IRC log for #asterisk on 20101012

00:06.14*** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net)
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01:39.54*** join/#asterisk jmmills (jason@204.16.200.135)
01:40.46jmmillscan someone explain to me why a sip provider would be sending (or asterisk be receiving) a plus symbol in an incoming calls extension?
01:41.50p3nguinSome weird providers think that's how it is supposed to be done.
01:42.29p3nguinYou have two choices.  Ask them to change it, or accept it as the extension they send calls to (change your dialplan).
01:43.13jmmillsis there a way I can just strip the leading + early in the context?
01:44.25p3nguinI wouldn't bother.  Just change your inbound extension.
01:44.50p3nguinIn such a condition, it's going to be purely cosmetic.
01:44.58p3nguinNow in a caller ID number, it's annoying for a few reasons.
01:47.33Guggein callerid i just change + to 00
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01:52.26*** join/#asterisk FreezingCold (~Arctic@unaffiliated/freezingcold)
01:54.16FreezingColdhttp://en.alldaytalk.com/Cell/ < Can I do something like this with a Asterisk box?
02:06.39p3nguinfreezingcold: Absolutely.
02:07.34p3nguinIt's very simple to do either type.
02:07.54FreezingColdI want to do the first
02:08.12p3nguinDo you have a cell phone plan with fave 5, my 5, my circle?
02:08.31FreezingColdYeah, I can get one number unlimited for $7/m
02:08.33FreezingColdI've very new to Asterisk, any guides?
02:08.39FreezingColdOr do you know SIPSorcery?
02:08.39p3nguin~book
02:08.39infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
02:09.28p3nguinHere's what I would do:  get a free DID with sipgate or ipcomms, add that number to your free airtime on your cell plan.
02:10.01p3nguinAre you wanting to make your outbound calls over a landline at home, or using an ITSP?
02:10.07FreezingColdYeah, I already kinda have that.  I have a freephoneline account (with SIP settings)
02:10.09FreezingColdI'm in Canada
02:10.18FreezingColdOn my cell phone
02:10.27MaliutaFreezingCold: where in canadia?
02:10.31FreezingColdOttawa
02:10.35FreezingColdUsing Telus
02:10.50MaliutaFreezingCold: My parents live in Fort MacMurray alberta
02:12.08FreezingColdNever been there
02:12.10p3nguinYou'll make free calls to your Asterisk system from your cell phone... then your Asterisk system will make phone calls out to someone else at a lower cost per minute than your airtime.
02:12.17FreezingColdYeah, that's the idea
02:12.24FreezingColdNow how would I go about doing that?
02:13.16MaliutaFreezingCold: is there a reason you don't just SIP from your phone directly into the * system?
02:13.50FreezingColdBecause I don't want to use up data on my cell phone
02:14.50p3nguin(2110.00) <p3nguin> Are you wanting to make your outbound calls over a landline at home, or using an ITSP?   <--- still waiting for the answer for this.
02:15.02FreezingColdITSP
02:15.13p3nguinDo you have one with low per-minute rates already?
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02:15.28FreezingColdI get unlimited in Canada & USA for free
02:15.29FreezingColdso yeah
02:15.37FreezingColdway cheaper then 20 cents a minute
02:15.59p3nguinAlright, so all you have to do is configure your DID onto your Asterisk system...
02:16.18p3nguinDevise a simple authenticate/DISA dialplan.
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02:16.28FreezingColdWhich isn't simple for someone like me =P
02:17.04FreezingColdAny premade type stuff?
02:17.05p3nguinI can do it for you.  Give me a minute to write an example.
02:17.09FreezingCold=0
02:17.10FreezingColdThanks!
02:17.44FreezingColdOn a totally different topic, should I put my ATA in DMZ?
02:18.01FreezingColdI've been having issues with one way audio sometimes
02:18.25p3nguinno
02:18.31FreezingColdWhy not?
02:18.34FreezingColdWhat's wrong with DMZ?
02:18.38p3nguinYou should never use DMZ.  Configure the port forwarding properly.
02:18.44FreezingColdalready did
02:18.50FreezingColdstill have the issue
02:19.06FreezingColdhttp://sipsorcery.wordpress.com/2009/08/05/nat-rtp-and-audio-problems/
02:19.33FreezingColdThat's what's happening to me
02:26.27p3nguinfreezingcold: http://pastebin.com/LUSCAdY1
02:26.55p3nguinfreezingcold: Let me explain it.
02:27.27FreezingColdalright
02:27.55p3nguinfreezingcold: This dialplan says that your DID number is 8005551212 (put your actual DID here).  If someone calls it, it will hangup, unless the callerID matches 3149691077 (use your cell phone number here).
02:28.14FreezingColdgo on
02:28.50p3nguinfreezingcold: If the caller ID matches, then it will run the Authenticate() command, which asks you to enter a PIN.  In my example, the PIN is store in the ast database (you can type the actual PIN in the dialplan if you want).
02:29.07p3nguinfreezingcold: If you provide the right PIN, it will give you a dial tone.
02:29.07FreezingColdgot it
02:29.34p3nguinCrap, I just realized I missed the underscores on the pattern matching for the magic context.
02:29.37FreezingColdI doubt you know anything about it, but could I port that plan over to SIPSorcery?
02:29.51p3nguinhttp://pastebin.com/MtF4rKU7    corrected
02:29.54FreezingColdThat way I don't need to leave my PC on 24/7
02:30.18p3nguin(I have a tendency to forget my underscores.)
02:31.18p3nguinWhen you get that dialtone, you dial the number.  DISA() sends the number into the [magic] context, where your Asterisk system dials out using a SIP peer by the name of your-itsp-peer in this example.
02:31.55p3nguinThat's the basic way to duplicate Plan A - Magic Number.
02:32.03FreezingColdNice
02:32.20FreezingColdHow could I take incoming calls as well?
02:32.35p3nguinI probably wouldn't ever give anyone your magic number, which you added to your free airtime.
02:32.52p3nguinIf you want to use it for dual-purpose, you just change the dialplan.
02:33.16p3nguinRather than running Hangup() when someone else calls, branch into something more useful... which does not provide them the dial tone.
02:33.25FreezingColdyeah
02:35.30p3nguinIt's pretty easy to change that behavior.  Instead of the hangup, you could use a Goto() to send them into a useful context.
02:35.45FreezingColdHave you used SIPSorcery before?
02:36.25p3nguinNope.
02:37.03FreezingColdHmm, guess I'm on my own for porting it
02:40.52p3nguinfreezingcold: Here's a variation for allowing other people to call in without having the ability to call back out:  http://pastebin.com/X5xetHvC
02:42.18p3nguinYou _could_ technically use this concept to set up something similar for friends/family giving everyone his own PIN and dial-out rules.
02:44.46Gibbyp3nguin, got it working thanks, can I add a 2nd D() option to dial a 3rd option?
02:45.35p3nguinfreezingcold: You could also do away with the Authenticate() command and use DISA's own PIN system from a file which determines the context based on the PIN entered.
02:45.50p3nguingibby: I actually don't know.  I never tried that.
02:46.13Gibbyok, i will let you know thanks
02:52.43Gibbywhat is the package for the CLI? i can issue asterisk -r and it connects but no other commands work then
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03:12.10p3nguingibby: Sounds broken.
03:12.43Gibbyp3nguin, that is what i thought, trying an update
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03:33.34Gibbyi am troubleshooting why my inbound calls are not coming through, i have forwarded 5060 and 10000-20000, i do a tcp dump on my router of my external IP and I see 5060 come through but nothing else, that seems to mean an issue with my SIP provider right?
03:34.02[TK]D-FenderGibby: What does "see 5060 come through" mean?
03:34.37[TK]D-FenderGibby: And just forwarding ports isn't enough
03:34.57Gibbyi can see the initial 5060 UDP packets come in from the internet from IP of my SIP provider and the correctly get forwarded to asterisk server but I do not see any other packets even attempt to come in
03:36.14[TK]D-FenderGibby: And who says the packets you see come in are meaningful?
03:36.40[TK]D-FenderGibby: You also fail to describe what other packets you are expecting, and why.
03:37.32Gibbyfrom what I was reading the 5060 packets are kinda of the initiator but the UDP channel traffic is carried over ports between 10000 and 20000
03:38.31[TK]D-FenderGibby: Where do I see that there is even a call coming in?  Or that * is trying to answer it?
03:38.46[TK]D-FenderGibby: Of that audio is supposed to be flowing?
03:40.33GibbyI am doing a TCP dump on the interface of my external traffic on my router, I set the source or destionation as the IP of my SIP provider, during an outbound call I see some UDP 5060 packets then packets on ports between 10000 and 20000, when i try to make an incoming call, i only see the UDP 5060 packets come in and I get a busy signal
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03:51.25[TK]D-FenderGibby: Maybe, just MAYBE you should be actually looking at the content of those packets in real detail.
03:51.44[TK]D-FenderGibby: 5060 is CALL CONTROL.  This could include a telling GTFO messagin in there somewhere.
03:51.47Gibbyzuj, i will check them
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04:17.02Gibby[TK]D-Fender, do you want to look at these packets?
04:17.48[TK]D-FenderGibby: Do you want an opinion on them?
04:17.55Gibbythat would be great
04:20.57Gibbytrying to send via IRC
04:20.59[TK]D-FenderGibby: PASTEBIN
04:21.01[TK]D-Fender!pb
04:21.03[TK]D-Fender~pb
04:21.03infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
04:22.49Gibbyhttp://pastebin.com/EKFaQP20
04:23.47[TK]D-FenderGibby: Trash.  Dow go get * SIP DEBUG from * CLI
04:23.50[TK]D-FenderNow*
04:24.24p3nguinWhat the heck is that from?
04:24.47Gibbytcpdump of raw data packets, easily viewed in Wireshark
04:26.30[TK]D-FenderGibby: Don't care about wireshark.  No go get a full capture of the attempt from * CLI with SIP DEBUG enabled
04:27.58Gibbygetting it now
04:30.06Gibbylooks like NAT issue, thanks guys
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05:08.52Gibbyinternet dropped, don't know if you got my last msg but, i see it NAT'ing now, still seeing SIP/2.0 401 Unauthroized though
05:09.36p3nguinAuthorize it.
05:10.23[TK]D-FenderNo.  Over half an hour later and still no pastebin as requested.  To late for help on my side.
05:10.27[TK]D-Fenderheads off to bed
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05:15.58GibbyAuthorizing it would be in the sip.conf right?
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05:17.50p3nguinIf the problem is that you don't have a peer matching what is sending the call, yes.
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05:19.37Gibbyok
05:20.07Gibbythat is the type= in sip.conf?
05:22.03Gibbylooks like options of type= are peer user and friend
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05:36.19Gibbyfyi got it working, took everything out of my router that I had configured for Asterisk port forwarding and sip, setup sip_nat.conf and added the following to my sip.conf, changed type to friend and added nat=yes, qualify=yes, insecure=very, dtmfmode=auto, dtmf=inbound, disallow=all and allow-ulaw&gsm
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05:55.27ssh-addgood morning
06:03.16radenhow can i tell asterisk to only process call files from 9 am to 6 pm ?
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06:27.22kaldemarraden: set the file creation time between 9 am and 6 pm before moving it to the spool directory.
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06:39.36ssh-addraden you can also do it in the dialplan
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07:01.23verywisemanwhat are family and key operators which are found in DB(family/key) function?
07:07.30kaldemarverywiseman: what you want them to be.
07:09.21kaldemarthe database has families with key-value pairs in them. by adding stuff you define the family and the key names at the same time.
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07:40.24hurdmanhi
07:41.35ssh-addhello
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08:07.34hrhrhris 1.6 feature frozen now? all updates being bug fixes?
08:13.20wdoekes2I've been told it is
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08:33.55hrhrhrchangelog doesn't make for easy reading does it
08:34.04hrhrhr10 merged with svnfeed
08:34.05hrhrhrgoto 10
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08:45.45nunnersSorry to ask a freepbx question, but it's a little but urgent... does anyone know of a problem when setting up ring groups on freepbx 2.8.0.3 with the dialparties.cgi script? I keep getting utils.c ast_carefulwrite: write() returned error: Broken pipe
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09:07.36SeTTleRhi
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10:29.43eMBeegood evening
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10:55.46rayno_bHi there everyone, I need some assistance.  My system is not showing the number where an incoming call comes from.  I have no clue how to correct this.
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11:03.22kaldemarrayno_b: how and where do you expect it to show the number?
11:04.23rayno_bkaldemar -> when the call is received (in CLI) it comes in from ''.  I'm expecting that DAHDI is not getting the number correctly from the Telco.  Eventually, I'd like to see the number where a call comes from on the SIP phones receiving the calls.
11:05.18kaldemarwhat technology are you using the interface telco? analog? PRI? BRI?
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11:06.56rayno_bBRI (Digium B410p Card)
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11:09.35kaldemarsee parameters usecallerid, cidsignalling and cidstart in chan_dahdi.conf.sample. "pri intense debug span X" (unless the command has changed in recent versions) will show you if there is a number to show in the signalling.
11:13.18WIMPypri set debug 2 span X
11:13.54rayno_bis this on the CLI?
11:14.00kaldemarrayno_b: yes
11:16.20rayno_bI get: No such command 'pri set debug 2 span X' (type 'help pri set' for other possible commands)
11:16.58kaldemarreplace X with your span number
11:16.59WIMPyThen it's the old version: pri intense debug span X
11:17.18WIMPyErr. Yes
11:17.47kaldemarpri show ... will get you the right span number. use tab completion.
11:18.31rayno_bwas: pri intensive debug span 1
11:18.35rayno_blet me phone in
11:20.37rayno_bI grepped the log file now for that number, but I don't think it's coming in with the number.  Could DAHDI be configured wrongly?
11:21.36WIMPyIf you don't get it, you don't get it.
11:23.16kaldemarthat's what the telco is sending you.
11:26.48SeTTleRthis should be stored in ${CALLERID(num)} correct? maybe you can explicitly print that to the console with the Verbose() app
11:27.02SeTTleRfrom your dialplan
11:27.30SeTTleRexten => s,n,Verbose(call came from ${CALLERID(num)})
11:27.34SeTTleRfor example
11:27.56SeTTleRthen you know, you don't get a callerid
11:28.11WIMPyIf there's nothing in the debug, it won't get into any variable.
11:28.27SeTTleR:-)
11:28.27SeTTleRthat's true
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11:37.04rayno_bI just want to make 100% sure - It can't maybe mean that there's just something wrong with DAHDI config?
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12:01.08SeTTleRis there a way to define custom log levels in asterisk? i want to create log entries from the dialplan with Log() and they should end up in separate files. of course, i can separate them with syslog, but maybe this can be done easily with asterisk
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12:33.05*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-rc3 (2010/10/07), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon
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12:45.47nicola_pavhello. I have asterisk servers. I connected two ip phones. I can make call from one another. everything is fine. upon running tcpdump on asterisk i get icmp unreachable errors
12:46.08nicola_pavtunning wireshark it seems icmp port unreachable code 3
12:46.15nicola_pavhow can i solve this?
12:48.25[TK]D-Fendernicola_pav: What are you doing with wireshark and it sounds like the call is WORKING, so what is the actual problem?
12:50.18nicola_pav[TK]D-Fender: the call is working yes
12:50.26nicola_pavbut why i am getting those icmp messages?
12:50.38nicola_pavi want a normal call flow without any errors
12:50.57SeTTleR? where do you get those messages? in wireshark? who cares?
12:51.03[TK]D-Fendernicola_pav: If your call is fine then you simply don't know how to use wireshark or don't know what you're supopsed to be looking for with it
12:51.15nicola_pavi run tcpdump and every 5 seconds i am having them
12:51.23nicola_pavi just analyzed it more with wireshark
12:51.26SeTTleRmaybe you have icmp blocked by a firewall or turned of somewhere in the proc/ fs
12:51.55nicola_pavis it normal while in the call flow to have every 5 seconds unreachable icmp errors?
12:52.10nicola_pavi dont want those errors to appear
12:52.27SeTTleRmaybe you have icmp blocked by a firewall or turned of somewhere in the proc/ fs <--
12:52.47nicola_pavwhat do u suggest?
12:52.55SeTTleRi really don't think that this is an asterisk issue
12:52.58nicola_pavwhat shall i do with proc/?
12:53.08nicola_pavis it ip phone issue?
12:53.39[TK]D-Fendernicola_pav: MAYBE YOU SHOULD show us
12:54.23nicola_pavof course
12:54.30SeTTleRwhat does cat /proc/sys/net/ipv4/icmp_echo_ignore_all say?
12:54.36SeTTleRon both machines
12:54.53nicola_pavi can run it on asterisk
12:54.56*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
12:54.57*** mode/#asterisk [+o malcolmd] by ChanServ
12:54.58nicola_pavthe other is ip phone
12:54.58SeTTleRif one says 1, then you have the problem
12:55.10SeTTleRok, try that
12:55.14nicola_pavon asterisk
12:55.16nicola_pavits 0
12:55.29nicola_pav14:39:14.396368 IP 192.168.199.11 > 192.168.199.201: ICMP 192.168.199.11 udp port 11794 unreachable, length 208
12:55.35nicola_pavthis message is from tcpdump
12:55.50nicola_pav192.168.199.11 is the IP address of the ip phone
12:56.14nicola_pavi get such messages each 5 seconds
12:56.32SeTTleRhuh.. i don't know whireshark that well, but why is there something with udp?
12:56.52nicola_pavthis message is from tcpdump
12:57.09nicola_pavi captured another call with wireshark to analayze more
12:57.17nicola_pavit seems CODE 3
12:57.23nicola_pavwhich means port unreachable
12:57.37SeTTleRwell this is not icmp not reachable
12:58.01nicola_pavit is
12:58.08nicola_pavicmp unreacable and code 3
12:58.11SeTTleRi don't know code 3 or something.. the error message above says something about udp?!
12:58.14nicola_pavwhich means port unreachable
12:59.06nicola_pavi got it from wireshark
12:59.24nicola_pavmaybe i am wrong
12:59.31nicola_pavi need to stop those messages
12:59.49SeTTleRi think the problem is another one.. the asterisk tries to reach the ip phone with udp port 11794 and this is closed on the phone which returns with the icmp message from above
12:59.50[TK]D-FendernicDo you?  You haven't told us a PROBLEM this is related to.
13:00.27nicola_pavi am afraid i did understand u
13:00.42nicola_pavy not stopping those messages?
13:00.48nicola_pavthey appear each 5 seconds
13:01.07SeTTleRcan be some kind of misconfiguration
13:01.20SeTTleRrtp ports or something.. i don't know
13:01.37SeTTleRasterisk tries to reach phone on that port.. port is closed -> message appears
13:01.46nicola_pavany hint?
13:02.22SeTTleRcheck the rtp ports in sip.conf and on the phone
13:03.32nicola_pavon the phone there is this parameter
13:03.40nicola_pavudp keep alive-message
13:03.47nicola_pavdoes it play a role?
13:03.54SeTTleRmaybe
13:04.56SeTTleRi don't know what phone you are using, but this sounds like "i am sending a udp message periodically every 5 seconds and trigger a message in wireshark with that" :D
13:04.57nicola_pavits disabled
13:05.03nicola_pavi will enable it and see
13:05.29nicola_pavbecause it happens each 5 seconds
13:05.42nicola_pavu say that sth is sending periodically?
13:05.44nicola_pavright?
13:05.53[TK]D-Fenderqualify=yse <-
13:05.55[TK]D-Fenderyse
13:06.34nicola_pavqualify=yes for the account?
13:06.37nicola_pavi will check now
13:08.34nicola_pavi have qualify yes already
13:09.19[TK]D-Fendernicola_pav: turn it OFF and see if it goes away
13:09.28nicola_pavok
13:10.55nicola_pavsame
13:11.02nicola_pavturned it off and same
13:11.05nicola_paveach 5 seconds
13:11.34[TK]D-Fendernicola_pav: Aside fromt he fact you can see that message in wireshark, is there an actual PROBLEM?
13:12.02nicola_pavare those messages normal?
13:12.13nicola_pavfor me they r not
13:12.15[TK]D-Fendernicola_pav: Maybe.  Is there a PROBLEM?
13:12.16*** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com)
13:12.26nicola_pavdepends :)
13:12.35nicola_pavfor me. there is a problem
13:13.50[TK]D-Fenderwrites nicola_pav off as the problem and moves on to more productive matters
13:14.50nicola_pavy can't anyone help me to figure it out?
13:15.50*** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk)
13:16.02[TK]D-Fendernicola_pav: So far you are looking at the RESULT, and not the CAUSE
13:16.39*** join/#asterisk UQlev (~Yuriy@212.50.99.8)
13:16.54nicola_pavcan anyone direct me to the cause?
13:19.17*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
13:20.35SeTTleRwell this is simple network debugging. look, what packet causes the phone to send the icmp message, observe with netstat, what process tries to send that packet and there you are..
13:21.43nicola_pavwhat options should i run with netstat?
13:22.08*** join/#asterisk sulex (~sulex@dynamic-adsl-94-39-192-128.clienti.tiscali.it)
13:26.47*** join/#asterisk telnettech (~telnettec@216.49.139.56)
13:27.53radenhow do i set the maximum amount of calls being placed at once for call files ?
13:28.01*** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt)
13:30.02[TK]D-Fenderraden: No such thing.  Stop placing them all there at once yoursle
13:30.04[TK]D-Fenderf
13:30.48radendoes each call need to be a seperate file ?
13:31.50radenso I should be basically having PHP write a file and then cron script move 1 file every minute to asterisk spool ?
13:32.12hurdmani'm looking for a sphinx 4 integration into asterisk, i have tried some tuto (and patch) found in the web, but it seems to be out-of-date, anyone have a recent link ?
13:32.22KattyHERRO WORLD
13:33.03[TK]D-Fenderraden: Create wahtever scheduling mechanism you feel like but if there are multiple in there they will ALL egt processed immediately
13:33.17[TK]D-FenderKatty: Mew.
13:33.42raden[TK]D-Fender, so if i put 4,000 calls in there they will all start calliing ?
13:33.50radenMorning Katty :)
13:34.14[TK]D-Fenderraden: Yes
13:34.40[TK]D-Fenderraden: And that many simultaneous might be a nasty load for your server and it could have a heart attack
13:34.54radentotally understandable
13:36.23*** join/#asterisk ariel_ (~chatzilla@63.214.236.169)
13:36.35hrhrhris there a way to remove the local card from the asterisk server for pri connections
13:36.44Kattyhi raden
13:36.46Kattyhugs raden
13:36.48hrhrhri.e. use some external device for interfacing with it
13:36.48Kattyhugs [TK]D-Fender
13:36.59Kattywould anyone like half a headache?
13:37.04hrhrhrhi fives Katty
13:37.12Kattyhrhrhr: Hello.
13:37.18[TK]D-Fenderhrhrhr: For interfacing with the PRI... or the CARD?
13:37.39hrhrhrinterfacing with the pri
13:37.43hrhrhran external card, if you like
13:37.44hrhrhror something
13:39.06[TK]D-Fenderhrhrhr: There are tons of SIP gateways, etc.  All very pricey by comparison
13:39.39hrhrhri want to retain the pri local interface
13:39.44hrhrhrbut
13:39.52hrhrhri dont want to have to open a machine if the pbx breaks
13:40.01hrhrhrnor do i want to buy 2 * pri per site
13:40.08*** join/#asterisk timahvo1 (~rogue@41.72.215.94)
13:40.12hrhrhr2 * pri card rather
13:40.42WIMPyhrhrhr: And what do you do if the gateway breaks?
13:40.47[TK]D-Fenderhrhrhr: Why have you jsut introduced multiple SITES into this picture?  Please clarify your situation
13:41.29hrhrhrhow does one build a hot/warm standby pbx
13:41.35hrhrhrwhen the interface card lives in the machine
13:41.51hrhrhrof course, if the interface card/device dies, that's something else
13:41.57hrhrhrbut i want to cover machine/server failure
13:42.22[TK]D-Fenderhrhrhr: Then get a PRI>SIP interface
13:42.40[TK]D-Fenderhrhrhr: eg AudioCodes Mediant series
13:43.46hrhrhrok cheers
13:48.45creativxman I am going crazy trying to figure out what is chopping up our SIP audio
13:53.41*** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2)
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14:00.09*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
14:04.24KattyNaikrovek: poke
14:05.17Naikroveksup
14:05.37Naikroveksorry lot of stuff going on today
14:05.48Naikrovekstressful stuff
14:07.11*** join/#asterisk Marquel (~Marquel@84.16.240.149)
14:07.18Marquelmorning.
14:07.32Marquelit seems my asterisk has a problem using time-triggered includes.
14:09.02Marquelsyntax for asterisk 1.6 is 'include => newcontext|08:00-10:00|mon-wed|*|*' to include newcontext every monday, tuesday and wednesday from 08:00 to 10:00, right?
14:09.06KattyNaikrovek: i saw your fb thing. was just curious if you were doign ok
14:12.20*** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-146-67-241.mia.bellsouth.net)
14:15.36*** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com)
14:15.37*** join/#asterisk xoveruk (~xover@193.220.59.2)
14:15.41xoverukhi all
14:18.02*** join/#asterisk theHub (~karl@69.177.93.21)
14:19.13*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
14:20.12*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
14:22.00*** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-212.dsl.net.pk)
14:22.11*** join/#asterisk krion (~seb@unaffiliated/krion)
14:22.21xoverukIf calls are conneting but then dropping after 20-30secs, what would that suggest is the issue or where should I start looking or clues?
14:23.01mechbangirchi one of my client says he has a shared ip so he can not forward port 5060 to an internal asterisk server. i do not know much about networking and the whole blame is on me
14:24.06xoverukmechbangirc: how can he have a shared IP, does he mean dynamic?
14:25.06*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
14:25.31mechbangircxoveruk: i doubt his knowldge and skill. i told him if he has dynamic ip its still can be done. though he keep saying it is not possible i dont have static ip my ip is sharred!!! i dont understand
14:25.58xoverukhe has a DHCP address from his IP.
14:26.05NaikrovekKatty: not really
14:26.06Naikrovekthanks though
14:26.32xoverukSo he needs to configure DDNS on his internet router and then use that DNS address for configuring the trunks or SIP accounts.
14:26.50Naikrovekpeople find what they can do to stress me out, then they do that one single thing only, over and over and over and over and over and over and over and over
14:27.11Naikrovekafter i tell them to stop, after i tell them what else to do, after i tell them that it's stressing me out
14:27.12Naikrovekover
14:27.13Naikrovekand over
14:27.17Naikrovekand i'm tired of it
14:27.27*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
14:27.36WIMPyNaikrovek: Isn't it nice if you can count on people? *eg*
14:27.56Naikrovekthe longer i live the more i see that you can't count on anyone for anything at all
14:28.02Naikrovekeven the simplest things
14:28.07xoverukNaikrovek: I 2nd that
14:28.14Naikroveki even have to tell my wife what to EAT half the time
14:28.17Naikrovekcan't you feed yourself
14:28.29Naikrovekcan FEEL his hair greying
14:28.48xoverukmechbangirc: Do you understand what I am suggesting?
14:28.50WIMPyYou still got hair?
14:29.50leifmadsenNaikrovek: amen :)
14:29.57leifmadsenNaikrovek: become a recluse -- it's easier
14:30.07mechbangircxoveruk: sorry i was away. internal server is on static ip (172.16.1.2) or so.
14:31.17p3nguinmechbangirc: I assume it's a shared server with only one IP address.
14:31.23*** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn)
14:31.41drmessanomechbangirc, are they using a wireless ISP?
14:31.52mechbangircxoveruk: on wan side his ip is dynamic i dont understand sharred ip at all.
14:32.01mechbangircdrmessano: no it is wired
14:32.23drmessanoWho is the provider?
14:33.51mechbangircdrmessano: hey i know for a fact the provider provides dsl connection not cable or something so how can it be sharred.
14:33.54p3nguinmechbangirc: (0931.17) <p3nguin> mechbangirc: I assume it's a shared server with only one IP address.
14:34.40mechbangircp3nguin: no it is a dedicated asterisk server with one private address
14:35.48xoverukhow can you have a shared IP, doesnt make any sense, how can you route internet traffic
14:35.55drmessanomechbangirc, just guessing, maybe he's got a single port DSL modem/router, and has a router behind it..
14:36.23drmessanoJust trying to guess what the client is talking about.. who knows how much he really knows
14:36.35xoverukcan someone tell me what the valid comments forms for asterisk config files. ';' and '#'?
14:36.51p3nguinmechbangirc: Then it's not a shared IP address.
14:37.06p3nguinmechbangirc: Tell him to learn some networking terminology before saying he can't comply with your request.
14:38.03mechbangircp3nguin: i guess i have to tell him. its hard to tell
14:38.18schmidtsxoveruk its ;
14:38.24p3nguinhttp://www.100best-free-web-space.com/articles/paid-web-hosting/shared-or-dedicated-ip-addresses-44.html
14:38.57p3nguinshared means SHARED, or not dedicated.
14:39.15p3nguinDynamic and shared are mutually exclusive terms.
14:39.50p3nguinYou could have a shared DYNAMIC address, or you could have a shared STATIC address.
14:40.09p3nguinSimilarly, you could have a dedicated dynamic address, or you could have a dedicated static address.
14:40.10*** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831)
14:40.25xoverukschmidts: what does the '#' denote/
14:40.53leifmadsenxoveruk: DAHDI related files use # for comments
14:40.56leifmadsenthe rest use ;
14:41.09leifmadsenit's a historical thing
14:41.50mechbangircp3nguin: i think shared add is good for web hosting companies not for small companies who do not provide public services over internet
14:41.51xoveruk@leifmadsen: so both are valid comment files, the later referring to relating files?
14:42.04leifmadsenxoveruk: they are both valid in the same file
14:42.10leifmadsenxoveruk: oops wrong
14:42.21leifmadsenxoveruk: I meant, they are not both valid in separate files, but not the same file
14:42.40mechbangircp3nguin: i would throw this stuff on him
14:42.43leifmadsenxoveruk: some files use ; while others use #. The ones that use # are pretty much all DAHDI related (i.e. chan_dahdi.conf)
14:43.04drmessanoSo DAHDI likes hash?
14:43.04leifmadsenxoveruk: you can tell which file uses what by looking at the sample files
14:43.12leifmadsendrmessano: who doesn't?
14:43.18drmessanoGood point
14:43.29leifmadsendrmessano: people I don't want to hang out with, that's who!
14:43.38schmidts:d
14:43.44drmessanolol
14:43.59*** join/#asterisk S-chum32 (~S-chum32@68.188.5.142)
14:44.06leifmadsenok time to get focused and get some work done before lunch
14:45.04drmessanoThere has to be an additional joke in there about DADHI, hash, and a semi-colon
14:45.21p3nguinmechbangirc: There's also an exclusivity between public and private addressing.
14:45.21drmessanoMy head is too stopped up to wrap around it
14:45.52hrhrhrwhat are your thoughts on tdmoe as opposed to a pri<>sip gateway
14:46.05WIMPydrmessano: Head already on dahdi-land?
14:46.34WIMPyin
14:46.37*** join/#asterisk [cannibalera] (~cannibale@201-41-239-137.fnsce703.dsl.brasiltelecom.net.br)
14:46.43p3nguins/addressing./addressing, and shared and dedicated./
14:46.55SeTTleRpigpen, in case you like to know: the problem with system() yesterday was fixed. was my fault, as the sh binary was not in the chroot. confusing error messages from asterisk though... but thanks for your help :D
14:47.36schmidtshrhrhr direct connection needed which can be hard if your customer wants a fallback system on different positions like min. 5 miles away ;)
14:47.54*** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-110.dsl.net.pk)
14:48.42pigpenSeTTleR, sure...glad to see you found it.
14:49.09SeTTleRme too :D
14:49.10WIMPyJo, forget about those ideas of using vi as login shell. Asterisk as login shell rulez!
14:50.44drmessanoWIMPy, my head feels like someone is crushing it with Asterisk 1.2, I have Zaptel running out my nostrils, and I am running to the restroom every 5 minutes to take a trixbox
14:52.56*** part/#asterisk [cannibalera] (~cannibale@201-41-239-137.fnsce703.dsl.brasiltelecom.net.br)
14:53.05kaldemarprescribe yourself 1.8 mg's of asterisk and some DAHDI.
14:53.29SeTTleR<PROTECTED>
14:58.09drmessanolol
14:59.40Naikrovekleifmadsen: you're no recluse.
15:00.20leifmadsenNaikrovek: shut your mouth
15:00.28theharnom nom donut!
15:00.33Naikrovek...
15:00.37leifmadsenthehar: nom nom is so 2007
15:00.48thehareat my ****
15:00.52thehar****sucker
15:00.54theharyeap
15:01.04*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
15:01.14leifmadsenand then we pink bellied The_Boy_Wonder!
15:01.20leifmadsenohai!
15:01.24theharlol
15:02.09*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
15:03.17p3nguinleifmadsen: So what's the new method of devouring yummy foods?
15:03.18*** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
15:03.23leifmadsenp3nguin: chewing?
15:03.29theharchewing is overrated
15:03.36leifmadsenI personally just thing nomnomnom is retarded in a bad way
15:03.42leifmadsens/thing/think
15:03.52*** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
15:04.11drmessanoYeah, nom nom sounds like you're ......
15:04.15p3nguinIs anything ever retarded in a good way?
15:04.22drmessanoAYBABTU was
15:04.36theharp3nguin: yes leif
15:04.39drmessanoI think even O RLY made sense
15:04.41theharcomma that somewhere
15:04.51leifmadsenO'Reilly?! O RLY!
15:05.01drmessanoO RRY was my fav
15:05.01p3nguinorally
15:05.30drmessanoBy the time we got to OMGWTFBBQ, I knew O RLY was over
15:05.40leifmadsenlol
15:05.53theharlol
15:09.29Kattydear universe.
15:09.35*** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net)
15:09.45Kattyplease stop tortuing me. my stomach hurts, that's enough now. can we please make up and be friends now?
15:09.48Kattylove, katty
15:09.57thehargets his magical wand
15:10.07theharuses it on Katty
15:10.08theharfixed.
15:10.13Kattyhoray!
15:10.35*** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
15:10.40*** part/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net)
15:13.49Marquelnobody?
15:15.02theharfor fun, I thought I'd try registering someschmuck.com, but some schmuck already owns it.
15:17.45p3nguinirony
15:17.49theharheh
15:27.09*** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net)
15:27.12*** join/#asterisk TobSnyder (~schneider@dslb-088-073-198-122.pools.arcor-ip.net)
15:28.11fullstopcan emailsubject and emailbody be set for an individual voicemail entry?  I have some vm contexts where I specify emailbody= and emailsubject=, but they are never used.
15:28.48fullstopI added some debug code to app_voicemail.c, and I don't see them being parsed..
15:29.13[TK]D-Fenderfullstop: parms aren't context based
15:30.28fullstop[TK]D-Fender: one moment
15:30.50*** join/#asterisk bsaxon (~bsaxon@12.107.149.61)
15:31.27fullstop[TK]D-Fender: http://svn.digium.com/svn/asterisk/branches/1.6.2/configs/voicemail.conf.sample
15:31.38fullstopsearch for Bianca
15:32.04[TK]D-Fenderfullstop: That's news....
15:32.30[TK]D-Fenderfullstop: seems to imply it is possible
15:32.51p3nguinfullstop: If you're using a version that supports it, why wouldn't it be possible?
15:33.09fullstopI'm asking because it is not working for me. :)
15:33.56fullstopI see where it looks for emailsubject and emailbody in the [global]
15:34.16fullstopand, I dump everything that it parses.. but it appears to only get that information from users.conf and not voicemail.conf
15:35.38[TK]D-Fenderfullstop: Sorry.. users.conf voids your warranty :p
15:35.52fullstop[TK]D-Fender: nothing in it except for [global]
15:35.55fullstoplet me remove it
15:36.23[TK]D-Fenderfullstop: http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf <--- seems to contradidict those options as being alloed at the box level anyway
15:37.37fullstop[TK]D-Fender: the patch mentioned appears as if it was applied to 1.6.x
15:38.17*** join/#asterisk Nwab (~Schnitzel@unaffiliated/benwa)
15:38.20[TK]D-Fenderfullstop: What patch?
15:38.25p3nguinspecifically, the 1.6.2 branch.
15:38.27fullstophttps://issues.asterisk.org/view.php?id=14372
15:39.18*** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net)
15:39.34[TK]D-Fenderfullstop: that patch implies REALTIME.  Are you using realtime?
15:39.43fullstopnegative
15:39.55rrb3942anyone familiar with issues between dialogic and asterisk, that can causes the rtp for the audio periodically just start/stop in a call?
15:40.07[TK]D-Fenderfullstop: then it sounds like a wish-list item that isn't real
15:40.24fullstop[TK]D-Fender: Are you sure that you are not German?
15:40.36[TK]D-Fenderrrb3942: Dialogic doesn't speak RTP last I checked.....
15:40.45[TK]D-Fenderfullstop: Entirely.
15:40.45rrb3942on a SIP call
15:40.56fullstop[TK]D-Fender: I don't see anything within the patch code which is realtime specific.  Let me look a bit further.
15:41.05rrb3942i thought they had some media gateway boxes that supported SIP?
15:43.46coppice[TK]D-Fender: its not the 20th century any more. most of dialogic's products speak RTP
15:47.18[TK]D-Fendercoppice: And he didn't specify what product he is dealing with.
15:47.37[TK]D-Fendercoppice: So 20th century is entirely possible
15:47.39rrb3942wish I knew, it is someone elses dialogic box
15:47.48[TK]D-FenderWOW... such awesome details.
15:48.04[TK]D-FenderHi, my Ford doesn't work... HOW TO FIX PLAESE!?!?
15:50.55rrb3942hey, I know the details suck, I've been trying to get information from these guys for two days now and its just circles with them
15:51.04drmessanoI can understand why
15:51.25coppice[TK]D-Fender trade it for a Honda
15:53.14neurosysor a personality :P
15:54.44[TK]D-Fenderneurosys: Putting the FUNF back in DYSFUNCTIONAL
15:54.47[TK]D-FenderFUN*
15:54.49*** join/#asterisk madduck (~madduck@debian/developer/madduck)
15:54.54neurosyshehe
15:57.20*** part/#asterisk sekil (~sekil@80.93.247.26)
15:59.53xoverukhow can I tell which port is being used for SIP?
16:00.18madducke.g. netatst -naup (5060/udp)
16:00.24madducknetstat even
16:00.48*** join/#asterisk _Sam-- (~sam@unaffiliated/sam--/x-573746)
16:01.13_Sam--i realize this is the asterisk channel, however, im wondering if anyone can give a thumbs up to the switchvox hardware devices?
16:02.13russellbgives a thumbs up
16:02.17*** join/#asterisk f00bar80 (~eng.debia@41.239.209.51)
16:02.19russellbalso works for digium ;-)
16:02.34f00bar80Just i'm asking can i have the vps hosting serving both web hosting and IP-pbx hosting if i'm gonna to install asterisk + freepbx and cPanel/whm ???
16:02.36russellb_Sam--: you can also try #switchvox
16:02.40_Sam--thank you.
16:05.06*** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby)
16:05.09wcselbyo/
16:05.17wcselbyanyone here use queuemetrics?
16:06.21*** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452)
16:07.50fullstop[TK]D-Fender: okay, I dug through the VM code a bit more.  emailsubject and emailbody are only applied if they are present in users.conf.
16:08.11fullstopConsidering something so evil, a lot of asterisk revolves around it.  :P
16:10.07*** join/#asterisk razu (~razu@razu.data.ee)
16:11.43xoverukmadduck: thanks
16:11.56xoverukI also used nmap and found the firewall settings, where can I set the port number for SIP?
16:14.10ManxPowerxoveruk, the sip port in your firewall?
16:14.49ManxPowerfullstop, you must be using some kind of GUI
16:16.03fullstopManxPower: No, I'm not.
16:16.18fullstopManxPower: look at the code.
16:16.30ManxPowerfullstop, emailsubject and emailbody do not, as far as I know, depend on users.conf
16:16.39fullstopManxPower: one moment
16:17.03fullstopManxPower: http://pastebin.com/ZihXzvq5
16:17.04ManxPowerif there is a users.conf what you see might be the case.
16:17.11xoverukManxPower: no for asterisk
16:17.12ManxPowerBut only crazy people use users.conf
16:17.18fullstopI am not using users.conf
16:17.27fullstopucfg represents users.conf
16:17.29fullstopand..
16:17.37ManxPowerxoveruk, if you do that none of your sip clients are likely to work anymore
16:17.45fullstopappy_options_full is only called if you are using users.conf or asterisk realtime.
16:17.54xoverukManxPower: For each sip device, the port numbered listed, what is that for? Connecting to the device?
16:18.04fullstopapply_options_full is what reads emailsubject and emailbody.
16:18.13*** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com)
16:18.31ManxPowerxoveruk, remember ALL IP has TWO ports.  the SOURCE port and the DESTINATION port.  The SOURCE port is chosen dynamically by the OS, the DEST port can't change or nothing will connect to it.
16:18.50ManxPowerfullstop, then file it as a bug, because it is one.
16:19.13ManxPowerthe ports listed in sip show peers would be the source port
16:20.26xoveruksource port meaning? the device that is connecting to asterisk?
16:20.36*** join/#asterisk rtert (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net)
16:20.41rterthi
16:20.44rtertis there anyone here ?
16:20.48theharnope.
16:20.50theharsleeping
16:21.00rtertok leave it
16:21.59ManxPowerxoveruk, the source of the connection, usually the device
16:22.21rtertlol
16:22.25rterthi there
16:22.28ManxPower~ask
16:22.29infobotQuestions 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.
16:22.40xoverukManxPower: ok great stuff
16:23.03rterti don't if it the write chnnel, i need an asterisk expert for a project
16:23.26rtertis there any channel for that ?
16:24.23ManxPoweryou can try #asterisk-biz if you are looking to hire a consultant
16:26.32rtertok thanks a lot
16:27.12*** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
16:29.23*** join/#asterisk jdwjr_ (~jon@66.195.223.170)
16:36.14*** join/#asterisk salviadud (~ralfalfa@189.224.63.254)
16:50.46*** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
16:57.49*** join/#asterisk b_d (~brian@mail.bsg.bm)
17:03.01*** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51)
17:03.30*** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51)
17:05.49*** join/#asterisk zplinux (~zplinux@213.8.57.217)
17:06.15zplinuxcan I get some help with the asterisk rpm spec file?
17:06.43zplinuxin line 110 I read: %{?_without_alsa:%if 0} what does that mean?
17:07.57Kattypeeks in
17:08.10Kattyohai
17:10.08Kattyzplinux: how're you today?
17:10.25zplinuxfine dear, and how are you?
17:10.34Kattyoh just peachy, thanks.
17:11.16zplinuxsay, where can I get help with the asterisk rpm spec file?
17:11.43Kattyi would imagine someone here would be able to help you eventually.
17:11.53Kattyyou will have to be patient tho.
17:12.00*** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk)
17:14.10KattyNaikrovek: we need to talk :<
17:14.49Naikroveknot really in the mood right now, katty
17:15.03Naikrovekgotta run cable through the office today
17:15.06Kattyk, well i'll take a raincheck.
17:15.06theharew
17:15.34f00bar80any comment ?
17:15.35f00bar80Just i'm asking can i have the vps hosting serving both web hosting and IP-pbx hosting if i'm gonna to install asterisk + freepbx and cPanel/whm ???
17:16.08*** join/#asterisk bsaxon (~bsaxon@12.107.149.61)
17:16.34theharare you going to do SIP signalling and RTP traffic on different ports while running both?
17:16.48*** join/#asterisk csnook (~chris@138.210.3.1)
17:21.26TobSnyderwhat is relaxdtmf for?
17:21.48[TK]D-FenderTobSnyder: Exactly what it says.  Relaxing the DTMF detection routine
17:24.08Kattyi'll relax your dtmf in a minute.
17:24.15thehari'll do it in 30 seconds
17:24.51Kattythat's quick.
17:25.13Kattyperhaps even premature.
17:25.45[TK]D-FenderThat's what SHE said
17:25.46coppiceanything below 36 weeks is considered premature
17:25.52theharlol
17:26.13Kattythat sounds painful.
17:26.26TobSnyder[TK]D-Fender: any more detail? use cases?
17:27.08[TK]D-FenderTobSnyder: Getting unreliable DTMF?  Try it
17:27.24TobSnyderjust have problems with DTMF on FXS Port
17:27.34drmessanoGet a new phone
17:27.48drmessanoTake that one back to Walmart and exchange it
17:28.10[TK]D-FenderTobSnyder: go try the option then.
17:28.10TobSnyderseems not to be related with phone, as it occurs on all analog telephone sets I have tested so far
17:28.11Kattywalmart :<
17:28.43TobSnyderand just wondered in dahdi show channel XX what "Relax DTMF: no" stands for
17:28.43Kattyi had some dtmf problems once, but it ended up somehow being motherboard related.
17:28.47*** part/#asterisk rtert (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net)
17:28.49carrarYou'd think walmart would have more walls in their store
17:29.20Kattyyes. they are the mart of wal.
17:29.37TobSnyderfirst I thought it might be a problem of double detection (as astribank an also detect dtmf) but this seems not to be the reason
17:29.38Kattystill not a fan tho. i feel like white trash the minute i park.
17:29.48TobSnyderso I am now checking relaxdtmf
17:29.48KattyTobSnyder: does it happen with a soft phone?
17:29.59drmessanoWalmart was named afer it's two founders, Walt Disney and Martin Short
17:30.01carrarKatty, you have to wear sweatpants there
17:30.05TobSnyderKatty: as far as I have tested not
17:30.05drmessanoEveryone knows that
17:30.06carrarand a white teeshir
17:30.07carrart
17:30.08Kattyi have sweat pants.
17:30.16Kattybut they say PINK on the back
17:30.18TobSnyderKatty: x-lite seemed to have no problems with dtmf
17:30.30carraryou have to wear them to walmart
17:30.33drmessanoTry with a different phone
17:30.42Kattydrmessano: he's tried with several hardware phones.
17:30.52KattyTobSnyder: what if you send digits with the dial plan?
17:30.54drmessanoGet a new FXS card
17:31.02KattyTobSnyder: and then have it playback them to you
17:31.05drmessanoMaybe you are listening too hard
17:31.11KattyTobSnyder: does it also have issues
17:31.58carrarI need to install my TV Antenna today
17:32.00KattyTobSnyder: what lvl is the dtmf getting snickerdoodled up?
17:32.09Kattycarrar: wanna set up a fta box for me?
17:32.19carrarFTA?
17:32.23Kattyfree to air
17:32.26carraroh
17:32.33carrarSURE
17:32.35carrarFly my out
17:32.38Katty:>>> k
17:32.39carrarme
17:32.44carrarI can install anything
17:32.46Kattymakes arrangements.
17:32.55carrarI have crimpers too
17:32.55[TK]D-Fender[13:30]<TobSnyder>Katty: x-lite seemed to have no problems with dtmf <- because it isn't AUDIO
17:33.14carrarand several free multimeters from harbor frieght :)
17:33.17Kattycarrar: crimpers?
17:33.26carrarto put ends on the RG6
17:33.29Kattyoh
17:33.32theharKatty: i still need to make snickerdooles
17:33.32Kattyi was thinking like....
17:33.38Kattyyou're going to do my hair?!
17:33.43carrarYES
17:33.48Kattyno.
17:33.54Kattyoh.
17:33.55carrarI might use RJ45 crimpers for that
17:33.58Kattyare you metro?
17:34.01carrarhell no
17:34.04Kattyif you are metro, you can do my hair.
17:34.16Kattythen keep your crimpers away from me.
17:34.19carrarI live in the metro downtown city for some unknown reason however
17:34.27Kattyi am sorry.
17:34.43carrarI can see 30+ Access Points from my place
17:35.02KattyEesha. i can only see 4
17:35.08KattyBut I also have trees.
17:35.15carrarI have trees
17:35.19carrar7 trees
17:35.21*** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net)
17:35.21theharlol
17:35.24Katty:<
17:35.24TobSnyderKatty: TobSnyder: what lvl is the dtmf getting snickerdoodled up? ??????
17:35.39carrar4 of them are 40+ feet
17:35.48carrarat least
17:35.48KattyOh. I forgot not everyone speaks Kat.
17:36.18TobSnyderactually when sending DTMF directly via Dialplan (e.g. as parameter of Dial Command) it worked
17:36.19KattyTobSnyder: from phone to asterisk box, from the fxs card (see log)...played back from the telco? i'd start swapping around hardware.
17:36.39KattyTobSnyder: again, that is not audio.
17:36.40TobSnyderI have created an extension that is playing back digits
17:36.54KattyTobSnyder: same with a software phone. that's not audio either.
17:37.24TobSnyder[dtmf-test-b]
17:37.24TobSnyderexten => 972,1,Dial(DAHDI/r1/01803001179,180,D(w0w7w1w6w4w1))
17:37.48TobSnyderattention, 3 lines are pasted:
17:37.50TobSnyder[dtmf-test-a]
17:37.50TobSnyderexten => 971,1,Answer
17:37.50TobSnyderexten => 971,2,Goto(saydigits,s,1)
17:37.55Kattycarrar: those aren't real trees :<
17:38.10carrarsure they are
17:38.13carrarthey are evergreens
17:38.15Katty:<
17:38.17Katty:<<<
17:38.29carrarmight be 60'
17:38.33carrarwho knows
17:40.34[TK]D-FenderTobSnyder: Your description is crap.  Where do I see the end that is FXS in there?  Where do I see a failed call?  Where do we see what the GOTO leads to?
17:40.45*** join/#asterisk angryuser_laptop (~Greg@90-156-167-83.reverse.alphalink.fr)
17:40.47Kattycarrar: http://www.flickr.com/photos/midmophil/2968628604/sizes/l/in/photostream/
17:40.55TobSnydersry
17:41.23carraryeah those are big
17:41.26angryuser_laptopGood day, when dahdi_cfg show FXS > its FXS or FXO ( it was inverted as i remember)
17:41.29Kattytrees = <3
17:42.09tzafrirangryuser_laptop: look at the output of lsdahdi
17:42.21tzafrirthe first column should normally get it right
17:42.24Kattycarrar: also http://www.flickr.com/photos/whitebuffalobk/3371824508/sizes/l/in/photostream/ <-
17:43.18carrarWhere is the photo of you hugging that tree?
17:43.18beardyHi Katty
17:43.53Kattyhi beardy
17:43.56Kattycarrar: there isn't one.
17:44.46TobSnyderso hwo to debug?
17:45.09carrarswitch to sip phones
17:45.18[TK]D-FenderTobSnyder: What debug?   Where's the failed CALL?  Where is the REST of your code?  PASETBIN <---------
17:45.19carrarand call it good
17:46.09beardyKatty: I'm about to have pizza, want some?
17:46.22carrarEpizza?
17:46.29angryuser_laptoptzafrir, thanks
17:46.52TobSnyder[TK]D-Fender: I don't have failed calls, the problem is that some IVR don't recognize DTMF correctly when calling from analog telephone set. When looking at logs you won't see anything interesting related to that problem there, as the call is directly bridged between FXS and BRI Port when it is established
17:47.03*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
17:47.19[TK]D-FenderTobSnyder: Show us this etst you are comparing
17:47.22[TK]D-Fendertest*
17:47.43angryuser_laptopTobSnyder, try to set relaxdtmf=yes
17:48.04[TK]D-FenderanyWe already told him to try it
17:48.11[TK]D-Fenderangryuser ^
17:48.16angryuser_laptopgood
17:48.19Kattybeardy: mmmm, what kind?
17:48.26[TK]D-Fenderangryuser_laptop: And he was the one who brought it up actually
17:48.39angryuser_laptop[TK]D-Fender, very good
17:49.23angryuser_laptoptry to make the dtmf duration bigger, and check if your regional settions is ok
17:49.28TobSnyderhttp://pastebin.com/2Dtu0upu
17:49.53*** join/#asterisk Tim_Toady (~moi@193.92.224.201.dsl.dyn.forthnet.gr)
17:50.32TobSnyderlet me just check relaxdtms
17:50.35TobSnyder*f
17:51.25*** join/#asterisk timahvo1 (~rogue@41.223.57.77)
17:53.05[TK]D-FenderTobSnyder: We told you to enable it half an hour agi
17:53.08[TK]D-Fenderago*
17:53.17TobSnyderwell but problem is that some digits are doubled
17:53.41TobSnyderjust thought relaxdtmf will result in more digits bying recognized
17:54.26[TK]D-FenderTobSnyder: Show us your test attempt
17:56.14TobSnyder[TK]D-Fender: you want to see full logs?
17:56.19beardyKatty: mincemeat, tomato, onion, spice tomato and garlic sauce, cheese :)
17:56.38beardyspicy*
17:57.29TobSnyder[TK]D-Fender: http://pastebin.com/zhzDRXrG
17:58.31TobSnyderrelaxdtmf did not change anything
17:58.41TobSnyderconcerning the issue
18:00.20TobSnyderargh
18:01.24*** join/#asterisk nort22 (~nort@208.104.101.168)
18:02.08TobSnyderafter restarting everythin I now encountered that relaxdtmf seems to solve the issue, I am going to do some more tests
18:02.25[TK]D-Fender.......
18:02.30TobSnydersorry I'm kinda confused
18:02.31[TK]D-Fenderhas wasted his time again
18:03.07TobSnyder[TK]D-Fender; perhaps I can spend you some beer some day
18:03.40*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
18:04.57TobSnyderok still have problems
18:05.14TobSnyderI hate those problems "sometimes it works, sometimes not" .......
18:07.21*** join/#asterisk DelphiWorld (~Delphi@41.200.13.124)
18:07.25zplinuxI still seek help with the asterisk spec file
18:07.36zplinuxwhat does %{?_without_tds:%if 0} mean?
18:07.59*** part/#asterisk DelphiWorld (~Delphi@41.200.13.124)
18:08.49Qwellzplinux: magic
18:09.28*** join/#asterisk DelphiWorld (~Delphi@41.200.13.124)
18:09.35DelphiWorld[TK]D-Fender: tu parle français?
18:09.59[TK]D-FenderDelphiWorld: Certainment
18:10.16DelphiWorld[TK]D-Fender: tré bien, je sais pas sa! je suis que vous êtes un canadiain
18:10.21fullstopleifmadsen: the patch I just submitted, you changed it to "needs license".  Is there something I need to do?
18:10.29DelphiWorld[TK]D-Fender: tré bien, je sais pas sa! je suis sure que vous êtes un canadiain
18:10.35[TK]D-FenderDelphiWorld: Il te manque un verbe la-dans....
18:10.36leifmadsenfullstop: no, just wait for the license to be approved
18:10.43fullstopleifmadsen: thanks
18:10.48DelphiWorld[TK]D-Fender: oui, j'ai just oublié
18:11.36*** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net)
18:11.38DelphiWorld[TK]D-Fender: just je veux te dit qu'on doit oublié notre problem
18:14.13*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
18:15.10[TK]D-FenderDelphiWorld: Quelle probleme?
18:15.20DelphiWorld[TK]D-Fender: qu'on a créer
18:15.52[TK]D-FenderDelphiWorld: Le seul qu j'ai note a date etait que t'es resistent a l'idee d'apprendre
18:17.25TobSnyder[TK]D-Fender: any other ideas instead of relaxdtmf ?
18:17.26DelphiWorld[TK]D-Fender: less dit sa dans un PM
18:17.48[TK]D-FenderDelphiWorld: Je m'en souviens mem-pas no plus...
18:17.51[TK]D-Fendernon*
18:18.17[TK]D-FenderDelphiWorld: Encore mieux de meme.
18:23.14*** join/#asterisk UQlev (~Yuriy@212.50.99.8)
18:24.19DelphiWorld[TK]D-Fender: le problem c'est que je suis pas bien familiar avec le format d'asterisk
18:26.42nort22Hello I'm new to this forum and have a question about asterisk
18:26.52nort22has anyone seen WARNING file.c: Failed to write frame, after  file.c: was Playing 'vm-intro.gsm'
18:27.04nort22the dial plan steps appears to complete as normal and voicemail message is stored
18:27.13nort22but i don't understand why i see "Failed to write frame" intermittently (1 every 400 calls)
18:27.42[TK]D-FenderDelphiWorld: Non, le problem c'est que vous-etait ici dupis plus qu'une an et demi et que tu refuse de LIRE.  Il-y-a autant de guides partout incluant une livre pas mal officiel et meme t'es pas rendu au meme point qu'une personne avec un ti-peut effort peut reussir dans 2 jours
18:27.50[TK]D-FenderDelphiWorld: PAS D'EXCUSE
18:28.14DelphiWorld[TK]D-Fender: ok pas de problem
18:28.32TobSnyderhttp://pastebin.com/xs3q88vh
18:29.32[TK]D-FenderDelphiWorld: "je le connait pas" n'est pas un excuse pour ne pas avoir LU LE CRISE DE LIVRE EN 18 MOIS OU MOINS
18:30.41TobSnyderI'll leave now, if someone out there has some great hints, pls let me know
18:31.10*** part/#asterisk zplinux (~zplinux@213.8.57.217)
18:31.10*** part/#asterisk DelphiWorld (~Delphi@41.200.13.124)
18:31.19TobSnydercurrently I am getting stuck with that problem
18:46.20*** join/#asterisk mandragor (~ergudicsu@70.158.116.38)
18:47.17mandragoris there a way to setup a queue so that the call is not connected unless someone in the queue group answers the call? currently I can set it up so that when they enter the queue music plays in the background
18:49.08[TK]D-Fendermandragor: With excessive trickery & scripting, yes
18:51.36mandragorbasically I have a service that sends calls to my company and I get charged if the call is answered so I have calls that are answered placed into the queue the customer then hangs up, I didn't deal with the customer but I still get charged
18:53.14mandragor[TK]D-Fender is there something you can point me to?
18:53.42f00bar80I'm asking your opinion regarding a vps serving both asterisk/freepbx and websites hosting, the vps specs are as follows BW: 500 GB and Disk space 30 GB, any advice ??
18:54.15beardyHow many percent of your future profits are you offering here?
18:55.59[TK]D-Fenderf00bar80: those mean nothing really
18:57.34*** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
18:59.20bmoraca_workje n'en sais foutrement rien!
19:00.31[TK]D-Fendermandragor: There are a lot of pieces.  Spawning new channels as place-holders in your queue, changing your agent dial's to local channels that will BridgeA() to the unanswered call's channel, etc
19:01.02ManxPowerf00bar80, the standard advice is don't run Asterisk in a VM.  If you insist on running Asterisk in a VM it might work, it might not work, it might be reliable, it might not be reliable, but don't expect support here.
19:02.15citywokespecially if you don't own the hardware it's running on and have no control / no visibility in to server load. you could spend time debugging issues that come down to overloaded server/network
19:03.05ManxPowercitywok, or the latency getting 1ms too high for clear audio
19:03.13mandragor[TK]D-Fender: Could I create a queue of max size 0?
19:03.26[TK]D-FenderManfMeaning?
19:03.33[TK]D-Fendermandragor: What is "size"?
19:03.39ManxPowermandragor, why don't you just not answer the call?
19:03.47citywoki'm assuming you don't mean network latency, we run with 200-250ms of latency to one intl call center without issue, tho jitter sometimes plays hell.
19:03.57[TK]D-FenderManxPower: He wants to queue incoming calls without answering until connected
19:04.00ManxPowermany providers support early media
19:04.09[TK]D-FenderManxPower: * isn't that smart
19:04.16ManxPower[TK]D-Fender, there isn't a noanswer option to queue?
19:04.18citywokWhy?  To avoid having to pay the provider for the hold time?
19:04.26[TK]D-FenderManxPower: totla news to me if there is
19:04.29citywokJust force it to stay ringing inbound non stop?
19:04.39[TK]D-FenderManxPower: I know DIAL has that....
19:04.39ManxPower[TK]D-Fender, no wonder queues suck so much
19:04.55bmoraca_workmandragor: how much is this actually "costing" you?  if it's pennies, I imagine the time being wasted here is costing more
19:04.56ManxPower[TK]D-Fender, so does background, read, etc
19:05.16citywokyea, unless you are talking thousands of minutes of hold time i wouldn't waste the energy
19:05.21[TK]D-FenderManxPower: Really?  I thought only playback did really...
19:06.18mandragorI was hoping it was something simple
19:06.31mandragorbut you think it would require a very complicated script?
19:06.39ManxPower[TK]D-Fender, I just confirmed background does have a noanswer
19:07.31[TK]D-FenderManxPower: I suppose that might work... I guess as soon as a digit is received that causes the telco to answer?  Because* has to trigger it somehow...
19:07.58ManxPowermandragor, also you should know that MANY providers disconnect ringing calls after X seconds (120 is common)
19:08.03[TK]D-FenderManxPower: Trying to think where Queue will ack that but not for prompts, etc, or if inbound DTMF automatically is processed by the telco
19:08.22[TK]D-FenderManxPower: I guess he's also expecting a high service level
19:08.40ManxPower[TK]D-Fender, Heh, n00bs, always thinking their telco will be reliable.
19:08.53breardo^^^^ learned about this TOO many times
19:09.01breardoNEVER EVER TRUST THEM!!!!
19:09.44breardoI especially love when Telco sales tells you they can do something.. and you sign the agreement.. and then the engineers say you cant do X
19:09.51[TK]D-FenderManxPower: Actually... they SHOULD be the most reliable part of the equation... and if they suck.. you are kinda fucked on a macro-leevl anyway
19:09.56*** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
19:09.56ManxPowerWe have a POTS line that Verizon has been trying to repair for TEN DAYS.
19:10.16[TK]D-FenderManxPower: Should have gotten a T1 w/ SLA :p
19:10.22mandragorManxpower: would I have to write a script to use the noanswer option? I am currently using fonality
19:10.39[TK]D-FenderManxPower: Then you ARE fucked.  You have NO control with that distro
19:10.39bmoraca_workwell, there's your first problem
19:10.42breardoyou're lucky..where I work, analog circuits have to be repaired by Qwest, but the ticket has to come from the state IT agency (MN OET).. its routine for it to take over two weeks for service to come on-site
19:10.44ManxPowermandragor, you mean the noanswer option that does not exist for queue (at least in the version of asterisk I'm using)
19:11.00breardoeven though I could call qwest direct and they'd be here in 12 hours..
19:11.02breardonooooooo
19:11.43ManxPower[TK]D-Fender, I don't believe Verizon offers SLAs to CLECs
19:12.07[TK]D-FenderManxPower: Life sucks, but rarely swallows ;)
19:12.29breardoi dont even care if life swallows..it could just smile once in awhile
19:13.42ManxPower[TK]D-Fender, what I think is *horrible* is that the PUC/PSC doesn't take complaints from CLECs and since the customer is a customer of the CLEC, not Verizon they can't file a complaint with the PUC/PSC either.
19:14.32ManxPowerwhat is nice is that is a customer files a complaint with the PUC/PSC against the CLEC if the CLEC can show the issue is Verizon, the PUC basically ignores the complaint
19:18.42bmoraca_workverizon is just about the most aweful company ever
19:19.09bmoraca_worktheir prices are insanely high, too
19:21.05*** join/#asterisk illizit (~illizit@c-76-110-117-217.hsd1.fl.comcast.net)
19:21.48bmoraca_workaround here, if you're a customer, the worst company to deal with is Telekenex...if you're an outside tech (or competing telco trying to port-out), then Telepacific is the worst
19:22.27jdwjr_I am seeing the following in our asterisk log WARNING file.c: Failed to write frame, after  file.c: was Playing 'vm-intro.gsm'
19:22.36jdwjr_Is this normal?
19:23.11*** join/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de)
19:24.52illizitHello, wondering if anyone knows what I am doing wrong. I am trying to setup Asterisk Realtime however, I am getting the following error: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available.
19:25.01illizitMy configs are as follows: http://pastebin.com/RZYTfzH7
19:25.26illizitasterisk 1.6.2.13
19:25.36illizitastersik addons 1.6.2.2
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19:29.20wcselbyo/
19:29.23*** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry)
19:29.24ghenrywhen a vendor talks about carrier mode and networking mode in a sip trunk, what do they mean?
19:29.47ghenryThis is on a NEC IP card
19:30.11bmoraca_workghenry: you'll have to ask him.  that sounds like proprietary lingo to me.
19:30.14ghenryyep
19:30.35bmoraca_workghenry: sight unseen, it seems like "carrier mode" would be trunk to a provider and "networking mode" might be a trunk to another PBX.
19:30.44bmoraca_workbut that is complete conjecture
19:30.51ghenryThey are asking if RFC2543 SIP is supported
19:31.06bmoraca_workonly a NEC sales rep would know what the terms really mean
19:31.10ghenryyep
19:31.23ghenry3261 is most current obviously plus additions
19:31.38ghenryand they are asking abotu 100rel which 1.4 doesn't have
19:32.06ghenryin fact it's disable on most softswitches I've used
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19:44.06illizitHello, wondering if anyone knows what I am doing wrong. I am trying to setup Asterisk Realtime however, I am getting the following error: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available.
19:44.34citywokdid you install the asterisk mysql module?
19:44.48*** join/#asterisk oryxtec (~test@119.152.50.78)
19:45.23oryxtechi all... can u use skype on asterisk ...with out using diguim solution.. coz that is really exp...
19:45.31oryxteci have found this link http://nerdvittles.com/?p=680
19:45.32citywokyou asked that yesterday
19:45.36citywoknobody had an answer for you
19:45.37oryxtechas any one tried this?
19:45.43illizityes, I did install the mysql module
19:45.44drmessanooryxtec, The other solutions are crap
19:45.45oryxtecwhy :(
19:45.56drmessanooryxtec, SFA is the only way to go
19:46.09oryxtecwht is SFA?
19:46.14illizit*CLI> realtime mysql status
19:46.14illizitgeneral connected to asteriskconfig@localhost, port 3306 with username asterisk for 30 minutes.
19:46.15drmessanoSKYPE FOR ASTERISK
19:46.34drmessanoThe one that is "really exp"
19:46.49oryxtechttp://nerdvittles.com/?p=680 --------- soo this will not work?
19:46.58leifmadsenhave you tried it?
19:47.03leifmadsengive it a shot and let us know
19:47.32oryxteci have not tried it.. b4 i will try.. i need expert point of views on it
19:47.40oryxtecand you guys are experts
19:47.45oryxtecthat is why i am asking you
19:48.10drmessanooryxtec, If you want to install X, then a 3rd party application, and route one concurrent call to the desktop client, there are ways
19:48.17drmessanoUnreliable as hell ways
19:48.49oryxtechummm
19:50.21oryxtecthanks :)
19:50.56drmessano$66 for a NATIVE channel driver that penetrates the Skype "bubble" is not bad at all
19:51.09drmessanoAlso comes with 1 G.729 license
19:51.15oryxtechumm
19:51.57drmessanoand it supports Digium, who pays the fine devs that code the Asterisk you use for free
19:52.14oryxtecdrmessano: i don't understand why skype sound quality is too good .. my servers are in UK as well..
19:52.30oryxtecwhen i use my server on same link... sound is ok .. not bad
19:52.42oryxtecbut when i use skype on same link.. sound is too good
19:52.50oryxtecwhy is that :S
19:53.24drmessanoSkype is using a swell codec, for one
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19:53.54f00bar80ManxPower, i've mentioned it's a VPS hosting, virtual private server
19:54.38oryxtecwht is swell codec?
19:55.40drmessanooryxtec, awesome, cool, the bomb, sweet, swell, nice, the bees knees
19:57.33oryxtec:)
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20:00.16illizitcity, any other ideas for my issue =D
20:02.41*** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com)
20:03.18ManxPowerf00bar80, unless you have proof otherwise, that is a VM
20:05.10mandragorhow can I check if a given extension is busy?
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20:08.18*** join/#asterisk Chodorenko (~chodorenk@by.one.by)
20:08.24ChodorenkoHello All
20:08.41ChodorenkoPlease consult me by SRTP
20:09.36Chodorenkoi read source code res_srtp and view  "key exchange can be activated
20:09.36Chodorenko<PROTECTED>
20:09.37Letoricmandragor: Look into 'hint'
20:10.28Chodorenkoi tried  on asterisk 1.8.rc3 and no can secured dial from asterisk to client
20:11.31Chodorenkocall from me go is secure, call from asterisk is no secure
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20:19.25f00bar80ManxPower, so the only option is a "Dedictaed server" ??
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20:41.00auti would like to set up a voip system for 10 users. it will be used for inbound customer service and regular outbound calls. can i just use any old sip trunk service? or do i need some kind of special connection to keep the latency low enough?
20:41.21citywokaut use whatever has good enough call quality for you.  test it.
20:41.23citywok~itsp
20:41.23infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
20:41.39autcitywok: can i use any isp so long as the latency/speed is enough?
20:41.53aut~itsplist-us
20:41.53infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
20:42.07citywokyou can use it... as long as it sounds good.
20:43.06autso there are never really any guarantees, eh? i guess im asking whether you need qos guarantees to legitimately use voip for a small call center
20:43.08citywokIf dial up sounds good enough for you, go for it
20:48.47ManxPowerf00bar80, the only option if you want help from most of the people on this channel.  You can do anything you please including using a turnip for your server.
20:49.51ManxPowerYou can get QoS if you get your internet service from your ITSP
20:51.53p3nguinThat doesn't sound like something that happens very often.
20:52.18autmanxpower: right. isn't that the "proper" way to accomplish it?
20:52.21citywokp3nguin: depends on your carrier, getting XO to work w/you without a connection from them is impossible
20:52.38citywokaut: the proper way is whatever works with the right quality vs cost.
20:52.39autmanxpower: the problem is that the itsp that can provide local internet service isnt usually a well-known company
20:53.19citywoki get pretty high call quality running with bandwidth.com on my datacenters 100mbit link for instance, without a dedicated link just for voice from an itsp
20:54.11citywoki might get a slightly lower dropped call rate, or slightly better call quality but the number of complaints i get per 10,000 phone calls is so low it doesn't justify the cost / extra work required to bring in a pipe for just calls
20:55.04autcitywok: interesting
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21:47.27docidok, slightly off topic and all, but i fuggure you gys might have experience with this, we've got many technologically limited users here at the office, we have a dieing meridian option-11, and a working stable asterisk install, using the standard meridian A II 4line phones, we want to move to asterisk before the meridian finishes dieing, so anybody have experence with what sip or iax phones would replace the meridian phones for useres obsti
21:47.28docidnante to learning anything new...
21:47.38*** part/#asterisk bsaxon (~bsaxon@12.107.149.61)
21:54.49Letoricdocid: I use Polycom SoundPoint IP670 phones, they work well for us
21:55.29docidthats the one i was just looking at...but im thinking the price will make the boss say no, except mabey one for his desk
21:55.39Letoricheh
21:55.47Letoricwell, you can go with the lower models of the same line
21:55.49[TK]D-Fenderdocid: Forget your old phones. They WILL be learning new stuff
21:56.38docidaye fender, im aware, but we have a few elders working here, so at least need 1 button pickup from transfer and hold
21:57.00[TK]D-Fenderdocid: Extremely unlikely
21:57.02*** join/#asterisk poison (~poison@78-21-102-160.access.telenet.be)
21:58.07poisonhi all, I'm createing some kind of escalation dial plan which rings number 1 on the list, and if that fails, it rings number 2, ... but the problem is that one of those numbers transfer directly to the voicemail (in case of a cell phone) then the call is answered and my application is broken
21:58.13poisonis there a way to overcome this?
21:58.33[TK]D-Fenderpoison: "core show application dial" <- M()
22:00.10poisonso there's something executed on the new called line, and there I can detect if it's a human voice or a voicemail?
22:00.52docidohh, im curious on that one too
22:04.23[TK]D-Fenderpoison: Nothing resembling reliable.  Promt them for CONFIRMATION.
22:04.27[TK]D-Fenderprompt*
22:05.17poisonso I should ask my users to press '1' if they are on some kind of voicemail?
22:06.34[TK]D-Fenderpoison: read the instructions again.  You seem to have completely missed it
22:07.23poisonI'm using asterisk 1.4.22-rc5 (from elastix), does your suggestion applies to that version too?
22:10.06[TK]D-Fenderpoison: To EVERY version.  However unless you're already working up custom dialplan for this you're screwed.
22:10.15Qwellpoison: Only 14 releases behind.
22:12.19poison<PROTECTED>
22:12.31poisonsorry I don't quite get it yet
22:14.57[TK]D-Fenderpoison: Call the Cell, as the CELL SIDE to press "1" to ACCEPT the call and it will be CONNECTED
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22:22.39poisonok I'll check, thanks for the hints
22:23.05poisonit's a pitty there's no special return code, I've read about machine detection modules but they don't see very reliable too
22:34.37poisonis there a way to play back sound to the party I just called, and not to the caller (the one currently in my dialplan application)
22:40.30[TK]D-Fender[17:58]<[TK]D-Fender>poison: "core show application dial" <- M()
22:58.00poisonhmz, [TK]D-Fender, I'm still struggling with your suggestion, I created a macro: macro-mustaccept, but it doesn't hangup the call if I don't press anything (http://pastebin.ca/1960488)
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23:00.07[TK]D-Fenderpoison: Read the instructions again <-
23:04.45[TK]D-FenderPoincare: Hint : there are CHANNEL variables you should be paying attention to.
23:23.04poisonis there a way to use the default outbound trunk in the Dial() program?
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23:26.48[TK]D-FenderPoiThre is no such thing as "default".  You tell it what to dial.  The end.
23:26.55[TK]D-Fenderpoison: Exactly as it appears.
23:29.23poisonwell, I have [outbound-allroutes] where I configured all my outbound routes, and now in each Dial() command I have to specifiy which route I want to use instead of using the logic in [outbound-allroutes]
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23:38.26[TK]D-Fenderpoison: there is no default logic to a context name
23:39.27[TK]D-Fenderpoison: And Dial() calls the specific resource you tell it to.  Again, this has what do do with a specific context?
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23:40.41poisonok, it's just that whenever I update my outbound route information in freepbx, I have to update my script to make sure my primary voice provider is used to forward my Dial() calls to
23:41.51*** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net)
23:43.58[TK]D-Fenderpoison: FreePBX?  it is NOT supported here and their junk is meaningless trash in here
23:44.02[TK]D-Fender~freepbx
23:44.02infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
23:44.04[TK]D-Fender^^^
23:44.55[TK]D-Fenderpoison: If you're under it's yoke then things get messy, fast.  Assuming that it doesn't force you to do everything manually if at all
23:48.26poisonwell, fortunately I'm creating my dialplan outside of freepbx (we just use it to enable non-technical ppl to do some basic stuff)
23:54.19*** join/#asterisk aut- (~aut@oenf-nat-96-8-4-42.beyond.dssitech.com)
23:55.28[TK]D-Fenderpoison: Well there is no "routes".  That means passing through their dialplan whose extra crap will FUBAR you
23:56.01[TK]D-Fenderpoison: So you are stuck doing it 100% manual for that segment.  Do not expect to use any of their structures for this

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