IRC log for #asterisk on 20100623

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00:56.49booduI try to configure isdn trunk with no success. Call Entrances is ok but call out failed. I don't see any channels ISDN
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01:01.20jsidhuso I've got both of my boxes connected via IAX2 trunks. this works if both sides have static ips since they both register to each other. How can i handle a situation where one side has a dynamic ip?
01:01.50chuckfjsidhu: dyndns is one way
01:03.10jsidhuchuckf: there's no other way besides have the remote side register? since the dynamic side is already registering, cant asterisk use that same trunk?
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01:11.15jsidhu..
01:13.51pabelanger-lapjsidhu: have your dynamic IAX2 IP register with your static IAX2 IP
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01:19.34jsidhupabelanger: yeah i was doing something wrong
01:19.41jsidhustarted over and followed http://pbxinaflash.com/forum/showpost.php?p=24595&postcount=3
01:19.45jsidhuworks fo rme
01:27.21*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)
01:28.31sawgoodHow could this be ... I yum removed all the asterisk RPM packages I installed, I rebooted the server, I look and saw the /etc/asterisk folder was empty, but yet I can still launch and run Asterisk?
01:28.53*** part/#asterisk |Rain| (rain@ev.il.net)
01:29.21chuckfsounds like a binary didn't get removed
01:30.02sawgood<PROTECTED>
01:30.08sawgoodI am looking now to see if there is anything left
01:30.37chuckf'which asterisk'
01:31.30sawgood1.6.2.8
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01:31.47sawgoodI found two RPM packages still installed ... I was able to remove one, but not the other
01:31.52sawgoodI did a core stop now from the CLI
01:32.04sawgoodasterisk16-core-1.6.2.8-1_centos5
01:32.19sawgoodthis RPM package will not come off with yum remove asterisk16-core-1.6.2.8-1_centos5
01:32.41sawgoodthere is goes ... finally ... gone
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01:34.18sawgoodeven if you do a yum remove of all Asterisk packages, it still leaves the /etc/asterisk directory with several saved copies of your Asterisk .conf files
01:34.25sawgoodasterisk.conf.rpmsave
01:35.04chuckfleaving behind user configured (or potentially configured) files is a good option to default to
01:36.06sawgoodyeah .. you are right about that
01:36.52sawgoodThe thing is ... I never installed FreePBX on this box, but yet after an install of the Asterisk RPM files (then removing them to see what is left) ... I see a /var/spool/asterisk/freepbx type file left behind
01:37.08sawgoodmaybe a long time ago this box had FreePbX on it, and I am just overlooking it
01:40.48chuckfhave you tried a 'locate asterisk' search?
01:43.47sawgoodI did that ... I think maybe this box had AsteriskNOW 1.5 on it a long time ago
01:44.28sawgoodthat is the only thing I can think ... I'm not sure ... because I would have simply put CentOS 5.4 on the box and build Asterisk from scratch or RPMs .... and FreePBX would have never been on the box
01:44.49sawgoodI am going to take the box from the co-lo; wipe the drive; reinstall CentOS
01:46.44sawgoodcheck this out ... the second I fire up Asterisk (after a fresh RPM build and no config files) ... I am flooded with this message from the CLI
01:46.46sawgood[Jun 22 18:46:11] NOTICE[3013]: chan_sip.c:21625 handle_request_register: Registration from '"user" <sip:user@65.49.22.226>' failed for '71.5.70.79' - No matching peer found
01:46.57sawgood1000s of the same message scrolling up the screen ...
01:47.35sawgoodI do not even have anything set in sip.conf or extensions.conf like that yet (it is a fresh build of the software)
01:48.56sawgoodmaybe someone is trying to 'hack' the box?
01:49.11sawgoodI have no idea who this 71.5.70.79 is
01:49.37pabelanger-lapsawgood: Do you have a firewall enabled?
01:49.58pabelanger-lapsawgood: I would recommend one
01:50.46sawgoodI can start iptables I guess ... but I do not know the sytax to stop this for sure (I might be able to figure it out)
01:51.01sawgoodits been a while since I wrote an iptables DROP statement
01:52.42sawgoodok got it stopped ...
01:52.50sawgoodno more messages scrolling by
01:52.50pabelanger-lapsawgood: Unprotected Asterisk box on a public network is never a good idea
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01:57.12ManxPowerWhen using the Zaptel/DAHDI and faxdetect=incoming I have to Answer() then wait.  Should I use a Wait(x) or a WaitExten(x)?
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02:04.32pabelanger-lapManxPower: Depends, add exten => fax,1,...  in your context. You'll have to play with the Wait() value
02:07.18ManxPowerSo I should use Wait instead of WaitExten?
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02:08.03pabelanger-lapManxPower: Are you expecting somebody to enter a extension?
02:10.52sawgood<PROTECTED>
02:10.58sawgoodin a nutshell what does this mean?
02:11.41pabelanger-lapsawgood: RTP is active
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02:13.29sawgoodI am not able to receive phone calls anymore (my trunk is registered) ... the only change is an upgrade to 1.6.2.8 from 1.6.2.6 (now when I call the DID) nothing populates in the CLI except that one statement
02:13.44WIMPysawgood: 802.1p for mor information
02:13.47sawgoodI can call outbound ... but I cannot receive calls
02:13.49WIMPy+e
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02:27.19sawgoodSo, I see on the Digium site the 'instructions' for installing Asterisk via RPM (very easy clean nice process) .. and now I know how to 'remove' Asterisk using the yum remove commands (so I can go from having Asterisk 1.6.2.8 to nothing and or from nothing to 1.6.2.8) in a matter of a few minutes
02:27.27sawgoodWhat I would like to know is if ....
02:27.53sawgoodis there a set of instructions for which yum install RPM packages are required if you want to go from nothing to 1.4.x
02:28.06sawgoodI think my box has stopped working with 1.6.2.8 ...
02:28.19sawgoodIt was working with 1.6.2.6 and earlier ...
02:28.59sawgoodthe yum RPM packages automatically install the latest 1.6.2.x release ... and I want to use 1.4.x (without having to build it from src)
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02:31.36booduis it possible to send a number to chan isdn from route number ?
02:35.04WIMPyboodu: Can you rephrase that?
02:35.32boodui can try :D
02:37.06booduI want a user compose a number as 0751044 and then the isdn call directly 751044. Actually user must dial 0 and after compose the real phone number
02:37.48booduIt's more understandable ?
02:38.18WIMPySounds like standard dilplan thing and nothing to do with any channel specific stuff.
02:40.36booduok, I must modify dialplan and don't use route with trunk for that
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02:44.54sawgoodwow!  I confirmed it ... I took Asterisk 1.6.2.8 off via the RPM remove process ... and I put 1.4.x on via the RPM process, and my box receives phone calls now ...
02:45.08sawgoodsomething 'broke' between 1.6.2.6 and 1.6.2.8
02:45.10sawgoodamazing
02:45.37sawgoodso, lesson here is ... if your box works, don't just run yum update because you want the latest release
02:46.04WIMPyWhat didn't work with 1.6.2.8?
02:46.55sawgoodWell, when I try to call the IP PBX with 1.6.2.8 installed, from the CLI ... I would only get the message about CoS 5 (then the line would go dead)
02:47.07sawgoodI got nothing else on the CLI (I could not receive a phone call)
02:47.18sawgoodWhen I had 1.6.2.6, calls flowed in just fine
02:47.42WIMPyOk, so nothing to worry about :-)
02:47.55sawgoodSo, I guess now ... I will need to get the SRC tarball for 1.6.2.6 and build that on the box
02:48.05sawgoodthen, if it works ... do not update it
02:48.06WIMPyjust upgraded from 1.6.2.0something to 1.6.2.9, but SIP works.
02:48.30sawgood1.6.2.9 is not in the RPM yet, right?
02:48.40WIMPynfi
02:49.20sawgoodActually, I'm glad this happened because with my excellent note taking ability ... I can overcome problems like this in the days ahead ..
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03:06.29Shaaananyone here do predictive dialing??
03:08.11[TK]D-FenderSIP always worked
03:10.47WIMPyUgh. Mu CLI just turned black on black.
03:14.20sawgood[TK]D-Fender: I read your archived 'frog' joke while researching Asterisk stuff on the net
03:14.23sawgoodfunny joke!
03:14.33[TK]D-Fendersawgood: Which?
03:14.46sawgoodfrog and the princess and the girlfriend
03:15.01[TK]D-Fendersawgood: Link it... I've clearly fogotten
03:15.34sawgoodno bookmark here ... it was during late night research on MWI lights and PFK buttons ...
03:15.52sawgoodIt was in a forum and/or chat room archive format though (on some web page)
03:17.42sawgoodDo you think Digium would be interested in knowing (which I have re-confirmed three times) ... my Asterisk box was working just fine under 1.6.2.6, but with a yum update to 1.6.2.8 (I was no longer able to receive an incoming SIP trunk call) ...
03:17.53sawgoodDo you think they would like to know about this in some form?
03:19.09sawgoodI have rolled back, and I am receiving calls just fine (I re applied 1.6.2.8 and could not receive a call) ... then back and it work ... etc .. etc.
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03:21.25[TK]D-Fendersawgood: You have absolutely nothing to show us and the weakest description this side of creation.
03:21.57[TK]D-Fendersawgood: Basically... you did something wrong or the packager did.  because the release versions would have caused a panic otherwise
03:36.11draivenhi, I am new user of asterisk, I install on ubuntu the asterisk 1.6.2.9 and asterisk-addons 1.6.1.4 from http://downloads.digium.com repositories and I has the error; "ERROR: __ASTERISK_SBIN_DIR__/asterisk not found" when I run service asterisk start
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03:38.14draivensomebody can help me?, and sorry for my english, i speak spanish
03:43.10draivensome body here?
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03:45.02WIMPyProbably noone hwo built those packages.
03:48.33draiveni unpack the files in /usr/src/ then i run sudo configure, sudo make, sudo make install, and sudo make samples
03:49.42WIMPyaye
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03:51.21draivenalso, I install before the zaptel and the libpri
03:51.43[TK]D-FenderAsterisk 1.6 does not WORK with Zaptel
03:52.02WIMPyzaptel has been replaced by dahdi and won't work with 1.6.
03:52.26draivenmmm
03:52.28draivenok
03:52.45WIMPyBut nfi on that 'service' thing.
03:53.01[TK]D-FenderAnd you need "make config" for init scripts
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03:55.14booduciao
03:55.42Kyoshwhats the best way to audit an asterisk box to find any problems to ensure no hackers (or at least minimize potential threats)?
03:56.52draivenI copy de init scripts from contrib/init.d/rc.debian.asterisk to /etc/init.d/asterisk
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04:09.41Kyoshis there any docs on working with T38 and asterisk (specifically v1.4)?
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04:10.22[TK]D-FenderKyosh: To do what?
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04:10.54coppiceguide space shuttles home. of course
04:11.27Kyoshwell i was wondering, i have asterisk v1.4 running.  right now i dont know how to get T38 faxing to work.  my provider supports T38 fax and my ATA's support T38 fax, but how do i get my asterisk box to support it as well?
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04:13.12_pepo_hi friends
04:13.50Kyoshhi pepo.  all my friends have been giving me money tonight.  want to be a friend? :)
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04:15.28_pepo_I pass, thanks
04:15.30_pepo_:D
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04:15.48Kyoshsome friend
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07:14.24derbaronhello @ all
07:15.18derbaroni have a question regarding analog line in the usa .... connecting to asterisk 1.6
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07:41.19stixHi guys. Is it impossible to manipulate with the EXTEN variable?
07:41.38stixTrying something like this: Set(EXTEN=11110${EXTEN})
07:41.45stixbut it wont change the variable
07:43.09ChannelZYou can re-write it but that will suddenly change the extension you would be executing if memory serves so your dialplan will probably misbehave
07:43.18ChannelZWhat exactly are you trying to accomplish?  There is probably a better way
07:44.13stixhmm okay
07:45.47stixwell the thing is, that the system I am trying to configure is using freepbx, so I can't just change what I want in the conf's. I have this "outbound route"/exten => _1111. and I want to add a 0 to all that is dialed here.
07:46.02stixEg exten => _1111.,n,Macro(dialout-trunk,5,${EXTEN:4},,) Becomes: exten => _1111.,n,Macro(dialout-trunk,5,0${EXTEN:4},,)
07:46.46kaldemarstix: you just did what you want
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07:47.32stixyes but I cannot change that line in the conf, freepbx will overwrite it. I can however add my custom code above, and there I wanted to rewrite the EXTEN variable
07:47.42stixguess that can't be done
07:48.43kaldemarask in #freepbx how that is done.
07:48.54stixyes I better
07:50.31kaldemarbtw, the EXTEN var cannot be set in the dialplan. only read.
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07:55.24stixokay thanks
07:59.04ChannelZAhhh I'm mis-remembering -- the case I'm thinking of was someone who was using exten qualifiers (not sure what they are called - like 555/111) and he was playing some games with CallerID causing the dialplan to fail
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09:04.20ndemirhere is a interesting question: can asterisk be used for voip infrastructure that will serve to 500.000 users?
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09:05.03*** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
09:05.27ndemirand how many servers should i use?
09:06.19mort_gibI have an B410P card, but I seem to only get 3 calls out of 3 ISDN lines???
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09:12.53tzafrirmort_gib, what would you expect to get? 6 calls?
09:13.02tzafrirOn which ports do you get those calls?
09:13.10mort_gibfor 3 ISDN2 circuits.. Yes
09:14.34mort_gibLooked like port 1-1
09:14.34mort_gibfor the first circuit, not in clients office now, just strange
09:15.34mort_gibtzafrir or, what does channel => 1-2 mean in chan_dahdi.conf if not channel one and two
09:16.02tzafriryes, 1 to 2
09:16.06ndemirno answer to me?
09:16.24mort_gibBut then I should be able to use channel one and two
09:16.33tzafrirndemir, "many"
09:16.34mort_gibndemir: What was your question??
09:17.02ndemirtzafrir , mort_gib : here is a interesting question: can asterisk be used for voip infrastructure that will serve to 500.000 users?
09:17.12tzafrirFor starters, how many concurrent calls do you expect? How many concurrent registrations?
09:17.13ndemirand how many servers should i use?
09:17.25mort_gibndemir yes, with careful planning
09:17.38mort_gibndemir and enough to carry the load
09:17.54ndemirmort_gib what are the important points?
09:18.09tzafrirndemir, that's the wrong question to ask. Start by understanding the exact details
09:18.34mort_gibhow many concurrent calls, what bandwidth you have, what codecs etc etc
09:19.33tzafrir"I have to move 500,000 KG from here to there. How many trunks do I need?"
09:19.37ndemirthat is the question i ask. suppose 300.00 concurrent calls i have. so what codec should i use for quality and what must be bandwidth?
09:20.40ndemirand i think i need load balancing, i should use may be 10 servers?
09:20.54mort_gibndemir interesting project, you do realize that you will need funding for proof of concept and proper testing, even before you start answering those questions
09:21.22mort_gibGet some really seasoned Asterisk gurus onboard too!
09:21.27ndemiryes, i know mort_gib.
09:21.42mort_gib:-) Sorry!
09:23.12ndemirmort_gib , why sorry? :)
09:23.16mort_gibndemir You will certainly need openSER and a clever infrastructure
09:23.46mort_gibBecause it is self evident that you will need people who know what they are doing when you do a 500KK user setup
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09:24.53mort_gibI'm looking into doing a 155 office setup now
09:25.06mort_gibWith between 15 and 25 users in each office
09:25.59ndemirmort_gib , can you tell what are you doing for infrastructure? basically?
09:26.07mort_gibIn a setup like that it's not only the amount of users, but also how many changes you get to the setup and how you handle them
09:26.17mort_gibWell, early stages
09:26.40mort_gibI will be using an external provider for the numbers
09:26.53mort_gibDDI's that is
09:27.00ndemirdou you need balancing for your project?
09:27.27mort_gibI'm playing with Asterisk running in a cloud, but waiting for the servers to arrive
09:27.50mort_gibThat could is for other projects too, I know there are issues getting Asterisk to run in a cloud
09:27.51tzafrir(openser has forked into kamailio and opensips)
09:27.59mort_gibtzafrir I know
09:28.14tzafrirndemir might not know
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09:28.31mort_gibWell thanks for correcting me :-)
09:28.38tzafrir<mort_gib> I'm playing with Asterisk running in a cloud, but waiting for the servers to arrive
09:28.52tzafrirSounds like a definition of "asterisk in the could" :-)
09:29.05mort_gibtzafrir Yes heh
09:29.25mort_gibWell I ordered 5 T310's from Dell last week
09:30.17mort_gibinteresting to see how it works, and if it scales nicer than just buying a monster server
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10:15.45Godfather_how can i see if my config in cdr_mysql.conf if connected to my asterisk?
10:15.56Godfather_to see the "status"
10:16.02Godfather_(1.6.0)
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10:18.00Godfather_No such command 'cdr mysql status' (type 'help cdr mysql status' for other possible commands)
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10:20.46kaldemarGodfather_: does "module show like cdr" list cdr_mysql.so?
10:21.39Godfather_kaldemar, http://pastebin.com/2JhJKCPc
10:22.12kaldemarcdr_addon_mysql.so seems to be the correct module name...
10:22.50Godfather_kaldemar, i installed it from addons yes...
10:23.11kaldemarunload the module and load it again to see if you get a warning of some sort.
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10:26.52Godfather_kaldemar, was a mistake on cdr_mysql.conf
10:27.02Godfather_i didnt uncomment [general] :|
10:27.08Godfather_Not currently connected to a MySQL server.
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11:07.45Godfather_kaldemar, http://pastebin.com/R5B0GS19
11:07.46Godfather_any ideas?
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11:53.32MartinblrDoes anybody knows BRI tone settings for greece?
11:56.00kaldemarMartinblr: indications.conf has some
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12:08.24Martinblrkaldemar: is there anything like power db for busy tone?
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12:13.27kaldemarMartinblr: what do you mean, exactly?
12:14.16Martinblrsomething like power db settings
12:14.43kaldemarand by power db you mean?
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12:17.57MartinblrI suspect it could be the power settings for the line
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12:56.41ndemirwhich codec should be used for ARM926 processor?
13:00.11kaldemarndemir: one that does not require the processor to transcode.
13:00.25pabelangerAgree
13:00.38ndemirhow is it transcoded?
13:01.03kaldemarif two legs of a call don't use the same codec.
13:02.11ndemirif two peers of calls use the same codec, the processor is not used. Is that true?
13:02.13pabelangerndemir: Ideally RTP should not be bridge via Asterisk.
13:02.41pabelangerndemir: correct, no transcoding is required
13:02.52kaldemarndemir: not true as you put it, but the processor is not used for transcoding.
13:03.11kaldemarndemir: which quite obviously was what you meant. :)
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13:15.15[TK]D-FenderRTP forwarding is a very small load.  Transcoding is not
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13:29.00radichttp://bravo.hopto.org/extensions.conf
13:29.31radicif I cakk 0900XXXXXX from context 373 It sohld be end in the context filter
13:30.12radicwhy asterisk let do the call?
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13:32.07wcselbyo/
13:32.16kaldemarradic: exten => _0ZXXX.,1,Dial(SIP/${EXTEN}@${OUTA}${CONTEXT}) is a better match than the include.
13:32.47pabelangerradic: *CLI> dialplan show 00900XXXXXX@373
13:32.51kaldemarradic: inside contexts, extensions are matched first, then includes in the order they appear in the dialplan.
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13:33.43radickaldemar: Is there a way that the include is processed befor the last extension?
13:34.01kaldemarradic: btw, is this: "exten => _090O.,1,GoTo(filter,1)" 090O and not 0900 on purpose?
13:34.11kaldemarradic: no.
13:34.40kaldemarradic: you need to put it in a separate context and include it before "filter".
13:36.54pabelangerradic: A debug log of your call will be help full too
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13:37.04pabelanger~collectdebug
13:37.14infobotit has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue.  Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
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13:50.44radicI have to list all nubers that shouldn't dialed in all 5 contexts?
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13:53.14E-bolaHave anybody ever heard/seen anybody use asterisk as a PA/Announcer type system. I have a new client who has an intercom system already, which allows the manager to "speak" to all basestations at the same time, so they can hear him in every room
13:53.38E-bolaI've setup asterisk and snom's intercom in combination which lets you do point A->B intercom
13:53.46*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
13:53.46E-bolabut they need point A->many
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13:56.41pabelangerE-bola: app_page?
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13:59.42[TK]D-Fenderradic: No, it looks in the DIRECT context first, then through the INCLUDE'sin the order you included them
13:59.49[TK]D-Fenderradic: Get your order right
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14:00.13[TK]D-Fenderpabelanger: Indeed
14:00.22[TK]D-Fenderpabelanger: They should name it something more obvious.
14:01.27ndemirhow to enable  SIP tcp transport?
14:02.10kaldemarndemir: you'll find the parameters, among others, in the sample sip.conf
14:02.14pabelangerndemir: transport=tcp,udp
14:13.31ndemirpabelengar thanx
14:14.44ndemirpabelanger: will it open tcp port 5060?
14:15.51puzzledhi
14:16.09pabelangerndemir: by default yes.  You can change it using binaddr=
14:16.33pabelangers/binaddr/bindaddr
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14:17.20ndemirpabelanger: ok i have to add tcpenable=yes
14:18.01pabelangerndemir: sip.conf.samples has all the information
14:23.04E-bolapabelanger: sounds like the right thing
14:23.13E-bolaalthough i get a WARNING[29159]: app_meetme.c:774 build_conf: Unable to open pseudo device
14:23.34pabelangerE-bola: DAHDI installeD?
14:23.58E-bolanot intentionally atleast. We've never used conferences before, but it looks like page needs that?
14:25.59pabelangerE-bola: not sure about app_page.c but your warning is from app_meetme.c.  Install DAHDI, recompile / install Asterisk
14:27.32WIMPyOr use ConfBridge instead.
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14:34.04[TK]D-Fenderapp_pagUses meetme.
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14:40.29pabelanger[TK]D-Fender: learn something new everyday
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14:46.20Godfather_hi
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14:57.13houmscan anyone help me with openfire asterisk-im? everything works great except for incoming call notifications in the spark client?
14:57.26Godfather_Im trying to connect cdr_mysql to asterisk, but i can't. i'm getting "failed to connect to database"
14:57.33Godfather_here is some output... http://pastebin.com/6aLaYQEX
14:58.09Naikrovekhoums: i have that working.  you need to associate an asterisk extension with the spark username
14:58.17Naikroveki need to update those, actually
14:58.21houmswhat do you mean?
14:58.26Naikrovekhang on
14:58.30houmscurrently I have the phone mappings done
14:58.34leifmadsenGodfather_: check the logs on the mysql side of things to see what part of the authentication is failing. From the Asterisk side you're not going to get much information.
14:58.35[TK]D-FenderGodfather_: sock=/tmp/mysql.sock <------ is it actually there?
14:59.11houmsand logging in to spark client works fine, when on call the status works properly, but no pop-ups show when incoming calls are received
14:59.40Godfather_[TK]D-Fender, vitto:/etc/asterisk# updatedb vitto:/etc/asterisk# locate mysql.sock
14:59.43Godfather_i get no output
15:00.14houmsNaikrovek how can i troubleshoot this? Or is there another configuration piece i am missing?
15:00.28Naikrovekhoums: i'm looking at mine now
15:00.45houmsthanks Naikrovek, your help is appreciated
15:01.07Godfather_leifmadsen, no output, http://pastebin.com/BwgLx3Ki
15:01.41radichow can I get the number auf a incoming call?
15:02.07Godfather_[TK]D-Fender, i see in my.conf ..
15:02.11Godfather_[client]
15:02.11Godfather_port            = 3306
15:02.11Godfather_socket          = /var/run/mysqld/mysqld.sock
15:02.36Godfather_should i remplace sock to /var/run/mysqld/mysqld.sock ?
15:03.49Godfather_[TK]D-Fender, http://pastebin.com/Gq3XNNiv
15:04.49Corydon76-digleifmadsen: aren't you on vacation?
15:05.02Naikroveksince when did vacation prevent people from working
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15:05.21Naikrovekhoums: i don't know why yours isn't working.  after looking at mine i'm not sure why it *is* working...
15:05.27leifmadsenCorydon76-dig: not really today... I'm supposed to be, but it's going to thunderstorm this afternoon and it rained yesterday, so we came back
15:05.37Godfather_[TK]D-Fender, you were right, now is loading, Loaded cdr_addon_mysql.so
15:06.01Godfather_Connected to asteriskcdr@localhost, port 3306 using table cdr for 49 seconds.
15:06.01Godfather_:D
15:06.52houmslol, I just setup the asterisk-im plugin and added the server. and added an entry to manager.conf. do you mind sharing what user the openfire server is using to talk to asterisk? are you using admin?
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15:07.06wcselbyfunny how putting thigns like the right path in your config files seems to make things work
15:07.10houmsNaikrovek are you using the admin user?
15:07.58wcselbyugh, i have a client that thinks the answer to any phone system issue is to reboot the phone server.  which wouldn't be so bad, if I didn't get emails notifying me of the fact, every time.
15:08.07[TK]D-FenderGodfather_: Sometimes you just have to read the big print with the giant neon sign and flashing arrow on it...
15:08.28Godfather_lol
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15:52.15wcselbymiracle goal in penalty time and USA advances
15:52.29wcselbywell, not a miracle, but a damn fine goal
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15:58.20ndemiris there a way to calculate maximum number of concurrent call for a asterisk server?
15:58.48Chainsawndemir: It is highly dependent upon whether any transcoding is required.
15:59.08[TK]D-Fenderndemir: Yes.  Reach it.
16:00.20ndemirChainsaw: suppose no transcoding is needed.
16:00.32Chainsawndemir: You've actively prevented it?
16:00.42ndemirChainsaw: yes.
16:01.01fenrusthere's benchmarking software that you can use :)
16:01.27ndemirfenrus: which benchmark is suitables?
16:02.41xhelioxUSA! USA! USA!  :)
16:02.47xhelioxThat was awesome.
16:04.35fenrusndemir, sipp
16:04.50fenrussipp.sf.net
16:05.26ndemirfenrus: thanks
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16:07.27ndemirI found this: The figure of 200 concurrent channels is based on a dual Xeon 2.8 Ghz system with 1 GB of RAM performing no echo cancellation and no codec transcoding.
16:07.38jcimsquick question.  should the reply to a SIP REGISTER command go to the source port that the SIP REGISTER command was sent from, or to port 5060 on the phone that originated the request.
16:08.11troffaskyI remember seeing something about source port brokenness on cisco handsets
16:08.18troffaskybut I can't find it now
16:08.29jcimsok...i'll look around.
16:08.31jcimsthanks
16:08.37troffaskyI've been searching for it since you asked in #cisco
16:09.01troffaskyI read it when I was setting up a 7940G to work wit asterisk
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16:09.51jcimslol, sorry about that. :)
16:10.56Bladerunner05hello all, with tdm410p and dahdi (at the latest version) if I do dahdi_scan I see the card and my 4 fxo port. but on cli if I do dahdi show channels I don't see any channel...
16:14.13russellbhave you configured it in both dahdi and asterisk?
16:14.22russellb<PROTECTED>
16:14.26[TK]D-Fenderclearly not
16:14.28russellbif not, do that.  :-)
16:14.58russellband if you have further trouble  getting the config right, http://www.digium.com/en/supportcenter/
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16:16.11Bladerunner05russelb: /etc/dhadi/system.conf yes, but leave untouched chan_dahdi..
16:16.25[TK]D-FenderBladerunner05: that is your job.  Go configure your channels
16:18.29russellbthat is your mission, should you choose to accept it.
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16:18.38russellbshould you not, no calls 4 u!
16:18.47carrarI thought the job of blade runner is to track do wn and terminate replicants?
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16:19.24p3nguinAre you asking him if that's what you thought?
16:21.34tzafrirBladerunner05, also: what's the output of lsdahdi ?
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16:27.18jcimstroffasky: gah! it was the voice_control_port setting buried in the sipdefault.cnf i was using.  it was set to 5061, which for some reason caused the phone to send queries from any port rather than 5061
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16:27.28jcimschanged it to 5060, reboot, phone works
16:27.31troffaskyah ok
16:27.37jcimsthanks for the help
16:27.43troffaskythere's a voip-info.org page about the 79xx series
16:28.19troffaskylisting all the interesting bugs in each release
16:29.04jcimsi'll check it out.  i'm on a crusty release, but no smartnet, so i'm just making due. :)
16:29.24jcimsthanks again...
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16:43.08sputnickhi there
16:44.12sputnickis there an implementation or a feature to automaticly and dynamicly change extensions for the lowest price route ?
16:44.41Qwellsputnick: sure.  it's called dialplan.
16:44.50sputnickor I need to be aware of prices from SIP providers and maintain a MySQL DB ?
16:45.20[TK]D-Fendersputnick: There is no AOC for SIP in *
16:45.52[TK]D-Fendersputnick: and * isn't psychic.  Indeed the diallpan does exactly what you tell it to.  If you have some table you can access for your vendors, then by all means get coding
16:46.14[TK]D-Fendersputnick: What you use as a pricing reference is up to you.  No-one else said you had to use SQL for this
16:47.33sputnick[TK]D-Fender: by AOC you mean http://en.wikipedia.org/wiki/Autonomic_Computing ?
16:48.07[TK]D-Fendersputnick: No.
16:48.12sputnicksorry
16:48.14[TK]D-Fendersputnick: Advice Of Charge
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16:48.55[TK]D-Fendersputnick: there is standard way to know which resource will be cheapest.  You'll have to mash it up yourself
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16:52.02sputnick[TK]D-Fender: by "standard way", what did you thought ?
16:52.11sputnickweb scraping ?
16:53.01[TK]D-Fendersputnick: Every vendor lists their prices however they feel like.  Some may be a parseable web page.  Other more direct CSV.  Some direct DB driven.  Who knows.  the thing is EVERY one will be different
16:53.08[TK]D-Fendersputnick: Good luck keeping up with them all.
16:53.16sputnick:)
16:53.20leifmadsenI read "keeping" as "sleeping"
16:54.09sputnickthanks for the explanation [TK]D-Fender
16:56.13sputnickI think to deal only with a small number of providers to start. If that's starts with 33686xxxxxx -> provider foo ;  If that's starts with 33656xxxxxx -> provider bar
16:56.47[TK]D-Fenderleifmadsen: Slut-tastic :-)
16:56.59*** join/#asterisk ndemir (~ndemir@94.121.167.175)
16:57.19[TK]D-Fenderleifmadsen: I know... I'm just whore-able :p
16:57.26[TK]D-Fenderturns his puns to "11"
16:57.27leifmadsenba-doom-chik
16:57.36ndemiranother question: when should i use isdn pri card?
16:57.48Qwellndemir: because you have an ISDN PRI
16:57.56[TK]D-Fenderndemir: When you want to use an ISDN PRI with Asterisk
16:57.58Qwellerr, when
16:58.02leifmadsenQwell: I was going to say something like that :)
16:58.32[TK]D-FenderDoctor, Doctor!  It hurts when I raise me arrrr..... awww FUKKIT
16:58.35[TK]D-Fendergives up
17:00.49ndemir<PROTECTED>
17:01.57[TK]D-Fenderndemir: That's the idea.
17:02.56*** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs)
17:03.04ndemirfor example in this picture (http://www.voip-info.org/img/wiki_up//asterisk_1.png), asterisk servers connect each other via ISDN. Why do i need this instead of connecting via IP address?
17:05.09[TK]D-Fenderndemir: Depends where A & B *are*
17:05.23[TK]D-Fenderndemir: SIP over the internet = massively unreliable
17:06.04[TK]D-Fenderndemir: locally I might be inclined to use SIP on a secondary NIC, or perhaps TDMoE if I found it necessary.  Neither requires a special card, just a NIC
17:08.05[TK]D-Fenderndemir: And your picture didn't show a fax at all BTW
17:08.34ndemir[TK]D-Fender: it is not my picture. i just found it.
17:09.36[TK]D-Fenderndemir: Hi, here's an unrelated picture, now what's wrong with ME systeM?!?!?!
17:09.39[TK]D-FenderMY*
17:10.25ndemir[TK]D-Fender: i am trying to understand when to use ISDN.
17:11.02[TK]D-Fenderndemir: When you need to interface with one.
17:11.24drmessanoDon't drive your car to the kitchen, the hallway won't like it
17:11.28*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
17:12.51*** join/#asterisk ruben23 (~ITadmin@125.212.40.2)
17:15.48ndemir[TK]D-Fender: now i have an asterisk working with SIP (over IP), it works. On the other hand, i have an ISDN card, i installed it, compiled kernel modules, created conf with genzatelconf. Now what to do?
17:17.07[TK]D-Fender~nowwhat
17:17.08infobotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk
17:20.29pabelangerinfobot: Get a pepsi
17:20.30infobotACTION fetches a pepsi
17:20.51*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
17:22.43*** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net)
17:25.20*** join/#asterisk Faithful (~Faithful@202.6.145.116)
17:28.40eduzimrs[TK]D-Fender is prefered to use IAX over internet?
17:30.12t_dot_zilladoes asterisk adjust the volume of MOH ? the MOH was too loud, so we lowered the volume in audacity and replaced the MOH in asterisk but it seems to be the same volume
17:30.28Qwellt_dot_zilla: the phones likely do
17:30.46Chainsawt_dot_zilla: It depends on how you produce the music-on-hold, but yes, the other end may be adjusting the volume as well (automatic gain control).
17:30.51pabelangereduzimrs: just easier to manage
17:31.01Chainsawt_dot_zilla: I have a volume adjustment on mine, which is pretty far down.
17:31.20t_dot_zillawe are calling in from TDM and it seems to be too loud
17:31.27WIMPyalso finds that he has to save MOH files at extremely low volume in order to get acceptable results.
17:31.29Chainsawt_dot_zilla: application=/usr/bin/dumbout /etc/asterisk/LINX/TheBlueValley.s3m -m -s 8000 -r 2 -v 0.2 -o -
17:31.59Chainsawt_dot_zilla: The -v switch on that application controls the volume. It seems to have quieted down the complaints of hold music being "excruciatingly loud".
17:32.40Qwells3m?
17:32.54ChainsawQwell: Yes, it's a sequenced music format. Consider it like MIDI with embedded samples.
17:33.09*** join/#asterisk knctrnl (~aembrey@76.164.169.130)
17:33.20t_dot_zillaChainsaw: i'm confused by your application=     i tried something like /usr/bin/play fpm-calm-river.wav | sox -r 16000 -t wav - -r 8000 -c 1 -t raw - vol 0.10
17:33.28t_dot_zillai got errors in asterisk
17:33.28ChainsawQwell: -rw-r--r-- 1 tony users 553K 2009-10-23 14:21 /etc/asterisk/LINX/TheBlueValley.s3m
17:33.37ChainsawQwell: Not bad for 12+ minutes of music.
17:33.55Chainsawt_dot_zilla: Do not pipe, use a single application.
17:34.05t_dot_zillaChainsaw: what is in /usr/bin/dumbout
17:34.26Chainsawt_dot_zilla: That is an application to play sequenced music files.
17:34.39Chainsawt_dot_zilla: /usr/bin/dumbout: ELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.9, stripped
17:35.20Chainsawt_dot_zilla: If you have an application called DUMB in your repository... I can get you the S3M and you can try that?
17:35.30[TK]D-Fendereduzimrs: IAX2 is preferred when it is needed.  When it isn't then SIP is
17:35.53Chainsaweduzimrs: IAX2 ability to penetrate NAT & firewalls is not to be underestimated.
17:36.58*** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)
17:37.40MiccI need to find a digium reseller in the Seattle area.
17:37.50MiccWe need to get a TE122p today if possible.
17:38.19QwellMicc: https://www.digium.com/en/ecosystem/resellers/locate.php
17:38.25*** join/#asterisk grinder13 (~grinder@146.176.165.57)
17:39.22[TK]D-FenderChainsaw: Was it good for you?
17:39.22*** join/#asterisk RobH (~robh@wikimedia/RobH)
17:39.24[TK]D-Fenderlights up
17:40.08grinder13hello! i 've compiled and installed asterisk from svn. while trying to connect to the asterisk CLI I get the following error: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)"
17:40.18Qwellgrinder13: does it exist?
17:40.32Chainsawgrinder13: What user is the Asterisk daemon running as? What user are you calling asterisk -r as?
17:40.40grinder13i discovered that the problem is here: "asterisk[1317]: segfault at d5 ip 08066032 sp bf904684 error 4 in asterisk[8048000+1af000]"
17:41.04Chainsawgrinder13: You've broken it already? That's quick. How did you get this one? Installed from a package manager?
17:41.05*** join/#asterisk MiserySoft (~Lee@nat66.mia.three.co.uk)
17:41.22grinder13as i said I compiled from svn
17:41.55grinder13any hints?
17:41.57*** part/#asterisk MiserySoft (~Lee@nat66.mia.three.co.uk)
17:41.58Chainsawgrinder13: Asterisk is developed by humans, not robots. Try the latest release to see if it a problem in the source or the way you've compiled it?
17:42.09Qwellgrinder13: what branch did you install?
17:42.21Chainsawtransfers the call to Qwell
17:42.29Qwellpresses DND
17:43.23grinder13that's what i thought Chainsaw. Qwell, I 've used the SVN trunk code from here: http://svn.asterisk.org/svn/asterisk/trunk/
17:43.40Qwelldon't use trunk
17:43.46Qwelluse branches/1.6.2/
17:44.04grinder13i need to use svn because of SRTP
17:44.11paulcwhile I'm on hold for some more bad customer service from Dell... How can I tell if dahdi_dummy is properly loaded, and where it's getting its timing source from?
17:44.27Qwellpaulc: what version of dahdi?
17:44.32*** join/#asterisk Draiven (~draiven@cable201-232-155-11.epm.net.co)
17:44.48Chainsawpaulc: The dahdi_dummy kernel module should show up in lsmod.
17:45.19*** join/#asterisk RobH (~robh@wikimedia/RobH)
17:45.30t_dot_zillain the musiconhold.conf, if you use the application=something, does mode=custom or can it be =files     will the application still run?
17:45.37Chainsawt_dot_zilla: mode=custom
17:46.11*** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net)
17:46.18paulc@Qwell 2.3.0.1+2.30
17:46.31Qwellthen dahdi_dummy does not exist
17:46.38Qwellif dahdi is loaded, it's providing timing
17:47.47paulclsmod shows dahdi but no dahdi_dummy
17:48.26Chainsawpaulc: Based on the version you have, you shouldn't have dahdi_dummy at all. Listen to Qwell.
17:48.31grinder13so regarding my segfault error in the syslog, no other tips/solutions apart from recompiling with the latest SVN release?
17:48.48Chainsawgrinder13: You should use the 1.6.2 branch, not trunk.
17:49.03Chainsawgrinder13: Or a release tarball, of course. Those work well.
17:49.16*** join/#asterisk sekil (~sekil@78.24.111.218)
17:49.22paulcSo just seeing "dahdi" in my lsmod is good enough and I'm all good for timing with no hardware
17:49.36Qwellyes
17:49.42Chainsawpaulc: Yes, the "core timer" is now enabled by default. You'll be fine.
17:50.02Chainsawpaulc: (I used to have to patch it in Gentoo, but it's the default)
17:50.10grinder13Chainsaw, as I said, I need the SRTP feature for my project. Unfortunatelly, I can't do without it. If it wasn't the SRTP I would install directly from the packages of my Linux distribution
17:50.31Chainsawgrinder13: And you are sure that this feature is not in the 1.6.2 branch?
17:50.44grinder13yes, it's not there
17:50.58Chainsawgrinder13: That's a shame. You are unlikely to get support on trunk builds.
17:51.05*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
17:51.28grinder13well, will see then
17:51.52grinder13i 'll try my luck with the latest SVN release
17:52.00Chainsawgrinder13: If you got me a backtrace or something more substantial...
17:52.03drmessanoThat's all you can hope for, is luck
17:52.38Chainsawgrinder13: Then perhaps we could all give you some educated guesses of what's going wrong. But saying "it exploded" and giving me a report that says "exhibit A exploded"... doesn't leave me with much.
17:53.01drmessanoChainsaw: It exploded and it was bad
17:53.30Chainsawdrmessano: It was bad? This is how it responded to "core restart gracefully" barely 4 versions ago.
17:53.40drmessanolol
17:53.41grinder13you are right, but how can I get a backtrace? sorry, I am not familiar with these programming stuff (feeling embarassed already). networking guy here...
17:53.47*** join/#asterisk Z_God (~julius@wlan224088.mobiel.utwente.nl)
17:53.57Chainsawgrinder13: Okay, make sure you haven't stripped the binaries and run it through gdb.
17:54.05t_dot_zillacan someone tell me what this means:
17:54.08t_dot_zilla[Jun 23 13:53:36] NOTICE[19880]: res_musiconhold.c:602 monmp3thread: Request to schedule in the past?!?!
17:54.10t_dot_zilla[Jun 23 13:53:36] WARNING[19880]: res_musiconhold.c:620 monmp3thread: Unable to send a SIGHUP to MOH process?!!: No such process
17:54.23[TK]D-Fendert_dot_zilla: means your timing is off and you're using MPG123 for MoH
17:54.35t_dot_zillai'm using /usr/bin/play
17:54.38[TK]D-Fendert_dot_zilla: change to NAtive MoH (mode=files)
17:54.45Chainsawgrinder13: Once it explodes, you'll get a gdb prompt. Type bt full and stick the result on pastebin somewhere.
17:54.53[TK]D-Fendert_dot_zilla: same diff
17:55.11t_dot_zilla<[TK]D-Fender>: i'm trying to use a customer app in MOH to lower the volume
17:55.20t_dot_zilla*custom
17:55.44Chainsawgrinder13: If we're lucky, you have debugging symbols and the output will be useful in determining where things derail.
17:56.06[TK]D-Fendert_dot_zilla: then jsut normalize your files
17:56.15Chainsawgrinder13: (Just "gdb asterisk" should get you going)
17:56.21[TK]D-Fendert_dot_zilla: You're wasting processing load, processes, etc
17:57.21grinder13thanx Chainsaw, I am reading the debbuging info on voip-info.org. will get back
17:57.29*** join/#asterisk southtel_ (~slester@c-24-126-177-12.hsd1.ga.comcast.net)
17:58.18*** join/#asterisk Acidshock (~none@cpe-75-84-10-22.socal.res.rr.com)
17:58.19Chainsawgrinder13: Okay.
17:59.20southtel_Can anyone suggest ways to troubleshoot CallerID problems for an analog system out in a rural area?
18:00.09southtel_The phone lines out here are not great, but with a regular phone, the CID shows up fine.
18:00.20paulcChainsaw / @Qwell : Thanks for your help (delayed reply, Dell support were consuming my energy.. aka will to live..)
18:00.27*** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br)
18:00.40southtel_But, on my * box (w/Digium wildcard), I only "see" the CallerID about 50% of the time.
18:00.49AcidshockCan anyone help me out with a dialplan issue I am having? I cant seem to get a sip trunk up and running. Keep getting error cause 3 no route to destitination errors
18:01.06Acidshocklol destination even :P
18:01.19[TK]D-FenderAcidshock: "sip show peer [yourpeerhere]" <- PASTEBIN it
18:01.21[TK]D-Fender~pb
18:01.21infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
18:01.22[TK]D-Fender^^^^^^^^^^^^^^^66
18:01.49grinder13hmm, as I am reading stuff in voip-info.org I noticed one thing: The damn fool I forgot to select the DON'T OPTIMIZE compiler flag!!!!! ARGHHHH!!!
18:02.04southtel_I've tried tweaking the cid_rxgain settings, but they don't _seem_ to make a difference.
18:02.10AcidshockFender, I am using IP Authentication not registration. Do you still want me to pastebin that for you?
18:02.22[TK]D-FenderAcidshock: Provide what I have requested
18:02.39grinder13will "make uninstall-all && make distclean && make" again...
18:03.06Acidshockjust making sure :)
18:04.02t_dot_zillaChainsaw: can you tell me more about how you've set up MOH?
18:04.09AcidshockFender, http://pastebin.com/VDn6ZVgh
18:04.19Chainsawt_dot_zilla: I use the DUMB suite of software to play sequenced music in S3M format.
18:04.27Chainsawt_dot_zilla: As I asked, do you have such software in your package manager?
18:04.41t_dot_zillano, i don't
18:04.42*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)
18:04.51t_dot_zillai'm running centos, i don't see it in yum search
18:05.12Chainsawt_dot_zilla: Right, you may have to compile that from source then. You can listen to my hold music if you call me.
18:05.26Chainsawt_dot_zilla: Perhaps tell me what the volume difference is like as well.
18:05.34t_dot_zillaok
18:06.20*** join/#asterisk iratik (~itariki@74.223.41.171.nw.nuvox.net)
18:07.48iratikHow can I make it so that calls cannot be made from outside our network and invite requests from our ITSPs? My initial thought is to make hosts.deny ALL and hosts.allow selectively permit local ip's and itsp ip's .. is there a setting in asterisk i can use?
18:08.20*** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net)
18:09.37southtel_Acidshock: can you access that ip from your machine?
18:09.44Acidshockyup
18:10.26southtel_Can you put on the pastebin the output from the cli when you try to dial using it?
18:11.18*** join/#asterisk theHub (~theHub@69.177.93.21)
18:11.37Acidshocksure 1 sec
18:12.50Acidshocksouthtel_, http://pastebin.com/B4Gsw1Qu
18:13.33Draivenhi,  i want install a cdr and realtime with my postgres db but it show me the error: found to engine 'pgsql', but the engine is not available
18:13.34*** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-77-81.home.otenet.gr)
18:14.07[TK]D-FenderAcidshock:   Addr->IP     : (Unspecified) Port 0 <--- You did nOT specify the HOST to call
18:14.19[TK]D-FenderAcidshock: Fix your peer.  Asterisk has nowhere to call now
18:14.39Draivensomebody can help me?
18:14.49Draivenplease
18:16.11[TK]D-FenderDraiven: PASTEBIn your configs, show us the module attempting to load.  Show that you're able to conenct locally to the DB, etc.
18:16.13[TK]D-Fender~pb
18:16.13infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
18:16.15[TK]D-Fender^^^^^
18:17.00southtel_D-Fender: how many times a day do you trigger that pastebin infobot?
18:18.04*** join/#asterisk dkirker-openmo-1 (~dkirker@openmobl/ceo/dkirker)
18:19.13kaldemarDraiven: res_config_pgsql.so is the module you're missing. you can also use a postgresql db via ODBC, as in the example in the book.
18:19.30DraivenI am conected to postrges localy with pgadmin
18:19.32[TK]D-Fendersouthtel_: Every time someone new needs to show us something and likely has no clue
18:19.45[TK]D-FenderDraiven: PASTEBIN <-------------------------
18:19.50*** join/#asterisk guilhermebr (~Guilherme@200.103.96.98)
18:20.22*** join/#asterisk uqlev (~yuriy@91.184.221.31)
18:20.58southtel_Good times.
18:21.03Draiveni dont have de res_config_pgsql module
18:21.06Draiven:(
18:21.41southtel_D-Fender: I'm looking elsewhere as well, but do you have any thoughts on tracking down a CID problem in a rural area?
18:21.57AcidshockFender, what would I use to specify that address because I am using host=<IP ADDRESS>
18:22.22kaldemarDraiven: how did you install asterisk?
18:23.14*** join/#asterisk hfb (~hfb@96.247.65.56)
18:23.34DraivenI download the repositories from http://downloads.digium.com
18:24.08Draivenasterisk 1.6.2.9 and asterisk-addons 1.6.1.4
18:24.26QwellDraiven: repositories?  downloads.digium.com?  one of those is wrong
18:24.42kaldemarDraiven: you compiled from source?
18:25.56Draiveni install runing ./configure
18:26.00Draivenmake
18:26.04Draivenmake install
18:26.39kaldemaryou need to install the dependency for res_config_pgsql, re-run configure and recompile.
18:27.22*** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net)
18:29.17[TK]D-FenderDraiven: Asterisk add-ons 1.6.1 is NOT meant to be used with Asterisk 1.6.2
18:29.39[TK]D-FenderDraiven: Stop putting Ford Mustang parts in your Toyota Prius
18:29.54[TK]D-FenderDraiven: Asterisk-Addons 1.6.2.1 <---------------------
18:30.07[TK]D-FenderCRAZY PEOPLE
18:30.27Acidshockcheerful group lol
18:31.02[TK]D-Fender[14:21]<Acidshock>Fender, what would I use to specify that address because I am using host=<IP ADDRESS> <--- ummm..
18:31.12[TK]D-Fender[14:14]<[TK]D-Fender>Acidshock: Addr->IP : (Unspecified) Port 0 <--- You did nOT specify the HOST to call
18:31.24[TK]D-FenderAcidshock: NO.  Or you are not showing me what I asked for.
18:31.56[TK]D-FenderAcidshock: Whatever you think you've done does not matcht eh evidence.
18:31.58Acidshocklet me PASTEBIN it for you :P
18:32.53grinder13i cannot take a backtrace. Asterisk crashes immediately after I issue the command "asterisk -r" (or "gdb asterisk" or "safe_asterisk" or whatever) and exits, so absolutely nothing is captured by gdb. any ideas?
18:33.16[TK]D-Fendergrinder13: start it MANUALLY
18:33.45grinder13but that's what I am doing
18:34.19AcidshockFender, http://pastebin.com/ZgQHSpyR
18:35.10[TK]D-Fendergrinder13: Nothing you have shown me is "manually".  "asterisk -gvvvvvvvvvvvvvc" is "manually"
18:35.46[TK]D-FenderAcidshock: type=friend <- should be "peer".
18:35.51grinder13but that's what I am telling you Fender
18:36.14[TK]D-FenderAcidshock: update.  Apply.  then PB "sip show peers", the dump for that specific peer again, and your actual failed call at verbose 10.
18:36.42[TK]D-Fender[14:35]<grinder13>but that's what I am telling you Fender <_ you did NOT do it properly, MANUALLY ilke I showed.  so go DO it, and then SHOW US
18:37.22grinder13how many times do I have to say that I did so???
18:37.48Draiven[TK]D-Fender, hehe, can i use asterisk-1.6.2.9 with addons-1.6.2.1?
18:38.06[TK]D-Fender[14:29]<[TK]D-Fender>Draiven: Asterisk-Addons 1.6.2.1 <---------------------
18:38.09[TK]D-Fender^^^^^ NOT OBVIOUS ENOUGH?
18:38.21[TK]D-Fenderlooks for the giant flashing neon sign again
18:39.02kaldemarwhat's the "<---------------------" in the version?
18:39.18grinder13http://pastebin.com/Cx5GUAfk
18:39.19kaldemaralmost .1 but not quite?
18:39.20wcselbyit's the awesome arrow version
18:39.32grinder13at the end of the above link you 'll see a segmentation fault
18:39.40*** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org)
18:39.41wcselbygrinder13 - have you tried starting asterisk with "asterisk -gvvvvvvvvc" from the command line?
18:39.53wcselbygrinder13 - nvm
18:40.02grinder13yeap, that's exactly what I did wcselby
18:40.28grinder13asterisk exits with a segmentation fault
18:40.32[TK]D-Fenderpulls another arrow from his quiver, nocks, draws, releases, and watches kaldemar drop like a sack of potatoes
18:42.21*** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt)
18:42.23[sr]howdy
18:42.53WIMPy[sr]!
18:43.07[sr]is there something that can send the information do the SIP phone's to learn the SIP server, a bit at the image of dhcp boot
18:43.10[sr]hi WIMPy
18:43.24[sr]dont know if this exists, but its interesting if yes
18:43.29AcidshockFender, http://pastebin.com/QARaZrFJ
18:43.31Bladerunner05wich is the dahdi command to make a call from cli ?
18:43.40*** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca)
18:44.31*** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net)
18:44.49AcidshockFender, slightly updated one with error from CLI http://pastebin.com/8m5N3mS5
18:45.10WIMPy[sr]: Depends on the phone, but most have some way of mass provisioning.
18:45.17kaldemarBladerunner05: there is no such dahdi command. either "console dial" or "channel originate".
18:45.40tzafrirBladerunner05, originate / channel originate
18:45.41Bladerunner05kaldemar: of course..
18:45.59[TK]D-FenderAcidshock: [Jun 23 11:43:48] WARNING[3954]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory <-- you calling channel is already broken.
18:46.06[TK]D-FenderAcidshock: Try again with a sane call
18:46.10[sr]WIMPy: thats the name for this protocol/method/techology?
18:46.22*** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
18:46.53WIMPy[sr]: can be anything. tftp, ftp, http or eben https. Check your phones manual.
18:47.01AcidshockFender, thats because I am dialing from CLI though correct? I can dial from a sip device and not receive that message
18:47.06AcidshockFender, trying again
18:47.08[sr]WIMPy: hum.. going to check :)
18:47.20[sr]WIMPy: for now i just have one yealink
18:47.39WIMPyNever heard of.
18:47.50[sr]yealink.com
18:48.29AcidshockFender, same problem http://pastebin.com/TLXespUL
18:48.37WIMPyLooks important
18:49.15[sr]WIMPy: they are accessible, i have the T28P
18:50.18*** join/#asterisk smooth_penguin (~smoove@59.95.54.237)
18:52.46WIMPyLooks interesting.
18:53.13[sr]112€+VAT
18:55.05ruben23hi guys can i set the directory manually of my recordings..? the dedault is /var/spool/asterisk/monitor/ can i put it like /var/spool/asterisk/monitor/archive
18:55.17WIMPy[sr]: Seems ok. I'm looking forward to your experiences with it.
18:56.01idespinner[sr], for autconfiguration of phones, it depends on the phones. polycoms can use DHCP + FTP to be provisioned
18:57.29grinder13i managed to get a backtrace. the problem is the libsrtp. damn it's the thing I need for my project!!!! will start hitting my had on the wall!!!
18:58.35*** join/#asterisk fish-bulb (~cstewart@nat/digium/x-gamfhoashrmsohrk)
19:01.37*** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com)
19:02.41[TK]D-FenderAcidshock: Where are your configs?  check your routing on your server
19:04.47*** join/#asterisk neurosys (~neurosys@69.199.204.33)
19:05.46*** join/#asterisk xxiao (~xxiao@140.242.26.81)
19:05.51*** join/#asterisk neurosys (~neurosys@69.199.204.33)
19:06.12xxiaonew to asterisk, what's the sip stack used by asterisk?
19:06.21QwellAsterisk
19:06.40xxiaogrep the source code and did not find any specifics
19:06.53[TK]D-FenderQwell: QUICK : What colour was Napoleon's white horse?!?!
19:06.53Qwellit's just Asterisk.
19:06.55xxiaodo you mean asterisk use its own sip stack
19:07.03[TK]D-Fenderxxiao: Clearly
19:07.34xxiaothanks. i was looking at osip2, pjsip and opensip and wondering if asterisk used them directly
19:07.57*** join/#asterisk radic (~radic@178.2.208.89)
19:08.13xxiaoneed find a small footprint sip stack on a portable device to work with asterisk
19:09.17[TK]D-Fenderxxiao: Perhaps you should look at osip2, pjsip and opensip
19:10.01xxiao[TK]D-Fender, thanks! i looked at openh323 and vovida 6 years ago and today with a new voip project, i found they all gone
19:10.26xxiaoand asterisk is just getting stronger
19:11.02[TK]D-Fenderxxiao: sofiasip <-
19:11.09[TK]D-Fenderxxiao: sofia-sip <-
19:11.21[TK]D-Fenderhttp://sofia-sip.sourceforge.net/
19:11.51xxiaothanks again, another new project
19:12.09xxiaowill compare them soon
19:12.14radicI've a dynamic IP. after a reconnect I'm not reachable until I did "sip reload"
19:12.34[TK]D-Fenderradic: READ <-
19:12.38[TK]D-Fender~sipnat
19:12.38infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
19:12.40[TK]D-Fender^^^^
19:14.47[sr]WIMPy: i'll tell, only have it for less than a day :p
19:15.02[sr]idespinner: hum... will check that
19:15.34WIMPyjust noticed, that the website only says copyright 2001-2009 and the firmware link is dead.
19:15.46radic[TK]D-Fender: that dosn't help me
19:16.25radicexternhost= is set in sip.conf
19:16.34[TK]D-Fenderradic: Fix your REFRESH
19:16.52[TK]D-Fenderradic: And verify your forwarding <-
19:17.39*** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net)
19:18.12*** join/#asterisk sekil (~sekil@78.24.111.218)
19:19.05ruben23hi guys can i set the directory manually of my recordings..? the dedault is /var/spool/asterisk/monitor/ can i put it like /var/spool/asterisk/monitor/archive
19:20.36knctrnlhas anyone ever got successfully integrated an avaya phone system with TN747B card with asterisk?
19:20.49[TK]D-Fenderruben23: No
19:21.49radicruben23: cd /var/spool/asterisk/monitor && ln -s /var/spool/asterisk/monitor archive
19:22.01[TK]D-Fenderknctrnl: Loks like a broing FXO interface.  Got a specific question pertaining to your goals?
19:22.08p3nguinWhy would you need to cd just to ln?
19:22.29[TK]D-Fenderradic: NO.  Go lookup RECUSION in the dictionary
19:22.33[TK]D-FenderRECURSION
19:22.51p3nguinI bet MixMonitor can use archive/<string>.<format> as the file name.
19:23.46p3nguinFor example:  MixMonitor(archive/${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.wav,a)
19:24.11knctrnlat one site we have an ancient avaya that we want to slowly transition to asterisk avaya will have pri to telco and asteirsk will have pri to telco. Telephone company will transition did's over as we need but we need to maintain 4 digit dialing.
19:25.07ruben23radic: but i want to put it on another mounted HDD, since its getting full on the 1st one
19:25.41dohdknctrnl: ancient avaya and mix with anything?! is that possible?
19:25.58WIMPyknctrnl: Maybe a 2nd pri for your asterisk would habe been a better choice than a 2nd pri from your telco.
19:26.13*** join/#asterisk Holos (~cosmond@static-pppoe-209-91-139-211.vianet.ca)
19:26.34radicruben23: mount it to /var/spool/asterisk/monitor
19:29.57*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
19:30.01knctrnlits a gs3i and I think there is a card in there that supports E&M
19:30.52knctrnli know if i look through the config ISDN is not enabled.
19:33.01x-demonwhat i need to configure in order to use sip url calls?
19:35.46*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)
19:36.14[TK]D-Fenderx-demon: exetnsions.conf dial command
19:37.21x-demon[TK]D-Fender, i know how to make dialplans... but i dunno how to configure rule for all SIP URLs
19:37.47*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)
19:37.53pabelangerx-demon: DNS SRV records?
19:38.16x-demonpabelanger, enabled and exists in on my server
19:38.44[TK]D-Fenderx-demon: You'll have to make a dialplan script to parse out the SIP header for your destination for the inbound call to strip the URI
19:38.59[TK]D-Fenderx-demon: "core show function SIP_HEADER
19:39.25x-demon[TK]D-Fender, "Not available"
19:39.43[TK]D-Fenderx-demon: show me
19:40.11x-demonin console i also see that xmldoc is not available, something like that
19:40.33x-demonWARNING[20991]: xmldoc.c:1720 xmldoc_build_field: Couldn't find application Park in XML documentation
19:40.49pabelangerx-demon: install libxml2-dev, recompile, install asterisk
19:41.05x-demonpabelanger, i use asterisk from debian repo
19:41.26x-demonbackported from sid, but i also saw such messages with lenny and squeeze builds
19:41.29[TK]D-Fenderx-demon: Meaningless.  get a VERSION NUMBER
19:41.50*** join/#asterisk neurosys (~neurosys@69.199.204.34)
19:42.10x-demon[TK]D-Fender, Asterisk 1.6.2.7-1.1
19:42.33[TK]D-Fenderx-demon: show me the check & error
19:42.40x-demonthe check?
19:43.25[TK]D-Fenderx-demon: I want to see you issuing the CLI request
19:44.08x-demonversion number?
19:45.35[TK]D-Fender[15:38]<[TK]D-Fender>x-demon: "core show function SIP_HEADER
19:46.17x-demon[TK]D-Fender, http://pastebin.com/E11UL011
19:46.59[TK]D-Fenderx-demon: Now spell it wrong and see if it looks the same.  Also I do not SEE you issuing the command in that PB
19:47.07elielx-demon: run this please: #dpkg -l |grep xml
19:47.40pabelangerx-demon: ls -la /usr/share/asterisk/documentation/
19:48.57x-demoneliel, libxml2 and libxml2-dev installed
19:49.19elielx-demon: asterisk from a debian package or from sources?
19:49.27x-demoneliel, backport from sid
19:49.49elieland that means? (from sources?)
19:49.50x-demonpabelanger, http://pastebin.com/niYwE32W
19:49.59x-demoneliel, debian package.
19:51.11pabelangerx-demon: core-en_US.xml is the file that lists all the commands.  For some reason, asterisk is not loading it.
19:51.24x-demonpabelanger, let me check locale...
19:51.38x-demonoh well, en_US
19:51.42pabelangerx-demon: *CLI> core show settings
19:51.45x-demonand no LC_ALL
19:51.51elielx-demon: cat /etc/asterisk/asterisk.conf |grep documentation_language
19:52.11elielsorry, already answered that
19:52.19x-demonoh well... now i understand why...
19:52.41pabelangerdrum-roll
19:54.03elielmaybe the libxml2 dependency was not met when installing asterisk (??)
19:54.10Kobazanyone know how to set up lldp on a dell powerconnect switxh
19:54.36pabelangereliel: core-en_US.xml exists though
19:55.30x-demoneliel, no i installed it before building backport
19:55.55x-demoneliel, btw, in which context i must put documentation_language = en_US
19:56.06x-demon[options]?
19:56.26elielyes
19:57.03x-demonsame error
19:57.18x-demondoes it need separate module? i believe no...
19:57.20elielx-demon: check file permissions
19:57.27elielx-demon: no, it is in the core
19:58.02x-demonpermissions 755
19:58.16elielasterisk running as ....?
19:58.21x-demonas asterisk
19:58.27x-demonwith group asterisk
19:58.28elielsu asterisk
19:58.36grinder13regarding my segfault with the libsrtp: i 've compiled libsrtp from source and it seems its fine now
19:58.39elielthen try to read the core-en_US.xml file
19:58.40x-demoni can't, this is no-login user
19:59.03x-demonsu -c "cmd" asterisk?
19:59.30elielok, stop asterisk and start it with: asterisk -vvvc and copy all the output to pastebin
19:59.59*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
20:02.00x-demoneliel, http://pastebin.com/QsasWwaM
20:03.14*** join/#asterisk CunningPike (~CunningPi@204.239.8.157)
20:03.30elielmmm
20:04.02x-demoni does not see any errors
20:04.48x-demonalso i can read file from asterisk user
20:05.05pabelangerx-demon: pb your asterisk.conf file
20:05.14elielx-demon: ls -al /var/lib/asterisk/documentation/
20:05.29*** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
20:05.43x-demoneliel, it's not in /var, it's in /usr/share
20:05.48pabelangereliel: /usr/share/asterisk/documentation/ :)
20:05.52pabelangerunder debian
20:06.01elielok, so cat /etc/asterisk/asterick.conf |grep astdatadir
20:06.22x-demon_damn_
20:07.06x-demonone second, i must check..
20:07.30elielpabelanger: good, i have never installed asterisk from packages :S
20:08.18x-demoneliel, changed to /usr/share/asterisk
20:08.21x-demonthanks
20:08.28pabelangereliel: Ya, need to ask tzafrir why Debian like /usr/share for data.
20:08.57elielx-demon: good :)
20:09.54x-demonby the way, if i have user 1000, and want this user to be reachable also via "me", i just should setup alias=?
20:13.35x-demonor i can just leave user id as "me" ?
20:16.03p3nguinWhat is this "user id" you're talking about?
20:16.25p3nguineliel: And don't use cat if you only need to grep.
20:16.56p3nguin"grep astdatadir /etc/asterisk/asterick.conf" is enough.
20:17.05x-demonp3nguin, what i put inside []
20:17.23elielp3nguin: i write faster cat /etc/asterisk/asterisk.conf |grep something, i am used to.. :D
20:17.38elielbut you are right
20:17.43p3nguinYou can't write more characters is a less amount of time.
20:20.22chuckfbut p3nguin the way you typed it won't work anyway
20:20.29chuckf:)
20:21.09eliellol
20:22.34[TK]D-Fenderchecout time, BBIAB
20:24.39*** join/#asterisk rustyclarkson (~rusty@u53.sutus.com)
20:29.07*** join/#asterisk cusco (~trilili@213.141.21.122)
20:29.10cuscohi
20:30.00cuscoIm trying to use SMS() app, but for some reason it fails to go trough, call is hangup from telco I guess... could ssomeone take a look at the span debug? --> http://paste.debian.net/78555/
20:34.29rustyclarksonHello, I want multiple queues to have their own unique MOH and then when the call is answered by an agent, if that agents puts the caller on hold, i would like a secondary MOH class to play dependent on the queue they've come from. I'm thinking the only way to do this is using the macro option in the Queue() command which I can use to Set(CHANNEL(musicclass)=...) when the call is answered. Does this sound about right or is there a better way?
20:35.18*** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-77-81.home.otenet.gr)
20:35.39cusco<PROTECTED>
20:35.45cuscothat should be the cause?!
20:39.35pabelangerrustyclarkson: Sounds right
20:42.14paulcMeetme: I have a SIP phone connected to a Meetme conf, listen only. I drop a call file that connects to that meetme via a Local channel, then connects to another Local channel that uses Playback to insert audio into that conference. DAHDI, no hardware. The audio stops after a while (could be seconds, could be minutes) but the channels are all still up/active. Any ideas what's going on?
20:48.43*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)
20:50.28x-demonthere is so much to learn about asterisk
20:50.47x-demonat least i now know basics and i have personal pbx
21:00.13*** join/#asterisk hfb (~hfb@pool-98-112-239-44.lsanca.dsl-w.verizon.net)
21:00.57radicwhat's chanmode c and r here?
21:01.25t_dot_zillawhat is the default attended transfer key?
21:04.16*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:09.31rustyclarksont_dot_zilla: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
21:09.39t_dot_zillathanks
21:10.56[TK]D-FenderFamiliar looking link
21:11.11[TK]D-FenderAlmost Like I provided it almost a half a dozen times before...
21:12.37*** join/#asterisk jeffrey (~zabbix@unaffiliated/Jeffrey)
21:12.40*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)
21:12.50*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
21:13.09[sr]the XML phonebook the SIP phones use, it's a standard or each trademark has its own format?
21:13.20jeffreyso my transfer buttons on my hardphone do not work when I use IAX2 instead of SIP
21:13.24jeffreyany clues?
21:14.48*** join/#asterisk nicoAMG (~nicoamg@201.237.49.131)
21:15.57[TK]D-Fender[17:13]<[sr]>the XML phonebook the SIP phones use, it's a standard or each trademark has its own format? <- every phone that even lets you store one, or access it from outside the phone has its own way.  No standards, many not even XML-like
21:15.57[TK]D-Fenderjeffrey: Your hardphone supports SIP & IAX2?
21:15.57jeffreyyes
21:16.02[TK]D-FenderjeffWhich?
21:16.10jeffreyCitel C-4110
21:16.13[TK]D-Fendersmells ATCOM crap
21:16.36[sr][TK]D-Fender: i'd like to have some example, i dont see that on my phone manual
21:16.53[sr][TK]D-Fender: or could i just make a xml file with two fields, on for name and other for number ?
21:17.07[TK]D-Fenderjeffrey: It should.  Check your manuals
21:17.14*** join/#asterisk freeedrich| (~eeePC@hansaserver.de)
21:17.18p3nguinchuckf: I just copied the information that he offered to the other person, but I used proper syntax.  Any problem with that, take it up with a more appropriate complaint department.
21:17.30[TK]D-Fenderjeffrey: And contact the manufacturer.  Noone I know would come within 10' of one
21:17.52[TK]D-Fender[sr]: And you haven't told me what you're USING <-------
21:17.55jeffreyyeah, unfortunately i don't make the purchasing decisions
21:17.56*** join/#asterisk freeedrich| (~eeePC@hansaserver.de)
21:18.32[sr][TK]D-Fender: sorry, will use remote url
21:25.14*** join/#asterisk cusco (~trilili@213.141.21.122)
21:25.31*** join/#asterisk thevoke (michiel@future.as3322.net)
21:26.20thevokegreetings, i'm working with asterisk manager api using http, and for some reasons i cannot get the variables in asterisk 1.6, do I need a special way of putting the global variables to make them visible?
21:31.24[TK]D-Fenderthevoke: Show us how you're setting them, and how you're trying to retrieve them
21:31.27[TK]D-Fender~pb
21:31.28infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
21:31.29[TK]D-Fender^^^^
21:32.16*** join/#asterisk miamiseb (~deigo@208.76.35.132)
21:34.36thevokeok
21:34.45thevokei'm trying quite a few methods now
21:34.51thevokeexten => 28030450,3,Set(lang=FR,g)
21:35.06miamisebHaving a problem on cisco 7960 that won't register. I double checked the line name/auth name/password and I'm still getting 401 unauthorized.
21:35.14miamisebsip debugs are at http://pastebin.com/3KKzgLdn
21:35.21thevokeand i'm reading it trough the asterisk AstConMan .net feature
21:36.02*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)
21:36.12*** part/#asterisk knctrnl (~aembrey@76.164.169.130)
21:36.14miamisebNone cisco devices (SPAs, polycom, aastra) seem to work fine.
21:36.21miamisebs/None/non/
21:38.10miamisebtried toggling qualify on and off.
21:39.30miamisebno way to actually see what it's sending as the password to match it up?
21:39.31[TK]D-Fenderthevoke: Not the right way.  GLOBAL()
21:39.47*** join/#asterisk sekil (~sekil@78.24.111.218)
21:39.53thevokeSet(GLOBAL(LANG=FR))
21:39.54thevokelike this ?
21:40.22[TK]D-Fenderthevoke: Close, but no
21:40.37thevokehow then? ;)
21:41.02[TK]D-Fenderthevoke: you don't have 2 things on either side of an "='.  Go read up on your FUNCTION basics
21:41.49miamisebhttp://www.the-asterisk-book.com/unstable/funktionen-global.html
21:41.59cuscoIm trying to use SMS() app, but for some reason it fails to go trough, call is hangup from telco I guess... could ssomeone take a look at the span debug? --> http://paste.debian.net/78555/
21:42.05cusco<PROTECTED>
21:42.11cuscothat would be the cause?
21:42.17miamisebaccording to ^ it would be Set(GLOBAL(LANG)=FR)
21:45.47thevokethats what i tried, however no variables in the channel or the linked channel
21:46.52cuscoanother question... while in queue, I hear the MOH if I produce noise on the microphone, if I stop producing noise on the microphone I stop hear MOH also...
21:47.05cuscolike asterisk is only sending me audio if he gets audio from me too
21:47.09cuscohow can I make it work?
21:52.34miamisebcusco: 1.4 or 1.6?
21:52.58*** join/#asterisk zeeesh (zeeesh@119.154.61.93)
21:53.08zeeeshhello everybody
21:53.27miamisebhttps://issues.asterisk.org/view.php?id=5374
21:53.41[TK]D-Fender[17:42]<miamiseb>according to ^ it would be Set(GLOBAL(LANG)=FR) <- Case errors
21:54.46miamisebthe caps for the variable name or what? A string should be able to be either case, and global is capitalized in the doc I was referring to.
21:55.49*** join/#asterisk JAMMAN2110 (~James@unaffiliated/jamman2110)
21:56.57cuscomiamiseb: .16
21:56.59cusco1.6
21:57.39zeeeshanybody has any idea about this feature. "call center setup, a person calling to call center, agents attend the call, supervisor is listening both conversation, is there any feature, that if supervisor give some instructions to his agent then only agent able to hear supervisor voice and the person who is calling at support center unable to hear supervisor voice"?
22:00.00[TK]D-Fenderzeeesh: chanspy <-
22:00.24zeeeshis that possible with asterisk ?
22:01.05spenguin[work]zeeesh: ofcourse!
22:01.16zeeeshoooh great
22:01.43zeeeshthanx bros... let me check
22:03.22*** join/#asterisk MiserySoft (~Lee@nat66.mia.three.co.uk)
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22:12.25*** join/#asterisk Guest24362 (mw3@88.151.97.220)
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22:18.21*** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca)
22:19.09citywokI frequently have ghost calls in MeetMe that have been running for a long time even though the channel is gone.  Any idea why this is happening?
22:19.51citywokSometimes the next time a person makes a call from their desk phone rather than make that phone call, they get put back in the conference that their prior call was stuck on. Hanging up and redialing repeats the same thing.  only way to get rid of it is soft hangup the sip channel.
22:21.50[TK]D-Fendercitywok: * begs to differ
22:22.20citywok?
22:29.25*** join/#asterisk Blackgibson (~inconnu@S01060015e912f559.vs.shawcable.net)
22:30.17*** join/#asterisk fas3r (~fas3r@90.25.broadband12.iol.cz)
22:30.19fas3rhello
22:30.36*** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc)
22:30.51booduhello
22:35.46fas3ri try to use my provider sip account services with asterisk. this is my conf file : http://pastebin.com/VDi8y7WR
22:36.51fas3rxlite register on asterisk with no problem but when i try to call outside i got this message : [Jun 24 00:29:02] NOTICE[17645]: chan_sip.c:14441 handle_request_invite: Call from 'fas3r' to extension '9555XXXXX' rejected because extension not found.
22:37.05*** join/#asterisk kotp (~vgoff@96.2.187.67)
22:37.06fas3rif you can help please ...
22:37.29*** join/#asterisk WintermeW (~clement@78.251.244.244)
22:39.25*** join/#asterisk seanjohn (~seanjohn@ns1.sheltoncomputers.com)
22:39.37WintermeWhi guys...well the channel is not very suited for my question but i can't find find any help. i need some people using asterisk-flite to do tts with their SIP servers , in order to know if flite supports SAPI 4/5 voices
22:40.05seanjohni'm using tos=0x18 in my sip.conf; now asterisk is complaining it's deprecated. What should I be using?
22:43.16Chainsawseanjohn: 42!
22:43.29*** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com)
22:43.35Chainsawseanjohn: On a more serious note: lowdelay, throughput, reliability, mincost or none
22:46.09*** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com)
22:46.44*** join/#asterisk fas3r (~fas3r@90.25.broadband12.iol.cz)
22:50.07seanjohnChainsaw: I don't get what you mean
22:50.24seanjohni Know I don't just type lowdelay in sip.conf
22:51.22Chainsawseanjohn: The new values are those textual ones. lowdelay, throughput, realibility, mincost or none. Specifying the bitfield by hand is what is deprecated.
22:51.28Chainsawseanjohn: And 42 is just a number I like.
22:51.46seanjohntos=lowdelay ??
22:53.47BlackgibsonI have a Linksys SPA3102 that is trying to register with the wrong extension. If im reading the log correctly, it is trying to use ext 100, which does not exist. how can I change that?
22:54.25Chainsawseanjohn: That sounds appropriate, yes.
22:54.43seanjohndidn't work, the entire tos= is deprecated
22:55.40seanjohntos_sip=cs3                    ; Sets TOS for SIP packets.
22:55.41seanjohn<PROTECTED>
22:55.41seanjohn<PROTECTED>
22:57.53seanjohnthose worked; I don't use h323 or any video
22:58.00Chainsawseanjohn: Ah, okay.
22:58.37seanjohni thought tos= was working for the last 4 years
22:58.40seanjohnlol
22:59.10seanjohni had the firewall marking the packets to be prioritized but asterisk wasn't marking them
23:03.00*** join/#asterisk jks (~jks@193.189.93.254)
23:03.17Chainsawseanjohn: I hope your performance increases :)
23:25.17*** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com)
23:27.24sputnickHi again. I use Asterisk 1.4.21.2 on Debian Lenny in a remote dedicated server with only SIP protocol, and ekiga3 as SIP client on my box@home with forwarded UDP ports 8000,5000-5100 with asterisk realms. I can call anybody with french phone or mobile phones via "freephonie", I listen to him, but I cannot talk to him. Any hint ?
23:30.52idespinnersputnick, check localnet, externip in sip.conf....
23:33.04sputnickthanks idespinner, I take a look
23:38.38sputnickidespinner: but I don't have any local non routables ip/vlan on my dedicated server. Only 1 interface with external IP.
23:51.47sputnickidespinner: I've tested with my local IP range ( box@home ) for "localnet=192.168.0.0/255.255.255.0" and my public adress for "externip=xx.xx.xx.xx" and test with "nat=yes" or "nat=no" but now I can't communicate on both sides.
23:58.16*** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa)
23:58.32a1fahey, whats a good hosted pbx that can take your number and provides iax or sip trunk?
23:59.18sputnickbrb

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