| 00:01.30 | lukejt | I'm assuming it's some kind of bug, or undocumented behaviour |
| 00:04.08 | WIMPy | When / from where exactely is that function called? |
| 00:05.47 | WIMPy | And look at the endbeforehexten in cdr.conf. |
| 00:10.39 | lukejt | sip user dials a number, the call is passed to my agi script. my agi script then dials out to pstn. once the dial command is finished and the call is hung up, asterisk SIGHUPs my script. That's when the code is called, essentially in a DeadAGI but the same functionality is built in to standard agi now. |
| 00:11.35 | WIMPy | I'm not sure that's too early. Did you set that option? |
| 00:11.59 | lukejt | so the CDR values should be populated. I did try changing the endbeforehexten to no avail. I'll try again now. |
| 00:12.27 | lukejt | The CDR() variables seem to work fine when I'm not using Dial (e.g. STREAM FILE then HANGUP) |
| 00:12.38 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-147-242.pools.arcor-ip.net) |
| 00:12.39 | WIMPy | Seems likely to me that you can't do it inside the same script. |
| 00:13.18 | WIMPy | It is clearly defined that you can only do it in the h extension IF that option is set. |
| 00:14.02 | WIMPy | And even a SIGHUPed AGi probably prevents the dialplan from entering the h extension. |
| 00:16.49 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:20.09 | lukejt | same thing happens with that option, ill check debug output |
| 00:21.14 | *** join/#asterisk ex-parrot (~ex-parrot@116-90-136-82.static.dsl.unleash.net.nz) |
| 00:22.19 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.81) |
| 00:22.20 | ex-parrot | hey all, I'm running asterisk 1.6.2.9 in Debian, and in spite of calling Set(CALLERID(all)="Foo" <123>); and being able to NoOp verify that, when I tcpdump the resulting SIP packets the CallerID is not set |
| 00:22.54 | p3nguin | There are no quotes in a caller id name. |
| 00:23.22 | ex-parrot | ok, that's a start. I'll remove that and see what happens :) |
| 00:23.23 | WIMPy | And setting 'all' might not be the best option anyway. |
| 00:24.17 | ex-parrot | all I really need to set is the number. the weird thing is I'm changing between VoIP providers and this config all worked with the old provider, but tcpdump shows I'm definitely not sending the callerID, so I don't know how it was working :/ |
| 00:25.11 | WIMPy | Then you should go and compare the peer definitions. |
| 00:25.42 | ex-parrot | aha, fixed it. the new provider instructed me to use "fromuser" in the sip.conf, which was overriding it |
| 00:26.00 | p3nguin | If you only need to set the number, only set the number. Set(CALLERID(num)=123) |
| 00:26.37 | ex-parrot | thanks guys, easy fix :) |
| 00:27.33 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 00:27.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 00:32.14 | *** part/#asterisk ex-parrot (~ex-parrot@116-90-136-82.static.dsl.unleash.net.nz) |
| 00:40.27 | *** join/#asterisk catphish (~charlie@5e060ff3.bb.sky.com) |
| 00:40.52 | catphish | does anyone know if there's a specific problem with cisco 79xx phones and nat? |
| 00:41.45 | *** join/#asterisk rootkit3 (~kvirc@bzq-79-176-192-84.red.bezeqint.net) |
| 00:42.33 | p3nguin | Sometimes. |
| 00:42.51 | p3nguin | Not all 7900 series phones have the problem. |
| 00:43.17 | p3nguin | 7912 does not; 7970 does. |
| 00:44.29 | rootkit3 | I'm trying to recall a dialplan app name which allows playing music while continuing dial plan execution. Could you tell which one is that? |
| 00:46.16 | catphish | p3nguin: thanks, the phone is identifying as Cisco-CP7975G/9.2.1 |
| 00:46.32 | p3nguin | It's probably going to have the NAT problem. |
| 00:46.38 | catphish | p3nguin: what is the problem? if you have a moment to explain? |
| 00:47.14 | p3nguin | It's something about ports. Either the phone expects the media to return on the same port that it was sent on, or something like that. |
| 00:47.15 | catphish | since my experience has been that with nat=yes and no ALG interference, nat is transparant to handsets |
| 00:48.06 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 00:48.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:48.12 | p3nguin | I'm thinking that it expects the peer to use the source port for returning traffic, when that isn't how asterisk and NAT work. |
| 00:48.54 | p3nguin | I could have that out of order, though. That should be enough for some search keywords. |
| 00:49.05 | catphish | thanks for the pointers |
| 00:49.33 | catphish | the problem that's been reported to me is a registration issue, the phone doesn't seem to be responding to 401 Unauthorized with a password |
| 00:49.48 | catphish | but i'm getting this information second hand :( |
| 00:50.19 | p3nguin | Set nat=no for that peer. |
| 00:51.17 | catphish | sorry for all the questions, but what effect does that have |
| 00:52.37 | p3nguin | Asterisk's nat/rport settings have some effect on port translation. Changing it from forcing rport to never forcing rport will make some difference, even if not completely solving the problem. Any change could be a step in the right direction in the troubleshooting process. |
| 00:53.33 | catphish | ok, thanks, found some docs that state nat=no enforces RFC 3581 behaviour, so i'll read up |
| 00:53.34 | catphish | thanks |
| 00:54.20 | catphish | i've had such good luck using nothing but nat=yes and killing ALGs |
| 00:54.42 | catphish | but i guess it's not a universal solution |
| 00:56.45 | p3nguin | It is for most devices, sans Cisco. |
| 01:01.40 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 01:03.02 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 01:05.22 | catphish | ok, thanks again |
| 01:05.26 | catphish | should sleep now :) |
| 01:13.44 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 01:13.52 | *** part/#asterisk deo (~deo@58.71.19.178) |
| 01:25.04 | rootkit3 | I'm trying to recall a dialplan app name which allows playing music while continuing dial plan execution. Could you tell which one is that? |
| 01:25.31 | *** join/#asterisk minotaur01 (~minotaur@bas8-hamilton14-3096451123.dsl.bell.ca) |
| 01:25.33 | WIMPy | Background |
| 01:25.50 | WIMPy | Well, to collect digits. |
| 01:26.04 | WIMPy | To relly continue in the Dialplan StartMusicOnHold. |
| 01:26.10 | p3nguin | What execution, specifically, are you looking for after the music begins? |
| 01:27.37 | rootkit3 | just some cURL functions |
| 01:28.00 | p3nguin | What wimpy said should take care of that for you. |
| 01:29.33 | rootkit3 | Thank you WIMPy, that's the one |
| 01:30.02 | rootkit3 | I red about it some time ago, but forgot the name |
| 01:30.26 | p3nguin | core show applications like music <------ |
| 01:30.34 | p3nguin | That could have given some ideas. |
| 01:31.39 | rootkit3 | fast answers by humans on IRC is better :) |
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| 02:10.59 | k1ng | <rootkit3> fast answers by humans on IRC is better :) |
| 02:11.04 | k1ng | quote of the day |
| 02:11.06 | jonno11 | p3nguin: got everything working, thanks man |
| 02:12.00 | jonno11 | I want to record some IVR prompts - What's the best quality audio codec to use with asterisk that's supported? |
| 02:12.35 | jonno11 | I'm on a mac |
| 02:14.30 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.81) |
| 02:14.34 | [TK]D-Fender | jonno11: The best codec to use is the one your calls will use |
| 02:16.45 | *** part/#asterisk deo__ (~deo@122.53.72.218) |
| 02:17.01 | jonno11 | [TK]D-Fender: what does that mean? |
| 02:17.15 | [TK]D-Fender | jonno11: Exactly what it sounds like |
| 02:17.32 | jonno11 | ie. how do I know what my calls will be using? |
| 02:17.35 | [TK]D-Fender | jonno11: Whatever codec your calls will use is the best codec to record your prompts in |
| 02:18.02 | [TK]D-Fender | jonno11: They are YOUR calls. You CONFIGURE what CODECs Asterisk uses |
| 02:18.49 | jonno11 | I got that bit |
| 02:19.25 | jonno11 | [TK]D-Fender: So how do I configure that? |
| 02:20.51 | [TK]D-Fender | jonno11: vi sip.conf |
| 02:20.57 | [TK]D-Fender | jonno11: vi iax.conf |
| 02:21.08 | jonno11 | So any codec I use will sound the same? |
| 02:21.19 | jonno11 | No codec will sound better than another? |
| 02:21.34 | [TK]D-Fender | jonno11: Of course not otherwise why would you need multiple choices? |
| 02:21.34 | jonno11 | [TK]D-Fender ^ |
| 02:21.47 | jonno11 | Right. That was my question |
| 02:21.53 | jonno11 | Which is the best? |
| 02:22.15 | [TK]D-Fender | jonno11: The best is thie higghest common quality you can afford to use |
| 02:22.26 | jonno11 | I can afford? |
| 02:23.55 | rootkit3 | yep. some codecs cost money |
| 02:23.58 | [TK]D-Fender | In terms of CPU usage, bandwidth and if BW is short, $ |
| 02:25.04 | p3nguin | Since your calls are coming from the PSTN, alaw would make sense to use. |
| 02:25.46 | jonno11 | Aha ok |
| 02:27.29 | [TK]D-Fender | Had to scroll way back to see a reference to what you were even using. |
| 02:27.47 | rootkit3 | what are the default, min and max values for the VOLUME dialplan function? |
| 02:28.22 | [TK]D-Fender | So in in EU, ALAW would probably be the best choice. Anything else would eventually get transcoded back to that by Sipgate and would degrade quality further than if you started there |
| 02:28.49 | [TK]D-Fender | rootkit3: Zero. This is relative db |
| 02:29.17 | rootkit3 | and min/max ? |
| 02:29.50 | rootkit3 | Set(VOLUME(TX)=3) |
| 02:29.58 | rootkit3 | what is maximum? |
| 02:30.42 | [TK]D-Fender | rootkit3: Is your question more than theoretical? |
| 02:31.12 | rootkit3 | yes |
| 02:31.20 | [TK]D-Fender | rootkit3: So what's the actual problem? |
| 02:31.23 | rootkit3 | and I can't find an answer at the book |
| 02:31.31 | rootkit3 | or google |
| 02:32.03 | [TK]D-Fender | rootkit3: No, that is an answer to the generic question... what is the the problem that USING that function is supposed to solve that playing around with values so far hasn't? |
| 02:33.27 | rootkit3 | I'm exploring SIP volunerabilities with 'progress' and chanspy |
| 02:33.43 | rootkit3 | this is complex setup |
| 02:34.20 | rootkit3 | but I need to raise the volume of a spied channel as high as possible |
| 02:35.52 | [TK]D-Fender | Where does volume of RTP have anything to do with a SIP vulnerability? |
| 02:36.30 | rootkit3 | you asked - I answered |
| 02:37.24 | [TK]D-Fender | I just asked another question which I don't see anything that would constitute an answer to... |
| 02:37.26 | rootkit3 | and in my case it has something to do with it |
| 02:38.27 | [TK]D-Fender | "Something" ....Care to answer in a less than "secret squirrel" generic way? |
| 02:39.27 | rootkit3 | I told you it's something resulting from a complex setup |
| 02:40.02 | rootkit3 | which I don't have the time, or the will, to share with you here |
| 02:40.19 | [TK]D-Fender | "will". Yes. That is becoming quite apparent |
| 02:40.19 | rootkit3 | I'm just asking a simple question |
| 02:40.35 | rootkit3 | what is max value for volume function? anyone? |
| 02:40.50 | [TK]D-Fender | I'm not seeing anything in the source |
| 02:41.29 | rootkit3 | yes I know my english is good, but not perfect |
| 02:41.44 | rootkit3 | but at least the question is clear |
| 02:41.53 | [TK]D-Fender | I haven't commented on your english... |
| 02:42.13 | rootkit3 | so about what? |
| 02:42.32 | [TK]D-Fender | So far all I see it type INT for them |
| 02:42.36 | [TK]D-Fender | So what .... 16K? |
| 02:42.48 | [TK]D-Fender | have you TRIED setting it higher and tested the output? |
| 02:43.06 | rootkit3 | not yet |
| 02:43.17 | rootkit3 | I'm going to try in the mourning |
| 02:43.20 | [TK]D-Fender | Sinply testing should have taken less time than asking. |
| 02:43.29 | WIMPy | In my experience, even setting it to 2 gives annoying distortions. |
| 02:43.31 | rootkit3 | is it possible there is no maximum? |
| 02:43.39 | [TK]D-Fender | rootkit3: INT <- |
| 02:43.55 | [TK]D-Fender | rootkit3: Give or take SLIN's max dynamic range |
| 02:44.29 | [TK]D-Fender | There is a technical maximum, but you have something you are trying to practically test |
| 02:44.31 | rootkit3 | ok. so it must be tested |
| 02:44.47 | rootkit3 | the results vary probably |
| 02:47.03 | rootkit3 | if a maximum is not set, I guess it depends on system capabilities |
| 02:50.46 | [TK]D-Fender | maximum is INT db |
| 02:52.45 | rootkit3 | INT decibels could be any integer, I don't see a maximum here |
| 02:54.46 | rootkit3 | so it must be tested to see how high I can go without destroying quality |
| 02:55.15 | [TK]D-Fender | INT is a FIXED data type with a FIXED range. |
| 02:55.44 | rootkit3 | what is the range? |
| 02:56.22 | [TK]D-Fender | Now you've taken a question of quantitative range and applied to to a qualitative expectation of "quality". |
| 02:57.30 | rootkit3 | of course |
| 02:57.32 | [TK]D-Fender | http://lmgtfy.com/?q=c+programming+int+range |
| 02:59.20 | [TK]D-Fender | heads out for a while.... |
| 02:59.36 | *** join/#asterisk jonno11 (~jonno11@cpc2-walt12-2-0-cust286.13-2.cable.virginmedia.com) |
| 03:00.06 | rootkit3 | INT is an integer |
| 03:02.06 | rootkit3 | and my question has nothing to do with your C programming paradigmes |
| 03:02.20 | rootkit3 | or with your will to show us hoq clever you are |
| 03:02.28 | rootkit3 | hoq=how |
| 03:05.30 | [TK]D-Fender | rootkit3: Asterisk is written in C. I actually LOOKED at the code. Have YOU? I see that the function passes an INT. That is an AUTOMATIC limitation. So you asked a theoretical limit. THAT WAS IT. |
| 03:05.40 | [TK]D-Fender | rootkit3: I answered your question rather directly. |
| 03:06.07 | [TK]D-Fender | rootkit3: This is also something YOU could have gotten if you looked for yourself. I don't even CODE in C let alone do dev work for Asterisk |
| 03:06.55 | [TK]D-Fender | Anyway, have fun with it. Lets see where your newer implication of "quality" comes into play with that..... |
| 03:07.08 | [TK]D-Fender | actually heads out the door.... |
| 03:12.40 | *** join/#asterisk Sean-Der (~sean@cpe-68-175-54-68.nyc.res.rr.com) |
| 03:14.11 | Sean-Der | I am running some C programs with SHELL, they should only be lasting .05 seconds in user time. It feels much longer in Asterisk though, any ideas? |
| 03:14.38 | Sean-Der | I might look into patching Asterisk to include my functions, it will be tough but would be worth it |
| 03:16.54 | Sean-Der | I could also use backticks, I doubt this is different though? |
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| 03:32.34 | Sean-Der | What is a good place to start learning how to write Asterisk modules? I have written a good amount of C, but have only administered Asterisk never developed for it |
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| 05:46.31 | kakain | Can anybody tell me a reputable DiD provider? There are so many to choose from, and i would like to stick with the community recomends |
| 05:57.54 | drmessano | Flowroute |
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| 06:52.00 | apb1963_ | So I'm not getting any audio on my calls (I only have incoming calls at this time) and I peeked in the logs where I found: "NOTICE[4321] chan_sip.c: Disconnecting call 'SIP/from-IPTEL-00000002' for lack of RTP activity in 31 seconds" Aha! Says me. At which point I scratched my head and said "hmmm". I'm reasonably sure it has something to do with the fact that I'm behind NAT. *1.8 on CentOS 5.7 running as a guest under VMware Player 5, with an XP P |
| 06:53.11 | kaldemar | ~sipnat |
| 06:53.11 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 06:58.21 | apb1963_ | I think I have 3, 4 & 5 |
| 06:59.22 | apb1963_ | oh did i mention I also have DHCP assigned IP from my ISP and my asterisk is on a private network? |
| 07:00.10 | *** join/#asterisk Chandrakant (~chandraka@115.252.76.195) |
| 07:00.20 | Chandrakant | hell |
| 07:00.22 | Chandrakant | hello |
| 07:00.43 | Chandrakant | Having problem with 64bit machine 'n OS with asterisk odbc |
| 07:01.30 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 07:02.15 | Chandrakant | any one please help me on this issue |
| 07:02.40 | Chandrakant | i couldn't understand its ODBC problem with 64bit machine or something else |
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| 08:09.45 | bulkorok | hi |
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| 08:15.19 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.53.93) |
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| 08:47.42 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:47.50 | bombev | hia ll |
| 08:51.39 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 08:52.25 | bombev | I am trying to make context for incoming calls, because I have connected to boxes, and I need to restrict calls to fixed and mobile phones |
| 08:53.51 | bombev | I made this [from-internal-test] |
| 08:53.52 | bombev | exten => _XXX,1,Dial(SIP/${EXTEN},40,TtRr) |
| 08:53.52 | bombev | exten => _XXX,n,Hangup |
| 08:54.22 | bombev | but there is something wrong.. |
| 08:55.02 | bulkorok | _XXX cathces calls to 000 to 999 |
| 08:55.39 | bulkorok | _XXX. for more numbers |
| 08:56.20 | bulkorok | and in Dial you write SIP/peer/${EXTEN} |
| 08:57.26 | apb1963_ | ~dmz |
| 08:57.27 | infobot | [~dmz] De-Militarized Zone, or usually a separate physical or logical network that has limited access to your internal systems and is accessible in limited ways from untrusted networks such as the Internet. Putting Asterisk in the DMZ is not an acceptable alternative to properly forwarding the appropriate ports, so don't do it. Plastic router appliances generally do not implement DMZ well. |
| 08:57.40 | bombev | bulkorok I dont need to catch more number than 999 |
| 08:57.47 | bulkorok | ok... |
| 08:57.54 | bombev | i made test |
| 08:58.00 | bombev | and i got this |
| 08:58.07 | bombev | <PROTECTED> |
| 08:58.30 | bombev | Call rejected by 212.XX.XXX.12: No such context/extension |
| 08:58.43 | kaldemar | apb1963_: don't go that way. forward sip and rtp ports to your asterisk and configure the nat settings under general (nat=yes, localnet, externaddr) and each peer that is behind a nat (nat=yes). |
| 08:59.08 | bombev | i will try this SIP/peer/${EXTEN} |
| 09:00.03 | bulkorok | bombev: check the context in your peer-config |
| 09:00.45 | bombev | under peer confing should [from-internal-test] as well |
| 09:00.59 | bulkorok | without the [ ] |
| 09:01.00 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.53.93) |
| 09:01.14 | bulkorok | context=from-internal-test |
| 09:01.16 | bombev | oh yes |
| 09:01.20 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-017-137.pools.arcor-ip.net) |
| 09:01.23 | bombev | i forgot about this |
| 09:01.24 | bombev | :) |
| 09:01.27 | bulkorok | :) |
| 09:01.36 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.118) |
| 09:05.53 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 09:06.10 | bombev | bulkorok I still got: Call rejected by 212.XX.102.XXX: No such context/extension |
| 09:06.18 | bombev | the second box reject the call |
| 09:07.51 | bombev | i am sure that extension is correct, but not sure about the context |
| 09:08.16 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.118) |
| 09:12.06 | *** part/#asterisk ziro_axis (~h.sabrey@62.240.32.90) |
| 09:17.10 | kaldemar | bombev: you're showing a dialplan that uses sip and output that shows iax. |
| 09:17.45 | kaldemar | bombev: your call does not use the extension you are modifying. |
| 09:18.27 | bombev | i got it |
| 09:18.32 | bombev | already |
| 09:20.54 | *** join/#asterisk alsuren__ (~dlaban@80.169.133.251) |
| 09:23.22 | bombev | this way it works good: exten => _XXXX,1,Dial(SIP/${EXTEN}) |
| 09:25.36 | bombev | kaldemar i am showing dialplan uses sip because my extensions are under sip |
| 09:25.44 | bombev | but the trunks are under IAX |
| 09:28.48 | bombev | if I add 30: exten => _XXXX,1,Dial(SIP/${EXTEN},30) is this mean that the extension will ring 30 seconds |
| 09:28.48 | bombev | ? |
| 09:31.20 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 09:34.47 | bombev | kaldemar well, you are right sorry |
| 09:35.11 | bombev | should i change instead SIP to put IAX |
| 09:36.32 | kaldemar | i didn't say that. you should be modifying the part that does not work. |
| 09:36.49 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 09:38.00 | bombev | which part |
| 09:38.04 | bombev | do you mean? |
| 09:38.46 | kaldemar | you're the one who knows that. are you trying to restrict calling into some numbers only or are you having trouble calling from one box to another? |
| 09:39.29 | bombev | i am trying to restrict some numbers |
| 09:39.50 | bombev | because if i am using from-internal context |
| 09:40.03 | bombev | Box A can call via box B to fixed and mobile phones |
| 09:40.21 | bombev | I want to restrict calls to mobile and fixed phones |
| 09:41.33 | bombev | i want to allow only BOX A to call only to extensions BoX B |
| 09:42.39 | kaldemar | and what have you done so far and what issues are you experiencing? |
| 09:44.33 | bombev | when I am using this context which i wrote [from-internal-test] |
| 09:44.33 | bombev | exten => _XXXX,1,Dial(SIP/${EXTEN},40,TtRr) |
| 09:44.33 | bombev | exten => _XXXX,n,Hangup |
| 09:44.49 | bombev | I got this error Call rejected by 212.XX.XXX.12: No such context/extension |
| 09:45.08 | bombev | Called IAX2/realelastix/161 |
| 09:46.05 | kaldemar | you're not using that extension. |
| 09:46.30 | *** join/#asterisk Rodeo65 (~Rodeo65@ip-37-188-232-72.eurotel.cz) |
| 09:47.51 | apb1963_ | kaldemar: if I have nat=no, never or route IT WORKS. With nat=yes I get inconsistent results. This is all w/out any extern{ip,addr,host} including the working one. If I use externhost with any nat setting it fails in various ways. Since my public IP is dynamic I've been avoiding hardcoding the IP with extern{addr,ip} - do you believe that would make the difference? I've got SIP & RTP ports open on the XP Pro firewall, but it's not possible to actu |
| 09:48.51 | bombev | here is the full asterisk log: http://pastebin.ca/2291057 |
| 09:49.13 | bombev | kaldemar how come i am not using that extension |
| 09:49.40 | kaldemar | bombev: -- Executing [2161@from-internal:1] |
| 09:49.55 | kaldemar | your extension is in from-internal-test. |
| 09:50.47 | kaldemar | ask in #freepbx how to do what you want properly. |
| 09:51.06 | apb1963_ | me? Did that. I was told to try it in the dmz |
| 09:51.13 | kaldemar | apb1963_: no, not you. |
| 09:51.17 | apb1963_ | lol |
| 09:51.18 | apb1963_ | ok |
| 09:51.47 | bombev | kaldemar |
| 09:52.01 | bombev | this is box A Executing [2161@from-internal:1] |
| 09:52.10 | kaldemar | apb1963_: you need to define externaddr or you will continue experiencing issues. one way is to use a name instead of an address. |
| 09:52.21 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.119) |
| 09:52.22 | bombev | I put context=from-internal-test into box B |
| 09:52.31 | apb1963_ | so externhost and externaddr work differently? |
| 09:52.43 | bombev | so when box B receives call from box A |
| 09:53.05 | bombev | the context=from-internal-test takes effect |
| 09:54.35 | kaldemar | apb1963_: externaddr is looked up once when loading the sip module, with externhost you can define an interval with externrefresh for looking up the address. externhost is really the better choice for you. |
| 09:54.54 | kaldemar | apb1963_: since your address may change. |
| 09:54.55 | apb1963_ | That's what i thought. But it fails. |
| 09:55.45 | kaldemar | bombev: doesn't look like it does. |
| 09:55.49 | apb1963_ | The only settings that seem to work are nat={no,never,route} with no externhost. |
| 09:55.58 | kaldemar | bombev: you're talking about box B and showing output from box A... |
| 09:56.51 | apb1963_ | i just don't understand the implications. |
| 09:57.04 | kaldemar | apb1963_: if you don't define externhost or externaddr, asterisk will put a private address in the SDP portion of SIP invites. that means it request the remote end to send audio into a private address and the audio goes to the wrong place or nowhere at all. |
| 09:58.09 | kaldemar | apb1963_: leaving externhost/externaddr out is not an option. elaborate on "But it fails." <-- what fails? what do you see? |
| 09:58.15 | apb1963_ | I understand what you're saying, but.... how come it works in that scenario? |
| 09:58.28 | apb1963_ | It fails in various ways. |
| 09:58.31 | apb1963_ | Such as... |
| 09:59.13 | kaldemar | what exact scenario does it work in? |
| 09:59.23 | apb1963_ | Voicemail the first time. Voicemail + softphone blipped the second time. Got audio the third time. 2 lines rang on my 3CX soft phone on the fourth time. Fifth & Sixth time got audio. |
| 10:00.38 | bombev | kaldemar this what I got from Box B |
| 10:00.38 | apb1963_ | those are the failures |
| 10:00.39 | bombev | <PROTECTED> |
| 10:00.39 | bombev | Rejected connect attempt from 213.91.132.40, request '161@from-internal-test' does not exist |
| 10:00.52 | apb1963_ | not counting fifth & sixth time |
| 10:01.06 | apb1963_ | if I have nat=no, never or route IT WORKS. |
| 10:01.16 | apb1963_ | as long as I do NOT define externhost |
| 10:03.21 | kaldemar | bombev: so you see "161@from-internal-test" and your pattern is _XXXX |
| 10:04.00 | bombev | so |
| 10:04.04 | bombev | my pattern is wrong |
| 10:04.12 | kaldemar | apb1963_: where are you dialing from, the same LAN or outside the NAT? |
| 10:04.45 | apb1963_ | I dial from a regular phone. |
| 10:04.55 | kaldemar | what do you mean by a regular phone? |
| 10:05.05 | apb1963_ | the kind they had before VOIP existed :) |
| 10:05.19 | kaldemar | and how is that connected to asterisk? |
| 10:05.22 | apb1963_ | it's not |
| 10:05.28 | apb1963_ | PSTN |
| 10:05.33 | apb1963_ | POTS |
| 10:05.39 | apb1963_ | Ma Bell |
| 10:05.43 | kaldemar | the device is irrelevant, connection to asterisk is what matters. |
| 10:05.59 | apb1963_ | ok you asked where I'm dialing from |
| 10:06.06 | kaldemar | how does your asterisk connect to PSTN? |
| 10:06.24 | apb1963_ | through a VOIP provider |
| 10:06.29 | apb1963_ | ... iptel.org |
| 10:06.40 | kaldemar | via SIP? |
| 10:06.43 | apb1963_ | yes |
| 10:07.42 | kaldemar | do you have a DNS name for your public address? |
| 10:08.03 | apb1963_ | yes |
| 10:08.23 | kaldemar | do you have a mechanism to automatically update the record when your IP address changes? |
| 10:08.44 | apb1963_ | I do on the host, but not on the guest just yet. |
| 10:08.55 | kaldemar | guest? |
| 10:09.17 | kaldemar | do you mean host and guest as in a VM host and guest? |
| 10:09.33 | apb1963_ | *1.8 on CentOS 5.7 running as a guest under VMware Player 5, with an XP Pro SP3 host. |
| 10:09.36 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 10:10.01 | kaldemar | what kind of a network adapter does the centos have? bridged? natted? |
| 10:10.05 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 10:10.08 | apb1963_ | nat |
| 10:10.30 | kaldemar | does the host have a public address? |
| 10:10.34 | apb1963_ | My ISP assigns one DHCP address and the host gets it. |
| 10:10.38 | apb1963_ | yes |
| 10:10.58 | apb1963_ | asterisk/CentOS/guest is on a private IP |
| 10:12.31 | kaldemar | no need to have any update mechanism on the guest then. let the host take care of that. you must get the traffic to your guest using the XP host though (UDP ports defined in sip.conf and rtp.conf). |
| 10:12.43 | kaldemar | then configure the nat settings for asterisk. |
| 10:13.16 | apb1963_ | 5060 and 10001-14000 updated in rtp.conf and opened on xp firewall |
| 10:13.41 | apb1963_ | also opened in iptables on guest |
| 10:14.06 | kaldemar | "opened" as in forwarded to the guest? |
| 10:14.18 | apb1963_ | but nobody is forwarding... xp does not provide a mechanism and I haven't yet found a good guide on how to do it in iptables. |
| 10:14.27 | apb1963_ | no |
| 10:14.51 | kaldemar | you can't do it in iptables on the guest. it must be done on the host. |
| 10:15.05 | kaldemar | until you can do that, don't expect anything to work. |
| 10:15.06 | apb1963_ | that's kinda what I thought... but I've reached the flailing point. |
| 10:15.24 | apb1963_ | Just flailing in all directions now :/ |
| 10:15.47 | apb1963_ | yeah, host has no mechanism to forward ports |
| 10:15.49 | kaldemar | you can also do yourself a favor and remove the XP from the equation. it will only cause you trouble. |
| 10:15.58 | apb1963_ | xp is running vmware |
| 10:16.05 | apb1963_ | which allows me to run centos |
| 10:16.46 | kaldemar | doesn't matter what you run on the XP, if you can't forward ports your setup is a dead end. |
| 10:17.44 | apb1963_ | this all worked great until I asked my ISP why I couldn't traceroute past a device on their network at which point he investigated and said "hmmmm... that's weird.. yo're supposed to only get 1 IP address" at which point my working world fell apart. I was bridging until then. |
| 10:18.35 | apb1963_ | Well... apparently there is in fact a network editor for VMware... but so far I don't see a way to add a port range... only one by one. |
| 10:18.56 | apb1963_ | And adding 4000 ports one by one is nuts. |
| 10:19.07 | apb1963_ | Even if I reduce that to what... 100? Still a little crazy. |
| 10:20.12 | *** join/#asterisk chris-NB (~chris@212095007037.public.telering.at) |
| 10:20.27 | apb1963_ | Let me ask you this just be certain... externhost... does that get set to the name server of to my public address? I would think my public address. |
| 10:20.39 | apb1963_ | s/of/or/ |
| 10:20.41 | v0lZy | whats the problem? |
| 10:20.54 | apb1963_ | cool bot! |
| 10:22.12 | apb1963_ | I have no mechanism to forward a range of ports.. but I'm not sure that's the actual problem?? |
| 10:22.22 | apb1963_ | Kinda lost track here :) |
| 10:22.31 | bombev | kaldemar |
| 10:23.02 | bombev | I think I did it with that context |
| 10:23.09 | bombev | exten => _XXX,1,NoOp() |
| 10:23.09 | bombev | exten => _XXX,n,Dial(SIP/${EXTEN},30) |
| 10:23.09 | bombev | exten => _XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) |
| 10:23.09 | bombev | exten => _XXX,n,Hangup() |
| 10:26.52 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
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| 10:28.57 | apb1963_ | well, I can't think straight anymore... 2:30 am here... g'night |
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| 10:49.02 | bombev | can somebody help me with that context above I want to make some improvements |
| 10:50.31 | bombev | is there option if somebody call other number which is not XXX to make playback(this is wrong number, you can call only extensions) |
| 10:50.33 | bombev | for example |
| 10:52.01 | *** join/#asterisk singler (~singler@beta.kirneh.eu) |
| 10:53.32 | kaldemar | bombev: use _X. as a pattern. |
| 10:57.45 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 11:00.14 | bombev | aha |
| 11:00.56 | bombev | exten => _X.,n,Playback(this is wrong number, you can call only extensions) |
| 11:01.10 | bombev | this line should be above this one correct? exten => _XXX,n,Dial(SIP/${EXTEN},30) |
| 11:02.22 | kaldemar | whether it is above or below is irrelevant. order does not matter with extensions in a context. |
| 11:02.43 | kaldemar | for a four digit number XXXX is a better match than X. and will be used. |
| 11:03.17 | kaldemar | what you should do is make a new extension with the different pattern, i.e. use priority 1 instead of n. |
| 11:04.52 | bombev | http://pastebin.ca/2291065 |
| 11:04.58 | bombev | is it correct this way |
| 11:05.25 | kaldemar | did you read a single word i just wrote? |
| 11:06.00 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.122) |
| 11:06.01 | bombev | yes |
| 11:06.03 | bombev | of course |
| 11:06.14 | bombev | this is what i did actually |
| 11:06.21 | *** join/#asterisk ziro_axis (~h.sabrey@62.240.32.90) |
| 11:06.28 | bombev | I did new extension with priority 1 |
| 11:07.12 | kaldemar | http://pastebin.ca/2291067 |
| 11:07.41 | kaldemar | do you actually have sound files by those names that you're giving app Playback? |
| 11:07.57 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 11:09.04 | *** part/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 11:09.31 | bombev | kaldemar i will make sound file |
| 11:09.41 | as001 | hello can someone help me with callerid issue with my device. It seems like device get callerid number from post but when I dial fxs device Dial(Dahdi/1) I can just see incomming at device. When I plug in device in wall I can see callerid which means Asterisk on device does not send it properly to phone. Here are relevant files: http://pastebin.ca/2291066 |
| 11:11.10 | bombev | kaldemar so line 7 is correct that way,, thanks |
| 11:11.21 | bombev | should I add extra line 8 with exten => _XXXX.,n,Hangup() |
| 11:11.23 | as001 | I can talk that is ok but I can not force this ip02 device to send callerid to phone |
| 11:12.16 | kaldemar | bombev: yes, use a hangup after the playback. btw, your _XXX extension will play back those sound files even when the call was successful. |
| 11:13.06 | bombev | kaldemar: why? |
| 11:13.25 | bombev | will play those sound files |
| 11:14.07 | kaldemar | because the Playback is after Dial. |
| 11:14.36 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.122) |
| 11:15.23 | kaldemar | bombev: usually the contents of DIALSTATUS is checked after dial to know if the call was answered or had some other status. |
| 11:15.57 | bombev | well |
| 11:16.33 | bombev | if i put this playback before DIAL: exten => _XXXX.,1,Playback(This is wrong number you can dial only extensions) |
| 11:17.15 | bombev | when somebody enter number as XXXX. will play that message |
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| 11:20.17 | kaldemar | bombev: i was not talking about that Playback, but the one in the _XXX extension. |
| 11:20.39 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 11:21.20 | kaldemar | you should do some reading on the dialplan structure: |
| 11:21.21 | bombev | well this exten => _XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) |
| 11:21.23 | kaldemar | ~book |
| 11:21.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:21.40 | bombev | will play if nobody pick up the phone after 30 seconds |
| 11:22.11 | kaldemar | and when the callee hangs up. and when the Dial app exits for any other reason. |
| 11:23.15 | bombev | the cal will hangs up after the playback message |
| 11:24.47 | kaldemar | sure. so what? |
| 11:25.26 | bombev | am nothing |
| 11:25.30 | bombev | thanks for the help :) |
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| 12:57.52 | Sean-Der | Does anyone knows the correct files to include or link too to compile an Asterisk module alone? I am just messing around with app_skel and am getting AST_MODULE undeclared here |
| 13:03.24 | leifmadsen | Sean-Der: try #asterisk-dev |
| 13:03.39 | leifmadsen | note it is still pretty early for much of North America |
| 13:04.07 | Sean-Der | leifmadsen: Ok thanks! |
| 13:06.02 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 13:06.24 | qakhan | hi all, i am using asterisk 1.4.38, |
| 13:06.45 | slash213 | Sean-Der, i tried to do that and failed miserably |
| 13:06.56 | qakhan | how i can get hold, un-hold event on manager |
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| 13:07.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:07.13 | slash213 | no matter what i was doing, asterisk kept sending me off with "module compiled within different environment" |
| 13:09.56 | Sean-Der | slash213: Yea you will get errors if they have different compile time iirc? But I think they will still load |
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| 13:11.39 | slash213 | Sean-Der, maybe there's some force-load option, i dunno. i messed with it for an hour and decided to recompile everything from scratch. i wanted to upgrade, anyway. |
| 13:16.31 | kaldemar | qakhan: have you enabled events in AMI? are you not seeing MusicOnHold event? |
| 13:16.59 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:17.29 | qakhan | kaldemar how i can enable events in AMI? |
| 13:17.58 | WIMPy | There's even an explicit hold event. |
| 13:21.20 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 13:21.40 | kaldemar | qakhan: "Events: on" in the login action. |
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| 13:23.41 | qakhan | do i go in manager.conf and enable it? |
| 13:24.15 | leifmadsen | qakhan: he is saying when you connect to the manager interface through your script, at login also do Events: on |
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| 13:25.47 | qakhan | oh ok |
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| 14:07.50 | JoTo | hi all |
| 14:08.19 | JoTo | anyone can enlighten me what i should do/check when i want to establish a video call (from a doorbell system with camera) |
| 14:08.26 | JoTo | and i find the following in the logs: |
| 14:08.47 | JoTo | [2012-12-10 15:06:44] WARNING[10472]: chan_sip.c:9048 process_sdp: Ignoring video media offer because port number is zero |
| 14:08.47 | JoTo | <PROTECTED> |
| 14:09.04 | JoTo | i dont find any way to set a portnumber |
| 14:09.32 | JoTo | ofc i have already enabled video in settings and chosen the correct codecs in both (asterisk and doorbell) |
| 14:09.43 | JoTo | I'm lost...any hint very appreciated |
| 14:09.54 | JoTo | using freepbx distro with asterisk 1.8 |
| 14:10.37 | JoTo | i also played a bit with payload type settings in the doorbell system (i found this hint via google) |
| 14:10.40 | JoTo | but no success |
| 14:10.46 | JoTo | an changed it back to default |
| 14:11.20 | JoTo | so if any kind soul can afford some minutes i'd be very grateful...TIA very much |
| 14:11.40 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:12.06 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:12.36 | JoTo | oh...forgot to say: endpoint device is 3CX Softphone |
| 14:12.57 | JoTo | No webcam installed on my pc...i only want to receive video (doorbell camera at the front door) |
| 14:13.02 | JoTo | so no sending necc. |
| 14:17.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 14:32.50 | rgsteele | Any recommendations for a successor to the tdm400p? Finally EOL-ing some ancient PBX's. I've been planning on grabbing some VPMADT032 bundles, but I'm open to alternatives if a better one exists. |
| 14:35.06 | rgsteele | As a related aside, will the tdm410 will fit a 1U? Haven't seen any form-factor measurements for it yet... looking around now to make sure the modules won't be too high. |
| 14:36.30 | *** join/#asterisk minotaur01 (~minotaur@bas8-hamilton14-3096451123.dsl.bell.ca) |
| 14:42.11 | p3nguin | Call the sales department and ask them for dimensions. |
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| 15:53.21 | PipBoy | I am getting a Got SIP response 603 "Declined" back from x.x.x.x:5060 -- SIP/xxxxxx-0000015d is busy on a PBX to PBX call. Been trying to figure out what I missconfigured for the last hour lol. The two PBX's are registered via Sip, and are type=peer . Any ideas? Id appreciate it |
| 15:57.47 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
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| 16:04.32 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 16:04.47 | [TK]D-Fender | PipBoy, You didn't answer the call. |
| 16:05.01 | [TK]D-Fender | PipBoy, And didn't pass back any other status |
| 16:06.14 | p3nguin | Could be a DND(busy) enabled on the phone. |
| 16:09.18 | [TK]D-Fender | p3nguin, No, that would be passed back as a "last status" |
| 16:10.01 | *** join/#asterisk camerin (hoax@46.4.214.112) |
| 16:11.16 | PipBoy | Hmm ok thanks |
| 16:11.52 | *** join/#asterisk Vince-0 (~Vince-0@41-135-218-85.dsl.mweb.co.za) |
| 16:12.01 | PipBoy | I found I have more issues than that. Just did a call, and it didnt write it to mysql lol. So, looks like im going to have a fun day :) |
| 16:12.11 | [TK]D-Fender | PipBoy, You should have seen this by actually looking at the calls on both servers.... |
| 16:12.32 | PipBoy | I did look at the call on both sides. Just couldnt make sense of it. |
| 16:12.48 | [TK]D-Fender | PipBoy, Byh default, unanaswered calls don't get written to CDR (don't just say "MySQL") |
| 16:13.29 | PipBoy | Well I found out one of my issues. I did an outbound call cross site. And it worked fine (out to my cellphone).. |
| 16:14.09 | PipBoy | and I do see it in /var/log/asterisk/full |
| 16:14.25 | PipBoy | I was under the impression I wouldnt if it was writing the cdr to mysql |
| 16:14.38 | Sean-Der | PipBoy: CDR has multiple backends |
| 16:15.01 | Sean-Der | PipBoy: Your correct in this case, but other people use PostgreSQL etc.. |
| 16:15.31 | PipBoy | I was aware of the multiple backends. But not aware that it would write to more than one at a time |
| 16:16.08 | PipBoy | if indeed it does |
| 16:17.27 | Sean-Der | PipBoy: I bet you could do some magic with odbc if you really wanted to :) No you can use different backends at seperate times |
| 16:17.40 | Sean-Der | Its just a naming choice (Linux vs GNU/Linux) thing |
| 16:18.16 | PipBoy | Haha me +magic is a bad idea. I learn new things very slowly unforunately. |
| 16:18.25 | PipBoy | but mysql is a registered backend http://pastebin.com/RZE84sB8 |
| 16:19.08 | *** part/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 16:19.39 | ghost75 | (17:12:48) [TK]D-Fender: PipBoy, Byh default, unanaswered calls don't get written to CDR <- this can be changed? |
| 16:20.14 | PipBoy | I am pretty sure it can in cdr.conf |
| 16:20.30 | PipBoy | ;unanswered = no to =yes |
| 16:20.37 | [TK]D-Fender | ghost75, Obviously. |
| 16:21.36 | ghost75 | when i use custom_cdr it will still use cdr.conf ? |
| 16:22.28 | [TK]D-Fender | ghost75, It should. |
| 16:27.05 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
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| 16:28.10 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:29.49 | PipBoy | Hmm.. well Ill be damned.. It appears my issues a simple username/pass issue. I was under the impression freepbx used no pass on mysql by default.. but nope, they have one and its randomly generated |
| 16:31.53 | Sean-Der | PipBoy: FreePBX <--- Found your problem :D |
| 16:32.46 | [TK]D-Fender | Not looking and making crazy assumptions <--- Found your problem :D |
| 16:34.41 | PipBoy | Pffft.. At least I openly admit my assumptions are out there lol |
| 16:36.33 | [TK]D-Fender | That's what elevates an assumption to "stupid". You know it's there and you not only do it knowingly, but then flaunt it ;) |
| 16:36.51 | [TK]D-Fender | pii11 steps to go! |
| 16:38.09 | PipBoy | you are really streching that one.. I say things under the pretext that I am not an expert on the subject and am possibly wrong.. |
| 16:41.15 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.186) |
| 16:42.18 | apb1963__ | You're married <--- Found your problem. |
| 16:42.37 | PipBoy | LoL |
| 16:42.51 | apb1963__ | Swoosh. Nothin' but net. |
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| 16:55.46 | tonikasch | ciao |
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| 17:18.32 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 17:18.47 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx |
| 17:20.08 | Sean-Der | Katty: Did you restart it |
| 17:28.50 | Katty | Sean-Der: hmm? |
| 17:29.18 | chuckf | Katty: pick up the phone and dial 202-456-2121 and tell them its an emergency and your phone doesn't work. They'll help you. |
| 17:29.25 | Katty | teehee |
| 17:29.29 | Katty | in other news. |
| 17:29.46 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 17:29.52 | Katty | i'm packing and such for the move, and i found 4 old laptops in the closet. i've gotten through 3 of them and they all boot. |
| 17:30.00 | Katty | granted they need new batteries, but they live. |
| 17:30.24 | chuckf | that's great. My kindle battery (I think) died over the weekend:( |
| 17:30.47 | Katty | yuck )= |
| 17:30.53 | Katty | can those be replaced? |
| 17:31.16 | chuckf | yeah. not too hard |
| 17:31.19 | Katty | i'm kind of at a loss on what to do with the laptops. |
| 17:31.29 | Katty | i don't need them, tho i will probably keep one just in case my macbook dies. |
| 17:31.35 | Katty | i don't really want to just give them away to goodwill. |
| 17:31.58 | Katty | i've had a couple friends pipe up that they want them, but they don't want to donate anything for them...they just want to take them off my hands for me. |
| 17:32.07 | Katty | which is kind of irritating. i have a bunch of mooch friends. |
| 17:32.10 | chuckf | maybe a non-profit can use them |
| 17:32.34 | Katty | yeah, that or perhaps family members |
| 17:32.42 | Katty | christmas present perhaps |
| 17:32.56 | chuckf | Ah, I thought you had already ruled out family members |
| 17:33.38 | Katty | well my family, yes. |
| 17:33.41 | chuckf | with your interest in critters and such, maybe donate them to a local animal rescue |
| 17:33.43 | Katty | the boy's family could maybe use one, idk |
| 17:36.53 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 17:38.16 | paulc | Katty: Craigslist? I had an old laptop and a guy snapped it up - he wanted something with a parallel port so he could control some milling machine.. modern laptops don't have parallel ports so he was super excited.. |
| 17:43.21 | Katty | that's not a bad idea |
| 17:45.14 | PipBoy | I still keep my old IBM 386 laptop.. When people forget their Serial-to-usb adapter I force them to use the laptop :D |
| 17:49.29 | leifmadsen | I use minicom and a usb -> serial adapter for connecting to my soekris box |
| 17:55.13 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 17:57.39 | chuckf | If I need a serial/parallel port for something, its too old and needs to be replaced. |
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| 18:38.05 | *** join/#asterisk jkroon (~jkroon@41.18.224.250) |
| 18:38.28 | jkroon | hi all, in spite of asterisk sending annexb=no on g729 there are still a few foolhardy phones that insist on sending CNG parameters. |
| 18:39.08 | jkroon | now, looking at the spec one option may be to just keep repeating the last 10 bit frame but gradually decreasing the 4 gain parameters in the frame. |
| 18:39.38 | jkroon | would tampering with specific bits in the stream (without actually encoding/decoding) violate the patent (if anybody perhaps know)? |
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| 19:41.43 | danfromuk | Hi, gotoiftime does not have a YEAR option. Is year available at all in the dialplan to make an IF statement? For example something like gotoif[${YEAR}==2012] ? |
| 19:42.38 | *** join/#asterisk wonderworld (~w@dsdf-4db5c375.pool.mediaWays.net) |
| 19:46.19 | [TK]D-Fender | danfromuk, there us atime function you can use for this... |
| 19:46.25 | [TK]D-Fender | danfromuk, Go look at your functions list |
| 19:48.04 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 19:48.47 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 19:49.35 | raub | What was the asterisk-sounds-extra package replaced with? Looking at http://downloads.asterisk.org/pub/telephony/sounds/ |
| 19:49.53 | Qwell | What do you mean what is it replaced with? |
| 19:52.02 | raub | Qwell: I am installing 1.8 and do not remember seeing it |
| 19:52.15 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 19:52.16 | Qwell | It's been there since the dawn of time. |
| 19:53.17 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:53.29 | [TK]D-Fender | Upgrade your memory. It's cheap now. |
| 19:53.53 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.109) |
| 19:57.14 | jkroon | hi guys, i'm seeing cases where remote clients (using g729) will still send CNG parameters, even with asterisk always sending annexb=no |
| 19:57.38 | jkroon | this exhibits itself as nasty format_g729.c: Invalid data length, 2, should be multiple of 10 messages |
| 19:58.15 | jkroon | after a bit of googling it seems that asterisk used to pass a=silenceSupp:off - - - - in the sdp, but will only pass that under specific circumstances now. |
| 19:58.59 | jkroon | the two clients in question is a yealink phone (can "fix" it by disabling in it's config) and the conaito soft phone (can't be fixed and producer is refusing to assist) |
| 19:59.43 | jkroon | any ideas on whether re-adding the a=silenceSupp: line might get the conaito client into a sane state? |
| 20:00.02 | qakhan | i add member in a queue through addqueuememember(), member get add but its state is unavailable in queue |
| 20:00.39 | qakhan | i used UnpauseQueueMember() but still unavailable |
| 20:01.30 | danfromuk | [TK]D-Fender: I dont see a time function in the list. |
| 20:03.03 | danfromuk | Think i've found it. STRFTIME |
| 20:04.27 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::1) |
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| 20:13.19 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 20:13.19 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:13.50 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 20:15.41 | qakhan | i added member in a queue through addqueuememember(), member get add but its state is unavailable in queue |
| 20:15.46 | qakhan | i used UnpauseQueueMember() but still unavailable |
| 20:16.51 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 20:22.14 | citywok | qakhan: is it a sip peer? is the peer connected? |
| 20:23.23 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 20:25.51 | Qwell | Did you add it properly? |
| 20:26.33 | [TK]D-Fender | Qwell, Are you going to trust the answer? :) |
| 20:26.49 | Qwell | [TK]D-Fender: I'd risk my life on the answer. |
| 20:27.56 | *** join/#asterisk doolittlework (~doolittle@197.81.23.25) |
| 20:29.18 | leifmadsen | also, is callcounter=yes enabled in sip.conf? (assuming it is a sip peer) |
| 20:30.12 | saint_ | hi all |
| 20:30.13 | [TK]D-Fender | Se we've gone from assuming it's right at all to even its type. And bids on the next characteristic? |
| 20:30.15 | saint_ | i m new with asterisk . |
| 20:30.34 | saint_ | can someone tell me what is the difference between register and type=peer ? |
| 20:30.50 | Qwell | saint_: The same difference between apples and night. |
| 20:30.52 | saint_ | it looks like when i remove the register, i cannot receive call anymore. |
| 20:31.07 | leifmadsen | because a register tells the other end where you live on the network |
| 20:31.09 | saint_ | but weither i have register or type=peer , i can make calls |
| 20:31.22 | leifmadsen | saint_: see asteriskdocs.org and read the first 5 chapters |
| 20:31.36 | saint_ | leifmadsen: i m at chapter 10 |
| 20:31.44 | leifmadsen | then I suggest starting again |
| 20:32.06 | saint_ | leifmadsen: but it s not too clear. it shows an example with register, an example with peer . i understand that. i just made tests removing the register, and i can still send calls . |
| 20:32.27 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DeviceConfig_id291973 |
| 20:32.30 | saint_ | so i was wondering what are people using them for exacly. |
| 20:32.49 | leifmadsen | yes, register has nothing to do with placing calls from asterisk |
| 20:33.11 | leifmadsen | register tells the other end where you live on the network so it can send you calls |
| 20:33.23 | leifmadsen | e.g. asterisk might register with an ITSP (provider) |
| 20:33.32 | leifmadsen | a phone might register with asterisk so asterisk can send it a call |
| 20:33.36 | saint_ | okay |
| 20:33.47 | leifmadsen | a type=peer is used for authenticating the actual call -- registration has nothing to do with call setup |
| 20:34.05 | saint_ | leifmadsen: outgoing or incoming ? |
| 20:34.13 | leifmadsen | try Chapter 5 again |
| 20:34.19 | leifmadsen | saint_: yes. |
| 20:34.54 | *** join/#asterisk cyborg-one (~cyborg-on@188.115.135.131) |
| 20:38.06 | qakhan | @Qwell here is my dialplan |
| 20:38.08 | qakhan | http://pastebin.com/6cY7skSq |
| 20:38.24 | qakhan | and CLI output http://pastebin.com/aPACZ7Lv |
| 20:39.20 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:39.52 | Qwell | That's great. Where's the agent config? |
| 20:41.08 | qakhan | you want my agents.conf |
| 20:41.27 | doolittlework | hi there anyone here familiar with the atcom ip04 i see this is using iclinux my question is i want to use this unit for the small customer install, but i need to ftp the data of the unist to my call management system |
| 20:42.59 | qakhan | @Qwell here is my agents.conf http://pastebin.com/0XqLSr7t |
| 20:45.21 | Qwell | okay, great, now where are your agents being logged in? |
| 20:46.32 | qakhan | regencytaxi queue |
| 20:47.23 | Qwell | ...show me |
| 20:49.10 | *** part/#asterisk Sean-Der (~sean@cpe-68-175-54-68.nyc.res.rr.com) |
| 20:50.30 | [TK]D-Fender | exten => s,4,UnpauseQueueMember(regencytaxi,Agent/,004) <- Bad comma |
| 20:50.41 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.99) |
| 20:50.53 | [TK]D-Fender | exten => h,1,DeadAGI,/data/scripts/call_log.php|update|CALLIN|${call_log_id}|${DIALSTATUS} <- Stop using | as a delimiter |
| 20:51.25 | [TK]D-Fender | exten => s,6,AGI,/data/scripts/call_log.php|insert|CALLIN c<- and like the other use () to enclose your parameters. Stop breaking standardized syntax |
| 20:52.54 | [TK]D-Fender | qakhan, And I see you adding your agent... but nowhere do I see you LOGGING IN. |
| 20:53.11 | danfromuk | Can anyone help with a dialplan issue? http://pastebin.com/mcTCMScA |
| 20:53.14 | [TK]D-Fender | qakhan, Without actually logging in why do you expect to be listed as "available"? |
| 20:53.39 | Qwell | danfromuk: What are you expecting to happen? |
| 20:54.01 | leifmadsen | danfromuk: 1) you don't need those _ in front of the literal numbers |
| 20:54.02 | Qwell | There is no priority 2 that matches. |
| 20:54.14 | [TK]D-Fender | ^ |
| 20:54.18 | danfromuk | Qwell: _409 matches 204409 |
| 20:54.21 | citywok | ^^ |
| 20:54.22 | Qwell | No it doesn't. |
| 20:54.29 | leifmadsen | Qwell: he's trying to make the _X.,1,NoOp() fall over to the multiple priority 2's depending on the actual extension |
| 20:54.35 | [TK]D-Fender | danfromuk, 409 is 409. |
| 20:54.39 | leifmadsen | danfromuk: oh ya, that certainly doesn't match at all |
| 20:54.44 | [TK]D-Fender | danfromuk, that number is NOT the same, |
| 20:54.44 | leifmadsen | it matches 409 literally |
| 20:54.49 | qakhan | @Qwell here is queues.confg |
| 20:54.52 | qakhan | http://pastebin.com/DPKkuHJ6 |
| 20:54.54 | danfromuk | Ah. you are correct. |
| 20:54.59 | [TK]D-Fender | qakhan, No need. You have not LOGGED IN |
| 20:55.02 | Qwell | qakhan: I don't care about queues. We're talking about agents. |
| 20:55.03 | leifmadsen | you have no actual pattern matching on priority 2 |
| 20:55.04 | leifmadsen | _409. maybe |
| 20:55.07 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:55.19 | danfromuk | thanks. didnt spot that. copied and pasted it from somewhere else in the dialplan. |
| 20:55.19 | [TK]D-Fender | Qwell, he's shown no login <- |
| 20:55.22 | leifmadsen | danfromuk: what version of asterisk are you using :\ |
| 20:55.25 | danfromuk | 1.4 |
| 20:55.28 | Qwell | [TK]D-Fender: way ahead of you |
| 20:55.39 | danfromuk | Its a box that i inheritted. the owner doesnt want to upgrade |
| 20:55.54 | leifmadsen | that's fine -- just your dialplan is like... 1.0 styles :) |
| 20:56.01 | danfromuk | Probably is |
| 20:56.03 | [TK]D-Fender | Qwell, Ah yes, had to read back and you did catch it... |
| 20:56.15 | danfromuk | Can I do _.409 ? |
| 20:56.18 | leifmadsen | no |
| 20:56.18 | Qwell | no |
| 20:56.22 | leifmadsen | :) |
| 20:56.29 | leifmadsen | . == match 1 or more digits |
| 20:56.38 | leifmadsen | the 409 would mean nothing in that case |
| 20:56.47 | Qwell | match 1 or more digits *and stop processing* :) |
| 20:56.54 | [TK]D-Fender | danfromuk, Chapter 5.... |
| 20:56.55 | danfromuk | How can I match any number ending in 409? |
| 20:56.55 | [TK]D-Fender | ~book |
| 20:56.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:56.58 | [TK]D-Fender | ^^^ |
| 20:57.03 | [TK]D-Fender | danfromuk, You can't |
| 20:57.11 | [TK]D-Fender | danfromuk, There is no "ending with" pattern |
| 20:57.11 | leifmadsen | danfromuk: _X409 _XX409 _XXX409 etc |
| 20:57.32 | qakhan | [TK]D-Fender can you tell me how i can login an agent? |
| 20:57.35 | [TK]D-Fender | danfromuk, You can have a fixed position ending though as leifmadsen pointed out, but not "variable" |
| 20:57.55 | danfromuk | ok. thanks. i'll try that out. |
| 20:57.59 | [TK]D-Fender | qakhan, AgentLogin () <------ |
| 20:58.06 | citywok | you could use dialplan and string comparisons to figure out if it's truly ending in 3 digits regardless of the length |
| 20:58.08 | [TK]D-Fender | qakhan, BOOK TIME. |
| 20:58.26 | citywok | but that'd be pretty ghetto :p |
| 20:58.55 | [TK]D-Fender | citywok, But that would match everything (or an excessively wide range) just to "fail". Not ideal... |
| 20:59.11 | [TK]D-Fender | citywok, Ghetto would be the most polite way of looking at it |
| 20:59.12 | leifmadsen | or just pass it to an AGI and do your matching with regex :) |
| 20:59.13 | citywok | yea, hence the ghetto. lol |
| 20:59.37 | Qwell | or fix your numbering plan. |
| 20:59.47 | qakhan | [TK]D-Fender i found on INternet that agentcallbacklogin() and addqueuemember() is same |
| 20:59.50 | citywok | Qwell: why would he do the most sensical thing? that's crazy talk |
| 20:59.53 | Qwell | qakhan: NO |
| 21:00.02 | [TK]D-Fender | qakhan, Not at all. |
| 21:00.05 | danfromuk | Found the reason that the original code was working. It was being calls with Goto(context,${EXTEN:3},1) |
| 21:00.06 | [TK]D-Fender | qakhan, You read wrong |
| 21:00.07 | qakhan | hmmm ok |
| 21:00.21 | qakhan | let me try with agentcallbacklogin() |
| 21:01.01 | qakhan | [TK]D-Fender what is alternate of agentcallbacklogin() after 1.4.38 |
| 21:01.10 | leifmadsen | dialplan |
| 21:01.53 | leifmadsen | somethign like: |
| 21:01.58 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html#ACD_id288626 |
| 21:03.08 | [TK]D-Fender | qakhan, chan_local <- |
| 21:03.18 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 21:04.51 | Robotman321 | has anyone successfully gotten asterisk data to be graphed with cacti? |
| 21:05.09 | leifmadsen | I did once |
| 21:05.18 | leifmadsen | so it does indeed work :) just make sure res_snmp is setup and working |
| 21:05.55 | [TK]D-Fender | Robotman321, I dot Asterisk data ont cacti one ... but getting the needles out after isn't worth the trouble.... |
| 21:06.10 | Robotman321 | I thought so true.. looking at the mibs give me a headache.. |
| 21:06.21 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:06.23 | Robotman321 | *too |
| 21:06.47 | qakhan | [TK]D-Fender can u give me agentcallbacklogin() syntex |
| 21:06.53 | Robotman321 | smacks rrittgarn |
| 21:11.39 | [TK]D-Fender | qakhan, There is none. It is GONE. Us chan_local to loop the dialplan. |
| 21:13.43 | Robotman321 | [TK]D-Fender: just a fyi, found a fairyly current template system: http://forums.cacti.net/about31296.html just for a reference if anyone else ever comes looking for the same.. |
| 21:13.51 | qakhan | no i am using 1.4.38 |
| 21:14.03 | [TK]D-Fender | qakhan, then the app is still there |
| 21:14.14 | qakhan | yes sir i know |
| 21:14.20 | qakhan | but i dont know syntex |
| 21:14.23 | [TK]D-Fender | qakhan, so read the instructions. |
| 21:14.25 | [TK]D-Fender | ^^^ |
| 21:15.25 | qakhan | ok thanks |
| 21:19.41 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 21:20.15 | *** join/#asterisk tzaltzul_ (~tzali@46.120.34.48) |
| 21:21.14 | tzaltzul_ | hello. I am a NB. need help with connecting digium 4 ports E1 card to 2 freepbx . who can asist ? |
| 21:23.15 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:38.10 | citywok | tzaltzul_: you will probably be better off by starting with what you've tried, where you've gotten stuck, and asking specific questions. |
| 21:43.02 | tzaltzul_ | thanks. i need to connect 2 pbx to one asterisk via e1. how should the dialplan looks like ? dont need a direct answer, but where to look for... |
| 21:43.57 | PipBoy | you WANT? an indirect answer ? ummm look at google? |
| 21:45.15 | Qwell | ~book |
| 21:45.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:46.02 | tzaltzul_ | .. went to every available page.. no luck finding two or more ports config. |
| 21:47.11 | WIMPy | tzaltzul_: The dialplan is usually very individual. So you have to finduout what you want to happen. |
| 21:50.21 | tzaltzul_ | thanks infobot.. will take a look.. |
| 21:52.17 | *** join/#asterisk hattkunskapz (~hatt@unaffiliated/hattkunskapz) |
| 21:54.14 | tzaltzul_ | but I will start with a simple Q. is every port equals a span ? |
| 21:54.38 | tzaltzul_ | when I issue >pri show spans, do I have to see 4 spans ? |
| 21:54.52 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 21:57.11 | WIMPy | Usually yes. |
| 21:57.36 | jonno11 | Is there a pre-configured forwarding system in place for asterisk? |
| 21:57.54 | WIMPy | jonno11: What does that even mean? |
| 21:58.11 | citywok | yea... what do you mean? |
| 21:58.29 | jonno11 | So say I'm in a call, I could dial *20* followed by an extension, then that would place the caller on hold and forward to the extension |
| 21:59.04 | WIMPy | That doesn't make sense. |
| 21:59.09 | WIMPy | Are you talking about call transfers? |
| 21:59.12 | jonno11 | Yep |
| 21:59.45 | citywok | i guess that's a feature code then. http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 21:59.57 | WIMPy | Transfers do exist, yes. In multiple ways. |
| 22:00.57 | tzaltzul_ | WIMPy thanks . and under which file I configure the DAHDI/g1,g0 etc. I did not get the relation between group, context and g0,g1,g2 etc. |
| 22:02.03 | WIMPy | chan_dahdi.conf. And gX are the groups. Contexts are a dialplan thing. |
| 22:07.46 | tzaltzul_ | Thanks WIMPy, last Q. where is the groups definition ?.. |
| 22:11.07 | WIMPy | chan_dahdi.conf |
| 22:12.26 | tzaltzul_ | OK,, will give it a try, you made some sense. thansk man. |
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| 23:39.26 | Billiard | is it possible to have a dial timeout which is disabled on any sign of progress: ringing, early media, answer? I want a "connection timeout" :/ |
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