| 00:00.18 | paulc | Eitan: copy all of /etc/asterisk to the new box, after installing the same version? |
| 00:11.15 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
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| 00:46.55 | titter | Off the wall question ... anyone ever pipe the CLI into a database? |
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| 01:18.28 | cj | is there anything I need to do in order to tell my sip clients to do peer-to-peer RTP rather than going through the asterisk box? |
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| 01:30.56 | p3nguin | Configure the peer definitions in Asterisk to use directmedia (canreinvite). |
| 01:35.07 | Katty | hi |
| 01:35.39 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 01:35.42 | jc319 | hi |
| 01:36.26 | Katty | hi |
| 01:37.48 | jc319 | If I am sure I will use only SIP do I need to compile libpri? (also dahdi but I think I need it in any case, right?) |
| 01:38.08 | Katty | you will need dahdi to make calls, most likely |
| 01:38.35 | Katty | i would go ahead and compile everything, if i were you |
| 01:39.41 | florz | jc319: no, you don't |
| 01:39.44 | Katty | i don't suppose you'd need libpri if you weren't going to use a pri/bri/isdn interface |
| 01:41.25 | jc319 | I compiled asterisk once so far, yesterday, following the guide @ http://astrecipes.net/index.php?n=398 it seemed to be okay as an intro, the only problem is -comparing to the free book- my dialplan had tons of things like EAL. I am trying to start from scratch and this time I intend to make this the permanent box, at least for a while. I even updated CentOS from 4.8 to 5.6 apparently it |
| 01:41.26 | jc319 | takes a lot of time on thin client. |
| 01:41.40 | jc319 | ok including dahdi, but not libpri, thanks |
| 01:43.39 | jc319 | I read the book offline a bit today, apparently I got device/extension/sip 'channel' confused. I have better hopes tonight, setup attempt version 0.2 in progress... |
| 01:54.16 | jc319 | "DAHDI tools installed successfully. ### If you have not done so before, install init scripts with: make config" << is this necessary only if I have a telephony hardware? |
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| 02:01.34 | Seeker2921 | any one willing to asist a newbe on MOH not working? if so, heres the details. freepbx indicates that i have files there. im using the defualt class. cli set to verbose 7 says moh class requested but not loaded. and i have a mp3, gsm, and wav pcm all in the /var/lib/asterisk/moh folder. im at a loss all my conf files: http://pastebin.com/EBbDdbXN |
| 02:01.46 | Seeker2921 | moh show files also shows nothing |
| 02:05.06 | *** join/#asterisk p_masho (~mash@84.45.225.28) |
| 02:07.34 | jc319 | Seeker2921: I don't know anything about it but examples on voip.info.org do not have trailing slash |
| 02:08.43 | p_masho | newbie questions.. I'm considering installing asterisk onto my dedicated server.. to use with the FlightGear flightsimulator .. for voice/atc http://wiki.flightgear.org/FGCom ie the machine has no soundcard/etc for voip only... is it 1) gonna work and 2) does it use a lot of resource ? |
| 02:08.47 | jc319 | official "musiconhold.conf" does not have it either... >> directory=/var/lib/asterisk/mohmp3 |
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| 02:09.58 | Seeker2921 | jc319: tried it. no dice. it would have figuared to be so simple. |
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| 02:20.25 | jc319 | I'm almost in make menuselect step, is there anything I should avoid in particular? so just to recap, for a sip-only deployment I want to keep it skinny by eliminating definitely unnecessary stuff, like ISDN/Hardware related packages). any ideas? [once I get the sip in/outbound working, I want to play with meetme, voicemail etc. so I want to keep feature specific extras] |
| 02:20.51 | jc319 | Do I need EAL package(s) I can check and give the exact name in a few minutes |
| 02:24.19 | ChannelZ | pretty much everything is built as dynamically loadable modules anyway, so rather than configuring out the stuff you think you don't need, just don't load them in your config. |
| 02:27.31 | jc319 | will do thx. when I went with the defaults last time, the dialplan show command in the books shows a few lines, my dialplan show, displayed a lot of stuff, how can I remove them? where do they come from I didn't note exact names but I remember a lot of EALs. that's the reason I was trying to filter out stuff.. |
| 02:35.13 | ChannelZ | I think you mean AEL |
| 02:37.00 | ChannelZ | just don't have an extensions.ael in your config and/or put a noload for pbx_ael.so in your modules.conf |
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| 02:48.41 | jc319 | will do thanks again. |
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| 04:02.23 | jc319 | okay I put "noload => pbx_ael.so" in modules.conf and that pbx_ael's are gone from the dialplan show output. now there are many pbx_config lines, do I need them or can I disable them? |
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| 04:11.49 | ChannelZ | sorry I'm wandering in an out. I'm not sure I understand your question |
| 04:12.42 | ectospasm | jc319: move all the extensions.* files to extensions.*.sample |
| 04:12.54 | ectospasm | also, users.conf if you have it |
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| 04:25.52 | jc319 | Just trying to start with zero extensions, I did with extensions.conf but still have more, ok renaming any other extension.* files now |
| 04:27.51 | ChannelZ | you will probably have a couple which are put in by other things, like if parking lots are turned on in features.conf |
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| 04:28.33 | titter | Off the wall question ... anyone ever pipe the CLI into a database? |
| 04:28.50 | jc319 | ok most is gone but I still have 3 extensions, app_queue / app_dial / features. just noload'ed app_queue, I think I got it know, hunting the remaining two |
| 04:28.58 | ectospasm | titter: I've piped it to a file, with tee |
| 04:29.02 | titter | Ya |
| 04:29.25 | titter | I am thinking tee to a txt, then parse, and insert to a db every 5-10 minutes or so. |
| 04:30.05 | ChannelZ | well you need app_dial |
| 04:30.05 | ectospasm | titter: what are you trying to capture? |
| 04:30.57 | titter | Just the CLI for things you can grab out of logs ... dialing errors, or call forwarding issues. Things users might do that won't show as a warning, but an error on their side. |
| 04:30.58 | ChannelZ | titter: do you need what the console outputs or are you trying to look at what other people might be doing IN the console? |
| 04:31.13 | titter | can't*** |
| 04:31.25 | jc319 | is it safe to remove (rename) features.conf, extensions*.conf, followme.conf? sorry if this is obvious for you but I don't want to break something with this and then waste time trying to fix it later |
| 04:32.00 | ChannelZ | jc319: yes. "break" is a relative term |
| 04:32.19 | titter | It would be interesting to see the CLI filtered as well |
| 04:32.23 | jc319 | [ Context 'parkedcalls' created by 'features' ] << why doesn't this go away, I moved features.conf to a backup/ dir |
| 04:32.35 | ChannelZ | did you reload? |
| 04:32.39 | titter | Basically we are trying to give our support staff overseas access to viewing the CLI without giving them direct access. |
| 04:33.07 | jc319 | I did 'core restart now' is it not greater than reload? |
| 04:33.16 | ChannelZ | It's not in the same format, but you can pretty much send everything to a log already via logger.conf |
| 04:33.40 | titter | Will it show everything similar to the CLI? Ringing, etc. |
| 04:33.50 | ChannelZ | I think there's a means to have the verbosity turned up for it but I might be wrong. |
| 04:34.26 | titter | Ya, this was kind of one of those "can" we do it, and I of course said ... it's not can we, but "how" |
| 04:34.38 | ChannelZ | I think you just have to launch your root asterisk process with the appropriate -v's |
| 04:35.14 | ChannelZ | or I think you can do it with asterisk.conf, setting the verbose to whatever in [options] |
| 04:35.40 | jc319 | ok removed almost all, two more remaining, NoOp [app_dial] and Park() [features] these look dangerous so I'll just leave them, it is blank enough for me now |
| 04:35.47 | ectospasm | titter: use the full log in logger.con |
| 04:35.52 | titter | Ya, I will mess with it a bit tomorrow. |
| 04:35.53 | ectospasm | s/$/f/ |
| 04:36.07 | titter | cool noted |
| 04:37.48 | ChannelZ | I just tried it, seems to work |
| 04:38.19 | ChannelZ | you just get it in more of a log format obviously with timestamps on each line and what module each event came from |
| 04:38.26 | titter | Also working on a CDR website similar to CDRstats that will include some other crap like loss and jitter data. Should be helpful to pull a call log easily from a user quality complaint and see if it was their connection, or if we pull a report for all calls around that time to see if multiple users had the same problems to pin the network issue. Centralized PBX crap is fun. |
| 04:38.30 | ChannelZ | But if you want to get trick you can parse it out however you want for display to the user |
| 04:38.36 | titter | Ya |
| 04:38.43 | ChannelZ | No colors ;) |
| 04:38.55 | titter | I just need to get the data to a DB, then figure how it should look, and let the programmers to their thing lol |
| 04:39.18 | jc319 | awesome, blank config to play with tomorrow. thanks |
| 04:39.19 | ChannelZ | In any event, just turn 'verbose' on for whatever logger.conf entry you want to use this for |
| 04:39.36 | titter | Nice |
| 04:39.56 | ChannelZ | "it's already in there already!" |
| 04:40.37 | titter | realtime for logger.conf ... challenge accepted. |
| 04:41.01 | titter | http://blog.amhill.net/wp-content/uploads/2011/05/challenge-accepted.png |
| 04:41.30 | ChannelZ | heh |
| 04:41.42 | titter | No clue what type of load this will add |
| 04:41.49 | titter | Should be interesting |
| 04:41.59 | titter | The server is overkill for an Asterisk instance |
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| 05:50.08 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 05:50.42 | doolittlework | sip providers in south africa sucks |
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| 05:53.43 | irroot | lol |
| 05:53.52 | irroot | doolittlework indeed |
| 05:54.02 | irroot | can i give you a account ?? |
| 05:54.19 | irroot | or colo and you set up your own ?? |
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| 05:56.47 | doolittlework | i have a box open to them with only the right ports open, someone logged on using my sip details and made a crap load of calls, now they say my server has been comppromised, i know they did not get access to my box |
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| 05:58.17 | irroot | doolittlework it happens |
| 05:58.37 | irroot | check the CDR's |
| 05:58.48 | irroot | the provider may have been compromised |
| 06:00.34 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:00.37 | schmidts | good morning |
| 06:03.25 | remx | Are there any mobile based asterisk managers? I'd really like an app or mobile browser based that uses ajax |
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| 06:18.25 | doolittlework | there is no cdr's |
| 06:18.57 | irroot | ah then insist the ITSP show you the calls and where they originated |
| 06:19.06 | doolittlework | had such a nice day planned now i have to go though system logs |
| 06:19.14 | kaldemar | doolittlework: how are you so sure that your box is not compromised? |
| 06:19.23 | doolittlework | someone from a 201 address registered |
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| 06:53.04 | mandla | Morning Everyone... |
| 06:53.59 | irroot | hey mandla coming right ?? |
| 06:54.15 | mandla | I came right my man. |
| 06:54.29 | irroot | awesome !!! |
| 06:55.22 | mandla | irroot, now im trying to configure so that calls can got outta office via PRI/BRI ports and PSTN, its so confusing. |
| 06:55.54 | irroot | yeah it gets confusing |
| 06:56.02 | irroot | but you will figgure it out |
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| 07:21.53 | mandla | irroot, mmmh, tell me something where can i get a document that has a step by step guide of configuring outgoing call via BRI ports |
| 07:22.19 | irroot | always check voip-info.org |
| 07:22.34 | mandla | irroot, i tried my friend Googly McGoogle, but he says nothing. |
| 07:22.41 | irroot | you need to grasp the concept of contexts and extentions |
| 07:23.07 | irroot | in extensions.conf |
| 07:23.17 | irroot | in "asterisk -r" |
| 07:23.25 | irroot | set "verbose 3" |
| 07:23.32 | irroot | "set verbose 3" |
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| 07:41.28 | *** join/#asterisk Polysics (~Polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:41.31 | Polysics | hello |
| 07:42.41 | Polysics | when calling SIP on SIP, shouldn't echo be impossible? |
| 07:43.00 | Polysics | both of us have headphones and the phone client we have has echo cancellation |
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| 07:46.38 | Cadey | Hi guys - any of you guys install or maintain asterisk in a windows end user enviroment? |
| 07:46.56 | irroot | polsics echo is introduced on a Hybrid/Balun or as a feedback loop |
| 07:47.06 | cneb3000 | Cadey: you mean the server on windows server? or like soft phones on windows machines? |
| 07:47.21 | irroot | if the design of the handset is bad echo is possible in the latter case |
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| 07:48.22 | Cadey | cneb3000 : SOrry, i mean mintain a linux asterisk install but where the clients end user enviroment is windows (AD's - Exchange - Windows) |
| 07:50.22 | Polysics | irroot: i tried calling a machine that has no headset and volume is turned down to 0, yet there is echo on the other side |
| 07:50.42 | Cadey | Cadey : Basicly we have built a suite of windows baised services that interact with AMI which means putting telco related features into win apps easy, it included a AMI proxy, Call stat generation, line monitoring (free, busy, receving) and single phone monitor for a more softphone type interface |
| 07:50.57 | Cadey | lol I put cadey : in my own message, doh |
| 07:51.03 | cneb3000 | haha |
| 07:51.24 | cneb3000 | cadey: isn't that SORT OF what that switchbox thing does? |
| 07:51.48 | Cadey | cadey : kind of but this is way more abstract |
| 07:51.56 | Cadey | lol again! |
| 07:51.58 | cneb3000 | haha |
| 07:52.06 | cadey_ | ;) |
| 07:52.07 | Cadey | lol |
| 07:52.09 | Cadey | thank |
| 07:52.10 | Cadey | s |
| 07:52.24 | Cadey | im going to put them up on codeplex today |
| 07:52.30 | Cadey | so ill give you a shout if your interested |
| 07:52.44 | cneb3000 | i'm always interested in what you're up to! |
| 07:52.55 | Cadey | lol |
| 07:53.22 | irroot | straw poll opensips vs kamalio |
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| 07:55.14 | Polysics_ | ls |
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| 08:14.38 | cneb3000 | irroot: i set up a bay to play wth kamalio.. but currently use openSIPS.. not sure if that counts as a vote :) |
| 08:15.07 | irroot | using kamalio ATM busy building opensips |
| 08:15.12 | Polysics | we have some sort of "bell" or "cymbal" sound on the line |
| 08:15.21 | Polysics | SIP to SIP using headphones |
| 08:15.29 | remx | In CLI, one trunk is showing "OK" under "Status" column, yet the rest are showing as Unmonitored. What is the difference between OK and Unmonitored?] |
| 08:15.44 | Polysics | can i do ANYTHING on * to help? |
| 08:15.51 | Polysics | or is it a client-only problem? |
| 08:19.31 | kaldemar | remx: one is monitored (qualify option in sip.conf is enabled) and responds, the rest are not monitored (qualify is not enabled). |
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| 08:25.12 | remx | kaldemar: thank you kindly |
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| 08:35.06 | ectospasm | Polysics: when does this unwanted sound play? |
| 08:35.21 | Polysics | ectospasm: at random during a call |
| 08:35.35 | Polysics | it does sound like a byproduct of echo |
| 08:35.41 | Polysics | it is probably the client's fault |
| 08:35.50 | Polysics | we are using a web-based Java phone by Mizutech |
| 08:35.54 | ectospasm | these are SIP users and trunks? |
| 08:36.01 | Polysics | yes, only SIP |
| 08:36.15 | ectospasm | yeah, echo on VoIP-only is usually acoustic echo |
| 08:36.35 | Polysics | if user A calls user B, and user A is on X-Lite while user B is on Webphone, only user A hears his own voice |
| 08:36.41 | ectospasm | ...which means that the mic is picking up the sound from the speakers, or from sounds in the ambient room |
| 08:37.47 | Polysics | to make things worse, if the above scenario is performed with user B having muted ALL sound, user A still hears echo |
| 08:37.58 | ectospasm | Polysics: in that case, it's probably the shoddy web phone that's retransmitting A's voice |
| 08:37.59 | Polysics | i would say the Webphone is doing something strange with the audio channels |
| 08:38.39 | ectospasm | Polysics: what if A calls C, another X-lite phone? |
| 08:39.11 | remx | Would qualify=yes prevent asterisk from re-connecting to a trunk after no response from peer? |
| 08:39.46 | ectospasm | remx: it will usually reconnect automatically. Qualify doesn't inhibit connection |
| 08:40.38 | remx | so it's just for information purposes? |
| 08:41.03 | ectospasm | it's so we don't try to send the endpoint a call when it's unreachable |
| 08:41.12 | remx | ahh i see |
| 08:41.15 | remx | thank you |
| 08:41.19 | ectospasm | qualify ensures that we know it's unreachable when it becomes so |
| 08:41.37 | ectospasm | otherwise, you try to send the call, and you get no response |
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| 08:41.57 | ectospasm | potentially retransmitting messages when the peer isn't available. |
| 08:42.19 | Polysics | ectospasm: X-Lite on X-Lite has totally zero problems |
| 08:43.28 | ectospasm | Polysics: then investigate (or ditch) Mizutech web phone. |
| 08:44.10 | Polysics | ectospasm: could you suggest an alternative, please? |
| 08:44.20 | Polysics | we need a web-based phone we can embed in a web page |
| 08:44.38 | Polysics | java would be better, and it needs a tunneling service to work ove r HTTP proxy |
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| 08:45.04 | ectospasm | Polysics: sorry, I have zero experience with many softphones, web-based or no |
| 08:45.05 | Polysics | we are probably better off asking the Mizu people to solve the echo problem, but i would gladly accept an alternative |
| 08:47.02 | *** part/#asterisk Polysics (~Polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
| 08:54.00 | remx | I have a local extension that has qualify=yes but in CLI I see the following |
| 08:54.10 | remx | 1001 (Unspecified) D A 0 UNKNOWN |
| 08:54.10 | remx | 29 sip peers [Monitored: 28 online, 1 offline Unmonitored: 0 online, 0 offline] |
| 08:54.51 | schmidts | remx the UNKNOWN state is cause asterisk didnt know where to find this peer (Unspecified) and so it can not ping it |
| 08:55.40 | remx | schmidts: I haven't connected to it so is the "UNKNOWN" and "1 offline Unmonitored" correct? |
| 08:55.47 | remx | I was worried I configured it incorrectly |
| 08:56.04 | remx | The extension is offline |
| 08:56.31 | schmidts | remx yes its correct |
| 08:57.41 | remx | I got confused because there's no period after offline :P |
| 09:00.05 | kaldemar | remx: UNKNOWN is a state when asterisk does not yet know whether the peer is OK or UNREACHABLE. |
| 09:00.17 | *** join/#asterisk davlefou (~david@41.225.89.192) |
| 09:06.06 | *** part/#asterisk NourSs (~gholzinge@LAubervilliers-151-13-22-64.w217-128.abo.wanadoo.fr) |
| 09:07.47 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:a957:70c6:73fa:4e8c) |
| 09:11.51 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 09:12.56 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 09:13.30 | mandla | Hey guys i cant seem to get my analog phone call outside via PSTN, help. |
| 09:13.59 | cneb3000 | mandla: what happens when you try to dial out on the phone? |
| 09:15.03 | mandla | cneb3000, There is nothing happening, when a press even 1 number it seems engaged. |
| 09:15.20 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 09:15.25 | cneb3000 | mandla: could you copy and paste the CLI output when you do this to pastebin.com? |
| 09:16.18 | mandla | cneb3000, what should i do? run pastebin.com on CLI? |
| 09:16.52 | cneb3000 | mandla: do you know how to access the Asterisk CLI? |
| 09:17.10 | mandla | yah. |
| 09:17.26 | mandla | asterisk -r, right? |
| 09:17.28 | cneb3000 | mandla: ok, so what you do is access the asterisk CLI (asterisk -r). |
| 09:17.32 | cneb3000 | yep :) |
| 09:17.44 | cneb3000 | then on the analog phone, push the 1 key or whatever |
| 09:17.51 | cneb3000 | and a bunch of text should fly on the cli |
| 09:18.10 | cneb3000 | (if it doesn't... it probably means you dont have logging on... type 'core set verbose 3' to turn it on when in the CLI |
| 09:18.18 | cneb3000 | anyway, copy paste that output to pastebin.com |
| 09:18.24 | cneb3000 | does that make sense? |
| 09:18.48 | mandla | yah, cant i paste it here/ |
| 09:19.01 | cneb3000 | it might be quite a few lines :D |
| 09:19.15 | cneb3000 | i think the rule of thumb, is if it's longer than 3 lines, use pastebin.com |
| 09:19.35 | cneb3000 | if asterisk for some reason only gives 1 line, then sure paste it here |
| 09:21.08 | *** join/#asterisk davlefou (~david@41.227.52.150) |
| 09:22.40 | mandla | cneb3000, iv pasted, note that where it says Ringing, i was calling a FXS port on the Astribank. |
| 09:22.52 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 09:23.01 | cneb3000 | what's the link mandla? |
| 09:23.08 | cneb3000 | to the pastebin paste |
| 09:23.31 | mandla | http://pastebin.com/31KYLWVG |
| 09:23.38 | mandla | cneb3000, http://pastebin.com/31KYLWVG |
| 09:24.20 | *** join/#asterisk jacobkiers (~jacobkier@82-168-215-74.ip.telfort.nl) |
| 09:24.49 | tzafrir | mandla, DAHDI/13 and DAHDI/14 are FXS phones? |
| 09:25.01 | mandla | Yah. |
| 09:25.28 | mandla | tzafrir, yah. |
| 09:25.43 | tzafrir | DAHDI/13 called DAHDI/14 . Did you pick up DAHDI/14 ? |
| 09:26.02 | *** join/#asterisk Polysics (~Polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
| 09:26.22 | tzafrir | hmm... is DAHDI/14 connected to an analog phone? |
| 09:27.27 | *** join/#asterisk sekil (~sekil@91.143.215.198) |
| 09:28.03 | mandla | tzafrir, nope, i was just calling a port, thats besides the point, the thing is i cant get DAHDI/13 to call outside the office. |
| 09:28.05 | *** join/#asterisk coppice (~chatzilla@m121-202-89-162.smartone-vodafone.com) |
| 09:29.12 | tzafrir | I'm confused. Where do you have a trace of DAHDI/13 calling outside? |
| 09:29.12 | mandla | tzafrir, like call other anolog phone at other companies. |
| 09:30.36 | mandla | tzafrir, thats the thing, it cnt call outside, because, when i try to dial on DAHDI/13 there is the tone, which does not allow me to even punch a number to dial. |
| 09:31.32 | tzafrir | So the issue is not with calling outside |
| 09:31.43 | tzafrir | It is with the local dialplan |
| 09:31.55 | tzafrir | Can you call anything else from that phone? |
| 09:32.54 | *** join/#asterisk flyman_ (~flyman_@mail.nationalfonds.org) |
| 09:33.17 | mandla | From DAHDI/13 i can call a x-lite softphone on my other laptop, and call a iphone hosting Bria. |
| 09:34.38 | mandla | Internally it seem to work well, on my network, but i think um missing something in trying to route call out via PSTN |
| 09:36.21 | mandla | Lines 9-12 shows when i was trying to call with DAHDI/13. |
| 09:40.16 | cneb3000 | to anyone thing of learning perl. spend some time going through this book, it's great ---> http://www.perl.org/books/beginning-perl/ |
| 09:40.27 | cneb3000 | (good for those who have never coded before) |
| 09:40.35 | cneb3000 | to anyone thinking of learning perl* |
| 09:40.47 | mandla | Are there any settings i should be doing on Asterisk to ensure that call can go out? |
| 09:41.37 | mandla | Perl, thats one thing i seem to know best. |
| 09:43.21 | *** join/#asterisk catphish (~catphish@charlie.office.atechmedia.net) |
| 09:43.58 | catphish | is there any way to use call parking on an asterisk instance shared by different organizations? |
| 09:45.28 | *** join/#asterisk davlefou (~david@196.203.145.203) |
| 09:46.37 | cneb3000 | mandla: sorry i cant help with your problem. but on the note of perl, it's great so far. what sort of stuff have you coded in the past? |
| 09:50.07 | mandla | Some port knocking projects, for my thesis. |
| 09:52.33 | cneb3000 | ooo interesting! |
| 09:54.08 | cneb3000 | how did you learn mandla? from books/internet or school? |
| 09:54.37 | mandla | internet. |
| 10:07.14 | *** join/#asterisk davlefou (~david@41.227.58.194) |
| 10:09.52 | tzafrir | mandla, what type of trunk? BRI? |
| 10:10.46 | mandla | BRI yes. |
| 10:13.46 | mandla | tzafrir, you there? |
| 10:15.01 | tzafrir | so, on what DAHDI ports? |
| 10:21.29 | mandla | I have one analog phone on DAHDI/13 (FXS port) |
| 10:22.14 | mandla | And i line from the PSTN on BRI_TE port on the Astribank. |
| 10:22.29 | mandla | And 1 line from the PSTN on BRI_TE port on the Astribank. |
| 10:22.34 | tzafrir | mandla, DAHDI ports 1 and 2? |
| 10:23.55 | mandla | Port B1 for the analog phone and Port BRI-1 for the line from the PSTN |
| 10:24.04 | catphish | is there a method to detect if a call is parked in a space or not? |
| 10:24.32 | mandla | catphish, I havent checked that. |
| 10:24.49 | mandla | Im still trying to get calls to go outside. |
| 10:24.52 | mandla | First |
| 10:26.21 | tzafrir | mandla, if so, try running the following from the asterisk CLI: |
| 10:26.47 | tzafrir | originate DAHDI/13 application Dial DAHDI/1/1234567890 |
| 10:27.01 | tzafrir | replace 1234567890 with the actual number you want to dial |
| 10:30.17 | Rufus | greetings. How can one an external SIP user, if the user is using a dinamic IP? |
| 10:30.56 | Rufus | *how can one add ... |
| 10:32.03 | mandla | tzafrir, ok what it does, DAHDI/13 rings and when i pick it out it rings on my mobile phone, i replaced 1234567890 with my mobile phone number. |
| 10:34.31 | ectospasm | Rufus: host = dynamic in users.conf, also you'll probably want to set nat = yes, and in the [general] section put localnet and externip... |
| 10:34.37 | ectospasm | er, not users.conf |
| 10:34.39 | ectospasm | sip.conf |
| 10:34.49 | ectospasm | ...well, it could go in users.conf, if you're using that |
| 10:35.03 | Rufus | thank you ectospasm. and except 5060, is there any other port I should allow in iptables? |
| 10:35.24 | ectospasm | Rufus: all the ports defined in rtp.conf |
| 10:35.34 | ectospasm | by default, 10000-20000 |
| 10:36.04 | Rufus | so all of those ? |
| 10:38.06 | tzafrir | mandla, right. So the devices work, and now you need to fix your dialplan |
| 10:38.30 | *** join/#asterisk MariusAgon (~aa@84.15.44.250) |
| 10:38.38 | ectospasm | Rufus: yes |
| 10:40.01 | Rufus | ectospasm I'm using asterisk 1.4.40. Could it be I need smth different than host=dynamic ? |
| 10:40.10 | *** join/#asterisk iulhk (~iulhk@119.152.224.91) |
| 10:40.14 | iulhk | hi all |
| 10:41.10 | ectospasm | Rufus: I dunno, what does SIP debug say when the external phone tries to connect? |
| 10:41.21 | ectospasm | connect/register |
| 10:42.24 | iulhk | using asterisk-1.4, call has been connected, how to disconnect the call after some specified time? anybody has any idea please ? |
| 10:42.25 | CaptainPants | [ASTERISK-1] [Status: Closed] SIP re-invites failing with certain proxies - https://issues.asterisk.org/jira/browse/ASTERISK-1 |
| 10:43.06 | Rufus | hmm moment ectospasm, iptables may be filtering me |
| 10:43.16 | mandla | tzafrir, you know what the problem is, the problem is that from DAHDI/13 i cant dial to make call, upon pressing a single digit, it gives back i engaged tone. |
| 10:43.43 | ectospasm | mandla: sounds like your dialplan doesn't have an extension that handles that digit. |
| 10:44.01 | ectospasm | mandla: does the CLI say anything about invalid number dialed, but no invalid handler? |
| 10:44.51 | mandla | ectospasm, i used the default dialplan, for Asterisk. |
| 10:45.04 | ectospasm | mandla: that means f' all to me |
| 10:45.12 | ectospasm | mandla: what does the CLI say when the call fails? |
| 10:46.34 | kaldemar | mandla: the sample dialplan is really just samples and unusable. remove it and make a working one instead. |
| 10:46.37 | mandla | ectospasm, From the CLI using this command originate DAHDI/13 application Dial DAHDI/1/1234567890, there is no invalid number error. |
| 10:46.59 | ectospasm | mandla: no, don't use the originate command, that's not testing your dialplan |
| 10:47.02 | *** join/#asterisk coppice (~chatzilla@m121-202-16-33.smartone-vodafone.com) |
| 10:47.16 | ectospasm | when you actually pick up an endpoint and try to call out, what does the CLI say? |
| 10:47.40 | mandla | ok let me try. |
| 10:50.14 | mandla | <PROTECTED> |
| 10:50.15 | mandla | <PROTECTED> |
| 10:50.15 | mandla | <PROTECTED> |
| 10:50.15 | mandla | [Jun 10 12:48:01] WARNING[8059]: channel.c:2556 ast_waitfordigit_full: Unexpected control subclass '9' |
| 10:50.15 | mandla | [Jun 10 12:48:02] WARNING[8059]: channel.c:2556 ast_waitfordigit_full: Unexpected control subclass '9' |
| 10:50.15 | mandla | [Jun 10 12:48:04] WARNING[8059]: channel.c:2556 ast_waitfordigit_full: Unexpected control subclass '9' |
| 10:50.17 | mandla | [Jun 10 12:48:07] WARNING[8059]: channel.c:2556 ast_waitfordigit_full: Unexpected control subclass '9' |
| 10:50.18 | mandla | <PROTECTED> |
| 10:50.20 | mandla | <PROTECTED> |
| 10:50.22 | mandla | <PROTECTED> |
| 10:50.24 | mandla | <PROTECTED> |
| 10:50.27 | mandla | <PROTECTED> |
| 10:50.28 | mandla | <PROTECTED> |
| 10:50.33 | mandla | <PROTECTED> |
| 10:50.35 | ectospasm | mandla: don't paste here |
| 10:50.35 | mandla | ----------------------------- |
| 10:50.36 | mandla | Thats what i get. |
| 10:50.46 | mandla | Sorry about that. |
| 10:50.55 | mandla | But thats what i get. |
| 10:50.59 | Rufus | ok that seemed to work ectospasm. Thanks you |
| 10:51.50 | ectospasm | mandla: what kind of device is DAHDI/13? |
| 10:52.13 | mandla | Um only able to make call to X-lite softphone in the network and iphone hosting Bria. |
| 10:52.30 | mandla | DAHDI/13 is an analog phone. |
| 10:52.55 | ectospasm | so when the phone picks up, you get a dialtone, but nothing happens when you dial? |
| 10:53.16 | ectospasm | Rufus: you're welcome. |
| 10:54.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:54.40 | mandla | ectospasm, when i pickup there is a dial tone, but when i try to call for example my mobile phone, when i click a 1 it produces an engaged tone. |
| 10:55.22 | ectospasm | mandla: are you able to make the call to the mobile phone? |
| 10:56.07 | mandla | ectospasm, no, but through that command i can. |
| 10:56.47 | ectospasm | mandla: and do you expect to be able to use that command in practice? |
| 10:57.07 | ectospasm | mandla: the originate command just tests to make sure the pieces are working OK |
| 10:57.09 | mandla | ectospasm, my mobile # is 71520005, but immediately upon pressing on 7, it produces the engaged tone. |
| 10:57.28 | ectospasm | mandla: right, look at the CLI when you press that 7 |
| 10:57.43 | ectospasm | mandla: have you edited the default configuration at all? |
| 10:57.56 | ectospasm | ...extensions.conf |
| 10:58.06 | ectospasm | (et. al) |
| 10:58.33 | mandla | ectospasm, i havent edited it at all. |
| 10:58.42 | ectospasm | mandla: then it's not going to work |
| 10:58.58 | ectospasm | mandla: what context is listed for that FXS channel in chan_dahdi.conf? |
| 11:00.07 | Rufus | are there any good iPhone SIP clients? preferably over both wifi and 3g? |
| 11:01.36 | cneb3000 | Rufus: I've used 3cxphone from time to time. |
| 11:01.44 | cneb3000 | Rufus: does the job! |
| 11:01.53 | Rufus | thank you cneb3000, I'll give it a shot |
| 11:02.11 | Rufus | i'm using the windows version so far of it ;) |
| 11:02.23 | cneb3000 | ahhh ;) |
| 11:05.48 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:05.48 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:06.00 | Chainsaw | Rufus: I use the Acrobits softphone. |
| 11:06.06 | Chainsaw | Rufus: Which appears to work. |
| 11:06.26 | Rufus | checking... ty Chainsaw ;) |
| 11:06.50 | Rufus | uh... 6$... darn |
| 11:06.50 | Chainsaw | Rufus: Didn't work very well over 3G, but I suspect that's my provider O2 "accidentally" breaking it. |
| 11:07.03 | Chainsaw | Rufus: iPhone apps cost money. This is not news. |
| 11:07.35 | Rufus | I hate apple for not allowing demos/trials |
| 11:08.33 | cneb3000 | Rufus: it's up to app developer to allow demos/trials ;) |
| 11:09.02 | cneb3000 | Rufus: silly example.. but only one i can think of. you can get a 'lite' version of the angry birds game for free. or buy the full version for money |
| 11:09.23 | Rufus | cneb3000 yeah, that'd be awsome with other aps to |
| 11:09.36 | Rufus | I think developers hide behind apple stopre though, re the free trials |
| 11:09.42 | cneb3000 | Rufus: Demos are certainly good idea i think. |
| 11:09.45 | cneb3000 | Rufus: Demos are certainly good idea i think |
| 11:09.50 | cneb3000 | ^^ yea maybe |
| 11:09.59 | Rufus | <-- mails steve heh |
| 11:10.13 | *** join/#asterisk wonderworld (~ww@port-92-201-90-205.dynamic.qsc.de) |
| 11:10.13 | cneb3000 | hehe |
| 11:13.30 | Chainsaw | cneb3000: Angry Birds is worth the money though. Well, the original. Not the 25 themed ones. |
| 11:14.12 | cneb3000 | Chainsaw: Pft!.. it's all about Fruit Ninja. |
| 11:14.15 | *** join/#asterisk acidfu (~nib@2001:470:1f0f:bb0:21e:37ff:fe16:6aa0) |
| 11:14.34 | mandla | Rufus, Bra is good. |
| 11:14.54 | mandla | Rufus, Brai is good. |
| 11:15.02 | Rufus | Host 'xx.xx.xx.xx' does not implement 'NOTIFY' <-- what's than? |
| 11:15.06 | mandla | Rufus, Bria is good. |
| 11:15.17 | Rufus | lol mandla ;) I'll check it too |
| 11:16.36 | mandla | ectospasm, what exactly should i look for under context. |
| 11:19.01 | ectospasm | mandla: do you understand what context= means in chan_dahdi.conf? It's basically where in the dialplan/extensions.conf a call originating from that endpoint will begin processing |
| 11:19.50 | ectospasm | so, if you have context=from-internal in chan_dahdi.conf, you should have a corresponding [from-internal] in extensions.conf. |
| 11:20.18 | ectospasm | mandla: have you read the book? |
| 11:20.22 | ectospasm | ~thebook |
| 11:20.22 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 11:21.10 | Rufus | mkay this is weird. any ideas why and external sip ( not from within the network) would only hear voice, but his doesn't go through? I checked mic |
| 11:21.24 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 11:21.36 | schmidts | sorry for this stupid question, which doesnt belong to asterisk at all |
| 11:21.44 | schmidts | can someone help me with a regex? |
| 11:22.23 | *** join/#asterisk irroot (~irroot@dsl-185-122-31.dynamic.wa.co.za) |
| 11:22.37 | irroot | irback |
| 11:23.36 | schmidts | i have this row in my log file (or more of this) and i want to get the DST and SRC ip: Jun 10 13:00:02 test-00 kernel: MYSQL IN=eth0 OUT= MAC=00:13:21:ae:68:8f:00:18:51:57:d0:3f:08:00 SRC=111.111.111.111 DST=222.222.222.222 LEN=60 TOS=0x00 PREC=0x00 TTL=64 ID=36203 DF PROTO=TCP SPT=47363 DPT=3306 WINDOW=5840 RES=0x00 SYN URGP=0 |
| 11:23.47 | ectospasm | Rufus: did you set nat = yes for that SIP peer like I said? |
| 11:23.53 | *** join/#asterisk Tim_Toady (~moi@195.74.224.42) |
| 11:24.03 | ectospasm | Rufus: one-way audio is a common SIP/NAT problem. |
| 11:25.53 | ectospasm | Rufus: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 11:27.56 | *** part/#asterisk jacobkiers (~jacobkier@82-168-215-74.ip.telfort.nl) |
| 11:28.00 | Rufus | yes nat is set as yes. reading your article now ectospasm |
| 11:32.15 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
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| 11:34.28 | kaldemar | schmidts: DST=(.*?)\s and SRC=(.*?)\s |
| 11:38.46 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 11:45.21 | irroot | anyone got opensips and MWI workin ?? |
| 11:59.35 | orn | schmidts: What programming langugage? |
| 12:00.43 | orn | schmidts: You might want to consider using awk instead if the fields are always in the same place |
| 12:03.50 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:05.59 | wdoekes2 | schmidts: depends on the regex-type, for basic (e.g. sed): s/.* SRC=\([^ ]*\) DST=\([^ ]*\) .*/\1,\2/ |
| 12:06.42 | *** join/#asterisk bchia (~Adium@nat/digium/x-sssfmeefqvvzrxrc) |
| 12:08.43 | wdoekes2 | irroot: one of our customers added a handle_publish() to the from_asterisk route block and was content with it (the users register and subscribe to the opensips) |
| 12:09.12 | irroot | cool thx |
| 12:09.28 | irroot | this is project .next i need to set up a ITSP |
| 12:09.43 | irroot | with centrix pbx functionality |
| 12:10.20 | irroot | opensips/kamalio will be doing the reg and inter domain stuff while asterisk will be doing the media bits |
| 12:16.48 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
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| 12:53.24 | riksta | Hi there, im trying to increment a variable in 1.6 exten => s,1,Set(BUSYCOUNT=$[1 + ${BUSYCOUNT}]) but I get an error "syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: 1 + " so it seems it doesn't parse out the variable when trying to add it. Can anyone assist please? |
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| 12:54.48 | riksta | my bad .. i forgot to initialise it! |
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| 12:56.17 | Cadey | Any windows admin interested in a suite of tools to help intergrate your asterisk system into windows applications? |
| 12:57.09 | *** join/#asterisk sereal (~sereal@2001:4830:116e:1:1e6f:65ff:fec3:cd0d) |
| 13:01.51 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 13:02.02 | sereal | Can someone point me to some information on programming the polycom 'softkeys' or explain to me a bit as to the process. I have been going threw the documentation but It seems like i'm missing something. |
| 13:03.24 | Chainsaw | sereal: Ah, you're finding that your settings in sip.cfg are ignored? |
| 13:03.38 | *** join/#asterisk irroot (~irroot@197.171.29.66) |
| 13:03.39 | Chainsaw | sereal: I had that. There's a feature flag you need to set (change from 0 to 1). |
| 13:03.43 | sereal | no, I just can't seem to figure out which buttons point to what. |
| 13:03.47 | sereal | I have some soft keys working |
| 13:03.59 | sereal | I have the feature flag, but it seems like i'm missing something where I figure out how to remap a key |
| 13:04.09 | sereal | like for instance I don't need a 'qsetup' on these phones |
| 13:04.41 | sereal | or like a 'MyStat' button |
| 13:05.29 | sereal | like I seem to be missing something in either the documentation or maybe it's just not there |
| 13:05.51 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 13:06.02 | sereal | <softkey softkey.3.enable="1" softkey.3.label="test3" softkey.3.action="blabla" softkey.3.use.idle="1"/> |
| 13:06.06 | jc319 | hello |
| 13:06.10 | sereal | but like what does softkey.3 refer to? |
| 13:06.30 | sereal | and how do I program the keys in the context sensitive stuff |
| 13:06.43 | Chainsaw | It can refer to anything you like, it depends on what action you pick for it. |
| 13:07.10 | Katty | morning |
| 13:07.17 | Chainsaw | ltns Katty :) |
| 13:07.25 | Katty | hugs Chainsaw |
| 13:07.33 | Chainsaw | Katty: *hug* How've you been? |
| 13:07.34 | Katty | Chainsaw: what you been up to, trouble? |
| 13:07.49 | sereal | so can I do something like softkey.qsetup or something similar? |
| 13:07.50 | Chainsaw | Katty: Oh yes. Patched a crasher bug though. |
| 13:07.56 | Katty | Chainsaw: ok i spose. some good days, some bad days. |
| 13:07.59 | Chainsaw | Katty: So at least I did something useful lately. |
| 13:08.06 | Katty | Chainsaw: most excellent |
| 13:08.19 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 13:08.42 | Chainsaw | Katty: Turned my T38 support off, which seems to have stopped the "it just sits there" bug after a week. |
| 13:08.52 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 13:09.06 | Katty | Chainsaw: fascinating |
| 13:09.40 | irroot | hey katty want a nother beer ?? zamalek this time ?? |
| 13:09.41 | Chainsaw | Katty: I can tell it's happened if "sip reload" just gives the prompt back. |
| 13:10.21 | sereal | is the polycom documentation all I really need? I have been struggling to find any examples of people's phone configs where they remapped the softkeys to do stuff |
| 13:10.58 | sereal | I only have a IP450 at the moment to test on, but from what I understand the configs are very similar for other phones like the 650 |
| 13:11.10 | Katty | irroot: i've never heard of zamalek, is it a lager or an ale? |
| 13:11.33 | Katty | irroot: perhaps a better question is how bitter does it taste? |
| 13:11.35 | irroot | its the local name for carling black label |
| 13:11.46 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 13:11.50 | hetii | Hello :) |
| 13:11.54 | Katty | morning |
| 13:12.09 | *** join/#asterisk maxJadi (~maxJadi@pontarius/mahdi) |
| 13:12.10 | hetii | please check my last logs about t38modem: http://pastebin.com/vhmE5MAC |
| 13:12.29 | irroot | dear customer please do not f**k with the lock or DND options |
| 13:12.42 | hetii | maybe some of you will had some idea what i can do else to run it on correct way. |
| 13:13.02 | Katty | so i'm going camping soon. |
| 13:13.23 | Katty | july1st-3rd |
| 13:14.11 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 13:15.44 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 13:15.55 | irroot | Katty enjoy it .... |
| 13:16.34 | irroot | http://www.oppikoppi.co.za/ <- may be going here camp / music festival |
| 13:17.15 | Cadey | http://amiproxy.codeplex.com/ |
| 13:17.37 | *** join/#asterisk maxJadi (~maxJadi@pontarius/mahdi) |
| 13:20.02 | irroot | hetii looks good |
| 13:20.11 | irroot | use mincom or similar |
| 13:20.18 | irroot | try ATD..... |
| 13:20.28 | irroot | and see output |
| 13:21.56 | neurosys | atom vs celeron for *... hmm.. |
| 13:22.32 | hetii | just ATD or ATD with args ? |
| 13:23.21 | irroot | ATD<NUMBER> |
| 13:23.28 | irroot | its the dial command |
| 13:23.55 | irroot | nuurosys i does favor atom |
| 13:24.08 | *** join/#asterisk anderson (~user@c-76-18-219-103.hsd1.fl.comcast.net) |
| 13:25.19 | *** join/#asterisk jonjonjon (~dka@dslb-188-097-083-254.pools.arcor-ip.net) |
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| 13:26.06 | neurosys | irroot: any particular reason why |
| 13:26.19 | irroot | deployed a few |
| 13:26.45 | irroot | the celeron concept of "wartered" down architecture is a turn off |
| 13:27.53 | hetii | this is what i got after type ATD100230 http://pastebin.com/eq7nfZ6N |
| 13:28.43 | hetii | OpalManCould not route a="modem:100600", b="100230@+/dev/ttyT38-0, call=Call[C73780eb87] |
| 13:30.59 | irroot | hetii look line 11 |
| 13:31.12 | irroot | it looks good |
| 13:31.28 | irroot | now do a sip debug in asterisk and see what is happening there |
| 13:33.04 | hetii | first of all this message is before about those route information, second on * i dons see any sip debug information |
| 13:33.35 | hetii | ok maybe not nothign but just this |
| 13:33.56 | hetii | http://pastebin.com/RpNLYvPJ |
| 13:34.42 | hetii | and about * configuration i use freepbx and inside i configure extension for this modem and set some password |
| 13:35.24 | hetii | but as you can see on my first pastebin log i don`t do any king of registration so on this stage i don`t belive that asterisk should do something with this call |
| 13:40.09 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
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| 13:41.46 | nellyb | hi guys, anyone around yet? i need some paid help with asterisk 1.6 dialplan. |
| 13:42.17 | nellyb | or even recomiling chan_sip.c |
| 13:42.43 | hetii | nice:> |
| 13:45.27 | nellyb | very helpful, hetii! |
| 13:51.51 | MariusAgon | Are there in Asterisk any simple options to make a dialplan, who counts, how many times one number called and if it called more than six times a day it'll be blacklisted? |
| 13:52.11 | Katty | i think you'd have to store a number in a database |
| 13:52.17 | Katty | in then do a query |
| 13:52.27 | Katty | so, as far as i know, i don't know of a Simple, Easy way |
| 13:52.32 | Katty | unless you're good with that sort of thing |
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| 13:52.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:54.41 | MariusAgon | You still have to clean a database every day |
| 13:55.11 | Katty | yep. pretty messy. |
| 13:55.17 | Katty | perhaps someone knows another way to do it tho. |
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| 13:56.11 | nellyb | Katty: how many numbers are we talking? |
| 13:56.26 | nellyb | Sorry - MariusAgon |
| 13:56.34 | Katty | ELEVENTY BILLION |
| 13:56.41 | nellyb | :-) |
| 13:57.32 | MariusAgon | It's ok |
| 13:57.33 | Deeewayne | MariusAgon, its not exactly 'in Asterisk', but I like to let Asterisk do what Asterisk does well and put stuff like that in Java (use Asterisk-Java) |
| 13:57.34 | MariusAgon | :) |
| 13:58.14 | MariusAgon | Deeewayne: thank you |
| 13:58.32 | sunfone | MariusAgnon - as far as a database goes to use from the dialplan, you could use the asterisk DB... just a simple key pair |
| 13:58.54 | sunfone | shell scripts can access it from CRON to "clean" it nightly |
| 13:59.01 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:59.01 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:04.18 | Deeewayne | bear hugs the Katty |
| 14:04.27 | MariusAgon | i know i just was wondering how scalable it would be |
| 14:05.16 | sunfone | The asterisk DB? |
| 14:05.52 | MariusAgon | and the problem is that in my situation nightly cron jobs to clean db wouldnt be suitable |
| 14:06.04 | MariusAgon | jap |
| 14:06.15 | MariusAgon | asterisk asterisk berkeley db |
| 14:07.14 | *** join/#asterisk Diffen (~diffen@c-fc73e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 14:07.26 | Deeewayne | add a 'dont call before' field; if the count is more than 6 update 'dont call before' w/ an epoch of 'tomorrow'; if you check a number that has a 'dont call before' less than 'now', clear it |
| 14:07.57 | jonumts | Hi, I'd like to talk about a feature request. |
| 14:08.22 | jonumts | Who can I talk to? |
| 14:08.30 | *** join/#asterisk cHarNe2 (thorn@newelite1.bshellz.net) |
| 14:08.37 | irroot | ~ask |
| 14:08.38 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:09.08 | Deeewayne | jonumts, jira if you have a patch; #asterisk-consultants or the mailing list if you need a patch |
| 14:09.46 | jonumts | Thanks. |
| 14:10.27 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:25e0:3907:7c8d:f734) |
| 14:10.54 | sunfone | MariusAgnon: If you are really concerned with scalabilty you should exit the dialplan and have the whole thing controlled by a *compiled* AGI |
| 14:11.25 | sunfone | But I think that the AST DB is actually very efficient, since you really need only a single key and a counter, there is no need for the full relational DB |
| 14:11.28 | jonumts | Asterisk is overwriting the User-Agent header field that a UAC has already set. Can I disable this "feature"? |
| 14:12.22 | Gugge | its making a new call, with all new headers, including the user-agent header |
| 14:12.41 | Gugge | and as far as i know you cant set the user-agent header yourself from the dialplan |
| 14:13.24 | sunfone | I suppose you could do something kind a fancy with the storage of the counter, appending a timestamp for the call, and automatically "clean" the counter on the next call, if sufficient time has passed. The storage is a string... |
| 14:13.41 | sunfone | that would avoid a cron job |
| 14:13.58 | jonumts | I'd like to see the user-agent field unchanged on the receiving side of the call. |
| 14:14.18 | Gugge | jonumts: it isnt changed, its a new call, with a new user-agent header :) |
| 14:14.34 | Gugge | asterisk is not a proxy :) |
| 14:15.13 | WIMPy | jonumts: You shold read about what a B2BUA is. That's what Asterisk is. |
| 14:15.52 | jonumts | b2bua = back-to-back user agent |
| 14:17.21 | MariusAgon | thanks and one more question. i noticed some diference in asterisk event depending on asterisk version, but thats what i could deel with. but what concerns me - is it possible that different channel drivers generates different event? in example is there difference in callerid in sip and dahdi(in asterisk 1.8 newchennel event theres calleridname and calleidnum)? |
| 14:17.27 | jonumts | b2bua operates both endpoints of a communication. But why does it have to rewrite the user-agent header field as the call originally is not initiated by Asterisk, I think. |
| 14:17.54 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:18.22 | WIMPy | jonumts: Gugge already explained that. Your question doesn't make sense for a B2BUA. |
| 14:18.24 | sunfone | MariusAgnon: no, there should be no difference in callerid handling |
| 14:18.40 | Katty | hugs Deeewayne |
| 14:19.22 | Gugge | jonumts: its not rewriting anything, its creating something |
| 14:19.23 | puzzled | hi |
| 14:21.37 | MariusAgon | so if in sip there is calleidnum/calleridname all other channels would have same pair, no matter protocol(iax, pri etc) doesnt have two same values? |
| 14:21.44 | jonumts | Gugge: but it is not creating a call from nowhere. Basically, it's passing on a call. |
| 14:21.58 | Gugge | jonumts: if you say so |
| 14:22.37 | WIMPy | jonumts: No. That's why I suggested, you read anout the operation of B2BUAs and do so again. |
| 14:22.43 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 14:23.25 | WIMPy | You're describing the operation of a SIP proxy, which Asterisk is not / does not and will not. |
| 14:23.54 | jonumts | thanks. |
| 14:24.31 | JerJer | the problem is asterisk is not exactly a b2bua either |
| 14:25.21 | JerJer | asterisk is a unique hybrid between a proxy and b2bua |
| 14:27.01 | JerJer | jonumts: if you require a sip proxy, look into kamailio |
| 14:27.03 | tuxxie | Is their a way to check Caller-id name on phone numbers I own. I have found that these names seem to change without notice and I am unsure how to insure our numbers are using the proper Caller-id name. |
| 14:27.37 | jonumts | Basically what I have is: 2 UAC connected to Asterisk. UAC1 is calling UAC2. UAC2 sees User-Agent field that was set from Asterisk. How can UAC2 find out "real" user-agent, i.e. user-agent that was set from UAC1? |
| 14:28.15 | JerJer | jonumts: you could pass a custom sip header |
| 14:28.50 | JerJer | you ~might~ be able to overwrite the asterisk set user-agent, header.... but i kinda doubt it |
| 14:30.23 | puzzled | JerJer: hi. did you find a solution to your question a while back (about some kind of proxy thing)? |
| 14:31.07 | JerJer | puzzled: heh - don't remember which question. too many customers for that |
| 14:31.07 | jonumts | Is it against any RFC to have the User-agent with peer-peer-semantics? |
| 14:31.08 | Gugge | <PROTECTED> |
| 14:31.17 | puzzled | JerJer: :) |
| 14:31.23 | JerJer | i have a new one though |
| 14:31.55 | JerJer | i have a multiple-VPN situation (meaning NAT), plus some not natted at all |
| 14:32.28 | *** join/#asterisk davlefou (~david@41.225.208.170) |
| 14:32.41 | jonumts | Gugge: do you think it is against RFC3261 |
| 14:33.03 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 14:34.00 | Gugge | jonumts: no idea, but its against the way asterisk works :) |
| 14:35.01 | puzzled | JerJer: that sounds like fun and makes my brain hurt. separate boxes for NAT and non-NAT? |
| 14:35.11 | jonumts | Gugge: what do you mean with "the way asterisk works"? |
| 14:35.39 | JerJer | kamailio+asterisk - kamailio is statefully forwarding requests (plus/record_route()) , but is generally transparent (if asterisk fails will kamailio will send the call elsewhere) |
| 14:35.40 | JerJer | http://pastebin.com/M2vQW8ZQ |
| 14:35.53 | Gugge | jonumts: i mean the fact that asterisk makes new calls, and that asterisk is the user-agent, and that it creates the call with its own user-agent set |
| 14:36.00 | JerJer | 'fixing' nat on kamailio, but NOT using rttproxy (just making sure SDP is correct) |
| 14:36.30 | JerJer | RTP flows like it is supposed to, then out of the blue ~something~ causes RTP to start sending from the private ip |
| 14:36.35 | JerJer | have not narrowed it down yet :( |
| 14:37.04 | JerJer | -t |
| 14:37.11 | *** join/#asterisk m_tadeu (~quassel@89-181-66-231.net.novis.pt) |
| 14:37.34 | jonumts | Gugge: I do not understand how that is different to a "forking sip proxy" that is doing call branching. Isn't this the same behaviour? |
| 14:37.39 | puzzled | JerJer: sorry, that kinda magic is not in my bag of tricks |
| 14:37.48 | JerJer | its a bitch :) |
| 14:38.31 | Gugge | jonumts: a forking proxy forks the call and send _that_ call to multiple destinations. it does not make new calls |
| 14:41.19 | hetii | irroot: any clue about my issue ? |
| 14:41.37 | catphish | is a specific module needed handle 'register' lines in sip.conf? |
| 14:41.53 | catphish | i have a sip setup but register lines are being ignored |
| 14:42.17 | wdoekes2 | jonumts: asterisk is a b2bua, it's not a proxy |
| 14:43.14 | jonumts | I am about to learn this :-) |
| 14:43.38 | jonumts | I am not yet clear what exactly the difference is |
| 14:43.54 | wdoekes2 | asterisk holds two legs of a call A->asterisk and asterisk->B |
| 14:44.06 | JerJer | wdoekes2: more correctly: asterisk is a b2bua that adds in relevant proxy features necessary to make everything work |
| 14:44.14 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 14:44.37 | wdoekes2 | a proxy would simply forward (or only set up) traffic between A and B |
| 14:45.03 | JerJer | and a proxy maintains only state information - has no concept of a 'call' |
| 14:46.09 | JerJer | asterisk bridges N channels together - keeping track of a lot more detail, including managing the RTP flow |
| 14:46.19 | wdoekes2 | JerJer: it might, e.g. opensips with the dialog module.. but now we're complicating things |
| 14:46.36 | JerJer | hence why the dialog 'module' |
| 14:46.51 | JerJer | and kamailio / ser are proxies first - opensips wants to do their own thing |
| 14:47.06 | JerJer | that is one of many problems in that community |
| 14:47.27 | jonumts | Ok, |
| 14:48.37 | jonumts | I guess, however, RTP flow is between UAC peers - if possible |
| 14:49.03 | WIMPy | jonumts: UA1 calls Asterisk. That call ends there. Asterisk then sets up a 2nd call to UA2. -- They may be linked together. once answered, but so far it's two different calls. |
| 14:49.22 | WIMPy | Depends on the configuration. |
| 14:49.30 | Katty | dances |
| 14:49.31 | jonumts | I think I am getting the image :-) |
| 14:50.55 | jonumts | I am just having the following curious case: UAC2 is nosy and wants to know what type of UAC the originating peer is. |
| 14:50.56 | MariusAgon | What is "line" in new channel event? what is it for? |
| 14:51.16 | jonumts | I now know, that asterisk has set up the call to UA2 |
| 14:51.41 | jonumts | However, this has not happend out of nowhere. Asterisk has done that because it has been called by UA1. |
| 14:52.20 | WIMPy | jonumts: That's why it will use UA1s caller ID. |
| 14:52.46 | WIMPy | But UA2 is called by Asterisk, not by UA1. |
| 14:52.58 | WIMPy | Just on behalf of UA1. |
| 14:52.59 | *** join/#asterisk davlefou (~david@41.225.250.22) |
| 14:53.19 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:54.11 | jonumts | Let me summarize: Asterisk is calling UA2 on behalf of UA1 and as such is using UA1 caller ID. |
| 14:54.20 | wdoekes2 | jonumts: but you can add extra headers if you want, before the Dial().. SIPAddHeader(X-Original-User-Agent: ${SIPHEADER(User-Agent)}) .. or similar |
| 14:55.00 | WIMPy | jonumts: yes |
| 14:55.34 | jonumts | My proposal is: Why not allow Asterisk use the User-Agent on behalf of UA1, too? This would be optional, of course. |
| 14:56.56 | WIMPy | jonumts: I wouldn't see anything wrong with your proposal, but I don't see what it could be good for, either. |
| 14:57.10 | jonumts | WIMPy: It would get you a bounty :-) |
| 14:57.56 | wdoekes2 | SIPAddHeader(User-Agent: ${SIPHEADER(User-Agent)}) |
| 14:58.12 | wdoekes2 | where's my bounty? |
| 14:58.55 | jonumts | wdoekes2: is this the solution? No code change involved, no patching? |
| 14:59.00 | *** part/#asterisk cHarNe2 (thorn@newelite1.bshellz.net) |
| 14:59.06 | wdoekes2 | I guess it works.. you probably get 2 headers |
| 14:59.49 | jonumts | wdoekes2: well, it would have to work, then you can write you bill. |
| 15:01.00 | jonumts | WIMPy: It's for optimized on video overlay |
| 15:01.49 | WIMPy | jonumts: Sorry. I can't imagine the relation. |
| 15:02.46 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 15:03.16 | wdoekes2 | jonumts: it was SIPAddHeader(User-Agent: ${SIP_HEADER(User-Agent)}) and you get 2 User-Agent headers.. the latter one being the one from the other UA |
| 15:03.18 | jonumts | WIMPy: It's for video telephony. UA2 overlays video image with graphical elements. Those are optimized depending on the type of UA1. This is already developed and working. |
| 15:03.45 | wdoekes2 | you can probably kill the original header with a one line fix in the chan_sip.c |
| 15:04.21 | catphish | can anyone tell me which module defines function HASH |
| 15:04.32 | wdoekes2 | grep |
| 15:04.41 | WIMPy | jonumts: Ah, that way. Well, that doesn't seem a very good way of doing it by those UAs, as you just found out. |
| 15:05.06 | jonumts | WIMPy: any better ideas are always welcome. |
| 15:06.07 | *** join/#asterisk cowboy_bebop (~b0n40@212.39.81.133) |
| 15:06.11 | jonumts | wdoekes2: we would need a way to overwrite existing headers. |
| 15:06.46 | wdoekes2 | set global_useragent= empty in sip.conf |
| 15:06.58 | WIMPy | jonumts: Put opttions in the media stream. |
| 15:07.07 | wdoekes2 | .. that would be the useragent= setting |
| 15:07.55 | jonumts | WIMPy: I have no influence on what UA1 does. sorry. |
| 15:08.47 | jonumts | wdoekes2: thanks, that sounds promising. |
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| 15:11.32 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 15:11.36 | skrusty | afternoon |
| 15:11.56 | skrusty | what's generally accepted to be the better conferencing solutions for * these days? |
| 15:12.53 | ickmund | how can the kindle version of * the def. guide be more expensive then paper-back? That's just moronic |
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| 15:19.11 | jonumts | thanks lads for the great help. |
| 15:19.46 | Katty | hmm |
| 15:20.30 | Katty | is quick cooking oats (oatmeal) consider processed or unprocessed food? |
| 15:22.51 | Chainsaw | Katty: I would consider that processed food. High in fibre though, so wouldn't class it as unhealthy. |
| 15:23.15 | Katty | and what, specifically, makes you put it in the processed category? |
| 15:23.21 | Katty | the fact that it's been somehow altered to cook faster? |
| 15:23.51 | Qwell | Katty: http://en.wikipedia.org/wiki/Oat#Processing |
| 15:23.53 | tzanger | Chainsaw: do you consider a steak processed as well? |
| 15:23.54 | ChannelZ | Anything that doesn't have dirty on it when you put it in your mouth, apparently. |
| 15:24.11 | ChannelZ | s/dirty/dirt/ |
| 15:24.21 | Chainsaw | tzafrir: No, that's cut out of a cow and slung into a pan. |
| 15:24.47 | Katty | Qwell: ty |
| 15:25.31 | Chainsaw | tzafrir: Not been through a whole production line; no industrial process has been applied. |
| 15:25.48 | Deeewayne | Katty, I make my oatmeal w/ honey, cinnamon, raisins, and milk. ...mmm.... |
| 15:26.01 | Katty | Deeewayne: do you use regular oatmeal, or quick cooking oatmeal? |
| 15:26.13 | tzanger | Katty: it doesn't say how quick cook oats are different... are they "dry" par-boiled or something? |
| 15:26.21 | Katty | tzanger: i've no idea. |
| 15:26.26 | Deeewayne | I use the big tube of Quaker Oats |
| 15:26.27 | Katty | tzanger: in the container they are dry |
| 15:26.28 | Qwell | Deeewayne: [Raisins] used to be fat and juicy and now they're twisted. They had their lives stolen. Well, they taste sweet, but really they're just humiliated grapes. I can't say I am a big supporter of the raisin council. |
| 15:26.45 | Chainsaw | Qwell: Humiliated grapes. LOL |
| 15:26.48 | tzanger | Katty: I know, that's why I said "dry" par-boiled, becuase oats expand like crazy when put in water |
| 15:26.55 | Katty | nods |
| 15:27.05 | Qwell | http://www.imdb.com/title/tt0106387/quotes I 5-starred that movie, and now Netflix thinks I only want to watch chick flicks. |
| 15:27.07 | Katty | i just don't know the process of making them Quick Cooking |
| 15:27.13 | Katty | googles |
| 15:27.17 | tzanger | my oatmeal gets a touch of cream and brown sugar. I'm a purist. |
| 15:27.58 | tzanger | last year when I was in detroit every week I'd have oatmeal for breakfast almost the entire winter |
| 15:28.03 | Chainsaw | Katty: Steaming. |
| 15:28.19 | tzanger | bastards thought it a good idea to put walnuts in my oatmeal, I almost threw the bowl at him |
| 15:28.29 | Katty | ah yes, just read that Chainsaw |
| 15:28.38 | Katty | steamed and then dehydrated is uppose |
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| 15:28.44 | tzanger | steam, really... I thought it would have caused them to become puffy |
| 15:28.55 | tzanger | so really the same process as parboiled rice |
| 15:29.06 | Katty | that's the instant rice, right? |
| 15:29.23 | Katty | i guess buying something pre-steamed isnt' /so/ bad |
| 15:29.46 | _Corey_ | Qwell: I have to vigilantly watch my Netflix queue to make sure my fiance hasn't rated anything stupid. Last night I unrated "Legally Blonde 2"... |
| 15:30.28 | tzanger | no, parboiled rice is different from instant |
| 15:30.47 | tzanger | _Corey_: hahaha, my wife doesn't bother, I am in control of the ratings |
| 15:30.49 | Katty | oh ok |
| 15:31.11 | tzanger | although the 15yo watches a TON of anime garbage, I hope netflix rating system can weed that garbage out |
| 15:31.16 | tzanger | so far the recommendations haven't been too bad |
| 15:31.16 | _Corey_ | We have it on the Tivo, so that damned thumbs up button is too easy for her |
| 15:31.50 | _Corey_ | I think you can watch whatever nonsense without affecting the recommendations as long as you don't rate it |
| 15:31.54 | _Corey_ | I could be wrong though |
| 15:32.07 | tzanger | hope so |
| 15:32.36 | tzanger | the 2yo watches a ton of thomas the train and franklin, 15yo watches anime...so far not a single animated movie shows up in ratings |
| 15:32.58 | *** join/#asterisk iamaham (~iamaham1@web2.supergreenhosting.com) |
| 15:32.59 | iamaham | greetings |
| 15:33.14 | iamaham | can you have 1 sip phone monitor a call on another sip phone? |
| 15:33.25 | iamaham | if so how or recommend documentation on doing so |
| 15:33.52 | _Corey_ | iamaham: "core show application ChanSpy" for documentation |
| 15:34.36 | iamaham | thanks |
| 15:35.20 | iamaham | hrm so you have to specify if the channel can be spied on in the conf ahead of time? |
| 15:35.45 | _Corey_ | not really, that's one way (but most common) |
| 15:36.37 | iamaham | sorry not great at this, basically learn as much as I need to kinda got dumped on me 2 years ago |
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| 15:38.51 | luckman212 | anyone know an easy way to tell if a Cisco 7940 is SIP or SCCP? |
| 15:39.12 | Qwell | boot it |
| 15:39.21 | _Corey_ | It'll say "Sip" in the upper right on the screen |
| 15:39.31 | luckman212 | it will? |
| 15:39.51 | _Corey_ | If it's SCCP it'll usually have a bunch of horizontal lines across the screen |
| 15:40.51 | Chainsaw | Yeah, SIP with a funky stylised S. |
| 15:41.13 | _Corey_ | check google images, you should be able to see some examples |
| 15:41.37 | _Corey_ | (There is always the menu too... ;) ) |
| 15:42.52 | luckman212 | yea but Im not in front of the phone, is the problem... i need to tell someone what buttons to press in the menus to see that status |
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| 15:45.01 | Chainsaw | luckman212: Get them to take a photo and e-mail it to you. |
| 15:45.09 | Chainsaw | luckman212: The main screen of the photo will give all the info you need. |
| 15:45.13 | luckman212 | k thanks guys |
| 15:45.15 | luckman212 | :) |
| 15:45.20 | Chainsaw | luckman212: Main screen of the phone, even. One day I'll learn to type. |
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| 15:54.17 | *** join/#asterisk mintee (~mintee@2001:470:7:a41::2) |
| 15:54.43 | mintee | hey all, i'm trying to log specific outbound calls by dialing a set a numbers before the actual phone number |
| 15:55.03 | mintee | i thought I could do this with exten => _1628NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 15:55.10 | mintee | but it's not working |
| 15:55.30 | Cadey | well that would dial 628NXXNXXXXXX |
| 15:55.48 | Cadey | dont you want EXTEN:4 to dial NXXNXXXXXX ? |
| 15:55.53 | irroot | adjust the :1 in ${EXTEN:1} |
| 15:56.32 | mintee | Oh... |
| 15:56.43 | mintee | tries |
| 15:57.00 | Cadey | can I get a Dialplan reload :P |
| 15:57.12 | Cadey | BWRAPP! :D |
| 15:57.19 | irroot | gives Cady a dialplan reload |
| 15:57.36 | Cadey | Na na... |
| 15:57.48 | mintee | hum |
| 15:57.49 | mintee | [Jun 10 11:57:24] NOTICE[29541]: chan_sip.c:15436 handle_request_invite: Call from 'linksysspa941' to extension '16282153708' rejected because extension not found. |
| 15:58.28 | Cadey | minitee, that number isnt long enough for that pattern |
| 15:58.28 | irroot | mintee you match is not right somewhere |
| 15:58.39 | Cadey | _1628NXXNXXX would work for that number |
| 15:58.49 | Cadey | but you may want to use this... _1628NXXNX. |
| 15:58.53 | irroot | double points for Cadey checking it :P |
| 15:59.01 | mintee | i was trying to dial 16282153708888 |
| 15:59.09 | Cadey | with the . so it can be any lenght afte the NXXNX |
| 15:59.17 | mintee | where the 215 is NXX |
| 15:59.40 | irroot | got opensips built and installed lets compare to kamalio |
| 15:59.49 | Cadey | but the error isnt showing you dialed the last 8's |
| 16:00.05 | irroot | seems similar but perhaps more features |
| 16:00.07 | Cadey | is your phone cutting them off with some kind of max dial string setting? |
| 16:00.10 | mintee | ya, the sip phone wouldn't allow me to |
| 16:00.24 | mintee | Cadey: ya, that might be in |
| 16:00.27 | irroot | there is a dialplan on the linksys |
| 16:00.40 | irroot | that needs to match as well |
| 16:00.44 | mintee | let me check it's configuation |
| 16:01.50 | *** join/#asterisk bchia (~Adium@nat/digium/x-ekceeoqindidqnym) |
| 16:02.39 | mintee | lol |
| 16:02.40 | mintee | (*xxx.|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 16:02.50 | mintee | that's the phones dialplan |
| 16:03.14 | Cadey | change it to |
| 16:03.20 | mintee | I have no clue how to decypher that |
| 16:03.21 | Cadey | (*xxx.|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxxxxx.) |
| 16:03.31 | Cadey | the last one has a max of 11 |
| 16:03.37 | Cadey | your number is 14 long |
| 16:03.39 | mintee | ah, i see |
| 16:04.18 | mintee | what is the 3469? |
| 16:04.23 | Cadey | does not use dialplans on his phones, set them to (.*) and lets asterisk work it out |
| 16:04.46 | mintee | ya, i think that's a good idea |
| 16:04.47 | Cadey | [3469]11 means 311, 411, 611 and 911 - amergency numbers |
| 16:04.53 | _Corey_ | you make your users hit Dial? |
| 16:05.13 | mintee | no, but i could |
| 16:05.22 | mintee | ah |
| 16:05.45 | _Corey_ | in my experience, they complain a lot when you do that |
| 16:05.52 | irroot | mintee there is a post dial delay to dial that is 5s ie after pause of 5s it dials |
| 16:06.23 | _Corey_ | "why does this phone system take so long to dial?"... "my old phone system dialed faster" etc |
| 16:06.27 | Cadey | _Coryey_ : what making them hit dial, I agree :) timeouts are how 99% of home phones work so its what they want |
| 16:07.10 | mintee | the extra xxx's didn't work |
| 16:07.41 | mintee | tries clearing the dialplan |
| 16:07.41 | sunfone | Cadey: I wouldn't agree - I would say 99% of home phones are subject to dial matching patterns, which were around far before VoIP |
| 16:08.01 | sunfone | If you don't want the complaints, I would use a dialplan. They aren't that hard to parse. |
| 16:08.01 | _Corey_ | did you remove: 1xxx[2-9]xxxxxxS0 ? |
| 16:08.14 | Cadey | sunfone : true I guess |
| 16:08.19 | mintee | no |
| 16:08.28 | _Corey_ | Well, you're probably matching that |
| 16:08.49 | irroot | mintee there better ways to do this |
| 16:08.53 | mintee | ya, cause of the 1 |
| 16:08.55 | _Corey_ | just remove everything that doesn't fit your situation and rebuild it |
| 16:08.57 | mintee | irroot: i'm all earys |
| 16:09.10 | irroot | prompt for a code is one |
| 16:09.15 | mintee | s/earys/ears/ |
| 16:09.29 | irroot | the other is to process the CDR's |
| 16:09.47 | mintee | i will be processing the CDR's. |
| 16:09.48 | irroot | im all beers |
| 16:09.58 | mintee | but we need to bill clients for outbound calls |
| 16:10.03 | Cadey | is thinking its nearly beer time here in the UK :D |
| 16:10.08 | _Corey_ | i'm about to be all beers... we're doing a cookout at the office today ;) |
| 16:10.11 | mintee | it's always beer time! |
| 16:10.22 | _Corey_ | in fact, they started the grill.. adios |
| 16:10.25 | mintee | so we have 2 clients we make random outbound calls for |
| 16:10.41 | irroot | mintee then SET(CDR(accountcode)=) |
| 16:10.43 | mintee | so we were going to dial a string+phonenumber |
| 16:10.46 | mintee | LOL |
| 16:10.48 | mintee | really? |
| 16:10.51 | mintee | it's that easy |
| 16:11.13 | irroot | can use the callerid perhaps |
| 16:11.27 | mintee | but how do we tell the difference in which client we made the call for? |
| 16:11.49 | mintee | oh, like set the CALLID before the outbound dial? |
| 16:11.58 | mintee | good idea too... |
| 16:12.07 | irroot | SET(CDR(accountcode)=${CALLERID(num)}) |
| 16:15.05 | mintee | but i still don't know how to show the difference between who we're calling for |
| 16:15.19 | mintee | say client 1 wants use to call 2155551234 |
| 16:15.33 | mintee | and then 10 minutes later, client 2 wants us to call the same number |
| 16:15.34 | irroot | the account code in the CDR |
| 16:15.54 | irroot | if you set the accountcode it shows in cdr |
| 16:16.08 | mintee | ya, i get that |
| 16:16.26 | irroot | then you can filter the CDR's |
| 16:16.36 | mintee | ya, but CALLERID isn't going to help |
| 16:16.52 | mintee | for arguement sake, let's say we only have 1 sip phone to dial out of |
| 16:17.10 | irroot | you could prompt for pin code ?? |
| 16:17.23 | mintee | i have _91NXXNXXXXXX as a default outbound |
| 16:17.45 | mintee | ya, but then they'd have to enter a pin for all outbound calls |
| 16:17.52 | mintee | and that's a pain |
| 16:18.28 | irroot | just options ... |
| 16:19.00 | irroot | need to see what best suits the l.user |
| 16:20.22 | mintee | exactly |
| 16:22.09 | irroot | have a extensions.conf that has bloated to near ridiculous ... trying to cater for them |
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| 16:46.11 | sunfone | For anyone who cares, my lab testing of LXC is going very well... adding a container adds a whopping 50M of RAM use with asterisk running inside... I've been on a conference call via the container now for over an hour on a Polycom IP650 with perfect audio quality. |
| 16:47.08 | *** join/#asterisk bchia (~Adium@nat/digium/x-vimobizsgavxkryx) |
| 16:47.56 | sunfone | Architecture is very clean - the host runs asterisk and provides access to all the containers to/from the PSTN. The containers have no direct Internet access, so very secure. |
| 17:06.40 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:11.58 | tzafrir | has been fighting with lxc today (regardless of Asterisk) and finds it under-documented |
| 17:12.29 | sunfone | tzafrir: you got that right |
| 17:14.10 | tzafrir | Yeah, Linux-VServer will not get merged. No idea about OpenVZ |
| 17:14.21 | tzafrir | So I'm moving to LXC |
| 17:14.36 | tzafrir | I suppose that by Debian Wheezy it will be fine |
| 17:15.12 | sunfone | At least on Debian/Ubuntu I understand OpenVZ will not get integrated |
| 17:15.26 | sunfone | I'm using Ubuntu Natty - 11.04 |
| 17:16.10 | sunfone | I'm still struggling with sharing dahdi to the containers though |
| 17:17.08 | tzafrir | Did you see what I wrote on the asterisk-users mailing list? |
| 17:17.22 | sunfone | hmm, no I must have missed it |
| 17:18.13 | tzafrir | http://lists.digium.com/pipermail/asterisk-users/2011-June/263599.html |
| 17:18.29 | Katty | herroes |
| 17:18.56 | sunfone | Excellent, thanks! I'll give that a shot. |
| 17:19.23 | sunfone | Not sure how I missed that... though I get in a hurry some days and most of the posts get the delete key without too much perusal |
| 17:19.28 | sunfone | Lot of noise there |
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| 17:59.58 | jc319 | wow this is a breakthrough, finally managed to get something from outside my room communicate with Asterisk |
| 18:00.10 | Qwell | long cable? |
| 18:00.22 | jc319 | [Jun 10 17:58:33] NOTICE[2903]: chan_sip.c:21619 handle_request_invite: Call from 'itsp_coms_home' to extension 's' rejected because extension not found in context 'incoming_calls'. |
| 18:00.53 | jc319 | does this not mean, direct all incoming calls to extension 201? [incoming_calls] exten => _X.,1.NoOp() exten => _X.,n,Dial(SIP/201) |
| 18:01.09 | Qwell | It does not. |
| 18:01.25 | Qwell | _X. does not match 's'. |
| 18:02.06 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 18:02.12 | irroot | jc319 need to have a s extenstion and possibly look at the to header for the exten it is routed too |
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| 18:28.24 | p3nguin | Not to mention _X.,1.NoOp() contains a typo that would likely prevent it from working if the extension matched the pattern. |
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| 18:38.22 | jc319 | just copied that from the * book, probably from the wrong section though. anyway I found a better one >> _. |
| 18:40.25 | jc319 | it worked - I got my first ring tone using my own server, a moment to remember |
| 18:40.37 | *** join/#asterisk jeremy_g (jeremy_g@c-6da0e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 18:40.40 | jeremy_g | yo babies |
| 18:40.48 | jeremy_g | where are the chickens |
| 18:41.15 | jc319 | now I need to create the proper dialplan, I am reading a lot but I don't know how to do this, can anyone please tell me how to implement what I want? I will type the long story if anyone is willing to have a look |
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| 18:41.41 | jeremy_g | jc319:sure my friend. i can help as much as i know which is little |
| 18:41.44 | jeremy_g | :( |
| 18:42.05 | jc319 | how little, I am a true * user for the last 18 minutes |
| 18:42.11 | jc319 | you must be more experienced :D |
| 18:42.22 | jeremy_g | jc319:shoot |
| 18:43.17 | jeremy_g | i am waiting... |
| 18:43.39 | jc319 | ok here's the case. this is a SOHO deployment, the only two users is me and my gf. we have 3 SIP DiDs, 1 for each and a 3rd one (shared-home #). I want to have on our 7960 phones homeline for each user in addition to their ownline |
| 18:45.09 | jc319 | I think I also need extensions in the traditional sense, like 201, 202 etc, so that we can talk for free using iPhone or PC SIP software when one of us is away |
| 18:46.06 | sunfone | tzafir: thanks for the tip - got the dahdi devices to show up in the LXC containers now ;) |
| 18:46.09 | jc319 | so if I get this correct, I need 3 hardphones (2 7960 deskphones and 1 home wireless SIP), 4 softphones (2x iphone softphones, 2x PC softphones) |
| 18:46.50 | jc319 | is this correct? do I create and assign an extension number to each device? and assign cisco's line buttons to different outbound lines? |
| 18:46.50 | jeremy_g | jc319:this sounds quite advanced stuff for me |
| 18:47.16 | jc319 | well this being my first config I get lost too. I was hoping someone tell me how to do this |
| 18:47.20 | jeremy_g | jc319:i am an old man living in a village and so religious that i only support married couples when it comes to providing services. |
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| 18:47.52 | jc319 | too bad |
| 18:47.58 | jc319 | thanks anyway |
| 18:47.58 | jeremy_g | for you |
| 18:48.20 | jeremy_g | yeah get married and then come back. :) |
| 18:48.26 | jc319 | ok |
| 18:48.38 | jc319 | but you improve your * skills too ok |
| 18:48.42 | jc319 | don't want to get married for nothing |
| 18:49.04 | jc319 | anyway I'll create about 10 extensions and see how it goes |
| 18:49.32 | n3hxs | jc319, create an extension for each phone. |
| 18:49.47 | jeremy_g | hehe |
| 18:50.10 | n3hxs | Though I am more of a FreePBX user. ;) |
| 18:50.41 | n3hxs | and not a Cisco phone user.... |
| 18:52.40 | jc319 | I like the handsfree voice quality of this 7940-7960s. I have seen and used one or two other phones in some temp offices but didn't do an extensive research on the subject |
| 18:54.00 | jeremy_g | is there any web based voip client that asterisk is known to work well with? |
| 18:54.50 | jeremy_g | secondly, any reports on how well asterisk works on amazon ec2 instances - any reading or recommendations |
| 18:56.11 | jeremy_g | third question, which web based provision client is recommended for production systems |
| 19:00.17 | jeremy_g | fourth question, can asterisk bring world peace |
| 19:00.58 | sunfone | jeremy_g: I'm curious to hear people's experience running on EC2 - I have heard that it can work well, but I would be very concerned about bandwidth costs. My asterisk instances get heavily attacked ALL the time, and though we use iptables to thwart them, the bandwidth is used anyway. On EC2 I think you would be charged for that over some threshold. |
| 19:03.13 | n3hxs | sunfone, seems to me that the billing for EC2 would be for call time, not for unsuccessful attempts. |
| 19:04.22 | sunfone | n3hxs: EC2 is just a virtual machine provider - they know nothing about what you are doing with the bandwidth. If a dedicated attacker consumes 10Mbps continuously for days, you can bet you will be over your bandwidth allowance for your VM instance. |
| 19:05.44 | n3hxs | But without authentication, there would be no way to bill your account. There are others on that service using the same IP address. Unless your user/pw is so simple that they are actually using your account. |
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| 19:06.52 | sunfone | Totally not following you. With EC2 you get a virtual machine and an IP for yourself - it would certainly not be shared. |
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| 19:20.41 | p3nguin | jc319: While _. may work, it isn't often the right choice. I would encourage you to learn and understand pattern matching and why _. matches extensions that you shouldn't match with patterns. |
| 19:21.27 | p3nguin | jc319: You'll want to create peers for the phones to use. You'll also have to create extensions which will dial the phones. |
| 19:23.44 | p3nguin | jc319: Depending on how you want to dial the phones in what many people call "internally," you may or may not want to have shortened extension numbers to call between phones. You can use the entire 10-digit extension to call from one phone to the other if that's how you choose to do it. I wouldn't do it that way, but you can if you want. I would use 3- or 4-digit extension numbers for calling between phones and use the ... |
| 19:23.49 | *** join/#asterisk scalex000 (~chatzilla@186.6.65.18) |
| 19:23.51 | p3nguin | ... 10-digit extension for calling inbound from the ITSP. |
| 19:23.58 | scalex000 | good afternoon |
| 19:24.35 | scalex000 | I have problem with my skype for asterisk behind iptables |
| 19:24.52 | scalex000 | which port I need to open |
| 19:27.28 | jc319 | p3nguin: I just wanted a quick method to match any and all incoming numbers just to ring once. I have this ip phones for months but due to several reasons was not able to set up this system, so just wanted to see it 'work', once, before I go in detailed configuration. I am doing that now. thanks for your assistance. I have already created 8 extensios ('phones'). the way I did is person1.deskphone, |
| 19:27.29 | jc319 | person1.iphone, person1.PC >> extensions 201,202,203. person2 the same for 211,212,213. and finally 221,222,223 (shared home line physical wireless phone, iphone softphone, PC softphone). so we have 3 different views to all lines using different extensions |
| 19:27.56 | jc319 | I have no idea how can say person1, on iphone, select outbound line-A or the other outbound line |
| 19:28.34 | jc319 | perhaps each extension should have a default outbound line, I will check sample configs to figure out |
| 19:28.55 | p3nguin | jc319: I know you want to match any and all incoming numbers, but don't do that. |
| 19:29.59 | jc319 | no I don't want to match any and all incoming numbers. I did wish that half an hour ago. now I want to inspect incoming calls, and if it belongs to person 1, ring their deskphone+softphone(iphone)+softphone(PC). |
| 19:30.10 | jc319 | the same for person2 or shared homeline. |
| 19:30.14 | p3nguin | jc319: Either explicitly define the DID numbers as the extensions, or use a better pattern that doesn't match EVERYTHING. _X. is a better pattern to match all incoming calls to real phone numbers. I don't recommend it, but it's effective. |
| 19:32.14 | p3nguin | jc319: You've used the term "inspect." If you use the DID number as the extension, you don't have to inspect anything; you simply accept the calls to their respective extensions and in turn Dial() your devices. |
| 19:32.37 | jc319 | [incoming_calls_home] exten => s,1,Answer exten => s,2,NoOp(${CALLERID}) exten => _.,n,Dial(SIP/221&SIP/222) |
| 19:32.56 | jc319 | would thisk work |
| 19:33.18 | p3nguin | Work, yes. Work in a way I would recommend, no. |
| 19:33.43 | p3nguin | You have no reason to Answer() the call only to Dial() a couple devices later. |
| 19:33.53 | p3nguin | And the extensions s and _. are very poorly used here. |
| 19:34.15 | jc319 | p3nguin: so there is one DiD number and there are 3 'locations' that I want this number to ring (desk+sofphone1+softphone2). what should I do then? |
| 19:34.40 | jc319 | maybe a better term for 'locations' is 'devices'? |
| 19:35.02 | p3nguin | Let us say that your DIDs are 3149691077, 3149691078, and 3149691079... |
| 19:35.17 | p3nguin | devices or phones, yes. |
| 19:35.20 | jc319 | ok |
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| 19:35.55 | p3nguin | exten => 3149691077,1,NoOp(shared phone number); |
| 19:36.22 | p3nguin | exten => 3149691077,n,Dial(SIP/phone1&SIP/phone2,30); |
| 19:36.52 | p3nguin | Now when a call comes in to your first (shared) phone number, it dials two devices at the same time. |
| 19:37.16 | jc319 | of course this can be three devices without any problems, just another &SIP/phone3, right? |
| 19:37.36 | p3nguin | correct, just add another device in the dial string. |
| 19:37.41 | jc319 | excellent |
| 19:37.58 | p3nguin | exten => 3149691078,1,NoOp(my phone number); |
| 19:38.16 | p3nguin | exten => 3149691078,n,Dial(SIP/phone1,30); |
| 19:38.36 | p3nguin | exten => 3149691079,1,NoOp(wife phone number); |
| 19:38.44 | p3nguin | exten => 3149691079,n,Dial(SIP/phone2,30); |
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| 19:39.22 | jc319 | so rather than 's' use the DiD, and rather than answer, just dial. got it thanks |
| 19:39.31 | p3nguin | You can even make each of your individual numbers ring that shared wireless phone by again adding &SIP/phone3 in the dial strings accordingly. |
| 19:39.34 | jc319 | what about the other way around, internal to external calls? |
| 19:40.08 | jc319 | more specifically, on 7960 phone, using the 'line buttons' can we call outbound from any of the 2 lines available to each person? |
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| 19:40.19 | p3nguin | Let me pastebin my outgoing extensions for you. |
| 19:40.26 | jc319 | great thanks |
| 19:40.40 | kaldemar | jc319: unless it is only s that you get in from your provider. but that is usually handled by register => user:secret@host/yourDID in sip.conf. |
| 19:41.23 | jc319 | one last thing I have on mind is the contexts, do I need simple incoming / outgoing or do I need to use contexts to divide them to logical units for any reason |
| 19:44.14 | p3nguin | jc319: http://pastebin.com/iPy2jYef |
| 19:44.58 | p3nguin | jc319: I have incoming, outgoing, and internal contexts with extensions appropriately placed within each. |
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| 19:45.41 | jc319 | would it be greedy to ask for that file too |
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| 19:46.22 | jc319 | thanks for this one, I will use this |
| 19:46.45 | p3nguin | I can't really provide you with my extensions.conf easily. I can tell you how to create extensions, though. |
| 19:47.06 | jc319 | do you still have a [default] section in your extensions? is it to be deleted when one starts writing his own config? |
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| 19:49.38 | [sr] | hi WIMPy |
| 19:55.29 | p3nguin | jc319: Asterisk seems to expect the default context to be present, so leave it, but leave it blank. Create your own new context to send anonymous calls into. |
| 19:56.20 | p3nguin | jc319: By "send anonymous calls into," I mean the context that you'll define in the general section of sip.conf. |
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| 19:58.55 | p3nguin | Let me see if I can find the old example extensions.conf I used to share. |
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| 20:03.48 | wwalker | I've got a voip provider using the acronym ULC, anyone know that one? He's using it in a context where LEC or CLEC might make sense. |
| 20:04.22 | wwalker | I think it's his term for upstream provider, but don't want to look like an idiot. |
| 20:05.38 | malcolmd | ULEC? unbundled local exchange carrier? |
| 20:10.04 | jc319 | p3nguin: thanks, appreciated. I keep typing and testing. |
| 20:10.18 | wwalker | malcolmd: thank you, sounds like he just left the E out, but he used the acronym multiple times... |
| 20:10.44 | malcolmd | could be a new one, but i've never heard of it, and google didn't help either |
| 20:11.09 | p3nguin | jc319: Here's a sample of how a fairly basic extensions.conf could look: http://pastebin.com/Piqv4Egj |
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| 20:11.27 | jc319 | I remember reading something like "last registered sip end point will ring, others will not receive any notifications" is that right? If so, for a single shared home DiD phone line, I will need to create many 'entry points' (internal extensions - one for physical wireless phone to access this DiD, 3 more points (internal extensions) for person1s use in 3 devices: deskphone,mobile,PC, and |
| 20:11.27 | jc319 | finally the same for person 2. is this unnecessary or the correct way to do it? |
| 20:12.11 | p3nguin | jc319: Every "line" on every device needs to have its own peer definition in sip.conf. |
| 20:12.33 | p3nguin | jc319: If you use the same peer name for several devices, the last registered device will get the call. |
| 20:13.58 | jc319 | cheers, I'm on the right track then |
| 20:14.19 | p3nguin | jc319: For a multi-line phone, I use the MAC address of the device and append -a, -b, -c, -d and so on for each line key that needs to have its own peer definition. |
| 20:14.44 | p3nguin | So 00001234FFFF-a would be the first line key on the device. |
| 20:15.05 | p3nguin | Then you dial it using Dial(SIP/00001234FFFF-a) |
| 20:16.55 | p3nguin | Not everyone agrees on device naming schemes, but this one seems to make sense to me. |
| 20:17.49 | p3nguin | Some people use the name of the person who uses the phone, others use the numeric matching the extension number. There are many naming schemes -- pick something that makes sense. |
| 20:18.38 | jc319 | yes I used extension numbers so far, but your style seems shorter to type due to copy paste factor |
| 20:19.34 | p3nguin | Some people don't like the MAC address scheme because they think it's harder to create the extension-to-device association. I think they're just being lazy. |
| 20:20.56 | p3nguin | Depending on how you handle the association (I use the AstDB), it doesn't have to be any harder than using an ID which happens to be the extension number. |
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| 20:33.00 | ideaman | Can anyone help tell me how on a single PRI card, set one span FXS so I can take it to a channel bank, and the other span, PRI. I know how to do one or the other, but in chan_dahdi, what about both? |
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| 20:36.58 | ideaman | anyone.... |
| 20:37.12 | _Corey_ | uh, you have a single PRI card (i.e. one port) ? |
| 20:37.19 | ideaman | no sorry, dual span |
| 20:37.50 | _Corey_ | Well, you can't "set the span FXS" because it's a T1 port |
| 20:38.01 | _Corey_ | what's the channel bank's capability? |
| 20:38.12 | ideaman | signalling is what I'm having the problem with on the asterisk box |
| 20:38.21 | ideaman | I can get fxs to the channel bank ok |
| 20:38.29 | ideaman | and I can get pri to my PBX ok |
| 20:38.44 | ideaman | I just don't know how to do both signallings on a dual span card |
| 20:38.58 | ideaman | in chan_dahdi |
| 20:39.05 | _Corey_ | so, you're just asking how to configure two ports with different signaling? |
| 20:39.09 | ideaman | yes sir |
| 20:39.14 | _Corey_ | ah |
| 20:39.49 | _Corey_ | well, just set your signaling and such, then define your channels and do it again and define the other channels |
| 20:40.24 | _Corey_ | i'll give you an example |
| 20:40.28 | ideaman | k |
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| 20:41.29 | _Corey_ | http://pastebin.com/ye2DSKNS |
| 20:46.05 | ideaman | so is it just in the sequence of doing signalling, then group, then channel, and you can have whatever you want mixed in? |
| 20:47.30 | _Corey_ | yeah, so pretty much what happens is you set a value and it's applied to whatever you specify at the end |
| 20:47.44 | _Corey_ | if you don't want it applied, you need to set it to something else |
| 20:48.10 | _Corey_ | you could do: |
| 20:48.11 | ideaman | I know I should I know this, but where does the group number come from ? system.conf? 2,2,1 lines? |
| 20:48.18 | _Corey_ | no |
| 20:48.25 | _Corey_ | the group number is arbitrary |
| 20:48.39 | _Corey_ | that stuff in system.conf is signaling and such |
| 20:48.48 | _Corey_ | (see the samples file for explanations) |
| 20:48.54 | ideaman | k |
| 20:49.58 | ideaman | let me see if I can get these to work really quick |
| 20:55.28 | ideaman | COREY!!! You saved me!! |
| 20:55.33 | ideaman | Thanks a million, works like a charm |
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| 20:59.48 | _Corey_ | yeah, no problem really |
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| 22:02.37 | jc319 | p3nguin: are you still here? |
| 22:02.42 | p3nguin | Yes. |
| 22:03.49 | jc319 | no luck, I am on my sixth config set (sip.conf & extensions.conf) and I think I will start from scratch |
| 22:06.31 | jc319 | in your 2nd (detailed) example, you have voipms in & outbound, also call-centric outbound. what is from-ipkall |
| 22:08.14 | p3nguin | It's another ITSP which sends calls inbound. |
| 22:08.46 | p3nguin | I would have liked to provide a better, more updated extensions.conf sample for you, but I was short on time. Maybe I can work on that pretty soon. |
| 22:10.27 | p3nguin | jc319: Who is your ITSP? |
| 22:10.33 | jc319 | coms.com |
| 22:10.49 | p3nguin | Okay, so you've configured a peer in sip.conf for it already? |
| 22:11.11 | p3nguin | [coms] |
| 22:11.11 | p3nguin | type=peer |
| 22:11.15 | jc319 | 2 minutes ago started from scratch, using your config, editing as I go |
| 22:11.19 | p3nguin | context=coms-inbound |
| 22:11.23 | p3nguin | et cetera |
| 22:11.32 | jc319 | I'll finish this and show you in a few mins |
| 22:12.34 | p3nguin | Here's a sample sip definition: http://pastebin.com/RAETbcNZ |
| 22:13.01 | p3nguin | a sample for VoIP.ms, that is. |
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| 22:15.21 | jc319 | 'username' should be 'fromuser' in the new * versions, right? |
| 22:15.41 | p3nguin | No, defaultuser, I believe. |
| 22:16.03 | p3nguin | fromuser is something different, and you'll only use it if your ITSP needs the value to be passed. |
| 22:16.44 | p3nguin | If you use "username" instead, you should get a verbose warning on the CLI that shows the new parameter's name. |
| 22:19.02 | jc319 | Note: This option is deprecated on Asterisk 1.6 which uses "defaultuser" |
| 22:19.08 | jc319 | ok using this |
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| 22:23.21 | pdtpatrick___ | is it possible to send information to a bash file from asterisk or a python file? lets say a number is dial, can agi send the input to a file? |
| 22:23.50 | pdtpatrick___ | currently we have it sending to a .agi file which is perl |
| 22:24.34 | sunfone | pdtpatrick: are you asking if you can code AGI in something other than perl? |
| 22:24.38 | sunfone | then yes |
| 22:25.01 | sunfone | but it would be best to make sure you have an AGI library for whatever language you choose |
| 22:25.19 | sunfone | IMO 'C' is good ;) |
| 22:27.07 | WIMPy | AGI library? What's that for? |
| 22:28.26 | p3nguin | That's where all the AGI books are kept. |
| 22:29.21 | WIMPy | There are AGI books? |
| 22:29.33 | p3nguin | I don't really know since I can't read. |
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| 22:30.29 | WIMPy | That's ok. I can't read either. But if you have books it looks like you could. |
| 22:31.31 | p3nguin | I usually use my books to hold up the end of my bed or to adjust the legs of tables. |
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| 22:35.20 | jc319 | p3nguin: do you not normally receive inbound calls from voipms-inbound? |
| 22:35.28 | jc319 | in other words are you DiDs elsewhere |
| 22:35.42 | p3nguin | jc319: I receive calls via DIDs from multiple providers. |
| 22:36.22 | p3nguin | Each ITSP will have its own sip.conf peer definition. |
| 22:37.20 | p3nguin | Also, in my configurations, ITSPs which have more than one IP address which calls could originate will have more than one peer definition. |
| 22:37.45 | jc319 | p3nguin: in this code ( http://pastebin.com/Piqv4Egj ), section beginning on 29... does it mean you do not 'expect' incoming calls from this inbound connection? |
| 22:39.17 | p3nguin | jc319: In this particular pasting, the voipms-inbound context does not have any explicitly configured phone numbers. What happens in this case is that any DIDs I purchase from VoIP.ms and route to my computer will end up on the "this number is not in service" recorded message. |
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| 22:39.45 | p3nguin | jc319: You can see in the from-ipkall context how I configure a phone number in a context. |
| 22:40.43 | p3nguin | jc319: For example, if I have four phone numbers with ipkall, only 2065237757 does anything useful, and any other phone number from ipkall will also end up on the not in service message. |
| 22:40.44 | jc319 | ok I think I got it now, please correct me if I'm misreading, but what I understand is... section beginning on line 37 is (one of the places) where you 'expect' inbound calls, you check if it matches the number if not with the next subsection (line42) you drop it |
| 22:40.58 | jc319 | cool |
| 22:41.18 | p3nguin | Line 37 begins the from-ipkall context. |
| 22:41.36 | jc319 | I didn't know how to do this before, I'll do the same then. but I will enter all my three DiDs in succession, the 4th section will be the not in service bit. I have high hopes now |
| 22:42.05 | p3nguin | Any call which matches the SIP peer definition in sip.conf will obey the context=from-ipkall parameter in the peer entry, sending the call into the from-ipkall context in extensions.conf. |
| 22:42.37 | p3nguin | If there is a matching extension, that extension will execute. |
| 22:42.38 | linuxgecko | i really might just be overlooking something obvious, but this is my first 1.8.x install. I'm trying to follow the howto on the wiki for using gooogle voice to make/take calls. all i have in my /etc/asterisk/ dir is the snippets on the wiki. i compiled my asterisk 1.8.4.2 with chan_gtalk and res_jabber. what am i missing? i don't see any channels, or sip options, and nothing in the dialplan except parking, |
| 22:43.18 | p3nguin | The extension 2065237757 is explicit, and you can see that it dials a SIP phone. Any other phone number not matching that extension will be matched by the _X. pattern, which plays the not in service message. |
| 22:44.03 | p3nguin | When matching extensions, inclusing patterns, the most specific match wins. |
| 22:44.29 | p3nguin | So an exact match of 10 digits wins despite the presence of the pattern which also matches. |
| 22:44.52 | p3nguin | Take out the explicit match and the pattern will win. |
| 22:46.20 | p3nguin | This is just my way (and I assume that other people use very similar if not identical method) of being able to send all my DID numbers to my PBX and handle them all accordingly, even if the numbers have no purpose at the current time. This is often useful when you buy blocks of phone numbers but only require a few of them. |
| 22:47.04 | p3nguin | Such as when you can buy a block of 20 numbers for the same price and 12 individual numbers... |
| 22:47.15 | jc319 | p3nguin: Thanks, this is very clear now. It should work now or at leats I will have a better config to paste now. many thanks |
| 22:47.22 | p3nguin | You'd want those 8 numbers for free, just in case you decide to use them for something later. |
| 22:48.32 | p3nguin | jc319: Feel free to paste your sip.conf and extensions.conf if you have any doubts. Mask your passwords before submitting the pastes. |
| 22:50.27 | jc319 | Shall do after I finish editing & testing. cheers |
| 22:51.29 | linuxgecko | am i missing something so big noone will even tel me RTFM? |
| 22:52.45 | p3nguin | linuxgecko: Did you create a dial plan to accept calls? Did you make the connection via jabber and gtalk? Is anything doing anything? |
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| 22:55.02 | p3nguin | The last time I used the wiki to configure google voice calling, it worked exactly as expected. |
| 23:03.20 | linuxgecko | p3nguin: i thought so, i thought setting up the files as listed in the wiki would do that, and it looks like nothing is. in that order. |
| 23:04.16 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 23:05.06 | linuxgecko | p3nguin: i only have the files listed in the wiki, and they are blank, except for the snippets in the wiki. |
| 23:05.22 | jc319 | p3nguin: I tried the modified code and of course it did not work :D here are the pastes, thanks. extensions @ http://pastebin.com/73unZ8Qu && sip @ http://pastebin.com/n5TdZTDK |
| 23:05.58 | jc319 | and this is the notice msg [Jun 10 23:02:03] NOTICE[3861]: chan_sip.c:21619 handle_request_invite: Call from '02071486267' to extension 's' rejected because extension not found in context 'comscom_home_inbound'. |
| 23:06.58 | p3nguin | jc319: They are incorrectly sending calls to extension s rather than to your phone numbers. |
| 23:07.29 | p3nguin | Also, they are not behind nat, so change nat=yes to nat=no in the peer definition. |
| 23:08.20 | jc319 | asterisk is behind not, phones and asterisk in the same lan. should it be no? |
| 23:08.24 | jc319 | nat* |
| 23:08.41 | jc319 | actually it is not behind nat, it has all the ports it needs open/redirected |
| 23:08.48 | jc319 | ok changing to no.. |
| 23:08.58 | p3nguin | I would also change all those type=friend values to type=peer. |
| 23:09.22 | p3nguin | The nat value in each peer definition is for that peer. The ITSP is not behind NAT, so nat=no is appropriate. |
| 23:09.48 | p3nguin | And if you have a private IP address on Asterisk, it _is_ behind NAT. |
| 23:10.13 | jc319 | ok. 1x nat changed to no. 3x friends changed to peer. |
| 23:10.13 | p3nguin | You'll need to configure the rest of sip.conf soon. |
| 23:13.31 | jc319 | when a call comes in through the only registered sip channel, it enters the dial plan in 'comscom_home_inbound' context, and that has this >> [comscom_home_inbound]; exten => 02071486267,1,Dial(SIP/201,30); exten => 02071486267,n,Hangup(); |
| 23:13.45 | jc319 | so does this not mean directly call local extension 201? |
| 23:13.51 | jc319 | that's what I intend at least |
| 23:14.49 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 23:19.24 | p3nguin | It means that a call to extension 02071486267 should dial that SIP phone. |
| 23:19.32 | p3nguin | It does not mean call local extension 201. |
| 23:20.15 | p3nguin | Asterisk has no concept of "local" or not local. That is a human concept. |
| 23:20.52 | p3nguin | But Dial(SIP/201,30) says to Dial() the SIP peer by the name of 201, and ring it for 30 seconds before giving up. |
| 23:22.12 | jc319 | Is that not the correct way of re-routing calls once they land in to Asterisk? And I did nat=no & friend>peer modifications, what else should I change? |
| 23:22.20 | p3nguin | http://pastebin.com/tER2jGnY |
| 23:22.56 | p3nguin | That's not "re-routing" a call. That is properly accepting a call to an extension and dialing the device. |
| 23:23.59 | jc319 | OK I will check the pasted config, modify & test |
| 23:28.09 | jc319 | Can you clarify this ITSP static/dynamic setting? My IP is static, ITSP is presumably static, so is my setting going to be static? Is this something else? |
| 23:31.57 | p3nguin | Are you supposed to register to your ITSP, or do you configure IP authentication in their user panel? |
| 23:33.46 | jc319 | I see. I'll do dynamic for all for the time being, if possible I can change it to static later. |
| 23:35.08 | p3nguin | That's the difference between dynamic and static. To Asterisk, devices sending and/or receiving REGISTER (SIP registration) is known as dynamic, even if the IP address of the device does not change. If the IP address of a device is statically configured within Asterisk, regardless of the device's IP configuration mechanism, that's when it is known as a static configuration. |
| 23:35.47 | p3nguin | For example, I might configure my phone with a static IP address, but it still registers to Asterisk. The peer must be set to host=dynamic if the phone sends registration. |
| 23:36.50 | p3nguin | On the other hand, I might turn off the registrations in the phone, and then I would have to set host=1.2.3.4 if the phone's IP address was 1.2.3.4 because Asterisk would have no other way to know how to reach the device. |
| 23:37.16 | p3nguin | SIP registration is used to tell a "server" how to reach a "client." |
| 23:37.36 | p3nguin | I quote those terms because they are really just user agents with slightly different roles. |
| 23:38.49 | p3nguin | Asterisk is a Back-to-Back User Agent (B2BUA), and functions as both a client and a server. |
| 23:40.47 | jc319 | I believe I did all the modifications according to sip.conf. is my extensions.conf still missing something? I still have that 's' problem. |
| 23:40.58 | p3nguin | That's the fault of hte ITSP. |
| 23:41.27 | p3nguin | They are sending your phone calls to the 's' extension rather than a sensible phone number. |
| 23:42.11 | p3nguin | When a person has only one phone number, you usually can rectify that problem by adjusting the register statement. |
| 23:42.36 | p3nguin | such as: register => 105696_90:2FsuonGrOuq4r@chicago.voip.ms:5060/8885551212 |
| 23:42.51 | p3nguin | The /8885551212 tells the remote side what extension to send calls to. |
| 23:43.27 | p3nguin | The problem that I see with this method shows up when you have more than one phone number with that provider under the same user name. |
| 23:44.19 | p3nguin | I, for example, have multiple DID numbers with voipms, and I have no reason to tack on a /extension on my register... they send my calls to whatever number the call arrived on. |
| 23:45.10 | jc319 | So nothing can be done on my side to fix this? No workarounds? Current configs: sip @ http://pastebin.com/CYmFpVuG && extensions @ http://pastebin.com/shqtSEZs |
| 23:45.38 | p3nguin | You may need to ask your ITSP technical support how you they expect you to distinguish between phone numbers. |
| 23:46.43 | p3nguin | externaddr should not have a port number (as far as I know). |
| 23:47.49 | jc319 | externaddr fixed now. I will contact them tomorrow and ask then. Their web site suggests they support Asterisk on customer side [ http://www.coms.com/ip-trunking.html ] |
| 23:48.36 | p3nguin | Are you not in USA? |
| 23:48.51 | jc319 | No, I'm in the UK |
| 23:49.50 | p3nguin | I didn't realize that when I was providing the inbound/outbound dialing examples. :/ |
| 23:50.29 | p3nguin | You'll need to redesign the extension used for outbound calling. I doubt you want _1NXXNXXXXXX for a pattern. |
| 23:51.25 | *** join/#asterisk wonderworld (~ww@port-92-201-110-39.dynamic.qsc.de) |
| 23:51.58 | *** join/#asterisk JuStIcIa_ (~justicia@190.167.52.240) |
| 23:51.59 | p3nguin | If you want to temporarily accept calls on the s extension just so you can get calls going, add extension s and make it do something useful. exten => s,1,Goto(02071486267,1); |
| 23:52.24 | jc319 | Oups, I changed it in the previous edits, but this one is still in _1NXX format. Changing now, thanks for heads up. So at least I can call people. Damn just noticed it's the weekend so can't get this fixed for another week |
| 23:52.27 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 23:52.43 | p3nguin | That will get your calls, which are incorrectly being delivered to you, into your actual DID number as you have it configured. |
| 23:53.41 | *** join/#asterisk dhartman (~dhartman@wilug/newlug/ricko73) |
| 23:54.19 | jc319 | I have 3 DiDs, with this method I can start using only one. Basically all 3 of them will be delivered to 'default destination' and no way to seperate them, right? |
| 23:54.23 | p3nguin | Depending on the UK's numbering plan, you may end up with an outbound extension pattern of _XXX. to simply accept calls which are at least four digits in length. I don't know the UK's numbering plan. |
| 23:54.49 | p3nguin | Until we can learn why they are sending your calls to 's' |
| 23:54.56 | p3nguin | and/or get them to fix it... |
| 23:55.39 | jc319 | I have some plans saved here I'll have a look. I hope they can fix this. I read some ISPs intentionally don't deliver all the flexibility to push clients to buy 'answering machines' and such great rocket science services |
| 23:56.30 | p3nguin | That's the point where the service provider and I part ways. |
| 23:56.39 | p3nguin | VoIP.ms has a PoP in London. |
| 23:56.43 | p3nguin | Just sayin'. |
| 23:57.55 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
