IRC log for #asterisk on 20080818

00:00.07[TK]D-Fendertvirus: it is recommended that if you value your faxes you will keep them as far away from * as possible.
00:00.59tvirusThe fax line carries the DSL, though, and the boss wants to use it. Why doesn't * play well with faxes?
00:03.55Elijah`[TK]D-Fender, could you take a look at this SIP debug and see what you think?  I thought I had the NAT problem resolved, but still not working... here's the pastebin: http://elijah.pastebin.com/m3ffe7d8e
00:05.21[TK]D-FenderElijah`: from the START of the call, and include your sip.conf masking only passwrds
00:05.21*** join/#asterisk brodiem (n=brodiem@e2.72.1243.static.theplanet.com)
00:06.09Elijah`That was from the start... but here I'll add in the sip.conf...
00:06.25axisys[TK]D-Fender: there are soo many emails.. i wish there is one article where it covers the setup.. i know wishful thinking
00:06.38[TK]D-FenderElijah`: No, that wasn't the start
00:07.15[TK]D-Fenderaxisys: JFGI
00:07.39axisysthe only thing is not working is when i receive a call from outside the pstn port redirects to lne port.. everyelse is working
00:07.50axisyswhats jfgi ?
00:09.21tvirusJust Fucking Google It
00:11.00Elijah`lol
00:11.17Elijah`ok here it is, I double checked and did another call, this is where it starts http://elijah.pastebin.com/d6b8fddf7
00:11.27axisystvirus: heh
00:11.46axisysman people are sooo angry.. wow!
00:11.54axisyssorry for asking questions
00:12.47axisysif anyone here can help w/o getting pissed of please help
00:12.55axisysthnx
00:13.09tvirusI'd help but I have no idea what I'm doing yet :)
00:13.22axisystvirus: appreciate it really
00:13.45Elijah`same... newbie here too LOL
00:14.06axisysElijah`: heh
00:14.32axisysi have been googling for about 2 weeks.. nothign came out to be same issue as mine
00:14.58axisysbut heh.. as long as I dont ask [TK]D-Fender i guess i am safe from another blasphemy.. heh
00:15.52jtoddaxisys: you've seen these, right?
00:15.53jtoddhttp://voxilla.com/tools/device-configuration-wizard/
00:15.54[TK]D-FenderElijah`: Where is your Cisco located relative to *?
00:16.12Elijah`it's behind a separate NAT
00:16.27axisysjtodd: yes.. i followed and it helped a lot
00:16.40axisysjtodd: i can call to my cell now from my line port
00:16.42[TK]D-Fenderaxisys: http://www.google.ca/search?hl=en&q=SPA-3102+setup+guide+asterisk&btnG=Search&meta=
00:17.03[TK]D-FenderElijah`: What do you have forwarded to *?
00:17.11axisysjtodd: i can recv call to.. but did not go thru sip proxy
00:17.17Elijah`port 5060, and the RTP ports 10000-10010
00:17.23Elijah`UDP
00:17.31clintctvirus: faxes won't work well if asterisk is trying to echo cancel
00:17.40[TK]D-FenderElijah`: You should widen that a healthy bit. I've seen things go bad due to over tightening
00:17.54[TK]D-FenderElijah`: But as we're not seeing an answer it may be an issue on the Cisco end
00:17.55Elijah`so widen the RTP to say, 50 or 100?
00:18.16Elijah`aah, ok makes sense... now thing is...
00:18.22Elijah`if the cisco places the call, things work fine
00:18.32tvirusSo the fax should just be left alone with the analog line.
00:18.45[TK]D-FenderElijah`:have you forwarded on their side?
00:18.47axisys[TK]D-Fender: thanks.. i will go throw the search results again.. thanks again
00:19.08Elijah`No, as I understood you shouldn't have to.. so no nothing is fwded
00:19.22clintctvirus: that's generally the easiest thing to do
00:19.28jtoddaxisys: I don't have any specific solutions; it's been a long time since I messed with a Sipura.  If I were you, I'd set up wireshark and look at what is happening on the ethernet between your * server and the Sipura.
00:19.55tvirusSounds good. So for a single analog line, would an ATA work just fine getting it setup in *?
00:20.19axisysjtodd: i already installed wireshark.. i will followup on your suggestion.. thnx
00:21.00clintctvirus: we never had good luck getting a fax to work with an ata... we send them out via an fxs configed port and that works well
00:21.09jtoddaxisys: change one thing at a time, note what you did, test further.  Too many options in the Sipura - easy to get confused.
00:21.23tvirusclintc: I meant for a separate analog line, not connected to the fax.
00:21.31*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-bf902bcc8805d395)
00:21.37clintctvirus: ah, sure
00:21.46Elijah`I would bet it's on the cisco end, because there's other phones registered elsewhere (behind other NATs) that work ok
00:21.52axisysjtodd: i know :-)
00:22.01jtoddaxisys: I think part of the "magic" for the configuration was "insecure=very" on the Asterisk side in sip.conf for that peer.
20:33.26*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
20:33.27*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev
20:33.43Nivexjbot: welcome back
20:33.44jbotGood to see you, back!
20:33.58Nivexheh
20:34.51ShaunWingand then what do I do after that?
20:35.14[TK]D-Fendereric_hill: Please dump your dialplan from CLI
20:35.19*** part/#asterisk iotashan (n=shan@adsl-71-150-254-145.dsl.mdsnwi.sbcglobal.net)
20:35.38*** join/#asterisk lanning (n=lanning@66.151.128.195)
20:36.28eric_hill[TK]D-Fender: It's huge.  I tried to boil it down to the smallest working demonstration of the problem rather than spamming this chat with 1000 lines of irrelevant crap.
20:37.23eric_hill[TK]D-Fender: And I can work around the problem by setting a variable before the first Gosub.  It just smells like a bug.  ARG1 gets dropped if returned from an invalid context extension.
20:37.37eric_hill[TK]D-Fender: But I guess it could be as-designed.
20:37.38[TK]D-Fendereric_hill: -- Executing [1002@foo-a:2] Return("Zap/16-1", "") <-- I stragnly see it starting at priority *2* and now I'm starting to complete not trust what you're showing me.
20:37.44[TK]D-FenderEither way heading home, BBIAB
20:38.09*** join/#asterisk MindTheGap (n=MindTheG@201.17.149.252)
20:39.33jeevFender. i did it
20:39.34jeevcrap
20:41.21drmessanoHA
20:41.25drmessanoSounds like it
20:41.45*** join/#asterisk barakuda (n=barakuda@91.78.92.63)
20:44.14jeevhmm, why aren't you ignored
20:53.43*** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net)
20:53.58LemensTSthis is repeating over and over in the cli: -- parse_srv: SRV mapped to host sphone.vopr.vonage.net, port 5061
20:54.07bkw_its doing SRV lookups
20:54.08bkw_thats normal
20:54.19kfifeQwell: func_odbc depends upon somethign called res_odbc.  Do you have any idea what CentOS/RHEL5 calls it in their repository?  I can't find a match.
20:54.34LemensTSbkw_ : ok thanks, never had it do that with other itsp's
20:54.35Qwellres_odbc is part of asterisk
20:54.51*** part/#asterisk gbr_ (n=gbr@200.103.96.98)
20:54.58bkw_LemensTS: if you have ITSP's without SRV records they are clearly doing it wrong
20:55.07bkw_SRV records are a must to do thing correctly
20:55.30kfifeI see.  I just need to re-run configure and make menuselect, and the dependencies will show as satisfied?
20:55.37ShaunWingSay: [root@messaging etc]# service asterisk start
20:55.38ShaunWingStarting asterisk:                                         [  OK  ]
20:55.40hsv-ali still cant get rid of this crap
20:55.40ShaunWing[root@messaging etc]# ps aux | grep asterisk
20:55.41ShaunWingroot      8437  0.0  0.0  65924   596 pts/0    S    22:52   0:00 /bin/sh /home/shaunw/asterisk-bin/sbin/safe_asterisk -U shaunw -G shaunw -C /home/shaunw/asterisk-bin/asterisk
20:55.43ShaunWingshaunw    8446  0.1  0.1 447312  9468 pts/0    Sl   22:52   0:00 /home/shaunw/asterisk-bin/sbin/asterisk -f -U shaunw -G shaunw -C /home/shaunw/asterisk-bin/asterisk -U shaunw -vvvg -c
20:55.44Qwellkfife: if you've got the deps installed, yeah :)
20:55.44ShaunWingroot      8482  0.0  0.0  61160   712 pts/0    S+   22:53   0:00 grep asterisk
20:55.45hsv-al[Aug 18 09:22:41] WARNING[9071]: res_smdi.c:1264 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.
20:55.52kfifeYou've just saved me a lot of time!  Thanks a million!
20:55.53ShaunWingIs this now running as non root?
20:55.53Qwellkfife: make sure you have unixODBC-devel, or whatever your distro calls it
20:55.55hsv-alhow do i prevent that module from loading up on start
20:56.17*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
20:57.22ManxPowerhsv-al: chances are the same way as you stop any other module from loading in Asterisk
20:57.51hsv-albut the thing is, its not set to load up
20:57.53ManxPowerYou had better hope no modules require res_smdi.so.
20:58.14ManxPowerhsv-al: pastebin your /etc/asterisk/modules.conf
20:58.30*** join/#asterisk grantm (n=grant@68.142.138.4)
20:59.05*** part/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net)
20:59.11ManxPowerhsv-al: you can always mv /usr/lib/asterisk/modules/res_smdi.so
20:59.17ManxPowerdrat
20:59.33ManxPowerhsv-al: you can always mv /usr/lib/asterisk/modules/res_smdi.so /usr/lib/asterisk/modules/disabled-res_smdi.so
20:59.38*** join/#asterisk gerphimum (n=trekkie@cpe-67-10-147-45.satx.res.rr.com)
20:59.48ManxPowerBut as I said, you had better hope no other modules require res_smdi.so
21:01.00*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
21:01.21ShaunWingsay when I run as non root:  asterisk -r I get back...
21:01.23ShaunWing-bash: asterisk: command not found
21:01.41QwellShaunWing: if you aren't root, you need to use the full path to asterisk
21:01.44ManxPowerShaunWing: then the directory asterisk is in is not i your path
21:02.03ShaunWingok, but how can I create a short cut for myself?
21:02.07ShaunWingplease
21:02.21ManxPowerOh, ShaunWing is one of *those* people.
21:02.39ManxPower(the people that expect us to tutor them in basic Linux stuff)
21:02.47drmessanoRight Click on the desktop
21:02.52drmessanoGo to New --> Shortcut
21:02.56seanbrightAlt-F4
21:02.57*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
21:02.57drmessanoGo to the C; Drive
21:03.05drmessanofind the exe file...
21:03.09drmessanoOh, wait
21:03.22ManxPowerdrmessano: no, no.  He said "short cut", not "shortcut.  For a short cut, do not cut as much
21:03.24hsv-alhttp://pastebin.ca/1178006
21:03.26seanbrightShaunWing: first, in your irc client type /server create-asterisk-shortcut
21:03.28drmessanoLOL
21:03.30hsv-alno reference to it
21:03.32hsv-alweird
21:03.57drmessanoShaunWing: freenode has a built in IRC Linux documentation server
21:03.59ManxPowerhsv-al: now you know what to do.
21:04.01ShaunWingTX 4 teh tip
21:04.02drmessanoYou need to add it to your server list
21:04.08drmessanodo /server linux-docs
21:04.30hsv-alwhat is that module typically used for
21:04.31seanbrighti get the medal if it works
21:04.39hsv-albut since its not in my system,, ill issue the command you pasted
21:04.58ManxPowerhsv-al: just add the noload line like I originally suggested
21:05.24ShaunWingUnknown host “linux-docs”
21:05.27ManxPowerI suspect you'll lose voicemail if you don't load res_smdi.so.
21:05.28seanbrightdrat
21:05.43eric_hillCan anyone else reproduce this problem? http://pastebin.com/d63f52f43
21:06.05seanbrighteric_hill: yes -> http://pastebin.com/m7fd1bce
21:06.24drmessanoShaunWing: Did you do it in this window?
21:06.31seanbrightan exact reproduction
21:06.33ShaunWingyes
21:06.39seanbrightstops being a smartass and goes home
21:06.48eric_hillneeds more beer on a Monday.
21:07.04drmessanotry /server linuxdocs
21:07.09*** join/#asterisk jks (n=jks@193.189.93.254)
21:07.27ShaunWing[ERROR]Unknown host “linuxdocs” connecting to irc://linuxdocs/,isserver (irc://linuxdocs/,isserver). [Help] Reconnecting in 15 seconds. [Cancel]
21:07.52drmessanoSounds like your IRC client isn't RFC2188 compliant
21:07.55drmessanoTry google
21:08.15drmessanoOh
21:08.21drmessanoYou have ChatZilla
21:08.22ManxPowereric_hill: I suspect you do not understand how gosub works.
21:08.45eric_hillManxPower, I'm just trying to figure out if this is a bug or a feature.
21:08.58ShaunWingyes
21:09.02drmessanoWith ChatZilla, ShaunWing, you can join channel 0 on here for linux help
21:09.02ShaunWingChatZilla
21:09.04sn9drmessano: doesn't work in xchat, either, and that *is* compliant
21:09.10drmessanoJust do a /join 0
21:09.17eric_hillManxPower, I wouldn't have expected the Macro argument to disappear, but saving it to a variable first works just fine.
21:09.23ManxPowereric_hill: It's working as intended (subroutines start with exten => s,1,Whatever
21:09.25[TK]D-Fendereric_hill: -- Executing [1002@foo-a:2] Return("Zap/16-1", "") in new stack <- why do we not see a PRIORITY 1 execute for this?
21:09.46ShaunWing*ChanServ*[#0] Welcome to #0 - the channel that shouldn't exist...
21:09.48ManxPower[TK]D-Fender: "s" is converted to real priorities at runtime
21:10.05[TK]D-FenderManxPower: No, read the PB
21:10.06jeevFender, i've succeeded in passing the variable with _XXX then $CHANNEL 1:4. now i've got to add _XXX to the dial plan also so the database get doesn't fail.
21:10.12sn9ShaunWing: i edited the /topic there
21:10.13ManxPower    -- Sent into invalid extension 's' in context 'foo-b' on Zap/16-1
21:10.14drmessanoShaunWing: Indeed not compliant
21:10.27drmessanoShaunWing: Try Ctrl-Alt-Shift-F4
21:10.33[TK]D-FenderManxPower: -- Executing [s@macro-example:2] Gosub("Zap/16-1", "foo-a|1002|1") in new stack followed IMMEDIATELY by :  -- Executing [1002@foo-a:2] Return("Zap/16-1", "") in new stack
21:10.53sn9drmessano: what is that supposed to do?
21:11.05eric_hill[TK]D-Fender - That was a verbose = 5, do you want a verbose 9?
21:11.14[TK]D-FenderManxPower: the first gosus is completely untrustoworthy
21:11.17ManxPower[TK]D-Fender: We are working on different problems.  The one I'm working on is the invalid gosub
21:11.17[TK]D-Fendergosub
21:11.39[TK]D-FenderManxPower: I'm working on #1 which tells me that what we are looking at is DOCTORED
21:11.41*** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com)
21:11.49ShaunWingits ##linux
21:12.11[TK]D-FenderManxPower: because #2 also would seem to indicate that the ${ARG1 } got wiped
21:12.11ManxPowerIf he changes exten => i,1, to exten => s,1,  I bet stuff will start working
21:12.30[TK]D-FenderManxPower: 2 problems, my impossible priority, and a MISSING var
21:12.35eric_hill[TK]D-Fender: You keep thinking I'm faking the result... http://pastebin.com/d40cac214
21:12.56hsv-altisk tisk @ "dr" messano for using
21:12.58hsv-althe alt-f4 trick
21:12.58ManxPowereric_hill: your pastebin of the dialplan executing does NOT match the CLI output.
21:13.18eric_hillWould you like to remote control my machine and see what I'm seeing?!
21:13.29eric_hillI'd be happy to set up a webex.  I'm not faking this.
21:13.50ManxPowerRemember when you do a macro inside a macro the arguement values are not saved.  I assume it is the same for gosub
21:14.27ManxPowereric_hill: That's why I think it is important to fix the obvious wrongness (as I told you just moments ago) than to try to find the hidden problems.
21:14.29[TK]D-FenderManxPower: That makes a logto of sense if they tried to make a cheap-out implementation of Gosub
21:14.37[TK]D-Fenderlot*
21:14.46ManxPower[TK]D-Fender: I got bitten by the issue years ago
21:15.18[TK]D-Fendereric_hill: It does look like the var is getting eaten by the gosub so far.
21:15.22*** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net)
21:15.26ManxPowerthe easiest thing is to NEVER EVER use ${ARG2}, ${ARG2} etc when calling a macro or gosub
21:15.53[TK]D-FenderManxPower: Actually its more like never call "gosub" from within a macro
21:15.54eric_hillChanging to exten => s,1 will cause the Gosub to never return on an invalid extension, hence the second Gosub will not be executed.
21:16.18ManxPowers/e ${ARG2}/e ${ARG1}/
21:16.23eric_hillOk, I'll write that down in the list of stuff not to do inside a dialplan.
21:16.27LemensTSI got vonage setup in asterisk using the soft phone account they offer, i was hoping to use it as DID like most other ITSP's offer...i called there tech support they dont know what DID is...do any of you know if they do this service or not?
21:16.58[TK]D-FenderLemensTS: DID = a phone number they receive that they route to you
21:17.00QwellLemensTS: don't all Vonage accounts have phone numbers?
21:17.11[TK]D-FenderLemensTS: and of course their tech support drones don't know shit about *
21:17.12ManxPower[TK]D-Fender: recursive macros are the heart of my macros! 9-)
21:18.04ManxPowereric_hill: the 2nd invocation of the macro/gosub will overwrite the ARG1, ARG2, etc and not restore them when returning to the original calling macro/gosub
21:18.18ManxPowerthis is by design as I understand it.
21:18.25[TK]D-FenderManxPower: And the look of revusion on my face upon seeing your spaghetti&meatsauce&ice-cream&brusselssprout&caviar code LEGENDARY
21:18.38LemensTSqwell: tkd: i mean so i can use one phone number with multiple outgoing lines...thats did isnt it...
21:18.46[TK]D-FenderLemensTS: NO
21:18.48ManxPowerLemensTS: no.
21:18.52[TK]D-FenderLemensTS: a DID is a PHONE NUMEBR
21:18.59eric_hillManxPower, tx.  Fender, classic, as always.
21:19.09[TK]D-FenderLemensTS: this has no impact over how it may be delivered to you.
21:19.09Qwellvoip doesn't really have the concept of "lines"..  presumably, they allow multiple incoming/outgoing calls at once
21:19.31*** part/#asterisk chackal_sjc (n=pet@200-168-20-85.dsl.telesp.net.br)
21:19.35*** join/#asterisk pascal_alm (n=pascal_a@66.11.93.129)
21:20.21ManxPowerQwell: They are called "trunks" in the VoIP world.  8-|
21:20.56LemensTSmanx : trunks yes that is the work i mean. I want a did with multiple trunks.
21:20.57[TK]D-Fender:p
21:21.04LemensTS*word
21:21.10ManxPowerchortles evily
21:21.21[TK]D-FenderLemensTS: You just got "Punk'd" by ManxPower
21:21.25jeevFender! http://al.pastebin.com/d22899ae9 uh, i dunno how to now make the number going out check the _XXX, extension then dial out after. do i need more variables?
21:21.32[TK]D-FenderManxPower: You are a very bad person....
21:21.39LemensTStkd: its fine i have a bad memory
21:22.53[TK]D-Fenderjeev: No, its PART of the exten that was dialed.
21:26.25*** join/#asterisk shinao1 (n=shinao1@62.173.48.176)
21:26.50LemensTShttp://www.voip-info.org/wiki/view/Vonage+Business+Plus   here is sip trunking with vonage i was looking for, helped when i used the work trunk.
21:27.22eric_hillIs there a switch somewhere to turn on outbound Caller ID sending over an ISDN line?  Inbound name/number is coming through, and outbound number are working fine.
21:27.36ManxPowereric_hill: yes
21:27.50roe_has anyone done any performance testing with the atom processor? I didn't see anything on the dimensioning page
21:28.09eric_hilloffers ManxPower a beer for the switch name.
21:28.16ManxPowereric_hill: If the outbound number works, then the telco has already set that option
21:28.49eric_hillManxPower: It's for an internal switch-to-switch call, not going over the PSTN.
21:28.51ManxPowereric_hill: the option is set by the telco, not the user.  Also most terminating telcos overwrite the name sent with whatever name the phone company has on file for that number.
21:29.05ManxPowereric_hill: then I have no idea.
21:30.08*** join/#asterisk nny_2 (n=Scott_My@64.203.244.146)
21:30.12nny_2hmmp
21:30.13nny_2h
21:30.33ManxPowereric_hill: you mean internal asterisk-asterisk call right?
21:31.06nny_2ha nm
21:31.08nny_2hi all
21:33.04eric_hillManxPower: Asterisk -> Nortel PBX
21:33.38eric_hillManxPower: Otherwise I'd just use SIP->SIP or IAX->IAX.
21:33.43ManxPowereric_hill: It must suck to be you.
21:34.02eric_hillI can't wait to get off this crappy PBX.
21:34.19ManxPowereric_hill: if you do a PRI debug I suspect you'll see Asterisk send the Caller*ID name.  If so then it is entirely up to the Nortel.
21:35.03*** join/#asterisk fury_ (n=flolry@ool-4b7fd6a2.static.optonline.net)
21:35.26eric_hillI've been debugging and see a *bunch* of stuff, including "Received caller name'8670 Eric H'", but I never see a "Sending caller name".
21:35.28ManxPowereric_hill: I believe Caller*ID name can be sent in one of two ways on PRI.  You might want to search the mailing list archives to see if Asterisk can be told to send it the other way.
21:35.42eric_hillGood to know.  Tx.
21:36.07eric_hillI also found a post that said I should try switching from Q.SIG to DMS100.  May try that tonight after-hours.
21:36.33fury_is there any addon or library or something that will help me with getting my phone system to be able to pronounce human names?
21:36.57[TK]D-Fenderfury_: "core show application record"
21:37.14eric_hillfury_: ran across http://www.cepstral.com/ today...
21:37.59fury_[TK]D-Fender, I don't think that's what I want
21:38.05fury_eric_hill: thanks
21:38.37[TK]D-Fenderfury_: Probable.  And I want $1,000,000.  Guess nobody is happy today.
21:38.52fury_no its no problem :)
21:39.05fury_just saying that's not what I asked. actually it will be useful anyway though.
21:39.10ManxPowerfury_: most text-to-speach programs allow you to spell names phonetically
21:39.18ManxPowerwords, not names
21:41.30fury_eric_hill: it has a demo, might work good enough :)
21:41.35fury_my name is kind of hard to pronounce
21:41.48eric_hillfury seems pretty easy to me...
21:42.14sn9Abby Normal?
21:43.58*** join/#asterisk ta^3 (n=tacvbo@189.146.172.3)
21:44.43ManxPowereric_hill: what if its pronounced "furry"?
21:46.04*** part/#asterisk nny_2 (n=Scott_My@64.203.244.146)
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21:47.18kfifeQwell: Func_odbc Installed! Thank you!
21:50.09kfifeDoes anyone know: soft hangup SIP/....  : How do I get the unique ID of the channel?  It's the string that comes across the CLI when a call comes in.  It's not the string that comes in when you type: core show channels or even sip show channels.  What command will show the string?
21:50.55kfife...you can't always draw a relationship between the channel you see that is hung, and the string that came across the CLI when the call originated.
21:51.14exothermcManxPower: I figured out a solution to my 503 ambiguity the other day.
21:51.45kfifeexothermic: 503 as in Grandstream HT503?
21:52.17[TK]D-Fenderkfife: "core show channels concise"
21:52.22exothermcNo sip response code 503 (congestion) I belive.
21:52.58kfifeThanks for the distinction.  I've got a question for anyone using the GS HT503.
21:53.24eric_hillMansPower: I think my CID problem may be solved with prilocaldialplan=local in the zapata.conf.  One hour and counting to a maintenance window... :)
21:53.31exothermcI was having the issue of not knowing how best to handle getting a 503 from a carrier (who wanted me to treat the 503 as a network wide response not just the server, which is how the RFCs say to use it for)
21:54.06exothermcSo basically they wanted me to route advance their network on a 503, not just keep hammering all their different end points for the call.
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21:59.06mltlnxDoes anyone know why when I restart my Sangoma 104 card my channel banks rings on all channels? Its really strange and we are deploying them for use in a hospital. Of course those lines will be in the patients rooms!
21:59.45denonhah
21:59.57denonI'd talk to Sangoma and see if they've got some tips
22:00.25denonthey've probably seen it a hundred times
22:00.45mltlnxCalled them this morning and they did not know.
22:03.09*** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com)
22:03.43kfife.
22:05.34mchouok, is this possible with asterisk? On an outbound call, dial plan extension pattern is matched (after the phone hsa sent the dial string).  I want to play a recording (more or less saying which extenion rule matched) before asterisk actually performs the Dial cmd.  What would be a good way to do this?
22:05.57mchouhas*
22:06.42[TK]D-Fendermchou: "core show application playback"
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22:15.53*** part/#asterisk beek (n=klinebl@65.211.106.242)
22:16.08jeevFender, did it again.. with the dial plan, used exten 4:11 and it picked it up, perfect!
22:16.12jeevgod i'm the best
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22:21.57*** join/#asterisk hadronzoo (n=user@m765f36d0.tmodns.net)
22:22.35hadronzooHello, how can I add some time before asterisk answers a call on either an inbound route or on an inbound trunk?
22:22.49riddleboxcan someone help me, when I make a call out, I can talk for a little bit, then it hangs up?     -- Executing [16186544596@DLPN_LocalLong:1] Macro("SIP/524-0822a840", "trunkdial-failover-0.3|Zap/g1/16186544596||trunk_1|") in new stack
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22:23.12jeevpastebin.com
22:23.14riddleboxshoot sorryt
22:23.16riddleboxkick me
22:23.23jeevgive me your adddress!
22:23.24riddleboxI did, but I guess it didnt copy the link
22:23.24wwalkerkicks riddlebox
22:23.37kd8iktleaps out and begins bludgeoning riddlebox
22:23.54riddleboxhow about that http://pastebin.ca/1178100
22:24.21wwalkercan yo7u bludgeon with a sharp object?  or only blunt ones?
22:24.33kd8ikthowbout i castrate you for not having atleast a unique nick
22:25.23riddleboxthank goodness [TK]D-Fender  hasnt seen that mistake yet ;)
22:26.18[TK]D-Fenderriddlebox: I have
22:26.21heedlyhadronzoo: Wait(time) before Answer or Dial.
22:27.02[TK]D-Fenderriddlebox: What are you dialing out of?
22:27.10hadronzooheedly: in what conf file?
22:27.21[TK]D-Fenderhadronzoo: extensions.conf, where else?
22:27.28[TK]D-Fenderhadronzoo: this is all DIALPLAN
22:27.34hadronzoo[TK]D-Fender: cool, thanks
22:28.08riddlebox[TK]D-Fender, a pots line
22:28.24Qwell~whatnow > Deeewayne
22:28.31Qwellsilly bot
22:28.37[TK]D-Fenderriddlebox: First guess "callprogress=yes", #2 = other side hung up.
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22:31.39hardwireanybody handle motorola bsr2000 cmts's here?
22:31.46hardwireit's.. slightly.. related.. :)
22:31.58heedlyhadronzoo: extensions.conf
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23:03.48*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
23:07.44lesouvageWhen I change the value of the variable "emailbody" in voicemail.conf it doesn't seem to effect the content of the e-mail send when a voicemail is received. I run asterisk 1.4.21.1. Isn't that where the variable "emailbody" is ment for?
23:10.27[TK]D-Fenderlesouvage: Did you restart *?
23:10.36*** join/#asterisk codefreeze-lap (n=murf@72.21.67.40)
23:11.12lesouvage[TK]D-Fender: no I did a reload, restart is needed?
23:11.22[TK]D-Fenderlesouvage: You need to reload the app at a minimum
23:15.06lesouvage[TK]D-Fender: I did a restart of Asterisk but still the default content of the e-mail ."New 0:24 long msg in box 505  from 505, on Tuesday, August 19, 2008 at 01:00:54 AM"
23:15.15*** join/#asterisk brodiem_mobile (n=brodiem_@32.171.51.154)
23:15.35[TK]D-Fenderlesouvage: pb your voicemail.conf
23:15.56lesouvagePoint is that the "from" is not correct.
23:16.25lesouvage[TK]D-Fender: thanks
23:16.33[TK]D-Fenderlesouvage: And that has WHAT to do with the "email body" exactly?
23:17.25[TK]D-Fenderlesouvage: Don't show me one thing and then pull a bait & switch like that
23:17.48lesouvage[TK]D-Fender: If the message that I can't change would have been correct there would have been no need for change.
23:18.34[TK]D-Fenderlesouvage: First you complain about the body now you complain about the "from" Holy shit make up your mind, and SHOW ME.
23:24.31jeevgod i'm so pissed, my friend called and bout 2 minutes into the conversation.. got disconnected.
23:24.36jeevi dunno if it's the ITSP or what..
23:25.16lesouvage[TK]D-Fender: I paste it on http://www.pastebin.be/13238
23:27.02unpaidbilllast t38
23:27.18*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
23:28.11[TK]D-Fenderlesouvage: And please PB a sample e-mail
23:29.19unpaidbill~opal
23:32.39*** part/#asterisk jkirby (n=jkirby@dsl-240-111-11.telkomadsl.co.za)
23:35.43kfifewhen I type CLI> show channels,  the last few characters of the channel name are always truncated.  SIP/ptassessor-b7d004a0 shows up as SIP/ptassessor-b7d00.   This means I have to complete the name before using a command such as soft hangup.  Granted it will autocomplete with a TAB press, but only if the first part of the name happens to be unique.   Is there a way to show the WHOLE channel name?
23:36.12*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
23:38.03[TK]D-Fenderkfife: "core show channels concise"
23:40.45*** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn)
23:40.56kfifeThanks!  You always seem to know!   You are the one that turned me on to local channels a while back which brought a great deal of efficiency to my dialplans.
23:40.58[TK]D-Fenderok, heading to the gym.  Back later
23:41.19kfife[TK]D-Fender: that thanks was for you!
23:41.34[TK]D-Fenderkfife: np
23:42.09*** join/#asterisk PepOSX (n=angeldav@190.72.129.75)
23:42.55anonymouz666[TK]D-Fender going to the gym?
23:42.58anonymouz666WTF
23:43.13lesouvage[TK]D-Fender: New 0:24 long msg in box 505  from 505, on Tuesday, August 19, 2008 at 01:00:54 AM
23:44.42anonymouz666[TK]D-Fender: you are gold medal of asterisk olympic
23:45.37lesouvage[TK]D-Fender: This is the whole e-mail.  It seems to be the pager message.
23:46.12anonymouz666imagine [TK]D-Fender asking for SIP DEBUG, extensions.conf and dmesg in the gym :P
23:50.14jeevhahaah
23:50.19jeevhe's the bear grylls of asterisk
23:57.59*** join/#asterisk spartan7 (n=spartan7@wsip-70-169-241-66.oc.oc.cox.net)
23:59.35lesouvage[[tK]D-Fender: sorry, it was a stupid typo. I have the voicemail to  e-mail working as it should  with msmtp instead of sendmail and a gtalk account. msmtp is very very easy to configure.

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