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| 09:50.55 | linocisco | is there any google employee ? I would like to ask which telephony system is being used. are they using asterisk? |
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| 10:12.57 | linocisco | is there any google employee ? I would like to ask which telephony system is being used at google. are they using asterisk? |
| 10:13.49 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 10:14.22 | Chainsaw | Yes, Google has employees. No, they will not tell a random member of the public about their inner workings. Next? |
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| 10:24.51 | Xaviertoor | http://www.digium.com/en/ |
| 10:25.19 | Xaviertoor | Google, Yahoo, ebay etc |
| 10:30.10 | linocisco | for example, HP is using avaya. cisco is using their own. just like that. I want to record and list. |
| 10:30.28 | linocisco | all US embassies are using Avaya |
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| 11:03.22 | zemmali-voip | hi guys ! please i need help to install asterisk on ec2 |
| 11:03.34 | zemmali-voip | You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. |
| 11:03.34 | zemmali-voip | make[1]: *** [oct612x-lib] Error 1 |
| 11:03.34 | zemmali-voip | make[1]: Leaving directory `/usr/src/lame-3.98.4/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux' |
| 11:03.34 | zemmali-voip | make: *** [all] Error 2 |
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| 11:06.15 | jp24 | Hi All, I was wondering whether 1.8.15-cert3 is going to be built into an RPM? Many thanks |
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| 12:25.24 | zemmali-voip | <PROTECTED> |
| 12:25.24 | zemmali-voip | Illegal instruction |
| 12:29.05 | zemmali-voip | i use centos and i find this pb please help me |
| 12:29.09 | zemmali-voip | <PROTECTED> |
| 12:29.09 | zemmali-voip | Illegal instruction |
| 12:29.57 | Chainsaw | This isn't a CentOS help channel. |
| 12:30.26 | WIMPy | Either you are using software unsuitable for your hardware or your hardware is broken. |
| 12:30.37 | Chainsaw | WIMPy: Mention of "Amazon EC2" earlier. |
| 12:30.54 | WIMPy | ok |
| 12:30.59 | WIMPy | Either you are using software unsuitable for your "hardware" or your "hardware" is broken. |
| 12:31.11 | Chainsaw | WIMPy: I see you like "cloud" solutions as much as I do. |
| 12:32.02 | WIMPy | There are uses, but far less than being used. |
| 12:32.34 | zemmali-voip | i use "Amazon EC2" |
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| 12:44.53 | Hunny | hi |
| 12:45.07 | Hunny | I did fail to ban for asterisk |
| 12:45.23 | Hunny | but its not blocking ip after ytries which i have mentioned |
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| 13:27.56 | tparcina | What has happen to asterisk-addons? |
| 13:28.47 | kaldemar | they're now included in the main source. |
| 13:30.03 | *** join/#asterisk linjan (~linjan@176.195.4.123) |
| 13:30.08 | tparcina | kaldemar: Since what version? |
| 13:30.49 | linjan | Hello! I have an strange question, but i dont know, where i can ask it. Anyone know, where i can but DID number in Belize? |
| 13:30.58 | linjan | i can buy* |
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| 13:37.30 | tparcina | kaldemar: It seams from version 1.8. |
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| 13:39.29 | saxa | what exactly means this |
| 13:39.32 | saxa | <PROTECTED> |
| 13:39.42 | saxa | UNKNOWN status ? |
| 13:40.00 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 13:40.01 | saxa | I can get calls but can not make them |
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| 13:42.48 | jploh | i moved /var/lib/asterisk but db.c still keeps looking at the old directory on version 11.6 |
| 13:43.20 | jploh | anyone else experiencing the same thing? it's looking for astdb.sqlite3 |
| 13:45.03 | kaldemar | saxa: nothing really. look at what happens before it. |
| 13:45.52 | kaldemar | jploh: did you change the dir in asterisk.conf? |
| 13:46.02 | jploh | yeah |
| 13:46.52 | kaldemar | care to pastebin the directories context? |
| 13:47.25 | jploh | http://pastebin.com/t1Ca5G6Z |
| 13:48.13 | saxa | kaldemar: to be honest not much |
| 13:48.14 | saxa | <PROTECTED> |
| 13:48.25 | kaldemar | jploh: are you familiar with (!)? |
| 13:48.43 | jploh | kaldemar: no. should i remove it? |
| 13:49.06 | jploh | kaldemar: thanks! |
| 13:49.15 | tparcina | Why is in the title "DAHD-linux 2.6.2" when there is dahdi-linux-2.7.0.1? |
| 13:49.22 | kaldemar | jploh: it disables the context. it is for defining templates that can be inherited from by other contexts. |
| 13:49.48 | jploh | kaldemar: got it |
| 13:49.50 | saxa | tparcina: probably they have not updated the web page |
| 13:51.15 | kaldemar | saxa: Wait does not make a call. you have at least two lines before that. |
| 13:51.21 | tparcina | saxa: I'm not talking about web page title, but the channel title. |
| 13:51.50 | kaldemar | tparcina: the channel title was updated nearly a month ago. just ignore it. |
| 13:52.56 | tparcina | kaldemar: Just it would be nice if it would show accurate information. |
| 13:53.21 | kaldemar | tparcina: sure. blame channel ops for that. :) |
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| 13:55.02 | tparcina | angler, LieutPants, pabelanger, Qwell: Please change the channel title ("DAHDI-linux 2.6.2" => "DAHDI-linux 2.7.0.1"). |
| 13:55.16 | tparcina | kaldemar: Hopefully someone will do something. :) |
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| 13:58.19 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:03.55 | saxa | kaldemar: just tried another thing and excluded a wrong context from my sip-phones context. |
| 14:04.03 | saxa | but now I'm getting this |
| 14:04.16 | saxa | http://pastebin.com/x2Bp91RX |
| 14:05.44 | saxa | and here is my extensions.conf |
| 14:05.48 | saxa | http://pastebin.com/Vv7RBqBh |
| 14:09.27 | *** join/#asterisk smirker (~x@101.162.177.251) |
| 14:11.25 | tparcina | I have Asterisk 1.6.2.16.2 that I need to upgrade to 11.6. |
| 14:12.00 | tparcina | Should I go 1.6 => 1.8 => 10 => 11 or |
| 14:12.00 | tparcina | 1.6 => 11.6? |
| 14:13.45 | *** join/#asterisk serafie (~erin@nat/digium/x-rdlflazgzfkvxkzo) |
| 14:15.43 | kaldemar | tparcina: no use doing the upgrades in between. just go to the version you're aiming for. |
| 14:16.15 | tparcina | kaldemar: Thank you. |
| 14:16.22 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 14:16.29 | kaldemar | saxa: enable pri debug |
| 14:16.48 | tparcina | And I hope that dialplan commands haven't changed much. :) |
| 14:17.30 | danfromuk | Hi. When using a queue, is there any way to get asterisk to put the answering agent into the cdr? |
| 14:17.33 | kaldemar | tparcina: they have, some. see UPGRADE*.txt in the source. |
| 14:18.19 | tparcina | kaldemar: Thank you. I didn't know about UPGRADE.txt |
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| 14:23.56 | saxa | tooth*CLI> pri set debug 10 |
| 14:23.56 | saxa | No such command 'pri set debug 10' (type 'core show help pri set' for other possible commands) |
| 14:24.18 | saxa | kaldemar: is it possible that pri is not enabled because of openr2 ? |
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| 14:29.00 | danfromuk | How do I reload agents.conf ? |
| 14:29.10 | saxa | agents reload ? |
| 14:29.29 | danfromuk | nope |
| 14:29.37 | saxa | ok just thought it :) |
| 14:30.32 | saxa | good, is there a way to catch the DNIS number in the extensions.conf ? |
| 14:31.04 | saxa | i can use this to make the direct calls to the extension I want correct ? |
| 14:31.33 | saxa | or is there an easier way to route the call to the desired extension directly ? |
| 14:34.49 | mirela666 | danfromuk: module reload chan_agents.so |
| 14:35.01 | mirela666 | danfromuk: module reload chan_agent.so < corection |
| 14:36.39 | saxa | kaldemar: i made a progress, changing the DAHDI/g0/{EXTEN} to DAHDI/1/{EXTEN} works on calling out. |
| 14:37.18 | saxa | so probably I have to set some things differently in my chan_dahdi.conf or see more in detail the group= paramenter I think. |
| 14:44.52 | kaldemar | you need to define the group above the channel lines in chan_dahdi.conf |
| 14:45.03 | saxa | it is afaik |
| 14:45.40 | saxa | but probably the problem can be sine I have only 10 channels enabled by my telco |
| 14:45.52 | saxa | not all of 31 available |
| 14:47.04 | saxa | ok, I got it, i had 2 group= lines |
| 14:47.16 | saxa | one was group=0 and the other group=1 |
| 14:47.22 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mvhovocrslyufank) |
| 14:47.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:59.38 | saxa | ok, strange, i have set also the trunkgroup= and span= paramenters in chan_dahdi.conf but still doesnt work. |
| 15:02.53 | saxa | good, sorted it out, I had a misconfiguration in here. |
| 15:03.13 | saxa | sorry for the noise, but its the first time i'm playing with this E1 thing. |
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| 15:06.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:20.10 | secesh | I'm having trouble with the record command |
| 15:20.15 | secesh | exten => s,n(record),Record(en/${recDir}/${callId}_comm.gsm,600,900,k) |
| 15:20.31 | secesh | recording always stops at 5 minutes |
| 15:21.27 | [TK]D-Fender | saxa: How many PRI do you have? |
| 15:21.27 | secesh | it seems I'm missing a timeout variable somewhere, but don't know what it might be. |
| 15:21.43 | *** join/#asterisk vlad_starkov (~vlad_star@217.26.6.70) |
| 15:22.24 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 15:37.03 | danfromuk | When an agent answers a call from a queue, is that recorded anywhere? I cant see it in the CDR. |
| 15:37.44 | [TK]D-Fender | You know queues have their own log, right? |
| 15:37.54 | danfromuk | No. I didnt. |
| 15:37.59 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 15:38.29 | [TK]D-Fender | This is part of the very basics on them. |
| 15:38.47 | [TK]D-Fender | There is a text file typically, and many pre-baked system set up SQL as well |
| 15:38.53 | [TK]D-Fender | it's in the docs |
| 15:39.06 | danfromuk | Ok, Taking a look now. Thanks for pointing that out. |
| 15:40.09 | [TK]D-Fender | it gives you the full detail of the time through as to entering, who answers, who ended the call, if they timed or bailed out, etc |
| 15:40.17 | linjan | Hello! I have an strange question, but i dont know, where i can ask it. Anyone know, where i can buy DID number in Belize? |
| 15:43.44 | saxa | [TK]D-Fender: i have one TE133 card with 10 channels enabled by the telco, but I have already sorted it out, I was missing the trunkgroup= and span= parameters in the chan_dahdi.conf , so setting them to the correct values, and setting the group=0 to group=1 it worked out. |
| 15:45.13 | [TK]D-Fender | saxa: You have no need of trunkgroup with a single PRI, that is for NFAS |
| 15:45.33 | [TK]D-Fender | saxa: Missing SPAN on the other hand is a DOA |
| 15:45.59 | [TK]D-Fender | (and that's for system.conf, not chan_dahdi.conf) |
| 15:46.33 | danfromuk | Is extconfig still reparsed when running asterisk -r ? |
| 15:46.55 | saxa | now I have this mess here :) http://pastebin.com/4uKMbiV7 and trying to sort out how to make the direct calls to the extensions. |
| 15:47.46 | pigpen | Anyone using Asterisk and the Polycom VVX phones with voice/video? |
| 15:47.52 | pigpen | Thoughts? |
| 15:48.02 | danfromuk | How do I reload extconfig.conf without a total reload? |
| 15:48.21 | secesh | dialplan reload |
| 15:48.34 | danfromuk | I dont think thats right |
| 15:48.44 | saxa | [TK]D-Fender: this is how it works for me right now, see here http://pastebin.com/229P1c0L |
| 15:49.45 | [TK]D-Fender | saxa: span=1,1,1 <- should probably be span=1,1,0. You shouldn't need LBO set unless you have a long run from your smart-jack to your card |
| 15:50.04 | saxa | what is a long run ? |
| 15:50.20 | [TK]D-Fender | 133'+ |
| 15:50.32 | saxa | ok I have nearly 1 meter :) |
| 15:50.40 | saxa | let me correct this. |
| 15:50.46 | [TK]D-Fender | saxa: Also half of what you've shown is for chan_dahdi.conf, not system.conf |
| 15:51.04 | [TK]D-Fender | either you've made a mess of your pastebin or you've mashed configs together inappropriately |
| 15:51.12 | saxa | i pasted both files |
| 15:51.17 | saxa | at once sorry |
| 15:51.32 | [TK]D-Fender | saxa: You also failed to specify your channels in system.conf |
| 15:51.34 | saxa | the system conf has only the lines after the # comments |
| 15:51.56 | [TK]D-Fender | line 7 onward is not system.conf stuff |
| 15:52.25 | [TK]D-Fender | Actually that whole thing is a mess |
| 15:52.29 | saxa | system.conf starts at line 43 |
| 15:52.32 | [TK]D-Fender | 51+ IS system.conf stuff |
| 15:52.39 | [TK]D-Fender | you mashed this up very badly |
| 15:53.02 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:53.04 | saxa | so I should correct the system.conf channels too ? |
| 15:53.37 | [TK]D-Fender | I can't tell how much is bad pastebin and how much is wrong things in the rwong file |
| 15:53.46 | [TK]D-Fender | Do it again clearly. |
| 15:54.13 | saxa | mfcr2 show channels shows the first 10 as active others are showed in an BLOCK state |
| 15:54.24 | saxa | ok [TK]D-Fender let me repaste it one by one |
| 15:55.28 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:55.30 | danfromuk | [TK]D-Fender: thanks. all sorted |
| 15:55.33 | saxa | http://pastebin.com/K4HXi5hN <- chan_dahdi.conf |
| 15:56.09 | [TK]D-Fender | saxa: 3-5 wrong for there |
| 15:56.19 | [TK]D-Fender | span does NOT belong in chan_dahdi |
| 15:56.26 | saxa | http://pastebin.com/ZFiqA3ah <- system.conf |
| 15:56.38 | danfromuk | [TK]D-Fender: actually,the queue log doesnt seem to show which extension answered the call. |
| 15:57.27 | [TK]D-Fender | danfromuk: it does |
| 15:58.07 | saxa | [TK]D-Fender: you are right it needs to be spanmap there. I just read it wrongly in the original chan_dahdi.conf |
| 15:58.52 | [TK]D-Fender | saxa: No. Spanmap is only for NFAS which you are not doing..... you have ONE PRI |
| 16:00.11 | *** part/#asterisk secesh (~matthew@c-71-229-49-222.hsd1.ga.comcast.net) |
| 16:01.25 | saxa | [TK]D-Fender: ok I will cut out trunkgroups and spanmap from that context |
| 16:03.02 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 16:03.18 | saxa | [TK]D-Fender: thanks for the corrections. |
| 16:04.36 | danfromuk | [TK]D-Fender: Got it. Any idea what format the data field is in? For example, event COMPLETEAGENT is "2|1|1" |
| 16:05.04 | [TK]D-Fender | that is the hangup.. not the answer.. |
| 16:05.13 | [TK]D-Fender | You should see an obvious line for the answer |
| 16:05.21 | danfromuk | CONNECT ? |
| 16:05.46 | [TK]D-Fender | correct |
| 16:05.54 | danfromuk | But what format is the data in? The CONNECT event is "2|{UNIQUEID}|1" |
| 16:06.34 | [TK]D-Fender | pastebin it. |
| 16:07.01 | danfromuk | Im using mysql. The data is 2|asterisk1-1384790089.20428|1 |
| 16:07.10 | danfromuk | asterisk1 is the name of this server. |
| 16:07.58 | [TK]D-Fender | danfromuk: dump the table |
| 16:08.07 | [TK]D-Fender | danfromuk: it should be the 4th parameter |
| 16:09.15 | danfromuk | I'm going to PM it you if thats ok. |
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| 16:37.18 | zafu | hi, does anyone remember the polycom setting to have the phone use the last way to dialing (headset/speaker) when dialing? |
| 16:39.47 | zafu | ah, up.headsetMode |
| 16:42.46 | *** join/#asterisk navaismo (~navaismo@189.241.101.4) |
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| 16:55.39 | Geek-Linux | Hello all, i want to configure ss7 on my asterisk boxes but a slight change is that i have same point codes(opc,dpc) for all my servers, Each server has its own signalling links, Now i want to configure each server separate with out loadbalancing. can some body help me.. |
| 16:56.58 | Geek-Linux | when i try to implement this i got and issue , some of my cis goes into ideal and some goes into idle reset pending ? |
| 16:59.34 | *** join/#asterisk CrashSys (~kumba@rrcs-97-76-33-146.se.biz.rr.com) |
| 17:00.02 | CrashSys | Is it little or big G that dials from a high channel down? like dial(dahdi/G1) would start at 23 and then go 22, etc? |
| 17:00.36 | outtolunc | g=low, G=high |
| 17:00.42 | CrashSys | ok thanks |
| 17:02.08 | outtolunc | remember there are the r/R's also |
| 17:02.31 | CrashSys | Yeah, I have a carrier in Canada who says that all calls must start on a high channel 23 and that's standard on all PRIs |
| 17:02.40 | CrashSys | told him he's full of it but couldn't remember which one did which |
| 17:02.55 | outtolunc | probably in/outbound circuits.. and wishing to neglect glare |
| 17:03.08 | outtolunc | er negate |
| 17:03.51 | CrashSys | I've just never seen a carrier require that before, most carriers we use r to eliminate glare on slow switches |
| 17:04.16 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:04.54 | outtolunc | the ones i've used in the past 30 years were always pick an end for each side.. and hope you don't meet in the middle (like they are suggesting) |
| 17:06.12 | CrashSys | I've never had a carrier care in the last 8 years or so, and it never brought down the D-Channel either |
| 17:08.18 | CrashSys | glare is more of a RBS thing though |
| 17:08.40 | CrashSys | this is a PRI |
| 17:15.41 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
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| 17:32.18 | *** join/#asterisk jeffasinger (~jeffasing@67.219.32.137) |
| 17:36.12 | jeffasinger | I'm looking through a dialplan, and it seems like there's a bunch of variables that should be loaded in from a mysql database, where would that database location normally be configured? |
| 17:37.55 | CrashSys | This carrier is now telling me he is going to change the PRI codec from g729 to G711 |
| 17:37.59 | CrashSys | this carrier needs to be shot |
| 17:38.02 | karl-s | huh... |
| 17:38.10 | karl-s | its probably a crappy flex circuit |
| 17:38.30 | karl-s | now watch as your faxes start to fail |
| 17:38.40 | CrashSys | It's voice |
| 17:38.44 | CrashSys | the d-channel is flapping |
| 17:39.08 | CrashSys | they are telling me the PRI has to dial high to low cause inbound is low to high (which is more of an RBF/LoopSTart thing) |
| 17:39.17 | CrashSys | and now telling me to reconfigure the codec from G729 to G711 |
| 17:39.44 | file | CrashSys, ... |
| 17:40.34 | CrashSys | I have no clue |
| 17:40.38 | CrashSys | this carrier is garbage |
| 17:40.39 | karl-s | :P |
| 17:42.34 | navaismo | jeffasinger, maybre from odbc.conf or res_config_mysql |
| 17:43.24 | CrashSys | The truly troubling thing is this is a call center. So the very core essence of their business was shopped around based on a google search for "Worlds cheapest long distance" |
| 17:44.03 | jeffasinger | hmmm, I don't have an obdc.conf, and res_mysql.conf is completely commented out |
| 17:45.44 | jeffasinger | navaismo: thanks for the pointer though, I think this may be related to the fact that I never built asterisk-addons |
| 17:46.17 | navaismo | uh an old version, I see |
| 17:48.10 | file | CrashSys, the cheapest long distance call is the one you never make... |
| 17:49.26 | CrashSys | and that's what they are doing ad nauseum :) |
| 17:49.48 | CrashSys | One thing I noticed is the call is placed out DAHDI/G1, but the CLI reverts it to DAHDI/I1 |
| 17:50.20 | jeffasinger | navaismo: I should've mentioned that, I'm trying to get this all to work on 1.4, because I know that's what this dial plan was originally working on |
| 17:52.43 | CrashSys | Maybe this is a newer behaviour in 1.8 |
| 17:55.23 | CrashSys | yeah |
| 17:55.24 | CrashSys | hmmm |
| 18:03.32 | Geek-Linux | Hello all, i want to configure ss7 on my asterisk boxes but a slight change is that i have same point codes(opc,dpc) for all my servers, Each server has its own signalling links, Now i want to configure each server separate with out loadbalancing. can some body help me.. when i try to implement this i got and issue , some of my cis goes into ideal and some goes into idle reset pending ? |
| 18:11.43 | *** join/#asterisk SupaYoshi (~supayoshi@ip4da5d319.direct-adsl.nl) |
| 18:11.55 | SupaYoshi | Does anyone have experience with resetting nortel 1140e devices? |
| 18:14.14 | jmetro | there might be an #avaya channel? |
| 18:15.04 | drmessano | Any ideas why a call to a DAHDI channel would generate a fast ringing at the far end? |
| 18:15.34 | drmessano | SIP <> Asterisk <> DAHDI <> Channel Bank |
| 18:16.16 | drmessano | I call any of the lines that ring on the channel bank, and rather than a long ring, followed by a long ring, I hear a fast ring-ring-ring-ring-ring |
| 18:16.59 | drmessano | End users think something is broken, so they hang up after the obscure rings |
| 18:16.59 | lvlinux | T1/E1 channel bank? |
| 18:17.02 | drmessano | Yep |
| 18:17.20 | drmessano | Rhino T1 channel bank, to be specific |
| 18:17.57 | lvlinux | sounds like maybe a setting in the bank, rather than * maybe? |
| 18:18.04 | drmessano | Hmmm |
| 18:18.41 | lvlinux | doesn't the bank send a ringing signal back through the T1 line, similiar to how a SIP ringing message gets sent? |
| 18:18.58 | lvlinux | (i could be wrong, but that's how i thought it worked) |
| 18:19.36 | drmessano | I recently rebuilt the system. Was fine before the rebuild. I copied the DAHDI config over from the old box. Nothing changed on the channel bank. Only difference is * 11 vs * 1.8 |
| 18:19.39 | jeev | drmessano |
| 18:19.47 | drmessano | (at least, thats the only difference I can see) |
| 18:20.06 | jeev | dude, i want to retire the as5300 i'm using t1 |
| 18:21.06 | jeev | it's just wasting rackspace and power.. what should i move the pri to |
| 18:21.46 | drmessano | jeev, I dont know. I don't have a PRI for inbound. |
| 18:22.06 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 18:23.13 | [TK]D-Fender | jeev: How much does it consume? What consumes less? How long would it take for the power savings to pay for the new solution? |
| 18:23.23 | [TK]D-Fender | jeev: How do you not already have this answer? |
| 18:23.57 | drmessano | [TK]D-Fender, any thoughts on my issue? |
| 18:24.19 | jeev | fender, i have some equipment, this wont cost me much, i believe it's using 2 or 3 amps, so about 24-36 bucks in power. it's also using 3U which I could use for something else. |
| 18:24.36 | jeev | i believe i have an adtran that can take the PRI and give me the same sip service. |
| 18:24.52 | jeev | or i can pop a pri card into a server and go direct into asterisk but i'm kind of scared to do that, |
| 18:25.37 | [TK]D-Fender | drmessano: If you're calling out... the ringing the far end seew shouldn't be any of your doing, should it? |
| 18:26.00 | drmessano | Calling in to one of the lines going to the channel bank.. Others work fine |
| 18:26.54 | [TK]D-Fender | jeev: No-one can solve the "I'm scared" part. So many of use these cards and have for untold years. |
| 18:27.21 | [TK]D-Fender | jeev: Yes, Adtran makes gateways. I'm sure they work fine. Many people here have used them before. |
| 18:27.22 | jeev | fender, that's not why i'm scared. |
| 18:27.37 | lvlinux | drmessano: have you tried checking/changing/setting the ring cadence in your chan_dahdi.conf? |
| 18:27.44 | [TK]D-Fender | jeev: You need to learn how to just make a decision. Facts don't seem to be the issue. |
| 18:27.56 | jeev | i can even just use a http://www.adtran.com/web/page/portal/Adtran/product/4212908L1/107 |
| 18:28.00 | saxa | hi, anybody can suggest how should I handle the incoming number to ring directly the selectes SIP device ? |
| 18:28.02 | jeev | i dont need the fx, i get ethernet and t1.. should do sip |
| 18:28.20 | [TK]D-Fender | saxa: it's your dilaplan.. make it dow hat you want |
| 18:28.32 | karl-s | jeev, i've been pretty happy with the G100 gateway. (although it had its growing pains early on) |
| 18:29.01 | jeev | hmm |
| 18:30.04 | Penguin | saxa: Do you mean all calls to your phone number will ring your phone? |
| 18:30.07 | lvlinux | saxa: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html |
| 18:30.29 | saxa | [TK]D-Fender: yeah i understand that, but what are the variables I have to use, I get the DNIS number correctly depending on the number called. |
| 18:30.44 | [TK]D-Fender | saxa: Go look at the call. |
| 18:30.57 | Penguin | saxa: Do you want only certain callers to go directly to your phone? |
| 18:31.31 | saxa | MFC/R2 call offered on chan 1. ANI = 7181962449, DNIS = 5903, Category = National Subscriber |
| 18:32.21 | [TK]D-Fender | saxa: And in the dialplan? |
| 18:36.19 | navaismo | not sure if trolling or real question |
| 18:37.11 | Penguin | After two of my questions were ignored, I'm not in the helping mood anymore. I would have told him exactly how to do what he was trying to do if he would have been clear in his goal. |
| 18:40.59 | saxa | Penguin: sorry i was busy over here |
| 18:41.36 | saxa | Penguin: I would like to make it as follows |
| 18:41.56 | saxa | i would like to use one number for the outgoing calls |
| 18:42.11 | saxa | or maximum 2 numbers from the 10 I have at my disposal |
| 18:42.33 | saxa | but would like that when somebody calls my direct number it rings only on my sip phone. |
| 18:42.56 | saxa | lvlinux: i'm reading that book right now, thx |
| 18:43.07 | Penguin | You use the word "somebody," but do you mean EVERYbody or only one? |
| 18:43.45 | saxa | [TK]D-Fender: I would like to know if I can get those ANI and DNIS variables in some way in the dialplan, as the teting dialplan I have handles this just with the _XXXX |
| 18:44.03 | saxa | Penguin: somebody is anybody who calls my number |
| 18:44.04 | [TK]D-Fender | saxa: "core show function CALLERID" <----------- |
| 18:44.30 | Penguin | Create an extension that uses Dial() to dial your device. That is all. |
| 18:44.35 | saxa | oh ok, is always with CALLERID() , will look at it. |
| 18:45.12 | saxa | Penguin: yes, but first I need to parse probably the incoming number |
| 18:45.20 | Penguin | Nope. |
| 18:45.28 | saxa | will * do that for me ? |
| 18:45.30 | Penguin | Becuase ALL callers go directly to your phone. |
| 18:45.51 | Penguin | If all callers go to your phone, we don't care who they are. |
| 18:47.23 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:47.25 | CrashSys | Anyone ever ran into an issue with a Cisco IAD2432 and PRI to Asterisk? |
| 18:48.12 | saxa | Penguin: no, there is 10 numbers available, so one of those 10 numbers will be mine, others will be from other people |
| 18:48.31 | CrashSys | Asterisk says the d-channel is flapping (up/down quickly), but the Cisco says everything is fine |
| 18:48.42 | saxa | so example, I have 11223300 as main number which will fall onto the central desk. |
| 18:48.56 | Penguin | That's what extensions are for. Create an extension for YOUR NUMBER that Dials YOUR PHONE. |
| 18:49.00 | saxa | but if somebody calls 11223303 will ring only on my desk. |
| 18:49.20 | saxa | ok got it. |
| 18:50.01 | saxa | but do i need to care if the DNIS will be 3303 ? |
| 18:50.30 | saxa | how * will know that if DNIS=3303 to ring only my sip extension ? |
| 18:50.56 | Penguin | There's no sip extension. There is only a sip phone. |
| 18:51.01 | [TK]D-Fender | saxa: because your dialplan should see that as the inbound extensions and YOU should ahve created a match for it |
| 18:51.17 | saxa | because with a global dialplan I have here right now, just for testing any number in range of 3300-3349 I call it always ring my testing phone. |
| 18:51.28 | [TK]D-Fender | saxa: Make a precise match |
| 18:51.33 | Penguin | Extensions run dial plan applications. In your case, the Dial() application puts the call to your phone. |
| 18:52.10 | saxa | ok so exten => 3303,1,Dial(SIP/saxa) ? |
| 18:52.45 | Penguin | If you call the number that you're working with, does asterisk look for extension 3303? |
| 18:52.46 | jmetro | exten => [whatever your server gets when its dialed],1,Dial([your tech]/[your peer]) |
| 18:53.27 | saxa | jmetro: ok, so the name of the extension should match the incoming number |
| 18:53.28 | jmetro | dial the number, see the error message asterisk gets that says "Extension [blahbllah] not found in context [your incoming context] |
| 18:53.36 | jmetro | make that your exten. |
| 18:53.41 | saxa | ok |
| 18:53.50 | Penguin | That's what extensions are for. |
| 18:53.56 | jmetro | some providers will let you choose what gets sent to your server. |
| 18:54.07 | saxa | will experiment a bit with it, I just tought I could use that DNIS part. |
| 18:54.32 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 18:54.33 | saxa | in some way, but if its not needed better :) |
| 18:54.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:54.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:54.48 | Penguin | Depends on what the call goes to when you call the number. Go look. |
| 18:55.23 | [TK]D-Fender | Looking appears to be the last resort regardless of being the first thing one is told to do. |
| 18:56.14 | saxa | http://pastebin.com/9HDRFqG3 as you see here, I just got some calls on channel 7 and 8 but the phone never ringed over here. |
| 18:57.15 | Penguin | Make a call to the number that you're trying to configure. |
| 18:57.47 | saxa | http://pastebin.com/0rJefN24 <- extensions.conf as it is now. |
| 18:58.13 | saxa | i think it should ring for any number that comes in as it is now. |
| 18:58.38 | [TK]D-Fender | You should probably enable CHANNEL LEVEL debug on that... |
| 18:58.38 | saxa | it should ring on SIP/ludmila&SIP/saladereuniao |
| 18:58.44 | Penguin | I see by looking at your dial plan that you want the people who call you to be able to do DTMF transfers. |
| 18:58.49 | [TK]D-Fender | As we see no details of what it is looking for. |
| 18:59.02 | [TK]D-Fender | And we don't see the line we get witht he other call that DID get accepted in the first place |
| 18:59.11 | Penguin | So if I call you, I'll transfer the call to someone else. |
| 18:59.17 | [TK]D-Fender | We are missing any kind of comparative analysis between these 2 calls... |
| 18:59.41 | saxa | i just saw that apeard that now on my phone |
| 18:59.52 | saxa | s/phone/asterisk CLI |
| 19:00.16 | [TK]D-Fender | And you should know the commands to get more debug for that specifica channel type |
| 19:00.45 | saxa | i have not enabled any debug yet, but will apply those command, need to find them out first :) |
| 19:01.06 | saxa | I know the sip set debug on , but for pri does not work |
| 19:01.24 | outtolunc | pri set debug on span 1 |
| 19:01.31 | saxa | mfcr2 also does not show much, but i have enbled debugging in the openr2 |
| 19:01.46 | saxa | outtolunc: let me try |
| 19:02.08 | saxa | tooth*CLI> pri set debug on span 1 |
| 19:02.09 | saxa | No PRI running on span 1 |
| 19:02.26 | outtolunc | then you are not config'd as a PRI |
| 19:02.34 | outtolunc | (on that span) |
| 19:06.01 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:08.12 | outtolunc | saxa: do a 'dahdi show status' |
| 19:08.31 | outtolunc | that will at least show what the system seeing |
| 19:09.17 | saxa | tooth*CLI> dahdi show status |
| 19:09.18 | saxa | Description Alarms IRQ bpviol CRC Fra Codi Options LBO |
| 19:09.21 | saxa | Wildcard TE133 Card 0 OK 0 -1 -1 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) |
| 19:09.45 | outtolunc | its CAS (like e&m, not PRI) |
| 19:10.06 | saxa | ok |
| 19:10.38 | saxa | there is no cas command in the cli |
| 19:11.10 | saxa | ok, I have to go now, l8r I'm back on that. |
| 19:11.14 | saxa | thx for now |
| 19:11.16 | saxa | c u |
| 19:11.16 | outtolunc | thats because there is no OOB signalling |
| 19:11.41 | saxa | ok have no idea, i'm first time using this kind of hardware. |
| 19:12.09 | saxa | c u |
| 19:12.15 | *** part/#asterisk saxa (~saxa@201-50-99-201.user.veloxzone.com.br) |
| 19:14.46 | *** part/#asterisk myself (~myself@ignignokt.mudkips.net) |
| 19:16.49 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 19:20.45 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 19:23.06 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 19:29.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:36.33 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 19:39.52 | cusco | http://www.masswerk.at/googleBBS/images/ |
| 19:39.56 | *** join/#asterisk crumb (crumb@gateway/shell/cadoth.net/x-jefeqkwsxngbtiqz) |
| 19:40.06 | crumb | asterisk -cvvv causes a segmentation fault |
| 19:40.13 | crumb | how do i find out what's causing it |
| 19:40.26 | WIMPy | Either you are using software unsuitable for your "hardware" or your "hardware" is broken. |
| 19:40.26 | cusco | have logger.conf write everything to full |
| 19:40.36 | crumb | cusco: me? |
| 19:40.39 | cusco | yes |
| 19:40.44 | cusco | and find out wich module is breaking it |
| 19:41.34 | Penguin | Try starting it with -ddddddddvvvvvvvv |
| 19:41.58 | crumb | gl_pathc 3 |
| 19:42.00 | crumb | Segmentation fault |
| 19:42.08 | Penguin | You may need to pastebin the ENTIRE output from the time you run the command until it seg faults. |
| 19:42.10 | *** join/#asterisk LoBoGoL (~LoBoGoL@200.162.194.189) |
| 19:42.49 | crumb | Penguin: http://pastebin.com/wGn5RFXN |
| 19:43.19 | Penguin | wtf |
| 19:43.23 | Penguin | There's nothing there! |
| 19:43.31 | cusco | lol |
| 19:43.34 | cusco | there is gl_pathc |
| 19:43.37 | cusco | what is that? |
| 19:43.45 | crumb | what's gl_patch 3 mean? |
| 19:44.08 | crumb | oh |
| 19:44.22 | jmetro | sounds like a typo |
| 19:44.52 | crumb | yeah, it's gl_pathc |
| 19:45.01 | cusco | seems xmldoc related |
| 19:45.04 | cusco | not to worry |
| 19:45.07 | cusco | perhaps your configs? |
| 19:45.13 | cusco | tried removing all from /etc/asterisk ? |
| 19:45.22 | cusco | also if all else fails: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 19:46.39 | Penguin | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:47.08 | crumb | i moved the entire /etc/asterisk dir, and it still segfaults |
| 19:48.40 | ChannelZ-Wk | Did you compile this yourself or is it a package? (or did I miss that part) |
| 19:49.56 | cusco | (as did I) |
| 19:51.17 | cusco | it is also worth noting, that distros are getting better at packagin software *for* *their* *own* *distro* - meaning, not to use some package you found on the web just because it uses the same packaging model as your distro |
| 19:51.38 | cusco | still, did you compile it from source? |
| 19:53.00 | crumb | my distro only provides a build script for asterisk which packages it |
| 19:53.00 | crumb | so yeah, it's built from source |
| 19:53.03 | Penguin | Which distro? |
| 19:53.06 | crumb | slackware |
| 19:53.11 | Penguin | Which asterisk version? |
| 19:53.18 | crumb | 1.8.20.1 |
| 19:53.55 | cusco | slackware can easely use rpm's watch out! |
| 19:54.27 | cusco | so... whats wrong with: ./configure, make menuselect, make, make install |
| 19:54.30 | ChannelZ-Wk | The " |
| 19:54.32 | ChannelZ-Wk | oops |
| 19:54.45 | cusco | bbl |
| 19:54.54 | crumb | http://slackbuilds.org/slackbuilds/14.0/network/asterisk/asterisk.SlackBuild |
| 19:55.05 | crumb | has the ./configure parameters ^ |
| 19:55.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.96) |
| 19:55.38 | ChannelZ-Wk | How old of a slackware install is it? I wonder if the compiler is a jacked version or something |
| 19:56.02 | crumb | gcc version 4.8.2 |
| 19:56.07 | Penguin | cusco: That's exactly what the slackbuild does. |
| 19:56.09 | ChannelZ-Wk | The 'No ethernet interface found' is troubling |
| 19:56.13 | Chainsaw | crumb: That's actually surprisingly new. |
| 19:56.40 | ChannelZ-Wk | The don't use selinux do they? Like it's locked off access to everythanng |
| 19:56.46 | crumb | why is it that when i mention slackware, everyone thinks it's something ancient |
| 19:57.00 | Penguin | They don't know. |
| 19:57.09 | ChannelZ-Wk | Crystal ball is broken. |
| 19:57.09 | WIMPy | Obviousely |
| 19:57.14 | Penguin | Feel free to enlighten them if they are under the wrong impression. |
| 20:00.08 | jmetro | why slack when you can debian |
| 20:00.14 | crumb | ok, i have gdb built |
| 20:01.39 | Penguin | Some of us don't necessarily like debian. Why do you live in a building instead of a paper carton? |
| 20:02.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:04.43 | crumb | http://pastebin.com/EBYuUkqw |
| 20:05.28 | *** join/#asterisk mirela666 (~mirko.bra@95.180.114.4) |
| 20:05.42 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:05.53 | [TK]D-Fender | This GDB was configured as "i486-slackware-linux". |
| 20:05.57 | crumb | http://pastebin.com/17cPBzsA |
| 20:06.03 | [TK]D-Fender | Does * even support i486 anymore? |
| 20:06.21 | crumb | it was tuned for i686 |
| 20:06.29 | WIMPy | Partially |
| 20:06.36 | WIMPy | (i.e. not really) |
| 20:07.18 | crumb | 0x081b33c0 in ast_xmldoc_load_documentation () |
| 20:07.25 | crumb | then.. 0xb747d133 in __libc_start_main () from /lib/libc.so.6 |
| 20:09.21 | crumb | http://pastebin.com/wvSLVcTZ |
| 20:09.37 | crumb | a bunch of No symbol table info available. |
| 20:12.29 | cmendes0101 | Would a digit only CNAM be valid? Sending it to the carrier and no error on it but been months and hasn't actually updated everywhere yet |
| 20:14.04 | crumb | http://pastebin.com/JMzuHL8z |
| 20:14.23 | Penguin | I wouldn't see a problem with a CNAM with only numbers. |
| 20:14.23 | ChannelZ-Wk | I don't see why not, but the CNAM system is kind of a mess |
| 20:15.14 | cmendes0101 | Thanks, I figured it "should" be ok but thought I would ask |
| 20:15.31 | WIMPy | Some software has trouble with names starting with digits. |
| 20:16.27 | ChannelZ-Wk | Or the year 2000 |
| 20:17.05 | WIMPy | Or the year 2010 (or was it 2011?). |
| 20:17.33 | WIMPy | Or the year 2038, off course. |
| 20:17.51 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:18.02 | Penguin | If it goes off course, it could wreck! |
| 20:19.49 | outtolunc | just like it's 19101 all over again ;) |
| 20:19.49 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.67) |
| 20:20.39 | dwayne | Anyone ever see (and know how to fix) Polycom 650's displaying "Caller ID Number" (line 1) followed by the DNIS (line 2) on the Caller ID screen even though Asterisk ${CALLERID(num)} and ${CALLERID(name)} contains properly received caller ID information? |
| 20:27.16 | *** join/#asterisk lordvadr (~lordvadr@jose-tc.ctc.biz) |
| 20:31.02 | Penguin | I wish there was some way to see detailed stats on items in ipset tables. |
| 20:31.36 | [TK]D-Fender | dwayne: Show us the call |
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| 20:33.35 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 20:34.36 | dwayne | [TK]D-Fender, ok, one sec |
| 20:35.01 | ChannelZ-Wk | Penguin: Detailed in what way |
| 20:37.36 | *** join/#asterisk g_r_eek (~g_r_eek@46-143-238.adsl.cyta.gr) |
| 20:37.43 | Penguin | I'm using iptables and match-set <myset>, which refers to ipset. My sets have lots of addresses, and I would like to see packets and/or bytes for each entry in the sets, much like the counters in iptables. |
| 20:38.30 | Penguin | If the address was matched by an iptables rule with -d addr, the counter would reflect things that have matched the rule. |
| 20:40.17 | Penguin | Since the iptables rule matches everything in a set, the iptables counter is cumulative. I would like to see detailed counters for the items in the set. |
| 20:40.51 | *** join/#asterisk dar123 (~DAR@2.90.101.148) |
| 20:43.49 | lordvadr | Hello, I need to get a macro or gosub or something run as soon as a channel is progressing during a Dial(). Like M(), or U(), but I need it to happen immediately after successful setup of a call. Anybody have any ideas? Asterisk 1.8.7. |
| 20:43.49 | Penguin | |
| 20:46.34 | Penguin | Isn't that what M and U options do? |
| 20:46.45 | lordvadr | No, they do that after the call is answered. I need it after it is setup. |
| 20:46.52 | lordvadr | e.g. progressing. |
| 20:47.17 | WIMPy | proceeding |
| 20:47.44 | lordvadr | On isdn it's called progressing, which is what I'm getting at...but yeah, thanks for that. |
| 20:48.17 | WIMPy | progressing is after proceeding. |
| 20:48.20 | Penguin | I guess I don't see a difference between "successful setup of a call" and "after the call is answered." |
| 20:48.26 | lordvadr | Specifically I'm trying to get at CHANNEL(dahdi_channel) while the call is ringing. |
| 20:49.00 | WIMPy | Maybe you should tell us what you want to do. |
| 20:49.05 | lordvadr | and, more importantly, depending on who hangs up, using g and F() to get at it doesn't work if the hangup comes from one direction. |
| 20:49.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:49.24 | WIMPy | I'm not sure what kind of events chan_dahdi throws. |
| 20:50.51 | crumb | Penguin, cusco, [TK]D-Fender .. it was -O2, i just swapped it with -g and no more segfault |
| 20:51.05 | lordvadr | I want to log span # and B channel # in CDR. On an inbound call, it's pretty easy, just put a Set(CDR(userfield)=${CHANNEL(dahdi_channel)}) during the "see if we're going to accept this" functionality. |
| 20:51.19 | WIMPy | crumb: I always use -O3 |
| 20:51.25 | Penguin | crumb: As in optimization? |
| 20:51.35 | crumb | yeah |
| 20:51.46 | WIMPy | What to you want the channel for? |
| 20:51.49 | lordvadr | On outbound, it's a whole lot more difficult. I've found no way to catch it if the call is never answered. F() works if the far end hangs up first. The 'g' flag functionality works, but the channel is already deallocated by then. |
| 20:52.53 | lordvadr | WIMPy: A number of reasons, mostly so I can put some graphs together and to find where there are buggy channels. |
| 20:53.17 | WIMPy | Buggy channels? What's that supposed to be? |
| 20:53.48 | WIMPy | Either the line works or it doesn't, but it's not down to specific channels. |
| 20:54.06 | lordvadr | We've had more than one instance where asterisk gets stuck in some sort of resetting state, or failed to deallocate the channel. Our PSTN provider then starts sending calls that get rejected by asterisk. We start hearing some complaints from customers and by the time we get enough info to go find it, it's a couple weeks later. |
| 20:54.48 | lordvadr | I'd like to already have the information to find it, or more importantly, that seeing calls on every channel throughout the day means everything is working correctly. |
| 20:54.55 | WIMPy | Ok, that was a known issue, but a simple upgrade should hopefully cure that. |
| 20:55.41 | WIMPy | You can do that without Asterisk. Just look at the D channel. |
| 20:56.14 | WIMPy | But you really should fix the issue instead of automating the detection of it. |
| 20:57.25 | lordvadr | We've actually seen it happen twice. First time I fixed it in source, filed a bug report with a patch, and got blown off. I still haven't pinned down the second issue yet, but this is a production system handling 20k calls a day. I can't just willy-nilly start shooting in the dark with upgrades. It was an upgrade that gave us this bug: https://issues.asterisk.org/jira/browse/ASTERISK-18899 |
| 20:57.26 | LieutPants | [ASTERISK-18899] [Status: Closed] [Assigned: lordvadr] Erroneous ISDN 44 Rejection Hangup() bug - https://issues.asterisk.org/jira/browse/ASTERISK-18899 |
| 20:57.27 | outtolunc | lordvadr: you might mod channel.c (look for AST_CONTROL_PROCEEDING and AST_CONTROL_PROGRESS) |
| 20:57.31 | dwayne | [TK]D-Fender, http://pastebin.ca/2477098 , re: Polycom CID Display |
| 20:58.34 | WIMPy | Well, that's why I prefer to use standard Linux drivers over the Digium ones. |
| 20:58.50 | lordvadr | There are "standard Linux drivers" for Digium hardware? |
| 20:59.02 | WIMPy | But I think they should also be ok by now. |
| 20:59.19 | WIMPy | No, not for Digium hardware, but for others. |
| 20:59.25 | [TK]D-Fender | dwayne: Nowhere in there do I see the debug going to the phone... |
| 20:59.35 | lordvadr | WIMPy: What are "they" that should also be ok by now? |
| 20:59.50 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:59.54 | lordvadr | And, what other hardware? We tried some OpenVOX hardware which promptly shit themselves. |
| 21:00.00 | WIMPy | dahdi/libpri/chan_dahdi |
| 21:00.18 | lordvadr | Where may I find these standard drivers? |
| 21:00.31 | WIMPy | The standard el cheapo stuff with CCD chips. |
| 21:00.53 | WIMPy | The famous HFC brand. |
| 21:01.19 | WIMPy | But there are others as well. |
| 21:01.23 | dwayne | [TK]D-Fender, sorry, coming up |
| 21:02.06 | lordvadr | Back when we built this system, there were only a handful of players in the quad PRI market. Digium and OpenVOX. I think there were some Sangoma cards as well, but I've never seen any el cheapo PRI cards. |
| 21:02.48 | WIMPy | The HFC ones are only available as single or dual span AFAIK. |
| 21:03.34 | WIMPy | But I found them to be a lot less trouble than the special Asterisk stuff. |
| 21:04.06 | Chainsaw | I use external hardware (Patton Smartnode gateways). |
| 21:04.33 | Chainsaw | And don't get me wrong, configuring them is hell on earth. Proper telco-grade "bring a consult or sacrifice a week of your life" stuff. |
| 21:04.40 | lordvadr | WIMPy: We've only had a handful of problems with the Digium hardware, and getting support on it is nice. I threw the openvox cards out the window. |
| 21:04.48 | WIMPy | But dahdi has more features by now. But not all the Linux drivers offer. So it relly depends on what you need. |
| 21:04.59 | Chainsaw | But if you need it to work always... it seems one of the only options. |
| 21:05.10 | Chainsaw | consultant, even. |
| 21:05.15 | Chainsaw | Typing is difficult. |
| 21:05.30 | lordvadr | Chainsaw: We've considered putting something in front of these boxes, but we do a lot of manipulation with the calls. |
| 21:05.49 | Chainsaw | lordvadr: It's just ISDN<->SIP really. |
| 21:05.52 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 21:05.58 | lordvadr | So far they've worked pretty well, just a few problems. ANYWAY, I only have what I have to work with. I'd like to log the B chan. Any ideas? |
| 21:06.02 | Chainsaw | lordvadr: You can manipulate all you want. It also does T38 correctly, which is a rare treat. |
| 21:06.12 | WIMPy | lordvadr: And losing calls is not much of a problem then? |
| 21:06.29 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 21:06.46 | WIMPy | As I said: Just look at the D channel and forget about Asterisk. |
| 21:06.47 | lordvadr | If I can fix it, it's not. It's when I can't and have to stay up all night trading patches with a guy in China that it becomes a problem. |
| 21:07.34 | WIMPy | I have hopes that that issue has been fixed. |
| 21:08.22 | lordvadr | I believe you. We had a similar but different issue recently (same symptoms at least) that having channel info in CDR would have been really handy. Also, upper mgmt wants to keep track of it. |
| 21:08.24 | WIMPy | But I think the issue causing both calls to be dropped on a "glare" situation still exists. |
| 21:08.53 | WIMPy | Channel numbers are pretty meaningless. |
| 21:09.10 | lordvadr | Not to the switch engineer at Windstream. |
| 21:09.37 | lordvadr | And we haven't seen any glare problems, even when we've tried to create them. Seems asterisk handles glare fine. |
| 21:09.55 | WIMPy | The CRNs are the omportant identifier. |
| 21:10.34 | WIMPy | As far as I know your issue is a consequence of the glare issue. |
| 21:10.36 | WIMPy | Or rather was. |
| 21:10.48 | lordvadr | Getting CRN might become something we want later. In any case, it's really easy to get span and channel on an incoming call. Outgoing is much more difficult. |
| 21:11.24 | WIMPy | From the dialplan. |
| 21:11.35 | lordvadr | We've got 4 PRI's in a trunk group from Windstream heading to two different servers. We'd really like to see what channel utilization looks like. I can get a count from the DAHDI/i[1|2], but nothing more. |
| 21:12.05 | WIMPy | What more do you need? |
| 21:12.24 | lordvadr | Ok, so that was the point of asking. If it can't be done well in the dialplan, I'll try to find another way to do it. |
| 21:12.36 | WIMPy | You know how many channels are in use per interface. |
| 21:12.53 | WIMPy | That's what I suggested. twice. |
| 21:13.14 | lordvadr | And if one never breaks N, I want to be able to pin down which channels are buggy. |
| 21:13.51 | lordvadr | Which was easy to do back before I asked Windstream to random-hunt in an effort to make similar problems in the future have less impact while we investigate. |
| 21:14.20 | lordvadr | No, there's no real predictability of where calls come in, but I can log them easy. I need to be able to log them out as well. |
| 21:14.30 | lordvadr | s/No,/Now,/ |
| 21:14.43 | lordvadr | dafuq? |
| 21:14.56 | lordvadr | Really, somebody wrote a bot to run sed commands? |
| 21:15.05 | Chainsaw | lordvadr: Oh yes. |
| 21:15.40 | Chainsaw | lordvadr: And it responds to some ~markers to help Fender fend off queries he doesn't want to answer. |
| 21:15.43 | jmetro | lordvadr: Look how useful it was. You just realized it without even using. |
| 21:15.56 | lordvadr | You know, I love the open-source community...some of them can be real shit-heads, but in general they're brilliant and friendly people. But every once in a while I wonder if they need better hobbies sometimes. |
| 21:16.02 | WIMPy | I could offer you a Tektronix 1205. That will answer all your questions. |
| 21:17.22 | lordvadr | I have an HST and a 2440, but I'd like a sleeker option. Thanks though. That being said, is there a good way to get the D-channel debug stuck in a different file the rest of the debug? |
| 21:17.48 | WIMPy | pri set debug file... |
| 21:18.25 | lordvadr | WIMPy: Thanks for that. I went looking for something like that but evidently missed it. |
| 21:18.50 | WIMPy | IIRC there was a n option added to do it completely outside of Asterisk as well. |
| 21:19.33 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 21:19.41 | WIMPy | IIRC there was an option added to dahdi to do it completely outside of Asterisk as well. |
| 21:20.57 | lordvadr | WIMPy: There's really interesting bug that pops up if you leave the kernel module in debug mode. |
| 21:22.33 | WIMPy | No, a usersapce app that writes pcap files. |
| 21:23.49 | lordvadr | WIMPy: I'll pass on that. Having a continuous log of the d channel would be really helpful for other reasons too, I just didn't know how to get it without it cluttering up the already cluttered verbose output. |
| 21:24.10 | lordvadr | So that's probably the route I'll go. Just wish I could do it in the dialplan also and get it into CDR. |
| 21:25.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.60) |
| 21:25.57 | *** join/#asterisk ickmund (~ickmund@h-91-126-243-207.wholesale.rp80.se) |
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| 21:34.50 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:35.56 | [TK]D-Fender | checkout time, BBIAB |
| 22:02.20 | *** join/#asterisk amessina_ (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 22:05.07 | *** join/#asterisk onizo (~onizo@wsip-70-166-5-159.sd.sd.cox.net) |
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| 22:29.34 | *** part/#asterisk crumb (crumb@gateway/shell/cadoth.net/x-jefeqkwsxngbtiqz) |
| 22:36.20 | *** join/#asterisk SupaYoshi (~supayoshi@ip4da5d319.direct-adsl.nl) |
| 22:36.28 | jeev | does an adtran channel bank or whatever this is, takes a t1, sends it out amphenol to a block which a nortel system is connected to.. does that pass date/time ? i unplugged and plugged back in and the time is off |
| 22:36.43 | SupaYoshi | does anyone here ahve experience with upgrading Nortel phones? I got everything setup but the display shows, Upgrades are availible |
| 22:36.46 | SupaYoshi | but it wont start the update... |
| 22:39.17 | *** join/#asterisk jploh (~textual@49.144.120.20) |
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| 22:44.08 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 22:46.48 | navaismo | try in #nortel |
| 22:47.23 | jeev | merci |
| 22:47.29 | jeev | oh |
| 22:47.31 | jeev | eh |
| 22:47.39 | jeev | there are -4 people in there |
| 22:55.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
| 22:57.24 | dar123 | my motherboard doesnt support pci 2.2 which is required by the telephony card i bought |
| 22:57.39 | dar123 | do i need to upgrade my pc ? :( |
| 22:58.07 | WIMPy | What card _requires_ 2.2? |
| 22:58.26 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-97-22.dynamic.qsc.de) |
| 22:58.32 | dar123 | its digium TDM410 |
| 22:58.39 | Chainsaw | WIMPy: Anything that will not do 5V signalling. |
| 22:59.32 | Chainsaw | WIMPy: http://upload.wikimedia.org/wikipedia/commons/6/6f/PCI_Keying.png |
| 23:00.10 | Chainsaw | WIMPy: The slot will appear to be 180 degrees reversed compared to the card, give or take. |
| 23:00.35 | WIMPy | But 3.3V has existed for longer, hasn't it? |
| 23:00.41 | Chainsaw | WIMPy: No. |
| 23:01.15 | dar123 | WMPy: i might be wrong but following the documentation it seems like that |
| 23:03.10 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 23:03.18 | WIMPy | According to wikipedia 5V support was removed in PCI 3.0. |
| 23:03.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:03.54 | Chainsaw | WIMPy: PCI 2.2 mandated 3.3V support. |
| 23:04.07 | Chainsaw | WIMPy: PCI 2.3 phased out 5V-only cards. |
| 23:04.37 | Chainsaw | WIMPy: We could talk shop about this all night if you wish, but it doesn't help dar123 fit a 3.3V-only PCI card into a 5V-only slot. Because that's what's going on here. |
| 23:05.13 | dar123 | is just did a "lspci -n" and wasnt able to find vendor id d161 |
| 23:05.22 | WIMPy | Well, it's not easy to find PC components that are compatible with each other. |
| 23:05.43 | Chainsaw | WIMPy: It is actually, but it requires prior study before purchase. |
| 23:05.58 | WIMPy | No, it's not. |
| 23:06.35 | WIMPy | So the card did fit physically? Then you shouldn't have an issue. |
| 23:07.03 | dar123 | yes the card fits perfectly |
| 23:10.19 | dar123 | i am new to linux, i might be missing some steps here. What do i need to do first for the card to work |
| 23:10.38 | WIMPy | Install the drivers. |
| 23:10.58 | Chainsaw | dar123: Based on Google image results, the TDM410p is a 3.3V/5V universally keyed card. Can I get lspci -vv output on pastebin please. |
| 23:13.15 | dar123 | whats pastebin, sorry for the question |
| 23:13.25 | Chainsaw | dar123: It's a website on pastebin.ca |
| 23:13.38 | Chainsaw | dar123: You can paste that long lspci -vv output there, and it'll give you a link that you can give me. |
| 23:14.26 | Chainsaw | dar123: Make it lspci -nnvv |
| 23:18.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
| 23:20.54 | dar123 | http://pastebin.ca/2477133 |
| 23:22.36 | Chainsaw | dar123: So you're running a VM. Have you set up PCI pass-through correctly? |
| 23:23.00 | WIMPy | Is that supported? |
| 23:23.41 | Chainsaw | WIMPy: Now there's a loaded question. |
| 23:24.25 | WIMPy | I was once told that it's not. |
| 23:24.35 | WIMPy | But that information might be out of date. |
| 23:25.34 | dar123 | i have VMware workstation 7 |
| 23:25.51 | Penguin | But does it have PCI pass-through? |
| 23:26.25 | dar123 | i do not have any option on it, let me see if version 10 has the option |
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| 23:31.12 | dar123 | ahh download is going to take some time |
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| 23:35.03 | Chainsaw | At any rate, three hurrahs for my friend lspci. |
| 23:35.20 | Chainsaw | As we'd all have spent a lot of time trying to figure out why that PCI card did not show up inside that VM. |
| 23:36.02 | *** join/#asterisk amessina__ (~amessina@50-196-241-78-static.hfc.comcastbusiness.net) |
| 23:38.52 | dar123 | :) |
| 23:40.07 | dar123 | is asterisk scf mature ? Just started reading about it |
| 23:40.28 | [TK]D-Fender | SCF died a while back |
| 23:41.02 | dar123 | seriously |
| 23:41.35 | [TK]D-Fender | yes |
| 23:42.00 | Chainsaw | Indeed, that ship has sailed. |
| 23:42.12 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 23:42.25 | [TK]D-Fender | Actually... that ship never launched |
| 23:42.26 | mjordan | and yet, the knowledge and lessons learned live on |
| 23:43.41 | dar123 | hmmm what if someone wants to build a very scalable asterisk deployment to cater millions of subscribers |
| 23:44.14 | [TK]D-Fender | Then they should rethink their business model |
| 23:44.15 | dar123 | was searching for a whitepaper or any howto on such deployment |
| 23:44.34 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 23:44.38 | [TK]D-Fender | Asterisk doesn't do "millions"., or tens of thousands. |
| 23:44.56 | [TK]D-Fender | You woul dneed a large infrastructure to spread such a thing over |
| 23:45.20 | dar123 | i have heard some people are using asterisk for services like viber |
| 23:45.50 | [TK]D-Fender | dar123: You'll find that paper on the 3rd shelf along with bigfoot, unicorns, and my raise from last year... |
| 23:46.00 | dar123 | offcourse not that big customer base, but still significant size. they put tons of servers to distribute the load |
| 23:47.38 | dar123 | i thought so too, asterisk is not designed to scale but the guy i met was very convincing |
| 23:48.28 | [TK]D-Fender | grabs a few of his pamphlets for "Ocean-front property for sale in Nevada" and prepares his pitch.... |
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| 23:49.49 | outtolunc | bridge_forsale.c |
| 23:49.58 | mjordan | define "scale" |
| 23:50.09 | Chainsaw | outtolunc: I was going to say something along those lines :D |
| 23:50.25 | [TK]D-Fender | mjordan: Someone's alrady gotten their hands on a few "ounces" ;) |
| 23:50.59 | mjordan | :-D |
| 23:51.32 | mjordan | It's funny. People always talk about scalability, but rarely do the definitions match up. |
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| 23:57.16 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mvhovocrslyufank) |
