IRC log for #asterisk on 20110613

00:01.39jc319Just learned a new word - 'endpoint' this makes it a tad bit easier for me to distinguish between 'extensions' and the other entity what-I-called-locations-before. places where a person or a group of persons (e.g.7936) can communicate. I can see endpoints everywhere.
00:05.59jc319p3nguin: You said you preferred to have all inbound calls in one single context I understand it would make it easier to manage in the config file editor. However, is it not better (at least the same) to have each sub-account their own inbound context? (because I would like to treat them as individual, separate customers if possible). Obviously I do not know the future impact of this, for
00:05.59jc319example for programmatic access or with advanced (for me) topics such as IVR, voicemail etc. I guess my question is -- Is it beyond doubt that combining all inbound calls into one context is better? [would it not be easier to manage while adding removing clients? or totally forwarding all their calls to somewhere else]
00:10.07WiretapWork_linuxgecko, modules.conf
00:15.24*** join/#asterisk digilink (~digilink@unaffiliated/digilink)
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01:22.08linuxgeckoWiretapWork_:  thanks for the help.. i think i have it working now :)
01:33.55linuxgeckoWiretapWork_:  arrgh..   MOSTLY working..  i can get teh sip to connect,  but i dont' see it trying to call via the google voice. when i conect with sipdroid, it comes back immediately with call failed, even though the peer is registered.  i only see `  == Using SIP RTP CoS mark 5
01:34.58linuxgeckoon the asterisk CLI when i watch it try to dial.   im @ verbosity of 10. what more should i doo to get more details on wwhere the weak link is?
01:35.20WiretapWork_linuxgecko, core set verbose 4
01:35.23WiretapWork_oh, right
01:35.24WiretapWork_okay
01:35.28WiretapWork_in that case, sip set debug on
01:35.35WiretapWork_you need to see what is going on
01:37.31linuxgeckoi think i know what the issue is ( for now)   gtalk show channels shows no channels.
01:42.57p3nguinjc319: It really depends on what you're doing.  You started out needing three phone numbers (one for you, one for someone else, and one for home), but now you're talking about customers.  If I serve "customers," I don't just give them a context, I give them an entire config FILE for extensions (which contains many contexts, because customers often have ridiculous demands).
01:43.27p3nguinjc319: endpoints = devices = phones
01:44.03p3nguinjc319: None of these three is the same as an extension.  Extensions are intangible.
01:47.39*** join/#asterisk Canabinoide (~fred@187.120.134.214)
01:48.50*** join/#asterisk coppice (~chatzilla@m180-219-242-200.smartone-vodafone.com)
01:51.21jc319At the moment I am setting up home office system but would be good to set it up in a way that it is small business ready, so I can just copy the config if I need to deploy in another SOHO
01:52.30jc319I have set up some of the extensions I have noted down 46 so far, but only 6 or so is live and it works good so far - except the multiple lines thing
01:53.02jc319Just noticed that the same problem happens on the PC with softphone. 1st line registers, 2nd does not (401 Unauthorized in CLI)
01:58.37jc319zoiper registers to local Asterisk if nat=yes. if nat=no it does not register. hmm
02:08.38WiretapWork_linuxgecko, channels only show when they are active (in use)
02:09.06WiretapWork_jc319, asterisk handles multiple lines via callwaiting
02:09.41WiretapWork_Cisco phones, at least, handle it quite gracefully
02:09.51WiretapWork_I established 8 calls through one line button on my 7970 to test
02:13.42p3nguinI've done the same, but he's wanting to use two lines for differing things.
02:13.51jc319WiretapWork_: What I am trying to do is to two things actually: 1) outbound part of the desired setup>> person1 can use deskphone or mobile softphone or pcsoftphone to call outbound using VoIP with a caller ID from any of the lines that he has access to (business line/home line/mobile line). to do this p3nguin gave me the config I need (setting OutPeer variable to use the correct channel &
02:13.51jc319using setvar=externalCID=xxx & callerid=yyy variables).
02:14.19WiretapWork_so you essentially want a mangled version of SLA?
02:14.34jc319The other part is inbound>> the same person can accept an inbound call on any of the endpoints (deskphone/pcsoftphone/mobilesoftphone) he has access to.
02:14.59jc319I don't know what it is called. I didn't even know what I wanted when I created the first sip.conf. learning as I go
02:15.06WiretapWork_you're not trying to register the same extension multiple times, but actually multiple lines per phone?
02:15.20WiretapWork_IIRC Ast doesn't support multiply registered extensions
02:15.27jc319thanks to p3nguin made a long way. however I can't still make additional lines to call outbound (yet).
02:15.28p3nguinExtensions don't register.
02:15.28jc319yes
02:15.40jc319I think I create multiple peers and register peers
02:16.03p3nguinThat's what you're supposed to do, anyway.  Create a peer for every different line you want.
02:16.16WiretapWork_fine, swap my use of the term 'extension' for 'peer'
02:19.31jc319here's my first design draft re extensions. http://pastebin.com/FswM9ASc
02:20.00jc3191st one was created when I was reading the asterisk book, chapter 5. 2nd part after talking to p3nguin :D
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02:28.18p3nguinI don't understand it, but I guess that's okay.
02:28.27linuxgeckoanyone else able to connect to google voice using the howto as wwritten??       it seems i'm not getting a connection to my google voice account,   but as yet, have not found enough debugging info as to why.
02:28.57WiretapWork_I'm not sure I understand the googletalk obsession
02:29.50p3nguin<PROTECTED>
02:29.57p3nguinSeems to be working for me.
02:30.24jc319p3nguin: Which part? Multiple peers to set different callerid's for outgoing calls. Having multiple peers also provides the ability to ring the correct subset of devices as required, no?
02:31.26WiretapWork_to be honest it sounds like jc319 wants ringgroups
02:32.03jc319'ring'+'groups' are these groups of peers that you can ring?
02:32.20linuxgeckop3nguin: then i must not have it configured right. how do i double-check?  i copy/pasted the wiki jabber.conf stuff, and the gtalk.conf stuff..  my extensionss.conf anf sip.cong seem to be reading in correctly.
02:32.44linuxgeckop3nguin:  well, obviously, i changed the credentialss in jabber.conf
02:37.31p3nguinjc319: Ring groups of groups of devices which can be dialed together.  You've already created a ring group when you put eight devices inside a Dial().
02:37.53p3nguinRing groups are human concepts, not a mechanism within Asterisk.
02:41.22p3nguinlinuxgecko: I copied and pasted the info from the wiki a few months ago.  I changed the email address and password to be my own.  It simply works for me.  What is the issue that you're experiencing?
02:44.35jc3198 devices in single bracket is not easy to read, today I discovered variables idea. I'll define 2 person ring choice variables and what endpoints they are interested to ring for each of their DID lines, and this way I hope I'll be able to put only one variable into dial application (rather than 8+ sip #s)
02:47.55p3nguinI do it from the AstDB and then I only have to reference a variable that reads the devices.
02:50.51jc319Yes I have noted it, in my long term todo list. I'll try to get the basic setup working first before building anything else though
02:51.06jc319At the moment #1 priority is fixing this additional lines cannot dial issue.
02:51.10jc319Any ideas?
02:51.27jc319(happens with 7960s and also zoiper) so it's a server side issue I guess
02:52.02jc319and this is newest message about this issue >> Found peer '000AB747A418-PB1' for '000AB747A418-PB3' from 10.9.8.71:50259       [Jun 13 02:49:19] WARNING[2230]: chan_sip.c:13698 check_auth: username mismatch, have <000AB747A418-PB1>, digest has <000AB747A418-PB3>
02:55.54WiretapWork_its not handling multiple peers on the same port/ip it seems
02:58.23jc319How to fix? This peer is dynamic, do I still type in port=nnnn in the peer definition?
03:00.42linuxgeckop3nguin:  when i dial +1xxxxxxxxxx  the call fails. it seems to connect to the my asterisk server, but it fails to gonnect to gtalk.  gtalk show channels shows no active  channels, and jabber show connections shows xxxxxxxxxx@gmail.com/Talk Disconnected
03:02.19WiretapWork_jc319, no
03:02.21jc319tried didn't work
03:02.40WiretapWork_linuxgecko, jabber set debug on?
03:02.51linuxgeckoWiretapWork_:  yeah..
03:02.59p3nguinjc319: Like I already told you, you don't need to change the friggin' ports.
03:03.34WiretapWork_p3nguin, you see it not matching the right peer though?
03:03.36*** join/#asterisk radic (~radic@dslb-094-216-248-133.pools.arcor-ip.net)
03:03.50p3nguinI'm tempted to switch my phone back to SIP just so I can give you my actual config that is in use and working.
03:04.19p3nguinSo I guess that's what I'm going to do.
03:04.33jc319p3nguin: OK not touching ports. Found a forum post in google, one guy suggests nat=never would fix this
03:04.53p3nguinI included that in my example that I gave you.
03:04.56jc319look at zoiper log window:
03:04.57jc31904:02:08 Line 1 : port resolved
03:04.57jc31904:02:08 Line 1 : outgoing call to '202'
03:04.57jc31904:02:09 Line 1 : failed (bearercapability notauth)
03:05.12jc319port resolved - (as if there was a problem with it)
03:05.22p3nguinYou should have had insecure=port and nat=no or nat=never in EVERY PEER DEFINITION FOR EVERY LINE ON THE PHONE.
03:09.43jc319Yes it was like that initially. However if I do not have "nat=yes", then zoiper does not register with Asterisk.
03:09.58WiretapWork_then quit using zoiper
03:10.27jc319(and by the way, yes I do have insecure=port for every phone. I also had nat=no until the abovementioned issue)
03:10.37jc319WiretapWork_: it's the same with 2 other softphones
03:10.52WiretapWork_"nat=no" != "nat=never"
03:11.19jc319OK I'll try never... I think I did already, but just to double check
03:12.49jc319insecure=port is still there. tried nat=never it still does not register
03:12.59jc31904:12:40 Timeout registration for '239@10.9.8.4'
03:13.00jc31904:12:44 Timeout registration for '238@10.9.8.4'
03:13.00jc31904:12:54 Timeout registration for '237@10.9.8.4'
03:13.11WiretapWork_timeout?
03:13.14WiretapWork_enable debug
03:14.38jc319that was from client (zoiper). it keeps trying until it gives up. and this is from asterisk >> http://pastebin.com/H4kgaKSg
03:16.30WiretapWork_the debug you sent from asterisk, is that the Zoiper cliehnt?
03:17.02WiretapWork_*looks at UA and realises it is*
03:17.14WiretapWork_well, it looks like it is registering to a public address
03:17.17WiretapWork_of course that will need NAT
03:18.18WiretapWork_this we call "failing to properly explain the situation"
03:18.24p3nguinI certainly wouldn't register my phone to a public address when I'm on the LAN where Asterisk is a local machine.
03:18.40p3nguinI'm switching my phone to SIP right now.
03:18.54WiretapWork_wait
03:18.57WiretapWork_hang on just a tick
03:19.11WiretapWork_your phones are registering to the outside address, while on the same lan as the inside address?
03:19.18WiretapWork_where's my shotgun...
03:19.57jc319oups
03:20.31jc319well, there was one single tiny nat, at the top
03:20.52jc319which happens to have [general] tag on it. it's really too high in the config file
03:21.01jc319so high that normal people wouldn't look
03:21.04jc319thanks :D
03:21.57*** join/#asterisk sourcode (~code@ppp-115-87-209-254.revip4.asianet.co.th)
03:21.59jc319[Jun 13 03:21:30] NOTICE[2230]: chan_sip.c:13740 check_auth: Correct auth, but based on stale nonce received from '"G Home"<sip:239@10.9.8.4;transport=UDP>;tag=6c2ae612'
03:22.03jc319WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01722b6a", stale=true
03:22.58jc3197960 still can't call outbound using programmable buttons. [which is supposed to set caller id and correct outbound channel]
03:23.19p3nguinWell, this is new.  My phone says UNPROVISIONED along the top and Phone Unprovisioned on the bottom.
03:23.41jc319It means that a SIP device is re-using an old authentication challenge. If it still registers and can place calls, there's no problem to worry about. It's just a warning.
03:23.49jc319so restart should fix
03:24.38WiretapWork_p3nguin, TFTP 404?
03:24.46p3nguinI'm about to check.
03:33.55p3nguinIt's so slow.
03:34.19p3nguinFirst I had to remember how to turn on logging in the tftpd.  Now I have to wait for the phone to try pulling files again.
03:34.24WiretapWork_lol
03:34.27WiretapWork_what type of phone?
03:35.22p3nguin7960G
03:36.31jc319Okay I removed the nat=yes in general context, zoiper successfully connected but then dropped the connection in a minute while I was testing with the hardphones. During this process hardphones at no point managed to successfully dial out using additional lines. The 1st button/line works all the time though.
03:36.50WiretapWork_p3nguin, you had it set up with SCCP?
03:37.13p3nguinYes.  I haven't used SIP on my phones in months.
03:37.18WiretapWork_ah
03:37.22WiretapWork_you find SCCP works better?
03:37.26p3nguinmuch
03:37.31WiretapWork_interesting
03:37.32WiretapWork_how so?
03:37.49p3nguinIt has more features, for one.  That's the biggest thing for me.
03:38.08WiretapWork_right, ast1.6 I assume, or have you some secret patch that works for 1.8 with chan_sccp-b?
03:38.21p3nguinI use 1.4
03:38.30WiretapWork_ah
03:39.33p3nguinThis is strange.  The phone pulls the SEP<MAC>.cnf.xml which tells it which firmware to load, then it pulls in the firmware file.  The end.
03:39.42WiretapWork_ah
03:39.51WiretapWork_make sure the capitalisation is correct on the Jar file
03:39.58p3nguinThere's no jar file.
03:40.11WiretapWork_well, the capitalisation on the first file it requests
03:40.23p3nguinIt's fine.
03:40.40WiretapWork_usually it is like 'jar70sip' and the phone requests 'Jar70sip'
03:40.58WiretapWork_I have also seen the same behaviour when the firmware versions are too different
03:41.12WiretapWork_i.e. <8.5(2) to >=8.5(4)
03:41.41p3nguinThis is a 7960, not a 7970.
03:42.13WiretapWork_this also applies to 7911s, and 7906s
03:42.22WiretapWork_that I've seen so far ayway
03:42.24WiretapWork_any*
03:43.20WiretapWork_the 7912 is different, and I had to 123456789*0# it to get itto take it
03:43.24*** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net)
03:43.26p3nguinI guess I can go do it on my 7940.  I used to switch it back and forth from SIP to SCCP a bunch when I was testing out SCCP.  Never had any trouble with that one going back and forth.  On my 7960 I went from MGCP to SIP to SCCP one time that I can remember.
03:44.48p3nguinIf I remember right, I was not able to go directly from MGCP to SCCP and I had to use SIP in between.
03:46.02p3nguinThe "SIP" indicator appears on the top right corner, so it took the firmware.
03:46.19p3nguinBut it remains unprovisioned.  Maybe I can do it manually.
03:49.29WiretapWork_should be able to do it through the UI on those IIRC
03:49.49WiretapWork_wait.... don't those versions look for SEP<MAC>.cnf, not .cnf.xml?
03:50.01p3nguinfixed
03:50.45p3nguinI went into Net Config and set the alternate tftpd address to be ... correct.  :/
03:50.50p3nguinAnd now it works.
03:50.59WiretapWork_lol
03:51.07WiretapWork_your DHCP doesn't hand out your TFTP?
03:51.32p3nguinIt's supposed to.  It always does in every other case.
04:06.09p3nguinThis is like learning how to use SIP on my phone all over again.
04:15.33p3nguinjc319: Do you have XMLDefault.cnf.xml and xmlDefault.CNF.XML in your tftproot?
04:18.20*** join/#asterisk huskydog (~matt@c-98-228-2-139.hsd1.il.comcast.net)
04:18.27huskydoggood evening all
04:18.53jc319p3nguin I think so, 1 sec
04:19.08WiretapWork_its funny how the chan_sccp-b devels still regard 1.8 as 'unstable' asterisk
04:19.22jc319not in the currently active 'tftp' directory.
04:19.39jc319I remember now - I had them originally, then I found out my current firmware does not use it at all. So I removed them
04:20.01jc319Why? If you need the files I might have a copy somewhere in archives
04:21.28jc319BTW I think I am on the right track re additional lines on 7960. According to a post, a potential solution could be "increasing cseq numbers for invite methods". Trying to find out how to do that.
04:21.39huskydogI have a strange issue with csipsimple for android.  The client registers to my asterisk server just fine, but when I try to dial an extension, the "Contact:" field in the sip packet is sending an ip address registered to the US DoD instead of the correct 4G ip address.
04:22.04WiretapWork_lol!
04:22.15huskydogany suggestions on where to look to figure out what the heck is going on?
04:22.21WiretapWork_pastebin?
04:22.30huskydogsure, hang on a sec
04:27.08huskydoghttp://pastebin.com/FrXNqdyL
04:27.43huskydogthe 26.243.8.160 is certainly not my mobil's address
04:29.48huskydogthe strange thing is the same thing happens with both csipsimple and sipdroid
04:30.06WiretapWork_huskydog, why are you sending a wireshark instead of an asterisk trace?
04:30.31WiretapWork_go to whatismyip.com on your phone
04:30.41huskydogyou never specified what kind of pastebin you wanted.
04:30.56WiretapWork_most people use asterisk's internal debugging
04:31.45huskydogmy phones address is not the one being sent in the sip packet.  I've confirmed that several ways.
04:32.23WiretapWork_well, your phone is getting hold of that address somewhere
04:34.43huskydogasterisk sip debug http://pastebin.com/iTjVsGNn
04:35.13*** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee)
04:35.14huskydoghas anyone heard of Tmobile screwing with VoIP packets?
04:35.53*** join/#asterisk luckman212_ (~irc@2001:470:1f07:1225:7c23:92b3:7e50:df74)
04:35.56WiretapWork_its certainly not Asterisk that is to blame
04:36.04huskydogno, I know that
04:36.26huskydogas you can see in the first few lines the 208.54.3.246 is the correct address
04:37.12WiretapWork_IIRC you can force asterisk to ignore the declared address
04:37.15huskydogI just can't for the life of me figure out where the other address is coming from, let alone why it's an address registered to the DoD
04:37.21WiretapWork_and to use the one connecting
04:41.56huskydognat=yes is supposed to ignore the contact uri, but it doesn't appear to be working that way.
04:42.01WiretapWork_I can't remember whether it is 'nat=yes' or 'nat=always'
04:42.04WiretapWork_try always
04:42.13huskydogok
04:42.58*** join/#asterisk irroot (~irroot@196.44.232.251)
04:44.03huskydognope
04:44.27WiretapWork_sounds like somebody is messing with your SIP packets
04:45.51jc319why does this not work?
04:45.58jc319in sip.conf in globals >> G_S_ENDPOINTS_FOR_BUSINESS=SIP/237&SIP/001A6CA3696C-PB1;
04:46.25jc319in dialplan >>> exten => <full no here>,Dial(${G_S_ENDPOINTS_FOR_BUSINESS},30)
04:46.55jc319Jun 13 04:45:05] WARNING[5150]: app_dial.c:1850 dial_exec_full: Dial requires an argument (technology/number)
04:47.13jc319so it does not receieve SIP/237? hmm
04:47.55kaldemarjc319: NoOp or Verbose the variable
04:49.16p3nguinHow about a priority number?  Did you omit it here or on your dial plan?
04:49.20huskydogI thought the FCC was pretty firm in that providers are not to interfere with data traversing the network
04:49.39huskydogI know they fined the hell out of comcast for just that a few years back
04:49.51jc319exten => 442033186544,1,NoOp(${G_S_ENDPOINTS_FOR_BUSINESS})
04:49.51jc319exten => 442033186544,n,Dial(${G_S_ENDPOINTS_FOR_BUSINESS},30)
04:49.51jc319exten => 442033186544,n,Hangup()
04:50.12kaldemarjc319: hmm.. then again, there is no globals in sip.conf. global dialplan variables are configured in extensions.conf.
04:50.12p3nguinAnd what appeared in your debug?
04:50.33p3nguinhmm
04:50.49WiretapWork_p3nguin, all I can suggest is trying via a tethered laptop with a softphone and doing a wireshark on both ends
04:50.57WiretapWork_err
04:50.59WiretapWork_huskydog,
04:51.55huskydogI can try that.
04:52.05jc319<------------>
04:52.05jc319<PROTECTED>
04:52.05jc319<PROTECTED>
04:52.05jc319[Jun 13 04:51:04] WARNING[5162]: app_dial.c:1850 dial_exec_full: Dial requires an argument (technology/number)
04:52.11p3nguinNow I'm really getting irritated.  I made some changes in the files on the tftpd and the phone never seems to request the files anymore.
04:52.26huskydogDo you think using tls will foil their packet rewriting attempts if in fact that's what's happening?
04:52.37*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-tllffiearjgrjszd)
04:52.40p3nguinNoOp("SIP/voipms_main-00000078", "")    <-- this says there is NO DATA in your NoOp().
04:52.48p3nguinNo data.
04:52.55p3nguinThat means your variable is null.
04:53.09jc319<jc319> in sip.conf in globals >> G_S_ENDPOINTS_FOR_BUSINESS=SIP/237&SIP/001A6CA3696C-PB1;
04:53.11p3nguinIt's null because of what kaldemar said.
04:53.17p3nguinThere is no globals section in sip.conf.
04:53.17jc319this is how it's in the book
04:53.19jc319hmm
04:53.21jc319there is
04:53.29p3nguinglobals go in your extensions.conf.
04:53.39jc319ouch
04:54.24p3nguinThere is not to be a globals section in sip.conf.  Delete it and anything you put within it.
04:55.06jc319cool it worked, thank you both
04:56.13jc319End of day report, outbound works from 1st line (programmable) button, not from others. The same with softphones. 1 host = 1 'active' authorization. Asterisk just ignores other registrations from the same host.
04:56.21huskydoghmm, according to several faqs I've found, Tmobile is not restricting sip over their data network.
04:56.34jc319inbound works too, with my new variables skill I'll write all inbound rules tomorrow
04:58.28p3nguinI've managed to jack up my phone with SIP.  It's not even trying to register, and now I can't get it to pull the confs from the tftpd again.
04:58.30WiretapWork_huskydog, see my debug recommendations for diagnosis
04:58.57WiretapWork_p3nguin, looks like you need to do a factory reset
04:59.19p3nguinI did the reset from in the menu.  Is there another way?
04:59.23WiretapWork_yep
04:59.28WiretapWork_when you plug the phone in
04:59.32WiretapWork_one light will light up
04:59.36WiretapWork_when it does hits
04:59.40WiretapWork_this*
04:59.43WiretapWork_hold the '#' key
05:00.00WiretapWork_it will, after a while, start either flashing the MWI lamp, or cycling through the line key lights
05:00.10WiretapWork_when it does this, enter 123456789*0#
05:00.16WiretapWork_this will completely erase the phone
05:00.41WiretapWork_and it will use TFTP as assigned by DHCP to pick up a firmware from 'term40.default.loads' (or in this case, term60
05:00.42p3nguinThis works when the phone already has SIP on it?  I know certain key sequences don't work on SCCP that do on SIP and vice versa.
05:01.01WiretapWork_this works for all cisco phones with all firmwares
05:01.05WiretapWork_even no firmwares
05:01.12p3nguinI don't have any term*.default.loads
05:01.12WiretapWork_as it is a bootloader-level control
05:01.28WiretapWork_you need to unzip an entire firmware package to your tftproot
05:01.34WiretapWork_it will contain these files
05:01.35p3nguinI did.
05:01.47WiretapWork_in that case, your phone will probably be looking for something else I'm not used to seeing
05:01.52p3nguinI use the sip 8.11 files.
05:01.58WiretapWork_not relevant
05:02.10WiretapWork_the phone looks for a file in this case based on its model
05:02.15WiretapWork_not on its firmware
05:02.15p3nguinOkay, good.
05:02.24p3nguinThen it'll be looking for the files I have.
05:02.28WiretapWork_yep
05:02.43p3nguinThe term* files I have seen on newer 7900 series phones, such as 7965.
05:02.46WiretapWork_watch the logs to see what files it requests, though it should show on the screen
05:02.54WiretapWork_p3nguin, they're on all phone that support CUCM
05:03.26p3nguinEvery time you mention a file that isn't the files I use, I stop myself from doing the reset.
05:04.38p3nguinI don't think I've ever had to perform a reset in this manner before.  I've also never had a phone refuse to access the tftpd after it already pulled the configs from it several times.
05:06.14jc319good luck with the phone p3nguin. see you later
05:06.59p3nguinAll this trouble just to try helping out someone else in need.
05:07.02irrootp3nguin morning slap the phone :P
05:07.14p3nguinI'll just go back to SCCP and call it a night.
05:07.14irrootyou rock
05:08.36p3nguinI switched my 7960G to SIP to try to help jc319 with his problem.  My phone is now being an asshole and not reading configs from the tftpd.  Meanwhile, he got at least one line on his phone working and was able to make a call, and is done for the day.
05:09.53irrootim just starting 7am and im off have a good down time
05:12.02p3nguinwiretapwork_: Do you have any idea if it is possible in SCCP to use config files to set ringtones on the lines?  I'd rather do it from files instead of the phone menu.
05:12.18WiretapWork_it is always possible
05:12.29WiretapWork_ringlist.txt or ringlist.xml depending on what the phone requests
05:13.14p3nguinWhat setting do I use to assign the ring to a line?
05:13.23WiretapWork_you do that through the phone menu
05:13.29p3nguin*sigh*
05:13.33WiretapWork_but you can tell it what rings are available via TFTP
05:17.13*** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za)
05:18.21p3nguinOkay, I did the factory reset.  It seems familiar, so I guess I've had to do it on some phone before.
05:22.11*** part/#asterisk awclin (~alinford@g0962184.demon.co.uk)
05:22.18p3nguinBut now it still does not want to pull files from the tftpd.
05:28.54p3nguinAlright, this is really pissing me off.  Fucking fail2ban banned my phone, which is on a local IP address, which is set in ignoreip in jail.conf.
05:29.21p3nguin# "ignoreip" can be an IP address, a CIDR mask or a DNS host. Fail2ban will not
05:29.25p3nguin# ban a host which matches an address in this list.
05:29.27p3nguinignoreip = 127.0.0.1 192.168.0.0/16
05:32.12*** join/#asterisk The_REV (~The_REV@76-217-185-124.lightspeed.frsnca.sbcglobal.net)
05:32.20p3nguinThat's enough to piss me off.
05:33.07The_REVHello room... First time in in a long while...
05:33.59p3nguinI whitelisted the network range in iptables rather than allowing everything that fail2ban isn't DROPping, and now the lines registered and everything looks normal.
05:35.09irrootyeah nothing like firewalling yourself to waste time ,....
05:35.51irrootfrustrating ^n
05:36.51p3nguinI never even THOUGHT that it would happen because I explicitly configured 192.168.0.0/16 in ignoreip.
05:36.57p3nguinIs that a fail2ban bug?
05:38.24p3nguinI'm using 0.8.4
05:42.59*** join/#asterisk jkroon (~jkroon@dsl-242-10-219.telkomadsl.co.za)
05:49.26*** join/#asterisk Juggie (~Juggie@unaffiliated/juggie)
05:51.33ChannelZmaybe, it was a bug in older versions where only the first one was being picked up
05:52.09ChannelZperhaps it's come back
05:52.19p3nguinThis appears to be the newest version.
05:52.22p3nguinAt least it's the newest version in my repo.
05:52.35ChannelZprobably, I think it hasn't been updated in years
05:53.38p3nguinWell, it sure needs to be.  This is a pretty fucked up bug.  I only thought of a firewall problem after I exhausted every other reasonable possibility.
05:54.17ChannelZseems like it's working here though..
05:56.11p3nguinIt sure jacked my phone.
05:56.49ChannelZwas this a recent change to your config or has that been in there for a long time?
05:57.21p3nguinIt has been there for as long as I can remember.  I copied the config over from the old system when I migrated.
05:58.08p3nguinSo I guess it has been years since I put it there.
05:58.21ChannelZHmm.
05:59.51p3nguinI don't know if there simply has never been that much traffic from one phone to where it gets banned, or if there's some other problem.
05:59.52ChannelZIs it in [DEFAULT]?
06:00.26p3nguinyes
06:00.32p3nguinJust a few lines from the top of the file.
06:00.58ChannelZHmph.  Dunno.  I just tried exactly the same thing and it seems to be workin
06:01.18*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
06:01.19p3nguinNow see if you can get your phone to hammer away to get fail2ban to ban it.
06:01.28p3nguintftpd, asterisk, and httpd
06:02.02ChannelZI'm not using it for SIP (the whole thing is firewalled off) but I use it for email, and I did test it locally and from work.. banned me properly from work but didn't make a peep for me locally.
06:02.29p3nguinEvery time I would reset the phone it would try to load files from tftpd, load the background image off the web server, and register to Asterisk.  I figure doing that time after time eventually pissed off fail2ban.
06:02.30*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
06:03.20p3nguinBut because I knew the local net was defined in ignoreip, I never bothered to check the firewall.
06:03.52ChannelZDo you know which service caused it?
06:04.10p3nguinI don't know if that gets logged or not, but I'll take a look.
06:04.13ChannelZI mean, shouldn't the fail2ban rule be triggered by a failure of some sort?
06:04.17ChannelZ/var/log/fail2ban.log here
06:04.47irrootthinks having a RETURN/ACCEPT ahead of f2b in iptables could be a good idea
06:06.07p3nguinHi,\n
06:06.07p3nguinThe IP 192.168.192.69 has just been banned by Fail2Ban after
06:06.07p3nguin6 attempts against ASTERISK.\n\n
06:07.29p3nguinI've got to fix the emailing of those alerts.  It tried to mail it to me, but sendmail -f ... seems to have failed.
06:08.54*** join/#asterisk SkoZombie (~quassel@60-240-15-173.tpgi.com.au)
06:11.29p3nguinIs there a single point of configuration for emailing from within fail2ban?  I don't use sendmail.
06:17.49irrootwill work for bitcoin
06:20.13p3nguineh
06:20.28p3nguinHow odd.
06:21.41*** join/#asterisk MarKsaitis (~MarKsaiti@cpc7-pool12-2-0-cust219.15-1.cable.virginmedia.com)
06:27.26ChannelZwhat mailer do you use?
06:28.40p3nguinmsmtp
06:29.24p3nguinThe only thing that really ever sends mails from this host is asterisk, so I use msmtp to relay through a gmail account.
06:29.35ChannelZI think it mimics the basic arguments to sendmail, you should be able to just symlink it
06:30.25p3nguinMaybe that's the problem.  There's no sendmail at all.  A symlink could do the trick.
06:30.56ChannelZmost other mailers will put symlinks to 'sendmail' in point at themselves ('mail' too)
06:31.01p3nguinln -s `which msmtp` /usr/sbin/sendmail
06:31.28irrootor ln -s /bin/true /usr/sbin/sendmail
06:31.41p3nguinHow is that going to help?
06:32.15*** join/#asterisk Buklov (~Buklov@mail.sapsun.su)
06:33.38p3nguinI just need to wait for something to trigger fail2ban and I'll see if it can mail out.
06:34.10*** join/#asterisk The_REV (~The_REV@76-217-185-124.lightspeed.frsnca.sbcglobal.net)
06:35.03p3nguinSometimes I go more than a week without a banning.
06:36.11*** join/#asterisk sgimeno (~sgimeno@163.117.206.10)
06:36.39ChannelZjust cat >> some bogus log entries onto one of your logs it monitors :)
06:37.08p3nguinInteresting idea.  I never even thought of that.
06:37.30p3nguinBut I restarted fail2ban and got an email, so I don't have to do that.
06:37.52ChannelZyay!
06:38.00p3nguinHi, The jail ASTERISK has been started successfully.
06:38.10p3nguinRegards,
06:38.11p3nguinFail2Ban
06:38.28p3nguinThat's kind of lame.
06:39.18ChannelZYou can customize it.. or turn it off
06:39.38p3nguinI'll have to.  I really only care about when I ban myself.
06:39.57ChannelZmail.conf in actions.d
06:40.37p3nguinThat was an easy solution to not having sendmail, though.  I'm glad it didn't require changing all the fail2ban crap.
06:42.40*** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap)
06:43.35p3nguinI hate to repeat, but I've got to tell wiretapmac what the hell went wrong.
06:43.44*** join/#asterisk MariusAgon (~aa@89.249.83.26)
06:43.46WiretapMacI did see
06:43.47WiretapMacdon't worry
06:43.51p3nguinoh
06:43.55WiretapMacyou'll note WiretapSeven is here
06:43.58WiretapMacmy home desktop :P
06:44.07p3nguinI didn't know you were present.
06:44.15WiretapMacI have a lot of machines
06:44.26p3nguinYou sit at all of them?
06:44.36WiretapMacif there isn't one present, wait 24 hours and call the emergency services if I'm not online :P
06:44.42p3nguinhaha
06:45.05WiretapMacfail2ban strikes me as very hacky
06:45.21p3nguinThat's because it's very hacky.
06:45.26WiretapMacI set my ACLs manually
06:45.40WiretapMacrestricting traffic to networks I expect it from
06:45.50p3nguinI only use fail2ban for active response.  In this case, the bug bit me.
06:48.17sxpertafter all, it's a crappy perl script reading logs and setting iptables....
06:48.33sxpertyou may get bitten ;)
06:49.34ChannelZit's a crappy Python script
06:49.46WiretapMaclulz
06:50.33irrootmore reason to get bitten !!! dangerous things pythons
06:54.36ChannelZwhat the...
07:02.59*** join/#asterisk ltd (~z@zx.io)
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07:12.25zknhi
07:12.51*** join/#asterisk Buklov (~Buklov@mail.sapsun.su)
07:13.00zknwhat was the CLI command to check what timing source is available/configured for Asterisk?
07:13.42*** join/#asterisk timahvo1 (~rogue@41.223.57.77)
07:13.51zknmy TAB key stops working soon, i just cannot seem to find it
07:18.09WiretapSevenI see only 'timing test'
07:21.48zknyep, but for sure there was a command that listed timing drivers known for Asrerisk and showed which was activated
07:24.18*** join/#asterisk mandla (~mandla@168.167.180.161)
07:24.39*** join/#asterisk Sertys (~sertys@vps121.webconnect.bg)
07:29.54zknmodule show like rest_timing_
07:30.02zknres*
07:31.23*** join/#asterisk cneb3000 (~Ben@87.127.15.145)
07:39.12mandlahello, where can i get an already made extensions.conf file?
07:39.27mandlaFor Asterisk with Astribank
07:50.09kaldemarmandla: you probably won't find one that will suit your purposes, and finding one may be a bigger effort than writing one yourself.
07:50.19*** join/#asterisk timahvo1 (~rogue@41.223.57.78)
07:50.34*** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za)
07:50.56mandlakaldemar, im new to this technology, and i need to get things done ASAP.
07:51.14*** join/#asterisk awclin (~alinford@static-93.158.79.70.got.public.icomera.com)
07:52.02cneb3000mandla: you may not feel like it's helpful. but you could honestly get something done within an hour without much experience by reading the right chapter in asterisk:The definitive guide
07:52.07cneb3000~thebook
07:52.07infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook
07:52.11doolittleworkneed some help someone got my sip_account details and was making calls of one of my servers, i 100% sure they did not log on to my server to get the sip_provider details, how do I prove this to my server provider?
07:53.09cneb3000doolittlework: what exactly are you trying to prove to your service provider?
07:53.31doolittleworkthere is no log of cdr's of the calls on my server
07:53.51doolittleworkso the calls did not originate of my server
07:55.06doolittleworkprovider said they will refund me if i can proof the hackers did not get access to my server
07:55.26cneb3000i think asterisk has the ability to PASS THROUGH calls, although i'm admitedly not to familiar with that
07:56.25doolittleworkbut on the provider billing engin the registered account has different ip address
07:56.37doolittleworkno my ip
07:56.45doolittleworksorry not my ip address
07:57.19*** join/#asterisk tamiel (~tamiel@213.30.183.226)
07:57.56mandlacneb3000, this book is been sold...
07:58.01cneb3000could it be they just used your account from a provider perspective?
07:58.19mandlacneb3000, i thought it was for free.
07:58.44doolittleworkhow do i proof this
07:58.59cneb3000mandla: http://ofps.oreilly.com/titles/9780596517342/
07:59.36cneb3000doolittlework: if you cant find the calls in the CDRs maybe there's something in your logs?
08:00.08tzafrir_laptopmandla, a basic thing is really simple. But it really needs to be a part of your PBX's dialplan
08:00.23cneb3000go to --> /var/log/asterisk --> find a file from the time of the supposed calls --> look for a reference to the numbers?
08:00.45doolittleworkthat is just it there is no details of this in the logs, the hacker linked straight to service provider
08:00.59cneb3000oh. well thats not fair...!
08:01.01tzafrir_laptope.g. I use a quick and dirty    exten => _4XXX,1,Dial(DAHDI/${EXTEN:1})
08:01.16cneb3000as in, not fair for them to ask YOU to prove it :/
08:01.28cneb3000you should ask them to prove it was you ¬_¬
08:01.33tzafrir_laptopto dial to analog phones. This works. But, well, it lacks one or two features you may need
08:01.42doolittleworklol wish it was that easy
08:01.52cneb3000who is the provider, if you dont mind me asking?
08:01.58doolittleworkclearvoice
08:02.04cneb3000..one to avoid? haha
08:02.10cneb3000i bet they're happy to get all that extra money
08:02.14kaldemarmandla: first step is to define what you want your dialplan to do. until then, you're hard to help.
08:02.36doolittleworki also think so, two of my accounts over the weekend, what is the odds
08:03.44cneb3000so some calls were made, they admit it wasn't your IP, and you say you cant find anything pass through your system. they wont let you access the logs or anything? you cant really do anything..!
08:03.54cneb3000your word against theres
08:03.58cneb3000theirs*
08:04.11doolittleworkmandla: you from south africa
08:04.24doolittleworkneed to cahnge providers
08:11.06*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
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08:27.11mandladoolittlework, nope, Botswana, why?
08:27.32*** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com)
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08:30.58mandlacneb3000, not so keen in reading.
08:31.20*** join/#asterisk lost_soul (shawn@cpe-67-249-130-106.twcny.res.rr.com)
08:32.24cneb3000well, you can give me £100 and i'll do it for you..? ;)
08:33.17cneb3000mandla: have you thought about using some of the free asterisk systems? the ones that come with easy to modify GUIs
08:33.41cneb3000i havn't had experience with them all, but there's stuff like pbxinaflash, asteriskNOW etc.
08:33.50mandlaYah um using it, AsteriskNow.
08:34.06cneb3000and what exactly are you trying to do?
08:34.37cneb3000there's probably a way of doing it in asterisknow without touch extensions.conf directly, thats sort of the point :)
08:34.43mandlaIm trying to make my asterisk system make call to the outside world.
08:35.53mandlacneb3000, for now everything is set internally, iv connected a line from the PSTN to the Astribank now i want to make call to the outside world
08:36.49cneb3000ok, and what have you done so far on the asterisk box?
08:36.57cneb3000in the software, not the hardware
08:37.07*** join/#asterisk engrxyz (~puitpyitr@212.23.51.7)
08:42.23tzafrir_laptopmandla, just like with any DAHDI. Read that book
08:42.41tzafrir_laptope.g.: DAHDI/g0/123456
08:43.28mandlatzafrir_laptop, what do you mean?
08:45.21tzafrir_laptopIf you use the FreePBX that came with AsteriskNow: set up a trunk, and outbound route, and such
08:46.25tzafrir_laptopIf not, you need to have it otherwise in your dialplan.
08:46.32tzafrir_laptopIt can be something as simple as:
08:47.03tzafrir_laptopexten => _9.,1,Dial(DAHDI/g0/${EXTEN:1})
08:47.47tzafrir_laptopWhich assumes that the dialed number is all digits, and that all the DAHDI channels you use for the trunk are in group 0
08:48.08tzafrir_laptop(group=0 , maybe with some other numbers, in chan_dahdi.conf)
08:48.36kaldemaror just ask freepbx help in #freepbx.
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09:00.59cneb3000-afknick /cneb3000
09:01.03cneb3000-afk>_<
09:01.25cneb3000working with avaya ip office 500 today. feel like i'm working with the devil.
09:02.28cneb3000and the mitel 3300. mu ahahah
09:02.36*** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16)
09:02.38*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-oaivzautlcaucxxj)
09:02.48aiksa[LV]Hi everyone. I am a bit stuck.
09:03.04aiksa[LV]with using the latest libpri, asterisk and dahdi
09:03.13*** join/#asterisk davlefou (~david@41.225.9.81)
09:03.37aiksa[LV]upon loading the tor card in as terisk I get "Unable to get span status: Inappropriate ioctl for device"
09:03.58aiksa[LV]quick google search pointed to to inconsistencies between various dahdi and astrisk versions
09:04.16aiksa[LV]however I have the latest (as in webiste releases) of both
09:04.35aiksa[LV]cat /proc/dahdi/* shows the spans and channels like it should under normal circumstances
09:04.46aiksa[LV]nothing fancy about dahdi-channels as well
09:05.36aiksa[LV]asterisk 1.8.4.2 and dahdi 2.4.1.2 2.4.1.
09:06.04aiksa[LV]the card: Tormenta 2 (PCI) Quad E1 Card
09:06.26aiksa[LV]any ideas?
09:07.25*** join/#asterisk irroot (~irroot@dsl-185-122-97.dynamic.wa.co.za)
09:08.09*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
09:11.44aiksa[LV]on a side note - is there a way to easily check loaded dahdi kernel and userspace versions?
09:12.00Chainsawaiksa[LV]: dahdi show version
09:12.08Chainsawaiksa[LV]: On the asterisk console will get you the userspace version.
09:13.04aiksa[LV]:) provided that dahdi is laoded
09:13.09Chainsawaiksa[LV]: dmesg | grep -i dahdi | grep -i version
09:13.22Chainsawaiksa[LV]: Well yes, it would have to be loaded for both the commands that I just suggested.
09:13.23aiksa[LV]Chainsaw: thanks. already found a line in syslog
09:13.36*** join/#asterisk timahvo1 (~rogue@41.223.57.77)
09:15.25aiksa[LV]hmmm
09:15.49aiksa[LV]strange, installed dahdi by make & make install. but still in syslog I see dahdi: Version: 2.2.1
09:16.05aiksa[LV]any method to remove prvious instalation completely
09:16.07aiksa[LV]...?
09:19.59Chainsawaiksa[LV]: I suspect one of your DAHDI installs is in prefix /usr, and the other in /usr/local
09:20.08aiksa[LV]yup exactly the case
09:20.08*** join/#asterisk sourcode (~code@ppp-115-87-209-254.revip4.asianet.co.th)
09:20.09aiksa[LV]...
09:20.29aiksa[LV]somebody had used the *magic* of debian packages before me
09:20.34aiksa[LV]ARGH!
09:21.05*** join/#asterisk timahvo1 (~rogue@41.223.57.72)
09:22.09aiksa[LV]Chainsaw: many thanks
09:22.20Chainsawaiksa[LV]: You're welcome :)
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10:08.56tzafrir_laptopRegarding aiksa[LV]'s question: the driver is basically tor2 . Though I hope it actually works. Bug reports are welcomed
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10:39.07*** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net)
10:39.15LantiziaHey is the use of the digits directory deprecated?
10:41.15ChainsawLantizia: No, it is not. But the layout of the languages in the sounds directory has changed.
10:49.08LantiziaChainsaw, if you have the time, may I expand on the larger issue I'm facing?
10:49.22ChainsawLantizia: Certainly.
10:49.35ChainsawLantizia: But basically, older versions had sounds/digits/en_GB
10:49.41ChainsawLantizia: The new convention is sounds/en_GB/digits
10:52.52LantiziaChainsaw, ok well I use the debian packaged version of asterisk that tzafrir maintains... and basically /usr/share/asterisk/sounds is where their kept... but directories like en and en_GB are symlinks to the debian sound package you have installed
10:53.27ChainsawLantizia: Okay.
10:53.31LantiziaChainsaw, I'm installing something (a2billing) that has it's own sounds... normally destined for the "en" directory... but obviously I'd be contaminating the debian sound package it's symlinked to
10:54.05LantiziaI do however... have a /usr/share/asterisk/sounds/custom   which goes to   /usr/local/share/asterisk/sounds
10:54.08ChainsawLantizia: Unless you overwrote files, I don't see how that would upset the package manager.
10:54.14*** join/#asterisk digilink (~digilink@unaffiliated/digilink)
10:54.15Lantiziaso I'm thinking stick my "en" directory in there
10:54.16ChainsawLantizia: Just adding should not be an issue.
10:54.46LantiziaChainsaw, ah but if the sound package was removed and replaced with another language - they'd vanish
10:55.08ChainsawLantizia: Your package manager would remove a non-empty directory? That seems broken behaviour.
10:55.20LantiziaI guess what I'm wondering is...... when referencing a sound like    en/hello.gsm    does asterisk check sounds/   _and_ also sounds/custom/
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10:55.46ChainsawLantizia: You'd have to ask others, sorry. I use just the sounds directory.
10:55.59Lantiziaok
10:57.22catphishcould anyone help me with this error? "Could not locate a FAX technology module with capabilities", I have compiled res_fax and the new res_fax_spandsp
11:00.08irrootwhat you trying to do catphish ??
11:00.43catphishexten => 12345,1,ReceiveFAX(/tmp/fax1)
11:00.49irrootthe capabilities are audio or T.38
11:00.56irrootand soon gateway
11:01.05irrootplease pastebin the bits
11:01.08catphishah that explains why receivefax is failing
11:01.09irrootalso version ??
11:01.13catphishsorry one moment
11:01.25catphishthe actual error is: [Jun 13 11:50:11] ERROR[24176]: res_fax.c:759 fax_session_reserve: Could not locate a FAX technology module with capabilities (RECEIVE)
11:02.18irrootyou sure res_fax_spandsp is loaded ??
11:03.50catphishno
11:04.24catphishand i'm running 1.8.3.2
11:04.25irrootres_fax.c is the generic bits
11:04.40irrootres_fax_spandsp is the technology
11:04.47catphishyes, that's obviously loading, but i'm not sure res_fax_spandsp is
11:05.39catphishthough i do have both load => res_fax_spandsp.so and autoload=yes
11:06.33irroot"module show like fax
11:06.35irrootModule                         Description                              Use Count
11:06.37irrootres_fax.so                     Generic FAX Applications                 1
11:06.39irrootres_fax_spandsp.so             Spandsp G.711 and T.38 FAX Technologies  0
11:07.01catphishyou're right, it's not loaded
11:07.24catphish[Jun 13 12:07:11] WARNING[24379]: loader.c:387 load_dynamic_module: Error loading module 'res_fax_spandsp.so': libspandsp.so.2: cannot open shared object file: No such file or directory
11:07.52catphishi'll try a ldreload
11:08.04catphishthat fixed it, thanks!
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11:38.50irrootneed a sanity check here [aka bit lazy to go find out self] app_queue cdr records
11:39.45irrootlooks like a cdr is generated and logged in queue_log but this does not tie up to cdr sometimes
11:41.53irrootthink forking the CDR before going into app_queue ??
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12:07.38jc319p3nguin: Hello, here?
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12:19.50cneb3000tell ya. sometimes there's nothing better than a cheese sandwich.
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12:51.27skrustyhmm, anyone know why the version of confbridge in 1.10 seems far better (from reading the docs) than the one in 1.8 (or 1.6) ?
12:51.42skrustyis this because it's being developed on a different branch or something?
12:52.40Chainsawskrusty: Yes, new features are developed in trunk first.
12:52.50Chainsawskrusty: And then they make it into the next release, rather than the current release.
12:53.25Chainsawskrusty: The idea is that development effort is then spent stabilising what's already there, rather than bolting on new features (which is cool & glamourous).
12:54.04skrustyok, understood
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13:07.42irrootok found my problem
13:08.16irroota call into a call transfered out then attend tranfer back in will not have a CDR for 2 entry
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13:23.07Sertysanyone here using a2billing?
13:23.24Sertysi tweaked it to work with 1.6.x and chan_mobile as a trunk
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13:24.05Sertysbut i'm stuck right now with balance checks on avail sim cards
13:29.48Kattyhi
13:36.20irrootkatty my CDR's in app_queue you fix it please
13:39.21Katty:<
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13:48.22irrootputnopvut dude just the expert i need to harass :P re app_queue update_cdr
13:48.49putnopvutirroot: oh dear, perhaps I can help
13:48.50irrootwhen a call is attended transfer to it the cdr is not nice im using ast_cdr_init
13:49.10irrootnow at it works the new cdr is fine
13:49.52irrootcall comes into queue gets transferd out and then att transfered back there is no cdr
13:49.53putnopvutpulling up code
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13:50.18irrootpulling out hair at same time
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13:50.55putnopvutHeh, the update_cdr option is something I've never seen before, to be honest. I'm reading aobut what it does right now.
13:50.57irrootjust after the "CONNECT"
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13:52.00irrootwithout adding ast_cdr_init(qe->chan->cdr, qe->chan);
13:52.11irrootit does not write out the CDR
13:53.24mha_Need to install asterisk for home use and for experiments, do I need a dedicated box?
13:53.48pabelangermha_: no, you can setup a VM
13:54.06putnopvutirroot: Well my knowledge of CDRs is not enough that I know if that is a safe thing to be doing there, to be honest. When a channel is created, ast_cdr_init() is called. I would suspect that calling ast_cdr_init() again after the call has connected could result in data on the channel's CDR to be lost.
13:54.11pabelangerunless you want to use actual hardware. EG: fxo / fxs
13:54.36irrootthat is indeed where i am @ putnopvut
13:54.39mha_pabelanger: on my fedora for example? I may use FXO/FXS later on
13:54.52irrootim looking at a option to Queue to fork a CDR
13:55.14irrootadd a new CDR onto the tail and init it
13:55.26irrootif the last CDR on the stack is not right
13:55.44pabelangermha_: under a VM, no. But a desktop, yes.  Just make sure you compile and install DAHDI first
13:56.11irrootie if qe->chan->uniqueid != cdr->(linkedid|uniqueid)
13:57.03mha_pabelanger: is it ok for newbie questions here?
13:57.15pabelangermha_: yes
13:57.18pabelanger~ask
13:57.18infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
13:57.21putnopvutirroot: my knowledge of CDRs is bad enough that I don't know if that logic makes sense or not.
13:57.47irrootputnopvut thx for the input its appreciated
13:58.28irrootill put a patch / review together
13:58.33putnopvutokay cool
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14:01.22mha_I have fedora15 on desktop (can be replaced by any other distro if necessary), and want to install Asterisk, is this OK?
14:01.35irrootsounds good mha_
14:01.45irrootwhat hardware you want to use
14:02.04irrootTDM hardware
14:02.04mha_no specific hardware now, just the soft clients
14:02.18irrootok then go for it
14:02.48irrootmha_ the "tricky" bits is to make sure you have drivers for tdm hardware
14:03.04mha_hardware later on maybe FXO/FXS, but I don think it is available here in Bahrain (persina gulf)
14:03.22mha_irroot: understood
14:04.55mha_persina=persian
14:05.18Canabinoideis gone.. autoaway after 15 min ..[cyp(l/on.p/on)]
14:06.20pabelangerCanabinoide: please disable your auto away message
14:09.06mha_is there a single groupinstall thing to install asterisk package?
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14:14.38pabelanger~packages
14:14.43*** join/#asterisk blop (~blop@217.112.179.12)
14:14.45blophello :)
14:15.14pabelangerinfobot: packages is  Asterisk is available for automated binary installation using APT for Debian and Ubuntu or YUM utility on CentOS 5 Linux and for RedHat Enterprise Linux 5: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
14:15.14infobot...but packages is already something else...
14:15.20pabelanger~packages
14:15.48pabelangerinfobot: packages are Asterisk is available for automated binary installation using APT for Debian and Ubuntu or YUM utility on CentOS 5 Linux and for RedHat Enterprise Linux 5: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
14:15.48infobot...but packages is already something else...
14:15.49blopcould someone fix the WCTDM driver for old tdm400 cards ? i have the same issue as : http://asteriskfaqs.org/2011/02/21/asterisk-users/problem-installing-fxs-module-in-old-digium-4-channel-tdm-card.html
14:16.10pabelangerinfobot: what is packages
14:16.18pabelangerinfobot: help
14:17.21pabelangerinfobot: tell me about packages
14:18.41pabelangerinfobot: asterisk-packages is Asterisk is available for automated binary installation using APT for Debian and Ubuntu or YUM utility on CentOS 5 Linux and for RedHat Enterprise Linux 5: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
14:18.41infobotokay, pabelanger
14:18.46pabelanger~asterisk-packages
14:18.47infobotsomebody said asterisk-packages was Asterisk is available for automated binary installation using APT for Debian and Ubuntu or YUM utility on CentOS 5 Linux and for RedHat Enterprise Linux 5: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
14:18.58pabelangermha_: ^
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14:21.16mandlahelp, http://pastebin.com/643HYDki
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14:27.06kaldemarmandla: "DAHDI/g1/71520005/00267". you shouldn't put / in a number.
14:28.20mha_pabelanger: I will look into it late as I have to go now.  Will be back in near future
14:29.03leifmadsenpabelanger: ping
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14:30.25pabelangerleifmadsen: pong
14:30.44leifmadsenpabelanger: I invited you to a channel :)
14:31.08pabelangerleifmadsen: did see it
14:31.11pabelangerdidn't
14:31.21leifmadsenah... you should be in #asterisk-testing
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14:37.55mandlakaldemar, i dnt understand how this is done.
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14:44.26jc319I still have the same problem, I wonder if you can enlighten me on this issue: Basically I am trying to use single 'device' (=7960 hardphone or a softphone) and multiple extensions. All is good for the extension/line defined at the top (let's call it 1st line). I can also receive incoming calls for other extensions/DIDs/lines too. Only problem is, when I want to use one of the additional
14:44.26jc319lines on the 'device' and attempt to call outbound, I recevie this error message>> http://pastebin.com/Aa2ECxaA (lines 39, 107 in particular)
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14:53.21leifmadsenjc319: that's because it is matching on IP address, and saying, "hey, IP address 192.168.x.x matches peer PB1, but you're trying to authentication with PB3"
14:54.01leifmadsenyou have to make sure everything is setup to match on username (From field I think) instead of letting it fall over to matching on IP address. This can be tested using type=user to make sure you can't match on IP address.
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15:04.01jc319leifmadsen: thank you this is now solved, by using type=friend and also setting insecure=no. I don't know which one fixed the isuse (or combination?). I will also try user. and if they both work I don't know what to do.
15:05.59mandlaI rest my case.
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15:22.58jayteemorning beek
15:23.03beekhello jaytee
15:29.19Canabinoideis gone.. autoaway after 15 min ..[cyp(l/on.p/on)]
15:29.29leifmadsenCanabinoide: please turn that off
15:34.26beekHmmm... leifmadsen.  If he's gone, how is he going to respond to your request?  ;)
15:34.37leifmadsenbeek: he will see when he returns :)
15:34.58leifmadsendon't use your logic on ME!
15:35.58beekWhy not kick him with a "Hey -- turn that damned thing off?"
15:36.13leifmadsenreeks of effort
15:36.19ChannelZhe probably has auto rejoin
15:36.29beekChannelZ: Good point.
15:36.59*** kick/#asterisk [Canabinoide!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (please turn off your auto-away msg)
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15:37.01leifmadsen:)
15:37.25beekWell, there is no longer any question about his "auto-join"
15:37.29cneb3000does anyone follow security news?
15:37.36ChannelZheh
15:37.38leifmadsenI bet at least 1 person does
15:37.44cneb3000whos that?
15:37.46cneb3000....
15:37.48cneb3000oh! ;)
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15:40.10cneb3000anyway. lulzsec, a little hacker group, set up a conference bridge. and they used asterisk for it. just thought it was kind of cool.
15:40.16cneb3000climbs back under his rock
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15:40.30_Corey_ugh
15:40.59cneb3000ugh?
15:41.46_Corey_I'm not a fan of online vandals
15:42.18_Corey_(although, the news about our friend Tupac was quite welcome)
15:43.09cneb3000funny thing about online vandals, i admire their technical prowess.. but certainly dont agree with their sinister intentions
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15:47.23Qwelltechnical prowess?  You mean the do_everything_for_me.sh script?
15:47.53ChainsawJust hex edit the IP address of your control server and you're good to go.
15:48.14_Corey_you mean 'rootkitwizard.exe'
15:48.33Chainsaw(Some even ship a whole mIRC binary so they can script in there...)
15:51.17cneb3000Qwell: Hm. good point.
15:51.35cneb3000Obviously, doesn't apply to all circumstances, but most I imagine so.
15:53.54p3nguinThe trick is to ban them, kick them, wait long enough that auto-rejoin won't work, then unban them.
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16:01.51XagoHello...why a iax2 extension don't register to Asterisk?
16:02.34XagoI'm seeing diff between two extension and except register zone is different.
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16:11.57michael-iHi all. When using automixmon to record calls, I'd like to set a CDR variable to indicate this has been happened. Anyone do something like that before?
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16:15.35atheosmichael-i what you're looking for is userfield http://www.voip-info.org/wiki/view/Asterisk+func+cdr
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16:17.28michael-iatheos: That looks good but I need this to happen when someone activates automixmon from features.conf. Maybe have to define my own application mapping?
16:18.34atheosmichael-i not real sure in that application.
16:18.37leifmadsenmake the application mapping use a Macro to do both things for oyu
16:19.39michael-ileifmadsen: makes sense...that way I can also change the record-on and record-off tones to be different sounds
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16:20.20puzzledhi
16:20.22leifmadsenyep
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16:21.34drmessanocneb3000, my gang has been using Asterisk for years to plot our world domination.. Sure, we're not as big as Anonymous or have the awesome Anonymous name, but "Unknown Caller" means business!
16:21.56irrootLOL
16:22.10puzzledheheh
16:22.11leifmadsenlol
16:22.13leifmadsendrmessano: +1
16:23.56irroot<PROTECTED>
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16:31.08neurosysleifmadsen:  I remember reading about a bug in the 'sip show channelstat' a while backm reporting massive packet loss and jitter, but wasnt accurate. has that been corrected?
16:31.16neurosysI couldn't find anything in the changelogs
16:31.22leifmadsennot sure
16:31.24leifmadsenprobably not
16:32.40neurosysOk :) I have techs freaking out seeing retarded amounts of jitter. :P I told them to ignore it
16:36.57irrootlol
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17:02.16sunfoneI think I may have stumbled over a new bug in app_dial
17:03.26sunfoneapp_dial seems to keep track of extensions that have been dialed during the span of the call
17:03.49sunfoneand if you bounce the caller around a bit and in the end try to dial the original extension they came in on, app_dial fails
17:04.00sunfonehmm I am having trouble explaining :)
17:04.48WIMPyMight be the same issue as I read on mantis. But I con't remember what the scenario there was.
17:04.48sunfonebugging #freepbx folks right now about it
17:05.04sunfoneWimpy - do you have the bug id?
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17:05.33WIMPyNo. It onvolved a transfer or something.
17:05.43sunfonetransfers definitely involved
17:05.43WIMPys/on/in/
17:06.29sunfoneIs mantis still up?  Its being migrated to something new, right?
17:06.47irrootyes it is but still there read only
17:11.14linuxgeckoi'm trying to get connected to gtalk/google voice with my asterisk server.  everything seems to be working, except for the jabber part (AFAIK), jabber debug keeps spiiting this out, repeatedly on the cli. what should i check/do next to fix this? http://pastebin.com/xjp3xBwn
17:12.17Phr0zeni'm having a dahdi issue, i get a MCE as soon as the service is started, does anyone know how I can trouble shoot this (asterisk 1.8.4.1)??
17:12.49linuxgeckoPhr0zen: MCE?
17:13.17Phr0zenmachine check exception
17:13.25Phr0zenfreezes the machine
17:13.46irrootkernel / dahdi version
17:13.54linuxgeckothought so,  but i was suprised.
17:14.02irrootlast MCE i got requried CPU replaced
17:14.19WIMPyI thought that was the only possible cause.
17:14.23Phr0zenit runs perfect until dahdi starts
17:14.38Phr0zente410p card
17:14.48WIMPyPhr0zen: compile gcc and see if that works.
17:15.05*** join/#asterisk lcat (~lcat@187.45.255.66)
17:15.42Phr0zenwhat would that do?
17:15.43drmessanolinuxgecko, are you using TLS in the google account config?  If so, did you compile iksemel yourself, and did you have GNUTLS available?
17:16.07WIMPyPhr0zen: It's the best hardware test I know.
17:16.23Phr0zeni did that previously no issues
17:16.40linuxgeckodrmessano: no, i did not compile iksemel myself, but if it would fix this issue,  i gladly would :)
17:17.00*** join/#asterisk justnulling2 (~jnull@ool-4b7fd02a.static.optonline.net)
17:17.12p3nguinIf you just want to stress the CPU, and don't want to compile gcc or something else large, check out cpuburn.
17:17.35drmessanolinuxgecko, what about the first part?  TLS directives in the account config?
17:18.18drmessanousetls=yes, usesasl=yes
17:18.36drmessanoNeed that for Google accounts in jabber.conf
17:18.54Phr0zeni read that disabling echocancellation in /etc/dahdi/system.conf can stop the nmi or mce.... think that could work?
17:19.04p3nguinI've had issues with iksemel not having gnutls support.
17:19.15linuxgeckodrmessano: yes. i have them..  my config is as nearly copy-paste from the wiki about google voice calling as i can get it to try and work.
17:20.46linuxgeckodrmessano:  where is the best place to grab iksemel sources to compile?
17:20.56*** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net)
17:21.17p3nguinI'd be surprised if google doesn't know the official site.  :)
17:21.59linuxgeckop3nguin:  hehe..   true..  i have a horrible tendency to be lazy, and ask to be spooon-fed  :)
17:22.29p3nguinOnce you figure out the site, you'll see why this was an attempt to be funny.
17:22.49drmessanolinuxgecko, lazy?  Spoon-fed?  Have you ever been on IRC before?
17:23.19drmessanoThat's like bringing a Windows machine to a Linux fest
17:23.36drmessano"You must be new around here, boy"
17:23.57p3nguinWhy does it need to take so long to ship stuff from Hong Kong to the US?
17:24.42justnulling2trying to setup ConfBridge with 1.8.4 but get rather cryptic error app_confbridge.c:435 join_conference_bridge: Conference bridge '1001' could not be created. and changing to load=yes doesn't help,
17:24.54irrootlol @ p3nguin try get anything from anywhere to lusaka zambia :P
17:25.14p3nguinjoin_conference_bridge: Conference bridge '1001' could not be created.     <-- nothing cryptic about this.
17:25.49p3nguin1001 is not able to be created.  You either didn't allow for dynamically creating conferences or didn't statically create conference 1001.
17:26.05michael-ijustnulling2: I had something like that because I hadn't loaded the bridge_* modules
17:26.33coppicep3nguin: If I order things from Amazon in the US, sending some by each of the mailing options, the cheapest one will normally arrive first
17:27.05p3nguinI'm making the uneducated guess (because I use MeetMe) that ConfBridge uses options similar to MeetMe for creation of conferences.
17:27.56*** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net)
17:28.34p3nguincoppice: When I need gadgets and shop on ebay for them, I often find them dirt cheap, but they come from HK.  There have been times that it takes a full month to get my purchase.
17:28.34*** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net)
17:28.50coppicethey will usually leave here very quickly. the delays are at the US end
17:28.59p3nguincustoms?
17:29.11coppicesending to the UK usually takes 3 or 4 days by normal airmail
17:29.31p3nguinI think the shortest time I have received something from HK was probably two weeks.
17:29.56coppiceI usually get things from HK the same day
17:29.58justnulling2michael-i, i have them all (i think) bridge_builtin_features.so bridge_multiplexed.so bridge_simple.so bridge_softmix.so are there are more?
17:30.13_Corey_I think the profit margin on most stuff manufactured in HK is blown if it's not shipped in a container on a boat...
17:30.22*** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz)
17:30.22_Corey_:)
17:30.35p3nguinIt's just piddly stuff like small tools or accessories, so it shouldn't need to be scrutinized to heavily at customs.
17:30.40coppicenothing is manufactured in HK
17:30.42michael-ijustnulling2: think that's it. then all I needed in extensions.conf was "ConfBridge(500,Mc,123)"
17:30.49coppiceexcept our kids
17:31.09_Corey_for children, by children eh?
17:31.26michael-ijustnulling2: ...oh, and Answer() the call before calling ConfBridge
17:31.28Scorpio2007i would like to bounce calls to a different context based on the amount of channels in use .. now i can do this by using Set(Groups() = "something") on incomming contexts and outgoing contexts and using group_count() to see if its over a certain # ..
17:31.31coppicehuh?
17:31.47Scorpio2007is there a better way to do this or ?
17:32.18linuxgeckop3nguin:  agreed :)     but grrrr..   iksemel is not seeing the libgnutls files, because mine don't have gnutls-config in it...  do i have to compile gnutls also?
17:32.58Scorpio2007for example in my from-pstn context i have exten => s,n,Set(GROUP()=CALL_GROUP)
17:32.58Scorpio2007and in my outbound context i have exten => s,n,Set(GROUP()=CALL_GROUP)
17:32.58Scorpio2007and exten => s,n,GotoIf($[${GROUP_COUNT()} > 17]?pri:overflow)
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17:33.54justnulling2michael-i, i have an answer and wait(2) and just tried it with "ConfBridge(500,Mc,123)" stil the same error
17:34.37michael-ijustnulling2: then I'm out of ideas...sorry
17:34.52Scorpio2007anyone?
17:34.56p3nguinlinuxgecko: I've experienced that same problem several times.
17:35.04Scorpio2007coppice: were you responding to me?
17:35.29justnulling2michael-i, thanks anyway
17:40.37*** join/#asterisk fulcan (~root@li345-191.members.linode.com)
17:41.28_Corey_Scorpio2007: It sounds like you have the right idea using the group count function...  is that your question?
17:41.45Scorpio2007yes i would like to make sure there is no "gotchas" lol
17:41.59*** join/#asterisk jetlag (~jetlag@pool-71-188-34-28.cmdnnj.east.verizon.net)
17:42.20_Corey_Just experiment to work out the details, but you've got the right idea
17:43.11Phr0zenmachine check exceptions still happening on dahdi
17:43.12Scorpio2007_Corey_: it works so i just wanted to make sure that its the right way of doing it .. i have a pri which i take into my asterisk server .. i use this for all inbound and all outbound .. but i want to keep track of how many channels are in use .. because once it reaches 75% i would like to then route all my calls to my sip provider .. so i have enough channels for inbound on the pri
17:43.13Phr0zensigh
17:43.48_Corey_Well, the other way to do it would be to define different channel groups for incoming/outgoing
17:44.10_Corey_That said, there are many ways to do the same thing :)
17:44.13*** part/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
17:44.30Scorpio2007_Corey_: i wanted incomming and outgoing on the same group because incomming or outgoing on a pri i only have 23 channels
17:45.05*** join/#asterisk jameswf (~james@unaffiliated/jameswf-home)
17:46.12_Corey_Keep in mind your chan_dahdi group definitions don't affect which channels your carrier chooses to send you calls
17:46.34Scorpio2007_Corey_: agree
17:46.51*** join/#asterisk afink (~afink@204.26.87.226)
17:48.50*** join/#asterisk banditti (~banditti@unaffiliated/banditti)
17:49.09Scorpio2007_Corey_: if you had to configure this would you be using groups or something else?
17:49.53bandittiI am bringing up a asterisk box, besides geoip, anything that I should consider?
17:50.32Scorpio2007banditti: is it realtime sip / dialplan? does it have dual power supply .. there are quite a few htings
17:50.42*** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net)
17:51.06Eitanlooking for a good TTS for a bank. Needs to be able to read amounts clearly. Anybody have any recommendations
17:51.46WIMPyEitan: Amounts? Shouldn't need anything special.
17:51.47*** join/#asterisk cerberus_za (~coert@196-210-151-122.dynamic.isadsl.co.za)
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17:52.39Eitanyeah account balances....
17:52.58Eitanlooked at ceptral, not quite clear enough
17:53.28leifmadsenCepstral is the best one
17:53.34Eitandamn
17:53.35Eitanlol
17:53.39leifmadsenuse the right voice, and it works great
17:53.51WIMPyWhy do you need TTS for that?
17:53.57Eitanyeah, we went through all the voices. I was ok with it, but the execs at the company didnt like it
17:54.05leifmadsenplus ya, just use the built in applications
17:54.15leifmadsenSayDigits()
17:54.19leifmadsenSayNumber()
17:54.31Eitanthe guy that is building the IVR recomended it, said that the built in sounds pooorly
17:54.58WIMPyTry it
17:55.02Eitani am new to asterisk, which is why we hried someone... he is building hte IVR to make API calls through our vpn connection to get balances and other account info
17:55.20Eitanill give it a shit
17:55.37Eitanshot
17:55.38Eitanhaha
17:55.42bandittilet us know how that works
17:55.43banditti:)
17:56.14bandittiI am bringing up a asterisk box, besides geoip, anything that I should consider?
17:57.19atheosEitan - I have an AGI that connects to an apple, using the apple speech synthesis. works pretty good, IMO. If you have access to an OSX box, do a : man say  then test out a few of the voices.
17:58.00Eitanatheos: yeah i have a new g5...
17:58.09Eitani could also install osx on a vm on our server
17:58.55atheosEitan, yea, I just shell to an xserve for one phone system, and a macpro for another. It's fast, and the voices sound good.
18:03.54Phr0zenso, after running dahdi_genconf the machine starts to chuck out machine check exceptions, but if i wipe out the conf files machines start fine.  so can a misconfigured dahdi cause this?
18:04.22WIMPyUnlikely.
18:05.15WIMPyWhat dabug info do you get?
18:05.18*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
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18:05.22WIMPys/da/de/
18:08.05Phr0zenwhere do i find the debug info?
18:08.32WIMPyAs you machine freezes, there would only be the screen.
18:08.48Phr0zencpu0 machine check exception 3894023840583490534859034
18:08.53Phr0zensomething like that
18:08.54Phr0zenchanges
18:08.58WIMPyTurn up the console log level. Alt-SysRq-KP9
18:09.25Phr0zenkp9 ?
18:09.26Phr0zenlolwut
18:09.34WIMPyKeypad-9
18:09.35Phr0zenkeypad 9?
18:09.37Phr0zenahh
18:09.45Phr0zenthat wont do anything
18:10.00Phr0zensince the machine totally freezes and cannot do anything put hard power off
18:10.15WIMPyBad kernel config. Then yu have to do it via klogd parameter.
18:10.41Phr0zenit says run mcelog --ascii
18:10.51Phr0zenwhich seems to provide no output after running that
18:10.56WIMPyWell, you should obviousely do that before you load the modules or at whichever exact point it throws the MCE.
18:11.13Phr0zenif i service dahdi start  -> bam system dead
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18:12.23Phr0zenits a complete lockup, nothing i do provides more detail than that
18:13.08WIMPyIf you turn up the console log level, you get everything on the screen that would otherwise turn up in dmesg.
18:13.24WIMPyThat should tell you exactely at which point it happens.
18:14.06Phr0zenjust by doing alt-sysrq-np9?
18:14.09sunfoneSo I found the JIRA issue for app_dial that I tripped over this morning... https://issues.asterisk.org/jira/browse/ASTERISK-17874.  The patch that fixes it is for 1.6.2.  So given that it isn't security related, is 1.4.X doomed to have this problem forever?
18:14.17WIMPyyes
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18:14.49sunfoneWow that sucks.
18:14.57*** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net)
18:15.06sunfoneEspecially since it is a regression.
18:15.41WIMPyHmm. Why can't I connect to jira from this machine?
18:16.09sunfoneHeh - looking at the diff its actually pretty small.  Going to try and backport it.
18:16.11Phr0zenshould i see anything on screen when i do that altsysrqnp9 deally?
18:16.23sunfoneIf you haven't logged into JIRA yet you have to reset your password.
18:16.29WIMPyPhr0zen: Yes
18:16.59sunfoneSo if I come up with a 1.4 diff, should I attach to the issue?
18:17.03sunfoneIs that still welcome?
18:17.08WIMPysunfone: Yesm I did. It works from the other machine. Here it always says "Timeout reading data".
18:17.16sunfoneweird
18:17.29WIMPyBut only under /jira/
18:17.42sunfonebrowser?
18:17.53WIMPyPhoenix
18:17.58WIMPy:-)
18:17.59sunfoneThink I recall a thread of problems with Safari
18:18.32sunfoneI would think firefox would be ok :):)
18:18.44WIMPyNot in this machine.
18:18.53sunfonebummer
18:19.04sunfoneAre people liking the JIRA move?
18:19.09sunfoneNo one likes change I guess.
18:20.01Phr0zenok so i did that alt_sysrq_np9 thing
18:20.04Phr0zenstarts fine
18:20.08WIMPyI find it less clearly arranged. But with some setting up, it's ok.
18:20.22Phr0zenthen it gets to the clock setting or something and machine locksup
18:21.03WIMPyDo you have the possibility to send your console to serial?
18:21.23Phr0zenactually i do
18:21.31Phr0zengot a cable here and a laptop
18:21.50Phr0zenwould have to use putty, dont think i have anything else on that laptop for it
18:21.56WIMPyCool. then you could paste the output.
18:22.32Phr0zen..if i can find my cable sigh
18:23.22Phr0zendamn cleaners moved it
18:23.24Phr0zensigh
18:23.36Phr0zengot it now, will be back once this ghetto lappy boots
18:26.48Phr0zenactually that pos dell laptop does not have a port, gettin another pos laptop to use (sigh)
18:27.51sunfonehmm ast_datastore_free() doesn't seem to exist in 1.4
18:27.52*** join/#asterisk wonderworld (~ww@port-92-201-11-198.dynamic.qsc.de)
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18:33.13Phr0zenlol this laptop is still booting
18:33.37Phr0zengonna break it over my knee like a breadstick after i get this issue sorted
18:33.59sunfoneShooting it is much more satisfying.
18:34.07sunfoneFather in law shot his monitor last year
18:34.10Phr0zenall i have is shotguns
18:34.12sunfoneSaid it made him feel great
18:34.18Phr0zenthat should suffice?
18:34.27sunfoneI would think so ;)
18:34.40sunfoneHe used a 9mm... just made holes I guess
18:34.46jayteeactually it's much more satisfying using a baseball bat while playing "Damn it feels good to be a gangsta" by the Gheto Boyz
18:34.53sunfonelol
18:34.55Phr0zenjust need someone to toss this lappy like a clay pidgeon
18:35.02Phr0zenempty the gun on it before it lands
18:35.17sunfoneI don't think there will be anything left to land!
18:35.20WIMPyIt's much more fun, when it's a good old CRT :-)
18:35.53sunfoneIn college someone entered the famous egg drop contest by dropping a CRT from the top of the engineering center (20 stories?) with an egg duct taped to it
18:35.57sunfoneThe egg didn't make it
18:36.26Phr0zenthis pc isnt gonna make it to get windows booted.. rage...building... starting...to...boil...
18:36.32_Corey_haha, great excuse to drop a crt
18:37.19Phr0zeni have never seen a latop so slow in my life
18:38.11Phr0zeno son of a random standby
18:38.15Phr0zentoday is not my day
18:38.56WIMPyOn some days you lose,
18:39.03WIMPyon some days the others win.
18:39.23Phr0zenoh and to top that off.. a java update nag
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18:42.39boazbjoin #vuc
18:42.43boazbwhoops
18:42.49boazbforgot the /
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18:48.43Phr0zencant get output to serial to work
18:48.55Phr0zenanything else i can do to get some logs before the machine strokes out?
18:49.15linuxgeckop3nguin:  where you have to compile EVERYTHING from source?
18:49.27senatorif i understand correctly, callprogress is the feature that, if enabled in chan_dahdi.conf, /might/ detect tones such as SIT if in the US for outbound dialing. correct so far?
18:49.44p3nguinlinuxgecko: What?
18:49.56WIMPyPhr0zen: I think there's an option to use the network. But I've never tried that.
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18:50.37cneb3000-laptopclear
18:50.38Phr0zeni can ssh into it of course, but once it goes down all subsequent messages are not displayed
18:50.39cneb3000-laptopwoops
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18:51.37WIMPyPhr0zen: Sure. I ment a network logging option. But then you never know how fast that might die.
18:51.49p3nguinsyslog?
18:52.06linuxgecko<PROTECTED>
18:52.10WIMPyFor klogs.
18:52.38p3nguinlinuxgecko: Oh.  No, I've experienced the problem with iksemel not caring that I installed gnutls.
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18:53.47Phr0zennah i cant get anything off that machine for erros
18:53.52Phr0zenblows up too fast
18:53.56Phr0zencanno believe there is nothing i can do
18:53.59Phr0zenanyone?
18:54.32WIMPyUse a camera?
18:56.45Phr0zentried by phone crashed fuckin LG
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18:57.06Phr0zenshould be replaced tomorrow with a bb or something lol
18:57.13Phr0zenjust a tech nightmare week for me so far
18:57.14Phr0zen:P
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19:02.55Phr0zenno chance this phone can snap a pic of that
19:03.03Phr0zentoo quick and phones are way to damn slow
19:03.08Phr0zenanything else i can do?
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19:03.27WIMPyDo the messages disappear?
19:03.46Phr0zenthe startup messages get overrun with a full screen of machine check exceptions
19:04.03WIMPyOh :-(
19:04.18Phr0zenlast think i see is oks for the dahdi start
19:04.30Phr0zenthen it does a time source thing or something i dunno
19:04.33Phr0zenthen bam dead
19:04.40Phr0zenmachine worked with zaptel but not dahdi
19:05.00senatorcallprogress, still considered buggy/incomplete? using asterisk 1.6.2.18
19:05.01WIMPyMaybe you should try to start the stuff by hand, without that script.
19:05.13Phr0zenwouldnt have a clue where to start
19:05.18WIMPyWouldn't be the forst script to load all kinds of stuff you don;t need.
19:05.38Phr0zenstarting the dahdi service?
19:05.46WIMPyyes
19:06.07Phr0zeni wouldnt know about that, not a linux guru over here
19:06.11otwieraczOK, is there anyone who can help me to set-up and test basic Asterisk config?
19:06.42WIMPyotwieracz: The book
19:06.46WIMPy~thebook
19:06.47infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook
19:06.55Phr0zencant find much help for this stuff, linux community is so fragmented its pathetic
19:07.02otwieraczWIMPy: I can't call book. :)
19:07.03WIMPyOtherwise ask
19:07.06WIMPy~ask
19:07.06infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
19:07.46Phr0zeni found on the digium site about adding boot options if u get nmis and such
19:08.21Phr0zenhowever they dont seem to work (nmi_watchdog=0, pci=norouteirq acpi=off pci=noapci noapic)
19:08.54Phr0zenis it possible to RMA a te410p card?
19:09.05Phr0zenperhaps i have one with bugs?
19:09.06WIMPyPhr0zen: Not the best starting point. You should modprobe the module you need for the card you've got. And then do a dahdi_cfg.
19:09.37WIMPyYou should contact Digium support on that one.
19:09.42Phr0zenhrmm, i did do a dahdi_genconf
19:10.02Phr0zenis there an online chat or any easy way to contact them that doesnt cost me $$
19:10.10WIMPyThat's only needed when changing hardware.
19:10.23WIMPyCall them?
19:10.48Phr0zenyea looks that way, gonna have to call.. (not even a toll free lolz)
19:13.20Qwellyou didn't look very hard, did you?
19:13.50Qwellhttp://www.digium.com/en/company/contact.php
19:14.57Phr0zenhehe i did
19:15.04Phr0zeni just didnt see the toll free on the right
19:15.07Phr0zenjust say the support
19:15.10Phr0zen*saw
19:16.05otwieraczhmm…
19:16.06Phr0zenfrustrated, didnt read it properly
19:16.17otwieraczEcho-test is not working in my Ekiga.
19:16.27otwieraczHow to check is that Ekiga problem or something with network?
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19:27.22p3nguinNo iNum for Digium?!  What kind of BS is that?
19:28.17doolittleworkwill i put load on cpu if i do a mysql query from the dialplan before every call?
19:28.30otwieraczOch, EchoTest works.
19:28.58_Corey_p3nguin: Digium's ISN # is 291, is that what you mean?
19:29.00WIMPydoolittlework: Sure, unless you know some magic.
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19:29.22p3nguinI don't even know what ISN is.
19:29.27russellbfreenum.org
19:29.32doolittleworki am no magician
19:29.49doolittleworkWIMPy: can you elaberate
19:30.12WIMPydoolittlework: I think you answered it yourself already.
19:30.20doolittleworklol
19:31.19*** part/#asterisk Corydon76-home (beige@c-69-137-80-31.hsd1.tn.comcast.net)
19:34.09*** join/#asterisk theron (~theron@ip244.scolloc.lh.net)
19:38.32jc319Is there a web based firewall test tool to check if UDP port is accessible from the Internet? It's open & redirected on appliance firewall but mobile sip clients on 3G can't communicate with Asterisk...
19:44.12Phr0zennever used it but would this work?  http://www.auditmypc.com/firewall-test.asp
19:44.29p3nguinDepending on the SIP client, it could be using TCP.
19:44.57tonsofpcstelnet?
19:44.57Nuggettelnet is eeeeeeevil!
19:45.00p3nguinTo get Asterisk to use TCP, you'll have to edit sip.conf slightly, then forward TCP 5060.
19:45.07tonsofpcsbot?
19:46.38jc319Firewall results Congratulations! None of the ports you scanned for were open!
19:46.40jc319haha
19:48.21p3nguinUDP is connectionless, so that's a typical result.
19:48.47p3nguinYou need to run an nmap udp scan from a remote host if you want better scan results.
19:50.02p3nguinI suppose someone here might help you out if you provide the host address to scan.
19:50.42jc31978.86.169.203
19:50.46jc319thanks
19:50.57jc319(UDP/5060)
19:51.52*** join/#asterisk markfoodyburton (~Adium@LMontsouris-156-24-44-5.w217-128.abo.wanadoo.fr)
19:53.57jc319p3nguin: thanks to your extended help, everything in my basic goals set works now, so far only in this local network. Before adding more features my next goal is to make it accessible from the Internet.. as of now I can use 7960s or softphones [on LAN] to call with caller id of my choice with sub-account (billing) of my choice. incoming calls also work and ring correct set of endpoints.
19:53.57jc319lovely.
19:54.21*** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18)
19:56.23p3nguinSounds like progress.
19:56.29p3nguin5060/udp open  sip     (SIP end point; Status: 200 OK)
19:56.59p3nguinNow check to see if your mobile SIP app requires the unusual SIP over TCP.
19:57.12*** part/#asterisk senator (lebbeous@nox.esilibrary.com)
19:59.37markfoodyburtonHi, I have a problem getting SIP working over 3G. My setup works fine over wifi, but over 3g I guess I have so much packet loss that the audio is so choppy to be useless. What I expected is that the audio would be delayed, but not toooo choppy…. I got as far as guessing that the TTL on the packets was casing the packets to be lost, so the jitter buffer wasn't helping, then I decided I need help (probably psychiatr
20:01.14anonymouz666I would say that 3G is not reliable
20:01.14michael-iCan variable initialization look sexier than this?! :) ExecIf(${ISNULL(${CDR(recordingcount)})}?Set(CDR(recordingcount)=0))
20:02.34markfoodyburtonanonymouz666: I guess your right, but I seem to have a reasonable 3g signal etc, so - the tech nurd in me wants to believe it should work? (even if it's not reliable all the time)
20:03.03otwieracz<PROTECTED>
20:04.02markfoodyburtonIs there some way of determining why I'm getting such choppy audio? - As I say, I'd expected, given that the registration succeeded, to get passable audio - if somewhat delayed...?
20:06.53bandittiI am bringing up a asterisk box, besides geoip, anything that I should consider?
20:07.34QwellThere is lots you should consider.  Can you be more specific?
20:10.40*** join/#asterisk Ad-Hoc (~nimbus@athedsl-392953.home.otenet.gr)
20:11.19bandittigeolock, other security / hardening options
20:15.04*** join/#asterisk mha_ (~mha@77.69.131.121)
20:15.34jc319p3nguin: A remote node (now a PC with Zoiper Classic) is trying to register, but with verbose L4, nothing is reported on Asterisk CLI - does this mean that packets don't even reach Asterisk?
20:17.06*** join/#asterisk infobot (~infobot@rikers.org)
20:17.06*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.2 (2011/06/04), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
20:19.34*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net)
20:19.35p3nguinNo, it does not mean that.  If you want to find out if the device is attempting to register, you have to use sip debug.
20:21.47WiretapSevenjc319, sip set debug on
20:22.19WiretapSevenmarkfoodyburton, choppy audio is dropped packets and/or jitter
20:22.58markfoodyburtonhi WiretapSeven, how can I check that the jitter buffer is on/working at both ends - is there a simple CLI command?
20:23.36markfoodyburtonon the handset, I have "jitterbuffer : 200" as an option, which is great - you can try any number you like, so long as it's 200 !
20:23.37markfoodyburton:-)
20:23.50leifmadsenmake sure you use the static jitterbuffer and not the dynamic one
20:24.25leifmadsenI think the dynamic one was fixed in perhaps the latest 1.8 branch and trunk, but not sure (I think The_Boy_Wonder may have fixed it), but I never had luck with dynamic. You should make sure it's static to start.
20:27.09markfoodyburtonleifmadsen:  I have:
20:27.09markfoodyburton<PROTECTED>
20:27.09markfoodyburton<PROTECTED>
20:27.09markfoodyburton<PROTECTED>
20:27.09markfoodyburton<PROTECTED>
20:27.09markfoodyburton<PROTECTED>
20:27.10markfoodyburton<PROTECTED>
20:27.17p3nguinUh
20:34.08WiretapSevenmarkfoodyburton, pastebin
20:34.10WiretapSeven~pastebin
20:34.10infobot[~pastebin] 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.
20:34.52Qwell^ 7 lines
20:35.12markfoodyburtonYeah - sorry - 7 was > 3 …..
20:35.28QwellI mean the correction.  Nevermind.  Irony.
20:36.10*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
20:36.14markfoodyburtonAd that was 6 :-) - if it had been 3, then, really, infobot shoudl point at a pastbin to tell you about pastbin :-)))
20:41.57jc319p3nguin & WiretapSeven: I run "core set verbose 4" && "sip set debug on" already but nothing came up about the remote SIP client trying to register. I unplugged all other hard/softphones, so it's the Asterisk only (and the external client). The only messages are about DID register challange/re-registrations. Nothing about the remote client.
20:42.14jc319I believe it does not even get to the local Asterisk. Any ideas?
20:42.21WiretapSevensounds like it doesn't
20:42.25WiretapSeventime for packet traces
20:43.24p3nguininfobot printed the entire message on a single line.
20:43.38p3nguinno cr, no newline
20:46.14*** join/#asterisk jkroon (~jkroon@dsl-242-10-219.telkomadsl.co.za)
20:47.14*** join/#asterisk slidesinger-lt (~jtatum@c-68-44-99-163.hsd1.nj.comcast.net)
20:47.53jkroonhi guys, i've got a setup with chan_dahdi, AEX800 with 2 x quad FXO (8 x FXO channels), four lines currently populated, one of the lines no echo, the other three lines insane echo.
20:48.12jkrooni've gone throught the fxotune routine, and I've set up oslec.
20:55.24*** join/#asterisk sourcode (~code@ppp-58-8-90-32.revip2.asianet.co.th)
20:56.18*** join/#asterisk remx (~remx@corporate.runecho.com)
21:03.16*** join/#asterisk mer_ge (~martin@chello062178156192.9.14.tuwien.teleweb.at)
21:05.49jc319<p3nguin> To get Asterisk to use TCP, you'll have to edit sip.conf slightly, then forward TCP 5060.
21:05.55jc319p3nguin I want to give this a try
21:06.09*** join/#asterisk cneb3000-laptop (~cneb30000@02da1dc1.bb.sky.com)
21:06.26jc319I have confirmed remote SIP peer windows machine and Internal Asterisk server can establish TCP/5060 connection
21:06.34p3nguinLook for tcpbindaddr in the sip.conf.sample and configure it, then sip reload.
21:06.58p3nguinMost SIP user agents do not use TCP.  Some do.
21:07.21*** join/#asterisk Ryushin (proxy@cl-412.phx-01.us.sixxs.net)
21:07.29p3nguinHow did you determine that yours established communication on TCP:5060?
21:10.35*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
21:11.15*** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl)
21:11.36jc319GNU nc (netcat) utility
21:12.16jc319UDP MODE server listens: netcat -l -u -p 5060 --source=10.9.8.4
21:12.16jc319UDP MODE client connects: nc -u -p 5060 78.86.169.203 5060
21:12.16jc319TCP MODE server listens: netcat -l -p 5060 --source=10.9.8.4
21:12.16jc319TCP MODE client connects: nc -p 5060 78.86.169.203 5060
21:12.44jc319When one types on one screen, it appears on the other one. With TCP mode. UDP did not work though
21:13.28michael-iIs it possible to use func_cdr to write values to the "common" cdr between two bridged channels? I'm trying to have two legs both work on a count variable which is then stored at call end.
21:13.31jc319(nc is windows nt build's executable name, netcat is the one for linux executable)
21:13.37michael-iSo far, each maintains its own count.
21:13.47p3nguinjc319: What exactly are you testing against?
21:14.15p3nguinjc319: You're ON THE LAN and testing against a public IP address again?
21:14.23RyushinI'm trying to make an ADT alarm system work through Asterisk PBX and a FXO card.  It worked prior to in 1.6, but 1.8 it's not.  I came across this post: http://www.voip-info.org/wiki/index.php?page_id=562&tk=41aef42dcf2c07164b4a&comments_page=1
21:14.24jc319Nothing related to SIP or Asterisk. Basic network connectivity between two computers. One computer runs netcat -l (listen). The other (remote) computer runs netcat connect.
21:14.32p3nguinjc319: What are you trying to do?
21:14.32RyushinAbout using DTMF = inband.
21:14.34jc319p3nguin: remote computer is remote desktop connection.
21:14.43*** join/#asterisk loconut (~loconut@173-16-61-8.client.mchsi.com)
21:15.00jc319<jc319> Nothing related to SIP or Asterisk. Basic network connectivity between two computers.
21:15.12jc319p3nguin: I am trying to test network connectivity.
21:15.15RyushinWhere can I set the DTMF inband setting?  chan_dahdi.conf does not seem to have an option for that.  I can't find any information on voip-info.org either.
21:15.24loconutanyone know how to set up a config (SIPDefault or something else) via TFTP to enable a DND toggle soft key on a 7960? I have the TFTP stuff already up just don't know how to deal with soft keys
21:15.24p3nguinWe already know the network works.  Move on.
21:15.32jc319How do you know that?
21:15.41p3nguinYou've been making calls.
21:16.11jkroonhi guys, having some serious problems getting rid of echo on some analog lines.
21:16.23jc319That only shows Asterisk successfully connects to its uplink (ITSP). How can you be sure that [any remote client] can successfully connect to Asterisk?
21:16.30*** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt)
21:16.32[sr]howdy
21:16.35[sr]hi WIMPy
21:16.50WIMPyHi [sr]
21:16.57[sr]i'd like to make a question, what is people using for appliance's?
21:17.02p3nguinjc319: Well... I would configure the phone and use it, but that's just me.
21:17.14[sr]build their own machine? use a pre-built appliance?
21:17.16p3nguinappliance's what?
21:17.48cneb3000-laptop[sr], do you mean server?
21:17.49p3nguinFor appliance's power cords, I use 6 feet ones.
21:17.53[sr]p3nguin: i mean, those RACK 4U with asterisk builtin
21:18.08[sr]p3nguin: ok my english killed me on that word, sorry
21:18.13p3nguinFor appliance's hardware, I use an embedded system.
21:18.14[sr]cneb3000: yes
21:18.51cneb3000-laptop[sr] no problem :)
21:19.02[sr]p3nguin: trademark?
21:19.13p3nguinHP Neoware thin client
21:19.32[sr]let me check
21:19.46[sr]some people use the XORCON one's, but i don't really have a feedback about it
21:19.48p3nguinYou can install CF card or DoM flash.
21:20.10p3nguinThey often come in 800 MHz or 1.0 GHz.
21:20.17jc319p3nguin: That's what I did first - being on the LAN, I'm using softphone over a remote connection running on a remote computer and that client cannot connect to local Asterisk. It cannot even produce single line of 'entry' in verbose/debug logs. Therefore I assume there is no network connectivity. That's why I switched to TCP now but it seems like it's not working with TCP either...
21:20.31p3nguinMost of them use normal SDRAM.
21:20.50[sr]p3nguin: maybe in my case i need something bigger, i need call recording
21:21.24p3nguinjc319: Out of 10 different user agents, maybe one will use the non-standard TCP.
21:21.37p3nguinZoiper is not one.
21:21.54p3nguinX-lite is not one (as far as I know).
21:22.19WIMPyEvery UA offering TLS.
21:22.34cneb3000-laptop[sr] how many people will use your server? and how many calls with happen at one time?
21:22.38p3nguinI have seen a couple of smart phone SIP apps that use TCP.
21:22.39jc319Noticed that, using mobile (iPhone) and Acrobits Softphone to test (it does support TCP - at least it is in the Advanced Configuration screen). It did not connect though...
21:23.09p3nguinAnd if you don't set up asterisk to use tcp and forward the port, it won't ever work.
21:23.15[sr]cneb3000: 20/30, and i need to insert at least 2x BRI card's... so the thin client won't do
21:23.25jc319p3nguin: Also this TCP is just a test, to prove network connection works and a remote client can register with it - but test failed.
21:23.39cneb3000-laptop20-30 calls?
21:23.44michael-iWorking on my recording macro with cdr logging, here is the conf so far: http://pastebin.com/9LUtti6E
21:24.01[sr]cneb3000: persons, at the same time, maybe 3/4 in the higher time on the day
21:24.04cneb3000-laptopthat's not many :) i've never used an xorcon but they look like a lot for what you want, i'd say.
21:24.08michael-iCDR still doesn't record the value (defined in sqlite3_custom.conf) and counts are still independent of each other...
21:24.28cneb3000-laptopmichael-i: thanks :)
21:25.48michael-iummm....you're welcome? I'm just trying to figure out how to merge the counts and actually SAVE the value :)
21:26.05cneb3000-laptop..just nice seeing dialplans shared ;)
21:27.03michael-imy pleasure sharing broken stuff
21:27.26luckman212_guys, an issue that I created on issues.asterisk.org has requested that I upload a patch file.  Sorry for this dumb question but would I just do a 'diff' on the 2 files to create this? or is there another way?
21:28.08*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
21:28.11WIMPyluckman212_: If you're usind svn version, just do 'svn diff'.
21:29.27luckman212_but I only want to patch against 1 file, I have modified quite a number of files....
21:29.54michael-isvn diff path/to/your/file
21:30.37luckman212_ah ok cool, thanks!
21:30.43jc319p3nguin: I did set up Asterisk to use TCP and forwarded the port from broadband router to Asterisk's (LAN) IP. Did not work...
21:32.10p3nguinI'll check it.
21:32.24p3nguin5060/tcp filtered sip
21:33.34p3nguinDid you make sure that Asterisk is listening?
21:33.45p3nguinlsof -i :5060
21:34.26p3nguinor if you don't have lsof,  netstat -lnptu|grep 5060
21:35.10p3nguinBoth of those commands will check for tcp and udp.
21:37.17jc319[root@thin ~]# lsof -i :5060
21:37.17jc319COMMAND   PID USER   FD   TYPE DEVICE SIZE NODE NAME
21:37.17jc319asterisk 2083 root   14u  IPv4   7495       UDP thin.grayzone.local:sip
21:37.17jc319[root@thin ~]# netstat -lnptu|grep 5060
21:37.17jc319udp        0      0 10.9.8.4:5060               0.0.0.0:*                               2083/asterisk
21:37.18jc319[root@thin ~]#
21:37.36p3nguinSo this confirms that Asterisk is *not* listening on tcp.
21:37.40jc319it's running with no roblems, I checked netstat before, however this only shows local data. This does not confirm remote hosts can connect.
21:38.17jc319problems* [i.e. comes up in the list]
21:38.25p3nguinI already confirmed that I can reach you on udp.
21:39.14p3nguinWhat did you do to enable tcp?  Do you even NEED to turn on TCP or are you just fiddling because you don't have anything else to do?
21:39.54jc319I don't _need_ to turn it, in fact I prefer to keep it UDP (assuming it would be faster). Turning to TCP option was just to test...
21:40.07jc319(UDP does not work at all - so let's see if TCP would work?)
21:45.59p3nguinjc319: Does sip show peer 243 show you anything?
21:48.14jc319It gives 2 pages of details from sip.conf peer definition. Are we looking for something in particular?
21:48.36p3nguinJust wanted to make sure it's configured.
21:49.35*** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap)
21:50.39jc319p3nguin: would binding to localIP:5060 create a problem?
21:52.45jc319p3nguin: In the pasted config files 2 tcpbind lines are active - actually currently running config have them commented out. All the other details are the same between the pasted config files & actual system.
21:53.25p3nguinNo, but there's probably no reason to not use 0.0.0.0 as the bindaddr.
21:53.52p3nguinThe peer you created for me needs adjustment.  It's not right for me.
21:53.58*** join/#asterisk brainiac (~mhauss@c-68-52-249-113.hsd1.tn.comcast.net)
21:54.44p3nguinYou duplicate options.
21:55.08p3nguinRemove the ACL permit if you're not going to use it.
21:55.17p3nguinset qualify=yes, nat=yes
21:55.23brainiacwhen I do 'pri show span 1', it says no PRI running on span 1.  How can I get it to work
21:55.33p3nguinsip reload, then I'll try to register again.
21:55.36jc319OK adjusted like this:
21:55.37jc319;tcpenable=yes
21:55.37jc319;tcpbindaddr=10.9.8.4:5060
21:55.37jc319;bindaddr=10.9.8.4:5060
21:55.37jc319bindaddr=0.0.0.0:5060
21:55.59jc319ACL is there because I'll use it later - currently commented out till the issues are fixed.
21:56.10jc319qualify=yes, nat=yes editing now..
21:56.15p3nguinIt's not commented out in the config you showed me.
21:57.17WIMPybrainiac: Check your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf.
21:58.28brainiacok
21:58.47jc319Oups you're right, it's active but allowing all hosts, so practically 'not in use' now, but as I said it will be so let's keep the line. Commented out it now & sip reloaded. (qualify & nat are set to yes for peer 243)
21:59.03jc319p3nguin: sip show peers gives this line for 243             243                        (Unspecified)                            D   N      0        UNKNOWN
22:00.32*** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net)
22:00.34jc319how about this setting, should it not chaneg? insecure=port
22:00.39jc319change*
22:01.28p3nguinGo ahead and leave it for now, but we may change it later.  sip set debug on
22:01.36p3nguinLet me know when you're watching it so I can try to register.
22:01.45linuxgeckop3nguin:  how did you get isemel to care that you have gnutls?
22:01.58linuxgecko*iksemel
22:02.00jc319watching now
22:02.08jc319debug L4, sip debug on
22:02.28*** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net)
22:02.39jc319Really destroying SIP dialog '737070ec77ec778440c49f556fb986d7@127.0.0.1' Method: REGISTER
22:02.47jc319I think this is the first time I see 127.0.0.1
22:03.38p3nguinAnything?
22:04.20jc319Nothing sorry
22:04.33jc319I thin I will need to come home to use voip
22:04.43sunfoneSo I backported a patch to 1.4 from 1.6.2, but the issue in JIRA is closed.  I don't seem to be able to attach my patch to the issue for any poor sap that has the same problem in 1.4... but then this is the first I have tried to work with JIRA... am I missing something?
22:04.44jc319s/thin/think/
22:05.08p3nguinAsterisk answers me when I nmap it.
22:06.32p3nguinhttp://pastebin.com/CHkcLwJj
22:06.56*** join/#asterisk ioioxoo (c2cbd7fe@gateway/web/freenode/ip.194.203.215.254)
22:07.49jc319So there's some kind of connection through this router but not the type that we want
22:08.01p3nguinActually that is the type I want.
22:08.16p3nguinThat was a SIP "connect" on UDP 5060.
22:09.21p3nguinYou're not seeing the attempts on sip debug?
22:10.13sunfoneUmm, it helps if you are logged in when you try to update a JIRA issue, apparently ;)
22:11.24leifmadsen:)
22:11.28jc319p3nguin: No I can't see anything at all. I keep scrolling up & down in putty window and nothing. Not to miss anything, I'm also CTRL+F searching for '243', nothing shows.
22:11.55p3nguinIs ANYTHING scrolling at all?
22:12.24p3nguinWith sip debug turned on, crap should be going past if you're got any SIP activity at all.
22:12.54jc319Yes, loads of stuff [4x DID registers refreshing, 2x Cisco 7960s doing some stuff etc], nothing related to 243 though.
22:13.38p3nguinMaybe you need to filter by my IP address.
22:15.29jc319How to filter, is it a CLI command? or do you mean log processing?
22:19.46jc319Anyway I keep copy & paste & searching, nothing so far..
22:20.13jc319Could it be anything related to disabling the following 3 modules?
22:20.15jc319;noload => pbx_ael.so
22:20.15jc319;noload => app_queue.so
22:20.15jc319;noload => app_dial.so
22:20.33jc319hmm they are not even disabled - ignore this sorry.
22:20.47jc319I think I'll change the router
22:33.34linuxgeckop3nguin:  did you have any insights as to how you got iksemel to compile with gnutls when it didn't want to see where i put it?
22:34.10*** join/#asterisk jkroon (~jkroon@dsl-242-10-219.telkomadsl.co.za)
22:34.51drmessanolinuxgecko, that's a hard one.. Some people have forced the path, others haven't.. even on the same distro and release.  I've never had to force it
22:35.06*** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net)
22:35.10linuxgeckodrmessano:  your distro?
22:35.57drmessanoI've compiled on both CentOS 5 and Ubuntu 11.04
22:36.26drmessanoActually, Ubuntu was a waste.. the packages actually don't suck with Ubuntu
22:36.56drmessanoCentOS is so far behind, all their Google packages are under "Backrub"
22:37.02drmessanoDon't get me started
22:37.35drmessanohttp://www.google.com/about/corporate/company/history.html  <--- to back up my obscure BackRub reference
22:38.40drmessanohttp://www.redhat.com/about/news/prarchive/2010/new-standard.html  <-- To back up my complaint that CentOS is so left behind, it's pic is on milk cartons
22:42.14linuxgeckodrmessano:  hehe ...  I'm on opensuse now, and very nearly a full convert to it. its the only "dead-Rat" based I've considered in 9 years.
22:43.14sunfoneUbuntu 11.04 rules
22:43.44linuxgeckosunfone:  that one has wayland in it,  yes?
22:43.46sunfoneI'll never go back to redhat based systems
22:43.52sunfoneof course ;)
22:44.31sunfonebut not on my servers...
22:44.31linuxgeckosunfone: ewww.   but that's presonal taste.
22:44.50sunfonefor my desktops I've been toying with Kubuntu and kind of like it
22:46.03drmessanoI have been happy with Ubuntu Server and Desktop.. though Natty is going to go down as the "Vista" of Ubuntu Desktop thanks to the half-complete Unity interface
22:46.10linuxgeckosunfone: i was a kde guy in gentoo,   then i moved to gnome with ubuntu because i needed to learn it for where i was doing some work,  now I'm back to kde with suse.  i never liked the way ubuntu did kde.
22:46.29sunfoneI've never tried SUSE
22:47.43drmessanoAfter Microsoft worked out that agreement with Novell that involved distributing Suse, I wrote it off
22:47.45linuxgeckoi like it..  especially because i had to remote-admin alot..  the cli admin interface is as nearly identical to the gui  interface, as you can have a CLI  and a GUI interface be the same..    BIG plus in my book.
22:47.59QwellAfter SUSE was invented and was awful, I wrote it off.
22:48.12sunfoneheh
22:48.24drmessanoIt's like when the kid you hate down the street buys something and it just ruins it for you
22:48.28Qwelldrmessano: I'm running out of distros to write off...
22:48.53brainiacI've done very well with Arch
22:48.56linuxgeckoQwell:  still got LFS on the list ??:)
22:48.58drmessanoQwell, you can't write off CentOS without a Delorean and a Flux Capacitor.
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22:49.14drmessanoI saw a presentation this weekend on LFS
22:49.15Qwelldrmessano: Years of psychiatric help.
22:49.45drmessanoIt was like "Why buy ground beef when you've got this cow pasture and this pointy stick"
22:49.52Qwelllinuxgecko: If I was going to create my own distro, I wouldn't use the LFS rules.
22:50.25linuxgeckodrmessano: for a long time ( when stage1 installs were still supported,) gentoo felt like LFS with training wheels :
22:50.28linuxgecko:)
22:50.36drmessanoNo, that's slackware
22:50.44WiretapWorkI just find ubuntu amusing because the next release has the same name as my company XD
22:50.48QwellI hear slackware is getting pam support next year.
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22:51.17linuxgeckogrrr.. i hat ethis configure error from iksemel.    where do i find/fake it???
22:51.18Qwellscrew it, I'm switching to yggdrassil
22:51.35drmessanoQwell, I hear Slackware is no longer going to be distributed on 344 floppy disks coming in 2013, thanks to Compact Disc support
22:51.43WiretapWorkrofl
22:51.54Qwelldrmessano: You shut your lying face.
22:52.07linuxgeckochecking for libgnutls-config... no
22:52.08linuxgeckochecking for libgnutls - version >= 0.1.0... no
22:53.02drmessanoSeriously, CentOS.. how long does it take to download sources, rip out some logos (a process you should have scripted years ago) and run a MAKE
22:53.13michael-i...and I'm back. Any pointers on why a CDR value in 'userfield' will not be saved? I'm setting with this code (http://pastebin.com/9LUtti6E) and using 1.8.4 with sqlite3 custom.
22:53.37michael-iIn my example I've named it recordingcount...but just tried with the standard userfield and no dice
22:54.50drmessanoI should build my own distro and release it under the "Lesser Don't Copy My Work Or I Will Shoot You In The Face License"
22:55.02WiretapWorklol
22:56.40linuxgeckodrmessano:  LOL :)
23:02.14linuxgeckooh guys..     fyi..  i found the kludge that will fix this, for suse 11.4. might be nice to have around if someone else like me has this issue
23:02.32linuxgeckohttp://code.google.com/p/iksemel/issues/detail?id=29
23:03.16linuxgeckosounds like the iksemel guys should fix that in the ./configure at some point :)
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23:08.11voxterAnyone experience an issue with polycom ip601's (firmware 3.1.7/asterisk 1.4) where after a call is answered, the person hears ringing tones continue in their earpiece?
23:12.46drmessanolinuxgecko,  I'll keep that in mind.  Don't know a lot of folks running Suse, but Iksemel questions are becoming more and more common
23:15.56linuxgeckodrmessano:  that's probably because it's the key to google voice asterisk calling :)
23:16.29linuxgeckodrmessano:  " why the #@$@#$@#$ can't i get this $@#$# thign working ...??!?!?" :)
23:19.27michael-iNothing I set with Set(CDR(blah)=shibby) is being recorded in this sqlite3 backend...even though testing the variable works in the dialplan. Any gotcha's?
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23:37.27michael-iA HA! I see that using Set(CDR(blah)=shibby) in a macro has ZERO effect on the CDR...
23:37.33kieppiehi guys
23:37.52michael-iEveryone knew that, right?
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23:38.56kieppieI'm trying to connect a Nokia SIP node to my Asterisk (Askozia variant), but getting failed registration issues. I'm not all that familiar with shell, so would appreciate some help getting connected, please
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23:39.46kieppiepopped the shell: `asterisk -vvvvvvvvvvr`
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23:42.11michael-i...and it seems my struggle today is related to an open issue: https://issues.asterisk.org/view.php?id=14662
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23:50.29michael-iAnyone have an alternative to using Macro() from features.conf's applciationmap which will let me execute a few lines of dialplan and set a CDR field successfully? Macro does not set CDR correctly
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