03:12.56 | *** join/#asterisk infobot (~infobot@rikers.org) |
03:12.56 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
03:13.05 | *** join/#asterisk ritztech (446c874c@gateway/web/freenode/ip.68.108.135.76) |
03:13.13 | ChrisInSydney | looks like someone made it work with a SPA3102 on the FXO port |
03:13.30 | ChrisInSydney | ThinkGNU- |
03:13.58 | ritztech | anyone know why when in asterisk cli that commands dahdi is not appearing (like dahdi show channels) and such |
03:15.28 | kaldemar | ritztech: chan_dahdi.so is not loaded. |
03:15.41 | ritztech | i loaded it like 4 times |
03:15.54 | ritztech | reloaded processes a couple |
03:16.46 | ritztech | SOB ive been trying to restart modules and asterisk processes for the past hour and a final reboot fixed it ..... fyi i was doing a service stop dahdi service stop asterisk and /etc/init.d/asterisk stop / start etc |
03:16.55 | ritztech | but a good ol rebooty tooty worked |
03:18.01 | ThinkGNU- | I just saw your message ChrisInSydney |
03:18.22 | ChrisInSydney | cool |
03:19.18 | ThinkGNU- | thanks :-) |
03:21.08 | ChrisInSydney | might help |
03:21.26 | ChrisInSydney | I'd better get back to what I am suposed to be doing |
03:22.09 | ChrisInSydney | managed to not pay attention and brought across a live system config onto a dev machine. Forgot to disable the SIP registrations |
03:22.11 | ChrisInSydney | oops |
03:22.19 | ChrisInSydney | only 1 call in |
03:22.26 | ChrisInSydney | then I notied |
03:23.10 | ChrisInSydney | s/notied/noticed/ |
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03:56.30 | *** join/#asterisk AliciaK (~AliciaK@142.167.221.85) |
03:56.49 | ChrisInSydney | hi all, was I haloucinatiing, but I thought I read somewhere about choosing a codec for outbound calls in the dial plan ? |
03:57.10 | ChrisInSydney | is that possible in 1.8.1x.x ? |
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04:35.28 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |
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04:55.05 | ThinkGNU- | I GOT IT!!!!! |
04:55.11 | ChrisInSydney | woohoo |
04:55.13 | ChrisInSydney | how |
04:55.14 | ChrisInSydney | ?? |
04:55.19 | ThinkGNU- | damn thing! |
04:55.48 | ThinkGNU- | I had to tell the ata that the LineInUseVoltageThreshold was 20 instead of 25 |
04:56.07 | ThinkGNU- | I'm going to ramp it up again until it doesn't work, just to see where the breaking point is |
04:56.51 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
04:57.37 | ChrisInSydney | so I was sort of useful with the voltage thingy |
04:57.41 | ChrisInSydney | :) |
04:57.42 | ThinkGNU- | YES! |
04:57.57 | ThinkGNU- | It took me a while to really pin it down in my configs |
04:58.08 | ChrisInSydney | analog is a bitch |
04:58.15 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
04:58.22 | ChrisInSydney | I'm slack and use SIP endpoints |
04:58.27 | ThinkGNU- | It sure can be |
04:58.31 | ThinkGNU- | it's good to know how to use though |
04:58.34 | ChrisInSydney | makes for less head aches |
04:58.44 | ChrisInSydney | helpful if you can justify the effort |
04:58.50 | ThinkGNU- | you never know how much antiquated stuff you might end up working with at one point or another |
04:59.08 | _Corey_ | ThinkGNU-: Congrats... :) Analog stuff is definitely harder than SIP |
04:59.20 | ChrisInSydney | I swing between both cheap but time consuming and hand over the cash |
04:59.31 | ChrisInSydney | it depends on how many I install |
05:00.06 | ChrisInSydney | I hardly ever do pagers so I just use SIP, whereas I have hacked away at DD-WRT to do many thiings you twould think it couldnt |
05:00.07 | ThinkGNU- | Well, I've spent a fair amount of time rewiring our buildings for this system to go in this weekend and I wasn't about to let this old pager set me off schedule |
05:00.19 | ChrisInSydney | he he |
05:00.21 | ThinkGNU- | dd-wrt is pretty awesome |
05:00.50 | ChrisInSydney | I got caught today. A 30 pair tie between two buildings on opposite sides of the street. |
05:01.02 | ThinkGNU- | nice |
05:01.06 | ThinkGNU- | gotta love those |
05:01.24 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-tsidwhakqsclwlrv) |
05:01.24 | ChrisInSydney | we had a pair of Cisco 878s running a 2Mb link over a pair. I upgraded the link to 2 pairs = 4.6Mbps |
05:01.42 | ChrisInSydney | but the lines were a little dodgy and the quick disconnect shorted |
05:01.48 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
05:02.01 | ChrisInSydney | = no phones for around half an hour in the sales dept |
05:02.08 | ChrisInSydney | not happy jan |
05:02.11 | ChrisInSydney | :-( |
05:02.50 | ThinkGNU- | We have some CAT 3 going bewteen two buildings that wouldn't make an ethernet link with anything but this old 3com switch. I tried everything but couldn't get it to link with any other hardware. just too long of an appearent loop distance |
05:02.54 | ChrisInSydney | It took me 26 minutes to realise what happened, find a single reliable pair and reprogram back to 2.3Mbps single pair |
05:03.10 | ThinkGNU- | I just did a wireless point to point link for mine |
05:03.11 | ChrisInSydney | I had to cross a 3 lane dual carrage way 4 timies to fix |
05:03.14 | ChrisInSydney | during peak hour |
05:03.32 | ThinkGNU- | the fun things you run accross huh? |
05:03.38 | _Corey_ | so, ChrisInSydney is in Sydney I presume... ThinkGNU- where in the world are you? |
05:03.49 | ChrisInSydney | if you want to run VoIP and you have existing runs of Cat3 |
05:03.58 | ChrisInSydney | you can back to back 2 SHDSL modems |
05:04.23 | ChrisInSydney | _Corey_: Not only certifiied in Asterisk, but observant as well ;-) |
05:04.32 | ChrisInSydney | damn i key |
05:04.32 | ThinkGNU- | _Corey_ I'm in Wyoming, USA |
05:04.38 | _Corey_ | ahh |
05:04.38 | ChrisInSydney | cool |
05:04.46 | ChrisInSydney | _Corey_ yuourself ? |
05:04.53 | _Corey_ | I'm in the Philadelphia area |
05:04.57 | ThinkGNU- | I was going to do two modems but the wireless is actually point-to-multipoint |
05:05.09 | ThinkGNU- | so I can hit a third building that's never had service |
05:05.14 | _Corey_ | just up late watching today's TiVO haul of the olympics ;) |
05:05.18 | lorsungcu_ | gnu, what mfg for your radios? |
05:05.31 | ThinkGNU- | Ubiquiti |
05:05.40 | lorsungcu_ | good stuff |
05:05.53 | lorsungcu_ | i use mikrotik routers on all my deployments |
05:06.00 | ThinkGNU- | Those are pretty good too |
05:06.09 | lorsungcu_ | wonderful things |
05:06.12 | ThinkGNU- | I was impressed to find out that the old canopy engineers are working on the Ubiquiti stuff |
05:06.23 | lorsungcu_ | yes that air fiber is impressive |
05:06.32 | lorsungcu_ | i need 6k and an excuse to buy some.. |
05:06.45 | _Corey_ | lorsungcu_: "The Dude" eh? |
05:06.57 | lorsungcu_ | yes :) |
05:07.00 | _Corey_ | lol |
05:07.06 | _Corey_ | damn nice gui |
05:07.06 | ThinkGNU- | the dude? What am I missing here? |
05:07.13 | lorsungcu_ | really? |
05:07.18 | _Corey_ | it's been a few years since we've used it |
05:07.35 | lorsungcu_ | i haven't found anything quite like it |
05:07.57 | _Corey_ | yeah, they haven't done a really good job promoting it |
05:08.12 | lorsungcu_ | maybe its just in the US, but yeah |
05:08.13 | _Corey_ | i found it about 5 years ago in "linux journal" |
05:08.30 | lorsungcu_ | found it after i'd already bought a rb 450 heh |
05:08.39 | lorsungcu_ | pretty sweet bonus |
05:08.57 | lorsungcu_ | $100 router does gigabit, mols, ppp |
05:09.06 | lorsungcu_ | +awesome management? |
05:09.10 | lorsungcu_ | over many devices? |
05:09.16 | lorsungcu_ | can't beat it. |
05:09.40 | _Corey_ | we found it was pretty abusive from a bandwidth perspective |
05:09.56 | lorsungcu_ | you can tone it down. its most nice for upgrades |
05:10.25 | lorsungcu_ | don't do any wireless, but I can imagine it'd be good for checking on links, etc. |
05:10.36 | _Corey_ | we had some tunnel issues when it was running, we should revisit though... it's been a while |
05:11.14 | lorsungcu_ | what sort of tunnels? |
05:11.23 | ThinkGNU- | Well, it's late here and I have to be up in about six hours to continue on tomorrow morning. Thanks again guys! |
05:11.58 | *** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com) |
05:12.05 | _Corey_ | ThinkGNU-: good luck |
05:12.21 | Sean-Der | I have tried Set(${LANGUAGE}=te) Set(${LANGUAGE()}=te) and a couple others :/ |
05:12.35 | lorsungcu_ | wat |
05:12.43 | Sean-Der | I can't seem to get at the language global variable! |
05:12.54 | ThinkGNU- | Thanks _Corey_ |
05:12.57 | *** part/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
05:13.29 | kaldemar | Sean-Der: Set(CHANNEL(language)=te) |
05:13.48 | ChrisInSydney | <Sean-Der> SET(LANGUAGE=te) |
05:14.29 | ChrisInSydney | use ${VARIABLE} when you want to look at the contents such as NoOp(${LANGUAGE}) |
05:14.31 | Sean-Der | kaldemar: Thank you! That worked perfectly |
05:14.47 | lorsungcu_ | +1 kaldemar |
05:14.57 | ChrisInSydney | -1 me :-/ |
05:15.04 | Sean-Der | ChrisInSydney: +2 |
05:15.11 | ChrisInSydney | thanks |
05:15.12 | _Corey_ | lol |
05:15.13 | Sean-Der | Thank you for trying to help! |
05:15.35 | ChrisInSydney | its 3:15pm here I have no real excuse |
05:15.41 | Sean-Der | If only Asterisk had PyDoc or something, I find myself wandering so much |
05:15.46 | ChrisInSydney | except that I am trying to work |
05:16.00 | kaldemar | the LANGUAGE variable was deprecated in 1.4 already. |
05:16.08 | _Corey_ | ChrisInSydney: Do you do telecom stuff where you work? |
05:16.13 | ChrisInSydney | for myself |
05:16.24 | _Corey_ | ah |
05:16.26 | Sean-Der | ChrisInSydney: Tell your boss that if you don't contrib Asterisk will fail... you need to help out on IRC |
05:16.28 | ChrisInSydney | Small - mid size |
05:16.42 | ChrisInSydney | I remind myself every day |
05:17.19 | _Corey_ | Sean-Der: well, we wouldn't want to portray Asterisk as some fragile thing not "business grade" would we? |
05:17.26 | ChrisInSydney | I used to post on the forums, but I havent done so for a couple of years now |
05:18.16 | _Corey_ | what forums? you mean the mailing lists? |
05:18.18 | Sean-Der | _Corey_: I have given up trying to convince people to use Free software at this point. People either get it or they don't it seems |
05:18.37 | ChrisInSydney | http://forums.asterisk.org/ |
05:18.44 | _Corey_ | ah |
05:19.13 | ChrisInSydney | that way, if its in the forums, it sticks around, and I can always go back and find what I forgot about |
05:19.28 | ChrisInSydney | reminid myself of how clever i "Used" to be |
05:19.35 | ChrisInSydney | :/ |
05:19.39 | Sean-Der | I have one boss that takes my advice and it works, the other DEMANDS that some overstarched soul-less company is behind it. |
05:19.41 | _Corey_ | Sean-Der: well, open source has consistently represented 18% of the north american telephony market for the past 3+ years |
05:19.58 | _Corey_ | Sean-Der: Asterisk represents 80+% of that |
05:20.02 | ChrisInSydney | Sean-Der: I had that same argument with my recent partners |
05:20.11 | Sean-Der | ChrisInSydney: I cry everytime I see my old forum posts... they are so bad |
05:20.15 | _Corey_ | that's more than Cisco, NEC, Mitel, etc |
05:20.32 | ChrisInSydney | wouldnt listen. Ended up sneaking some FOSS on a few jobs. No issues what soever |
05:20.41 | _Corey_ | if people are dismissing open-source or Asterisk, they just don't get it |
05:21.18 | _Corey_ | I have Fortune-500 and ivy-league University customers running Asterisk |
05:21.24 | ChrisInSydney | _Corey_: And the Open Source part has been critical in augmenting some of the features for commercial deployments |
05:21.44 | _Corey_ | scratch the F on FOSS |
05:21.59 | _Corey_ | open-source is many things... free is not one |
05:22.05 | _Corey_ | FLEXIBLE is the keyword |
05:22.08 | ChrisInSydney | _Corey_ True, its never ever "free" |
05:22.21 | ChrisInSydney | it always comes at a cost |
05:22.24 | Sean-Der | _Corey_: Asterisk has gotten a lot more power behind the brand, but people who have been out of the industry for a little while still only accept Cisco, IBM, Mircosoft. At home I am an adamant Free Software guy ... at work I lax |
05:22.31 | ChrisInSydney | mine si tmie with my family |
05:22.47 | Sean-Der | Make sure not to release under BSD or they will just steal it all :p |
05:22.52 | ChrisInSydney | so put Ciisco handsets on tehir desks and switches in their racks |
05:23.14 | _Corey_ | Sean-Der: IBM gave a keynote 2 years ago at Astricon about their choice of Asterisk on their SMB appliances |
05:23.24 | ChrisInSydney | s/ so put Ciisco handsets on tehir desks and switches in their racks/ so put Cisco handsets on their desks and switches in their racks/ |
05:23.46 | ChrisInSydney | s/so put Ciisco handsets on tehir desks and switches in their racks/so put Cisco handsets on their desks and switches in their racks/ |
05:23.50 | _Corey_ | Cisco switches are fine... stay away from their handsets |
05:24.01 | lorsungcu_ | yeah... |
05:24.03 | ChrisInSydney | depends on the model |
05:24.09 | _Corey_ | they haven't supported SIP well since... um, like 2007 |
05:24.12 | lorsungcu_ | had a customer that wanted ~400 handsets moved to SIP |
05:24.17 | Sean-Der | _Corey_: And * is only gonna grow from there. |
05:24.18 | lorsungcu_ | this was last year |
05:24.31 | ChrisInSydney | Done a heap of SPA525G2s and SPA504s for a client. Thats what they wanted |
05:24.33 | _Corey_ | unless you're talking about the SPA stuff, which has suffered since the Cisco acquisition |
05:24.41 | lorsungcu_ | these were all 7940 |
05:25.02 | _Corey_ | well, if they have the investment in the 79xx stuff it is what it is |
05:25.08 | lorsungcu_ | its crap, either way |
05:25.17 | _Corey_ | I have a customer that has a few hundred endpoints on it |
05:25.17 | ChrisInSydney | Got a client next fortnight wit around 30 7971G-GEs and another 40 or so 7940s |
05:25.25 | Sean-Der | Luckily I don't have to do any hardware side stuff :) I enjoy being a software developer.. I really should checkout the Asterisk repo, I doubt I could do much though :( |
05:25.30 | ChrisInSydney | found a box of 7912s as well they gave me |
05:25.39 | lorsungcu_ | 7912 is a bitch |
05:25.44 | _Corey_ | 7912s don't really support SIP well |
05:25.45 | ChrisInSydney | really ? |
05:25.47 | lorsungcu_ | and 7905 |
05:25.48 | Sean-Der | The biggest C project I worked on was Fish-shell |
05:26.29 | _Corey_ | yeah, the 7940s and 7960s are the only Cisco "enterprise grade" handsets that support SIP |
05:26.38 | lorsungcu_ | support is used loosely. |
05:26.45 | _Corey_ | 7941s and 7961s are VERY sketchy |
05:26.48 | lorsungcu_ | they _do_ make and receive calls. |
05:26.49 | ChrisInSydney | Sean-Der: its not too hard to tweak things, such as back ports and upgrading patches to newer versions |
05:27.14 | _Corey_ | lorsungcu_: No, they're not bad... comparatively speaking |
05:27.21 | ChrisInSydney | _Corey_ lorsungcu_ The 7971s are working now on SIP |
05:27.40 | lorsungcu_ | yes the 79x1 works on sip for a while |
05:27.42 | ChrisInSydney | I did play with chan_sccp-b a while ago |
05:27.50 | Sean-Der | Excuse my ignorance, but what is Cisco pushing with their handsets? I have only ever encountered IAX*, RTP & SIP |
05:27.55 | _Corey_ | lorsungcu_: The features are limites on the 79x1 |
05:27.57 | lorsungcu_ | but users expecting cisco proprietary functionality on cisco sip firmware are always disappointed. |
05:28.02 | ChrisInSydney | they worked better on those |
05:28.14 | ChrisInSydney | Sean-Der: skinny |
05:28.25 | _Corey_ | Sean-Der: well, the world is moving to SIP and Cisco gets that |
05:28.29 | ChrisInSydney | = not SIP = cisco proprietory |
05:28.42 | ChrisInSydney | but they are moving to "Cisco SIP" |
05:28.44 | _Corey_ | they're still not 100% there with their Call Manager deployments |
05:28.58 | ChrisInSydney | = wont work properly with Asterisk SIP handsets |
05:29.17 | _Corey_ | but the SPA series (formerly Sipura) stuff is all designed for SMB and is SIP |
05:29.21 | ChrisInSydney | true |
05:29.22 | lorsungcu_ | and pretty decent |
05:29.34 | _Corey_ | eh |
05:29.41 | _Corey_ | I've found them to be buggy |
05:29.48 | ChrisInSydney | reaonably. Mind you we just ripped out 60 faulty 525G2s which randomly lock and reboot |
05:29.50 | lorsungcu_ | yes, but for $80, you can get a good handset. |
05:30.06 | Sean-Der | ChrisInSydney: Do they have a pretty open bug tracker and the such? I should check out the code and close a few bugs to get involved. Even if its stupid stuff (docs) |
05:30.08 | ChrisInSydney | 525G2s are a little more than $80 |
05:30.08 | _Corey_ | lorsungcu_: Yeah, you can get Digium phones |
05:30.12 | lorsungcu_ | bleh |
05:30.16 | lorsungcu_ | not a fan |
05:30.30 | *** join/#asterisk SQS (~SQS@94.20.72.138) |
05:30.44 | ChrisInSydney | _Corey_ lorsungcu_: +1 Snom |
05:30.52 | lorsungcu_ | yes |
05:30.59 | lorsungcu_ | 370 is my main bitch |
05:31.03 | _Corey_ | The Digium phones stack up better than Polycom or Snom and cost 20-30% less |
05:31.16 | _Corey_ | plus, they offer TONS of extra Asterisk features |
05:31.35 | ChrisInSydney | Sean-Der: Not sure, I just play with the released stuff, haven't published my "mods"...yet |
05:31.45 | ChrisInSydney | not too sure how to present the .patch file |
05:31.55 | lorsungcu_ | _Corey_: someday ill find time to learn how they work |
05:32.08 | lorsungcu_ | until then, i'll keep using the tons of other phones that _all work the same_ |
05:32.23 | _Corey_ | lorsungcu_: They have an XML spec that's pretty sweet, and then there's the DPMA... |
05:32.36 | lorsungcu_ | yeah, i've got a few here |
05:32.52 | lorsungcu_ | had a short look at them, just don't have time to get into it |
05:32.57 | lorsungcu_ | until i do, i like my snoms :) |
05:33.02 | _Corey_ | Don't forget... 80% of the Asterisk development is done by Digum on its own $... let's try to support them when we can |
05:33.10 | lorsungcu_ | i will |
05:33.28 | _Corey_ | especially when they actually make a better product than their competitors! |
05:33.38 | lorsungcu_ | but the way they handled the phones bugs me a bit. pm me if you want to hear it, not going to get into it out here.. |
05:33.42 | _Corey_ | that's how we keep Asterisk free at the end of the day |
05:34.31 | coppice | Digium is left to do a lot of the development because they actively drove other developers away |
05:36.01 | Sean-Der | coppice: What did they do? |
05:36.38 | coppice | various things. leaving contributions to rot has been one of their more popular moves |
05:37.32 | Sean-Der | Thats sad.. |
05:39.51 | ChrisInSydney | There have been some nice patches such as the codec negotiation patch from the Sippy people that would have been nice |
05:40.32 | ChrisInSydney | As I always say, if you dont like how an ope source project is run, you can always go and "fork yourself" |
05:40.35 | ChrisInSydney | :D |
05:41.14 | coppice | freeswitch is largely developed by people who wrote a lot of asterisk, but finally got pissed off and moved on |
05:43.17 | Sean-Der | coppice: Is freeswitch a fork or a rewrite? |
05:43.54 | Sean-Der | They use git... that might be enough reason for anyone to favor it haha |
05:43.56 | coppice | there is no common code in freeswitch, other than things both projects absorbed from elsewhere |
05:44.10 | _Corey_ | Freeswitch is another product altogether |
05:44.17 | ChrisInSydney | The anglican church was created by Henry VIII because he finally got pissed off with the catholic church and moved on |
05:44.50 | coppice | he forked. he didn't start from scratch |
05:44.55 | _Corey_ | well, Catholicism still has more souls |
05:44.57 | _Corey_ | ;) |
05:44.59 | ChrisInSydney | thats right |
05:45.14 | ChrisInSydney | I thiink it was his forking around that pissed off the pope |
05:45.20 | Sean-Der | And in the end they all killed each other over a mystical man in the sky..... |
05:45.36 | ChrisInSydney | at least that is what catholic historians would like us to believe |
05:45.58 | Sean-Der | haha |
05:46.02 | ChrisInSydney | Sean-Der. No they killed eachother over money, land weath and power |
05:46.08 | coppice | most of europe was pissed off with the pope at that time |
05:46.12 | ChrisInSydney | they just used God as an excuse |
05:46.44 | _Corey_ | Well, anyhow, Asterisk (and Digium) has a community, a support infrastructure, as well as great annual conference... |
05:46.45 | Sean-Der | coppice: Sorry for bothering you so much, but how different are the paradigms of VoIP in Freeswitch compared to Asterisk? |
05:47.24 | Sean-Der | It is pretty easy to jump back and forth, or did they attack problems differently? |
05:47.36 | coppice | oooh, you're just trying to start some rants :-) |
05:47.47 | ChrisInSydney | _Corey_: True. You heading south to Astricon this year ? |
05:48.07 | ChrisInSydney | +1 coppice |
05:48.29 | _Corey_ | ChrisInSydney: yeah, I'm doing a session |
05:48.45 | ChrisInSydney | Sean-Der: More so than OpenSIPS v Kamailio |
05:48.58 | ChrisInSydney | _Corey_ cool, what on > |
05:48.59 | ChrisInSydney | ? |
05:49.25 | ChrisInSydney | Sean-Der: Just threw that one in for you ;-) |
05:49.49 | Sean-Der | coppice: hahha maybe I am... No just curious I have fooled around with stuff like (Bluebox?), but Asterisk is the one I always come back to |
05:49.57 | Sean-Der | ChrisInSydney: Thank you :p |
05:49.57 | lorsungcu_ | Sean |
05:49.58 | lorsungcu_ | http://freeswitch.org/node/117 |
05:50.31 | _Corey_ | ChrisInSydney: It's a talk about a case study of a large call center running asterisk... definitely worth seeing if you'll be there :) |
05:50.52 | ChrisInSydney | bit of a haul, but I'll see what the $$$ are like |
05:51.24 | ChrisInSydney | not too good right now, and I have a few family things to do in Oct / Nov |
05:51.46 | ChrisInSydney | holiday budget and my training budget is already spent |
05:52.03 | Sean-Der | I wish I could go to AstriCon! Maybe next year |
05:52.03 | ChrisInSydney | would like to go though, catch up with some of the VUC folks too |
05:52.29 | ChrisInSydney | Sean-Der: http://vuc.me |
05:52.50 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
05:53.05 | ChrisInSydney | oh yeah, and a house to repair |
05:53.27 | ChrisInSydney | thats the discressionary spend gone |
05:53.49 | Sean-Der | Well its 2 AM here, so I have to get up for work in 5 hours sighhh.. Thanks for the link ChrisInSydney I am looking it right now |
05:54.30 | ChrisInSydney | download some of the podcasts, there is some good stuff there, Asterisk, Freeswitch, YATE, SER/Kamailio, SIPS |
05:54.55 | ChrisInSydney | plus heaps of good VoIP / SIP / online communication stuff |
05:55.05 | Sean-Der | Cool! Thanks |
05:55.06 | ChrisInSydney | (shameless plug) |
05:55.44 | ChrisInSydney | I must also get back to it. |
05:56.01 | ChrisInSydney | cheers all |
05:56.12 | lorsungcu_ | later |
05:56.29 | ChrisInSydney | one thing before I go |
05:56.36 | ChannelZ | BABABOOEY |
05:57.09 | ChrisInSydney | a post from before: |
05:57.13 | ChrisInSydney | <ChrisInSydney> hi all, was I haloucinatiing, but I thought I read somewhere about choosing a codec for outbound calls in the dial plan ? |
05:57.13 | ChrisInSydney | <ChrisInSydney> is that possible in 1.8.1x.x ? |
05:57.40 | ChrisInSydney | save the translation when calling out on g711 but internally on 722 |
05:57.49 | ChrisInSydney | anyone ? |
05:58.04 | ChrisInSydney | Hi ChannelZ |
06:00.04 | kaldemar | ChrisInSydney: there are variables for it in chan_sip. |
06:00.33 | ChrisInSydney | kaldemar: So set the variable, then dial ? |
06:00.48 | kaldemar | ChrisInSydney: SIP_CODEC and SIP_CODEC_OUTBOUND |
06:00.51 | ChrisInSydney | https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables |
06:00.55 | ChrisInSydney | :D |
06:01.03 | ChrisInSydney | ill play and report back |
06:01.06 | ChrisInSydney | thanks |
06:01.54 | ChrisInSydney | I dont care what they say about you on #freeswitch. Youre alright :D |
06:02.18 | kaldemar | say what? |
06:02.40 | ChrisInSydney | he he he he |
06:03.23 | ChrisInSydney | ;-) |
06:09.35 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
06:19.02 | ChrisInSydney | hi, a quick "best practices" question |
06:19.32 | lorsungcu_ | go on |
06:20.06 | ChrisInSydney | using the '-' character is discouraged in peer names, what about context names. ? |
06:20.15 | ChrisInSydney | I am tidying up legacy code |
06:20.19 | _Corey_ | tmm |
06:20.21 | _Corey_ | umm rather |
06:20.27 | _Corey_ | it's pretty common in context names |
06:20.42 | _Corey_ | FreePBX uses it extensively |
06:21.13 | ChrisInSydney | they do too, and we all know what they say about kaldemar ;-) |
06:21.30 | ChannelZ | I use them |
06:22.03 | ChannelZ | [handle-incoming] and [fax-in] etc |
06:22.08 | _Corey_ | Yeah, I'm not sure what you mean but don't let the FreePBX thing throw you then... they're fine |
06:22.31 | ChrisInSydney | I was using them in peer names, which made for a lots of changes to code to cope with them being used asa seperator character |
06:23.03 | ChrisInSydney | _Corey_ A previous post when kaldemar helped me |
06:23.15 | _Corey_ | ah |
06:23.26 | ChannelZ | hmm I use them in my peer names too |
06:24.01 | ChrisInSydney | Z: II find it buggers up some reports and packages that manage channels and CDRs |
06:24.15 | ChrisInSydney | s/II/I/ |
06:24.55 | ChrisInSydney | s/II/I Must get my keyboard fixed as well/ |
06:25.20 | ChrisInSydney | all cool |
06:25.22 | ChrisInSydney | cheers |
06:27.42 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
06:31.41 | kaldemar | ChrisInSydney: make up your mind already. freeswitch of freepbx? :) |
06:31.55 | ChannelZ | FREEBIRD! |
06:32.09 | ChrisInSydney | ANGRYBIRD |
06:33.10 | ChrisInSydney | +1 <kaldemar> |
06:33.34 | ChrisInSydney | FREEBEER (my favourite) |
06:33.43 | kaldemar | freebeard! |
06:36.05 | ChrisInSydney | its the beard without the mo' that has always creeped me out ever since I was a kid |
06:37.24 | ChrisInSydney | found something with dialplan logic and pattern matching... |
06:38.24 | ChrisInSydney | ... if you have a pattern in a context and an included context with an exact match, the pattern in the original context is used first. It ignores the exact match in the included context |
06:38.42 | ChrisInSydney | eg [context1] |
06:38.50 | ChrisInSydney | include => context2 |
06:39.18 | ChrisInSydney | exten => _123X,1,Playback(tt-monkeys) |
06:39.24 | ChrisInSydney | [context2] |
06:39.43 | ChrisInSydney | exten =>1234,1,Playback(tt-weasels) |
06:39.49 | ChrisInSydney | it plays monkeys |
06:39.58 | ChrisInSydney | ast 1.8.13.something |
06:41.42 | ChrisInSydney | it doesnt matter if the include is before or after the pattern |
06:42.17 | lorsungcu_ | hmm |
06:42.35 | lorsungcu_ | i think that is on purpose |
06:43.41 | kaldemar | ChrisInSydney: that's how the order works. when in a context, extensions are matched first, then includes and switches. |
06:43.59 | ChrisInSydney | its a bit of a pain. i am trying to separate out different tenant's inbound numbers, but I want a catch all if I miss one |
06:44.14 | ChrisInSydney | each tenant's inbound is in its own file |
06:44.23 | kaldemar | ChrisInSydney: and includes in the order they appear in the context, they are not handled as a single entity. |
06:44.34 | ChrisInSydney | kaldemar: thanks |
06:45.07 | ChrisInSydney | a coffee and a short rethink is in order |
06:45.20 | ChrisInSydney | back to it |
06:47.10 | *** join/#asterisk Maliuta (~nobusines@59.167.214.92) |
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06:53.09 | *** join/#asterisk ahmadhasssani (~ahmad.2.h@203.130.22.202) |
06:53.33 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
06:53.34 | schmidts | good morning |
06:53.41 | lorsungcu_ | hi |
06:54.07 | ahmadhasssani | gudmorning |
06:54.26 | ahmadhasssani | i am preparing asterisk server 1.4 |
06:54.45 | ahmadhasssani | can somone guide me |
06:54.52 | ahmadhasssani | coz i am tottaly new |
06:54.55 | ChrisInSydney | ~book |
06:54.56 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
06:56.56 | ahmadhasssani | hello |
06:57.05 | ahmadhasssani | can someone plz guide me |
06:57.08 | ChrisInSydney | ahmadhasssani: There are a heap of hello world examples to get Asterisk going. Check out the googles. The book is an excellent place to start. There are also some Pakt publications wich walk you through step by step |
06:59.15 | ahmadhasssani | i am setting up a server |
06:59.24 | ahmadhasssani | <PROTECTED> |
06:59.37 | ahmadhasssani | i have downloaded all pakes |
06:59.46 | ahmadhasssani | *pakages |
07:00.05 | ahmadhasssani | using centod 5.5 |
07:01.05 | lorsungcu_ | why 1.4? |
07:02.05 | kaldemar | ahmadhasssani: you might want to reconsider the version: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
07:02.17 | ahmadhasssani | any one there |
07:02.18 | ahmadhasssani | ? |
07:02.48 | ahmadhasssani | i am using 1.4.13 |
07:02.58 | ahmadhasssani | asterik 1.4.13 |
07:03.11 | ahmadhasssani | need help in insttalation |
07:03.35 | li3p | what's problems? |
07:03.59 | ahmadhasssani | problem is that when i insttal libpre pakage |
07:04.11 | ahmadhasssani | it says permission denied |
07:05.03 | ahmadhasssani | libpri-1.4.11.5 |
07:05.50 | ChrisInSydney | kaldemar: I put the catchall into an included context. If you cant beat them, driink them under the table after |
07:06.18 | li3p | you install the program as root? |
07:06.29 | rolandow | [Aug 2 09:05:44] WARNING[6237]: app_dial.c:2444 dial_exec_full: Invalid timeout specified: '-1'. Setting timeout to infinite |
07:06.34 | rolandow | how can i set this properly? |
07:06.38 | rolandow | will 0 be infinite as well? |
07:07.20 | ChrisInSydney | ouch !!! I just core dumped on a dialplan reload |
07:07.38 | ChrisInSydney | Asterisk 1.8.13.0 |
07:07.48 | lorsungcu_ | Chris, i think it just drank you under the table. |
07:08.01 | ChrisInSydney | last drinks |
07:08.24 | ChrisInSydney | actually I had a few at the pub the other night |
07:08.37 | ChrisInSydney | got home, tapped the wife on the shoulder. |
07:08.57 | ChrisInSydney | She said "You only want to have sex with me when you are drunk" |
07:08.59 | ChrisInSydney | I replied.... |
07:09.11 | ChrisInSydney | "no, sometimes when I am drunk, I want a kebab |
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07:09.19 | gg608f | haha |
07:09.39 | lorsungcu_ | love kebab |
07:09.43 | lorsungcu_ | sober too.. |
07:09.43 | ChrisInSydney | thats a true joke |
07:09.53 | ChrisInSydney | ive finished with it |
07:10.08 | lorsungcu_ | i have not found a similar thing in america |
07:10.08 | ChrisInSydney | +1 lorsungcu_ |
07:10.20 | kaldemar | rolandow: or just don't set it. |
07:10.24 | lorsungcu_ | only when i go to europe. then i get fat til i return |
07:10.25 | ChrisInSydney | yeros = greek kebab |
07:10.46 | ChrisInSydney | no shwarmas ? |
07:10.49 | ChrisInSydney | heaps here |
07:10.53 | lorsungcu_ | nada |
07:10.56 | rolandow | kaldemar: yes, but i put it in a variable which depends on an if statement |
07:11.04 | lorsungcu_ | i swear if i made it, i would be a millionaire. |
07:11.07 | gg608f | mexican tacos al pastor is similar to kebab |
07:11.10 | lorsungcu_ | drive through kebab |
07:11.22 | rolandow | kaldemar: something like: if you can find second-line SIP, timeout 30 sec, otherwise, don't timeout. |
07:11.23 | ChrisInSydney | Cleavelan street has so many turksish, lebanese, etc places |
07:11.25 | kaldemar | rolandow: 0 and empty translate to 136 years or something. |
07:11.33 | gg608f | good kebab replacement in america |
07:11.44 | rolandow | kaldemar: ok so 0 would do |
07:12.05 | lorsungcu_ | never had it, gg608f |
07:12.10 | ChrisInSydney | maybe thats what I need to be doing, launch my kebab chain at AstriCon |
07:12.15 | lorsungcu_ | do it |
07:12.18 | lorsungcu_ | i will cook |
07:12.23 | lorsungcu_ | we will bank |
07:12.34 | rolandow | [Aug 2 09:12:16] WARNING[7067]: app_dial.c:2444 dial_exec_full: Invalid timeout specified: '0'. Setting timeout to infinite |
07:12.43 | lorsungcu_ | roland |
07:12.45 | rolandow | hm... what is the NULL variable |
07:12.46 | lorsungcu_ | make it empty |
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07:13.08 | ChrisInSydney | back to work for me |
07:13.13 | rolandow | <PROTECTED> |
07:13.24 | rolandow | how do i make that empty?? should i say "":30 at the en? |
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07:13.28 | ChrisInSydney | now I feel hungry |
07:13.29 | rolandow | or NULL:30 ? |
07:13.41 | Roelt | ?:30 might work |
07:14.08 | lorsungcu_ | or an empty variable, if that doesnt? |
07:14.26 | rolandow | ok "" works |
07:14.36 | lorsungcu_ | hm |
07:14.36 | rolandow | Roelt: :30 wouldn't set it if the if is true |
07:14.43 | gg608f | yeah tacos al pastor is an evolution of kebab, made by lebanese in mexico, they add onion, cilantro and spicy sauce |
07:14.45 | gg608f | sometimes pineapple |
07:14.48 | rolandow | Roelt: and it's in a loop, so it could be set previously :) |
07:15.09 | lorsungcu_ | where do i get this mexikebab, gg |
07:17.04 | gg608f | any mexican place |
07:17.05 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
07:17.16 | gg608f | i live in california, so bunch of mexican places |
07:17.16 | ChrisInSydney | cilantro = coriander ? |
07:17.21 | lorsungcu_ | im in MN |
07:17.27 | lorsungcu_ | there are only some. |
07:17.42 | lorsungcu_ | but i will be on the lookout |
07:17.54 | gg608f | yep, coriander |
07:18.00 | gg608f | i may try kebab :) |
07:18.31 | ChrisInSydney | we have a mex place in staggering distance. He goes back to Mexico to record with a bunch of his mates each year. |
07:18.42 | ChrisInSydney | I have a few of his CDs |
07:19.00 | ChrisInSydney | he is a good guitarist, not a bad voice |
07:19.12 | lorsungcu_ | i used to have a similar setup |
07:19.15 | lorsungcu_ | but it was taco bell... |
07:19.20 | ChrisInSydney | :-? |
07:19.32 | lorsungcu_ | did many drunk drive through-walks |
07:19.36 | lorsungcu_ | :/ |
07:19.38 | ChrisInSydney | :D |
07:19.49 | gg608f | nice |
07:19.51 | ChrisInSydney | done that, in a shopping trolley |
07:19.55 | lorsungcu_ | man |
07:20.15 | lorsungcu_ | its maybe the funniest thing, ever |
07:20.16 | gg608f | , i heard couple of mexican gutarists in coachella "rodrigo y gabriela" amazing |
07:20.19 | lorsungcu_ | esp if you make car sounds.. |
07:20.20 | gg608f | hahhaa |
07:20.38 | *** part/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
07:20.46 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net) |
07:21.43 | coppice | gg608f: the heavy metal nylon guitarists :-) |
07:22.40 | gg608f | :) |
07:22.58 | lorsungcu_ | here is something important |
07:23.00 | lorsungcu_ | http://en.wikipedia.org/wiki/List_of_sandwiches |
07:27.38 | gg608f | man…irc is making me fat |
07:28.26 | gg608f | dagwood.... |
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07:48.09 | ChrisInSydney | exten => _04XXXXXXXX,n,Set(CHANNEL(SIP_CODEC_OUTBOUND)=alaw) |
07:48.09 | ChrisInSydney | <PROTECTED> |
07:48.27 | ChrisInSydney | is that correct, or am I missing something ? |
07:49.08 | kaldemar | CHANNEL is a function that has certain specified fields. it is not used to set any channel variable. |
07:49.26 | kaldemar | SIP_CODEC_OUTBOUND is a variable, not a field of the CHANNEL function. |
07:49.27 | ChrisInSydney | what about these: https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables |
07:49.32 | ChrisInSydney | ahh |
07:49.41 | kaldemar | so Set(SIP_CODEC_OUTBOUND=alaw) |
07:50.08 | ChrisInSydney | cool |
07:50.12 | ChrisInSydney | brb |
07:52.15 | ChrisInSydney | back |
07:52.16 | ChrisInSydney | nup |
07:52.43 | ChrisInSydney | tried _SIP_CODEC_OUTBOUND and got [Aug 2 19:05:17] NOTICE[17760]: chan_sip.c:6455 try_suggested_sip_codec: Changing codec to 'alaw' for this call because of ${SIP_CODEC} variable |
07:52.43 | edgars | yello |
07:52.54 | ChrisInSydney | but |
07:53.05 | ChrisInSydney | 192.168.231.201 HTI_extn202 3ec4263c2814-qr 0x1000 (g722) No Rx: ACK HTI_extn20 |
07:53.09 | ChrisInSydney | still transcoding |
07:53.19 | ChrisInSydney | edgars: Hey |
07:53.44 | edgars | yo ChrisInSydney |
07:54.15 | ChrisInSydney | so 722, then alaw on the handset and alaw only on the SIP trunk |
07:56.36 | ChrisInSydney | exten => _04XXXXXXXX,n,Set(_SIP_CODEC_OUTBOUND=alaw) |
07:56.53 | ChrisInSydney | isnt good enough |
07:56.59 | ChrisInSydney | exten => _04XXXXXXXX,n,Set(_SIP_CODEC=alaw) |
07:57.00 | ChrisInSydney | exten => _04XXXXXXXX,n,Set(_SIP_CODEC_OUTBOUND=alaw) |
07:57.03 | ChrisInSydney | works :-) |
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08:10.16 | Scapal | Cisco SPA500DS : extension pannel with LCD labels. I've bee waiting for those. I could only find traces of those in the latest Release Note of the phones firmware (20120724). |
08:10.35 | Scapal | http://www.static-cisco.com/assets/prod/pct/spa500ds.jpg |
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08:11.37 | Scapal | Not yet released apparently |
08:12.00 | schmidts | wow looks nice scapal |
08:12.22 | Scapal | There is no full product description yet but you can get some by comparing it with the 500S: http://www.cisco.com/en/US/products/ps10499/prod_models_comparison.html |
08:13.25 | schmidts | i guess it will just be the same like the 500S but with display ;) |
08:13.44 | schmidts | hmm maybe it will also work to display a callerid on a ringing blf key ;) |
08:13.54 | schmidts | then this stuff will really rocks |
08:20.21 | Roelt | show waiting calls in a queue, number of voicemails |
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08:46.25 | aberrios | I see the MYSQL application has a connection timeout argument but there doesn't seem to be a way to timeout a query that takes too long to respond. Would I have to sort this out at the MySQL end? |
08:49.46 | dexteruk | I have a stupid problem with AGI using a php script |
08:50.13 | *** join/#asterisk dexteruk (~dexteruk@77.70.92.171) |
08:51.02 | dexteruk | hi im trying to write a simple AGI script using PHP |
08:51.14 | dexteruk | i cannot get it to STEAM FILE |
08:51.29 | dexteruk | it runs the command, but i hear no sound |
08:52.05 | dexteruk | if i PLAYBACK before i execute the AGI the audio file plays |
08:52.29 | dexteruk | i am running php -q |
08:52.45 | dexteruk | to stop any HTML codes being returned to asterisk |
08:53.22 | kaldemar | dexteruk: perhaps the AGI stream command does not answer the line like Playback does and you need to use the answer command before it. |
08:53.23 | dexteruk | Does anyone have any experiance with AGI scripts? |
08:54.09 | dexteruk | there is an Answer at the begining before it calls the AGI script |
08:54.30 | dexteruk | or you mean in the AGI script itself |
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08:55.56 | kaldemar | dexteruk: enable agi debug with "agi set debug on" and pastebin the CLI output of a call. |
09:05.22 | dexteruk | http://pastebin.com/KbxB5g3Y |
09:05.26 | dexteruk | here is the output |
09:06.55 | dexteruk | here is the AGI script itself http://pastebin.com/hAbUSkkY |
09:07.48 | dexteruk | when it says the number it works fine |
09:07.54 | dexteruk | just when it comes to streaming the file |
09:08.02 | dexteruk | it fails |
09:08.18 | dexteruk | i have tried with full path to the file too |
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09:55.17 | sbw_DvD | dexteruk: you could try the phpagi class http://phpagi.sourceforge.net/ |
09:55.59 | sbw_DvD | they have some better output handling which avoids the broken pipe errors |
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10:07.13 | awk | hmm, anyone using this amr codec from sourceforge... I'm not sure about the 1.8.*_crypto_offer patch, it remove SHA1_80 encryption offer used by SRTP (Polycom, Aastra, Snom, etc) |
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10:27.47 | ChanSpy | I have an issue with Chanspy / Extenspy |
10:28.44 | ChanSpy | It works if it's started before any spied channel |
10:29.17 | ChanSpy | if i start it after, it connects to the channel, but no sound is coming |
10:30.04 | ChanSpy | whisper / quiet mode doesn't change anything to this issue |
10:32.20 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
10:32.29 | *** join/#asterisk jurij1 (~jurij@ip-43-82.sn2.eutelia.it) |
10:32.46 | jurij1 | hello i'm using elastix. is there a way i can have fax clients see only their received/sent faxes? |
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10:41.18 | *** join/#asterisk pbxMan (c335d9a2@gateway/web/freenode/ip.195.53.217.162) |
10:43.41 | pbxMan | does anybody know whether is possible to change the installation directory for all asterisk folders without using links? |
10:44.40 | pbxMan | is it right to change the default.h file? |
10:45.55 | ChrisInSydney | II think you can do that during make ? :-/ |
10:46.07 | bulkorok | check ./confgure --help |
10:46.53 | bulkorok | there you'll find Installation directories: --prefix=PREFIX --exec-prefix=EPREFIX |
10:50.08 | pbxMan | OK cheers mate |
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10:58.08 | dax_roc | <PROTECTED> |
11:15.54 | *** join/#asterisk danfromuk (~IceChat77@2.27.1.139) |
11:16.24 | danfromuk | Hi, how can I convert origtime of a voicemail message to standard time format? |
11:16.35 | *** join/#asterisk Sidrov (~sid@85.186.159.201) |
11:16.38 | danfromuk | what format is origtime in? |
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11:19.29 | Sidrov | hello all |
11:23.29 | Sidrov | is there any active user ? |
11:25.48 | leifmadsen | eh? |
11:31.04 | dax_roc | Can you call a skype user via sip like channel originate SIP/skype/username extension ### ? |
11:31.27 | leifmadsen | perhaps if you have Skype For SIP |
11:31.55 | dax_roc | I get a 484 , "address incomplete" |
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12:17.36 | ChrisInSydney | hey, can someone help me with call park on 1.8.x |
12:17.37 | ChrisInSydney | ? |
12:19.26 | ChrisInSydney | porting from my legacy 1.4 systems, I have parking defined between 70000-79920 (long story) |
12:19.59 | ChrisInSydney | usually I would do a Set(PARKINGEXTEN="70202") followed by a Park() |
12:20.02 | ChrisInSydney | but it doesnt work |
12:20.05 | ChrisInSydney | .... |
12:20.19 | ChrisInSydney | becuase I have "" in the Set :-/ |
12:20.24 | ChrisInSydney | just noticed that |
12:20.36 | ChrisInSydney | is embarrassed |
12:20.56 | ChrisInSydney | is crawling back into his hole |
12:24.23 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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12:28.03 | pouyou | hi |
12:28.08 | ChrisInSydney | hey |
12:28.36 | pouyou | i have a quick question someone might be able to answer |
12:29.08 | ChrisInSydney | shoot |
12:29.30 | ChrisInSydney | but not like in a batman movie |
12:29.36 | pouyou | let's say i have a sip call between user A and B. Is it possible at some point via the dialplan to play a certain sound or video file to one of them while they are on the call |
12:29.52 | dax_roc | Anyone here worked with a matrix comsec GS12? |
12:29.52 | pouyou | bang ! |
12:29.56 | ChrisInSydney | look in features |
12:30.27 | dax_roc | *GS12 GE12 |
12:30.53 | ChrisInSydney | pouyou: Just had this page open myself: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AdditionalConfig_id256654.html |
12:32.00 | ChrisInSydney | there is some DTMF stuff there for playiing recordings or running AGI |
12:32.15 | ChrisInSydney | I have played with it and it worked on my old 1.4 |
12:32.32 | ChrisInSydney | works with 1.8 as well. Just a bit better I guess |
12:32.58 | leifmadsen | pouyou: try looking at my presentation from AstriCon last year |
12:33.23 | pouyou | sure, do you have a link ? |
12:34.07 | leifmadsen | http://www.astricon.net/videos/Cooking-with-Asterisk.html |
12:34.25 | pouyou | ChrisInSydney, do not really see something I could use in the additionalConfig so far |
12:34.27 | leifmadsen | it's the 5th recipe I show |
12:35.31 | ChrisInSydney | pouyou: additionalConfig...smells like freePBX |
12:35.55 | ChrisInSydney | ? |
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12:36.35 | pouyou | i was mentioning the link you gave me :) |
12:36.37 | leifmadsen | unfortunately the player kind of sucks and doesn't have a time stamp on it, so I can't really find and tell you where to start watching |
12:36.52 | ChrisInSydney | ahh |
12:37.07 | ChrisInSydney | I was only reading the parking stuff |
12:37.10 | ChrisInSydney | :/ |
12:37.42 | leifmadsen | pouyou: leifmadsen.com should also have my presentation from last year that matches the video link I gave you and might be easier to get to what you want |
12:37.43 | pouyou | yeah |
12:38.03 | pouyou | i kinda was trying to dl the video as the player sucked so much :P |
12:38.14 | leifmadsen | indeed |
12:38.25 | WIMPy | never managed to get at the videos. |
12:38.33 | ChrisInSydney | [applicationmap] |
12:38.35 | leifmadsen | really? it's just a link.... |
12:38.41 | ChrisInSydney | testfeature => #9,peer,Playback,tt-monkeys |
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12:38.49 | ChrisInSydney | [myGroupName] |
12:38.55 | ChrisInSydney | testfeature => #9 |
12:39.20 | ChrisInSydney | then put Set(DYNAMIC_FEATURES=myGroupName) in the dial plan |
12:39.43 | ChrisInSydney | leifmadsen is a far better cook than me. |
12:40.06 | rolandow | leifmadsen: why not use vimeo or youtube? i can't even play the video |
12:40.07 | ChrisInSydney | I can only cook toast |
12:40.11 | WIMPy | A link to a page wit a video none of my download helpers or video plugis detects. |
12:40.27 | leifmadsen | rolandow: I had zero say or interaction with creating the video other than to do the presentation |
12:40.54 | rolandow | too bad.. maybe you can rip it and upload it to youtube under a different nickname ;-) |
12:41.01 | leifmadsen | maybe you can |
12:41.34 | WIMPy | Maybe not that easy if we can't even watch it. |
12:41.44 | leifmadsen | I was watching it just fine |
12:41.53 | rolandow | it doesn't play without stuttering |
12:41.59 | rolandow | or what is it called in english .. |
12:42.00 | leifmadsen | Fedora 17 w/ Google Chrome |
12:42.03 | leifmadsen | no issues here |
12:42.04 | kaldemar | leifmadsen: did you do it with applicationmap or with originating a call to ChanSpy? |
12:42.16 | leifmadsen | kaldemar: not sure, you'll have to check my presentation |
12:42.21 | leifmadsen | it's been over a year since I looked at what I did |
12:42.29 | kaldemar | leifmadsen: i tried, but gave up. :P |
12:42.33 | leifmadsen | I probably used stuff and things |
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12:45.52 | *** join/#asterisk jkroon (~jkroon@dsl-244-38-51.telkomadsl.co.za) |
12:46.06 | jkroon | is it possible to disable's asterisk's internal log rotation? |
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12:46.19 | rolandow | how can you be so sure if you don't remember |
12:48.01 | rolandow | maybe you used some Techniques .. who knows? |
12:51.40 | leifmadsen | rolandow: I know because I wrote it down and tested it |
12:51.46 | ChrisInSydney | hmmm. Just playing with park and such and even though the console says "Will timeout back to extension [t02-parkinglot] s, 1 in 120 seconds" it throws the call back to the extension who parked the call |
12:51.50 | leifmadsen | not remembering doesn't mean it didn't happen |
12:52.14 | ChrisInSydney | leifmadsen: Ive had nights like that |
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12:52.25 | leifmadsen | ChrisInSydney: :) |
12:52.25 | rolandow | of course, but how can you be sure you used stuff and things then .. |
12:52.37 | leifmadsen | because I use stuff and things in everything i do |
12:52.50 | rolandow | i see |
12:52.52 | rolandow | i rest my case then |
12:53.29 | rolandow | just curious, where do you draw the line between stuff and things? |
12:53.51 | leifmadsen | rolandow: I tend to use more stuff than things, but it depends on the implementation |
12:54.12 | ChrisInSydney | you can always use a thin veneer of BS when you get stuck |
12:54.37 | rolandow | do you use anythings as well? |
12:54.46 | rolandow | or more somethings |
12:54.48 | leifmadsen | rolandow: nah, totally overrated |
12:55.31 | ChrisInSydney | I used to have a good supply of whatcha-ma-giggys |
12:55.34 | rolandow | taking notes |
12:57.05 | ChrisInSydney | but the supplier started supplying lots of you know and didnt have room for both |
12:57.11 | rolandow | i just used two apples |
12:57.22 | rolandow | trying to get the hunger feeling go away |
12:57.35 | rolandow | Junami apples .. the Elstars aren't sold anymore .. i guess Elstar season is over. |
13:00.06 | Roelt | is currently jailbreaking an apple |
13:00.37 | Roelt | but, had some grapes this morning as breakfast |
13:00.51 | ChrisInSydney | is is going to grab an orange |
13:01.26 | ChrisInSydney | 2 bags of Aussie Navels for $5. Cant complain about that |
13:06.03 | Katty | hello my asterisk does not work at all how to fix??? answer plz. |
13:06.17 | malcolmd | Katty: did you try turning it off and on again? |
13:07.03 | Katty | what is turn off???? *hee* |
13:08.11 | ChrisInSydney | Katty: Hold down the shift key |
13:09.34 | ChrisInSydney | Do you think I should take my son to Astricon for his birthday. He is turning 3 that week |
13:09.59 | Katty | no |
13:10.00 | ChrisInSydney | hmmm, how many hours from Sydney to LA |
13:10.07 | Katty | i think that's a terrible idea for a 3 year old |
13:10.18 | Katty | they need things to entertain them, and naps |
13:10.24 | ChrisInSydney | thats what I thought too |
13:10.26 | Katty | and if he starts bawling in the middle of a talk, no one wants to hear that. |
13:10.42 | ChrisInSydney | The reminder email turned up |
13:10.52 | ChrisInSydney | looked at the date, already booked |
13:11.08 | Katty | take him to someplace COOL |
13:11.49 | ChrisInSydney | good idea. I'll take him for a ski this year |
13:12.18 | ChrisInSydney | hes old enough and end of September / OCtober the weather is niceish |
13:12.26 | Katty | are there lots of fun Hands On stuff at sking? |
13:13.40 | ChrisInSydney | my dad took me skiing at 2.5 |
13:14.15 | ChrisInSydney | he held on to me and I skied between his legs |
13:14.28 | ChrisInSydney | get him started early |
13:15.30 | ChrisInSydney | but not too long. It has to be fun |
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13:59.54 | areay | hi all.. when installing asterisk 1.8.15.0 i'm getting an error with libspeex1: libspeex1 conflicts with asterisk (<= 1:1.4.18.1~dfsg-1) |
14:00.13 | areay | does this mean i need to downgrade libspeex1? |
14:00.59 | ChrisInSydney | Katty: Seriously, with a toss up between Astricon and hanging around with my son and a bunch of three year olds running around on a sugar high....Ill take the sugar high any day |
14:01.41 | pabelanger | areay: you need a newer version of libspeex |
14:01.53 | pabelanger | what version of Debian? |
14:02.10 | areay | pabelanger it's ubuntu 10.04 |
14:02.28 | pabelanger | where are you installing asterisk from? |
14:02.41 | areay | i'm installing from source and using checkinstall |
14:02.55 | areay | compiling rather. sorry it's early |
14:09.47 | Kobaz | anyone have problems with polycom phones losing their mwi? like there is 2 messages waiting, and after restarting asterisk then there's no message light |
14:10.10 | areay | pabelanger, what's the best way of upgrading libspeex? |
14:14.49 | ChrisInSydney | Kobaz: are you using Asterisk Realtime ? |
14:15.20 | ChrisInSydney | also, FYI, the MWI message is a SIP notify message from the Asterisk box to the handset to set the MWI |
14:15.59 | Kobaz | yeah |
14:16.12 | Kobaz | it should redo the NOTIFY's on startup |
14:16.27 | Kobaz | i'm using static realtime voicemail |
14:16.34 | ChrisInSydney | I think there was somethng I read where the handset would request the MWI, while others relied on a push |
14:16.49 | Kobaz | well there's voicemail subscribe |
14:16.50 | ChrisInSydney | try without the realtime voicemail |
14:17.00 | Kobaz | that's not really an option |
14:17.35 | Kobaz | the phone will subscribe to the box which will then tell asterisk to send it mwi's |
14:17.38 | ChrisInSydney | there is a line in the blues brothers movie I could use right now |
14:18.07 | Kobaz | what i should do is some tests and do some packet dumps on restart and see if it's resending mwi |
14:18.41 | ChrisInSydney | I ran into this with someone else, ast realtime VM and dodgy MWIs |
14:18.49 | ChrisInSydney | cant remember how it was resolved |
14:18.55 | ChrisInSydney | sorry |
14:18.57 | ChrisInSydney | :( |
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14:21.10 | Kobaz | http://www.voip-info.org/wiki/view/Asterisk+Realtime+MWI+Hacks |
14:22.12 | Kobaz | what i can do, is add a timer to chan_sip to send out periodic notifys if it doesn't do so already |
14:22.47 | ChrisInSydney | sounds right |
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14:26.27 | bulkorok | hi... if I delete the txt and WAV files in the "Old" vm-folder, are they gone, so that the box is at zero!? |
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14:30.17 | ChrisInSydney | bulkorok: you can trash the whole extension folder and it will be recreated when Asterisk starts |
14:32.59 | bulkorok | I caonnot restart asterisk :-( |
14:33.45 | WIMPy | hands bulkorok a kill -9 |
14:34.04 | ChrisInSydney | you can, but you dont want to |
14:34.08 | ChrisInSydney | go on |
14:34.15 | bulkorok | :) |
14:34.36 | bulkorok | I'll try with a new box :-D |
14:34.39 | ChrisInSydney | restart when convenient |
14:34.58 | bulkorok | but good to know anyway ... |
14:35.10 | *** join/#asterisk uskerine (~Uske@231.Red-79-157-169.dynamicIP.rima-tde.net) |
14:35.31 | ChrisInSydney | the trouble is convenient for the system and convenient for me sometime are different |
14:35.45 | WIMPy | Don;t say you can;t if you can, but don't want to. |
14:36.07 | ChrisInSydney | yes, you can always choose the bullet |
14:36.51 | ChrisInSydney | goes away on a short brain break |
14:39.52 | uskerine | hi, i have installed asterisk package in my distro and i have noticed that some extensions and dialplan (demo-ael) are installed by default |
14:40.16 | uskerine | aprat from the files I have modified (sip.conf, extensions.conf, queues.conf, chan_dahdi.conf, etc.) |
14:40.22 | uskerine | what else should be kept in /etc/asterisk? |
14:40.53 | mirela666 | uskerine: probably extensions.ael file |
14:41.51 | uskerine | i also see pbx_lua and pbx_config while (re)loading dialplan |
14:41.58 | uskerine | is that also added stuff? |
14:43.19 | uskerine | Registered extension context 'app_queue_gosub_virtual_context'; registrar: app_queue |
14:45.14 | mirela666 | uskerine: all the config files in /etc/asterisk are mostly needed to run normaly, the number of them depends on modules and apps you chose on 'installation' ./menuselect |
14:46.24 | [TK]D-Fender | Which he didn't do |
14:46.32 | mirela666 | :) |
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14:48.05 | uskerine | ok thks |
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14:50.57 | bulkorok | info: I moved the wav and txt files out of the Old folder and the voice in of voicemialmain told me there are no messages... problem solved without restart :-) |
14:51.16 | ChrisInSydney | cool |
14:51.51 | bulkorok | still the problem why the option to delete messages was not announced :( |
14:52.08 | bulkorok | there were 123 messages... |
14:52.13 | ChrisInSydney | ? |
14:52.15 | ChrisInSydney | :/ |
14:53.26 | bulkorok | ah ok... I don't investigate this...: version 1.4.23.2 |
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15:07.35 | straterra | I know its not an asterisk question, but you guys are knowledgeable about phone stuff in general.. |
15:08.27 | straterra | I need a device that will take a 3.5mm audio input and let me stream it over a network to a decoder with 3.5mm. The two sites are connected by fiber (no copper) and need to have shared paging..any recommendations from experience? |
15:08.54 | uskerine | recorded audio with Mixmonitor sounds strange (like metalical sound) |
15:09.18 | uskerine | so human voice sounds a bit like a robot |
15:09.22 | uskerine | any idea on what could it be? |
15:10.24 | straterra | Point to multipoint would be awesome too |
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15:22.34 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
15:22.39 | mcolombo | hi all |
15:23.04 | Katty | hihi |
15:25.14 | slav3_kitten | yo Katty *waves* |
15:25.22 | Katty | ohai |
15:25.24 | Katty | how'rechu dear? |
15:27.17 | mcolombo | i've a problem with hangupcause. In the SIP CANCEL Request the field "cause" is not present! |
15:27.38 | mcolombo | I've already add in my dialplan this : NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})}) |
15:28.38 | mcolombo | and add use_q850_reason=yes in sip.conf |
15:29.00 | mcolombo | but does not work, anyone can help me? |
15:29.01 | mcolombo | Thanks |
15:29.49 | VultureZ | Anyone utilizing OpenSIPS? |
15:29.58 | VultureZ | for LB |
15:31.03 | *** join/#asterisk defswork (~andy@cpc17-sutt4-2-0-cust175.perr.cable.virginmedia.com) |
15:34.10 | dax_roc | Can I do a pass through on a pri interface with asterisk? |
15:35.13 | dax_roc | * I want to filter only some calls and reinject them to an exsisting pabx via a man in the middle setup. Which would put us in a better place to migrate group by group ( if that makes sense ) |
15:36.07 | dax_roc | Pri -> Asterisk(pass - through) -> Legacy (PABX) |
15:38.20 | [TK]D-Fender | Call it ... acll out. |
15:38.22 | [TK]D-Fender | call* |
15:38.25 | [TK]D-Fender | al there is to it |
15:38.56 | carrar | You can do anything with asterisk! |
15:44.44 | straterra | Can I call an extension and have my coffee maker make me a cup? |
15:44.51 | Qwell | straterra: Yes, yes you can. |
15:44.58 | Qwell | It has been done (more than once). |
15:47.23 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
15:51.28 | [TK]D-Fender | Yes,, that is what my * does for me already |
15:51.32 | [TK]D-Fender | and is my jukebox... |
15:52.47 | [TK]D-Fender | http://purl.rikers.org/%23asterisk/20100423.html.gz |
15:53.30 | [TK]D-Fender | http://ibot.rikers.org/%23asterisk/20081111.html.gz |
15:53.34 | [TK]D-Fender | Oooohh, 2008.. |
15:53.41 | [TK]D-Fender | wonder just how long ago I did that... |
16:02.58 | ChrisInSydney | strange thing. CONNECTEDLINE stopped working on my dev system, but it was working :-? |
16:03.25 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
16:04.44 | Katty | i have ham and potatoes in my fridge. |
16:04.47 | Katty | what should i make for dinner |
16:05.40 | chuckf | that sounds like a good dinner to me, maybe make them hash browns |
16:07.37 | dax_roc | eggs ham and spam ... |
16:09.43 | newtonr | just microwaving a sweet potato is cool sometimes |
16:11.39 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
16:16.52 | vastina | newtonr: how modest |
16:17.10 | newtonr | AND THEN ADD BACON |
16:17.23 | vastina | \o/! bacon makes everything better |
16:36.19 | ChrisInSydney | Wierd. I have two accounts on the same box with the same phone |
16:37.10 | ChrisInSydney | "Set(CONNECTEDLINE(name,i)=Something" works on one reg but not the other |
16:37.12 | ChrisInSydney | :-/ |
16:38.05 | ChrisInSydney | sip is the same except for alaw v g722 being first in the list |
16:38.14 | ChrisInSydney | clues anyone ? |
16:38.29 | ChrisInSydney | same profile on the handset |
16:38.34 | ChrisInSydney | handset is a Snom 970 |
16:38.36 | ChrisInSydney | 870 |
16:38.46 | ChrisInSydney | s/970/870/ |
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16:56.01 | kuku | I can't get cdr adaptive odbc to come up as a cdr module.. any ideas ? I'm using 1.8 - and after confnigure cdr adbaptive odbc has XXX - and its written that its in the core. |
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17:04.49 | numenory | having bri issues on a b200p, supplier says configs are correct asterisk 1.8.15, libpri-1.4.12, openvox-dahdi-linux-complete-2.6.1+2.6.1, call connects briefly is crackly, then cuts off, console outputs.... SIP/33-00000001 answered DAHDI/i2/07952015843-1 |
17:04.51 | numenory | [2012-08-02 10:59:12] ERROR[7682]: chan_dahdi.c:14135 dahdi_pri_error: PRI Span: 1 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established) |
17:04.52 | numenory | == Primary D-Channel on span 1 down |
17:04.54 | numenory | [2012-08-02 10:59:12] WARNING[7682]: sig_pri.c:1126 pri_find_dchan: Span 1: D-channel is down! |
17:04.55 | numenory | PRI Span: 2 TEI=0 MDL-ERROR (C): UA (F=1) in state 7(Multi-frame established) |
17:04.57 | numenory | == Primary D-Channel on span 1 up |
17:04.58 | numenory | -- Span 1: Channel 0/1 restarted |
17:04.59 | numenory | [2012-08-02 10:59:12] ERROR[7683]: chan_dahdi.c:14135 dahdi_pri_error: PRI Span: 2 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established) |
17:05.01 | numenory | -- Span 2: Channel 0/1 restarted |
17:05.03 | numenory | == Primary D-Channel on span 2 down |
17:05.04 | numenory | [2012-08-02 10:59:12] WARNING[7683]: sig_pri.c:1126 pri_find_dchan: Span 2: D-channel is down! |
17:05.06 | numenory | after that you cannot connect anymore until you restart dahdi |
17:05.08 | numenory | debian 6, box |
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17:14.21 | Kobaz | aaaaah |
17:14.38 | Kobaz | so asterisk does not resend sip notify after restart |
17:14.47 | Kobaz | using realtime voicemail |
17:14.59 | Kobaz | in 1.8.12 anyway |
17:15.49 | Kobaz | ChrisInSydney: should be easy to fix |
17:16.16 | ChrisInSydney | you would hope so |
17:16.39 | ChrisInSydney | you can use the AMI to send SIP NOTIFY messages, but your content has to be in reverse |
17:16.42 | ChrisInSydney | last line first |
17:17.39 | Kobaz | no i mean like |
17:17.48 | Kobaz | just make app_voicemail send NOTIFYs on load |
17:19.14 | Kobaz | that's the easy way |
17:22.16 | ChrisInSydney | cool |
17:22.32 | ChrisInSydney | my Park BLFs dont work |
17:23.01 | ChrisInSydney | regular BLFs do work |
17:23.12 | Kobaz | ooooh, more interesting tidbits |
17:23.21 | Kobaz | it actually does send a notify, but it's inaccurate |
17:23.33 | Kobaz | Messages-Waiting: no |
17:23.34 | Kobaz | <PROTECTED> |
17:25.24 | ChrisInSydney | these phones are on drugs. Now Park 1 works, park2 nahda |
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18:42.43 | xSmurf | hey all, I'm trying to get asterisk 1.6.2.9 and meetme working in a domU, but all the instructions I find seem outdated, zaptel is gone and the dahdi source do not include USE_RTC anywhere.... |
18:43.10 | xSmurf | everything else I find is "google, the answer is there".. super helpful considering all the answers on google do not work |
18:43.33 | sruffell | xSmurf: if you can load / compile DAHDI you're good. No more need for dahdi_dummy, etc... |
18:43.41 | xSmurf | I see |
18:44.26 | xSmurf | I already have the dahdi pkgs but I can't seem to modproble dahdi |
18:44.32 | xSmurf | *modprobe |
18:44.40 | xSmurf | (Debian squeeze) |
18:45.17 | sruffell | what does it say when you try to modprobe? |
18:46.17 | xSmurf | module not found |
18:47.11 | xSmurf | starts a 'find / -name "dahdi.so"' |
18:47.18 | xSmurf | yeah, non existant |
18:47.22 | sruffell | use .ko not .so |
18:47.41 | xSmurf | thx, I tried so cause "module load dahdi" returned "Error loading module 'dahdi': /usr/lib/asterisk/modules/dahdi.so: cannot open shared object file: No such file or directory" |
18:47.44 | xSmurf | no .ko either :/ |
18:48.16 | xSmurf | maybe I need to install the non-free pkg?! |
18:48.22 | sruffell | I would just build from source then. |
18:48.36 | sruffell | I'm not that familiar with the debian packages. |
18:49.16 | xSmurf | building from source is not the greatest for production usage |
18:49.39 | xSmurf | I can be harsed to go and manually check every other day for new security fixes and recompile everytime |
18:49.45 | VultureZ | Anyone utilizing OpenSIPS as a load balancer for multiple * deployments? |
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18:53.52 | xSmurf | sruffell: nevertheless, thanks for the help, it was enlightening |
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19:16.25 | sruffell | xSmurf: yeah…I hear you there. |
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19:52.04 | UnixDev | hi everyone, I'm using realtime peers but for some reason mohsuggest does not seem to be pulled from the database… is there something I don't know about this? |
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20:01.13 | navaismo | Hi if anyone is interested in a call monitor program like YAACID here is another one, for linux and windows http://dl.dropbox.com/u/1277237/DM_CallMonitor.tar.gz |
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20:05.59 | sebastianpersic | I didn't create the asteriskpbx user before install... what should I do |
20:06.24 | [TK]D-Fender | make it after if you care |
20:06.34 | [TK]D-Fender | it's all int he book |
20:07.21 | sebastianpersic | ok.. because I saw some text during boot up my system that it didn't knew user asterisk and group asterisk |
20:17.01 | [TK]D-Fender | sebastianpersic, and we don't know what your base OS is, or how you installed *, and what any of it is expecting |
20:25.22 | [TK]D-Fender | checkout time, later all |
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20:43.27 | WIMPy | numenory: Looks like you've got a bad connection. Either in hardware or in software. |
20:48.50 | *** join/#asterisk Mango45 (~Mango45@d209-89-215-139.abhsia.telus.net) |
20:49.26 | Mango45 | In what variable is the audio IP for an in-progress call? |
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20:55.59 | kuku | someone please help me with getting the adaptive odbc showing up as a registered backend. pretty please. |
20:58.14 | doolittlework | what do you mean kuku? |
20:58.17 | kuku | I've spend 3 hours on trying to get it to work. I have odbc.ini settings, res_odbc.conf, cdr_adaptive_odbc.conf, cdr.conf, odbcinst.ini configured.... Running 1.8, so adaptive is part of "core" and doesn't ahve to be compiled. |
20:58.48 | kuku | "cdr show status" shows cdr-custom and csv only |
20:58.48 | mjordan | kuku: uhm, no, it still has to be compiled |
20:59.19 | kuku | mjordan: which option in menuselect? |
20:59.44 | xSmurf | so "|" was changed to ",", is the warning just a deprecation warning or is | actually broken?! |
20:59.45 | kuku | cdr_Adaptive_odbc has XXX where other options have [*] |
21:00.12 | xSmurf | sruffell: on top of that dahdi includes binary blobs which is really unacceptable in my situation |
21:00.23 | mjordan | kuku: then that would mean it can't be compiled |
21:00.42 | mjordan | kuku: highlighting it should tell you what the dependencies are |
21:00.43 | xSmurf | for the |... I should specify from 1.4 > 1.6 |
21:00.57 | kuku | mjordan: res_odbc |
21:01.11 | mjordan | kuku: okay, is res_odbc enabled? |
21:01.27 | kuku | aaaaaaaaaaaaaaaa :) |
21:01.57 | kuku | its not enabled, because it needs generic_odbc, and ltdl |
21:03.00 | sruffell | xSmurf: cool…yeah, I guess just need to get the debian packages installed |
21:03.13 | kuku | let me run ./configure again... |
21:04.14 | xSmurf | sruffell: I mean, the dahdi-non-free pkg didn't install the appropriate kernel module... building from source downloads a whole bunch of binary blobs |
21:04.17 | kuku | mjordan: :) can I kiss you ? I've had a very bad day |
21:04.22 | *** join/#asterisk Micc_ (~Micc@50.125.113.34) |
21:04.24 | kuku | and now it works!! |
21:04.39 | mjordan | good :-) Glad to hear it |
21:04.40 | navaismo | 0_o |
21:04.50 | mjordan | (but I'm happily married, thanks) |
21:04.58 | kuku | now i can go and do the other 200 things I have to do |
21:05.26 | sruffell | xSmurf: there is only one binary blob that is code..the rest is firmware for the various cards. You can edit Kbuild to only build dahdi.ko if you just want software mixing. |
21:05.39 | kuku | mjordan: much thank you. two days wasted. |
21:05.47 | xSmurf | sruffell: thanks, that's good to know |
21:05.48 | sruffell | (and that binary blob is to load firmware too….no way around that unfortunately) |
21:06.29 | *** part/#asterisk Mango45 (~Mango45@d209-89-215-139.abhsia.telus.net) |
21:08.23 | sruffell | xSmurf: if you're only just using software mixing too…and you're ok with building…change DAHDI_CHUNKSIZE in include/dahdi/kernel.h from 8 to 40 in order to mix audio in 5ms increments instead of 1. Much more efficient if you don't need to talk to hardware. |
21:08.38 | sruffell | when all the audio is coming in at typically 20ms intervals anyway. |
21:09.00 | xSmurf | wow man, I'm gonna have a lot of documenting to do in our wiki ;) |
21:09.13 | sruffell | :) …or not. |
21:09.31 | xSmurf | well I a) need to remember all this b) should make this available to all more easily |
21:09.48 | xSmurf | we usually have pretty hefty docs on things we use and a lot of people seem to use it |
21:10.19 | sruffell | http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51777 |
21:10.45 | xSmurf | Danke! |
21:10.57 | sruffell | xSmurf: why do you even need DAHDI in a virtual machine? You can't use app_confbridge? You need page app? or some other reason? |
21:11.25 | xSmurf | hmmm maybe, we were using app_meetme for conferencing |
21:11.39 | xSmurf | I'm new here, and was charged with migrating the voice boxen |
21:11.51 | xSmurf | (I think they wanted to laugh at me banging my head on the desk :P) |
21:12.16 | sruffell | ok…afaik with a newish kernel (that supports timerfd) and asterisk-10….app_page is the only functionality that can't be done without dahdi currently. |
21:12.41 | xSmurf | when I tried app_meetme bitched at the lack of pseudo device :/ |
21:12.47 | xSmurf | *when I tried without dahdi |
21:12.52 | mjordan | sruffell: and in Asterisk 11, that won't be true either! |
21:13.18 | sruffell | yeah…app_meetme does all the mixing in the kernel. It was easier that way in the beginning (I'm assuming, since I wasn't there) since the drivers already implemented mixing in order to bridge channels between cards. |
21:13.47 | sruffell | mjordan: woot! |
21:14.41 | xSmurf | sweet |
21:15.45 | Qwell | sruffell: SLA :( |
21:15.53 | Qwell | but we don't talk about that |
21:16.23 | sruffell | ? I'm not following. Shared Line Appearence? |
21:16.38 | sruffell | xSmurf: a little Zaptel / DAHDI history http://thread.gmane.org/gmane.linux.kernel/40267 |
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21:16.56 | sruffell | regarding why mixing in the kernel. |
21:18.13 | Qwell | sruffell: yeah, app_meetme has some weird SLA stuff still. confbridge will never do that. |
21:18.23 | Qwell | (because it's a terrible idea) |
21:18.36 | sruffell | ahhhhhh.... |
21:18.46 | xSmurf | sruffell: sorry for the dumb question... what do you mean by "page app" |
21:19.11 | sruffell | if you wanted to broadcast a page to a bunch of endpoints… |
21:19.23 | xSmurf | ah I see |
21:19.31 | sruffell | basically it will connect to a bunch of endpoints, and drop them in a DAHDI conference to do the broadcast. |
21:19.54 | xSmurf | good question.. we might use it for our nagios integration for oncall admins |
21:20.03 | sruffell | wait for Asterisk-11 |
21:20.05 | sruffell | :) |
21:20.05 | xSmurf | (I just can't wait to get my name on that list ;p /s ) |
21:21.33 | Qwell | what list? |
21:22.19 | _Corey_ | there's a mailing list ;) |
21:22.51 | Qwell | _Corey_: I haven't heard you shouting from the rooftops about how awesome DPMA 1.2 is. I'm sad. :( |
21:22.58 | _Corey_ | haha |
21:23.05 | _Corey_ | I loaded it the other day |
21:23.20 | Qwell | well, I'm not hearing you shout at me either, so I guess it went okay. |
21:23.48 | _Corey_ | no problems to speak of... haven't had time to shake out the new features yet |
21:23.56 | Qwell | close enough |
21:24.16 | _Corey_ | it's a little unclear how the Queue app is supposed to work, so I was planning to poke at it a bit later this evening |
21:24.36 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-znqdipkdooqaquyn) |
21:24.47 | Qwell | it's a little funky, just know that you need a separate type=queue for each phone *and* for each Asterisk queue. |
21:24.54 | Qwell | so 3 users with 3 queues = 9 entries |
21:25.05 | Qwell | (assuming all 3 users are in all 3 queues) |
21:25.14 | xSmurf | Qwell: the on call admins ;) |
21:25.23 | Qwell | xSmurf: oh. yeah, good luck with that. |
21:25.27 | xSmurf | so I gotta make sure that asterisk box is rock solid ;) |
21:25.50 | _Corey_ | xSmurf: You don't want to be on that list... hire an answering service |
21:25.56 | xSmurf | :p |
21:26.17 | xSmurf | what do you think asterisk is for :P |
21:26.53 | _Corey_ | Qwell: makes sense... my mistake was trying to reverse engineer the FreePBX stuff you guys added for that |
21:27.03 | numenory | wimpy: still there? tried another isdn2 box on the wall same result, both boxes function ok with the existing pbx (not asterisk) |
21:27.17 | _Corey_ | if it works, I'm sure I'll figure it out... |
21:27.24 | Qwell | _Corey_: You probably don't need the login/logout_exten |
21:27.59 | Qwell | FreePBX uses them, because it has a bunch of magic it needs to do. |
21:28.45 | numenory | wimpy: not sure if i need the molex power plugged in, in te mode, and whether having it plugged in could cause the issue |
21:31.32 | _Corey_ | Qwell: hmm, I have type=application,application=queue in the queue entries in res_digium_phone_applications.conf |
21:32.00 | Qwell | fyi, nothing actually reads that file - in FreePBX, it's included |
21:32.04 | Qwell | #included* |
21:32.32 | Qwell | ie; just put it in res_digium_phone.conf, unless you've done #include magic |
21:32.44 | _Corey_ | yeah, res_digium_phone.conf #includes it |
21:32.46 | Qwell | if that's not the problem, I'd be happy to look over your config if you pastebin parts of it |
21:33.02 | Qwell | really, just the queue contexts |
21:33.17 | _Corey_ | thanks, I'll poke around a bit with it later when I have more time... I may bother you about it tomorrow :) |
21:33.43 | Qwell | also don't forget your application=contextname in type=phone contexts |
21:35.33 | _Corey_ | I'll let you know what happens. I'm having load balance problems at the moment... |
21:43.39 | WIMPy | numenory: Don't know that card, but it's unlikely you need extra power. Receiving garbage is often caused by IRQ issues, however. |
21:43.56 | WIMPy | Faulty cabling would be another possibility. |
21:44.05 | WIMPy | (i.e. wrong termination) |
21:45.39 | numenory | wimpy: sitting on irq 21 on its own |
21:46.23 | numenory | wimpy: was going to try a different slot |
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21:53.29 | numenory | winpy: do get this in dmesg - IRQ 21/b4xxp: IRQF_DISABLED is not guaranteed on shared IRQs |
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22:19.26 | rocksfrow | Hi |
22:19.41 | rocksfrow | I was hoping somebody could enlighten me |
22:20.00 | navaismo | ~ask |
22:20.00 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
22:20.16 | rocksfrow | i'm typing! heh |
22:21.04 | rocksfrow | I have a callmonitor script setup that's watching queue_log, and I didn't realize it but a few months ago the structure of queue_log changed... on CONNECT events the agent used to show in the format including the extension, for ex: Local/1234@from-queue |
22:21.28 | rocksfrow | Now I notice this has changed a few months ago and instead the fullname of the extension is being output there instead of the device/extension |
22:21.40 | rocksfrow | Is this configuration that was added, or changed default behavior? |
22:22.54 | rocksfrow | so anyways, the script was relying on the extension being there (via regex), but now instead of 'Local/5200@from-queue' being the value, it's changed to 'John Doe' |
22:23.36 | rocksfrow | Right now i'm looking at modifying the script and doing a lookup based on name instead which isn't a huge deal, but obviously my preferred solution would be a configuration change to cause * to log the extension # instead of name in queue_log |
22:23.43 | rocksfrow | Suggestions please? |
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22:33.39 | rocksfrow | Anyboddy? :-/ |
22:37.21 | rocksfrow_afk | plz any tips would be nice, bbs. |
22:43.21 | Micc | Does anyone still have the aastra problem that the phone never stops ringing sometimes? |
22:44.34 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
22:51.09 | newtonr | rocksfrow_afk: you hit the non-busy time, you might try posting on the asterisk-users list |
22:53.16 | vastina | hey phreaks... how can I get a serial number from a digium card from the CLI? |
22:53.30 | vastina | tried dahdi_scan() but couldn't find serial |
22:53.51 | WIMPy | shutdown -h now; use screwdriver |
22:54.11 | vastina | WIMPy: you assume so damned much when I speak... remote site... |
22:54.15 | vastina | :) |
22:54.21 | citywok | rocksfrow_afk: adapt your script to work either way :p -- lol |
22:54.24 | WIMPy | ok |
22:54.34 | WIMPy | shutdown -h now; drive there;use screwdriver |
22:54.42 | citywok | vastina: shutdown now -h; call onsite person; have htem use screwdriver |
22:54.43 | vastina | facepalms |
22:54.58 | WIMPy | Maybe better drive there before shutting down. |
22:55.11 | citywok | vastina: look at the invoice from when you bought it? the SN is probably on there. |
22:55.24 | WIMPy | Again, call before doing the shutdown. |
22:55.26 | vastina | ok so there's no way from software |
22:55.28 | vastina | thanks. |
22:56.04 | citywok | vastina: probably not, that would require embedding it in each chip which seems like more work. |
22:56.05 | WIMPy | I don't think they have a serial number other than in ink on paper. |
22:56.40 | citywok | the more softwarey devices have it, but that's because they have memory. (sip/pri gateway) |
22:58.02 | doolittlework | lol srewdriver |
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