| 00:00.19 | rene- | Strom_C: couldnt reach digium today.. |
| 00:00.34 | Strom_C | rene-: ? |
| 00:00.39 | Strom_C | +1 256 428 6000 |
| 00:01.14 | JT | does Echo() introduce a certain echo component, or just the minimum caused by bandwidth and computing power? |
| 00:01.14 | rene- | couldnt reach tech support |
| 00:01.14 | rene- | no tech available |
| 00:01.16 | Strom_C | rene-: odd; you didnt wait in the queue? |
| 00:01.25 | rene- | straight to vm |
| 00:01.26 | rene- | no queue |
| 00:01.38 | rene- | they are probably at a party or something :) |
| 00:01.43 | Strom_C | or just busy |
| 00:01.54 | rene- | probably |
| 00:02.12 | *** join/#asterisk lters (n=tech@mrtcdsl-433.mis.net) |
| 00:02.49 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:05.33 | rene- | no slips/alarms/resets for 5 and a half hours. hopefully stays that way for another 10 |
| 00:05.45 | rene- | i have resetinterval to never |
| 00:06.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:06.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:06.40 | *** join/#asterisk flujan (n=flujan@201-42-124-150.dsl.telesp.net.br) |
| 00:07.25 | rene- | probably udma mode 3 is working a small miracle here |
| 00:08.15 | rene- | asterisk can go nuts for all i care as log as it does it from 9PM to 6 Am |
| 00:08.21 | rene- | s/log/long/ |
| 00:09.54 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:11.44 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 00:14.20 | flujan | hi guys... i enable ami in my manager.conf... how can I use telnet to log into it? |
| 00:15.48 | russellb | telnet asteriskhost 5038 |
| 00:16.19 | russellb | ^] |
| 00:16.22 | russellb | set crlf |
| 00:17.23 | russellb | Action: Login |
| 00:17.27 | russellb | Username: whatever |
| 00:17.33 | russellb | Secret: whatever |
| 00:17.38 | russellb | hit enter again ... |
| 00:17.59 | blitzrage | russellb: wow, that's useful to know |
| 00:18.08 | russellb | blitzrage: :) |
| 00:19.31 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:20.14 | russellb | Action: Ping |
| 00:20.22 | russellb | that's pretty useful. |
| 00:21.27 | opus_ | Action: getTheseMotherFuckinSnaeksOffTheMotherFuckingPlane |
| 00:21.43 | blitzrage | russellb: thanks, going to post that to my site |
| 00:21.56 | russellb | opus_: lol |
| 00:21.59 | russellb | blitzrage: sweet |
| 00:22.00 | blitzrage | venomous serpants upon thine aerocoach |
| 00:22.08 | opus_ | heh |
| 00:22.08 | blitzrage | russellb: you'll get props obviously |
| 00:22.12 | russellb | yay! |
| 00:22.31 | flujan | ok... :) |
| 00:22.31 | flujan | rhanks russellb |
| 00:22.31 | flujan | now that i know it is working i have a doubt... :) |
| 00:22.35 | flujan | i am using realtime queues |
| 00:22.45 | flujan | but the statistics in the queues table are not been updated |
| 00:22.53 | rene- | read upon the wiki about asterisk manager actions |
| 00:22.59 | flujan | hi rene |
| 00:23.01 | flujan | :) |
| 00:23.03 | flujan | me again |
| 00:23.05 | rene- | hi flujan |
| 00:23.14 | flujan | i am having actions in the ami |
| 00:23.18 | russellb | blitzrage: yeah, the "set crlf" is key ... |
| 00:24.24 | russellb | blitzrage: once we have SIP over TCP support, you could set up a session with telnet, too :) |
| 00:24.43 | russellb | i'm not sure why you would ... but ... |
| 00:24.46 | rene- | flujan: issue a QueueStatus action |
| 00:25.20 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:25.38 | rene- | i dont know if realtime queues update this information but you could try |
| 00:25.55 | rene- | Action: QueueStatus |
| 00:26.07 | blitzrage | russellb: yah... I'm not really sure why its useful, but hey :) |
| 00:26.51 | flujan | rene-, i am always getting this messages: http://pastebin.ca/156762 |
| 00:26.54 | russellb | blitzrage: i just use it sometimes if i have to debug a problem with manager events |
| 00:28.10 | blitzrage | russellb: makes sense |
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| 00:29.00 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 00:29.09 | rene- | ok oficially my asterisk box pri cant past 6 hours without going into reset channel hell |
| 00:29.25 | russellb | rene-: support@digium.com |
| 00:29.37 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
| 00:30.10 | awe6 | What parameter in extensions.conf will search both last and first names using *411? |
| 00:30.22 | flujan | guys, another question... any of yours are getting problems with a TE406P board? |
| 00:30.57 | *** join/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 00:34.13 | *** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek) |
| 00:35.14 | opus_ | flujan, you should contact digium support |
| 00:35.18 | flujan | rene-, please take a look at http://pastebin.ca/156777 |
| 00:35.20 | opus_ | or try the mailling list |
| 00:36.18 | flujan | opus_, hum... I am in Brazil... The reseller said these boards have problems and digium is recalling them... is it true? |
| 00:37.13 | flujan | rene-, ? As you can see the queues are been updated... But not the table queues in my database. |
| 00:37.31 | tessier__ | Anyone know what determines which SIP identity is used on a Snom phone when you just start dialing as opposed to using one of the function buttons? |
| 00:37.38 | rene- | flujan: i believe |
| 00:37.58 | rene- | that you need to have at least one member in one of those realtime queues for them to report |
| 00:38.23 | opus_ | flujan, i do not use digium hardware so i am not sure |
| 00:38.25 | flujan | rene-, but i have a member in this queue. |
| 00:38.38 | flujan | opus_, which hardware do you recommend to work with asterisk? |
| 00:38.50 | opus_ | sangoma |
| 00:38.56 | JT | i doubt they'd tell you there was a problem with one of their boards if they didn't really think so |
| 00:40.48 | flujan | JT, for sure... But you never knows... It is sad to digium put such boafd in the market... :( |
| 00:41.10 | flujan | rene-, You mean a online member? |
| 00:41.22 | Strom_C | flujan: Digium is not recalling anything |
| 00:41.49 | JT | everyone puts out a dud every once in a while, if you're paranoid, might be wise to hold off using a new piece of hardware until people have been using it for a while |
| 00:41.50 | rene- | flujan: yes |
| 00:41.55 | rene- | logged in to the quee |
| 00:42.49 | flujan | JT, I am not paranoid... but once my boss get angry because the call center stopped... :P |
| 00:42.57 | JT | heh |
| 00:42.57 | *** join/#asterisk ToyMan (n=stuq@pool-70-22-158-57.bos.east.verizon.net) |
| 00:43.04 | JT | a reason to be paranoid... |
| 00:43.35 | l-fy | sangoma rulez |
| 00:44.04 | flujan | JT, lol... :) |
| 00:44.04 | *** join/#asterisk dalekurt (n=dalekurt@port0033-afw-adsl.cwjamaica.com) |
| 00:46.45 | dalekurt | Anyone on FWD |
| 00:46.52 | Strom_C | I am, theoretically |
| 00:47.02 | dalekurt | Wanna test my setup |
| 00:47.13 | dalekurt | can you call me or vice versa |
| 00:49.42 | flujan | rene-, no success... i tried it with a logged member |
| 00:51.42 | flujan | rene-, http://pastebin.ca/156790 |
| 00:51.59 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:52.18 | rene- | flujan: dont know what to do next man, what i would try is generating a file with the queue database data, #including it fromo queues.conf and issuing a reload app_queue.so from cli whenever you made changes |
| 00:53.53 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:54.05 | *** join/#asterisk Mathesis (n=chatzill@unaffiliated/mathesis) |
| 00:54.11 | Mathesis | hello |
| 00:54.27 | *** join/#asterisk mog_home (n=mogorman@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:54.27 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 00:54.40 | Mathesis | where is a channel spanish? |
| 00:55.09 | Strom_C | I think I crashed my polycom phone :/ |
| 00:56.18 | Qwell | Strom_C: nub :p |
| 00:56.39 | Strom_C | nub indeed |
| 01:01.24 | flujan | rene-, I get what was wrong. |
| 01:01.41 | flujan | realtime queues do not store information about the queues in the data base... just read it. |
| 01:02.05 | flujan | I were waiting it to record queues statistics in the database... |
| 01:02.18 | flujan | anyone can say if is it possible? |
| 01:02.34 | l-fy | flujan > not in asterisk |
| 01:03.42 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:05.22 | flujan | l-fy, man... now you break my heart... :( |
| 01:06.00 | flujan | l-fy, hum, but I can create a app to read the ami output and store it... Is it correct? |
| 01:06.13 | l-fy | flujan > no is not corect |
| 01:06.16 | l-fy | i'm not a man |
| 01:06.48 | flujan | l-fy, ops... I am sorry. :) |
| 01:07.19 | flujan | l-fy, so you break my heart twice... :D |
| 01:07.54 | *** join/#asterisk sdenike (n=shelby@h-66-166-5-222.lsanca54.covad.net) |
| 01:07.57 | l-fy | well, this why we have women for |
| 01:08.06 | l-fy | to break people heart |
| 01:08.17 | l-fy | i know a few people which felt like you when i've started yate |
| 01:09.53 | Qwell | flujan: Why not read from the queue log? |
| 01:10.30 | sdenike | can someone here help me out getting my zaptel module to compile correctly on a trixbox setup? |
| 01:10.43 | Strom_C | sdenike: see the following: |
| 01:10.45 | Strom_C | ~centosbug |
| 01:10.56 | jbot | somebody said centosbug was a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 01:12.02 | sdenike | Still the same issue |
| 01:12.09 | opus_ | well, its not really a centos bug.. it was a red hat bug |
| 01:12.14 | sdenike | i am getting a few different errrors when compiling |
| 01:12.17 | sdenike | <PROTECTED> |
| 01:12.17 | sdenike | /usr/src/zaptel/zaptel.c:188: warning: 'fcstab' defined but not used |
| 01:12.17 | sdenike | <PROTECTED> |
| 01:12.17 | sdenike | <PROTECTED> |
| 01:12.17 | sdenike | /usr/src/zaptel/torisa.c:1145: warning: 'set_tor_base' defined but not used |
| 01:12.29 | Strom_C | sdenike: thats not a fatal error |
| 01:12.34 | opus_ | centos just removed 'red hat' from the files and replaced it with 'centos' |
| 01:12.39 | *** part/#asterisk Mathesis (n=chatzill@unaffiliated/mathesis) |
| 01:13.11 | flujan | l-fy, I truly believe we have women to love... :D |
| 01:13.19 | flujan | Qwell, where i can grab this information? |
| 01:13.32 | Qwell | flujan: /var/log/asterisk/queue_log |
| 01:14.03 | sdenike | Storm_C: if thats the case that leads me to my next issue. I am able to dial into the system but I get no audio feedback at all. Also the zttool shows that the Alarm is RED for span 1, and Alarm is UNCONFIGURED for span 2 |
| 01:14.06 | opus_ | l-fy, freeswitch4live |
| 01:14.14 | opus_ | l-fy, freeswitch4lif3 :) |
| 01:14.33 | opus_ | l-fy i hear that you guys got into some kind of argument where bkw was banned from #yate |
| 01:14.48 | Qwell | Strom_C: so, should I assume a raincheck for yesterday? :D |
| 01:15.01 | Strom_C | um...yes! |
| 01:15.22 | harryvv | How long has cvs.digium.com been down? |
| 01:15.30 | Strom_C | months? |
| 01:15.34 | harryvv | :) |
| 01:15.37 | Qwell | Strom_C: let me know when you aren't busy - preferably earlier in the day. :D |
| 01:15.43 | Strom_C | alright |
| 01:15.52 | Strom_C | this week suddenly turned busy, sadly |
| 01:15.52 | Qwell | rather, preferably let me know earlier |
| 01:15.55 | harryvv | I was trying to get some more sounds and have not used cvs in a long time. |
| 01:16.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:16.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:16.12 | opus_ | *** #yate RoyK!*n=roy@80.239.107.70 l-fy!n=pchitesc@yate/developer/l-fy 1156839880 |
| 01:16.17 | opus_ | *** #yate *!*n=bkw_@*asterisk/friend-and-developer/bkw killall-9!n=paulc@diana.null.ro 1157064369 |
| 01:16.19 | harryvv | Strom you know of a alternative repository of sounds I can download |
| 01:16.24 | opus_ | haha |
| 01:16.29 | Qwell | harryvv: svn |
| 01:16.29 | Strom_C | harryvv: use svn.digium.com |
| 01:16.33 | harryvv | k |
| 01:16.34 | pyrom | Strom_C, Think that there might be something wrong with the linksys spa3102, might be the firmware, getting another tomorrow going to thest it before i upgrade the firmware. |
| 01:17.17 | harryvv | and /usr/svnroot ? |
| 01:17.20 | opus_ | *** Total number of bans on #asterisk - 40 |
| 01:17.29 | opus_ | *** Total number of bans on #yate - 6 |
| 01:17.53 | opus_ | *** Total number of bans on #freeswitch - 2 |
| 01:18.04 | sdenike | Strom_C: any idea where I should check to fix that? I have been at this all day and haven't progressed past that point |
| 01:18.39 | Strom_C | sdenike: you have a TE205/210P? |
| 01:18.48 | sdenike | 06:00.0 Communication controller: Digium, Inc. Wildcard TE205P (rev 02) |
| 01:19.14 | sdenike | Strom_C: yeah |
| 01:19.21 | russellb | opus_: are you trolling? |
| 01:19.26 | Strom_C | what's on the other end of span 1? |
| 01:19.51 | sdenike | Strom_C: other end as in the provider? |
| 01:20.06 | Strom_C | other end as in the cable |
| 01:20.12 | *** part/#asterisk cornet (n=nathan@cpc1-heck3-0-0-cust893.hudd.cable.ntl.com) |
| 01:20.50 | harryvv | whats the export. It comes up asking for a password and using anoncvs or should it be anoncvn |
| 01:21.04 | Strom_C | harryvv: www.asterisk.org |
| 01:21.07 | Strom_C | read directions |
| 01:21.09 | sdenike | Strom_C: 23 channel+ T1 |
| 01:21.10 | harryvv | k |
| 01:21.19 | Strom_C | sdenike: a PRI? |
| 01:21.24 | sdenike | Strom_C: yes |
| 01:21.32 | Strom_C | sdenike: this is a large business install? |
| 01:21.48 | sdenike | Strom_C: yeah its a phone system linking our 2 offices |
| 01:21.50 | harryvv | Strom was refering to more of a out of date site thanks :) |
| 01:22.02 | Strom_C | sdenike: and you're running TRIXBOX??? |
| 01:22.37 | sdenike | Strom_C: Not by my choice but yeah. Do you suggest other? I would have went with a gentoo isntall myself but it was not my call |
| 01:22.48 | sdenike | Strom_C: you have a better suggestion? |
| 01:22.51 | Strom_C | sdenike: trixbox is a TOY |
| 01:23.11 | Strom_C | start with asterisk 1.2 on [linux-distro-of-your-choice] and go from there |
| 01:23.46 | Strom_C | I would strongly advise you not to run a business on trixbox |
| 01:24.02 | harryvv | sdenike are you the admin or consultant |
| 01:24.06 | sdenike | Strom_C: why so can you give me some reasons to back it up with so that I can try to get them to change away from it |
| 01:24.33 | sdenike | harryvv: I am jsut a jr. admin told to use this setup by the sr. admin ... |
| 01:24.40 | harryvv | make the pbx as thin as possible :) |
| 01:24.45 | harryvv | fast |
| 01:25.05 | sdenike | its running on a decent machine that should be able to handle it. 2.8ghz 1 gig ram sata drives |
| 01:25.09 | Strom_C | The GUI severely limits your flexibility; trixbox's extra garbage adds a LOT of overhead; administration of such a system is difficult at best and impossible at worst |
| 01:25.20 | Strom_C | plus support is harder to come by |
| 01:25.47 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 01:26.07 | sdenike | Once up and running it needs to have an easy to use GUI as the call center supervisor will be the one managing it once its setup |
| 01:26.18 | harryvv | for some reason tar runs in the background of AAH and eats up cycles killing a otherside good conversaion. I dont know if thats a factor in latter editions including trixbox. |
| 01:27.01 | Strom_C | sdenike: if the call center supervisor can't manage asterisk, I suggest either sending him/her in for training immediately or finding someone who can manage asterisk |
| 01:27.09 | sdenike | hehe |
| 01:27.21 | opus_ | russellb, yeah i'm a troll alright |
| 01:27.25 | JT | what is manage? |
| 01:27.30 | harryvv | I agree man what would happen if something in asterkis dies or has issues? |
| 01:27.35 | russellb | opus_: oh, ok, just checking |
| 01:27.36 | JT | do they need to redesign the dialplan at will? |
| 01:27.50 | sdenike | well manage as in setting up new extentions, Q's etc. |
| 01:27.51 | Strom_C | I hear the same excuse every time, and it basically boils down to "someone who is not qualified to administer the phone system is being pressed into administering the phone system" |
| 01:27.53 | russellb | carry on. |
| 01:27.55 | opus_ | :) |
| 01:28.28 | JT | they're things that should be done by a phone system administrator who knows what they're doing |
| 01:28.34 | harryvv | Strom There is somone here who administers a City halls asterisk phone system and thats his only job ;) |
| 01:29.01 | sdenike | Strom_C: well i mean the person that will be making the changes at this point has enough sckill but over expansion the next person to take there place when that person gets moved up might not be as fluent |
| 01:29.01 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 01:29.19 | Strom_C | sdenike: it's still a flimsy excuse |
| 01:29.23 | Qwell | sdenike: Then the new person shouldn't be in that position |
| 01:29.24 | harryvv | sdenike, how many phones will this system run |
| 01:29.26 | opus_ | i always make sure my call center managers know how to use gdb and debug deadlocks, it would be stupid if they didn't have that experience. |
| 01:29.44 | sdenike | harryvv: right now about 6 in one office and a dozen in the other |
| 01:29.57 | JT | with a traditional PABX, you don't ask a call centre manager to install new cables for a new extension and program the |
| 01:29.58 | harryvv | not many to loose your head |
| 01:29.59 | opus_ | i look for managers with 15+ erlang and assembly skills experience only |
| 01:30.00 | JT | PABX |
| 01:30.06 | JT | why should asterisk be any different? |
| 01:30.17 | Qwell | JT: because it's free, right? |
| 01:30.25 | Qwell | That's the excuse I always hear. :) |
| 01:31.02 | Strom_C | JT: because a lot of people get it into their heads that just because asterisk is free means that they should also be able to hire some underpaid underqualified moron to administer it. |
| 01:31.11 | JT | heh |
| 01:31.58 | opus_ | Storm_C those same people end up spending 10 times more then they would if they just went with a closed sourced solution and somebody half as comptitent |
| 01:32.01 | blitzrage | yah... asterisk isn't trivial |
| 01:32.09 | blitzrage | and we're talking about a phone system -- you get what you pay for |
| 01:32.26 | blitzrage | if your phones aren't important to your business, then feel free to cheap out on it |
| 01:34.39 | opus_ | a call center manager is kind of a mcjob anyway though |
| 01:35.31 | blitzrage | call centre manager probably shouldn't be doing anything that isn't nub proof though |
| 01:37.11 | opus_ | hey wait i just realized something.. open source development is kind of a mcjob too |
| 01:37.38 | blitzrage | uh huh |
| 01:39.15 | russellb | opus_: hey troll ... stfu, kthx |
| 01:39.43 | *** join/#asterisk Brijn (n=Bas@S0106004063c0fa1f.vn.shawcable.net) |
| 01:40.10 | Altair256 | it's half our fault too russellb... we shouldn't be feeding the trolls |
| 01:40.17 | opus_ | russellb, bite me |
| 01:40.40 | Altair256 | what a waste to come into a support channel and complain about stuff like this opus_, please get a life |
| 01:40.53 | *** mode/#asterisk [+b %opus_!*@*] by russellb |
| 01:41.05 | techie | oh oh |
| 01:41.20 | Altair256 | lol, score one for the good guys |
| 01:41.23 | anthonyl | that makes me happy :) |
| 01:41.26 | *** join/#asterisk Brijn (n=Bas@S0106004063c0fa1f.vn.shawcable.net) |
| 01:41.38 | Altair256 | still gotta kick him |
| 01:41.44 | anthonyl | why |
| 01:41.44 | russellb | no need. |
| 01:43.07 | Altair256 | well... good times, everyone have a good night! :) |
| 01:43.28 | *** part/#asterisk opus_ (i=opus@pabstblueribbon.net) |
| 01:44.56 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:49.44 | _DAW | is dns srv properly implemented in trunk? |
| 01:51.30 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 01:53.06 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 01:53.12 | Rahail | hello people |
| 01:53.57 | *** part/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 01:54.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:54.35 | file | [TK]D-Fender: go to bed |
| 01:55.46 | [TK]D-Fender | file : In about 1/2 hour... waking up at 5:30 tomorrow |
| 01:57.52 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 01:57.52 | DasTech | ok hello |
| 01:57.52 | DasTech | having issue |
| 01:57.57 | DasTech | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 01:57.57 | DasTech | See `config.log' for more details. |
| 01:59.04 | DasTech | but the log looks fine |
| 01:59.57 | *** join/#asterisk dalekurt (n=dalekurt@port0033-afw-adsl.cwjamaica.com) |
| 02:00.45 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 02:01.53 | blitzrage | DasTech: yum install gcc4-c++ |
| 02:02.28 | DasTech | ok |
| 02:03.45 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:04.10 | DasTech | gcc version 3.4.6 20060404 (Red Hat 3.4.6-3) |
| 02:04.21 | DasTech | so do I do3 insted of 4 |
| 02:04.55 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 02:05.14 | blitzrage | well, I'm using Fedora Core 5, so you might not have version 4 |
| 02:06.58 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.25 | DasTech | I am on centos |
| 02:07.37 | DasTech | so I just did gcc-c++ |
| 02:07.46 | DasTech | and its installing |
| 02:08.11 | Rahail | i need help |
| 02:08.24 | Rahail | how do i see trunk name on asterisk clit |
| 02:08.26 | Rahail | cli |
| 02:09.31 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 02:10.13 | blitzrage | lol |
| 02:10.17 | blitzrage | show version |
| 02:10.45 | Strom_C | hahaha |
| 02:10.46 | Strom_C | asterisk clit |
| 02:10.52 | Strom_C | classic |
| 02:11.04 | blitzrage | :D |
| 02:11.27 | blitzrage | the asterisk cli is the closest to a clit I bet a bunch of people in here get :) |
| 02:11.36 | Rahail | lol |
| 02:11.37 | Rahail | my bad |
| 02:11.47 | Strom_C | hahahaha, oh man, that's hilarious AND horrible |
| 02:11.53 | blitzrage | I try |
| 02:12.15 | blitzrage | my work here is done |
| 02:12.36 | dalekurt | lol |
| 02:15.10 | [TK]D-Fender | blitzrage : ! ! ! |
| 02:15.10 | blitzrage | [TK]D-Fender: not tonight.. I'm tired |
| 02:15.22 | [TK]D-Fender | blitzrage : A headache that lasts 3 weeks is something serious... see a doctor! |
| 02:15.29 | file | now now you two, get a room |
| 02:15.45 | [TK]D-Fender | file : Is it billed by the hour? |
| 02:15.56 | file | by the minute |
| 02:19.47 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 02:22.46 | shmaltz | anybody here tried remaping keys on the Polycom phones? |
| 02:24.27 | Rahail | any one here using a2billing |
| 02:24.46 | dalekurt | StromC |
| 02:28.32 | Strom_C | hi |
| 02:28.32 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:31.46 | [TK]D-Fender | shmaltz : I've remapped the DND on a 301 to "Messages" one |
| 02:31.53 | [TK]D-Fender | once* |
| 02:34.03 | shmaltz | D-Fender, how would I remap to speed dial? and is it possilbe to remap the soft buttons? |
| 02:34.04 | [TK]D-Fender | shmaltz : to my awareness you can't do either |
| 02:34.35 | shmaltz | D-Fender, I'm trying to create a handsfree intercome option on a button, I have an extension that symply sets a DB value that my DP looks up and based on that sets the Alert_Info, how can I map a button to that feature? |
| 02:36.21 | [TK]D-Fender | shmaltz : Dont use all of your line-keys for appearances and save it for a speed-dial. that'd be 1-touch. |
| 02:36.54 | shmaltz | Fender, nah, they wont be happy, I'm using the 501, and they want 3 :( |
| 02:38.59 | [TK]D-Fender | shmaltz how many calls are these morons using at a time? Its not like this needs to be a limitation even.... |
| 02:39.40 | shmaltz | D-Fender, here: |
| 02:39.42 | shmaltz | http://ww.thinkexist.com/quotation/there_is_only_one_boss-the_customer-and_he_can/263196.html |
| 02:41.06 | Strom_C | bullshit |
| 02:41.11 | Strom_C | there is not only one boss :) |
| 02:41.33 | Strom_C | as a consultant, your job is to provide expert advice, not to enable the customer to shoot himself in the foot |
| 02:41.48 | Strom_C | if the customer does not want to listen to you, that's when you walk away |
| 02:42.12 | [TK]D-Fender | Strom_C : no, its be your job to offer the best solution he can afford from you :) |
| 02:42.57 | [TK]D-Fender | Strom_C : Quite often consultant = whore (in this category feel free to add : graphic artist, Web site designer, phone support monkey, etc) |
| 02:43.04 | shmaltz | Strom_C, while I agree with you most of the time, I can't stop looking at that familys business, and if that quote got it going we should then follow |
| 02:43.53 | Strom_C | [TK]D-Fender: I believe we prefer the term "conslutant" to "whore" |
| 02:45.16 | [TK]D-Fender | Strom_C : Which makes the statement no less true.... |
| 02:45.24 | Strom_C | it was a joke, silly |
| 02:45.27 | Strom_C | humor |
| 02:45.30 | Strom_C | you're supposed to laugh |
| 02:45.38 | [TK]D-Fender | I too friggen tired..... |
| 02:46.35 | [TK]D-Fender | websae : Didn't get to call but that last item is done... I'm on vacation as of now and won't likely be online till late sunday ro monday. |
| 02:53.37 | shmaltz | this is quite interesting, I get the following phishing link in my emails: |
| 02:53.38 | shmaltz | http:// 0xc2669cc4/index.php |
| 02:53.40 | shmaltz | and it translates to: http://195.149.199.137:8888/ |
| 02:53.41 | shmaltz | How is that? |
| 02:54.55 | [TK]D-Fender | shmaltz : hex>4-byte IP |
| 02:55.15 | shmaltz | D-Fender, could be, but it doesn't ranslate to the same IP |
| 02:57.02 | [TK]D-Fender | shmaltz : maybe it doesn't break by byte but does a continuous number broken by 3-decimal increments |
| 02:57.17 | shmaltz | D-Fender you are right, some stupid PHP redirects me |
| 02:57.59 | shmaltz | it's hex>4-byte IP, each 2 digits make out the hex of the dec value for that octec of the IP address |
| 02:58.10 | JT | yep |
| 02:59.18 | *** part/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 02:59.54 | *** join/#asterisk Katty (n=Angela@dialup-4.244.123.12.Dial1.StLouis1.Level3.net) |
| 03:00.34 | *** join/#asterisk lordgoth (n=will@ip68-12-81-212.ok.ok.cox.net) |
| 03:00.44 | lordgoth | what are some good voip wholesalers? |
| 03:01.19 | Katty | voip-supply.com |
| 03:01.22 | shmaltz | lordgoth, define good |
| 03:02.03 | lordgoth | good as in reliable service, allows asterisk |
| 03:02.16 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 03:02.17 | Strom_C | level3 :) |
| 03:02.29 | shmaltz | Strom_C :P |
| 03:02.31 | Katty | also, hi. |
| 03:02.37 | lordgoth | hi |
| 03:02.38 | lordgoth | =) |
| 03:02.39 | smackus | doesnt a red blinking light on a port of my digium t1 card mean a lack of carrier voltage? |
| 03:02.56 | Katty | it means the signal has to stop and wait for a green light |
| 03:03.00 | Katty | and yellow means go veryvery fast. |
| 03:03.07 | smackus | nice |
| 03:03.11 | Nivex | Katty: *chuckle* |
| 03:03.15 | Strom_C | smackus: it means no signal can be found |
| 03:03.34 | Qwell | Strom_C++ |
| 03:03.39 | Strom_C | hi hi |
| 03:03.45 | Katty | Qwell: (= |
| 03:03.47 | Qwell | for continuing the pun, AND successfully answering the question |
| 03:04.01 | Juggie | it can mean alot of things |
| 03:04.07 | shmaltz | smackus, yes it does, so add a 12 volt bettery from your car to the line |
| 03:04.10 | [TK]D-Fender | Katty : And in Quebec : "We may not be able to turn right on a red light but tabarnac we can go right THROUGH IT!@" |
| 03:04.31 | Katty | [TK]D-Fender: :> |
| 03:04.49 | Katty | [TK]D-Fender: thanks with that call recording help the other day (= |
| 03:05.05 | smackus | ok.. .here is what I am trying to do. I have two different phone systems. one asterisk, one something else. both systems are using the exact same t1 configurations. I have been slowly moving the t1s from the old server to asterisk. I want to have the last two t1s plugged into ports 1 and 2 of my asterisk box, and then send out the last few numbers that have to go to the old system out on ports 3 and 4 of asterisk into the t1 card on the other system. i ge |
| 03:05.06 | [TK]D-Fender | Katty : Np, anytime.... |
| 03:05.11 | smackus | ports 1 and 2 in asterisk are correct. |
| 03:05.23 | smackus | and if i move those t1s to 3 and 4 they work there also. |
| 03:05.41 | smackus | i thought i could just put a crossover between ports 3 and 4 and the other phone system and dial out to them. |
| 03:05.49 | smackus | am i wrong? |
| 03:06.01 | Strom_C | smackus: what happens with a straight-through cable? |
| 03:06.03 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:06.05 | shmaltz | smackus, you still got to configure it |
| 03:06.05 | smackus | same thing |
| 03:06.19 | Strom_C | you got the framing and line coding correct? |
| 03:06.26 | Strom_C | and are you using a T1 crossover cable? |
| 03:06.42 | Strom_C | remember, T1 and ethernet don't use the same pins |
| 03:06.50 | smackus | shit... that could be it. |
| 03:07.00 | JT | lol |
| 03:07.02 | smackus | can anyone send me a link to the color patterns for a t1 crossover? |
| 03:07.09 | Strom_C | smackus: one moment |
| 03:07.11 | JT | search voip-info.org |
| 03:07.16 | shmaltz | smakcus, it should be on voip-info.org |
| 03:07.16 | JT | there's 2 pages on it |
| 03:07.31 | smackus | ok.. heading there. thanks |
| 03:07.32 | shmaltz | but anyhow here it is: |
| 03:07.34 | shmaltz | pin 1 > pin 4 |
| 03:07.36 | shmaltz | pin 2 > pin 5 |
| 03:07.46 | JT | like "t1 crossover cable" and "crossover cable t1" or somethin |
| 03:08.07 | JT | 1 > 4 |
| 03:08.08 | Katty | i've never heard of a t1 crossover cable. |
| 03:08.12 | JT | 2 > 5 |
| 03:08.17 | shmaltz | http://www.google.com/search?hl=en&q=crossover+site%3Avoip-info.org&btnG=Google+Search |
| 03:08.17 | JT | 4 > 1 |
| 03:08.18 | shmaltz | http://www.voip-info.org/wiki/view/T1+Crossover+Cable |
| 03:08.21 | Katty | i've heard of a regular cross over cable, which is cat5 |
| 03:08.27 | JT | 5 > 2 |
| 03:08.38 | smackus | thanks for the overwhelming response. JT, I am reading it, thanks |
| 03:09.09 | JT | Katty: they're both cat5, but TIA568A (ethernet) is not the same |
| 03:09.20 | JT | technically T1 connectors are RJ-48C |
| 03:09.35 | JT | they're the same physical 8P8C plugs |
| 03:09.51 | Strom_C | smackus: http://stromcarlson.com/wiring/t1_crossover.png |
| 03:11.06 | JT | Strom_C: is that the image that is meant to appear on voip info? |
| 03:11.12 | Katty | what a bunch of weirdos. |
| 03:11.21 | Strom_C | JT: no, that's one I drew myself |
| 03:11.23 | JT | broken image on voip-info last i checked |
| 03:11.25 | JT | Strom_C: okay |
| 03:11.36 | JT | humans are weird |
| 03:11.46 | shmaltz | Strom_C, you a designer? |
| 03:11.52 | Strom_C | no |
| 03:11.56 | Strom_C | I'm a phone phreak |
| 03:12.04 | anthonyl | ;) |
| 03:12.26 | JT | he needs a big printed reference image when he's out beigeing T1 lines |
| 03:12.43 | Strom_C | idiot, you can't "beige" T1s |
| 03:12.52 | Strom_C | anyway, beige boxes are for kiddies |
| 03:13.00 | shmaltz | Strom_C, with asterisk you could :P |
| 03:13.01 | JT | so you don't like the colour |
| 03:13.05 | Strom_C | i don't do any of that toll fraud garbage |
| 03:13.08 | shmaltz | Strom_C, no it's not |
| 03:13.13 | JT | hey, i can "beige" them if i wanted to |
| 03:13.16 | JT | not that i do |
| 03:13.18 | shmaltz | it's not fraud, its a way to not pay :P |
| 03:13.25 | JT | portable communications analyser |
| 03:13.44 | JT | that does T1/E1, CRO, serial etc |
| 03:14.00 | shmaltz | a butt set is a beige box, fluke makes one that can do it for T1s |
| 03:14.02 | Katty | it's amazing i don't know a thing about what you're talking about right now. |
| 03:14.24 | shmaltz | Katty, you not missing anything you can go to sleep |
| 03:14.28 | Strom_C | shmaltz: "beige box" implies you spent less than $10 on everything |
| 03:14.30 | JT | shmaltz: i have a little think like a chunky laptop which will do it |
| 03:14.36 | Katty | shmaltz: but it's only 10:30 :< |
| 03:14.39 | JT | it cost $12000 in the early 90s |
| 03:14.49 | shmaltz | ok |
| 03:15.18 | shmaltz | Katty, here print it out, take some coocies goto bad and read this: |
| 03:15.20 | JT | of course i used the power of ebay to get it for much less |
| 03:15.20 | shmaltz | http://www.exegesis.uklinux.net/gandalf/phreak/ukbeige.htm |
| 03:15.32 | Strom_C | the power! of ebay! |
| 03:15.42 | JT | indeed |
| 03:15.44 | Katty | shmaltz: i'll have nightmares after i read that. |
| 03:15.50 | JT | all heil |
| 03:15.50 | shmaltz | JT, with something like a VIA, it's possible to make on your own |
| 03:15.56 | shmaltz | Katty, for years |
| 03:16.05 | JT | shmaltz: i'm sure, but effort... |
| 03:17.25 | *** join/#asterisk PCadach[work] (n=PCadach[@www.east.telecom.kz) |
| 03:18.16 | shmaltz | I remember the red boxing years, it was huge |
| 03:18.40 | shmaltz | these stupid dialers used to go for around $200 after the mods |
| 03:18.48 | lordgoth | red boxing was fun |
| 03:19.03 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 03:19.34 | shmaltz | Who overhere lives under big V teritory? |
| 03:20.11 | *** join/#asterisk crashsys (n=kumba@210-208.124-70.tampabay.res.rr.com) |
| 03:20.33 | crashsys | Is Asterisk with a Digium TE205p known to have any problems on i875-based chipsets? |
| 03:20.58 | shmaltz | crashsys, what mobo? |
| 03:21.04 | crashsys | Asus P4C800 |
| 03:21.11 | shmaltz | crashsys, the intel mobos have trouble usualy |
| 03:21.14 | crashsys | 2.4g P4 |
| 03:21.16 | shmaltz | well asus should be ok |
| 03:21.21 | crashsys | Intel-Style? or intel-brand? |
| 03:21.23 | crashsys | ahhh, ok... |
| 03:21.32 | crashsys | is CAS more proned to errors then PRI? |
| 03:21.32 | shmaltz | crashsys, what distro? |
| 03:21.38 | crashsys | Slackware 10.2 |
| 03:21.41 | shmaltz | why do you think you got problems? |
| 03:21.41 | crashsys | 2.6 kernel |
| 03:21.56 | shmaltz | what type of errors? |
| 03:21.57 | crashsys | Cause my span goes down, then instantly back up... all 24-channels go red... |
| 03:22.09 | crashsys | then instantly are back up... |
| 03:22.14 | shmaltz | looks like a loose connection to me |
| 03:22.23 | shmaltz | did you crimp the cable urself? |
| 03:22.26 | crashsys | and the few times I was able to catch it in zttool it looked like the recieve bits all went 1... |
| 03:22.54 | shmaltz | crashsys, did you crimp the cables? |
| 03:23.10 | *** join/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 03:23.21 | crashsys | Yeah, I crimped the cables... but the T1 crossover cable has no problems going to the channel bank... |
| 03:23.40 | crashsys | and the GigE cable has no problems... the T1 cable tested clear... |
| 03:23.47 | shmaltz | crashsys, doesn't matter, you used solid or strended wire? |
| 03:23.55 | crashsys | Solid Cat3 |
| 03:24.08 | crashsys | from the smart-jack to the card |
| 03:24.18 | crashsys | should I have used flat-cable? |
| 03:24.24 | shmaltz | crashsys, you have special crimps or regular? |
| 03:24.52 | crashsys | Mmmm... special crimps meaning the tool or the connector? |
| 03:25.01 | shmaltz | connector |
| 03:25.03 | crashsys | the tool is just a regular RJ45 crimp tool... |
| 03:25.11 | crashsys | the connector is RJ45... |
| 03:25.26 | shmaltz | crashsys, if you crimp solid wire you need solid connectors like these: |
| 03:25.28 | shmaltz | http://froogle.google.com/froogle?q=rj45+solid+crimps&hl=en&lr=&sa=X&oi=froogle&ct=title |
| 03:25.47 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 03:25.55 | shmaltz | otherwise the crimping snaps the 26 awg wire/conducter |
| 03:26.13 | shmaltz | but hold on just enough signal that you don't notice it all the time |
| 03:26.25 | crashsys | The connector's i'm using are RJ45 ethernet for solid core's... |
| 03:27.13 | crashsys | it slides down the side of the wire and strips it back for connection... |
| 03:27.18 | shmaltz | crashsys, so you are sure that the cables are good, have you changed them to make sure that the cable is not the problem? |
| 03:27.28 | shmaltz | which is how it should be |
| 03:27.34 | shmaltz | so the connector is the right one |
| 03:27.46 | crashsys | shmaltz: Yes, i've even used a pre-machined cable... |
| 03:27.52 | crashsys | err pre-made |
| 03:27.56 | shmaltz | so lets assume the wiring is good |
| 03:28.08 | shmaltz | what cb do you have connected to it? |
| 03:28.25 | crashsys | CAC... it's the half-rack 96-channel one... lemme find the model number... |
| 03:29.06 | shmaltz | I never knew they make a 96 channel one |
| 03:29.13 | shmaltz | unless it's 2 adit 600s |
| 03:29.15 | crashsys | The channel bank never goes down... just the incoming PSTN T1... |
| 03:29.23 | JT | i know it's all you need, but curious why you used Cat3 when Cat5 is so abundant? |
| 03:29.38 | crashsys | jt: I had a box of cat 3 cable that needed to be used up... |
| 03:29.46 | JT | ah |
| 03:30.04 | JT | and another thing, solid is more likely to snap a conductor, especially if used as a patch cord |
| 03:30.05 | crashsys | been sitting in a closet for a while... and that was to run a new cable from the smart-jack to the card... there was a pre-existing run with Cat 5... |
| 03:30.08 | JT | so that could be it |
| 03:30.09 | shmaltz | crashsys, so only the PRI goes down? |
| 03:30.22 | shmaltz | JT, cat3 is excellent for this |
| 03:30.23 | crashsys | It's not PRI... it's CAS... but yes... |
| 03:30.39 | shmaltz | ~cas |
| 03:30.45 | JT | shmaltz: i'm sure cat3 has sufficient bandwidth |
| 03:30.51 | crashsys | Yeah, CAC 600... |
| 03:30.58 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 03:30.59 | shmaltz | how does cas work? does it have a signaling channel? or it's like RBS? |
| 03:31.04 | crashsys | RBS |
| 03:31.18 | JT | shmaltz: channel associated signalling |
| 03:31.19 | JT | yeah |
| 03:31.25 | crashsys | with RBS |
| 03:31.51 | shmaltz | hmmmm, I can't help you, the only time I had to work with one like that was when I integrated a televantage system to asterisk |
| 03:31.56 | shmaltz | I hate them |
| 03:32.03 | crashsys | I ripped my provider a new one... and they ran intrusive testing for a week, then supposively fixed a cable... |
| 03:32.17 | crashsys | I guess i'm gonna have to bitch more till they let me out of my contract, or run a new wire from the head-end... |
| 03:32.22 | shmaltz | crashsys, then it's their fault |
| 03:32.34 | crashsys | They keep telling me the damn line is clear... |
| 03:32.40 | shmaltz | crashsys, why don't you upgrade it to PRI? |
| 03:32.58 | shmaltz | also, what does zttest tell you? |
| 03:32.59 | JT | i'm confused, crashsys, is the faulty T1 to a channel bank or provider? |
| 03:33.04 | crashsys | Cause if i'm having cable issues, then i'm stuck with randomn span drops... |
| 03:33.17 | crashsys | jt: It's faulty from PSTN to me... |
| 03:33.25 | crashsys | from me to the channel bank it's fine... |
| 03:33.29 | JT | right |
| 03:33.30 | crashsys | that span never has issues... |
| 03:33.38 | shmaltz | crashsys, usualy does issues come up on the last mile connection, in which case you will hava this problem with every provider |
| 03:33.42 | crashsys | I'm migrating them from a Toshiba PBX to asterisk |
| 03:33.45 | JT | you to channel bank is a different T1 yeah? |
| 03:34.04 | crashsys | jt: Me to a channel bank is the second T1 port of my TE205p card... yes... |
| 03:34.14 | JT | cool |
| 03:34.20 | crashsys | I've even RMA'd the card to digium thinking the hardware had gone faulty... |
| 03:34.31 | shmaltz | crashsys, zttest resluts are what? give min, max and avg |
| 03:34.41 | JT | so you to the provider faulty opens up a lot more possibilities |
| 03:34.48 | crashsys | ATA's suck / were too exspensive... plus my provider already provided me with the channel bank... |
| 03:35.20 | crashsys | so I just put asterisk in between the channel bank the the provider, and it's doing simple channel routing... |
| 03:35.28 | shmaltz | crashsys, you have any clue if it comes in on fiber or copper? |
| 03:35.30 | JT | ah |
| 03:35.36 | crashsys | shmaltz: copper hdsl |
| 03:35.56 | shmaltz | hmmmm, that might be the problem, you know how far you are from the CO? |
| 03:36.18 | crashsys | Nope... i'm a mushroom... they keep me in the dark, feed me shit, and tell me it'll be ok... |
| 03:36.27 | shmaltz | lol |
| 03:36.34 | crashsys | I've asked for a cutover sheet I dont know how many times to make sure my settings are right... I still dont get that... |
| 03:36.47 | CunningPike | crashsys: Who's the telco? |
| 03:36.50 | crashsys | My next move is to hit 'em with a breach in the SLA and cancel my contract and go shopping... |
| 03:36.52 | crashsys | XO |
| 03:36.56 | shmaltz | crashsys, you don't know whare you are? every neighborhood I come into, the first thing I check is how far from CO |
| 03:37.04 | crashsys | they've been pretty good about everything else... |
| 03:37.22 | CunningPike | Except the service provision part |
| 03:37.29 | crashsys | shmaltz: I just haven't asked... I know i'm on a repeater... |
| 03:37.57 | shmaltz | crashsys, what is the NPA-NXX? |
| 03:38.18 | shmaltz | I'll tell you where the CO is then you can use mapquest to figure out milage, add around 10% |
| 03:38.34 | crashsys | 727-530 |
| 03:39.11 | crashsys | Verizon sucks complete ass down here too... |
| 03:39.29 | [shodan] | is there a way to ask asterisk if echo cancel is running on a fxo ? because I've got one bad echo problem :\ |
| 03:40.03 | crashsys | I thought the hardest part about learning asterisk would be the dial plan... not my damn physical pipe :( |
| 03:40.31 | *** part/#asterisk PCadach[work] (n=PCadach[@www.east.telecom.kz) |
| 03:41.36 | crashsys | Now... the part that makes me question if it's me is that if I take the T1 from the PSTN, and plug it directly into the channel bank, it works fine, and doesn't drop... |
| 03:42.05 | crashsys | I can only deduce that because the channel bank is not a smart piece of equipment, it is converting signal's wether one is being supplied or not... |
| 03:42.33 | CunningPike | [shodan]: something like 'zap show channel nnn' |
| 03:42.50 | CunningPike | [shodan]: But it will only show e/c on during a call |
| 03:42.54 | crashsys | am I right in thinking that? |
| 03:43.33 | JT | that would imply that the problem is less likely to be in te provider leg |
| 03:43.53 | crashsys | It's also randomn... with no pattern... |
| 03:44.19 | crashsys | some days it wont drop at all, other days it'll drop every 10 minutes... |
| 03:44.41 | smackus | ok, so now with the CORRECT kind of corssover cable my asterisk server is happy. but on the other server it shows the t1s are out of service. do i have to do anything different with my config to send t1 from asterisk instead of receive t1 from asterisk? |
| 03:45.04 | shmaltz | crashsys, here: |
| 03:45.06 | shmaltz | http://www.mapquest.com/maps/map.adp?areacode=727&exchange=530 |
| 03:45.07 | shmaltz | zoom in all the way, that will show you the street where the star is located, |
| 03:46.29 | shmaltz | smackus, with PRIs you got to set pri_net instead of pri_cpe |
| 03:46.46 | smackus | ok... |
| 03:46.48 | smackus | let me try that. |
| 03:46.53 | smackus | is that the only change? |
| 03:47.16 | crashsys | mapquest cant input my street address now... hmmm |
| 03:47.20 | shmaltz | smackus, yes |
| 03:47.39 | shmaltz | crashsys, what is the zttest output results? |
| 03:47.50 | crashsys | 2.64-miles... |
| 03:48.14 | crashsys | I dont remember... i'm not in front of the machine unfortunately... |
| 03:48.34 | shmaltz | http://www.google.com/search?hl=en&q=2.64+miles+in+feet&btnG=Google+Search |
| 03:49.07 | crashsys | 2.64mi + 10% = 2.9mi * 5280 = 15312 |
| 03:49.09 | crashsys | feet |
| 03:49.10 | shmaltz | that makes (with the 10% overhead): |
| 03:49.12 | shmaltz | 15 333.12 |
| 03:49.55 | shmaltz | which around here Verizon would sometimes (depending on the neighborhood) pull fiber for such a haul |
| 03:50.03 | JT | do you measure the distance as the crow flies, or by roads |
| 03:50.13 | crashsys | I know i'm on atleast one set of repeaters... |
| 03:50.16 | *** join/#asterisk kph100 (n=kph100@206-248-128-164.dsl.teksavvy.com) |
| 03:50.18 | shmaltz | JT, of course roads |
| 03:50.19 | kph100 | hello |
| 03:50.21 | crashsys | jt: that's by major roads... |
| 03:50.28 | JT | yeah i thought so |
| 03:50.39 | JT | 10% wouldn't be enough otherwise :) |
| 03:50.48 | shmaltz | JT, the wiring also travles on major roads |
| 03:50.55 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 03:51.04 | [shodan] | I ran show channel zap/1-1 , I got some information but nothing about echo cancellation , also I tried making a call with echocancel and echocancelwhenbridged on or off and it makes no difference ?! |
| 03:51.06 | kph100 | I need to dial an extension, but then abort and return back to main script if there was no answer/busy |
| 03:51.27 | kph100 | how can i do this? |
| 03:51.38 | crashsys | shmaltz: i'd be more then happy to give you access to the box, but without me driving up there to plug in a loopback or it into the PSTN there's no point... :) |
| 03:51.47 | crashsys | Unless you wanna hang around for 30 minutes... |
| 03:51.51 | crashsys | heh |
| 03:52.43 | crashsys | sounds like mae-east :) |
| 03:52.55 | crashsys | rent a backhoe, cripple the internet... heh |
| 03:53.05 | shmaltz | lol |
| 03:53.53 | shmaltz | ok, guys gtg |
| 03:53.56 | shmaltz | cya around |
| 03:54.06 | shmaltz | a few more minutes and my wife is locking me out :P |
| 03:54.08 | joelsolanki | Hi all. anybody using didx.org with asterisk perfectly ? |
| 03:54.09 | shmaltz | just kidding |
| 03:54.55 | crashsys | Hmmm... it appears I have the cross-over cable connecting the two spans... heh... |
| 03:55.08 | crashsys | still dont help with zttest |
| 03:55.12 | joelsolanki | Anybody plz. i need to make asterisk+didx.org working ... |
| 03:57.09 | crashsys | Is there a way to relax the red-alarm detection in zaptel? |
| 03:57.43 | crashsys | or is that not feasible? |
| 03:57.53 | russellb | relax? |
| 03:58.04 | russellb | i mean ... red alarm is pretty ... not relaxable |
| 03:58.20 | Qwell | russellb: :D |
| 03:58.25 | crashsys | didn't think so... figured i'd just throw it out there for desperation :) |
| 03:59.41 | smackus | so when dialing out on a crossover t1 to the old phone system, would I still dial using zap? for example: |
| 03:59.44 | smackus | exten => 8017037001,1,Dial(Zap/g2/${EXTEN:0}) |
| 03:59.50 | joelsolanki | Qwell: Hello |
| 04:00.02 | russellb | smackus: yes ... though the ":0" doesn't do anything :) |
| 04:00.24 | *** join/#asterisk bmg505 (n=leon@c1-243-1.rndf.isadsl.co.za) |
| 04:00.29 | smackus | here is what I get: |
| 04:00.29 | smackus | <PROTECTED> |
| 04:00.30 | smackus | <PROTECTED> |
| 04:00.30 | smackus | Aug 31 21:55:52 NOTICE[29191]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 04:00.30 | smackus | <PROTECTED> |
| 04:00.30 | smackus | <PROTECTED> |
| 04:00.32 | smackus | <PROTECTED> |
| 04:00.34 | smackus | <PROTECTED> |
| 04:00.36 | smackus | whoa.. sorry for the flood |
| 04:00.48 | Qwell | russellb: ${EXTEN:0:${LEN(${EXTEN})}} |
| 04:00.56 | joelsolanki | Qwell: Hello |
| 04:01.03 | Qwell | joelsolanki: hi |
| 04:01.03 | russellb | Qwell: AHHHHHHH |
| 04:01.12 | Qwell | russellb: You know you like it. :) |
| 04:01.21 | russellb | yeah, pretty nice. |
| 04:01.22 | Katty | i would just like to state i hate boys. |
| 04:01.25 | Katty | kthx. |
| 04:01.30 | smackus | and they hate you |
| 04:01.33 | Katty | please return to your regularly scheduled chan stuff. |
| 04:01.34 | smackus | just kidding... |
| 04:01.35 | smackus | not nice |
| 04:01.39 | russellb | Katty: me too |
| 04:01.42 | Qwell | Katty: men > boys |
| 04:01.46 | Katty | smackus: sorry, had to get that off... |
| 04:01.52 | Katty | smackus: me and my probably soon to be ex... |
| 04:01.52 | smackus | good for you |
| 04:01.57 | joelsolanki | Qwell: need little guide with asterisk + didx.org. |
| 04:02.01 | Qwell | umm...I seriously hope nobody takes that the wrong way |
| 04:02.06 | Katty | smackus: he doesn't want to stop raiding so that he can drive 15 miles to come get me. |
| 04:02.26 | Katty | smackus: instead, he wants me to drive the additional 15, plus the 90 i'm already driving, and wait around for him to get done raiding so i'm convient for him. |
| 04:02.35 | Katty | smackus: i'm about ready to smackus his tail. |
| 04:02.41 | smackus | lol |
| 04:02.45 | Qwell | Katty: PK him |
| 04:02.53 | russellb | question ... what is raiding? |
| 04:02.59 | Katty | russellb: WoW |
| 04:02.59 | Qwell | russellb: WoW :p |
| 04:02.59 | JT | indeed |
| 04:03.04 | JT | oh |
| 04:03.05 | JT | ... |
| 04:03.05 | Katty | russellb: online gaming, etc |
| 04:03.11 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 04:03.13 | russellb | lame |
| 04:03.16 | smackus | ok.. back to my question.... |
| 04:03.16 | Katty | very |
| 04:03.19 | crashsys | I like my AC still :( |
| 04:03.20 | Katty | he's not...grown up yet |
| 04:03.56 | russellb | smackus: is this a digium card? |
| 04:03.59 | Katty | i don't understand why i'm having such a hard time finding a non defective male :< |
| 04:04.03 | smackus | yes |
| 04:04.06 | Qwell | ooo |
| 04:04.08 | russellb | if so, i would save yourself the stress and contact support@digium.com tomorrow |
| 04:04.09 | Katty | i'm not ugly...i'm not dumb. |
| 04:04.09 | crashsys | Katty: RMA him? |
| 04:04.12 | JT | try find a non-defective human |
| 04:04.15 | JT | good luck |
| 04:04.18 | Qwell | russellb: even better ${EXTEN:${LEN(${EXTEN})}:0} |
| 04:04.20 | Katty | crashsys: i'd love to! |
| 04:04.31 | Katty | i don't have a /whole/ lot of baggage. |
| 04:04.34 | russellb | Qwell: but that's ... |
| 04:04.35 | Katty | hell i'm a geek! |
| 04:04.37 | russellb | Qwell: nothing |
| 04:04.39 | Qwell | russellb: exactly! |
| 04:04.58 | smackus | Qwell: ? |
| 04:04.59 | Qwell | That's how you take the LONG WAY AROUND to NULL |
| 04:05.02 | crashsys | CRCCIT is the only kernel option I need for zaptel to work properly correct? |
| 04:05.19 | JT | mm, so does anyone know is Echo() actually adds any echo that isn't caused by the system? |
| 04:05.21 | Katty | file: i need to put in a request for a non defective male :< |
| 04:05.32 | Katty | file: can i get a credit or do i need to have this one rmaed? |
| 04:05.34 | Qwell | JT: it copies the packets, and sends them right back out |
| 04:05.38 | Katty | file: trade? |
| 04:05.44 | Qwell | hell, it probably doesn't even copy the packets |
| 04:05.52 | JT | so the echo should be fairly representative of the system delay |
| 04:05.53 | JT | hmm |
| 04:05.58 | file | Katty: well, if you leave a valid credit card you can cross ship it |
| 04:06.00 | Qwell | yes, generally |
| 04:06.11 | JT | cool |
| 04:06.17 | smackus | I would love to wait till tomorrow, but I am able to take the other system off line tonight. |
| 04:06.23 | Katty | file: i'm going to be the little old lady on 8th street with the nine cats, aren't i? |
| 04:06.31 | smackus | let me paste my configs and see if you guys can see any errors |
| 04:06.31 | JT | sounds like i have about 200ms delay on my lan all up |
| 04:06.32 | JT | using iax |
| 04:06.33 | file | Katty: no, ten cats |
| 04:06.39 | file | Katty: nice even amount |
| 04:06.42 | Katty | file: oh, well...that's not so bad then. |
| 04:07.22 | russellb | ${GLOBAL(${EVAL(${EXTEN:${LEN(${EXTEN})})})} |
| 04:07.26 | Katty | file: cept the single part :< |
| 04:07.26 | JT | crashsys: where is the crccit option? :P |
| 04:07.27 | JT | :) |
| 04:07.40 | file | Katty: eBay? |
| 04:07.50 | Katty | ebay! why didn't i think of that! |
| 04:07.52 | crashsys | libraries |
| 04:07.58 | Qwell | russellb: I.. I.. lose |
| 04:08.04 | russellb | Qwell: that could get pretty silly |
| 04:08.10 | JT | that's what i thought, couldn't find it in mine, weird |
| 04:08.14 | JT | will have to try again |
| 04:08.15 | russellb | when you start using stuff that will accept blank args :) |
| 04:08.21 | Qwell | heh |
| 04:08.29 | file | Qwell: you need to increase your silly level |
| 04:08.36 | file | although russellb is a pro |
| 04:08.52 | russellb | i'm kind of a big deal ... |
| 04:08.56 | Qwell | heh |
| 04:09.19 | Katty | russellb: like a value meal? |
| 04:09.26 | Katty | russellb: supersized? |
| 04:09.27 | russellb | ha, exactly. |
| 04:09.34 | russellb | just for 39 cents more |
| 04:09.45 | file | what a bargain! |
| 04:09.49 | Katty | cheap price for a heart attack! |
| 04:09.57 | Katty | we loves it when you be smilin. |
| 04:09.59 | Qwell | Katty: totally worth it |
| 04:10.06 | smackus | here are my configs... http://pastebin.ca/156919 i am trying to dial from my cell phone 8017037001 and get the error: http://pastebin.ca/156921 |
| 04:11.20 | CunningPike | Katty: Get a dog |
| 04:11.39 | Qwell | plants > dogs |
| 04:12.05 | smackus | plants die, dogs run away... less quilt with a dog |
| 04:12.12 | smackus | guilt |
| 04:12.12 | docelmo | KATTY! |
| 04:12.21 | CunningPike | I'd like to see the plant that can get the paper for you |
| 04:12.35 | smackus | id like my dog to get thepaper |
| 04:13.08 | JT | dogs in a car |
| 04:13.08 | smackus | so do my configs look jacked? or am i on the right track? |
| 04:13.48 | Katty | CunningPike: i like doggies. |
| 04:13.57 | Katty | CunningPike: but they don't hold the door open very often. |
| 04:14.14 | Katty | CunningPike: would you believe that none of the guys i've dated have opened the car door for me? |
| 04:14.17 | CunningPike | Katty: They don't leave their socks on the floor either |
| 04:14.19 | Katty | docelmo: hey. |
| 04:14.30 | Katty | CunningPike: eh, i don't care about socks on the floor. |
| 04:14.50 | Juggie | smackus, your trying to dial out the channel that just received the call. |
| 04:16.04 | smackus | ok, i do need caffeine. I am totally messed up here. thanks for the catch. |
| 04:16.25 | *** join/#asterisk tabmeister1 (n=troy@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 04:16.28 | smackus | wait... hang on. |
| 04:16.37 | smackus | the call should be coming in on g1. |
| 04:16.42 | tabmeister1 | i seem to get a no authority found error when initiating an IAX call, any ideas? |
| 04:16.43 | smackus | and going out on g2. |
| 04:17.01 | smackus | g2 is cross over t1 cables plugged into the other phone system. |
| 04:17.07 | smackus | g1 is the inbound t1s. |
| 04:17.10 | Juggie | is g2 up? |
| 04:17.16 | Juggie | because it has no channels |
| 04:17.19 | smackus | yes... |
| 04:17.25 | smackus | thats why i think there is a config error |
| 04:17.29 | Juggie | check your group setup |
| 04:17.44 | smackus | what do you mean? |
| 04:18.01 | smackus | all channels are green |
| 04:18.08 | smackus | and g2 is set up as pri_net |
| 04:18.35 | Juggie | what does 'zap show status' say |
| 04:18.55 | Katty | show status of zaptel cards or channels, me thinks. |
| 04:19.00 | smackus | dev-asterisk*CLI> zap show status |
| 04:19.00 | smackus | Description Alarms IRQ bpviol CRC4 |
| 04:19.00 | smackus | T4XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 04:19.00 | smackus | T4XXP (PCI) Card 0 Span 2 OK 0 0 0 |
| 04:19.00 | smackus | T4XXP (PCI) Card 0 Span 3 OK 0 0 0 |
| 04:19.01 | smackus | T4XXP (PCI) Card 0 Span 4 OK 0 0 0 |
| 04:19.02 | Katty | oh. |
| 04:19.09 | Katty | i love how i just read that as the wrong question ;) |
| 04:19.21 | file | Katty: I am sleepy! |
| 04:19.23 | Katty | way to be awake Katty! |
| 04:19.30 | Katty | file: come nap. |
| 04:19.38 | file | I'm already in bed :( |
| 04:19.44 | Katty | then go nap! |
| 04:19.50 | file | I go nini |
| 04:19.57 | Qwell | file: pfft |
| 04:20.11 | file | Qwell: back to work! |
| 04:20.20 | Juggie | smackus, can you dial from the other phone system into *? |
| 04:20.20 | Katty | file: ctrl+a+d kthx. |
| 04:20.25 | smackus | nope |
| 04:20.35 | smackus | but the t1s there are green also |
| 04:20.41 | smackus | so i have to be close |
| 04:21.36 | Juggie | do a 'pri debug span 3' 'pri debug span 4' |
| 04:21.40 | *** join/#asterisk dalabera (n=dalabera@71.196.61.57) |
| 04:21.40 | Juggie | and then try the call again |
| 04:21.44 | *** part/#asterisk dalabera (n=dalabera@71.196.61.57) |
| 04:24.10 | smackus | here is all i got for an inbound and an outbound call: http://pastebin.ca/156927 |
| 04:24.46 | smackus | inbound call from cell phone, and outbound call from the old phone system plugged into asterisk ports 3 and 4 |
| 04:25.32 | *** join/#asterisk flujan (n=flujan@201-42-98-156.dsl.telesp.net.br) |
| 04:25.59 | Juggie | ok, pri intense debug span 3 & span 4 |
| 04:26.03 | Juggie | and do the same thing |
| 04:27.10 | flujan | guys... my asterisk install stop working... |
| 04:28.05 | flujan | i just change the queue_log to be a pipe using mkfifo |
| 04:28.42 | DasTech | http://pastebin.ca/156928 need help with asterisk-addons |
| 04:28.46 | DasTech | wont build |
| 04:28.56 | smackus | Juggie: http://pastebin.ca/156929 |
| 04:28.57 | DasTech | svn trunk |
| 04:29.34 | DasTech | anyone |
| 04:30.07 | smackus | too vague... what is the issue |
| 04:30.10 | smackus | DasTech: |
| 04:30.16 | smackus | tell us more |
| 04:30.42 | smackus | what distro |
| 04:30.51 | DasTech | i pasted the error on pastebin |
| 04:30.55 | DasTech | centos |
| 04:32.17 | DasTech | http://pastebin.ca/156928 |
| 04:32.42 | DasTech | my whole box is now fubared |
| 04:32.51 | DasTech | realtime wont work |
| 04:33.07 | DasTech | I need the msql stuff from addons |
| 04:33.26 | tabmeister1 | for some reason automated menus arent' registering my DTMF tones from a cisco ip phone |
| 04:34.05 | smackus | Juggie: any thoughts on my output? |
| 04:37.17 | smackus | <PROTECTED> |
| 04:37.24 | smackus | errr |
| 04:37.28 | smackus | not tabmeister1 |
| 04:37.35 | smackus | <PROTECTED> |
| 04:37.42 | tabmeister1 | heh |
| 04:40.16 | DasTech | nope no help |
| 04:41.33 | [shodan] | what's a "briged TDM call" ? and should I care if my setup only include sip phones, ATAs and the pstn ? |
| 04:44.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:44.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:45.16 | *** join/#asterisk negativecreep (n=xaeem@host210-2-170-89.isb.dancom.net.pk) |
| 04:45.22 | negativecreep | hi guys |
| 04:45.32 | smackus | Juggie, you still there? |
| 04:50.47 | Juggie | yah |
| 04:50.51 | Juggie | dealing with another problem sorry. |
| 04:50.59 | smackus | oh, no prob. |
| 04:51.03 | *** part/#asterisk negativecreep (n=xaeem@host210-2-170-89.isb.dancom.net.pk) |
| 04:51.03 | Juggie | someone stole 880$ from my bank account |
| 04:51.08 | smackus | thanks for the efforts thus far |
| 04:51.10 | Juggie | at a bank machine |
| 04:51.13 | Juggie | in montreal |
| 04:51.17 | Juggie | i am in newfoundland :) |
| 04:51.38 | smackus | damn Canadians, freakin hosers |
| 04:52.34 | [shodan] | you probably used a tapped bank terminal |
| 04:52.45 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 04:52.55 | Juggie | probally. |
| 04:53.43 | [shodan] | that's why I always use the teller and keep my balance at 0$ |
| 04:57.19 | Juggie | i only use bank machines |
| 04:57.31 | Juggie | i never ever pay debit @ a store |
| 04:57.40 | Juggie | maybe mcdonalds were they dont take visa |
| 05:00.03 | *** join/#asterisk jefrey (n=jefrey@202.190.203.200) |
| 05:00.17 | jefrey | anyone uses cdr_radius here? |
| 05:01.58 | smackus | Juggie: should I still be assigning a D channel on the t1s going out to the old pbx? or should i be using all 24 ports? |
| 05:02.37 | Juggie | depends on what the other end is configured as. |
| 05:02.54 | Juggie | still busy, on the phone w/ bank |
| 05:03.01 | smackus | sorry.. |
| 05:04.09 | JT | if you're talking PRI, 23B + 1D |
| 05:04.18 | JT | otherwise 24B with CAS |
| 05:07.54 | *** join/#asterisk DuckWizrd (i=DuckWizr@user-10cmagk.cable.mindspring.com) |
| 05:08.42 | DuckWizrd | SIP calls work fine on my asterisk box, but playing sounds with Playback() or Background() yields nothing but silence. /var/log/asterisk/messages shows no potential problems. |
| 05:08.47 | DuckWizrd | any idea what could cause this? |
| 05:09.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:09.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:09.40 | JT | DuckWizrd: are you Answering the calls first? |
| 05:09.48 | DuckWizrd | JT: Yes. |
| 05:09.54 | [shodan] | is there any way to tune the echo cancelling on a x100p ? I have echocancel=256 and echotraining=4000 , and I'm getting as much echo as if I set them both to 0 , but the settings at least for echotraining seem to do something because since I added it to zapata.conf the call takes longer to start and when I run ztmonitor -v I can see the training pulses |
| 05:10.00 | JT | actually i dunno if that would matter, last time i tried, didn't need to Anwer() Playback() |
| 05:10.12 | JT | Answer() |
| 05:10.52 | DuckWizrd | My dialplan for that extension goes ringing, Answer(), Wait(3), Playback(some-sound-that-exists) |
| 05:11.06 | DuckWizrd | err actually wait then answer |
| 05:11.09 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 05:11.28 | JT | does anything come up in the console? try starting the console with at least 4 vs |
| 05:11.42 | hads | Give it 5 for good measure :) |
| 05:12.34 | DuckWizrd | I have actually never had success with the console. What's the proper command line (could never get an interactive console to come up) |
| 05:12.58 | hads | asterisk -r |
| 05:13.10 | CunningPike | DuckWizrd: sudo /usr/sbin/asterisk -r |
| 05:13.15 | tengulre | Hi,all |
| 05:13.19 | JT | don't forget the vs |
| 05:13.23 | tengulre | which raduis supported by asterisk?? |
| 05:13.57 | CunningPike | DuckWizrd: Are these your own sounds, or the default asterisk ones? |
| 05:14.06 | hads | Bah, don't need v's just use 'set verbose 5' :) |
| 05:14.49 | DuckWizrd | CunningPike: I have tried both custom sounds (16bit signed mono WAV at 8KHz) and default asterisk sounds (GSM) |
| 05:15.27 | CunningPike | DuckWizrd: What codecs are you using? |
| 05:16.11 | DuckWizrd | CunningPike: I have not changed any codec settings. I'm running the debian experimental asterisk package. |
| 05:16.30 | JT | DuckWizrd: try watching in the console |
| 05:16.44 | JT | it'll be a lot more useful than our random guesses, with a bit of data |
| 05:16.52 | DuckWizrd | ok, I got the console up. I'll try calling in once I get a softphone installed on my home machine |
| 05:17.49 | DuckWizrd | bah, it wants me to reboot... thanks for the tips; I'll be back :) |
| 05:18.36 | tessier | Anyone know what is best practice for setting up the dialplan for dialing local extensions and for outbound dialing? I am told that with a modern PBX you don't need to set it up to dial 9. But then how does the phone know if you are dialing a local extension vs dialing an outbound number so it can know how many digits to expect and to launch the call when you are done dialing? Surely users are not happy with the cell phone beh |
| 05:18.36 | JT | the softphone i will assume |
| 05:18.36 | tessier | avior of having to press the "dial" button every time. |
| 05:18.44 | [shodan] | hmm what's the characteristic impedance of CAT5 ? |
| 05:19.15 | JT | 120ohm i thought |
| 05:19.32 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 05:19.45 | CunningPike | tessier: You can do it by designing your dialplan to avoid conflicts. For example, I am in area code 604 - we won't ever create a local extension that starts 604X |
| 05:19.45 | rene- | hey |
| 05:19.56 | JT | 100ohm sorry |
| 05:19.57 | *** part/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 05:20.15 | tessier | CunningPike: And when I want to dial a different area code that does match a local extension? |
| 05:20.38 | tessier | CunningPike: And won't you usually dial a 1 before dialing an area code? |
| 05:20.44 | CunningPike | tessier: You put a 1 in front of it, if it's long distance |
| 05:21.02 | tessier | And if it isn't long distance? Just a normal 7 digit number? |
| 05:21.11 | CunningPike | tessier: We're a little weird here - we have 2 intermingled area codes, so we have 10-digit dialling |
| 05:21.34 | tessier | If you have to put a 1 in front of everything you can't have any extensions that begin with 1 like 100 or 1000. That would be weird. |
| 05:21.43 | CunningPike | tessier: Hmm - with 7 digit dialing, you might be better off with a 9 prefix |
| 05:22.20 | Qwell | putting 9 in front of all "external" calls is a "best practice" |
| 05:22.20 | tessier | And the problem with the 9 prefix to get an outside like is that the Snom 320 doesn't continue with dialtone after you press the 9. |
| 05:22.34 | Qwell | tessier: I'm sure there is an option for it to |
| 05:22.41 | tessier | Qwell: Snom told me otherwise. Their phones apparently cannot continue to make dialtone after pressing 9. |
| 05:22.42 | JT | asterisk can actually deal with multiple numbers, with some shorter versions of longer ones |
| 05:22.45 | JT | with timeouts |
| 05:22.49 | JT | but it'd a bit of a kludge |
| 05:22.51 | tessier | JT: I know that. But that is annoying. |
| 05:22.56 | Qwell | tessier: that's kinda stupid |
| 05:23.02 | tessier | I am trying to find The Right Way. |
| 05:23.12 | JT | is 0 for outside dialling no good? |
| 05:23.22 | CunningPike | tessier: The Right Way (tm) is to use 9 in your situation |
| 05:23.27 | [shodan] | oh so if I used about 10 meters of cat5 to carry my pstn line that is 600 ohm that could be the cause of the echo I'm getting right ? |
| 05:23.33 | hads | Why is it a "best practice"? |
| 05:23.42 | Qwell | hads: notice the quotes ;) |
| 05:23.57 | Qwell | hads: It's just a generally accepted thing you do |
| 05:23.59 | tessier | CunningPike: That's what I thought. Hrm...which is annoying since the snom doesn't handle it well. So far that is the only deficiency I have found in these phones. They are the best I have used otherwise. |
| 05:24.07 | Qwell | the alternative, is to hardcore every local prefix |
| 05:24.22 | hads | Qwell: Oh. I get it :) (excuse me, it's beer o'clock here) |
| 05:24.32 | Qwell | and if one ever gets added that collides with your internal extensions, then...well...you're changing your extensions |
| 05:24.39 | Qwell | s/hardcore/hardcode/ |
| 05:25.03 | CunningPike | tessier: Pity - our Polycoms will do it, but we're lucky to have 10-digit dialing, so we can dispense with the 9 |
| 05:25.14 | tessier | CunningPike: I know Cisco's can do it also. |
| 05:25.16 | Qwell | of course, we in LA have it easy. Everything is 10 digit dialed, so...pfft |
| 05:25.30 | CunningPike | tessier: But we have enough users that press the 9 anyway, that our dialplan does both |
| 05:25.43 | CunningPike | (Hangover from our legacy PBX) |
| 05:25.48 | tabmeister1 | Qwell, only 10 digits? :) |
| 05:25.57 | Qwell | sorry, 10+1 |
| 05:26.02 | Qwell | or 1+10 as it were |
| 05:26.16 | Qwell | I'm stealing hads "it's late" excuse |
| 05:26.40 | tessier | I am also wondering if I can just use different identities on the phone for outside lines and put them into different context. |
| 05:26.59 | Qwell | tessier: You could, but that seems like a huge hack |
| 05:27.19 | Qwell | regardless of what snom told you, I'd be shocked if they didn't support a secondary dialtone |
| 05:27.29 | CunningPike | tessier: I think your users would like that less than having no second dial tone |
| 05:27.34 | hads | It's Friday 17:27 here, but I cheated by a few minutes and am on my second beer :) |
| 05:27.47 | CunningPike | hads: Where are you? |
| 05:27.49 | CunningPike | Oz? |
| 05:28.02 | hads | NZ - two hours ahead of AU |
| 05:28.12 | JT | heh |
| 05:28.14 | CunningPike | hads: Cool - a Kiwi |
| 05:28.26 | JT | i was just thinking "waaait, it's only 1528 here" |
| 05:28.32 | tessier | Qwell: I have googled all over and asked Snom. No secondary dialtone. |
| 05:28.36 | hads | Well, that depends on where you are in AU of course. |
| 05:28.52 | JT | somewhere that matters :P |
| 05:28.52 | CunningPike | It's 2228 on Thursday here |
| 05:29.02 | JT | living in the past ;) |
| 05:29.05 | CunningPike | :D |
| 05:29.11 | tessier | CunningPike: Would like what less? Having to press a button with a different identity to get an outside line? They already have to do that on the current phone system. |
| 05:29.39 | CunningPike | tessier: Really? Wow |
| 05:30.44 | tessier | Actually they say the dialtone has to be signalled to the phone by the pbx. |
| 05:31.03 | tessier | Which means I could use DISA. But then the numbers dialed do not show up on the phones screen. |
| 05:31.23 | *** join/#asterisk asterisk_bounty (n=pcm@72.146.14.209) |
| 05:31.57 | *** join/#asterisk HaMYaI (n=hamyai@ppp-58.8.9.82.revip2.asianet.co.th) |
| 05:32.37 | tessier | I guess I will just have to say screw the secondary dialtone. |
| 05:32.44 | tessier | And I will use 9 to get an outside line. |
| 05:32.52 | CunningPike | tessier: Good choice |
| 05:33.35 | tessier | sx-wks: And you have local extensions? And the phone knows when the number is complete? How? |
| 05:33.51 | sx-wks | well... all outside numbers start with 0 :D |
| 05:33.58 | tessier | bah |
| 05:34.34 | HaMYaI | anyone knows what can be used to interrupt the "Background", apart from DTMF? |
| 05:34.51 | tessier | Ok, I guess I have this figured out. It's not ideal but it's good enough. |
| 05:35.06 | sx-wks | HaMYaI: there's annother version that listens for "something" |
| 05:35.22 | HaMYaI | sx-wks: what is it? |
| 05:35.26 | sx-wks | erh |
| 05:35.39 | HaMYaI | sx-wks: that's what I'm really looking for |
| 05:35.53 | HaMYaI | sx-wks: is it MusicOnHold? |
| 05:36.08 | sx-wks | BackgroundDetect |
| 05:36.41 | HaMYaI | sx-wks: umm, yeah but that's still for DTMF I think |
| 05:36.48 | HaMYaI | sx-wks: let me check again |
| 05:36.48 | sx-wks | nah |
| 05:37.11 | sx-wks | according to the book, it looks for "non silence" |
| 05:37.56 | HaMYaI | sx-wks: yeah, you are right that's what it said |
| 05:38.29 | HaMYaI | sx-wks: but what I want to do is to check for the status of something like "an existence of a file" |
| 05:38.56 | HaMYaI | sx-wks: and when the condition is met I will stop the Background Play |
| 05:39.15 | HaMYaI | any other possible methods? |
| 05:39.33 | sx-wks | erh, add an option to background then :D |
| 05:39.48 | sx-wks | after all, it's free software :D |
| 05:41.14 | HaMYaI | from the wiki it said "Background: Play a sound file while processing other commands " |
| 05:41.38 | HaMYaI | so I guess there must be a way to do it |
| 05:42.06 | *** part/#asterisk jefrey (n=jefrey@202.190.203.200) |
| 05:42.32 | HaMYaI | as you said, it may be easier modifying the Background function |
| 05:52.00 | tengulre | which raduis supported by asterisk?? |
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| 05:52.35 | docelmo | none |
| 05:52.39 | docelmo | You have to hack it.. |
| 05:53.00 | docelmo | And RADIUS is a protocol anyone compliant with the RFC will work either way either server or client |
| 05:56.34 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:01.51 | HaMYaI | sx-wks: got it done |
| 06:02.08 | sx-wks | wow, that was fast :D |
| 06:02.11 | HaMYaI | it's a lot easier really |
| 06:02.31 | sx-wks | now, get yourself a disclaimer and post the feature in the mantis :D |
| 06:09.01 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 06:15.19 | benjk | real hackers don't sign disclaimers |
| 06:15.36 | benjk | real hackers release under BSD or MIT terms |
| 06:16.42 | benjk | or GPL/LGPL if they feel they need to restrict distribution |
| 06:17.32 | sx-wks | benjk: heh |
| 06:17.58 | sx-wks | benjk: man, you're a troll |
| 06:18.10 | benjk | ditto |
| 06:19.08 | benjk | people who try to apply group pressure on their peers are more dangerous then people who are ridiculed as trolls |
| 06:19.14 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 06:19.35 | Qwell | than |
| 06:20.39 | MGSsancho | i listen to the person who has a gun up to my head |
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| 06:42.47 | asterisk_bounty | is trunk supposed to compile ? |
| 06:43.23 | sx-wks | asterisk_bounty: it did last night, but sip is unuseable |
| 06:43.26 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 06:49.43 | asterisk_bounty | sx-wks: I tried to compile now and it's wierd |
| 06:50.01 | asterisk_bounty | sk-wks: first stops on astman .. complains about newt functions .. |
| 06:50.11 | sx-wks | not here |
| 06:50.21 | asterisk_bounty | so I disable all the util files ... then it complains about res/* files ... |
| 06:50.47 | asterisk_bounty | so if I try to checkout 'last nights' version I should be ok ? |
| 06:51.18 | asterisk_bounty | sx-wks; should it present me with menuselect or should I call menuselect myslef ? |
| 06:51.33 | hads | asterisk_bounty: libnewt? |
| 06:51.56 | asterisk_bounty | hads; yes, I have it installed along with -devel ... but it's no go |
| 06:52.22 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 06:52.35 | asterisk_bounty | hads: but when I disable the astman and other utils then trunk stops compiling on the first res/res_* file |
| 06:52.42 | hads | I just updated mine (haven't for a while) and am compiling to see. It usually compiles though. |
| 06:52.44 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 06:55.42 | asterisk_bounty | hads: i'm trying version from 20060830 |
| 06:56.13 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 06:57.51 | hads | asterisk_bounty: Trunk compiled fine here - current as of now. |
| 06:58.27 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 06:58.42 | asterisk_bounty | ok |
| 06:58.53 | asterisk_bounty | hads: what distribution do you use ? |
| 06:59.05 | hads | Debian testing |
| 06:59.22 | asterisk_bounty | ok |
| 06:59.27 | asterisk_bounty | I'll try centos |
| 06:59.34 | asterisk_bounty | i was using an old fedora1 box |
| 07:02.40 | *** part/#asterisk tabmeister1 (n=troy@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
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| 07:05.57 | asterisk_bounty | hads: ok, it compiles now... |
| 07:06.07 | asterisk_bounty | thanks ... |
| 07:06.21 | hads | :) I didn't do anything |
| 07:07.51 | asterisk_bounty | you told me that it should work :) |
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| 07:16.26 | sx-wks | asterisk_bounty: lol |
| 07:16.40 | sx-wks | that changed everything for sure :D |
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| 07:21.29 | Aurs | hehe |
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| 07:22.28 | BeBoo | In queues.conf, will a member => @1 be enough for all in group 1? |
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| 08:04.30 | swytch | hello. im wandering if a digium wildcard with two physical ports can have the ports addressed and configured as separate units in * config. |
| 08:04.58 | Strom_C | swytch: what do you mean "separate units"? |
| 08:05.11 | Strom_C | and are you talking about analog ports or T1 ports? |
| 08:06.05 | swytch | Strom_C: that i can have one dialplan associated with port #1 and separately configure port #2 as for example a peer for outgoing calls. |
| 08:06.27 | swytch | Strom_C: digital, for example PRI |
| 08:06.46 | *** join/#asterisk xyklopz (n=xyklopzi@86.122.8.28) |
| 08:06.47 | Strom_C | swytch: yes you can do that easily |
| 08:07.31 | swytch | Strom_C: ok, i guess the sample config for the card contains an example once i buy a card. |
| 08:07.50 | Strom_C | asteris comes with a sample config |
| 08:08.08 | Strom_C | and digium will help you configure and install the card for free |
| 08:08.17 | *** join/#asterisk duderock (n=natrix@CPE-72-131-35-138.wi.res.rr.com) |
| 08:08.56 | swytch | Strom_C: what im going to do is to try improve sound quality by passing my sip-to-sip traffic thru a wildcard having the octasic voice quality enhancement. |
| 08:09.05 | Strom_C | swytch: you can do that |
| 08:09.13 | xyklopz | i need help setting up my sip.conf in mysql |
| 08:09.17 | Strom_C | swytch: how well it will work, i have no idea |
| 08:09.22 | xyklopz | i have compiled/installed the *-addons package |
| 08:09.49 | swytch | Strom_C: this is against all logic going to improve the quality. we have observed this improvement by passing rtp thru another rtpproxy that we cannot know anything about. |
| 08:10.00 | xyklopz | and find quite different SQL schemas online ... I'm using the latest 1.2.11 |
| 08:12.52 | swytch | Strom_C: ill try out the TE212P then. and doing this crazy thing. thank you. (-8 |
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| 08:25.58 | benjk | who are you and why are you trying to drag me into this madzone room? |
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| 08:30.03 | darkskiez | benjk: coz Akamurdera invited everyone, twonk |
| 08:31.05 | darkskiez | benjk: him, not you. |
| 08:31.18 | benjk | I suspected that, but I didn't want to let it stand without expressing my objection |
| 08:31.43 | benjk | maybe he should be kicked |
| 08:31.55 | foxmjay | good morning all , plz, how can i use the CHANNEL STATUS agi command in a php , i did fwrite($stdout,"CHANNEL STATUS ZAP/1-1\n"); but it gives me the result -1 instade of giving me 0 |
| 08:32.06 | benjk | depening on whether or not you consider it a form of spam |
| 08:33.08 | foxmjay | the others status like line is up & ringing, are correct |
| 08:33.20 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 08:35.04 | foxmjay | someone have used the CHANNEL STATUS command !! |
| 08:35.19 | benjk | I think you'll find its the wrong time of day |
| 08:35.21 | EmleyMoor | I'm probably moving my asterisk to another machine in a few weeks (existing one is too slow, really) but I haven't decided whether to go for the same version or downgrade to 1.0.7 (as that's in Debian stable) - what are the pros and cons? |
| 08:36.46 | tessier | EmleyMoor: I wouldn't downgrade. |
| 08:36.50 | tessier | EmleyMoor: Asterisk has improved a lot. |
| 08:37.03 | tessier | EmleyMoor: And if you ask anyone about problems you have with 1.0.7 they will tell you to upgrade |
| 08:37.18 | benjk | well, the opposite here |
| 08:37.22 | EmleyMoor | So you'd say go "testing" and then stabilise at the next upgrade? |
| 08:37.25 | tessier | debian stable is an ancient pile or horse crap anyhow. Use mdoern software. |
| 08:37.28 | benjk | 1.0.10 works best for me |
| 08:37.51 | benjk | plus various mods of course |
| 08:38.12 | *** join/#asterisk CSerpent (i=daemon@62.49.253.91) |
| 08:38.21 | benjk | there are always to sides to it, at the end of the day you have to find out for yourself what's best for you |
| 08:38.26 | CSerpent | morning |
| 08:38.37 | benjk | two sides |
| 08:41.00 | *** join/#asterisk duderock (n=natrix@CPE-72-131-35-138.wi.res.rr.com) |
| 08:41.16 | duderock | hi room! anyone running * on an ubuntu-server? |
| 08:42.23 | swytch | duderock: i did think ubuntu was best for desktop. but as it is based on debian its great yes. just disable X etcetera. |
| 08:43.36 | swytch | duderock is a little quick for me. |
| 08:45.32 | benjk | duderock, yes Ubuntu server 6.06 |
| 08:45.44 | benjk | ooops |
| 08:45.58 | *** join/#asterisk duderock (n=natrix@CPE-72-131-35-138.wi.res.rr.com) |
| 08:45.59 | benjk | how long did he wait for an answer? |
| 08:46.04 | benjk | doh |
| 08:47.17 | tessier | Does anyone still use python in the dialplan with asterisk 1.2? I haven't heard anything about it in quite a while. |
| 08:47.50 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 08:48.12 | duderock | sorry to repeat myself, switched to console... any ubuntu users here? |
| 08:48.34 | benjk | benjk: duderock, yes Ubuntu server 6.06 |
| 08:49.05 | duderock | excellent. i tried out trixbox, but it was too....eh, fedora-like... any problems with *? |
| 08:49.25 | benjk | trixbox != * |
| 08:49.42 | tessier | PyAstre seems to be defunct. :( |
| 08:49.46 | tessier | The websites don't work anymore |
| 08:50.27 | duderock | sorry, benjk don't follow your syntax.... |
| 08:50.48 | benjk | of course its based on Asterisk, but the front end of it is FreePBX and you won't find many people in here who use that, so you will probably not get a lot of feedback |
| 08:51.03 | benjk | but there is a channel just for that |
| 08:51.18 | benjk | I think its #freepbx, but not sure |
| 08:51.26 | duderock | ah. I did install FreePBX, but mainly for FOP... and to allow others to modify some settings. |
| 08:51.55 | benjk | in that channel you'll get lots of feedback on FreePBX and trixbox |
| 08:52.42 | benjk | I am not saying it is no good, or whatever, I am only saying that the folks who use it hang out in their own channel and not typically in here |
| 08:52.53 | duderock | ah, see trixbox kept upsetting my AMD X2. |
| 08:52.57 | duderock | i don't like it. |
| 08:53.52 | duderock | CentOS' most recent kernel has a bug that causes a kernel panic on dual-core machines... |
| 08:54.13 | benjk | anyway, I used to built SuSE based * boxes for almost 3 years, simply because of the convenience of YaST |
| 08:54.27 | duderock | How does YaST compare to APT? |
| 08:55.10 | benjk | but since Novell took over there have been some policy changes and now they make it difficult for you to build your own kernel modules, which means you have to go through a whole bunch of extra nonense to prepare your machine to build zaptel drivers etc |
| 08:55.18 | benjk | so I switched distros |
| 08:55.25 | benjk | about 3-4 months ago or so |
| 08:55.31 | duderock | And Ubuntu/Zaptel play well together? |
| 08:55.45 | benjk | I tried out a couple of things and settled for Ubuntu Server |
| 08:55.51 | benjk | yes, works like a charm |
| 08:56.36 | duderock | how did you obtain your zaptel module? compile from * svn? |
| 08:56.49 | benjk | btw, YaST is not a package manager, its a user interface for all things to do with system admin |
| 08:57.08 | duderock | oh, sorry - never used SUSE before... |
| 08:57.10 | benjk | its got a curses front end so you can use it over ssh |
| 08:57.17 | benjk | this was the main attraction for SuSE |
| 08:57.22 | duderock | similar to webmin then? |
| 08:57.31 | benjk | well, its all character based |
| 08:57.46 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 08:57.50 | swytch | duderock: i did think ubuntu was best for desktop. but as it is based on debian its great yes. just disable X etcetera. |
| 08:57.55 | benjk | log in over ssh, fire up YaST and you get a kind of midnight commander for sysadmin stuff |
| 08:58.01 | duderock | ah |
| 08:58.15 | benjk | other than that, SuSE isn't really for servers |
| 08:58.18 | *** join/#asterisk RoyK (n=roy@ti211110a081-1433.bb.online.no) |
| 08:58.19 | duderock | I haven't loaded X |
| 08:58.26 | benjk | no I don't even install X |
| 08:58.51 | benjk | I used Ubuntu Server, its a single CDROM for a server install, no X |
| 08:58.54 | duderock | though on all of my personal machines, i did the ubuntu-server install, then installed X on top of it. |
| 08:58.59 | benjk | hi RoyK |
| 08:59.08 | duderock | ('cause I use fluxbox) |
| 08:59.22 | swytch | duderock: also disable rpc, nfs, samba etc etc etc. for security. (-8 |
| 08:59.48 | duderock | I'm positive none of those are pre-installed on ubuntu server |
| 08:59.59 | duderock | well, not totally sure about rfc... |
| 09:00.15 | benjk | rpc is disabled by default |
| 09:00.50 | duderock | not familiar with rpc... |
| 09:01.47 | duderock | benjk: did you use the zaptel src from asterisk's svn? |
| 09:02.05 | benjk | indirectly |
| 09:02.09 | benjk | I use BRIstuff |
| 09:02.15 | swytch | duderock: its a ber-encoded remote procedure call thing (like xml-rpc but "binary") needed by nfs, the network file system. dont know anything else that uses it. |
| 09:02.50 | sumasuma | Is there is any C Implementation for ICE (Interactive Connectivity Establishment) ? |
| 09:05.22 | CSerpent | has anyone got BRIstuff working on Zap 1.2.8 yet? |
| 09:05.53 | benjk | there is an experimental version of BRIstuff for 1.2.x, not sure how high up you can use that though |
| 09:07.14 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.228) |
| 09:08.05 | CSerpent | ah fair enough, I know the PRE-1s build does work on Zap 1.2.6 but after trying to build it with 1.2.8 it does go through but no calls get through |
| 09:08.15 | CSerpent | that was a few days ago admittedly :) |
| 09:08.33 | duderock | a simple question: I'm fairly new to telephony.. If I am using VOIP service only, do I need zaptel hardware? (on a data T1 line) |
| 09:08.34 | benjk | well, I am fine with BRIstuff stable |
| 09:08.47 | benjk | no |
| 09:08.49 | benjk | but |
| 09:08.51 | duderock | i.e. TE110p? |
| 09:09.08 | benjk | on Asterisk there are several dependencies on zaptel hardware's timing source |
| 09:09.36 | benjk | you can use various workarounds but zaptel hardware timing appears to still be better |
| 09:10.35 | duderock | well, I do plan to use a card w/ 2 FXO & 2 FXS.... |
| 09:10.56 | duderock | My company wishes to retain 2 POTS lines for emergency dialing/faxing |
| 09:11.15 | duderock | and power failures.. |
| 09:11.50 | benjk | well, if the power is down and you don't have UPS for your * box, it doesn't really help you |
| 09:12.10 | benjk | so in that regard there is little difference between POTS and ISDN |
| 09:12.21 | duderock | I have a UPS & a generator for extended failure |
| 09:12.53 | benjk | then you don't require POTS lines as a measure against power failure |
| 09:13.40 | duderock | True, unless I run out of gas ;) |
| 09:14.03 | duderock | though I work for a large livery service, so that's a lot of gas tanks to syphon! |
| 09:14.07 | benjk | I don't know where you are, but where I am if there is a power failure it is usually short |
| 09:14.17 | duderock | Usually it is.... |
| 09:14.23 | CSerpent | I think by that time people will have gone home ;) |
| 09:14.35 | duderock | Well, I sure hope so... |
| 09:14.47 | duderock | But we are a 24/7 service.. |
| 09:14.48 | benjk | anyway, POTS is not for the faint of heart |
| 09:15.11 | duderock | Oooh, a challenge! ;) |
| 09:15.16 | benjk | especially not if you do it for bosses who insist on professionality, like having backups etc |
| 09:15.32 | *** join/#asterisk vlt (n=dm@p54B33DE9.dip0.t-ipconnect.de) |
| 09:15.36 | benjk | if you can avoid POTS, try to avoid it |
| 09:16.20 | duderock | Like I said, it's just for emergencies. They want a phone on the wall that will work even if the computers are down. |
| 09:16.29 | benjk | cell phones are usually better backups against landline phone outages anyway |
| 09:16.39 | tessier | POTS is fine if all you are going to do is plug an analog phone or fax into it |
| 09:16.40 | duderock | And I had planned to purchase the TDM400 to hook up a couple of fax machines anyhow.. |
| 09:16.51 | tessier | Avoid POTS interface hardware like the plague. |
| 09:16.54 | benjk | in that case, get a single POTS line and DO NOT EVEN THINK ABOUT CONNECTING IT TO ASTERISK |
| 09:17.16 | benjk | plug in that analog phone set directly and leave it alone |
| 09:17.25 | benjk | absolutely |
| 09:17.35 | *** join/#asterisk Splat (n=Splat@220-253-136-216.TAS.netspace.net.au) |
| 09:17.44 | duderock | hey, that's why I'm here - good advice. |
| 09:17.51 | benjk | I always tell any of my clients who want fax to use their ADSL line as a dedicated fax machine line |
| 09:17.58 | benjk | wirthout any connection to the PBX |
| 09:18.07 | MrChimpy | if something so awful happens that you're relying on POTS your asterisk box is probably toast |
| 09:18.16 | benjk | just the fax machine into the splitter along the ADSL modem and be done |
| 09:18.27 | MrChimpy | yep. good plan. |
| 09:18.46 | duderock | I don't think this will work with a T1.. |
| 09:18.55 | benjk | fractional T1 |
| 09:19.03 | benjk | if you dont need a full one |
| 09:19.05 | duderock | which is currently what we have. |
| 09:19.33 | benjk | then get a single POTS line connected directly to an analog phone and fax machine, done |
| 09:19.54 | duderock | We have around 12-15 simultaneous workstations max, will 1 T1 be enough for 12 VOIP channels, and data? |
| 09:20.10 | duderock | I already have the budget for 2.. |
| 09:20.33 | duderock | I had planned to put the asterisk box on it's own... |
| 09:20.36 | benjk | do you intend to use the T1 for PRI or for data or for both? |
| 09:21.07 | duderock | I had planned on data only |
| 09:21.20 | duderock | I hadn't planned on using it for PRI |
| 09:22.03 | benjk | so you want to get all your phone "lines" from a VoIP provider over then net then? |
| 09:22.21 | duderock | My plan was to have each box on it's own (data) T1, and use a provider, such as Teliax for VoIP. |
| 09:22.34 | benjk | sure you can do that |
| 09:22.53 | duderock | then use Squid to prioritize traffic over the LAN for VoIP |
| 09:23.19 | benjk | however, if your bosses are so concerned about backup that they want an anlog line and phone that still works if all the computers are down, then you probably shoudl look into getting at least a few voice channels over the T1 |
| 09:23.38 | benjk | find out what the smallest bundle is you can get from your telco |
| 09:23.43 | duderock | Then do I need a channel bank? |
| 09:23.53 | benjk | not necessarily |
| 09:24.08 | benjk | if you want to use analog phone sets then a channel bank can be a good idea |
| 09:24.33 | duderock | I see. Well I think I'm gonna try 1 T1 & 2 POTS lines and see how it goes.. |
| 09:24.40 | benjk | but if you use IP phones (not computer softrware, real IP phones) then you can use LAN switches instead |
| 09:25.21 | benjk | anyway, my recommendation would be to order one of those T1s as a fractional T1, split between voice (PRI) and data |
| 09:25.35 | benjk | llike say 4 channels PRI and the remaining capacity for data |
| 09:26.13 | benjk | then you know you will always have phone service even if those VoIP providers cannot deliver |
| 09:26.37 | duderock | And will the digium card handle the voice & data then? |
| 09:26.47 | benjk | because it is many many many times more likely that you can't get phone service on all your VoIP providers' services than it is to have a power outage |
| 09:27.21 | benjk | Both Digium and Sangoma's T1 cards do both voice and data, and do so in split mode |
| 09:27.29 | duderock | interesting. |
| 09:28.08 | duderock | But if I wanted to plug in a fax machine, and didn't want the headache of configuring a TDM-400 with 4 FXS ports, then I'd want a channel bank, no? |
| 09:28.24 | benjk | could use a channel bank yes |
| 09:28.37 | benjk | a bit overkill though for a single fax machine |
| 09:28.37 | MrChimpy | tessier: i'm waiting for mine |
| 09:28.42 | duderock | ok. this is making sense now.. glad I stayed up late... |
| 09:29.01 | MrChimpy | just get a single POTS line and straight T1. make it easy on yourself. |
| 09:29.05 | benjk | if they want an analog backup line, why not just plug the fax machine into the analog backup line and be done with it? |
| 09:29.12 | duderock | Actually it will be 1 hylafax server... |
| 09:29.43 | benjk | in that case you may want to look into OpenPBX and Unicall/SpanDSP with T38 support |
| 09:30.09 | duderock | yeah, i'm just not sure if 1 T1 will satisfy our VoIP & data requirements... |
| 09:30.20 | benjk | its a fork off Asterisk which has the advantage that it can embed GPL only libs which Digium don;t want because of their dual licensing |
| 09:30.22 | benjk | ie Unicall |
| 09:30.23 | duderock | we use FlightVIEW for live flight tracking.. |
| 09:31.25 | duderock | Is OpenPBX similar to config? |
| 09:31.37 | duderock | I'd hate to waste all I've learned about * |
| 09:31.49 | benjk | It's afork of Asterisk |
| 09:32.02 | benjk | same code base, with mods |
| 09:32.37 | benjk | but GPL only, so there is no hesitance to embed GPL only libs, like Unicall/SpanDSP |
| 09:33.10 | benjk | are the newer cards crap still? |
| 09:33.24 | benjk | the newer post-Zapata designs I mean? |
| 09:33.31 | MrChimpy | i've lost trust in them after my TE411P |
| 09:34.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:34.21 | MrChimpy | i just had the TE411s sucking all the CPU off well specced servers |
| 09:34.39 | sumasuma | MrChimpy: i lost trust in them when i bought x100p itself |
| 09:34.42 | MrChimpy | well, there were 2 in the box. |
| 09:34.56 | duderock | wow, i'm glad I came here before I bought any hardware... (well, besides the server itself.) |
| 09:35.03 | sumasuma | MrChimpy: you tried Sangoma cards ? |
| 09:35.07 | MrChimpy | i'm doing a high load app. it managed about 600 calls a min then started to fall apart |
| 09:35.37 | MrChimpy | sumasuma: i've ordered sangoma two 4 port jobs to evaluate |
| 09:35.54 | MrChimpy | from what i've read they're about 50% more efficient CPU wise |
| 09:36.02 | benjk | duderock, it depends on the time of day |
| 09:36.15 | benjk | right now its night time in most of the US |
| 09:36.20 | MrChimpy | still, working with the digium cards made me optimise the entire app which is a good thing :) |
| 09:36.30 | benjk | during the US daytime this channel is full of Digium fanboys |
| 09:36.33 | duderock | wow.. the sangoma cards are really reasonable |
| 09:36.57 | benjk | but if you are in any other part of the world, you are less likely to like Digium because Digium is less likely to care about the place you live |
| 09:37.11 | sumasuma | MrChimpy: Mark Spencer really lost trust in giving quality hardware to customers |
| 09:37.24 | duderock | well, I sleep during the day to avoid such fanboys! ;) |
| 09:37.30 | benjk | heh |
| 09:37.57 | sumasuma | MrChimpy: I paid $150 to buy a single x100p card, which was worster than ever i bought for $6.95 in ebay |
| 09:38.05 | benjk | there are two factions, the Digium fanboys and the Digium critics |
| 09:38.19 | benjk | the critics are in #openpbx and #freeswitch |
| 09:38.31 | MrChimpy | sus: we spent about $7500 on digium cards |
| 09:38.56 | MrChimpy | it's not my money though, and i don't mind funding them a bit |
| 09:39.12 | benjk | then you wont mind to fund Sangoma either I guess |
| 09:39.23 | benjk | for work on libSangoma and Unicall for example |
| 09:39.26 | sumasuma | MrChimpy: You got g729 licenses from them also ? |
| 09:39.36 | MrChimpy | but through having shitty E1 hardware they're losing about $250k's worth of hardware purchases from me |
| 09:39.45 | duderock | Heh, my company spent $6000 last year on a system that only records phone calls. I'm building the entire PBX for less than that! |
| 09:39.50 | duderock | with phones! |
| 09:40.11 | MrChimpy | sum: i don't use g729 yet |
| 09:40.59 | MrChimpy | dude: the system i wrote is replacing commercial boxes that cost £70,000 (+ support) with ones that cost £5000 |
| 09:41.21 | MrChimpy | and it does better call volumes, even when crippled by digium cards. |
| 09:41.42 | benjk | MrChimpy, you are going to get yourself banned from #asterisk |
| 09:41.43 | benjk | :D |
| 09:41.45 | MrChimpy | and we have about 20 of these boxes, which I can replace with about 4 of these things :) |
| 09:41.46 | duderock | They were even more thrilled when I gave them webmail for $300 earlier this year. |
| 09:41.52 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 09:42.07 | MrChimpy | benjk :) |
| 09:42.17 | benjk | you're such a troll |
| 09:42.19 | benjk | :P |
| 09:42.28 | benjk | and a heretic |
| 09:42.33 | sumasuma | MrChimpy: Pay money and get banned ;) |
| 09:42.40 | benjk | I should know |
| 09:42.47 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 09:42.48 | benjk | because I am a heretic and a troll, too |
| 09:42.57 | duderock | hey, i got kicked from the ubuntu channel for recommending Webmin! |
| 09:43.04 | benjk | heh |
| 09:43.11 | MrChimpy | hmm. a bit hairy... |
| 09:44.20 | benjk | well, duderock, if you ever get banned here, you can joint #openpbx and #freeswitch, its very difficult to obtain that privilege there |
| 09:44.38 | duderock | sounds like a challenge ;) |
| 09:44.39 | duderock | lol |
| 09:44.45 | AsteriskAlbania | grandstream phones freezes when call ends . This happens from the clients that are on the net other than my office, where may be the fault, linux , system , asterisk , codecs , internet line? How can I trace the problem |
| 09:44.47 | benjk | although we almost kicked yate diana out this week |
| 09:45.40 | sumasuma | diana yate why ? |
| 09:45.49 | benjk | diana from #yate |
| 09:46.10 | benjk | she was hurling personal attacks and insults at RoyK and Coppice |
| 09:46.14 | benjk | repeatedly |
| 09:46.50 | duderock | AsteriskAlbania: Do you have other phones connected outside your LAN or just the Grandsteams? |
| 09:47.16 | benjk | we are ok with heated arguments and different opinion fight to the bone, but insults are not on |
| 09:48.31 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 09:48.38 | benjk | hi zoa |
| 09:48.46 | zoa | hey ho |
| 09:48.59 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:48.59 | benjk | hows your iax phone coming along? |
| 09:49.32 | zoa | good good |
| 09:49.34 | zoa | sip in beta |
| 09:49.47 | benjk | which stack are you using for that? |
| 09:51.26 | sumasuma | Is anyone selling g729 licenses other than digium ? |
| 09:52.13 | zoa | no |
| 09:52.46 | benjk | its 50.000 USD prepaid minimum |
| 09:52.59 | zoa | reciprocate |
| 09:53.24 | benjk | need to get a bit of volume before its worthwhile getting a g729 royalty license from the g729 consortium |
| 09:53.38 | benjk | ah, ok |
| 09:54.09 | benjk | the patents for g729a run out in a year and a bit |
| 09:54.43 | zoa | i thought a sip phone with reciprocate would be easy |
| 09:54.48 | zoa | :( |
| 09:54.51 | benjk | haha |
| 09:55.12 | tessier | benjk: g729 will be free in a year or so? Cool! |
| 09:55.23 | benjk | if you get stuck ... the freeswitch guys are now all excited about that sip stack they found |
| 09:55.26 | benjk | Sofia or so |
| 09:55.42 | zoa | oh its finished already so |
| 09:55.46 | benjk | oh well |
| 09:55.53 | benjk | g729a |
| 09:55.55 | RoyK | benjk: she has those every now and then |
| 09:55.58 | RoyK | hi, zoa |
| 09:56.00 | RoyK | long time |
| 09:56.24 | benjk | not g729b |
| 09:56.49 | benjk | RoyK, you mean miss yate? |
| 09:57.44 | zoa | hey ho roy |
| 09:57.53 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:57.54 | benjk | RoyK, when you have the time, maybe you want to check out the design/arch docs I checked in |
| 09:58.13 | benjk | its about removing CLI and Manager Interface |
| 09:58.23 | RoyK | benjk: what sorts? where can i find them? |
| 09:58.26 | benjk | let me know what you think |
| 09:58.33 | benjk | check the revision log on trac |
| 09:58.36 | RoyK | removing cli/manager sounds like a brilliant idea :) |
| 09:58.40 | RoyK | ok |
| 09:59.56 | tessier | Removing the manager interface from asterisk?! |
| 10:00.12 | benjk | yep |
| 10:00.15 | tessier | Why? |
| 10:00.20 | benjk | well, not from asterisk, openpbx |
| 10:00.22 | RoyK | i beleive benjk is talking about openpbx |
| 10:00.25 | tessier | oh |
| 10:00.28 | benjk | http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx-cli/ARCHITECTURE |
| 10:00.32 | RoyK | the manager and cli interfaces are crap |
| 10:00.38 | tessier | Because I have a ton of code that depends on the manager interface in asterisk. :) |
| 10:00.46 | tessier | Hopefully you intend to replace them with something better. |
| 10:00.56 | benjk | we're not going to replace it without a replacement |
| 10:01.02 | benjk | read the doc |
| 10:01.37 | RoyK | 'replace it without a replacement' :) |
| 10:01.37 | RoyK | ~lart benjk |
| 10:01.37 | benjk | stating the obvious |
| 10:01.37 | benjk | :D |
| 10:01.45 | tessier | It's deja vu all over again |
| 10:01.58 | benjk | deju vu? |
| 10:02.18 | tessier | OpenPBX is based on asterisk? |
| 10:02.27 | RoyK | it started off as a fork |
| 10:02.29 | benjk | its a fork of asterisk |
| 10:02.56 | RoyK | but work is being put into opbx to make it what asterisk should have been in the first place |
| 10:02.59 | RoyK | not just a pile of code |
| 10:03.13 | tessier | For example? |
| 10:03.17 | benjk | pile of Commodore VIC-20 BASIC code |
| 10:03.30 | benjk | for example removing the CLI and the manager interface :) |
| 10:03.47 | benjk | and replace it with a true client/agent/server architecture |
| 10:03.51 | benjk | pluggable |
| 10:03.54 | tessier | How does that help anything? And if it did help why wouldn't you just contribute that to asterisk instead of forking? |
| 10:03.55 | MrChimpy | sounds promising, aspects of asterisk make me want to barf |
| 10:03.58 | RoyK | and what is this new storage stuff in opbx again? |
| 10:04.06 | benjk | SOAP, XML-RPC, Jabber, remote-cli |
| 10:04.17 | MrChimpy | tessier: asterisk's design is fundamentally flawed and it's a bit messty |
| 10:04.21 | benjk | replaceing all those horrible linked lists with hashtables |
| 10:04.33 | MrChimpy | it's a byproduct of how it was written |
| 10:04.40 | tessier | So far this all sounds like stuff that can be fixed in asterisk instead of a fork |
| 10:04.44 | tessier | MrChimpy: How was it written? |
| 10:04.45 | benjk | and replacing all those suck ass strcmp() and strcasecmp() calls with hash comparisons |
| 10:04.48 | RoyK | tessier: as long as asterisk isn't pure gpl, you can't use all sorts of existing stuff, so it won't work |
| 10:04.58 | benjk | tessier, we have tried that |
| 10:05.02 | benjk | didin't work |
| 10:05.17 | benjk | because there are those Digium fanboys who resist any sane proposal |
| 10:05.24 | MrChimpy | tessier: stuff chucked on top of other stuff by many people with some rather short sighted design |
| 10:05.39 | MrChimpy | AMI/AGI stuff is vile in places |
| 10:06.24 | tessier | RoyK: According to LICENSE it is pure GPL. |
| 10:06.39 | benjk | also, did you know that if you log in to the Asterisk CLI over a slow modem link your calls will stop for as long as it takes to finish the console output with every command? |
| 10:06.54 | pigpen | Not. |
| 10:07.02 | RoyK | tessier: except that any code needs to be owned by digium |
| 10:07.17 | RoyK | tessier: you need to disclaim all contributions to digium to get it in there |
| 10:07.33 | benjk | tessier, Digium wouldn't use any pure GPL libs because they want to have their cake and it it too |
| 10:07.33 | benjk | for example Unicall |
| 10:07.40 | tessier | hrm |
| 10:07.46 | AsteriskAlbania | duderock: Almost Grandstream maybe some XLITE but very few |
| 10:07.47 | RoyK | tessier: so if asterisk was to implement video transcoding, they can't simply use libavcodec, because other people have the copyright for that code |
| 10:07.52 | benjk | they wanted it but because its GPL and they cant have it signed over to them for free, they won't have it |
| 10:08.07 | tessier | I see. That is rather shortsighted of them. |
| 10:08.15 | tessier | Digium impresses me less and less. |
| 10:08.16 | RoyK | tessier: you can say that again |
| 10:08.43 | benjk | With all the cash Digium make these days, why wouldn't they have offered steve some money to purchase the rights to Unicall? |
| 10:08.51 | benjk | thats pure greed |
| 10:08.57 | tessier | benjk: How do you know they are making a lot of cash? |
| 10:09.22 | benjk | they were making 10 million USD per year before they stopped telling the public how much they make |
| 10:09.32 | tessier | And what were their expenses? |
| 10:09.33 | benjk | and now they say they double revenue every year |
| 10:09.51 | benjk | so they are probably at about 25 million a year |
| 10:09.54 | CSerpent | :O |
| 10:10.01 | benjk | how many expenses can they have |
| 10:10.09 | CSerpent | Mark's salary? :) |
| 10:10.10 | benjk | a couple of VON shows and that |
| 10:10.12 | tessier | Developing hardware is super expensive |
| 10:10.27 | benjk | developing hardware isn't that expensive |
| 10:10.33 | benjk | not their kind of hardware |
| 10:10.49 | benjk | its an expense, but peanuts if you have 10m revenue p.a. |
| 10:10.51 | JT | all they're really doing is stringing ASICs together |
| 10:10.52 | CSerpent | if they sub their hardware building to some place in taiwan it will be very reasonable |
| 10:11.08 | JT | it's not like they're inventing some super duper algorithm DSPs |
| 10:11.10 | tessier | CSerpent: Over what volume? You have no idea. |
| 10:11.23 | benjk | and their developers are working for a mattress in the basement and a daily ration of red bull |
| 10:11.40 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 10:11.51 | duderock | mmm. red bull. |
| 10:12.58 | AsteriskAlbania | Dont care for the money they are wininning , care-ing for the money I am loosing, can someone help: grandstream phones freezes when call ends . This happens from the clients that are on the net other than my office, where may be the fault, linux , system , asterisk , codecs , internet line? How can I trace the problem |
| 10:13.13 | benjk | sip debug is your friend |
| 10:13.48 | duderock | AA: log into your CLI and see what it says when the client disconnects that should give you a starting point |
| 10:14.00 | benjk | sip debug |
| 10:14.10 | tessier | AsteriskAlbania: Grandstream? Do you not like this client or something? |
| 10:14.15 | AsteriskAlbania | benjk: there is a lot of debuging |
| 10:14.27 | benjk | sip debug ip .... |
| 10:14.30 | AsteriskAlbania | It has worked perfectly for 2 months |
| 10:14.36 | duderock | hey, I personally <3 my GXP-2000. |
| 10:14.42 | AsteriskAlbania | I dont know whether is Grandstream fault |
| 10:14.51 | benjk | sip debug will tell you |
| 10:15.56 | CSerpent | tessier: too true, don't get me wrong I think the fact that * and Digium's bringings into the open source market of * is great, but $10m/yr profit.. that's an awful lot - and a lot of money for their directors |
| 10:16.27 | AsteriskAlbania | benjk: I have activated for 1 peer |
| 10:16.54 | benjk | directors? |
| 10:17.08 | benjk | I think there's Mark and Greg and that's probably it |
| 10:17.29 | CSerpent | well they'll get dividends from that profit |
| 10:17.43 | CSerpent | but that's the sign of a successful business |
| 10:17.57 | benjk | and the fanboys even collected money to buy him a whirlpool |
| 10:18.03 | benjk | its not a company |
| 10:18.06 | benjk | its a cult |
| 10:18.09 | CSerpent | oh right.. my mistake |
| 10:18.12 | CSerpent | lol |
| 10:18.27 | CSerpent | have to get the BBC to put it on their cult site ;) |
| 10:18.36 | benjk | heh |
| 10:18.38 | RoyK | tessier: they're not really developing much hardware. they use existing chipsets and lego them |
| 10:18.59 | benjk | the BBC is a bunch of terrorist, anarchist and communist propagandists |
| 10:19.04 | benjk | didn't you know that? |
| 10:19.06 | duderock | so with the Sangoma PCI cards, you can chain 4 of them on 1 PCI slot? |
| 10:19.12 | CSerpent | ssh.. MI-5 will be after you! |
| 10:19.22 | RoyK | benjk: yeah. and the white house always say the truth |
| 10:19.35 | CSerpent | Wierd Al speaks the truth ;) |
| 10:19.35 | benjk | that's what I was told a short while ago by some Redneck here |
| 10:19.46 | RoyK | rotfl |
| 10:20.07 | RoyK | BBC is right wing stuff compared to the papers I read |
| 10:20.10 | benjk | the casualties of Israeli air raids on Lebanon were all hezbollah terrorists |
| 10:20.15 | benjk | so they said |
| 10:20.21 | RoyK | yeah |
| 10:20.22 | benjk | and I said, 1000? |
| 10:20.30 | RoyK | three year old terrorists |
| 10:20.36 | RoyK | or to-be-terrorists |
| 10:20.36 | benjk | they said, 1000? where did you get that number from? |
| 10:20.37 | RoyK | or something |
| 10:20.39 | CSerpent | BBC News, powered by the Morning Star? :) |
| 10:20.45 | benjk | I said it was on BBC news |
| 10:21.07 | benjk | they said it was a terrorist, anarchist, communist propaganda outlet not to be trusted |
| 10:21.18 | RoyK | smart bombs pinpointing locations? |
| 10:21.27 | RoyK | smart cluster bombs over civilians is nice |
| 10:21.44 | *** join/#asterisk raggi (n=ra66i@skynet.cx) |
| 10:21.54 | CSerpent | well it's always been admitted that the BBC is manipulated by the UK Govt |
| 10:21.57 | duderock | That's why I get all of my news from The Onion. |
| 10:22.04 | benjk | hahaha |
| 10:22.07 | CSerpent | oh the Onion.. THE source |
| 10:22.47 | CSerpent | I prefer El Reg :) |
| 10:23.10 | benjk | I think you won't find any more objective mainstream news network on the planet |
| 10:23.37 | benjk | the wheather forecast on BBC World are always totally off though |
| 10:23.50 | benjk | that really can't be trusted |
| 10:24.16 | CSerpent | "in this area where Typhoon 34E is, it will be mostly cloudy and you can expect it to rain a little" |
| 10:24.18 | benjk | they report sunshine for Tokyo and we get rain |
| 10:24.21 | duderock | I think that can be said for weather forcasts in general.. |
| 10:24.42 | CSerpent | Use Aviation Forecasts (TAFs).. if it says PROB30, you dismiss it ;) |
| 10:24.43 | benjk | especially if you attempt to cover the entire planet in like 90 seconds |
| 10:25.27 | *** join/#asterisk dalekurt (n=dalekurt@port0033-afw-adsl.cwjamaica.com) |
| 10:25.53 | duderock | life in Wisconsin, USA teaches you to never trust meteorology. |
| 10:26.21 | *** join/#asterisk gabry (n=gabry@host162-9.pool80181.interbusiness.it) |
| 10:26.38 | benjk | but when Israel violated the ceasefire right away, it wasn't mentioned on CNN until 55 mins after the hour, it was the last thing just before the final hour commercials |
| 10:27.11 | duderock | yeah, and someone probably got fired for putting it on the teleprompter |
| 10:27.16 | benjk | it was on the front pages and number 1 items on every TV channel on the planet outside of the US (and presumably Israel, not sure about that though) |
| 10:28.59 | benjk | I have heard though that the Israeli media are more objective and critical about their own governemnt than the US media |
| 10:29.34 | duderock | That doesn't surprise me... |
| 10:29.43 | benjk | anyway, BBC is considered communist in the Imperial United States |
| 10:29.48 | duderock | We're fed information as if we're 4 years old... |
| 10:30.33 | AsteriskAlbania | Retransmitting #6 (NAT) to 217.24.246.103:54947: |
| 10:30.34 | AsteriskAlbania | BYE sip:1008@192.168.0.215 SIP/2.0 |
| 10:30.34 | AsteriskAlbania | Via: SIP/2.0/UDP 217.24.244.137:5060;branch=z9hG4bK0f0c4c43;rport |
| 10:30.34 | AsteriskAlbania | From: <sip:35547400074@217.24.244.137;user=phone>;tag=as10d494b8 |
| 10:30.34 | AsteriskAlbania | To: "Dritan Muka" <sip:1008@217.24.244.137;user=phone>;tag=882f1ba4cd5b1078 |
| 10:30.34 | AsteriskAlbania | Contact: <sip:35547400074@217.24.244.137> |
| 10:30.36 | AsteriskAlbania | Call-ID: 22b47cfeeaad1527@192.168.0.215 |
| 10:30.38 | AsteriskAlbania | CSeq: 102 BYE |
| 10:30.40 | AsteriskAlbania | User-Agent: Asterisk PBX |
| 10:30.42 | AsteriskAlbania | Max-Forwards: 70 |
| 10:30.44 | AsteriskAlbania | Content-Length: 0 |
| 10:30.45 | MrChimpy | the BBC is considered communist by the UK government :) |
| 10:30.53 | AsteriskAlbania | sorry for 11 lines, this is what it makes on debug sip |
| 10:31.35 | AsteriskAlbania | benjk: What can I understand here |
| 10:31.40 | CSerpent | but the BBC is puppeted in forms by the UK govt |
| 10:31.47 | CSerpent | and the UK govt is puppeted in form by the EU and US |
| 10:31.54 | MrChimpy | during the shock + awe bit of the iraq war they were all over the place. they wouldn't call people terrorists (and quite rightly so) and then talked about liberation. |
| 10:31.57 | benjk | well, usually you use pastebin.ca for that but its not buys here now, so I guess its ok |
| 10:32.02 | benjk | busy |
| 10:32.17 | AsteriskAlbania | :) |
| 10:32.50 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:32.55 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 10:32.59 | benjk | that doesn't tell much |
| 10:33.12 | MrChimpy | nah, BBC isn't too bad. keeps biting the goverment. sky is more pro blair cos of murdoch. |
| 10:34.02 | yatesy | i've always liked the BBC for remainingg more impartial than the rest of them |
| 10:35.05 | CSerpent | I had Canadian satellite put in when I was at a friend's in Texas.. he was shocked by the different reporting to what he saw on domestic channels |
| 10:35.19 | benjk | heh |
| 10:35.22 | MrChimpy | the daily moose news? :) |
| 10:35.30 | MrChimpy | forecast for lumberjacks? |
| 10:35.58 | MrChimpy | or maybe the lack of reports on gun crime shocked him :) |
| 10:36.25 | CSerpent | plenty of gun crime in Canada |
| 10:36.36 | MrChimpy | much less than US |
| 10:36.38 | gabry | Hi all...can anyone help me in FOP configuration? |
| 10:36.44 | duderock | Q: Is it possible to register a sip/iax2 device to a queue like you would an extension? |
| 10:37.05 | CSerpent | well, of course, anywhere where you're allowed to bear arms is gonna have more of a gun problem |
| 10:37.05 | duderock | (so users can monitor, but not dial out/receive calls?) |
| 10:37.25 | duderock | I think they should change it to "arm bears" |
| 10:38.12 | gabry | nobody wants to help me? |
| 10:38.30 | benjk | Intel is feeling the Apple curse now |
| 10:38.36 | CSerpent | oh? |
| 10:38.42 | benjk | 20.000 jobs at intel will be axed |
| 10:38.56 | *** join/#asterisk aixa (n=Miranda@www.crediweb.lv) |
| 10:38.58 | CSerpent | that's hardly an apple curse, it's called lagging behind AMD :) |
| 10:39.02 | aixa | hi |
| 10:39.08 | CSerpent | hi aixa |
| 10:39.12 | benjk | the going joke is that every chip company Apple will team up with has its fortunes go down |
| 10:39.15 | aixa | everyone ever tried to use asterfax |
| 10:39.17 | aixa | ? |
| 10:39.28 | CSerpent | I love my MacBook Pro.. works great |
| 10:39.40 | aixa | or txfax and rxfax applications |
| 10:39.41 | benjk | I love my MacBook Pro too :) |
| 10:39.51 | duderock | I wish I could afford a mac......So I could buy 4 PC's! |
| 10:40.01 | benjk | but its nevertheless funny how that decades old "curse" appears to hold true |
| 10:40.17 | benjk | that's a myth though |
| 10:40.30 | benjk | Macs are no longer any more expensive than PCs |
| 10:40.39 | aixa | txfax crashes the asterisk when it cant understand the format of tiff file is that normal? |
| 10:40.48 | duderock | Aren't macbooks ~1800USD? |
| 10:40.49 | HaMYaI | what's the wait() in c? |
| 10:40.58 | raggi | benjk: but nor are they much better. |
| 10:41.09 | benjk | aixa, coppice has been wokring on some stuff this week and I think that was onw o the problems |
| 10:41.21 | aixa | oh, okay |
| 10:41.22 | benjk | but I also seem to remember that it was fixed |
| 10:41.33 | benjk | just from listening in in #openpbx |
| 10:41.33 | aixa | so i managed to install the software at the right time :) |
| 10:41.47 | benjk | I think it was fixed last night or the night before |
| 10:41.54 | duderock | i thought fax over VoIP was unusable....no? |
| 10:41.54 | benjk | and it may not be for asterisk yet |
| 10:41.57 | aixa | benjk: , so i should realy checkout latest from spandsp |
| 10:41.58 | aixa | ? |
| 10:42.16 | benjk | Unicall and SpanDSP is developed first on OpenPBX, then ported to Asterisk |
| 10:42.24 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 10:42.32 | benjk | or more like first tested etc |
| 10:42.43 | CSerpent | Macbook Pros are expensive, or you can get student discount on them and get 20% off :) |
| 10:42.57 | benjk | you may want to come to #openpbx a little later (in about 2-3 hbours) and ask coppice |
| 10:43.21 | aixa | duderock: fax over over VOIP would cause some problems (unless you use losless transport codecs) |
| 10:43.24 | benjk | they are only expensive compared to cheapo computers below their own class |
| 10:43.29 | aixa | benjk thanks |
| 10:43.41 | benjk | if you want to go low cost then you have to compare the MacBook |
| 10:43.49 | CSerpent | no matter which way I look at it, it's been more reliable, better connectors, works well |
| 10:43.56 | aixa | duderock: : however - we use asterisk with isdn bri card |
| 10:44.19 | duderock | so over your LAN the faxing capabilities are okay? |
| 10:44.19 | CSerpent | and the magneto power clip is fantastic, number of times I've knocked power cables! |
| 10:44.25 | aixa | and want to get rid of that old fax machine of monsterous size |
| 10:44.30 | benjk | my power connector is about to come off, the cable is |
| 10:44.45 | duderock | aixa: I wish I could get my users to quit feeding paper into that thing! |
| 10:44.45 | benjk | that's after using it daily since Feb 15 |
| 10:45.05 | benjk | usually it takes 1 year for an Apple power supply to have its cable break |
| 10:45.07 | aixa | duderock :))) I have rather young crowd |
| 10:45.15 | duderock | We've had Hylafax up & running for 3 mos, and only recently I have made it mandatory to receive faxes with it.. |
| 10:45.18 | benjk | or any other notebook power supply for that matter |
| 10:45.37 | CSerpent | usually cos it gets damaged in the socket and or beaks the connector inside |
| 10:45.43 | CSerpent | breaks* |
| 10:45.46 | benjk | indeed |
| 10:45.58 | benjk | so I wish they would make the cord pluggable too |
| 10:45.58 | aixa | duderock - most of them using skype, ims, stuff like that; it is not that hard to get them to use new methods |
| 10:46.08 | benjk | not only the cord between the power supply and the wall |
| 10:46.20 | CSerpent | well hence the macbook's system |
| 10:46.21 | benjk | but also the cord between the power supply and the computer |
| 10:46.25 | duderock | aixa: I work with a fairly young crowd as well, but they don't react well to new technology. |
| 10:46.30 | CSerpent | it's a magnetic holder with touch contact to charge |
| 10:46.31 | benjk | no, its worse with the new system |
| 10:46.46 | duderock | aixa: they all freaked out when I made them start using Zimbra webmail.. ;) |
| 10:46.50 | benjk | I am telling you its almost broken through after only 6 months |
| 10:47.00 | benjk | I have had my MBP since Feb 15 |
| 10:47.19 | benjk | the cable is coming off right behind the magnetic connector block |
| 10:47.35 | HaMYaI | can anyone tell why sleep() or wait() can't be used in pbx.c ? |
| 10:48.00 | duderock | HaMYaI: Your copy of asterisk has ADHD. |
| 10:48.01 | aixa | duderock - what have they been using before? |
| 10:48.16 | duderock | aixa: pop3/OE |
| 10:49.02 | duderock | aixa: but some of our users share/use multiple workstations so I wanted saved/sent mail accessible from any terminal |
| 10:49.38 | aixa | okay |
| 10:49.43 | duderock | plus Zimbra rules :) |
| 10:49.53 | aixa | looking at it right now :) |
| 10:51.26 | aixa | duderock: if they were so stressed out because of changes in interface why not give them imap? |
| 10:51.33 | aixa | instead of pop |
| 10:52.01 | duderock | because I didn't want them storing stuff on local machines. w |
| 10:53.03 | benjk | ? |
| 10:53.16 | benjk | pop stores locally, imap only caches |
| 10:53.25 | aixa | exactly |
| 10:53.55 | aixa | and good old UI in most cases is kept |
| 10:54.25 | duderock | oh, yeah. |
| 10:54.31 | vlt | Hello. Can I import blf hints from another asterisk machine? And can I create blf hints for non-SIP connections? |
| 10:54.35 | pigpen | Zimbra is great...if you like Mysql |
| 10:54.46 | aixa | nevertheless the exchange database of 50 GiGs is also a problem :P |
| 10:54.52 | aixa | at least to backup |
| 10:54.54 | zoa | whats the main difference between macbook and macbook pro ? |
| 10:55.09 | pigpen | zoa, what I have and what my wife has. |
| 10:55.13 | aixa | zoa the price and neat "pro" sign |
| 10:55.14 | pigpen | :) |
| 10:55.29 | aixa | just kidding |
| 10:55.34 | benjk | the Pro has bigger screens |
| 10:55.37 | CSerpent | metal cover (pro) vs plastic (macbook), screen size up to 17" WS (pro) vs 13.1" WS (macbook) are the main two |
| 10:55.43 | zoa | and weights about 10 times more ? |
| 10:55.44 | zoa | :) |
| 10:55.49 | CSerpent | higher CPUs in the Pro |
| 10:55.57 | aixa | dont forget about cute "pro" logo |
| 10:55.57 | MrChimpy | pro has dedicated graphics |
| 10:55.58 | pigpen | I love my macbook pro, and my quad, and my dual... |
| 10:55.59 | benjk | also the Macbook has only got glossy screens, not everybodys first choice that |
| 10:56.09 | zoa | aha |
| 10:56.16 | zoa | do they both have wireless ? |
| 10:56.20 | MrChimpy | yep |
| 10:56.22 | CSerpent | hmm I had a macbook the other week in here and it had a matt screen |
| 10:56.40 | pigpen | you can order it with glossy (default) |
| 10:56.41 | MrChimpy | oh yeah, and the non-pro will probably break quite quickly |
| 10:56.48 | pigpen | but you can get it with non-glossy... |
| 10:56.54 | MrChimpy | and both are capable of cooking small meals :) |
| 10:57.04 | CSerpent | I've had macs since I had a PowerMac 650 :D |
| 10:57.04 | pigpen | you are also the "cool guy" when you need help at the apple store. |
| 10:57.07 | CSerpent | I work on both |
| 10:57.18 | pigpen | yep. |
| 10:57.43 | pigpen | I used to curse mac's on a network... |
| 10:57.50 | CSerpent | Apple have had a market share rise from 8% to 17% of the market predicted by mid 07 |
| 10:57.54 | pigpen | now, -sigh- I am a mac user. |
| 10:58.03 | pigpen | and Gentoo linux. |
| 10:58.05 | aixa | well apple finnaly got their marketing and product promotion and design (like features compatability not the looks) right |
| 10:58.06 | pigpen | :) |
| 10:58.07 | zoa | why would the non pro break quickly ? |
| 10:58.18 | pigpen | zoa, not as good quality |
| 10:58.38 | benjk | the only thing that will break quickly is the cable of the power supply unit |
| 10:58.44 | pigpen | you will also fondle the mac book pro more.... |
| 10:58.47 | zoa | its not that magnet thing ? |
| 10:58.47 | benjk | takes about 6 months to 1 year |
| 10:58.55 | benjk | not the magnet, but the cable |
| 10:58.57 | zoa | aha |
| 10:59.06 | zoa | so they just moved the problem somewhere else :) |
| 10:59.17 | benjk | unless you use it as a desktop where the cable never moves' |
| 10:59.19 | zoa | i was thinking of buying the smallest model |
| 10:59.25 | pigpen | oh..the stiff cable? |
| 10:59.27 | zoa | i mean cheapest model |
| 10:59.42 | CSerpent | Anyone up for a newton? :D |
| 10:59.43 | benjk | if you do work out of the office everyday, then it breaks fairly quickly |
| 10:59.53 | MrChimpy | newton rocked! |
| 10:59.55 | benjk | I still have two Newton 2000K |
| 10:59.57 | benjk | 2K |
| 11:00.00 | CSerpent | I remember playing with the first newton when the college I was working at got one |
| 11:00.11 | MrChimpy | i remmeber the original, mostly cos it was ARM based |
| 11:00.24 | duderock | <PROTECTED> |
| 11:00.37 | benjk | the original wasn't all that good, it had too many issues with yet unmature tech |
| 11:00.41 | benjk | immature |
| 11:01.21 | benjk | the MP2K was almost perfect, just a little too thick |
| 11:01.56 | raggi | exit |
| 11:01.59 | raggi | oops |
| 11:02.01 | benjk | it wouldn't easily get in and out of the inner breast pocket of a men's suit's jacket |
| 11:02.25 | MrChimpy | the original was crap and huge but very impressive |
| 11:02.29 | CSerpent | I just want Sat Nav for my Blackberry! :'( |
| 11:02.42 | benjk | it was cool as a kind of proof of concept for early adopters |
| 11:04.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:04.44 | hads | duderock: I threw my thinkpad across a room when I was drunk and it still goes fine :) |
| 11:05.30 | duderock | I just wish I could get mine to dock/undock properly with ubuntu.. |
| 11:05.46 | duderock | i'm close... it undocks without locking up now at least.. |
| 11:07.19 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 11:11.41 | nettie | Hi guys, anyone know if comforn noise enabled at my sip voip carrier might cause call quality problems considering asterisk doesnt support it? |
| 11:12.32 | Strom_C | nettie: yes it will - have your carrier turn off cng / vad |
| 11:13.55 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:14.27 | nettie | Strom_C what kind of problems I might expect? |
| 11:14.46 | nettie | Strom_C I keep getting Sep 1 11:45:20 NOTICE[24161] rtp.c: Comfort noise support incomplete in Asteris |
| 11:14.46 | nettie | k (RFC 3389). Please turn off on client if possible. Client IP: 88.149.128.60 |
| 11:14.46 | Strom_C | dropped frames and other general audio quality problems |
| 11:15.05 | benjk | general discomfort |
| 11:15.26 | RoyK | Strom_C: using what? |
| 11:15.26 | nettie | hey roy |
| 11:15.26 | nettie | :) |
| 11:15.26 | RoyK | hi, nettie |
| 11:15.44 | benjk | cheers |
| 11:23.18 | Arno[Slack] | hum, what is a decent value for the maxjitterbuffer option in iax.conf ? |
| 11:24.27 | Strom_C | Arno[Slack]: are you having jitter issues to begin with? |
| 11:25.11 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 11:25.20 | Arno[Slack] | no no but i'm coding an interface for the asterisk configuration and i wonder what type of value i should show to users as a good example |
| 11:25.45 | Arno[Slack] | and since i have no jitter problems i have never try this option :) |
| 11:25.51 | Strom_C | Arno[Slack]: if there's no jitter on the circuit, then jitterbuffering should not be used |
| 11:26.03 | benjk | 42 |
| 11:26.14 | Arno[Slack] | 42 yes of course ;) |
| 11:26.35 | HaMYaI | Strom_C, hi yesterday I was asking you about processing something else while let the callers listening to the Background or MusicOnHold |
| 11:26.37 | Strom_C | otherwise, you slowly increment the jitterbuffer to the point where it solves your jitter issue; too much and you just add needless latency into the connection |
| 11:27.00 | Strom_C | HaMYaI: yes, I remember, and I still have no idea what you're trying to do |
| 11:27.07 | HaMYaI | Strom_C, ok |
| 11:27.30 | HaMYaI | Strom_C, let me explain again |
| 11:27.42 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 11:28.11 | Arno[Slack] | ok thanks |
| 11:28.30 | HaMYaI | Strom_C, when the caller calls into my system I run an agi script to play the sounds to callers |
| 11:28.53 | HaMYaI | using exec('Background','filename') |
| 11:29.40 | HaMYaI | while the caller is listening to the file I wanna be able to process something else |
| 11:30.10 | HaMYaI | and be able to stop the music previously played to that caller when the condition is met |
| 11:30.24 | Strom_C | HaMYaI: it sounds to me like you're trying to compensate for bad system design...what are you processing that takes so long that you need to pacify callers with music? |
| 11:31.22 | HaMYaI | Strom_C, yes, they will have to wait like 5-10 minutes actually |
| 11:31.43 | Strom_C | HaMYaI: five to ten MINUTES?? |
| 11:31.51 | Strom_C | what the hell are you processing in the background? |
| 11:32.20 | HaMYaI | Strom_C, not because that the processing takes to long |
| 11:32.35 | HaMYaI | Strom_C, but the condition isn't met easily |
| 11:32.42 | Strom_C | what condition? |
| 11:32.54 | Strom_C | give me a clearer idea of what you're having the caller wait for |
| 11:33.06 | HaMYaI | Strom_C, what I do is actually sending a http post to another system to check if the conference room is available |
| 11:33.23 | *** join/#asterisk r_marvin (n=r_marvin@86.126.53.60) |
| 11:33.41 | Assid | HaMYaI: that doesnt take more than a few seconds |
| 11:33.54 | HaMYaI | Strom_C, is they say "YES" then I send my call through and start to charge the caller |
| 11:34.25 | Strom_C | HaMYaI: it sounds like you'd be better off using queues... |
| 11:34.26 | HaMYaI | Strom_C, yes but it's a consulting service |
| 11:34.40 | HaMYaI | Strom_C, the advisor can only talk to one person at a time |
| 11:34.57 | Assid | then use queues |
| 11:35.01 | Strom_C | HaMYaI: just use queues |
| 11:35.07 | Assid | agents and queus.. very simple |
| 11:35.11 | Strom_C | queues already do exactly what you need to do |
| 11:35.34 | HaMYaI | Assid, yeah but I will need to put them through my billing system |
| 11:35.48 | HaMYaI | Assid, with queue we can't do that easily |
| 11:35.56 | Assid | sure you can |
| 11:36.27 | Assid | when the advisor is ready to answer the call have him do a conference call witht he billing section where he punches his username/password |
| 11:36.53 | RoyK | can someone explain why asterisk rewrites RTP timestamps by default when bridging rtp? |
| 11:36.55 | Assid | that way the client only pays for his session |
| 11:37.15 | r_marvin | helo, i want to automatically record meetme conferences, however, using (s,1,monitor; s,2,meetme) produces a different file for each party |
| 11:37.38 | Strom_C | r_marvin: show application meetme |
| 11:37.57 | Strom_C | instructions for recording meetme conferences will show up :) |
| 11:38.23 | HaMYaI | Assid, they are on different systems |
| 11:38.52 | Assid | are they both using asterisk? |
| 11:38.59 | RoyK | 'restart when highly inconvenient' |
| 11:39.06 | r_marvin | Strom_C: oh, the "r" flag. sorry, i saw that before, but missed that |
| 11:39.08 | r_marvin | sorry |
| 11:39.16 | HaMYaI | Assid, the caller is waiting on my server and the advisor belongs to my partner's |
| 11:39.45 | Assid | does the queue take place in the advisors' box? |
| 11:39.56 | Assid | as well |
| 11:40.10 | Assid | or do you hold onto the queue and then pass on 1 at a time? |
| 11:40.17 | Assid | either which way.. you can still do it |
| 11:40.22 | HaMYaI | Assid, nope they're on my box |
| 11:40.37 | HaMYaI | Assid, yes only pass one at a time |
| 11:40.38 | Assid | okay so have the advisor call an extension on their box for their billing |
| 11:40.58 | *** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg) |
| 11:41.04 | Assid | and have them punch in the user/pass as usual |
| 11:41.22 | Assid | let the advisor conference the call with the billing box |
| 11:41.47 | HaMYaI | Assid, so the cdr records are on their side only |
| 11:43.08 | Assid | the billing will take place whereever the billing box is |
| 11:43.11 | Assid | just call that box |
| 11:43.14 | nettie | Strom_C I have call forward on my polycom phone but unfortunately it doesnt work when the phone is disconnected or the internet link is down, do you know where I can find some documentation on how to configured it on asterisk please? I would like the server to handle such things instead of single remote phones. thanx. |
| 11:43.20 | HaMYaI | Assid, right now I'm having on both sides |
| 11:43.29 | Strom_C | nettie: write it in a macro |
| 11:43.43 | nettie | ahh a macro.. |
| 11:44.04 | nettie | because I rember I read somethign about *70s oemthign.. |
| 11:44.08 | HaMYaI | Assid, I mean both boxes keep their own records |
| 11:44.17 | nettie | *70 does something.. but maybe was just a custom dialplan I think |
| 11:44.30 | Strom_C | *70 disables call waiting on a per-call basis |
| 11:44.42 | Strom_C | *72 enables call forwarding |
| 11:45.08 | Strom_C | http://nanpa.com/number_resource_info/vsc_assignments.html |
| 11:45.41 | nettie | because in my country we normally yse *21*NUMBERHERE# to enable it |
| 11:46.16 | nettie | of so, based on that napa assignments I should code my macro |
| 11:46.17 | nettie | right? |
| 11:46.26 | HaMYaI | ohh, the man is gone |
| 11:46.31 | Strom_C | nettie: which country are you in |
| 11:46.36 | nettie | italy |
| 11:46.39 | Strom_C | ah |
| 11:46.45 | nettie | :) |
| 11:46.47 | Strom_C | you should conform to your local numbering plan |
| 11:46.50 | nettie | yeah |
| 11:47.03 | nettie | good point.. but the idea is the one you suggested anyway right? |
| 11:47.20 | Strom_C | yes - write a macro which checks for call forwarding and whatnot |
| 11:47.21 | nettie | find the standard ones and code a macro using them |
| 11:47.25 | nettie | ok |
| 11:47.26 | nettie | thanx |
| 11:47.36 | nettie | I foudn somethign here as well to get started http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forwarding |
| 11:47.37 | HaMYaI | Strom_C: have you had a look at "pbx_builtin_background()" in pbx.c at all? |
| 11:47.47 | Strom_C | HaMYaI: I do not code |
| 11:47.56 | HaMYaI | Strom_C: ok |
| 11:49.39 | Muck- | can someone help me with capi drivers? i got a eicon diva server bri card and after installing the eicon driver i can see capiinfo but after rebooting it said capi not installed - No such file or directory (2) |
| 11:56.05 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-b027d81cb4b8c6b3) |
| 11:56.21 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.96) |
| 11:59.59 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 11:59.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:00.13 | *** join/#asterisk laichzeit (n=immortal@dsl-145-177-33.telkomadsl.co.za) |
| 12:00.21 | *** join/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 12:00.54 | *** join/#asterisk jpsaman (n=jpsaman@ip545278d7.speed.planet.nl) |
| 12:01.11 | laichzeit | hi, is it possible for asterisk to print the current context it is in to CLI? I'm having difficulties tracing a dial-plan |
| 12:01.13 | jpsaman | Does asterisk work together with ekiga? |
| 12:02.25 | RoyK | hm. is there a way to check the SIP response after a Dial_ |
| 12:02.27 | RoyK | ? |
| 12:02.41 | RoyK | as in "Dial, got 404, ".... |
| 12:03.33 | Sonderblade | have anyone purchased asterisk voice prompts? |
| 12:04.44 | zoa | nopez |
| 12:05.05 | *** join/#asterisk crCernier_ (n=crochat@adsl-84-227-2-252.adslplus.ch) |
| 12:07.24 | zoa | well somebody probably did, but i didnt :p |
| 12:07.37 | RoyK | :P |
| 12:08.39 | zoa | i use dtmf morse code |
| 12:08.44 | zoa | as voice prompt |
| 12:08.45 | RoyK | :) |
| 12:08.50 | Sonderblade | but they are so cheap... |
| 12:08.54 | RoyK | zoa: app_morse |
| 12:09.03 | zoa | oh allison is good |
| 12:09.36 | zoa | http://www.theivrvoice.com/ |
| 12:11.26 | zoa | you can buy a clone in a little box |
| 12:13.19 | stoffell_h | is it impossible (on E1) to get the telco error message in case a wrong number or no-longer-existing number is called ?? |
| 12:13.31 | zoa | yes |
| 12:13.38 | zoa | its called isdn release cause |
| 12:14.02 | stoffell_h | zoa: yes, but is it impossible to not use the hangupcause and let the telco play it's default message? |
| 12:14.19 | zoa | the telco could still play it if they want |
| 12:14.23 | zoa | but they have to enable it |
| 12:14.26 | zoa | as early media |
| 12:14.27 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 12:14.44 | zoa | and on an E1 they usually dont to my experience |
| 12:15.22 | stoffell_h | zoa: okay, but it's enabled, on the 'old' pbx the messages came from the telco. now (asterisk) it looks like i should prvide them.. |
| 12:17.09 | *** part/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 12:18.35 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 12:18.38 | RoyK | anyone that knows how i can use ethereal to detect packet loss? i have a 50MB trace, but how can I find when packets are lost? without going through the whole shite manually..... |
| 12:19.03 | RoyK | the jitterbuffer says its interpolating lost packets, but how can I know the jitterbuffer isn't fscked? |
| 12:21.13 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:22.00 | *** join/#asterisk Un1x|laptop (n=Un1x_lap@bas2-toronto63-1128742491.dsl.bell.ca) |
| 12:22.06 | Un1x|laptop | how's everyone today? |
| 12:22.41 | ambriento | g00d. tnx for asking :) what about you? |
| 12:22.41 | Un1x|laptop | Good i Hope :p? |
| 12:22.52 | Un1x|laptop | heh bored waiting here for delivery to arrive |
| 12:23.00 | Un1x|laptop | and doing some WarSitting LOL! |
| 12:23.19 | ambriento | what delivery? |
| 12:23.30 | Un1x|laptop | WarSitting = Abbreviation for when you're sitting and loggin/hacking wireless networks rather then driving :P |
| 12:23.46 | Un1x|laptop | My Ubiquiti SRC Card with a 60inch cable.. |
| 12:23.47 | Un1x|laptop | :( |
| 12:23.56 | Un1x|laptop | I'm well waiting on it to come my way :) |
| 12:23.58 | Un1x|laptop | hehehehe |
| 12:24.03 | Un1x|laptop | I cant wait :) |
| 12:24.06 | ambriento | hahah |
| 12:24.12 | laichzeit | how do I enable priniting of current context in asterisk CLI? |
| 12:24.25 | Un1x|laptop | you know aht i really want a WRT54G at this very moment run openWRT or Dd-WRT on it :D |
| 12:24.38 | Un1x|laptop | and then be able to do stuff i couldn't with crap linux firmware one it :) |
| 12:24.52 | Un1x|laptop | i was up for a entire night reading about themm and mods people have done to them |
| 12:25.02 | Un1x|laptop | i"m probably going to do a custom mode tho :D |
| 12:25.33 | Un1x|laptop | Frig i hope that Stupid UPS guy gets here soon or i'm going to die sitting here eagerly.. |
| 12:25.40 | stoffell_h | Un1x|laptop: the wrt54gl is cool, it has a controllable LED also :) |
| 12:25.51 | Un1x|laptop | laichzeit why not just rea the file? |
| 12:26.00 | Un1x|laptop | stoffell_h what controllable LED? |
| 12:26.09 | Un1x|laptop | whats it do By the way? |
| 12:26.41 | Un1x|laptop | i cant beleive bell is this fast, heh, ive got like 5 unsecred wireless networks just around my house |
| 12:26.51 | Un1x|laptop | this is with just a plain Factory Wnic as well... |
| 12:26.59 | Un1x|laptop | those ones onboard |
| 12:27.28 | Un1x|laptop | i cant imagine what i will get with a Ubiquiti SRC with a High Gain Antenna :D |
| 12:28.24 | stoffell_h | Un1x|laptop: it's a led/button on the wrt (normally used to configure it..), you can switch it off/on, fun to use |
| 12:28.54 | Un1x|laptop | you must havve fun with that :P |
| 12:29.05 | Muck- | after a reboot diva4linux doesn't start because of "kernel version, missing files" but if i do in /usr/lib/eicon/diva the ./Config command first, i can ./Start afterwards and capiinfo works |
| 12:29.11 | Muck- | what a fuck |
| 12:29.27 | stoffell_h | Un1x|laptop: it's nice to use for enabling/disabling specific things (in our situation: control the VPN and display status via led colouring..) |
| 12:29.54 | Un1x|laptop | well i was looking at this LCD mod wich i want to do just i cant find the Guide for it |
| 12:30.07 | Un1x|laptop | i got a RV08 Linksys router that 8 port Dual wan one :) |
| 12:30.12 | Un1x|laptop | i wanna mod that once i get it :) |
| 12:30.17 | Un1x|laptop | if i can that is :) |
| 12:31.16 | Un1x|laptop | stoffell_h im going to talk to you from my other computer i cant sit out here for few hours wiating for gay UPS |
| 12:33.13 | *** part/#asterisk HaMYaI (n=hamyai@ppp-58.8.9.82.revip2.asianet.co.th) |
| 12:33.50 | Un1x | pl |
| 12:33.54 | Un1x | Ok i'm back :) |
| 12:38.29 | stoffell_h | lol |
| 12:41.22 | *** join/#asterisk tparcina (n=tparcina@lns01-0149.dsl.iskon.hr) |
| 12:41.28 | tparcina | hi channel |
| 12:41.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.96) |
| 12:41.42 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.190) |
| 12:41.49 | tparcina | i need to delete first for digits from caller id numeber, how do i do that? |
| 12:42.04 | tparcina | Set(CALLERID(number)=13473${CALLERID(number)}) |
| 12:42.37 | stoffell_h | tparcina: there's a strip function for that, can't remember the name, but it's on voip-info.. |
| 12:42.53 | Strom_C | Set(CALLERID(num)=${CALLERID(num):4}) |
| 12:43.30 | tparcina | yes, i know there is on woip info, and I have been searching for an hour and I couldn't find it. and i'm sure it's there, i have seen it several times... :)) |
| 12:43.33 | tparcina | thank you strom! |
| 12:46.07 | tparcina | Strom_C: do you have link to that page on voip-info, because I allways have problem to find that page |
| 12:46.23 | Strom_C | what page on voip-info? |
| 12:46.30 | Strom_C | i didn't have to look that one up |
| 12:46.53 | tparcina | where they have explained that with extensions |
| 12:48.03 | Un1x | Wird i thought my laptop turned off seems it hasn't |
| 12:48.57 | *** part/#asterisk tparcina (n=tparcina@lns01-0149.dsl.iskon.hr) |
| 12:51.58 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 12:53.22 | *** join/#asterisk pablus (n=nn@test.conama.cl) |
| 12:53.24 | pablus | morning |
| 12:54.01 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 12:54.08 | Muck- | why could it be, that capiinfo shows my eicon card but if i start asterisk i get " chan_capi.c: CAPI not installed, CAPI disabled!" |
| 12:54.27 | Muck- | has somebody any clou? |
| 12:56.18 | *** join/#asterisk Bambr (n=Bambr@213-35-238-90-dsl.end.estpak.ee) |
| 12:59.47 | BeBoo | I was thinking of linking two asterisk servers together. Is it somehow possible to make them announce all the phone numbers the server handles in a specific context, so i dont have to configure every IAX user? |
| 13:00.26 | Strom_C | yes |
| 13:00.30 | Strom_C | it's called DUNDi |
| 13:02.13 | BeBoo | okay thanks |
| 13:02.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:03.11 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
| 13:04.55 | *** join/#asterisk Nix (n=Nix@195.175.21.46) |
| 13:06.00 | benjk | there's also the switch=> statement in the dialplan |
| 13:06.16 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 13:08.25 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 13:11.21 | *** part/#asterisk Nix (n=Nix@195.175.21.46) |
| 13:12.46 | *** join/#asterisk Synoptic (i=Synoptic@modemcable236.180-203-24.mc.videotron.ca) |
| 13:12.52 | Synoptic | hi there |
| 13:12.56 | aixa | hi |
| 13:13.03 | Synoptic | ok i'm having a little issue |
| 13:13.18 | Synoptic | I have 2 trunks cnnectec to two othe asterisk, wich worked fine for days |
| 13:13.24 | Synoptic | IAX2 trunks |
| 13:13.43 | aixa | okay, and what happened? |
| 13:13.51 | Strom_C | they started baking cakes instead |
| 13:13.53 | Strom_C | that's what happened |
| 13:14.12 | Synoptic | now, for unkown reason, I can call the trunlks, but cant receive calls, when calling my server it rings and after maybe 5 - 6 rings, it says al lcircuits are busy. I,m trying to fnd an error log that could be usefull to look at, but dont see anything really wrong. |
| 13:14.18 | benjk | no, it's Friday evening, they went to the pub |
| 13:14.25 | Synoptic | lol |
| 13:14.46 | Strom_C | Synoptic: did you change /anything/? |
| 13:15.13 | Synoptic | Strom_C: not that I can really remeber, no updates, no extension creation, no routing shit... |
| 13:15.25 | Synoptic | it was working on wednesday btw |
| 13:15.28 | Synoptic | or tuesday |
| 13:15.31 | Synoptic | anyway |
| 13:15.38 | benjk | yeah, but wednesdy isn't pub day |
| 13:15.59 | aixa | okay how - "unchangable without your knowladge" is the env. between those servers? |
| 13:16.18 | Synoptic | pretty much aware of |
| 13:16.32 | Synoptic | we're 3 friends in pre-prod mode |
| 13:16.54 | Synoptic | so we know it's not time to mess around. when it works, we stop playing. |
| 13:16.56 | benjk | did you try iax debug? |
| 13:17.00 | aixa | if you manually send udp packet to that port and server does it arrive there? |
| 13:17.06 | Synoptic | oh |
| 13:17.08 | Synoptic | i could try |
| 13:17.10 | Synoptic | tell me how |
| 13:17.16 | benjk | on the cli |
| 13:17.23 | benjk | iax2 debug |
| 13:17.47 | aixa | and launch network sniffers at both ends |
| 13:18.03 | Synoptic | hmm |
| 13:18.04 | aixa | could be some weired nat issue |
| 13:18.05 | Synoptic | ok |
| 13:18.12 | aixa | if you have any |
| 13:18.14 | Muck- | hmm do i need /dev/capi or /dev/capi20? because i don't have them and chan_capi can't find my capi of which capiinfo says that it works... |
| 13:18.19 | Synoptic | maybe, i'm behind a NAT, but i'm ion the DMZ. |
| 13:19.12 | aixa | voip-info had some links to debug/test tools for iax2 |
| 13:19.43 | Synoptic | yeah |
| 13:20.06 | aixa | http://www.voip-info.org/wiki/view/IAX2+Analysis+using+Unsniff+Network+Analyzer |
| 13:20.23 | aixa | http://www.voip-info.org/wiki/view/IAX2+ping+tool |
| 13:20.37 | *** join/#asterisk mover (n=dlu@83.125.8.7) |
| 13:21.05 | *** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg) |
| 13:21.27 | Synoptic | thx |
| 13:22.18 | Synoptic | oh |
| 13:22.28 | Synoptic | the client is only ssh accessible by me |
| 13:24.02 | Synoptic | ok, the perl script doesnt work |
| 13:24.05 | Synoptic | let me have a look |
| 13:24.57 | Synoptic | ok |
| 13:25.17 | Synoptic | the tool is working, just that it need an ip address. |
| 13:25.19 | Synoptic | let me run it |
| 13:25.46 | Synoptic | hmm |
| 13:25.47 | Synoptic | timeout |
| 13:25.56 | Synoptic | ok |
| 13:25.57 | Synoptic | FW |
| 13:25.58 | Synoptic | issue |
| 13:26.02 | Synoptic | will reboot the router |
| 13:26.33 | aixa | by the way - as regards controlable env. in which astyerisk resides |
| 13:26.59 | aixa | I once supported one IVR which had E1 attached to it |
| 13:27.07 | aixa | worked fine for a couple of months |
| 13:27.38 | aixa | then - blah out of the blue it is not responding one day |
| 13:27.50 | Synoptic | k |
| 13:27.52 | Synoptic | hey |
| 13:27.52 | Synoptic | brbr |
| 13:27.56 | Synoptic | will reset the router |
| 13:28.29 | aixa | I and the telco at the other side of E1 spent 2 days trying to understand the issue |
| 13:28.33 | aixa | no good |
| 13:28.40 | suma | what is the best way to get g729 license ? |
| 13:28.51 | benjk | www.digium.com |
| 13:29.01 | suma | it is $10 per channel |
| 13:29.02 | benjk | in fact its the only way |
| 13:29.09 | suma | is there anywhere i can get cheaper ? |
| 13:29.17 | benjk | are you kidding? |
| 13:29.23 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.164) |
| 13:29.27 | suma | nope, really asking |
| 13:29.30 | *** join/#asterisk Synoptic (i=Synoptic@modemcable236.180-203-24.mc.videotron.ca) |
| 13:29.37 | benjk | its usually 50.000 USD |
| 13:29.41 | Synoptic | ok |
| 13:29.45 | Synoptic | lets see if it worked |
| 13:29.48 | suma | i got an email saying i can get license for $5 per channel |
| 13:29.51 | HaMYaI | I 'm having this error "ZT_CHANCONFIG failed on channel 25: No such device or address (6)" |
| 13:29.52 | aixa | at the very end it turned out that the line provider (the one who connected us with telco) had changed the hw and the new hw started to to print its own shit on bchan |
| 13:30.03 | HaMYaI | anyone knows what causes this? |
| 13:30.11 | benjk | its 50.000 USD for 50.000 devices or 50.000 channels |
| 13:30.12 | HaMYaI | on a TE110P |
| 13:30.13 | benjk | prepaid |
| 13:30.26 | suma | from digium ?? |
| 13:30.26 | benjk | so you are getting a good deal with $10 per channel |
| 13:30.34 | HaMYaI | and unconnected |
| 13:30.36 | benjk | no, from the G729 consortium |
| 13:30.43 | Synoptic | nope |
| 13:30.50 | benjk | Digium paid their 50.000 USD to them |
| 13:30.50 | suma | $50 usd for each device or whole ? |
| 13:30.53 | Synoptic | packets cant get here. |
| 13:31.24 | benjk | its 50.000 USD for the smallest license and that includes 50.000 devices or 50.000 channels |
| 13:31.30 | benjk | fifty thousand |
| 13:31.52 | benjk | so if you can have it for 10 USD for a single channel, thats' not too bad, is it?! |
| 13:31.59 | aixa | we proved to line provider that their hw was f&*&^&^ up only by making loopback with e1 wire |
| 13:32.17 | suma | I got email saying $5 per channel ! |
| 13:32.23 | suma | acceptable ? |
| 13:32.41 | Strom_C | suma: email from who? |
| 13:32.48 | benjk | with Asterisk the only company which can sell you g729 legally is Digium |
| 13:32.52 | suma | from the internet |
| 13:32.54 | suma | i c |
| 13:32.58 | *** join/#asterisk Synoptic (i=Synoptic@modemcable236.180-203-24.mc.videotron.ca) |
| 13:33.00 | Synoptic | AH ! |
| 13:33.00 | aixa | suma - I get emails every day saying $5 for adobe photoshop |
| 13:33.02 | Synoptic | found it |
| 13:33.02 | Strom_C | the entire internet sent you an email? |
| 13:33.13 | benjk | you probably got spam |
| 13:33.16 | suma | but it worked for $5 |
| 13:33.28 | benjk | its probably a scam |
| 13:33.28 | suma | i got g729 working ! |
| 13:33.28 | Synoptic | I dont know WHY but in my Dlink router, the GAMING mode was enabled. and I know gaming mode causes issue |
| 13:33.39 | benjk | if its not from Digium its not legal |
| 13:33.50 | CSerpent | DLink? <spit> |
| 13:33.51 | suma | i just want to test it |
| 13:33.52 | benjk | because nobody else can grant an exception |
| 13:34.26 | Synoptic | CSerpent : yeah, no comments on this. but I have a WRT54G at work ;P |
| 13:34.38 | Synoptic | will swap the routers i think |
| 13:34.48 | benjk | unless the software is somehow decoupled from Asterisk, like through a socket interface or somthing |
| 13:34.49 | mover | hi all |
| 13:34.53 | *** join/#asterisk trelane` (n=trelane@unaffiliated/trelane) |
| 13:34.53 | benjk | unlikely though |
| 13:34.53 | aixa | Synoptic - so, basicly issue solved? |
| 13:35.00 | mover | i have a question about invites |
| 13:35.27 | CSerpent | Dlink sent out ADSL modems a while ago, and they wouldn't take the full login string required in the UK, so they say "upgrade the firmware".. if you upgrade the firmware it breaks the modem :D great! |
| 13:35.52 | aixa | CSerpent - :D |
| 13:35.55 | mover | if i call a registered sip client in the invites TO header are user@ip adress it is correct? |
| 13:36.11 | mover | it seems there must be username@realm or i am wrong? |
| 13:36.30 | RoyK | http://isitfriday.net/ |
| 13:36.48 | benjk | yeah, time for Guinness, RoyK |
| 13:37.00 | RoyK | yeah |
| 13:37.17 | Synoptic | aixa : yes , issue resolved, i had that feeling it was something about the firewall, i'm just lucky to have RTFM about my dlink's gaming mode feature, even though the asterisk is in the DMZ.. |
| 13:37.50 | benjk | gaming mode? asterisk? |
| 13:37.57 | Synoptic | benjk : no, |
| 13:38.00 | benjk | oh well, come to think of it |
| 13:38.07 | RoyK | yesterday was a day of Karoshi |
| 13:38.11 | RoyK | or could have been |
| 13:38.15 | benjk | its a geek's game |
| 13:38.17 | aixa | asterisk -gamin |
| 13:38.29 | benjk | Karoshi? whazzat? |
| 13:38.35 | aixa | throws random exceptions, swaps extensions |
| 13:38.46 | Synoptic | benjk : in my dlink router, there is an option I can check, called Gaming mode. Inthis mode, the router process packet diferently to help the ping time in multiplayer games. I do not play games, and for unkown reason, this mode was activated. |
| 13:38.51 | RoyK | http://en.wikipedia.org/wiki/Karoshi |
| 13:39.04 | BeBoo | Hmm, anyone tried x-lite together with asterisk? Isn't it possible to get dtmf working within an active call? |
| 13:39.10 | benjk | the ast-gremlins did it |
| 13:39.11 | *** join/#asterisk ComputerWarm (n=dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 13:39.15 | RoyK | BeBoo: works for me (tm) |
| 13:39.19 | BeBoo | hmm |
| 13:39.24 | aixa | BeBoo - i had dtmf working on xlite but that was a while ago |
| 13:39.25 | ComputerWarm | Hello all anyone know where i can get a 18XX number for canada? |
| 13:39.31 | aixa | like 1/2 a year or so |
| 13:39.54 | benjk | ahh |
| 13:39.55 | benjk | 過労死, |
| 13:39.58 | aixa | benjk - swaped extensions |
| 13:40.00 | aixa | ? |
| 13:40.11 | nettie | hey guys anyone have any call forwarding macro to show me please? all the examples I was able to find are for older * versions and not longer supported. thanx |
| 13:40.29 | mover | i guess its a bug |
| 13:40.38 | nettie | my concerns are moslty regardign the db/setdb/getdb stuff which has been deprecated |
| 13:40.41 | BeBoo | Hmm, cannot get it to work here at all - tried auto/rfc/inband nothing works for me |
| 13:40.42 | *** join/#asterisk Synoptic (i=Synoptic@modemcable236.180-203-24.mc.videotron.ca) |
| 13:40.50 | Synoptic | what was my quit message? |
| 13:41.07 | mover | Signoff: Synoptic (Read error: 104 (Connection reset by peer)) |
| 13:41.07 | CSerpent | aixa: 800/888/877/866 should generally work across the US and Canada - if not then try googling "toll-free canada IAX" |
| 13:41.13 | Synoptic | weird |
| 13:41.27 | benjk | 866 won't work in CA if its a US 866 number |
| 13:41.41 | CSerpent | they becoming selective now? eesh |
| 13:41.56 | benjk | no, its always been national |
| 13:41.58 | ComputerWarm | benjk thats not totally true there is alot of 1866 numbers i call and i am a half a hour from Toronto |
| 13:42.11 | Synoptic | hey |
| 13:42.12 | Synoptic | weird |
| 13:42.14 | benjk | you have to pay extra for international 8xx numbers |
| 13:42.16 | Synoptic | on conference |
| 13:42.27 | Synoptic | if I choose channel 1 for all people it doesnt work |
| 13:42.28 | benjk | there are some mapping codes, but 866 isn't covered by those yet |
| 13:42.30 | Synoptic | is that normal ? |
| 13:42.52 | ComputerWarm | benjk true. :-) i want to get a 800 or something thats not going to cost me 4 cents a minue do you know of any? |
| 13:43.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:43.19 | HaMYaI | can anyone confirm how I change from T1 to E1 mode in TE110P? |
| 13:43.34 | brookshire | strap off :) |
| 13:43.39 | aixa | HaMYaI - didnt it have a jumper? |
| 13:43.39 | benjk | Did you try NuFone? should be 2 cents and you get it via IAX |
| 13:43.57 | HaMYaI | aixa: yes there is |
| 13:43.59 | ComputerWarm | benjk i had one of theirs a few years ago didn`t work for canada |
| 13:44.04 | HaMYaI | aixa: I'm just unsure |
| 13:44.05 | RoyK | HaMYaI: load asdf.ko t1e1override=15 |
| 13:44.10 | ComputerWarm | trying to reach jerjer to see if they do now but no replies |
| 13:44.10 | benjk | ah ok |
| 13:44.20 | HaMYaI | RoyK: oh, ok |
| 13:44.28 | brookshire | http://www.digium.com/en/docs/TE110P/te110p_config.php |
| 13:44.34 | HaMYaI | RoyK: is that a software solution? |
| 13:44.35 | RoyK | HaMYaI: i've had boxes installed with the strappings wrong as well :) |
| 13:44.38 | RoyK | yes |
| 13:44.43 | brookshire | sorry.. jumper on |
| 13:45.00 | HaMYaI | RoyK: ok, thanks |
| 13:45.15 | aixa | i would rather prefer jumper |
| 13:45.15 | RoyK | t1e1override = 15 == binary 1111 == e1 e1 e1 e1 |
| 13:45.18 | aixa | but nevermind |
| 13:45.25 | RoyK | HaMYaI: with one port, =1 will prolly do |
| 13:46.14 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 13:46.25 | HaMYaI | RoyK: ok, I have just one |
| 13:47.00 | HaMYaI | aixa: =) I was gonna open the box too |
| 13:47.07 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 13:47.17 | aixa | any news on spandsp ? |
| 13:47.18 | *** join/#asterisk scage (n=xzen@202.63.226.41) |
| 13:47.36 | aixa | i understood today they were going to fix txfax problem |
| 13:47.59 | HaMYaI | aixa: I talked to coppice two days ago , there was still problems |
| 13:48.13 | aixa | HaMYaI, okay |
| 13:48.32 | aixa | crushed production server yesterday night :PPP |
| 13:48.44 | HaMYaI | aixa: he said a lot of people reported a success with 1.2.9 tho |
| 13:49.01 | aixa | i can send *some* files |
| 13:49.22 | aixa | but for example provided test tiff brought the system down |
| 13:49.26 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 13:49.46 | RoyK | aixa: use openpbx. steve has spandsp 0.0.3 with lots of nice stuff for opbx, but not for asterisk because of licensing issues |
| 13:50.42 | aixa | i dont fancy the idea of using software which isnt even in beta version |
| 13:50.44 | benjk | not really licensing issues, but the Digium repository gods have pissed him off |
| 13:51.03 | aixa | for production env. |
| 13:51.05 | benjk | asterisk has been pre beta for ages |
| 13:51.10 | benjk | still is |
| 13:51.14 | benjk | all perception |
| 13:51.18 | aixa | ok |
| 13:51.26 | benjk | you give it a fancy number and people get excited |
| 13:51.32 | aixa | :))) |
| 13:51.37 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 13:51.44 | aixa | microsoft have been doing this to DOS for ages |
| 13:51.51 | CSerpent | of course we do, we're techies :) |
| 13:51.59 | aixa | WinXP |
| 13:52.35 | aixa | benjk I was planing to give openpbx a try |
| 13:53.21 | aixa | but first I have to finish this one asterisk project and get the company up and running |
| 13:53.37 | aixa | then I can start to experiment |
| 13:55.27 | aixa | hmm this looks mutch like asterisk`s dialplan: http://wiki.openpbx.org/tiki-index.php?page=OpenPBX.org+tricks |
| 13:55.39 | brookshire | it is asterisk |
| 13:55.50 | brookshire | openpbx is a fork of asterisk |
| 13:56.58 | mover | how i can configure asterisk to invite a peer so the to header contain user@realm instead of user@ip? |
| 13:58.12 | aixa | brookshire - i knew it is a for, yet (fromIdontKnowWhere) I had this idea that it has been redone rather different |
| 13:58.37 | aixa | guess from voip-info page |
| 13:58.55 | benjk | aixa, the dialplan and the configs are going to change |
| 13:59.28 | benjk | right now the changes are still mostly internal |
| 14:00.11 | aixa | benjk - anywhere where I could read about that? |
| 14:00.20 | aixa | about planned changes I mean |
| 14:00.33 | benjk | http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk/opbx_plist_syntax.txt |
| 14:00.43 | brookshire | aixa: debian also has a project of packaging asterisk and spandsp for asterisk among other object |
| 14:01.00 | brookshire | s/object/applications |
| 14:01.43 | bkw_ | you can't legally package asterisk with spandsp |
| 14:01.43 | bkw_ | tiz a GPL violate |
| 14:01.43 | aixa | brookshire, i am slackware guy, so no debian |
| 14:01.43 | bkw_ | er violation |
| 14:02.03 | brookshire | bkw: they are separate |
| 14:02.15 | benjk | no, that's not so bkw |
| 14:02.22 | benjk | Digium could, but they don't want to |
| 14:02.40 | brookshire | http://packages.debian.org/stable/source/asterisk-spandsp-plugins |
| 14:02.41 | benjk | because then their GPL version would have something their commercial non-GPL cant have |
| 14:03.16 | benjk | and they don't want to pay for a license to be able to bundle Unicall/SpanDSP with their commercial version either |
| 14:03.32 | benjk | they've got enough cash, they could easily do it |
| 14:03.37 | benjk | strategic investment |
| 14:03.45 | benjk | they just don't want to |
| 14:03.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:04.02 | brookshire | benjk: i don't think that's true |
| 14:04.05 | blitzrage | I am the troll you are looking for! |
| 14:04.26 | benjk | what is not true? |
| 14:04.38 | *** join/#asterisk SomethingIsODD (n=dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 14:04.41 | benjk | Unicall/SpanDSP is GPL licensed, that's ture |
| 14:05.01 | benjk | thus, the GPL version of asterisk could easily bundle it |
| 14:05.08 | benjk | just not the non-GPL version |
| 14:05.11 | brookshire | it is true that Steve Underwood is a GPL nazie |
| 14:05.25 | benjk | well, it always cuts both ways |
| 14:05.45 | benjk | you might as well say that "its true Mark Spencer is a dual licensing nazie" |
| 14:05.50 | benjk | so? |
| 14:05.58 | brookshire | so it has nothing to do with money |
| 14:06.05 | benjk | of course it has |
| 14:06.12 | blitzrage | nazi has no 'e' |
| 14:06.17 | *** join/#asterisk ToyMan (n=stuq@pool-70-19-239-233.bos.east.verizon.net) |
| 14:06.26 | benjk | talk to steve and you'll find out |
| 14:06.39 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:06.53 | brookshire | blitzrage: grammar nazi :) |
| 14:07.01 | benjk | :) |
| 14:07.14 | benjk | I guess that would rather be spelling nazi |
| 14:08.19 | blitzrage | brookshire: it's true! |
| 14:08.22 | benjk | in any event, if you don't make any offer to somebody who says he isnt going to sign over his house for a wet handshake, you won't find out if you can buy the house |
| 14:08.33 | blitzrage | JunK-Y: question for you |
| 14:09.30 | blitzrage | or I guess anyone else who can answer it... :) |
| 14:09.37 | blitzrage | is there a way I can tell Dial() to not continue trying to setup a connection if there is no response in X number of seconds, but keep the ringing timeout a separate value? |
| 14:11.41 | file | benjk: I have to ask this... why have you been trolling on Asterisk and Digium so much lately? it seems every day there's something... and since you have a creative outlet in OpenPBX.org for your ideas, why continue? |
| 14:11.43 | aixa | ok, bye everyone and have a nice weekend :) |
| 14:11.45 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 14:11.49 | aixa | oh no |
| 14:11.54 | aixa | not yet :))) |
| 14:12.04 | blitzrage | coppice: morning |
| 14:12.11 | aixa | coppice morning |
| 14:12.22 | coppice | evening |
| 14:12.39 | JunK-Y | blitzrage: not that i know, ringing timeout is the connection timeout. |
| 14:12.39 | aixa | i understand ur the person to some extent responsible for spandsp? |
| 14:12.56 | aixa | or in knowledge of dev. |
| 14:13.25 | benjk | file I am not trolling, thats all I can say |
| 14:13.48 | benjk | but perhaps I have been trolling when I told some guy that he should get his g729 licenses from Digium |
| 14:13.50 | benjk | dunno |
| 14:14.05 | benjk | you pick only the stuff that you don't like and you call it trolling |
| 14:14.13 | benjk | when I say something that doesn |
| 14:14.22 | benjk | t sound uncomfortable to you you overlook that |
| 14:14.38 | blitzrage | JunK-Y: yah... issue I'm having is that I have several softswitches which I run my traffic through. If one of them is down, then the connection just hangs for a pretty long time before it times out, then it rolls over to the next connection (based on my dialplan). However, if I set the ring timeout low, then of course if it does connect, then it'll only ring for a very short period of time. The only thing I've been abl |
| 14:15.03 | blitzrage | doesn't feel -- is ;) |
| 14:15.40 | suma | how much is benjk paid every month by digium ? |
| 14:15.47 | JunK-Y | dial ur 2 switches at once, if u want it transparent for the end-user? |
| 14:15.52 | SomethingIsODD | i hate to ask again does anyone know any cheap place to get 1800 numbers that will work in canada and usa? |
| 14:16.11 | JunK-Y | SomethingIsODD: telus? |
| 14:16.23 | JunK-Y | :P |
| 14:16.26 | *** part/#asterisk scage (n=xzen@202.63.226.41) |
| 14:16.28 | SomethingIsODD | JunK-Y :-p that works on sip |
| 14:16.34 | blitzrage | JunK-Y: hrmmm... actually... the might be the simplest solution... I wonder if it "answers" the call... wouldn't want it to try calling the device 4-5 times at once :) |
| 14:16.34 | aixa | blitzrage perhaps you could do some cron |
| 14:16.37 | aixa | regular checks |
| 14:16.43 | blitzrage | aixa: cron? I don't see how that would help |
| 14:16.55 | aixa | okay - you do regular checks |
| 14:17.03 | blitzrage | checking for what? |
| 14:17.11 | JunK-Y | SomethingIsODD: i think asterlink.com can provides ya stuff in both countries. |
| 14:17.21 | SomethingIsODD | JunK-Y ok thanks |
| 14:17.25 | aixa | blitzrage connect to ami and issue dial |
| 14:17.34 | file | benjk: okay, let's try this... why do you continue to make negative comments about Asterisk / Digium - be it programming/licensing when you can go elsewhere, like to OpenPBX.org where that doesn't matter? |
| 14:17.54 | aixa | to test number on the switch |
| 14:17.55 | JunK-Y | blitzrage: why not dial the 1st one, with LOW timeout, and if it fails, dial the oher one? |
| 14:18.07 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:18.11 | aixa | then if you cant access this you could alter dialplan |
| 14:18.12 | blitzrage | JunK-Y: but if it connects, then the call will only ring for that short period of time... |
| 14:18.26 | benjk | this is probably news to you but there is freedom of expression and all that |
| 14:18.29 | coppice | benjk: being at Digium seems to have a strange effect on people. seems the only one there not deeply paranoid is Kevin |
| 14:18.40 | blitzrage | JunK-Y: let me try Dial(SIP/1&SIP/2...) and see if I get multiple calls :) |
| 14:18.56 | suma | how much is benjk paid every month by digium ? |
| 14:19.02 | blitzrage | who cares |
| 14:19.07 | l-fy | suma > for what? |
| 14:19.12 | l-fy | benjk works for openpbx |
| 14:19.18 | JunK-Y | blitzrage: not sure to understand, u get some ringing, stopings, then some ringing(on the other switch) again? |
| 14:19.22 | suma | I-fy: for marketing digum products and asterisk here |
| 14:19.24 | benjk | in fact I did respond to some comment that SpanDSP wasnt legal to be distributed with Asterisk |
| 14:19.38 | l-fy | sumasuma > who cares, still sangoma is better |
| 14:19.42 | blitzrage | JunK-Y: well, the switch is going to try and setup a call to a phone, I'm not just calling the switch directly (its routing the call) |
| 14:19.55 | coppice | you can distribute spandsp with *, as long as you are distributing * under GPL terms |
| 14:19.57 | benjk | I set the record straight that there is no such legal impediment, which in fact is a positive thing for asterisk users |
| 14:20.16 | benjk | at least for those asterisk users who use the GPL version |
| 14:20.31 | l-fy | coppice > still learning c? |
| 14:20.32 | aixa | and i was just wanting to know what caused the crash of asterisk after executionof txfax |
| 14:20.33 | suma | I-fy: still sangoma is costlier ! |
| 14:20.43 | aixa | so its me to blame for starting this conv. |
| 14:21.30 | l-fy | suma > what do you mean? |
| 14:21.49 | blitzrage | suma: I don't remember us talking about hardware |
| 14:22.41 | suma | blitzrage: you can refer to the chat before, I-fy mentioning about the Sangoma |
| 14:22.42 | RoyK | I keep hearing digium currently makes lots of money... does anyone know how much of this is license money? |
| 14:23.06 | JunK-Y | RoyK: in what thats important. |
| 14:23.15 | suma | RoyK: may be, you want some from them ? |
| 14:23.27 | RoyK | not really |
| 14:23.32 | blitzrage | suma: maybe I meant to reply to him then :) |
| 14:23.36 | RoyK | i just wonder why they keep that stupid licensing |
| 14:23.47 | JunK-Y | contact sangoma and ask them how much money they make on a specific product, im not sure they will answer ur question. |
| 14:23.52 | JunK-Y | like any other companies. |
| 14:23.54 | blitzrage | RoyK: only people at digium would know that, and that would be priviledged information |
| 14:24.04 | RoyK | exactly |
| 14:24.11 | JunK-Y | so why are ya asking? |
| 14:24.12 | benjk | for the same reason why files that are entirely commented in Espanol are headed "written by Mark Spencer" I presume |
| 14:24.13 | blitzrage | then why the hell are you asking? |
| 14:24.14 | JunK-Y | :) |
| 14:24.17 | blitzrage | heh |
| 14:24.18 | RoyK | I really really dislike that disclaiming shite |
| 14:24.21 | RoyK | ~disclaimer |
| 14:24.24 | jbot | I disclaim all of you!, or "fortune -m 'Void where'" |
| 14:24.25 | blitzrage | RoyK: so? |
| 14:24.32 | JunK-Y | blitzrage: french canadian are still faster huh! |
| 14:24.38 | blitzrage | JunK-Y: in bed! |
| 14:24.54 | JunK-Y | ya, i know ur gf always saying that to me. |
| 14:25.34 | *** join/#asterisk kernelbee (n=Naveed@202.63.226.41) |
| 14:26.15 | aixa | okay, bye everyone and have a nice weekend |
| 14:26.37 | benjk | see ya |
| 14:26.48 | kernelbee | bye |
| 14:27.49 | blitzrage | JunK-Y: I have no g/f... just yours :) |
| 14:28.14 | JunK-Y | shes obssed by twisted, not by ya, sorry! |
| 14:28.37 | file | JunK-Y: oh baby you're hurting me |
| 14:28.49 | JunK-Y | exactly! |
| 14:29.42 | JunK-Y | file: i just realized i was listening michelle branch! |
| 14:29.56 | file | ha |
| 14:29.57 | blitzrage | JunK-Y: she likes twisted more than me? hard to believe :) |
| 14:30.02 | coppice | aixa: I don't think you actually asked any questions |
| 14:30.19 | RoyK | hi coppice |
| 14:30.23 | JunK-Y | blitzrage: she likes u too, dont worry! |
| 14:30.27 | blitzrage | lol |
| 14:30.27 | coppice | hi |
| 14:30.32 | RoyK | coppice: do you know if there'll be any 0.0.3 interface working with asterisk? |
| 14:31.02 | suma | 0.0.3 ?? |
| 14:31.14 | RoyK | spandsp 0.0.3 |
| 14:31.15 | suma | is 9p protocol supported by asterisk ?/ |
| 14:31.27 | coppice | I intend to update app_rxfax and app_txfax this weekend to work with * and 0.0.3 |
| 14:31.34 | RoyK | nice |
| 14:31.39 | RoyK | what's new in 0.0.3 anyway? |
| 14:32.20 | coppice | T.38 and ECM are the big things |
| 14:32.46 | blitzrage | tasty |
| 14:33.20 | coppice | you won't get the full T.38 stuff with * just yet, though |
| 14:33.44 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:34.16 | brookshire | JunK-Y: cantaloupe |
| 14:34.39 | JunK-Y | hey matt! |
| 14:34.45 | JunK-Y | whats up? |
| 14:34.54 | brookshire | in training class :/ |
| 14:35.10 | blitzrage | coppice: pass through or termination? |
| 14:35.28 | coppice | you don't need a thing like spandsp for passthrough |
| 14:35.55 | RoyK | what's the t.38 stuff you won't get for asterisk? |
| 14:35.57 | JunK-Y | what kind of training class? |
| 14:37.43 | coppice | termination should work with * head, apart from the issues the * code has with the rest of the world not following specs. t.38 gateway works, but only with a kind of fudged interface to the system. a proper means of integration is still needed. something * doesn;t really have a framework for, but matt said he is working on |
| 14:38.21 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 14:43.25 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 14:49.17 | spr1te | hello, do youw know softswitch on similarity MVTS... |
| 14:49.21 | spr1te | for tests=) |
| 14:49.39 | *** join/#asterisk hellor (n=50ccb724@c-67-160-249-95.hsd1.ca.comcast.net) |
| 14:49.48 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:50.37 | hellor | anyone out there? |
| 14:51.16 | hellor | i need some help on nway calling |
| 14:51.18 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 14:51.23 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 14:52.09 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:52.13 | spr1te | hello!! i need some help=) do you now some free billing (softswitch) for voip.. for tests.. |
| 14:52.28 | hellor | okay.. so this channel is alive |
| 14:52.32 | hellor | whew !! |
| 14:52.44 | hellor | could someone respond to this pls.... |
| 14:53.19 | benjk | you should just ask your question, and then it should be a detailed question |
| 14:53.31 | hellor | okay....sure... |
| 14:53.49 | benjk | not as much as to flood the channel though ;) |
| 14:54.50 | hellor | I want to setup a reverse conference calls. meaning instead of folks calling into the bridge, if there was a way for the server to call set conference parties at a set time then there is no reason for outlook reminder and remembering the pin/conf# etc |
| 14:55.14 | mover | exist a solution to rewrite a sip header on asterisk? |
| 14:55.35 | hellor | Going beyond, if there is a way for the receiving party to accept/deny a conference call invite, that would help |
| 14:56.29 | hellor | any ideas if such a thing exist, can be done? |
| 14:57.47 | benjk | check out call files which you can drop into /var/spool/asterisk to initiate calls with given destinations at a given time |
| 14:58.06 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 14:58.38 | hellor | i see.. once done with the call, i could delete this file and done.. |
| 15:02.13 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 15:03.25 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:04.12 | flujan | hi guys... :) how are you doing? |
| 15:04.34 | *** join/#asterisk CSerpent (i=daemon@62.49.253.91) |
| 15:04.53 | benjk | you won't need to delete the spool files |
| 15:05.44 | flujan | benjk, ? how can I achieve this behavior? |
| 15:06.00 | benjk | read up on asterisk call files |
| 15:06.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:06.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:06.10 | benjk | you can write a scrip that creates them |
| 15:06.18 | benjk | then drop them into the spool dir |
| 15:06.34 | benjk | there should be examples on voip-info.org |
| 15:12.17 | *** join/#asterisk Doodluv (n=brad@24.214.206.158) |
| 15:14.08 | SomethingIsODD | geeze i can`t believe the headachs trying to find a canada toll free number over sip or iax |
| 15:14.11 | Muck- | what is the easiest way to test my connection with the CAPI ISDN Card and the BRI? |
| 15:16.00 | *** join/#asterisk cilly (i=pluto@unaffiliated/cilly) |
| 15:17.30 | designdream | anyone in canada? |
| 15:18.31 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 15:18.46 | SomethingIsODD | designdream i am |
| 15:22.16 | designdream | SomethingIsODD: i have an employee in calgary and i wanted to get him an iaxy adaptor.. is there a way i can get him one asap? any store sell them or have them in inventory for quick shipping? |
| 15:22.52 | designdream | i am looking for voip suppliers in calgary.. i would even attempt to figure out how to get him on via SIP if necessary |
| 15:24.21 | *** join/#asterisk daysmen3 (n=primus@host81-154-137-104.range81-154.btcentralplus.com) |
| 15:26.28 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 15:26.32 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 15:29.15 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 15:31.32 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-3-246.nwrknj.east.verizon.net) |
| 15:33.07 | mover | any sip professionals here? why asterisk add a to header with user@1.2.3.4:5060 to intives instead of To: <sip:user@realm> ????? |
| 15:39.36 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:40.37 | kpettit | I have some g729 license and I upgraded asterisk and now the codec seems to be messed up |
| 15:40.51 | kpettit | anytihng special you need to do with g729 license when you upgrade asterisk? |
| 15:40.52 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 15:41.31 | pollohawk | Is there a way to generate a random number in a dial plan? |
| 15:43.24 | Nugget | sure, either with an AGI or the random() application |
| 15:44.30 | pollohawk | Nugget: how would you do it with the random application, it only jumps you to another context |
| 15:44.45 | Nugget | if you don't find that sufficient, write an AGI. |
| 15:45.03 | benjk | use the last one or two digits of TIMESTAMP |
| 15:45.18 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:45.55 | benjk | for most uses that's random enough |
| 15:45.57 | *** part/#asterisk adminguru (n=atze@u6-177.dsl.vianetworks.de) |
| 15:46.01 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:46.20 | pollohawk | benjk: I will give that a try |
| 15:47.32 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:47.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:52.24 | *** join/#asterisk dalekurt (n=DaleKurt@65.183.3.229) |
| 15:54.01 | dalekurt | Hey StromC got a question for you |
| 15:56.06 | *** join/#asterisk adorah (n=Administ@84.94.145.201.cable.012.net.il) |
| 15:56.44 | *** join/#asterisk bugz (n=will@72.54.193.13) |
| 15:57.00 | bugz | asterisk is not sending voicemail to email |
| 15:57.06 | bugz | what could cause this? |
| 15:57.12 | bugz | i have dozens of boxes that do |
| 15:57.16 | bugz | and the configs are basically the same |
| 15:57.24 | mut | O_ |
| 15:57.25 | mut | o |
| 15:57.29 | bugz | it just never tries to send the mail |
| 15:57.43 | bugz | looking at the log, there is nothing invoking any kind of mail command |
| 15:57.49 | bugz | attach=yes |
| 15:57.51 | bugz | is set |
| 15:58.35 | RoyK | bugz: is yesdoattachdonotmesswithmeanddontfuckupnoreallyimeanit=yes set? |
| 15:58.59 | aydiosmio | anyone know how I can get the STDOUT/STDERR from AGI applications? |
| 15:59.15 | aydiosmio | one of my perl scripts is failing and I don't know why |
| 16:01.04 | mut | bugz: tail -f the mail logs |
| 16:01.09 | mut | does it spit out anything? |
| 16:01.18 | bugz | mut: it does not invoke any mail command at all |
| 16:01.22 | bugz | there is nothing in the mail logs |
| 16:01.34 | bugz | nothing, but i can send mail from the cli all day long without any problems |
| 16:01.42 | bugz | i get my alert mail from it at 2 am |
| 16:01.53 | bugz | so its not a mail program issue |
| 16:01.58 | bugz | mailcmd is not set though |
| 16:02.04 | bugz | but its not set on any other box either |
| 16:02.08 | bugz | and they work just fine |
| 16:03.44 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 16:03.51 | mut | well is your sendmail file in the default place? |
| 16:04.08 | mut | i dunno what it defaults to |
| 16:13.39 | *** join/#asterisk DFiber (n=bwarner@65.113.208.30) |
| 16:15.50 | bugz | im not using sendmail |
| 16:16.06 | bugz | im using courier, which works fine on 78 other systems just like this |
| 16:16.12 | bugz | but someone got on here and jacked with some configs |
| 16:16.17 | bugz | and im trying to figure out what they changes |
| 16:17.19 | _alex_mx_ | bugz, in voicemail.conf is attach=yes set for each one? |
| 16:18.16 | bugz | yes |
| 16:18.34 | bugz | 9390 => 7777,Some Guy,email@mycompany.com,,|tz=central|attach=yes|saycid=yes|review=yes|envelope=yes |
| 16:19.01 | bugz | man.. nothing |
| 16:19.11 | bugz | i put mailcmd=/bin/mail |
| 16:19.12 | bugz | no go |
| 16:19.24 | bugz | but mailcmd is not specified on any other system |
| 16:20.06 | *** join/#asterisk enioreh (n=ceb@paris.coldev.org) |
| 16:20.10 | enioreh | Hi ppl |
| 16:20.20 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 16:20.47 | rene- | hi |
| 16:21.08 | rene- | hello ppl |
| 16:21.13 | bugz | omfg i hate asterisk |
| 16:21.22 | enioreh | I'm having a strange behavior from asterisk .. when two phones configured with the same account both answered an incoming call at the same time .. then, rtp streams seems to be mixed .. as both phones answered with a sip message 200 OK with session description |
| 16:21.23 | bugz | im going to work at burger king |
| 16:21.38 | rene- | there was a talk yesterday on how asterisk was a mc job |
| 16:21.49 | *** join/#asterisk Niklas- (n=Niklas@0x55513f37.adsl.cybercity.dk) |
| 16:22.32 | rene- | i disabled mysql in the machine i was having problems with but it is still happening, on a 24hr lapse i see 5 events around half hour each where my board goes mad |
| 16:23.12 | eKo1 | rene-: do you have another board to test with? |
| 16:23.25 | rene- | no :( |
| 16:23.51 | rene- | is it true that this can happen when a board has gone bad? |
| 16:24.23 | rene- | i have tested with two machines now |
| 16:24.52 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 16:24.57 | SpaceBass | anyone using IAX2 and voxbone? |
| 16:25.52 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:28.24 | *** join/#asterisk xxarmiexx (n=armie@arm.enter.net) |
| 16:29.40 | coppice | voxbone sounds like a telephone sex service :-\ |
| 16:29.46 | syzygyBSD | bugz: have you tried to sendmail from the commandline? |
| 16:29.55 | syzygyBSD | what does mailq say? |
| 16:30.10 | xxarmiexx | i was wondering if anyone knows the answer to this, for some reason the people in queues dont get queue calls unless they are marked as 'idle' in the flashpanel, is there a way to not have to wait till they become idle |
| 16:31.46 | *** join/#asterisk sonium (n=Sonium@p54928EA3.dip0.t-ipconnect.de) |
| 16:32.11 | Qwell | coppice: wow, that totally does |
| 16:36.46 | *** join/#asterisk foxmjay (n=root@ll81-144-114-192-81.ll81.iam.net.ma) |
| 16:37.30 | foxmjay | hi, plz, is the CHANNEL STATUS used just for the fxs channels? |
| 16:37.42 | Qwell | Where is .ma? |
| 16:37.48 | foxmjay | morocco |
| 16:37.55 | Qwell | ahh |
| 16:37.55 | foxmjay | :) |
| 16:38.12 | foxmjay | so any idea about the channel status |
| 16:38.53 | coppice | i think .ma is with .pa |
| 16:39.02 | foxmjay | looool |
| 16:39.59 | foxmjay | plz does anyone have use the channel stauts with php |
| 16:41.38 | foxmjay | when i use it for an fxo channel , it gives me -1 (failed) ,instade of giving me 0 ( channel is down, and available) |
| 16:42.20 | foxmjay | the others status 1 2 3 4 5 6 worke fine !!!!! |
| 16:45.05 | syzygyBSD | foxmjay: no, channel status is used for all channels |
| 16:45.28 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 16:46.03 | sonium | hi. are the asterisk packets from ubuntu okay or should I compile asterisk myself? |
| 16:46.21 | aydiosmio | the asterisk packets from ubuntu are dirty |
| 16:46.24 | aydiosmio | they need to be cleaned with |
| 16:46.33 | aydiosmio | packetwasher |
| 16:46.45 | sonium | dirty in what meaning? |
| 16:46.46 | aydiosmio | I think you can get that from sourceforge.net |
| 16:47.05 | aydiosmio | dirty in that they need to be cleaned |
| 16:47.48 | sonium | but self-compiling is better, isn't it? |
| 16:48.25 | aydiosmio | it makes the packets cleaner, but you need to keep packetsoap updated to the latest version |
| 16:48.32 | aydiosmio | or else linux will core dump |
| 16:55.19 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:55.50 | *** part/#asterisk chexum (i=chexum@gateway/tor/x-b027d81cb4b8c6b3) |
| 16:56.41 | SpaceBass | can anyone help look at an iax2 debug and see why I'm not getting an inbound call? |
| 16:57.19 | SpaceBass | http://pastebin.ca/157537 the cli doesnt show the call coming in at all |
| 16:57.39 | syzygyBSD | does anyone here use rxfax? |
| 16:58.28 | syzygyBSD | lol, well why not? |
| 16:59.24 | syzygyBSD | SpaceBass: to me it looks like the phone isnt' authing correctly |
| 16:59.38 | syzygyBSD | or the calling application |
| 16:59.51 | SpaceBass | syzygyBSD, its an inbound trunk, but that could be it |
| 17:00.07 | syzygyBSD | call it what you want to call it |
| 17:00.12 | SpaceBass | :) |
| 17:00.17 | _alex_mx_ | SpaceBass, need a few more lines to see what happened with AUTHREQ |
| 17:00.26 | SpaceBass | leme grab em |
| 17:00.59 | aydiosmio | anyone know how I can get the STDOUT/STDERR from AGI applications? |
| 17:01.28 | SpaceBass | http://pastebin.ca/157543 |
| 17:01.35 | syzygyBSD | lol.. well that depends on the language you use |
| 17:01.43 | aydiosmio | perl |
| 17:01.49 | syzygyBSD | print |
| 17:02.00 | SpaceBass | aydiosmio, I had to do that a while ago with a bash script...i think you should just be able to redirect in the call to the script |
| 17:02.13 | _alex_mx_ | SpaceBass, looks like the incoming context is not being found |
| 17:02.15 | SpaceBass | _alex_mx_, thanks for taking a look....i'm better with SIP than iax |
| 17:02.15 | syzygyBSD | in bash it is easy |
| 17:02.18 | aydiosmio | the perl script is failing. I need the error. |
| 17:02.36 | syzygyBSD | echo "blah" |
| 17:02.36 | SpaceBass | _alex_mx_, what line tells you that, so I can learn a little here :) |
| 17:02.40 | foxmjay | syzygyBSD: but why the result is -1 , the line is ok and available, any idea about that ? |
| 17:02.48 | syzygyBSD | oh.. you mean you want to redirect stderr to std out? |
| 17:02.52 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 17:02.58 | _alex_mx_ | SpaceBass, under REJECT it tells you why |
| 17:02.59 | syzygyBSD | hah, the line isn't ok |
| 17:03.13 | aydiosmio | syzygyBSD: it would seem so |
| 17:03.14 | _alex_mx_ | SpaceBass, CAUSE and CAUSE CODE |
| 17:03.17 | SpaceBass | _alex_mx_, i see the no such context/exten ... but is there a way to see which one its looking for? |
| 17:03.18 | syzygyBSD | if it was it wouldn't be -1 would it... |
| 17:03.21 | foxmjay | :) , i test it |
| 17:03.26 | syzygyBSD | what does show channels show for that line |
| 17:04.09 | _alex_mx_ | SpaceBass, do you have a context line in iax.conf for that number? |
| 17:04.17 | syzygyBSD | can you pastebin more of what you are trying? |
| 17:04.22 | SpaceBass | yeah from-pstn |
| 17:04.27 | foxmjay | resrved ,dialing , .. up |
| 17:04.38 | SpaceBass | this is on a trixbox box....from-pstn is the catch all incoming context |
| 17:04.43 | syzygyBSD | the code |
| 17:05.45 | foxmjay | the line is : fwrite($stdout,"CHANNEL STATUS ZAP/1-1 \n"); |
| 17:05.51 | foxmjay | is this correct ? |
| 17:06.12 | _alex_mx_ | SpaceBass, this between 2 * boxes? |
| 17:06.18 | foxmjay | i have an fxo in ZAP/1-1 and it's connected to gsm gatway |
| 17:06.25 | SpaceBass | _alex_mx_, no, its a trunk from Voxbone.com |
| 17:07.12 | _alex_mx_ | SpaceBass, is 14049694655 defined anywhere? |
| 17:07.31 | SpaceBass | _alex_mx_, yeah, doing inbound routing on it |
| 17:07.45 | SpaceBass | _alex_mx_, basically dials a sip phone on my desk |
| 17:07.46 | *** join/#asterisk frenzy (n=frenzy@196.46.104.106) |
| 17:07.54 | *** part/#asterisk frenzy (n=frenzy@196.46.104.106) |
| 17:07.56 | SpaceBass | this damn tropical storm...power keeps blinking |
| 17:08.20 | foxmjay | syzygyBSD: so what do you think ?!! |
| 17:08.36 | syzygyBSD | I know you should drop the last -1 in that |
| 17:08.41 | syzygyBSD | it is zap/1 |
| 17:09.16 | foxmjay | i tried zap/1 too, but its always -1 for all status :) |
| 17:09.25 | docelmo | SpaceBass where do you live? |
| 17:09.33 | SpaceBass | Richmond VA |
| 17:09.56 | aydiosmio | ah duh |
| 17:10.08 | syzygyBSD | this is Agi right? |
| 17:10.09 | aydiosmio | this example perl script has // instead of # for it's comment lines |
| 17:10.10 | aydiosmio | wtf |
| 17:10.18 | foxmjay | yyep |
| 17:10.21 | foxmjay | php |
| 17:10.46 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 17:10.47 | *** join/#asterisk azzie (n=az@azzie.net) |
| 17:11.06 | azzie | anybody used to run asterisk under VMWare virtual server ? |
| 17:11.21 | syzygyBSD | foxmjay: from http://www.voip-info.org/wiki-Asterisk+AGI " channel status: Returns status of the connected channel" |
| 17:11.29 | syzygyBSD | are you trying the connected channel? |
| 17:11.37 | docelmo | SpaceBass Could be worse you could be in florida where this happens every other week or so.. :) |
| 17:11.52 | SpaceBass | docelmo, true! |
| 17:12.06 | syzygyBSD | nm, looked at it more |
| 17:12.27 | syzygyBSD | foxmjay: do you have a Zap/1? |
| 17:12.28 | foxmjay | azzie: how connected channel!!!! ,what's that mean, the gsm gatway is connected |
| 17:12.35 | foxmjay | yes |
| 17:13.12 | foxmjay | i make call's and get call's :D |
| 17:13.53 | syzygyBSD | well, I tend to use manager for things that don't have to deal with the current line, |
| 17:14.04 | syzygyBSD | maybe someone else can help you |
| 17:14.39 | foxmjay | o thanks a lot |
| 17:14.45 | foxmjay | tnx |
| 17:17.53 | *** join/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net) |
| 17:18.09 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 17:18.36 | foxmjay | syzygyBSD: could be somthing that i have to change in extensions.conf or zapata.conf ? |
| 17:20.18 | tasat- | A couple days ago, someone here was recommending a specific linksys router for passing through SIP/RTP traffic -- anyone know what this could have been? This would be for a home net, but should still work and be reliable... |
| 17:21.34 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 17:22.39 | *** join/#asterisk brijn (n=bas@204.244.176.116.net-conex.com) |
| 17:22.44 | brijn | hello all |
| 17:24.57 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 17:30.21 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 17:30.33 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 17:30.39 | jbalcomb | Can Asterisk be made to get its sound file from MySQL and is that a retarded idea? |
| 17:31.10 | *** join/#asterisk svijaykr (n=siddhart@203.197.247.136) |
| 17:31.27 | aydiosmio | jbalcomb: it's better to put a reference to a local file in the database |
| 17:31.34 | jbalcomb | I want to have a central location for configs, sounds, voicemail, etc. because I have two servers for fail-over redundancy |
| 17:31.35 | aydiosmio | storing binary data is always kinda dumb |
| 17:31.35 | Qwell | jbalcomb: not currently, but it's probably not a horrible idea |
| 17:31.54 | Qwell | jbalcomb: I'd certainly be interested to see a patch that does that |
| 17:31.59 | jbalcomb | aydiosmio: that is a silly statement and request you to only if you actually have somethign relevant |
| 17:32.13 | aydiosmio | me to only? |
| 17:32.36 | aydiosmio | and it's easily done with a custom application |
| 17:33.19 | jbalcomb | Qwell: ok, for the time being do you know if and how others are doing this? I am assuming NFS it a fair option. |
| 17:33.38 | hmmhesays | bahhh |
| 17:33.43 | jbalcomb | aydiosmio: let me know when you have that ready |
| 17:33.59 | Qwell | jbalcomb: nfs can be...meh |
| 17:34.02 | *** part/#asterisk azzie (n=az@azzie.net) |
| 17:34.29 | jbalcomb | Qwell: the other part of this is that I'd like users to be able to interface with the recording feature via a webpage |
| 17:34.44 | Qwell | jbalcomb: that wouldn't be too difficult |
| 17:34.50 | jbalcomb | Qwell: especially for holiday greetings and announcements for our call queues |
| 17:35.00 | aydiosmio | #!/usr/bin/perl |
| 17:35.06 | aydiosmio | my $sql = "SELECT * |
| 17:35.07 | hmmhesays | heh |
| 17:35.18 | hmmhesays | when not just do it in C for the speed |
| 17:35.18 | aydiosmio | $AGI->stream_file(" |
| 17:35.20 | aydiosmio | tada |
| 17:35.26 | aydiosmio | there's your custom application |
| 17:35.29 | aydiosmio | smartass |
| 17:35.50 | *** join/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 17:36.16 | aydiosmio | not like piping data from a sql query in perl through EAGI is hard or anything |
| 17:36.22 | svijaykr | can I get some help with installing zaptel on ubuntu 5.10 ? |
| 17:36.23 | jbalcomb | aydiosmio: good man. you are forgiven and thanked. |
| 17:36.33 | svijaykr | is this the right channel for that ? |
| 17:36.45 | aydiosmio | you're welcome, we love doing all the work for you |
| 17:37.38 | aydiosmio | he's some information on perl and blobs |
| 17:37.40 | aydiosmio | http://www.james.rcpt.to/programs/mysql/blob/ |
| 17:38.35 | aydiosmio | http://www.voip-info.org/wiki/view/Perl+EAGI |
| 17:38.40 | jbalcomb | aydiosmio: well, i do pay for the work when someone ponies up with complete and quality work |
| 17:38.42 | aydiosmio | and how to send raw audio data to asterisk |
| 17:38.48 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 17:39.45 | svijaykr | I downloaded and compiled zaptel - did a make install and all that .. but when I try a modprobe zaptel - it says that it can't find the module .. |
| 17:40.19 | jbalcomb | hmmhesays: i would definitely do it in C if i could remember anything I learned 10 years ago. ;) |
| 17:40.26 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp26-194.adsl.forthnet.gr) |
| 17:40.32 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:40.52 | svijaykr | I can see the modules in /lib/modules .. |
| 17:40.53 | jbalcomb | hmmhesays: PERL is all I know at this point but I'm being moved towards Python by our developers |
| 17:40.58 | aydiosmio | jbalcomb: but like I said, file references are better for binary data |
| 17:41.13 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-d3d972a615fadf19) |
| 17:41.26 | jbalcomb | aydiosmio: but would that not leave me back at NFS for my redundancy? |
| 17:41.55 | aydiosmio | there are better filesystems for distributed use |
| 17:42.24 | aydiosmio | e.g. afs |
| 17:42.50 | aydiosmio | I have a firend who works with openafs and prefers it |
| 17:43.56 | jbalcomb | yeah, I used to AFS when I was at VerIO but here everything is NFS |
| 17:44.10 | jbalcomb | I've been toying with SSHFS but it seems troubled still |
| 17:44.24 | eKo1 | troubled? |
| 17:44.49 | *** join/#asterisk worldentropy (i=a@host217-41-100-163.in-addr.btopenworld.com) |
| 17:45.20 | syzygyBSD | can someone explain to me the n option for Queues? "n' — no retries on the timeout; will exit this application and go to the next step." |
| 17:45.27 | syzygyBSD | isn't that the default behavior? |
| 17:46.02 | worldentropy | guys, is there a free/open source/GPL'd g729a codec that will work with Asterisk accept the INTEL IPP one ? |
| 17:46.14 | worldentropy | I have a P4, and that is not compatible with the Itanium code |
| 17:46.43 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:46.51 | syzygyBSD | hmm, i thought the IP owners of g729 didn't allow free versions |
| 17:47.04 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 17:47.33 | worldentropy | well, Intel releases a version that is made freely available, and some kind smaritans have gotten a patch out to make that Asterisk-savvy |
| 17:47.54 | worldentropy | trouble is that it will only run on Itanium-type CPUs and all I have is a straight P4 |
| 17:49.15 | rene- | worldentroy: that is untrue |
| 17:49.33 | rene- | there are compiled versions for most cpu types |
| 17:50.12 | rene- | dont have the url handy |
| 17:50.15 | rene- | but google for them |
| 17:50.24 | rene- | i downloaded P3 versions |
| 17:50.29 | ghenry | Hi all. Anyone form the UK here who can recommend a good VoIP provider (SIP), that offers the ability to resell accounts for use with *? |
| 17:50.51 | worldentropy | rene: thanks for that .. I'll try searching again |
| 17:51.08 | worldentropy | ghenry: depends on the business |
| 17:51.13 | worldentropy | BT Igite do this |
| 17:51.19 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 17:51.21 | worldentropy | amongst a whole string of others |
| 17:51.22 | *** part/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 17:51.22 | kink0 | hi |
| 17:52.01 | ghenry | ok worldentropy . We've tried sipgate.co.uk, but tech support if e-mail only and no reseller accounts. Also, can't even get past the "Security Code" when trying to add Credit Card Credit |
| 17:52.28 | worldentropy | ghenry: register yourself with BT WholeSale |
| 17:52.41 | worldentropy | startup is slow (about 2-4 weeks) but you get the service |
| 17:52.58 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 17:52.59 | kink0 | when I call system(some_external_program &) that I need to be running in background, that program is floked under asterisk thread, so if for some reason the channels is hanged, the background process will still there fews minutes ( I have set an alarm to kill it after 2 minutes ), but that results in a lot of threads in my asterisk |
| 17:53.01 | ghenry | worldentropy: Fine with *, yeah |
| 17:53.02 | worldentropy | and a 24/7 desk, manned by humans, here in the UK not half-way round the palnette |
| 17:53.14 | ghenry | worldentropy: exactly! ;-) |
| 17:53.26 | worldentropy | the service is SIP/H323 compatible |
| 17:53.34 | worldentropy | we've used it, and it works very well |
| 17:53.38 | worldentropy | + it's completely hosted |
| 17:53.39 | kink0 | any idea how to call system() without be folked in a way that if channel is hanged did not still ussing a thread |
| 17:53.41 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:53.59 | ghenry | worldentropy: can easily be an asterisk trunk and incoming to *? |
| 17:54.34 | worldentropy | its bi-directional |
| 17:54.45 | ghenry | sweet |
| 17:54.50 | worldentropy | in other words .. you register with their SIP proxy ..etc |
| 17:54.59 | worldentropy | username/pass auth |
| 17:55.01 | jbalcomb | eKo1: yeah, i have /var/log/asterisk sshfs mounted to the queuemetrics server and occasionaly it errors that the queue_log can not be found |
| 17:55.02 | worldentropy | nothing complicated |
| 17:55.35 | worldentropy | ghenry: what do you do ? are you a carrier ? |
| 17:56.42 | ghenry | worldentropy: We are starting to sell * packages in Scotland, and what to resell VoIP connections whilst using our * product. i.e. by a box from us and we'll sign you up to a VoIp provider, but make some cash from the resell and of course the * services |
| 17:56.59 | *** join/#asterisk ad0le (n=PhilMcre@66.239.217.42.ptr.us.xo.net) |
| 17:57.03 | ghenry | nothing complicated ;-) |
| 17:57.10 | worldentropy | cool gh .. DCC me |
| 17:58.48 | ad0le | I need a high level overview of asterisk, is it possible to use asterisk as a sort of vocal UI to take choices from a user (tone based), send custom voice menus based on thier response, and enter said responses in a database? |
| 17:58.53 | *** join/#asterisk ComputerWarm (n=dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 17:59.05 | ghenry | ad0le: Of course! |
| 17:59.24 | ad0le | is that supported outta the box? |
| 17:59.29 | ad0le | dynamic menuing |
| 18:00.04 | ad0le | or would I have to use a C api to tap into an existing server? |
| 18:00.28 | brijn | A question: what and how do you monitor on your * box |
| 18:00.42 | ComputerWarm | hello does anyone know of any toll free providers that will allow me to start off with 5 or 10 dollars? |
| 18:01.18 | ad0le | brijn, are you talking to me? |
| 18:01.48 | brijn | ad0le: No :) General Q |
| 18:01.53 | rene- | ad0le: |
| 18:01.56 | rene- | no C |
| 18:02.55 | rene- | you can write that easily in either the traditional dial plan programming or the newer asteriske xtension language. |
| 18:02.55 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 18:02.55 | ad0le | rene-, does it have functionality to write to a database? |
| 18:03.57 | ad0le | taking responses, dynamically changing directions based on response, and saving result in a database that is. |
| 18:04.07 | ad0le | is all that supported via the xtension language. |
| 18:04.15 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:07.32 | *** part/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 18:12.24 | rene- | ad0le: yes |
| 18:12.35 | rene- | there is direct access to mysql from the dialplan |
| 18:12.50 | ad0le | well then, looks like im using * from my newest project |
| 18:13.06 | ad0le | any suggestions for books to get started? |
| 18:13.09 | rene- | if you use an external programming language then you have access to any database |
| 18:13.15 | rene- | ~book |
| 18:13.17 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:13.31 | *** join/#asterisk aaugustf (n=aaugustf@copper-14.dynamic2.rpi.edu) |
| 18:14.04 | ad0le | ummm, does the extension language work as a scripting host in other languages like javascript does in browers? |
| 18:14.26 | ad0le | meaning, can it be hosted independently |
| 18:14.33 | ad0le | ill just read :) sorry |
| 18:14.50 | aaugustf | how can i setup a dialing pattern such that when a user enters 9 followed by 4 numbers, asterisk dials out 1212646, followed by the 4 numbers? |
| 18:15.21 | rene- | you can use extensions.conf it is programmable but it s a bit like a configuration file.. on the other hand AEL is more like a programming language with loops and things |
| 18:15.43 | rene- | from any of those you can link to external scripts in your language of choice |
| 18:16.05 | ad0le | one last question |
| 18:16.06 | rene- | that approach is called AGI.... i also believe there was a res_js for javascript |
| 18:16.13 | Qwell | ad0le: think of it this way. it's just a language. mozilla et al had to implement javascript. You could implement whatever language you like in your app |
| 18:16.29 | ad0le | how is the text to speech? |
| 18:16.35 | rene- | two options |
| 18:16.37 | rene- | festival free |
| 18:16.42 | bkw_ | app_cepstral from pbxfreeware.org |
| 18:16.43 | rene- | and cepstral that is unexpensive |
| 18:16.55 | rene- | cepstral has good quality |
| 18:16.57 | bkw_ | the app on pbxfreeware is a BSD licensed module for asterisk |
| 18:17.06 | bkw_ | which uses the SWIFT api to stream |
| 18:17.07 | ad0le | which would you reccomend? |
| 18:17.13 | bkw_ | cepstral |
| 18:17.17 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 18:17.20 | rene- | cepstral |
| 18:17.38 | ad0le | BSD or linux? |
| 18:17.41 | ad0le | or both? |
| 18:18.45 | eKo1 | I use cepstral. It rocks |
| 18:22.05 | crochat | When I call from another IP phone in my LAN, the CallerID is displayed on my analog phone which is connected on the AG168V FXS ! But when I receive an incoming call through my SIP provider, Asterisk tell the CallerID on all the IP phones, but on the phone behind the AG168V, it's just displayed "Incoming call" !! WHY ?? |
| 18:22.59 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 18:27.13 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 18:28.50 | funkmaster | <PROTECTED> |
| 18:29.16 | _deg_ | is there any way to change de voicemail password by phone and change it on the database? Im using asterisk realtime static |
| 18:29.51 | _deg_ | i mean, change it on the phone and update it on the database and not on the conf file. |
| 18:32.56 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 18:34.23 | brijn | A question: what and how do you monitor on your * box |
| 18:35.59 | jbalcomb | brijn: we use Nagios |
| 18:36.58 | jbalcomb | brijn: also, logging remotely via syslog-ng. Haven't set up a log analyzer yet but it will be one of either Logsurfer, SEC, or Lire |
| 18:37.41 | brijn | jbalcomb: But are there specific * parameters you monitor. Like sip peer response times etc |
| 18:37.42 | jbalcomb | brijn: Nagios is just pinging the host for the time being but we quest for a script to connect as a call for checking SIP and RTP |
| 18:37.58 | jbalcomb | brijn: not as of yet |
| 18:38.17 | rene- | can asterisk do a transfer over sip and stay out of signalling and media path? can it act as a stateless proxy, router? |
| 18:38.33 | brijn | I didn't find to much on it, so I might start working on my own.. My primary target is SIP latency as reported by * |
| 18:38.39 | jbalcomb | brijn: i have recently found that there are QoS stats in the BYE sip_header on some phones and someday we will gather that |
| 18:39.02 | brijn | Put it in a graph so that I can see if my provider stars playing tricks on VoiP traffic :) |
| 18:39.09 | aydiosmio | jbalcomb: I'm working on that script |
| 18:39.12 | aydiosmio | should be done in a few |
| 18:39.33 | brijn | jbalcomb: QoS stats? Things as delivery rate, latency? |
| 18:39.43 | jbalcomb | aydiosmio: right on. i'm so busy working on my auto provisioner that i haven't gotten there yet |
| 18:39.52 | jbalcomb | brijn: yes, loss, latency, etc |
| 18:40.11 | jbalcomb | aydiosmio: are you going public with that |
| 18:40.12 | brijn | jbalcomb: That info is not available on * side? |
| 18:40.33 | jbalcomb | brijn: it may be but i assume having both sides is better |
| 18:40.44 | aydiosmio | jbalcomb: I dunno |
| 18:40.44 | brijn | jbalcomb: Always :) |
| 18:41.10 | brijn | aydiosmio: what exactly will you monitor? Or was it the SIP/RTP test that jbalcomb described |
| 18:41.37 | aydiosmio | I'm writing a script to play audio stored in mysql blobs |
| 18:42.37 | brijn | aydiosmio: And what do you do with the audio then? |
| 18:42.59 | aydiosmio | well right now I'm writing it to /tmp and using STREAM FILE |
| 18:43.39 | crochat | Aaargghhhh !! How can I really use the debug output of PalmTool ? It's crazy ! I cannot save anything, and it's too fast to see anything... and after a very short time, all the messages are cleared and replaced by new messages... it's unusable !! |
| 18:43.42 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:46.50 | Zodiacal | anyone know if you can dial internationaly from the u.s. with out having to dial 011? someone once said that you can dial canada with just an area code. is this true for other countries? |
| 18:47.19 | Strom_C | Zodiacal: canada can be dialed on a 1+ basis from the US because the two countries share the same numbering plan |
| 18:47.41 | Strom_C | Zodiacal: you can dial internationally on mobile phones by prefixing the country code with "+" |
| 18:47.41 | *** join/#asterisk LoneShadow (n=duh@59.92.140.210) |
| 18:47.48 | hellor | that's it.... spent the entire morning getting it work and no luck yet. Anyone out there helps me with the Auto Dial Out and dump the call to a conf number (provieded in the .call file) gets $10 in the paypal account :-) |
| 18:48.05 | Strom_C | hellor: $10? that's it? |
| 18:48.16 | hellor | can not afford more :-( |
| 18:48.49 | hellor | need to have a call file and a conf extension... thats a;; |
| 18:48.52 | hellor | all |
| 18:49.00 | Strom_C | in summary: I should not wake up and then immediately IRC |
| 18:49.24 | hellor | so how long is it to do the trick??? |
| 18:50.00 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:50.14 | Strom_C | hellor: well, I tend to have a minimum charge of one hour |
| 18:51.23 | hellor | that's a bit more for my wallet... |
| 18:51.35 | hellor | will keep in mind for some 1 hour worth of work |
| 18:51.41 | hellor | when I can afford it |
| 18:52.21 | hellor | http://pastebin.ca/157705 |
| 18:52.23 | *** join/#asterisk sevard (n=sev@c-67-182-119-152.hsd1.ca.comcast.net) |
| 18:52.26 | hellor | that's my .call file |
| 18:52.38 | hellor | just need a context [trycall] |
| 18:52.39 | Strom_C | sevard, however, will work for bacon scraps! |
| 18:53.08 | sevard | damn dog trim your shit |
| 18:53.25 | eKo1 | that is disgusting... |
| 18:53.34 | sevard | you're disgusting. |
| 18:54.04 | sevard | So I was just noticing today that I have two instances of mpg123 eating up 99% of my cpu |
| 18:54.20 | sevard | and I couldn't remember what some of you astriskphiles recommended |
| 18:54.30 | eKo1 | Sounds like someone didn't shutdown asterisk properly. |
| 18:54.40 | hellor | could someone tell me (1) if the http://pastebin.ca/157705 looks okay? (2) could give me the trycall context for conference dumping |
| 18:55.02 | sevard | eKo1: I _alway_ shutdown asterisk with stop when convient |
| 18:55.25 | *** join/#asterisk volp (n=volp@201.210.82.39) |
| 18:55.25 | sevard | s/convient/convenient/ |
| 18:55.38 | sevard | s/alway/always/ |
| 18:55.45 | sevard | well that's gay. |
| 18:55.51 | Strom_C | you're gay. |
| 18:55.54 | sevard | you wish. |
| 18:56.00 | Strom_C | yes I do. |
| 18:56.03 | sevard | lulz |
| 18:56.05 | eKo1 | ewwww |
| 18:56.06 | Strom_C | ?) |
| 18:56.08 | Strom_C | er :) |
| 18:56.12 | sevard | i'm effin sexy |
| 18:56.21 | *** part/#asterisk volp (n=volp@201.210.82.39) |
| 18:56.29 | eKo1 | You're name is effin sexy? |
| 18:56.30 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 18:56.30 | Nugget | calling things "gay" when you really mean "stupid" is gay. |
| 18:56.37 | sevard | but unless my gf does something nasty i'll stick with her. |
| 18:56.40 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 18:56.42 | sevard | Nugget: that's gay |
| 18:57.04 | hellor | any taker?? |
| 18:57.13 | sevard | i'll take her |
| 18:57.20 | rene- | a taker would be so gay |
| 18:58.04 | Zodiacal | anyone have any ringtone sites preferably for a polycom? but .wav or whatever format will be fine.. |
| 18:58.14 | hellor | <PROTECTED> |
| 18:58.36 | rene- | hellor: where are you from? |
| 18:58.38 | hellor | <PROTECTED> |
| 18:58.49 | hellor | California |
| 18:59.00 | Strom_C | hellor: please stop repeating things you said minutes ago |
| 18:59.25 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:59.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:59.25 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:59.32 | hellor | Strom_C, the post got lost in some out of context stuff ... perhaps that should stop... NO? |
| 18:59.33 | rene- | hellor: what is minimal wage in CA. 8/hr? |
| 19:00.02 | Zodiacal | rene i think 6.25 |
| 19:00.04 | LoneShadow | $8.95 ? |
| 19:00.09 | Zodiacal | last i heard, and they are trying to raise it to 7 somthing |
| 19:00.12 | sevard | hellor: check msg foo |
| 19:00.16 | Zodiacal | crazy people |
| 19:00.39 | LoneShadow | I thought it was 8.xx |
| 19:01.25 | Zodiacal | 6.75 sorry |
| 19:01.26 | Zodiacal | http://www.dol.gov/esa/minwage/america.htm#California |
| 19:02.25 | Luke-Jr | is tf.voipmich.com broken? |
| 19:02.55 | hellor | okay... i got help.. |
| 19:03.08 | LoneShadow | good |
| 19:03.15 | LoneShadow | now hang out here and help 2 more people :D |
| 19:05.35 | LoneShadow | any spa3k experts ? |
| 19:05.52 | Zodiacal | rene maybe it would make products cheaper, and motivate people not to want to work at those types of jobs forever and to move up? |
| 19:06.00 | Zodiacal | rene if they removed it that is.. |
| 19:06.18 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 19:08.51 | rene- | maybe people that has low education and family to support cant afford to be out of job while looking for better jobs? |
| 19:09.25 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917D7A.dip.t-dialin.net) |
| 19:09.39 | rene- | Zodiacal: my individualistic self tells me you are right, but somehow you sound wrong |
| 19:10.01 | rene- | maybe not wrong just evil heh |
| 19:11.45 | designdream | bleh.. where do i buy an iax adaptor in canada? |
| 19:14.47 | aydiosmio | jbalcomb: http://pastebin.ca/157745 |
| 19:15.20 | *** part/#asterisk s1gny|wrk (n=s1gny@p54917D7A.dip.t-dialin.net) |
| 19:15.54 | aydiosmio | If anyone is interested, this is the script to play audio from a mysql database |
| 19:15.57 | aydiosmio | http://pastebin.ca/157745 |
| 19:16.05 | Zodiacal | rene maybe they shoudn't have a family if they cant' support them.. just my thoughts tho |
| 19:16.07 | Zodiacal | maybe im crazy |
| 19:17.23 | Zodiacal | evil = world some times, can't just ignore it.. owell , back to werk i go |
| 19:17.38 | rene- | it is true |
| 19:19.28 | rene- | it is nature and it is evolution, survival of the fittest, in information society you must study to be successful.. and the truth is that most people that dont will be able to reach an old age... they just wont own a business jet |
| 19:19.57 | *** join/#asterisk kronos003 (n=kronos00@c-68-43-231-70.hsd1.mi.comcast.net) |
| 19:21.13 | rene- | but in other countries, the children of those people who work for minimal wages should be allowed to go to school and not be forced to work as childs so they have a shot at buying their own hose and have credit cards and blah blah blah what not |
| 19:21.22 | designdream | i am currently trunking with outside voip providers via sip and have all the ports mapped so that it works through the NAT.. I now need to give an offsite employee an extension via sip... i shouldnt have to change anything on my end for it to work.. correctly? |
| 19:21.28 | designdream | s/correctly/correct/ |
| 19:23.27 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:24.14 | Zodiacal | rene in an ideal world, but we don't live in one... im more pragmatic, than they "should" have those things.. |
| 19:24.37 | Zodiacal | if they make minimum wage, maybe they shoudn't have kids they can't feed? |
| 19:25.06 | rene- | how would you enforce it? |
| 19:25.12 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:25.18 | aydiosmio | now where did that guy go |
| 19:25.31 | aydiosmio | he was all like "solve my problem" and I did, then he disappeared |
| 19:25.34 | syzygyBSD | anyone had a network cable go bad so you only got 40k out of it? |
| 19:25.54 | aydiosmio | it's possible, not likely |
| 19:26.09 | syzygyBSD | it might have been the switch, i replaced them both, but i was getting 700 out of it |
| 19:26.14 | aydiosmio | I've had bad cables, but I didn't note their throughput |
| 19:26.23 | Zodiacal | rene- good point, but having them demand a minimum wage isn't fair either |
| 19:26.40 | syzygyBSD | I went from a speed test of 40 k an hour ago to 25mbit now |
| 19:26.42 | aydiosmio | they mostly exibit large retries, packet loss |
| 19:26.49 | Zodiacal | they should get paid by how much someone is willing to pay for their services |
| 19:26.57 | aydiosmio | run ping for a few minutes and check your drop rate |
| 19:27.09 | syzygyBSD | ya, that was what seemed odd, all the packets went though, just took forever, I think it was the switch |
| 19:27.24 | hmmhesays | time to play some flatout 2 |
| 19:28.07 | Luke-Jr | How do I get Asterisk to authenticate with WWW-Authenticate instead of Proxy auth |
| 19:28.07 | Luke-Jr | ? |
| 19:28.22 | syzygyBSD | now the real question is who do i want to pawn this switch off to |
| 19:28.37 | hmmhesays | haha, what switch? |
| 19:29.42 | file | Luke-Jr: why are you asking that? |
| 19:29.44 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 19:30.24 | *** join/#asterisk RoyKa (n=roy@a217-118-45-74.bluecom.no) |
| 19:30.31 | grabeez | Hi, is it normal behavious for asterisk queues to still pass calls to an agents voicemail if that user is on the phone and call-limit=1. It seems that asterisk is not detecting any extensions in use |
| 19:30.43 | rene- | Zodiacal: no, wages must be fair, a fair day of work should be rewarded by a fair pay |
| 19:31.10 | rene- | Zodiacal: that thinking of yours has caused many revolutions before |
| 19:31.26 | rene- | Zodiacal: the minimal wage is a gimmick to prevent that |
| 19:31.35 | aydiosmio | heh |
| 19:31.39 | aydiosmio | yeah really |
| 19:31.52 | aydiosmio | free market dicitates what "fair" wages are |
| 19:32.06 | aydiosmio | we're lucky to have minimum wage laws |
| 19:32.18 | Zodiacal | rene the freemarket system works, why mess with it.. |
| 19:32.21 | aydiosmio | because their are obviously jobs that would pay less otherwise |
| 19:32.23 | rene- | free market isnt free you know |
| 19:32.26 | aydiosmio | and do (waitressing) |
| 19:32.42 | aydiosmio | free market economics, that is |
| 19:33.02 | Zodiacal | im sure there are people that will work for less than minimum wage.. why not let them |
| 19:33.14 | Zodiacal | = more production, cheaper prices, everyone benifits |
| 19:33.26 | rene- | they do: they are called illegal immigrant workers |
| 19:33.42 | rene- | and to a degree it kind of works the way you say |
| 19:34.07 | Zodiacal | they don't pay taxes tho, so its probably kinda like minimum wage legal workers |
| 19:34.16 | aydiosmio | yes they do |
| 19:34.23 | rene- | as this people support a substantial portion of some third world countries |
| 19:34.41 | aydiosmio | employers pretty much universally impose payroll taxes even on illegal aliens |
| 19:35.01 | aydiosmio | no one pays taxes |
| 19:35.06 | rene- | and you always pay taxes on cigarretes, manufactured goods, gasoline, and alcohol |
| 19:35.10 | aydiosmio | they're taken from us before we get our checks:) |
| 19:36.07 | Zodiacal | aydiosmio how can an employer pay taxes for an employee with out a ss#? |
| 19:36.25 | rene- | anyway in my country at least people that make low wage dont pay income tax by law. |
| 19:36.39 | Zodiacal | and the poor can by food with out paying taxes. |
| 19:36.46 | Zodiacal | acctualy anyone can buy food |
| 19:37.40 | Zodiacal | rene what country? |
| 19:37.44 | rene- | mexico |
| 19:39.07 | Luke-Jr | file: remote SIP server won't work with proxy auth |
| 19:39.16 | rene- | aydiosmio: where are you from? |
| 19:40.04 | Zodiacal | rene one last thing and ill stop ranting.. why tax the rich more? doesn't that loose incentives to become rich? |
| 19:40.25 | file | Luke-Jr: well, WWW is only used for REGISTER and SUBSCRIBE... |
| 19:40.53 | Luke-Jr | file: well, I need it for INVITE? |
| 19:42.31 | rene- | tax the rich more because they can afford it. nobody has become poor by paying taxes.. and nobody should have to choose between eat or health care or paying taxes, or even buy something for themselves.. |
| 19:42.50 | rene- | tax the rich more because they are profiting from the system, |
| 19:43.04 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 19:43.21 | file | Luke-Jr: okay, so what exactly is the scenario? |
| 19:43.37 | sx-wks | rene-: "paying something for themselves" such as food and clothing so that they don't die |
| 19:43.40 | Luke-Jr | file: the caller SIP server only supports WWW-Auth for INVITE |
| 19:43.49 | file | Luke-Jr: no, that's the problem - not the scenario |
| 19:44.02 | Luke-Jr | ... |
| 19:44.12 | file | Luke-Jr: "a remote SIP server is sending a call to me" would be the scenario or "I am placing a call to a remote SIP server" |
| 19:44.45 | rene- | because rich people for the most part will use their money for speculation and will only invest a part of their money in real investments that benefit the community |
| 19:44.49 | Luke-Jr | they're calling via me |
| 19:45.02 | file | via you? they're sending an INVITE to you? |
| 19:45.07 | Luke-Jr | yes |
| 19:45.16 | file | you're probably going to have to modify code then |
| 19:46.09 | Luke-Jr | or just disable passwords, I guess |
| 19:46.12 | file | yup |
| 19:46.31 | aydiosmio | rene-: new york city |
| 19:48.16 | aydiosmio | Zodiacal: fake information |
| 19:48.30 | aydiosmio | I saw a figure somewhere that 50% of illegal immigrants work on the books |
| 19:49.00 | aydiosmio | and pay in about $4200 a year in taxes per household |
| 19:49.04 | Qwell | aydiosmio: This is incredibly off topic |
| 19:49.12 | aydiosmio | thanks |
| 19:49.35 | Qwell | at night, when less people are asking questions, it's a little easier to get away with, but during the day (and currently) people do ask questions |
| 19:49.39 | aydiosmio | I'll remember that next time someone else starts a conversation and I answer a question |
| 19:50.07 | Qwell | aydiosmio: okay, well you're the only one I saw talking, without looking at my scrollback. It obviously goes for everybody. |
| 19:50.17 | Qwell | didn't mean to single you out |
| 19:50.58 | Filar | lol |
| 19:51.34 | Luke-Jr | how can I get the IP of the remote server in a call? |
| 19:51.39 | aydiosmio | phpmyadmin allows you to upload blobs, that's cool |
| 19:52.14 | pdt | any polycom experts, I need to remap a key and can't for the life of me find a reference of the key numbers to do the re-assignment. I could reconfig the phone 40 times I guess, but there has to be an online reference for the key indexes somewhere |
| 19:53.54 | kronos003 | hey what do you guys think of * manager http://astman.sourceforge.net/ |
| 19:54.03 | aydiosmio | assman? |
| 19:54.07 | aydiosmio | hrm |
| 19:54.14 | *** part/#asterisk jpsaman (n=jpsaman@ip545278d7.speed.planet.nl) |
| 19:55.06 | pdt | kronos003: had pretty good luck with it so far, it's pretty flexible |
| 19:55.08 | rene- | manager protects you from crashing your box by sending too much messages to the manager interfase |
| 19:55.29 | aydiosmio | I really like freepbx |
| 19:55.35 | rene- | s/manager protects/ astman protects/ |
| 19:55.36 | aydiosmio | never tried manager |
| 19:55.51 | kronos003 | i see - what do i need to make it work other than asterisk and a default perl installation |
| 19:56.08 | rene- | nothing if you use default mode |
| 19:56.09 | Qwell | kronos003: not a whole lot |
| 19:56.18 | rene- | it is transparent |
| 19:56.20 | pdt | kronos003: it should have install docs, it may bitch about some ssl stuff |
| 19:56.22 | blitzrage | heck, you can telnet into AMI :) |
| 19:56.24 | rene- | perl wont tell the difference |
| 19:56.40 | pdt | blitzrage: bad things |
| 19:56.42 | kronos003 | i was looking at freepbx but it looks fairly invasive |
| 19:56.43 | pdt | hehe |
| 19:57.09 | kronos003 | i need something that i can still run a webserver on the same box |
| 19:57.24 | blitzrage | pdt: well, the only reason I mention that is that AMI is not fancy -- just test messages being passed around |
| 19:57.29 | blitzrage | s/test/text |
| 19:57.32 | pdt | right |
| 19:57.58 | pdt | ya I use some custom socket stuff against it myself, pretty simple protocol |
| 19:58.08 | blitzrage | not that I'm advocating using telnet :) |
| 19:58.25 | kronos003 | so basicly i just put the scripts somewhere and im in business? |
| 19:58.38 | Qwell | kronos003: It's very invasive |
| 19:59.06 | kronos003 | Qwell: freepbx or astman? |
| 19:59.09 | Qwell | freepbx |
| 19:59.17 | Qwell | astman isn't at all - it just sits there looking pretty |
| 19:59.48 | Qwell | I like astman - it's really nice |
| 20:00.00 | pdt | the name could use some work |
| 20:00.04 | rene- | why |
| 20:00.06 | aydiosmio | indeed |
| 20:00.11 | kronos003 | Qwell: cool so with astman i just drop the astman scripts in my cgi dir of one of my ssl hosts and im good? |
| 20:00.11 | pdt | too close to assman for me :) |
| 20:00.21 | Qwell | kronos003: pretty much... |
| 20:00.32 | kronos003 | Qwell: cool i like that |
| 20:00.37 | Filar | laffy taffy bananna just tastes odd |
| 20:00.52 | Qwell | pdt: I've heard far worse... |
| 20:01.01 | rene- | well assman save my ass, man |
| 20:01.08 | kronos003 | now if only someone could start a webmin plugin |
| 20:01.23 | kronos003 | that was OSS |
| 20:03.04 | kronos003 | ok - now that ive got a plan for the front end more or less, now i need to figure out what else i need |
| 20:03.37 | kronos003 | if i want sip phones, voip trunk, and voice mail what all do i need to get other than the asterisk base tarball |
| 20:04.35 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:05.20 | pdt | you ned to by some phones :) |
| 20:05.58 | kronos003 | oh yeah that too - i think i want a polycom ip 601 - that looks like a sweet phone |
| 20:06.07 | aydiosmio | there is a webmin for asterisk |
| 20:06.32 | Qwell | aydiosmio: isn't it like $600 or so? |
| 20:06.42 | kronos003 | aydiosmio: yeah but it costs an arm and a leg - I HATE licencing bs |
| 20:06.47 | *** join/#asterisk Glanzmann (n=sithglan@mini.glanzmann.de) |
| 20:07.18 | Glanzmann | Does somebody know which ethernet chipset is in snom 320? |
| 20:08.02 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 20:08.03 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 20:08.07 | kronos003 | hell i would be HAPPY to donate $$$ to a GPL version just because its GPL and shoretell wanted like 7500 bux plus an M$ server for their stuff |
| 20:08.15 | a1fa | anyway to get sip working on a blackberry? |
| 20:08.45 | syzygyBSD | did they change the way thevoice website works? |
| 20:08.58 | aydiosmio | i mean this one |
| 20:08.59 | aydiosmio | http://ftp.digium.com/pub/telephony/asterisk/webmin/ |
| 20:09.12 | aydiosmio | but it's pretty much useless. |
| 20:09.31 | kronos003 | oh i see |
| 20:09.39 | kronos003 | is it GPL? or what |
| 20:09.42 | syzygyBSD | vi is the best |
| 20:09.47 | kronos003 | LOL |
| 20:10.10 | aydiosmio | no idea |
| 20:10.15 | a1fa | Anyway to get BlackBary to use SIP via EDGE? |
| 20:10.16 | kronos003 | best maybe but certainly not the easiest |
| 20:10.16 | aydiosmio | nano >vi |
| 20:10.37 | syzygyBSD | I hate nano, it doesnt' let you do anything and treats you like a gui user |
| 20:10.37 | *** join/#asterisk findlay (n=justin@67.137.24.115) |
| 20:10.45 | findlay | does anybody here use broadvoice? |
| 20:10.47 | kronos003 | gvim nuf said |
| 20:10.52 | rene- | vim |
| 20:10.52 | aydiosmio | a1fa: with the right java application, it might be possible, but that way beyond the scope of this channel |
| 20:11.01 | syzygyBSD | gvim if I used x... but i don't |
| 20:11.03 | a1fa | aydiosmio : i wonder if anybody done it |
| 20:11.21 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:11.27 | aydiosmio | there's java sip applications, but it's going to vary by hardware |
| 20:11.36 | aydiosmio | so one for your specific device is unlikely |
| 20:12.01 | a1fa | hm |
| 20:12.04 | a1fa | crap dude |
| 20:12.04 | kronos003 | is that digum webmin module Open source or closed |
| 20:12.13 | aydiosmio | BlackBerry 7270 has native SIP support |
| 20:12.26 | aydiosmio | kronos003: well it's obviously not closed |
| 20:12.32 | Qwell | kronos003: Digium webmin module? |
| 20:12.33 | aydiosmio | cause the source is right there |
| 20:12.39 | kronos003 | Qwell: yup |
| 20:12.43 | Qwell | link? |
| 20:12.53 | a1fa | I cant belive no body hacked Blackberry to use voip |
| 20:12.59 | file | it's from a time long ago |
| 20:13.00 | kronos003 | Qwell: http://ftp.digium.com/pub/telephony/asterisk/webmin/ |
| 20:13.16 | aydiosmio | a1fa: well Blackberry did it |
| 20:13.19 | aydiosmio | now there's no reason to |
| 20:13.23 | a1fa | hm |
| 20:13.32 | a1fa | not for all their blackberries |
| 20:13.44 | a1fa | and only through WIFI |
| 20:13.50 | a1fa | not through GPRS/EDGE |
| 20:13.55 | aydiosmio | why would you use SIP through your USB cable? |
| 20:13.58 | kronos003 | if its opensource we should put it up on sourceforge and call it artermin or something |
| 20:14.00 | aydiosmio | oh |
| 20:14.07 | aydiosmio | nm |
| 20:14.08 | kronos003 | astermin that is |
| 20:14.26 | a1fa | yeah |
| 20:14.31 | a1fa | i want to pipe sip through GPRS |
| 20:14.38 | a1fa | buy one of those unlimited data plans for $44 |
| 20:14.41 | aydiosmio | yeah |
| 20:14.45 | aydiosmio | that's why it doesn't work that way. |
| 20:14.52 | a1fa | yeah.. bastards |
| 20:14.57 | *** join/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 20:16.02 | kronos003 | so what the deal with the digium module - whats it missing |
| 20:16.13 | aydiosmio | a1fa: I'm not seeing anything |
| 20:16.41 | aydiosmio | to interface with the handheld's mic and speaker, decode whatever RTP codec you were usign in software |
| 20:16.44 | aydiosmio | that looks tough |
| 20:16.54 | *** join/#asterisk RoyK (n=roy@ti211310a080-1716.bb.online.no) |
| 20:18.00 | a1fa | nah |
| 20:18.02 | a1fa | its easy |
| 20:18.16 | kronos003 | aydiosmio: what is missing in that webmin module you showed us |
| 20:18.23 | a1fa | webmin module sucks |
| 20:18.28 | a1fa | dont use that shit |
| 20:18.31 | a1fa | you will never learn anything |
| 20:18.35 | kronos003 | why - whats wrong with it |
| 20:18.36 | aydiosmio | I mean could the blackberry's CPU handle bidirectional g729? |
| 20:18.48 | a1fa | aydiosmio : g711 maybe |
| 20:18.50 | aydiosmio | kronos003: I couldn't even get it to install, I don't know |
| 20:18.55 | kronos003 | hmm |
| 20:19.05 | brijn | aydiosmio: The new 8700r's have pretty gast CPU (~ 300Mhz I think) |
| 20:19.12 | aydiosmio | a1fa: think you can stream g711 over GPRS? |
| 20:19.25 | kronos003 | aydiosmio: did it kill your webmin install too |
| 20:19.30 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:19.37 | aydiosmio | kronos003: no, just didn't install |
| 20:19.41 | kronos003 | hmm |
| 20:19.53 | kronos003 | gonna try it now to see what happens |
| 20:20.24 | jbalcomb | aydiosmio: that is in fact the most excellent reply I have ever received to a inquiry. thank you. |
| 20:23.13 | kronos003 | hmm no module.info |
| 20:23.18 | *** join/#asterisk alawguy (n=mike@85-124-232-191.work.xdsl-line.inode.at) [NETSPLIT VICTIM] |
| 20:23.34 | aydiosmio | jbalcomb: no problem |
| 20:24.01 | aydiosmio | EAGI doesn't support sending audio back out, so you need to improvise with the temporary file, but this should work great for your purposes |
| 20:25.27 | syzygyBSD | anyone here do fax with asterisk? |
| 20:25.46 | aydiosmio | I heard of some interesting plugins for it |
| 20:26.13 | syzygyBSD | oh, I have it setup, i was just wondering if there were issues with certain machines, options for store and forward, etc |
| 20:27.38 | syzygyBSD | basically we need to make sure we recieve 100% of the faxes, whether forwarding them on to a real machine or keeping them locally |
| 20:27.39 | aydiosmio | what mode are you using? ATA with g711? email-to-fax? |
| 20:28.00 | syzygyBSD | no, have zap in, then a fxo card to a local fax machine |
| 20:28.06 | syzygyBSD | pri in |
| 20:29.15 | syzygyBSD | I have it so it will optionally recieve to a webpage (tiff, then pdf) but I am not sure that rxfax works with all machines, I remember hearing about some problems a while back and wonder how it is now |
| 20:29.35 | kronos003 | who is the coder for this module? |
| 20:29.45 | kronos003 | digum webmin |
| 20:30.35 | aydiosmio | ah |
| 20:31.26 | Niklas- | nice :) |
| 20:32.13 | sx-wks | I implemented a realtime shared blacklist :D |
| 20:32.33 | kronos003 | made it install - it was NOT happy |
| 20:32.53 | sx-wks | now for some more complicated features :d |
| 20:33.41 | aydiosmio | blacklist of what? |
| 20:34.23 | sx-wks | aydiosmio: like callerid based blacklist |
| 20:34.37 | sx-wks | we don't want certain persons from connecting |
| 20:37.06 | aydiosmio | ah |
| 20:37.09 | aydiosmio | neat |
| 20:38.22 | sx-wks | thing is I'm planning ahead there. we'll have only one asterisk server, but that's like the prototype to replace more boxes |
| 20:38.52 | sx-wks | and we want a common blacklist :D |
| 20:40.01 | aydiosmio | interesting |
| 20:40.15 | aydiosmio | all my database stuff has been through perl |
| 20:40.28 | *** join/#asterisk dflow (i=pch@h82-143-131-199-static.e-wro.net.pl) |
| 20:40.39 | aydiosmio | mssql/mysql |
| 20:42.31 | sx-wks | aydiosmio: this is like brand new, was committed 3 days or so ago :D |
| 20:43.11 | *** join/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 20:43.34 | Cubber | does anyone know about how many sip connections a DSL connection can support? |
| 20:44.10 | Cubber | the connection is also used for web and email traffic also |
| 20:44.15 | sx-wks | depends on the codec and the connection |
| 20:44.23 | Cubber | g711 codec |
| 20:44.36 | sx-wks | how much up-going bandwidth do you have ? |
| 20:44.48 | Cubber | around 1.5 meg |
| 20:45.01 | aydiosmio | what? |
| 20:45.03 | sx-wks | so that's about 15 connections tops |
| 20:45.03 | aydiosmio | now ay. |
| 20:45.08 | Cubber | going to do a dsl reports fast |
| 20:45.13 | sx-wks | ok |
| 20:45.22 | sx-wks | it's about 100kbit/s per stream |
| 20:45.25 | aydiosmio | residential dsl is like 512 max |
| 20:45.36 | Cubber | this is commercial |
| 20:45.42 | sx-wks | why do I get 18M/1M then ? |
| 20:45.51 | sx-wks | (with tv and sip ?) |
| 20:46.07 | sx-wks | for 30€ a month ? |
| 20:46.09 | RaYmAn-Bx | aydiosmio: you can't say anything general about that. You can easily get 20/1.5mbit residential here. |
| 20:46.15 | aydiosmio | sounds like FIOS |
| 20:46.18 | Cubber | bah proxy is blocking dsl reports |
| 20:46.21 | Qwell | 30/5 FIOS <3 |
| 20:46.36 | aydiosmio | sx-wks: 30 what? |
| 20:46.38 | aydiosmio | hahaha |
| 20:46.43 | Strom_C | damn you, qwell |
| 20:46.45 | sx-wks | aydiosmio: 30 EUR |
| 20:46.47 | Qwell | Strom_C: no, I wish |
| 20:46.48 | Cubber | g711 is 80k a call right? |
| 20:46.52 | aydiosmio | ah |
| 20:46.52 | Qwell | Strom_C: I've got like 3/384 :P |
| 20:47.00 | Strom_C | Cubber: 64kbls plus overhead |
| 20:47.00 | Qwell | stupid charter |
| 20:47.03 | Strom_C | er |
| 20:47.05 | Strom_C | kbps |
| 20:47.08 | Cubber | yah 16k overhead |
| 20:47.11 | aydiosmio | Cubber is in the US |
| 20:47.14 | aydiosmio | our DSL sucks |
| 20:47.14 | Cubber | yup US |
| 20:47.17 | Strom_C | and the overhead depends on which protocol and packet size you use |
| 20:47.23 | sx-wks | aydiosmio: this explains that :D |
| 20:47.26 | Cubber | last dls report i did was around 1500 |
| 20:47.36 | sx-wks | Cubber: how much do you pay again ? :D |
| 20:47.48 | aydiosmio | Cubber: this is upstream, not down right? |
| 20:47.55 | Cubber | no 1500 down |
| 20:48.00 | sx-wks | erf |
| 20:48.01 | Cubber | up is prolly about 512 |
| 20:48.06 | aydiosmio | yeah, see |
| 20:48.10 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 20:48.17 | aydiosmio | you need symmetic calculations Cubber |
| 20:48.23 | aydiosmio | traffic goes both ways |
| 20:48.25 | sx-wks | ok. so that's like 6 comms max |
| 20:48.33 | Cubber | yah I realize that |
| 20:48.47 | Cubber | 6 comms w/out any other internet traffic or with? |
| 20:48.50 | RaYmAn-Bx | more likely 5 |
| 20:48.54 | aydiosmio | so why bother telling us your download bandwidth, haha |
| 20:49.00 | sx-wks | and that wouldn't leave anythign for the web traffic |
| 20:49.15 | Cubber | so say 3 with qos will allow me to have other traffic? |
| 20:49.18 | sx-wks | yeah |
| 20:49.30 | aydiosmio | eh |
| 20:49.43 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 20:49.56 | Cubber | thanks |
| 20:50.08 | *** join/#asterisk razu_ (n=razu@80-235-90-218-dsl.prn.estpak.ee) |
| 20:50.13 | Strom_C | Cubber: also, iax2 trunking will reduce packet overhead |
| 20:50.27 | Strom_C | Cubber: and g726 sounds almost like ulaw |
| 20:50.57 | Cubber | have to see what my SIP provider uses for codecs, i know they have g711 and ulaw not sure what else |
| 20:51.06 | Strom_C | ulaw /is/ g711 |
| 20:51.06 | aydiosmio | if they have it, use g729 |
| 20:51.10 | Strom_C | oh god no |
| 20:51.14 | Strom_C | g729 sounds like ass |
| 20:51.30 | Cubber | g711 is the most compatable with pstn isnt it? |
| 20:51.38 | Strom_C | g711 is what the pstn uses |
| 20:51.40 | aydiosmio | comparable |
| 20:51.44 | Cubber | thought so |
| 20:51.52 | aydiosmio | not compatible |
| 20:51.56 | sx-wks | Cubber: you may want to use gsm. it's pretty decent |
| 20:52.07 | sx-wks | aydiosmio: how so? |
| 20:52.41 | sx-wks | Cubber: if you want to go g711, use ulaw, as you're in the US |
| 20:52.42 | kpettit | iLBC is pretty cool |
| 20:52.46 | kpettit | speex isn't too bad either |
| 20:52.47 | aydiosmio | the word he was looking for |
| 20:52.53 | aydiosmio | he was comparing call quality |
| 20:52.58 | kpettit | more people/phones seem to support g729 though |
| 20:53.02 | Cubber | telasip only offers g729 and g711 |
| 20:53.10 | aydiosmio | most carrier support g711 and g729 |
| 20:53.15 | aydiosmio | carriers |
| 20:53.17 | sx-wks | kpettit: that requires a non-free licence though |
| 20:53.19 | Cubber | I am using g711 atm |
| 20:53.20 | kpettit | g711 is better quality |
| 20:53.24 | Strom_C | Cubber: be aware that g729 is patent-encumbeed and requires a license |
| 20:53.26 | kpettit | and free |
| 20:53.35 | Strom_C | so you have to pay for crappier audio :) |
| 20:53.35 | Cubber | ahh ok prolly will stick with g711 then |
| 20:53.38 | Cubber | haha |
| 20:53.41 | *** join/#asterisk LoRez_ (i=lorez@freenode/staff/lorez) |
| 20:53.42 | aydiosmio | $10 a line is fine if you wanna stuff 5 calls into a DSL line |
| 20:53.44 | sx-wks | Strom_C: rotfl |
| 20:53.47 | kpettit | Ughh I found a bug today in 1.2.10, g729 codec dosen't work |
| 20:54.02 | kpettit | or rather the licenses don't work, always shows maxed out, first time a person gets on thep hone |
| 20:54.04 | aydiosmio | kpettit: FIX IT FIX IT |
| 20:54.14 | sx-wks | kpettit: even worse, you have to pay for *non_working* audio !!! |
| 20:54.16 | kpettit | I verified the big with Digium, they couldn't fix it either. They told me to down or upgrade |
| 20:54.26 | kpettit | I downgraded to 1.2.9 and it worked again |
| 20:54.40 | kpettit | it really tickes me off because 1.2.9 has a voicemail bug |
| 20:54.56 | sx-wks | kpettit: upgrade, after all 1.2.11 is out :D |
| 20:55.17 | kpettit | yeah I was trying to create a ebuild on gentoo for it, but haven't go it figured out yet |
| 20:55.38 | kpettit | to many weird ass package dependencies |
| 20:57.35 | Cubber | thanks again for the info guys, headin out |
| 20:57.46 | *** part/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 20:57.52 | sx-wks | kpettit: just compile the source :D |
| 20:58.12 | kpettit | it's too nasty to keep track off, it's alot better to use a package manager |
| 20:58.22 | kpettit | espically when i have so many machines. If it was just one box I wouldn't worry about it |
| 20:58.26 | kpettit | but I've got like 40 |
| 20:58.31 | *** join/#asterisk eldu (n=damajor@nysa.e-geek.org) |
| 20:58.35 | eldu | hi |
| 20:58.41 | sx-wks | kpettit: you need to run off the same nfs root dude :D |
| 20:58.54 | kpettit | they are remote machines all over the place |
| 20:59.02 | kpettit | that would be nice though |
| 20:59.31 | eldu | is there someone that runs iaxmodem or uses libiax2 on a sparc ? |
| 20:59.42 | aydiosmio | hm, faxing over the internet using g726, that's cool |
| 20:59.44 | sx-wks | (we are working on massive clusters at work... netboot galore there, using linuxbios and shit) |
| 20:59.51 | kpettit | eldu I use iaxmodem with hylafax, it kicks ass |
| 21:00.07 | eldu | kpettit: on which kind of hardware ? |
| 21:00.12 | kpettit | I've got a machine that's done almost 80k faxes with it so far |
| 21:00.30 | *** join/#asterisk amer (n=ame@202.163.121.116) |
| 21:00.32 | eldu | x86 or other ? |
| 21:00.34 | kpettit | I'm using a SIP connect on this box |
| 21:00.36 | kpettit | x86 |
| 21:00.38 | sx-wks | and they say 'it's a toy, it's not ready, blah...' |
| 21:00.41 | eldu | ok :/ |
| 21:00.50 | amer | how do i turn off realtime support in astpp? |
| 21:01.04 | kpettit | sx-wks, the iaxmodem/hylafax combo is for sure better than spandsp/rx-txfax |
| 21:01.18 | sx-wks | kpettit: heh |
| 21:01.34 | kpettit | eldu, I have a point2point t-1 and that provider (cbeyond) gives us SIP, so we don't have to worry about latency/jitter |
| 21:02.13 | eldu | kpettit: i dont have to worry to latency too ;) the problem is that iaxmodem is crashing |
| 21:02.14 | sx-wks | kpettit: how do faxes get sent through sip ? as a digital image file ? |
| 21:02.19 | aydiosmio | that's crazy stuff |
| 21:02.29 | eldu | i think it comes from libiax on the sparcx |
| 21:02.34 | kpettit | I have DID's through my sip provider, and I sip connect to them. |
| 21:02.48 | aydiosmio | kpettit: what codec? |
| 21:02.59 | kpettit | I use iaxmodem to answer the fax and hylafax to fax2email or whatever |
| 21:03.38 | kpettit | slinear or ulaw |
| 21:03.52 | eldu | alaw should work too |
| 21:04.03 | kpettit | ulaw normally |
| 21:04.14 | kpettit | my sip connect to my provider is ulaw |
| 21:05.03 | kpettit | but iaxmodem/hylafax gives me about a 99% success, on the same setup spandsp/tx-rxfax only gave me about 60-75%. I did testing for weeks on that crap |
| 21:05.33 | aydiosmio | what hardware? |
| 21:05.46 | kpettit | just x86, or x86_64 |
| 21:06.00 | kpettit | sometimes using a sip connect other connections with a Digium PRI |
| 21:06.03 | MikeJ | kpettit.. new revisions of spandsp out very soon will have error correctionworking right. that should be the main diff between those |
| 21:06.04 | eldu | yeah dont use sparc with slowlaris |
| 21:06.07 | sx-wks | (obviously) |
| 21:06.13 | sx-wks | slowarisk :D |
| 21:06.19 | eldu | :) |
| 21:06.31 | aydiosmio | ah, I was reading how the digium cards aren't so good at fax reception |
| 21:06.37 | kpettit | iaxmodem does some tricks as well. The developer for iaxmodem has submited some patches for spandsp that i hope wer included as well |
| 21:06.45 | aydiosmio | some timing issue |
| 21:07.02 | kpettit | aydiosmio, I for sure like the sip connet over a t-1 rather than a PRI |
| 21:07.23 | kpettit | have timing issues when using the PRI cards, never have to worry about it with SIP |
| 21:07.25 | MikeJ | redder86 and coppice are working fairly closely on that.. |
| 21:07.27 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 21:07.37 | aydiosmio | right |
| 21:07.39 | kpettit | I have to buy a specific motherboard to get rid of the timing isues with zaptel/pri combo |
| 21:07.45 | aydiosmio | unless you drop/jitter:) |
| 21:08.14 | aydiosmio | I need to try that out |
| 21:08.16 | kpettit | oh yeah redder86 is who I contracted to initially set it up for us |
| 21:08.19 | kpettit | he kicks ass |
| 21:08.23 | aydiosmio | I'm sure one of my clients will ask about faxing someday |
| 21:08.44 | kpettit | we did one setup across a 800ms vsat connection on a floating oil right that worked flawless |
| 21:09.15 | aydiosmio | ... you faxed over SIP over satellite!? |
| 21:09.15 | kpettit | basically we had hylafax store the fax image then scp it to the server that had a good connection so it could fax it out |
| 21:09.28 | kpettit | yeah, it was kind of a hack between two asterisk boxes but it worked well |
| 21:09.29 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-34-65.nwrknj.east.verizon.net) |
| 21:09.31 | aydiosmio | ah |
| 21:09.56 | kpettit | on the oil rig they had real fax machines and faxed to the local box, then we scp'd the tiff file with allthe fax info then on shore the system there would actually fax it |
| 21:10.15 | kpettit | same way going the other direction as well. works pretty well, becuase you can pick up where you left off even if your connection goes down |
| 21:10.16 | aydiosmio | i see |
| 21:10.27 | kpettit | kind of our own hack on t.37 I guess |
| 21:11.10 | aydiosmio | how long did that take to figure out and test? |
| 21:11.38 | kronos003 | i just sent a note to mpatton - the initial developer for the webmin module - hopefully we can put it up on sourceforge and make something that works well with asterisk |
| 21:11.56 | aydiosmio | ah cool |
| 21:11.58 | kpettit | not too long |
| 21:12.13 | syzygyBSD | tell me if this sounds like an ugly hack or if it should work |
| 21:12.25 | kpettit | we new we'd have to do it like that, and redder86 is the master of all things fax so it was done pretty quick |
| 21:12.25 | MikeJ | it sounds like an ugly hack! |
| 21:12.47 | syzygyBSD | in order to recieve all faxes i send them to a fax machine which forwards them to rxfax with ECM turned on |
| 21:13.12 | kpettit | I'm outta here,have a nice weekend |
| 21:13.32 | MikeJ | huh? |
| 21:15.41 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
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| 21:16.21 | *** join/#asterisk dxll (n=mian@174.Red-83-58-145.dynamicIP.rima-tde.net) |
| 21:16.38 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 21:19.41 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 21:20.21 | *** join/#asterisk Jug (n=Jug@f287b.ttorni.tontut.fi) |
| 21:24.56 | tessier__ | After my AGI exits with exit code 0 I get this error: |
| 21:24.58 | tessier__ | <PROTECTED> |
| 21:25.06 | *** join/#asterisk Iam8up (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 21:25.13 | tessier__ | Why does it try to go to outgoing,804,1? |
| 21:25.30 | tessier__ | I don't have anything after the AGI invocation. I used to have a Hangup there but that didn't change anything. |
| 21:25.41 | Iam8up | can anyone tell me which of the mailing lists i could use to ask for a paying job? |
| 21:25.52 | MikeJ | biz |
| 21:25.58 | Iam8up | biz? |
| 21:26.08 | tessier__ | asterisk-biz |
| 21:26.09 | tessier__ | google for it |
| 21:26.18 | MikeJ | lists.digium.com |
| 21:26.37 | tessier__ | I just put the hangup back in there and it never gets run |
| 21:26.51 | *** part/#asterisk dxll (n=mian@174.Red-83-58-145.dynamicIP.rima-tde.net) |
| 21:27.08 | Iam8up | ya lists.digium.com isn't workign for me |
| 21:27.17 | Iam8up | half of the internet isn't working... |
| 21:27.31 | Iam8up | i can resolve the IPs and all..just can't reach them |
| 21:27.35 | MikeJ | I guess your out of luck then |
| 21:28.17 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:28.47 | Iam8up | ya |
| 21:29.30 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:29.30 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:30.31 | kronos003 | hmm just installed this module - it looks like someone fixed the digium module and put it up on sourceforge http://sourceforge.net/projects/asterisco-em |
| 21:38.24 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:40.23 | tessier__ | exten = 840,1,Hangup |
| 21:40.27 | tessier__ | WHY would that not work? |
| 21:40.34 | blitzrage | question about queues: is it possible to call a "group" of people if I setup a member => to use the Local/ channel? |
| 21:40.34 | Qwell | => ? |
| 21:40.40 | Qwell | blitzrage: yep! |
| 21:40.42 | tessier__ | It gives me the same error as when I tried my AGI. So my AGI is not the problem. |
| 21:40.43 | blitzrage | tessier__: exten => missing the > |
| 21:40.54 | blitzrage | Qwell: exxxxxxxcellent |
| 21:41.09 | blitzrage | Qwell: because the queue() will think the Local chan is always avail right? |
| 21:41.16 | Qwell | blitzrage: Dial(Local/mygroup@mycontext), [mycontext] exten => mygroup,1,Dial(SIP/100&SIP/101) |
| 21:41.32 | blitzrage | Qwell: perfect |
| 21:41.33 | tessier__ | blitzrage: Oops....typo. It is actually there. |
| 21:41.38 | Qwell | it would surprise me immensely if that didn't work |
| 21:41.47 | blitzrage | Qwell: me too |
| 21:45.29 | blitzrage | Qwell: craaaaaaassssshhhhh! |
| 21:45.44 | blitzrage | hard :) |
| 21:46.26 | Qwell | nice |
| 21:47.30 | blitzrage | http://pastebin.ca/157994 |
| 21:48.17 | Qwell | blitzrage: interesting |
| 21:48.24 | Qwell | blitzrage: 1.2? |
| 21:48.34 | blitzrage | Qwell: trunk |
| 21:48.44 | blitzrage | as of like a day ago |
| 21:48.47 | blitzrage | yesterday It hink |
| 21:49.02 | blitzrage | Qwell: lets take this to -bugs |
| 21:49.04 | Qwell | blitzrage: fluke check it... |
| 21:49.12 | Qwell | ie; try it again :D |
| 21:49.15 | blitzrage | checking |
| 21:50.15 | blitzrage | Qwell: yah... seems to be going this time.... |
| 21:50.17 | blitzrage | but eeeeesh |
| 21:50.20 | Qwell | weird |
| 21:50.25 | blitzrage | I hate that.... |
| 21:50.32 | Qwell | hey, at least you got a core |
| 21:50.36 | blitzrage | yah |
| 21:50.59 | blitzrage | that doesn't make me feel very good though :) |
| 21:51.08 | blitzrage | really wish I knew what it was... |
| 22:00.50 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 22:02.11 | C6Vette | I have an agi script question. if i do the following $AGI->set_variable('somevar','abc 123 321'); somevar only gets set to abc. |
| 22:02.24 | C6Vette | Is this supposed to be that way? |
| 22:02.36 | blitzrage | I don't think you can have spaces |
| 22:02.42 | C6Vette | It doeslike spaces |
| 22:02.49 | C6Vette | hmmm. |
| 22:02.59 | blitzrage | or try double quotes? |
| 22:03.49 | blitzrage | either that, or you might have to put escaped double quotes around your string... but that just seems odd :) |
| 22:03.49 | C6Vette | I have tried quotes. Im using set_variable('something',$something) |
| 22:04.28 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 22:05.09 | C6Vette | that worked set_variable('something',"\"$something\"") |
| 22:05.14 | C6Vette | thanks |
| 22:05.50 | tecnico | Hey. In the sound file "record_help: press 1 to switch to playback mode, press 8 to truncate the file and start over," does truncate mean to wipe the file and start again in this context ? Can anyone explain ? |
| 22:06.08 | tecnico | by context I mean, the meaning and purpose of the sentence |
| 22:06.18 | blitzrage | truncate means delete |
| 22:06.26 | tecnico | tnx. |
| 22:13.12 | jbalcomb | If I wanted to have two Asterisk servers redundant fail-over, I should only need to NFS mount /etc/asterisk, /var/lib/asterisk, /var/log/asterisk, and /var/spool/asterisk correct? |
| 22:14.23 | *** join/#asterisk dflow (i=pch@h82-143-131-199-static.e-wro.net.pl) |
| 22:15.58 | C6Vette | jbalcomb, what about voicemail? You using dns for failover |
| 22:16.52 | jbalcomb | C6Vette: voicemail is in /var/spool yes? |
| 22:17.10 | jbalcomb | C6Vette: actually look at using haeart beat monitor |
| 22:17.31 | jbalcomb | C6Vette: not sure what i'm doing about the PRI lines yet |
| 22:17.48 | kronos003 | has any one seen voiceone http://www.voiceone.it/ |
| 22:18.42 | jbalcomb | C6Vette: I'm also hoping that as I go I will be moving configs, sounds, and voicemail to MySQL |
| 22:19.14 | *** join/#asterisk ComputerWarm (n=dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 22:19.27 | ComputerWarm | Hello is the creator or anyone using A2billing around? |
| 22:19.56 | jbalcomb | not I sir |
| 22:20.12 | ComputerWarm | ok ty anyway |
| 22:20.27 | blitzrage | MySQL? ewwwwwwww ;) |
| 22:21.13 | ComputerWarm | no postgresql :-p |
| 22:21.25 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:21.41 | blitzrage | pgsql is the cats meow :) |
| 22:22.18 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:22.21 | ComputerWarm | seems good i don`t use it alot. but i think something is wrong with the way i configure a2bill but not sure |
| 22:24.12 | C6Vette | postgresql has nothing on mysql |
| 22:24.49 | ComputerWarm | lol |
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| 22:39.51 | kronos003 | heres a question - is * multithreaded - IE can it use SMP |
| 22:40.10 | mog | yes |
| 22:41.01 | kronos003 | looks like the folks at astGUIclient need to update their faq "since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down" |
| 22:41.15 | *** join/#asterisk opus_ (i=opus@pabstblueribbon.net) |
| 22:41.19 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 22:41.34 | mog | asterisk has been multi threaded always |
| 22:41.39 | *** part/#asterisk opus_ (i=opus@pabstblueribbon.net) |
| 22:42.03 | kronos003 | hmm is ther any truth in that statment about transcoding? |
| 22:42.36 | *** join/#asterisk [shodan] (n=shodan@ip186.96-113-216.pppoe1.joliette.intermonde.net) |
| 22:43.00 | mog | there is one central proccess for asterisk, the core that starts and uses everything |
| 22:43.17 | mog | but you can and will transcode on multiple procs if you have em |
| 22:43.24 | kronos003 | hmm |
| 22:43.34 | eKo1 | well, each call that uses codec_* gets its own thread so it is multithreaded |
| 22:44.24 | kronos003 | i see |
| 22:46.04 | bkw_ | no two call legs live in that thread |
| 22:46.57 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-a13580f97fba664a) |
| 22:47.24 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:48.53 | jbalcomb | kronos003: you should not use a hyperthreading CPU according to Digium because of the interrupts |
| 22:49.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:49.31 | tessier__ | I have a feeling hyperthreading will have been a short lived technology anyway |
| 22:49.35 | tessier__ | Much better off with separate cores |
| 22:49.42 | jbalcomb | tessier__ agreed |
| 22:49.52 | rene- | i have ht disabled . my smp kernel refuses to start with ht enabled |
| 22:50.08 | kronos003 | what about dual core athlons |
| 22:50.12 | rene- | it is not a custom built kernel, it is debian 2.6.8-2 smp for 686 |
| 22:50.33 | jbalcomb | kronos003: i was not aware that AMD had anything dual core |
| 22:52.50 | kronos003 | operon 265 i thnk also the FX 62 |
| 22:53.46 | dflow | jbalcomb: what's wrong with hyperthreading CPU for transcoding? |
| 22:54.09 | bkw_ | hyperthreading is a fake CPU |
| 22:54.17 | bkw_ | you're splitting the pipeline into two smaller pipes |
| 22:54.29 | kronos003 | FX 62 http://www.newegg.com/Product/Product.asp?Item=N82E16819103861 opteron 285 http://www.newegg.com/Product/Product.asp?Item=N82E16819103004 |
| 22:54.54 | mog | hyperthreading is just bad in my opinion |
| 22:55.46 | mog | but time to leave for the day byes |
| 22:55.52 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:56.23 | rene- | mmm |
| 22:56.40 | [shodan] | I have a big echo problem, I'm going to the nearby town to buy some supply , what kind of hardware should I get while I'm there that could help me with that ? (new cables, baluns etc... ) ? |
| 22:57.29 | *** part/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 22:57.54 | rene- | [shodan]: what setup |
| 22:58.07 | rene- | analog? |
| 22:58.44 | [shodan] | yep, sip phones -> asterisk box -> x100p clones -> pstn |
| 22:59.00 | rene- | x100p have much echo |
| 22:59.13 | [shodan] | they got bad hybrids ? |
| 22:59.32 | rene- | but you could try to change the echocanceler before zapata compilation |
| 22:59.39 | rene- | s/zapata/zaptel/ |
| 23:00.15 | [shodan] | is there anything I can do to solve the problem at the source ? |
| 23:00.20 | rene- | you go ahead and tweak zconfig |
| 23:00.31 | rene- | what sipphones? |
| 23:00.40 | [shodan] | polycoms |
| 23:00.48 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:00.51 | [shodan] | also a spa-2102 |
| 23:01.04 | rene- | calls between polycoms or spa-2102 have any echo? |
| 23:01.06 | [shodan] | inter-office call have no echo |
| 23:01.23 | [shodan] | nope , just small amount of lag |
| 23:01.34 | [shodan] | it's only the outbound calls |
| 23:01.49 | rene- | what about the PC you are using |
| 23:01.57 | rene- | who assembled it? |
| 23:02.13 | [shodan] | a * dedicated athlonxp 2400 512mb built by me |
| 23:02.42 | rene- | [shodan]: there could be a problem of EMI with your case and also the x100p is famous for giving echo problems |
| 23:03.24 | rene- | [shodan]: having said that i think that you can try to select a different echo canceler in zconfig.h inside zaptel sources and enable it in zapata.conf |
| 23:03.55 | [shodan] | k, I'll try that , but what about using a balancing transformer between my x100p and the pstn ? |
| 23:04.04 | rene- | if you have elevated the rx/tx volumes by too much then you can have an echo problem |
| 23:04.22 | rene- | i dont know what a balancing transformer is |
| 23:04.29 | [shodan] | my gains are at 0/0 |
| 23:04.36 | rene- | ok |
| 23:04.47 | [shodan] | it's a device to match the impedance between a transceiver and it's transmission line |
| 23:05.10 | [shodan] | but I don't know if those would help on a phone line |
| 23:05.23 | rene- | i recall something about the x100p clones lacking the ability to have change the impedance |
| 23:05.29 | [shodan] | or if they exist for phone lines and if they exist I don't know what are the values |
| 23:05.39 | [shodan] | http://en.wikipedia.org/wiki/Balun |
| 23:06.14 | [shodan] | my guess is the x100p is set to 600ohm resistive and my line is -something else- |
| 23:06.58 | eKo1 | I use baluns to go from 75 ohm coaxial pair to 120 ohm rj48 |
| 23:08.51 | [shodan] | I'll check the hardware store's electronics section , maybe they have something I can use |
| 23:09.00 | rene- | i believed that the echo canceling code for x100p would change the impedance on the line according to line conditions |
| 23:09.09 | rene- | but that may or may not be truth |
| 23:09.21 | rene- | good luck |
| 23:10.32 | [shodan] | thanks I'll need it , if I can't fix that my whole system is useless and I'll be in a lot of trouble |
| 23:11.08 | rene- | well, unrelated but i have a problem with my zaptel card... the telco suggested to set the second pri port clock to 2 since it is unused.. they say that it could cause troubles with the line |
| 23:11.21 | rene- | with the other port that it is connected |
| 23:11.42 | rene- | that was overlooked by digium but well i dont care whos right as long as they are right |
| 23:12.16 | [shodan] | worst case I'll try to call the telco but they'll probably say "not our problem" |
| 23:12.26 | *** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net) |
| 23:13.03 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:13.33 | rene- | [shodan]: can you afford a tdm400 ? |
| 23:14.25 | [shodan] | nope, the polycoms ate all my budget |
| 23:14.33 | rene- | damn |
| 23:15.20 | [shodan] | I'll go buy some new wiring and maybe some baluns before the stores closes and I'll try to sort this out later |
| 23:17.56 | reneeeeeeeeeeeee | he |
| 23:18.53 | R | realtime queues are not a good idea since you can not get queue statistics in the database nor in ami right? |
| 23:19.32 | R | Qwell: you are the Queue expert, can you tell me if realtime queues can be queried for statistics??/ |
| 23:20.19 | R | i think that everything is best handled by means of static files that you include in queues.conf and reload whenever the app queue gets reloaded |
| 23:20.24 | *** join/#asterisk snappy (i=naveen@zerowing.calpop.com) |
| 23:21.32 | R | the data can be kept in a database one only needs to create the file again rom the db contents and reload the application from command line using asterisk -rtx 'reload app_queue.so' |
| 23:21.47 | R | is everybody out? |
| 23:25.32 | *** join/#asterisk goozbach (n=goozbach@brooks.netradius.com) |
| 23:26.11 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 23:26.49 | R | hi hi hi hi hi hi hi |
| 23:27.13 | *** join/#asterisk rickb|server (n=admin@71.66.110.248) |
| 23:31.09 | tessier__ | Anyone know if QueueAdd or AgentCallbackLogin is the preferred way of getting someone into a queue? |
| 23:31.20 | tessier__ | And if you are using AgentCallbackLogin how do you specify which queue they are to go into? |
| 23:34.08 | Qwell | tessier__: agentcallbacklogin is about to be deprecated |
| 23:34.43 | tessier__ | Qwell: Oh really? So how should I be doing it? |
| 23:35.22 | tessier__ | Should I use QueueAdd? It does not appear to support an AgentID so I'm not sure how I would associate calls with different agents. |
| 23:40.07 | R | Qwell: what about plain agentlogin? |
| 23:40.35 | R | will that be deprecated too? |
| 23:49.26 | *** part/#asterisk rickb|server (n=admin@71.66.110.248) |
