| 00:01.38 | McL0VIN | lanning: still same issue |
| 00:02.10 | McL0VIN | lanning: note i do have ExtenWait() after the background() |
| 00:02.13 | *** join/#asterisk juanIMP (n=juan@201.244.45.216) |
| 00:02.21 | McL0VIN | http://pastebin.ca/1468188 |
| 00:03.09 | wdoekes | # |
| 00:03.12 | wdoekes | exten => Wait(2) |
| 00:03.34 | wdoekes | should be: exten => _XX.,n,Wait(2) |
| 00:04.18 | McL0VIN | wdoekes: what is the dif between WaitExten() and Wait() |
| 00:04.35 | lanning | wait just waits |
| 00:04.43 | McL0VIN | ah ok |
| 00:04.45 | ISO9001 | background will answer() if the call hasn't been already... but I'm not sure what happens with the wait(2) if the call hasn't been answered first. |
| 00:04.49 | lanning | waitexten waits for an extension to be dialed |
| 00:04.56 | ISO9001 | try answer(), wait(2), background(...) |
| 00:05.40 | wdoekes | (answer accepts milliseconds to wait as arg1, afaik) |
| 00:06.17 | ISO9001 | answer's arg is wait time before answering, not after. |
| 00:07.12 | wierdo | McL0VIN, waits for an ext to be dialed and could provide MoH while waiting |
| 00:07.12 | wdoekes | ah |
| 00:07.29 | McL0VIN | still, i am getting the same result and it just hangup http://pastebin.ca/1468191 |
| 00:07.52 | ISO9001 | WaitEten... |
| 00:08.03 | McL0VIN | errr |
| 00:08.09 | lanning | hahah |
| 00:08.19 | lanning | not a literal "n" |
| 00:08.36 | ISO9001 | lanning: literal n works fine. |
| 00:08.45 | wdoekes | McL0VIN: Answer() first |
| 00:09.52 | lanning | does "n" auto-increment? |
| 00:09.57 | wdoekes | yes |
| 00:10.08 | wdoekes | you need the 1 though |
| 00:10.23 | lanning | ah, cool, I will have to remember that... |
| 00:11.16 | McL0VIN | wdoekes: it should like like that thu, exten => _XX.,1,Answer() |
| 00:11.18 | McL0VIN | right? |
| 00:11.54 | lanning | yes |
| 00:12.51 | McL0VIN | ok its working, but still not that clear |
| 00:14.08 | McL0VIN | can i make it wait 1.5 secs |
| 00:14.24 | wierdo | yes you can |
| 00:15.22 | lanning | when a call is answered, it takes a little time for the media to be negotiated. so, you have to split up the answer and the playback, then insert a delay, to allow the negotiation. |
| 00:17.26 | rue_mohr | tzafrir_laptop, I still have -10dbm I dont know if the 1db difference is coinncodince or not |
| 00:18.02 | rue_mohr | I made the change, recompiled, installed and rebooted (just to make sure) |
| 00:19.35 | rue_mohr | .dtmf_high_level = 0, .dtmf_low_level = 0, |
| 00:19.56 | rue_mohr | hmm whats mfr... |
| 00:24.06 | rue_mohr | no even with that at 0 the 1mw is -10dbm |
| 00:31.51 | rue_mohr | I wonder if a signed/unsigned mismatch happening twice could cause this |
| 00:50.14 | McL0VIN | lanning: still not playing right |
| 01:22.12 | lanning | McL0VIN: still missing the first part of the recording? |
| 01:22.30 | McL0VIN | yes |
| 01:23.18 | lanning | can you pastebin your current dialplan? |
| 01:30.46 | McL0VIN | lanning: http://pastebin.ca/1468285 |
| 01:31.44 | lanning | change the Answer() to "n" from "1" for the priority. (you have two priority 1's) |
| 01:33.59 | carrar | and how long do you want to wait for a keypress (waitexten) |
| 01:35.39 | comfrey | 9[2-5]xxxxxxxxx should serve for 9 for outside line and then 9 digits... no? |
| 01:35.47 | comfrey | this is in polycom sip.cfg |
| 01:36.07 | comfrey | for some reason it does not wait for the last digit before it dials |
| 01:36.18 | carrar | no |
| 01:37.08 | McL0VIN | lanning: i gtg now , will come back later |
| 01:37.15 | *** part/#asterisk McL0VIN (n=chatzill@unaffiliated/dacs) |
| 01:37.53 | carrar | try 9,[2-9]xxxxxxxxx |
| 01:38.03 | carrar | outside numbers are 10 digits btw |
| 01:38.11 | carrar | 9,1[2-9]xxxxxxxxx |
| 01:40.40 | comfrey | thanks carrar, trying it now |
| 01:41.00 | carrar | the comma gives you the second dialtone (outside line effect) |
| 01:42.44 | comfrey | yes, it works very nicely. thanks |
| 01:55.31 | *** join/#asterisk Gnoll (n=Gnoll@ppp-71-3.98-62.inwind.it) |
| 01:55.36 | Gnoll | hi |
| 02:00.16 | Gnoll | i have these problem: |
| 02:00.16 | Gnoll | [Jun 21 03:58:29] WARNING[7507]: chan_sip.c:2994 create_addr: No such host: 7272 |
| 02:00.16 | Gnoll | [Jun 21 03:58:29] WARNING[7507]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 02:00.17 | Gnoll | [Jun 21 03:58:29] WARNING[7507]: app_voicemail.c:4011 leave_voicemail: No entry in voicemail config file for '2000' |
| 02:00.58 | Gnoll | can you help me please? |
| 02:02.52 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 02:02.56 | ruben23 | anyone have idea on this error i have on my asterisk CLI http://pastebin.com/m4d316022 |
| 02:10.15 | comfrey | Gnoll: what are you registering to in sip.cfg? |
| 02:10.31 | comfrey | seems like host does not exist ... maybe a typo |
| 02:11.00 | Gnoll | umh one moment |
| 02:11.21 | comfrey | ruben23: looks like an invalid destination in extensions.conf or somethiing |
| 02:12.31 | ruben23 | comfrey:yes sip trunk |
| 02:14.05 | Gnoll | i tried to redirect a call to another internal, now this problem: |
| 02:14.05 | Gnoll | [Jun 21 04:13:07] WARNING[8266]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 02:14.05 | Gnoll | [Jun 21 04:13:07] WARNING[8266]: app_voicemail.c:4011 leave_voicemail: No entry in voicemail config file for '2000' |
| 02:15.34 | Gnoll | the app_voicemail is normal (not setted) |
| 02:35.18 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 03:03.00 | eppigy | DONDE ESTA |
| 03:05.46 | jaytee | Que? |
| 03:05.51 | Nugget | ESTOY AQUI |
| 03:07.26 | Gnoll | please speak english, ( per favor hablate ingles) ... |
| 03:14.31 | eppigy | NO ME GUSTA |
| 03:14.57 | Gnoll | O_o |
| 03:15.24 | Gnoll | puede hablar espanhol en #asterisk-es |
| 03:20.13 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
| 03:27.40 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
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| 03:43.58 | *** join/#asterisk phurl (n=mdupont@82.114.75.252) |
| 03:50.12 | blaxthos | so i added a new extension today, polycom 501, trixbox/latest |
| 03:50.26 | blaxthos | can dial from it to any internal extension, or out a trunk no problem |
| 03:50.33 | blaxthos | but can't dial into that extension from anywhere |
| 04:12.10 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 04:16.00 | Micc | does voicemail have a callback feature? |
| 04:17.06 | Micc | aha, it does. just need to enable in voicemail.conf |
| 04:23.35 | blaxthos | <PROTECTED> |
| 04:23.35 | blaxthos | <PROTECTED> |
| 04:23.35 | blaxthos | <PROTECTED> |
| 04:23.35 | blaxthos | <PROTECTED> |
| 04:23.50 | blaxthos | wtf can't i csll this new extension ? |
| 04:29.01 | rue_mohr | how does a dummy_dahdi driver work, can I set up channels on it and use dahdi_monitor on them? |
| 04:31.28 | rue_mohr | ... configure it as a span |
| 04:32.06 | rue_mohr | how can I get a dummy_dahdi channel to make a call? |
| 04:45.23 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 04:45.55 | rue_mohr | heh, I get stuck on chan_dahdi.conf, I dont know how to specify which span the channels are for... |
| 04:46.38 | [TK]D-Fender | rue_mohr: You can't, its just a dummy timing source |
| 04:47.08 | [TK]D-Fender | rue_mohr: And chan_dahdi.conf is almost exactly the same as zapata.conf |
| 04:47.17 | [TK]D-Fender | rue_mohr: You never specify channels by span. |
| 04:47.18 | rue_mohr | oh, I cant trick the system into putting the 1mw to it and running dahdi_monitor against it? |
| 04:47.31 | rue_mohr | hmm, so much for that idea |
| 04:47.37 | [TK]D-Fender | rue_mohr: it depends what order multiple cards are ordered and what kind of card you're working with |
| 04:47.47 | rue_mohr | T100P |
| 04:48.11 | [TK]D-Fender | rue_mohr: You only have 1 span then, and your channels are 1-X depending on what you configured |
| 04:48.24 | rue_mohr | from the milliwatt app, to the t1 line there isn't much code, so this has to be easy to find |
| 04:48.56 | rue_mohr | I'm scrutinizing chan_dahdi.c but its context is beyond me |
| 04:51.30 | rob0 | Funny though, I get all kinds of calls from dummies. |
| 04:51.38 | rue_mohr | dont we all |
| 04:55.06 | rue_mohr | "Asterisk has detected a problem with your DAHDI configuration" = "load your dahdi_transcoder module" |
| 04:57.56 | rue_mohr | ~help |
| 04:58.37 | rue_mohr | infobot, Asterisk has detected a problem with your DAHDI configuration is "load your dahdi_transcoder module" |
| 04:58.37 | infobot | rue_mohr: that's too long |
| 04:58.39 | drmessano | thinks someones PBX has a butterface |
| 04:58.55 | rue_mohr | infobot, Asterisk has detected a problem is "load your dahdi_transcoder module" |
| 04:58.55 | infobot | okay, rue_mohr |
| 04:59.25 | [TK]D-Fender | ~asterisk |
| 04:59.25 | infobot | extra, extra, read all about it, asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 05:01.37 | rue_mohr | I really want to say the problem I found is part of codec_dahdi.c, but I cant. |
| 05:06.55 | rue_mohr | I know the ulaw.c code isn't he issue cause it didn't change |
| 05:07.01 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 05:07.42 | rue_mohr | the only changes to playtones seems to be that there was extra code added to deal differently with multi versus single tone generation |
| 05:08.47 | rue_mohr | I cant actaully find where the tone data is generated, and the codec_dahd.c looks ok, but seems to be the only piece in the middle to blame |
| 05:10.44 | rue_mohr | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg43484.html <-- OH HELLO |
| 05:11.32 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 05:12.34 | rue_mohr | hey its my -10db all over again! |
| 05:13.51 | rue_mohr | I'm _NOT_the only one! this is a real problem! |
| 05:15.09 | rue_mohr | my zaptel drivers that didn't ahve the problem were zaptel-1.2.13 I dont know if that post or predates his drivers |
| 05:15.38 | rue_mohr | so app_volume was ultimitly created cause of a bug |
| 05:15.46 | rue_mohr | that still isn't fixed |
| 05:17.10 | [TK]D-Fender | rue_mohr: Nope. App_volume was created because sometimes you're stuf with a provider whose gains are shit and you don't want to fubar all your equipment just to compensate for them. |
| 05:17.30 | [TK]D-Fender | stuck* |
| 05:17.33 | rue_mohr | to me that sounds like the email that origionated app_volume |
| 05:18.07 | drmessano | https://issues.asterisk.org/view.php?id=2023 |
| 05:18.07 | rue_mohr | he posted it cause he thought an 11db loss wasn't part of his phone equipment |
| 05:18.59 | [TK]D-Fender | rue_mohr: that e-mail is dated 2004. app voluem is an * 1.6 app |
| 05:20.00 | drmessano | Seems the issue was normalization between formats, per that bug report |
| 05:20.46 | rue_mohr | and I'm saying there is a 10db loss between milliwatt and my t1 line |
| 05:21.44 | rue_mohr | what do both situations have in common? |
| 05:22.26 | drmessano | a random number like 10db |
| 05:22.47 | rue_mohr | I measured between 11 and 9.8 |
| 05:23.00 | drmessano | because the bug report and subsequent comments show the difference that email reported wasnt 10db consistently |
| 05:24.03 | rue_mohr | this is driving me nuts, somewhere something broke between zaptel-1.2.13 and dahdi-linux-complete-2.1.0.4+2.1.0.2 that causes a 10->11db loss |
| 05:24.18 | rue_mohr | http://eds.dyndns.org/~ircjunk/asterisk/missing_11db.html |
| 05:24.41 | [TK]D-Fender | rue_mohr: Why are we not hearing about you testing 1.4? |
| 05:24.59 | rue_mohr | ? |
| 05:25.12 | rue_mohr | I upgraded them at the same time |
| 05:25.53 | rue_mohr | I wasn't looking for this, I didn't expect it |
| 05:26.37 | rue_mohr | I wanted to know what the dahdi_monitor level was for 0db, so I could work out why the tdm800 install seemed to be about 11db short |
| 05:27.16 | rue_mohr | I almost had it too! if, back in the day, I'd installed the zaptel-tools, I could have had it |
| 05:27.35 | rue_mohr | but I looked for a while and couldn't find the right version, so I did a completel upgrade |
| 05:28.01 | [TK]D-Fender | rue_mohr: Well if you're going to do a job, do it right. |
| 05:28.05 | [TK]D-Fender | rue_mohr: Be complete |
| 05:28.14 | [TK]D-Fender | on that note, its checkout time. Later all |
| 05:28.15 | rue_mohr | ? didn't I? |
| 05:28.24 | drmessano | Upgrade to 1.6 |
| 05:28.33 | *** join/#asterisk scooby2 (n=scooby2@pdpc/supporter/active/scooby2) |
| 05:28.49 | rue_mohr | so you suspect this is an issue with asterisk and not dahdi? |
| 05:28.51 | Qwell | rue_mohr: I've still never seen you say what your exact setup is |
| 05:29.00 | rue_mohr | http://eds.dyndns.org/~ircjunk/asterisk/missing_11db.html |
| 05:29.09 | drmessano | No but 1.4 with dahdi is a crutch, IMO |
| 05:29.09 | rue_mohr | what else would you like to know? |
| 05:29.14 | Qwell | That only describes the interfaces, not the call path |
| 05:29.24 | rue_mohr | ok |
| 05:29.27 | rue_mohr | just a sec |
| 05:29.45 | rue_mohr | ... would extensions.conf do you? |
| 05:30.28 | rue_mohr | or do you want to see dahdi show channel |
| 05:31.45 | drmessano | Honestly, 1.4 with dahdi scares me a bit. I've always felt 1.4 was like the quiet kid down the street who you knew would kill his parents one day, and I also feel like 1.6 with dahdi will always be a better match.. nobody likes a backport. |
| 05:32.19 | rue_mohr | ok, I'm not sure I'm really ready to completely rewrite extensions.conf, which is why I'm avoiding 1.6 |
| 05:32.27 | drmessano | Besides, 1.4 is 2 branches back now.. it's "very old", not just "OLD" |
| 05:32.37 | drmessano | Completely rewrite? |
| 05:32.47 | rue_mohr | iirc, its not at all compatible |
| 05:32.49 | drmessano | Have you bothered to check whats involved? No |
| 05:32.50 | rue_mohr | ok, let me post that |
| 05:32.55 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
| 05:32.57 | drmessano | No, youre not correct |
| 05:33.02 | rue_mohr | that would be good |
| 05:34.19 | rue_mohr | http://www.pastebin.ca/1468422 |
| 05:34.28 | rue_mohr | thats my extenstions.conf |
| 05:34.44 | rue_mohr | I'm trying to remember how to get call path data out of the console |
| 05:35.12 | rue_mohr | got it |
| 05:35.48 | rue_mohr | http://www.pastebin.ca/1468423 <-- call path information |
| 05:36.26 | rue_mohr | let me grab a bridged call to show you one with no loss |
| 05:37.35 | rue_mohr | http://www.pastebin.ca/1468425 <-- there is no loss in this call |
| 05:40.13 | rue_mohr | drmessano, Qwell ? |
| 05:41.37 | rue_mohr | ok so, if I drop in 1.6 and still have the 11db loss, then what? |
| 05:43.32 | rue_mohr | racks his brain on how to test the codecs |
| 05:43.37 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
| 05:44.56 | rue_mohr | there are two places the transcoding could be done, the dahdi codec or hte ulaw codec |
| 05:47.57 | *** join/#asterisk af_ (n=getsmart@88-149-230-49.dynamic.ngi.it) |
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| 05:54.34 | rue_mohr | ok so the transcoder must be shot, its gonna take me weeks to find, might as well work on some of my other projects for the night |
| 05:55.39 | rue_mohr | there are two linear to ulaw transcoders in there.... |
| 05:55.49 | rue_mohr | er codecs |
| 05:56.00 | rue_mohr | damn my obseviveness |
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| 06:00.29 | rue_mohr | can I do ${tech} where tech is DAHDI and channels are specified as ${tech}/3 etc? |
| 06:00.56 | rue_mohr | arg |
| 06:01.23 | rue_mohr | is leaving to try to obsess about one of his own projects for a while, one of you see if you can find the codec screwup |
| 06:01.25 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
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| 08:04.42 | thehar | doot doot |
| 08:04.45 | thehar | flashin phonessszzz |
| 08:43.40 | *** join/#asterisk r3c0n (n=recon102@212.28.233.34) |
| 08:43.53 | r3c0n | hello everyone |
| 08:50.28 | tzafrir_laptop | thehar, and what was that supposed to mean? |
| 08:52.03 | thehar | workin late |
| 08:56.49 | r3c0n | so im trying to get my hands wet with asterisk by implementing a solution for my home. I've got a telephone line that i would like to hook up to my asterisk and everytime i make an international call i would like for it to be done through my voip application on my pc rather than phone line. Otherwise all local area calls should go through asterisk. any guidance would be appreciated.. I'm assuming for the phone line i need 1xFXS and NxFXO card (depending o |
| 08:57.22 | r3c0n | Otherwise all local area calls should go through FXS |
| 08:58.00 | r3c0n | is this remotely possible with a small budget? |
| 09:04.19 | tzafrir_laptop | ~fxsfxo |
| 09:04.19 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 09:05.20 | tzafrir_laptop | r3c0n, define small. E.g. it is certainly possible with a tiny fraction of the national budget |
| 09:09.22 | coppice | to GM tiny was 30 odd billion to tide them over until their bankrupcy filing was complete |
| 09:10.56 | r3c0n | tzafrir_laptop, small meaning 1 phone line hooking up 3x FXO devices that the family uses.. |
| 09:11.57 | r3c0n | all my phones are obviouslly analogue devices and all thats required is that when a special *number is typed that the PBX sends the request to the resident voip application because that *number would designate someone attempting to make an international call |
| 09:12.27 | tzafrir_laptop | r3c0n, and the budget you had in mind was? |
| 09:13.34 | r3c0n | its my understanding that the FXS plug in wall would connect to the FXO plug in pbx and then the pbx must also have multiple FXS plugs to provide the analogue phonelines connectivity.. but in my case is an FXO gateway needed? |
| 09:14.25 | r3c0n | 500 bucks.. ive already got the analogue phones and a dedicated PC for exclusively running the asterisk software |
| 09:15.01 | r3c0n | so far and out of my lack of experience im guessing i just need 1 PCI card which has (1xFXO and 3xFXS) |
| 09:15.35 | tzafrir_laptop | you need one FXS port and one FXO port |
| 09:16.02 | r3c0n | hmm.. i was under the impression that each telephone device needs to be plugged into an FXS |
| 09:16.08 | tzafrir_laptop | 3 FXS ports will give you the ability to connect 3 phones at your house |
| 09:17.01 | r3c0n | here's how i understood it based on a basic course i took in school (telephony is not my major).. the plug in the wall or the FXS from phone company connects to the FXO in pbc and then pbx would act as a central system by connecting all telephones through its FXS ports |
| 09:17.13 | r3c0n | pbx* |
| 09:17.30 | r3c0n | yes that's what i need.. ive got 3 phone devices in my house.. |
| 09:17.36 | r3c0n | all analogue |
| 09:20.31 | r3c0n | Is a gateway required for my setup? |
| 09:21.41 | *** join/#asterisk angryuser_ (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 09:22.53 | *** part/#asterisk angryuser_ (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 09:26.35 | r3c0n | so pretty much this = FXS(from phone company) ---> FXO (asterisk machine) ---> 3x FXS (3x phone devices) |
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| 09:52.38 | ectospasm | r3c0n: theoritically, if you had two telephone circuits in your house you could have one be serviced by the telco, and connect one FXS connect to the other circuit |
| 09:53.00 | ectospasm | 3 phones is less than 4REN or whatever it's called (this is a late night guesstimate, so I may be off) |
| 10:05.05 | *** join/#asterisk Gnoll (n=Gnoll@ppp-71-3.98-62.inwind.it) |
| 10:06.48 | Gnoll | excuseme, for convert .wmv in .gsm i write this: sox file.wav -r 8000 -c 1 gsmfile.gsm resample -ql |
| 10:06.50 | Gnoll | it gives me back an error: sox soxio: Can't open output file `gsmfile.gsm': unknown file type `gsm' |
| 10:07.25 | Gnoll | how can i do? |
| 10:32.19 | tzafrir_laptop | Gnoll, I'm not sure sox knows .wmv |
| 10:32.28 | tzafrir_laptop | anyway, what Linux distribution do you use? |
| 10:32.41 | Gnoll | i'm use debian lenny and ubuntu 8.04 |
| 10:33.34 | tzafrir_laptop | on both: install either just libsox-fmt-gsm or libsox-fmt-all |
| 10:33.46 | Gnoll | ok one moment |
| 10:33.56 | tzafrir_laptop | but I suspect sox doesn't know wmv |
| 10:34.12 | tzafrir_laptop | maybe you need ffmpeg or something similar for that conversion |
| 10:35.34 | Gnoll | ok, its all right :D |
| 10:35.34 | ISO9001 | I doubt sox knows wmv... use something else to extract the audio, then use sox to mangle it. |
| 10:35.41 | Gnoll | thanks :D |
| 10:37.05 | *** join/#asterisk joobie (n=joobie@124-168-12-192.dyn.iinet.net.au) |
| 10:41.25 | coppice | you need something on windows or something that ignores patent issues to handle wmv |
| 10:42.25 | Gnoll | umh, the file extist, but i always see the same written on the monitor: [Jun 21 12:38:59] WARNING[32428]: pbx.c:5725 pbx_builtin_background: ast_streamfile failed on SIP/83.211.227.23-0819e3e0 for aaa |
| 10:47.32 | joobie | guys anyone know how to reset the stats stored in asterisk AMI for queues? stats seem to sit there for ages |
| 10:47.45 | joobie | want to like null them once a day or something |
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| 10:53.07 | tompaw | hi |
| 10:53.24 | tompaw | is the information at http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql up to date? |
| 10:53.30 | tompaw | it says it's for 1.2 |
| 10:53.41 | tompaw | especially regarding the database structure |
| 10:54.22 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 10:57.44 | barbacha | tompaw: looks like the table I have for 1.4 |
| 10:57.52 | tompaw | barbacha: thx |
| 11:02.58 | joobie | argh |
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| 11:03.01 | joobie | guys |
| 11:03.14 | joobie | cant seem to find a way to null the Queue stats in 1.4 |
| 11:03.22 | joobie | like abandonded calls for example |
| 11:03.24 | joobie | anyonek now how? |
| 11:04.56 | joobie | a reload does it, but seems a bit much just to reset the queue stats |
| 11:08.21 | Gnoll | bye |
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| 13:39.01 | henk | hi i have a question about caller ids. i have a asterisk box that registers with two voip providers. an incoming call on one initiates a call over the other. i'd like to set the callerid of that call to the number that's calling asterisk. is that possible? |
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| 13:47.34 | [TK]D-Fender | henk: depends if your providers LET you set callerid. If so and you've set your peers up not to interfere with that it will "just work" |
| 13:49.34 | henk | [TK]D-Fender: any quick way to find out if my provider lets me? apart from asking him. did i get it right, that both the incomin calls callerid and the outgoing ones are in the same variable? |
| 13:51.02 | [TK]D-Fender | henk: when you call out its the callerID of the channel that is used if configured to no override it |
| 13:51.31 | henk | when i read ${CALLERID(all)} i get the incoming and this http://www.voip-info.org/wiki/view/Setting+Callerid says to set the same variable. |
| 13:51.52 | henk | [TK]D-Fender: i can't remember telling it to override. where would i do that? sip.conf? |
| 13:52.29 | [TK]D-Fender | henk: make sure to set "sendrpid=yes" and "trustrpid=yes" |
| 13:52.57 | [TK]D-Fender | henk: And YES you should get off your ask and actually confirm that your provider supports letting you set your own CID |
| 13:54.10 | henk | :-p |
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| 14:00.38 | ruben23 | hi |
| 14:14.40 | *** join/#asterisk ruben23 (n=RPL@203.177.91.193) |
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| 14:27.34 | [TK]D-Fender | BBIAB |
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| 16:01.06 | McL0VIN | Good morning all |
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| 16:22.14 | ruben23 | <PROTECTED> |
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| 16:58.00 | McL0VIN | i guess i need a free multi line softphone |
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| 17:20.06 | FreakGuard | how to setup a forwarding depending on daytime? |
| 17:20.37 | eppigy | ~book |
| 17:20.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:21.17 | FreakGuard | eppigy: I suppose that's an rtfm for me. |
| 17:22.39 | eppigy | yes indeed |
| 17:23.53 | FreakGuard | eppigy: oke. just wanted to know if it's possible ;) |
| 17:24.40 | FreakGuard | thanks |
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| 18:54.59 | Jumpie | sup guys |
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| 19:25.48 | Jumpie | wow is every asterisk channel on this network totally idle now? |
| 19:31.47 | carrar | WHAT |
| 19:32.38 | Jumpie | lol |
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| 19:42.33 | KavanS | yes |
| 19:42.41 | KavanS | I'm afk atm |
| 19:42.54 | KavanS | (I was afk while typing that, that's how pro I am) |
| 19:43.28 | carrar | residual key presses? |
| 19:44.12 | engien | idle timer |
| 19:44.59 | KavanS | no, both wrong |
| 19:45.04 | KavanS | telekenetic powers |
| 19:45.15 | KavanS | I got them from playing WoW |
| 19:45.38 | Jumpie | heh |
| 19:45.51 | *** part/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust923.oxfd.cable.ntl.com) |
| 19:46.03 | Jumpie | hmm is there some special way to signal my server that im trying to log in with my phone? |
| 19:46.16 | Jumpie | the extension is made, the new firmware is supposedly loaded to the phone, but i know its offline |
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| 20:15.05 | KavanS | damn it, just found out that I got a counterfeit pap2 adapter |
| 20:16.45 | drmessano | counterfeit? |
| 20:18.44 | KavanS | http://text.broadbandreports.com/forum/r21331513-Equipment-Counterfeit-Linksys-PAP2 |
| 20:18.50 | KavanS | been troubleshooting some issues with a sip adapter |
| 20:19.21 | KavanS | and mac address doesn't match outside, sometimes goes in a funk...I got it from a company down in California, but they might be getting from China |
| 20:30.58 | ISO9001 | :/ |
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| 22:09.46 | kn0x | does anyone have the default Cisco 7940 xml configs? |
| 22:16.39 | carrar | 7940's don't use XML |
| 22:16.50 | carrar | for their sip co nfig |
| 22:26.21 | kn0x | but my tftp logs see it going for SEP[MAC].cnf.xml |
| 22:26.28 | kn0x | I used to have .cnf files |
| 22:26.42 | kn0x | im guessing new firmware version? |
| 22:27.27 | kn0x | carrar: are you including version 7.X |
| 22:30.56 | carrar | .cnf are not XML |
| 22:31.17 | carrar | all Cisco 7940 SIP versions are just a plain text file |
| 22:31.44 | carrar | same for 7960 |
| 22:31.53 | Chainsaw | SEP prefix would suggest the phone has had an SCCP load, at any rate. |
| 22:32.06 | Chainsaw | (SIP firmware will request SIP[MAC].cnf.xml instead) |
| 22:32.15 | carrar | yeah load SIP on it |
| 22:36.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:37.21 | carrar | even SIP 7940's will look for a SEP file |
| 22:37.46 | carrar | newer SIP firmware versions |
| 22:38.18 | carrar | as well as a CTLSEPMAC.tlv file |
| 22:39.16 | carrar | Settings, status, firmware versions on the menu |
| 23:01.55 | kn0x | Chainsaw / carrar do I just rename the .cnf to .cnf.xml ? |
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| 23:03.26 | kn0x | carrar: are you saying .cnf.xml are actually just .cnf files? |
| 23:17.57 | kn0x | does anyone have cisco SIP firmware? |
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