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| 01:11.26 | Katty | rawr |
| 01:15.06 | igcewieling | hey Katty |
| 01:15.15 | Katty | herro |
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| 01:55.34 | TriJetScud | throws Katty in the water |
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| 02:07.18 | phix | werd |
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| 02:42.10 | thx2000 | Anyone know why line-in streaming moh would be choppy using arecord? Works ok streaming shoutcast w/ mpg123. |
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| 05:32.34 | iprouteth0 | would anyone have an idea why I don't hear ringing for outbound calls? |
| 05:32.57 | iprouteth0 | same condition for both IAX and SIP calls |
| 05:34.33 | [TK]D-Fender | lack of progress from provider. |
| 05:34.49 | iprouteth0 | using google voice, asterisk 11 |
| 05:34.54 | [TK]D-Fender | Lack of indications.conf for Answered inband. |
| 05:35.02 | [TK]D-Fender | Go look at the actual call. |
| 05:35.09 | [TK]D-Fender | and GV is neither SIP nor IAX2 |
| 05:35.39 | iprouteth0 | not sure I've ever opened indications.conf |
| 05:36.03 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 05:36.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 05:36.14 | [TK]D-Fender | eek |
| 05:36.29 | iprouteth0 | it is present |
| 05:36.44 | iprouteth0 | Never had this trouble with asterisk 1.8' |
| 05:38.02 | carrar | eek |
| 05:39.19 | [TK]D-Fender | iprouteth0: If you want advice you need to show the call |
| 05:44.18 | iprouteth0 | http://pastebin.com/AayWFfPD |
| 05:46.53 | [TK]D-Fender | Motif is not indicating ringing |
| 05:47.22 | [TK]D-Fender | Eitehr they don't support it, or where it got routed didn't support it in the chain,et c which can happen depending where you;re calling |
| 05:47.50 | iprouteth0 | and IAX |
| 05:47.52 | iprouteth0 | http://pastebin.com/kj6YWrCz |
| 05:50.22 | [TK]D-Fender | iprouteth0: You don't seem to be getting it |
| 05:50.35 | [TK]D-Fender | iprouteth0: GOOGLE VOICE is not passing back a RINGING STATUS |
| 05:50.43 | [TK]D-Fender | What YOU call with isn't the issue |
| 05:51.16 | [TK]D-Fender | You can force ringing to your phone with "" if you want the confort of hearing something... |
| 05:51.19 | [TK]D-Fender | "r" |
| 05:52.03 | [TK]D-Fender | but that may be misleading if they his delays in routing or end up leading o some sort of other message |
| 05:53.11 | carrar | Using google is always misleading :) |
| 05:53.23 | carrar | Oh woo, we have reader |
| 05:53.26 | carrar | oh no |
| 05:58.14 | iprouteth0 | same account on an Asterisk 1.8 box |
| 05:58.15 | iprouteth0 | http://pastebin.com/AGEnaj6N |
| 05:59.18 | [TK]D-Fender | iprouteth0: And chan_gtalk is not chan_motif |
| 05:59.28 | [TK]D-Fender | Do not compare them |
| 05:59.46 | [TK]D-Fender | Could be that Motif lacks call progress indications entirely |
| 06:00.06 | [TK]D-Fender | But don't assume any similarity jus because "Google" is the back-end |
| 06:01.04 | [TK]D-Fender | and ... it's way past bed-time... |
| 06:01.09 | [TK]D-Fender | heads off |
| 06:01.14 | [TK]D-Fender | night all... |
| 06:01.50 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:02.45 | iprouteth0 | I think it has to do with changes in motif and I'm thinking it's just a configuration issue. Anyone running asterisk 11 with google voice? |
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| 06:07.15 | iprouteth0 | 'nailed it! |
| 06:07.33 | iprouteth0 | I forgot to put in the new directive "transport=" |
| 06:07.37 | iprouteth0 | used google-v1 |
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| 06:09.33 | iprouteth0 | in motif.conf |
| 06:11.37 | deo | hello guys.. is there a particular command in asterisk cli to show connected PSTN on the TDM card?? |
| 06:13.05 | deo | i mean like to show the list of connected lines in a TDM800P card, is there any ? |
| 06:14.11 | iprouteth0 | pri or fxs? |
| 06:14.41 | deo | fxs iprouteth0 |
| 06:16.54 | jkroon | hi, would someone be able to assist me with access to the reviewboard? |
| 06:17.23 | jkroon | for some reason my login from issues.asterisk.org doesn't pull through |
| 06:17.44 | iprouteth0 | zap show channels or zap show status perhaps |
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| 06:18.32 | iprouteth0 | never used fxs or pri with asterisk myself |
| 06:18.58 | jkroon | dahdi show channels :) |
| 06:20.05 | deo | jkroon: iprouteth0 dahdi show channels it is.. |
| 06:20.07 | jkroon | in the case of fxs it will not give you much details other than it's configured, you can however do dahdi show channel X which will give you more details. |
| 06:20.23 | jkroon | for fxo channels it will show alarms if there is not battery on the line |
| 06:22.10 | deo | thanks jkroon i though it will give what PSTN number is connected on that X |
| 06:23.50 | iprouteth0 | I've never had the $ for dadhi hardware....Always worked with sip, google voice, and a bit of IAX |
| 06:24.29 | deo | iprouteth0: you have servers in different countries? |
| 06:25.09 | iprouteth0 | only US with a few different geographic locations |
| 06:25.15 | deo | i have in mind that if ive put a asterisk server in USA and another asterisk server in beijing.. |
| 06:25.32 | deo | would my calls from US to beijing be free?? |
| 06:25.42 | iprouteth0 | trunk them together with no provider |
| 06:25.43 | jkroon | deo, no, but if you really need to you can do some really cool stuff with dialing and CLI |
| 06:26.01 | deo | i mean i route my calls from asterisk-USA to asterisk-Beijing.. iprouteth0 |
| 06:26.09 | deo | yeah no provider used iprouteth0 |
| 06:26.20 | deo | jkroon: how ??? |
| 06:26.45 | jkroon | for example, i've got a feature on my PBX systems where it will iterate through all FXO channels, dial one of my fixed numbers over the PSTN, wihch will then play back the receive CLI using DTMF, which is then read by the caller and stored in the astdb :) |
| 06:26.56 | jkroon | using call origination. |
| 06:27.08 | deo | ohhhh that was cool jkroon |
| 06:27.49 | iprouteth0 | yeah he does good work. Still doing gentoo voip dev? |
| 06:28.00 | jkroon | jip |
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| 06:28.52 | jkroon | ok well, good work or not probably depends on who you ask (thanks anyway), either which way, i'm off ... need to go climb a mast |
| 06:28.56 | deo | so it will be like this iprouteth0 : me in USA connected to my asterisk-US will call my asterisk-Beijing via iax2 and then from asterisk-Beijing ill call local numbers in Beijing.. so my call from USA to Beijing will be free??? is that right jkroon iprouteth0 ??? |
| 06:30.30 | iprouteth0 | never done it like that though I suppose it's possible |
| 06:32.04 | bulkorok | hi |
| 06:32.43 | bulkorok | deo: this is possible |
| 06:33.07 | iprouteth0 | might be able to use DISA somehow and make it only available to internal extensions |
| 06:33.10 | deo | bulkorok: possible that it will be free? :0 |
| 06:33.43 | deo | asterisk-US and asterisk-Beijing will be connected through iax.. and no provider used |
| 06:33.44 | bulkorok | deo: depends on your rates in bejing, but the IAX calls from *US to *CN are free(TM) |
| 06:34.11 | deo | yeah that was i meant by the free bulkorok |
| 06:34.20 | bulkorok | well, as free, as your internet-line costs are :-) |
| 06:34.30 | deo | not from beijing to local numbers there hehe |
| 06:34.36 | bulkorok | sure :-) |
| 06:35.01 | deo | im planning to setup asterisk in US and another in Beijing lol hehe |
| 06:35.09 | deo | thanks bulkorok iprouteth0 |
| 06:35.28 | bulkorok | great.... but be aware that you need a RTT less then 300ms |
| 06:36.52 | deo | for what purpose bulkorok ? |
| 06:37.56 | bulkorok | image a call that needs about 500ms when the callers hear each other... that's not that comfortable |
| 06:50.36 | *** join/#asterisk rasheed (~rasheed@115.109.4.63) |
| 06:50.41 | rasheed | hello |
| 06:51.45 | rasheed | with queue i noticed that one of the sip channel was hung in inuse status.. |
| 06:52.10 | rasheed | finally i cleared it using the command channel request hangup |
| 06:52.25 | rasheed | is there any measures which i can take , to avoid this situation in future |
| 06:53.06 | rasheed | it will be of a great help if any one can help me on this.. as we are runnig a critical call centre.. |
| 06:53.30 | rasheed | do you think that rtp time out wil help? |
| 07:00.50 | rasheed | i would really appreciate any help on this regard.. please |
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| 07:39.51 | smirker | could somebody please clarify a quick CDR question for me. Asterisk 1.8. say an incoming call comes in at time A, i Answer() and Wait(10) seconds, then Dial() another party and they answer at time B. is the "answer" time in the CDR time A (when I Answer()ed) or time B (when two parties were bridged) ? |
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| 07:59.22 | andross | the cdr has 2 time fields |
| 07:59.29 | andross | one shows when the call comes into the server |
| 07:59.35 | andross | and the other shows when the call is answered |
| 08:00.00 | andross | so you will see two times |
| 08:00.08 | andross | maybe one is 38 seconds and the other is 30 |
| 08:00.14 | andross | call rang for 8 seconds |
| 08:00.27 | andross | was answered and then hung up 30 seconds from answer |
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| 08:13.34 | smirker | andross: when happens if I Answer() the call into the server straight away, and later on (after some prompts) Dial() a number and they answer? if "answer" the time the call in was Answer()ed, or the time the final Dial() was answered by party B? |
| 08:13.47 | smirker | s/if/is |
| 08:15.03 | smirker | wondering which has precidence |
| 08:15.05 | andross | if you are concerned with billing incoming and outgoing calls, i believe the best way to do is it have separate servers |
| 08:15.10 | andross | have one server collect all incoming |
| 08:15.14 | andross | and one collect all outgoing |
| 08:15.32 | andross | that way the cdrs dont show the time for all the internal ivr stuff |
| 08:15.57 | *** join/#asterisk proute (~proute@146.255.170.33) |
| 08:16.11 | proute | Hi everybody |
| 08:17.25 | proute | I use Asterisk 1.6.0.28 (I know, it's an old release) who works fine. But I have a problem with Meetme. I use Meetme with Dahdi (without card) on trunk SIP. Sometime during the conf, I have no sound during 10s... |
| 08:17.50 | smirker | the information i need to extract is when the incoming call was *actually* answered (200 OK), as i need this time to match CDRs across external CDRs |
| 08:17.55 | smirker | andross: ^ |
| 08:18.09 | proute | It seems that this problem come from when a user enter and leave a conf |
| 08:18.29 | proute | But I'm not sure. Do you have any idea about this problem with meetme? |
| 08:18.33 | proute | Thanks for your help |
| 08:22.08 | ectospasm | proute: Asterisk 1.6.0 is EOL (End Of Life), and has been for a very long time. You are strongly encouraged to upgrade to something current before we provide any assistance. |
| 08:22.27 | ectospasm | proute: let me guess, you're using Trixbox, in this day and age? |
| 08:22.39 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 08:22.57 | proute | ectospasm: No i don't use Trixbox. The conf files are edited manually |
| 08:24.10 | ectospasm | ah, OK. The last version I ever saw Trixbox use was 1.6.0.28, so I guessed wrong. |
| 08:25.33 | proute | No Problem, I use Asteisk 1.6.0.x Since 3 years... And In my company, it's not possible to upgrade to new release. This release works fine for me, but I have problems with meetme. |
| 08:26.02 | proute | Just a question, I use meetme with dahdi (without dahdi hadware). Is it possible to have a timing problem? |
| 08:26.57 | ectospasm | possible, yes. |
| 08:27.39 | ectospasm | I'd start planning a migration away from that version as soon as possible. You are probably full of security holes which will NOT be fixed. |
| 08:35.35 | andross | the following command should allow you to begin a new cdr before calling out from an ivr or dialing from an already initiated call http://www.voip-info.org/wiki/view/Asterisk+cmd+ResetCDR |
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| 08:42.10 | *** join/#asterisk nova34353 (bab2494d@gateway/web/freenode/ip.186.178.73.77) |
| 08:42.20 | nova34353 | Hello! |
| 08:43.01 | nova34353 | Is anyone on? |
| 08:45.17 | Faustov | no, shut the door, I want to sleep |
| 08:45.24 | nova34353 | I need help with No matching peer found for a user I know for a fact is in the DB |
| 08:46.08 | nova34353 | I just don't know what to check. Is there any way to ask asterisk to list all the peers it does know about? |
| 08:46.44 | nova34353 | Like make it query the DB and check? BTW this is Asterisk 1.8 on Ubuntu 12.04 and the sippeer was created by A2Billing |
| 08:47.22 | nova34353 | Install via apt-get from the Ubuntu repos. Not sure if that makes a difference or not. |
| 08:51.20 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 08:52.43 | nova34353 | Here is the output of core show config mappings |
| 08:52.46 | nova34353 | sip*CLI> core show config mappings Config Engine: mysql ===> extensions (db=general, table=cc_sip_buddies) ===> sippeers (db=general, table=cc_sip_buddies) ===> sipusers (db=general, table=cc_sip_buddies) ===> iaxpeers (db=general, table=cc_iax_buddies) ===> iaxusers (db=general, table=cc_iax_buddies) |
| 08:55.57 | nova34353 | There is one warning /error when I do a reload |
| 08:55.58 | nova34353 | <PROTECTED> |
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| 09:26.24 | andross | i have no idea what you are talking about nova what exactly are you trying to do |
| 09:26.25 | *** join/#asterisk deo (~deo@122.52.128.123) |
| 09:26.43 | andross | can you see the sip or iax peer avail from the cli? |
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| 10:11.20 | y4h0 | hey there |
| 10:11.40 | y4h0 | i have a gigaset a510 IP phone with 5 headsets |
| 10:12.10 | y4h0 | *handsets |
| 10:12.27 | y4h0 | randomly 2 of the handsets are busy and can't accept incoming calls |
| 10:12.54 | y4h0 | the message in asterisk console is Got SIP response 486 "Busy Here" back from ... |
| 10:13.11 | y4h0 | any help you might provide ? |
| 10:14.36 | WIMPy | Have you tried turning it off and on again? |
| 10:23.26 | y4h0 | i tried sip reload |
| 10:24.34 | WIMPy | How could that change anything on the phone? |
| 10:25.41 | y4h0 | oh the phone you mean ... yes |
| 10:26.46 | WIMPy | It's the phone that says it's busy. Nothing oyu could change baout that on Asterisk. |
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| 10:36.28 | linocisco | hi all , I followed http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/ to install asterisk on ubuntu server, I would like now to install Freepbx or AsteriskNow GUI, how to install? |
| 10:37.03 | y4h0 | i have an ip station, the station says it is busy |
| 10:37.14 | ectospasm | linocisco: don't use Ubuntu, and use a distribution that ships with FreePBX natively (such as AsteriskNOW) |
| 10:37.40 | y4h0 | the station has an ip .146 and that is reported as busy |
| 10:37.46 | linocisco | ectospasm, no. I am just familiar with ubuntu . not centOS |
| 10:37.52 | ectospasm | note that FreePBX is not supported here. |
| 10:38.17 | ectospasm | linocisco: I don't know of any FreePBX installations which work on Ubuntu |
| 10:38.26 | ectospasm | They may exist, I just know of none. |
| 10:38.38 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 10:38.44 | linocisco | ectospasm, ok thanks anyway |
| 10:43.00 | *** join/#asterisk MarKsaitis (~MarKsaiti@194.168.230.106) |
| 10:44.05 | WIMPy | y4h0: No, the station is not reported as busy, the *station* reports itself as being busy. |
| 10:45.10 | WIMPy | So that leaves no doubt as to where something can be done about that. |
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| 11:40.56 | aurs | what should I do if I have a caller in a queue, with a specified channel, and that channel does not exist on the system? (So it is not possible to soft hangup the channel) |
| 11:41.15 | aurs | the non-exising call is blocking the queue from receiving calls |
| 11:44.00 | aurs | hmm... nevermind |
| 11:45.04 | aurs | note to self: "sip show channel" != "core show channel" :) |
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| 12:21.55 | jkroon | hi all, given that some providers are totally braindead re telephone-event having to have a specific type number (96 in particular, whereas asterisk uses 101), would a patch to make this value configurable be accepted? |
| 12:22.19 | jkroon | what really bugs me - what if I have to deal with two different such braindead providers that uses different values ... |
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| 12:30.08 | pzn | I have problems with a home-made implementation of sip protocol and asterisk... mainly it "unregisters" my sip device some times per day, and in the next register, it registers again. I'd like to debug that. how can I "turn on" debugging in asterisk to know the reason of an unregister (timeout, network lag)? |
| 12:32.41 | kaldemar | pzn: expiry is the keyword. |
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| 12:34.44 | infernix | i have an android phone that registers to asterisk with iax just fine when coming in over a public IP; wireshark shows the phone sending REGREQ and asterisk replying with REGAUTH. |
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| 12:35.20 | infernix | when I put that phone in a LAN with the same asterisk server, it sends a REGREQ, and asterisk receives it (shown by iax debug on and debug level 15) but it never responds with a REGAUTH |
| 12:35.31 | infernix | anyone have the slightest idea why? |
| 12:40.31 | infernix | it will work fine when I register to the LAN IP (client 192.168.0.x, asterisk server 192.168.0.1) but not when registering to the public IP (client 192.168.0.x, asterisk server 1.2.3.4) |
| 12:45.24 | infernix | huh. fixed by binding only to the public ip, and not 0.0.0.0 |
| 12:48.45 | jkroon | rofl |
| 12:49.00 | jkroon | ok, would it make sense to make the rtp payload number for telephone-event configurable? |
| 12:49.16 | jkroon | currently it's hard coded to 101, which breaks my DTMF on some peerings ... |
| 12:49.39 | WIMPy | jkroon: I think you answered that yourself already. |
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| 13:03.56 | jkroon | WIMPy, trying to rally some support here ... |
| 13:04.14 | jkroon | currently i have to custom patch those servers ... which is annoying to say the least |
| 13:05.22 | WIMPy | You don't need to tell me. Organizing patches can become a real struggle. Even with git and we don;t even have that here. |
| 13:06.36 | jkroon | hehe, ok, so file a bug, submit a patch and wait... |
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| 13:11.12 | coppice | its a dynamic payload type. how can 101 be wrong? |
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| 13:34.43 | vixmemon | i need help in asterisk configuration... i m badly stuck. |
| 13:34.55 | vixmemon | can someone help me here? |
| 13:35.57 | vixmemon | @pabelanger ... i need help |
| 13:36.12 | bulkorok | ~ask |
| 13:36.12 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:36.28 | pabelanger | vixmemon: what bulkorok said |
| 13:36.45 | vixmemon | I am using TrixBox on my local network. i can connect and register softphones through defined extensions. |
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| 13:37.22 | WIMPy | ~trixbox |
| 13:37.22 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 13:37.33 | vixmemon | I am trying to access trixbox from outside my local network. I have tried to bind a live domain with the ip of the server on 5060 but i can not still register softphones. |
| 13:38.17 | pzn | kaldemar, sorry... I searched for "expiry" but couldn't find what I expect... I want to see some debug about the unregister, if it was "disconnected by qualify timeout" or if it was disconnected by "timeout of the register command" |
| 13:38.24 | WIMPy | vixmemon: We cannot support GUIs here. You need to ask in the support channels of the vendor. |
| 13:38.34 | vixmemon | are there any configurations i need to do in TrixBox to be able to make it accessible from outside my network. |
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| 13:43.11 | vixmemon | Isnt it Asterisk related question.. do i need any configurations in Asterisk to be able to access it (register softphones) from outside my local network? |
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| 13:43.44 | WIMPy | Your Asterisk is configured by Trixbox, not by you. |
| 13:43.53 | WIMPy | If you do it yourself you might even break it. |
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| 13:51.30 | protocoldoug | we have customers complaining about some strange phone calls |
| 13:52.09 | chuckf_ | protocoldoug: I find all phone calls strange |
| 13:52.12 | protocoldoug | seems these callers set a random originating ANI, and like to call VM on a customer side, and then they play some DTMF digits -very- slowly |
| 13:52.14 | protocoldoug | hahahah, as do i |
| 13:52.34 | protocoldoug | then... after a little while, sometimes..... they'll start playing some Asian sounding music |
| 13:53.00 | kaldemar | pzn: "disconnected by qualify timeout" does not make sense. asterisk does not unregister a peer even if the qualify functionality judges it unreachable. "timeout by register command" must be something in your client, sip debug is all you have on the asterisk side. |
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| 13:53.20 | WIMPy | Wait for the next bill. Maybe they have some million minutes to asia. |
| 13:53.45 | chuckf_ | protocoldoug: sounds like someone trying to hack your phone system |
| 13:54.34 | WIMPy | Maybe it would be better not to wait for the bill. Better check what else might be going on there. |
| 13:54.50 | protocoldoug | i'm sure it's a phreaking operation of some type, but... it's happening after I tandem a call, it's happening on the bridge, not a lot they can do. and luckily there's no int'l trunks outbound for me, can't even make one of those calls, just 1+ dialing |
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| 14:09.09 | pzn | kaldemar, there is a "qualify package that asterisk is sending (sip option package)", if I don't reply to it then asterisk unregisters me because of this. that is what I mean by "qualify timeout". other thing: when I register, in the (sip register package) there is a field "expires" which a time in seconds... |
| 14:09.47 | pzn | kaldemar, thanks, "sip debug" is showing me important information to solve this issue |
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| 14:15.45 | bightf_ | Hello, I just upgraded from Asterisk 1.8.17.0 to 11.3.0 in order to correct a bug with pbx_lua, which was continuing execution of lua dialplan after a gosub was called. Now that this bug is resolved, when I call a gosub from my extensions function in the lua dialplan, this function stops executing after the gosub has returned, but I want to continue execution of this same function ! So, does anybody got an idea ? |
| 14:16.13 | kaldemar | pzn: yes, the expiry is in the sip message. that's why i said expiry earlier, it is configurable in asterisk. see expiry related options in sip.conf.sample. |
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| 14:18.29 | pgrace | Hi there, I'm having a hard time getting an AudioCodes MediaPack FXS gateway to register to asterisk. I've got a sip debug (https://gist.github.com/PeterGrace/e2fc0b8cab1726cee0ed) that I've been staring at and I just can't seem to figure out why it's failing. Asterisk usually alerts on bad password for a user, right? Can someone take a look and just sanity check me, I've been staring at it too long. |
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| 14:26.14 | pzn | kaldemar, looking at network package debug, I noticed that asterisk sends one "sip options" package per second if client is down. how can I configure this "retry" to be 5 seconds, not 1 second |
| 14:26.16 | pzn | ? |
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| 14:31.06 | igcewieling | pzn: there is a qualifyfreq= option, check the sip.conf.sample included in your asterisk source code |
| 14:32.11 | pzn | igcewieling, I found this option, it sets "how frequently send OPTIONS", but when options fail, it resends every 1 second (i didn't find config to increase this 1 second time) |
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| 14:32.36 | igcewieling | pzn: you may either have to put up with it or change it in the source code. |
| 14:32.47 | igcewieling | or turn off qualify entirely |
| 14:33.55 | pzn | igcewieling, mmm... lets test turning it off |
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| 15:06.36 | WIMPy | This is what happens whe O2 removes GSM equipmentt and leaves behind their garbage in form of cable ties. The doves adapted quickly: |
| 15:06.44 | WIMPy | https://lh4.googleusercontent.com/-eFo5iPSyXIQ/UVsVA4NcOnI/AAAAAAAAAcY/-OoiMjHlGWY/s902/2013+-+1 |
| 15:10.41 | linuxgeek | lol |
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| 15:12.59 | Onyx47 | hello, quick question, wiki for Asterisk 11 lists AMI events such as Pickup and QueueCallerAbandon but I'm not receiving them when I connect to AMI on my install... do I need to do anything special to enable them? |
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| 15:52.55 | bightf_ | Hello, I just upgraded from Asterisk 1.8.17.0 to 11.3.0 in order to correct a bug with pbx_lua, which was continuing execution of lua dialplan after a gosub was called. Now that this bug is resolved, when I call a gosub from my extensions function in the lua dialplan, this function stops executing after the gosub has returned, but I want to continue execution of this same function ! So, does anybody got an idea ? |
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| 16:06.10 | igcewieling | bightf_: you might want to ask on the asterisk-users mailing list. I've not seen LUA questions on this channel in years |
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| 16:14.41 | leifmadsen | people still use lua? |
| 16:14.46 | leifmadsen | s/still// |
| 16:19.36 | igcewieling | checks his pocket watch and adjusts his bow tie. |
| 16:22.13 | sruffell | looks like your monocle has a smudge on it. |
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| 16:37.49 | bightf_ | Yes some of us still use lua and ok, I will try on the users mailing liist, thx. |
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| 16:45.07 | drmessano | Why u mad, bro? |
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| 17:13.02 | qakhan | Hi all, what is data1, data2, data3, data4, rowcount and lastpointer meaning in queue_log |
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| 17:23.02 | qakhan | data1 shows holdtime, data2 shows talk time/CallerID, |
| 17:23.35 | qakhan | i didnt get what remaining fields shows, does anyone know? |
| 17:24.12 | [TK]D-Fender | qakhan, there is DOC in your tarball for this |
| 17:29.43 | qakhan | [TK]D-Fender what is tarball? |
| 17:36.02 | Joel | qakhan: don't mind [TK]D-Fender he's always sucked at helping |
| 17:36.04 | Joel | qakhan: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#ACD_id289031 |
| 17:36.22 | Joel | qakhan: check out completeagent for example |
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| 17:38.17 | Joel | GameGamer43: My uncle now lives in Oshkosh, prepare to lose to me this summer. |
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| 17:40.55 | qakhan | Thanks Joel. please clean my confussion |
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| 17:42.40 | qakhan | data1, data2, data3, data4 etc.. are related to events |
| 17:42.55 | qakhan | like COMPLETECALLER(holdtime|calltime|origposition) |
| 17:43.23 | Joel | yes. |
| 17:44.09 | qakhan | Thanks |
| 17:45.20 | qakhan | as per [TK]D-Fender concern i dont mind him/her. he/she is good person. |
| 17:46.12 | Joel | He's clearly been burned out on helping people for years. |
| 17:46.31 | Joel | needs a hug, a puppy, or both. |
| 17:46.56 | qakhan | always help to others, and i like his behavior to point the solution and ask us go and get it |
| 17:47.22 | qakhan | through this we learn new things and have good experience :) |
| 17:47.39 | qakhan | LOVE You!!! [TK]D-Fender :) |
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| 18:12.57 | citywok | he was banned from here for a while for good reason :) |
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| 18:21.20 | [TK]D-Fender | Joel, I told him it was in a doc in the tarball.. That is all that I said. Nothing more. Your comment is unwarranted |
| 18:22.03 | Joel | [TK]D-Fender: want a hug? |
| 18:22.18 | [TK]D-Fender | Joel, No need of one right now, thanks. |
| 18:23.04 | [TK]D-Fender | <citywok> he was banned from here for a while for good reason :) <- feel free to follow the discussion that went on and following that day. Because the specifics and other related events probably won't match up with the view-point you may have of the people or events involved. |
| 18:23.34 | [TK]D-Fender | Joel, qakhan : http://svnview.digium.com/svn/asterisk/branches/1.4/doc/queuelog.txt?revision=58955&view=markup |
| 18:23.37 | fkurkowski | Is this an issue on a TDM410P: codec_dahdi.c: Failed to open /dev/dahdi/transcode? Or maybe is it only used by newer cards? |
| 18:23.52 | [TK]D-Fender | Linked right off the SVN which is in the tarball... there is the DOC folder as advertised. |
| 18:23.57 | [TK]D-Fender | I did say it was right there. |
| 18:24.49 | Joel | [TK]D-Fender: sorry, are you going on about that thing I answers quite some time ago? |
| 18:24.57 | Joel | s/answers/answered/ |
| 18:25.59 | [TK]D-Fender | Joel, I was validating that what I told him was a direct answer for all of those values and not some wild goose chase. |
| 18:26.30 | [TK]D-Fender | Joel, so no, this was not "sucking at helping" |
| 18:26.39 | Joel | [TK]D-Fender: sorry, are you going on about that thing I answerd quite some time ago? |
| 18:26.58 | Joel | Did I miss that link the first time around? |
| 18:27.06 | [TK]D-Fender | Joel, No. This is shooting down your attempt at a cheap shot at me. |
| 18:27.26 | Joel | I must have missed that link the first time around. |
| 18:28.00 | [TK]D-Fender | Just because I didn't provide a link vs telling him its in the doc folder does not make the answer any less direct or valid for where the list was |
| 18:29.29 | Joel | you don't think it makes it less direct of an answer? :3 |
| 18:29.56 | [TK]D-Fender | Stop being a troll. |
| 18:30.09 | Joel | You can always ignore me, if you feel I'm trolling. |
| 18:30.24 | elguero | fkurkowski: http://lists.digium.com/pipermail/asterisk-users/2011-April/260909.html |
| 18:31.43 | fkurkowski | yeah I also found this but was not sure if it would also apply to other cards |
| 18:31.50 | [TK]D-Fender | Joel, I tell him its in the doc folder in the source tarball. You just fire off a "he sucks at helping." You are being a troll and an asshole. |
| 18:32.00 | Joel | You can always ignore me, if you feel I'm trolling. |
| 18:32.15 | [TK]D-Fender | Yeah, I see what kind of person I'm dealing with here... |
| 18:32.32 | fkurkowski | I hard 'newer ones' actually need it, but newer is quite ambiguous concept |
| 18:32.50 | fkurkowski | s/hard/heard |
| 18:32.59 | fkurkowski | oh crap |
| 18:34.00 | Joel | [TK]D-Fender: a sensible person who's aware of how to help people and use irc clients? |
| 18:34.33 | qakhan | [TK]D-Fender did thank you |
| 18:34.54 | qakhan | i got your point but i take some time to understand you |
| 18:38.49 | edong23 | well, i have personally witnessed [TK]D-Fender being very helpful for many. i havent seen anything that was uncalled for. Occasionally after being told several times, i have seen [TK]D-Fender a little annoyed |
| 18:38.53 | edong23 | but we all where |
| 18:39.11 | edong23 | anyway, i agree joel is trolling |
| 18:39.20 | Joel | You can always ignore me, if you feel I'm trolling. |
| 18:41.35 | edong23 | this seems to be an isolated event... i just feel like that was uncalled for. thats all |
| 18:41.38 | edong23 | we are all adults |
| 18:41.40 | edong23 | maybe... |
| 18:42.02 | Joel | ^ |
| 18:43.48 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 18:45.25 | newtonr | Joel: I think the act of trying to help is what is most important. It's doing something like acknowledging that someone is trying to help and then saying they "suck at helping" which is annoying. That is neither helpful nor appreciated by anyone. If you want to criticize the *how* of someone helping, you can do that in private. |
| 18:46.59 | chuckf_ | Joel: being told to look in the tarball is a perfectly valid, helpful answer. That qakhan doesn't know what a tarball is is not the helper's fault. |
| 18:47.14 | chris_n | is there a way to coerce the cdr module to log the cid after the privacy app has set it? |
| 18:48.42 | Joel | You can always ignore me, if you feel I'm trolling. |
| 18:50.22 | chuckf_ | Joel: I don't think you're trolling, I know you're wrong about your opinion on this matter. |
| 18:50.32 | Joel | You can always ignore me, if you feel I'm trolling. |
| 18:54.26 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:54.26 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:03.48 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 19:09.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:09.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:14.37 | *** join/#asterisk bobb_WU (~bobb@206.74.211.18) |
| 19:14.38 | Katty | hello my asterisk does not work at all how to fix plz, srs aswers only thx |
| 19:15.04 | Joel | Katty: can haz dial tonez? |
| 19:15.13 | bobb_WU | hello all, i'm having trouble getting a phone to register to asterisk. got a doc for putting in the details of a sip user? |
| 19:15.31 | zerohalo | lol @ Katty |
| 19:15.43 | chuckf_ | Katty: turn the receiver the right way around |
| 19:17.44 | newtonr | bobb_WU: the sample sip.conf file has some examples in it. This http://shop.oreilly.com/product/9780596517342.do book has info on it and if you google you'll find a dozen tutorials on setting up SIP users in Asterisk. |
| 19:19.38 | Katty | calls chuckf_ with phone upside down |
| 19:19.38 | newtonr | bobb_WU: we accept payment in Aztec gold only |
| 19:19.42 | Katty | HERRO CAN U HER ME |
| 19:24.43 | chuckf_ | wat? |
| 19:25.03 | Katty | TURN UP UR HERING AID |
| 19:25.15 | newtonr | wat-son can you hear me? |
| 19:25.25 | chuckf_ | wat? can you turn the tv down? |
| 19:25.37 | Katty | giggle fits right out of chair |
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| 19:30.20 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 19:30.37 | jeffspeff | is there a way to abort the "core restart when convenient" command? |
| 19:31.26 | igcewieling | jeffspeff: yes |
| 19:31.33 | jeffspeff | how? |
| 19:31.40 | chuckf_ | If you're a coder and haven't watched 'wat' yet take a few minutes... http://www.youtube.com/watch?v=kXEgk1Hdze0 |
| 19:32.20 | drmessano | Too funny |
| 19:32.32 | jeffspeff | igcewieling, found it. |
| 19:32.34 | drmessano | I just had a problem with a young man hearing me |
| 19:32.38 | jeffspeff | didn't think it'd be that simple |
| 19:32.44 | drmessano | I wanted to stab him |
| 19:37.16 | igcewieling | We were having such a good day, then the ipsec tunnel to verizon went down and all hell broke loose. *sigh* |
| 19:39.42 | jeffspeff | igcewieling, do ya'll do services for verizon? our tunnel has been doing good. we do telephony engineering for VZ. |
| 19:42.35 | igcewieling | jeffspeff: we use VZB as our primary SIP upstream |
| 19:43.00 | jeffspeff | ah, got ya |
| 19:43.05 | igcewieling | I hope you don't do engineering for VZ in the north east, they REALLLY SUCK |
| 19:44.20 | jeffspeff | lol, yes, we do some work in the north east |
| 19:44.24 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 19:44.38 | jeffspeff | we're not their only egninerring and design vendor |
| 19:44.48 | igcewieling | jeffspeff: You don't happen to be involved in the group who handles IPSec tunnels for their wholesale SIP service, do you? |
| 19:45.07 | jeffspeff | no, not me |
| 19:45.19 | jeffspeff | VZ is much better than ATT though. |
| 19:45.24 | jeffspeff | everything about ATT sucks |
| 19:45.26 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:45.30 | jeffspeff | they don't even offer SIP service |
| 19:45.43 | iprouteth0 | [TK]D-Fender: Figured out my ringing issue with google voice and my sip clients. Still no ringing with IAX, but I havent really investigated that fully yet |
| 19:46.59 | igcewieling | We also use Level 3, but they are expensive. |
| 19:47.07 | iprouteth0 | I've always wanted to implement strongswan for IPsec tunnels.. I've only ever used ipsec with cisco routers |
| 19:47.20 | drmessano | iprouteth0: Are you using Asterisk 11 for GV> |
| 19:47.21 | drmessano | ? |
| 19:47.21 | iprouteth0 | Level 3 is awesome though. So much easier to deal with than a lot of other carriers |
| 19:47.26 | iprouteth0 | they are really on their game |
| 19:47.33 | igcewieling | VZB REQUIRES SIP signlaing to go over a vpn tunnel to them |
| 19:47.38 | iprouteth0 | drmessano: yes, asterisk 11 on an OpenWRT router |
| 19:47.43 | drmessano | Cool cool |
| 19:48.21 | iprouteth0 | there is a new configuration directive in motif.conf that was not present in jabber.conf |
| 19:48.27 | drmessano | Because when it comes to GV theres Asterisk 11 and not-doing-it-at-all. I don't even consider the older chan_gtalk to be an option |
| 19:48.39 | igcewieling | Level 3 is expensive for incoming compared to VZB. VZB is expensive for the circuit, but not for incoming |
| 19:48.46 | iprouteth0 | still works fine in asterisk 1.8 systems for me. |
| 19:49.02 | iprouteth0 | the directive I was missing in motif.conf was "transport=google-v1" |
| 19:49.09 | drmessano | Ah.. yep |
| 19:49.23 | iprouteth0 | I wasnt getting ringing on outbound calls without it |
| 19:50.17 | iprouteth0 | Malcolm Davenport was kind enough to update the wiki for calling with google for me |
| 19:50.21 | drmessano | Well, as I understand it, the total lack of audio with chan_gtalk was resolved prior to 11 being released, but motif is so much better AND SUPPORTED (!!!!). |
| 19:50.33 | drmessano | So long pulling hair out every time google changes something |
| 19:50.34 | iprouteth0 | I've found much of the trouble with incoming calls with asterisk and google voice has to do with the priority setting |
| 19:50.36 | iprouteth0 | More about Priorities |
| 19:50.37 | iprouteth0 | As many different connections to Google are possible simultaneously via different client mechanisms, it is important to understand the role of priorities in the routing of inbound calls. Proper usage of the priority setting can allow use of a Google account that is not otherwise entirely dedicated to voice services. |
| 19:50.37 | iprouteth0 | With priorities, the higher the setting value, the more any client using that value is preferred as a destination for inbound calls, in deference to any other client with a lower priority value. Known values of commonly used clients include the Gmail chat client, which maintains a priority of 20, and the Windows GTalk client, which uses a priority of 24. The maximum allowable value is 127. Thus, setting one's priority option for the XMPP peer |
| 19:50.39 | iprouteth0 | <PROTECTED> |
| 19:50.42 | iprouteth0 | Outbound calls are unaffected by the priority setting. |
| 19:50.55 | iprouteth0 | malcolmd: Thanks BTW! |
| 19:51.23 | drmessano | Ahh |
| 19:51.26 | drmessano | I didn't know that |
| 19:51.58 | iprouteth0 | I even found in the asterisk book thats around an indication that a lower priority numeric value meant a higher priorityu |
| 19:51.58 | drmessano | I understand XMPP priorities, but didn't know about how they affected call flow |
| 19:52.03 | malcolmd | iprouteth0: thanks to you |
| 19:52.15 | iprouteth0 | gotta go. Will sign back in when I get to work. |
| 19:52.23 | *** part/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
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| 20:06.01 | igcewieling | 70% packet loss 8-( |
| 20:08.40 | *** join/#asterisk iprouteth0 (ccf60469@gateway/web/freenode/ip.204.246.4.105) |
| 20:09.11 | iprouteth0 | malcolmd: Really the thanks should go to Bill Simon who runs a YATE google voice to sip gateway |
| 20:10.05 | iprouteth0 | https://simonics.com/2012/11/gv-gateway-registration-open-new-features/ |
| 20:10.59 | iprouteth0 | That blog post pointed my research in the correct direction, and I think the verbiage you choose for the wiki is quite excellent |
| 20:13.00 | malcolmd | credit to my high school english professor |
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| 20:31.31 | luckman212 | On an Asterisk 11 system, what's the "right" way to set the default 'verbose' level for CLI sessions to 3 instead of 0? |
| 20:31.40 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 20:31.59 | iprouteth0 | I always just connect with |
| 20:32.02 | iprouteth0 | asterisk -rvvv |
| 20:32.10 | luckman212 | I see a verbose= setting in /etc/asterisk.conf and also a 'core set verbose 3 = yes' in cli.conf -- not sure which to use (both??) |
| 20:32.34 | iprouteth0 | core set verbose X I think is the actual CLI command to adjust it |
| 20:32.53 | luckman212 | iprouteth0: yes you're right, core set verbose 3 does it, but only temporarily |
| 20:33.37 | luckman212 | trying to make 3 the default .... it's a little weird that you can 'set' the verbose level but you cant 'get' or 'show' it. |
| 20:33.58 | luckman212 | so it's hard to tell if my config changes are working, argh |
| 20:34.04 | igcewieling | which ever method you prefer |
| 20:36.34 | iprouteth0 | asterisk shows you the verbosity level whe you connect to it |
| 20:37.04 | luckman212 | iprouteth0: mine doesn't (ast 11.1.2) |
| 20:37.10 | iprouteth0 | at least I thought it used to |
| 20:37.34 | luckman212 | yep. 1.8 did |
| 20:38.58 | iprouteth0 | yep. My 1.8 does. 11 doesnt. I'm still learning the differences in 11 yet. Just installed it a few nights ago |
| 20:40.57 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:43.17 | luckman212 | seems like after reading https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 and https://issues.asterisk.org/jira/browse/ASTERISK-20281 that the correct place to set default CLI verbosity is in /etc/asterisk/cli.conf |
| 20:43.18 | LieutPants | [ASTERISK-20281] [Status: Open] "core set verbose" behaves strangely, can't alias it, cli.conf example broken - https://issues.asterisk.org/jira/browse/ASTERISK-20281 |
| 20:47.46 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 21:22.41 | _Corey_ | faxing is evil. |
| 21:22.45 | _Corey_ | (just saying) |
| 21:23.57 | leifmadsen | I <3 me some faxing goodness |
| 21:26.02 | _Corey_ | I'm seriously juggling multiple unrelated customer fax issues right now... one related to a document converter, one is definitely an e-mail issue... everything faxing touches turns brown and dies |
| 21:26.16 | _Corey_ | <PROTECTED> |
| 21:26.20 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:1ce:aee2:1dcc:52ac) |
| 21:28.29 | igcewieling | telnet |
| 21:28.44 | igcewieling | the one time I want Nugget's script to kick in.... |
| 21:29.11 | igcewieling | _Corey_: Your experience is not unusual. |
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| 22:38.01 | jkroon | newtonr, seems you don't keep all of my bugs ( https://issues.asterisk.org/jira/browse/ASTERISK-21382), yea, not a bug, but would appreciate some help. |
| 22:38.01 | LieutPants | [ASTERISK-21382] [Status: Closed] make (at least some of) the payload numbers used by dynamic payload types configurable - https://issues.asterisk.org/jira/browse/ASTERISK-21382 |
| 22:41.32 | newtonr | jkroon: It sounds like a frustrating issue :( If you need some help building a work-around I would ask on the asterisk-dev list or in #asterisk-dev |
| 22:42.32 | newtonr | jkroon: the mail list is probably better to avoid the issue of missing people when they are away from the desk or work |
| 22:42.34 | jkroon | i've got a patch that I can apply on the raw code ... but that means a custom patch set that I have to "maintain". which is something I don't like. |
| 22:42.50 | jkroon | blasts of an email, thanks. |
| 22:44.50 | newtonr | once you get the patch ready then just upload it to the issue and ping #asterisk-bugs, then someone can re-open the issue |
| 22:45.19 | newtonr | and of course throw it on reviewboard if possible |
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| 22:51.04 | jkroon | re reviewboard, i'd love to but i can't seem to log in |
| 22:52.03 | newtonr | jkroon: login to #asterisk-dev and ask mjordan - he may know if there is an issue |
| 22:52.40 | mjordan | jkroon: newtonr: I lurk here as well |
| 22:53.01 | mjordan | jkroon: e-mail me at mjordan@digium.com . I'll get an account set up for you |
| 22:53.12 | mjordan | jkroon: it isn't integrated (yet!) with Crowd, so your Jira credentials won't work |
| 22:54.00 | *** join/#asterisk dijib (~root@24-231-78-197.eastlink.ca) |
| 22:56.09 | newtonr | mjordan: i was just trying to push the conversation over into -dev, but that ended up being short :) |
| 22:58.24 | phix | Morning |
| 23:14.34 | igcewieling | jkroon: posting a bounty to asterisk-biz mailing list could get you a patch cheaply |
| 23:14.45 | igcewieling | then you can post it to the feature request in jira |
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| 23:47.52 | Katty | helllllllllllllllllllllloooooooooooooooooooooooooo asterisk. |