IRC log for #asterisk on 20100520

00:02.16norrecidk if it does, i dont see it in my cdrs
00:03.59Naikrovekmaybe that's just my wonky-as-all-getup trixbox setup
00:04.37norreclol
00:04.55norrecwell would u mind pbing the cdr conf files?
00:05.09norrecso i can see if urs is different than mine
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00:33.08joobieNaikrovek, sorry man - got caught up.. at work
00:33.43joobieNaikrovek, it's important because it conflicts with another firewall rule (got some fairly complex routing in that i dont want to complicate further to accomodate for this condition if possible)
00:34.16Naikroveksource ports aren't really a part of any firewall rules i've seen.  destination ports are
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00:34.40Naikrovekthe OS usually picks a port at random or via some selection mechanism I don't know about
00:34.56Naikrovekpoint being you can never count on a part going unused or consistently being used, for any service
00:35.21Naikrovekso, people don't base firewall rules on the source port.  destination port, though, you can change
00:35.44Naikroveks/part/port/ (two lines up)
00:36.27Naikrovekdestination ports are static, though
00:36.32Naikrovekare you sure you're not thinking of source port
00:36.34Naikrovekummmm
00:36.39NaikrovekDESTINATION port
00:37.46Naikrovekpolycom may have it set up so that source port and destination port are the same.  if you change the port asterisk listens on to 5061 or something, maybe the polycom will change its source port to that as well
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00:40.38joobieNaikrovek, they are apart of my firewall rules
00:40.43joobiewe have some complex stuff going on..
00:40.48Naikroveki guess
00:40.51joobieI dont know if it's completely random
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00:40.59joobiethe src port that is
00:41.09joobiesome phones appear to be completely random
00:41.12Naikroveki used to do firewalls for a living (well, cisco access lists for 300 customers) and never once did i filter based on source port
00:41.14Naikrovekbut whatever
00:41.15joobiesome seem to want to use 5060 as their src
00:41.33Naikrovekif you change the destination port number via ... you know what
00:41.37Naikrovekhow do you configure the phones
00:41.41joobiesrc port filtering is common
00:41.43joobiesuch as FTP
00:41.56joobiebut less common compared to dport filtering
00:42.02Naikrovekokay
00:42.08Naikrovekhow do you configure the phones
00:42.10joobiei configure this via https://
00:42.15Naikrovekweb interface
00:42.16joobiebut on a boot server
00:42.24joobienaa just use https as the transport for the boot server
00:42.34joobieused to use web interface but cant do what i need via that anymore
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00:42.48Naikrovekokay so you know all about sip.cfg and stuff
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00:43.02Naikrovekhttps://  ... do you generate the config files dynamically
00:43.08Naikroveki think that could be useful
00:43.19Naikrovekhave a webapp that generates the phone configs as they're requested
00:43.31Naikrovekanyway
00:43.41Naikroveksip.cfg may have a setting for source port
00:45.45Naikrovekthere's a fella at work i wanna prank.  every 3rd or 4th time his phone boots i want his custom ringtone to be "i'm a barbie girl" or something
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01:27.02BugKhaMis the /etc/modprobe.d/zaptel required to start zaptel?
01:27.25BugKhaMthought it's done through an init script
01:30.06joobieNaikrovek, funny :)
01:30.12joobieNaikrovek, you should do it ..
01:30.20joobieNaikrovek, any idea what the sip.cfg setting is?
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02:46.16norrecdoes anyone know how to modify cdr output?
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02:48.16p3nguin/etc/asterisk/cdr_custom.conf
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02:56.24rocksfrowam i missing something I need to allow through my firewall for external sip connections?
02:56.34rocksfrowi have the appropriate ports unblocked, afaik..
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02:56.54rocksfrowmy softphone is registering fine, and i can even dial out..pick up my cell phone, it shows i picked up..but no audio on either side
02:56.57rocksfrowand the call eventually drops
02:57.01norrecp3nguin: do you know where i can find some documentation on writing that file, cause i know nothing about cdrs...
02:57.06rocksfrowany clue?
02:57.27rocksfrowor where I could start debugging? everything looks normal in asterisk debug
02:57.37norrecwhere is ur asterisk server
02:57.56rocksfrowmy office
02:58.00rocksfrowexternal
02:58.03TJNII~sipnat
02:58.04infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
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02:58.24norrecis ur asterisk box natted?
02:58.38rocksfrowumm...
02:58.45rocksfrowyeah.
02:59.04norrecok so its asterisk - firewall - pub internet - firewall - client
02:59.05rocksfrowi have the ports forwarded to the internal IP
02:59.13rocksfrowyerp
02:59.30rocksfrowand i have the appropriate ports setup, well..atleast I can make sip registration
02:59.33rocksfrowand make calls
02:59.40rocksfrowjust no audio
02:59.41norrecok, well you need to make sure asterisk know's its natted and that the client is natted
03:00.24rocksfrowokay, i guess i should give this a read through
03:00.47rocksfrowi don't want to mess up any of the internal phones though
03:00.48BugKhaMis the /etc/modprobe.d/zaptel required to start zaptel? or only an init script?
03:00.48norrecyeah look at #4
03:01.03rocksfrowah hah.
03:01.12rocksfrownorrec, let me confirm... this won't break inside connections through LAN
03:01.21rocksfrowprobably a stupid question.
03:01.30norreci cant promise anything, but in my experience no
03:01.50TJNIIYou can break internal phones if you really hose it up, but if you have two brain cells to rub together you should be fine.
03:01.52rocksfrowugh..the howto link is a 404, lol
03:02.05rocksfrowi have port forwarding setup and working..but dono about STUN
03:02.10rocksfrowguess that's where the issue is
03:02.37norreclol, hold on and i'll let u know which things u should prob set
03:03.02rocksfrowTJNII, lol, nice.
03:04.12drmessanoOuch.. If you're going to setup your tea maker and coffee maker on X10, make sure you remember which one is which
03:04.41norrecin global, set nat=yes, externip=external ip addr, and then nat=yes on each non-lan client
03:04.44drmessanoHTCPCP 418: I'm a teapot
03:04.48drmessano:(
03:05.50norrecalso set localnet = net.ip.addr/subnet.mask (ex 192.168.1.0/255.255.255.0)
03:06.01TJNIII was just about to call you on that.
03:06.28drmessanoand throw a canreinvite=no in there
03:07.12norreclol TJNII, i couldnt remember if netip/subnet was correct
03:07.15norrechad to look it up first lol
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03:08.22norrecdoes anyone have expeirence with modifying cdr outputs?
03:08.22TJNIInorrec: No worries.  I was quickly grepping my sip.conf before I said anything.
03:08.30rocksfroweek, this is a freepbx box
03:08.50TJNII~freepbx
03:08.51infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
03:08.55rocksfrowlol
03:09.17TJNIIrocksfrow: Come back and see us if you ditch the GUIs, otherwise it's their way or the highway.
03:09.43rocksfrowTJNII, well, i'm actually on the verge of doing that... but not sure
03:09.51TJNIIDo it.  It's fun!
03:09.59rocksfrowi have a second box that i'm about to put asterisknow on for redundancy... and was considering going straight install
03:10.36drmessanoThose settings for NAT awareness are the same
03:10.52rocksfrowdrmessano, within an extension some of them don't exist
03:10.53drmessanoJust need to place them in sip_general_custom.conf
03:11.08rocksfrowdrmessano, i'm afraid if I manually edit my changes will just be overwritten
03:11.11drmessanoDont worry about the extension
03:11.16drmessanoI just told you how to do it
03:11.19rocksfrowohh...
03:11.33rocksfrowsip_general_custom..interseting
03:11.50drmessanonat=yes, externip=, localnet=, and canreinvite=
03:11.50rocksfrowit's empty :-p
03:11.55drmessanoYes it is
03:12.04rocksfrowwell i don't see how these are static
03:12.06drmessanoJust put those global settings in that file
03:12.08rocksfrowi mean global**
03:12.11drmessanoThey are
03:12.17rocksfrowi have internal clients as well
03:12.28drmessanoWhich is why you set localnet
03:12.46rocksfrowbut, externip is not just on IP necessarily
03:12.49drmessanonat=yes is a behavioral setting based on the other parms
03:12.53rocksfrowthis could be 20 different IPs
03:12.54drmessanonat=yes is a behavioral setting based on the other parms
03:12.58rocksfrowone SIP from each
03:13.15rocksfrowone**
03:13.19drmessanoYour Asterisk box has 20 IPs?
03:13.29rocksfrowoh
03:13.33rocksfrowthat hosts public ip
03:13.35rocksfrowsorry.
03:14.28rocksfrowso, localnet=10.0.0.0/255.255.255.0 look right?
03:14.44drmessanoIf that's your local subnet and it's mask
03:15.11rocksfrowwell i just dont know if it should be 10.0.0.1/255.255.255.0
03:15.15rocksfrowor is it supposed to be .0
03:15.21drmessanoYes
03:15.23drmessano.0
03:15.40drmessanoBut you're asking if that looks right..
03:16.04rocksfrowafter looking at my extensions in freepbx, they all currently have nat=yes
03:16.05drmessanoSure, or it could be wrong if that's not the subnet your boxes are on and you're not using a /24
03:16.08rocksfrowi think that's a default value
03:16.25drmessanoThere is no default value
03:16.46rocksfrowwell, i mean on the 'add extension' form within freepbx, the prefilled 'default' values, lol
03:16.53rocksfrowjust saying, they're all set to nat=yes and have been working fine
03:16.56rocksfrowhow come?
03:17.17rocksfrowmaybe i can fix my fax problems by turning nat off on that extension? :-p
03:17.27drmessanoNo
03:17.37maxagazhi
03:17.50rocksfrowdrmessano, restart asterisk and give it a go?
03:17.55drmessanosure
03:19.21maxagazI still have the problem of a zero added between the extension set to call outside and the number I need to call, I can't see this zero in the logs not in the console, does someone have an idea ?
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03:19.52spenguin[work]TEST
03:19.54p3nguin~freepbx
03:19.55infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
03:20.22rocksfrowduuuuude
03:20.33rocksfrowdrmessano, norrec, etc...you da mannn
03:20.36rocksfrowdamn.
03:20.39rocksfrowthat easy lol
03:21.12rocksfrowso i had the port forwarding working fine, but the externip i think is what i was missing
03:21.18drmessanoYes, I am well aware.  I get lots of high 5's from the bro's and "oh my god, no man has ever.. " from the ladies.
03:21.36rocksfrowkinda scary i just have to wonder if everything is working in the office still
03:21.47rocksfrowdrmessano, so i should probably update the internal extensions to nat=no? or does it matter
03:21.52drmessanoNO
03:22.07rocksfrowwhoa
03:22.14drmessanoI told you twice
03:22.25TJNIIdrmessano: International man of mystery, intrigue, and telephone repair.
03:22.36drmessanoNAT=yes is an "ALLOW" based on other conditions, one of those being localnet
03:23.10rocksfrowokay.
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03:25.02norrecrocksfrow: + ive set lan clients as nat w.o using localnet and not had any issues
03:25.22rocksfrownorrec, yeah thx
03:25.34rocksfrowi think the externip was what i needed
03:25.38drmessanoYou would only change that to "never" if you needed some device to ALWAYS use the public IP
03:25.52rocksfrowgotcha.
03:26.01drmessanoas in, a LAN device
03:26.07drmessanoon the 10.0.0.0 subnet
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03:26.37drmessanoas in "I know you're local, but screw you, use the external IP for call setup because I hate you, bro"
03:27.01drmessanoIm sure there's a reason under some condition to do that, but I have never not set extensions to nat=yes
03:27.38rocksfrowdrmessano, i guess that explains the default setting in freepbx
03:27.56rocksfrowthanks very much for the explanation
03:28.03rocksfrowsorry to have you say it three times. :-/
03:28.05drmessanoCorrect, because it acts more like an allow and not forcing some condition, it's a safe default
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03:33.01Naikrovekjoobie: no idea what the setting is, but look for 5060 in sip.cfg and see if one look like it might be it
03:33.24joobietried that :/
03:33.55joobie.. unless im plain outright missing it
03:34.44joobievoIpProt.local.port
03:34.50joobieunless it's that? but not clear by the docs
03:34.58Naikrovekthat sounds like it
03:35.10joobieLocal port for sending and receiving SIP signaling packets.
03:35.11joobieIf set to 0 or Null, 5060 is used for the local port but it is not advertised in the SIP signaling.
03:35.11joobieIf set to some other value, that value is used for the local port and it is advertised in the SIP signaling.
03:35.21joobieergh - scuse the long paste.. thought it would be a one liner.
03:35.26joobiethat above is the description for that option tho
03:35.35Naikrovekwon't worry about it.  change it - i bet it changes the source port
03:36.00Naikrovekor maybe it changes where it listens for SIP connections from asterisk
03:36.03Naikrovekwon't know til you try
03:36.11Naikrovekah yes you will
03:36.16Naikrovekit says "Sending and receiving"
03:36.20Naikrovekthere's your answer
03:36.30Naikrovekthat is how you change the source port
03:38.42drmessanoWhy would set up firewall rules based on SOURCE port?
03:38.49Naikrovekthat was my question
03:38.53Naikrovekhe says its common
03:39.02Naikroveki disagree but i'm no firewall expert so i moved on
03:39.43joobiei lost you Naikrovek - is voIpProt.local.port the answer?
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03:39.54drmessanoI'm sure web browsers are pissing them off then
03:39.58joobieit's very common
03:40.03joobiejust look at the FTP protocol
03:40.17joobieit's just not as common as dport
03:40.37joobiethe reason im doing it is i have a connection at this site that i can make free local / national calls from
03:41.07joobieso ive some funky nat setup so that i can route an asterisk box at a another site via VPN+NAT and emulate the connection as if it was originated locally
03:41.47joobiecould just substitute this with another asterisk box at this site..
03:42.34joobiemy old job, we had *the* most complex iptables setup going across 3 different datacenters
03:43.30joobieit sucked when you wanted to amend things in the ruleset - easy to break something else - but at the same time steps up your knowledge of firewalling / nat / routing 10 fold.
03:44.53joobieBTW Naikrovek, that voIpProt.local.port set the src port correctly :) Thnx
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06:05.39ChannelZCan GoDaddy suck any more ass?
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06:17.58p3nguinnot easily
06:18.09p3nguinThey've done a pretty bang-up job already.
06:19.27p3nguinI bought my first domain name with them in 2003, and they weren't that bad at that time.
06:19.40p3nguinNo, actually, I transferred in to them.
06:20.13p3nguinfrom some junky-ass registrar that didn't even let me control my DNS.
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06:26.49carrarIf they don't have the ability to do glue records they aren't worth using
06:28.06carrarIPv6 glue records that is
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06:33.53Miladis there anyway to add sip header in 1.4 ? it seems remove after 1.2, right ?
06:34.43carrarSipAddHeader()
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06:35.34Milad#  Asterisk func SIPAddHeader: Typically used to set Alert-Info information, e.g. ring tone .wav files. (1.2)
06:35.55Miladow it mean add from 1.2 !
06:35.56Miladtnx
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06:40.37aceiohow do i turn on callerid
06:41.59aceioi am waching cli console  Accepting call from '' to 'xxxxx' on channel 0/1, span 1
06:42.41aceioand i would like too see the incoming numbers
06:47.03ChannelZyou have to print it...  NoOp(${CALLERID(all)})
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06:57.14aceiothx ChannelZ
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06:59.19maxagazI still have the problem of a zero added between the extension set to call outside and the number I need to call, I can't see this zero in the logs nor in the console, does someone have an idea ?
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07:00.19carrarSounds like a dialplan typo
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07:00.52carrarhow do you know it's there if it's not in the logs
07:00.58ChannelZyeah need to see your Dial() and a call
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07:46.32maxagazcarrar, I'm in Beijing, the server is in Shanghai, I compose 9 followed by the number I need to call, for mobile phone outside shanghai, I have to compose 0 first
07:47.57maxagazcarrar, so, I did the following tests: I called my mobile phone number in Beijing by just composing 91580135**** and it works, while it should be 9015801****
07:49.04maxagazcarrar, when I compose shanghai's mobile, it says no need to compose 0 before, and then it doesn't work
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08:05.13carrarSounds like a dialplan issue
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08:45.13tengulrehi,all
08:45.41tengulreanybody build call center under asterisk ?
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09:01.13maxagazcarrar, I can't see any error in my dialplan...
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09:04.40Polysicshello
09:04.55Polysicsasterisk 1.6.1 - can i implement some sort of "single user queue"?
09:05.13Polysicsaka. if i direct call an user and he is busy, can i put the call in a sort of "waiting list"?
09:05.58Polysicsa problem might be that users are also part of queues
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09:28.51kaldemarPolysics: make a queue with a single member
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09:31.53Polysicskaldemar, then i would have each user in two queues
09:32.02Polysicsdoes that work?
09:32.11kaldemarthere's no problem with that.
09:32.37Polysicsone other thing: i have two users, which are also members of queues, that are talking to each other
09:32.58Polysicsuser A is member of queue 1. let's say i have a third user, member of queue 1 too
09:33.27Polysicswhen someone calls queue 1, the queue does not ring user C, instead it waits for A to finish the call, then calls user A
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09:38.51Polysicsany idea why that happens?
09:40.39Polysicsbtw, i think i could just turn on call waiting on my extensions, if i could find the syntax anywhere
09:40.42kaldemarbecause of your queue strategy or some other configuration?
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09:42.43Polysicsstrategy is rrmemory - but shouldn't the queue call the first available member'
09:42.45Polysics?
09:43.10Polysicsi mean, the member the queue "wants" to call is busy, but there is one that is not
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09:58.05Polysicsi think the only solution could be pausing the user from queues
09:58.54Polysicsthen on the hangup event i unpause him
09:59.08Polysicsi suppose the queue does not "know" the user is being called
09:59.41GopalPolysics: try to change the queue ringing strategy as round robin
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10:01.16jkroonhi guys, got a problem with MOH not acting as it should.
10:01.48jkroontransfer happens, when user presses "transfer" the caller goes to MOH, but as soon as it's then supposed to start ringing (destination channel signals ringing) it's just silent.
10:02.15jkroonasterisk 1.6.1.18, confirmed with incomin channels of IAX/2 and DAHDI, transferring channel is SIP, destination tech doesn't matter.
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10:03.24Gopaljkroon: codec might be problem...
10:03.38jkroonGopal, explain ?
10:04.11Gopaljkroon: if you have different codec in the destination when compared to transferring line
10:04.19Gopaljkroon: that could be the prob
10:04.31Gopaljkroon: are you transferring to same network or to outside network
10:05.52jkroonwell, in this particular case it's incoming on DAHDI (PRI/T1) -> SIP.  SIP then puts DAHDI on hold by pressing the transfer button (and MOH plays), at this point she dials the new destination (usually another SIP extension on the same network, sometimes an external DAHDI destination), and as soon as that phone starts ringing the caller gets silence.
10:06.33PolysicsGopal, no, the same problem persists
10:06.47Polysicsi would say it is a matter of pausing users
10:09.32Gopaljkroon: have you given transfer = yes in chan_dadhi.conf?
10:10.20Polysicswhen a user is unpaused from queues, and he is hte only agent in a queue, does the queue get registered as "empty"?
10:10.30jkroonGopal, was set to no.  what exactly does it change?
10:10.36jkroonPolysics, no.
10:10.45jkroonthis is considered a feature, not a bug :(
10:11.23GopalPolysics: yes
10:11.37Polysicsjkroon, which means i could pause users taht are engaged in a "direct" call, and calls to the queue would properly "wait"?
10:11.42PolysicsGopal, oh lol :-)
10:12.20GopalPolysics: not for your question just
10:12.35jkroonPolysics, when a user is unpaused there will be >0 agents in the queue.  there is a queue option that will prevent app_queue from sending calls to in-use consumers from the queue.
10:12.59jkroonringinuse=no <-- should do what you want.
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10:14.21Polysicsjkroon, ringinuse is already set (to 0, not to "no" since queues are in realtime)
10:14.54jkroonok well, works for me so far.
10:14.55jkroonGopal, what does transfer = yes change?
10:15.41Gopaljkroon: it will transfer for flash-hook http://www.voip-info.org/wiki/view/chan_dahdi.conf
10:15.57jkroonGopal, you don't understand the problem then.
10:16.07jkroonthat transfer i want on no
10:16.24jkroonthe issue is that the incoming caller (from dahdi) doesn't hear ringing when the sip side transfers.
10:17.07jkroonyou don't want people that call in to your system to be able to initiate transfers ... imho.
10:17.08Gopaljkroon: ok
10:17.27Gopaljkroon: ok I understood
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10:18.06Gopaljkroon: can you able to transfer a call from SIP to SIP without using dahdi
10:18.06jkroono.O
10:18.24ampilogov_a_hi there :)
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10:20.36Polysicsjkroon, basically, my queues do not honor ringinuse
10:20.38jkroonok, let me rephrase again.  some external to my asterisk system user calls in coming in off my DAHDI channel coming from external.  This goes into a Queue() that gets answered by a SIP/ channel.  So now I have a bridge between DAHDI/x and SIP/y.  At this point y presses "transfer" (putting x on hold) and dials a new number (let's say z), which ends up going to SIP/z, now as soon as SIP/z signals ringing DAHDI/x should hear ringing, but in
10:20.39jkroonstead it gets silence.
10:20.43ampilogov_a_can you help with asterisk? I have trixbox system with SIP phones. There is FOP in the trixbox where you can connect 2 SIP phones by drag&drop. I want to do the same but programmaticaly on PHP (for examp php-agi)
10:21.17Polysicsampilogov_a_, you need to use AMI originate
10:21.35Polysicsyou can do that from php-agi
10:21.43ampilogov_a_does it work for ineranl SIP numbers? i can't dial external numbers
10:21.56Polysicsjust be careful as Originate is a bit difficult at first
10:22.02Polysicsit sure does
10:22.27ampilogov_a_ooh, allright, will read documentation
10:23.20Gopaljkroon: but in the otherend SIP/z there is a ringing, only in dhadi/x there is no ring back tone, correct?
10:23.30jkrooncorrect!
10:23.51Gopaljkroon: what is your tx and rx gain value ?
10:25.38Gopaljkroon: hey whats your dialroute you have given to reach the SIP/z extension?
10:25.56Gopaljkroon: you need to add "r" in the dial route to hear the ringback tone
10:26.26jkroonDial(SIP/z)
10:26.49Gopaljkroon: try this Dial(SIP/z,Tr)
10:26.55jkroonwhy would I need an explicit r?  surely the fact that I receive Ringing() from SIP/z will cause asterisk to pass that back to the DAHDI channel?
10:27.29Gopaljkroon: to hear the ringback tone you need to give r for TDM/PSTN
10:27.38jkroonGopal, surely you mean Dial(SIP/z,,Tr) ?
10:27.45Gopaljkroon: yes
10:27.59Gopaljkroon: I faced the same prob in PSTN after giving r it worked for me
10:28.16jkroon<PROTECTED>
10:28.17jkroon<PROTECTED>
10:28.24jkroonjust the r then.
10:29.12ampilogov_a_hmm, i'm not shure i've done all correct, so is there simple test phone using php-agi script, for instance, to call my SIP phone?
10:29.13ampilogov_a_require_once('/var/lib/asterisk/agi-bin/phpagi-asmanager.php');  $number = '222';  $asm = new AGI_AsteriskManager(); if($asm->connect('localhost:5038', 'admin', 'amp111')) {     echo "connect";         $call = $asm->send_request('Originate',                     array('Channel'=>"SIP/$number",                       'Context'=>'default',                       'Priority'=>1,                       'Callerid'=>$number));      
10:33.42jkroonGopal, it works, however, i've got a concern regarding it, what happens if the phone isn't available?  does it wait until receiving ringing from the new SIP/ channel before sending ringing or does it just send it immediately after receiving 100 Trying or even before that?
10:34.53Gopaljkroon: if the phone is not avaialable it will return back some different tone
10:35.28jkrooni'm still sceptical.  i recall having had false rings with that option before.
10:36.35Gopaljkroon: by default it has to ring without "r" option, I too tried a lot and did the same... but need to check is it giving false ring...:D
10:37.19Gopaljkroon: if SIP phone is not there defnitely it will not ring as I suspect...
10:37.32Gopaljkroon: since the SIP registration will not be there...
10:48.18jkroonGopal, then for internal extensions it should be OK.
10:48.58Gopaljkroon: ok
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10:49.34jkroonhowever, my case that I recall being problematic was a system where I'd receive a call via IAX/2 or SIP, and then call out to an upstream via SIP doing Dial(SIP/upstream/${EXTEN},,r) ... this would start ringing immediately, and then 3 seconds later drop the call when hitting a 404 or something.  so I dropped the r and the problem went away.
10:49.45oktayhello. anybody have the thomson 780wl ? (or know whether the antenna is removable?)
10:52.24Gopaljkroon: so now you are not using r?
10:52.34jkroonnot on that particular switch no.
10:52.49jkroonthat r and m option has been confusing the crap out of me.
10:53.05jkroonbut that switch also doesn't deal with transfers, ever.
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11:01.02oktayit is not removable. thanks for listening :)
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11:17.34Gopaljkroon:ok
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11:20.32Bartockbatzhey folks - a fast and easy way to have the asterisk server email the last 5 incoming call records - so far, I am using a shell script to awk out the last 5 records -
11:21.09Bartockbatzinstead of reinventing the wheel, I wanted to check if there are any tools available for 1.4xx
11:21.35jkroontail instead of awk?
11:21.49jkroonunless you're re-formatting .... :p
11:22.03Bartockbatzoh yeah - used tail
11:22.10Bartockbatzlet me show you what I ue
11:22.12Bartockbatzuse
11:22.39Bartockbatztailf  -n 5 /asterisk/log/csv_file
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11:23.18Bartockbatzawk -F',' ( I would have to look at the fields that I grabbed - basically the incoming call, the time and date)
11:23.32Bartockbatzthen redirect that to a date stamped file
11:23.41Bartockbatzand email the text to an email address
11:24.22Bartockbatzit works, but I was wondering if there is a 'way to build a better mousetrap'
11:24.37jkroondepends on what you want to do but it sounds about right.
11:25.27Bartockbatzso, no fancy tools that I can grab that will save me the scripting for formatting the output? not that I mind doing it, but I like to work smarter, not harder
11:26.29BartockbatzI saw a tool called 'Phone Genie' - but I am wondering if it would suit my needs
11:27.57BartockbatzWell, I am going to format it to make it look pretty, just to keep the client happy ad then look around for a tool
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11:45.34Gopalwhat are all the ports to open if i have a firewall?
11:45.34GopalTCP and UDP?
11:46.50beefpastryGopal: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules
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11:51.19hajkymhi can me sombody help with this?
11:51.24hajkymhttp://pastebin.org/257506
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11:51.58hajkymi want create one user which can only make call and user which can only receive call
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12:03.12Gopalbeefpastry: do you have any idea about RTCP.. do we also enable port for that...
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12:24.00thomasAloha!
12:24.33ThoMeIs it posible with the asterisk-console or API/manager? check if the user logged in?
12:25.40[TK]D-FenderThoMe: What "user"?  Logged into what?
12:26.15ThoMe[TK]D-Fender: login via SIP in asterisk, sorry.
12:26.53tzafrirThoMe, sip show peers?
12:26.57tzafrirsip show registry?
12:27.21[TK]D-Fendertzafrir: Too obvious....
12:27.35ThoMe[TK]D-Fender: what is better? ;) :P
12:27.42ThoMetzafrir: i think show peers or?
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13:39.13rrb3942Are their any characters that cannot be used in a SIP secret? or should be avoided?
13:40.05ManxPowerrrb3942, Is there a specific character that you have in mind?
13:41.13rrb3942Mostly wondering if any special characters will cause a problem
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13:42.23rrb3942I am creating a script to generate a csv for use with the freepbx bulk extension module and right now it just makes a passwords from any printable characters
13:44.18rrb3942So ',' is a problem since it is a csv, but I don't know if maybe '\/#+' would be an issue for the sip secret in asterisk
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13:50.51krionanyone using thomson st2022 with fw v4.69 ?
13:51.13krioni know it's not asterisk related but i have quality problem with it...
13:51.21krionwonder if anyone already fix it
13:56.14ManxPowerI suspect # is the only printable char that is not supported.  " is not a problem in CSV
13:59.04rrb3942Alright, sounds good
13:59.46rrb3942Ill add in quotes for good measure as well
13:59.48rrb3942thanks
14:00.09ManxPoweryou will need to escape the " of course
14:00.52rrb3942yep
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15:20.50LemensTSdoes nat=yes send keep-alives or options?
15:21.02pabelangerLemensTS: no
15:21.44pabelangerLemensTS: only real solution is to lower your SIP registration / qualify timers for nat-keepalive
15:22.15pabelangeror port-forwarding on your router
15:22.35LemensTSthis is for phone device
15:23.08pabelangerLemensTS: Same rules apply
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15:23.30LemensTSThat is done with qualify=XXX right
15:23.42pabelangerLemensTS: correct
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15:24.21LemensTSok. im having a problem with it going unreachable. I had it on qualify=yes nat=yes....what ms you think i should try?
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15:27.59pabelangerLemensTS: 200ms is the default
15:28.11pabelangerLemensTS: Also, what version of Asterisk you using?
15:28.26LemensTS1.4.25.1
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15:29.58pabelangerLemensTS: you should be fine in the 1.4 branch
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15:34.45LemensTSpabelanger: http://pastebin.com/1hMDSFdr   does retrasmitting like this mean it is sending options to the device, but the device isn't responding so it is trying again (retransmitting)
15:35.21LemensTSit trys it 4 times than it destroys the sip deialog
15:35.40pabelanger~sipnat
15:35.41infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
15:35.48pabelangerLemensTS: ^^^
15:36.05pabelangerLemensTS: Likely a configuration issue
15:36.30Belgarathuse dafaultexpirey
15:36.35Belgarathand lower the limit to 10 sec
15:36.47Belgaraththen go up until you find highest working value
15:38.12*** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com)
15:38.27rrb3942some routers also have udp session timeout options, increasing those to several minutes can help sometimes
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15:46.10ZnuffNGEN
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15:54.37puzzledtzafrir: about http://docs.tzafrir.org.il/dahdi-linux/README.html#_oslec : even on CentOS 5.5 with kernel 2.6.18 do you still need to take a newer kernel like 2.6.28 as it says in your doc?
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15:55.17vk4akpHI. Looking for some help with the ChanSPy command.
15:55.24tzafrirpuzzled, "need newer kernel" is for getting oslec from that kernel
15:55.36tzafrirIf you don't, just use any kernel (supported by DAHDI)
15:55.48puzzledtzafrir: got it. thanks
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16:00.31vk4akpexten => _88XXXX,1,Chanspy(SIP/${EXTEN:2}|b)
16:01.00vk4akpHow can I change this so teh number entered after the 88 can be of variable length. Like 1-4 long instead of having to be 4 long?
16:01.11LemensTSif there is a defaultexpirey, why is there a maxexpirey?
16:01.32LemensTSi dont see how it could be a range
16:03.13*** join/#asterisk shader (~40846872@gateway/web/freenode/x-tlrpyxnujngkhutf)
16:04.51shaderdo I need to implement things like congestion tones myself when I dial out to an external number, or will the remote end handle that?
16:06.31idespinnervk4akp, i would just make 4 entries: _88x, _88xx, _88xxx, _88xxxx
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16:08.01vk4akpI don't seem to be able to make this feature work anyhow.
16:08.03[TK]D-Fenderexten => _88., <- variable length
16:08.17vk4akpHave you used this ChanSpy feature?
16:08.28idespinner[TK]D-Fender, except that includes 1-infinity digits
16:08.29[TK]D-Fendervk4akp: What ver of * are you using?
16:08.36idespinneri think he wants 1-4 digits only
16:08.42vk4akpYes [TK]D-Fender, I ended up doing it with the DOT . thanks.
16:08.49[TK]D-Fenderidespinner: My statement remains entirely accurate :)
16:08.57vk4akpUm. I am using the special one for APP_RPT
16:09.09[TK]D-Fendervk4akp: Now try expressing that with NUMBERS.
16:09.20vk4akpAsterisk SVN--r588M built by root @ shazam on a i686 running Linux on 2010-04-01 14:19:13 UTC
16:09.40[TK]D-Fendervk4akp: What branch is that from?
16:09.53vk4akpYOu would have to tell me how to find out.
16:10.09vk4akpIt's a special one built for APP_RPT it runs into a aradio network.
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16:11.39vk4akpI think it is 1.4.somethign with stuff added to fix bugs.
16:12.43shadervk4akp: were you the one that compiled it?
16:12.50vk4akpNo.
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16:13.05vk4akpthe APP_RPT / All-STar guys do all that.
16:13.09vk4akpI am just a user.
16:13.16shaderok
16:13.28vk4akpWell. I guess technically it was compiled on our system.
16:13.40shaderbut you didn't do the installation yourself
16:13.43vk4akpIts a special version for Amateur Radio use.
16:13.56vk4akpOriginally I installed it with a alot of problems.
16:14.08vk4akpBut this release was installed by one of the nice ALL_Star guys
16:14.23vk4akpThey updated the system for me a month or so back.
16:14.40shader[TK]D-Fender: if you dial out to an external number, say via a sip service provider, do they provide features like busy tones, or do I have to implement that myself?
16:14.56shadervk4akp: ok
16:15.15shadervk4akp: did they check out the source onto your system? it might be in /usr/src
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16:15.23shaderin which case you could probably figure out the version
16:15.27vk4akpSo anyhow. I am looking for ways to monitor different extensions on teh PBX.
16:15.39vk4akpI currently have issues where the system is being abused and I need a way to monitor.
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16:17.08shaderwhat kind of monitoring are you looking for?
16:17.21shaderChanSpy and ExtenSpy seem to be for recording calls
16:19.38p3nguinExtenSpy() spies based on the extension, ChanSpy() spies on the channel.  Neither are for recording.
16:20.33p3nguinMixMonitor() and Monitor() are used to record calls.
16:22.01shaderfine, but they seem to be for listening in on a channel, with the option for recording
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16:22.19shaderwhich may or may not be what vk4akp is looking for
16:22.29ManxPowervk4akp, "core show applications" is your friend.
16:22.52ManxPower(might even be a friend with benefits....of knowing Asterisk if you play your cards right
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16:23.50p3nguinYeah, I didn't mean to mislead by saying the aren't for recording.  I should have said their main purpose is not recording.
16:24.25ManxPowershader, with VoIP (any protocol) or ISDN (any form) you get back a status code, it is up to you to play Busy, Congestion or Whatever.
16:25.29shaderok
16:25.45shaderhow do you have yours set up?
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16:35.39BANSALcan anybody suggest me any ebook or web link which can provide me complete documentation about asterisk ?
16:35.51Qwell~book
16:35.52infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
16:36.39vk4akpSorry back.
16:36.46vk4akpExtenSpy might do what I need then.
16:39.56shadervk4akp: what do you need?
16:41.00vk4akpThe ability to monitor an extension that someone is on.
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16:58.34Kevin`vk4akp: like chanspy?
17:02.00vk4akpYea like that.
17:02.05vk4akpI couldn't make it work though.
17:02.25vk4akpMainly being able to monitor a meetme conference without chiming in.
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17:03.37rocksfrowis it possible to configure a polycom phone to ring forever based on specific route?
17:04.14ManxPowerrocksfrow, route?
17:04.23rocksfrowwhat i meant by that was, cid i guess
17:04.31rocksfrowbasically, i want the phone to ring forever if coming from the calling queue
17:04.36rocksfrowbut ring for x seconds if anything else
17:04.40rocksfrowwithout using two extensions
17:04.41rocksfrowpossible?
17:04.48ManxPowerFigure out what you really want then ask again
17:04.54rocksfrow....
17:04.58rocksfrowdid what i just say not make sense
17:05.15ManxPowerrocksfrow, set the polycom to ring (almost) forever, then do all your timeouts in the Asterisk dialplan
17:05.39shaderor play ring tones using asterisk
17:05.54shaderunless you mean timeout built into the phone
17:06.00ManxPowershader, that makes no sense whatsoever
17:06.09rocksfrowlol
17:06.23rocksfrowwell, i have my calling queue setup to ring forever, instead of ringing / timing out / ringing / timing out
17:06.35rocksfrowbut, the phones stop ringing after a bit, bc of the timeout on the phone i assume
17:06.54rocksfrowi was just curious if anybody knew a way to configure this directly within the phone
17:07.21ManxPowerrocksfrow, YES!
17:07.36ManxPowerThe Polycom Admin Guide - the same place ALL information on Polycom setup is located.
17:08.24BANSALIs there any difference using asterisk if I install it using fedora repo or if I install it using tar ball ?
17:09.18*** join/#asterisk shader (~40846872@gateway/web/freenode/x-tcwhbtllrxqfruem)
17:10.11BANSALactually I have install it from fedora repo .. and now I don't know how can I use additional module if I want ...
17:10.13rocksfrowManxPower, lol, you're such a smartass
17:10.23rocksfrownothing comes easy from you lol
17:10.40rocksfrowbut thank you!
17:13.00shaderrocksfrow: I'm somewhat confused as to why the phones would time out. Don't you have asterisk answer the connection before it puts them in a queue?
17:13.33rocksfrowshader, they're in the queue..
17:13.51rocksfrowwhile they're in the queue, the agents are rang over and over
17:13.58bmoraca_workdoes anyone here happen to have a copy of the Cisco Unity Express Script Editor v2.3?
17:14.09rocksfrowif i configure it to timeout after 15 seconds and restart, it works fine... but at the same time leaves extra records in the history
17:14.22rocksfrowso, i reconfigured it to ring forever, but need the phones to also allow ringing forever
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17:15.38shaderrocksfrow: so this is for the agents' phones?
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17:17.46shaderwell, that does sound like something that would be on the phone itself
17:18.00shaderunfortunately, there isn't a #polycom channel
17:18.09Naikrovekthis is as close as you'll get
17:18.20Naikrovekthere really isn't a need for #polycom with us in here :)
17:18.32rocksfrowNaikrovek, :-p
17:18.51rocksfrowim thinking about just making a second extension
17:18.59shaderand yet, no one seems to want to answer rocksfrow's polycom specific questions
17:19.19Naikrovekwe all have other things to do.  most of us anyway
17:19.19beefpastryBANSAL: have to get the src rpm and rebuild if it's missing modules (in other words, might as well compile from source, anyway)
17:19.31rocksfrowNaikrovek, thanks, lol
17:19.56Naikrovekhow long does the polycom ring before it stops
17:20.42Naikrovekah i'm reading the backlog now
17:20.45Naikrovekone moment
17:21.15Naikrovekokay
17:21.28[TK]D-Fender[13:06]<rocksfrow>but, the phones stop ringing after a bit, bc of the timeout on the phone i assume <- Correct... this is an assumption
17:21.40Naikrovekwhat manxpower said is right, i think
17:21.46rocksfrow[TK]D-Fender, a pretty accurate one...
17:21.55rocksfrow[TK]D-Fender, i'm pretty positive, actually lol
17:22.00rocksfrowpretty common sense
17:22.07Naikrovekred flag
17:22.11[TK]D-Fenderrocksfrow: People are usually very sure of things they aren't really looking at
17:22.16rocksfrowlol..
17:22.18rocksfrowyes, i understand that
17:22.26Naikrovekwhenever someone says something is common sense, red flag
17:22.26rocksfrowyou have questions for me to confirm?..
17:22.30BANSALbeefpastry: actually I am using fedora 12 and installed it from repo ... now for libpri and dahdi installation should I do all the work to /etc asterisk or anything else ?
17:22.30rocksfrowhaha..
17:22.32rocksfrowwhatever
17:22.36rocksfrowshall i explain?
17:22.46[TK]D-Fenderrocksfrow: Got a real call and real configs to show us?
17:22.50Naikrovekshow us your config where you've set it to ring forever
17:22.57Naikrovekand show us the call log showing that it rings forever
17:23.00rocksfrowit's a freepbx system XO
17:23.02rocksfrowhere it comes...
17:23.04Naikrovekyep
17:23.04[TK]D-Fenderrocksfrow: I don't do "stories", I do "configs & debug"
17:23.05rocksfrowlol
17:23.06Naikrovekhere it comes
17:23.13BANSALbeefpastry: /etc/asterisk
17:23.44rocksfrowthis is a polycom specific question
17:23.54rocksfrowi'm simply asking how i can modify the ring timeout on the polycom
17:23.56Naikrovekthen show us the polycom specific config files
17:24.00Naikrovekconfig files
17:24.04Naikrovekyou modify config files
17:24.09Naikrovekshow us your existing config files
17:24.15rocksfrowyeah..i was wondering if anybody knew the config off the top of their head
17:24.20Naikrovekjesus
17:24.27Naikrovekthis is why you don't get an answer
17:24.30rocksfrowi'm provisioning via FTP thanks to somebodys help with some scripts here
17:24.39Naikrovekthat might have been me
17:24.41shader[TK]D-Fender is really strict about that stuff. If you sneezed on him, he probably wouldn't give you a tissue until after you showed him the logs :D
17:24.41rocksfrowNaikrovek, ...are you serious? lol ...i'm bringing up the config damn
17:24.42Naikrovekshow us the sip.cfg
17:24.55[TK]D-Fender13:23]<rocksfrow>i'm simply asking how i can modify the ring timeout on the polycom <- there is no such thing
17:25.14[TK]D-Fenderrocksfrow: You want to let a call ring pretty much forever?  Go right ahead
17:25.38Naikrovekif [TK]D-Fender is right, as per usual, then its freepbx that is stopping the ringing
17:25.44Naikrovekso it ceases to become a polycom only question
17:26.02rocksfrowokay how about this
17:26.07rocksfrowi'll change it from unlimted to 3 minutes
17:26.13rocksfrowand watch the polycoms stop rining at the same exact time
17:26.26[TK]D-Fenderrocksfrow: sure,... go show us
17:32.09rocksfrowyeah man..
17:32.53rocksfrowso, i changed the retry timeout from unlimited, to 60 seconds, and the phone does as expected...it rings for a little less than 60 seconds..then restarts ringing a couple seconds after that
17:33.02rocksfrowbecause it restarts the ringing process..
17:33.14rocksfrowi'm trying to set this to unlilmited so the phones don't go through that brief hesitation
17:33.23rocksfrowwhen i do that, the phones never start ringing again after the first cycle
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17:41.21chazzamrocksfrow: you still don't have logs or config...
17:41.54rocksfrowchazzam, bc im testing something
17:42.01rocksfrow#call.offeringTimeOut.label=
17:42.01rocksfrowcall.offeringTimeOut.description=Time in seconds to allow an incoming call to ring before dropping the call,  \
17:42.01rocksfrow<PROTECTED>
17:42.06rocksfrowfound that...^
17:42.17[TK]D-Fenderrocksfrow: And what was it set to?
17:42.31rocksfrowits set to 60, im making it 0 and reprovisioning to test
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17:44.43LemensTSI have ata/sip phones deployed around the USA off my asterisk server, im wanting to build a provision/firmware server. I have option of TFTP, HTTP, HTTPS on cisco SPA2102 what do you guys suggest I use
17:45.11LemensTSIm using Polycom Soundpoint phones also
17:45.51LemensTSIve not read the provisioning docs on them, ive done a tftp server for polycom phones before...http and https im not sure on them
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17:47.19Naikrovekare http and https supported?  yes
17:47.43Naikrovekftp, tftp, ftps, http, https
17:48.10LemensTSWhat do you prefer?
17:49.00Naikrovekftp
17:49.02LemensTSGuess ill have to have ftp/tftp for the firmware on both...
17:49.17LemensTSyou like ftp for the .cfg also?
17:49.20Naikrovekyeah
17:49.35Naikroveki put everything in a ftp directory, then tell the phones to log into that server
17:49.44Naikrovekeverything goes in root directory from the phone's POV
17:49.49Naikrovekeverything just works
17:50.12LemensTSWhy not tftp?
17:50.24Naikrovekbecause i want to enforce passwords
17:50.25LemensTSTFTP seems finicky from what I have used..
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17:52.07[TK]D-FenderI don't treat my PBX as "trivial"
17:57.43*** join/#asterisk BANSAL (~BANSAL@117.207.82.156)
17:59.48leifmadsenok this is the first time I've really had an issue like this, so I'm curious if I should be trying something like "relaxdtmf" or the like. Problem is that I have a PRI on a Digium TE220p card. For some reason sometimes I get either duplicate DTMF on the first digit (sometimes) or it misses the first digit entirely. This seems to most often happen with '2' (oddly enough). It's not consistent, and I don't believe the card h
17:59.48leifmadsenas an echo cancel board on it (not 100% sure, researching that now). Configuration information here: http://pastebin.com/p1m2mcpV
18:00.54rocksfrowNaikrovek, do configurations overwrite if they're previously set..or no? CONFIG_FILES="preferences.cfg, ringer.cfg, server.cfg, 0004f22b1224-phone.cfg, phone1.cfg, sip.cfg"
18:01.04leifmadsenconfirmed just now, no octasic echo cancel daughter card
18:01.05rocksfrowis a configuration in sip.cfg going to override something in xxxx-phone.cfg?
18:02.03[TK]D-Fenderrocksfrow: So far you're drilling without looking at the simple proof.
18:03.02*** join/#asterisk Canabinoide (~eu@189.96.54.38)
18:03.29rocksfrow[TK]D-Fender, i think you're right
18:03.58*** join/#asterisk glwgoes (~guilherme@200.175.61.250)
18:07.09BANSALhey guys I am having this problem installing dahdi .. http://pastebin.org/258654
18:07.18BANSALplease take a look ...
18:08.48[TK]D-FenderBANSAL: You do not appear to have the sources for the 2.6.32.12-115.fc12.i686.PAE kernel installed. <-
18:09.01[TK]D-FenderBANSAL: Go install the pre-requisites that the docs tell you you need
18:09.18BANSAL[TK]D-Fender: I have install kernel-devel ..
18:10.22BANSALbut still the same problem ..
18:10.29[TK]D-FenderBANSAL: You need the devel, and the headers.  Go prove precisely which of each you have installed.
18:10.48BANSAL[TK]D-Fender: yup I have install both ...
18:11.04Slugs_he said prove
18:12.27leifmadsenBANSAL: you've installed the same version for the currently running kernel?
18:12.30leifmadsenlikely not
18:12.53BANSALhttp://pastebin.org/258682
18:12.55*** join/#asterisk k-man (~jason@unaffiliated/k-man)
18:12.56leifmadsenif you're running and older kernel than what is currently available, then the packages you install will be for the NEWEST kernel and not what you're running
18:13.10leifmadsenBANSAL: uname -a
18:13.10BANSALleifmadsen: how can I check it ?
18:13.18leifmadsenwhat i running?
18:13.23leifmadsenwhat is running?
18:13.34[TK]D-Fender[14:08]<[TK]D-Fender>BANSAL: You do not appear to have the sources for the 2.6.32.12-115.fc12.i686.PAE  kernel installed. <-
18:13.36[TK]D-Fender^^^6
18:14.09[TK]D-FenderBANSAL: None of those packages you show are for the PAE kernel
18:14.11BANSALleifmadsen: Linux localhost.localdomain 2.6.32.12-115.fc12.i686.PAE #1 SMP Fri Apr 30 20:14:08 UTC 2010 i686 i686 i386 GNU/Linux
18:14.19leifmadsenwhat [TK]D-Fender said
18:14.24leifmadsen.PAE is your problem
18:14.28leifmadsenthey do not matc h
18:14.34BANSAL[TK]D-Fender: which package .. can you tell ...
18:14.42leifmadsenboth
18:14.51leifmadsenlook at them - they are not the exact same string
18:14.58leifmadseni686 != i686.PAE
18:15.51BANSAL<PROTECTED>
18:16.08leifmadsenBANSAL: the correct version of those packages
18:16.59BANSAL<PROTECTED>
18:17.14*** join/#asterisk atis_work (~atis_work@193.238.212.171)
18:18.02pabelangerBANSAL: rpm -q kernel-devel
18:18.34BANSALkernel-devel-2.6.32.12-115.fc12.i686
18:18.35BANSALonly this result ...
18:19.32pabelangerBANSAL: like leifmadsen said, you are running PAE kernel, but don't have source installed for it
18:20.05BANSAL<PROTECTED>
18:20.08*** join/#asterisk dotnetted (~dotnetted@75.138.79.27)
18:20.25BANSALthis is not in repo I think ...
18:20.44leifmadsenyou must be running a custom kernel or something -- google is likely helpful at this point
18:21.06dotnettedhey all - I just changed my asterisk users UID to 999 to hide it from the login choices on Ubuntu/Gnome and after a reboot asterisk seems to be missing all the sip commands - what might have happened? thanks
18:21.26pabelangerBANSAL: install the kernel headers.    kernel-PAE-devel-2.6.32.12-115.fc12.i686.rpm
18:21.43[TK]D-Fenderdotnetted: Did you change the owner on all of *'s files?
18:21.54[TK]D-Fenderdotnetted: Because files actually store the NUMBER, not the name.
18:22.13[TK]D-Fenderdotnetted: So if you just changed the mapping, the files remain with the old number and no name to match
18:22.25pabelangerBANSAL: http://lmgtfy.com/?q=fc12+pae+kernel-devel
18:22.35dotnettedoh doh - that would explain it heh - thanks
18:22.54dotnettednow to try to find all of the files... ;)
18:23.24pabelangerdotnetted: look at your asterisk.conf for the paths
18:23.49dotnettedpabelanger: thanks
18:25.27*** join/#asterisk joako_ (~joako@opensuse/member/joak0)
18:27.11dotnettedafter changing all the permissions whats the best way to reload asterisk (as it would on reboot) without rebooting?
18:27.37shaderwhich dahdi package is required for meetme conferencing only: dahdi-linux, or dahdi-linux-complete?
18:29.52*** join/#asterisk scalex000 (~chatzilla@190.166.188.12)
18:29.57scalex000Hi guys
18:30.21[TK]D-Fendershader: The first includes al the drives.  The latter includes toos you may never need regardless
18:30.48*** join/#asterisk maddhat (~MadHatter@173-26-185-193.client.mchsi.com)
18:31.21[TK]D-Fenderdrivers*
18:31.43jayteesuspect TK's fingers are tired
18:32.05maddhathi everyone. sorry for the newbie question... Wanting to setup an answering machine from my PSTN and i have 2x PCI modems laying around for it.. will either of these work? Gateway 6001761 Modem 17510, Compaq 56K Modem - CIS WS/M1-5614PM3 PM1560024001
18:32.31jayteeI see a soldering gun and years of frustration in someone's future
18:35.42Naikrovekmaddhat: won't work.  modems don't do voice
18:35.53jaytee~savemoney
18:35.54infobot<Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards.
18:36.23pabelangerjaytee: lol
18:36.27pabelangernice macro
18:36.35*** join/#asterisk dotnetted (~dotnetted@75.138.79.27)
18:36.40jayteeit's just infobot being a genius
18:36.53*** join/#asterisk dohd (~Xaa@nala.dohd.org)
18:36.54pabelangerinfobot: :)
18:36.55infobot(:
18:37.09jaytee~botsnack
18:37.09infobotaw, gee, jaytee
18:37.36pabelangerjaytee: any way for infobot to list all of his commands?
18:37.50jayteepabelanger, not sure really
18:37.54dotnettedwhats the module called that provides sip functionality?
18:38.04jayteechan_sip.so
18:38.05Naikroveksip module?
18:38.21BANSALpabelanger: I have installed dahdi ... but now there is error installing asterisk ...
18:38.40pabelangerinfobot: what's up?
18:38.40infobotUp is the direction away from the central point of gravity.
18:38.40BANSALpabelanger: make: *** [makeopts] Error 1
18:38.55pabelanger~pb
18:38.56infobot[~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
18:38.56rocksfrowis it possible to override this for a specific extension?
18:38.59pabelangerBANSAL: ^^
18:39.10BANSALyup .. sure ..
18:39.32maddhatNaikrovek: ok. darn :-/  well thank you for the information
18:39.55Naikrovekwelcome
18:40.11rocksfrowNaikrovek, [TK]D-Fender im just blind..
18:40.14rocksfrowand retarded
18:40.21BANSALpabelanger: http://pastebin.org/258729
18:40.28Naikrovekrocksfrow: what's the solution?
18:40.46rocksfrowwas missing the ringtime setting specific to an extension
18:40.57rocksfrowmy blindass was looking for 'timeout'
18:40.58Naikrovekgotcha
18:41.01Naikrovekheh
18:41.03rocksfrownot ringtime, heh
18:41.07Naikrovekfreepbx setting, then
18:41.16rocksfrowhence my apologee :-)
18:41.26rocksfrowbut there is a setting on the polycom phone too though :-)
18:41.28rocksfrowthat sets to 60
18:41.34rocksfrowso my bet is, when i jump this up..ill hit that limit
18:44.12dotnettedafter changing asterisks UID I screwed up the permissions for many of the files it uses - I fixed all I could find but now "module load chan_sip.so" fails without much info - debugging is on and nothing useful was logged to log/messages or log/full - whats the best way to debug why the module failed to load?
18:44.42[TK]D-Fenderdotnetted: Go prove the owners are right on everything
18:45.00BANSALpabelanger: got anything ?
18:46.00dotnetted[TK]D-Fender: do you know which files would need to be set (at the very least) for chan_sip.so to load? I cant seem to find any that have the wrong permissions anywhere
18:46.21[TK]D-Fenderdotnetted: Modules, configs...
18:47.23dotnetted- /usr/lib/asterisk/modules/* and /etc/asterisk/* all have the right permissions (and all the modules show up under 'module show')
18:47.55[TK]D-Fenderdotnetted: better start showing us...
18:48.24pabelangerBANSAL: ./configure
18:48.47BANSALpabelanger: done ..
18:49.02pabelangerBANSAL: make
18:49.25BANSALpabelanger : after ./configure ... make ?
18:49.42pabelangerBANSAL: yes. read the INSTALL file
18:49.51*** join/#asterisk Alagar (~Administr@122.164.38.169)
18:49.53BANSALpabelanger: done .. same results
18:50.08pabelangerpastebin your config.log file
18:50.12BANSALpabelanger: am I still missimng something ?
18:51.35BANSALpabelanger: http://pastebin.org/258757
18:51.56*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
18:53.17pabelangerBANSAL: you are missing your compilers, install them
18:53.32BANSALas ... yum install ...
18:54.13pabelangerBANSAL: yes
18:54.16pabelangerBANSAL: http://lmgtfy.com/?q=asterisk+fedora+howto
18:54.25pabelangerfollow the guides listed there
18:57.32BANSALpabelanger: which compiler I have to install ?
18:57.45BANSALpabelanger: I have installed gcc
18:58.55pabelangerBANSAL: config.log tells you what you are missing
18:59.15pabelangerBANSAL: http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_fedora.html
18:59.20*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
19:04.53*** join/#asterisk Z_God (~julius@wlan230085.mobiel.utwente.nl)
19:08.13Qwelllooks at puzzled
19:08.30QwellSMTP error from remote mail server after end of data: [83.163.53.136]: 550 5.7.1 No spam accepted
19:08.57puzzledhmm lemme check that. which To: ?
19:09.12Qwellpatrick@
19:12.59puzzledQwell: please try again. I forgot to remove a temp block for some Chinese spam bot going crazy that used your name in From:
19:13.11Qwellseriously?
19:13.31puzzledonly first name, not surname or first + surname
19:13.34Qwelloh
19:14.42*** join/#asterisk knot (yiffstar66@unaffiliated/devemo)
19:15.22puzzledQwell: got it. thanks for the response. will have a look
19:16.20maddhatNaikrovek: from what im reading here.. softmodems can be used: http://linuxgazette.net/120/smith.html
19:16.35maddhatthere something im missing?
19:17.21leifmadsenmaddhat: not any soft modem
19:17.23Qwellmaddhat: That specific one.  but it's crap.
19:17.31leifmadsensuper crap
19:17.36maddhati see
19:17.47leifmadsenyou're way better off with an FXO -> SIP device
19:17.48*** join/#asterisk atis_work (~atis_work@193.238.212.171)
19:18.07idespinnerspa3102 is only 60$
19:18.09maddhatbut looking at the price of the asterisk cards.. yikes!
19:18.16idespinner1fxo 1 fxs
19:20.16Qwellmaddhat: if that scares you, go look at how much "real" PBX hardware costs.
19:21.09maddhatim not arguing that its an extremely low priced alternative.. but for a hobbyist like me who just wants simple features it seems like overkill
19:25.18Corydon76-digmaddhat: it's actually quite reasonable for hardware
19:25.29pabelangermaddhat: TDM400P is sub $200
19:25.41pabelangerdepending on how many ports you get
19:25.48Corydon76-digTDM410 is what he wants
19:25.54pabelangerSPA3102 is another option
19:25.56maddhatwhat about just something like this: http://cgi.ebay.com/X100PSE-FXO-PCI-Digium-Asterisk-Trixbox-Elastix-FreePBX-/130339643798?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item1e58d92196
19:26.52Corydon76-digmaddhat: I have one of those cards.  Trust me.  You want the TDM410 card
19:27.18[TK]D-FendermadWhat are you looking to do exactly?
19:27.22pabelangerIs the X100p even supported any more?
19:27.36Corydon76-digpabelanger: we've never removed support for it in DAHDI
19:27.37[TK]D-Fendermaddhat:  What are you looking to do exactly?
19:27.57pabelangerCorydon76-dig: gotcha
19:29.10maddhat[TK]D-Fender: id like to setup a simple voicemail box... and possibly look into getting a SIP phone
19:29.45[TK]D-Fendermaddhat: For your single analog line?
19:29.49maddhatcorrect
19:31.15Corydon76-digmaddhat: if you can accept a card that works to a degree, but I don't think will work to your complete satisfaction, then get the X100P
19:31.45lirakisis there a config option to tell asterisk where sounds reside?
19:31.50Corydon76-digBut don't get it if you think it's going to be the last telephony card you'll ever need
19:31.55*** join/#asterisk shader (~40846872@gateway/web/freenode/x-bwojznmwmbgntksw)
19:31.58maddhatis voice quality the difference between the two?
19:32.20Corydon76-digmaddhat: and echo, callerid success...
19:32.21[TK]D-Fendermaddhat: The SPA-3102 may be a better bet for you.  You'll get the FXO side out of the way, reduce system requirements, and get an FXS interface to use an analog hone with your system immediately.
19:32.26shaderany good references for providing IMAP access to asterisk voicemail?
19:33.46lirakisastvarlibdir  ?
19:33.58Corydon76-dig[TK]D-Fender: for someone who's not yet familiar with Asterisk, I'm not sure I'd recommend an external FXO device
19:34.33shadernvm, I found a section on it in The Book
19:34.42Corydon76-digMost of the time, people want to have a low-effort audio path, then they can branch out
19:34.48pabelangerlirakis: yes
19:34.58[TK]D-FenderCorydon76-dig: Compare filling in a few small blanks VS the giant story we got from this other guy a few minutes ago who can't compile shit, and all those with bitchy kernel, ones who get updaetd all the time , etc.
19:35.08[TK]D-FenderCorydon76-dig: Seriously... almost a non-issue
19:35.28lirakishrm ... asterisk is saying it cant open tt-monkeys (ulaw) ... but its def. there, and the sound file directory is correct
19:35.46maddhati enjoy tinkering... and am no stranger to networking configuration.. im sure i can manage setup
19:35.52[TK]D-FenderCorydon76-dig: there are certain things that a DAHDI driven card can do better for sure... but in many other cases I figure it isn't worth it for the "el-cheapo" side
19:36.00lirakisand ... i have ulaw sounds... and that is what my codec has been negotiated to... i wonder if there is some i/o prob.
19:37.08pabelangerlirakis: pastebin output
19:37.47maddhatthanks for the options everyone. ill look into the 3102 more seriously as an option.
19:37.59lirakispabelanger, http://pastie.org/969987
19:39.26pabelangerLinuturk: ls /var/lib/asterisk/sounds/en/ | grep tt-monkeys
19:39.31*** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br)
19:40.08lirakispabelanger, http://pastie.org/969992
19:40.55lirakispabelanger,  ... asterisk.conf ...  http://pastie.org/969995
19:41.13lirakispabelanger, my sounds are not in the en folder
19:41.17lirakis... do they need to be?
19:41.22pabelangerlirakis: yes
19:41.25lirakisi guess if i have the directory prefix thingy set
19:41.28pabelangerif that is your default language
19:41.28lirakisok .. ill move
19:41.35italorossiHello all, I'm getting this message: "audiohook_read_frame_both: Read factory 0xa27c300 was pretty quick last time, waiting for them.". Does anyone know the cause of this? I've tested with different asterisk 1.4 versions (26, 28, 31).
19:42.34filethat message is primarily for debugging ...
19:42.40*** part/#asterisk maddhat (~MadHatter@173-26-185-193.client.mchsi.com)
19:44.24lirakispabelanger, do i need to set language=en in sip.conf ??
19:44.56italorossiok.. this can cause some strange behaviors in recorded calls? Calls with silence or repeating the audio at the end?
19:45.06pabelangerlirakis: no, it is the default
19:45.53lirakishmmm
19:45.53fileitalorossi, it won't cause strange behaviors - it's a perfectly normal message to have under normal circumstances
19:45.53lirakispabelanger, still getting the same thi8ng
19:46.30*** join/#asterisk lanning (~lanning@208.87.235.224)
19:48.11lirakisi think there is an i/o issue ... this was working with voicemail before
19:49.31*** join/#asterisk david456 (~david7345@118.172.90.13.adsl.dynamic.totbb.net)
19:49.42lirakisand now i get the same crap for voicemail ... saying that it the vm sound files dont exist
19:50.03muiroDoes the 'allowmultiplelogin' setting in manager.conf disallow multiple logins based on the manager user ID or the host/ip?
19:50.07pabelangerlirakis: what did you change on your system, if this used to work
19:50.36lirakispabelanger: i changed what you just suggested .. moved files to the en directory
19:50.45italorossifile: thanks! I'm having some problems with call recordings and this message called my atention. I have the same configuration (boards, asterisk version, call flow) in other client and I don't see the same behavior.
19:51.26lirakispabelanger, this is running on a hacked wrt54g router.... i put a 2 gig sd card in running serial i/o over free gpio pins on the cpu .... so.... my guess is there is some issue with reading
19:52.06lirakisis waiting for my fanless 1u server to show up so i can stop running on 13mb of ram
19:52.07muiroEvery time I set 'allowmultiplelogin' to 'no', no manager is able to connect to the system at all. Not even the first one
19:52.21*** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7 (2010/05/04), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs
19:52.48pabelangerlirakis: maybe, then move your files back to the previous directory for VM to work again
19:53.03leifmadsenAsterisk 1.6.1.20 and 1.6.0.28 are now available. These are the final maintenance releases for the 1.6.0 and 1.6.1 branch. Please see the release announcement: http://www.asterisk.org/node/51358
19:53.28Naikrovekfinal?  call them 1.6.0.99 and 1.6.1.99
19:53.40Naikrovekkidding
19:53.43Naikroveki know it doesn't work that way
19:53.59pabelangermuiro: There must already be a manager login used.   ;IF set to no, rejects manager logins that are already in use.
19:54.00sprite--Has anyone ever experienced syslog.cron killing asterisk? http://gist.github.com/407996
19:55.15*** join/#asterisk sprite-- (~sprite@c-98-251-108-29.hsd1.ga.comcast.net)
19:55.18sprite--sorry got disconnected
19:56.36muiropabelanger: the very first manager login that I attempt fails on could not authenticate. If I set allowmultiplelogins to yes, it does not fail. I even check 'manager show connected' before trying this and no use is connected.
19:56.46ryanlinrejected because extension not found.
19:57.11ryanlindoes anyone know why?
19:57.28ryanlinthis is coming from our callmanager to asteriks
19:57.30ryanlinasterisk
19:57.36[TK]D-Fenderryanlin: Because there is no extension to match hat was dialed in the context the call is looking in <-
19:57.40[TK]D-Fenderryanlin: Just like it says
19:58.06ryanlininteresting
19:58.15ryanlinexten => 3300,1,Macro(corpuser,3300)  ;test-phone
19:58.21ryanlini have this entry defined in extensions.conf
19:58.32ryanlinand the appropiate entry defined in sip.conf for the actual phone config
19:58.53ryanlinthis should be sufficient..i don't know why it's complaining that the extension is not found
19:58.55[TK]D-Fenderryanlin: Nowhere do I see a COMPLETE call with SIP DEBUG showing what CONTEXT its looking in, nor enough dialplan to prove that that is the one its looking for, and that its in the right place
19:59.01[TK]D-Fenderryanlin: PASTEBIN is your friend.
19:59.03[TK]D-Fender~pb
19:59.03infobot[~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
19:59.04ryanlinone sec
19:59.04[TK]D-Fender^^^^^^^^
19:59.12ryanlinhold
20:01.14ryanlinhttp://internetworkpro.org/pastebin/6386
20:01.38ryanlinthe initial invite looks okay
20:02.28pabelangerryanlin: what context did you setup in your sip.conf?
20:02.33david456Hello, friends.  Is there a preference order that asterisk has for sound file formats?  Or does it just depend on the order in which the modules loaded?
20:02.39[TK]D-Fenderryanlin: Where is your dialplan to match?
20:03.03[TK]D-Fenderpabelanger: We already see that answer
20:03.07shaderfor only meetme conferencing, which dahdi package do I need, dahdi-linux-complete, or just dahdi-linux?
20:03.18[TK]D-Fendershader: Latter.  I already answered this
20:03.27shaderyeah, I think I disconned for some reason
20:03.38shaderso I didn't get to hear, sorry
20:04.11shaderwill I need dahdi-tools?
20:04.52[TK]D-Fendershader: Probably not
20:05.36italorossiSetting the AUDIOHOOK_INHERIT before or after the MixMonitor makes any difference?
20:05.53shaderok, thanks
20:06.41pabelangerdavid456: it all depends on what format you are providing to your phones.
20:07.19pabelangerdavid456: if your phone need ulaw, then install .ulaw. You don't want to be transcoding audio
20:07.19sprite--http://gist.github.com/408010 why the hell does sysklogd restart my asterisk?
20:07.25TrixboxerHi
20:07.46Trixboxerhow can I dial a SIP URI from my dialpla ?? any idea ?
20:07.51Trixboxerdialplan*
20:08.11pabelangersprite--: Because you told syslogd to restart it.  check your syslogd scripts
20:08.35pabelanger~book
20:08.36infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
20:08.40pabelangerTrixboxer: ^^^
20:09.16Trixboxerthnx pabelanger.. will have a look at it...
20:10.01ryanlinone sec
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20:19.21bent_screwdriveris there a way to change which wav format MixMonitor uses so I can use the wav format that makes for a smaller file?
20:20.12[TK]D-Fenderbent_screwdriver: It records in precisely the format you specify
20:21.37bent_screwdriver[TK]D-Fender: looked on http://www.voip-info.org/wiki/view/Asterisk+cmd+MixMonitor and didn't see that opotion. Can you give me an example syntax?
20:22.39[TK]D-Fenderbent_screwdriver: the file extension specifies the format
20:24.13ryanlin[TK]D-Fender: are you therE
20:24.13bent_screwdriver[TK]D-Fender: ahhh so to get wav49 do i do .wav or .WAV?
20:24.40bent_screwdriverryanlin: TK's always here as far as I can tell ;)
20:24.53[TK]D-Fenderryanlin: Yes.. now where is the pastebin I asked for over 20 mins ago?
20:25.04[TK]D-Fenderbent_screwdriver: Good for noticing that WAV != wav
20:25.12ryanlin[TK]D-Fender: http://internetworkpro.org/pastebin/6387
20:25.23[TK]D-Fenderbent_screwdriver: Feel free to read up on which is which, or do the 10 second test youself
20:25.40[TK]D-Fenderryanlin: that is NOT your dialplan
20:26.03bent_screwdriver[TK]D-Fender: will do. thanks for the info!
20:26.13ryanlinsorry
20:26.14ryanlinhttp://internetworkpro.org/pastebin/6388
20:26.16ryanlinhere it is
20:26.44[TK]D-Fenderryanlin: WTF is that?
20:26.49ryanlinisn't that the one?
20:26.54[TK]D-Fenderryanlin: That doesn't look like a proper extensions.conf
20:27.00ryanlinahhh
20:27.07ryanlinshit..okay  let me get all the stuff in extensions.conf
20:27.08[TK]D-Fenderdefault => 3258,1,Dial(SIP/$) <-- PARDON?
20:27.22[TK]D-Fenderryanlin: Where did you just pull that mess from?
20:27.30ryanlinin extensions.conf
20:27.34ryanlinit was there
20:27.43ryanlini commented it out
20:27.52*** join/#asterisk boch (~boch@200.61.191.9)
20:27.57[TK]D-Fenderryanlin: I don't see contexts with braces around them, I see a single mashed line with junk in it.
20:28.01[TK]D-FenderRyTry again.
20:29.02ryanlinhmmm..in this case, i did not specify any dial plans for the extension 3300
20:29.06bochHi all, im having a problem, i have a DID point to my asterisk and the exten for that DID is configured OK, the fact is sometimes goes perfect but sometimes the call is rejected because exten was not found. never had a problem like this, what may i be doing wrong ?
20:29.13ryanlinor unless there is a default dial plan
20:29.20[TK]D-Fenderryanlin: No such thing.
20:29.25ryanlinok
20:29.29ryanlinthat explains it
20:29.34[TK]D-Fenderryanlin: And you are reversing things again.
20:29.41[TK]D-FenderAn extension IS dialplan.
20:29.55ryanlinexten => 3300,1,Macro(corpuser,3300)  ;test-phone
20:29.58ryanlinthis is it then ?
20:30.05ryanlinthis is what's in the extensions.conf
20:30.11[TK]D-Fenderryanlin: That is an extension.  The question is.. WHERE is it?
20:30.18ryanlinin extensions.conf
20:30.27[TK]D-FenderWHAT FUCKING **CONTEXT**
20:30.33[TK]D-Fenderbursts a vein
20:30.56ryanlinhmm
20:31.10ryanlinit's probably not defined then
20:31.12[TK]D-Fenderryanlin: pastebin the entire file.
20:31.15ryanlinsure
20:31.22[TK]D-FenderRy"probably"?  Why is this a guess?
20:31.23*** join/#asterisk xheliox (~jeff@i216-58-41-253.cybersurf.com)
20:35.56[TK]D-FenderOk, times up...
20:35.58[TK]D-Fendercheckout time.
20:36.00[TK]D-FenderBBL
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21:17.46ruben23hi there any sofphones used iax2 with g729 codec..?
21:18.41ruben23i would like to used for remote callers
21:23.24*** join/#asterisk AndreBrasil (~staastis@187.65.196.165)
21:25.19leifmadsenruben23: zoiper likely
21:25.20ruben23anyone have idea..?
21:25.29leifmadsenpatience grasshopper
21:25.48ruben23leifmadsen:ok sorry, yeah zoiper free dont have g729
21:25.59leifmadsenruben23: that's because g729 is not a free codec
21:26.17leifmadsenit requires a license to use it
21:26.37leifmadsenyou won't find a free softphone with g.729 support
21:26.54Corydon76-diguntil 2014, when the associated patents expire
21:26.55LemensTSwhere do you get firmware for PAP2T? All I can find is the spc tool
21:27.18ruben23<PROTECTED>
21:27.31leifmadsenruben23: define "best"
21:27.41Corydon76-digruben23: lpc10
21:27.44ruben23currently im suing ulaw/alaw but getting bad connection and call drop
21:27.55Corydon76-digruben23: It's actually even better than g729
21:28.05leifmadsensnickers
21:28.17ruben23<PROTECTED>
21:28.19*** part/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net)
21:28.35leifmadsenruben23: potentially gsm I guess. Quality is not as good though.
21:28.43*** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net)
21:28.52leifmadsenthere is a reason lots of people want to use g.729 instead of a free codec
21:28.58ruben23Corydon76-dig:its added by default on asterisk and most of the softphones right..?
21:29.16leifmadsenruben23: he's yanking your chain -- lpc10 makes you sound like a robot
21:29.20Corydon76-digI believe so, yes
21:29.32Corydon76-digleifmadsen: it's the best codec, though...
21:29.39leifmadsenit certainly makes me laugh :)
21:30.54*** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net)
21:31.21shaderleifmadsen: is the stuff in the section in your book on imap voice mail really all you have to do to get imap working?
21:31.39leifmadsenshader: when we wrote it, yes
21:31.47leifmadsenI haven't done a lot of IMAP stuff and we actually had someone else write that
21:32.05shaderso, there's no need to do any more configuration on dovecot?
21:32.17leifmadsenyou ask a question I do not know the answer to
21:32.23shaderok
21:32.43leifmadsencheck the imap documentation in the doc/ directory of your asterisk source
21:32.49leifmadsenit contains some more up to date information
21:34.07*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:38.34italorossiIt's ok to define twice AUDIOHOOK_INHERIT in this case: Caller dial 7089 and 7089 executes an attended transfer to 7099 ? Example: http://pastebin.com/R0xGv6Tm
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21:41.41scalex000hello guys, I need to handle incoming call, but I have dude using queue.  I don't if I need to dial first and when all represantive is busy to queue a call what is the suggestion
21:42.11leifmadsenGotoIf($[${DIALSTATUS} = BUSY]?send_to_queue)
21:42.31*** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com)
21:42.34leifmadsenif I at all understand what you just said
21:42.55*** join/#asterisk nargon (mike@217.194.139.22)
21:43.03[TK]D-Fenderleifmadsen: Yeah, that's mostly what he said... but likely not what he'll need :)
21:43.06scalex000check this
21:43.11leifmadsen[TK]D-Fender: probably
21:43.15[TK]D-Fenderleifmadsen: Which won't be any fault of yours naturally ;)
21:43.18leifmadsengoes back to doing work
21:44.00*** part/#asterisk Canabinoide (~eu@189.96.54.38)
21:44.12scalex000leifmadsen, http://pastebin.com/8WGCUnnJ
21:44.12nargonanyone know what to do, bv keeps disconnecting me after a few mnts.. i've changed qualify to = 3600.. i'm blocked right now on all the serves.. very annoying
21:45.19*** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net)
21:45.27[TK]D-Fenderscalex000: thats kinda retarded... Starting MixMonitor and immediately hanging up the call...
21:46.16scalex000TK, sorry my mistake
21:46.31scalex000TK, anyway my confusion is over Dial
21:46.47italorossiany comments about this: http://pastebin.com/R0xGv6Tm ?
21:46.53scalex000TK, how asterisk run gotoif
21:47.23[TK]D-Fenderscalex000: Rephrase.  You aren't making any sense?
21:47.32[TK]D-Fenders/?/./
21:47.37scalex000ok
21:47.54nargonitalorossi is line 6 supposed to be n or 1 ?
21:48.12*** join/#asterisk x303 (~X303@97.102.28.28)
21:48.33italorossin, sorry!
21:48.50italorossiIt's ok to define twice AUDIOHOOK_INHERIT in this case: Caller dial 7089 and 7089 executes an attended transfer to 7099 ?
21:49.19scalex000TK, I mean I want before agent pickup the phone playback, active mixmonitor so when all are busy queue
21:50.24scalex000TK, I think I found the answer, I need to dial to Agent not to SIP extension, what do you think? remember my english is poor
21:50.34[TK]D-Fenderscalex000: Just look at the order you are doing things in.  Clearly this isn't doing what you want.  Go change it
21:50.35leifmadsenwe noticed :)
21:50.51scalex000leifmadsen, thx
21:51.06[TK]D-Fenderscalex000: You should be starting Monitor before some other kind of call.
21:51.36scalex000ok
21:51.51*** join/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net)
21:52.41nargoncan anyone comment registering a peer to broadvoice for incoming? the peer registeres somtimes but then goes to unregistered after a couple minutes then the server blocks my ast box for an unkown amount of time... i see posts on this on the net but i can't figure out a fix
21:53.16*** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no)
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21:55.12[TK]D-Fendernargon: you should pastebin the SIP debug of your actual registration attempts...
21:55.16[TK]D-Fender~pb
21:55.17infobot[~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
21:55.56scalex000TK, fix it http://pastebin.com/d4rRwqyB
21:57.00nargon[TK]D-Fender yeah its not really a problem registering its that the broadvoice servers block my registration when I register more than once every 60-120 seconds...  occasionally they will accept my registration and then drop it after a few mnts... so all you would really see would be repeating reg attempts.. its not a problem with the config other than the picky bv registration crap..
21:57.44[TK]D-Fenderscalex000: Fix what?  You haven't expressed what is FAILING there,
21:57.58[TK]D-Fenderscalex000: And your GotoIF is bad.  You have quotes on ONE SIDE only.
21:58.14[TK]D-Fenderscalex000: the left side of your expression will never have quotes in the reulst
21:58.21*** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com)
21:59.15nargon2142421756/2142421756      147.135.8.128               5060     OK (756 ms)  registered now... will see how long it stays that way
21:59.16scalex000TK, yes its terrible
21:59.47nargondon't laugh at my latency i'm on a satellite connection :)
22:00.42[TK]D-Fenderresult*
22:00.43codefreeze-lapLOL.... oops! Sorry!  (murf shuts back up)
22:00.50nargon2142421756/2142421756      147.135.8.128               5060     UNREACHABLE   << yeah its dead now again
22:01.35*** join/#asterisk trelane (~trelane@funtoo/staff/trelane)
22:01.38[TK]D-Fendernargon: that is only your peer status, it has nothing to do with your registration.
22:01.56ManxPowerWhat is funny is that your users apparently tolerate 756 ms of latency.
22:02.01trelaneneed a recommendation on a single port FXO (receiving an inbound POTS line) that registered back to asterisk on the back end
22:02.08ManxPower"sip show registry" is what you want, I think.
22:02.20trelane(registers back via IAX or SIP)
22:02.30[TK]D-Fendertrelane: SPA-3102
22:03.41nargonManxPower yeah show sip registry is fine.. it shows the status... not much i can get from that
22:03.57trelane[TK]D-Fender, config's similar to a linksys phone or a pap-2002?
22:04.14nargontrelane yeah we are in iraq.. actually not that bad.. somtimes i call people and they tell me its the clearest call they have every been on and I sound like i'm right down the street..
22:04.42trelanenargon, cool
22:05.58*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
22:06.05*** join/#asterisk niekie_ (~niek@CAcert/Assurer/niekie)
22:06.05*** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek)
22:06.31[TK]D-Fendertrelane: Similar
22:07.28trelane[TK]D-Fender, many thanks
22:07.33trelanethe FXS on that actually helps as well
22:07.35*** join/#asterisk Mw3_ (mw3@mw3.hu)
22:07.39trelaneoddly useful little device
22:07.57*** join/#asterisk viq_ (~viq@unaffiliated/viq)
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22:09.24*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
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22:21.52*** join/#asterisk bcrisp (~bcrisp@70.102.242.138)
22:22.39bcrispbook
22:23.13bcrispcould someone drop a link to the free * ebook into chat?
22:23.43*** join/#asterisk blaines (~blaines@75-171-72-110.phnx.qwest.net)
22:25.50idespinneryou mean this: http://www.asteriskdocs.org/?
22:26.16bcrispnah, got it but thanks
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22:27.46*** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
22:28.41ManxPower~answers
22:28.42infobot[~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt
22:28.55bcrisp~book
22:28.56infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
22:29.03bcrispok i forgot my tilde
22:29.17nargonhow can i set my registration timeout from 120 to 3600 on my peer to bv ?
22:29.51nargondefaultexpirey=1800
22:29.52ManxPowernargon, remove the change you made to the default of 3600 seconds?
22:29.53nargonmaxexpirey=3600
22:30.00bcrispwhats the default digittimeout?
22:30.07nargonManxPower that was for qualify
22:30.20nargonshould i just set default and max to 3600 ?
22:30.20ManxPowerbcrisp, no idea, but ONLY applies to IVRs in Asterisk, not dialing
22:30.21Corydon76-digbcrisp: 5
22:30.30bcrispCorydon76-dig: thanks
22:30.36ManxPowernargon, no, qualify defaults to something tless than 3600
22:30.56nargonManxPower i mean i put qualify=3600
22:31.03nargonbefore i never modified the registration timeout
22:31.32Corydon76-digqualify defaults to 2000
22:32.07Corydon76-dig3600 tells me you probably think it's doing something other than what you think
22:32.08ManxPowernargon, next time try SAYING "qualify" and not "registration"
22:32.37Corydon76-digqualify ==  number of milliseconds a response needs to arrive before it's considered too late.
22:32.53nargonManxPower i did :(
22:33.02ManxPower<nargon> how can i set my registration timeout from 120 to 3600 on my peer to bv ?
22:33.11ManxPowerThen you started talking about qualify
22:33.31ManxPowerTell us again what you are trying to do.
22:33.33nargonright because you refered to an earlier comment where i was talking about qualify
22:33.48ManxPowernargon, no, I wasn't.
22:33.57nargonok sorry thought you were
22:35.09nargonCorydon76-dig my default reg seems to have this line in the sip debug Expires: 120 i added maxexpirey and defaultexpiry = 3600 let me check debug again
22:35.43ManxPowernargon, I suspect BV is telling you to use 120
22:36.54*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
22:37.16nargonManxPower BV has a problem with you registering to often at 60 or 120 thats why i'm trying to set it to 3600
22:37.40nargonits part of the kind of picky system they use to protect their servers i guess...
22:38.13nargonafter you reg to often they drop all reg attempts for an seemingly random amount of time (this is what i'm getting of google)
22:38.50Corydon76-dignargon: the default expiry for your registration is specified in your 'register' line, right after the '~'
22:38.55nargonhttp://www.voip-info.org/wiki/view/Broadvoice
22:39.15nargonok let me try to look at that
22:39.20ruben23hi i have an asterisk server whihc uses 10 Mbps this is shared betwwen data and voice, and got 13 phone extensions only, problem with calls are sometimes choppy voice and when someone call they cant hear each other.
22:39.37ruben23this asterisk server is public and not in a NAT environment.
22:40.05*** join/#asterisk knot (yiffstar66@unaffiliated/devemo)
22:40.06ruben23si this caused by the data traffic..? but i have 10 Mb bandwidth.
22:40.09Corydon76-dignargon: those other values are for regulating INCOMING registration requests from phones, not OUTGOING
22:41.05nargonCorydon76-dig right now i'm just trying to get the incoming peer to stay registered
22:41.53nargonruben23 you can check your jitter by looking at stdev on a ping -c 100 yourhost.com
22:41.56Corydon76-dignargon: No, if you're dealing with broadvoice, you're working with the OUTGOING peer
22:42.42nargonCorydon76-dig in trixbox I have a user set up for outgoing and a peer set up for incoming
22:43.05Corydon76-dignargon: there's your overarching problem.  IT's trixbox
22:43.12Corydon76-dig~trixbox
22:43.13infoboti heard trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY!
22:43.32Get_The_Fishruben23 what type of connection tech are you on?
22:43.58nargonwell its to late now i use it for our office server.. i use pure asterisk for our production stuff for our customers but I don't want to reprogram the 120 some odd extensions for our office users :(
22:44.08nargonit worked before untill they switched me to a business account
22:45.28nargonaside from the stupid web interface its astrisk underneath so.. it just has some stupid stuff going on with the "UI" right..
22:46.24ruben23Get_The_Fish:what you mean..?
22:46.50Get_The_Fishruben23: t1, fiber, cable modem, etc
22:46.52ruben23nargon: the host is the voip provider switch..?
22:46.54nargoni blame BV rather than  "sh1tb0x" ..
22:47.10ruben23Get_The_Fish:im using coppe
22:47.15ruben23copper i mena
22:47.25nargonruben23 i think your voip provider should not have latency issues i'm talking about your box on the 10Mbps link
22:47.49Get_The_Fishruben23: ok, copper what?  whats the transport?
22:48.04nargonyou could start by making your net connection is solid by pinging your trix from somwhere on the net..
22:48.13nargonmaking sure*
22:48.55nargons/trix/asterisk box/
22:49.03ruben23Get_The_Fish: copper 10/100 Mbps
22:51.14Get_The_Fishso it's ethernet?  Who is the carrier?
22:52.34Get_The_FishDifferent technologies have different characteristics with regards to WAN connectivity.  Some technologies are most burst than others, etc... a data T1 may not have much bandwidth but the underlying technology makes for a relatively jitter free connection.
22:53.08ruben23nargon: --->http://pastebin.com/HTWukkKr
22:53.54ruben23Get_The_Fish:yes ethernet, provider of the line bandwidth or the voip..?
22:54.44Get_The_Fishruben23: provider of the line.  The issue that you describe is related to jitter, or variations in the time between packets arriving at an endpoint.
22:55.18Get_The_FishA connection that has a solid 80ms response will generally sound better than one that oscillates between 20 and 80 rapidly.
22:56.51ruben23<PROTECTED>
22:56.55nargonruben23 the 74. ip is your ast box and you pinged this from where ?
22:57.17ruben23what causes the jitter..? is it my data traffic..?
22:57.46ruben23nargon: i ping the voip carrier switch on US
22:57.55nargon3.735 ms is your stdev/mdev so thats not much what codec are you using ?
22:57.55ruben23im using my asterisk box
22:58.24nargoni suppose that would work f9.. you don't have much jitter really
22:58.48*** join/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net)
22:58.51nargonmaybee your ast server hardware is weak how many calls are you processing at the same time ?
22:59.02LemensTSanyone know where i can get the lastest PAP2T firmware?
22:59.08Get_The_Fishnargon, out of curiosity where did you get that stdev number?
22:59.35nargonGet_The_Fish got it from his pastebin at the bottom of the ping output
22:59.39*** part/#asterisk Scorcerer (scor@czlug.icis.pcz.pl)
22:59.42ruben23nargon: how about this------> http://pastebin.com/XhzkB4cj
23:00.20Get_The_Fishoh he sent me a different pastebin is why... :) just checking
23:01.05Get_The_Fishruben23, there are some hops there that are a little shaky but nothing too bad...
23:01.17ruben23nargon: worst ping is 600 plus ms.
23:01.18Get_The_Fishruben23 who is the itsp?
23:01.26nargoni added minexpirey, maxexpirey, defaultexpirey = 3600 to my peer definition but its still trying to register with a 120 s timeout
23:01.55ruben23Get_The_Fish:im in the philippine sim using bayantel
23:02.52Get_The_Fishruben23, mtr can be a little deceiving, as it is pinging each hope on the route, and most routers are equiped to prioritize traffic... obviously ICMP gets a lower priority
23:03.02*** part/#asterisk devmod_ (~devmod__@c-76-100-208-204.hsd1.md.comcast.net)
23:03.15nargon125.212.40.1 ruben23 is this your box ?
23:03.17Get_The_Fishare you doing conferencing at all?  What is your timing source?
23:03.35nargonruben23 give me your box ip i'll ping it from my nms in germany
23:03.52Get_The_FishI can do the same here from the US
23:05.41riddleboxhaha been fighting a phone not registering for an hour then I finally figured I should check iptables and the ip is blocked lol
23:05.57*** join/#asterisk jks (jks@193.189.93.254)
23:06.01nargonnice..
23:06.07ruben23nargon: my box IP is 125.212.40.6
23:06.42ruben23Get_The_Fish: doing conference call -yes and my timing osurce is ztdummy.
23:07.09Get_The_Fishthat can potentially be an issue, depending on the accuracy of the system clock
23:07.12*** join/#asterisk knot (yiffstar66@unaffiliated/devemo)
23:07.32nargonround-trip min/avg/max/stddev = 390.664/404.720/416.555/5.655 ms with 2% loss.. from germany...
23:07.35Get_The_Fishztdummy or dahdi dummy?  What version of zaptel/dahdi/asterisk
23:08.26nargonlooks horrible compaired to the ping between my germany and US NMSs
23:09.06nargonwhich is round-trip min/avg/max/stddev = 137.457/137.510/137.772/0.080 ms with 0% loss
23:09.14Get_The_Fishyeah mine gets pretty rough right before it hits the pacific
23:10.07nargonwhat codec are you using and will is sustain 2% loss with 6ms of jitter i guess is your question but i'm not the one to answer that..
23:10.56idespinner6ms jitter is fine
23:11.08idespinnerusually below 20 is OK
23:11.19idespinnerlatency is ok until you hit around 50-100
23:11.30*** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no)
23:12.06nargoni run ulaw at 600ms latency on sat links with no real issue other than a slight delay 2% loss though might cause an issue...
23:12.37idespinneraslong as its stable and your buffer is large enough you should be ok
23:12.50idespinnerbut i'd be people notice delay when talking
23:12.57idespinners/be/bet/
23:13.00ruben23<PROTECTED>
23:13.05nargonwhats your proccessor look like ?
23:13.23ruben23im uisng xeon dual core 2.4 Ghz, 4Gb ram
23:13.26*** join/#asterisk x303 (~X303@28.28.102.97.cfl.res.rr.com)
23:13.35nargonruben23 i mean what is the utilization
23:13.47nargontry doing an mpstat -P ALL to get a realtime readout
23:14.56nargonruben23 try ulaw you wil use more bandwidth i think but you might bet better call quality ( i'm know very little off hand about codecs and there tolerance to loss)
23:16.08nargonidespinner yeah i notice the delay.. there is just enough delay to interupt people when talking..
23:18.09Get_The_Fishruben23, the absolute best way to troubleshoot this IMHO is to use tcpdump on the asterisk box, then analyze the results with wireshark... it will give you a clearer picture as to what is going on when you have issues.
23:22.39*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
23:23.52nargoncan anyone tell me how to set the registration expiry for my peer, when i debug i see the sip registration attempts have expirty set to 120 even though i specified min max defaultexpiry = 3600
23:30.28*** join/#asterisk xuser (~xuser@unaffiliated/xuser)
23:31.03nargonalso tried registertimeout=3600 not working
23:34.23*** join/#asterisk glwgoes (~guilherme@189.114.202.44.dynamic.adsl.gvt.net.br)
23:38.22nargondefaultexpiry = 600
23:38.41nargoncan be set in global under sip.conf but that probably sets it for all peers..
23:39.01nargonit workes though
23:45.13*** part/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net)
23:46.07[TK]D-Fendernargon: Of course it has to be under [general], this is not a peer option
23:46.25[TK]D-Fendernargon: PEERS have nothing to do with registering
23:48.54*** join/#asterisk Ta^3 (~tacvbo@189.146.181.250)
23:52.45*** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net)
23:52.58*** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485)
23:54.31*** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk)
23:58.33saisomahey guys, quick question.  I'm getting complaints that the polycom 331s are picking up too much background on my * 1.6.2.7 server.  anyone know a setting off the top of their head for adjustment?
23:59.53*** join/#asterisk dauergast (~sag@188-193-136-54-dynip.superkabel.de)

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