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| 00:28.53 | jShaf | asterisk.org down? |
| 00:30.34 | WIMPy | www seems down, yes. |
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| 00:55.46 | tacloban | is anyone available to help with some google voice integration? |
| 00:59.41 | tacloban | everyone must be eating some nice christmas eve dinner |
| 01:01.31 | *** join/#asterisk btel (~btel@c-98-234-115-234.hsd1.ca.comcast.net) |
| 01:02.18 | tacloban | sup btel |
| 01:03.11 | btel | hi tacloban |
| 01:05.01 | tacloban | no dinner for you? |
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| 03:30.22 | Freeaqingme | Hey folks, I'm busy setting up asterisk using the definitive handbook, but I'm running into some sort of problem now (rather something I'm just unable to find). I can make outbound calls just fine (going out of my trunk), but when I call my trunk, I get a "number busy" signal. How do I debug that, and how do I get it to redirectto a nice menu? |
| 03:34.07 | *** part/#asterisk Freeaqingme (~Freeaqing@2001:980:45b3:1:216:3eff:fe00:5) |
| 03:34.10 | WIMPy | Set verbos to at least 3, make a call and see what it has to tell you. |
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| 05:15.08 | sizzers | hey guys, can anyone point me to a nice simple outbound dial macro that checks the status of each trunk before it tries to send a call out it. |
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| 05:28.03 | NightMonkey | Howdy. I'm slowly working through calming down daemons from writing to the flash RAM I'm using. I found that Asterisk is writing to /var/spool/asterisk/outgoing/ 120 per 60 seconds. |
| 05:28.22 | NightMonkey | What is stored here, and can I reduce the amount of writes Asterisk makes there? |
| 05:29.25 | WIMPy | It writes there? |
| 05:29.42 | WIMPy | You could disable pbx_spool, I guess. |
| 05:30.15 | WIMPy | But generally I wouldn't use flash storage read-write at all. |
| 05:30.33 | NightMonkey | WIMPy: Hrm, I'll need to look up that module's purpose. |
| 05:30.56 | WIMPy | call files |
| 05:31.10 | NightMonkey | WIMPy: Yeah, I could symlink it to /dev/shm, but I'd like to see how far I can get in calming writes. |
| 05:31.21 | NightMonkey | WIMPy: Hrm. No calls are in progress. |
| 05:31.49 | WIMPy | It is searching for call files there. |
| 05:32.03 | WIMPy | I don;t have a clue why it writes, tho. |
| 05:32.23 | NightMonkey | WIMPy: Ah, my bad! |
| 05:32.43 | NightMonkey | WIMPy: I wasn't reading inotifywatch carefully. It was just reading there. |
| 05:33.03 | WIMPy | Ok |
| 05:33.10 | NightMonkey | WIMPy: OK, that makes me feel better about *. :) Now, I need to calm Postfix. :) |
| 05:33.40 | NightMonkey | WIMPy: Sorry for the noise. Thanks for helping. |
| 05:34.22 | WIMPy | np |
| 05:35.39 | tacloban | anyone here familiar with the asterisk wiki article on google voice integration? |
| 05:35.55 | tacloban | i am looking to do a minimal asterisk config for google voice |
| 05:36.07 | tacloban | but, I seem to be having problmes |
| 05:58.41 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
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| 06:51.08 | raden | why is asterisk not deleting my call files upon completion ? |
| 06:51.26 | WIMPy | permissions? |
| 06:54.17 | raden | hmmm |
| 06:54.32 | p3nguin | Typically, that's the reason. |
| 06:54.55 | raden | call file i have channel: SIP/303 |
| 06:55.03 | raden | [Dec 25 00:52:37] NOTICE[9093]: channel.c:3524 __ast_request_and_dial: Unable to request channel SIP/303 |
| 06:59.08 | raden | OK how do i make asterisk not call all 2000 call files at once ? |
| 06:59.32 | p3nguin | Don't dump 2000 call files into the spool directory at one time. |
| 06:59.50 | raden | how am i going to control that ? |
| 07:00.03 | p3nguin | Just don't do it. That takes care of not doing it. |
| 07:00.03 | raden | with php |
| 07:00.49 | carrar | Change your file mod time on the call file |
| 07:00.52 | carrar | to some future date |
| 07:00.53 | p3nguin | Better coding could prevent writing 2000 files all at once. |
| 07:01.21 | carrar | and add google to your "ask first" list |
| 07:01.27 | carrar | :) |
| 07:01.30 | carrar | Merry Christmas |
| 07:02.42 | raden | carrar, thanks |
| 07:02.59 | raden | why cant i dial SIP/100 etc from a call file |
| 07:03.12 | p3nguin | SIP/100 must not exist. |
| 07:03.55 | p3nguin | If the device entry doesn't exist or if the device isn't available, your call to it will fail. |
| 07:03.57 | carrar | SIP/100 is hard coded to be blocked |
| 07:04.03 | p3nguin | hahaha |
| 07:04.04 | carrar | don't use it |
| 07:04.09 | raden | lol |
| 07:04.14 | p3nguin | Oh, you didn't get the SIP/100 license, I guess. |
| 07:04.24 | raden | for some reason none of phones at office were registered |
| 07:04.31 | raden | p3nguin, haha :P |
| 07:04.38 | p3nguin | You'll have to purchase that from Digium and use their register utility. |
| 07:05.06 | raden | got it working |
| 07:05.27 | p3nguin | So... I bought myself a Christmas present. |
| 07:05.33 | raden | me 2 :) |
| 07:05.36 | raden | what u get yourself |
| 07:05.51 | p3nguin | I decided I needed an Escort Passport 9500ix radar detector. |
| 07:06.12 | raden | I bought me one of them for my birthday my 7500 finally died this spring |
| 07:06.18 | raden | I love the 9500 |
| 07:06.36 | raden | I bought a 8k winch for my truck for christmas :) |
| 07:06.41 | p3nguin | Quite expensive, but I figured what the hell, it's only money. |
| 07:06.47 | carrar | How about just not speed |
| 07:07.00 | raden | the cost of the detector far outweighs what a ticket does financially |
| 07:07.02 | p3nguin | I don't always speed with intention. |
| 07:07.09 | raden | has same problem |
| 07:07.13 | raden | even with cruise |
| 07:07.18 | carrar | get a lawyer that will get your tickets dropped |
| 07:07.28 | raden | carrar, again detector cheaper |
| 07:07.34 | p3nguin | The detector costs less than an attorney. |
| 07:07.40 | carrar | lawyer is funner |
| 07:07.51 | raden | p3nguin, I look at a escort as a cheap insurance policy :) |
| 07:08.01 | carrar | make the state spend more |
| 07:08.09 | raden | I have over 1500 in radar detectors and jammers but almost 1 million miles ticket free |
| 07:08.19 | tyman | Does anyone know a repo for fedora 14 for mpg123 and it's dependencies? |
| 07:08.28 | carrar | Buy a radar TX |
| 07:08.31 | carrar | and jam |
| 07:08.53 | carrar | since you're gonna break the law anyways |
| 07:08.56 | raden | Im amazed by how simple call files are |
| 07:09.10 | raden | tyman, dont know anyone using fedora bro |
| 07:09.10 | p3nguin | I might know about _its_ dependencies, but I don't really care too much about it. |
| 07:10.13 | p3nguin | yum install mpg123? It's in rpmforge. |
| 07:10.36 | tyman | raden: yeah...just using it on my laptop vm for devel. Haven't done desktop RH since they f'ed up the window managers with that "blue curve" hybrid gnome/kde crap. |
| 07:11.04 | raden | lol |
| 07:11.51 | p3nguin | Install rpmforge. That will take care of a lot of missing packages -- including mpg123. |
| 07:12.39 | tyman | p3nguin: I'm there now...looking for fedora rpmforge rpm |
| 07:15.24 | p3nguin | I think it would be nice if they could make an official rpmforge rpm available in extra, which would ease the step of installing it to gain access to rpmforge. |
| 07:15.44 | tyman | agreed |
| 07:15.45 | p3nguin | Maybe they were thinking of a recursion problem and chose to not do it. |
| 07:16.20 | raden | Whats a virtual PRI ? |
| 07:17.43 | p3nguin | As far as I can tell, it's a bunch of channels which are unmetered. |
| 07:18.02 | raden | Interesting |
| 07:20.15 | carrar | no such thing as a virtual pri |
| 07:20.40 | p3nguin | There is according to Flowroute (and probably other ITSPs). |
| 07:20.54 | carrar | then it's SIP channels |
| 07:21.29 | carrar | or possibly TDMA over Ethernet |
| 07:21.46 | tyman | probably marketing bs to sell you the same number of SIP trunks as pri bearer channels |
| 07:21.55 | carrar | very possible |
| 07:22.23 | carrar | buy 23 SIP Channels when you really only need 5 |
| 07:22.24 | p3nguin | Right, it's SIP channels. |
| 07:22.55 | tyman | i love how all these ITSPs want to sell you 20 concurrent channels, each with unlimited minutes.... |
| 07:24.36 | p3nguin | If you use LOTS of minutes and need lots of concurrent calls, it makes sense to go unmetered. |
| 07:24.55 | tyman | if you read some of their EULAs, it states "unlimited" is not more than experienced be their typical users...in other words, unlimited == "we have a limit, but we're not going to tell you until after you sign up and hit it" |
| 07:25.20 | p3nguin | When dealing with metered channels, unlimited is often around 3000 - 3500 minutes. |
| 07:25.51 | p3nguin | Unmetered, on the other hand, is not limited nor charged for minutes of use. |
| 07:26.49 | tyman | p3nguin: there's nothing with sip itself that keeps you from making unlimited concurrent calls thru 1 channel really. I wish they'd drop the trunk carry over bs and just charge 10 people at 1 min the same as 1person at 10mins. |
| 07:27.27 | p3nguin | You can only make one concurrent call per channel. |
| 07:27.43 | raden | i just need 2 channels unmetered |
| 07:27.52 | raden | we have 20 metered channels at $0.0065 |
| 07:28.10 | raden | when asterisk calls out by a call file how do i control the codex ? |
| 07:28.27 | p3nguin | How many minutes total, on all combined channels, do you use each month? |
| 07:29.03 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:29.06 | tyman | p3nguin: My understanding of sip is that is just implemented (sometimes not even) for their billing purposes....so you'll buy multiple trunks. |
| 07:29.08 | raden | slow months 3k-4k total busy season we can push 2k a day between 4 people |
| 07:29.34 | raden | lines get stacked then having call files going more inbound etc.... I have no clue how many were going to start using |
| 07:29.45 | p3nguin | First of all, there is no such thing as a "SIP trunk." Second of all, each call creates one channel. |
| 07:30.34 | tyman | p3nguin: agreed. kind of my point. Should charge per channel |
| 07:30.52 | p3nguin | If they say "trunk," they just mean a grouping of channels between their system and yours. |
| 07:31.15 | p3nguin | They do charge per channel... or per minute, depending on what you've agreed to. |
| 07:31.53 | raden | we have 20 some channels i just wish we had a few unmetered channnels |
| 07:32.00 | p3nguin | I pay per minute that I have a call active. If I have 100 one minute calls all at the same time, I've used 100 minutes and I pay 100 cents. |
| 07:32.04 | raden | no big deal though |
| 07:32.22 | tyman | p3nguin: thats the way to go... |
| 07:32.35 | raden | p3nguin, thats what we have now but 100 minutes 65 cents |
| 07:32.43 | p3nguin | On the other hand, if I have one 100-minute call, I still pay 100 cents for it. |
| 07:32.52 | tyman | p3nguin: i was just commenting on the marketing games used with sip "trunking" |
| 07:32.53 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:32.53 | raden | exactly |
| 07:33.35 | p3nguin | But if you have the virtual PRI, it's going to be unmetered and have a limited number of channels you have available. |
| 07:33.53 | p3nguin | You might be able to make ONLY 12 concurrent calls, but those calls don't cost you per minute of use. |
| 07:34.07 | p3nguin | Call number 13 will give congestion tones. |
| 07:34.34 | raden | yea ill just add more channels for now |
| 07:34.54 | raden | i was going to set it up so outbound on VPRI but screw it |
| 07:35.17 | raden | how can i set codec asterisk uses when processing a call file |
| 07:35.19 | p3nguin | I have an unlimited number of channels available, so my concurrent call limit will be imposed by my bandwidth and hardware limitations. |
| 07:35.26 | carrar | Just make virtual calls |
| 07:35.51 | *** join/#asterisk epaphus (~epaphus@ec2-50-16-226-177.compute-1.amazonaws.com) |
| 07:35.54 | epaphus | Hello all. |
| 07:36.38 | epaphus | Is it possible to create an extension that when dialed instead of getting a "ringing" tone , i get music untill it is picked up? |
| 07:36.57 | tyman | p3nguin: may i ask who you are using as an ITSP? |
| 07:37.11 | p3nguin | epaphus: Use the 'm' option in the Dial() command. |
| 07:37.55 | p3nguin | I primarily use VoIP.ms, but I play around with Flowroute. |
| 07:37.59 | tyman | epaphus: also look at the FollowMe dial application |
| 07:38.47 | tyman | p3nguin: I'll check them out. thx |
| 07:38.51 | p3nguin | The m options plays the configured MusicOnHold music rather than ringing sound. |
| 07:38.59 | p3nguin | core show application Dial |
| 07:39.30 | epaphus | p3nguin iam new to asterisk.. so where is the Dial() command? |
| 07:39.38 | p3nguin | It's in the extension. |
| 07:39.46 | epaphus | extension.conf, ok |
| 07:40.19 | tyman | epaphus: http://ofps.oreilly.com (great reference) |
| 07:40.57 | p3nguin | I got the feeling that you didn't need to create a new extension, but rather wanted to change an existing extension from ringing to music. |
| 07:41.04 | epaphus | p3nguin, or could i also set a queue for 1 extension only? or that pretty much is a dumb thing? |
| 07:41.21 | epaphus | p3nguin ahh right.. yeah thats all i need |
| 07:41.22 | p3nguin | It's rather useless, when the m option is made to do exactly what you're after. |
| 07:41.43 | raden | Im thinking of getting a 32" monitor |
| 07:42.19 | tyman | raden: I have a 30" cinema display at the office and home...I frickin' love them |
| 07:42.27 | p3nguin | If you have something like Dial(SIP/turd,30,r) , just change it to Dial(SIP/turd,30,m) and save/exit, dialplan reload, done. |
| 07:42.51 | raden | Acera has there 32" monitor with TV tuner for $329 right now seems like a good deal 25" seems so small these days |
| 07:43.03 | epaphus | p3nguin so ok.. suppose i have my extension set with music instead of ringing. is it possible to embed recordings like.. "please wait, you will be assisted shortly" ..? |
| 07:43.34 | p3nguin | Yes, but you'll have to create a musiconhold class with your own files that play those messages. |
| 07:43.46 | epaphus | ok. thanks.. |
| 07:44.06 | p3nguin | Well, it's better to use one long file that contains the music and the periodic voice messages. |
| 07:44.45 | p3nguin | If you start getting into waiting and periodic announcements, queues start looking better to me. |
| 07:45.34 | epaphus | Suppose I did all that, and i want to create a dialplan for that extension so that it tries to ring 2 remote SIP devices.. or else go to voicemail. are there examples of that on the net? |
| 07:46.16 | p3nguin | Dial(SIP/turd&SIP/ferguson,30,m) |
| 07:46.26 | p3nguin | VoiceMail(mailbox@context) |
| 07:46.43 | epaphus | easy. cool. |
| 07:46.56 | p3nguin | "mailbox" would need to be a mailbox that both people can easily check if the message could be for both people. |
| 07:47.04 | epaphus | tnx |
| 07:47.42 | p3nguin | I'm not sure if it is possible to record one voicemail into more than one mailbox at a time. |
| 07:47.54 | epaphus | iam the only guy, so no problem |
| 07:48.31 | p3nguin | Oh, alright. That makes it easier. I was going to say that I use a group mailbox and subscribe each member to his own mailbox plus the group one. |
| 07:48.56 | epaphus | So, suppose that instead of one of those SIP devices.. i want it to try to dial through the trunk the call to the PSTN. I sthat possible also? :) p3nguin |
| 07:49.58 | p3nguin | Sure. Dial(SIP/turd&DAHDI/g0/13149691077,30,m) should do it. |
| 07:50.44 | tyman | p3nguin: I thought you could record to multiple vmail boxes with the single VoiceMail dial app. It just uses the greeting from the first listed. |
| 07:51.22 | p3nguin | Or if by "trunk," you didn't really mean it and meant another via another SIP peer, SIP/itsp/3149691077 would be the way to do that. |
| 07:51.30 | epaphus | p3nguin if i do it that way.. and i decide to do it with a queue.. then i imagine that I would need to hardcode myself to the queue.. right? |
| 07:52.14 | p3nguin | Just list the queue member as being either that, or use a local channel and make a special extension to take care of the dialing. |
| 07:52.29 | p3nguin | I'd go for the local channel method. |
| 07:53.11 | p3nguin | I have a context called queue-devices that I use for my queue members which are local channels. |
| 07:53.18 | epaphus | wonders what a local channel is. |
| 07:53.32 | p3nguin | In the queue-devices context, I set up my Dial commands. |
| 07:53.52 | p3nguin | member => Local/762@queue-devices |
| 07:53.56 | ChannelZ | Local channels are psuedo-channels that can call the dialplan |
| 07:54.15 | p3nguin | then extension 762 in the queue-devices context does the actual dialing of a device for me. |
| 07:54.34 | epaphus | ok ill read more on that tnx again |
| 07:58.05 | epaphus | p3nguin.. so youre saying that a queue member can be something like Dial(SIP/turd&DAHDI/g0/13149691077,30,m) ... right? |
| 07:58.13 | p3nguin | One of my primary reasons to use a Local channel as a queue member is so I can do more than just dial a device from a queue. For instance, I want to use JabberSend() to IM some call info, set a couple new variables, and then finally dial the devices from the queue. |
| 07:58.19 | p3nguin | no |
| 07:59.07 | p3nguin | The member could be SIP/turd, Agent/54321, DAHDI/g0/13149691077, or Local/762@queue-devices. |
| 08:00.04 | p3nguin | SIP, Agent, DAHDI, and Local ... all being the channel technology used to reach the member. |
| 08:00.37 | p3nguin | IAX2, SCCP, Console, ... all other possible channel technologies used for reaching a member. |
| 08:01.08 | epaphus | cool. |
| 08:01.10 | epaphus | got it. |
| 08:03.41 | epaphus | night. |
| 08:04.57 | epaphus | wonders how setting a queue member being dadhi on a cell phone.. how would it react if the cellphone is out of range and it gives exactly that message. Would asterisk route the call ? |
| 08:06.05 | raden | I miss SUSE 10.X |
| 08:08.02 | epaphus | anybody know if that would account as if the member actually answered the call? thus the party hearing that annoying message? |
| 08:11.59 | raden | is there a way to wait till someone talks and finishes talking before a message is playd ? |
| 08:19.27 | *** join/#asterisk coppice (~chatzilla@210.17.219.137) |
| 08:37.45 | tyman | p3nguin: is there a way to run an apparent continuous stream of music while handling a call? |
| 08:38.19 | tyman | p3nguin: ie prompts over top |
| 08:43.45 | *** join/#asterisk joelsolanki (~joelsolan@124.125.149.128) |
| 09:00.21 | raden | OMFG women around wisconsin are retarted most common thing I have seen in profiles for like to dos is play beer pong and party |
| 09:05.54 | tyman | Let's here it for the "retarted" girls! Half of us wouldn't be here without them. |
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| 09:17.09 | cmnky | tyman, amen brother ;) |
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| 11:46.52 | al_nz1 | join #openssh |
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| 12:14.05 | [sr] | hohoho |
| 12:14.28 | [sr] | many many asterisk boards on your shoes as gifts :p |
| 12:19.57 | ariel_ | Morning Marry Christmas everyone |
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| 13:49.11 | coppice | Christmas is fun for one day a year, but I wouldn't want to marry it |
| 13:49.36 | syncer | Marry Chrtistmas to all |
| 13:50.10 | syncer | how possible to check if transcoding is works g711->g729? |
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| 15:34.13 | tacloban | merry christmas |
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| 16:02.56 | raden | wow quiet in here today |
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| 16:23.36 | epaphus | Hello all. Iam creating a queue where a member is a remote dahdi that dials a cell phone. How to control calls passed to it when the cell phone rings to a "cell phone is out of range" automatic message ? |
| 16:26.00 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 16:26.40 | Benwa | HI, i'm trying to install xivo but i get this error : http://paste.debian.net/103173/ (in french, sorry). Any idea ? |
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| 17:03.14 | syncer | how to set 34 no circuit available instead of temp. failure by default? |
| 17:03.21 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
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| 18:01.50 | *** join/#asterisk twanny796 (~twanny@a171.201.adsl.nextweb.net.mt) |
| 18:02.43 | twanny796 | where can I find a trixbox extensions.conf file? |
| 18:03.17 | p3nguin | ~trixbox |
| 18:03.17 | infobot | from memory, trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 18:03.27 | p3nguin | Maybe #trixbox knows. |
| 18:05.07 | twanny796 | infobot: I'm after an extensions.conf file with some functions |
| 18:05.24 | p3nguin | Maybe #trixbox knows. This is #asterisk, an Asterisk channel. |
| 18:06.03 | twanny796 | p3nguin: extensions.conf is asterisk, yes? |
| 18:06.29 | p3nguin | The standard location of extensions.conf in Asterisk is /etc/asterisk/extensions.conf. |
| 18:15.35 | twanny796 | does an ignorepat => 9, mean that the dialling tone will be heard after dialling 9? |
| 18:15.44 | twanny796 | in extensions.conf |
| 18:15.45 | p3nguin | No. |
| 18:15.57 | p3nguin | The phone is responsible for that operation. |
| 18:16.33 | twanny796 | p3nguin: what should I do in the phone? |
| 18:16.52 | p3nguin | But don't do that anyway. There's very rarely a "good" reason to add a 9 on the front of your phone number. |
| 18:16.56 | p3nguin | Just dial the phone number normally. |
| 18:17.02 | p3nguin | Configure an extension that does something with it. |
| 18:17.25 | p3nguin | Usually you will have a pattern match for outbound calls, since you don't know all possible phone numbers that will be called. |
| 18:18.05 | p3nguin | exten => _1NXXNXXXXXX,1,Dial(SIP/itsp/${EXTEN},60) |
| 18:18.31 | syncer | how i can set 34 no circuit available instead of temp. failure for outbound route if all tranks are busy or failed?? |
| 18:19.21 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 18:19.28 | AliRezaTaleghani | hi |
| 18:20.32 | AliRezaTaleghani | how can i get the Queue Max Length (maxlen variable) with a Dialplan? |
| 18:21.10 | twanny796 | p3nguin: using exten => _9NXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1},,R) |
| 18:21.41 | p3nguin | It could be what you're using, but that doesn't make it a good idea. |
| 18:21.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:22.24 | twanny796 | p3nguin: could you please paster what you told me before, cause I cleared the screen |
| 18:22.52 | p3nguin | It's actually kind of 1985-ish to WANT to prefix a phone numbe with additional numbers. |
| 18:22.59 | p3nguin | exten => _1NXXNXXXXXX,1,Dial(SIP/itsp/${EXTEN},60) |
| 18:23.30 | p3nguin | Or using your info, exten => _1NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},60) |
| 18:23.47 | AliRezaTaleghani | HELP :^) |
| 18:23.50 | AliRezaTaleghani | plz |
| 18:25.22 | p3nguin | This allows you to JUST DIAL THE NUMBER YOU WANT TO CALL. |
| 18:25.23 | twanny796 | p3nguin: _1NXXNXXXXXXX, does not make sense to me, out numbering plan is 2XXXXXXX, and 9 to make an outside call |
| 18:25.53 | p3nguin | I made the assumption you are in North America where I am because you didn't tell me otherwise. |
| 18:26.16 | p3nguin | Change the pattern if it is wrong for your national numbering plan. |
| 18:26.27 | p3nguin | Either way, just dial the number you want to call. |
| 18:27.55 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 18:29.35 | p3nguin | Maybe you don't comprehend the concept. I'll be more specific: If I want to call 1 314 969 1077, I am going to dial 13149691077 on my phone; I am NOT going to dial 913149691077 because THAT is not the number I want to call. |
| 18:30.41 | twanny796 | p3nguin: changed to exten => _NXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},,r) |
| 18:31.07 | twanny796 | p3nguin: don't know why I used the 9 in the first place, prob to imitate a PBX |
| 18:31.10 | p3nguin | Are you having problems with ringing sound not working when you call out? If not, drop the r option. |
| 18:31.35 | p3nguin | r adds ringing sound when none exists. |
| 18:31.50 | twanny796 | p3nguin: it's ringing ok |
| 18:32.01 | p3nguin | Then you don't need ,,r in your Dial(). |
| 18:32.22 | p3nguin | That is to add ringing when it isn't already working. |
| 18:33.02 | AliRezaTaleghani | no, idea! :-/ how can i get the Queue Max Length (maxlen variable) with in a Dialplan? |
| 18:34.10 | twanny796 | p3nguin: yep, I removed the r also |
| 18:34.49 | twanny796 | p3nguin: new topic, is there a way to change the telephone tone when there's mail? |
| 18:34.57 | twanny796 | or some other indication |
| 18:35.10 | p3nguin | You might be able to set up stutter dial tone. |
| 18:35.31 | p3nguin | I have a big red light on my phone, so I don't need stutter. |
| 18:35.43 | twanny796 | p3nguin: does that depend on the phone? |
| 18:36.09 | p3nguin | I would think so, but I'm not totally sure. |
| 18:36.41 | twanny796 | p3nguin: how did you set yours? ;) |
| 18:36.56 | p3nguin | I get a stutter dial tone on my phone connected by ATA, but my IP phone has a big red light for message waiting. |
| 18:37.19 | p3nguin | In the ATA, it is configured in the ATA device, not in Asterisk. |
| 18:44.28 | *** join/#asterisk Tim_Toady (~moi@77.49.3.151.dsl.dyn.forthnet.gr) |
| 18:46.23 | atan | twanny796, any chance you started a thread on the alternative voicemail dialtone? |
| 18:46.32 | atan | twanny796, that is something I was wondering about the other day as well. |
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| 18:57.45 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 18:59.28 | p3nguin | atan: If you are using an IP phone, you will have to configure the phone set to play a stutter dial tone when there is a message waiting. It is not done in Asterisk with IP phones. |
| 18:59.54 | atan | p3nguin, I suppose that would make sense. =) |
| 18:59.59 | p3nguin | This is a features of your phone's Message Waiting Indicator (MWI). |
| 19:00.34 | p3nguin | If the phone cannot set the stutter dial tone as MWI, you won't get the stutter. |
| 19:21.01 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
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| 20:30.17 | *** join/#asterisk my007ms (~my007ms@email.msamir.net) |
| 20:31.33 | my007ms | hello , i install g729 and g723 and it work fine |
| 20:31.42 | p3nguin | Nice! |
| 20:32.02 | my007ms | however it's not working when i do transcoding between g711u and g729 |
| 20:32.36 | my007ms | it only work the call coming in g729 and it call endpoint that support g729 |
| 20:32.37 | p3nguin | Then you probably forgot to install your license(s) for the patented codecs. |
| 20:32.47 | my007ms | :( i use the free version |
| 20:32.54 | p3nguin | There is no free version. |
| 20:33.12 | Benwa | :) |
| 20:33.45 | p3nguin | If you got it for free, that's what we call "stolen" property. |
| 20:33.55 | my007ms | no no |
| 20:33.59 | my007ms | let me how you the link |
| 20:33.59 | p3nguin | Yes. |
| 20:34.02 | my007ms | it's free |
| 20:34.40 | my007ms | http://asterisk.hosting.lv/ |
| 20:34.52 | p3nguin | stolen property. |
| 20:35.00 | p3nguin | aka, not free. |
| 20:35.01 | florz | it's neither property nor stolen |
| 20:35.11 | florz | don't tell such propaganda bullshit |
| 20:35.20 | Benwa | my007ms: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. |
| 20:35.26 | florz | it's not licensed, if anything |
| 20:35.59 | my007ms | i have no problem i will buy it if it's really stolen i was think this is free version |
| 20:36.04 | my007ms | as thy provides source code too |
| 20:36.10 | p3nguin | There's no free version. |
| 20:37.01 | my007ms | so asterisk support codec transcoding between g711u and g729 and my problem that i use stolen version |
| 20:37.08 | my007ms | ? |
| 20:37.25 | p3nguin | Normally, it is required that you purchase licenses from Digium and install them in Asterisk to be able to transcode. |
| 20:38.56 | p3nguin | You can use g729 between your devices and your ITSP, but no transcoding will be possible unless Asterisk has been allowed to transcode to/from another codec. |
| 20:40.44 | my007ms | and how i allow this ? |
| 20:41.23 | p3nguin | Purchase the licenses and install them along with the codec they give you. |
| 20:42.31 | my007ms | p3nguin are you sure that site i give to you now " stolen code " |
| 20:42.44 | my007ms | really i have to advice many friend that use it |
| 20:43.01 | p3nguin | Just because it is stolen does not mean it won't work. |
| 20:43.39 | p3nguin | I'm just telling you that it is not legal to use those files for transcoding without having a license to use them. |
| 20:44.52 | my007ms | transcoding consume many CPU ? |
| 20:45.00 | p3nguin | If your question was "Will those files work?", then the answer is "Yes, they have the ability to work." |
| 20:45.12 | p3nguin | Transcoding does use CPU cycles. |
| 20:46.13 | my007ms | i find G.729 in http://www.digium.com/en/products/ however i don't find G 723 |
| 20:46.23 | my007ms | is there any other vendor ? |
| 20:46.37 | my007ms | i need Transcoding from both |
| 20:48.51 | my007ms | i buy one channel for testing first |
| 20:51.27 | p3nguin | Which Asterisk version are you using? |
| 20:53.17 | my007ms | 1.6.2.5 |
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| 21:11.08 | my007ms | p3nguin |
| 21:11.10 | my007ms | [Dec 25 13:25:12] WARNING[17070]: loader.c:392 load_dynamic_module: Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: ast_register_translator |
| 21:11.10 | my007ms | [Dec 25 13:25:12] WARNING[17070]: loader.c:787 load_resource: Module 'codec_g729a.so' could not be loaded. |
| 21:11.30 | p3nguin | yeah |
| 21:11.58 | my007ms | i follow readme and every thing was going fine till this step |
| 21:12.29 | my007ms | when i try load the module i generate the licenses already |
| 21:15.47 | my007ms | p3nguin any idea what is the problem ? |
| 21:17.34 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 21:20.10 | my007ms | can someone pleas help me in register codec_g729 |
| 21:37.08 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
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| 22:02.41 | tyman | p3nguin: thanks for turning me on to flowroute...finally have a plan I like without the bs |
| 22:03.33 | p3nguin | Hope it works out for you. |
| 22:07.20 | *** join/#asterisk epaphus (~epaphus@ec2-50-16-226-177.compute-1.amazonaws.com) |
| 22:07.33 | epaphus | hello. Anybody know of a mirror i get download asterisknow ISO.. since asterisk.org is down |
| 22:10.36 | p3nguin | I probably have a copy of it if you want it. |
| 22:11.12 | p3nguin | Do you have an FTP where I could upload it, or do you need to download it from me? |
| 22:23.53 | *** join/#asterisk syncer (~syncer@opensuse/member/andamasov) |
| 22:24.00 | syncer | hi |
| 22:24.48 | syncer | how can i notify other side of incomming trunk that on my asterisk no available outgoing tranks? |
| 22:25.20 | p3nguin | What? Can you repeat this using terms that I understand? |
| 22:26.19 | syncer | p3nguin: sorry, i'm new in all this stuff |
| 22:26.29 | p3nguin | Ah, it was apparent. |
| 22:26.36 | syncer | i have asterisk 1.6 box |
| 22:26.46 | p3nguin | 1.6.what.what? |
| 22:26.56 | syncer | i have asterisk 1.6.2.15 |
| 22:27.01 | p3nguin | Okay, good. |
| 22:27.22 | syncer | there are 2 sip trunks |
| 22:27.29 | syncer | 1 from provider |
| 22:27.37 | syncer | 2 to gsm gateway |
| 22:27.41 | p3nguin | Okay, there is no such thing as a sip trunk. |
| 22:27.52 | syncer | sorry |
| 22:29.12 | syncer | when call come from 1 and on 2 no available channels, i need decline call so provider can reroute that call by different route |
| 22:29.31 | p3nguin | Asterisk should do that automatically. |
| 22:30.01 | p3nguin | If no channels are available, the call will fail with congestion tones. |
| 22:30.29 | syncer | p3nguin: i guess so, maybe problem with softswitch of provider |
| 22:30.34 | WIMPy | As long as you have a Hangup() in your dialplan. |
| 22:31.07 | syncer | because their tech guy told me that call declinen after 6 seconds |
| 22:31.39 | WIMPy | From where to where? |
| 22:31.55 | syncer | when they pass call to my trunk |
| 22:32.12 | p3nguin | You want the provider to reroute the call when you fail to accept it? |
| 22:32.13 | syncer | and asterisk can't route it because no channels available |
| 22:32.17 | syncer | yes |
| 22:32.22 | p3nguin | That's something they are responsible for. |
| 22:32.27 | *** join/#asterisk Tili (~Tili@cm161.eta193.maxonline.com.sg) |
| 22:33.05 | p3nguin | My ITSP gives me the ability to configure that on their system. |
| 22:33.09 | syncer | p3nguin: i think so too, but they telling that i accept call |
| 22:33.23 | p3nguin | I see. |
| 22:33.34 | WIMPy | Maybe you should show us your dialplan. |
| 22:34.27 | syncer | sure, which part? |
| 22:34.31 | syncer | or all? |
| 22:35.02 | WIMPy | The part that's involved in those calls. |
| 22:35.37 | p3nguin | Start with at least the entire context where the call goes in. |
| 22:37.34 | syncer | i completelly lost( |
| 22:38.44 | syncer | on that machine installed elastix, this is what i got |
| 22:38.51 | syncer | not really happy, but.. |
| 22:40.35 | syncer | http://pastebin.com/XWFncDGf |
| 22:41.05 | WIMPy | There is probably not much hope that this is readable. |
| 22:42.00 | WIMPy | Well, it is, but doesn't say much. |
| 22:42.45 | syncer | should i show dialout-trunk macro? |
| 22:44.22 | syncer | http://pastebin.com/uXybR4Er |
| 22:44.33 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 22:47.07 | WIMPy | Lots of macros and agis. Pretty impossible to say what's going on there. |
| 22:47.16 | *** join/#asterisk Tili (~Tili@cm161.eta193.maxonline.com.sg) |
| 22:49.15 | syncer | maybe provider's switch also stupid or missconfigured, because when i make call for example, it's not accepted, softphone continue show calling while playing message "all service are busy now " |
| 22:49.57 | WIMPy | That clearly means you are not rejecting the call. |
| 22:50.06 | WIMPy | At least not with SIP. |
| 22:51.06 | syncer | this is my problem i guess |
| 22:51.22 | syncer | Progress() can be reason? |
| 22:51.22 | WIMPy | definitely |
| 22:52.10 | WIMPy | Yes, but it's really the announcement. |
| 22:53.01 | syncer | http://pastebin.com/Ap1YxKwM |
| 22:53.07 | syncer | here is call ending |
| 22:53.23 | syncer | at least i think so |
| 22:53.59 | WIMPy | Could be. You could try to comment out that playback. |
| 22:54.52 | p3nguin | Playback() has an implied Answer built in, unless you specify the noanswer option. |
| 22:55.21 | WIMPy | I don't think that would help anyway. |
| 22:55.43 | WIMPy | There is a noanswer. |
| 22:57.22 | syncer | well, i knew that |
| 22:57.27 | syncer | thanks anyway |
| 23:01.31 | syncer | WIMPy: p3nguin can you tell me what should happen with softphone if all is correct |
| 23:01.40 | syncer | it should drop call immediatly? |
| 23:01.56 | WIMPy | yes |
| 23:03.17 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
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| 23:10.39 | syncer | http://pastebin.com/7uzmTKyE |
| 23:11.11 | syncer | strange, but i still have message, but not see any playback in trace |
| 23:11.26 | syncer | how hard to be noob( |
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| 23:28.52 | ChannelZ | Generally speaking PCI Express 2.0 graphics cards aught still work in 1.x slots right just slower? |
| 23:31.44 | syncer | finally if i trying call directly via my asterisk with sofphone, call rejected without immedialty |
| 23:32.09 | syncer | but if i call trough provide, i still hear message |
| 23:44.06 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 23:44.43 | sawgood | Merry X-mas everyone! |
| 23:48.01 | ChannelZ | inded |
| 23:48.06 | ChannelZ | s/inded/indeed/ |
| 23:50.47 | sawgood | How long now has the electronic version of, "Asterisk the Definitive Guide" been out? I see a copyright date of 2011 |
| 23:50.57 | sawgood | this information is very SOLID and a super nice read |
| 23:53.11 | p3nguin | A PCI-E 2.0 card ought to work in a PCI-E 1.0a/1.1 slot. |
| 23:55.16 | ChannelZ | sawgood: the 3rd edition isn't quite out yet but the first two have been out of several years |
| 23:56.50 | sawgood | This is totally different from the "Future of Telephony" |
| 23:56.59 | sawgood | I knew out the other releases but not this new one |
| 23:57.03 | p3nguin | A copyright date of 2011 probably means that someone is very optimistic. |
| 23:57.30 | sawgood | It is more than well done (and written) .. |
| 23:57.32 | sawgood | Its good |
| 23:58.35 | p3nguin | sawgood: Where did _you_ get the electronic copy which you are reviewing? |
| 23:59.08 | sawgood | http://ofps.oreilly.com/titles/9780596517342/index.html |