| 00:11.49 | *** part/#asterisk nicknam12322 (021d147d@gateway/web/freenode/ip.2.29.20.125) |
| 00:18.44 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 00:20.47 | slav3_kitten | i hate cisco i hate cisco i hate cisco |
| 00:26.47 | jeffspeff | hey, just wanted to spread the word, found some cheap servers on ebay that'll make pretty good * boxes. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=290737584521 |
| 00:27.00 | jeffspeff | no, i'm not selling, just found them and bought 2. :) |
| 00:31.18 | jpsharp | Not a bad deal at all, assuming you can hunt down drive trays. |
| 00:31.28 | *** join/#asterisk jeffspeff (~Jeff@67.231.40.120) |
| 00:33.38 | *** join/#asterisk nobodyshome (~dijib@208-96-84-35.eastlink.ca) |
| 00:34.25 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 00:48.39 | ChannelZ | U320 SCSI? Is anyone still making drives? |
| 00:49.54 | jeffspeff | ChannelZ, you talking about that server i linked to? |
| 00:50.11 | dijib | anybody know lvm? |
| 00:50.51 | jeffspeff | dijib, as in logical volume managment? |
| 00:53.52 | jpsharp | ChannelZ: I have a dozen 320GB U320 drives on a shelf in the office. |
| 00:54.27 | jpsharp | Nowait, I'm sorry. They're only 72GB. |
| 00:54.57 | ChannelZ | yeah |
| 00:55.15 | ChannelZ | I have an old drive array that is U320 but I don't even remember now what drives are in them. |
| 00:55.34 | ChannelZ | 160gb maybe (x 8) |
| 00:56.36 | jeffspeff | i want to find some 1tb ones |
| 00:56.54 | jpsharp | You'll pay a pretty penny for those. |
| 00:57.11 | jpsharp | Unless you can find someone who is shitcanning an old server. |
| 00:58.37 | jeffspeff | my personal server has 8 hotswap bays, and i would love to put 8 1tb drives in it and put them in a raid 1or maybe 5 |
| 00:59.01 | jeffspeff | either way, it'd be nice... but a nice dream for now |
| 00:59.03 | jeffspeff | lol |
| 01:00.21 | dijib | jeffspeff: yes |
| 01:00.50 | jpsharp | I'm going to go 4GB fiberchannel when I build out a storage system. |
| 01:00.59 | jeffspeff | dijib, what kind of problem are you having? |
| 01:01.53 | dijib | ive purchased and install a 3tb drive. pvmoved the data off the old ones to the new one. and now have physically removed the old ones. everything is working well and my data is there. but...... |
| 01:02.46 | slav3_kitten | jeffspeff, i had one of those power edge's |
| 01:02.50 | slav3_kitten | i loved it |
| 01:03.09 | jeffspeff | i just bought 2 of them |
| 01:03.16 | WIMPy | Hmm. Why does teh keyboard of the PC have Caps lock and Scroll lock on? And why don;t I get any video any more? |
| 01:03.22 | jeffspeff | starting a new company on the side, they'll be good for that |
| 01:03.54 | jpsharp | Just make sure that you don't have anything on the workbench when you fire them up. They spool up like a jet turbine when you first power them on. |
| 01:04.00 | jeffspeff | dijib, but?? |
| 01:04.01 | dijib | its not really a problem its just in the Webmin LVM management consoles in the LV details " Physical volumes allocated sdb 355.75 GB , sdb 110 GB , sdb 232.88 GB , sdb 231.37 GB" those were the former drives capacities |
| 01:04.15 | jeffspeff | jpsharp, i'd imagine so |
| 01:04.20 | dijib | i cannot repordude that with pvdisplay lvdisplay or vgdisplay |
| 01:04.39 | jpsharp | Blew a bunch of papers off my desk. |
| 01:04.41 | dijib | just wondering if i can combine all of those into one so its cleaner looking |
| 01:04.50 | jeffspeff | dijib, did you do a fresh install? |
| 01:04.58 | jeffspeff | when you the drives in? |
| 01:05.29 | WIMPy | *phew* must have been the keyboard itself. |
| 01:08.29 | jpsharp | I spilled coke on it. Sorry. |
| 01:08.50 | dijib | fresh instal of what? |
| 01:09.22 | jeffspeff | your OS |
| 01:11.16 | jeffspeff | it sounds like you'll need to rebuild your lvm structure. i'd hop on the channel for your linux distro |
| 01:14.06 | jeffspeff | think of an lvm as a software raid. you can't just use the dd command and copy an lvm to dissimilar hardware |
| 01:16.41 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 01:41.16 | *** join/#asterisk dgeary2 (~david@49.176.99.91) |
| 01:42.10 | dgeary2 | my ISP has recently started allocating private IP addresses to its customers and NATting their connections |
| 01:42.51 | dgeary2 | does anyone know of a simple test to prove that this breaks SIP? |
| 01:43.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 01:43.50 | [TK]D-Fender | It's obvious and documented everywhere |
| 01:44.07 | [TK]D-Fender | If you tell the other side your IP is some private value then they may trust that and you get no audio. |
| 01:45.08 | [TK]D-Fender | You need RTP forwarded as well as SIP and your server needs to know the IP to advertise. |
| 01:49.17 | dgeary2 | my ISP is very trying. what's interesting is that i am able to receive incoming SIP calls with 2-way audio (the SIP control connections is unencrypted) |
| 01:50.08 | dgeary2 | i suspect their NAT device has some kind of application helper for SIP which snoops on the control connection and forwards the required ports |
| 01:52.00 | dgeary2 | are you aware of any public SIP echo-test server? |
| 01:52.45 | *** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca) |
| 01:56.44 | slav3_kitten | the 7940 and 7960 use the same firmware right? |
| 02:02.11 | jpsharp | I believe so, yes. |
| 02:05.48 | slav3_kitten | hmmm |
| 02:05.48 | slav3_kitten | i hate cisco |
| 02:07.48 | jeffspeff | lol |
| 02:12.58 | slav3_kitten | you laugh... |
| 02:13.35 | jpsharp | meh. I bent the 7940s to my will years ago :) |
| 02:13.51 | slav3_kitten | mine are bending me to their will |
| 02:14.46 | slav3_kitten | spent all day in the most uncomfortable chair listening to the whine of my server closet while on the phone with my friend who is a ccnp telling me how my phone should work hooked up to a port with an access an voice vlan |
| 02:14.56 | slav3_kitten | turns out you need to set it up as a trunk line |
| 02:15.02 | slav3_kitten | no idea why |
| 02:16.08 | slav3_kitten | huzzah got the fing thing upgraded |
| 02:16.17 | jpsharp | If you don't connect the phone to a trunk port, the switch isn't looking for 802.1q vlan tags on the incoming/outgoing frames. |
| 02:16.48 | slav3_kitten | apparently new model switches use cdp to auto configure it as a trunk |
| 02:16.57 | slav3_kitten | or so i'm told |
| 02:17.24 | jpsharp | Assuming you have CDP enabled on the switch. I forget if it is default on or not. I think it is. |
| 02:17.30 | slav3_kitten | by 3 different ccnps i know, none of which have used anything as old as a 3500xl |
| 02:17.44 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 02:18.00 | slav3_kitten | yea cdp is enabled |
| 02:18.27 | slav3_kitten | phone 2/6 provisioned |
| 02:18.37 | slav3_kitten | well upgraded, not provisioned |
| 02:19.14 | jpsharp | I vaguely recall there being a voice vlan config on the 3500xl. I never split the voice/data onto separate vlans on the networks I built. |
| 02:19.42 | WIMPy | remembers that as well. |
| 02:19.44 | slav3_kitten | ah i decided it all needed to be fancy like |
| 02:20.02 | jpsharp | So all the switches had a gigabit port set for trunk and then I used native vlan 'switchport access vlan blah' for each individual port. |
| 02:20.02 | slav3_kitten | i think this is because i'm a glutton for punishment |
| 02:20.11 | WIMPy | I retired mine last year. |
| 02:20.22 | jpsharp | Since I had so much crap that didn't grok 802.1q. |
| 02:22.12 | slav3_kitten | as soon as i get all these things upgraded i'll in theory start setting up the asterisk server to actually allow them to make calls |
| 02:22.21 | jpsharp | And I had a mix of Crisco phones and Polycom phones. |
| 02:22.27 | jpsharp | And Cisco too. |
| 02:22.32 | slav3_kitten | once that's setup... i'l then work on getting a static IP from my isp |
| 02:23.04 | jpsharp | What do you need a static for? devices external to your network? |
| 02:23.49 | slav3_kitten | i had read i needed static... |
| 02:24.24 | jpsharp | Not unless you have devices external to your network that need to register. And even then, static isn't mandatory. |
| 02:24.32 | slav3_kitten | i'd like to be able to have some external soft phone extensions to run |
| 02:25.04 | slav3_kitten | p.s. typos will make phones not do what you want |
| 02:25.28 | jpsharp | My asterisk box here does just fine with a dynamic ip + dyndns. I have a softphone on my iphone that registers to my home Asterisk machine. |
| 02:26.48 | slav3_kitten | ok i should rephrase that |
| 02:26.57 | slav3_kitten | i need a public IP to be able to work it |
| 02:27.25 | slav3_kitten | currently i'm natted about 3 times previous to the "modem" from my ISP which hands my router an all that a 192.168.1.100 ip |
| 02:27.39 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 02:27.45 | jpsharp | Oh, yes. Dispense with the NAT. |
| 02:28.07 | slav3_kitten | well there will be nat 1x from the router to voice vlan |
| 02:28.10 | slav3_kitten | but that's all |
| 02:28.22 | slav3_kitten | appropriate ports forwarded an all that |
| 02:28.53 | jpsharp | Thats why I'm glad I figured out how to put my DSL modem into bridge mode. No NAT. |
| 02:29.34 | jpsharp | For a while, their DHCP server would had out an address to whatever I had connected, but now it is smarter and limiting me to 2 dynamic public IPs. |
| 02:29.52 | jpsharp | I had half a dozen VMs, all with external global IPs. |
| 02:30.04 | slav3_kitten | lol |
| 02:30.29 | slav3_kitten | i only get one public IP |
| 02:30.34 | slav3_kitten | an it costs me 5/month |
| 02:31.12 | slav3_kitten | that's in addition to the 60/m it costs me for 3mbps |
| 02:31.21 | jpsharp | Ow |
| 02:31.31 | jpsharp | Remind me to never move there. |
| 02:31.46 | pppingme | slav3_kitten you're getting charged just to get a real ip, or is that for a 2nd ip? |
| 02:32.42 | jpsharp | That's why "real internet connectivity" is near the top of the list of house-searching requirements. |
| 02:33.06 | slav3_kitten | that is for 1 real IP |
| 02:33.19 | pppingme | slav3_kitten OUCH! |
| 02:33.25 | jpsharp | Oh, honey, look at this house....Oh, only crappy verizon DSL. |
| 02:33.26 | jeffspeff | i'd love to only have to pay $5 for an IP |
| 02:33.30 | jeffspeff | i pay $15 |
| 02:33.38 | jeffspeff | for a static |
| 02:34.04 | slav3_kitten | jpsharp, it was for me too... comcast said 8 times by 3 people that we could get business class cable internet at the location of our prospective house |
| 02:34.34 | slav3_kitten | we closed on the property, called them to come install it, they came out an said "oh hell no we can't get you service at this house" |
| 02:34.49 | jeffspeff | oh wow! |
| 02:34.59 | slav3_kitten | so now i have a WISP |
| 02:35.21 | slav3_kitten | so when it rains really heavy the 2.4ghz wifi to the water tower 2 miles of gets a little slow as hell |
| 02:35.37 | jeffspeff | lmao |
| 02:35.51 | slav3_kitten | no laughing... |
| 02:36.37 | slav3_kitten | i just end up wanting to cry |
| 02:36.44 | jpsharp | When we find a house, we look at the google street view and I look at the wiring on the poles. 'Yep, there's a cable plant...nope, only see phone trunks'. |
| 02:37.08 | slav3_kitten | hindsight |
| 02:37.54 | pppingme | I'm missing 1gb/s by just a block |
| 02:38.26 | jeffspeff | wow, where do you live pppingme ? |
| 02:38.40 | pppingme | KC! |
| 02:38.45 | pppingme | google fiber |
| 02:38.47 | jeffspeff | kansas city? |
| 02:38.59 | pppingme | yep |
| 02:39.07 | jeffspeff | i've got friends that live there |
| 02:39.15 | jeffspeff | sounds like i might be moving my servers. lol |
| 02:39.25 | jpsharp | For that, I'd be out there with a pickaxe running a piece of condiut that block my own damn self. |
| 02:39.50 | pppingme | yeah, thats gig in each direction, one of this slow down on the upload side |
| 02:40.03 | jpsharp | Or making real good friends with my next door neighbor. |
| 02:40.04 | jeffspeff | agrees with jpsharp |
| 02:40.05 | pppingme | none of this slow down on the upload side |
| 02:40.33 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 02:41.53 | jpsharp | I don't complain about 25/2 DSL here. Solid, I can almost hit the CO with a rock. |
| 02:42.06 | slav3_kitten | so jpsharp this CTL file, how important is it |
| 02:43.06 | jpsharp | Depending on what version of firmware is on the phone, it'll need to be loaded to tell the phone to go load new firmware. |
| 02:43.16 | jpsharp | Especially if you're converting phones from SCCP to SIP. |
| 02:43.34 | jpsharp | And/or converting old firmware to ones based on the Universal loader. |
| 02:43.56 | slav3_kitten | hmm |
| 02:44.50 | slav3_kitten | well i'll beat out this 7911 issue in a bit |
| 02:45.35 | jpsharp | I've not touched a 7911, so I can't help much there. |
| 02:46.25 | slav3_kitten | the 7940's an 7960 go into unprovisioned with the correct name attached so i know it's reading the config files |
| 02:46.37 | slav3_kitten | 7911 is not sure. gotta setup the sip.con to test em i guess |
| 03:25.50 | slav3_kitten | hmmm i got a phone unprovsioned error |
| 03:37.17 | *** join/#asterisk j4m3s (~j4m3s@pdpc/supporter/active/j4m3s) |
| 03:37.47 | j4m3s | anyone have any success installing res_digium_phone on centos5 with asterisk10 rpms? |
| 03:46.05 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.251) |
| 03:59.40 | *** join/#asterisk tehrabbitt (~tehrabbit@unaffiliated/tehrabbitt) |
| 03:59.49 | tehrabbitt | hey, is there a freepbx style GUI for Asterisk 11? |
| 04:00.42 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 04:19.41 | infinity_ | tehrabbitt: i think there is a beta of freepbx w/11 |
| 04:27.15 | tehrabbitt | infinity_: yeah, I just found out freepbx supports version 10 so i'm going to use that :D |
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| 04:59.24 | slav3_kitten | why the hell is my phone unprovisioned |
| 04:59.29 | slav3_kitten | grumbles |
| 05:11.11 | slav3_kitten | ok got the phone working |
| 05:19.14 | jpsharp | You just have to glare at it hard enough. |
| 05:20.17 | WIMPy | Glare? Wasn't that this special dahdi feature that wasn't known before? |
| 05:21.59 | jpsharp | Either that or threaten to toss it into the litter box. |
| 05:25.15 | slav3_kitten | apparently it's the solution to cisco crap. turns out the quotes in the config were killing things |
| 05:26.52 | slav3_kitten | can't seem to get the logo url thing to work though |
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| 06:03.42 | crackguy | is there a way to extract Cisco SPA-504G settings to an XML file? |
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| 07:07.09 | slav3_kitten | wtf changed... |
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| 07:41.22 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 07:43.40 | deo | hi all, can i ask for question here |
| 07:44.46 | deo | how do we declare a route in asterisk? |
| 07:45.39 | carrar | ~ask |
| 07:45.39 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:46.13 | carrar | What do you mean declare a route? |
| 07:46.40 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 07:46.41 | deo | hi carrar :) |
| 07:46.46 | carrar | hi |
| 07:47.01 | deo | the scenario is this > i have two asterisk on different location |
| 07:47.43 | deo | one from here.. and one from US,.. i want to route calls from here to US pbx |
| 07:47.51 | deo | if dialling a US number... |
| 07:48.05 | carrar | make a extension rule |
| 07:48.13 | carrar | _NXXNXXXXXX |
| 07:48.18 | carrar | and route that to your US box |
| 07:48.19 | deo | btw., im using trixbox...:) |
| 07:48.24 | carrar | sucks to be you |
| 07:48.30 | deo | ??? |
| 07:49.05 | carrar | You're not gonna get much help in here for that crap |
| 07:49.15 | deo | :( |
| 07:49.28 | carrar | But general generic stuff you can |
| 07:49.37 | carrar | go back and compile from Asterisk source |
| 07:49.40 | carrar | and do it right |
| 07:50.15 | kaldemar | or go to #trixbox |
| 07:50.24 | carrar | but the same pattern matching extension plan will need to be used |
| 07:50.53 | deo | ok thanks btw... carrar kaldemar |
| 07:52.55 | slav3_kitten | well hell, i may have killed one of my phones |
| 07:53.17 | slav3_kitten | it's like stuck eternally upgrading and sitting at the main cisco screen |
| 08:01.50 | slav3_kitten | cisco docs "give it 20 mins" |
| 08:02.01 | slav3_kitten | http://www.cisco.com/en/US/ts/fn/620/fn62949.html |
| 08:02.29 | deo | hi carrar , how about if you dial that number ex:) a US number, but you will not hear anything... |
| 08:03.53 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:04.44 | kaldemar | deo: how did you dial? |
| 08:05.29 | deo | 1-805-XXX-XXXX |
| 08:05.35 | deo | ecluding the - |
| 08:05.39 | deo | *excluding |
| 08:05.47 | deo | kaldemar: ^ |
| 08:10.39 | deo | sometimes, after adjusting the extension rule to _NXX ,, it wills say all circuit are busy now... |
| 08:10.51 | deo | but we can dial other US number, fyi. |
| 08:13.25 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-kjjdexrnvuhsesiu) |
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| 08:18.33 | carrar | try _1NXXNXXXXXX |
| 08:19.47 | deo | i find it as the call walked through because ive seen on the terminal that the call went through.. |
| 08:20.02 | deo | however i dont hear any sounds.. |
| 08:20.04 | deo | carrar: ^ |
| 08:20.09 | carrar | So your UK box is sending the calls |
| 08:20.17 | carrar | and your USA box is not answering it? |
| 08:20.20 | deo | yes... |
| 08:20.25 | carrar | so fix the usa box |
| 08:20.26 | deo | but some number would be |
| 08:20.38 | deo | i can dial any number aside from 1805 |
| 08:20.41 | deo | carrar: ^ |
| 08:21.11 | carrar | You should connect the two boxes with IPSEC |
| 08:21.21 | carrar | so the SIP & RTP is encrypted |
| 08:21.26 | deo | hmmnn |
| 08:21.40 | deo | i also find this on the terminal is making progress passing it to |
| 08:22.39 | wdoekes | 1-805-[01].. ? |
| 08:23.23 | carrar | perhaps you have some other pattern matching going on? |
| 08:29.33 | deo | wdoekes: carrar the number 1-805-XXX-XXXX |
| 08:29.41 | deo | any number on that pattern |
| 08:30.02 | deo | :( |
| 08:32.42 | wdoekes | if I understand correctly, you've managed to setup the pattern matching, but the connection with the US box yields no sounds, right? |
| 08:33.24 | wdoekes | in that case, you must rephrase your problem |
| 08:34.38 | deo | yes wdoekes , actually the scenario is > i can call some US number on any pattern except 1-805 |
| 08:35.11 | deo | i did what carrar stated above with the dial pattern matching... _1NXXNXXXXXX |
| 08:36.09 | wdoekes | deo: sounds != call setup.. you need to figure out if the call setup works |
| 08:36.18 | wdoekes | if it does, then pattern matching has 0 to do with the problem |
| 08:38.11 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 08:45.35 | deo | the call went through as ive seen on the terminal wdoekes .. however i didnt hear any sounds, even rings :( |
| 08:46.10 | wdoekes | in that case you'll need to debug that.. sip set debug on, rtp set debug on |
| 08:46.41 | slav3_kitten | i think i need to go to sleep. it took me 15 minutes to figure out that wdoekes was a person here an not some brit term i've never heard |
| 08:46.42 | deo | ok will try |
| 08:52.37 | wdoekes | deo: it could be that the 1805 numbers require PRACK, a feature to acknowledge early media |
| 08:53.00 | deo | hhmmmm ive seen early media shows on my zoiper |
| 08:53.03 | deo | wdoekes: ^ |
| 08:53.26 | deo | whenever i tried to call that number |
| 08:54.44 | wdoekes | asterisk doesn't to PRACK.. |
| 08:54.47 | wdoekes | http://svn.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/ |
| 08:55.29 | wdoekes | s/to/do./ |
| 08:59.40 | deo | thanks will read that.. |
| 08:59.57 | deo | any headsup to PRACK?? what is it actually... |
| 09:00.00 | deo | wdoekes: ^ |
| 09:00.44 | wdoekes | 09:52 < wdoekes> deo: it could be that the 1805 numbers require PRACK, a feature to acknowledge early media |
| 09:00.58 | wdoekes | deo: ^ |
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| 09:02.34 | wdoekes | "what is a feature?" "what is early media?" "why does it need to be acknowledged?" all perfectly legal questions |
| 09:03.45 | wdoekes | "what is it actually..." however, is not. it doesn't even have a question mark |
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| 10:00.40 | bulkorok | somebody knows what this could cause: WARNING[29448]: channel.c:1507 __ast_queue_frame: Exceptionally long voice queue length queuing to SIP/... |
| 10:00.54 | bulkorok | it comes after dialplan-app: Answer(3000) |
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| 10:35.28 | [sr] | ei WIMPy, found the reason about HW echo, the driver on dahdi for that openvox card doesn't support it, i need to use a openvox dahdi mofified version |
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| 11:56.55 | mirela666 | Hi, is there any other way to get phone-context from the sip url except extension patterns |
| 11:59.10 | mirela666 | for example 111;phone-context=dispatch@execute-local to automaticlly send 111 to [dispatch] context and not [execute-local] |
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| 12:18.18 | deo | hi all, any suggestion if what to use > asterisk compile from source or those packages(elastix,trixbox) ..need your ideas guys |
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| 12:25.08 | fjfalcon | Hello all. Can symbol "#" be read by Read function? |
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| 12:39.46 | mirela666 | deo: always better from source |
| 12:40.07 | biomorph | Hi all. |
| 12:40.24 | mirela666 | deo: but if you need quick solution, than trixbox is good (but abandoned procject) |
| 12:40.49 | biomorph | I have set up an asterisk box which successfully receives calls from and routes outgoing calls through sipgate. |
| 12:41.20 | biomorph | But what I want to do is route the incoming call dependent on the number that was dialled. |
| 12:41.33 | biomorph | I can't find a variable that gives me that. |
| 12:43.25 | biomorph | So my external telephone number is 01234567, my sip number is 7654321. I can determine the sip number 7654321, but not the 01234567 |
| 12:43.31 | biomorph | Any ideas? |
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| 12:44.37 | kaldemar | biomorph: you just described what dialplan does. |
| 12:45.20 | kaldemar | biomorph: when a call comes in, it lands in the context you have defined in sip.conf. in that context you put extensions that match different dialed numbers. |
| 12:46.26 | biomorph | in extensions.conf I have the sip number. But I want to know what number the incoming person dialled. |
| 12:47.00 | biomorph | So I may have ten different PSTN numbers, but only four SIP numbers. |
| 12:47.28 | biomorph | It's the PSTN number I want to use to route the call once I have received it. |
| 12:48.01 | biomorph | Sorry if my terminology is a bit woolly.... |
| 12:48.06 | kaldemar | it is. :) |
| 12:49.06 | kaldemar | in extensions.conf you put patterns or extensions that match incoming numbers, i.e. what the user dialed. end of story. |
| 12:50.21 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:50.35 | biomorph | kaldemar, Thanks for the help here. But the number the customer is dialling is 01234567, but the number my asterisk server is getting is 7654321 which is my sip number. |
| 12:50.43 | kaldemar | you have accounts for the provider, and you probably need to register to them. in the registration statement in sip.conf you put /exten which defines what they send to you when a call comes in. |
| 12:51.46 | biomorph | Ah - light appears above biomorph's head...... |
| 12:54.37 | biomorph | OK - that makes sense. But I can register with a sip provider to give me three phone numbers, but just one trunk. |
| 12:55.05 | biomorph | In that situation I would still like to be able to determine the number the customer has actually called. |
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| 12:57.07 | kaldemar | "just one trunk" <-- we're not talking about copper pairs here. |
| 12:57.50 | kaldemar | if you get three numbers, the situation is exactly the same. you register to them with three different credentials then. |
| 12:58.23 | kaldemar | if not and the provider only sends you one incoming number, then there's nothing YOU can do about it. |
| 13:02.34 | biomorph | kaldemar, Thanks for your help. I was missing the meaning of the last entry on the register..... |
| 13:03.02 | biomorph | Now I can talk to the sip provider to see what they can provide. |
| 13:03.36 | WIMPy | fjfalcon: No. I once did a patch for that. It's somewhere on jira. |
| 13:13.11 | SeRi | WIMPy: I have a question for you. |
| 13:13.26 | SeRi | or any body in the chann |
| 13:13.47 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat62.beelinegprs.ru) |
| 13:14.13 | SeRi | Is it to much to at least expect a carrier to communicate to their customer that a hughe ass Hurricane is coming their way and that they mostlikely will be fully off line? |
| 13:14.59 | SeRi | That we will not attempt to mirror our site because the time and effort will be spent on employee safety |
| 13:15.10 | SeRi | Is that unreasonable? |
| 13:15.56 | kaldemar | depends on your contract. what's the point? |
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| 13:16.34 | WIMPy | In my experience the only thing you can expect, is to get billed. |
| 13:16.42 | SeRi | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:16.58 | SeRi | My point is that due to Sandy CC went fully of the grid |
| 13:17.14 | SeRi | They had no DR in place and they not even try to get ready for what was coming |
| 13:17.45 | SeRi | they just went by by |
| 13:17.49 | WIMPy | look at the contracts today. They define very clearly what, when and how you have to play, but don't refine what you can expect to get for that. |
| 13:18.19 | Maliuta | The only thing most telco's do well is bill |
| 13:18.24 | WIMPy | define |
| 13:18.42 | Maliuta | the rest of there stuff is basically bend over and take it as it is |
| 13:18.50 | Maliuta | s/there/their/ |
| 13:19.06 | SeRi | no shit |
| 13:19.24 | WIMPy | And even if tehy define what you can expect (as my telco partially does) that doesn't mean thay will be able to deliver that. And no, they won't care. |
| 13:20.12 | WIMPy | My telco is definitely unable to fulfill their contract. But they know that I know that all others are just as bad so I have no choice to get anything better anyway. |
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| 13:20.27 | WIMPy | That's how business works these days. |
| 13:20.47 | Maliuta | SLA's are good for that, they can't deliver? They pay you $$$'s |
| 13:21.07 | Maliuta | You still have to face the down time though |
| 13:21.56 | WIMPy | They're selling features their equipment doesn't support. |
| 13:21.57 | SeRi | I see. I was blinded I guess. |
| 13:22.12 | WIMPy | So it never has worked according to contract and probably never will. |
| 13:22.21 | SeRi | I expected so much but in general is just not that way |
| 13:22.44 | SeRi | good to know everybody's view on this and experience |
| 13:22.52 | Maliuta | WIMPy: I've been there, I worked at an ISP (years when DSL was shiny new in .au) where our upstream was changed and we were put onto a billing plan that didn't exist yet |
| 13:23.44 | Maliuta | that ended up in a huge dispute over the something like $28k in over billing, that I had to sort out by going through the day to day usage for 6 months of usage |
| 13:24.00 | SeRi | damn. |
| 13:24.08 | SeRi | sucks to be you that day |
| 13:24.17 | WIMPy | Oh yes. That's nice. I also had a telco once where they used different price lists for billing than they cave the customers. And they took over half a year to correct that, even though not only I had |
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| 13:24.25 | WIMPy | some heavy discussions with them about that. |
| 13:24.38 | Maliuta | can't wait for the Australian NBN to come to his neighborhood. FTH all the way. |
| 13:24.56 | SeRi | nice |
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| 13:25.01 | Maliuta | SeRi: well it took me all day to sort out that we owed them $5k, and then a week to get them to play ball |
| 13:25.22 | SeRi | Maliuta: damn. |
| 13:25.34 | WIMPy | Yes, I was one of at least two people to write an alternative billing software for their service to calculate what you had to pay according to contracts. |
| 13:26.12 | SeRi | well... My question came about here because I am arguing with forum fanatics about how bad CC did regarding communcations, DR, or planning, |
| 13:26.27 | Maliuta | WIMPy: I think I've written at least 3 ISP billing systems now, and a bunch of stuff to munge info from upstreams to stop getting ripped off |
| 13:26.52 | Maliuta | meh. time for bed |
| 13:27.34 | SeRi | g/n |
| 13:27.34 | WIMPy | A company I know quite well went throug that procedure every month. They demanded corrected billd from the telco until the difference was just a few thousands at which point they stopped because the |
| 13:27.41 | WIMPy | time spent on the issue was no longer worth it. |
| 13:28.12 | WIMPy | What does DR mean? |
| 13:28.38 | SeRi | WIMPy: Disaster Recover |
| 13:28.47 | SeRi | s/Recover/Recovery |
| 13:28.53 | SeRi | damn |
| 13:28.54 | SeRi | lol |
| 13:29.44 | WIMPy | I don't think may think about that. They are happy if things *seem to* go to plan under normal conditions. |
| 13:30.37 | SeRi | Yea I just figure that out, Though voip.ms did had a plan in place |
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| 13:31.09 | WIMPy | Since the telco market has been opened to competition the quality has dropped to a point of practical non-existance. |
| 13:31.29 | SeRi | as soon as they where informed that their main DC where their site and NYC pop are hosted where not going to be online due to a gen been offline under water they moved their servers and site to a back up DC. |
| 13:31.45 | SeRi | WIMPy: I see. |
| 13:32.31 | WIMPy | And yes, prices have fallen as well, but not nearly as much as quality. |
| 13:33.58 | SeRi | I was not around for byod services until the last 2yrs so I would not know... |
| 13:34.13 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216.191.106.165) |
| 13:34.29 | SeRi | My first voip service ever vonage about 7yrs a go or maybe longer |
| 13:34.46 | WIMPy | Everyone can get an impression on his home phone line, really. |
| 13:35.13 | file | my telco just throws in phone over my fiber for really no extra price, so meh I just use it |
| 13:35.27 | WIMPy | Today you can dial and then wait seconds until the call connects. |
| 13:35.30 | file | SIP back to analog then back to SIP :D |
| 13:36.43 | SeRi | lol |
| 13:36.56 | SeRi | WIMPy: I actually dont seattle for that. |
| 13:37.17 | WIMPy | That's BTW the reason I use Asterisk. Not that I like it, but it's still better than what the professionals offer nowadays. |
| 13:37.55 | SeRi | what would you guys consider comcast tel service to be? |
| 13:38.17 | SeRi | to me looks like the infrastructure is as what file just discribe |
| 13:38.39 | file | it's VoIP on a tightly controlled network, using a derivative of MGCP |
| 13:38.52 | SeRi | file: Thanks for the info. |
| 13:39.13 | SeRi | Is that way my alarm system works ok on their system but not through regular voip? |
| 13:39.34 | file | probably |
| 13:39.47 | [TK]D-Fender | end-to-end QoQ |
| 13:39.49 | [TK]D-Fender | QoS* |
| 13:40.12 | WIMPy | Possibly it's not even VOIP. |
| 13:40.17 | file | VoIP isn't inherently bad/jittery/whatever, it's the network that it is used on ultimately that dictates the experience |
| 13:40.23 | SeRi | WIMPy: Those where my thoughts |
| 13:40.34 | WIMPy | There are good working NGN solutions out there. Unfortunatly not many telcos use them. |
| 13:40.37 | SeRi | file: agree |
| 13:40.53 | file | http://en.wikipedia.org/wiki/PacketCable |
| 13:40.56 | WIMPy | Here exactely none :-( |
| 13:41.13 | file | I can never remember "PacketCable", I always have to Google it |
| 13:42.48 | SeRi | interesting |
| 13:43.22 | SeRi | [TK]D-Fender: I was googling your term. Interesting. Seems like wireless carrier also use it for video delivery |
| 13:43.58 | WIMPy | I've been told in .dk they use VoDSL, i.e. doing it at ATM level which oddly enough is ecately what ATM was made for. |
| 13:44.27 | file | WIMPy, wow... that's... something |
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| 13:51.16 | coppice | WIMPy: do you know who makes the boxes for that? |
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| 13:52.36 | WIMPy | The danish ones? |
| 13:52.47 | coppice | yes |
| 13:53.03 | WIMPy | No. But I can try to find out. |
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| 14:26.48 | skrusty | afternoon, i wonder if anyone could point out why, all of a sudden (after upgrading one of our servers to 1.6.2.24) dialling 800 brings up no console output... but causes all phones to pickup! it's like it's some sort of paging system! I've never ever seen this happen before! |
| 14:27.02 | skrusty | there's no output at the console at all for the call |
| 14:27.34 | WIMPy | You upgraded to an ancient version? |
| 14:27.38 | skrusty | yeah :) |
| 14:27.56 | skrusty | we're in process of getting ready to move to 1.8, but we're not quite there yet! |
| 14:28.06 | skrusty | non the less, have you ever seen this before? |
| 14:28.53 | WIMPy | I don't know your dialplan. |
| 14:29.01 | skrusty | t's not in our dialplan |
| 14:29.02 | skrusty | i |
| 14:29.16 | skrusty | and there's no console output |
| 14:29.29 | skrusty | but dailling 800 calls every phone and they all pickup |
| 14:29.42 | skrusty | like one massive conf call |
| 14:30.08 | *** join/#asterisk markit (~marco@151.78.74.112) |
| 14:31.13 | skrusty | i even put a phone into a context with nothing in it, and it still happens |
| 14:31.48 | skrusty | the only thing in features.conf (which is i might note commented out) is a reference to parking on 800-850 |
| 14:34.51 | *** join/#asterisk galba (~pif@ll62-62-242-251-62.ll62.iam.net.ma) |
| 14:39.28 | markit | hi, I'm confused about how my sip provider can allow me do pstn calls through it. I've read that the registration process is only for be "located", so seems important for incoming calls. But in registration I provide username and password, does it work that way for every invite too? |
| 14:40.02 | markit | (and "nonce" is generated by server at first 401?) |
| 14:40.36 | markit | since sip is on udp, there is not a "session" on so the invite is inside that, initiated with registration |
| 14:43.12 | WIMPy | yes |
| 14:43.48 | WIMPy | But the registration does not initiate anything. |
| 14:43.57 | kaldemar | markit: the credentials you provide upon registration are not re-used when you make calls. you need to configure those to a peer in sip.conf. |
| 14:43.58 | WIMPy | As you said, it's only to tell the peer where to find you. |
| 14:44.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:44.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:44.55 | WIMPy | Unless you register from within a peer. |
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| 14:46.50 | markit | ok, but I mean, when I dial a PSTN number,the sip server that provides me the pstn trunk has to bill me for that |
| 14:46.58 | markit | so needs to know that I'm me |
| 14:47.34 | markit | so does INVITE get a "authentication challenge" like "register"? |
| 14:47.46 | WIMPy | Yes, each call is authenticated individually. |
| 14:47.51 | WIMPy | yes |
| 14:47.54 | markit | I've never noticed with wireshark (but can't test right now, I've asterisk at home while I'm at work now) |
| 14:48.15 | markit | AH |
| 14:49.11 | markit | wondering if there is an "how things really work" document that explains all this... troubleshooting SIP seems a lot complicated to me |
| 14:50.00 | markit | thnaks a lot |
| 14:51.11 | kaldemar | you'll find a basic specification on SIP in RFC 3261. |
| 14:55.15 | [TK]D-Fender | skrusty, PASTEBIN <- |
| 14:55.17 | [TK]D-Fender | ~pb |
| 14:55.17 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:00.07 | WIMPy | coppice: So far I only found out that the ISP in question does no longer exist. |
| 15:05.30 | *** join/#asterisk notze (~notze@31-18-135-48-dynip.superkabel.de) |
| 15:05.33 | notze | hi there |
| 15:06.10 | notze | is there anyway to create escalations on asterisk, like if one user doesnt pick up the next one is beeing called= |
| 15:07.22 | kaldemar | use a timeout in Dial and do another Dial in the next priority. |
| 15:07.46 | coppice | WIMPy: maybe it didn't work out so well :-) |
| 15:12.36 | notze | kaldemar can it be configured by gui? |
| 15:14.46 | skrusty | [TK]D-Fender, there's nothing to paste |
| 15:15.01 | [TK]D-Fender | notze, queuerules.conf <- this is the only FreePBX option for this. Everything else is hand-coding. |
| 15:15.07 | WIMPy | coppice: Yes, I gueass anything that works just isn't competitive. But I don't know anythig about the story. |
| 15:15.35 | [TK]D-Fender | skrusty, If there is a call in yoursystem, there IS. "core set verbose 10" |
| 15:15.50 | [TK]D-Fender | skrusty, Phones don't magically start ringing for no reason. Go look at your CLI properly |
| 15:16.43 | skrusty | i understand that, verbose was 30 |
| 15:16.48 | skrusty | i'll set it to 10 now |
| 15:17.11 | [TK]D-Fender | No functional difference |
| 15:17.22 | skrusty | ok - so yet there was no output at all.... |
| 15:17.29 | [TK]D-Fender | So go wait for a call like you described |
| 15:17.39 | [TK]D-Fender | And enable SIP DEBUG and watch |
| 15:17.46 | skrusty | will do |
| 15:17.56 | skrusty | i have wireshark running on a mirror port |
| 15:18.02 | *** join/#asterisk italorossi (~textual@189.124.196.68) |
| 15:18.05 | jeffspeff | anybody know where i can find something like the UPA-1 device listed here http://www.broad-tel.com/products/phoneadapter.php ? it doesn't look like this company is still in business. |
| 15:18.26 | j4m3s | upon loading res_digium_phone.so on asterisk 10.7.1 (from Digium's centos5 repo) i get "res_digium_phone.so: undefined symbol: ast_presence_state" Is there a special version of Asterisk 10 i need for the digium phone resource? |
| 15:24.43 | skrusty | ok - well that's explained! :) Cisco group paging feature! |
| 15:24.46 | skrusty | fs... |
| 15:25.00 | skrusty | no idea why it's only just started to happen though |
| 15:25.11 | chris_n | is there any way that the change to standard time over the weekend could be associated with extensions suddenly responding to invites with 603 Declined? |
| 15:25.34 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:25.34 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:25.35 | skrusty | it's on the phones not asterisk :) |
| 15:26.40 | x2s | Hi. I'm wondering if there is some kind of fax manager for asterisk out there, which has equal functionality like hylafax. |
| 15:27.40 | x2s | Because receiving faxes is easy, but sending can be a bit painful, if you need a queue and error mangement and you've got to write everything yourself |
| 15:35.50 | coppice | x2s: a couple of people have started hylafax compatible managers, but don't know of anyone who saw it through to completion |
| 15:39.42 | x2s | coppice: what a shame |
| 15:39.46 | drmessano | Anyone have the SILK codec working in Asterisk 10 or 11? |
| 15:39.54 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:39.59 | x2s | then I've got to write my own manager |
| 15:45.09 | notze | i want to call from asterisk to outside |
| 15:45.16 | notze | and if no andwer next person is beeing callef |
| 15:45.19 | notze | is tehre anything |
| 15:45.24 | notze | even commercial |
| 15:45.37 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 15:45.53 | mjordan | drmessano: what are you running into? |
| 15:46.17 | jeffspeff | notze, so you want to place a call to an outside number and then if it's not answered call somebody else in a list? |
| 15:47.49 | [TK]D-Fender | notze, http://www.vicidial.org/vicidial.php |
| 15:54.52 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 15:55.10 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:55.14 | *** join/#asterisk gremlin054 (57547e41@gateway/web/freenode/ip.87.84.126.65) |
| 15:56.17 | gremlin054 | Hi all. A quick question - is there any way to set fromstring in voicemail.conf on a per context basis? |
| 16:00.16 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:03.50 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:06.24 | *** join/#asterisk krotos (~d3v1l@host139-51-dynamic.252-95-r.retail.telecomitalia.it) |
| 16:06.27 | krotos | hi all guy |
| 16:08.46 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:13.25 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:17.01 | jeffspeff | gremlin054, i would have the peers that belong in a certain context set a special variable that's referenced from the voicemail. if the global variables allow it. not sure, never tried it |
| 16:30.16 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:30.17 | slav3_kitten | well i've gotten 5 of my 6 phones working an may have bricked one of them |
| 16:31.14 | WIMPy | You can still use it as door stop. |
| 16:31.27 | Qwell | unless it's a Grandstream. |
| 16:31.36 | Qwell | those couldn't keep many doors open... |
| 16:31.43 | coppice | get some more and make an actual brick wall with them |
| 16:31.43 | slav3_kitten | but it's a doorstop i need working and can't really afford to replace |
| 16:31.55 | *** join/#asterisk nicknam12322 (021d145d@gateway/web/freenode/ip.2.29.20.93) |
| 16:32.13 | slav3_kitten | reading about it it seems this odd timing issue only presents in the 7911 ad not the 7906 that uses the same firmware |
| 16:35.56 | slav3_kitten | i'm trying the recovery procedure on it right now but i'm none to hopeful |
| 16:37.40 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 16:38.15 | *** join/#asterisk gtTuna (~Tuna@d149-67-253-75.try.wideopenwest.com) |
| 16:38.35 | gtTuna | Does anyone know if Aastra phones support DHCP Options for FTP (161/162) ? |
| 16:40.22 | gtTuna | trying to move away from TFTP |
| 16:40.35 | slav3_kitten | what's wrong with tftp gtTuna ? |
| 16:42.59 | Qwell | That app seems fishy. |
| 16:43.13 | gtTuna | nothing, really, and I know FTP really isnt that much more secure |
| 16:43.36 | gtTuna | but on a couple servers we need to keep TFTP open to the world for some clients |
| 16:44.20 | _Corey_ | security is trivial... |
| 16:45.08 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:45.33 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 16:45.49 | jeffspeff | anybody in here up to forking an opensource project? i'm getting the feeling this is more than i can do by myself http://www.bigbluebutton.org/ |
| 16:46.17 | jeffspeff | one of the main things i want to change is migrating the voice and video back to asterisk and away from freeswitch |
| 16:46.54 | jeffspeff | also make it more configurable with other asterisk configurations, etc. to just play nicer over-all and be more flexible |
| 16:48.19 | gremlin054 | jeffspeff, Thanks for that - I will take a look! |
| 16:48.49 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:55.02 | slav3_kitten | does anyone know what the inside of a tlv file should look like |
| 16:57.35 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 16:58.19 | slav3_kitten | hell yea got that phone to unbrick |
| 16:59.02 | WIMPy | I hope you have another door stop to replace it :-) |
| 16:59.21 | slav3_kitten | WIMPy, i have an old 266mhz laptop |
| 17:01.55 | *** join/#asterisk italorossi (~textual@189.124.196.68) |
| 17:10.33 | *** join/#asterisk d_preston215 (~chatzilla@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 17:12.26 | *** join/#asterisk mobile_gordita (~Robert@66-87-93-161.pools.spcsdns.net) |
| 17:16.17 | *** join/#asterisk nicknam12322 (021d145d@gateway/web/freenode/ip.2.29.20.93) |
| 17:18.40 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 17:22.29 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:24.11 | d_preston215 | Anyone ever hear of being able to place an outgoing call from an extension while it had a ringing incoming call, all with Call Waiting off? |
| 17:25.06 | jpsharp | On an analogue line or a SIP phone? |
| 17:25.31 | jpsharp | It cannot happen on an analogue line. |
| 17:27.19 | d_preston215 | SIP |
| 17:28.50 | jpsharp | I wouldn't see why it couldn't happen then. |
| 17:29.52 | jpsharp | If the phone is smart enough or has "multiple lines". |
| 17:30.40 | d_preston215 | Cisco 7960 |
| 17:31.25 | d_preston215 | Asterisk says that phone is busy in because it has call waiting disabled. |
| 17:43.58 | SeRi | d_preston215: look at the phones option. |
| 17:44.04 | SeRi | It could b DND while on call |
| 17:44.22 | SeRi | My Polycom has an option to turn off call waiting and set the user in DND while on a call. |
| 17:45.22 | d_preston215 | DND isnt enabled. |
| 17:47.01 | *** join/#asterisk italorossi (~textual@189.124.196.68) |
| 17:47.06 | SeRi | Than check for phone options to see if call waiting is turned off. |
| 17:47.51 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-ddijliodzksrktqi) |
| 17:51.09 | d_preston215 | Call waiting is off. |
| 17:53.54 | *** join/#asterisk brettnem (~brett@76-216-205-249.lightspeed.austtx.sbcglobal.net) |
| 17:54.50 | brettnem | Hey all, can anyone recommend a 1RU box for a small pbx install (20-30 concurrent, no recordings or transcoding) :) |
| 17:55.19 | jpsharp | The cheapest thing you can get your hands on? |
| 17:55.55 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 17:56.07 | brettnem | sure.. as long as it doesn't immediately catch on fire and kill everyone |
| 17:56.11 | brettnem | :) |
| 17:56.38 | SeRi | d_preston215: Than thats your problem |
| 17:56.52 | jpsharp | Pretty much. Anything more than a Pentium 133 with that call load will look at you and say "Well, when am I getting real work?" |
| 17:57.13 | brettnem | 133.. haha |
| 17:57.19 | nicknam12322 | : - ) |
| 17:57.38 | d_preston215 | I know. I just wondered if anyone ever had that going in that way. |
| 17:57.53 | slav3_kitten | HUZZAH! |
| 17:58.07 | brettnem | I was considering a super micro atom based server.. I actually don't mind paying a little more for something that isn't crap. I'd like to be able to install a modern os (not stripped down) on it.. |
| 17:58.08 | jpsharp | Someone beat his 7940 into submission. |
| 17:58.24 | slav3_kitten | i overlooked the simple line of "register with proxy" in the damn config for hours |
| 17:58.29 | drmessano | SuperMicro Atom box will handle that load just fine |
| 17:58.41 | jpsharp | yep. What drmessano said. |
| 17:58.47 | slav3_kitten | jpsharp, nah all the 79x0's were beat into submission last night early on |
| 17:58.53 | d_preston215 | lol |
| 17:58.56 | drmessano | I dare say it's almost overkill |
| 17:59.00 | slav3_kitten | it's the 7911g that was beating me up |
| 17:59.28 | drmessano | I've put the same number of calls on lesser hardware and had no problems. |
| 17:59.39 | slav3_kitten | it's not perfect but it makes a call, can't seem to get the ntp to work |
| 17:59.40 | drmessano | BUT DONT |
| 17:59.54 | drmessano | slav3_kitten: are you using CNTP |
| 18:00.04 | d_preston215 | I'm really starting to hate these 7960s again. |
| 18:00.06 | brettnem | drmessano: yeah, same here.. If I can keep it under $500 and make it look respectable, that'd be great :) |
| 18:00.29 | slav3_kitten | cntp? |
| 18:00.58 | drmessano | CNTP: Where all time is referenced to July 22, 1982, all traffic is on port 65540, and you need a CNTP license from Cisco to even look at your watch to set the time on the server |
| 18:01.31 | slav3_kitten | shouldn't be.. but i honestly am unsure |
| 18:01.40 | drmessano | </joke> |
| 18:02.05 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 18:02.23 | brettnem | So I'm really not trying to be lazy.. I can get a 1ru barebones supermicro and piece it together.. but is there a good canned setup out there anyone can recommend? Anything will likely be overkill |
| 18:02.27 | drmessano | Common format .. Take simple tech, add a letter, some unusal configuration, and throw in licensing, and you have Cisco's variant |
| 18:02.47 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
| 18:03.04 | slav3_kitten | stares evil like at drmessano |
| 18:03.21 | d_preston215 | lol |
| 18:04.34 | drmessano | Wait until you try CSIP. How you know your IPX/SPX |
| 18:04.58 | drmessano | Crap, now I need to mail Cisco a check for even mentioning it |
| 18:05.06 | *** join/#asterisk theron (~Theron@ip66-43-220-25.static.ishsi.com) |
| 18:08.38 | slav3_kitten | hmm maybe i have something sideways on my configuration |
| 18:16.47 | slav3_kitten | why would a phone not register with asterisk? |
| 18:17.27 | *** join/#asterisk Micc_ (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 18:18.02 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 18:23.14 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-nwgngbnlajrzymuj) |
| 18:26.12 | SeRi | drmessano: lol |
| 18:26.46 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-pxdukimojjrmvdvz) |
| 18:31.13 | *** join/#asterisk nicknam12322 (021d145d@gateway/web/freenode/ip.2.29.20.93) |
| 18:32.12 | jeffspeff | slav3_kitten, because you're holding it wrong |
| 18:32.28 | slav3_kitten | jeffspeff, i mean seriously |
| 18:33.55 | slav3_kitten | http://pastie.org/private/sp2gxtbdwm7whcaiaee7q |
| 18:33.58 | jeffspeff | bad auth, bad digest, bad realm, realm doesn't match digest realm in auth, bad server setting on the phone, phone set to use auth_id but auth_id is not set, maybe outbound proxy settting on the phone, asterisk isn't listening on the right network, asterisk isn't listening on the right port, nat issues, localnet not defined |
| 18:34.18 | slav3_kitten | that's what i'm getting transaction wise. phone is just sitting there saying registering |
| 18:34.48 | jeffspeff | slav3_kitten, this is what i get when i view that page |
| 18:34.50 | jeffspeff | Application error |
| 18:34.50 | jeffspeff | Change this error message for exceptions thrown outside of an action (like in Dispatcher setups or broken Ruby code) in public/500.html |
| 18:35.00 | slav3_kitten | :/ |
| 18:35.11 | jeffspeff | wati, 500 refreshes, and it worked |
| 18:35.59 | slav3_kitten | sorry, i won't use that site again. anyhow can you see the problem off the bat? |
| 18:37.01 | jeffspeff | the from and to addresses are the same |
| 18:37.07 | jeffspeff | that doesn't seem right |
| 18:37.46 | jeffspeff | so, is 192.168.1.2 your asterisk box and .64 your phone? |
| 18:37.53 | slav3_kitten | 1.2 is asterisk |
| 18:37.57 | slav3_kitten | .64 is phone |
| 18:39.03 | slav3_kitten | it's gotta be something in the phone config |
| 18:39.31 | jeffspeff | the phones sip user, is it "test" or "livingroom" ? |
| 18:39.56 | slav3_kitten | err test |
| 18:40.05 | slav3_kitten | thought i sanitized that post better |
| 18:40.08 | jeffspeff | Contact: <sip:livingroom@192.168.1.64:5060;transport=udp>;expires=3600 |
| 18:40.10 | slav3_kitten | whoops |
| 18:40.31 | slav3_kitten | yea that was me missing to replace an instance of livingroom with test |
| 18:40.41 | jeffspeff | oh, o |
| 18:40.46 | jeffspeff | *ok |
| 18:40.59 | slav3_kitten | basically to and from should not be the same |
| 18:41.07 | slav3_kitten | that's what i'm kinda seeing |
| 18:41.17 | jeffspeff | no, they really shouldn't lol |
| 18:41.35 | jeffspeff | check your outbound proxy settings and auth_id settings and that stuff |
| 18:45.16 | slav3_kitten | i found that the ntp settings will be ignored if it can't find the locale files. and i can't find any information on how to create them so that explains that. |
| 18:45.32 | slav3_kitten | supposedly if i get it to register it'll set the time from the 200 response |
| 18:49.43 | slav3_kitten | are you freaking kidding me |
| 18:50.31 | slav3_kitten | apparently the phone was like "hell no, i'm not pulling down the latest config, i'll use the old one that i have from when you managed to recover me from the dead |
| 18:50.43 | slav3_kitten | so factory reset solved the problem |
| 18:51.23 | slav3_kitten | down side: i have no idea why it refused to pull the updated config |
| 18:51.40 | nicknam12322 | : - ) |
| 18:52.15 | slav3_kitten | up side: factory reset is going to happen any time the phone acts up after changing a config |
| 18:53.37 | slav3_kitten | jeffspeff, the from and to lines are still the same .1.2 address but it registers now |
| 18:54.13 | jeffspeff | that's odd |
| 18:54.33 | slav3_kitten | yep, i got no decent answer other then i hate the 7911 |
| 19:06.31 | slav3_kitten | it mostly works, i'm going to call it a win |
| 19:11.36 | *** join/#asterisk af_ (~getsmart@78.134.25.32) |
| 19:13.14 | slav3_kitten | huzzah all phones are working i think |
| 19:14.33 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:17.46 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 19:27.20 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 19:30.46 | slav3_kitten | ok they all register an call the hello world extension |
| 19:31.02 | slav3_kitten | i'm calling that a huge victory and treating myself to some cancer sticks |
| 19:31.20 | nicknam12322 | Interesting. Nice. |
| 19:35.29 | nicknam12322 | Leaving channel. |
| 19:35.33 | nicknam12322 | Bye - all. |
| 19:35.42 | nicknam12322 | Nice discussion. Thanks. |
| 19:43.05 | *** join/#asterisk navaismo (~navaismo@189.241.62.150) |
| 19:53.35 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 20:06.40 | *** join/#asterisk nobodyshome (~dijib@208-96-84-35.eastlink.ca) |
| 20:07.16 | chris_n | would a remote gateway responding with a 503 rather than a 200 to a qualify from * cause * to drop registration of the gw? |
| 20:07.32 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 20:07.56 | *** join/#asterisk b3nt_pin (~quassel@stjhnf0148w-142134079200.dhcp-dynamic.FibreOP.nl.bellaliant.net) |
| 20:08.27 | [TK]D-Fender | Qualify has nothing to do with registration |
| 20:08.59 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 20:09.44 | pzn | I'm a newbie in configuration... I just bought an grandstream gxs4108 and will connect my incoming analog lines on it. any hint about how to configure this in asterisk? |
| 20:09.47 | chris_n | [TK]D-Fender: thanks |
| 20:11.28 | [TK]D-Fender | pzn, Plenty of guides out there already withing 5 seconds of a google search |
| 20:11.34 | [TK]D-Fender | pzn, Including : http://www.asteriskguru.com/tutorials/gxw4108.html |
| 20:14.21 | pzn | [TK]D-Fender, I already read this link without success... what is 1-stage dial or 2-stage dial? I think an incoming call reaches FXO, then it re-dials some internal extension? I'm too newbie to know some caveeats of the text |
| 20:15.39 | [TK]D-Fender | You should probably be showing your actual attempts to set it up along with console output with SIP debug, etc... |
| 20:20.51 | pzn | [TK]D-Fender, what I could do until now is this: one FXO line is registered in asterisk/sip. when an incoming call reaches the line, the phone rings once, then I hear one "click" like it is answering the call, then I heard an line audio tone and that is it. no internal extension rings... |
| 20:21.56 | [TK]D-Fender | pzn, And we see none of this. |
| 20:22.12 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 20:22.44 | pzn | [TK]D-Fender, you mean that I should post the output of "asterisk -rvvvv" ? |
| 20:22.54 | [TK]D-Fender | clearly |
| 20:23.15 | [TK]D-Fender | show us real calls with real debug so we can see what's happening |
| 20:26.01 | SeRi | pzn: sip set debug on |
| 20:26.15 | pzn | [TK]D-Fender, got how to discover the problem! I will pastebin it... |
| 20:31.08 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 20:31.27 | *** join/#asterisk artyx (U2FsdGVkX1@junction.googleplex.net) |
| 20:34.23 | pzn | here is the information... http://pastebin.com/8DJxZEXg can anyone help me? |
| 20:35.15 | SeRi | pzn: Thats not what [TK]D-Fender ask you for |
| 20:35.22 | *** join/#asterisk Galen (~Galen@rrcs-76-79-170-42.west.biz.rr.com) |
| 20:35.47 | SeRi | he ask for debuging information |
| 20:36.02 | SeRi | in astrisk cli do sip set debug on |
| 20:36.06 | SeRi | make the call |
| 20:36.11 | SeRi | sip set debug off |
| 20:36.25 | SeRi | anc pastebin the output |
| 20:36.30 | SeRi | s/anc/and/ |
| 20:39.13 | artyx | lol |
| 20:39.40 | pzn | SeRi, [TK]D-Fender, I think my question is more about GXW4108 than about asterisk. I'll go home... read its manual with attention and try again tomorrow. If I have problems, then I came back with a more elaborate question with debugs too. thanks folks |
| 20:39.59 | *** join/#asterisk BenC[UK] (~bcummins@cpc10-lock3-2-0-cust14.6-1.cable.virginmedia.com) |
| 20:40.23 | BenC[UK] | hi guys, don't suppose anyone can think of a reason why every now and again my server seems to ignore anything send on the management interface? |
| 20:40.40 | BenC[UK] | its been working fine for months, and now 10 times since friday I've had to restart asterisk |
| 20:41.37 | BenC[UK] | I see nothing in the console when its down.. full.log is showing this: http://pastebin.com/E3ZxndHA |
| 20:42.33 | BenC[UK] | if I type "reload" when its down, it does nothing |
| 20:43.25 | slav3_kitten | hmm my cat seems to be stuck in hunter mode |
| 20:45.50 | *** join/#asterisk v4x (~v4x@unaffiliated/v4x) |
| 20:46.26 | BenC[UK] | my st bernard is squaring up to a jack russel at the moment :D |
| 20:47.45 | slav3_kitten | that won't end well |
| 20:47.53 | jpsharp | I just have one cat asleep on the floor and one on top of the laser printer. |
| 20:48.27 | slav3_kitten | jpsharp, yea we have that problem with out cats an the printer |
| 20:48.39 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:49.44 | jpsharp | I suspect 4 of the other 5 are out on the back porch and the last one is asleep under my bed. |
| 20:51.07 | *** join/#asterisk ag4ve (~ag4ve@96.26.67.194) |
| 20:54.08 | *** join/#asterisk ag4ve (~ag4ve@96.26.67.194) |
| 20:55.42 | BenC[UK] | we're lookign after gf's dog.. it snot going well! |
| 20:56.29 | BenC[UK] | gfs sisters dog even |
| 20:56.35 | BenC[UK] | put them back in their cages now |
| 20:56.39 | BenC[UK] | that was… fun |
| 20:58.44 | BenC[UK] | anyway, back to my original issue... |
| 20:58.57 | BenC[UK] | Anyone got any suggestions as to what I can try to find out whats going on? |
| 21:05.41 | SeRi | BenC[UK]: what version of *? |
| 21:06.02 | BenC[UK] | Asterisk 1.8.7.2 |
| 21:06.20 | SeRi | quick google: https://www.google.com/search?sugexp=chrome,mod=0&sourceid=chrome&ie=UTF-8&q=audiohook.c%3A+Read+factory |
| 21:06.40 | SeRi | Seems to be a few issues out there |
| 21:06.49 | SeRi | looks like related to MixMonitor |
| 21:07.42 | [TK]D-Fender | BenC[UK], You are 10 releases behind current in your branch.... |
| 21:08.03 | BenC[UK] | Just don't understand why its broken now |
| 21:08.09 | BenC[UK] | its been working fine for months |
| 21:08.52 | BenC[UK] | im going to try deletng some of the history |
| 21:09.01 | BenC[UK] | the call recordings |
| 21:09.06 | BenC[UK] | got 90,000 of them |
| 21:12.44 | [TK]D-Fender | That's also only part of ONE SECOND worth of logs... |
| 21:13.40 | BenC[UK] | in the link I send TK ? |
| 21:13.59 | BenC[UK] | I have several minutes of that |
| 21:14.03 | BenC[UK] | before I restart asrterisk |
| 21:14.09 | BenC[UK] | and its all fine again for a 1-4 hours |
| 21:42.18 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 21:44.57 | artyx | So ive defined a dahdi trunk channel 1 with a cid of (number), i can use the trunk to dial out no problem. |
| 21:45.21 | artyx | When dialing in, i get ssnoservice, its currently mapped to a user extension, the user's phone is registered and i can call the users phoen manually |
| 21:45.39 | artyx | If i remove the did (match everything), the call goes tot their phone, but never picks up |
| 21:46.21 | artyx | <PROTECTED> |
| 21:52.21 | artyx | Th eproblem im having is the did not matching |
| 21:53.48 | drmessano | Thanks for getting 11.0.1 out so quiet, guys! |
| 21:53.53 | drmessano | GAH |
| 21:53.56 | drmessano | Thanks for getting 11.0.1 out so quick, guys! |
| 21:54.01 | drmessano | Mind auto complete fail |
| 22:00.09 | artyx | extensions_additional.conf:exten => s,1,Noop(Entering from-dahdi with DID == ${DID}) |
| 22:12.27 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 22:15.08 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:15.09 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:36.01 | *** join/#asterisk navaismo (~navaismo@189.241.62.150) |
| 22:40.47 | SeRi | dijib: you around? |
| 22:55.45 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
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| 23:00.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:05.55 | slav3_kitten | hmmm so bee reading (dangerous i know) but passing through nat maybe i should look for an IAX did provider with good international rates to england and the US |
| 23:06.02 | slav3_kitten | been even |
| 23:06.53 | SeRi | not many out there. |
| 23:07.23 | SeRi | voip.ms is the only one that I know that has IAX2 Trunks |
| 23:07.35 | slav3_kitten | sad |
| 23:08.18 | WIMPy | IAX seems to have gone out of fashion :-( |
| 23:09.37 | slav3_kitten | it only used one port though... |
| 23:10.00 | slav3_kitten | voip.ms is spendy, 12 cents a minute |
| 23:10.21 | SeRi | wow 12 cents. |
| 23:10.39 | BenC[UK] | hmm, its not that much |
| 23:10.49 | SeRi | are you sure you are reading it right? |
| 23:10.52 | SeRi | :P |
| 23:10.54 | BenC[UK] | I am usin git |
| 23:11.02 | BenC[UK] | using it |
| 23:11.11 | BenC[UK] | calling from where to where? |
| 23:11.29 | slav3_kitten | SeRi, it is for 447748 |
| 23:12.14 | BenC[UK] | thats a mobile number though.. |
| 23:12.31 | slav3_kitten | and the number i'm going to be calling most often |
| 23:13.23 | BenC[UK] | get them to get a voip client :d |
| 23:14.34 | slav3_kitten | anveo is 7.5 cents/min |
| 23:15.27 | slav3_kitten | didforsale is 7.1 |
| 23:16.44 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:16.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 23:16.54 | WIMPy | But on a mobile thay should be rather standard unless branded away. |
| 23:17.20 | slav3_kitten | WIMPy, iirc her mobile phone is a non smart phone |
| 23:17.35 | slav3_kitten | i know my mobile phone gets calls, no text no data |
| 23:17.42 | WIMPy | Then you know what a good X-mas present would be. |
| 23:17.57 | BenC[UK] | what about getting her to call a local number her end to goto voip :P |
| 23:18.15 | WIMPy | Not a bad idea. |
| 23:19.40 | BenC[UK] | siptalk.co.uk I think in the uk gives free inbound |
| 23:20.18 | WIMPy | Are there ones that don't? |
| 23:20.32 | slav3_kitten | hmm that might be an option |
| 23:21.13 | slav3_kitten | still it'd be nice to not get an ass pounding @ 12c/min when i want to call her. 7.1 or 7.5 is doable |
| 23:22.36 | slav3_kitten | ok BenC[UK] what's the difference between a geographical and non geographical uk number? |
| 23:23.09 | BenC[UK] | geographical means it relates to the area you're calling |
| 23:23.19 | BenC[UK] | for example, 02380 is a Southampton, UK number |
| 23:23.39 | BenC[UK] | non-geographical are numbers for like call centers etc, and are like 0845 or 0870 |
| 23:24.29 | BenC[UK] | but it could go anywhere... |
| 23:25.15 | slav3_kitten | so which would be a better did to get for her to be calling me on? the uk rate thing is quite confusing since in the states we pretty much have no distinction between mobile numbers an the such |
| 23:25.49 | BenC[UK] | geographic |
| 23:26.04 | BenC[UK] | most of the time, they are included in the included minutes of the contract |
| 23:26.30 | BenC[UK] | yeah, I find that strange about the states |
| 23:27.12 | slav3_kitten | it's simpler for us old people terrified of changeo |
| 23:27.17 | slav3_kitten | change* |
| 23:27.23 | BenC[UK] | I run a few call centers in the states, and we collect phone numbers from leads providers… and we want to drop text messages for customers we can't reach, but we end up sending to pstn etc because we can't tell the difference\ |
| 23:28.01 | WIMPy | That's not supported there? |
| 23:28.03 | slav3_kitten | lol i have text services disabled on my cell |
| 23:28.54 | BenC[UK] | WIMPy: you mean text->pstn? |
| 23:29.00 | BenC[UK] | most networks seem to reject it |
| 23:29.13 | WIMPy | yes |
| 23:29.37 | BenC[UK] | in the UK, we get a computer generated voice... |
| 23:29.40 | slav3_kitten | we are still the best country on the planet :P |
| 23:30.30 | WIMPy | Most dect handsets support text messages as well. |
| 23:31.10 | slav3_kitten | oh i have an odd question actually |
| 23:31.36 | slav3_kitten | starts with) does anyone have experience with wifi sip phones |
| 23:32.13 | WIMPy | Yes. Nice toy when you're on your way. But not something I'd use on a daily basis. |
| 23:33.01 | slav3_kitten | and then b) how does dedicated voice lan work with wifi phones |
| 23:33.03 | BenC[UK] | I have dect phone which has SIP support |
| 23:33.33 | slav3_kitten | i'd assume you'd have to make a virtual access point on the voice vlan for the tagged packets to go to the right places |
| 23:34.03 | WIMPy | The ones I've seen are rather basic. |
| 23:34.25 | WIMPy | Better go for dect/cat-iq. |
| 23:35.22 | slav3_kitten | i got an old cisco 7921g in rough but working condition |
| 23:35.44 | slav3_kitten | but after the nightmare getting that 7911 to work right i'm kinda afraid of the damn thing |
| 23:37.16 | BenC[UK] | I had a cisco 7960 but just sold it, as I found it a right mare to get working right |
| 23:37.21 | BenC[UK] | was perfect when it was working properly |
| 23:37.50 | slav3_kitten | is a right mare the british version of nightmare? |
| 23:38.08 | BenC[UK] | yeah, :) |
| 23:38.20 | BenC[UK] | well.. its slang |
| 23:38.29 | slav3_kitten | but yea i spent the last 3 days having a small war with 4x 7940s, 1x7960, and 1x 7911 |
| 23:38.52 | slav3_kitten | the 7911 took a whole day to hammer out as it bricked itself during a firmware load |
| 23:39.29 | slav3_kitten | then i found that it's a bug with bad timings on the 7911 but not the 7906 which uses the same firmware |
| 23:40.16 | slav3_kitten | and as soon as i find my damn crimpers i'll make up a few cables and connect the phones after setting up the ports on the switch |
| 23:44.35 | BenC[UK] | all good fun :D |
| 23:44.42 | BenC[UK] | I've found the ciscos to have the best voice quality |
| 23:44.53 | BenC[UK] | especially when it comes to loud speaker |
| 23:46.19 | slav3_kitten | well that's a small bonus for the anguish of setting them up |
| 23:48.13 | slav3_kitten | that doesn't look right... works though but it lookks off |
| 23:49.25 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-htumdwsvpbcmvfsj) |
| 23:52.49 | *** join/#asterisk Nivex (~kjotte@atlantis.home.nivex.net) |
| 23:54.09 | Nivex | realising "sucks to be you" is a likely answer, I'll ask around here anyway: I bought a SPA-2000 at a hamfest and it still seems to be locked to voicepulse. Anyone have an admin password I can try to perform the full factory reset? |
| 23:58.14 | BenC[UK] | tried this? http://forum.voxilla.com/threads/factory-reset-spa-2000.15692/ |
| 23:59.26 | Nivex | BenC[UK]: oh yes, I've been over many forum posts. The problem is that where it says "Press 1 to confirm" I get "Enter password" instead |