| 00:00.34 | tuxinator_linux | thanks for the cookie file |
| 00:00.39 | cj | ketanp: ooh, ooh, can I be a piece? |
| 00:01.04 | cj | ketanp: http://washingtonwireless.org/ |
| 00:01.31 | cj | what's the footprint of an asterisk server? |
| 00:02.31 | cj | I've got 64M of memory and 16M of flash drive capacity. is that enough to service about 100 people? |
| 00:03.41 | cj | authentication could be done via an external ldap server, if that makes a difference |
| 00:04.37 | [av]bani | _Sam--: yep, fixed |
| 00:06.01 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.84.104.myacc.net) |
| 00:06.18 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-25-222.cybersurf.com) |
| 00:06.38 | _Sam-- | there are only 2 files in the .zip? |
| 00:08.40 | bjames | Zaptel hates AMD64 |
| 00:11.32 | [av]bani | ztdummy likes amd64 here |
| 00:12.07 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 00:12.23 | cpm | apparently so does everyone who pays attention at all. |
| 00:12.31 | cpm | hehe |
| 00:12.40 | cpm | it does have it's tricks though. |
| 00:13.51 | tzafrir_laptop | cj, quite minimal. Your starting point should probably be astlinux.org |
| 00:14.12 | xachen | cj, uh no :) |
| 00:14.19 | xachen | * would crash if you had 100 people on 64M of RAM |
| 00:14.29 | xachen | you'd need at least 512 |
| 00:14.32 | xachen | and thats if your optimized good |
| 00:14.46 | tzafrir_laptop | 64MB may be just about enough if you don't have too any concurrent calls. |
| 00:14.52 | cpm | and a really really fast bus |
| 00:15.13 | tzafrir_laptop | 100 people or 100 concurrent calls? |
| 00:16.26 | tzafrir_laptop | nah, don't need that much memory (512MB) even for 100 concurrent calls. But it never hurts... |
| 00:19.26 | tzafrir_laptop | in fact, if the system had just a kernel, a minimal asterisk, busybox it might just fit in 16MB. But it would take stressing. And there are probably one or two other things to put on the system |
| 00:19.37 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 00:20.18 | *** join/#asterisk geesus (n=paul@220-244-218-250.static.tpgi.com.au) |
| 00:21.21 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:21.53 | _Sam-- | i dont think you could fit a kernel * and busy box in 16 |
| 00:21.56 | _Sam-- | but i could be wrong |
| 00:22.01 | mzo | heh 1 call maybe :P |
| 00:22.12 | cj | tzafrir_laptop: 100 people |
| 00:22.13 | justinu | is there a dialplan variable that gets filled in with the mailbox number of a SIP peer? |
| 00:22.20 | justinu | i looked at README.variables and didn't see it. |
| 00:22.38 | geesus | Hi all, im having a slight bit of trouble thats more a networking question then an asterisk question. I have 2 iaxy's on an upstairs network, my asterisk box is on a downstairs network. there is a linux box between them both, the ip for upstairs is 192.168.0.2, the ip for downstairs is 192.168.2.2, how can I set it up to forward packets from the iaxy's to the asterisk server? |
| 00:22.57 | cj | tzafrir_laptop: quagga, an embedded http server, the NoCat utils |
| 00:24.39 | lithi | justinu: I don't believe there is |
| 00:24.46 | justinu | i might have to add one |
| 00:25.04 | lithi | justinu: Unless you set accountcode to the same as the vmb number |
| 00:25.06 | [av]bani | _Sam--: openwrt fits a complete system in 4m |
| 00:25.08 | WasPhantom | heh - just watched the second ep of the it crowd over lunch. Christ it is funny. |
| 00:25.15 | justinu | in my setup, the callerID is often different than the mailbox number |
| 00:25.31 | _Sam-- | 4 including kernel? |
| 00:25.37 | lithi | justinu: not callerid but accountcode |
| 00:25.47 | [av]bani | and i fit a kernel and * and busybox in the 16 of openwrt |
| 00:25.55 | [av]bani | kernel, *, busybox, ntp, wlan, etc |
| 00:26.04 | [av]bani | on my wrt54gs |
| 00:26.53 | lithi | [av]bani: how many calls can you put though that? |
| 00:27.02 | [av]bani | lithi: a lot, if you use reinvite |
| 00:27.19 | [av]bani | with wrt54gs in the media path, about 4 |
| 00:27.26 | lithi | [av]bani: yea I wonder what happens if you need to transcode |
| 00:27.27 | lithi | ah |
| 00:27.31 | litage | i'm configuring ser and serweb and noticed that serweb expects some tables that the ser database doesn't have, such as lcr and uuidaliases. what's up with that? |
| 00:27.33 | [av]bani | no transcoding allowed :) |
| 00:27.49 | lithi | [av]bani: Did you setup QoS on your wrt54g yet? |
| 00:28.18 | geesus | [av]bani: less * and more etpro plz ;) |
| 00:28.22 | _Sam-- | most sip providers support reinvite? |
| 00:28.36 | tzafrir_laptop | cj, actually busybox has its own httpd. Though it is probably worthless |
| 00:28.51 | _Sam-- | i use the busybox httpd on my tivo...it works |
| 00:29.09 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:29.23 | lithi | [av]bani: where do you store voicemail? |
| 00:30.12 | justinu | any of you q931 wizards know what the "screening" is in the called/calling party IEs? |
| 00:32.33 | [av]bani | lithi: wherever you want |
| 00:32.50 | [av]bani | geesus: O RLY |
| 00:33.14 | geesus | [av]bani: YA RLY, SRSLY |
| 00:34.24 | Dr-Linux | hi all |
| 00:34.27 | *** join/#asterisk JUNBUG (n=bouboun@c-69-180-110-32.hsd1.fl.comcast.net) |
| 00:34.36 | geesus | hi2u |
| 00:34.58 | JUNBUG | bah... wheres ariel when ya need hi,... |
| 00:35.06 | Dr-Linux | justinu: how are youuuu ? :) |
| 00:36.07 | cj | tzafrir_laptop: http://metrix.net/support/dist/ |
| 00:36.08 | cj | http://metrix.net/metrix/products/packages |
| 00:36.20 | [hC] | hmm... odd. my sipura 941 keeps repeating itself every once in a while |
| 00:36.23 | [hC] | seen it happen twice today |
| 00:36.27 | [hC] | wonder why that'd happen |
| 00:36.49 | Junbug | looking for good sip/iax -> pstn gateway providers ,,, got my box colo'd |
| 00:38.09 | justinu | Dr-Linux: good.... i guess no one knows about q931 here :) |
| 00:38.59 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 00:39.06 | Qwell[] | Junbug: asterlink or nufone are good |
| 00:39.43 | detatch | repeating itslef |
| 00:39.47 | detatch | whatchu talkin bout willis |
| 00:40.58 | Junbug | Qwell: ahh thats the name i couldnt remember asterlink .... thx |
| 00:42.39 | *** join/#asterisk benjk (n=benjamin@66-215-63-81.dhcp.atsc.ca.charter.com) |
| 00:43.12 | benjk | has anybody here used the Grandstream Ht-488 for outbound PSTN calls? |
| 00:43.37 | detatch | ht-488 |
| 00:43.40 | tzafrir_laptop | cj, if you find pebble's software obsolete, consider "voyage linux", which is somewhat similar in scope |
| 00:43.41 | benjk | yep |
| 00:43.44 | detatch | is that a model? |
| 00:43.54 | benjk | that is the model number of the adapter |
| 00:44.01 | detatch | ah its an ata? |
| 00:44.02 | benjk | HandyTone HT-488 |
| 00:44.09 | benjk | 1FXO and 1FXS |
| 00:44.24 | benjk | more than an ATA, it's got an FXO port |
| 00:44.27 | detatch | oh, no havent used it, i have an office full of budge tone's |
| 00:44.46 | detatch | and sipura's |
| 00:44.50 | benjk | I can receive incoming calls on the FXO but outbound dialing doesn't do anything |
| 00:45.02 | justinu | spa-3000 is sipura's ATA w/ FXO and FXS |
| 00:45.29 | benjk | I know about the SPA-3000 but trust me, the box I am looking at right now is not a Sipura |
| 00:45.31 | [hC] | any of you guys connected to canadian pri? |
| 00:45.33 | benjk | I can read |
| 00:45.35 | *** part/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 00:45.38 | [hC] | specifically to allstream? |
| 00:45.47 | benjk | It says HandyTone-488 on it |
| 00:45.57 | benjk | so clearly it is *not* a Sipura |
| 00:46.29 | mzo | grandstreams need more lights |
| 00:46.38 | hypa7ia | hey [hC] |
| 00:46.55 | benjk | its a little black box, it doesn't need any lights |
| 00:47.07 | detatch | what u need lights for |
| 00:47.11 | detatch | flare |
| 00:47.17 | WasPhantom | question people, what is a fair price on a cisco 7960g that has been flashed with the sip image? |
| 00:47.25 | [hC] | hey hypa! |
| 00:47.27 | WasPhantom | comes with power supply |
| 00:47.29 | [hC] | me too! |
| 00:47.33 | [hC] | What are you doing here?? |
| 00:47.39 | justinu | benjk: i believe you, i was just not paying to the conversation really... just throwing out stupid comments |
| 00:47.51 | justinu | s/paying/paying attention/ |
| 00:47.57 | benjk | well, I need to get this thing to dial out |
| 00:48.06 | justinu | i could help you if it was a SPA-3000 ;) |
| 00:48.29 | benjk | I hate two stage dialing boxes |
| 00:48.44 | benjk | the SPA-3K is as bad on that as the HT-488 |
| 00:49.03 | hypa7ia | [hC]: i'm here for http://www.oscms-summit.org/ |
| 00:49.08 | hypa7ia | at robson place |
| 00:49.16 | benjk | whats wrong with dialing sip:18885551234@adapter |
| 00:50.05 | benjk | no, these imbeciles have to pick up the line and read DTMF |
| 00:50.13 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 00:50.20 | benjk | as if SIP couldn't pass the dialed number |
| 00:50.23 | mzo | <PROTECTED> |
| 00:50.29 | [hC] | hypa7ia: ah, right on.. didnt know that was going on.. how is it? |
| 00:50.34 | wundaboy | im trying to get my pbx to Answer and then Echo, i have an voip provider who transfers call's to my pbx over IAX. when i call it i get this: |
| 00:50.34 | wundaboy | Feb 7 16:52:50 NOTICE[24324]: chan_iax2.c:6776 socket_read: Rejected connect attempt from 66.227.100.30, who was trying to reach '15033341400@' |
| 00:50.44 | [av]bani | benjk: its simepl to make spa3k 1 stage |
| 00:50.48 | [av]bani | simple |
| 00:50.49 | wundaboy | what does that mean? |
| 00:50.56 | hypa7ia | [hC]: it's good, lots of cool people. focus is mostly on drupal |
| 00:51.02 | [av]bani | of course the AEC on the spa3k is complete shit, but... |
| 00:51.05 | mzo | So I was reading about the mysql thing, is that license business even legal that they can have a dual license? like, me using Asterisk @ home it's that actually legal? :P |
| 00:51.07 | [av]bani | that is another story |
| 00:51.13 | [av]bani | mzo: asterisk has a dual license |
| 00:51.26 | hypa7ia | mzo: back up a second there |
| 00:51.43 | [hC] | hypa7ia: nice :) I just got back from costa rica with some friends last night, got a cold already :S doh.. |
| 00:51.55 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 00:51.59 | hypa7ia | [hC]: shitty :/ |
| 00:52.08 | [hC] | hypa7ia: how long is the conference? |
| 00:52.16 | tzafrir_laptop | mzo, no, AFAIK, but IANALATINALA |
| 00:52.23 | benjk | well, the HT-488 documentation says "In the web configuration page, if the Route to PSTN field is configured, the second stage dialing is eliminated. That is, after users dial the FXO SIP account number, the PSTN number will be called automatically." |
| 00:52.33 | [av]bani | ht488 is poo, so is spa3k |
| 00:52.59 | mzo | omg, long acronym. :P |
| 00:53.02 | benjk | indeed, but its the only thing that can be used for now since we haven't got any drivers yet |
| 00:53.11 | [av]bani | drivers for what |
| 00:53.18 | Qwell[] | benjk: Did you see it earlier? |
| 00:53.21 | cpm | the truck |
| 00:53.26 | tzafrir_laptop | mzo, but you know what it means |
| 00:53.27 | benjk | for telephony hardware like PCI cards and modems |
| 00:53.33 | [hC] | Why would i be getting double ring on calls to my pri, even if im not specifying the 'r' argument to Dial()? Its a standard canadian pri, I wonder if i have to adjust the tone zone settings? |
| 00:53.36 | CunningPike | I have a Sipura SPA-3000 that we have configured for VOIP-to-PSTN dialing. It all works fine, except there is an approximately 9 second delay between when it answers the call from Asterisk and when the line starts ringing. Is there anything that can be done to reduce that? |
| 00:53.37 | [av]bani | oh thats right, you're mr. bsd |
| 00:53.43 | benjk | Qwell, see what earlier? |
| 00:53.52 | Qwell[] | benjk: The noob getting his 7960 working |
| 00:53.59 | Qwell[] | one step upgrade from sccp 3.1 to sip 7.5 |
| 00:54.25 | Qwell[] | took all of 5 minutes for him to work out the tftp issue |
| 00:54.34 | benjk | well, it didn't work here |
| 00:54.51 | Qwell[] | [hC]: Did you have any problems one step'ing your 7960's? :) |
| 00:54.58 | mzo | tzafrir_laptop, i do. I'm a huge noob. :P |
| 00:55.17 | geesus | Hi all, im having a slight bit of trouble thats more a networking question then an asterisk question. I have 2 iaxy's on an upstairs network, my asterisk box is on a downstairs network. there is a linux box between them both, the ip for upstairs is 192.168.0.2, the ip for downstairs is 192.168.2.2, how can I set it up to forward packets from the iaxy's to the asterisk server? |
| 00:55.18 | Qwell[] | benjk: I'm pretty sure it's just you |
| 00:55.18 | mzo | haha, that phone is still broken? Mail it to me, i need a spare cisco :P |
| 00:55.32 | benjk | and judging by some rather weird quirks, it is very possible that it didn't work because the phone didn't see any DNS or default router or NTP server or whatever |
| 00:55.41 | [hC] | Qwell: nope. none. |
| 00:55.55 | Qwell[] | benjk: Ahh, so it said something in the logs I told you to check? |
| 00:56.07 | [hC] | Qwell: I may have had to do some xmldefault.cnf.xml shit to get it to pick up the firmware, but never had to do special version loads in order. |
| 00:56.08 | mzo | i feel like stealing a 7940 from work and messing it up, getting really drunk and then doing the firmware update |
| 00:56.21 | Qwell[] | mzo: $10 says it'll work |
| 00:56.23 | benjk | for example, when upgrading from 2.3 to 3.0 you have to make sure that you do not have a CRLF character after the image name in the SIPDefault.cnf file |
| 00:56.29 | mzo | im sure. $10 buys a lot of booze :p |
| 00:56.39 | WasPhantom | heh - I done that to a USR modem once. Never worked again. |
| 00:56.43 | Qwell[] | So don't uprade from 2.3 to 3.0 |
| 00:56.52 | Qwell[] | upgrade from the stock 3.1 sccp image (which you had) to 7.5 SIP |
| 00:57.11 | benjk | things like that mean that you can never be entirely sure how the phones will behave in a give environment that may be different from what you normally encounter |
| 00:57.27 | mzo | ahah, i did that to my USR 8054 router, it's still sitting as a doorstop. |
| 00:57.36 | benjk | well, in this environment upgrading from SCCP 3.1 was impossible |
| 00:57.40 | Qwell[] | mmhmm |
| 00:57.42 | benjk | probably due to such quirks |
| 00:57.46 | WasPhantom | I'll never flash a device while drunk ever again |
| 00:57.47 | WasPhantom | heh |
| 00:57.48 | Qwell[] | so fix the quirks |
| 00:57.55 | benjk | that's not my job |
| 00:57.56 | Qwell[] | WasPhantom: It's FUN! |
| 00:58.01 | mzo | All those upgrades just work. Every time they didn't was because i was stupid and like, didn't have it plugged in right, or had a firewall running (or I was drunk) |
| 00:58.02 | Qwell[] | benjk: Then shut up about it |
| 00:58.09 | cpm | heh |
| 00:58.09 | benjk | and besides I don't have the Cisco firmware source code |
| 00:58.18 | benjk | that/s a ridiculous comment |
| 00:58.29 | benjk | since when is Cisco's firmware open source? |
| 00:58.32 | mzo | benjk, send me your ciscoz plz as you aren't going to be using them. :P |
| 00:58.41 | benjk | I don't have any Ciscos |
| 00:58.43 | Qwell[] | mzo: indeed |
| 00:58.52 | benjk | and I am indeed not going to use them |
| 00:58.54 | mzo | um, aren't you trying to upgrade a 7940 phone? :P |
| 00:59.07 | mzo | send me!!! |
| 00:59.07 | benjk | they are my customer |
| 00:59.07 | mzo | oh. |
| 00:59.08 | benjk | s phones and he is going to use them |
| 00:59.10 | Qwell[] | they aren't gonna be using them either |
| 00:59.11 | mzo | tell them to sell them to me =) |
| 00:59.15 | mzo | i want a few |
| 00:59.20 | benjk | they are using them already |
| 00:59.24 | mzo | say they are broken. =) |
| 00:59.27 | benjk | with 7.5 SIP images |
| 00:59.42 | benjk | and yes, Cisco' |
| 00:59.47 | *** join/#asterisk rephorm (n=rephorm@216-188-230-85.dyn.grandenetworks.net) |
| 00:59.51 | benjk | s firmware does have some nasty quirks |
| 00:59.54 | mzo | so how did they get to work if the upgrade didn't work? |
| 01:00.30 | benjk | SCCP 3.1 to SIP 2.0, 2.2, 2.3, 3.0, 3.2, 4.0, 6.3, 7.5 |
| 01:00.36 | Qwell[] | hilarious |
| 01:00.39 | benjk | indeed |
| 01:00.46 | Qwell[] | sccp 3.1 > sip 7.5 |
| 01:00.46 | mzo | <PROTECTED> |
| 01:00.54 | mzo | they had me on some VOIP call taht was totally popping and staticy |
| 01:01.01 | mzo | i felt like telling them to reboot their callmanager(TM) |
| 01:01.09 | benjk | and from 2.3 to 3.0 you have to eliminate any CRLF after the image name in SIPDefault.cnf |
| 01:01.20 | dudes | or "dude, you're getting asterisk" |
| 01:01.25 | mzo | um, you can go fro sccp 3.1 > sip7.5 |
| 01:01.27 | mzo | it was kinda sad. |
| 01:01.33 | mzo | i had to call back like four times to get a decent connection |
| 01:01.34 | benjk | Qwell, not in this environment |
| 01:01.39 | mzo | and I talked to "esteban" from "texas" |
| 01:02.01 | mzo | the faint smell of curry was coming through the voip connection, so I'll never know where he really was :P |
| 01:02.06 | rephorm | hi. i've got an X100P FXO card hooked up to my home phone line (using * as an answering machine). it works well, except that the zap channel doesn't detect when a caller hangs up before asterisk answers. so i get a few minute recording of dialtone when this happens. any ideas? |
| 01:02.31 | benjk | anyway, does anybody know how the HT-488 PSTN dialing works? |
| 01:02.48 | mzo | i'm happy my budgettone voicemail works :) |
| 01:02.48 | dudes | add a h extensions? |
| 01:03.17 | mzo | how to torture asterisk, call your voice mail, put it on hold. repeat |
| 01:03.36 | geesus | hahah |
| 01:03.47 | mzo | i was trying to crash it, but no luck, it just works |
| 01:03.54 | rephorm | dudes: ah. thanks. worked :) |
| 01:04.31 | mzo | my grandstreams sound so wonky when i hit dtmf tones, they sound all trance-music-y. |
| 01:04.49 | CunningPike | Nobody with any SPA-3000 knowledge? |
| 01:05.53 | rephorm | dudes: or not... |
| 01:06.35 | rephorm | dudes: if you hang up at the right time it still goes wonky |
| 01:06.38 | mzo | i donut have an spa-3000 sorry =( |
| 01:07.58 | mzo | hehe, memory is leaking hardcore ;) |
| 01:08.11 | rephorm | are all US pots lines signalled kewlstart? |
| 01:10.02 | mzo | here's an obscure question. |
| 01:10.14 | cpm | anything that's been done in the last 30 years will probably loop start, so kewlstart is a good option |
| 01:10.14 | mzo | Anyone have an old AMD Slot A motherboard laying around? |
| 01:10.19 | cpm | mzo, yes |
| 01:10.23 | mzo | working? :P |
| 01:10.26 | cpm | yes |
| 01:10.27 | rephorm | socket A you mean? |
| 01:10.28 | mzo | gimme! |
| 01:10.34 | mzo | no the cartridge style? |
| 01:10.36 | mzo | old stuff |
| 01:10.38 | rephorm | ahh |
| 01:10.40 | cpm | yeah, old stuff |
| 01:10.52 | mzo | i have a 700mhz athlon that i got as a pullout. |
| 01:10.56 | cpm | but it's still in production, in a laying around kinda way |
| 01:11.18 | cpm | you can still get them, |
| 01:11.39 | mzo | oh, i'll look for some then. regular PC-100 memory? |
| 01:11.44 | Qwell[] | heh, slot a... |
| 01:11.55 | Qwell[] | What was AMD thinking? |
| 01:11.58 | mzo | i have tons of old crap laying around. |
| 01:12.33 | benjk | the HT-488 doesn't even always give a dialtone if you just connect to it leaving the second stage dialing for manual |
| 01:12.51 | cpm | http://shop.store.yahoo.com/justdeals/k7v.html |
| 01:12.52 | benjk | about 1 out of 10 times it gives you a dialtone, otherwise not |
| 01:12.58 | mzo | wow, asus still sells stuf. crazy |
| 01:13.14 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 01:13.18 | cpm | the ubiquitous k7v |
| 01:13.24 | rephorm | so no other clues about getting a cheap-o fxo to detect hangups while Waiting before Answering? |
| 01:13.26 | mzo | i'll pick one up. |
| 01:13.40 | Qwell[] | mzo: a new cpu would be cheaper :p |
| 01:13.46 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:13.47 | mzo | it's $23. ;) |
| 01:13.53 | Qwell[] | as I said...heh |
| 01:13.58 | mzo | i have enough old parts laying around that i won't need anything 'new' |
| 01:14.11 | mzo | my beaten-to-death p2-300 is finally dead. |
| 01:14.23 | mzo | that tyan like, was turned on for years non-stop. |
| 01:14.43 | cpm | bummer! |
| 01:15.05 | mzo | haha, i collect crap |
| 01:15.10 | mzo | as that photo of my 'telco wiring' proves :P |
| 01:15.13 | dudes | I have a Athlon 850 that has ran since 99' (except for a two day power outage from a ice storm) that went through a nasty house fire =) |
| 01:15.17 | Qwell[] | mzo : link? |
| 01:15.21 | mzo | um. Anyone want two quad xeon servers? :P |
| 01:15.30 | Qwell[] | mzo: Sure, send em over |
| 01:15.32 | dudes | And upgrades =) it was originally a 600 |
| 01:15.35 | mzo | www.trenchenko.com/nest.jpg |
| 01:15.51 | mzo | Qwell[], where is here? They are 220v. xeon 550s |
| 01:15.56 | Qwell[] | heh |
| 01:16.02 | mzo | they are like 80 pound cases |
| 01:16.02 | Qwell[] | I can replace the PS :P |
| 01:16.07 | Qwell[] | Southern CA |
| 01:16.12 | mzo | ow, shipping would destroy me |
| 01:16.34 | Qwell[] | if you're just giving them away, I could pay shipping :p |
| 01:16.36 | mzo | it's slow. |
| 01:16.42 | mzo | i'm gonna try to craigslist em |
| 01:16.45 | mzo | then im gonna dump em |
| 01:16.49 | Qwell[] | That's some ugly wiring... |
| 01:16.51 | mzo | i have 0 need for htem |
| 01:16.55 | mzo | i did say i collected crap :P |
| 01:17.26 | cpm | love it! |
| 01:17.28 | Qwell[] | mzo: If you end up just wanting to dump them, let me know. I can give them a good home :p |
| 01:17.32 | mzo | ok. |
| 01:17.38 | mzo | i had to wipe the drives, so you'd be reinstalling |
| 01:17.43 | mzo | but they do run linux, and whatever |
| 01:17.46 | mzo | it's intel SHV motherboards |
| 01:17.47 | Qwell[] | They'd run Gentoo anyhow |
| 01:17.51 | mzo | oh they'd run fine |
| 01:17.59 | mzo | i had to wipe them before i was able to take them out of the buildling |
| 01:18.02 | Qwell[] | yeah |
| 01:18.13 | mzo | if they weren't 220V I'd actually use them |
| 01:18.18 | mzo | but replacement PS is a nighmare |
| 01:18.52 | dudes | Quad Xeons for the price of shipping, hmm. |
| 01:19.10 | mzo | put it this way, i had to get them on pallets to get them home |
| 01:19.17 | Qwell[] | heh |
| 01:19.24 | wundaboy | i cant get asterisk to pickup from my VOIP provider ... every time i call it i get this in the asterisk console: |
| 01:19.28 | wundaboy | Feb 7 17:19:50 NOTICE[24324]: chan_iax2.c:6776 socket_read: Rejected connect attempt from 66.227.100.30, who was trying to reach '15033341400@' |
| 01:19.41 | wundaboy | help? |
| 01:19.45 | dudes | that's how they ship transmissions |
| 01:19.57 | mzo | yteah, same stuff |
| 01:20.07 | mzo | i borrowed a van that day and unpacked them and loaded them in |
| 01:20.14 | mzo | i sold my rack otherwise i would have used them |
| 01:21.18 | mzo | my old athlon 1.1 is tarting to go, too. *sniff* |
| 01:22.14 | dudes | I got a 1.2 Duron mobo/CPU I don't use ... a K7S5A (I think it's a decent board). Good for overclocking mobile XP's in any event |
| 01:23.01 | wundaboy | the ecs k7s5a is the worst motherboard ive ever seen |
| 01:23.02 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 01:23.12 | dudes | What's wrong with it |
| 01:23.14 | wundaboy | no offense |
| 01:23.21 | dudes | Works good for me =o |
| 01:23.23 | wundaboy | a) its made by ecs |
| 01:23.23 | Qwell[] | heh, I have that board |
| 01:23.26 | Qwell[] | it's complete crap |
| 01:23.26 | wundaboy | b) via chipset |
| 01:23.30 | wundaboy | c) made by ecs |
| 01:23.31 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 01:23.32 | Qwell[] | might be the k7s6a though |
| 01:23.33 | wundaboy | d) made by ecs |
| 01:23.41 | wundaboy | e) made by ecs |
| 01:23.57 | wundaboy | i had it for a while and it sucked :P |
| 01:24.08 | Ayano | I have a tdm400, and when I make outbound calls, and only out bound calls, the recieving parties cannot hear me... Any clue? |
| 01:24.11 | [hC] | Should i care about upgrading to 1.2.4 from 1.2.1? Important enough? |
| 01:24.19 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-6-79-138.msy.bellsouth.net) |
| 01:24.19 | dudes | I haven't had any complaint with the board |
| 01:24.28 | Qwell[] | [hC]: vm stuff, and memleaks |
| 01:24.36 | Qwell[] | (you already have the vm patch though) |
| 01:24.41 | [hC] | yeah |
| 01:24.50 | [hC] | ah, ill upgrade anyways |
| 01:25.07 | dudes | wundaboy - it looks like you need to have context=context in iax.conf |
| 01:25.50 | mzo | hahaha made by ECS is slang for 'embodies the powar of complete shit' |
| 01:26.55 | dudes | that'd be EPCS |
| 01:27.04 | mzo | the p is for POWAH |
| 01:27.22 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 01:27.22 | _Sam-- | the hint command in the dialplan doesnt need a priority? |
| 01:27.49 | Qwell[] | _Sam--: The hint IS the priority |
| 01:27.56 | Qwell[] | s,hint,something |
| 01:28.07 | _Sam-- | thanks...i see |
| 01:28.12 | _Sam-- | never used hint before |
| 01:28.18 | Qwell[] | exten,hint,whattowatch |
| 01:28.21 | Qwell[] | so, I have |
| 01:28.28 | dudes | I preferr Asus boards but the K7S5A runs my mobile XP, so I can't bitch about it |
| 01:28.32 | Qwell[] | 6265730000,hint,SCCP/6265730000 |
| 01:28.39 | _Sam-- | easy enough |
| 01:28.49 | _Sam-- | thanks Qwell |
| 01:28.51 | rob0 | haha I have a couple of cheapo K7S5A's |
| 01:29.06 | rob0 | I should say, I have one. The other died. |
| 01:29.10 | dudes | I have one running a XP mobiile and another I don't use |
| 01:29.32 | benjk | geez |
| 01:29.48 | benjk | HT-488 forgets half of its SIP configuration after firmware upgrade |
| 01:30.15 | rob0 | I agree, ECS is junk, I tried to save a buck and I regret it now. |
| 01:30.38 | dudes | I have no regrets on the board ... Been a champ for well over a year |
| 01:30.41 | Qwell[] | I've been regretting that dollar for a little over 4 years |
| 01:30.48 | TooMe | i'm having a lotta problems getting System() to run any command i give it. They work fine from the command line, but not with a /bin/sh -c in front... |
| 01:30.53 | detatch | funny |
| 01:30.59 | mzo | i have two dead abits in the closet somewhere. |
| 01:31.00 | detatch | ive been regretting getting married for about the same time |
| 01:31.13 | rob0 | The K7S5A is my workstation also doubling as Internet router now. |
| 01:31.22 | mzo | haha my p2-300 is so old it uses a XT keyboard connector. i wonder if i have any cables for that |
| 01:31.23 | dudes | that's why you don't get married |
| 01:31.28 | detatch | f |
| 01:31.29 | detatch | now you tell me |
| 01:31.34 | dudes | heh |
| 01:31.57 | rob0 | "Workstation" loosely speaking, I use a laptop and connect to the K7S5A's kdm. |
| 01:32.42 | benjk | Oh no |
| 01:32.48 | dudes | I think with the K7S5A's it can depend the version you get |
| 01:33.00 | dudes | they have like 5 revisions |
| 01:33.02 | benjk | you can reenter the settings and click update, but it will forget them again |
| 01:33.22 | detatch | the k7s5a i thought was a good board |
| 01:33.38 | Qwell[] | Here's what I find funny... |
| 01:33.47 | Qwell[] | I just bought a MB today, been eyeing that one for weeks |
| 01:33.51 | Qwell[] | I can't tell you the model of it |
| 01:33.51 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 01:34.05 | *** part/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 01:34.05 | Qwell[] | but you bring up that pile of garbage that ecs put out...and I can tell you everything about it :p |
| 01:34.17 | mzo | i think it's gonna be time to recycle this p2-300 |
| 01:34.21 | Qwell[] | and I'm sure I'm not the only one |
| 01:34.22 | mzo | it doesn't even post |
| 01:34.27 | rob0 | I'd say, K7S5A is sort of adequate for non-crucial uses. :) I am okay with the one I have, but I'll never buy another ECS. |
| 01:34.42 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 01:34.59 | dudes | what about the FIC SD-11 heh |
| 01:35.07 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 01:35.16 | rob0 | We have a slightly better ECS with a Via chipset too. |
| 01:36.37 | dudes | I got a XFX Mach4 KT400A chipset (I believe), but I couldn't find a reference to it working with Mobiles. So it's just sitting collecting dust. |
| 01:36.52 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 01:37.01 | rob0 | My next system, a server, is on order: SuperMicro H8SSL-i. |
| 01:38.04 | rob0 | also getting a 3ware 9550SX-4LP SATA RAID controller with it. |
| 01:39.47 | dudes | I'm eager to play with a Opteron box |
| 01:40.50 | mzo | you will be happy |
| 01:40.55 | geesus | i'm eager to go home :P |
| 01:41.00 | mzo | ooh i found my athlon 1.1 and it has diablo2 still installed! Back soon :P |
| 01:41.09 | iDunno | opteron++ |
| 01:41.14 | iDunno | amd64++ |
| 01:41.30 | iDunno | intel-- |
| 01:41.38 | [av]bani | 6502+++++ |
| 01:41.45 | detatch | 8086+++++ |
| 01:41.47 | detatch | ha! |
| 01:41.50 | justinu | Z80 :P |
| 01:41.55 | dudes | I've wondered how a 8 way Opteron would perform /w * |
| 01:42.05 | _Sam-- | [av]bani: blf on the gxp is 1 line + 2 fields in gxp |
| 01:42.14 | _Sam-- | works fine, and gives you 1 touch dial too |
| 01:42.21 | _Sam-- | with phone on hook |
| 01:42.32 | dudes | Or even what a Dual Opteron could do with transcoding g729 to ulaw |
| 01:42.39 | justinu | i have a customer who wants gxp2000s |
| 01:42.46 | justinu | what firmware should I be using on them, sam? |
| 01:42.48 | dudes | But I've never found anything |
| 01:43.37 | _Sam-- | justinu: i dont think there is any downside to running the latest beta <1.0.2.8> |
| 01:43.50 | justinu | ok, is that available to me, or do I have to warez it |
| 01:44.13 | _Sam-- | no its right out there on the GXP wiki page...http://www.voip-info.org/tiki-index.php?page=GXP-2000 |
| 01:44.27 | _Sam-- | check the mac address of the phones before upgrading |
| 01:44.31 | justinu | _Sam--: i believe this setup is similar to yours, asterisk, sangoma single span T1 card, PRI, 20ish phones. |
| 01:44.34 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:44.39 | ketanp | does the gxp do HTTP xml provisioning yet? |
| 01:44.45 | _Sam-- | er it looks like mac address problems are fixed |
| 01:45.06 | [av]bani | _Sam--: ? |
| 01:45.13 | _Sam-- | justinu: thats about right...that was how i started at least, with a PRI...then switched to T1 / remote gw |
| 01:45.20 | justinu | ok |
| 01:45.25 | _Sam-- | [av]bani: there were problems with 1.0.2.6 with mac address lower than a certani number |
| 01:45.27 | justinu | T1 meaning data? |
| 01:45.35 | justinu | cuz PRI runs on T1 also ;) |
| 01:45.35 | _Sam-- | they would have the display blank / and scrolling issues |
| 01:45.40 | _Sam-- | yes data T1 |
| 01:45.46 | [av]bani | _Sam--: the display corruption? reports are that 1.0.2.8 has the same issues |
| 01:45.48 | _Sam-- | t1 = data t1 pri = 23b + D :) |
| 01:45.56 | *** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:46.03 | _Sam-- | they took the notes off the page |
| 01:46.07 | [av]bani | ?! |
| 01:46.14 | _Sam-- | i helped robin_sz diagnose hers the other day |
| 01:46.58 | _Sam-- | [av]bani: do you think there's any downside to running the latest firmware? |
| 01:47.01 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:47.10 | _Sam-- | if you are not affected by the low address problem |
| 01:47.11 | Dr-Linux | _Sam--: today i gonna start to learn something :P |
| 01:47.11 | justinu | _Sam--: you're running them in production now? |
| 01:47.22 | justinu | with the latest firmware? |
| 01:47.37 | _Sam-- | yeah about 15 out of 50 total phones are on latest |
| 01:47.41 | [av]bani | _Sam--: how did you diagnose it? |
| 01:48.02 | justinu | _Sam--: mad propz to you for blazing the trail on these phones |
| 01:48.03 | _Sam-- | i read the description, asked her to tell me her mac address...and sure enough she was in the range of problematic macs |
| 01:48.07 | justinu | some of us appreciate it |
| 01:48.21 | justinu | some of my customer have the money for polycom and love it, and some of my customers don't |
| 01:48.22 | benjk | anybody using a Grandstream adapter with 1.0.2.16 firmware? |
| 01:48.30 | geesus | what happens when the world runs out of mac addresses? |
| 01:48.41 | _Sam-- | thanks! i appreciate your appreciation! they are good phones...mostly. today i learned more that the speakerphone must suck (2 people told me) |
| 01:48.42 | justinu | MACs only have to be unique on LAN segments |
| 01:48.48 | geesus | do we start macv6 or something? :P |
| 01:48.52 | Mark_Halverson | ok - i'm prob doing something stupid and forgot something...but I can't find it..in IAX.CONF i have a context defined - but when I attempt to dial I am getting the following: Feb 7 17:46:05 NOTICE[14770]: chan_iax2.c:6782 socket_read: Rejected connect attempt from (ip addr removed), who was trying to reach '18006396111@' |
| 01:49.00 | geesus | so there is duplicate mac addresses? |
| 01:49.09 | justinu | there can be |
| 01:49.19 | justinu | you can also change your MAC to whatever you want |
| 01:49.27 | _Sam-- | i dont think with these phones |
| 01:50.09 | _Sam-- | [av]bani: <dont yell ~pb>>> its only a few lines...NOTE: (Feb07/06) Upon further research it's not a sp-phone bug, it just got triggered when I hit SPKR. This is the same bug discussed above that made beta pull this ver off his TFTP, my MAC addy is 00.0B.82.03.xx.xx so I am affected. See: here if you still want a screenshot. -Helix |
| 01:50.21 | _Sam-- | that is the mac... |
| 01:50.33 | _Sam-- | if your mac is higher than that, you wont be affected by the screen problems |
| 01:50.34 | justinu | 13:37:DE:AD:BE:EF is my favorite mac to run |
| 01:51.06 | [av]bani | _Sam--: so its not fixed, phones in 03.xx.xx are affected |
| 01:51.21 | _Sam-- | yes, but the notes at the top of the page are removed |
| 01:51.28 | _Sam-- | there were specific notes about 1.0.2.6 |
| 01:51.31 | _Sam-- | and those macs |
| 01:51.56 | _Sam-- | it was in the yellow warning part |
| 01:52.05 | _Sam-- | from adrian2k |
| 01:52.08 | _Sam-- | put now its changed/gone |
| 01:52.15 | _Sam-- | s/put/but/ |
| 01:52.36 | [av]bani | i imagine theres a hw revision change around 03.xx.xx and the firmware uses wrong memory timings or something |
| 01:52.45 | [av]bani | so 03.xx.xx gets memory corruption |
| 01:52.54 | justinu | did GS fix the problem of having to type in the TFTP server address manually yet? |
| 01:53.12 | _Sam-- | justinu: not yet...but if bani has his way it wont be very long. |
| 01:53.17 | _Sam-- | like another couple weeks at this rate |
| 01:53.18 | justinu | sweet |
| 01:53.21 | [av]bani | i put it on the request page |
| 01:53.24 | justinu | there's just no excuse for that |
| 01:53.28 | [av]bani | <PROTECTED> |
| 01:53.40 | _Sam-- | they have been responsive |
| 01:53.44 | justinu | great |
| 01:53.46 | Dr-Linux | why people use GS, is it cheap or what? |
| 01:53.48 | _Sam-- | that one might be a little harder |
| 01:53.57 | justinu | yeah, nothing else can touch it's pricepoint Dr-Linux |
| 01:54.15 | ketanp | justinu: and what approx is the price point? |
| 01:54.19 | Dr-Linux | wow |
| 01:54.20 | justinu | 85 USD |
| 01:54.20 | [av]bani | its a reasonable phone for $70 |
| 01:54.23 | justinu | 70? |
| 01:54.25 | justinu | where? |
| 01:54.28 | _Sam-- | thats a damn good deal :) |
| 01:54.29 | Dr-Linux | justinu: how much GS cost? normally |
| 01:54.37 | justinu | Dr-Linux: 85USD for me |
| 01:54.39 | [av]bani | er $80 |
| 01:54.40 | _Sam-- | i guess if you got dealer pricing from atacommm it would be about that |
| 01:54.47 | justinu | k |
| 01:54.49 | [av]bani | https://www.discountvoipoutlet.com/store/product.php?productid=2&cat=1&page=1 |
| 01:55.03 | justinu | you happy with that reseller? |
| 01:55.21 | [av]bani | we got from voipsupply |
| 01:55.22 | justinu | i have to have 15 of them by saturday |
| 01:55.32 | Dr-Linux | makes sense |
| 01:55.46 | Dr-Linux | i think cisco phone is expensive than all others |
| 01:55.53 | justinu | yeah, cisco is the most expensive |
| 01:55.55 | _Sam-- | damn justin...im about a week too late... |
| 01:55.56 | justinu | afiak |
| 01:56.03 | _Sam-- | this is my latest project |
| 01:56.04 | _Sam-- | http://www.voipserv.com/index-dev.php?action=view_catalog_list&category_id=100001&page=1&max_item_count=15)http://www.voipserv.com/index-dev.php?action=view_catalog_list&category_id=100001&page=1&max_item_count=15 |
| 01:56.13 | _Sam-- | should be running in another few days |
| 01:56.27 | justinu | you gonna start reselling that stuff? |
| 01:56.40 | _Sam-- | yeah, i already do mail/order online biz of other stuff |
| 01:56.45 | _Sam-- | so to add that is really nothing |
| 01:56.46 | justinu | i'd definitely buy from you if you could get them to me by sat!! |
| 01:56.50 | justinu | :P |
| 01:56.56 | _Sam-- | where are you located? |
| 01:56.59 | justinu | Los Angeles |
| 01:57.01 | [av]bani | so ive given up on the spa3k as a fxo |
| 01:57.09 | justinu | bani: that bad? |
| 01:57.13 | [av]bani | the echo is horrible |
| 01:57.17 | [av]bani | its hopeless |
| 01:57.17 | justinu | :( |
| 01:57.48 | _Sam-- | justin if you have any hard time getting a hold of the phones let me know, i have a couple dealer accounts, but pricing isnt so hot. |
| 01:57.54 | justinu | how much? |
| 01:58.02 | _Sam-- | bani's place sells them for close to what i would pay |
| 01:58.24 | [av]bani | the dlink dvg-3004s looks nice, but nobody seems to have it yet |
| 01:58.28 | _Sam-- | i will check it out for you, not looking to make a dime, if i can help you out i will. |
| 01:58.29 | *** join/#asterisk psi_force (n=mark@marksnb.eng.unimelb.edu.au) |
| 01:58.34 | [av]bani | http://bani.anime.net/gateways/ |
| 01:58.37 | geesus | im gonna eat my iaxy soon |
| 01:59.28 | justinu | i gave this customer 4 gxp's to test with |
| 01:59.33 | psi_force | hi all does anyone know why a sip user's default account code (set in sip.conf) is not being set? |
| 01:59.41 | _Sam-- | they may be disappointed unless you upgrade them |
| 01:59.44 | justinu | they're supposedly going to give me a decision on whether they're acceptable or not |
| 01:59.44 | _Sam-- | the old firmware is pretty old |
| 01:59.55 | _Sam-- | and i know it echos pretty badly |
| 02:00.02 | justinu | hmm |
| 02:00.05 | Dr-Linux | justinu: what is gxp ? |
| 02:00.12 | justinu | gxp-2000 |
| 02:00.15 | justinu | grandstream |
| 02:00.25 | Dr-Linux | oo ok |
| 02:00.35 | _Sam-- | they implemented (i think) alot of things to help with the echo |
| 02:00.37 | justinu | i told them if they find the echo unacceptable, they'll have to go with polycom ip301s |
| 02:00.55 | [av]bani | the speakerphone echo is the main issue |
| 02:01.05 | [av]bani | fixed in 1.0.1.12 |
| 02:01.15 | WasPhantom | sweet - getting myself a cheap Polycom IP500.... |
| 02:01.17 | _Sam-- | there was some echo in the earpiece |
| 02:01.18 | justinu | ok, i don't care about speakerphone |
| 02:01.26 | [hC] | Why the heck does this spa-941 reboot itself occasionally... |
| 02:01.46 | _Sam-- | but the firmware could influence their decision, in my opinion. |
| 02:01.58 | _Sam-- | this new firmware sounds better, again, in my opinion. |
| 02:02.11 | justinu | i'm not sure if i'm willing to go thru the headache of trying to upgrade these things remotely. |
| 02:02.14 | ketanp | [hC]: are you using it with a 9000? |
| 02:02.18 | _Sam-- | its not that bad |
| 02:02.22 | _Sam-- | i upgraded 10 remotely yesterday |
| 02:02.31 | Dr-Linux | [dc]: there must be a virus in the spa 941 :P |
| 02:02.46 | _Sam-- | out of the 10, i had to ask the customer to pull the plug on 1 |
| 02:02.47 | Dr-Linux | [hC]:/ |
| 02:02.53 | _Sam-- | it wouldnt get the time right after it was done |
| 02:03.16 | psi_force | does anyone have any ideas about my account code problem? |
| 02:03.17 | _Sam-- | but they updated themselves ok |
| 02:03.21 | justinu | k |
| 02:03.26 | _Sam-- | i tested it first on my deskotp to make sure they would |
| 02:03.29 | [hC] | ketanp: nope. |
| 02:03.30 | justinu | i'm just gonna see what they have to say about 1.0.1.9 |
| 02:03.34 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 02:03.42 | WasPhantom | are they any good? |
| 02:03.42 | *** join/#asterisk watchy (n=watchy@70.238.56.18) |
| 02:03.45 | watchy | Jan 7 00:59:12 NOTICE[11150]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 02:03.53 | watchy | what would call that? |
| 02:04.01 | justinu | _Sam--: aren't you the customer? :P |
| 02:04.11 | WasPhantom | watchy, an unknown error |
| 02:04.18 | _Sam-- | im someones customer |
| 02:04.32 | _Sam-- | i would be willing to wager a few dollars that they will say the phone sucks |
| 02:04.40 | _Sam-- | because it kind of does with that firmware |
| 02:04.51 | justinu | k |
| 02:04.53 | ketanp | [hC]: hmm, is it at regular intervals, or random? if it's random, the syslog output might tell you why |
| 02:05.08 | _Sam-- | what phones are they used to? |
| 02:05.14 | justinu | pots phones |
| 02:05.25 | watchy | was: hmm |
| 02:05.29 | watchy | it was working yesterday |
| 02:05.34 | [hC] | ketanp: Im not sure wether its an interval or not, ive only seen it happen twice so far. Is there an internal syslog on the phone or do i hve to tell it to syslog externally to retrieve it? |
| 02:05.37 | _Sam-- | this is a whole new install...new pri, new pbx, new phones? |
| 02:05.47 | justinu | yep, all new. |
| 02:05.55 | _Sam-- | hosted pbx or server there? |
| 02:06.00 | justinu | on site |
| 02:06.29 | watchy | Jan 7 01:04:21 NOTICE[11180]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 02:06.48 | watchy | i keep getting this shit |
| 02:06.54 | watchy | it was working yesterday |
| 02:07.12 | ketanp | [hC]: i think you'd have to setup a syslog server, there are some simple windows ones available (and obviously linux ones)... then set the Syslog_Server and Debug_Level settings... if it's random, the unit's probably bad... the first few batches were a little weak |
| 02:08.04 | [hC] | ketanp: cool, ill give the syslog server a go if it persists. I just turned off provisioning, see if maybe that corrects it. |
| 02:08.10 | [hC] | bbl.. |
| 02:08.39 | _Sam-- | justinu: you will be responsible for remotely managing the server? |
| 02:09.10 | justinu | initially |
| 02:09.16 | justinu | i'll outsource it to someone else tho |
| 02:09.58 | Dr-Linux | justinu: they know about asterisk stuff? like how it works? |
| 02:10.06 | justinu | nope |
| 02:10.13 | Dr-Linux | they will be able to handle * issue laterly? |
| 02:10.14 | _Sam-- | i have a few of those arrangements...but i like to put solid state equipment at the customer side |
| 02:10.35 | justinu | yeah, a drive failure would take their PBX down |
| 02:10.42 | _Sam-- | you are solid state? |
| 02:11.19 | _Sam-- | if you ever want to try it...i have a CF version |
| 02:11.27 | Dr-Linux | justinu: initialy you will manage their server for free? then after fixed time you will change them? |
| 02:11.33 | _Sam-- | uses CF as the / root...and backs up to USB thumb |
| 02:11.37 | justinu | Dr-Linux: not for free |
| 02:11.46 | justinu | _Sam--: i might take you up on that |
| 02:12.05 | ketanp | justinu: have you considered just being a reseller for fonality? |
| 02:12.13 | justinu | no |
| 02:12.17 | justinu | i don't know what fonality is |
| 02:12.35 | shido6 | anyone configured an avaya switch? |
| 02:12.48 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 02:12.57 | *** part/#asterisk litage (n=nick@203.220.55.70) |
| 02:13.03 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 02:13.08 | _Sam-- | why would you need fonality? |
| 02:13.16 | ketanp | just an option |
| 02:13.30 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 02:13.35 | _Sam-- | i think you could make more money building your own server / setup than trying to resell theirs? |
| 02:13.40 | _Sam-- | at least i think i could! |
| 02:14.46 | shido6 | thats a lot of time to put into the gui |
| 02:14.55 | ketanp | it's a tradeoff between spending time coding/doing support and spending time selling, i guess... (i'm not pushing the fonality solution, just throwing the idea out there) |
| 02:15.17 | _Sam-- | most of my customers arent qualified to try to operate a gui and configure their own stuff |
| 02:15.24 | ketanp | it's always worth weighing your options before diving into months worth of development which only you can support going forward |
| 02:20.07 | *** part/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 02:20.09 | *** join/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 02:20.22 | *** part/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 02:20.47 | *** join/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 02:24.04 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 02:24.23 | clyrrad | How can you change what ports remote sip phones connect on? |
| 02:24.31 | _Sam-- | hey bani what snom did you get |
| 02:25.11 | *** join/#asterisk mattwj2005 (n=matt@dialup-4.159.59.94.Dial1.Chicago1.Level3.net) |
| 02:25.49 | Dr-Linux | clyrrad: do you want to change SIP port? other than 5060 ? |
| 02:26.26 | *** join/#asterisk cassio (n=cassio@c91102b1.rjo.virtua.com.br) |
| 02:26.49 | clyrrad | Dr-Linux, yes but only for remote sip phones |
| 02:26.54 | cassio | what is happening, I have some broadvoice lines, one register but the others dont, this is the problem |
| 02:26.55 | cassio | Got SIP response 404 "Not found" back from 147.135.0.128 |
| 02:26.57 | clyrrad | not for internal phones they can stay at 5060 |
| 02:27.30 | clyrrad | cassio set your user=friend in iax.conf in the context that is giving you the problem |
| 02:27.53 | clyrrad | Dr-Linux, is it possible to set a range of ports for remote sip phones to connect on? |
| 02:27.58 | cassio | clyrrad I dont use iax for broadvoice |
| 02:28.05 | Dr-Linux | clyrrad: do PAT from your router |
| 02:28.06 | clyrrad | then do it in sip.conf |
| 02:28.16 | clyrrad | PAT? |
| 02:28.16 | Dr-Linux | or do it on the same machine with iptables |
| 02:28.29 | Dr-Linux | PAT= port address translation |
| 02:28.46 | cassio | clyrrad ok I will try |
| 02:28.52 | clyrrad | oh, but, the problem is I dont know what range of ports it will come in on to map them |
| 02:29.05 | _Sam-- | hey bani the manual for the snom360 shows all the config through the phone keypad...does it have a web interface? |
| 02:29.30 | Dr-Linux | clyrrad: it listens on only 1 port |
| 02:29.47 | justinu | it does have a web interface |
| 02:30.14 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:30.46 | _Sam-- | i think i have to buy one |
| 02:33.12 | _Sam-- | damn, bani is on the snom360 wiki page too, that guy gets around |
| 02:33.29 | cassio | suddenly my connections get lost, I dont have a firewall, can someone take a look at this paste? |
| 02:33.30 | _Sam-- | bani...how do you get any work done with all those feature requests |
| 02:33.40 | cassio | http://pastebin.com/544346 |
| 02:33.57 | [av]bani | _Sam--: i dont |
| 02:34.27 | _Sam-- | is the snom twice as good the gxp? |
| 02:34.30 | clyrrad | Dr-Linux do you know what port? I have 3 remote sip phones all registerd on different ports 60016, 51476, 60613, there must be a config file that specifies what range they can work on, or am I wrong? |
| 02:35.07 | _Sam-- | i need to start working with other phones so i can keep learning |
| 02:35.26 | cassio | does anyone know a good place to rent a dedicated server that works with asterisk? |
| 02:35.43 | clyrrad | cassio pm me |
| 02:35.52 | _Sam-- | i dont run asterisk at this place, but they have good dedicated servers and bandwidth: www.ev1servers.net |
| 02:35.56 | _Sam-- | i have a few machines there |
| 02:36.14 | cassio | sam do they work with asterisk? |
| 02:36.15 | Dr-Linux | cassio: the phone are registered? |
| 02:36.18 | Dr-Linux | sorry |
| 02:36.23 | Dr-Linux | clyrrad* ? |
| 02:36.24 | _Sam-- | they run linux...there's no reason they wouldnt |
| 02:36.58 | _Sam-- | this is one of their network backbone graphs: |
| 02:36.59 | _Sam-- | http://www.ev1servers.net/datacenters/dc1graphs.asp |
| 02:37.00 | clyrrad | Dr-Linux, i am trying to figure out how i can set what range of ports remote sip phones will connect on |
| 02:37.14 | cassio | sam: I rented a box at godaddy.com, its linux but this is whats going on, my connections are droping |
| 02:37.17 | _Sam-- | very well connected, in my opinion. |
| 02:37.25 | cassio | _Sam-- here http://pastebin.com/544346 |
| 02:37.29 | Dr-Linux | clyrrad: but you said those phone are registered |
| 02:38.07 | Dr-Linux | clyrrad: why you want to use other ports too ? |
| 02:38.26 | _Sam-- | clyrrad: do you have your own machine, or just a virtual machine? |
| 02:38.31 | _Sam-- | er cassio i mean |
| 02:38.50 | clyrrad | Dr-Linux so I can open the firewall |
| 02:38.51 | cassio | _Sam-- its a box at godaddy.com |
| 02:38.55 | Dr-Linux | clyrrad: my server listens on 2 UDP ports 5060 and 8080 |
| 02:38.58 | clyrrad | and have more than one asterisk server running off the same IP |
| 02:39.36 | _Sam-- | i dunno cassio...i pull in about 10mil a year in annual sales on servers that sit on ev1. |
| 02:39.40 | _Sam-- | and their crap works fine for me :) |
| 02:39.58 | _Sam-- | i cant speak for godaddy |
| 02:40.22 | _Sam-- | i think they spent too much money on their superbowl ads, now they cant have enough bandwidth for you :) |
| 02:40.48 | *** join/#asterisk _deg_ (n=deg@201.22.46.150.adsl.gvt.net.br) |
| 02:40.49 | Dr-Linux | clyrrad: simply use any other UDP port like 8585 and forward this port to 5060 in firewall |
| 02:41.04 | IronHelix | good morning |
| 02:41.38 | Dr-Linux | so in this case your server will listen on 2 ports |
| 02:41.54 | cassio | _Sam-- I have the same idea as you, now I am looking for a new server |
| 02:42.04 | *** part/#asterisk mattwj2005 (n=matt@dialup-4.159.59.94.Dial1.Chicago1.Level3.net) |
| 02:42.27 | _Sam-- | i think you could also look at rackspace.com |
| 02:42.33 | _Sam-- | but i dont have any personal experience with them |
| 02:42.38 | geesus | can a bridge have 2 ip's? |
| 02:42.40 | IronHelix | expensive+++ from what i recall |
| 02:42.52 | clyrrad | Dr-Linux, I understand that part, but here is what I am saying, if i knew what UDP port the remote phones where going to choose to connect on I could do that, but I dont know where they are choosing their port, so to setup that forward you are telling me about is impossible, know what i mean? |
| 02:43.16 | Mark_Halverson | i am using cari.net for hosting FC4 64 bit...works great and no prob with timing |
| 02:43.25 | Mark_Halverson | $59 / month |
| 02:43.59 | cassio | Mark_Halverson do you have asterisk there? |
| 02:44.10 | Mark_Halverson | sure do |
| 02:44.33 | Mark_Halverson | and here is my plug...order thru me I get credit...but I can also get you a 10mb unlimited port |
| 02:44.47 | _Sam-- | i would like to see their network backbone graphs, like ev1s |
| 02:45.14 | Mark_Halverson | i run 100+ sip calls and doesn't miss a beat |
| 02:45.27 | clyrrad | all at the same time? |
| 02:45.32 | Mark_Halverson | yes |
| 02:45.41 | clyrrad | what are your server specs? |
| 02:45.52 | Dr-Linux | clyrrad: sorry my english is not good |
| 02:46.09 | _Sam-- | iff you pay 59 bucks, its a celeron D 64bit |
| 02:46.28 | _Sam-- | and they give you 1200GB a month of transfer |
| 02:46.31 | clyrrad | Dr-Linux, all I want to know is where do you set the ports that remote sip phones will use to connect on |
| 02:46.32 | Mark_Halverson | $59 1gb ram / 160 HD / Cel D 64 bit --- that's correct sam |
| 02:46.38 | clyrrad | USD right? |
| 02:46.40 | [av]bani | _Sam--: the snom definitely sounds miles better than the gxp, and the construction is miles better |
| 02:46.56 | clyrrad | also do they give you RAID? |
| 02:47.07 | ketanp | clyrrad: what model phones? |
| 02:47.17 | Mark_Halverson | not for $59 bucks....but i have been very happy for several months |
| 02:47.19 | clyrrad | ketanp GNET |
| 02:47.19 | _Sam-- | clyrrad: yeah for the dual procs |
| 02:47.53 | clyrrad | do you own the server or just rent it? |
| 02:47.55 | _Sam-- | their prices are cheap at cari.net...but you usually do get what you pay for...but if it is working well for you , i cant argue with that. |
| 02:48.01 | ketanp | clyrrad: and are you talking about for the sip traffic or rtp? |
| 02:48.11 | clyrrad | ketanp for the rtp |
| 02:48.13 | Dr-Linux | clyrrad: sip has only one port by default 5060 ... |
| 02:48.29 | Dr-Linux | and there is rtp range 10000:20000 |
| 02:48.30 | Mark_Halverson | i have put several clients on it...i mean for $59 bucks...it's atleast worth a try |
| 02:48.34 | clyrrad | when i sip show peers i get different ports on the remote phones |
| 02:48.45 | Dr-Linux | ooo |
| 02:48.54 | _Sam-- | Mark_Halverson: what is an IP of your machine at cari.net? |
| 02:48.58 | clyrrad | Dr-Linux, yes i have made the change in rtp.conf, but they phones are NOT using the ports with-in that range |
| 02:48.59 | Dr-Linux | clyrrad: that doesn't matter .. those are source ports |
| 02:49.00 | _Sam-- | id like to see the routes to the place |
| 02:49.00 | justinu | that's the phone's source port. |
| 02:49.33 | clyrrad | Dr-Linux, i guess it's actually the SIP port i want to set not the rdp ports, but only for remote phones |
| 02:49.42 | clyrrad | they are connecting on all these IP's that I have not defined anywhere |
| 02:49.50 | clyrrad | sorry not IP's i mean ports |
| 02:49.54 | justinu | lol, this is such bullshit: " The Grandstream GXP-2000 is a next generation enterprise IP telephone based on industry open standards. Built on innovative technologies, Grandstream GXP-2000 features market leading superb sound quality, rich functionalities and excellent manageability at an affordable price." |
| 02:49.57 | Mark_Halverson | i have two ... which one you wanna hack??? lol |
| 02:50.07 | Dr-Linux | clyrrad: yeah i already told you the solution |
| 02:50.14 | _Sam-- | justinu: [av]bani wrote that |
| 02:50.17 | ketanp | clyrrad: that's the expected behavior |
| 02:50.17 | _Sam-- | j/k |
| 02:50.20 | justinu | lol |
| 02:50.30 | Dr-Linux | i suggest do not change the SIP port .. just PAT one more port to sip |
| 02:50.44 | clyrrad | Dr-Linux, perhaps i misunderstood you? I dont know what ports will be chosen, so to forward is impossible, please clarify what you meant |
| 02:50.45 | Mark_Halverson | i know...lol |
| 02:50.52 | IronHelix | haha |
| 02:50.53 | _Sam-- | Mark_Halverson : can you give me an IP or the name of your server at cari.net so i can traceroute there |
| 02:50.57 | _Sam-- | i want to see the connectivity |
| 02:51.03 | IronHelix | the GXP has decent hardware they just need to get their firwmare in order |
| 02:51.21 | clyrrad | ketanp, how can i change what ports it will use? I run 2 * boxes on this IP, and each need to have a range of ports forwareded to the machine of choice |
| 02:51.35 | IronHelix | they're trying though, i think the wiki page has been very helpful to them |
| 02:52.16 | IronHelix | sip.conf change the port, then rtp.conf change the ports.... but that doesnt change the phone, the phone will use whatever ports it wants to |
| 02:52.37 | clyrrad | IronHelix, yea thats my problem..... |
| 02:52.51 | IronHelix | that you have to fix on the phone itself |
| 02:53.02 | IronHelix | not all phones have a local SIP port or local RTP port option |
| 02:53.06 | Dr-Linux | clyrrad: you have 2 servers behind the firewall right? |
| 02:53.13 | justinu | some phones call it symmetric rtp |
| 02:53.14 | clyrrad | Dr-Linux yes |
| 02:53.22 | Dr-Linux | and you have public ip is same for both right? |
| 02:53.25 | ketanp | clyrrad: what is the core problem you are having? ignoring the port issue |
| 02:53.31 | clyrrad | and I loose voice becase i cant direct pakets properly |
| 02:54.16 | clyrrad | voice rtp packets i need to control what server they end up at, thats why I am trying to set the port the phones connect on, that way I can say between range A-B go to server 1, between C-D go to server 2 |
| 02:54.33 | [av]bani | so what FXO gateways everyone using? |
| 02:54.47 | ketanp | clyrrad: what is your router/firewall? |
| 02:54.54 | clyrrad | m0n0wall |
| 02:55.34 | Dr-Linux | clyrrad: then change serverB sip port or assign one other port beside the default one |
| 02:55.45 | clyrrad | Dr-Linux I have done that already |
| 02:55.56 | clyrrad | Server 1 is on 5060, Server 2 is on 5061 |
| 02:56.42 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 02:56.53 | websae | does anyone have a ATA adapter connected to their home phone system? |
| 02:57.08 | websae | I am wondering what the best adapter is to do this with? |
| 02:57.21 | IronHelix | anything by Sipura will do quite well |
| 02:57.32 | websae | my sipura 841 phone is terrible |
| 02:58.10 | IronHelix | the 841 is not great |
| 02:58.15 | IronHelix | but the ata's are very nice |
| 02:58.44 | websae | so the ata will work fine wiring it to home phone system? |
| 02:58.58 | IronHelix | yes just make damn sure that the telco interface is unplugged |
| 03:00.09 | *** join/#asterisk talljon84 (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 03:00.36 | websae | i heard that sometimes the ata is not powerful enough to make the phones all ring |
| 03:00.41 | websae | so i was just curious about that |
| 03:00.44 | IronHelix | ah |
| 03:00.46 | IronHelix | that depends |
| 03:00.51 | IronHelix | look up the REN of each of your phones |
| 03:01.00 | IronHelix | REN is how much ringing power the phone absorbs |
| 03:01.06 | IronHelix | old bell phones = 1.0REN |
| 03:01.16 | *** join/#asterisk jontow (i=jontow@bsd.adminforrent.com) |
| 03:01.17 | ketanp | clyrrad: i think you might be focusing on the wrong thing... i think you should be able to get this working without having to force the ports the remote phones use |
| 03:01.23 | websae | so cordless shoudl be fine |
| 03:01.27 | IronHelix | new electronic phones usually = 0.2-0.5REN |
| 03:01.39 | IronHelix | the ATA puts out as i recall 3.0-4.0REN |
| 03:01.46 | IronHelix | yeah a cordless phone or two will be fine |
| 03:02.05 | talljon84 | Anyone have suggestions on how to fix crackling/popping when MOH is playing? Voice works great and the problem only happens when playing music on hold. Using 1.2.4 with make mpg123. |
| 03:02.54 | SibRphrek | talljon84: going with RAW audio instead |
| 03:03.09 | websae | what can you get to make the ringing work if it doesn't? |
| 03:03.11 | websae | what sort of adapter |
| 03:03.25 | fugitivo | talljon84: use native moh |
| 03:03.30 | SibRphrek | websae: a lightbulb |
| 03:03.30 | tuxinator_linux | talljon84, I agree with SibRphrek, but I don't have any personal experience |
| 03:03.33 | IronHelix | i think Radio Shack has a ring booster adapter which adds like 5-8REN to your line |
| 03:03.49 | websae | are those expensive? |
| 03:03.51 | SibRphrek | talljon84: there's a great walk thru for it on voip-info.org or w/e that site is |
| 03:03.57 | Qwell | websae: not needed |
| 03:04.05 | Qwell | You could practically run your whole house on a single ATA |
| 03:04.13 | Qwell | especially if they're cordless |
| 03:04.13 | _Sam-- | file you there you geek? |
| 03:04.17 | websae | so sipura should be just fine then |
| 03:04.18 | websae | okay |
| 03:04.20 | Qwell | _Sam--: he's always there |
| 03:04.20 | IronHelix | not really, like $30, but qwell is right its highly unlikely that you will need one |
| 03:04.23 | talljon84 | SipRphrek: I'll take a look at voipinfo but I was not aware of any other MOH options than the musiconhold.conf |
| 03:04.43 | websae | thank you both for your help with the matter |
| 03:04.55 | IronHelix | no problem, good luck with your install! |
| 03:05.48 | *** join/#asterisk grumpie (n=vijay@dsl093-139-122.sfo4.dsl.speakeasy.net) |
| 03:07.06 | justinu | i wonder what kind of REN an old candlestick phone has |
| 03:08.26 | justinu | any of you ever run a softphone over an ssh tunnel? |
| 03:08.27 | tuxinator_linux | justinu, what about a tin cans and string, what's the REN on those? |
| 03:08.35 | justinu | specifically putty |
| 03:08.49 | justinu | tuxinator_linux: infinity? |
| 03:08.51 | *** join/#asterisk afu888 (n=jkr888@c-69-248-26-95.hsd1.nj.comcast.net) |
| 03:09.07 | justinu | i don't think string conducts electricity that well :P |
| 03:09.23 | _Sam-- | file why dont you fix (name) your IP so incoming calls look proper: Dial("IAX2/66.250.69.11:4569-8", "SIP/300&SIP/310& |
| 03:09.24 | tuxinator_linux | encrypting your communications, you might have the FBI show up on your door |
| 03:09.44 | justinu | it's more for network security, but |
| 03:09.51 | _Sam-- | or maybe its not about naming the IP |
| 03:09.52 | *** join/#asterisk kimosabe (n=kimosabe@dsl-201-128-128-191.prod-infinitum.com.mx) |
| 03:09.58 | _Sam-- | but my other incoming calls dont come in looking that way |
| 03:10.11 | kimosabe | is any one here running asterisk on an wrt54g model linksys ?? |
| 03:10.26 | Dr-Linux | _Sam--: irc and #asterisk are the enemy of your wife? |
| 03:10.31 | tuxinator_linux | you know, and I might have the FBI show up for saying it, but why are the terrorists not using encrypted communications |
| 03:10.37 | _Sam-- | how did you know?? that and motorcycles |
| 03:10.42 | file | what is with people and channel names lately? |
| 03:10.47 | justinu | tuxinator_linux: they are, i'm sure |
| 03:10.58 | _Sam-- | sorry |
| 03:11.01 | tuxinator_linux | justinu, I'd assume as much |
| 03:11.11 | file | espically billing off of the generated channel name |
| 03:11.15 | tuxinator_linux | then the whole phone tapping crap is useless |
| 03:11.30 | _Sam-- | alright...its just the first time i had seen the name as an IP...so i was like who is calling this way |
| 03:11.33 | justinu | der, it's just to catch stupid people |
| 03:11.43 | _Sam-- | i guess the client is finally starting to switch to you for origination |
| 03:11.55 | justinu | any real terrorist wouldn't be caught blabbing about an operation on a clear channel connection |
| 03:12.24 | tuxinator_linux | talking to my wife, I encrypt |
| 03:12.27 | justinu | it's the fact that most people are too stupid to realize it that's the problem. |
| 03:12.30 | Dr-Linux | _Sam--: i bet one day your wife will leave you bcoz of these things |
| 03:12.36 | Mark_Halverson | in the navy we used VoIP over encrypted VPNs on cisco routers |
| 03:12.38 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 03:12.39 | file | tuxinator_linux's wife is c00l |
| 03:12.40 | tuxinator_linux | stupid criminals are funny |
| 03:12.43 | _Sam-- | dr-linux that would be your lucky day, eh? :) |
| 03:12.50 | justinu | tuxinator_linux: in former east germany (GDR), they'd love to listen to husband and wife converse. |
| 03:12.56 | afu888 | Hello, I have a problem with one of the channel of digium TDM400P, all fxo, but some how one channel was bad, after reboot, I can only use the channel once, after that I get a noisy everytime i tried to dial...any help or direction much appreciated. |
| 03:13.06 | justinu | it gave them the most reliable info, because usually married people talk about everything |
| 03:13.08 | Dr-Linux | _Sam--: NO? i'm not after your wife |
| 03:13.12 | _Sam-- | lol! |
| 03:13.14 | tuxinator_linux | justinu, voyour's? |
| 03:13.25 | justinu | no, they just wanted to find out who was an enemy of the state |
| 03:13.32 | tuxinator_linux | ahh |
| 03:13.36 | Mark_Halverson | lol |
| 03:13.37 | justinu | who didn't like Dear Comrade Erich Honeker |
| 03:13.37 | file | I smell women, smell them in the air, think I'll drop my anchor in that harbor over there |
| 03:13.43 | _Sam-- | Dr-Linux: it seems your english AND your asterisk are both improving! |
| 03:14.01 | tuxinator_linux | file, women are good, don't want to live without oen |
| 03:14.16 | justinu | i like mine |
| 03:14.35 | Dr-Linux | _Sam--: but believe me you gonna lose her ... :S |
| 03:14.36 | tuxinator_linux | my wife likes to wear her "I LOVE MY GEEK" shirt |
| 03:14.41 | _Sam-- | it really IS amazing they put up with us, and all our IRC and phone obsessing |
| 03:15.00 | _Sam-- | my wifes been through this before when i started the two other businesses |
| 03:15.04 | _Sam-- | "im working" |
| 03:15.06 | Mark_Halverson | my wife want's to know why we can't have a regular phone....lol |
| 03:15.07 | justinu | it's amazing what a woman will put up with for a little financial security :P |
| 03:15.15 | tuxinator_linux | Mark_Halverson, Mines also |
| 03:15.16 | file | lvoely ladies! |
| 03:15.27 | tuxinator_linux | justinu, true |
| 03:15.34 | Mark_Halverson | and the phone company keeps asking, "you have what....a PBX in your house?" |
| 03:15.52 | justinu | you got a PRI at home yt? |
| 03:15.55 | justinu | yet |
| 03:16.02 | Mark_Halverson | TDM DS3 |
| 03:16.05 | justinu | heh |
| 03:16.06 | tuxinator_linux | I don't even have my apartment wired for phones |
| 03:16.07 | Mark_Halverson | and IP DS3 |
| 03:16.10 | Dr-Linux | _Sam--: i think you don't care is she leaves you, i think your eyes are on someone else as well :S |
| 03:16.16 | [TK]D-Fender | Mark_Halverson : My GF FLIPPED when she lost a call I tried trnsferring.... I pulled the whole setup down... |
| 03:16.24 | Mark_Halverson | 6 weeks..install man comes....can't wait |
| 03:16.36 | justinu | for a DS3 at home? |
| 03:16.44 | Mark_Halverson | correct ds3 at home |
| 03:16.48 | justinu | what'd that cost? |
| 03:16.51 | tuxinator_linux | Mark_Halverson, can we all move in? |
| 03:16.53 | Mark_Halverson | tdm |
| 03:16.58 | Mark_Halverson | To Damn Much |
| 03:17.00 | Mark_Halverson | lol |
| 03:17.07 | Mark_Halverson | DS3 as a CLEC is cheap |
| 03:17.23 | Mark_Halverson | IP DS3 I got a deal from a friend who owns an ISP...1800 |
| 03:17.31 | justinu | what about loop? |
| 03:17.32 | _Sam-- | should be interesting when they have to drag fiber to your house |
| 03:17.36 | tuxinator_linux | what's 1800? |
| 03:17.38 | Mark_Halverson | coax |
| 03:17.41 | Mark_Halverson | not fiber |
| 03:17.44 | _Sam-- | oh i see |
| 03:17.49 | file | I'm not that obsessive... about my telephony and internet needs... |
| 03:17.50 | justinu | yeah, but what is the ILEC charging for the loop? |
| 03:17.51 | IronHelix | 1800 is the monthly fee i think... |
| 03:18.00 | justinu | 1800? |
| 03:18.11 | Mark_Halverson | business....make money gota spend money |
| 03:18.17 | Mark_Halverson | brb...wife calling |
| 03:18.38 | justinu | you gotta generator to back all that HICAP equipment up? |
| 03:19.42 | _Sam-- | in the old days (5-10 yrs ago) all DS3 used come in on fiber? |
| 03:20.00 | _Thor | I've got a dedicated fiber optic line to my office |
| 03:20.16 | Dandan | i want one too! |
| 03:20.17 | [av]bani | _Sam--: no, fiber is a recent phenomenon for that. it was almost always coax |
| 03:20.29 | _Sam-- | but oc3 was fiber? |
| 03:20.37 | _Thor | not at all |
| 03:20.41 | [TK]D-Fender | y0 file... |
| 03:21.01 | _Thor | you can fit I don't know how many OC3s in a fiber optic... too many |
| 03:21.30 | _Sam-- | i thought OC stood for optical carrier |
| 03:23.45 | [av]bani | its a framing standard |
| 03:24.11 | _Sam-- | so , an OC3 is merely any line that does 155mbps and uses that framing? |
| 03:24.23 | _Sam-- | it has nothing to do with fiber? |
| 03:24.43 | [av]bani | sorta like ds3 can be fiber or coax |
| 03:24.54 | [av]bani | or ethernet can be fiber |
| 03:25.10 | _Sam-- | i get it |
| 03:25.18 | [av]bani | hell, you can even do fddi over coax... |
| 03:25.39 | _Sam-- | how fast can can coax go? |
| 03:25.48 | [av]bani | how fast you want to go? |
| 03:26.04 | _Sam-- | it can go OC48 fast? |
| 03:26.07 | file | I don't want to get stopped by the cops |
| 03:26.13 | [av]bani | ask your cable company |
| 03:26.23 | Dr-Linux | _Sam--: i was kidding i hope you will not mind |
| 03:26.33 | _Sam-- | Dr-Linux : not mind 1 bit, i enjoy it! |
| 03:26.52 | Dr-Linux | _Sam--: great! :) |
| 03:26.56 | [av]bani | how much bandwidth is ther ein the cable feeding 400 channels of video into your house? |
| 03:27.25 | _Sam-- | how many kbps to do a digital channel? |
| 03:27.29 | Qwell | 4 |
| 03:27.43 | _Sam-- | for a digital channel on your cable tv? 4kbps? |
| 03:27.48 | _Sam-- | that is efficient! |
| 03:27.51 | Qwell | no |
| 03:31.43 | _Sam-- | how active are the asterisk mailing lists? |
| 03:31.48 | _Sam-- | 100s of messages per day? |
| 03:32.19 | AlexCTI | <PROTECTED> |
| 03:34.38 | [TK]D-Fender | iptables :) |
| 03:35.55 | tholo | OpenBSD/pf |
| 03:37.26 | [TK]D-Fender | Mastercraft Industrial Wire-Cutters? |
| 03:39.57 | Mark_Halverson | okay back |
| 03:40.48 | Mark_Halverson | why is it a wife's honey doos never include computer time |
| 03:40.49 | Mark_Halverson | lol |
| 03:41.32 | fafnir | is there any sort of pgpish encryption for asterisk? |
| 03:42.33 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 03:42.36 | Flyboy-SR22 | Hey Everyone - ANyone using A@H with Sugar to dial out...? |
| 03:42.49 | *** part/#asterisk AlexCTI (n=alex@weston-69.65.84.104.myacc.net) |
| 03:42.53 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 03:42.58 | cassio | guys I need a place to rent a dedicated server for asterisk, anyone know any other? |
| 03:43.04 | tuxinator_linux | Mark_Halverson, my wife said "cause your on there all the time" |
| 03:43.29 | tuxinator_linux | cassio, asterlink? |
| 03:43.59 | tuxinator_linux | s/your/you're |
| 03:44.11 | file | we don't rent dedicated servers sadly |
| 03:44.12 | Flyboy-SR22 | I can make calls no problem from my extension, but when I hit the Call button in SugarCRM, I get an Invalid Channel error |
| 03:44.26 | tuxinator_linux | file, but you co-location, right? |
| 03:44.33 | file | yes |
| 03:44.54 | Qwell | I need a colo |
| 03:45.06 | Mark_Halverson | Flyboy: I used to...not anymore |
| 03:45.27 | Flyboy-SR22 | humm... |
| 03:45.29 | tuxinator_linux | Did I see that Asterlink has toll free origination for .02 USD? |
| 03:45.38 | Qwell | tuxinator_linux: yes |
| 03:45.39 | Flyboy-SR22 | can't seem to find any logs that tell me what the problem is.... |
| 03:45.55 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 03:46.15 | tuxinator_linux | is the toll free origination for calls coming in on the toll free number (I think I understand it now)? |
| 03:46.17 | Flyboy-SR22 | Qwell, Where do you want to colo..? |
| 03:46.18 | kpettit | I just started trying to do the RealTime stuff with extensions.conf |
| 03:46.22 | file | tuxinator_linux: aye |
| 03:46.25 | Qwell | Flyboy-SR22: my apartment |
| 03:46.29 | Flyboy-SR22 | HA |
| 03:46.33 | file | Qwell: Qwelllllllllllllllll |
| 03:46.35 | kpettit | were do I get the debug info for that? I'm getting MySQL errors but I know the MySQL login info is right |
| 03:46.35 | Flyboy-SR22 | I have a T1 to my house :-) |
| 03:46.42 | Qwell | file: Can you get me a DS3 to my house? :( |
| 03:46.49 | file | Qwell: yes, but it'll cost you! |
| 03:46.49 | Flyboy-SR22 | Its only $$$ |
| 03:46.56 | Flyboy-SR22 | the company pays for my T1 |
| 03:46.57 | Qwell | file: MikeJ said he'd pay! |
| 03:47.00 | Flyboy-SR22 | only reason I have it |
| 03:47.02 | kpettit | ERROR[8275]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on.... |
| 03:47.04 | file | Qwell: rrrrrrrrrrright |
| 03:47.08 | Qwell | fine :( |
| 03:47.09 | tuxinator_linux | Flyboy-SR22, what is the reason for your T1? |
| 03:47.46 | Flyboy-SR22 | tuxinator_linux, Just to have it...I'm the CTO here, so its kinda a perk...working for an ISP has its good points ! |
| 03:47.56 | tuxinator_linux | ahhh, got it |
| 03:48.14 | Flyboy-SR22 | tuxinator_linux, but its nothing to the Seven Gig backbones we have here in the data center :_) |
| 03:48.31 | tuxinator_linux | I did web hosting 10 years ago, and had a T1 to my house, a bit more expensive then, and that was before broadband for homes |
| 03:48.40 | *** join/#asterisk bmg505 (n=leon@dsl-146-18-213.telkomadsl.co.za) |
| 03:48.58 | Flyboy-SR22 | tuxinator_linux, I could do with cable modem no problem, but the T1 gets me behind the firewall without a VPN, so its kinda nice |
| 03:49.03 | *** join/#asterisk daguerro (i=daguerro@lss-67-68.ee.itb.ac.id) |
| 03:49.14 | Flyboy-SR22 | tuxinator_linux, plus I am the canary in the cave so to speak sometimes |
| 03:49.14 | tuxinator_linux | Flyboy-SR22, that would be handy |
| 03:49.54 | tuxinator_linux | with some many community getting fiber to the house, it's crazy, bandwith is dirt cheap |
| 03:50.05 | tuxinator_linux | s/some/so |
| 03:50.36 | Flyboy-SR22 | I am getting limited logging in when I run asterisk -rvvvvvvvvv I am trying to figure out why calls from SugarCRM are failing with an Invalid Channel, but I am not sure what logs to look in, anyone have any ideas..? |
| 03:50.50 | tuxinator_linux | pretty soon our refrigerators will be talking to the toaster in china |
| 03:50.52 | Flyboy-SR22 | Here in California, they are putting fiber in faster than you can imagine |
| 03:51.14 | tuxinator_linux | Flyboy-SR22, I am in SoCal |
| 03:51.37 | Flyboy-SR22 | tuxinator_linux, I live in Escondido and our Data Center is in the San Diego Tech Center off the 5 and 805 |
| 03:51.48 | *** join/#asterisk ryguy (n=rgwillia@c-68-85-72-17.hsd1.fl.comcast.net) |
| 03:51.50 | Nugget | Everyone remember to track their flights on flightaware.com! :) |
| 03:51.58 | file | Nugget: you *know* I am already :P |
| 03:52.03 | Nugget | yay |
| 03:52.13 | Qwell | from the air? |
| 03:52.18 | tuxinator_linux | Nemesis760, I do |
| 03:52.22 | tuxinator_linux | oops |
| 03:52.28 | tuxinator_linux | Nugget, I do |
| 03:52.30 | Nugget | yay |
| 03:53.46 | clyrrad | This is the damn stragest thing I have seen as of yet. I have 2 phones connected to my PBX, they are able to ring each other no problem. There is NO audio, and even stranger is when Phone A rings Phone B, and you answer on B, you still hear the ringing sound on A. However when B ring A, and you answer on A, you dont hear ringing sound, but you still have no audio. Any suggestions :) |
| 03:53.58 | tuxinator_linux | Nugget, why are you advertising flightaware? |
| 03:54.03 | Nugget | it's my site |
| 03:54.08 | Qwell | eh? |
| 03:54.13 | file | Nugget's cool so he can do that |
| 03:54.20 | Qwell | Where do you get your data? |
| 03:54.24 | Nugget | the faa |
| 03:54.32 | ryguy | does anyone know of any bugs in asterisk that would cause random hangup's when upgrading from 1.0.9 to 1.2.1 (same with 1.2.3) on both sip lines and zap lines |
| 03:54.52 | tuxinator_linux | Nugget, why were you interested in doing that project? |
| 03:55.03 | Nugget | It seemed like a fun thing to do at the time |
| 03:55.08 | tuxinator_linux | oh, look, you do IFR also |
| 03:55.13 | Nugget | naturally |
| 03:55.20 | Nugget | commercial traffic is boring. :) |
| 03:55.21 | Qwell | Nugget: When do we get to see live webcams? |
| 03:55.26 | tuxinator_linux | I need to get IRF certified |
| 03:55.37 | Flyboy-SR22 | :-) |
| 03:55.59 | Nugget | I need to actually suck it up and finish my ASEL. I'm just a staller (har har) student. |
| 03:56.05 | *** part/#asterisk cassio (n=cassio@c91102b1.rjo.virtua.com.br) |
| 03:56.08 | Nugget | stalled, even. |
| 03:56.14 | talljon84 | I'm still working on the crackle/pop music on hold problems. I have tried using mode => native and mode => custom with an application => /usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s but both give the same crackle/pop. If I am reading it correctly, the first is the "native" mode. What else could be wrong or am I missing something? |
| 03:56.17 | Flyboy-SR22 | Nugget, HA |
| 03:56.17 | tuxinator_linux | I'm looking for a CFI. |
| 03:56.25 | ryguy | nugget, we have a customer that does similar (airports.com) but there are quite a few cool features on your site that they do not have |
| 03:56.26 | Qwell | http://flightaware.com/live/flight/SKW6178 9 minute flight? |
| 03:56.47 | Flyboy-SR22 | tuxinator_linux, It fun, but not a lot of time...I keep my CFI/CFII current doing BFRs with other insturcotrs :-) |
| 03:56.51 | talljon84 | Correction, mode => files and mode => custom |
| 03:57.01 | dlynes | talljon: do you have any hardware supported by the zaptel drivers? |
| 03:57.18 | talljon84 | TDM400, one FXO module install so far. |
| 03:57.19 | Nugget | heh |
| 03:57.25 | tuxinator_linux | Flyboy-SR22, I just need a couple hours every few weeks, still working on finishing my private |
| 03:57.25 | tronix | (stall recovery) |
| 03:57.42 | tronix | should do an * fly-in (hahaha) some time |
| 03:57.43 | file | Nugget: you're like that every day |
| 03:57.43 | *** join/#asterisk zekeonfir3 (n=zekeonfi@c-67-164-250-173.hsd1.ut.comcast.net) |
| 03:57.58 | tuxinator_linux | Nugget is flying a too high of an altitute, needs supplimental oxygen |
| 03:58.10 | Flyboy-SR22 | tuxinator_linux, WHat airport do you fly out of..? |
| 03:58.13 | dlynes | talljon: no idea from me, then |
| 03:58.18 | tuxinator_linux | I know Ariel_ is also a pilot |
| 03:58.34 | tronix | justinu too |
| 03:58.40 | tuxinator_linux | I used to fly out of KIWA, in Arizona |
| 03:58.51 | Nugget | strange that we have so many pilots in here, given the totally unrelated channel focus. |
| 03:58.52 | Flyboy-SR22 | http://richard.sears.net/?p=19#more-19 |
| 03:58.54 | dlynes | talljon: well, other than that your mp3 that you're playing for music on hold might just sound crappy to begin with |
| 03:58.55 | tuxinator_linux | my last fights have been out of Cable and Fullerton |
| 03:59.06 | Flyboy-SR22 | ah... |
| 03:59.08 | trelane | anyone using cisco 7905 phones? |
| 03:59.15 | talljon84 | dlynes: I'm using the default music files that come with Asterisk. |
| 03:59.23 | talljon84 | trelane: no but I do have a 7912 |
| 03:59.26 | Flyboy-SR22 | one of my jobs is to fly my business partners and the sales people around...heading to Ontario on Thursday |
| 03:59.38 | tuxinator_linux | that's by me |
| 03:59.43 | dlynes | talljon: no idea then, sorry...but then again i haven't dealt much with MOH, either |
| 03:59.44 | tronix | did fly from a tiny strip SE of Phoenix once at 5am for a nice view and avoid insane summer heat |
| 03:59.45 | trelane | talljon84, those are fairly similar, are you using TFTP? |
| 03:59.54 | tronix | shortest/smallest runway I ever saw |
| 03:59.58 | talljon84 | yeah |
| 04:00.03 | tronix | 1700ft by 40ft |
| 04:00.09 | ryguy | can anyone point me to where I should be looking to find out why I would be dropping calls on both sip (connecting to 7940's and to a t1 via an as5400 for inbound call termination) as well as zap lines connected to both a t1 card and a channelbank for an analog extension |
| 04:00.17 | Qwell | tuxinator_linux: fly us to VON next month :P |
| 04:00.27 | tuxinator_linux | Qwell, yes, I would do that |
| 04:00.31 | Qwell | except? |
| 04:00.35 | tuxinator_linux | no except |
| 04:00.41 | tuxinator_linux | I have my VON ticket |
| 04:00.43 | Qwell | oh |
| 04:00.56 | Qwell | You are flying there yourself? heh |
| 04:00.59 | tuxinator_linux | just need to crash at someones hotel |
| 04:01.04 | Flyboy-SR22 | ouch |
| 04:01.06 | talljon84 | trelane: are you using a Skinny or SIP software image on the 7905? |
| 04:01.26 | tuxinator_linux | I would love to carpool/planepool |
| 04:01.40 | file | WELL |
| 04:01.52 | file | Qwell is doing the same as you :) |
| 04:01.59 | Qwell | tuxinator_linux: If you're flying/driving, let me know if you want company |
| 04:02.00 | trelane | talljon84, sip |
| 04:02.02 | file | so how fitting! |
| 04:02.16 | tuxinator_linux | Qwell, I was planning on driving and would like company |
| 04:02.21 | trelane | talljon84, I'm having registration issues with the phones, 2 days now and it's driving me batty. |
| 04:02.26 | Qwell | probably a lot cheaper than flying.. |
| 04:02.34 | Nugget | but less fun :) |
| 04:02.35 | Qwell | commercial flight, that is |
| 04:02.43 | file | Qwell and tuxinator_linux are geeks |
| 04:02.44 | tuxinator_linux | and I can take a sleeping back with me in the car, so I can crash on someones floor |
| 04:02.50 | file | they'll have GPRS |
| 04:02.51 | tuxinator_linux | file, Aye, that I am |
| 04:02.55 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 04:02.57 | talljon84 | trelane: the 7912/7905 use binary configuration files, even in SIP mode. You need the editor to "build" their config file |
| 04:03.12 | trelane | talljon84, I have that, I need a functional config as a reference so I can figure out where mine's borked |
| 04:03.13 | znoG | is there a way, with AEL, to call a "label" and return when it's done? |
| 04:03.16 | tuxinator_linux | sleeping back, not back |
| 04:03.21 | Qwell | heh |
| 04:03.27 | file | tuxinator_linux: anyway finding somewhere for you to sleep shouldn't be a problem |
| 04:03.41 | trelane | talljon84, two letter prefix mac address.txt and cfgmkr.linux <filename> |
| 04:03.59 | tuxinator_linux | file, be safe if I just show up? |
| 04:03.59 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 04:04.02 | trelane | talljon84, believe me man I've put in the config time, this phone is driving me bloody bonkers |
| 04:04.04 | znoG | eg. i might wanna call a label like: receptionists: { ... }; and I want it to return and keep execution when its done |
| 04:04.08 | file | tuxinator_linux: I'll find something for you tomorrow |
| 04:04.35 | tuxinator_linux | Happy tux I am:-) |
| 04:04.46 | file | I just need to talk to someone |
| 04:05.14 | talljon84 | trelane, you should have two files just to confirm a gkdefault.cfg and a gk<MAC>.cfg .. or whatever the prefix for 7905 is |
| 04:05.23 | trelane | right |
| 04:05.34 | dahunter3 | I have this weird telco that will be for e&m wink, will be sending a wink, then *10-12 digit callerid*DIDnumber* Will asterisk be able to catch that information appropriately? |
| 04:05.38 | trelane | and it loads the image and the config via tftp |
| 04:06.00 | trelane | and then tries to register to the PBX but it's not sending the right login info, it's trying to authenticate (as best as asterisk tells me as username = phone) |
| 04:06.13 | trelane | I'm either missing an option or have a typo and I can't determine which |
| 04:06.15 | file | dahunter3: you might have to mangle it in your dialplan... I'm not an e&m wink expert |
| 04:06.22 | trelane | a reference/working config will help me diagnose and fix the problem |
| 04:07.05 | trelane | plaintext config |
| 04:07.08 | dahunter3 | file: I'm mainly just trying to make sure that I can catch that callerid information appropriately... would I still need to mess with my dialplan? |
| 04:07.14 | trelane | the .txt not the .cfg |
| 04:07.35 | file | dahunter3: I dunno how it'll come in really :) |
| 04:07.37 | talljon84 | trelane, correct, the LoginID parameter must be the SIP name. |
| 04:07.57 | talljon84 | trelane, my bad, the UID field |
| 04:08.09 | file | ah the lion king... |
| 04:08.19 | talljon84 | trelane, example: UID:200 |
| 04:08.58 | talljon84 | trelane, also besure SIPRegOn:1 it has to register to asterisk |
| 04:11.04 | talljon84 | trelane, I'm not sure if this is standard with the config utility but I have a sipexample.txt file which lists all of the parameters if you want it. |
| 04:11.05 | trelane | right I realize that |
| 04:11.18 | trelane | talljon84, that'd be immensely appreciated |
| 04:11.44 | trelane | sipregon:1 is active |
| 04:12.48 | znoG | so how would you guys do it? say you have a routine to follow to dial operators (ie. dial one guy, then try another guy, then another guy, and finally some other guy. I want to call this routine from several places, but return to the place they were called from once they're done. Can it be done easily in AEL? |
| 04:12.52 | trelane | talljon84, mind e-mailing it to trelane@trelane.net? |
| 04:13.16 | talljon84 | trelane, yeah, give me a min or two to get it off of the asterisk box. |
| 04:14.04 | trelane | talljon84, many thanks man, this has been a two day 16 hour war |
| 04:14.28 | talljon84 | haha. np. now answer my bloody MOH problem and we'll all be set! |
| 04:15.39 | *** join/#asterisk Myconid3 (n=myconid@69-164-122-221.sbtnvt.adelphia.net) |
| 04:16.48 | *** part/#asterisk Myconid3 (n=myconid@69-164-122-221.sbtnvt.adelphia.net) |
| 04:18.04 | znoG | anyone using AEL2 in production? |
| 04:18.07 | Flyboy-SR22 | If I wanted to force this $exten = $PhoneCall; to add a '1' in the front of the number represented by $PhoneCall, any idea how I could do it..? |
| 04:18.26 | Flyboy-SR22 | so $exten = $PhoneCall; = $exten = 1$PhoneCall; |
| 04:21.39 | talljon84 | trelane, it's on it's way |
| 04:21.52 | trelane | talljon84, again many thanks! |
| 04:22.06 | talljon84 | good luck with the options, it's a long list. |
| 04:23.51 | trelane | talljon84, do you have a plaintext config for asterisk by chance? |
| 04:25.37 | trelane | for your current phones (lose the passwords of course) |
| 04:26.36 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 04:26.51 | *** join/#asterisk xblurone (n=me@202.42.66.33) |
| 04:26.59 | xblurone | Hello |
| 04:27.46 | talljon84 | trelane, let me look to see if I still have one. If not I need to config one anyways so give me a little while |
| 04:27.51 | Flyboy-SR22 | Hey xblurone |
| 04:27.52 | xblurone | Can somebody please give me the link (I remember reading it somewhere before) on how to read the cuase of an asterisk crash from the core dump? |
| 04:28.08 | trelane | talljon84, the corrosponding sip.conf entry would also be useful, again, thanks! |
| 04:28.11 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-42-138.cybersurf.com) |
| 04:28.29 | xblurone | running SVN-trunk-r7498 |
| 04:28.41 | talljon84 | trelane, the sip.conf entry is a normal "friend" entry as you would with any sip endpoint. nothing special there |
| 04:28.49 | xblurone | [Feb 8 11:48:26] NOTICE[5529]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Local' (cause 0 - Unknown) |
| 04:28.50 | xblurone | asterisk*CLI> |
| 04:28.50 | xblurone | Disconnected from Asterisk server |
| 04:28.50 | xblurone | Attempting to reconnect for 30 seconds |
| 04:28.50 | xblurone | Reconnect succeeded after 0.050 seconds |
| 04:29.09 | trelane | talljon84, then disregard that part |
| 04:29.11 | xblurone | so, I suspect app_dial.c |
| 04:30.14 | Flyboy-SR22 | figured it out |
| 04:30.16 | Flyboy-SR22 | l> $exten = preg_replace("/^([^1])/", "1\\1", $PhoneCall) |
| 04:30.19 | Flyboy-SR22 | woohoo |
| 04:30.53 | afu888 | Hello, I have a problem with one of the channel of digium TDM400P, all fxo, but some how one channel was bad, after reboot, I can only use the channel once, after that I get a noisy everytime i tried to dial...any help or direction much appreciated. |
| 04:30.55 | watchy | i love you |
| 04:30.56 | xblurone | ?? |
| 04:31.00 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 04:31.30 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 04:31.38 | tuxinator_linux | watchy, who's your daddy |
| 04:32.33 | xblurone | Sorry, my question too stupid? :) |
| 04:32.34 | watchy | you are |
| 04:32.41 | tuxinator_linux | xblurone, maybe do a pastebin |
| 04:32.48 | afu888 | where should i start to look, just don't know where to start ... from cli, it look like nothing wrong |
| 04:32.55 | tuxinator_linux | xblurone, easier to read then |
| 04:33.21 | xblurone | tuxinator, please clarify pastebin ? |
| 04:33.34 | tuxinator_linux | ~pb |
| 04:33.36 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 04:33.59 | xblurone | ok, sorry |
| 04:35.31 | znoG | AEL2 looks quite nice, the guy behind it seems to be pretty active |
| 04:35.33 | xblurone | pasted it there under name xblurone |
| 04:35.55 | tuxinator_linux | xblurone, what's the link |
| 04:36.16 | xblurone | http://pastebin.com/544482 |
| 04:37.17 | tuxinator_linux | xblurone, now ask your question |
| 04:38.13 | xblurone | ok, my question is, asterisk just core-dumped and restarted... I have the core file. Obviously this is a bug somewhere (I suspect app_dial.c) which causes this to happen so now and then (every day 2-3 times) |
| 04:38.49 | xblurone | ... how do I read the core to assertain where it went wrong and how can I be helpful submitting this the best way? |
| 04:38.57 | tuxinator_linux | which version are you running? |
| 04:39.18 | xblurone | SVN-trunk-r7498 |
| 04:39.25 | *** part/#asterisk rephorm (n=rephorm@216-188-230-85.dyn.grandenetworks.net) |
| 04:40.34 | tuxinator_linux | if you're learning, best to be using stable (1.2.4) |
| 04:41.04 | xblurone | I am running linux already for more than 1.5 years, not learning by any means |
| 04:41.16 | xblurone | just want to have this bug fixed :-) |
| 04:41.17 | tuxinator_linux | learing asterisk, that is |
| 04:41.51 | xblurone | Stable 1.2.4 would have all that is included in CVS until that date, correct? |
| 04:42.08 | Telamon | xblurone: Not necessarily. |
| 04:42.13 | xblurone | hmmmm |
| 04:42.16 | tuxinator_linux | stable is a branch, if I understand it correctly |
| 04:42.29 | X10ZION | Anyone by chance why I might be getting "Action: Logoff Response: Error Message: Missing action in request" in AMI |
| 04:42.37 | Telamon | xblurone: There are feature patches and such that go into head that don't go into what gets marked as stable. |
| 04:43.11 | tuxinator_linux | unless you can code in C, you're better off just using stable |
| 04:43.52 | tuxinator_linux | too much trouble otherwise |
| 04:45.27 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 04:46.21 | Corydon76-home | tuxinator_linux: there's no such thing as "stable" |
| 04:46.33 | Corydon76-home | There's "trunk" and "release" |
| 04:47.57 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 04:48.41 | xblurone | Well, since I have been running cvs forever now and written various applications for it i do not know what would happen if i switch to stable |
| 04:49.08 | xblurone | so, how to report this bug and "read" the core dump? (I can read&write C) |
| 04:49.09 | tuxinator_linux | Corydon76-home, I stand corrected |
| 04:51.12 | tuxinator_linux | are you running the lasted trunk? |
| 04:51.34 | tuxinator_linux | if I am not mistaken, they don't want bug reports unless you are running the latest trunk |
| 04:53.05 | tuxinator_linux | as the problem may have already been fixed |
| 04:53.11 | benjk | ~seen jerjer |
| 04:53.24 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 20d 7h 22m 30s ago, saying: 'thanks again'. |
| 04:53.24 | dogtanian | !seen jerjer |
| 04:53.35 | dogtanian | ;) |
| 04:53.38 | benjk | !seen shido6 |
| 04:53.43 | tuxinator_linux | jerjer is MIA |
| 04:54.11 | benjk | does anybody know for how long the NuFone DID ordering is offline? |
| 04:54.45 | tronix | xblurone: make sure asterisk was compiled with -g then run 'gdb asterisk core' or something, and do 'bt' command to get a stack backtrace |
| 04:55.27 | dogtanian | i use ipkall to get incoming DID |
| 04:55.31 | dogtanian | it's free too |
| 04:56.00 | Flyboy-SR22 | benjk, been awhile... |
| 04:56.08 | Flyboy-SR22 | benjk, I have been waiting for some time |
| 04:56.13 | *** join/#asterisk stealthmethod (n=nop@adsl-9-62-45.mia.bellsouth.net) |
| 04:58.12 | X10ZION | Anyone ever get from "Action: Logoff" - "Response: Error" "Message: Missing action in request" in Asterisk Call Manager |
| 04:58.24 | tuxinator_linux | dogtanian, how does ipkall afford to connect to the PSTN network and give out free DID's? |
| 04:58.35 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:59.06 | talljon84 | trelane, the other file is on the way |
| 04:59.34 | trelane | talljon84, again thanks |
| 05:00.15 | zekeonfir3 | benjk et al.: you use NuFone? Have you checked out iax.cc (sixtel)? I'm with them... $1.49/mo/DID I've been happy so far. |
| 05:00.48 | xblurone | ok, got it... SegFault... hmmm... |
| 05:01.41 | *** join/#asterisk daguerro (i=daguerro@lss-67-68.ee.itb.ac.id) |
| 05:02.02 | xblurone | let me upgrade to the latest trunk and see what happens... |
| 05:05.27 | tronix | xblurone: might be useful to strace -f -o /path/to/truss/logfile -p <asterisk pid> or to pastebin the gdb 'bt' output |
| 05:06.15 | tronix | segfault only tells you that you have a bad pointer somewhere... bt would at least narrow down to which function it's dying in |
| 05:06.21 | tronix | and with what arguments, etc |
| 05:06.38 | dogtanian | tuxinator_linux: I have no idea - they may pick fractions of a cent/minuite by terminating incoming calls, but i'n not sure what their profit-model is |
| 05:07.46 | dogtanian | tuxinator_linux: needless to say i'm not inclined to hand-out my US ipkall number as my 'permenant' contact number... just in case thir lack of money suddenly dawns on them |
| 05:07.51 | tuxinator_linux | dogtanian, when you see companies like that, you have to think, they must really be operating a porn site underneath to offord to give away so much stuff |
| 05:07.58 | dogtanian | lol |
| 05:08.01 | dogtanian | quite |
| 05:08.09 | xblurone | tronix, it's in http://pastebin.com/544531 |
| 05:09.12 | dogtanian | although if i'm getting a free number, i can deal with porn activities... as long as people who call me don't accidently get connected to a sweedish fetish chat-line |
| 05:10.20 | tronix | xblurone: what distro is it? |
| 05:10.28 | tronix | xblurone: fedora core 4? |
| 05:10.32 | xblurone | SVN-trunk-r7498 |
| 05:10.48 | tronix | xblurone: ahh.. meant the "operating system" :) debian, gentoo, etc |
| 05:11.02 | xblurone | CentOS release 4.1 (Final) |
| 05:11.11 | tronix | xblurone: ok. one sec |
| 05:11.21 | tronix | anyone familiar with CentOS 4 -- does it require NPTL? |
| 05:11.32 | tronix | for all the packages |
| 05:11.36 | xblurone | NPTL? |
| 05:11.42 | tronix | native posix threading library |
| 05:11.49 | tronix | that's the /lib/tls stuff in pb output |
| 05:12.01 | xblurone | mmmkay |
| 05:12.01 | tronix | it's infamous for causing this kind of problem |
| 05:12.25 | tuxinator_linux | I have run * on Centos 4.1 |
| 05:12.31 | tronix | hm. ok. hmm. |
| 05:12.45 | tronix | guess that makes sense... asterisk @ home is CentOS 4-based, after all |
| 05:12.51 | tuxinator_linux | ;-) |
| 05:13.01 | tronix | was just not sure if NPTL was optional or mandatory with CentOS |
| 05:13.10 | tronix | optional on Gentoo, mandatory on FC4 (iirc) |
| 05:13.17 | xblurone | there's no package with posix installed anyway. |
| 05:13.28 | xblurone | <PROTECTED> |
| 05:13.32 | xblurone | gives me nothing |
| 05:13.49 | tronix | xblurone: it's a little more complex than that. I'm not sure if all of CentOS applications |
| 05:13.52 | tronix | are linked against libtls or not |
| 05:14.18 | tronix | if CentOS can run without NPTL then asterisk might work if you mv /lib/tls /lib/tls.disabled |
| 05:14.21 | xblurone | only have gnutls-1.0.20-3.2.1 |
| 05:14.24 | tronix | but I'm not sure if it will break your system |
| 05:15.01 | xblurone | yeah, a bit tricky on a live system, but I have a backup, lemme see if it breaks the other idle one... |
| 05:15.08 | tronix | ok |
| 05:16.08 | *** join/#asterisk N0S3 (n=cristian@host210.200-43-204.telecom.net.ar) |
| 05:18.09 | xblurone | well, the other system boots :-) |
| 05:21.26 | benjk | zekeonfir3: how do they charge? do they accept Paypal? |
| 05:21.33 | Flyboy-SR22 | Good Night Everyone - time to head home :-) |
| 05:22.01 | Ayano | When I call out my zap card it takes about 15 seconds befor the other side can hear me, any clue why? |
| 05:22.45 | tuxinator_linux | Nigh Flyboy-SR22 |
| 05:22.52 | *** part/#asterisk N0S3 (n=cristian@host210.200-43-204.telecom.net.ar) |
| 05:23.02 | Flyboy-SR22 | see you all tomorrow - happy flying |
| 05:24.17 | tuxinator_linux | Wish I could do more of it, flying |
| 05:24.34 | tuxinator_linux | need a CFI, so I can finish my private |
| 05:24.52 | tuxinator_linux | but go home, or you'll never leave |
| 05:26.26 | Flyboy-SR22 | :-) |
| 05:29.00 | zekeonfir3 | benjk: they have variable rates from 1.7c/min to 1.47c/min you pay in advance toll-free#s are .20c/min vanity toll-free are $10 setup & $.20/mo @1.9c/min |
| 05:29.15 | zekeonfir3 | benjk: they do accept paypal |
| 05:29.21 | zekeonfir3 | and credit cards |
| 05:30.48 | zekeonfir3 | I've been a little disappointed. I ordered a vanity toll-free # like 3 weeks ago, but no dice. I put in a trouble ticket & no one has gotten back to me. |
| 05:37.02 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 05:37.27 | *** join/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 05:37.36 | JerJer | alright who's bitchin? :) |
| 05:37.44 | JerJer | i can smell it |
| 05:39.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:42.47 | tuxinator_linux | JerJer, you've been idle for 20 days (according to jbot). and look ... here you are |
| 05:43.01 | tuxinator_linux | no bitchin from me though |
| 05:43.27 | JerJer | idle ? |
| 05:43.30 | JerJer | perhaps in this channel |
| 05:43.42 | JerJer | i couldn't deal with the SNR in here |
| 05:43.50 | JerJer | so i've been hangin in -dev |
| 05:44.16 | h3x | #asterisk-paddedroom |
| 05:44.16 | tuxinator_linux | yep, I it is a bit noisy in here |
| 05:44.18 | Corydon76-home | And then they bring the noise to -dev |
| 05:44.46 | JerJer | grr yeah |
| 05:45.04 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 05:45.08 | tuxinator_linux | that channel isn't advertised, is it? |
| 05:45.13 | aminorex | how can i get a local DID in China? |
| 05:45.23 | JerJer | aminorex: talk chinglish |
| 05:45.40 | JerJer | tuxinator_linux: i didn't say -dev i meant #moosepenis |
| 05:45.41 | JerJer | :) |
| 05:45.52 | tuxinator_linux | JerJer, much better |
| 05:45.55 | aminorex | my chinglish is poor :( |
| 05:46.15 | tuxinator_linux | aminorex, pay someone to speak chinglish for you |
| 05:46.33 | aminorex | yeah, i need an agent in china |
| 05:46.43 | tuxinator_linux | I can't help you there, I only speak Spanglish |
| 05:47.05 | JerJer | i can swear in russain |
| 05:47.10 | JerJer | :P |
| 05:47.15 | Corydon76-home | "Oh, stewardess... I speak Jive." |
| 05:47.35 | JerJer | i picked the wrong week to stop sniffing glue |
| 05:47.40 | rob0 | LOL |
| 05:47.46 | h3x | if only i had ops. |
| 05:47.47 | rob0 | my favorite movie |
| 05:47.59 | h3x | topic |
| 05:47.59 | h3x | haha |
| 05:48.03 | tuxinator_linux | Corydon-w, airplain, he he |
| 05:48.12 | JerJer | h3x: can you sell me an oc-12 yet ? |
| 05:48.27 | h3x | you need an oc12? |
| 05:48.34 | tuxinator_linux | denon is alive? |
| 05:48.35 | h3x | wtf for |
| 05:48.38 | h3x | ptp sharing? heh |
| 05:48.46 | tuxinator_linux | porn, it's always porn |
| 05:48.55 | rob0 | My kids see that movie and don't get all the dated humor. Like the casting of Barbara Billingsley as a "jive" translator! |
| 05:48.56 | denon | tuxinator_linux: no, i'm just a bot |
| 05:49.10 | JerJer | phone calls, yo |
| 05:49.33 | tuxinator_linux | rob0, I'm 25, and I don't understand half of it |
| 05:49.56 | JerJer | i wouldn't mind 702-777-6969 :) |
| 05:49.57 | rob0 | Oh how sad, it is a masterpiece. |
| 05:50.02 | tuxinator_linux | JerJer speaks Jivelish |
| 05:50.41 | tuxinator_linux | denon, then this won't hurt |
| 05:50.48 | h3x | its already taken |
| 05:50.53 | h3x | thats the first one i thought of |
| 05:52.10 | JerJer | sucks |
| 05:52.21 | JerJer | 702-777-8647 :) |
| 05:53.11 | h3x | im selling 777-taxi i think |
| 05:53.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:53.17 | JerJer | sweet |
| 05:53.20 | h3x | im trying to find somebody to give me an appraisal |
| 05:53.30 | h3x | the domain name appraisal people dont wanna touch it |
| 05:53.48 | JerJer | list it on ebay :P |
| 05:54.59 | denon | I'll save ya the time .. I'll give ya $20 for it right now |
| 05:55.09 | h3x | i have an interested buyer, a new cab company |
| 05:55.12 | denon | no muss no fuss, I'll even paypal ya the cash |
| 05:55.33 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:55.39 | tuxinator_linux | denon, bots have cash? |
| 05:55.48 | denon | uh well .. |
| 05:55.52 | denon | AI is very resourceful |
| 05:56.00 | tuxinator_linux | ah, good point |
| 05:56.20 | JerJer | bots have to survive somehow |
| 05:58.07 | JerJer | exit with non-zero on 'Jerjer/1-1' |
| 05:58.14 | denon | hah |
| 05:58.19 | denon | JerJer needs better error trapping <G> |
| 05:59.16 | denon | long live freebsd :) |
| 05:59.41 | JerJer | i think this one is AIX :) |
| 05:59.51 | JerJer | or NextStep |
| 05:59.54 | denon | hah |
| 06:00.02 | denon | upgrade to SunOS :) |
| 06:00.06 | Qwell | silly JerJer running ReactOS |
| 06:00.31 | h3x | reactos is useless. |
| 06:01.20 | denon | er, I mean .. in an idle loop |
| 06:01.34 | Qwell | denon: ast_safe_sleep? |
| 06:01.44 | JerJer | perhaps StarOS maybe :) |
| 06:01.47 | denon | som'n like that |
| 06:01.53 | denon | DoNothing() |
| 06:02.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:02.09 | Corydon76-home | NoOp ? |
| 06:02.24 | JerJer | that's too asm |
| 06:02.29 | Qwell | while (1) DoNothing(); |
| 06:02.40 | JerJer | for (;;) |
| 06:02.45 | JerJer | while 1 waists time |
| 06:02.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-112-192.red.bezeqint.net) |
| 06:03.00 | Qwell | void DoNothing() { for (int j; j < 1000000; j++); } |
| 06:03.02 | JerJer | no need to test anything :) |
| 06:03.17 | Qwell | s/DoNothing/DoNothingInefficiently/ |
| 06:03.40 | JerJer | lol |
| 06:03.46 | Corydon76-home | but that doesn't do nothing... that does something... |
| 06:03.48 | Qwell | I've seen (and written) code like that. |
| 06:03.49 | JerJer | jbot does search and replace now |
| 06:03.53 | JerJer | someone was bored |
| 06:04.10 | *** part/#asterisk zekeonfir3 (n=zekeonfi@c-67-164-250-173.hsd1.ut.comcast.net) |
| 06:04.19 | Qwell | test |
| 06:04.27 | Corydon76-home | ~timriker |
| 06:04.29 | jbot | it has been said that timriker is my owner http://rikers.org/ mailto:Tim@Rikers.org mailto:TimR@Debian.org maintainer of BZFlag, member of a ton of open source projects http://www.advogato.com/person/timriker/ http://sourceforge.net/users/timriker/ the guy who GPL'd SCO's ABI files, giving every Linux user the right to use them ;-), or a very cool guy. |
| 06:04.35 | Qwell | s/e(s)//g |
| 06:04.41 | Qwell | lame |
| 06:05.26 | AlexLee | Help!!!I focused on Asterisk for several months, and I am using asterisk@home now, and also I am planning to develop a client, and hopfully the client may intergrate both Iï¼ and VOIP call functions. It is similar with MSN or Yahoo Messager, and surely I hope the asterisk@home provide with Voip function on MY CLIENT, and the other server provide with IM function. Actually, MSN and yahoo messager use two independent services to provide IM or voip call |
| 06:05.27 | AlexLee | <PROTECTED> |
| 06:05.27 | AlexLee | By now my problem is that: I have a website, after the users register on my website, I should send their User ID and password to users, but the most important is that I hope I can send a extension No.to the users, in this case, I have to wirte the registered information into database asterisk server when writing to the database on my website. By doing this, the databases of Asterisk and website are keeping same, and the users can login with my IM and |
| 06:05.29 | AlexLee | VOIP client to make call or send message. |
| 06:05.31 | AlexLee | To make the above idea into reality, the problem have to be resolved is that: we have to write registering information into the database on Asterisk when writing it into database on my website. I read most all of the document of asterisk on several website, but there is no solution. |
| 06:05.35 | AlexLee | By now, the extension No. for users is created by Administrator of Asterisk server, and the users can not apply for extensions by themselves from web or other program. If so, I think it is limited for collaboration application between Asterisk and other services, and the result is that the application of asterisk is too limited. Anyone have proposals or suggestions on this problem? How to write the registered information into the database on Asterisk |
| 06:05.42 | AlexLee | when writing registered information into other database on other services, or writing information into the database on Asterisk from other program? |
| 06:05.54 | lo_tech | this |
| 06:05.58 | lo_tech | s/this/that/ |
| 06:06.02 | Corydon76-home | denon: how about a +q for the flood? |
| 06:06.31 | lo_tech | someone definitely bored... bot love the regex |
| 06:06.50 | JerJer | how about a +o on Corydon76-home :) |
| 06:07.09 | Corydon76-home | Heh, I've tried, to no avail |
| 06:09.06 | JerJer | suckage |
| 06:09.29 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 06:09.30 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 06:25.50 | tzafrir_laptop | AlexLee, A@H is a distribution that includes Asterisk |
| 06:26.01 | JerJer | poorly |
| 06:26.12 | Qwell | s/includes/butchers/ |
| 06:26.25 | tzafrir_laptop | And not a very extendible one, too |
| 06:27.37 | tzafrir_laptop | There are a bunch of such portal programs. I figure that what you need is a way to integrate one with Asterisk |
| 06:27.56 | JerJer | use JTAPI and JBOSS |
| 06:28.07 | JerJer | yeah |
| 06:28.10 | tzafrir_laptop | It depends on your Asterisk config: it may reside (partially) in a database |
| 06:35.06 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
| 06:35.40 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 06:44.21 | Poincare | anyone knows how to set COLP (Connected Line Identification Presentation) for incomming calls? |
| 06:44.29 | Qwell | file: ^? |
| 06:44.37 | Qwell | I bet file knows how to set COLP |
| 06:44.52 | JerJer | or at least spell it |
| 06:45.06 | JerJer | so COLP is not DNIS ? |
| 06:45.24 | Poincare | DNIS = ? |
| 06:46.05 | trixter | dialed number information string |
| 06:46.07 | JerJer | Dialed number information service |
| 06:46.07 | trixter | ~dnis |
| 06:46.09 | jbot | it has been said that dnis is A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination ... |
| 06:47.16 | JerJer | lol - well don't search for "asterisk colp" in google |
| 06:47.24 | JerJer | you won't get what you expect :) |
| 06:47.25 | Poincare | :-) |
| 06:47.42 | Poincare | seems to be a famous guy that mr colp |
| 06:47.54 | tuxinator_linux | like saying don't look at the naked chich |
| 06:47.56 | Poincare | looks like dnis is slightly different... |
| 06:47.56 | Corydon76-home | ~google for asterisk colp |
| 06:48.02 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 06:49.25 | tuxinator_linux | crazy wind, rattleing the windows |
| 06:51.49 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 06:52.35 | JunK-Y | mr colp is a terrible man. |
| 06:52.47 | Qwell | JunK-Y: indeed |
| 06:52.52 | Qwell | pure EEEVIILLL |
| 06:53.06 | JunK-Y | hhehe |
| 06:53.23 | Corydon76-home | sweet like candy |
| 06:53.36 | Corydon76-home | He da candyman |
| 06:54.08 | JunK-Y | Corydon76-home: u make me sad, i thought I was ur candyman! |
| 06:54.15 | Qwell | tmi |
| 06:54.16 | brookshire[home] | drunk-y!! |
| 06:54.31 | JunK-Y | hey brookshire[home] ! |
| 06:54.39 | Corydon76-home | Uh, you might not want to be my candyman... |
| 06:54.41 | JunK-Y | long time, no see ya, whats up? |
| 06:55.08 | JunK-Y | whatever u want man ! :P |
| 06:55.13 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 06:55.18 | justinu | is it normal for * to restart b channels periodically? |
| 06:55.28 | JerJer | PRI - yes |
| 06:55.31 | justinu | k |
| 06:55.33 | JunK-Y | justinu: resetinterval yes |
| 06:55.36 | justinu | thx |
| 06:55.39 | JerJer | thats an adtran trick |
| 06:55.40 | JunK-Y | like once/hr? |
| 06:55.44 | justinu | looks like it |
| 06:56.03 | JunK-Y | u can change it via the resetinterval in ur zapata.conf |
| 06:57.09 | justinu | i just wanted to make sure it was normal behavior |
| 06:57.29 | JerJer | its a way to ensure the switch doesn't loose track of B channels |
| 07:00.22 | *** join/#asterisk dZen|n| (n=AbasCatu@cpe.atm2-0-1081053.0x50a4f886.arcnxx10.customer.tele.dk) |
| 07:03.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:04.37 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 07:15.52 | benjk | jerjer, when will your DID ordering be online again? |
| 07:16.19 | justinu | is sam here? |
| 07:18.33 | JerJer | when its back online |
| 07:18.54 | benjk | are we talking hours, days, weeks? |
| 07:19.08 | JerJer | perhaps years |
| 07:19.16 | JerJer | maybe a decade |
| 07:19.25 | benjk | well, I don't have that much time to wait |
| 07:19.55 | benjk | I will rather tell those folks to get a number elsewhere then so I can set this up before I go back home |
| 07:20.02 | JerJer | sure |
| 07:20.17 | JerJer | its either fix our shit or let people order new services |
| 07:20.18 | JerJer | hmm |
| 07:20.23 | JerJer | fix shit |
| 07:20.45 | benjk | I am not complaining, just wanted to know if it is worth waiting |
| 07:21.03 | JerJer | i have learned that i really suck at estimating coding efforts |
| 07:21.24 | benjk | yeah, same here, I always need at least twice as long ;) |
| 07:21.27 | JerJer | espcially when you have random people wire transfering you $5,000 to solve their problems - without even asking if 5g is enough |
| 07:22.15 | JerJer | so i have been seriously laying low lately, so real development can get done |
| 07:22.58 | h3x | ive learned that even tho im a good programmer im better off paying some schmuck 65k a year and give him a list of things to do |
| 07:22.58 | h3x | heh |
| 07:23.09 | h3x | and then just crack the whip |
| 07:23.23 | JerJer | if my current crack supply holds out we should be done in a few weeks |
| 07:23.31 | *** join/#asterisk latinbamm (n=spam@lgb-cust-208.57.68.90.mpowercom.net) |
| 07:23.31 | h3x | rofl |
| 07:23.35 | benjk | some things, if you want them done properly you almost got no other choice than doing them yourself |
| 07:23.39 | latinbamm | can someone help me really quick.. |
| 07:23.49 | JerJer | only if you ask a specific question |
| 07:23.50 | latinbamm | I keep getting HDLC Abort 6 errors |
| 07:24.03 | latinbamm | HDLC Bad FCS Errors |
| 07:24.16 | JerJer | got your time sync value right in zaptel.conf ? |
| 07:24.23 | JerJer | using a known good cable ? |
| 07:24.35 | latinbamm | yes i changed like 4 cables already.. |
| 07:24.39 | JerJer | using current zaptel / asterisk ? |
| 07:24.43 | latinbamm | whats a good sync value? |
| 07:24.50 | JerJer | depends on what your doing |
| 07:24.59 | latinbamm | asterisk 1.2.5 i belive |
| 07:25.15 | JerJer | damn you got skills |
| 07:25.28 | latinbamm | all the channels keep on restarting |
| 07:25.35 | latinbamm | sorry.. |
| 07:25.50 | JerJer | is this a known good T-1 you are using ? |
| 07:25.50 | benjk | probably because you are running future software ... |
| 07:26.01 | latinbamm | It was working fine in the morning and all of the sudden it stop working.. |
| 07:26.10 | benjk | time space continuum side effects |
| 07:26.12 | latinbamm | Yes we have been using the system for a bout 2 months now. |
| 07:26.13 | JerJer | sharing interrupts ? |
| 07:26.40 | latinbamm | No i disabled all that stuff in the bios i have the TE110P card interrupt 11 |
| 07:26.47 | latinbamm | and nothing is using that.. |
| 07:26.59 | JerJer | to rule out the telco, put a loop on and have them run tests |
| 07:27.18 | JerJer | perhaps pull and reseat the PCI card |
| 07:27.28 | latinbamm | whant me to post my cat /proc/interrupt |
| 07:27.32 | JerJer | and make sure its screwed down tight |
| 07:27.35 | JerJer | not really |
| 07:27.44 | JerJer | just as long as there is no sharing |
| 07:28.06 | latinbamm | well there is a 185 on it and i dont understand why. |
| 07:28.13 | latinbamm | it whould be interrupt 11 |
| 07:28.24 | latinbamm | when i run zttest my lowest it 99.95% |
| 07:28.37 | JerJer | paste just the line for your zaptel card |
| 07:28.49 | latinbamm | 185: 640189 11192 627649 0 IO-APIC-level wcte11xp |
| 07:29.02 | JerJer | you can try turning off apic |
| 07:29.08 | JerJer | i always do from the start |
| 07:29.29 | latinbamm | how do i do that if you can please let me know.. |
| 07:29.39 | justinu | any of you guys have a standard maintenance contract you use for customers? |
| 07:29.41 | latinbamm | on the kernel line before it boot right.. |
| 07:29.44 | justinu | i'd be willing to pay you for its use |
| 07:30.47 | *** join/#asterisk julien[re] (n=mactouch@AStDenis-103-1-11-187.w80-8.abo.wanadoo.fr) |
| 07:30.56 | JerJer | put noapic nolapic on your kernel line in your grub config |
| 07:30.56 | julien[re] | hi there |
| 07:30.58 | JerJer | and reboot |
| 07:31.19 | latinbamm | ok let me try it.. |
| 07:31.30 | latinbamm | thank you so much it let you know how it goes.. |
| 07:31.31 | *** join/#asterisk newl (n=newlook@203-59-210-244.dyn.iinet.net.au) |
| 07:31.32 | julien[re] | i've got a caller id problem on my BRI, the first 0 is deleted for local calls |
| 07:31.45 | julien[re] | but i receive 00 for international calls |
| 07:32.04 | justinu | so? |
| 07:32.09 | JerJer | and? |
| 07:32.14 | julien[re] | so i'd like to add the 0 to the local numbers |
| 07:32.16 | julien[re] | automatically |
| 07:32.23 | julien[re] | i do this in extensions.conf |
| 07:32.32 | JerJer | to the caller*id or the number they dial ? |
| 07:32.32 | julien[re] | but i'd rather have it done by zapata |
| 07:32.38 | julien[re] | the callerid |
| 07:33.02 | JerJer | Set(CALLERID(number)=0${CALLERIDNUM}) |
| 07:33.04 | JerJer | i believe |
| 07:33.07 | julien[re] | exten => 7860,1,SetCallerID(0${CALLERIDNUM}) |
| 07:33.14 | julien[re] | yeah that's what i use |
| 07:33.15 | JerJer | that's the old way |
| 07:33.31 | julien[re] | but if some1 dials from abroad |
| 07:33.37 | julien[re] | the number is 00044123456 |
| 07:33.55 | julien[re] | in fact i'd like to check if the number begins by 0 |
| 07:34.01 | julien[re] | and add a zero if not already here |
| 07:34.11 | JerJer | have fun |
| 07:34.48 | julien[re] | any1 experienced this problem by the way (missing first 0)? |
| 07:34.59 | justinu | ask your telco |
| 07:35.39 | JerJer | zaptel/asterisk won't mess with it unless you tell it to do so |
| 07:35.43 | Qwell | JerJer: 0${CALLERID(number)}, would be better even |
| 07:35.59 | latinbamm | im still getting those errors.. |
| 07:36.02 | julien[re] | oki but can i add the 0 only for numbers |
| 07:36.09 | latinbamm | i did notice that this though. |
| 07:36.11 | julien[re] | where the 0 is not already here |
| 07:36.14 | latinbamm | <PROTECTED> |
| 07:36.22 | latinbamm | it changed from 185 to 10 |
| 07:36.32 | JerJer | yes - that's the non apic way to rap with the processor |
| 07:36.41 | AlexLee | Help!!!I focused on Asterisk for several months, and I am using asterisk@home now, and also I am planning to develop a client, and hopfully the client may intergrate both Iï¼ and VOIP call functions. It is similar with MSN or Yahoo Messager, and surely I hope the asterisk@home provide with Voip function on MY CLIENT, and the other server provide with IM function. Actually, MSN and yahoo messager use two independent services to provide IM or voip call |
| 07:36.41 | AlexLee | <PROTECTED> |
| 07:36.41 | AlexLee | By now my problem is that: I have a website, after the users register on my website, I should send their User ID and password to users, but the most important is that I hope I can send a extension No.to the users, in this case, I have to wirte the registered information into database asterisk server when writing to the database on my website. By doing this, the databases of Asterisk and website are keeping same, and the users can login with my IM and |
| 07:36.44 | AlexLee | VOIP client to make call or send message. |
| 07:36.45 | JerJer | grrr |
| 07:36.46 | AlexLee | To make the above idea into reality, the problem have to be resolved is that: we have to write registering information into the database on Asterisk when writing it into database on my website. I read most all of the document of asterisk on several website, but there is no solution. |
| 07:36.50 | AlexLee | By now, the extension No. for users is created by Administrator of Asterisk server, and the users can not apply for extensions by themselves from web or other program. If so, I think it is limited for collaboration application between Asterisk and other services, and the result is that the application of asterisk is too limited. Anyone have proposals or suggestions on this problem? How to write the registered information into the database on Asterisk |
| 07:36.51 | JerJer | AlexLee: STOP |
| 07:36.55 | AlexLee | when writing registered information into other database on other services, or writing information into the database on Asterisk from other program? |
| 07:36.58 | JerJer | nobody is going to answer you like that |
| 07:37.04 | julien[re] | guess so :D |
| 07:37.13 | julien[re] | and does any1 has an idea for me? |
| 07:37.18 | latinbamm | is going even faster now thoug hwith the restarting.. |
| 07:37.26 | AlexLee | oh.why? |
| 07:37.28 | julien[re] | since i dont want international numbers to have three 0 |
| 07:37.36 | JerJer | AlexLee: cuz nobody wants to read all that shit |
| 07:37.41 | JerJer | ask a simple question |
| 07:37.42 | latinbamm | it restarts fine and then all of the sudden the HDLC errors start. |
| 07:37.48 | AlexLee | this is first time for me to use xchat |
| 07:37.53 | JerJer | if you cannot ask a simple question, post that shit on the mailing list |
| 07:38.00 | AlexLee | :) |
| 07:38.04 | latinbamm | anything you mightwanna look at that you thinkg might help.. |
| 07:38.10 | latinbamm | I really appreciate you helping me.. |
| 07:39.10 | JerJer | AlexLee: http://www.ircbeginner.com/ircinfo/etiquette.html |
| 07:40.22 | AlexLee | oh |
| 07:40.52 | JerJer | latinbamm: swap PCI slots |
| 07:41.10 | AlexLee | i suppose I can find someone to discuss... |
| 07:41.17 | latinbamm | ok i will try it right now.. |
| 07:46.20 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 07:46.56 | latinbamm | changed the pci card to another slot the irq changed to 7 but i still get those errors.. |
| 07:46.58 | latinbamm | :( |
| 07:47.14 | JerJer | do you have any other known working T-1 gear ? |
| 07:47.26 | JerJer | or its time to put a loop on the circuit and call the telco |
| 07:47.44 | JerJer | i would rule out the telco first |
| 07:47.46 | latinbamm | another card no. I have another T1 line but its data only |
| 07:47.55 | JerJer | then if that tests ok, then i would call Digium tech sup |
| 07:48.11 | latinbamm | how do i do the loop test. |
| 07:48.27 | JerJer | put a loop on the line, then call your provider |
| 07:48.51 | JerJer | i would hard loop it right at the smart jack |
| 07:48.58 | JerJer | to rule out any local hardware issues |
| 07:49.04 | latinbamm | were the adtran is right.. |
| 07:49.09 | JerJer | adtran ? |
| 07:49.11 | latinbamm | the little white box.. |
| 07:49.25 | JerJer | you have this into a dsu/csu ?? |
| 07:49.34 | JerJer | or does adtran make smartjacks ? |
| 07:50.12 | latinbamm | well on the back of the room there is this white box with to RJ45 cables comming out one is our T1(data) and the other is PRI |
| 07:50.29 | *** join/#asterisk dZen|n| (n=AbasCatu@cpe.atm2-0-1081053.0x50a4f886.arcnxx10.customer.tele.dk) |
| 07:50.33 | JerJer | hmm ok |
| 07:50.35 | latinbamm | so i make the loop test on that right? |
| 07:51.04 | JerJer | yes i would eliminate any local issues |
| 07:51.14 | latinbamm | ok.. |
| 07:51.45 | latinbamm | i just dont understand how it was working perfectly for almost 2 months and all of teh sudden it starts throwing those errors. |
| 07:51.46 | latinbamm | :( |
| 07:52.13 | latinbamm | Im thinking about reinstalling and backing up the inportant configuration files.. |
| 07:52.14 | JerJer | line cards can go bad |
| 07:52.16 | latinbamm | oh.. |
| 07:52.22 | JerJer | or the Zaptel card went bad |
| 07:52.34 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:52.42 | latinbamm | i hope not :( |
| 07:52.44 | JerJer | this an E-1 or T-1 ? |
| 07:52.50 | latinbamm | T-1 |
| 07:52.52 | JerJer | ok |
| 07:53.00 | latinbamm | Thank you so much for you help.. |
| 07:53.25 | latinbamm | Ill keep you in mind next time i need help and if you like i can even pay you if necessary.. |
| 07:53.43 | JerJer | does this system have an E1000 based NIC? |
| 07:53.56 | latinbamm | Yes (Dell 1600SC) |
| 07:54.24 | latinbamm | Dual Xeon 2.8 |
| 07:54.54 | latinbamm | think i should nuke the nic and install a pci one? |
| 07:55.13 | JerJer | http://www.digium.com/index.php?menu=compatibility |
| 07:55.34 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 07:55.51 | latinbamm | Oh great.. |
| 07:56.15 | latinbamm | So i guess im screwed with this one.. |
| 07:57.52 | latinbamm | thank you so much... |
| 07:58.02 | latinbamm | i appreciate your help.. ill see what i can do.. |
| 07:58.08 | brookshire[home] | latin: call digium support |
| 07:58.15 | brookshire[home] | latin: they might be able to help you out |
| 07:58.31 | latinbamm | Thank i will tomorrow first thing in the morning.. |
| 07:58.44 | latinbamm | thank you so much for taking the time to help me out.. |
| 07:58.44 | brookshire[home] | awesome :D |
| 07:58.53 | latinbamm | have a goodnight.. |
| 08:02.00 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:02.26 | *** join/#asterisk eobanb (n=eoban@149-159-112-169.dhcp-bl.indiana.edu) |
| 08:03.00 | eobanb | hi all; does anyone know if it's possible to use a rockwell RCV336ACF/SP with asterisk? |
| 08:03.42 | brookshire[home] | probably not... |
| 08:03.46 | JerJer | by a real FXO interface |
| 08:03.50 | JerJer | buy |
| 08:03.59 | brookshire[home] | or get a sip did from jerjer ;) |
| 08:04.16 | brookshire[home] | iax i mean |
| 08:04.23 | brookshire[home] | iax/sip/h323 |
| 08:04.24 | JerJer | Ordering of DIDs are offline |
| 08:04.26 | Qwell | nufone does sip too, no? |
| 08:04.27 | brookshire[home] | awh :( |
| 08:04.32 | JerJer | sip/iax |
| 08:04.35 | JerJer | no h.323 bullshit |
| 08:04.39 | Qwell | no sccp? ;] |
| 08:04.40 | brookshire[home] | but! |
| 08:04.46 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62-90-99-68.barak.net.il) |
| 08:04.50 | brookshire[home] | what kind of voip provider are you |
| 08:04.53 | brookshire[home] | no h323 |
| 08:04.57 | brookshire[home] | *pffft* |
| 08:05.08 | JerJer | one that has seen the light |
| 08:05.28 | Qwell | bed |
| 08:05.46 | eobanb | well the rockwell is what i have at the moment. it IS a voice modem but apparently it can't do an 8000 Hz sampling rate. |
| 08:05.58 | Qwell | FXO != voice modem |
| 08:07.37 | eobanb | look i dont need to plug into the LATA i just want to try out asterisk outside a voip-only environment |
| 08:07.50 | JerJer | then buy a real FXO interface |
| 08:07.59 | JerJer | voice modems will not cut it |
| 08:08.04 | JerJer | or buy a SPA-3000 |
| 08:08.34 | JerJer | or TDM01B |
| 08:08.38 | eobanb | why would a voice modem not work? |
| 08:08.49 | JerJer | (03:09:06) Qwell: FXO != voice modem |
| 08:09.41 | eobanb | okay if i told you i dont *really* understand the difference, would that get you to explain why i cant use a voice modem? |
| 08:09.52 | JerJer | (03:09:06) Qwell: FXO != voice modem |
| 08:09.54 | brookshire[home] | can you write the driver? |
| 08:10.01 | brookshire[home] | do you know c? |
| 08:10.17 | JerJer | for one most 'voice modems' are half duplex |
| 08:10.31 | JerJer | then they also cannot deal with no-hook audio |
| 08:10.33 | JerJer | on |
| 08:10.36 | benjk | eobanb: there is no real difference, it's just a label |
| 08:10.39 | JerJer | s/no/on |
| 08:10.50 | JerJer | benjk: bullshit |
| 08:10.51 | benjk | well as far as the hardware is concerned anyway |
| 08:11.09 | benjk | some FXO boards also have EC on board |
| 08:11.25 | benjk | which voice modems don't, so you need to do that in software |
| 08:11.47 | JerJer | that's like saying there is no difference than a Chevette versus a Corvette since they are both cars |
| 08:11.52 | benjk | however, you cannot expect to use a modem driver for using the modem as an FXO board |
| 08:12.01 | eobanb | my rockwell is full duplex. |
| 08:12.03 | benjk | because a modem driver is made for data and fax, not for voice |
| 08:12.08 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-234.claranet.co.uk) |
| 08:12.21 | benjk | the modem but not the driver |
| 08:12.23 | JerJer | eobanb: then write a zaptel hook for it |
| 08:12.24 | brookshire[home] | eobanb: then write the driver and leave us be :) |
| 08:12.56 | benjk | so, it comes down to the driver software, not the hardware |
| 08:13.24 | JerJer | benjk: you cannot say that for sure |
| 08:13.42 | brookshire[home] | it could be possible... |
| 08:13.52 | brookshire[home] | but not possible right now :) |
| 08:13.55 | benjk | most voice modems these days are soft modems |
| 08:14.02 | Nugget | unless your time is only worth a few pennies an hour, it's almost certainly a better decision to buy a "real" FXO device. |
| 08:14.21 | benjk | real FXO devices suck as much as the modems do |
| 08:14.30 | benjk | unless you're talking channel banks |
| 08:14.46 | Nugget | it's what us ghetto-dwellers have to deal with, though. |
| 08:22.43 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:26.16 | [av]bani | spa3k doesnt cut it |
| 08:27.40 | JerJer | why not ? we use them for 911 solutions all over the place |
| 08:27.51 | JerJer | and backup to internet failure |
| 08:29.34 | [av]bani | because the EC is pathetic |
| 08:30.42 | [av]bani | they might suffice for 911, but not daily use in say, a call center |
| 08:30.42 | [av]bani | the EC is poo |
| 08:32.30 | JerJer | perhaps in the early revs |
| 08:32.43 | [av]bani | early revs of what? |
| 08:32.49 | [av]bani | hardware? firmware? |
| 08:33.08 | JerJer | but i've put s3ks on lines where real, digium purchased X100Ps wouldn't deal at all |
| 08:33.21 | [av]bani | oh, no doubt. x100p has no EC at all. just software |
| 08:33.28 | [av]bani | x100p is poo on top of poo |
| 08:33.28 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 08:33.36 | [av]bani | but spa3k isnt much better, its still poo |
| 08:33.47 | JerJer | been rock solid in my experience |
| 08:34.08 | [av]bani | maybe on the shortest loops. the max echo tail on them is 8ms and its g.165, not g.168 |
| 08:35.06 | [av]bani | i'm guessing you just leave all the defaults on, including sipura's 'baseball bat' solution to EC |
| 08:35.34 | JerJer | well then you must have an early rev |
| 08:35.45 | [av]bani | lollerskates |
| 08:35.46 | JerJer | the docs on the one i have here say G.168 |
| 08:36.40 | [av]bani | er... what docs? |
| 08:37.04 | JerJer | the little print out that gives technical specs |
| 08:37.18 | [av]bani | from what, your vendor? |
| 08:37.30 | JerJer | that came inside of the s3k packaging |
| 08:37.34 | [av]bani | yeah, vendor. |
| 08:37.40 | [av]bani | sipura doesnt provide anythin g:) |
| 08:37.51 | JerJer | this is the sipura manual YO |
| 08:38.11 | [av]bani | oh you mean the quick install card? |
| 08:38.29 | JerJer | its a 5-10 page booklet |
| 08:38.48 | JerJer | says SIPURA on the front, with a picture of the unit |
| 08:41.56 | [av]bani | oh you mean this: http://www.sipura.com/Documents/SPA-3000.pdf |
| 08:42.12 | JerJer | no, but i was just about to point that out too |
| 08:42.50 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 08:43.05 | [av]bani | well let me ask you this, have you changed any of the units from default settings IRT echo cancellation |
| 08:43.29 | JerJer | possibly |
| 08:43.37 | JerJer | in most cases, no |
| 08:43.46 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
| 08:44.06 | [av]bani | have you ever wondered why the volume jumps up and down when you dial out and hear the remote ringing? |
| 08:44.13 | JerJer | because echo has never been a problem when we have used s3k |
| 08:44.21 | JerJer | volume ? |
| 08:44.22 | JerJer | no |
| 08:44.24 | [av]bani | oh, its not a problem if you leave it default |
| 08:44.29 | [av]bani | but that causes other issues |
| 08:45.03 | JerJer | i have literally not received a single complaint about the s3k in perhaps three dozen systems |
| 08:45.23 | [av]bani | i'm guessing they are not heavily used |
| 08:45.39 | JerJer | all local calls |
| 08:45.41 | [av]bani | echo is the #1 complaint about the spa3k |
| 08:45.42 | JerJer | fax calls |
| 08:45.52 | JerJer | inbound calls on legacy numbers |
| 08:45.55 | JerJer | sure they are used |
| 08:46.01 | danzig | EHLO * gurus |
| 08:46.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@85-65-124-16.barak-online.net) |
| 08:47.26 | [av]bani | do you have an spa3k there now? |
| 08:47.42 | JerJer | yes |
| 08:48.02 | [av]bani | do you have it set to dial out ? |
| 08:48.10 | JerJer | of course |
| 08:48.39 | bsdfreak | i don't have any problems with echo on my sp3k |
| 08:49.05 | [av]bani | go to pstn line in the webinterface and tell me what Echo Canc Enable, Echo Canc Adapt Enable, Echo Supp Enable are set to |
| 08:49.31 | [av]bani | bsdfreak: if you use the defaults, then you likely wont. but that causes other problems :) |
| 08:49.38 | bsdfreak | such as? |
| 08:49.53 | [av]bani | do you have your spa3k there now? |
| 08:49.57 | bsdfreak | yep |
| 08:50.02 | danzig | I have about 100 accounts with a SIP provider (musimi.dk), i.e. 100 peers in sip.conf. I have one main number with the same provider, all incoming call come in on (ie a register in sip.conf that sends to extension 'dialtone'). Calls coming in on my main number Asterisk logs as channel SIP/aRandomSipAccount-xxxx . It is not the first defined peer/register. It is not the last defined peer/register. |
| 08:50.04 | danzig | Why does Asterisk coosee that chanel to log it under? Is there any way I can specify a channel in the register? |
| 08:50.09 | [av]bani | can you dial out with it right now? |
| 08:50.12 | bsdfreak | echo canc enable: yes |
| 08:50.17 | bsdfreak | echo canc adapt enable: yes |
| 08:50.20 | bsdfreak | echo supp enable: yes |
| 08:50.24 | bsdfreak | heh |
| 08:50.27 | bsdfreak | sure |
| 08:50.37 | [av]bani | ok, whats your fxo impedance and spa to pstn gain |
| 08:50.48 | bsdfreak | 600 impedence |
| 08:50.51 | bsdfreak | 0 gain |
| 08:50.56 | [av]bani | so you havent touched it from the box |
| 08:50.59 | [av]bani | ok |
| 08:51.05 | bsdfreak | i messed with the gain and imp quite a bit |
| 08:51.08 | bsdfreak | found those settings worked the best |
| 08:51.51 | [av]bani | ok, do you hav a regular landline you can dial out on thats not connected to the spa3k |
| 08:51.55 | bsdfreak | sure |
| 08:51.59 | [av]bani | ok |
| 08:52.52 | bsdfreak | heh |
| 08:52.53 | [av]bani | dial out on the non spa3k landline, listen to how the remote ring sounds, should be pretty constant |
| 08:53.07 | [av]bani | remote doesnt have to answer |
| 08:53.12 | bsdfreak | k |
| 08:53.31 | [av]bani | shoud just be a normal rrrrrrrrrrr rrrrrrrrrr very constant, no changes |
| 08:53.40 | JerJer | never changes here |
| 08:53.48 | bsdfreak | are you basically suggesting the echo supp functionality messes with remote ringing? |
| 08:53.52 | [av]bani | no |
| 08:53.56 | bsdfreak | (if you're dialing out from the spa3k) |
| 08:53.59 | bsdfreak | ok what then hehe |
| 08:54.06 | [av]bani | just walk with me |
| 08:54.21 | bsdfreak | heh |
| 08:54.34 | [av]bani | ok, tested remote ringing on non spa3k line? |
| 08:54.42 | JerJer | yes |
| 08:55.11 | [av]bani | ok, now dial out on the spa3k. listen to the ring. is it perfectly constant? maybe tap the handset a little while its ringing |
| 08:55.27 | [av]bani | or a little background noise to make the echo supp kick in |
| 08:55.33 | [av]bani | breathing |
| 08:55.41 | JerJer | sounds the same |
| 08:56.01 | [av]bani | no volume fluctuation at all? |
| 08:56.06 | JerJer | no |
| 08:56.06 | bsdfreak | not that i can hear |
| 08:56.16 | JerJer | i even have mp3s pretty loud |
| 08:56.23 | JerJer | and it was solid |
| 08:56.29 | [av]bani | if you have constant background noise it will always be kicked in |
| 08:56.34 | [av]bani | so it wont change |
| 08:56.43 | JerJer | ok silent |
| 08:56.46 | JerJer | breathing |
| 08:56.47 | [av]bani | but if you have a quiet room with occasional noise it will kikc in and out |
| 08:56.47 | JerJer | ring |
| 08:56.50 | JerJer | breathing |
| 08:56.51 | JerJer | ring |
| 08:56.52 | JerJer | same |
| 08:57.09 | JerJer | lets listen to a miliwatt tone |
| 08:57.12 | JerJer | and blow into the phone |
| 08:57.20 | JerJer | nothing |
| 08:57.23 | [av]bani | oh heh, milliwat will screw it totally |
| 08:57.24 | JerJer | its the same |
| 08:57.33 | [av]bani | it wont kick in at all |
| 08:58.07 | JerJer | ok TV on other side of room |
| 08:58.13 | JerJer | ring |
| 08:58.18 | JerJer | breatch |
| 08:58.19 | JerJer | -c |
| 08:58.28 | [av]bani | yeah, if you have noise in the background it will be permanently kicked in so there will be no change |
| 08:58.32 | JerJer | no change in ring tone |
| 08:58.54 | JerJer | the TV is all the way on the other side of the room, tuned to a 24 hour news channel |
| 08:59.03 | JerJer | so this should simulate an agent in the next cube |
| 08:59.56 | [av]bani | ok just for grins, try turning off Echo Supp Enable |
| 09:00.05 | [av]bani | if the EC is real, you should hear no echo at all |
| 09:00.26 | [av]bani | (echo supressor != echo canceller) |
| 09:00.34 | JerJer | like i said, (how many hours ago?), i have taken an X100P that totally could not deal with the line and dropped in an s3k and its been rock solid |
| 09:00.38 | danzig | Is there a way to get * to log the IP number a call comes from? While I could log the output of 'sip show peers', someone can place a call without being registered, and I cannot find out what IP number the call came from. Solutions? |
| 09:01.09 | rabelais | danzip: turn off autocreatepeer |
| 09:01.28 | [av]bani | echo supp enable does a funky AGC on each side, not echo cancellation |
| 09:01.30 | JerJer | or come up with a solution that doesn't require knowing the IP address |
| 09:01.59 | [av]bani | JerJer: turn off echo supp enable, then dial and see if you get echo when you speak while the other end rings |
| 09:02.09 | JerJer | rebooting |
| 09:02.10 | [av]bani | if its really g.165/g.168 there should be none |
| 09:02.20 | JerJer | there is always NO echo |
| 09:02.21 | JerJer | none |
| 09:02.25 | JerJer | neither side |
| 09:02.41 | *** join/#asterisk koperniqs (n=koperniq@129.187.15.199) |
| 09:02.42 | koperniqs | hi |
| 09:02.43 | [av]bani | callers shouldnt hear any yes, its the outgoing side that would |
| 09:02.56 | danzig | rabelais>>autocreatepeer is off. That doesen't prevent someone from just calling sip/extension@myserver.com without being a peer/registered? |
| 09:02.57 | [av]bani | since EC would be done from the ilec or from their end |
| 09:03.00 | JerJer | neit |
| 09:03.05 | JerJer | sounds exactly the same |
| 09:03.32 | [av]bani | so you have echo supp enable No and you hear no echo when talking while remote is ringing |
| 09:03.35 | rabelais | danzig: unless they know the specific exention, no they can't call |
| 09:04.06 | JerJer | correct |
| 09:04.18 | [av]bani | under pstn line, not line 1 |
| 09:04.26 | JerJer | again i have never heard an echo from the s3k |
| 09:04.52 | JerJer | i am calling out via the POTS line interface yes |
| 09:05.01 | JerJer | via SIP on my 7960 |
| 09:05.03 | [av]bani | what firmware you using? |
| 09:05.08 | danzig | rabelais>>The extensions are public knowledge. Anyway, what I want to know is how to get * to log like any other server, apache etc - I want the IP number of anyone who connects to me. |
| 09:05.52 | rabelais | danzig: I think there is a per call environment variable that has the IP of the caller |
| 09:06.10 | *** join/#asterisk many (i=many@krikkit.ukeer.de) |
| 09:06.11 | JerJer | <PROTECTED> |
| 09:07.15 | rabelais | danzig: http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 09:07.36 | rabelais | I think the one you may want is ${SIPUSERAGENT} |
| 09:07.38 | [av]bani | what is your hardware version? |
| 09:07.52 | rabelais | if not that then ${SIPCALLID} |
| 09:08.24 | JerJer | i'm guessing 3.0 |
| 09:08.38 | [av]bani | guessing? |
| 09:08.40 | JerJer | since that ver matches up with whats on sipura.com |
| 09:08.51 | *** join/#asterisk draga (n=draga@host209-235.pool8253.interbusiness.it) |
| 09:08.52 | [av]bani | you cant see in your webinterface? |
| 09:09.05 | JerJer | i just closed the damn window, for the second time |
| 09:09.14 | danzig | rabelais>> Ok, I counld use that, but it has 2 big disadvantages: I have to implement it in my extensions.conf, many places, and it will not log failed attempts - for example a voip spammer trying random extensions... Is there really no way to get * to log this natively? Or perhaps I will have to log at the firewall (which also sucks as the firewall can't look inside SIP) |
| 09:09.44 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
| 09:09.44 | rabelais | danzig: none that I'm aware of....maybe? someone else? |
| 09:09.46 | bsdfreak | heh |
| 09:10.50 | rabelais | danzig: if there were, then I want one too |
| 09:11.00 | rabelais | cause I'm dying over here without a decent logging system |
| 09:11.02 | [av]bani | bsdfreak: what hardware rev. does it say your spa3k is in the info page? |
| 09:11.07 | rabelais | and I haven't gotten around to building my own yet |
| 09:11.13 | [av]bani | Hardware Version: |
| 09:11.47 | danzig | I use screen to capture the output of asterisk -r - but even that doesn't include all the info I want... |
| 09:11.55 | JerJer | oh god |
| 09:12.01 | JerJer | danzig: why in hell ? |
| 09:13.22 | danzig | without being registered. |
| 09:14.40 | danzig | ups... jerjer>> Because I have a user claiming that calls logged as her she is not placing. So I need to be really sure that the calls are not coming from another IP number, with her user name/pass, without being registered |
| 09:17.15 | danzig | Maybe I need to use creen to capture the output of asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvv ;-((( Or give up on * having a proper security log, and use snort or tcpdump or something... Any ideas? |
| 09:18.23 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 09:19.11 | JerJer | how about setting up a softphone or some other device with this users account info |
| 09:19.19 | JerJer | and attempt to pass a call thru without being registered |
| 09:19.24 | JerJer | and see what happens |
| 09:20.31 | [av]bani | JerJer: can you check the Hardware Version: on your spa3k? |
| 09:21.09 | danzig | jerjer>> unless you know something I don't, it will definately work. in SIP one can place calls without being registered, no problem. One just has to be registered to be able to _recieve_ calls |
| 09:21.26 | JerJer | ok |
| 09:21.42 | JerJer | so then lock the account in question down to the known IP |
| 09:21.58 | JerJer | and change the secret |
| 09:23.13 | af_ | what is sctp? |
| 09:23.57 | JerJer | smells like an SS7 protocol |
| 09:24.48 | danzig | jerjer>> fixed IP is a good idea - although I would rather have a good long term solution, so the next time this happens I can just look in the log and tell the user: that call came from your IP number, with your user/pass, and arpwatch says your ip number was on your MAC address, and the switch says the mac address was in your room. So the call definately came from your phone |
| 09:25.04 | JerJer | then do that |
| 09:25.34 | danzig | but I need asterisk to log the IP number of each call... |
| 09:25.40 | JerJer | then log it |
| 09:25.47 | danzig | how? |
| 09:26.07 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 09:28.15 | Hertell | has anyone got openwengo to work with asterisk? |
| 09:28.28 | Hertell | mainly the free-part of it |
| 09:28.45 | bsdfreak | never heard of it until now |
| 09:29.17 | Hertell | wengo will be a skhype-killer :-) |
| 09:29.21 | Hertell | works on sip |
| 09:29.22 | bsdfreak | heh |
| 09:29.27 | bsdfreak | interesting |
| 09:29.33 | Hertell | openwengo.org |
| 09:29.35 | bsdfreak | well if their pocketpc version works well i'll use it :] |
| 09:29.36 | Hertell | or was it com |
| 09:29.39 | bsdfreak | it's .com |
| 09:29.42 | [av]bani | bsdfreak: what hardware rev. does it say your spa3k is in the info page? |
| 09:29.47 | bsdfreak | heh |
| 09:29.52 | JerJer | danzig: doesn't the ${SIPCALLID} contain what you need? |
| 09:30.14 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:30.18 | bsdfreak | Software Version:2.0.13(GWg) Hardware Version:2.0.1(1813) |
| 09:30.43 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 09:30.59 | [av]bani | 2.0.13 :() |
| 09:31.05 | bsdfreak | eh? |
| 09:31.35 | [av]bani | software version |
| 09:31.46 | bsdfreak | yes, what about it |
| 09:32.21 | [av]bani | the latest is 3.1.7 |
| 09:33.23 | bsdfreak | for 3.0 hardware.. right? |
| 09:34.02 | [av]bani | you have 3.0 hardware |
| 09:34.06 | bsdfreak | eh? |
| 09:34.13 | bsdfreak | hardware version is 2.0.1 |
| 09:34.15 | [av]bani | ver 0.x = rev 1 |
| 09:34.17 | [av]bani | ver 1.x = rev 2 |
| 09:34.19 | [av]bani | ver 2.x = rev 3 |
| 09:34.59 | [av]bani | nobody has a spa3k which says 'hardware version:3.x.x(bla)' |
| 09:35.40 | [av]bani | and in any case, all versions work on all hw revs |
| 09:36.13 | danzig | jerjer>>yes, I was just hoping to not have to alter a lot of things in extensions.conf :-( What is the cleverest way of logging it? Using 'system' to write to a text file or what? |
| 09:36.23 | JerJer | hell no |
| 09:36.27 | JerJer | add it to the CDR |
| 09:36.56 | JerJer | or write your own CDR handler and suck in the channel vars |
| 09:37.15 | bsdfreak | ah i see, neat, av |
| 09:38.41 | draga | Hello everybody, a question: is it possible to have t38 pass-through on asterisk? I've a starcom ata and t38 support from the SIP provider but asterisk doesn't support it... |
| 09:38.50 | JerJer | yes |
| 09:40.31 | draga | JerJer: how? I've found some patches and tried them but on 1.2.4 it they fail to apply |
| 09:40.38 | danzig | jerjer>>sorry for stupid question, but where do i add it to the cdr? link? |
| 09:43.07 | JerJer | you need to check out file's passthrusdp branch |
| 09:43.27 | JerJer | research the CDR Userfield |
| 09:44.18 | danzig | jerjer>>thanx |
| 09:45.08 | *** join/#asterisk masked (n=cocaine@static-203-87-16-192.vic.chariot.net.au) |
| 09:45.33 | danzig | jerjer>>ok, can't use userfield - it is not even in cvs head, and I am on * 1.0.9 |
| 09:45.34 | masked | hi anyone familiar with the x100p? |
| 09:45.54 | masked | X-Rob: are you about? |
| 09:46.17 | draga | JerJer: thank you. I'm searching for it now |
| 09:46.31 | JerJer | danzig: sucks to be you then |
| 09:46.44 | JerJer | this is why you should have upgraded a year ago |
| 09:46.51 | masked | lol |
| 09:47.21 | danzig | Dont't fix things that aint broke :-) I do have a development server, though |
| 09:47.27 | JerJer | it is broke |
| 09:47.36 | JerJer | you are in here right now looking for a fix |
| 09:48.41 | danzig | We haven't noticed before now :-) And there are many workarounds. It is completely stable - and I am not risking that for one user... |
| 09:48.58 | JerJer | lol |
| 09:49.01 | JerJer | that's funny |
| 09:49.04 | masked | anyone have an idea why the zap modules load but wont create a channel for the x100p? |
| 09:49.15 | JerJer | if it was completely stable, why has there been so many MAJOR fixes ? |
| 09:49.23 | JerJer | including SIGNIFICANT MEMORY LEAKS |
| 09:49.38 | CaT[tm] | stable != bugfree |
| 09:49.46 | masked | its not hard to upgade danzig |
| 09:51.41 | danzig | Probably modules I'm not using ;-) Uptime 12 weeks, 3 days, 8 hours, 5 minutes, 24 seconds. Its not hard to upgrade, but we have a lot of custom integration, which would need to be tested. |
| 09:51.45 | draga | JerJer: that branch will only pass t38? is it aligned with the 1.2.4? |
| 09:52.02 | JerJer | that's what you get for doing 'custom' integration |
| 09:52.18 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:52.29 | JerJer | draga: it adds an option to the sip.conf to pass thru sdp |
| 09:52.33 | JerJer | if you want it |
| 09:53.17 | bsdfreak | heh |
| 09:53.23 | JerJer | draga: so your 'custom' integration was really a hack |
| 09:53.29 | JerJer | now you get to live with that hack |
| 09:53.34 | [av]bani | bsdfreak: what voip phones you use? |
| 09:54.56 | draga | JerJer: great, thank you! |
| 09:55.35 | draga | (last newbie question) |
| 09:55.53 | draga | how do I checkout that svn branch? I'm not familiar with svn |
| 09:59.13 | masked | what does this mean: line 223: Cannot get number of tones chanel 1 ? |
| 10:01.12 | draga | found, thank you! |
| 10:01.53 | masked | nevermind. fixed |
| 10:13.25 | masked | should a zap's alarm change from red once a live line is plugged in? |
| 10:15.50 | glm2k | yep |
| 10:20.01 | *** join/#asterisk dZen|n| (n=AbasCatu@cpe.atm2-0-1081053.0x50a4f886.arcnxx10.customer.tele.dk) |
| 10:20.32 | *** join/#asterisk vattern (n=vattern@dsl-146-148-171.telkomadsl.co.za) |
| 10:20.59 | vattern | Hi |
| 10:21.18 | vattern | I am looking for someone to test my ipkall DID if possible |
| 10:21.37 | trixter | why cant you test it yourself? |
| 10:22.38 | trixter | of course no one can ever test it if you dont give out the number |
| 10:22.49 | glm2k | hehehe |
| 10:23.38 | glm2k | hmmm, that reminds me, i never got my ipkall did to work. |
| 10:23.59 | vattern | 360-226-6878 |
| 10:24.16 | glm2k | get ready to be spammed |
| 10:24.19 | glm2k | j/k |
| 10:24.29 | vattern | I cannot test it because I am in south africa |
| 10:24.43 | vattern | the pabx I am wanting to replace doesn't allow INTL calls |
| 10:25.41 | trixter | it doesnt appear to be answering |
| 10:25.45 | vattern | mmmm |
| 10:25.51 | *** join/#asterisk Bambr (n=Bambr@213-35-237-161-dsl.end.estpak.ee) |
| 10:25.54 | trixter | took forever to ring |
| 10:26.33 | trixter | at least it works, just took forever to actually ring - I only heard one ring |
| 10:26.33 | *** join/#asterisk fsa (i=fsa@221.128.182.69) |
| 10:26.45 | fsa | hello guys I've setup asterisk on my machine and now I'd like to connect my POTS to my PC, how do I go abt it? I've got external voice modem will tht do the job or I need something else? |
| 10:27.02 | vattern | ok thanxit only rang once . than I picked up :) |
| 10:27.07 | vattern | Thanx a TON |
| 10:27.21 | trixter | yeah just saying that it was like 30 seconds to get that first ring |
| 10:27.24 | trixter | dunno where the lag was |
| 10:27.36 | vattern | probably here i South africa .. |
| 10:27.38 | trixter | my guess is it was at ipkasll trying to connect to your box |
| 10:27.52 | trixter | recalling to see |
| 10:28.15 | vattern | ipKall -> FWD via IAX -> box in South Africa |
| 10:28.17 | danzig | I have about 100 accounts with a SIP provider (musimi.dk), i.e. 100 peers in sip.conf. I have one main number with the same provider, all incoming call come in on (ie a register in sip.conf that sends to extension 'dialtone'). Calls coming in on my main number Asterisk logs as channelSIP/aRandomSipAccount- xxxx . It is not the first defined peer/register. It is not the last defined peer/register. |
| 10:28.19 | danzig | Why does Asterisk coosee that chanel to log it under? Is there any way I can specify a channel in the register directive? |
| 10:28.39 | trixter | ahh well its fwd sip -> fwd iax as I understand their set up |
| 10:28.50 | trixter | you can have ipkall directly connect to your box if you want |
| 10:28.56 | trixter | eliminate any lag fwd may introduce |
| 10:29.30 | vattern | mmmmm |
| 10:29.37 | vattern | that sounds like a plan |
| 10:30.14 | vattern | tho I would have to fix the damn firewall :) |
| 10:30.48 | vattern | urggh ... tarceroute to FWD : 364.244 ms |
| 10:31.09 | trixter | that is no good |
| 10:31.28 | *** join/#asterisk backblue (n=igor@87-196-10-190.net.novis.pt) |
| 10:31.58 | backblue | hi, anyone with hfc card in nt mode? |
| 10:32.28 | fsa | hello guys I've setup asterisk on my machine and now I'd like to connect my POTS to my PC, how do I go abt it? I've got external voice modem will tht do the job or I need something else? |
| 10:33.04 | trixter | odds are high you will need something else |
| 10:33.10 | trixter | at least if you want to use asterisk |
| 10:33.24 | backblue | fsa: buy a fxo card |
| 10:33.27 | fsa | do I need to get FXO |
| 10:33.28 | vattern | I am just playing around, testing things |
| 10:33.31 | backblue | and connect your ports there |
| 10:33.47 | backblue | simple |
| 10:34.16 | fsa | but the problem is I stay in India and if I go and tell these guys I need Foreign eXchange Office they'll direct me to Currency office |
| 10:34.45 | trixter | then you are asking in the wrong place |
| 10:35.30 | fsa | anyone has any idea as to how much will it cost me to get decent quality card maybe digium frm US? |
| 10:37.19 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:38.08 | danzig | fsa: I would reccomend ethernet attached FXO - for example sipura SPA3000 - more flexible, fewer driver problems. |
| 10:38.13 | backblue | fsa: which card? |
| 10:38.23 | *** part/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:38.41 | fsa | backblue: FXO card |
| 10:38.44 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:39.30 | backblue | anyone know if i need to terminate the BRI circuit in a card (whatever) in NT mode? |
| 10:39.33 | fsa | danzig: how many line-in does it have |
| 10:40.12 | danzig | fsa: 1 in, 1 out. If you need (a few) more, you could look at Fritz box |
| 10:40.59 | fsa | ok thnkx I'll google and find more info |
| 10:42.12 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:44.40 | RoyK | morning |
| 10:44.43 | *** join/#asterisk _deg_ (n=deg@201.22.46.150.adsl.gvt.net.br) |
| 10:50.51 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 10:51.19 | RoyK | anyone have any idea what JerJer is talking about when he speaks about a good alternative to realtime? |
| 10:53.12 | *** join/#asterisk Hmmmm (n=Hmmmm@221.135.51.19) |
| 10:53.27 | JerJer | give it up |
| 10:53.48 | JerJer | so some of your own research |
| 10:54.02 | JerJer | and since when have i ever spoken of any alternative ? |
| 10:55.08 | *** join/#asterisk fugitivo (n=ajf@201.255.178.247) |
| 10:55.14 | mut | JerJer: any idea why an adtran atlas 550 would strip off callerid name? |
| 10:55.56 | RoyK | fuck you, JerJer |
| 10:56.06 | JerJer | no thanks |
| 10:56.20 | JerJer | mut: strip? are you waiting long enough to answer the line ? |
| 10:56.29 | mut | yea |
| 10:56.45 | mut | the number shows but name dissapears |
| 10:56.54 | JerJer | are you sure the name shows up ? |
| 10:57.13 | JerJer | like is the telco sending it ? |
| 10:57.21 | mut | i'm sending it from a te405p |
| 10:57.44 | *** join/#asterisk fulgas (n=fulgas@209.8.233.241) |
| 10:57.49 | JerJer | not sure PRI deals with CID name |
| 10:58.01 | RoyK | it does not |
| 10:58.02 | JerJer | i have never tried to set it and push it out |
| 10:58.17 | mut | hm |
| 10:58.42 | mut | k, well one other question then, i can't get my as5350 to pass CID name to asterisk |
| 10:58.49 | mut | i see the Facility msg come in |
| 10:58.55 | mut | but it's not passed along |
| 10:59.11 | mut | this is from the telco |
| 10:59.15 | *** join/#asterisk g0mb0 (n=g0mb0@external.micom.mng.net) |
| 10:59.32 | *** join/#asterisk saftsack (n=saftsack@p54A7CA87.dip.t-dialin.net) |
| 10:59.34 | JerJer | can't help ya there |
| 10:59.37 | RoyK | ASN.1 notation sends only digits packed into four bits per digit |
| 10:59.48 | JerJer | i have always avoided configuring 5300s |
| 11:00.23 | mut | i'de like to |
| 11:00.27 | mut | but i get echo on my 405p |
| 11:00.33 | mut | so i had to put the cisco back in there |
| 11:01.06 | JerJer | echo !? |
| 11:01.21 | mut | yes echo?! |
| 11:01.34 | RoyK | yes echo?! |
| 11:01.42 | fugitivo | hell echo! |
| 11:01.46 | *** join/#asterisk Himeko (i=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 11:01.48 | JerJer | i simply do not understand why so many people have problem with echo, yet every project I have done, I have never experienced echo that could not be dealt with in asterisk |
| 11:02.02 | mut | well heres the thing |
| 11:02.03 | JerJer | and never had echo on PRI |
| 11:02.05 | fugitivo | JerJer: lucky you |
| 11:02.18 | JerJer | i don't think its luck - i have done thousands of systems |
| 11:02.19 | mut | it's only when i call out to a verizon phone |
| 11:02.29 | JerJer | so then its verizons problem |
| 11:02.33 | fugitivo | JerJer: i only had problems with x100p |
| 11:02.36 | mut | sure |
| 11:02.43 | mut | but the cisco can echo cancel it |
| 11:02.56 | mut | so it's also the zap echo cancellers problem |
| 11:03.03 | JerJer | that doesn't make sense |
| 11:03.14 | mut | yea, not everything does |
| 11:03.39 | mut | i spent a few days adjusting taps and gains and echo cancellers |
| 11:03.43 | mut | and nothing gets rid of it |
| 11:04.00 | Hmmmm | does anyone know of any cheap phone adapters? |
| 11:04.02 | JerJer | fugitivo: i have had echo on x100ps too - but most of the time it could be dealt with |
| 11:04.18 | RoyK | JerJer: 'done thousands of systems' lol |
| 11:04.35 | JerJer | keep pointing |
| 11:04.40 | fugitivo | JerJer: i'd found that with the x100p, at least in Argentina, the echo and volume problems depends on who is calling you |
| 11:04.50 | JerJer | someday your finger will get bit |
| 11:05.09 | mut | so anyway |
| 11:05.15 | trixter | that depends on where he sticks his finger I guess |
| 11:05.22 | mut | i'm putting together a couple tellab 2572 cancellers |
| 11:05.28 | mut | and using those with my 405p |
| 11:07.14 | JerJer | something strange has to be going on |
| 11:07.26 | JerJer | 2572s are way overkill |
| 11:07.35 | mut | dunno, the cisco can get rid of it |
| 11:07.44 | mut | but the zap software cant |
| 11:07.47 | JerJer | on the same PRI ? |
| 11:07.50 | mut | yea |
| 11:08.02 | mut | i plug it into my cisco and use sip to asterisk |
| 11:08.12 | danzig | jerjer>> I have figured out how to log IP numbers easily - 'sip show channels' shows the IP numbers, so I'll just run a cron job every minute to log that to a file :-) [hack ;-)] |
| 11:08.42 | trixter | ugly hack, and will cause higher load than is required |
| 11:08.45 | JerJer | and asterisk having to put up with all that bullshit |
| 11:08.54 | trixter | why not use the manager interface so its at least event triggered? |
| 11:09.00 | JerJer | danzig: just upgrade |
| 11:09.17 | RoyK | computers are like air conditioners. they stop working when you open windows.... |
| 11:09.29 | *** join/#asterisk tbs_ (n=ingen@fw.sg12.dk) |
| 11:09.33 | trixter | or if you are set on a sip show channels method, why not write a client that will use the asterisk stream pipe and not restart asterisk every minute? |
| 11:09.53 | JerJer | or just upgrade |
| 11:10.02 | trixter | the code on how to do that is in asterisk.c and cli.c ... |
| 11:10.07 | trixter | upgrading isnt always an option |
| 11:10.14 | JerJer | bullshit |
| 11:10.23 | mut | how isn't it |
| 11:10.52 | trixter | it depends on applications that interface, it depends on the platform that you are running it on, there are a few variables |
| 11:10.57 | JerJer | only if they hacked the shit out of that asterisk source |
| 11:10.59 | trixter | not everyone runs stock asterisk on linux |
| 11:11.00 | RoyK | JerJer: please try to understand that upgrading is not always an option. i know it's hard, but try to think a little... |
| 11:11.07 | JerJer | then that's what they get for hacking bullshit |
| 11:11.17 | mut | heh |
| 11:11.20 | RoyK | trixter: ignore the monkey :) |
| 11:11.21 | JerJer | then they shouldn't be running asterisk |
| 11:11.37 | mut | hah |
| 11:11.37 | JerJer | esp back in 1.0 days |
| 11:11.47 | trixter | becuase they dont want to run stock asterisk on linux they shouldnt run it? seems a bit biased |
| 11:11.48 | tbs_ | Hi everyone... I have a periodical problem, which I hope you are able to help me with. Asterisk 1.2.4 with zaptel/zapata cards. Siemens Gigaset ISDN handsets connected to two of the cards. Occasionally, I am not able to ring out from any of the handsets on one card. I get the following error in console: |
| 11:11.48 | tbs_ | http://pastebin.ca/40694 |
| 11:12.08 | JerJer | asterisk was wrote on linux for linux |
| 11:12.09 | tbs_ | It appears that asterisk thinks that both ISDN-lines are busy, but they are not |
| 11:12.19 | RoyK | was wrote..... |
| 11:12.22 | RoyK | not written..... |
| 11:12.22 | JerJer | only as an afterthought did other *NIX options came in |
| 11:12.43 | JerJer | notice i said was fucking wrote - and you say i should think? |
| 11:12.58 | danzig | My concrete problem is one user who says she is not placing calls which are logged to her. Surely asterisk -rx 'sip show channels' is almost no load, just connects to the already running asterisk, should not cause any problems. I don't really feel like upgrading a perfectly functioning system, or specifying custom CDRs or writing manager interface things just for this one, probably transient, prob |
| 11:13.00 | danzig | lem. |
| 11:13.04 | *** join/#asterisk _deg_ (n=deg@201.22.46.150.adsl.gvt.net.br) |
| 11:13.13 | RoyK | methinks someone _wrote_ asterisk for linux, but that it _was_written_ for linux |
| 11:13.15 | trixter | ok, well back to the cross platform floating license server for g.729 |
| 11:13.19 | JerJer | danzig: keep thinking that |
| 11:13.37 | JerJer | RoyK: so you a teacher now ? |
| 11:13.39 | tbs_ | The pastebin-example was made calling from the console, but the error messages are the same |
| 11:14.26 | Hmmmm | guys, does anyone know of any cheap phone adapters? |
| 11:14.42 | RoyK | quite a few, yes :) |
| 11:14.46 | RoyK | but the cheaper, the worse |
| 11:15.17 | Hmmmm | RoyK, but is there a relatively cheap one that you could recommend? |
| 11:15.35 | JerJer | spa-2002 |
| 11:15.50 | JerJer | get it from store.voxilla.com |
| 11:16.09 | RoyK | we're doing testing on some really neat netgear ATAs now with two FXS ports or one FXS port + WLAN. They rock! and with really good jitterbuffer + PLC. we tested the new jitterbuffer patch with those boxes on a 704/128kbps ADSL link while having 30 downloads running, and the sound was crystal |
| 11:16.56 | JerJer | too bad PLC does nothing for asterisk |
| 11:17.15 | JerJer | but i'm sure you already knew that |
| 11:17.50 | *** join/#asterisk daguerro (n=asdasd@lss-67-33.ee.ITB.ac.id) |
| 11:18.03 | JerJer | i won't even bother commenting on the '30 downloads' |
| 11:19.31 | *** part/#asterisk lahaine (n=qzxcd@2.67.119-80.rev.gaoland.net) |
| 11:20.00 | *** join/#asterisk heroine (n=heroine@2.67.119-80.rev.gaoland.net) |
| 11:22.04 | *** join/#asterisk heroine (n=heroine@2.67.119-80.rev.gaoland.net) |
| 11:22.22 | heroine | hi |
| 11:25.44 | RoyK | JerJer: you really are stupid :) |
| 11:28.37 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 11:36.44 | *** join/#asterisk Thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 11:44.00 | daguerro | hi all,, i got a problem starting asterisk.. it says "Unable to set normal priority Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl:... bla bla... " does somebody know why?? |
| 11:44.23 | trixter | prolly a permission problem |
| 11:44.32 | trixter | like you arent running it as root and you didnt set it up to run as any other user |
| 11:45.20 | trixter | asterisk defaults to run as root cause its easier that way, and for home users you dont have less than fully trusted people fiddling with it so some but not all of the security concerns arent present |
| 11:47.32 | *** join/#asterisk SGM (n=stoyan@213.240.228.198) |
| 11:47.40 | SGM | hi guys |
| 11:47.42 | daguerro | i know the root password. how to run the asterisk whitout login as root? i'm sorry. i'm new to linux.. |
| 11:48.31 | fugitivo | daguerro: http://www.voip-info.org/wiki-Asterisk+non-root |
| 11:49.05 | daguerro | ok... |
| 11:49.05 | RoyK | fugitivo: i beleive what he needs is a linux howto :) |
| 11:49.06 | daguerro | thx |
| 11:49.17 | fugitivo | errr |
| 11:49.18 | SGM | does anybody know why I'm getting "registration refused" from fwd |
| 11:49.21 | RoyK | daguerro: just su to root and start asterisk from there |
| 11:49.31 | fugitivo | daguerro: also, there's a book called "linux in a nutshell" |
| 11:49.38 | daguerro | hehehe... could be... |
| 11:50.43 | daguerro | but i want to start * on web,, so with php i write "system('/asterisk -vvvgc');" and that is the result... |
| 11:51.22 | fugitivo | daguerro: you need the webbrowser to run with the same user as asterisk |
| 11:51.54 | daguerro | i run it with root actually |
| 11:52.15 | fugitivo | the webserver? |
| 11:52.17 | daguerro | how to check my user? |
| 11:52.19 | daguerro | yes |
| 11:52.28 | fugitivo | you're brave |
| 11:53.00 | fugitivo | ps axu |grep httpd |
| 11:53.06 | SGM | daguerro: are you sure you're running your web server as root? :) |
| 11:53.24 | daguerro | sgm: how to chek that? |
| 11:53.35 | fugitivo | ps axu |grep httpd |
| 11:53.41 | SGM | ^^ |
| 11:53.41 | SGM | :) |
| 11:53.57 | trixter | even if you run the web server as root *most* will drop root after they bind to port 80 for security purposes |
| 11:54.09 | trixter | specifically to prevent arbitrary code from running as root via the webserver |
| 11:54.16 | fugitivo | daguerro: i strongly recommend you to read some linux basics |
| 11:55.27 | trixter | so launching it from php without it being suid root (generally not a good idea for other reasons) means it will exec as the webserver user, which will give you exactly that problem |
| 11:56.53 | daguerro | fugitivo: okey... |
| 11:57.40 | fugitivo | daguerro: you shouldn't run services as root, keep that in mind |
| 11:57.55 | fugitivo | i don't know why asterisk still uses user root as default |
| 11:59.33 | trixter | I was told about amonth ago that its easier to make asterisk run as root than to educate people about /dev/zap permissions, and the stream pipe permissions and all that |
| 12:00.07 | trixter | rather than have a proper init script that will fix all of that at start up or something |
| 12:03.09 | trixter | that probably is true, it probably is easier to do it that way than educate everyone who might use asterisk but I still think its wrong becuase it can be set up such that it doesnt need to do that.. there appears to be some work to change this eventually, just not today |
| 12:08.26 | fugitivo | well, we're not talking about games |
| 12:08.38 | fugitivo | this is a service that will run the communications of a company |
| 12:08.51 | fugitivo | it should do the right thing, not the easier thing |
| 12:08.57 | trixter | been there, done that, its a lost cause :) |
| 12:09.38 | trixter | in the end you get yelled at saying that any compromise means that it doesnt matter if its root or not becuase the user asterisk runs at is broken thus the communications are broken into |
| 12:09.52 | trixter | which I think is partly true, partly short sighted, but meh |
| 12:10.35 | fugitivo | true, but with root access they can do more things than break communications |
| 12:11.11 | trixter | been there done that, in the end you get yelled at and are told you are wrong |
| 12:11.12 | trixter | :P |
| 12:11.32 | fugitivo | that's like saying that it's the same running apache as root or apache user |
| 12:11.33 | fugitivo | it's not |
| 12:11.43 | fugitivo | there's a reason for not running apache as root |
| 12:12.03 | fugitivo | and ssh, and databases, and all services :) |
| 12:12.21 | fugitivo | i think that asterisk is the only serious service that runs as root |
| 12:12.33 | fugitivo | do you know any other? |
| 12:12.51 | trixter | sshd |
| 12:12.59 | trixter | so it can switch users once authenticated |
| 12:12.59 | trixter | :P |
| 12:13.10 | fugitivo | right, it runs as root |
| 12:13.18 | fugitivo | but for a very good reason :) |
| 12:13.41 | trixter | at least the mentality of the 80s and early 90s is gone ... run *EVERYTHING* as nobody which gives the user nobody a lot of power |
| 12:13.52 | fugitivo | hehehe |
| 12:14.01 | trixter | and at least stuff in /bin isnt owned by bin anymore which means a non-root user can tool the box |
| 12:14.23 | trixter | those only took about 20 years to correct |
| 12:14.30 | *** join/#asterisk coppice (n=chatzill@184.168.17.210.dyn.pacific.net.hk) |
| 12:15.30 | fugitivo | hmm, my dns isn't working |
| 12:17.57 | [av]bani | \o> |
| 12:20.15 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 12:28.06 | daguerro | yes,, httpd runs using user apache actually,, |
| 12:28.41 | trixter | that varies on distribution and operating system |
| 12:28.46 | trixter | debian for example runs it as www-data |
| 12:28.52 | trixter | but at least its easily set |
| 12:29.28 | fugitivo | daguerro: well, user apache can't stop and start asterisk |
| 12:29.30 | trixter | it will start as root becuase it binds to a port < 1024 and on unix like systems only root can do that by default (most systems have a kernel tunable that lets you set that port) |
| 12:30.45 | [av]bani | no problem to run on port 8080 :) |
| 12:31.30 | coppice | but its a little more modern to run it on port 8086 :-) |
| 12:31.45 | trixter | www.nsa.gov runs a dedicated program on port 80 that only does a 302 redirect to port 8080 for that reason |
| 12:31.52 | daguerro | hmm... |
| 12:32.12 | trixter | very small code bases are easier to evaluate for security problems |
| 12:32.39 | [av]bani | there's always selinux :) |
| 12:32.47 | [av]bani | its not too bad really |
| 12:33.33 | daguerro | selinux?? |
| 12:33.33 | trixter | hopefully the nsa hid all the backdoors they put into that :P |
| 12:33.34 | trixter | security enhanced linux |
| 12:33.41 | trixter | NSA creation based on older technology, it segregates power a bit more |
| 12:33.49 | trixter | if you didnt want to go selinux systrace is a really good option |
| 12:34.02 | trixter | and systrace works on non-linux platforms |
| 12:34.23 | [av]bani | based on? its rbac... rbac is rbac |
| 12:34.27 | trixter | http://www.citi.umich.edu/u/provos/systrace/ |
| 12:34.36 | [av]bani | you want old, default unix permissions is far older |
| 12:35.58 | [av]bani | systrace is also rbac |
| 12:37.18 | trixter | its slightly different in how it works though |
| 12:37.26 | [av]bani | same goal |
| 12:37.34 | trixter | to a point |
| 12:37.39 | [av]bani | same system, different internal architecture |
| 12:37.42 | trixter | it depends on how broadly you define the goal |
| 12:37.52 | [av]bani | its a watch, it keeps time, only the gears are different |
| 12:38.15 | trixter | there is actually a DoD spec that predates SELINUX that SELINUX was based on originally, not true with systrace |
| 12:38.32 | trixter | why I said that selinux was based on older technology, specifically that DoD spec |
| 12:38.42 | trixter | systrace lets you do things that the DoD spec wouldnt let you do |
| 12:38.49 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:38.56 | trixter | sidewinder firewall was also based on that same spec |
| 12:39.05 | trixter | although you dont hear much about them anymore, wonder if they are even still around |
| 12:39.27 | trixter | their contests to break the firewall were biased and they didnt pay prizes at least in the mid 90s when people won and posted proof of their win |
| 12:39.32 | trixter | which upset many in the industry |
| 12:39.41 | trixter | dont promise a leather bomber jacket and then not pay |
| 12:39.53 | trixter | or promise $10,000 cash and then say that you cant use any preexisting holes |
| 12:39.56 | *** join/#asterisk L|NUX (n=linux@202.5.145.57) |
| 12:40.28 | trixter | and their contests were to break into the firewall not past it so that is biased from a real world perspective anyway |
| 12:42.47 | [av]bani | trixter: what, you're saying selinux is based on orange book? |
| 12:43.17 | [av]bani | it's not. selinux isnt even C2. it's a completely different approach, though the capabilities overlap orange book somewhat |
| 12:43.52 | trixter | I never said that |
| 12:44.01 | trixter | there are more DoD standards than the rainbow series |
| 12:44.06 | [av]bani | its not dod spec at all, its not based on dod spec |
| 12:44.15 | trixter | and C2 requires not only software but specific hardware to be certified, down to the firmware versions |
| 12:44.26 | trixter | so you cant make a generic OS C2 |
| 12:44.39 | trixter | actually it is |
| 12:44.42 | [av]bani | its a completely from scratch system, not based on anything, but it is rbac in design |
| 12:44.54 | trixter | its based on the same spec sidewinder was based on |
| 12:45.52 | [av]bani | no.. selinux incorporated type enforcement which sidewinder had, but selinux is again separate and unique |
| 12:46.08 | [av]bani | thats like saying linux and bsd are the same, they arent. they both have posix though |
| 12:47.09 | trixter | sidewinder was based on the DoD spec as well |
| 12:47.19 | trixter | I didnt say that selinux and sidewinder are the same becuase they have similar properties |
| 12:47.27 | [av]bani | what dod spec is that? |
| 12:47.29 | fugitivo | openbsd is secure and doesn't use selinux (at least the last version I used) |
| 12:47.36 | trixter | I said that selinux and sidewinder based their original designs off a spec that predated both products |
| 12:47.41 | trixter | sidewinder was even up front about it |
| 12:47.55 | backblue | when you connect one rdis phone into your NT mode bri card in asterisk, where do i configure the extension of the phone? that it sends? |
| 12:53.30 | *** part/#asterisk SGM (n=stoyan@213.240.228.198) |
| 12:54.05 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
| 12:54.06 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:57.37 | trixter | This enhanced architecture was named Flask. The NSA has now integrated the Flask architecture into the Linux operating system to transfer the technology to a larger developer and user community. http://www.nsa.gov/selinux/info/ |
| 12:57.43 | trixter | NSA is DoD and FLASK > RBAC |
| 12:57.49 | trixter | RBAC is one component of FLASK |
| 12:58.02 | trixter | now I can eaisly get back to work |
| 12:59.59 | holmeh | What is the most usual way for VoIP providers to connect to the PSTN? |
| 13:00.25 | trixter | via a tier 1 carrier unless they are crappy voip providers then through someone else that resells voip service |
| 13:00.42 | trixter | some tier 1 interconnects are voip some are tdm |
| 13:01.08 | trixter | some connect via smaller providers but usually they provide regional coverage |
| 13:03.17 | [av]bani | so it wasn't a dod spec |
| 13:03.33 | [av]bani | selinux uses technoligies, its not written to meet a spec |
| 13:04.28 | trixter | please show me where I siad it was written to meet a spec |
| 13:04.47 | [av]bani | <trixter> its based on the same spec sidewinder was based on |
| 13:04.50 | [av]bani | <trixter> sidewinder was based on the DoD spec as well |
| 13:04.50 | trixter | I must have been abducted and replaced by an alien when that was typed becuase I personally never said that |
| 13:04.58 | trixter | based on and written to are totally different |
| 13:05.06 | trixter | it uses technologies, thus its based on |
| 13:05.17 | trixter | I really dont know why you are trying to argue this point so much |
| 13:06.16 | [av]bani | i would ask you the same question |
| 13:06.33 | trixter | twice you said I said things I didnt say |
| 13:06.43 | trixter | and you kept insisting that selinux is just rbac yet its more than that |
| 13:06.52 | trixter | rbac is one component of FLASK, which is what selinux was based on |
| 13:06.55 | [av]bani | just let it go, youre getting bent out of shape over nothing |
| 13:07.04 | trixter | FLASK was done in part by the NSA a division of the DoD |
| 13:07.05 | fugitivo | mysql is a poor database! and is evil! |
| 13:07.17 | trixter | I am getting bent out of shape? |
| 13:07.24 | trixter | I am not the one that has to lie to try to prove a point |
| 13:07.36 | [av]bani | try decaf |
| 13:07.37 | trixter | nor am I the one that didnt understand things all along |
| 13:07.52 | trixter | so since you have proven your worth you should be able to let it go now |
| 13:07.54 | trixter | please let it go |
| 13:08.28 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:09.43 | coppice | holmeh: they usually connect using 2 cans and a taught optical fibre |
| 13:09.55 | *** join/#asterisk draga (n=draga@82.53.235.209) |
| 13:10.26 | *** join/#asterisk pengyong (n=lala@222.185.16.174) |
| 13:11.29 | cpm | for long haul, very taught optical fibre |
| 13:11.50 | kippi | If I have (managers|n|||12) is the 12 for the amount of time that it will ring that queue? |
| 13:11.52 | draga | Hello everybody. I've downloaded, installed and run Asterisk passthrousdp branch. How can I configure the T38 passthrou? My adapter uses the port 49170 for T38...I can't find any doc about it |
| 13:12.04 | cpm | And large coffee cans instead of the smaller juice cans. Broadband uses the big peach cans. |
| 13:12.15 | Flyboy-SR22 | good morning everyone |
| 13:12.18 | trixter | when you tie the string into the can you want to use a button |
| 13:12.19 | cpm | morn'n |
| 13:12.24 | trixter | greatly iumproves audio quality |
| 13:12.42 | cpm | trixter, thanks, good clue! I'll give that one a shot. |
| 13:12.45 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:12.54 | trixter | cpm: did you see the cysco cans? |
| 13:12.58 | trixter | the ones on ebay |
| 13:13.04 | cpm | nope. |
| 13:13.11 | trixter | cat 5 + 2 cysco (real food manufacturer) cans |
| 13:13.51 | trixter | called it a cysco voip system, and advertised that it would carry your voice down the cat5 was quite funny |
| 13:14.00 | coppice | trixter: you can't tie fibre around buttons. it fractures. You wrap duct tape aroound it, till its thick enough not to pull through the hole in the can. gee some people are so out of date |
| 13:14.06 | trixter | ebay canceled their auctions a few times but they insisted they really did have 2 such cans for sale at $50 or something |
| 13:14.09 | cpm | good point. I have some acoustic couplers around here somewhere, I might try putting a data call across cans at some point. |
| 13:14.16 | trixter | coppice: I said string :P |
| 13:14.24 | cpm | I think that's a worthy effort |
| 13:14.33 | trixter | ha |
| 13:14.37 | trixter | odds are the cans will echo a bit |
| 13:14.45 | trixter | gotta get an echo canceler for those cans |
| 13:15.07 | cpm | yeah, but the near-end delay is so short, the combfiltering effect is nominal |
| 13:15.14 | trixter | the button really does increase the rxgain |
| 13:15.21 | coppice | echo won't matter for the modem standards used with acoustic couplers |
| 13:15.47 | trixter | cpm: get webspace at some provider you hate write that up then /. it |
| 13:16.07 | trixter | be sure to put banners on the page so at least you can pay your hosting and materials for that project :) |
| 13:16.19 | fugitivo | coppice: what is your website for unicall? |
| 13:16.29 | coppice | acoustic couplers can carry 1200bps V.23. Use the DoD 600bps MELP codec, and your away. VoIPo2cansand string |
| 13:16.44 | coppice | fugitivo: www.soft-switch.org |
| 13:16.47 | trixter | does unicall support the sangoma cards where you can tell em to drop 20ms frames instead of a constant stream? |
| 13:16.57 | fugitivo | thanks |
| 13:17.37 | *** join/#asterisk adibar (n=adibar@217-162-123-170.dclient.hispeed.ch) |
| 13:17.48 | trixter | I think cpm went idle while he is looknig for cans in his trash |
| 13:18.22 | coppice | trixter: are you asking if it supports sangoma's native interface the answer is not yet. sangoma haven't finish it |
| 13:18.29 | fugitivo | coppice: is the 0.0.3pre8 ok for asterisk 1.2.4? |
| 13:18.48 | coppice | should be |
| 13:18.57 | trixter | cpm: for your next project get a parabolic and at its focal point put a large washer type device (flat metal disc with large hole) anything said through that will be spread out across the parabolic and not as audible, but the other end will still be able to hear it |
| 13:19.10 | trixter | if you can get the couplers working with that then you can have a wireless rig |
| 13:19.38 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:20.01 | trixter | if you stand in between those dishes you wont really hear anything, but whispers can travel, providing the dish is large enough and the other end close enough of course |
| 13:20.16 | trixter | the science center in san diego had that set up when I was um 5 or so and went there on a field trip |
| 13:20.23 | trixter | I thought it was the coolest thing ever |
| 13:20.39 | trixter | course I was also 5 and back then computers didnt do much :P |
| 13:20.57 | coppice | St Paul's Cathedral in London has a more fun example of that |
| 13:22.44 | cpm | well, i've got a bunch of ts43 fieldphones, that I have run a single strand of 22g wire, twisted together for over 2 miles, with 4 phones on it, and they all worked fine |
| 13:23.14 | cpm | that's with 1 pole on the single strand, the the other pole tied to a 22d copper nail stuck in the ground |
| 13:23.35 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:23.40 | cpm | btw, the audio quality was far superior to any cellphone, |
| 13:23.42 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 13:23.49 | coppice | the pole will severly attentuate the high frequencies :-) |
| 13:23.59 | trixter | cpm: seriously do the tin can acoustic coupler thing and /. it, then get some old satellite tv dishes and make it wireless.. |
| 13:24.06 | trixter | that would be the funniest thing ever |
| 13:24.07 | cpm | hehe |
| 13:24.11 | cpm | that would be fun. |
| 13:24.24 | trixter | get some banner ads so you can pay for hosting and materials |
| 13:24.27 | trixter | cj.com |
| 13:24.32 | cpm | I might have some time for that over the summer. would be worth a try, as i do have the couplers |
| 13:24.36 | MikeJ[Laptop] | silly silly! |
| 13:24.57 | MikeJ[Laptop] | I know somone who was working on that... |
| 13:25.46 | cpm | well, running on fibre, you'd know instantly if someone tapped it, as the carrier would drop. |
| 13:25.54 | cpm | less so for 'wireless' |
| 13:25.59 | sergeus | i have an issue with SIP client behind NAT :)) |
| 13:26.00 | cpm | so, fibre is more secure |
| 13:26.14 | sergeus | Please don't tell me RTFM :)) |
| 13:26.14 | coppice | if its really uselss and dumb, there's bound to be somebody working on it :-) |
| 13:26.23 | cpm | heh |
| 13:26.38 | MikeJ[Laptop] | hehe |
| 13:26.44 | MikeJ[Laptop] | sergeus, DRTFM! |
| 13:26.46 | MikeJ[Laptop] | :P |
| 13:27.04 | sergeus | i have nat=yes in sip.conf, bu i see that asterisk keeps sending RTP to internal address - why ? |
| 13:27.04 | MikeJ[Laptop] | sergeus, no one can help you with that little information... we need details |
| 13:27.17 | cpm | sergeus SIP and NAT don't play well together. |
| 13:27.23 | cpm | ~stun |
| 13:27.34 | jbot | methinks stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs |
| 13:27.34 | MikeJ[Laptop] | do you have your external and localnet set right? |
| 13:27.34 | MikeJ[Laptop] | externalip |
| 13:28.06 | MikeJ[Laptop] | cpm, it can work... in many situations just fine.. it is only an issue if BOTH client and server are behind nat.. |
| 13:28.14 | sergeus | MikeJ[Laptop], nope :) but all over clients works well :) ok i'll read abot this 2 options and i'll play with it |
| 13:28.27 | MikeJ[Laptop] | sergeus, RTFM :P |
| 13:28.42 | cpm | best luck I've had is when using a stun server. |
| 13:29.42 | sergeus | MikeJ[Laptop], yeah yeah i knew it ;))) |
| 13:29.54 | draga | Hello everybody. I've downloaded, installed and run Asterisk passthrousdp branch. How can I configure the T38 passthrou? My adapter uses the port 49170 for T38...I can't find any doc about it |
| 13:30.14 | MikeJ[Laptop] | http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?annotate=9034 around line 160 |
| 13:30.47 | MikeJ[Laptop] | draga, read the bugnote on that one.. there were notes in there |
| 13:31.01 | draga | thank you MikeJ[Laptop]! |
| 13:31.09 | MikeJ[Laptop] | http://bugs.digium.com/view.php?id=5090 |
| 13:31.56 | trixter | coppice if its really uselss and dumb, there's bound to be somebody working on it :-) |
| 13:32.08 | trixter | strangely the more useless and more dumb something is the more people work on it and the more effort goes into it |
| 13:32.18 | trixter | almost an exponential return |
| 13:32.19 | coppice | how do you tell linux which driver to use when a USB device is plugged in and there are multiple drivers that are prepared to pick it up |
| 13:32.28 | trixter | hotplug ? |
| 13:32.45 | MikeJ[Laptop] | coppice, with a baseball bat? |
| 13:32.53 | MikeJ[Laptop] | :D |
| 13:32.57 | trixter | normally its based off the device id and all that which there should only be one driver loaded for a given id set |
| 13:33.07 | AndyCap | coppice: /etc/hotplug/usb.*map |
| 13:33.16 | MikeJ[Laptop] | I like my answer better. |
| 13:33.21 | trixter | you have the manufacturer id and the device id so that combo should be unique to one driver |
| 13:33.34 | sergeus | localnet and externip won't help me, |
| 13:33.37 | AndyCap | trixter: not really. :-) |
| 13:33.37 | trixter | MikeJ[Laptop]: your answer goes with the state of this channel a few minutes ago :P |
| 13:33.48 | AndyCap | trixter: but ideally. |
| 13:33.49 | coppice | trixter: except for any HID device the generic hid drivers will pick it up, and they seem to do this before the specialised driver that should |
| 13:33.52 | trixter | well it should, I didnt say that it was |
| 13:33.54 | MikeJ[Laptop] | sergeus, why not? |
| 13:34.05 | trixter | becuase if you have multiple drivers all looking for the same thing weird things happen |
| 13:34.24 | AndyCap | HID and usb-rs232 devices are probably offenders |
| 13:34.39 | sergeus | MikeJ[Laptop], 1. my asterisk has only 1 real ip so externip won't affect it, right? |
| 13:34.49 | coppice | trixter: that's the normal case. things like the generic HID driver wil pick up anything in the HID class, regardless of its ID |
| 13:34.57 | trixter | it needs both so it knows what is and what isnt natted |
| 13:35.10 | trixter | coppice: well that is just broke |
| 13:35.20 | coppice | that's USB :-) |
| 13:35.28 | trixter | it shouldnt do that, but I dont really use usb so I dont know .. I use it for my iupaq and um um um that is about it |
| 13:35.35 | MikeJ[Laptop] | sergeus, ok... so, the asterisk box is not behind nat, just the client? |
| 13:35.49 | sergeus | MikeJ[Laptop], yes |
| 13:36.05 | draga | MikeJ[Laptop]: I can't find any interesting config line in http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?annotate=9034 |
| 13:36.15 | draga | perhaps I'm missing something, what should I expect to find? |
| 13:37.19 | coppice | I put an entry in /etc/hotplug/usb.handmap for my device. It doesn't seem to get used, though. It there anything else I need to do? |
| 13:37.31 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 13:38.27 | MikeJ[Laptop] | draga, huh? |
| 13:38.29 | *** join/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu) |
| 13:38.39 | MikeJ[Laptop] | that was for sergeus |
| 13:38.43 | pycsusz | Hi Everybody! |
| 13:39.05 | draga | MikeJ[Laptop]: I can't find the sip.conf option to enable t38 |
| 13:39.51 | draga | Mike: found :) |
| 13:41.13 | sergeus | MikeJ[Laptop], maybe if you have some free time you'll take a look ? :) http://pastebin.ca/40720 |
| 13:41.39 | sergeus | MikeJ[Laptop], i'm almost sure that i'm doing something wrong, but i can't se what |
| 13:43.07 | sergeus | MikeJ[Laptop], i can't understand why asterisk sending RTP to internal address rom INVITE packet - as far as i understand if nat=yes - enabled then asterisk should ignore ip addres from SIP packet and send all replyes to source, right? |
| 13:43.11 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:44.48 | pycsusz | I have a question, and I would like to get answer from so user, who already has got solution for this problem. |
| 13:45.28 | pycsusz | well, I have got an asterisk server, and a lot of extens on different subnetwork |
| 13:45.32 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:45.46 | MikeJ[Laptop] | sergeus, canreinvite=no, host=dynamic |
| 13:46.04 | MikeJ[Laptop] | you have nat=yes on the peer? |
| 13:46.17 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:46.18 | pycsusz | how can I configure the asterisk and the phones(grandstreem) how it work? |
| 13:46.19 | MikeJ[Laptop] | is it matching the right one? |
| 13:46.25 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:46.29 | sergeus | MikeJ[Laptop], yes |
| 13:46.41 | MikeJ[Laptop] | try those other options.. |
| 13:46.47 | MikeJ[Laptop] | onthe peer |
| 13:47.01 | MikeJ[Laptop] | somthing else isn't right inthe conf |
| 13:47.09 | sergeus | MikeJ[Laptop], yes i have nat-yes, canreinvite=no,host=dynamic,cancallforward=no |
| 13:47.13 | MikeJ[Laptop] | what nat=is in your globals? |
| 13:47.34 | sergeus | MikeJ[Laptop], nat doesn't set in globals |
| 13:47.43 | asteriskmonkey | sergeus.. you have nat-yes not nat=yes |
| 13:47.43 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 13:47.43 | *** join/#asterisk Ariek48 (n=ariek@141.252.221.218) |
| 13:47.58 | sergeus | asteriskmonkey, nat=yes :) |
| 13:47.59 | MikeJ[Laptop] | heh.. that would do it. |
| 13:48.23 | MikeJ[Laptop] | sergeus, scrub the passwords out of your sip.conf and pastebin it |
| 13:48.24 | sergeus | MikeJ[Laptop], ok i'll try to enable nat=yes in globals |
| 13:52.02 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:53.03 | *** join/#asterisk Tili (n=Tili@202-133-67-98-dialup.sat.net.pk) |
| 13:56.47 | *** join/#asterisk jaiger (n=jaiger@c-67-165-4-34.hsd1.ct.comcast.net) |
| 13:57.11 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 13:57.11 | *** mode/#asterisk [+o denon] by ChanServ |
| 13:58.32 | sergeus | MikeJ[Laptop], sorry, it took some time http://pastebin.ca/40730 |
| 13:59.45 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:01.12 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 14:07.39 | Katty | hmm. |
| 14:07.48 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 14:07.57 | iDunno | hmm? |
| 14:09.20 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 14:10.01 | sergeus | MikeJ[Laptop], global option nat=yes doesn't help |
| 14:10.04 | danzig | I have about 100 accounts with a SIP provider (musimi.dk), i.e. 100 peers in sip.conf. I have one main number with the same provider, all incoming call come in on (ie a register in sip.conf that sends to extension 'dialtone'). Calls coming in on my main number Asterisk logs as channel SIP/aRandomSipAccount-xxxx . It is not the first defined peer/register. It is not the last defined peer/register. |
| 14:10.06 | danzig | Why does Asterisk choose that chanell to log it under? Is there any way I can specify a channel in the register directive? |
| 14:12.06 | *** join/#asterisk rene- (n=rene@dsl-201-128-115-190.prod-infinitum.com.mx) |
| 14:12.39 | docelm0 | mew |
| 14:14.48 | bweschke | danzig: it's going to pickup whatever registration was last successful. |
| 14:17.12 | danzig | bwes>>thanks. It is a bit ugly, that all calls coming in on the main number arer logged as channel <one of the normal users>, but I guess it does not really do any harm... |
| 14:17.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.01 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 14:19.12 | MRH2 | Hi can I record files in g729 format for AgentMonitorOutgoing, Monitor and MixMonitor? if so is the extension option "g729"? |
| 14:19.14 | sergeus | MikeJ[Laptop], Any ideas ? :) |
| 14:22.51 | sergeus | MikeJ[Laptop], you the only person who can help me, pleeeeeeaaaseee :) |
| 14:25.08 | [TK]D-Fender | Katty: Mew. |
| 14:25.19 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:27.12 | Katty | [TK]D-Fender: (= |
| 14:28.30 | kippi | If I have (managers|n|||12) is the 12 for the amount of time that it will ring that queue? |
| 14:30.13 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:31.51 | Falle | adibar: there? |
| 14:32.01 | rene- | hi. |
| 14:32.11 | adibar | Falle: Yope |
| 14:32.29 | Falle | adibar: is your sidetone still working with the latest beta? :) |
| 14:32.29 | rene- | does anyone knows if spandsp, txfax/rxfax are bundled in a debian pkg? |
| 14:33.13 | adibar | Falle: Have tested it right now... Yes, it does. |
| 14:34.16 | adibar | Falle: and I'm using 1.0.2.3 |
| 14:34.17 | Falle | adibar: hmm. is there some setting for it then? With 1.0.2.3 it worked but my phone had some echoproblems. Now in 1.0.2.8 its gone and so is the echo. |
| 14:34.32 | Falle | oh |
| 14:34.55 | adibar | Falle: seems to be that I missed 1.0.2.8 |
| 14:35.13 | Falle | http://www.voip-info.org/users/31/26031/images/421/GXP2000_Release_1.0.2.8.zip |
| 14:35.39 | adibar | Falle: is .8 also such a looping bastard as .3 ? |
| 14:35.47 | Falle | no |
| 14:36.17 | Falle | that problem was fixed in .6 |
| 14:36.36 | adibar | Falle. Gimme a mom to upgrade ;-) |
| 14:36.45 | Falle | ait :) |
| 14:38.04 | adibar | ...booting |
| 14:38.43 | adibar | ...rebooting ? |
| 14:38.50 | adibar | done ! |
| 14:39.10 | adibar | ring still working |
| 14:39.43 | Falle | and sidetone? |
| 14:40.15 | adibar | Falle: Programm: .8 Bootloader: .3 Ring works |
| 14:42.01 | adibar | Falle: what do U mean with sidetone ? |
| 14:44.45 | Falle | adibar: http://en.wikipedia.org/wiki/Sidetone |
| 14:45.09 | adibar | Falle. Mom... |
| 14:45.26 | Falle | thats sort of the major thing with the beta releases.. they are trying to get the sidetone support working correctly :P |
| 14:46.57 | sergeus | MikeJ[Laptop], you here? :) |
| 14:47.34 | adibar | Falle: Gotcha... Mine is away too :-( |
| 14:48.33 | Falle | ok, atleast its not a "my phone only" problem then :) |
| 14:50.14 | adibar | Falle: Right... Now as U forced me to update it's also mine ;-) |
| 14:50.35 | Falle | adibar: hehe, sorry :P |
| 14:51.04 | adibar | Falle: I will never ever forgive U taht ;-) |
| 14:51.15 | Falle | adibar: I think i can live with that ;) |
| 14:51.38 | adibar | Falle. Do U remeber the BETA-Link of the Grandstream-Page ? |
| 14:52.20 | Falle | adibar: actually i think i like it better without the sidetone support until its working correctly. It would be great to be able to turn it of and on for beta testing. |
| 14:52.36 | Falle | adibar: http://www.voip-info.org/wiki/view/GXP-2000 |
| 14:53.47 | adibar | Falle. Did not mean that. I found it: http://www.grandstream.com/BETATEST/ |
| 14:54.12 | Falle | adibar: that link dont contain the latest betas.. ever :) |
| 14:54.23 | _Sam-- | it does this time |
| 14:54.54 | adibar | Falle: True... it's still .3 |
| 14:55.04 | Falle | yep.. its not updated at all |
| 14:55.05 | Falle | :) |
| 14:55.06 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:55.08 | _Sam-- | er...i thought you meant the gxp wiki page! |
| 14:55.13 | _Sam-- | i didnt look at the link close enough! |
| 14:55.18 | Falle | :) |
| 14:55.23 | Falle | The wiki page is great |
| 14:56.21 | adibar | Falle: Horible if an external open community is more actual than the vendoe/producer of a good :-( |
| 14:56.39 | _Sam-- | they are receptive though |
| 14:56.48 | _Sam-- | they specifically implemented a feature request i put in there |
| 14:57.15 | Falle | nice to have a "direct link" to the devel folks :) |
| 14:57.26 | sergeus | any experts in NAT and SIP ? :) |
| 14:57.27 | _Sam-- | and the phone, at least in my opinion is alot better than it was 8 weeks ago |
| 14:57.32 | Falle | oh yes |
| 14:57.44 | sergeus | please somebody help me! :) |
| 14:58.00 | adibar | definitly. I like tha feature of the auto-hangup if remote is off |
| 14:58.58 | _Sam-- | i like the 1 touch dial with the phone on hook, and the blf, and the improved soung quality |
| 14:59.11 | Falle | _sam--: i suppose the sidetone support is gone in .8 for you aswell? |
| 14:59.18 | adibar | less echo ! |
| 14:59.32 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:59.39 | _Sam-- | i only have one on .8 right now, the rest are .6, but i didnt notice anything |
| 15:00.36 | Falle | I just want to find out if its related to hardware version of the phone or if its generally turned off in the firmware again. |
| 15:00.53 | adibar | I currently have only one GXP2000 right now, the rest is future-thinking ;-) |
| 15:02.55 | _Sam-- | i have a stupid and unrelated question....but why wouldnt 'hint' be on by default for any extension...like why do oyu have to specify it in extensions.conf per extension.... |
| 15:03.02 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:03.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:03.09 | _Sam-- | does it take many resources? |
| 15:03.38 | _Sam-- | er i guess it wouldnt know how to hint |
| 15:03.46 | _Sam-- | if it didnt know what channel to look at |
| 15:04.28 | Falle | I cant help you there.. haven't tried hints yet :/ |
| 15:04.39 | _Sam-- | works well on the gxp |
| 15:04.43 | Falle | ok |
| 15:04.50 | _Sam-- | what do you use the buttons on the side for? |
| 15:05.04 | Falle | right now. nothing |
| 15:05.15 | _Sam-- | you will like it |
| 15:05.34 | adibar | speeddialing |
| 15:05.34 | rene- | is a quad xeon strong enough for 6xE1 + 300 sip users, no transcoding, with recording? 150 of the users will be agents so the system is half pbx half predictive dialer |
| 15:06.03 | _Sam-- | i have system widw speeddial |
| 15:06.06 | Falle | I was planning to use BLF support with my DND function in asterisk. just havent started figguring out how yet :) |
| 15:06.16 | _Sam-- | it takes about 2 minutes |
| 15:06.32 | _Sam-- | add a line like this to extensions.conf: |
| 15:06.34 | _Sam-- | exten => 103,hint,sip/103 |
| 15:06.41 | _Sam-- | and set the phone to check 103 / blf for the button |
| 15:06.44 | adibar | U can set it in the webiface |
| 15:06.54 | _Sam-- | you need to do both...have to set it in extensions.conf |
| 15:07.22 | _Sam-- | and for some reason, i think you have to restart asterisk all the way for it to work, not just extensions reload |
| 15:07.25 | _Sam-- | i could be wrong. |
| 15:07.34 | Falle | can hints be used with variables? |
| 15:07.48 | _Sam-- | i dont see why not |
| 15:08.05 | Falle | if $DND1000 = 1, gxp light up |
| 15:08.44 | _Sam-- | i see what you're saying...im not sure how to do that. |
| 15:09.28 | Falle | thats my problem.. havent really figgured that out yet. :) |
| 15:09.44 | _Sam-- | just call an extension for DND |
| 15:09.53 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 15:09.53 | _Sam-- | exten => 1000, answer |
| 15:09.59 | _Sam-- | then dial 1000 and leave |
| 15:10.04 | _Sam-- | probably easier anyway |
| 15:10.18 | Falle | Im dialing a set of extensions on my asterisk to set this DND1000 variable and then use the variable in my dialplan to route calls. |
| 15:10.26 | asterisk99 | Anyone here using Asterisk on a Athlon? (specifically an Athlon shuttle?) |
| 15:11.03 | _Sam-- | if the last part of the dialpan didnt disconnect you when you called to set DND1000, then you would light up as busy on all the phones |
| 15:11.17 | _Sam-- | but i do see what you're saying. |
| 15:11.44 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 15:12.07 | mrtwister|mobile | hello, what settings for sip dtmf mode should i use? in cisco it is dtmf-relay rtp-nte h245-signal h245-alphanumeric |
| 15:12.11 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 15:17.09 | Splas | _Sam--: I'm using the hint stuff /w a Polycom 601 + Attendant Console.. Works pretty good, cept the phone gets real confused if you reload asterisk and requires a reset before it'll properly monitor the hints again |
| 15:18.40 | _Sam-- | hmmm i havent used it long enough with my phones to know if they will work that same way |
| 15:18.53 | _Sam-- | even just from the cli "reload" breaks it? |
| 15:18.57 | _Sam-- | and you have to restart the phone? |
| 15:19.47 | Falle | bye adibar |
| 15:20.00 | adibar | Falle. cya later |
| 15:20.32 | [TK]D-Fender | _Sam-- : Yeah, reload breaks HINTS.... |
| 15:20.56 | adibar | file: don't hesitate... Do it... Living without feeding yourself can be a painfull experience ;-) |
| 15:21.08 | *** join/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch) |
| 15:21.09 | [TK]D-Fender | asterisk99 : I run * on them... (not Shuttle specifically, but Gigabyte MB's + AMD XP's) |
| 15:21.13 | arkanis | hi |
| 15:21.22 | arkanis | I have a rather big problem *g* |
| 15:21.40 | [TK]D-Fender | arkanis : Try #drphil |
| 15:21.45 | Falle | haha :D |
| 15:21.52 | arkanis | Whenever I call outbound the callde person doesnt hear me, but I can hear them |
| 15:21.52 | _Sam-- | [TK]D-Fender : restart gracefully will leave the phones fine? |
| 15:22.45 | [TK]D-Fender | _Sam-- : Not sure... that'd be a thing to test, but depending on call volume would create a potentially LONG wait before changes would take effect. |
| 15:22.52 | *** join/#asterisk coppice (n=chatzill@1.168.17.210.dyn.pacific.net.hk) |
| 15:23.12 | _Sam-- | yeah we are just a pretty small place ...restart gracefully = never longer than 5 minutes |
| 15:23.18 | _Sam-- | but i understand what you're saying |
| 15:23.20 | droops | hey arkanis aer you using sip or iax |
| 15:23.24 | mut | I FOUND IT! |
| 15:23.37 | docelm0 | My weather script is done! woo hoo! |
| 15:23.42 | arkanis | I'am using SIP |
| 15:23.50 | mut | problem with my dslam & loading voip config/webpages |
| 15:23.55 | droops | is either one of you behind nat |
| 15:24.04 | arkanis | yes me |
| 15:24.19 | droops | you need to forward through your router |
| 15:24.22 | arkanis | but I have client accept all, server accept all |
| 15:24.27 | arkanis | in firehol.conf |
| 15:24.29 | _Sam-- | bkw__: thanks for sending me to asterlink...its all good except that guy file. |
| 15:24.32 | droops | sip doesnt work with nat |
| 15:24.37 | arkanis | It worked until today |
| 15:24.54 | droops | thats all i got man |
| 15:24.58 | arkanis | whats really curious is, that one of our employees actually can phone |
| 15:25.04 | file | yes, I'm absolutely horrible |
| 15:25.11 | _Sam-- | :) |
| 15:25.16 | MikeJ[Laptop] | indeed |
| 15:25.45 | _Sam-- | bkw__: i cant stop telling everyone about the guy at asterlink that answered my stupid question at 11:30pm on a friday night |
| 15:25.55 | _Sam-- | its pretty rare, at least it has been for me. |
| 15:26.24 | *** part/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 15:26.39 | [TK]D-Fender | _Sam-- : They outsourced him from somewhere overseas where it's really 11:00am and he starts at 10am :D |
| 15:26.47 | _Sam-- | hahah! |
| 15:27.15 | [TK]D-Fender | <- Only SOMEWHAT horrible |
| 15:28.08 | file | why can't my bills stay paid?!? |
| 15:28.14 | arkanis | @droops we used firehol |
| 15:28.25 | arkanis | at it worked until today :-/ |
| 15:29.19 | [TK]D-Fender | file : Have you tried discontinuing your accounts with them? That sometimes works.... |
| 15:29.39 | file | [TK]D-Fender: I somehow doubt that would work... they'd probably send collection agencies after me |
| 15:29.42 | file | "You owe us 10 cents!" |
| 15:30.00 | _Sam-- | hey file do you guys run your credit card batch at a certain time? |
| 15:30.03 | file | I would then send them 11 cents, and demand my 1 cent back |
| 15:30.12 | _Sam-- | im just wondering if it matters when i should i make payment |
| 15:30.25 | file | _Sam--: it runs every few minutes or something I think |
| 15:30.39 | file | so, just like make a payment and it'll happen eventually :) |
| 15:31.11 | _Sam-- | if i pump a lot of money into my current account, when i setup a second one, can you transfer any funds between them (i know im a pain, just asking) |
| 15:31.21 | file | yes |
| 15:31.38 | _Sam-- | k, thanks |
| 15:31.46 | file | I can manually adjust the balance and credit/debit them accordingly |
| 15:32.52 | arkanis | does somebody know why they can't hear me, when I'm phoing |
| 15:32.55 | arkanis | phonig |
| 15:34.20 | _Sam-- | hey file...just an idea...some pagination on the transaction histroy page or way to select a range would be a nice option! |
| 15:38.15 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj2j.dialup.mindspring.com) |
| 15:38.34 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:43.34 | *** join/#asterisk Prival (i=user69@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 15:44.46 | Prival | Anyone has a PRI and can dial the 514-935-8435 to see if they can navigate their menus? I have a PRI install and we we dial this number, we can't naviagte their menus... |
| 15:45.39 | Prival | BTW, from a standard analog line it works, but looks like through a PRI it does not |
| 15:48.17 | arkanis | strange, it seems that we can't call mobile-phones |
| 15:48.44 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 15:48.53 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj2j.dialup.mindspring.com) |
| 15:49.37 | Ahrimanes | anyone using rxfax here? |
| 15:49.44 | sergeus | any more experts in SIP+NAT ? :) |
| 15:49.59 | *** join/#asterisk TallAndy (i=TallAndy@82.163.6.217) |
| 15:50.01 | Ahrimanes | sergeus: i have some experience |
| 15:50.10 | Prival | sergeus: was able to get it working also |
| 15:50.18 | Prival | Anyone has a PRI and can dial the 514-935-8435 to see if they can navigate their menus? I have a PRI install and we we dial this number, we can't naviagte their menus... |
| 15:50.20 | *** join/#asterisk astnoob_ (n=astnoob@static-209-137-161-171.t1.cavtel.net) |
| 15:50.31 | TallAndy | Hi, does anyone know how I could bridge to calls, channels together in asterisk, pref using phpagi? |
| 15:50.40 | *** join/#asterisk SwK (n=SwK@64.89.118.139) |
| 15:50.55 | Ahrimanes | i'm receiving faxes just fine, but looks like rxfax thinks it should save them in landscape instead og portrait |
| 15:51.03 | Ahrimanes | anyone seen this behaviour? |
| 15:51.14 | sergeus | here is my question: |
| 15:52.04 | sergeus | why with NAT=yes option asterisk keeps trying to send RTP packets to NATed address? (SIP client behind NAT) |
| 15:52.48 | *** join/#asterisk coppice (n=chatzill@87.162.17.210.dyn.pacific.net.hk) |
| 15:53.04 | *** join/#asterisk meljack (n=esculli@host172138.metrored.net.ar) |
| 15:53.08 | sergeus | i.e. why Symmetric RTP doesn't work - what can cause such missbehaviour? |
| 15:54.47 | sergeus | any ideas? :) |
| 15:55.19 | malverian[work] | What method do you guys use for recording a specific channel on demand? |
| 15:55.27 | iCEBrkr | werd. |
| 15:55.45 | malverian[work] | Seems like ChanSpy + Record would work.. |
| 15:55.55 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 15:55.56 | iCEBrkr | asterisk99: Use pastebin next time! |
| 15:56.06 | malverian[work] | But seems a bit hackish.. surely there's a better way? |
| 15:56.19 | asterisk99 | Anyone here using Asterisk on a Athlon? |
| 15:56.44 | brad_mssw | sure |
| 15:56.54 | brad_mssw | ours runs on an athlon xp 2200+ |
| 15:56.56 | Katty | and now it's time for, spot the american tourist! |
| 15:57.23 | Katty | what /not/ to wear on vacation to the uk. |
| 15:57.33 | Katty | input? :) |
| 15:57.54 | malverian[work] | Oh.. interesting.. |
| 15:58.23 | malverian[work] | Chanspy(SIP/foo,r) |
| 15:58.30 | malverian[work] | Records to /var/spool/asterisk/monitor :-D |
| 15:58.32 | malverian[work] | Uber. |
| 15:58.37 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:59.14 | iDunno | Katty: don't wear shorts and a t-shirt in the middle of winter, that'd be a dead giveaway ;) |
| 15:59.27 | Katty | iDunno: ha. |
| 16:00.19 | iDunno | also, don't have huge numbers of cameras laying about your person, just the one will do ;) |
| 16:00.43 | Katty | hey! i'll be 20lbs lighter then! |
| 16:00.52 | *** join/#asterisk HamYaI (i=HamYai@125.24.7.134) |
| 16:01.18 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 16:01.52 | *** join/#asterisk TelcoDE (n=ed@c-68-82-75-221.hsd1.de.comcast.net) |
| 16:02.33 | austinnichols101 | anyone got their hands on an SPA941 or SPA942 yet? |
| 16:02.50 | kll | I'm having problems distinguishing the differences between a sip friend/user/peer... I'm currently using friend, but someone told I should use type=peer instead, however when switching I get an authentication error |
| 16:03.57 | HamYaI | Can anyone comment on "S100-FX v2.0" and "Sipura SPA-3000"? |
| 16:04.27 | jaiger | HamYaI, I've used the spa3000 with good results |
| 16:04.54 | TallAndy | Can I link 2 channels together from the asterisk manager? |
| 16:04.58 | HamYaI | jaiger: the "S100-FX v2.0" has the same features, right? |
| 16:05.15 | dpryo | TallAndy: Two active channels? |
| 16:05.22 | malverian[work] | Hm.. |
| 16:05.27 | TallAndy | dpryo: Yes |
| 16:05.33 | dpryo | TallAndy: You could originate a call to both of them. |
| 16:05.35 | malverian[work] | Now to figure out how to _schedule_ recording of a specific channel.. |
| 16:05.36 | HamYaI | jaiger: I mean one FXO and one FXS |
| 16:05.46 | iCEBrkr | malverian[work]: gotoiftime() |
| 16:06.07 | TallAndy | dpryo: Make call to A party they decide if the want to accept, then B party is called, then calls are bridged. |
| 16:06.20 | dpryo | TallAndy: Yeah, that's easy. |
| 16:06.46 | dpryo | TallAndy: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate |
| 16:06.55 | TallAndy | dpryo: I'm using phpagi at the mo for so I can excute commands through that |
| 16:07.03 | TallAndy | Ok so you can link using originate |
| 16:07.10 | dpryo | Yes |
| 16:07.15 | *** join/#asterisk hmerry (n=hmerry@66.0.23.226) |
| 16:07.23 | TallAndy | dpryo: Thanks! |
| 16:07.34 | HamYaI | <PROTECTED> |
| 16:07.51 | hmerry | I am looking for a suggestion for a wireless VOIP hardphone that works well with asterisk? |
| 16:08.33 | dpryo | cisco 7920 might work with sccp? |
| 16:08.49 | dpryo | I've lost my charger, so haven't tested it ;P |
| 16:09.23 | malverian[work] | iCEBrkr, Well.. I'd actually have to use a cron job to start a call to do the chanspy I think... |
| 16:09.57 | [TK]D-Fender | austinnichols101 : I own an SPA-941 |
| 16:10.04 | malverian[work] | iCEBrkr, Or I guess I could use gotoiftime any time they dial an extension (and call monitor if it's in between that time) |
| 16:10.17 | [TK]D-Fender | Prival : IVR seems fine on that number |
| 16:10.18 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 16:10.54 | TallAndy | dpryo: Sorry dpryo, so I would use originate twice to make two calls, then orignate a third time to bridge them? |
| 16:11.06 | iCEBrkr | malverian[work]: I was assumeing the second solution |
| 16:11.12 | kll | could someone please outline the differences between user, peer and friend entries in the sip.conf file? |
| 16:11.19 | iCEBrkr | malverian[work]: I just record all calls :P |
| 16:11.47 | dpryo | TallAndy: You send one originate request. |
| 16:11.48 | iCEBrkr | [root@chrome monitor]# ls -la | wc -l |
| 16:11.48 | iCEBrkr | 73 |
| 16:11.48 | iCEBrkr | [root@chrome monitor]# du -sh |
| 16:11.48 | iCEBrkr | 11M . |
| 16:11.53 | hmerry | The 7920 is definitely beta quality, at most: I have |
| 16:11.53 | hmerry | experienced dropped connections as well as the phone ignoring remote |
| 16:11.53 | hmerry | hangup, on more that 50% of my calls in a 100% Cisco environment, with a |
| 16:11.53 | hmerry | Cisco AiroNet access point within 3m, direct line-of-sight. So I guess it |
| 16:11.53 | hmerry | will take a year or two (just as with the 7960/7940) before Cisco gets the |
| 16:11.54 | hmerry | firmware in a state that can be considered production quality. |
| 16:12.08 | hmerry | from http://lists.digium.com/pipermail/asterisk-dev/2004-January/002793.html |
| 16:12.25 | Nugget | that was two years ago, so I guess they work perfectly now! ;) |
| 16:12.48 | hmerry | thanks ;-P |
| 16:13.02 | *** join/#asterisk Jizzbug (n=derekm@199.227.154.26) |
| 16:13.06 | asterisk99 | Anyone here using Asterisk on a *ATTHLON*? |
| 16:13.22 | dpryo | TallAndy: You could see if you could get anythint out of this: http://hn.iw.no/originate.txt |
| 16:13.27 | Nugget | My coomputer iss ann atthlon. |
| 16:13.27 | asterisk99 | (egad... I can't type... ATHLON) |
| 16:13.29 | TallAndy | ok |
| 16:14.00 | dpryo | TallAndy: Forgot to add $first_number to the $channel var.. |
| 16:14.47 | TallAndy | dpryo: I will take a look, what happens is call 1 is made and there maybe upto a minute before the second calls is generated |
| 16:15.08 | jaiger | HamYaI, I don't have any issues with the spa3000s. I am happy with Sipura hardware so far. |
| 16:15.10 | TallAndy | dpryo: so originate for 1st call, then orignate again for 2nd call linking to call 1 |
| 16:15.15 | docelm0 | asterisk99, yes why? |
| 16:15.32 | TallAndy | dpryo: does that make sense? :P |
| 16:15.34 | dpryo | TallAndy: When the first call is answered, it starts to call the other one, so the first user will hear a ringing tone |
| 16:15.48 | asterisk99 | _docelm0: Works fine? I am deciding whether to stick with Pentium or risk Athlon |
| 16:15.57 | dpryo | TallAndy: You only do the originating command once. |
| 16:15.57 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 16:16.09 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 16:16.28 | docelm0 | Over all I prefer performance of Intel over AMD for Asterisk |
| 16:16.43 | TallAndy | dyro: Basically call 1 will be answered and if the person types say "*1" that will accept and ring the designated number and then bridge the calls |
| 16:17.02 | TallAndy | so far its all be done through phpagi |
| 16:17.27 | dpryo | TallAndy: No. The first one won't get to press a key or something. |
| 16:17.43 | dpryo | TallAndy: Then you'd have to originate him to an extension/service (agi) first. |
| 16:17.56 | asterisk99 | docelm0: Your Athlon works fine? I am deciding whether to stick with Pentium or risk Athlon |
| 16:17.59 | KranZ | anyone having issues checking out of svn? |
| 16:18.01 | dpryo | TallAndy: And do the checks, and then dial the other party |
| 16:18.15 | KranZ | asterisk99: i use a dual opteron setup |
| 16:18.18 | KranZ | no problems |
| 16:18.29 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 16:18.36 | stas | hi, we have 3 numbers that can be dialed how can i see or send to a phone to seee which number is dialed ? Is this possible in asterisk.. This phone desk.. |
| 16:18.54 | TallAndy | dypro: ok but then how would i join the two calls? |
| 16:18.59 | *** join/#asterisk xhoms (n=XHOMS@234.Red-80-34-155.staticIP.rima-tde.net) |
| 16:19.09 | *** part/#asterisk xhoms (n=XHOMS@234.Red-80-34-155.staticIP.rima-tde.net) |
| 16:19.10 | asterisk99 | Anyone here using Asterisk on a Shuttle XPC? (That's thems thar teeny tiny boxes from shuttle.com) |
| 16:19.30 | bweschke | asterisk99: yes |
| 16:19.33 | KranZ | i have it running on a mini-itx board |
| 16:19.53 | asterisk99 | KranZ: Which Digium card? |
| 16:20.08 | KranZ | no card on the itx board |
| 16:20.13 | MRH2 | Hi I am recording calls under a single directory( linux2.6 / ext3) is there a practicle upper limit on the number of recording files i should have under a single directory? |
| 16:20.15 | KranZ | its a remote host |
| 16:20.31 | KranZ | got a te405 on the dual opteron setup |
| 16:21.00 | KranZ | asterisk99: tho i dont see why it couldnt handle one |
| 16:21.01 | *** join/#asterisk Flyboy-SR22 (i=rsears@gateway.adnc.com) |
| 16:21.16 | KranZ | asterisk99: besides the reason the case i got is so small you need a 2.5" HD |
| 16:21.47 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
| 16:21.50 | asterisk99 | KranZ: Me neither... but there have been some reported problems with Athlons and Digium cards |
| 16:22.21 | KranZ | only issue i had with the opterons was a Bus issue with the te405 card and the onboard sata |
| 16:22.39 | KranZ | couldnt move slots so i bought a pci sata card with the same chipset as the onboard |
| 16:22.40 | jaiger | asterisk99, I'm using mini-itx with a T100P card for 5 channels |
| 16:22.41 | asterisk99 | KranZ: You have a small case? You have a link? |
| 16:23.06 | jaiger | asterisk99, it's a VIA C3 cpu tho |
| 16:23.26 | kll | could someone please outline the differences between user, peer and friend entries in the sip.conf file? |
| 16:23.51 | jaiger | ~peer |
| 16:23.52 | jbot | i heard peer is the most elusive script kiddie this side of Jupiter |
| 16:23.58 | jaiger | ~user |
| 16:23.59 | jbot | extra, extra, read all about it, user is currently detached. Talk to this user upon their return. You will now be ignored. [HackFactor Elite 2.0], or a synonym for moron |
| 16:24.12 | KranZ | jaiger: im using an itx board with a 1.2ghz pIII |
| 16:24.25 | TallAndy | 1. Make call 2. If user enters "*1" make second call 3. Bridge(call 1, call 2) |
| 16:24.26 | asterisk99 | jaiger: What case are you using? (I'm looking for small & sexy ) :) |
| 16:24.50 | KranZ | asterisk99: im using the cubid 3677 |
| 16:24.51 | kll | jaiger: huh? neither one seems very attractive. |
| 16:25.07 | jaiger | asterisk99, I have a 1U case but I forget the brand |
| 16:25.42 | jaiger | kll, user can place calls and peer can accept calls from your *. friend is a union of user+peer |
| 16:27.20 | kll | alright |
| 16:27.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:27.29 | [TK]D-Fender | Dammit, I have a D-Channel thats constantly dropping, anyone know how I can try and force it back up? |
| 16:27.40 | docelm0 | shut no shut |
| 16:27.47 | nestar | lol |
| 16:27.50 | [TK]D-Fender | ? |
| 16:27.56 | docelm0 | its a cisco thing |
| 16:29.44 | iCEBrkr | It's a Cisco thing, you wouldn't understand. |
| 16:30.58 | [TK]D-Fender | &^%@#%&^*@#&%^%^&@#%&^@#%&^*&%^*@# @#^&% &^*% &^*% ^&%^%$^% ^$%$^%&$^%& $%^& $%^& $%^& ^$%& $%^& $^%& |
| 16:32.12 | Nivex | [TK]D-Fender: is that Perl? |
| 16:32.16 | tzafrir_laptop | [TK]D-Fender, is that some UTF-8 garbage? |
| 16:32.26 | rene- | hi |
| 16:33.47 | rene- | for the network ninja out there: i have a cgi that spits xml configuration for ip phones. i assume that if my environment variables tell me that remote port is different than 80 then it has been changed by a natp device and the device is behind nat, am i right or not? |
| 16:34.50 | bweschke | rene- : no. not necessarily |
| 16:35.16 | *** join/#asterisk SenorAmor (n=me@ns1.accu-com.com) |
| 16:35.19 | Bakermd | Anyone here used the Fonality product offering? |
| 16:35.22 | rene- | bweschke: please elaborate |
| 16:35.30 | SenorAmor | When I make outgoing calls (out my T1), my recipient gets a bunch of static and rapid beeps (like morse code). Any ideas on what I can do to improve the quality of my calls? |
| 16:35.36 | SenorAmor | Everything sounds a-ok on my end. Just the recipient has quality issues. |
| 16:36.03 | SenorAmor | (I'm an asterisk newbie) :( |
| 16:36.36 | bweschke | Bakermd: what do you need to know about it? |
| 16:37.13 | Katty | woah |
| 16:37.17 | Katty | they found a cure for hiv/aids |
| 16:37.19 | Bakermd | bweschke: Is it a product to recommend? Are the features functional? Does it expand if needed to multi- server setups |
| 16:37.28 | Bakermd | Katty: Abstenance? |
| 16:37.30 | austinnichols101 | katty: possible cure |
| 16:37.35 | meljack | I am developing a application, this use “Manager API” to connect with Asterisk. But when I call to an external number (over a zap channel), I don’t receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? |
| 16:37.39 | Katty | austinnichols101: better than nothing ((((= |
| 16:37.47 | Katty | Bakermd: you can be born with it, dear. |
| 16:38.03 | bweschke | rene-: behind nat or not, your remote port contacting a web server at port 80 is not like to be port 80 on the other side |
| 16:38.18 | Bakermd | Katty: Not being 100% serious of course |
| 16:38.34 | austinnichols101 | katty: hasn't even passed peer review yet. That puts the news just above 'rumor' |
| 16:38.48 | Katty | austinnichols101: better than nothing ((((= |
| 16:39.13 | bweschke | Bakermd: it's a mature product as far as GUI goes... I don't believe that it does support "multi server" setups, but you'd need to confirm that with them. The other thing is that you don't really have the GUI running on the server you purchase from them. The GUI runs at their colo and then they push configs to your box so your box must always be internet connected |
| 16:39.23 | meljack | I am developing a application, this use “Manager API” to connect with Asterisk. But when I call to an external number (over a zap channel), I don’t receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? |
| 16:39.48 | bweschke | meljack: what kind of zap channel? fxo? pri? |
| 16:40.01 | Bakermd | bweschke: That makes it a LOT clearer. Thank you very much. |
| 16:40.16 | rene- | bweschke: so, is there other way? |
| 16:40.24 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 16:40.34 | SenorAmor | When I make outgoing calls (out my T1), my recipient gets a bunch of static and rapid beeps (like morse code). Any ideas on what I can do to improve the quality of my calls? |
| 16:40.41 | meljack | i have a tdm400 |
| 16:41.32 | bweschke | rene-: to know whether or not they're behind NAT from a web hit? well... that's tough... it's alot easier with SIP (because devoid of STUN attempts on the UA side, the agent plugs in their internal IP into the SIP request and then it's pretty easy to know) |
| 16:41.52 | malverian[work] | iCEBrkr, What format do you record to? |
| 16:41.57 | bweschke | rene-: with a web hit you're not going to see that internal IP. the web server is going to log the IP of the external side of the NAT device |
| 16:42.05 | Flyboy-SR22 | How does FOnality compare to A@H in features and usability..? |
| 16:42.23 | Flyboy-SR22 | I did not like the fact that you had to buy the Fonality "white-box" server |
| 16:42.36 | Flyboy-SR22 | I like to deploy production * on Dell equipment myself |
| 16:42.43 | Flyboy-SR22 | they have no option for that... |
| 16:42.58 | bweschke | Flyboy-SR22: It's got some outlook integration and click to dial stuff that A@H can't offer. However, you're spot on. You've got to buy their boxen and you don't ever get away from their managed service offering |
| 16:43.05 | austinnichols101 | Flyboy-SR22: I spoke with them before installing AAH. Found them quite expensive. Features seemed to be more-or-less similar |
| 16:43.29 | bweschke | Flyboy-SR22: they'll quote you a Dell box.. you just need to ask them.. |
| 16:43.41 | bweschke | Flyboy-SR22: or at least they used to |
| 16:43.43 | TallAndy | Is there a asterisk manager or AGI function that can bridge calls? |
| 16:43.47 | docelm0 | hay whats the command for collecting digits in asterisk? |
| 16:44.03 | Flyboy-SR22 | I have A@H doing Outlook dial |
| 16:44.14 | malverian[work] | What formats do you all use for voicemail? |
| 16:44.28 | bweschke | right - but you did that through your own integration work, right? |
| 16:44.30 | Flyboy-SR22 | and SugarCRM |
| 16:45.15 | bweschke | I think with Fonality the idea was that it's a "turnkey" solution... they don't really want you going in and adding your own stuff like you can with AAH and the regular distribution.. so if your goal is to tinker - you're probably on the right road with AAH |
| 16:45.16 | docelm0 | anyone? |
| 16:45.20 | austinnichols101 | Pbxtra Call Center Edition = 2k + annual maintenance |
| 16:45.59 | austinnichols101 | not that there's anything wrong with charging for your product/services |
| 16:46.13 | docelm0 | Can someone help me with getting digits from a caller? I just need to know the command |
| 16:46.18 | bweschke | Read |
| 16:46.27 | bweschke | docelm0 that is |
| 16:46.28 | bweschke | :) |
| 16:46.51 | docelm0 | ohh ok Sorry.. never used it before.. Just know it was out there .. :) |
| 16:47.00 | SenorAmor | When I make outgoing calls (out my T1), my recipient gets a bunch of static and rapid beeps (like morse code). Any ideas on what I can do to improve the quality of my calls? |
| 16:47.03 | SenorAmor | Anyone? |
| 16:48.12 | bweschke | SenorAmor: what kind of echo training configuration do you have turned on that trunk? |
| 16:49.04 | Flyboy-SR22 | bweschke, Thanks - I am looking for something for my customers...I like the A@H becuase of the CRM and others tools already integrated |
| 16:49.28 | Flyboy-SR22 | bweschke, I had heard about Fonality and SomethingVOX that both require you to use their hardware |
| 16:49.42 | Flyboy-SR22 | bweschke, but never found anyone who is actually using them |
| 16:49.50 | SenorAmor | echocancel=128 |
| 16:49.50 | SenorAmor | echocancelwhenbridged=yes |
| 16:49.56 | SenorAmor | Is that what you're asking, bweschke |
| 16:50.34 | SenorAmor | I'm an Asterisk rookie, unfortunately. |
| 16:50.50 | bweschke | Flyboy-SR22: it depends how "hands on" you want to be with your customer's installs. if you're looking for an outsource - I'd look at reselling fonality or switchvox. If not, you may do OK with A@H or building your own solution |
| 16:51.08 | *** join/#asterisk Garak_ (n=garak@209.5.171.170) |
| 16:51.28 | Flyboy-SR22 | :-) Hands on = $$ per hour right..? |
| 16:51.30 | Flyboy-SR22 | :-) |
| 16:51.51 | Flyboy-SR22 | They are expensive in the grand scheme of thing it seems |
| 16:52.02 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:52.19 | bweschke | SenorAmor: that's part of it |
| 16:52.28 | bweschke | but what does echotraining= say? |
| 16:52.40 | SenorAmor | I don't have that line |
| 16:52.50 | SenorAmor | What's the syntax? |
| 16:52.51 | Garak_ | How do you configure /etc/zaptel.conf for both a T100P and a TDM400P? |
| 16:53.17 | bweschke | well |
| 16:53.34 | bweschke | it sounds like you might be getting echotraining that's a bit agressive at the beginning of the call |
| 16:53.45 | bweschke | you can try setting echotraining=no and see if that helps |
| 16:54.12 | bweschke | Flyboy-SR22: sure. professional services is how my company makes most of it's money |
| 16:54.41 | SenorAmor | ok. echotraining is also in zapata.conf, I presume? |
| 16:54.55 | bweschke | it's only in zapata.conf - not in zaptel.conf |
| 16:55.19 | Flyboy-SR22 | bweschke, Does fonality or switchvox have a crm solution builtin like A@H that you know of..? |
| 16:55.21 | SenorAmor | Right. I meant 'also' as in 'also with the other echo lines' |
| 16:55.21 | bweschke | Garak_: just setup channels 1-24 for the T100P and 25-28 for the TDM400P |
| 16:55.35 | bweschke | Flyboy-Sr22 - yes I believe switchvox is integrated with sugar |
| 16:55.55 | Garak_ | bweschke: thats what I thought, but I was getting an error, maybe it wasn't related |
| 16:56.23 | *** join/#asterisk vattern (n=vattern@dsl-146-150-218.telkomadsl.co.za) |
| 16:56.45 | Flyboy-SR22 | bweschke, thanks - I guess I will take a closer look..if they will do Dell, then I may look at it...with switchvox do you have to have their update service..? |
| 16:57.23 | bweschke | Flyboy-SR22: I'm really not sure. You'd need to ask them. |
| 16:57.57 | *** join/#asterisk prog (n=prog@ip-85-160-89-226.eurotel.cz) |
| 16:58.13 | bweschke | Flyboy-SR22: they just seem like good folks (SwitchVox that is) - they're at the shows with Digium when we go and they seem well respected by the Asterisk community and that's generally a good thing when looking for a partner |
| 16:58.23 | SenorAmor | bweschke: adding 'echotraining=no' didn't change anything. Have any other ideas? :) |
| 16:58.34 | *** part/#asterisk Garak_ (n=garak@209.5.171.170) |
| 16:58.58 | bweschke | SenorAmor: is it a PRI? |
| 16:59.12 | file | I like the SwitchVox people |
| 16:59.16 | Flyboy-SR22 | bweschke, good deal - I think I will talk to them today...how is their support..? |
| 16:59.33 | *** part/#asterisk prog (n=prog@ip-85-160-89-226.eurotel.cz) |
| 17:00.10 | bweschke | Flyboy-SR22: I haven't any personal experience with their support, but I've heard nothing negative about them - which says something in this small community |
| 17:00.29 | Flyboy-SR22 | bweschke, that is very true !!! |
| 17:01.05 | SenorAmor | bweschke: no, e&m wink |
| 17:02.24 | bweschke | you could try setting echocancel=no and echocancelwhenbridged=no along with echotraining=no and see if you still have the problem with static and beeping. if you do, it may not be you- but maybe your carrier or someone in the middle doing that |
| 17:04.14 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 17:06.06 | SenorAmor | bweschke: Ok, setting all 3 to 'no' now puts echo on my end. I'm fairly certain that the problem lies with me and not my provider, as it worked perfectly before, and I'm now having problems since reinstalling everything on my voip box. |
| 17:06.55 | *** join/#asterisk nocnoc (n=cvenegas@216.241.1.58) |
| 17:07.17 | bweschke | right |
| 17:07.40 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 17:07.45 | bweschke | agree that it won't take away your echo problem :) -- however... does the static and beeping now go away when you turn that off? |
| 17:07.59 | *** join/#asterisk Reputation001 (n=Miranda@p548872D6.dip.t-dialin.net) |
| 17:08.37 | Reputation001 | hello |
| 17:09.01 | Reputation001 | i need help with indications.conf |
| 17:09.31 | SenorAmor | No. In fact my callee said the beeping was more prevalent now. Plus, I could now also hear beeping on my end. |
| 17:10.54 | Reputation001 | i tried to change the tones generated by Playtones() by setting country to 'de' in the general section |
| 17:11.07 | Reputation001 | but i still have us indications |
| 17:11.18 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
| 17:12.58 | mdave | how do I control what extension context is used if I receive an inbound call over SIP from another server that i'm not registered with? |
| 17:13.08 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 17:13.10 | mdave | or can I/ |
| 17:13.19 | mdave | er |
| 17:13.20 | mdave | nvm |
| 17:13.22 | mdave | i see it |
| 17:13.28 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 17:13.43 | bweschke | SenorAmor: don't really know what to tell you w/o being hands on with the system. You might try switching your EC algorithm in zconfig.h to MG2 and then try echotraining=400 instead of echotraining=no. |
| 17:14.27 | bweschke | "beeping" makes me think there's something more at the tdm network level that's going on that your carrier isn't happy with |
| 17:14.41 | SenorAmor | I currently have echocancel=128. I set it to 'no' and then to '256', both with horrible results. |
| 17:14.57 | markit | anyone here using visdn? |
| 17:15.03 | bweschke | have you cat /proc/interrupts to make certain your not sharing interrupts between your tdm card and anything else? |
| 17:15.12 | jaiger | SenorAmor, is 256 a valid value? |
| 17:15.17 | bweschke | jaiger: yes |
| 17:15.29 | SenorAmor | Yes, they appear to have their own IRQs |
| 17:15.45 | SenorAmor | jaiger: I'm not entirely sure. I thought it was factors of 2. |
| 17:15.49 | bweschke | 'they' ? |
| 17:15.56 | bweschke | how many cards do you have in the machine? |
| 17:16.02 | *** join/#asterisk ken____ (n=garak@209.5.171.170) |
| 17:16.11 | SenorAmor | Sorry. I have a T100 and TDM400 |
| 17:16.23 | _Sam-- | hmm what does this mean: |
| 17:16.24 | _Sam-- | Asterisk ended with exit status 139 |
| 17:16.27 | _Sam-- | er |
| 17:16.32 | *** join/#asterisk AlexCTI (n=alex@64.251.9.44) |
| 17:16.33 | SenorAmor | I'm only trying to go out the T1 card at the moment. |
| 17:16.35 | _Sam-- | usr/sbin/safe_asterisk: line 83: 23226 Segmentation fault (core dumped) asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY} </dev/${TTY} |
| 17:16.42 | _Sam-- | sucks |
| 17:17.11 | markit | is it possible to grab asterisk-1.2 svn at a certain date? I would like to try the code of some days ago... |
| 17:17.19 | *** join/#asterisk repuHell (n=me@p548872D6.dip.t-dialin.net) |
| 17:17.26 | bweschke | markit: sure - just find the revision you want and pull it |
| 17:18.07 | markit | bweschke: how can I tell the revision number? I just know the date... |
| 17:18.12 | bweschke | what date? |
| 17:18.28 | bweschke | trunk or 1.2 ? |
| 17:18.29 | markit | bweschke: the svn ok was of 2006/02/03 |
| 17:18.37 | markit | 1.2 |
| 17:18.42 | Reputation001 | do i have to change something else than country in indications.conf? |
| 17:19.05 | markit | bweschke: I use this syntax now: svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 17:19.49 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:20.27 | bweschke | markit: u can pull r9086 |
| 17:20.36 | markit | bweschke: thanks a lot! |
| 17:20.38 | bweschke | that's from 2/2 |
| 17:20.52 | markit | perfect |
| 17:21.18 | Reputation001 | *crying* i am all alone and with wrong indications |
| 17:21.18 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:21.26 | shmaltz | any problems with the list? |
| 17:21.44 | *** join/#asterisk ken____ (n=garak@209.5.171.170) |
| 17:21.47 | bweschke | SenorAmor: what's in the TDM400? FXS's or FXO's - where are the phones that this person is calling into from the T1? or are you calling out? |
| 17:22.21 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 17:22.25 | markit | I've problems in compiling visdn, it does not find version.h. I fixed it with "cp -a /usr/src/asterisk-1.2/include/asterisk/version.h /usr/include/asterisk/", but wondering if is an asterisk 1.2 svn bug in the install script |
| 17:22.38 | ken____ | . |
| 17:22.40 | jpablo | hey people, is there a way to load modules to asterisk at runtime ? i need to load chan_alsa |
| 17:23.35 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 17:23.47 | SenorAmor | bweschke: FXOs in the TDM400. I am making outgoing calls out the T100P to a standard POTS line |
| 17:24.18 | bweschke | sure |
| 17:24.27 | bweschke | jpablo: load chan_alsa.so |
| 17:24.32 | *** join/#asterisk ken____ (n=garak@209.5.171.170) |
| 17:25.58 | bweschke | SenorAmor: sorry - I know we've reviewed this already, but wctxxp in /proc/interrupts is on a line with an interrupt by itself? |
| 17:26.45 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:26.59 | *** join/#asterisk KEN____ (n=garak@209.5.171.170) |
| 17:27.08 | astnoob_ | Wondering if anyone has seen this before - I was running 1.0.x (4?) and recently upgraded to 1.2.4. Nothing changed in our config files other than upgrading extensions. We keep getting sip Username/Auth name mismatch errors for our 7960 users. Here are the erros and sip.conf as well as .cnf for the phone. Any help is appreciated! |
| 17:27.08 | astnoob_ | http://paste.uni.cc/8558 |
| 17:27.31 | *** join/#asterisk denon (i=denon@tor/session/x-e5bc45e791b5f6f0) |
| 17:27.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:28.23 | *** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de) |
| 17:28.38 | bweschke | astnoob: remove the double quotes around username |
| 17:28.41 | bweschke | in sip.conf |
| 17:28.43 | bweschke | and authname |
| 17:29.20 | astnoob_ | Thanks bweschke, I'll try that. |
| 17:29.50 | bweschke | and then pastebin a sip debug of "sip debug ip 172.16.0.120" on a registration attempt if you're still having trouble |
| 17:30.28 | SenorAmor | bweschke: yes (interrupts) |
| 17:30.57 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 17:31.02 | *** join/#asterisk Skid (i=cm@unaffiliated/skid) |
| 17:31.02 | bweschke | SenorAmor: this is a brand new T100P from Digium? |
| 17:31.44 | astnoob_ | bweschke: leave doublt quotes in Sip000xxx.cnf? |
| 17:31.48 | astnoob_ | er .. double |
| 17:32.00 | SenorAmor | bweschke: Yup. They even configured it for me. |
| 17:32.03 | bweschke | astnoob: not sure. one sec I'll check mine. |
| 17:32.15 | Skid | hiya, I'm having issues trying to get my head around contexts.. i've an outgoing provider, of which i'd only like my dialplan 'chrisdialplan' access to (i.e. so my user only would have access to make calls outgoing, and not others on the asterisk box) can anyone give a good example url/config please? |
| 17:32.20 | bweschke | SenorAmor: have you gone back to support with regard to the beeping? what do they say about it? |
| 17:32.25 | Skid | (yet also , still be able to access the stuff in default) |
| 17:32.44 | [TK]D-Fender | _Sam-- : I just seg faulted today too with * 1.0.9 CVS something or other.... |
| 17:32.59 | kll | how do I make dialplan "routing" descisions based on the time of the day and which day it is? |
| 17:33.15 | bweschke | astnoob: yes - double quotes in SIP00xxx.cnf |
| 17:33.18 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:33.31 | bweschke | kll: GotoIfTime |
| 17:33.55 | kll | thanks |
| 17:34.14 | _Sam-- | i was 1.2.4 |
| 17:34.19 | _Sam-- | got me worried |
| 17:34.35 | SenorAmor | bweschke: They keep giving me the same copy and paste response email. It's ungodly long and I've gone thru it twice. Stuff like the interrupts and making sure dma mode is enabled and such. |
| 17:34.39 | SenorAmor | I've done it all. |
| 17:35.06 | [TK]D-Fender | _Sam-- : I have a LOT of eyes (not the good kind) aimed at me today over it.... |
| 17:35.12 | bweschke | Sam: if you're seg faulting with 1.2.4 - make sure you've recompiled Asterisk with "make valgrind" - get the core dump and upload it to bugs.digium.com so someone can look at it |
| 17:35.23 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 17:35.51 | _Sam-- | if it happens once more i'll just go to back to 1.2.1 |
| 17:36.07 | znoG | you'll probably be helping the project if you supply the core dump |
| 17:36.13 | _Sam-- | we are a small biz here...but when it segfaulted we still lose about 10 calls |
| 17:36.19 | _Sam-- | znoG: i have it on pastebin if you want it :) |
| 17:36.27 | _Sam-- | er some gdb output at least |
| 17:36.28 | bweschke | _Sam-- : careful with that - you may hit the timebomb bug depending on what version of 1.2.1 |
| 17:36.34 | _Sam-- | my 1.2.1 was always fine |
| 17:36.40 | _Sam-- | <famous last words> |
| 17:36.43 | _Sam-- | but i ran it for a while |
| 17:36.47 | znoG | i'm not a * developer, unfortunately, but you probably want to upload it to bugs.digium.com as bweschke said |
| 17:37.08 | nocnoc | has anybody compiled the t38 thingie from www.ionidea.ua? |
| 17:37.16 | bweschke | Sam- u have the gdb output pastebin'd ? |
| 17:37.28 | _Sam-- | yep |
| 17:37.29 | Skid | how can I only allow certain users to dial out, via a context in my sip.conf please chaps, I can't find any docs that explain it detailed enough for me to grasp |
| 17:37.41 | _Sam-- | let me find the url |
| 17:38.05 | Skid | I've had a check on the voip-info wiki, but it doesn't detail outgoing - only really shows an example for incoming |
| 17:38.05 | _Sam-- | http://pastebin.com/545203 |
| 17:39.18 | [TK]D-Fender | Skid : the context DEFINES what they are allowed to dial. if the context includes entries that lead to outgoing resources, then thats what muct be changed. |
| 17:39.40 | [TK]D-Fender | s/muct/must |
| 17:40.07 | Skid | so if a user is in the default context, I need to create a new 'outgoing' context, put the user in that context, and include the default context into the outgoing context? |
| 17:40.34 | znoG | are the gdb outputs really helpful to debug crashes? i never really had a good look at one |
| 17:40.37 | *** join/#asterisk KEN____ (n=garak@209.5.171.170) |
| 17:40.51 | znoG | they show each step it took that it lead' up to the crash, right? |
| 17:41.04 | _Sam-- | znoG: to some people, like file on thsi channel.., i guess they understand something there :) |
| 17:41.12 | _Sam-- | to me it doesnt mean anything |
| 17:41.16 | znoG | yeah, agreed |
| 17:41.46 | znoG | wouldn't mean much to me either but it would show the function calls and stuff like that, so people who work on * code would likely make sense of it |
| 17:42.11 | znoG | would be nice though to understand the * code to be able to help out, i had a go at understanding it a few times but it doesn't make much sense |
| 17:42.21 | *** join/#asterisk lorinc (n=ang@caracas-3415.adsl.interware.hu) |
| 17:42.33 | KEN____ | how to delete the" ____ " behind my name |
| 17:42.34 | znoG | cause you not only have to know C (C++?) pretty well, you have to know about telephony as well |
| 17:42.49 | znoG | KEN____: actually its after your nick, not behind. try /nick KEN |
| 17:42.59 | znoG | ok so there's already a ken |
| 17:43.07 | znoG | ok there's all sorts of KEN's |
| 17:43.11 | znoG | so you're stuck with ken____ |
| 17:43.19 | znoG | try /nick _KEN |
| 17:43.19 | KranZ | what's the frequency kenneth |
| 17:43.53 | KEN____ | thx |
| 17:44.24 | KranZ | or /nick KEN` |
| 17:44.31 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 17:45.50 | SenorAmor | bweschke: It appears the static, reverb, and beeping is caused by my computer somehow |
| 17:46.03 | SenorAmor | When I make calls from my laptop, they're perfect. |
| 17:46.06 | Skid | hmm, i feel like a fool as I can't understand this :P |
| 17:46.13 | *** join/#asterisk KEN____ (n=garak@209.5.171.170) |
| 17:46.14 | SenorAmor | So, now, fix my computer :P |
| 17:46.21 | bweschke | SenorAmor: you're calling from a softphone? |
| 17:46.24 | SenorAmor | Thanks a ton for the help, man :) |
| 17:46.27 | SenorAmor | Yup. |
| 17:46.29 | SenorAmor | X-Lite. |
| 17:46.30 | KEN____ | nick_KEN |
| 17:46.36 | KranZ | heh |
| 17:46.37 | SenorAmor | On both my computer and laptop. |
| 17:46.52 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 17:47.06 | *** part/#asterisk KEN____ (n=garak@209.5.171.170) |
| 17:47.28 | *** join/#asterisk KEN____ (n=garak@209.5.171.170) |
| 17:47.35 | *** join/#asterisk JCC_ (n=chatzill@207.41.92.131) |
| 17:47.51 | bweschke | SenorAmor: how healthy is the cpu on the computer? is it getting taxed? if you're not getting the cycles it needs to run - you're going to get sound breakdown |
| 17:48.03 | bweschke | is xlite competing for the sound card with another application? |
| 17:48.10 | KEN____ | guys /nick KEN is not work |
| 17:48.20 | bweschke | you've opened up a wealth of new potential problems. :) |
| 17:48.37 | KranZ | ./slap KEN____ |
| 17:49.05 | KranZ | KEN____ that's because there is already a KEN |
| 17:49.06 | KEN____ | ./slap KEN___ |
| 17:49.16 | KEN____ | ok |
| 17:49.28 | KranZ | pick a different name |
| 17:49.44 | KEN____ | ok |
| 17:50.01 | SenorAmor | bweschke: CPU is good. There's gotta be something else going on with it. No biggie, however. Maybe it's just the network cable. |
| 17:50.43 | [TK]D-Fender | Skid : Yeah, that pretty much sums it up |
| 17:51.08 | KEN____ | nick_KEN |
| 17:51.24 | Skid | hmm, but surely |
| 17:51.31 | Skid | wait, gotta try and get my head arond it :) |
| 17:52.00 | Skid | dont think it'd work heh ngh |
| 17:52.47 | Skid | would you mind if I PM'd you a second, [TK]D-Fender ? |
| 17:53.45 | bweschke | Sam--: were you doing stuff on the CLI when you segfaulted? |
| 17:54.01 | _Sam-- | nope |
| 17:54.03 | _Sam-- | the cli was open |
| 17:54.07 | _Sam-- | on a different screen |
| 17:54.09 | bweschke | hmm... |
| 17:54.14 | bweschke | that's where the segfault is coming from |
| 17:54.21 | bweschke | on a read from cli input in asterisk.c |
| 17:54.51 | [TK]D-Fender | Skid : Sure |
| 17:54.55 | Skid | thanks |
| 17:55.01 | _Sam-- | i run 'screen' my ssh session...i flipped to that screen and noticed it had segfaulted |
| 17:55.07 | _Sam-- | i wasnt on it or near it |
| 17:55.41 | _Sam-- | but many thanks to you for looking at it. |
| 17:57.52 | [TK]D-Fender | ~pb |
| 17:57.54 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:59.05 | markit | with svn 1.2 table, is it normal not having /usr/include/asterisk/version.h ? or is the installation script fault? |
| 18:00.31 | *** part/#asterisk SenorAmor (n=me@ns1.accu-com.com) |
| 18:00.51 | bweschke | Sam -- which thread was gdb on when you opened it? |
| 18:01.57 | znoG | not sure what you guys know about yacc (using byacc) but I'm getting this error when building AEL2: |
| 18:02.00 | znoG | yacc: e - line 133 of "ast_expr2.y", syntax error |
| 18:02.03 | znoG | %pure-parser |
| 18:02.03 | znoG | ^ |
| 18:02.16 | znoG | which looks to me like % is not supported in my version of yacc, or something |
| 18:03.25 | bweschke | _Sam-- what do you get when you do "info threads" on gdb? |
| 18:05.11 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 18:06.28 | Netgeeks | mail lists down again? |
| 18:09.00 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 18:09.04 | g4m | join #python |
| 18:09.15 | g4m | oops sorry |
| 18:09.52 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 18:11.06 | Netgeeks | ... |
| 18:11.07 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.16.10.telnor.net) |
| 18:13.41 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 18:14.21 | *** join/#asterisk HuaFeng (n=garak@209.5.171.170) |
| 18:14.45 | AlexCTI | Hi guys, Do you recomend RedHad enterprise as a OS for set ASterisk on a DellServer SC1425? |
| 18:15.10 | Nugget | Any Unix is fine. Any linux if you need zaptel. |
| 18:15.33 | mroth_imm | AlexCTI: There is a thread on the users list all about your question right now : ) |
| 18:16.08 | znoG | and if you wait a day or two, there'll be a fresh new one! |
| 18:16.12 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:16.12 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 18:16.14 | Nugget | heh |
| 18:16.15 | AlexCTI | Cool |
| 18:16.45 | mroth_imm | If you are considering Asterisk Business Edition, you will find that RHEL3 or FC3 are the only real choices |
| 18:17.28 | mroth_imm | we're running FC3 on a Dell PowerEdge 6850 with no problems though, so you should be okay |
| 18:17.31 | AlexCTI | i have RHEL3 |
| 18:17.45 | mroth_imm | They are very similar in my experience |
| 18:18.00 | znoG | nice, upgraded bison and all is well now |
| 18:18.11 | tzafrir_laptop | RHEL3 ~~ Centos 3.x, right? |
| 18:18.14 | AlexCTI | and wich package should i choose on tha version? |
| 18:18.19 | znoG | ok not all, another compile problem |
| 18:18.32 | tzafrir_laptop | also, what's wrong with Debian (for ABE)? |
| 18:19.53 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 18:19.56 | robin_sz | meep? |
| 18:20.20 | mroth_imm | ABE installs asterisk and libpri via RPMs...the install script specifically looks for FC3 or RHEL3 and aborts if neither is found |
| 18:20.45 | mroth_imm | you could probably work around it, but then there is the tech support issue...full tech support is only available for FC3 or RHEL3 |
| 18:21.06 | mroth_imm | AlexCTI: i don't understand your package question... |
| 18:21.38 | AlexCTI | I mean, when you start to install it ask you for hundred of packages.. |
| 18:21.59 | *** part/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 18:22.15 | mroth_imm | if you're using RH, you should be able to download the packages listed here and be fine <http://asterisk.org/download> |
| 18:22.39 | AlexCTI | ok.. |
| 18:23.00 | mroth_imm | on other distros, i had to do some searching to find the right packages, but it's pretty straight forward for RH/FC |
| 18:23.20 | fulgas | humm |
| 18:23.25 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 18:23.26 | fulgas | just got a * crash |
| 18:23.29 | fulgas | on chan_sip |
| 18:23.42 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 18:23.57 | jhiver | ah bloody hell voipjet sucks so badly it's unreal |
| 18:24.29 | jhiver | why do I still have so much credit with them? it was stupid of me to transfer that money :) |
| 18:25.02 | mroth_imm | AlexCTI: good luck...gotta work now : ( |
| 18:25.31 | jhiver | I've got stupidly long connection times (to france and uk), no ringback tone, and audio cuts off after about 10 minutes or so |
| 18:25.39 | jhiver | moan moan moan |
| 18:25.44 | Katty | moaner. |
| 18:25.54 | jhiver | moan moan :) |
| 18:26.01 | jhiver | and also grump I would add |
| 18:26.25 | jhiver | grumpy moaner, I like the sound of this :) |
| 18:26.35 | AlexCTI | Thanks.mroth_imm |
| 18:26.42 | robin_sz | ooh, lookie! .. my GXP2000 display it australian ... |
| 18:26.45 | robin_sz | is |
| 18:27.05 | robin_sz | upside down .. and apparently pissed |
| 18:27.39 | *** join/#asterisk chaos1 (n=sa@82.196.211.247) |
| 18:27.40 | chaos1 | hi |
| 18:27.43 | Katty | i've yet to see a pissed off aussie. |
| 18:27.52 | jhiver | no, no |
| 18:27.56 | jhiver | he said "pissed" |
| 18:28.06 | robin_sz | Katty: in UK, pissed = drunk |
| 18:28.11 | Katty | oh. |
| 18:28.16 | iCEBrkr | Katty: Yeah, DUH |
| 18:28.16 | Katty | k |
| 18:28.22 | Katty | iCEBrkr: you betta don't |
| 18:28.25 | Katty | iCEBrkr: NEXT |
| 18:28.36 | iCEBrkr | Katty: Hey! I haven't picked on you in how long? |
| 18:28.44 | iCEBrkr | I got some catching up to do! |
| 18:28.45 | [TK]D-Fender | Like... SECONDS |
| 18:28.47 | robin_sz | sigh .. it looks all scred up |
| 18:28.56 | Katty | delicious coffee anyone? </isaac> |
| 18:29.07 | znoG | ast_expr2f.o(.text+0xe0f): In function `ast_expr': |
| 18:29.07 | znoG | /usr/local/src/asterisk-1.2.4/ast_expr2.fl:152: undefined reference to `ast_yyparse' |
| 18:29.09 | g4m | i seem to be having trouble getting DTMF tones from my cell phone while dialing in, i think its because the tones are too short or something, where can i tweak the DTMF settings? |
| 18:29.11 | znoG | any ideas what might cause that? |
| 18:29.17 | Katty | bkw__: you watch Isaac? |
| 18:29.27 | *** join/#asterisk trelane_ (n=trelane@asterisk.sosdg.org) |
| 18:29.34 | iCEBrkr | g4m: I blame the cellphone |
| 18:29.56 | chaos1 | hi folks |
| 18:30.08 | tzafrir_laptop | znoG, not sure. Version of bison? (is that related?) |
| 18:30.31 | znoG | tzafrir_laptop: maybe, using the latest (straight from the bison site) |
| 18:30.35 | chaos1 | is setting up of conference made impossible in 1.2 |
| 18:30.39 | _Sam-- | robin_sz : did you fix your gxp display with the new firmware? |
| 18:30.49 | znoG | tzafrir_laptop: maybe I need a later version of flex? |
| 18:30.50 | _Sam-- | er maybe they didnt fix that yet, ..actualyl i dont think they did |
| 18:30.51 | chaos1 | show application meetme ? |
| 18:30.55 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 18:30.59 | iCEBrkr | chaos1: eh? |
| 18:31.00 | robin_sz | _Sam--: no, I fixed the going blank ... |
| 18:31.10 | _Sam-- | you got the 1.0.2.8? |
| 18:31.12 | jaiger | g4m, how are you connected to the pstn? (eg. zaptel, ata, voip service) |
| 18:31.15 | robin_sz | _Sam--: now its upside down and scrambled |
| 18:31.15 | g4m | iCEBrkr: i get something like this in the CLI "WARNING[13706]: pbx.c:2394 __ast_pbx_run: Invalid extension '40', but no rule 'i' in context 'mainmenu'" |
| 18:31.22 | g4m | jaiger: voip service |
| 18:31.32 | iCEBrkr | g4m: Ok.. and? |
| 18:31.33 | robin_sz | _Sam--: and its still missing its MWI |
| 18:31.37 | g4m | jaiger: it happens when i use a standard telephone as well, if i type the numbers in too quickly |
| 18:31.47 | g4m | i get the previous error after typing in 4101 |
| 18:31.56 | iCEBrkr | g4m: That's just telling you there's no extension '40' and there's no 'i' (invalid) extension to land in.. |
| 18:32.07 | _Sam-- | robin: lucky you! :) |
| 18:32.13 | iCEBrkr | g4m: I still blame the cellphone. |
| 18:32.30 | g4m | iCEBrkr: but i just tried on my Cisco 7940 with the same results |
| 18:33.26 | jaiger | g4m, then I think your voip service isn't getting/hearing the DTMF properly from your cell. maybe voip service, cell service, cell phone etc |
| 18:33.33 | iCEBrkr | g4m: Now it sounds like a user error :P |
| 18:34.42 | jhiver | shido6, hi greg! |
| 18:34.56 | g4m | can i shorten what asterisk thinks is a DTMF tone |
| 18:35.17 | shido6 | ? |
| 18:35.28 | shido6 | hello |
| 18:35.49 | Cinen | Can anyone recommend a good billing solution. We have the cdrs but no easy way to rate them. I have tried several options but no one seems to confident/realiable. Any help would be greatly appreciated. |
| 18:35.55 | jhiver | how are things at nufone? |
| 18:36.04 | shido6 | stay tuned :) |
| 18:36.17 | jhiver | that don't mean jack :) |
| 18:36.33 | jhiver | do you mean, are you going to have an A-Z rate sheet again? :) |
| 18:36.41 | shido6 | maybe |
| 18:36.57 | shido6 | its bigger than that tho |
| 18:36.58 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 18:37.12 | websae | where is the CLI console located in asterisk? |
| 18:37.13 | jhiver | oh come on, you've said too much and not enough now :) |
| 18:37.21 | jhiver | websae, asterisk -r |
| 18:37.25 | websae | <PROTECTED> |
| 18:37.35 | _Sam-- | what does this mean (the ip is my phone, i am not using my phone, it is getting hints)... |
| 18:37.36 | _Sam-- | <PROTECTED> |
| 18:37.41 | jhiver | websae, depends where it's installed |
| 18:37.49 | jhiver | try locate asterisk |grep bin |
| 18:37.53 | shido6 | what is so fascinating about sending little sheets of paper through a machine and watching them appear somewhere else in the world |
| 18:38.22 | jaiger | shido6, you have a transporter? |
| 18:38.25 | jhiver | shido6, you've got a grudge against faxes? |
| 18:38.33 | chaos1 | Cinen:portaone rocks |
| 18:38.35 | jhiver | you're going to add T38 support to *? |
| 18:38.36 | mroth_imm | does anyone have experience with Native MOH |
| 18:38.39 | mroth_imm | ? |
| 18:38.55 | *** part/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:39.00 | chaos1 | i am really having a problem with setting up a conference room on asterisk 1.2 |
| 18:39.19 | jhiver | shido6, so? |
| 18:39.38 | chaos1 | it does not accept the meetme construct |
| 18:40.10 | chaos1 | it was bitch simple in * 1, just add the rooms in meetme.conf and call meetme in the extension.conf |
| 18:40.24 | trelane_ | I'm attempting to modprobe ztdummy running 2.6.15.1 vanilla and am getting an input/output error when running modprobe |
| 18:40.26 | websae | <PROTECTED> |
| 18:40.26 | websae | Feb 8 10:40:04 WARNING[4807]: chan_sip.c:4836 check_auth: Stale nonce received from '"Brandon" <sip:100@sip.websae.com>' |
| 18:40.40 | Nugget | You need a fresher Brandon. |
| 18:41.05 | chaos1 | mroth_imm:man, i am at it |
| 18:41.08 | *** join/#asterisk salviadud (n=ralfalfa@201.137.161.198) |
| 18:41.20 | Nugget | websae: http://lists.digium.com/pipermail/asterisk-users/2005-July/117209.html is what google found for me. What did you find when you googled for that error? |
| 18:41.35 | mroth_imm | chaos1: you have native moh experience? |
| 18:41.35 | websae | i use yahoo |
| 18:41.57 | chaos1 | mroth_imm:man! i been trying that for last two hours |
| 18:42.08 | chaos1 | mroth_imm:lame,sox ed mp3s into raw |
| 18:42.12 | Nugget | websae: technically you use google, too. You just expect us to do all the typing. :) |
| 18:42.28 | chaos1 | mroth_imm:set mode=files and directory where raw files were placed |
| 18:42.30 | mroth_imm | oh, maybe we could help each other than, just got mine setup yesterday and was wondering about scalability |
| 18:42.46 | chaos1 | called musiconhold in extensions.conf and tested it by making a call thru the sip channel |
| 18:42.55 | _Sam-- | [av]bani you around |
| 18:42.59 | *** join/#asterisk loick (n=loick@APuteaux-151-1-22-149.w82-124.abo.wanadoo.fr) |
| 18:43.04 | Katty | Nugget: :> |
| 18:43.06 | Katty | file: :> |
| 18:43.09 | chaos1 | mroth_imm:cli shows executing it but i cant listen it :) |
| 18:43.28 | chaos1 | i used to be such a happy man until i got bitten by 1.2 |
| 18:43.48 | mroth_imm | i got it running on abe and 1.21 yesterday... |
| 18:44.14 | mroth_imm | what happens if you say "moh show classes" or "moh show files" at the CLI |
| 18:44.57 | chaos1 | thanks one million shido6 |
| 18:45.11 | chaos1 | i just found out that you can't do conferencing without a zaptel interface |
| 18:45.13 | chaos1 | excellent |
| 18:45.21 | robin_sz | chaos1: you can |
| 18:45.24 | chaos1 | i have got only sip channels, so i am doomed |
| 18:45.33 | chaos1 | robin_sz:how? |
| 18:45.35 | _Sam-- | file: please fix your stuff |
| 18:45.38 | _Sam-- | we got no outoign calls |
| 18:45.47 | file | eh? |
| 18:45.51 | _Sam-- | no outgoing |
| 18:45.53 | _Sam-- | we send the calls to you |
| 18:45.57 | _Sam-- | but your side never gets the call out now |
| 18:45.57 | robin_sz | is file with voipgate? |
| 18:46.04 | _Sam-- | want me to paste? |
| 18:46.36 | file | grrrrrrrrr |
| 18:46.48 | _Sam-- | exactly! |
| 18:46.51 | file | someone broke it |
| 18:47.08 | _Sam-- | just let me know when so i can appease the angry sales force |
| 18:47.25 | mroth_imm | file: could you tell me the scaling points of native moh? |
| 18:48.14 | _Sam-- | i just cant win with remote gateways...and nobody say 'told ya so' either. |
| 18:48.21 | _Sam-- | live and learn |
| 18:48.33 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 18:48.38 | file | is it working now? |
| 18:48.51 | fugitivo | "someone" broke it |
| 18:48.54 | _Sam-- | nope |
| 18:49.02 | file | too many windows open... |
| 18:49.03 | fugitivo | $someone broke it |
| 18:49.21 | file | oh |
| 18:49.30 | file | I have an idea |
| 18:50.17 | file | yes, someone did break it |
| 18:50.22 | robin_sz | does it involve goats blodd and chicken bones? |
| 18:50.22 | file | so I'm grepping the logs to see who! |
| 18:50.30 | _Sam-- | seeing who doesnt FIX it though! |
| 18:50.49 | robin_sz | but blame attribution is important :) .. |
| 18:50.51 | file | it's being fixed right now |
| 18:50.59 | _Sam-- | thanks |
| 18:52.17 | _Sam-- | its fixed. |
| 18:52.47 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 18:52.55 | fugitivo | file: i bet that "someone" did a modification to the wrong window |
| 18:52.58 | [av]bani | _Sam--: \o/ |
| 18:53.26 | _Sam-- | [av]bani: you never tried BLF on your gxp yet? |
| 18:53.31 | _Sam-- | i think i found a bug of some kind |
| 18:53.42 | websae | anyone her familiar with e911? |
| 18:54.03 | kippi | how well does asterisk run on fedora core 3? |
| 18:54.18 | mut | just as well as any other linux distro |
| 18:54.19 | mroth_imm | very (YMMV) |
| 18:54.32 | [av]bani | _Sam--: i have it set up, havent used it much |
| 18:54.38 | [av]bani | it seems to work so far |
| 18:54.38 | _Sam-- | check your console |
| 18:54.44 | _Sam-- | i get these all over |
| 18:54.45 | _Sam-- | <PROTECTED> |
| 18:54.54 | R3DB0x | where is a good place to buy a Wildcard TDM400P for 3 analong lines? |
| 18:54.57 | _Sam-- | 201 = my phone...its not on any calls |
| 18:55.00 | _Sam-- | its just getting hints |
| 18:55.24 | _Sam-- | the blf on the phones work |
| 18:55.32 | _Sam-- | but i get a zillion of those messages |
| 18:55.35 | [av]bani | _Sam--: i dont see anything |
| 18:55.40 | Jizzbug | R3DB0x, OpenVox.com.cn (ppl 'round here will chastize you, but nothing is free [or cheap, in this case]) |
| 18:55.56 | [av]bani | _Sam--: my console is silent |
| 18:56.07 | _Sam-- | if you search this string on google "Incoming call: Got SIP response 415 "Unacceptable Content-Type" the only result is a gxp result |
| 18:56.33 | [av]bani | how do you have your hints setup? |
| 18:56.35 | R3DB0x | Jizzbug, im not looking really looking for free or even cheap im looking for a reputable place to buy from |
| 18:56.48 | _Sam-- | exten => 100,hint,SIP/100 |
| 18:56.53 | hypa7ia | R3DB0x: try store.digium.com then |
| 18:57.21 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 18:58.13 | bweschke | R3DB0x: how many FXO/FXS ports are you looking for on the TDM400P card? |
| 18:58.22 | websae | anyone here have e911 going? |
| 18:58.31 | docelm0 | yes.. |
| 18:58.40 | R3DB0x | i have 3 telelphone lines |
| 18:58.44 | Katty | no |
| 18:58.49 | bweschke | so you need 3 fxo ports? |
| 18:59.02 | R3DB0x | yes |
| 18:59.06 | [av]bani | _Sam--: and in sip.conf ? |
| 18:59.09 | websae | docelm0:what do you have going? |
| 18:59.16 | _Sam-- | i dont have anything in sip.conf that i know of |
| 18:59.17 | docelm0 | You mean who am I using? |
| 18:59.19 | docelm0 | GX |
| 18:59.21 | websae | yes |
| 18:59.25 | [av]bani | _Sam--: you need a subscribecontext |
| 18:59.37 | bweschke | can do $275 + shipping for a TDM400P with 3 fxo modules |
| 18:59.40 | _Sam-- | i tried it with that yesterday and it didt work |
| 18:59.42 | [TK]D-Fender | [av]bani : Had a chance to give all of your new phones a good test? |
| 18:59.43 | docelm0 | it comes with the DID when I purchase them |
| 18:59.47 | _Sam-- | i took out the subscribe context, and it worked |
| 18:59.57 | [av]bani | _Sam--: and it needs to be the same context as the context for your extensions |
| 18:59.58 | Jizzbug | hypa7ia: you a chick geek? or just a fan of the library of alexandria? |
| 19:00.03 | [av]bani | _Sam--: until i did that, i got errors |
| 19:00.07 | hypa7ia | Jizzbug: both |
| 19:00.08 | _Sam-- | i will give it a shot, THANKS> |
| 19:00.14 | _Sam-- | \o/! |
| 19:00.39 | Jizzbug | clever nick |
| 19:00.40 | _Sam-- | i like the blf and the one touch on hook dialing with those buttons |
| 19:00.59 | [av]bani | _Sam--: the hints need to be in the same context as your extensions, the subscribecontext needs to state that context also |
| 19:01.08 | [av]bani | _Sam--: its a bit weird but ... it worked for me |
| 19:01.12 | websae | who is GX |
| 19:01.22 | [av]bani | _Sam--: and subscribecontext needs to go in [general] |
| 19:02.00 | _Sam-- | thanks will sip reload do enough, or it needs a full reload? |
| 19:02.08 | [av]bani | sip reload is enough |
| 19:02.26 | websae | rdocelm0: what's their website? |
| 19:03.06 | _Sam-- | im not sure i think it may even need a restart |
| 19:04.08 | benjk | does anyone know if it is possible to stop a Cisco 7960 from trying to register with the sip accounts for line appearances 2, 3, 4 etc while still registering with the one for line appearance 1 ? |
| 19:04.23 | websae | docelm0*:what's their website, GX.? |
| 19:04.34 | docelm0 | um, GLOBAL CROSSING |
| 19:04.46 | astnoob_ | <PROTECTED> |
| 19:04.46 | astnoob_ | I am able to place and receive calls on this extension. |
| 19:05.08 | astnoob_ | <PROTECTED> |
| 19:05.41 | docelm0 | They have minimums if you want you can purchase them thru me once account has things settled.. |
| 19:05.48 | docelm0 | I have most of the details.. Just not all |
| 19:05.56 | bweschke | astnoob: your cisco is trying to register as "-" |
| 19:05.58 | bweschke | that's bad. :) |
| 19:06.12 | bweschke | what does the SIP00xxx.cnf look like again in pb? |
| 19:07.04 | astnoob_ | <PROTECTED> |
| 19:07.27 | astnoob_ | <PROTECTED> |
| 19:07.45 | astnoob_ | <PROTECTED> |
| 19:07.49 | *** join/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net) |
| 19:07.54 | diclophis | howdy all |
| 19:08.02 | diclophis | so.. i have a agi/record file problem |
| 19:08.02 | astnoob_ | <PROTECTED> |
| 19:08.14 | diclophis | it seems to be also recording the # keypress at the end of all recordings |
| 19:08.29 | bweschke | astnoob: yep - that's it - just make it "" |
| 19:08.37 | astnoob_ | <PROTECTED> |
| 19:08.37 | bweschke | otherwise the cisco will try to register with that |
| 19:08.40 | bweschke | and that's what it's doing |
| 19:08.41 | diclophis | is this default behaviuor.. or perhaps a echo cancelazation bug? |
| 19:08.50 | astnoob_ | <PROTECTED> |
| 19:09.05 | *** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
| 19:09.30 | benjk | I have ... |
| 19:09.30 | jsaunders | Is there an easy way to set an extension that will allow me to listen to moh? |
| 19:09.32 | benjk | # Proxy Registration (0-disable (default), 1-enable) |
| 19:09.33 | benjk | proxy_register: 1 |
| 19:09.44 | benjk | but that effects all accounts |
| 19:09.48 | *** join/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
| 19:11.18 | *** join/#asterisk hamish (n=hamish@196-28-87-71.wdsl.co.za) |
| 19:11.38 | _Sam-- | exten => 6000,1,Answer |
| 19:11.38 | _Sam-- | exten => 6000,2,MusicOnHold() |
| 19:11.48 | Overworked554 | hello, does anyone know of or recommend any outsourced live operator services that can take calls over sip/iax that can handle limited overflow call handling and night services? |
| 19:12.05 | bweschke | astnoob: if you want to have multiple call appearances - or register to other sip accts, sure |
| 19:13.30 | *** join/#asterisk saftsack (n=saftsack@p54A7F7B3.dip.t-dialin.net) |
| 19:13.50 | salviadud | sam, i would go with n |
| 19:13.54 | salviadud | instead of 2 |
| 19:14.06 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 19:14.07 | salviadud | that way, you can add a bunch of other things to do... |
| 19:14.18 | _Sam-- | he just wanted to listen to MOH |
| 19:14.22 | _Sam-- | which you can do fine with that |
| 19:14.28 | _Sam-- | but im not saying there may not be better ways |
| 19:14.31 | salviadud | well, yeah, i guess |
| 19:14.41 | jsaunders | Your suggestion salviadud? |
| 19:14.42 | diclophis | ... so does anyone know anything about recording files with asterisk? |
| 19:14.48 | jsaunders | Tnx _Sam-- btw. :) |
| 19:14.48 | benjk | proxy1_register: 1 |
| 19:14.49 | benjk | proxy2_register: 0 |
| 19:14.53 | benjk | doesnt work |
| 19:14.53 | _Sam-- | sure thing |
| 19:14.55 | salviadud | an n instead of a 2 |
| 19:15.02 | benjk | how braindead |
| 19:15.08 | jsaunders | 1,Answer and n,MusicOnHold()? |
| 19:15.20 | salviadud | yep |
| 19:15.23 | brif8 | I'm wanting to play with the idea of storing certain calls their information into my own MySql Database file. Does anyone know of good DB examples that I could read to follow? |
| 19:15.32 | jsaunders | Huh. Don't know the difference, but will try. |
| 19:15.34 | jsaunders | Tnx muchly. |
| 19:15.42 | benjk | is there anything one can do to prevent a Cisco phone from registering on all its accounts? |
| 19:15.43 | PupenoL | Hello. |
| 19:15.52 | astnoob_ | <PROTECTED> |
| 19:16.32 | jaiger | brif8, do you mean the CDR info? |
| 19:16.58 | brif8 | jaiger: yes number called, number of caller, time, duration etc... |
| 19:17.19 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 19:18.04 | chaos1 | Is it possible to run music on hold on a sip channel? |
| 19:18.12 | _Sam-- | [av]bani : thanks! \o/ |
| 19:18.18 | mroth_imm | chaos1: yes |
| 19:18.49 | mroth_imm | doing it now with some slight timing hiccups here and there...wanna try putting some zap hardware in to see if it fixes them (using ztdummy now) |
| 19:18.53 | astnoob_ | <PROTECTED> |
| 19:18.58 | chaos1 | is it possible to setup a production class conferncing server without zaptel hardware using only ztdummy |
| 19:19.00 | brif8 | jaiger: why ? |
| 19:19.04 | g4m | iCEBrkr, jaiger: thanks for the help, ended up being that Asterisk wasn't listening long enough, or something of that nature, either way some massaging of DTMF response times made it all work. |
| 19:19.16 | websae | I do not have e911 yet, and I provide VoIP to a select few customers --- one customer i know wants e911, what's the best way to implement it |
| 19:19.17 | websae | ? |
| 19:19.20 | iCEBrkr | g4m: cool |
| 19:19.32 | bweschke | astnoob: np |
| 19:19.43 | chaos1 | mroth_imm:could you tell me what codec to allow= for? |
| 19:19.43 | bweschke | chaos1: yes |
| 19:19.47 | chaos1 | ulaw,alaw |
| 19:19.59 | chaos1 | ast 1.2 |
| 19:20.05 | mroth_imm | all of our calls are u-Law <-> u-Law |
| 19:20.37 | mroth_imm | so we "disallow=all" |
| 19:20.50 | mroth_imm | "allow = ulaw" |
| 19:20.58 | bweschke | chaos1: the www.bargainconference.com site is running a pre-paid conferencing service off of an asterisk 1.2 base w/o zaptel hardware |
| 19:21.10 | chaos1 | mroth_imm:is it necessary to have an mp3 player on asterisk 1.2 to play the files? demo-instruct.gsm can be listened while it plays but musiconhold is not audible |
| 19:21.19 | *** join/#asterisk ramy (n=ramy@62.135.122.240) |
| 19:21.19 | lo_tech | diclophis: yar |
| 19:21.23 | chaos1 | bweschke:wow |
| 19:21.28 | jaiger | brif8, your question wasn't clear. look on voip-info.org for the CDR pages. they detail using SQL databases. I use postgres myself |
| 19:21.37 | mroth_imm | no, not with native moh, you should be able to hear the music without mpg123 |
| 19:21.46 | hamish | New to * and trying to piece it together. sip.conf has [jane] and username=jane. How do I associate jane with extension 100 in extensions.conf |
| 19:22.09 | iCEBrkr | WHY DO PEOPLE ASSOCIATED EXTENSIONS WITH PEOPLES NAMES!!!!!!! |
| 19:22.09 | hamish | would it be with extn ->100,1,SIP/jane ? |
| 19:22.10 | chaos1 | mroth_imm:i have learnt the same from the documentation that 1.2 has its own player. but it doesnt work |
| 19:22.13 | JCC_ | hello all, can anyone tell me what would put this error in the logs "chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 102 Cmd SIP/2.0" |
| 19:22.22 | _Sam-- | hamist: exten=> 100,1,DIAL(SIP/jane) |
| 19:22.25 | bweschke | hamish: in the context you've assigned jane to, exten => 100,1,Dial(SIP/jane) |
| 19:22.26 | iCEBrkr | hamish: make it [100] :P |
| 19:22.30 | chaos1 | mroth_imm:i have files=native, directory=/var/.../mohmp3 |
| 19:22.51 | diclophis | lo_tech... so sometimes the '#' dtmf gets recorded along with the the voice |
| 19:22.51 | _Sam-- | there are other options you can add as well, but that will make jane forever until she answer it or you hang up :) |
| 19:22.56 | diclophis | is it an echo problem |
| 19:23.08 | [av]bani | _Sam--: works? \o/ |
| 19:23.11 | chaos1 | hamish: add /100 at end of register in sip.conf |
| 19:23.14 | diclophis | or is there a setting or something to not record the escape digit at the end of a rrecording |
| 19:23.14 | Netgeeks | iCEBrkr: um, so you have problems with people thinking that extensions are tied to human beings and not instances on a phone? |
| 19:23.18 | mroth_imm | our music on hold came from cd ---CDEX---> Microsoft wavs ---sox "infile.wav" -t ul -r 8000 -b -c 1 "outfile.pcm"---> uLaw pcm |
| 19:23.20 | _Sam-- | [av]bani: yep, like a million bucks. THANKS. |
| 19:23.23 | ramy | anybody suggests a starting point for someone who knows nothing about asterisk or SIP and want to gt started ? |
| 19:23.27 | iCEBrkr | Netgeeks: Make it one to one man.. It's easier |
| 19:23.39 | mroth_imm | musiconhold.conf looks like: |
| 19:23.41 | iCEBrkr | Netgeeks: Besides, if you recycle phones, there's no need to reconfigure the phone |
| 19:23.42 | [av]bani | _Sam--: well, payback for un-bricking my gxp the other day :D |
| 19:23.48 | mroth_imm | [classes] |
| 19:23.52 | JCC_ | try asterisk@home |
| 19:23.57 | mroth_imm | [moh_files] |
| 19:23.58 | [av]bani | _Sam--: http://www.voip-info.org/wiki/view/GXP-2000+Disaster+Recovery |
| 19:23.58 | brif8 | jaiger: thanks I'll look further so I had drawn a blank so I'm looking more |
| 19:24.00 | hamish | ok, sorry iCEBrkr. New here. So I recreate [jane] to [100]. What about the username then if I don't associate it with jane? |
| 19:24.06 | *** join/#asterisk WasPhantom (n=neil@198-48-0-224-dhcp.cafenet.co.nz) |
| 19:24.08 | mroth_imm | default => /var/lib/asterisk/moh-native/latin/afternoon_in_rio,r |
| 19:24.14 | Netgeeks | I agree with the users, a person should have a single contact id... dial it and you always find that person... |
| 19:24.22 | iCEBrkr | hamish: I personally think it's easier to associate everyone one to one to one.. |
| 19:24.43 | iCEBrkr | Extension 100's sip.conf is [100], the sip registry on the phone is 100 as well. |
| 19:24.46 | _Sam-- | wow, thanks for the props! |
| 19:24.52 | benjk | any Cisco 79XX guru here? |
| 19:24.59 | chaos1 | mroth_imm:i applied lame on asterisk mp3s to convert to wave, applied sox on wave to get raw files just as you did |
| 19:25.03 | chaos1 | but it wont get played |
| 19:25.06 | lo_tech | diclophis: have a pastebin of extensions.conf? |
| 19:25.18 | chaos1 | mroth_imm:unaudible,though cli shows it playing it |
| 19:25.27 | [av]bani | _Sam--: did you figure out recovery yourself or did grandstream give you hints? |
| 19:25.28 | _Sam-- | i couldnt get it to work when i connected directly via ethernet the cable to the computer. |
| 19:25.31 | _Sam-- | i had to get it over a hub |
| 19:25.31 | iCEBrkr | hamish: and like whoever it was said.. in your [main] put exten => 100,1,Dial(SIP/100) |
| 19:25.33 | Netgeeks | iCE: the app I wrote ^H^H^H^H^H am writing associates extensions to humans and not to phones... |
| 19:25.35 | diclophis | lo_tech it is from AGI |
| 19:25.36 | diclophis | ... |
| 19:25.38 | mroth_imm | that is strange... |
| 19:25.45 | _Sam-- | [av]bani: that was courtersy of bill @ gs |
| 19:25.54 | diclophis | the "RECORD FILE" command to be exact |
| 19:25.55 | [av]bani | _Sam--: i have no idea who he is :) |
| 19:25.57 | jsaunders | Hrmm: MusicOnHold() channel is dead quiet. :( |
| 19:26.00 | mroth_imm | if you do "moh classes show" and "moh files show" does the output look correct |
| 19:26.05 | diclophis | which iirc is the only way to record files using agi |
| 19:26.06 | hamish | iCEBrkr. What you mean? |
| 19:26.13 | hamish | One to one to one? |
| 19:26.22 | iCEBrkr | Netgeeks: dial plans can be made simple with _XXXX,1,Dial(SIP/${EXTEN}) as well.. |
| 19:26.24 | [TK]D-Fender | [av]bani : How are the new phone model tests going? |
| 19:26.27 | diclophis | other than perhaps a Zap command, executing through AGI with EXEC |
| 19:26.28 | lo_tech | diclophis: AGI is still exec-ing a command to record, I need to know which one you're using |
| 19:26.42 | diclophis | ... uh as far as i can tell "RECORD FILE" |
| 19:26.42 | iCEBrkr | hamish: Keep everything lines up for debuging and order. |
| 19:26.46 | jsaunders | "moh files show" is empty. |
| 19:26.47 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 19:26.51 | Netgeeks | or even _XXXX,1,Dial(${GEORGE}) |
| 19:26.53 | diclophis | the line is using the Zaptel interface |
| 19:26.57 | [av]bani | [TK]D-Fender: new model? |
| 19:26.58 | diclophis | if that explains anything |
| 19:27.02 | *** join/#asterisk EvlHimeko (i=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 19:27.03 | iCEBrkr | Netgeeks: That's gay. Seriously. |
| 19:27.08 | Netgeeks | hahaha |
| 19:27.09 | _Sam-- | somethig you could put on that disaster recovery page bani...is the IP the phone is looking for |
| 19:27.10 | iCEBrkr | Netgeeks: If you want some monkey to maintain the system.. |
| 19:27.11 | _Sam-- | its always the same |
| 19:27.13 | mroth_imm | what about after a "moh reload" |
| 19:27.25 | iCEBrkr | Netgeeks: I mean, if you're application does all that behind the scenes? SURE! Rock'on! |
| 19:27.26 | _Sam-- | so they could just setup the pc to that IP, load tfptd, plug in the cable and go |
| 19:27.32 | _Sam-- | they wouldnt even need ethereal |
| 19:27.38 | _Sam-- | if it works, it works |
| 19:27.45 | Netgeeks | I like monkeys maintaining systems, keeps me from having to deal with lusers.... |
| 19:27.50 | iCEBrkr | Netgeeks: All I'm saying is maintain [JOEB] in a 500 user system. |
| 19:27.53 | jsaunders | moh reload> 1 class reloaded |
| 19:27.54 | diclophis | http://pastebin.com/545398 is my extensions.conf |
| 19:28.16 | iCEBrkr | Oh, shit there's already a [JOEB] so I'll make it [JOEB2] |
| 19:28.18 | _Sam-- | the only thing the ethereal/tcpdump tells you is if your phone is sending out the requests or not |
| 19:28.33 | Netgeeks | *nod* I was being simplistic |
| 19:28.36 | _Sam-- | you dont really need to know that...if it gets the tftp files, you know its sending requets :) |
| 19:28.45 | diclophis | http://pastebin.com/545400 is the ragi code that i call from my script |
| 19:28.57 | *** part/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
| 19:28.58 | jsaunders | Does mpg123 need access to a sound card to function properly? |
| 19:29.04 | [TK]D-Fender | [av]bani : You were testing out 7940G, SNOM260, etc now, weren't you? |
| 19:29.08 | diclophis | i call it with beep=true and silencedetect=2 |
| 19:29.14 | [TK]D-Fender | [av]bani : And a Polycom or two? |
| 19:29.33 | _Sam-- | bani is too busy updating feature requests for all those phones to reply. |
| 19:29.35 | [av]bani | [TK]D-Fender: no |
| 19:29.47 | [av]bani | [TK]D-Fender: put down the bong, you're confusing me with someone else :D |
| 19:29.49 | [TK]D-Fender | [av]bani : Must be mixing people up.... |
| 19:29.52 | mroth_imm | jsaunders: what is the output of "moh classes show" (does it look correct)? |
| 19:30.09 | _Sam-- | my favorite 4 letter word |
| 19:30.10 | jsaunders | Class: default |
| 19:30.10 | jsaunders | <PROTECTED> |
| 19:30.10 | jsaunders | <PROTECTED> |
| 19:30.10 | jsaunders | <PROTECTED> |
| 19:30.11 | [TK]D-Fender | [av]bani : Actually I'm not on drugs.... maybe I should start X.X |
| 19:30.35 | [TK]D-Fender | [av]bani : Have you tried a Snom 360 by chance? |
| 19:30.53 | _Sam-- | he's got a 360 |
| 19:31.03 | mroth_imm | jsaunders: you're not using native moh, right? |
| 19:31.11 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 19:31.39 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:31.40 | *** join/#asterisk jjhall (n=chatzill@94-253.69-92-cpe.cableone.net) |
| 19:31.58 | Tall-guy | ...just setup an analog phone with a sipura spa-1001...took 4 minutes, start to finish..HONEST! (for anyone who cares) |
| 19:31.58 | _Sam-- | [av]bani: for the record, when you add subscribecontext to sip.conf, it necessitates a full restart. |
| 19:32.19 | jsaunders | mroth_imm: load=res_musiconhold.so |
| 19:32.29 | mroth_imm | if not, make sure you're mp3s are in "/var/lib/asterisk/mohmp3" and make sure you are specifying quietmp3 as the class |
| 19:32.56 | mroth_imm | i don't think the moh command would've worked from the cli without the moh resource |
| 19:33.08 | mroth_imm | s/command/commands |
| 19:33.46 | jsaunders | mroth_imm: mp3's are in directory, queitmp3 selected. Could it be the fact that linux has no drivers setup up for sound card? |
| 19:33.48 | benjk | Those Cisco phones are really silly ... |
| 19:33.49 | benjk | Registration from 'sip:20032@0.0.0.0' failed for '192.168.0.233' |
| 19:34.17 | WasPhantom | so has anyone here used a Polycom IP500? |
| 19:34.19 | benjk | I set the proxy to 0.0.0.0 so it wont bother the Asterisk box with registrations, but it comes in anyway |
| 19:34.27 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 19:34.29 | mroth_imm | i guess that is possible...that's outside my understanding though |
| 19:35.02 | jsaunders | mroth_imm: I'm wondering if mpg123 requires sound card access. It says it's running when 'ps ax'. |
| 19:35.04 | Defraz | What would be the best way to implement if my ATA looses connectivity to my Astrisk box, the asterisk box would forward the call to my cell phone. |
| 19:35.11 | [av]bani | _Sam--: weird |
| 19:35.16 | mroth_imm | that was going to be my next question... |
| 19:35.26 | [av]bani | [TK]D-Fender: i _have_ a snom 360, remember i was comparing the poly601 to it |
| 19:35.36 | mroth_imm | you could try native moh or the rawplayer method to find out |
| 19:35.55 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 19:36.03 | jsaunders | mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 <files...> |
| 19:36.07 | Defraz | I think I might be able to do a call forward on unavailable. |
| 19:36.09 | jsaunders | is what ps ax shows |
| 19:36.17 | Defraz | or is unavailable mean I don't answer. |
| 19:36.23 | *** part/#asterisk gambolputty (n=root@64.74.225.131) |
| 19:36.41 | Tall-guy | defraz: a simple diaplan should do the trick. |
| 19:36.48 | Tall-guy | diaLplan |
| 19:37.05 | Defraz | :) What is simple for you isn''t so simple for me. |
| 19:37.11 | Defraz | But I can look that direction. |
| 19:37.15 | mroth_imm | maybe if you looked for some info on mpg123 you could find out if it was dependant on a sound card |
| 19:37.37 | mroth_imm | two workarounds: native moh, rawplayer |
| 19:38.00 | Tall-guy | Defraz: give yourself some time to understand the dialplan and its variables, and you're away...don't try and run before you walk....make a dialplan, screw with it, break it, and then go wild :) |
| 19:38.03 | jsaunders | mroth_imm: What is native moh? |
| 19:38.21 | Tall-guy | jsaunders: aboriginal chanting on hold :) |
| 19:38.48 | Defraz | sounds good |
| 19:39.01 | mroth_imm | you convert the mp3s to the native codec of your calls, then asterisk plays them directly without an outside process |
| 19:39.06 | jsaunders | Oh boy. Heheh. |
| 19:39.20 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 19:39.36 | mroth_imm | well, the conversion is tricky, but it doesn't have to be perfect to tell if mpg123 is your problem |
| 19:39.39 | jsaunders | morth_imm: Huh. Ain't that somethin'. |
| 19:39.42 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 19:39.57 | ctooley | Is there a way to overwrite the cdr dst field so that a call file's destination is correct? |
| 19:40.00 | mroth_imm | for that purpose, rawplayer would be easier |
| 19:40.08 | jsaunders | I'm gonna upgrade mpg123 and see what happens. I did a 'locate mpg123' and found just the bin, not even sure where it came from. |
| 19:40.16 | jsaunders | Found mpg321 too. |
| 19:40.24 | jsaunders | Dunno if that can be used or not. |
| 19:40.30 | [TK]D-Fender | [av]bani : Presence on all the buttons working good for you? I'm due to decide what to do with the IP601 I got my receptionist and would like a testemonial on my alternative. |
| 19:40.50 | mroth_imm | i *think* that has it's own issues |
| 19:40.56 | mroth_imm | give this a shot: http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it |
| 19:40.56 | [av]bani | [TK]D-Fender: i havent tried it yet, nor have i tried it on the ip601 (and i dont really look forward to mangling the 601 to do it) |
| 19:41.09 | [TK]D-Fender | [av]bani : Its that the 601 has issues right now which should be solved eventually, but this decision could save me some money as well. |
| 19:41.09 | [av]bani | [TK]D-Fender: it "just works" on the gxp, as does everything else on it |
| 19:41.13 | mroth_imm | if you still don't hear anything, hit the users list...someone will know : ) |
| 19:41.15 | jsaunders | Oh, it is mpg321 that's being used. Huh. It's a symbolic link. |
| 19:41.18 | [TK]D-Fender | :/ |
| 19:41.24 | [av]bani | [TK]D-Fender: the gxp is about as plug and play with * as you get.. everything "just works" |
| 19:41.28 | *** join/#asterisk prog (n=prog@ip-85-160-89-226.eurotel.cz) |
| 19:41.28 | mroth_imm | that right there could be a problem |
| 19:41.29 | [av]bani | [TK]D-Fender: no mangling required |
| 19:41.38 | jsaunders | mroth_imm Indeed. |
| 19:41.44 | mroth_imm | it may not accept the same switches...try installing mpg123 and give it another go : ) |
| 19:41.56 | jsaunders | Will do. |
| 19:42.04 | hamish | From my reading sip.conf is for registering the phone to * for making calls from the phone, extensions.conf is for registering extensions to receive calls. |
| 19:42.08 | mroth_imm | gotta earn my paycheck now... |
| 19:42.12 | lo_tech | diclophis: what version of Asterisk::AGI are you using? checking my version, I dont have silencesuppression as a passed value... so I don't think we're using the same version |
| 19:42.41 | [TK]D-Fender | [av]bani : Ok, guess I'll have to wait on someone elses case-study then.... thanks anyways.... |
| 19:42.53 | [TK]D-Fender | [av]bani : How do you find the 601 in comparison for what you did test? |
| 19:42.59 | *** part/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net) |
| 19:43.32 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:44.15 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 19:45.56 | *** join/#asterisk azc-uk (n=vircuser@84-45-206-139.no-dns-yet.enta.net) |
| 19:46.25 | jsaunders | mroth_imm: Sweet! :) |
| 19:46.31 | azc-uk | hi all, is this an appropriate place to ask about AMP, in particular FOP please? |
| 19:46.44 | jsaunders | mroth_imm: Now how do I crank the volume on this sucker. |
| 19:46.55 | mroth_imm | jsaunders: nice...glad i could help : ) |
| 19:47.04 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 19:47.13 | iCEBrkr | azc-uk: I don't think there's an appropriate place for AMP :D |
| 19:47.22 | azc-uk | heh ok |
| 19:47.24 | iCEBrkr | azc-uk: But someone might be able to answer your questions |
| 19:47.33 | [TK]D-Fender | ~amp |
| 19:47.41 | jbot | it has been said that amp is NOT supported here! people using it should join #amportal |
| 19:47.47 | *** join/#asterisk ToTo (n=ToTo@host5-158.pool874.interbusiness.it) |
| 19:47.52 | iCEBrkr | [TK]D-Fender: LOL |
| 19:48.09 | [av]bani | [TK]D-Fender: well, what do you want to know specifically? |
| 19:48.10 | azc-uk | well, i set up a@h, all is working dandy, changed my passwords for maint etc, but when i fire up FOP, all the icons are just flashing red/green |
| 19:48.21 | azc-uk | does that mean it is not connected properly? |
| 19:48.30 | iCEBrkr | [TK]D-Fender: We need one for a@h |
| 19:48.41 | _Sam-- | hey file you around? i need the CIDName lookerupper |
| 19:48.41 | iCEBrkr | ~a@h |
| 19:48.49 | _Sam-- | i cant find the url |
| 19:48.52 | iCEBrkr | ~astrisk@home |
| 19:48.54 | *** join/#asterisk FlyboySR22 (i=rsears@gateway.adnc.com) |
| 19:48.55 | file | ok |
| 19:49.06 | FlyboySR22 | zeeze |
| 19:49.16 | [TK]D-Fender | [av]bani : Well for your comparison basically which you liked better in general and any highlight. For the Snom 360 Its all about presence for me right now. |
| 19:49.27 | [TK]D-Fender | iCEBrkr : Need what for A@H? |
| 19:50.25 | iCEBrkr | [TK]D-Fender: Not supported message and where to go :P |
| 19:50.56 | [TK]D-Fender | iCEBrkr : A@H = AMP AFAIAC. |
| 19:51.18 | iCEBrkr | [TK]D-Fender: Hrr, I guess. |
| 19:51.25 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:51.27 | iCEBrkr | But you can install AMP without A@H |
| 19:52.01 | [av]bani | [TK]D-Fender: 601 phone ui is better, 360 has backlight, 360 has non braindamaged web ui, 601 has better audio, snom has non braindamaged firmware policy, you can reasonably configure snom wihtout having to mangle xml |
| 19:52.20 | *** join/#asterisk jsuther (n=jay@coal.acdmis.com) |
| 19:52.23 | tainted_ | what does this mean: Nat with Independend Mapping and Dependent Filter |
| 19:52.27 | [av]bani | [TK]D-Fender: 360 has more buttons |
| 19:53.52 | [TK]D-Fender | [av]bani : Ok, that confirms everything I've suspected and heard so far. Since the XML doesn't really peave me I'm better off with my Polycom's with the sole exception of my recptionist (who needs the buttons) |
| 19:54.13 | [TK]D-Fender | [av]bani : You concur on audio quality? |
| 19:55.01 | [TK]D-Fender | iCEBrkr : Well A@H may as well be just CentOS + basic * + AMP. Since you can build on the base I don't consider its packaging to be the problem, only *1* package on it :D |
| 19:55.28 | iCEBrkr | hah |
| 19:55.31 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:57.03 | [TK]D-Fender | iCEBrkr : I think I've been fair.... |
| 19:57.20 | *** join/#asterisk santiago (n=santiago@63.245.86.215) |
| 19:57.39 | [av]bani | [TK]D-Fender: well, i'd say polycom handset is 90% that of polycom, but the polycom's speakerphone is much better |
| 19:57.45 | [av]bani | [TK]D-Fender: er snom handset |
| 20:00.59 | *** join/#asterisk CrazyYoss (n=CrazyYos@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 20:02.02 | CrazyYoss | This is a shot in the dark but has anyone in here worked with MMS gateways and cell phone carriers? |
| 20:02.24 | *** join/#asterisk nmsclera (n=arthurh@invinine.com) |
| 20:03.17 | nmsclera | When checking sip qualify peers from the CLI using SIP SHOW PEERS, what exactly is the last column "Status" measuring? |
| 20:03.39 | websae | E911 service...anyone familiar with it or know of inexpensive wholesale providers? |
| 20:04.02 | nmsclera | and why on local ethernet is it outrageously high? (Say, greater than 120ms) |
| 20:04.29 | *** part/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
| 20:05.48 | Defraz | Dash-911 |
| 20:05.53 | Defraz | they seem reasonable. |
| 20:08.11 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 20:15.18 | [TK]D-Fender | [av]bani : Makes it tough to switch..... |
| 20:15.22 | [TK]D-Fender | dangit |
| 20:16.10 | benjk | any Cisco 79XX guru here? |
| 20:16.26 | [av]bani | [TK]D-Fender: ? |
| 20:17.09 | [av]bani | nmsclera: because your device takes a long time to respond to the qualify queries. it's not a ping |
| 20:17.39 | [av]bani | nmsclera: typically, the cpu in phones is not uber. and they're not designed to respond to those queries at high priority |
| 20:17.40 | _Sam-- | bani i really do think there is a problem with the BLF/hints |
| 20:17.49 | [av]bani | _Sam--: o_O |
| 20:17.55 | _Sam-- | i still get those errors when i am on the phone, and new calls come in |
| 20:18.19 | [av]bani | _Sam--: hm, never tested that |
| 20:18.25 | [av]bani | i'll test |
| 20:18.42 | sevard | Can someone please point me to a document regarding how Asterisk talks with Perl + how it would use perl as a medium to communicate with MySQL |
| 20:18.54 | *** join/#asterisk apardo (n=apardo@87.218.45.8) |
| 20:18.59 | [TK]D-Fender | [av]bani : To hear the quality diff + interface will count against me. Makes me think I'm better to pay the extra right now to keep my 601 +2 Mods and jsut wait till it gets worked out... |
| 20:19.43 | [av]bani | _Sam--: do you get it if you call yourself? |
| 20:19.50 | _Sam-- | let me see |
| 20:19.59 | _Sam-- | i dont have a hint on my extension |
| 20:20.02 | _Sam-- | so i would assume not |
| 20:20.02 | [av]bani | _Sam--: eg, call an extension, put it on hold. then call yourself |
| 20:20.12 | [av]bani | i didnt get any error |
| 20:21.02 | _Sam-- | its not about calling myself, i am not well versed at this so i cant explain it right...the blf buttons are set to check extensions here.... |
| 20:21.09 | *** part/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 20:21.13 | _Sam-- | if i am on the phone, and those extensions ring....then i get that error |
| 20:21.18 | _Sam-- | or message, not error |
| 20:21.24 | [av]bani | _Sam--: oh, if someone elses extension rings? |
| 20:21.30 | _Sam-- | now i am getting them all the time |
| 20:21.34 | _Sam-- | yes when someone elses extension rings |
| 20:21.36 | [av]bani | k |
| 20:21.46 | [av]bani | let me do a test |
| 20:22.22 | _Sam-- | and after i get 1 error...then they ALL start happening if i am or am not on the phone |
| 20:22.25 | _Sam-- | it was fine for about 30 minutes |
| 20:22.33 | _Sam-- | no every call that comes in, i get those errors |
| 20:22.39 | [av]bani | i cant do it from here... i call an extension, put it on hold, then ring another extension |
| 20:22.46 | [av]bani | the blfs go blinky blinky but no error on console |
| 20:23.08 | _Sam-- | they go blinky blnky...i use ring all...so ALL the blfs blink at once |
| 20:23.09 | [av]bani | i have myself on a blf on my own phone and it lights up correctly |
| 20:23.24 | _Sam-- | and i get about 7 errors at once |
| 20:23.24 | [av]bani | hmm, might be that the phone is getting too many infos at once |
| 20:23.48 | [av]bani | i'm still setting up our pbx soi havent tested that yet |
| 20:24.06 | _Sam-- | it seems relatively harmless really...the blfs work properly, and so does my asterisk |
| 20:24.13 | _Sam-- | at least, as far as i can tell |
| 20:24.18 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 20:24.25 | [av]bani | _Sam--: file a mantis ticket |
| 20:25.13 | franck | I have two sip phones registered with same extension and password, when the extension gets called only one phone is called |
| 20:25.19 | *** join/#asterisk De_Mon (n=de_mon@fl-71-0-154-141.dyn.sprint-hsd.net) |
| 20:25.25 | [av]bani | franck: you cant do that |
| 20:25.53 | franck | [av]bani: ok, how do I select which phones get called? |
| 20:25.53 | _Sam-- | you need to DIAL(SIP/1&SIP/2) |
| 20:26.02 | [av]bani | _Sam--: he said theyre the same extension... |
| 20:26.13 | [av]bani | franck: you give the phones different extensions |
| 20:26.19 | _Sam-- | but it sounds like he just wants both phone to ring when someone calls |
| 20:26.28 | _Sam-- | just make it dial both phones |
| 20:26.32 | [av]bani | _Sam--: both phones are SIP/1 according to him |
| 20:26.35 | _Sam-- | heh |
| 20:26.47 | _Sam-- | i see...isee |
| 20:27.03 | franck | [av]bani: ok let me put it differently then. I have a phone at work but when I go on travel I would like to use my softphone. How do I deregister the phone at work? |
| 20:27.15 | [av]bani | franck: unplug it |
| 20:27.30 | _Sam-- | or make taht extension in extensions.conf ring both phones |
| 20:27.36 | _Sam-- | and register them on different sip accounts |
| 20:27.45 | [av]bani | franck: the best way would be to give your softphone a different extension, then ring them both always |
| 20:27.56 | [av]bani | franck: eg 4000 for work, 5000 for softphone |
| 20:28.08 | [av]bani | Dial(SIP/4000&SIP/5000) always |
| 20:28.32 | franck | [av]bani: interesting... |
| 20:28.33 | [av]bani | just use qualify=yes and * will track the status of the extensions |
| 20:28.55 | Skid | argh /me gives up |
| 20:30.17 | Skid | coffee needed |
| 20:30.17 | franck | ok |
| 20:30.17 | franck | I will play around a little... The problem with unplug the phone, is that usually you leave the office and forgets to unplug it |
| 20:30.17 | [av]bani | franck: you could also just make agents log into a queue |
| 20:30.17 | [av]bani | franck: but that means you'd have to "login" on your office phone |
| 20:30.32 | franck | [av]bani: I have 80 people, can't do that for everybody |
| 20:30.37 | [av]bani | franck: you dont have to, just the people you want to roam |
| 20:31.05 | [av]bani | but it means the roaming people would have to login always |
| 20:31.16 | [av]bani | sorta inconvenient, but meh |
| 20:31.33 | [av]bani | franck: you could always make a pseudo-extension or ivr which disables extensions |
| 20:31.37 | franck | I use amp, I think there is some login stuff in it... |
| 20:31.44 | [av]bani | dial *4000 to turn off an extension |
| 20:31.47 | [av]bani | oh, amp.. heh |
| 20:31.58 | [av]bani | yeah, there's agent and queues in amp, you can set it up |
| 20:32.18 | franck | I have set up a queue for the switchboard |
| 20:32.20 | [av]bani | franck: ahh... you can just set DND on your phone |
| 20:32.23 | _Sam-- | frank: the easy way is to just add a seperate SIP account for the 2nd phone |
| 20:32.31 | _Sam-- | and just ring them both with DIAL(SIP/1&SIP/2) |
| 20:32.33 | franck | but I saw also you can log in to any phone... |
| 20:32.34 | fugitivo | ~amp |
| 20:32.36 | jbot | i heard amp is NOT supported here! people using it should join #amportal |
| 20:32.41 | [av]bani | franck: then just punch DND when you leave |
| 20:32.58 | *** join/#asterisk WasPhantom (n=neil@198-48-0-224-dhcp.cafenet.co.nz) |
| 20:33.04 | [av]bani | thats what DND function is for :) |
| 20:33.13 | franck | [av]bani: cool |
| 20:33.18 | franck | I will try all that... |
| 20:33.22 | [av]bani | what phone you using? |
| 20:33.27 | [av]bani | hardphone |
| 20:33.37 | franck | my hardphone is a www.planet.com.tw |
| 20:33.48 | tainted_ | hardphone |
| 20:33.50 | franck | VIP153PT |
| 20:33.52 | tainted_ | there must be a better term |
| 20:34.17 | [av]bani | tainted_: you want to change hardware/software too then? |
| 20:34.39 | tainted_ | no one says hardphone |
| 20:34.44 | [av]bani | sure they do |
| 20:34.50 | tainted_ | i will smack them |
| 20:34.55 | franck | [av]bani: thanks for the info... |
| 20:34.59 | *** part/#asterisk santiago (n=santiago@63.245.86.215) |
| 20:35.05 | _Sam-- | tainted, what do you call them? |
| 20:35.10 | tainted_ | that's like people who say 'gooey' when they mean GUI |
| 20:35.39 | tainted_ | why not just phone |
| 20:35.41 | [av]bani | franck: you just have to remember to hit DND on your VIP153PT when you leave the office. but it shouldnt ring at all |
| 20:35.48 | _Sam-- | i think an IP phone would probably be fine...and a softphone is a softphone |
| 20:36.05 | [av]bani | franck: so if you have SIP/4000&SIP/5000 in your dial, and sip/4000 is DND then it should still ring SIP/5000 |
| 20:36.10 | tainted_ | softmodem |
| 20:36.15 | [av]bani | which would be your softphone |
| 20:36.17 | tainted_ | softvideo |
| 20:36.25 | [av]bani | tainted_: hard disk |
| 20:36.38 | [av]bani | hardcopy |
| 20:36.48 | _Sam-- | hardwired |
| 20:36.52 | [av]bani | i'm sorry, but you are overruled |
| 20:36.53 | justinu | hardcore |
| 20:36.59 | [av]bani | YUO LOES! |
| 20:37.06 | [TK]D-Fender | hardly <_ |
| 20:37.07 | tainted_ | next time u say hardphone in real life - question yourself |
| 20:37.13 | [av]bani | _Sam--: not all softphones speak IP ... |
| 20:37.19 | [av]bani | tainted_: make me |
| 20:37.24 | tainted_ | i think IP phones make more sense |
| 20:37.31 | [av]bani | tainted_: not all softphones speak IP |
| 20:37.34 | justinu | i call them IP phones, myself |
| 20:37.35 | tainted_ | IP vs telephones |
| 20:37.44 | tainted_ | yea |
| 20:37.46 | _Sam-- | IP phones are not softphones, they are hardphones! |
| 20:37.49 | tainted_ | no one says hardphones |
| 20:37.57 | shido6 | Analog Phones |
| 20:37.58 | [av]bani | and... my snom 360 speaks ip. is it then an IP phone? |
| 20:38.01 | [TK]D-Fender | Lets call the whole thing off! |
| 20:38.06 | tainted_ | shido6 ther eu go |
| 20:38.15 | _Sam-- | yeah...your 360 is an IP phone...but xlite is a softphone |
| 20:38.16 | shido6 | or |
| 20:38.21 | shido6 | tha Murda Weapon |
| 20:38.50 | shido6 | its got a cord to trap knife attacks |
| 20:38.53 | [av]bani | hardphones.. hardware... softphones... software |
| 20:38.58 | [av]bani | makes perfect sense to me |
| 20:39.09 | fugitivo | soft = software phone = telephone |
| 20:39.11 | _Sam-- | it does, i dont have any problem with it myself |
| 20:39.26 | shido6 | a nice grip and most phones are constructed perfectly for blunt force trauma |
| 20:40.22 | [av]bani | shido6: i find VGA monitors work better |
| 20:40.58 | _Sam-- | what are you doing with VGA monitors |
| 20:41.26 | justinu | blunt force trauma? |
| 20:41.43 | justinu | baseball bat or billy club is pretty much optimized for that |
| 20:42.08 | _Sam-- | justinu: do the people who are testing the GXPs want a baseball bat or billy club yet? |
| 20:42.17 | justinu | yes |
| 20:42.30 | _Sam-- | figured |
| 20:42.51 | justinu | i told them that I'll try the new firmware, but if they continue to complain they'll just have to buy polycoms |
| 20:43.21 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:43.23 | _Sam-- | the threat of higher priced hardware usually makes them more accepting |
| 20:43.29 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 20:43.57 | [av]bani | justinu: not happy with gxp? |
| 20:44.09 | justinu | customer isn't |
| 20:44.15 | justinu | they complain about echo |
| 20:44.16 | _Sam-- | they are running 1.0.1.9 |
| 20:44.18 | [av]bani | ha |
| 20:44.19 | _Sam-- | what do you think |
| 20:44.19 | tainted_ | fugitivo good point |
| 20:44.25 | [av]bani | 1.0.1.9, lollerskates |
| 20:44.29 | [av]bani | of cours theres echo on that |
| 20:44.30 | tainted_ | soft = softWARE not soft to the touch |
| 20:44.39 | [av]bani | its only about a year old firmware |
| 20:44.42 | justinu | i don't understand why there's echo, but I'll upgrade them |
| 20:44.45 | iCEBrkr | tainted_: LOL |
| 20:45.05 | tainted_ | [av]bani you must change your 'hardphone' ways |
| 20:45.05 | Qwell[] | asterisk is silky smooth...very soft |
| 20:45.06 | [av]bani | justinu: old firmware didnt echo cancel the speakerphone |
| 20:45.09 | [av]bani | tainted_: make me |
| 20:45.21 | _Sam-- | lol |
| 20:45.35 | tainted_ | ain't my clients laughing at me |
| 20:45.36 | [av]bani | justinu: anything 1.0.1.12 or better shouldnt have any echo issues |
| 20:45.58 | justinu | bani: they're not even using the speakerphone |
| 20:46.05 | *** join/#asterisk Libila (n=vye@ip68-8-174-154.sd.sd.cox.net) |
| 20:47.15 | _Sam-- | there were also problems if you had the earpiece volume turned up too high, it induced echo |
| 20:47.20 | iCEBrkr | Man, it's sad when you have 11ms ping times on your own local network. |
| 20:47.21 | _Sam-- | because the mic would pick it up |
| 20:47.33 | [av]bani | yeah, but thats the problem with any phone, once you clip you can forget EC |
| 20:47.46 | _Sam-- | no, the mic was actually picking up the earpiece soudns |
| 20:47.47 | [av]bani | iCEBrkr: you going by the #'s in sip show peers? |
| 20:48.11 | iCEBrkr | [av]bani: Yea, but then when I did a regular ping, they were MUCH less. :P |
| 20:48.28 | iCEBrkr | [av]bani: But ya know, I wouldn't put that latency past this network. It's total shit |
| 20:48.39 | [av]bani | iCEBrkr: yes, sip show peers isnt a measurement of network latency |
| 20:48.53 | *** join/#asterisk lst2 (i=liquid@12-221-233-172.client.insightBB.com) |
| 20:48.54 | [av]bani | iCEBrkr: it's a measurement of the latency inside your phone |
| 20:49.13 | [av]bani | hint: the embedded cpus in hardphones are not ghz pentiums |
| 20:49.27 | iCEBrkr | :) |
| 20:49.27 | lst2 | any of you guys dealt with Affinity VOIP Telecom? |
| 20:49.29 | WasPhantom | -> gone |
| 20:50.13 | [av]bani | iCEBrkr: the polycom on our network has the highest # in the sip show peers, its very slow to respond |
| 20:50.33 | [av]bani | iCEBrkr: 85ms, but other phones on the same ethernet segment show 5ms |
| 20:50.49 | iCEBrkr | ohh |
| 20:51.05 | iCEBrkr | See that would have freaked me out on my home network or something |
| 20:51.10 | iCEBrkr | But here at work, our network is total junk |
| 20:51.19 | [av]bani | but you ping the phone, it comes back 0.1ms |
| 20:51.22 | iCEBrkr | 85ms would be expected :P |
| 20:52.30 | [av]bani | or to compare, a snom 360 hardphone takes 15ms on 100bt, but the snom 360 softphone on a 3ghz pentium takes 2ms |
| 20:52.35 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
| 20:53.00 | Qwell[] | Place your bets everybody...will Charter die again at 1:25 for the fourth day in a row? |
| 20:53.01 | [av]bani | the difference between a 150mhz cpu and a 3000mhz cpu :) |
| 20:53.07 | [hC] | anyone experienced double ringing when dialing over a PRI and NOT passing 'r' to Dial() ? |
| 20:53.14 | Qwell[] | I'll give 3:1 odds |
| 20:53.31 | [hC] | I cant figure out what causes this. and it only seems to happen on some models of handsets. cisco's dont do it, sipura's do |
| 20:53.32 | [av]bani | Qwell[]: what do i win? |
| 20:53.44 | mogorman | ill take your bet qwell |
| 20:53.47 | justinu | i just cancelled my cable modem service |
| 20:54.00 | Qwell[] | I'm fairly sure it'll die again... |
| 20:54.08 | justinu | too many problems that just fix themselves |
| 20:54.32 | _Sam-- | qwell put in me for 10 |
| 20:54.35 | _Sam-- | i'll take 10 says you're fine |
| 20:54.39 | Qwell[] | :p |
| 20:55.00 | [av]bani | [hC]: upgraded the sipura firmware? |
| 20:55.06 | _Sam-- | im a sucker for odds |
| 20:55.11 | Qwell[] | we should know in 30mins here |
| 20:55.21 | [av]bani | Qwell[]: "user error" |
| 20:55.26 | [hC] | [av]bani: its actually a linksys spa-941 (sipura) and its the newest firmware. also happens on aastra phones |
| 20:55.35 | Qwell[] | [av]bani nobody is home to error it :p |
| 20:55.38 | [hC] | [av]bani: im pretty sure its asterisk's fault somehow, since it only happens when i call out my PRI |
| 20:56.18 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 20:57.06 | [av]bani | probably, * might be sending remote ring indications confusing the sipura/aastra |
| 20:57.31 | [av]bani | [hC]: do you answer() then dial() ? |
| 20:58.23 | [av]bani | Qwell[]: doesnt matter, its always "user error" when it comes to cable companies |
| 20:58.42 | Qwell[] | [av]bani Nah, Sunday it was an outage.. |
| 20:58.43 | [hC] | [av]bani: no. why would i do that? :P |
| 20:58.50 | Qwell[] | no clue what its been since |
| 20:58.51 | [av]bani | [hC]: as an experiment |
| 20:58.59 | [hC] | [av]bani |
| 20:59.01 | [hC] | oops |
| 20:59.06 | Qwell[] | but if it happens again, I'm gonna get a free month |
| 20:59.07 | [hC] | [av]bani: ah.. nope havent tried that. |
| 20:59.24 | [av]bani | [hC]: try it ? :) |
| 20:59.34 | [hC] | [av]bani: k. :) |
| 20:59.45 | [av]bani | i'm guessing it might fix it |
| 20:59.52 | [av]bani | at least for pri |
| 21:00.09 | [hC] | [av]bani: that doesnt seem like a 'proper' fix though.... i mean it may cause it to work how i want, but that seems to only band aid the real problem |
| 21:00.21 | [av]bani | [hC]: yeah, shrug.. worth a try though? |
| 21:00.32 | Qwell[] | might help narrow down the problem |
| 21:00.35 | [hC] | [av]bani: ive seen this sort of thing happen before when the tone zone definitions in zaptel were incorrect.. its a canadian pri, but it behaves just like any other pri in the US... |
| 21:00.42 | [av]bani | with sip and voip there are lots of hacks and few "proper" fixes :) |
| 21:00.44 | iCEBrkr | [hC]: Whutup |
| 21:00.49 | [hC] | sup timmeh |
| 21:00.54 | iCEBrkr | :-/ |
| 21:05.46 | *** join/#asterisk jyukes (n=jameshot@pool-141-150-181-246.atc.east.verizon.net) |
| 21:06.35 | *** join/#asterisk TallAndy (n=dfa@81-178-225-109.dsl.pipex.com) |
| 21:07.02 | RoyK | ~nickometer j4m3s |
| 21:07.33 | mroth_imm | ~nickometer RoyK |
| 21:07.49 | mroth_imm | ~nickometer mroth_imm |
| 21:08.06 | mroth_imm | wth is that? |
| 21:08.46 | [TK]D-Fender | ~nickometer [TK]D-Fender |
| 21:08.53 | [TK]D-Fender | yay! |
| 21:09.36 | mroth_imm | i want to achieve 100% lameness |
| 21:09.54 | [TK]D-Fender | mroth_imm : Install A@H and come back here ;) |
| 21:10.03 | mroth_imm | ~nickometer sL1m_sH4dY |
| 21:10.22 | mroth_imm | oh my, you beat l33t slim shady! |
| 21:11.08 | mroth_imm | ~nickometer markster |
| 21:11.09 | Qwell[] | ~nickometer IIIIIII111111 |
| 21:11.16 | Qwell[] | I win |
| 21:11.34 | _Sam-- | [av]bani: sales guys have a gxp issue....maybe you know the answer smart guy.... |
| 21:11.39 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 21:11.43 | [av]bani | O RLY |
| 21:11.50 | _Sam-- | calls come in all at once...sometimes lines 1 2 and 3 are all ringing at once, ringall... |
| 21:11.58 | _Sam-- | one sales guy picks up line1.... |
| 21:12.04 | _Sam-- | but lines 2 and 3 dont ring any longer on the other phones |
| 21:12.06 | _Sam-- | they flash |
| 21:12.16 | [TK]D-Fender | ~nickometer [av]bani |
| 21:12.16 | _Sam-- | and the sales guys say sometimes they dont realize they are ringing |
| 21:12.25 | [TK]D-Fender | Far better than I! |
| 21:12.38 | clyrrad | I have AMP running, and phones can ring each other, but no voice, and this is all on a local network, phones and Asteriks server, I have checked rtp.conf and opned those ports on the Firewall |
| 21:12.48 | [TK]D-Fender | ~amp |
| 21:12.49 | jbot | i guess amp is NOT supported here! people using it should join #amportal |
| 21:13.05 | clyrrad | the prolbem is not with AMP, its the voice |
| 21:13.19 | [av]bani | _Sam--: i think its because of ringall, if one extension is answered the rest stop ringing |
| 21:13.20 | clyrrad | its not passing to the phones, i changed rtp.conf, do i need to change anyting else? |
| 21:13.36 | [av]bani | _Sam--: try a different ring strategy |
| 21:13.36 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 21:13.55 | _Sam-- | i dont know if you understand...lines 2 and 3 are ringing on the phone |
| 21:13.57 | clyrrad | does voice not go over UDP? |
| 21:13.58 | _Sam-- | they havent been answered |
| 21:14.01 | _Sam-- | but they just flash |
| 21:14.22 | [av]bani | maybe the blf subscription confuses them into thinking the local extension has been answered |
| 21:14.31 | [av]bani | when a remote is |
| 21:15.01 | _Sam-- | gonna turn the hints off see if it changes anything...the sales guy say the dont remember this behavior before i ugpraded the phones |
| 21:15.08 | [av]bani | try taking a phone off blf and see if it changes the ring problem |
| 21:18.40 | sevard | I'm trying to find example scripts, or scripts that anyone wrote that deal with asterisk+perl+mysql, where might I find these |
| 21:20.26 | jhiver | sevard, check out astcc |
| 21:20.34 | jhiver | it's asterisk+perl+mysql |
| 21:20.43 | sevard | thank you |
| 21:24.38 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:24.43 | generalhan | whats going on everyone ?! |
| 21:25.44 | generalhan | can some one tell me ... : is there a way to identify, automatically, the phone that picks up a call from the queue ? |
| 21:26.37 | _Sam-- | maybe using FOP? |
| 21:26.48 | generalhan | well ... what i mean is ... |
| 21:26.50 | _Sam-- | have you tried the flash operator panel |
| 21:27.23 | _Sam-- | i think it does show which person connected to the queued call, but im not positive, havent looked at it for a while |
| 21:27.40 | generalhan | i do use the FOP |
| 21:27.45 | generalhan | i love it but what i need this for is ... |
| 21:27.55 | clyrrad | Can someone give me some tips? I can get my phones to ring each other, but I get no voice from phone to phone, but from the Asteriks Server to the phone there is voice (ie... when leaving voice mail you can hear the auto attendant) |
| 21:28.47 | generalhan | I record all calls from a queue as the call comes into the queue. so those calls are saved into a directory called "recorded_calls" then they have the phone number of the person that called in. but i want to save it as the person who picked up the phone but i dont know how to figure that out |
| 21:29.24 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 21:30.01 | redder86 | how do I kill a call (SIP) ? |
| 21:30.14 | redder86 | ... from the CLI ? |
| 21:30.40 | austinnichols101 | redder86: soft hangup ... |
| 21:30.52 | KranZ | hangup |
| 21:31.00 | generalhan | _Sam--: you see where im going with this? have any idea how to take care of something like that ? |
| 21:31.13 | KranZ | redder86: try "soft hangup " |
| 21:32.05 | redder86 | thank you that worked |
| 21:32.29 | KranZ | np |
| 21:33.11 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 21:38.03 | fifer | I know this may be a longshot, but anyone here using the latest version of IPSwitchboard? |
| 21:44.11 | clyrrad | can anyone help me with the "no voice" problem? I can get phones to ring each other but no VOICE traffic, I have checked the firewall its completely off |
| 21:44.34 | Netgeeks | 913 active SIP channels |
| 21:44.43 | Netgeeks | with audio! |
| 21:44.56 | clyrrad | Can you help me? Im not sure what I am doing wrong |
| 21:45.15 | saftsack | are some germans here because of german sip provider? |
| 21:45.39 | *** join/#asterisk Samoied (n=Samoied@201.3.237.29) |
| 21:45.59 | clyrrad | Netgeeks, what can be stopping my audio from going phone to phone? |
| 21:47.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 21:52.22 | *** join/#asterisk ToTo (n=ToTo@host5-158.pool874.interbusiness.it) |
| 21:53.55 | *** join/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 21:54.38 | *** join/#asterisk razu (n=razu@ip49.cab19.mus.starman.ee) |
| 21:54.42 | razu | hi ... |
| 21:54.52 | propagandhi | does anybody know of any software that can run on a client computer that listens for calls coming in and can execute specific commands, or display a notification of the call coming in |
| 21:55.20 | razu | have anyone had luck installing Sweex X100P clone card ??? |
| 21:55.41 | generalhan | propagandhi: i have heard a lot of people talking about "jabber" ive never used it but incase no one answers you look into that |
| 21:55.52 | propagandhi | generalhan: ok, sweet thankyou |
| 21:55.54 | iCEBrkr | propagandhi: I wrote a CallerID notification app for Windows |
| 21:56.06 | propagandhi | iCEBrkr: yeah, cool would u let me have it |
| 21:56.18 | iCEBrkr | propagandhi: it's listed on the Asterisk 3rd party apps |
| 21:56.24 | iCEBrkr | iBCallerID |
| 21:56.28 | iCEBrkr | lemme get you a link |
| 21:56.32 | generalhan | iCEBrkr: how does the phone extension link with that persons workstation ? |
| 21:56.33 | propagandhi | iCEBrkr: thankyou |
| 21:56.45 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/?page_id=5 |
| 21:56.48 | iCEBrkr | Scroll to the bottom |
| 21:57.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-80-13-7.red.bezeqint.net) |
| 21:57.17 | iCEBrkr | generalhan: It listens on the manage port |
| 21:57.29 | iCEBrkr | generalhan: There's some setup to do, it just doesn't work out of the box :P |
| 21:58.04 | generalhan | iCEBrkr: thats fine i dont mind doing the setup i was just wondering if i had to have the phone plugged into that computer or if there was a way to set it up otherwise |
| 21:58.45 | iCEBrkr | generalhan: Like I said, it monitors asterisk events. |
| 21:59.16 | iCEBrkr | At the moment it's more for personal use, since it's insecure in the fact that someone could possibly set it up to monitor other extensions.... |
| 21:59.24 | generalhan | Ok so it monitors the events so i just have to tell it what PC is at what Extension ? |
| 21:59.26 | iCEBrkr | well, maybe not.. Since you have to have a manage login |
| 21:59.51 | iCEBrkr | generalhan: Nope, it logs into the manage port, you assign the program the extension to look for events for. |
| 22:00.21 | iCEBrkr | generalhan: http://www.voip-info.org/wiki/view/iBCallerID |
| 22:00.23 | generalhan | ohh that makes sense ... duh LOL. so i just set that extension at each workstation as i install the app on each station |
| 22:00.43 | propagandhi | iCEBrkr: it keeps asking me to specify the preferences, then restart the app, but thats an endless loop for me |
| 22:01.00 | iCEBrkr | propagandhi: Well then you didn't fill in a blank |
| 22:01.11 | propagandhi | all preferences are filled |
| 22:01.12 | iCEBrkr | propagandhi: http://www.voip-info.org/wiki/view/iBCallerID |
| 22:01.23 | iCEBrkr | Read the HOW-TO man.. |
| 22:01.31 | iCEBrkr | Then it probably couldn't login. |
| 22:01.35 | iCEBrkr | for OBVIOUS reasons |
| 22:01.52 | ramy | I want to setup asterisk to receive phone calls using X100P FXO, and allow the caller to use DTMF to make SIP calls. Any poiters to HOW-TOs or a good starting point ? |
| 22:02.23 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 22:02.43 | iCEBrkr | Hrrm, I'm going to have to update the documentation for 1.2.x |
| 22:02.43 | propagandhi | iCEBrkr: understood, sorry to have come across as ignorant |
| 22:02.54 | propagandhi | iCEBrkr: did u write this in C# |
| 22:02.57 | iCEBrkr | since ${CALLERIDNUM} is now a function. |
| 22:03.13 | iCEBrkr | propagandhi: nope |
| 22:03.30 | propagandhi | iCEBrkr: which language did u use |
| 22:03.43 | razu | ramy : for configuring the card or extensions ? |
| 22:03.44 | justinu | foul, probably |
| 22:04.51 | generalhan | can anyone tell me is there a variable for "receiving channel" or something like that ? |
| 22:05.18 | Qwell[] | generalhan: What are you trying to do? |
| 22:05.21 | ramy | razu: for the card |
| 22:05.33 | generalhan | i need to be able to print the extension of a worker that took a call from the queue, but i cant use $CHANNEL cause that will show me the called in on cahnnel |
| 22:05.46 | Qwell[] | generalhan: manager interface can get that |
| 22:05.51 | rene- | how well supported is the cisco 7960 telephone? |
| 22:05.56 | Qwell[] | there is an event when a queue is answered |
| 22:06.00 | Qwell[] | rene-: SIP or SCCP? |
| 22:06.23 | rene- | they are in SCCP now but i would like to switch them to sip |
| 22:06.30 | generalhan | Qwell[]:i record all the calls comming out of this queue by the queue name (i.e. /recorded_calls/) but i want to save them in the directory of the person that took that call (i.e. /ext7111/) |
| 22:06.32 | Qwell[] | well, both work... |
| 22:06.58 | generalhan | rene-:i use 15 of those phones in here on the SIP firmware and i love them ! |
| 22:07.07 | Qwell[] | generalhan: hack up the code to set a channel var |
| 22:07.21 | rene- | do i have to pay extra for getting those phones to speak SIP? |
| 22:07.40 | Qwell[] | rene-: YOu technically need to get a smartnet contract for them. You can get them for like $9 though |
| 22:08.27 | rene- | Not expensive at all |
| 22:08.31 | generalhan | Qwell[]:http://generalhan.pastebin.ca/40769 that is what i have to cut the channel for my outgoing but i dont think this will work for incoming |
| 22:08.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:09.49 | Skumling | rene-: my experience is that 7960 works quite well with SIP (and asterisk) |
| 22:09.59 | Qwell[] | They work better with SCCP ;] |
| 22:10.20 | Skumling | Qwell: heh okay, I didn't try it with SCCP :) |
| 22:10.58 | Qwell[] | try again |
| 22:11.06 | file | :( |
| 22:11.11 | generalhan | Qwell[]: did you take a look at my pastebin ?? is this what you meant for the Cut channel ? |
| 22:11.21 | Qwell[] | generalhan: I never said anything about cut... |
| 22:11.30 | Qwell[] | generalhan: hack the code to add a channel variable when it gets answered |
| 22:11.47 | _Sam-- | hey file, if my asterisk keeps segfaulting like i showed you...do you have any suggestions |
| 22:11.48 | generalhan | Qwell[]: how so ? thats what i dont understand how to do. |
| 22:11.59 | Qwell[] | I'll rephrase then |
| 22:11.59 | _Sam-- | this just started with 1.2.4...and it was fine until all the sudden today |
| 22:12.03 | file | _Sam--: not really |
| 22:12.11 | Qwell[] | generalhan: pay somebody to hack the code to add a channel variable when it gets answered |
| 22:12.16 | _Sam-- | how it has segfaulted like 3 times today |
| 22:12.24 | Netgeeks | well, I'm load testing a system here, and here is the best I've done so far... 1198 active SIP channels |
| 22:12.39 | Netgeeks | seems I'm hitting the 100Mb ethernet port limit |
| 22:12.43 | mroth_imm | anyone know if a basic init script exists for asterisk? |
| 22:12.46 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:12.48 | iCEBrkr | iBCallerID Wiki page all updated. w00t |
| 22:12.51 | Qwell[] | mroth_imm: make config |
| 22:12.52 | rene- | Qwell: they play better (7960's) with SCCP in asterisk? |
| 22:12.58 | Qwell[] | rene-: For me they do |
| 22:13.05 | justinu | netgeeks: i could never get my systems to go beyond about 170 channels |
| 22:13.06 | Qwell[] | with chan_sccp2 |
| 22:13.10 | justinu | Netgeeks: what's the secret? |
| 22:13.23 | Netgeeks | justinu: it's a sun |
| 22:13.36 | Qwell[] | Linux or Solaris? |
| 22:13.39 | Skumling | Qwell[]: what improvements do you get? better stability or added features? |
| 22:13.46 | Qwell[] | Skumling: kinda both |
| 22:13.47 | Netgeeks | nothing more than that.. Aurora linux ' |
| 22:13.57 | Qwell[] | aurora <3 |
| 22:14.05 | Qwell[] | Netgeeks: 1.9x? |
| 22:14.31 | Netgeeks | Linux dev-1 2.6.13-1.1603sp8smp #1 SMP Tue Nov 15 10:36:20 EST 2005 sparc64 sparc64 sparc64 GNU/Linux |
| 22:14.32 | _Sam-- | bweschke : can you check out another core? |
| 22:14.47 | mroth_imm | Qwell[]: thanks |
| 22:15.41 | Netgeeks | here is the fun part, Qwell: Brought up 10 CPUs |
| 22:15.42 | Netgeeks | Total of 10 processors activated (7987.20 BogoMIPS). |
| 22:16.05 | Qwell[] | Only 10? :) |
| 22:16.26 | Qwell[] | what speed procs? |
| 22:16.48 | Netgeeks | E4500 with 10 400MHz US-2's 10G mem |
| 22:17.03 | Qwell[] | Netgeeks: You should post your findings to the asterisk-dev mailing list. I bet John Todd would be interested to see those |
| 22:17.07 | oogle | that's a waq server |
| 22:17.20 | Netgeeks | JT has an account on the server :) |
| 22:17.45 | Qwell[] | Netgeeks: ahh |
| 22:18.05 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:18.10 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 22:18.14 | Netgeeks | I've got a second 100Mbps interface, going to bond them and see if I can hit 2k calls |
| 22:18.42 | Netgeeks | if so, it would be nice, I got this box for $950 on ebay |
| 22:18.55 | justinu | nice |
| 22:18.59 | Qwell[] | Netgeeks: Go ethernet |
| 22:19.06 | Qwell[] | or try local channels |
| 22:19.10 | justinu | too bad they're a PITA to rack |
| 22:19.48 | Netgeeks | and this is not pass-through, the test dialplan is pretty simple, it answers, sets a group prints the group count to cli, plays back a small wav file and goes into echo |
| 22:20.08 | file | how are you doing the test? |
| 22:20.37 | Netgeeks | sipp using a scenerio file, and a 5 minute long RTP stream to generate the traffic |
| 22:20.45 | file | good good |
| 22:20.56 | file | verified it's doing the RTP? |
| 22:21.31 | Netgeeks | yep, iftop on teh test generation system (AMD X2 4800+) reports 91.7Mbps for 1198 calls |
| 22:21.39 | file | excellent |
| 22:21.39 | rene- | Qwell: do you have a recommendation for a supplier of the firmware. (btw is this even legit?) |
| 22:21.52 | Qwell[] | rene-: You get the firmware from cisco, after you buy a smartnet |
| 22:21.54 | file | did the dialplan execution go wonky at all? (ie: suddenly stuff started to be executed in bursts) |
| 22:21.59 | rene- | Qwell:other than getting it directly from cisco, |
| 22:22.13 | *** join/#asterisk cyburdine (n=jburdine@208.2.145.2) |
| 22:22.15 | Netgeeks | and the sun reports in ifconfig - RX bytes:16509413568 (15.3 GiB) |
| 22:22.16 | Qwell[] | rene-: That's the only legal way to get it |
| 22:22.22 | Qwell[] | jeebus |
| 22:22.30 | Qwell[] | Netgeeks: You need gbit |
| 22:22.44 | Netgeeks | yeah, no decent sbus gbit cards available |
| 22:22.45 | Qwell[] | or a bonded QFE at least, heh |
| 22:22.54 | Netgeeks | I was thinking a bonded QFE would work |
| 22:23.03 | Netgeeks | should get me to about 5k ulaw calls |
| 22:23.05 | Qwell[] | I loves me my QFE |
| 22:23.09 | cyburdine | hey gang |
| 22:23.19 | MikeJ[Laptop] | hey homies |
| 22:23.25 | Netgeeks | file: the dialplan held up fine, I did get alot of warning... |
| 22:23.31 | rene- | ah so the other way would be buying it from a friendly asterisk user who has a smartnet contract |
| 22:23.34 | Netgeeks | many avoided deadlock warnings |
| 22:23.45 | rene- | i will try SCCP2 |
| 22:23.58 | Qwell[] | avoiding deadlocks is good, heh |
| 22:24.04 | MikeJ[Laptop] | hehe |
| 22:24.08 | MikeJ[Laptop] | hey mr. Qwell. |
| 22:24.11 | Qwell[] | Netgeeks: Don't type show channels |
| 22:24.13 | De_Mon | <PROTECTED> |
| 22:24.15 | Qwell[] | MikeJ[Laptop] y0 |
| 22:24.22 | Qwell[] | De_Mon: QFE = Quad Fast Ethernet |
| 22:24.25 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 22:24.44 | Netgeeks | and some of these: Feb 8 14:07:45 WARNING[6000]: chan_sip.c:11271 do_monitor: chan_sip: ast_sched_runq ran 26 all at once |
| 22:24.58 | file | Netgeeks: can we talk in privmsg? |
| 22:26.04 | TallAndy | Hi, in Asterisk + PHP there is a function called "asterisk_bridge_call" does anyone know how you would do this in an agi script e.g. phpagi? I need to 1. Make a call if the recipient presses *1 it would dial another number then bridge the 2 calls together. |
| 22:28.29 | Katty | file: it's time to abandon work, and search for muffinery. |
| 22:28.45 | Qwell[] | no e in muffinry |
| 22:28.50 | file | MUFFINS! |
| 22:29.10 | Katty | hey Netgeeks (= |
| 22:29.13 | _Sam-- | maybe you should have them for breakfast instead of pizza bites |
| 22:29.23 | Katty | Qwell[]: you clearly can't spell in the angeliverse. |
| 22:29.26 | justinu | _Sam--: blasphemy |
| 22:29.31 | _Sam-- | lol |
| 22:29.53 | cyburdine | in this line... register => 8015551212:blah@myprovider.com/8015551212 can I make myprovider.com anything I want? or does that have to be a valid domain? |
| 22:30.10 | cyburdine | I'm trying to figure out how to configure 4 sip trunks to my provider |
| 22:30.24 | cyburdine | it would seem I need 4 register lines |
| 22:30.37 | rene- | cyburdine: i think so |
| 22:31.00 | rene- | myprovider.com can be a host or an entry in sip.conf |
| 22:31.47 | rene- | disregard the last message pls |
| 22:32.05 | cyburdine | gotcha... but for each trunk I'd need a register line that is specific to a [config] |
| 22:32.16 | Netgeeks | Gonna see if 1.2.4 performs any differently compared to the version I was running |
| 22:34.05 | Netgeeks | Qwell: I just have to make sure I get the QFE card that has 4 separate FE engines instead of the one that has one engine and 4 shared ports |
| 22:34.15 | clyrrad | anyone here know why my phones can ring each other but no voice gets transmitted? |
| 22:34.44 | Netgeeks | cylrrad: one way or no-way audio usually indicates nat or firewall issues... usually |
| 22:34.53 | rene- | sip protocol is used for signaling and rtp for actual data |
| 22:34.53 | Netgeeks | clyrrad, even |
| 22:35.06 | rene- | something it is blocking the actual voice |
| 22:35.09 | rene- | s/data/voice |
| 22:35.16 | clyrrad | yea, but i have the firewall completely disabled |
| 22:35.44 | clyrrad | can it be a codec problem? |
| 22:35.47 | hypa7ia | clyrrad: are you behind nat? |
| 22:35.52 | rene- | it then might be codec negotiation, |
| 22:35.59 | Hymie | you're a codec problem, clyrrad... YOU"RE A CODEC PROBLEM!!!!!! |
| 22:36.00 | rene- | if you are reinviting |
| 22:36.04 | clyrrad | yes, I am behinde the NAT so is the server and all the phones |
| 22:36.19 | hypa7ia | clyrrad: nat is likely your problem |
| 22:36.29 | hypa7ia | clyrrad: sip right? |
| 22:36.32 | rene- | the phones and server are behind the same nat? |
| 22:36.35 | clyrrad | Yep SIP |
| 22:36.39 | Hymie | well, turning off re-invite is a good test |
| 22:36.44 | clyrrad | rene, yes all behinde the same NAT |
| 22:36.51 | rene- | then you are on a lan |
| 22:36.54 | clyrrad | i have canreinvite=no |
| 22:36.55 | rene- | there is no nat |
| 22:37.02 | clyrrad | right they on the same LAN |
| 22:37.07 | _Sam-- | hey file...if we can call HI through asterlink, HI can also can in to us on the toll free? |
| 22:37.19 | _Sam-- | <i would like to only make outgoing to HI and AK> |
| 22:37.26 | _Sam-- | but not a big deal, just curious. |
| 22:37.29 | rene- | i dont think you are having codec issues since a call shouldnt stablish if the codec negotiation fails, more so if asterisk is in the media |
| 22:37.31 | clyrrad | but just as a test I disabled the firewall anyway |
| 22:37.39 | file | _Sam--: no |
| 22:37.59 | file | _Sam--: two separate rateplans for inbound and outbound :) |
| 22:38.00 | clyrrad | ok so I have nat=never, and canreinvite=no and still no audio |
| 22:38.09 | rene- | your phones need not to use stun |
| 22:38.16 | _Sam-- | perfect...im monitoring an incoming call from HI....wasnt sure if he came in on the toll free |
| 22:38.19 | clyrrad | right, I do NOT have stun on any phones |
| 22:39.08 | *** join/#asterisk Soul (n=Soul@87-196-41-220.net.novis.pt) |
| 22:39.10 | _Sam-- | poor guy from teliax is probably footing that bill |
| 22:39.13 | rene- | i would go tru the sip debug payin attention to the sdp parts |
| 22:39.37 | clyrrad | rene I have done that I see no errors in the sip debug |
| 22:40.25 | rene- | ok, then please recheck the codecs issue, is the only thing left at this point.. |
| 22:40.39 | clyrrad | is there a diference between nat=no and nat=never? |
| 22:41.06 | clyrrad | Also i see on the CLI Attempting native bridge of SIP/201-8fa5 and SIP/300-60e6 |
| 22:41.14 | clyrrad | but only after I answer the call |
| 22:41.18 | rene- | that is normal |
| 22:41.32 | rene- | it is the bridging of media |
| 22:41.46 | clyrrad | but nothing is getting bridged by the looks of it |
| 22:41.51 | rene- | you said canreinvite=no |
| 22:41.56 | clyrrad | Right |
| 22:42.02 | clyrrad | and nat=never |
| 22:42.26 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 22:42.57 | rene- | never is for some uniden phones |
| 22:43.08 | clyrrad | set it to no? |
| 22:43.32 | rene- | you can, im not sure if its relevant |
| 22:43.58 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 22:44.06 | rene- | when you checked the logs, did you saw in the sdp parts that both parties shared common codecs? |
| 22:44.16 | clyrrad | same problem :( what about qualify=yes? |
| 22:44.22 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 22:44.23 | elg | I'm confused by queue's timeout |
| 22:44.28 | clyrrad | rene yes they both have the same codecs |
| 22:44.34 | elg | I want the queue to timeout and go to voicemail after 20 seconds |
| 22:44.34 | srt | nicht sauer sein also :) |
| 22:44.34 | rene- | it is mostly the same is a workaround for nat devices |
| 22:44.37 | elg | but it doesn't seem to do that |
| 22:44.37 | srt | nicht sauer sein also :) |
| 22:44.55 | elg | queue(everybodyq,rwtn,,,20) |
| 22:45.32 | clyrrad | so shoudl it be set to no or yes? |
| 22:45.44 | clyrrad | right now qualify=yes |
| 22:46.54 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 22:48.32 | De_Mon | I haven't used cvs in ages how do I update my sources? cvs update -- right? |
| 22:49.47 | De_Mon | oh hey, zaptel is building now. happy day |
| 22:51.14 | mogorman | svn |
| 22:54.26 | De_Mon | svn? |
| 22:59.04 | Qwell[] | De_Mon: Kill your cvs dir, and get the source from svn |
| 22:59.29 | _Sam-- | how come when safe_asterisk starts asterisk it doesnt assign a high priority to the asterisk process? |
| 22:59.43 | De_Mon | is svn more up-to-date? I don't have it installed |
| 23:00.05 | _Sam-- | asterisk mpg123 gets started with a higher priority |
| 23:00.26 | Netgeeks | kk |
| 23:00.36 | *** join/#asterisk GNu_Joe (n=joe@bi-02pt1.bluebird.ibm.com) |
| 23:01.13 | *** join/#asterisk AlexCTI (n=alex@64.251.9.44) |
| 23:01.22 | Qwell[] | De_Mon: cvs is old school, svn is the new shit |
| 23:01.46 | tzafrir_laptop | De_Mon, svn update |
| 23:02.17 | tzafrir_laptop | De_Mon, generally: try 'svn <cvs command-line>' |
| 23:02.41 | De_Mon | will svn talk to cvs repositories? I still have a few projects I might revisit some day ;P |
| 23:02.48 | Qwell[] | De_Mon: no |
| 23:03.58 | Netgeeks | no response? |
| 23:04.36 | Netgeeks | 1200 active calls, cpu is at 13.1% |
| 23:04.48 | Qwell[] | yikes |
| 23:05.01 | Qwell[] | still 100mbit? |
| 23:05.13 | Netgeeks | yes |
| 23:05.31 | Netgeeks | limit is 1250 calls on that interface |
| 23:05.42 | Netgeeks | but it looks like the realistic limt is about 1200 |
| 23:06.01 | De_Mon | http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
| 23:06.02 | Qwell[] | take out your video card, and drop in 3 QFEs |
| 23:06.11 | De_Mon | doesn't have svn instructions just cvs |
| 23:06.15 | Qwell[] | That should get you about... |
| 23:06.22 | Netgeeks | and my test box (athlon X2 4800+) starts falling apart at 1200 calls - both cpus up there at 100% |
| 23:06.29 | De_Mon | his videocard isn't taking up 3 slots! |
| 23:06.32 | Qwell[] | 9100 |
| 23:06.42 | Netgeeks | no vid card, serial console only |
| 23:06.56 | KranZ | sweetness |
| 23:06.56 | Qwell[] | 2 QFEs would max your CPU... |
| 23:06.58 | Netgeeks | I have 4 available sbus slots (each it's own sbus) |
| 23:06.58 | KranZ | how much ram |
| 23:07.02 | Netgeeks | 10G ram |
| 23:07.06 | Soul | good evening |
| 23:07.09 | *** part/#asterisk GNu_Joe (n=joe@bi-02pt1.bluebird.ibm.com) |
| 23:07.14 | KranZ | raid HD's? |
| 23:07.22 | Qwell[] | You could do 7.633 times the current amount |
| 23:07.30 | Netgeeks | nope, just one FW2 scsi |
| 23:07.41 | Soul | i hope somebody here has used BRI interfaces with asterisk, we really need some help |
| 23:07.54 | Netgeeks | on the sun, not the test box... the amd test box is 2G and ata hf |
| 23:07.54 | Qwell[] | Netgeeks: Go buy QFEs :P |
| 23:07.56 | Netgeeks | hd |
| 23:08.06 | Netgeeks | already got 2 coming |
| 23:08.09 | Qwell[] | get 7 more dualcore AMDs... |
| 23:08.12 | De_Mon | does zaptel even have an svn repository? All I've found is bsd stuff |
| 23:08.17 | Qwell[] | That'll about max you out |
| 23:08.31 | Qwell[] | De_Mon: Yes |
| 23:09.17 | Qwell[] | Netgeeks: What's the load like on that box with 1200 active calls? |
| 23:09.25 | *** join/#asterisk hamish (n=hamish@196-28-87-71.wdsl.co.za) |
| 23:09.31 | Qwell[] | I'd be curious about both boxes, actually |
| 23:09.48 | Nugget | wow, my fxo from digium, shipped ups ground, arrived in 1 day. |
| 23:09.52 | Netgeeks | at the time I had 1200 calls the user cpu load was 13.1% and the another 5.1% between system and interrupt load |
| 23:09.59 | [av]bani | Nugget: what card? |
| 23:10.03 | Netgeeks | memory loading looked to be just about 1G |
| 23:10.07 | Qwell[] | Netgeeks: What about the load though? Hit w |
| 23:10.11 | [av]bani | moving packets doesnt take much cpu |
| 23:10.21 | Netgeeks | ah, process load? |
| 23:10.24 | Qwell[] | yeah |
| 23:10.38 | Netgeeks | can't say, I was compiling a couple things on it at the same time |
| 23:10.43 | Qwell[] | wtf |
| 23:10.46 | Netgeeks | lol |
| 23:10.46 | [av]bani | even shifting 1mpps is nothing anymore, it used to be magic and require $1m ciscos, now you can do it with $300 PCs |
| 23:10.55 | Qwell[] | hmm |
| 23:11.14 | Netgeeks | I'll fire up the test again and watch close this time |
| 23:11.52 | Qwell[] | [av]bani: I can't with my ss5 |
| 23:12.03 | Qwell[] | I can barely get 30mbit out of my QFE |
| 23:12.12 | watchy | anyone here know about mibs? |
| 23:12.17 | Netgeeks | bani: you asked about a card? |
| 23:12.30 | watchy | can you use snmp with asterisk? |
| 23:12.43 | Qwell[] | watchy: If you write an snmp interface for it, sure |
| 23:12.44 | _Sam-- | i use mtrg w/asterisk |
| 23:12.57 | [av]bani | <Nugget> wow, my fxo from digium, shipped ups ground, arrived in 1 day. |
| 23:13.01 | [av]bani | <[av]bani> Nugget: what card? |
| 23:13.02 | watchy | sam: need to compile asterisk anyway special for it? |
| 23:13.05 | _Sam-- | i graph my number of active channels |
| 23:13.08 | azzie | Netgeeks, sorry... what software was processing 1200 calls?.. |
| 23:13.20 | [av]bani | Qwell[]: ss5, ew. i'm sorry to hear that |
| 23:13.21 | De_Mon | svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 |
| 23:13.23 | De_Mon | found it |
| 23:13.27 | Netgeeks | asterisk 1.2.4 on a sun E4500 |
| 23:13.31 | Qwell[] | [av]bani: I abuse that poor thing |
| 23:13.43 | Netgeeks | Qwell: amd test box load avg 53.50 cpu 100% on both |
| 23:13.47 | Qwell[] | Netgeeks: What's the ETA on the QFEs? |
| 23:13.48 | azzie | Netgeeks, did asterisk proxy RTP as well ? |
| 23:13.54 | Qwell[] | wow |
| 23:14.00 | Netgeeks | sun box: load 2.53 cpu 13.7% |
| 23:14.00 | [av]bani | Qwell[]: i was able to get 600mbit out of my athlon 1200 and crappy chinese clone GbE pci card |
| 23:14.03 | Qwell[] | What about on the E4500? |
| 23:14.11 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 23:14.41 | AlexCTI | Hi, Anyone can help me i have a server with 3 boards 4T1 each one, and if i remove any one the config is gone, so what do i need to do to get back the config with only onle board, i set the zaptel with the first 4 span and zapata,conf as well but still the unconfig on zttool |
| 23:15.08 | Netgeeks | on the E4500 load is at 4.20 cpu 10.9% stable now |
| 23:15.15 | azzie | Netgeeks, i suppose the call flow was sip-to-sip ? |
| 23:15.18 | Qwell[] | That's it? damn |
| 23:15.26 | Netgeeks | and there si an asterisk compiling also |
| 23:15.29 | Qwell[] | I expected far higher |
| 23:15.55 | Qwell[] | die damnit, die |
| 23:16.17 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-42-138.cybersurf.com) |
| 23:16.38 | _Sam-- | how does oreilly make any money on the book if you can download it for free |
| 23:16.46 | Netgeeks | azzie: test platform is an adm x2 running sipp with a 4 minute ULAW stream it calls the sun, who answers, plays back a short file, and then goes into echo for the remaining 4 minutes |
| 23:16.54 | [av]bani | they hope you'll buy the books from them that arent free |
| 23:17.02 | *** join/#asterisk Soul (n=Soul@87-196-41-220.net.novis.pt) |
| 23:17.13 | Netgeeks | at 1200 calls the ethernet card is reporting around 90Mbps traffic |
| 23:17.26 | azzie | Netgeeks, what is cpu speed on e4500 ? |
| 23:17.37 | watchy | hey Sam |
| 23:17.37 | _Sam-- | i guess i should contribute some money, i would have paid 30 bucks for it |
| 23:17.42 | _Sam-- | hi watchy |
| 23:17.47 | Qwell[] | _Sam--: buy it |
| 23:17.50 | watchy | you got info on how to do asterisk with snmp |
| 23:17.51 | Qwell[] | It's worth it |
| 23:18.07 | _Sam-- | its more fun to read online..i can irc at the same time |
| 23:18.17 | Netgeeks | azzie: 10 CPUs @400MHz each 8MB caches |
| 23:18.37 | _Sam-- | watchy: sure, one sec...are you at all familiar with mtrg already? |
| 23:18.39 | De_Mon | the oriely asterisk book? |
| 23:18.39 | azzie | Netgeeks, heh, same one as we've got here :-) gotta switch it from mysql to * :) |
| 23:18.40 | Qwell[] | Wow, he came back |
| 23:18.43 | Qwell[] | Qwell: bastard |
| 23:19.04 | Qwell[] | CTCP PING reply from Qwell: 435.038 seconds |
| 23:19.15 | azzie | Netgeeks, this is impressive... do you have any stats on RTP packet loss? |
| 23:19.21 | _Sam-- | hey qwell do i owe you 10 or you owe me 30? :) |
| 23:19.21 | watchy | saM: i am with cacti a little |
| 23:19.29 | azzie | I would guess * can loose heck a lot of RTP under such load |
| 23:19.33 | Qwell[] | _Sam--: Well, it did technicaly die for about 8 minutes there :p |
| 23:19.56 | Netgeeks | azzie: not yet, sipp reports no errors at all for the 1200 calls, but I'll need to get something else to check rtp loss |
| 23:20.06 | mogorman | hey i want my money Qwell |
| 23:20.16 | Netgeeks | azzie: at 1200 calls with the FE card at 90Mbps I suspect some loss has to be there |
| 23:20.17 | Qwell[] | freenode just didn't let him timeout :P |
| 23:20.23 | Qwell[] | and it was an hour and a half late |
| 23:20.30 | mogorman | in small unmarked bills |
| 23:20.47 | websae | anyone using a E911 provider here? |
| 23:20.55 | Qwell[] | I hate Charter... |
| 23:21.16 | azzie | Netgeeks, yeah, or loss/latency during * being late for processing RTP packets in case process scheduling is not precise somewhere |
| 23:21.26 | _Sam-- | watchy: this is the page i used to set up mine ...but it is fairly mrtg specific (from the asterisk wiki pages): http://karlsbakk.net/asterisk/ |
| 23:22.00 | fifer | <PROTECTED> |
| 23:22.17 | *** part/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 23:22.17 | RoyK | i've wanted to |
| 23:22.18 | *** join/#asterisk _Paulo_ (n=pirch@201-27-20-209.dsl.telesp.net.br) |
| 23:22.19 | RoyK | :) |
| 23:22.29 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 23:22.31 | watchy | thanks |
| 23:22.36 | RoyK | fifer: but i haven't gotten through to take a deep look into what it does |
| 23:23.08 | Netgeeks | <-- got a conference call to get on, I'm away for about 1 hour, demoing billing product |
| 23:23.14 | fifer | I have looked at it in the past, but never realy tried to use it untill now, and, ofcourse, now that I WANT it to work, it is giving errors ;-) |
| 23:23.27 | fifer | Hopfuly something simple, but checked most of the simple stuff |
| 23:23.29 | azzie | Netgeeks, this is very interesting and and useful testing. Keep us posted :) |
| 23:23.45 | azzie | Netgeeks, or publish your results on voip-info (or I can do it for you) |
| 23:24.25 | Netgeeks | I'll do some more involved testing this coming weekend and I'll publish it somewhere, which can be linked/posted to mail lists, etc. |
| 23:24.55 | Netgeeks | I'll also test the amd box as the ast server as well as a dual xeon and a couple single proc machines for comparison |
| 23:25.31 | azzie | that would be so cool |
| 23:25.31 | meshuga | anyone ever done a sip trunk with CCM into asterisk? I'm getting congestion tone immediately on the * side |
| 23:25.43 | meshuga | same issue the guy in the wiki post comments about CMM integration has in fact |
| 23:25.44 | meshuga | CCM^ |
| 23:25.52 | meshuga | http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration |
| 23:25.53 | _Sam-- | this is interesting |
| 23:25.53 | _Sam-- | It should be noted that evidence now suggests that connecting |
| 23:25.54 | _Sam-- | together two completely separate, single-CPU systems may provide far |
| 23:25.54 | _Sam-- | more benefits than simply using two processors in the same machine. |
| 23:25.59 | Soul | im sorry, who was i talking to about BRI interfaces in asterisk ? |
| 23:26.15 | RoyK | god |
| 23:28.23 | Qwell[] | Netgeeks: Did you already say how fast the CPUs were in the E4500? |
| 23:28.56 | Netgeeks | 400MHz Ultrasparc-II |
| 23:29.07 | Netgeeks | on the conference call now, so I may be slow to respond |
| 23:29.17 | brookshire | _sam: it all depends on the application |
| 23:29.25 | _Sam-- | that is from the book! |
| 23:29.43 | [av]bani | Feb 8 15:17:35 NOTICE[15218]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received |
| 23:29.46 | [av]bani | hmm |
| 23:31.45 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 23:32.22 | _Sam-- | i need to start learning how to build multiple systems that operate together to handle alot of calls |
| 23:38.02 | _Sam-- | did anyone from here help write the book? |
| 23:40.12 | _Sam-- | i saw bkw got some props in the early part |
| 23:40.37 | mogorman | are you talking about the book _Sam-- ? |
| 23:40.52 | _Sam-- | yessir |
| 23:41.29 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 23:42.40 | Telamon | Can anyone recommend a good recorder script? Basically I want to send a filename, have it record a new file, allow preview, and then ask to replace the old one or abort. I'm just too lazy to write it myself. :) |
| 23:43.14 | Qwell[] | _Sam--: blitzrage and jsmith |
| 23:43.19 | mogorman | isnt that what app_dictate is for? |
| 23:43.27 | Qwell[] | and no clue whether Jim IRCs |
| 23:44.02 | bweschke | _Sam--: do you still have that core up? |
| 23:44.10 | _Sam-- | i have a new core, havent run gdb |
| 23:44.35 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 23:45.03 | _Sam-- | i think it is related to this error: (sorry will be a few lines, but not pb material) |
| 23:45.04 | _Sam-- | [msg(file)] -- Registered SIP '104' at 207.245.69.59 port 1494 expires 3600 |
| 23:45.04 | _Sam-- | [msg(file)] Feb 8 17:09:11 ERROR[9599]: chan_sip.c:10790 handle_request_subscri context default |
| 23:45.15 | _Sam-- | it segfaulted immediately after that |
| 23:46.47 | JunK-Y | run gdb and submit a bugs on the bugs tracker please. |
| 23:46.51 | Telamon | mogorman: Hmm, do you have a link to some docs on app_dictate? Nothing on voip-info.com and google just brings up the source code and mailing list of dev work. |
| 23:47.07 | mogorman | show application dictate |
| 23:47.08 | mogorman | ^_^ |
| 23:47.41 | Telamon | Ahh, forgot about that. :) |
| 23:47.53 | bweschke | _sam--: no chance you had sip debug on did you? |
| 23:48.02 | _Sam-- | no i didnt |
| 23:48.07 | _Sam-- | but it was related to hints |
| 23:48.15 | _Sam-- | i didnt have a hint on an extension |
| 23:48.18 | bweschke | because looking closer at your last crash - thread #1 had an address violation too out of chan_sip.c |
| 23:48.18 | _Sam-- | but a phone was looking for it |
| 23:48.25 | _Sam-- | caused that message |
| 23:48.31 | bweschke | yep |
| 23:48.38 | *** join/#asterisk NeonLevel (n=NeonLeve@dsl-201-133-181-39.prod-infinitum.com.mx) |
| 23:48.59 | bweschke | we need to see who's sending what that's causing you to crash |
| 23:49.10 | [av]bani | gxp2000 |
| 23:49.15 | _Sam-- | yeah i turned it all off |
| 23:49.19 | [av]bani | you could just have him register one of his phones to your asterisk :) |
| 23:49.21 | _Sam-- | and have had o problems |
| 23:49.32 | [av]bani | _Sam-- could crash your asterisk for you :) |
| 23:49.34 | NeonLevel | good day everyone, i have * version 1.2.4 but application Meetme doesn't show up??? did it change its name? |
| 23:49.40 | Nugget | http://slacker.com/~nugget/asterisk12.php |
| 23:49.45 | Nugget | (for those who were asking earlier) |
| 23:49.52 | [av]bani | Nugget: what fxo did you buy from digium? |
| 23:49.55 | JunK-Y | Nugget: the stuff from last week? does it works fine? |
| 23:49.57 | Nugget | I've got a TDM400P |
| 23:50.04 | Nugget | I added another FXO to it toda |
| 23:50.05 | Nugget | +y |
| 23:50.08 | bweschke | NeonLevel - if you don't have Zaptel loaded on your machine app_meetme won't build |
| 23:50.10 | [av]bani | oh, another module |
| 23:50.13 | Nugget | right |
| 23:50.23 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:50.26 | NeonLevel | bweschke: oh i see... |
| 23:50.26 | Ariel_ | hello everyone |
| 23:50.30 | _Sam-- | bweschke which command should i run from the gdb prompt that would be best |
| 23:50.33 | NeonLevel | thanks for your help |
| 23:50.53 | bweschke | let's do what you did before |
| 23:51.03 | [av]bani | Nugget: i get timeout on slacker.com |
| 23:51.05 | Nugget | JunK-Y: yeah, works fine |
| 23:51.12 | Nugget | [av]bani: using linux 2.6.8? |
| 23:51.15 | fifer | Hey, does FOP always eat up most CPU resources? something does not seem right, though I'm seeing this on multiple computers |
| 23:51.16 | JunK-Y | what was the model again? |
| 23:51.20 | bweschke | plus 'info threads' |
| 23:51.22 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:51.22 | Nugget | AU-600 |
| 23:51.22 | [av]bani | Nugget: 2.6.14 |
| 23:51.29 | Nugget | strange, maybe they reintroduced the bug. |
| 23:51.30 | JunK-Y | im gonna buy one for my alone week-ends :) |
| 23:51.30 | _Sam-- | heh, file gave me the command i used last time...hey file can you help a dumb fool out |
| 23:51.59 | Nugget | Linux 2.6.8 (and maybe 2.6.14 too) creates malformed mtu packets that can't get through many types of scrubbing firewalls. |
| 23:52.41 | bweschke | thread apply all bt full |
| 23:52.52 | _Sam-- | ty |
| 23:52.54 | [av]bani | hmm |
| 23:52.58 | cyburdine | any one seen this error? "Not a local SIP domain" when trying to register a sip phone to asterisk. |
| 23:53.27 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 23:53.33 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 23:53.55 | [av]bani | Nugget: eh... 2.6.13.2 is fine, 2.6.14.3 is not |
| 23:54.02 | [av]bani | AHAHAHAHA |
| 23:54.04 | [av]bani | no wonder... |
| 23:54.23 | [av]bani | you are in the infamous 68/8 ip space |
| 23:54.29 | Nugget | what's infamous about that? |
| 23:54.30 | [av]bani | there is a busted firewall between me and you |
| 23:54.30 | Telamon | cyburdine: I'm guessing you have an @ in your username in the SIP phone, or the default context in your sip.conf entry doesn't exist. Just a guess though. |
| 23:54.57 | Nugget | got any references to "the infamous 68/8 space"? I've never heard of that. |
| 23:55.08 | cyburdine | cool thanks for the pointer... |
| 23:55.42 | _Sam-- | bweschke : http://sam.pastebin.com/545854 thanks. |
| 23:55.56 | [av]bani | Nugget: 68/8 and 69/8 were recently assigned, but a lot of old 'network guides' list it as bogon space |
| 23:56.14 | Nugget | what's "recently" in your context? |
| 23:56.27 | [av]bani | Nugget: well, jun 01, aug 02 respectively |
| 23:56.31 | Nugget | *nod* |
| 23:56.41 | Nugget | care to test something for me? |
| 23:56.43 | [av]bani | Nugget: sadly those spaces are still massively blocked |
| 23:56.50 | [av]bani | http://puck.nether.net/~jared/papers/69-paper.html |
| 23:57.08 | Nugget | Try to hit slacker.com from your 2.6.14.3 box now. |
| 23:57.19 | [av]bani | nothing |
| 23:57.21 | [av]bani | nada |
| 23:57.22 | bweschke | _Sam-- : ya - you're crashing on the same call to get_header in chan_sip again |
| 23:57.25 | Soul | i am having problems with BRI interfaces in NT mode and an alcatel interface with BRI interfaces in TE mode. the asterisk BRI interfaces work with one siemens isdn phone in TE mode. can somebody help ? |
| 23:57.31 | bweschke | we're gonna need a sip debug |
| 23:57.35 | Nugget | ok, thanks. It's not the malformed packet issue, then. |
| 23:57.45 | [av]bani | its some idiot firewalling it between us |
| 23:57.45 | _Sam-- | give me another hour or so after we close up shop. |
| 23:57.48 | Nugget | whew. I was worried that Linux had reintroduced that bug, which would suck |
| 23:57.53 | _Sam-- | then i will put it back to make it crash |
| 23:58.05 | [av]bani | Nugget: you're unfortunate enough to be on 68/8, which is teh suck. |
| 23:58.29 | Netgeeks | I'm in 68/8 as well! |
| 23:58.32 | Nugget | I've been in 68/8 for about four years now and this is the first I've ever heard of this, so I don't think the problem is at all widespread. |
| 23:58.40 | _Sam-- | i maintain it is the phone, not asterisk...but asterisk shouldnt crash on it |
| 23:58.51 | [av]bani | Nugget: it is though. its just that most failures are silent... people just assume your site is dead and move on. |
| 23:59.12 | Nugget | presumably I'd be hitting it all the time trying to reach sites, though, which I don't. |
| 23:59.39 | Nugget | can you send me a traceroute to the point where it dies? |
| 23:59.42 | [av]bani | looks like its being filtered somewhere in sprintlink |