| 00:00.35 | rcahilig | Hi, how to create 3 way calling in Asterisk? |
| 00:01.36 | hardwire | maqr is now "in the know" |
| 00:01.42 | hardwire | no more trying to screw with him. |
| 00:01.50 | plla | Hello. I am trying to output cdr(customname)=customvalue to postgresql. So I decided to modify cdr_pgsql.c in order to work with the extra fields. |
| 00:02.05 | adorah | Hi have anyone replaced hard key buttons in Polycom 501? |
| 00:02.05 | maqr | lol hardwire |
| 00:02.32 | SteveTotaro | who is really in the "know" |
| 00:02.33 | [TK]D-Fender | rcahilig, you don't "create" 3-way calling. Thats a phone feature. What kind of phone are you using? |
| 00:02.50 | [TK]D-Fender | adorah, I've done it on IP 301's |
| 00:02.57 | [TK]D-Fender | adorah, What are you looking to do? |
| 00:03.01 | plla | However, I stumble into the "struct ast_cdr" if I try to call cdr->customname. |
| 00:03.30 | rcahilig | I'm using zoiper free softphone in Ubuntu |
| 00:03.36 | plla | I was wondering if anyone could give me pointers to solve this without modifying asterisk/cdr.h |
| 00:03.36 | disposable | i have a [template](!) in sip.conf in which i define qualify=yes, but out of the hundreds of extensions i have, a handful need qualify=no. can i still leave qualify=yes in the template and just override it in those friends' stanzas? (they will still make use of the template) in other words, will it override the template's qualify=yes or conflict it? |
| 00:03.47 | adorah | <[TK]D-Fender>I've changed the functions now i want to replace the buttons with the transparent ones supplied with the package |
| 00:03.47 | Qwell | disposable: yep |
| 00:03.53 | [TK]D-Fender | rcahilig, if it doesn't offer 3-way calling * isn't going to be able to offer it for you. |
| 00:03.58 | Qwell | that's precisely the reason to use a template |
| 00:04.21 | adorah | <[TK]D-Fender>I've changed them to speed dial |
| 00:04.34 | disposable | Qwell: "tea or coffee sir?" "yes, please" |
| 00:04.46 | Qwell | :) |
| 00:04.51 | [TK]D-Fender | adorah, you're making it sound like you've already done it. Whats your actual question? |
| 00:04.51 | Qwell | yes you can override templates |
| 00:05.01 | disposable | Qwell: thank you |
| 00:05.08 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:05.16 | Qwell | disposable: it parsed as a binary question |
| 00:05.30 | plla | How does cdr_custom.c call extra variables? |
| 00:05.33 | adorah | <[TK]D-Fender>I want to phisically remove the current buttons and replace it with other ones.. |
| 00:06.03 | disposable | [TK]D-Fender: when do you sleep? |
| 00:06.08 | adorah | <[TK]D-Fender>I am not sire how to pull them out without braking anything |
| 00:06.15 | adorah | =sure.. |
| 00:06.27 | SteveTotaro | needle nose pliars |
| 00:06.45 | Qwell | disposable: 3:30-4:00, and 8:00-8:30, I think |
| 00:07.05 | [TK]D-Fender | adorah, I just used my hunting knife just now. If you're not an incompetant klutz you'll be fine... |
| 00:07.35 | disposable | Qwell: certainly looks like it. i've talked to him 15-16 hours ago, 8 hours ago and he's still here. |
| 00:07.44 | [TK]D-Fender | adorah, For everything else there's Polysporin and gauze ;) |
| 00:08.18 | adorah | <[TK]D-Fender>I'm very near being a klutz in such matters but I'll give it a try..thx |
| 00:09.08 | adorah | Perhaps with a screwdriver..here we don't hunt hehe |
| 00:09.23 | SteveTotaro | pliars |
| 00:09.28 | SteveTotaro | don't grip too tight |
| 00:12.58 | *** join/#asterisk codehaxor (n=info@124.6.153.226) |
| 00:13.35 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 00:14.11 | hsv-al | hello fellow internet addicts - are we all looking forward to another long & glorious night on the internet & irc? :) |
| 00:18.55 | codehaxor | hi guys |
| 00:19.04 | *** join/#asterisk mrtelnet (n=mrtelnet@c-67-173-191-235.hsd1.in.comcast.net) |
| 00:19.06 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:20.01 | mrtelnet | Fax over asterisk using T.38 is (to my knowledge) unsupported, I still have to use ulaw to send a fax |
| 00:20.17 | phix | I still have to use a fax machine :) |
| 00:20.45 | mrtelnet | using a FXS card or ata you can. |
| 00:22.22 | *** join/#asterisk hmodes (n=hmodes@ipv6.matrix.gs) |
| 00:22.25 | mrtelnet | has anyone implemented asterisk running on LinuxHA? or done any other type of high avalibility asterisk? |
| 00:23.35 | maqr | shouldn't ulaw be pretty capable of delivering faxes? my understanding was that most PSTN's use ulaw |
| 00:24.12 | SteveTotaro | or alaw |
| 00:24.16 | mrtelnet | ulaw is consistant for me. It just takes alot of bandwidth. |
| 00:24.19 | SteveTotaro | depends on what part of the world |
| 00:24.35 | SteveTotaro | ulaw/alaw=same bandwidth |
| 00:24.55 | mrtelnet | but T.38 Bandwidth < ulaw/alaw bandwidth |
| 00:25.33 | SteveTotaro | mrtelnet: i am doing openser with redfone in the very near future |
| 00:25.34 | Nugget | telnet is eeeeeeevil! |
| 00:25.41 | mrtelnet | very true |
| 00:26.02 | *** part/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 00:26.47 | mrtelnet | SteveTotaro: do you have 2 * boxes and the openser to route between the two? |
| 00:27.12 | SteveTotaro | that is the plan, it is only in planning phase right now |
| 00:27.44 | SteveTotaro | at least two boxes for openser and redfone TDMoE |
| 00:28.10 | *** join/#asterisk anthm (n=anthm@68-29-199-4.area4.spcsdns.net) |
| 00:28.45 | SteveTotaro | both of which i have no experience (yet) |
| 00:29.38 | *** join/#asterisk asteriboy (n=younssig@41.251.7.121) |
| 00:30.04 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 00:30.14 | mrtelnet | Ahh, are you planning on implementing load balancing or just a Active/Passive config? |
| 00:30.21 | mackes | Hey Everyone |
| 00:30.37 | mrtelnet | Hi |
| 00:30.47 | mackes | The I have news |
| 00:31.03 | mackes | The Beta for the Polycom WiFI SIP phone has come to an end |
| 00:31.13 | SteveTotaro | load balancing and failover with dispatcher module |
| 00:31.19 | mackes | So, it is now availible to the public |
| 00:31.27 | mackes | http://www.polycom.com/usa/en/products/voice/mobile/wi-fi/handsets/spectralink_8002_wireless.html |
| 00:31.37 | kimo_sabe | mackes: neat |
| 00:31.42 | *** join/#asterisk twelveca (n=twelve@cx-246.cx.voinetworks.net) |
| 00:31.48 | mackes | Yeah |
| 00:31.51 | SteveTotaro | http://www.openser.org/docs/modules/1.2.x/dispatcher.html |
| 00:31.54 | mrtelnet | awesome |
| 00:32.12 | mrtelnet | mackes: any idea how expensive |
| 00:32.19 | mackes | It is a very cool phone- Drop rated, and very good quality |
| 00:32.32 | mackes | I'm not sure. I think around $350 |
| 00:32.39 | mackes | But they are worth it |
| 00:32.56 | mackes | They are made my a company named spectralink, the Polycom now owns |
| 00:33.05 | mackes | They make heavy duty WiFi phones |
| 00:33.13 | mackes | This is the first full SIP model |
| 00:33.18 | mrtelnet | mackes: interesting, how is the wifi on them? good range? |
| 00:33.21 | mackes | And it is certified on Asterisk |
| 00:33.36 | mrtelnet | thanks stevetotaro, ill give that a shot |
| 00:33.45 | twelveca | URL for the wifi phone? |
| 00:33.55 | mackes | I think so. On on Linksys AP with DDWrt installed, it easily covered my home and my yard |
| 00:34.01 | mackes | http://www.polycom.com/usa/en/products/voice/mobile/wi-fi/handsets/spectralink_8002_wireless.html |
| 00:34.18 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a603bc6161a2698f) |
| 00:34.54 | twelveca | mmmm, looks good I have a client who is looking to replace cell phones for their employees with wifi SIP phones.. ( Apartment complex ) |
| 00:35.13 | twelveca | They spend hundreds per month on local calling to their people while they are in the buildings |
| 00:35.38 | twelveca | thnx for the url |
| 00:35.39 | mackes | We beta'd them in our office building- 4 floors, large building |
| 00:35.56 | twelveca | how many AP did you run for the 4 floors? |
| 00:36.00 | twelveca | 1 per 2 floors ? |
| 00:36.02 | mackes | Worked perfect. However we have 12 AP's (4 per floor) |
| 00:36.12 | twelveca | Cisco running mesh ? |
| 00:36.41 | mackes | Alcatel System- It has a central controller, and the AP's are dumb clients of the controller. |
| 00:36.47 | *** join/#asterisk Mitchell (i=8ea131ae@wikimedia/Greeves) |
| 00:36.48 | mackes | IT allows for roaming |
| 00:36.55 | mackes | But |
| 00:37.15 | mackes | Like I said, on a normal Linksys wth DDWRT, it ran very well |
| 00:38.37 | twelveca | might pick one up and give it a try. |
| 00:38.51 | mackes | it requires the AP have WMM, which is a new protocal in Wireless AP's for multimedia devices that stream. A normal Linksys does not have this option in the firmware. But DDWRT firmware does |
| 00:38.57 | SteveTotaro | DECT |
| 00:39.10 | twelveca | thnx mackes will check it out |
| 00:39.20 | mackes | Cisco AP's work well with the right Cisco firmware as well |
| 00:39.26 | mackes | Cool, thanks |
| 00:42.08 | mrtelnet | off topic, but does ddwrt support the newer routers from linksys |
| 00:46.37 | kimo_sabe | mrtelnet: the L or S ones at least. The basic WRT54G is too gimpy, but Linksys still makes the WRT54GL specifically to be able to run linux-based firmwares |
| 00:51.16 | outtolunc | yes, new wrt54g (v8.2) will run the dd-wrt micro code |
| 00:51.19 | *** part/#asterisk mike-ekim (n=uideqzer@24.238.28.248) |
| 00:51.30 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:51.57 | mrtelnet | great, thanks |
| 00:52.06 | outtolunc | the antennas aren't detachable tho <G> |
| 00:52.59 | codehaxor | hi guys what hardware will be needed for asterisk to be a recording gateway for 50 simultaneous calls |
| 00:55.31 | outtolunc | you would need to be more specific than that, what tech type, codecs, etc |
| 00:55.52 | SteveTotaro | use orecx |
| 00:56.02 | codehaxor | sip, ulaw, g729 going to the voip provider which supports g729 thanks in advance |
| 00:56.05 | SteveTotaro | forget recording in asterisk with any kind of volume |
| 00:56.13 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:58.41 | codehaxor | ok.. |
| 00:58.47 | codehaxor | how do i use that one |
| 00:59.39 | disposable | when you do e.g. Background(enter-ext-of-person), what is the actual filename that gets played? is it enter-ext-of-person.whatever_codec_doesn't_need_recoding ? |
| 01:00.32 | disposable | s/you/i |
| 01:02.21 | Strom_C | yes |
| 01:02.35 | Strom_C | or (i believe) the codec which will incur the least overhead to transcode |
| 01:04.06 | disposable | Strom_C: thank you. if i need to record my own prompts/sounds, where do i place them? where the original files are? or can i specify my own sounds repository in asterisk.conf? (please don't rtfm me, it's 3am here and i haven't slept for 30+ hours :) |
| 01:05.24 | mackes | We recording in Asterisk about 5 channels at a time. ULaw to Wav. Very good results. |
| 01:05.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:06.49 | disposable | Strom_C: no need to answer it. i've found it. thanks |
| 01:07.47 | mackes | Codehaxor: use a multicore machine and scale up slowly. Start with 10 calls, on a true Dual Core, 2.6 Ghz, with 3GB of RAM, and watch your utilization |
| 01:07.53 | SteveTotaro | five channels is not a problem |
| 01:08.27 | SteveTotaro | at ~60 calls you will have an I/O bottleneck and all calls with get choppy |
| 01:08.39 | SteveTotaro | two work arounds, ramdisk or orecx |
| 01:08.42 | Strom_M | disposable: perhaps I should instead tell you to gtba (go to bed already) |
| 01:08.43 | voxter | this is messed up.. on my link at my office (happens to be wireless) calling out via IAX2 sounds choppy and jittery, but calling out over SIP sounds fine. |
| 01:08.47 | mackes | Yep, five was no problem . We have a Dual dual core, and it doesnt show .2 usage in TOP |
| 01:08.50 | SteveTotaro | i like orecx since it is "passive" |
| 01:15.11 | codehaxor | <mackes> Codehaxor: use a multicore machine and scale up slowly. Start with 10 calls, on a true Dual Core, 2.6 Ghz, with 3GB of RAM, and watch your utilization -> Im on a intel core2duo 1.8GHZ with 2GB ram 150 GB SATA 2 HDD, im outputting the file as a WAV |
| 01:15.40 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9) |
| 01:18.25 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 01:18.44 | SteveTotaro | with a c2d @ 1.8 ghz i could capture 200 calls with orecx |
| 01:18.45 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:18.58 | SteveTotaro | asterisk would crap out at ~60-70 |
| 01:19.12 | codehaxor | my target is just 51 calls |
| 01:19.24 | SteveTotaro | take your chances then |
| 01:19.40 | SteveTotaro | hope your HD doesn't fill up |
| 01:19.41 | mackes | codehaxor: And how is it working out? |
| 01:20.30 | codehaxor | <mackes> codehaxor: And how is it working out? --> i tried it on trixbox a while ago, the cpu usage gone haywire, now im trying it on pure asterisk |
| 01:20.48 | codehaxor | im having problems with our current dialers recording server, lots of missing recordings |
| 01:20.56 | *** join/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 01:21.22 | *** join/#asterisk saftsack (n=oliver@p4FC771DB.dip.t-dialin.net) |
| 01:21.25 | drmessano | I have found two things |
| 01:21.52 | saftsack | hi. which revision of the asterisk-addons svn trunk is known to work with chan_mobile-1.4.x.patch? |
| 01:22.00 | drmessano | 1) Asterisk doesn't miss recording calls.. your server tends to get "forgetful" when it's running at 100% CPU and burning swap |
| 01:22.29 | drmessano | 1) Asterisk doesn't forget a ringall of 100 calls.. your server tends to get "forgetful" when it's running at 100% CPU and burning swap |
| 01:22.35 | drmessano | Err.. that was 2 |
| 01:22.37 | drmessano | Bad paste |
| 01:22.39 | drmessano | BAD BAD BAD |
| 01:22.45 | codehaxor | noted... |
| 01:25.38 | seanbright | saftsack: where did you get the patch from? |
| 01:26.05 | saftsack | www.chan-mobile.org |
| 01:26.06 | saftsack | ;) |
| 01:26.14 | seanbright | looks |
| 01:26.34 | SteveTotaro | i think you can find it on matis |
| 01:26.41 | seanbright | saftsack: "Checkout the trunk version of asterisk-addons from Digiums SVN repository" |
| 01:26.53 | SteveTotaro | at least you could when i first started playing with chan_mobile a year ago |
| 01:27.05 | SteveTotaro | chan_bluetooth back then |
| 01:27.06 | seanbright | so http://svn.digium.com/svn/asterisk-addons/branches/1.4 |
| 01:27.10 | seanbright | that's my guess |
| 01:27.26 | saftsack | seanbright, "Download the actual linux kernel and compile it against the actual 5.25 floppy driver" ;) |
| 01:27.31 | saftsack | you know what i mean ^^ |
| 01:27.39 | seanbright | saftsack: you got me |
| 01:27.48 | *** join/#asterisk dustybin (i=subx@microsoft.devilcode.net) |
| 01:27.53 | saftsack | seanbright, i will test it but i think that there isnt a chan mobile in 1.4 |
| 01:28.05 | codehaxor | <SteveTotaro> with a c2d @ 1.8 ghz i could capture 200 calls with orecx --> can it sniff iax? |
| 01:28.15 | saftsack | i had a look at the patch and there were corrections made which seems that the original one was for 1.6 |
| 01:28.18 | dustybin | im going to buy a new cellphone what is capable of making calls over WLAN |
| 01:28.41 | dustybin | i would like it to auth against asterisk, is that possible? |
| 01:28.42 | hsv-al | ?msg xcell they picked up a few of them last week, the nissan place is getting 5 gt-r's, the car is to muscular for you, since your metro :) |
| 01:28.45 | seanbright | saftsack: you are correct. trunk it is. |
| 01:29.14 | seanbright | saftsack: http://svn.digium.com/view/asterisk-addons/trunk/ |
| 01:29.38 | seanbright | s/view/svn/ |
| 01:30.22 | saftsack | downloaded this trunk but this trunk is to new for the patch. now i try revision 530. maybe i have luck with it |
| 01:30.48 | SteveTotaro | if only you could do more than one phone per bluetooth device |
| 01:31.12 | seanbright | saftsack: the patch fails? i can try to cook up one that works if you want |
| 01:32.04 | saftsack | that would be nice but thanks i am able to do it to. and i dont want other people to work just because im lazy ;) |
| 01:32.15 | mackes | SteveTotaro: I checked out the Orecx website, but I am not sure how it works- Does it sniff for RDP traffic and grab it off the wire on the way to Asterisk? Kinda like Wireshark? |
| 01:32.17 | seanbright | saftsack: sounds good. |
| 01:32.27 | hsv-al | saftsack |
| 01:32.35 | hsv-al | so is your name an alternative way of referencing something :) |
| 01:32.44 | saftsack | ;) |
| 01:33.05 | seanbright | yeah, because its *really* subtle |
| 01:33.06 | seanbright | heh |
| 01:33.54 | saftsack | revision 470 seems to work :) |
| 01:34.01 | seanbright | saftsack: nice |
| 01:34.22 | hsv-al | saftsack, your name is as insane as the Joseph A Bank commercials |
| 01:34.24 | hsv-al | for mens suits |
| 01:34.44 | SteveTotaro | it works like wireshark |
| 01:34.47 | saftsack | hrhr |
| 01:35.04 | SteveTotaro | you have to mirror a port |
| 01:35.08 | SteveTotaro | or use a switch |
| 01:35.20 | SteveTotaro | i mean a hub if you can find one |
| 01:35.29 | mackes | That is cool, and creepy. Its a wiretap? Damn |
| 01:35.31 | saftsack | [May 22 03:35:22] ERROR[4592]: chan_mobile.c:2056 load_module: No Bluetooth device found. Not loading module. |
| 01:35.37 | saftsack | seems to work with asterisk-1.4.19 :) |
| 01:35.42 | SteveTotaro | yup, wiretap it is |
| 01:35.53 | mackes | So, you can grab all Voip off a wire, as long as it passes by your port. |
| 01:36.06 | SteveTotaro | that is correct |
| 01:36.20 | SteveTotaro | sip, iax, skinny, 3com..... |
| 01:36.35 | mackes | That is creepy! I use Asterisk on a Cable modem at home. Anyone on my loop could be grabing my Voip Traffic |
| 01:36.39 | SteveTotaro | not sure about mgcp |
| 01:36.49 | seanbright | mackes: its more creepy than snooping smtp, xmmp, or http traffic? |
| 01:36.50 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 01:37.25 | SteveTotaro | my a5 gsm scanner is more creepy than any wired device |
| 01:37.28 | seanbright | mackes: i don't see the distinction. personally i would much rather someone listen to my phone calls then see the stuff i browse for on the interweb ;-) |
| 01:37.47 | mackes | Well. Yeah. I mean I expect my data traffic is out there, but voice traffic, I had hoped was not as easily reassembled. |
| 01:37.49 | *** join/#asterisk jsolis (n=jimmy@190.41.82.1) |
| 01:38.01 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 01:38.23 | seanbright | mackes: packets are packets. |
| 01:38.34 | mackes | <PROTECTED> |
| 01:38.41 | SteveTotaro | your porn collection sean? |
| 01:38.54 | jsolis | hey guys any have experence connecting avaya 8700 with asterisk using PRI |
| 01:38.57 | SteveTotaro | orca seems to be in many distros |
| 01:39.15 | SteveTotaro | i have done a definity g3 |
| 01:39.44 | SteveTotaro | e&m_w |
| 01:41.04 | mrtelnet | Cable modems filter content according to the packet addresses and do not allow you to view your neighbor's data without using a hacked modem. |
| 01:41.23 | mrtelnet | even then, you can only see the data on your local node. |
| 01:41.26 | SteveTotaro | mac address |
| 01:41.45 | mrtelnet | which are carried in the packet's header |
| 01:41.48 | SteveTotaro | local node in the city |
| 01:41.49 | *** join/#asterisk RoyK (n=roy@ip-103-24-149-91.dialup.ice.no) |
| 01:41.57 | mrtelnet | making them the packet's addresses |
| 01:42.11 | *** part/#asterisk RoyK (n=roy@ip-103-24-149-91.dialup.ice.no) |
| 01:42.22 | mrtelnet | no, typically the node only will supply a neigborhood |
| 01:42.46 | SteveTotaro | yes a "neigborhood" is much more dense in a city.... |
| 01:45.12 | SteveTotaro | jameswf said he had software that could sniff image files |
| 01:45.44 | mrtelnet | Wireshark will allow you to reconstruct most data streams including voip and save them to file. |
| 01:46.04 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 01:46.17 | SteveTotaro | orecx will handle most voip out of the box |
| 01:48.18 | SteveTotaro | uses libpcap |
| 01:48.43 | mrtelnet | same as wireshark. |
| 01:49.26 | mrtelnet | Wireshark is just more generalized and supports HTTP, SIP, DHCP, and even Jabber |
| 01:49.49 | SteveTotaro | preaching to the choir |
| 01:50.04 | SteveTotaro | but if you want something to grab audio out of box, then it is orecx |
| 01:50.15 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 01:50.21 | SteveTotaro | not just sip but iax2, skinny... |
| 01:51.04 | mrtelnet | definately if you are looking to capture just audio, that would be the way to go. Personally, I'm more nosy! |
| 01:51.32 | SteveTotaro | then you should check out the USRP and GSM scanner |
| 01:51.52 | SteveTotaro | capture voice, sms, data |
| 01:52.14 | SteveTotaro | gnuradio |
| 01:53.38 | SteveTotaro | http://www.cnet.com/8301-13505_1-9734052-16.html |
| 01:53.48 | mrtelnet | lol |
| 01:53.50 | SteveTotaro | http://technology.timesonline.co.uk/tol/news/tech_and_web/article3945496.ece |
| 01:53.52 | mrtelnet | great |
| 01:54.53 | SteveTotaro | well the gsm scanner takes 2 tb of harddrive to crack a5 |
| 01:55.00 | mrtelnet | eek |
| 01:55.15 | SteveTotaro | but that is the first iteration in the general public |
| 01:55.21 | SteveTotaro | this is N SA stuff |
| 01:55.47 | *** join/#asterisk katiamong (n=denmasbo@202.154.57.15) |
| 01:56.06 | katiamong | hi.. |
| 01:56.39 | SteveTotaro | bonjour |
| 01:56.45 | SteveTotaro | bon nuit |
| 01:57.41 | katiamong | bounjour .. |
| 01:58.01 | SteveTotaro | ca va? |
| 01:58.05 | katiamong | i'm sorry i don't understand french |
| 01:58.13 | katiamong | just say hi |
| 01:58.30 | SteveTotaro | "good day" actually |
| 01:59.39 | katiamong | oke steve |
| 01:59.51 | katiamong | have you ever use dundi for load balancing ? |
| 01:59.55 | katiamong | or anyone |
| 01:59.56 | katiamong | ? |
| 01:59.59 | SteveTotaro | no |
| 02:00.02 | SteveTotaro | openser |
| 02:02.19 | lmadsen | katiamong: device 'load balance' |
| 02:02.26 | lmadsen | s/device/define |
| 02:03.29 | lmadsen | although using the distribution module in openser is probably the best way to do traditional load balancing between two asterisk servers |
| 02:03.36 | katiamong | yes |
| 02:03.55 | katiamong | i've already try to make load balance with openser |
| 02:04.08 | katiamong | openser - -------- > asterisk |
| 02:04.12 | lmadsen | aye |
| 02:04.22 | katiamong | but i've got a proble "loop detected" |
| 02:07.52 | *** join/#asterisk DAHDI (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
| 02:08.38 | katiamong | on asterisk i've got message "482 Loop Detected" |
| 02:08.38 | maqr | how does number portability work with ITSPs? if i wanted to move my number, do they have any legal obligation to let me? (US) |
| 02:09.26 | lmadsen | maqr: I believe they do... your other ITSP you want to move to should be able to request it from them... however I've never done it... (but it seems logical) |
| 02:09.33 | lmadsen | I moved my cellphone number to an ITSP |
| 02:10.20 | maqr | lmadsen: yeah, i'm pretty sure it's legally protected to transfer from the major providers, but i don't know about from ITSPs... it'd make sense, but then again, this is america |
| 02:12.18 | SteveTotaro | http://www.openser.org/docs/modules/1.2.x/dispatcher.html |
| 02:13.16 | katiamong | steve => but i think the asterisk must be patch first |
| 02:13.45 | SteveTotaro | to take calls from a proxy? |
| 02:14.02 | katiamong | because i try to forward the INVITE message from oepnSER to asterisk |
| 02:14.15 | SteveTotaro | dispacher does load balancing with failover |
| 02:14.33 | katiamong | and then i've go message "482 Loop Detected" |
| 02:14.46 | SteveTotaro | http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-registar-sip-proxy-and-far-end-nat-traversal-for-media/ |
| 02:14.53 | SteveTotaro | maybe that will help |
| 02:15.16 | SteveTotaro | do you have reinvite turned on? |
| 02:15.23 | SteveTotaro | that can cause loop detected |
| 02:16.05 | SteveTotaro | i fixed this for an itsp in germany |
| 02:16.14 | SteveTotaro | it was just canreinvite=yes |
| 02:16.21 | SteveTotaro | should be =no |
| 02:16.23 | katiamong | oohh |
| 02:16.29 | katiamong | just it ? |
| 02:16.48 | katiamong | i've read the article http://www.openser.org/pipermail/users/2007-September/013320.html |
| 02:16.49 | SteveTotaro | that's what fixed their issue with openser |
| 02:17.07 | SteveTotaro | if you have reinvite turned on try turning it off |
| 02:17.19 | katiamong | wait a moment steve |
| 02:17.23 | katiamong | i look for it |
| 02:17.26 | *** join/#asterisk osiris (n=osiris@c-71-205-9-42.hsd1.mi.comcast.net) |
| 02:17.31 | katiamong | i think i leave it to default value |
| 02:18.19 | katiamong | ;canreinvite=yes |
| 02:18.31 | katiamong | the default value was cmmented |
| 02:18.39 | katiamong | commented |
| 02:18.41 | SteveTotaro | not sure what it defaults to |
| 02:18.46 | [TK]D-Fender | DEFAULTS? |
| 02:18.47 | SteveTotaro | i would set it to no |
| 02:18.53 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 02:19.14 | [TK]D-Fender | What are these magical "defaults" I'm hearing about? Last I heard you wrote your own configs.... |
| 02:19.26 | SteveTotaro | the mail you linked to was very specific as to the situation |
| 02:19.45 | SteveTotaro | but certain things default to yes or no if you don't set it |
| 02:20.16 | edibrac | hmm my dialplan's GotoIF() seems to be not working the way I want to: http://pastebin.org/37658 |
| 02:20.52 | edibrac | basically, if the outbound call is from a DID line, i want to have caller id set correctly, but from a non-DID, i just want to show the office |
| 02:21.12 | SteveTotaro | most people starting out do make configs and then use those to build their own |
| 02:21.29 | [TK]D-Fender | edibrac, What ver of * are you using? |
| 02:21.36 | SteveTotaro | building configs from scratch is more advanced |
| 02:21.38 | edibrac | [TK]D-Fender: 1.2.12 |
| 02:22.11 | SteveTotaro | why 1.2.12 |
| 02:22.21 | katiamong | steve => so i must patch it first and add the canreinvite=yes |
| 02:22.37 | SteveTotaro | no, i would not patch |
| 02:22.43 | katiamong | yes |
| 02:22.45 | edibrac | SteveTotaro: it's what rpath has released |
| 02:22.58 | SteveTotaro | and no, it is canreinvite=no |
| 02:23.05 | SteveTotaro | not yes |
| 02:23.07 | katiamong | because i rather to use debian way |
| 02:23.07 | edibrac | so until i figure out out to do conary packaging, then I'm going to stick with 1.2.12 |
| 02:23.10 | [TK]D-Fender | edibrac, GotoIf($["${CALLERID(num):0:1}"="4"]?20:30) |
| 02:23.12 | katiamong | with apt |
| 02:23.24 | katiamong | it could my life become easier :D |
| 02:23.27 | DAHDI | patching is evil. |
| 02:23.36 | [TK]D-Fender | edibrac, you on ABE? |
| 02:23.40 | SteveTotaro | DADDY |
| 02:23.41 | katiamong | patching is evil ??? |
| 02:23.47 | edibrac | ABE? |
| 02:23.52 | DAHDI | SUGAR DAHDI for you |
| 02:23.54 | [TK]D-Fender | ~abe |
| 02:23.54 | jbot | from memory, abe is Asterisk Business Edition, a commercial version of Asterisk from Digium, Inc. See http://www.digium.com/ for more information. |
| 02:24.07 | SteveTotaro | he is getting loop detected with openser |
| 02:24.18 | SteveTotaro | i told him to turn off reninvites |
| 02:24.27 | edibrac | ah - no, we run rpath for all our production servers (web, database) and it's been decided that we standardize on rpath |
| 02:24.41 | SteveTotaro | for some reason he thinks this applies http://www.openser.org/pipermail/users/2007-September/013320.html |
| 02:24.49 | katiamong | yesterday i'll try to pacth, and i wasting a lot of time to wait what error could come |
| 02:24.53 | [TK]D-Fender | edibrac, odd choice. You saying you can't compile on your setup? |
| 02:25.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:25.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:25.28 | SteveTotaro | trust me, canreinvite=no |
| 02:25.38 | SteveTotaro | bye bye loop detected |
| 02:25.43 | katiamong | ok steve ... |
| 02:25.47 | katiamong | i'll try it |
| 02:25.57 | katiamong | thks everybody |
| 02:26.06 | katiamong | i hope we will chat soon |
| 02:26.19 | edibrac | [TK]D-Fender: it can/will compile by source - but the company standard is nothing by source, everything by package. in this case, conary. |
| 02:26.21 | SteveTotaro | yes, let me know if it works |
| 02:26.37 | katiamong | ok steve |
| 02:26.40 | [TK]D-Fender | edibrac, Now... if you best friend jumped off a bridge.... |
| 02:26.45 | SteveTotaro | <PROTECTED> |
| 02:26.59 | katiamong | can i have your email please ? |
| 02:27.10 | SteveTotaro | sure, stotaro@asteriskhelpdesk.com |
| 02:27.25 | katiamong | ok thnk a lot steve |
| 02:27.25 | edibrac | well i don't set the standards so -- since I don't lead, I either follow or get out of the way |
| 02:28.25 | SteveTotaro | show them the security fixes since 1.2.12 and whatever else is in the changelogs |
| 02:28.26 | edibrac | [TK]D-Fender: see, that GotoIf($["${CALLERID(num):0:1}"="4"]?20:30) , which is your version, does the same thing -- when I call my cell phone from my internal office phone, the internal line gets sent. |
| 02:28.51 | edibrac | this is so odd. why can I not get a IF/then expression working. |
| 02:28.57 | *** join/#asterisk niZon (n=niZon@tande.voinetworks.net) |
| 02:29.07 | SteveTotaro | K sensitive? |
| 02:29.09 | SteveTotaro | If |
| 02:29.13 | SteveTotaro | not IF |
| 02:29.15 | SteveTotaro | ? |
| 02:29.30 | edibrac | oh wait |
| 02:29.45 | edibrac | goddamit. i was reading his wrong |
| 02:30.11 | [TK]D-Fender | edibrac, forget to reverse the #'s didn't you :) |
| 02:30.13 | seanbright | better than reading his wong. |
| 02:31.08 | SteveTotaro | especially if he reads braille |
| 02:32.46 | *** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net) |
| 02:33.06 | edibrac | no wait.. it is the same - always going to [longdistance-did] http://pastebin.org/37659 |
| 02:33.59 | [TK]D-Fender | edibrac, make priority #1 NOOP it |
| 02:34.16 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 02:34.51 | edibrac | noop what? |
| 02:35.07 | JayTee52 | NoOp? |
| 02:35.40 | edibrac | [TK]D-Fender: from the current Noop() it shows for both 1 and 0, ..this is weird |
| 02:35.57 | edibrac | "exten => _91NXXNXXXXXX,1,Noop($[${CALLERIDNUM:0:1}=4]" ..............." |
| 02:37.11 | edibrac | i guess i should just use a astdb instead - this is sort of a hack since by convention we have a range of DID's that start with a 4 |
| 02:37.25 | edibrac | does info in astdb survive a reboot? |
| 02:37.50 | SteveTotaro | no |
| 02:38.09 | edibrac | then i guess an AGI will have to do |
| 02:40.30 | edibrac | asterisk is so fucking cool |
| 02:40.40 | [TK]D-Fender | edibrac, no, NoOp the CALLERID <- |
| 02:41.06 | [TK]D-Fender | edibrac, screw the evaluation, I don't trust the SOURCE. |
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| 02:42.04 | edibrac | hmm i'll try that when I get home. though I know i'm testing from an internal phone, and I expect the CALLERID to be what I think it should be |
| 02:42.15 | [TK]D-Fender | edibrac, EXPECT? |
| 02:42.19 | [TK]D-Fender | ~assume |
| 02:42.19 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 02:42.36 | edibrac | no you should quote Under Siege 2 |
| 02:43.14 | [TK]D-Fender | edibrac, Nothing he did since the first was worth anything. |
| 02:43.25 | edibrac | like when the henchman comes back to tell the boss that he thinks Steven Segal is dead..but he didn't see a body.... the boss' response was ""assumptions are the mother of all fuckups" |
| 02:45.59 | drmessano | That's a memorable quote? |
| 02:46.13 | drmessano | You need to get out more |
| 02:51.08 | JayTee52 | that's not a memorable quote. memorable quotes are more like, "I think we're gonna need a bigger boat!" from Jaws or "Looks like she doesn't like the cornbread either!" from Aliens. |
| 02:52.58 | maqr | hmm, callcentric is telling me i have to pay some kind of '911 cost recovery fee', wtf is up with that? are all providers doing that now? |
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| 02:54.08 | mackes | If your 911 isnt setup correctly and you dial 911- you are sent to a middleman 911 center that cost mush buck |
| 02:54.25 | maqr | mackes: but i'm never going to dial 911 |
| 02:54.49 | maqr | A setup fee of $2.03 and a monthly fee of $2.37 apply to each Callcentric account requiring 911 service. These fees are used to recover costs directly associated with developing, implementing and maintaining 911 service in compliance with FCC regulations. |
| 02:54.53 | maqr | that's really irritating |
| 02:55.23 | mackes | Yeah. Vitelity lets you op out of 911 |
| 02:55.29 | maqr | If you are located in the United States and have purchased a rate plan for outbound calling, or a phone number to receive calls we must provide you with 911 service as per FCC regulations. |
| 02:55.36 | maqr | hmm, maybe i'll go with them |
| 02:55.44 | JayTee52 | well, they couldn't actually call it "Passthecostontothecustomer_fee" or "WeLikeSodomizingYou_Fee" |
| 02:56.05 | mackes | I LOVE the 911 feature. |
| 02:56.39 | mackes | 2.37 isnt all that much |
| 02:57.33 | mackes | We are on the fringe of VoIP ITSP right now. Everything is going to go up as soon as the Govt catches on as to how much they are losing in taxes |
| 02:57.57 | maqr | mackes: uhh, why would you like the 911 feature? |
| 02:57.59 | maqr | makes no sense |
| 02:58.09 | maqr | i don't need to pay an extra $2.37 for something i will never ever use |
| 02:58.17 | maqr | especially for pay-as-you-go, it's the principle of the matter |
| 02:59.32 | mackes | Without that feature, I need an ATA for 911 |
| 02:59.47 | mackes | How do you dial 911? |
| 03:00.46 | outtolunc | first you dip you left arm in gasoline |
| 03:00.58 | drmessano | I always call the local number for the dispatch center |
| 03:01.07 | maqr | mackes: well, my residence is not voip... this is just for a business line |
| 03:01.13 | drmessano | 911 is waste.. too much guessing anyway |
| 03:01.16 | maqr | mackes: so i have absolutely no need for 911 |
| 03:01.24 | maqr | plus, why would i ever call 911 on anything besides my cell phone? |
| 03:01.30 | maqr | i'm not going to stay trapped in a burning building to make a 911 call |
| 03:02.00 | drmessano | If you run to a neighbors house to call 911, you still have to give them the address |
| 03:02.07 | drmessano | So may as well use a cell |
| 03:02.12 | drmessano | or call the local number |
| 03:02.15 | mackes | People can be hurt, and without 911. well.. Yada Yada. |
| 03:02.16 | maqr | vitelity looks like they don't suck, have you guys had good luck with them? |
| 03:02.30 | maqr | mackes: no one is accidentally going to get trapped inside my VPS and need to make a 911 call |
| 03:02.40 | mackes | I like them. They are a pro outfit. |
| 03:02.47 | maqr | vitelity it is then |
| 03:02.49 | mackes | ok, sounds good |
| 03:03.42 | maqr | they should really give an IP to ping against though |
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| 03:06.40 | mackes | mapr: Do a DNS lookup against the Vitelity address, you will find something to ping |
| 03:07.22 | maqr | mackes: what's the sip address you use for them? i can just ping that if i know what it is |
| 03:08.59 | mackes | proxy01.sipphone.com |
| 03:09.14 | mackes | Nope. |
| 03:09.14 | mackes | Wait |
| 03:09.54 | mackes | inbound1.vitelity.net |
| 03:10.19 | maqr | hmm, 40ms |
| 03:11.07 | maqr | mackes: can you pick any host to use? i ping 20ms to inbound10, which is theplanet |
| 03:12.52 | mackes | ? |
| 03:13.11 | maqr | mackes: inbound10.vitelity.net has great ping for me |
| 03:13.12 | mackes | I'm not sure what you are asking? |
| 03:13.32 | maqr | mackes: did they assign you inbound1, or could you have used a different host of theirs, like inbound10? |
| 03:13.48 | mackes | They tell everyone inbound1.vitelity.net |
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| 03:14.21 | maqr | hrm |
| 03:16.26 | maqr | well, it'll probably work |
| 03:16.27 | maqr | signs up |
| 03:17.10 | timothy_jones | hello, sorry for interrupting, i am having an issue trying to interface * to nec neax 2400 using sangoma a101 and I am wondering if someone here might be able to help out |
| 03:17.55 | timothy_jones | I am using e&m signalling. If I have s,1,Answer() s,n,Echo() in from-pstn, all channels connect and stay Up (show channels)... when I call in from nec, the channel hangs up and calls given exten |
| 03:18.34 | timothy_jones | after that it goes off show channels, but shows busy on nec system |
| 03:20.44 | SteveTotaro | did you use ./setup Install to create your configs on the Sangoma? |
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| 03:21.28 | timothy_jones | SteveTotaro: I used wancfg_zaptel to create all config files |
| 03:21.40 | timothy_jones | is that what you mean? |
| 03:22.26 | SteveTotaro | did you also do wancfg alone? |
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| 03:23.11 | timothy_jones | hmmm, no... it wasn't in their directions |
| 03:23.27 | SteveTotaro | i had sort of the same issues with e&m_w |
| 03:23.34 | SteveTotaro | their scripts are geared to PRI |
| 03:23.52 | SteveTotaro | so check all the configs, zaptel and wanpipe |
| 03:24.33 | SteveTotaro | even though it asked about e&m_w it still configured for PRI |
| 03:24.39 | timothy_jones | I did go through them a few times, but there are some things I am not fully tracking with, should I go ahead and do wancfg? |
| 03:24.48 | SteveTotaro | cannot remember what config file though |
| 03:25.05 | maqr | signs up for vitelity |
| 03:25.06 | SteveTotaro | yes, you should be able to view the setup with wancfg |
| 03:25.22 | SteveTotaro | vitelity is good |
| 03:25.25 | SteveTotaro | been a happy customer of iax.cc for years |
| 03:25.55 | timothy_jones | wondering if vitelity can do 200 channels |
| 03:26.26 | mackes | Yep. We have many, many DID's and are doing as much as 48,000 a month in Minutes |
| 03:26.27 | SteveTotaro | not sure, i just use them for testing and a few toll frees that just go to my cell |
| 03:26.49 | SteveTotaro | seems at that level you could do better than vitelity |
| 03:26.56 | SteveTotaro | unless they are giving you a deal |
| 03:27.36 | mackes | Actual we are going to double to 100,000 minutes by the end of the summer. |
| 03:27.46 | mackes | Per month |
| 03:27.57 | timothy_jones | Anybody knows how they compare to eg callcentric or nexvortex? |
| 03:28.17 | SteveTotaro | nah, i tried voicepulse, nufone, and a couple others |
| 03:28.26 | maqr | i was just about to sign up for callcentric, then i saw they charge $2.50/mo for 911 service |
| 03:28.29 | maqr | and it's mandatory |
| 03:28.29 | SteveTotaro | was not happy until i found iax.cc/vitelity |
| 03:28.35 | maqr | so screw them |
| 03:29.09 | SteveTotaro | my cell phone is fine for 911 it should be optional if waived |
| 03:29.50 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 03:29.50 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 03:29.50 | *** join/#asterisk vitasoy (n=vitasoy@adsl-69-109-124-91.dsl.pltn13.pacbell.net) |
| 03:29.50 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 03:29.50 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:29.50 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:29.50 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
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| 03:29.52 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 03:29.52 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 03:29.52 | timothy_jones | i've been trying to find decent iax provider to patch up all the holes in firewall |
| 03:30.06 | *** join/#asterisk sack (n=sack@33.Red-81-32-160.dynamicIP.rima-tde.net) |
| 03:30.06 | SteveTotaro | funny thing is iax.cc recommends sip |
| 03:30.13 | SteveTotaro | confirms my feelings that iax2 sucks |
| 03:30.23 | mackes | I don't think we can do better the Vitality. No PRIs needed, so no loop charges. That is the savings |
| 03:31.07 | timothy_jones | iax2 sure makes it guys happy :) |
| 03:31.09 | maqr | and i am now a vitelity customer |
| 03:31.11 | SteveTotaro | if you want, i can check SIP pricing with my carrier |
| 03:31.12 | maqr | that was easy |
| 03:31.34 | SteveTotaro | i have qwest resold by usadigital |
| 03:31.59 | SteveTotaro | i have real PRIs but they are dirt cheap $150/mo and a penny a minute |
| 03:33.05 | mackes | That is VERY cheap |
| 03:33.15 | mackes | REALLY Cheap for full PRI |
| 03:33.17 | SteveTotaro | it was a no brainer for me |
| 03:33.23 | timothy_jones | going to check if wancfg is going to save his day or make it more miserable |
| 03:33.47 | SteveTotaro | wancfg is a little strange to get around |
| 03:33.58 | SteveTotaro | but maybe it will help |
| 03:34.13 | SteveTotaro | you can pb your zaptel.conf and zapata.conf |
| 03:34.42 | SteveTotaro | ~pb |
| 03:34.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:36.40 | SteveTotaro | btw, sangoma is great if you finally give in and want help |
| 03:36.40 | SteveTotaro | just give em ssh |
| 03:36.46 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 03:37.51 | maqr | mackes: do you know if the inbound DIDs are only one channel? |
| 03:39.20 | timothy_jones | SteveTotaro: the thing which is throwing me off is CommPort=PRI in wanpipe1.conf file. does that mean the card is using PRI? |
| 03:39.49 | timothy_jones | I am gonna pastebin my zap*.conf files |
| 03:45.16 | timothy_jones | here are my configs seem to be pretty straightforward: http://pastebin.ca/1025563 |
| 03:47.18 | timothy_jones | the only one that is suspicious is wanpipe1.conf -- the CommPort=PRI option: http://pastebin.ca/1025565 |
| 03:50.10 | mackes | No limit on how many calls come in per did |
| 03:50.21 | mackes | maqr |
| 03:50.34 | mackes | You just pay for the usage |
| 03:50.54 | mackes | No limit on the number of channels |
| 03:52.09 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
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| 03:54.34 | mackes | ~mackes |
| 03:54.35 | jbot | it has been said that mackes is the man |
| 03:54.44 | maqr | mackes: oh, ok, awesome |
| 03:56.11 | maqr | mackes: how about outbound caller id? |
| 03:56.29 | mackes | You will always get the number |
| 03:56.49 | mackes | <PROTECTED> |
| 03:56.56 | maqr | err, outbound |
| 03:57.08 | drmessano | ~mackes |
| 03:57.08 | jbot | Don't ask me about mackes. Really. |
| 03:57.10 | maqr | will it always show 'unknown'? |
| 03:57.15 | maqr | ~mackes, really |
| 03:57.18 | maqr | :/ |
| 03:57.25 | mackes | Ohhhh.. Set it to what ever you like |
| 03:58.06 | maqr | mackes: i don't see it in the control panel, i guess i have to have a DID before i can set the outbound name/number? |
| 03:58.13 | maqr | like, so i just don't set a random number? |
| 03:58.30 | mackes | Random number? |
| 03:58.39 | mackes | You can set it to anything you would like |
| 03:58.45 | mackes | Or, nothing |
| 03:59.13 | mackes | If you dont tell it anything, it will show the Vitelity local access number |
| 03:59.26 | mackes | ~mackes |
| 03:59.27 | jbot | you are, like, simply fantastic |
| 03:59.36 | mackes | Thanks jbot |
| 04:00.25 | drmessano | thinks mackes still doesn't get the whole "Setting the reply for your own nickname is kinda douche.. most everyone elses was set for them by someone else..." |
| 04:01.33 | mackes | Umm ok drmessano |
| 04:01.44 | mackes | So I should set them for others. |
| 04:01.50 | mackes | ~drmessano |
| 04:01.51 | jbot | extra, extra, read all about it, drmessano is a douche |
| 04:02.13 | maqr | mackes: hmm, i didn't set it, and it just says "Unknown" on my cell when i call |
| 04:02.23 | mackes | Ahh.. Lucky. |
| 04:02.31 | mackes | Its not always that way |
| 04:02.36 | maqr | mackes: i don't see a place i can set it |
| 04:02.38 | drmessano | Yeah.. the one that was in there was much funnier |
| 04:02.39 | mackes | and it depends who you call |
| 04:02.45 | *** part/#asterisk mrtelnet (n=mrtelnet@c-67-173-191-235.hsd1.in.comcast.net) |
| 04:02.58 | mackes | Add a line to your sip.conf, or in your dialplan |
| 04:03.39 | drmessano | ~mackes |
| 04:03.40 | jbot | http://www.youtube.com/watch?v=P7v7uBA6LW8 |
| 04:03.45 | *** join/#asterisk BBHoss (n=BBHoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 04:04.32 | mackes | You can change your caller iD based on what number you are calling |
| 04:07.03 | maqr | mackes: oh, i didn't realize that was a sip option |
| 04:08.00 | mackes | It is, you just need to add it above and beyond the sampled vitelity gives you. They are one of the last ITSP that allows you to set your own outbound caller ID |
| 04:09.12 | maqr | mackes: that's pretty awesome, i'm gonna try that right now |
| 04:09.47 | BBHoss | mackes, i've seen a ton that still do |
| 04:11.04 | timothy_jones | never mind on CommPort=PRI in wanpipe1.conf. PRI means primary in this case |
| 04:11.27 | mackes | Yeah. Vitelity is a Wholesaler for some very large other very large SIP services. They are actually the product of some other large data organizations. When we spoke with them, they explained that we could drop direct T's in there XO data centers. |
| 04:12.14 | drmessano | *their* |
| 04:12.50 | mackes | ? |
| 04:13.10 | drmessano | their XO data center |
| 04:13.16 | drmessano | not *there* |
| 04:13.23 | mackes | ah yes. Thank you. |
| 04:14.24 | drmessano | Vitelity is still too expensive, IMO |
| 04:14.38 | mackes | Who is less? I like inexpensive |
| 04:15.00 | drmessano | Les.net for one.. 1.44 cents a min for termination sucks |
| 04:16.13 | *** join/#asterisk codehaxor (n=info@124.6.153.226) |
| 04:16.16 | codehaxor | hi guys |
| 04:16.34 | codehaxor | how do i set the ${TIMESTAMP} variable in ast 1.4? |
| 04:16.56 | drmessano | $7.95 for an unlimited DID is awful.. I have seen cheaper |
| 04:17.02 | drmessano | In quite a few places |
| 04:17.03 | mackes | Les.net lists .015 |
| 04:17.16 | mackes | http://les.net/products/product_iptermnorth.php |
| 04:17.18 | drmessano | Yeah, you can do cheaper.. just need to work with them |
| 04:17.27 | drmessano | You dont need to link me.. I use the service, thanks |
| 04:18.05 | mackes | wow. why. why do you always have an attitude? |
| 04:19.05 | maqr | mackes: how exactly are you supposed to specify callerid? |
| 04:19.27 | [TK]D-Fender | codehaxor, You don't, its read-only |
| 04:20.08 | mackes | You can add a line to your SIP.conf setup for Vitelity or in your extensions.conf for your dialing plan that will be directing calls towards Vitelity |
| 04:20.29 | mackes | Vitelity acts just like a SIP phone on system |
| 04:20.45 | maqr | mackes: what's the line look like? |
| 04:20.58 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 04:21.01 | maqr | i tried doing "callerid=<(555) 555-5555>" with my cell number, but it didn't seem to do anything |
| 04:21.59 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
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| 04:28.56 | Ritzerisk | helolo |
| 04:29.06 | Strom_M | omg, a lol is invading that hello |
| 04:29.11 | Strom_M | SOMEBODY DO SOMETHING |
| 04:29.23 | Ritzerisk | :) |
| 04:31.25 | codehaxor | in recording under asterisk which is better to use MixMOnitor() or Monitor()? |
| 04:31.42 | Strom_M | MixMonitor, generally |
| 04:32.00 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:32.15 | codehaxor | ok.. |
| 04:32.20 | codehaxor | which eats more cpu? |
| 04:32.35 | Strom_M | that I don't remember |
| 04:32.43 | Strom_M | but the bottleneck is going to be I/O, not CPU |
| 04:33.36 | mosty | depends what format you are writing out as |
| 04:34.05 | codehaxor | .gsm |
| 04:34.08 | Strom_M | disk space is cheap. don't transcode. |
| 04:34.25 | mosty | i have a voicemail message that was recorded with asterisk, it's all stuttery and garbled, i'm tryng to figure out if there's any other explanation besides "it's a bug in asterisk". anyone have any ideas? |
| 04:34.30 | [TK]D-Fender | Monitor with the join option should be lower than MixMonitor for being a post-call merge |
| 04:34.31 | codehaxor | ok.. so ill be using wav instead of gsm? |
| 04:34.51 | mosty | Strom_C, i have some installs where that's not feasible, even with a large NAS :( |
| 04:35.23 | [TK]D-Fender | mosty, Yeah, go take some drugs. Your imagination is severly stunted and proving a developmental liability. |
| 04:35.33 | mosty | Strom_C, although i do it in post-processing |
| 04:35.36 | Strom_M | well, the only thing I can say to that is "holy fucking caramel-coated jesus christ on a popsicle stick with pralines on top" |
| 04:36.05 | mosty | [TK]D-Fender, the call came in on a PRI line, and went to voicemail, what could be the cause? |
| 04:36.50 | [TK]D-Fender | mosty, how many formats did * write it out as? did you test each of them? Did you consider spying on the channel as the VM is left? Considered that maybe your CALLER |
| 04:36.53 | [TK]D-Fender | s end suck? |
| 04:37.18 | BBHoss | maqr, i always use extensions.conf to set CID |
| 04:37.42 | [TK]D-Fender | mosty, I could probably come up with another dozen points if I had to, but I think you should take this unique opportunity to lift your skirt, grab your balls and man up while the getting's good... |
| 04:37.50 | [TK]D-Fender | :p |
| 04:37.50 | Ritzerisk | anyone ever use the vicidial.... |
| 04:37.56 | codehaxor | ne |
| 04:37.59 | codehaxor | me i have vicidial |
| 04:38.24 | mosty | [TK]D-Fender, the file was a .wav, this only happened once in hundreds of voicemail messages, but the person wants me to give them a reason why it happened |
| 04:39.06 | [TK]D-Fender | mosty, caller's fault. End of story. |
| 04:39.10 | BBHoss | mosty, just tell him the disk ran out of BW and couldnt be written to fast enough :) |
| 04:39.16 | mosty | it's unlikely to be the caller's end, since it came in on PRI and everything sounded fine on their end |
| 04:39.17 | Ritzerisk | really is it able to do a transfer after the message was answered ... like have the recoreded message say if your interested dial 1 to be transfered..... |
| 04:39.22 | [TK]D-Fender | mosty, what a phenominal waste of good theorizing. |
| 04:39.30 | Ritzerisk | andtransfer to my pbx .... |
| 04:39.55 | mosty | [TK]D-Fender, i agree, but i don't have a good excuse to give the client. maybe i'll use the solar flares excuse |
| 04:40.09 | [TK]D-Fender | mosty, if if comes in on a PRI its almost guanteed to be the other end... yours is DIGITAL <- |
| 04:40.15 | Strom_M | proof that telemarketers are the scum of the earth: can't spell "recorded" or use "you're" properly ;) |
| 04:40.36 | maqr | BBHoss: what's the line look like in extensions.conf? |
| 04:41.10 | BBHoss | maqr, 1.4 right? |
| 04:41.19 | maqr | BBHoss: yup |
| 04:41.32 | mosty | [TK]D-Fender, isn't PRI digital too? |
| 04:42.11 | BBHoss | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=15555551212)) |
| 04:42.26 | BBHoss | yours may be a bit different, depending on ITSP |
| 04:42.33 | maqr | hmm |
| 04:42.43 | Strom_M | pfft, the standard fake caller ID number is +1 311 555 2368 |
| 04:42.56 | Strom_M | know it...love it |
| 04:43.47 | Strom_M | jesus fuck, Internet. Learn to spell. Thanks. |
| 04:45.23 | maqr | BBHoss: ty, i'll play with that and see if i can get it to work |
| 04:46.53 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 04:48.38 | maqr | BBHoss: so 1 would be the Set(), 2 would be the Dial(), right? |
| 04:48.45 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 04:52.11 | maqr | doesn't seem to do much here, maybe i do need a DID first |
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| 05:00.11 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 05:10.53 | *** join/#asterisk CCFL_Man2 (i=2fce89f5@pool-70-105-211-134.scr.east.verizon.net) |
| 05:12.24 | CCFL_Man2 | hah, this voice gateway puts every line in a ringing state on the T1 until the individual fxs ports come up |
| 05:12.35 | CCFL_Man2 | it rings constantly for like 30 seconds |
| 05:12.41 | CCFL_Man2 | no cadence |
| 05:13.09 | *** join/#asterisk steliosk (n=Stelios@athedsl-109108.home.otenet.gr) |
| 05:15.49 | Strom_M | that's good. you can test to see that all the ringers work |
| 05:16.19 | Strom_M | and turn the office into some poor receptionist's worst nightmare |
| 05:19.50 | CCFL_Man2 | Strom_M: when you pick up the phone these is no one on the line of course, and when you put it back on the hook it still rings |
| 05:20.08 | CCFL_Man2 | you have more pictures of your phones |
| 05:20.09 | Strom_M | mwahahahahaha |
| 05:20.21 | Strom_M | im off to the diner |
| 05:20.24 | Strom_M | back on from my mobile |
| 05:20.37 | Strom_M | oh, grr, namespace |
| 05:21.19 | CCFL_Man2 | ahh |
| 05:21.25 | CCFL_Man2 | tis kind of late |
| 05:21.54 | Strom_C | it's only twenty past ten |
| 05:22.01 | Strom_C | and the diner is 24/7 |
| 05:22.38 | *** join/#asterisk Strom_M (n=pocketir@m5f0e36d0.tmodns.net) |
| 05:22.48 | Strom_M | there! |
| 05:23.26 | CCFL_Man2 | lol |
| 05:24.12 | CCFL_Man2 | Strom_C: i have to replace the transmitter in my candlestick phone, it sounds like crap |
| 05:24.55 | Strom_M | have you tried just banging it against a hard surface? |
| 05:25.07 | Strom_M | loosen up the carbon granules |
| 05:25.22 | CCFL_Man2 | ahh, was just going to ask that |
| 05:25.24 | CCFL_Man2 | i did |
| 05:25.26 | Strom_M | bang the broad flat plastic side, not the edge |
| 05:25.42 | CCFL_Man2 | they're very loose actually |
| 05:26.03 | codehaxor | hi guys |
| 05:26.06 | CCFL_Man2 | i can even unscrew the capsule and fill it up with new ones |
| 05:26.19 | codehaxor | can anyone give me an example of a cron job that converts wav files to mp3's |
| 05:27.29 | mosty | codehaxor, google has heaps. use lame |
| 05:27.31 | CCFL_Man2 | Strom_C: i was going to replace it with a T1 transmitter |
| 05:30.31 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 05:31.20 | Strom_M | that may or may not work |
| 05:31.35 | codehaxor | ok... |
| 05:31.39 | CCFL_Man2 | Strom_C: oldphoneworks says it does |
| 05:31.46 | Strom_M | not familiar with the requisite electrical jiggerypokery |
| 05:32.23 | CCFL_Man2 | Strom_C: they sell a kit, comes with everything required, even the transmitter |
| 05:32.24 | *** part/#asterisk DAHDI (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
| 05:32.28 | codehaxor | anyway im storing my recordings as a WAV file then after the campaign i will convert them to mp3's |
| 05:32.30 | Strom_M | hah |
| 05:32.49 | CCFL_Man2 | i read on the net that thats the transmitter to use |
| 05:33.01 | Strom_M | codehaxr: thats going to sound horrendous unless you upsample first |
| 05:33.02 | CCFL_Man2 | the receiver is a different story |
| 05:33.30 | CCFL_Man2 | codehaxor: vote democrat, please! |
| 05:33.40 | codehaxor | <Strom_M> codehaxr: thats going to sound horrendous unless you upsample first --> what do you mean? |
| 05:33.52 | codehaxor | <CCFL_Man2> codehaxor: vote democrat, please! --> im from the phils |
| 05:34.24 | CCFL_Man2 | codehaxor: philipines? |
| 05:34.34 | codehaxor | <CCFL_Man2> codehaxor: philipines? --> yep |
| 05:34.43 | Strom_M | codehaxor: im not in the mood to describe all the requisite digital audio theory at the moment |
| 05:34.46 | CCFL_Man2 | ahh |
| 05:35.09 | Strom_M | just trust me on this one :) |
| 05:35.12 | CCFL_Man2 | codehaxor: i forget what rate and format asterisk records at |
| 05:35.28 | Strom_M | ccfl: 8khz mono 16 bit wav |
| 05:36.21 | codehaxor | ok |
| 05:36.23 | codehaxor | thanks.. |
| 05:36.23 | CCFL_Man2 | Strom_C: ulaw? |
| 05:36.38 | CCFL_Man2 | or whatever the codec is? |
| 05:36.44 | codehaxor | anyway im just using the MOnitor() instead of mixmonitor() and outputting my files as raw WAV |
| 05:36.46 | codehaxor | lesser CPU |
| 05:36.50 | Strom_M | no....everything is slin internally |
| 05:36.57 | CCFL_Man2 | ahh |
| 05:37.11 | CCFL_Man2 | Strom_M: http://www.oldphoneworks.com/antique-phone-parts/by-type/part.asp?currency=USD&PhonePart=900&PhonePartType=134 |
| 05:37.25 | Strom_M | ccfl: im on my phone |
| 05:37.27 | CCFL_Man2 | or you can't view urls right now |
| 05:37.29 | CCFL_Man2 | ahh |
| 05:37.54 | Strom_M | well I can, just dont want the hassle right now |
| 05:38.03 | CCFL_Man2 | yeah, it's a pain |
| 05:38.18 | Strom_M | im sick, im at the diner, and i just want my udon, damnit |
| 05:38.19 | CCFL_Man2 | one of these days we should call eachother's sip extention |
| 05:38.28 | Strom_M | yeah |
| 05:38.32 | CCFL_Man2 | whoa, udon at the diner? |
| 05:38.46 | Strom_M | also, its spelled extension |
| 05:38.47 | Strom_M | yep |
| 05:38.53 | Strom_M | welcome to los angeles |
| 05:39.05 | CCFL_Man2 | nice |
| 05:39.20 | drmessano | Looks like some oper pissed off a script kiddie |
| 05:40.57 | Strom_M | thats been happening all night |
| 05:41.12 | drmessano | Yep |
| 05:41.54 | CCFL_Man2 | you know, i was playing with the ceramic phone block downstairs and the damn adsl wic retrained again |
| 05:42.02 | drmessano | Probably will continue until something is rerouted, or the 10 yr old has to go to bed |
| 05:42.08 | CCFL_Man2 | wires have to be too long |
| 05:42.51 | drmessano | too long? |
| 05:43.00 | CCFL_Man2 | i'm sure the 1970 Ma'Bell grey twisted pair doesn't help either |
| 05:43.21 | CCFL_Man2 | err, two pair, i doubt if there is a twist in it |
| 05:43.26 | drmessano | I doubt length is your issue |
| 05:43.32 | BBHoss | drmessano the MOTD says the servers are split |
| 05:43.33 | drmessano | or the twist |
| 05:44.07 | CCFL_Man2 | drmessano: the pair inside, too much of the outer jacket was stripped off |
| 05:44.22 | drmessano | I mean.. DSL travels miles down some of the crappiest copper on earth.. Not having a "data twist" inside the house and an extra few feet of wire really aren't going to hurt it |
| 05:44.36 | Strom_M | ddfl: barking up the wrong tree...dsl was designed to work with that crap |
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| 05:45.02 | CCFL_Man2 | Strom_M: my adsl is fast path, so i get retrains more often |
| 05:45.12 | drmessano | BBHoss: So? |
| 05:45.17 | CCFL_Man2 | on interleaved it hasn't retrained once |
| 05:45.32 | CCFL_Man2 | and my CO is right down the street |
| 05:45.34 | drmessano | BBHoss: MOTD's mean nothing.. they don't exactly announce "Hey, we're being attacked.. hang on" |
| 05:45.55 | BBHoss | true |
| 05:46.21 | drmessano | I was 450 feet from my CO once |
| 05:46.26 | drmessano | and I would still go down |
| 05:46.27 | Strom_M | i dont remember "fast path" in my everything you never wanted to know about xdsl book |
| 05:46.29 | CCFL_Man2 | the wire coming into my house is the vintage 1940s kind, black, rubber insulated, rectangle shaped outer jacket |
| 05:46.30 | drmessano | Still have line issues |
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| 05:46.40 | BBHoss | why do people say CO, is that what they mean where the DSLAM is? |
| 05:46.46 | drmessano | Who cares what kind of block it is lol |
| 05:46.50 | drmessano | Is it metal? |
| 05:46.52 | drmessano | Does it hold a wire? |
| 05:46.56 | drmessano | Good. |
| 05:47.17 | drmessano | BBHoss: I hope not |
| 05:47.26 | drmessano | BBHoss: I hope he means the central office |
| 05:47.42 | CCFL_Man2 | if the wiring block is messy looking i always fix it right to kelp with the signal |
| 05:48.01 | CCFL_Man2 | i hear some people tell me adsl needs atleast cat3 |
| 05:48.08 | drmessano | LOL |
| 05:48.11 | drmessano | Uh huh |
| 05:48.23 | Strom_M | ccfl: did you learn wiring imr from the urban legend fairy? |
| 05:48.31 | CCFL_Man2 | on fast path adsl i'm begining to believe it |
| 05:48.41 | drmessano | Because, there's cat3 all over the poles outside |
| 05:48.44 | BBHoss | i have seen DSLAMs in the fiber termination point |
| 05:49.07 | drmessano | What is "fast path" DSL? |
| 05:49.09 | CCFL_Man2 | Strom_M: i never believed it |
| 05:49.41 | Strom_M | ccfl: what is this "fast path" gibberish? |
| 05:50.48 | CCFL_Man2 | http://www.dslreports.com/faq/1238 |
| 05:52.03 | Strom_M | sounds like someone made up that name |
| 05:52.14 | Strom_M | because i dont remember reading that in my dsl book |
| 05:52.27 | Strom_M | but ill check when i get home |
| 05:52.27 | CCFL_Man2 | my adsl stats in my cisco router have interleaved and fast path stats |
| 05:52.58 | drmessano | interleaving and non-interleaving have to do with data transmission methods.. they're not a type of DSL |
| 05:53.03 | drmessano | ADSL is ADSL |
| 05:53.20 | CCFL_Man2 | bits are spread in time and interleaved with bits from other packets |
| 05:53.28 | CCFL_Man2 | drmessano: right |
| 05:53.42 | drmessano | You keep calling it "Fastpath DSL" |
| 05:54.00 | CCFL_Man2 | i should say fast path mode |
| 05:54.07 | drmessano | So I should refer to my cable as "DOCSIS 2.0 Cable" |
| 05:54.16 | CCFL_Man2 | or fast path transmission |
| 05:54.33 | drmessano | Sounds to me like you learned a new term and are looking to wear it out |
| 05:55.05 | drmessano | Enough with the fastpath nonsense.. You don't need Cat3 to get DSL to work, and your wiring block nonsense is bollocks |
| 05:55.25 | drmessano | What happens at your house has little to do in the way of affecting your DSL signal |
| 05:56.03 | Strom_M | like i said: the wiring urban legend fairy |
| 05:56.08 | drmessano | you can do like I did when I was younger and run CAT5 outside for it, and then watch nothing happen |
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| 05:56.42 | drmessano | Because putting Cat5 on your DSL line to speed it up is like wiping vaseline on the hood of your car |
| 05:57.06 | CCFL_Man2 | i don't want to speed it up, i want to lower the noise level |
| 05:57.15 | drmessano | Oh god |
| 05:57.37 | drmessano | beats head on desk |
| 05:57.49 | drmessano | Go put some CAT5 on it |
| 05:57.50 | CCFL_Man2 | or do i have the audiophool syndrome? |
| 05:57.54 | Strom_M | let me just take this conversation to its logical extreme now: MONSTER CABLE !!!!!! |
| 05:57.59 | drmessano | HA |
| 05:58.06 | drmessano | Yeah, get some monster cable |
| 05:58.30 | CCFL_Man2 | Strom_M: i'm a fan of comtran |
| 05:58.57 | Strom_M | ? |
| 05:58.57 | drmessano | Monster Cable is a farse |
| 05:58.58 | drmessano | Someone put two audio experts in a room... and ran two tests.. One with monster cable |
| 05:59.21 | drmessano | and one with COAT HANGERS |
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| 05:59.23 | drmessano | It was double blind |
| 05:59.38 | CCFL_Man2 | Strom_M: comtran telco cable |
| 05:59.38 | drmessano | and they couldnt tell the difference |
| 05:59.38 | CCFL_Man2 | drmessano: of course |
| 05:59.48 | Strom_M | drmessano: well yeah, coathangers are nice heavy gauge :) |
| 05:59.57 | drmessano | CCFL_Man2: Yet, you're talking about putting CAT3 on DSL |
| 06:00.42 | CCFL_Man2 | drmessano: i wasn't going to do it, i was saying it's what some people told me |
| 06:00.42 | drmessano | If you have that much noise on the line, start kicking over pedestals |
| 06:00.57 | drmessano | Find where the nearest Slick is |
| 06:01.13 | Strom_M | SLC |
| 06:01.16 | Strom_M | not slick |
| 06:01.28 | drmessano | Oh, who doesn't call it a "slick" |
| 06:01.29 | Strom_M | its just pronounced "slick" |
| 06:01.35 | drmessano | Actually |
| 06:01.38 | drmessano | It's an SLC-96 |
| 06:01.43 | Strom_M | yeah, and lets call it a scuzzy drive |
| 06:01.44 | drmessano | If you want to be specific |
| 06:02.02 | drmessano | Whatever |
| 06:02.12 | Strom_M | cocks |
| 06:02.36 | CCFL_Man2 | 14db noise margain on downstream |
| 06:02.57 | drmessano | CCFL_Man2: Do you know what those numbers mean? |
| 06:03.09 | CCFL_Man2 | yes, it needs to be higher |
| 06:03.39 | drmessano | Is that the SNR, and is that in sinad? |
| 06:04.00 | CCFL_Man2 | Noise Margin: 14.0 dB |
| 06:04.08 | drmessano | Is that the SNR, and is that in sinad? |
| 06:04.23 | CCFL_Man2 | i have no idea exactly what it is |
| 06:04.38 | drmessano | Exactly. |
| 06:05.12 | drmessano | You're trying to fix a problem that you're not even sure if it exists or where it exists |
| 06:05.13 | CCFL_Man2 | it is the snr |
| 06:05.20 | drmessano | If you have a problem, call the phone company |
| 06:06.02 | CCFL_Man2 | it needs to be high |
| 06:06.02 | drmessano | Figure out where the nearest, AHEM, SLC is.. and backtrace it to your house |
| 06:06.24 | drmessano | How high does it need to be? |
| 06:06.36 | CCFL_Man2 | enough so i don't get retrains |
| 06:06.47 | CCFL_Man2 | crc errors in atm packets are fine |
| 06:07.20 | CCFL_Man2 | i do understand signal to noise ratio |
| 06:07.42 | drmessano | Well, 14db is not unacceptable |
| 06:08.01 | drmessano | Do you know how much attenuation you have? |
| 06:08.05 | CCFL_Man2 | i know that |
| 06:08.13 | CCFL_Man2 | 31.5db |
| 06:08.23 | drmessano | Which is normal |
| 06:08.34 | drmessano | [02:05] <CCFL_Man2> it needs to be high |
| 06:08.48 | drmessano | Even though you agree 14db is acceptable |
| 06:08.49 | CCFL_Man2 | in bad weather retrains are often, like maybe once or twice in a day |
| 06:09.40 | drmessano | If you have an issue once or twice in a day during bad weather, I would be happy |
| 06:10.07 | CCFL_Man2 | i know, i'm such a perfectionist at this |
| 06:10.20 | drmessano | You're spinning your wheels |
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| 06:10.31 | CCFL_Man2 | when it's interleaved i have not a single retrain |
| 06:10.49 | CCFL_Man2 | but i don't want the latency |
| 06:11.50 | CCFL_Man2 | i also want to get a cisco 1841 |
| 06:11.51 | drmessano | If you can notice that much of a difference, you're more than a perfectionist... that's OCD |
| 06:12.06 | CCFL_Man2 | i notice retrains |
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| 06:12.26 | CCFL_Man2 | it doesn't happen often, but it happens at the worst times |
| 06:13.06 | drmessano | Alright.. well, good luck on getting some Cat3 in place or whatever it is you want to do.. I'm sure it will help a lot and be worth it. |
| 06:13.30 | CCFL_Man2 | i'm not replacing anything |
| 06:13.38 | drmessano | I'm gonna go shorten the cord on my fridge to see if it makes my drinks cooler |
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| 06:14.13 | CCFL_Man2 | you can't compare it to that |
| 06:14.28 | CCFL_Man2 | bad lines in the house can lead to dsl issues |
| 06:15.15 | Strom_M | CCFL_Man2: don't make me hit you over the head with the DSL book |
| 06:15.18 | drmessano | If you're having problems during bad weather |
| 06:15.22 | drmessano | Do you REALLY |
| 06:15.24 | drmessano | REALLY |
| 06:15.43 | drmessano | REALLY think the problem can be fixed INSIDE the house? |
| 06:15.57 | CCFL_Man2 | no, it cannot |
| 06:16.07 | Strom_M | MONSTER CABLE !!!!!!!!!!!!!!!!!!!!!!!!!! |
| 06:16.23 | drmessano | Do you REALLY think shortening a perfectly harmless ball of twisted pair is gonna throw your SNR through the roof? |
| 06:16.24 | drmessano | Come on.. |
| 06:16.45 | CCFL_Man2 | might change it by 1db |
| 06:16.49 | drmessano | ROFL |
| 06:16.55 | Strom_M | a whole dB |
| 06:16.56 | Strom_M | hahahahahhha |
| 06:16.58 | CCFL_Man2 | but thats nothing |
| 06:17.00 | drmessano | If you say so |
| 06:18.00 | Strom_M | go to the computer store and buy that $25 Belkin ADSL RJ-14 cable instead of the 99c satin-flex cord you have now |
| 06:18.02 | BBHoss | aren't most places on DLCs now? |
| 06:18.14 | CCFL_Man2 | actually everything is wired properly and neatly in the house |
| 06:18.14 | Strom_M | BBHoss: depends, but generally no |
| 06:18.16 | drmessano | I have run 6 T1 lines 3 feet from a 4000 watt AM transmitter with a 300 foot tower 50 feet away from it off one one of the messiest DMARC's you have ever seen |
| 06:18.20 | drmessano | and had few problems |
| 06:18.33 | BBHoss | thats all i've seen here |
| 06:18.46 | Strom_M | where's "here"? |
| 06:18.48 | CCFL_Man2 | T1s can take it |
| 06:18.50 | drmessano | I dont think your small length of wire is a DSL magic bullet |
| 06:18.53 | BBHoss | north alabama |
| 06:19.00 | BBHoss | huntsville |
| 06:19.04 | Strom_M | BBHoss: yeah, but thats mostly new construction |
| 06:19.07 | drmessano | CCFL_Man2: Whats the difference between a T1 and DSL? |
| 06:19.09 | drmessano | Oh wait.. |
| 06:19.35 | CCFL_Man2 | drmessano: actually T1s are send over hsdl nowadays |
| 06:19.47 | BBHoss | from the looks they run off of one or two pairs of SM fiber, no longer T1 lines |
| 06:19.50 | CCFL_Man2 | so there is no difference |
| 06:19.50 | drmessano | I forgot that SDSL changed the laws of physics that ADSL isn't bound to |
| 06:20.12 | drmessano | No, there isnt |
| 06:20.15 | BBHoss | CCFL_Man2, come again? t1 over dsl? |
| 06:20.21 | drmessano | [02:18] <CCFL_Man2> T1s can take it <--- holds no water |
| 06:20.39 | CCFL_Man2 | right |
| 06:20.50 | CCFL_Man2 | i forgot that |
| 06:20.51 | drmessano | Most T1s are over SDSL or HDSL |
| 06:21.23 | CCFL_Man2 | every single smart jack where i work are all hsdl smart jacks |
| 06:21.44 | drmessano | So again.. How is your little length of copper gonna change all that? |
| 06:21.51 | CCFL_Man2 | it won't |
| 06:21.53 | drmessano | You keep contradicting yourself |
| 06:22.03 | BBHoss | drmessano, but is it still a t1 "modem" at the customer location? |
| 06:22.05 | CCFL_Man2 | i'm learning now |
| 06:22.24 | drmessano | Theres a SmartJack that makes it all happen |
| 06:22.47 | drmessano | Pair of copper in, RJ-45 on the top/bottom |
| 06:22.51 | CCFL_Man2 | smart jacks are sexy looking |
| 06:23.19 | CCFL_Man2 | having one in an old garden shed makes your e-Penis huge |
| 06:24.09 | drmessano | I had fireants eating through pairs and pairs of copper before taking out a T1.. I would imagine the noise was horrid before it started dropping enough frames to actually down the line.. |
| 06:24.24 | drmessano | Most DSL can handle a good amount of noise.. its meant to |
| 06:24.27 | drmessano | Its Telco |
| 06:24.36 | drmessano | Old copper, stapled to wooden poles |
| 06:24.44 | drmessano | Dont forget that |
| 06:24.53 | CCFL_Man2 | drmessano: was it a DS1 or DSX1? |
| 06:24.58 | drmessano | It's not gigabit Ethernet |
| 06:25.10 | drmessano | DS1 |
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| 06:25.35 | CCFL_Man2 | shame the little bastards never got zapped |
| 06:27.37 | bobnormal | hi, i want to set up a way to have a program select a destination for a call (AMI or AGI) based on the current availability of agents logged in to a web-based system. should i look at AMI or AGI? and do i still need a queue? my goal is to have channels down unless a call is active, and queues seem to die if there's 'nobody listening' (ie: registered in the queue). whats the right approach? im happy to code agi or ami. |
| 06:27.51 | bobnormal | ideally also if there's no agents available the user can listen to on-hold music until one becomes available |
| 06:28.22 | CCFL_Man2 | bobnormal: the backstreet boys? |
| 06:28.30 | Strom_M | ...dude, you dont need AMI/AGI for that |
| 06:28.35 | drmessano | SSMTMFA |
| 06:28.48 | drmessano | Someone Stab Me, Too Many Freakin Acronyms |
| 06:29.12 | CCFL_Man2 | backstreet's back, ALRIGHT! |
| 06:30.15 | bobnormal | strom_m: how can i query a complex mysql state without ami/agi? its a multi-table thing, as it has to match operator language to incoming call language |
| 06:30.18 | CCFL_Man2 | would be an evil joke to put hansen on there |
| 06:30.34 | CCFL_Man2 | imagine how many people would mmmbop |
| 06:30.46 | Strom_M | bobnormal: sounds like youve got one massively overengineered solution there |
| 06:31.05 | CCFL_Man2 | Strom_M: i do that sometimes |
| 06:31.11 | bobnormal | strom_m: thanks but actually its required since the system is multinational and the database is already written |
| 06:31.28 | bobnormal | strom_m: could you tell me the best way to go about it? |
| 06:31.46 | CCFL_Man2 | the carbon microphone is dated november 1927 |
| 06:35.10 | bobnormal | err does anyone have some suggestions |
| 06:35.45 | Strom_M | bobnormal: wouldnt even begin to be able to tell you without a thorough analysis of how the system works |
| 06:37.00 | bobnormal | strom_m: ok so to simplify, assume you want to route a call in a 'queue-like-thing' based on some external program, while playing music to the remote side, and you might need to wait+retry in some cases, how do you do it |
| 06:37.52 | Strom_M | well, i could tell you how to do it entirely with Queue() |
| 06:38.20 | Strom_M | but it sounds like youve got so much complexity on top of that that I'd probably be wasting our time by doing so |
| 06:38.47 | bobnormal | i can easily write a simple program that returns a valid SIP address or nothing to route to, would that be good for Queue() use? |
| 06:38.55 | Strom_M | no |
| 06:39.32 | bobnormal | the thing is, i need to execute other code to initialise the telephonist's web interface when a call comes through |
| 06:39.38 | bobnormal | i think last time i looked at this, queue() couldn't let me do that |
| 06:39.57 | Strom_M | again, I think your question is way beyond the scope of #asterisk |
| 06:40.21 | Strom_M | i would recommend you talk to the engineer who put it together |
| 06:40.24 | bobnormal | i've sorted the initialise-browser thing out, i've got ajax polling going to the server |
| 06:40.25 | Strom_M | or hire a consultant |
| 06:40.34 | bobnormal | pfft thanks |
| 06:40.52 | bobnormal | probable case: you dont know complex real world situations well enough to answer |
| 06:41.39 | Strom_M | no, it's more that I don't know your specific complex real-world situation well enough to answer :) |
| 06:42.19 | bobnormal | ok so has anyone else here ever done web interface/asterisk integration for a call center |
| 06:42.22 | bobnormal | im sure its not uncommon |
| 06:45.42 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
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| 06:45.42 | *** mode/#asterisk [+o file] by irc.freenode.net |
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| 06:46.55 | CCFL_Man2 | Strom_M knows his shit |
| 06:49.15 | Strom_M | ive only been living and breathing telephony since 2001...asteisk since 2004 |
| 06:49.24 | Strom_M | s/asteisk/asterisk/ |
| 06:49.36 | Strom_M | a mere drop in the bucket, I'm sure |
| 06:50.41 | bobnormal | well if you live and breathe how come you cant give me a straight answer on what surely must be a relatively common use scenario |
| 06:50.57 | bobnormal | i think asterisk could go a lot further if this sort of thing was better documented |
| 06:51.09 | bobnormal | i guess there aren't many people that require this kind of functionality and use asterisk though |
| 06:51.28 | bobnormal | i guess probably almost all of the complex asterisk installs are controlled with a lot of custom code |
| 06:51.31 | bobnormal | would you agree? |
| 06:51.48 | *** join/#asterisk stormix (i=scream@mailserver.shebanet.net) |
| 06:51.51 | bobnormal | i mean almost any non-trivial application with database interaction |
| 06:52.11 | bobnormal | therefore people don't share it as they consider the code commercially significant |
| 06:52.31 | bobnormal | stops guessing and gets back to trying to find a decent queues explanation in the asterisk tfot book |
| 06:53.25 | Strom_M | bobnormal: because you've provided so little information about the architecture of your custom system that any of my guesses at this point would likely not even begin to approach the scope of the answer that you actually need |
| 06:54.46 | bobnormal | strom_m: basically i just want to know, "whats the easiest way to run a custom piece of code to decide where to send a user who is listening to music, to leave them listening to music if that returns false, to retry periodically, and to run another bit of code in the case that a connection is made" ... what's the magic mix of AMI/AGI/queues? |
| 06:55.10 | Strom_M | queue() can do that all by itself, if I understand you correctly |
| 06:56.05 | Strom_M | but as soon as I started to try to explain it, you shot back with all of your database / ajax / ami / agi jiggerypokery, which convinces me that you don't really want an answer. |
| 06:57.34 | *** join/#asterisk Sparc__ (n=chatzill@c-76-126-162-83.hsd1.ca.comcast.net) |
| 06:58.06 | Strom_M | that's the kind of nonsense I'm willing to put up with as a paid consultant, but for free? forget it. |
| 06:58.20 | Sparc__ | does asterisk realtime capture the unique_id anywhere in a table? |
| 06:58.58 | Sparc__ | i'd like to query a database by unique_id to see if an action sent via ami was successful or not..and if so, what is it's current state |
| 06:59.17 | bobnormal | strom_m: looking at the queue() definition in TFOT, it states lost of people "report that their solutions required customisation", and the documentation is a little opaque when it comes to running code when a queue agent / user are connected. i think this requires AMI. |
| 07:08.36 | *** join/#asterisk LuisTorres (n=chatzill@bl6-207-60.dsl.telepac.pt) |
| 07:12.06 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 07:13.33 | yang | ~portech |
| 07:13.41 | yang | ~poortech |
| 07:18.55 | CCFL_Man2 | anyone want to do a test sip call to me? |
| 07:19.12 | *** join/#asterisk sarmisak (n=omer@212.156.220.171) |
| 07:19.59 | yang | CCFL_Man2: what is your number |
| 07:20.40 | *** part/#asterisk sarmisak (n=omer@212.156.220.171) |
| 07:24.43 | stormix | can any one provide me some good documentation on AGI script? |
| 07:25.56 | bobnormal | stormix: there's a chapter on it for php/perl AGI in asterisk - the future of telephony (free pdf download) |
| 07:26.22 | stormix | thanks |
| 07:26.44 | Strom_M | ~thebook |
| 07:26.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 07:26.48 | bobnormal | stormix: you might need AMI instead though, depending on whether you just want to launch from the dialplan or whether you want event monitoring or something |
| 07:33.07 | *** join/#asterisk tamato (n=Marius@41-195-68-70.access.uunet.co.za) |
| 07:33.21 | tamato | hi all |
| 07:34.02 | tamato | anyone here? |
| 07:34.08 | bobnormal | yep |
| 07:34.17 | tamato | oh kewl |
| 07:34.32 | tamato | im new here first time on irc |
| 07:34.41 | bobnormal | haha :) welcome |
| 07:35.01 | tamato | a bit of a mission to get connected but got it now:) |
| 07:35.01 | yang | tamato: I have to warn, it could be addictive |
| 07:35.08 | tamato | lol |
| 07:35.17 | bobnormal | tamato: you sound like a bot :) |
| 07:35.28 | tamato | huh? |
| 07:35.40 | tamato | why is that? |
| 07:35.55 | bobnormal | some people have strange sense of humour |
| 07:36.14 | tamato | ok then |
| 07:36.43 | tamato | can i ask anything here? about asterisk |
| 07:37.21 | yang | yep |
| 07:38.05 | gr0mit | tamato, asterisk comes with a Health Warning. |
| 07:38.26 | gr0mit | as yang says - it is Highly Addictive |
| 07:39.32 | tamato | lolo |
| 07:39.42 | tamato | letme tel my story |
| 07:40.01 | tamato | ive sucsessfully setup a asterisk box and is working |
| 07:40.33 | tamato | ive even setup sip extensions for remote sites and it works aswell so happy |
| 07:41.10 | tamato | anyway i want to know how do you actually save money |
| 07:41.28 | tamato | if you still need to use the pstn line for dialing out |
| 07:42.08 | tamato | i cant just start one huge network and connecting everyone to asterisk to phone each other |
| 07:42.25 | yang | tamato: there are a bunch of voip providers, which can give you cheaper rates for dialing out |
| 07:42.41 | yang | and you can also get numbers from them |
| 07:42.49 | tamato | but how do you do it yourself |
| 07:42.57 | tamato | thats the big quistion |
| 07:43.03 | stormix | bobnormal: i want create an IVR and extract data from database depending on user key press |
| 07:44.35 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9) |
| 07:46.16 | tamato | stormix look at this http://www.linuxbsd.com.br/guias_livros/AsteriskTFOT.pdf |
| 07:46.59 | stormix | thanks tomato |
| 07:47.10 | tamato | yang: dont voip providers go through pstn lines aswell |
| 07:47.17 | tamato | stormix: np |
| 07:48.11 | tamato | or let me ask this is there anyone here from south africa?? |
| 07:48.33 | Strom_M | tamato: well, let me put it this way: I'm no stranger to Mrs. Ball's chutney |
| 07:48.36 | JT | voip providers can get better rates than you |
| 07:48.49 | tamato | hehehehe |
| 07:49.14 | stormix | i already have that book but i am looking for more AGI scripting help with example |
| 07:49.39 | tamato | i know there is alot of providers but who do i use |
| 07:49.42 | Strom_M | tamato: but, basically, you're never going to get "free" telephone service, as all phone calls to and from the PSTN will incur some sort of charge |
| 07:50.37 | tamato | i know that |
| 07:51.02 | tamato | strom_M: do you work for a voip provider? |
| 07:51.12 | Strom_M | no. I'm an independent consultant |
| 07:51.50 | tamato | just making sure because i came acros telfree but they dont want me to use asterisk |
| 07:52.08 | Strom_M | they're a voip provider? |
| 07:52.28 | tamato | jip i met the ceo of the company |
| 07:52.50 | Strom_M | "jip"? |
| 07:52.51 | tamato | but they limit you so much and said that they are gonna be the biggest in sa |
| 07:52.59 | Strom_M | yeah, screw them |
| 07:53.10 | Strom_M | find a real ITSP :) |
| 07:53.16 | tamato | i like flexability |
| 07:53.27 | tamato | in SA im not sure about that |
| 07:54.20 | tamato | btw jip = yes or ya whatever |
| 07:54.43 | Strom_M | i only know about takkies and bokkies |
| 07:54.47 | JT | no-one knows what jip mean |
| 07:55.04 | tamato | lol sorry guys |
| 07:55.42 | tamato | anyway who can i use then |
| 07:56.03 | tamato | for local calls and international |
| 07:56.09 | Strom_M | beats me; i dont know the south african ITSP market well at all |
| 07:56.48 | tamato | because it will get complicated to use a overseas company |
| 07:57.04 | tamato | with the exchange rates ant things |
| 07:59.11 | tamato | what i need is a company that could let me connect via iax with good prices to calls |
| 07:59.28 | tamato | local and international |
| 08:00.29 | JT | sip is fine |
| 08:01.13 | tamato | ok or even sip but just for dialing to pstn |
| 08:01.50 | tamato | if i google there is so many but who is reliable and dont offer prepaid |
| 08:05.24 | tamato | anyone know of a ITSP |
| 08:06.49 | *** join/#asterisk E-bola2 (n=sdf@87.63.94.170) |
| 08:07.20 | Strom_M | BREAKFAST CANCER |
| 08:07.45 | tamato | that is ? |
| 08:09.01 | E-bola2 | When our receptionist does blind xfer's, sometimes she realises the person isnt there, if she does an xfer and nobody picks up, the call returns to her phone after 30 seconds. But if she realises she's amde a mistake and want to pull back the call actively, how could i do that? (Im using newest 1.4 and a snom360) |
| 08:09.54 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 08:28.33 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:39.30 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:42.15 | LuisTorres | Hi..., Im getting loads of Dial("SIP/anonymous.invalid- ) do you know what could cause this? |
| 08:48.17 | yang | Why is the sip show peers qualify time longer than the ping response to that device? |
| 08:49.43 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 08:53.57 | Curus | yang: The phone has to actually process the request |
| 08:54.47 | Curus | And Asterisk has made an unfortunate choice of request to do it. Processing that request according to the RFC is as expensive as handling a Dial |
| 08:56.30 | yang | Anyone here uses the Grandstream phones? |
| 08:57.28 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:20.17 | *** join/#asterisk genius_boy2006 (n=genius_b@85.185.91.3) |
| 09:20.32 | genius_boy2006 | hi |
| 09:20.40 | genius_boy2006 | iam new to asterisk |
| 09:20.47 | genius_boy2006 | can anyone help me ? |
| 09:21.44 | genius_boy2006 | Çhello? |
| 09:21.51 | km2 | ~thebook |
| 09:21.52 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 09:22.11 | genius_boy2006 | well |
| 09:22.17 | genius_boy2006 | i have installed the Asterisk now |
| 09:22.33 | genius_boy2006 | it works fine in my local network |
| 09:22.45 | genius_boy2006 | ( i have just changed the ip address ) |
| 09:22.54 | genius_boy2006 | configured some users and so |
| 09:22.58 | genius_boy2006 | but the problem is |
| 09:23.23 | genius_boy2006 | when i assign it a Valid ip |
| 09:23.30 | genius_boy2006 | none of my devices register |
| 09:24.45 | genius_boy2006 | it is a Asterisk now |
| 09:24.49 | km2 | after you changed the server's ip address? |
| 09:24.53 | genius_boy2006 | yes |
| 09:24.56 | genius_boy2006 | before that |
| 09:25.00 | genius_boy2006 | it was 192.168 |
| 09:25.11 | genius_boy2006 | but after i change it to a Valid ip |
| 09:25.28 | genius_boy2006 | i thought i could connect anydevice from internet to it right ? |
| 09:25.56 | km2 | the phones probably still have private ips |
| 09:26.14 | genius_boy2006 | the only thing is i havent change anything from the default setting |
| 09:26.19 | E-bola2 | wonders why some extensions are yellow in the flash operator panel |
| 09:26.26 | km2 | e.g. 192.168.* |
| 09:26.37 | genius_boy2006 | well yes they have |
| 09:26.43 | genius_boy2006 | but they are behind a NAT server |
| 09:26.47 | genius_boy2006 | which works fien |
| 09:27.21 | genius_boy2006 | and i think they should be able to communicate |
| 09:27.47 | genius_boy2006 | i dont know if i should change anything for that ? |
| 09:27.50 | genius_boy2006 | i mean the Server |
| 09:28.09 | genius_boy2006 | in the webconfiguration i could change many things |
| 09:28.27 | km2 | SIP and NAT do not get along well. if you want your PBX on the internet, your best bet is to get another network card. give one a public IP and one a private LAN IP |
| 09:28.58 | km2 | see http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 09:29.20 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9f7496dff27b67d7) |
| 09:30.25 | genius_boy2006 | so Asterisk server with the default configuration and a Valid Ip address can serve all the internet without any problem? |
| 09:31.03 | genius_boy2006 | and what should i put in Outbond and Inbond proxy setting ? |
| 09:31.12 | genius_boy2006 | the valid ip i assigned to the server ? |
| 09:32.38 | km2 | i'm not sure about that, but i can say that usually "local" phones/devices register through the asterisk server's internal ip |
| 09:33.17 | genius_boy2006 | and how about using it globally |
| 09:33.32 | genius_boy2006 | i mean having a asterisk server with |
| 09:34.00 | genius_boy2006 | a valid ip is enough for me to be a server for as many Sip clients as i want ? |
| 09:34.16 | genius_boy2006 | no Proxy servers and stuff needed? |
| 09:36.15 | km2 | yes, SIP clients can register from anywhere, but due to complications between SIP and NAT, often you can only have one SIP device behind a NAT device. if you have more than one SIP device behind NAT, they will all want to use the same port which confuses things |
| 09:36.20 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 09:36.47 | km2 | the IAX protocol works around this limitation because it can multiplex a bunch of devices through one firewall port, which SIP cannot do |
| 09:37.46 | km2 | however, not many phones support IAX (yet?) |
| 09:38.35 | km2 | our network includes a bunch of retail stores which all have phones behind NAT, but they're all part of a big VPN, which is another way to sidestep the NAT issue |
| 09:39.49 | genius_boy2006 | well theoritically they should work fine |
| 09:40.32 | genius_boy2006 | since the source port remains the NAT server |
| 09:40.54 | *** join/#asterisk tapirkopi (n=denmasbo@202.154.57.15) |
| 09:40.57 | genius_boy2006 | and right now i am using more than 4 devices behind a Local natserver |
| 09:41.12 | tapirkopi | hello |
| 09:41.20 | genius_boy2006 | they are connected to a |
| 09:41.25 | tapirkopi | is there steve ? |
| 09:41.28 | genius_boy2006 | high end |
| 09:41.30 | genius_boy2006 | server |
| 09:41.39 | genius_boy2006 | i dont know what it is |
| 09:41.43 | genius_boy2006 | but really works fine |
| 09:41.52 | genius_boy2006 | i wanted to do the same with asterisk |
| 09:42.11 | genius_boy2006 | but it really bugs me |
| 09:42.25 | genius_boy2006 | i dont know where did i go wrong |
| 09:42.45 | genius_boy2006 | i have 3 computers |
| 09:43.01 | genius_boy2006 | having same invalid ip |
| 09:43.07 | genius_boy2006 | 192.168.*.* |
| 09:43.10 | km2 | i won't claim to be an expert, but i will say you're not the first to be frustrated by SIP and NAT together |
| 09:43.31 | genius_boy2006 | Locally they work fine |
| 09:43.49 | genius_boy2006 | when my asterisk Ip is 192.168.*.* |
| 09:43.56 | genius_boy2006 | they all work smoothly |
| 09:44.09 | genius_boy2006 | but when i change the server ip to Valid ip |
| 09:44.22 | genius_boy2006 | none of my devices can register |
| 09:44.38 | genius_boy2006 | <PROTECTED> |
| 09:44.50 | genius_boy2006 | i thought maybe there is another setting |
| 09:44.54 | genius_boy2006 | that i should change |
| 09:45.01 | genius_boy2006 | is there really ? |
| 09:46.25 | *** join/#asterisk tuxx- (n=tuxx@nightshade.nl) |
| 09:47.10 | tuxx- | Hello, anyone have any information about how to setup asterisk with an AVM fritz ISDN card? Can't find much on the net.. |
| 09:47.42 | genius_boy2006 | ok Thanx KM2 |
| 09:47.45 | genius_boy2006 | i have to go |
| 09:47.55 | genius_boy2006 | youve been helpful really |
| 09:48.04 | patrick-- | Hey, im currently receiving faxes with app_rxfax of spandsp. sometimes the faxes are not received properly and then cannot be opened. is there anything you guys can think of to fix this ? |
| 09:48.09 | km2 | SIP packets contain addressing information, but NAT doesn't rewrite this info because it's not that smart (through some firewalls are "SIP-aware" now). so the SIP packets that reach your public asterisk server say "communicate with me back at 192.168.*.*" but the asterisk box, sitting on the internet, has no way of getting data back to that address. NAT handles readdressing the outer layer but doesn't drill down into SIP |
| 09:48.56 | km2 | google "sip nat" for lots more info |
| 09:49.05 | genius_boy2006 | Ok |
| 09:49.08 | genius_boy2006 | Thanx alot |
| 09:49.19 | genius_boy2006 | i think i need to search more |
| 09:49.23 | genius_boy2006 | and yes |
| 09:49.26 | km2 | the info is out there |
| 09:49.45 | genius_boy2006 | i am now beggining to inderstand |
| 09:49.54 | genius_boy2006 | ok thanx |
| 09:49.59 | km2 | you're welcome |
| 09:50.09 | genius_boy2006 | really appriciate it |
| 09:50.24 | genius_boy2006 | do you have any msn or sth for |
| 09:50.40 | km2 | any what? |
| 09:50.45 | genius_boy2006 | Messenger |
| 09:50.54 | genius_boy2006 | as a matter of fact i am new in this IRc |
| 09:51.06 | genius_boy2006 | is there any kinda friend list i can add you to it? |
| 09:51.20 | km2 | msn: bobby@recchiuti.com, aim: cellsdividing, irc: km2 |
| 09:51.46 | genius_boy2006 | genius_boy2006@yahoo.com |
| 09:51.59 | genius_boy2006 | thank you for everything |
| 09:52.06 | genius_boy2006 | hope to get in touch later |
| 09:52.12 | genius_boy2006 | <PROTECTED> |
| 09:52.20 | genius_boy2006 | bye =; |
| 09:54.48 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 09:55.00 | casix | hello |
| 09:57.51 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 09:58.50 | patrick-- | Hey, im currently receiving faxes with app_rxfax of spandsp. sometimes the faxes are not received properly and then cannot be opened. is there anything you guys can think of to fix this ? |
| 10:00.27 | tuxx- | Hello, anyone have any information about how to setup asterisk with an AVM fritz ISDN card? Can't find much on the net.. |
| 10:15.19 | *** join/#asterisk Q3Man (n=Steve@wsip-24-234-155-62.lv.lv.cox.net) |
| 10:17.09 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:27.20 | *** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net) |
| 10:27.30 | *** join/#asterisk shpaq (i=shpaq@gentoo/user/shpaq) |
| 10:29.04 | shpaq | does somebody can recommend me a good/cheap voip operator which provides uk inbound numbers? |
| 10:29.40 | shpaq | i don't have any experience and i just don't know what i'm suppose to search for |
| 10:30.57 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:32.37 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:35.05 | DarnoQ | start searching from here: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential |
| 10:35.38 | Rico29 | can somebody help me for cepstral+asterisk |
| 10:35.48 | shpaq | DarnoQ: thx |
| 10:35.53 | Rico29 | i'm trying to make it working for 4h... |
| 10:36.11 | Rico29 | and I get an error while compiling asterisk |
| 10:36.24 | Rico29 | i followed http://www.automated.it/asterisk/ |
| 10:36.34 | tzafrir | http://paste.debian.net/3758/ <-- pointless tests in qozap ? |
| 10:36.53 | shpaq | DarnoQ: i need also recommendation |
| 10:36.57 | tzafrir | Rico29, could you please pastebin that error? |
| 10:37.02 | Rico29 | yes |
| 10:37.03 | Rico29 | sure |
| 10:37.07 | tzafrir | I think that the text you follow is slightly dated |
| 10:38.16 | Rico29 | http://debian.pastebin.com/meb82ab8 |
| 10:38.34 | Rico29 | i put app_cepstral.c in the app directory of asterisk source |
| 10:38.46 | Rico29 | and add lines to apps/Makefile |
| 10:39.17 | tzafrir | # |
| 10:39.17 | tzafrir | In file included from app_cepstral.c:15: |
| 10:39.17 | tzafrir | # |
| 10:39.17 | tzafrir | /home/asterisk/sources/asterisk-1.4.19.1/include/asterisk/file.h:27:2: error: #error You must include stdio.h before file.h! |
| 10:39.17 | tzafrir | # |
| 10:39.19 | tzafrir | app_cepstral.c:25:25: error: ../asterisk.h: Aucun fichier ou répertoire de ce type |
| 10:39.21 | tzafrir | # |
| 10:39.25 | tzafrir | oops, sorry |
| 10:39.32 | Rico29 | :) |
| 10:39.48 | tzafrir | is that app_cepstral.c ready for asterisk 1.4? |
| 10:40.06 | Rico29 | i hope... |
| 10:40.15 | Rico29 | that's the only one i found |
| 10:41.17 | Rico29 | if anyone have one which works on 1.4... |
| 10:46.12 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 10:47.22 | *** join/#asterisk incorrect (n=incorrec@212.44.62.238) |
| 10:50.17 | DarnoQ | shpaq: can't help you there, you can try at - http://groups.google.com/group/uk.telecom.voip/topics?hl=pl&lnk=gschg |
| 10:51.17 | shpaq | DarnoQ: thx again |
| 10:57.36 | tzafrir | Rico29, on what system do you build it? |
| 10:57.44 | Rico29 | debian 4 |
| 10:57.54 | *** join/#asterisk liri (n=liran@bzq-179-128-126.static.bezeqint.net) |
| 11:01.47 | disposable | i have a strange problem. a number of our clients with polycom phones (ip430 and ip650 with sip2.2.0) started having incoming sip phonecalls from sip:5199362832664@xxx.xxx.xxx.xxx (the ip address being phone's own ip). when they pick up the receiver, it's silent (ip650) or freezes the phone (ip430). the phonecalls don't even appear in our logs. just one (out of more than 20): |
| 11:02.01 | disposable | "multiplay","7018","5199362832664","multiplay_trusted","""Stewart Fletcher"" <7018>","SIP/multiplay_ip601a-28c4","","Hangup","","2007-10-05 09:34:08","2007-10-05 09:34:08","2007-10-05 09:34:10",2,2,"ANSWERED","DOCUMENTATION" |
| 11:02.09 | disposable | has anyone seen anything like it? |
| 11:02.15 | *** join/#asterisk stuffcorpse (n=pickle@60-234-225-122.bitstream.orcon.net.nz) |
| 11:04.43 | Rico29 | tzafrir> do you have a correct app_cepstral.c file ? |
| 11:05.01 | tzafrir | no |
| 11:05.42 | tzafrir | Rico29, do you use debs? if so, you may need to provide a symlink for asterisk.h |
| 11:05.53 | Rico29 | k |
| 11:12.01 | Superbartt | I'm looking for the following solution: I want my current ISDN lines to connect to my server with asteriks and let incoming and outgoing calls go over that line, is that even possible with asteriks, and what hardware should i be looking for? |
| 11:13.18 | Rico29 | tzafrir> i get the same error, excluding asterisk.h error |
| 11:16.43 | *** part/#asterisk shpaq (i=shpaq@gentoo/user/shpaq) |
| 11:16.54 | Rico29 | :( |
| 11:18.17 | *** join/#asterisk xcsuperfly (n=dtortora@static-96-234-33-9.nwrknj.fios.verizon.net) |
| 11:19.08 | *** join/#asterisk shpaq (i=shpaq@gentoo/user/shpaq) |
| 11:19.57 | shpaq | does somebody has any expieriences with http://www.transcom.net/index.html |
| 11:24.21 | *** join/#asterisk Zyna (n=IceChat7@p54BCD847.dip.t-dialin.net) |
| 11:24.24 | Zyna | hi folks |
| 11:24.34 | *** join/#asterisk talntid (n=eric@c-67-185-237-158.hsd1.wa.comcast.net) |
| 11:25.21 | Zyna | Could somebody give me a quick hand please? This is what I got, but how do I now tell teh dialplan to establish outbound calls? |
| 11:25.24 | Zyna | http://pastebin.com/d1d685a60 |
| 11:26.44 | Zyna | anybody? |
| 11:26.46 | Zyna | please! |
| 11:27.07 | Zyna | there's gotta be just a simple exten => instruction to dial external numbers...? |
| 11:28.42 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:33.05 | Rico29 | http://debian.pastebin.com/m5ae93534 |
| 11:34.02 | Rico29 | nobody can help me ? with cepstral+asterisk ? |
| 11:38.51 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-230-250.unitz.ca) |
| 11:52.30 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 12:00.18 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 12:00.43 | synthetiq | are agi perl scripts meant to only excute one asterisk command or multiple commands? |
| 12:03.47 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:03.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:05.07 | synthetiq | ? |
| 12:07.56 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:11.39 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 12:12.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:12.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:12.56 | Vec | Hi, what kind of problems would I could I expect to experiance if I don't have ztdummy installed on a asterisk box that does not have any zaptel hardware ? |
| 12:13.41 | smps | Vec, there would be no problems at all |
| 12:14.12 | Oy90 | Vec: MeetMe conferencing requires ztdummy |
| 12:14.19 | tzafrir | syntaxman, they can execute multiple commands |
| 12:14.31 | lmadsen | Vec: you couldn't use meetme or iax2 trunking |
| 12:14.39 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:14.57 | lmadsen | Vec: and if you hadn't built asterisk with it prior, you wouldn't even get the modules built that required timing |
| 12:15.15 | lmadsen | now wonders why you wouldn't install a timing device anyways... |
| 12:16.42 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 12:17.22 | Vec | Thanks all |
| 12:17.42 | *** join/#asterisk blinky42 (n=steveb@67.106.175.146.ptr.us.xo.net) |
| 12:18.09 | Curus | It's a bit sad that Asterisk doesn't just use the timing that the kernel can provide these days |
| 12:18.31 | Curus | I guess I should convince management to put up a bounty |
| 12:21.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:21.33 | *** join/#asterisk Weasel[DK] (n=Weasel[D@0x503fbea6.kd4nxx16.adsl-dhcp.tele.dk) |
| 12:21.51 | Vec | Curus : I guess ztdummy is a convoluted way of using the timing that the kernel provides |
| 12:22.46 | Weasel[DK] | how do i make asterisk play .wav files... ? running Ubuntu 8.04 and getting this -> (format 0x4 (ulaw)): No such file or directory |
| 12:22.58 | Weasel[DK] | trust me the file does exist |
| 12:23.25 | Curus | Vec: Yes, but that doesn't work in a virtual server |
| 12:23.28 | [TK]D-Fender | Weasel[DK]: Show us in full gory detail |
| 12:23.39 | [TK]D-Fender | ~pb |
| 12:23.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:23.41 | [TK]D-Fender | ^^^^^ |
| 12:24.08 | Rico29 | need help for that please : http://debian.pastebin.com/d1d8b6646 |
| 12:24.16 | Weasel[DK] | [TK]D-Fender, yes sir... right away |
| 12:26.02 | Weasel[DK] | [TK]D-Fender, here we go -> http://pastebin.com/dc2d314b |
| 12:26.10 | *** join/#asterisk deStone (n=cclayton@unaffiliated/destone) |
| 12:27.17 | Weasel[DK] | does the player depend on sox ? or is it sox ??? |
| 12:27.34 | Rico29 | need help for that please : http://debian.pastebin.com/d1d8b6646 |
| 12:27.55 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 12:29.12 | [TK]D-Fender | Weasel[DK]: [May 22 14:25:13] WARNING[8237]: format_wav.c:156 check_header: Unexpected freqency 44100 |
| 12:29.24 | [TK]D-Fender | Weasel[DK]: You have not converted it to settings * can read. |
| 12:29.32 | [TK]D-Fender | Weasel[DK]: must be 8khz mono <- |
| 12:29.42 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 12:29.47 | [TK]D-Fender | Weasel[DK]: You don't jsut feed it CD quality settings |
| 12:31.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:31.42 | Weasel[DK] | [TK]D-Fender, ok, i tried to change it with sox but the default package on Ubuntu 8.04 does not support very much it seems |
| 12:32.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 12:32.43 | Weasel[DK] | [TK]D-Fender, i could try sox on another machine... |
| 12:33.13 | [TK]D-Fender | Weasel[DK]: go get audacity. Think that'll do it. |
| 12:33.30 | *** join/#asterisk havokb8 (n=havokown@c-71-56-193-205.hsd1.co.comcast.net) |
| 12:33.33 | havokb8 | omfg |
| 12:33.39 | *** part/#asterisk havokb8 (n=havokown@c-71-56-193-205.hsd1.co.comcast.net) |
| 12:34.04 | russellb | o.O |
| 12:34.05 | russellb | that was random |
| 12:34.39 | Curus | I actually have wav files in the right format which Asterisk won't play. They have some kind of author header |
| 12:34.52 | Curus | sox file.wav file-stripped.wav "fixes" it |
| 12:36.09 | Oy90 | Weasel[DK]: apt-get install libsox* should help to get sox working in hardy |
| 12:37.04 | deStone | anyone using an asterisk / vmware type solution for redundancy? |
| 12:38.37 | Weasel[DK] | Oy90, wow what a bunch.... thx |
| 12:38.44 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:42.13 | dustybin | anyone here use a cellphone on a wireless network what hooks up with asterisk? |
| 12:42.54 | *** part/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 12:48.25 | Curus | Yes |
| 12:49.01 | Curus | dustybin: I can't recommend it |
| 12:49.28 | dustybin | why not |
| 12:50.12 | Curus | Well you can possibly get it to work at home |
| 12:51.18 | Curus | But I haven't found something that works reliably. E.g. I can get my E70 to work at home on my Linksys router, but at work with a fancy Aruba network or with a HP network, no go |
| 12:51.42 | Curus | Then sometimes outgoing calls won't work, but an incoming call will unstick it and make outgoing work again |
| 12:51.56 | Curus | And forget roaming between access points |
| 12:53.28 | Curus | Dedicated WLAN phones are more reliable, but worse in other ways |
| 12:53.35 | Curus | DECT works |
| 12:54.23 | tzanger | panasonic has a great dect phone now |
| 12:54.29 | tzanger | $100 at future shop though, sucks |
| 12:54.43 | Curus | Does it do SIP on the base? |
| 12:54.45 | tzanger | tehre's an interesting vtech dect phone too.. fully keyboard on it |
| 12:54.47 | tzanger | Curus: no |
| 12:54.49 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:54.51 | tzanger | I'm looking for something that does that |
| 12:54.59 | Curus | Ah ok, there are a ton of good DECT analog phones |
| 12:54.59 | dustybin | hmm ok interesting |
| 12:55.07 | Curus | Not so many good DECT SIP phones |
| 12:55.08 | tzanger | the vtech dect one has usb, I just can't justify buying it to experiment |
| 12:55.14 | dustybin | so cellphone + WLAN = gimmick |
| 12:55.16 | tzanger | Curus: I like aastra's offering |
| 12:55.35 | Curus | We tend to use Kirk base stations with Siemens or Kirk handsets |
| 12:55.38 | tzanger | except the actual hardware didn't implement proper ringtones |
| 12:55.43 | [TK]D-Fender | Aastra's is good if you don't mind them being tied at the hip |
| 12:55.50 | tzanger | "tied at the hip" ? |
| 12:56.13 | Curus | For home use I'd probably go with one of the Siemens SIP DECT phones |
| 12:56.26 | tzanger | All I want's a fucking dect sip solution that has a few soft buttons and a polycom-like xml running screen, decent ringtones, vibrate and bluetooth. that's it. why's that so fucking hard? |
| 12:57.13 | Curus | No XML or soft buttons, but otherwise there's a Siemens phone which does the rest. Sound quality is sadly not so good on that particular model |
| 12:57.40 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:57.56 | hsv-al | #asterisknow is dead, but how do you set that box in the CLI a static iP? |
| 12:58.07 | hsv-al | its auto-taking 192.168.1.1 |
| 12:59.06 | Corydon76-dig | You don't set the IP via the CLI |
| 12:59.48 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:00.54 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:01.38 | hsv-al | i tried finding dox online, but nothing states where to set this manually, the menu selector doesnt have the option either. |
| 13:01.54 | maqr | can someone clue me in as to how to set my outbound callerid? i've tried a few things, none of them seem to work though. my provider is vitelity. |
| 13:02.28 | Curus | maqr: Your provider probably overrides whatever you're sending |
| 13:02.31 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 13:02.43 | Curus | We do, at least |
| 13:03.13 | maqr | Curus: even for the number? it just always says 'unknown' |
| 13:03.38 | dustybin | <PROTECTED> |
| 13:04.05 | Curus | Yes, especially for the number |
| 13:04.07 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 13:04.58 | Curus | But they should show the number you want if you ask them to |
| 13:05.34 | file | dustybin: O.o do you mean voice or data... |
| 13:05.40 | dustybin | data |
| 13:06.06 | file | GPRS, 1XRTT, EVDO, EDGE, UMTS, HSDPA |
| 13:06.07 | maqr | Curus: oh wait, i was just using the wrong format in asterisk |
| 13:06.18 | maqr | Curus: if i just set number, it doesn't work, but if i set name/number, it does |
| 13:06.23 | dustybin | aye ok thanks |
| 13:06.35 | Curus | Ah excellent, maqr |
| 13:07.14 | maqr | Curus: that's kinda freaky that i can just spoof ANI like that |
| 13:07.24 | maqr | you'd think that'd be more tightly controlled |
| 13:07.53 | Curus | In some countries it's more regulated than others |
| 13:08.17 | rob0 | hsv-al, that one truly is an #asterisknow specific question, because the IP address[es] are set and managed by the OS. You could override it with ifconfig(8) or ip(8) (ip addr help) commands, but that only works until you reboot. |
| 13:08.27 | coppice | ANI is hard to spoof, but in some countries CLI is trivial to spoof |
| 13:09.18 | maqr | CLI is different than ANI? |
| 13:09.27 | maqr | oh, ANI includes the ability to set the name? |
| 13:09.31 | coppice | yes. odd, isn't it |
| 13:10.02 | maqr | well, whatever it is that i can set, i'm surprised it's so easy to set it |
| 13:10.46 | Curus | We filter the caller ID's of our customers, and our providers filter us in turn |
| 13:11.17 | coppice | in many countries you can only set it to one of the numbers you've been allocated. in some places (e.g. north america) they lock down ANI, but let you stuff anything in the originating number/CLI field |
| 13:11.34 | maqr | coppice: so i'm really setting CLI? |
| 13:12.09 | coppice | it would appear so, though the naming can be a bit vague in some places |
| 13:12.31 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 13:13.55 | *** join/#asterisk robevans (n=robevans@OL6-231.fibertel.com.ar) |
| 13:14.12 | maqr | interesting |
| 13:15.57 | *** join/#asterisk railsmunky (n=nick@82-70-72-101.dsl.in-addr.zen.co.uk) |
| 13:17.48 | railsmunky | hey people is there a way to configure asterisk to ring extensions in order one after another BUT pickup where it left off on the order so [call 1] ring 1 -> 2 -> 3 then [call2] ring 3 -> 4 -> 1 -> 2 ? |
| 13:17.56 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 13:19.00 | [TK]D-Fender | railsmunky: How many times would a call go around? |
| 13:20.00 | railsmunky | I guess with that config indefinitely or unless there's some way of dropping out after 4 iterations for example |
| 13:22.50 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 13:22.56 | *** part/#asterisk shpaq (i=shpaq@gentoo/user/shpaq) |
| 13:23.08 | [TK]D-Fender | railsmunky: Go setup a queue then with strategy "rrmemory" |
| 13:23.58 | railsmunky | aha of course. Thanks. [TK]D-Fender |
| 13:24.15 | hsv-al | d-fender, do you have any syringes? |
| 13:24.26 | hsv-al | I have some high potency rocket fuel cofee, for you to inject your veins with :) |
| 13:26.26 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 13:31.44 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 13:32.12 | Faustov | hi, could anyone recommend a SIP provider that supports uk inbound numbers? |
| 13:33.05 | eric2 | Got SIP response 486 "Busy Here" my pri provider is giving me this message... any reasons for this? |
| 13:33.49 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 13:33.59 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-0729398bbe8c7f5f) |
| 13:34.58 | rcahilig | guys please help me, my queue is ringing instead of playing MOH |
| 13:35.21 | iamhrh | are yoiu using the 'r' option? |
| 13:36.41 | rcahilig | yes |
| 13:36.47 | iamhrh | well stop |
| 13:36.48 | [TK]D-Fender | Faustov: Some of my customers have used Gradwell |
| 13:37.51 | rcahilig | iamhrh: I tested it locally and its playing MOH |
| 13:38.11 | Faustov | [TK]D-Fender: thanks, checking |
| 13:38.43 | iamhrh | rcahilig: do you have the relevant extension.conf information posted? (pastebin.ca) |
| 13:40.45 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-6338a480458b13bc) |
| 13:40.49 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:40.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:41.13 | iamhrh | i'm playing with the Pickup() application, trying to make an extension that I can dial that will just ring indefinately, and then another extension that I can call from another phone to get pick it up... this is what I was trying (http://pastebin.ca/1025872), but it doesn't work. anyone have a suggestion? |
| 13:42.52 | rcahilig | here is my queue in extensions.conf http://www.pastebin.ca/1025873 |
| 13:43.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:43.14 | [TK]D-Fender | iamhrh: How are you supposed to know to pickup that call? |
| 13:43.22 | deStone | off topic, if I wanted to learn about SAN, what's a good resource that has the most current technologies? |
| 13:43.40 | deStone | besides obviously googling, just figured there might be a site that goes into that stuff |
| 13:44.03 | [TK]D-Fender | deStone: SAN sure doesn't sound like telephony. |
| 13:44.11 | Faustov | [TK]D-Fender: thanks, checking |
| 13:44.14 | [TK]D-Fender | deStone: Go visit EMC's site |
| 13:45.16 | iamhrh | [TK]D-Fender: because I'm dialing it on one phone, then walking over to another to pick it up :-) |
| 13:45.32 | [TK]D-Fender | iamhrh: And what is your longer term goal? |
| 13:46.03 | *** join/#asterisk mikebee (n=jcrum@66-193-179-217.static.twtelecom.net) |
| 13:46.41 | [TK]D-Fender | rcahilig: would be nice if you showed us the CLI output of your failed attempt at verbose 10, and your queues.conf as well.... |
| 13:46.51 | iamhrh | [TK]D-Fender: i'd like to have the members of a queue answer a call, then have to press something to actually be connected to the caller |
| 13:47.36 | [TK]D-Fender | iamhrh: Why is that? |
| 13:47.50 | iamhrh | having annoying issues with members who are on cell phones with their voicemail answering the queue calls |
| 13:48.21 | [TK]D-Fender | iamhrh: Using PSTN to call out for agents? Thats a boat-load of trouble. |
| 13:48.38 | [TK]D-Fender | iamhrh: use the "M()" dial option and ask to confirm. |
| 13:48.43 | [TK]D-Fender | iamhrh: and say a prayer |
| 13:48.56 | iamhrh | [TK]D-Fender: haha nice |
| 13:49.04 | gr0mit | Faustov, i can provide UK numbers |
| 13:49.05 | [TK]D-Fender | iamhrh: And "pickup" has nothing to do with queues. |
| 13:49.35 | iamhrh | [TK]D-Fender: yeah, but if the queue members were local/xxx@internal ... |
| 13:49.59 | [TK]D-Fender | iamhrh: Doesn't matter |
| 13:50.48 | iamhrh | [TK]D-Fender: I'm sure there are several scenarios you're thinking of with the pstn for agents thing - but what has you saying its so much trouble? |
| 13:51.00 | gr0mit | Faustov, which type of numbers are you looking for? |
| 13:51.40 | [TK]D-Fender | iamhrh: If a cell phone is out of range,e tc VM will pick up instantly, etc. Call progress detection, and a million other things. You go try. |
| 13:52.36 | iamhrh | [TK]D-Fender: ah yes, those are the things I'm trying to work around & with. Thanks for the idea with the M() option. |
| 13:52.44 | *** join/#asterisk tobias (n=tobias@cpe-069-134-035-018.nc.res.rr.com) |
| 13:53.16 | [TK]D-Fender | iamhrh: And "Ringing" does not put a call into an actual "ringing" state. |
| 13:53.30 | [TK]D-Fender | iamhrh: so taht pickup idea is pretty dead in the water. |
| 13:54.05 | iamhrh | [TK]D-Fender: well then. bummer. time to get back to the drawing board I suppose |
| 13:55.13 | rcahilig | [TK]D-Fender: Here is my CLI output, queues.conf and extensions.conf http://www.pastebin.ca/1025886 |
| 13:57.21 | [TK]D-Fender | rcahilig: looks like MoH is playing... |
| 13:59.04 | rcahilig | [TK]D-Fender:I can hear only rings, but if I test it locally I hear music on hold |
| 13:59.43 | [TK]D-Fender | rcahilig: pastebin a local attempt |
| 14:03.34 | *** join/#asterisk a-s (n=user@89.38.174.194) |
| 14:03.40 | a-s | hello |
| 14:04.39 | rcahilig | [TK]D-Fender: Here is my full extensions.conf http://www.pastebin.ca/1025896 |
| 14:05.10 | a-s | when I am in a queue and I initiate a transfer of the calling user, can I get back and continue to speak with the calling user? |
| 14:08.09 | *** join/#asterisk intralanman (n=lanman@209.85.58.2) |
| 14:08.34 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 14:08.41 | znoG_ | heya ... if I wanted to dial a PBX and an extension all on its own ... how do I do this? (basically I want to dial a number .. wait 3 seconds .. dial the extension .. complete) |
| 14:09.02 | [TK]D-Fender | rcahilig: wasn't what I asked for... |
| 14:09.25 | [TK]D-Fender | znoG_: show us how you're dialing it currently. |
| 14:09.42 | kannan | znoG_ , you want to wait 3 secs and then pass DTMF? pre-set the DTMF? |
| 14:10.05 | [TK]D-Fender | a-s: You mean abort the transfer? |
| 14:10.19 | [TK]D-Fender | a-s: (before its completed) |
| 14:11.07 | *** join/#asterisk tobias (n=tobias@cpe-069-134-035-018.nc.res.rr.com) |
| 14:11.13 | hsv-al | corydon76-dig |
| 14:11.16 | hsv-al | its all fixed now |
| 14:13.40 | a-s | [TK]D-Fender: No, I mean: |
| 14:14.08 | *** join/#asterisk ronr (n=ron@82-204-104-172.fttx.bbeyond.nl) |
| 14:14.20 | iamhrh | [TK]D-Fender: that M() option is amazing! I was SO far from understanding what it did before now...for some reason I thought the channel got hung up immediately when it returned |
| 14:14.28 | iamhrh | [TK]D-Fender: thanks again |
| 14:15.15 | ronr | somehow my asterisk server doesn't accept incoming calls on the pri any more, a voice tells me all lines are busy, outgoing calls are still ok, enabling debug on pri doesn't show anything on incoming calls, any ideas? |
| 14:15.19 | jaytee | anyone notice the voip-info.org Asterisk wiki has changed? |
| 14:16.28 | a-s | X calls into a queue. An agent A answers to X in the queue. Then A wants to transfer X to a given number. But maybe A reminds, and he wants to continue speaking with X. How can I do for A to be able to speak with X? |
| 14:17.43 | *** join/#asterisk Weasel[DK] (n=Weasel[D@0x503fbea6.kd4nxx16.adsl-dhcp.tele.dk) |
| 14:20.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:20.54 | znoG_ | [TK]D-Fender: right now just a standard Dial() .. i want to dial a particular number a certain way ... dial, wait 3 secs, send a sequence of DTMF .. then the dial is complete |
| 14:21.16 | znoG_ | kannan: yes .. i want to dial, wait 3 secs, send DTMF .. then bridge |
| 14:21.28 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:23.43 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 14:23.54 | kannan | znoG_ : i think Dial with D option, use w for 1 sec pause then the digit to be transmitted |
| 14:24.52 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 14:25.05 | *** join/#asterisk r0land (n=roland@212.28.233.21) |
| 14:25.08 | r0land | hello all |
| 14:25.34 | r0land | could some1 guide me to a way of configuring extensions.conf to answer an incoming pstn call and plays a msg asking the caller to input an extension |
| 14:25.53 | znoG_ | kannan: something like this? Dial(Zap/g0/number,D(:w110)); |
| 14:26.35 | kannan | like Dial(SIP/something, 55, D(www1w2w3)) |
| 14:26.46 | kannan | znoG_ : yes i think so |
| 14:26.56 | [TK]D-Fender | jaytee: Yeah some dumb-ass mangled it. |
| 14:26.58 | znoG_ | kannan: thanks |
| 14:27.04 | kannan | you have to know the amt of time to wait tho |
| 14:27.20 | jaytee | [TK]D-Fender, that's what I thought. |
| 14:27.49 | [TK]D-Fender | r0land: Go lookup "asterisk ivr tips" on the WIKI, and go read the book. |
| 14:27.50 | [TK]D-Fender | ~wikis |
| 14:27.51 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:27.52 | [TK]D-Fender | ~book |
| 14:27.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 14:31.02 | *** join/#asterisk SteveTotaro (n=Administ@b.ccbccf.client.atlantech.net) |
| 14:31.21 | a-s | [TK]D-Fender: solved the problem. just using parkingcall can solve the problem |
| 14:31.42 | [TK]D-Fender | a-s: glad to hear. |
| 14:31.52 | ronr | anyone any idea where I can look when incoming calls never seem to reach asterisk at all (pri E1) |
| 14:33.38 | kannan | ronor : maybe the telco has to be contacted. Previosly, what dialplan extension pattern was used fr the incoming? |
| 14:34.33 | kannan | ronr |
| 14:34.48 | [TK]D-Fender | ronr: start up PRI debug on max and take a look. |
| 14:35.25 | ronr | kannan: I made no changes to the dailplan, I did however do a recompile of zaptel / libpri / asterisk and a kernel upgrade (among with other lib upgrades) (needed the security updates) |
| 14:36.01 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 14:36.05 | ronr | [TK]D-Fender: pri debug span 1 max and tried to call in, nothing |
| 14:36.44 | [TK]D-Fender | ronr: It was working, no change, then just stopped? Any error lights? |
| 14:37.46 | ronr | [TK]D-Fender: well, changes with the kernel and libs update and the recompile, but that's all, no error lights (outgoing call on the same PRI are working btw) |
| 14:38.01 | rcahilig | [TK]D-Fender: sorry for the delay, here the CLI of my local attempt http://www.pastebin.ca/1025928 |
| 14:38.11 | ronr | [TK]D-Fender: somehow, however I did overwrite zaptel.conf, but I restored it from a backup |
| 14:39.04 | [TK]D-Fender | rcahilig: Your MoH classes are not the same <-- |
| 14:39.30 | [TK]D-Fender | rcahilig: Local used "default", inbound used well.. "inbound" and I'm betting that just isn't valid.. |
| 14:40.09 | [TK]D-Fender | ronr: recompile of zaptel and * were after the kernel upgrade? |
| 14:40.24 | [TK]D-Fender | ronr: does ztcfg -vvvv look normal? |
| 14:40.44 | ronr | [TK]D-Fender: yes, after the upgrade |
| 14:40.57 | ronr | [TK]D-Fender: yes |
| 14:41.04 | ronr | [TK]D-Fender: looks normal that is |
| 14:41.15 | [TK]D-Fender | ronr: ok, I' pastebin absolutely every test you can think of and take a closer look. |
| 14:41.46 | ronr | [TK]D-Fender: it does seem that outgoing calls are cut off after a while, the console says cause 31 |
| 14:41.53 | ronr | [TK]D-Fender: that would be great thx |
| 14:42.08 | [TK]D-Fender | ronr: if calls are failing hmm.. what zaptel ver? |
| 14:42.23 | ronr | [TK]D-Fender: 1.4.7 |
| 14:42.27 | [TK]D-Fender | ronr: pastebin "cat /pro/interrupts", and what card are you using? |
| 14:42.27 | ronr | [TK]D-Fender: .1 |
| 14:43.11 | ronr | [TK]D-Fender: http://pastebin.com/m776b9b90 |
| 14:43.55 | ronr | [TK]D-Fender: the card is a TE12XP |
| 14:44.11 | ronr | TE120 P |
| 14:44.56 | [TK]D-Fender | ronr: feel free to paste up your configs. |
| 14:45.13 | ronr | [TK]D-Fender: which, zaptel.conf and zapata.conf? |
| 14:45.52 | ronr | http://pastebin.com/m24e120d9 |
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| 14:46.30 | rcahilig | http:/ |
| 14:46.52 | rcahilig | [TK]D-Fender: how can I fix this? |
| 14:46.58 | ronr | http://pastebin.com/m3660cbd2 |
| 14:47.40 | [TK]D-Fender | rcahilig: You're setting a MoH class in your dialplan. Why is ti different? Whya re you even setting it? |
| 14:48.05 | [TK]D-Fender | ronr: and the config files themselves... |
| 14:48.16 | ronr | [TK]D-Fender: pri intense debian span 1 still doesn't show any output during the incoming calls |
| 14:48.27 | ronr | [TK]D-Fender: which config files do you mean? |
| 14:48.38 | coppice | multi-touch is nice |
| 14:48.59 | coppice | oops. wrong window |
| 14:49.11 | [TK]D-Fender | ronr: zapata / zaptel ... |
| 14:49.31 | ronr | [TK]D-Fender: those are in http://pastebin.com/m24e120d9 |
| 14:49.34 | coppice | daddy / mommy |
| 14:50.47 | znoG_ | kannan: next problem I'm having is that if I use that D(:digits) suggestion, it waits until the channel is answered before sending the DTMF (which is actually logical, except that Asterisk doesn't detect that the PBX has actually picked up the call) |
| 14:50.59 | [TK]D-Fender | ronr: ok.. a partial PRI... this WAS working jsut fine before on the same setup, right? |
| 14:51.28 | ronr | [TK]D-Fender: yes |
| 14:51.39 | ronr | [TK]D-Fender: I just found this in the logs NOTICE[31903] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 14:51.41 | [TK]D-Fender | ronr: Ok, I'm all out of ideas... |
| 14:51.48 | [TK]D-Fender | ronr: WELL thats not good. |
| 14:51.59 | [TK]D-Fender | ronr: tried shutting down and completely restarting? |
| 14:52.17 | ronr | [TK]D-Fender: yes |
| 14:52.36 | ronr | [TK]D-Fender: I guess I could try the latest zaptel and libpri (if there are any newer version) |
| 14:53.03 | [TK]D-Fender | ronr: guess so... |
| 14:53.16 | kannan | znoG_ : i will try, i have a working conf of this somewhere, but cant seem to find it now |
| 14:58.04 | ronr | [TK]D-Fender: new version of zaptel and libpri give ZT_SPANCONFIG failed on span 1: Invalid argument (22) at the last line of ztcfg -vvvv |
| 14:58.59 | ronr | [TK]D-Fender: I'm gonna try another full redmond solution (reboot) |
| 14:59.07 | [TK]D-Fender | ronr: Good idea. |
| 14:59.52 | kannan | znoG_, i called an IVR and made it connect to me, at the particular sub-menu i wanted to go to. |
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| 15:05.39 | maqr | anyone have tips for recording wavs to be used as ulaw asterisk messages? |
| 15:06.09 | *** part/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 15:06.20 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 15:06.37 | ronr | [TK]D-Fender: still the same ZT_SPANCONFIG error in ztcfg -vvvv (after the reboot) |
| 15:06.57 | [TK]D-Fender | ronr: modprobe the driver again, and try ztcfg again |
| 15:07.37 | ronr | [TK]D-Fender: dmesg shows a lot of these wcte12xp: disagrees about version of symbol zt_receive |
| 15:08.57 | [TK]D-Fender | ronr: when in doubt, trash your source folders and reextract to recompile & install |
| 15:09.24 | tzafrir | ronr, if you get such an error, there's a mismatch between your zaptel.conf and the actual state of your system |
| 15:09.57 | tzafrir | If it is in spanconfig: grep ^span /etc/zaptel.conf |
| 15:10.15 | tzafrir | and now look at the list of spans in /proc/zaptel/* |
| 15:10.33 | kannan | is it possibly to state in general what needs to be done in a kernel re-copile for zaptel? |
| 15:10.39 | kannan | possible |
| 15:11.00 | ronr | tzafrir: zaptel.conf: span=1,1,0,ccs,hdb3 and /proc/zaptel/1 Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1" (MASTER) (so that not right) |
| 15:11.25 | [TK]D-Fender | ronr: rmmod ztdummy |
| 15:11.28 | tzafrir | ronr, the driver has failed to detect a device |
| 15:11.42 | tzafrir | or is it in /proc/zaptel/2 ? |
| 15:11.53 | [TK]D-Fender | ronr: then modprobe your driver again |
| 15:12.23 | ronr | no /proc/zaptel/2 (trying the rmmod now) |
| 15:13.09 | ronr | [TK]D-Fender: tzafrir: FATAL: Error inserting wcte12xp (/lib/modules/2.6.18-6-686/misc/wcte12xp.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 15:13.18 | ronr | recompilen zaptel now |
| 15:13.31 | tzafrir | ronr, see dmesg? |
| 15:13.36 | [TK]D-Fender | ronr: make sure you've got headers to match your kernel |
| 15:13.38 | tzafrir | or the kernel logs |
| 15:13.43 | tzafrir | what distribution is it? |
| 15:13.51 | tzafrir | ah, Debian |
| 15:13.54 | ronr | tzafrir: debian |
| 15:13.56 | tzafrir | tail /var/log/kern.log |
| 15:14.08 | ronr | it's messages like wcte12xp: disagrees about version of symbol zt_receive |
| 15:14.25 | tzafrir | right. You have an older version of Zaptel still loaded |
| 15:14.33 | ronr | uname -a and /usr/src/linux have the same header |
| 15:14.38 | tzafrir | My guess: either you forgot to unload zaptel |
| 15:14.53 | tzafrir | or you failed to unload zaptel because Asterisk was still running |
| 15:14.54 | ronr | tzafrir: I used /etc/init.d/zaptel restart and I've rebooted the machine |
| 15:15.05 | tzafrir | to check the version of the running zaptel module: |
| 15:15.13 | tzafrir | cat /sys/module/zaptel/version |
| 15:15.17 | kannan | does Debian have the same issue like xentos, where iyum install kernel, kernel-devel, alone not sufficient and needs the actual src rpm for zaptel? |
| 15:15.21 | kannan | centos |
| 15:15.48 | ronr | 1.4.10.1 (which is the version I just recompiled) |
| 15:16.09 | tzafrir | kannan, the name of the package is different: linux-headers-<version> where <version> is the name of the kernel version (the output of uname -r) |
| 15:16.45 | tzafrir | kannan, and it's actually written in the Zaptel README |
| 15:17.06 | kannan | tzafrir : oh ok |
| 15:17.09 | kannan | :) |
| 15:19.34 | ronr | [TK]D-Fender: tzafrir: fixed the module loading error (it was http://bugs.digium.com/view.php?id=12262&nbn=6) |
| 15:20.53 | jaytee | [TK]D-Fender, what codec do you recommend for Polycoms other than 729? |
| 15:20.53 | kannan | ronor, then the pri is works now fine? |
| 15:21.46 | [TK]D-Fender | jaytee: G.711. |
| 15:22.15 | ronr | kannan: [TK]D-Fender: tzafrir: and I'm back at the same problem I was before recompiling zaptel and libpri, I can call out, but not in and when calling in there's no output even in pri intense debug span 1 |
| 15:22.18 | jaytee | [TK]D-Fender, that ulaw, right? |
| 15:22.31 | [TK]D-Fender | jaytee: if the phone is local you shouldn't be using G.729 anyways |
| 15:22.56 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 15:23.04 | tzafrir | ronr, ok. And how can you be sure htat you dial the right number? |
| 15:23.11 | jaytee | yeah, from what I've read the only time the cost is justified is if you're using a ITSP |
| 15:23.12 | kannan | ronr, the modprobe wcte12xp works fine right? |
| 15:23.31 | [TK]D-Fender | jaytee: No, I still usually let * do the transcoding. |
| 15:23.35 | rbd | hi guys, I have an AMI handler set up for a meetme conference leave event. is there a way to see if the caller left because of a hangup (vs. a transfer out of the room or some other event)? |
| 15:23.57 | ronr | tzafrir: I'm sure it's the right number (unless the telco changed it without telling us and it just happens to happen just after I did some server upgrades) |
| 15:24.19 | *** join/#asterisk mukudo (n=jgreen@58.251.227.213) |
| 15:24.25 | [TK]D-Fender | rbd: Do you see a "left meetme for reason X" messaging standard anywhere? |
| 15:24.45 | *** join/#asterisk LuisTorres (n=chatzill@bl6-207-60.dsl.telepac.pt) |
| 15:24.54 | tzafrir | ronr, outgoing calls work? |
| 15:25.00 | ronr | tzafrir: yes |
| 15:25.04 | tzafrir | no alarms and such now? |
| 15:25.15 | hsv-al | [May 22 10:24:40] ERROR[5466] config.c: Future versions of Asterisk will treat a #include of a file that does not exist as an error, and will fail to load that configuration file. Please ensure that the file '../misdn-init.conf' exists, even if it is empty. |
| 15:25.16 | hsv-al | ? |
| 15:25.21 | mukudo | Hi, I have asterisk acting as PSTN gateway to openser. In the middle of the call session, Asterisk issue a Cancel message to openser. What's wrong? |
| 15:25.26 | rbd | [TK]D-Fender: well right now I'm not testing it live. this is with the starpy lib (python bindings to ami/agi) so I do get an event hash with those values parsed out. I can check though.... |
| 15:25.44 | ronr | tzafrir: none I can find |
| 15:25.49 | mukudo | The log said "Auto-congesting SIP/sipproxy-... circuit-busy |
| 15:25.58 | tzafrir | hsv-al, touch it if you want to include it |
| 15:26.07 | tzafrir | an ugly trick around it: |
| 15:26.28 | tzafrir | #include {/dev/null,missing_file.conf} |
| 15:26.30 | mukudo | how come? |
| 15:26.35 | tzafrir | (haven't tried it, though) |
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| 15:27.32 | mukudo | could somone please help me out? |
| 15:28.27 | tzafrir | mukudo, it may help to pastebin a more complete trace |
| 15:28.28 | rbd | [TK]D-Fender: now that I think about it, if the caller hangs up while in the conference room, a Hangup event should be generated. I won't get this event though if the caller is transferred out of the room (to another asterisk box, for instance), right? |
| 15:29.57 | mukudo | tzafrir: hi, please take a look at http://pastebin.org/37786 |
| 15:30.25 | mukudo | tzafrir: openser ( port 5060 ) and asterisk ( port 5061 ) in the same box |
| 15:30.43 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 15:31.03 | mukudo | tzafrir: in line 196, asterisk suddenly sends the Cancel |
| 15:32.13 | ronr | tzafrir: [TK]D-Fender: looks like it's a telco problem |
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| 15:34.23 | [TK]D-Fender | rbd: dunno. |
| 15:34.49 | tzafrir | mukudo, isn't that the remote side notifying you that it is busy? |
| 15:35.47 | mukudo | mukudo: no, I call in with my cell phone, my cell phone can hear the test music |
| 15:36.26 | *** join/#asterisk rgsteele||work (n=chatzill@74.94.57.81) |
| 15:36.43 | mukudo | cell phone <-> pstn <-> digium card <-> asterisk <-> openser <-> B2BUA is my setup |
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| 15:37.36 | mukudo | tzafrir: but in the middle, Asterisk issues the Cancel to openser, which then issues the cancel message to the other leg |
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| 15:37.48 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:41.33 | mukudo | tzafrir: it is so strange |
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| 15:53.21 | AlexDenton | im having trouble with incoming calls, it tells me that it failed to authenticate invite, im calling my DID, i already put canreinvite=no, im using only g729, plase help. |
| 15:53.25 | AlexDenton | please* |
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| 15:58.54 | spokra | Alex what do you have for insecure in sip config? |
| 15:59.12 | spokra | and what version of asterisk? |
| 15:59.14 | [TK]D-Fender | AlexDenton: PASTEBIN the CLI output of a failed call at verbose 10 and SIP debug enabled from beginning to end. |
| 15:59.16 | [TK]D-Fender | ~pb |
| 15:59.17 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:59.18 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
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| 16:02.23 | mukudo | could someone please help me out... with a strange asterisk problem http://pastebin.org/37786 |
| 16:03.13 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 16:08.16 | rbd | I'm looking for a set of AMI events I can hook it to to handle both when a call starts on the box and when it ends on the box (whether the caller hung up or was transferred out) --- would the Newcallerid and Hangup events take care of all of that? (e.g. is Hangup generated for transfers off the box?) |
| 16:10.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:16.35 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 16:17.31 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:20.31 | Yourname` | Hi. This is weird. Agents logged into support queue are rcving calls suppoed to go to sales queue.. there is no queue cascading or nothing of that sort, what could cause this? |
| 16:22.52 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:32.32 | Qwell | Yourname`: misconfiguration |
| 16:33.10 | Yourname` | Qwell: I've checked all the extensions, queues.conf, and agents.conf numerous times and I don't see any misconfig. Infact, config was not even changed. |
| 16:33.30 | Qwell | so it just suddenly started happening? |
| 16:33.36 | Yourname` | Qwell: Yeah! |
| 16:33.44 | Yourname` | That's why I was calling it weird, lol |
| 16:33.45 | Qwell | so you changed the config |
| 16:33.58 | Yourname` | haha but I didn't.. anyway, I'll go look again. |
| 16:34.07 | Faustov | hi, i'm having an inbound number registered from a sip provider, however whenever someone dials into that number and presses #, he turns on the transfer option - which is nowhere enabled in my featuremap or dialplan - is it the provider's fault? |
| 16:34.16 | Yourname` | Yesterday the only thing that was REALLY strange was I tried to 'reload' but it stopped ALL action on CLI. |
| 16:34.25 | Yourname` | No agents were able to make calls or anything until I did a restart. |
| 16:34.29 | Yourname` | That was another weird thing. |
| 16:34.46 | mukudo | hi, could someone help me with a strange problem? Asterisk sends out Cancel message to sip proxy for no reason |
| 16:35.26 | Yourname` | Qwell: reload app_queue.so should reload only queues.conf, right? |
| 16:35.29 | Strom_M | asterisk and your sip proxy don't like each other very much |
| 16:35.35 | [TK]D-Fender | Faustov: pastebin comprehensive CLI debug for the call |
| 16:35.42 | Qwell | Yourname`: no, it reloads app_queue.so |
| 16:35.59 | [TK]D-Fender | Yourname`: pastebin everything. |
| 16:36.02 | Yourname` | Qwell: If I make some changed to queues.conf, how can I reload just that? |
| 16:36.20 | Qwell | 1.4? |
| 16:36.20 | Kobaz | what's the best way to deal with a call when you don't have disconnect supervision |
| 16:36.26 | Yourname` | [TK]D-Fender: I was just getting ready for that.. BUT, as Qwell pointed out, I noticed a _MASSIVE_ oversight on my part. |
| 16:36.29 | Yourname` | Qwell: Yessir |
| 16:36.34 | Qwell | then you can't |
| 16:36.41 | Kobaz | is there a way you can detect dead silence? or something similar |
| 16:37.08 | *** join/#asterisk danlock2 (n=bean@wikipedia/danlock2) |
| 16:37.10 | Strom_M | how do I make asterisk talk to jell-o pudding pops? thanks in advance |
| 16:37.23 | *** join/#asterisk Vec (n=Vec@host-87-74-7-57.dslgb.com) |
| 16:37.41 | Yourname` | gah |
| 16:37.43 | [TK]D-Fender | Kobaz: "callprogress=yes" and say a prays because its also synonymous with "Randomly Disconnect My Calls" |
| 16:37.43 | Yourname` | A reload then |
| 16:37.54 | Kobaz | [TK]D-Fender: ah |
| 16:38.00 | [TK]D-Fender | prayer* |
| 16:38.31 | Kobaz | we're doing a setup with a box hooked up to a nortel pbx with a message waiting line card |
| 16:38.47 | Vec | I am trying to use a variable within a variable NoOp(${AGENTBYCALLERID_${CALLERIDNUM}}), running asterisk 1.2, any idea how I could do this, I found a way on voip-info, but its a horrible hack, anyone got something more elegent ? |
| 16:38.57 | Kobaz | it doesn't support any form of disconnect supervision... they've said they haven't had any problems with any other vendors products on their nortel pbx |
| 16:39.04 | Qwell | Vec: that's perfectly valid |
| 16:39.16 | Kobaz | i wonder how these other apps handle getting rid of a call that's gone away |
| 16:40.18 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 16:40.30 | Yourname` | Qwell: What could STALL a reload? As soon as I do reload, there's NOTHING on CLI anymore.. |
| 16:40.55 | Qwell | lots of things |
| 16:41.04 | Yourname` | Qwell: Thanks for the config part again, everything looked so right that I just wanted to blame other stuff, lol.. it WAS a config issue, btw. |
| 16:41.11 | Yourname` | Qwell: Any main thing? |
| 16:41.23 | Qwell | Yourname`: Please feel free to request my paypal address. |
| 16:41.37 | Yourname` | Put it on your away message haha |
| 16:41.48 | Yourname` | I'm big on the anonymous thing. |
| 16:42.31 | Faustov | [TK]D-Fender: http://pastebin.com/m15cb6b42 |
| 16:43.00 | *** join/#asterisk mikebee (n=jcrum@66-193-179-217.static.twtelecom.net) |
| 16:43.01 | Faustov | [TK]D-Fender: i've included sip debug from that call but i can't see anything more than it already is in the normal core debug |
| 16:43.39 | *** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com) |
| 16:43.50 | shmaltz | anybody know how to reset an adit 600 to defaults? |
| 16:43.53 | Vec | @Qwell : hehe, thanks, just tested it. |
| 16:43.54 | shmaltz | no access to console |
| 16:44.08 | Qwell | Vec: can also nest functions and such like that. |
| 16:44.27 | mikebee | Hi, has anyone had problems with 1.4.20 or .21.1? I upgraded from .18 but it seems I'm getting some odd behavior - Asterisk becomes unresponsive, soon I have to kill the process. |
| 16:44.50 | jer | i can paste dialplan if needed, but i'm just curious, what usually causes a failure to authenticate on invite? |
| 16:45.10 | Qwell | jer: invalid username/password |
| 16:45.22 | jer | that's what i thought |
| 16:46.18 | jer | but the username and password are correct, it's weird |
| 16:46.42 | jer | happens on two peers but not a third, all are sip, and pretty much set up the same way |
| 16:47.02 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 16:47.02 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:50.49 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:50.57 | kannan | whats the better option when it comes to termination services for USA for about 250 simultaneous calls - get T-1s from a telco, or go to a termination service provider. Bith from the point of view of reliablity and price |
| 16:51.08 | kannan | both |
| 16:51.48 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 16:55.00 | shmaltz | anybody know how to reset an adit 600 to defaults? |
| 16:56.46 | mikebee | kannan: I'd use T's if you're concerned about reliability and voice quality. |
| 16:56.47 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 16:56.54 | madduck | what is this trying to suggest to me? |
| 16:56.54 | madduck | SIP/2.0 403 Authentication user name does not match account name |
| 16:57.04 | madduck | the account name is e51 in sip.conf |
| 16:57.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.06 | jer | anyone want to look over this for me, see if they spot a problme? http://pastebin.com/me6a268c |
| 16:57.14 | kannan | mikebee : thanks, any suggestions for telco carriers? |
| 16:57.19 | madduck | and that's what is in the REGISTER auth string too |
| 16:57.51 | mikebee | kannan: time warner or verizon, only two I really have experience with, but both have been ok |
| 16:58.11 | jer | the "VOICENETWORK" and "PHONEFX" macros expand to hostnames |
| 16:59.33 | muiro | oh I just found aelparse -w. I'm so happy |
| 17:01.14 | kannan | mikebee : thanks |
| 17:03.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:04.20 | *** join/#asterisk svenna_ (n=svenna@p548D189B.dip0.t-ipconnect.de) |
| 17:05.05 | Vec | @Qwell : I am having trouble nesting this "Set(${AGENTCODE}=$[${AGENTBYCALLERID_${CALLERIDNUM}}])" ? the NoOp(${AGENTBYCALLERID_${CALLERIDNUM}}) works fine ? |
| 17:05.25 | Qwell | ${AGENTCODE} will be expanded before being set |
| 17:05.36 | Qwell | so Set(1234=1) |
| 17:05.48 | Qwell | $[] is also useless there |
| 17:07.13 | Vec | @Qwell : oh yeh, forgot when u set variable u don't use the ${}, trying... |
| 17:09.23 | Vec | @Qwell, thanks works perfect |
| 17:09.56 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 17:12.02 | [TK]D-Fender | Faustov: Nowhere in there do I see a transfer being initiated. when features.conf is triggered you'd see a CLI output of the attempt to play the transfer message. So whatever is happening is happening at the client's side. |
| 17:13.27 | shmaltz | anybody know how to reset an adit 600 to defaults? |
| 17:13.29 | Faustov | [TK]D-Fender: exactly, i've been viewing all kinds of logs over and over again, i'm pretty sure it's something on the provider's site, but they blame my setup. People connecting to another inbound number (which goes to the same extension) dont have this problem |
| 17:13.46 | Faustov | [TK]D-Fender: neither people that dial in from local phones (that register with my asterisk) |
| 17:14.06 | [TK]D-Fender | Faustov: Time to accept they fact their heads are up their asses |
| 17:14.29 | Faustov | [TK]D-Fender: ok, thanks, i was just trying to collect enough evidence that they are in fact, wankers |
| 17:14.30 | Faustov | :> |
| 17:16.10 | *** join/#asterisk danlock2 (n=bean@wikipedia/danlock2) |
| 17:16.44 | danlock2 | when I make a call into my PBX system, the phone rings, and i can pick it up, but all i hear on my cell phone calling in is ringing still... any suggestions? |
| 17:18.21 | shmaltz | anybody know how to reset an adit 600 to defaults? |
| 17:34.42 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:34.57 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-94-228-tpr-esr-2.dynamic.isadsl.co.za) |
| 17:35.16 | *** join/#asterisk golumn (n=golumn@190.6.195.108) |
| 17:35.50 | golumn | Hi guys, Which program can I use to save files in .gsm. I convert from wav to gsm usign sox, but the quality of the sound reduce alot |
| 17:36.19 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:36.45 | tzafrir | golumn, that's inherent |
| 17:37.27 | tzafrir | Inherent to the format |
| 17:38.24 | golumn | tzafrir, which will be the best format, for sound quality ulaw, alaw, gsm, etc.. |
| 17:38.34 | Qwell | golumn: ulaw |
| 17:38.39 | tzafrir | Keep it as wav |
| 17:38.41 | Qwell | (or alaw) |
| 17:38.58 | tzafrir | Why do you want to change the format? |
| 17:39.07 | Qwell | tzafrir: presumably, it isn't 8khz |
| 17:39.23 | golumn | tzafrir, I save the file in wav, but asterisk reproduce it. It is 8khz |
| 17:39.26 | golumn | mono |
| 17:39.29 | golumn | 16 bits |
| 17:39.37 | golumn | sorry |
| 17:39.42 | golumn | asterisk don't reproduce it |
| 17:40.39 | golumn | I use wav file with compressor gsm |
| 17:40.59 | golumn | or do I need to use wav as compresor also? |
| 17:46.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:53.24 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 18:05.19 | *** join/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote) |
| 18:15.49 | *** join/#asterisk josephp (n=josephp@shell4.sea5.speakeasy.net) |
| 18:16.32 | *** join/#asterisk josephp (n=josephp@shell4.sea5.speakeasy.net) |
| 18:16.34 | josephp | interesting |
| 18:17.36 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:19.36 | jaytee | I've got to start a blog about this project. I should call it "I Am Damn Proud Of Creating The Ugliest VOIP Solution That Ever Worked" |
| 18:19.56 | *** join/#asterisk ToTo (n=ToTo@151.57.70.76) |
| 18:20.05 | jaytee | course once I publish I'll get an email from some guy in Australia saying, "Bugger off, mate! I've got SIP Ethernet to Cans n' String running here in the Outback." |
| 18:20.28 | *** join/#asterisk esk (i=esk@esk.cs.usu.edu) |
| 18:21.33 | *** join/#asterisk josephp (n=Joseph_P@edgewater.cgi-bin.org) |
| 18:21.44 | josephp | blah .. ok hopefully now I wont get flooded |
| 18:21.53 | mocker | Does anyone know the database structure for agents to be in realtime? |
| 18:22.59 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 18:23.02 | josephp | anyways - I've got an asterisk 1.4.9 that was installed haphazardly and I've been trying to maintain, but the powers that be dont want me spending much time on it.. The question I have is, how do I see and/or change the codec being used for certain extensions? |
| 18:23.32 | josephp | i.e. my home extension goes across a PIX and VPN connection so I dont want to be wasting bandwidth on voip |
| 18:24.25 | Vorbote | josephp, you can use "disallow=all" and "allow=blah" in a extension by extension basis. |
| 18:25.10 | josephp | okay so it looks like I've got ulaw and alaw allowed for all |
| 18:25.20 | josephp | what would be best for saving bandwidth? |
| 18:25.26 | mocker | Hmmm, maybe agents.conf doesn't support realtime. |
| 18:27.30 | Vorbote | josephp, gsm and speex work OK in my limited experience. |
| 18:28.13 | josephp | Vorbote: okay thanks for the help, I'll play around a bit =) |
| 18:33.14 | [TK]D-Fender | josephp: What phones are you using? |
| 18:33.39 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:33.44 | Vorbote | I have a little puzzle. I have a TDM-400 with three FXO modules connected to the POTS. I need to set up the system so that it always picks ups the calls in two channels, never the third one. How can I do this? |
| 18:34.07 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 18:34.24 | [TK]D-Fender | Vorbote: in zapata.conf set your 3rd channel to point to a context that jsut calls "hangup" |
| 18:34.56 | Vorbote | [TK]D-Fender, That's an excellent idea. I hadn't thought of that. Thanks. |
| 18:35.10 | josephp | [TK]D-Fender: Polycom IP650 |
| 18:35.20 | jaytee | that why they pay him the big bucks |
| 18:35.27 | [TK]D-Fender | josephp: and they are connected to * on the other end of your VPN? |
| 18:36.43 | josephp | [TK]D-Fender: The asterisk is at the datacenter sitting next to the Cisco VPN3000 concentrator, at my home I have the Polycom 650 connected to a PIX505 which has a VPN tunnel to the VPN3000. |
| 18:36.59 | hsv-al | pix to concentrator? |
| 18:37.02 | Bananaskin | hey [TK]D-Fender received 4 faxes that I sent from work via TDM/IAXModem/Hylafx |
| 18:37.09 | [TK]D-Fender | josephp: Ok, then your only option is G.729a which is a licensed codec. |
| 18:37.28 | hsv-al | josephp, sha1/3des? |
| 18:37.32 | [TK]D-Fender | Bananaskin: And are you satisfied with the experience? |
| 18:37.56 | [TK]D-Fender | josephp: You need to pay per-channel to have * support your remote phones like that. |
| 18:38.07 | Bananaskin | Not entirely, have to see if I can now send via the TDM/IAXModem/Hylafax without the use of a call file, is this possible ? |
| 18:38.09 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:38.24 | josephp | hsv-al: 3des lan to lan network extension |
| 18:38.31 | [TK]D-Fender | Bananaskin: Just have Hylafax place the call. the rest goes through your dialplan. |
| 18:38.36 | josephp | [TK]D-Fender: ouch that hurts.. heh |
| 18:38.37 | [TK]D-Fender | Bananaskin: no * scripting needed. |
| 18:38.45 | hsv-al | hmac-160? |
| 18:39.24 | Bananaskin | but will it use the ZAP device to make the outbound call ? |
| 18:39.28 | josephp | [TK]D-Fender: is there a codec that will cost less bandwidth wise than ulaw without paying for licenses? (the powers that be are cheapasses and they have been using it with ulaw for years and just dealing with the occasional jitter) |
| 18:39.53 | hsv-al | josephp, can you degrade the tunnel enc alg? |
| 18:39.55 | [TK]D-Fender | josephp: No, Polycom only supports ULAW, ALAW, and G.729. |
| 18:41.09 | josephp | hsv-al: I was thinking about QoS but after it goes through the pix it gets combined with other traffic from a DMZ before passing through a Cisco 1721 and finally out the DSL |
| 18:41.19 | hsv-al | custom bw policy |
| 18:41.20 | hsv-al | on the conc? |
| 18:42.01 | hsv-al | forget, but isnt there two locations on the 3K for auth |
| 18:42.04 | josephp | hsv-al: the concentrator end isn't the end I'm worried about, it has an 100mb connection to the internet provider |
| 18:42.06 | hsv-al | and IKE proposals, for enc overhead? |
| 18:42.08 | hsv-al | oh |
| 18:42.51 | josephp | hsv-al: it's the DSL end that I want to choke down.. and I'm apparently the only person who wants to do this |
| 18:42.55 | [TK]D-Fender | josephp: Well you've got your options. |
| 18:44.23 | *** join/#asterisk oserr (n=oserr@c-ee74e255.016-134-73746f32.cust.bredbandsbolaget.se) |
| 18:44.40 | Weasel[DK] | [TK]D-Fender, thanks for the help earlier today with the playback stuff... finaly i got a little time to try you advise, and it worked :) |
| 18:44.48 | josephp | [TK]D-Fender: thanks |
| 18:45.00 | hsv-al | weasel, fender does not have human blood |
| 18:45.03 | [TK]D-Fender | Weasel[DK]: Glad to help |
| 18:45.06 | hsv-al | he has caffeine flowing in his veins |
| 18:45.07 | [TK]D-Fender | josephp: np |
| 18:45.15 | Weasel[DK] | Lols |
| 18:45.30 | hsv-al | dcc's d-fender some starbucks |
| 18:46.46 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
| 18:54.13 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 18:55.42 | Bananaskin | [TK]D-Fender, is it possible to use the likes of the winprint hylafax device then to send faxes ? |
| 18:56.21 | [TK]D-Fender | Bananaskin: Hylafax is still just Hylafax. It does what it does, and IAXModem is just that, a modem, nothing more. |
| 18:56.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:57.15 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.07 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 19:10.24 | *** part/#asterisk josephp (n=Joseph_P@edgewater.cgi-bin.org) |
| 19:13.26 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-94-228-tpr-esr-2.dynamic.isadsl.co.za) |
| 19:20.17 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 19:23.02 | znoG_ | hey, got a problem with dialing one particular number that Asterisk doesn't detect that the call has actually been answered (it just goes back to "nobody picked up after xxx ms") .. the call *is* actually being answered. What parameters should I play with? |
| 19:23.39 | [TK]D-Fender | znoG_: Show us comprehensive backup of the failed call with full debug (SIP, etc as appropriate) in a pastebin. |
| 19:24.10 | znoG_ | [TK]D-Fender: this is when going out a Zap channel |
| 19:26.04 | [TK]D-Fender | znoG_: pastebin it |
| 19:27.01 | znoG_ | [TK]D-Fender: sure .. which part is the most relevant? |
| 19:27.22 | [TK]D-Fender | zapata.conf & cli at verbose 10, core debug 10 |
| 19:29.41 | esk | we have a TE-220 connected to 2 did e&m t1's, I can make calls out just fine, but when I try to call in it doesn't seem to pick it up until after I hang up |
| 19:31.03 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:31.17 | esk | any ideas on what might be going wrong? |
| 19:33.18 | tzafrir | esk, no. So try providing a more verbose trace? |
| 19:34.45 | esk | from the cli? |
| 19:37.24 | esk | tzafrir: when I hang up it says " -- Starting simple switch on 'Zap/1-1'", followed by: "[May 22 13:33:51] WARNING[3434]: chan_zap.c:5837 ss_thread: getdtmf on channel 1: Success" and then "Hungup 'Zap/1-1'" |
| 19:47.09 | maqr | is there a reason why Wait()'s seem to be needed to not have audio cut off at the beginning/end of automated messages? |
| 19:47.37 | mvanbaak | rtp setup |
| 19:48.49 | maqr | could you elaborate? :) |
| 19:49.06 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 19:49.38 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 19:50.20 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
| 19:50.36 | hmmhesays | What up folks |
| 19:51.12 | SteveTotaro | anyone ever play with the NEC DTERM IPK phone? Trying to get into the manual settings |
| 19:52.11 | hmmhesays | sounds fancy |
| 19:52.46 | SteveTotaro | it is fancy |
| 19:53.04 | hsv-al | steve, |
| 19:53.08 | hsv-al | are those the black/silver ones? |
| 19:53.11 | SteveTotaro | yes |
| 19:53.13 | hsv-al | i think we got like 10 of them |
| 19:53.22 | hsv-al | what about em? |
| 19:53.22 | SteveTotaro | how do i get into manual config |
| 19:53.31 | hsv-al | http into unit ip? |
| 19:53.37 | SteveTotaro | i tried conf, hold, *, # , during initializing as the manual said but it no workie |
| 19:53.53 | hsv-al | brb |
| 19:53.54 | SteveTotaro | i don't think these have browsers |
| 19:53.57 | hsv-al | they dont |
| 19:54.14 | hsv-al | i have the manual for it in other desk |
| 19:54.26 | hsv-al | try to google i have to go afk 15min |
| 19:54.26 | SteveTotaro | nice, figured a long shot |
| 19:54.40 | Bananaskin | [TK]D-Fender, faxing (sending) working the best :) using the winprint hylafax print util, cheers for help |
| 19:55.08 | *** join/#asterisk Serge8888 (n=Serge@76.66.214.130) |
| 19:55.08 | SteveTotaro | google says conf, hold, *, # during initializing |
| 19:55.22 | Serge8888 | hi all |
| 19:55.37 | SteveTotaro | the only other link keeps it a "secret" |
| 19:56.19 | SteveTotaro | http://www.sundance-communications.com/cgi-bin/ultimatebb.cgi?/ubb/get_topic/f/14/t/001616.html#000007 |
| 19:57.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:59.14 | Serge8888 | I have a problem with a queue, in high volume of calls agent might get a couple concurent calls in one time, like I have agent with 2 lines both lines start ringing. Is any solutions for ths? |
| 19:59.30 | [TK]D-Fender | Bananaskin: got a link to it? |
| 19:59.51 | [TK]D-Fender | Serge8888: check to see if they're on the phone before actually dialing them. |
| 20:01.13 | Bananaskin | [TK]D-Fender, 2 secs |
| 20:01.34 | Serge8888 | no they not on the phone in that moment. |
| 20:02.04 | Bananaskin | [TK]D-Fender, http://winprinthylafax.sourceforge.net/ |
| 20:03.26 | [TK]D-Fender | Bananaskin: Looks nice for basic use. Got somethig more comprehensive for mass-faxing by any chance? |
| 20:05.42 | *** join/#asterisk RoyK (n=roy@ip-103-24-149-91.dialup.ice.no) |
| 20:06.42 | Bananaskin | [TK]D-Fender, sorry mate no, nothing that supports hylafx, I imagine that a script could be made if you want to go down that route of the callfile option? |
| 20:06.56 | Bananaskin | is this mass fax spam style ? |
| 20:07.46 | [TK]D-Fender | Bananaskin: I need the hylafax client. Call-files never factory into Hylafax + IAXModem, only with "txfax" dialplan app which becomes completely defunct in this case. |
| 20:07.58 | [TK]D-Fender | Bananaskin: this is for sending out TOC changes to our customer, etc. |
| 20:12.10 | [TK]D-Fender | Bananaskin: well i suppose I could script queueing them up from linux CLI once I know the file is in the right format VS trying to print from a driver like that. |
| 20:13.52 | SteveTotaro | you can drop outbound pdfs into the hylafax monitored directory via FTP |
| 20:14.06 | SteveTotaro | the filename specifies the coverpage and the dial string |
| 20:14.11 | Bananaskin | thought callfiles worked with hylafax... and the sending of a tif file |
| 20:14.33 | Bananaskin | its a tif ?? I believe that has to be put in the outbound dir |
| 20:14.39 | SteveTotaro | i don't know, i have only used pdf and hylafax's native monitored directory |
| 20:14.59 | esk | in fact, my problem looks very similar to http://forums.digium.com/viewtopic.php?t=20169 |
| 20:15.02 | [TK]D-Fender | Bananaskin: No, callfiles is only when you need * to initiate the call. When hylafax uses iaxmodem to dial, its just an IAX call so to *, hylafax is a phone and just places the call. |
| 20:15.17 | [TK]D-Fender | Bananaskin: call-files only apply to TXFAX. |
| 20:16.01 | SteveTotaro | yeah, like i said, name a pdf (i assume tiff as well) with the correct format and drop it into hylafax outbound directory it will just dial and also format your cover page for you |
| 20:16.03 | [TK]D-Fender | Bananaskin: hylafax comes with its own CLI tools for sending and uses whatever modems you have set up with it. |
| 20:16.20 | Bananaskin | trying to find link, 2 secs |
| 20:16.27 | *** join/#asterisk ipkaf (n=chatzill@dyn-88-121-29-225.ppp.tiscali.fr) |
| 20:16.30 | [TK]D-Fender | SteveTotaro: yeah, I've got lots to learn about it. I AM looking for a good GUI manger for it though. |
| 20:17.07 | eric2 | is this still valid in the dial plan: RealTimeUpdate(<family>|<colmatch>|<value>|<newcol>|<newval>) ?? |
| 20:17.28 | eric2 | I keep getting: [May 22 16:14:46] ERROR[6978]: pbx.c:1523 ast_func_read: Function REALTIMEUPDATE not registered |
| 20:17.47 | SteveTotaro | http://www.hylafax.org/man/current/hylafax-server.4f.html |
| 20:17.53 | eric2 | trying to update the database from the dial plan :( |
| 20:21.43 | *** join/#asterisk znoG (n=gs@host106.190-225-68.telecom.net.ar) |
| 20:22.40 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 20:25.28 | Bananaskin | [TK]D-Fender, just found that link and you were right about txfax, its just that it was in amongst the hylafx/IAXmodem info.. :) |
| 20:28.10 | Bananaskin | [TK]D-Fender, one option might be to use email2fax for your mass faxing |
| 20:28.46 | Bananaskin | http://wpkg.org/email2fax |
| 20:29.27 | Bananaskin | Hy-email2fax - allows you to send faxes through hylafax |
| 20:30.50 | SteveTotaro | i already told you the easiest way |
| 20:30.52 | [TK]D-Fender | ok, well I'll just google up my alternatives, just wondering what people here were actually using. Not asking to do random searching for me... |
| 20:31.18 | *** join/#asterisk km2 (n=x@c-24-23-252-175.hsd1.ca.comcast.net) |
| 20:31.26 | SteveTotaro | setup a samba share to the monitored hylafax dir or ftp |
| 20:31.27 | [TK]D-Fender | SteveTotaro: well yeah CLI could work. its fugly, and its not something I could hand to an end-user here, but its doable for a few one-shot deals. |
| 20:31.28 | Bananaskin | [TK]D-Fender, was looking at the emai2fax option fro myself and stumbled over the hy-email2fax :) |
| 20:31.42 | SteveTotaro | no, it is not cli |
| 20:31.57 | SteveTotaro | the spool dir is monitored |
| 20:31.58 | [TK]D-Fender | will follow up on all of this shortly, but for now, must be off home. Thanks to all. |
| 20:32.10 | [TK]D-Fender | SteveTotaro: gotcha, will inspect that one shortly. |
| 20:39.55 | kannan | bye all |
| 20:44.04 | jer | hrmm.. anyone have any recommendations for international DID termination? specifically, i need a mexican DID |
| 20:47.46 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:50.41 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:53.37 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:54.00 | Kobaz | i always like when my asterisk reproduces |
| 20:54.07 | Kobaz | oh, whoops |
| 20:54.25 | Kobaz | i was replying to the position i had in my scrollback buffer |
| 20:54.27 | Kobaz | heh |
| 20:58.25 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:07.11 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 21:07.35 | bsdwarrior | I changed the callgroup on a phone, reboot the phone and pruned the ext, but this phone is still unable to *0 calls. any suggestions ? |
| 21:09.22 | maqr | if i was going to try and optimize a wav file for mulaw, any idea how i'd go about doing it? |
| 21:11.14 | km2 | anyone remember the keypad shortcut to reboot a polycom phone? |
| 21:11.39 | [TK]D-Fender | maqr, sox |
| 21:12.24 | maqr | [TK]D-Fender: yeah, but there must be some preprocessing that can be done for a better sound |
| 21:12.27 | [TK]D-Fender | km2, [menu] 3,2,4,5,6,[enter],3 {yes] |
| 21:12.37 | maqr | [TK]D-Fender: maybe mulaw has some range of frequencies i should be optimizing for? |
| 21:13.05 | km2 | [TK]D-Fender, great, thank you |
| 21:13.58 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 21:15.46 | maqr | [TK]D-Fender: i got a fair amount of sound studio stuff, i just never worked towards a codec before |
| 21:16.02 | [TK]D-Fender | km2, there is a simultaneous-hold combo, but thats model specific, and I can do this one blindfolded even faster :) |
| 21:17.02 | bsdwarrior | how can I troubleshoot the pickupgroup not working. |
| 21:17.20 | km2 | [TK]D-Fender, i didn't know if that was a myth or not, but yes, i'm getting used to those key presses too |
| 21:17.49 | km2 | [TK]D-Fender, do you know how to reboot a polycom phone with packets? i've seen this scripted before but never delved into the code |
| 21:18.08 | [TK]D-Fender | km2, its clearly written on the WIKI |
| 21:18.17 | km2 | [TK]D-Fender, i'll take a look |
| 21:31.34 | bsdwarrior | I cant seem to fix my pickup group problem. it works fine one all phones but one |
| 21:35.26 | SteveTotaro | The House Commerce Committee is looking at a proposal from Representative Joe Barton that would raise the cost of your phone bill. The bill would collect the Universal Service Fund (USF) fee using a flat-fee system, where everyone pays the same amount, per telephone number. This means that even if you hardly use your phone for long-distance calling, you will pay the same as a person or business making two, ten or a hund |
| 21:36.25 | SteveTotaro | sorry, wrong window |
| 21:37.35 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 21:44.38 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 21:45.21 | edibrac | is asterisknow and/or trixbox easy to integrate onto a new existing "config-files-only" asterisk? of course, if I do it, i'm going to test it on a separate box |
| 21:50.09 | [TK]D-Fender | edibrac, No, they completely ursurp your configs. |
| 21:50.19 | [TK]D-Fender | edibrac, and turn your system into a "toaster" |
| 21:50.34 | edibrac | they should make a happy medium |
| 21:50.35 | danlock2 | anyone know anything about this error |
| 21:50.35 | danlock2 | <PROTECTED> |
| 21:50.50 | *** join/#asterisk chazzm (n=chazz@216.207.245.1) |
| 21:50.56 | Qwell | with AsteriskNOW, you can at least set the level of toastiness. |
| 21:51.05 | Qwell | I don't know the technical term for that knob on most toasters |
| 21:51.13 | Qwell | brownness level? |
| 21:51.45 | danlock2 | no one knows anything about this error? zt_handle_event: Ring/Off-hook in strange state 6 |
| 21:52.52 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 21:55.30 | wichitalineman | i was told it was for the adjustment of "knobbiness" |
| 21:55.37 | chazzm | danlock2: have you tried looking on lists.digium.com, : http://lists.digium.com/pipermail/asterisk-users/2003-September/013690.html |
| 21:56.14 | chazzm | http://lists.digium.com/pipermail/asterisk-users/2003-September/013679.html |
| 21:56.25 | wichitalineman | here is my lament... I have been burning ISOs since before God... I cannot for the life of me create a bootable sipX disk |
| 21:56.41 | sp00kz | Qwell, were you part of the dcap that happend 2 weeks ago? |
| 21:56.47 | Qwell | no |
| 21:57.10 | wichitalineman | am I not supposed to be curious? |
| 21:58.17 | danlock2 | yes, i looked into i |
| 21:58.38 | danlock2 | chazzm: you realize that is from '03 right |
| 21:58.57 | chazzm | yeah, but all I got is the one line from your request |
| 21:59.18 | chazzm | and all I did was a really really quick google search |
| 21:59.53 | danlock2 | yeah, i know |
| 22:00.03 | danlock2 | oh well.... It will maybe work itself out tomorrow |
| 22:00.35 | chazzm | hopefully so then |
| 22:07.36 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 22:08.43 | km2 | this may sound silly, but is it possible to install/compile zaptel before you have a card installed? and just insert it later? |
| 22:12.51 | seanbright | km2: yup |
| 22:13.54 | tzanger | km2: of course |
| 22:14.06 | wichitalineman | km2: the zaptel drivers will just pick up a newly installed card when you restart * |
| 22:14.53 | km2 | excellent, thank you. i didn't know if it compiled in a certain way depending on the hardware it found |
| 22:15.22 | wichitalineman | that's a valid concern |
| 22:25.43 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 22:26.00 | *** join/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
| 22:26.13 | lzhang | does anybody know what OPTIONS 324 SIP/2.0 means? |
| 22:29.48 | *** join/#asterisk joobie (n=joobie@58.108.192.59) |
| 22:31.58 | *** join/#asterisk rsc-232 (i=mrdigita@65-78-101-58.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 22:32.37 | hmmhesays | in regards to what? |
| 22:34.00 | lzhang | I'm debugging a sip registration by a phone I have... |
| 22:34.59 | lzhang | it sends the REGISTER, then asterisk responds with trying, and then 401 UNAUTHORIZED... then the phone reregisters, and asterisk responds with trying, then OPTIONS 324 SIP/2.0 |
| 22:35.17 | lzhang | the phone doesn't know what to do with it... it sends back 400 bad request |
| 22:36.00 | lzhang | I'm using an aastra 57i on the same network as the asterisk server |
| 22:37.45 | CCFL_Man2 | can a cisco ip phone with sip firmware accept incoming sip calls if a username is defined? |
| 22:44.18 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 22:56.07 | *** part/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 22:59.22 | *** part/#asterisk RoyK (n=roy@ip-103-24-149-91.dialup.ice.no) |
| 23:12.19 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:22.06 | drmessano | karma drmessano |
| 23:22.11 | drmessano | ~karma drmessano |
| 23:22.11 | jbot | drmessano has karma of 1 |
| 23:22.16 | drmessano | 1? |
| 23:24.31 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:27.46 | *** join/#asterisk synchris (n=synchris@athedsl-07871.home.otenet.gr) |
| 23:28.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:31.43 | SteveTotaro | ~karma steve totaro |
| 23:31.43 | jbot | steve totaro has neutral karma |
| 23:31.45 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:31.53 | SteveTotaro | ~karma stevetotaro |
| 23:31.53 | jbot | stevetotaro has neutral karma |
| 23:32.00 | SteveTotaro | better than mine |
| 23:34.46 | mackes | Whats happening |
| 23:34.59 | SteveTotaro | ~karma mackes |
| 23:34.59 | jbot | mackes has neutral karma |
| 23:35.17 | SteveTotaro | ~karma Qwell |
| 23:35.17 | jbot | qwell has karma of 9 |
| 23:37.50 | lanning | lanning++ |
| 23:37.58 | lanning | ~karma lanning |
| 23:37.58 | jbot | lanning has neutral karma |
| 23:38.08 | mackes | ~mackes |
| 23:38.09 | jbot | http://www.youtube.com/watch?v=P7v7uBA6LW8 |
| 23:38.33 | SteveTotaro | gross |
| 23:38.40 | mackes | Yeah' |
| 23:38.51 | SteveTotaro | ~stevetotaro |
| 23:38.51 | jbot | it has been said that stevetotaro is an IRC nub |
| 23:40.17 | SteveTotaro | http://www.youtube.com/watch?v=51vtQLydDxs&NR=1 |
| 23:41.27 | *** part/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:41.32 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:41.51 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 23:42.16 | SteveTotaro | http://www.youtube.com/watch?v=51vtQLydDxs&NR=1 |
| 23:42.29 | SteveTotaro | you in buffalo mackes? |
| 23:42.39 | mackes | Yep |
| 23:42.43 | SteveTotaro | my extended family is in Hamburg |
| 23:42.47 | mackes | Sure am |
| 23:42.50 | mackes | Ahh. |
| 23:42.52 | SteveTotaro | i go up there all the time |
| 23:42.56 | mackes | I am in Hamburg |
| 23:43.03 | SteveTotaro | really, small world |
| 23:43.05 | mackes | Do you know Delta Sonic |
| 23:43.23 | mackes | The car wash chain? |
| 23:43.29 | SteveTotaro | no, i have not been up since my cousin's wedding a few years ago |
| 23:43.33 | mackes | Ahh |
| 23:43.50 | SteveTotaro | do you know the "Best Pizzaria"? |
| 23:44.13 | mackes | Yep |
| 23:44.16 | mackes | Sure do |
| 23:44.57 | SteveTotaro | actually it is the Now Pizzaria |
| 23:44.58 | SteveTotaro | my uncle owns that |
| 23:45.04 | mackes | Neat |
| 23:45.20 | SteveTotaro | lots of family dating way back on main st |
| 23:45.33 | SteveTotaro | i remember alot from when i was a kid |
| 23:45.42 | SteveTotaro | how old are you? |
| 23:45.49 | mackes | 31 |
| 23:46.09 | SteveTotaro | did you go to high school around there? |
| 23:46.19 | mackes | Williamsville |
| 23:46.27 | SteveTotaro | do you John Andreef? |
| 23:46.35 | mackes | Na |
| 23:46.48 | SteveTotaro | used to be small town, guess it still is but not like it was |
| 23:47.33 | SteveTotaro | my fathers side of the family was from west seneca |
| 23:49.23 | *** join/#asterisk frieze (n=frieze@pool-71-251-13-242.nycmny.fios.verizon.net) |
| 23:50.13 | frieze | So I'm writing my first sip.cfg for my polycom phone. Are there any sample ones on the web anywhere? The admin guide explains all the parameters, but I have no idea what the file is supposed to look like. I hate xml |
| 23:50.57 | SteveTotaro | http://www.ca8.uscourts.gov/opndir/03/10/023842P.pdf my uncle :-( damn RICO statutes |
| 23:51.52 | mackes | Cool |
| 23:52.09 | mackes | frieze: |
| 23:52.15 | SteveTotaro | not really |
| 23:52.25 | mackes | It all depends on the version of your Polycom Software |
| 23:52.33 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 23:52.35 | mackes | Why was he a defendant? |
| 23:52.36 | frieze | is it an xml file? |
| 23:52.48 | frieze | do I just do parameter=value; |
| 23:52.56 | mackes | No, The Version of your Polycom Firmware |
| 23:53.05 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-174-254.telkomadsl.co.za) |
| 23:53.27 | *** join/#asterisk jsolis (n=jimmy@201.240.55.69) |
| 23:53.31 | xacatecas | hi all, is there anybody here that can explain to me how/what amaflags is useful for? |
| 23:53.47 | frieze | the bootrom is 3.2.3.0002 |
| 23:54.20 | frieze | so I'm reading the sip3.0 firmware, being in best guess mode |
| 23:55.35 | mackes | Polycom phones need a few files to FTP/ TFTP provision |
| 23:55.38 | jameswf-home | ~weather phx |
| 23:55.50 | frieze | though I may have wandered into the soundstation documentation, since they wisely decided to hash the documentation filenames |
| 23:55.59 | mackes | ~weather buff |
| 23:56.09 | rob0 | try kphx |
| 23:56.16 | mackes | ~weather buf |
| 23:56.29 | jameswf-home | ~weather KFFZ |
| 23:56.46 | mackes | neato |
| 23:56.56 | jsolis | some advise to take a dcap exam... |
| 23:56.58 | lanning | ~weather sjc |
| 23:57.01 | mackes | And no one as yelled yet |
| 23:57.09 | mackes | is the dcap worth it? |
| 23:57.20 | mackes | I was thinking od going to boot camp |
| 23:57.50 | jameswf-home | ~weather KFuq |
| 23:57.54 | jameswf-home | heh |
| 23:57.57 | lanning | ~weather sjo |
| 23:58.05 | *** part/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:58.28 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:59.19 | lanning | ~weather ksjc |
| 23:59.42 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 23:59.46 | jameswf-home | 1,min to go |