distinctive ring support |
| 20:52.36 | austinnichols101 | agreed: it's just who came up with the arbitrary limit of 7. Why not 6 or 8... |
| 20:52.44 | austinnichols101 | it's a lazy way to program things |
| 20:52.51 | [av]bani | SplasPood: snom, if you want >7 working blf. cisco. |
| 20:53.12 | xyklopz | Helix is afk |
| 20:53.14 | Renacor | hmm I don't see anything other than timeout value in the queues.conf to limit queue wait time but that doesn't do anything either |
| 20:53.21 | SplasPood | [av]bani: and the cisco solution is SIP? |
| 20:53.32 | [av]bani | SplasPood: depends on which model you buy |
| 20:53.42 | stoffell | SplasPood, you could also use chan_sccp |
| 20:53.45 | SplasPood | [av]bani: well whatever model is gonna let me show 20 lines :) |
| 20:53.48 | MstlyHrmls | austinnichols101: aye, 7 is dumb, and who knows why it was set at that. |
| 20:53.55 | SplasPood | stoffell: is it really considered production ready? |
| 20:53.58 | [av]bani | SplasPood: 7940/7960 with sidecar then |
| 20:54.29 | Qwell[] | 7940 with sidecar is silly |
| 20:54.38 | Qwell[] | in fact, I don't even know if it works |
| 20:54.45 | stoffell | SplasPood, depends, very active development, i'm using it with some Kirk phones, but I don't update if i don't need to :) |
| 20:54.47 | [av]bani | knowing cisco, probably not? |
| 20:55.44 | SpaceBass | can someone help me with calling a URI? I have a service trying to call an exten@myip and I'm not sure what I need to do on my end |
| 20:55.59 | [av]bani | if the gxp exp. port is for sidecar, i dont see how grandstream expects the modules to stay attached |
| 20:56.01 | harryvv | can anyone give me a voice quality rating comming from xo? |
| 20:56.52 | stoffell | does snom have licensing issues like cisco? (like, you pay for a license and stuff?) |
| 20:56.57 | [av]bani | no |
| 20:57.11 | stoffell | ok, tnx, cool |
| 20:57.13 | [av]bani | you shouldnt have to pay for cisco licenses though |
| 20:57.16 | SplasPood | ok well this 7 limit /w polycom has depressed me greatly... Anyone wanna cheer me up with some LCR info? :) |
| 20:57.23 | [av]bani | use sccp out of the box |
| 20:57.39 | stoffell | [av]bani, ah, ok, you're right |
| 20:59.02 | xyklopz | can someone help me with setting up a sipphone virtual # |
| 20:59.23 | xyklopz | i already have the sip peer setup |
| 20:59.28 | xyklopz | and it's registering through the firewall |
| 20:59.42 | xyklopz | now I want to purchase a virtual # for them to accept incoming |
| 21:00.33 | De_Mon | xyklopz whats your question? |
| 21:00.44 | De_Mon | s/ts/t's/ |
| 21:00.57 | De_Mon | ^_~ |
| 21:01.09 | xyklopz | okay .. the # is 1404XXXXXXX |
| 21:01.25 | xyklopz | I assume they handle the routing |
| 21:01.31 | xyklopz | from PSTN to IP |
| 21:01.50 | xyklopz | then it'll come on my sipphone # (my softphone account # 174XXXX..) |
| 21:01.53 | De_Mon | xyklopz is the provider registered in asterisk? |
| 21:02.03 | harryvv | xo is on a roll. It is probebly going to reach over 2 billion minutes this first quarter. |
| 21:02.14 | xyklopz | the peer is there |
| 21:02.55 | [av]bani | Qwell[]: sccp can invite an rtp stream from a sip phone, or does it need to be bridged through * ? |
| 21:04.02 | De_Mon | xyklopz ok, so someone calls that provider (any number) and it reads the config and is sent into the specified contex in extensions.conf |
| 21:04.12 | [av]bani | hmm. the zoom fxo echo tail is 10 milliseconds... bizarre |
| 21:04.19 | De_Mon | from there you told it to do whatever you wanted it to... what is your QUESTION? |
| 21:04.54 | [av]bani | De_Mon: what is the airspeed of an unladen swallow? |
| 21:05.05 | De_Mon | [av]bani 13 |
| 21:05.06 | Igbothom_III | african swallow? |
| 21:05.10 | Nivex | [av]bani: get out of my head! |
| 21:05.41 | [av]bani | Nivex: but it was so spacious and empty |
| 21:06.07 | Renacor | anybody got a good example on how to send a call to vm after it's been sitting in a queue for 2 min? |
| 21:06.32 | [av]bani | Renacor: set queue timeout to 2 min, then have it fall to voicemail(bla) |
| 21:06.36 | xyklopz | anyone have a sipphone account? |
| 21:06.42 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:07.01 | mischko | Is Vonage proprietary or using SIP? |
| 21:07.02 | Renacor | [av]bani: the timeout in queues.conf right? |
| 21:07.06 | De_Mon | Igbothom_III 13 too |
| 21:07.07 | [av]bani | Renacor: yes |
| 21:07.18 | Qwell[] | mischko: Both |
| 21:07.26 | Renacor | [av]bani: I tried that, and added another priority after the call was answered in the extensions.conf but it didn't work |
| 21:07.31 | Qwell[] | SIP with proprietary logins |
| 21:07.32 | De_Mon | xyklopz probably most people in here... |
| 21:07.44 | [av]bani | Renacor: i think you need to queue(bla) with some parameters for it to timeout and fall through |
| 21:07.48 | Renacor | [av]bani: do you have to restart asterisk for that or can you just do a reload ? |
| 21:07.52 | xyklopz | I just want to see if it rings through properly |
| 21:07.57 | [av]bani | Renacor: reload app_queue.so |
| 21:08.13 | De_Mon | xyklopz dial the number? |
| 21:08.53 | xyklopz | 17476380060 if it's called from another sipphone account it's free |
| 21:09.10 | xyklopz | i want to make sure it's going through * correctly to my lan peer I told it to ring to |
| 21:10.00 | De_Mon | Is sipphone some VoIP provider I've never heard of? |
| 21:10.08 | xyklopz | sipphone.com |
| 21:10.14 | xyklopz | incoming virtual #'s only $35/year |
| 21:10.21 | xyklopz | no fee on incoming and only 2c/min outgoing |
| 21:10.24 | xyklopz | pretty cheap |
| 21:11.11 | mischko | I assume WiFi SIP phones are available? |
| 21:11.16 | Qwell[] | mischko: yep |
| 21:11.19 | afrosheen | sorta |
| 21:11.19 | De_Mon | oh.. in that case I retract my previous comment about most people having it |
| 21:11.28 | xyklopz | {{ £åügHîñg Øüt £öüÐ }} ... |
| 21:11.32 | De_Mon | xyklopz if you can make outgoing calls you can call it yourself |
| 21:11.34 | afrosheen | ow my eyes |
| 21:11.35 | Qwell[] | mischko: If you get me a cisco 7920, I'll make sure it works with chan_sccp ;) |
| 21:12.18 | mischko | Qwell[], I wish. What's chan_sccp? |
| 21:12.23 | mikefoo | I am sitting next to four cisco as5400's hah |
| 21:12.45 | Qwell[] | mischko: sccp channel driver for * |
| 21:13.54 | KranZ | anyone messed with SRTP? |
| 21:14.45 | harryvv | mischko have been around for a while |
| 21:15.01 | mischko | harryvv, not around VOIP. |
| 21:15.36 | [av]bani | mischko: all the wifi phones seem to suck |
| 21:15.48 | mischko | [av]bani, how come? |
| 21:15.50 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
| 21:15.51 | Qwell[] | 7920 doesn't suck |
| 21:16.07 | [av]bani | Qwell[]: someone with a 7920 was complaining the other day its crap like a rebranded chinese clone |
| 21:16.13 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:16.32 | Qwell[] | [av]bani: Just like the guy who said a 7960 was better than a 7970? |
| 21:16.33 | Qwell[] | mmhmm |
| 21:16.37 | [av]bani | no |
| 21:16.48 | [av]bani | someone in this channel was having problems with his 7920 |
| 21:16.50 | *** join/#asterisk justinu (n=justin@72.18.13.34) |
| 21:16.57 | [av]bani | said it was rebranded chinese clone junk |
| 21:16.58 | mikefoo | I have six 7960's on my desk, hah |
| 21:16.59 | Qwell[] | so immediately it sucks...yes, of course |
| 21:17.04 | Primer | I have a 7920 |
| 21:17.06 | Primer | works fine |
| 21:17.15 | Primer | using chan_sccp |
| 21:17.30 | PakiPenguin | do sangoma analog cards support fax correctly? |
| 21:17.31 | Primer | but I got mine from a friend that works at cisco |
| 21:17.41 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 21:17.54 | [av]bani | Primer: is it worth the $550+ cisco price tag? |
| 21:18.05 | [TK]D-Fender | Hey, need a hand here... I've got a PRI thats "Status: Provisioned, Down, Active" |
| 21:18.05 | Primer | damn, I didn't realize it cost that much |
| 21:18.10 | [av]bani | Primer: yep |
| 21:18.12 | Primer | perhaps I should ebay it |
| 21:18.16 | [av]bani | heh |
| 21:18.17 | Qwell[] | I don't think they do...not from cisco |
| 21:18.18 | [TK]D-Fender | How do I cycle it so I can try to re-initializee the link |
| 21:18.28 | justinu | d-fender: restart asterisk? |
| 21:18.48 | Primer | [av]bani: well, the battery doesn't seem to last long, but as far as performance, it works fine |
| 21:18.52 | justinu | asterisk doesn't seem to understand the concept of placing things OOS/INS |
| 21:19.04 | [TK]D-Fender | :/ |
| 21:19.18 | Primer | [av]bani: could be that this one is just old, it's pretty beat up |
| 21:19.21 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 21:21.00 | xyklopz | okay, * isn't accepting the call from sipphone.com! |
| 21:21.01 | xyklopz | :-( |
| 21:21.07 | xyklopz | it's registering |
| 21:21.09 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:21.48 | KranZ | sip debug ip [ipofsipphone.com] |
| 21:22.42 | iCEBrkr | I should just go home |
| 21:22.48 | iCEBrkr | I can't think anymore |
| 21:23.23 | [av]bani | Qwell[]: they are more expensive from cisco. 550+ is street. retail is $675.99 |
| 21:23.31 | [av]bani | http://www.voipsupply.com/product_info.php?products_id=57 |
| 21:23.31 | justinu | 4:20 anyways |
| 21:23.32 | justinu | :P |
| 21:23.37 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 21:24.18 | Qwell[] | nope...sorry...you're wrong :p |
| 21:24.26 | [av]bani | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-29579092224.htm |
| 21:24.40 | Qwell[] | $406 + $49 + $29 |
| 21:24.47 | Qwell[] | phone + battery + charger |
| 21:25.04 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
| 21:25.08 | [av]bani | oh.. $406 phone with no batter and no charger. excellent |
| 21:25.15 | [av]bani | i stand corrected |
| 21:25.17 | Qwell[] | < $500 |
| 21:25.22 | Qwell[] | 500 < 550 |
| 21:25.27 | [av]bani | $406 phone you cant use |
| 21:25.32 | [av]bani | super |
| 21:25.36 | Qwell[] | $406 + $49 + $29 |
| 21:25.38 | Qwell[] | phone + battery + charger |
| 21:25.43 | greendisease | how do you reload extensions? |
| 21:25.43 | Qwell[] | That's still < 500 |
| 21:25.46 | [av]bani | where from? |
| 21:25.48 | malverian[work] | extensions reload |
| 21:25.50 | Qwell[] | greendisease extensions reload |
| 21:25.51 | Qwell[] | [av]bani cisco |
| 21:25.57 | malverian[work] | asterisk -r -x "extensions reload" |
| 21:26.03 | [av]bani | cisco sells it for less than voipsupply retails it? |
| 21:26.08 | Qwell[] | umm...duh? |
| 21:26.17 | greendisease | i figured |
| 21:26.17 | Qwell[] | How else is voipsupply going to make a profit? |
| 21:26.19 | greendisease | thank Qwell |
| 21:26.22 | Qwell[] | YOu think they sell stuff below cost? |
| 21:26.40 | iCEBrkr | justinu: exactly |
| 21:26.59 | [av]bani | dealer cost is generally < retail |
| 21:27.07 | [av]bani | why else have resellers? |
| 21:27.27 | Qwell[] | ask voipsupply |
| 21:27.31 | [av]bani | if cisco will sell direct to end users for less than resellers... sort of makes resellers pointless |
| 21:27.35 | Qwell[] | They're the tools who add $150 onto everything |
| 21:27.40 | [av]bani | Qwell[]: atacomm? |
| 21:27.44 | Qwell[] | $200 |
| 21:28.17 | trixter | http://www.thevoipconnection.com doesnt seem to rape as badly, they also dont charge what is generally listed on their webpage, or at least I have never paid that much.. and I have always gotten a price below voipsupply |
| 21:28.24 | Qwell[] | how much is voipsupply selling 7970's for? or even 7960's? |
| 21:28.26 | xyklopz | it's not calling!!! |
| 21:28.29 | Abydos313 | i'm editing the Makefile to compile my ztdummy driver and the line the book says to look for is not there |
| 21:28.31 | trixter | plus mike is a good guy, quick and very responsive ... |
| 21:28.46 | xyklopz | okay, configure the softphone directly to sipphone.com; call the virtual #; softphone rings |
| 21:28.51 | [av]bani | Qwell[]: 549 for 7970, 299 for 7960 |
| 21:29.04 | xyklopz | configure asterisk as sipphone.com peer; call the virtual #; get sipphone.com's vmail system |
| 21:29.21 | Qwell[] | cisco just raised their prices on the 7970 it seems...nice |
| 21:29.33 | [av]bani | excellent |
| 21:29.35 | _Sam-- | whats this mean: |
| 21:29.36 | _Sam-- | Feb 17 16:29:06 WARNING[21881]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x410015f0', 10 retries! |
| 21:29.47 | mischko | [av]bani, I just hit both the links above for the 7920 and they show $575??? |
| 21:29.51 | Qwell[] | 270 for the 7960 |
| 21:29.59 | [av]bani | mischko: yes |
| 21:30.03 | Dr-Linux | hi guys |
| 21:30.10 | mischko | I just called Cisco and list on that phone is 475 (no battery/charger). |
| 21:30.12 | Dr-Linux | sam i'm happy today :P |
| 21:30.18 | [av]bani | mischko: liar! |
| 21:30.24 | _Sam-- | Dr-Linux : happy day to you |
| 21:30.24 | mischko | battery is 75 or 95 and charger is 45. LIST. |
| 21:30.28 | Qwell[] | mischko: retail :) |
| 21:30.37 | mischko | [av]bani, I JUST called Cisco. |
| 21:30.39 | _Sam-- | im happy every day...or at least every day when my remote asterisk gateways work |
| 21:31.20 | [av]bani | mischko: so what is Qwell[] babbling about $406+49+29 then? |
| 21:31.29 | Qwell[] | [av]bani: direct from cisco |
| 21:31.29 | mischko | I don't know where he got those numbers. |
| 21:31.39 | [av]bani | Qwell[]: end users? |
| 21:31.41 | Abydos313 | nm guess it would help to open the correct Makefile..heh |
| 21:31.55 | mischko | [av]bani, Cisco's list price is more than I'd expect to find it for on the street. |
| 21:31.56 | _Sam-- | Dr-Linux : did you fix your wake up? |
| 21:32.13 | [av]bani | mischko: i guess he's quoting dealer cost, and berating voipsupply for selling dealer cost + markup |
| 21:32.41 | mischko | [av]bani, for a retailer to sell that far above list is completely out of line. They should be 20-30% below list. |
| 21:33.14 | [av]bani | mischko: i cant seem to find anyone selling the 7920 for anything lower |
| 21:33.25 | [av]bani | most dont even sell the 7920... |
| 21:33.26 | justinu | iCEBrkr: it's only 1:36 here :( |
| 21:33.43 | mischko | [av]bani, You can get it direct from Cisco for less than it's on those pages. |
| 21:33.55 | _Paulo_ | here 19:33h. Time to grab a six-pack. |
| 21:34.04 | [av]bani | mischko: maybe that explains why so few resellers list it |
| 21:34.22 | mischko | I'd call people who have Cisco product line and ask 'em for a quote. |
| 21:35.07 | mikefoo | iCEBrkr: hows the campaings going? |
| 21:35.11 | Dr-Linux | _Sam--: no, everything is done, but the alarm file doesn't copy to /var/spood/asterisk/outgoing dir |
| 21:35.24 | _Sam-- | check your permissions |
| 21:35.33 | mischko | I've gotta run. Thanks for all the help [av]bani , Qwell[] , etc. |
| 21:35.36 | _Sam-- | if asterisk is not running as root, it may not have the right permissions to write there |
| 21:36.03 | *** part/#asterisk mischko (n=Scott@cvo-cr1-200-242.peak.org) |
| 21:36.24 | NovceGuru | how come when dialing out some of my numbers work and some say "all circuits are busy now" ? |
| 21:37.55 | *** join/#asterisk Simon- (i=fictitio@80.193.211.68) |
| 21:38.05 | Dr-Linux | _Sam--: you are talking to me? |
| 21:38.07 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 21:38.10 | GerbilWrk | has anyone found a way to imporve QoS with Teliax? |
| 21:38.30 | *** join/#asterisk cassio (n=cassio@c91133b9.rjo.virtua.com.br) |
| 21:38.33 | _Sam-- | GerbilWrk : avbani found a way....get your terminating calls from someone else |
| 21:38.50 | GerbilWrk | ;[ |
| 21:38.54 | xyklopz | I can't get incoming to dial two extensions |
| 21:39.02 | iCEBrkr | mikefoo: Working on quota management |
| 21:39.05 | _Sam-- | xyklopz : DIAL(sip/1&sip/2) |
| 21:39.05 | cassio | does anyone know any provisioning software for sipura and linksys? |
| 21:39.11 | xyklopz | exten => s,1,Dial(SIP/1002&SIP/1005,20,r) doesn't work |
| 21:39.33 | _Sam-- | then it only doesnt work because it doesnt know how to reach sip/10002 or 1005 (whichever one isnt ringing) |
| 21:39.37 | iCEBrkr | mikefoo: It paces itself and obviously records the data. But we have quotas to meet and need to quit dialing |
| 21:39.40 | _Sam-- | one of them probably isnt registered to * |
| 21:39.45 | xyklopz | but they can talk to each other |
| 21:39.53 | iCEBrkr | _Sam--: Tonights the night. I'm gonna try to start the ol' dinosaur! |
| 21:39.54 | xyklopz | the dialplan seems right |
| 21:39.54 | _Sam-- | they need to register to receive incoming calls from * |
| 21:39.56 | xyklopz | pm me? |
| 21:40.01 | _Sam-- | sorry i have work to do |
| 21:40.02 | xyklopz | they are both in sip show peers |
| 21:40.07 | _Sam-- | sip show registry |
| 21:40.10 | mikefoo | any decent answering machine detection out there? |
| 21:40.12 | _Sam-- | er |
| 21:40.15 | _Sam-- | sip show peers will be fine too |
| 21:40.18 | iCEBrkr | mikefoo: app_machinedetect.c |
| 21:40.18 | _Sam-- | if it shows their ip |
| 21:40.24 | xyklopz | which it does |
| 21:40.28 | mikefoo | iCEBrkr: word.. thanks |
| 21:40.28 | iCEBrkr | mikefoo: google it |
| 21:40.30 | xyklopz | i reloaded the config |
| 21:40.32 | SplasPood | has there been ANY development as to SIP-B + Asterisk? |
| 21:40.44 | mikefoo | iCEBrkr: yah i googled before, seen a few, but bad reviews on them all. |
| 21:40.48 | _Sam-- | iCEBrkr: good luck! |
| 21:40.49 | [av]bani | SplasPood: hasnt started afaik |
| 21:40.51 | mikefoo | iCEBrkr: later |
| 21:40.52 | iCEBrkr | _Sam--: :) |
| 21:41.00 | SplasPood | or anything that does SIP-B that comes without a gigantic invoice... |
| 21:41.00 | [av]bani | SplasPood: coming summer 2006 to an * near you |
| 21:41.09 | SplasPood | [av]bani: ya I saw the post :) |
| 21:41.31 | SplasPood | No bounties? Anyone we could offer some cash to to get some movement on that earlier? |
| 21:41.40 | *** join/#asterisk mischko (n=Scott@cvo-cr1-200-242.peak.org) |
| 21:41.47 | mischko | [av]bani, http://www.mtmnet.com/CP-7920-AP-K9_New.htm |
| 21:41.50 | mischko | $460 |
| 21:42.01 | cassio | does anyone know any provisioning software for sipura and linksys? |
| 21:42.38 | _Sam-- | cassio: did you try the [AV]baniMAKER (TM)? |
| 21:42.59 | _Sam-- | <sorry couldnt help it> |
| 21:42.59 | justinu | bani: i just tested the tones |
| 21:43.11 | justinu | two comments: the snom "congestion" tone, has the wrong cadence |
| 21:43.20 | justinu | and it sounds brighter, or harsher |
| 21:44.17 | mischko | What's it mean on the Cisco 7920 " without User License"? |
| 21:44.46 | justinu | means you can buy it, but it's illegal to use :P |
| 21:44.59 | mischko | What do you need a license for? |
| 21:45.02 | [av]bani | justinu: yes, exactly |
| 21:45.11 | [av]bani | justinu: snom congestion is raspy and bizarre |
| 21:45.24 | *** part/#asterisk Simon- (i=fictitio@80.193.211.68) |
| 21:45.26 | justinu | but i wouldn't have noticed it unless you pointed it out |
| 21:45.33 | [av]bani | justinu: compared dialtone too? |
| 21:45.45 | [av]bani | same thing for me, snom dialtone is raspy and brighter |
| 21:45.46 | NovceGuru | http://pastebin.com/560231 I assume that problem isn't on my end? |
| 21:45.46 | justinu | no, i'll try that... |
| 21:46.11 | [av]bani | justinu: while playtones(dial) is smooth |
| 21:46.19 | xyklopz | okay, incoming sip calls aren't being displayed in the debug output |
| 21:46.27 | xyklopz | if i register with a softphone everything works fine |
| 21:46.27 | [TK]D-Fender | SplasPood : SIP-B for * 1.4 |
| 21:46.41 | justinu | bani: yeah, the difference in dial tone is more apparent |
| 21:46.42 | xyklopz | something isn't right with asterisk registering to sipphone.com as a peer |
| 21:46.54 | [av]bani | justinu: k, just making sure its not just me or "user error" :) |
| 21:46.59 | justinu | seems not |
| 21:47.06 | [av]bani | justinu: i'd like to get recordings somehow, since snom seems incredulous |
| 21:47.11 | justinu | heh |
| 21:47.14 | [av]bani | justinu: "you are mistaken sir" |
| 21:47.21 | justinu | i wonder if they're sampled in the rom, or if it's a tone generator function |
| 21:47.23 | [av]bani | justinu: "our tones are correct, please put down your bong" |
| 21:47.35 | De_Mon | Abydos313 are you going to use meetme with ztdummy? |
| 21:48.00 | [av]bani | justinu: dunno, would be silly to be sampled though.. would make it a pita to support arbitrary zones |
| 21:48.08 | NovceGuru | anybody? |
| 21:48.10 | justinu | yeah, but c'mon... it's snom |
| 21:48.13 | [av]bani | :) |
| 21:48.39 | [av]bani | nice hardware, shame about the software |
| 21:48.50 | justinu | snom is also the only phone i have that doesn't support remote-party-id in the 180/183 responses |
| 21:48.51 | jeebusroxors | can anyone recomend a cheap-o phone? its for home use.... |
| 21:49.02 | justinu | jeebusroxors: grandstream gxp2000 |
| 21:49.04 | justinu | $85 |
| 21:49.20 | mischko | What do these "phone technology licenses" cost? |
| 21:49.37 | jeebusroxors | justinu; thanks alot - any chance of picking one of those up localy? best buy etc |
| 21:49.38 | [av]bani | justinu: remote party id? |
| 21:49.44 | justinu | jeebusroxors: nope |
| 21:49.50 | jeebusroxors | justinu; figures :) |
| 21:49.59 | justinu | bani: rpid is how asterisk can tell the callee who he's calling |
| 21:50.04 | justinu | s/callee/caller/ |
| 21:50.05 | [av]bani | jeebusroxors: nobody sells sip phones retail, except maybe fry's |
| 21:50.18 | justinu | i looked at fry's last time I was there |
| 21:50.21 | justinu | they didn't have any |
| 21:50.45 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 21:50.45 | jeebusroxors | that sucks....are other VOIP phones more costly? |
| 21:51.08 | justinu | yeah, there is nothing at the gxp2000's pricepoint |
| 21:51.12 | justinu | with the kind of features it has |
| 21:51.19 | [av]bani | gxp2k is best price/features at that price point |
| 21:51.20 | justinu | you can step down to an ATA, or budgetone 101 |
| 21:51.21 | SplasPood | [TK]D-Fender: I know, I was wondering if some cash given to the right people might speed it up :) |
| 21:51.34 | justinu | you can step up to polycom, or cisco |
| 21:51.43 | [av]bani | it's not a great phone, but it's great value for the money |
| 21:51.44 | justinu | or snom ;) |
| 21:51.58 | justinu | is there an RFC for SIP-B? |
| 21:51.59 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:52.01 | [av]bani | for a home phone for playing with *, totally acceptable |
| 21:52.30 | justinu | one of my beta IP centrex customers loves his |
| 21:52.37 | [av]bani | jeebusroxors: actually walmart sells a crappy soyo voip phone. they dont stock it though, you need to order it on their website to be delivered to your local store |
| 21:52.39 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 21:52.40 | justinu | my PRI customers hate them |
| 21:52.49 | jeebusroxors | [av]bani; thats cool. eventually id like to have my family hooked up on * |
| 21:52.55 | harryvv | hate what justinu |
| 21:53.00 | justinu | the gxp2000 |
| 21:53.05 | [av]bani | jeebusroxors: but if youre going to order from the web, might as well order a gxp2000 |
| 21:53.16 | harryvv | ohh |
| 21:53.27 | harryvv | why poor quality? |
| 21:53.27 | jeebusroxors | [av]bani; yea ive actually seen those hanging around.... |
| 21:53.28 | [av]bani | jeebusroxors: you could spend a little more and get a polycom 301 though |
| 21:53.59 | jeebusroxors | the budget 101 is actually more than the gxp2000 heh |
| 21:54.07 | justinu | harryvv: poor interoperability with the PRI, i suppose |
| 21:54.10 | [TK]D-Fender | SplasPood : Sorry.. wish I could as I'm a Polycom user who'd benifit from it, but I'm not a coder.... |
| 21:54.11 | PakiPenguin | anyone here is from the chicago area? |
| 21:54.13 | harryvv | i see |
| 21:54.16 | justinu | jeebusroxors: budgetone 101 should be about $50 |
| 21:54.28 | PakiPenguin | i'd like to ask about any t1 provider ( voice + data both ) |
| 21:54.29 | jeebusroxors | i saw it at 89 heh |
| 21:54.31 | afrosheen | I've seen some junk sip phones at Fry's before |
| 21:54.37 | [av]bani | SplasPood still upset over the 7 buddy revelation? |
| 21:54.39 | afrosheen | Asus or someone was selling them |
| 21:54.39 | [av]bani | :) |
| 21:54.42 | harryvv | polycom should make vidio security cat5 based and wifi cameras :) |
| 21:54.46 | [av]bani | i just ruined SplasPood's week :) |
| 21:54.57 | jeebusroxors | and the soyo is 90 bucks *rolls eyes* |
| 21:55.03 | mischko | [av]bani, Those licenses are about $150 apparently so the prices cited earlier may be good enough if they include a license. |
| 21:55.10 | jeebusroxors | looks like gxp2000 it is |
| 21:55.10 | [TK]D-Fender | [av]bani : Well Cisco's 7914 only works in CCP, so that leaves SNOM and their flakeyness for presence :) |
| 21:55.12 | afrosheen | jeebusroxors: yeah that's the one, the Soyo |
| 21:55.15 | [av]bani | jeebusroxors: yeah, like i said nothing comes close to the gxp2k in terms of price/features |
| 21:55.22 | SplasPood | [av]bani: oh yea totally. I'm a total idiot for never trying more than 7 |
| 21:55.35 | jeebusroxors | well thanks alot for the advice |
| 21:55.37 | [av]bani | SplasPood: happy to help :) |
| 21:55.39 | SplasPood | [av]bani: And we were totally hoping to standardize on pcom |
| 21:55.44 | Qwell[] | mischko: $98 for most licenses, actually. |
| 21:55.46 | JCC_ | the sipura 3000 ata at $99 worked out well for me as a home system, I keep the local # and use VoIP for LD |
| 21:55.54 | afrosheen | might as well go with a decent phone and an iAXy if you're gonna be cheap |
| 21:56.01 | JCC_ | and keep the portable phone |
| 21:56.08 | [av]bani | mischko: i would imagine they do |
| 21:56.14 | justinu | jcc: you having luck with the FXO side of it? |
| 21:56.17 | afrosheen | I did that with my portable phone at home |
| 21:56.28 | mischko | Are these licenses a Cisco-only thing or something else? |
| 21:56.30 | [av]bani | mischko: i checked around, nobody seems to sell it cheaper. if they list a price thats cheaper it always ends up being without battery or PS |
| 21:56.38 | Qwell[] | you need a license to use it with cisco |
| 21:56.43 | JCC_ | yes, a little awkward at first, bt there are resources on the www |
| 21:56.53 | justinu | SplasPood: is there an RFC for sip-b? |
| 21:56.55 | _Sam-- | anyone using any wireless headsets that can answer/hangup calls from a softphone? |
| 21:57.12 | [av]bani | mischko: so the voipsupply and atacomm prices do not seem to be out of line with anyone else |
| 21:57.26 | SplasPood | justinu: yes I think so... I believe it's linked to on voip-info.org |
| 21:57.38 | mischko | [av]bani, seems like it, if the include the magic license. Is this some sort of Cisco price gouge? |
| 21:57.42 | mischko | (the license) |
| 21:57.47 | [av]bani | _Sam--: maybe a bluetooth headset? |
| 21:57.51 | Qwell[] | voipsupply doesn't include licenses |
| 21:57.56 | JCC_ | not sure what you mean, but I tried a bluetooth unit with Xten and didn't care for it, too many drops |
| 21:58.00 | _Sam-- | i dont think they will answer/hangup the calls remotely |
| 21:58.02 | [av]bani | mischko: cisco loves licensing.. and smartnet |
| 21:58.06 | Qwell[] | except maybe on the 7920 |
| 21:58.08 | _Sam-- | i have a wirless usb plantronics one |
| 21:58.19 | _Sam-- | but it only answer/hangs up like 2 softphones remotely |
| 21:58.33 | justinu | searching for SIP-B rfc comes up with nothing |
| 21:58.36 | _Sam-- | guess i have to buy eyebeam |
| 21:58.37 | mischko | [av]bani, so you're paying an extra $100 for a cisco phone. That's nuts. What do they offer that makes it worth it to go there? |
| 21:58.38 | *** join/#asterisk darby_t (i=darby_t@dkt82.neoplus.adsl.tpnet.pl) |
| 21:58.48 | justinu | mischko: name recognition |
| 21:58.58 | Qwell[] | mischko: If you buy it from voipsupply...you still need a license... |
| 21:58.59 | [av]bani | jeebusroxors: its proably an optional annex to the main sip rfc |
| 21:59.04 | justinu | and a phone that people say works |
| 21:59.04 | JCC_ | and good reliability, but expensive |
| 21:59.05 | Qwell[] | to use it with ccm anyhow |
| 21:59.05 | Dr-Linux | justinu: heyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy :P |
| 21:59.10 | [av]bani | s/jeebusroxors/justinu/ |
| 21:59.13 | justinu | Dr-Linux: :P |
| 21:59.14 | [TK]D-Fender | SplasPood : I'd still go ahead if you can livewithout the full-service receptionist for a while |
| 21:59.24 | justinu | bani: i haven't been able to find it yet, but i'd like to read it. |
| 21:59.28 | mischko | Qwell[], so if you use it standalone, you don't need a license? |
| 21:59.35 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 21:59.36 | Qwell[] | as far as I'm aware |
| 21:59.44 | mischko | what's ccm provide you? |
| 21:59.47 | Egonis | I can't get device nodes to show for my sangoma card, although wancfg finds the card, can anyone help me/ |
| 21:59.52 | justinu | ccm is a softswitch |
| 21:59.59 | Qwell[] | mischko: a big hole where your bank account used to be? |
| 22:00.02 | justinu | similar in functionality to asterisk, i suppose |
| 22:00.04 | mischko | lol |
| 22:00.09 | justinu | i heard it runs on windows tho |
| 22:00.12 | Qwell[] | it does |
| 22:00.13 | justinu | which is frightening |
| 22:00.15 | mischko | yuk. |
| 22:00.32 | Qwell[] | its funny though...even cisco direct, they won't sell you a 7920 without a UL |
| 22:00.34 | mischko | So you can use the 7920 with * just fine and it won't need the nasty license. |
| 22:00.42 | Qwell[] | if they did...you'd be able to get it for $308 |
| 22:00.43 | justinu | you still need a SIP license |
| 22:00.48 | Qwell[] | nah |
| 22:00.48 | justinu | no? |
| 22:00.51 | Qwell[] | keep it with sccp :p |
| 22:00.54 | Primer | 7920 is sccp |
| 22:00.57 | justinu | so it's legal to buy and use with SCCP? |
| 22:00.58 | Qwell[] | get an $8 smartnet |
| 22:01.03 | Primer | there is no official sip firware for it |
| 22:01.04 | Primer | afaik |
| 22:01.05 | Qwell[] | as far as I'm aware, heh |
| 22:01.10 | mischko | SIP devices require a license? Is that some Federal tax? |
| 22:01.14 | Qwell[] | I think you only need a license to use it with CCM |
| 22:01.16 | justinu | no, it's a cisco tax |
| 22:01.29 | SplasPood | [TK]D-Fender: We will... Although a receptionist console of some sort will be a requirement for some customers.. |
| 22:01.38 | JCC_ | cisco gives away nothing... |
| 22:01.44 | [av]bani | except grief |
| 22:01.47 | Qwell[] | So...if you can get cisco to sell you a spare 7920...you could get away with < $400 |
| 22:01.54 | mischko | JCC_, I'd say they charge you for nothing. The license is nothing. |
| 22:01.59 | justinu | "you mean you want software for you phone? that's another 100 bucks..." |
| 22:02.03 | afrosheen | lol |
| 22:02.09 | Qwell[] | cp-7920-fc= would be the product number, if it existed |
| 22:02.13 | [TK]D-Fender | SplasPood : You a reseller? |
| 22:02.23 | [av]bani | justinu: welcome to 1994? cisco has done that since forever |
| 22:02.34 | justinu | yep, xyxel was always like that too |
| 22:02.38 | mischko | someone said to stay away from Grandstream. How come? |
| 22:02.48 | Hmmhesays | anyone in here ever lived in argentina? |
| 22:02.50 | justinu | "you bought a terminal server off ebay? that's just the hardware... the firmware image will cost you $800" |
| 22:02.50 | [av]bani | mischko: because grandstream are cheap, like sipura used to be |
| 22:02.57 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 22:02.59 | SplasPood | [TK]D-Fender: Well we're pushing a voip service offering.. as part of it we supply the phones, yea |
| 22:03.02 | jhiver | 'nite all |
| 22:03.05 | [TK]D-Fender | mischko : Because they are cheap flimsy phones with flakey firmware.... |
| 22:03.13 | Qwell[] | oh, wait... |
| 22:03.19 | JCC_ | sipura used to be cheap... until cisco bought them |
| 22:03.21 | Qwell[] | cp-7920-fc-k9= |
| 22:03.21 | [av]bani | as opposed to well constructed phones with flakey firmware... |
| 22:03.26 | [TK]D-Fender | SplasPood : Ok, well there are PC based optiosn if they can live with it for a bit...\ |
| 22:03.29 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 22:03.37 | SplasPood | [TK]D-Fender: like FOP? |
| 22:03.42 | [TK]D-Fender | Sipura was always AFFORDABLE, not "cheap". |
| 22:03.44 | mischko | [av]bani, what brand do you recommend? |
| 22:03.47 | afrosheen | well I love our polycoms, the 38 out of 40 that don't reboot mysteriously |
| 22:03.54 | [TK]D-Fender | SplasPood : Yeah or any of the other AMI based options. |
| 22:03.57 | JCC_ | better put, you're right |
| 22:03.59 | [av]bani | mischko: depends on your requirements and budget |
| 22:04.16 | Egonis | Has anyone else had /dev problems with Sangoma Cards? |
| 22:04.20 | *** join/#asterisk bhima (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
| 22:04.23 | [av]bani | [TK]D-Fender: the 841 was cheap |
| 22:04.27 | jhiver | mischko, I like my Fritz!FonBox integrated DSL / VoIP device |
| 22:04.37 | [av]bani | [TK]D-Fender: which is why everyone crreamed their pants over hte 941 |
| 22:04.41 | jhiver | talks SIP, no NAT issue, just connect some phones on it |
| 22:04.43 | SplasPood | [TK]D-Fender: Do you have a personal fav? |
| 22:04.46 | [TK]D-Fender | [av]bani : Ok, barring that one :) It's nearly the same boat as the GXP :) |
| 22:05.15 | [av]bani | [TK]D-Fender: spa3k is also cheap |
| 22:05.21 | bhima | I'd like to make an asterisk call go through with audio in one direction only. Is there an easy way to do that? |
| 22:05.24 | [av]bani | [TK]D-Fender: and now, basically unsupported :/ |
| 22:05.40 | Qwell[] | okay, they do exist. $309 for a 7920, plus the battery/charger |
| 22:05.46 | jhiver | bhima, yes, use SIP + NAT :)))) |
| 22:05.53 | [av]bani | Qwell[]: cisco refurb? |
| 22:05.53 | Qwell[] | with no license |
| 22:05.56 | [av]bani | ew |
| 22:05.57 | Qwell[] | brand new |
| 22:06.01 | jhiver | just kidding really :) |
| 22:06.02 | bhima | jhiver: SIP and NAT were working for me. :P |
| 22:06.05 | ManxPower | http://www.ashlux.com/?postid=14 |
| 22:06.22 | [av]bani | Qwell[]: cisco direct? i cant find any cisco resellers who will sell for that low |
| 22:06.29 | [av]bani | neither can mischko |
| 22:06.30 | Qwell[] | [av]bani yeah |
| 22:06.40 | [av]bani | makes you woinder why cisco bothers with resellers then |
| 22:06.47 | [av]bani | if they are just going to undercut their retail channel |
| 22:06.56 | Qwell[] | list is about $550 |
| 22:07.24 | mischko | Qwell, list is $475 for part# CP-7920-AP-K9 |
| 22:07.38 | [av]bani | thats with license right? |
| 22:07.41 | [av]bani | -ap- |
| 22:07.42 | Qwell[] | mischko: change AP to FC...that'll work in the US |
| 22:07.44 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:07.50 | Qwell[] | no, with the license is -ch1 |
| 22:07.53 | PakiPenguin | i was wondering what sort of t1 i'd need to carry both voice and data if i go for a t1 from some provider and what card / hardware would i need ? |
| 22:08.11 | mischko | Qwell[], fellow at Cisco knew I was calling for US usage and gave me the wrong part #. Nice. |
| 22:08.17 | Qwell[] | heh |
| 22:08.26 | Qwell[] | I don't know what the difference is, but...pretty sure you want -fc |
| 22:08.35 | Qwell[] | fc=fcc |
| 22:08.41 | Qwell[] | must be fcc approved...dunno |
| 22:08.50 | Qwell[] | or follows fcc regs |
| 22:08.57 | Qwell[] | ap is for eu iirc |
| 22:09.07 | mischko | Ok. I'm off again. |
| 22:09.22 | ketanp | jhiver: does that fritz box provision over https? |
| 22:09.49 | jhiver | ketanp, I don't think it does support provisioning |
| 22:09.53 | jhiver | you have to configure it |
| 22:10.02 | jhiver | and I don't think you can lock it either |
| 22:10.05 | ketanp | jhiver: ok, thanks |
| 22:10.07 | Qwell[] | brb |
| 22:10.11 | ketanp | always a catch... haha |
| 22:10.18 | jhiver | but it's such a sweet device ): |
| 22:10.19 | jhiver | :) |
| 22:11.40 | bhima | so, anybody know how to mute one side of a call? |
| 22:11.45 | JCC_ | are they available in the US? |
| 22:13.13 | [TK]D-Fender | PakiPenguin : You'd just need a mixed mode T1 and any of the std T1 cards supported by * |
| 22:13.39 | PakiPenguin | [TK]D-Fender, i see , how'd i get data out of it? |
| 22:14.10 | PakiPenguin | i mean the hardware needed for that |
| 22:14.19 | PakiPenguin | would be provided by the teleco? |
| 22:16.56 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
| 22:17.07 | [TK]D-Fender | PakiPenguin : the T1 card does the work... |
| 22:17.15 | PakiPenguin | awesome! |
| 22:17.16 | [TK]D-Fender | be it Digium or Sangoma |
| 22:18.12 | ManxPower | I prefer doing the muxing/demuxing outside of Asterisk. That way the internet and frame relay doesn't go down everytime I run ztcfg or reboot the Asterisk server, or stop Asterisk. Expecially if you are SSHing into the system via that internet link |
| 22:20.14 | Mavvie | ASTERISK_FILE_VERSION(__FILE__, "$Revision$") |
| 22:20.15 | Mavvie | Wonder why this isn't expanded by SVN. |
| 22:20.15 | PakiPenguin | ManxPower, what can i use to do that? i mean the hardware? |
| 22:20.54 | ManxPower | PakiPenguin, Adtran TA 850 will do it, and there are several Adtran TSUs that will do it. |
| 22:21.15 | ManxPower | Lemme check a min. |
| 22:21.23 | ManxPower | naw, I'm too lazy for that. |
| 22:21.41 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 22:21.50 | ManxPower | Adtran makes several boxes that will do it. We got one that take a T-1, splits out channels to a DXS-1 port (to Asterisk) and V.35 port (to the Cisco router) |
| 22:22.32 | *** join/#asterisk file[laptop] (n=jcolp@206.13.96.155) |
| 22:22.48 | iCEBrkr | eh? |
| 22:24.01 | ManxPower | TSU120 is what I think we have. |
| 22:24.18 | PakiPenguin | i see |
| 22:24.27 | *** join/#asterisk Libila (n=vye@ip68-8-174-154.sd.sd.cox.net) |
| 22:24.30 | ManxPower | come to think of it a TA750 should be able to do it too, but I've not tried that. |
| 22:25.43 | harryvv | I wonder if its possible to force a extention to ask for a 1800 number and dial it on a usa bound termination point. lots of 1800 us bound numbers dont work up here in canada. |
| 22:26.01 | *** join/#asterisk Tamarisk (n=adrian@user-6887.lns5-c11.dsl.pol.co.uk) |
| 22:26.35 | *** join/#asterisk Ahrimanes (n=michael@aronsen.dk) |
| 22:27.08 | ManxPower | harryvv, trivia |
| 22:27.10 | ManxPower | trivial |
| 22:28.10 | ManxPower | You just have to make sure that users can't call USA toll free numbers that bill your phone number |
| 22:28.16 | *** part/#asterisk JCC_ (n=john@207.41.92.131) |
| 22:28.31 | Libila | I've got this chunk in my extensions.conf (http://rafb.net/paste/results/HEA11x69.html) and when I call a normal 7 digit phone number I hear all this cracking and buzzing, asterisk says: -- Executing Dial("SIP/user1-dd98", "Zap/g1/1234567|20|t") in new stack, then says -- Called g1/1234567. So why do I not hear the number I called ringing? Then when I hang up the phone starts ringing and I answer it to hear the dial tone. |
| 22:28.44 | ManxPower | i.e. "Call 1-800-BLOW-JOB, $2.99/min will be billed to your phone." |
| 22:28.54 | bhima | manx: those exist? |
| 22:29.23 | Eggplant | bhima, oh yes they exist, and they are evil |
| 22:29.36 | Qwell[] | Libila: Are you calling through the right type of ports? |
| 22:29.42 | Eggplant | AOL's 1800 dialup number does that |
| 22:30.47 | Libila | Qwell[]: I'm pretty new to all this. I think your talking about the FXO ports I have on my TDM04B? I have the first port hooked up to my normal telephone line. |
| 22:30.57 | Qwell[] | heh |
| 22:31.07 | Qwell[] | What is the phone plugged into? |
| 22:31.19 | Libila | LAN |
| 22:31.41 | Libila | it's registered with asterisk, I can make normal extension calls to another phone. |
| 22:32.13 | Tamarisk | Hi can someone tel me what libpri-1.2.2.tar.gz provides? |
| 22:33.45 | ManxPower | Tamarisk, it provides support for PRI lines |
| 22:34.56 | robin_z | meep! |
| 22:35.06 | Tamarisk | ManxPower. Thanks so for no ISDN cards I should not need it? |
| 22:35.32 | ManxPower | Tamarisk, I always install it anyway, but you should not need it unless you are using ISDN PRI |
| 22:36.10 | Tamarisk | OK just trying to figure why I have make errors and no package at the end. |
| 22:36.34 | robin_z | hey, watch out for this scam thats going around the supermarkets at the moment .. two girls offer to pack your bags and carry them in exchange for a lift to another nearby shop, once in your car one of them gives you a blow-job, afterwards you are so distracted, you fail to notice they steal your shopping whne they get out. A nasty trick indeed. |
| 22:36.39 | Tamarisk | I am just trying to install the asterisk 1.2.4 No Zaptel or any add-ons |
| 22:36.42 | robin_z | I fell for it last Tuesday |
| 22:36.46 | robin_z | wednesday |
| 22:36.49 | robin_z | friday |
| 22:36.53 | robin_z | and twice on saturday |
| 22:37.07 | bhima | http://www.ftc.gov/bcp/conline/pubs/tmarkg/tollfree.htm |
| 22:37.31 | mishehu | robin_z: if they're hot and you bought some cheap ass shopping, it'd pay off. |
| 22:37.42 | bhima | Billing you without a prior agreeement and a security method of some kind is illegal. |
| 22:38.47 | mishehu | robin_z: REALLY. I didn't know that. |
| 22:38.57 | bhima | robin_z: Given how much I think blow jobs probably cost, and how much my groceries cost, I'm not sure that's a bad scam. |
| 22:39.38 | mishehu | robin_z: but alas, if it was a joke and you need to label as a joke, it obviously wasn't a ha ha funny type of joke. |
| 22:40.27 | robin_z | see! |
| 22:40.39 | bhima | yeah, I thought it was funny too. |
| 22:41.07 | crusher | Thanks for the advice robin_z. Too bad I don't own a car.... |
| 22:41.35 | robin_z | anyway ... ;) |
| 22:42.04 | *** part/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 22:42.13 | Tamarisk | You can always try a push bike and have one sit on the handlebars! |
| 22:42.43 | robin_z | I am reasonably certain I wold crash ... |
| 22:43.30 | Tamarisk | stabilizers! |
| 22:44.20 | mikefoo | How would I be able to call into a pbx, then fwd to another party, and completely disconnect from the two callers, hence not be billed for their talk time. this possible? |
| 22:44.29 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:44.50 | Qwell[] | mikefoo: No |
| 22:45.05 | mikefoo | ahh any way that I can make that happen to some degree? |
| 22:45.12 | crusher | Unless your SIP provider offers free calls |
| 22:45.13 | Qwell[] | and not be billed? no |
| 22:45.21 | jhiver | mikefoo, if you're using SIP and reinvite it could be ok |
| 22:45.28 | jhiver | if you're using PSTN, no way |
| 22:45.38 | jhiver | you'll have to pay for both ways |
| 22:45.51 | Qwell[] | SIP will still charge you |
| 22:45.56 | jhiver | guys have you tried out hamachi? |
| 22:46.06 | mikefoo | my pbx would be * on sip, with a providers doing the terminating for me. |
| 22:46.23 | jhiver | it looks like quite interesting because it's able to establish NAT <=> NAT VPN sessions and that would be pretty handy for VoIP |
| 22:46.28 | mikefoo | so no way huh... |
| 22:46.34 | Mavvie | hmmm... wonder which dialplan variable is holding the context on which the calls comes in. |
| 22:46.44 | Qwell[] | ${CONTEXT} ? |
| 22:47.10 | Mavvie | Qwell[]: nope, that one holds the current include or macro. |
| 22:47.23 | Mavvie | <PROTECTED> |
| 22:47.39 | Qwell[] | ${MACROCONTEXT} |
| 22:47.54 | ManxPower | Mavvie, SetVar(SAVED_CONTEXT=${CONTEXT}) |
| 22:48.17 | ManxPower | README.variables is your friend |
| 22:49.10 | jhiver | I wonder how they do it though |
| 22:49.41 | NovceGuru | http://pastebin.com/560231 I assume that problem isn't on my end? |
| 22:51.05 | Mavvie | ManxPower: yes, but then I have to add it to all numbers before they jump into the macros. |
| 22:52.01 | ManxPower | Mavvie, I'll bet MACRO_CONTEXT would work, but again, README.variables is your frien |
| 22:52.01 | ManxPower | d |
| 22:54.05 | Dr-Linux | justinu |
| 22:54.09 | NovceGuru | it gives me the "all circuits are busy now" error |
| 22:54.28 | Mavvie | ManxPower: that one still gives me the calling context, not the initial context. |
| 22:55.12 | jhiver | http://en.wikipedia.org/wiki/Hamachi |
| 22:55.24 | Mavvie | there is no such a thing yet. |
| 22:55.44 | ManxPower | Mavvie, Yup. Just do this, make the context the call comes into very limited, i.e. an exten => line to match any call, do the setvar, then go a goto ${EXTEN} in the real context to do everything else. |
| 22:56.22 | ManxPower | You can pretty much to any processing before call routing doing it that way |
| 22:56.55 | ManxPower | to == do |
| 22:57.07 | Mavvie | aha, that's a nice workaround. thanks. |
| 22:57.07 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 22:57.37 | ManxPower | Mav have you ever used variable "subscripts" in extensions.conf? |
| 22:57.58 | Mavvie | I'm not familiar with the term. |
| 22:58.34 | Mavvie | (but that doesn't mean that I might not have used it) |
| 22:59.39 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 23:00.39 | ManxPower | Mavvie, http://pastebin.ca/42002 |
| 23:00.59 | ManxPower | Mavvie, they are NOT subscripts, but they look like them |
| 23:01.56 | Dr-Linux | ManxPower: whats path of README.variable ? |
| 23:02.04 | Mavvie | to me it looks like a macro, but my guess is that it's different. |
| 23:02.33 | ManxPower | <PROTECTED> |
| 23:02.54 | Dr-Linux | hhm.. cool |
| 23:02.54 | Mavvie | you call this with "include => auto-attentdent" ? |
| 23:03.05 | ManxPower | sorry, that's not a good paste for the example |
| 23:04.30 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 23:04.36 | ManxPower | This is much more complex: http://pastebin.ca/42003 |
| 23:04.38 | *** join/#asterisk santiago (n=santiago@63.245.86.219) |
| 23:04.51 | justinu | what does this message mean? |
| 23:04.52 | justinu | Feb 16 08:47:06 ERROR[21778] app_dial.c: Could not stop autoservice on calling channel |
| 23:05.09 | *** part/#asterisk santiago (n=santiago@63.245.86.219) |
| 23:05.40 | *** join/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17) |
| 23:05.46 | TuckerAdelaide | #thebook |
| 23:05.51 | Dr-Linux | justinu: did you google it? |
| 23:05.54 | justinu | no |
| 23:06.09 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-113-60.dsl.tulsok.swbell.net) |
| 23:06.12 | *** part/#asterisk bkw_ (n=brian@ppp-70-128-113-60.dsl.tulsok.swbell.net) |
| 23:06.14 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 23:06.16 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-113-60.dsl.tulsok.swbell.net) |
| 23:06.18 | Dr-Linux | looks new to me |
| 23:06.59 | thazza | ~thebook |
| 23:07.01 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 23:07.38 | Winkie | is that the ololly one? |
| 23:08.19 | thazza | if you mean oriely.. Yep. |
| 23:08.33 | Winkie | excellent i didn't know if it was easily available online or if i'd have to google |
| 23:08.37 | Winkie | i'm going to hardcopy it at some point |
| 23:09.14 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 23:09.35 | TuckerAdelaide | ~thebook |
| 23:09.36 | jbot | [thebook] Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 23:09.46 | tzafrir_laptop | What is "the creative commons license"? |
| 23:10.17 | Winkie | it's a licensed designed by web 2.0 retards |
| 23:10.18 | thazza | ~google |
| 23:10.20 | jbot | somebody said google was a search engine found at http://www.google.com/ |
| 23:10.34 | Mavvie | http://en.wikipedia.org/wiki/Creative_Commons |
| 23:10.41 | Mavvie | just had it open for somebody else. |
| 23:10.49 | thazza | lol |
| 23:10.50 | tzafrir_laptop | There are plenty of CC licenses. That specific one actually does not allow commercial distribution (e.g.: to include it in a CD you sell for the cost of burning) |
| 23:11.19 | tzafrir_laptop | Not that I don't consider the release of the bok under that license very generous and helpful |
| 23:12.09 | tzafrir_laptop | how do I edit a jbot item? |
| 23:12.29 | MikeJ[Laptop] | jbot: help? |
| 23:12.31 | jbot | i guess help is /msg jbot help |
| 23:12.39 | MikeJ[Laptop] | heheh\ |
| 23:12.51 | *** part/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17) |
| 23:12.55 | MikeJ[Laptop] | jbot tzafrir is a guy who needs help |
| 23:12.56 | jbot | ...but tzafrir is already something else... |
| 23:12.58 | tzafrir_laptop | didn't get me very far last time I tried it |
| 23:13.10 | MikeJ[Laptop] | jbot: tzafrir |
| 23:13.11 | jbot | extra, extra, read all about it, tzafrir is http://tzafrir.org.il/ |
| 23:13.26 | Dr-Linux | whos owner of jbot ? |
| 23:13.28 | tzafrir_laptop | ~help |
| 23:13.29 | MikeJ[Laptop] | jbot: no tzafrir needs help |
| 23:13.52 | MikeJ[Laptop] | jbot: tzafrir |
| 23:13.53 | jbot | from memory, tzafrir is http://tzafrir.org.il/ |
| 23:13.57 | MikeJ[Laptop] | jbot: no tzafrir needs help |
| 23:14.13 | MikeJ[Laptop] | jbot: tzafrir |
| 23:14.15 | jbot | somebody said tzafrir was http://tzafrir.org.il/ |
| 23:14.19 | MikeJ[Laptop] | grrr |
| 23:14.22 | MikeJ[Laptop] | oh |
| 23:14.23 | [av]bani | http://www.vonmag.com/images/issue/jul05/depts/tjo-avaya.jpg |
| 23:14.26 | [av]bani | looks like a nice phone |
| 23:14.27 | MikeJ[Laptop] | jbot: no tzafrir is needs help |
| 23:14.29 | jbot | MikeJ[Laptop]: okay |
| 23:14.33 | MikeJ[Laptop] | there we go |
| 23:14.36 | MikeJ[Laptop] | jbot: tzafrir |
| 23:14.38 | jbot | it has been said that tzafrir is needs help |
| 23:14.43 | MikeJ[Laptop] | :P |
| 23:15.14 | FuriousGeorge | anyone using mog's asterisk-xmpp? |
| 23:15.32 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 23:15.53 | Dr-Linux | anyone is using wake up call? |
| 23:16.09 | tzafrir_laptop | jbot, forget tzafrir |
| 23:16.39 | tzafrir_laptop | jbot, no, tzafrir is http://tzafrir.org.il/ |
| 23:16.41 | jbot | tzafrir_laptop: okay |
| 23:16.47 | Tamarisk | Help with install from source file. I get: - checking for tgetent in (various) Configure error termcap support not found |
| 23:17.21 | Tamarisk | I have termcap libs installed but can not see what provides tgetent? |
| 23:17.29 | Winkie | Tamarisk: do you have termcap devel libs? |
| 23:18.03 | Mavvie | ncruses-devel. |
| 23:18.04 | Tamarisk | Winkie. I will check what I have in yast please hold! |
| 23:18.22 | Mavvie | ncurses-devel |
| 23:19.20 | Tamarisk | Sorry is ncurses-devel what i need? |
| 23:19.21 | *** part/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 23:19.27 | Mavvie | yes |
| 23:19.28 | harryvv | yes |
| 23:19.32 | tzafrir_laptop | ~thebook |
| 23:19.34 | jbot | it has been said that thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 23:19.37 | Mavvie | see the README in the root directory. |
| 23:19.40 | Tamarisk | OK looking for that as well |
| 23:19.47 | tzafrir_laptop | (note the "a") |
| 23:20.40 | tzafrir_laptop | Mavvie, you probably know that "ncurses" is "new curses". |
| 23:21.09 | Tamarisk | I have the termcap libary and termcap debug info installed |
| 23:21.15 | tzafrir_laptop | Which probably comes natural with the build process |
| 23:21.19 | Tamarisk | I am now installing ncurses-deval |
| 23:21.35 | Tamarisk | and debug info? |
| 23:21.49 | tzafrir_laptop | Not needed for building |
| 23:21.50 | klasstek | Why does IAX -> SIP -> ZAP suck so much when SIP -> SIP -> ZAP works great? |
| 23:22.21 | tzafrir_laptop | klasstek, is there transcoding in the SIP>SIP? |
| 23:22.29 | Tamarisk | the debug info not needed? |
| 23:22.36 | tzafrir_laptop | no |
| 23:22.53 | klasstek | no transcoding but the media path is not released for other reasons |
| 23:23.32 | Tamarisk | OK may be a bit late to stop now but should not hinder. I hope |
| 23:25.19 | Tamarisk | getting further then ever before |
| 23:26.18 | Tamarisk | plenty of missing prototypes but still going |
| 23:28.04 | Mavvie | hey... looks like the dial command has become smarter. |
| 23:28.25 | Mavvie | you can now do more than one trunk there. |
| 23:28.33 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 23:29.29 | Tamarisk | Bummer! Cannot find -lssl collect2: id returned 1 exit status |
| 23:30.10 | Mavvie | Tamarisk: check the README, search for ncurses-devel and see what other libraries you need to install. |
| 23:30.12 | justinu | bani: it's weird that the snom internal ring tone sounds just like the asterisk generated tones |
| 23:30.19 | justinu | but not the dial/busy/congestion |
| 23:30.31 | Tamarisk | OK |
| 23:31.05 | tzafrir_laptop | Tamarisk, openssl-devel or something in the neibourhood |
| 23:32.39 | *** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net) |
| 23:33.01 | Tamarisk | <PROTECTED> |
| 23:34.28 | *** join/#asterisk ctooley (n=ctooley@c-67-187-102-122.hsd1.tx.comcast.net) |
| 23:34.39 | ctooley | Anyone in the Dallas area looking for work? |
| 23:35.54 | Tamarisk | missing the openssl-deval had openssl but not the libs thanks will now install and try again |
| 23:36.19 | *** part/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 23:36.24 | FuriousGeorge | is there any way to check the version of the zaptel module that is laoded |
| 23:37.07 | tzafrir_laptop | Tamarisk, generally if there is one missing prototype you're missing some headers and usually that means a missing -devel package (or an rpm-based system) |
| 23:37.23 | backblue | klasstek: i have sip-iax-zap and it works great |
| 23:37.36 | klasstek | other way |
| 23:37.41 | klasstek | iax-sip-zap |
| 23:37.47 | tzafrir_laptop | klasstek, what version of *? |
| 23:37.51 | klasstek | zap is pri on t400p |
| 23:37.51 | backblue | i dont use iax phones |
| 23:38.00 | backblue | my zap is bri |
| 23:38.02 | klasstek | hrmmm.... one moment |
| 23:38.19 | klasstek | iax-sip is stable 1.2.4 |
| 23:38.25 | backblue | i even do sip-iax2-sip-zap |
| 23:38.27 | Tamarisk | Ok thanks for that information I will try and file away in the gret matter. Suse10 provides 1.09-4 all docs or articles relate to 1.2.4 hence trying to install from source |
| 23:38.28 | backblue | and it works great |
| 23:38.52 | klasstek | sip-zap shows CVS-v1-0-06/04/05-21:26:28 |
| 23:39.30 | klasstek | iax-sip works fine and sounds great |
| 23:39.39 | klasstek | sip-sip-zap works fine and sounds great |
| 23:40.00 | klasstek | iax-sip-zap sounds terrible to the zap side. sounds fine on the iax side |
| 23:40.31 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 23:40.52 | *** join/#asterisk saftsack (n=saftsack@p54A7FEB2.dip.t-dialin.net) |
| 23:40.53 | saftsack | hi |
| 23:41.02 | *** join/#asterisk robbie2 (n=rob@howzat.dsl.onthenet.net) |
| 23:41.09 | robbie2 | mornings |
| 23:41.10 | Tamarisk | tzafrir_laptop. just reading your comment through again when you say a rpm based system. Suse is I think is that an issue |
| 23:41.53 | robbie2 | i have a TE110P, everything is working excellent, except people tell me there is a background crackling when i call them |
| 23:42.05 | robbie2 | its connected to an ISDN10 in australia |
| 23:42.10 | robbie2 | using asterisk 1.2.4 |
| 23:42.15 | tzafrir_laptop | Tamarisk, get the asterisk 1.09 srpm and make sure you have all the build dependencies installed |
| 23:43.11 | Tamarisk | I can and have installed 1.09 from Suse it does install. I then removed it all to start fresh with 1.2.4 |
| 23:43.17 | tzafrir_laptop | That is: when you run: rpmbuild -t (is that the right flag for extracting only) it will not fail |
| 23:43.41 | tzafrir_laptop | This is something that can be done by a normal user in a different dir. |
| 23:44.14 | tzafrir_laptop | (At least the equivalent in Debian is the recommended practice even for non-package builds) |
| 23:44.45 | saftsack | hi is somebody here how has a hfc chipset based isdn card? |
| 23:45.28 | Tamarisk | rpmbuild, I can just about remember that from RH many years ago. Yast is the automatic installer in Suse |
| 23:45.39 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-37.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:46.00 | ctooley | If anyone's interested in the Dallas area job email me at ctooley@gmail.com |
| 23:46.31 | Tamarisk | It manages the download off the web for me etc and make life very simple and I need it at this rate |
| 23:51.49 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 23:52.57 | *** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl) |
| 23:54.20 | *** join/#asterisk ome3 (n=ome@69.90.135.67) |
| 23:54.38 | ome3 | Anyone know of any cheap providers which would allow me to move my lines from BS to them? |
| 23:55.05 | FuriousGeorge | anyone using bristuff? |
| 23:55.41 | Tamarisk | Whooop Whooop!! |
| 23:55.51 | Tamarisk | Asterisk Installation complete |
| 23:56.11 | ome3 | anyone here moved phone numbers to a provider? i juts asked in #nufone although nobody seems to be there |
| 23:56.37 | FuriousGeorge | i changed zaptel-1.2.3 to 1.2.4 in the bristuff install script, and eveyrthing went ok with the patching, but i got to wondering if it was a bad idea |
