IRC log for #asterisk on 20131212

00:01.06*** join/#asterisk cmendes0101 (~cmendes01@office.phone.com)
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00:35.48phixPenguin, outtolunc: ok so I need to do that for each external dial plan entry I have?
00:36.02PenguinAgain, I'm not seeing the whole picture.
00:36.05*** join/#asterisk MrGate (~MrGate@74-140-14-32.dhcp.insightbb.com)
00:36.08MrGatehi
00:36.23MrGateon one of my extensions it says the forceport is not set
00:36.29MrGatewhen i use command sip show peers
00:36.40MrGateis there a way i can set that
00:36.40PenguinExtensions don't have ports, but devices do.
00:36.43phixPenguin: would you like me to pastebin my entire sip.conf and extensions.conf?
00:37.03MrGateim using x-lite softphone
00:37.07MrGateon the local network
00:38.28Penguinphix: If you want to, but I don't think it would help anything.
00:38.52PenguinI'll take a look at what you've already done, though, if you paste it.
00:44.24MrGateis there a way to set forceport
00:45.04PenguinI think maybe you are looking for nat=force_rport in your sip peer entry.
00:45.21MrGatewhere the sip peer entry
00:45.40PenguinBut if you are on the same LAN as asterisk, you don't need to force rport.
00:45.59PenguinThat's a NAT setting, and there is no NAT between your phone and asterisk if you are on the same subnet.
00:46.18MrGateso turn off the nat for the phones on the same subnet
00:46.19MrGate:)
00:46.25MrGatehere the issue
00:46.27MrGate3366 works
00:46.29Penguinusually, yes
00:46.33MrGatecalling out to a number works
00:46.43Penguinnat=no in most cases where you are on the same subnet.
00:46.44MrGatebut calling from a ext to a ext there is no audio
00:46.57PenguinYou can't call "from" an extension.
00:47.10MrGateit worked before
00:47.13PenguinPhones send calls TO extensions only.  What asterisk does next is up to you.
00:47.57MrGatelike this ext is trying to use port 61736 ?
00:48.04PenguinExtensions don't use ports.
00:48.16MrGatesip show peers
00:48.18MrGatesorry
00:48.24MrGatethe sip
00:48.27PenguinWhat about sip show peers?
00:48.37PenguinThe peer is using the wrong port?
00:48.46PenguinIf so, go fix the peer entry in sip.conf.
00:49.06MrGatemy sip.conf just includes a load of other configs
00:49.23Penguin~freepbx
00:49.23infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
01:06.12XandriXis apparently really good at breaking things
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01:34.43phixPenguin: extensions.conf: http://pastebin.com/c6zq3K7D , sip.conf: http://pastebin.com/ncSePRV8
01:36.36phixextensions.conf: http://pastebin.com/XfYg2VsU  (ignore the other one :))
01:44.27PenguinUh, that isn't an extensions.conf.
01:49.30*** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net)
01:59.05phixhaha oops I meant ignore the sip one :P
01:59.18phixthe first one that is, Ihad a typo in it
01:59.24phixPenguin: Thoughts?
02:01.30Penguinphix: It would be so easy to just set a variable for the peer you want to use to dial out and another for the callerid number.  See this paste for concept: http://pastebin.com/dg2cCuAL
02:04.35PenguinYou're REALLY complicating it.  It's not that hard.
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02:24.32XandriXPenguin: is there something special i need to do with my sip trunk for proper outgoing callerid ?
02:25.09PenguinSet the CALLERID(num).
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02:26.44XandriXooh woops i did this completely wrong in my defined peers i would do something like this callerid<desired callerid>
02:27.00PenguinYep, that would be incorrect.
02:27.35PenguinWell... let me rephrase.
02:27.59WIMPyThere's a = missing.
02:28.18PenguinYou can set the callerid value per peer, but most people don't want a 10-digit phone number showing up when they call another local phone.
02:29.35XandriXi might be looking at the wrong option i wanted something along the lines of if i call from said extension lets say it displays xandrix mobile sort of like when a company calls you and it displays the company name
02:29.53phixPenguin: I am complicating it?
02:30.12PenguinYou don't call FROM extensions.  I'm kind of getting tired of saying this and I would prefer you learn it quickly.
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02:32.43phixPenguin: so I would need to repeat those 5 lines for each of the external dial plan numbers?
02:33.31phix_NXXNXXXXXX doesn't cover all of the external phone numbers
02:34.17phixPenguin: wouldn't it make more sense to call a macro for each external extension pattern when doing the dialing?
02:34.50phixso I dont have to repeat the 5 lines, that may change in the future, wouldnt it be better to change it at one stop instead of multiple times/
02:35.26phixstop = spot
02:36.11WIMPymacros are deprecated
02:36.25WIMPyAny reason not to just use a Goto()?
02:36.53*** join/#asterisk jasonwert (~w3rt@71.89.137.28)
02:37.16phixno, I don't mind using Goto if macros are deprecated
02:37.48phixso, _XX.,1,goto(dialout,s,1) ?
02:38.06WIMPyI didn't read the beginning, so I was just asking.
02:38.33phixI don't have to use a macro but I dont want to repeat myself when I can stick it in a function or something that I can call on
02:38.41WIMPyBut that doesn't look like you need multiple extensions.
02:39.25WIMPyAnd _XX. only makes sense if you use overlap dialling.
02:39.36phixWIMPy: That is what I have atm: extensions.conf: http://pastebin.com/c6zq3K7D , sip.conf:  http://pastebin.com/XfYg2VsU
02:41.19WIMPyWell, as Penguin suggested, setting the callerID via a variable might be easier, at least for sip devices.
02:41.21phixWIMPy: how would I pass an argument to a context? this for example:   exten => _+XXX.,1,macro(dialout, ${EXTEN:1})
02:41.55WIMPyAnd for which part is that pattern/macor or not thing?
02:42.06phix?
02:42.27phixhow would I get the same effect as exten => _+XXX.,1,macro(dialout, ${EXTEN:1})
02:42.35Penguinphix: What is your obsession with macro?
02:42.44WIMPyWhat part of your dialplan are we talking about here?
02:42.50phixIt makes sense
02:42.59phixWIMPy: the bottom, matching for external numbers
02:43.14phixPenguin: Coming from a programing background I like functions :)
02:43.17WIMPyYou can use Gosub() instead of Macro(), if you want to return.
02:43.29phixno I want to pass in an argument
02:43.36phixor am I going to use globals for that?
02:43.45WIMPyIt's already in a variable.
02:43.55PenguinSo far you haven't shown anything that you need to pass in as an arg.
02:45.00phixPenguin: ${EXTEN:1}
02:45.05WIMPyYou just Goto(othercontext,${EXTEN},1)
02:45.21phixso I then have to repeat the match?
02:45.29phix_+XXX.
02:46.02WIMPyOr you don't do anything and just put that stuff in the internal context, starting at priority 2.
02:46.10WIMPyLots of possibilities.
02:46.11Penguinwimpy: This was the concept I suggested for his specific application: http://pastebin.com/dg2cCuAL
02:46.34WIMPyI just told him that it was a good Idea.
02:47.00PenguinAh, I was trying to do other things as well as read this.  I thought maybe you missed the paste.
02:47.40phixwhen I use set within a context, if the variable doesn't exist am I created it in the scope of the context? or is it now global?
02:47.48WIMPyOr use AstDB to store internal to external callerID mappings.
02:48.10WIMPyYou're mixing things up.
02:48.19WIMPyVariables belong to channels.
02:48.22phixWIMPy: I would like to modify my add user script to setup the phone for them as well
02:48.29WIMPyGlobal variables are accessible from all channels.
02:48.58WIMPyThere's nothing more local than a channel.
02:49.14phixso when I use n,Set(DialPeer=${IF($[${EXISTS(${dialoutpeer})}]?${dialoutpeer}:peer3)})   DialPeer is only acceptable within outbound context?
02:49.27phixa channel is a context?
02:49.28PenguinNope.
02:49.39PenguinThe variable is carried on the channel.
02:50.10WIMPyThere's no relation between variables and contexts.
02:50.11phixThe channel being the SIP phone that is dialing?
02:50.24PenguinThe phone creates the channel.
02:50.27phixok
02:50.29WIMPyNo, the channel the call created.
02:50.29phixI get that now
02:51.23phixso if I specify a global then use set within a context / channel, does the global then change globally or only within that channel?
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02:51.44WIMPy>>Global variables are accessible from all channels.
02:52.06WIMPySee function GLOBAL
02:52.14phixWIMPy: so if I re-define a global variable then it applies to all channel from that point on?
02:52.49WIMPyAll variables are local to the channel unless you use GLOBAL().
02:54.36WIMPyBut all global variables are copied as local variables when a channel is created.
02:57.41phixok
02:58.50phixso same => is a thing now?
02:59.25PenguinIf I used it in the example, it is.
02:59.35phixso sub's accept variables?
03:00.02PenguinSubroutines are just part of dialplan; dialplan allows for variable expansion.
03:00.25phixok, well I will go the subroutine approach
03:00.31phixit looks like a macro anyway
03:00.33PenguinIt isn't needed.
03:00.40phixit isn't?
03:00.54PenguinStop obsessing over subroutines.
03:01.06phixbut your example doesn't cover patterns that dont match _NXXNXXXXXX
03:01.14PenguinChange the pattern!
03:01.27phixI have 5 of them, so I am repeating myself 5 times?
03:01.34PenguinI gave you an EXAMPLE.  Change and add what you need to change and add.
03:01.35smirkersmirks
03:01.35phixinstead of calling a subroutine?
03:01.48PenguinYou have 5 different patterns?
03:01.52phixyes
03:02.07PenguinCan you list them now?
03:02.15phixin the paste
03:02.17atanAnyone in here subscribe to some Cisco package the lets you access firmware?
03:02.47PenguinIf you'll tell me the five different patterns, I'll incorporate them into the example I gave you.
03:02.51phixhttp://pastebin.com/c6zq3K7D
03:04.06phixsorry there are 9 of them: 000, _0NXXXXXXXX, _1[23]XXXX, _1300XXXXXX, _1800XXXXXX, _190XXXXXXX, _XXXXXXXX, _+XXXXX., _0011X.
03:04.47PenguinDo they all do the exact same things after the match?
03:04.55smirkerHi guys. CDR question. I have a context that sets CDR(userfield)=A, then sends the caller into a queue (consisting of SIP peers). The call gets answered, and the sip peer does a blind transfer (REFER). This initiates another context (on the same channel). My problem is that in this context, on the first line, if I view CDR(userfield) it equals "AA;A"
03:05.01phix000 I want it to basically establish a connection no matter what, using any trunk, and disconnecting an existing connection if needed
03:05.01smirkerany ideas what would be causing this?
03:05.08PenguinIs the extension pattern the only thing that differs?
03:05.10phixPenguin: no
03:05.33smirkerAsterisk 11.6.0
03:06.11phix_190XXXXXXX for example doesn't do a dial, international calls do the same thing atm but I will be using a different trunk in the future for those
03:06.25phixsame with 12, 13, 1300 and 1800 numbers
03:06.48phix000 being the Australian emergency number
03:07.30WIMPyugh, yet another one.
03:07.40phixsup WIMPy ?
03:08.06WIMPyEmergency number.
03:08.07*** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com)
03:08.15phixah :)  yeah
03:08.28PenguinIf each extension pattern will do something totally different, create another independent extension for it.
03:08.56PenguinIf only the first line of it is different, feel free to use a Goto().
03:08.57phixPenguin: they may do one thing different, then do the same 5 lines as in your example
03:09.06PenguinGoto()
03:09.11WIMPyI don't see them doing different things so far.
03:09.21PenguinI haven't either.
03:09.35PenguinIn the case of _190XXXXXXX, he did say it won't Dial().
03:09.53PenguinSo you'll have an extension for _190XXXXXXX that does something else.
03:09.59WIMPyYes, but the others are the same.
03:10.00PenguinStill no place for a subroutine.
03:10.28phix190X being premium rate services like phone sex (if that still even exists any more due to Internet), TV / radio competitions, etc..
03:10.59phixPenguin: it is if I am repeating myself
03:11.06PenguinNope.
03:11.10PenguinThat's what Goto() is for.
03:11.35phixexample please
03:11.40PenguinOne moment.
03:11.59WIMPyJust replace your Macros with Gotos.
03:12.33phixand I need to especitly call hangup?  or isn't fallthrough set by default?
03:13.18WIMPyTher's always the chance that some other parrern gets you if you don't Hangup explicitely.
03:13.44WIMPys/rr/tt/
03:15.44phixok
03:18.46Penguinexample  http://pastebin.com/dgPmqHuV
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03:25.03PenguinAny question?
03:25.22phixPenguin: ok looks good except I will change _1NXXNXXXXXX to dialout, as I want to match 12 and 13 numbers which are 6 in length
03:25.52PenguinI'm only providing examples for NANP.  Your patterns will be whatever your people dial.
03:25.58phixthe problem there is I can only use that within that context, which is fine I suppose, I just use goto in other contexts
03:26.15phixNANP? that a US thing?
03:26.22Penguin~nanp
03:26.42PenguinWhat?!  No tidbit in the bot?
03:26.50phixThe bots dont even know :PO
03:26.53PenguinNorth American Numbering Plan
03:26.57phixah
03:27.14phixyeah I am in AU but I have those numbering plans :)
03:27.32PenguinWhy do you need more than one context for the extension?
03:27.54phixactually i cant use dialout as the exten name because I dont know what extension I am in
03:28.02phixand I cant use _X. as it will match too early :\
03:28.13WIMPy?
03:28.57PenguinI have a feeling you don't understand extensions.
03:29.08WIMPy_X. won't match "dialout", only extensions starting with digits. What are yu mixing up there?
03:29.13phixand simutaneous calls differs depending on the out going trunk
03:29.25phixWIMPy: I know
03:29.28Penguin_X. is one digit 0-9 AND one or more additional characters.
03:29.29phixthat isn't what I said
03:29.33phixPenguin: yes I know that
03:29.46PenguinSo it doesn't match too early like you said.
03:29.51phix= 14:25:22 < phix> Penguin: ok looks good except I will change _1NXXNXXXXXX to dialout, as I want to match 12 and 13 numbers which are 6 in length
03:29.55phix= 14:27:54 < phix> actually i cant use dialout as the exten name because I dont know what extension I am in
03:30.05Penguin_X. matches both 10 and 1000000000000000000000.
03:30.33Penguinand many other things.
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03:31.30phixPenguin: correct
03:31.47phixany way I will try and modify what I have done so far and see if it works
03:31.50PenguinI'm not saying _X. is the best pattern, but I'm saying that what you said about it wasn't accurate.
03:31.52phixthnx for the assistance
03:32.34PenguinIf you need 10 different patterns that all do the same thing after the match, use Goto() like I have in the example.
03:32.49phixPenguin: also do I need to specify the overlimit count in extension?  or can I get that from my sip definition?
03:32.55PenguinIf you need 10 different patterns that all do different things, create 10 different extensions.
03:32.57phixas the count differs depdning on the trunk used
03:33.23PenguinFeel free to set a variable per device and refer to it in dialplan.
03:33.32WIMPyDefine them as global variables with the trunk number as index or use AstDB to store it.
03:33.52PenguinBe sure to write a failsafe into it for when you forget to assign it.
03:34.08phixok
03:34.16PenguinYou don't want it to barf when checking the limit if the variable is null.
03:34.33PenguinYou can do the check just like I did the callerid check.
03:34.54PenguinIF() and EXISTS()
03:35.23PenguinIf it exists, use it.  If it does not exist, set it to a hard-coded value.
03:35.45PenguinThe hard-coded value could be a global variable.
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03:38.35phixok
03:41.17PenguinI can visualize using a gosub to do checking and setting of data like that if you need to do it in multiple (different) extensions.
03:41.41PenguinThat would be a valid use of a subroutine.
03:44.49smirkerOK guys. Prepare to wrap your minds around THIS: there is MASTER_CHANNEL().  is there anything like PARENT_CHANNEL() ?  I have channel A spawning B spawning C, and in C i want to set a variable in B. -_-
03:45.17smirkerand that variable is actually CHANNEL(musicclass)
03:45.59PenguinYou want to set it in B and then what?
03:46.18smirkerGet paid.
03:46.29PenguinThat is outside the scope of asterisk.
03:46.37smirkersmirks
03:47.20smirkerJust trying something lads. I'm wondering is MASTER_CHANNEL is actually PARENT_CHANNEL. In that case, I might want to achieve something else and could use MASTER_CHANNEL(MASTER_CHANNEL()) ;p please hold.
03:52.37smirkerAh ok, no, my original question holds.
03:53.43smirkerPreviously I had A->B. A had CHANNEL(musicclass)=intro, and when the queue was answered executed a gosub routine to set MASTER_CHANNEL(CHANNEL(musicclass))=normal, so when the peer that answered the call put the caller on hold, it would play different music.
03:53.46smirkerWorked perfectly.
03:54.13smirkerNow, I've had to implement A->B(intro music,queue)->C(on answer, sets musicclass to normal)
03:54.29smirkerbut in C, MASTER_CHANNEL(CHANNEL(musicclass)) changes it on the A channel, when I need to change the B channel.
03:55.09smirkerso ideally I'd like to access the variable space or channel variables of the parent channel... or if possible, any channel specified by name.
03:57.20dar123anyone knows how to run debugs for wctdm24xxp
03:58.29Penguinsmirker: Are you saying that within C, MASTER_CHANNEL() is A rather than B?
03:59.13smirkerPenguin: yes, most definitely.
03:59.17PenguinThat almost seems like a bug.
03:59.25smirkerIt defies the documentation.
03:59.40smirkerLet me run one more sanity check -- 60 seconds
03:59.42PenguinMASTER_CHANNEL() is supposed to be the channel that created the current channel.  In your case, it should be B.
04:00.06Penguins/your case/the case of C/
04:01.10smirkerMASTER_CHANNEL(test) is CHANNEL A
04:02.10smirkerI set test=CHANNEL A on the first channel, Dial(Local/..../n) to second channel and set test=CHANNEL B, then Queue(blah,,,,,,,,gosub-routine) to reach third channel.  In third channel, ${MASTER_CHANNEL(test)) gives CHANNEL A
04:03.26smirkerobviously the first two channels are separate channels, because they have their own variable space (otherwise test would be overridden)
04:03.55PenguinThat just seems wrong.
04:04.11smirkerI know right. It's causing me headaches.
04:04.46PenguinI think you need to talk to the devs about it and probably file an issue on jira.
04:05.19smirkerPenguin: Will do. Should I file an issue first? Never done this before.
04:05.58PenguinI would talk to the devs tomorrow to see if they feel it does in fact need an issue filed.  They will probably say it does.
04:06.17smirkerHow do I reach the devs?
04:06.25Penguinpabelanger might even be listening now.
04:06.53pabelangerhuh
04:07.25smirkerpabelanger: hi, see above starting from *smirker smirkers
04:08.27smirkerIn the case where channel A spawns channel B which spawns channel C. Running MASTER_CHANNEL() from channel C accesses channel A. There is no way to access channel B. Documentation states: Allows access to the channel which created the current channel, if any. If the channel is already a master channel, then accesses local channel variables.
04:08.53PenguinIf channel A creates channel B, MASTER_CHANNEL() should be channel A.  If channel B creates channel C, I think within C MASTER_CHANNEL() should be B.  But it is going all the way back to A instead.
04:08.58pabelangerNot sure, but all I do know is you should be using MASTER_CHANNEL() there is a performance hit to using it
04:09.24PenguinHmm?  Restate?
04:10.19pabelangermy bad: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
04:10.22pabelangerit was SIP_CAUSE
04:10.39pabelangereither way, can't help right now. Doing a customer install
04:10.48PenguinWould you recommend filing an issue regarding MASTER_CHANNEL()?
04:11.19PenguinOr raise the topic again tomorrow during business hours?
04:13.10smirkerPenguin: I think it would be best to redefine MASTER_CHANNEL in the documentation to specify its current behaviour, or you might risk breaking people's dialplans.  And additionally, implement PARENT_CHANNEL() which exhibits the behaviour we're expecting.
04:14.25PenguinI think I'd bring up the topic tomorrow in here and/or in #asterisk-dev during business hours.
04:15.07smirkerWhat timezone are business hours specified in?
04:15.25PenguinUS central
04:16.10Penguin1500-2300 GMT
04:16.37smirkerAlright, great.
04:16.57PenguinI don't know your TZ, so I figured going from GMT might be easier.
04:17.05smirkerMost definitely :)
04:18.20smirkerSo in the meantime, would there be a way to set channel variables on a channel specified by name?
04:18.50smirker(Ignoring all the potential synchronisation issues)
04:21.04smirkerSHARED() isn't very useful in this case ;p
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04:46.16flingAre you logging to postgresql or mysql?
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04:59.38atan~sipnat
04:59.38infobot[~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet.
04:59.58atan^ top link in that is 404 :(
05:01.13PenguinHe knows.
05:01.29PenguinJust follow the "typically, you must..." part.
05:04.35[TK]D-FenderI do.
05:04.43[TK]D-FenderI'll re-write it this weekend
05:07.50atan:-) am just hitting trouble with a device not working as expected after awhile
05:08.19atanOh now I'm loving these Cisco phones ^_^ I'm sure once I get it right they'll be great
05:09.52atanIs there anything in http://pastebin.com/cNzu924Z that jumps out as being wrong or needing adjustment? The trouble is the Cisco phones drop off after awhile and can't rx calls
05:11.01ChannelZif it's behind NAT it might be timing it out of the nat table.
05:11.40atanAsterisk is sitting out in the open, the connecting clients are not most are behind some type of consumer router which I have little if any control over
05:11.58ChannelZThat's what I mean, the phone
05:12.07atanIs there a setting I can force on the peer to reconnect after X or what have you?
05:12.08ChannelZYou can try using qualify
05:12.22atanLooking in the wiki for qualify now
05:12.38ChannelZsip.conf, it's in there.
05:14.03atanSilly question, am I even in the right spot? https://wiki.asterisk.org/wiki/display/AST/Configuration+and+Operation
05:16.12ChannelZI don't think it's documented there in any complete way. Just look at the sample sip.conf.
05:17.56PenguinJust set qualify=yes in each phone entry to turn it on.
05:18.30ChannelZThough you might need to make it a lower value if it IS nat timeout.
05:18.44PenguinThat setting won't affect that.
05:18.45ChannelZor wait that one defaults to once a minute doesn't it
05:18.55ChannelZwhich is probably fine
05:19.36Penguinqualifyfreq is the one that keeps the nat active, which should default to 60.  And it will be used once qualify is set to yes or a numeric value.
05:19.57ChannelZyes, that
05:20.05Penguinqualify=yes is the same as qualify=4000 if I remember right.
05:21.26atanDoes this apply to TCP clients?
05:22.02atanand is there any reason you wouldn't put this in your general section? (need to be set on each peer?)
05:24.01PenguinIf you want it to apply to all peers, I guess you could put it in general.
05:24.33PenguinYou might have some peers that you don't want it enabled, and then you would have to set qualify=no for those.
05:24.56PenguinI would probably suggest applying it to a template for natted phones.
05:25.22atanAwesome. Thanks! Learning as I go :)
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05:26.23atanI have one issue I'm trying to sort out that might not be related to this, hard to say. The phone will ring but the line will "hang open" if the line is not answered by the phone.
05:26.47atanSo if it ends up in voicemail or whatever, it'll still be showing on the phone despite the dialplan having moved on
05:27.06atanIs that a setting in my Dial() perhaps, or some other network setting I might look at?
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05:43.34ChannelZWhat do you mean it'll still be showing on the phone -- ringing?
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05:50.53atanHmm, no, not really. Let me take a picture quickly because it makes no sense otherwise
05:53.10atanhttp://imgur.com/qTHMGFs on this phone the bottom one there is ringing still, the top 2 are "missed" calls that went to voicemail
05:53.59atanWhen it's in this state you can't edit any config settings on the device. Seems to treat it as an active call in some way. Same sort of thing if you have an active call and enter the prefs, they're just grey and you can't touch them
05:54.06atanOnly way to resolve this right now is to reboot the device
05:55.45atanThere was once, and only once, where it somehow ended up stuck ringing. Now they just kind of hang there like that. I was thinking about trying to upgrade the firmware but I don't know how Cisco handles that. Seems I need some login, which I assume is connected to some costly contract of some kind (I did hear something about $13 somewhere, but didn't find it)
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05:59.05ChannelZDunno if it's missing packets or something..
05:59.39ChannelZyou'd have to capture some sip debugs and see if asterisk is maybe retransmitting messages, or the device is replying back confused perhaps
06:00.04atanI've only found one other report of something like it where the user said this same thing happened "if they called themself from the phone"
06:01.30atanOne moment I'll do it again but grab a debug. Last time I looked it sent a request cancelled, and then ack
06:02.43atanPerhaps unrelated but I notice "copy_header: No field 'Call-ID' present to copy" showing up in the console
06:07.10atanhttp://pastebin.com/vDg0Pphe is what happens near the end, if that is of any help
06:08.37ChannelZwell you need to capture what is possibly causing it to get stuck the first time
06:08.44smirkerHi again. Quick question that I haven't been able to find in the documentation. My server has multiple IP addresses in the same subnet, e.g. 1.1.1.1, 1.1.1.20, and 1.1.1.150.  I want to set up an entry in sip.conf that tells asterisk to send traffic from one of those specific IP addresses. Is this possible?
06:09.33ChannelZWell you can bind it only to one
06:09.58ChannelZudpbindaddr
06:10.34smirkerI have multiple CIDR ranges and providers sending me traffic ;[
06:11.50PenguinYou can bind to only one single address or 0.0.0.0 which means all addresses on all interfaces.
06:12.49smirkerI have it bound to 0.0.0.0. Then for one of my many entries in sip.conf, I need to specify which address to send from, and it just so happens I need to have multiple addresses in that range.
06:12.58smirkerIf not possible, I'll set up some more rules on the other side.
06:14.12ChannelZSounds like a routing job. I'm assuming that's multihomed on a single interface and not discreet
06:14.25ChannelZeth0:1 and the like
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06:15.05smirkerIf only.
06:15.54smirkerJust trying to make it neat for our pacemaker cluster / IP addressess / passing traffic between the clusters.  I've just added entries for all possible IP addresses on the other end and it works fine now :)
06:16.00smirkerDeadline drawing near ;p
06:17.53ChannelZI can't remember if pjsip is supposed to finally solve that in *12
06:23.22ChannelZI think so..
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09:02.53flingChannelZ: is not it possible to log to postgresql?
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09:04.06flingfound it > https://wiki.asterisk.org/wiki/display/AST/PostgreSQL+CDR+Backend
09:04.35bulkoroknice one!
09:04.37flingHow to log only what I want
09:04.45flingbulkorok:  :>
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09:05.05bulkorokjust create the coloums you need in the table!?
09:05.28flingbulkorok: no, I mean not all the calls
09:05.29bulkorokmaybe you can configure it in the cdr_pgsql.conf
09:05.40flingright, doing
09:06.04flingI want to log few contexts to this base, not all
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12:02.03SepulturaHallo
12:03.59SepulturaAsterisk is a software which I can use to make IP Calls?
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12:08.29bacobartasterisk is a telephony server (pbx). it is used to make voip calls yes (but you will still need a voip phone to connect to asterisk)
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12:09.15Sepulturabacobart: does Asterisk support Encryption?
12:10.16bacobartyes
12:10.36bacobartfyi, so far your questions could've been answered by yourself with a 2 second google query :P
12:11.16Sepulturabacobart: I found Asterisk by Google and Wikipedia but I wanted to be sure the Informations are correct
12:14.01bacobartSepultura: ok. http://cdn.oreilly.com/books/9780596510480.pdf there's a free book with loads of information about asterisk. it'll get you started if you want to know about what asterisk can do.
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12:18.47Sepulturabacobart: I am thinking about making an own phonecall Server.
12:21.50bacobartwell. i guess you could. butthat's a huge task.
12:22.00bacobartoh
12:22.07bacobartwait you mean setting up your own server
12:22.14bacobartor developing your own software?:P
12:24.57Sepulturaset up my own server
12:25.08Sepulturafor encrypted telephony
12:26.20bacobartah ok. well asterisk sounds like a good fit then ;p
12:26.58SepulturaEkiga can connect to Asterisk
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12:39.24SepulturaI hope it supports Video conferencing?
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13:44.16gianyIs there a way I can change the T38maxBitRate parameter?
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13:53.44TobSnyderI want to use cidname / displayname from caller and not from DB
13:57.39[TK]D-FenderTobSnyder: It's your call, do whatever you want with it...
13:57.51TobSnyderactually the problem is I am using freepbx
13:58.04TobSnyderwhere I am creating all those extensions, with some fancy names
13:58.11[TK]D-FenderTobSnyder: And you're asking in the wrong channel
13:58.48TobSnydermh
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14:28.47Jinxed-Anyone have any recommendations for a non-bloated free softphone for windows to do some testing with a local asterisk box
14:29.05Jinxed-I just installed linphone wasn't too impressed
14:29.21Jinxed-and 3cx has changed their interface and won't allow me to "add accounts"
14:29.29Jinxed-I think they are requiring you to use their phone system now
14:30.15navaismoblink
14:30.18navaismozoiper
14:30.24Penguinzoiper
14:31.06Jinxed-navaismo, http://icanblink.com/?
14:31.30pabelangerzoiper
14:31.55Jinxed-zoiper it is
14:32.12navaismoJinxed-, yes that blink
14:34.13Jinxed-well that was easy
14:34.15pabelangerblink is good too, I just had issue getting it to work on ubuntu 12.04
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14:35.36Jinxed-nice
14:35.38Jinxed-it worked well
14:35.41Jinxed-zoiper
14:35.57gianyIs there a way I can change the T38maxBitRate parameter, anyone knows??
14:39.54Jinxed-I thought all you had to do to enable a meetme conference was have conf => thenumber
14:40.03Jinxed-do you need to add something in extensions.conf as well
14:40.27[TK]D-Fendergiany: "core show function FAXOPT" <---
14:40.39Penguinjinxed-: All you have to do to enable a meetme conference is run the MeetMe() application, usually within an extension.
14:41.20Penguinjinxed-: How would adding a conf to meetme.conf allow a phone to call it?
14:41.49Jinxed-no idea, seemed odd to me
14:42.10Jinxed-never have tried it before, and the google response I found didn't mention anything
14:42.14PenguinWhen you pick up the phone and dial a number, where does the call go to start processing it?
14:42.21Jinxed-extensions.conf
14:42.28PenguinSo put your conf there.
14:42.53Jinxed-exten => 5555,1,MeetMe(5555)
14:43.18PenguinThe MeetMe() application can dynamically create conferences or use confs defined in meetme.conf.
14:43.25Penguincore show application MeetMe
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14:48.37Jinxed-Your application(s) is (are) not registered
14:48.43Jinxed-Command 'core show application MeetMe' failed
14:50.12[TK]D-FenderHelps to have it compiled...
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14:50.32[TK]D-FenderGood odds you're missing the important prerequisites
14:50.41[TK]D-FenderDAHDI <-
14:51.34WIMPyThe best workaround is to use ConfBridge.
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14:56.37Jinxed-[TK]D-Fender, Does it not come compiled in the asterisk with ubuntu 12.04 ? Seems odd they would have the meetme.conf file already there
14:56.59[TK]D-Fenderjinkdahdi would be a completely separate package
14:57.08[TK]D-FenderJinxed-: dahdi would be a completely separate package
14:58.54Jinxed-the closest thing I have to Dahdi is when I do a core show applciation d(tab complete) I see DAHDIScan
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15:00.00WIMPyDAHDI is/are kernel modules.
15:02.39[TK]D-FenderJinxed-: Another UBUNTU package....
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15:06.09PenguinJust install any dahdi packages that are available and see where that takes you.
15:08.13Jinxed-installing asterisk-dahdi had some errors hmm
15:08.41Jinxed-Is confBridge or MeetMe the prefered way of conferencing? What I eventually want to do is to have a number that joins two or more phones to a conference, and then a number which can destroy that conference
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15:12.41[TK]D-FenderJinxed-: What version of * are you running?
15:13.22[TK]D-FenderJinxed-: 1.8 ConfBridge was inferior functionality compared to MeetMe.  It improved massively in 10 and 11 is highly recommended
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15:14.20Jinxed-1.8.13
15:16.19[TK]D-FenderJinxed-: 1.8.24.0 is current in your branch so you're somewhat behind as it is
15:16.53[TK]D-FenderIt'll do for the basics, but there are a lot of administrative things that changed in future versions
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15:19.07Jinxed-[TK]D-Fender, How do you update beyond what ubuntu provides?
15:19.15Jinxed-ppa?
15:19.47PenguinConfBridge in 1.8 does provide very basic conferencing.  Any special things may need to be handled in dial plan or by AMI, though.
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15:20.57[TK]D-FenderJinxed-: Don't know what other repos are out there and good.
15:21.21Jinxed-All I need to do is add/remove users to a conference by dialing different numbers
15:21.42Jinxed-or remove all users from all conferences by dialing a number
15:27.27PenguinYou should be able to accomplish that with ConfBridge() in 1.8.  While it is not very robust, it does do basic conferencing well.
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15:35.19giany[TK]D-Fender: core show function F
15:35.19gianyFIELDNUM         FIELDQTY         FILE             FILE_COUNT_LINE  FILE_FORMAT      FILEDELETE       FILTER           FRAME_TRACE
15:35.27gianyseems i don't have that.
15:35.48[TK]D-Fendergiany: And what verions are you running?
15:35.52[TK]D-Fenderversion*
15:36.14giany1.8.9.2
15:37.05[TK]D-Fendergiany: Clearly missing prerequisites for it
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16:05.53oquidavehello am getting this error while trying to start asterisk asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory
16:06.02oquidavewhat could be the problem? thanks
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16:09.05rrittgarnim guessing that file doesn't exist or has wrong permissions
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16:14.23oquidaverrittgarn: doest that mean i've to recompile asterisk
16:14.38rrittgarnmore likely you didn't compile it correctly in the first place
16:14.50rrittgarnmake sure you don't get any errors when you build it.
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16:15.01rrittgarnthe book has a walk through on how to build and correct errors if you need
16:15.03rrittgarn~book
16:15.04infobotAsterisk: 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
16:15.24Max_EIs there a way to debug AMI in the asterisk cli? manager set debug on ==> nada
16:16.35oquidaverrittgarn: i compiled it without errors
16:17.01oquidaverrittgarn: actually am upgrading from 1.8 to 11... that previous installation was flaweless
16:17.15rrittgarnah, did you move your modules directory then rebuild?
16:17.25rrittgarnyou could have modules that don't need to be there in there...
16:18.48oquidaverrittgarn: no i didn't
16:19.56rrittgarni would suggest that and then rebuild
16:20.38kaldemaroquidave: did you install it?
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16:22.29Max_Emmm i guess ill stay with ngrep
16:22.57oquidavekaldemar: asterisk said it installed successfuly after i compiled it. though am compiling it again this time movign the module dir first
16:25.40oquidavestill having the same problem
16:27.58oquidaveive looked at the modules directory and "libasteriskssl.so.1:" doesn't exist
16:28.29Penguinasterisk: error while loading shared
16:28.29Penguinlibraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory
16:28.35PenguinYep, that's what it said alright.
16:29.46oquidavebtw, am installing the latest astrisk asterisk-11.6.0 ...is it stable?
16:30.17oquidavebecause 1.8 worked without trouble
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16:36.50XandriXPenguin: quick question auto attendant and ivr = 2 completely different things right ?
16:37.41Penguinxandrix: Yes, they are different, but many people think everything is IVR.
16:38.31XandriXi was checking to make sure i remember talking to you about something i wanted to do and you had mentioned that an auto attendant would do the job for my application and that there was a big difference between it and ivr and that ivr for my situation would have been overkill hehe
16:40.09XandriXand i also wish to apolagize for the calling from an extension thing you had explained it to me and i completely forgot ....
16:41.50PenguinI don't know if I said overkill.  IVR just is a totally different idea from a digital receptionist.
16:43.38XandriXnoted :) currently reading chapter 15 of the 3rd revision of the asterisk book
16:43.47XandriXill takle the hole caller id thing some other time
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16:44.24XandriX*whole
16:44.49PenguinIVRs collect information from the caller for processing.  The digital receptionist (auto attendant) executes new extensions.
16:45.00oquidavewhen i recompile 1.8, asterisk runs perfectly well again
16:45.09*** join/#asterisk DynamicFail (~b0ot@147.177.62.50)
16:45.13oquidavewhat's the right procedure of upgraading!!
16:45.26DynamicFailHow out of date is asterisk 1.8.5 in terms of either MeetMe or ConfBridge
16:46.22WIMPyMeetMe is out of date and 1.8 as well. 1.8 has the old and rather restricted ConfBridge. the new ConfBridge came with 10.
16:47.02DynamicFailI heard there was a bug with ConfBridge in 1.8 in which if two people left at the same time
16:47.04DynamicFailit would crash
16:48.10DynamicFailshould conf bridge show up as a registeredf application
16:48.16DynamicFailin the core show applications
16:48.18DynamicFailcommand
16:48.51WIMPysure
16:50.00XandriXhmmm i should maybe consider building a package of asterisk for pfsense that is version 10.x or 11.x instead of the default 1.8 one
16:50.14[TK]D-FenderDynamicFail: "heard" doesn't give us anything specific to go on, is likely no longer relevant if it was ever accurate
16:51.10DynamicFailgoogle seems to indicate that even ubuntu 13.10 (latest one) still comes with asterisk 1.8
16:51.14DynamicFailwith the apt-get
16:51.44XandriXDynamicFail: in many situations i prefer building stuff myself than using apt-get / yum and other package managers
16:51.50[TK]D-FenderDynamicFail: What Ubuntu packages is not our concern unfortunately
16:52.04DynamicFailjust surprised
16:52.42DynamicFailXandriX, I like that package managers can keep the software uptodate and handles dependancies
16:52.50XandriXisint surprised of anything canonical does
16:52.59DynamicFailI find that when I build things myself shortly after it no longer works
16:53.03[TK]D-FenderDynamicFail: 1.8 is an LTS branch so it isn't a "bad" policy per-se
16:53.07DynamicFailbecause of upgrades in dependancies
16:53.35DynamicFailso is Asterisk 11 though isn't?
16:53.45XandriXDynamicFail: nothing stops you from having a local repository and build the source code as a deb package and have a tiny script that checks to see if a newer version of your desired application version is out
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16:54.16DynamicFailother than not knowing what you just said
16:54.18DynamicFailmeans
16:54.35XandriXregardless i get your point of view DynamicFail and my sentence made no sense so no biggy
16:55.02DynamicFailAny linux versions play nicely with asterisk as far as staying up to date?
16:55.14kaldemaroquidave: make uninstall in the old source dir. then configure, make and make install in the new one.
16:55.32[TK]D-FenderDynamicFail: Yes, 11 is LTS... so no reason they shouldn't move on.. but nothing forcing them to either...
16:55.52[TK]D-Fender[11:54]DynamicFailAny linux versions play nicely with asterisk as far as staying up to date? <- any when configured properly
16:55.58XandriXcompile asterisk 10 build a package with it cuz ubuntu has the tools to make a .deb file and just have a script running in crontab or something that checks to see if said software has a newer source version available and make it warn you so you can do the same process with the newer source version
16:56.19XandriXOR exactly what kaldemar said
16:56.23XandriXwich is way simpler
16:56.45[TK]D-FenderAsterisk 10 = DEAD
16:56.58fileAsterisk 10, we hardly knew you
16:57.10XandriX[TK]D-Fender: so either use 11 or 1.8 branches up until 1.8 is no longer supported ?
16:57.14[TK]D-Fender#plausibledeniability
16:57.27[TK]D-FenderXandriX: Correct
16:57.51DynamicFailhow much longer is 1.8 supported for?
16:58.08*** join/#asterisk dumby (~dumby@204.246.140.162)
16:58.25kaldemarDynamicFail: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
16:59.17DynamicFailok
16:59.21DynamicFailso still a few more years
17:00.03DynamicFailSo I'm a little confused, even though 1.8 is LTS it doesn't include a lot of the improvements to ConfBridge?
17:00.34WIMPyIt may be LTS, but it's still out of date.
17:00.39oquidavekaldemar: am doing that
17:01.09[TK]D-FenderDynamicFail: branches don't get improvements.. they get FIXES
17:01.21[TK]D-FenderDynamicFail: New branches get new features
17:01.58DynamicFail:(
17:03.30*** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net)
17:03.36polysicshello
17:03.42polysicsdumb question time
17:03.53polysicshow do I originate a call to another box I do not have a peer on?
17:04.14polysicsie. how do I originate to an arbitrary SIP URL
17:04.39WIMPySIP/user@host
17:04.48kaldemarpolysics: see dial strings in the sample sip.conf
17:04.49PenguinSIP/extension@host.com
17:06.27XandriXPenguin: i cleared up my extensions.conf and took the time to properly diferentiate extensions from peers as you had suggested
17:06.34*** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz)
17:06.44XandriXalso did alot of cleaning and using same => n,
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17:10.27polysicshmm, I have been trying with the simple SIP/user@host but it looks like I am missing something
17:10.41PenguinShow me the command you've been running.
17:10.55[TK]D-FenderAnd the debug
17:11.19PenguinThe command syntax could be wrong.  We'll start there.
17:13.28[TK]D-Fenderpolysics: a peer ON... or a peer TO?
17:13.55[TK]D-Fenderpolysics: How is the other side set up?  Are they configured to accept your "random call"?
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17:14.49oquidavekaldemar: i uninstall 1.8 and installed 11, but am still having the same problem!!!!
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17:17.20kaldemaroquidave: how are you starting it?
17:18.00oquidavekaldemar: asterisk -vvvc
17:18.11kaldemaroquidave: asterisk -V
17:18.32oquidavesame thing
17:18.33polysics[TK]D-Fender: got it. I needed an extension
17:19.12[TK]D-Fenderyou are still completely vague about what side you're talking about or how that terminology is even applied
17:19.28*** join/#asterisk gerhard7 (~gerhard7@77.172.47.159)
17:20.30kaldemaroquidave: what exact commands did you use to configure, compile and install?
17:20.56oquidavesudo ./configure,  sudo make,  sudo make install
17:20.58polysicssimply using channel originate SIP/123@asterisk.local-dev.mojolingo.com extension 123@default
17:21.04polysicsI can also use an application
17:21.32PenguinLooks like you've got it covered.
17:21.41kaldemaroquidave: ls -la /usr/lib/libasteriskssl.so.1
17:21.51polysicsI had the syntax wrong, it looked like extension was the place where to put my target extension, instead it is the actual word "extension"
17:21.54polysicsthat was all
17:22.05oquidavekaldemar: -rwxr-xr-x 1 root root 677575 Dec 12 20:13 /usr/lib/libasteriskssl.so.1
17:23.52kaldemaroquidave: in the 11.6.0 source dir: ldd main/asterisk
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17:26.18*** join/#asterisk vader-- (~me@204.183.88.101)
17:26.20vader--hello
17:26.37oquidavekaldemar: i can see " libasteriskssl.so.1 => not found"
17:26.53kaldemaroquidave: sure. what else?
17:27.08oquidavekaldemar: the others are available libxml2.so.2 => /usr/lib64/libxml2.so.2 (0x00007fa41b259000)
17:27.44kaldemaroquidave: what else?
17:27.56oquidavekaldemar: http://pastebin.com/yHVpqxxQ
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17:28.05vader--any of you guys happen to have experience with Cisco ATA 186/188 devices? I have one that i am setting up but I can't get a dial tone. I can push the button and the IVR comes on. I can call the device. It registers correctly with SIP. Both POTS lines are dead when it comes to a dial tone. On the web interface when i pick the phone up it shows predial as the status, when i press a number the
17:28.06vader--web interface shows dialing.
17:28.22vader--when i call the number assigned to the line, the phone rings.
17:28.25ruben231hi guys any insight adn idea on how fax on asterisk works..
17:28.35vader--just no dialtone when i pick the phone up
17:28.41*** join/#asterisk felipealmeida (~user@177.41.225.200)
17:30.07oquidavekaldemar: i copied the cp /usr/lib/libasteriskssl.so.1 /usr/lib64/ and it worked
17:30.16kaldemaroquidave: your library paths are twisted.
17:31.11kaldemarasterisk installed in /usr/lib but shared libraries are not searched under it.
17:31.13oquidavekaldemar: my machine/os is 64 bit so iit has /usr/lib and /usr/lib64...and i think asterisk assumes it's a 32 bit
17:35.37oquidavekaldemar: meanwhile thanks for the help
17:37.07kaldemaroquidave: what os are you on?
17:45.37kaldemaryou should probably configure asterisk with "./configure --libdir=/usr/lib64" or run ldconfig after installing. copying the library by hand is not a good idea.
17:46.32*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)
17:51.28PenguinI would prefer to see it configured with the right path prior to make.
17:55.42*** join/#asterisk Villalobos (villalobos@95.85.24.26)
17:56.24VillalobosAnyone even had trouble with SRV records? Not accepting calls from proxy B while setting up calls to proxy A?
17:56.53VillalobosReturning SIP 403 to the provider while it shouldn't for incoming calls.
17:57.15kaldemari'd prefer to see the distro use /usr/lib for libraries, be it 64-bit or not.
17:57.55PenguinI'm not sure if that's feasible in a multilib setup.  It seems like they always separate them.
17:59.39kaldemarnot necessarily for all libraries. but a 64-bit system can use /usr/lib by default and have other dirs for 32-bit libraries for example.
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18:21.32PenguinAh, I think I see what you mean.  Instead of /usr/lib/ for 32-bit and /usr/lib64/ for 64-bit, you could have /usr/lib/ for 64-bit and /usr/lib32/ for the rare 32-bit stuff you may need?
18:25.47*** join/#asterisk Jinxed- (~b0ot@147.177.62.50)
18:26.24Jinxed-kaldemar, Do you have any good links on how one might uninstall 1.8 and install asterisk 11?
18:28.03Penguinyum remove asterisk not good enough?
18:28.32Jinxed-I can apt-get remove asterisk
18:28.42Jinxed-I was planning on follwoing this for the install: http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/
18:30.52pigpenI posted this yesterday, but this is an issue that few will be knowledgable about.
18:31.01pigpenso, anybody using realtime?  I an extension in realtime (psql) that pages a list of about 43 sip phones.  Seems it limits target (list of extensions) to about 120 char.
18:31.29pigpenanybody know how to lengthen the appdata "window" of realtime?
18:31.53pigpenso, regards to my realtime issue, i found pbx/pbx_realtime.c, line 60:  #define EXT_DATA_SIZE 256
18:32.10pigpenI am unsure if this is the right place.
18:32.34Max_Etry in the dev channel
18:32.44pigpenasterisk-dev?
18:33.22Max_Eyes
18:33.34pigpenk.  yeah, just found it.  good idea.
18:33.36pigpentks.
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19:40.06apb1963I think I may be bloated.  http://www.bufferbloat.net/projects/bloat/wiki/Introduction
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20:15.07ChannelZapb1963: on cable, more than likely
20:15.53ChannelZhttp://netalyzr.icsi.berkeley.edu/  (it's java though)
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20:46.29[SySteM]Please, i got a problem, i search a solution since 3 days but.. nothing
20:46.43[SySteM]im in France and im brother is in qatar
20:47.09[SySteM]He got 2 clients for sip (a sipurabox and a iphone client)
20:47.17[SySteM]this 2 clients worked when he was in france
20:47.29[SySteM]but in qatar, he got a 1Mo connexion internet (skype and webcam work)
20:47.35polysicswhere is the server?
20:48.06[SySteM]but with sip, he can talk 1 or 2 secs and after, i cant hear corretly him, i get lots lots lots of cut, and him can ear me correctly
20:48.08[SySteM]in france
20:49.30vader--any of you guys happen to have experience with Cisco ATA 186/188 devices? I have one that i am setting up but I can't get a dial tone. I can push the button and the IVR comes on. I can call the device. It registers correctly with SIP. Both POTS lines are dead when it comes to a dial tone. On the web interface when i pick the phone up it shows predial as the status, when i press a number the
20:49.30vader--web interface shows dialing.  when i call the number assigned to the line, the phone rings. just no dialtone when i pick the phone up
20:49.34[SySteM]i could understand than bad connexion could probably make that but... facetime, skype, lots of things work correctly on audio with webcam..
20:50.34[SySteM]and why only me cant understand him correctly and him can have good audio
20:59.15[TK]D-Fender[SySteM]: Qatar ISP's are known to actively disrupt SIP as a way to control access to the PSTN
20:59.59[SySteM]yes but worked when my brother was at qatar hostel
21:00.05[SySteM]and it was the same isp
21:00.21[TK]D-Fenderhttps://www.google.ca/#q=qatar+blocking+sip
21:00.39[TK]D-Fenderhttps://forums.digitalpoint.com/threads/how-to-bypass-voip-blockage-from-isp-in-blocking-countries.2006125/
21:00.55[TK]D-Fender[SySteM]: sometimes maybe you get lucky
21:01.17[TK]D-Fender[SySteM]: But the country & its providers seem to be mostly hostile
21:01.48[SySteM]but skype & facetime work correctly :/
21:02.13[TK]D-Fender[those are different protocols.
21:02.13Max_Enon sip
21:02.20[TK]D-Fenderand factetim isn't used for PSTN
21:03.29Max_Enow is listenning--->http://www.youtube.com/watch?v=5NV6Rdv1a3I
21:03.41[SySteM]u thing i need to change udp port ?
21:03.59[SySteM]sorry i dont see previus link thx
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21:20.28KattyHELLO CUPCAKES
21:20.36bsdiceyes
21:21.00Kattyno.
21:21.16bsdicemay I call you
21:21.22bsdiceKat?
21:24.08Kattysure! i used to by Kat on slashnet.org.
21:24.14Kattyspeaking of such
21:24.18KattyNugget: telnet
21:24.18Nuggettelnet is eeeeeeevil!
21:24.24KattyNugget: do you want a christmas card this year dear?
21:24.32Nuggethuggles Katty
21:24.38Kattyhugs on Nugget
21:25.09bsdicepleased to meet you Kat, my name is Dr Sbaitoso, I am here to help you
21:25.23bsdiceSbaitso even
21:25.27NuggetI am a talking parrot, please say something!
21:25.34Kattybergawk.
21:25.46bsdiceNugget you must be OLD :)
21:26.08Nuggetindeed
21:26.22Kattyshould i take that as a yes or a no? :P
21:26.30Kattyno is ok!
21:27.03NuggetI survive on a fixed diet of hugs and snark.  Cards are unnecessary :)
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21:27.28*** join/#asterisk Jinxed- (~b0ot@147.177.62.50)
21:27.52Jinxed-Anyone ever use the Zultys 2x2 phone with asterisk? I can't even get the phone to attempt to register with *box
21:28.43KattyNugget: righto, sir.
21:28.50Kattyapplies christmas hugging to Nugget liberally
21:28.56Nuggethooray
21:29.22Kattywhich means the only other person on my list is jaytee
21:29.46Kattywhere does jaytee hang out these days?
21:30.16vader--any of you guys happen to have experience with Cisco ATA 186/188 devices? I have one that i am setting up but I can't get a dial tone. I can push the button and the IVR comes on. I can call the device. It registers correctly with SIP. Both POTS lines are dead when it comes to a dial tone. On the web interface when i pick the phone up it shows predial as the status, when i press a number the
21:30.17vader--web interface shows dialing.  when i call the number assigned to the line, the phone rings. just no dialtone when i pick the phone up...
21:30.34Max_Eis that a bot with a cron job inside? ^^
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21:40.51newtonrJinxed-, I have not. If you post a pastebin to an Asterisk log( http://tinyurl.com/3ldrxnn) with "sip set debug on" enabled, then we can look what is happening.
21:41.48newtonrJinxed-, though if it is truly not attempting at all to register, that is, if there is no SIP traffic being sent to Asterisk, the logs would be useless and the solution would be fixes on the phone or network.
21:42.13newtonrAt least looking at the logs would tell you what Asterisk sees.
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21:53.42*** join/#asterisk wolrah (~wolrah@24.239.210.140)
21:54.17bsdicevader-- Look here http://spakonfig.de/ -> Regional -> Dial tone
21:56.36*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2)
21:56.41Kattyfender bender.
21:57.14*** join/#asterisk vlad_starkov (~vlad_star@109.188.124.67)
21:58.14bsdicebored, Katty? :)
21:58.25Kattyno i'm always like this.
21:59.21bsdiceme too, but then I got a new life and quit chat for good for many years
22:01.14bsdiceand now I am back... checking out voip channels
22:06.21*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
22:06.21*** mode/#asterisk [+o pabelanger] by ChanServ
22:07.53*** join/#asterisk smirker (~x@CPE-123-211-202-192.lnse4.woo.bigpond.net.au)
22:08.03*** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net)
22:13.17drmessanobsdice, Asterisk is cooler and more stable, some of us idiots are now older idiots, and Skype still sucks and is evil
22:13.25apb1963ChannelZ: Thank you, I actually tried that the other day and it agreed I may be bloated.  However.... I'm not sure what to do about it.    And yes... it's a cable modem.... however my ISP is actually fairly responsive to me once I yank the right strings so if there's something that can be done....?
22:13.37drmessanochecks the Digium blog to make sure we haven't announced we like Skype again
22:14.29*** join/#asterisk vlad_starkov (~vlad_star@91.206.59.133)
22:16.52hardwirehah
22:22.06Jinxed-what would be the format to have a number dial an ip address?
22:22.08Jinxed-via sip
22:22.21drmessanoSIP/user@ip
22:22.28Jinxed-drmessano, no username
22:22.30Jinxed-just the IP
22:22.35drmessanoO.o
22:22.51Jinxed-the device is only able to do IP to IP calling (SIP)
22:22.58Jinxed-can't register, has no idea of usernames
22:23.13*** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.67)
22:23.22drmessanoSIP/@IP?
22:24.52*** join/#asterisk vlad_st__ (~vlad_star@109.188.127.92)
22:27.13drmessanoJinxed-, have you tried SIP/IP?
22:27.19Jinxed-I tried both
22:27.23drmessanoI see IAX2 supports IAX2/host
22:27.38Jinxed-My phone rings, but the end device doesn't ring
22:27.53Jinxed-not sure
22:28.01*** join/#asterisk Kobaz (~kobaz@its.kobaz.net)
22:28.14drmessanoDid you try SIP/nonya@IP or something?
22:28.28Jinxed-not yet
22:28.37Jinxed-i tried SIP/IP and SIP/@IP
22:28.38drmessanoWondering if the username satisfies Asterisk and the endpoint wont care
22:28.49drmessanoI would try a dummy username
22:28.51[TK]D-Fendersip/ip should do it for *
22:29.43Jinxed-I think all 3 methods are working from my original phone and asterisk
22:29.50Jinxed-my guess its the endpoint that isn't working
22:30.02Jinxed-very unique device
22:30.06drmessanoWhat is it?
22:30.13Jinxed-a type of radio
22:30.25drmessanoACU2000 or something?
22:30.29Jinxed-nah
22:30.44drmessanoTwo-Way radio?
22:31.16Jinxed-you could say
22:31.36*** join/#asterisk jhlavacek (~jirka@ALyon-253-1-132-167.w90-14.abo.wanadoo.fr)
22:31.45drmessanoIs it a radio or a console?
22:32.46Jinxed-hang on be back later
22:34.26[TK]D-Fenderhow wonderfully vague
22:40.31*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
22:40.32*** mode/#asterisk [+o pabelanger] by ChanServ
22:41.30bsdicedrmessano I am using * with TLS and SRTP or over a VPN tunnel by mobile phone. Well, I like it.
22:41.56bsdicestarted with 0 configs and did everything by hand
22:42.49bsdiceworks well with snom 821 voip phone and csipsimple
22:43.36bsdicewith tweaks even over UMTS using speex codec at 11 kbps
22:44.34bsdiceSIP adapters and SRTP are mostly worthless though
22:44.47bsdiceMicroSIP ok too (for windows)
22:45.10Jinxed-hahahahaha
22:45.15Jinxed-I was looking at the wrong device
22:48.31*** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-npgaeckzzqvwjcvw)
22:55.10*** join/#asterisk areski (~areski@80.174.128.76.dyn.user.ono.com)
23:00.35*** join/#asterisk StanAccy (~StanAccy@unaffiliated/stanaccy)
23:00.52StanAccyhow do I restart asterisk (or reload the configs) from the command line
23:01.24StanAccy'restart gracefully' isnt recognized
23:02.17Max_Esipr reloa, dialplan reload, manager reload etc, but if there is no 'name' reload go for the module reload <modulename>.so
23:02.53StanAccythanks
23:05.33Max_Es/sipr reloa/sip realod/
23:05.38Max_Eargh
23:08.31*** join/#asterisk ruben231 (~OpenDial@112.198.90.252)
23:08.47*** part/#asterisk ruben231 (~OpenDial@112.198.90.252)
23:10.04*** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001)
23:14.00bsdiceStanAccy /sbin/coldreboot :)
23:14.13bsdiceservice asterisk restart
23:14.31bsdicefor upstart systems (Ubuntu 12 etc)
23:21.21phixbsdice: /etc/init.d/asterisk restart still works in 12 but gives a warning message
23:22.19bsdicewell, using my own upstart init-script
23:24.02*** join/#asterisk dxd828 (~dxd828@host217-43-217-142.range217-43.btcentralplus.com)
23:24.24Max_Ewaiting for ubuntu/debian users/fanboys saying how awesome is the: service xxx start|stop|restart|status thanks lord centos/redhat/fedora switch that for systemclt
23:27.55bsdicewhat
23:28.28*** join/#asterisk theron (~theron@66.220.144.81)
23:30.12bsdiceI don't care, really. bsd-ice means I could also do with /usr/local/etc/rc.d/asterisk restart
23:30.21bsdiceor how about inittab ;-)
23:30.48bsdicenet stop asterisk / net start asterisk hehe
23:38.36Jinxed-Anyway to route a call placed to the IP address of the call manager
23:38.51Jinxed-so if my * box is 10.1.1.1
23:39.12Jinxed-route a call placed to 10.1.1.1 to a specified destination?
23:40.47*** join/#asterisk timahvo1 (~rogue@197.237.174.93)
23:46.31*** join/#asterisk felipealmeida (~user@177.41.225.200)

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