IRC log for #asterisk on 20090123

08:34.34*** join/#asterisk jbot (i=ibot@rikers.org)
08:34.34*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.3 (2009/01/06), 1.4.23-rc4 (2009/01/16), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev
08:34.46TimRikerjbot: wb
08:34.46jbotIt's great to be back!
08:37.10TrentCreek~google jbot
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08:41.28drmessanoYAY
08:41.33drmessano~drmessano
08:41.34jbot[drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily
08:42.16drmessanoYou brought my best friend back
08:42.19drmessanoTY TY TY
08:42.20dan__theh
08:42.25drmessanoMY BROTHER
08:42.33drmessanoJBOT: I HEART YOU
08:42.34jbotYou HEART YOU?
08:42.45drmessanoslaps jbot around
08:42.50drmessano<3
08:43.13drmessano~polycommunist
08:43.14jbotA polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world.  They may also be getting a 10% kickback.
08:43.28drmessano:D
08:43.40drmessano~~~~~
08:43.40jbotgrrrr
08:43.45drmessano:D
08:43.58drmessanoI feel whole again
08:44.01drmessano~book
08:44.02jbotit has been said that book is probably 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://tfot.leifmadsen.com or see ~buybook
08:44.13drmessano^^^^^ YAY 4 LESS TYPING
08:44.59drmessanojbot went down, and I was like "Fine, may as well take Google down too and I can swallow the whole bottle"
08:45.07drmessanoBut now, yay
08:45.38drmessanowonders where the Paypal donate button is for jbot
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09:00.08tamseelclear
09:00.08dan__tblahblahblah does not exist in any format ... ... .. generally an indication that I'm doing something dumb with codecs?
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09:00.08tamseelhi <dan_t> can u solve ma problem plz
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09:01.18dan__tI cannot.
09:01.18dan__tSorry.
09:01.42tamseelany body else who can?
09:02.54TrentCreektamseel: wait a few more hours when more people will be here
09:03.03tamseel<dan_t> i am asking that any body else who can solve my problem with zaptel
09:04.31torrikfttamseel whats the problem?
09:04.58tamseeli have reinstall the zaptel card but now it do'nt make any outgoing call
09:05.29torrikftwas it working before?
09:05.52tamseelyah but due to some problem i have reinstall it
09:06.35tamseelthe problem was that with old configuration that some time it make a call but some time it generate problem i mean not make call
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09:07.39virtualme123Does using the SIP jitterbuffer have no effect when you are using the dahdi_dummy with jitterbuffer turned off?
09:07.59tamseeli have also make the changes to /etc/asterisk/zapata.conf
09:08.06tamseelbut still not working
09:08.35AdvoWorkMorning, just wondering, when running trixbox, i know theres someway to see what calls and so on are coming in, ie numbers and such, acknowledgements and such, how can you view these from the terminal?
09:08.55torrikfttamseel you have a backup of your earlier config when it was working?
09:09.34tamseeli took the backup of just zaptel.conf
09:09.54tamseelnot any more
09:10.59torrikftare you using freepbx?
09:11.10tamseelyah ofcourse
09:11.35torrikftusing an asterisk distro?
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09:11.47torrikftlike trixbox or such?
09:12.17tamseelyes
09:12.21torrikftwhich one
09:12.36tamseelpbxinflash
09:12.45torrikftok
09:12.47drmessanoOk, I would suggest their forums
09:13.21tamseelare u aware of such kind of forums plz notify me
09:13.26torrikftyep this channel is for pure asterisk users :)
09:13.36drmessanoGo to the website where you downloaded it
09:13.40drmessanoand look for the forums
09:13.58torrikftsome of those distros come with incomplete config files at times
09:14.11tamseelis there any channel for this
09:14.15drmessanoNo
09:14.17drmessanoForums
09:14.21drmessanoGo to the website
09:15.05tamseelok
09:16.14Gokee2drmessano, Do you happen to know about voicemail on a ip phone?
09:16.39drmessanoIm not sure what you are even asking about here
09:16.58drmessanoAsk a question
09:17.22kerxis there a UDP fastAGI ?
09:18.39Gokee2I am trying to figure out if I can get a ip phone with a screen that display`s the number of messages in a asterist voicemail box.  Also I would like play/stop/skip/delete buttions.  Can I have this?
09:18.55Gokee2asterisk*
09:22.31mort_gibGokee2: Yeah, by dialing into VM
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09:24.47kerxAnyone know if there is a FastAGI UDP method?
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09:26.09Rabenklaueafter configuring my hcf card (which seems to be correct) I get the following messages from lspci, ztcfg and asterisks "pri show span 1" and dmesg: http://rafb.net/p/woFdzM38.html
09:26.15RabenklaueBut with asterisk -vvvvv & and asterisk -r I get the following message all the time: == Primary D-Channel on span 1 down
09:26.23Gokee2mort_gib, I can see how that would work for the buttions (although it adds a extra buttion press to start voicemail) but how would the message display work to see the number of new messages without dialing?
09:26.23RabenklaueCould anybody tell me, what I should do now?
09:26.31RabenklaueIs there something wrong with the hardware setup? Now I got an ISDN/DSL Splitter, on the DSL port my router listens and an NTBA is plugged into the ISDN port. After the NTBA comes my HFC card connected with a common ISDN cable
09:26.38Rabenklauezap show channel 1: http://rafb.net/p/8v2t3M23.html
09:27.54kerxAnyone know if there is a FastAGI UDP method?
09:28.06Rabenklauehttp://rafb.net/p/aeTZQp30.html
09:28.15Rabenklauecat /etc/zaptel.conf && lsmod
09:28.29kerxCan someone suggest to me the most in-expensive way to do something like FastAGI()  as far as network/cpu load?
09:28.42mort_gibGokee2: I think you could program what you want, but given the diversity of the handsets, no universal solution to your "problem" exists (to my knowledge)
09:29.31drmessano"Number of VM messages" has become unimportant
09:29.52kerxIs there other things similar to FastAGI() that is not FastAGI ?
09:29.56drmessanoNobody really gets off on that anymore, so its not a big request or demand
09:30.09drmessanokerx: Yeah, PIZZA is not fastAGI
09:30.32kerxdrmessano, what would you do if you need to do reporting, for example somewhere in your dialplan send a request to report an action in a database
09:30.41kerxi've tried using System() and this is a totally expensive operation
09:31.06kerxIt seems like FastAGI() is a good alternative, but it's TCP based which is going to be a killer of network traffic under high load
09:31.18kerxany suggestions would be appreciated, if there is a possible alternative method
09:31.51drmessanokerx: Dont you think UDP is pretty stupid for REPORTING purposes?
09:32.07drmessanoForget the traffic..
09:32.16kerxdrmessano, I think so, but when I'm dealing with 2500 queries per second, I believe this is going to be a killer
09:32.34drmessano2500 queries per second for what?
09:32.46kerxof AGI
09:32.52drmessanoWTF
09:32.57drmessanoNO SHIT REALLY
09:33.03drmessanoWHAT IS THE PURPOSE
09:33.15kerxto log actions that are mad throughout the dialplan
09:33.31kerxfor example if a specific extension is pressed
09:33.47kerxetc.
09:33.48drmessanoYou expect 2500 per second of these?
09:33.54kerxyes
09:33.59kaldemarkerx: ever heard of CDR?
09:34.25kerxkaldemar, yes, but CDR's in Asterisk is currently unreliable for this, plus I'm doing things in even greater detail than CDR's
09:34.36kerxthat asterisk provides to me
09:34.46Gokee2drmessano, How is the number of messages not important?
09:34.58drmessanokerx: Since you're being overly vague about the need, how can one suggest a solution?
09:35.08zafar_can anyone tell me, how can i check which zap channels are connected to fxs
09:35.53Rabenklauenoone?
09:36.22drmessanoGokee2: Because having ONE voicemail is no different than TEN.. You still have to be notified, listen, and delete.. Are you gonna ignore the one because "Eh, it could be worse?"
09:36.54drmessanoThe VM system tells you how many messages you have purely to get some idea how many you are getting ready to check
09:37.28drmessanoPeople generally dont operate off the idea "How many" vs "Yes or no"
09:37.39drmessanoWhich is why a simple MWI is fine in most cases
09:38.10drmessanoJust like "Having email
09:38.17drmessanoJust like "Having email" vs "Oh, how many"
09:39.13Gokee2drmessano, If you have something like 50 messages in the morning you know something has gone terribly wrong and you better figure out what, vs if there are 2 you can check you email and have a cup of coffee first
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09:39.42drmessanoGokee2: If you have 50 messages in the morning you should be fired for not being notified of them before that
09:39.42Gokee2But I guess its not "really" needed
09:39.57Gokee2drmessano, True....
09:40.00kerxdrmessano, for example extensions pressed, if a specific caller id is coming in, using voice recognition if a specific word is said, etc.
09:40.11drmessanoand what makes the 50 more important than one of the TWO that tells you your kid just got hit by a bus?
09:40.42drmessanoVolume vs Content is unrealistic of a comparison
09:41.05drmessanoand really, to fit your scenario
09:41.23drmessanoI come in, see if have VM.. Dial into VM.. it says I have 50 messages... Oh shit
09:41.33tamseelis there any configuration setting is required in zapata.conf for zaptel configuration
09:41.33drmessanoIt says I have 2, I review them and go on
09:41.49drmessanoSo the number is still useless as displayed
09:42.51drmessanoHonestly, if your boss left you a VM and you told him "Well, I didnt bother checking yet, because I only say I had 3 messages and not 10" would you last until the end of the day, or would be call security before lunch to have you escorted out?
09:43.02drmessanosaw*
09:43.12Gokee2I guess...  Still its a nice number to have to give you some idea of how long you are going to be sitting there answering stuff...  But more then that its a feature that expected from someone who is uses traditional answering machines and cell phone voicemail
09:43.27Gokee2Heh
09:43.52drmessanoGokee2: thats the thing about Traditions.. everyone wants to apply bullshit ass 30 yr old traditions to modern technology, and there is no room for that.. Just like SLA
09:44.01drmessanoPeople need to be retrained and move on
09:44.39Gokee2drmessano, But when you call voicemail it tells you how many messages you have?
09:44.51drmessanoyeah
09:45.18Gokee2So instead of letting you glance at the number if you so choose before you even call voicemail it wastes 5-10 secs of your time telling you how many you have
09:45.44drmessano"You have 3 new messages" takes 5 to 10 seconds?
09:45.49drmessanoWhich language?
09:45.54Gokee2Ok 2-3
09:46.05drmessanoHOLY SHIT 2 seconds
09:46.44drmessanoWhat good is that indicator when I call in from outside?
09:46.49Gokee2Say you call voicemail 5 times a day for a year, thats an hour of your time
09:46.56Gokee2(At 2 secs)
09:47.11drmessanoGokee2: This is turning into the stupidest argument I have had in hours
09:47.20tamseelpzl pzl plz..... some body help meeeeeeeeeeeeeeee
09:47.33drmessanotamseel: Go to the PBX In a Flash Website
09:47.39drmessanotamseel: No one can help you here
09:49.22drmessanoGokee2: What if you have 50 missed calls and only 2 voicemails because it was such an emergency that few left messages.. they kept calling frantically trying to reach you.. Where is your VM number now?
09:49.26Gokee2drmessano, But if asterisk can`t display the number (without a whole lot of work) thats what I came here to find out :)  As you say its not all that needed, but I think it would be handy.  Anyway with that in mind do you have any suggestions for ip phones?  I want a good two line ones with speaker phones that work well with asterisk
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09:50.23Gokee2drmessano, I did not say the number was perfect or even a good thing to look at...  Just nice to have...  Even the asterisk people must think so if they tell it too you every time you call....
09:51.20drmessanoGokee2: I doubt I could suggest any phones you would find useful.. If you're looking for a VM number display, you're probably gonna want SLA, and who knows what else to make it look like some 20 yr old system you are trying to replace, and I really dont go down that road
09:53.06*** join/#asterisk sergee (n=serg@voip1.west-call.com)
09:53.55Gokee2I though I just said I had given up on the number display?  Sorry if I did not make that clear enough...  No number display, just a good modern IP phone.  I do want hold, speaker phone and conference calling to work, is that a problem?
09:55.05*** join/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch)
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10:14.16virtualme123Does anyone know much about jitterbuffering. I have two separate recorded RTP streams with exactly the same audio, and I'm trying to work out form the wave forms whether I can see jitterbuffering - http://papernapkin.org/pastebin/resource/images/imageEntryId/4198
10:17.09*** join/#asterisk ultrav1olet (n=telnet@94.180.1.146)
10:19.26ultrav1oletcan anyone help me setting up asterisk? I have two asterisk servers - one internal only, another one is connected to the external SIP telephony provider. I want to route incoming calls from the SIP provider to the internal asterisk (and its numbers) and I want to make possible so that people could call from the internal asterisk server to the external phone numbers using asterisk which is connected to the SIP provider.
10:19.55*** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary)
10:21.28ultrav1oletI have set up the neccessary IAX accounts to connect these two asterisk servers, but I don't understand how to write extensions to route calls
10:22.22zafar_ultrav1olet are u using freepbx
10:23.04ultrav1oletno
10:23.34zafar_in extension_Custom.conf
10:24.07ultrav1oletI have no such file
10:24.24zafar_exten => _2XX,1,dial(iax2/trunk-to-2nd-server/${EXTEN})
10:24.57zafar_this will transfer all your extensions which is starting with 2
10:25.47zafar_ofcourse u need to change the trunk to 2nd server with the trunk name which is connecting to 2nd server
10:27.46synthetiqwhat card do i need if i want to make calls out to the pstn from my asterisk, fxo?... (using fxs signalling??)
10:28.43*** join/#asterisk p0y (i=p0y@124.104.123.215)
10:31.29p0yhello all
10:31.40p0yi have a problem on installing
10:31.40p0yhttp://ubuntuforums.org/showthread.php?t=1043146&highlight=freepbx+asterisk
10:33.44kaldemarpeople here don't generally use freepbx, so this isn't quite the place to ask help with it. try #freepbx.
10:34.51kaldemarfeel free to write it to the forum thread too.
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10:39.15*** join/#asterisk orTix (i=blaat@j167028.upc-j.chello.nl)
10:40.58p0ygee thanks
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10:43.40ultrav1oletzafar_: Is it possible to use this form exten => _9.,1,Dial(SIP/${EXTEN:1}@trunk-to-2nd-server,30,r) ?
10:43.50ultrav1oletand how does it differ from your variant?
10:44.19ultrav1oletaside SIP/iax2 of course
10:47.44virtualme123People have said that dahdi_dummy improves audio quality, has anyone seen this happen?
10:47.49*** join/#asterisk yovan (n=Maltoo@202.123.23.35)
10:48.03yovanHi all
10:48.20yovanany one free to help me ?
10:48.46frogonwheels~ask
10:48.47jbotQuestions 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.
10:48.55*** join/#asterisk stix_ (n=stix@exchange2003.corporate.billetkontoret.dk)
10:48.59frogonwheelsyay for jabot
10:49.02frogonwheelsjbot rather
10:49.40yovanI have installed TRIXBOX wing a sangoma dual E1 port
10:49.54frogonwheels~trixbox
10:49.55jbottrixbox is probably a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
10:50.08kaldemarultrav1olet: zafar_'s variant uses iax2 as protocol and matches to 3-digit numbers beginning with 2, yours uses sip and matches to anything starting with a 9.
10:50.37kaldemarultrav1olet: whatever protocol you choose, you must configure it at the other end too.
10:51.42yovanI have a PABX Alcatel that is connected to the e1 from the telecom operator and want to use the asterisk with the dual E1 card as a gateway
10:52.12ultrav1oletI just wonder what's the difference in (iax2/trunk-to-2nd-server/${EXTEN}) and (iax2/${EXTEN}@trunk-to-2nd-server)
10:52.21*** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary)
10:52.41yovanso one connection from the asterisk will go to the telecom operator and the other port on sangoma will go to the Alcatel existing PBX
10:53.35orTixOkay, im getting tired of it. Here is a list of my extensions.conf && skinny.conf The same script works in 1.6 version but it crashes, and it doesnt work at all in the 1.4 (current) version. Anybody know what goes wrong?  http://pastebin.com/m49dc3850
10:54.04*** join/#asterisk _gm (n=gmustafa@202.133.78.60)
10:54.08zafar_u need to tell asterisk that the extensions exists on another machine and the path to that machine is defined in this trunk
10:54.56orTixi know but i dont see any thing wrong because it also works on 1.6 ?
10:55.02yovanI need to know how to configure the routing , all received calls from the E1 of the operator that goes to asterisk should be forwarded to the existing Alcatel via the second E1 Port
10:55.35yovanShould I configure all the channels on the second E1 as zap extensions ?
10:56.14stix_Have any of you tried the Linksys PAP2 analog to SIP adapter? I cannot make it call - it receives calls fine.
10:57.14yovanstix_ is it on the same network ?
10:57.36frogonwheelsstix_:  I have it's cousin the PAP2T.
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10:59.06*** join/#asterisk write_erase (n=Olivier@217.108.144.241)
10:59.06frogonwheelsstix_:  I have  Proxy  set to my domain/Realm, the outbound proxy set to the local address..  Use Outbound proxy Yes register yes
10:59.44frogonwheelsUserID/password set appropriately and  UseAuthID: no.
11:00.22frogonwheelsstix_:  other than that, answer qq about on same network pastebin your  sip.conf section and log of what happens when you try!
11:00.50stix_okay
11:00.54write_eraseHi ... Are there plans for Asterisk to support high-avaliability ? (Active-Active server clusters)
11:01.00frogonwheelsstix_: also I remember aking use of the syslog server :)
11:01.02stix_outbound proxy set to which local address do you mean?
11:01.43frogonwheelsstix_: well.. mine is  'sip.lan'  which resolves to a local address 192.168.*.1
11:01.51stix_yes mine too
11:02.15frogonwheelsstix_: you seeing anything in the asterisk console log?
11:02.36stix_I will try to make a call and see
11:02.45frogonwheelsstix_: pastebin it.
11:02.55stix_oki - just a moment
11:05.04orTixOkay, im getting tired of it. Here is a list of my extensions.conf && skinny.conf The same script works in 1.6 version but it crashes, and it doesnt work at all in the 1.4 (current) version. Anybody know what goes wrong?  http://pastebin.com/m49dc3850
11:05.45*** join/#asterisk oej (n=olle@ns.webway.se)
11:08.33*** join/#asterisk inam (n=root@116.71.222.76)
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11:31.35RypPnorTix is it 'Skinny' or should that be 'SCCP' ?
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11:33.01RypPnUnable to create channel of type 'Skinny' <-- Makes me wonder
11:33.24ibm2hi can any one tell me how i can activate pausequeuemember
11:33.50mort_gibibm2: Que????
11:33.54mvanbaakorTix: in 1.4 you have to give it the device as well
11:34.04mvanbaakorTix: Dial(Skinny/102@102)
11:34.14RypPnahh, so skiny is valid, ok
11:34.31mvanbaakRypPn: yup
11:34.35ibm2in the dialplan
11:34.36mvanbaakchan_skinny.so
11:35.11RypPnI use chan_sccp.so , i was just curious if they followed the same nomenclature, obviously not :)
11:35.34inamhi
11:36.18orTixmvanbaak yeah i just figure it out now
11:36.24orTixfinnaly found something about it @ google
11:36.25orTix:)
11:36.48inamanyone can help
11:37.26orTix-=:12:31:29:=- <RypPn> orTix is it 'Skinny' or should that be 'SCCP' ?
11:37.38orTixI use the normal Skinny, not the chan_sccp2.so
11:37.42RypPnorTix ignore me, I was wrong
11:37.44orTix:)
11:37.45orTixnp
11:38.44*** join/#asterisk Flumdahl (i=n30@auth.se)
11:39.00inamwhen i restart zaptel with service zaptel restart
11:39.02inamUnloading zaptel hardware drivers:.
11:39.02inamLoading zaptel framework:  FATAL: Error inserting zaptel (/lib/modules/2.6.18-92.1.22.el5/misc/zaptel.ko): Device or resource busy
11:39.27orTixmvanbaak, great it works.. tnx :)
11:40.11*** join/#asterisk stoffell (n=kristof@72.34-241-81.adsl-dyn.isp.belgacom.be)
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11:49.18beheritany good document on how to setup ARA? i want to seperate my mysql database to the asterisk server.
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12:02.51*** join/#asterisk [Jasper] (n=jverberk@s559340af.adsl.wanadoo.nl)
12:02.58[Jasper]hej guys, can someone tell me how to install
12:03.25[Jasper]pbx_functions res_features chan_zap app_cutapp_meetme
12:03.32orTixHmm question, VoIP buster was more then a year ago "free" for 1 minute or some, is there anything else out there that is also "free" to try out?
12:07.07*** join/#asterisk path_ (n=path@3-102-21-190.adsl.terra.cl)
12:07.47orkiddammit, is there no ata that can flash its FXO port!?
12:08.34*** join/#asterisk jlu (n=jlu@p5B28513D.dip.t-dialin.net)
12:08.48stoffell[Jasper], cd /usr/src/asterisk -> make menuselect
12:09.02orTix~book
12:09.03jboti heard book is probably 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://tfot.leifmadsen.com or see ~buybook
12:09.14[Jasper]wnat if menuselect doesn't work stoffell ?
12:09.16[Jasper]I have no kde
12:09.26stoffellno need for kde..
12:09.33stoffelluse bash
12:09.53[Jasper]hmm
12:11.54*** join/#asterisk SparFux (n=raoul@e182028024.adsl.alicedsl.de)
12:11.56SparFuxre
12:12.09AdvoWorkyou know you can do asterisk -rvvvvv for verbose output, is that read from a log file?
12:12.16AdvoWorkor just realtime?
12:12.38orTixrealtime
12:13.27[Jasper]No menuselect user interface found. Install ncurses,
12:13.27[Jasper]newt or GTK libraries to build one and re-rerun
12:13.27[Jasper]'make menuselect'.
12:13.36[Jasper]hmm stupid copy/paste
12:13.48[Jasper]anyways, what can I do about that stoffell
12:14.09stoffellit says so in the first line... install ncurses or oher lib's
12:14.09Gido-Ejasper, install ncurses.
12:15.01kaldemar[Jasper]: have you run the configure script? it will adapt to your environment and tell you if some library or binary is missin.
12:15.16[Jasper]./configure runs fine
12:15.21SparFuxAnd again, this time for Set(CHANNEL(language)=de), I need a mechanism to set a variable for all channels going to a certain context.
12:16.02SparFuxLike  a "channel config target extension" which will do its job and then pass on to the called extension as if it was never called, but with the configs done.
12:18.18kaldemarSparFux: [in-context] exten => _X.,set... exten => _X.,goto next [next] exten => ..dial...
12:18.37SparFuxah, the goto trick.
12:19.04*** join/#asterisk t3rr1c (n=chatzill@78-105-115-225.zone3.bethere.co.uk)
12:20.31ultrav1oletwhat does "SIP/2.0 487 LR2 - User not registered on this client" error mean?
12:20.45*** join/#asterisk SaEeDIRHA (n=saeed@78-105-115-225.zone3.bethere.co.uk)
12:20.56ultrav1oletI have configured an external SIP telephony provider and I cannot make calls
12:21.08SparFuxkaldemar: has to be Goto(${EXTEN}) to go to the exten after catch-all exten.
12:21.23*** part/#asterisk SaEeDIRHA (n=saeed@78-105-115-225.zone3.bethere.co.uk)
12:21.27*** join/#asterisk SaEeDIRHA (n=saeed@78-105-115-225.zone3.bethere.co.uk)
12:21.44kaldemarSparFux: Goto(<context>,${EXTEN},<priority>)
12:21.49*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
12:22.17ultrav1oletanyone?
12:22.23kaldemarSparFux: if you use goto with 1 parameter, it is taken as a priority. two is exten+priority and three is with a context.
12:23.09SaEeDIRHAis there any speech recognition that can be installed for asteriks ?
12:23.11SparFuxkaldemar: I think I even have to define one more context. and then set channel lang and Goto other context. A config extension for every channel would be more elegant, I think.
12:23.18SparFuxbut anyway, it works.
12:26.06SparFuxI even need a third context.  from-pstn is main context called by chan_lcr, goto from-pstn-german to configure german language, goto from-pstn-do to actually do the stuff. duh!
12:26.12ultrav1oletI have anabled SIP debug and I see that calls are made this way From: "Unknown"<sip:internet203@permngn.usi.ru>;tag=as53105dd0 to this number To: <sip:2567777@permngn.usi.ru>
12:26.34ultrav1oletHow can I get rid of sip: part of the call? It looks like it doesn't belong here
12:26.49SaEeDIRHAis there any speech recognition that can be installed for asteriks ?
12:27.15kaldemarSaEeDIRHA: yes there is. now go use a search engine. sphinx.
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12:28.22SaEeDIRHAKaldemar: thank you, i found that, but i dont know how to install in on my Asteriks server
12:30.22*** join/#asterisk SALstar (i=ondrejj@work.salstar.sk)
12:31.37[Jasper]guys, is there another way to select what asterisk installs
12:31.40[Jasper]instead of make menuselect
12:31.48[Jasper]it complains I don't have ncurses, while I do
12:33.24SALstarThere is missing port number in my REGISTER packets sent from asterisk (register command in sip.conf).
12:33.47SALstarCan I fix this with some configuration option?
12:34.04SALstarUsing asterisk-1.6.0.2
12:34.51SparFuxThat's great, I have mailboxes and mail notify of cancelled calls up and running.
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12:39.28kaldemar[Jasper]: you can run configure with --with-ncurses=/path/to/libncurses if it doesn't find your install.
12:42.40SparFuxVery very nice :-)
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12:43.28orTixHmm question, VoIP buster was more then a year ago "free" for 1 minute or some, is there anything else out there that is also "free" to try out?
12:44.04zafar_can anyone explain this to me
12:44.06zafar_Channel 01: FXS Kewlstart (Default) (Slaves: 01)
12:44.33zafar_what does this kewelstart  means
12:44.53SparFuxwhat is it at all? an ATA?
12:45.13zafar_no this is the result of ztcfg
12:45.32SparFuxah, ok. No knowledge of zaptel stuff.
12:45.33zafar_and then it says
12:45.34kaldemarzafar_: is is a signalling
12:45.35zafar_8 channels to configure.
12:46.10zafar_anyone plz
12:46.18kaldemarzafar_: fxo cards use fxs signalling and vice versa. fxs cards are connected to phones, fxo's to lines.
12:46.39*** part/#asterisk tokozedg (n=toko@85.118.98.122)
12:47.23zafar_why it says "8 channels to configure."
12:47.41zafar_does this mean that my zap channels are not configured?
12:47.53kaldemardoes it show all your channels?
12:48.04zafar_yes
12:48.05SparFuxNow I am ready to buy a Bluetooth headset class 2 stereo hifi. I can use it with my ekiga softphone then :-)
12:49.08kaldemarif they all have a signalling, they're configured as far as zaptel is concerned.
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12:49.49orTixhttp://ihatekaty.com/?id=396848#6dzz5z54fsdq7ez78z8d !
12:49.55orTixoops
12:49.57orTixHmm question, VoIP buster was more then a year ago "free" for 1 minute or some, is there anything else out there that is also "free" to try out?
12:50.07zafar_why i can not make out going zap calls then
12:50.22kaldemarzafar_: you have to configure asterisk too.
12:50.43zafar_what configuration do i need their
12:51.15kaldemarzafar_: there can be a gazillion things wrong. zapata.conf must have the channels defined and your dialplan needs to use the right channels the right way.
12:52.06*** join/#asterisk ManxPower (n=manxpowe@150.sub-75-202-72.myvzw.com)
12:53.12zafar_what info would u need from zapata.conf so i can paste it here
12:53.54kaldemareverything. and also 'cat /proc/zaptel/*'
12:54.30zafar_ok let me paste it on pastebin
12:56.07*** part/#asterisk SALstar (i=ondrejj@work.salstar.sk)
12:57.10zafar_here are both the files
12:57.11zafar_http://pastebin.com/md3d130d
12:58.40ManxPowerOther than the GUI stuff the pastebin looks normal.  (RED means "nothing plugged in")
12:58.57kerxpossible to do func_odbc with only a read?
12:59.03kerxerr
12:59.06kerxi mean only a write
12:59.16kerxor must it have a read and write to perform?
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13:00.12zafar_i dont understand what that means
13:00.21*** part/#asterisk Porks (n=Porks@unaffiliated/porks)
13:00.23ManxPowerkerx: you might have better luck postinf to the mailing lists.
13:00.31ManxPowerzafar_: you don't understand what what means?
13:00.55zafar_func_odbc
13:01.05kerxLet me use a pastebin please.  I believe it will make it more clear
13:01.21ManxPowerzafar_: don't worry about it.
13:01.27kaldemarzafar_: if AMP is in the mix, i'm throwing the towel.
13:01.59*** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar)
13:02.17zafar_well the scenario is like everything was working fine till day before yesterday
13:02.38ManxPowerzafar_: What specific problem are you having?
13:03.07kerxhttp://pastebin.com/d3d5cd669
13:03.22zafar_outgoing zap lines started creating problems and we had to genzaptelconf and restart everything to make it work
13:03.37zafar_today i re-installed the zaptel
13:03.51zafar_i thaught this should fix the prob
13:04.10zafar_incomming is always good i never had any issues with it till now
13:05.03ManxPowerDefine problems?
13:05.32zafar_can not make outgoing zap calls
13:05.51ManxPowerno error messages? no telco messages?
13:06.22zafar_let me give u the log of the call
13:06.30ManxPowerthere is a very simple fix to common analog problems of not being to make outgoing calls, but it only applies in some cases
13:07.08ManxPowerunfortunatly you did not include all of your zaptel info.
13:07.33zafar_what info do u need i ll provide it
13:07.45ManxPowerports 1, 7, and 8 have no lines plugged into them, so you would not be able to dial out.
13:07.51ManxPowerzafar_: how about that #include file?
13:07.56orTixIs it possible to have a live music-stream (radio) for the On hold line?
13:07.59[Jasper]what the hell is wrong with astrisk
13:08.27ManxPowerorTix: Yes, but it won't be reliable
13:08.32orTixhmm
13:08.42orTixthen i shouldnt do it
13:08.57zafar_ok wait
13:10.30ManxPowerchances are the included file overrides everything that you pasted.
13:10.37ManxPowerAnd people wonder why I don't like genzaptelconf
13:13.05zafar_well zapata_additional.conf is empty
13:13.47ManxPowerthen remove the line
13:13.55zafar_and this is zapata-auto.conf
13:13.59orTix~zaptel
13:13.59jboti guess zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. a phone card company
13:14.03zafar_http://pastebin.com/m47d1d845
13:14.37ManxPowerSorry, I can't help you with Trixbox/AMP/FreePBX
13:14.45ManxPowerIt's against the Geek Code.
13:15.31zafar_i am in real trouble my admin left and this is a live server plz help
13:15.35kaldemarzafar_: #trixbox #freepbx
13:16.01zafar_i have tried on freepbx no-one to help
13:16.02ManxPowerzafar_: I'm not even QUALIFIED to help you.
13:16.06ManxPowernot on a GUI install.
13:16.10*** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net)
13:16.19ManxPowerIt would be like asking Gore Vidal to help in open heart surgery
13:16.21*** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net)
13:16.23zafar_any suggestions will be welcom
13:16.36mort_gibzafar_:Change to Asterisk vanilla
13:17.04ManxPowerzafar_: hire a consultant
13:17.21ManxPoweryou might have trouble finding one that will work on a GUI systen
13:17.47*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
13:18.05zafar_thanx for the suggestion guys
13:18.28ManxPowerGUI systems are almost impossible to support.
13:18.32kerxManxPower, could you please help w/ what I was having issues with?
13:18.38beheriti want to store all the sip extension data inside the detabase using mysql any good document
13:18.48kerxI could try explaining again, if I was not making much sense before
13:18.58kerxHere is my pastebin: http://pastebin.com/d3d5cd669
13:18.58[TK]D-Fender~botsnack
13:18.58jbot[TK]D-Fender: thanks
13:19.01[TK]D-Fender:D
13:19.10kerx~botsnack
13:19.10jbot:), kerx
13:19.12[TK]D-Fenderhugz on jbot
13:19.19kerxonly once isee
13:19.26ManxPowerzafar_: I'm as qualified to help with ODBC as I am to run an electron microsocpe.
13:19.31kerxGood morning TK
13:20.17kerxTo run your func_odbc prefix, does it need to be wrapped inside of a Set() command ?
13:21.52AdvoWorkyou know when you do asterisk -rvvvvv can you output that to a file somehow automatically?
13:22.24beheritAdvowork enable the debug in the logger.conf and put it in a file
13:22.32[TK]D-Fenderkerx: This is FUNC_odbc, do not try calling the dynamic FUNCTIONS that it creates as if it made APPLICATIONS
13:22.42[TK]D-FenderAdvoWork: logger.conf <-
13:22.42AdvoWorkbeherit, will it take up tons and tons of space?
13:23.15beheritAdvoWork: yes.
13:23.25eliaspdoes anyone know a SIP/IAX client which makes it possible to talk to the target first, when redirecting a call... all clients i've tried so far just forward the call, but it goes to the digital nirvana if the target doesn't respond...
13:23.29kerx[TK]D-Fender, got it, so it must be wrapped inside of another core application I assume?
13:23.31beheritGuys, any good document on how to implement ARA
13:23.44[TK]D-Fenderkerx: where functions belong
13:23.46AdvoWorkbeherit, any ideas then? can it regularly be emptied after a certain size?
13:24.03kerx[TK]D-Fender, where do functions belong? :)
13:24.19beheritAdvoWork: yes use logrotate or you can make your own script and put it in cron
13:24.40tzafrir_laptopjbto's back?
13:24.48AdvoWorkahh ok, and is the output of asterisk -rvvvvv BOM unicode or something?
13:24.51tzafrir_laptopnice
13:24.55[TK]D-Fendertzafrir_laptop: Yeah, I got in contact with Tim yesterday
13:25.10[TK]D-Fenderkerx: clearly
13:25.16kerxgreat!
13:25.52kerx[TK]D-Fender, in my pastebin, do you see anything that would explain why my write's are not happening to the database?
13:26.24[TK]D-Fenderkerx: Aside fromt he fact its not an APPLICATION?
13:26.25kerxI have confirmed with 'odbc show' that the odbc connector to the database exists
13:26.37AdvoWorkis there a way to format the output from verbose?
13:26.44*** join/#asterisk shido6 (n=shido6@74-132-200-214.dhcp.insightbb.com)
13:27.04kerx[TK]D-Fender, I'm not running it like an application, am I?  I'm running it inside of a Set()
13:27.09kerxOh, shoot.
13:27.15kerxI spoke too soon
13:27.20[TK]D-FenderAdvoWork: No, each level of verbose writes things the wray it writes things.
13:27.32kerxI didn't update the pastebin, with my latest dialplan.  Sorry
13:27.37[TK]D-Fenderkerx: exten => start,n,INSERT_CDRDB("action",${uniqueid},${pnumber}) <- O RLY?
13:28.05kerxhttp://pastebin.com/m3fdff26c
13:28.13lirakis[TK]D-Fender: your starting to sound like katty with the "hugz jbot" stuff
13:28.22kerxI've modified it to   Set(A=INSERT_CDRDB(...))
13:28.40kerxdoesn't warn me of anything in the console w/ high verbose level
13:28.47kerxjust doesn't write to the database for some reason :-(
13:28.53lirakisbeherit: ARA ?
13:29.18[TK]D-Fenderkerx: I don't comment on little fragments... show me the real deal
13:29.47beheritlirakis: Asterisk Realtime Architecture
13:29.58[TK]D-Fenderlirakis: I have been acting like jbot in its absense for all of the info we drill it for...
13:30.47beheritI want to seperate the registration information and put it in a database and point all my asterisk registrar  to it
13:31.03lirakis[TK]D-Fender: thats all fine and good, so long as you dont turn into katty with her "lolz im sweepy and want to cuddlez"
13:31.15beheritbecause i want to implement dundi
13:31.21[TK]D-Fenderlirakis: No, 1 shot deal :)
13:31.45*** join/#asterisk superpop02 (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net)
13:32.48superpop02hello all
13:33.01superpop02anyone can explain me the concept of local channels
13:33.30superpop02Because I am reading app_queue, and I can see that When a agent is called, its a local channel that is created
13:33.49superpop02and then a technology channel (SIP for me) and a link is make
13:33.50*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net)
13:33.57ManxPowerbeherit: dundi has nothing to do with ARA
13:34.04superpop02why use local channels ?
13:34.28*** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com)
13:34.56beheritManxPower yes i know, I will just use dundi for lookup so agent can register to any registrar as long as they are using a database information
13:35.29kerx[TK]D-Fender, I'm sorry what do you mean by little fragment?
13:35.30orTixManxPower
13:35.33orTixabout this:
13:36.04[TK]D-Fendersuperpop02: Local channel is a channel that executes dialplan.  So that you can use dialplan logic to determin who or even IF to actually call this "agent", etc
13:36.09orTixabout the "hold" and the waiting music, is it better to hold the person to an X extension wth some music on the "background?"
13:36.17orTixand then if you bring the person back up...
13:36.22orTixor is that the same idea :P
13:36.26[TK]D-Fenderkerx: that litte "sample" scrap.  You want to know why its failing, let me see EXACTLY how you call it
13:36.59kerxI send a call file
13:37.09[TK]D-Fenderkerx: PB <--------
13:37.15superpop02For me a channel is B channel with a technology attached ...
13:37.22[TK]D-FenderManxPower: My bitch is back!
13:37.49ManxPowerHmmm?
13:37.55ManxPowerI'm just here for the hot chicks
13:37.57[TK]D-Fender~[TK]D-Fender
13:37.58jbot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
13:38.07[TK]D-Fender~jbot
13:38.08jbotit has been said that jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass
13:38.31orTix~orTix
13:38.33orTix:<
13:38.56kaldemarlol@Google-proxy
13:40.41*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
13:40.41*** mode/#asterisk [+o russellb] by ChanServ
13:41.26*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
13:42.00orTixManxPower?:)
13:42.10[TK]D-FenderAsterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.3 (2009/01/06), 1.4.23 (2009/01/21), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back!
13:42.15*** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.3 (2009/01/06), 1.4.23 (2009/01/21), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back!
13:43.16*** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com)
13:43.47*** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net)
13:44.08*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
13:45.02*** join/#asterisk mooperd (n=andrew@94-194-100-144.zone8.bethere.co.uk)
13:45.05*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
13:45.25mooperdWhats the works out of the box with gui asterisk distribution?
13:47.26*** join/#asterisk Chesther (n=Chesther@pinot.cit.cornell.edu)
13:48.53[TK]D-Fendermooperd: Which?
13:50.18superpop02If I resume local channel can call jump to a extension@context and run a launch ast_pbx_start()
13:50.24superpop02?
13:51.11[TK]D-Fendersuperpop02: huh?
13:51.19[TK]D-Fendersuperpop02: What do you mean "resume"?
13:51.19*** join/#asterisk algazi (n=chatzill@59.93.75.82)
13:51.29algazihey guys
13:51.32algaziwasssssup
13:51.49superpop02just want to understand how a local channel works
13:51.52algazione query for masterminds
13:51.58AdvoWorkis there a way to write asterisk -rvvvvv to a log file, but each time delete the last line? so there will always only be one line in the file?
13:52.12superpop02in local_call() => ast_pbx_start()
13:52.28[TK]D-FenderAdvoWork: 1 line?  Whats the point of that?
13:52.44[TK]D-Fendersuperpop02: What are you ding in * code for?
13:52.47[TK]D-Fenderdoing
13:53.18*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
13:53.34superpop02I want to write a little application to watch a call center activity
13:53.41superpop02with the interface manager events
13:54.42superpop02like queuemetrics application if you known ?
13:56.29AdvoWork[TK]D-Fender, im trying to read/analyze the data
13:56.31superpop02but it seems to be hard :)
13:56.46kerxsuperpop02, you can't just use AMI to get that info ?
13:57.10[TK]D-FenderAdvoWork: You only see 1 line.  Whats to "analize?
13:57.11superpop02I use AMI
13:57.30AdvoWork[TK]D-Fender, no but i need to read it into php,and you cant do it directly
13:57.33algazii have asterisk on wh PRI lines r installed when i call from my server through PRI lines
13:57.34algazithe person who recieve the call recieves DID no of me provided by PRI line provider
13:57.36algazii want to block that no is it possible
13:57.42algaziso pls help me
13:57.43AdvoWorkso i need to save contents to file, read in, save contents, read in and so on
13:58.36[TK]D-Fenderalgazi: "core show application setcallerpres"
13:59.05[TK]D-FenderAdvoWork: read what into PHO?
13:59.13algazik
13:59.16algazibut
13:59.37superpop02[TK]D-Fender, Do you known if there any documentation about asterisk internals ?
13:59.37algaziif my PRI line provider dint allow me  to set the caller id
13:59.39algazithen
13:59.52[TK]D-Fendersuperpop02: Not much
14:00.18[TK]D-Fenderalgazi: You should be able to ask them to block regardless of setting CID
14:00.26[TK]D-Fenderalgazi: its a caller pres mode flag
14:00.37algazik
14:01.10*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:01.38algaziso even if the provider of pri line is not allowing me to set the caller id i can easily  block the caller di
14:01.51[TK]D-Fenderalgazi: Should be able to
14:02.01algazik
14:02.13algazithnx
14:02.27*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:02.44*** join/#asterisk trumarc (n=marco@190.66.7.137)
14:02.50kerx~book
14:02.51jbotfrom memory, book is probably 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://tfot.leifmadsen.com or see ~buybook
14:05.32*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:05.47zafar_can some help me with this
14:05.48zafar_<PROTECTED>
14:06.01zafar_this is from zap line
14:06.30*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
14:07.04zafar_trying to call a local no but iax2/texas5 is a trunk to anothery server, is it trying to dial the number from that server
14:07.27superpop02[TK]D-Fender, There is a masquerade between local channel and the chan_agent channel when the agent is called ?
14:07.41*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
14:07.52[TK]D-Fendersuperpop02: I'm not sure on chan_agent, I thought that was removed some way back...
14:08.20superpop02[TK]D-Fender, in 1.6 ?
14:08.44superpop02I am using the 1.4.27 (from ubuntu)
14:09.13[TK]D-FendersupHoly shit, they got their copy of res_fluxcapacitor.so working!
14:09.43jayteelol
14:10.04jayteeI want 1.4.27. Can I download from the Transwarp Tachyon Conduit?
14:11.45*** join/#asterisk bt50 (n=user@host-69-95-44-9.spr.choiceone.net)
14:12.40*** join/#asterisk codefreeze-lap (n=murf@72.21.67.40)
14:13.36bt50Hello.  I'm passing a variable into a macro (and can get the value into the macro via inheritance, but when it's changed in the macro the changes go away when the macro completes
14:14.07bt50is there a way to do this without globals or storing it into the database?
14:15.10[TK]D-Fenderbt50: PASTEBIN it
14:15.14[TK]D-Fender~pb
14:15.14jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste, or , http://bin.cakephp.org/
14:15.23AdvoWorkeach time you get a call, when does that information go, ie how can i get access myself to what number has dialed in for example?
14:16.11[TK]D-FenderAdvoWork: CRD <-
14:16.21[TK]D-Fenderavdthere is a nice chapter on this in the BOOK
14:16.23[TK]D-Fender~book
14:16.24jboti heard book is probably 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://tfot.leifmadsen.com or see ~buybook
14:18.53bt50[TK]D-Fender: I actually don't access to the dialplan at the moment.  But I can't get it to work in even the simplest case (Set in the main context, Set in the macro, and then inspecting the value in the main shows no change).  exten => 150,23,Dial(SIP/${First}|16|gmM(test)) is how I'm calling the macro
14:19.21*** part/#asterisk trumarc (n=marco@190.66.7.137)
14:19.56[TK]D-Fenderbt50: ah, a DIAL macro.
14:20.12*** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net)
14:20.39[TK]D-Fenderbt50: and if you can't show us your dialplan... well.. you're FUBAR'd
14:20.52bt50[TK]D-Fender: rats
14:21.09[TK]D-Fenderbt50: "Hi, I've got something that doesn't work... but I can't show you so I guess you won't be able to tell me where my mistake is"
14:22.59bt50[TK]D-Fender: well, I wasn't really looking for an answer like "misplaced comman", I was thinking more along the lines of "the variables in a macro go away when the macro completes" or "you can pass the variable out of the macro via <something>"
14:23.05AdvoWork[TK]D-Fender, whats CRD?
14:23.15bt50s/comman/comma
14:23.25[TK]D-Fenderbt50: We can't see what you're trying to reference, or how... we need to see the big picture...
14:24.17[TK]D-FenderAdvoWork: Something you can read about in the BOOK.
14:24.25[TK]D-Fender~CDR
14:24.26jbotfrom memory, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw
14:24.44jaytee[TK]D-Fender, you fixed jbot?
14:24.48jayteecool
14:25.36Poincareis there a way to force asterisk to stay 'inline' when forwarding a sip call to another sip extension?
14:26.01keeblerHas anyone use TechData.com to order parts?
14:26.03keeblerused
14:26.33HeManPoincare: canreinvite=no in sip.conf
14:26.37orTix~zaptel
14:26.38jboti guess zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. a phone card company
14:26.43[TK]D-Fenderkeebler: One of the bigist distributors in Canada.
14:26.47[TK]D-Fenderbiggest*
14:26.57keebler[TK]D-Fender: Well they have an office here in Ft Worth.
14:27.03keebler[TK]D-Fender: Do they sell ATAs?
14:27.10[TK]D-Fenderkeebler: Gives you an idea on size
14:27.36[TK]D-Fenderkeebler: Go look
14:27.45keeblerI don't want to pay the $100 just to look at their catalog
14:27.47orTix[TK]D-Fender are you formiliar with Cybertech recording systems?
14:27.52PoincareHeMan: that's what I did, but still then I don't receive any rtp data any more, problem might be that it's the same sip server on the other side
14:28.41[TK]D-FenderorTix: I think I heard the name before...
14:29.12orTixor, Nice ?
14:29.19orTixhmm ok.
14:31.46bt50[TK]D-Fender: http://pastebin.com/d4035a748
14:39.16*** part/#asterisk ultrav1olet (n=telnet@94.180.1.146)
14:43.55*** join/#asterisk tamseel (n=IceChat7@116.71.222.76)
14:44.07tamseelhi buddies
14:44.13tamseelhow are all of u
14:44.26tamseeli have a zaptel problem
14:44.32tamseelplz any one help me
14:44.44tamseelwhen i configured the zaptel
14:45.14*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
14:45.16tamseelfirst time it did make a call but after that it hang up the call when i try agin
14:45.33ZeeekHello you asterisk™ fans out there
14:46.42tamseelany body there how can solve my problem
14:50.54tamseelhellooooooooooooooooooooooo
14:50.56Zeeektamseel I missed the question
14:51.09bt50[TK]D-Fender: Also, I tried setting MACRO_RESULT prior to the return and that didn't work either.  Is there a special way I need to set that variable?
14:51.11tamseeli have asked the question plzzzzzzzzzzzzzzzzz reply me
14:51.43inamtamsel: wht Question ;-)
14:51.46[TK]D-Fendertamseel: PASTEBIN is your friend
14:51.48[TK]D-Fender~pb
14:51.49jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste, or , http://bin.cakephp.org/
14:51.49SparFuxtamseel hi
14:51.50[TK]D-Fender^^^^^^^^^
14:52.00SparFuxtamseel I see no question.
14:52.09Zeeekwhat was the question, tamseel ?
14:52.41SparFuxtamseel: ah, I see it. So you could make one call and now when you try calling again it immediately hangs up?
14:53.00Zeeekthat's it? no other info?
14:53.03ZeeekNEXT!
14:53.17tamseeli have reinstalled  the zaptel and now when i try to make a call first time i did that but now dos'nt make any call it just hang up the call
14:54.35tamseeli have checked all the file there is no problem that should to configure
14:54.51Zeeektamseel: what does the CVLI say?
14:54.54ZeeekCLI
14:55.29*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:55.35*** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca)
14:55.55tamseelwell i am try to send the logs to u ok
14:56.16*** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-21a1c8377105a1da)
14:59.37*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-3bbab33fcf5aa475)
14:59.37*** mode/#asterisk [+o putnopvut] by ChanServ
15:00.08tamseelhttp://pastebin.com/m68ef4905
15:00.29tamseeljust check the url here all the logs are available
15:01.49[TK]D-Fenderzafar, yay
15:01.58[TK]D-Fendermultiple clueless people working ont he same project
15:02.07[TK]D-Fenderand with FREEPBX NO LESS
15:02.11[TK]D-Fender~freepbx
15:02.13jbot[~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
15:02.14[TK]D-Fender^^^^^^^^^
15:02.42zafar_yes dude
15:03.04zafar_we are looking all around but no help:'(
15:03.41[TK]D-Fender-- Executing [s@macro-dialout-trunk:20] Dial("Local/90505515897@from-internal-bc38,2", "ZAP/g0/0505515897|300|") in new stack
15:03.54[TK]D-Fenderzafar_: You're chopping the "9" off the front of your DIAL <-
15:04.27[TK]D-Fenderzafar_: And zafar_ is that appropriate?
15:04.45*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
15:04.49tamseelhellowwwwwwwww zeek where are u i am still waiting
15:05.24zafar_few line above this one it shows 9 as well
15:06.15zafar_<PROTECTED>
15:06.22*** join/#asterisk MindTheGap_ (n=MindTheG@201.80.82.57)
15:07.22[TK]D-Fenderzafar_: does ti loko right to dial this number? "0505515897"
15:07.27[TK]D-Fenderit look*
15:07.49tamseelwhere are you zeek
15:08.53*** part/#asterisk bt50 (n=user@host-69-95-44-9.spr.choiceone.net)
15:08.54keeblerHas anyone run asterisk on a !GB CF FLASH card?
15:08.58keebler1gb
15:09.09keeblerI should say, Asterisk+nix
15:09.40[TK]D-Fenderkeebler: People have gotten away with far less
15:09.53keeblerWell I know they've installed it on lInksys routers
15:10.10*** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman)
15:10.11keeblerBut iirc it was only able to handle a few calls?
15:10.16[TK]D-Fenderkeebler: So next question please :)
15:10.17keeblerat a time
15:10.50keeblerWell, my question really goes, how much space does asterisk need for logs?
15:11.09keeblerHow large does a typical log reach, say with 10,000 calls a month
15:11.34[TK]D-Fenderkeebler: What kind of logs?
15:11.52[TK]D-Fenderkeebler: What kind of hardware for the rest of the system?
15:11.58keeblerWell, I'd like to setup time stamped logs, duration of call, and what extention was called
15:12.06keeblerIntel ATOM mobo, 1GB ram
15:12.38*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)
15:12.38keeblerI'm trying to use some extra hardware I have lying around for an extra system.
15:12.45*** join/#asterisk telnettech (i=telnette@gw.percipia.com)
15:12.57keeblerRight now I've got the same mobo/ram/proc running with a 4GB CF Flash just fine
15:13.14keeblerBut not in the work environment yet.
15:13.33[TK]D-Fenderkeebler: Lok at your CDR file and you tell me...
15:13.47[TK]D-Fenderkeebler: size / lines
15:13.57keeblerHeh, its installing right now. haha
15:14.17keeblerI decided to go back to BSD+*
15:14.25keeblerTried Trixbox.
15:15.20keeblerI've probably reformatted this system 6 or 7 times this last month trying different options.
15:15.49*** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net)
15:18.58Nuggetanyone have any experience with chanspy or extenspy?  I think I'm doing it right but it's griping about "File spy-dahdi does not exist in any format"
15:19.38telnettechanybody know what the default IP is for the GXP2000 Grandstream.....We dont do DHCP and I need to know what IP the phone will use if it doesnt get 1 from a DHCP server?
15:19.54SparFuxtelnettech: no, but you might nmap it!
15:19.54Nuggettelnet is eeeeeeevil!
15:20.04*** join/#asterisk remont (n=reMont@200.73.192.226)
15:20.11*** join/#asterisk DarkRift (n=dark@65.92.166.119)
15:20.35*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
15:22.59Nasraany1 know any service provider in the Dom.Republic?
15:23.36macros73telnettech: http://www.voiptalk.org/products/gxp_setup.html
15:23.44Zeeekso yeah... #voip-users-conference is and in 90 minutes the Conference begins
15:25.30randulo_is_zeeekas I was saying before I was called into service
15:26.07randulo_is_zeeekhttp://tr.im/joinvoip will give you the info to call and join us. We're talking to Randal Schwartz today about various Open Source and asterisk and all
15:26.46telnettechmacros: Thanks but that is just the instructions on how to configure the phone.....i need to know what the default IP that the phone uses when it boots up if there is no DCP server...I have a idiot for a maint man who cant even use the freaking buttons on the phone to put an IP in it
15:26.56telnettechI need to get into the web gui to program it
15:27.14telnettechI dont want to send a tech to the site cause it is like $150 per hour
15:28.02[TK]D-Fendertelnettech: I've never head of a phone that has a default IP and don't fall to DHCP before.
15:28.26[TK]D-Fendertelnettech: And if the guy is too dumb to push what you say button for button, then fire him
15:28.37keeblerwho was it hear that said they used this http://www.mini-box.com/Mini-Box-M200-LCD ? And what sucked about it?
15:28.40keeblerhere
15:29.00*** join/#asterisk [Jasper] (i=Jasper@s559340af.adsl.wanadoo.nl)
15:29.03[Jasper]hej guys, I'm getting
15:29.04[Jasper]chan_sip.c:5593 process_sdp: No compatible codecs, not accepting this offer!
15:29.06[Jasper]how can I solve this
15:29.20telnettechTK: We dont use DHCP to assign ip addresses. The phone is set to default to DHCP. So you are saying that if it is set to DHCP and doesnt get an IP address that there is no default?
15:29.23Zeeek_is_randulowhat are allowed and what do you have?
15:30.32[Jasper]anyone?
15:30.34[Jasper]got a clue?
15:32.26Zeeek_is_randuloWHat codecs do you allow?
15:32.32[Jasper]where can I ese that?
15:32.50Zeeek_is_randuloin the sip.conf file
15:33.26Zeeek_is_randuloif you don't know what that is or where it is, you need to ask [TK]D-fender about it
15:33.27macros73telnettech: Those instructions tell you how to find the IP from the phone itself
15:33.40*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
15:33.46macros73telnettech: If he can at least read he can tell you the IP it got :D
15:33.52Dr-Linux|homehhm..
15:34.02[TK]D-Fendertelnettech: Its a PHONE. a STUPID phone.  Don't expect it to be a router.
15:34.24Dr-Linux|homewhat is the best loadblancer hardware for SIP that we can use with Asterisk?
15:34.51[Jasper]what header in sip.conf Zeeek_is_randulo ?
15:35.10*** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
15:35.17[Jasper]dtmfmode=rfc2833
15:35.22[Jasper]that's all that is in there
15:35.23Zeeek_is_randuloJasper sip.conf is where the codec info will be
15:36.25*** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net)
15:36.59*** join/#asterisk abatista (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
15:37.32[Jasper]no codec info in there Zeeek_is_randulo
15:39.27keeblerOkay, more questions related to infrastructure... Anyone use meraki mesh networks?
15:40.28Zeeek_is_randuloJAsper you need to find a basic asterisk text and read it
15:41.08*** join/#asterisk mv2 (n=maverick@83.240.229.38)
15:41.11Zeeek_is_randulo[Jasper] like this one: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
15:41.34*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
15:41.34*** mode/#asterisk [+o russellb] by ChanServ
15:41.52mv2how can I obtain all the logs from the Display panel in twinkle
15:41.54mv2?
15:44.22*** part/#asterisk awk_r (n=awk_r@nat/digium/x-dc36a65285965f4f)
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15:52.42[Jasper]Zeeek_is_randulo it's allready working
15:52.47*** join/#asterisk manxpower (n=Administ@router.asteriasgi.com)
15:52.50[Jasper]now I'm getting extension not found...but I added a dial-out
15:53.06[Jasper]with _X0.
15:53.16[Jasper]so that should pickup all the numbers starting with 0 right?
15:53.21Zeeek_is_randulo[Jasper] I am turning your case over to one of our more capable agents: manxpower
15:53.23[Jasper]why does it say extension not found that?
15:53.38[Jasper]?
15:53.46[Jasper]my questions not difficult enough?
15:54.06Zeeek_is_randuloI think they indicate some reading of existing docs
15:54.17[TK]D-Fender[Jasper]: because you don't have a match in the CONTEXT its looking in
15:54.21*** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca)
15:54.22Zeeek_is_randuloplus, if you get shirty people will not help you at all
15:54.55[TK]D-Fender[10:53]<[Jasper]>so that should pickup all the numbers starting with 0 right? <- and, no.
15:55.08[TK]D-Fender[Jasper]: You should read Chapter 5 again..
15:55.25Zeeek_is_randuloI leave you in good Hans
15:55.26DarkRift_X0. means start with any 0-9 digit then followed by a 0 + any number of digits after that
15:55.41[TK]D-FenderdarkClose
15:55.47[TK]D-FenderDarkRift: Close
15:56.01DarkRift1-9 ?
15:56.23DarkRiftI'll have a look as well
15:56.25[TK]D-FenderDarkRift: keep trying :)
15:56.27DarkRiftNah Z is for 1-9
15:57.33*** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar)
15:57.40*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-3567f6d96fc1fefe)
15:57.40*** mode/#asterisk [+o Deeewayne] by ChanServ
15:58.09*** join/#asterisk Subdolus (n=subby@subby.afraid.org)
15:59.31[Jasper]jees
16:03.04DarkRiftTrue close, but the last part is 1 or more characters, not "any number" (which can mean 0 as well)
16:03.48[TK]D-FenderDarkRift: There you go :)
16:04.22[TK]D-FenderDarkRift: And thats "1 or more CHARACTERS, not necessarily DIGITS
16:04.28DarkRiftyeah
16:04.37DarkRiftNoticed as well, see the correction in my last explanation :P
16:05.00[TK]D-FenderWe should completely ditch dialplan pattern logic for pur REGEX
16:05.07DarkRifttrue
16:05.15*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)
16:05.20DarkRiftREGEX is awesome imo
16:05.26*** join/#asterisk garo (n=garo-irc@d54C3F35B.access.telenet.be)
16:05.37zafar_can any one tell me the difference btw zaptel and dahdi
16:05.39DarkRiftuse more processing tho
16:06.22*** join/#asterisk _gm (n=gmustafa@202.133.78.60)
16:06.51DarkRiftdahdi is the digium version of zaptel drivers since the owner/creator of zaptel wanted it that way (to put it simply)
16:07.39virtualme123When I do sip show channels there is a column for last message, is there somewhere I can look to find out what each term means?
16:07.49[TK]D-FenderDarkRift: Umm.. no
16:08.30kaldemarDarkRift: http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/
16:08.34[TK]D-FenderDAHDI is the new NAME for the Zaptel drivers (always made by Digium), because the NAME was trademarked by another company
16:08.34kaldemarzafar_: ^
16:08.50*** join/#asterisk bmoraca (n=bmoraca@209.60.253.58)
16:08.55DarkRiftah
16:09.23zafar_ooah
16:10.04zafar_where can i get good help on these
16:11.49DarkRiftI Thought the zaptel driver was made by someone else, my bad
16:12.08[TK]D-FenderDarkRift: Silly wabbit
16:13.05*** join/#asterisk usam (i=nyrk@silenceisdefeat.com)
16:13.33*** part/#asterisk usam (i=nyrk@silenceisdefeat.com)
16:14.02[Jasper]so [TK]D-Fender you mean page : http://openfacts.berlios.de/index-en.phtml?title=DeStarInstall/BudgetPhone
16:14.06[Jasper]is wrong concerning their settings
16:14.24virtualme123There seem to be Rx: messages in that last message column and wondered what that referred too.
16:14.26[Jasper]their dial-out pattern
16:17.15*** join/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net)
16:17.59virtualme123Jasper: Who me ?
16:18.26*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
16:19.46*** join/#asterisk neurosys (n=vinix@173.9.159.180)
16:20.16*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
16:20.27[Jasper]no that's meant towards [TK]D-Fender
16:20.32[Jasper]and Zeeek
16:21.10virtualme123Jasper: sorry ...
16:22.07*** join/#asterisk inam (n=inam@116.71.222.76)
16:22.13[TK]D-Fender[Jasper]: Where do they say what that pattern represents?
16:23.18[TK]D-Fender[Jasper]: they suggest a pattern with no explanationg of why or how it will be processed and we're supposed to take it at face value?
16:23.19*** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr)
16:27.13eliaspdoes anyone know a free (OpenSource if possible) SIP/IAX client which supports "attended transfer"?
16:30.58path_what's the application that allows to execute the 'do not disturb' mode ?
16:30.58*** join/#asterisk flashnet (n=flashnet@200-127-213-146.cab.prima.net.ar)
16:31.57[TK]D-Fenderpath_: what "mode"?
16:32.08[TK]D-Fenderpath_: This is not a state, its a reaction
16:32.10De_Moneliasp counterpath has a free sip phone. it's not open source tho
16:32.16path_duh
16:32.17De_Moneliasp try voip-info
16:32.24path_Do-not-disturb (DND), found it [TK]D-Fender
16:32.37path_but it seems more a SIP-side feature than a PBX feature
16:32.48path_I meant softphone/hardphone
16:32.52[TK]D-Fenderpath_: Correct
16:32.55*** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn)
16:32.58eliaspDe_Mon: yes, i'm already crawling through voip-info.org lists all the time... tried now ~25 different clients, none of them had attended transfer yet... i'll take a look at counterpath, thx
16:33.14[TK]D-Fenderpath_: It can be done server-side in DIALPLAN of course
16:33.24[Jasper][TK]D-Fender actually before I did a reinstall that worked
16:33.29[Jasper]exten=_0X
16:33.36[Jasper]that waas selected for numbers beginnen with 0
16:33.38[Jasper]I don't know why
16:33.46manxpower_0X would me Zero followed by one of any digit
16:34.04manxpowerIn other words it would match 00 - 09
16:34.09[TK]D-Fender[10:53]<[Jasper]>with _X0.
16:34.11[TK]D-Fender[10:53]<[Jasper]>so that should pickup all the numbers starting with 0 right?
16:34.17*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
16:34.23[Jasper]is that incorrect?
16:34.24[TK]D-Fender[Jasper]: Please notice that you wrote it BACKWARDS before <-
16:34.30[Jasper]....
16:34.31[Jasper]my mistake
16:34.36[Jasper]still this doesn't work..
16:35.03[TK]D-Fender[Jasper]: And you aren't showing us the failed call with SIP debug, nor your complete dialplan context
16:35.15[TK]D-Fender~pb
16:35.16jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste, or , http://bin.cakephp.org/
16:35.17[TK]D-Fender^^^^^^^^^^^^^^^^
16:37.19[Jasper]hmm [TK]D-Fender
16:37.24[Jasper]you mean running asterisk with -vvvc ?
16:38.38path_[TK]D-Fender, ok thanks :)
16:40.19*** join/#asterisk _gm (n=gmustafa@202.133.78.60)
16:40.37bmoraca[Jasper]:  record the output of a call after issuing the "sip set debug peer XXXX" where XXXX is your SIP extension you're trying to call from.  Then pastebin what you recorded.  Use putty or something and enable logging.
16:43.06[Jasper]sip set debug peer _0X.
16:43.06[Jasper]No such peer '_0X.'
16:43.09[Jasper]hmm
16:43.13bmoracathat's not your peer
16:43.33bmoracathat's the extension you're dialing to, not the extension FROM WHICH you are dialing
16:43.53[Jasper]oh k
16:43.58*** join/#asterisk Mark17 (n=mark@freenode/sponsor/mark17)
16:44.23[Jasper]that IS the extension I'm dialing to
16:44.29[Jasper]I'm dialing a mobile number
16:44.31[Jasper]or at least tryin
16:44.52bmoracaright, but the sip set debug peer command needs to know what extension you're dialing FROM
16:44.53Mark17hello, is there somewhere an bluetooth dongle chipset reference available? the one available on http://www.voip-info.org/wiki/view/chan_mobile isnt available anymore (as mentioned at that page)
16:45.03[Jasper][Jan 23 17:44:35] NOTICE[2730]: chan_sip.c:14489 handle_request_invite: Call from 'verberk' to extension '06182XXXXX' rejected because extension not found.
16:45.40Mark17does extension 06182xxxx exist?
16:45.41bmoraca[Jasper]: ok, now pastebin your extensions.conf and your sip.conf
16:45.46Mark17what asterisk version do you use?
16:45.51*** join/#asterisk flashnet (n=flashnet@200-127-213-146.cab.prima.net.ar)
16:45.52*** join/#asterisk CunningPike (n=arodgers@204.239.10.119)
16:46.05[TK]D-Fenderbmoraca: Brace for regret...
16:46.13bmoracahaha
16:46.17[Jasper]k
16:46.19bmoracai can try to do my good deed for the day...
16:46.46[TK]D-Fenderbmoraca: "No good deed goes unpunished"
16:46.55bmoracawell, that's true
16:47.15bmoracabut i've only got about 5 minutes...so the punishment won't be very bad
16:48.43*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
16:48.47[TK]D-FenderI'm out the door for lunch... back later
16:48.50Mark17[Jasper]: what asterisk version do you use? 1.2.x or 1.4.x or 1.6.x?
16:49.12[Jasper]bmoraca <- sip.conf : http://pastebin.com/m44f23e84
16:49.54bmoracaok, that's a start
16:50.24[Jasper]bmoraca <- extensions.conf : http://pastebin.com/m4ef19111
16:50.34[Jasper]1.4 Mark17
16:51.02*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
16:51.08Mark17[Jasper]: is that your total config?
16:51.17*** part/#asterisk ibm2 (n=Administ@196.203.44.3)
16:51.22Mark17i am missing some information for as far as i can see
16:51.22*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
16:51.25bmoraca[Jasper]:  your problem is that the extension really doesn't exist.
16:51.39Mark17and there is no option for calling out ;)
16:52.01bmoraca[Jasper]:  either that, or you didn't give me everything I need
16:52.36Mark17i am missing some parts from extension.conf (for calling out at your config)
16:52.39*** join/#asterisk errr (n=mike@fedora/errr)
16:52.49bmoraca[Jasper]:  in what you pasted, there is nothing that will match 06182xxxx
16:52.51Mark17for as far as i can see
16:53.08[Jasper]I want a match for _0X.
16:53.29[Jasper]exten=_X,1,
16:53.31[Jasper]what';s that then?
16:53.56*** join/#asterisk demiv (n=demiv@190.158.82.133)
16:54.47bmoraca[Jasper]:  is there anyway you could post your entire extensions.conf without modifying it?  your arbitrary numbers of X's used for masking are making it rather difficult to tell what's goingon
16:55.32bmoracagotta go
16:55.35bmoracabye!
16:57.56Mark17[Jasper]: could you give the total extencion.conf (only modifying paswords and replacing them by {pass} or something like that)?
16:58.05Mark17so i can look for your problem
16:58.26Mark17also sending it by /msg or /q is no problem (also dutch is no problem)
16:59.33Mark17and dont use XXX for hidding nummbers/other info (use HHH in stead, because XXX is used at asterisk)
17:00.47*** join/#asterisk telecos (n=sergio@107.167.219.87.dynamic.jazztel.es)
17:05.52*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
17:09.08*** join/#asterisk telnettech (i=telnette@gw.percipia.com)
17:10.24manxpower[Jasper]: Try reading some docs
17:14.43*** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com)
17:22.07orTix[Jasper]
17:22.09orTix~book
17:22.10jbotwell, book is probably 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://tfot.leifmadsen.com or see ~buybook
17:22.18orTixi used that to, for my dialplan & extensiosn
17:23.11manxpowerYou could also try the Super Secret Docs (SSD) located in /path/to/src/asterisk/doc but don't tell anyone about it.
17:23.47manxpowerthere's another place to  /path/to/src/asterisk/configs with the Super Secret Config Examples (SSCE)
17:30.30[Jasper]manxpower could you be mor funny?
17:30.35[Jasper]or is this your best try?
17:32.17*** join/#asterisk kisu_ (n=kisu@2001:5c0:1100:9900:c135:858f:6466:f6b4)
17:38.33hardwiresigh.
17:38.43hardwireanybody got the goods for a bunch of professional wave ringtones?
17:39.30demivhello, is possible  to disable callwaiting generally  for all sip users ?
17:39.51hardwireuse groups in the dialplan, gives you a lot of flexibility.
17:40.01hardwirevs disabling call-waiting on the phones itself.
17:40.10hardwireor through the calllimit in sip.conf
17:40.19hardwiregroups are teh bomb
17:41.38inamhello
17:41.41demivif I set call-limit=1 then I do not transfer the call
17:42.14demivI cant transfer a call
17:42.46hardwiredemiv: hi
17:42.48hardwireuse groups.
17:43.02*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:43.11rob0I have a dialplan which was kludged along since pre-1.0 days, and it seems in 1.6 the use of "|" as parameter separator is broken, at least for Goto(). Is that true for other commands as well?
17:43.11demivI will try with groups
17:43.15*** join/#asterisk BipBip (n=BipBip@194.65.5.235)
17:43.51demivhardwire: Im trying with this command: database put CW TC4656 DISABLED
17:43.56demivbut it fail.
17:44.39manxpowerrob0: Go read the upgrade files
17:44.57manxpowerAll of them
17:45.35NoxIn-<PROTECTED>
17:47.00demivNoxIn-: No, no error message. Just dont work. The call waiting is stil working.
17:47.16manxpoweryou should disable call waiting IN THE PHOINE
17:47.37demivmanxpower: I use the famous EyeBeam
17:47.55manxpowerdemiv: Welcome to the world of softphones.  It sucks to be you.
17:48.21demivheh
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17:49.15*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
17:49.50FruitBasketI'm running Asterisk 1.4.22. We just had a power outtage and one of our routers ate itself. We swapped all our phones to a different router with a different IP address, and _NONE_ of the phones will register. The fix is to unplug EVERY PHONE for 5 minutes and asterisk will "reset" some sort of data.. what the FUCK?
17:50.36manxpowerasterisk does not "reset some data" when the phone is unplugged
17:50.49manxpowerAsterisk won't even know the phone is unplugged, just that it is unreachable
17:52.23*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
17:52.50FruitBasketthe phones reregister once every 3 minutes. I have three different brands, and _none_ of them register until they've been unplugged for 5 minutes.
17:53.15FruitBasketThere's something with ASterisk. There's nothing else it can be. Any time a phone changes IP's and immediately tries to reregister
17:55.21FruitBasketI turn on sip debug... not a damn message comes out.
17:55.25FruitBasketI tcpdump.. I see sip traffic.
17:56.07*** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de)
18:03.30watchyanyway to change the volume of a sip device from *?
18:04.39*** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com)
18:05.05watchyi got 3 mediatrix units in place but apparently they think the volume is to high
18:07.57*** join/#asterisk tamseel (n=IceChat7@116.71.222.76)
18:08.04tamseelhi all
18:08.13tamseeli have a problem
18:08.33tamseelhow to disable a zaptel channel
18:08.50watchyfrom *?
18:08.57tamseelasterisk
18:08.59watchyuse zap destroy i think
18:09.06watchyit'll remove it from *
18:09.37tamseeli want to disable not to revmove it
18:09.45watchyi don't think you can
18:09.57watchyjust reload zaptel when you want it back
18:11.08tamseelis there any other option wit zap destroy
18:11.24tamseeli mean i want to disable a specific channel not to all
18:11.32watchythats what it does
18:11.37watchyit gets rid of a certain channel
18:11.40watchynot all of it
18:11.55NoxIn-tamseel: like you have an E1 of 30 channels but you wan't the chanel 18 to be never used ?
18:12.14watchyremove it from zapata
18:12.26watchyor if you just wanna do it temp destroy it
18:12.28tamseelyes
18:12.38NoxIn-then remove it in zapata.conf and reload
18:12.38tamseelnoxin
18:13.15NoxIn-but if it's used in incoming... you may still receive calls on it
18:13.34*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
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18:14.41tamseelwell i should just remove that specific channel
18:14.51tamseelfrom zapata.conf
18:14.59NoxIn-yes
18:15.25NoxIn-but if the E1 comes from a telco you shoud also tel it to the telco so the don't send calls to you on this channel
18:16.25tamseel<PROTECTED>
18:18.07tamseeli have 1-8 channels
18:18.18tamseeland i want not to use 1,7 and 8
18:18.52tamseelwhat should be the configuration now
18:19.05tamseeleither it should be just like this 2-6 now
18:19.16tamseelor either some thing different
18:19.40NoxIn-<PROTECTED>
18:20.01tamseelok i am just trying it
18:21.49NoxIn-you can even have something like channel =>4-15,17-19,25-31
18:22.08NoxIn-(not very usefull usually)
18:32.07*** join/#asterisk tmjb (n=tmjb@212.200.239.230)
18:36.48tmjbHello all I am trying to sepereate b chanels my dsp isdn  card with misdn.conf but I dont know what to enter in for msn= should this be my two number or some thing else.Tnx
18:37.05*** join/#asterisk beek (n=klinebl@65.211.106.242)
18:37.39manxpowerNoxIn-: That format is used quite often in the real world
18:38.31watchyhmm
18:39.02NoxIn-manxpower: the format yes, but cutting an E1 in slices like I did I don't think so
18:39.18NoxIn-or maybe if you have a very strange config
18:39.45manxpowermaybe not E-1s, but it's common with T-1s, especially with mixed PRI/DATA/FXOFXS
18:39.57NoxIn-oki
18:39.58NoxIn-ok
18:43.08tamseelwell noxin i have do as u told but still the channel 1 hang up the call when i call on that channel
18:44.51*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:44.52tamseelthe problem is same before the current configuration
18:45.19tamseelstill same situation with channel 1
18:45.29[Jasper]hej guys, I have a question when I dial via a certaintrunk which I have told to redirct to a phone
18:45.37[Jasper]for some reason [default] get called
18:46.48tamseelthe configuration is as follows setting in zapata.conf =>2-6 and reload all the zaptel services
18:46.51*** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178)
18:47.04tamseellike zaptel reload , service zaptel restqart etc
18:47.23[Jasper][default] get called instead of the [budgetphone]
18:47.37ruben23hi
18:48.49tamseelif still something is missing plz notify that
18:49.35ruben23what si the difference between dtmfmode=inband with dtmfmode=rfc2833....
18:49.48ruben23what is the difference between dtmfmode=inband with dtmfmode=rfc2833....
18:50.08*** join/#asterisk Shaun2222 (n=shaun@ip68-5-154-128.oc.oc.cox.net)
18:50.18Shaun2222is there a wireless head set for the polycom 550's?
18:50.55*** join/#asterisk SparFux (n=raoul@f050021092.adsl.alicedsl.de)
18:51.22SparFuxCan I connect a device which has a DSL modem included to some ethernet device instead of dsl?
18:51.40orkidanyone havfe experience with sangoma's U100? (usb2fxo)
18:51.42[Jasper]how can I tell a incoming call to go to a certain context?
18:51.46[Jasper]like [incoming]
18:51.52[Jasper]how can I tell a trunk to go there?
18:51.54tamseelyah u can connect that with ethernet
18:52.19*** join/#asterisk citywok (n=chatzill@corpnet.csgopenline.com)
18:52.29citywokdoes anybody have a trick for making fax + sip work?
18:53.02tamseelyou must specify trunk for a specific context
18:54.14[Jasper]how?
18:54.22[Jasper]by giving it the same name?
18:54.40tamseeland then u specify an outbound route to go a specific extension
18:54.56FruitBasketfax + sip = no
18:54.57[Jasper]tamseel how?
18:55.00tamseelyah
18:55.15tamseelline the context name should be same
18:55.19tamseelon both side
18:56.03[Jasper]hmm
18:56.11[Jasper]weird, why does default get called then?
18:57.51tamseelbecause they are configured already and most of the people use the default context i.e from-internal
18:58.15*** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary)
18:59.25citywokoh, i had GSM turned on, GSM compression not good for faxing, lets try it without :P
19:00.24orkidfor sure!
19:00.28orkidtry 711u if u can
19:02.14*** join/#asterisk TrentCreek (n=kvirc@adsl-70-254-120-235.dsl.hrlntx.swbell.net)
19:02.35SparFux711u can transport faxes?
19:02.45[Jasper]hmm I have a trunk called [budgetphone]
19:03.13[Jasper]why does this go to [default] instead of [budgetphone] context
19:03.15SparFuxunfortunately the asterisk-app-fax package in debian is broken since ages. and I think they just want to wait for asterisk 1.6
19:03.42*** join/#asterisk flashnet (n=flashnet@200-127-213-146.cab.prima.net.ar)
19:04.04citywokwell, my main PSTN fax line is not working, i keep getting com errors on two fax machines, so i figured i'd try out my SIP, also errors out, using the default ulaw
19:04.38tamseelwell i can't tell u untill i see ur configuration
19:05.29tamseelbecause there might be some problem that's why..
19:05.35*** join/#asterisk peppux (n=peppux@151.67.64.150)
19:05.37peppuxhello
19:05.41citywokfax detect was turned off in zapata.conf
19:05.49peppuxsomeone can help me please?
19:06.07tamseelreload ur zaptel configuration
19:06.20[Jasper]tamseel : http://pastebin.com/m44f23e84 <- sip.conf      and    extensions.conf : http://pastebin.com/m4ef19111
19:06.23citywokjust module reloaded it, gonna go test it out again
19:06.57[Jasper]for some reason my trunk of budgetphone has context in-budgetphone ... but it goes to [default] in extension instead of [in-budgetphone]
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19:16.41manxpower~ask
19:16.42jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
19:18.07orkidplease suggest an inexpensive (<$100) fxo device that supports sending hookflash to the pstn :)
19:18.21*** join/#asterisk datagen24 (n=craig@c-75-68-48-81.hsd1.nh.comcast.net)
19:18.26[TK]D-Fenderorkid: X100P
19:18.43orkidthat's fxs isn't it?
19:19.02[TK]D-Fenderorkid: Do you really want to ask me questions twice?
19:19.35orkidyou're implying that i asked you something twice, i did not. i simply asked for confirmation
19:19.39[Jasper]I can't find anything about this in the manual
19:19.43ruben23hi if i used inbound and outbound calls for my asterisk.....what would be my dtmfmode=?
19:20.00[TK]D-Fenderorkid: No.  I just lied to you.  It couldn't be that I meant my answer.
19:20.02datagen24question regarding install: rpm install via yum errors out needing libtds.so.5, installed latest freetds same error is their a particular package it is looking for?
19:20.17orkidwhat should i look for to not get burned by a bad quality x100p?
19:20.32[TK]D-Fender[Jasper]: and I do not see you pastebinning the SIP debug of your failed call.
19:20.46[TK]D-Fenderorkid: Roll of the dice.  Every time.
19:21.17[TK]D-Fenderruben23: Whatever your provider uses
19:22.44[Jasper]I have no clue how to make a debug [TK]D-Fender
19:23.13[TK]D-Fender[Jasper]: Go to * CLI and "sip set debug"
19:23.36Kobazso it looks like originate doesn't use the callerid from the channel driver configuration when establishing a call
19:24.02Kobazor is there a way to get it to do that.. without having to always specify the callerid (if i want it to just be the callerid that's defined already)
19:24.54tamseeli think your threre is nothing wrong in that but i thnk you should be carefull with pbx
19:25.17Kobaztalking to me?
19:25.43[TK]D-FenderKobaz: Defined where?
19:25.50Kobazdefined in sip.conf/iax.conf
19:25.57Kobazcallerid=
19:26.11orkidwhat in these X100P varies that makes one good and another bad? or is that unknown
19:26.18ruben23[TK]D-Fender: so ill my voip provider what they support...thanks
19:27.06[TK]D-FenderKobaz: it can't pull the CID from your SIP device because you could be passing ANYTHING to the channel.
19:27.44[TK]D-FenderKobaz: You need a CID set because the FIRST channel is CALLED by the call-file and upon answer will then connect tot he dialplan.  It was not really started by the device
19:28.04[TK]D-Fenderorkid: Lower quality parts
19:28.16Kobaz[TK]D-Fender: yeah, but it will use the channel driver to place the call, it could grab the callerid from the config
19:28.19*** join/#asterisk Micc (n=dotirc@166.129.107.108)
19:28.19Kobazcouldn't it?
19:28.53[TK]D-FenderKobaz: May be possible, but what if you point to a local channel?  No CID then.
19:28.55MiccI'm onsite at a customer location installing an spa8000, but I'm not sure how to identify the lines in the punch down block.
19:29.00Kobaztrue
19:29.09[TK]D-FenderKobaz: And besides it has the CID you pass so that the decive you call firs t has somethng to see.
19:29.12MiccI'm got a test analog phone and a phone cord I spliced the end and I have a red and green wire.
19:29.19Kobazyeah
19:29.24MiccWhich wire is tip and which is ring?
19:29.58MiccI'm trying to touch the bare wires to the punchdown pins but I'm not getting any dialtone.
19:30.18[TK]D-FenderMicc: Does the phone work directly jacked?
19:30.21orkidare there any decent X100P cars out there, or are they really not suited for being used in a 'production' environment because of general quality issues (even with 'good' cards)
19:30.43MiccTKD-Fender, yeah it works directly jacked to the spa8000.
19:31.02[TK]D-Fenderorkid: You wanna be a cheapskate don't think you'll necessarily get a good one.
19:31.04MiccSo I'm looking on wiki and green is the stripped pin.
19:31.17Micctip
19:31.22[TK]D-Fenderorkid: And all this theory means nothing.  Usually the part that failes is CallerID BTW
19:31.44MiccI wonder if these are digital lines. They are going into a card in an NEC pbx system.
19:32.05[TK]D-FenderMicc: Trying to fry the SPA are you?
19:32.17[TK]D-FenderMicc: These are not questions you should be asking yourself right now...
19:33.31Miccno, I'm just using an analog phone and trying to patch into the punch down block.
19:34.01orkid[TK]D-Fender: tx for the info
19:36.16*** join/#asterisk propellerhead (n=yogurt2u@host42.200-45-219.telecom.net.ar)
19:37.19manxpoweranyone know the default root password for Trixbox?
19:37.54manxpoweramazing what a person will do for money.
19:38.11jjshoemanxpower ce or pro?
19:38.16Nuggetheh
19:38.44Nuggetisn't that our cue to beat manxpower over the head with a grandstream phone and chase him off to #trixbox?  :)
19:39.02[TK]D-FenderNugget: Possibly...
19:39.08jjshoeNugget not when it takes less time to answer his question then shove a dildo in your ass, step on a soap box, and preach.
19:39.54manxpowerjjshoe: pro and I have enough good karma here I think I can burn a little of it.
19:41.18Nuggetoh stuff it, jjshoe.  you're hanging out on irc so it's obvious you don't care if your time is wasted.  and it's spelled "than", not "then"
19:41.19[Jasper][TK]D-Fender <--http://pastebin.com/m709a8aca
19:43.02[TK]D-Fender[Jasper]: Found no matching peer or user for '81.23.228.150:5060'
19:43.25[Jasper][TK]D-Fender I can't make a peer for everyone who dials now can I?
19:43.36[Jasper]I gave the trunk a context...
19:43.54[TK]D-Fender[Jasper]: Did I say PER NUMBER?  No.
19:43.56*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
19:45.34[Jasper]I thought incoming calls would go the the trunks specified context?
19:45.38[Jasper]not correct?
19:45.51[TK]D-Fender[Jasper]: it would if it MATCHED your peer
19:46.22[TK]D-Fender[Jasper]: http://www.budgetphone.nl/forum/viewtopic.php?t=46
19:47.24jjshoemanxpower log in as ip/ipo
19:47.26jjshoemanxpower log in as ip/ip
19:47.42jjshoeNugget see, soap box and dildo, much easier for me to just answer :)
19:47.57jjshoemanxpower from there you should be dropped into a perl script which lets you change the root password
19:48.05jjshoemanxpower that login only works at local console though
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19:53.22orkidpeople report "CallerID transfer" failure.. that's on an inbound call? why do they call it transfer?
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20:00.30MiccI got it.
20:00.32mnicholsonTrentCreek, so the script is publicly accessible?
20:00.34Micchappy customer.
20:00.57TrentCreeknot yet...but once I realse it I need to have precautions in place
20:01.25mnicholsonTrentCreek, well, it's not really an AMI problem if it's just some one executing a script
20:02.11TrentCreekmnicholson: okay, I was just seeing if there was a method within Asterisk I could implement
20:02.38mnicholsonTrentCreek, no, there is not really anything in asterisk that gives you that for free, you are going to have to handle that in the script
20:02.55TrentCreekmnicholson: okay. thanks a lot for the info
20:03.03mnicholsonTrentCreek, np
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20:57.36bleblebleare there any good trixbox / asterisk(paid) support options? I've heard the fonality support is really bad. We are moving our office pbx to asterisk / trixbox and would like to have someone on retainer or in case of emergency can call for some help
20:58.13*** join/#asterisk manxpower (n=Administ@24.214.206.254)
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21:09.22Kobazhmm
21:09.28SparFuxHi.
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21:13.25jjshoemanxpower that work for you?
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21:13.41manxpowerjjshoe: no.
21:13.42shido6ZzzZz
21:13.58manxpowerapparently you can't change the root password remotly and it is set on install.
21:14.11jjshoemanxpower right, you can't change it unless you already have it.
21:14.26shido6buffer oberflow might help
21:14.30jjshoemanxpower but having a user set one with local access is pretty easy.
21:14.34shido6yes a v goes there
21:14.35Qwellshido6: there are plenty of those
21:15.29shido6dip chicken fingers in ranch dressing and carefuly spread it over the server motherboard. Guaranteed to (up)root something.
21:15.32shido6or someone
21:16.12manxpowerjjshoe: How would you do that?  the only thing we have is the GUI password
21:16.36manxpowerthe original installer is no longer available.
21:16.36jjshoemanxpower hook up monitor/keyboard
21:16.44jjshoemanxpower login as ip/ip, as I said earlier.
21:16.51manxpowerjjshoe: Do you know where I can get a 400 mile long KVM cable?
21:17.09jjshoemanxpower yes 400milelongkvmcable.com
21:17.14manxpowerthought so.
21:17.21*** join/#asterisk mphill (n=mphill@204.14.193.163)
21:17.40jjshoealternatively you can have fonality do it for you, but you'll have to beg hard.
21:18.16jjshoethe few months I worked support at fonality before moving on to engineering I wouldn't change a root password remotely no matter how much someone begged
21:18.50manxpowerjjshoe: My goal is to do the min required and then work out my anger at being shanghaied into this boondoggle piece of shit software by a sales guy.
21:19.03jjshoemanxpower have support on it?
21:19.08manxpowerAnd also make the customer wait as much as I can.
21:19.24SparFuxIt would really be the best thing for me to have the pc beeper signal incoming calls. Is it possible to do so with asterisk?
21:19.25manxpowerjjshoe: I would have to ask the "customer"
21:19.33jjshoemanxpower what's the server id?
21:19.46shido6yes
21:19.49manxpowerlook.  I've gotten to the main page.  that is all.
21:20.01shido6why not give you callerid w/ beeping morse code
21:20.08shido6now thats so much cooler and annoying
21:20.14manxpowerI'll make the customer put a keyboard and monitor on it, boot into single user mode, and reset the password
21:20.26jjshoemanxpower it's simpler then that...
21:20.34jjshoeI've told you twice how to do it with a keyboard and monitor
21:20.37manxpowerbut right NOW I am looking for some clip art for my anti-trixbox sign for my desk.
21:20.40jjshoewith the server-id I might be able to get someone to do it for you
21:20.42jjshoebut *shrug*
21:20.44jjshoeyou're not listening
21:20.56manxpoweranyone know where I can get an 8.5x11 red circle with a line thru it.
21:21.10manxpowerjjshoe: and I don't WANT to get it done right now.  I want to make the customer wait a while.
21:21.13SparFuxshido6: I have severe problems with the permissions to the console.
21:21.45SparFuxshido6: I would have to run cw in suid root mode. And then the problem remains, that it will go on beeping, no matter wether the call has been picked up.
21:23.26jjshoeyeah perhaps #trixbox is the place for you.
21:23.54manxpowerI'll work on it on monday
21:24.06SparFuxHow can I run a command with system( <command> & ) in the background and kill it as soon as a call has been picked up?
21:26.05*** join/#asterisk iMacGyver (i=harmon@pico.dreamhost.com)
21:26.18shido6what?
21:26.45[TK]D-FenderSparFux: Launch it in an AGI and set a var with the PID.  Then do your dial with M() and call the System() to kill it
21:26.58shido6SparFux - dont u have a dev  or staging environment?
21:27.39SparFuxshido6: what's dev or staging env?
21:27.59shido6great nick by the way :)
21:28.36SparFux:-D
21:28.55[TK]D-Fenderheading home, BBIAB
21:30.10iMacGyverhow does asterisk determine 'no answer' when the phone rings? is there a way to increase the number of rings?
21:30.36mvanbaakiMacGyver: Dial(SIP/bla,45)
21:30.45mvanbaakthat will ring it for 45 second-
21:30.49mvanbaakseconds
21:31.11SparFuxFender: ah, yes, an agi it has to be.
21:31.45shido6have i been under a rock? has it changed to seconds from thousand milliseconds/
21:32.13shido6;)
21:32.14iMacGyvermvanbaak: ah thanks, using trixbox right now, is there a place in the web interface to change that for a ring group or ?
21:32.32mvanbaakiMacGyver: I have no idea
21:32.44mvanbaakiMacGyver: I use vim
21:32.48iMacGyvermvanbaak: doh just saw it
21:32.54iMacGyverit was right in front of my face
21:33.13mvanbaakgui's do that to you
21:33.16iMacGyverweird they say the max is 60 seconds
21:33.19iMacGyvertrue
21:37.16*** join/#asterisk D3b|4n (n=D3b_4n@unaffiliated/lynxnica)
21:40.34D3b|4ni need help
21:42.21manxpowerMaybe I need a job where I do stuff I enjoy most of the time rather than doing stuff I don't enjoy most of the time.
21:47.29iMacGyverso are there any free/cheap sip trunk providers?
21:50.16D3b|4ni have a problem
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22:00.01frogonwheels~ask
22:00.03jbotQuestions 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.
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22:01.41jjshoe~itsp-list
22:01.42SimonTekhello
22:01.49jjshoehrm
22:06.34*** join/#asterisk sack (n=sack@79.148.188.224)
22:06.44*** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn)
22:08.30*** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it)
22:10.51SimonTekis reading the trixbox and asterisks manuals before i ask a question
22:11.49[TK]D-Fender~trixbox
22:11.50jbothmm... trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
22:13.08*** join/#asterisk Dwev (n=dx@ppp118-208-135-42.lns10.mel4.internode.on.net)
22:13.19manxpowerI just can't get away from it.
22:13.52frogonwheels[TK]D-Fender: it's nice to have the jbot back :)
22:13.54Dwevhello
22:14.20Dwevis this channel friendly to people that don't know much about asterisk at all?
22:14.25SimonTeki know. but I also have an asterisks box. (test box) I use
22:15.00SimonTekI also have the trixbox channel open as well.
22:15.10mmlj4hey manxpower
22:15.46*** join/#asterisk martyn-dev (n=admin@190.24.134.154)
22:15.51martyn-devHi.
22:16.34martyn-devI'm compilling asterisk 1.2.14 .. so, when i try compile on centOS, i get this error: tds.h: No such file or directory
22:16.43martyn-devdo you know something about it ?
22:17.11SparFux1.2.14??
22:17.24SparFuxthat's old!
22:17.43beekWell, Teliax managed to fuck up DTMF detection again.
22:18.01nix8n82why do you say that beek?
22:18.28martyn-dev<PROTECTED>
22:18.42beekI use their NYC site and a client uses Denver.   Dial any number that requires DTMF (paging, IVR, etc) and have fun.
22:18.42mmlj4/me dances a little jig... he just got his first new asterisk client a few minutes ago
22:18.42martyn-devSparFux: so it's installed on my companny :( ..
22:18.44*** join/#asterisk Dwev (n=dx@ppp118-208-135-42.lns10.mel4.internode.on.net)
22:19.08*** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
22:19.11beeknix8n82: I just tried sending a page to my pager.   I keyed 1234567890, I got 111223344456667889
22:19.27mmlj4[root@samson asterisk]# rpm -qa | grep tds
22:19.27mmlj4docbook-dtds-1.0-30.1  # this is on centos, martyn-dev .... apparently used for documentation
22:19.33beeknix8n82: They had problems, seemed to have them fixed, and now we're back to this again.
22:19.54nix8n82oh that sucks
22:20.20beekAnd the client having the problem is a paging company.  They can't even send test messages to the pager.
22:20.39nix8n82how do you collect the dtmf? agi?
22:21.03*** join/#asterisk abatista (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
22:21.14beeknix8n82: The target in this case is a paging terminal.    I dial out through Teliax and the terminal receives the DTMF.
22:21.18mmlj4tone sniffers work well, iirc
22:21.29martyn-devmmlj4: rpm -qa | grep tds
22:21.29martyn-devfreetds-0.63-1.2.el4.rf
22:21.29martyn-devfreetds-devel-0.63-1.2.el4.rf
22:21.57nix8n82oh ok
22:21.58beekI have the same problem dialing from home into my office IVR (Asterisk).   I get all kind of crap in the IVR.
22:22.13*** join/#asterisk makkksimal (n=makkksim@91-66-249-217-dynip.superkabel.de)
22:22.34martyn-devlet me install docbook-dtds.noarch mmlj4 ...
22:23.27mmlj4that's not your problem, though... I have no idea what's happening for you, though
22:23.59frogonwheelsDwev: On a good day it's friendly.
22:24.13*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
22:24.14Dwevthanks frog
22:24.54martyn-devtds.h: No such file or directory :(
22:25.05frogonwheelsDwev: as long as you follow up on information given you wont get chewed up .. much..
22:25.16Dwevokey doke
22:25.18frogonwheels~book
22:25.19jboti heard book is probably 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://tfot.leifmadsen.com or see ~buybook
22:25.38Dwevyeah, i was looking at getting that
22:26.06Dwevgood for beginner?
22:26.08frogonwheelsDwev: read the pdfs - it at least gets you on track with what's what.
22:26.12frogonwheelsDwev: yep
22:26.39Dwevoh!
22:26.46Dwevi didn't realise it was a full pdf
22:26.53frogonwheelsyep -all there :)
22:26.56Dwevcool
22:26.57Dwev:)
22:27.05Dwevdoes it go through hardware?
22:27.09frogonwheelsDwev: you might find the hardcopy more comforting.
22:27.29bmoracai always prefer hardcopy myself
22:27.29Dwevpossibly still might get hardcopy
22:27.30frogonwheelsDwev: Hmm.. don't think so much.
22:27.38Dwevbut happy to look through the content first
22:29.08Dwevi'm trying to work out the minimum hardware required to run two voip/sip lines with a pstn fallback
22:29.39Dwevi'm thinking it's 2xATA, one with FXS port
22:30.07Dwevwhich i don't really need asterisk for, but i want to see if i can set up something that i can expand in the future
22:30.29frogonwheelsDwev: I have a home asterisk box running on my router - connected to a PAP2T
22:31.21Dwevsorry, pap2t?
22:31.23SimonTekI will say I am amazed how well asterisk works on a T1 line
22:31.53frogonwheelsDwev: google it. it's a linksys ata
22:31.53Dwevah ok
22:32.02*** join/#asterisk TheCompWiz (n=tmealey@wsip-68-109-200-102.mc.at.cox.net)
22:32.20TheCompWizwhat's the meaning behind exten => s,n(something),command
22:32.28TheCompWizwhat the heck is the (something) portion?
22:32.42*** part/#asterisk SimonTek (n=SimonTek@69.229.246.101)
22:32.50*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
22:32.54orkiddoes the motorola 62412-51 chip work as an x100p card?
22:33.05orkidthe modem suppoesdly says U26280.2-0.6
22:34.39Dwevfrog: is this a sipura in different clothes?
22:34.50Dwevor the next model up from a 3102?
22:35.00Dwevor something different again?
22:35.46TheCompWizsighs... anyone know anything about asterisk syntax?
22:36.09*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
22:36.10frogonwheelsTheCompWiz:  it's a label
22:36.30TheCompWizok.... for the most part does that mean it's only cosmetic? and non-functional?
22:36.42frogonwheelsTheCompWiz:  it means you can   Goto(something)  rather than Goto(3)
22:36.51TheCompWizah.
22:36.57TheCompWizthanks frogonwheels.
22:37.20frogonwheelsTheCompWiz: But in the case of where you are use 'n' instead of 2 3 4 .. then it's essential
22:37.27TheCompWizyep
22:37.31TheCompWiz1 last question...
22:38.00TheCompWizhow about exten s,1+998,blah .... does that mean the priority is 999? or something else?
22:38.28frogonwheelsshrugs
22:38.52frogonwheelsI doubt it.
22:39.33fogoI've determined that a sporadic crash of asterisk coincides with the following message in the log: "translate.c: No bits set? 0" Any ideas what may be causing this?
22:39.54*** join/#asterisk interfaith (n=chatzill@ip67-88-184-130.z184-88-67.customer.algx.net)
22:40.58Dwevok, i'm having difficulty understanding the difference between a linksys spa3102 and a pap2t
22:41.10interfaithhello iax users ? when iax call setup is done from iax1 to iax3 via iax2 will the iax go direct iax1 to iax3 ?
22:41.12Dwevis the latter just a 2 port version of the former?
22:41.29frogonwheelsDwev: oh.. talking to me? you should tabcomlete my full nick so I can see :)
22:41.33*** join/#asterisk TrentCreek (n=kvirc@ppp-70-247-66-31.dsl.hrlntx.swbell.net)
22:41.59Dwevsorry frog, yes
22:42.00frogonwheelsDwev: the pap2t is a redressed Sipura, yes.  I don't know exactly the versions or details.
22:44.19Dwevfrogonwheels: so it's possible to use the pap2t with a couple of voip accounts and a DID
22:44.54Dwevand have it work ok, then add an asterisk server later to get more advanced functionality?
22:45.01*** join/#asterisk reneger (n=reneger@p3EE2FE8B.dip.t-dialin.net)
22:45.08frogonwheelsDwev: yep
22:45.30*** join/#asterisk otherwiseguy (n=otherwis@72.128.38.41)
22:45.36frogonwheelsDwev: Ithink you should even be able to put some dialplan logic in to route certain calls out the landline
22:46.01frogonwheelsThat's how I started.
22:46.10Dwevcool
22:46.17Dwevthis is for a friend of mine
22:46.25Dwevi have a spa3000
22:46.25Dwevand have been playing with that
22:46.34Dwevbut he wanted two virtual lines
22:47.07Dwevi think the pap2t would be the best solution
22:48.50frogonwheelsDwev: I have 2 lines on my Pap2t.. I use my mobile as a SIP over WLAN client...
22:49.04frogonwheelsDwev: and then 2 more softphones connected to asterisk.
22:50.00Dwevso your mobile becomes an extension when it's within your wifi network?
22:50.15frogonwheelsDwev: yep :)
22:50.27frogonwheelsDwev: it sucks power a bit though.
22:50.39Dwevcool
22:50.41frogonwheelsDwev: but I use it to call out on a lot 'cause it has my phonebook
22:51.04Dwevso you have client software on the mobile obviously?
22:51.08Dwevis it a WM device?
22:51.11frogonwheelsDwev: it comes with it.
22:51.19Dwevoh?
22:51.28frogonwheelsDwev: Nokia Symbian based
22:51.37Dwevoh ok
22:51.47Dwevthere is still much for me to learn
22:51.50Dwev:)
22:51.58frogonwheelsso I just use the default SIP client
22:52.09Dwevah right
22:52.10*** join/#asterisk dan__t (n=dant@nagios.mojohost.com)
22:52.13frogonwheelsJust need to make sure I have reinvite=no
22:52.54*** part/#asterisk TheCompWiz (n=tmealey@wsip-68-109-200-102.mc.at.cox.net)
22:53.05Dwevso what would i need to begin playing with asterisk?
22:53.06Mark17hello, is there somewhere an bluetooth dongle chipset reference available? the one available on http://www.voip-info.org/wiki/view/chan_mobile isnt available anymore (as mentioned at that page), or is there an option so i can check it with my laptop (running Windows Vista)?
22:53.14QwellDwev: A computer.
22:53.16Dwevapart from a the asterisk server
22:53.25Dwevis a single ata enough?
22:53.36Qwelldon't even need that
22:53.56Dwevwell, i'd need some kind of hardware cards in the computer, no?
22:54.04Qwellno
22:54.16Dwev...
22:54.29QwellWhat do you want to DO with it?
22:54.37Dwevi guess that is the question
22:54.47Dwevi want to play with it in a functional environment
22:54.49*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)
22:55.06Dwevplay with ivrs and voicemail and voip routing etc
22:55.13Qwelldepends on your definition of functional..  got a PC with a soundcard?
22:55.22Dwevi don't have the pc yet
22:55.29Qwellnot the server
22:55.34Dwevi use a mac
22:55.47Dwevbut if that counts, then yes
22:55.49Qwelldoes it have a soundcard?
22:55.54Mark17can you connect a headphone (speaker + mic) to it?
22:55.58Dwevyes
22:56.03Qwellthen there you go - there's your phone
22:56.29Dwevsoftphone?
22:56.30theharwould you guys push sangoma or digium fxo/fxs cards? in your opinion. i use digium but just curious of opinions.
22:56.34Qwellyep
22:56.38Dwevah
22:56.52Qwellthehar: I can give you a biased opinion.  Digium cards all the way, of course
22:56.58Dwevso the mac sits on the network, and the * server sits on the network
22:57.03theharyeah i'm biased toward digium as well.
22:57.09thehari wanted 3rd party. haha
22:57.27Dwevand then the mac is like an extension on the asterisk system?
22:57.40QwellDwev: yes
22:57.44Dwevi see
22:57.47Mark17Dwev: the mac is just the phone for asterisk
22:57.52Dwevright
22:58.00Dwevso if i also have an ata
22:58.06Dwevthat can be another extension?
22:58.09Mark17yes
22:58.25Mark17it shouldnt be the same, just make an extra extension for it
22:58.49*** join/#asterisk BadHAL (n=wut@cpe-72-179-194-139.stx.res.rr.com)
22:58.55Dwevand if i have two voip accounts, i can set it up that i can make two calls simultaneously from each extension routed through the * server?
22:59.22Qwellif the "voip accounts" only let you have one simultaneous call, yes
22:59.32Dwevhmm
22:59.35Dwevinteresting
22:59.35Dwevok
22:59.42Dwevthat helps me understand
22:59.56Dwevnow, on a completely different topic
23:00.02Qwellpotatoes
23:00.05Dwevi've been told to find an Indian DID
23:00.23Dwevis there a marketplace somewhere that lists companies that do DIDs?
23:00.41drmessanoGoogle does
23:01.01DwevUK and US DIDs are easy, but i'm having trouble (even with google) finding an Indian one
23:01.01Mark17where is your server located and to where do you call (mainly)?
23:01.09Dwevserver in Australia
23:01.30Mark17and to where do you call mainly?
23:01.30Dwevbut would like a phone presence in India
23:01.38Dwevcall to australia and india
23:01.45Dwev(this is for a friend)
23:02.18drmessanoIf my operations in Tunis weren't suspended, I would see what I can do
23:02.39Mark17for that phone presence in india you should look for an DID that offers it (I know that it is for Belgium almost impossible to get a normal number)
23:02.43drmessanoBut sadly, it will be at least six months before Raolo gets out of prison.. If he's even still alive :(
23:03.02Dwevthat's what i was looking for
23:03.29Dwevjust the local phone number in india as DID
23:03.37drmessano"Look for a DID that offers it"
23:03.50drmessanoA DID _is_ the number
23:03.51Dwevyes
23:04.05Dwevso i look for a number that offers a number?
23:04.08Dwev?
23:04.14drmessanoApparently
23:04.51drmessanoWhat you're looking for is an ITSP
23:05.00drmessanoSomeone who can provide a DID
23:05.05drmessanoThey are not "a DID"
23:05.43drmessanoThey are no more a "DID Provider" than the phone company is a "Phone number company"
23:06.06Dwevok
23:06.19DwevITSP = Internet Telephony Service Provider?
23:06.22drmessanoYes
23:06.36Dwevand is there a directory of those somewhere?
23:06.52Dwevi'm scouring sipbroker.com at the moment
23:07.13Dwevbut it's hard to narrow down by country
23:14.03*** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat)
23:15.43watchyanyone here work with Mediatrix hardware?
23:16.37Dwevhmm
23:16.57Dwevit seems Indian DIDs are actually illegal or something
23:17.05Dwevthe Indian govt doesn't like them
23:17.18Dwevthat would explain some things
23:17.53watchyi need some help with audio on a mediatrix
23:29.30*** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
23:30.39seanbrightmartyn-dev: what version of asterisk?
23:30.46martyn-dev1.2.14 seanbright
23:30.56seanbrightugh
23:31.04seanbrightand what version of freetds?
23:31.09*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)
23:31.14martyn-devwhy can i see it seanbright ?
23:31.24seanbrightthat's a good question
23:31.38*** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-0-78.dsl.hrlntx.sbcglobal.net)
23:31.44martyn-dev:)
23:31.46martyn-devlet me see
23:31.51martyn-devsean
23:31.54martyn-devseanbright:
23:31.56martyn-devsee it
23:31.57martyn-devfreetds-0.63-1.2.el4.rf
23:32.01martyn-devrpm -qa | grep freetds
23:32.02martyn-devfreetds-0.63-1.2.el4.rf
23:32.02martyn-devfreetds-devel-0.63-1.2.el4.rf
23:32.13martyn-dev0.63-1.2 :D seanbright
23:32.13seanbrightok, hold on
23:32.18martyn-devok..  seanbright
23:33.09seanbrightyou should only see that warning with .62 or earlier
23:33.27seanbrightdo you need cdr_tds?
23:34.01*** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178)
23:34.20martyn-devseanbright: yeah
23:34.58seanbrightare you installing asterisk from source or from a package?
23:35.08martyn-devfrom source .tar.gz
23:35.21seanbrightok, pastebin your config.log file
23:35.30seanbrightor just post it somewhere
23:35.36seanbright(but not in the channel)
23:36.02martyn-devyea seanbright.. let me ..
23:36.39seanbrighterr
23:36.44seanbrightyou don't have a config.log
23:36.49seanbrightbecause you're running an old version
23:36.54seanbrightwhich doesn't have configure
23:37.11martyn-devi have tail -f editline/config.log seanbright
23:37.25seanbrightyeah, that won't help
23:37.35martyn-dev:(
23:37.48seanbrightlet me look at something
23:37.51*** join/#asterisk troubled (n=troubled@unaffiliated/troubled)
23:37.57martyn-devbut i've been installed this same .tar.gz on other centOS and its works :(
23:38.01martyn-devyeha. tell me seanbright
23:38.29seanbrightthat's strange
23:39.24seanbrightfind your tdsver.h file
23:39.34martyn-devlet me..
23:39.35seanbrightit's either somewhere in /usr/include or /usr/local/include
23:39.52martyn-dev./usr/include/freetds/tdsver.h
23:39.53martyn-dev./usr/include/tdsver.h
23:39.56seanbrightahhh
23:40.00martyn-dev?
23:40.01seanbrighthmmm
23:40.04seanbrightyou have two?
23:40.09seanbrightyou shouldn't have two.
23:40.43martyn-devyeah :( because when i try copmpile,.. the message that show is no such file or directory.. so i copy the file on /usr/include/ :S
23:40.48seanbrightah
23:40.51seanbrightdon't do that
23:40.54martyn-devoh!
23:40.56beekHas anyone installed 1.6.0.5 yet?  I just compiled it and it's reporting as 1.6.0.3 when I connect to it with 'asterisk -r'
23:41.42seanbrightmartyn-dev: grep TDS_VERSION_NO /usr/include/freetds/tdsver.h
23:42.08martyn-devgrep TDS_VERSION_NO /usr/include/freetds/tdsver.h
23:42.09martyn-dev#define TDS_VERSION_NO "freetds v0.63"
23:42.28martyn-devis corret version ..
23:42.39*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
23:43.00martyn-devif I try compile.. i get this error:
23:43.02martyn-devtds.h: No such file or directory
23:43.02martyn-devcdr_tds.c:68:24: tdsconvert.h: No such file or directory
23:43.35martyn-dev:(
23:43.44seanbrightmake CFLAGS=-I/usr/include/freetds
23:43.48seanbrightsee if that works
23:43.56martyn-devlet me
23:44.05beekseanbright: As a bug marshall you may want to note that the just-released 1.6.0.5 isn't at that level.
23:44.17beek./include/asterisk/version.h:#define ASTERISK_VERSION "1.6.0.3"
23:44.44martyn-devseanbright: so, asterisk.h: No such file or directory
23:44.44martyn-devcdr_tds.c:73: error: syntax error before string constant
23:44.52seanbrightmartyn-dev: ah.  doh.
23:44.55*** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat)
23:45.33jayteebug marshall? is that anything like cat herder?
23:45.43beekHi jaytee ...
23:45.47*** join/#asterisk Ineluctable (n=Inelucta@adsl-76-242-91-181.dsl.lsan03.sbcglobal.net)
23:45.47jayteehi beek
23:46.27Ineluctablehardware is only need if I want to use a lan phone correct?
23:46.31beek1.6.0.3 had a regression over 1.6.0.1 in the management interface.  I was excited to see 1.6.0.5 come out, but it reports as 1.6.0.3 and has the same bug.
23:46.48beekjaytee: so that is what I'm working on.
23:46.52martyn-devseanbright:  :(
23:46.53jayteeIneluctable, a lan phone?
23:47.05*** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net)
23:47.34icebrew54I am utilizing the monitor function in asterisk 1.4.18.1, and am getting scratchy sound quality to the wav files before and after joining them
23:47.40seanbrightmartyn-dev: still looking
23:47.51icebrew54can anyone point me in the right direction to troubleshoot such a problem?
23:47.52IneluctableI mean a analog of digital phone
23:48.13[TK]D-FenderIneluctable: even less understandable
23:48.26jayteelol
23:48.32beekEvening [TK]D-Fender
23:49.34seanbrightbeek: did you grab it from the tarball or from svn?
23:49.51Ineluctableok I have only the hardware that came with my computer.  do I need to buy anything to make asterisk work?
23:49.52beek<PROTECTED>
23:50.13*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
23:50.13*** mode/#asterisk [+o russellb] by ChanServ
23:50.19[TK]D-FenderIneluctable: No.  * is just a piece of software sitting on a computer
23:50.23beekseanbright: http://downloads.digium.com/pub/asterisk/asterisk-1.6.0.5.tar.gz
23:51.35seanbrightbeek: cat /path/to/asterisk/source/.version
23:51.41[TK]D-Fenderseriously... WTF happened to 1.6.0.4?
23:51.57seanbrightwe skipped it
23:52.05seanbrightcuz we're crrraaaaazzzzzy
23:52.06beek[asterisk@corp-asterisk asterisk-1.6.0.5]$ cat .version
23:52.06beek1.6.0.3
23:52.09Ineluctablewill I be able to have a phone number?
23:52.18beekseanbright: ^^^^^
23:52.24seanbrightbeek: the tarball i just downloaded has the correct value in .version
23:52.31Ineluctableor do I still have to suscribe to a voip service?
23:52.35jayteelet's just skip all the rest and go straight to 1.7.5, that ought to be stable and bug free!
23:52.42seanbrightbeek: you probably grabbed it just before it was fixed
23:52.44[TK]D-FenderIneluctable: * can connect to physical phone lines you have with special hardware., or to ITSP's
23:52.46[TK]D-Fender~itsp
23:52.47jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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
23:53.00*** join/#asterisk SparFux (n=raoul@e182030239.adsl.alicedsl.de)
23:53.08SparFuxHello!
23:53.25SparFuxDoes anybody successfully use iaxcomm of debian unstable (sid)?
23:53.42Ineluctable~itsplist-us
23:53.43jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
23:53.46beekseanbright: WTF!!  I'll download it again and see what happens.  Thanks
23:54.09seanbrightbeek: just check the .version file before doing anything :)
23:54.14seanbright(on the  new download)
23:54.24beekseanbright: will do.   Leave it to me to jump the gun.
23:54.37seanbrightbeek: i blame myself
23:55.20seanbrightmartyn-dev: can you paste the compile line that is failing?
23:55.29*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
23:55.35martyn-devseanbright: yeah. let me
23:55.46beekseanbright: Are there more changes to that download other than 0.3 -> 0.5?
23:55.55seanbrightbeek: not according to svn
23:56.19*** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back!
23:56.20beekCrap.  Then the bug still exists.  The fix was to be in 1.6.0.3-RC2, which never happened.
23:56.29seanbrightbeek: what bug?
23:56.35martyn-devseanbright: http://asterisk.pastebin.ca/1316467
23:56.54beekseanbright: When using the management interface to dial a call you'd get a "channel not specificed."
23:57.05seanbrightbeek: #?
23:57.24beekLooking now.l..
23:58.04beekThe message from Mark Michelson is:
23:58.11beekThanks for pointing this out. I have located the erroneous code and have fixed
23:58.11beek> it in subversion, revision 161490. The next rc of 1.6.0 will not have this bug.
23:58.26seanbrightwhat issue number is it?
23:58.33seanbrightdon't make me beg
23:59.08beekseanbright: I'll have to dig it up.  I didn't file the bug, only replied to the message from Mark saying it was fixed and letting him know that it's not.  Let me go to bugzilla and see if I can locate it.
23:59.18seanbrightoh, gotcha.
23:59.31IneluctableI am new to this so please bear with me.  If I have to pay for a voip account then this means I can just a sip program right?

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