| 00:01.39 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 00:01.45 | *** join/#asterisk Bloudermilk (~Bloudermi@dsl081-234-075.lax1.dsl.speakeasy.net) |
| 00:02.07 | GTXComm | I think I figured it out P3. |
| 00:02.23 | Bloudermilk | What's the story behind this? https://wiki.asterisk.org/wiki/display/TOP/SIP+Stack+Research |
| 00:02.32 | Bloudermilk | And this? https://wiki.asterisk.org/wiki/display/TOP/RTP+Stack+Research |
| 00:02.34 | capitan__ | hmm... is nat only for nat? for example, if i have a multi-homed server, and i want asterisk to bind to the second ip (without a default gateway), and i'm just doing sip... can i lie to asterisk, set externip to the ip of the second device, and i'll be good to go? |
| 00:02.52 | Bloudermilk | Is Asterisk considering dropping the already-implemented SIP/RTP portions of the application? |
| 00:03.30 | serafie | Bloudermilk: that's Asterisk SCF, not Asterisk. |
| 00:03.38 | Bloudermilk | Ohhh |
| 00:05.36 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 00:10.08 | capitan__ | looked like that fixed it... phew! |
| 00:10.51 | p3nguin | capitan__: extenip is only for NAT configuration, and it has nothing to do with which IP or interface Asterisk uses to bind to. |
| 00:12.08 | capitan__ | p3nguin, hmm... thanks for the info... i tried bindaddr by itself, but i was getting calls dropping after 8 seconds or so, and tried to act quickly... |
| 00:12.20 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:12.23 | capitan__ | maybe i should mention that the first network interface is locked down to high hell... does that make a difference? |
| 00:13.30 | p3nguin | capitan__: Typically, we leave the bindaddr set to 0.0.0.0 so Asterisk will use all interfaces/addresses, then use iptables to hide the one you don't want to be used. |
| 00:15.12 | capitan__ | hmmm... i had that setup at first... then i was afraid that with sip registration, with the default gateway interface locked down, packets would smack into the firewall |
| 00:15.30 | capitan__ | maybe my restart fixed something, and it was just a coincidence? |
| 00:16.39 | capitan__ | i did notice that when i tried to restart, it said "asterisk is already running"... maybe i had two processes running at first and didn't know it, and that was the culprit... |
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| 00:32.33 | *** join/#asterisk luisfelice (~luisfelic@190.79.35.108) |
| 00:33.13 | luisfelice | Hi, I am having a static noise problem with a TDM400P card on the FXS ports, any idea? |
| 00:34.51 | *** join/#asterisk Lord_Rahl (~Lord_Ralh@c-98-243-8-24.hsd1.mi.comcast.net) |
| 00:35.37 | luisfelice | A friend told me that maybe is a problem with the power |
| 00:36.23 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 00:37.08 | luisfelice | anyone? |
| 00:37.16 | sawgood | If I have a Sangoma card installed in an Asterisk solution, how can I tell if the card is 'active' and can make calls (I ran wancfg_dahdi) already |
| 00:37.25 | sawgood | It is a PRI T1 card |
| 00:38.23 | IsUp | sawgood: do 'wanrouter hwprobe' and 'wanrouter status', it'll tell you "CONNECTED/DISCONNECTED" |
| 00:38.33 | IsUp | sawgood: you have to check signalling first |
| 00:38.54 | IsUp | sawgood: and it sounds like Sangoma related problem or support. check #sangoma or contact their tech support. they are really good. |
| 00:41.14 | sawgood | Thank you ... I am looking at it now |
| 00:41.29 | sawgood | wanpipe1 | AFT TE1 | N/A | Disconnected | |
| 00:42.14 | sawgood | what do you mean by checking signalling first? |
| 00:42.29 | Lord_Rahl | anyone familiar with astgui? I found that every work fine except the save button on chrome and firefox 4. I want to see if I can fix that, bit I dont know what file it in |
| 00:43.04 | IsUp | sawgood: disconnected means, your card is not in LINK UP state with your provider |
| 00:43.26 | IsUp | sawgood: run Asterisk and try using "pri show spans" command on CLI |
| 00:43.52 | sawgood | Well, the system has been down for 2 weeks (using a SIP trunk since it went down) ... so should it come back live on its own? (since I reinstalled * and wanpipe drivers)? |
| 00:44.07 | *** part/#asterisk Bloudermilk (~Bloudermi@dsl081-234-075.lax1.dsl.speakeasy.net) |
| 00:44.21 | sawgood | PRI span 1/0: Provisioned, In Alarm, Down, Active |
| 00:44.21 | sawgood | "this is the output from pri show spans" |
| 00:46.59 | *** part/#asterisk Lord_Rahl (~Lord_Ralh@c-98-243-8-24.hsd1.mi.comcast.net) |
| 00:47.06 | *** join/#asterisk Blue-Dragon (~asdf@dbeuchert.com) |
| 00:47.35 | Blue-Dragon | Hi, is there someone who can answer me mISDN-related question? |
| 00:48.19 | Blue-Dragon | I'm wondering because mISDNv2 doesn't list any AVM-ISDN-Cards on their "Supported Cards"-list. Are those cards really only supported by mISDN 1? |
| 00:48.23 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:49.21 | IsUp | sawgood: probably something wrong with your configuration. if you are using wancfg_* its overwrites your current configuration |
| 00:49.44 | IsUp | sawgood: have you ever talked to your provider? |
| 00:49.56 | sawgood | IsUp: I am starting a call to them now |
| 00:50.26 | *** join/#asterisk rizwank (~rizwank@76.89.131.47) |
| 00:50.30 | *** part/#asterisk rizwank (~rizwank@76.89.131.47) |
| 00:50.39 | IsUp | sawgood: exactly. they'll tell you status of your link |
| 00:51.10 | sawgood | I am on hold now |
| 00:51.52 | *** join/#asterisk rizwank (~rizwank@76.89.131.47) |
| 00:53.00 | rizwank | I've got a number of Asterisk servers already; but it's time to add a new one; preferably using Asterisk 1.8x - is there a recommended OS distro that keeps Asterisk packages handy? I'm usually an CentOS guy, but most of what I've seen out there requires rebuilding Asterisk by hand each time; and when you're managing 10+ machines, you try to avoid that. |
| 00:54.27 | p3nguin | You should either be using the Asterisk repository that AsteriskNOW uses to get your RPMs, or you should be packaging your own builds into RPMs to distribute across your machines. |
| 00:54.31 | p3nguin | rizwank: ^^^ |
| 00:54.47 | sawgood | IsUp: I have an open trouble ticket with the vendor now |
| 00:54.59 | sawgood | When they call, they probably won't know much about Sangoma ... |
| 00:55.14 | sawgood | but they can tell me if they have a good working signal to the building, right? |
| 00:55.21 | IsUp | sawgood: doesnt matter, they'll tell you if they have signal or not. |
| 00:55.22 | IsUp | sawgood: yes |
| 00:56.13 | sawgood | cool ... so even if my card config files are 'wrong', bad, or incorrect, they will tell me they have a working signal to the 'smart jack' on the wall? |
| 00:56.47 | sawgood | Once I've confirmed the good signal to the building, then I guess I start from square one with the wancfg_dahdi scripting process ... |
| 00:56.57 | IsUp | sawgood: probably. PM me and i'll send you other debug commands of wanrouter utility |
| 00:56.59 | sawgood | or, have Sangoma give me advice tomorrow ... |
| 00:57.15 | sawgood | IsUP: I will (once the PRI is confirmed good) ... thank you! |
| 00:57.25 | IsUp | sawgood: no problem |
| 00:59.00 | sawgood | yes 100% ok with me! |
| 00:59.30 | luisfelice | Hi, I am having a static noise problem with a TDM400P card on the FXS ports, any idea? |
| 01:01.46 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
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| 01:25.56 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 01:26.11 | Aut0Exec | why is it that the only way to install asterisk is compile? |
| 01:26.18 | p3nguin | It's not. |
| 01:26.18 | Aut0Exec | why arent their binaries for distros? |
| 01:26.22 | Aut0Exec | like say debian |
| 01:26.22 | p3nguin | You there are. |
| 01:26.30 | p3nguin | aptitude install asterisk |
| 01:26.31 | Aut0Exec | oh really |
| 01:26.33 | Aut0Exec | latest? |
| 01:26.36 | Aut0Exec | 1.8? |
| 01:26.37 | p3nguin | yum install asterisk |
| 01:26.42 | Aut0Exec | 1.8? |
| 01:26.43 | Aut0Exec | no |
| 01:26.52 | Aut0Exec | prolly 1.6 |
| 01:26.56 | p3nguin | Don't ask me, look for yourself. |
| 01:27.30 | Aut0Exec | k |
| 01:27.46 | p3nguin | Asterisk 1.6.2.17 isn't any older than 1.8.3 |
| 01:27.52 | p3nguin | They were released on THE SAME DAY. |
| 01:27.58 | p3nguin | So 1.8.3 is not newer. |
| 01:28.25 | p3nguin | Do you have a problem compiling software? |
| 01:29.14 | Aut0Exec | yup 1.6 |
| 01:29.26 | Aut0Exec | yeah I like gtalk support |
| 01:29.31 | Aut0Exec | only with 1.8 right? |
| 01:29.42 | p3nguin | Oh, you'll want 1.8 for that. |
| 01:29.45 | Aut0Exec | bingo |
| 01:29.47 | Aut0Exec | :) |
| 01:30.16 | Aut0Exec | had such a hard time finding the dependencies for things like alsa, gtalk |
| 01:30.19 | Aut0Exec | etc etc |
| 01:30.19 | Aut0Exec | omg |
| 01:31.05 | Aut0Exec | and i'm a nub so I select lots of unecessary sound files... like all baiscally... en,es, the works.. so now I have to wait forever for comple now :( |
| 01:50.27 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:51.48 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
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| 02:45.39 | devdvd | anyone using a linksys pap2t ata? Just got one and having trouble making 3way calling work. Wanted to know if anyone had any ideas as to what might be wrong. What happens is, I call the first person i want to talk to, then i press the flash button on my phone and dial the second person, but when i press flash again, the first person is gone, gets disconnected |
| 02:46.27 | devdvd | s/pap2t/papt2 |
| 02:47.18 | p3nguin | you were close. |
| 02:47.26 | p3nguin | s/close/almost there/ |
| 02:48.01 | p3nguin | PAP2T |
| 02:48.42 | devdvd | yea, eyes and brain are a bit buggy tonight, 3 days without sleep will do that to a person |
| 02:50.01 | pigpen | Hi all, back when I was doing realtime extensions, the priority was 1,2,3,4 and so on. Now, I have in my normal txt extensions.conf, 1,n,n,n,n. To add to it, I also use 1,n(start),n(stop),n,n....and so on. |
| 02:50.58 | pigpen | does anyone know if I can designate the priority in a realtime database such as "n(start)" ? |
| 02:51.14 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-hckfeixmniefdbsx) |
| 02:54.35 | pigpen | priority int4 NOT NULL default 0, <<<<< nevermind, no hints for me. |
| 02:57.58 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 02:58.02 | sequencer | hi all |
| 02:58.25 | sequencer | is there any way to disable fax signal detection ? using asterisk 1.6.2 and frepbx 2.8 |
| 02:59.01 | sequencer | the reason for that is sometimes for whatever reason, asterisk detects a fax signal and drops a random active call |
| 02:59.26 | sequencer | anyone has any similar incidents? |
| 02:59.46 | sequencer | i have already unloaded the res_fax and res_fax_diguim |
| 03:01.57 | pigpen | faxdetect=no in /etc/dahdi/system.conf or /etc/asterisk/zapata.conf depending on the ver ( I can't remember when dahdi came around.....bad memory....late.... |
| 03:02.16 | sequencer | thnx |
| 03:02.58 | sequencer | what happens if i disabled dahdi itself? |
| 03:03.50 | sequencer | my system uses sip only for inbound/outbound calls |
| 03:07.47 | pigpen | I wouldn't. You are likely using it for timing in some sort. |
| 03:08.32 | sequencer | anything else i need to be aware of? |
| 03:08.41 | pigpen | I think in 1.6.2.13 they started splitting off some real timing modules, which caused other odd issues through 1.8.2.3, fixed in 1.8.2.4 |
| 03:08.46 | sequencer | i disabled the faxing in both trunks i have |
| 03:09.07 | pigpen | yeah, having sip only, the dahdi setting will have no affect. |
| 03:09.23 | pigpen | you may want to debug and see what is really happening. |
| 03:09.38 | sequencer | it just says fax signal |
| 03:09.43 | sequencer | then a random call drops |
| 03:10.05 | sequencer | it gets crazy whn you have a zombie call every 20-30 minutes for no reason |
| 03:10.16 | sequencer | esp. in a very busy environment |
| 03:10.21 | sequencer | 30+ ppl on phone :s |
| 03:10.26 | pigpen | now, please remember, you are using freepbx. there is -allot- that is happening behind the scenes. |
| 03:10.35 | sequencer | right |
| 03:10.47 | p3nguin | allot? Do you mean "a lot?" |
| 03:10.56 | pigpen | this channel is for the more core asterisk person. |
| 03:11.06 | pigpen | p3nguin = union spell checker |
| 03:11.09 | pigpen | ;-) |
| 03:11.34 | pigpen | Dam Education System!!!! heh |
| 03:12.29 | pigpen | sequencer, if you have 30+ calls at a time, you may want to consider setting up a real deployment on the side and move over to a platform you can really control. |
| 03:13.15 | pigpen | just my 2 cents worth. |
| 03:13.21 | pigpen | right p3nguin !!?? |
| 03:23.20 | eMBee | good morning |
| 03:23.38 | carrar | ood evening |
| 03:23.40 | carrar | good |
| 03:24.24 | eMBee | is looking at a wierd problem: since this morning calls made to a queue don't play a ringing, that is the call works and the phones ring, but the caller hears nothing until an agent talks |
| 03:24.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:26.36 | pigpen | eMBee, what ver? |
| 03:27.16 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-4354d123.dyn.optonline.net) |
| 03:27.50 | eMBee | 1.6 |
| 03:30.06 | pigpen | I think you really don't want it just to sit there and ring. |
| 03:30.21 | pigpen | that would drive anybody insane (unless this is the idea) |
| 03:30.22 | FuriousGeorge | so out of nowhere today my sangoma using dahdi cant take a call on my server. I recently (8 weeks ago) upgraded from zaptel, so I can't remember if this is typical for dahdi but: |
| 03:30.28 | FuriousGeorge | handle_alarms: Detected alarm on channel 7: Red Alarm |
| 03:30.32 | pigpen | consider setting up a musicclass |
| 03:30.41 | FuriousGeorge | my guess is that it has something to do with that |
| 03:31.21 | pigpen | FuriousGeorge, we have had something similar. We had to go shutdown the box, disconnect power, and bring it all back up. |
| 03:31.36 | pigpen | hence why we are starting to use external sip boxes, such as audiocodes |
| 03:32.01 | FuriousGeorge | pigpen: i did find this http://lists.digium.com/pipermail/asterisk-users/2007-February/179354.html |
| 03:32.04 | FuriousGeorge | but ill try that anyway |
| 03:32.41 | pigpen | yeah, try that, because nothing is worse than chasing your tail for an easy fix. |
| 03:38.57 | *** join/#asterisk eugeneoden (~goden@63.133.138.10) |
| 03:39.21 | eMBee | pigpen: what do you mean? |
| 03:39.53 | eMBee | long ringing not a good idea? |
| 03:39.57 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:40.14 | pigpen | music. |
| 03:41.09 | pigpen | it soothes the soul. Something like Guns and Roses, Iron Maiden, maybe Metalica. ;-) |
| 03:41.26 | pigpen | ie: music on hold. |
| 03:41.48 | pigpen | you setup a musicclass in the moh, then reference it in the queues |
| 03:41.57 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:41.59 | pigpen | (from my vague memory) |
| 03:43.42 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:47.18 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:48.21 | eMBee | ok, i agree with that, however the waiting times in this queue are not long so it's ok, right now there is no sound at all which is rather disturbing for the caller |
| 03:49.09 | pigpen | I don't thing a ring tone is an option. but, I am sure you can find some wav file out there that is a ring tone which you can play in a music class. |
| 03:49.15 | p3nguin | Answer the line. Play some sounds. |
| 03:49.36 | eMBee | well, actually, it has been ringing before, it's not ringing now though |
| 03:50.41 | pigpen | I had a bad experience with a call queue when I was a child...so I avoid them when I can.....and no, I don't want to talk about it. |
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| 04:05.57 | eMBee | :-) |
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| 04:25.00 | Blue-Dragon | Hi again! I'm using asterisk 1.8.3 and 3cxphone as sip-client. When I got called I can't hear the one on the phone but he can hear me. When I edit my dialplan so that he has to speak to the voicemail, it works. I can also call my voicemail and listen to his message. Any ideas? Does this seem to be an asterisk-problem or a problem with my SIP-client? |
| 04:26.17 | pigpen | is your asterisk box on the same network as the softphone? |
| 04:26.52 | Blue-Dragon | yes |
| 04:27.21 | pigpen | it sounds like you have a nat issue. |
| 04:27.49 | Blue-Dragon | even if I can hear my asterisk when putting a Playback(hello-world) in the dialplan? |
| 04:29.45 | pigpen | show us the call topology: ie: sip-softphone<---->asterisk<----(fw/router)----<SIP Trunk>----Online provider.....etc |
| 04:35.54 | Blue-Dragon | It should be like that |
| 04:36.01 | Blue-Dragon | But NAT is actually enabled |
| 04:37.21 | pigpen | here is a general rule of thumb: a sip session should only have one NAT. Anymore, and it will screw up. (and the term is lost in my mind at the moment...too much realtime) |
| 04:43.21 | Blue-Dragon | is it wrong when I say "my router is the only NAT"? |
| 04:43.43 | pigpen | depends. |
| 04:43.47 | Blue-Dragon | On what? |
| 04:43.50 | pigpen | is the other side natted? |
| 04:43.54 | pigpen | nat'ed |
| 04:43.57 | pigpen | whatever. |
| 04:44.40 | pigpen | I can't spell tonight |
| 04:45.36 | jmordica | anyone got some space or know of anyone that can handle a midtower? all i need is 1mb 95percentile burstable to 10 or 100mb |
| 04:45.39 | jmordica | ?? |
| 04:47.38 | pigpen | what area of the country (or world)? |
| 04:48.33 | Blue-Dragon | the other side conventional phone |
| 04:48.48 | Blue-Dragon | is a* |
| 04:49.16 | Blue-Dragon | calling a provider who then calls my asterisk over SIP |
| 04:50.03 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:50.14 | pigpen | I would try a real phone or such, just to rule out the softphone. |
| 04:52.02 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 04:52.11 | Blue-Dragon | I'll try to set up another softphone using another extension on my asterisk |
| 04:53.00 | Blue-Dragon | I wonder what happens when I call sofphone 1 with softphone 2 |
| 04:54.28 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 04:55.18 | *** join/#asterisk bmg505 (~leon@196-209-99-182.dynamic.isadsl.co.za) |
| 04:59.27 | Blue-Dragon | It works |
| 04:59.43 | Blue-Dragon | Ah, I think, I got it |
| 05:05.39 | Blue-Dragon | pigpen, I really had the option nat=yes in my config of the SIP-provider. But setting it to no or removing it didn't really help, so I'm still facing the same problem |
| 05:06.02 | Blue-Dragon | Do you have any other ideas how I can find out what the problem is? |
| 05:07.19 | pigpen | debug sip |
| 05:10.58 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 05:12.54 | Blue-Dragon | But how is the magic question. I don't think that I have enough knowledge about all that stuff, thats why I'm asking here :\ |
| 05:13.32 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qezdxuzazvcgyozp) |
| 05:18.13 | tyman | I've got a bit of an emergency...my asterisk server has had no configuration changes, verified by diffing the /etc/asterisk with version control. Yet, for an unknown period of time, we're not able to take calls. Outbound calls are fine... please see this http://pastebin.com/HQMxDcRU |
| 05:19.03 | tyman | i'm handling a major networking cutover for a client right now and realized my phones were completely down...i'm in a pickle |
| 05:19.43 | tyman | I have never seen the 66.42.120.184 addr before...this is not my itsp |
| 05:20.24 | pigpen | is pacwest your provider? |
| 05:20.25 | tyman | the calling number (559).... is my cell |
| 05:20.30 | tyman | flowroute |
| 05:20.46 | pigpen | they may just be reselling pacwest. |
| 05:20.46 | tyman | s/$/is my itsp/ |
| 05:21.18 | pigpen | pacwest has been pushing to this, hence why they lost our business. |
| 05:21.29 | tyman | think they made some changes?...and now my sip config is incompatible as well as my sip authentication does not allow them |
| 05:22.01 | pigpen | yeah, I would open a ticket with your provider. |
| 05:22.10 | tyman | pigpen: pacwest pushed flowroute to resell? |
| 05:22.40 | pigpen | no, their business plan. |
| 05:23.03 | pigpen | ie: we had many lines with a provider here in Texas. PacWest came in and bought them out. |
| 05:23.46 | pigpen | then, as part of their business plan, they pretty much told all of us ISP's to piss off and did nothing but screw up circuits and every bit of billing they could. |
| 05:23.54 | pigpen | hence, driving us away. |
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| 05:24.08 | pigpen | but, they are good at it: This won't be the first time they have gone bankrupt. |
| 05:24.19 | pigpen | imho. |
| 05:26.34 | tyman | pigpen: PacWest going bankrupt? think i'm following |
| 05:26.58 | tyman | did u recognize that number as a pacwest number? |
| 05:27.20 | pigpen | I don't know if they are in the process of it, but I have talked to employees that are bailing the company in texas. |
| 05:27.22 | tyman | sorry...says it right there plain as day |
| 05:27.31 | pigpen | yeah, the resolve shows pacwest. |
| 05:27.42 | pigpen | their techs are stupid. |
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| 05:33.41 | pigpen | we moved our pri's to TWTelecom. Upstream is with TWTelecom & Cogent. |
| 05:33.53 | tyman | pigpen |
| 05:33.57 | pigpen | tyman |
| 05:34.04 | tyman | checked my email... |
| 05:34.12 | pigpen | I am glad. |
| 05:34.13 | tyman | there's a notifcation in ther |
| 05:34.35 | pigpen | Oh...late on the plastic surgery payment? |
| 05:34.38 | pigpen | ;-) |
| 05:34.39 | tyman | pigpen: sorry...overwhelmed with people after me...service is down |
| 05:35.05 | tyman | http://pastebin.com/d517txud |
| 05:35.07 | pigpen | ah...well there you go. One thing off your plate. |
| 05:35.13 | tyman | def |
| 05:35.31 | tyman | geez...thanks for the info...sorry for the crazy posts |
| 05:35.33 | tyman | later |
| 05:35.54 | pigpen | good luck. HAIL PAC WEST!!!! |
| 05:35.55 | tyman | this DURING a bgp multihoming cutover for an isp |
| 05:36.05 | pigpen | nice. |
| 05:36.25 | tyman | they think i'm slaving away on their gear |
| 05:36.27 | tyman | :-) |
| 05:36.37 | pigpen | all billable baby. |
| 05:36.51 | pigpen | hmm...what if they are in this channel...hmm.... |
| 05:36.51 | tyman | y, nite |
| 05:36.57 | tyman | gulp |
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| 06:11.37 | awk | Hi guys, any advice here.. I am playing with an asterisk box in a VMware env.. Windows Server -> Vmware Workstation -> CentOS -> Asterisk 1.4.39-1 .. RTP is clear, however any prompt is so slow and lagged, any idea what could be causing all prompts to be so slow? I have disabled zaptel and started asterisk and tried changing Asterisk versions to 1.4.35.. Could there be some setting in the |
| 06:11.38 | awk | BIOS causing this or a VMWare setting? |
| 06:13.16 | pigpen | you could have a loop or something goofy in your dialplan. |
| 06:13.28 | pigpen | make it _real_ simple. |
| 06:13.48 | pigpen | also, get on 1.6, or 1.8 |
| 06:14.04 | awk | Our GUI runs 1.4 |
| 06:14.09 | pigpen | nobody will really help until you get on something current. |
| 06:14.19 | awk | We have it at hundreds of sites. |
| 06:14.30 | pigpen | ah... |
| 06:14.39 | pigpen | yeah, I bet you dont want to update at whim. |
| 06:14.44 | awk | 1,6 and 1,6 is nowhere near stable enough for some of our clients (banks) etc |
| 06:14.52 | pigpen | have the devs write it better. |
| 06:14.54 | awk | err 1.6 and 1.8 |
| 06:14.54 | pigpen | ;-) |
| 06:15.14 | pigpen | yeah, 1.8 is to be the answer..... |
| 06:15.18 | awk | There is something, I wonder if it has something to do with ACPI |
| 06:15.20 | pigpen | when it matures. |
| 06:15.27 | pigpen | yeah, sounds like a loop. |
| 06:15.33 | pigpen | a goofy include |
| 06:15.35 | pigpen | something |
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| 06:29.20 | ajmcello | i have an unauthorized SIP user who connects as SIP/IP_ADDRESS |
| 06:29.36 | ajmcello | it isnt listed in sip.conf, and i cannot figure out how they are connecting. i have full logging turned on and i never see it register |
| 06:29.43 | ajmcello | could someone please help? |
| 06:30.24 | ajmcello | it started once i upgraded from 1.4 to 1.8.3 |
| 06:30.59 | p3nguin | ajmcello: Change "allowguest=yes" or ";allowguest=no" to "allowguest=no" in sip.conf |
| 06:32.05 | ajmcello | lol |
| 06:32.08 | ajmcello | cool, thanks. |
| 06:32.20 | ajmcello | why is that on by default? |
| 06:32.31 | p3nguin | I'm not sure. |
| 06:32.46 | ajmcello | lame. |
| 06:32.54 | ajmcello | wonder how much LD i have racked up in overseas calls. |
| 06:33.00 | p3nguin | Sorry, I can't know everything. |
| 06:33.40 | ajmcello | hehe. |
| 06:39.03 | kaldemar | ajmcello: registration is not needed to make calls, so better not look for registrations if you have unauthorized connections. |
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| 06:45.50 | ajmcello | kaldemar: allowguest=no will fix that? |
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| 06:47.20 | pigpen | but to even make a call, although not registered, a valid name/secret must be used. |
| 06:47.36 | kaldemar | ajmcello: nothing to fix really. registration is just a way to let the other end know where you are. it's not like a login kind of procedure. |
| 06:49.07 | pigpen | something you could do, at least to help at the moment, is to block the ip address they are connecting from at the firewall level. |
| 06:49.43 | pigpen | if you are not a public sip provider, I would secure your sip sessions behind a firewall and within vpn's. |
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| 07:05.39 | ChannelZ | and for god sakes put some pants on! |
| 07:09.10 | pigpen | That is certainly wise advice. I am positive it comes from first hand experience. |
| 07:09.42 | pigpen | shit. I said "hand". |
| 07:14.36 | kleszcz | morning |
| 07:15.34 | ajmcello | interesting |
| 07:15.53 | ajmcello | with allowguest=yes, if they guessed a SIP name, they could make calls without knowing the password? |
| 07:17.26 | kaldemar | ajmcello: no |
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| 07:18.03 | kaldemar | ajmcello: allowguest=yes allows such calls that do not match a device defined in sip.conf. |
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| 08:08.15 | schmidts | good morning |
| 08:11.06 | kleszcz | schmidts: mornin' |
| 08:11.43 | asterisk-learner | schmidts: morning |
| 08:15.44 | asterisk-learner | in ChanSpy() application pressing '*' will "stop spying and look for another channel to spy on." |
| 08:15.55 | asterisk-learner | does this means it randomly switches between active calls ? |
| 08:16.01 | shapr | try it? |
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| 08:19.43 | schmidts | asterisk-learner it depends on what you start chanspy application |
| 08:20.21 | schmidts | if you use it like ChanSpy(SIP-123) then it will rotate over each channel which starts wich SIP-123... and normaly asterisk starts with the newest channel first |
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| 08:20.54 | asterisk-learner | i am using it on smthg like this |
| 08:20.55 | asterisk-learner | exten => _XXXX,n,ChanSpy(IAX2/${EXTEN:5},qb) |
| 08:21.29 | asterisk-learner | what do u mean by : "normaly asterisk starts with the newest channel first" |
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| 08:21.45 | asterisk-learner | if i have 5 channels and i am listening to one, then i press * |
| 08:22.01 | asterisk-learner | many times, will it cycles thru the others or just pick the saem one each time |
| 08:22.03 | asterisk-learner | ? |
| 08:22.38 | schmidts | asterisk-learner which version do you use? if you use something lower than 1.8 then the internal channel list starts with the newest channel after this you have the second newest and the last one is the one which is the oldest |
| 08:23.22 | schmidts | if you have other channels which match your search like IAX2/12345 then yes, this should work ;) |
| 08:23.30 | asterisk-learner | schimdts: ok, yes i am using 1.4.36 |
| 08:23.54 | asterisk-learner | ok good then it will wake thru the list of channels |
| 08:24.32 | schmidts | asterisk-learner just do a core show channels and see which channel is the first one, this is the same order you will have in chanspy ;) |
| 08:25.40 | asterisk-learner | schimdts: ok thx :-) |
| 09:01.49 | *** join/#asterisk UnixDev_ (~UnixDev@unaffiliated/unixdev) |
| 09:03.15 | UnixDev_ | how can I change the system default unavailable message for a user without having to set one for each? |
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| 09:07.24 | jkroon | when using queues, the leastrecent strategy - does it take agent penalties into consideration at all? |
| 09:08.17 | jkroon | my situation is that i've got three technicians to which I'd like to send calls in a least recent fashion, but if they are all three engaged let it overflow to some of the more senior guys which I don't like to normally take calls. |
| 09:10.55 | kaldemar | UnixDev: which message are referring to, exactly? when does your user get this message in your dialplan? |
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| 09:11.08 | Sertys | yeah |
| 09:11.14 | Sertys | there's no "default" message |
| 09:11.55 | Sertys | jkroon: well, 2 Dial()s with 2 callgroups |
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| 09:12.42 | jkroon | Sertys, please explain how that addresses my requirements, I fail to see how to use 2 Dial()s to get what I want. |
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| 09:13.34 | jkroon | how do i queue people doing that should all the senior technicians (two of us) also be engaged already? |
| 09:14.59 | Sertys | well, it's a natural fallback scenario |
| 09:16.30 | Sertys | ext,1,Dial(sip/tech1&sip/tech2&sip/tech2,60) |
| 09:16.31 | Sertys | ext,2,Dial(sip/officer1&sip/officer2) |
| 09:16.48 | Sertys | u're all obsessed with queues |
| 09:16.52 | jkroon | no, that doesn't do what I asked. |
| 09:17.08 | jkroon | that will ring all three guys for 60 seconds |
| 09:17.22 | Sertys | yep |
| 09:17.34 | jkroon | then go to the other two after that, it also means i must disable call waiting on those three phones, which is not desirable. |
| 09:17.45 | kaldemar | jkroon: if all members are busy, you could maybe check if the senior guys already are in the queue, and if not, add them temporarily to the queue and remove after the call. |
| 09:18.34 | jkroon | kaldemar, ok, so upon entering the queue, check if there are members available, if not, add the senior guys, and after the call remove them again? |
| 09:18.43 | jkroon | doesn't that open me up for a bunch of race conditions? |
| 09:19.29 | kaldemar | jkroon: well, a matter of testing if it is feasible. |
| 09:21.17 | jkroon | i still figure it may be better to get it done in app_queue |
| 09:21.48 | SiNGLer | there is other solution: create 2 queues, one without seniors, and one with. check if members in first available, if not, then send to second queue |
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| 09:23.31 | Sertys | SiNGLer: yea |
| 09:23.36 | Sertys | makes sense |
| 09:25.21 | Sertys | such complex scenario always miss my mind |
| 09:25.32 | Sertys | *scenarios |
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| 09:44.29 | UnixDev | kaldemar: im talking about the unavailable messages in the voicemail application |
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| 09:49.46 | jkroon | SiNGLer, that could work. |
| 09:49.58 | kaldemar | UnixDev: find the corresponding sound file somewhere under /var/lib/asterisk/sounds and change it to what you want. |
| 09:50.38 | kaldemar | UnixDev: the voicemail files are named vm-* |
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| 09:51.16 | jkroon | SiNGLer, kaldemar - how do i check whether there are any available members in a queue (ie, at least one member that has a status of NOTINUSE)? |
| 09:52.24 | jkroon | Sertys, that is still _simple_ ... you should see some of the requirements some of my clients cook up. |
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| 10:00.10 | tuxx- | Hey guys, my dahdi setup doesn't seem to be working. First i had problems loading pri_cpe, that seems to fixed now, but i'm getting the error that dahdi can't load channel 1. Anyone got any hints which way i should look? I have recompiled the whole bunch (libpri, dahdi, asterisk) a couple of times, and now im stuck on this error. http://pastie.org/1673712 |
| 10:02.57 | kleszcz | show me /etc/asterisk/chan_dahdi.conf |
| 10:03.05 | tuxx- | it's in the paste |
| 10:03.31 | tuxx- | made it a little more clear with ------ on the side |
| 10:03.32 | tuxx- | :) |
| 10:04.37 | kleszcz | change Dial(DAHDI/r0/${EXTEN}) |
| 10:04.56 | tuxx- | it's not a dial, it's when starting asterisk and loading chan_dahdi.so |
| 10:10.54 | WIMPy | tuxx-: You've got only 15 channels? |
| 10:11.03 | WIMPy | Is that a new install or an upgrade? |
| 10:11.23 | WIMPy | Do you have some old zaptel stuff floating around? |
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| 10:14.17 | tuxx- | new install WIMPy |
| 10:14.19 | tuxx- | no zaptel stuff |
| 10:14.29 | UnixDev | kaldemar: I also want to stop the default unavailable message from speaking the extension... how can I do that? |
| 10:15.04 | WIMPy | UnixDev: Maybe you want MiniVM instead of VoiceMail? |
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| 10:16.06 | tuxx- | probably a jumper problem |
| 10:16.07 | tuxx- | >_< |
| 10:16.32 | UnixDev | WIMPy: can minivm be accessed through the legacy ivr menu interface? |
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| 10:23.47 | WIMPy | I only know that the idea of MiniVM is that it does not provide much so that you get the chance to maximum customisation. |
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| 11:37.47 | awk | Ok guys here my prediciment.. I want to have a Incoming -> Queue (caller sits in the call) when he is in the queue I want him to get sent to an AGI script where he will then be able to answer questions, however when it moves to top of the queue it must then transfer it to the agent to handle the call.. it is a way to by pass time and gather information at the same time? |
| 11:37.57 | awk | I was thinking of maybe a meetme room inside a queue? |
| 11:38.02 | awk | or what do you guys recommend.. |
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| 11:54.41 | CRCinAU_ | p3nguin: ping? |
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| 12:25.08 | jkroon | hi guys, i was wondering, with FXS ports, is it theoretically possible to detect (with assistance from kernel drivers if need be) to detect the difference between a port that is not connected vs a port that is onhook? |
| 12:26.10 | jkroon | awk, queue a Local/ channel into the queue, somehow detect when it gets picked up by the agent and "steal" the channel from the games app? |
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| 12:33.13 | jkroon | awk, yea, originatinig a Loca/ channel that sits in the queue, using some AGI or even a simple macro/gosub on the Queue() call of that Local/ channel might well do the trick for you. |
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| 12:52.09 | jkroon | ok, looking at the source for app_queue it looks like the app should be using the penalty for each individual queue member to adjust the metric (calculated value based on lastcall + 100000 * penalty), but this doesn't seem to actually be what's happening. |
| 12:52.42 | TJNII | jkroon: No. A traditional phone opens the circuit when on hook. It is electrically equal to no connection. |
| 12:52.46 | jkroon | so I believe the _intended_ behaviour for app_queue is that which I want, but the actual behaviour is in conflict. |
| 12:53.14 | jkroon | TJNII, thanks. that confirms what I thought would be the case. wish there was a way though. |
| 12:53.31 | TJNII | Some phones may draw some current for things like CID, but you won't be able to reliably detect that. |
| 12:53.40 | jkroon | was hoping that the ringer might in fact be a partial short that could be detected. |
| 12:53.53 | TJNII | I forgot about the ringer. |
| 12:54.07 | TJNII | I think that will depend on the phone, though. |
| 12:54.24 | TJNII | Modern solid-state ringers will be a lot harder to detect than a old coil ringer. |
| 12:54.28 | jkroon | so a NC would be infinite resistance (ie, no connection), the ringer some (low) current. |
| 12:54.54 | jkroon | and a off-hook is "high" current, something to that effect, but it was a really long shot. |
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| 12:55.01 | TJNII | The ringer is in parallel with the rest of the phone, but the rest of the phone is disconnected by the hook switch. |
| 12:55.28 | jkroon | that is what i figured, which is why when the FXS port generates ring it actually rings. |
| 12:55.38 | TJNII | The phone is a constant current device, so when it goes off hook the line voltage drops drastically. |
| 12:56.17 | jkroon | I don't the technical details, but yea, constant current would necessitate a voltage drop. |
| 12:56.48 | jkroon | the question remains: is the Digium FXS cards sensitive enough to detect the difference between no connection and a ringer? |
| 12:57.19 | jkroon | essentially generate a RED alarm on no connection, and onhook/offhook otherwise. |
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| 13:00.42 | tzafrir_laptop | jkroon, not really sure |
| 13:02.03 | tzafrir_laptop | basically there should be some difference. But I'm not really sure how sure you can be of it. |
| 13:02.34 | tzafrir_laptop | You can query the SLIC directly for current and voltage |
| 13:03.28 | jkroon | tzafrir_laptop, ok - how can I query that? |
| 13:05.53 | TJNII | That may not work either as the ringer is AC coupled, though |
| 13:06.07 | TJNII | So if the line is pure DC the ringer will draw no current. |
| 13:06.24 | jkroon | it still has to have _some_ resistance, which if connected in DC will produce current. |
| 13:06.29 | TJNII | I think you may be able to do this, but every phone is going to give you different readings, if you can do it at all. |
| 13:06.32 | jkroon | a ringer fires when it _detects_ AC. |
| 13:07.05 | TJNII | No, some are just connected through a cap. So pure DC will just charge the cap. That's how the old phones worked. |
| 13:07.36 | TJNII | You have the internal resistance of the cap itself, but will that be low enough to give you a usable reading? That's your challenge. |
| 13:07.38 | coppice | if the ringer draws DC current it will permanently loop the line |
| 13:07.45 | jkroon | ah ok. hmm, caps leaks. |
| 13:08.19 | TJNII | Anyways, best of luck. I have to go to work. |
| 13:08.27 | jkroon | coppice, that doesn't matter. as long as it's a really, really small current it makes no real difference, I think (i'm not an electrical engineer, just a CS grad student with some interest in electronics) |
| 13:09.13 | coppice | it sure is really really small :-) |
| 13:09.27 | tzafrir_laptop | jkroon, there's an ioctl to get values of all registers. This should help you |
| 13:09.52 | jkroon | tzafrir_laptop, existing tool that can dump this values for me? |
| 13:09.53 | tzafrir_laptop | Get the spec of the chip from Silicon-Labs's site |
| 13:10.38 | jkroon | this feels like one of the crazier ideas i've had in a while. |
| 13:10.43 | *** join/#asterisk TimeRider (~steve@92.27.131.175) |
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| 13:11.23 | tzafrir_laptop | jkroon, dahdi_diag is a small program that sends that IOCTL |
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| 13:13.48 | coppice | jkroon: what are you trying to detect? |
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| 13:18.48 | jkroon | thanks tzafrir_laptop |
| 13:19.48 | *** part/#asterisk mintee (1000@static-64-115-220-213.isp.broadviewnet.net) |
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| 13:35.06 | jkroon | thanks tzafrir_laptop - it's a no go though, the info output to dmesg definitely doesn't contain any identifying for NC vs Onhook, and the only difference to offhook is that it gets a current tone ... |
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| 14:35.50 | gentoo_fun2 | so question |
| 14:35.56 | gentoo_fun2 | does Device does not match ACL |
| 14:36.10 | gentoo_fun2 | mean i have implemented my sip.conf's permit, deny correctly? |
| 14:36.47 | *** join/#asterisk leafartn (~chatzilla@190.145.253.51) |
| 14:38.31 | gentoo_fun2 | I am getting a ton of script kittens attn trying to login as peers to my lil asterisk box |
| 14:39.14 | kaldemar | gentoo_fun2: you better block them at your firewall already. many people use fail2ban for that. |
| 14:39.32 | *** join/#asterisk nestAr (nester@blackbox.ninjaz.net) |
| 14:41.05 | leafartn | Hi! When I add a member in a queue, if I type "queue show" command, in the members shows me always the memebers with status (dynamic)(not in use) although is in the middle of a call, so if the queue receives another call, this member receives this call |
| 14:41.05 | gentoo_fun2 | kaldemar: that is the next step |
| 14:41.20 | gentoo_fun2 | but i am trying to properly reject access with asterisk as well |
| 14:41.41 | gentoo_fun2 | ive been manually blocking with iptables for a couple of days now |
| 14:42.02 | gentoo_fun2 | I just cant figure out what that message means |
| 14:42.39 | leafartn | ...and the call should wait meanwhile the member ends the actual call |
| 14:44.09 | kaldemar | gentoo_fun2: and yes, you get "Device does not match ACL" when you use a denied address. |
| 14:44.16 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:44.18 | *** join/#asterisk jshriver (~jshriver@72.240.39.37) |
| 14:44.20 | gentoo_fun2 | cool how did u find that out |
| 14:44.22 | jshriver | greetings :) |
| 14:44.52 | jshriver | Is it possible to record live phone calls and if so have leads? A girl at work has been getting some harasing calls and need/want to record them. |
| 14:45.17 | jshriver | already told her she'd haev to inform him legally due to recording laws, so now it's a matter of tech |
| 14:47.27 | nestAr | leafartn: sounds like a call waiting issue.. the phone has to tell * that it's on another call and not to bother it. |
| 14:47.28 | jshriver | hrm anyone on :) |
| 14:47.35 | kaldemar | gentoo_fun2: by looking at the source. :P |
| 14:47.40 | nestAr | jshriver: Monitor() |
| 14:48.14 | gentoo_fun2 | kaldemar: appreciate your time |
| 14:48.24 | jshriver | No such command hrm |
| 14:48.38 | nestAr | jshriver: it's a dialplan command |
| 14:48.52 | gentoo_fun2 | ya u gota check your extensions |
| 14:49.08 | nestAr | jshriver: ala: exten => s,4,Monitor(wav,${CALLFILENAME},m) |
| 14:49.25 | jshriver | ty looking for examples now. |
| 14:49.39 | jshriver | Is there a way a user can say press a combo and it starts to record? |
| 14:51.13 | leafartn | nestAr: but in asterisk queue show this member appears as (not in use) |
| 14:51.58 | nestAr | leafartn: i understand, because * doesn't keep track of it.. for better or worse, it doesn't know that phone is on a call, once it hands over the call |
| 14:52.04 | nestAr | jshriver: check out features.conf |
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| 14:54.45 | *** join/#asterisk madwill (~dude@teluq37.teluq.uquebec.ca) |
| 14:54.51 | madwill | Hi |
| 14:55.33 | madwill | Anybody here ever tryed to transcode h323 to something lik h264 in order to provide "connectivity" with web players ? |
| 14:56.05 | madwill | Is there a payable plugins already in place ? |
| 14:56.07 | nestAr | leafartn: http://pastebin.com/kbUc7ZzS |
| 14:56.22 | nestAr | leafartn: that's what i was using, specifically for this issue on polycom phones |
| 14:56.55 | madwill | i intend to send it to red5 so i would need something like h263/h264 |
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| 14:58.43 | jshriver | I enabled automon *1; in features.conf but nothing is getting put in the /var/log/asterisk/monitor |
| 14:58.48 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 14:58.49 | jshriver | er /var/spool/asterisk/monitor |
| 14:59.36 | madwill | there is no such things as a xuggler for asterisk? |
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| 15:00.05 | madwill | asterisk support h263 by default but does not do transcoding |
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| 15:01.06 | zkn | Any australians lurking in here at this hour? |
| 15:02.00 | nestAr | jshriver: yeah, i've never actually used it.. i just record everything. |
| 15:02.23 | jshriver | have a website that shows how to set that up? |
| 15:02.37 | nestAr | might try the asterisk wiki |
| 15:02.49 | nestAr | wiki.asterisk.org |
| 15:03.10 | chazzam | or the newbook |
| 15:03.12 | chazzam | ~newbook |
| 15:03.12 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 15:04.41 | madwill | what i'm asking for is stupid right ? |
| 15:04.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:05.22 | madwill | transcoding live h323 video from hardware videoconference rooms using sip is just asking for trouble later on |
| 15:06.07 | tzafrir_laptop | zkn, I'm not one, but I suggest asking a more direct questions |
| 15:06.30 | tzafrir_laptop | zkn, I also suspect this is not the best time of the day to aim for them |
| 15:07.24 | *** join/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 15:08.25 | zkn | yeah, thought so, i know there are at least a few australians around here... worth a try... i'm curious about australian ITSPs, but I don't want to clutter the main screen here with OT questions |
| 15:09.25 | zkn | will check back later..err.. or elarier to pick a few brains for recommendations |
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| 15:21.13 | jshriver | thank god for IRC, an * subscription is at the minimum $595 wow |
| 15:22.32 | Qwell | jshriver: That's quite inexpensive compared to other vendors. Also, not everybody is willing/able to do IRC support. |
| 15:22.44 | Qwell | and, IRC support, as you've seen, is not always...kind. |
| 15:23.03 | nestAr | indeed |
| 15:23.16 | jshriver | heh |
| 15:23.35 | nestAr | i can put up with a lot of BS to save $600 though. :D |
| 15:23.42 | jshriver | eh generally kind just not always the answer. automon must not be a very common thing. Not many sites talk about it and none so far work |
| 15:23.57 | jshriver | nestAr: same, plus my boss would never pay that just to have someone to ask questions |
| 15:23.58 | Qwell | Press the keys faster. |
| 15:24.06 | Qwell | (That'll be $595. Thanks.) |
| 15:24.07 | jshriver | brb |
| 15:24.25 | jshriver | do you know what it means to put wW in dial plan? |
| 15:24.49 | Qwell | 'core show application Dial' does |
| 15:26.37 | nestAr | indeed it does |
| 15:26.46 | jshriver | Does this look remotely correct? exten => 302,2,Dial(SIP/302,20,rtwW) |
| 15:27.10 | nestAr | looks ok to me |
| 15:27.26 | jshriver | ok wasn't sure what rt was or if I could just add wW there with rt present |
| 15:27.41 | nestAr | yeah, the core show application dial will explain all those |
| 15:28.12 | jshriver | one more question, can you reload the conf files while the server is running or do you have to /etc/init.d/asterisk restart every change? |
| 15:28.31 | jshriver | was hoping there would be a graceful way to restart or reload dynamically |
| 15:28.38 | nestAr | if it's just dialplan, just reload that.. |
| 15:28.43 | nestAr | other stuff depends.. |
| 15:28.48 | jshriver | erm ok how :) |
| 15:28.55 | nestAr | but reload from the CLI should work |
| 15:28.59 | nestAr | without a full restart |
| 15:29.10 | jshriver | ok googling "asterisk cli reload" |
| 15:29.49 | russellb | *CLI> dialplan reload ... or *CLI> core reload |
| 15:29.55 | russellb | reload dialplan or reload everything |
| 15:30.19 | nestAr | see also: module reload res_features.so |
| 15:30.26 | nestAr | if you made a change to features.conf |
| 15:30.36 | nestAr | etc etc etc |
| 15:30.56 | jshriver | brb ty |
| 15:31.49 | jshriver | appreciate it gentleman it works |
| 15:31.57 | jshriver | now back to my real job lol |
| 15:32.01 | nestAr | lol |
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| 15:50.52 | leafartn | nestAr: thanks |
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| 16:03.46 | no1peanut | Hi - I have a single party call to asterisk and am trying to log it to cdr. Will I not get the fields answer and end when making a single party call ? |
| 16:04.39 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
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| 16:05.30 | Aut0ExeC | does pstn allow for putting calls in queue or call park? |
| 16:05.39 | Aut0ExeC | with basic ATA setup |
| 16:05.41 | Aut0ExeC | of course |
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| 16:42.08 | NovceGuru | Hey guys, I know this is an asterisk channel, which is my pbx of choice, but in this sutation i'm forced to support a 3cx install :( Can anybody tell me if the call parking system in 3cx is as stupid as I think it is? If I understand correctly, the user has to know if anybody is parked on *00 through *09 ? What if someone "double parks" ? |
| 16:42.26 | NovceGuru | just a shot in the dark if someone else has had to deal with 3cx... |
| 16:50.05 | weta | Is there a global call timeout setting? I can see the dial option. If possible, I'd like to set it one place only. Thanks. |
| 16:50.27 | weta | sorry, call time limit. |
| 16:58.24 | devdvd | Hi all, have a bit of a problem. Running asterisk 1.6.2.15 and having an issue sending dtmf. What i notice is when i call something (like a place with an automated attendant), and it tells me to enter say my account and i type 12345, the system will read it back to me as like 11135 (or other combinations most of the time the digits are repeated) |
| 16:58.29 | devdvd | How do i solve this? |
| 16:59.21 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 17:05.06 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 17:05.35 | Joe_CoT | So previously I would get a notification on my phone when I had new voicemail. After moving from 1.4 to 1.8, I no longer get that. Has something changed? |
| 17:05.43 | Joe_CoT | Well, has something related changed |
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| 17:16.49 | superduty | @devdvd what are your dtmf sip settings? |
| 17:17.12 | devdvd | rfc2833 is the mode |
| 17:17.30 | devdvd | and i called my itsp and thats what they are using as well |
| 17:17.41 | devdvd | or are you looking for different information? |
| 17:18.37 | *** join/#asterisk oquidave_ (~oquidave@41.190.129.127) |
| 17:19.02 | oquidave_ | hello people |
| 17:21.01 | oquidave_ | am configuring a digium TDM pci card for my asterisk box |
| 17:21.27 | oquidave_ | i have installed asterisk on opensuse 11.3 with the dahdi drivers |
| 17:22.16 | Qwell | opensuse? My condolences. |
| 17:22.27 | oquidave_ | when i enter the asterisk CLI, and type dahdi show status to test the card, i get Alarms field set to RED |
| 17:22.31 | oquidave_ | what does that mean |
| 17:22.41 | Qwell | It means it's not connected to the other side. |
| 17:23.04 | oquidave_ | which side? |
| 17:23.12 | Qwell | The side that isn't you. |
| 17:23.33 | Qwell | What is it plugged in to? |
| 17:23.44 | oquidave_ | PCI slot |
| 17:23.50 | Qwell | ...the cables |
| 17:24.09 | oquidave_ | a t1 |
| 17:24.21 | Qwell | It cannot communicate with the other side. |
| 17:24.45 | oquidave_ | OK...so i need to first connect the T1? |
| 17:25.02 | oquidave_ | plus the RED lights are not blinking! |
| 17:25.47 | oquidave_ | on the TDM card..they only blink or light at startup |
| 17:25.51 | Qwell | RED alarm means that it cannot talk to the other side. If the cables are not connected, then of course it won't be able to. |
| 17:26.07 | oquidave_ | ok |
| 17:26.30 | oquidave_ | but the lights on the card don't light...wat does that mean? |
| 17:29.32 | oquidave_ | mr Qwell please...could you help with that? |
| 17:29.46 | Qwell | Plug in a cable and we'll talk... |
| 17:29.52 | oquidave_ | okay |
| 17:30.11 | oquidave_ | let me do it |
| 17:31.03 | *** join/#asterisk Lann (ab457dc8@gateway/web/freenode/ip.171.69.125.200) |
| 17:32.00 | Lann | Hello |
| 17:33.39 | Lann | I came here maybe a couple months ago asking about the feasibility of using asterisk for an audio MUD. Some people here reccomended using ChanSpy() in order to send multiple audio streams to the same connected user. Is it possible to use chanspy to both an individual, AND to a conference call full of people (including the individual) at the same time? |
| 17:34.56 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
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| 17:41.10 | raden | anyone have a idea why MOH dont work with aastra phones ? on 1.8 works fine in 1.6 |
| 17:41.17 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
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| 17:50.19 | _Corey_ | Lann: I'd probably pipe the sounds into a MeetMe using AMI or something |
| 17:50.38 | _Corey_ | Lann: But as with anything else, there are many ways to get at the same thing... |
| 17:54.32 | Janos | hello, got a simple question, when using asterisk's default pickup feature '*8', all that is needed to "group" the channels are the pickupgroup and callgroup parameters in the channels right ? |
| 17:55.41 | oquidave_ | okay mr. qwell, i have got to do go somewhere |
| 17:56.05 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 17:56.07 | oquidave_ | we will continue tomorow if you will be around...thanks for the help though |
| 17:56.52 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 17:58.39 | paulc | DAHDI under Centos: running dahdi_cfg I get "line 0: Unable to open master device '/dev/dahdi/ctl'" - the devices exist, same permissions as on an Ubuntu box (where it works fine). I'm stumped as to what in Centos is causing the problem. Can anyone offer a pointer in the right direction? |
| 18:02.55 | tvc123 | paulc: could be SELinux |
| 18:03.19 | tvc123 | try disabling it to check .. if thats the case you should write a rule to allow the access |
| 18:04.40 | paulc | tvc123: Hmm.. yeah, good call - I'd forgotten about that.. but doing a sestatus shows it as disabled already, so that's not it :-( |
| 18:05.23 | tvc123 | doh |
| 18:17.38 | chazzam | you are running it as root? |
| 18:18.29 | paulc | chazzam: yup |
| 18:19.20 | chazzam | and you have dahdi and <driver for your card> in output of lsmod? |
| 18:20.18 | paulc | I think half the problem is the kernel version, having 2.6.18-194.el5PAE reported from uname -r, and having 2.6.18-194.31.1.el5PAE as well in sources where stuff is installed |
| 18:20.44 | paulc | chazzam: no, I don't see dahdi in lsmod because dahdi won't load/start. And there's no hardware - it's just for timing for meetme via SIP |
| 18:20.50 | *** join/#asterisk frawd (~francois@3.Red-81-44-76.dynamicIP.rima-tde.net) |
| 18:23.42 | paulc | ok.. depmod solved it - seems ok now |
| 18:38.11 | *** join/#asterisk shortcircuit (~shortcirc@rosettacode.org) |
| 18:39.40 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 18:53.04 | *** join/#asterisk duna (~dreams@190.96.69.243) |
| 18:53.09 | duna | hi guys! |
| 18:53.17 | *** join/#asterisk sawgood (~sawgood@ip-69-33-230-149.sjc.megapath.net) |
| 18:54.33 | duna | question: what kind of server do you recommend for audio recording with asterisk + 8E1 (PRI) ? |
| 18:59.04 | tzafrir_laptop | paulc, "the device exists" but is the driver loaded? |
| 18:59.38 | paulc | tzafrir_laptop: It wasn't, but it is now - problem solved, we're all good :) |
| 18:59.40 | *** join/#asterisk angeld (~sdiofj@213.134.98.242) |
| 18:59.43 | angeld | hi |
| 19:00.01 | angeld | im having some trouble getting the datacard module to work.. |
| 19:00.08 | paulc | duna: a beefy one - with lots of RAM (hint: record to RAM then move the recording to disk, rather than recording directly to disk) |
| 19:00.28 | angeld | WARNING[22556] chan_sip.c: No such host. 123456 |
| 19:00.46 | angeld | trying to use it with the auto-dial (.call files) |
| 19:02.22 | duna | paulc yeah, i thought that too |
| 19:02.46 | paulc | angeld: pastebin one of your call files? |
| 19:03.13 | angeld | ok |
| 19:05.14 | angeld | paulc: http://www.pastebin.se/203503 |
| 19:05.44 | angeld | what i dont get is how i tell asterisk to use the datacard0 device? |
| 19:06.27 | wdoekes2 | do you have host=123456 for your sip peer? |
| 19:06.56 | angeld | its a 3g modem that connected |
| 19:06.59 | angeld | via usb |
| 19:07.51 | paulc | angeld: I'm confused - is that like a SIP device/end point? I'm not clear on what you're trying to do exactly |
| 19:08.16 | angeld | Im trying to make nagios notifications through my 3g modem at /dev/ttyUSB0* |
| 19:08.32 | angeld | it shows the 3g modem inside asterisk |
| 19:09.03 | angeld | but how i format the .call file and sip.conf so that i can adjust the phonenumber it rings with the .call file? |
| 19:09.51 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:10.44 | angeld | is it possible to choose which device and phonenumber to use on the channel line of the .call file? |
| 19:13.54 | duna | i'm confused about what angeld want to do |
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| 19:22.33 | *** join/#asterisk _zoom_ (~Administr@41.218.59.37) |
| 19:23.09 | _zoom_ | hello, stopplaytones doesnt work am using asterisk 1.6 |
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| 19:45.19 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 20:05.27 | atan | Anyone here use one voip.ms account to route premium + value calls? |
| 20:09.19 | p3nguin | yep |
| 20:10.47 | benngard | yay, it works but is it the correct way to do it? exten => 9,n,Originate(OOH323/901@Avaya,app,SendFAX,/tmp/fax.tiff), Answer before and Hangup after, i do receive the fax |
| 20:11.42 | p3nguin | I don't know what you were working on, but you probably don't need to answer before you send a fax. |
| 20:11.50 | benngard | with t38 over ooh323 :) |
| 20:15.52 | benngard | what i am looking of is something like, a wants to fax b, a is a sip ata fax, b is a pstn fax, i have problems with the timing from a to b, so i think like this: a calls b, hits dialplan, ReceiveFAX( and then my Originate line, ofc need to fix the "9" and so on |
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| 20:21.53 | atan | p3nguin, you do? They told me I can't route NA calls through premium + value using one account. |
| 20:22.07 | atan | They said I need to open a second account. Not just a sub-account, but an entirely seperate account. |
| 20:22.37 | atan | I thought adding 044 would trigger the premium route for NA so 04419055551212 and such would route over premium but it's still defaulting back to value. |
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| 21:33.48 | p3nguin | atan: I don't see a reason that sub accounts wouldn't be able to be used. I can adjust my routes in my sub accounts. |
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| 21:49.27 | Janos | can anyone tell me where to find an answer to this question, what does "DEBUG[19599] chan_zap.c: Started VLDTMF digit '1'" means ? |
| 21:50.27 | raden | anyone know how to fix MOH issue with aastra phones in asterisk 1.8 ? |
| 21:52.31 | paulc | raden: still no joy with that one eh? Have you done a Wireshark trace of 1.6 vs 1.8 and seen what's different? |
| 22:03.07 | raden | paulc, no and unfortunately havenot had time that would be a good idea though |
| 22:03.32 | raden | when I hit hold on my phone SIP debug throws a few lines at asterisk but no hold info that i see |
| 22:17.43 | paulc | raden: if you want to pastebin that stuff, I'll take a look if you like |
| 22:18.04 | raden | give me one second lemee call myself and get a debug |
| 22:22.12 | raden | paulc, http://pastebin.com/i2mv5bCu |
| 22:22.25 | raden | thats from time hold is hit till hangup |
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| 22:24.40 | nightwalk | I have a call group set up via a variable that boils down to this, once expanded: exten => s,n,Dial(dahdi/g1&SIP/9,20,wW) |
| 22:24.59 | paulc | raden: hmm.. not sure.. I can do something similar with my gear here, but not till later on.. to test/compare.. |
| 22:25.15 | nightwalk | Is there anything obvious that's wrong with that? The dahdi connection rings just fine, but the SIP connection does not. |
| 22:26.03 | raden | paulc, much appreciated can i pm my email address so we can keep in better touch ? |
| 22:27.01 | paulc | sure |
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| 23:49.07 | p3nguin | nightwalk: Do you have a definition in sip.conf for a peer by the name of 9? Most people wouldn't. |
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| 23:50.20 | p3nguin | nightwalk: Also, do you want both the caller and the person being called to be able to start and stop the recording of the call? |
| 23:50.37 | p3nguin | nightwalk: Again, most people don't. |
| 23:53.10 | nightwalk | p3nguin: actually, I found the answer 15 minutes or so after I posted that. I had an error elsewhere that asterisk didn't bother to tell me about. Oops |
| 23:54.34 | nightwalk | is not really thrilled about asterisk 1.8's overly-permissive attitude :/ |
| 23:55.03 | p3nguin | To which aspects are you referring? |