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| 00:29.39 | igcewieling | One of these days I'll learn not to try .0 releaes of Asterisk 8-| |
| 00:30.47 | igcewieling | srl295: Generally 911 fee on an inbound only number does not make sense. Are you sure it is inbound only? |
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| 02:38.04 | greenwolf | helloo all... |
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| 03:02.46 | srl295 | igcewieling, callcentric confirmed the 911 fee is required even for their free DID account. FCC reg. |
| 03:07.16 | sawgood | well ... that is an interesting point |
| 03:07.57 | sawgood | FCC says "charge 'em all for 911 and send us the money"? |
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| 03:09.31 | srl295 | sawgood, if true, it means no truly $0.00 test accts for US residents |
| 03:09.38 | srl295 | sawgood, by anyone |
| 03:09.46 | sawgood | I know this guy who had an ex-girlfriend that called sandwhiches, "sammy's" ... that would drive me crazy |
| 03:09.54 | sawgood | would you like a roast beef sammy? |
| 03:11.30 | srl295 | What if your name was 'sammy'? |
| 03:11.37 | leifmadsen | my Dad calls them sandridges |
| 03:12.01 | srl295 | sammy, would you like a roast beef sammy, sammy? |
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| 03:14.36 | sawgood | its about time to head out for dinner, so I think I'll have soup and a sammy |
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| 05:12.41 | sorressean | I have a quick question, an organization I work with is using free confirence calls, but they're usually about as stable as win95. Is there somewhere good to either host an asterisk/freeswitch service with inbound calling that wouldn't cost much or a service that would do all of this? |
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| 05:23.43 | greenwolf | ipkall will give u a free incoming phone number..just setup ur own asterisk conference and point ur number to the room |
| 05:24.20 | greenwolf | then u can host and run the conference yourself and not have to worry about freeconference calling hosting your conferences anymore |
| 05:24.46 | greenwolf | i can build u something along those lines if your interested plz email me at unixlost@gmail.com |
| 05:28.11 | sorressean | hrm. so I'd just have to pay for voip. |
| 05:31.04 | sorressean | might use something like didwww.us or whatever that was. |
| 05:34.32 | sorressean | greenwolf: I appreciate the offer, but that looked like I'd be paying you. I'm not sure how I feel about paying someone money that doesn't have a professional domain and uses "ur and plz" when talking about doing business. |
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| 05:52.09 | ChannelZ | Keep in mind you need decent enough internet to handle as many people are in the conference. |
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| 07:44.46 | x1user | Shit happens on my PBX... I got Using SIP RTP CoS mark 5 message and cant figure out why.. |
| 07:44.55 | x1user | In the asterisk CLI as ana err |
| 07:45.13 | ChannelZ | It's not really an error |
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| 07:52.44 | x1user | Yes but cause me a lot of toble |
| 07:52.55 | x1user | trouble*, sometimes the call goes sometimes it doesnot |
| 07:53.05 | ChannelZ | Well that's a separate issue |
| 07:54.25 | ChannelZ | you'd have to look at a SIP debug to know for sure what's going on. Maybe an auth problem. |
| 07:55.20 | kaldemar | the RTP CoS print is not an error. not even a warning. just verbosity. |
| 07:56.07 | ChannelZ | It is a delicious cookie. |
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| 07:57.10 | fling | kaldemar: ChannelZ: hello :p |
| 07:57.34 | kaldemar | fling: hi. |
| 07:57.45 | fling | I've setup few asterisk servers, everything works, asterisk rocks! |
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| 08:00.07 | ChannelZ | ahoy |
| 08:00.13 | ChannelZ | and hurray! |
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| 08:03.44 | kaldemar | fling: good to hear. |
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| 08:09.40 | jkroon | if dahdi_scan outputs "alarms=BLU/YEL/RED" as part of the output from a TE122 card, what would that even mean? RED indicates no signalling at all, and BLUE indicates receiving of all 1's (which means that there is an upstream issue) - so how can we have both BLUE and RED at the same time?!? |
| 08:09.47 | jkroon | dahdi version 2.6.1 |
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| 08:12.01 | coppice | if you have a red alarm you should ignore the other alarms, their status is meaningless until the red alarm goes away |
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| 08:25.13 | ectospasm | jkroon: BLU/YEL/RED usually means you have your CRC4 parameter wrong in system.conf... |
| 08:26.06 | jkroon | ectospasm, ok |
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| 08:26.50 | jkroon | so just update it, rerun dahdi_cfg and restart asterisk? |
| 08:29.09 | jkroon | ok, switching crc3 off does not fix the problem, so with it on or off, same issue. |
| 08:29.20 | jkroon | ectospasm, any other ideas? |
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| 08:38.43 | ectospasm | I would try unloading DAHDI and reloading it, I can't remember if CRC4 is an option that can be cleared/set by dahdi_cfg... |
| 08:38.49 | ectospasm | jkroon: ^ |
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| 08:50.03 | jkroon | ectospasm, already done. |
| 08:50.35 | jkroon | is unloading / reloading the module good enough, or do I need to reboot/power cycle perhaps? the link did work flawlessly for quite some time - so we actually suspect of upstream issue. |
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| 08:53.20 | ectospasm | negative, a DAHDI restart should be enough |
| 08:53.40 | jkroon | ok, so leaving for upstream to confirm that the PRI itself is fine. |
| 08:53.50 | ectospasm | ...yeah, strange alarm states are usually the CRC4 option set incorrectly, a problem with the cable, or a problem with or beyond the provider. |
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| 08:54.14 | jkroon | ok, on a totally different note, when examining RTCP packets, the sent octets and bytes is not *end-to-end* but hop-to-hop? |
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| 08:55.06 | FluxiFlax2023 | hi all, I do have a panasonic IP PBX with many network based IP phones "not sip phones", is there any possiblity to setup an asterisk box and have it act like a ip phone or do I need extra hardware for that ? |
| 08:55.18 | jkroon | meaning that if I have two call legs, A -> B, and B -> C, then if frames from A->B gets lost it won't affect the RTCP values between B and C? B will simply report a lower "sent" value than A->B (and yes, rtp is going via B, not direct, yes, there is no way around this for me in this particular scenario) |
| 08:55.33 | jkroon | FluxiFlax2023, depends on the protocol in use. |
| 08:55.56 | FluxiFlax2023 | jkroon, c an you please elaborate ? |
| 09:01.22 | jkroon | FluxiFlax2023, what protocol does panasonic then use if it's not sip? |
| 09:01.24 | jkroon | unistim? |
| 09:01.26 | jkroon | h323? |
| 09:01.39 | jkroon | something else? |
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| 09:04.07 | kaldemar | FluxiFlax2023: "network based IP phones" does not really mean much. find out what protocols the panasonic pbx supports. if it speaks SIP, you can use asterisk with it. |
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| 09:10.54 | PbxMan | hello |
| 09:11.17 | ChannelZ | ohell |
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| 09:27.05 | schmidts | good morning |
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| 10:41.43 | jacekowski | hi people |
| 10:42.13 | jacekowski | i'm trying to reproduce behaviour of our old phone system on asterisk |
| 10:43.02 | jacekowski | basically our old system, displayed a message that person you are calling is on another call |
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| 10:45.15 | jacekowski | so i'm looking for some kind of text call waiting indication on the phone |
| 10:46.49 | WIMPy | Is there a "to" missing? |
| 10:47.03 | WIMPy | It's up to your phone, not Asterisk. |
| 10:47.36 | jacekowski | well, asterisk has to send some info to the caller that called party is on the phone |
| 10:47.53 | jacekowski | and then phone has to do something about it |
| 10:48.30 | WIMPy | Ok, so you want the "call is a waiting call" notification? |
| 10:48.41 | jacekowski | yes |
| 10:49.18 | WIMPy | If you phones can display text messages that should be possible by checking the device state before dialling. |
| 10:49.37 | WIMPy | But it would be limited to that account, off course. |
| 10:49.49 | jacekowski | well, i'm using digium phones |
| 10:50.15 | WIMPy | I haven't got that far with them, yet. |
| 10:50.47 | jacekowski | it's probably possible with new firmware |
| 10:51.18 | WIMPy | I haven't found any new firmware, either. |
| 10:51.29 | jacekowski | there is that beta version |
| 10:52.42 | WIMPy | Ok, yes, I saw a beta mentioned. But it still seems strange that there is no more than the original and a beta for a phone that has been on the market for quite some months now. |
| 11:01.15 | jacekowski | is there any standard call waiting indication that is sent to the caller via sip? |
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| 11:01.59 | WIMPy | Asterisk doesn't do such a thing and as far as I know there is no ISP (sub) standard to do that, either. |
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| 11:33.07 | schmidts | WIMPy imho there is something using notifys but i am not sure if this is used for normal call waiting indication or only in groups for pickup indication |
| 11:35.00 | dtcrshr | hello everyone! Iv seen on the wiki thats ubuntu recommended to use asterisk. Im my very low knowledge of linux, i bet debian would be much more "stable" and reliable than ubuntu for such an important server. Why should I go to ubuntu, isnt debian recommended? |
| 11:35.49 | WIMPy | It's recommended that you use what YOU're comfortable with. |
| 11:36.00 | kaldemar | dtcrshr: use a distro that you're most familiar with. |
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| 11:39.42 | dtcrshr | im pretty confortable on debian though |
| 11:40.30 | dtcrshr | i just gob bit by the wiki that points up to ubuntu, which is debian based, cared to ask here first what really differs from ubuntu to be preffered on the wiki |
| 11:40.47 | dtcrshr | maybe its just to make the install more friendily or something |
| 11:41.16 | WIMPy | What wiki are you talking about, BTW? |
| 11:41.30 | WIMPy | But you surely want to use a current version of Asterisk. |
| 11:41.39 | dtcrshr | wiki.asterisk.org |
| 11:42.09 | WIMPy | Interesting. I would have expected to read about Centos there. |
| 11:43.32 | dtcrshr | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 11:44.58 | kaldemar | which part says that ubuntu is recommended? |
| 11:46.22 | dtcrshr | THATs what im looking for |
| 11:46.58 | dtcrshr | the sysadmin claims that ubuntu IS the recommendated system, I only found that page relating to apts |
| 11:47.20 | dtcrshr | but the WIMPy comments already suited me to use the one im confortable |
| 11:47.30 | dtcrshr | i just want to encourage him to NOT use ubuntu as a server |
| 11:49.37 | WIMPy | Well, let me re-ohrase that. |
| 11:49.39 | dtcrshr | well, thanks anyway. ill try to work him out of ubuntu |
| 11:50.06 | WIMPy | It's recommended that you use the distribution that the one who is going to administrate it will be comfortable with. |
| 11:50.23 | dtcrshr | bingo |
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| 13:34.14 | BorjaGVO | Hi, where can I find the different arguments that "queue reload" can take? |
| 13:34.26 | BorjaGVO | I see {parameter | member |
| 13:34.40 | BorjaGVO | but tha doesn't help much |
| 13:36.03 | kaldemar | keep on hitting that tab. |
| 13:36.22 | kaldemar | and then "core show help queue reload members" etc. |
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| 13:44.21 | BorjaGVO | kaldemar: thank you |
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| 13:47.32 | danfromuk | Hi, i'm trying to upload a new greeting to mysql for use with voicemail using odbc storage. However when I try to dial in to the mailbox, i just get the standard alison greeting and the cli says app_voicemail.c:3462 retrieve_file: Unable to truncate 'file path removed' Invalid argument |
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| 14:18.23 | Katty | i have coffee :> |
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| 14:20.41 | cusco | I do too |
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| 14:27.55 | SuperNull | i could use a coffee. but i will wait till im done drinking my red bull |
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| 14:39.47 | schmidts | SuperNull mix it to get awaken |
| 14:41.46 | danfromuk | What format is required for voicemail greetings? |
| 14:44.27 | SuperNull | ntfs. |
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| 15:03.55 | rue_house | http://community.polycom.com/t5/PSTN/Soundstation-2-Volume/td-p/2535 <-- I got the same runaround regarding audio levels for the phones I got |
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| 15:25.24 | Elleni_ | hi all, could someone assist me in enabling to send message from one softpphone to another through asterisk server, please ? Is that easily configurable ? |
| 15:25.37 | Katty | looks in |
| 15:25.53 | Katty | HI LADS. |
| 15:25.56 | Katty | i broke my asterisk again. |
| 15:26.07 | Katty | it makes this noise: zzrrrggbbbb, zrgggbbbb. what is wrong plz??? |
| 15:28.54 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 15:29.43 | schmidts | Katty you have to change oil, first do a shutdown -h now, then open your pc with a screwdriver, insert some native olive oil and then it should be fine again |
| 15:30.18 | Katty | *hee* |
| 15:30.18 | [TK]D-Fender | rue_house, You said yours were too weak. that is also a speakerphone and analog. We'[re pretty much apples to apples, and there is a question fo what a person considers "right". When in doubt, people are crazy, possibly stupid. |
| 15:30.53 | schmidts | hehe :) |
| 15:31.34 | [TK]D-Fender | Apples & oranges. |
| 15:31.40 | [TK]D-Fender | Wow, not enough caffiene at all |
| 15:33.12 | Elleni_ | anyone a direction to push me? trying to find out how I could send messages from softphone to softphone, connected to my asterisk srv :) |
| 15:34.13 | [TK]D-Fender | "More info, my soundstation2 is network attached via a cisco ATA 186. Listening to bridge call when experiencing loud volumes, if I listen on a voip phone to same bridge I also hear one person with pretty loud volume, on lowest setting on voip phone as well....?" <- basically their caller themselves are loud. He wan't a confidential meeting and the other party is blasting. This guy is pretty much a confirmed twit. |
| 15:34.44 | [TK]D-Fender | You want a small quiet conference then you don't need a boomerang conference phone. |
| 15:34.48 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:35.00 | [TK]D-Fender | Wrong tool being USED by a tool. |
| 15:36.19 | *** join/#asterisk rambusac (~rambusac@chello062178212009.4.15.vie.surfer.at) |
| 15:37.19 | Katty | fender. dear. |
| 15:37.21 | Katty | be nice. |
| 15:37.27 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:37.27 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:42.17 | beardy | Tool ate. |
| 15:44.11 | Katty | hi beardy |
| 15:44.39 | beardy | Hello Katty |
| 15:45.30 | Katty | we should totes hug. |
| 15:45.31 | [TK]D-Fender | high-5's beardy |
| 15:45.45 | [TK]D-Fender | beardy, Worthy pun |
| 15:46.23 | beardy | high-5's [TK]D-Fender and hugs Katty |
| 15:47.02 | Katty | hugs beardy |
| 15:47.29 | Katty | [TK]D-Fender: come back after you've had another cup of coffee. |
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| 15:52.12 | navaismo | Hi anyone here using webrtc with sipml5 & webrtc2sip as media gateway? I have a one way audio issue on localnet. |
| 15:59.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:01.25 | Greenlight | Is there any way at present in Asterisk 11 to use the MixMonitorID's from the AMI ? |
| 16:01.46 | Greenlight | I can't see a way to "get" or "set" them... |
| 16:02.20 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:02.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:04.42 | jeev | is it extremely frowned upon to get a second hand PRI card ? |
| 16:08.18 | *** join/#asterisk serafie (~erin@nat/digium/x-rzipwugnsgkvjfeh) |
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| 16:13.26 | SuperNull | jeev i would rather have second hand PRI than second hang analog any day. |
| 16:13.39 | igcewieling | jeev: 1) second hand cards seem to cost almost much as a new card. |
| 16:13.52 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 16:14.11 | SuperNull | pri has dropped a lot its not like $1500 for a 1 port card anymore .. right? |
| 16:14.45 | [TK]D-Fender | It was never anywhere near there. |
| 16:14.56 | [TK]D-Fender | not in the past decade. |
| 16:15.34 | SuperNull | lies. |
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| 17:18.34 | navaismo | never mind stupid headset or stupid me |
| 17:18.44 | navaismo | now time to see if vp8 works |
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| 19:30.53 | WIMPy | Why does Asterisk negotiate SRTP even if it's not supported? or what does "No SRTP module loaded, can't setup SRTP session." mean? |
| 19:31.02 | WIMPy | It certainly means no audio. |
| 19:31.22 | iwanttobefreak | Hello, I have plug a PCI GSM card, how can I configure with dahdi? |
| 19:31.45 | WIMPy | RTFM |
| 19:32.03 | WIMPy | Does it work with dahdi at all? |
| 19:32.44 | iwanttobefreak | yes, I have make the trunnk and route like my FXO but doesn't work |
| 19:33.57 | navaismo | iwanttobefreak, its very possible that you need the drivers from your vendor |
| 19:34.43 | WIMPy | Most probably |
| 19:34.53 | WIMPy | Does it show up as dahdi device, yet? |
| 19:36.37 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:37.50 | iwanttobefreak | i think no... |
| 19:38.24 | iwanttobefreak | dahdi show channels |
| 19:38.24 | iwanttobefreak | <PROTECTED> |
| 19:38.24 | iwanttobefreak | <PROTECTED> |
| 19:38.56 | ChannelZ | dahdi_hardware in a shell (assuming you have dahdi-tools installed) |
| 19:39.02 | WIMPy | dahdi_scan, dahdi_hardware |
| 19:44.38 | iwanttobefreak | umhhh this is another server, maybe not have installed dahdi-tools |
| 19:44.57 | *** part/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
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| 19:48.29 | ChannelZ | well if there is no DAHDI driver that can see the card you're hosed so you need to start there |
| 19:49.25 | iwanttobefreak | I have dahdi-tools installed but not command dahdi |
| 19:49.26 | WIMPy | "think", "maybe"? That is all very vague. |
| 19:49.28 | iwanttobefreak | # dahdi_scan |
| 19:49.29 | iwanttobefreak | [1] |
| 19:49.29 | iwanttobefreak | active=yes |
| 19:49.29 | iwanttobefreak | alarms=UNCONFIGURED |
| 19:49.29 | iwanttobefreak | description=DAHDI_DUMMY/1 (source: Linux26) 1 |
| 19:49.33 | iwanttobefreak | name=DAHDI_DUMMY/1 |
| 19:49.35 | iwanttobefreak | manufacturer= |
| 19:49.37 | iwanttobefreak | devicetype=DAHDI Dummy Timing |
| 19:49.39 | iwanttobefreak | location= |
| 19:49.41 | iwanttobefreak | basechan=1 |
| 19:49.43 | iwanttobefreak | totchans=0 |
| 19:49.45 | iwanttobefreak | irq=0 |
| 19:49.46 | WIMPy | ~pb |
| 19:49.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:50.47 | iwanttobefreak | sorry :( |
| 19:51.48 | WIMPy | And dahdi_hardware? |
| 19:52.24 | iwanttobefreak | pci:0000:00:0a.0 wctdm- e159:0001 Wildcard TDM400P REV E/F |
| 19:52.49 | WIMPy | Which is another card you have? |
| 19:52.59 | iwanttobefreak | lspci: 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 |
| 19:53.09 | iwanttobefreak | is in another server I remember now |
| 19:53.53 | iwanttobefreak | And I have the anothe server power off now |
| 19:53.55 | WIMPy | What's that about the other server? |
| 19:54.14 | WIMPy | And that does not look like a gsm card. |
| 19:56.41 | *** join/#asterisk BriGuy (~BriGuy@74.115.41.6) |
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| 20:01.21 | *** join/#asterisk kilgorex (~rob@82-69-63-23.dsl.in-addr.zen.co.uk) |
| 20:02.35 | igcewieling | no, it looks like an X100P |
| 20:03.13 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 20:04.54 | navaismo | which softphone for linux with video support do you recommend, aparto of linphone ? |
| 20:10.49 | navaismo | ¬¬ |
| 20:12.06 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:12.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 20:12.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 20:17.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:19.49 | leifmadsen | I like jitsi |
| 20:20.54 | *** join/#asterisk ghghz (~ton@kluonis.kvb.lt) |
| 20:21.43 | ghghz | Hello. Is it possible to use call files and have failover? I mean I do Channel: SIP/trunk/number and want to know DialStatus. If Status unavailable, then Dial via another trunk. |
| 20:26.08 | [TK]D-Fender | ghghz, No. Dial a local channel so you can do it in the dialplan directly |
| 20:26.36 | ghghz | Channel: Local/something ? |
| 20:29.47 | ghghz | [TK]D-Fender: http://p.defau.lt/?Vny3IT_uDuNnUBa75K7KSQ |
| 20:29.51 | ghghz | I tried liek this |
| 20:29.58 | ghghz | but I don't get NoOp |
| 20:30.57 | igcewieling | ghghz: generally you do not want both legs of the call going to the same extension/context |
| 20:33.32 | ChannelZ | Maybe he has no one else to talk to. |
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| 20:43.46 | ghghz | tried like this |
| 20:43.48 | ghghz | Channel: Local/666 |
| 20:43.48 | ghghz | Context: default |
| 20:43.48 | ghghz | Application: PlayBack |
| 20:43.48 | ghghz | Data: Radistai |
| 20:43.49 | ghghz | Archive: yes |
| 20:43.52 | ghghz | MaxRetries: 0 |
| 20:43.55 | ghghz | WaitTime: 20 |
| 20:43.57 | ghghz | nothing happens |
| 20:47.14 | ghghz | sorry, it's ok. |
| 20:47.41 | *** join/#asterisk niluje (~niluje@bdv75-4-82-227-67-242.fbx.proxad.net) |
| 20:48.13 | igcewieling | ghghz: correct. The app won't run until the CHANNEL ANSWERS |
| 20:48.47 | igcewieling | so put in an answer in your 666 dialplan |
| 20:49.09 | [TK]D-Fender | igcewieling, Ummm.. NO. |
| 20:49.15 | [TK]D-Fender | Definitely do not do that. |
| 20:49.18 | igcewieling | [TK]D-Fender: hmm? |
| 20:49.28 | ghghz | it answers automaticaly |
| 20:49.37 | igcewieling | I've never gotten the near leg to work until the far leg answers |
| 20:49.47 | [TK]D-Fender | He wants to do the dial in there and THEN connect to the other spot in the dialplan. |
| 20:50.00 | ghghz | yes |
| 20:50.03 | igcewieling | [TK]D-Fender: right, but the last thing he posted just had a noop in it for the dialplan |
| 20:50.03 | [TK]D-Fender | If you answer then the other end starts processing before the call is answereed. |
| 20:50.19 | [TK]D-Fender | igcewieling, Yeah, well that's just a starting test to look at the idea I presented |
| 20:50.24 | igcewieling | ghghz: are you using analog ports to dial out of or T-1/PRI/SIP? |
| 20:50.31 | [TK]D-Fender | igcewieling, It was never meant to be a full real attempt |
| 20:50.34 | ghghz | SIP |
| 20:51.20 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 20:52.54 | navaismo | leifmadsen, I can't make it work on Fedora 17, the rpm freeze the PC, the Bin installation can't make a call. |
| 20:55.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kkqnbwttgedzpiqn) |
| 20:55.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:06.23 | *** join/#asterisk asteriskmonkey (~philip@206.51.27.151) |
| 21:06.36 | asteriskmonkey | I seem to get alot of warnign on my cli in asterisk 10 about db.c:329 ast_db_put: Couldn't execute statment: SQL logic error or missing database |
| 21:06.50 | asteriskmonkey | how can I trace whats causing that |
| 21:13.56 | [TK]D-Fender | missing astdb file and possibly permissions error in trying to recreate it |
| 21:17.17 | *** join/#asterisk lukejt (95f1c7bd@gateway/web/freenode/ip.149.241.199.189) |
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| 21:22.14 | leifmadsen | navaismo: sorry, not sure what you're talking about |
| 21:22.23 | leifmadsen | oh, jitsi |
| 21:22.27 | navaismo | yep |
| 21:22.28 | leifmadsen | sorry, not sure |
| 21:22.58 | navaismo | np |
| 21:23.42 | lukejt | I'm creating realtime a call rating script using the AGI, trying to match a number e.g. 442034567890 to a destination in a sql database - in this case '442'. At the moment I'm doing a recursive search on the database removing a character from the end until I get a match. |
| 21:23.56 | lukejt | Can anyone suggest a more efficient way? |
| 21:25.52 | Katty | if i have one more efficient ocd moment today |
| 21:25.53 | Katty | i'm gonna snap |
| 21:26.05 | Katty | i swear i will put post it notes all over the office! |
| 21:26.24 | Katty | demands to be lazy bum rest of the day |
| 21:32.56 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
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| 21:35.10 | danfromuk | Hi, we've got a charity running a family safety information line, receiving around 150 concurrent calls. They have an IVR with 8 choices. Is it possible to add a custom field to the CDR containing all the choices a caller selected so they can see which features are more popular? Like this cdr(userfield) = cdr(userfield) + item selected |
| 21:37.58 | navaismo | your userfield is already in use? |
| 21:39.39 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:40.31 | navaismo | SET(CDR(userfield)=${EXTEN}) before jumping to the selected choice |
| 21:41.38 | danfromuk | Thanks. I'll try it out. I'm not using it yet. |
| 21:43.54 | *** part/#asterisk asteriskmonkey (~philip@206.51.27.151) |
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| 22:00.24 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 22:01.03 | killown | I have a Wildcard X100P Board and I'd to know how to check if my analog line is connected? |
| 22:01.35 | killown | http://bpaste.net/show/QBcVCn9F7zIsBGoWKK7n/ |
| 22:03.41 | Tim_Toady | killown: 'dahdi show status' will be 'RED' if its not connected |
| 22:04.57 | killown | Tim_Toady, Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) and the line is disconnected |
| 22:05.46 | Tim_Toady | it usually returns 'RED' instead of 'OK' when it is disconetced |
| 22:06.19 | killown | there is no line and still returns ok :/ |
| 22:06.57 | killown | ok I will reboot the pabx |
| 22:10.39 | killown | Tim_Toady, http://bpaste.net/show/P4seFe1QzQgH89PhY4lB/ no analog line is connected |
| 22:10.43 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 22:10.54 | killown | I rebooted the system and even so giving ok |
| 22:11.34 | Tim_Toady | when you say not connected you mean...? no calble connected? or you just cant call |
| 22:12.11 | killown | Tim_Toady, sorry, I mean no cable connected |
| 22:16.07 | igcewieling | killown: what brand of card? |
| 22:16.51 | killown | igcewieling, this one http://www.x100p.com/products/FXO.php |
| 22:17.06 | leifmadsen | pukes a little in his mouth |
| 22:17.42 | igcewieling | killown: if went searching for the worst card to use with Asterisk, your card would be the one you would use. |
| 22:18.02 | killown | igcewieling, why? |
| 22:18.14 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 22:18.28 | igcewieling | killown: they don't work very well. I suspect they simply don't generate a red alarm. |
| 22:18.40 | *** part/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 22:19.39 | danfromuk | Hi, what does Call to timerfd_gettime() error mean? |
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| 23:04.36 | *** join/#asterisk Smak (~adam@c-98-226-137-241.hsd1.in.comcast.net) |
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| 23:23.32 | iwanttobefreak | WIMPy, ChannelZ Thank you for all. I blacklist driver wctdm and all ok |
| 23:34.00 | iwanttobefreak | Can I make a test call from CLI to a external phone with FXO o GSM card? |
| 23:35.59 | *** join/#asterisk s14ck (~s14ck@190.203.177.134) |
| 23:36.09 | danfromuk | Currently, anyone can send an call to the incoming call context and as long as the extension is correct, the call is accepted. |
| 23:36.32 | danfromuk | Is it possible to check against a list of ip addresses to ensure incoming calls are only from the suppliers that we deal with? |
| 23:37.22 | s14ck | Yes, this is a fact. |
| 23:39.14 | Smak | I would guess I am not the first person to ask opinions on cheapish trunking service providers. Rates seem to vary greatly and at this point I am just playing around with asterisk and am no looking to invest a lot into experimenting, nor do I expect it to be free. So any recommendations besides google talk? |
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