| 00:00.06 | cmnky | was still cheaper than a new one ... but if i had to do it over ... id have just bought a cisco phone |
| 00:00.33 | citywok | ~phones |
| 00:00.33 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 00:01.57 | cmnky | any used those ooma "free voip" boxes ? |
| 00:02.02 | cmnky | er..anyone |
| 00:02.35 | ruben23 | guys any setting on my chan_dahdi.conf i would chaneg how should it take effect-ill restart dahdi..? or asterisk..? |
| 00:03.32 | *** join/#asterisk rrb3942 (~rbullock@67.242.215.62) |
| 00:06.23 | cmnky | citywok, looking at neweggs offerings now ... seen the 79xx ... not the others ... thx |
| 00:07.07 | citywok | my personal preference is the aastra 57i devices |
| 00:07.22 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 00:08.50 | cmnky | citywok, those show up on ebay reasonable ? |
| 00:09.01 | citywok | they are like $195 new |
| 00:09.18 | citywok | i'm buying for a business so i just order them from my local vendor and they show up |
| 00:10.19 | cmnky | sure ... im buying for a business too ... but i own the business, i have no local vendors, no money, and no it staff ... so i pray alot ;) |
| 00:10.38 | citywok | i'm buying for a call center |
| 00:11.12 | cmnky | ah ha ! your the reason everyone needs asterisk ... to block telemarketing calls |
| 00:12.20 | citywok | b2b call center :) |
| 00:12.28 | citywok | not a harassment collections agency call center lol |
| 00:14.18 | cmnky | i have the 2001 phone book DVD records in ASCII format |
| 00:14.25 | cmnky | when i get asterisk working i might run for president |
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| 00:25.23 | jsjc | p3nguin look what i found: app_exec.c:222 execif_exec: Deprecated syntax found. Please upgrade to using ExecIf(<expr>?(null)((null))) |
| 00:30.17 | talntid | citywok, what call center do you do? i'm also a b2b call center :) |
| 00:30.59 | talntid | we do the advertisements on the back of the reciept tape in grocery stores nation wide... call businesses around the grocery stores to set up appointments for the managers to go in and talk to 'em |
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| 00:39.12 | cmnky | thanks [TK]D-Fender ... you rock |
| 00:42.55 | Docfxit | Hi. |
| 00:43.05 | Docfxit | I would like to resolve a problem with a sip phone. The error is Server did not respond to register. I posted my config files and error logs at http://pastebin.com/U7hdPNBv |
| 00:47.00 | Docfxit | A packet capture on the router of the sip phone shows it's passing data. |
| 00:48.14 | citywok | talntid: we do channel marketing for tech co's, and we have a SaaS product for partner management |
| 00:53.25 | cmnky | my spa2102 has no dial tone because ______ {fill in the blank} |
| 00:53.51 | citywok | you touch yourself at night |
| 00:54.03 | cmnky | thats besides the point |
| 00:57.04 | cmnky | answer: ... because its not talking to asterisk yet |
| 00:59.16 | cmnky | 192.168.5.23 ... i gots da ip |
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| 00:59.31 | ecrane | _Corey_: hmm... it seems there IS a way to set variables via .call files... my mistake. "Setvar: <varname:value>" |
| 01:01.51 | Docfxit | Could I please get somebody to take a look at my configuration? |
| 01:06.43 | cmnky | sweet ... i got a web admin interface to my phone ... how cool is that |
| 01:06.51 | cmnky | <--- {easily impressed} |
| 01:09.30 | leifmadsen | citywok: I built call centres too :) I only build inbound call centres, never outbound |
| 01:09.48 | leifmadsen | been asked a few times to build dialers, and I tell them I refuse to build them out of moral implications :) |
| 01:10.28 | cmnky | build them on a fee per call basis ... with an annual morals review ... if they are using it immorally ... up your fee ;) |
| 01:12.17 | citywok | leifmadsen: you missed my poking fun at you and russell further up |
| 01:12.40 | citywok | yea thank god we don't do that dialing crap, microsoft won't allow us to do anything like it. |
| 01:13.04 | cmnky | citywok, you work at ms ? |
| 01:13.15 | citywok | no, we're a vendor |
| 01:13.37 | cmnky | i almost applied at ms awhile back ... when i was in tacoma ... thank god i came to my senses ;) |
| 01:13.48 | citywok | heh, i grew up in university place |
| 01:14.05 | citywok | now as you can imagine i live and work in redmond |
| 01:14.38 | greezmunkey | citywok: you must like heavy traffic :) |
| 01:14.52 | citywok | i live 1.1 miles from work and my commute takes 3 minutes. 5 if i miss both lights. |
| 01:15.00 | greezmunkey | nice |
| 01:15.30 | citywok | i sped this morning (60 in a 40) and got here in 2 |
| 01:15.47 | cmnky | citywok, so are you saying the federal gov didnt really need to build that bridge for ms ? |
| 01:16.04 | citywok | i just don't deal with that side of things. i work on 148th and live on 140th :P |
| 01:16.18 | citywok | @ 51st |
| 01:16.59 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 01:17.25 | citywok | i didn't realize there were so many redmond people here haha |
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| 01:17.51 | cmnky | im on the other side of the us now ... this was 10+ years ago |
| 01:18.20 | citywok | 10 years ago i was in highschool :P |
| 01:18.38 | greezmunkey | Docfxit: looking at your extensions.conf, lines 27 and 28 in the pastebin, seems like you've crafted a nice loop for Gary. If I'm wrong, please elaborate. |
| 01:20.04 | citywok | greezmunkey: i don't think that would create a loop. the call should come in under s,1 right? no extension is being passed to gary |
| 01:20.22 | cmnky | sweet ... just locked myself out of my SPA2102 ... luvin it |
| 01:20.41 | citywok | but the lack of exten => is messed up on lines 18 through 24 |
| 01:20.49 | greezmunkey | citywok: the number dialed is 120@sip.... |
| 01:22.08 | citywok | Yea, i'm not really sure what he's trying to accomplish. it looks messed up so the functionality can't exactly be great. |
| 01:22.12 | p3nguin | line 18-24 of what? |
| 01:22.25 | citywok | Docfxit's pastebin above, like 30min ago |
| 01:22.51 | p3nguin | I looked at it, but I didn't see exten missing anywhere. |
| 01:22.58 | citywok | he's missing the > |
| 01:23.03 | citywok | on a handful of lines |
| 01:23.11 | Docfxit | I have a software sip phone trying to connect remotely with asterisk. |
| 01:23.32 | p3nguin | oh. It's not required, but I don't know it it's okay to switch from not having it to having it mid-extension or not. |
| 01:23.51 | Docfxit | Which lines are missing the > |
| 01:23.58 | p3nguin | 18-24 |
| 01:24.32 | Docfxit | I'll put those in right now. |
| 01:24.56 | citywok | and it does look like you could have a loop if you are dialing sip/gary and it comes in on 120 which then loops back and dials again |
| 01:24.56 | p3nguin | jsjc: Cool. That's quite useful that it knew. |
| 01:27.48 | cmnky | i have 0 clue what im looking at ... this is great |
| 01:30.11 | Docfxit | I added > to 18-24 |
| 01:30.42 | Docfxit | citywok How can I change it so there isn't a loop? |
| 01:30.51 | citywok | Fix the logic |
| 01:32.00 | Docfxit | citywok Could you please let me know what changes would fix the logic? |
| 01:32.02 | cmnky | anyone have the skinny on a US based dirt cheap/free voip provider ? |
| 01:32.11 | citywok | ~itsp |
| 01:32.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:32.42 | cmnky | is the usa list gonna spam the channel ... if so, can i find it elsewhere ? |
| 01:32.47 | citywok | Docfxit: i'm not going to fix your logic for you. But you should figure out how you want your call routing to work and then look at how it routes |
| 01:32.49 | citywok | ~itsplist-us |
| 01:32.49 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 01:33.15 | p3nguin | cmnky: Are you looking for termination service? |
| 01:33.38 | cmnky | origination and termination ... ie .. trying to emulate 1 pots line |
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| 01:33.45 | Docfxit | citywok Thanks for your help. |
| 01:33.49 | citywok | Alright i'm taking off for the day. |
| 01:34.10 | cmnky | thanks for all the help citywok |
| 01:34.16 | citywok | yup |
| 01:35.57 | cmnky | i read before about free #, free origination, 0.01 per min termination ... but that was 2 years ago |
| 01:36.54 | p3nguin | There are a few companies that will give you a fee DID and free incoming calls, and I pay about 1 cent per minutes for termination through VoIP.ms. |
| 01:37.02 | p3nguin | s/fee/free/ |
| 01:37.38 | cmnky | any leads on the free DID/incomming ... or just check the US list ? |
| 01:37.42 | p3nguin | Did you consider Google Voice? |
| 01:37.59 | cmnky | i dont like using google for anything if i can avoid it |
| 01:38.07 | p3nguin | IPKall, Sipgate, IPcomms |
| 01:38.24 | cmnky | their TOS says they p0wn all your data ... id like privacy to some degree |
| 01:38.31 | cmnky | or at least .. legal rights to my data |
| 01:40.40 | cmnky | thnks p3nguin |
| 01:43.59 | cmnky | should i prefer an IAX2 vendor ? |
| 01:44.18 | p3nguin | It's nice to have the option. |
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| 01:50.28 | Docfxit | I read someone connected to MagicJack for $20/yr/line for all calls in US and CAN. |
| 01:50.52 | cmnky | they made asterisk pretend to be a magic jack ? |
| 01:51.13 | Docfxit | I don't know how they did it. |
| 01:51.48 | p3nguin | There's at least one forum where people talk about hacking Magic Jack with Asterisk. |
| 01:52.10 | cmnky | i called magickjack support to ask about linux once ... and got the expected response |
| 01:52.22 | p3nguin | "What's that?" |
| 01:53.08 | cmnky | they just kept repeating "Currently, mj only supports Windows operating systems at this time" |
| 02:02.27 | cmnky | when a voip providers plan talks about "channels" .. is that the equivalent of "simultaneous calls" ? |
| 02:04.20 | coppice | "Linux is engineered, but we only work with magic" |
| 02:10.12 | p3nguin | cmnky: Each leg of a call requires a channel. If you have two incoming calls, you're using two incoming channels. If you put both of those callers on hold or in a queue and then place an outbound call, now you are using two inbound channels and one outbound channel. |
| 02:10.54 | cmnky | right ... a channel is therefore, a POTS line, without a number ... thanks p3nguin |
| 02:13.23 | Corydon76-home | Well, sometimes a POTS line occupies two channels, depending upon whether you have call waiting or not |
| 02:13.32 | Corydon76-home | only one concurrent channel, though |
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| 02:38.14 | pabelanger | Odd.... FRACK!, Failed assertion s != NULL, id=9 (0) at line 1790 in iax2_destroy_helper of chan_iax2.c |
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| 02:45.40 | pabelanger | nevermind, my issue |
| 02:48.35 | cmnky | i would help ... if it were within my abilities ;( |
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| 03:01.52 | jshupe | I am having some issues with WaitExten() on Asterisk 1.6 using Broadvoice as my DID provider. |
| 03:02.03 | p3nguin | What's the problem? |
| 03:02.27 | jshupe | It doesn't seem to be registering anything I type, and ends up timing out. |
| 03:02.42 | p3nguin | What dtmf mode are you using? |
| 03:03.18 | jshupe | inband... I was wondering if that was the problem but am unsure what to set it to. |
| 03:03.54 | p3nguin | Set it to rfc2833. |
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| 03:04.36 | jshupe | I apologize for my ignorance.... but where, exactly? Under my [sip.broadvoice.com] context in sip.conf, or somewhere else? |
| 03:06.02 | p3nguin | I'd set it in my general section as well as every peer. |
| 03:06.12 | p3nguin | just to be thorough. |
| 03:06.42 | p3nguin | (in sip.conf, yes) |
| 03:09.32 | jshupe | I just tried setting every instance of ^dtmf to rfc2833 and restarting Asterisk, but my keys are still being ignored. |
| 03:10.11 | p3nguin | dtmfmode=rfc2833 |
| 03:10.13 | jshupe | <PROTECTED> |
| 03:10.13 | jshupe | [Nov 30 21:09:07] WARNING[88501]: pbx.c:8788 pbx_builtin_waitexten: Timeout but no rule 't' in context 'broadvoice-in' |
| 03:10.46 | jshupe | adams# grep ^dtmf sip.conf |
| 03:10.46 | jshupe | dtmf=rfc2833 |
| 03:10.46 | jshupe | dtmfmode=rfc2833 |
| 03:12.39 | jshupe | Is it possible that my extensions.conf is incorrect? It is very simple- 1,Answer() ... 2,WaitExten(10) on Asterisk 1.6.2.13 on FreeBSD. |
| 03:13.42 | p3nguin | I would expect it to work. Care to try a BackGround() of some file and see if it responds? |
| 03:14.03 | jshupe | sure... I'll give that a try. |
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| 03:19.15 | jshupe | seems to work fine with BackGround() |
| 03:19.29 | jshupe | any idea why that would make a difference? |
| 03:20.01 | p3nguin | My concern was that WaitExten() wasn't actually listening when it was preceeded only by Answer(). |
| 03:20.46 | jshupe | well, thanks for the help. I'm going to go figure out the rest of my configuration now. |
| 03:20.47 | p3nguin | Try a Playback(silence/1) before your WaitExten(10) and see if it works that way. |
| 03:21.52 | jshupe | seems to work great that way as well. |
| 03:22.42 | p3nguin | I wonder if there's any documentation on WaitExten that mentions it won't work the way you were using it. |
| 03:23.31 | p3nguin | It could be a bug, or it could be a design limitation. |
| 03:23.38 | jshupe | Not that I have seen, I've been Googling for hours. (I'm relatively new to Asterisk, though.) |
| 03:24.01 | jshupe | Either way I can work around it. I really appreciate your help. |
| 03:24.34 | p3nguin | I'm glad we were able to overcome the problem so easily. |
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| 04:00.26 | cmnky | <PROTECTED> |
| 04:00.26 | cmnky | <cmnky> i know there are exceptions to that rule ... just speaking in general |
| 04:08.49 | ChannelZ | that's usually how most telcos operate yes |
| 04:09.18 | cmnky | ok ... thnx ... trying to plan my own personal "friends and family" network around free local DIDs |
| 04:24.47 | p3nguin | cmnky: Long distance toll charges are usually based on distance from a telephone company's central office. |
| 04:26.11 | p3nguin | According to an AT&T customer service representative, a number "based" more than 15 miles from me is long distance, and long distance charges would apply if I call it. |
| 04:26.21 | cmnky | p3nguin, yeah ... its not that tho ... im just trying to use up my free DID numbers/incomming/accnts in a price effective manner for the few people that would be calling me |
| 04:26.41 | cmnky | i know area code -> area code is not a hard and fast rule ... just a general approximation |
| 04:30.32 | pabelanger | cmnky: you will usually here, 'this is a long distance call, hang up and dial again' |
| 04:30.46 | cmnky | anyone useing voxox with asterisk ? |
| 04:30.47 | pabelanger | rather then the telco charging you |
| 04:31.17 | cmnky | pabelanger, yes ... but im not trying to avoid long distance calls ... im trying to guarantee the #'s i get are local ... similar but different ;) |
| 04:34.50 | cmnky | U.S. & Canada Unlimited Calling Plan $2.45 per month (voxox) |
| 04:35.34 | pabelanger | Free North America calls with GoogleVoice ATM |
| 04:35.52 | cmnky | i dont want to use google voice ... they'll own my phone call content if i do |
| 04:36.07 | pabelanger | o.0 |
| 04:36.29 | cmnky | the TOS for every one of their services gives them IP rights to *your* content ... i don't like that |
| 04:37.15 | cmnky | if they speech-to-text your business conversation, they can steal your ideas and theres nothing you can do about it ... doesn't sit well with me |
| 04:37.45 | pabelanger | Google does not claim any ownership in any of the content that you or your callers upload, transmit or store in your Google Voice account. We will not use any of your content for any purpose except to provide you with the Service. |
| 04:38.09 | cmnky | ah ... right ... "except to provide you with the Service" |
| 04:38.20 | cmnky | which means if you talk about a new voip codec ... its theirs |
| 04:38.33 | cmnky | it sounds kosher ... but it isnt |
| 04:39.43 | cmnky | they have stated publicly ... that "googles goal is to know more about you than you do" |
| 04:40.17 | sawgood | Would it be possible to combine the following to extensions.conf lines into one line? |
| 04:40.19 | sawgood | exten => 4155551212,1,Set(CALLERID(num)=(415) 555-1212 / ${CALLERID(num)}) |
| 04:40.19 | sawgood | exten => 4155551212,n,Dial(SIP/1212/${EXTEN}) |
| 04:40.20 | cmnky | you won't catch me paying $300 for google tv so they can monitor my viewing habits either |
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| 04:41.05 | candrews | Does gtalk use res_stun_monitor or does it do it's own thing for stun? |
| 04:41.20 | pabelanger | sawgood: no |
| 04:41.21 | p3nguin | candrews: its own |
| 04:41.26 | candrews | I'm wondering if setting "stunaddr" in gtalk.conf is redundant with res_stun_monitor.conf |
| 04:41.39 | candrews | Thanks - I'll set both. |
| 04:42.01 | candrews | Out of curiosity... why is that? If SIP and IAX use res_stun_monitor, why doesn't gtalk? |
| 04:46.19 | sawgood | Thing is I can get the two lines working in one context (which is one ITSP), but the same lines with different numbers do not work in another context (different ITSP) |
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| 04:46.34 | sawgood | same syntax, different numbers, different ITSP |
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| 04:51.02 | cmnky | voxox use with * looks sketchy ... im passing for now |
| 04:53.35 | leifmadsen | candrews: chan_gtalk is an older module before res_stun_monitor existed |
| 04:57.23 | candrews | thank you. makes a lot of sense. |
| 04:59.28 | trollasaurus | I've been using Google Voice with Asterisks for a few weeks... I haven't seen any problems. |
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| 05:02.19 | candrews | does anyone have a handy way to attempt to route calls over enum, and fall through if unsuccessful? I've had tremendous trouble trying to make enum work. |
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| 05:37.52 | cmnky | okay ... since finding other voip providers is really starting to be a complicated drag ... im opting to use google voice for testing ... it works with asterisk yes ? |
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| 05:57.57 | ChannelZ | sawgood: that first Set line is a mess and pretty much totally invalid |
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| 06:27.21 | Kobaz | what's the longest value you can put in a astdb entry |
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| 07:00.57 | greezmunkey | lo |
| 07:01.10 | ChannelZ | Bonzai! |
| 07:02.22 | greezmunkey | ChannelZ: I loaded another centos system, all good. I have one issue with music on hold, but I'll deal with that later. Other than that it's all good! |
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| 07:17.12 | schmidts | good morning |
| 07:17.17 | ChannelZ | Hurray |
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| 07:22.26 | ruben23 | hi guys is it possible like to call a certain state in the us and automatically my callerid based on teh state area code..? like if i call canda-ill get canda callerID adn if US-chicago-i get chicago-if CA i get CA-caller ID also. |
| 07:24.17 | p3nguin | If I understand you correctly, that would be done with pure dialplan. |
| 07:25.25 | p3nguin | Set the caller ID num based on the extension pattern you're dialing. It will take a bit of work to determine which areas get which caller ID numbers, but it can certainly be done. |
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| 07:44.45 | sawgood | http://pastebin.com/dDxXvUtj |
| 07:45.06 | sawgood | Any chance someoone could read the short bit I wrote at pastebin.com and give me an opinion? |
| 07:45.42 | p3nguin | You'd be better off using Verbose() instead of NoOp(). |
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| 07:46.08 | sawgood | p3nguin: thank you |
| 07:46.28 | sawgood | do those commands go into extensions.conf like NoOp does? |
| 07:46.55 | kaldemar | sawgood: "TOP of the context" includes an invalid assumptions. it doesn't matter where an extension is in a context, but how good a match it is. use n priority, you can comment those out as you wish. |
| 07:46.57 | p3nguin | Yes, Verbose() is a dial plan application. |
| 07:48.01 | sawgood | So, in a context, could it start with a priority 1 statement, followed by 10 n statements, followed by another priority 1 statement, etc. |
| 07:48.25 | p3nguin | no |
| 07:49.19 | sawgood | does there have to be a priority 1 set (or some number) first and then everything else could be n? |
| 07:49.32 | p3nguin | Each extension must start with priority 1 and a subsequent priority 1 is only used in very special cases and would be right after the first one. |
| 07:49.45 | kaldemar | unless you have a priority 1 in a matching extension, nothing will happen. |
| 07:49.56 | p3nguin | The case where you would have a second 1 might be for old-style caller ID matching. |
| 07:50.29 | kaldemar | old-style? |
| 07:50.32 | sawgood | ok ... what if i wanted a NoOp or Verbose output to happen before the priority 1? |
| 07:50.44 | p3nguin | Under normal circumstances, you'll start every extension with priority 1 and never have a second priority 1 in it. |
| 07:50.54 | kaldemar | sawgood: there is no such thing as "before 1". |
| 07:50.59 | p3nguin | There is nothing before 1. |
| 07:51.01 | *** join/#asterisk mintos (~mvaliyav@114.143.166.232) |
| 07:51.05 | p3nguin | 1 is where it begins. |
| 07:51.23 | sawgood | Say I have a priority 1 extension line ... followed by n statements ... |
| 07:51.56 | kaldemar | the first n becomes 2 when the parser reads the config. the second n becomes 3 and so on. |
| 07:52.08 | sawgood | Now, I need something to be done prior to the priority 1 (just on a troubleshooting basis) ... would have to change the 1 to an n (and then make the noop a 1)? |
| 07:52.21 | p3nguin | If you're having a problem with commenting out the priority 1 when you want to enable your debug lines, use a NoOp() on priority 1 and never touch it. |
| 07:52.42 | p3nguin | exten => 123,1,NoOp() |
| 07:52.43 | kaldemar | sawgood: yes. NoOp as 1 and the previous 1 as n. |
| 07:53.00 | p3nguin | exten => 123,n,Verbose(stuff to see on the CLI) |
| 07:53.30 | p3nguin | Then never touch the priority 1, but comment out the other lines as needed. |
| 07:54.18 | sawgood | makes more sense ... what is the advantage/difference between the NoOp and Verbose lines you gave as examples (one started with a 1 followed by a n) |
| 07:54.45 | p3nguin | NoOp() does nothing, so no problem using as many as you want. |
| 07:55.02 | p3nguin | Verbose() prints things on the console. |
| 07:55.02 | kaldemar | sawgood: core show application Verbose, core show application NoOp |
| 07:55.36 | sawgood | p3nguin: If I used a NoOp call, and then my dialplan stopped working, it would probably be more related to the priority settings I was using? |
| 07:55.45 | p3nguin | yes |
| 07:56.03 | sawgood | I'm glad I took a 2 hour break (sometimes breaks help)! |
| 07:56.23 | p3nguin | Because each extension has to start with priority 1, and you usually won't have another pri 1 at all, ever, in the same extension. |
| 07:56.57 | p3nguin | I can show you an example of where you'll duplicate the priority, though. |
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| 07:57.20 | SeTTleR | hi |
| 07:57.43 | sawgood | So, if I have a priority 1 (very first line of the context) which is a NoOp call, could I NOT use a _X. pattern then (since priority 1 is done first) and the NoOp call would perform the action? |
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| 07:59.02 | p3nguin | Like I mentioned, NoOp() is an application which does nothing. |
| 07:59.18 | p3nguin | Like Dial() does something, NoOp() does nothing. |
| 07:59.22 | kaldemar | sawgood: it doesn't matter if the priority is the first line of the context. it can be the 100th line if the extension matches. |
| 08:00.01 | sawgood | kaldemar: got it ... very strong point |
| 08:00.33 | sawgood | let me show you two examples ... (writing them out now) |
| 08:01.04 | kaldemar | sawgood: if you have something else by priority 1 that is a better match than _X., your _X. will not be executed. order of extensions in a context is irrelevant. includes are used in the order they appear in the context though. |
| 08:01.43 | sawgood | kaldemar: that answers my question and next question too |
| 08:01.48 | sawgood | thank you |
| 08:02.58 | sawgood | summary: once a priority 1 matches (no other priority 1 lines would be used) ... got it ... but would the next step be to process priority 2 and/or n lines? |
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| 08:03.55 | p3nguin | Here's an example of duplicating a priority. In this case, I have three priority 1s. http://pastebin.com/5jEeaHxS |
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| 08:04.24 | p3nguin | priority n means NEXT. If you have 1, n, n ... that means 1, 2, 3. |
| 08:05.00 | sawgood | you duplicated priority 1 because they had nothing to do with each other, but once a match was found (all things need #2 and or the n lines) |
| 08:05.24 | kaldemar | sawgood: after 1 comes 2. |
| 08:05.35 | p3nguin | Each of my three pri 1 lines is unique. |
| 08:05.39 | ChannelZ | Enough with your lie-telling! |
| 08:07.09 | sawgood | The use of priority 1, and the either 2, or a n (could be a huge difference) |
| 08:07.31 | sawgood | you might want something processed through a bunch of lines (so stick with n) |
| 08:07.32 | p3nguin | Don't get in the habit of using 2, 3, etc. Use n. |
| 08:08.49 | ChannelZ | Unless you long for the days of Commodore 64 BASIC |
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| 08:09.02 | sawgood | When something enters the context (and has been processed by a priority 1 for example) and after the priority 1 line comes 25 line with n, it is possible the item could match on all, some, or none of the n lines |
| 08:09.23 | p3nguin | You jumped from context to priority. |
| 08:09.30 | p3nguin | You totally bypassed the EXTENSION. |
| 08:10.21 | sawgood | What is the correct term for something data coming into the extensions.conf file, entering a context, and started to be processed? |
| 08:10.23 | p3nguin | The context is just a container for zero or more extensions. |
| 08:10.55 | sawgood | would it be fair to say ... an 'extension' is parsed through extensions.conf |
| 08:11.08 | kaldemar | sawgood: a call will match an extension. |
| 08:11.29 | sawgood | calls enter the file to be processed and matched by extensions! |
| 08:11.32 | sawgood | got it! |
| 08:11.43 | p3nguin | You were right in saying that the call is directed to a context, but once it gets to that context it has to match an extension for anything to happen. |
| 08:12.10 | p3nguin | Once the extension is matched, the priorities are used as a "list of commands" to execute. |
| 08:12.19 | ChannelZ | Priorities sequence an extension's actions together; Extensions are grouped in contexts. |
| 08:12.43 | kaldemar | calls enter your dialplan, extensions.conf is just a storage that asterisk reads to create a dialplan. "dialplan show" in CLI will show you what the dialplan has. |
| 08:12.46 | sawgood | So, if a call arrives to a context, and several extension lines match the call (all having a n priority) each and everone one of the n lines would be processed in the order they are in? |
| 08:12.55 | p3nguin | I was just realizing that I have way too many contexts with just extension 's' in them. |
| 08:13.31 | p3nguin | The extension match happens and the extension starts processing the call at priority 1. |
| 08:14.42 | p3nguin | As I said a few minutes ago: 1, n, n equals 1, 2, 3. |
| 08:14.59 | p3nguin | The call starts at 1, then goes to 2 which was written literally as n. |
| 08:15.24 | p3nguin | pbx_config converts each n to the Next priority. |
| 08:15.46 | p3nguin | "dialplan show" will show you how it converts them all to numbers. |
| 08:15.52 | sawgood | cool |
| 08:16.42 | p3nguin | Well, it'll show you that it has converted them to numbers... rather than showing HOW it converts them. |
| 08:16.45 | p3nguin | Bad wording on my part. |
| 08:19.19 | sawgood | even though I am a definite dialplan rookie, I can see where you might want a priority 1 entry (definitely), and then a priority 2, and then use n for everything else (so you could sort of control how a call is handled) |
| 08:19.37 | p3nguin | 1, n, n |
| 08:19.44 | p3nguin | There's no need for 1, 2, n. |
| 08:20.43 | sawgood | ok ... |
| 08:20.59 | p3nguin | You _could_ go back to how it worked years ago. n wasn't always valid -- you had to number each line. |
| 08:22.24 | p3nguin | But evolution has given us priority n and labels so we don't have to number every line, so why not take advantage of it? |
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| 08:23.56 | sawgood | take this line for example: |
| 08:23.59 | sawgood | exten => s,1,Set(Var_TO=${SIP_HEADER(TO)}) |
| 08:24.06 | sawgood | how would this ever match anything? |
| 08:24.14 | sawgood | there is no _X. |
| 08:24.16 | p3nguin | It matches extension 's' |
| 08:24.22 | sawgood | oh |
| 08:24.53 | p3nguin | It could have just as easily been exten => poopyhead,1,NoOp() |
| 08:25.31 | sawgood | what is this 'extension' s (I see it mentioned in dialplans often) |
| 08:25.42 | p3nguin | _ means you're building a pattern. X is a number. . means one or more characters. |
| 08:26.12 | p3nguin | _X. matches one number and one or more characters. |
| 08:26.33 | p3nguin | If you called 2b on your phone, it would match _X. |
| 08:27.30 | p3nguin | ~standard-extensions |
| 08:27.30 | infobot | [~asterisk standard extensions] http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 08:27.42 | p3nguin | Maybe that'll help. |
| 08:28.18 | sawgood | 100% |
| 08:28.51 | p3nguin | Extension s is often used in IVR/attendant dialplan where you don't need to send a phone number, and it is used on channel technology where there is no actual phone number being called. |
| 08:29.32 | p3nguin | It isn't a wildcard nor a catch-all, though. Lots of people mistake it for that. |
| 08:29.48 | sawgood | I would have |
| 08:29.52 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 08:29.55 | schmidts | p3nguin its just the default if nothing else match |
| 08:30.04 | p3nguin | sometimes |
| 08:30.14 | sawgood | so, I think I have the answer to why my NoOp calls were breaking the rest of the extension.conf context processes |
| 08:30.32 | p3nguin | Saying "if nothing else matches" makes it seem like a catch-all, which it isn't. |
| 08:30.55 | sawgood | once a call is processed by a single priority 1 match, no other priority 1 matches will touch it (only 2, or n) |
| 08:31.16 | p3nguin | That seems like a fair assessment. |
| 08:31.42 | sawgood | back to the drawing board |
| 08:31.48 | sawgood | oh wait ...one more thing |
| 08:31.50 | p3nguin | Other conditions can be created where you'll end up at 1 again. |
| 08:31.51 | schmidts | p3nguin what exactly do you think of? which extensions |
| 08:32.06 | p3nguin | Goto(), GotoIf(), etc. |
| 08:32.35 | p3nguin | schmidts: I don't understand your question. |
| 08:33.01 | schmidts | p3nguin what extensions do you think of when you mean its not a catch-all |
| 08:33.11 | schmidts | s/mean/say/ |
| 08:33.35 | p3nguin | Extension s does not match simply because no other extension matched. |
| 08:33.37 | sawgood | would this be a valid line: exten => _X.,1,NoOp(Set(Var_TO=${SIP_HEADER(TO)}) |
| 08:33.51 | p3nguin | no |
| 08:34.05 | p3nguin | NoOP() is an application, and Set() is an application. |
| 08:34.30 | sawgood | I simply want the TO field in the header to be shown on the screen |
| 08:34.58 | p3nguin | If you used it as-is, you'd see Set(Var_TO=${SIP_HEADER(TO)} printed in your verbose/debug output if you have it enabled. |
| 08:35.17 | p3nguin | Verbose(${SIP_HEADER(TO)}) |
| 08:35.29 | sawgood | really? |
| 08:35.42 | p3nguin | If ${SIP_HEADER(TO)} parses and contains info, the info will be printed. |
| 08:35.43 | sawgood | Do verbose commands require a priority and all |
| 08:35.56 | p3nguin | Every line requires a priority. |
| 08:36.24 | p3nguin | I merely provided you with the application and the app data. |
| 08:36.35 | sawgood | exten => _X.,1,Verose(${SIP_HEADER(TO)}) |
| 08:36.40 | sawgood | Verbose (sorry) |
| 08:36.47 | kaldemar | sawgood: you don't seem to have a clue on what extensions, contexts and applications are. maybe you should take a while and read some documentation on them. |
| 08:37.02 | greezmunkey | . |
| 08:37.18 | kaldemar | ~thebook |
| 08:37.18 | infobot | thebook is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 08:37.25 | p3nguin | If ${SIP_HEADER(TO)} contains a value and you've matched that extension, Verbose() will print the value on your console. |
| 08:37.36 | sawgood | ty! |
| 08:38.10 | p3nguin | Every application and function on your system will have usage information. core show application Verbose |
| 08:38.13 | greezmunkey | p3nguin: Since you all are on the subject, Does the Verbose app do anything special in the logs? |
| 08:38.43 | p3nguin | I wouldn't expect it to. |
| 08:39.08 | greezmunkey | p3nguin: my thoughts as well, 'cuz I didn't see anything. |
| 08:39.26 | greezmunkey | p3nguin: thanks. |
| 08:39.53 | kaldemar | unless you define asterisk to log verbose messages in logger.conf. |
| 08:39.54 | schmidts | p3nguin http://www.asterisk.org/doxygen/trunk/pbx_8c-source.html#l04783 |
| 08:39.58 | p3nguin | I have been mistaken before, though, so play with the logger setting and see what you get. |
| 08:40.17 | greezmunkey | kaldemar: Ah, ok. I'll look into that. |
| 08:40.28 | schmidts | p3nguin i read this as if nothing else match fall back to s ;) |
| 08:41.13 | p3nguin | Maybe if autofallthrough is enabled, maybe, just maybe, would that occur. |
| 08:41.28 | schmidts | :D i can live with this ;) |
| 08:41.38 | greezmunkey | schmidts: If that were true couldn't the nothing else match case just as easily hit on the i ? |
| 08:42.46 | schmidts | greezmunkey ast_exists_extension would give you i as proper extension if it exists |
| 08:43.23 | greezmunkey | schmidts: there's a logic twist. |
| 08:43.55 | greezmunkey | ok, enough homework for me, good night! |
| 08:44.08 | schmidts | gn8 |
| 08:44.14 | p3nguin | gneight! |
| 08:44.25 | p3nguin | (whatever gneight means) |
| 08:46.58 | schmidts | good night :D |
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| 08:49.30 | sawgood | http://pastebin.com/jcnrc8Nj |
| 08:49.55 | sawgood | So, any call to the (808) number goes through fine, but any call to the (503) number fails |
| 08:50.20 | sawgood | If I comment out the NoOp line, all calls work just fine |
| 08:50.49 | p3nguin | If you call the 503 number, what happens? |
| 08:50.56 | p3nguin | core set verbose 4 |
| 08:50.58 | p3nguin | make the call |
| 08:51.03 | p3nguin | show me the output. |
| 08:51.12 | sawgood | I get output in the console which says, "auto follow through" failed |
| 08:51.50 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 08:51.53 | sawgood | == Auto fallthrough, channel 'SIP/11659-00009c48' status is 'UNKNOWN' |
| 08:52.08 | p3nguin | Do you have sip peers by the names of 4470, 7085, and 7610? |
| 08:52.24 | sawgood | yes, definitely (if the noop is ; out everything works) |
| 08:52.58 | p3nguin | When you set verbose to 4 and made the call, you only got that one single line? |
| 08:53.12 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:53.13 | p3nguin | I'm having a hard time believing there wasn't more. |
| 08:53.17 | sawgood | I have core set verbose at 5 (I was told that is the highest I'd ever need) |
| 08:54.01 | Dovid | anyone know if Asterisk 1.8.X works with SS7 ? |
| 08:54.52 | p3nguin | How could "Set(CALLERID(num)=(808) 854-4470 / ${CALLERID(num)})" do anything at all other than choke/puke? |
| 08:55.04 | sawgood | amazing? |
| 08:55.05 | p3nguin | or possibly die |
| 08:55.22 | sawgood | I wrote that myself (by the way) ... |
| 08:55.27 | p3nguin | I can tell. |
| 08:55.54 | p3nguin | I guess if your phone is okay with it, I'm okay with it. |
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| 08:56.23 | p3nguin | What kind of devices are those sip peers? |
| 08:56.35 | sawgood | yes ... Asterisk boxes |
| 08:56.46 | p3nguin | hmm |
| 08:56.47 | kaldemar | sawgood: see that "dialplan show" looks like. |
| 08:57.41 | sawgood | neat you can break it down by context even! |
| 08:58.01 | kaldemar | you see how the priorities get parsed? |
| 08:58.12 | sawgood | The output is not in 1,2,3,4,5 order though in the dump |
| 08:58.16 | sawgood | 2,3,5,4,1 |
| 08:58.49 | kaldemar | irrelevant. look at the priorities. |
| 08:59.09 | sawgood | I see ... very nice tip! |
| 08:59.23 | kaldemar | you're not supposed to have 1, 2, 3, 4 and 5, but a 1, two 2's and two 3's. |
| 08:59.42 | sawgood | I have 1,2,3,4,5 |
| 08:59.59 | kaldemar | http://pastebin.com/agS29NJD will always work. |
| 09:00.20 | kaldemar | change your extensions conf like that, do a dialplan reload and look at dialplan show again. |
| 09:01.34 | p3nguin | For the sake of being consistent, I would change that last line from n to 2 as well. |
| 09:02.33 | p3nguin | I guess actually it would be required. |
| 09:05.27 | sawgood | I changed everything from n to 2 (and most everything works now) |
| 09:05.29 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 09:05.51 | p3nguin | Line 7 in the pastebin should have remained n. |
| 09:06.08 | sawgood | let me try that ... |
| 09:06.09 | p3nguin | 1, 2, n, 2, 2 |
| 09:06.50 | p3nguin | The reason you have three 2s is because you have three different extensions that branched after priority 1. |
| 09:07.28 | WIMPy | Good morning |
| 09:07.35 | schmidts | morning wimpy |
| 09:07.45 | p3nguin | But the one extension has an additional priority, which could be n or 3. |
| 09:07.46 | Gugge | the CALLERID(num) is pretty strange too ... i would use name for a callerid like that |
| 09:08.01 | sawgood | 100% working 1,2,n,2,2 |
| 09:08.04 | sawgood | very sexy! |
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| 09:11.03 | sawgood | p3nguin: thank you so much (you too kaldemar) |
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| 09:40.06 | kjs | Guys, I am trying to setup iptables on my asterisk box... When the iptables is started the phones fail to register. |
| 09:40.17 | kjs | I have the following two lines: |
| 09:40.22 | kjs | iptables -A INPUT -p udp --dport 5060 -j ACCEPT |
| 09:40.33 | kjs | iptables -A INPUT -p udp --dport 10000:20000 -j ACCEPT |
| 09:41.37 | kjs | default INPUT & FORWARD is set to DROP and the default OUTPUT is set to ACCEPT.. |
| 09:44.22 | kjs | ahh don't worry sorte dit. |
| 09:44.33 | kjs | didn't save the rules |
| 09:44.36 | schmidts | :D |
| 09:44.49 | kjs | puts his n00b hat on and sits in the corner. |
| 09:54.11 | shamelessn00b | joins kjs |
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| 10:13.36 | WIMPy | Does anyone have any pointers on how to get the misdn fork from digium svn to work? |
| 10:15.30 | WIMPy | Or does it have the same restrictions as the original, i.e. no chance on kernels >2.6.25? (or something in that region) |
| 10:20.18 | ectospasm | WIMPy: what card is this again? |
| 10:21.32 | WIMPy | is talking about software, not hardware. |
| 10:22.10 | ectospasm | yes, and I'm trying to figure out what fork you're talking about |
| 10:22.54 | WIMPy | There is a fork of the old misdn1 on svn.digium.com. |
| 10:23.20 | WIMPy | That's needed for full functionality of chan_misdn as included in 1.8.0. |
| 10:23.34 | ectospasm | yeah, Digium maintained one until they released the wcb4xxp |
| 10:23.56 | ectospasm | ...but if you've got a card that depends on mISDN, I don't know what to tell you. |
| 10:24.24 | WIMPy | wcb4xxp doesn't support NT mode on standard HFC-4S/-8S cards. |
| 10:25.04 | WIMPy | But the point is that I was rather disappointed by chan_dahdi so I wanted to try the next option. |
| 10:25.26 | ectospasm | I don't know about anything other than chan_dahdi |
| 10:26.07 | ectospasm | ...and I don't know anything about HFC-4S/-8S cards. |
| 10:26.35 | WIMPy | From what I read in the source of cahn_misdn, it might be very interesting. |
| 10:27.28 | WIMPy | But when trying to biuld it I only get tons of missing or mismatching definitions. |
| 10:28.56 | WIMPy | Just like with the original misdn1, which was the reason I changed to misdn2/lcr/chan_lcr |
| 10:30.32 | ectospasm | yeah, Digium hasn't maintained mISDN in some time, but a developer should be able to give a more definitive answer. |
| 10:30.37 | WIMPy | Even wiki.asterisk.org only describes the situation as it has been three years ago. I can;t find any information on the fork for 1.8. |
| 10:31.32 | EmleyMoor | Is there a good guide to getting fax<->email working with Asterisk 1.6? Everything I've read seems to be 1.4 or even 1.2-centric |
| 10:31.37 | WIMPy | That fork has additions especially for the new featues of Asterisk 1.8, so that's current. |
| 10:31.56 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:32.09 | WIMPy | But I can't find any howto or the like. |
| 10:32.59 | ectospasm | EmleyMoor: you can check out the Free Fax for Asterisk (FFA) stuff, that will work with Asterisk 1.6.2 and 1.8 |
| 10:35.17 | EmleyMoor | Unfortunately hat looks like a way to go from "free, works but is not enough" to "costs" |
| 10:35.42 | ectospasm | EmleyMoor: you get what you pay for, as always... |
| 10:35.47 | EmleyMoor | Hovewer, I will keep it in mind |
| 10:36.20 | shamelessn00b | >.> |
| 10:42.51 | *** join/#asterisk mintos (~mvaliyav@114.143.164.120) |
| 10:42.54 | EmleyMoor | Alternatively, is there a way to do IP-based faxing via hylafax? (T38modem?) |
| 10:43.32 | shamelessn00b | yes >.> |
| 10:44.12 | EmleyMoor | I'm used to hylafax - my old setup, which no longer works, used that and IAXmodem |
| 10:44.42 | shamelessn00b | wonders why people still use fax these days -_- |
| 10:44.54 | ectospasm | shamelessn00b: because lawyers are retarded |
| 10:45.06 | EmleyMoor | shamelessn00b: Mainly, it's my partner who needs it for communicating with medics |
| 10:45.07 | *** part/#asterisk Marquel (~Marquel@84.16.240.149) |
| 10:46.06 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:47.05 | EmleyMoor | shamelessn00b: I'm surprised an IPP-based solution to the problem hasn't taken off |
| 10:47.43 | shamelessn00b | EmleyMoor: totally agree |
| 10:50.07 | ectospasm | it's more difficult to hijack a POTS line, afaict |
| 10:50.28 | Chainsaw | You pretty much need physical access for that, yes. |
| 10:50.50 | EmleyMoor | For what technologies does Asterisk support T38? |
| 10:51.10 | ectospasm | EmleyMoor: FFA and FAX (Fax for Asterisk) support T.38 |
| 10:51.37 | Chainsaw | ectospasm: But on what technologies (line types, channel drivers) can one use them? |
| 10:52.07 | ectospasm | Chainsaw: Only SIP for T.38. IAX2 isn't geared for T.38 IIRC |
| 10:52.32 | ectospasm | EmleyMoor: ^ |
| 10:52.39 | EmleyMoor | I understand "not H323" at present - SIP, yes. Not IAX2? Shame but I can cope with SIP |
| 10:54.18 | ectospasm | EmleyMoor: IAX2 would require low-level protocol changes to allow for T.38 |
| 10:54.30 | ectospasm | ...don't ask me what those would be, though |
| 10:55.33 | Dovid | Does 1.8.X support T.38 as a gateway ? |
| 10:55.47 | ectospasm | Dovid: not with FFA/FAX |
| 10:57.04 | coppice | there are T.38 gateway patches in the bug tracker which many people use successfully |
| 10:58.44 | Chainsaw | You should really see the bug tracker as a prime hunting ground for both bug fixes and feature additions. |
| 10:59.01 | Chainsaw | If you're a distribution packager, I don't believe you can do your job without fishing out a few of those for every package you roll... |
| 11:03.38 | mpe | I have a problem with Local channels that I think has a bug |
| 11:03.39 | mpe | if you use LOCAL_NO_OPTIMIZATION, reinvite dont happen |
| 11:03.39 | mpe | is this by designe or is it a bug |
| 11:05.34 | *** join/#asterisk debuggerboy (~anish@59.98.128.250) |
| 11:05.42 | debuggerboy | hi |
| 11:06.07 | mpe | as the reason for the /n is the second Dial uses som options that otherwise wont work |
| 11:06.17 | debuggerboy | How can I create a variable (extensions.conf) with out of system command? |
| 11:06.36 | debuggerboy | How can I create a variable (extensions.conf) with output of system command? |
| 11:06.50 | ectospasm | debuggerboy: what kind of variable? Does the Set() application not work for you? |
| 11:07.04 | eMBee | good evening |
| 11:07.12 | debuggerboy | nubers only |
| 11:07.12 | *** join/#asterisk glam (~glam@116.24.116.128) |
| 11:07.17 | debuggerboy | numbers |
| 11:07.51 | kaldemar | debuggerboy: core show function SHELL |
| 11:07.59 | mpe | if you use "#exec" to get dialplan for a external program |
| 11:08.55 | eMBee | has a wierd problem. our phone server had 6 dahdi trunks, now we added a 7th one. the 6 trunks work fine, the new one fails to do outgoing calls, asterisk claims it is busy. incoming works |
| 11:09.12 | eMBee | can not figure out what would be different on the new trunk |
| 11:09.13 | debuggerboy | I have a perlscript, Its will give me a number (its a SIP user), I want to save the output of this script as a variable. |
| 11:09.35 | debuggerboy | I am calling the perlscript with System command |
| 11:10.33 | debuggerboy | inside extensions.conf |
| 11:11.51 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 11:13.33 | debuggerboy | I tried to create a varaiable called Set(SIPNUM=System(/var/lib/asterisk/agi-bin/ssipl.pl ${EXTEN}) |
| 11:13.59 | kaldemar | debuggerboy: did you see my previous message? |
| 11:14.34 | kaldemar | debuggerboy: and, applications can't be nested like that. only functions inside applications. |
| 11:15.06 | debuggerboy | No function by that name registered. |
| 11:15.13 | kaldemar | what version of asterisk are you using? |
| 11:15.33 | debuggerboy | 1.4.29 |
| 11:16.17 | kaldemar | func SHELL was introduced with 1.6.0. |
| 11:16.24 | debuggerboy | oh |
| 11:17.52 | debuggerboy | All I need is a variable whose value will be the output of "/var/lib/asterisk/agi-bin/ssipl.pl ${EXTEN}" |
| 11:18.00 | kaldemar | well.. you're options are something like saving the output of the script to a file and read it with app Readfile or something as ugly. |
| 11:19.14 | WIMPy | Use AGI and "set variable". |
| 11:20.36 | debuggerboy | Thanks brothers, let me dig more... thanks for the quick help.. c u |
| 11:24.43 | EmleyMoor | Hmmm... not great |
| 11:28.30 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-144-010.pools.arcor-ip.net) |
| 11:31.21 | debuggerboy | which command is used to store varaiable in AGI script |
| 11:31.57 | WIMPy | Use AGI and "set variable". |
| 11:33.50 | debuggerboy | ok |
| 11:33.56 | debuggerboy | thanks |
| 11:59.05 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 12:01.47 | WIMPy | To answer my question from earlier: Yes the current trung version of misdn1 for Asterisk 1.8 requires a Kernel <= 2.6.25. Bummer. |
| 12:02.16 | *** join/#asterisk fofware (~Fabian@host69.200-117-29.telecom.net.ar) |
| 12:18.53 | *** join/#asterisk coppice (~chatzilla@m121-202-18-242.smartone-vodafone.com) |
| 12:22.41 | Chainsaw | WIMPy: That seems very strange. Does it just need some include love like the DAHDI code? |
| 12:23.38 | Chainsaw | WIMPy: Or is it closed proprietary code with a driver shim, in which case I reserve the right to revert to point & laugh mode. |
| 12:23.51 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 12:27.48 | WIMPy | It needs the old includes. |
| 12:30.06 | WIMPy | This is all very *censored* |
| 12:35.14 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 12:40.00 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:43.44 | Chainsaw | WIMPy: That seems fixable though. Do they have DAHDI-style KERNEL_VERSION defines? |
| 12:47.35 | *** join/#asterisk jsjc (~jsjc@152.red-80-28-172.adsl.static.ccgg.telefonica.net) |
| 12:48.24 | cmnky | gm peeps |
| 12:50.33 | WIMPy | Chainsaw: No idea what DAHDI style looks like, but I guess there isn't much about version at all. That's why I stoped using it a long time ago. It was a contant struggle to get it working with |
| 12:50.38 | WIMPy | different kernels. |
| 12:51.01 | Chainsaw | WIMPy: I just port it to newer kernels as necessary and give my patches to Digium. |
| 12:51.05 | *** join/#asterisk coppice (~chatzilla@m121-202-18-242.smartone-vodafone.com) |
| 12:51.31 | Chainsaw | WIMPy: They tend to apply them reasonably quickly now, but you can always check the bug tracker. |
| 12:52.03 | WIMPy | Not sure it's easy to fix. |
| 12:52.20 | Chainsaw | WIMPy: It's easy to fix if I can break backward compatibility. |
| 12:52.30 | Chainsaw | WIMPy: If not, I need some sort of ifdef that tells me what kernel I'm on. |
| 12:52.41 | cmnky | reading other peoples C code partially liquifies my brain |
| 12:53.29 | cmnky | my C code is beautiful, a work of art, and reads like Beowulf with a mathematics degree |
| 12:53.44 | WIMPy | Hmm. Not sure. There seem to be quite some incompatibilities. |
| 12:54.21 | cmnky | {insert laugh track here} |
| 12:54.31 | *** join/#asterisk ik_5 (~ik@109.226.17.43) |
| 12:54.34 | ik_5 | hello |
| 12:54.36 | WIMPy | It's strange however that it was updated to support the new features of 1.8 but needs a kernel at least 3 years old. |
| 12:54.47 | Chainsaw | WIMPy: Well that tells you a lot about their mindset. |
| 12:55.06 | Chainsaw | WIMPy: And that they are running Red Hat Expensive Linux version 3 or so. |
| 12:55.28 | WIMPy | Yes :-( |
| 12:55.47 | cmnky | gm ik_5 |
| 12:55.55 | WIMPy | But unfortunaletly all of the available options have issues :-( |
| 12:56.06 | Chainsaw | WIMPy: So remind me, why do you need this "mISDN" stuff? |
| 12:56.18 | ik_5 | I need to connect sip trunk from a nextone pbx to asterisk, and asterisk does not like the other side pbx headers and it keep sending invite. does anyone know how i can debug this issue ? |
| 12:56.48 | WIMPy | I don't need it, I want to try it out, as it promises to ultilise 1.8s new features. |
| 12:57.11 | Chainsaw | WIMPy: It also promises that if you don't run RHEL or CentOS, it is never going to work. |
| 12:58.32 | WIMPy | Maybe. |
| 12:58.50 | WIMPy | That just adds to an overall rather unsatisfactory situation. |
| 12:59.13 | kaldemar | ik_5: enable sip debug on the asterisk side and look at the sip trace. |
| 12:59.46 | ik_5 | kaldemar, that's on and so does tshark (that show a bit different information) |
| 13:00.29 | kaldemar | ik_5: pastebin a trace of a failed call then. |
| 13:00.48 | WIMPy | now tries with mixed includes from an old and a new kernel. Do I go to hell for that? |
| 13:00.57 | ik_5 | kaldemar, I'll |
| 13:01.21 | *** join/#asterisk jazzy-J (~jazzy-J@92.249.107.65) |
| 13:04.49 | leifmadsen | sooooooooooooo tired |
| 13:07.48 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:08.30 | ik_5 | kaldemar, the trace : https://gist.github.com/debbccd75c2230a6c27c |
| 13:13.10 | WIMPy | It's running \o/ |
| 13:13.22 | cmnky | hurray |
| 13:14.25 | cmnky | if i wanted to be my own voip provider ... what would i need ... my own T1 and some "im a telephone company license" |
| 13:14.40 | *** join/#asterisk kmd_mlc (~Kd@75.99.197.242) |
| 13:15.32 | Holos | cmnky: just use a larger upstream voip terminator... |
| 13:16.11 | cmnky | Holos, its a curiosity ... not planning on selling phone cards or retailing anything |
| 13:16.57 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.58) |
| 13:17.02 | cmnky | while i was hunting for a voip provider last night ... i realized ... they all seem to pretty much suck in 1 way or another ... yet i still *need* one |
| 13:17.10 | Holos | cmnky: you could get a PRI and have up to 23 concurent channels, but it's cheaper to get an upstream provider. and typically cheaper to terminate over SIP. |
| 13:17.42 | cmnky | how do you terminate over sip to the PTSN |
| 13:17.45 | Holos | cmnky: There are a lot of good ones, you just have to do 100,000 minutes a month to talk to them :) |
| 13:18.14 | cmnky | Holos, which makes them suck ;) |
| 13:20.42 | mandragor | I'm trying to connect one asterisk box to another over sip but I keep gettign a notice that says Failed to authenticate on INVITE to '"1000" I tried everything can someone help me? |
| 13:21.51 | leifmadsen | [OtherServer] |
| 13:21.53 | leifmadsen | type=friend |
| 13:21.59 | leifmadsen | fromuser=MyServer |
| 13:22.03 | leifmadsen | defaultuser=MyServer |
| 13:22.05 | leifmadsen | ... |
| 13:22.08 | leifmadsen | other side |
| 13:22.10 | leifmadsen | <PROTECTED> |
| 13:22.13 | leifmadsen | [MyServer] |
| 13:22.16 | leifmadsen | type=friend |
| 13:22.22 | leifmadsen | fromuser=OtherServer |
| 13:22.23 | kaldemar | pb! ;) |
| 13:22.30 | leifmadsen | defaultuser=OtherServer |
| 13:22.31 | leifmadsen | ... |
| 13:22.33 | leifmadsen | <PROTECTED> |
| 13:22.34 | schmidts | +1 kaldemar :D |
| 13:22.39 | leifmadsen | I'm just typing it, not copy/pasting |
| 13:22.39 | mandragor | let me try that |
| 13:22.50 | leifmadsen | and when you can kick me out of the room, you can tell me what to do :) |
| 13:23.33 | kaldemar | it would take an unreasonable amount of effort. :) |
| 13:23.40 | leifmadsen | agreed :) |
| 13:30.46 | mandragor | that worked thanks leifmadsen |
| 13:30.54 | leifmadsen | tips hat |
| 13:31.04 | leifmadsen | fromuser and defaultuser being the important parts :) |
| 13:31.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:35.09 | cmnky | gm [TK]D-Fender |
| 13:38.25 | *** join/#asterisk fenrus (fenrus@drickheroin.se) |
| 13:39.33 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 13:41.42 | *** join/#asterisk bip (~bip@unaffiliated/bip) |
| 13:42.14 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:42.40 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 13:48.37 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 13:48.37 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 13:49.41 | cmnky | mmmmm coffee |
| 13:50.20 | bip | yeah you cannot be a bofh w/out caffeine ;-) |
| 13:50.48 | bip | or any other sort of nerdish klife form ;-) |
| 13:51.32 | cmnky | i really need a voip provider |
| 13:51.58 | cmnky | its a crazy sea of money and numbers and minutes |
| 13:52.37 | ik_5 | solved it myself |
| 13:56.18 | bip | here the phones keep ringing even after hang up, it s bothering, can this be fixed ? |
| 13:59.01 | beek | unplug the phone. |
| 13:59.09 | beek | s/phone./phones./ |
| 13:59.35 | leifmadsen | cmnky: I like bandwidth.com |
| 13:59.58 | bip | beek: i don t think they will like that solution |
| 14:00.15 | beek | bip: It's the quickest I could come up with. ;) |
| 14:00.16 | *** join/#asterisk Kyosh (whoa@96.246.171.27) |
| 14:00.36 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:00.58 | cmnky | leifmadsen, they want $18 a month for metered or $30 for unlimited |
| 14:01.23 | cmnky | im looking for cheap / free atm ... just so i can make asterisk do something useful |
| 14:01.39 | bip | yes very quick beek :) |
| 14:02.36 | cmnky | waiting for ipcomms to open ... they have a free deal ... but i have to call them to get it ... no web signup |
| 14:03.20 | leifmadsen | cmnky: use google voice |
| 14:03.46 | cmnky | leifmadsen, i started down that road as well ... but so far as i could tell, gtalk is only in asterisk 1.8 ? |
| 14:04.02 | leifmadsen | ya.... |
| 14:04.09 | leifmadsen | if you're just playing around, why aren't you using 1.8? |
| 14:04.12 | cmnky | yeah, thats an issue |
| 14:04.33 | cmnky | im not just "playing around" ... im a startup business ... new to asterisk |
| 14:04.45 | leifmadsen | then you should definitely be using 1.8 |
| 14:04.49 | leifmadsen | it's the next long term support |
| 14:04.59 | leifmadsen | support for everything else goes away early next year |
| 14:05.11 | cmnky | sure ... but i would have to use a kernel with security holes |
| 14:05.18 | cmnky | when they fix the bugs, i'll upgrade |
| 14:05.40 | leifmadsen | what does a kernel have anything to do with it? |
| 14:06.59 | cmnky | my asterisk box ... is not dedicated to only asterisk |
| 14:07.18 | leifmadsen | then you should get another bx |
| 14:07.25 | leifmadsen | especially since they are crazy cheap |
| 14:07.45 | *** join/#asterisk mlkaaa (~chatzilla@193.252.118.23) |
| 14:07.49 | cmnky | i appreciate the advice leifmadsen ... but i dont need another box ;) |
| 14:08.00 | leifmadsen | anyways, it sounds like you're making this a lot harder on yourself than necessar |
| 14:08.01 | mlkaaa | hi |
| 14:08.20 | leifmadsen | ohai |
| 14:08.30 | mlkaaa | i got 2 provider |
| 14:08.36 | cmnky | leifmadsen, call ipcomms, tell them to let me signup over the web, so it wont be so hard |
| 14:08.59 | mlkaaa | where i signed for making incomming outgoing calls |
| 14:09.10 | mlkaaa | where i have to configure my provider ? |
| 14:09.16 | mlkaaa | on my asterisk system ? |
| 14:11.36 | mlkaaa | hello is there anyone here ? |
| 14:11.53 | leifmadsen | cmnky: why would I do that? |
| 14:12.17 | cmnky | leifmadsen, don't worry about it man ... i appreciate the advice |
| 14:12.24 | cmnky | leifmadsen, but im not *that* lost |
| 14:12.32 | leifmadsen | shrugs |
| 14:13.27 | leifmadsen | I still don't get what the kernel has to do with anything since 1.8 would compile against nearly any somewhat recent 2.6 kernel |
| 14:13.27 | neurosys | if nat is bad, should i atleast see the rtp traf going out to the proveder? |
| 14:13.56 | leifmadsen | neurosys: maybe... asterisk might not end it until it sees traffic though |
| 14:14.02 | leifmadsen | s/end/send |
| 14:14.08 | mlkaaa | well i try to rexplain correctly |
| 14:14.15 | cmnky | leifmadsen, because im using, and require, and settled on ... 2.6.25-r12 |
| 14:14.27 | leifmadsen | cmnky: .... ok, so just compile against that |
| 14:14.43 | leifmadsen | in that case, the kernel has no bearing on the inabilty to use 1.8 |
| 14:14.55 | cmnky | people were saying last night that 1.8 doesnt work on that kernel, and that 1.8 has bugs, and is "very bloody" ... in their words |
| 14:15.13 | cmnky | also, 1.8 is currently hard masked in gentoo portage tree |
| 14:15.21 | leifmadsen | I'm deploying a call centre on 1.8 right now, and my production machine has been running 1.8 since RC1 |
| 14:15.30 | leifmadsen | don't use packages to deploy asterisk |
| 14:15.31 | cmnky | and hard masked packages are normally hard masked for good reason ... they dont yet work |
| 14:15.32 | leifmadsen | compile from source |
| 14:15.52 | mlkaaa | i got 2 SIP account with sip provider, where i have to register my provider on my asterisk ? |
| 14:15.54 | leifmadsen | most of the people who say bad things about 1.8 tend to have not tried it |
| 14:15.59 | cmnky | thats great leifmadsen ... but you are an asterisk expert / guru ... and im a noob ... you have superman like abilities with asterisk that i don't yet, and probably wont ever, possess |
| 14:16.00 | WIMPy | has been running 1.8 on two production system for some weeks now. After I disables srtp they are all well. |
| 14:16.02 | leifmadsen | mlkaaa: in sip.conf |
| 14:16.19 | mlkaaa | have you got any example of that ? |
| 14:16.29 | leifmadsen | ~docs |
| 14:16.29 | infobot | Asterisk documentation is available at http://wiki.asterisk.org (Official Asterisk Documentation Wiki), the Voip-Info wiki at http://voip-info.org (~voip-info) or Asterisk: The Future of Telephony (~book) |
| 14:16.33 | leifmadsen | ~book |
| 14:16.33 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 14:16.39 | leifmadsen | it's all in the documentation |
| 14:16.50 | leifmadsen | also look at sip.conf.sample |
| 14:17.24 | mlkaaa | what i mean is : is it same like registering user for registering provider ? |
| 14:17.58 | leifmadsen | you have to register to the provider so it knows where to send a call to. You ALSO need to define the authentication for the incoming call. They are separate things. |
| 14:18.18 | leifmadsen | i.e. register => username:pass@provider |
| 14:18.18 | leifmadsen | and |
| 14:18.18 | leifmadsen | [ITSP] |
| 14:18.18 | leifmadsen | type=friend |
| 14:18.18 | leifmadsen | etc... |
| 14:18.25 | WIMPy | Oh, and don't use chan_misdn either. That segfaults on me as well. |
| 14:20.50 | mlkaaa | i just only want to make outgoing calls |
| 14:20.59 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:21.12 | leifmadsen | then don't bother with the register part |
| 14:21.24 | leifmadsen | you need to authenticate the outgoing call as if you were calling another SIP device |
| 14:21.25 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 14:22.08 | mlkaaa | i just only want to make outgoing calls to PSTN lines |
| 14:23.39 | Katty | when we do it right, gettin slizzard |
| 14:23.50 | mlkaaa | i really need an example to be based on |
| 14:24.01 | Katty | oh |
| 14:24.07 | Katty | when we drink we do it right gettin slizzard. |
| 14:25.15 | WIMPy | Don't you just love it when your phone says "protocol error"? |
| 14:25.51 | Katty | Dear Decemember, go away. Love, Katty. |
| 14:26.00 | Katty | ..december, i mean. |
| 14:27.23 | WIMPy | Thought so. Same bug as in cahn_dahdi. |
| 14:32.20 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 14:32.27 | *** join/#asterisk iratik (~Adium@74-84-99-12.client.mchsi.com) |
| 14:34.14 | iratik | Just a quick PSA: Please install fail2ban if you have an internet facing asterisk box or at least lock incoming sip connections into only known sip trunks/peers ⦠Installed fail2ban last week on a particular asterisk server , since 1 week ago we've had over 20 intrusion attempts from various IPs using "friendly-scanner" |
| 14:35.59 | leifmadsen | iratik: +1 |
| 14:36.42 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
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| 14:56.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:00.52 | Docfxit | What does this mean? WARNING[5543]: chan_dahdi.c:3784 handle_alarms: Detected alarm on channel 1: Red Alarm |
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| 15:30.42 | pabelanger | Docfxit: you have a red alarm on your T1 / E1 |
| 15:31.23 | echinos | DEFCON 2! |
| 15:31.38 | cmnky | shall we play a game |
| 15:31.53 | echinos | haha |
| 15:31.57 | *** join/#asterisk AlexForster (~AlexForst@75.147.73.151) |
| 15:32.16 | echinos | Joshua |
| 15:32.25 | echinos | how about a nice game of chess? |
| 15:32.33 | WIMPy | Global thermonuclear war? Or just trying to interface Asterisk to ISDN? |
| 15:32.46 | AlexForster | quick question- can I use |
| 15:32.46 | AlexForster | requirements = createclose with odbc? |
| 15:33.01 | *** part/#asterisk w____ (~weezr@209.159.158.150) |
| 15:33.02 | AlexForster | i have no idea why that split across two lines, sorry |
| 15:39.37 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 15:41.27 | mishari | hi, I have a problem with my asterisk. quite suddenly today it has stopped responding to any packets |
| 15:41.59 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:42.00 | mishari | tcpdump shows register attempts coming in from my sip client, but no response |
| 15:42.14 | mishari | sip ip debug registers nothing, but my firewall is disabled |
| 15:42.21 | WIMPy | mishari: does the *CLI respond? |
| 15:42.43 | mishari | yep.. cli response.. |
| 15:43.09 | WIMPy | Including the Tab key? |
| 15:43.23 | mishari | yep |
| 15:43.30 | WIMPy | That's usually the danger sign for me when using teh tab key stalls the *CLI. |
| 15:44.13 | mishari | oh, i also swapped ISPs today, but since the asterisk box is internal it shouldn't make a difference, but i thought i'd mention it |
| 15:45.14 | WIMPy | You've got an external IP configured in sip.conf? |
| 15:46.20 | mishari | nope |
| 15:47.22 | WIMPy | Might be time to dig out tcpdump and see how far the packets get. |
| 15:47.32 | mishari | i did.. |
| 15:48.24 | mishari | REGISTER comes in.. and nothing happens after that |
| 15:48.25 | WIMPy | So they do reach the Asterisk box? |
| 15:48.59 | mishari | yep |
| 15:49.48 | mishari | however, it doesn't show up in sip debug |
| 15:50.28 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:51.47 | mishari | oh wait |
| 15:51.52 | mishari | i think it started working again |
| 15:53.39 | *** join/#asterisk sbszulu (~dundubala@41.16.3.163) |
| 15:58.09 | mishari | now that i'm here.. the external ip... can i set it as a hostname? |
| 16:02.21 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
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| 16:23.44 | cmnky | w00t ... dyndns seems to be working ... one step closer to world domination |
| 16:28.49 | *** join/#asterisk nickaugust (~nick@li181-40.members.linode.com) |
| 16:29.27 | cmnky | anyone have any ssl certificate authority recommendations ? |
| 16:30.12 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 16:31.11 | ChannelZ | Anywhere but from those wanks at GoDaddy |
| 16:31.42 | cmnky | godaddy = wankers ... noted ;) |
| 16:31.46 | russellb | cmnky: I think we've been using rapidssl |
| 16:32.23 | ChannelZ | yeah I used one of those resellers.. I can't remember which now. Hmm. |
| 16:32.48 | cmnky | rapid looks alright ... but they want money ;( |
| 16:33.06 | russellb | heh, um, yes, most do |
| 16:33.47 | cmnky | russellb, im being cheap atm ... just to get going |
| 16:33.57 | russellb | to get going, you can use a self signed cert |
| 16:34.17 | cmnky | yeah ... i know |
| 16:38.01 | cmnky | www.startssl.com seems to have a free deal that'll work for now |
| 16:38.45 | cmnky | at least for me, in startup/test mode |
| 16:38.58 | cmnky | wouldnt run a public corporation or gov on it ... but im not that |
| 16:39.25 | Katty | drmessano: you be actin like you're drunk |
| 16:39.59 | cmnky | Katty, i heard a rumour that he is a genius ... maybe he's doing it on purpose |
| 16:40.16 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 16:40.28 | Katty | cmnky: drmessano? a genius? |
| 16:40.38 | Katty | cmnky: if by genius you mean ladies man |
| 16:40.44 | Katty | cmnky: then sure |
| 16:43.34 | cmnky | the little bird didnt define 'genius' |
| 16:46.21 | *** join/#asterisk carrar (~tim@2604:5000:11:1::3) |
| 16:47.37 | Katty | hai carrar |
| 16:47.45 | carrar | hai Katty!! |
| 16:48.09 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 16:48.11 | carrar | How is your HUMP day going? |
| 16:48.15 | thehar | Katty: ! |
| 16:48.16 | thehar | hugs Katty |
| 16:48.20 | Katty | not humpy enough, sadly |
| 16:48.26 | Katty | it's a bit........frustrating |
| 16:48.26 | thehar | lol |
| 16:48.28 | Katty | hugs thehar |
| 16:48.36 | thehar | Katty: !i have an address! |
| 16:48.47 | Katty | excellent |
| 16:48.51 | carrar | What on earth is frustrating you on this day! |
| 16:49.05 | Katty | uhmmm ^_^ |
| 16:49.28 | AlexForster | okay, i'll try another question |
| 16:49.48 | AlexForster | is it possible to have all asterisk servers in a cluster be dundi registration servers? i'm trying to avoid the single point of failure |
| 16:50.08 | Katty | single point of failure? they make pills for that...right? |
| 16:50.28 | carrar | is that the blue pill? |
| 16:50.50 | Katty | carrar: you just keep the bumpers up constantly don't you |
| 16:50.59 | beek | hugs Katty |
| 16:51.00 | Katty | carrar: i need to borrow the bumpers today |
| 16:51.04 | Katty | hugs beek |
| 16:51.12 | carrar | I do, it's a crazy boat ride, you never know what you are gonna run into!! |
| 16:51.32 | carrar | keep those bumpers on!! |
| 16:51.34 | Katty | thehar: you get it tho, don't you |
| 16:51.56 | Katty | thehar: cause you just coo like that |
| 16:52.11 | thehar | rawr |
| 16:52.35 | Katty | furreals. |
| 16:52.51 | thehar | i need to find the card site again |
| 16:52.53 | thehar | so i can update addy |
| 16:53.03 | *** part/#asterisk AlexForster (~AlexForst@75.147.73.151) |
| 16:55.02 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 16:55.19 | Katty | i'll update your addy in a min |
| 16:57.21 | carrar | Are you sending out Christmas cards? |
| 16:58.06 | Katty | yes'r |
| 16:58.21 | Katty | and if anyone is interested in being on the list please /query me |
| 16:58.34 | Katty | cutoff for shipping is around dec 10 or 15th |
| 16:58.50 | carrar | shipping? Are sending everyone a free iPad? |
| 16:59.04 | Katty | no silly, cards. |
| 16:59.08 | Katty | the kind you write on, and put in the mail |
| 16:59.13 | carrar | Why not iPads? |
| 16:59.16 | Katty | possibly with glitter |
| 16:59.49 | Katty | i'll give you an ipad |
| 16:59.57 | Katty | oh wait, syntax. |
| 17:00.01 | Katty | i'll give YOU an ipad. |
| 17:00.01 | carrar | heh |
| 17:01.22 | carrar | waits for his free iPad from Katty |
| 17:01.39 | thehar | lol |
| 17:01.55 | carrar | with glitter |
| 17:02.14 | Katty | that'd be messy. |
| 17:02.32 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 17:02.40 | thehar | aaaanddd updated |
| 17:04.23 | *** join/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 17:04.40 | *** join/#asterisk gucki (~gucki@134.Red-83-35-209.dynamicIP.rima-tde.net) |
| 17:04.59 | gucki | hi. i just upgraded from 1.4 to 1.6 but now sip isnt working anymore :( |
| 17:05.40 | gucki | when i go "asterisk -r" and do "sip show peers" it tells me "No such command 'sip show peers' (type 'core show help sip show' for other possible commands)" |
| 17:08.01 | gucki | anybody alive? ;) |
| 17:08.15 | thehar | heh |
| 17:08.20 | thehar | do you have chan_sip loaded? |
| 17:10.47 | gucki | thehar: do i have to put this into modules.conf? |
| 17:10.55 | thehar | it should just be loaded |
| 17:11.35 | thehar | unless you've set noload => chan_sip.so |
| 17:12.02 | gucki | i did "module load chan_sip" and now it seems to work |
| 17:12.09 | gucki | but i wonder why it doesn't get autoloaded |
| 17:12.15 | gucki | i didn't put noload anywhere.. |
| 17:12.25 | thehar | add it in modules.conf to preload |
| 17:12.27 | *** part/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 17:17.21 | gucki | thehar: ok, seems the modules.conf was missing completely. i created it and added "autoload=>true", now it seems to work :). tnx! |
| 17:20.26 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:24.55 | gucki | thehar: ok, sip works now but i cannot dial. i get 'use extension not found in context 'gucki-in''...do i have to load some more modules? :) |
| 17:25.20 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:26.16 | *** join/#asterisk trumee (~trumee@cpc5-cmbg14-0-0-cust982.5-4.cable.virginmedia.com) |
| 17:27.54 | trumee | Guys, i have installed asterisk1.6 in Ubuntu Server 10.10. I am unable to start asterisk unless i specify noload => app_voicemail.so in /etc/asterisk/modules.conf. But then how do i get voicemail running? |
| 17:28.09 | Qwell | trumee: Uninstall the other voicemail modules you installed. |
| 17:28.28 | trumee | Qwell: i also have noload => app_voicemail_odbc.so and noload => app_voicemail_imap.so |
| 17:28.42 | Qwell | then you can load app_voicemail.so without problem |
| 17:29.30 | trumee | Qwell: ok, you mean in cli i do, module load app_voicemail.so ? |
| 17:29.40 | Qwell | No, I mean don't noload it |
| 17:30.05 | trumee | ah i see. |
| 17:32.18 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
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| 17:36.14 | gucki | mh, this upgrade really sucks |
| 17:36.22 | trumee | Qwell: unfortunately, voicemail is hanging up on me. http://pastebin.com/cWUiH6dR |
| 17:36.27 | gucki | no with autoload asterisk tells to it's going to 204 modules will be loaded. |
| 17:36.36 | gucki | why doesn't it only load what it needs? |
| 17:36.44 | gucki | i only want simple sip + voicemail... |
| 17:39.58 | yonahw | I have a user that initiated a chanspy via FOP. They hungup and the channel is still active for 15 minutes now. I have tried sending a hangup request via CLI to no avail. any ideas? |
| 17:43.19 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
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| 17:58.00 | *** join/#asterisk muddd (superstraw@gateway/shell/bshellz.net/x-uunbjqrscronitwo) |
| 17:59.56 | muddd | is asterisk able to be installed between a key unit and it's phones for recording all telephone conversations in an office? |
| 18:00.10 | muddd | the lines are digital, coming from a norstar MICS |
| 18:01.02 | Qwell | Not like that, no. Asterisk doesn't support "digital" phones. It's not too difficult to create a route that would force it to go through Asterisk though. |
| 18:02.43 | muddd | Hmm, ok then would it be possible to equip all desks with voip phones that go through to the asterisk machine, then out to our analog connection leading to the t1's we have? |
| 18:02.57 | muddd | I'll explain more about the t1's analog setup: |
| 18:03.48 | muddd | 2 t1's coming in, converted to analog by adtran routers, which lead to 2 66 punchblocks. I know I can get a t1 card for asterisk, but would analog outputs be cheaper? |
| 18:03.52 | leifmadsen | so I'm trying to think of a good way of injecting some audio to a caller who is speaking with an agent. The caller should have the ability to press '#' that would trigger a webservice to be called via CURL() and then a prompt played back to them via Playback, SayDigits, or something. Anyone have ideas how I should approach that? |
| 18:04.24 | WIMPy | muddd: No you definitely want to tap the T1s. |
| 18:04.28 | leifmadsen | I was thinking features.conf, but of course I can't do anything like Macro() or GoSub() with it, so I'm wondering if I could somehow call an AGI() to do all the logic, passing it the channel name perhaps. How might I inject audio to the channel? |
| 18:04.46 | *** join/#asterisk SparFux (~raoul@g231006074.adsl.alicedsl.de) |
| 18:06.11 | Katty | hai |
| 18:06.23 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 18:07.03 | Qwell | leifmadsen: confbridge? |
| 18:07.07 | muddd | all the voip phones I've dealt with have been with 8x8 and vocalocity. But I can have my own voip phones on my own internal network, no subscribing to outside providers, and put the calls out using our t1 then? |
| 18:07.31 | leifmadsen | Qwell: well the call is going to an Agent channel so the channel will already be active and I was hoping to avoid just putting people into a ConfBridge() right away... |
| 18:07.39 | leifmadsen | although that might be the only way to do it.... |
| 18:07.58 | leifmadsen | I'd prefer to have something where maybe I attach via ChanSpy() and whisper... |
| 18:08.08 | *** part/#asterisk SparFux (~raoul@g231006074.adsl.alicedsl.de) |
| 18:08.29 | WIMPy | muddd: If you want to replace all phones... There are lots of possibilities. |
| 18:15.45 | muddd | we have these phones that work with our norstar, and also have a couple boxes of older phones: tel-plus pc-1648. are those analog? |
| 18:16.05 | muddd | they went to our last pbx/keysystem whatever it was. a dinosaur |
| 18:16.43 | muddd | i want voip phones and cable modems, but the boss is stuck on t1's and key systems |
| 18:17.35 | Katty | ohaithar |
| 18:17.57 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 18:22.05 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 18:24.38 | nny | quick question. I have a dialplan that runs 4 or 5 companies through it. Overall it's "one company, many faces" I had in the past been using unique contexts for each IVR, etc, but starting to think variabilized gosubs for each uniqe aspect (time conditions, set CID, set incoming CID name to signify which it is) maybe cleaner.. anyone else handle this differently? |
| 18:25.36 | yonahw | how can I figure out what is holding up a channel from being dropped. Noone is actively on the channel and all requests to hangup are ignored. |
| 18:25.39 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.128.6) |
| 18:26.08 | leifmadsen | yonahw: could be a deadlock, in which case you could do "core show locks" if DEBUG_THREADS was compiled in |
| 18:29.19 | yonahw | leifmadsen: must not have DEBUG_THREADS compiled in. I get no such command core show locks |
| 18:29.35 | leifmadsen | aye |
| 18:29.46 | leifmadsen | which wouldn't help you much other than to determine if a lock was really be held |
| 18:29.54 | leifmadsen | probably have to restart asterisk to clear it |
| 18:30.05 | yonahw | leifmadsen: thanks will do |
| 18:30.13 | nny | some examples of what I asked would be using an initial variable in my default inbound context that then defines how other variables are set later. (COMPANY=JOES) (later set CID${COMPANY${JOES}} Not sure if you can GoSub(timecheck${JOES},s,1) or the like. Excuse any formatting errors, just trying to convey the notion |
| 18:31.01 | nny | right now I just have contextcompanyone subcontextcompanyone etc. Lot's of repetitive routines each crafted for the company using it |
| 18:31.20 | nny | i guess i'll have to just play with it and see what pitfalls or obstacles I hit |
| 18:31.33 | yonahw | leifmadsen: will core restart when convenient work or will it just wait forever for that deadlocked channel? |
| 18:31.49 | leifmadsen | not sure, I don't like using those commands. I prefer to monitor manually :) |
| 18:31.58 | leifmadsen | I've had it restart on me when I didn't want it to |
| 18:32.09 | yonahw | oh, good to know |
| 18:34.20 | p3nguin | If you used restart gracefully, it would prevent new calls, so you'd eventually either be left with that one stuck channel or it would restart when all others were gone. |
| 18:34.54 | p3nguin | If you watch it, you'd see the channels decreasing as people end their calls. |
| 18:34.57 | WIMPy | bet's you couldn't even kill it without -9. |
| 18:35.14 | citywok | i use restart now, restart gracefully waits for hung calls, and sometimes i end up with a SIP call that gets orphaned for some reason and sits open forever |
| 18:35.17 | WIMPy | -' |
| 18:35.17 | p3nguin | When it's down to just 1, assume it is hung on your stuck channel. |
| 18:35.46 | Katty | hugs leifmadsen |
| 18:35.52 | citywok | so at midnight every night i have a cron restart now. it's helped cut down on the asterisk crashes. |
| 18:36.47 | p3nguin | I'd probably move it to 3 or 4 am, since I'm likely to be using the phone at midnight. |
| 18:37.01 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:37.46 | WIMPy | uses core stop when convenient. Then when it won't go after some time, I throw a kill -9 at it. |
| 18:38.02 | *** join/#asterisk digilink (~digilink@70.90.228.194) |
| 18:38.51 | citywok | yea, we're idle from 10PM to 5AM so i can put it anywhere in there. |
| 18:44.06 | p3nguin | tries to figure out how to increase the top speed of the Wheego all electric car. |
| 18:46.05 | nny | p3nguin: hemi |
| 18:46.12 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 18:46.16 | nny | p3nguin: and a deathwish |
| 18:46.53 | nny | although on a serious note I bet it has a speed governor on it |
| 18:48.02 | *** join/#asterisk trumee (~trumee@cpc5-cmbg14-0-0-cust982.5-4.cable.virginmedia.com) |
| 18:48.16 | nny | p3nguin: looks like it does, 35 mph. So you just need to figure out how to tell it to ignore that feature :D |
| 18:48.43 | p3nguin | Top speed on the Wheego LiFe is 70 MPH. |
| 18:50.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:50.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:50.49 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
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| 18:53.28 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 18:54.11 | nny | p3nguin: think this was for the Whip |
| 18:55.43 | p3nguin | whip life |
| 18:57.36 | kerx | hi all, i'm having major difficulty with my Polycom IP phone, it's not getting the time settings from the NTP server plugged into the configuration (pool.ntp.org). |
| 18:58.03 | kerx | it was working fine, until i rebooted my phone, now it's stuck on a date 30 day's ahead |
| 18:59.52 | Qwell | 1/1/11? |
| 19:05.22 | Katty | ohai |
| 19:05.23 | Katty | mister Qwell |
| 19:05.32 | Katty | Qwell: we haven't hugged in days |
| 19:05.36 | Katty | Qwell: i think i'm having withdrawl |
| 19:12.02 | Qwell | I didn't do it |
| 19:16.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 19:17.42 | *** join/#asterisk angav (~angav@189.251.20.126) |
| 19:18.06 | *** join/#asterisk mlsmith9999 (~store@adsl-76-240-11-130.dsl.ltrkar.sbcglobal.net) |
| 19:18.10 | angav | Hi, just want to know if anyone can advise me on a general VOIP channel, thanks |
| 19:19.24 | mlsmith9999 | afternoon. I have to reinstall after my trixbox CE install got hacked, what is the best asterisk package for beginners? I like have CLI access but still need a gui to configure things correctly. and it has to go fast, it's a production machine. |
| 19:19.36 | mlsmith9999 | (For a small company, mine) |
| 19:19.40 | russellb | AsteriskNOW |
| 19:19.49 | mlsmith9999 | why? |
| 19:19.59 | russellb | because it's provided by the same team that is charge of Asterisk itself |
| 19:20.10 | russellb | and the Asterisk packages are kept up to date with Asterisk releases |
| 19:20.21 | mlsmith9999 | hmmm ok, nice interface? |
| 19:20.41 | russellb | depends what you mean, if you mean the GUI for Asterisk, it includes FreePBX by default, the Asterisk-GUI is also an option |
| 19:20.44 | russellb | (or no GUI at all) |
| 19:21.13 | mlsmith9999 | ok, which of those GUI's would you recommend to a rookie. |
| 19:21.30 | russellb | FreePBX is most popular. |
| 19:21.35 | Katty | you're most popular. |
| 19:21.38 | russellb | <3 |
| 19:21.40 | Katty | yeah, i said. |
| 19:21.41 | thehar | yes he is |
| 19:21.43 | Katty | it |
| 19:21.48 | thehar | i think Katty is more popular tho |
| 19:21.53 | russellb | thehar: i agree |
| 19:22.03 | thehar | she is pretty awesome too |
| 19:22.18 | mlsmith9999 | Why is it most popular? |
| 19:22.47 | russellb | i dunno, it just is? It's the most featureful of the open source GUIs |
| 19:22.51 | Katty | because all the nice people are |
| 19:23.36 | mlsmith9999 | ok.. I'll look closer at it then. |
| 19:23.47 | russellb | If you used Trixbox, that's the GUI you were using. |
| 19:23.52 | russellb | a hacked up version of it anyway |
| 19:24.49 | leifmadsen | darn, it seems I can't pass or forward channel variables to features.conf |
| 19:24.55 | leifmadsen | (or to a feature triggered via features.conf) |
| 19:24.59 | thehar | fail |
| 19:25.07 | leifmadsen | indeed |
| 19:25.12 | leifmadsen | now how do I get around this.... |
| 19:25.16 | Qwell | func_shared? |
| 19:25.22 | leifmadsen | hmmm, looking! |
| 19:25.29 | Qwell | or whatever the heck its called |
| 19:25.37 | russellb | I don't think the args are evaluated at all |
| 19:25.39 | russellb | that's the problem |
| 19:25.45 | russellb | good feature request right there |
| 19:26.00 | leifmadsen | Qwell: SHARED() |
| 19:26.06 | leifmadsen | yes, this may work... trying! |
| 19:26.10 | leifmadsen | thanks for the tip! |
| 19:26.14 | leifmadsen | russellb: correct :) |
| 19:26.16 | Katty | Qwell: :< |
| 19:26.19 | Katty | Qwell: no, you didn't do it |
| 19:26.21 | Qwell | Katty: :( |
| 19:26.23 | Katty | Qwell: what's the dealio with that |
| 19:26.38 | Katty | Qwell: ARE YOU RATIONING MY HUGS SIR |
| 19:26.42 | russellb | leifmadsen: issue tracker the request, we should totally do that |
| 19:26.44 | mlsmith9999 | one more question, what kernel version is Asterisk NOW 1.7 using? |
| 19:26.45 | Qwell | yes, they might run out. |
| 19:26.46 | russellb | that's borderline a bug TBH |
| 19:26.52 | Qwell | mlsmith9999: Latest CentOS 5.5 kernel |
| 19:26.54 | leifmadsen | russellb: ok! I'll try and get that filed |
| 19:27.09 | mlsmith9999 | Sorry, I mean Asterisk Core. |
| 19:27.17 | mlsmith9999 | 2.8? |
| 19:27.26 | leifmadsen | 1.8? |
| 19:27.35 | nny | SOLD! |
| 19:27.38 | mlsmith9999 | or that.. |
| 19:27.43 | Qwell | mlsmith9999: 1.4, 1.6.2, or 1.8 |
| 19:27.46 | nny | hands mlsmith9999 his fish |
| 19:27.52 | mlsmith9999 | hehe |
| 19:28.01 | nny | :) |
| 19:28.16 | Katty | cries. |
| 19:28.31 | mlsmith9999 | So AsteriskNow 1.7 runs Asterisk 1.8? |
| 19:28.42 | mlsmith9999 | sorry this version crap can get confusing.. |
| 19:29.48 | nny | is reminded of the scene from Spaceballs |
| 19:30.13 | mlsmith9999 | please don't feel stupid enough as it is.. :P |
| 19:30.22 | nny | nah it's cool just making humor |
| 19:30.55 | *** part/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 19:31.13 | nny | mlsmith9999: i am looking for a page that might have a changelog to help you in the future, not sure if it exists :D |
| 19:31.28 | mlsmith9999 | Yeah I know. So am I correct in that *NOW 1.7 runs * 1.8 |
| 19:31.32 | nny | mlsmith9999: http://www.asterisk.org/AsteriskNOW-1.5-QuickStart |
| 19:31.44 | nny | mlsmith9999: nm |
| 19:31.49 | nny | mlsmith9999: that only shows 1.6 |
| 19:32.11 | mlsmith9999 | nny: :) you see the reason for my question... |
| 19:32.44 | nny | mlsmith9999: Qwell: mlsmith9999: 1.4, 1.6.2, or 1.8 |
| 19:33.00 | nny | Qwell is usually 110% correct, which defies mathematics |
| 19:33.05 | p3nguin | In the amount of time you've spent asking you could have installed it and found out what it offers. |
| 19:33.10 | mlsmith9999 | I see what I can use it for, how to install it, but no spec's, where my sales slick with the spec's.. |
| 19:33.30 | nny | mlsmith9999: probably cause it's an OS and specifications would be too numerous to list |
| 19:34.04 | nny | php version X, kernel version Y.. etc you can see how you can't cherry pick it. I agree it should say what asterisk versions it uses somewhere, maybe worth a suggestion to Digium |
| 19:34.33 | mlsmith9999 | p3nguin: nice.. that's helpful. If I have a problem figuring that out, then I'm geussing there are more, and my pointing it out might help inspire somebody to make it easier to figure it out on the webpage. |
| 19:34.53 | nny | you can also roll your own using Freepbx directly and insert distro X here |
| 19:35.10 | mlsmith9999 | nny: yep.. I'll send them an email. In the meantime I'll follow Penguin's advice. |
| 19:35.17 | leifmadsen | Qwell: ya SHARED() looks like it might not work because the caller could press # at any time, so SHARED() would always be changing and likely not contain the value I want |
| 19:35.25 | leifmadsen | (or expect) |
| 19:35.28 | mlsmith9999 | nny: yeah and add to my problems :) |
| 19:35.44 | leifmadsen | thanks for the idea though, but I can't see how I can make it work :\ |
| 19:35.56 | mlsmith9999 | bbl |
| 19:35.58 | nny | mlsmith9999: well, teach a man to fish etc. I started with Asterisknow 4 years ago or so, and prefer my own rolled vanilla asterisk installs now. |
| 19:36.00 | nny | mlsmith9999: gl |
| 19:36.12 | mlsmith9999 | nny: thanks.. |
| 19:36.56 | *** part/#asterisk angav (~angav@189.251.20.126) |
| 19:37.13 | leifmadsen | hmmm, I don't quite get how SHARED() is different from GLOBAL() :) |
| 19:39.11 | nny | leifmadsen: not sure if it helps, but I have a cellphone transfer feature that uses channel variables as well as ones I set a macro itself. I can't call the variables from the command in features.conf, but I can call the Macro. I know macros are old news, but it works well |
| 19:39.53 | WIMPy | leifmadsen: I guess it's the parameter. Looks like you can create namespaces as you like. |
| 19:39.56 | leifmadsen | hmmm, the note says you can't use Macro() from features.conf.... which is why I didn't try it :) |
| 19:39.58 | nny | leifmadsen: see http://pastebin.com/GdYZ21iK |
| 19:40.31 | nny | leifmadsen: this is Asterisk 1.6.2.6 fwiw |
| 19:40.34 | leifmadsen | calling a Macro() would actually makes this pretty much trivial |
| 19:40.37 | leifmadsen | ya using 1.8 here |
| 19:40.41 | leifmadsen | maybe I should have just tried it :) |
| 19:40.46 | nny | hehe |
| 19:40.46 | leifmadsen | maybe that note is just old news |
| 19:42.28 | leifmadsen | nny: if you're using 1.6.2 I think Originate() exists, which would avoid you having to connect to the Asterisk console |
| 19:42.31 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 19:45.39 | Corydon76-home | nny: not really. The namespaces are attached to the channel |
| 19:46.18 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 19:46.34 | nny | leifmadsen: yeah I run it that way so if it fails it's not picked up nothing happens |
| 19:46.44 | nny | er |
| 19:46.47 | nny | if it's not picked up* |
| 19:46.51 | nny | parse fail |
| 19:47.12 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.141.242.dsl.dyn.forthnet.gr) |
| 19:47.20 | nny | the call rings the cell phone, but doesn't actually transfer until I answer (ala Google Voice) |
| 19:47.24 | Corydon76-home | When the channel goes away, the related SHARED() space also goes away |
| 19:47.37 | leifmadsen | Corydon76-home: I don't get what the 2nd argument to SHARED() is for |
| 19:48.03 | Corydon76-home | leifmadsen: for the channel name for which you want to access the SHARED space for |
| 19:48.19 | nny | Corydon76-home: yeah I hear you, I was just basing it on his original query "(2:23:37 PM) leifmadsen: darn, it seems I can't pass or forward channel variables to features.conf" |
| 19:48.25 | leifmadsen | ah, so basically get the value inside the shared space for the channel I request |
| 19:48.26 | WIMPy | Hmm, but if they go away with the channel that created them, how are they different from _variables? |
| 19:48.48 | leifmadsen | WIMPy: you can read them from other channels |
| 19:48.52 | Corydon76-home | leifmadsen: get AND set |
| 19:49.04 | leifmadsen | gotcha, ok starting to become more clear now |
| 19:49.10 | Corydon76-home | You can already ready arbitrary variables from other channels with IMPORT() |
| 19:49.26 | WIMPy | Ok, so you cann pass them back 'upwards'. |
| 19:49.27 | nny | interesting, good to know |
| 19:50.03 | Corydon76-home | leifmadsen: basically, the idea is to allow a channel's normal channel variables to be inviolable in the dialplan |
| 19:50.05 | WIMPy | This variable thing can be quite confusing. |
| 19:50.07 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 19:50.22 | leifmadsen | heck ya :) |
| 19:50.54 | Corydon76-home | and then allow a channel to export certain ones that it will accept as allowed to be changed from elsewhere |
| 19:51.09 | nny | Corydon76-home: I just realized based on your username we may or may not be the same age :D |
| 19:51.47 | Corydon76-home | nny: You're 1,934 years old, too? |
| 19:51.51 | WIMPy | Variables with permissions. I think I need to get used to that. |
| 19:52.10 | Corydon76-home | WIMPy: think public/private |
| 19:52.14 | nny | Corydon76-home: oh nm, older |
| 19:53.46 | *** join/#asterisk bneff (~bneff@12.104.144.2) |
| 20:03.26 | leifmadsen | got it! |
| 20:03.40 | leifmadsen | just had to use the ol trick of passing data via the extension |
| 20:03.49 | leifmadsen | _spy-.,1,Set(...) |
| 20:10.01 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 20:11.48 | citywok | leifmadsen: yea i love doing that. are you originating via features and sending it to the context that way? |
| 20:12.42 | leifmadsen | citywok: 2008002 |
| 20:12.44 | leifmadsen | oops |
| 20:12.49 | leifmadsen | citywok: http://pastebin.ca/2008002 |
| 20:12.51 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:13.27 | leifmadsen | citywok: the URIENCODE() is because the channel name has a / in it |
| 20:13.36 | leifmadsen | URIENCODE() is my new favourite function this week |
| 20:14.40 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 20:15.29 | citywok | you have to escape the / even though it's coming through a variable? |
| 20:16.17 | *** join/#asterisk Tim_Toady (~moi@178.128.21.207.dsl.dyn.forthnet.gr) |
| 20:17.14 | leifmadsen | I'm passing it via an Originate() |
| 20:17.20 | leifmadsen | which is going to be like passing a / to Dial() |
| 20:17.40 | leifmadsen | Local/spy-SIP/9999@.... |
| 20:17.53 | leifmadsen | so I use URIENCODE() to hide the / from Originate() |
| 20:17.59 | leifmadsen | tune URIDECODE() to get ti back later |
| 20:18.10 | leifmadsen | s/tune/then |
| 20:18.10 | citywok | oh okay i get it |
| 20:18.28 | citywok | i've never had to do anything like that |
| 20:19.11 | yonahw | I have calls which get answered in thrown into an IVR. Caller can dial extensions of users which will in turn call Dial(SIP/${EXTEN},25,wW). Everything works as expected except caller does not hear ringing. I have tried adding 'r' no help. 'w' will play music however. How can I make sure caller hears ringing? |
| 20:19.16 | leifmadsen | citywok: this was my first time :) |
| 20:19.36 | leifmadsen | the other reason I used it this week was for passing data to a webservice that had spaces in it |
| 20:20.01 | leifmadsen | yonahw: try Progress() first? |
| 20:20.43 | *** join/#asterisk Dances (~Alex@pool-74-102-158-194.nwrknj.east.verizon.net) |
| 20:21.47 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
| 20:22.47 | Dances | This is a really stupid question but not sure who else to ask. I plugged the phone line that went to my office phone into my computer and I cant make calls fine using windows Dialer.exe program. Is there a way to dial an internal extension here from teh Dialer.exe program or do I need something more complicated? I tried just typing the extension in directly and dialing that but nothing happens. Any ideas? |
| 20:28.19 | yonahw | leifmadsen: thanks, unfamiliar with it, will look it up and try |
| 20:28.27 | marl_scot | Dances, not really an asterisk thing, but if you could dial an extension from your office phone, then i would have thought you would be able to dial from the computer (presuming that you could dial external numbers from the computer, and it was using your office line, and not another modem on the pc) |
| 20:29.19 | citywok | Dances: is your office phone a digital phone or an analog phone? It may be a special type of phone or something. |
| 20:29.38 | citywok | Our old inter-tel phones you couldn't have just plugged the jack in to your ocmputer and had it work |
| 20:30.16 | Dances | hmm its a pretty new Avaya phone |
| 20:30.20 | Dances | so im guessing digital |
| 20:30.33 | citywok | most likely |
| 20:30.56 | citywok | you need an analog phone jack to plug in to your PC, you may be able to connect one to your PBX if you have those jacks. |
| 20:31.19 | _Corey_ | We had a customer's IT vendor plug in a bunch of POTS lines to their ethernet switch once and wonder why they couldn't use them to dial out via Asterisk if that makes you feel any better. |
| 20:31.27 | Dances | hmmm |
| 20:31.37 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:31.45 | citywok | _Corey_: time for a new vendor |
| 20:32.17 | _Corey_ | ;) we really couldn't believe it |
| 20:32.23 | Dances | i dont know shit about phone systems so i guess ill have to look into this deeper, maybe ask my phone guy |
| 20:32.30 | Dances | thanks anyway |
| 20:33.24 | citywok | Dances: good luck |
| 20:33.39 | citywok | i'm guessing you aren't an asterisk user then? |
| 20:33.58 | yonahw | leifmadsen: tried progress as well as ringing with no luck |
| 20:34.00 | *** join/#asterisk diemos (~diemos@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 20:34.03 | diemos | herroz |
| 20:34.08 | yonahw | any other ideas? |
| 20:34.09 | leifmadsen | not sure |
| 20:34.16 | leifmadsen | usually it "just works" for me |
| 20:34.18 | diemos | I'm wanting to connect a remote VoIP phone to a PBX behind a NAT'd firewall without using VPN or SSH. Any ideas? |
| 20:34.29 | Dances | citywork: nah but i i know u guys know alot about phones thats why i asked here |
| 20:34.30 | citywok | diemos: forward the ports |
| 20:34.39 | leifmadsen | diemos: and use externip / localnet |
| 20:34.41 | leifmadsen | in sip.conf |
| 20:35.04 | diemos | I don't want to open the ports, I've been getting slammed with bruteforce attacks -.- |
| 20:35.12 | yonahw | "just works" is kind of what I would expect |
| 20:35.14 | WIMPy | diemos: you can't use ssh. VPN is possible. |
| 20:35.45 | diemos | because it's eating up bandwidth on the remote site and slowing thier system down for one |
| 20:36.13 | diemos | for two i don't have a phone with built in VPN support |
| 20:36.15 | leifmadsen | diemos: use iptables and just open the ports for the remote IP addresses |
| 20:36.24 | diemos | ah |
| 20:36.26 | leifmadsen | also use fail2ban |
| 20:36.37 | citywok | the problem is you're still receiving the traffic so by opening the ports you aren't really adding much bandwidth |
| 20:36.39 | diemos | wow i overlooked the obvious |
| 20:36.56 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 20:37.10 | citywok | i banned some 50gb of traffic over the course of a week from one host that was brute forcing thousands of attempts per second |
| 20:37.31 | diemos | yeah, that's what someone was doing to me, he got in and actually made some international calls |
| 20:37.35 | diemos | so i'm weary now |
| 20:37.47 | citywok | even though i had firewalled the traffic the retard still racked up that much. i ended up having my ISP drop the traffic for me. |
| 20:38.09 | citywok | and as leifmadsen use fail2ban with aggressive limits, it will protect you against brute forcing |
| 20:38.24 | citywok | a couple attempts in a minute ban for an hour or a month or something |
| 20:38.36 | diemos | hm I hadn't heard of fail2ban |
| 20:38.36 | leifmadsen | citywok: heh, need to cut off some digits on the "cents": "Total cost of your call is, 11, dollars, 70 million, 234 thousand, cents" |
| 20:38.51 | diemos | lol |
| 20:38.54 | citywok | hahahaha |
| 20:39.08 | citywok | that's a big bill! |
| 20:39.12 | leifmadsen | only 11 dollars! |
| 20:39.20 | leifmadsen | (and 70 million cents) |
| 20:39.23 | citywok | 70 million 234 thousand cents is a LOT |
| 20:39.27 | leifmadsen | :D |
| 20:39.38 | citywok | shift the decimal a couple places.. |
| 20:39.44 | citywok | crap! .7 million dollars! EEK |
| 20:39.57 | citywok | that's an expensvie 900# |
| 20:40.11 | citywok | what is the business use for this? lol |
| 20:41.12 | Qwell | time machine. gives you the lottery numbers from 3 days ago. |
| 20:41.35 | diemos | so, what if you have remote users whose IP address changes frequently? |
| 20:41.36 | WIMPy | This call costa 1$ and 999¢ per minute. |
| 20:42.13 | citywok | diemos: personally i would use the XML browser in my phone to hit a webpage on port 80, which would then open the requesting IP address to 5060 |
| 20:42.36 | leifmadsen | citywok: clever -- we have a client who makes people check their email which triggers a script |
| 20:42.45 | leifmadsen | they login to IMAP, then IP is opened for them |
| 20:43.03 | diemos | hm, i'll do some reading up and get back to you guys. You're saviours haha. |
| 20:43.03 | citywok | leifmadsen: my aastra phones have a polling itnerval of 5 minutes where they auto grab a script on my server. i don't actually lock down by IP like this but if i wanted to. that's how i'd do it. |
| 20:43.10 | Qwell | leifmadsen: nifty |
| 20:43.22 | leifmadsen | citywok: indeed -- polycom could do the same thing too |
| 20:43.58 | citywok | yea, that'd be my suggestion for the super secure route. i've considered doing it, but fail2ban works pretty well for me. |
| 20:46.21 | citywok | Hmm. I have a non profit with 25 extensions that wants a phone system. I'd like to install something with a GUI so i can train their admin on how to update extensions, but haven't used anything other than plain ol' vanilla *. suggestions? * Now, trixbox? |
| 20:46.48 | malcolmd | doesn't seem to be much activity in trixbox land. anything else is probably a better idea |
| 20:46.56 | leifmadsen | OT: how do you list files recursively with ls? |
| 20:47.01 | citywok | ls -R |
| 20:47.11 | leifmadsen | weird... that doesn't work for me |
| 20:47.13 | citywok | rm -Rf works too |
| 20:47.37 | citywok | is it some stripped down embedded appliance? |
| 20:47.40 | leifmadsen | I see the file in the letters/ subdir but doesn't show up when I do: ls -R /var/lib/asterisk/sounds/en/*doll* |
| 20:47.46 | leifmadsen | no, CentOS |
| 20:47.54 | leifmadsen | oh I know |
| 20:47.57 | citywok | if you do it without the *doll* does it work? |
| 20:48.01 | leifmadsen | stupid CentOS creating aliases |
| 20:48.08 | leifmadsen | I need to remove those |
| 20:48.18 | Qwell | leifmadsen: it'll only be recursive through anything that matches *doll*/ |
| 20:48.21 | Qwell | (which is none) |
| 20:48.26 | Qwell | use find |
| 20:48.34 | leifmadsen | Qwell: oh ok.... quick tutorial on find? :) |
| 20:48.36 | leifmadsen | never used it |
| 20:48.44 | citywok | or just grep the output |
| 20:48.45 | Qwell | find /path/ -name '*doll*' |
| 20:48.46 | leifmadsen | suprisingly enough |
| 20:48.49 | leifmadsen | thanks |
| 20:49.10 | Qwell | probably add a "-type f" too |
| 20:49.17 | leifmadsen | got it, thanks! |
| 20:49.20 | citywok | d = directory? |
| 20:49.23 | Qwell | mmhmm |
| 20:49.26 | leifmadsen | how intuitive |
| 20:49.33 | leifmadsen | /var/lib/asterisk/sounds/en/digits/dollars.g729 |
| 20:49.33 | leifmadsen | /var/lib/asterisk/sounds/en/letters/dollar.wav |
| 20:49.39 | citywok | yea, i've never used find for much more than find ./ lol |
| 20:49.55 | leifmadsen | digits vs letters |
| 20:50.08 | Qwell | wtf |
| 20:50.29 | citywok | and why is one g729 and the other wav? or did you just prune the output? |
| 20:51.49 | marl_scot | hi, I am having problems getting CID to work with LCR/Asterisk 1.4, anyone able to give me some pointer on where to look to sort it out? if i ',NOOP(CALLERID : ${CALLERID(all)})' in the dialplan for the incoming call, i get nothing displayed :( |
| 20:52.24 | leifmadsen | citywok: just pruned as those were right beside each other |
| 20:52.30 | leifmadsen | it was the dir that was weird :) |
| 20:52.50 | leifmadsen | marl_scot: via SIP? |
| 20:53.06 | marl_scot | isdn -> asterisk -> sip |
| 20:53.14 | leifmadsen | ah, then I don't know |
| 20:53.17 | marl_scot | but asterisk isnt showing the CID via the callp[lan |
| 20:53.18 | leifmadsen | I don't use isdn |
| 20:53.27 | leifmadsen | not very often anyways :) |
| 20:54.38 | marl_scot | outbound is fine, have all calls set to put the caller id at the base DDI number, works a treat, was told originally that there was no CLI on these lines, but have now been told 'Yes there is, why isnt it working????' just typical :( |
| 20:54.40 | leifmadsen | there isn't a function in Asterisk I'm not thinking about that can convert seconds to minutes/seconds ? |
| 20:55.02 | marl_scot | and they cant wait a week or so more for me to replace the box with a 1.8 system :( |
| 20:55.03 | leifmadsen | rather than doing it with MATH() :) |
| 20:55.18 | citywok | lol, does * provide mod? |
| 20:55.37 | leifmadsen | yes it does |
| 20:55.48 | citywok | then you are totally good with math :P |
| 20:55.50 | leifmadsen | via MATH() :) |
| 20:55.58 | leifmadsen | citywok: no, I failed math :) |
| 20:56.03 | citywok | hahaha |
| 20:56.14 | citywok | you probably could have written it in the last 2 minutes using math :P |
| 20:56.20 | leifmadsen | probably |
| 20:56.48 | citywok | :) |
| 20:58.32 | citywok | or as tk points out you could have done it in 120 seconds :P |
| 20:58.42 | citywok | mod 60 = 2 minutes and 0 seconds |
| 20:59.45 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 21:01.34 | citywok | leifmadsen: i suggest you add a function for it :P |
| 21:02.08 | leifmadsen | doesn't it just leave the remainder? |
| 21:02.13 | leifmadsen | oh well, I have it figured out I think :) |
| 21:02.28 | citywok | can you ROUND(TOTAL/60,0) |
| 21:02.40 | citywok | whatever the * syntax would be for something like that |
| 21:02.49 | citywok | that would give you minutes, then use mod to get the remainder seconds |
| 21:03.46 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 21:03.59 | citywok | I guess you want FLOOR(total/60) |
| 21:04.07 | diemos | citywok: might i ask you for help building that script you spoke of? or ask for a copy of some sort? |
| 21:04.31 | citywok | the dirtiest way ever i could write in one line of php. lol |
| 21:05.49 | citywok | something liek this would work if you had sudo set up. If you don't want the gaping security hole their you'd have to write to a database and have a daemon running as root poll that DB to add the rules. <?PHP exec("sudo iptables -A INPUT -s {$_SERVER['REMOTEHOST']} -p udp --dport 5060 -j ACCEPT"); ?> |
| 21:05.59 | citywok | s/their/there |
| 21:06.34 | WIMPy | And how do you close them again? |
| 21:06.38 | citywok | and s/REMOTEHOST/REMOTE_ADDR |
| 21:07.03 | diemos | hm |
| 21:07.55 | citywok | WIMPy: with this dirty hack you don't, using the polling method it would be simple |
| 21:08.35 | diemos | what do you use to poll the db? |
| 21:09.19 | diemos | well, I have a linux ClearOS gateway setup with MySQL on it |
| 21:09.38 | diemos | and this is what I'm using to control all the traffic as well as do webserver stuff as well |
| 21:10.27 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:12.22 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 21:14.49 | *** join/#asterisk intrin (~notrin@host-174-44-155-166.hln-mt.client.bresnan.net) |
| 21:15.09 | intrin | anyone heard of asterCRM ? i was looking at it, and i cant find the system requierments |
| 21:17.43 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 21:21.11 | Katty | hi |
| 21:21.41 | Katty | my asterisk does not work at all how to fix plz???? |
| 21:21.50 | fenrus | :D |
| 21:22.11 | fenrus | did you reboot the server twice? :) |
| 21:22.22 | Katty | what is reboot?? |
| 21:22.22 | p3nguin | You're supposed to reboot it three times. |
| 21:22.33 | WIMPy | open bin, throw Asterisk into bin, close bin |
| 21:22.42 | Katty | k hold on i will try |
| 21:22.57 | Qwell | what means recycle bin full? |
| 21:23.00 | Katty | asterisk does not fit in bin??? |
| 21:23.15 | WIMPy | use force |
| 21:23.28 | Katty | that's what... |
| 21:23.30 | Katty | nevermind. |
| 21:23.40 | fenrus | my asterisk is in bin.. |
| 21:23.41 | fenrus | $ whereis asterisk |
| 21:23.41 | fenrus | asterisk: /usr/sbin/asterisk |
| 21:23.50 | fenrus | well, sbin then (: |
| 21:24.31 | *** join/#asterisk delx (~delx@dslb-092-073-186-106.pools.arcor-ip.net) |
| 21:24.33 | delx | Hey |
| 21:25.26 | leifmadsen | I think I may have found a bug... damn |
| 21:26.05 | WIMPy | The first today? :-) |
| 21:27.11 | citywok | diemos: you'd have to write everything. it's a custom hack to do what you want :P |
| 21:28.34 | delx | I just set up asterisk on my root server. Outgoing calls with ekiga work, but incoming calls to not get through. "sip show registry" doesn't show anything, but "sip show peers" shows sipgate with status OK. My configuration is: http://nopaste.info/8c93a05032.html Does anyone know what's wrong? :) |
| 21:28.40 | leifmadsen | WIMPy: first since using 1.8 actually |
| 21:28.56 | leifmadsen | PBXA*CLI> group show channels |
| 21:28.56 | leifmadsen | Channel Group Category |
| 21:28.56 | leifmadsen | Local/9998@QueueNumbers-5ad8;2 9998 agents |
| 21:29.07 | leifmadsen | although no Local channels exist, or active calls |
| 21:29.20 | citywok | i get ghost sip channels every couple days ever since 1.6.0 |
| 21:29.42 | citywok | although the ghost meetme channels that were super weird seem to have gone away |
| 21:29.56 | citywok | previously if you had a ghost meetme channel, next time you made any call you got put back in your meetme conference. lol |
| 21:32.40 | WIMPy | had 6 so far |
| 21:32.41 | WIMPy | Two of which I couln't pinpint, yet. |
| 21:32.46 | WIMPy | But then I'm usually pretty good at breaking things. |
| 21:34.33 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 21:35.03 | diemos | Katty: Go purchase Arm+Hammer, pour it on yourself and asterisk, then drench yourself in too much perfume and smile :D |
| 21:36.56 | *** join/#asterisk wimt (wimt@freenode/staff/wikipedia.wimt) |
| 21:37.33 | *** join/#asterisk Mhaddog (~Mhaddog_@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 21:37.33 | *** join/#asterisk tish (~dna6@unaffiliated/tish) |
| 21:37.33 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 21:39.17 | leifmadsen | aha! resolved! |
| 21:39.23 | leifmadsen | and I think I know why too, which is good to know |
| 21:39.23 | citywok | did you use math? lol |
| 21:39.29 | leifmadsen | lol I did earlier :) |
| 21:39.31 | citywok | oh, your ghost channel nm |
| 21:39.34 | leifmadsen | I used deductive reasoning this time |
| 21:39.35 | leifmadsen | ya |
| 21:39.37 | citywok | how did you get minutes? |
| 21:39.51 | leifmadsen | citywok: sloppy, like your wife! |
| 21:40.02 | citywok | oh crap, i have a wife?!?!?!? |
| 21:40.13 | citywok | lol |
| 21:40.13 | leifmadsen | :D |
| 21:40.14 | leifmadsen | same => n,Set(TimeUsed=${CUT(TimeCost,:,2)}) |
| 21:40.14 | leifmadsen | same => n,Set(MinutesUsed=${CUT(MATH(${TimeUsed} / 60),.,1)}) |
| 21:40.14 | leifmadsen | same => n,Set(SecondsUsed=$[${TimeUsed} - $[${MinutesUsed} * 60]]) |
| 21:40.51 | citywok | Heh that works. You could use floor though and round down the math instead of cutting on the . |
| 21:41.00 | leifmadsen | floor? |
| 21:41.32 | citywok | FLOOR(x) rounds x down to the nearest integer. |
| 21:41.41 | citywok | http://www.voip-info.org/wiki/view/Asterisk+Expressions#Arithmeticoperators |
| 21:42.02 | leifmadsen | citywok: the ghost channel was because features.conf was being executed on the peer channel, and it wasn't going away in memory after hang up (just the group count part for some reason). Switching to to run on the 'self' channel caused it to not hang up in memory anymore |
| 21:42.16 | leifmadsen | there is no FLOOR() function :) |
| 21:43.05 | citywok | stupid voip-info for being wrong. lol |
| 21:43.52 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 21:44.21 | leifmadsen | well it wasn't working for me when I had it on 'self' before, so I switched it and it was working, but then switching it bad it is still working, so away we go :) |
| 21:44.53 | citywok | was it the /n which is supposed to keep all the variables around? |
| 21:45.01 | leifmadsen | nope, I tried removing that too |
| 21:45.05 | leifmadsen | no change |
| 21:45.10 | leifmadsen | it was the change from peer to self |
| 21:45.14 | citywok | ah, weird. is it a bug? |
| 21:45.19 | leifmadsen | I think it could be.... |
| 21:45.28 | leifmadsen | the thing is that I'm using chan_agent as well |
| 21:45.34 | leifmadsen | so the agent never "hangs up" |
| 21:45.49 | leifmadsen | if I was using a SIP channel via Dial() I imagine I wouldn't have even noticed it |
| 21:46.17 | citywok | i've never used chan_agent |
| 21:46.24 | leifmadsen | ya this is my first time |
| 21:46.33 | leifmadsen | I hate it, but they insist they need the agents to be online all the time |
| 21:46.51 | citywok | what's the difference from using queues? |
| 21:47.00 | citywok | oooh |
| 21:47.08 | citywok | i see, it's liek the old agentcallbacklogin thing i guess |
| 21:47.48 | citywok | why do they want to hold hte line open and waste bandwidth/cpu? |
| 21:49.30 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:50.44 | leifmadsen | citywok: because they don't want the few seconds delay and agent could waste hitting the answer button, or not answering the call |
| 21:51.30 | leifmadsen | I'm going to have to ask them how much they are paying for bandwidth, and to then have them do a cost analysis to determine if what they are paying agents and the amount of time they could potentially waste would be enough to cover the costs of the increased bandwidth costs :) |
| 21:52.02 | citywok | queues w/ auto-answer headers & a 1 second warning beep? lol |
| 21:52.21 | citywok | use a local context channel member for the queue and use that to send an auto-answer |
| 21:52.37 | leifmadsen | citywok: ya this call center they are using is using SJphone unfortunately |
| 21:52.40 | leifmadsen | that was my suggestion too |
| 21:53.06 | citywok | Ah, i've never used SJPhone, sucky that it doesn't accept autoanswer. is it a freebie client that you have to get the paid version for that feature? |
| 21:53.54 | delx | I just set up asterisk on my root server. Outgoing calls with ekiga work, but incoming calls to not get through. "sip show registry" doesn't show anything, but "sip show peers" shows sipgate with status OK. My configuration is: http://nopaste.info/8c93a05032.html Does anyone know what's wrong? :) |
| 21:54.13 | leifmadsen | citywok: it's just a shitty softphone :) |
| 21:54.17 | leifmadsen | I think it's old and not updated |
| 21:54.30 | citywok | Ah. Sucky. Time to get them to switch to something that is slightly less suckarific. |
| 21:54.45 | leifmadsen | delx: if you don't register with sipgate how are they going to know where to send the call to? |
| 21:55.12 | leifmadsen | citywok: no kidding, unfortunately this is a sub-contracted call centre, so they don't own the phones, agents, etc. |
| 21:55.19 | leifmadsen | they just have the switch part |
| 21:55.24 | delx | leifmadsen, you are completely right. I used various debug options, but I don't see that my asterisk sends an REGISTER command to sipgate - but why!? |
| 21:55.36 | WIMPy | delx: Do you have the register in [general]? |
| 21:55.38 | citywok | delx: for lack of a register statement probably |
| 21:55.59 | leifmadsen | delx: because you didn't configure it to |
| 21:57.04 | delx | WIMPy, citywok, leifmadsen, I added "[general]" above "# register => 1133382t0:ZXY@sipconnect.sipgate.de/1133382t0 " - restarted asterisk -> no change. |
| 21:57.18 | leifmadsen | why do you have # ? |
| 21:57.24 | leifmadsen | sip show registry |
| 21:57.25 | delx | no "#" |
| 21:57.26 | leifmadsen | sip reload |
| 21:57.56 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 21:58.32 | delx | http://nopaste.info/dba6fbd563.html |
| 21:58.55 | p3nguin | You also should consider sending your phone number as the extension, rather than your username. |
| 21:58.57 | delx | leifmadsen, you say I didn't configure it? Isn't that register line enough? |
| 21:59.14 | delx | p3nguin, yes - but this doesn't affect the registry? |
| 21:59.43 | p3nguin | no, it just makes them send calls to the wrong exten. |
| 22:00.18 | delx | I'm going to fix it after the registry works, I promise! :) |
| 22:01.56 | delx | leifmadsen, did i configure it? |
| 22:02.17 | leifmadsen | delx: I don't know, I stopped paying attention a few minutes ago when I started programming again |
| 22:03.05 | delx | leifmadsen, is that registry statement enough? I'd appreciate it if you have a look.. :) |
| 22:03.15 | citywok | wtf, my * just hung for 90 seconds |
| 22:03.16 | leifmadsen | I can't look now |
| 22:03.25 | citywok | something blocked, when the block cleared everything caught up |
| 22:03.47 | marl_scot | citywok, what does top say? |
| 22:04.05 | marl_scot | was it just an * block or did cpu usage go through the roof? |
| 22:04.28 | WIMPy | Or does your harddisc cry for replacement? |
| 22:04.29 | citywok | no idea, i didn't catch it in time. |
| 22:04.39 | delx | no one knows why my registry isn't working? Why asterisk is not sending REGISTRY requests to sipgate? |
| 22:04.49 | marl_scot | top gives last 1 5 and 15 min load avg. normally |
| 22:04.54 | citywok | 88 seconds of blocking, i can see it by looking at the logs |
| 22:05.28 | marl_scot | how r u recording CDR? if its not csv, could you have a problem with your cdr backend? |
| 22:05.56 | citywok | It's CSV |
| 22:06.14 | marl_scot | oh well, that rules out one thing :) |
| 22:06.31 | marl_scot | (had that happen to me once, with a bad sql table) |
| 22:06.33 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 22:07.19 | russellb | Gosh, it sure is nice to be able to look in this channel throughout the day and not see people being called idiots all the time. |
| 22:07.22 | marl_scot | ever noticed a block before? |
| 22:07.46 | citywok | marl_scot: not in months |
| 22:07.58 | citywok | russellb: yes, it is. he PMd me earlier but he hasn't said anything in the channel. |
| 22:08.02 | marl_scot | hard to trace then :( |
| 22:08.58 | marl_scot | while im back in, anyone used LCR/* 1.4 with CID? cant et CID to work for incoming calls ISDN -> asterisk |
| 22:09.12 | citywok | yea, i can see the last thing it did by looking at the logs, but a single perl script that is actually just a die; at the top shouldn't cause that. |
| 22:09.59 | WIMPy | marl_scot: Only with 1.6 and 1.8. Do you see them in lcr.log? |
| 22:10.02 | marl_scot | if you are just killing the perl script would it not be better to return TRUE/FALSE rather than die? |
| 22:10.21 | delx | I get this error while asterisk talks with sipgate. Really destroying SIP dialog '3d8f87ec4b6fe3ed67eb13e54ebf3a74@85.214.253.251' Method: OPTIONS <- Is this a hint? |
| 22:10.28 | Qwell | test |
| 22:10.31 | Qwell | better. |
| 22:10.36 | citywok | marl_scot: i removed it from the extensions to not even call it since i'm not using it at the moment. lol |
| 22:10.45 | citywok | that's what i SHOULD have done :P |
| 22:11.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:12.12 | marl_scot | WIMPy, i dont appear to have a lcr.log at the moment :( have just enabled it |
| 22:12.59 | WIMPy | marl_scot: You can also view it with lcradmin state. |
| 22:13.29 | marl_scot | cant see my number appear in lcr.log |
| 22:13.32 | WIMPy | And it should be obvious for active cahnnels. |
| 22:14.06 | marl_scot | i can see 'caller id number=' but nothing shown |
| 22:14.28 | WIMPy | Looks like you're not receiving it then. |
| 22:15.42 | marl_scot | could there be any other config options in * that would prevent it being shown? cant be certain everythi gis configured corectly, althow most of it works :) |
| 22:16.38 | marl_scot | is there anythig that says a telco in the uk sends differant CID details to say a US telco? |
| 22:16.50 | citywok | international CID is hit & miss |
| 22:19.41 | WIMPy | marl_scot: Look at the lines with "MT_SETUP INDICATION U<-N" |
| 22:20.17 | marl_scot | 01.12.10 22:12:55.171 CH(1): MT_SETUP INDICATION U<-N port 3 called_pn type=0 plan=1 number=323232 complete channel_id exclusive=0 channel=1 bearer coding=0 capability=16 mode=0 rate=16 multi=-1 user=3 |
| 22:20.42 | marl_scot | does that make any sence? |
| 22:20.58 | WIMPy | Looks like the "calling_pn" part is missing. |
| 22:21.26 | marl_scot | ok where do i find the calling_pn? |
| 22:21.40 | WIMPy | In that line if it exists. |
| 22:22.30 | *** join/#asterisk trumee (~trumee@cpc5-cmbg14-0-0-cust982.5-4.cable.virginmedia.com) |
| 22:22.56 | WIMPy | So that looks like you aren't getting any. On in an incompatible format, but I couldn;t imagine what that could be. |
| 22:23.09 | WIMPy | Maybe you need to subscribe to some option? |
| 22:23.35 | WIMPy | Or you can use misdn_log to capture the raw packets. |
| 22:24.46 | *** join/#asterisk jsjc (~jsjc@230.Red-83-52-217.dynamicIP.rima-tde.net) |
| 22:25.18 | *** join/#asterisk Zipper_32 (~Zipper_32@CABLE-72-53-35-89.cia.com) |
| 22:26.47 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
| 22:27.12 | marl_scot | ok, lost connection there, local exchange doesnt like temps below -5 :( |
| 22:27.35 | WIMPy | So that looks like you aren't getting any. On in an incompatible format, but I couldn;t imagine what that could be. |
| 22:27.39 | WIMPy | Maybe you need to subscribe to some option? |
| 22:27.42 | WIMPy | Or you can use misdn_log to capture the raw packets. |
| 22:27.44 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 22:27.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:27.45 | marl_scot | WIMPy, did you say anything after 'looks like the calling_pm part is missing |
| 22:29.18 | marl_scot | wat level do i need to set debug to in misdn.conf to get hte required packet data? |
| 22:29.32 | marl_scot | and can it be grepped for the number or is it encoded? |
| 22:30.02 | WIMPy | You don't Just start misdn_log -cport |
| 22:32.43 | marl_scot | ok, only thing i can see is that id changes from 00af to 00ae in the output (01.12.2010 22:31:08.992751 received 12 bytes prim = 2001 id=00ac0000 00 01 01 01) |
| 22:33.30 | *** join/#asterisk diemos (~diemos@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 22:33.33 | WIMPy | Look for a call. That should give you a longer row of numbers. |
| 22:33.59 | marl_scot | nope, that line was the longest line dispalyed :( |
| 22:35.21 | WIMPy | Do you have more than one interface? |
| 22:35.37 | marl_scot | the id changing from 00ac0000 to 00ae0000 was the only appearnt differance between idle log entrys and entries when calling into * |
| 22:35.47 | marl_scot | one card, 4 ports :) |
| 22:36.27 | WIMPy | Wrong port? |
| 22:38.06 | marl_scot | is ther ea way to monitor all ports at once? |
| 22:38.56 | WIMPy | Not with misdn_log. There might be with lcradmin trace. Not sure about that. |
| 22:39.58 | WIMPy | But 0-3 shouldn't be too much choice to find out. |
| 22:41.45 | marl_scot | :) looks like the cables are wrong way round, -c3 gave me a long line on inbound call |
| 22:42.04 | marl_scot | 01.12.2010 22:40:39.283064 received 38 bytes prim = 2002 id=00000000 02 01 24 24 08 01 73 05 a1 04 03 90 90 a3 18 01 |
| 22:42.05 | marl_scot | <PROTECTED> |
| 22:42.34 | WIMPy | That's clearly only one number. |
| 22:43.01 | marl_scot | what do u mean? |
| 22:43.12 | marl_scot | not sure what im looking at in this log |
| 22:43.51 | WIMPy | You can easily spot digits as they have a 3 in the left nibble. |
| 22:44.34 | WIMPy | So the only number there is 323232. |
| 22:44.46 | WIMPy | I guess you have to talk to BT. |
| 22:44.51 | marl_scot | ok, going by that rule, then the only number displayed is 323232, which is the DDI number that i am dialing into |
| 22:45.16 | marl_scot | sh*t, thats me onto another 1.5 hour phone call :( |
| 22:46.34 | marl_scot | 15 minites to get to speek to a human being, then another hour to get to speak to someone who knows what there talking about then 15 mins to get them to look at a screen to say if there is any problem with the CID :( |
| 22:46.42 | WIMPy | Don't they have a contact form online? |
| 22:47.28 | marl_scot | lol, yup if you want ot wait for a year to get something thats been converted from english to japaneese to koreen to hungerian to german and back to english :) |
| 22:48.16 | WIMPy | But here you wouldn't have a cahnce to ever talk to someone who understands the issue. |
| 22:48.19 | marl_scot | may just wait till i find an engineer at a terminal block on the street and hold him at knife point to press 15 keys to get the info i need :) |
| 22:48.37 | WIMPy | The trick is to just tell them to write down what you want and pass it on. |
| 22:49.00 | marl_scot | lol, thats if they understand the words you tell them :) |
| 22:49.11 | marl_scot | BT is TERIBLE, and has always been :( |
| 22:49.37 | WIMPy | They don't. Just spell it out. |
| 22:49.56 | marl_scot | I used to charge cusomters £5 p/h to talk to BT about phone line problems, as i was garentied to make £50+ from each problem :) |
| 22:49.58 | WIMPy | Takes a little more time, but gets the job done. |
| 22:50.20 | WIMPy | ugh |
| 22:52.15 | marl_scot | they are the worst telco on the planet for customer service :( i knew more about phone lines when i was 10 years old than most of the people who talk on the phone ifor BT, the only way i have ever got anything done with them is to shout loudly and then find a frendly engineer who wants a couple of cans oif beer to fix the problem :) |
| 22:53.03 | WIMPy | They are all prett bad, I think. |
| 22:53.05 | WIMPy | +y |
| 22:54.41 | WIMPy | The worst I cam across are Kabel Deutschland. They always respond with "we are aware of the problem ad are working at high pressure on a solution". They don't seem to think it somehow funny to write |
| 22:54.49 | WIMPy | that for over a year. |
| 22:55.12 | Katty | hellloooooooooooooooooooooo nurse. |
| 22:56.39 | marl_scot | lol, BTs normal is 'I can not see any problems with your line, it must be your equipment, and there fore we will charge you £150.00 to come out and hanve a look', only to have the problem 'magiclly' correct itself 10 mins later without doing single thing! |
| 22:57.01 | WIMPy | classic |
| 22:57.29 | marl_scot | i have had well over 100 of those 'magical' fixes in the last 18 months! |
| 22:57.56 | _Corey_ | jeez, that never happens over here in the states ;) |
| 22:59.40 | marl_scot | _Corey_, BT are too used to having the manopoly here, its only only in the last couple of years that its been posible to tell BT to get knotted, and even then they cause problems with moving numbers from them to a new telco :( |
| 23:00.43 | _Corey_ | yeah, exactly the same story here... the competitive providers are getting just as bad |
| 23:01.23 | _Corey_ | I have a theory that there are actually between 1-5 engineers at any carrier that really understand what they're doing |
| 23:01.52 | _Corey_ | I *know* this to be the case for several of the bigger CLECs and their VoIP offerings |
| 23:01.56 | WIMPy | You're probably right about that. |
| 23:02.19 | marl_scot | i even had BT blame another telco for not plugging in my ISDN cable at the exchange, when the only engineers with access to that exchange where BT!!!! |
| 23:03.15 | marl_scot | 1-5??? id say about 1 for most now :( they lay off the 'old school' engineers, and take on new 'swap it out' engineers |
| 23:04.04 | delx | p3nguin, still here? :) |
| 23:04.06 | _Corey_ | depends on the size, I can say definitely _1 guy_ at a major national CLEC (who shall remain nameless) on their SIP offer |
| 23:04.37 | _Corey_ | I gotta believe Verizon has at least a few more than that worldwide. ;) |
| 23:05.19 | marl_scot | over here, we have one telco that provides a good service, and that is it, and even they have BIG problems, as BT still has controll over the copper cable :( (or aluminum if your REALLY unlucky)!!! |
| 23:06.13 | _Corey_ | That's still the case here in most markets outside major cities |
| 23:07.20 | _Corey_ | I'm never surprised when a carrier does the "blame the other guy" game though |
| 23:07.32 | delx | "Call from '1133382t0' to extension '49234163556527' rejected because extension not found." <- Then I must name my extension "[49234163556527]" - right? |
| 23:07.46 | _Corey_ | I've just been shocked at how truly stupid and untrained most of the support people are at every level at most carriers |
| 23:07.47 | citywok | One thing i love about dealing 100% with qwest is they can't blmae the other guy. THEY are the other guy so end to end it's their problem. |
| 23:07.57 | citywok | delx: correct |
| 23:08.29 | _Corey_ | We had an engineer here call Level-3 yesterday with a customer to get their SIP credentials for the trunk and the engineer responded confidently that it was "ESF/B8ZS" |
| 23:08.48 | delx | citywok, I fixed the registry problem. It works now. But I need to configure my extensions. Could you have a look on my config files please? |
| 23:08.54 | delx | I post them..one moment |
| 23:09.02 | _Corey_ | has finished venting |
| 23:09.30 | citywok | lol t1 settings on a sip trunk |
| 23:09.45 | _Corey_ | seriously |
| 23:09.47 | citywok | and L3 will talk to you? you must have some serious minutes. they wouldnt talk to me b/c i dont have 500k / mo |
| 23:09.49 | nny | anyone know of a command line tool (transcode?) to convert g711 to wav or similar? |
| 23:10.09 | nny | _Corey_: heh |
| 23:10.18 | _Corey_ | this is actually a smaller customer believe it or not |
| 23:10.25 | _Corey_ | end-user school district |
| 23:10.33 | citywok | nny transcoding- |
| 23:10.40 | citywok | weird, paste fail. http://keystoneit.wordpress.com/2006/05/02/sox-wrap-for-transcoding-asterisk-sounds/ |
| 23:10.51 | _Corey_ | it's usually the wholesale programs that have the heavy monthly usage commits |
| 23:11.01 | nny | _Corey_: in that right now with a missing 180/183 from SIP on outgoing calls |
| 23:11.13 | citywok | whoops this may be better nny http://astrecipes.net/index.php?n=152 |
| 23:11.15 | delx | citywok, http://nopaste.info/b525dda623.html |
| 23:11.28 | _Corey_ | nny: good luck getting them to understand _that_ |
| 23:11.53 | citywok | delx: put [sipgate_in] instead of [4923....] |
| 23:12.01 | citywok | and change the 114138 to the 492341 |
| 23:12.03 | nny | _Corey_: they say they are sending it, yet tcpdump on my end shows nothing except 100->200 :\ |
| 23:12.07 | citywok | does that make sense? |
| 23:12.18 | nny | _Corey_: but yeah, fun stuff |
| 23:12.19 | citywok | nny ask them for their tcpdump |
| 23:12.25 | _Corey_ | nny: Maybe they're sending it somehwere else... :) |
| 23:12.33 | citywok | yea, ask them for the dump :P |
| 23:12.35 | nny | _Corey_: hmm good point |
| 23:13.01 | nny | citywok: will do. They were transcoding g729 to g729 at their media gateway last and dropping DTMF. That took lots of pain to get fixed |
| 23:13.11 | nny | they said it took less overhead to transcode |
| 23:13.15 | nny | i have no idea why that is |
| 23:13.40 | citywok | how do you transcode from -> to the same thing? |
| 23:13.43 | _Corey_ | sounds like a critical step... transcoding from g729 to g729 |
| 23:13.44 | citywok | decompress it then recompress? |
| 23:13.47 | _Corey_ | lol |
| 23:13.56 | nny | lol |
| 23:13.57 | nny | idk |
| 23:14.15 | citywok | delx: also, dial(SIP/101,20) or 30, you want a timeout or it will ring into oblivion |
| 23:21.15 | delx | citywok, incoming calls work, great! But now outgoing calls have a little problem :) |
| 23:21.29 | delx | handle_response_invite: Failed to authenticate on INVITE to '"1133382t0" <sip:1133382t0@85.214.xx.xx>;tag=as35f52fe8' |
| 23:21.49 | citywok | what does your outbound dialing context look like |
| 23:23.41 | delx | citywok, http://nopaste.info/d2f8d4d59b.html |
| 23:24.22 | WIMPy | delx: That Set doesn't make much sense. |
| 23:24.28 | citywok | yea |
| 23:24.35 | citywok | Dial(SIP/1133382t0) |
| 23:24.40 | delx | ok I remove it |
| 23:24.54 | citywok | well, sip/${EXTEN}@sippeer |
| 23:24.55 | WIMPy | delx: Dial the peer name from your sip.conf. The way you do it, you're missing a password. |
| 23:26.12 | *** join/#asterisk ectospasm (ectospasm@188.72.223.139) |
| 23:27.42 | delx | citywok, WIMPy, YOU ARE MY HEROS! :) It's working both: Incoming AND outgoing now :) |
| 23:28.01 | delx | where should I transfer the 10.000 $ ? |
| 23:28.19 | citywok | lol |
| 23:28.22 | citywok | my bank account :P |
| 23:28.27 | WIMPy | delx: Sparda-Bank Hamburg :-) |
| 23:28.44 | delx | WIMPy, du kommst aus deutschland? :) |
| 23:28.56 | WIMPy | ja |
| 23:29.06 | delx | machst du das mit asterisk beruflich? :) |
| 23:29.15 | WIMPy | So Euros would be preferred :-) |
| 23:29.25 | WIMPy | ja |
| 23:29.46 | WIMPy | We have an #asterisk.de, BTW. |
| 23:30.10 | delx | ah, cool |
| 23:32.02 | citywok | i'm already in enough #asterisk channels lol |
| 23:32.13 | citywok | and delx you're welcome. |
| 23:32.27 | citywok | do you now understand HOW those changes we made you make work? |
| 23:32.47 | delx | yes, I understand more and more |
| 23:33.05 | delx | It took me 6 hours now to fully base-setup my server :) |
| 23:33.14 | delx | s/server/asterisk/ |
| 23:33.16 | citywok | Yea, the first one takes a lot longer :P |
| 23:33.31 | citywok | wow infobot is actually responding now. the last couple times he didnt |
| 23:33.49 | delx | he only responds to me,hehe |
| 23:34.05 | citywok | s/infobot/botinfo/ |
| 23:34.10 | citywok | pfft, ignorer |
| 23:34.11 | ecrane | _Corey_: that SIP credential is sweet. |
| 23:34.19 | delx | citywok, haha :) |
| 23:34.33 | _Corey_ | delx: 8 hrs isn't bad... at that rate you'll be able to take your dCAP test in a couple weeks |
| 23:34.35 | delx | s/haha/hahaha/ |
| 23:34.50 | citywok | what a butt |
| 23:34.54 | citywok | s/butt/buttt/ |
| 23:35.02 | citywok | ah, it ignores if you use its name :P |
| 23:35.08 | delx | _Corey_, or years :) |
| 23:35.31 | citywok | heh, i suppose i could have taken the dcap test, but i don't really need it for anything so... |
| 23:35.43 | citywok | and it would have taken away from my drinking time at astricon |
| 23:35.54 | _Corey_ | you missed a fancy breakfast |
| 23:35.59 | _Corey_ | there were omlettes |
| 23:36.35 | _Corey_ | although it was the morning after the party, so... it could have been rough for many |
| 23:36.49 | citywok | yea i didn't get up until like 11. we didnt go to bed until 3... |
| 23:37.12 | _Corey_ | lol, i was speaking at 11 so i had to curb my intake |
| 23:37.28 | citywok | what was your topic? |
| 23:37.37 | _Corey_ | it was a call center case study |
| 23:38.00 | delx | Now i need the next feature: If my number is 4040 - I'd like to append a code to this number (4040 - 12345) and this code (12345) should relate to a phone number (x) to which the phone call should be forwarded. Any hints what topic I should search on google? :) |
| 23:38.23 | delx | The code->phone number relation should be stored in a database |
| 23:38.45 | delx | Could I use AGI for that? |
| 23:39.19 | _Corey_ | delx: you could dip into mysql or something with an agi and just set a channel variable |
| 23:39.37 | delx | I'm familiar with mysql - that would be fine |
| 23:39.39 | *** join/#asterisk digilink (~digilink@irc.stephennet.net) |
| 23:39.43 | delx | http://stackoverflow.com/questions/69676/mysql-asterisk-dialplans-and-call-forwarding |
| 23:39.47 | delx | this might work, right? |
| 23:40.38 | _Corey_ | yeah, looks about right |
| 23:40.49 | _Corey_ | you just deal with the ${FWD_TO} variable |
| 23:40.57 | _Corey_ | you'll want to check to make sure it contains something though |
| 23:41.34 | delx | yes - where do I have to store that script which I can call with AGI(script) ? |
| 23:41.53 | _Corey_ | <PROTECTED> |
| 23:42.01 | _Corey_ | (unless you specify a path) |
| 23:42.28 | delx | ah, ok..path is fine |
| 23:43.09 | carrar | moo |
| 23:44.47 | *** join/#asterisk d-fens (bungojungo@inter.netsucks.net) |
| 23:55.08 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 23:55.32 | *** join/#asterisk d-fens (~jscalia@su-nat.int.smq.datapipe.net) |
| 23:55.55 | Docfxit | Hi, |
| 23:55.59 | *** join/#asterisk Tim_Toady (~moi@178.128.21.207.dsl.dyn.forthnet.gr) |
| 23:59.46 | Docfxit | The phones connected to an Asterisk box can't make or receive calls. The Ubuntu/Asterisk can ping to the phones. The phone icon is usually black now they are clear . Digium tested Asterisk and it can make and receive calls from the CPU. I have rebooted one phone and that didn't fix it. The phones do have the current date and time. Why wouldn't the phones be communicating to Asterisk? |