| 00:00.35 | fujin | Innatech: kind of |
| 00:00.48 | flenders | Innatech: traffic will still go through a router if you have different subnets |
| 00:00.52 | fujin | yes |
| 00:00.57 | fujin | you'll still need a router between the subnets |
| 00:01.09 | flenders | get gigabit ports and gigabit NICs on the router and you're set |
| 00:01.11 | fujin | (or a virtual interface in the subnets) |
| 00:01.40 | flenders | still goes through router |
| 00:02.14 | fujin | hardly need gigabitz for VoIP. |
| 00:02.26 | fujin | maybe for trunks between points of presence |
| 00:02.41 | flenders | I meant for data too |
| 00:02.45 | Innatech | OK....so I can't plug phones on 192.168.55.x into the same switch as desktops on 192.168.50.x and provide uplinks to a router (192.168.50.1) and to Asterisk (192.168.55.2) on separate ports? All traffic would need to pass through the router? |
| 00:03.15 | fujin | uh |
| 00:03.19 | fujin | are you planning a network? |
| 00:03.25 | fujin | or trying to add things to an existing one |
| 00:03.29 | Innatech | new network. |
| 00:03.31 | fujin | ok |
| 00:03.36 | fujin | start drawing some pictures |
| 00:03.38 | fujin | :) |
| 00:03.40 | Innatech | oh, I have some. |
| 00:03.52 | Innatech | In fact, I have 2^3 . |
| 00:04.05 | flenders | Innatech: you can plug all them on the same switch |
| 00:04.05 | Innatech | covered all of the permutations. :) |
| 00:04.35 | flenders | Innatech: but you need different VLANs on the switch to achieve what you're describing |
| 00:05.17 | flenders | the throughput on modern switches is great, so you wouldn't have problems... |
| 00:05.22 | fujin | got access to reasonable hardware, Innatech ? |
| 00:05.22 | Innatech | OK, cool. That much makes sense. |
| 00:05.23 | fujin | ciscoz..? |
| 00:05.31 | Innatech | Nah, not that reasonable. |
| 00:05.35 | flenders | yeah, or HP procurves |
| 00:05.39 | fujin | lol |
| 00:06.20 | fujin | or not :P |
| 00:06.25 | Innatech | Switches are going to be one of the more modest lines---probably 3com or dlink smart switches. |
| 00:06.30 | Innatech | The routers will be Linux. |
| 00:06.43 | flenders | even those have pretty decent throughputs |
| 00:06.43 | fujin | cool |
| 00:06.46 | Innatech | 4 or 6 PCI-E GbE NICs each. |
| 00:06.47 | Kwakwa | I've been having trouble setting up rxfax on one of my asterisk boxes on 1.4.5. Its mainly used for IAX2 trunking so if I install 1.2.20 on it instead will it be fine communicating over IAX2 to my other 1.4.* boxes? |
| 00:07.11 | fujin | we should just get rid of faxes altogether |
| 00:07.13 | Kwakwa | I'm assuming all the IAX2 fixes are backported to 1.2 also? |
| 00:07.21 | fujin | I wouldn't. |
| 00:07.26 | flenders | my routers here are dells 1850 with 6 Gbit cards each |
| 00:08.07 | Kwakwa | At the moment we have two fax modems on analogue using hylafax, I'd be happy getting this working with asterisk :) |
| 00:08.27 | fujin | I run cisco 3560's for switchfabric |
| 00:08.42 | Innatech | I'm building these from Commel mainboards. I'm looking forward to that part of the project. |
| 00:08.42 | fujin | I think that our network dudes put in a 2950 or a 2950g or xl for the router |
| 00:11.50 | flenders | Innatech: linksys managed switches are quite decent |
| 00:12.00 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 00:12.02 | flenders | better than D-Links, IMHO |
| 00:12.08 | Innatech | yeah, I've heard their config pages only work reliably in IE. |
| 00:12.12 | Innatech | have you seen that? |
| 00:12.36 | snuff-work | i hate linksys web managed switches |
| 00:12.40 | jcaceres | hello whe i do "sip show users" i get field called ALC what does it stands for? |
| 00:12.51 | snuff-work | ACL = access control list |
| 00:13.13 | jcaceres | thanks |
| 00:14.31 | k31th | any of you guys know of a decent asterisk distro for a solid state PBX im attempting to build? |
| 00:15.00 | fujin | ubuntu |
| 00:15.25 | Innatech | OK, so my takeaway here is that I can go with a single managed PoE switch, VLAN the VOIP and standard traffic, connect both the Asterisk server and the regular LAN servers to the switch and the back router will only worry about passing traffic between the back LAN subnet and the firewall subnet. The phones will see the Asterisk box across the VLAN/switch w/o routing since they share a subnet. Same thing with the dekstops and the ser |
| 00:15.25 | Innatech | vers. Sound about right? |
| 00:15.54 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 00:15.57 | flenders | Innatech: you can telnet to them I think |
| 00:16.31 | Innatech | Umm. What? |
| 00:17.21 | flenders | Innatech: yeah, that's right |
| 00:17.30 | flenders | I was talking about the linksys switches before |
| 00:17.31 | Innatech | Oh, telnet to the linksys |
| 00:17.33 | Innatech | yeah, gotcha. |
| 00:17.36 | Innatech | thanks. |
| 00:17.39 | flenders | hey, I also have a Dell managed switch here... |
| 00:17.48 | flenders | the OS is VERY similar to cisco's ISO |
| 00:17.54 | flenders | cisco's IOS |
| 00:17.58 | snuff-work | hehe.. dell loves rebranding |
| 00:18.06 | flenders | and you run ssh on them too |
| 00:18.11 | Innatech | Dell is a frankenstein of rebadged products. |
| 00:18.24 | flenders | well, they're cheap |
| 00:18.34 | flenders | and you made it clear you're not willing to pay for a cisco |
| 00:18.35 | snuff-work | dells SAN switches are brocade from memory.. |
| 00:18.53 | Innatech | Heh. Then there's my spec for the "subtenant" AKA "legacy" network: nothing but DD-WRT's, bay-be! |
| 00:19.12 | Innatech | yeah, I'm looking at Dell too. |
| 00:19.19 | fujin | openwrt! |
| 00:19.25 | Innatech | Is it better? |
| 00:19.29 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:20.04 | Innatech | I'm ashamed to admit I still have alchemy firmware kicking around. |
| 00:20.14 | flenders | Innatech: they're pretty much the same... DD-WRT and openwrt |
| 00:20.21 | fujin | hardly |
| 00:20.26 | fujin | dd-wrt still has that webui and shit |
| 00:20.31 | Innatech | Well, there must be some reason for the fork, eh? |
| 00:20.39 | flenders | but you can also ssh into it |
| 00:20.44 | fujin | yeah true |
| 00:20.51 | fujin | always had better luck with openwrt though |
| 00:21.02 | jcaceres | how can use an acl, where do i configure it? |
| 00:21.11 | holiday_42 | with openwrt they make i really easy to roll you own customer firmware |
| 00:21.16 | jcaceres | i have done it with cisco |
| 00:21.18 | Innatech | interesting. |
| 00:21.40 | jcaceres | but in asterisk sounds me new |
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| 00:23.06 | *** join/#asterisk yotta_ (n=yotta@adsl-69-236-170-102.dsl.pltn13.pacbell.net) |
| 00:24.43 | yotta_ | Hi, I'm looking for a way to get 4 or 8 fxo ports, perferable in a device that works over ethernet. anyone have reccomendations? |
| 00:25.11 | fujin | get a PRI |
| 00:25.12 | fujin | :D |
| 00:25.17 | __DAW | Audiocodes MP118 |
| 00:28.26 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:34.24 | snuff-work | analog is yucky ;) |
| 00:35.10 | snuff-work | yotta_, you could also look at astribank i think they are called.. |
| 00:35.44 | yotta_ | well |
| 00:35.59 | yotta_ | I want something that I can use from two asterisk servers with hot failover. |
| 00:36.43 | fujin | ho ho |
| 00:36.48 | fujin | I just finished setting that up yesterrrday |
| 00:36.56 | fujin | I still have a single point of failure though |
| 00:36.57 | fujin | or a few |
| 00:37.01 | fujin | no, just one |
| 00:37.02 | fujin | ;] |
| 00:37.04 | jcaceres | hello, what is the diference betwen an URA and an IVR? |
| 00:37.21 | fujin | yotta_: I use heartbeat v1 to hot/cold my asterisks. |
| 00:37.39 | yotta_ | fujin: I use heartbeat |
| 00:37.44 | yotta_ | on some servers at work |
| 00:37.46 | yotta_ | it rocks. |
| 00:37.55 | yotta_ | anyway |
| 00:38.12 | fujin | I'm just goin to get two PRI's for the as5400 |
| 00:38.17 | yotta_ | i want one piece of hardware that i can hook up to two asterisk boxes |
| 00:38.18 | fujin | and then work out some cisco loadbalancin. |
| 00:38.20 | yotta_ | don't need pri |
| 00:38.32 | yotta_ | bossman doesn't want pri |
| 00:38.36 | fujin | sorry then I dunno anything <pri |
| 00:38.47 | fujin | pri is proper good man |
| 00:39.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:39.15 | yotta_ | yeah |
| 00:39.20 | yotta_ | but expensive |
| 00:39.29 | yotta_ | we need only 4 to 6 phone lines |
| 00:39.37 | yotta_ | probably 4 |
| 00:41.19 | flenders | yotta_: in AU, it makes more sense to get a 10 channel PRI than 6 pots lines |
| 00:41.37 | yotta_ | in us |
| 00:41.39 | flenders | a full PRI would be too much, I agree |
| 00:41.52 | flenders | but check the prices for fractional PRIs |
| 00:42.05 | yotta_ | that involves calling people |
| 00:42.40 | flenders | yeah |
| 00:42.41 | flenders | so? |
| 00:42.56 | flenders | getting POTS lines also involves people |
| 00:43.07 | flenders | and buying lunch every day also involves people |
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| 00:46.57 | yotta_ | we have pots lines already |
| 00:47.44 | flenders | seriously, PRI is A LOT better than pots lines |
| 00:48.12 | flenders | one of my boxes has 2 TDM400s with 4 FXO modules on each. nightmare |
| 00:48.40 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 00:48.52 | flenders | the other one has a 10ch PRI... sooo much better |
| 00:49.44 | fujin | yea |
| 00:49.47 | fujin | I break my pri out to sip |
| 00:49.49 | fujin | with an as5400 |
| 00:49.53 | fujin | saves a whole motherfucker |
| 00:49.54 | fujin | of trouble |
| 00:50.40 | flenders | so, with heartbeat, when it fails, it just STONITH, and bring the other asterisk up? and as it's all sip, it's easy, right? |
| 00:53.26 | *** join/#asterisk friedrich| (n=friedric@e177244164.adsl.alicedsl.de) |
| 00:53.38 | fujin | flenders: eh, I don't even fuck with STONITH |
| 00:53.46 | fujin | it just does /etc/init.d/* stuff. |
| 00:53.55 | fujin | never have come across a situation when i need to stonith. |
| 00:54.15 | flenders | so it just shuts down asterisk on the problematic box |
| 00:59.12 | flenders | fujin: and what's your point of failure? |
| 01:02.08 | flenders | holy mother of god... I had never checked the prices on an AS5400 before |
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| 01:31.27 | Hmmhesays | does asterisk store the originally dialed number as a constant somewhere? |
| 01:32.15 | [TK]D-Fender | CDR |
| 01:32.41 | Hmmhesays | [TK]D-Fender: doesn't that change if the exten changes? |
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| 01:39.12 | [TK]D-Fender | Hmmhesays, not the originally dialed number |
| 01:39.27 | tengulre | anybody know which g729 codecs is free and not connected limited? |
| 01:39.31 | [TK]D-Fender | brb |
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| 01:40.55 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 01:43.34 | Hmmhesays | what is a good way to check if a variable is null, if it isn't then check what the value is set to |
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| 01:54.27 | fujin | flenders: the as5400 |
| 01:54.44 | fujin | flenders: yes, if the box stops responding ove rthe c/over cable |
| 01:54.47 | fujin | it'll shut down asterisk |
| 01:54.51 | fujin | dnsmasq (dhcp) |
| 01:54.59 | fujin | and some other things.. all configurable |
| 01:55.41 | dlynes_laptop | ~t38 |
| 01:55.42 | jbot | t38 is probably see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 01:56.38 | alrs | Hmmhesays: NoOp |
| 01:57.08 | flenders | so, second asterisk comes up with the same IP as the first one, and all phones re-register? |
| 01:57.52 | fujin | yes |
| 01:57.55 | fujin | it aliases the same ip |
| 01:57.59 | fujin | does an arp broadcast |
| 01:58.06 | fujin | the phones are configured to resync every 20 secs |
| 01:58.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:03.04 | flenders | wonder what I'd need to do to get heartbeat to work with asterisk+PRI cards |
| 02:03.43 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 02:03.45 | fujin | a mission |
| 02:03.47 | fujin | fuck that |
| 02:03.54 | fujin | you gotta have somethign upstream from asterisk to reroute calls |
| 02:03.56 | RyanW | How can i log pri debug out to a file ? |
| 02:04.05 | fujin | see, our as5400 is told to point all sip calls @ 192.168.108.1 |
| 02:04.08 | fujin | which is *always* one of the boxes |
| 02:04.08 | flenders | problem would be to have both cards connected to the same CSU/DSU |
| 02:04.48 | flenders | yeah, but an as5400 costs shit loads of money |
| 02:07.03 | *** part/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 02:08.07 | fujin | heh |
| 02:08.09 | fujin | gotta spend it to make it |
| 02:08.26 | __DAW | is there anything for SIP like zapbarge? |
| 02:08.44 | fujin | what's that do? |
| 02:08.49 | Hmmhesays | chanspy |
| 02:09.24 | __DAW | chanspy |
| 02:09.27 | __DAW | thanks |
| 02:09.38 | Hmmhesays | [TK]D-Fender: I'm not seeing that variable |
| 02:10.26 | [TK]D-Fender | Hmmhesays, its a FUNCTION. |
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| 02:16.22 | *** join/#asterisk dexpdx (n=dex@66-162-134-242.static.twtelecom.net) |
| 02:17.09 | dexpdx | Hey, I just upgraded from 1.2.13 to 1.2.20 and now I'm getting "pri_find_dchan: No D-channels available!" messages |
| 02:17.12 | dexpdx | is this normal? |
| 02:17.56 | Hmmhesays | [TK]D-Fender: oooooh |
| 02:18.16 | shmaltz | dexpdx, no |
| 02:18.24 | Hmmhesays | [TK]D-Fender: CDR gets or sets a cdr variable... |
| 02:18.27 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:18.27 | dexpdx | Will it cause problems? |
| 02:18.30 | shmaltz | dexpdx, if you doh't have any dchannels then you don't have a PRI |
| 02:18.30 | Hmmhesays | how does that help me |
| 02:18.37 | dexpdx | I have pri's :) |
| 02:18.39 | shmaltz | dexpdx, update zaptel as well |
| 02:18.42 | dexpdx | wanrouter see's them |
| 02:18.52 | dexpdx | shmaltz: already done, latest stable |
| 02:19.28 | kolian123 | Hello |
| 02:19.43 | dexpdx | oh weird |
| 02:19.54 | dexpdx | Primary D-channel: 24 |
| 02:19.55 | dexpdx | Status: Provisioned, Down, Active |
| 02:20.14 | kolian123 | Does anybody know what to set in zaptel config for echocancel for the card with hardware cancellation? |
| 02:24.34 | shmaltz | kolian123, yes, nothing |
| 02:25.09 | kolian123 | hi shmaltz, thanks, just omit it or set to =no? |
| 02:25.29 | shmaltz | kolian1234, you want to disable it? |
| 02:26.08 | kolian123 | shmaltz, i would like to use hardware and disable software...what would you recommend? |
| 02:26.52 | shmaltz | kolian123, if you have the hardware module, then it uses the hardware and NOT the software, no need to disable that |
| 02:27.39 | kolian123 | Alright, thanks so it would use it automatically? |
| 02:27.45 | dexpdx | ok this is weird, when I do a pri show span 1 |
| 02:27.49 | dexpdx | it shows up, then down |
| 02:27.51 | kolian123 | yes i have vpm400 on the board. |
| 02:27.55 | dexpdx | wtf |
| 02:28.21 | kolian123 | shmaltz, zap show channel 66 shows that echo disabled |
| 02:28.39 | shmaltz | kolian123, is channel 66 in use? |
| 02:28.52 | shmaltz | kolian123, what does dmesg show? |
| 02:28.59 | kolian123 | dmesg is ok |
| 02:29.18 | shmaltz | kolian123, does it show that vpm is persent? |
| 02:29.33 | kolian123 | VPM400: Span 0 U-law mode |
| 02:29.45 | kolian123 | shmaltz, dmesg is good |
| 02:29.53 | kolian123 | no channel 66 not in use |
| 02:30.28 | kolian123 | shmaltz |
| 02:30.31 | kolian123 | Echo Cancellation: 1 taps unless TDM bridged, currently OFF |
| 02:30.49 | kolian123 | just wondering if it's normal |
| 02:32.32 | kolian123 | It will turn on but says: Echo Cancellation: 1 taps unless TDM bridged, currently ON |
| 02:32.48 | kolian123 | shmaltz, just wondering about 1 taps |
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| 02:43.57 | Mercestes | How do I set an Aastra for timezone -5 GMT Eastern? |
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| 02:51.17 | flenders | ~seen JT |
| 02:51.19 | jbot | jt is currently on #asterisk #slug. Has said a total of 977 messages. Is idling for 1d 10m 25s, last said: 'yes, you haven't noticeD? :)'. |
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| 02:55.02 | J4k3 | should I be suprised that a P3-600 "choked" on a 4-way SIP conference, all g711u? |
| 02:55.03 | rue_mohr | does anyone know how to check analog lines for things like impedence mismatches? |
| 02:55.45 | rue_mohr | I want to know if my co card is really puttin on a 600R load |
| 02:55.54 | *** part/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com) |
| 02:56.31 | J4k3 | rue_mohr: they're never perfect. The important part are no short-to-ground on either wire (unless your local system has one side grounded, which generally sucks for performance) |
| 02:56.47 | J4k3 | and that you have enough talk power to hear the other side, and vice versa |
| 02:59.50 | *** join/#asterisk ManxPower (n=manxpowe@015-819-449.area5.spcsdns.net) |
| 03:00.31 | rue_mohr | hmm |
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| 03:00.35 | rue_mohr | tip is - right? |
| 03:01.14 | rue_mohr | there is something wrong here, I find to find out what |
| 03:02.32 | riddlebox | is there anything special you need to get disa working, I used to have it working, on 1.2 but now the same setup doesnt work with 1.4.x |
| 03:02.39 | ManxPower | rue_mohr: What is the specific issue? |
| 03:02.49 | rue_mohr | Tip is the ground side (positive) and Ring is the battery (negative) side of a phone circuit |
| 03:02.56 | ManxPower | riddlebox: The first thing to check is DTMF. |
| 03:03.18 | ManxPower | Call into your system and then log into voicemail. does it work? If so, then you prolly don't have a DTMF issue. |
| 03:03.21 | rue_mohr | sound isn't right, and I get cbc on the fxo line |
| 03:03.35 | ManxPower | call in from the same phone you are trying to use DISA with |
| 03:03.37 | madcap | tip is positive. |
| 03:03.38 | rue_mohr | as in CBC |
| 03:03.42 | rue_mohr | the radio station |
| 03:03.58 | ManxPower | Canadian Broadcasting Company |
| 03:04.11 | ManxPower | The Canadian NPR |
| 03:04.16 | ManxPower | or BBC as the case may be |
| 03:04.37 | rue_mohr | thats why I suspect that the fxo cxard is really high impedence |
| 03:04.50 | ManxPower | rue_mohr: you don't here it using a plain analog phone? |
| 03:04.55 | riddlebox | ManxPower, dtmf works fine, I have an auto attendant setup which works perfectly and voicemail works as well |
| 03:04.56 | ManxPower | hear that is |
| 03:05.14 | rue_mohr | no |
| 03:05.28 | rue_mohr | if I plug a phone into the incomming line its clear |
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| 03:05.40 | rue_mohr | when I use the channelbank, we get cbc on the inside |
| 03:05.45 | ManxPower | riddlebox: Good. What does the CLI show when you try using DISA? (use pastebin.ca if you have to paste the CLI output) |
| 03:05.45 | ManxPower | rue_mohr: I assume you ARE in Canada, right? |
| 03:05.51 | rue_mohr | yup |
| 03:05.52 | ManxPower | Oh! |
| 03:05.59 | ManxPower | A channel bank. |
| 03:06.12 | ManxPower | I've been fighting hum on a long analog loop into a channel bank. |
| 03:06.30 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 03:06.38 | ManxPower | rue_mohr: have you tried swapping the two wires in the pair. |
| 03:06.42 | aptura | Manx would a toleroid work on it? |
| 03:06.48 | *** join/#asterisk iBuMp- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 03:06.58 | rue_mohr | was just checking that |
| 03:07.04 | ManxPower | tolleriod? |
| 03:07.08 | aptura | filter |
| 03:07.16 | rue_mohr | ooo |
| 03:07.21 | rue_mohr | k try that after |
| 03:07.49 | riddlebox | ManxPower, as soon as it hits the disa part I get the busy tone |
| 03:07.54 | riddlebox | http://pastebin.ca/605336 |
| 03:07.54 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 03:07.54 | ManxPower | Ah. |
| 03:08.10 | ManxPower | rue_mohr: Do you live near the station you are hearing? |
| 03:08.51 | RyanW | Hello, can someone alalyze these logs and explain why the call terminated. http://pastebin.ca/605330 http://pastebin.ca/605337 |
| 03:08.57 | RyanW | please |
| 03:09.07 | J4k3 | ManxPower: load coils. :) |
| 03:09.09 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:09.13 | rue_mohr | no |
| 03:09.17 | rue_mohr | the really odd thing is |
| 03:09.21 | J4k3 | the phone company didn't put load coils in the phone system for the fun of it. |
| 03:09.44 | rue_mohr | if I pick up the analog phone on the line, so I get a dialtone |
| 03:09.49 | riddlebox | rue_mohr, fm re-tuning is easy, first, are you sure it is coming on the co? |
| 03:09.51 | J4k3 | a lot of modems work better as crystal radios than they do modems |
| 03:10.06 | rue_mohr | then get the channelbank to pick up the same line, all I hear is a nasty 60Hz buzz |
| 03:10.18 | J4k3 | some old USR Sportster's would pick up the local AM station (MF band, 1290 khz) loud and clear |
| 03:10.22 | J4k3 | and the station was pushing a measly 250W |
| 03:10.31 | rue_mohr | riddlebox, I dont get the radio station with an analog phone put directly on the line |
| 03:10.41 | aptura | rue, make sure some of the cableing is no where close to any ballast if that may be the case. |
| 03:10.51 | rue_mohr | no ballasts |
| 03:10.57 | aptura | k |
| 03:11.02 | riddlebox | rue_mohr, the first thing to do is to ground all pairs not needed |
| 03:11.08 | rue_mohr | the 60Hz only comes thru when I do that |
| 03:11.15 | rue_mohr | normally we dont get it |
| 03:11.18 | rue_mohr | hmm |
| 03:11.25 | rue_mohr | heh |
| 03:11.28 | ManxPower | riddlebox: hold on |
| 03:11.50 | ManxPower | J4k3: I can see that. |
| 03:11.52 | rue_mohr | I have a pair connected to a line to the road which isn't used... |
| 03:11.57 | rue_mohr | hmmm |
| 03:12.10 | rue_mohr | but I still dotn get it using a regular analog phone |
| 03:12.14 | ManxPower | based on my understanding of loading coils and RF, I can see how they would help |
| 03:12.40 | riddlebox | rue_mohr, is the it happening on all stations? or is it happening on an analog station? |
| 03:13.32 | riddlebox | rue_mohr, http://www.sandman.com/rf.html |
| 03:14.28 | riddlebox | I have used the cb filters and it drastically reduced the interference on the stations, but that was for a digital station....on an old toshiba perception e |
| 03:14.31 | aptura | Manx us hams use them all the time to cut out the RFI in the cabeling to the phones. But does not always work simply because of cheap non FCC ciritfied phones. Neighboors have always been at odds with the hams because of the interfearence problem. |
| 03:14.38 | ManxPower | riddlebox: And you have a ]from-incoming |
| 03:14.42 | ManxPower | <PROTECTED> |
| 03:14.57 | riddlebox | ManxPower, yes |
| 03:15.12 | ManxPower | with the same number of "n" and "m" s |
| 03:15.40 | aptura | There is a cirtain amount of wraps you need to put around the filters to cut out the interference where ever it is comming from. To bad you did not have a old o-scope lying around to sniff the 60hz signal |
| 03:15.58 | riddlebox | ManxPower, yes |
| 03:16.04 | ManxPower | Chances are it is a harmonic of 60Hz |
| 03:16.16 | aptura | Possible. |
| 03:16.27 | aptura | mabey 120 or 180 hz |
| 03:16.33 | ManxPower | pastebin the first 10 lines of that context |
| 03:16.42 | riddlebox | www.sandman.com is a really good source for interference issues |
| 03:16.51 | ManxPower | aptura: My hum was more or less 120 and 180Hz |
| 03:17.04 | aptura | okay |
| 03:17.55 | ManxPower | The HPEC seemed to help massivly |
| 03:18.01 | aptura | I know if you over drive a trancivers audio front end over 100% it can create harmonics but in standard AC circuits really dont see what would cause it. |
| 03:18.12 | rue_mohr | changing polarity -> didn't work. disconnecting unused line from street -> didn't work. adding 4 turns of ferrite -> didn't work |
| 03:18.38 | rue_mohr | wonder what I get with a dead line |
| 03:18.41 | ManxPower | rue_mohr: how about disconnecting the problem line at the dmarc |
| 03:19.03 | riddlebox | rue_mohr, are you sure it is on the co side? is it happening on an analog phone? or an IP phone? |
| 03:19.04 | rue_mohr | its onyl a problem with the channelbank |
| 03:19.23 | rue_mohr | its not happening with an analog phone plugged into the telco |
| 03:19.28 | ManxPower | riddlebox: I believe on a PSTN line |
| 03:19.29 | rue_mohr | its onyl a problem with the channelbank |
| 03:19.45 | rue_mohr | analog phone to pstn line is fine, clear as a bell |
| 03:20.57 | riddlebox | so you are taking an analog line from co and putting it into a channel bank? |
| 03:21.20 | rue_mohr | pstn -> channelbank -> asterisk -> channelbank -> analog phones |
| 03:21.31 | ManxPower | I assume it is the same channel bank |
| 03:21.40 | rue_mohr | if I disconnedct the incomming pair from the pstn, its clear |
| 03:21.42 | rue_mohr | no CBC |
| 03:21.52 | rue_mohr | yes, same channelbank |
| 03:21.54 | ManxPower | Have you checked the ground on the channel bank? |
| 03:22.07 | rue_mohr | I could be using much heavier wire |
| 03:22.37 | riddlebox | rue_mohr, are you getting the interference on all phones? |
| 03:22.56 | ManxPower | rue_mohr: Did you buy the channel bank new? |
| 03:23.45 | rue_mohr | noits a mainstreet 3624 |
| 03:24.24 | J4k3 | haha |
| 03:24.25 | J4k3 | :| |
| 03:24.26 | ManxPower | have you tried conning support out of them |
| 03:24.43 | ManxPower | J4k3: It is odd, so many RF and telco geeks around here. |
| 03:25.19 | ManxPower | Usually people are like "what's an FXO?" |
| 03:25.26 | J4k3 | asterisk is where many different kind of geek meet. |
| 03:25.33 | J4k3 | radio geeks, telco geeks, computer geeks |
| 03:25.40 | russellb | yay geeks. |
| 03:25.43 | J4k3 | because it all fits |
| 03:25.43 | [TK]D-Fender | rue_mohr, so... you get HBO on that thing yet? ;) |
| 03:25.49 | J4k3 | hell, even circus geeks |
| 03:26.08 | ManxPower | I've never been fond of small animals |
| 03:26.26 | J4k3 | ManxPower: so bite their heads off! :) |
| 03:26.27 | russellb | o.O |
| 03:26.40 | russellb | i really don't even like phones ... |
| 03:26.47 | [TK]D-Fender | ManxPower, Way to go Ozzy! |
| 03:26.50 | russellb | they are quite annoying |
| 03:26.57 | J4k3 | I don't talk on the phone much, but mostly because I don't have a decent phone |
| 03:27.00 | rue_mohr | [TK]D-Fender, :/ |
| 03:27.03 | rue_mohr | I hate CBC |
| 03:27.03 | ManxPower | [TK]D-Fender: don't get me excited |
| 03:27.07 | J4k3 | I test my * via cellular because I'm too cheap to buy *myself* a phone. |
| 03:27.10 | russellb | i like making them work ... just not using them |
| 03:27.15 | J4k3 | (it also makes a great excuse to say "I'll call them back" |
| 03:27.20 | ManxPower | I don't even HAVE a non-cell phone at the moment. |
| 03:27.31 | [TK]D-Fender | rue_mohr, CBC Radio : The cure for insomnia (or the ultimate punishment) |
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| 03:30.42 | rue_mohr | I'm suspicious of the fxo cards |
| 03:30.50 | rue_mohr | their 100% solid state |
| 03:31.32 | rue_mohr | ie non-isolated |
| 03:31.33 | rue_mohr | hmmm |
| 03:31.52 | riddlebox | ManxPower, I figured my disa problem out, I didnt include any outgoing access from the incoming context |
| 03:31.58 | [TK]D-Fender | rue_mohr, Yes, we all know that the best radio amplification is with TUBES ;) |
| 03:32.07 | [TK]D-Fender | riddlebox, SMRT |
| 03:32.22 | riddlebox | :) |
| 03:32.24 | rue_mohr | hehe, a tube pbx |
| 03:32.33 | ManxPower | riddlebox: you shouldn't |
| 03:32.51 | riddlebox | ManxPower, how else am I going to be able to use the disa line? |
| 03:34.37 | rue_mohr | I suppose nobody else has or has had a newbridge mainstreet 3624? |
| 03:35.00 | ManxPower | riddlebox: My systems have these contexts: [incoming] is where calls from untrusted sources land, like calls from the PSTN. [toll-trunks] and [local-trunks] is where the access to the outside world is. [toll-access] is where devices that can call out is located. [extensions] is where internal phones are listed. |
| 03:35.53 | ManxPower | riddlebox: do outside, untrusted calls ever land in in that context? |
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| 03:36.49 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:37.05 | riddlebox | the only thing that a caller would do is press 1 or 2 there, I have *18 built just for me, if the press any other digit, it starts the auto attendant over again, and loops it three times, if they dont get anything right in three times it hangs up |
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| 03:56.57 | J4k3 | hmm, what the world needs is a USB DECT multi-handset system direct connectability to * |
| 03:57.38 | russellb | there are multiple vendors selling SIP to DECT systems |
| 03:57.58 | russellb | wait, you said USB |
| 03:58.07 | russellb | it's late... |
| 03:59.08 | J4k3 | well, sip to dect is fine too, but usb would be cheaper to implement most likely |
| 03:59.23 | J4k3 | no need for a seperate power supply or even much intelligence in the base station. |
| 04:00.56 | FuriousGeorge | russellb: are you still awake? |
| 04:02.22 | FuriousGeorge | in regards to the bug reports ive filed over the last few weeks, i got that core dump you asked about |
| 04:02.27 | russellb | FuriousGeorge: somewhat, hey |
| 04:02.30 | russellb | ok, cool |
| 04:03.11 | russellb | can you point me to the bug number? |
| 04:03.18 | FuriousGeorge | one sec |
| 04:04.23 | FuriousGeorge | http://bugs.digium.com/view.php?id=9889 <--- this is the original. as per your suggestion i upgraded to 1.4.5 after that. then i filed this one: |
| 04:05.30 | FuriousGeorge | hmm, i think qwell closed out the other one asking for a bt against 1.4.6. at the time asterisk was crashing without dumping a core |
| 04:05.44 | FuriousGeorge | i notice that it crashed to day and dumped a core though |
| 04:05.46 | Qwell | I don't think I closed it |
| 04:05.55 | FuriousGeorge | its not in my view for some reason |
| 04:06.04 | russellb | did you get a backtrace from the core today? |
| 04:06.23 | FuriousGeorge | i think i can do that |
| 04:06.32 | russellb | gdb /usr/sbin/asterisk core.12345 |
| 04:06.37 | russellb | (gdb) bt ... (gdb) bt full |
| 04:06.41 | russellb | then pastebin.ca |
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| 04:09.17 | FuriousGeorge | http://pastebin.ca/605375 |
| 04:09.32 | FuriousGeorge | looks like a segfault |
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| 04:11.47 | *** part/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 04:12.41 | russellb | yeah ... and it's not a trivial one ... can you post it to a bug, please? |
| 04:13.00 | FuriousGeorge | russellb: i guess i should upgrade to .6 first, no? |
| 04:13.16 | russellb | yeah |
| 04:13.22 | FuriousGeorge | i changed the way my users parked aswell. its better now because asterisk totally dies and automatically restarts |
| 04:13.35 | FuriousGeorge | dont know if those things are related |
| 04:13.44 | russellb | heh |
| 04:13.44 | FuriousGeorge | but this damn computer has not worked right for months |
| 04:13.49 | russellb | well, it's still not right ... |
| 04:14.05 | FuriousGeorge | russellb: thanks for the time |
| 04:14.57 | russellb | can you make this happen? or is it random? |
| 04:21.33 | russellb | hrm ... |
| 04:25.18 | FuriousGeorge | russellb: its always been random |
| 04:25.22 | FuriousGeorge | sorry for delay responding |
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| 04:25.52 | russellb | no problem |
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| 04:28.44 | FuriousGeorge | but the "symptoms" always change. with 1.2.X it was a lot of avoiding initial deadlock. With 1.4.4 there would be no such error though a deadlock-like condition would occur. in 1.4.5 (pre- my changes to parking) it would just die and not dump a core. now it dumps a core. |
| 04:28.55 | FuriousGeorge | i also have an earlier core dump, if you would like to see that |
| 04:30.01 | FuriousGeorge | i dont know if you recall my telling you this, but my users were using a "ParkOrbit" feature of the phone, which would BlindTx the person to the parking extension. now i make them park it with a normal manual blind or AtTx |
| 04:30.35 | FuriousGeorge | russellb: so what am i doing then? upgrade to 1.4.6, see if it happens again, and if it does, make a bug report? |
| 04:31.45 | *** join/#asterisk ManxPower (n=manxpowe@26.sub-70-197-93.myvzw.com) |
| 04:33.02 | russellb | yeah, i remember that |
| 04:33.24 | russellb | i would say go ahead and post a bug, as well as update |
| 04:33.51 | russellb | i would like to investigate that backtrace some more |
| 04:34.04 | russellb | if you have an older backtrace, post that as well |
| 04:34.08 | russellb | the more info the better |
| 04:34.19 | FuriousGeorge | sip.conf and extensions.conf too, then? |
| 04:34.37 | russellb | yeah, without any private information of course |
| 04:34.50 | FuriousGeorge | naturally ;) |
| 04:35.07 | FuriousGeorge | ok, thats what i'll do then |
| 04:35.10 | russellb | msg me the bug number so i can monitor it |
| 04:35.13 | ManxPower | My Sprint PCMCIA card keeps cycling between EVDO Rev A, 1xRTT, and something called "Circuit Data". Way to go Sprint |
| 04:35.14 | rue_mohr | ok, why would a office card look like it has no isolation and a subscriber card have isolation transformers and relays? |
| 04:35.24 | FuriousGeorge | russellb: you got ti |
| 04:35.25 | FuriousGeorge | it |
| 04:35.31 | *** part/#asterisk OrNix (n=ornix@b-internet.87.103.245.217.snt.ru) |
| 04:35.37 | russellb | cool, we'll get it fixed eventually ... |
| 04:35.41 | russellb | g'night |
| 04:36.20 | russellb | rue_mohr: FXS modules have more junk on them because they do the ringing |
| 04:36.35 | rue_mohr | hmm ok |
| 04:36.54 | rue_mohr | yea there is no way I could have it backwards |
| 04:38.02 | rue_mohr | I tried some nice big ground wire right to the steak, didn't help at all |
| 04:38.33 | rue_mohr | I'll try inserting soem resistors and see what the currents look like |
| 04:38.41 | rue_mohr | I think their really wrong |
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| 05:17.56 | JoelSolanki | Good Evening all |
| 05:21.05 | JoelSolanki | I want to display live calls in browser in php. i guess we can get the live calls from astdb. |
| 05:21.12 | JoelSolanki | can anybody provide inputs on this plz. |
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| 06:00.47 | OloBola | I can only find DVD ISO's for Fedora core 7.. ? |
| 06:04.09 | J4k3 | somebody should get movie rental joints to rent open source operating system disks. |
| 06:04.33 | J4k3 | for those folks with shitty broadband, or none. |
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| 06:53.00 | J4k3 | soo, it appears I can get pda phones for $160ish off ebay, unlocked w/o contract. woohoo :) |
| 06:53.06 | J4k3 | (and they're compatable with my cellular provider) |
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| 06:55.53 | Supaplex | spifty |
| 06:56.10 | J4k3 | err, wifi equipped pda phones |
| 06:56.17 | J4k3 | (for sip-over-wifi use) |
| 06:56.36 | *** join/#asterisk vlt (n=dm@port-87-234-95-227.dynamic.qsc.de) |
| 06:58.26 | vlt | Hello. I "serve" a BRI/PTP line to a legacy ISDN PBX and want to use the 3rd B channel to phone out. What does " -- Ignoring callwaiting SETUP on channel 255/255 span 3 -1" appearing on *CLI mean? |
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| 07:02.43 | J4k3 | yeah, I'ma get an sch-i730 it appears, unless I find out theres some showstopping problem between now and tomorrow ;) |
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| 07:14.58 | [[blah]asfd | what would cause this error: |
| 07:15.01 | [[blah]asfd | Jul 5 19:01:10 WARNING[27795] chan_iax2.c: Maximum trunk data space exceeded to 10.2.0.10:4569 |
| 07:15.13 | [[blah]asfd | I have trunking set to yes on both ends of my connection |
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| 07:36.24 | santoshr | is it possible to start asterisk as a different user and connect to the console from a different user |
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| 08:06.30 | steliosk | santoshr : if the 2 users are in the same group it should work |
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| 08:08.59 | Swat2 | can anyone explain what insecure=very does? |
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| 08:11.01 | Swat2 | http://pastebin.ca/605532 |
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| 08:29.57 | vlt | Hello. What is the difference between "user=", "username=" and "fromuser=" in a "type=peer" section of sip.conf? |
| 08:30.07 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:30.41 | Andri[DK] | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 08:32.01 | vlt | Andri[DK]: Thank you. |
| 08:33.53 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:35.56 | santoshr | steliosk: they are in the same group. i get this Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?).. whereas when i do it from the user that started asterisk or root i am able to get through |
| 08:36.25 | Andri[DK] | probably due to permissions of the socket? |
| 08:38.14 | santoshr | but the permission is 755 .. i mean the group has execute permission to the file |
| 08:38.59 | *** join/#asterisk matsk (n=mk@194.68.102.171) |
| 08:41.10 | Andri[DK] | not sure, but I'll have to deal with this later today soon though |
| 08:42.07 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:43.41 | dominic1 | Can anybody help me with the internal webserver of asterisk? I wanted to login to the webinterface with http://mymachine/asterisk/manager?action=login&username=foo&secret=bar ,but it doesn't work |
| 08:43.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:44.52 | dominic1 | can I get a list of http commands? |
| 08:45.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:45.20 | vlt | Andri[DK]: I've read your link. So there's no option "user=", right? "username=" is used to authenticate while "fromuser=" replaces the CALLERID. Why doesn't the SIP packet sent to the peer contain the username at all that is set in sip.conf? |
| 08:45.30 | Andri[DK] | dominic1: manager show commands i think |
| 08:46.30 | Andri[DK] | vlt: depends on your version, you can always include the username in the dial command though, like Dial(SIP/myuser@mypeer,911) |
| 08:46.50 | Andri[DK] | I'm mostly guessing here though |
| 08:49.33 | vlt | Andri[DK]: Where to put the extension in your example's Dial() cmd? |
| 08:49.56 | vlt | Andri[DK]: nm, it's the 911, right? |
| 08:50.01 | Andri[DK] | just make a test extension in your default context |
| 08:50.14 | Andri[DK] | heh, 911 is the phonenumber, and i recommend you change that ;) |
| 08:50.34 | Strom_M | uh, no, 911 is the timeout unless you replace , with / |
| 08:50.42 | vlt | Andri[DK]: That would be no problem here in .de |
| 08:50.55 | Andri[DK] | good point Strom |
| 08:51.24 | Andri[DK] | vlt: I recommend that you look at www.voip-info.org, there are alot of configuration examples there |
| 08:52.38 | vlt | Strom_M: The cmd should be "Dial(SIP/[user@]peer/exten)"? That looks very IAXy ... hmmm ... |
| 08:52.53 | Andri[DK] | looks very Asterisk, I'd say ;) |
| 08:53.23 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 08:57.20 | *** join/#asterisk OloBola (n=not@74.95.13.57) |
| 08:58.06 | OloBola | is it possible to reinstall asterisk without losing all my current setup? |
| 08:58.16 | OloBola | I need to run 1.4.3 |
| 08:58.26 | OloBola | for lumenvox |
| 08:58.35 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 09:00.36 | Andri[DK] | I think that depends on your installation. If you don't have any interface cards I think the configuration transition should be pretty easy. At least I'm not have much problems with moving from 1.0 to 1.4 atm |
| 09:01.31 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:01.37 | OloBola | I don't have any cards |
| 09:03.04 | Andri[DK] | Then it shouldn't be that hard. I f.e. installed a new server and I run it in parallel with the old one but route unknown extensions between the two boxes through IAX2 while I'm in transition. |
| 09:03.21 | Andri[DK] | then when I'm ready I'll just turn the old one off |
| 09:03.33 | steliosk | santoshr : if i remember correctly it has to do with the permissions of asterisk.ctl |
| 09:04.38 | OloBola | ok, great. So a new installation shouldn't overwrite my conf files etc? |
| 09:04.49 | Andri[DK] | OloBola: I'm using seperate machines |
| 09:04.50 | steliosk | santoshr : either fix them or use sudo. I used the later with some cgi scripts for a web interface for asterisk we did |
| 09:05.19 | *** join/#asterisk bmg505 (n=leon@196.209.176.213) |
| 09:05.49 | OloBola | I see. I just pulled a drive from an old machine and put it in a new one. Everything seems to be ok. I just to switch to 1.4.3 |
| 09:10.05 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:15.15 | dominic1 | is it not possible to administrate meetme conferences via manager api? |
| 09:19.56 | oej | Should be, dominic1 |
| 09:20.07 | oej | If you have the module loaded |
| 09:21.08 | dominic1 | there are just two commands meetmemute and unmute |
| 09:21.15 | dominic1 | but I want more.... |
| 09:21.55 | oej | You can reach the CLi commands with the "command" action |
| 09:22.05 | oej | What more do you need? |
| 09:22.36 | dominic1 | list of conferencerooms and users |
| 09:22.58 | dominic1 | kick users |
| 09:29.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:30.16 | *** join/#asterisk snuff-work (n=na@61.29.30.137) |
| 09:30.47 | OloBola | so I backed up: var-lib-asterisk and etc-asterisk |
| 09:30.59 | OloBola | I don't have any voicemail |
| 09:31.09 | OloBola | should I be ok? |
| 09:39.11 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 09:40.26 | Zeeek | slowly I turned |
| 09:42.36 | oej | dominic1: Yes, those are good ideas. |
| 09:53.39 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 09:54.13 | Zeeek | hi oej |
| 09:54.42 | Uatec | hi |
| 09:54.48 | Zeeek | hi |
| 09:55.35 | oej | hi |
| 09:58.31 | *** join/#asterisk saftsack (n=saftsack@pD9E06F25.dip.t-dialin.net) |
| 10:00.37 | *** join/#asterisk mjmarrio (n=mike@210.19.201.38) |
| 10:00.56 | mjmarrio | hello all I am having a problem sending dtmf from iaxcomm |
| 10:01.31 | mjmarrio | Asterisk is ok cause it is sending DTMF via the D option in the Dial command |
| 10:01.41 | mjmarrio | also a SIP phone is working ok |
| 10:01.51 | mjmarrio | but the iaxcomm does not seem to work |
| 10:02.02 | mjmarrio | I am not sure if it is an Asterisk config or iaxcomm |
| 10:02.07 | *** join/#asterisk codejunky (n=jan@codejunky.org) |
| 10:02.13 | mjmarrio | I have tried all the codes in iaxcomm |
| 10:02.17 | mjmarrio | anyone help? |
| 10:02.18 | *** part/#asterisk codejunky (n=jan@codejunky.org) |
| 10:03.09 | mjmarrio | I called another iaxcomm phone and sent dtmf but the other user could not hear |
| 10:03.55 | mjmarrio | the xlite phone sends tone back to the earpiece as well as to the remote party |
| 10:04.10 | mjmarrio | so no problem there |
| 10:13.28 | *** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk) |
| 10:39.15 | tzafrir_home | mjmarrio, what is the other user? |
| 10:39.30 | tzafrir_home | mjmarrio, try calling an Echo() extension |
| 10:39.36 | Zeeek | hello tzafrir_home |
| 10:39.48 | tzafrir_home | (echo test: sends you back what it gets) |
| 10:39.52 | Zeeek | I was wondering if you solved your Talkshoe problem? |
| 10:40.09 | tzafrir_home | haven't had the time to look at it. I'll try today |
| 10:40.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:40.32 | Zeeek | you can always listen to the stream and watch the video at http://www.atserisktv.com |
| 10:40.37 | Zeeek | shit |
| 10:40.48 | Zeeek | video at http://www.asterisktv.com |
| 10:41.19 | tzafrir_home | better than atrisktv... |
| 10:41.20 | Zeeek | AT 16:30 GMT Mark will be live on the vid and conference |
| 10:41.49 | Zeeek | I don't know if I can pull off the technology of this thing, but I'll sure try |
| 10:43.53 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 10:44.29 | mjmarrio | oh sorry I was away |
| 10:45.15 | mjmarrio | ok let me see how to set that up |
| 10:47.51 | mjmarrio | but I have tested that a sip phone sends and is ok. Do you know if I have to set anything for iax in asterisk for dtmf signalling? |
| 10:48.31 | mjmarrio | I can see DTMF 1 being printed at the console of iaxcomm so it thinks it is sending dtmf |
| 10:49.01 | mjmarrio | is their a parameter in asterisk specific to iax for receiving and passing dtmf |
| 10:49.06 | mjmarrio | perhaps? |
| 10:49.10 | Zeeek | have you tried sniffing the network? |
| 10:49.50 | mjmarrio | Zeeek: well sniffing the iax packets to see if the dtmf is included you mean? |
| 10:50.01 | Zeeek | ya |
| 10:50.15 | mjmarrio | you have a nice sniff command I can follow? |
| 10:50.23 | Zeeek | what OS? |
| 10:50.44 | mjmarrio | Linux FC4 or FC6 |
| 10:51.09 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 10:51.22 | Zeeek | there is ip* like ipdump - I can't remember look up network monitoring |
| 10:51.41 | Zeeek | ethereal of course |
| 10:51.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:52.04 | mjmarrio | ok so what am I looking for in the iax packets? what command line options should I use? |
| 10:52.53 | Zeeek | doesn't idefisk have a linux version? |
| 10:53.48 | mjmarrio | don't know |
| 10:54.17 | Zeeek | have you tried calling a bank with DTMF? |
| 10:54.31 | mjmarrio | so I guess there is no specific config in asterisk I should be looking at? |
| 10:54.45 | Zeeek | if the dtmf mode is set, no |
| 10:55.12 | mjmarrio | well I have tried calling a known IVR system that I installed (Asterisk) and that works fine with xlite |
| 10:55.33 | mjmarrio | I also called another iaxcomm on my network from the xlite and that also worked fine |
| 10:55.44 | Zeeek | look at the iax.conf stuff in the sample file? |
| 10:55.50 | mjmarrio | but the iaxcomm does not seem to sent |
| 10:55.54 | mjmarrio | yeah I have done that |
| 10:56.10 | mjmarrio | tried to extend the length to 300 in zapata.conf also |
| 10:56.42 | mjmarrio | I have iaxfriends in a mysql table also |
| 10:56.50 | mjmarrio | not relevant though I know |
| 10:58.10 | mjmarrio | hmm yum search sniff comes up with lots of hits so If I really have to sniff traffic I will but seems a bit of an overkill at this stage |
| 10:59.31 | mjmarrio | is there a dtmf mode for iax.conf? |
| 11:00.59 | mjmarrio | I couldn't find a param for that. I am using 1.4 |
| 11:01.15 | Zeeek | look here: http://www.voip-info.org/tiki-index.php?page=IAX+versus+SIP |
| 11:01.36 | Zeeek | IAX always sends DTMF out of band so there is never any confusion about |
| 11:01.36 | Zeeek | what method is used. |
| 11:01.36 | mjmarrio | looking now |
| 11:01.57 | *** join/#asterisk sadara (n=sadara@203-59-87-43.dyn.iinet.net.au) |
| 11:05.33 | mjmarrio | well I think it is pretty clear that iaxcomm uses out of band. Especially if the GSM codec is used. But is that important? I mean if it is out of band then the signalling is within the protocol and therefore asterisk does something about it like get the digium card to send appropriate dtmf tone. |
| 11:06.03 | mjmarrio | if it is in band it is then part of the voice stream which should also work |
| 11:06.30 | Zeeek | who can make a call for me in the usa? |
| 11:07.32 | mjmarrio | hmm perhaps you may be on to it... |
| 11:07.44 | Zeeek | call 1 (724) 444-7444 enter 4296# 4444444444# (that 10 4s) |
| 11:07.47 | mjmarrio | I am using a TE205P card |
| 11:08.04 | Zeeek | mjmarrio try using ulaw to test |
| 11:08.11 | mjmarrio | done that |
| 11:08.18 | mjmarrio | trie all the codecs |
| 11:08.23 | mjmarrio | no joy |
| 11:08.28 | mjmarrio | I was thinking |
| 11:08.47 | mjmarrio | if the xlite is using in band signalling and the iaxcomm is not... |
| 11:09.01 | Zeeek | tell the client to use inband |
| 11:09.06 | Zeeek | that seems obvious |
| 11:09.19 | Zeeek | of course it's supposed to always do that |
| 11:09.25 | Zeeek | so there's nothig to tell :) |
| 11:09.29 | mjmarrio | not sure what you mean |
| 11:09.33 | Zeeek | nevermind |
| 11:09.48 | mjmarrio | hmm |
| 11:10.02 | Zeeek | can someone make a free SIP call to my meeting to test it ? |
| 11:10.25 | rbd | say a SIP call is transferred from asterisk server A to server B. Will server B generate a NewChannel AMI event when it gets the call, or something else? |
| 11:10.28 | Zeeek | tzafrir if you had a second maybe this will test your thing as well |
| 11:11.05 | mjmarrio | if an iaxcomm client send out of band dtmf then asterisk can either choose to send the tone in band or out of band depending on the hw config I guess |
| 11:11.21 | mjmarrio | but either way the dtmf should get to the other end |
| 11:11.56 | Zeeek | tzafrir_home TEST CALL: Dial (SIP/123@66.212.134.192,60,D(4296#5555555555#)) |
| 11:12.08 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 11:12.13 | mjmarrio | looking at zapata.conf |
| 11:12.29 | Zeeek | you're calling on a ZAP phone? |
| 11:13.32 | mjmarrio | asking me? |
| 11:13.43 | Zeeek | yes, why zapata? |
| 11:14.04 | Zeeek | you seem a little confused |
| 11:14.24 | obnauticus | . |
| 11:14.28 | mjmarrio | well ur right if I cannot send to another iaxcomm on same asterisk machine it has nothing to do with zapata |
| 11:14.50 | Zeeek | someone make a test call to me: Dial (SIP/123@66.212.134.192,60,D(4296#5555555555#)) |
| 11:14.51 | mjmarrio | just grasping |
| 11:15.39 | mjmarrio | I am confused as to what asterisk does when a client sends an out of band dtmf signal |
| 11:16.05 | Zeeek | what do you want it to "do"? |
| 11:16.26 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 11:16.50 | tzafrir_home | Forbidden |
| 11:17.03 | tzafrir_home | Zeeek, still "forbidden" |
| 11:17.05 | Zeeek | tzafrir_home which was where we left off, right? |
| 11:17.11 | tzafrir_home | right |
| 11:17.27 | Zeeek | This is the first time I've ever heard of that happening |
| 11:17.30 | mjmarrio | well if the destination is not iax2 then it needs to send the dtmf on either out of band if it is a digitial line such as ISDN or in band if it is analog |
| 11:17.56 | Zeeek | mjmarrio I assume it will do what you instruct it to do on that channl |
| 11:18.19 | mjmarrio | I assume that if an iaxcomm client calls a POTS line the dtmf is converted to in band tone by asterisk |
| 11:18.24 | Zeeek | tzafrir_home do you have a way to call US cheap? I guess not |
| 11:19.06 | mjmarrio | if another iaxcomm client is called then the client recognises the dtmf in the protocol and generates a tone?? |
| 11:19.09 | tzafrir_home | international calls from here are not *that* expensive, actually |
| 11:19.23 | tzafrir_home | a bit more than cellular calls |
| 11:19.29 | *** join/#asterisk berktr (n=cn@teknopet.com) |
| 11:19.36 | Zeeek | I pay less that $0.01/min usually and the max is 3c/min which is still not much |
| 11:19.38 | tzafrir_home | but this has become a chalange for me now |
| 11:19.55 | Zeeek | you want me to send you a debug of a call that works? |
| 11:20.06 | tzafrir_home | yes, please |
| 11:20.16 | Zeeek | ok, just a sec |
| 11:22.29 | Zeeek | I'm going to PM, ok? |
| 11:23.04 | mjmarrio | hmmm pretty frustrating. iaxcomm to iaxcomm you think would have no problem |
| 11:23.11 | mjmarrio | sending dtmf |
| 11:23.32 | mjmarrio | when one party presses the key pad the other should hear a tone |
| 11:25.04 | tzafrir_home | how do I allow floods? |
| 11:25.07 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 11:25.12 | Zeeek | heh |
| 11:25.21 | Zeeek | Ok, let me di a pastebuin |
| 11:26.19 | Zeeek | tzafrir_home http://pastebin.ca/605678 |
| 11:27.33 | Zeeek | tzafrir_home note that I called an invalid code, so although the call itself works and I hear the message, it did send a "Busy Here" at one point |
| 11:31.52 | tzafrir_home | "MyIPAddress" is your internal IP address, right? |
| 11:32.09 | Zeeek | no external |
| 11:32.15 | Zeeek | I am behind NAT though |
| 11:32.16 | tzafrir_home | Anyway, instead of the immediate 200 (OK) answer, I get an 403 answer |
| 11:32.38 | Zeeek | could it be a DTMF issue? |
| 11:33.04 | Zeeek | I've called that SIP number from every machine I own |
| 11:33.13 | tzafrir_home | nah. It didn't even establish a connection |
| 11:33.24 | *** join/#asterisk perf3kt (i=perf3kt@iupui-vpn-32-94.noc.iupui.edu) |
| 11:33.37 | Zeeek | never had any problem, direct (PC client) or through asterisk |
| 11:34.04 | Zeeek | I hate mysteries like this |
| 11:35.43 | berktr | what does this mean => moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 11:36.59 | *** join/#asterisk gardo (n=gardo@121.97.199.100) |
| 11:38.18 | Zeeek | TGF |
| 11:39.08 | Zeeek | for example: http://forums.digium.com/viewtopic.php?t=7931 |
| 11:39.37 | tzafrir_home | http://pastebin.ca/605694 |
| 11:40.14 | tzafrir_home | berktr, no zaptel timing |
| 11:40.41 | tzafrir_home | berktr, do you have zaptel installed (e.g: ztdummy)? |
| 11:42.12 | *** join/#asterisk cayorde (n=flexable@87.19.162.237) |
| 11:43.39 | Zeeek | tzafrir_home remove fromuser |
| 11:44.18 | tzafrir_home | I get basically the same, only with 6003 as the fromuser |
| 11:44.30 | tzafrir_home | Want a post? |
| 11:45.23 | tzafrir_home | I don't quite get the interaction between fromuser and the callerid |
| 11:45.25 | Zeeek | mwait a sec, let me look at mine |
| 11:45.41 | tzafrir_home | 6003 is the callerid |
| 11:46.13 | Zeeek | try using your PIN as acallerid as in callerid="Me <1234567890>" |
| 11:46.49 | Zeeek | and add canreinvite=no even if you're not behind NAT |
| 11:46.54 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:50.47 | berktr | zaptel is enabled and installed |
| 11:51.11 | berktr | tzafrir_home, what do you suggest |
| 11:52.00 | tzafrir_home | Zeeek, authentication still fails |
| 11:52.13 | tzafrir_home | canreinvite is only with regards to RTP issues |
| 11:52.16 | tzafrir_home | right? |
| 11:52.36 | tzafrir_home | berktr, ls -l /dev/zap/pseudo |
| 11:52.56 | rbd | say a SIP call is transferred from asterisk server A to server B. Will server B generate a NewChannel AMI event when it gets the call, or something else? |
| 11:52.57 | berktr | %ls -l /dev/zap/pseudo |
| 11:52.57 | berktr | crw------- 1 root wheel 0, 113 Jul 6 14:45 /dev/zap/pseudo |
| 11:53.20 | tzafrir_home | berktr, so only root can read from it. |
| 11:53.44 | tzafrir_home | berktr, what distribution / kernel do you use? (to see if you use udev) |
| 11:53.46 | Zeeek | tzafrir_home yes for canreinvite, you're right it shouldn't matter |
| 11:54.11 | Zeeek | berktr I highly recommend Google. THere are hundreds of pages about this, you'll see some specific suggestions |
| 11:54.17 | berktr | <PROTECTED> |
| 11:54.56 | Zeeek | and? WHat did you see? |
| 11:55.20 | berktr | people suggested using file variable instead of others in moh.conf |
| 11:55.31 | tzafrir_home | Zeeek, give him a break. THere's also some confusing stuff there... |
| 11:55.36 | berktr | however when i use file, it uses mpg123, which freezes sometimes |
| 11:56.00 | Zeeek | I know, I'm just determining what he's alrady seen |
| 11:57.32 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 11:58.15 | tzafrir_home | berktr, what distribution do you use? |
| 11:58.18 | Zeeek | Here's an interesting page: http://www.asteriskguru.com/tutorials/warnings.html |
| 11:58.41 | Zeeek | The timing mechanism required by several Asterisk applications is a 1KHz interrupt. If you |
| 11:58.42 | Zeeek | are using recent kernels ( > 2.6.12), make sure to check in the kernel configuration, you |
| 11:58.42 | Zeeek | selected in processor type and features the timing to be 1000Hz fixed. |
| 11:58.42 | Zeeek | " |
| 11:59.03 | tzafrir_home | asteriskguru pages tend to frequently turn up in searches due to comments. Some of them are simply unanswered questions |
| 11:59.22 | Zeeek | the above is a possible thing to investigate |
| 11:59.41 | Zeeek | I never hear of it before |
| 11:59.54 | tzafrir_home | Zeeek, actually, recent kernels ( >= 2.6.15 ) on i386 and amd64 have RTC that is also good enough |
| 12:00.22 | Zeeek | which is why they suggest checking the 1000Hz fixed thing, right? |
| 12:00.51 | tzafrir_home | no. It is a simple permissions issue. But if he has udev, he needs to fix the udev rules and chmod/chown. Otherwise he just needs to chmod / chown. |
| 12:01.03 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:01.13 | tzafrir_home | Ah, and check that timing actually works (zttest, or by reading from /dev/zap/pseudo ) |
| 12:02.30 | lilalinux | what signalling to I take for TE mode? the one with ptmp or without? |
| 12:02.49 | tzafrir_home | lilalinux, depends on what you want to connect to. |
| 12:03.03 | lilalinux | tzafrir_home: it's an ordinary Euro ISDN |
| 12:03.15 | tzafrir_home | chances are it is ptmp, if this is a standard connection |
| 12:03.22 | lilalinux | thx |
| 12:03.27 | lilalinux | and for my NT? |
| 12:03.36 | lilalinux | I want to connect an NTBA and a Gigaset |
| 12:04.18 | tzafrir_home | if you don't need to connect several phones on the same ISDN lines, I figure both ptmp and ptp will work just as well. You just need to agree with the other party. |
| 12:04.50 | lilalinux | My Gigaset has multiple phones, so I guess ptmp |
| 12:05.11 | tzafrir_home | what is a gigaset? |
| 12:05.34 | Zeeek | what is ztspeed supposed to tell you? |
| 12:06.02 | tzafrir_home | ptmp allows several devices to share the same phisical link. Just like several analog phones can be connected to the same FXS device on the same line. |
| 12:06.17 | lilalinux | tzafrir_home: it's a tlephone system |
| 12:06.22 | tzafrir_home | ztspeed? I have no idea what it is good for. Ignore it |
| 12:08.34 | Zeeek | ~ztspeed |
| 12:09.02 | lilalinux | I was offline for a minute O_o Did I miss a msg? |
| 12:09.03 | *** join/#asterisk jm|laptop (n=jm|home@sentry.flags.co.uk) |
| 12:09.23 | rbd | the ID of asterisk SIP channels take the form of SIP/peer-id ...they say that 'id' part is 'randomly' generated...however, when I make a call, disconnect, and make the call again, I get the same id component. Is the channel name randomly generated only each time asterisk has to allocate a channel? ....what I mean is, can I count on the pool of ids being relatively small? |
| 12:15.48 | berktr | what is this |
| 12:15.49 | berktr | abstract_jb.c:321 ast_jb_put: SIP/1007-08785000 recieved frame with invalid timing info: has_timing_info=1, len=0, ts=54140 |
| 12:17.08 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-40-34.lns3.syd7.internode.on.net) |
| 12:17.26 | *** join/#asterisk FliTTi (n=chatzill@212.218.65.233) |
| 12:17.41 | FliTTi | hey experts. i have an question: |
| 12:18.07 | FliTTi | my asterisk has uncloses (Down) Channels from MISDN. How can i destroy, close these hanging channels? |
| 12:18.11 | FliTTi | please help me |
| 12:20.48 | berktr | abstract_jb.c:321 ast_jb_put: SIP/1007-08785000 recieved frame with invalid timing info: has_timing_info=1, len=0, ts=54140 |
| 12:20.50 | berktr | please help |
| 12:20.55 | berktr | this happens when i force jb |
| 12:21.49 | oej | tss. A swedish magazine just publiched a notice about a new IP-phone with VoIP functions... Show me an IP phone without VoIP. |
| 12:22.11 | FliTTi | no body an idear what i can do with down channels? |
| 12:22.19 | FliTTi | so that they are closed? |
| 12:23.10 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 12:23.11 | oej | "soft hangup" in the CLI |
| 12:23.42 | santoshr | 1,n,ExecIfTime(17:00-18:00|*|*|*?Queue[desk|tT|200]) >>>> is anything wrong with this line |
| 12:24.41 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 12:25.11 | santoshr | >>>>Jul 6 17:55:54 WARNING[15652]: pbx.c:5594 pbx_builtin_execiftime: Cannot locate application Queue[desk |
| 12:25.16 | santoshr | i get the above error |
| 12:26.29 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 12:27.26 | Zeeek | santoshr show application queue |
| 12:27.38 | Zeeek | it look like it's not loaded |
| 12:28.55 | santoshr | no found the issue it wants the arguments to the application queue in comma seperated format. |
| 12:30.44 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:31.08 | Zeeek | so much the better |
| 12:34.28 | rbd | is it possible to get a SIP header variable through AMI's GetVar command? (e.g. like SIP_HEADER(X-customvar) ) |
| 12:37.31 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:37.32 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:38.36 | berktr | abstract_jb.c:321 ast_jb_put: SIP/1007-08785000 recieved frame with invalid timing info: has_timing_info=1, len=0, ts=54140 |
| 12:38.38 | berktr | this happens when i force jb |
| 12:38.57 | santoshr | thank you |
| 12:38.58 | santoshr | exit |
| 12:39.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:40.58 | dominic1 | question about meetme: I want that meetme does a playback of a soundfile |
| 12:41.03 | Zeeek | quit |
| 12:41.23 | dominic1 | like "you are now entering the conference" after the pin and before joining the conference |
| 12:41.33 | dominic1 | is it possible to adjust this setting? |
| 12:44.36 | k31th | jesus on this ubuntu system i dont have /boot/grub/menu.lst |
| 12:44.40 | k31th | how is this even working ? |
| 12:49.41 | tzafrir_laptop | lilo? |
| 12:50.12 | tzafrir_laptop | lilo -q (to query the installed boot manager IIRC) |
| 12:52.23 | tzafrir_laptop | dominic1, in the worst case, move the PIN question into the dialplan |
| 12:55.56 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:56.03 | dandre | hello, |
| 12:57.22 | dandre | I use the manager interface to get/set asterisk configuration from a gui tool. I have seen that GetConfig action doesn't send international characters (ie accentuated). Is there any workaround? |
| 12:58.21 | [TK]D-Fender | dandre: GCC |
| 12:58.36 | Zeeek | [TK]D-Fender-Alert |
| 12:59.07 | Zeeek | I skipped those iphone downloads :) |
| 12:59.49 | mocker | Heh. |
| 13:00.17 | dandre | [TK]D-Fender: Am I the only guy concerned with this? |
| 13:00.19 | Zeeek | I can only admire the success of the marketing |
| 13:00.35 | Zeeek | dandre no I hate when shit doesn't work with international characters |
| 13:00.45 | Zeeek | it's a real pain |
| 13:01.02 | Zeeek | as for a way round it... |
| 13:01.28 | [TK]D-Fender | dandre: Welcome to the world of AMERICAN software :) |
| 13:01.34 | Zeeek | yeah |
| 13:02.41 | dandre | ok so I will have to leave manager interface and write my one ini like file modification :-( |
| 13:03.21 | dandre | what is strange thought is that the manager can write international chars but can't read them ;-) |
| 13:03.47 | Zeeek | chmod 1777 manager |
| 13:03.57 | Zeeek | <joke> |
| 13:04.10 | *** join/#asterisk Op3r (n=op3r@121.97.196.30) |
| 13:04.32 | mocker | I need to figure a simple QoS solution for my house. |
| 13:04.44 | mocker | So my downloads stop making calls sound like total crap. |
| 13:05.09 | *** join/#asterisk basty (n=basty@212.218.65.246) |
| 13:05.29 | basty | Hi, anyone familar with mISDN with Asterisk ? |
| 13:05.33 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 13:05.47 | mocker | Aww, iaxtel.com is down. |
| 13:06.10 | Zeeek | mocker - router qos? |
| 13:06.18 | mocker | Zeeek: Something. |
| 13:07.02 | Zeeek | my old linksys has qos that seems to work in that way |
| 13:07.14 | mocker | I have DD-WRT firmware on my WRT54G |
| 13:07.15 | Zeeek | I think it even works at the MAC level |
| 13:07.21 | mocker | It has QoS, I just need to play w/ it. |
| 13:07.22 | mocker | <PROTECTED> |
| 13:07.30 | Zeeek | oh, so that wasn't a question |
| 13:07.45 | mocker | heh, no.. |
| 13:07.51 | dandre | more seriously, is this limitation on international chars only for the manager or is there any other side effect if they are in the config files? |
| 13:07.53 | mocker | Just babbling while drinking coffee.. |
| 13:07.58 | Zeeek | does it use MAC filter? I can't remember and my phone is on hub with PC |
| 13:08.05 | mocker | I think it can. |
| 13:08.22 | Zeeek | the one I had here at the office died |
| 13:09.00 | basty | I have strange problem using Asterisk with mISDN. Everytime I dial a number with my ISDN Phone, connected to a NT Port of the Beronet Bn8S0 Card - it cuts off the Number into a Number with max 5. |
| 13:10.05 | [TK]D-Fender | mocker: Yeah, install a DIY linux setup on your router and install something like PacketShaper or : http://www.faqs.org/docs/Linux-HOWTO/ADSL-Bandwidth-Management-HOWTO.html#AEN166 |
| 13:10.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.27 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 13:11.28 | dominic1 | basty, had this problem too, solved it with a wait(3) in the dialplan |
| 13:11.38 | dominic1 | I think a Waitfordigits will be better |
| 13:11.47 | Zeeek | [TK]D-Fender what's wrong with the router qos? |
| 13:12.24 | [TK]D-Fender | Zeeother solution let you better seperate by protocol / IP / MAC, etc. |
| 13:12.42 | Zeeek | the router will do a lot |
| 13:12.56 | [TK]D-Fender | Zeeek: And not require tagging or other such nonsense. |
| 13:13.01 | basty | dominic1: okay..i will try that..thanks! |
| 13:13.13 | Zeeek | including which port you're in which makes it very easy |
| 13:13.16 | x86_ | what kind of headset do I need for a polycom IP 301? |
| 13:13.19 | [TK]D-Fender | Zeeek: I'm not saying "don't try it", but I'm just providing do-able alternatives. |
| 13:13.35 | Zeeek | and there are many: the first is "don't download while using voip" |
| 13:13.44 | [TK]D-Fender | x86_: You'll need an RJ9 adapter cable, or AMP |
| 13:13.46 | Zeeek | but I digress |
| 13:14.19 | Zeeek | I have a cheap Plantronics headset on my Poly ip 155 |
| 13:14.34 | Zeeek | s/ip 155/ip500/ |
| 13:15.06 | x86_ | [TK]D-Fender: so any headset, plus an RJ9 adaptor cable? |
| 13:15.22 | Zeeek | most won't have enough juice w/o preamp |
| 13:15.40 | x86_ | hmm |
| 13:15.41 | [TK]D-Fender | x86_: The cable is a rare find, You'll likely have to mail-order it. AMP add to the cost but really come through on quality |
| 13:16.11 | x86_ | [TK]D-Fender: isn't there RJ9 headsets available? |
| 13:16.13 | Zeeek | I like mine. It has a qucik disconnect that puts the person on hold while you go smoke a cig or sthing :) |
| 13:16.30 | [TK]D-Fender | x86_: Indeed un-amped is usually pretty damn wimpy. My call center here uses IP 600's + Plantronics M22 Amps, and Plantronics H263 binaural headsets |
| 13:16.49 | [TK]D-Fender | x86_: Straight? VERY rare. Don't recall seeing any personally. |
| 13:16.51 | Zeeek | but the headset/amp is more expensive than the phone itself in some cases |
| 13:17.14 | x86_ | holy crap! the M22 is expensive by itself... |
| 13:17.42 | x86_ | for that kind of money there is the Plantronics CS55 |
| 13:17.46 | marl | hi, can anyone help with with the following, i am running trixbox/a@h/freepbx 2.2.2 and am trying to setup a ring group that dials my mobile number via one of my tdm ports, i need to dial 1470 first thow and have a pause before it dials the mobile number, is there a way to insert this pause into the dial sequence? (preferable with the freepbx web interface) |
| 13:17.51 | [TK]D-Fender | x86_: Yup, my headset combo cost more than your phone easily :D |
| 13:18.03 | [TK]D-Fender | ~trixbox |
| 13:18.04 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 13:18.12 | [TK]D-Fender | marl: ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:18.13 | Zeeek | here we go |
| 13:18.27 | [TK]D-Fender | marl: You've definately come to the WRONG place... |
| 13:18.27 | x86_ | [TK]D-Fender: why not just get the CS55? I used to use one of those at one of my old jobs, loved it |
| 13:19.13 | marl | [TK]D-Fender, thanks, had tried the freepbx irc and no ones around thought i would try here just incase, will try the trixbox irc as well |
| 13:19.43 | mocker | Is that phone conference today? |
| 13:20.02 | [TK]D-Fender | x86_: ummm... its over $200..... |
| 13:20.15 | [TK]D-Fender | x86_: wireless is nifty, but the quality vastly inferior. |
| 13:20.31 | Zeeek | mocker yeah, phone and bideo http://asterisktv.com |
| 13:20.37 | [TK]D-Fender | x86_: And lifters = ass |
| 13:20.46 | mocker | Zeeek: Do many people actually join up? |
| 13:20.48 | Zeeek | bideo = boring video :) |
| 13:21.05 | Zeeek | Usually there are about ten people plus the invisible listeners |
| 13:21.18 | mocker | werd. |
| 13:21.22 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:21.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:21.22 | Zeeek | Today, Mark Spencer will be on, so I expect more and better |
| 13:21.28 | mocker | really? |
| 13:21.32 | mocker | That's pretty cool. |
| 13:21.33 | Zeeek | really what? |
| 13:21.42 | Zeeek | Mark has been on the conf before |
| 13:22.17 | Zeeek | You can listen here: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 |
| 13:22.27 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 13:22.54 | Zeeek | Mark was in April |
| 13:25.38 | Zeeek | hmmmm, my 1.2 is a little behind |
| 13:25.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:26.47 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-ad7ba600dbb86e77) |
| 13:29.40 | Zeeek | is anyone else getting a zillion infected PDF a day with the subject [variable].PDF |
| 13:30.21 | blitzrage | I think I saw one of those |
| 13:30.27 | blitzrage | luckily I run linux, so it doesn't affect me :) |
| 13:30.37 | Zeeek | I get about 20 per day on several different accounts |
| 13:30.48 | Zeeek | it doesn't effect anyone with a brain either |
| 13:30.58 | Zeeek | errr |
| 13:31.05 | Zeeek | not that you don't have a brain |
| 13:31.15 | blitzrage | ouch Zeeek ..... ouch |
| 13:31.22 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 13:31.23 | Zeeek | I wouldn't know, having been in the same room with you and not having met you! |
| 13:32.04 | jart | would anyone like to test the latest voice changer before i make a release announcement? http://www.lobstertech.com/code/voicechanger/ |
| 13:33.43 | *** join/#asterisk flujan (n=flujan@201-43-212-42.dsl.telesp.net.br) |
| 13:34.29 | jart | please email bugs to jtunney@gmail.com |
| 13:34.39 | *** join/#asterisk joe-e (n=joe@megan.healthnet.co.uk) |
| 13:34.46 | flujan | hi guys... I wil install asterisk on a machine with a valid IP on the net. I will put secrets for the sip extensions I have. I want to know what exactly insecure=very will do... |
| 13:34.59 | flujan | will it give me crypt passwords? |
| 13:36.26 | Zeeek | [TK]D-Fender will answer that question any second |
| 13:36.56 | krdian_ | flujan: insecure=very ; To allow registered hosts to call without re-authenticating |
| 13:37.41 | joe-e | Hi, ive just got a Sangoma A101D PCI/ex with Echo cancellation.. going though their wiki to set it up and got to the stage to run wancfg_zaptel but getting error "Can't locate Filter/Util/Call.pm in @INC...." anyone set these cards up before? know what might be happening here? What is "Call.pm" ? |
| 13:37.52 | flujan | krdian_: You mean that every time a extension place a call it will authenticate the user again? |
| 13:40.25 | dominic1 | can I execute selects on a ODBC/mysql database from the dialplan? |
| 13:40.44 | jart | dominic1: i think you can in 1.4 |
| 13:41.06 | jart | there was a way to do it in earlier versions but it was very unreliable |
| 13:41.11 | jart | why not just use an AGI app? |
| 13:41.31 | jart | dialplan is really more for static switching |
| 13:42.00 | [TK]D-Fender | flujan: insecure=very means it WON'T re-auth for every call. |
| 13:42.06 | krdian_ | flujan: no, i mean that will be not authenticated again |
| 13:42.50 | flujan | [TK]D-Fender: so I can conclude that is better to DON'T use this options, right? |
| 13:43.01 | [TK]D-Fender | flujan: Thats my take |
| 13:43.06 | dominic1 | how? |
| 13:43.39 | dominic1 | I just want to get a password from the database and don't want to write a agi for that |
| 13:44.03 | krdian_ | dominic1: show function DB |
| 13:44.26 | krdian_ | dominic1: sorry, i'm wrong |
| 13:44.54 | krdian_ | ODBC_SQL |
| 13:45.26 | dominic1 | my asterisk has support for odbc_sql and is getting two sip users from there |
| 13:45.52 | dominic1 | now I want to add a new custom table and get passwords for my conferences from there |
| 13:45.52 | flujan | [TK]D-Fender: since you are the asterisk master. :) Which is the best secure policy o use with asterisk. I cannot use the host=IP because i have a dhcp on my network, |
| 13:46.11 | flujan | and every login the users changes the ip address. |
| 13:46.14 | blitzrage | anyone have any *ins* at Mitel? |
| 13:46.36 | [TK]D-Fender | flujan: Doesn't mean you can't set static. |
| 13:46.40 | Zeeek | Habla espanol? http://www.saghul.net/blog/2007/07/04/conferencia-televisada-sobre-asterisk/ |
| 13:46.41 | blitzrage | I have 12 Mitel 5220's that got a bad flash... |
| 13:46.49 | krdian_ | dominic1: www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc |
| 13:47.02 | blitzrage | (not physically... the software screwed up in installation -- nothing wrong with the hardware... but I'd like to figure out how to fix them) |
| 13:47.11 | blitzrage | FUNC_ODBC!!! |
| 13:50.08 | [TK]D-Fender | flujan: Most secure? Box with a GIANT PCI back-plane loaded with multi-port NIC's whree each phone is only allowed to connect to a given NIC. THEN you MAC filter everything, IP filter, COFFEE filter, hand-transcribe the packets onto Post-It (tm) notes, BURN them, use an old printer cover to begin pulsing out the smoke signals to be picked up by a neigboring high-rise. From there they will... |
| 13:50.09 | [TK]D-Fender | ...telegraph the signal back with a hash to the tin-cup transceiver station... then.... |
| 13:50.10 | *** join/#asterisk ReDNeQ- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 13:50.35 | [TK]D-Fender | ok.... I'll stop now :) |
| 13:50.36 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 13:50.37 | flujan | [TK]D-Fender: lolo |
| 13:50.38 | Zeeek | insecure=anal |
| 13:51.03 | Zeeek | shifted 5 bit baud code |
| 13:51.13 | flujan | [TK]D-Fender: for sure... I don't think my boss will allow it... Brazilian indians are not good with smoke signals... :P |
| 13:51.42 | [TK]D-Fender | flujan: its ALWAYS a personnel issue... |
| 13:51.52 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:51.56 | flujan | [TK]D-Fender: so, you think that setting static will be a good security policy? |
| 13:52.27 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 13:52.30 | Zeeek | anyone use digg? You can help: http://digg.com/podcasts/Asterisk_Users_Live_Conference_Podcast |
| 13:52.43 | [TK]D-Fender | flujan: I personally am in the "barely gives a shit" category. I set a user/pass, normal security, and thats about it. |
| 13:53.09 | Zeeek | user=user secret=secret |
| 13:53.13 | [TK]D-Fender | flujan: You MIGHT want to add a hostmask to the entry jjust to ensure they are local. |
| 13:53.29 | [TK]D-Fender | flujan: You could go 1 psycho step further and check the UA on EVERY call.... |
| 13:53.40 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:53.56 | *** join/#asterisk littleball (n=littleba@bb220-255-68-9.singnet.com.sg) |
| 13:53.56 | flujan | how can I check the UA every call? |
| 13:54.22 | [TK]D-Fender | Zeeek: Following our first conference : |
| 13:54.23 | [TK]D-Fender | ~sipnat |
| 13:54.24 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://aocomputing.net/wordpress/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:54.31 | [TK]D-Fender | flujan: Check the header in the dialplan. |
| 13:55.04 | flujan | [TK]D-Fender: thank you so much for the tips... I will give it a try. |
| 13:55.23 | *** join/#asterisk twitchnln (n=twitch@70.43.112.117.nw.nuvox.net) |
| 13:55.36 | [TK]D-Fender | flujan: think about me when it comes to the tin-can & string transceivers ;) |
| 13:55.45 | littleball | hi, i have install zaptel on centos 4.4, when running ztcfg, i got this error msg, ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 13:55.48 | flujan | ok... |
| 13:55.48 | twitchnln | good morning all |
| 13:55.49 | flujan | :D |
| 13:55.51 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:55.55 | littleball | who can help? |
| 13:56.16 | [TK]D-Fender | littleball: modprobe your modules first, and check dmesg |
| 13:56.20 | coppice | OFDM over string will one day replace FTTH |
| 13:56.52 | twitchnln | littleball: check dmesg and see if the card is there, if not try reseating the card |
| 13:57.12 | littleball | the card is there. if i run modprobe two times, then the problem gone |
| 13:58.04 | twitchnln | anyone had any luck with getting snmp mibs working with * |
| 13:58.50 | *** join/#asterisk bintut (n=bintut@cm25.gamma178.maxonline.com.sg) |
| 13:59.22 | twitchnln | i keep getting unknown object identifier (sub-id not found) when i try to snmpwalk the box |
| 14:00.52 | *** join/#asterisk Bu5hm4nn (n=Miranda@pD9E0DCE6.dip0.t-ipconnect.de) |
| 14:03.12 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 14:03.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:04.22 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 14:06.19 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 14:06.25 | littleball | it works perfectly after i run make config |
| 14:07.57 | lilalinux | I have asterisk-bristuff installed (debian/etch) and get the following messages: pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! |
| 14:08.00 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:08.13 | lilalinux | why does is say "pri"? |
| 14:08.17 | lilalinux | s/is/it |
| 14:08.25 | naitram | in the features.conf file what are the arguments to an application seperated with, example. Monitor,wav myoutputfile mb. are the arguments just seperated by spaces or what |
| 14:10.18 | *** join/#asterisk eth01 (n=crash@unaffiliated/eth01) |
| 14:10.44 | *** join/#asterisk Skarmeth (n=Skarmeth@201009016009.user.veloxzone.com.br) |
| 14:12.24 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.13 | [TK]D-Fender | lilalinux: pastebin your configs. |
| 14:15.15 | [TK]D-Fender | ~pb |
| 14:15.15 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 14:15.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 14:15.46 | *** join/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net) |
| 14:16.05 | trevarthan | hello, is it possible to get asterisk-addons for asterisk business edition? |
| 14:17.31 | [TK]D-Fender | trevarthan: I'm sure Digium will ansewr that one for free, but I can't imagine why not... |
| 14:18.05 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:18.46 | trevarthan | hoping someone from digium was on the list. oh well. thanks. |
| 14:18.48 | *** part/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net) |
| 14:19.46 | tzafrir_home | lilalinux, bri is often called "pri" in chan_zap |
| 14:23.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:25.24 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:25.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:26.38 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 14:27.25 | krdian_ | <PROTECTED> |
| 14:27.49 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 14:28.14 | *** join/#asterisk Vec (n=Vec@dsl-243-85-143.telkomadsl.co.za) |
| 14:29.29 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 14:29.58 | lilalinux | thx |
| 14:31.58 | twitchnln | so noone in here uses snmp to monitor their *??? |
| 14:32.25 | Juggie | its a new feature |
| 14:32.45 | naitram | in the features.conf file what are the arguments to an application seperated with, example. Monitor,wav myoutputfile mb. are the arguments just seperated by spaces or what |
| 14:34.08 | joe-e | Anyone here got a Sangoma A101D card and set it upt for use in Asterisk ? |
| 14:39.45 | [TK]D-Fender | joe-e: You seem to be missing a CPAN module or something (PERL related) that I believe their setup uses. |
| 14:39.58 | [TK]D-Fender | joe-e: Screw the automator. Does Wanrouter start? |
| 14:40.26 | joe-e | fender: fixed that ;) |
| 14:40.36 | creativx | john the automator |
| 14:40.58 | joe-e | hmmm, managed to run wancfg_zaptel now,, but doesnt seem to have done anything. |
| 14:41.42 | joe-e | Fender: "WanRouter start" -> showed 24 channes, then had and error message ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 14:41.54 | joe-e | Configuring interfaces: w1g1 w1g1: unknown interface: No such device |
| 14:43.49 | [TK]D-Fender | joe-e: pastebin your wanpipe1.conf |
| 14:43.54 | [TK]D-Fender | ~pb |
| 14:43.54 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 14:44.36 | joe-e | well, this is what i dont think ive set up right ;) |
| 14:45.28 | joe-e | http://pastebin.ca/605887 |
| 14:46.02 | [TK]D-Fender | TDMV_DCHAN = 24 |
| 14:46.07 | [TK]D-Fender | change to "0" |
| 14:46.19 | lilalinux | what is "hangup cause 26"? |
| 14:46.33 | lilalinux | (using faxreceive.agi) |
| 14:46.47 | joe-e | i was using wancfg gui to modify it. |
| 14:46.50 | [TK]D-Fender | joe-e: And then verify "wanrouter --hwdetect" to make sure the PIC & slot were right |
| 14:48.34 | [TK]D-Fender | joe-e: wancgf usually does a fine job. |
| 14:48.35 | joe-e | no --hwdetect option. |
| 14:49.02 | joe-e | wancfg detects "AFT-A101-SH SLOT=4 BUS=4 IRQ=16 CPU=A PORT=1 HWEC=32 V=30" |
| 14:51.43 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:54.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:54.50 | *** part/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com) |
| 14:58.15 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 14:59.29 | [TK]D-Fender | joe-e: no, stop wanrouter, and restart it. |
| 15:00.28 | joe-e | let me pastebin stuff for this. |
| 15:01.08 | joe-e | http://pastebin.ca/605904 |
| 15:01.40 | irule | what program may help me determine what ports are being used by another program? |
| 15:02.00 | drako | irule, netstat |
| 15:04.23 | [TK]D-Fender | joe-e: Think you're missing your interface setup in wancfg. |
| 15:04.28 | [TK]D-Fender | joe-e: Go review it |
| 15:04.35 | [TK]D-Fender | Configuring interfaces: w1g1 w1g1: unknown interface: No such device |
| 15:04.37 | [TK]D-Fender | ^^^^^^^^^ |
| 15:04.39 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 15:05.41 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:06.00 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 15:06.52 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 15:07.53 | *** join/#asterisk Tusker (n=tusker@mx.node6503.gplhost.com) |
| 15:11.08 | Tusker | heya guys, i'm having a strange problem with a VoIP provider. When I make a call through the provider it seems to involve a 3rd party IP in the conversation, which seems to confuse both Asterisk, and SPA IP phones. Ie, I register on sip.pfingo.com, but when I make a call, I get an ACK from a different IP |
| 15:11.28 | Tusker | is there any way to tell asterisk that the ACK is OK for that call ? |
| 15:11.53 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:14.47 | *** join/#asterisk mightnare (n=mike@p4218-ipad06motosinmat.mie.ocn.ne.jp) |
| 15:16.26 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 15:16.37 | BSD_Tech | Morning |
| 15:16.56 | BSD_Tech | is there a script tp convert asterisk.con to asterisk.ael |
| 15:17.01 | BSD_Tech | tp/to |
| 15:17.38 | naitram | here is the syntax for features in features.conf <FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]] |
| 15:18.12 | naitram | <PROTECTED> |
| 15:20.05 | [TK]D-Fender | Tusker: * is pretty dumb. May need to run a proxy in front. |
| 15:20.34 | Tusker | [TK]D-Fender: so this isn't typical behaviour that I can config around ? :) |
| 15:20.50 | Tusker | [TK]D-Fender: what if I set the outboundproxy to that IP I see in the tcpdump ? |
| 15:21.06 | *** join/#asterisk zapp-branigan (n=zapp-bra@128.23.220.87.dynamic.jazztel.es) |
| 15:21.30 | [TK]D-Fender | Tusker: Sorry, can't suggest more than I did... |
| 15:21.41 | Tusker | :) |
| 15:22.08 | naitram | test, test. Is this thing on? Tap, Tap, Tap. |
| 15:24.55 | Capps- | comedian. |
| 15:26.25 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 15:26.25 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:30.01 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:30.24 | *** join/#asterisk pois0n (i=Andretii@adsl-75-22-21-45.dsl.chcgil.sbcglobal.net) |
| 15:30.38 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 15:31.17 | santoshr | when an agent recieves a call he cannot use the transfer. how to change this beahviour |
| 15:32.32 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:32.35 | santoshr | have a >>> Dial(Agent/${EXTEN},30|tTwW) in the extension, but when an a normal extension dials an agent id, the agent cannot use transfer key frrom the features.conf |
| 15:34.07 | *** join/#asterisk JoJo_ReloadeD (n=incom@84.124.137.19.dyn.user.ono.com) |
| 15:34.07 | JoJo_ReloadeD | hi |
| 15:34.35 | naitram | so, just in case some others have this same issue. I found the answer to my question. I looked at the source code. To seperate arguments to an application in the features.conf file script, use the | pipe. so Monitor,wav|myfile|mb. If this was not answered because it was just a dumb question then my appologies. |
| 15:34.45 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
| 15:34.47 | JoJo_ReloadeD | it is allowed to ask questions about asterisk configuration in the channel ? |
| 15:34.55 | Qwell[] | JoJo_ReloadeD: That's the point of the channel - yes |
| 15:35.03 | JoJo_ReloadeD | ok |
| 15:35.40 | JoJo_ReloadeD | i'm trying to configura a quadbri and a tdm801b together, the quadbri goes ok (with asteriskbristuffed), but the tdm801b does nothing |
| 15:36.13 | JoJo_ReloadeD | the card is recognized by the system, but i cant load his module |
| 15:36.26 | JoJo_ReloadeD | it says 'no such device or address' |
| 15:37.17 | JoJo_ReloadeD | i'm using channels 1-12 for the quadbri (8B+4D) and channel 16 for the only fxo port of the tdm |
| 15:37.26 | JoJo_ReloadeD | i've also tried channel 16 |
| 15:37.50 | JoJo_ReloadeD | even channel 1, but using 1 it says there's a conflict, because that's a channel of the quadbri |
| 15:37.54 | JoJo_ReloadeD | any suggestions ? |
| 15:41.27 | santoshr | when an agent recieves a call he cannot use the transfer. how to change this beahviour |
| 15:42.44 | JoJo_ReloadeD | santoshr, do you have 'transfer=yes' in your zapata.conf ? |
| 15:43.05 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 15:45.02 | Zeeek | [TK]D-Fender reading your blog post. I still have not solved the (apparently NAT) problem on the 1.4 box |
| 15:46.12 | [TK]D-Fender | Zeeek: Show me the money. |
| 15:46.32 | Zeeek | nice commenting |
| 15:46.44 | Zeeek | callers within this range will be presented our IP. |
| 15:46.44 | irule | while true; do sudo netstat -anp|grep -i xten; sleep 1 ; clear ; done and while true; do sudo netstat -anp|grep -i asterisk; sleep 1 ; clear ; done I see no matching ports, is this normal? |
| 15:47.01 | Zeeek | should read "local IP" |
| 15:47.09 | Zeeek | callers within this range will be presented our LOCAL IP. |
| 15:47.17 | Zeeek | but the though was there |
| 15:47.46 | santoshr | JoJo_ReloadeD: a normal extension when recieves a call on the underlying extension he can transfer the call only an agent when recieves on his agent id he cannot |
| 15:47.47 | Zeeek | I blame the router |
| 15:48.52 | Zeeek | there actually are people watching http://asterisktv.com |
| 15:49.11 | Zeeek | great way to waste your employer's money and time |
| 15:50.34 | Qwell[] | Zeeek: Way to be anti-open standards :) |
| 15:51.55 | codefreeze | BSD_Tech: For extensions.conf to AEL, look at bug # 7638 |
| 15:51.55 | Zeeek | how so? |
| 15:52.05 | Zeeek | the Flash? |
| 15:52.08 | Qwell[] | yeah |
| 15:52.12 | Qwell[] | "Click here to download plugin." |
| 15:52.31 | Zeeek | The day an open standard is available that does what this platform does, I'll be the first to EMBRACE it |
| 15:52.38 | Qwell[] | avi |
| 15:52.43 | Zeeek | 90% of the people on the net have it |
| 15:52.52 | Zeeek | but let's not go there :) |
| 15:52.54 | mog | mpeg stream |
| 15:53.14 | mog | mms stream, even mplayer can do that |
| 15:53.19 | Zeeek | None of this is possible using someone else's bandwidth free with open standards |
| 15:53.23 | mog | can gnash play it? |
| 15:53.32 | mog | or is it flash9 |
| 15:53.42 | Zeeek | But if anyone decides to do what these guys are doing, I'm ready |
| 15:53.51 | Zeeek | Flash 9 unfortunately |
| 15:54.01 | mog | oh well |
| 15:54.10 | *** join/#asterisk jarrod (i=anon@theos.org) |
| 15:54.10 | Zeeek | I'm also recording it though and hope to present an mp4 version or whatever |
| 15:54.24 | jarrod | how do i stop asterisk from displaying 'asterisk' on private calls |
| 15:54.28 | Zeeek | yeah the fact of flash 9 really sucks |
| 15:54.33 | Qwell[] | s/ 9/ |
| 15:54.38 | mog | flash 9 is of satin |
| 15:54.44 | mog | lol Qwell |
| 15:54.45 | Qwell[] | s/ 9// |
| 15:54.52 | Qwell[] | umm, yeah |
| 15:54.57 | Zeeek | it is, in the words used often about asterisk releases "unstable", yes :) |
| 15:56.40 | Corydon76-work | Zeeek: you realize the irony here, right? |
| 15:56.52 | Zeeek | Sure I do |
| 15:57.04 | Zeeek | and I wallow in it like a pig in sh^H^H mud |
| 15:57.25 | Zeeek | The Zen answer is, "it does not matter Grasshopper" |
| 15:58.00 | Zeeek | If this is a *bad* thing, who are the 8 people watching right now? |
| 15:58.12 | Qwell[] | Windows users. nuff said |
| 15:58.17 | Zeeek | muhahaha |
| 15:58.36 | Zeeek | Digium wants corporate clients. They're all seeing the web via Windows |
| 15:59.25 | santoshr | how to make the agent use the features.conf file |
| 15:59.28 | Corydon76-work | Yes, but Digium is not the Asterisk community |
| 15:59.35 | Zeeek | indeed |
| 15:59.47 | Zeeek | but I am part of the asterisk community |
| 15:59.53 | Zeeek | albeit a small part |
| 16:00.07 | BSD_Tech | ok |
| 16:00.27 | Corydon76-work | Considering that most of the developers are running Linux, none of them can see it |
| 16:01.07 | BSD_Tech | why |
| 16:01.13 | BSD_Tech | whats wrong zeek |
| 16:01.22 | Zeeek | I think this is where you miss the point. The video is probably not going to be of interested to developers anyway |
| 16:01.36 | mog | Corydon76-work, the ones that hate freedom can see it ^_^ |
| 16:01.45 | Zeeek | hahah |
| 16:01.47 | Corydon76-work | mog: rofl |
| 16:02.14 | Zeeek | this is more about bringing new people in boyz |
| 16:02.24 | Corydon76-work | Zeeek: the reason you want developers to see it is that people are going to ask about something they may have seen in the presentation |
| 16:02.29 | mog | new people who hate freedom ^_^ |
| 16:02.32 | Zeeek | $peope*, not developers :) |
| 16:02.44 | Corydon76-work | Zeeek: if we can't see it, we can't answer their questions directly |
| 16:02.49 | mog | what videos are up Zeeek ? |
| 16:02.51 | Zeeek | I'll direct them to the IRC channel! |
| 16:03.08 | BSD_Tech | ? |
| 16:03.15 | BSD_Tech | what channel |
| 16:03.25 | mog | asterisktv.com |
| 16:03.29 | mog | or something like that right? |
| 16:03.29 | Zeeek | anyway, the audio is a free stream, I'm sure even developers can figure out how to listen to it and call in via SIP should they wish to do so :) |
| 16:03.45 | Corydon76-work | What about IAX? |
| 16:03.53 | mog | heh |
| 16:03.55 | Zeeek | http://asterisktv.com "for the new people who hate freedom" |
| 16:04.14 | mog | heh |
| 16:04.15 | BSD_Tech | updating my asterisk-now server |
| 16:04.26 | BSD_Tech | andding features and plugins |
| 16:04.26 | mog | you should have it show a message if one doesnt have flash |
| 16:04.28 | Zeeek | IAX? Sure trancode at your end :) Asterisk is good at that |
| 16:04.32 | mog | reccomending that they hate freedom |
| 16:04.34 | mog | and install it |
| 16:04.44 | Zeeek | thanks for helping me relax :) |
| 16:04.46 | Corydon76-work | Transcoding isn't about the protocol |
| 16:05.00 | BSD_Tech | its about cpu and memmory |
| 16:05.43 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 16:05.49 | BSD_Tech | whats the asterisk conf channel |
| 16:05.57 | BSD_Tech | zeek |
| 16:06.13 | Corydon76-work | #asterisk-conf I think |
| 16:06.47 | BSD_Tech | empty |
| 16:07.13 | Corydon76-work | That's a good sign |
| 16:07.21 | Corydon76-work | </sarcasm> |
| 16:07.30 | Zeeek | for the Mark has entered the building |
| 16:07.47 | Zeeek | memmory? Mammary? |
| 16:08.09 | Zeeek | #asterisk-users-conference |
| 16:08.22 | BSD_Tech | KRAM is in the house. |
| 16:08.24 | Zeeek | look at http://x2z.eu all that info is there |
| 16:08.24 | BSD_Tech | run |
| 16:08.36 | Zeeek | indeed kram is in the house |
| 16:09.03 | mog | people watching to much marko cam |
| 16:09.07 | BSD_Tech | it does nto give the irc channel |
| 16:09.16 | Zeeek | x2z.eu gives it |
| 16:09.58 | Corydon76-work | Zeeek: the extensions.conf fragment in there is deficient |
| 16:10.10 | BSD_Tech | its in a small text |
| 16:10.15 | Corydon76-work | There's no "exten =>" |
| 16:11.42 | jarrod | No application 'SetAccount' for extension - why am i getting this on 1.4 ? |
| 16:12.19 | Corydon76-work | jarrod: because SetAccount was deprecated |
| 16:12.34 | Corydon76-work | jarrod: in 1.2... Use Set(CDR(accountcode)=...) instead |
| 16:12.35 | jarrod | what is to be used now |
| 16:12.48 | jarrod | ok |
| 16:12.51 | jarrod | ill try, thanks |
| 16:13.00 | *** join/#asterisk friedrich| (n=friedric@e177244164.adsl.alicedsl.de) |
| 16:16.59 | Zeeek | what's wrong with the extensions conf? |
| 16:17.29 | Zeeek | people are supposed to figure that part out |
| 16:17.56 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:19.10 | Zeeek | going live in a few for the conference. IRC #asterisk-users-conference |
| 16:22.20 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 16:26.09 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 16:28.35 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 16:32.15 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-87.ph.ph.cox.net) |
| 16:33.00 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 16:33.31 | Voicemeup | .20 broke cisco phones BTW.. seems phone cant sent the ack back to asterisk .17 is perfec |
| 16:33.52 | Voicemeup | cisco stucks on ringing 180.. then cant take more call till reboot.. |
| 16:35.05 | twitchnln | any snmp guru's in here? |
| 16:35.14 | Qwell[] | ~ask |
| 16:35.14 | jbot | 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. |
| 16:37.23 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:39.31 | mog | Zeeek, flash plugin works, its just not free as in freedom |
| 16:39.48 | Qwell[] | or usable on 64-bit |
| 16:39.52 | *** join/#asterisk vgster (n=vgster@group.navonline.net) |
| 16:40.00 | mog | you can Qwell you just need 32 bit firefox |
| 16:40.07 | Qwell[] | I'm lazy |
| 16:40.07 | mog | and 32 bit oss |
| 16:40.21 | mog | i mean its so easy no wonder its number 1 |
| 16:40.27 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 16:40.58 | citats | hey everyone, checking in for my required minute, gotta go head out now |
| 16:41.02 | mog | i bet gnash can play it, i need to get the beta version working on the lappy |
| 16:49.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:51.01 | *** join/#asterisk kirberich (n=robert@i538715FA.versanet.de) |
| 16:51.04 | kirberich | hi |
| 16:51.59 | shido6 | hi |
| 16:53.57 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:54.49 | kirberich | i have a problem regarding setting of caller id, but i'm having some trouble articulating it ;) |
| 16:55.03 | *** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl) |
| 16:55.08 | kirberich | i have 3 msns, and three users making outgoing calls |
| 16:55.29 | kirberich | my telco bills each msn individually, so that every user only has to pay his own calls |
| 16:55.45 | kirberich | but now when i use asterisk to call out, it always uses the first msn |
| 16:55.57 | kirberich | is there any way to change that? (i'm using capi btw) |
| 16:56.47 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:56.53 | x86 | <PROTECTED> |
| 16:56.54 | x86 | <PROTECTED> |
| 16:57.01 | x86 | anyone see a problem with that? |
| 16:57.24 | x86 | getting this: Jul 6 11:57:14 WARNING[29640]: config.c:525 process_text_line: No '=' (equal sign) in line 355 of extensions.conf |
| 16:57.32 | x86 | line 355 is the ZapBarge line |
| 16:58.04 | Qwell[] | x86: exten => ? |
| 16:58.19 | x86 | jesus christ i'm dense sometimes |
| 16:58.20 | x86 | haha |
| 16:58.23 | x86 | thanks man ;) |
| 16:58.24 | Qwell[] | ;) |
| 16:58.32 | Qwell[] | That'll be $49.99 |
| 16:59.40 | jarrod | how do i stop asterisk from displaying "asterisk" on calls with no caller-id? |
| 17:02.02 | russellb | jarrod: defaultcallerid option in sip.conf |
| 17:02.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:05.23 | jarrod | i dont see defaultcallerid |
| 17:05.54 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
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| 17:13.39 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 17:13.44 | *** join/#asterisk cayorde (n=flexable@87.19.162.237) |
| 17:16.37 | *** part/#asterisk cnet2 (n=felipe@201.192.35.137) |
| 17:17.56 | BSD_Tech | ok |
| 17:18.17 | BSD_Tech | I was working to update and build pkgs for asterisknow and I have a few issues |
| 17:18.42 | BSD_Tech | what pkg/dep does res_speech require |
| 17:18.52 | BSD_Tech | it seems to have issues now |
| 17:19.12 | BSD_Tech | and deps for res_crypto |
| 17:19.29 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:21.37 | mvanbaak | BSD_Tech: where can I find the patches for asterisk+bsd ? |
| 17:22.02 | jarrod | just run it on linux |
| 17:22.03 | mvanbaak | I want to know what you think needs patching |
| 17:22.19 | jarrod | how do i stop asterisk from displaying "asterisk" on calls with no caller-id? |
| 17:22.19 | mvanbaak | because stock asterisk runs fine on obsd here |
| 17:22.32 | mvanbaak | jarrod: you can set that in sip.conf |
| 17:22.41 | jarrod | whats the directive? |
| 17:23.11 | mvanbaak | dont know from memory |
| 17:23.16 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:23.20 | mvanbaak | haven't used sip for months now |
| 17:23.20 | jarrod | ive looked at all the options |
| 17:23.56 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 17:24.29 | [[blah]asfd | One of my servers had a crash on it today. Looking at the logs I came across this: http://pastebin.ca/606082 Has anyone ever seen this or know what may have cause this? |
| 17:25.43 | BSD_Tech | the ports tree |
| 17:25.58 | mvanbaak | jarrod: you can also do it in the dialplan |
| 17:26.03 | BSD_Tech | I am orking on newer patches |
| 17:26.06 | BSD_Tech | brb |
| 17:26.08 | mvanbaak | using the CALLERID() dialplan function |
| 17:26.15 | mvanbaak | BSD_Tech: obsd as well ? |
| 17:26.34 | BSD_Tech | the port sjould work but I only work on freebsd |
| 17:26.37 | mvanbaak | ah yeah |
| 17:27.10 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 17:29.03 | mvanbaak | found it |
| 17:29.09 | mvanbaak | ok, off to have dinner |
| 17:29.22 | mvanbaak | and poker |
| 17:30.19 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:31.30 | BSD_Tech | [Jul 6 10:31:11] WARNING[3580]: loader.c:360 load_dynamic_module: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_unregister_file_version |
| 17:31.30 | BSD_Tech | [Jul 6 10:31:11] WARNING[3580]: loader.c:360 load_dynamic_module: Error loading module 'func_odbc.so': /usr/lib/asterisk/modules/func_odbc.so: undefined symbol: ast_register_file_version |
| 17:31.30 | BSD_Tech | Segmentation fault (core dumped) |
| 17:31.34 | BSD_Tech | we have issues |
| 17:32.35 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 17:33.21 | BSD_Tech | updating fubared my system |
| 17:33.23 | BSD_Tech | grrr |
| 17:33.27 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:34.01 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:37.35 | Zeeek | Mark not yet kleft the building |
| 17:37.42 | Zeeek | but I'm about to! |
| 17:37.44 | Zeeek | bye |
| 17:37.46 | BSD_Tech | bye |
| 17:38.06 | BSD_Tech | I missed today due to asterisk technical difficaulties |
| 17:39.13 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 17:39.27 | generalhan | hey all ! |
| 17:39.55 | Mercestes | lo |
| 17:40.10 | generalhan | im having an issue registering a remote server using IAX. could some one take a look at my config/error and give me some insight ? http://generalhan.pastebin.ca/606115 |
| 17:41.30 | generalhan | both servers are really on the same network, but i want to test this way before i send it out |
| 17:42.15 | Cyon | Not exactly sure how you have the host set to the same IP on both boxes...but that's just me. |
| 17:42.39 | generalhan | Cyon: i dont |
| 17:42.43 | [TK]D-Fender | generalhan: that is not an "error" |
| 17:42.55 | generalhan | [TK]D-Fender: sorry NOTICE then |
| 17:42.57 | Cyon | Wow I'm blind, sorry |
| 17:43.15 | [TK]D-Fender | generalhan: NOTICE[9971]: chan_iax2.c:5132 register_verify: Peer 'Admin' is not dynamic (from 192.168.0.64) <- |
| 17:43.26 | [TK]D-Fender | generalhan: You don't register if you have a fixed bloody host! |
| 17:43.28 | [TK]D-Fender | DUH! |
| 17:43.32 | BSD_Tech | ok what are the needed deps for res_krypto |
| 17:43.33 | generalhan | hmm |
| 17:43.38 | BSD_Tech | ok what are the needed deps for res_crypto |
| 17:43.54 | [TK]D-Fender | Registering is to INFORM them of your IP. That suer can't SET an IP! |
| 17:44.04 | generalhan | gotcha |
| 17:44.09 | generalhan | makes sense |
| 17:48.14 | BSD_Tech | ok what are the needed deps for res_crypto |
| 17:48.20 | Qwell[] | BSD_Tech: check menuselect |
| 17:48.23 | BSD_Tech | its needs updating |
| 17:48.58 | *** join/#asterisk Rhiliam (n=user@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 17:50.28 | Rhiliam | When a person leaves a voicemail, is there a way to tell asterisk to call a phone number and assuming the call is answered, tell them that they have a voicemail waiting? |
| 17:53.58 | Mercestes | Rhiliam, Why not send them a txt message? |
| 17:55.48 | BSD_Tech | what is osptk short for ? |
| 17:56.15 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 17:56.39 | Qwell[] | ~osptk |
| 17:56.48 | BSD_Tech | I cant find a uptodate pkg for it |
| 17:56.56 | *** join/#asterisk some_dude (n=Miranda@adsl-66-140-4-177.dsl.hstntx.swbell.net) |
| 17:56.58 | Corydon76-work | Open Something Protocol Tool Kit |
| 17:57.07 | Qwell[] | settlement |
| 17:58.14 | coppice | as in "Contents may settle after shipping" |
| 17:58.16 | some_dude | I've got 2 locations, and a reg phone system. I wanted to patch one of my lines into a voip system to call the other location, Asterisk at both locations looks like a good option, but what sort of hardware do i need ? |
| 17:58.36 | Qwell[] | coppice: during shipping |
| 17:59.10 | Corydon76-work | some_dude: dedicated T1 |
| 17:59.24 | coppice | what do that care, provided they can claim "its not our fault. we didn't do it" |
| 17:59.28 | some_dude | for just one line ? |
| 17:59.31 | Rhiliam | That would be ideal, however their current voicemail system does this, and they want to keep this function if at all possibe |
| 17:59.37 | coppice | s/that/they |
| 17:59.38 | Corydon76-work | Oh, only 1 line? |
| 17:59.43 | Qwell[] | Rhiliam: You could do it in dialplan logic |
| 17:59.55 | Qwell[] | or call an external script, which drops a call file, which calls them |
| 17:59.55 | Corydon76-work | some_dude: is it mission-critical? |
| 17:59.59 | some_dude | no |
| 18:00.06 | Corydon76-work | some_dude: voip is fine, then |
| 18:00.11 | Rhiliam | Great - How would I do this. Capture a return code and then dial out? |
| 18:00.21 | Rhiliam | or something similar |
| 18:00.24 | Qwell[] | Rhiliam: look at the externnotify option in voicemail.conf |
| 18:00.37 | Corydon76-work | some_dude: at least one or both of the Asterisk servers needs a public IP |
| 18:01.31 | some_dude | I'm not sure i understand what asterisk is, I'm going to have a linux box, with a phone pluged in to it. Then then Astrisk will send the message over IP to the other server, where it's converted back. correct ? |
| 18:01.52 | BSD_Tech | ltdl |
| 18:01.55 | BSD_Tech | ? |
| 18:02.20 | Rhiliam | Qwel: So essentialy write some external code? I was hoping this functinality was built in. |
| 18:02.28 | Corydon76-work | some_dude: that's basically how it works, yes |
| 18:03.04 | Qwell[] | Rhiliam: the code would be small |
| 18:03.06 | Yomer | [TK]D-Fender : you there? |
| 18:03.15 | [TK]D-Fender | Yomer: Yup |
| 18:03.16 | Qwell[] | maybe 2-3 lines of bash script |
| 18:03.22 | some_dude | BUT also, I can plug a bunch of phone into it, and let them talk to each other. and even patch in external phone lines, right ? |
| 18:03.30 | Corydon76-work | BSD_Tech: Lib Tool Dynamic Library |
| 18:03.31 | Mercestes | .0000000000000000000000000.000 |
| 18:03.38 | Yomer | [TK]D-Fender : hi, you remeber my SPA400 problems yesterday? |
| 18:03.50 | [TK]D-Fender | Yomer: Yeah I think I saw a msg from you on thier site today. |
| 18:03.54 | BSD_Tech | ok |
| 18:04.04 | BSD_Tech | just trying to fix broken deps |
| 18:04.16 | *** join/#asterisk andyd (n=andyd@212.183.134.208) |
| 18:04.18 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 18:04.24 | Yomer | [TK]D-Fender : well im still stuck with... have oyu worked with SPA400? |
| 18:04.30 | joetester | Questions: 1. What is the official IAX2 packet to say "The channel is ready"... what is sent exactly? 2. When my 1.4 machine is connected to a 1.2 machine (through IAX), it sends packets DTMF not DTMF_B and DTMF_E, that normal? |
| 18:04.39 | *** join/#asterisk stoffell_w (n=stoffell@fw.catsanddogs.com) |
| 18:04.40 | Yomer | [TK]D-Fender: im tring to get a hold of a sip log from it |
| 18:04.55 | [TK]D-Fender | some_dude: * is a telephpny toolkit that you can use to build a PBX, dialout system, call center, answering machine, etc. But whatever card you have in that PC that your line is plugged into is probably WORTHLESS |
| 18:04.55 | Yomer | [TK]D-Fender : to compare with mine...and see if i spot whats wrong |
| 18:04.56 | Rhiliam | Qwell : Looking at this a bit more, so essentially, create a .call file with the appropropriate information in the context. Then put some code in the dialplan? I am getting this right? |
| 18:05.05 | Qwell[] | Rhiliam: yep |
| 18:05.07 | rene- | hey, about the extensionstate manager event, should it fire itself whenever an extension that has a hint changes state? or is one supposed to be polling asterisk to know the extensions states? |
| 18:06.04 | some_dude | What do you mean that the card is worthless. |
| 18:06.11 | Rhiliam | Qwell: I think I got it. By any chance would you have an example call file and dialplan entry :) |
| 18:06.15 | Qwell[] | nope |
| 18:06.39 | Rhiliam | Never hurts to ask :) - thanks for the help |
| 18:06.40 | Qwell[] | ~wikis |
| 18:06.40 | jbot | hmm... wikis is http://www.voip-info.org |
| 18:06.41 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:06.42 | Qwell[] | try there though |
| 18:06.56 | some_dude | I need special modems to make it work ? |
| 18:07.34 | [TK]D-Fender | some_dude: only 1 kind really, and they kinda suck |
| 18:07.59 | [TK]D-Fender | some_dude: go to the WIKi and check out the ahrdware compatability list. Then DL * and install it with Zaptel and see if you're remarkably lucky. |
| 18:08.22 | some_dude | I don't have a box built for it yet. |
| 18:08.39 | *** join/#asterisk Chris-NB (n=chris@ip.tech.t-mobile.at) |
| 18:09.03 | Corydon76-work | some_dude: you're better off getting a TDM800 |
| 18:09.13 | [TK]D-Fender | some_dude: Well in the mean-time go download.... THE BOOK and get reading to understand what * is and can do. |
| 18:09.16 | [TK]D-Fender | ~book |
| 18:09.16 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:09.18 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 18:09.34 | BSD_Tech | unixodbc requires qt bloddy hell |
| 18:10.35 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 18:10.47 | some_dude | I want a TDM800 |
| 18:11.04 | Corydon76-work | BSD_Tech: no, it doesn't |
| 18:11.05 | BSD_Tech | why at that poit just get a t1 |
| 18:11.16 | Corydon76-work | BSD_Tech: you can disable the gui at compile time |
| 18:11.26 | BSD_Tech | configure: checking for Qt |
| 18:11.26 | BSD_Tech | checking for Qt headers... no |
| 18:11.26 | BSD_Tech | configure: error: cannot find correct Qt headers! |
| 18:11.34 | Corydon76-work | ./configure --without-x I think |
| 18:11.41 | BSD_Tech | that in unixodbc |
| 18:11.47 | Corydon76-work | Correct |
| 18:12.17 | Corydon76-work | Ah, no, it's --disable-gui |
| 18:12.19 | BSD_Tech | ok rerunning |
| 18:12.23 | *** part/#asterisk msetim (n=marcos@200.195.161.164) |
| 18:12.31 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 18:12.31 | twitchnln | can someone gimme a hand figuring out the correct oid to graph sip channels with snmp? |
| 18:12.55 | Nugget | Is it normal for a PRI to show fairly frequent restarts? I'm getting: |
| 18:12.55 | Nugget | <PROTECTED> |
| 18:13.04 | Nugget | on all the channels a few times a day. |
| 18:13.05 | Voicemeup | yes |
| 18:13.11 | Voicemeup | its clearing status of used lines etc |
| 18:13.11 | Corydon76-work | Nugget: once per hour, by default |
| 18:13.12 | [TK]D-Fender | some_dude: before even thinking about ahrdware, come to use with the specs of what you want and we'll make some more comprehensive suggestions for you then. |
| 18:13.14 | Voicemeup | unused |
| 18:13.16 | Nugget | super, thanks. |
| 18:13.28 | Voicemeup | wat i HATE is this ARNING[3064650672]: chan_zap.c:8136 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
| 18:13.31 | Voicemeup | then it kills the pri |
| 18:13.34 | [TK]D-Fender | some_dude: I would not recommend that card if you want to plug PHONES in, only LINES. |
| 18:13.50 | Voicemeup | of course its always following channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb675e3b8', 10 retries! |
| 18:14.01 | [TK]D-Fender | Corydon76-work: TDM800 looks interesting with mixed modules capability |
| 18:14.07 | some_dude | what's for phone ? |
| 18:14.33 | Corydon76-work | [TK]D-Fender: works better for fax, too |
| 18:14.55 | [TK]D-Fender | some_dude: http://www.telephonydepot.com/product_p/105-054-212.htm Supports 2 analog phones to be used as SIP devices |
| 18:15.15 | [TK]D-Fender | Corydon76-work: Will take your word for it. Is it documented anywhere I can read? |
| 18:15.44 | Corydon76-work | Is what documented? |
| 18:15.58 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 18:17.31 | Mercestes | that it supports fax. >.> |
| 18:17.46 | Corydon76-work | No, it's just anecdotal |
| 18:18.25 | Mercestes | ah, damnit. |
| 18:18.33 | Mercestes | one can hope, right? |
| 18:18.38 | [TK]D-Fender | Hey I know..... lets start a rumor! |
| 18:18.44 | [TK]D-Fender | </snicker> |
| 18:19.00 | Corydon76-work | and when I say anecdotal, I mean that it's my experience that it works better with fax |
| 18:19.10 | coppice | if you really want FAX to work you need to do something like sangoma have just done far too late - a link between cards to sync them |
| 18:20.35 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 18:20.38 | [TK]D-Fender | coppice: By far too late, do you mean that its a hardware solution, not firmware? |
| 18:20.49 | BSD_Tech | iaxmodem and hylafax |
| 18:20.51 | Corydon76-work | and I've heard a good explanation for what was the problem that they corrected in the TDM800. Apparently, the PCI chip interface on the TDM400 sucked |
| 18:21.15 | coppice | its hardware for just some of their cards. by far too late I mean its an obvious thing no card should ever have been shipped without |
| 18:21.42 | joetester | Does chan_iax know anything about the version of chan_iax on the remote machine? |
| 18:22.14 | [TK]D-Fender | coppice: Something that can be added to an existing card, or does it require a complete replacement? |
| 18:22.15 | *** join/#asterisk madcap (i=madcap@unaffiliated/madcap) |
| 18:22.43 | Corydon76-work | In the test case I've worked up, the fax is passing twice through a Digium digital card and twice through a Digium analog card, and the fax still works fine |
| 18:22.56 | coppice | not sure if there is an upgrade option. however, they say only cards with hardware EC are offered with it even now. that sucks |
| 18:23.25 | coppice | Corydon76-work: without hardware sync, that's luck not engineering |
| 18:23.46 | Corydon76-work | Fax - FXS - - FXO - channel bank - TE410P - - TE410P - PSTN |
| 18:24.15 | Corydon76-work | coppice: I'll take whatever I can get |
| 18:24.39 | *** join/#asterisk diskfree (n=SimmerMo@www3.datarack.nl) |
| 18:24.40 | Corydon76-work | I never claimed to be a hardware engineer |
| 18:25.10 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:25.38 | lesouvage | . |
| 18:26.07 | coppice | it puzzles me why the card makers are all determined to make their products work so badly. its not like software, where you can milk support charges out of people. its just a loss with hardware |
| 18:27.06 | joetester | Development time? Development costs? |
| 18:27.18 | Corydon76-work | The continuing struggle between profit and hardware cost? |
| 18:27.22 | coppice | Incompetance? |
| 18:27.38 | Mercestes | de-evolution? |
| 18:28.15 | diskfree | coppice: what card is this about, or are you talking 'in general' |
| 18:28.56 | Corydon76-work | diskfree: he was complaining about a Sangoma product decision |
| 18:29.00 | coppice | digium, sangoma, or any of the others. sangoma is the first to provide any syncing, and still only offer it in a limited way |
| 18:29.48 | diskfree | coppice: ok, cause we were going to use sangoma (single E1, the A101d) |
| 18:30.02 | diskfree | we're now using cisco voicerouters |
| 18:30.22 | diskfree | anybody else using cisco voicerouters here? |
| 18:30.27 | polerin | coppice: I know! everyone's favorite: Avarice. (ugh spelling. I really don't care today) |
| 18:31.07 | crimethinker | I like my Digium cards. |
| 18:31.23 | joetester | So do I! |
| 18:31.27 | Corydon76-work | diskfree: I have a customer using them. I can't say anything about the quality of the cards, but the quality of his Cisco tech is seriously lacking |
| 18:31.37 | file | actually the TE4XXP stuff you can get a timing cable to link them together for sync |
| 18:31.44 | Corydon76-work | His Cisco tech tried configuring inband DTMF with G.729 codec |
| 18:32.10 | Corydon76-work | err, s/cards/routers/ |
| 18:32.14 | coppice | file: can you sync any of the analogue cards to that? |
| 18:32.25 | file | no |
| 18:32.25 | diskfree | @Corydon76-work: what protocol are you using between the cisco's and *? |
| 18:32.51 | Corydon76-work | diskfree: I believe it's using PRI |
| 18:33.11 | Corydon76-work | diskfree: in case you didn't understand, inband and G.729 don't mix |
| 18:33.25 | coppice | file: well, that tends to be the critical one. the digitals can usually be synced by having at least one port on each card to the same point (e.g. PSTN) the analogue cards are always isolated |
| 18:33.26 | Corydon76-work | You can only do inband DTMF with ulaw or alaw |
| 18:33.49 | diskfree | @Corydon76-work: I generally like cisco equipment for the quality, but it looks like the sip implementation between router/IOS versions differ |
| 18:34.21 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 18:34.27 | diskfree | @Corydon76-work: In understand that G729 part, I am using g711ulaw myself |
| 18:35.09 | coppice | diskfree: I generally liek the cisco equipment for landfill, but that's just from too many callouts due to bugs |
| 18:36.15 | diskfree | coppice: hehe, well, I was mainly talking about switches. I am having serious doubts regarding voicerouters. |
| 18:36.32 | *** part/#asterisk twitchnln (n=twitch@70.43.112.117.nw.nuvox.net) |
| 18:36.49 | macli | Hi, print "STREAM FILE eh \"\"\n"; but asterisk does not play eh sound file ?? |
| 18:37.18 | macli | print "SAY NUMBER 100 \"\"\n"; works |
| 18:37.51 | diskfree | coppice: And when you think about using sangoma and you check here, somebody's disgusted with digium/sangoma cards and recommends another brand... That doesn't help really, in decision making |
| 18:38.34 | diskfree | I know that digium cards are picky about the hardware they're placed in |
| 18:38.47 | coppice | diskfree: does it help when someone says everything is wonderful? |
| 18:39.07 | [TK]D-Fender | Denial : It's not jsut a river in Egypt <---- |
| 18:39.13 | polerin | "Can haz eh Lisehn Plz?" (joking.) |
| 18:39.16 | Corydon76-work | diskfree: it's picky to ask that motherboards implement PCI to spec? |
| 18:39.42 | [TK]D-Fender | polerin: I'm SOOO going to lolcat that. |
| 18:39.53 | polerin | go for it :D |
| 18:40.21 | polerin | ... you know... dial plans would be PERFECT for LOLCODE |
| 18:41.58 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 18:42.02 | Qwell[] | polerin: I CAN HAZ DIALTONE? |
| 18:42.13 | polerin | Qwell[]: bingo. |
| 18:42.30 | Corydon76-work | Qwell[]: I've dealt with too many people like that for that to be funny. |
| 18:43.07 | polerin | hehe, I can just see Corydon76-work look at a dialplan written in kitteh and his head just exploding in pure disgust |
| 18:43.08 | vooduhal | Does chan_agent support extconfig in 1.2 and if so, can someone point me to configuring it? I can't find anything via google. We have sip and voicemail using extconfig and we've written a rails interface for managing voicemail but we would like to not have to edit agents.conf as well. |
| 18:43.26 | Corydon76-work | I've actually stopped people to ask "You're putting me on, right? Nobody can be THAT stupid, right?" |
| 18:43.50 | Corydon76-work | Sad to say, they weren't putting me on |
| 18:43.59 | polerin | Corydon76-work: I wish I could cut people off like that |
| 18:44.05 | coppice | I liked those old iridium ads - "we take a dial tone to the four corners of planet earth". If they thought the earth was square its not surprising they failed |
| 18:44.07 | polerin | Corydon76-work: unfortuantly thats a firing offense here :/ |
| 18:44.29 | Corydon76-work | polerin: okay, I'm a bit nicer than that... |
| 18:45.03 | polerin | heh |
| 18:45.08 | polerin | I can't even be close to that |
| 18:45.10 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:45.10 | Corydon76-work | Still, it's one of those cases where if you get an awkward silence, you realize they're complete idiots |
| 18:45.19 | polerin | seriously our techs are fucking stupid and twichy sometimes |
| 18:45.47 | Corydon76-work | Puts me in teaching-a-2yo-mode... except that I've known smarter 2yo's |
| 18:45.53 | polerin | "Hi I Are call you 4 hlp, You n0 tell me how to do my job!" |
| 18:46.07 | polerin | **headdesk** |
| 18:46.14 | [TK]D-Fender | Corydon76-work: "A child of fivie could do this.... fetch me a child of five!" |
| 18:46.31 | coppice | dumb people are more pleasant to handle than the politically screwed up bloody difficult ones |
| 18:48.04 | irule | hi, what is the difference between hangup and softhangup? |
| 18:48.10 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 18:48.37 | coppice | padded rope around your neck |
| 18:49.05 | rene- | heheh |
| 18:49.12 | rene- | velvet rope |
| 18:49.40 | polerin | coppice: you're in a pleasent mood aren't you ;P |
| 18:57.59 | diskfree | @Corydon76-work: are you only managing * for that customer with cisco routers? |
| 19:01.37 | jart | crisco! |
| 19:07.21 | *** join/#asterisk erthnet (n=erthnet@66.206.86.107) |
| 19:07.59 | vooduhal | Does chan_agent support extconfig in 1.2 and if so, can someone point me to configuring it? I can't find anything via google. We have sip and voicemail using extconfig and we've written a rails interface for managing voicemail but we would like to not have to edit agents.conf as well. |
| 19:10.18 | *** join/#asterisk RedComet- (i=RedComet@71.13.113.239) |
| 19:10.25 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 19:13.54 | jart | feel free to test the new release candidate for my voice changer: http://www.lobstertech.com/code/voicechanger/ Report bugs to jtunney@gmail.com |
| 19:14.05 | vooduhal | exit |
| 19:15.04 | *** join/#asterisk saftsack (n=saftsack@pD9E07966.dip.t-dialin.net) |
| 19:16.02 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:16.09 | coppice | jart: if you can change a thick Karnataka accent to something more western you could make a fortune in Bangalore's call centres |
| 19:16.58 | diskfree | jart: nice, will check it out later this weekend |
| 19:17.55 | coppice | its a fun toy. I wonder if anyone applies it to serious work? |
| 19:18.01 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 19:19.02 | JerJer | on 1.4 has anyone else seen Queue attempt to send two new calls to one queue member ? |
| 19:20.00 | JerJer | using least called and ringall - separate queues. |
| 19:20.35 | jart | hehe |
| 19:21.04 | JerJer | we are using dial modifier M to pause the queue member |
| 19:21.04 | Mercestes | Can I use thsi software to pick up 13 year olds in myspace? |
| 19:21.12 | jart | Mercestes: yes |
| 19:21.16 | Mercestes | Yes! |
| 19:21.20 | Mercestes | I love it already |
| 19:21.33 | JerJer | Mer |
| 19:22.15 | Mercestes | ? |
| 19:22.19 | jart | at Lobster Technologies, Inc., we strive to provide technologies that enable people to wreak havoc upon humanity |
| 19:22.36 | JerJer | Press 1 to hear a toilet flush |
| 19:22.45 | Mercestes | 1 |
| 19:23.06 | JerJer | Press 2 to use your pimp hand |
| 19:25.03 | Mercestes | 2 |
| 19:25.16 | Mercestes | 2 2 2 2 2 2 2 Where's my money, biotch! |
| 19:28.20 | BSD_Tech | humans bullshit eachother to gt by |
| 19:29.17 | coppice | this sounds like its gonna get a little hip-hop-ish |
| 19:29.52 | BSD_Tech | just stating facts |
| 19:30.26 | coppice | high quality bullshit has always been the most valuable of commodities |
| 19:31.15 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 19:31.17 | Supaplex | now with enhanced bioflavonoids |
| 19:31.34 | Mercestes | and antioxidants |
| 19:34.59 | coppice | does it have hexagonal water molecules? |
| 19:35.24 | Mercestes | only if they're gay |
| 19:36.18 | *** join/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 19:37.21 | jart | so how do you all feel about FreeSWITCH, CallWeaver, OpenSER, etc? |
| 19:38.02 | Corydon76-work | jart: off-topic, take it somewhere else, please |
| 19:38.15 | Mercestes | and that's all we have to say about that. |
| 19:38.17 | luckyone | question, how do you setup a dialplan which allows you to access voicemail externally |
| 19:38.46 | luckyone | like get access to your voice mailbox VoiceMailMain |
| 19:38.51 | Mercestes | luckyone, Same way you do internally, except you either use an external number to access it directly or set it up as an extension as part of an IVR. |
| 19:39.02 | jart | sorry, i don't mean to start a fight |
| 19:39.34 | luckyone | Mercestes: can you have an IVR is listening while ringing is playing? |
| 19:39.37 | Mercestes | jart: Yea, that convo is destined to start a fight. |
| 19:39.40 | coppice | jart: take care, or the secret police will come for you in the night |
| 19:39.42 | Corydon76-work | jart: we try to avoid flame wars as best as possible. |
| 19:40.04 | Mercestes | luckyone, if you want. Just play ringing to them. |
| 19:40.06 | Corydon76-work | Sometimes it's unavoidable, though |
| 19:40.14 | Mercestes | Sorry, I'll try to be better. |
| 19:40.35 | *** join/#asterisk Hymie (n=Hymie@l8r.net) |
| 19:40.45 | Hymie | hmm.. anyone want to buy 12 uniden phones :Þ |
| 19:41.13 | coppice | Hymie: let me guess. you did? |
| 19:41.20 | luckyone | Mercestes: /leave |
| 19:41.24 | luckyone | hehe |
| 19:41.24 | Mercestes | Hymie: try #callweaver |
| 19:41.25 | luckyone | woops |
| 19:41.28 | *** part/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 19:41.29 | jart | yea i can see it being touchy, sorry i brought it up |
| 19:41.46 | Mercestes | gah, even the n00bs are trying to run me off. :( |
| 19:41.50 | Hymie | what's up with this mercestes "person" |
| 19:42.01 | Mercestes | I'm a bot. |
| 19:42.10 | Hymie | !shoot mercestes |
| 19:42.11 | Corydon76-work | Hymie: we think he's a "closet" case |
| 19:42.15 | Hymie | !leave mercestes |
| 19:42.19 | [TK]D-Fender | ~Mercestes |
| 19:42.20 | jbot | mercestes is definitely a total nub |
| 19:42.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 19:42.30 | Corydon76-work | ~lart mercestes |
| 19:42.30 | jbot | beats the living hamstercrap out of mercestes |
| 19:42.56 | Mercestes | ~lart mercestes |
| 19:42.56 | jbot | puts on a hockey mask and jumps out at mercestes |
| 19:43.07 | Hymie | [TK]D-Fender: I think you were telling me last week, that you wanted to buy some uniden phones? |
| 19:43.20 | Corydon76-work | Hymie: try ebay |
| 19:43.20 | [TK]D-Fender | Hymie: 5$ each, deal! |
| 19:43.21 | Hymie | [TK]D-Fender: I can help you! |
| 19:43.58 | jart | wait 5 dollar phones? |
| 19:44.11 | Hymie | wait, you'll really give me $5? |
| 19:44.12 | Hymie | jart! |
| 19:44.17 | Hymie | step into my officve! |
| 19:44.17 | Mercestes | Hymie: don't forget to put in a minimum bid. |
| 19:44.33 | Mercestes | They're great phones. |
| 19:44.36 | coppice | not US$5. some less valuable dollars |
| 19:44.47 | Mercestes | aren't those the ones with auto-jitter correction? |
| 19:44.52 | Corydon76-work | What, AUS$5 ? |
| 19:45.10 | jart | eek, am i in trouble again? :( |
| 19:45.41 | Corydon76-work | jart: trouble is as trouble does |
| 19:45.41 | Mercestes | jart: Why would you be in trouble? |
| 19:46.35 | jart | Mercestes: because i'm the crazy asterisk hacker who writes all the wacky applications |
| 19:46.47 | jart | like Phone Parrot lol |
| 19:46.55 | JerJer | http://merkwork.com/images/LOL.jpg |
| 19:46.58 | [TK]D-Fender | jart: SER is not OT here, the * forks, and FreeSWITCH tend to spark flame-wars. |
| 19:47.18 | [TK]D-Fender | jart: Personally if you keep it clean I wouldn't consider them OT. |
| 19:47.32 | jart | i'm not talking about that anymore |
| 19:47.48 | JerJer | [TK]D-Fender: does someone need to lay off the crack pipe? :) |
| 19:47.58 | Mercestes | R O F L M A O |
| 19:48.14 | jart | i wouldn't troll but Corydon76 is probably right about it being to risky to even bother in a public chat |
| 19:48.23 | [TK]D-Fender | JerJer: Lots of people :) |
| 19:48.48 | [TK]D-Fender | jart: Jusdge yourself, or we'll do it for you :) All part of the service! |
| 19:48.59 | JerJer | imho, OpenSER totally complements Asterisk's skills |
| 19:49.27 | jart | JerJer: i'm currently using openser to load balance across asterisk boxes, it works really well |
| 19:49.35 | Mercestes | I concur, OpenSER is well worth looking at. |
| 19:49.49 | Mercestes | I'm being serious this time. |
| 19:49.55 | JerJer | in fact i've been playin with some fancy IMS stuff using OpenSER - which has very serious potential to disrupt some major business models |
| 19:49.57 | jart | asterisk has some problems talking to my voip provider sometimes, but besides that things are smooth |
| 19:50.13 | JerJer | jart: what kind of problems ? |
| 19:50.24 | JerJer | authenticated invite crap ? |
| 19:50.29 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 19:50.42 | murdmath | Howdy all. I have a quick context question. |
| 19:50.53 | jart | channels sometimes don't hangup when they should |
| 19:50.59 | JerJer | you have 19 seconds |
| 19:51.30 | JerJer | jart: are you not getting a BYE? |
| 19:51.31 | murdmath | Do all contexts need to be unique. If you have a iax extension named for instance 1234 and another context named 1234 in your extensions.conf that is bad right? |
| 19:51.40 | JerJer | right |
| 19:51.54 | murdmath | JerJer: Were you replying to me? |
| 19:51.56 | Corydon76-work | murdmath: context != extension |
| 19:52.00 | JerJer | right |
| 19:52.08 | jart | JerJer: i'm not entirely sure why. we're so understaffed here i don't have time to figure it out |
| 19:52.09 | [TK]D-Fender | murdmath: There is no such thing as an IAX extension. |
| 19:52.15 | jart | maybe it's my fault |
| 19:52.35 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:52.40 | JerJer | jart: firewall / proxy / very old network edge device ? |
| 19:52.48 | Mercestes | murdmath, your grossly abusing terms but, that should be fine to have an exten => 1234 and a context [1234] elsewhere. |
| 19:52.55 | jart | sip proxy |
| 19:53.06 | Mercestes | murdmath, very bad practice because it's confusing and ambiguous, but syntatically, there is nothing wrong with it. |
| 19:53.10 | murdmath | Mercestes: Sorry for the grossness :) |
| 19:53.30 | [TK]D-Fender | murdmath: An extension is something yuo can DIAL in your DIALPLAN (extensions.conf). a DEVICE is something you set up in your channel driver fille (eg sip.conf, iax.conf) and can use the "Dial" application to make RING. |
| 19:53.31 | JerJer | murdmath: your dialplan should be laid out using a simple hierarchy |
| 19:53.46 | Mercestes | murdmath, It's ok, I have low standards |
| 19:54.29 | murdmath | Mercestes: Thanks. I'm just trying to debug an Asterisk based embedded system and I want to make sure I'm not reporting a false bug. |
| 19:55.01 | *** join/#asterisk saftsack (n=saftsack@pD9E07966.dip.t-dialin.net) |
| 19:55.18 | murdmath | Mercestes: It is my understanding that a context is something in brackets such as [incoming] |
| 19:55.38 | Mercestes | murdmath, correct. |
| 19:55.55 | murdmath | Mercestes: If such a thing exists in iax.conf is it still a context? |
| 19:56.02 | Mercestes | murdmath: It is not. |
| 19:56.04 | *** join/#asterisk saftsack (n=saftsack@pD9E07966.dip.t-dialin.net) |
| 19:56.08 | murdmath | Mercestes: ok. |
| 19:56.10 | Mercestes | then it becomes a peer name |
| 19:56.23 | polerin | or a friend name or.. |
| 19:56.39 | Mercestes | however, if there is no correlation between extension 1234, and peer 1234, then it is a bad practice for human legibility, but it is syntactically correct. |
| 19:56.46 | Mercestes | polerin: You have no friends. |
| 19:56.49 | Mercestes | ... |
| 19:56.52 | Mercestes | I'm sorry |
| 19:57.03 | polerin | Mercestes: duh. I meen come on, who on IRC actually has friends. Freak. |
| 19:57.03 | murdmath | Mercestes: And a peer/friend name can be the same as a context found in extensions.conf |
| 19:57.05 | polerin | ;) |
| 19:57.07 | Corydon76-work | Mercestes: yes, she does |
| 19:57.14 | Mercestes | I know that... |
| 19:57.16 | Mercestes | I'm her friend. :) |
| 19:57.21 | Mercestes | sorta... |
| 19:57.26 | Mercestes | it's more one sided... |
| 19:57.39 | Mercestes | me: "hi friend." Polerin: *stabs in the eye with a spork* |
| 19:58.05 | Corydon76-work | She's nice, but she can be a little moody |
| 19:58.17 | Corydon76-work | even IRL |
| 19:58.23 | Mercestes | that's all women |
| 19:58.33 | Mercestes | and some effeminate men. |
| 19:59.19 | polerin | ... |
| 19:59.29 | Corydon76-work | ANYway |
| 19:59.43 | polerin | heh |
| 20:00.41 | Mercestes | pirates? |
| 20:00.48 | polerin | I don't think I'm moody so much as tempremental |
| 20:00.59 | Mercestes | and a good speller |
| 20:01.05 | Corydon76-work | It's the hormones |
| 20:01.07 | polerin | stfu :P |
| 20:01.18 | polerin | that and the stress I guess :P |
| 20:01.21 | Corydon76-work | Menopause came early |
| 20:01.27 | polerin | :/ |
| 20:01.38 | Mercestes | ... |
| 20:01.45 | Mercestes | sed/opause//g |
| 20:01.49 | Mercestes | >.. |
| 20:01.52 | Mercestes | >.> even |
| 20:02.14 | polerin | Mercestes: I'm not even commenting on that in this channel |
| 20:02.14 | polerin | lol |
| 20:02.19 | Mercestes | lmao |
| 20:03.05 | J4k3 | I read that as "Mercestes came early" which was even funnier. |
| 20:03.16 | Mercestes | J4k3, rofl. >.> uhh..who told? |
| 20:03.18 | *** part/#asterisk daguz (n=leo@208-1-63-34.celito.net) |
| 20:04.18 | Mercestes | It's the hormones. |
| 20:04.22 | Mercestes | that and the stress, I guess. |
| 20:05.36 | polerin | anyway. murdmath as a newb to * as well, It took me a bit to get used to [name] as a specification for everything. The best way I can think about it right now is that the use of the file specifies the meening of the [name] |
| 20:06.18 | murdmath | polerin: Thanks. |
| 20:06.30 | Mercestes | well.... |
| 20:06.42 | *** join/#asterisk joetester2 (n=joeteste@216.191.34.13) |
| 20:06.42 | Mercestes | yea |
| 20:07.08 | Mercestes | [name] in extensions.conf is the name of a group of extensions. [name] in iax.conf is the name of an iax2 device, [name] in sip.conf is the name of a sip device. |
| 20:07.15 | Mercestes | [name] in zaptel.conf is a syntax error. |
| 20:08.23 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 20:08.29 | shido6 | unless you are talking about [general] in iax.conf or sip.conf this applies as a default if not mentioned in the device stanza for the most part. |
| 20:10.34 | murdmath | Mercestes: Perfect. |
| 20:12.04 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 20:12.49 | Katty | allo (= |
| 20:12.53 | murdmath | So here is another thing. |
| 20:12.59 | Katty | i'm having a bit of an unusual problem. at least to me. |
| 20:13.12 | Katty | when i make an internal call, everything is just peachy. |
| 20:13.18 | Katty | i call ext whatever, it works. |
| 20:13.30 | Katty | if i make an outgoing call, with a _Nxxxxx thingy |
| 20:13.36 | Katty | i just get a fast busy, and nothing at the CLI |
| 20:13.39 | Katty | but those lines do work. |
| 20:13.47 | Katty | I get incoming calls on them all the time |
| 20:13.49 | shido6 | sip phone? |
| 20:13.52 | Katty | yes'r |
| 20:14.07 | Katty | this nothing at the CLI thing has me confuzzled. |
| 20:14.13 | shido6 | sip debug and watch for "looking for (whateveryoudialedhere) in context (somecontextthatphonehasaccesstoin sip.conf)" |
| 20:14.17 | murdmath | If I was to have one asterisk box be a provider for another and I use iax should the iax device be named the same as the username? |
| 20:14.25 | Katty | ok |
| 20:14.27 | shido6 | or a 484 address incomplete |
| 20:14.29 | shido6 | or similar |
| 20:14.34 | shido6 | asterisk will tell on itself |
| 20:14.39 | shido6 | if you let it |
| 20:15.44 | Mercestes | Katty!!! |
| 20:15.49 | Katty | Mercestes!!! |
| 20:15.55 | Mercestes | Is your dialplan correct in your phone? |
| 20:16.20 | shido6 | Looking for 3344439 in downstairs (domain 192.168.0.8) ? |
| 20:16.36 | shido6 | so what does [downstairs] look like? |
| 20:16.46 | J4k3 | beige, its mostly beige |
| 20:16.48 | Katty | i'll pastebin it |
| 20:16.56 | shido6 | :) |
| 20:18.37 | shido6 | wow |
| 20:19.04 | shido6 | ok between 37 and 43 |
| 20:19.09 | shido6 | is that in the upstairs context ? |
| 20:19.49 | shido6 | ok the first sip debug |
| 20:19.54 | Katty | nope, that's in the downstairs context |
| 20:20.03 | Katty | thats the included file |
| 20:20.06 | shido6 | you just added it in the paste :) |
| 20:20.07 | shido6 | ok |
| 20:20.15 | Katty | the actual catch all thingy |
| 20:20.26 | Katty | just dumped it at the bottom |
| 20:20.36 | shido6 | in "/etc/asterisk/downstairsdialout" |
| 20:20.39 | shido6 | <PROTECTED> |
| 20:20.39 | Katty | it's really not there :P |
| 20:20.39 | Katty | yes |
| 20:20.52 | shido6 | ok here's the thing |
| 20:21.00 | shido6 | do you have any named contexts |
| 20:21.02 | shido6 | in the other files |
| 20:21.03 | shido6 | <PROTECTED> |
| 20:21.11 | shido6 | "/etc/asterisk/downstairsautoattendant" |
| 20:21.17 | shido6 | or "/etc/asterisk/speeddialdownstairs" |
| 20:21.22 | shido6 | or "/etc/asterisk/sipdownstairs" |
| 20:21.24 | [TK]D-Fender | Katty: Mew. |
| 20:21.34 | Katty | yes |
| 20:21.36 | Katty | and those all work |
| 20:21.52 | Katty | i can dial 05 and get my cellphone, or 112, and get my coworker |
| 20:21.52 | shido6 | because if you have a context in those other files |
| 20:22.03 | shido6 | this downstairs dial out section u have in the pastebin may be in another context |
| 20:22.22 | Katty | no, the other files are just dumped in there for organizational stuff |
| 20:22.28 | shido6 | okie dokie |
| 20:22.29 | Katty | there's no further contexts in there |
| 20:22.58 | shido6 | add a _ |
| 20:23.01 | shido6 | in front of 3344439 |
| 20:23.03 | Katty | ok |
| 20:23.06 | shido6 | and do an extensions reload |
| 20:23.08 | shido6 | and try again |
| 20:23.09 | jart | http://merkwork.com/images/LOL.jpg <-- wait isn't that wrong? isn't that e^(i*pi) instead of e^(2*pi) which is euler's formula returning -1 so it ends up being 0.002 + -1 + 1 which is just 0.002 |
| 20:23.57 | shido6 | and your 911 one add a few more lines in case you have a heart attack and have issues dialing single digits and then a double for example, 911 is a great start but add 9911 9991111 9111 |
| 20:24.14 | shido6 | :) |
| 20:24.31 | shido6 | have you tried the Tim Hortons Cheese Danish |
| 20:24.41 | Katty | hmmm... i didn't see it listed... |
| 20:24.49 | shido6 | the cheese danish or 911 ? |
| 20:24.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:25.10 | Katty | nono, when i did extensions reload |
| 20:25.17 | Katty | it gives you that big long list of what's reloaded |
| 20:25.21 | shido6 | yeah |
| 20:25.27 | shido6 | and u didnt catch it eh.. |
| 20:25.29 | Katty | it wasn't shown in there |
| 20:25.31 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:25.31 | Katty | yeah |
| 20:26.13 | Mercestes | jart: 1.10121 to be exact. |
| 20:26.38 | jart | Mercestes: how do you figure? |
| 20:26.41 | joetester2 | jart: at the bottom is it's written e^j2pi but on the check it's jpi |
| 20:26.42 | Mercestes | e^(2*pi) I mean. |
| 20:26.54 | Katty | AH HA! |
| 20:26.59 | Katty | shido6: it was a context problem |
| 20:27.06 | jart | i think they read the check wrong |
| 20:27.07 | Katty | shido6: the auto attendant DID have an additional context |
| 20:27.13 | Mercestes | pi = 3.1415926535897932384626433832795 |
| 20:27.13 | Katty | shido6: so i moved them around and did a reload. |
| 20:27.20 | x86 | Qwell[]: was it you that told me the Polycom IP301 sucks? |
| 20:27.26 | shido6 | cool |
| 20:27.27 | x86 | Qwell[]: like, compared to a 501 or 601? |
| 20:27.28 | Mercestes | 2*3.1415926535897932384626433832795 = 6.283185307179586476925286766559 |
| 20:27.31 | Katty | shido6: my poor wittle server thought it was under my auto attendant >.< |
| 20:27.34 | Qwell[] | x86: no |
| 20:27.36 | joetester2 | It's e^jpi not e^pi dude |
| 20:27.43 | diskfree | @Corydon76-work: is your customer using h323 or sip with their cisco routers? |
| 20:27.46 | joetester2 | It's euler not just e^something |
| 20:27.46 | x86 | Qwell[]: ah... was just gonna say you were totally right ;) |
| 20:27.55 | jart | http://en.wikipedia.org/wiki/Euler's_formula |
| 20:28.13 | Katty | shido6: so how do i solve that problem for good? is there a Prefered Way, or should i stick it at the bottom |
| 20:28.14 | Mercestes | 0.002^6.283185307179586476925286766559 = 1.1012129862274081175242718331433e-17 |
| 20:28.44 | shido6 | if it works im not going to touch it - figure out the better way when you have to :) |
| 20:28.58 | joetester2 | jart : I don't think he gets it :S |
| 20:29.05 | Mercestes | It is not jpi |
| 20:29.12 | *** part/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 20:29.12 | shido6 | unless you are bored. |
| 20:29.18 | joetester2 | i = j = sqrt(-1) |
| 20:29.23 | shido6 | and if you are just write a blog with adsense |
| 20:29.26 | Katty | shido6: kk, i just stuck it as the last item under the [downstairs] |
| 20:29.27 | Mercestes | It says, 002 + e^(2pie) = 535 |
| 20:29.28 | shido6 | make some money :) |
| 20:29.31 | nohop | hey people :) |
| 20:29.38 | Katty | shido6: i'll do the same with my other one too |
| 20:29.45 | shido6 | oh! |
| 20:29.49 | shido6 | but please make the 911 changes |
| 20:29.55 | Mercestes | There is no J |
| 20:30.01 | nohop | is there a way to make asterisk dial 2 numbers, and connect both together from the remote command line ? |
| 20:30.04 | nohop | from the console, that is |
| 20:30.04 | joetester2 | Dude! It's e^i*pi! |
| 20:30.05 | jart | are you sure it's 2 pi? it looks like i*pi |
| 20:30.27 | joetester2 | Oh he's right, my eyesight sucks |
| 20:30.33 | jart | i remember reading that guys story about the debate between parts of a cent versus parts of a dollar |
| 20:30.37 | jart | it was sooo funny |
| 20:30.40 | Mercestes | I am forgettng e tho |
| 20:30.59 | Katty | shido6: you mean put a _ in front of it? |
| 20:31.01 | joetester2 | That's supposedly a two |
| 20:31.10 | Mercestes | His handwriting is a little crappy too |
| 20:31.10 | shido6 | no.. |
| 20:31.16 | Katty | shido6: i wasn't paying attention. |
| 20:31.19 | shido6 | 911, 99111 9991111 91111 91111 |
| 20:31.27 | Katty | oh? why? |
| 20:31.30 | shido6 | in case you cant focus for 911 |
| 20:31.32 | Mercestes | shido6, That's a little silly |
| 20:31.32 | Katty | are those all 911 numbers? |
| 20:31.39 | Katty | no |
| 20:31.41 | Katty | it's a good idea |
| 20:31.47 | Katty | if i was panicing, i'd be doing stuff like that too |
| 20:31.48 | shido6 | they can dial the same 911 |
| 20:31.57 | Mercestes | Might as well throw some 8's in there too then. |
| 20:31.57 | joetester2 | Sorry Mercestes, my eyesight sucks apparently |
| 20:32.02 | shido6 | but yuo can enter them differently |
| 20:32.04 | Mercestes | and some *'s. |
| 20:32.06 | *** join/#asterisk huey23 (n=huey23@66.17.218.10) |
| 20:32.17 | Mercestes | if you can't dial "911" then you can't expect them to hit the right keys either.. |
| 20:32.36 | Mercestes | Wouldn't it be a *better* idea to make a bright, red, button labeled "help me" that auto dials 911? |
| 20:32.52 | huey23 | "help me" is a little vague |
| 20:32.59 | Mercestes | huey23, how is "help me" vague? |
| 20:33.19 | huey23 | help me can refer to a lot of things...not just emergencies |
| 20:33.47 | Mercestes | I guess you could create lots of buttons, "help, I'm having a heartattack" "help, I'm having a stroke" "help, I'm being stabbed" "help, I'm suffering from erectile dysfunction" but...Unless she has a polycom 601 she's likely to run out of keys. |
| 20:33.53 | huey23 | :) |
| 20:33.56 | huey23 | there ya go |
| 20:34.19 | huey23 | you forgot "help me, my e-mail's broke |
| 20:34.24 | huey23 | " |
| 20:34.37 | Katty | help, i'm dumb. |
| 20:34.45 | Katty | that should cover all of it. |
| 20:34.45 | Mercestes | Have a beer. |
| 20:34.52 | Katty | ewww, beer |
| 20:34.53 | Katty | oh! |
| 20:34.55 | Katty | ohhh!! |
| 20:34.58 | Katty | i'm getting a puppy! |
| 20:34.58 | Mercestes | So, what's your name? |
| 20:34.58 | jart | take care everyone! time to go someplace far away from phones, walk up a mountain and drink lots of wine |
| 20:34.58 | huey23 | "help me, I need a smoke: see..to vague |
| 20:35.11 | Katty | a german shepherd! |
| 20:35.12 | Mercestes | Katty: Nice....Gratz! |
| 20:35.13 | Katty | tonight! |
| 20:35.16 | Mercestes | oh. |
| 20:35.17 | *** part/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 20:35.18 | Katty | in...uh 3 hours |
| 20:35.18 | Mercestes | no gratz. |
| 20:35.25 | Mercestes | German shepherd? ...why? |
| 20:35.29 | Katty | guard pup |
| 20:35.33 | Mercestes | .... |
| 20:35.39 | huey23 | good dogs |
| 20:35.42 | Katty | and their ears are cute |
| 20:35.45 | Katty | and very trainable |
| 20:35.46 | Mercestes | Wouldn't an attack horse be a little more economic?? |
| 20:35.52 | Katty | nah |
| 20:36.00 | Katty | if i wanted economic, i'd just train my two ferrets |
| 20:36.00 | shido6 | well, Mercestes, thats what lumenvox is for |
| 20:36.11 | Mercestes | ....attack ferrets.... |
| 20:36.13 | Mercestes | bwahahaha |
| 20:36.13 | shido6 | speak your emergency |
| 20:36.16 | Katty | they already attack the bookshelf |
| 20:36.23 | Mercestes | "Please state the nature of your medical emergency." |
| 20:36.30 | shido6 | I'm dying |
| 20:36.46 | shido6 | Im taking a crap and it hurts.... |
| 20:36.49 | shido6 | to think |
| 20:36.58 | Mercestes | I'm dying. Disambiguation page. Did you mean "i'm dying" the song, "I'm dying" the emergency, or "I'm dying" the euphamism? |
| 20:36.59 | huey23 | fart and clear your mind |
| 20:37.11 | tzanger | farting cetainly won't clear hte air |
| 20:37.19 | Mercestes | will clear your mind tho. |
| 20:37.29 | Mercestes | and the room |
| 20:37.32 | huey23 | absolutely |
| 20:39.21 | Katty | also, i hit 314 mining today. |
| 20:39.26 | Katty | and 313 herbalism. |
| 20:39.28 | Katty | if anyone cares. |
| 20:39.32 | Qwell[] | umm, yeah |
| 20:39.39 | Mercestes | ... |
| 20:39.42 | Mercestes | I hit...420 herbalism. |
| 20:39.45 | *** join/#asterisk Snible (n=Miranda@pD9E0A152.dip.t-dialin.net) |
| 20:39.48 | Katty | 375 is the max :< |
| 20:39.51 | Mercestes | and 14 jailbaite. |
| 20:40.07 | Katty | not enough people play wow |
| 20:40.17 | Mercestes | Oh...WoW |
| 20:40.19 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-cf83692581ac44fb) |
| 20:40.24 | Mercestes | World of Warcrack |
| 20:40.33 | bkruse | Katty: I just tried to tab complete your name and it crashed my pidgin :[ |
| 20:40.49 | Mercestes | A colt45 and two zig zags, baby that's all we need. |
| 20:40.54 | Katty | bkruse: my pidgin is crashy too |
| 20:40.57 | Katty | bkruse: alll day long |
| 20:40.58 | Mercestes | bkruse, That's because your not good enough for katty |
| 20:41.05 | Mercestes | katty > j00 |
| 20:41.07 | Katty | ^_- |
| 20:41.09 | Katty | hush up |
| 20:41.11 | bkruse | Mercestes: :[ |
| 20:41.14 | bkruse | yay |
| 20:41.37 | Katty | "can't we all just get along?!" -Fire Imp. |
| 20:43.05 | Katty | anywho, i need a name for the new pup |
| 20:43.07 | Katty | something german. |
| 20:43.18 | Katty | i've been thinking Fuhrer, jeiger, zeek, and uber. |
| 20:43.23 | Mercestes | Jager |
| 20:43.30 | Katty | however you spell it |
| 20:43.33 | Mercestes | Meister Jager |
| 20:43.35 | Katty | the stuff in the green bottle. |
| 20:43.40 | Mercestes | Yes.... |
| 20:43.44 | Katty | it's sweet. |
| 20:43.45 | Mercestes | ....omg..Jager... |
| 20:43.46 | Katty | and licoricy |
| 20:43.54 | Mercestes | threesomemy... |
| 20:43.54 | diskfree | Jagermeister |
| 20:43.56 | Katty | very yummy on the rocks. |
| 20:44.02 | Katty | diskfree: yes, precisely. |
| 20:44.08 | diskfree | :) |
| 20:44.11 | Katty | diskfree: but jager for short. |
| 20:44.17 | Mercestes | it tastes better than teen spirit. |
| 20:44.20 | Katty | diskfree: cause a pup needs something short and sweet to understand. |
| 20:44.20 | Mercestes | Yes, name your dog Jager... |
| 20:44.35 | Mercestes | Or Mr. Tinkles |
| 20:44.35 | diskfree | Katty: lol |
| 20:46.33 | tzanger | My last dog's name was casey |
| 20:46.36 | tzanger | perfect name for a dog |
| 20:46.40 | tzanger | cat was jake |
| 20:46.48 | tzanger | they need something short and sharp |
| 20:46.52 | tzanger | easy for them to recognize |
| 20:46.55 | Mercestes | Name him Evinrude |
| 20:47.00 | tzanger | you could name him "Gunnerstchoffel" |
| 20:47.04 | tzanger | oh man |
| 20:47.09 | tzanger | from the kid's movie The Rescuers |
| 20:47.13 | Mercestes | Nice |
| 20:47.23 | tzanger | the dragonfly that made their leaf boat run |
| 20:47.24 | Mercestes | Didn't think anyone would catch *that* reference. lol |
| 20:47.26 | tzanger | his name was evenrude |
| 20:47.29 | Mercestes | Yup |
| 20:47.32 | Mercestes | wow. |
| 20:47.35 | tzanger | I LOVED that |
| 20:47.38 | Mercestes | Showing your age *and* your movie preferences. :D |
| 20:47.38 | tzanger | giggled for most of the movie |
| 20:47.58 | Mercestes | hehe |
| 20:48.01 | Mercestes | It was awesome. :D |
| 20:48.02 | tzanger | Katty: Reinhold |
| 20:48.07 | tzanger | or Helmut |
| 20:48.32 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:48.57 | Mercestes | R E S C U E, rescue aid so-cie-a-te, heads held high, touch the sky, you mean everything to me..... |
| 20:49.21 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:49.39 | tzanger | Mercestes: yep |
| 20:49.45 | tzanger | that bianca mouse was hot |
| 20:49.51 | tzanger | mind you I had a thing for gadget too from rescue rangers |
| 20:49.59 | Mercestes | Gidget. |
| 20:50.06 | Mercestes | she was hot. |
| 20:50.17 | Qwell[] | she was a cartoon mouse |
| 20:50.22 | Mercestes | she was HOT. |
| 20:50.29 | Qwell[] | she was a CARTOON MOUSE |
| 20:50.39 | tzanger | http://www.pionier-clan.de/Bilder/RitterdesRechts/Trixi/gadget1.gif |
| 20:50.40 | tzanger | hmm |
| 20:50.42 | tzanger | I remember her cuter |
| 20:50.59 | Mercestes | oh, maybe it was gadget |
| 20:51.18 | _DAW | Wasnt sally fields gidget? |
| 20:51.27 | Mercestes | I dunno |
| 20:51.41 | huey23 | she is hot |
| 20:52.02 | tzanger | I dunno... a chick who's into mechanics and whose smart... that's hot |
| 20:52.12 | Mercestes | http://captainkalen.tripod.com/images5/penny_coo-cooclockcaper66.jpg |
| 20:52.14 | Mercestes | she was hot too |
| 20:52.19 | Mercestes | wait... |
| 20:52.21 | Mercestes | shouldn't have used that pic |
| 20:52.26 | tzanger | hahaha |
| 20:52.32 | Mercestes | http://www.geocities.com/hollywood/screen/7219/12.jpg |
| 20:52.33 | tzanger | I EFFING HATED INSPECTOR GADGET |
| 20:52.33 | Mercestes | there |
| 20:52.38 | Mercestes | I did too |
| 20:52.41 | huey23 | lol |
| 20:52.41 | Mercestes | but his daughter was hot |
| 20:52.47 | diskfree | Penny |
| 20:52.48 | tzanger | I didn't like penny, sorry |
| 20:52.55 | Mercestes | oh... |
| 20:53.01 | Mercestes | not furry enough? |
| 20:53.04 | [TK]D-Fender | Mercestes, Boy you're going to extreme measured to ensure virginity.... |
| 20:53.07 | diskfree | Gadget was her uncle, wasn't he? |
| 20:53.24 | Mercestes | uncle, father, whatever. |
| 20:53.39 | tzanger | uncle |
| 20:53.39 | diskfree | not that it matters |
| 20:53.41 | tzanger | he was a shmuck |
| 20:53.46 | Mercestes | agreed. |
| 20:53.47 | tzanger | damn I hated that hsow |
| 20:53.52 | diskfree | It was kinda a funny |
| 20:53.54 | tzanger | speaking of shows |
| 20:53.58 | tzanger | I gotta see that transformers movie |
| 20:53.59 | diskfree | in 1988 |
| 20:54.02 | _DAW | Always wanted to see mr. claw though :( |
| 20:54.10 | Mercestes | I saw him. |
| 20:54.11 | tzanger | looks totally different than the cartoons, but I think the boys'd like it |
| 20:54.15 | Mercestes | He's just a hand attached to a chair. |
| 20:54.17 | diskfree | next time gadget... |
| 20:54.33 | diskfree | maybe this where jart's voice application comes in handy |
| 20:54.34 | tzanger | now it's time to reminisce about voltron |
| 20:54.39 | diskfree | +is |
| 20:54.45 | Mercestes | Pidge was hot.... |
| 20:54.46 | Mercestes | damnit... |
| 20:54.49 | Mercestes | I meant Princess Aluria. |
| 20:54.56 | tzanger | ? |
| 20:54.56 | Mercestes | really. |
| 20:54.58 | Mercestes | >.> |
| 20:55.08 | diskfree | Aluria? |
| 20:55.12 | Mercestes | Yea |
| 20:55.17 | Mercestes | Alaria...Aluria... |
| 20:55.20 | Mercestes | the pink chick |
| 20:55.20 | iCEBrkr | ASTERISK SUCKS!!! |
| 20:55.23 | iCEBrkr | Just kidding! |
| 20:55.23 | huey23 | malaria |
| 20:55.24 | tzanger | iCEBrkr: agreed |
| 20:55.27 | iCEBrkr | tzanger: haha |
| 20:55.40 | iCEBrkr | It's been kind lately.. seemless upgrades. |
| 20:55.43 | iCEBrkr | weee! |
| 20:56.09 | Mercestes | Princess Allura/Princess Farla (ファーラ姫, FÄra-hime?): Princess Allura of the planet Arus is the ruler of the Kingdom of Altair |
| 20:57.13 | diskfree | http://cbjam.tripod.com/voltron/voltcharacters.html ? |
| 20:57.41 | Qwell[] | /part #cartoons |
| 20:58.13 | Mercestes | Qwell[] Don't be a hater. |
| 20:58.23 | iCEBrkr | haha |
| 20:58.37 | jarrod | is heartbeat supposed to configure the alias ip address when started? |
| 20:59.40 | bkruse | jarrod: yay failover |
| 21:00.11 | Mercestes | ... |
| 21:00.22 | Mercestes | if I call my wife by some weird cartoon name she recognizes...I will hate you all. |
| 21:00.45 | iCEBrkr | Is that some different level of furry? |
| 21:00.48 | iCEBrkr | o.O |
| 21:01.02 | Mercestes | I've got Penny stuck in my head now. |
| 21:01.06 | bkruse | call her la blue girl |
| 21:01.19 | x86 | <PROTECTED> |
| 21:01.25 | Mercestes | I'll call her D-fender... |
| 21:01.32 | Mercestes | she'll think I'm referring to the guitar and it will be ok. |
| 21:01.33 | bkruse | x86: they didnt specify a context? |
| 21:01.43 | x86 | why does it say '7697@' when i tell it the context is 'admin' in the peer/user definition in iax.conf?? |
| 21:01.46 | bkruse | they would have a context if they had an entry in iax.conf |
| 21:02.00 | x86 | they do |
| 21:02.01 | bkruse | is he dynamic/registered or static ip? |
| 21:02.09 | x86 | tried it both ways |
| 21:02.10 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 21:02.12 | bkruse | ;[ |
| 21:02.17 | bkruse | what does your dial statement look like |
| 21:02.31 | bkruse | dial(iax2/hostnameorpeer/number)? |
| 21:02.43 | x86 | Dial(IAX2/user:pass@server/${EXTEN}|1000|tTR) |
| 21:03.12 | bkruse | hmm |
| 21:03.22 | x86 | i tried also doing just the peer name specified in iax.conf |
| 21:03.24 | bkruse | do you have a friend/peer/user defined on the machine calling? |
| 21:03.28 | x86 | yes |
| 21:03.29 | bkruse | dang |
| 21:03.32 | x86 | friends on both sides |
| 21:03.35 | bkruse | thats wierd :/ |
| 21:03.39 | x86 | yeah |
| 21:03.41 | bkruse | right, then just dialing like iax2/peername/exten |
| 21:03.42 | bkruse | hmm |
| 21:03.50 | x86 | yeah no workie |
| 21:04.42 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:04.55 | generalhan | ok, i have a new issue ! im trying to get this phone to register and it wont, so i turned on sip debugging and got some information, but until it says "detroying" i dont see any issues with the information. but i would never claim to know exactly what im looking at either ! can some one take a look please? http://generalhan.pastebin.ca/606433 |
| 21:05.41 | x86 | bkruse: well site A is running 1.0.10 |
| 21:06.09 | x86 | bkruse: site B is running 1.2.12.1 |
| 21:06.14 | generalhan | umm, well without any further information from the sip debug, now its registered ??? wow am i ever confused |
| 21:06.28 | tzanger | hahahahhahhaha |
| 21:06.30 | tzanger | 17:06 < sgi> wtf: my math teacher staples burger king applications to failed tests |
| 21:06.33 | Siya | anyone ever tried to use justvoip.com with asterisk? |
| 21:06.40 | x86 | tzanger: roflmao |
| 21:06.58 | [TK]D-Fender | generalhan, "SIP/2.0 401 Unauthorized" <- says it all |
| 21:07.03 | diskfree | tzanger: lol |
| 21:07.05 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 21:07.21 | generalhan | [TK]D-Fender: what does that mean though ? cause now its actually working |
| 21:07.23 | bkruse | x86: woah |
| 21:07.28 | bkruse | update please! |
| 21:07.36 | bkruse | at least site A |
| 21:07.47 | [TK]D-Fender | generalhan, bad user/pass |
| 21:07.59 | x86 | bkruse: yeah, working on it :P |
| 21:08.01 | generalhan | [TK]D-Fender: well how would it just fix itself all the sudden ? |
| 21:08.14 | sakic | who is a good voip provider for asterisk, quality and price wisE? |
| 21:08.19 | [TK]D-Fender | generalhan, define "fix". |
| 21:08.36 | x86 | bkruse: but site A (1.0.10) can call site C (1.2.17) just fine! |
| 21:08.50 | x86 | bkruse: so the problem has to be some kind of config on site B, right? |
| 21:09.09 | x86 | not protocol / version mismatch |
| 21:09.28 | generalhan | [TK]D-Fender: i cant define fix. but in the amount of time it took for me to post that pastebin, it magically registered |
| 21:09.38 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 21:09.40 | generalhan | now the phone is working perfectly |
| 21:09.57 | generalhan | but i didnt touch anything between the time that i posted in here, and the time that it started working |
| 21:10.21 | *** join/#asterisk saftsack (n=saftsack@pD9E07966.dip.t-dialin.net) |
| 21:11.43 | bkruse | x86: it COULD be, there have been so many changes to chan_iax you know? |
| 21:12.02 | bkruse | 1.2 chan_iax was isn't even threaded right? |
| 21:12.08 | j0 | I keep getting this error on IAX trunks (on SIP it works fine) right around a WaitExten: [2:09.28p] * Topic is 'Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.20, 1.4.6 (June 29, 2007 |
| 21:12.26 | bkruse | clipboard > j0 |
| 21:12.37 | j0 | bkruse? |
| 21:12.42 | j0 | oops.. haha |
| 21:12.49 | j0 | NOTICE[14042]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. |
| 21:12.50 | bkruse | I will help if I get the full error message :D |
| 21:12.58 | bkruse | oh, ive seen that. What version of asterisk? |
| 21:13.02 | bkruse | 1.2? |
| 21:13.07 | j0 | 1.2.18 |
| 21:13.11 | bkruse | oh. |
| 21:13.14 | bkruse | are you talking to a 1.4 box? |
| 21:13.31 | j0 | it used to be a 1.2... maybe they've changed! |
| 21:13.34 | bkruse | I have seen that when going from 1.2 -> 1.4 |
| 21:13.51 | bkruse | let me look up exactly where thats spitting that out... h/o |
| 21:14.40 | j0 | its not being caused by WaitExten.. I can't quite pinpoint where it goes wrong. I've tried all sorts of things.. like not using Answer at the beginning, or no background sounds, longer wait times |
| 21:14.56 | bkruse | hmm |
| 21:14.56 | bkruse | right |
| 21:15.08 | tzafrir_home | x86: iax or sip between hosts? |
| 21:15.32 | bkruse | hmm, thats wierd.... |
| 21:15.42 | bkruse | tzafrir_home: iax? |
| 21:16.14 | bkruse | .read = iax2_read, |
| 21:16.30 | bkruse | iax2_read just has that notice, then returns NULL |
| 21:16.38 | bkruse | From what I can tell, it does not actually hangup |
| 21:16.48 | j0 | is there an easy way to check the remote version? |
| 21:16.53 | tzafrir_home | I recall some problems connecting 1.0 and 1.2 through IAX, that were resolved when the 1.2 host has disabled jitter-buffer support |
| 21:17.09 | bkruse | tzafrir_home: right, thats what I thought it was, just to far off versions |
| 21:17.15 | bkruse | im thinking the remote is older, or 1.4 |
| 21:17.20 | x86 | tzafrir_home: iax2 |
| 21:17.33 | bkruse | he was assuming iax2 i believe |
| 21:17.37 | bkruse | x86: does it ACTUALLY hangup the call? |
| 21:17.48 | x86 | it rejects it with unknown context |
| 21:18.11 | bkruse | Try to update to 1.2.20, though i think you will find better results with your endpoint updating. |
| 21:18.14 | x86 | works from site A (1.0.10) to site C (1.2.17), but not from site A (1.0.10) to site B (1.2.12.1) |
| 21:18.47 | bkruse | im still sure its because 1.0.10 |
| 21:18.49 | bkruse | thats old |
| 21:18.55 | x86 | dude |
| 21:19.00 | x86 | it works to 1.2.17 just fine |
| 21:19.15 | x86 | how do you logically explain that it doesn't work to 1.2.12 ? |
| 21:19.23 | bkruse | i cannot logically explain it ;] |
| 21:19.27 | bkruse | well, do the obvious |
| 21:19.31 | bkruse | what boxes do you have control over? |
| 21:19.41 | diskfree | x86: have you tested with C and B? |
| 21:19.51 | x86 | it might make sense if it worked between 1.0.10 and 1.2.12, but not from 1.0.10 to 1.2.17, but that's not the case |
| 21:20.07 | x86 | diskfree: B and C can talk all day long |
| 21:20.13 | x86 | diskfree: D too ;) |
| 21:20.26 | russellb | 1.0 w00t |
| 21:20.28 | russellb | that is all |
| 21:20.40 | diskfree | x86: :) ok |
| 21:20.59 | x86 | i'll update site A from 1.0.10 to 1.2.20 this weekend and see what I break |
| 21:21.01 | x86 | >:) |
| 21:21.10 | diskfree | x86: so along the way in 1.2.12.1 -> 1.2.17 something got fixed |
| 21:21.38 | russellb | just be sure to read UPGRADE.txt in 1.2 when upgrading from 1.0 to 1.2 |
| 21:21.40 | diskfree | x86: probably more then 1 thing :) |
| 21:22.08 | *** join/#asterisk apardo (n=deal@33.145.217.87.dynamic.jazztel.es) |
| 21:22.27 | x86 | russellb: yeah i've upgraded from 1.0 to 1.2 before... that's why i was looking for a quick fix instead of jumping at that solution already ;) |
| 21:23.32 | russellb | cool |
| 21:26.02 | j0 | is it possible to check the * version remotely? |
| 21:26.21 | Mercestes | j0: ssh |
| 21:26.27 | j0 | darn |
| 21:26.44 | rudholm | anyone here using the TDM800 for FXO? |
| 21:28.37 | j0 | ok.. so my provider is using 1.2.16 and i'm on 1.2.18 |
| 21:28.38 | *** join/#asterisk osiris250 (i=brdz4ioy@bsd02.evansengineering.net) |
| 21:29.05 | Mercestes | j0: ok... |
| 21:31.32 | bkruse | that would be a cool thing, to check it remotely, but could also turn into a subnet scanning exploit bot :/ |
| 21:31.42 | j0 | bkruse: that's what i figured |
| 21:31.55 | bkruse | but would be useful for situations like this non the les |
| 21:32.00 | bkruse | s/les/less/g |
| 21:42.23 | Jingles | do you have to specify if a sip extension can transfer a call or not? |
| 21:43.05 | Mercestes | Jingles, Tt |
| 21:43.14 | Jingles | that's in the dial string... |
| 21:43.28 | Jingles | but what if it's the sip extension itself that's trying to dial out? |
| 21:43.33 | Jingles | I'm probably not being clear. |
| 21:43.56 | Jingles | I have an IVR - and I need it to dial a number, and if the user presses 3, to transfer the call to some other number. |
| 21:44.35 | Jingles | the asterisk box itself takes 'flash hook' just fine via the TDM cards. |
| 21:44.48 | Jingles | however, the IAXys won't transfer a call, and neither will this ATA. |
| 21:45.34 | shido6 | iaxys will xfer a call |
| 21:45.44 | shido6 | if you enable T/t and features.conf |
| 21:46.00 | Jingles | right. I can do it, via an IAXy - if I do it by 'hand' (analog phone, dialing #s, etc) |
| 21:46.10 | Jingles | but we're using a product called 'PhoneHerald' as an IVR |
| 21:46.16 | Jingles | and it *won't* with an IAXy. |
| 21:46.23 | Jingles | something about timing or whatever. |
| 21:46.28 | shido6 | timing? |
| 21:46.32 | shido6 | odd. |
| 21:46.44 | shido6 | do you want it to? |
| 21:46.46 | Jingles | yeah - the stupid software just hangs up the phone on the IAXy. |
| 21:46.53 | shido6 | opensource? |
| 21:46.57 | *** join/#asterisk mitcheloc (n=mitchelo@rrcs-64-183-110-250.west.biz.rr.com) |
| 21:47.13 | Jingles | it sends the flash over the line, then click |
| 21:47.33 | shido6 | can it send a "#" ? |
| 21:47.42 | Jingles | personally, I think this is a dumb idea, all the way around. |
| 21:47.48 | Jingles | but what do I know - just the IT 'grunt'. |
| 21:48.04 | Jingles | hmm. |
| 21:50.30 | Jingles | testing to see if the problem is the ATA. I'm betting it is. |
| 21:51.29 | *** join/#asterisk friedrich| (n=friedric@e177243203.adsl.alicedsl.de) |
| 21:53.10 | Innatech | anyone have strong feelings on dlink vs netgear web-managed switches? ( http://www.dlink.com/products/?pid=541 vs. http://www.netgear.com/Products/Switches/SmartSwitches/FS752TPS.aspx?detail=Specifications ) The dlink is a little cheaper. Am I missing some worthwhile feature difference? |
| 21:53.35 | bkruse | managed switches in general, bleh |
| 21:53.57 | bkruse | i had a netgear managed switch, or 2, at my school, they were expensive and not worth it |
| 21:54.04 | bkruse | i would have to go with dlink just because my bad experience |
| 21:54.08 | Innatech | Well, I have to buy one or another. Heh. |
| 21:54.14 | mitcheloc | pfft, netgear > dlink |
| 21:54.22 | bkruse | and because that page is written in aspx. |
| 21:54.41 | bkruse | mitcheloc: lot of networking experience? |
| 21:54.43 | mitcheloc | bkruse, that was uncaled for =P |
| 21:54.47 | bkruse | :] |
| 21:54.51 | mitcheloc | a mild amount |
| 21:54.59 | bkruse | we should ask jscott! |
| 21:56.01 | bkruse | s/we/I/g |
| 21:56.12 | bkruse | :D |
| 21:56.29 | mitcheloc | wierdo |
| 21:56.41 | bkruse | dude jbot_ owns. |
| 21:58.46 | j0 | any gotchas for upgrading from 1.2.18 to 1.2.20? |
| 21:59.07 | bkruse | CHANGES |
| 21:59.26 | j0 | yes.. changes that break things |
| 22:00.04 | bkruse | somtimes,. yes |
| 22:00.19 | j0 | wtf.. fixed some of my problems |
| 22:00.28 | j0 | still get the strange "you should never see this", but it continues on after that |
| 22:01.23 | Jingles | ok. I'm going to try the IAXy again, I guess. |
| 22:01.58 | Jingles | what do I have to put in iax.conf for call transfers? |
| 22:02.44 | k31th | is it best to source install asterisk ? |
| 22:02.59 | k31th | and wat distro is prefered |
| 22:04.45 | bkruse | i love debian |
| 22:04.49 | bkruse | apt-get build-dep asterisk |
| 22:04.52 | bkruse | then build your source |
| 22:08.12 | generalhan | ok all i am sooo close to getting this remote box all setup. i can make phone calls from the remote machine by connecting it to the host machine, that part works just fine. but i cant get calls to go the other direction. To, lets say, dial an extension on the remote machine. i have pasted the configs for both machines, if someone could take a look. http://generalhan.pastebin.ca/606522 |
| 22:08.42 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) |
| 22:08.49 | k31th | bkruse: iirc with debian i could build that src package as a dpkg ? |
| 22:09.09 | k31th | so i coud remove it etc if needed? |
| 22:09.12 | bkruse | k31th: yes, but its harder than it looks :P |
| 22:09.15 | bkruse | k31th: yes |
| 22:09.22 | bkruse | i used to do some packaing with tzafrir |
| 22:09.28 | k31th | bkruse: how do you go about doing it ? |
| 22:09.34 | bkruse | packaging something like asterisk can get harry |
| 22:09.39 | k31th | ok. |
| 22:09.46 | bkruse | k31th: google, its a lot. |
| 22:09.48 | bkruse | build dpkg |
| 22:09.51 | k31th | ok fine well this box is only going to do asterisk |
| 22:10.06 | [TK]D-Fender | generalhan, bad : exten => 7654,Dial(IAX2/Asterisk:Passw0rd@192.168.0.64/${EXTEN}) |
| 22:10.13 | k31th | so i dont need a package just a way of updating asterisk if and when i need to. |
| 22:10.27 | [TK]D-Fender | generalhan, good : exten => 7654,Dial(IAX2/Asterisk/${EXTEN}) |
| 22:12.10 | generalhan | [TK]D-Fender: sweet thanks ! why is that anyway? when i dial form the remote box to this one i have to put in the password to make it work |
| 22:12.13 | [TK]D-Fender | generalhan, and it'd be nice to see the call ATTEMPT + debug |
| 22:12.29 | [TK]D-Fender | generalhan, What the hell do you think you are making a PERR enry for! |
| 22:12.32 | [TK]D-Fender | PEER* |
| 22:12.40 | [TK]D-Fender | generalhan, So you DON'T put that in the DIALPLAN |
| 22:12.56 | generalhan | hmm |
| 22:13.09 | [TK]D-Fender | generalhan, "user THIS peer", "dial THIS number" |
| 22:13.16 | [TK]D-Fender | generalhan, You have missed the point again! |
| 22:13.18 | generalhan | im going to take that stuff off of the other box like i did before, and see if it works now. cause before it would work without the pass |
| 22:15.12 | k31th | is it easy to upgrade to the next version of asterisk from source ? |
| 22:15.19 | k31th | when updates occour |
| 22:22.15 | shido6 | yes |
| 22:22.27 | shido6 | unless you are going from 1.2 to 1.4 :) |
| 22:22.35 | shido6 | then the logic changes a tad. |
| 22:22.41 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:22.52 | shido6 | there are only 4k changes from 1.2 to 1.4 so dont freak out too much |
| 22:25.26 | russellb | only 4k? |
| 22:25.31 | JerJer | ok - i have a guy that is hell bent on not reloading asterisk... Is real-time able to pull the mailbox information dynamically ? |
| 22:25.54 | russellb | JerJer: in 1.4, yeah, you can put voicemail in realtime (i think) |
| 22:26.08 | russellb | yes, you can |
| 22:26.19 | JerJer | no the mailbox=100@foo on a sip peer |
| 22:26.34 | russellb | oh, then yeah ... |
| 22:26.45 | russellb | but only when you put all of the sip peer stuff in there |
| 22:26.55 | russellb | you can't only do one options |
| 22:27.35 | JerJer | one options? |
| 22:27.40 | russellb | one option |
| 22:27.50 | russellb | as in, mailbox= for sip peers, it has to be all information for all peers in realtime |
| 22:28.06 | JerJer | yeah - all the relevant stuff - host, secret, name, etc |
| 22:28.07 | JerJer | yeah |
| 22:28.11 | russellb | yep |
| 22:30.23 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:31.29 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 22:32.06 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 22:32.55 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 22:34.17 | Qwell[] | JerJer implementing realtime? |
| 22:34.21 | Qwell[] | I never thought I'd see the day. |
| 22:34.25 | Qwell[] | ~realtime |
| 22:34.25 | jbot | from memory, realtime is a feature of Asterisk starting with 1.2 which allows you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
| 22:34.48 | JerJer | "i have a guy that is hell bent on not reloading asterisk." |
| 22:35.00 | Qwell[] | I figured you'd fire him first :P |
| 22:35.22 | k31th | lol |
| 22:35.23 | JerJer | not my system |
| 22:35.34 | russellb | you can fire customers |
| 22:35.34 | Qwell[] | JerJer: I'm just messing with you :) |
| 22:35.37 | JerJer | not my decision |
| 22:35.39 | russellb | :) |
| 22:35.44 | JerJer | russellb: yes, yes i can |
| 22:35.47 | k31th | quit ? |
| 22:35.52 | Qwell[] | k31th: It's his company. |
| 22:35.56 | Qwell[] | that would be less than ideal |
| 22:36.04 | k31th | ahh well i know how that feels... |
| 22:36.26 | k31th | JerJer: where you at? |
| 22:36.35 | JerJer | Qwell[]: not just 'mine' any more.... i now have a board and shareholders to keep happy |
| 22:36.43 | Qwell[] | ahh |
| 22:36.45 | JerJer | k31th: in the bat cave |
| 22:36.46 | russellb | ooh |
| 22:36.50 | k31th | lol |
| 22:36.57 | Qwell[] | JerJer: You should totally give russellb and I stock. |
| 22:37.01 | Qwell[] | (kidding, of course) |
| 22:37.02 | k31th | ok you dont want to tell me |
| 22:37.25 | JerJer | how about options? :) |
| 22:37.33 | Qwell[] | I'll take options :p |
| 22:37.57 | k31th | JerJer: you do asterisk solutions? |
| 22:37.57 | Mercestes | I'll take options |
| 22:38.07 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 22:38.21 | russellb | yeah, i'll take options :-D |
| 22:38.32 | russellb | you're right here? --> http://www.ohiobarns.com/othersites/signs/nc/33-45batcave.jpeg |
| 22:38.35 | russellb | be there in a few |
| 22:38.49 | JerJer | i totally want that sign |
| 22:39.17 | Qwell[] | That's very large... |
| 22:39.25 | russellb | it's right here: http://tinyurl.com/2cbm6l |
| 22:39.37 | JerJer | k31th: i am one of many asterisk consultants |
| 22:40.12 | k31th | JerJer: and its working out well i take it? |
| 22:40.20 | Innatech | Best to be multifaceted. |
| 22:40.34 | tzanger | hahaha |
| 22:40.38 | tzanger | I'm looking up the LD50 of water |
| 22:40.44 | tzanger | and one report is giving it in depth (cm) |
| 22:40.49 | Innatech | hehe. |
| 22:40.53 | tzanger | cat: 38.6cm |
| 22:40.59 | tzanger | cat (in bag with rock) 16.5cm |
| 22:41.15 | Qwell[] | what is LD50? |
| 22:41.24 | Innatech | lethal dose, 50% of population |
| 22:41.27 | Qwell[] | oh |
| 22:42.06 | k31th | its possible to dissable PXE boot on most NICS right ? |
| 22:42.19 | tzanger | k31th: yes, but it's fairly well known, so why? |
| 22:42.20 | k31th | i cant find an option in this biox |
| 22:42.27 | k31th | bios. |
| 22:43.45 | tzafrir_laptop | tzanger, hmmm.. should be around 3 litters or even more |
| 22:44.05 | tzanger | I know it's more than that |
| 22:44.08 | tzanger | considerably more |
| 22:44.28 | tzafrir_laptop | It also depends on how dried you originally were |
| 22:46.04 | tzanger | yeah |
| 22:46.41 | tzanger | 3L is what's generally recommended and that is above and beyond what is normally taken in with food |
| 22:50.26 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:51.16 | *** join/#asterisk c6vette (n=Cori@ip70-176-167-93.ph.ph.cox.net) |
| 22:52.12 | c6vette | what would cause this: http://www.pastebin.ca/606569 I've seen it before but cant remember the solution |
| 22:53.16 | *** join/#asterisk andyd (n=andyd@212.183.134.130) |
| 22:57.29 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 22:58.40 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 22:59.10 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:05.16 | [TK]D-Fender | c6vette, the other side doesn't like your QUALIFY packets. |
| 23:08.53 | c6vette | So to get rid of it would be to turn off Qualify? Any other options? |
| 23:10.29 | *** join/#asterisk type0 (n=type0@239-9-178-69.static.gci.net) |
| 23:10.45 | type0 | has anyone gotten a linksys 'vonage' router to connect via sip to an asterisk box? |
| 23:12.05 | [TK]D-Fender | c6vette, convince their SIP router to not nag you :) |
| 23:12.21 | [TK]D-Fender | type0, Sure. UNLOCK it and reconfigure for your server. |
| 23:12.42 | [TK]D-Fender | type0, www.voxilla.com <- check out their forum, an best of luck, you may NEED it. |
| 23:13.30 | c6vette | type0: I have 30+ of them working with Asterisk, just need to unlock them.. :) |
| 23:13.49 | type0 | is it an easy thing to do? |
| 23:14.15 | type0 | ie, next 8 hours if i go and buy one right now? |
| 23:14.46 | [TK]D-Fender | type0, Why for the love of God are you looking to BUY trouble!? |
| 23:15.07 | [TK]D-Fender | type0, just order a normal never-locked ATA like the rest of the sane world! |
| 23:15.34 | generalhan | [TK]D-Fender: ok i have fixed the dialplan for the IAX2/ Dial, but its still not working ... the asterisk CLI doesnt even show an attempt but i do have the sip debug for the phone i tried to call out on. i also see no attempt on the remote server, so i dont think that its leaving here at all. http://generalhan.pastebin.ca/606593 |
| 23:16.13 | [TK]D-Fender | generalhan, # |
| 23:16.13 | [TK]D-Fender | Reliably Transmitting (no NAT) to 192.168.0.78:5061: |
| 23:16.13 | [TK]D-Fender | # |
| 23:16.13 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 23:16.20 | [TK]D-Fender | generalhan, your dialplan is WRONG. |
| 23:16.23 | [TK]D-Fender | generalhan, go fix it |
| 23:16.30 | generalhan | whats "Not Found" ? |
| 23:17.02 | [TK]D-Fender | generalhan, Means "Where the ^&@& is that exten in [internal]" <------------ |
| 23:17.15 | [TK]D-Fender | Looking for 8654 in internal (domain 192.168.0.42) |
| 23:17.44 | generalhan | why is that though ? i have the IAX configs setup to push it into the [incoming] context, not internal |
| 23:18.12 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 23:18.40 | type0 | [TK]D-Fender.. There arent any available here in my city, and I need it tonight |
| 23:18.53 | [TK]D-Fender | generalhan, You aren't even GETTING to dial the IAX peer! Your SIP end is being rejected! |
| 23:19.05 | [TK]D-Fender | type0, And why the huge rush? |
| 23:19.35 | [TK]D-Fender | type0, Wasting money & time on a very possible DEAD end is ridiculously stupid. |
| 23:19.49 | *** join/#asterisk canberk (n=canberk@85.103.108.250) |
| 23:21.44 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 23:22.16 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:23.18 | canberk | hello |
| 23:23.32 | generalhan | [TK]D-Fender: im soo confused, at first i was thinking this could be an issue with two different versions of *, but if youre saying its not even leaving my side to get to the IAX remote side ... i dont know what to think |
| 23:23.33 | canberk | even though i searched google, i couldn't understand what does jitter buffer really do |
| 23:24.00 | canberk | if anybody can be so kind to explain it in one sentence, i'll be glad |
| 23:24.01 | BSD_Tech | ok I have open ssl installed but why is ./configure not seeing it |
| 23:24.18 | [TK]D-Fender | generalhan, yupy dialplan is bad. Can I yell it any louder?!?!? |
| 23:24.21 | [TK]D-Fender | your* |
| 23:24.28 | k31th | ahh |
| 23:24.33 | k31th | channel activity |
| 23:24.38 | k31th | BSD_Tech: wats cracking |
| 23:24.57 | BSD_Tech | trying to update a box |
| 23:25.12 | BSD_Tech | and it seems asterisk ./configure is not seeing all the libx |
| 23:25.15 | generalhan | [TK]D-Fender: you *could* yell it louder ... but that wouldnt help anymore. why is my dialplan bad? thats what im trying to figure out. i heard you say it the first time, but i cant see where i went wrong |
| 23:25.16 | BSD_Tech | libs |
| 23:25.28 | Zodiacal | will asterisk's Shared Line Appearance feature allow users to see who is on a line? |
| 23:25.34 | Zodiacal | not just that the line is in use? |
| 23:26.53 | [TK]D-Fender | generalhan, pastebin it. |
| 23:26.59 | [TK]D-Fender | hfkjhfasfdlnquioewynrvqoewurynquioweyrnvqwrevn8907rejf1f |
| 23:27.10 | generalhan | the whole thing ? or just the part for the IAX Dial ? |
| 23:27.12 | [TK]D-Fender | Zodiacal, No, Presence will |
| 23:27.14 | *** join/#asterisk zapp-branigan (n=zapp-bra@124.22.220.87.dynamic.jazztel.es) |
| 23:27.26 | k31th | BSD_Tech: on BSD? |
| 23:27.33 | [TK]D-Fender | Zodiacal, *'s SLA = Hack of very limited usefullness |
| 23:27.35 | canberk | even though i searched google, i couldn't understand what does jitter buffer really do |
| 23:27.37 | BSD_Tech | no asterisknow |
| 23:27.43 | k31th | ohhhh |
| 23:27.46 | BSD_Tech | rpath |
| 23:27.47 | k31th | hows that going |
| 23:28.05 | BSD_Tech | zaptel libpri and everything else is ther |
| 23:28.10 | Zodiacal | tkd-fender presence? is that like hints? or is that another way of doing SLA? |
| 23:28.18 | BSD_Tech | but its not finding ssl |
| 23:28.29 | BSD_Tech | yet openssl is installed |
| 23:28.31 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/606597 |
| 23:28.45 | type0 | im assuming its in your path? |
| 23:29.12 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:29.37 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 23:29.43 | k31th | anyone in conf? |
| 23:29.43 | BSD_Tech | hould be I installed threw the conary |
| 23:29.59 | BSD_Tech | brb doorbell |
| 23:30.06 | k31th | before i bother pressing redial. |
| 23:30.33 | type0 | gimme infro to conf |
| 23:31.17 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:32.34 | k31th | 888@elastix.kicks-ass.net |
| 23:34.09 | tzafrir_home | if anybody wants to review my changes to zaptel's README . There's some nonsense there, I think: |
| 23:34.09 | tzafrir_home | http://svn.digium.com/view/zaptel/branches/1.4/README?r1=2703&r2=2702&pathrev=2703 |
| 23:34.09 | k31th | <PROTECTED> |
| 23:34.09 | k31th | <PROTECTED> |
| 23:34.09 | k31th | <PROTECTED> |
| 23:34.14 | k31th | upps |
| 23:34.32 | k31th | soz |
| 23:36.38 | [TK]D-Fender | <[TK]D-Fender> generalhan, Means "Where the ^&@& is that exten in [internal]" <------------ |
| 23:36.38 | [TK]D-Fender | <[TK]D-Fender> Looking for 8654 in internal (domain 192.168.0.42) |
| 23:36.38 | [TK]D-Fender | <generalhan> why is that though ? i have the IAX configs setup to push it into the [incoming] context, not internal |
| 23:36.50 | BSD_Tech | O have oh323 and ssl and ./configure is not finding them grrr |
| 23:37.18 | generalhan | [TK]D-Fender: ? |
| 23:37.23 | [TK]D-Fender | generalhan, You're ***SIP*** phone is looking at *****[internal]***** and that exten you showed me is in ****** [incoming] ****** |
| 23:37.27 | tzafrir_home | how nice: running my zaptel device inside qemu. "8192 samples in 82333 sample intervals -805.041504%" |
| 23:37.46 | [TK]D-Fender | generalhan, You're phone has NO FRIIGEN EXTEN TO FIND <------------------ |
| 23:37.51 | tzafrir_home | (the things there are very unoptimized, I must say. I don't even use kqemu) |
| 23:38.07 | [TK]D-Fender | generalhan, this has NOTHING to do with IAX |
| 23:38.17 | generalhan | [TK]D-Fender: my SIP phone is looking at the remote server's [internal] ? |
| 23:38.51 | generalhan | becasue i have exten => 8654 included in my local [internal] context |
| 23:39.41 | [TK]D-Fender | generalhan, look you must be ABSOLUTELY blind here. Your phone points to a context on the server its CONNECTED TO. That context "[internal]" doesb't have an exten named 8654 in it. |
| 23:39.52 | [TK]D-Fender | generalhan, pastebin the whole damned thing. |
| 23:40.15 | generalhan | [TK]D-Fender: yes it does ... and that was in the last pastebin i posted |
| 23:40.16 | [TK]D-Fender | gen it is screaming in your face that your dialplan is no good, and exectly WHERE. I've REPEATED it. |
| 23:40.22 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:40.24 | [TK]D-Fender | # |
| 23:40.25 | [TK]D-Fender | [incoming] |
| 23:40.25 | [TK]D-Fender | # |
| 23:40.25 | [TK]D-Fender | exten => 8654,1,Dial(SIP/8654,20,tT) |
| 23:40.35 | [TK]D-Fender | does that look like INTERNAL to you?! |
| 23:40.36 | generalhan | [TK]D-Fender: LOOK CLOSER ! |
| 23:40.46 | generalhan | that is CLEARLY labeled as the REMOTE machine |
| 23:40.50 | [TK]D-Fender | <[TK]D-Fender> [incoming] <----------------- |
| 23:41.00 | [TK]D-Fender | dsjs;d |
| 23:41.02 | [TK]D-Fender | ugh |
| 23:41.10 | generalhan | the LOCAL or HOST machine show [internal] |
| 23:41.14 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:41.15 | file | [TK]D-Fender: be calm! serenity now! |
| 23:41.20 | [TK]D-Fender | Ah, I see! |
| 23:41.22 | anonymouz666 | [TK]D-Fender flooder ;) |
| 23:41.37 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:41.47 | [TK]D-Fender | generalhan, # |
| 23:41.47 | [TK]D-Fender | exten => 8654,Dial(IAX2/Asterisk/${EXTEN}) <- wheee the hell is your PRIORITY on this line?!?! no WONDER it got IGNORED. |
| 23:41.58 | [TK]D-Fender | BINGO |
| 23:42.01 | generalhan | ahh , simple ! |
| 23:42.14 | generalhan | see something that i was soo easily over looking, you made the same mistake ! |
| 23:42.20 | [TK]D-Fender | generalhan, Put. Down. The. Crack. Pipe! (c) JerJer |
| 23:42.20 | generalhan | i appologize for all the confusion |
| 23:42.48 | [TK]D-Fender | *gasp*WHEEZE*choke*GUFFAW*lol*PUKE* |
| 23:43.02 | *** join/#asterisk [hC] (n=hardcore@s209-121-69-32.bc.hsia.telus.net) |
| 23:43.20 | tzafrir_home | generalhan, if you suspect something like this, comparge the dialplan you wrote to what you see in 'show dialplan CONTEXT' |
| 23:44.51 | generalhan | this whole process is killing me. it was soo easy to drop a couple of remote users with sip phones. but this server -> server stuff is way harder ... nothing is going correctly |
| 23:46.19 | [TK]D-Fender | generalhan, ... and imagine you couldn't even get a call into your server before you got all nuts on trying to make it OUT. |
| 23:47.11 | generalhan | i thought i had a pretty good grasp on this for local use only. |
| 23:47.23 | generalhan | i figured it would be just as easy to get a remote server setup |
| 23:47.54 | [TK]D-Fender | it is, but you went and screwed up the FIRST step :) |
| 23:47.55 | generalhan | and i could receive calls in, i cant make calls out only |
| 23:48.16 | [TK]D-Fender | ok, out for a bit |
| 23:48.18 | [TK]D-Fender | BBL |
| 23:48.29 | generalhan | and now im still getting errors ... but ill see what i can find |
| 23:51.54 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:51.54 | *** mode/#asterisk [+o mog] by ChanServ |
