| 00:00.03 | DevWork_ | router does one to one nat. |
| 00:00.15 | Neozonz | ah |
| 00:00.16 | DevWork_ | internally, they are like 10.x.x.Y |
| 00:00.41 | Neozonz | [TK]D-Fender, Gonna probably tackle conference calling tomorrow, if I need help I might drop by again |
| 00:00.45 | Neozonz | thanks again for the help |
| 00:01.27 | [TK]D-Fender | You're welcome |
| 00:02.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.211) |
| 00:08.45 | DevWork_ | Are there any recent guides for centos and asterisk? last I could find was 6.0 |
| 00:13.23 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 00:13.25 | [TK]D-Fender | doesn't really change with decimals... |
| 00:17.20 | DevWork_ | Oh ok. Not a huge centos nerd so I don't know how it works. I know how versioning schemes should work :) Some follow, some don't. |
| 00:18.20 | DevWork_ | Think I'm more comfortable with the src than any instruction set too. |
| 00:19.56 | DevWork_ | I have a book on asterisk 1.6, the recipe one which I think is based on 1.8, is asterisk 11 the release after 1.8? How relevant is the 1.6 and 1.8 material. |
| 00:21.55 | WIMPy | 11 is the current stable/LTS |
| 00:23.43 | [TK]D-Fender | 12 is current, but won't be long-lived |
| 00:23.53 | [TK]D-Fender | lets say 2 years-ish |
| 00:24.07 | WIMPy | But is it stable? |
| 00:24.54 | *** join/#asterisk xaristax (~xaristax@189.242.191.250) |
| 00:26.08 | xaristax | Hi can someone help me i need two make play some audio after a Dial and sounds o the two sides i already try U parameter an make a Playback but only sounds on one side what can i do to make it work? |
| 00:32.11 | DevWork_ | Are magic jacks sip devices? |
| 00:32.23 | DevWork_ | sip with FXS on one side? |
| 00:42.37 | *** join/#asterisk jsjc (~Adium@61.Red-79-150-66.dynamicIP.rima-tde.net) |
| 00:49.27 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 00:53.38 | [TK]D-Fender | No, its USB to your PCS |
| 00:53.56 | [TK]D-Fender | and their client is essentially SIP, but the account locked down. |
| 00:59.27 | *** join/#asterisk vtone_ (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
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| 01:05.37 | ipengineer | I have my asterisk 12 box with a public IP and trying to get it to work with a device that is behind nat. The devices registers but it cannot receive calls and when I try to place a call I get no RTP. Do I need to do something to my transport section in pjsip.conf? |
| 01:11.59 | *** join/#asterisk vtone (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
| 01:13.49 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 01:14.20 | *** join/#asterisk D30__ (~deo@58.71.19.178) |
| 01:16.15 | Neozonz | [TK]D-Fender, hmm, it seems if I connect to openfire now, via an admin manager it doesn't work anymore :( |
| 01:16.43 | [TK]D-Fender | Look at the call... |
| 01:17.38 | Neozonz | https://gist.github.com/arajakul/8180750e1c0f5c5d7778 |
| 01:25.01 | [TK]D-Fender | We've already gone through the kinds you need... |
| 01:25.03 | [TK]D-Fender | that's not it |
| 01:25.57 | ipengineer | [TK]D-Fender: Would you mind looking at this sip trace and pointing me in the right direction as to why I am not getting an RTP audio? https://share.voipmonitor.org/view/AkkiCC8Xk1XhdsjClhztcnWG2b7lSWV_CkgOTJ5_WTW3hpcS9-K30Rw,, |
| 01:27.12 | [TK]D-Fender | Same goes for you... |
| 01:27.16 | [TK]D-Fender | look at the call |
| 01:27.27 | [TK]D-Fender | and that means ASTERISK SIDE |
| 01:31.13 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 01:36.20 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 01:36.20 | *** mode/#asterisk [+o file] by ChanServ |
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| 01:50.17 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 01:50.17 | *** mode/#asterisk [+o file] by ChanServ |
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| 02:14.28 | photog3 | can anyone recommend a SIP/TLS provider that does CLI passthrough in low volume? I need to use work CID on my home trunks for work calls, but personal for calls from non work extensions. |
| 02:18.26 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.118) |
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| 02:32.17 | *** join/#asterisk beanie (~beanie@87.113.216.163) |
| 02:33.15 | beanie | hello @n3hxs, @n3glv |
| 02:35.11 | *** join/#asterisk chare (322f51ac@gateway/web/freenode/ip.50.47.81.172) |
| 02:35.20 | chare | ok guys so how do we get free voip now? |
| 02:35.35 | beanie | hi Chare, are you new to this? |
| 02:35.46 | [TK]D-Fender | You know asking the same question every day sure is getting tiring.... |
| 02:36.15 | chare | I want to use google voice from my phone |
| 02:36.17 | chare | how do I do that |
| 02:36.24 | chare | whats the latest way to do it |
| 02:36.37 | [TK]D-Fender | "my phone" is not a specific thing |
| 02:36.44 | [TK]D-Fender | And this isn't a Google Voice channel |
| 02:36.48 | [TK]D-Fender | You're in the wrong place |
| 02:37.16 | [TK]D-Fender | When hangouts finishes taking over there is no more here until they open something up |
| 02:37.25 | [TK]D-Fender | And you won't see it appear overnight. |
| 02:38.29 | [TK]D-Fender | beanie: He's been asking the same question for weeks. |
| 02:38.34 | beanie | ah meh :-( |
| 02:38.59 | chare | give me a better answer then |
| 02:39.35 | [TK]D-Fender | chare: The facts do not care that you do not like them and will not change on your behalf. |
| 02:39.56 | chare | so in other words you don't know how to setup voip |
| 02:40.26 | beanie | Chare - you need to ASK your question in a way that 1) relates to the scope of this room 2) ask a question that you are sure you are going to understand the answer of |
| 02:40.45 | chare | setup asterisk with google voice to get free voip |
| 02:40.46 | [TK]D-Fender | please try to at least construct qualified statements and questions.... |
| 02:41.10 | [TK]D-Fender | "Free voip" isn't a thing. |
| 02:41.14 | [TK]D-Fender | Stop abusing words |
| 02:41.23 | [TK]D-Fender | VoIp is a CONCEPT. |
| 02:41.29 | [TK]D-Fender | it doesn't have a price. |
| 02:41.51 | beanie | well the other side of this |
| 02:42.11 | [TK]D-Fender | As for setting up * with GV.... for those for whom chan_motif works at all at this poitn... it's documented. |
| 02:42.26 | [TK]D-Fender | And with a death-clock hanging over it |
| 02:42.38 | beanie | dare I complicate it, free call routing between voip users is possible but that does not mean you can get free calls to standard telephone connections (POTS) |
| 02:43.18 | [TK]D-Fender | yes well he keeps throwing around vague terms trying to dodge the reality that termination is what costs |
| 02:43.50 | [TK]D-Fender | And asks why the rest of the world doesn't implement his pipe-dream mesh network to replace the entire system because he's impossibly cheap. |
| 02:43.58 | beanie | I guess this might have been one of the reasons Chare is pursuing the concept |
| 02:44.13 | chare | so beanie what do you use for your voip setup |
| 02:44.21 | [TK]D-Fender | beanie: The word you're looking for is "slactivist" |
| 02:44.31 | beanie | aha |
| 02:44.48 | beanie | Chare, I use Asterisk and FreePBX |
| 02:44.55 | beanie | I use a SIP trunk provider |
| 02:45.07 | beanie | and then my own hardware and a fibre optic internet connection |
| 02:45.31 | beanie | SIP Trunk = much credit to TK D-Fender who helped me for hours and firewall help from apb1963 |
| 02:45.33 | chare | what sip trunk provider |
| 02:45.41 | beanie | I'm not prepared to say |
| 02:45.50 | beanie | if you google them, there are many out there |
| 02:46.03 | chare | just tell me |
| 02:46.08 | beanie | no |
| 02:46.11 | Penguin | ~trunk |
| 02:46.11 | infobot | methinks 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. the nose of an elephant |
| 02:46.15 | chare | so you'll get sued if you talk |
| 02:46.39 | Penguin | The term you are looking for is ITSP. |
| 02:46.42 | Penguin | ~itsp |
| 02:46.43 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 02:46.47 | beanie | TK D-Fender can we have a definition for troll? |
| 02:47.12 | chare | penguin what free itsp do you use |
| 02:47.27 | [TK]D-Fender | beanie: No, I think he's just straight up delusional & retarded... but not an actual troll... |
| 02:47.31 | Penguin | None. Termination isn't free. |
| 02:47.42 | [TK]D-Fender | beanie: mustn't forget cheap... |
| 02:48.31 | beanie | damn, does the definition feature allow multiple words? |
| 02:48.42 | beanie | (TK D-Fender) |
| 02:48.49 | *** join/#asterisk vlad_sta_ (~vlad_star@h082.porttelecom.ru) |
| 02:49.00 | [TK]D-Fender | beanie: Not sure what you mean there... |
| 02:49.01 | Penguin | I do use Google Voice for now, since they gave that to me, but I'll take my numbers and go to my regular ITSP when they stop giving it to me. See voip.ms and flowroute.com. |
| 02:49.30 | beanie | Chare: You won't get ITSP for free |
| 02:49.46 | chare | I WILL USE GOOGLE VOICE |
| 02:49.51 | chare | so stop trying to stop me |
| 02:49.51 | beanie | not besides Google and |
| 02:50.00 | Penguin | You'll only use it as long as Google allows you to use it. |
| 02:50.28 | beanie | this is the thing, at the end of the day Chare, there's tons of different SIP Providers out there - all private companies |
| 02:50.41 | Penguin | ITSP. The term is ITSP. |
| 02:50.45 | beanie | many people have more than one in different countries - often allows taking advantage of cheap internetal calling |
| 02:50.53 | beanie | sorry Penguin - I stand corrected :-) |
| 02:51.14 | beanie | SIP Trunk Providers I meant |
| 02:51.22 | Penguin | ~trunk |
| 02:51.22 | infobot | somebody said trunk was 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. the nose of an elephant |
| 02:51.24 | Penguin | ^ |
| 02:51.40 | Penguin | Saying "SIP provider" is like saying "HTTP provider." It isn't really a thing. |
| 02:52.04 | beanie | hell ok - ITSP - what does it stand for? |
| 02:52.09 | [TK]D-Fender | Penguin: At least not as bad as saying HTTP protocol ;) |
| 02:52.11 | Penguin | ~itsp |
| 02:52.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 02:52.34 | Penguin | ATM machine, DSL line, VIN number ... |
| 02:52.58 | Penguin | People just don't get it. |
| 02:53.00 | chare | ~itsplist-us |
| 02:53.00 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 02:53.04 | beanie | Penguin - that somehow doesn't sit right with me thats' like saying Flowroute are the same as skype |
| 02:53.06 | [TK]D-Fender | The redundant backup department bureau of redundancy! |
| 02:53.13 | chare | WHY ISN'T GOOGLE VOICE ON THE LIST |
| 02:53.22 | Penguin | Google isn't an ITSP. |
| 02:53.23 | [TK]D-Fender | "respected" <- |
| 02:53.30 | chare | so google voice is a .... |
| 02:53.35 | [TK]D-Fender | DEAD DUCK |
| 02:53.44 | [TK]D-Fender | </aqualung> |
| 02:54.03 | beanie | A duck that has ducked the dead only to become dead |
| 02:54.40 | Penguin | Google Voice is a communication product which employs Voice over Internet Protocol. |
| 02:55.07 | chare | and connects to itsp |
| 02:55.15 | Penguin | I doubt they connect to an ITSP. |
| 02:55.21 | chare | THEY ARE THE ITSP |
| 02:55.37 | Penguin | They probably have a close relationship with an ILEC. |
| 02:55.42 | chare | ~ilec |
| 02:55.42 | infobot | i guess ilec is Typically the carrier that was granted the right to provide service as a result of the breakup of AT&T. These providers are also referred to as RBOCs (Regional Bell Operating Companies) or Baby Bells. |
| 02:56.31 | beanie | but an internet telephony service provider means different things by the layperson and anyone knowing anything about this line of work |
| 02:56.54 | Penguin | I'll give you that much. |
| 02:57.14 | Penguin | To someone who has no idea, Google Voice may appear to provide the same service as an ITSP. |
| 02:57.20 | beanie | it's a bit vague - I could understand someone saying the like of Skype is ths same as Teliax |
| 02:57.31 | [TK]D-Fender | beanie: The layperson is a moron and has no clue what the terms they throw around even mean. Don't drag this to the lowest common denominator, it's depressing ;) |
| 02:57.31 | Penguin | While it may be similar, it isn't the same overall. |
| 02:57.32 | chare | penguin I make a call with google voice |
| 02:57.34 | chare | thats all that matters |
| 02:57.51 | beanie | Chare: Don't try and cut down on what you think is relevant |
| 02:58.00 | Penguin | Skype can do the same type of things, but Microsoft is not an ITSP. |
| 02:58.14 | beanie | it's way too complicated - you'll need to set aside a lot of time |
| 02:58.19 | beanie | otherwise, just use skype man |
| 02:58.43 | Penguin | If you want to use Google Voice with asterisk, hop to it. It is well documented on the asterisk wiki how to set it up. |
| 02:59.00 | Penguin | Use it while it's still available. I do. |
| 02:59.05 | chare | but thats only temporary |
| 02:59.11 | Penguin | That's right! |
| 02:59.11 | chare | xmpp going away for google voice right? |
| 02:59.18 | [TK]D-Fender | [21:57]beanieChare: Don't try and cut down on what you think is relevant <- his bubble demands it.... |
| 02:59.19 | chare | SO WHATS THE LONG TERM FIX |
| 02:59.26 | Penguin | an ITSP |
| 02:59.30 | Penguin | but it is not free. |
| 02:59.33 | beanie | indeed |
| 02:59.37 | [TK]D-Fender | chare: "Get the hell outta Dodge" |
| 02:59.45 | chare | NO ITS TOO COMPLICATED TO USE BOTH GOOGLE VOCIE AND ANOTHER THING |
| 02:59.49 | [TK]D-Fender | This isn't a "breakage" |
| 03:00.13 | Penguin | Asterisk is only as complicated as you make it out to be. |
| 03:00.15 | beanie | ****Chare's wallet flies down the road chased by Chare screaming at the idea of a penny falling out**** |
| 03:00.57 | chare | so penguin whats the latest way to use google voice with asterisk NOT USING XMPP but using hangout thingy |
| 03:01.09 | beanie | there's 8 asterisk's there ^ |
| 03:01.11 | Penguin | I don't know of any way. |
| 03:01.12 | Penguin | I use xmpp still. |
| 03:02.07 | [TK]D-Fender | chare: there is NO Google Hangouts support. NONE. There is no "lastet" way because there is NO WAY. NIEN! NYET! |
| 03:02.08 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 03:02.32 | Penguin | You mean there isn't a way? You weren't being clear there. |
| 03:02.43 | [TK]D-Fender | http://catmacros.files.wordpress.com/2009/10/emo.jpg |
| 03:03.15 | chare | come on penguin there needs to be a way around xmpp |
| 03:03.18 | chare | tell me you know it |
| 03:03.21 | [TK]D-Fender | No there doesn't |
| 03:04.48 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 03:04.49 | Penguin | I'm not a software developer, so you're looking at the wrong guy. |
| 03:04.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 03:04.56 | chare | WTF |
| 03:05.03 | chare | HOW CAN YOU NOT KNOW HOW TO PROGRAM |
| 03:05.09 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:06.55 | chare | someone here must have a way to get around xmpp disappearing |
| 03:07.03 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 03:07.07 | [TK]D-Fender | yes, it's called "getting over it" |
| 03:07.14 | [TK]D-Fender | And we did so LOOOOONG ago |
| 03:07.21 | beanie | Chare: Re your earlier question - http://lmgtfy.com/?q=use+google+voice+with+asterisk+using+hangout |
| 03:09.06 | Penguin | I don't even know if it is worth my time to google for those terms. |
| 03:10.35 | beanie | ah but it does bring up a nice easily readable research list to get a feel for the topic |
| 03:10.52 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 03:10.56 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 03:12.25 | *** join/#asterisk starks (~Eric@69.74.61.3) |
| 03:12.36 | chare | the search gives no useful results |
| 03:12.56 | starks | jesus christ... why does digium disable serial and lock down ssh on their switchvox. i'm manually adding users from a usb boot chroot |
| 03:13.21 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 03:13.45 | Penguin | Jesus just left. |
| 03:15.34 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 03:16.51 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 03:18.48 | [TK]D-Fender | chare: Sure it did. |
| 03:18.50 | [TK]D-Fender | http://www.obitalk.com/forum/index.php?topic=6867.msg45992#msg45992 |
| 03:18.57 | [TK]D-Fender | That sums things up nicely |
| 03:19.05 | [TK]D-Fender | Considering you also have an Obi |
| 03:22.31 | Penguin | Is it because of obi that google has decided to get rid of their xmpp services? |
| 03:22.42 | [TK]D-Fender | Google never cared about becoming a telco. They data-mined people all they cared. Free ride over. They don't care about your goals. They are not shared. |
| 03:23.08 | [TK]D-Fender | doubt directly. Perhaps the reality of there being no value to free-loaders |
| 03:23.17 | [TK]D-Fender | and Obi was just another one./ |
| 03:23.33 | Penguin | That post kind of alluded to it, so I didn't know. |
| 03:23.36 | [TK]D-Fender | GV = MS BOB |
| 03:25.07 | Penguin | Did Digium contribute to it, or did Google permit it? |
| 03:27.42 | [TK]D-Fender | "did nothing about it" more likely |
| 03:28.09 | [TK]D-Fender | Sort of the same way humans barely notice when they step on ants |
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| 03:29.02 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 03:29.56 | chare | I'm panicing here |
| 03:30.00 | chare | no alternative to google voice |
| 03:30.02 | chare | CRY CRY CRY |
| 03:30.37 | [TK]D-Fender | no FREE alternative |
| 03:30.46 | Penguin | Mmmmm pan icing... |
| 03:30.54 | beanie | Starks - hell it's no expensive you know |
| 03:31.14 | beanie | starks - don't know why that pasted |
| 03:31.15 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 03:32.21 | beanie | look at the actual prices - pennies per minute to pots, many itsp's offer free voip calls between users with separate networks that use the same itsp as you and you can still contact your network for free using asterisk |
| 03:32.56 | beanie | my itsp asks me to pay them for an inbound telephone number i use, they then route my calls to me for free |
| 03:33.46 | beanie | well to my CentOS box which then routes to asterisk ivr and call queues and then to agents |
| 03:34.32 | starks | so uh... is there any way to force asterisk to register a cisco phone. i've heard all kind of crazy things about certain firmware disabling UDP, forcing USECALLMANAGER, NAT troubles, using illegal characters in SUPPORTED: |
| 03:34.32 | starks | etc |
| 03:34.41 | starks | i keep getting 401 rejections |
| 03:35.12 | beanie | starks - i dont know the first thing but is flashing the cisco phone with some decent freeware a good start? |
| 03:35.23 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 03:35.34 | starks | beanie, that exists? |
| 03:35.38 | beanie | even, would asterisk run the on the phone itself? |
| 03:35.39 | starks | for the 79xx series? |
| 03:35.57 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 03:36.01 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 03:36.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:36.04 | beanie | well ive heard its possible to flash phones but I have only heard this - just about everyone knows a million times more than me |
| 03:36.19 | *** join/#asterisk Neo (~neo@opennic/Neo) |
| 03:36.21 | starks | if there's firmware for the phone that has a proper sip implementation, i want ti |
| 03:36.30 | Penguin | I simply use chan_sccp-b on asterisk and SCCP images on my phones. |
| 03:36.38 | beanie | hello sruffell - why do the channel ops never speak in here |
| 03:36.41 | Penguin | There's also MGCP and SIP. |
| 03:38.23 | Penguin | I heard that chan_skinny received some attention after 1.8, but I haven't tested that out yet. |
| 03:41.27 | beanie | penguin - is it difficult to flash a cisco phone - how would you get it running asterisk but still operating as a usable phone? |
| 03:41.42 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 03:42.56 | sruffell | hello beanie |
| 03:43.19 | sruffell | I just don't have much to say. WIMPy and TKD-Fender normally have it covered much better than I could possibly. |
| 03:43.51 | sruffell | [TK]D-Fender (whoops) |
| 03:43.51 | beanie | howdo sruffell :-) ah that's a little mad... you would think that they'd end up being the channel ops... |
| 03:44.37 | beanie | but nice of you to say hi :-) |
| 03:44.38 | sruffell | Yeah…they probably could be…I'm only a channel ops since I've been around awhile as a DAHDI maintainer. But most of the questions in here have to do with Asterisk / deployments….and that's not my personal strong suit. |
| 03:45.03 | [TK]D-Fender | Knowledge alone isn't grounds for a basis of goverment apparently. |
| 03:45.20 | [TK]D-Fender | much like saucy tarts handing out swords from a lake ;) |
| 03:45.29 | sruffell | But I always log in when I'm working in case someone knows I can help with something specific…otherwise, I don't really read too much. |
| 03:45.38 | beanie | ahhh fair play :-) Well i suppose the more the more democratic :) |
| 03:45.40 | sruffell | lol |
| 03:46.05 | *** join/#asterisk soy_el_pulpo (~soy_el_pu@181.64.133.94) |
| 03:46.07 | beanie | i just never notice any of you in conversations |
| 03:46.26 | beanie | so many people here never say anything |
| 03:46.33 | beanie | most* |
| 03:46.35 | [TK]D-Fender | beanie: they care much less at night since most ops have daytime obligations related to * that has them here |
| 03:47.03 | beanie | oh right so asterisk insist that there's ops here... |
| 03:47.18 | beanie | sorry i say that, i mean digium i suppose |
| 03:47.45 | chare | sruffell tell me you have a fix for the xmpp google voice going away problem |
| 03:47.58 | beanie | God loves a tryer :) |
| 03:48.18 | sruffell | nope…I have no fixes related to google voice. |
| 03:48.46 | sruffell | [TK]D-Fender probably should be a channel dev…but sometimes…he gets a little too mad.... |
| 03:49.04 | beanie | no comment |
| 03:49.06 | beanie | :-p |
| 03:49.07 | [TK]D-Fender | I'm much better than I used to be :) |
| 03:49.08 | chare | sruffell is there any way to get around the problem |
| 03:49.19 | sruffell | heh |
| 03:49.34 | [TK]D-Fender | Chait isn't a "problem". Nothing is "wrong". They are simply dumping you on the curb. |
| 03:49.40 | [TK]D-Fender | chare: it isn't a "problem". Nothing is "wrong". They are simply dumping you on the curb. |
| 03:49.45 | [TK]D-Fender | BAI BAI |
| 03:49.47 | beanie | pretty much a no go it seems...Chare - embrace the fact you are a bit stuffed and stop wasting your time trying to find a cure to something uncureable and start researching a new workaround not involving google |
| 03:49.59 | chare | god damn it does it mean i need to use vitality |
| 03:50.04 | sruffell | chare: sorry I can't help but I really do not know anything about that issue. I would just end up searching the issue tracker (http://issues.asterisk.org) and the recent commits to see if there was anything google talk / xmpp related. |
| 03:50.05 | [TK]D-Fender | There is nothing to fix because it isn't BROKEN. Theya re taking their toys back |
| 03:50.10 | [TK]D-Fender | NO SOUP FOR YOU! |
| 03:50.52 | [TK]D-Fender | sruffell: He's been trolling this channel for weeks whining about losing his free-ride and insisting there must be another free ride or way to steal his old one back. |
| 03:51.11 | sruffell | ahh... |
| 03:51.17 | beanie | Chare - I know it might suck, but you're better to move on |
| 03:51.26 | beanie | it really doesn't cost that much to pay you know :) |
| 03:51.29 | chare | I'm not a troll |
| 03:52.00 | [TK]D-Fender | chare: You are reaching levels of stupidity that make that almost too much to believe. |
| 03:53.27 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 03:53.28 | beanie | sruffell.. so do you literally have to sit watching these boards for hours then |
| 03:53.36 | beanie | or do you rely on pm's |
| 03:53.44 | [TK]D-Fender | There is no fix. It has nothing to do with Asterisk. Google changed their mind. Too bad. No hacks. No back-door. No way around. Game Over. |
| 03:54.25 | sruffell | nope…I just log in….my client is configured to beep if someone says my name…otherwise I personally pretty much ignore. |
| 03:54.27 | ChannelZ | Second verse, same as the first! |
| 03:54.39 | coppice | [TK]D-Fender: with Google its rarely game over. its game endlessly changing in near random ways |
| 03:54.50 | beanie | Chare: pick yourself up, dust yourself, get back up, restrategise, look at potential options, look at cost, then work out how to research then test and then deploy |
| 03:55.09 | sruffell | I wouldn't get anything done if I kept up with the traffic in hear, asterisk-dev, the internal chat rooms, the mailing lists, the RSS feed from the forums, etc… in real-time. |
| 03:55.16 | sruffell | s/hear/here/ |
| 03:55.25 | [TK]D-Fender | Copp when the game changes.. it ceases to be the game it was. That game is over. Welcome to "data miniming the planet for PROFIT 103.7!" |
| 03:55.33 | [TK]D-Fender | mining* |
| 03:55.35 | beanie | Coppice - you are basically describing Google's biggest asset - "Google Random" |
| 03:55.37 | [TK]D-Fender | coppice: * |
| 03:55.38 | [TK]D-Fender | gat |
| 03:55.49 | [TK]D-Fender | gah* |
| 03:56.00 | [TK]D-Fender | there go my typing skills again... |
| 03:56.46 | beanie | sruffell - so how do you do it :) |
| 03:57.15 | coppice | Ah, the Google company song: |
| 03:57.16 | coppice | <PROTECTED> |
| 03:57.35 | coppice | as they try to appear snow white |
| 03:58.06 | beanie | well that's just it, their idea of "off to work we go" is pressing the on button on their tablet in bed |
| 04:01.03 | beanie | ok i'm going to ping myself down to my router as i need to reset it |
| 04:01.16 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 04:01.18 | beanie | i'll be back in about 10 mins |
| 04:01.31 | beanie | nice chatting to you Sruffell if you're gone again |
| 04:02.06 | sruffell | :) I did have my other window open. |
| 04:02.37 | coppice | as in "My other window's a porn site"? |
| 04:03.10 | Penguin | beanie: It is not difficult to load software onto a Cisco 7900 series phone. But that has nothing to do with asterisk at all. |
| 04:03.39 | chare | sruffell you are familiar enough with the google hangouts protocol to fix asterisk to be able to use google voice when xmpp goes away? |
| 04:05.07 | sruffell | my other window is a shell (as if I really needed to say that….) and no…I know about device drivers and the linux kernel. I don't have an incentive to spend any time chasing Google. For my home system I just buy minutes from a SIP provider. |
| 04:05.47 | sruffell | coppice: speaking of drivers. Was it you who wrote the write up about CUDA / OpenCL for transcoding and why it doesn't work so well? |
| 04:05.56 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 04:06.14 | sruffell | I remember reading something on that…but I'm not sure where it was. Someone was asking me about it last week for another project. |
| 04:06.14 | coppice | I have written something about that in the past |
| 04:06.32 | sruffell | Ok…I'll go search your site. |
| 04:06.35 | sruffell | thanks. |
| 04:07.15 | coppice | very few descriptions of GPU performance are honest |
| 04:09.05 | *** join/#asterisk beanie (~beanie@bgareth2.plus.com) |
| 04:09.25 | Penguin | beanie: It is not difficult to load software onto a Cisco 7900 series phone. But that has nothing to do with asterisk at all. |
| 04:09.30 | beanie | ha@Coppice - what was Shruffell's reply :-p |
| 04:09.45 | beanie | Penguin - eek - whole new load of learning right? |
| 04:10.01 | Penguin | Which part? |
| 04:10.33 | Penguin | The loading of software on a Cisco phone amounts to reading the instructions on the Cisco site or any of the other sites that replicate the instructions. |
| 04:10.48 | Penguin | The configuration of Asterisk is also fairly well-documented. |
| 04:13.06 | beanie | is it a timely job for a newbie |
| 04:13.17 | beanie | context is important here :) |
| 04:13.47 | Penguin | Most people don't learn how to configure all aspects of asterisk in a day. |
| 04:14.00 | Penguin | Many people will learn how to configure a few of them in just a day. |
| 04:15.42 | beanie | i havent googled but i guess there are online vids to get a taste? |
| 04:15.47 | Penguin | ~book |
| 04:15.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:16.00 | beanie | ah im not a book person |
| 04:16.05 | beanie | visual learner :) |
| 04:16.16 | Penguin | Why would you need a video to describe how to type some words and numbers? |
| 04:16.19 | [TK]D-Fender | Books are highly visual. |
| 04:16.29 | beanie | anyone offering free food, board and travel tickets for work experience :-p |
| 04:17.16 | Penguin | Some stuff just doesn't require a YouTube video. |
| 04:17.40 | beanie | :-) I've suddenly hit point where i feel physically sick tired |
| 04:17.49 | beanie | it's 04:17 GMT |
| 04:17.51 | [TK]D-Fender | Most of the people making youtube videos know jsut enough to be dangerous |
| 04:18.08 | beanie | TK D-Fender - Wise warning |
| 04:18.16 | Penguin | It might be 04:18 GMT, but I don't feel that tired. |
| 04:18.35 | beanie | are you in the UK? |
| 04:18.38 | Penguin | Nope. |
| 04:18.46 | beanie | working to UK Time? |
| 04:18.52 | Penguin | Negative. |
| 04:19.10 | beanie | no need to feel that way |
| 04:19.15 | beanie | na i'm messing |
| 04:19.22 | Penguin | I was just pointing out that time is relative. |
| 04:19.34 | Penguin | It's only 2219 here. |
| 04:19.43 | beanie | :-) human's are also restricted by it |
| 04:19.46 | beanie | reet bed for me |
| 04:19.54 | beanie | nuce chatting to you all :) |
| 04:19.58 | beanie | nice |
| 04:20.00 | coppice | 22:19 is soooo yesterday |
| 04:20.07 | beanie | :-p |
| 04:20.10 | Penguin | I do have a meeting at 0900, so I can't stay up all night. |
| 04:20.31 | beanie | bye :) |
| 04:21.33 | [TK]D-Fender | [23:19]coppice22:19 is soooo yesterday <- or later today |
| 04:22.15 | coppice | [TK]D-Fender: that's everyone's future |
| 04:22.23 | Penguin | In this case, it's yesterday. |
| 04:23.39 | [TK]D-Fender | coppice: Some might not make it..... |
| 04:23.58 | Penguin | Or they might turn into pumpkins. |
| 04:27.48 | *** join/#asterisk lok-jpr-india (~jpr-in@202.191.157.74) |
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| 06:52.23 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 07:16.48 | arvind_khadri | Hi, I need some help with astersik-fu, how can I present a user with the same menu when he presses a key. |
| 07:19.22 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 07:21.58 | Penguin | Like... the same menu he just heard? |
| 07:22.55 | Penguin | But only if he pressed a key that was not a valid choice? |
| 07:23.35 | Penguin | arvind_khadri |
| 07:23.37 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:24.03 | arvind_khadri | Penguin, Umm, more like repeat menu on a key. What am looking for is, that when a user presses a key, e.g 5 for 15 times say, that key press should get recorded. |
| 07:24.41 | arvind_khadri | So right now, what I have working is that when a user presses 5, it gets logged only once, and not after that |
| 07:25.05 | arvind_khadri | and no other key press works after that. What am I missing? Penguin |
| 07:26.44 | Penguin | I don't understand what you are describing, and now I'm going to go to bed. I thought I was going to get to tell you about extension i and be finished. |
| 07:34.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 07:34.15 | arvind_khadri | Penguin, aah I see. Thanks :) |
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| 07:41.56 | bsdice | what is the restricted PQ prefix in Jira? Like PQ-1396 |
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| 08:40.51 | verywiseman | Hi, i established TLS/SRTP/SIPS connections between 2 mobile devices , all things appear good , but no one can hear the other end point . what may the problems ? Also are there specific codecs for encrypted call ? |
| 08:46.36 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 09:15.25 | *** join/#asterisk eg0x (~eg0x@unaffiliated/eg0x) |
| 09:21.33 | eg0x | hey guys. i`m currently a little bit stuck. i want to to a paging to some phones, initiated with a call file in /var/spool/asterisk/outgoing - my .call file: http://pastebin.com/PZPaxc9x - my asterisk log: http://pastebin.com/bSadndRf |
| 09:21.58 | eg0x | i made a paging / intercom group (1339) |
| 09:22.28 | WIMPy | You are calling a sip device 1339, not an extension. |
| 09:22.35 | WIMPy | Use a local channel to call an extension. |
| 09:22.52 | eg0x | ok, syntax is Local/1339 instead of SIP/1339 then? |
| 09:22.57 | WIMPy | ("notfall" doesn't sound like naguage=en) |
| 09:23.06 | eg0x | yeah ;) its just for testing |
| 09:23.13 | eg0x | german word for emergency |
| 09:23.14 | WIMPy | local/extension@context |
| 09:23.33 | WIMPy | #asterisk-de :-) |
| 09:23.42 | *** join/#asterisk YannikS (~Miranda@31-18-193-221-dynip.superkabel.de) |
| 09:24.11 | eg0x | WIMPy: nice, thanks. one more thing: what does @context mean? |
| 09:24.19 | eg0x | is that a hostname or something? |
| 09:24.25 | eg0x | or the name of the sip trunk? |
| 09:24.43 | WIMPy | The context of your extensions.conf. |
| 09:24.50 | eg0x | ah :) |
| 09:25.14 | WIMPy | For the basics try the |
| 09:25.19 | WIMPy | ~book |
| 09:25.19 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:25.27 | eg0x | :D |
| 09:25.29 | WIMPy | or http://das-asterisk-buch.de/1.6/ |
| 09:25.33 | eg0x | i love freenode. thanks!!! |
| 09:25.45 | WIMPy | A little out of date, but not bad anyway. |
| 09:26.41 | eg0x | it will fit my needs :) thanks for your great help |
| 09:26.59 | eg0x | bye + have a nice day |
| 09:27.01 | *** part/#asterisk eg0x (~eg0x@unaffiliated/eg0x) |
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| 09:30.00 | arvind_khadri | Hi, I have an asterisk setup which is functional. I want to just log the number of times a key is pressed, where is the best place I can do it? In the AGI scripts? |
| 09:35.28 | kaldemar | the place where you handle the call. |
| 09:37.36 | kaldemar | if you have an IVR in dialplan, use a channel variable that you increment on each key press. |
| 09:41.16 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 09:41.48 | arvind_khadri | kaldemar, ok, so I have a python script which answers the call. What I have achieved is having to detect the key pressed, but it only works once and then the server does not respond for any key press |
| 09:43.29 | kaldemar | are you using AGI? |
| 09:43.34 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 09:44.00 | phpboy | hi all, I cannot seem to get the peers specified in sip.conf using register = to actually register |
| 09:44.14 | phpboy | they all stay in unregistered state with no debug info |
| 09:44.34 | phpboy | what can I do to debug it or force it to try register besides issuing 'sip reload' |
| 09:44.38 | WIMPy | phpboy: Peers or register => ? |
| 09:44.48 | WIMPy | That's the way. |
| 09:44.50 | phpboy | sorry: register -> |
| 09:44.51 | arvind_khadri | kaldemar, yes |
| 09:44.56 | phpboy | register => |
| 09:45.08 | phpboy | it appears in sip show registry |
| 09:45.17 | phpboy | they all just stay unregistered |
| 09:45.21 | phpboy | :\ |
| 09:45.26 | WIMPy | Ok, I haven't used register for ages in favour of doing it in the peers. |
| 09:45.42 | *** join/#asterisk Kaotiko (~0xFF@unaffiliated/kaotiko) |
| 09:45.46 | phpboy | well, it's not working with type=peer either |
| 09:45.50 | WIMPy | Does the host resolve? |
| 09:45.55 | phpboy | it does |
| 09:46.48 | phpboy | I'm using register to make debuging this problem easier |
| 09:46.55 | phpboy | but that doesn't help :\ |
| 09:47.07 | WIMPy | Why should it make it easier? |
| 09:47.29 | WIMPy | If you use a peer, you can qualify it. |
| 09:48.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:48.07 | WIMPy | BTW: Do current versions still try to resolve the host, even if there's an outboundproxy configured? |
| 09:48.08 | phpboy | qualify=yes |
| 09:48.09 | phpboy | ? |
| 09:48.17 | WIMPy | yes |
| 09:48.18 | kaldemar | arvind_khadri: then it's up to your agi to read keypresses and do what you want with them. |
| 09:48.23 | phpboy | plese hold |
| 09:49.05 | arvind_khadri | kaldemar, aah ok :) |
| 09:49.19 | arvind_khadri | kaldemar, has got nothing to do with extensions right? |
| 09:49.48 | WIMPy | Your AGI is (part of) an extension. |
| 09:52.15 | phpboy | WIMPy: ok, I did two things now, changed host to use IP (which I can contact) and qualify=yes |
| 09:52.19 | phpboy | still doesn't work |
| 09:52.37 | phpboy | what debug should I be checking for? no errors are staring me in the face |
| 09:52.39 | WIMPy | And is the peer reachable? |
| 09:53.17 | WIMPy | I think verbose 2 is enough for registration issues. |
| 09:54.11 | phpboy | ah wait, SIP debug is showing me the light |
| 09:54.15 | phpboy | kinda |
| 09:54.29 | phpboy | it seems it's sending the calls to itself instead of to the provider |
| 09:55.33 | kaldemar | arvind_khadri: once you start an AGI, it handles the call until it exits. |
| 09:55.45 | phpboy | which is odd because I'm definitely dialing the right SIP context |
| 09:56.02 | arvind_khadri | kaldemar, oh ok :) |
| 10:03.00 | phpboy | WIMPy: Do you know why it would try dial itself instead of the provider? the sip context is correct (and simple) |
| 10:05.31 | phpboy | sip show peers 'the peer is registered' |
| 10:13.26 | *** join/#asterisk StaRetji (~vencuga@178.79.6.17) |
| 10:15.31 | StaRetji | Hello all. Is it possible to exten => 100,n,Read(digits,,4) send to something like this exten => 100,n,GotoIf(${DB_EXISTS(pintemp/digits)}?correct,${EXTEN},1:incorrect) |
| 10:16.10 | StaRetji | I know how to get CALLERID(num) and CALLERID(dnid) and check if exists in b |
| 10:16.27 | StaRetji | but is there a way to check entered digits? |
| 10:16.33 | *** join/#asterisk evil_gordita (robert@70.188.56.12) |
| 10:16.57 | StaRetji | you see, idea is to ask for a pin, if user enters 1234, asterisk should check in astdb |
| 10:17.09 | StaRetji | any help would be highly appreciated |
| 10:18.51 | phpboy | ok, now I've gotten as far as getting a reject from the provider |
| 10:20.46 | WIMPy | StaRetji: Wow. Have you heard of the general concept of using variables? the "digits" you supply to Read() is one. |
| 10:22.12 | WIMPy | And you get the value of a variable with ${variable_name}. |
| 10:23.54 | Faustov | StaRetji: you can add DTMF to log level |
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| 10:41.30 | StaRetji | WIMPy: and Faustov, thanks for reply |
| 10:42.38 | StaRetji | Read(digits) then $(digits) ? |
| 10:43.04 | WIMPy | Lookalike |
| 10:43.11 | WIMPy | {} not () |
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| 10:43.48 | StaRetji | please forgive me, I am trying to understand it |
| 10:44.28 | StaRetji | for example, this works GotoIf($["${digits:0:4}" = "1234"] |
| 10:44.54 | WIMPy | I wonder how you survived for so long without having made use of those things before. |
| 10:44.56 | StaRetji | instead of "1234" I need DB_EXISTS |
| 10:45.12 | StaRetji | lol |
| 10:45.20 | WIMPy | Go back to your original question. |
| 10:45.37 | WIMPy | Just use ${digits} instead of just digits. |
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| 11:10.37 | StaRetji | WIMPy: thanks mate, that was it! |
| 11:10.48 | StaRetji | I was almost there :) |
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| 13:17.21 | Weezey | Having a weird issue where recordings to a certain extension are being delayed before playback. Not happening on other numbers on the same box, even a working recording starts playing a couple seconds in on this context vs another. |
| 13:17.45 | [TK]D-Fender | pastebin the complete call with appropriate debug |
| 13:17.54 | Weezey | k |
| 13:25.33 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 13:28.47 | Weezey | [TK]D-Fender: as always, you rule the roost. In trying to make the debug I think I may have found the issue, looks like that DID exists elsewhere in my dialplans and is bouncing around before ending up on that last server for playback, adding quite a few legs to the call before it starts playback. |
| 13:29.51 | [TK]D-Fender | Silly me and my "looking at issues". I'm terrible.... |
| 13:30.04 | Weezey | [TK]D-Fender: terribly awesome. |
| 13:30.28 | [TK]D-Fender | better than awesomely terrible I guess |
| 13:31.10 | Weezey | [TK]D-Fender: on a completely unrelated topic, is it possible to specify a specific UDP port for RTP for a single SIP peer? |
| 13:32.05 | [TK]D-Fender | nope |
| 13:32.30 | [TK]D-Fender | well... maybe to a degree |
| 13:32.54 | Weezey | got someone stuck in a coworking space with a crap router. |
| 13:32.55 | [TK]D-Fender | That being that on certain SIP phones you can restrict the range, so that might tighten THAT end, but there is no reservation on *'s end |
| 13:33.13 | Weezey | k, didn't think so. |
| 13:33.22 | [TK]D-Fender | Um, how bad a router? |
| 13:33.30 | [TK]D-Fender | typically standard NAT handling of it works fine... |
| 13:33.40 | *** join/#asterisk henny199 (~henny199@host217-37-52-85.in-addr.btopenworld.com) |
| 13:33.59 | Weezey | It's showing up as the public IP in asterisk's peer info where it would show public and private |
| 13:35.35 | [TK]D-Fender | ? |
| 13:35.52 | [TK]D-Fender | PB it |
| 13:38.14 | *** join/#asterisk apurvtwr (b64722fb@gateway/web/freenode/ip.182.71.34.251) |
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| 13:39.41 | Weezey | other peers show Addr->IP: publicIP:port and under Reg. Contact: sip:peername@privateIP:port;transport=udp this one shows the public IP for both. |
| 13:41.13 | apurvtwr | hi everyone. I am using asterisk 1.8 and while making outbound calls it gives the following error intermittently. |
| 13:41.16 | apurvtwr | " chan_sip.c: Autodestruct on dialog '313ccdc00ca1a2f778ebe98d2972c9fd@10.10.10.10:5060' with owner SIP/trunkName-00000a1f in place (Method: BYE). Rescheduling destruction for 10000 ms" |
| 13:41.28 | henny199 | Hi All, Having issue with an Incomming DDI , all looks good in freePBX,but looking at the log i can't seem to see the incimmoing call being taged with the DDI any ideas where to look? |
| 13:42.29 | [TK]D-Fender | henny199: "core set verbose 10", "sip set debug on" <- PASTEBIN the complete inbound call for us to examine. |
| 13:42.31 | [TK]D-Fender | ~pb |
| 13:42.31 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:42.33 | [TK]D-Fender | ^^^ |
| 13:44.02 | apurvtwr | The issue seems to have been encountered by some one here http://forums.asterisk.org/viewtopic.php?f=1&t=86221 too |
| 13:44.29 | henny199 | [TK]D-fender btw it an isdn line coming in threw DADHI |
| 13:44.59 | *** join/#asterisk tedstriker (~tedstrike@92.40.248.220.threembb.co.uk) |
| 13:45.13 | apurvtwr | but david55 suggested asterisk 1.8.21 which I am already using. |
| 13:45.34 | [TK]D-Fender | apurvtwr: 4 versions behind... |
| 13:45.34 | WIMPy | henny199: Do you have overlap enabled? |
| 13:50.04 | apurvtwr | [TK]D-Fender: I am reluctant in switching asterisk version unless absolutely necessary since we have done alot of custom development around this already. |
| 13:50.31 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 13:50.48 | apurvtwr | Is this an unclosed bug of that release? |
| 13:51.09 | [TK]D-Fender | apurvtwr: Go check the changelog & tracker |
| 13:51.26 | [TK]D-Fender | apurvtwr: locking yourself in on a specific version is a BAD move |
| 13:53.12 | henny199 | [TK]D-Fender - see http://pastebin.com/kZ0PYikK note ive swaped out the incomming number to 12345678910 rater than posting my mobile number |
| 13:55.58 | apurvtwr | [TK]D-Fender: Thanks. We did that because it was an LTS I suppose. |
| 13:56.38 | [TK]D-Fender | henny199: disable the core debug in there, it's pointless and floods the log, and that is not a complete call from the start... |
| 13:56.42 | apurvtwr | Could you please also tell me, what this error really means. |
| 13:56.57 | [TK]D-Fender | apurvtwr: the BRANCH is LTS< not that specific decimal release |
| 13:57.52 | [TK]D-Fender | apurvtwr: 1.8.22 came out because 1.8.21 had issues..... |
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| 13:59.16 | apurvtwr | hm. My bad . |
| 14:01.56 | Katty | morning |
| 14:01.59 | henny199 | whats the command to disbale debug |
| 14:02.11 | leifmadsen | core set debug 0 |
| 14:02.22 | leifmadsen | or if you're talking sip debug: sip set debug off |
| 14:02.57 | WIMPy | pri debug would help, however. |
| 14:03.25 | WIMPy | hex is good enough for me. Otherwise use intense. |
| 14:04.21 | apurvtwr | [TK]D-Fender: thanks. I am installing Asterisk 11.6-cert1. |
| 14:05.01 | WIMPy | The story ends with the first line of dialplan. |
| 14:05.31 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 14:11.50 | henny199 | http://pastebin.com/1qHc7tkt |
| 14:13.46 | WIMPy | Apart from the faild attempt to enable pri debug, that does not look like it includes the start of the call, which would be the only interesting bit. |
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| 14:14.28 | WIMPy | oh, the start comes later on. |
| 14:14.59 | WIMPy | Well, the issue is right there in your face: |
| 14:15.04 | WIMPy | -- Going to extension s|1 because of immediate=yes |
| 14:15.22 | henny199 | where would this be set? |
| 14:15.29 | henny199 | miniPBX:456 |
| 14:15.42 | WIMPy | chan_dahdi.conf |
| 14:15.44 | teeteewhy | h |
| 14:16.26 | WIMPy | And make sure to have overlapdial=yes |
| 14:16.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 14:16.57 | WIMPy | Hmm. If you have that, immediate=yes should be ok, shouldn't it? |
| 14:17.47 | WIMPy | Or in other words, my first answer is still the current one. |
| 14:19.15 | henny199 | so wht am i changing ? |
| 14:19.33 | WIMPy | overlapdial=yes |
| 14:20.31 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 14:20.36 | *** mode/#asterisk [+q chare!*@*] by Qwell |
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| 14:21.07 | henny199 | do i just need to add that line? |
| 14:21.28 | WIMPy | If it doesn;t exist, yes. |
| 14:21.37 | henny199 | under channels? |
| 14:21.43 | WIMPy | Before the channels that use DDI. |
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| 14:21.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:23.03 | Katty | PEANUT BUTTER PRETZELS |
| 14:23.32 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 14:24.04 | [TK]D-Fender | Katty: FTMFWY! |
| 14:25.05 | henny199 | so just check before i save |
| 14:25.06 | henny199 | http://pastebin.com/E00NM1EC |
| 14:25.54 | *** join/#asterisk Neozonz|DIsc (~arajakul@unaffiliated/neozonz) |
| 14:26.22 | Neozonz|DIsc | I'm having really bad coppy/bad quality audio on asterisks, any tips on what to look at? |
| 14:28.26 | WIMPy | henny199: After [channels], just before any actual channel. |
| 14:28.45 | WIMPy | So you probably should take a look at the included files. |
| 14:29.23 | [TK]D-Fender | NeoClealr describe All of the pieces involved. |
| 14:29.29 | WIMPy | BTW: Is there any use for being able to set overlapdial to no? |
| 14:29.37 | [TK]D-Fender | Neozonz|DIsc: Clearly describe all of the pieces involved. |
| 14:29.54 | [TK]D-Fender | 's coffer is slow to take hold this morning... |
| 14:29.58 | [TK]D-Fender | coffee* |
| 14:30.08 | Qwell | not enough hold, apparently |
| 14:30.24 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 14:30.26 | WIMPy | henny199: And then ask in #freepbx how to make the change permanent. |
| 14:31.07 | WIMPy | (or why you have to change it yourself in the first place) |
| 14:31.35 | Neozonz|DIsc | [TK]D-Fender, Spark + Asterisk + Openfire |
| 14:31.47 | Neozonz|DIsc | Spark <-> Spark VoIP Call via Asterisk |
| 14:32.17 | [TK]D-Fender | I'm talking protocols, what networks, what is your server running on, what's on both ends...... |
| 14:33.14 | Neozonz|DIsc | Spark (NAT) <-> (NAT) Asterisk <-> Spark (NAT) using GSM |
| 14:34.18 | [TK]D-Fender | Spark is not a protocol, and GSm is vague |
| 14:34.22 | WIMPy | I hope "GSM" is not to be taken literally. Otherwise that would be more than normal. |
| 14:35.33 | Katty | hugs Qwell |
| 14:35.37 | Katty | file: how are your tonsils doing? |
| 14:35.39 | Qwell | O.o |
| 14:36.05 | file | Katty, much better! |
| 14:36.11 | Katty | file: horay! |
| 14:36.45 | Qwell | It's a wonder what happens when you can go see a doctor when you start feeling sick. Stupid Canada.</jealousy> |
| 14:36.59 | Neozonz|DIsc | You wait, usually |
| 14:37.00 | Katty | nods sadly |
| 14:37.02 | MaliuatLap | Katty: you still believing in "squirrels"? ;) |
| 14:37.03 | Neozonz|DIsc | for a long long time |
| 14:37.16 | Katty | Neozonz|DIsc: you have to wait down here in the us, too |
| 14:37.21 | Katty | Neozonz|DIsc: walk in clinics can take hours |
| 14:37.29 | Katty | Neozonz|DIsc: or you make an appt with a GP and wait days |
| 14:37.32 | Neozonz|DIsc | oh but here, you go to ER and you wait maybe 12 hours |
| 14:37.45 | wonderworld | is PJSip considered as stable as chan_sip ? |
| 14:38.12 | MaliuatLap | Neozonz|DIsc: depends on what you have, and what your history with that hospital is |
| 14:38.13 | Neozonz|DIsc | [TK]D-Fender, Using gsm on SIP on clients and server |
| 14:38.26 | Katty | [TK]D-Fender: have you ever had to wait 12 hours in ER? |
| 14:38.50 | WIMPy | Neozonz|DIsc: What codec? And again: Really GSM? |
| 14:39.04 | Neozonz|DIsc | WIMPy, Codec is gsm |
| 14:39.16 | Neozonz|DIsc | allow=gsm |
| 14:39.17 | MaliuatLap | Neozonz|DIsc: I walk into RBWH, they go get my chart, and the Doctor asks me what I need (script pad in hand, whatever dosages I want - he doesn't want to read my chart) |
| 14:39.25 | WIMPy | Oh, so you're not using GSM? |
| 14:39.28 | [TK]D-Fender | Neozonz|DIsc: Already one of the worst. |
| 14:39.40 | Neozonz|DIsc | ? |
| 14:39.52 | [TK]D-Fender | Neozonz|DIsc: GSM is low bandwith.. AND low quality. |
| 14:39.52 | Neozonz|DIsc | i attempted ulaw, but same thing... figured gsm has less network overhead |
| 14:40.05 | [TK]D-Fender | GSM immediately cripples your quality. |
| 14:40.09 | Katty | i just need a good run of spring, i think. |
| 14:40.14 | [TK]D-Fender | it's a concession to lack of bandwidth |
| 14:40.24 | WIMPy | The overhead is the same. The payload is less. |
| 14:40.26 | MaliuatLap | [TK]D-Fender: well it's not speex |
| 14:40.27 | Neozonz|DIsc | but the voice was choppy |
| 14:40.29 | MaliuatLap | :) |
| 14:40.39 | Neozonz|DIsc | so I was thinking it was a bandwidth issue |
| 14:40.53 | WIMPy | And always remember that GSM codec in VOIP land is not what's used in actual GSM. |
| 14:40.54 | Neozonz|DIsc | like speaking to a martian |
| 14:40.59 | MaliuatLap | Neozonz|DIsc: do you know what your up/down limits are? |
| 14:41.05 | [TK]D-Fender | Neozonz|DIsc: choppy= JITTER |
| 14:41.09 | [TK]D-Fender | and PL |
| 14:41.20 | Neozonz|DIsc | MaliuatLap, I don't believe they are the issue since I can VoIP over X-Lite perfectly fine |
| 14:41.42 | [TK]D-Fender | Neozonz|DIsc: G.729 and iLBC handle those better... but shitty network quality is what it is.... |
| 14:41.44 | Neozonz|DIsc | but for some reason when I use spark I have choppy/martian voice |
| 14:41.55 | Katty | maybe you ARE a martian |
| 14:42.03 | Katty | MAYBE there are hooman skeletons on mars. |
| 14:42.04 | Neozonz|DIsc | [TK]D-Fender, it doesn't make sense for one client to work fine and another client to not work |
| 14:42.07 | Neozonz|DIsc | between same locations... |
| 14:42.11 | [TK]D-Fender | Neozonz|DIsc: Could be there is a software issue as well I suppose... I'd hate to think that was it;... |
| 14:42.12 | wonderworld | outer limits situation. |
| 14:42.14 | MaliuatLap | Katty: careful, his name may be Marvin ;) |
| 14:42.36 | Neozonz|DIsc | interesting bunch for a voip channel, you are |
| 14:42.47 | Neozonz|DIsc | jumps back on asterisk to modify codecs |
| 14:43.10 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 14:43.10 | MaliuatLap | we're all just nuts ... if Katty could get her hands on us she could feed her "squirrels" for eternity :) |
| 14:43.23 | Katty | nods. |
| 14:43.27 | WIMPy | LOL |
| 14:43.36 | wonderworld | WIMPy: what are the differences between "voip gsm" and "real gsm" ? |
| 14:43.46 | WIMPy | The codec. |
| 14:43.57 | wonderworld | a completely different thing? |
| 14:44.07 | MaliuatLap | gsm as a codec vs. gsm as a form of radio communication |
| 14:44.11 | WIMPy | There are several of them. The one we use (or don't) is the oldest and worst. |
| 14:44.24 | Katty | if you were getting a playstation for valentines day, what would you want to go with it? |
| 14:44.29 | Katty | sorry, PSP. |
| 14:44.34 | Katty | i should have clarified. |
| 14:44.49 | [TK]D-Fender | Katty: My guess: games |
| 14:44.50 | wonderworld | jailbreak it. run gameboy games |
| 14:44.57 | Katty | [TK]D-Fender: i was hoping more specific than just Games |
| 14:45.04 | carrar | Space Invaders! |
| 14:45.16 | carrar | Flowers? |
| 14:45.20 | Katty | they made a space invaders for psp? |
| 14:45.20 | wonderworld | revenge of gator pinball! |
| 14:45.20 | carrar | and chocolates? |
| 14:45.27 | Katty | he won't want chocolate and flowers. |
| 14:45.31 | MaliuatLap | carrar: no, you come here for "space invaders" |
| 14:45.37 | Katty | i was looking for Game Suggestions, or accessories. |
| 14:45.50 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 14:45.56 | MaliuatLap | Katty: a naked woman is a good accessory ;P |
| 14:46.00 | wonderworld | isn't PS VITA the next gen PSP? |
| 14:46.03 | WIMPy | A hammer is always a good accessory. |
| 14:46.08 | Katty | wonderworld: he already has one. |
| 14:46.21 | Katty | and i got him a wii u for his bday |
| 14:46.23 | wonderworld | doesn't it run all the PSP games as well? |
| 14:46.36 | Katty | i have no idea. |
| 14:46.38 | carrar | Katty, get something that requires going outside |
| 14:46.47 | Katty | i know that the cd drive on his psp died, which is why i'm getting him another. |
| 14:46.48 | MaliuatLap | Katty: have you considered getting him a life outside gaming? |
| 14:46.52 | wonderworld | you should better check ;) |
| 14:47.05 | Katty | why would i? |
| 14:47.09 | Katty | he's a collector. |
| 14:47.14 | Katty | stop asking me why. |
| 14:47.17 | Katty | give me game suggesetions. |
| 14:47.24 | WIMPy | hast to think of the lunch break in the park on the IT crowd. |
| 14:47.26 | MaliuatLap | Go |
| 14:47.34 | MaliuatLap | it's about the only game I play |
| 14:47.42 | Katty | MaliuatLap: is that for PSP? |
| 14:47.46 | MaliuatLap | but you need a board and stones :) |
| 14:47.54 | carrar | goes off and makes espresso |
| 14:47.56 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:48.01 | Katty | carrar: :> |
| 14:48.10 | MaliuatLap | I'm sure there is something for psp |
| 14:49.31 | wonderworld | Katty: http://www.ign.com/articles/2011/12/28/the-top-25-psp-games |
| 14:50.12 | Katty | wonderworld: ty (= |
| 14:50.38 | MaliuatLap | Katty: why can't you be like a normal girlfriend and just give him sex for valentines day? :) |
| 14:50.50 | MaliuatLap | Katty: instead of making us do all this work |
| 14:51.00 | Katty | being a normal girl is boring. |
| 14:51.06 | Katty | if i were a normal girl, i wouldn't be in #asterisk |
| 14:51.18 | MaliuatLap | there _is_ that |
| 14:52.02 | MaliuatLap | having problems seeing straight, should probably go to bed |
| 14:52.16 | Katty | probably |
| 14:52.24 | *** join/#asterisk vtone (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
| 14:52.27 | wonderworld | i guess no normal person would be in #asterisk |
| 14:52.56 | MaliuatLap | <- gothic freak sysadmin |
| 14:52.58 | *** join/#asterisk vtone (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
| 14:53.12 | wonderworld | <- freak freelancer |
| 14:53.53 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 14:54.41 | *** join/#asterisk vtone (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
| 14:55.15 | *** join/#asterisk vtone (~vtone@cpc35-bagu10-2-0-cust138.1-3.cable.virginm.net) |
| 14:56.41 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 14:56.45 | Neozonz|DIsc | ok looks like gsm is the only codec that i get sound on... |
| 14:56.50 | Neozonz|DIsc | all others I get nothing :( |
| 15:06.45 | *** join/#asterisk gmalsack (~gmalsack@162.210.116.242) |
| 15:07.02 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 15:08.55 | gmalsack | ok all. running sip peers in realtime db. rtcachefriends=yes hints work, but phone registrations eventually crap out with asterisk saying the password is invalid. rtcachefriends=no, phone registrations work fine, but hints stop working.... thoughts? |
| 15:12.23 | Neozonz|DIsc | man i don't get it x-lite works fine but using spark = choppyness |
| 15:13.18 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 15:15.53 | wonderworld | sounds like a problem with sparks config? |
| 15:22.18 | Neozonz|DIsc | yeah but nothing to really config... |
| 15:25.05 | wonderworld | maybe try other softphones as well to nail the problem down to sparks |
| 15:26.07 | wonderworld | my current fav is Zoiper on android |
| 15:27.59 | wonderworld | never failed on me, configurable, nice interface, can do IAX and is always in my pocket |
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| 15:28.17 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:46.53 | Neozonz|DIsc | anyone know if spark has any voip plugins? |
| 15:48.16 | [TK]D-Fender | What is this "Spark" you're talking about? |
| 15:49.17 | WIMPy | Well, that took an interesting twist. |
| 15:50.49 | navaismo | maybe a car, maybe a xmpp client |
| 15:51.38 | Qwell | [TK]D-Fender: a very dead IM client |
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| 15:53.33 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-218-093.pools.arcor-ip.net) |
| 15:54.12 | ghost75 | is it possible to change the allow features in sip message header? |
| 15:57.28 | navaismo | where? I mean at which point? |
| 15:57.56 | ghost75 | i was trying to add allow:UPDATE |
| 15:58.09 | ghost75 | in sip invite |
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| 16:00.29 | navaismo | hmmm nope I dont know |
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| 16:02.22 | Qwell | ghost75: if the endpoint supports it, it'll add it |
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| 16:19.10 | ghost75 | from what i understand it doesnt add: http://asterisk.inmte.com/viewtopic.php?t=24654&sid=f39eb015f34dbc049c2617e023b20add |
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| 16:22.56 | Qwell | ghost75: I'm not sure how that contradicts what I said. |
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| 16:23.50 | Qwell | As much as is wrong with chan_sip, it's doing exactly the right thing here. |
| 16:24.01 | ghost75 | it says: Asterisk does not advertise support for the UPDATE method |
| 16:24.09 | Qwell | Because it doesn't support it. |
| 16:24.29 | ghost75 | doesnt that mean then there will be ne UPDATE in sip invite? |
| 16:25.06 | Qwell | Rephrase your question. |
| 16:26.05 | ghost75 | its similar to here: http://forums.asterisk.org/viewtopic.php?f=1&t=88153&start=0 |
| 16:26.42 | Qwell | It's not supported. |
| 16:28.56 | ghost75 | then i dont understand: if the endpoint supports it, it'll add it <- if * not supports, how it should add |
| 16:29.06 | *** join/#asterisk wolrah_ (~wolrah@50.54.40.127) |
| 16:29.39 | Qwell | You will have to write the code to make Asterisk support UPDATEs. |
| 16:29.50 | ghost75 | yes |
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| 16:32.17 | ghost75 | it could explain why isp peer blocks invite with sip 480 because it expects UPDATE |
| 16:33.06 | apurvtwr | . |
| 16:34.26 | Qwell | I would sure as hell hope that's not what they respond with, on a failure. |
| 16:34.52 | ghost75 | couldnt find any other reason yet |
| 16:34.57 | Qwell | Ask them. |
| 16:35.12 | ghost75 | they do not answer such questions |
| 16:35.18 | Qwell | Then find a new provider. |
| 16:35.50 | ghost75 | takes much time |
| 16:35.57 | Qwell | rolls his eyes |
| 16:36.19 | ghost75 | 2 years on contract |
| 16:37.04 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 16:38.26 | verywiseman | are all codecs work with SRDP/TSL ? Also are all codecs work with all types of connections (i.e WIFI , 3G ,etc) ? |
| 16:39.12 | [TK]D-Fender | verywiseman: packets are packets |
| 16:40.03 | Penguin | Some packets are larger than others. |
| 16:41.07 | Penguin | And some data types require more or less packets to move the information. |
| 16:41.20 | [TK]D-Fender | It's not the size of the packet it's the quality of the payload |
| 16:41.34 | *** join/#asterisk DanielSa (~daniel@198.147.23.156) |
| 16:41.43 | [TK]D-Fender | likes big buffers and he cannot lie |
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| 16:42.07 | Penguin | I know, it's not the size of the pencil, it's how you leave your autograph. |
| 16:43.23 | verywiseman | Ok , but i am facing a problem , i have 2 * servers , one of them in LAN , the other is global , and i am using TLS/SIPS/SDRP , and i have 2 smart phones , when they register to local server , they are working good , but when they register to global server , the callee can't hear each other , what may the problem ? |
| 16:43.40 | [TK]D-Fender | I went to a disco the other night. They played 'Jump' and I jumped. They played 'The Twist', I did the twist. They played 'Come on Eileen'....got kicked out after that one. |
| 16:44.04 | Penguin | I'd imagine. |
| 16:44.09 | [TK]D-Fender | verywiseman: any number of things related to the networking.] |
| 16:45.02 | verywiseman | [TK]D-Fender, i didn't understand u |
| 16:45.17 | navaismo | no one does |
| 16:45.29 | navaismo | he is very complicated |
| 16:46.05 | [TK]D-Fender | verywiseman: What's not to understand? parkets either aren't being sent where they need to go, or are being blocked from doing so. |
| 16:46.25 | navaismo | & omnipres |
| 16:47.45 | Penguin | Without knowing all the details, I would look at the RTP ports. |
| 16:47.55 | navaismo | or nat settings |
| 16:49.16 | [TK]D-Fender | Without knowing anything about the call.... I'd start by realizing that nothing has been shown |
| 16:49.38 | Penguin | Oh come on... THAT part was obvious! |
| 16:49.44 | [TK]D-Fender | And I see no start having been made to actually LOOK at why since we were asked to GUESS why. |
| 16:49.57 | [TK]D-Fender | Why isn't it working? |
| 16:50.00 | [TK]D-Fender | 42 <---------------- |
| 16:50.03 | Penguin | That's the fun part about guessing. |
| 16:50.15 | navaismo | isn't tha for life |
| 16:50.20 | Penguin | Spend all your time guessing and no time getting anything DONE. |
| 16:50.21 | navaismo | or now it covers everything? |
| 16:50.28 | *** join/#asterisk karl-s (~karl.s@pool-173-51-92-61.lsanca.fios.verizon.net) |
| 16:50.37 | Penguin | Life, the universe, and EVERYTHING |
| 16:50.58 | navaismo | hmmm |
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| 17:55.32 | gmalsack | ok all. running sip peers in realtime db. rtcachefriends=yes hints work, but phone registrations eventually crap out with asterisk saying the password is invalid. rtcachefriends=no, phone registrations work fine, but hints stop working.... thoughts? |
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| 18:48.23 | pabelanger | gmalsack, what does the debug logs show |
| 18:48.30 | pabelanger | it should give you a reason for it failing |
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| 19:24.02 | dan_j | Hi. does anyone know of a cordless sip phone with BLF? |
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| 19:41.48 | rafuch0 | hi all |
| 19:41.56 | rafuch0 | a quick question |
| 19:42.31 | Chainsaw | rafuch0: Yes? |
| 19:42.59 | carrar | What color is the sky |
| 19:43.12 | rafuch0 | you make a call to internal phone you can hear the ringing all good yo make a call to outside you cant hear the ringing tone |
| 19:43.23 | rafuch0 | this is the log |
| 19:43.24 | rafuch0 | http://pastebin.com/RmyLMsvP |
| 19:43.26 | Penguin | It has an orange methane glow. |
| 19:43.32 | carrar | heh |
| 19:43.58 | rafuch0 | what cut it be? |
| 19:45.11 | rafuch0 | i had checked the parameters progressinband=yes and is not set in the sip.conf |
| 19:45.22 | rafuch0 | or custom |
| 19:45.36 | rafuch0 | this cut be a carrier problem? |
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| 19:56.24 | wonderworld | rafuch0: you could try to experiment with Ringing() and Progress(). |
| 19:56.58 | rafuch0 | how is that? |
| 19:57.45 | rafuch0 | wonderworld: |
| 20:01.17 | *** part/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269307.dsl.bell.ca) |
| 20:02.56 | Neozonz|DIsc | Anyone use Spark for voip? |
| 20:09.27 | *** join/#asterisk Skriza (~Skriza@200.195.149.61) |
| 20:09.45 | Skriza | hello people |
| 20:10.13 | Skriza | i need some help for setting new variables on asterisk 1.8.15 .. can anybody help me? |
| 20:10.58 | gmalsack | skirza: sure, that's why we are all here.... better than scratching our balls raw.... ;-) lol |
| 20:11.18 | Skriza | hehehe |
| 20:11.34 | Skriza | so Im tryng to set a new field on my cdr table |
| 20:11.50 | Skriza | when I use a fixed value, it goes like a charm |
| 20:11.52 | gmalsack | csv or stored via sql? |
| 20:11.56 | Skriza | mySQl |
| 20:12.00 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 20:12.11 | Skriza | when I put a variable, it does note store ... |
| 20:12.23 | Skriza | look at my extensions.conf line |
| 20:12.38 | gmalsack | pastebin sample code. |
| 20:12.46 | Skriza | exten => _XXXXXXX,n,Set(CDR(nmcont)=${nmcont}) |
| 20:13.12 | Skriza | if I Use Set(CDR(nmcont)=TEST) ... IT GOES PERFECTLY.. |
| 20:13.21 | Skriza | its crazy, isnt it? |
| 20:13.50 | gmalsack | where are you setting the value of ${nmcont} |
| 20:14.15 | gmalsack | try putting this one line ahead and check your logs to ensure the variable is being set... |
| 20:14.18 | Skriza | its a number, just line "12312312" |
| 20:14.43 | Skriza | yeah, i did it.. the variable is coming |
| 20:14.54 | gmalsack | exten => _XXXXXX,n,NoOp(nmcont -> ${nmcont}) |
| 20:15.42 | Skriza | <PROTECTED> |
| 20:15.45 | Skriza | it is coming.. |
| 20:16.00 | Skriza | but, its not storing just when it comes from variables... |
| 20:16.17 | Skriza | if I set a fixed value, there is no problems.. |
| 20:17.08 | gmalsack | my question is does ${nmcont} actually contain data when you are trying to set it's value to CDR(nmcont)... |
| 20:17.28 | [TK]D-Fender | There is no issue with variables |
| 20:17.34 | [TK]D-Fender | that isn't the problem |
| 20:17.53 | Skriza | yes, it is.. |
| 20:18.04 | gmalsack | ok |
| 20:18.05 | Skriza | Set("Local/6085@default-7280;1", "CDR(nmcont)=660164587") => that is the console line |
| 20:18.12 | [TK]D-Fender | No. Your sample is where you see a difference, but the fact there is a variable there is not it |
| 20:19.58 | gmalsack | oh. so in the console you're seeing a value where the variable is. so the variable is doing it's job correctly... check that you can add that value to your database table manually. |
| 20:20.33 | Skriza | yeah, i can do it |
| 20:20.41 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 20:20.46 | Skriza | manually |
| 20:21.12 | Skriza | the crazy thing is that if I set a fixed value, just like SET(CDR(nmcont)=TESTE) |
| 20:21.22 | Skriza | it goes to the table normally |
| 20:21.26 | gmalsack | try enabling debugging and check your full log. |
| 20:22.35 | gmalsack | like tk said, the problem isn't with the variable, because you can see in the console where the variable is replaced with a value. so the problem is elsewhere. you need to get past your hangup of the issue being the variable or you'll never find the problem. |
| 20:24.55 | Skriza | Ill make another tests... |
| 20:25.43 | Skriza | ill be back to tell you guys the results.. |
| 20:26.10 | gmalsack | newtonr: are the support guys really busy today? |
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| 20:32.54 | newtonr | gmalsack, no idea. I'm not in the office, we've had ice/snow issues, so many people may not be in. |
| 20:33.40 | gmalsack | ah. yea, lovin this "southern" winter hey! feel like I was back up in Wisconsin... lol |
| 20:34.21 | newtonr | :D |
| 20:34.37 | gmalsack | you know, I moved to lower alabama for a reason.... to get away from this crap!!! |
| 20:34.43 | gmalsack | :D |
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| 20:50.28 | dan_j | Hi, if I want to set a list of variables a few places in the dialplan, whats better, gosub or macro? |
| 20:52.06 | kaldemar | macro is deprecated, gosub is not. |
| 20:54.05 | dan_j | Ok. Thanks. I knew one of the was, but couldnt see it in the wiki |
| 20:57.02 | dan_j | Is there any real difference between the two? |
| 20:57.03 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Application_Macro warns you to use a GoSub. https://wiki.asterisk.org/wiki/display/AST/app_macro+Deprecation are the notes on the actual work that went into deprecated app_macro |
| 20:58.25 | dan_j | mjordan: thanks |
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| 21:01.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:10.34 | paulc | Is there a way to clear the Asterisk CLI history? |
| 21:12.38 | [TK]D-Fender | It has none. |
| 21:12.45 | [TK]D-Fender | Everything is shell CLI |
| 21:13.03 | wonderworld | i am trying to setup a webrtc peer with asterisk 12 and pjsip. is there documentation available? i am having a hard time to find the correct values for transport and enpoint in pjsip.conf for the peer. the websocket connection works, but i get UNAUTHORIZED no matter what i try... |
| 21:14.21 | *** part/#asterisk Zogot (~Leon@26.sub224.ddfr.nl) |
| 21:15.06 | paulc | [TK]D-Fender: I don't understand. In *CLI> I can cursor up and see history.. of CLI commands only.. and in the shell I invoke "asterisk -r" from, I can cursor up and see shell history (but not *CLI> commmands).. so.. what am I missing? |
| 21:15.22 | paulc | Or to ask another way: Where does the history of commands issued in *CLI> come from? |
| 21:15.58 | [TK]D-Fender | ah, COMMAND history... not sure |
| 21:18.19 | Qwell | paulc: ~/.asterisk_history |
| 21:19.24 | paulc | @Qwell: That's it! |
| 21:19.29 | paulc | doffs hat - Thanks very much :-) |
| 21:23.40 | [TK]D-Fender | checkout time, BBIAB |
| 21:24.50 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 21:35.14 | Kobaz | anyone seeing any fun attacks from the 188.138.0.0/16 subnet? |
| 21:35.34 | Kobaz | two different servers so far, someone is trying a specific extension bruteforce with thousands of requests per second |
| 21:35.49 | Kobaz | one attempt from 188.138.26.185 and one attempt from 188.138.125.63 |
| 21:37.29 | Kobaz | serverloft.com is the source |
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| 21:44.29 | gmalsack | later all... enjoy your evening... |
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| 21:49.18 | *** join/#asterisk ionface (~ionface@unaffiliated/ionface) |
| 21:49.24 | ionface | hello |
| 21:50.41 | ionface | I need help finding a service for a small business using an in-house asterisk box |
| 21:50.56 | *** join/#asterisk jmetro (ada52bb5@gateway/web/freenode/ip.173.165.43.181) |
| 21:50.57 | jmetro | ahoy |
| 21:51.03 | ionface | what's the service called again? I recall some kind of list on a wiki, but have no idea if that's current or not |
| 21:51.15 | ionface | ahoy jmetro |
| 21:51.34 | jmetro | ~book |
| 21:51.34 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:51.52 | ionface | <ionface> I need help finding a service for a small business using an in-house asterisk box |
| 21:51.52 | ionface | * jmetro |
| 21:52.04 | ionface | jmetro, does this link apply to me? |
| 21:52.19 | jmetro | ionface: No, that link was for me, i wanted the oreilly link |
| 21:52.24 | ionface | oh okay :) |
| 21:54.10 | jmetro | Ionface: I'm guessing youre looking for someone to maintain your asterisk box? |
| 21:54.42 | ionface | mmm, no, I'm looking for an upstream server to connect to POTS |
| 21:54.47 | ionface | or whatever it's called these days |
| 21:54.48 | jmetro | or are you looking for a .. as D-Fender would put it "Sip trunk provider" |
| 21:54.53 | ionface | that's the word |
| 21:55.04 | jmetro | depends on your area |
| 21:55.08 | jmetro | ~siptrunk |
| 21:55.08 | ionface | isn't there a better protocol than sip though? |
| 21:55.10 | mjordan | wonderworld: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 21:55.36 | jmetro | SIP is okay, it depends who you want to use i guess. |
| 21:55.46 | mjordan | wonderworld: WS connections are always a bit more complex. Before you jump into that, have you gotten a phone able to register and make calls? That's probably a better starting point than WebRTC connections (although they should work) |
| 21:56.16 | ionface | yeah, but there's a difference in what I have and what I want |
| 21:57.04 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 21:57.06 | ionface | I have 4 phone lines going through a sip to pstn card |
| 21:57.21 | ionface | and I basically want the reverse |
| 21:57.56 | jmetro | ionface, if youre VOIP and you want to put yourself on the POTS network and able to make POTS calls, a sip trunking provider is probably the best bet. you can google them and find one in your area. |
| 21:57.57 | ionface | I'd be routing incoming calls to those I believe... |
| 21:58.10 | ionface | yeah, okay |
| 21:58.11 | Chainsaw | ionface: So facing analog telephones instead of a phone exchange. |
| 21:58.18 | ionface | right |
| 21:58.23 | Chainsaw | ionface: That's "FXS". |
| 21:58.45 | Chainsaw | ionface: If you're sticking cards on a PCI card, those are green. Instead of FXO (for phone exchanges), which are red. |
| 21:58.58 | Chainsaw | modules I should have said really. |
| 21:59.06 | ionface | it's been a year or two since I've touched that machine, probably need to reinstall things and learn all over, but I'm estimating my costs now |
| 21:59.48 | ionface | Chainsaw, that's right |
| 22:00.03 | ionface | mkay, so I'll probably need more sip/voip phones |
| 22:00.40 | ionface | recommend any headsets? |
| 22:03.25 | ionface | hmm, maybe a few FXS cards are cheaper... though I have the PCs. |
| 22:03.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:03.38 | ionface | well, you've given me a lot of search terms, thanks a bunch guys |
| 22:08.59 | wonderworld | mjordan: thanks, got it working. was just a typo |
| 22:12.31 | *** join/#asterisk gmalsack (~gmalsack@162.210.116.242) |
| 22:13.46 | gmalsack | leifmadsen: did you use any linux os clustering components when you built up your * clusters? |
| 22:14.33 | leifmadsen | gmalsack: I have used linuxHA before |
| 22:14.44 | leifmadsen | that was for failover mostly |
| 22:15.10 | leifmadsen | the way I'd do it now is quite a bit more complex, but really, realtime sip peers (cached) and func_odbc based dialplan is a pretty good start |
| 22:16.29 | gmalsack | yea, that cached part is what's killing me. for some reason the phones register and randomly start dropping off with invalid password errors.... damn problems kicking my ass... lol |
| 22:29.08 | DevWork_ | just compiled and tried to start asterisk |
| 22:29.21 | DevWork_ | I get no logs generated and asterisk dead but subsys locked when starting via init.d |
| 22:33.29 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 22:38.56 | *** join/#asterisk SupaYoshi (~SupaYoshi@ip4da5d319.direct-adsl.nl) |
| 22:39.10 | SupaYoshi | any help converting a mp3 to wav for asterisk IVR? |
| 22:39.27 | SupaYoshi | Ive been trying, getting files out, but I keep getting different errors like ;) format_wav.c:94 check_header_fmt: Not a wav file 49 |
| 22:39.57 | SupaYoshi | Im trying from mp3 to wav. I have libsox-fmt-mp3 installed but still nothing. |
| 22:43.56 | SupaYoshi | nvm i got it |
| 22:53.44 | *** join/#asterisk vlad_starkov (~vlad_star@78.182.78.71) |
| 22:59.39 | wonderworld | SupaYoshi: ffmpeg -i ./input.mp3 -ar 8000 -ac 1 -ab 64 ./output.wav |
| 23:02.00 | gmalsack | should I be able to specify an opensips server as the outbound proxy on a sip device and expect asterisk to match the call to a configured peer? |
| 23:03.34 | gmalsack | i.e. I want to use the opensips server to load balance outgoing calls from my sip devices across multiple asterisk servers, and have the asterisk server know which sip device is making the call. |
| 23:04.14 | gmalsack | right now the asterisk servers all think the opensips server is making the call... kinda make sense... |
| 23:05.08 | gmalsack | but I can't think of any other way of doing so. Right now, the opensips server monitors the asterisk servers to know which ones are active and which ones are not, and will load balance the traffic across the active servers only. |
| 23:07.45 | Penguin | Can you make the proxy pass the SIP username? If so, try changing to type=friend rather than type=peer. |
| 23:08.19 | Penguin | type=peer matches on IP/port. type=friend tries username first. |
| 23:08.40 | wonderworld | gmalsack: maybe somehow with rewritehost() in opensips.cfg? (just a vague idea) |
| 23:09.20 | gmalsack | penguin: already type=friend |
| 23:09.31 | gmalsack | wonderworld: thanks! I'll look at that one... |
| 23:09.50 | Penguin | That's my only idea. I don't know very much about opensips. |
| 23:14.59 | ghost75 | when remote peer answers to sip read with 403 forbidden, how asterisk knows if the session registration is ok? |
| 23:16.56 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 23:20.37 | wonderworld | can you recommend any good read on SIP? i want to understand it. it still puzzles me. i am happy when it works but when i need to debug a SIP session i am lost most of the time. |
| 23:22.44 | DevWork_ | whats a decent softphone for testing all this stuff? |
| 23:23.42 | *** part/#asterisk mjordan (~matt@nat/digium/x-difwjtlziprxcyrg) |
| 23:26.59 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 23:28.07 | wonderworld | DevWork_: OS? |
| 23:28.20 | DevWork_ | windows 7 mostly. |
| 23:28.44 | DevWork_ | scripting support would be nice :) |
| 23:30.09 | wonderworld | sorry, i habe no idea about windows stuff, but i hear many people talking about xlite on windows |
| 23:30.17 | DevWork_ | yea im trying that now. |
| 23:30.53 | DevWork_ | I can't quite get it to link up with asterisk yet, all I did was install asterisk and add an [user] entry in sip.conf |
| 23:44.56 | *** join/#asterisk xaristax (~xaristax@189.242.191.250) |
| 23:46.06 | xaristax | Hey can someone help me im trying to transmit an audio once the Dial command is execute i already try U and Playback but only one of the channels ear the audio and the other one just silence |
