| 00:08.27 | p3nguin | How can I check to see if a call between two devices has been reinvited? |
| 00:13.37 | etfonhomey | p3nguin, look at the channel details from the CLI maybe? |
| 00:14.43 | p3nguin | Are you guessing, or do you actually expect that information to be there? |
| 00:15.01 | Katty | peeks in |
| 00:15.10 | etfonhomey | p3nguin, reinvited would have 2 channels, probably. Channe1 = station1 -> * and Channel2 = * -> station2 (canreinvite=no) OR station1 -> station2 (canreinvite=yes) |
| 00:15.39 | etfonhomey | p3nguin, I would be surprised if the endpoint IP's are not in the channel details. |
| 00:15.42 | p3nguin | Alright, that looks like useful information. Let me look at a call and see if it compares to yours. |
| 00:16.04 | *** join/#asterisk albertoandrade (n=albertoa@189.101.117.92) |
| 00:16.28 | etfonhomey | p3nguin, sorry, I said that wrong. A reinvited call should only have 1 channel. Sorry. |
| 00:16.36 | Katty | ello. |
| 00:17.28 | etfonhomey | p3nguin, I'll test as well. |
| 00:17.47 | p3nguin | cool |
| 00:20.00 | p3nguin | I'm seeing two channels active, so I guess there's no reinvite on that call. |
| 00:21.14 | etfonhomey | p3nguin, I have canreinvite=no and I have 2 channels as well. I'm gonna change it to canreinvite=yes and try again. One sec... |
| 00:23.24 | *** join/#asterisk titter` (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 00:23.52 | titter` | say hi, and cable modem reboots ... odd! |
| 00:24.12 | Katty | hai der |
| 00:24.17 | p3nguin | sip show channel <my channel> reveals that the "audio IP" is that of my * system, so that indicates to me that there is no reinvite. |
| 00:27.50 | p3nguin | Does that sound right? |
| 00:27.53 | *** join/#asterisk cjp (i=5f4ac434@gateway/web/freenode/x-kjulknscvipnhahl) |
| 00:28.07 | cjp | back |
| 00:28.12 | Katty | anyone know about abc news? |
| 00:28.20 | Katty | is it like fox news or.... bbc |
| 00:28.28 | *** join/#asterisk coppice (n=chatzill@234.157.17.210.dyn.pacific.net.hk) |
| 00:28.37 | k-man | Katty: which country? |
| 00:28.42 | Katty | america. |
| 00:29.16 | etfonhomey | p3nguin, I believe if you do a "core show channel _____" and look at the Direct Bridge and Indirect Bridge values. The values were the other endpoint IP's when I had canreinvite=yes. Checking it now with canreinvite=no. |
| 00:29.41 | p3nguin | I would expect that ABC is going to be just like NBC, CBS, and Fox. |
| 00:29.56 | Katty | :< |
| 00:30.02 | Katty | what about CNN? |
| 00:30.56 | p3nguin | I'm not sure. What info are you trying to gain about the news services? |
| 00:31.06 | Katty | whether it's liberal or conservative |
| 00:31.48 | p3nguin | They're all biased and should not be trusted. |
| 00:31.58 | cjp | no comments out there on my question from before? |
| 00:32.35 | Katty | ^_- |
| 00:33.20 | ChannelZ | pretty much all of them are liberal except fox news which is both but probably leans more right |
| 00:33.33 | Katty | i can't stand fox news. |
| 00:33.51 | etfonhomey | p3nguin, I found it. |
| 00:33.56 | p3nguin | do tell |
| 00:34.32 | k-man | cjp: no, no one answered your question while you were gone |
| 00:34.33 | etfonhomey | p3nguin, during an active call, look at one of the channels for one of your endpoints and look at the "Audio IP" value. If canreinvite=yes , that value should be your other endpoint. |
| 00:34.50 | ChannelZ | well then maybe you should watch MSNBC and read The New York Times |
| 00:34.56 | k-man | cjp: you should consider writing to the mailing lists also |
| 00:35.01 | ChannelZ | Obama buttkiss, all the time |
| 00:35.40 | cjp | using a .call file to connect Channel: Local/1005@ext-internal with Context:ext-id Extension:7000, extension looks like this: exten => 1005,1,Dial(${TRUNKTYPE1}${TRUNKNAME1}/${PHONENUMBER1}&Skype/1005|30), I need to identify which of the 2 Dialled parties picks up the call and pass it on to Extension in an agi_ for example |
| 00:36.38 | Katty | ChannelZ: i'm just tired of hearing Terrorism every other word, Michael Jackson, and everyone's interjected opinons. |
| 00:37.06 | Katty | ChannelZ: and they twist the crap out of things. |
| 00:37.55 | p3nguin | etfonhomey: That is consistent with my assumption mentioned earlier. |
| 00:38.15 | dlynes_laptop | Katty, people are saying michael jackson is a terrorist now? |
| 00:38.17 | p3nguin | etfonhomey: On the other hand, I can't figure out why my devices don't want to reinvite. |
| 00:38.22 | Katty | dlynes_laptop: no. |
| 00:38.51 | ChannelZ | Whose still talking about michael jackson? |
| 00:38.51 | dlynes_laptop | Katty, oh...nvm. |
| 00:39.01 | dlynes_laptop | Katty, the completely unbiased opinions of fox news reporters |
| 00:39.16 | Katty | heh |
| 00:39.33 | Katty | it's terrible...i don't get why it's the #1 watched news in america. |
| 00:39.50 | ChannelZ | There's also a difference between hard news and news analysis shows, on any network |
| 00:39.58 | Katty | yes. |
| 00:40.00 | dlynes_laptop | Whenever I want to see what's on the latest american propogandist podium, I tune into fox news |
| 00:40.03 | Katty | in news analysis it's expected |
| 00:40.12 | Katty | i don't need their opinions when they're delivering Events |
| 00:40.25 | p3nguin | cjp: The BRIDGEPEER variable value could be of some use. |
| 00:40.31 | dlynes_laptop | fox news is right up there with the national enquirer |
| 00:40.43 | Katty | agreed. |
| 00:40.48 | ChannelZ | That's just silly |
| 00:40.54 | coppice | Fox is so blatant it should be ineffective. Strange that its still watched. Most highly biased news sources are more subtle |
| 00:40.59 | dlynes_laptop | ChannelZ, are you kidding? |
| 00:41.09 | ChannelZ | although if you want to pander in comparisons like that, didn't National Enquirer get the John Edwards story right? |
| 00:41.10 | dlynes_laptop | ChannelZ, fox isn't news...it's pure propoganda |
| 00:41.24 | etfonhomey | p3nguin, are your devices on the same subnet? |
| 00:41.30 | p3nguin | etfonhomey: yes |
| 00:41.35 | dlynes_laptop | erm propaganda, even |
| 00:41.45 | p3nguin | etfonhomey: same physical switch, even. |
| 00:42.06 | Katty | fox news had everyone up in arms about terrorism |
| 00:42.14 | Katty | every other word out of their cast was terror this and terror that |
| 00:42.16 | Katty | and omgomgomg |
| 00:42.32 | *** join/#asterisk Truenos (i=Truenos@dont.mind.if.im.in.ur.ignorelist.com) |
| 00:42.33 | Katty | but that's what they wanted. |
| 00:42.47 | Katty | don't get me started on fox news. |
| 00:42.48 | p3nguin | etfonhomey: The only thing I can think of is that the w and W Dial() options break reinvites in the same way that t and T do. |
| 00:42.51 | coppice | Katty: the real terrorist would be those trying to terrorise the population |
| 00:43.07 | Katty | coppice: they take the cake for that. |
| 00:43.36 | etfonhomey | p3nguin, I can test it on my end. What do the w and W options do? I can't remember off the top of my head. |
| 00:43.37 | dlynes_laptop | Katty, does dubbaya own fox news? |
| 00:43.40 | ChannelZ | Well I guess if you don't want opinion, turn off the TV, because even PBS doesn't get it right |
| 00:43.45 | coppice | Katty: making people scared is the basic strategy for anyone trying to control people. |
| 00:43.50 | p3nguin | etfonhomey: They are used for automon. |
| 00:44.04 | Katty | i'm not sure who owns fox news. |
| 00:44.04 | ChannelZ | coppice: like "We need healthcare reform YESTERDAY or the world is going to end"? |
| 00:44.10 | Katty | actually isn't it mister murdock? |
| 00:44.40 | Katty | yeah rupert murdock |
| 00:44.41 | dlynes_laptop | ChannelZ, How about your buddy that wants to bring back in the depression era banking system that clinton abolished in 1988(?) |
| 00:44.51 | p3nguin | etfonhomey: I use them so I can press *1 to start recording during a call. I could take them out and retest to see if that changes anything. |
| 00:44.58 | coppice | ChannelZ: the only thing that don't treat as an emergency are the real emergencies |
| 00:45.02 | ChannelZ | What buddy is that? |
| 00:45.11 | dlynes_laptop | ChannelZ, Obama |
| 00:45.15 | Katty | watches some natgeo |
| 00:45.22 | etfonhomey | p3nguin, Does automon record the files on the * filesystem? |
| 00:45.30 | ChannelZ | He's not my buddy |
| 00:45.32 | p3nguin | etfonhomey: yep |
| 00:45.47 | etfonhomey | p3nguin, if so, I could definitely see it breaking the reinvite. * would need to stay in the audio path in order to record the audio. |
| 00:45.51 | dlynes_laptop | ChannelZ, i kinda figured that...already had you pegged for a die in the boots republican :) |
| 00:45.53 | Katty | why are we talking about Obama bringing back teh dperession era? |
| 00:46.01 | Katty | when it was mister bush who got us into this situation |
| 00:46.02 | ChannelZ | I'm not that either. |
| 00:46.22 | dlynes_laptop | Katty, clinton was actually the one that removed the controls from the banks |
| 00:46.42 | dlynes_laptop | Katty, the controls that stated investment banks and commercial banks cannot be one and the same |
| 00:46.50 | Katty | i wasn't actually refering to subprime loans |
| 00:46.59 | Katty | but that's another major problem. |
| 00:47.02 | coppice | Katty: Bush did plenty of bad things, but the lack of regulation that has caused the greater magnitude of economic swings predates him |
| 00:47.07 | p3nguin | etfonhomey: I guess I or someone else needs to make a list of which Dial() options can be used and reinvites still work. A quick reference would have been useful right now. |
| 00:47.17 | dlynes_laptop | Katty, nah...this is more the banks pissing away money on risky hedge funds |
| 00:47.28 | dlynes_laptop | Katty, i.e. corporate greed |
| 00:47.34 | Katty | so how is that Obama's fault? |
| 00:47.44 | Katty | i'm not following |
| 00:47.48 | dlynes_laptop | Katty, it's not...Obama wants to reinstate the laws that prevented that |
| 00:47.50 | ChannelZ | Apparently nothing is his fault anyway |
| 00:48.07 | titter` | It isn't his fault, it's how he is handling the situation |
| 00:48.08 | dlynes_laptop | Katty, Clinton axed the laws that allowed that |
| 00:48.24 | Katty | he just inherited a bad situation |
| 00:48.46 | dlynes_laptop | Katty, http://blogs.villagevoice.com/runninscared/archives/2010/01/obamas_new_bank.php |
| 00:49.01 | coppice | dlynes_laptop: there were a series of stupid deregulation steps going back to the 80s. Remember the Savings and Loans? |
| 00:49.08 | Katty | ChannelZ: i think the country is just thrilled to not have bush in office ;) |
| 00:49.16 | dlynes_laptop | coppice, yeah...I remember the savings and loan crisis of the 80's |
| 00:49.35 | dlynes_laptop | coppice, I can't even remember how many banks and savings and loan companies went tits up then |
| 00:49.35 | titter` | Katty: I would argue more are less thrilled to have Obama in the office |
| 00:49.41 | p3nguin | I remember a lot of things in the '80s. |
| 00:49.54 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.35.31) |
| 00:49.59 | p3nguin | Like gasoline being 88 cents per gallon. |
| 00:50.04 | Katty | titter`: why do you say that? |
| 00:50.06 | ChannelZ | All politicians are pretty much a wreck |
| 00:50.14 | titter` | I was born in the 80's *shrugs* |
| 00:50.14 | coppice | dlynes_laptop: well, that monumental success with deregulation inspired all the later ones |
| 00:50.26 | p3nguin | not the '80s? |
| 00:50.37 | ChannelZ | Obama is just another crap politician in a long line of crap politicians |
| 00:50.52 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 00:50.56 | Katty | way to be mister negativity |
| 00:50.59 | dlynes_laptop | ChannelZ, we've got them going all the way back to diefenbaker in Canada :) |
| 00:51.14 | titter` | Things haven |
| 00:51.24 | titter` | Things haven't improved at all |
| 00:51.27 | p3nguin | All politicians are crap. Some are just less crappy than others. |
| 00:51.28 | titter` | They have gotten worse |
| 00:51.40 | dlynes_laptop | titter`, i don't know about that |
| 00:51.43 | Katty | titter`: rome wasn't built in a day |
| 00:51.52 | Katty | titter`: we're not going to have a magically recovery over night |
| 00:52.02 | Katty | titter`: it will take several years probably |
| 00:52.09 | dlynes_laptop | titter`, the headlines the last couple of years are they same as they were in the mid 90's, the same as they were in the early 80's, the same as they were in the early 30's, ... |
| 00:52.26 | dlynes_laptop | titter`, history repeats itself, and nobody seems to remember |
| 00:52.27 | titter` | December sales down, unemployement rises again |
| 00:52.35 | titter` | Same old same old |
| 00:52.46 | Katty | well it hits closer to home for me |
| 00:52.49 | titter` | http://finance.yahoo.com/news/Strategic-Defaults-and-the-usnews-2190373684.html?x=0&mod=loans |
| 00:52.51 | Katty | my dad's been unemployeed for over a year now |
| 00:52.55 | dlynes_laptop | so on that note, everyone should just have a beer and relax |
| 00:52.57 | p3nguin | I remember news in the '90s, the '80s, and I wasn't alive in the '30s. |
| 00:53.01 | coppice | titter`: this is normal. the main things affecting the average person happen as an economy is climing out of recession. the lag is something like 2 years |
| 00:53.46 | dlynes_laptop | Katty, that's the exact reason I became a contractor...so I don't need to depend on someone else for a living |
| 00:53.48 | titter | I am more worried about the about the effects of inflation that we will see in the next 2 years |
| 00:54.10 | Katty | dlynes_laptop: my dad worked for GM. |
| 00:54.16 | Katty | dlynes_laptop: for nearly 30 years. |
| 00:54.17 | ChannelZ | to moving from one toxic topic to a completely different one, does silicon caulk (like for bathrooms) interact badly with latex (like gloves)? |
| 00:54.27 | Katty | dlynes_laptop: he was 6 months away from retiring :/ |
| 00:54.57 | p3nguin | channelz: it should not. |
| 00:55.30 | titter | There is too much debt and too much money printed in the past year for inflation to not be a problem in the near future. |
| 00:55.34 | dlynes_laptop | Katty, eww...sorry to hear that |
| 00:55.37 | p3nguin | At most, it might deteriorate the latex. |
| 00:55.52 | ChannelZ | I need to do an emergency patch job on a shower drain |
| 00:56.43 | Katty | i'll ask ryan |
| 00:56.51 | dlynes_laptop | ChannelZ, and why would latex and silicone interaction matter to you? |
| 00:56.52 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 00:57.05 | dlynes_laptop | ChannelZ, are you going to line it with latex before siliconing it? |
| 00:57.35 | Katty | ChannelZ: ryan says latex will be fine |
| 00:57.44 | ChannelZ | no just wondering if I could wear some disposable gloves without some chemical reaction fusing to my fingers permenantly or setting on fire or something |
| 00:58.02 | Katty | ChannelZ: you suggested using something to scoop out the excess tho |
| 00:58.04 | ChannelZ | I need to push the stuff into a place I can't easily see |
| 00:58.08 | Katty | ChannelZ: s/you/he/ |
| 00:58.17 | dlynes_laptop | ChannelZ, nah...silicone's pretty inert...just rub it off your hands after you're finished |
| 00:58.19 | p3nguin | Heck, I wouldn't even bother. If you get some on your fingers, wipe it off or wait until it dries and peel it off. |
| 00:58.33 | ChannelZ | I just don't want to be picking it out of my fingernails for 3 days |
| 00:58.43 | dlynes_laptop | ChannelZ, i've never used gloves when using silicone |
| 00:59.20 | dlynes_laptop | ChannelZ, nah...just use some of the cleaner that mechanics use...the stuff with pumice in it |
| 00:59.31 | ChannelZ | lava |
| 00:59.32 | p3nguin | RTV silicone is much stickier. |
| 00:59.33 | dlynes_laptop | ChannelZ, or orange clean works too (you can buy it at home depot) |
| 01:00.02 | dlynes_laptop | p3nguin, rtv? |
| 01:03.12 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 01:04.00 | p3nguin | RTV silicone seems to be more sticky than things like acrylic latex caulk or some other type. I'm guessing most of your "at home" silicones are going to be RTV. |
| 01:06.15 | p3nguin | I still wouldn't bother with gloves if I had any doubt there could be chemical reaction. Gloves are just going to be a convenience to keep your fingers clean rather than having to use cleaner afterwards. |
| 01:06.44 | titter | I've never used gloves with silicone |
| 01:12.21 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:25.02 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:26.35 | *** join/#asterisk andrew` (n=andrew@70-36-140-59.dsl.dynamic.sonic.net) |
| 01:27.47 | andrew` | hi, could anyone suggest any of the best currenttly available SIP phones for my friend's small business...I'm thinking of maybe something like the SPA-841 from a few years ago...nothing too fancy, but decent quality |
| 01:29.56 | k-man | andrew`: i have an spa942 thats pretty nice imho |
| 01:34.16 | andrew` | looks pretty decent, thank you |
| 01:38.38 | andrew` | anyone else have any suggestions for comparison? |
| 01:41.11 | k-man | i think the 942 might be discontinnued and there is a newer equivalent in its place now, not sure |
| 01:44.03 | fenrus | what about SPA 504G ? |
| 01:48.52 | *** join/#asterisk jhirley (n=jhirley@adsl-3-194-220.mia.bellsouth.net) |
| 01:54.48 | andrew` | hmm it looks very similar |
| 01:57.42 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
| 02:11.04 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:15.26 | *** join/#asterisk jhirley_ (n=jhirley@adsl-3-67-180.mia.bellsouth.net) |
| 02:15.58 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:18.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:22.14 | *** join/#asterisk timholum_ (n=chatzill@168.sub-97-241-58.myvzw.com) |
| 02:29.14 | random_mike | can anyone advise how to change the Contact header in a registration packet from Asterisk to another SIP server? |
| 02:30.29 | p3nguin | What is your end goal? |
| 02:32.29 | random_mike | to remove the s@123.123.123.123 as the contact :) |
| 02:33.00 | random_mike | alter it to have username@123.123.123.123 |
| 02:33.17 | hardwire | check out "fromuser" in your sip peer configuration |
| 02:33.33 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:33.51 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 02:34.28 | random_mike | hardwire, currently in sip.conf I have register => username:password@sipprovider.com |
| 02:34.32 | random_mike | is this what you refer to? |
| 02:34.36 | hardwire | no |
| 02:34.51 | hardwire | check out the sip.conf example that comes with the asterisk source |
| 02:35.01 | hardwire | search for fromuser.. it applies to user/peer/friends. |
| 02:35.10 | random_mike | ok |
| 02:35.11 | hardwire | well.. peers. |
| 02:38.32 | random_mike | oh this isnt for outbound calls |
| 02:38.43 | random_mike | just the registration when astrisk acts like a sip client |
| 02:40.06 | hardwire | r u doing it wrong? |
| 02:41.00 | p3nguin | Is 123.123.123.123 your IP address or your ITSP's? |
| 02:52.02 | *** join/#asterisk gnufan (n=hardev@71-93-139-56.dhcp.bbcy.ca.charter.com) |
| 02:52.48 | gnufan | have you guys seen this before? WARNING[4505]: chan_skinny.c:6315 get_input: Skinny Client sent less data than expected. |
| 02:52.58 | p3nguin | I have. |
| 02:53.11 | gnufan | any fix? |
| 02:53.24 | p3nguin | Yes. Convert to a SIP image on your phone. |
| 02:54.06 | gnufan | actually im trying to dial out from sugarcrm application using VoiceRD |
| 02:54.21 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:56.01 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-40-42.acanac.net) |
| 02:57.27 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 02:57.46 | gnufan | anyone has experience integrating Asterisk and SugarCRM using VoiceRD? |
| 02:58.36 | ChannelZ | hardwire: his problem is the remote side isn't giving the extension in the INVITE, it's putting 's' in and putting the extension in the To: field |
| 02:59.04 | random_mike | p3nguin, what is an ITSP ? |
| 02:59.16 | p3nguin | I solved that by putting /myphonenumber on the end of the registration statement. |
| 02:59.31 | p3nguin | ~itsp |
| 02:59.32 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 02:59.34 | voipmonk | internet telephone service provider |
| 03:00.03 | ChannelZ | p3nguin: yeah but that just sends everything to one extension |
| 03:00.18 | p3nguin | And 's' isn't just one extension? |
| 03:00.28 | random_mike | yea p3nguin just got that too |
| 03:01.16 | p3nguin | I'm just saying how I solved the problem of the calls being delivered to 's' when they should have been delivered to my phone number. |
| 03:01.27 | ChannelZ | yeah, but if I call SIP/you/666 you might want it to go to 666. |
| 03:02.05 | p3nguin | That's a completely different matter. |
| 03:02.34 | gnufan | What is default port for asterisk 1.6? is it 2000? |
| 03:02.50 | ChannelZ | Port for what? |
| 03:02.50 | p3nguin | SCCP's port is 2000. |
| 03:03.41 | p3nguin | Adding the /phonenumber onto the register statement changed the fact that it was "Looking for 's' in context" to now be "Looking for 5551212 in context" |
| 03:03.50 | p3nguin | I thought that is what you said there was a problem with. |
| 03:04.18 | gnufan | K. if a client is asking for asterisk port #, what should i put? |
| 03:04.34 | ChannelZ | re: "his problem is the remote side isn't giving the extension in the INVITE, it's putting 's' in and putting the extension in the To: field" |
| 03:05.09 | ChannelZ | gnufan: there is no 'port for asterisk'. SIP is usually 5060 |
| 03:05.29 | ChannelZ | IAX is usually 4569 |
| 03:05.49 | ChannelZ | RTP uses a whole range.. so the question is a little too vague |
| 03:06.16 | gnufan | ChannelZ: 5038 is used for what? |
| 03:06.33 | ChannelZ | the * manager interface |
| 03:07.21 | gnufan | Thanks ChannelZ |
| 03:09.08 | *** part/#asterisk Truenos (i=Truenos@unaffiliated/truenos) |
| 03:21.15 | p3nguin | Actually, IAX is port 5036. |
| 03:21.27 | p3nguin | IAX2, which we currently use, is what is on 4569. |
| 03:23.38 | random_mike | is it possible to use regex within extensions.conf ? |
| 03:23.42 | random_mike | for example: |
| 03:23.46 | random_mike | exten => atas,n,NoOp(Dialed Number = ${SIP_HEADER(To)}) |
| 03:24.08 | random_mike | [Jan 25 13:52:06] VERBOSE[26166] logger.c: -- Executing [atas@atas:2] NoOp("SIP/sipdev-00000001", "Dialed Number = *123 <sip:*123@xxx.xxx.xxx.xxx:5060>") in new stack |
| 03:24.29 | random_mike | i'm just after the *123 not the <sip:*123@xxx.xxx.xxx.xxx:5060> |
| 03:26.19 | ChannelZ | not that I know of |
| 03:27.01 | *** join/#asterisk youngproguru (n=youngpro@cpe-76-180-188-78.buffalo.res.rr.com) |
| 03:28.18 | random_mike | :( |
| 03:30.14 | ChannelZ | you could do it externally with an AGI or something |
| 03:30.27 | ChannelZ | but maybe you could just get an ITSP that isn't crazy? |
| 03:30.32 | gnufan | What does this mean? ERROR[4547]: utils.c:1175 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 03:30.40 | random_mike | ChannelZ, if only! |
| 03:32.08 | p3nguin | random_mike: What do you want to change? What is the problem? Stop being cryptic and start explaining in detail what you need to do. |
| 03:32.21 | ChannelZ | He did. |
| 03:32.23 | *** join/#asterisk aceking5 (n=aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 03:33.57 | ChannelZ | His ITSP is sending calls to 's' but shoving the 'real' extension in the SIP To: header. He's trying to get at the 'real' extension. This means parsing ${SIP_HEADER(To)} |
| 03:34.34 | random_mike | basically ^ |
| 03:34.54 | random_mike | p3nguin, basically the SIP_HEADER(To) is reporting as "*123 <sip:*123@xxx.xxx.xxx.xxx:5060>" but all I want to do is pull the "*123" off the front of the header. |
| 03:35.03 | ChannelZ | What is this perl piece of vomit they are running again? |
| 03:35.17 | random_mike | in perl I could use regex as suggested by ChannelZ, but I'd rather not break into AGI if I coudl help it |
| 03:35.36 | random_mike | http://www.iagu.net/products.html |
| 03:35.59 | random_mike | We run Slipper - by IAGU |
| 03:36.17 | p3nguin | What is the purpose of pulling "the *123 off the front of the header?" What will you do with it once you "pull" it? |
| 03:37.02 | random_mike | I can use a gotoif based on the dialed number (as opposed to the whole header) to make for a nicer extensions.conf |
| 03:37.51 | p3nguin | If you just make them send you the dialed number, you can use the dialed number as an extension and stop trying to rig it. |
| 03:38.16 | p3nguin | Did you even TRY what I said? |
| 03:38.20 | random_mike | how would I do that p3nguin when I have more than 1 number assigned to that username? |
| 03:39.21 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 03:39.22 | p3nguin | That I do not know. I don't see anywhere that you said you have more than one DID with that ITSP. |
| 03:39.50 | random_mike | Thank I am sorry for not explaining myself better. |
| 03:40.53 | ChannelZ | In general an ITSP is going to send all calls to a single extension to you.. usually that's your DID number |
| 03:41.33 | p3nguin | I guess I am lucky enough that my ITSP doesn't make my figure out how to do it. They just send the number with every call. |
| 03:42.56 | ChannelZ | random_mike: lets assume that everything is working how it's supposed to. Where would *123 be coming from? Is that something static you're trying to tell your ITSP to dial when you get a call on a certain phone number? |
| 03:43.34 | *** join/#asterisk girlny (n=girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 03:43.48 | random_mike | ChannelZ, I run the ITSP - just so we're all aware |
| 03:44.22 | random_mike | The idea is, to allow anyone registered to our ITSP to dial *xxx (and number we decide to use) and have that call directed to Asterisk, and have Asterisk do something based upon the number dialed. |
| 03:44.25 | girlny | were in my files i can find [macro-stdexten] i need to add some lines to it |
| 03:44.42 | random_mike | such as dial *123 and read the weather, dial *99 and get voicemail, etc, etc, etc |
| 03:44.43 | *** part/#asterisk gnufan (n=hardev@71-93-139-56.dhcp.bbcy.ca.charter.com) |
| 03:45.02 | voipmonk | Ë |
| 03:45.07 | voipmonk | Ëbook |
| 03:45.11 | voipmonk | Ëthebook |
| 03:45.18 | voipmonk | and that is not a tilde |
| 03:45.35 | random_mike | voipmonk, are you speaking to me/ |
| 03:45.37 | ChannelZ | it is not |
| 03:45.57 | voipmonk | nope Im on my way to making an ass of myself right now :) but you should read the book |
| 03:46.12 | ChannelZ | random_mike: but this 'slipper' is doing all of the front-end work (and why?) |
| 03:46.12 | voipmonk | and experiment with the dialplan logic you need to do what you want |
| 03:46.44 | random_mike | Slipper is the impliented SIP server my company use, I had no choice in this matter it was put in place prior to my start at this company |
| 03:47.00 | random_mike | im simply trying to add some additional features to the service. |
| 03:47.23 | girlny | how do i add some lines to [macro-stdexten] ??????????????? |
| 03:47.29 | girlny | were do i find it |
| 03:47.41 | p3nguin | ~freepbx |
| 03:47.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:47.49 | p3nguin | girlny: ^^^^^^^ |
| 03:48.30 | random_mike | As for the book, I have a copy of Asterisk: The future of telephony 2nd ed |
| 03:48.46 | p3nguin | random_mike: Why can't you use NORMAL extensions? The person calls *123, it matches in your dialplan, and runs some application. |
| 03:48.58 | ChannelZ | random_mike: so what are you doing in slipper to make *123 call your asterisk box? |
| 03:50.06 | random_mike | p3nguin, I have 1 registration to my ITSP. I dont know how to force calls to either DID to my dialplan |
| 03:50.26 | random_mike | ChannelZ, Basically theres a database that looks up dialled numbers, and assigns them to usernames |
| 03:50.37 | p3nguin | random_mike: I thought you were the ITSP. |
| 03:50.41 | random_mike | in my case, I have *99 and *123 directed to the username |
| 03:50.49 | random_mike | p3nguin, I am the ITSP yes. |
| 03:51.00 | p3nguin | But heck, you didn't even know what an ITSP was until 50 minutes ago. |
| 03:51.15 | random_mike | The abriviation ITSP correct. I had no idea what that mean |
| 03:51.20 | random_mike | meant* |
| 03:51.28 | random_mike | over here we call them SIP Providers. |
| 03:51.29 | ChannelZ | well with this retarded setup on slipper's side, the path of least resistance is to just create different SIP peers for each number.. direct them to different contexts ors omething |
| 03:51.49 | random_mike | Yes ChannelZ I fear that's what I'll end up having to do |
| 03:51.57 | p3nguin | So if you are the ITSP, why are you worried about your Asterisk's registration to your ITSP? |
| 03:52.34 | p3nguin | The calls you said you were having issues with were those of people calling in to you. |
| 03:52.41 | ChannelZ | random_mike: but I dunno what the big deal with writing a small AGI to parse the To header otherwise and jump to that extension.. |
| 03:52.44 | p3nguin | So use normal dialplan logic. |
| 03:53.23 | random_mike | I'd prefer to use the one registration to keep life simple. I'm dancing here with you guys to see if that is possible. But it appears that's not the case. |
| 03:53.26 | ChannelZ | p3nguin: * is NOT the front-end of his setup. |
| 03:53.34 | p3nguin | Why does that matter? |
| 03:53.55 | p3nguin | Are you saying that Asterisk doesn't ultimately control the call? |
| 03:53.57 | ChannelZ | because the bit that IS the front-end doesn't send him calls in the form that he needs to just 'use normal dialplan logic' on the * side |
| 03:54.40 | random_mike | p3nguin, because if I want to assign more numbers for more features in the future, it means I need to create more accounts to register to. Obviously this would be optimum to keep it to one registration and have multiple numbers directed to the one username. |
| 03:54.50 | ChannelZ | The person dials *123, "slipper" (his SIP front-end) is programmed to send the call to his * box. But not as extension *123, so he can't easily see what the person originally dialed |
| 03:55.15 | p3nguin | Now we're finally making some sense! |
| 03:55.53 | ChannelZ | This was all explained awhile ago |
| 03:56.14 | random_mike | Well to p3nguin's credit, Im not the most verbose person when it comes to fine details. |
| 03:58.18 | voipmonk | thats what an itsp is all about :) the details... the cdrs... down to a fraction of the second... you sure you're in the right business ? |
| 04:00.04 | random_mike | heh I was the next best thing once the guy incharge of our voip system retired |
| 04:00.10 | random_mike | so now its time to get my hands dirty. |
| 04:00.19 | ChannelZ | wait a minute |
| 04:00.48 | random_mike | I'll go down the path of multiple registrations atleast I know this will work as I can simple force a registration to an extension in Asterisk. |
| 04:02.59 | random_mike | thanks for your assistance ChannelZ and p3nguin and voipmonk - we'll get there in the end (including blood sweat and tears) |
| 04:03.12 | ChannelZ | if you keep all your extensions kind of consistent, you could do this probably just with simple substrings in * |
| 04:03.29 | ChannelZ | (that is, always the same number of digits) |
| 04:03.43 | random_mike | that can be done |
| 04:04.34 | ChannelZ | ${foo:0:4} |
| 04:04.49 | ChannelZ | would be the first 4 characters of variable 'foo' |
| 04:05.16 | ChannelZ | so ${${SIP_HEADER(To)}:0:4} would be what you want perhaps |
| 04:10.45 | ChannelZ | ....so....in your 's' extension, Goto(${${SIP_HEADER(To)}:0:4},1) would jump to the extension, priority 1 |
| 04:19.30 | *** join/#asterisk alrs (n=lars@h-74-3-2-122.lsanca54.static.covad.net) |
| 04:21.44 | ChannelZ | WHOA there is a regex operator. |
| 04:24.08 | ChannelZ | Set(regx="(\*[0-9]+ ") |
| 04:24.47 | ChannelZ | Set(realexten=$["${SIP_HEADER(To)}" : ${regx}]) |
| 04:25.21 | ChannelZ | would work too, assuming the extensions were *### (with any number of numbers) |
| 04:27.01 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 04:29.38 | ChannelZ | pinches random_mike |
| 04:30.12 | random_mike | sorry fixing a mailserver issue |
| 04:30.13 | random_mike | well |
| 04:30.21 | random_mike | a staff )!@# up |
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| 04:55.52 | *** join/#asterisk harisalialiali (n=harisali@58.65.193.26) |
| 04:59.24 | harisalialiali | Hi all I need guidence for MOH via Line-in of sound card, can any body plz help me ? |
| 05:00.54 | harisalialiali | Hi all, I need some assistance on live stream via MOH, please |
| 05:04.22 | ChannelZ | you need an app that spits out audio data at the right samplerate from your card |
| 05:05.25 | harisalialiali | I did some work wanna to share with you |
| 05:05.33 | ChannelZ | here try this http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf#Usingasoundcardasthesource |
| 05:06.58 | harisalialiali | ok thanks , I had tried to configure server , by study different web sites I am using asterisk 1.6 |
| 05:08.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-fpwrpynedlghgomv) |
| 05:09.12 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 05:15.33 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 05:16.24 | imcdona | what defines the order of the music played in musiconhold.conf? timestamp of file? |
| 05:19.54 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 05:22.52 | ChannelZ | imcdona: depends on what version of * and what you have set |
| 05:23.39 | ChannelZ | 1.6 has a couple more options for sorting |
| 05:24.27 | imcdona | asterisk 1.4.21.2 "mode=files" and then the directory. All the sounds files are in the native asterisk WAV format |
| 05:25.41 | imcdona | The first sound file in the music on hold tells callers we are busy and alternative options to get account balance etc. |
| 05:25.59 | ChannelZ | in 1.4 the order is in 'directory order' which is however the filesystem returns the files to * in a listing |
| 05:27.48 | imcdona | an "ls -als" shows teh files in the correct order. how can I determine what * sees the order as? |
| 05:32.05 | ChannelZ | try ls -alsU |
| 05:32.38 | ChannelZ | but I dunno if that is even right - I remeber when I was screwing with it last year I gave up getting a listing that showed the same order as * was playing them |
| 05:34.16 | imcdona | that listing looks to be correct. thanks channelz |
| 05:35.32 | ChannelZ | now how you get them into a different order is another question |
| 05:35.49 | ChannelZ | 1.6 has a sort=alpha option |
| 05:36.37 | imcdona | hmm. looks like I am going to have to get creative here with 1.4 to get this to work. |
| 05:36.51 | *** join/#asterisk aidinb (n=Aidin@32.171.37.40) |
| 05:39.12 | imcdona | I could take all the wav file and combine them into one big one. That would solve it ;) |
| 05:39.47 | imcdona | actually so far the order looks to be file size, smallest file first |
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| 05:44.36 | imcdona | nope...not size either...spoke too soon |
| 05:47.45 | *** join/#asterisk harisalialiali (n=harisali@34.158.70.202.dynamic.max.com.pk) |
| 05:49.34 | ChannelZ | It's 'directory order'.. you might try moving all the files out of there, and then copying them back into the directory, one by one, in the order you want |
| 05:49.40 | harisalialiali | Hi i need assiatance on MOH via soundcard Line-in, I have implemented instruction on different web sites like http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf but unable to listen stream |
| 05:49.40 | harisalialiali | kindly help me I am using asterisk 1.6 |
| 05:51.00 | imcdona | I am not familiar with how to do that. However, you can plug your source into another computer and use a shoutcast stream to asterisk |
| 05:53.59 | ChannelZ | harisalialiali: are you trying to use 'arecord'? |
| 05:54.23 | harisalialiali | yap i am trying arecord n alsamixer |
| 05:54.39 | ChannelZ | does arecord -l show your card? |
| 05:55.46 | harisalialiali | yes it shows,,, card 0: I82801DBICH4 [Intel 82801DB-ICH4] |
| 05:56.18 | harisalialiali | and i am able to listen mic or lin-in sound on head phone to |
| 05:57.18 | harisalialiali | an there is no error in script event not on cli [new stack |
| 05:57.21 | harisalialiali | <PROTECTED> |
| 05:57.37 | harisalialiali | but still no sound |
| 05:58.30 | ChannelZ | well I'd try running arecord manually (but with -t wav blah.wav instead of -t raw), let it run for a few seconds, and then play the resulting wav file, make sure it's working at all |
| 05:58.53 | imcdona | http://www.trixbox.org/forums/trixbox-forums/help/music-hold-line-sound-card |
| 05:59.43 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 05:59.55 | ChannelZ | yeah if you aren't running asterisk as root, it might be that whatever user it's running as doesn't have access to the audio card |
| 06:00.54 | *** join/#asterisk [8none1]_ (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 06:03.16 | harisalialiali | I am running asterisk as root, I installed asterisk by yum, & asterisk by default install as root ? |
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| 06:07.14 | harisalialiali | ok I replace -t raw with -t wav in my script but still no sound [#!/bin/bash |
| 06:07.14 | harisalialiali | ] |
| 06:07.35 | harisalialiali | #!/bin/bash |
| 06:08.01 | harisalialiali | ""/usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t wav" |
| 06:08.49 | harisalialiali | Is asterisk 1.6 required decoder for mp3 ? as i read its work on wav now ? |
| 06:10.30 | ChannelZ | I didn't mean to replace it in your script... but to run the whole commandline manually to see if it even works for you |
| 06:10.39 | ChannelZ | also remove -q and see if it complains of something |
| 06:10.52 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
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| 06:34.46 | *** join/#asterisk cvnet (n=cvnet@dsl-69-172-67-153.acanac.net) |
| 06:34.48 | cvnet | hello all |
| 06:38.19 | cvnet | when a call comes in to the system, whats the variable that holds the DID (incoming numbers) value? |
| 06:39.13 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:41.33 | p3nguin | cvnet: Possibly EXTEN |
| 06:42.40 | cvnet | lol |
| 06:42.44 | cvnet | ya i feel stupid now |
| 06:42.54 | cvnet | logical |
| 06:44.16 | ChannelZ | jesus audio on linux is a mess |
| 06:45.32 | Faustov | right... |
| 06:46.15 | cvnet | exten => 0,1,SipAddHeader(P-Asserted-Identity: "My name" <sip:+${EXTEN}@192.168.1.1>) |
| 06:46.17 | cvnet | didnt work for me |
| 06:56.56 | ChannelZ | hmm I guess arecord only likes my device if I do it in stereo. |
| 07:01.32 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
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| 07:27.22 | benngard | hm... why does latest trunk crash? the only error i see is "[Jan 25 08:22:30] ERROR[30084] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory" |
| 07:33.58 | sun28 | moin \o/ |
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| 07:51.15 | benngard | Jan 25 08:50:05 sip kernel: [1750086.150667] asterisk[30587]: segfault at 94 ip b7cce9a0 sp b64a7204 error 4 in libpthread-2.7.so[b7cc7000+15000] |
| 07:51.21 | benngard | :( |
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| 08:51.59 | mlarsen | Is it possible to prefix a callerid with the selection from an ivr menu? |
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| 10:33.55 | enriquemora | hello |
| 10:37.02 | enriquemora | We have an Asterisk 1.6.1.8 that is crashing occasionaly with a segfault at a4 in libpthread. I would really appreciate if someone can point me in the right direction. Should I open an issue at issues.asterisk.org? |
| 10:37.43 | Chainsaw | enriquemora: I would suggest that you update to 1.6.1.13 before you try anything else. |
| 10:38.20 | enriquemora | I'll start looking at that right now |
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| 10:48.03 | benngard | enriquemora: wellcome to the club, my box have started to do the same |
| 10:48.44 | benngard | but i did play around pretty hard with it lately so im not sure what part is causing the error |
| 10:49.10 | benngard | but i know that i upgraded to latest dahidi, what version are u running? |
| 11:00.49 | *** join/#asterisk lordmortis (n=lordmort@203-59-87-141.dyn.iinet.net.au) |
| 11:04.34 | enriquemora | hi benngard. We're running DAHDI 2.2.0.2 |
| 11:05.38 | enriquemora | I also saw something about activating the DONT_OPTIMIZE compiler option. We have it deactivated |
| 11:11.38 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 11:12.29 | benngard | enriquemora: i did downgrade to that version but the same fault |
| 11:18.31 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 11:20.55 | benngard | in my case it seems that ooh323 is the bad guy :( |
| 11:21.21 | benngard | did remove that channel driver and restarted no crashes so far |
| 11:23.21 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:23.32 | *** join/#asterisk Obeliks (i=obeliks@gentoo/contributor/Obeliks) |
| 11:24.45 | enriquemora | benngard, we too have ooh323 loaded... but were not using it and we can remove |
| 11:25.02 | enriquemora | where did you see that ooh323 was the culprit? |
| 11:26.20 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 11:27.24 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 11:29.05 | Whtsup | hello |
| 11:31.44 | benngard | we did some heavy ooh323 test last night so i took i wild guess |
| 11:32.13 | Whtsup | anyone can help regarding a2billing issue |
| 11:35.04 | *** join/#asterisk Akiraa (n=Akira@92.81.195.213) |
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| 11:50.10 | Akiraa | What's a multi-port (4) FXO interface you would recommend? |
| 11:50.40 | enriquemora | btw DONT_OPTIMIZE has nothing to do with the segfault. I just needs to be deactivated so that the crashdump will contain debugging symbols. Since I'm not going to open an issue just yet, we wont be touching this for now. |
| 11:51.31 | enriquemora | Akiraa, are you looking for a card or a SIP FXO Gateway |
| 11:51.33 | enriquemora | ? |
| 11:52.36 | Akiraa | enriquemora: for now, I am looking at PCI/PCIe cards, but was wondering if standalone devices (SIP/IAX2 FXO gateways) exist |
| 11:53.32 | Akiraa | other than the 1 port (FXO+FXS) spa 3102 |
| 11:54.47 | enriquemora | There are many standalone devices with 4 ports. However, I personally cannot recommend one. I can however recommend AGAINST one. We are very unsatisfied with the SPA400 |
| 12:04.06 | *** join/#asterisk [netman] (n=netman@195.Red-88-23-119.staticIP.rima-tde.net) |
| 12:11.09 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 12:11.42 | *** part/#asterisk harisalialiali (n=harisali@116.71.62.153) |
| 12:16.22 | enriquemora | Akiraa, one of my engineers says that the Vegastream 4 FXO works well |
| 12:17.48 | *** join/#asterisk coppice (n=chatzill@234.157.17.210.dyn.pacific.net.hk) |
| 12:17.58 | enriquemora | Has anyone used a Fritz!Box with Asterisk? |
| 12:47.29 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
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| 13:00.35 | *** join/#asterisk Zeeshan_M (i=develope@o.je) |
| 13:02.31 | Zeeshan_M | Hiya chaps. I am looking for a stand-alone hardware product I can purchase which uses Asterisk internally. I am looking to buy the product at http://www.orchid-telecom.com/Diallers/v4.html but would love an open source alternative. |
| 13:03.34 | Zeeshan_M | I do not mind if I am required to use my PC to program the device and then make use of it standalone, but I do not wish to have my computer running 24/7 in order to run Asterisk to reroute or translate telephone numbers I call. |
| 13:03.51 | Zeeshan_M | Any recommendations? :-) |
| 13:04.28 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 13:04.32 | Akiraa | Zeeshan_M: you could try to install Asterisk on a ddwrt-capable router |
| 13:06.19 | Akiraa | Zeeshan_M: or a small form factor PC running without a monitor attached |
| 13:07.01 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:07.18 | Zeeshan_M | I really don't want to buy a small mini-factor PC for this. I am a simple home user! |
| 13:07.35 | Zeeshan_M | I am checking to see if my Linksys router thingy is 'DD-WRT' capable |
| 13:07.42 | Akiraa | well, a hardware IP pbx is probably too expensive |
| 13:08.10 | Zeeshan_M | Are there any alternatives to the product I noted via the URL above? |
| 13:08.15 | Akiraa | but check out some small computers, they shouldn't break the bank really |
| 13:08.22 | *** join/#asterisk [Outcast] (n=anonymou@64.202.62.5) |
| 13:08.24 | Akiraa | compared to specialized voip hardware |
| 13:10.31 | Zeeshan_M | is curious |
| 13:10.51 | Zeeshan_M | How would a 'DD-WRT' capable router allow me to connect my telephone into its back sockets? |
| 13:11.00 | Zeeshan_M | I thought those are only meant for CAT5e cables? |
| 13:11.29 | Akiraa | Zeeshan_M: an IP phone uses ethernet |
| 13:11.30 | *** join/#asterisk hhkahya (n=hulusika@88.247.127.66) |
| 13:11.53 | Akiraa | if you want to connect your analog phone, you will need an adaptor (ATA) |
| 13:12.05 | Zeeshan_M | Oh. Pants. I didn't note I wanted calls routed through the normal telephone wiring, not via the Internet. |
| 13:12.33 | Zeeshan_M | (I am in the UK, and make use of a ADSL modem that is connect to the Internet. This is connected to a ADSL filter which splits the telephone wire to let me connect my ADSL modem and my normal telephone lead) |
| 13:12.53 | Akiraa | Zeeshan_M: just use skype :) |
| 13:12.59 | [TK]D-Fender | Akiraa: I'm only familiar with ONE piece of something I'd call "VoIP Hardware' and virtually no-one uses it. |
| 13:13.01 | Zeeshan_M | Heh, I am have considered that too. |
| 13:13.23 | Zeeshan_M | That isn't suitable for my needs either. Maybe I'll just buy the £15 unit from http://www.orchid-telecom.com/Diallers/v4.html and code it. |
| 13:14.07 | Zeeshan_M | There's a good small community for the device at http://forums.moneysavingexpert.com/showthread.html?t=1477019 and it seems to have resolved most of my coding needs using MS Excel and the need for a 56k modem to transfer the details to the device. |
| 13:14.28 | [TK]D-Fender | Zeeshan_M: What are you trying to do exactly? |
| 13:14.32 | Akiraa | [TK]D-Fender: I was checking some prices; hardware IP-PBX runs somewhat against the concept of Asterisk; one offering actually required a license fee per each SIP terminal; fail |
| 13:15.00 | *** join/#asterisk korihor (n=korihor@190.205.251.97) |
| 13:15.06 | [TK]D-Fender | Akiraa: That is "proprietary PBX", not "VoIP Hardware". |
| 13:15.18 | Zeeshan_M | I am looking to connect a device to my landline which I connect my normal telephone to, upon me dialing certain high cost numbers, I want to translate/reroute these numbers. |
| 13:15.39 | [TK]D-Fender | Akiraa: The fact it can spake a VoIP protocol is a little besides the point |
| 13:15.57 | Zeeshan_M | I need to be able to do this based on time of day and what day type it is (weekday/weekend). I also require a small pause eventing after connection and then the functionality to add additional numbers. |
| 13:16.09 | [TK]D-Fender | Zeeshan_M: So you want Asterisk to connct both to your analog LINE as well as an analog PHONE then? |
| 13:16.19 | Zeeshan_M | Event handling like 'after call is connected, wait 5 seconds, press #, enter 12345, press #' |
| 13:16.29 | [TK]D-Fender | Zeeshan_M: and based on what you dial, use either your landline, or an ITSP for example? |
| 13:16.29 | Zeeshan_M | Yes |
| 13:16.38 | Zeeshan_M | No, none of that 'ITSP' stuff |
| 13:17.00 | [TK]D-Fender | ~itsp |
| 13:17.00 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 13:17.11 | [TK]D-Fender | Zeeshan_M: taht is that LCR you were referring to.... |
| 13:17.16 | Zeeshan_M | If I dial 0800 800 150 using my telephone, I want the asterisk to notice it's Monday morning and the call will cost more to route through my telephone provider. |
| 13:17.24 | [TK]D-Fender | Zeeshan_M: use a VoIP termination provider instead of your landline |
| 13:17.30 | Zeeshan_M | So it'll prefix the number with say 1500 and then call 0800 800 150 |
| 13:18.03 | [TK]D-Fender | Zeeshan_M: Anyway, here's an option for you : Linksys SPA-3102 |
| 13:19.13 | Zeeshan_M | Any recommendations which don't invoice VOIP or any kind of routing through the Internet? |
| 13:19.24 | Zeeshan_M | I wish to only translate/reroute numbers on my analog telephone |
| 13:19.57 | Zeeshan_M | invoice => involve |
| 13:20.13 | ManxPower-work | Zeeshan_M: then you will spend money on a telephony card. |
| 13:20.33 | Zeeshan_M | A PCI card that connects to a computer? |
| 13:20.42 | Zeeshan_M | That's going to force a PC to be on 24/7 :( |
| 13:20.58 | ManxPower-work | Zeeshan_M: You are in the wrong place then. |
| 13:21.00 | Zeeshan_M | connects => goes into, even. Which I can connect my analog telephone |
| 13:21.16 | Zeeshan_M | heh. It seems like I misunderstood what asterisk is intended for :-) |
| 13:21.29 | ManxPower-work | Zeeshan_M: Asterisk is a toolkit that allows you to build a phone system. |
| 13:21.46 | Zeeshan_M | Yep, I thought it could also be used for routing only. |
| 13:21.51 | ManxPower-work | Now, if you don't want to have your phone system powered on 24/7 then there really isn't much we can help you with. |
| 13:22.33 | Zeeshan_M | Surely there are standalone devices which run asterisk that can be connected to telephone line to reroute/translate the calls |
| 13:22.39 | Zeeshan_M | Devices like: http://www.orchid-telecom.com/Diallers/v4.html |
| 13:23.34 | ManxPower-work | Zeeshan_M: If that device is a PC, runs Linux, and has supported telephony interface cards, then it might work. |
| 13:23.59 | Zeeshan_M | I see. |
| 13:24.25 | [TK]D-Fender | [08:19]<Zeeshan_M>I wish to only translate/reroute numbers on my analog telephone <- the unit I already suggested to you |
| 13:24.46 | ManxPower-work | It's not hard to make Asterisk run on non-Intel platforms, but because the PSTN interface cards are not supported, you are limited to VoIP. |
| 13:25.43 | [TK]D-Fender | Zeeshan_M: [08:24]<ManxPower-work>It's not hard to make Asterisk run on non-Intel platforms, but because the PSTN interface cards are not supported, you are limited to VoIP. <- Clarification : Limited to using VoIP between ASTERISK and the DEVICE that lets you plug in your analog line <- |
| 13:25.55 | [TK]D-Fender | Zeeshan_M: This has nothing to do with the public internet |
| 13:26.11 | Zeeshan_M | Aaah |
| 13:26.25 | Zeeshan_M | I saw the terms VOIP in the product specification you noted and ran for the hills |
| 13:26.32 | [TK]D-Fender | Zeeshan_M: So router w/ DD-WRT + SPA-3102 = enough |
| 13:26.50 | Zeeshan_M | Linksys SPA-3102 on its own is not sufficent to program? |
| 13:27.04 | [TK]D-Fender | Zeeshan_M: No. |
| 13:27.07 | Akiraa | Zeeshan_M: what exactly fo you want to do? |
| 13:27.25 | [TK]D-Fender | Akiraa: Already all answered.... I just asked this question. |
| 13:28.22 | Akiraa | ah, multi-line cost-optimizing phone dialers |
| 13:28.41 | Zeeshan_M | Akiraa, I wish to use my analog telephone to make calls to my analog phone line (no VOIP/SIP), based on the time of the day, if its the weekend or a weekday and based on what I just input as the number to call, I would like to reshape the telephone number and add prefixes to it or call another number and upon connection, dial the initially given number. |
| 13:28.49 | Zeeshan_M | Yep. |
| 13:30.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:30.05 | Zeeshan_M | I was hoping that I could use Asterisk::LCR::* modules on CPAN to write some basic scripts in Perl to handle my needs on a device running Asterisk. It seems the devices I need are much more complex and costly than I expected. |
| 13:31.15 | Zeeshan_M | Anyhow, thank you chaps for your help and clarifications. I appreciate your time and patience. |
| 13:31.23 | Zeeshan_M | Have a good day/evening :) |
| 13:31.25 | ManxPower-work | Telephony is always more complex and costly than you expect. Unless, of course, you are familiar with telecom, then Asterisk appears simple and cheap. |
| 13:31.34 | *** part/#asterisk Zeeshan_M (i=develope@o.je) |
| 13:32.29 | [TK]D-Fender | If you can't recoup a $68 USD expense on an SPA-3102 then the projet can't be worth doing at all |
| 13:32.36 | coppice | anything real time gets interesting |
| 13:34.47 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 13:35.44 | ManxPower-work | Telecom Person: Wow! I can get 24 T-1 channels for less than $700? That's cheap! n00b: What do you mean I have to spend $70 on an ATA? I thought this was supposed to be free!" |
| 13:38.58 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:42.50 | beek | ManxPower-work: Very nice. And dead-on accurate. |
| 13:47.11 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
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| 14:04.37 | ManxPower-work | Only 7 days until the Polycom SDK is supposed to be releases. Yay! |
| 14:05.04 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:05.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:05.05 | Gido-E | GPL license? |
| 14:05.18 | ManxPower-work | Gido-E: I guess we'll see when they release it. |
| 14:05.29 | coppice | ManxPower-work: what kind of software development is that for? |
| 14:06.20 | [TK]D-Fender | BRB |
| 14:07.24 | ManxPower-work | coppice: not sure. I'm just hoping it has some more documentation. http://www.polycom.com/support/voice/spip_ssip_vvx_SDK_availability.html |
| 14:07.45 | Tim_Toady | any idea to check if a remote iax port is open and asterisk is listening? |
| 14:07.46 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 14:07.55 | Tim_Toady | netcat? nmap? |
| 14:08.16 | ManxPower-work | Tim_Toady: netstat -an |
| 14:08.33 | Tim_Toady | ManxPower-work remote port not on my local machine |
| 14:08.54 | Gido-E | netstat -utn |
| 14:09.00 | ManxPower-work | Tim_Toady: It sucks to be you. |
| 14:09.05 | Gido-E | a gives you also the unix sockets. |
| 14:09.32 | ManxPower-work | is the remote server behind nat? |
| 14:09.38 | Tim_Toady | yes |
| 14:09.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 14:09.46 | ManxPower-work | did you forward UDP/4569? |
| 14:10.07 | Tim_Toady | its supposed to be forwared |
| 14:10.11 | Tim_Toady | thats what i wanna check |
| 14:10.25 | ManxPower-work | that was not your original question |
| 14:10.47 | Tim_Toady | my original question was 'if a remote iax port is open and asterisk is listening?' |
| 14:11.00 | Tim_Toady | and it still is |
| 14:11.08 | ManxPower-work | Yup. No way to tell if Asterisk is really listening without seeing the local machine. |
| 14:11.30 | Tim_Toady | can i send some garbage with nc and get any responce? |
| 14:11.33 | ManxPower-work | So, based on the limited stuff you actually CAN do, I'd nmap it. |
| 14:11.41 | Tim_Toady | nmap didnt help |
| 14:11.57 | creativx | try to register to it.. |
| 14:12.02 | ManxPower-work | as I said, it sucks to be you. |
| 14:12.24 | ManxPower-work | why did nmap not help? |
| 14:12.36 | Tim_Toady | that's supposed to be some funny line? |
| 14:12.54 | ManxPower-work | No. If nmap shows the port not open, then the port is not open. If it shows the port open, then the port is open. |
| 14:13.12 | ManxPower-work | Not exactly sure how either of those is "not help" |
| 14:13.23 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:13.24 | benngard | nmap is benngard's best friend! |
| 14:13.30 | Katty | hi |
| 14:13.39 | Tim_Toady | it reposts 4569/udp open|filtered unknown for all ports, even in local machies that 4569 is open |
| 14:13.43 | benngard | welcome onboard |
| 14:14.03 | Katty | my asterisk does not work at all |
| 14:14.07 | Katty | how to fix pls??? |
| 14:14.14 | creativx | Katty: plz halp!!!!!!!1 |
| 14:14.27 | Katty | creativx: autoooooooowashh |
| 14:14.28 | benngard | hire a consultabt for $1000 per hour! |
| 14:14.29 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-40-42.acanac.net) |
| 14:14.34 | benngard | consultant* |
| 14:14.40 | Gido-E | katty, jump 2 times, facing north. |
| 14:14.40 | Katty | hello mister monk! how's the daughter? |
| 14:14.46 | Katty | benngard: now why would i do that? |
| 14:14.57 | Katty | benngard: i'd get more use out of a house cleaner. |
| 14:15.05 | benngard | :) |
| 14:15.06 | Katty | benngard: do you do laundry and dishes and dust? |
| 14:15.23 | benngard | Katty: i am a father to 2 sons, what do u think? |
| 14:15.23 | Gido-E | katty, woman are born for that kind of work. |
| 14:15.54 | Katty | benngard: i'm going to guess yes, and more |
| 14:16.02 | Katty | Gido-E: you're cruisin for a bruisin |
| 14:16.13 | Katty | rolls up the sleeves |
| 14:17.34 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:17.59 | Gido-E | Katty nope, just explaining how the world works. |
| 14:18.07 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 14:18.51 | ManxPower-work | The way the world works is that everyone is trying to get an advantage over everyone else. |
| 14:19.11 | Katty | well my squirrel feeders are most definately the advantagious ones. |
| 14:19.17 | voipmonk | the daughter is fine :) hello ! |
| 14:19.32 | Katty | voipmonk: when do you get photos of the ultrasound? |
| 14:19.49 | voipmonk | i have photos now :) |
| 14:19.53 | voipmonk | and a movie |
| 14:19.53 | Katty | ! |
| 14:19.56 | voipmonk | at 20 weeks |
| 14:19.59 | Katty | you never shared them :< |
| 14:20.00 | Katty | cries |
| 14:20.05 | voipmonk | oh... |
| 14:20.05 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 14:20.08 | Polysics | hello |
| 14:20.10 | Polysics | i need an idea |
| 14:20.16 | Polysics | i have my system that is mostly working |
| 14:20.30 | Katty | i suggest cheese. |
| 14:20.33 | Katty | and pasta. |
| 14:20.40 | Katty | two wonderful ideas that will get you closer to a lovely dinner. |
| 14:20.43 | Polysics | but i would like some users to be able to make calls to the outside through the sip provider we are using |
| 14:20.54 | Polysics | cheese and pasta always work |
| 14:20.55 | ManxPower-work | Quick! Patent the idea! |
| 14:20.56 | Polysics | :-) |
| 14:21.11 | Katty | ManxPower-work: i'll Patent you in a minute. |
| 14:21.24 | Polysics | what is usually done there? provide an extension tha switches to the outside context? |
| 14:21.33 | Polysics | like, dial 0 then the outside number? |
| 14:21.34 | ManxPower-work | You use contexts |
| 14:21.45 | Katty | Polysics: there are a million ways to do it |
| 14:21.52 | Katty | Polysics: the question is, how do you, and your users, want to do it |
| 14:21.54 | ManxPower-work | phones with different "permissions" means phones in different contexts |
| 14:22.06 | Polysics | as of now all phones are in the same context |
| 14:22.14 | Polysics | let's simplify it to "all users" |
| 14:22.18 | ManxPower-work | Polysics: then they all have the same permissions. |
| 14:22.45 | ManxPower-work | contexts are THE way to set what a phone can and can't do in the dialplan |
| 14:22.52 | Polysics | and they call each other by dialing their nomber, which is a progressive from 1000 |
| 14:23.32 | ManxPower-work | For example I may have phones in the toll-access context that include => contexts that allow dialing 9+1+ac+phone number, but other phones are in the exten-access context, those phones are only allowed to dial extensions |
| 14:24.31 | ManxPower-work | Polysics: you're not using some form of Asterisk GUI, are you? |
| 14:24.46 | Polysics | no, doing everything by hand |
| 14:24.52 | ManxPower-work | good. |
| 14:24.53 | Polysics | so i at least learn something :-) |
| 14:25.08 | ManxPower-work | Chances are you'll have to totally redesign your dialplan. |
| 14:25.29 | Polysics | my extensions.conf mostly contains a reference to Realtime, plus a few test extensions |
| 14:25.45 | ManxPower-work | Oh. I can't help you then. I wish you the BEST of luck. |
| 14:26.03 | Polysics | realtime is unused at the moment, btw :-) |
| 14:26.09 | Polysics | there are no extensions in the db :-) |
| 14:26.11 | [TK]D-Fender | Polysics: contexts.... learn 'em |
| 14:28.24 | Polysics | so, the answer to the above, if i did know about context, would be "have 0 switch to a context for outgoing calls"? |
| 14:28.42 | ManxPower-work | Polysics: no. |
| 14:28.51 | Polysics | good :-) |
| 14:29.00 | ManxPower-work | make the phones land in the restricted context by default by using the context= line in sip.conf for each restricted phones. |
| 14:29.37 | Polysics | and then those restricted context includes the "normal" one? |
| 14:29.42 | Polysics | *that |
| 14:30.00 | Polysics | ie. the one that has the various test extensions |
| 14:30.06 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 14:30.45 | [TK]D-Fender | Polysics: Contexts separate what a given device or call has access to at any given point. |
| 14:31.20 | ManxPower-work | The restricted contexts would only include => contexts that allow you do do whatever it is you want to do, like only dial extensiosn |
| 14:31.28 | [TK]D-Fender | Polysics: If you want only CERTAIN people have access to certain things then put them in separate contexts that have separate extensions |
| 14:32.07 | Polysics | conversely, if i have a "basic" context that gives everyone some needed functions, i include THAT into other more specialized contexts? |
| 14:33.18 | voipmonk | contexts can include other contexts with include => contextnamehere - but be careful - if u need to make another context to exclude some function, you can :) |
| 14:33.44 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:33.48 | ManxPower-work | Polysics: contexts are both the most difficult things about Asterisk and also one of the most important things about Asterisk. |
| 14:33.49 | voipmonk | but you dont have to rewrite contexts.... just include them... or exclude them... keep this in mind when writing your own |
| 14:34.02 | [TK]D-Fender | ManxPower-work: Hardly difficult. |
| 14:34.55 | voipmonk | so many cooks in the kitchen.... just dive in and try what you think works... experiment... and use pastebin.ca to reply with some debug or your sip.conf and extensions.conf |
| 14:35.39 | Polysics | ok, i managed to move one SIP account to a separate context |
| 14:35.56 | Polysics | one thing: while using include, how are conflicts for the same extension resolved? |
| 14:36.24 | ManxPower-work | Polysics: since each context is TOTALLY separate, no problem. |
| 14:36.50 | ManxPower-work | now if you include a context with a duplicate extension, I think the first one matches, but it's a bad idea to do that. |
| 14:36.56 | Polysics | ok, that's it |
| 14:37.00 | Polysics | just that :-) |
| 14:37.44 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 14:39.17 | *** join/#asterisk iamdharma (n=iamdharm@static-68-162-250-125.bos.east.verizon.net) |
| 14:48.15 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.18.7) |
| 14:49.56 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 14:51.24 | Akiraa | Are there any attempts at devices that combine a classical telephone with an IP one? Perhaps with FXO gateway capabilities as well |
| 14:51.38 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:52.41 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:52.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:53.20 | [TK]D-Fender | Akiraa: SPA-3102 |
| 14:54.03 | Akiraa | [TK]D-Fender: I mean something like a phone+spa-3102 all in one device |
| 14:54.40 | [TK]D-Fender | Akiraa: Nothing that isn't some cheap POS |
| 14:55.19 | Polysics | wow, it works :-) |
| 14:58.49 | Akiraa | [TK]D-Fender: actually, I'd be interested in cheap devices for remote small offices |
| 14:59.54 | *** join/#asterisk darkskiez_ (n=dz@62-50-207-183.client.stsn.net) |
| 15:01.33 | [TK]D-Fender | Akiraa: Why would each phone be mixed mode? Whats the point? |
| 15:01.52 | Akiraa | why not? also, fewer cables etc make for a neater office |
| 15:01.53 | Katty | "There is no point, i just like the story" - Grumpy Old Men |
| 15:02.11 | leifmadsen | I like that line:) |
| 15:02.55 | Katty | hugs leifmadsen |
| 15:03.05 | leifmadsen | syn/ack hugs Katty |
| 15:03.11 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:04.37 | [TK]D-Fender | Akiraa: Fewer cables? Pardon? You d mode phones. Each phone wil require 2 wires |
| 15:04.47 | [TK]D-Fender | Akiraa: Sure doesn't make sense |
| 15:04.52 | [TK]D-Fender | Akiraa: Whats the POINT? |
| 15:05.20 | Skeeter- | a2billing is a nightmare, it must be the GUI part of it... |
| 15:05.23 | Katty | [TK]D-Fender: simmerdown cranky pants. |
| 15:05.50 | Katty | [TK]D-Fender: you're gonna get yourself all worked up |
| 15:07.34 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 15:09.50 | *** part/#asterisk dmast (n=dmast@exchange.newpointe.org) |
| 15:09.59 | *** join/#asterisk dmast (n=dmast@exchange.newpointe.org) |
| 15:10.31 | Katty | anyone see on reddit where a guy is trying to buy a 35 year old company so it can run for president? |
| 15:13.25 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:14.38 | Polysics | hmm |
| 15:14.52 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 15:15.03 | Polysics | any common cause for calls that do work, but one of the parties can only listen, ie. no microphone |
| 15:15.26 | Polysics | it's not the PC or the extensions, it's the fact that the party that can't talk is on a different, web-based softphone |
| 15:15.32 | coppice | Katty: what is the connexion between buying a company, and running for president? president of the company is a given if he buys it |
| 15:15.40 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:15.54 | Polysics | i would like to exclude * from the equation, and blame the web phone |
| 15:16.01 | *** join/#asterisk Drcarumas (n=carumas@a81-84-246-201.cpe.netcabo.pt) |
| 15:16.13 | Katty | coppice: supreme court ruled that a company is now a person |
| 15:16.16 | [TK]D-Fender | Polysics: NAT issues, sound-card issues, etc. take your pick |
| 15:16.17 | ManxPower-work | coppice: in the USA corporations are "people", therefore corporations should be able to become president. Yes, that is the logic they use. |
| 15:16.20 | Katty | coppice: and a us citizen, with full rights. |
| 15:16.35 | voipmonk | silent partners can use the president as a front end |
| 15:16.36 | coppice | so the company will run for president? |
| 15:16.38 | ManxPower-work | Katty: maybe full rights, but not full responsibilities. |
| 15:16.41 | Katty | coppice: so someone wants to take it to the extreme and run for president iwth a coporation |
| 15:16.41 | [TK]D-Fender | Polysics: test the device to * DIreCTLY, not using another phone. if that works bothe ways, then its likely a reinvitie issue as well betweent hem |
| 15:16.54 | Katty | http://i.imgur.com/c5VDl.jpg <- United States of America, LLC |
| 15:16.58 | Katty | ^- compliemnts of reddit. |
| 15:17.08 | coppice | Katty: could be the first "in the black" president |
| 15:17.23 | Katty | i just wonder how long it will take for someone to really try it |
| 15:17.29 | Polysics | [TK]D-Fender, it works both way using a normal softphone like X-Lite |
| 15:18.04 | *** join/#asterisk davevg-btwtech (n=davevg__@75.97.64.33.res-cmts.senj.ptd.net) |
| 15:18.20 | [TK]D-Fender | Polysics: .... |
| 15:18.34 | [TK]D-Fender | Polysics: test the web one direct |
| 15:19.05 | Polysics | as in "web to web"? |
| 15:19.10 | Polysics | there is no audio in that case |
| 15:19.39 | ManxPower-work | Polysics: you are making no sense. "web to web"? |
| 15:19.46 | [TK]D-Fender | Polysics: As in web -> *. How many more times to I have to say it? |
| 15:20.03 | Polysics | [TK]D-Fender, oh, like with the Echo app, sorry |
| 15:20.03 | *** join/#asterisk fatnasty1 (n=chatzill@ext-52.sagetelecom.net) |
| 15:20.12 | [TK]D-Fender | Polysics: as in RECORD and PLAYBACK |
| 15:20.28 | Katty | http://30.media.tumblr.com/tumblr_kwq5ac9NSO1qz4d4bo1_500.jpg <- BunBun has a picnic |
| 15:21.07 | fatnasty1 | In the dial plan, can I match on most specific rather than first match? |
| 15:21.15 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 15:21.24 | Drcarumas | Hi guys, i'm having this problem using asterisk AMD , Asterisk 1.4.24.1 . Could you please check this pastbin. I don't have a clue what happening i've searched, but nothing about this. http://pastebin.com/d1c4a6b2a |
| 15:21.24 | ManxPower-work | fatnasty1: that is the default |
| 15:22.40 | fatnasty1 | ManxPower-work: in my phones context, I include 2 other contexts. the first matches on _X. and the second on 100, if I dial 100 from the phone it matches on the _X. rather than 100. |
| 15:23.05 | Kobaz | any idea what would cause random t1 audio dropouts: http://www.kobaz.net/misc/dropout.wav |
| 15:23.13 | [TK]D-Fender | fatnasty1: INCLUDE |
| 15:23.22 | Kobaz | and all the counters are clean, no frame slips, no crc errors, no nothing |
| 15:23.23 | [TK]D-Fender | fatnasty1: INCLUDE's are searched in ORDER of "include" |
| 15:23.43 | [TK]D-Fender | fatnasty1: they are not clumped together for matching |
| 15:23.45 | ManxPower-work | fatnasty1: also extensions in the local context will always have priority over extensions in an include => 'd context |
| 15:24.07 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 15:24.24 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:24.49 | ManxPower-work | usually, needing an _X. shows poor dialplan design |
| 15:25.08 | Kobaz | ManxPower-work: i would disagree |
| 15:25.50 | fatnasty1 | ManxPower-work: Im just testing, I need to build osme logic that will playback number values to callers. I figured this logic would be out there allready, but I cant find it. |
| 15:26.00 | ManxPower-work | Kobaz: You are always welcome to disagree. You're wrong, but you can still disagree. 8-) |
| 15:26.11 | Gido-E | fatnasty1 saydigits? |
| 15:26.12 | Polysics | ok |
| 15:26.16 | Kobaz | well in that case we're both wrong |
| 15:26.27 | Kobaz | heh |
| 15:26.34 | Polysics | [TK]D-Fender, a normal softphone works both with a Record/Playback app and with an Echo demo |
| 15:26.35 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:26.50 | ManxPower-work | Polysics: then the problem would be with your "web phone". |
| 15:26.51 | fatnasty1 | Gido-E: No I need 100,300 to play back as "one hunderd thousand three hundred" |
| 15:27.03 | ManxPower-work | fatnasty1: SayNumber |
| 15:27.20 | fatnasty1 | ManxPower-work: Really? let me check |
| 15:27.37 | Gido-E | fatnasty1 welcome in the world of the wonders of asterisk :-) |
| 15:27.41 | ManxPower-work | fatnasty1: "core show applications" for the official list of applications for YOUR Asterisk. |
| 15:27.42 | [TK]D-Fender | fatnasty1: SayNumber() |
| 15:27.45 | Drcarumas | yep and if your in english it's already done |
| 15:27.46 | fatnasty1 | ManxPower-work: oh, hell yeah |
| 15:27.48 | Polysics | i don't see anything i can recognize ad an error though |
| 15:27.52 | Polysics | *as |
| 15:27.56 | fatnasty1 | ManxPower-work: Is there a saydate by anychance? |
| 15:28.03 | [TK]D-Fender | Polysics: You said web-client.... |
| 15:28.08 | ManxPower-work | fatnasty1: what does "core show applications" say? |
| 15:28.17 | fatnasty1 | ManxPower-work: nope |
| 15:28.25 | ManxPower-work | fatnasty1: then it's not there. |
| 15:28.38 | ManxPower-work | Odd, since I see something that would act like "saydate" |
| 15:28.38 | Polysics | [TK]D-Fender, the web client is basically connecting to the * using mjsip on the server... it's probably some sort of reinvite mismatch |
| 15:29.00 | Polysics | since the extension is defined as "nat=yes" but it actually has no nat |
| 15:29.01 | [TK]D-Fender | Polysics: "connecting to the * using mjsip on the server" <- huh? |
| 15:29.08 | fatnasty1 | ManxPower-work: ? |
| 15:29.14 | Drcarumas | guys let me check with you again my question: i'm trying to use AMD and getting some errors. Here's the pastbin link with the error: http://pastebin.com/d1c4a6b2a. What do you thing it could be the problem? |
| 15:29.21 | ManxPower-work | fatnasty1: read the list of applications again |
| 15:29.27 | Polysics | [TK]D-Fender, it's like a softphone running on the * machine itself |
| 15:29.32 | ManxPower-work | Drcarumas: I doubt many people here use AMD. |
| 15:29.57 | Polysics | shouldn't a reinvite problem of some sort show in the logs? |
| 15:30.01 | ManxPower-work | Polysics: every word out of your mouth makes your description more and more bizarre. |
| 15:30.31 | [TK]D-Fender | Polysics: ..... WTF is running client-side? |
| 15:30.41 | Polysics | how can i clarify it without turning into a burlesque VOIP act? :-) |
| 15:30.45 | Drcarumas | ManxPower-work: i'm aware that could be the case, even so maybe the error doenst got to do nothing with AMD. if some one could help that would be great. I've always get good tips from you guys :) |
| 15:30.59 | Polysics | [TK]D-Fender, Flash, and the audio echo test for Flash itself does work |
| 15:31.13 | Drcarumas | *anything to do with... |
| 15:31.18 | ManxPower-work | Polysics: no. Connect your web phone to Asterisk. You are not doing that. You are connecting your phone to something called "mjsip". |
| 15:31.20 | [TK]D-Fender | Polysics: there shouldn't be anything running on your * server..... should be SIP from client direct to * |
| 15:31.35 | [TK]D-Fender | Polysics: Whats this middleman crap? |
| 15:31.47 | ManxPower-work | [TK]D-Fender: I think he's just trolling. |
| 15:31.57 | TheDavidFactor | is there an easy to tell which module provides which applications once asterisk loaded? I know it's displayed during startup but I was wondering if it was possible to find from the CLI without restarting |
| 15:32.25 | Polysics | ManxPower-work, the Java client built on mjsip runs in a Red5 server to allow users to make and receive calls through a Flash in terface |
| 15:32.32 | voipmonk | Drcarumas: turn on debug in /etc/asterisk/logger.conf then start asterisk manually with a vvvgcd then engage your application and report back with a pastebin url... this will force asterisk to show its soul and not just verbose , we want to see what everything is doing. |
| 15:32.36 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 15:32.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:32.36 | Polysics | not trolling, it's (almost) working |
| 15:32.56 | Kobaz | I'm seeing an audio dropout of exactly 250ms on random intervals over t1/pri on a sangoma card. Any idea what would cause random t1 audio dropouts: here's a sample, http://www.kobaz.net/misc/dropout.wav |
| 15:33.00 | ManxPower-work | TheDavidFactor: no, but you don't see that list on start either. You can only see what modules are loaded "show modules". You can't see what application the modules ACTUALLY provides. |
| 15:33.01 | Drcarumas | voipmonk: thanks will do that |
| 15:33.34 | voipmonk | what slot is the sangoma card plugged into, Kobaz ? |
| 15:33.41 | Kobaz | voipmonk: the only pri slot that the machine has |
| 15:33.44 | TheDavidFactor | ManxPower-work, actually if you start asterisk with -cvvv it will show you each module that's loaded and what applications or functions it provides |
| 15:33.45 | Kobaz | it's a u1 server |
| 15:33.51 | Kobaz | er, i mean pci slot |
| 15:33.56 | ManxPower-work | app_dial.so, for example provides the Dial application. res_indications, provides Playtones application |
| 15:33.57 | TheDavidFactor | you might have to scroll up a lot ;-) |
| 15:34.02 | Kobaz | i can't type today... 1u server |
| 15:34.23 | *** join/#asterisk grayhame (n=miller@74-94-250-169-Nashville.hfc.comcastbusiness.net) |
| 15:34.40 | Kobaz | people are telling me it's getting progressively worse |
| 15:34.42 | voipmonk | Kobaz: read through http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 15:34.57 | Kobaz | it started on sunday and now it's happening more often |
| 15:34.57 | Kobaz | k |
| 15:35.03 | voipmonk | then read through http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 15:35.08 | Kobaz | k |
| 15:35.09 | voipmonk | then come back |
| 15:35.26 | Kobaz | heh |
| 15:36.38 | Kobaz | k |
| 15:38.09 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:38.32 | *** join/#asterisk [Outcast] (n=anonymou@64.202.62.5) |
| 15:40.20 | Kobaz | <PROTECTED> |
| 15:40.25 | Kobaz | hmm, that looks bad |
| 15:40.40 | Kobaz | all the other latency values for other pci cards are sub 60 |
| 15:41.08 | Kobaz | oh wait, the higher the better |
| 15:41.10 | Kobaz | never mind |
| 15:42.59 | Kobaz | Choosing a different filesystem might also be a good idea, reiserFS is recommended. |
| 15:43.03 | Kobaz | haha |
| 15:43.21 | Kobaz | voipmonk: this doc is old |
| 15:44.06 | ManxPower-work | Kobaz: you should expect that |
| 15:44.26 | Chainsaw | Kobaz: Most of voip-info.org is old and/or wrong. |
| 15:44.56 | Kobaz | yeah i know |
| 15:45.12 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:45.27 | coppice | voip-inaccurate-info.org |
| 15:49.07 | fatnasty1 | ManxPower-work: SayUnixTime(1264433703,-6,bd) thanks! |
| 15:49.27 | voipmonk | Kobaz: http://pastebin.ca/1765254 |
| 15:50.29 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 15:51.59 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:52.12 | *** join/#asterisk darkskiez (n=dz@62-50-207-156.client.stsn.net) |
| 15:53.30 | Kobaz | heh |
| 15:55.00 | Kobaz | all the test stuff from voipinfo and the asteriskguru site checks out so far |
| 15:55.30 | Kobaz | average from dahdi_test is 99.995799, no irq conflicts, no io contention with the hard drive |
| 15:55.55 | Kobaz | rebooting seems to fix the problem, at least in the past |
| 15:56.06 | Kobaz | haven't did one yet, since i wanna have sangoma look at it |
| 15:56.59 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:56.59 | *** join/#asterisk arossouw (n=arossouw@41.30.85.10) |
| 15:57.32 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 15:57.59 | *** part/#asterisk arossouw (n=arossouw@41.30.85.10) |
| 15:58.03 | Katty | haven't did one? |
| 15:58.22 | *** join/#asterisk kruemeltee (n=Maddin@port-92-198-62-82.static.qsc.de) |
| 15:58.24 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 15:58.34 | kruemeltee | hello all together |
| 15:58.48 | Katty | also! I'm looking for a big barn feeder which i can stick on that pole. if anyone has a suggestion, lemme know (= |
| 15:59.24 | Kobaz | Katty: havent did a reboot |
| 15:59.28 | Kobaz | did/done |
| 15:59.49 | Kobaz | it's too early to be gramatically correct |
| 16:00.03 | Katty | i ain't got none of dem der grammars. |
| 16:00.25 | ManxPower-work | You must be from Alabama. |
| 16:00.26 | Naikrovek | "Them there grammars." |
| 16:00.34 | Naikrovek | :D |
| 16:00.44 | Katty | grammatorials. |
| 16:00.48 | Katty | where's dave |
| 16:00.49 | Katty | i miss dave |
| 16:01.13 | Naikrovek | I saw someone online once say they had a PhD in Grammar. |
| 16:01.14 | Naikrovek | har |
| 16:01.33 | *** join/#asterisk eppigy (n=Dave@216-139-241-102.aus.us.siteprotect.com) |
| 16:01.34 | eppigy | hello |
| 16:01.36 | eppigy | i am dave |
| 16:01.40 | Katty | :>>>>>>>>>>>>>>>>> |
| 16:01.43 | eppigy | HEARTLES |
| 16:01.49 | beek | hugs Katty |
| 16:01.49 | Katty | hugs on eppigy |
| 16:01.55 | eppigy | huggles |
| 16:01.56 | Katty | eppigy: you've been gone forever. |
| 16:01.58 | Katty | hugs beres |
| 16:01.59 | Katty | oh |
| 16:02.03 | Katty | hugs beek, too |
| 16:02.15 | eppigy | yeah for some reason I get disconnected |
| 16:02.19 | eppigy | every now an dthen |
| 16:02.37 | Katty | it's okay. I KNOW WHERE YOU LIVE |
| 16:03.14 | ManxPower-work | I always assume epiggy was female. |
| 16:03.27 | Gido-E | or wants tobe |
| 16:03.54 | Katty | he doesn't look very female to me |
| 16:04.21 | beek | A female named dave? |
| 16:05.17 | bmoraca_work | like a boy named sue? |
| 16:05.29 | Naikrovek | I knew a dude named Angie once. |
| 16:05.33 | Katty | creepy |
| 16:05.39 | Katty | how dare he steal my name. |
| 16:05.40 | *** join/#asterisk arossouw (n=arossouw@41.31.17.227) |
| 16:05.43 | Polysics | ok, i think i found what the problem is: how/can i handle SIP accounts that can be both bhind NAT or not? |
| 16:05.47 | Naikrovek | His name was Angelo, but he preferred Angie |
| 16:05.58 | Katty | angelo? |
| 16:06.01 | Naikrovek | yeah |
| 16:06.06 | Naikrovek | eye-talian |
| 16:06.07 | Katty | michael angie |
| 16:06.12 | Katty | intttttttteresting. |
| 16:06.41 | bmoraca_work | Polysics, configure the endpoint correctly. nat=yes isn't required by Asterisk if the endpoints are properly configured. |
| 16:06.42 | ManxPower-work | Polysics: nat=yes will not nat if there isn't any nat |
| 16:07.27 | *** part/#asterisk ChannelZ (i=channelz@burner.com) |
| 16:07.41 | bmoraca_work | Polysics, right...that, too. nat=yes simply tells asterisk to prefer the ip:port combo that it received the packet from, rather than the URI contained within the SIP packet itself |
| 16:07.45 | Naikrovek | Anyone know a good 2-line ATA that allows you to set up autodialing? pick up the phone and it auto dials something else |
| 16:07.58 | bmoraca_work | Naikrovek, i believe the PAP2T will do that |
| 16:08.06 | Naikrovek | need to set up some outdoor weatherproof enclosures |
| 16:08.11 | Naikrovek | bmoraca_work: thanks |
| 16:08.29 | Katty | Naikrovek: oh? |
| 16:08.36 | Katty | Naikrovek: enclosures for what? cameras? |
| 16:08.36 | bmoraca_work | Katty, worst named person I ever knew was a girl in high school named Anita Johnson. it's childish, but i always felt bad for her |
| 16:08.51 | Katty | bmoraca_work: that's not soo bad... |
| 16:08.59 | Katty | bmoraca_work: if my name was Anita, i'd just go by Annie |
| 16:09.22 | Katty | bmoraca_work: actually i wouldn't. Anita's an awesome name |
| 16:09.22 | ManxPower-work | Better than a guy I once knew, named Jack Hoff. |
| 16:09.40 | Polysics | bmoraca_work, what do you mean with configuring the endpoints correctly? |
| 16:09.42 | bmoraca_work | Naikrovek, if you need something like an intercom or gate control, there are IP-based phones that will do that as well. |
| 16:09.54 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 16:10.04 | coppice | There are lots of sad real names: Rick Shaw, Arthur Yard, Arthur Dyer who insisted on answering the phone "Dyer 'ere" |
| 16:10.06 | bmoraca_work | Polysics, endpoints...as in SIP user agents...as in the devices that will be registering to asterisk that you're concerned about. |
| 16:10.20 | Naikrovek | bmoraca_work: don't need that, just need for people outside to be able to call inside, if they have no knowledge of anyone's extension. such as delivery drivers or people that show up before we officially open and the doors unlock |
| 16:10.32 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 16:10.48 | bmoraca_work | Naikrovek, gotcha. |
| 16:11.57 | *** part/#asterisk arossouw (n=arossouw@41.31.17.227) |
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| 16:14.29 | bmoraca_work | Polysics, generally, unless you're engaging some kind of ALG (layer 7 gateway), you will need to tell the endpoint that it is behind a NAT otherwise it will reject packets from your server because they are addressed to the outside IP address. if you configure it correctly, you shouldn't need to make Asterisk aware of the fact that the endpoint is behind a nat |
| 16:15.06 | bmoraca_work | Polysics, likewise, having nat=yes on all peers isn't necessarily a bad thing...though some endpoints don't like that (Cisco 7940s are touchy about that) |
| 16:15.20 | Polysics | bmoraca_work, i think i basically understood 1% of what yo usaid :-) |
| 16:15.50 | Polysics | bmoraca_work, let me elaborate: this particular client is a Java app on the same machine that hosts * that acts as a proxy for Flash web-based clients |
| 16:15.53 | [TK]D-Fender | thinks Polysics understood 1/2 of what he claimed... |
| 16:15.56 | Kobaz | Polysics: your nat config is screwy |
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| 16:16.33 | bmoraca_work | Polysics, you have a java sip phone that you allow people access to? |
| 16:16.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.32 | Polysics | bmoraca_work, no, the Java app runs on the server, people connect to it through RTMP, it then registers with * and routes the audio streams from RTMP to SIP |
| 16:17.57 | Polysics | which boils down to "the clients run on the same machine as *" |
| 16:18.13 | eppigy | Katty: you should stop by for dinner some time then |
| 16:18.15 | bmoraca_work | then it's always a local connection. what's the problem? |
| 16:18.19 | Katty | eppigy: kay. |
| 16:18.20 | Polysics | RTMP audio works, record/playback tests on * work |
| 16:18.30 | eppigy | I will get some really good takeout |
| 16:18.38 | Katty | lol |
| 16:18.46 | titter | Katty: mer mer mer hi. |
| 16:18.55 | Katty | Katty: mer? |
| 16:18.56 | Katty | oh |
| 16:18.58 | Katty | titter: mer? |
| 16:19.00 | Katty | titter: also, hi! |
| 16:19.07 | Polysics | bmoraca_work, the Flash clients have no microphone, and there are no particular errors, and i am under the impression there is some sort of network setup problem |
| 16:19.08 | titter | Katty: lol mer. |
| 16:19.19 | Katty | titter: mer does not parse. please try again. |
| 16:19.19 | Polysics | bmoraca_work, no microphone = don't send audio |
| 16:20.09 | titter | Katty: so I hate traveling, now I am stuck eating bad food for a few days |
| 16:20.12 | Polysics | while using a normal softphone everything works, but from inside NAT |
| 16:20.33 | bmoraca_work | Polysics, so are you trying to set up an alternative or are you trying to troubleshoot this third-party fustercluck? if the former, fix your NAT settings...if the latter, you're looking in the wrong place |
| 16:20.33 | Katty | titter: just because you travel doesn't mean you have to eat bad food. |
| 16:20.49 | bmoraca_work | Polysics, have you read the relevent portions of the book and looked at the sipnat tutorials? |
| 16:20.52 | bmoraca_work | ~sipnat |
| 16:20.53 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:20.54 | bmoraca_work | ~book |
| 16:20.55 | 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 |
| 16:21.10 | Polysics | bmoraca_work, i would just be happy with figuring out which part of the chain is causing problems |
| 16:21.26 | Polysics | and making sure NAT settings are ok is part of it |
| 16:22.24 | titter | Katty: its texas ... everything is made with some sort of fat |
| 16:22.25 | bmoraca_work | Polysics, yes, it is. it's the most important part of it. and there are two sides to the NAT settings: the server and the client. the server-side can be negated by a properly configured client. additionally, your port forward may not be correct and your clients' nat routers may not support SIP passtrhough |
| 16:23.44 | Polysics | bmoraca_work, the NATed clients work perfectly, only, the SAME SIP accounts don't work when connecting through the RTMP server, that is, locally instead of across the NAT |
| 16:24.37 | bmoraca_work | why are you using a third-party proxy server in the first place? |
| 16:25.56 | Polysics | bmoraca_work, to provide web-based calling |
| 16:26.11 | Polysics | and Flash has some advantages over activeX if i can ever make it work |
| 16:26.34 | bmoraca_work | i thought you said you were abandoning the flash because it didn't have a microphone? |
| 16:26.54 | bmoraca_work | Polysics, if you're having problems with a third party application, you need to look at that third party application's support. |
| 16:26.58 | Polysics | bmoraca_work, no, i mis-expressed myself, the flash client doesn't transmit audio |
| 16:27.13 | bmoraca_work | that sounds like a problem with the flash client |
| 16:27.18 | Katty | sighs |
| 16:28.13 | Polysics | it could be, but then again, my original question was if there was a surefire way to take NAT issues out of the picture |
| 16:28.33 | Katty | i really hate being toshiba certified |
| 16:28.35 | Polysics | as the Java part is the most complicated of the whole chain :-) |
| 16:28.36 | Katty | i really, really hate it. |
| 16:30.22 | titter | Katty: Does that mean you can fix my vcr. |
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| 16:30.31 | bmoraca_work | Polysics, the only surefire way to take NAT out of the picture is to not use NAT. |
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| 16:31.04 | bmoraca_work | Polysics, if all connections and registrations that happen with Asterisk are coming from a Proxy residing ont he same LAN as the Asterisk box, there is no NAT as far as asterisk is concerned. |
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| 16:33.01 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 16:33.15 | Polysics | bmoraca_work, what is complicating the picture is that the same SIP account could be in theory used to connect using a "standard" softphone, which would be behind NAT |
| 16:33.28 | Polysics | but i think the first test I should run is "nat=no" on one of those clients and see what happens |
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| 16:34.57 | p3nguin | If it doesn't work with nat=yes, I can't imagine how it could work with nat=no, since nat=yes just gives asterisk the ability to translate the addresses found inside the localnets setting. |
| 16:35.09 | Polysics | canreinvite=yes too, i'd say |
| 16:35.23 | p3nguin | Definitely put canreinvite to no. |
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| 16:39.37 | Polysics | so, if i were to ask "where do you think the problem lies?" you would say "in the Java or Flash part"? |
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| 16:40.56 | ManxPower-work | Polysics: I would say "Yes". |
| 16:40.57 | [TK]D-Fender | Polysics: Why is there a Java AND a Flash part? |
| 16:41.20 | ManxPower-work | You're not going to get much help with a "web phone" here. |
| 16:41.23 | Polysics | [TK]D-Fender, Flash can't connect directly to SIP, it needs "something" to act as a proxy |
| 16:41.45 | Polysics | ManxPower-work, i was just trying to rule out * as the source of errors |
| 16:41.49 | [TK]D-Fender | Polysics: which end is where in the scheme of things? |
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| 16:42.36 | ManxPower-work | Polysics: then use a phone we are familiar with. |
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| 16:43.41 | Polysics | [TK]D-Fender, the Flash client connects to the Java server using an RTMP stream. the Java server registers itself to * and pipes the RTMP audio to the * channels |
| 16:43.51 | [TK]D-Fender | Polysics: .... |
| 16:44.08 | [TK]D-Fender | Polysics: describe the #&^$ing chain including what side each PIEce is on. |
| 16:44.19 | Polysics | side of the NAT? |
| 16:44.58 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 16:44.58 | Polysics | Flash client -> Outside of NAT (every user's browser) - uses ports 5080 and a few others to connect to RTMP server |
| 16:45.21 | Polysics | s/Outside/Inside |
| 16:45.37 | Polysics | RTMP server -> runs on same machine as * |
| 16:45.50 | Polysics | that's really all |
| 16:45.59 | [TK]D-Fender | Polysics: RTMP <- what other tools do you ahve to test this? |
| 16:46.16 | Polysics | an Echo app similar to the * one, which does work |
| 16:46.32 | Polysics | and a few others i haven't tested tbh, but if one works all should |
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| 16:50.31 | [TK]D-Fender | Polysics: do record & playback work as I told you to? |
| 16:50.50 | Polysics | yes, using a normal softphone, not with the flash client (obviously) |
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| 16:52.00 | [TK]D-Fender | Polysics: Then RTMP has an issue. its on the same box as *. Shouldn't ahve a firewall issue local to the server (which you should check anyway). that aisde we have no idea about the SW you are using. |
| 16:52.17 | Polysics | [TK]D-Fender, that's partially good news |
| 16:52.32 | drmessano | Does PJSIP have a configuration, such as which RTP ports it uses? |
| 16:54.04 | Polysics | at least it restricts the operating field, which is still something good |
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| 16:55.19 | drmessano | Um ok |
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| 17:20.38 | Polysics | i might end up giving up... |
| 17:20.48 | carrar | Never Give Up |
| 17:21.01 | Polysics | there's a very good activeX client out for sale, it works wonderfully :-) |
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| 17:25.41 | voipmonk | have you tried zoipers web phone? |
| 17:25.58 | p3nguin | Is it free? |
| 17:27.07 | benngard | nice feature on the ooh323 channel, when i call from a h323 phone (avaya cm) h323 (asterisk) to a sip phone, when sip phone answers the display of the h323 phone changes to the "name of" the sip phone :) |
| 17:30.20 | leifmadsen | p3nguin: I'm not sure if really any of the web phones I've come across are free |
| 17:30.46 | Polysics | voipmonk, i have tried Zoiper, on leifmadsen recommendation too - it is actually great |
| 17:30.56 | Polysics | i just decided to give Flash one last go |
| 17:31.17 | Polysics | but apparently the project is half-dead, and it's very hard to find anyone that knows much about it |
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| 17:31.30 | leifmadsen | never heard of it |
| 17:31.30 | Polysics | so i think i will just have to go with Zoiper |
| 17:31.40 | Polysics | leifmadsen, red5phone |
| 17:32.00 | Polysics | it's not far away from being usable, but some things are just too hard to debug |
| 17:32.08 | Polysics | too many parts to the stack |
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| 17:33.23 | leifmadsen | ah, might be using this, or that web phone works with this patch: https://issues.asterisk.org/view.php?id=15484 |
| 17:34.57 | ManxPower-work | someone saying "hard to debug. too many parts to the stack" should not be trying to use aflashphone |
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| 18:05.55 | klochan | hi, is there some way to call mysql stored routines directly from asterisk? =) |
| 18:09.53 | Naikrovek | yeah i think so |
| 18:09.57 | Naikrovek | i've never done it |
| 18:10.29 | Naikrovek | can't you just use the stored procedure as a table name in the SQL query? select * from store_procedure |
| 18:11.43 | klochan | in mysql i have to "call myprocedure" |
| 18:12.50 | Naikrovek | how would you call it via sql |
| 18:13.09 | klochan | mysql cli>call my_procedure; |
| 18:13.24 | klochan | that's default procedure exec |
| 18:14.37 | Naikrovek | what is the point of a stored procedure if you can only use it at the mysql_client |
| 18:14.42 | Naikrovek | there must be another way |
| 18:15.14 | bmoraca_work | klochan, func_odbc can do that |
| 18:17.16 | klochan | Naikrovek, e.g. i can call stored routines from freeradius =) |
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| 18:17.50 | bmoraca_work | klochan, you can use func_odbc to call your stored procedures from within dialplan. http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/9079823.html has an example further down the page |
| 18:18.43 | klochan | ooh, thank you! ) |
| 18:19.24 | klochan | thank's a lot! )) |
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| 18:26.31 | [TK]D-Fender | *b00m* |
| 18:27.32 | Naikrovek | uh huh |
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| 18:39.46 | shaun_ | hello |
| 18:40.54 | shaun_ | anyone feel up to answering a few perl/cgi/asterisk questions? |
| 18:41.09 | voipmonk | ask, shaun_ |
| 18:42.08 | Naikrovek | just ask |
| 18:42.42 | shaun_ | I'm triing to set up a phone system for an answering service, I'd like it to filter calls by callerid, forward them to the correct extenstion, and display a form on a web page with information about the caller |
| 18:42.51 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:44.16 | shaun_ | I'm triing to find documentation on sending data from asterisk to another program and I'm not getting what I'm looking for |
| 18:45.06 | ManxPower-work | shaun_: AGI() or System() |
| 18:45.14 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 18:45.18 | Qwell | or use AMI |
| 18:45.35 | ManxPower-work | I guess you could fire off an AMI script from the dialplan. |
| 18:46.06 | Qwell | no need. All the necessary stuff is already sent out |
| 18:46.09 | ManxPower-work | I'm assuming he wants to do it in the dialplan |
| 18:46.54 | ArtemMakhutov | Hello, is is possible to enable jitterbuffer (jbenable=yes) on a peer basis? |
| 18:46.57 | dlynes_laptop | ManxPower-work, well, the web page with information about the caller sounds like an AMI interface, to me |
| 18:47.04 | ManxPower-work | i.e. SEND the info to the browser, not have the browser constantly polling |
| 18:47.29 | ManxPower-work | dlynes_laptop: "sending data to a browser based on the callerID" sounds like an AGI to me. |
| 18:47.34 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 18:48.02 | shaun_ | brb, I'm going to hop on a computer with more than one monitor |
| 18:49.16 | Naikrovek | multiple monitors are underrated |
| 18:49.27 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:50.50 | *** part/#asterisk shaun_ (n=shaun@cblmdm72-241-199-247.buckeyecom.net) |
| 18:50.57 | russellb | Naikrovek: agreed |
| 18:51.14 | Naikrovek | laptops need dual hdmi ports for this reason |
| 18:51.35 | russellb | I just use a docking station for that purpose :-) |
| 18:51.43 | Naikrovek | that would also work |
| 18:52.13 | bmoraca_work | ManxPower-work, the way I've always done that in the past is an application that runs on the local workstation and monitors AGI and then pops up a browser window with whatever info is needed (HTTP GET variables are helpful) |
| 18:54.05 | *** join/#asterisk shaun_ (n=lmiwtf@cblmdm72-241-199-247.buckeyecom.net) |
| 18:55.12 | shaun_ | has anyone ever used a DID truck? |
| 18:55.32 | shaun_ | trunk rather |
| 18:55.38 | [TK]D-Fender | shaun_: No, DID's are car-only options |
| 18:56.05 | [TK]D-Fender | shaun_: And that entire term, even as corrected i s vague and meaningless |
| 18:56.09 | bmoraca_work | "DID trunk"? do they still even have those? to my knowledge, no asterisk hardware supports analog did trunks. |
| 18:56.31 | [TK]D-Fender | shaun_: DID is jsut a phone number. the question is what is the CALL delivered to you over. |
| 18:56.31 | shaun_ | rhino fxs cards do aparently |
| 18:56.39 | ManxPower-work | ~trunk |
| 18:56.40 | infobot | i heard trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 18:56.54 | [TK]D-Fender | shaun_: FXS is for plugging in PHONES. Since when does a PHONE send a "DID"? |
| 18:57.07 | bmoraca_work | [TK]D-Fender, taken literally, he means an analog phone line that has more than one phone number and can receive DNIS information |
| 18:57.34 | [TK]D-Fender | bmoraca_work: Yes well there are several ways to do it. Typical is inband DTMF upon answer. |
| 18:57.44 | [TK]D-Fender | bmoraca_work: Easy little script to pull that before processing. |
| 18:57.47 | bmoraca_work | the voltages and signalling are very different from POTS analog lines and require specialized hardware |
| 18:57.50 | [TK]D-Fender | bmoraca_work: I've done it before |
| 18:58.04 | shaun_ | it's what someone at rhino told me, when you use a did line you have to use a FXS card with wink start |
| 18:58.17 | [TK]D-Fender | bmoraca_work: I've seen it on boring analog requiring no Zaptel setup or specific cards |
| 18:58.46 | [TK]D-Fender | shaun_: First thats FXO, not FXS. Second you need to confirm with your **TELCO** as to what their standard requires |
| 18:59.42 | bmoraca_work | shaun_, or just make it easy on yourself and use a real digital transport |
| 18:59.45 | shaun_ | you have to provide voltage to the line so I was told you have to use an fxs card |
| 19:00.31 | bmoraca_work | shaun_, talk to your telco |
| 19:01.26 | p3nguin | I can't imagine that you would be the one providing voltage on your phone line. That seems like something the telco normally takes care of when you pay the bill. |
| 19:01.40 | ManxPower-work | DID trunks can be set up in one of many ways. |
| 19:02.02 | ManxPower-work | p3nguin: the telco provides voltage on NORMAL lines. |
| 19:02.15 | p3nguin | yeah |
| 19:02.26 | ManxPower-work | Special access / special provisioned analog lines could be different. And an analog DID could be one of those types of lines. |
| 19:02.27 | p3nguin | Is his line an abnormal one? |
| 19:02.47 | ManxPower-work | p3nguin: not really a big deal. If he plugs the line in the wrong port chances are he'll just blow the port and have to replace it. |
| 19:03.00 | p3nguin | fun stuff. |
| 19:03.04 | ManxPower-work | Then maybe he'll contact the telco first. |
| 19:04.36 | ManxPower-work | When you're talking about analog DID, none of the normal telco rules apply. |
| 19:06.48 | dlynes_laptop | shaun_, are you referring to what's called an "analog DID trunk"? |
| 19:07.11 | drmessano | Is that like a unicorn? |
| 19:07.53 | ManxPower-work | drmessano: more like an honest politician. Rare, but rumored to exist. |
| 19:08.45 | drmessano | When he plugs that FXS card into the wall his module will be a rumor too |
| 19:08.45 | dlynes_laptop | drmessano, no...it's a group of a minimum of four analog lines that supply incoming DIDs, but you cannot specify an outgoing DID |
| 19:09.00 | [TK]D-Fender | [13:59]<shaun_>you have to provide voltage to the line so I was told you have to use an fxs card <- you do not give voltage to the TELCO, they give it to YOU |
| 19:09.03 | dlynes_laptop | drmessano, i don't know about other telcos, but Telus supplies these DID trunks |
| 19:09.31 | drmessano | Ok, how is that a trunk? |
| 19:09.43 | dlynes_laptop | drmessano, it's a popular option for some businesses that want some of the power of a pri, without needing the capacity of a pri |
| 19:09.44 | shaun_ | I want to use SIP connections anyways, DID trunk are just an industry stardard for answering services |
| 19:09.55 | ManxPower-work | [TK]D-Fender: there are types of lines where you provide voltage to the telco |
| 19:10.22 | [TK]D-Fender | ManxPower-work: W-T-F |
| 19:10.34 | dlynes_laptop | drmessano, and don't ask me how it's called a DID trunk....ask Telus...that's what they call it on your phone bill |
| 19:10.42 | bmoraca_work | shaun_, how many concurrent calls are you expecting? most answering services I deal with (generally in the medical field) are pretty high volume. probably want something along the lines of a PRI |
| 19:10.44 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 19:10.48 | drmessano | dlynes_laptop: No different than a SIP trunk |
| 19:11.09 | shaun_ | they have 120 small business customers |
| 19:11.11 | drmessano | bmoraca_work: 10,000 calls. 1 at a time |
| 19:11.14 | bmoraca_work | ManxPower-work, an OPX line, I can see that. |
| 19:11.20 | dlynes_laptop | drmessano, you can call it a potato or a tomato...I don't really care....but Telus calls it a DID trunk, so that's what all of their customers call it, too |
| 19:11.22 | ManxPower-work | bmoraca_work: *nod* |
| 19:11.46 | drmessano | dlynes_laptop: Great, so lets fire up the SIP trunk debate again too |
| 19:12.18 | shaun_ | I was thinking about having them get a t1 and use SIP with G.729, they'll be able to handle about 60 calls at once then correct? |
| 19:12.29 | bmoraca_work | "trunk" is an idiom in the telephony world. it's taken to meaning "way of reaching the PSTN or connecting PBXes". let people do what they need. |
| 19:13.01 | shaun_ | the provider wants me to use 711 but that will limit me to like 18 calls at once |
| 19:13.05 | dlynes_laptop | drmessano, anyways...I talk to telcos all day, so I need to speak their terminology, not yours |
| 19:13.05 | bmoraca_work | shaun_, if their current internet cannot handle the SIP traffic, you're not going to save any money by using SIP accounts AND paying for extra internet accounts. |
| 19:13.19 | drmessano | dlynes_laptop: It isn't *MY* terminology |
| 19:14.02 | drmessano | dlynes_laptop: and i don't care if you sleep in a cardboard box outside their headquarters, its still wrong |
| 19:16.07 | *** join/#asterisk fatnasty1 (n=chatzill@ext-52.sagetelecom.net) |
| 19:16.41 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 19:16.43 | fatnasty1 | how do i set up sip session timers? i cant find any reference to it, I assume somewhere in sip.conf |
| 19:16.53 | ManxPower-work | "Dialogic® Brooktrout® TR1034 Fax Boards do not use an external power supply, but instead get the DID voltage from the PCI bus. Once the DID line is activated, -48Vdc power must be continuous, or the telephone company may disconnect the DID service. The Brooktrout TR1034 DID Fax Board will supply the voltage to the DID line as long as the PC power is on and the Fax Board gets the voltage from the PC bus; the phone company should not put any v |
| 19:16.58 | dlynes_laptop | fatnasty1, which version of asterisk are you using? |
| 19:17.00 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.47 | fatnasty1 | dlynes_laptop: 1.6 |
| 19:17.56 | dlynes_laptop | fatnasty1, 1.6.? |
| 19:18.03 | fatnasty1 | 1.6.1.6 |
| 19:18.07 | *** join/#asterisk jdoe (i=jdoe@falseprophet.ca) |
| 19:18.09 | voipmonk | AAAAAAAAH!!!!!!!!!!! |
| 19:18.17 | ManxPower-work | fatnasty1: there were no mention of sip timers in the sip.conf.sample? |
| 19:18.19 | dlynes_laptop | fatnasty1, yeah...that version definitely supports it...one sec |
| 19:18.21 | coppice | ManxPower-work: those things are a pain. you normally want a 48V float battery to ensure they stay up |
| 19:18.25 | ManxPower-work | voipmonk: running IPv6 I see. |
| 19:18.46 | fatnasty1 | ManxPower-work: i didnt make samples, Im looking on voip-info |
| 19:18.49 | p3nguin | "outgoing DID" I have to remember that; that's a good one. |
| 19:18.49 | ManxPower-work | coppice: *nod* But it is a situation where YOU are providing voltage TO the telco. |
| 19:19.08 | ManxPower-work | fatnasty1: stop looking at old docs and read the sip.conf.sample. "make samples" will overwrite your configs |
| 19:19.20 | dlynes_laptop | fatnasty1, /usr/local/src/path/to/your/asterisk/source/code/asterisk-1.6.1.6/configs/ |
| 19:19.25 | voipmonk | smacks ManxPower-work on the head with a styrofoam box........ BOOKTROUT!!!!! BROOKTROUT!!!!!! AAAAAAAAAAAAAAAAAAAAAAH! |
| 19:19.47 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 19:19.58 | coppice | ManxPower-work: yep. its one of the reasons PBXes in many places are required to have >8 hour battery power |
| 19:20.03 | ManxPower-work | I was making a joke aboutIPV6 AAAAAAAAAAAA records (or however many A's they use) |
| 19:20.10 | drmessano | grabs a pole. some worms, and a 24 pack of budweiser |
| 19:20.11 | Naikrovek | AAAA |
| 19:20.32 | fatnasty1 | How about that, didnt know the configs/ directory existed... thanks. |
| 19:20.50 | ManxPower-work | fatnasty1: wait until you discover the docs/ directort |
| 19:20.54 | ManxPower-work | directory |
| 19:20.57 | voipmonk | the first time i held a brooktrout card I was reminded by my boss that they cost more than a few months pay for me at the time |
| 19:21.07 | dlynes_laptop | fatnasty1, where did you think make samples got the configs from? embedded docs inside the C code? |
| 19:21.10 | voipmonk | had to be like 97 or 98 ish |
| 19:21.37 | fatnasty1 | dlynes_laptop: Never crossed my mind |
| 19:21.53 | ManxPower-work | fatnasty1: the only official docs are the docs that are part of Asterisk |
| 19:22.41 | shaun_ | gahh |
| 19:23.06 | shaun_ | everything I'm reading is about managing existing calls or placing outgoing |
| 19:24.07 | ManxPower-work | shaun_: you're not going to find much info about doing what you want to do. |
| 19:25.19 | shaun_ | I'm decent with perl and there are 3 products on the market using asterisk that provide a simular service |
| 19:25.42 | dlynes_laptop | shaun_, what kind of service is that? |
| 19:25.47 | dlynes_laptop | shaun_, answering machine? |
| 19:26.04 | ManxPower-work | shaun_: Yes. Much like billing, CRM is usually a costly thing to buy. |
| 19:26.04 | shaun_ | answering service |
| 19:26.27 | [TK]D-Fender | shaun_: Sounds petty to make in * |
| 19:26.43 | ManxPower-work | shaun_: first you have to figure out how to send data to the browser -- this is not an Asterisk thing. |
| 19:27.16 | shaun_ | I've got it to filter calls by caller ID and forward them to the correct extenstion |
| 19:27.32 | ManxPower-work | shaun_: but you are ignoring the hardest part. |
| 19:27.40 | *** join/#asterisk vally (i=kvirc@ip-92-50-113-184.unitymediagroup.de) |
| 19:27.51 | shaun_ | I'm getting to it, not ignoring it |
| 19:28.13 | bmoraca_work | shaun_, don't most answering services provide each customer with their own DID for routing calls? seems that will be much more reliable than using callerid... |
| 19:28.22 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 19:29.31 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 19:29.42 | shaun_ | when you forward calls to the sip number the caller ID shows the number that was called |
| 19:30.06 | shaun_ | I'm working with a 10k budget to get this up for the first month |
| 19:30.09 | ManxPower-work | shaun_: maybe for YOU, but not for most people |
| 19:30.17 | benngard | can u help me find the english/american word for "a box that u attach to pstn, load money into it, and can make stamps) if i just translate the swedish word(s) it will be stamp-machinne |
| 19:30.27 | shaun_ | what do you mean, maybe for me? |
| 19:30.37 | ManxPower-work | benngard: "postage machine" |
| 19:30.37 | bmoraca_work | shaun_, i'll set it up for 5k and then use the next 5k to provide you with DIDs and trunking for 6 months. |
| 19:30.42 | benngard | thx |
| 19:30.49 | p3nguin | postage meter? |
| 19:30.56 | ManxPower-work | shaun_: I've never had a number forwared to sip that set the callerid to the dialed number |
| 19:31.08 | ManxPower-work | I'm not aware of any SIP providers that do that. |
| 19:31.16 | p3nguin | benngard: http://www.pitneyworks.com/ |
| 19:31.39 | shaun_ | you get the number that placed the call in the first place? |
| 19:31.47 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 19:31.50 | dlynes_laptop | shaun_, that only applies if you to an attended transfer to another number |
| 19:31.56 | ManxPower-work | shaun_: Yes. |
| 19:32.08 | bmoraca_work | DNIS vs. ANI |
| 19:32.55 | benngard | i have like 20 "postage machines" that i need to reload, vithout pstn, gonna be fun tomorrow to do it by some ata's and asterisk ;) |
| 19:33.42 | benngard | p3nguin: exactly that kind of stuff i mean |
| 19:34.00 | benngard | learned a new word |
| 19:34.56 | shaun_ | shit, I think the sales guy lied to me |
| 19:35.04 | Naikrovek | that NEVER happens |
| 19:35.10 | ktwilight | never... |
| 19:35.10 | Naikrovek | hates sales guys |
| 19:35.16 | benngard | sales guy never lie! |
| 19:35.24 | benngard | but sales girl do ;) |
| 19:35.26 | shaun_ | I've tested it though and it worked with my phone |
| 19:36.33 | shaun_ | I forwarded my number to the sip line they let me test with, called if from another phone and my number came up |
| 19:36.55 | dlynes_laptop | shaun_, it's because you're doing a forwarding |
| 19:36.58 | dlynes_laptop | shaun_, not a transfer |
| 19:37.24 | dlynes_laptop | shaun_, erm....nvm...misunderstood what you said |
| 19:37.26 | p3nguin | benngard: Are your machines printing postage onto envelopes based on weight, or just putting out stamps? |
| 19:37.37 | bmoraca_work | shaun_, depends how you forwarded the number and what your provider does. |
| 19:37.38 | shaun_ | that's what the customers will be doing, forwarding all of their calls to us after hours |
| 19:38.05 | bmoraca_work | shaun_, if they're doing it at the telco level, the callerid of the originall caller will be passed |
| 19:38.12 | bmoraca_work | (most of the time) |
| 19:38.21 | dlynes_laptop | shaun_, are you Answer()'ing the call before calling the other device? |
| 19:38.24 | paulc | shaun_ So it's fine if they have a specific DID each, you'll get the caller's caller ID |
| 19:38.47 | bmoraca_work | shaun_, the only way to accurately route like this is by DNIS. each business will need their own DID. DIDs are dirt cheap, though. |
| 19:38.49 | paulc | but you can't get all your customers to forward to the same numebr unless you get the RDNIS from yoru provider |
| 19:38.55 | benngard | p3nguin: based on weight, the small envelope, i can just slide them through and the stamp get prited, the bigger i have to make "stamps" for them |
| 19:39.06 | benngard | printed* |
| 19:39.10 | p3nguin | benngard: That's called a postage meter. |
| 19:39.18 | shaun_ | I don't really care about the callers callerid |
| 19:39.20 | paulc | Or a franking machine |
| 19:39.32 | shaun_ | so it looks like I'm going with DID |
| 19:39.33 | paulc | Neopost make them.. I used to sell them (long story!) |
| 19:39.46 | bmoraca_work | shaun_, i understand that, but you cannot accurately route this way based on callerid information |
| 19:39.59 | benngard | frankeringsmaskin in swedish, wtf didnt i wrote that word |
| 19:40.15 | benngard | we are using neopost |
| 19:40.19 | shaun_ | looks like I'm starting from scratch again, heh |
| 19:40.26 | paulc | In the UK we call them Franking Machines. No one in North America calls them that though. |
| 19:40.26 | bmoraca_work | shaun_, a "DID" is just a telephone number. a PRI or a SIP account can generally have multiple telephone number associated, and you can have asterisk route differently based on them |
| 19:41.58 | Katty | mmmm, cajun |
| 19:42.19 | shaun_ | I still need to figure out how to pass live call information to a program |
| 19:42.31 | Katty | i got a Bird Feeder Pole accessory whiel i was at the store. |
| 19:42.39 | bmoraca_work | shaun_, what program? |
| 19:42.41 | Katty | hopefullly it fits my pole..it has 3 hooks on it rather than two (= |
| 19:42.50 | Katty | i got a hanging bird bath for it |
| 19:42.58 | dlynes_laptop | Katty, btw...your camera was severely overexposed yesterday |
| 19:43.02 | bmoraca_work | shaun_, and what "live call information"? |
| 19:43.05 | Katty | dlynes_laptop: yes, i know. |
| 19:43.08 | shaun_ | I wanted to use perl with a browser |
| 19:43.08 | Katty | dlynes_laptop: and i told you why, too |
| 19:43.13 | Katty | dlynes_laptop: i guess you didn't get the message. |
| 19:43.14 | dlynes_laptop | Katty, oh...must've missed it |
| 19:43.59 | shaun_ | I want the people on the extenstion to see what customer's calling before they answer the phone |
| 19:44.03 | ktwilight | frama :) |
| 19:44.16 | *** join/#asterisk QubeZ (n=qube@64.128.254.34) |
| 19:44.17 | shaun_ | and have a form for them to take a message and email it to the client |
| 19:44.20 | QubeZ | hello all |
| 19:44.24 | shaun_ | the second part is easy |
| 19:44.25 | p3nguin | shaun_: People aren't on extensions. |
| 19:44.45 | shaun_ | the folks answering the call |
| 19:44.47 | Katty | p3nguin: who says. |
| 19:44.54 | p3nguin | PBX law |
| 19:44.56 | bmoraca_work | shaun_, so what's stoping you? there are a couple ways to do this...application running on the user's system (AMI) which pops up a web browser (there are currently apps built to do this (snap-a-number is one))...or you can build a webpage that polls a database or AJAX engine...tons of ways to do that |
| 19:44.58 | QubeZ | if i have a pc connected though my polycom's "pc" port and polycom "lan" is connected to my router. how do I get stun client running on my laptop to show my phone its public ip? |
| 19:45.07 | *** join/#asterisk cjp (i=4a73a210@gateway/web/freenode/x-keocludarrsmhyxr) |
| 19:45.09 | bmoraca_work | p3nguin, don't start with that shit again |
| 19:45.10 | Katty | p3nguin: show me your badge. |
| 19:45.12 | Katty | p3nguin: officer. |
| 19:45.22 | cjp | hi - how can i check which modules are loaded from the CLI? |
| 19:45.35 | QubeZ | cjp show modules |
| 19:45.41 | cjp | merci |
| 19:45.50 | p3nguin | modules show |
| 19:45.55 | QubeZ | show modules like func <-- will show you all the func modules |
| 19:46.07 | p3nguin | modules show! |
| 19:46.11 | QubeZ | p3nguin or 'module show' |
| 19:46.13 | Katty | p3nguin: simmer. |
| 19:46.16 | Katty | p3nguin: down. |
| 19:46.18 | QubeZ | i dont see a modules show |
| 19:46.27 | QubeZ | its not plural |
| 19:46.42 | [TK]D-Fender | QubeZ: You usually don't have to set anything on Polycom's to work behind NAT. Only seetings are * side |
| 19:46.44 | shaun_ | snap a number seems to do the reverse of what I need |
| 19:47.01 | p3nguin | You're right. I got carried away on the keyboard and made it plural when it isn't. |
| 19:47.11 | Katty | p3nguin: what's got your panties in a bunch today. |
| 19:47.28 | QubeZ | [TK]D-Fender i have nat=yes and externip set on my asterisk but i keep getting these errors: [Jan 25 14:13:04] WARNING[28287]: chan_sip.c:1981 retrans_pkt: Maximum retries exceeded on transmission 9fead017-b8477625-56e055f2@192.168.1.66 for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt. |
| 19:47.32 | bmoraca_work | shaun_, no, snap-a-number does EXACTLY what you need. it will launch a web browser with callerid information as HTTP GET variables pointed to whatever script you want. |
| 19:47.34 | cjp | i'm having the stranges problems since installing chan_skype.so . first of all, i get a crash on reload about 70% of the time. secondly, my iax2 devices won't register |
| 19:47.54 | [TK]D-Fender | QubeZ: Show me an actual call with actual SIP debug, and actual configs and maybe we'll find the problem. |
| 19:48.01 | ManxPower-work | QubeZ: try using localnet= too |
| 19:48.06 | QubeZ | [TK]D-Fender my phone is not registering |
| 19:48.17 | QubeZ | ManxPower-work i have 192.168.0.0/255.255.0.0 in my localnet as well |
| 19:48.21 | [TK]D-Fender | QubeZ: And where's the SIP DEBUG of that for me to look at? |
| 19:48.34 | shaun_ | alright, I'm going to do some deeper reading into ADA |
| 19:48.38 | shaun_ | thanks for all of the help |
| 19:48.45 | cjp | i have reported the crash on reload to digium, as it's freakin serious. we've been dropping channels all day. it was happening on 1.4.25 and we did a rebuild to 1.4.29 but it didn't solve the problem |
| 19:48.55 | bmoraca_work | shaun_, ADA is a huge hunk of crap. see if you can find the original Snap-A-Number anywhere |
| 19:49.00 | cjp | what's got me know though is why the hell my IAX2 channels won't register |
| 19:49.11 | shaun_ | damnit |
| 19:49.16 | cjp | this is a stable system that has been running for more or less 4 years |
| 19:49.44 | shaun_ | where's a good place to order poloycom phones? |
| 19:50.10 | bmoraca_work | telephonydepot.com seems pretty decent |
| 19:50.10 | QubeZ | [TK]D-Fender how do i get that for you? |
| 19:50.22 | [TK]D-Fender | QubeZ: * CLi |
| 19:50.35 | QubeZ | set verbose 100 ? |
| 19:50.40 | [TK]D-Fender | SIP DEBUG |
| 19:50.44 | [TK]D-Fender | QubeZ: "help sip" |
| 19:50.57 | Katty | why can't someone make a bird feeder which can easily accomodate a blue jay |
| 19:51.00 | Katty | why are they all so /tiny/ |
| 19:53.44 | *** join/#asterisk brezular (n=brezular@adsl-dyn161.91-127-129.t-com.sk) |
| 19:56.57 | *** join/#asterisk Alagar (n=Administ@122.164.34.213) |
| 19:59.49 | QubeZ | [TK]D-Fender http://pastebin.com/m5b32ce15 |
| 20:00.12 | cjp | any idea on why IAX2 devices suddenly don't register? it's not like i'm complete newbie here, I have been running this system for 4 years |
| 20:00.45 | cjp | dialplan, in other words, is fine |
| 20:01.00 | bmoraca_work | cjp, i'm guessing an IAX2 debug might shed a little more light on what's going on. |
| 20:01.09 | [TK]D-Fender | QubeZ: Now try looking at the COMPLETE COMMUNICATION |
| 20:01.12 | cjp | yeah, it does very little |
| 20:01.14 | cjp | here: |
| 20:01.43 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 20:01.51 | cjp | when i try to register, here is the debug: |
| 20:01.51 | cjp | Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00003ms SCall: 03777 DCall: 00001 [95.95.184.198:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00003ms SCall: 00468 DCall: 00000 [74.115.162.16:39251] USERNAME : 107 REFRESH : 3600 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK |
| 20:02.00 | bmoraca_work | ~pastebin |
| 20:02.01 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:02.12 | QubeZ | [TK]D-Fender check this out: http://pastebin.com/m11c5c972 |
| 20:02.18 | QubeZ | complete communication |
| 20:02.21 | cjp | unfortunately, i can't read that |
| 20:03.05 | *** join/#asterisk klochan (n=klochan@95-27-73-123.broadband.corbina.ru) |
| 20:04.02 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:04.19 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
| 20:04.33 | [TK]D-Fender | QubeZ: There is no register in there |
| 20:04.54 | QubeZ | thats all that scrolls by |
| 20:05.23 | *** part/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 20:05.58 | [TK]D-Fender | QubeZ: You tell me it fails to register. You aren't showing me it failing to register. I don't see peer status. I don't see a call attempt from it. I don't see a call attempt to it. |
| 20:06.20 | QubeZ | peer status: 12410 (Unspecified) D N 0 Unmonitored |
| 20:06.45 | [TK]D-Fender | QubeZ: restart the phone and watch the regiswter attempt |
| 20:07.53 | QubeZ | via the sip debug? sorry, im very new to this and trying to get a phone to register over DSL |
| 20:08.28 | *** join/#asterisk ChrisWi (n=admin@mx2.wwserver.net) |
| 20:08.52 | QubeZ | [TK]D-Fender rebooting phone now |
| 20:09.24 | cjp | what should i see when a phone tries to register over IAX2? |
| 20:09.45 | ChrisWi | how is it possible to compile asterisk with h323, when I try to build a rpm ? |
| 20:12.35 | p3nguin | chriswi: Use checkinstall to install it after building from source. checkinstall should be able to roll your files into an rpm for you. You'll just need to set the prefix and sysconfdir options when you configure it. |
| 20:13.27 | benngard | h323 and asterisk thats what i am struggling with |
| 20:13.29 | cjp | what should i see when a phone tries to register over IAX2? |
| 20:13.36 | ChrisWi | I have pwlib and openh323 coming via rpm, |
| 20:14.20 | ChrisWi | the configure succeed, but the channels/h323/Makefile fails. |
| 20:14.35 | p3nguin | Did you already install those dependency rpms? |
| 20:14.38 | ChrisWi | because it wants to include the openh323.mak |
| 20:14.40 | benngard | ChrisWi: maybe some are gonna kill me but go for addons/ooh323 instead |
| 20:15.00 | ChrisWi | whats that, deps ? |
| 20:15.07 | Qwell | ChrisWi: what actually happens when you try? |
| 20:16.25 | benngard | ChrisWi: i have the addons/ooh323 up and running, it doesnt need any pwlib and stuff like that |
| 20:16.35 | p3nguin | chriswi: If your build will need h232, then you obviously need to install the h323 rpm first. |
| 20:16.47 | p3nguin | s/h232/h323/ |
| 20:17.44 | ChrisWi | I did, and the ast configure finds /usr/include/openh323/h323.h and /usr/include/ptlib.h |
| 20:17.56 | p3nguin | Is that acceptable? |
| 20:18.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:18.32 | benngard | ChrisWi: what version of asterisk are u trying to compile? |
| 20:18.40 | ChrisWi | but I hassle with this: include $(OPENH323DIR)/openh323u.mak |
| 20:18.48 | ChrisWi | benngard: 1.4.29 |
| 20:18.58 | benngard | sorry to old for me |
| 20:19.03 | p3nguin | include $(OPENH323DIR)/openh323u.mak fails? |
| 20:19.09 | p3nguin | benngard: OLD? It's brand new. |
| 20:19.13 | ChrisWi | i also looked into 1.6.2.1, but this stuff didn't changed |
| 20:19.17 | benngard | but did u do a proper EXPORT? |
| 20:19.20 | p3nguin | It was just released within the past 48 hours. |
| 20:19.28 | ChrisWi | so it makes no difference wich version to take |
| 20:19.38 | Naikrovek | "Config file error error is 0x0" |
| 20:19.41 | Naikrovek | hm |
| 20:19.45 | benngard | the asterisk yes, but the h323 driver is old' |
| 20:20.37 | p3nguin | chriswi: If you know where openh323u.mak is, then set your OPENH323DIR environment variable to be the path to it. |
| 20:20.39 | ChrisWi | yes, I already reconized this. branched to "opalvoip" and "h323+" |
| 20:21.33 | ChrisWi | this file came with the openh323 rpm and resides in /usr/share/openh323/ |
| 20:22.17 | ChrisWi | but inside there are settings for: OPENH323DIR = /usr/src/packages/BUILD/openh323_v1_19_0_1 |
| 20:22.41 | ChrisWi | which does not exists, it has existed, when buildung openh323 rpm |
| 20:23.01 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-131.gtconnect.net) |
| 20:23.47 | ChrisWi | isn't there a way to build the "chan_h323.so", with -I/usr/inlude/openh323 ? |
| 20:24.06 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 20:24.13 | ChrisWi | I think the make files should be rewritten, to get this stuff work |
| 20:24.37 | ChrisWi | for me it looks like having openh323 as source, not as rpm. |
| 20:25.47 | Corydon76-lap | ChrisWi: that's because it uses a set of files to get the driver to build that is not installed with the regular targets |
| 20:26.08 | Corydon76-lap | but very few people use h323 anymore anyway |
| 20:27.00 | *** join/#asterisk stix (n=stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:27.16 | benngard | i do |
| 20:27.26 | ChrisWi | Corydon76-lap: so your advice would be not to compile h323 in ? |
| 20:27.47 | benngard | if u dont need h323 dont |
| 20:27.59 | benngard | but if u need it... |
| 20:28.31 | Corydon76-lap | ChrisWi: that, or get a distro that installs the .mak file correctly (like Ubuntu) |
| 20:28.37 | ChrisWi | thougt it is for video conferencing, isn't it ? |
| 20:28.57 | Corydon76-lap | ChrisWi: No, it's a separate protocol. Most people use SIP nowadays |
| 20:29.39 | Corydon76-lap | Video is a codec, which is completely separate and agnostic about what protocol is used to carry it |
| 20:29.46 | p3nguin | I can't believe you would advise to completely change operating systems because of a single file. |
| 20:29.50 | QubeZ | can someone help me troubleshoot what is happening here: http://pastebin.com/m54a69ce9 -- my phone is not registering over the internet using SIP. I've opened the ports on PIX (5060, 10000-20000). Used externsip, nat=yes and localnet on my * server. |
| 20:30.07 | p3nguin | Don't like the color of front door on your house? MOVE! |
| 20:30.19 | Corydon76-lap | p3nguin: I advise changing from using RH purely because it's RH, not because of a single file. |
| 20:30.32 | voipmonk | move? u mean go to home depot and change it? |
| 20:30.40 | [TK]D-Fender | QubeZ: SIP/2.0 403 Forbidden (Bad auth) |
| 20:30.46 | [TK]D-Fender | QubeZ: Has nothing to do with NAT |
| 20:30.49 | ChrisWi | Corydon76-lap: so there is no really need to have h323 ? |
| 20:31.05 | [TK]D-Fender | QubeZ: NAT would stop the packets from making it from A to B. Its arriving and * is saying "GTFO" |
| 20:31.10 | Corydon76-lap | p3nguin: that's like saying if you don't like living in a garbage heap, you should repaint your front door. |
| 20:31.21 | Corydon76-lap | ChrisWi: no need at all |
| 20:31.41 | QubeZ | [TK]D-Fender hmm bad auth, we're using the same user/pass for all the phones here |
| 20:31.49 | Corydon76-lap | Well, other than for people who need to integrate with existing h323 endpoints |
| 20:31.52 | [TK]D-Fender | QubeZ: BAD |
| 20:31.54 | ChrisWi | thanks, I will skip it, then. |
| 20:32.03 | carrar | hahah |
| 20:32.06 | ChrisWi | you made me happy :) |
| 20:32.09 | [TK]D-Fender | \qubSet them sanely, and individually |
| 20:32.16 | carrar | every phone is extension 100 and pass 100 |
| 20:32.21 | carrar | thats rocks |
| 20:32.34 | QubeZ | well no, its <ext> for username and same pass for password |
| 20:32.50 | QubeZ | so in this case its user: 12410 pass: fppass |
| 20:33.08 | p3nguin | You should put more numbers in your extensions. |
| 20:33.25 | carrar | at least 10 |
| 20:33.26 | p3nguin | Make it more than the actual DID phone number. |
| 20:33.31 | Corydon76-lap | You should also make your passwords sufficiently difficult to GUESS |
| 20:33.44 | QubeZ | i understand but this is a lab |
| 20:33.46 | QubeZ | im testing |
| 20:34.16 | p3nguin | Learning the right way while still in testing would be the best. |
| 20:34.16 | Corydon76-lap | usernames are nothing. They're generally in plaintext in the protocol. It's the PASSWORD that needs to be long, random, and difficult to predict |
| 20:34.17 | QubeZ | all my lab phones have basic setup so i can test registration via internet but its not working. Bad auth *shrug* |
| 20:35.00 | carrar | QubeZ, remove the internet part and just get a local phone working |
| 20:35.21 | QubeZ | carrar i have several 124xx phones working locally |
| 20:35.39 | QubeZ | 12400 and 12406 are working and have been. Just trying to get 12410 to work over internet. |
| 20:36.58 | carrar | QubeZ, you're not using sip fixup |
| 20:37.01 | carrar | on the pix |
| 20:37.05 | *** join/#asterisk giantrobot (n=giantrob@74-133-4-226.dhcp.insightbb.com) |
| 20:37.09 | carrar | shouldn't be using |
| 20:37.22 | QubeZ | shouldn't be using sip fixup? im not, just inspect sip is the only config line |
| 20:38.14 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 20:38.20 | murdock_ut | Howdy all. |
| 20:38.27 | QubeZ | ...udp any host PUBLIC_IP eq sip <-- assuming here sip is 5060 |
| 20:39.00 | carrar | back in 2007 I used a pix once and had to add 'fixup protocol sip 5060' & 'fixup protocol sip udp 5060' |
| 20:39.09 | carrar | not sure if that will help you at all |
| 20:39.19 | carrar | err no fixup protocol sip 5060 |
| 20:39.25 | carrar | and no fixup protocol sip udp 5060 |
| 20:39.35 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:39.52 | carrar | pix code has change I'm sure since then |
| 20:42.11 | carrar | get 12410 working locally first |
| 20:44.01 | cjp | can someone tell me if my dialplan says this: exten => 106,4,Dial(SIP/106,60) |
| 20:44.19 | cjp | why the CLI shows this: -- Executing [106@from-internal:4] Dial("SIP/107-0000000d", "SIP/106|60") in new stack |
| 20:44.45 | cjp | i don't recall seeing that first part of the dial statement previously - has something been changed to make this now visible? |
| 20:44.57 | voipmonk | are you using a device named 107 to call 106? whats before the 4th priority ? show your dialplan on pastebin.ca |
| 20:45.03 | cjp | yes |
| 20:45.29 | murdock_ut | I friend who sells PBX for a living was showing me a cool feature and I was wondering if such an animal was in asterisk. I know it could be accomplished using agi, but I thought I would ask if there was something more native. In a dial plan you somehow set variables and their values and then tell the system to send these variables to a webpage via post or get. In that webpage the user... |
| 20:45.31 | murdock_ut | ...would have whatever logic they wanted and then return any information need back via xml. The pbx would then receive that xml and out create and populate any variables in that xml and make make it available to the dialplan. |
| 20:46.07 | [TK]D-Fender | murdock_ut: System() |
| 20:46.11 | murdock_ut | shish, bad grammer. |
| 20:46.51 | cjp | voipmonk: http://pastebin.ca/1765658 |
| 20:46.59 | cjp | thanks |
| 20:47.04 | voipmonk | murdock_ut: yes.... but there are better ways of doing that |
| 20:47.25 | voipmonk | great, cjp now show some sip debug |
| 20:47.51 | bmoraca_work | murdock_ut: yes, System() could do that...or I believe there's a CURL module for Asterisk now...that would do it too |
| 20:47.55 | murdock_ut | voipmonk: I know you could use something like curl I think in an php agi script |
| 20:48.08 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 20:48.20 | voipmonk | no i would probably do something else... :) |
| 20:48.20 | [TK]D-Fender | No need for AGI |
| 20:48.49 | murdock_ut | Personally I would like to make use of Asterisks native functions and avoid agi if at all possible. |
| 20:48.50 | Geminizer | Hello all. Has anyone ever used voicetrading? |
| 20:49.45 | Katty | Geminizer: no, but i recently tried Blistex's medicated lib balm with sunscreen. |
| 20:49.52 | Katty | Geminizer: it's not all that great :< |
| 20:50.06 | p3nguin | lol |
| 20:51.06 | Naikrovek | lol |
| 20:51.07 | Katty | Carmex Moisture Plus (with SPF 15) is lots better |
| 20:51.14 | Naikrovek | carmex > * |
| 20:51.22 | Naikrovek | imho |
| 20:51.30 | Katty | well, i mostly agree with that |
| 20:51.30 | Geminizer | Yeah... finding out the hard way :) I have followed their documentation to set up a sip trunk, and keep getting "Everyone is busy/congested at this time" |
| 20:51.39 | Katty | except that Blistex bought out a company named Lip Medex |
| 20:51.45 | Katty | it's a little blue tin |
| 20:51.48 | Naikrovek | has a lesson for all Polycom users. double check your config files before you reboot all your phones. |
| 20:51.49 | p3nguin | I'll stick to my Chap Stick medicated. |
| 20:51.53 | voipmonk | Geminizer: show your debug - use pastebin.ca |
| 20:51.53 | Katty | Lip Medex > * |
| 20:52.04 | Katty | p3nguin: try some lip medex. i bet you won't switch back |
| 20:52.10 | Geminizer | http://pastebin.com/m6146d8d7 |
| 20:52.19 | p3nguin | If it comes in a stick, I might consider it. |
| 20:52.50 | Katty | idk if it comes in a stick |
| 20:53.00 | Katty | http://www.wekenshop.com/images/66139.jpg <- google seems to think it does. |
| 20:53.12 | Katty | but idk if that's the same as the stuff in the tin. the stuff in the tin is amazing |
| 20:53.58 | [TK]D-Fender | Geminizer: Meaningless. Go look at SIP DEBUG for the call |
| 20:54.45 | paulc | Hmm.. CURL vs AGI.. discuss? |
| 20:54.59 | bmoraca_work | Geminizer: decent rates, but how can they not be losing their shirt with 1-second billing intervals calling places like Mexico which has 60/60 local billing? |
| 20:55.02 | Katty | will there be heat damage? |
| 20:55.39 | bmoraca_work | Geminizer: nevermind, I was looking at the EU rates and thinking they were in USD. those rates aren't very good :) |
| 20:57.46 | Naikrovek | anyone know of a way to remotely reboot polycom phones without using the webui to do it |
| 20:58.50 | p3nguin | That's one thing I like about using SCCP on Cisco phones. I can reset the phones from * CLI. |
| 20:59.11 | Naikrovek | cruising the entire building rebooting phones is not the best way to spent 30 minutes |
| 20:59.15 | Naikrovek | would rather do it remotely |
| 21:00.44 | [TK]D-Fender | Naikrovek: PoE switch. Unplug. Plug. |
| 21:00.55 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 21:01.00 | Naikrovek | [TK]D-Fender: yeah that's not a bad idea. gotta get the poe switch though |
| 21:01.26 | [TK]D-Fender | Naikrovek: You mean you aren't running that on a rack with a networked PDU? |
| 21:01.43 | Naikrovek | yup |
| 21:01.59 | Naikrovek | man i inherited this setup and the owners are SUPER cheapasses |
| 21:03.33 | *** join/#asterisk aidinb (n=Aidin@166.190.249.120) |
| 21:04.09 | Naikrovek | i have switches scattered all over because they were shocked at what cat5e cost on a spool |
| 21:04.09 | Naikrovek | ! |
| 21:04.26 | Naikrovek | i wanna consolidate all those, of course, but am getting pushback |
| 21:04.39 | TheDavidFactor | I upgraded to 1.6.2.1 and I'm seeing a weird issue. Pastebin: http://pastebin.com/d2acfb06b the macro argument comes in and is present on line 6, but is empty on line 14. Line 7 was just something I tried. Can anyone give me any suggestions? |
| 21:05.41 | Naikrovek | TheDavidFactor: what did you upgrade from |
| 21:06.19 | voipmonk | whats your debug tell you , TheDavidFactor ? |
| 21:06.23 | [TK]D-Fender | TheDavidFactor: You don't make IVR's in macro's. Shoot On Sight Capital Offense |
| 21:06.36 | *** join/#asterisk bio-tty (n=c@62.70.2.252) |
| 21:06.40 | Naikrovek | SOSCO |
| 21:06.41 | bio-tty | i have a sip question -- is an invite server transaction erased once a 2xx has been sent from TU? |
| 21:06.45 | bio-tty | and if this is the case (and therefore a new server transaction created if the same INVITE is received again due to retransmit) then how is the completed dialog supposed to detect and handle the retransit? cseq? upper via;branch? if the latter, then the concept (of rfc 3261) seem to duplicate transaction-logic in the dialog. |
| 21:06.53 | TheDavidFactor | 1.4.2x to 1.6.1.6 to 1.6.2.1 but I don't think I ever tried using this particular macro on 1.6.1.6 |
| 21:07.17 | [TK]D-Fender | TheDavidFactor: And you are jumping out of "s" and thinking a a macro will survive. Once you hard-goto youa re asking for trouble. |
| 21:07.51 | TheDavidFactor | ok, thanks! |
| 21:10.09 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:10.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:12.16 | dlynes_laptop | [TK]D-Fender, why are ivr's a bad thing in macros? |
| 21:12.35 | Qwell | dlynes_laptop: because it changes the extension |
| 21:12.54 | Qwell | exiting the macro becomes rather tricky |
| 21:12.55 | dlynes_laptop | Qwell, ok, but you can pass the extension as an argument to the macro |
| 21:13.03 | ManxPower-work | The first thing I do in almost all my macros is to change the extension. |
| 21:13.22 | ManxPower-work | exten => s,1,Goto(${MACRO_EXTEN},1) |
| 21:13.35 | ManxPower-work | Used to do that for CDRs, but I don't think it's really needed anymore. |
| 21:13.37 | dlynes_laptop | ManxPower-work, what's wrong with the 's' extension? |
| 21:13.47 | ManxPower-work | dlynes_laptop: "s" doesn't tell much in the CDRs |
| 21:13.52 | bio-tty | doesnt people use ael? |
| 21:13.59 | ManxPower-work | dlynes_laptop: use a gosub instead of a macro if using 1.6 |
| 21:14.15 | dlynes_laptop | ManxPower-work, oh...thought the cdr would've recorded the extension before it went to the macro |
| 21:14.47 | dlynes_laptop | ManxPower-work, or you're talking for outbound calling? not inbound? |
| 21:14.49 | ManxPower-work | dlynes_laptop: there were changes (1.6?) which made CDRs accurate for macros |
| 21:15.15 | dlynes_laptop | ManxPower-work, ah...I didn't actually start using macros much until after I started using 1.6 |
| 21:15.40 | dlynes_laptop | ManxPower-work, previously, I mostly used includes |
| 21:15.43 | ManxPower-work | dlynes_laptop: You know Macro() is deprecated, right? |
| 21:15.49 | dlynes_laptop | ManxPower-work, it is? |
| 21:16.07 | dlynes_laptop | ManxPower-work, odd that it doesn't give you a warning about it, then |
| 21:16.09 | ManxPower-work | and in 1.6 there is no reason to not use a Gosub. |
| 21:16.14 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 21:16.17 | paulc | GoSub is the new preferred method? I think I read that somewhere? |
| 21:16.21 | dlynes_laptop | ManxPower-work, gosubs are cdr accurate? |
| 21:16.30 | ManxPower-work | dlynes_laptop: It's such an important app I doubt they are going to remove it anytime soon. |
| 21:16.33 | ManxPower-work | dlynes_laptop: I do not know. |
| 21:17.00 | dlynes_laptop | ManxPower-work, well, you said it was deprecated, which warrants a warning, not an error...obsoleted would warrant an error |
| 21:17.11 | ManxPower-work | I last visited the issue before Gosub accepted options so it made it a non-starter for me. I first visited the issue before Gosub even existed. |
| 21:17.37 | dlynes_laptop | so everything macro does, gosub also does? |
| 21:18.21 | ManxPower-work | dlynes_laptop: in 1.6.x+ yes, not for 1.4.x |
| 21:18.26 | *** join/#asterisk rizwank (n=rizwank@76.89.131.47) |
| 21:19.00 | ManxPower-work | The only thing missing, as far as I know is a GOSUB_EXTEN like like MACRO_EXTEN. |
| 21:19.12 | rizwank | Can I get some sort of status back from a Dial() to react to how the call ended? For instance, I want to detect when the Dial(Skype ...) completes a call versus fails due to capacity or something. (I assume the Dial() is blocking - I can't run any code during it.) |
| 21:19.24 | ManxPower-work | rizwank: "core show application Dial) |
| 21:19.35 | ManxPower-work | notice the variables that Dial sets. |
| 21:20.08 | rizwank | check. thanks. |
| 21:20.33 | rizwank | Are commands like Dial and playback usually blocking (unless the application specifically states that it's waiting for dtmf) |
| 21:20.34 | ManxPower-work | rizwank: you won't find much help for Skype here. |
| 21:20.49 | leifmadsen | ManxPower-work: we don't remove applications for backwards compatibility |
| 21:21.15 | leifmadsen | GoSub() is preferred over Macro() |
| 21:21.23 | ManxPower-work | leifmadsen: just functions and variables and priority jumping, I guess applications are the only thing not changed./ |
| 21:21.26 | paulc | rizwank: yes - they block. See dial parameter "g" to continue on in the dialplan after call end |
| 21:21.41 | rizwank | awesome. thanks. |
| 21:21.47 | rizwank | time to read over that |
| 21:22.00 | leifmadsen | ManxPower-work: previously things got removed -- that is no longer the case within the last year or so |
| 21:22.03 | dlynes_laptop | ManxPower-work, ah...thanks...I never use MACRO_EXTEN, anyways |
| 21:22.12 | ManxPower-work | * The CallerPres application has been removed. Use SetCallerPres |
| 21:22.12 | ManxPower-work | <PROTECTED> |
| 21:22.24 | ManxPower-work | I suspect I can find more examples. |
| 21:22.29 | leifmadsen | ManxPower-work: what version was that introduced in |
| 21:22.34 | dlynes_laptop | leifmadsen, ummmm....yes you do remove applications :) |
| 21:22.34 | p3nguin | Is there anything like Dial()'s 'g' option to use with Queue() so that after one side of the calls hangs up, it can continue, or does that fall entirely onto the 'h' extension? |
| 21:22.35 | leifmadsen | ManxPower-work: like I said -- previously we removed |
| 21:23.01 | dlynes_laptop | leifmadsen, ah....when is 'previously'? |
| 21:23.06 | leifmadsen | no one reads |
| 21:23.14 | leifmadsen | look at what I just said 2 lines ago |
| 21:23.20 | dlynes_laptop | ah...last year or so |
| 21:23.31 | ManxPower-work | that's pretty recently. |
| 21:23.38 | dlynes_laptop | i would say |
| 21:23.49 | ManxPower-work | leifmadsen: backwards compatible for ever? |
| 21:23.51 | leifmadsen | that would be correct |
| 21:24.12 | ManxPower-work | That's sad. |
| 21:24.13 | leifmadsen | ManxPower-work: features that have better methods will remain, but may not be supported |
| 21:24.31 | leifmadsen | ManxPower-work: that other comment was directed at: "that's pretty recently" |
| 21:24.32 | ManxPower-work | leifmadsen: like DBGet/DBPut/DBDel? |
| 21:24.52 | ManxPower-work | Just how DO you delete an entry in AstDB anymore? |
| 21:24.58 | leifmadsen | even if the code exists, that doesn't mean we can't change the default compile time options in menuselect to not be selected |
| 21:25.08 | leifmadsen | ManxPower-work: DB_DELETE() I believe |
| 21:25.29 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:25.34 | bmoraca_work | all the AstDB stuff got changed to functions instead of apps...weird dichotamy, but i guess it works |
| 21:25.47 | dlynes_laptop | leifmadsen, DBDel() and DBdeltree() |
| 21:26.01 | dlynes_laptop | leifmadsen, or are those removed as of asterisk 1.6.2? |
| 21:26.17 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 21:26.22 | dlynes_laptop | leifmadsen, they still exist in 1.6.1 |
| 21:26.24 | leifmadsen | dlynes_laptop: look at menuselect to determine -- I don't have the list of available applications and functions memorized for each version |
| 21:26.42 | ManxPower-work | pbx*CLI> core show FUNCTION DBDel |
| 21:26.42 | ManxPower-work | No function by that name registered. |
| 21:26.47 | leifmadsen | that's not a function |
| 21:26.50 | ManxPower-work | nevermind, that's not all caps |
| 21:26.50 | leifmadsen | that's an application |
| 21:27.01 | leifmadsen | plus the caps |
| 21:27.32 | p3nguin | [Jan 25 15:26:34] WARNING[6442]: app_ivrdemo.c:98 skel_exec: skel requires an argument (filename) |
| 21:27.34 | dlynes_laptop | ManxPower-work, DBDel()/DBdeltree() are applications, DB_DELETE(), DB(), DB_EXISTS() are all functions |
| 21:28.10 | p3nguin | This is what happens when calling IVRDemo... but core show application IVRDemo does not state anything about skel filename as an arg. |
| 21:28.44 | leifmadsen | p3nguin: look at the code -- sounds like someone developed IVRDemo from app_skel.c and didn't change something |
| 21:28.59 | leifmadsen | p3nguin: app_skel.c is a skeleton application for an example of how to start building your own application |
| 21:30.04 | bmoraca_work | that sounds like FUN! |
| 21:30.14 | dlynes_laptop | Kinda silly though that DBDel() and DB_DELETE() both exist, though |
| 21:30.16 | p3nguin | This seems like a bug in the application description. |
| 21:30.21 | leifmadsen | right |
| 21:30.41 | leifmadsen | dlynes_laptop: why? if you have an old dialplan about 10,000 lines long, it's nice when you can continue using the old method |
| 21:30.56 | dlynes_laptop | and DB_DELETE() should delete a tree if you specify a tree, or a key if you specify a key |
| 21:31.02 | leifmadsen | dlynes_laptop: hence the backwards compatibility -- if you don't need DBdel(), then just don't compile it |
| 21:31.17 | dlynes_laptop | leifmadsen, well, because your old dialplan will break anyways, if DBPut()/DBGet() are gone |
| 21:31.27 | leifmadsen | dlynes_laptop: hence why we offer both... |
| 21:31.31 | leifmadsen | I don't get your point |
| 21:31.46 | ManxPower-work | I'm just worried about code bloat |
| 21:31.57 | p3nguin | Back to my previous comment, "core show application IVRDemo" does not state that it requires a file name nor a skeleton application. It doesn't state that there are ANY args for IVRDemo at all. |
| 21:31.59 | leifmadsen | ManxPower-work: everything is a module -- just don't compile what you don't need |
| 21:32.01 | dlynes_laptop | leifmadsen, why not get rid of all the old db remnants instead of only some of them, thus confusing people like ManxPower-work |
| 21:32.50 | leifmadsen | dlynes_laptop: because when stuff was removed, people got all bitchy they couldn't upgrade without having to refactor their dialplan, and when we don't remove stuff, people get all bitchy that we have provided too many options |
| 21:33.15 | ManxPower-work | dlynes_laptop: I suspect my misunderstanding came about from the mailing lists discussions back when DB() was being discussed. I'm trying to update my info from 1.2 to 1.6.x, but it's a slow process. |
| 21:33.18 | dlynes_laptop | leifmadsen, ok...weird |
| 21:33.22 | p3nguin | So keep the deprecated apps and always issue warnings on the CLI. |
| 21:33.43 | leifmadsen | people are welcome to file bugs as always |
| 21:33.57 | Geminizer | ok, the SIP DEBUG dump --> http://pastebin.com/m439f8d2a <-- even more cryptic than basic verbosity :) |
| 21:34.07 | dlynes_laptop | leifmadsen, if you deprecate stuff and then keep it deprecated for two major versions before obsoleting it, it should give everyone plenty enough warning on what to upgrade in their dialplans |
| 21:34.31 | dlynes_laptop | leifmadsen, as long as you warn about the deprecations, much like Java does |
| 21:34.31 | ManxPower-work | dlynes_laptop: that's what they used to do. |
| 21:34.40 | leifmadsen | dlynes_laptop: ya you'd think so, but there is always something to complain about -- you're welcome to bring this up on a mailing list for discussion by the community |
| 21:34.46 | ManxPower-work | Personally if you are upgrading between major versions you should just shut up and fix your dialplan. |
| 21:34.53 | dlynes_laptop | ManxPower-work, no kidding |
| 21:35.00 | p3nguin | I agree. |
| 21:35.09 | dlynes_laptop | ManxPower-work, java programmers don't seem to have a problem with it...don't know why asterisk programmers would |
| 21:35.16 | p3nguin | Major versions, absolutely. |
| 21:35.19 | leifmadsen | moves onto something more productive |
| 21:35.32 | dlynes_laptop | snickers at leifmadsen. |
| 21:36.10 | Geminizer | Asterisk doesn't suck per-se... just the availability of reliable sip providers |
| 21:36.11 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 21:36.17 | ManxPower-work | I must admit I'm surprised that the new release model does seem to have fewer horribly broken releases. |
| 21:36.25 | ManxPower-work | Geminizer: how long ago did someone ask you for that pastebin? |
| 21:36.48 | Geminizer | hmmm.. 20 min at least |
| 21:36.50 | dlynes_laptop | ManxPower-work, well, based on my experience, 1.6.1 series is very stable compared to previous versions |
| 21:36.58 | p3nguin | geminizer: What country are you in? |
| 21:37.01 | Geminizer | US |
| 21:37.16 | p3nguin | geminizer: What's the problem with VoIP.ms or Flowroute? |
| 21:37.18 | leifmadsen | I have at least 2 SIP providers who are very reliable |
| 21:37.32 | leifmadsen | bandwidth.com for US, Unlimitel.ca for Canada -- never have issues |
| 21:37.33 | [TK]D-Fender | [16:13]<ManxPower-work>exten => s,1,Goto(${MACRO_EXTEN},1) <- now you're assuming you always want to return the the 1st priority, |
| 21:37.37 | p3nguin | geminizer: Those are the two cheapest ones I can think of right now, and they are very reliable. |
| 21:37.47 | dlynes_laptop | Geminizer, vitelity.net's pretty good, too. ManxPower-work suggested it to me, and I've been using it ever since |
| 21:37.49 | [TK]D-Fender | checkout time, BBIL |
| 21:37.51 | ManxPower-work | I seldom have problems with Vitelity |
| 21:38.04 | p3nguin | bandwidth.com doesn't even compare to the rates of the two I mentioned. |
| 21:38.16 | leifmadsen | oh, so you want cheap AND awesome |
| 21:38.30 | p3nguin | VoIP.ms is a Vitelity reseller AND they have lower rates. Hard to beat that. |
| 21:39.05 | dlynes_laptop | p3nguin, you mean they're in vitelity's affiliate program? |
| 21:39.10 | ManxPower-work | dlynes_laptop: I tried to develop for 1.6 and backport to 1.4, but they are just too different. |
| 21:39.20 | p3nguin | dlynes_laptop: No, that's not what I mean. |
| 21:39.20 | leifmadsen | Cheap, Fast, and Good -- pick any two. |
| 21:39.48 | Geminizer | I suppose I have been spoiled by seeing voicetrading's low call rates... but if their service is proportional to their rates (that is, both being cheap), then it's probably not worth it |
| 21:40.01 | dlynes_laptop | p3nguin, so they charge you, and feed you back through their backend connection to vitelity, so you're still effectively going through their equipment and vitelity's? |
| 21:40.07 | bmoraca_work | Geminizer: voicetrading's rates aren't that low |
| 21:40.23 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:40.23 | dlynes_laptop | ManxPower-work, develop dial plans, or develop C code? |
| 21:40.27 | ManxPower-work | Maybe I just like to spend money, but 1.2/cents/min is cheap enough for me. |
| 21:40.29 | p3nguin | geminizer: They are higher than the ones I suggested to you. |
| 21:40.31 | ManxPower-work | dlynes_laptop: dialplan |
| 21:40.44 | dlynes_laptop | ah |
| 21:40.48 | ManxPower-work | and almost everyone has 1.2 - 1.9 cents/mon |
| 21:41.01 | p3nguin | manxpower-work: inbound or outbound? |
| 21:41.06 | dlynes_laptop | I'm getting under 1c/min from vitelity |
| 21:41.17 | ManxPower-work | p3nguin: I would have to check. |
| 21:41.41 | p3nguin | Flowroute's termination rates from the US to other US numbers is $0.0098/minute. |
| 21:41.43 | dlynes_laptop | It's something like 0.8c/min if I remember correctly |
| 21:42.07 | p3nguin | Vitelity's retail termination rate is $0.014/minute. |
| 21:42.11 | bmoraca_work | i average $0.0065/min nation-wide |
| 21:42.22 | ManxPower-work | I'm paying 1.2 - 1.6 cpm, depending on the DID. |
| 21:42.57 | ManxPower-work | I don't know how much for outbound. |
| 21:45.08 | dlynes_laptop | oh...nvm 1.44c/min |
| 21:45.16 | p3nguin | right |
| 21:45.30 | dlynes_laptop | but for my other stuff I'm getting 0.55c/min |
| 21:45.42 | dlynes_laptop | so I only use vitelity for inbound for certain dids |
| 21:45.53 | Geminizer | p3nguin: what would you suggest for low outbound international call rates? |
| 21:46.01 | Geminizer | US to Dominican Republic, for example |
| 21:46.31 | p3nguin | geminizer: I only have personal knowledge of VoIP.ms for international calling, since that is who I use. I'll look at flowroute in a minute. |
| 21:46.32 | dlynes_laptop | Geminizer, have you tried looking on calltermination.com? |
| 21:47.14 | maximCH | http://backsla.sh/betamax ... compares all Betamax companies... they usually have the lowest rates. |
| 21:47.16 | p3nguin | geminizer: Give me the first 5 numbers of a phone number that you would call in Dom. Rep. |
| 21:47.33 | p3nguin | 243XX |
| 21:49.07 | Geminizer | 82999 |
| 21:49.25 | Geminizer | or 80948 |
| 21:50.51 | p3nguin | $0.00365/min to 82999 |
| 21:51.25 | p3nguin | $0.0364/min to 80948 |
| 21:51.54 | p3nguin | Wait, I think that first one has a misplaced decimal. |
| 21:52.07 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:52.14 | Geminizer | not bad at all... which provider is that from? |
| 21:52.17 | p3nguin | $0.0365/minute to 82999 |
| 21:52.21 | p3nguin | VoIP.ms |
| 21:52.32 | p3nguin | under 4c/min is okay, I guess. |
| 21:52.37 | bmoraca_work | i like voip.ms's user interface |
| 21:52.39 | maximCH | 0.014145 with calleasy.com or dialnow.com |
| 21:52.56 | *** join/#asterisk cadmium (i=mike@217.194.139.22) |
| 21:53.32 | cadmium | hi, how can i see the result of the noOp() function? |
| 21:54.52 | p3nguin | Flowroute is $0.1272/minute to 82999 and 80948 |
| 21:55.16 | p3nguin | almost 13c/m |
| 21:55.22 | p3nguin | That's a lot more than VoIP.ms. |
| 21:56.57 | p3nguin | I use VoIP.ms for both termination and toll-free DID. I haven't been dissatisfied with them yet. |
| 21:57.35 | Geminizer | I have used them recently, and am impressed with their customer service - very helpful.. |
| 21:58.47 | Chainsaw | cadmium: Just turning verbosity up to 10 should do it. |
| 21:59.22 | Geminizer | p3nguin... where did you go on the voip.ms site to get those rates for outbound international rates? |
| 21:59.45 | Geminizer | (pardon redundancy) |
| 22:00.39 | hardwire | (12:59:53 PM) Shane R. Spencer: eacn VNC connection takes up around 300kbps |
| 22:00.39 | hardwire | (01:00:00 PM) Shane R. Spencer: we have a 1500kbps link |
| 22:00.43 | hardwire | (01:00:27 PM) michael.horton: how do we change that |
| 22:00.46 | hardwire | ... ? |
| 22:00.57 | hardwire | (01:00:47 PM) michael.horton: the speed from 300kbps to 600kbps |
| 22:01.12 | eppigy | Katty: http://www.ustream.tv/sfshiba#more |
| 22:01.18 | hardwire | wrnog window |
| 22:01.56 | cadmium | Chainsaw got it thanks |
| 22:02.52 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 22:03.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:05.17 | bmoraca_work | lol...mispastes are fun |
| 22:05.54 | p3nguin | geminizer: Go to the main page, then click on International rates on the left-side navigation menu. |
| 22:06.22 | p3nguin | geminizer: Then put in 1809 in the search box and press the Search button. |
| 22:07.49 | p3nguin | geminizer: Oh, sorry... click on Termination Rates on the main page. |
| 22:08.06 | p3nguin | I had "international" on the brain for a minute. |
| 22:08.45 | Kobaz | ack |
| 22:08.48 | Kobaz | asterisk go byebye |
| 22:09.00 | p3nguin | waves to it |
| 22:09.28 | Kobaz | local channels tend to crash asterisk quite a bit |
| 22:09.38 | *** join/#asterisk box_ (n=lyle@75-147-236-238-Sacramento.hfc.comcastbusiness.net) |
| 22:09.48 | Kobaz | http://pastebin.ca/1765755 |
| 22:09.54 | cadmium | i'm trying to terminate inbound calls originating from DIDWW does anyone know how to get DIDWW to include the DID # in the uristring and then reference the DID in the dialplan? |
| 22:10.13 | Geminizer | p3nguin... are those the charges for local calls (e.g. DR to DR), or international (e.g. US to DR, DR to US)? |
| 22:10.50 | p3nguin | geminizer: They are assuming you are using their servers in US and Canada to terminate a call in DR. |
| 22:11.07 | p3nguin | That's what termination rates are. |
| 22:11.48 | box_ | hey, I just compiled and installed asterisk from source (over an existing trixbox install ). everything including SIP seems to be working fine so far, but IAX registration fails. It looks like * is sending only an AUTH after the REGREQ instead of the expected REGAUTH. Any idea what could cause this or debug approaches? |
| 22:11.57 | p3nguin | Now if you are in DR using the servers in US/Canada and calling a local (another DR) number, the same rate still applies. |
| 22:12.26 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:12.26 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:13.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:16.09 | *** join/#asterisk batphone (n=will@rrcs-24-153-211-180.sw.biz.rr.com) |
| 22:16.45 | batphone | i have a vendor on one side and a developer on the other arguing about what field in the SIP packet to route the call with |
| 22:16.58 | batphone | the vendor thinks we should be routing the call based on the "To:" field |
| 22:17.12 | batphone | the developer thinks we should be routing the call based on the R-URI |
| 22:17.45 | batphone | i can't seem to pin down the part of any of the dozens of SIP RFCs that define this... |
| 22:18.01 | batphone | because its an implementation issue. do you guys have any opinions on this? |
| 22:19.35 | *** join/#asterisk upp (n=upp@p57A77064.dip.t-dialin.net) |
| 22:19.53 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 22:20.09 | box_ | whyyy would this damn thing not see the REGREQ and send a REGAUTH? |
| 22:21.32 | box_ | i suppose there's not an increased debugging level for IAX2 |
| 22:27.01 | paulc | batphone: "To:" seems to be sensible, especially if the vendor's agreeing, no? |
| 22:27.17 | batphone | one would think |
| 22:27.30 | paulc | cluebat for the developer? ;-) |
| 22:27.51 | batphone | whats bad is sometimes vendors disagree |
| 22:27.58 | batphone | and it causes security models to break in some cases |
| 22:28.07 | batphone | thus leaving people un-interconnectable |
| 22:28.15 | paulc | true, which is why you want the RFC for reference.. but like you said - easier said than done sometimes isn't it |
| 22:28.35 | batphone | especially with SIP. how many are there? i lost count about 3 years ago. |
| 22:32.08 | *** join/#asterisk pietro (n=pietro@88-149-224-77.dynamic.ngi.it) |
| 22:32.49 | *** part/#asterisk pietro (n=pietro@88-149-224-77.dynamic.ngi.it) |
| 22:37.29 | box_ | hey, I just compiled and installed asterisk from source (over an existing trixbox install ). everything including SIP seems to be working fine so far, but IAX registration fails. It looks like * is sending only an AUTH after the REGREQ instead of the expected REGAUTH. Any idea what could cause this or debug approaches? |
| 22:57.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:57.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:00.28 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 23:03.43 | riddlebox | is mg2 echocan the best to use? |
| 23:04.01 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-40-42.acanac.net) |
| 23:04.42 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:06.22 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 23:21.41 | ruben23 | hi nayone have idea or used gsm channel bank for asterisk..? |
| 23:21.44 | *** join/#asterisk linuxh (n=LordVAXe@201.82.16.138) |
| 23:21.58 | linuxh | greetings ppl from earth |
| 23:22.08 | teknoprep | hi |
| 23:22.22 | teknoprep | how do i get a Cisco SPA series phone to hang up right away |
| 23:22.46 | teknoprep | when a person hangs up it stays connected..... then i get a busy signal... then it hangs up after a bit |
| 23:22.56 | teknoprep | i asked here a long time ago... and someone knew the answer |
| 23:23.00 | teknoprep | i forget the setting on the phone |
| 23:23.45 | linuxh | having a problem with distored audio after a second or so, between 2 x100p channels.... if iax <-> any of them it works fine, but between the 2 fxo channels i get good audio for a second or so.. then the gains seems to go crazy (monitoring with dahdih tool) and all gets distored.... asterisk 1.6.2 dahdih .2 and no tdm bridging echo cancel. |
| 23:24.19 | paulc | teknoprep: CPC timer.. can't remember the exact name in the interface but I think it references CPC |
| 23:27.59 | *** part/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 23:28.22 | paulc | teknoprep: CPC Delay and CPC Duration :-) - reduce CPC Delay to 0 or 1 and Duration to 1 or greater, works great for me |
| 23:28.52 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 23:32.24 | linuxh | my poor soul couldn't find any help using the regular "ways".... anyone ? ( = |
| 23:33.10 | voipmonk | are you using two different cards, linuxh ? |
| 23:33.32 | linuxh | yup, no conflicts nothing |
| 23:33.57 | linuxh | also, if i enable tdm bridging things gets better, but still odd and bad quality |
| 23:34.12 | linuxh | i'm about to try the 1.4.... as i can't find any reference of similar problems.. |
| 23:34.26 | voipmonk | really... |
| 23:34.32 | linuxh | (i tried several revisions of 1.6 and dahdi) |
| 23:34.46 | linuxh | the interesting part is.. the gains seems to go crazy after some seconds |
| 23:35.04 | linuxh | you can see the normal/good convo... then it just hits the limit... and all gets distorted.. |
| 23:35.09 | linuxh | like an AGC... :/ |
| 23:35.26 | voipmonk | disabled unused motherboard ports already, yes? |
| 23:35.29 | dlynes_laptop | linuxh, I think i've figured out yoru problem |
| 23:35.42 | linuxh | voip, i tried to change computers, even... |
| 23:35.45 | linuxh | dllynes ? |
| 23:35.47 | dlynes_laptop | linuxh, you're using two x100p cards |
| 23:35.50 | linuxh | yup |
| 23:36.02 | dlynes_laptop | linuxh, 1 x100p card is bad enough in one machine, but you're using two |
| 23:36.15 | dlynes_laptop | linuxh, you're a masochist, aren't you? |
| 23:36.15 | teknoprep | pal c |
| 23:36.21 | dlynes_laptop | ~x100p |
| 23:36.22 | infobot | methinks x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 23:36.24 | teknoprep | paulc, can't find that setting |
| 23:36.42 | linuxh | dlly.. its for a friend who can just affort that... i have never had a problem with 2 x100p... |
| 23:36.58 | dlynes_laptop | linuxh, are you sharing interrupts at all? |
| 23:37.27 | dlynes_laptop | linuxh, the x100p is quite picky about what systems it'll even work on |
| 23:37.31 | paulc | teknoprep - what model of ATA are you using? |
| 23:37.41 | teknoprep | cisco spa525g |
| 23:37.42 | dlynes_laptop | linuxh, most of the super cheap machines it won't work properly on |
| 23:37.56 | linuxh | dlly, no... not sharing interrupts |
| 23:37.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:38.00 | dlynes_laptop | linuxh, because it shares the interrupt with the network card |
| 23:38.03 | linuxh | its a dual PIII xeon |
| 23:38.11 | linuxh | no int. sharing, i'm sure |
| 23:38.16 | paulc | teknoprep: ah, it's a phone not an ATA |
| 23:38.18 | dlynes_laptop | linuxh, you sure? cat /proc/interrupts, and show me |
| 23:38.37 | paulc | teknoprep - don't have the settings handy but I recall from the old SPA941's that it always gave busy tone then disconnected |
| 23:38.41 | linuxh | <PROTECTED> |
| 23:38.42 | dlynes_laptop | linuxh, just grep out the lines that match your x100p card |
| 23:38.47 | linuxh | <PROTECTED> |
| 23:39.07 | dlynes_laptop | wow...that's a amazing |
| 23:39.12 | dlynes_laptop | first time I've ever seen that |
| 23:39.29 | linuxh | i would not come to here bug you ppl without trying all i could imagine |
| 23:39.30 | linuxh | ( = |
| 23:39.42 | voipmonk | pastebin your interrupts |
| 23:39.50 | dlynes_laptop | i guess you're able to tell pci cards what interrupts to use specifically? |
| 23:40.11 | voipmonk | are you using a dual core system |
| 23:40.48 | linuxh | i tested with one, now i switched to an other system |
| 23:40.52 | dlynes_laptop | voipmonk, why does dual core matter? |
| 23:40.57 | linuxh | trying to get the problem figured |
| 23:41.16 | linuxh | with at lest 3 machines, same issues... |
| 23:41.51 | dlynes_laptop | linuxh, which particular brand of x100p cards are you using? |
| 23:42.35 | linuxh | clones... |
| 23:42.35 | linuxh | 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 23:42.36 | linuxh | <PROTECTED> |
| 23:43.00 | dlynes_laptop | linuxh, i didn't ask whether they were clones, or what lspci told you they were...i asked what brand they were |
| 23:43.09 | linuxh | just for the record.. both works fine.. at the same time or iax <-> fxo |
| 23:43.10 | dlynes_laptop | linuxh, i already know they're clones |
| 23:43.20 | linuxh | no brand dude, i got them at the junk yard. |
| 23:43.24 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 23:43.30 | voipmonk | garbage in, garbage out |
| 23:43.31 | dlynes_laptop | linuxh, and most of them show up as tiger jet 3xx |
| 23:43.34 | linuxh | just a cheap and dirty modem. |
| 23:43.50 | dlynes_laptop | linuxh, the x101p clones show up as intel modems |
| 23:44.03 | linuxh | lemme check the chipset them, if that helps |
| 23:44.36 | linuxh | sure.. are those ambient md3200 |
| 23:44.45 | *** part/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 23:44.58 | linuxh | i did pick about 20 of them for 3 U$ |
| 23:44.59 | linuxh | ( = |
| 23:45.29 | QubeZ | anyone have experience with PIX and Asterisk server? Do I need to disable sip inspect? |
| 23:45.33 | dlynes_laptop | linuxh, ah....those are the shittiest clones |
| 23:45.50 | linuxh | yes.. but as i explained.. both can work at the same time... |
| 23:46.07 | linuxh | so.. what is bridging them is the cause.... and as it isn't a direct connection... |
| 23:46.27 | linuxh | i guess u get what i mean.... specially as they both works fine at the same time (2 iax <-> 2 fxo) |
| 23:46.37 | linuxh | the problem is when "native bridged" |
| 23:46.59 | linuxh | or some sort of nasty echo, as when i enable the native bridging echo cancelation, i see "some" improvement |
| 23:47.05 | linuxh | :/ |
| 23:49.44 | bmoraca_work | linuxh: it is an unfortunate side effect of using analog ports...timing is very touchy and can give you some pretty crappy results when bridge between ports in different PCI slots |
| 23:50.23 | linuxh | thats true |
| 23:50.41 | linuxh | also, glueing it with something found during my searchs |
| 23:50.45 | linuxh | it can be the cause... |
| 23:50.53 | linuxh | the damn board missing interrupts |
| 23:51.22 | linuxh | all my machines were of similar power, even the dual |
| 23:51.35 | linuxh | so its a possibility, lemme run the dahdih speed tests |
| 23:57.50 | linuxh | i guess you are at least right... the iax <-> fxo seems to be more forgiven |
| 23:58.02 | linuxh | and running the dahdih tests, it gives lower than 99.98... |
| 23:58.09 | linuxh | 99.97 ~ usually |
| 23:59.38 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 23:59.47 | riddlebox | wtf, why do people ask these dumb questions.... |
| 23:59.48 | riddlebox | http://forums.digium.com/viewtopic.php?f=13&t=72911&p=141253&sid=17f43dc2482f4ceb38be0e0ac2492d4f#p141253 |
