| 00:02.34 | Zopiac | well I set up asterisk on a virtual machine and can connect to it fine from my main computer |
| 00:03.53 | Zopiac | just tried from the windows computer, it worked too |
| 00:08.13 | Zopiac | when I try to call, though, I get a "user not fount" error |
| 00:08.49 | Zopiac | er, I forgot to set up extensions |
| 00:17.22 | Zopiac | second user can't accept any calls... when it tries to call itself it gets a "remote user busy" error |
| 00:22.01 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 00:28.28 | Zopiac | and after restarting ekiga, the windows computer can no longer connect... |
| 00:31.08 | Zopiac | never mind, windows was just being annoying. restarted it again and it's OK |
| 00:31.14 | Zopiac | still calls itself busy though |
| 00:32.20 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:32.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:33.00 | *** part/#asterisk mazpe (~mazpe@c-174-61-76-85.hsd1.fl.comcast.net) |
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| 00:34.21 | slav3_kitten | sighs |
| 00:34.25 | bluregard | good evening all |
| 00:34.38 | Zopiac | 'evening |
| 00:34.41 | WIMPy | Zopiac: That's normal for most phones. |
| 00:34.59 | Zopiac | interesting |
| 00:35.16 | bluregard | I know this isn't exactly Asterisk related, but has anyone successfully gotten push notifications to work to Cisco SPA phones via the /CGI/Execute URL? |
| 00:35.30 | Zopiac | not sure why it's giving me that problem on this computer but not the other |
| 00:35.37 | Zopiac | same softphone software |
| 00:37.43 | slav3_kitten | have you tried threatening it with a high voltage capacitor? :) |
| 01:53.37 | slav3_kitten | guess it was not funny |
| 01:53.38 | slav3_kitten | sounded good in my head |
| 02:01.32 | jpsharp | That's always the way it works. |
| 02:02.25 | slav3_kitten | so true |
| 02:02.39 | slav3_kitten | then again i've threatened projects with the cap before when frustrated |
| 02:13.36 | jpsharp | I dont even have to threaten it. I just glare. Sort of like the "mom" look for code. |
| 02:14.21 | jpsharp | And I'm about ready to pull that glare with my Polycom. It won't auto answer. |
| 02:16.14 | slav3_kitten | lol |
| 02:16.30 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:19.34 | slav3_kitten | so jpsharp know of any decent sip phones you can get dirt cheap used? |
| 02:20.10 | jpsharp | The polycom 300 like what I have are $50 a pop. |
| 02:20.34 | slav3_kitten | googles |
| 02:20.58 | jpsharp | The grandstream Budgetone 200s are $50 a pop too. |
| 02:21.12 | slav3_kitten | that's an odd thing |
| 02:21.27 | jpsharp | I don't know about the grandstream phone, but the polycom 300s don't do POE. |
| 02:21.43 | slav3_kitten | that bites :( |
| 02:21.57 | jpsharp | You're going to be hardpressed to find a cheap phone that does POE. |
| 02:22.51 | slav3_kitten | eh personally i'd spend the money on a phone that does poe *points at the cisco gear he has* but... friend i am going to help setup after i learn asterisk has been pestering me about hard phones on the cheap |
| 02:23.41 | jpsharp | You can occasionally find the Cisco 7940/60s for $150 each, they do Cisco POE. |
| 02:23.41 | slav3_kitten | so he can deal with power bricks, me i hate anything i have to feed with power bricks and shove UPSs all over for |
| 02:24.02 | slav3_kitten | yea i have a 40 and 60 as well as an 11 |
| 02:24.15 | slav3_kitten | and a cisco 3500 poe switch |
| 02:24.29 | WIMPy | Wow. I thought even the cheap phones had POE.. |
| 02:24.34 | slav3_kitten | then again my home network closet has a 42u rack bolted to the floor |
| 02:24.44 | slav3_kitten | so i am a touch insane |
| 02:25.30 | jpsharp | This place seems to carry cheap 7940s. |
| 02:25.31 | jpsharp | http://www.metrolinedirect.com/cisco-7940-ip-telephone.html |
| 02:25.41 | jpsharp | $70 each. |
| 02:26.18 | jpsharp | No, you're not insane. A friend of mine lives in an 18000 sq ft warehouse. |
| 02:26.46 | jpsharp | Computer equipent ranging from PDP-8s to Cray SV-1s. |
| 02:26.48 | WIMPy | That sounds nice. |
| 02:26.48 | slav3_kitten | that's awesome, i live in a 2500sq ft manufactured home (glorified trailer) |
| 02:27.14 | slav3_kitten | and i ran like 1200 feet of cat 5e and 500 feet of rg6 for tv |
| 02:27.34 | slav3_kitten | ebay has those polycom phones for 30 bucks |
| 02:27.45 | jpsharp | I'll be doing that late next year when we buy a house. |
| 02:27.46 | slav3_kitten | which is likely what he's going to go with because he's a cheap ass |
| 02:28.27 | slav3_kitten | jpsharp, my only suggestion is this... |
| 02:28.46 | jpsharp | I like mine. POE isn't a deal killer and now that I've got it auto answering it does what I need. |
| 02:29.55 | jpsharp | Now, when my wife is in the office and doesn't have her cell on her, I can dial into the asterisk box and page her. |
| 02:30.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:30.32 | slav3_kitten | for me personally PoE is a deal killer because i like everything to be right in one location for ease of powering an what not |
| 02:32.31 | jpsharp | Unfortunately, the houses around here are solid slab. No crawlspace/floor access. Crawl spaces are unheard of in Florida. |
| 02:32.56 | slav3_kitten | that sucks! |
| 02:33.14 | jpsharp | Hell, *all* of the walls in my current house are concrete block. Interior and exterior. |
| 02:33.45 | jpsharp | Hurricane/bulletproof, but damn inconenvient when it comes to running cable & hanging things. |
| 02:33.46 | slav3_kitten | do you live in a bunker? |
| 02:34.23 | jpsharp | 95% of the houses down here are concrete block for the exterior. Hurricane resistant and quick to build. |
| 02:34.27 | slav3_kitten | nah, make it like some fallout bunker, run 2" conduit home runs on the ceiling with drops to boxes below and swap all your lighting for hazardous location lamp |
| 02:36.22 | slav3_kitten | then again wife may not approve of industrial/military design styling |
| 02:37.19 | jpsharp | She's pretty tolerant of me. |
| 02:38.02 | slav3_kitten | she geeky herself? |
| 02:42.22 | Maliuta | I would like to live in a converted missle silo ... only no ICBM's were ever deployed in .au |
| 02:43.05 | slav3_kitten | Maliuta, i'll marry you if you move to america a buy a missile silo |
| 02:43.58 | Maliuta | slav3_kitten: the moving to america is a deal breaker ... I might consider one closer to the ancestoral home (which is just outside of Kiev) |
| 02:44.17 | slav3_kitten | damn what's wrong with america? |
| 02:44.41 | Maliuta | it's full of americans :) |
| 02:44.59 | Maliuta | oh, and the politics really suck |
| 02:45.16 | slav3_kitten | aside from that we have great gun laws :) |
| 02:46.15 | Maliuta | yeah, but so does Nigeria ... but I'm not look to move there anytime soon either :) |
| 02:47.10 | Maliuta | And in most of the hell holes with less restrictive gun laws than .au I could collect fully auto's too |
| 02:47.24 | Maliuta | <-- slight military fetish |
| 02:47.36 | Maliuta | mostly I wear mil-spec clothing |
| 02:47.47 | slav3_kitten | <-- owns a lot of guns |
| 02:48.04 | slav3_kitten | Maliuta, i picked up a nice emergency rappel belt the other day |
| 02:48.29 | Maliuta | slav3_kitten: where abouts in the US are you? got a conceal and carry? |
| 02:48.59 | slav3_kitten | mid south, had one but it lapsed since i moved and i've not had the money to get it again |
| 02:50.03 | Maliuta | While I like the idea of more relaxed gun laws the problem is that people with guns are more inclined to use them, I prefer the ability to fight my personal wars with duct tape and things you can buy from a hardware store |
| 02:50.47 | Maliuta | Firearms should be a method of last resort |
| 02:51.24 | Maliuta | although I do like guns and would like to seriously collect (if I had the oodles of cash) |
| 02:51.52 | Maliuta | planning on getting my shooters licence, and then a pistol licence |
| 02:52.07 | Maliuta | the main issue with owning here is the cost of properly securing them |
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| 03:03.31 | WIMPy | tries to flash a new main-fs... |
| 03:04.22 | *** part/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 03:04.41 | WIMPy | Takes a little longer via serial... |
| 03:16.11 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:19.33 | WIMPy | Jay! It's booting again. |
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| 04:38.15 | slav3_kitten | once upon a time there was a guy, he built a bitching home network, decked it out with all the latest backup power supplies mounted it all in a rack an had it set to be defended against power outages |
| 04:38.41 | slav3_kitten | his ISP which he pays 60/m for 3 meg down 1.5 meg up couldn't be bothered to backup their shit |
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| 05:13.12 | din3sh | gd mrning all |
| 05:19.42 | slav3_kitten | yo din3sh |
| 05:28.25 | Maliuta | slav3_kitten: $60US/mnth for 3Mb/s down??? sounds like a ripoff to me. I pay about that for 24Mbs/2Mbs with a /29 and an IPv6 /56 |
| 05:28.56 | slav3_kitten | Maliuta, i'm taking it the ass because it's a WISP and the only game in town |
| 05:29.12 | slav3_kitten | i also need to pay them 5/m additional if i want a public IP |
| 05:29.23 | slav3_kitten | it's like triple natted an some odd load balancer setup |
| 05:37.49 | *** join/#asterisk din3sh (~din3sh@41.212.202.90) |
| 05:46.31 | volga629 | Maliuta which ISP ? |
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| 06:06.21 | trulsk | move to norway and ill sell you best effort vdsl2 with a static ip and native ipv6 for that prize :) |
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| 06:16.25 | slav3_kitten | huzzah i have a phone registered! |
| 06:20.38 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:20.46 | v0lZy | lo |
| 06:33.21 | *** join/#asterisk atekinalp (~alper@85.105.28.77) |
| 06:34.01 | atekinalp | hi how can i implement one button call hold/call parking? |
| 06:38.00 | v0lZy | what is one button call hold? |
| 06:41.30 | atekinalp | v0lZy: say i am talking with someone then i press a button like * or ** and callee is taking to hold and listen music then i again press same button/s and continue talking |
| 06:42.12 | v0lZy | u dont have a hold button on your phone? |
| 06:42.32 | resist0r | re: features |
| 06:44.03 | atekinalp | i use a softphone and no i couldnt seen any options like that |
| 06:44.50 | v0lZy | which softphone? 3cx is a good one, has hold button |
| 06:45.16 | atekinalp | microsip |
| 06:45.52 | v0lZy | never heard of it |
| 06:46.07 | v0lZy | but what you are mentioning should be doable i think |
| 06:46.25 | v0lZy | what you could do is |
| 06:46.44 | v0lZy | have an extension for call parking |
| 06:47.36 | v0lZy | ParkedCall(extension) picks up a parked call |
| 06:48.48 | v0lZy | I guess all you need to have is an extension defined for parking a call |
| 06:48.50 | v0lZy | like |
| 06:48.58 | v0lZy | *444 or something |
| 06:49.00 | v0lZy | and u type |
| 06:49.04 | v0lZy | *444 to park the call |
| 06:49.18 | v0lZy | and *444 to pick it up if there is a call parked |
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| 07:10.43 | atekinalp | v0lZy:thanks i will try |
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| 08:29.02 | zamba | is there a way to force all my sip clients to move to another server? |
| 08:44.14 | ChannelZ | other than reconfiguring them? |
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| 08:48.48 | zamba | well.. i've updated the DNS records |
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| 09:08.58 | davlefou | Hi, |
| 09:09.21 | davlefou | i would like get call number in my web site, how can i do? |
| 09:09.50 | bulkorok | hi... is there anybody who uses the (free) res_fax module with spandsp in a productive environment? |
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| 09:38.26 | v0lZy | hi guys |
| 09:38.33 | v0lZy | how would i check if a number is internal or not? |
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| 10:10.16 | kaldemar | v0lZy: "internal" does not mean anything to asterisk, that's something you need to define yourself in your dialplan structure. |
| 10:10.37 | v0lZy | yeah |
| 10:10.40 | v0lZy | i have it defined |
| 10:10.46 | v0lZy | [internal] |
| 10:10.48 | v0lZy | exten => etc |
| 10:10.54 | v0lZy | i just dont know how to check that |
| 10:11.04 | kaldemar | then use func DIALPLAN_EXISTS |
| 10:11.20 | v0lZy | let me read up on it |
| 10:11.58 | v0lZy | doesnt appear to be in themanual |
| 10:12.01 | v0lZy | core show functions? |
| 10:12.30 | kaldemar | ${IF($[${DIALPLAN_EXISTS(context,exten,priority)}]?true:false)} |
| 10:12.39 | kaldemar | core show function DIALPLAN_EXISTS |
| 10:13.11 | v0lZy | nope |
| 10:13.12 | v0lZy | dont have it |
| 10:13.47 | kaldemar | why do you need such a check? |
| 10:13.52 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:13.59 | v0lZy | sNo function by that name registered. |
| 10:14.00 | v0lZy | Command 'core show function DIALPLAN_EXISTS' failed. |
| 10:14.07 | v0lZy | well, its my call forwarding situation |
| 10:14.54 | v0lZy | I have a draytek phone and enabling administratior password locks the call forwarding menu |
| 10:15.09 | v0lZy | which makes sense because u dont want users to forward calls to their mobile phones and have the company pay for that. |
| 10:15.12 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:15.30 | v0lZy | now, i have a headset button but noone uses headsets here |
| 10:15.38 | v0lZy | so im putting a sticker over it saying 'call forward' |
| 10:16.06 | v0lZy | and basically, what it is is that it redirects the sound in and sound-out to a nonexistant headset... so its a dead line, right |
| 10:16.20 | v0lZy | now what i want to do is that when a user presses that and dials a **44 |
| 10:16.41 | v0lZy | asterisk redirects all calls coming to ${CALLERID(num)} to ${EXTEN:2} |
| 10:17.01 | v0lZy | but i dont want to do that if the number is not in the asterisk dialplan, so that users cant divert outside my pbx. |
| 10:17.22 | v0lZy | in order to do that, id like to check if the extension exists or not |
| 10:20.25 | v0lZy | i have such a check for voicemail |
| 10:20.31 | v0lZy | curtesy of you guys here |
| 10:20.32 | v0lZy | n,GotoIf($["${ARG1}" = "${CUT(AST_CONFIG(voicemail.conf,default,${CALLERID(num)}),",",1)}"]?voicemail:hangup) |
| 10:20.46 | v0lZy | i was thinking of making this |
| 10:21.58 | v0lZy | ${CUT(AST_CONFIG(extensions.conf,internal,${EXTEN:2},thisiswhereimstuck,1))} |
| 10:22.09 | v0lZy | because the input there is |
| 10:22.20 | kaldemar | AST_CONFIG is really not usable with extensions.conf. |
| 10:22.21 | v0lZy | exten => number,... |
| 10:23.19 | kaldemar | use func SHELL to do something outside dialplan. |
| 10:23.29 | kaldemar | or get DIALPLAN_EXISTS... |
| 10:24.30 | kaldemar | you're using some pre 1.6.0 version, right? |
| 10:26.05 | v0lZy | Asterisk 1.8.4.4 built by root @ on a i486 running Linux on 2012-04-25 16:59:15 UTC |
| 10:26.33 | kaldemar | compile func_dialplan |
| 10:26.56 | kaldemar | or load it if it's just unloaded. your version has the module. |
| 10:27.30 | v0lZy | sec |
| 10:27.32 | v0lZy | modules.conf |
| 10:28.57 | v0lZy | how do i see available modules? |
| 10:30.39 | v0lZy | what directory are the modules in? |
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| 10:33.47 | kaldemar | v0lZy: /usr/lib/asterisk/modules |
| 10:35.17 | v0lZy | func_dialplan.so ? |
| 10:39.31 | v0lZy | ah, got it |
| 10:39.32 | v0lZy | ok |
| 10:39.39 | v0lZy | DIALPLAN_EXISTS.. sec |
| 10:40.32 | v0lZy | ok so |
| 10:41.19 | v0lZy | ${IF($[${DIALPLAN_EXISTS(internal,$EXTEN:2)}]?true:false)} |
| 10:42.02 | v0lZy | ${IF($[${DIALPLAN_EXISTS(internal,$EXTEN:2)}]?Goto(wherever):Hangup())} |
| 10:42.05 | v0lZy | like this? |
| 10:42.12 | v0lZy | and then |
| 10:42.17 | v0lZy | n(whatever),blahlbha |
| 10:42.19 | v0lZy | ? |
| 10:45.50 | v0lZy | n,${IF($[${DIALPLAN_EXISTS(internal,${EXTEN:2})}]?Goto(waitloop):Hangup())} |
| 10:46.01 | v0lZy | missing commas/parenthesis something |
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| 10:54.30 | v0lZy | n,SayDigits(${DIALPLAN_EXISTS(internal,${EXTEN:2})}) works up to here |
| 10:54.36 | v0lZy | so |
| 10:54.48 | v0lZy | ${DIALPLAN_EXISTS(internal,${EXTEN:2})} part is ok |
| 10:55.41 | kaldemar | n,${IF($[${DIALPLAN_EXISTS(internal,${EXTEN:2})}]?Goto(waitloop):Hangup())} is missing an application entirely. |
| 10:55.44 | v0lZy | ${IF($[ ${DIALPLAN_EXISTS(internal,${EXTEN:2})} ]?Goto(wherever):Hangup())} part doesnt work though. |
| 10:56.03 | kaldemar | sure it does, but you're not using it right. |
| 10:56.14 | v0lZy | what application should i put that in? |
| 10:56.24 | v0lZy | eval? |
| 10:56.38 | v0lZy | wait.. thats another function.. |
| 10:56.53 | kaldemar | you're most likely thinking of app Exec |
| 10:56.54 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:57.16 | v0lZy | yeah |
| 10:57.28 | kaldemar | i'd personally use GotoIf($[${DIALPLAN_EXISTS(internal,${EXTEN:2})}]?waitloop:hangup) |
| 10:57.28 | *** part/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:57.48 | v0lZy | hm.. |
| 10:57.49 | kaldemar | and have priorities n(waitloop) and n(hangup) |
| 10:57.59 | v0lZy | but wouldnt hangup be a priority in there then? |
| 10:58.05 | v0lZy | ah, ok, yes, that answers my question |
| 10:58.07 | v0lZy | ty |
| 10:58.09 | v0lZy | but |
| 10:58.19 | v0lZy | theres another thing |
| 10:58.23 | v0lZy | i have to preform something once this is hungup |
| 11:00.26 | v0lZy | ok, one part is working now |
| 11:00.36 | v0lZy | how do i preform something after i hangup? |
| 11:00.48 | v0lZy | h,n,dosomething() ? |
| 11:01.11 | v0lZy | what i need to do is destroy a redirect |
| 11:01.27 | v0lZy | i have a n,Set(DB(CFI/${CALLERID(NUM)})=${EXTEN:2}) |
| 11:01.28 | kaldemar | is the hangup done by on-hook? |
| 11:02.00 | v0lZy | what do you mean by 'done by on-hook'? |
| 11:02.06 | v0lZy | (sorry, not familiar with this language |
| 11:02.07 | v0lZy | ) |
| 11:02.38 | kaldemar | is the hangup goto(hangup) or the user putting the phone on-hook, i.e. putting the handset back to its place? |
| 11:04.33 | v0lZy | user putting the phone on hook |
| 11:06.57 | davlefou | v0lZy: why you don't use the pastebin? |
| 11:07.13 | v0lZy | thats ok |
| 11:07.21 | v0lZy | i've got it up to the on-hook hangup thing |
| 11:07.36 | v0lZy | all i need to do is unset the redirec twhen the user hangsup |
| 11:07.46 | v0lZy | basically this DBdel(CFI/${CALLERID(num)}) |
| 11:07.54 | v0lZy | once the user puts the phoen back on hook |
| 11:09.13 | v0lZy | someone said i shoudl do it like |
| 11:09.19 | v0lZy | h,1,DBdel(CFI/${CALLERID(num)}) |
| 11:09.26 | kaldemar | use a hangup extension. DBDel is deprecated, use func DB_DELETE. |
| 11:10.15 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 11:10.23 | kaldemar | h,1,Noop(Removed ${DB_DELETE(CFI/${CALLERID(num)})}) |
| 11:10.56 | v0lZy | i put this into the same context? |
| 11:11.47 | v0lZy | can i use a macro or something? |
| 11:11.56 | v0lZy | because the web interface doesnt allow me to do the h,1, ... |
| 11:12.35 | kaldemar | i'm not taking guesses on what some web interface lets you do. |
| 11:15.32 | v0lZy | no i mean |
| 11:15.57 | v0lZy | is there a way to define this within the 's', extension |
| 11:17.45 | v0lZy | i guess i can live with h,1,Noop(Removed ${DB_DELETE(CFI/${CALLERID(num)})}) |
| 11:17.49 | v0lZy | can i put it in the same context? |
| 11:20.16 | v0lZy | btw this also deprecated? Set(DB(CFI/${CALLERID(NUM)})=${EXTEN:2}) |
| 11:21.48 | kaldemar | v0lZy: you can do whatever you want in the s extension, but the call must end up in it, and that won't happen automatically as with h. |
| 11:22.32 | *** join/#asterisk hwt (hwt@curb.thorkildssen.com) |
| 11:23.08 | kaldemar | where you put it depends on the rest of your dialplan and what is the "current" context for the call. the current context is the one where the call first landed or some other the call went to with a goto. |
| 11:23.16 | kaldemar | DB() is not deprecated. |
| 11:23.39 | v0lZy | kaldemar: do you perhaps have teamviewer? |
| 11:23.43 | kaldemar | no |
| 11:23.48 | hwt | can anyone explain to me why both inbound and outbound sip proxies should be defined as "peer" on asterisk 1.8+? |
| 11:23.59 | v0lZy | shux.. hm. |
| 11:24.01 | kaldemar | v0lZy: use pastebin if you want to show something. |
| 11:24.15 | v0lZy | kaldemar: I cant exactly paste it coherently ... |
| 11:24.17 | v0lZy | but ok |
| 11:24.19 | v0lZy | one moment |
| 11:26.44 | v0lZy | if you do ctrl+f on the web page and search for 'Redirect a call' |
| 11:26.49 | v0lZy | you'll get to the part im talking about |
| 11:28.00 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 11:28.02 | schmidts | Hi folks |
| 11:28.35 | v0lZy | kaldemar: context name is [DIALPLAN-APPLICATION-14958000124fcc572a2d711] |
| 11:33.25 | *** join/#asterisk tuxx- (tuxx@pantoff0l.nl) |
| 11:37.51 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:38.47 | kaldemar | v0lZy: what page? give a link. |
| 11:38.57 | v0lZy | i noticed you the link |
| 11:39.42 | *** join/#asterisk CommaCrazy (1000@nikola.office.bvcom.net) |
| 11:43.00 | kaldemar | eww |
| 11:43.43 | v0lZy | yeah |
| 11:43.54 | v0lZy | its a bit messy since its handled by a frontend |
| 11:45.30 | kaldemar | put the h exten in the same context. |
| 11:45.46 | v0lZy | k |
| 11:46.16 | v0lZy | is this line ok btw?: Set(DB(CFI/${CALLERID(NUM)})=${EXTEN:2}) |
| 11:47.03 | kaldemar | depends on what EXTEN has in it. |
| 11:47.22 | v0lZy | **44 |
| 11:47.29 | v0lZy | it supposed to redirect to 44 |
| 11:47.32 | v0lZy | though im not sure about CFI |
| 11:47.35 | v0lZy | and CF... |
| 11:48.01 | kaldemar | CFI or CF is just a family name in the DB, it doesn't mean anything to asterisk. |
| 11:49.37 | v0lZy | it just made a difference :D |
| 11:50.03 | Digweed | I'm having a problem with Asterisk 1.4, "Address family not supported by protocol". I get this when using the outboundproxy realtime option. I think it's not supported by Asterisk 1.4 or at least has bugs. Is there a way to get this working without upgrading to Asterisk 1.6? I have this problem on 20 media gateways and upgrading is currently not an option. |
| 11:51.00 | schmidts | Digweed yes uprade to a version which is not outdated like 1.8. 1.6 is also allready dead like 1.4 |
| 11:52.21 | Digweed | We are replacing our old OpenSER with a new OpenSIPS, this will give us loadbalancing functionality but the Asterisk 1.4 media gateway do not like realtime outboundproxy switching. |
| 11:52.30 | Digweed | I was hoping there is a quick, dirty, fix. |
| 11:54.02 | v0lZy | kaldemar: the h,1,Noop(Removed ${DB_DELETE(CFI/${CALLERID(num)})}) doesnt work |
| 12:02.27 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 12:04.04 | v0lZy | also found this |
| 12:04.05 | v0lZy | macro hangup-calls() { Verbose( - - - This is how you hangup call in a macro when using AEL - - - ); catch h { ResetCDR(); Hangup(); };}; |
| 12:04.39 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-219.bois.qwest.net) |
| 12:08.02 | kaldemar | v0lZy: how does it not work? |
| 12:08.16 | v0lZy | it doesnt kill the redirect |
| 12:08.24 | kaldemar | does the extension not execute? |
| 12:08.26 | v0lZy | i don think it runs |
| 12:08.37 | kaldemar | look at CLI |
| 12:10.23 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:11.18 | v0lZy | -- Executing [h@DIALPLAN-APPLICATION-14958000124fcc572a2d711:1] Answer("SIP/16-00000076", "") in new stack |
| 12:11.20 | v0lZy | <PROTECTED> |
| 12:11.56 | v0lZy | i guess it did run |
| 12:12.00 | v0lZy | but non-zero exit |
| 12:12.06 | v0lZy | that sounds like an error |
| 12:13.06 | v0lZy | exten => h,1,NoOp(${DB_DELETE(CF/${CALLERID(num)})}) |
| 12:13.07 | v0lZy | btw |
| 12:13.15 | v0lZy | does this h,1 run if the callee drops the call? |
| 12:14.02 | [TK]D-Fender | should if either |
| 12:14.19 | v0lZy | uh... |
| 12:14.26 | v0lZy | thats not gonna work for my purposes then i suppose |
| 12:14.28 | v0lZy | http://pastebin.com/QMwf5mKc |
| 12:14.29 | [TK]D-Fender | Answering a call in "h" is counterproductive |
| 12:14.30 | v0lZy | this is what i have |
| 12:15.01 | [TK]D-Fender | that is a terrible pattern... I believe I already went through all of this with you. |
| 12:15.22 | v0lZy | [TK]D-Fender: what i need is that I dial into an application... and as long as im dialing to teh application, all incoming calls to my number are redirected to another number.. when i drop the line to my application, i want the redirection to drop. |
| 12:15.36 | v0lZy | [TK]D-Fender: yeah, but i didnt finish it on last friday |
| 12:15.42 | [TK]D-Fender | voiYes, and I told you how to do this |
| 12:15.45 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:15.49 | v0lZy | I got it to work... up to the point where i call **XX |
| 12:15.57 | [TK]D-Fender | Get rid of that pattern |
| 12:16.11 | [TK]D-Fender | Actually I even GAVE you a complete code sample |
| 12:16.27 | v0lZy | [TK]D-Fender: I cant get rid of that _[0-9] stuff.. its how the interface works |
| 12:16.36 | v0lZy | (perhaps i should show the entire file...) |
| 12:16.37 | [TK]D-Fender | Get out of that context. |
| 12:16.48 | kaldemar | v0lZy: h@DIALPLAN-APPLICATION-14958000124fcc572a2d711:1] Answer("SIP/16-00000076", "") is clearly not executing NoOp(${DB_DELETE(CF/${CALLERID(num)})}) |
| 12:16.51 | [TK]D-Fender | Which I also showed you |
| 12:16.56 | kaldemar | v0lZy: fix your dialplan. |
| 12:17.11 | [TK]D-Fender | _[0-9a-zA-Z*#]!,1,Answer() <---- because THIS also matches |
| 12:17.26 | [TK]D-Fender | v0lZy, like I said that pattern is stupidly dangerous |
| 12:17.31 | [TK]D-Fender | v0lZy, Get OUT of that context |
| 12:17.39 | v0lZy | so... |
| 12:17.40 | v0lZy | gotosub? |
| 12:17.47 | [TK]D-Fender | v0lZy, GOTO |
| 12:18.02 | v0lZy | ok then from my link |
| 12:18.19 | v0lZy | i just move everything below the if to a new context? |
| 12:18.49 | v0lZy | and i need to fix the hangup and createredirect stuff i guess |
| 12:20.58 | *** join/#asterisk DaneoShiga (~dshiga@kraz.dreamhost.com) |
| 12:22.14 | v0lZy | [TK]D-Fender: http://pastebin.com/dMwb3RpL |
| 12:22.20 | v0lZy | this is the entire extensions.conf file |
| 12:23.52 | v0lZy | In sip.conf each phone gets its own context. |
| 12:24.26 | v0lZy | as does each provider. |
| 12:25.35 | [TK]D-Fender | ... |
| 12:25.47 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:25.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:26.12 | v0lZy | each phone in the context includes the 'internal' context |
| 12:26.15 | DaneoShiga | hi, i'm learning about dialplan, i got into a dbout, would have better performance to use "same =>" instead of the prefix pattern? or just avoid typo errors? |
| 12:26.26 | [TK]D-Fender | what part of "make the first step of this dialplan application get the hell out of that context and away for the dangerous pattern match" am I not being entirely clear about? |
| 12:26.42 | v0lZy | and in the internal context all definitions are held and a main macro is executed at the end of each one or something |
| 12:27.03 | v0lZy | [TK]D-Fender: I understand what you are saying, but wont that break everything? |
| 12:28.02 | WIMPy | v0lZy: Maybe you should get rid of whatever webinterface you've got there. They are usually at least a real PITA if you want to do something yourself, if not a show stopper. |
| 12:28.15 | [TK]D-Fender | v0lZy, this is your APPLICATION. That is the only point of that context |
| 12:29.18 | v0lZy | ; application : Redirect a call |
| 12:29.19 | v0lZy | exten => _**XX,1,NoOp(internal calling application: Redirect a call) |
| 12:29.21 | v0lZy | exten => _**XX,n,Set(CDR(InternalCalleridNum)=_**XX) |
| 12:29.23 | v0lZy | exten => _**XX,n,Goto(DIALPLAN-APPLICATION-14958000124fcc572a2d711,${EXTEN},1) |
| 12:29.24 | v0lZy | this is where the patteren is defined |
| 12:29.32 | v0lZy | and this is what the application matches really |
| 12:29.34 | v0lZy | **XX |
| 12:29.42 | leifmadsen | yes |
| 12:29.47 | v0lZy | but that sends it to DIALPLAN-APPLICATION-gibberish |
| 12:29.52 | leifmadsen | if you're typing it yourself, then same => is sane |
| 12:30.10 | leifmadsen | that's a context name |
| 12:30.18 | [TK]D-Fender | v0lZy, I've said it 5 times now. |
| 12:30.23 | v0lZy | in that gibberish context, apparently, it has that all matching patteren... will this effect anythign if i delete it? |
| 12:30.25 | leifmadsen | my guess, autogenerated via a web interface |
| 12:30.25 | [TK]D-Fender | v0lZy, that earlier bit does not matter. |
| 12:30.27 | v0lZy | and what should i replace it with TK? |
| 12:30.32 | leifmadsen | v0lZy: it probably will |
| 12:30.35 | leifmadsen | don't use web gui's |
| 12:30.43 | leifmadsen | if you do, don't change anything manually |
| 12:30.46 | [TK]D-Fender | v0lZy, in your other context, go save the exten in another var and then GOTO out to a saner context |
| 12:31.18 | leifmadsen | because it won't matter, since as soon as you change anything and make a change in the web interface, it'll just overwrite your changes anyways |
| 12:31.18 | v0lZy | [TK]D-Fender: ok... can i use gotoif to switch into a nother context? |
| 12:31.40 | v0lZy | leifmadsen: no, actually what [TK]D-Fender is saying makes sense |
| 12:31.45 | v0lZy | if i create a different cotnext |
| 12:31.47 | v0lZy | it will preserve it |
| 12:31.56 | leifmadsen | whatever floats your boat |
| 12:32.03 | v0lZy | it does some kind of diff magic |
| 12:32.22 | v0lZy | but im a bit stuck... once at gotoif thing |
| 12:32.34 | v0lZy | i just need some unrelated context somewhere i think |
| 12:32.55 | [TK]D-Fender | There is no IF. |
| 12:32.57 | [TK]D-Fender | You MUST |
| 12:34.23 | v0lZy | ok... |
| 12:34.28 | v0lZy | let me paste then |
| 12:36.51 | v0lZy | fuck |
| 12:36.56 | v0lZy | i think i just killed.... everything. |
| 12:37.56 | v0lZy | entire exten.conf ... ehehe |
| 12:38.09 | v0lZy | thank god for pastebin |
| 12:41.35 | *** join/#asterisk awk (~awk@trancehouse.co.za) |
| 12:41.41 | v0lZy | ; Wait for hangup (on-hook) |
| 12:41.42 | v0lZy | [WaitHangup] |
| 12:41.44 | v0lZy | exten => s,n(waitloop),Wait(10) |
| 12:41.45 | v0lZy | exten => s,n,SayDigits(${EXTEN:2}) |
| 12:41.47 | v0lZy | exten => s,n,Goto(waitloop) |
| 12:41.48 | v0lZy | exten => h,1,Noop(Removed ${DB_DELETE(CFI/${CALLERID(num)})}) |
| 12:41.59 | v0lZy | like this [TK]D-Fender ? |
| 12:42.26 | v0lZy | and where i do the pattern matching |
| 12:42.27 | v0lZy | i do this |
| 12:42.30 | v0lZy | 1,Answer() |
| 12:42.32 | v0lZy | n,GotoIf($[${DIALPLAN_EXISTS(internal,${EXTEN:2})}]?createredirect:hangup) |
| 12:42.33 | v0lZy | n(createredirect),Set(DB(CF/${CALLERID(NUM)})=${EXTEN:2}) |
| 12:42.35 | v0lZy | n,Goto(WaitHangup) |
| 12:42.36 | v0lZy | n(hangup),Hangup() |
| 12:42.46 | awk | hi guys, the codec for a blackberry is that AMR? is there a way to load it on a pre-compiled system... aka something that was compiled by yum |
| 12:42.55 | awk | or would I have to modify the yum package to include the patches? |
| 12:45.03 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 12:45.06 | WIMPy | There is AMR support from somewhere? |
| 12:45.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gcmzoxvgblkrhqrv) |
| 12:45.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:47.39 | bulkorok | hey guys... does fax for asterisk from digium uses spandsp? |
| 12:48.56 | v0lZy | hm |
| 12:48.58 | v0lZy | ok |
| 12:49.03 | v0lZy | i think i got it the way u suggested [TK]D-Fender |
| 12:49.07 | v0lZy | but its not working really. |
| 12:49.09 | leifmadsen | bulkorok: no, they are different things |
| 12:49.18 | leifmadsen | bulkorok: you either use res_fax_digium or res_fax_spandsp |
| 12:49.57 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:49.57 | *** join/#asterisk mbrit (~mbrit@186.120.97.195) |
| 12:49.59 | bulkorok | leifmadsen: ok.. because asterisk segfaults sometimes... so I thought I give fax_digium a try if it doesn't use spandsp |
| 12:50.04 | *** join/#asterisk Norrec (~anarchy@99-10-121-68.lightspeed.sndgca.sbcglobal.net) |
| 12:50.15 | v0lZy | ; Wait for hangup (on-hook) |
| 12:50.17 | v0lZy | [WaitHangup] |
| 12:50.18 | leifmadsen | bulkorok: it doesn't |
| 12:50.18 | v0lZy | exten => s,1(waitloop),Wait(10) |
| 12:50.20 | v0lZy | exten => s,n,Goto(waitloop) |
| 12:50.21 | v0lZy | exten => h,1,Noop(Removed ${DB_DELETE(CF/${CALLERID(num)})}) |
| 12:50.23 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 12:50.23 | v0lZy | but my line drops. |
| 12:51.16 | bulkorok | leifmadsen: great... right now I use freeswitch for faxing, but it's on the same machine like asterisk... i don't like this... do you know what are the costs for the licences? |
| 12:51.44 | leifmadsen | bulkorok: check digium.com for that nfo |
| 12:51.45 | *** join/#asterisk lcotte (~luis.carl@186.209.79.11) |
| 12:51.48 | bulkorok | k |
| 12:53.22 | v0lZy | [TK]D-Fender: it just disconnects right away |
| 12:54.25 | lcotte | DaneoShiga, tenso a coisa |
| 12:54.27 | lcotte | rsssss |
| 12:55.42 | DaneoShiga | lcotte: haeuahuehaeu |
| 12:55.46 | v0lZy | <PROTECTED> |
| 12:55.48 | v0lZy | <PROTECTED> |
| 12:55.49 | v0lZy | <PROTECTED> |
| 12:55.53 | v0lZy | its totally not waiting for anything :| |
| 12:56.31 | [TK]D-Fender | <v0lZy> exten => s,1(waitloop),Wait(10) <--- missing COMMA |
| 12:56.42 | [TK]D-Fender | or is it.. |
| 12:56.45 | [TK]D-Fender | or jsut not loaded. |
| 12:56.53 | [TK]D-Fender | go prove you even erloaded your dialplan for it |
| 12:57.08 | [TK]D-Fender | "dialplan show WaitHangup" |
| 12:57.25 | [TK]D-Fender | and pastebini all of this new dialplan as well for us along with that dump |
| 12:57.44 | v0lZy | askoziapbx*CLI> dialplan show WaitHangup |
| 12:57.45 | v0lZy | [ Context 'WaitHangup' created by 'pbx_config' ] |
| 12:57.47 | v0lZy | <PROTECTED> |
| 12:57.49 | v0lZy | -= 1 extension (1 priority) in 1 context. =- |
| 12:58.12 | v0lZy | it ignores everything with an s in front of it |
| 12:58.16 | [TK]D-Fender | pastebin... |
| 12:58.32 | [TK]D-Fender | and reload it first |
| 12:58.33 | v0lZy | what do u want to see? |
| 12:58.35 | v0lZy | the entire file? |
| 12:58.49 | v0lZy | ah |
| 12:58.49 | [TK]D-Fender | sure, why not |
| 12:58.50 | v0lZy | ok |
| 12:58.52 | v0lZy | <PROTECTED> |
| 12:58.53 | v0lZy | reloaded |
| 12:58.58 | [TK]D-Fender | who file, PLUS that dump... after reloading the dialplan |
| 12:59.00 | v0lZy | ok |
| 12:59.02 | v0lZy | probably works now |
| 12:59.05 | v0lZy | let me test |
| 13:01.10 | bulkorok | I opend an issue in the jira that asterisk segfaults when receiving a fax using res_fax_spandsp(19762)... can someone tell me that if I use res_fax_digium asterisk won't segfault in this way?! |
| 13:01.26 | v0lZy | finally |
| 13:01.28 | v0lZy | it works |
| 13:01.30 | v0lZy | thanks [TK]D-Fender !! |
| 13:01.49 | *** join/#asterisk atekinalp (~alper@85.105.28.77) |
| 13:02.16 | v0lZy | it works as expected now |
| 13:03.45 | wonko | ugh, what am i missing that inbound google voice calls don't work? |
| 13:11.29 | [TK]D-Fender | wonko, perhaps yo should be showing us the call attempt.... unless you meant that to be rhetorical. |
| 13:13.08 | wonko | see, that's the problem, i'm seeing NOTHING in asterisk, so i don't even know where to start debugging this |
| 13:13.34 | [TK]D-Fender | wonko, Verify if you're even authenticated first |
| 13:14.20 | wonko | i am, and i can make outbound calls just fine |
| 13:14.34 | wonko | and this google account isn't logged in anywhere else |
| 13:17.43 | lcotte | wonko, inbound dialplan its configured? |
| 13:19.14 | wonko | exten => work.phone@4amlunch.net,1,Answer |
| 13:19.14 | wonko | exten => work.phone@4amlunch.net,n,Wait(8) |
| 13:19.14 | wonko | exten => work.phone@4amlunch.net,n,SendDTMF(1) |
| 13:19.14 | wonko | exten => work.phone@4amlunch.net,n,Dial(IAX2/laptop) |
| 13:19.23 | bitwize | wonko: maybe you cannot see any output becouse of the verbose level? core set verbose XXX |
| 13:19.49 | wonko | higher is better? |
| 13:20.22 | WIMPy | set it to 3 |
| 13:20.22 | wonko | i see other calls though (like the outbound one) |
| 13:20.27 | wonko | it's already at 3 |
| 13:21.03 | lcotte | try insert the s exten for debug |
| 13:22.08 | lcotte | try 99 verbose, ;) |
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| 13:34.21 | hypknight | Hey Guys, in Asterisk 1.8.10 -- I'm trying to setup a queue_log realtime mysql db... I found a couple guides that make reference to changing a couple of files (/etc/asterisk/res_config_mysql.conf & /etc/asterisk/extconfig.conf) that my installation doesn't possess... Ayone have any tips to getting queue_log realtime setup for a remote database in 1.8.10? |
| 13:34.58 | *** join/#asterisk serafie (~erin@nat/digium/x-efwkvdidhpstyhmf) |
| 13:35.34 | [TK]D-Fender | hypknight, Specify the host... |
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| 13:36.20 | hypknight | right, but the point is I don't even see those files -- so where would I alter to even turn ON realtime queue_log? |
| 13:37.00 | hypknight | ie. queue_log => mysql,asterisk is supposed to go into extconfig.conf -- that I don't have |
| 13:38.21 | [TK]D-Fender | So go get it |
| 13:38.42 | [TK]D-Fender | This is also quite likely because of a lack of dependencies |
| 13:38.50 | [TK]D-Fender | So it never installed |
| 13:38.55 | [TK]D-Fender | I'd check that first |
| 13:39.47 | hypknight | is there something somewhere that tells the dependencies for just the queue_log realtime module? I wasn't able to find anything like that -- so I assumed it was just built in.. |
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| 13:41.15 | [TK]D-Fender | How did you install it in the first place? |
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| 13:42.29 | hypknight | [TK]D-Fender: check #freepbx |
| 13:42.33 | [TK]D-Fender | already there |
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| 13:44.07 | bulkorok | hypknight: http://etel.wiki.oreilly.com/wiki/index.php/Realtime_Queue_and_Agent_Management |
| 13:44.22 | hypknight | cute -- |
| 13:44.34 | bulkorok | works great :) |
| 13:44.53 | bulkorok | ah wait... u wanted queue_log... |
| 13:45.00 | bulkorok | but the dependencies are the same |
| 13:45.03 | hypknight | the only dependencies listed there are mysql |
| 13:45.05 | hypknight | ... |
| 13:45.08 | [TK]D-Fender | Yeah, it helps when the solution fits the request.... |
| 13:45.08 | hypknight | I have mysql installed... |
| 13:45.17 | bulkorok | libmysqlclient15off |
| 13:45.28 | bulkorok | try to install the devs too |
| 13:45.29 | hypknight | which is installed |
| 13:45.40 | bulkorok | libmysqlclient-dev |
| 13:45.47 | hypknight | it's all installed |
| 13:45.59 | hypknight | the config files it references aren't there... that's the issue |
| 13:46.13 | bulkorok | did you choose it in make menuselect? |
| 13:46.16 | wonko | lcotte: that didn't help either. :) |
| 13:47.31 | bulkorok | hypknight: I use this with odbc |
| 13:47.59 | hypknight | bulkorok: use what? queue_log realtime? |
| 13:48.25 | bulkorok | yes |
| 13:48.47 | bulkorok | wait... I had another link |
| 13:49.02 | hypknight | yea that'd be fine... the mechanism I use to connect to mysql i'm not too fussed about... just put out that the files aren't there... |
| 13:52.24 | bulkorok | first: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html |
| 13:53.48 | bulkorok | second: http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc for setting up odbc for asterisk... |
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| 13:54.08 | bulkorok | but instead of using cdr_odbc.conf check res_odbc.conf |
| 13:54.32 | bulkorok | take a look at the sample config files in the sources for explanation |
| 13:55.55 | bulkorok | and at least... for storing the queue_log in mysql use the ~ast_src/contrib/realtime/mysql/queue_log.sql to create the table and enable it in the extconfig.conf |
| 13:56.19 | bulkorok | there must be sth like queue_log => odbc,asterisk,queue_log |
| 13:56.43 | slav3_kitten | yawns widely |
| 13:56.46 | hypknight | bulkorok: extconfig.conf doesn't exist in my installation -- that's the whole problem. |
| 13:56.55 | bulkorok | well... |
| 13:57.12 | bulkorok | look at the samples in the sources |
| 13:57.38 | bulkorok | or issue make samples BUT THE CURRENT CONFIG WILL BE DELETETD! |
| 13:58.03 | bulkorok | sry for screaming... just wanted to be sure you understand! |
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| 13:58.29 | bulkorok | ah I forgot this is working with 1.8 |
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| 14:10.51 | din3sh | hey all |
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| 14:28.30 | lcotte | wonko, sorry, i back now |
| 14:29.28 | lcotte | wonko, you get more errors and information activating the full log |
| 14:29.44 | lcotte | in /etc/asterisk/logger.conf |
| 14:32.18 | wonko | i don't have one of those so i'll have to create it. :) |
| 14:32.42 | wonko | but i don't see why calls outbound (and internally to other extension) would show up in the console but not inbound calls from google |
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| 15:04.23 | tix | why do sip phones have "extensions" on them? |
| 15:04.30 | tix | sorry, lines |
| 15:04.43 | tix | i see like "5-line IP phone" and shit advertised |
| 15:05.05 | tix | what does that mean when you have trunks rather than analog phone lines? |
| 15:05.11 | tix | how can a sip phone have lines at all? |
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| 15:06.49 | [TK]D-Fender | tix, typically it represents the number of unique registrations it can have |
| 15:06.50 | singler | tix: ifor example it may be able to register to more than one server |
| 15:07.19 | [TK]D-Fender | tix, on some it ties to the number of simultaneous calls it can handle, sometimes as divvied up by the number of registrations it supports |
| 15:07.31 | tix | ah, okay |
| 15:07.45 | slav3_kitten | i thought it as like a traditional phone where it allows you to put people on hold an transfer calls to the phone etc |
| 15:07.58 | tix | but for a small office "single line" sip phones should be fine, right? |
| 15:08.07 | tix | you can still transfer calls to other extensions with those, right? |
| 15:08.16 | tix | because that's handled by the PBX, not the phone itself |
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| 15:08.51 | tix | i'm just confused by the terminology, i guess |
| 15:08.56 | tix | because they're not really lines |
| 15:08.58 | [TK]D-Fender | Again a single line phone may not imply how many CALLS it can support on it |
| 15:09.26 | [TK]D-Fender | An ancient Polycom IP301 is listed as a 2-line phone. You can juggle TEN calls on it however |
| 15:09.58 | edge | in the chan_dahdi configuration, is the setting "usedistinctiveringdetection=yes" for when the phone company puts two phone numbers on the same line? (like a regular # and then a fax #) ? |
| 15:10.18 | tix | also, for a small office of 6 employees (where some will have phones at home) do you think phonebooth (hosted) or sipstation (self-hosted) is better? |
| 15:10.24 | tix | that's the decision i have to make |
| 15:10.48 | tix | i think paying per-trunk rather than per-extension is cheaper when we scale up, idk |
| 15:13.40 | [TK]D-Fender | I think IN-HOUSE is better |
| 15:14.21 | tix | yeah, i think so too |
| 15:14.29 | tix | but then again we're going to have employees with homes at phone too |
| 15:14.37 | tix | so perhaps a hosted PBX is better |
| 15:14.53 | tix | since only 50% of the phones in the system will be in the office where the PBX would be located |
| 15:14.56 | tix | idk |
| 15:16.38 | tix | so i'll go with 2-line sip phones (even though we're going to have 4 trunks) |
| 15:16.42 | tix | shouldn't be a problem, right? |
| 15:16.59 | [TK]D-Fender | trunk is a very poor term... |
| 15:17.17 | [TK]D-Fender | Ans what the phone handles may have little to do with whatever PSTN connectivity your PBX has |
| 15:17.37 | [TK]D-Fender | You could have 100 DID's, but each phone may only require a sigle identity on thye server |
| 15:18.16 | tix | we're going to use sipstation for trunks |
| 15:19.03 | tix | with just a single DID |
| 15:19.04 | tix | and 4 trunks |
| 15:19.11 | tix | and roughly 6-8 phones in the office |
| 15:19.18 | tix | so 2-line sip phones should be fine, right? |
| 15:19.24 | [TK]D-Fender | 4 CHANNELS you mean then... |
| 15:19.35 | tix | they call them trunks, but yeah |
| 15:19.37 | [TK]D-Fender | What models are you looking at? |
| 15:19.45 | tix | polycom ip 335 |
| 15:19.49 | tix | looks good |
| 15:19.56 | [TK]D-Fender | Very good phones |
| 15:20.20 | [TK]D-Fender | What kind of call voluem are you expecting? |
| 15:20.21 | tix | sipstation calls them trunks, but they're actually channels right? |
| 15:20.28 | [TK]D-Fender | ~trunk |
| 15:20.28 | infobot | somebody said trunk was a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 15:20.32 | tix | probably 4 simultaneous calls max |
| 15:20.52 | [TK]D-Fender | tix, I means how many calls/day, how much time on the phone? |
| 15:21.15 | [TK]D-Fender | are we talking very infrequent use, etc |
| 15:21.37 | tix | lol bandwidth.com calls them sip trunks, and they're the largest VoIP provider. |
| 15:21.40 | tix | so.... |
| 15:21.47 | edge | Does anybody know where some documentation exists for distintive ringing? |
| 15:22.01 | tix | we'll be on the phone all day, [TK]D-Fender |
| 15:22.08 | tix | it's a sales floor |
| 15:22.14 | [TK]D-Fender | tix, they dumb things down for the masses... and they are often terribly overpriced |
| 15:22.23 | [TK]D-Fender | tix, You may want to consider headsets... |
| 15:22.34 | tix | with softphones? |
| 15:22.39 | ChannelZ | edge: that really depends on the phone, I don't think there is really any "standard" for it |
| 15:22.41 | [TK]D-Fender | edge, It's specific to every manufacturer. |
| 15:23.00 | tix | [TK]D-Fender: who do you recommend over phonebooth/sipstation/bandwidth.com? |
| 15:23.28 | [TK]D-Fender | tix, I'd need a better impression of the exact offer you have, and your expected usage. |
| 15:24.14 | tix | sipstation is $25/mo per "trunk" |
| 15:24.19 | tix | we're going to get 4 trunks |
| 15:24.26 | tix | well, 4 channels |
| 15:24.39 | tix | and our usage is going to be moderate to heavy |
| 15:25.19 | edge | [TK]D-Fender: ChannelZ : i'm thinking i might be better to call the phone company and have them disable the "two #'s on one line" and make them two seperate and that will solve my problems. |
| 15:25.47 | [TK]D-Fender | not so bad I guess |
| 15:26.15 | [TK]D-Fender | edge, So now you're referring to the analog LINE service? |
| 15:26.56 | tix | oh, sipstation doesn't have international calling |
| 15:26.57 | tix | wtf |
| 15:27.07 | tix | they're under bandwidth.com and phonebooth has intl, why doesn't sipstation? |
| 15:27.12 | tix | ugh, this changes everything |
| 15:27.28 | [TK]D-Fender | so maybe get a 2nd provider for international |
| 15:28.10 | tix | no, that's too confusing |
| 15:28.18 | tix | two trunks, two bills, two providers to deal with |
| 15:28.22 | tix | fuck that |
| 15:28.48 | tix | i think i'll just go with phonebooth.com for now |
| 15:28.57 | edge | [TK]D-Fender: What i mean is, the office we're moving into i wanted to put an asterisk system into. We have 3 phone numbers but only two physcial lines comming in. So when a fax comes in it rings differently on the second line, vs ringing normally if its a human calling the second number. Am I better off disabling that with the phone company? |
| 15:29.59 | [TK]D-Fender | Depends. What are you interfacing that line with? |
| 15:30.17 | ChannelZ | DAHDI can do distinctive ring detection but I'm not sure how it actually manifests |
| 15:30.30 | slav3_kitten | tix, i've been looking into... didforsale... they are like 20 channels iirc |
| 15:31.07 | edge | [TK]D-Fender: I'd like to have it go to a DAHDI card we have and then to the fax. If it double rings it goes to fax, single ring call queue. |
| 15:31.11 | slav3_kitten | an like .4 cents/min incoming for 8k then .875 c/m incoming and outbound is between .4 cents an .875 cents/min |
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| 15:31.26 | ChannelZ | I guess you have to setup all these "dring" patterns to match the ring |
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| 15:32.54 | [TK]D-Fender | edge, So you don't have your hardware yet? |
| 15:36.01 | edge | [TK]D-Fender: I have my DADHI hardware and server and everything is installed and i configured it , but then realized that we have this special ringing condiction |
| 15:36.07 | edge | [TK]D-Fender: kind of figured it out at the last minute.. |
| 15:36.39 | [TK]D-Fender | edge, ChannelZ pointed out that it is the "dring" settings you'll wnt to take a look at |
| 15:38.08 | slav3_kitten | an silence from tix |
| 15:39.03 | edge | [TK]D-Fender: ChannelZ Thanks guys. I'll give this a wirl, if I fail, i'll just seperate the lines and never look back. |
| 15:40.34 | tix | slav3_kitten: huh? |
| 15:40.37 | tix | silence? |
| 15:41.10 | tix | i don't like the look of didforsale, i don't trust companies with sites that haven't been redesigned since 1998 |
| 15:41.26 | slav3_kitten | you got quiet, i'd check into didforsale.com i'm really looking into them for my UK and US needs |
| 15:42.42 | slav3_kitten | personally i'd rather deal with a company that has been concentrating on growth and stability, than a company that paid some guy from cali to make them a beautiful site but has crap service |
| 15:42.56 | tix | from didforsale.com's website: |
| 15:43.00 | tix | "Why to use our service ?" |
| 15:43.08 | tix | "Reliable VoIP Servcie provider" |
| 15:43.19 | tix | "No minimum commitments on voip services" |
| 15:43.41 | tix | this must have been written by a 2nd grader |
| 15:44.03 | tix | how can you trust a company with a website that has so many blatant spelling errors? |
| 15:44.06 | tix | lol |
| 15:44.12 | tix | also, the design of their site is atrocious |
| 15:44.31 | tix | i would NEVER trust these guys with phone service that is critical to our business |
| 15:44.46 | slav3_kitten | tix, first off they aren't american and their first language isn't english |
| 15:45.03 | tix | so when i call them for support, will they be speaking broken english? |
| 15:45.09 | tix | exactly my point |
| 15:45.11 | slav3_kitten | i do a lot of work with chinese, despite spelling and grammar errors they do good work |
| 15:45.26 | slav3_kitten | so i'm apt to deal with a little broken english for good service |
| 15:45.34 | slav3_kitten | but that's just me personally |
| 15:45.47 | tix | yeah, i'm not dealing with a chinese voip provider for my US based phone needs |
| 15:45.48 | slav3_kitten | you're free to do as you please i merely offered a cost effective option |
| 15:45.50 | tix | fuck that |
| 15:45.55 | tix | thanks for the suggestion though |
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| 15:55.17 | blizzow | So this weekend, calls into my asterisk box started getting fast busy and calls out were getting "all circuits busy". I called my PRI provider and they said they detected trouble, but after a few hours they started to blame my equipment saying it was returning all 0's. My equipment is a red-fone fonebridge2 and a decent dell server. I'm running * 1.8.12 and dahdi 2.5.1. I enabled pri debug span 1 and here's a pastebin of that output: http://pastebin.co |
| 15:55.17 | blizzow | How can I tell if the problem is with my PRI provider, my tdMOe box (redfone), dahdi, or asterisk? |
| 15:57.02 | blizzow | I'm pretty worried as I saw a fresh set of yellow alarms in my logs this morning about an hour ago, and another span 1 D-channel is down message. |
| 15:57.51 | slav3_kitten | blizzow, your patebin link got cut off |
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| 16:13.08 | blizzow | slav3_kitten: which one? |
| 16:13.23 | blizzow | http://pastebin.com/0951NnQD is the dahdi restart output. |
| 16:13.47 | blizzow | http://pastebin.com/Nyker4kg is some of the pri debug span output. |
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| 16:33.54 | edge | Is there a way to cause DAHDI to re-read its configuration without restarting asterisk? |
| 16:34.09 | edge | i can do it with the dial plan (dialplan reload) |
| 16:35.06 | [TK]D-Fender | module reload chan_dahdi.so |
| 16:36.11 | edge | thanks [TK]D-Fender . I'm thinking i'm getting close to this seperate ringer.. |
| 16:36.42 | slav3_kitten | awesome netsplit |
| 16:39.54 | edge | <PROTECTED> |
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| 16:43.40 | oyugik | morning all |
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| 16:46.37 | oyugik | hey guys how do you configure landline/modem to port 4, the FXO port. |
| 16:47.57 | kaldemar | oyugik: what hardware do you have? |
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| 16:52.47 | oyugik | I have a tdm400p |
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| 16:56.16 | kaldemar | oyugik: does it have an FXO module in the 4th slot? |
| 16:59.01 | oyugik | please correct me if I am wrong but the fxo modules are usually red in color right? |
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| 16:59.30 | [TK]D-Fender | yes |
| 17:00.05 | kaldemar | yes, and FXS modules are green. |
| 17:00.09 | [TK]D-Fender | oyugik, in system.conf : fxsks=4 |
| 17:00.43 | [TK]D-Fender | oyugik, in chan_dahdi.conf : signalling=fxs_ks , channel => 4 |
| 17:01.11 | sruffell | oyugik: The easiest way is "/etc/init.d/dahdi stop; modprobe wctdm; dahdi_genconf; dahdi_cfg" That will automatically configure /etc/dahdi/system.conf and there will also be a /etc/asterisk/dahdi-channels.conf" that you can include in /etc/asterisk/chan_dahdi.conf. |
| 17:01.34 | sruffell | (that assumes you have dahdi-linux, dahdi-tools, and asterisk all installed properly) |
| 17:02.30 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:02.40 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 17:04.01 | citywok | Has anybody used Asterisk in Hyper-V in a production environment with something like 100 users? What was your experience like? |
| 17:05.32 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:06.39 | oyugik | thanks guys let me try out all you suggestions and get back to you in a few |
| 17:12.16 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:15.23 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:15.33 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:17.18 | oyugik | "# channel 4, WCTDM/4/3, no module." I have this line in my system.conf |
| 17:18.37 | slav3_kitten | oyugik, exciting |
| 17:18.37 | oyugik | -- Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 4: Invalid argument (22) |
| 17:18.50 | slav3_kitten | btw i'm sorry you've got some frustrating problem |
| 17:19.18 | oyugik | that is the error I get when I start dadhi,, and ideas |
| 17:19.30 | sruffell | when you load wctdm, what is the output in dmesg? It appears that the driver is not recognizing your module? |
| 17:20.07 | oyugik | Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 4: Invalid argument (22) |
| 17:20.35 | slav3_kitten | woah that's a cool card. *reads more* |
| 17:21.48 | sruffell | oyugik, /etc/init.d/dahdi stop; dmesg -c > /dev/null; modprobe wctdm; dmesg > /tmp/dmesg.txt then link the output of /tmp/dmesg.txt so we can see what the driver is saying when it loads. |
| 17:24.13 | slav3_kitten | sruffell, that's some nice console fu |
| 17:24.34 | sruffell | :) |
| 17:26.44 | slav3_kitten | when i get back home in july i'm going to have to dick around with getting my fxo working |
| 17:27.38 | slav3_kitten | it's a cisco 2621xm nm-2v with a vic-2fxo card |
| 17:28.47 | oyugik | This is the output -- I think this is the relevant bit |
| 17:28.47 | oyugik | Module 0: Installed -- AUTO FXO (FCC mode) |
| 17:28.47 | oyugik | Module 1: Installed -- AUTO FXO (FCC mode) |
| 17:28.47 | oyugik | Module 2: Installed -- AUTO FXS/DPO |
| 17:28.47 | oyugik | Module 3: Not installed |
| 17:30.53 | sruffell | yes..so nothing in port 4 |
| 17:31.37 | sruffell | so, dahdi_genconf shouldn't have set anything up like that in /etc/dahdi/system.conf. How did you create the config file you were running when you used dahdi_cfg before? |
| 17:32.06 | oyugik | that means that the fxo module is either non existent or it is being used by something else |
| 17:32.36 | sruffell | It's non-existant or malfunctioning. It would not print Not INstalled on driver load if it was in use by something else. |
| 17:36.09 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-86-140.user.veloxzone.com.br) |
| 17:37.10 | oyugik | thanks @sruffell I now know exactly what is causing the problem. Thnks man. Cheers |
| 17:37.31 | sruffell | cool, glad to hear it. |
| 17:37.53 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-dxbizprceyxflafw) |
| 17:43.58 | slav3_kitten | oyugik, what was it? |
| 17:44.22 | *** join/#asterisk lcotte (~luis.carl@186.209.79.11) |
| 17:45.55 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-imkkaeqzfsdosqiw) |
| 17:48.03 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:51.24 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:51.24 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:51.48 | oyugik | I have 3 modules 2 FXO and 1 FXS - the 2 FX0 are in use the . The 4th are non-existent |
| 17:51.59 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.161) |
| 17:52.07 | oyugik | *is non existent * sorry |
| 17:53.20 | *** join/#asterisk jarrod (~jarrod@chaw.ipglobal.net) |
| 17:53.30 | jarrod | anyone know of a polycom macro that will enable another softkey? |
| 17:53.32 | slav3_kitten | ah, so you thought chan 3 but it was chan 0 in reality |
| 17:54.11 | slav3_kitten | i've done similar before :) |
| 17:55.08 | oyugik | you can say that |
| 17:56.29 | oyugik | remember there are various ways you can route your fxo, it doest have to be a landline modem only ...that the hint to my underlying problem cum realisation |
| 17:57.11 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:57.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:02.20 | [TK]D-Fender | Please stop using the word "modem".... |
| 18:09.11 | slav3_kitten | foreign exchange office.... |
| 18:10.03 | WIMPy | Trafficing organization? |
| 18:12.18 | pabelanger | when using the polycom 3.1.8 legacy release, do you have to explicitly default the sip.cfg defaults (meaning include the xml file) or does the phone have the defaults in memory? |
| 18:12.37 | pabelanger | From some initial testing, I think I need to include that file |
| 18:13.00 | pabelanger | which is different then in the 3.3.4 release |
| 18:14.32 | oyugik | sorry [TK]-Defender |
| 18:15.10 | pabelanger | Guess so |
| 18:16.18 | *** join/#asterisk fubada (~fubada@ool-4573155d.dyn.optonline.net) |
| 18:16.35 | fubada | hi, does anyone know of a gui call manager app? |
| 18:16.46 | fubada | soemthing like isymphony but open source? |
| 18:17.21 | fubada | aka switchboard? |
| 18:19.41 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:20.48 | *** join/#asterisk brettnem (~brett@76-216-205-249.lightspeed.austtx.sbcglobal.net) |
| 18:20.54 | brettnem | hey all |
| 18:23.32 | brettnem | wow, is the asterisk group normally this quiet these days? |
| 18:24.41 | Nugget | everyone is busy making prank phone calls. Why else do you think we wanted voip? |
| 18:25.44 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 18:26.00 | fubada | is there any alternative to 'Fop2" |
| 18:26.06 | fubada | for gui call manager |
| 18:26.15 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 18:26.29 | brettnem | Nugget: you in austin? I thought you were for some reason.. |
| 18:26.38 | Nugget | moved to houston a few years ago |
| 18:26.54 | brettnem | seriously? oh the horror! |
| 18:27.00 | Nugget | don't remind me! |
| 18:27.05 | brettnem | I'm from houston.. so I can say that :) |
| 18:27.25 | Nugget | I'm in montrose and never leave the loop, so it's actually not too bad. just expensive |
| 18:27.46 | brettnem | expensive? You mean, going out to eat? :) |
| 18:27.49 | Nugget | so it's not like I live in sugarland or anything |
| 18:28.04 | Nugget | well the rents are high too |
| 18:29.00 | Nugget | and buying a house is out of the question unless your standards are impossibly low or your job title is "Vice President of Embezzlement" at ExxonMobil or something |
| 18:29.31 | brettnem | are you sure you're talking about Texas? Sounds like california to me :) |
| 18:29.31 | Nugget | Don't mess with Texas. |
| 18:29.47 | fubada | you guys need to see housing prices in NYC metro area |
| 18:30.04 | fubada | Fairfield county CT |
| 18:30.36 | brettnem | ok, well I'll just throw this out. Got a dahdi issue. T1's won't come up green on this box. Something auto-updated, server rebooted now all is red. If I move the T1 to another server, it's green. I've update dahdi-linux/tools to latest version, no difference. I was told by someone else who worked on this that the newer versions of dahdi did this and that older versions work (wtf?). I've also tried 2.3 and 2.4 with same resu |
| 18:31.00 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 18:31.02 | brettnem | well hey, I'm happy with housing here in Austin. it's a good place to be |
| 18:31.29 | brettnem | I just drove thru houston yesterday.. so much freeway.. as far as the eye can see |
| 18:32.22 | fubada | keyword is FREEway :) |
| 18:32.32 | fubada | how do 9$ tolls sound? |
| 18:32.57 | brettnem | sounds like a bid for public transportation |
| 18:33.05 | brettnem | do you get a lotto ticket with that> |
| 18:33.06 | brettnem | ? |
| 18:33.30 | fubada | you get to drive into a pothole the size of your car :) |
| 18:34.13 | brettnem | so any thoughts on my dahdi issues? :D |
| 18:35.05 | fubada | no sorry, i dont know/use dahdi |
| 18:35.12 | fubada | any experience with FOP2? |
| 18:35.54 | *** join/#asterisk edge (~edge@74-84-85-171.client.mchsi.com) |
| 18:36.48 | brettnem | Yeah, I've used FOP2 |
| 18:37.00 | brettnem | It's pretty good for what it is |
| 18:37.11 | fubada | any alternatives? |
| 18:37.17 | brettnem | fop1 ? ha |
| 18:37.21 | fubada | ehe |
| 18:37.28 | brettnem | I don't know of any real good alternatives |
| 18:37.29 | fubada | iSymphony? |
| 18:37.35 | fubada | ok thanks |
| 18:37.45 | brettnem | there was some reason I didn't like iSymphony.. but it's been a real long time since i looked |
| 18:38.10 | brettnem | HUD used to be a contender, but.. well.. |
| 18:38.10 | fubada | appreciate the info |
| 18:38.32 | edge | I setup in the chan_dahdi to listen for a specific ring pattern and route it to an extension within my dialplan, but i'm wondering what i should put inside that extension if i want asterisk to do nothing. I don't want it to pick it up, i just want it to leave that ring pattern alone (fax) |
| 18:38.43 | brettnem | maybe it was the 19.95/mo subscription?! sheesh |
| 18:39.13 | brettnem | edge: is it an analog line? |
| 18:39.14 | [TK]D-Fender | edge, Nothing at all.. or just hangup... |
| 18:39.26 | edge | brettnem: yes analog. |
| 18:39.37 | edge | [TK]D-Fender: just put Hangup() in there? |
| 18:40.00 | brettnem | You could probably put a Wait() in there.. Ringing might answer an analog line, I'm not entirely sure |
| 18:40.41 | edge | brettnem: This is a fax line, so the fax machine will be picking up after a few ring (providing asterisk didn't think it was ringing for it and picked it up already) |
| 18:40.50 | [TK]D-Fender | edge, Or nothing |
| 18:42.15 | edge | [TK]D-Fender: i don't mind it doing nothing as long as it is a defined behavior and not undefined. |
| 18:44.49 | edge | also should the ring pattern change wildly? |
| 18:48.22 | *** join/#asterisk edge (~edge@74-84-85-171.client.mchsi.com) |
| 18:48.32 | edge | sorry about that, lag pushed me off |
| 18:48.56 | edge | Should the ring pattern move wildly ? |
| 18:51.32 | *** part/#asterisk oyugik (~oyugik@41.84.143.52) |
| 18:52.50 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 18:54.02 | brettnem | bah, really looks like this T1 is just down.. but somehow it's not.. |
| 18:54.38 | *** join/#asterisk sarsaeol (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 19:01.06 | jpsharp | Down enough to not pass traffic, but not down enough to go into red alarm? |
| 19:01.35 | WIMPy | So what does it (not) do? |
| 19:02.02 | *** join/#asterisk showy (~showy@186.188.21.15) |
| 19:03.04 | showy | hi, i've compiled asterisk 10 and none of my functions are registered , how can i solve this ? |
| 19:03.25 | jpsharp | Did you install config files? |
| 19:03.30 | showy | yes |
| 19:03.57 | showy | asterisk.conf also has [directories] pointing out to the mod dir, and the module.conf has autoload=yes |
| 19:04.43 | jpsharp | Try loading a module by hand and see what it complains about. |
| 19:04.47 | showy | starting asterisk i can see functions being loaded, but at use, i got error funct not register' |
| 19:06.11 | *** join/#asterisk navaismo (~navaismo@201.123.49.213) |
| 19:06.13 | showy | sorry my bad, putting an space before the function name -.- |
| 19:09.56 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:18.08 | brettnem | jpsharp: it's in red alarm. Stays in red alarm. If I plug it into another (working) server, it goes green |
| 19:18.20 | navaismo | the fax for asterisk works fine using a "Sip trunk"? |
| 19:18.43 | brettnem | dmesg actually reports a yellow |
| 19:19.13 | WIMPy | brettnem: Same cable? |
| 19:19.17 | brettnem | yeah |
| 19:19.18 | WIMPy | Do you have a loop plug? |
| 19:19.31 | brettnem | WIMPy: I'm not on site, I'm trying to get someone to test that |
| 19:20.08 | WIMPy | So one server says yellow, another green, and nothing else changes? |
| 19:22.18 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 19:34.38 | jpsharp | If you plug the T1 into another server, the T1 goes green? Or if you plug the Asterisk box into another server it goes green? |
| 19:36.55 | jpsharp | navaismo: I use the SpanDSP fax application on Asterisk over a SIP provider. |
| 19:37.14 | jpsharp | I never could get the Digium fax application to work with it. |
| 19:39.08 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 19:45.08 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 19:46.56 | navaismo | thxs jpsharp |
| 19:48.22 | jarrod | what is the variable for custom idle display text on polycom? |
| 19:50.51 | gusto | btw. what is jitter? |
| 19:50.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:51.03 | jpsharp | Jitter is the change in latency over time. |
| 19:51.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:51.35 | jpsharp | Causes packets to show up delayed more than expected or out of order. |
| 19:51.57 | gusto | http://en.wikipedia.org/wiki/Packet_delay_variation ? |
| 19:52.20 | jpsharp | Yep, that's it. |
| 19:52.58 | gusto | i noticed that it is not important for asterisk as only the exchange |
| 19:53.13 | gusto | and far more for the endpoints - for example ata's |
| 19:53.52 | gusto | i have no jitter configured on my asterisk, but i had to change it to "hard" on my ata in order to hear anything |
| 19:54.02 | gusto | or "high" |
| 19:54.23 | jpsharp | Do you have high latency ip connections? |
| 19:54.27 | gusto | yes |
| 19:54.33 | gusto | since i have no dsl |
| 19:56.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:57.15 | gusto | does someone have asterisk running with ipv6 connections? |
| 19:57.42 | jpsharp | I do. |
| 19:59.38 | jpsharp | I did it inadvertantly. I set up an ipv6 tunnel and Asterisk started using it for my Google Talk calls. |
| 20:02.27 | wonko | fuck i'm doing that |
| 20:02.32 | wonko | i don't know what the problem is |
| 20:02.44 | wonko | and of course i'm not getting much help from the asterisk irc channel |
| 20:02.52 | wonko | so it's either ipv6 or asterisk forums. :) |
| 20:03.12 | wonko | what do i need to run ipv6 on the 18xx series routers (and the 3560g) |
| 20:03.26 | jpsharp | A tunnel provider. |
| 20:03.46 | trulsk | ipadvancedservices |
| 20:03.58 | jpsharp | That too. |
| 20:04.40 | trulsk | ipv6 unicast-routing enabled and youre set |
| 20:06.23 | jpsharp | And it is definitely using IPv6: talk.l.google.com has IPv6 address 2001:4860:800a::7d |
| 20:06.29 | jpsharp | tcp6 0 0 2001:470:5:47e:20:33167 2001:4860:800a::7d:5222 ESTABLISHED |
| 20:06.42 | wonko | i guess that wiki page needs to be updated then |
| 20:08.15 | wonko | hmm, the 3560g only has ipservices, guess i'll have to upgrade that |
| 20:08.44 | jpsharp | Your sqitch doesn't care if you run ipv6. |
| 20:08.49 | jpsharp | And neither does your switch. |
| 20:09.06 | wonko | it's doing layer 3 routing, so it might care |
| 20:09.06 | wonko | :) |
| 20:09.18 | trulsk | sorry my bad |
| 20:09.34 | trulsk | im running :c3560-ipservicesk9-mz.122-58.SE1.bin, with ipv6 |
| 20:09.40 | wonko | oh |
| 20:09.42 | wonko | good |
| 20:09.42 | wonko | :) |
| 20:09.44 | trulsk | so no need for adv |
| 20:10.02 | wonko | i have no ipv6 command though? |
| 20:10.06 | wonko | Cisco IOS Software, C3560 Software (C3560-IPSERVICESK9-M), Version 12.2(55)SE4, RELEASE SOFTWARE (fc1) |
| 20:10.19 | trulsk | i never remember what its called |
| 20:10.29 | trulsk | but you have to enable it by switching the memory mode to allow it |
| 20:11.02 | trulsk | router(config)#sdm prefer dual-ipv4-and-ipv6 |
| 20:11.21 | trulsk | (then write and reload) |
| 20:12.19 | wonko | do i want to set any biases other than that? |
| 20:13.03 | trulsk | but your 3560 cant do tunnels, if that was what you were trying to do(?) |
| 20:13.15 | wonko | no, the tunnel will happen on the 1841 |
| 20:13.19 | trulsk | ah of course |
| 20:13.23 | wonko | it's just the 3560 routes between vlans |
| 20:13.29 | wonko | so it'll need to know how to do ipv6 as well |
| 20:13.54 | trulsk | your home setup this? |
| 20:13.55 | gusto | googletalk is a good ida |
| 20:13.57 | gusto | ide |
| 20:13.59 | gusto | a |
| 20:14.13 | gusto | but google ist not the only one ... also jabber can do xmpp (voice) |
| 20:14.27 | jpsharp | googletalk is xmpp. |
| 20:14.36 | gusto | so ... would it be possible to do calls from asterisk (over SIP) to XMPP (jabber)? |
| 20:14.49 | wonko | trulsk: yes |
| 20:15.43 | trulsk | wonko: nothing like a home setup flourishing of vlans! my coworkers mock me for the same |
| 20:16.15 | wonko | your coworkers are silly. :) |
| 20:16.27 | jpsharp | I have them as well. Just not nearly as extensive as wonko's. |
| 20:17.44 | wonko | not counting transit networks i actually only have two VLANs right now that get any real use |
| 20:17.47 | wonko | ok, three |
| 20:17.50 | wonko | :) |
| 20:20.21 | jpsharp | Your 1841 has gigabit ethernet on it, yes? |
| 20:20.44 | trulsk | it doesnt come with gig if not by extnsion module |
| 20:20.54 | trulsk | two fe |
| 20:21.41 | wonko | yeah, just two fe |
| 20:21.52 | wonko | and it doesn't have any modules so that's all it has at all |
| 20:22.26 | trulsk | you could probably find a cheap dsl-or-whatever-youre-using-for-uplink-module at ebay |
| 20:23.26 | jpsharp | Yes, you're right. I was thinking of the 2821 with its onboard gigabit. |
| 20:23.54 | wonko | trulsk: cable modem and sadly no DOCSIS 3 module for ciscos |
| 20:23.57 | wonko | i already checked. :) |
| 20:24.02 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 20:24.35 | trulsk | hehe |
| 20:27.43 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 20:38.30 | *** join/#asterisk danfromuk (~IceChat77@2.27.7.133) |
| 20:38.45 | danfromuk | Hi, i'm currently seeing this repeatedly in my CLI. |
| 20:38.45 | danfromuk | -- Nobody picked up in 0 ms |
| 20:39.03 | danfromuk | Is there any way to tell which channel is causing it so that I can 'soft hangup' it? |
| 20:40.42 | danfromuk | also, whats the 'soft hangup' command now in 1.8? |
| 20:40.58 | WIMPy | channel request hangup |
| 20:41.48 | danfromuk | perfect. thanks |
| 20:43.36 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-9.phlapa.fios.verizon.net) |
| 20:47.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:47.45 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.0 (2012/06/04), 1.8.13.0 (2012/06/04), 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 |
| 20:50.01 | gusto | heh? |
| 20:50.14 | gusto | today 10.5.0 came out |
| 20:50.34 | gusto | hmm |
| 20:54.27 | *** join/#asterisk Kalavera (~Kalavera@aquiles.novelix.com.pe) |
| 20:54.40 | Kalavera | greetings guys I am configuring some extensions on a grandstream Ip phone and I keep getting the following error on my asterisk |
| 20:55.17 | Kalavera | [Jun 4 15:47:59] NOTICE[3479]: chan_sip.c:15049 handle_request_register: Registration from '<sip:ext@ipserver>' failed for '192.168.24.25' - Wrong password |
| 20:55.29 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ocozrhmkvbnbyzjj) |
| 20:55.38 | eduzimrs | exit |
| 20:55.42 | Kalavera | retyped the password at least 5 times and I am pretty sure it was correct , is there anything I could be missing? |
| 20:55.44 | *** part/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:56.05 | leifmadsen | Kalavera: it's defintely a misconfiguration of the password somewhere |
| 20:56.35 | leifmadsen | Kalavera: might be some problem with incorrect characters, or unsupported characters, but something in the password is wrong for the peer it is matchin gagainst in asterisk |
| 20:58.47 | Kalavera | leifmadsen: ok it is a grandstream ip phone |
| 20:59.19 | leifmadsen | it sure is? |
| 21:00.05 | *** join/#asterisk serafie (~erin@nat/digium/x-apigripmaojkdwnx) |
| 21:01.31 | Kalavera | ok registered now |
| 21:01.46 | Kalavera | the guy that manage the asterisk wasn't updated the password there |
| 21:01.49 | Kalavera | lol |
| 21:01.58 | Kalavera | leifmadsen: thank you anyways |
| 21:05.47 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
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| 21:25.34 | wonko | trulsk: ipv6 enabled. :) |
| 21:25.59 | trulsk | wohho |
| 21:26.10 | trulsk | now go bug them for native |
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| 21:26.14 | trulsk | :) |
| 21:26.16 | wonko | heh |
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| 21:27.52 | trulsk | it can be done so why not. except for switching to a different kind of dslmodems we havent spent much calories on deploying ipv6 for all customers |
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| 21:46.41 | jpsharp | wonko: Have a tunnel up yet? I recommend Hurricane Electric/Tunnelbroker.net |
| 21:47.30 | wonko | jpsharp: i got the switch rebooted with ipv6 support, that's as far as i've gotten |
| 21:47.35 | wonko | still need to setup the 1841 |
| 21:47.38 | wonko | and the openbsd box |
| 21:47.48 | wonko | although the openbsd box is just enabling ospf6 |
| 21:48.38 | jpsharp | tunnelbroker will give you a Tunnel config you can copypasta directly into your router. |
| 21:48.55 | wonko | nice |
| 21:50.06 | jpsharp | And you've got a static ip, so Bob's your uncle. |
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| 21:54.32 | Jouva | newtonr: Not sure if you're there, but if you recall from a week or two ago, I was having very very weird SIP issues with a new asterisk install |
| 21:55.14 | newtonr | I do. Is that one that looked like it was possibly a RTP routing difficulty ? |
| 21:55.44 | Jouva | Yeah some sort of routing issue. But more just how it was trying to authenticate and not getting the 401 error and timing out |
| 21:56.16 | Jouva | Now here's the crazy part about it. I got it to work and I don't know WHY this is the issue |
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| 21:57.08 | Jouva | But it didn't like that IP. If I used the main IP of the box it worked fine. But the hostname pointed to another IP and it never worked on that one. Yet netstat says port 5060 is being listened on by 0.0.0.0 so it SHOULD work on ALL IPs |
| 21:57.43 | Jouva | And yes I'm sure the IP in the DNS is correct as it wouldn't even contact it at all if it wasn't correct |
| 21:58.38 | newtonr | strange. I'm no networking expert. What about iptables or something along the lines of software firewalling on the system? |
| 21:58.59 | Jouva | There's no firewall setup on there |
| 22:00.51 | newtonr | so when you got it to work, in the phones interface you plugged in the IP of the asterisk system, instead of the hostname? |
| 22:01.00 | Jouva | Yep |
| 22:01.51 | Jouva | But now I have an even weirder problem. THAT system works. Hard phone (Aastra 6755i) works perfectly on that one. But on ANOTHER server, the same phone logs in fine and accepts calls, but trying to make outgoing calls won't work. And the extension.conf files are nearly identical. Soft phone works trying to make calls on the 2nd system |
| 22:02.33 | Jouva | Asterisk logs nothing. It fails on the end of my phone for some reason. |
| 22:02.54 | navaismo | fax in voip enviroment drive me crazy |
| 22:04.08 | newtonr | Jouva: for that kind of issue, i'd always look at a VERBOSE and DEBUG log when making the call. If nothing relevant there, then look at a SIP packet trace to see if the phones INVITE ever reaches asterisk |
| 22:04.22 | newtonr | if it does, then Asterisk should be saying something in VERBOSE and DEBUG |
| 22:04.36 | Jouva | what should I set verbose and debug to? 7? |
| 22:04.41 | Jouva | in the console that is |
| 22:04.53 | newtonr | at least 5, i dont think they use much above that |
| 22:06.10 | Jouva | Yeah it logged nothing |
| 22:06.13 | Jouva | hmmm hold on |
| 22:06.23 | Jouva | how do I turn on the SIP packet trace stuff again? |
| 22:06.47 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 22:06.55 | newtonr | probably sip set debug on |
| 22:08.16 | newtonr | can't remember what version you are on, but that should work for 1.6 and higher |
| 22:08.38 | Jouva | ok I got some debug info, one moment |
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| 22:12.49 | Jouva | Ok I've replaced the domain with example.com to hide the identity of the server |
| 22:12.51 | Jouva | http://pastebin.com/AqFLGmSZ |
| 22:13.01 | Jouva | Otherwise everything is exactly intact |
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| 22:23.11 | *** join/#asterisk danfromuk (~IceChat77@2.27.7.133) |
| 22:24.43 | danfromuk | Hi, I've discovered that some calls are not being recorded to cdr mysql. Is there any way to debug this? |
| 22:25.15 | danfromuk | Does csv work 100% of the time? I'm considering switching to that, or adding it as a failsafe to cover the missing details. |
| 22:26.57 | pithagorians | looks like a bug https://issues.asterisk.org/view.php?id=18689 |
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| 22:28.53 | danfromuk | Im using 1.8. |
| 22:29.37 | pithagorians | the calls not registered are unanswered? |
| 22:30.44 | agisamentum | I'm having a bit of trouble with a short circuit of TDM => TDM calls where the CDR is not picking up what's actually happening in the cabinet |
| 22:30.55 | pithagorians | did you try CEL in combination with CDR? |
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| 22:31.04 | danfromuk | Sorry, CEL? |
| 22:31.19 | agisamentum | apparently there's a messaged on the D channel called "2 B Channel Transfer" that tells the other end I am tying two B channels together |
| 22:31.40 | agisamentum | where under DAHDI tools would I find this message? |
| 22:32.05 | pithagorians | channel event logging |
| 22:32.07 | agisamentum | excuse me if I'm vague, I'm still a bit green |
| 22:32.08 | WIMPy | agisamentum: If it's enabled, Dial() will use it. |
| 22:33.58 | agisamentum | thank you, looking now |
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| 22:36.23 | pithagorians | <WIMPy> hi |
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| 22:37.19 | danfromuk | pithagorians: thanks. i'll take a look at it. |
| 22:37.26 | pithagorians | any working sample on how incoming calls from sipgate should be routed according to the DDIs? |
| 22:38.35 | WIMPy | pithagorians: What's special about that? |
| 22:39.12 | pithagorians | for ex send the call to the proper DDI |
| 22:39.35 | WIMPy | Send? |
| 22:39.41 | pithagorians | when there are many DDIs |
| 22:39.59 | pithagorians | receive* |
| 22:40.25 | WIMPy | That's what your dialplan is there for. |
| 22:40.32 | agisamentum | WIMPy: So under the CEL i'll be looking for BRIDGE_START? |
| 22:40.56 | agisamentum | as the type of event |
| 22:41.52 | WIMPy | agisamentum: Yes, but I'm not sure there's a way to find out if it's handed out to the switch. |
| 22:44.53 | agisamentum | what's interesting is the call will remain active but the channels are now free... and I'm not sure if the message indicates if the call has ended or to tie them back together through 2 B channels so we can record it for CDR accountability or simply end it. just hangs |
| 22:45.11 | agisamentum | facepalms |
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| 22:46.41 | WIMPy | agisamentum: Not sure what your question is, but, yes you continue to receive information about the call even if it doesn't use any of your channels. |
| 22:47.11 | WIMPy | Just as with forwarded calls. |
| 22:49.25 | agisamentum | sorry for my lack of clarity, I do have a message on the D channel, the 2 B channel Transfer message telling us two B channels are being tied together, the call remains active, but the channel frees up |
| 22:49.49 | agisamentum | the CDR only records about 5 seconds and the call remains active but asterisk appears to free the channel up if that makes sense |
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| 22:50.50 | michael-i | Is there a way to send an AMI event to a specific AMI client/username? |
| 22:50.54 | agisamentum | I need the channel to remain tied and not freed until the call has actually ended for accuracy of the CDR |
| 22:51.28 | WIMPy | agisamentum: Yes, Asterisk probably can't handle that situation. |
| 22:51.53 | WIMPy | I guess you can only disable transfers or do some wild coding. |
| 22:52.17 | agisamentum | hmm |
| 22:53.51 | agisamentum | WIMPy: I appreciate your help, as I've mentioned I'm relatively green, apart of the Asterisk 3rd edition book, any pointers to some pertinent documentation I can try? |
| 22:54.18 | WIMPy | On what exactely? |
| 22:54.48 | agisamentum | how I can try to get asterisk to behave in synchronisation with my cabinet |
| 22:55.20 | agisamentum | assumes by the "asterisk probably can't handle that sit" comment, maybe not |
| 22:55.56 | WIMPy | As I said: disable transfers in chan_dahdi.conf. |
| 22:56.18 | WIMPy | That way you won;t bw able to free the channels, but Asterisk will know what's going on. |
| 22:56.53 | agisamentum | that makes sense, hope it works with the techs in the back... thanks again :) |
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| 23:14.15 | agisamentum | WIMPy: okay I had a bit better of an explanation of what's happening here... if i have a call being xferred between two B channels, how would I tell the CO that both B channels are in progress over the PRI or TDMoE? |
| 23:14.56 | agisamentum | so if i get a new call coming in, the channels are tied and new calls to either of the channels will be sent to an exception handler/voicemail |
| 23:15.52 | agisamentum | so chalk what I previously sent, as I wasn't fully comprehending the issue until now |
| 23:15.52 | WIMPy | That didn't make sense to me. |
| 23:17.40 | WIMPy | What exactely is it you're having an issue with? Or what do you want to change? |
| 23:18.42 | agisamentum | alright, the project is a cabinet and I have two B channels xferring a call between them, I need to be able to tell the CO over PRI or TDMoE that both channels are busy or in progress and instead of the channel being released, I need new inbound calls to either of the channel to be sent to voicemail |
| 23:19.05 | agisamentum | did that make better sense or do I need to rest my head for the end of the day and think of a better way to express my problem |
| 23:19.08 | agisamentum | :p |
| 23:19.14 | agisamentum | thank you for your patience with me |
| 23:19.29 | WIMPy | That doesn;t make sense. Channels are assigned randomly. |
| 23:19.45 | WIMPy | (not exately but that's what you should assume) |
| 23:20.18 | WIMPy | This feature exists just to free up the channels. |
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| 23:23.32 | agisamentum | thanks again WIMPy I'll have to sit and mull on this one and hopefully more efficiently express it tomorrow |
| 23:24.00 | agisamentum | s/efficiently/effectively/ |
| 23:24.12 | WIMPy | What exactely is it you're having an issue with? Or what do you want to change? |
| 23:25.21 | WIMPy | If you've got an issue with your CDRs, switch the feature off at the cost of not freeing the channels. |
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