| 00:00.16 | igcewieling1 | they also point to perquisite |
| 00:00.58 | ageis | :P |
| 00:01.25 | ageis | who can we ask that'd be the ultimate authority on this matter? |
| 00:01.28 | igcewieling1 | goes back to socket programming in PHP |
| 00:01.54 | ageis | igcewieling1: ever work w/ websockets? |
| 00:02.44 | igcewieling1 | ageis: no. Too new, too poorly supported. |
| 00:02.56 | ageis | support is improving. |
| 00:02.59 | apb1963 | Webster made a mistake. |
| 00:03.05 | apb1963 | Please check your aglets. There has been a rash of aglet theft lately. |
| 00:03.22 | igcewieling1 | IP sockets have been around for 20+ years. That is stable enough for me. |
| 00:03.23 | ageis | looks up what an aglet is |
| 00:09.55 | apb1963 | Well this is just disheartening.... flibbertigibbety is at the bottom 50% of lookups on webster |
| 00:10.27 | apb1963 | What is this country coming to? |
| 00:11.44 | apb1963 | Maybe I'm looking at this all wrong... maybe so many people already know what it means that there's no need to look it up. Yes, that must be it. |
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| 00:13.23 | apb1963 | now once again happy with the status quo, returns to beating his head against a wall. |
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| 00:19.00 | igcewieling1 | apb1963 *nod* The reason nobody talks to me in the bars is because I'm so hot they are intimidated. 8-| |
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| 00:22.33 | apb1963 | That's easily fixed... get out there on the dance floor and jiggle your booty. In no time at all you'll attract lots of blind people that don't care what you look like. |
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| 00:25.12 | apb1963 | This form of dancing is an offshoot of "belly dancing". It's known as "booty dancing". It's known only to a select few, commonly known as the "in crowd". Once a member of the "in crowd", you'll be besieged with unlimited requests for your money. |
| 00:26.32 | apb1963 | It's a very powerful technique and with great power comes great responsibility. Booty dance wisely. |
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| 00:27.32 | apb1963 | Do not misuse the booty dance or you will be cast out of the "in crowd" and returned once more to your dreary life. |
| 00:49.11 | igcewieling1 | I believe you are referring to a "club" |
| 00:49.21 | igcewieling1 | bars don't have dance floors. |
| 00:51.35 | apb1963 | In that case, feel free to use a table if available, pool table, or climb up on the bar itself. |
| 00:51.47 | apb1963 | Alternatively, go to a club. |
| 00:51.51 | apb1963 | with a dance floor |
| 00:53.27 | apb1963 | This technique can actually be used anywhere. At a supermarket, 7-11, bowling alley, or at a stop light. |
| 00:54.40 | apb1963 | It is most effective when used consistently, especially if a stop light is your chosen venue. |
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| 02:55.11 | igcewieling1 | I really hate race conditions in code |
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| 03:03.00 | igcewieling1 | suddenly has a lot more sympathy for when those conditions get into the Asterisk code |
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| 05:08.59 | carrar | *Y*A*W*N* |
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| 05:46.45 | dorphalsig | Hi |
| 05:47.23 | dorphalsig | I just bought a Natural Microsystems card (Quad E1) |
| 05:49.08 | *** join/#asterisk linuxgeek (~linuxgeek@v6.home.uppal.mobi) |
| 05:49.26 | dorphalsig | I just bought a Natural Microsystems card (Quad E1) I thought it would work with DAHDI and it doesnt :(. I've read good amount of sites and I've found three main solutions. 1) Use Chan_CAPI 2)use some obscure chan_nms 3) use some kind of driver that Dialogic (they seem to have bought nms) has developed for dialogic's cards. |
| 05:49.43 | dorphalsig | Can anybody actually help me do the right thing? |
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| 05:50.26 | Rahail | Is it possible to put answer delay on asterisk even my sip trunk provider gave answer signal |
| 05:50.48 | dorphalsig | wait(x) then answer, no? |
| 05:51.03 | Rahail | they reason i need that.. they are using CDMA gateway as soon as call hit there end they give answer signal.. and I am having problem with that |
| 05:51.19 | Rahail | i try that way didnt work |
| 05:51.42 | dorphalsig | ur recieving call? |
| 05:51.51 | Rahail | i am calling out |
| 05:53.14 | dorphalsig | So let me see if I understand. You place a call, then wait a couple of seconds (for it to be answered), then you do something and hang up, right? |
| 05:54.29 | dorphalsig | I just bought a Natural Microsystems card (Quad E1) I thought it would work with DAHDI and it doesnt :(. I've read good amount of sites and I've found three main solutions. 1) Use Chan_CAPI 2)use some obscure chan_nms 3) use some kind of driver that Dialogic (they seem to have bought nms) has developed for dialogic's cards. Can naybody help me choose the right thiing to do because I am lost. |
| 05:54.44 | Rahail | WE place call.. then even we got answer signal from sip trunk provider we do not want give answer to our customer |
| 05:54.49 | Rahail | we need to put x s delay |
| 05:57.11 | *** join/#asterisk linuxgeek (~linuxgeek@v6.home.uppal.mobi) |
| 05:58.34 | dorphalsig | but what if the line is busy? |
| 05:58.43 | dorphalsig | I mean, if it rings it gets the answered flag? |
| 05:59.01 | Rahail | with this provider doesntmatter |
| 05:59.13 | Rahail | busy or answer or cancel as soon as call hit it send answer |
| 06:00.25 | dorphalsig | but how will you detect that? |
| 06:00.54 | Rahail | if they hear busy they will hang up |
| 06:00.58 | Rahail | busy tone |
| 06:01.01 | dorphalsig | because if you put a delay, it will be a blind delay. I mean, it will always execute whatever goes next |
| 06:01.10 | dorphalsig | ok |
| 06:01.57 | Rahail | thats the only way i can have my customer stop screaming at me |
| 06:02.08 | Rahail | i will just put 45S delay |
| 06:02.11 | Rahail | or maby more |
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| 06:02.20 | dorphalsig | and what happens when you do wait()? |
| 06:02.45 | Rahail | i try this |
| 06:02.52 | Rahail | exten => s,1,Wait(45) |
| 06:03.00 | Rahail | xten => s,n,Answer( ) |
| 06:03.06 | Rahail | sim like this didnt work |
| 06:03.13 | Rahail | exten => _X,1,Wait(45) |
| 06:03.20 | dorphalsig | I dont understand why are you answering if you are dialing |
| 06:03.42 | dorphalsig | it may be the time, but Im lost |
| 06:03.52 | Rahail | which you would do it |
| 06:04.11 | dorphalsig | exten => s,1,dial(TheNumberToBeCalled) |
| 06:04.21 | dorphalsig | exten =>s,n,Wait(455) |
| 06:04.34 | dorphalsig | exten=>s,n,DoSomethingCoolHere() |
| 06:04.50 | Rahail | ~pastebin |
| 06:04.50 | 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. |
| 06:05.50 | Rahail | http://bin.cakephp.org/view/643393060 |
| 06:06.02 | Rahail | this what i am using now and i need to implent on this |
| 06:08.12 | Rahail | http://bin.cakephp.org/view/643393060 @@@ dorphalsig |
| 06:08.14 | Rahail | can you see this |
| 06:08.18 | dorphalsig | exten => _X.,n,Dial(SIP/trunk4/${EXTEN},45) |
| 06:08.26 | dorphalsig | after that put the wait |
| 06:09.04 | dorphalsig | but it wont really do anything |
| 06:09.22 | Rahail | means |
| 06:09.26 | dorphalsig | because it will just stay on the dial untill the caller hangs up |
| 06:09.55 | Rahail | even after x second |
| 06:10.00 | Rahail | it will not send answer ? |
| 06:10.12 | Rahail | if they stay more then 45 second |
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| 06:13.34 | dorphalsig | Rahail: You are saying it did answer |
| 06:14.05 | dorphalsig | you are saying your provider is sending you an answer flag (which doesnt make much sense...) |
| 06:14.16 | dorphalsig | as soon as you place the call |
| 06:14.22 | Rahail | yes |
| 06:15.37 | dorphalsig | so as far as * is concerned the call is placed... |
| 06:16.01 | dorphalsig | that timeout is how long to wait for the answer flag |
| 06:16.31 | Rahail | so if i put that wait will it going to wait that time and then send answer to my users |
| 06:16.40 | dorphalsig | no |
| 06:16.46 | Rahail | oh |
| 06:16.59 | dorphalsig | its like the 'ring time' |
| 06:17.03 | Rahail | then is there way is it possible in asterisk dialplan |
| 06:17.19 | Rahail | i have ring time set it to 45s |
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| 06:17.48 | dorphalsig | yeah, but its how long asterisk will wait for an answer |
| 06:17.55 | dorphalsig | but its getting it right away |
| 06:18.11 | Rahail | yeap.. its getting answer right way |
| 06:18.21 | Rahail | i need to put delay on it |
| 06:18.26 | Rahail | if possible |
| 06:20.57 | dorphalsig | I dont think thats possible |
| 06:21.03 | dorphalsig | I dont know |
| 06:21.25 | igcewieling1 | Rahail: the solution is to tell your provider to stop immediatly answering the call |
| 06:21.38 | dorphalsig | thats what I was going to say |
| 06:21.44 | Rahail | igcewieling1 they do not have option in there GW |
| 06:21.51 | dorphalsig | scream to your provider |
| 06:22.03 | igcewieling1 | Rahail: then you cannot do what you want to do |
| 06:22.10 | Rahail | they giving me 20s graceperiod |
| 06:22.25 | Rahail | however this not helping me getting scream from our people |
| 06:22.28 | igcewieling1 | your provider just wants to bill you even when the call is not anwered by the far end |
| 06:23.01 | dorphalsig | dumb question but cant you just switch providers? |
| 06:23.11 | igcewieling1 | dorphalsig: that is his only other option. |
| 06:23.39 | Rahail | then i would have to go with tier1 provide however there rate is 3 times more then his rate |
| 06:24.01 | igcewieling1 | any decent provider of any tier does not unconditionally answer calls. |
| 06:24.38 | dorphalsig | Rahail: Could it be that the provider's equipment is damaged? |
| 06:24.56 | Rahail | its CDMA gw |
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| 06:25.07 | Rahail | and CDMA network as soon as call hit they give answer |
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| 06:25.12 | Rahail | they are different then GSM |
| 06:25.27 | Rahail | If you are in USA you will see the same use ATT and USE verizon or metropcs |
| 06:25.54 | Rahail | as soon as you clal on VZ or Metro they start showing timer however with ATT they dont show timer untill call is answer |
| 06:26.43 | igcewieling1 | there is not technical reason for that |
| 06:26.55 | igcewieling1 | s/not/no/ |
| 06:27.32 | dorphalsig | <PROTECTED> |
| 06:28.27 | igcewieling1 | I should say there is no technical reason between GSM and CDMA. Many SIP/cellular gatewats or even PRI/cellular gateways do that, but it is a design flaw in the gateway. |
| 06:28.49 | igcewieling1 | now, WHY are you using cellular for your calls when normal SIP ITSPs are so inexpensive |
| 06:28.53 | Rahail | igcewieling1 yes some gw have option you put delay but some dont |
| 06:29.03 | Rahail | overseas |
| 06:29.10 | igcewieling1 | Rahail: not a delay, just don't send answer until the far end answers. |
| 06:29.18 | igcewieling1 | Rahail: "overseas" is not a reason. |
| 06:29.52 | igcewieling1 | dorphalsig: return the card and buy something which is supported by DAHDI |
| 06:30.50 | dorphalsig | igcewieling1: I think I'll just buy something taht works... thats what I get for buying USD 50 cards... |
| 06:32.23 | dorphalsig | Althought it would be nice to be able to have it working |
| 06:32.42 | igcewieling1 | dorphalsig: likely you would have to write a DAHDI driver for it yourself |
| 06:32.59 | dorphalsig | :s |
| 06:34.24 | Rahail | <igcewieling1> Rahail: not a delay, just don't send answer until the far end answers. |
| 06:34.31 | Rahail | its hard to detect with this provider |
| 06:35.23 | igcewieling1 | Rahail: if you provider does not SIGNAL an answer there is nothing you can do. |
| 06:35.40 | igcewieling1 | It is like trying to make a cake when the only ingredients you have are eggs and dirt. |
| 06:35.59 | igcewieling1 | it won't matter what you do, you can't make a cake from eggs and dirt. |
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| 06:36.28 | Rahail | Ok thanx ... some one told me asterisk had option to on delay to send answer signal |
| 06:36.30 | Rahail | i guess they are wrong |
| 06:36.44 | igcewieling1 | Rahail: they are wrong. |
| 06:37.19 | igcewieling1 | Asterisk as a feature called call progress detection, but that option ends up randomly disconnecting your call and it only works with DAHDI interfaces |
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| 10:44.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 13:55.49 | vi390 | hi, I'm stuck at the following point - needing inspiration. Having a running AGI script. Reacts on Key's pressed. having problem with #-Key Works in GSM Phones (probably GSM-inband?) but not on VoipPhones (rfc2833?). Its always recognized as a "transfer" action. (which is what # means mostly. but) tried to switch that off (transfers off) but it does not work. Also tried dtmfrelax and dtmf.. auto. Any idea how I can get the # Key not being the key for "t |
| 13:55.51 | vi390 | ransfer calls"? |
| 14:00.35 | mmlj4 | you can use any key you want for transfers, if you're writing your own AGO dialplan |
| 14:02.45 | *** join/#asterisk bitwize (~bitwize@c83-253-251-219.bredband.comhem.se) |
| 14:05.25 | vi390 | mmlj4: ok, but if I want to disable transfers? My problem is, that the hash-Key sign does not come through to the agi-env... Ok found that i can disable it with T , maybe that will work |
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| 14:17.44 | vi390 | hmm, ok reading , that the "T" sign in extensions.conf makes a transfer by pressing # possible, but I do not have a T sign in the conf - hmm |
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| 14:25.50 | vi390 | ok also tried in features.conf blindxfer => #1;Its still using the # - hash as a "call-transfer" : I need to switch that OFF - How can I do that? |
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| 14:39.22 | vi390 | ok - it must be in some way, that "transfer on #" is activated by accident somehow. I dont have a T sign in my extension nor anywhere else explicitly defined to allow transfers when pressing # - Any Idea where to look to deactivate transfers when has Key is pressed? |
| 14:46.03 | vi390 | why is it not working adding : blindxfer => ###111 to features.conf => this should disable the "transfer call when press #" - right? It still activstes transfer, when I press # during a call. And cant figure out why |
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| 15:11.04 | igcewieling1 | go into the settings for your cell phone and set dtmf length to long |
| 15:11.49 | igcewieling1 | on MY droid 3 that is settings/call settings/dtmf tones. On your phone it is likely different |
| 15:12.42 | volga629 | Hello Everyone, I have question about asterisk presence |
| 15:12.46 | volga629 | https://fpaste.networklab.ca/JJCb/ |
| 15:13.49 | volga629 | I getting 489 that not understand request, Is any specific setting might missing ? |
| 15:25.26 | cusco | hi folks |
| 15:26.00 | cusco | using gmail gtalk... and asterisk chan_motif |
| 15:26.17 | cusco | when dialing from gmail to asterisk, then asterisk to pstn via SIP gateway |
| 15:26.29 | cusco | I'm not hearing the ringing in gtalk |
| 15:27.54 | cusco | the dial(sip-gateway) has the r option... |
| 15:29.52 | igcewieling1 | the "r" option almost never helps |
| 15:43.50 | *** join/#asterisk gusto (~gusto@ppp-212-114-234-56.dynamic.mnet-online.de) |
| 15:48.05 | atan | I see Fax For Asterisk doesn't support color. Anyone here use some other module or something to handle color faxes? |
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| 15:48.24 | igcewieling1 | none that I am aware of |
| 15:49.07 | atan | Was "color faxing" one of those things that just a few companies played with, but no adoption? |
| 15:49.32 | igcewieling1 | I suspect it has lots of patents on it. |
| 15:49.54 | atan | All I could come up with was ITU T.30e... hmm |
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| 15:52.03 | atan | Well thank you for the reply either way :-) going to check out this Hylafax thing, I've never used the thing before... want to know more! |
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| 16:07.16 | vi390 | igcewieling1: thanks for the answer about dtmf length: I can not adjust them in the Phone I use somehow. So need to go the way of disabling transfer call detection - somehow, its a weird behaviour from asterisk, to allways want to interpret # as transferCall |
| 16:12.08 | igcewieling1 | vi390: in my pre-coffee haze I though you were having trouble detecting DTMF. |
| 16:12.27 | igcewieling1 | Why don't you disable DTMF based transfers entirely? |
| 16:12.59 | vi390 | igcewieling1: yes if I would know how, that would be the solution :) |
| 16:13.40 | igcewieling1 | vi390: remove any T t W and w options from your Dial lines. |
| 16:14.13 | vi390 | igcewieling1: tried in features.conf blindxfer => ##### didnt work... |
| 16:14.41 | igcewieling1 | vi390: remove any T t W and w options from your Dial lines. and rename your features.conf file to some other name and restart asterisk |
| 16:15.07 | vi390 | igcewieling1: hmm yes , and thats the weird thing. Because I dont have any TtWw or such in the extension - Its directed to an agi script, without any options, but still wants do transfers |
| 16:15.21 | vi390 | ok renaming is good idea |
| 16:15.54 | igcewieling1 | I do not recommend dialing from inside an AGI. Make debugging hell to do and often doesn't work as you expect. |
| 16:16.54 | igcewieling1 | vi390: also enable agi debug to see exactly what the agi is doing |
| 16:17.51 | vi390 | igcewieling1: the agi works, it does what it should. Inband DTMF is working, but rfc2833 is not |
| 16:18.05 | igcewieling1 | vi390: I doubt that is your issue. |
| 16:18.34 | vi390 | ok, trying the rename fetaures, thats good idea, deactivates everything in it |
| 16:19.01 | igcewieling1 | enable agi debug and look at exactly what the Dial line being executed is. |
| 16:19.33 | vi390 | igcewieling1: ok maybe good idea @ debug -thx |
| 16:19.34 | igcewieling1 | vi390: maybe you have a channel or global variable set for DYNAMIC_FEATURES |
| 16:21.20 | vi390 | igcewieling1: well usually not, the setup should be very minimal, all what the setup does is handling to the Agi script, thats it. And agi is working as long as inband is used |
| 16:22.00 | vi390 | igcewieling1: if i use the phone with the rfc283 - boom there is the transfer bug |
| 16:22.26 | vi390 | igcewieling1: all other DTMF keys are working by the way, even in rfc2833 |
| 16:22.36 | vi390 | igcewieling1: only # is not |
| 16:23.19 | igcewieling1 | vi390: seems to me the exact opposite. rfc2833 dtmf works so the # is detected. With inband dtmf might not be working so astterisk doesn't detect the # |
| 16:24.02 | igcewieling1 | vi390: asterisk does not use features.conf unless told to do so. You need to find out where it is being told to do so. |
| 16:24.02 | vi390 | igcewieling1: hmm good point - but why does the agi script then pick up the "#" as value with inband? |
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| 16:24.30 | igcewieling1 | vi390: any number of possibilities, all of them very hard to figure out. 8-| |
| 16:24.36 | vi390 | igcewieling1: well I asume it is not using features.conf I have renamed it, and nothing has changed in behaviour |
| 16:24.37 | igcewieling1 | what codec are you using? |
| 16:24.55 | vi390 | igcewieling1: yep - very hard to figure out. |
| 16:25.09 | igcewieling1 | I thought you said renaming features.conf fixed the issue. |
| 16:25.14 | igcewieling1 | (12:18:34 PM) vi390: ok, trying the rename fetaures, thats good idea, deactivates everything in it |
| 16:25.28 | igcewieling1 | did you restart asterisk after renaming the file? |
| 16:25.46 | vi390 | codec = G711a-NB |
| 16:26.04 | vi390 | igcewieling1: shure @ restart /etc/init.d/asterisk restart |
| 16:26.22 | igcewieling1 | good. you should not expect inband DTMF to work with any codec other than g711 ulaw or alaw |
| 16:26.31 | igcewieling1 | you are using alaw |
| 16:27.05 | igcewieling1 | vi390: enable agi debug, set your verbose to 99 and set your debug to 99 and set dtmf logging in logger.conf |
| 16:28.02 | vi390 | igcewieling1: hmm - Well the script was used so far only for GSM phones - so maybe was no problem - but now has to be used for softphones and normal phones as well |
| 16:28.29 | vi390 | igcewieling1: ok doing debug |
| 16:28.39 | igcewieling1 | vi390: Cell phones use out of band dtmf when sending from the phone to the carrier. |
| 16:28.52 | igcewieling1 | then the carrier converts them to inband for transport to the rest of the PSTN |
| 16:29.06 | igcewieling1 | unless your GSM phone is connected directly to asterisk with no carrier involved |
| 16:29.08 | vi390 | igcewieling1: ok @ GSM , thats weird than, because GSm was always working |
| 16:29.26 | vi390 | there is carrier involved, |
| 16:29.29 | igcewieling1 | you can do a sip show channels while a call is up to see the codec the call is using |
| 16:31.04 | vi390 | ok if carrier converts to inband, then this explains why gsm was always working |
| 16:32.07 | igcewieling1 | how is asterisk connecting to the carrier? |
| 16:32.27 | igcewieling1 | using inband DTMF is the best way to have dtmf randomly fail. |
| 16:32.33 | vi390 | @connected = SIP-Protocol |
| 16:32.47 | igcewieling1 | vi390: likely your sip carrier is converting it to rfc2833 |
| 16:32.51 | vi390 | ok @ randomly fail; good to know |
| 16:33.17 | igcewieling1 | for example if you have a lost packet during inband dtmf then asterisk may detect the digit twice. |
| 16:33.19 | vi390 | igcewieling1: woah @ converting it to rfc2833 , thats very possible |
| 16:39.09 | igcewieling1 | sip show peer X where x is the peer name of your provider and your sip phone will tell you the configured dtmf mode. |
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| 16:42.12 | vi390 | ok nice to know. trying to debug agi, - Switched to rfc2833 on the calling phone now no keys are working. Also not showing in the Debug |
| 16:45.32 | vi390 | ok @ no key working, having had switched it to inband only ;) |
| 16:46.33 | igcewieling1 | where did you switch, on the phone or in the asterisk peer setting? they should both have the same dtmf mode setting |
| 16:47.26 | vi390 | igcewieling1: the sip channel has dmfmode=auto now; the phone has for testing rfc2833; |
| 16:47.36 | igcewieling1 | never set auto |
| 16:48.45 | vi390 | igcewieling1: its weird, the debug does not show any keys pressed, only streamed voicefiles (core debug 99) ; Another weird thing is - realy all keys are working even with rfc2833; except the # Key |
| 16:49.07 | igcewieling1 | maybe the # key is broken? |
| 16:49.07 | vi390 | ok @ never set auto; well I have to make it compatible with most setups out there |
| 16:50.09 | vi390 | hehe @ # key :) => if I press it, and its in rfc2833 mode, then asterisk starts "transfer call" instead of letting the key value through to the Agi script |
| 16:50.55 | vi390 | thats the whole problem. Maybe it is not a problem with dtmf, since I guess its coming in correctly. Its just Asterisk which wants to interpret it as Transfer call instead of letting it through |
| 16:51.52 | vi390 | I think so, because all other keys are working exactly how they should |
| 16:52.55 | vi390 | @transferCall: If it would be possible to switch OFF that feature, it would be worth a try. But havent found any possibility to switch off transfers when pressing # |
| 16:53.21 | vi390 | however only is a problem when switched to rfc2833, not inband |
| 16:54.03 | vi390 | on inband asterisk does not interpet it as transfer; on rfc283 it is interpreted as transfer call; weird I think |
| 16:55.55 | igcewieling1 | DTMF based transfers are configured in features.conf |
| 16:57.50 | vi390 | which is renamed to features_deactivated.con at the moment; in extensions.conf I have exten => _X.,1,AGI(/index.py) thats it, so there is no T or t or something. |
| 16:59.27 | vi390 | thats the only line |
| 16:59.39 | igcewieling1 | what does AGI debug show? |
| 17:00.01 | vi390 | only the played voice streams and their configuration |
| 17:00.55 | igcewieling1 | I wish you the best of luck |
| 17:01.11 | vi390 | hehe - well thanks :) |
| 17:06.01 | vi390 | also thank you for help |
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| 17:49.26 | atan | With the key you get for Free Fax For Asterisk, if you want to move from one server to another... it reads as if you can re-register once before needing to call. Can you just purchase another key from the store @ $0 and use that? Not trying to get free lines or anything, just avoid wasting their time on a freebie. |
| 17:50.22 | igcewieling1 | I don't believe you can re-register the free fax for asterisk, only the paid ones |
| 17:51.39 | atan | So it seems I will need to buy a second one, hmm. Could be worse. :-) |
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| 20:58.07 | Elico | hey I installed asterisk on centos 6.3 and when I send start command it states: |
| 20:58.08 | Elico | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 20:58.08 | Elico | Starting asterisk: [ OK ] |
| 21:01.43 | *** join/#asterisk liamjfoy (~liamjfoy@unaffiliated/liamjfoy) |
| 21:02.09 | liamjfoy | Hello folks, I'm struggling to register my devices on a new asterisk server. |
| 21:03.14 | liamjfoy | Keeps saying: SIP/2.0 401 Unauthorized when i 'set sip debug on' |
| 21:03.27 | *** join/#asterisk TooTall (~stephen@ip68-97-32-252.ok.ok.cox.net) |
| 21:03.34 | liamjfoy | However, sip.conf and extensions.conf look OK to 'me' - anyone give any pointers? |
| 21:05.43 | krapper | have you tried using a soft phone? that way you can copy and paste password from config and test that way? |
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| 21:13.48 | liamjfoy | krapper: just tried my android phone |
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| 21:18.27 | liamjfoy | http://pastebin.com/Ur39EyPq |
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| 21:39.52 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 21:41.37 | krapper | liamjfoy, any NAT in play? everything local? |
| 21:41.58 | [TK]D-Fender | public_ip = 178.79.147.XXX |
| 21:42.03 | [TK]D-Fender | not a valid parameter name. |
| 21:42.15 | [TK]D-Fender | And the dialplan syntax is 1.0 standard and certain to fail |
| 21:42.38 | [TK]D-Fender | The system is not setup to work from behind NAT at all currently |
| 21:46.57 | liamjfoy | The remote server isn't behind NAT, no. |
| 21:48.09 | liamjfoy | The server has a remote IP and the phone is behind NAT |
| 21:48.28 | liamjfoy | I've tried 'yes' in the 'nat' parameter but no luck. |
| 21:56.54 | igcewieling1 | liamjfoy: then remove the public_ip setting |
| 21:57.49 | igcewieling1 | also you want disallow=all and then allow= only the few codecs you use |
| 21:58.11 | [TK]D-Fender | your endpoints may be behind NAT and should be probably set as such regardless |
| 21:58.18 | [TK]D-Fender | And you need to disable directmedia |
| 21:58.38 | igcewieling1 | liamjfoy: did you read the Asterisk Book? |
| 21:58.39 | igcewieling1 | ~book |
| 21:58.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:59.07 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:06.35 | liamjfoy | igcewieling1: I tried removing it with no luck |
| 22:06.49 | liamjfoy | No I haven't read the book. |
| 22:06.53 | igcewieling1 | you should. |
| 22:07.08 | igcewieling1 | pastebin the output of a failure with sip debug enabled |
| 22:07.30 | igcewieling1 | also the output of sip show peer 2000 or 2001 whichever one is failing |
| 22:09.43 | liamjfoy | http://pastebin.com/dBtVm8U8 |
| 22:10.55 | igcewieling1 | liamjfoy: all sip transactions start with a request, a 401 with nonce, then a re-request with the password encoded with the nonce. I only see the first two packets there |
| 22:12.44 | liamjfoy | http://pastebin.com/fNUni7Ak |
| 22:12.45 | liamjfoy | second |
| 22:14.07 | igcewieling1 | DirectMedia : Yes you've not disabled direct media. You still have all codecs allowed. |
| 22:15.00 | igcewieling1 | even if the register worked the calls would fail because of those two things which we said to change |
| 22:16.34 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 22:16.47 | liamjfoy | http://pastebin.com/WV2XbKzZ |
| 22:20.12 | igcewieling1 | liamjfoy: either the passwords don't match between asterisk and the sip client or the client is doing something wrong. Have you tried a different sofrphone? Which softphone are you using? |
| 22:20.31 | liamjfoy | Default sip client on cyanogenmod |
| 22:21.44 | liamjfoy | igcewieling1: I see this in /var/log/asterisk/messages - [May 5 23:19:44] WARNING[19911] chan_unistim.c: Your OS does not support IP_PKTINFO, you must set public_ip. |
| 22:22.22 | igcewieling1 | liamjfoy: put noload => chan_unistim.so in your /etc/asterisk/modules.conf |
| 22:22.45 | igcewieling1 | you are using the SIP protocol, not the Unistim (Nortel) protocol |
| 22:23.03 | *** join/#asterisk gusto (~gusto@ppp-62-216-201-224.dynamic.mnet-online.de) |
| 22:23.12 | igcewieling1 | do a module unload chan_unistim.so in the Asterisk cli to remove it without restarting asterisk |
| 22:23.54 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 22:30.10 | liamjfoy | igcewieling1: OK, I'll try now - that error is when you start asterisk, not when I try to register |
| 22:30.32 | igcewieling1 | try a different sip client too |
| 22:30.45 | igcewieling1 | the error has nothing to do with your problem |
| 22:31.34 | liamjfoy | OK |
| 22:33.00 | liamjfoy | ha! Telephone on Mac OS X registers |
| 22:33.10 | liamjfoy | So it's something to do with this phone |
| 22:34.27 | igcewieling1 | or an incorrect setup on the phone. Often times you need to put the username and/or password in multiple places in the softphone config |
| 22:38.34 | liamjfoy | igcewieling1: yep! csip client on android works =) |
| 22:38.48 | liamjfoy | thanks! I appreciate it - it's a start =) |
| 22:38.59 | igcewieling1 | np. Now go read The Book |
| 22:39.57 | liamjfoy | Yeah, I'll look - if only I'd knew it would be the damn phone! |
| 22:45.08 | *** join/#asterisk italorossi (~italoross@201.76.144.193.digi.com.br) |
| 22:46.24 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-152-106.pools.arcor-ip.net) |
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| 23:15.12 | *** part/#asterisk Elico (~Thunderbi@109.65.188.44) |
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| 23:39.24 | *** join/#asterisk theqkash (58c76278@gateway/web/freenode/ip.88.199.98.120) |
| 23:39.27 | theqkash | hello |
| 23:39.39 | theqkash | can anybody help me with setting up asterisk gui? |
| 23:39.47 | theqkash | i have some problem here and don't know what to do |
| 23:42.22 | drmessano | Asterisk GUI is kinda extinct |
| 23:42.47 | theqkash | so what to use instead? |
| 23:43.10 | theqkash | I'm new into this and don't want to go through configuration files to do something simple |
| 23:43.12 | drmessano | FreePBX |
| 23:46.34 | igcewieling1 | theqkash: there is no such thing as a "simple setup" for a PBX. |