| 00:02.26 | *** join/#asterisk raphWRKN (n=raph@203.63.223.17) |
| 00:02.53 | raphWRKN | anyone here able to help me with iax issues? |
| 00:03.57 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 00:05.52 | *** join/#asterisk ipso (n=ipso@S010600a0d1b92a08.ok.shawcable.net) |
| 00:05.59 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 00:08.44 | raphWRKN | trying to get diax to register from an external ip through nat |
| 00:08.48 | raphWRKN | and having 0 joy |
| 00:09.08 | Omer | whats diax? |
| 00:09.57 | raphWRKN | soft iax phone |
| 00:09.59 | raphWRKN | freeware |
| 00:10.03 | *** join/#asterisk Newbie___ (n=Newbie__@218.111.68.114) |
| 00:10.59 | raphWRKN | i must be doing something wrong in iax.conf, but cant work out what |
| 00:12.30 | riddlebox | raphWRKN, you setup the iax user right? |
| 00:15.57 | Newbie___ | hi, can a single Te110P support 1 adit600 with 48 FXS / |
| 00:16.27 | aptura | raphWRKN let me see it. |
| 00:16.57 | *** join/#asterisk OneBinary (n=d@64.161.217.18) |
| 00:17.24 | OneBinary | when trying to login to a queue, I'm getting the following. any ideas on what this means? |
| 00:17.24 | OneBinary | Dec 6 16:05:43 WARNING[19702]: chan_agent.c:1849 __login_exec: Extension '364' is not valid for automatic login of agent '364' |
| 00:17.32 | JT | Newbie___: no |
| 00:17.35 | raphWRKN | riddlebox: afaik |
| 00:17.48 | JT | 48 FXS is more than 1 E1 or T1 worth of channels |
| 00:18.09 | riddlebox | itsa ok |
| 00:18.22 | aptura | raphWRKN what is the issue |
| 00:18.41 | Newbie___ | JT: another word i need dual T1 then |
| 00:18.43 | raphWRKN | aptura: its not registering |
| 00:18.46 | raphWRKN | so cant call out |
| 00:19.00 | *** join/#asterisk h3x (n=hex@64.192.116.17) |
| 00:19.29 | JT | Newbie___: i have no idea how the adit600 connects, but dual T1 sounds right, as a T1 is 24 channels in CAS mode |
| 00:19.31 | raphWRKN | entry in iax.conf |
| 00:19.38 | JT | so 2 * 24 = 48 |
| 00:19.40 | raphWRKN | [6999] |
| 00:19.41 | raphWRKN | type=friend |
| 00:19.42 | raphWRKN | host=dynamic |
| 00:19.43 | raphWRKN | username=6999 |
| 00:19.44 | raphWRKN | callerid=6999 |
| 00:19.46 | aptura | raphWRKN okay then do you have and not here! |
| 00:19.49 | raphWRKN | peercontext=local |
| 00:19.50 | raphWRKN | auth=plaintext,md5,rsa |
| 00:19.51 | raphWRKN | secret=6999 |
| 00:19.52 | raphWRKN | context=local |
| 00:19.53 | aptura | send it to pastebin.ca |
| 00:19.53 | raphWRKN | notransfer=yes |
| 00:19.54 | raphWRKN | qualify=yes |
| 00:19.55 | aptura | stop |
| 00:19.55 | raphWRKN | disallow=all |
| 00:19.56 | raphWRKN | allow=alaw |
| 00:19.57 | raphWRKN | permit=0.0.0.0/0.0.0.0 |
| 00:20.00 | raphWRKN | aah sorry |
| 00:20.01 | JT | raphWRKN: stop that |
| 00:20.06 | OneBinary | raphWRKN: try changing qualify to no |
| 00:20.16 | JT | ~pb |
| 00:20.18 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 00:20.21 | Newbie___ | JT: never seen a adit 600 before, sure hope its got 2 T1 plug |
| 00:20.25 | OneBinary | i had a similar problem until i turned quality to no |
| 00:20.41 | JT | Newbie___: well it'd be a good idea to do some technical research before buying one |
| 00:21.07 | JT | the TE110P is only a single T1/E1 card |
| 00:21.43 | raphWRKN | hmmm still no joy... |
| 00:21.59 | raphWRKN | was trying to get it going through sip, got connection but only 1 way audio |
| 00:22.03 | raphWRKN | nat sucks for that |
| 00:22.03 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 00:23.22 | OneBinary | anybody know why i can't login to a queue? WARNING[19702]: chan_agent.c:1849 __login_exec: Extension '364' is not valid for automatic login of agent '364' |
| 00:28.40 | te_lo_meto_mami | yo raphWRKN your lines that you put in iax.conf raphWRKNdisallow=all |
| 00:28.41 | te_lo_meto_mami | raphWRKNallow=alaw con i use the codec g729 allow only that? can providers support any codec that a peer setsup in their config? |
| 00:28.42 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 00:29.20 | raphWRKN | yah you can use only g729 but you need a valid license for it |
| 00:29.39 | te_lo_meto_mami | you know the cost off the top? |
| 00:29.50 | BlepsoaF | anyone ever have any issues with DTMF not working when enabling automon in features.conf & Ww in the dial cmd |
| 00:29.52 | raphWRKN | nah, something like US$10 a channel i think |
| 00:29.56 | JT | USD$10 or so |
| 00:30.07 | JT | per transcode session, not channel |
| 00:30.11 | te_lo_meto_mami | ahh ill find the link thanks |
| 00:30.16 | raphWRKN | i own 50 of them, but they are all in use, and its not part of my issue |
| 00:30.26 | JT | sometimes it's possible to to use more than one transcoding session per call |
| 00:30.39 | JT | te_lo_meto_mami: digium.com |
| 00:30.47 | te_lo_meto_mami | i know , im a newbie also just thought you coudl shed some light |
| 00:30.49 | te_lo_meto_mami | thanks |
| 00:30.52 | te_lo_meto_mami | Jt thanks |
| 00:31.07 | te_lo_meto_mami | wait i dont have a digium card |
| 00:31.11 | te_lo_meto_mami | voip only |
| 00:31.17 | te_lo_meto_mami | still need it right? |
| 00:31.25 | te_lo_meto_mami | digium is just a source |
| 00:31.40 | te_lo_meto_mami | codec im talkking about |
| 00:31.51 | JT | digium sell the codec licences |
| 00:31.54 | te_lo_meto_mami | k |
| 00:31.58 | JT | digium write asterisk |
| 00:32.00 | JT | mainly |
| 00:32.12 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 00:32.15 | te_lo_meto_mami | i love digium |
| 00:32.28 | te_lo_meto_mami | man i just dove into asterisk and i learned so much in two days |
| 00:32.34 | te_lo_meto_mami | i feel like a crack head |
| 00:32.47 | JT | yes it is a bit insane |
| 00:33.04 | JT | i wish asterisk had inbuilt stable bri support |
| 00:33.04 | te_lo_meto_mami | its like neverending the things you can do with it |
| 00:33.17 | te_lo_meto_mami | you not in North america? |
| 00:33.21 | JT | yeah it's fairly flexible |
| 00:33.23 | JT | i'm not |
| 00:33.27 | te_lo_meto_mami | figured |
| 00:33.31 | mosty | does asterisk support array variables? |
| 00:33.32 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 00:33.46 | JT | arrays, i don't think so |
| 00:33.47 | mosty | or variable arrays, if you prefer |
| 00:33.55 | mosty | d'oh |
| 00:34.00 | JT | it has ast db |
| 00:34.05 | JT | which is key:value |
| 00:34.14 | te_lo_meto_mami | like a registry in windows? |
| 00:34.16 | te_lo_meto_mami | kinda |
| 00:34.18 | JT | and realtime, which allows you to use an rdbms |
| 00:34.18 | te_lo_meto_mami | ? |
| 00:34.26 | JT | the ast db is similar to a registry |
| 00:34.33 | te_lo_meto_mami | yuk windows by the way |
| 00:34.36 | JT | non-volatile db |
| 00:34.42 | mosty | i would prefer not to use realtime just for this |
| 00:34.50 | mosty | astdb might be ok |
| 00:34.57 | JT | what do you need an array for? |
| 00:35.01 | BlepsoaF | just use agi |
| 00:35.07 | JT | it has global variables as well as channel variables |
| 00:35.25 | BlepsoaF | then use any programming lanuage you want |
| 00:35.34 | BlepsoaF | language |
| 00:35.49 | mosty | i want a series of mappings from DID to local extension, so i can simplify one of my contexts |
| 00:35.58 | JT | we don't even know what he wants to do yet |
| 00:36.08 | JT | agi introduces performance penalties though |
| 00:36.11 | JT | hrm |
| 00:36.17 | BlepsoaF | JT: true, but I do all my dialplan stuff in AGI |
| 00:36.28 | BlepsoaF | easier to build a framework and go nuts |
| 00:36.36 | mosty | i would prefer not to use AGI in this case too |
| 00:36.37 | JT | you absolutely don't want did mappings in the dialplan? |
| 00:36.56 | JT | BlepsoaF: sounds like unnecessary overhead for most stuff |
| 00:37.40 | BlepsoaF | re-phrase -> I do most IVR stuff in AGI |
| 00:37.48 | mosty | i want did mappings in the dialplan in one place (i was hoping for an array), then i could write one set of rules to route each of the did's, instead of the same thing for every did |
| 00:38.00 | JT | mosty: macros? |
| 00:38.08 | BlepsoaF | macros would be the way to go |
| 00:38.32 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.169.68) |
| 00:38.41 | BlepsoaF | anyone ever have any issues with DTMF not working when enabling automon in features.conf & Ww in the dial cmd |
| 00:39.02 | BlepsoaF | sounds like an asterisk media path thing to me, but I dont know why |
| 00:40.15 | Newbie___ | JT: thank you |
| 00:40.17 | mosty | jt, hrm let me look at that then |
| 00:40.36 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:41.18 | JT | mosty: read the book? |
| 00:42.03 | mosty | the o'reilly one? i skimmed through it a while back |
| 00:42.20 | JT | it mentions all these major principles like macros |
| 00:44.33 | aptura | has asterisk and linux been installed on single board computers before? |
| 00:46.02 | mosty | yes but transcoding is painful |
| 00:46.03 | JT | a lot of people have done that sort of stuff |
| 00:46.18 | aptura | JT, was that for me? |
| 00:46.24 | JT | yes |
| 00:46.39 | Un1x | Hye for Sangoma cards |
| 00:46.47 | Un1x | are the green modules FXS and red FXO |
| 00:46.52 | Un1x | there opposite of Digium? |
| 00:47.56 | Un1x | yes they are just looked at handbook |
| 00:47.56 | Un1x | :P |
| 00:49.29 | aptura | very very very odd. My asterisk box called my extension using my CID name. |
| 00:51.07 | [hC] | anyone know why my polycom 601 w/ 3 expasion modules might be rebooting? |
| 00:51.10 | [hC] | on its own |
| 00:51.31 | wunderkin | [hC], ive been having rebooting problems too but i have a 430 |
| 00:51.31 | aptura | my asterisk box just called me on its own figure that one out :) |
| 00:52.01 | mosty | jt: is is possible to take the value of a variable and use that as a variable name? eg something like ${$foo} ? |
| 00:53.08 | JT | sure |
| 00:53.14 | *** join/#asterisk Phoenix^ (n=Phoenix@phoenixphire.org) |
| 00:53.32 | mosty | in that case i can emulate arrays |
| 00:53.51 | mosty | $hash_${keyname} |
| 00:53.59 | mosty | i set that to $value |
| 00:54.18 | JT | why do you need to do that? |
| 00:54.36 | mosty | i want a map from did to sip extension |
| 00:54.47 | [hC] | wunderkin: any ideas? |
| 00:55.07 | JT | mosty: i have no idea why you'd do it like that |
| 00:56.10 | mosty | then i can have a context which tries to do something like this: $did,1,Dial($hash_${did}) ; $did,1,Goto(voicemail,$hash_${did},1) |
| 00:56.24 | mosty | er, the second one would be priority 2 |
| 00:56.35 | tessier | Anyone know why calls would get stuck with channels open even though nobody is actually on the phone? |
| 00:56.45 | mosty | tessier, what kind of channels? |
| 00:56.50 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-110-113.dsl.tulsok.swbell.net) |
| 00:57.22 | mosty | jt: i have quite a few did's, and i would like this mapping defined in a single place in the dialplan |
| 00:57.23 | tessier | mosty: SIP channels |
| 00:57.33 | tessier | mosty: This has been a problem in my callcenter for a couple of months. |
| 00:57.35 | JT | mosty: why wouldn't you use ast db or global variables? |
| 00:57.57 | tessier | Every now and then someone will stop getting calls. It's because there is a channel stuck. Even if the phone has been rebooted. Sometimes they will have a number of channels stuck open. |
| 00:58.11 | tessier | Mostly channels going into an ivr I built but occasionally other things. |
| 00:58.32 | mosty | jt: the natural way of expressing this map in most languages is to use an array, i want to use a global array (or something similar) |
| 00:58.34 | tessier | Seems like there should be a timeout of some sort that would kill them. |
| 00:58.52 | JT | mosty: there is no arrays, as far as i know |
| 00:59.02 | JT | so you'll have to do it in some key:value format |
| 00:59.04 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:59.04 | mosty | tessier: look at the t extension |
| 00:59.08 | tessier | I wonder if I should put the timeout() application in my dialplan somewhere... |
| 00:59.12 | JT | i fail to see why that's a problem |
| 00:59.21 | tessier | mosty: Isn't that only for when the user doesn't enter a selection in an ivr? |
| 00:59.22 | JT | otherwise you could use realtime |
| 00:59.47 | mosty | jt: yes, i will use a format like that, but it would be simpler if asterisk simply supported arrays |
| 01:00.10 | JT | mosty: i guess realtime is intended for that sort of stuff |
| 01:00.26 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 01:00.30 | mosty | tessier: it would help for the ivr case. where else do your zombie calls live? |
| 01:00.47 | mosty | jt: i suppose, but i am not a fan of realtime |
| 01:00.53 | tessier | hmm...there is no t extension in that ivr... |
| 01:00.57 | mosty | it's overkill for this case |
| 01:01.00 | tessier | Hopefully that will fix it. |
| 01:01.02 | mceGEEK | what is provisioning ? |
| 01:01.06 | tessier | I actually looked at that before and have t extensions elsewhere. |
| 01:01.17 | tessier | But I didn't realize this particular IVR didn't have one |
| 01:01.32 | *** join/#asterisk emphyrio (n=stryfe@dsl254-076-201.nyc1.dsl.speakeasy.net) |
| 01:02.15 | tessier | hmm...but it already does seem to be timing out by itself. |
| 01:03.35 | tessier | SIP/141-b7b9fa20 1@company-outbound-d Up DISA(no-password|drjays-outgoi |
| 01:03.42 | tessier | That is a typical hung channel. |
| 01:05.22 | Phoenix^ | Does anyone know of an IP phone that has an 802.1x supplicant built in? |
| 01:06.38 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:07.14 | Un1x | tessier you doing DISA? |
| 01:08.15 | tessier | Un1x: Yes |
| 01:09.43 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 01:09.53 | JT | mosty: your problem isn't that hard |
| 01:10.03 | JT | just use a pattern that will match all DIDs |
| 01:10.17 | Un1x | tessier i'm trying to do somethin similar but, have no luck with it yet... |
| 01:11.16 | JT | and have either that extension, or a macro it calls do something like Dial(Blah/DID_${EXTEN}) |
| 01:11.40 | JT | and have global variables setup with DID_<DID number> |
| 01:11.44 | JT | or you could use ast db |
| 01:12.12 | JT | as ${DID} as the extension name would not work |
| 01:12.29 | mosty | i'm using global vars of the form DIDtoSIP_<DID>=<SIP> |
| 01:12.37 | mosty | already halfway done |
| 01:12.56 | JT | yeah so you only need one pattern in your dialplan to catch it most likely |
| 01:13.13 | JT | it's easy if all the DIDs are the same number of digits |
| 01:13.52 | mosty | yes |
| 01:13.52 | *** join/#asterisk ManxPower (n=manxpowe@73.sub-70-216-51.myvzw.com) |
| 01:14.04 | mosty | otherwise i would need lots of extra contexts |
| 01:14.18 | JT | or a matchall ;) |
| 01:14.30 | JT | i'm doing something similar |
| 01:14.40 | mosty | in that case i would use an sql database, and either generate dialplan files or use realtime |
| 01:14.41 | JT | except i just retransmit the DID/MSN |
| 01:14.47 | JT | as it interfaces to an isdn pabx |
| 01:15.10 | *** part/#asterisk raphWRKN (n=raph@203.63.223.17) |
| 01:15.13 | JT | i don't see why you'd need to, just due to differing amounts of digits |
| 01:18.03 | DrCron | man, it seems all the cool voip->dect stuff is made in china, and they cant write up the specs for crap |
| 01:18.19 | DrCron | and the manuals are even worse |
| 01:18.30 | mosty | jt: it helps in terms of management, when i have encountered that in the past |
| 01:18.38 | ManxPower | The worst phone I ever used is one of the USA DECT phones. |
| 01:18.44 | *** join/#asterisk alamantia (n=alamanti@72.146.23.242) |
| 01:19.14 | mosty | i write a script which queries the db, sorts by prefix length, creates contexts in decreasing length of prefix length and generates the contexts in a file which is #included by extensions.conf |
| 01:19.32 | DrCron | i just want a system where each dect phone has a sip address, and can register to asterisk |
| 01:20.06 | De_Mon | Im trying to get asterisk and exchange 2007 to talk to eachother, anyone had any success? |
| 01:20.24 | JT | mosty: is that more due to the volume of dids? because the exact same solution for fixed length numbers would be applicable to variable length numbers |
| 01:20.46 | JT | talk to each other... i think you will need to be more specific |
| 01:20.51 | De_Mon | I've got openser on the asterisk server on port 5061, and so far things are not going very well |
| 01:20.53 | ManxPower | De_Mon: I didn't know Exchange talked SIP |
| 01:21.08 | De_Mon | it's called Unified Messaging |
| 01:21.09 | ManxPower | De_Mon: Does it work if they are on different servers? |
| 01:21.27 | mosty | jt: yes. i had a huge number of dids in that case, it was a fairly complex dialplan |
| 01:21.27 | De_Mon | duno, don't have another server to find out with |
| 01:21.38 | ManxPower | De_Mon: Oh! Like Cold Fusion, the tooth fairy, and pepertual motion machines! |
| 01:22.11 | De_Mon | ManxPower no, thats just what ms calls their autoatendent and phone features |
| 01:22.19 | ManxPower | De_Mon: What specific problem are you hving? |
| 01:23.18 | De_Mon | exchange 2007 only supports TCP, so Ive got openser proxying the protocols and it's not working... |
| 01:23.49 | De_Mon | <PROTECTED> |
| 01:24.28 | De_Mon | .225 is the asterisk server, so something is amiss |
| 01:25.59 | ManxPower | De_Mon: 302 is call has been forwarded. |
| 01:25.59 | ManxPower | De_Mon: good luck with it. |
| 01:25.59 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-44.mtco.com) |
| 01:26.11 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 01:27.57 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:28.03 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:30.04 | De_Mon | yeah, thats why I was looking for someone with some previous experience in the area |
| 01:31.00 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 01:31.02 | djflux | since Exchange 2007 is fairly new I doubt you're going to find someone with experience :) |
| 01:31.30 | ManxPower | I have a new policy. Anyone with questions on Exchange goes on the ignore list. |
| 01:32.00 | djflux | wow ... harshness |
| 01:33.18 | ManxPower | djflux: GAIM's /ignore seems to only be for the current session. |
| 01:34.04 | djflux | hrm ... you use GAIM for IRC? |
| 01:34.38 | djflux | I tried it for a bit and wasn't fond of it ... went back to xchat |
| 01:35.01 | DrCron | hmm, what would you guys (and gals) sugest for a wireless phone interface to asterisk? |
| 01:36.55 | ManxPower | DrCron: an ATA with a 900Mhz DSS |
| 01:39.04 | ManxPower | 900Mhz DSS is next to impossible to find, but if you need range and have lots of plants around it's the only way to do. |
| 01:39.21 | ManxPower | The last one we had reached almost 1/2 a mile. |
| 01:40.57 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:41.25 | DrCron | the dect handset we had last (base just died) reached nearly that, with buildings in the way no less |
| 01:41.48 | DrCron | and i like the fact there are dect handsets with bluetooth |
| 01:44.59 | djflux | De_Mon: check out http://bugs.digium.com/view.php?id=4903 ... looks like there is some TCP SIP support but you have to patch * |
| 01:45.05 | djflux | now I'm REALLY going home |
| 01:47.44 | ManxPower | DrCron: you must not be in the USA |
| 01:52.20 | DrCron | is dect on a diffrent freq here? |
| 01:52.40 | DrCron | i was using a gigaset 2400 |
| 01:52.46 | De_Mon | djflux I saw it, it's still rather experimental though |
| 01:52.54 | ManxPower | In the usa it is in the 1900 range |
| 01:53.25 | ManxPower | I imagine it COULD be in the 900 and 2.4ghz rane as those are unlicenses. |
| 01:53.26 | De_Mon | im loosly following: http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Integration&view_comment_id=12495 |
| 01:53.42 | DrCron | in the US and in europe its in the 1900mhz band |
| 01:54.02 | ManxPower | But all the "DECT 6.0" phones I've seen were 1900 range, it is a new unlicensed range in the USA |
| 01:54.50 | ManxPower | They got like 300 ft and that was it. |
| 01:55.07 | ManxPower | I really witch I could find a non-Sony 900Mhz DSS phone. |
| 01:55.32 | ManxPower | The Sony one we had was AWESOME, but kept dropping calls even if you were near the base. |
| 01:55.55 | DrCron | huh, well the seimens phone we had rocked |
| 01:56.03 | DrCron | and it was dect |
| 01:56.11 | ManxPower | DrCron: Usa or Euroland? |
| 01:56.13 | De_Mon | is there a simple test I could perform using telnet or something to check my connections? |
| 01:56.28 | DrCron | um, it was a USA model |
| 01:56.37 | De_Mon | Im mostly interested in the CLIENT -> OpenSER -> Asterisk path |
| 01:57.03 | De_Mon | (tcp)CLIENT -> OpenSER ... |
| 01:57.19 | ManxPower | We have almost 50 acers to cover |
| 01:57.32 | ManxPower | prolly 40 actually need coverage |
| 01:57.43 | DrCron | open? |
| 01:58.14 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 01:58.18 | ManxPower | hell no. |
| 01:58.24 | ManxPower | hills, woods |
| 01:58.38 | ManxPower | the 900Mhz covered most of the 40 acers |
| 01:58.52 | ManxPower | pretty much a 2.4Ghz phone's worst nightmare. |
| 01:59.00 | mceGEEK | hi bkw |
| 01:59.00 | DrCron | yhea |
| 01:59.19 | ManxPower | hence my fetish for 900Mhz DSS |
| 02:07.53 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
| 02:08.35 | *** join/#asterisk loldongs (n=kingzork@kbhn-vbrg-sr0-vl204-122.perspektivbredband.net) |
| 02:16.21 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:17.06 | *** join/#asterisk jmacz (n=jmacz@190.24.97.151) |
| 02:24.26 | puzzled | RoyK: ping |
| 02:24.50 | RoyK | s/ping// |
| 02:25.29 | puzzled | RoyK: if I use Slav's JB patch does that mean that there is no JB on SIP-SIP calls? |
| 02:26.03 | RoyK | sip/sip should't be dejittered |
| 02:26.11 | RoyK | dejittering happens at endpoints |
| 02:27.00 | puzzled | RoyK: ah so if the * server is in the middle it basically should just pass the stuff on and leave it to (in my case) the 7961's to take care of dejittering? |
| 02:27.25 | RoyK | yes |
| 02:27.31 | puzzled | got it. thanks |
| 02:27.36 | RoyK | but without the jitterbufferpatch |
| 02:27.41 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 02:27.45 | RoyK | asterisk will rewrite rtp timestamps |
| 02:28.02 | RoyK | which will fuck up any possibility to dejitter the data |
| 02:28.11 | puzzled | right. just what I was thinking |
| 02:28.53 | RoyK | but the jitterbufferpatch includes a little one that passes on the timestamps |
| 02:29.10 | puzzled | any way I can see SIP-ZAP JB stats during a call? |
| 02:29.41 | RoyK | asterisk -gvvvvvc |
| 02:29.57 | RoyK | and then a green line comes up when the jb kicks in |
| 02:30.13 | RoyK | just make sure you enable jb in chan_zap |
| 02:30.27 | puzzled | never seen that one but this is on a local lan with hardly any traffic |
| 02:30.32 | puzzled | the green line I mean |
| 02:31.03 | *** join/#asterisk robeph (n=robf@user-24-236-88-244.knology.net) |
| 02:31.20 | RoyK | puzzled: you'll see it if you configure it correctly |
| 02:31.43 | puzzled | RoyK: do I need to set jb-log=yes? |
| 02:32.20 | RoyK | no. that only creates rubbish in /tmp |
| 02:32.37 | puzzled | lol |
| 02:33.03 | RoyK | good for debugging, but that's it |
| 02:34.46 | *** part/#asterisk loldongs (n=kingzork@kbhn-vbrg-sr0-vl204-122.perspektivbredband.net) |
| 02:35.13 | groogs | Does anyone know anything about chan_bluetooth? the official svn for it seems to be gone, it was last updated in 2005 .. has anyone taken it over? (the 'lastest' snapshop doesn't compile against 1.2.13) |
| 02:35.26 | puzzled | RoyK: ok thanks. will mess with it a bit more |
| 02:35.53 | RoyK | hehe |
| 02:36.21 | RoyK | puzzled: it works, beleive me, we've been using that jb code in production since may or so |
| 02:37.26 | puzzled | RoyK: don't doubt that :) |
| 02:42.42 | Un1x | Hmm, i wonder if i can switch the 2 FXO modules i have in my TDM400P with FXS ones |
| 02:42.50 | Un1x | or if there actualy solderied onto the board |
| 02:42.58 | Qwell | Un1x: They're modules |
| 02:43.18 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.94) |
| 02:55.31 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 02:55.36 | TripleFFFF | anyone use a cisco phone? |
| 02:55.58 | tim27dr | me |
| 02:56.11 | TripleFFFF | you use with multiple providers ? |
| 02:56.17 | TripleFFFF | if so what firmware |
| 02:56.22 | tim27dr | no |
| 02:56.25 | TripleFFFF | coz me its alwyas registering them to #1 |
| 02:56.37 | tim27dr | :( |
| 02:56.45 | TripleFFFF | http://www.freeworlddialup.com/community/forum/viewtopic.php?p=677&sid=5d4508a6259a965a40564b1a63e5cef5 |
| 02:56.51 | TripleFFFF | that user seems to have it to work |
| 02:57.33 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 02:57.41 | tim27dr | why just dont connection your provider to your asterisk box... |
| 02:57.54 | km- | hey, does anyone here happen to know what context I'm supposed to pop outbound calls to on my nufone account? |
| 02:57.58 | TripleFFFF | <PROTECTED> |
| 02:58.00 | TripleFFFF | hmm |
| 02:58.08 | TripleFFFF | coz i got none @ home |
| 02:58.12 | TripleFFFF | installing now lol |
| 02:58.25 | tim27dr | just install asterisk |
| 02:59.16 | tim27dr | trixbox have a great GUI based to config your phones etc |
| 02:59.54 | TripleFFFF | nah |
| 02:59.57 | TripleFFFF | no way |
| 02:59.57 | TripleFFFF | lol |
| 03:00.03 | TripleFFFF | ill install openpbx with a one line |
| 03:00.08 | TripleFFFF | extensions to my phone |
| 03:00.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:01.04 | te_lo_meto_mami | funny newbie mistake |
| 03:01.17 | te_lo_meto_mami | i just wasted 10 dollars on g729 codec |
| 03:01.21 | te_lo_meto_mami | and im using freebsd |
| 03:01.24 | te_lo_meto_mami | grinning |
| 03:01.29 | *** join/#asterisk scropia (n=Scorpia@80.224.14.6) |
| 03:01.41 | scropia | hola |
| 03:01.54 | scropia | tengo un problema |
| 03:02.14 | scropia | algien puede ayudarme? |
| 03:05.22 | mceGEEK | i'm seeing msgs about SIP Express Router |
| 03:05.24 | mceGEEK | what is that |
| 03:06.24 | TripleFFFF | now the hmm show config is cached in linksys ? its not redownloading the config from tftp |
| 03:06.29 | TripleFFFF | but i see tftp serving it |
| 03:06.35 | TripleFFFF | i mean in cisco |
| 03:07.38 | *** join/#asterisk afernandez (n=Ayax@201.230.170.235) |
| 03:09.38 | *** join/#asterisk jmacz (n=jmacz@190.24.97.151) |
| 03:13.10 | mosty | te_lo_meto_mami, do the non-digium g729 modules compile on freebsd? |
| 03:13.44 | mosty | or alternatively, doesn't freebsd have some sort of linux executable compatibility library? |
| 03:16.24 | DrkShdw | yes, it does. |
| 03:23.19 | *** join/#asterisk mhnoyes_ (n=mhnoyes@dialup-4.246.21.1.Dial1.SanJose1.Level3.net) |
| 03:32.54 | *** part/#asterisk afernandez (n=Ayax@201.230.170.235) |
| 03:34.47 | *** part/#asterisk mithraen (n=mithraen@87.228.121.245) |
| 03:34.59 | *** join/#asterisk jorge_ (n=te_lo_me@209.200.133.237) |
| 03:35.02 | *** join/#asterisk mithraen (n=mithraen@87.228.121.245) |
| 03:35.10 | DrCron | why use g729? |
| 03:37.27 | Un1x | Because, its Bandwidth efficient |
| 03:37.28 | [TK]D-Fender | DrCron : Top 2 reasons : Bandwidth, and maybe he has hardware that ONLY uses that codec. |
| 03:37.30 | Un1x | for home users specificly... |
| 03:37.50 | Un1x | because if you have a corproate line you really dont need it unless you wanna cut costs on bandwidth, |
| 03:37.55 | benjk | bandwidth alone is not a good reason to use g729 |
| 03:37.59 | benjk | there's speex |
| 03:38.02 | benjk | its free |
| 03:38.06 | Un1x | but home users id assume would want it if hardware supports it due to it being bandwidth efficient like me :P |
| 03:38.19 | benjk | the only reason to use g729 is if you are forced to use it |
| 03:38.26 | Un1x | i got a 100KB upload so i prefer to use g729 then i can have more calls active as to with ulaw itd be getting 1 call only... |
| 03:38.31 | Un1x | benjk |
| 03:38.34 | DrCron | speek or gsm |
| 03:38.36 | Un1x | you familiar with DISA? |
| 03:38.38 | DrCron | speex |
| 03:38.40 | benjk | speex |
| 03:38.55 | benjk | *CLI> show application disa |
| 03:39.40 | JT | Un1x: there's not much to DISA |
| 03:39.56 | Un1x | JT, well err i got that but the context is whats making me weary |
| 03:40.02 | JT | ? |
| 03:40.09 | Un1x | because it needs a new context |
| 03:40.21 | Un1x | nevermind let me read then i'll know what im talking about :P |
| 03:40.46 | JT | well it needs to look at a context to extension match |
| 03:41.29 | benjk | Un1x, there are also some examples for DISA on Voip-Info.org |
| 03:43.27 | *** join/#asterisk [hC-] (n=hardcore@66.119.167.162) |
| 03:43.35 | *** join/#asterisk [hC] (n=root@69.90.99.195) |
| 03:43.53 | *** join/#asterisk dacleric (n=dacleric@p5482285F.dip0.t-ipconnect.de) |
| 03:44.11 | [hC] | can anyone explain why zttest would return either 100% across the board, or some crappy numbers, differing every time you simply kill and restart zttest? |
| 03:45.01 | JT | do you reboot between attempts? |
| 03:45.06 | [hC] | nopr |
| 03:45.07 | [hC] | nope |
| 03:45.20 | [hC] | its in a dell that also returns /dev/rtc issues upon boot |
| 03:45.29 | [hC] | if that is any indication |
| 03:45.35 | JT | hmm |
| 03:45.39 | JT | weird |
| 03:45.41 | JT | any zap hw? |
| 03:45.52 | [hC] | ya, sangoma a200 |
| 03:46.02 | Un1x | heh sorry guys its not DISA disas got like 4 lines to it its all its context and call routing that scares me because i want disa accissble, only by a person who dails the correct did not the other dids i have |
| 03:47.50 | DrCron | so set up a context for that DID |
| 03:49.10 | [hC] | any ideas? |
| 03:50.02 | Supaplex | [hC]: not involving duct tape and three crazed weasels? |
| 03:50.25 | [hC] | yeah... unfortunately ive only got two weasels in stock.... |
| 03:50.34 | Supaplex | oh ok. |
| 03:50.42 | [TK]D-Fender | Un1x: You'll want to password it as well as seperate by DID. You could alternately do DID + CID. |
| 03:50.48 | Supaplex | and they're only partially crazy. |
| 03:52.08 | *** join/#asterisk Fatty123 (i=sean@gateway/gpg-tor/key-0x9C54163E) |
| 03:52.26 | *** join/#asterisk bmg505 (n=leon@c1-103-9.rndf.isadsl.co.za) |
| 03:56.21 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 04:06.22 | Un1x | D-fender for sure :P |
| 04:08.55 | De_Mon | <-- SIP read from 66.192.107.225:5070: |
| 04:08.55 | De_Mon | OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0 |
| 04:08.55 | De_Mon | Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=fa65c1f0e8> |
| 04:08.55 | De_Mon | FROM: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=fa65c1f0e8 |
| 04:09.09 | De_Mon | ack thankyou irssi |
| 04:09.26 | De_Mon | http://pastebin.ca/270181 |
| 04:09.53 | De_Mon | can someone translate that debug log for me |
| 04:10.26 | De_Mon | It looks like eo-dev.webcode.com is telling Asterisk SIP/2.0 404 Not Found |
| 04:10.51 | De_Mon | but im not sure why asterisk is looking for s in local |
| 04:12.18 | JT | probably to do with sip.conf? |
| 04:13.35 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.mn.comcast.net) |
| 04:13.37 | De_Mon | eo-dev isn't supposed to be doing anything but checking the connection |
| 04:14.04 | De_Mon | IP Peer pbx.elephantoutlook.com did not respond to the PING request. The error code returned is 500 and the error text is I'm terribly sorry, server error occurred (1/SL) |
| 04:14.12 | converx | anyone here can please share the sendmail.mc file? |
| 04:14.51 | De_Mon | converx try #sendmail |
| 04:16.52 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 04:18.13 | DrCron | does asterisk have a module for challenge response authentication? |
| 04:20.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:20.35 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 04:27.30 | JT | authentication for what? |
| 04:36.32 | DrCron | voicemail, that disa thing |
| 04:36.42 | DrCron | when you call in from a remote ptsn terminal |
| 04:39.26 | JT | so what do you mean challenge/response? |
| 04:42.06 | DrCron | um, it gives you a n digit code, you punch it into a device, you get a n digit code back to enter via dtmf |
| 04:42.18 | *** part/#asterisk Phoenix^ (n=Phoenix@phoenixphire.org) |
| 04:42.40 | mosty | drcron: you could write an agi script to do that, if you know what the algorithm is |
| 04:43.23 | JT | yeah it doesn't come with that, a little too difficult for most users |
| 04:44.45 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.12) |
| 04:49.19 | *** join/#asterisk fx0 (n=ident@voip.terrorist.net) |
| 04:50.30 | mosty | if i have the name of a variable $x stored in $y how can i find the value of $x ? |
| 04:51.27 | awannabe | can someone tell me why that doesnt work!?! exten => _XX,1,Set(CALLERID(ALL)=${IF($[ ${EXTEN} = 400]?Foo BAR<12343216554>:Normal Name<1234565555>} |
| 04:55.57 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:59.01 | *** join/#asterisk leprechau (i=leprecha@c-68-53-17-135.hsd1.tn.comcast.net) |
| 05:05.16 | leprechau | can anyone reccomend a good third party PSTN connection service? |
| 05:05.25 | leprechau | are there any 3rd party commercial IAX services? |
| 05:06.49 | leprechau | I have a client that I would like to do an askerisk install for....but they are not interested in bringing in any phone lines....they only have data services |
| 05:07.08 | leprechau | and are currently using aptela ...but would like to have more control themselves |
| 05:10.30 | awannabe | leprechau: teliax does IAX |
| 05:10.38 | awannabe | i use them on a very very small scare with no issues |
| 05:10.48 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 05:12.58 | Un1x | hey |
| 05:13.00 | Un1x | where is a good place |
| 05:13.08 | Un1x | to get 18** dids |
| 05:13.09 | Un1x | ? |
| 05:13.41 | fx0 | black market. |
| 05:14.17 | Qwell | I don't think 8** is valid NANPA |
| 05:14.24 | Qwell | NANP rather |
| 05:14.58 | Un1x | nanpa? |
| 05:15.02 | Qwell | ~nanpa |
| 05:15.04 | jbot | it has been said that nanpa is North America Numbering Plan Administration: an integrated telephone numbering plan serving 19 North American countries that share its resources. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. ... |
| 05:15.04 | Qwell | ~nanp |
| 05:15.13 | *** join/#asterisk wunderkin- (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 05:15.13 | Un1x | :O |
| 05:15.24 | [hC-] | is there a way to boost vmail volume when people listen via handset? |
| 05:15.31 | [hC-] | i have a patch for email |
| 05:15.35 | *** join/#asterisk fx0_ (n=ident@voip.terrorist.net) |
| 05:16.32 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 05:17.45 | leprechau | awannabe, what have you heard about VoIPJet ?? |
| 05:17.56 | awannabe | leprechau: used em in the past, were total crapo |
| 05:18.01 | Un1x | anyone been having problems with voicemail on Slackware? |
| 05:18.04 | fx0_ | nothing but nice things about john |
| 05:18.08 | leprechau | ouch :/ |
| 05:18.10 | Un1x | due to the mpg321 instead of mpg123 |
| 05:18.14 | *** join/#asterisk LoneShadow (n=duh@c-67-180-81-124.hsd1.ca.comcast.net) |
| 05:18.21 | awannabe | Un1x: ive had none yet |
| 05:18.32 | leprechau | awannabe, so your reccomendation would be teliax? |
| 05:18.35 | Un1x | i dont know man i dont get any sound its pissing me off |
| 05:18.40 | Qwell | Why would voicemail be using mpg123? |
| 05:18.50 | Un1x | I dont know ask fx0_ |
| 05:19.12 | leprechau | the current usage load will be about 20-30 total phones and prolly 15 max simultaneous conversations |
| 05:19.27 | leprechau | nothing large |
| 05:19.48 | awannabe | well they charge 2 cents a minute |
| 05:20.32 | awannabe | per channel |
| 05:20.33 | awannabe | ive had good luck with them, had a account for about a year or so |
| 05:21.06 | leprechau | and they are worth the double price in your oppinion over someone like voipjet? |
| 05:21.12 | mosty | does anyone know of a good little soho router that has reasonable qos? this is for a 2M/2M cable internet connection |
| 05:21.12 | awannabe | i never really found a ton of companies that did IAX, tons of SIP, but IAX is kind of iffy it seems |
| 05:21.39 | awannabe | voipjet sounded like you were making calls while on a jet! its a great name for them! |
| 05:21.53 | leprechau | well would i be better off having the local asterisk box SIP proxy out instead of IAX out? |
| 05:21.56 | awannabe | i had more issues then i could count, crappy termination, disconnects, etc, etc |
| 05:22.06 | awannabe | IAX can tranverse a firewall uber nice |
| 05:22.28 | leprechau | well the asterisk box will have direct T1+ connectivity |
| 05:22.29 | awannabe | and good for wan links cause it can compress |
| 05:22.33 | leprechau | so that's not a huge issue |
| 05:22.42 | awannabe | is this housed in like a data center/ |
| 05:22.52 | leprechau | office |
| 05:23.22 | leprechau | guy has a bonded T1 data connection... 3Mbps link |
| 05:23.51 | awannabe | i would check to see if you get good ping times to their sites |
| 05:24.08 | leprechau | but doesn't want to bring in any voice lines....and wants voip internally for sure...so i gotta do voip external to someone for PSTN connectivity |
| 05:24.28 | awannabe | yeah, i think your going to spend a arm and leg though |
| 05:25.00 | leprechau | well right now he is using aptela and spending $20/mo per phone |
| 05:27.54 | awannabe | i guess it depends on usage |
| 05:33.25 | leprechau | teliax has a 2500 minute plan for $40/mo |
| 05:33.30 | leprechau | that's pretty decent |
| 05:33.48 | awannabe | but, thats one concurrent call |
| 05:33.55 | *** join/#asterisk NghtLght (n=ircproxy@202.179.3.171) |
| 05:34.21 | leprechau | right...but don't multiple concurrent calls counted the same |
| 05:34.31 | leprechau | ie...total minutes of all calls == total minutes? |
| 05:35.11 | mosty | does anyone know how to do the equivalent of php's $$x with asterisk variables? |
| 05:35.36 | mosty | ie use the value of a variable to access another variable |
| 05:35.39 | leprechau | like if sam, sally and joe all are on the phone at the same time.... sam talks 5 ... sally 4, and joe 10 .... total minutes are 19 .... right? |
| 05:38.19 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.12) |
| 05:41.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:42.50 | *** part/#asterisk fx0 (n=ident@voip.terrorist.net) |
| 05:50.04 | [TK]D-Fender | leprechau : 1 concurrent call means if you try a 2nd call you get refused. |
| 06:01.53 | Un1x | how long is 1500 MS |
| 06:01.54 | Un1x | in seconds |
| 06:02.01 | Un1x | exten => s,1,Answer() |
| 06:02.01 | Un1x | exten => s,2,Dial(Zap/1,60) |
| 06:02.01 | Un1x | exten => s,3,Playback(vm-isunavail) |
| 06:02.01 | Un1x | exten => s,4,Playback(vm-intro) |
| 06:02.01 | Un1x | exten => s,5,Voicemail(s89) |
| 06:02.03 | ShadowHntr | 1.5 seconds |
| 06:02.05 | Un1x | thats my contex but |
| 06:02.09 | ShadowHntr | ms = milliseconds |
| 06:02.11 | Un1x | - Nobody picked up in 15000 ms |
| 06:02.11 | Un1x | <PROTECTED> |
| 06:02.13 | ShadowHntr | 1/1000 second |
| 06:02.13 | Un1x | this happens |
| 06:02.19 | ShadowHntr | 15000ms = 15 secs |
| 06:02.24 | Un1x | so why is it cutting the phone off in 15 seconds |
| 06:02.28 | Un1x | wen i have 60 |
| 06:02.35 | Un1x | exten => s,2,Dial(Zap/1,60) |
| 06:02.39 | ShadowHntr | dunno |
| 06:03.55 | JT | you don't know what a milisecond is? |
| 06:04.16 | Un1x | i know what a mili second is dont know how to convert into seconds |
| 06:04.17 | Un1x | :p |
| 06:04.43 | JT | well you can't if you don't know how to convert it, mili = thousandth |
| 06:04.50 | ShadowHntr | drop 3 zeroes |
| 06:04.54 | JT | so divie by 1000 |
| 06:04.57 | JT | divide |
| 06:05.02 | ShadowHntr | or move the decimal place to the left three places. |
| 06:05.02 | Corydon76-home | Perhaps you're looking at the wrong context |
| 06:05.58 | *** join/#asterisk apardo (n=apardo@87.217.147.146) |
| 06:06.04 | Corydon76-home | Do you have a context where the Dial timeout is 15 seconds? |
| 06:07.10 | *** join/#asterisk alerios (n=alerios@190.24.97.151) |
| 06:11.55 | [TK]D-Fender | Its like I always say.... there are 3 kinds of people in the world... those that know math.. and those that don't... |
| 06:13.17 | JT | there are 2 types of people in the world, those that say "maths" correctly, those that don't :D |
| 06:14.07 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.mn.comcast.net) |
| 06:14.19 | benjk | there are 10 types of people, those that understand binary and those that don't |
| 06:14.41 | Un1x | guys any idea why this isn't working |
| 06:14.42 | Un1x | http://pastebin.ca/270272 |
| 06:14.44 | [TK]D-Fender | benjk : base 13 FTW |
| 06:14.47 | Un1x | please help guys :P |
| 06:14.50 | benjk | heh |
| 06:15.21 | [TK]D-Fender | Un1x : How about you tell us what PART isn't working there... |
| 06:15.31 | benjk | there are actually only two types of people ... those who divide people into two types of people and those who don't |
| 06:15.36 | Un1x | i called that did |
| 06:15.39 | Un1x | and i get person unavailable |
| 06:16.05 | Un1x | http://pastebin.ca/270274 |
| 06:16.08 | Un1x | here is my extensions.conf |
| 06:16.09 | [TK]D-Fender | Un1x : what you showed us means NOTHING then. Check your inbound context. |
| 06:16.21 | Un1x | for some reason when i call my did it says person unavailable |
| 06:16.27 | Un1x | and im in console as well i see no activity |
| 06:17.02 | [TK]D-Fender | Un1x : The problem is current in your channel definition. |
| 06:17.16 | Un1x | so how do i fix it? |
| 06:17.30 | Un1x | outgoing calls are working perfectly just the incoming in the default context are having problems |
| 06:17.39 | Un1x | when i remove the did beside the answer and everything calls come in fine |
| 06:17.44 | [TK]D-Fender | Un1x : You haven't shown it yet. Time to wake up.... |
| 06:18.42 | Un1x | D-Fender i just pasted you all of my extensions.conf man :( |
| 06:18.47 | Un1x | its in there |
| 06:19.00 | *** join/#asterisk SwK[Work] (n=SwK@br0.asteriasgi.com) |
| 06:19.18 | [TK]D-Fender | Un1x : And I just told you that your problem is in your CHANNEL definition. that means sip.conf, zapata,conf, etc, depending on where the call is coming in from. |
| 06:19.28 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 06:19.42 | [TK]D-Fender | Un1x : Your extensions.conf isn't even coming into play. |
| 06:19.47 | Un1x | see if i replace this exten => s,1,Answer() |
| 06:19.47 | Un1x | exten => s,2,Dial(Zap/1,45) |
| 06:19.47 | Un1x | exten => s,3,Playback(vm-isunavail) |
| 06:19.47 | Un1x | exten => s,4,Playback(vm-intro) |
| 06:19.47 | Un1x | exten => s,5,Voicemail(s89) |
| 06:19.53 | Un1x | sorry |
| 06:19.57 | Un1x | if i replace the s with my did |
| 06:20.14 | Un1x | or is the s supposed to be the number im calling from? |
| 06:20.22 | heison | hi there... i have cdr_odbc running on one box which I setup years ago... i'm trying to replicate on a new box but i can't recall how cdr_odbc was built... |
| 06:20.44 | [TK]D-Fender | Un1x : Look at you damned sip.conf entires to see where they point. then if everything looks right, turn on sip debug and see what your provider is calling to send you calls |
| 06:21.02 | heison | I check asterisk/cdr/Makefile on the machine that has cdr_odbc.so, and compare that to my new box... they are identical |
| 06:21.24 | [TK]D-Fender | Un1x : GET YOUR ASS OUT OF EXTENSION.CONF. Is that clear enough for you? You ar barking up the wrong tree. |
| 06:21.35 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 06:21.58 | Un1x | yes |
| 06:22.19 | [TK]D-Fender | Un1x : pastebin your sip.conf substituting only passwords. |
| 06:23.59 | Un1x | ok |
| 06:24.13 | *** join/#asterisk BSD_Tech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
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| 06:24.42 | Un1x | http://pastebin.ca/270276 |
| 06:24.49 | Un1x | [TK]D-Fender ther you go |
| 06:26.06 | [TK]D-Fender | Un1x : You did not tell them what exten to dial into you with in your register line |
| 06:26.32 | [TK]D-Fender | Un1x : register => 10085:passss@voipgw1.splitinfinity.net/yourdidhere <- |
| 06:26.48 | [TK]D-Fender | Un1x : * most likely told them to send calls to "s". |
| 06:27.04 | BSDTech | what does asterisk use from x11 ? |
| 06:27.09 | [TK]D-Fender | Un1x : For which you have no match in [default] |
| 06:27.19 | BSDTech | is the lib really neede ? |
| 06:27.24 | Un1x | ahh so i just gotta add my did after the .net? |
| 06:27.38 | Un1x | but D-Fender i have more then one DID... |
| 06:27.46 | Un1x | then if i put one did there where will the others go? |
| 06:29.22 | *** join/#asterisk diclophis-work (n=jbardin@comet.cisdata.net) |
| 06:29.29 | Un1x | [TK]D-Fender : so how do i solve this problem how do i tell them what exten to dail into instead of using 's' |
| 06:29.52 | [TK]D-Fender | Un1x : more than 1 DID on that registration? |
| 06:30.08 | orlock | Un1x: have you specified anything on your REGISTER line? |
| 06:30.12 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:30.12 | Un1x | can i do more then one did on that registration? |
| 06:30.13 | orlock | uhh, register, even |
| 06:30.35 | Un1x | how do devide them : | ; or , |
| 06:30.47 | orlock | Un1x: how many numbers are assigned to that account? |
| 06:30.53 | Un1x | register => 10085:passss@voipgw1.splitinfinity.net/4168480516/4168485516 |
| 06:30.55 | Un1x | like that? |
| 06:31.03 | [TK]D-Fender | Un1x : no, you can only pass 1 value |
| 06:31.06 | orlock | nah |
| 06:31.08 | orlock | just the one |
| 06:31.18 | orlock | Un1x: do you have multiple DID's on that one SIP account? |
| 06:31.36 | [TK]D-Fender | Un1x : You didn't answer my question. Do you have more than 1 DID coming inbound from that provider? |
| 06:32.02 | orlock | Un1x: asterisk determines the DID based on the account in the INVITE, not the To:, and your ISP will probably be specifying the DID in the To: |
| 06:32.10 | orlock | i actually think thats a bug in asterisk |
| 06:32.31 | orlock | but i am not sure if the RFC specifys it, and there is a workaround |
| 06:32.46 | Un1x | [TK]D-Fender yes i have 2 dids coming from that provider |
| 06:32.54 | orlock | Un1x: on the one account? |
| 06:32.57 | Un1x | and one from did-ww as you see at the bottom |
| 06:32.58 | Un1x | yes |
| 06:33.00 | hads | Parse the header |
| 06:33.10 | Un1x | how so? |
| 06:33.32 | heison | ah... found it... cdr_odbc.so need odbcinst.h which I do not have installed... it's part of unixodbc-dev (I have only installed unixodbc) |
| 06:33.35 | orlock | SipGetHeader and SipSetHeader, or theres another function now i think |
| 06:34.13 | Un1x | [TK]D-Fender ? |
| 06:34.28 | [TK]D-Fender | Un1x : did-ww and splitinifinity appear to be 2 different places.... |
| 06:34.41 | Un1x | Yes, i have 2 dids from splitinfinity and 1 from did-ww |
| 06:34.54 | Un1x | so total of 3 but to you answer youre question about splitinfinity yes i got 2 from them |
| 06:34.59 | [TK]D-Fender | Un1x : Also you context setups in sip.conf are backwards. you use "user" to recieve calls against, and "peer |
| 06:35.10 | [TK]D-Fender | " to send calls. you have defined them backwards. |
| 06:35.58 | Un1x | D-fender the did from did-ww is only incoming not outgoing service i can only receive calls from that provider |
| 06:36.14 | [TK]D-Fender | Un1x : If your provider has a guide, or are listed on the WIKi I suggest you go over their samples again. |
| 06:36.21 | [TK]D-Fender | Either way I've got to hit the sack... |
| 06:36.28 | Un1x | alright man gnite |
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| 07:16.07 | Un1x | Sigh anyone know of 8** did provider |
| 07:16.10 | Un1x | toll free did providers |
| 07:17.14 | rob0 | Oh I think there are many. Asterlink, for one. |
| 07:17.35 | rob0 | By 8** ITYM toll-free? |
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| 07:17.55 | Un1x | rob0 yes |
| 07:17.57 | Un1x | ~nanpa |
| 07:17.59 | jbot | methinks nanpa is North America Numbering Plan Administration: an integrated telephone numbering plan serving 19 North American countries that share its resources. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.net/ |
| 07:18.18 | *** join/#asterisk redebr2818 (n=rede@201.63.15.202) |
| 07:18.39 | DrCron | actually only a 800 and 888 are toll free in the 8xx block |
| 07:19.00 | rob0 | 877, 866? |
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| 07:19.20 | Un1x | so list some... |
| 07:19.22 | Un1x | :P |
| 07:19.50 | DrCron | almost any DID provider will do it |
| 07:20.46 | DrCron | voipjet, myphonecompany.com, broadvoice |
| 07:27.49 | benjk | DrCron, not true |
| 07:28.39 | benjk | 800, 855, 866, 877, and 888 are all for toll free use, 855 not being deployed yet, 866 still lacking overlay codes, but all of them are toll free |
| 07:29.42 | benjk | it depends on the service provider how they customise their offer, what you pay for incoming calls on those though |
| 07:30.44 | benjk | in order to keep cost as low as possible, many providers offer 8xx toll free services such that only callers from within the same country can call the number, so as to avoid the international charges for calls from other NANPA countries |
| 07:31.44 | benjk | this means that in some cases you cannot call a toll free number directly, but you have to use an overlay code, in which case both the caller and the called party pay |
| 07:33.00 | benjk | for example if you want to call an 800 number restricted to US48 from Canada, you dial an overlay code instead of the 800 code and then it is not toll free to you, but the 800 number is toll free from within US48 area codes |
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| 07:39.41 | DrCron | sorry, i forgot 55 and 66 |
| 07:39.56 | DrCron | but most of the 8xx codes are not |
| 07:40.09 | DrCron | right? |
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| 07:48.03 | Un1x | fuck i have bad lucky |
| 07:48.22 | Un1x | i cant even find one god damn 18** provider that doesn't want u to sign up to there shit like voicermail etc |
| 07:48.24 | Un1x | or other services |
| 07:50.01 | DrCron | check voip-info |
| 07:50.07 | DrCron | they have a listing |
| 07:51.16 | Un1x | there listing has been truncated recently |
| 07:51.19 | Un1x | i clicked all |
| 07:51.19 | Un1x | lol |
| 07:51.23 | Un1x | well the retailer ones |
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| 07:57.38 | benjk | DrCron, only 800, 855 (not implemented yet), 866 (overlay codes not implemented yet), 877 and 888, yes |
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| 08:13.18 | E-bola | Morning |
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| 08:38.31 | HaMYaI | has anyone had a problem where outgoing sip calls get sent to default context? |
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| 08:39.59 | HaMYaI | seems like it's the authentication problem |
| 08:40.30 | Un1x | hamyai paste youre extensions.conf and we can see what might the problem be |
| 08:40.38 | Un1x | otherwise no one will even both responding to you :P |
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| 08:48.01 | phonetalk | hello everybody |
| 08:48.09 | phonetalk | i m having a little problem in ss7 |
| 08:50.13 | Un1x | Okay, guys how would i get a specific phone on a specific channel to use a specific carrier what would the syntax be for the context |
| 08:50.40 | phonetalk | anybody here experience with asterisk-ss7 ? |
| 08:54.44 | *** part/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 08:55.58 | JT | phonetalk: what on earth are you using ss7 for? |
| 08:56.23 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.2.16.revip2.asianet.co.th) |
| 08:57.14 | HaMYaI | Un1x: should I paste the sip.conf instead? |
| 08:57.38 | phonetalk | JT: its because of my PSTN |
| 08:58.18 | HaMYaI | Un1x: the call hasn't actually been authenticated |
| 08:58.21 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 09:00.44 | HaMYaI | and there's no link between user's context and the default context either |
| 09:01.47 | Un1x | HaMYaI ask JT |
| 09:02.02 | HaMYaI | OK |
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| 09:03.26 | HaMYaI | JT: I am having a problem where outgoing sip calls get sent to the default SIP context but it only happens not very frequently |
| 09:04.06 | HaMYaI | JT: you know where I should start looking at in order to solve this problem |
| 09:04.38 | HaMYaI | Un1x: well, don't think he's awake |
| 09:06.35 | JT | what do you mean by an outgoing sip call? |
| 09:08.07 | phonetalk | anybody experienced with ss7 here ? |
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| 09:12.17 | HaMYaI | JT: I am using SIP softphone to dial out |
| 09:12.54 | HaMYaI | added this user to sip.conf and set context=user-context |
| 09:14.47 | HaMYaI | JT: This works fine generally but sometimes the calls were sent to default context set in the [general] section of the sip.conf |
| 09:15.42 | HaMYaI | no realm parameter is set in user's setting but in [general] section |
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| 10:13.52 | merbanan | if asterisk 1.4 doesn't open the sip port what do I need to configure ? |
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| 10:20.32 | phonetalk2 | hello |
| 10:21.01 | phonetalk2 | anybody have experience with asterisk-ss7 ? |
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| 10:22.46 | vooduhal | Anyone alive? |
| 10:22.46 | darkskiez | probably not many folk, best get support with digium for that specialist topic |
| 10:23.57 | zoa | i dont think digium supports it yet |
| 10:24.07 | zoa | its best to subscribe to the asterisk-ss7 mailinglist |
| 10:24.08 | zoa | and try there |
| 10:24.28 | dlynes_laptop | and the asterisk-ss7 list seems to be sparse at best |
| 10:24.42 | vooduhal | Can anyone suggest a way to do the whole, "Enter the extension you'd like to dial followed by the # key" and have it either wait for the # and then dial the extension or timeout and dial what was entered? My dialplan currently just as _XXXX and as soon as they enter the 4 digits it starts going but the prompt we have recorded says to follow it with #. |
| 10:25.04 | darkskiez | _XXXX# :) |
| 10:25.16 | vooduhal | That doesn't account for the timeout though. |
| 10:25.56 | vooduhal | If they enter just 1234 it will jump to either i or t depending. |
| 10:26.24 | vooduhal | Nm.. Just found a way. Looks like Read can do what I want. |
| 10:28.39 | MrChimpy | good morning |
| 10:28.51 | MrChimpy | anyone running asterisk in x86_64 build? |
| 10:29.11 | zoa | me |
| 10:29.18 | zoa | for 2 year or so already |
| 10:29.20 | zoa | works fine |
| 10:29.23 | MrChimpy | do you use zaptel? |
| 10:29.37 | zoa | yes |
| 10:30.02 | zoa | aha |
| 10:30.07 | zoa | check your kernel timing |
| 10:30.13 | MrChimpy | ooh |
| 10:30.22 | zoa | ah i have no experience with wanpipe |
| 10:30.34 | MrChimpy | kernel timing would sound feasible |
| 10:31.12 | zoa | http://www.asteriskguru.com/tutorials/timingwarnings.html |
| 10:31.17 | MrChimpy | it's playing alaw sample, the stuff works on 32 bit build on other boxes |
| 10:31.28 | MrChimpy | sample plays about 4x too slow with loads of clipping |
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| 10:32.22 | zoa | try the kernel timing thing |
| 10:32.27 | zoa | ive seen similar things before |
| 10:33.46 | merbanan | I get Registration error 405 when I try to register to the asterisk server, the accounts are listed correctly, what can be wrong ? |
| 10:34.56 | jeremy_g | merbanan:what does 405 mean? |
| 10:37.04 | merbanan | it is a sip error code for the "Method Not Allowed" message |
| 10:42.08 | jeremy_g | MrChimpy: :) how do you do |
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| 10:47.02 | tparcina | hi channel! |
| 10:47.29 | tparcina | how can I tell competer to reboot tonight at 2AM? |
| 10:48.15 | dlynes_laptop | 0 2 * * * /usr/sbin/shutdown -r now Now!!! |
| 10:48.18 | dlynes_laptop | in your crontab |
| 10:49.20 | tparcina | dlynes_laptop: anything else except crontab? |
| 10:50.03 | dlynes_laptop | tparcina: you could run something silently in the background as a daemon that wakes up at 2am |
| 10:50.13 | dlynes_laptop | tparcina: but why bother when you have cron? |
| 10:50.24 | dlynes_laptop | tparcina: also, you can use the 'at' command |
| 10:52.05 | tparcina | dlynes_laptop: echo reboot | at 2am |
| 10:52.15 | jeremy_g | hi tparcina |
| 10:52.31 | tparcina | dlynes_laptop: I think I'll use that one |
| 10:52.43 | tparcina | jeremy_g: hi jeremy |
| 10:52.56 | jeremy_g | tparcina:hows your work going |
| 10:53.07 | jeremy_g | tparcina:yeah use at, cuz u need one time rebboot |
| 10:53.21 | tparcina | jeremy_g: what my work? |
| 10:53.27 | dlynes_laptop | tparcina: is it every night you want to do that, or only tonight? |
| 10:53.42 | tparcina | dlynes_laptop: only tonight |
| 10:53.49 | dlynes_laptop | tparcina: ah..yeah...use at then |
| 10:55.21 | tzafrir | tparcina, why not with a cron job? |
| 10:55.39 | tzafrir | @reboot sleep 86400; reboot |
| 10:55.49 | tzafrir | ;-P |
| 10:56.25 | *** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br) |
| 10:56.42 | tparcina | tzafrir: thank you for sugestion |
| 10:57.59 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 10:58.43 | tzafrir | tparcina, the @reboot one? it was not serious. Consider what happens if you reboot once during the day |
| 10:58.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:00.36 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 11:01.36 | *** join/#asterisk CleanerX (n=nix@p54A394F8.dip0.t-ipconnect.de) |
| 11:04.16 | *** join/#asterisk asterisk_baby (n=shady@202.61.50.75) |
| 11:04.56 | asterisk_baby | hi |
| 11:05.07 | *** join/#asterisk _omer (i=_omer@202.38.55.125) |
| 11:05.34 | asterisk_baby | im having problems registering sip users on my godaddy server (fedora 4) |
| 11:06.50 | _omer | hi |
| 11:07.01 | _omer | any suggestions?? http://pastebin.ca/270442 |
| 11:08.06 | jeremy_g | _omer:you seem to be asking whether asterisk supports refer method? |
| 11:08.27 | _omer | yep. |
| 11:10.34 | *** join/#asterisk SoftIce (n=softice@vc-196-207-45-253.3g.vodacom.co.za) |
| 11:11.21 | SoftIce | hi could anyone tell me here, if I install freepbx for gui for asterisk will it mess with my zaptel conf ? as its taken me ages to get the isdn modem working and all the modifications ive mad to zaptel |
| 11:11.48 | jm|work | SoftIce, just back up /etc/asterisk and /etc/zapata.conf ? |
| 11:11.49 | asterisk_baby | http://pastebin.ca/270448 |
| 11:11.54 | asterisk_baby | help plz |
| 11:12.12 | *** join/#asterisk Modcuts (n=Moducts@88-111-138-170.dynamic.dsl.as9105.com) |
| 11:12.18 | *** join/#asterisk AuPix (n=root@mail.aupix.com) |
| 11:12.38 | jeremy_g | _omer:asterisk does support refer method |
| 11:13.16 | SoftIce | jm|work: ye I ca do that, but overwriting /etc/asterisk/zapata.conf and /etc/zaptel.conf with what ever freepbx does |
| 11:13.22 | SoftIce | will that cause freepbx not work? |
| 11:14.36 | zapp-branigan | hi the Open Source G.729 ? where can i donload the license ? the txt say http://www.intel.com/software/products/ipp/ register to intel site but i can't find this ... |
| 11:15.08 | asterisk_baby | http://pastebin.ca/270448 |
| 11:16.35 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-115-171.vic.westnet.com.au) |
| 11:17.27 | *** join/#asterisk dorel__ (n=liran@80.179.31.43.static.012.net.il) |
| 11:20.55 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 11:21.21 | *** join/#asterisk jm|work (n=jamiem@sentry.flags.co.uk) |
| 11:21.29 | *** join/#asterisk tparcina_ (n=tomo@wr-lama.iskon.hr) |
| 11:23.51 | asterisk_baby | guys |
| 11:23.51 | asterisk_baby | http://pastebin.ca/270452 |
| 11:23.54 | asterisk_baby | need help |
| 11:24.58 | RoyK | asterisk_baby: if unable to register sip users, first doublecheck everything, then try to make a sip debug |
| 11:26.09 | hads | Some clients have trouble with non-standard ports. |
| 11:26.17 | *** join/#asterisk xAD (i=doddo@82.56.174.227) |
| 11:27.16 | *** part/#asterisk xAD (i=doddo@82.56.174.227) |
| 11:27.48 | asterisk_baby | royk: i just compiled everything again..checked the sip debug too its still the same |
| 11:28.17 | RoyK | what does 'sip debug' say? |
| 11:29.26 | asterisk_baby | i see nothing.. no activity |
| 11:29.48 | *** join/#asterisk Omer (i=Omer@202.133.79.19) |
| 11:30.38 | RoyK | well, then no data reaches the box |
| 11:30.50 | asterisk_baby | yup |
| 11:32.32 | RoyK | asterisk_baby: and don't contact people privately unless it's private. this is support |
| 11:32.53 | *** join/#asterisk vIR_uS (n=I@p54B16954.dip.t-dialin.net) |
| 11:34.57 | asterisk_baby | alright RoyK. the requests are coming on my pc |
| 11:35.10 | Omer | FOP is not showing anything that who is calling or who is on the line |
| 11:36.21 | RoyK | ~fop |
| 11:36.27 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 11:39.14 | *** join/#asterisk jm|work (n=jamiem@sentry.flags.co.uk) |
| 11:41.01 | asterisk_baby | i meant the requests are NOT coming on the server |
| 11:41.07 | asterisk_baby | royk: any ideas? |
| 11:41.22 | Omer | i just checked it |
| 11:41.25 | *** join/#asterisk jm|work (n=jamiem@sentry.flags.co.uk) |
| 11:41.45 | Omer | its working fine for local users but not for external users |
| 11:42.12 | asterisk_baby | http://pastebin.ca/270452 |
| 11:42.15 | asterisk_baby | help me guys :( |
| 11:42.19 | asterisk_baby | please |
| 11:42.30 | RoyK | asterisk_baby: fix your network |
| 11:42.53 | RoyK | if the packets aren't arriving at the server, there is NOTHING asterisk can do about it |
| 11:43.38 | Omer | indeeed |
| 11:43.53 | zoa | does anybody here have sample support contracts for PBX'es ? |
| 11:44.26 | asterisk_baby | i tried registering it on the server itself.. register => .. but even that doesnt works |
| 11:45.16 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:53.20 | asterisk_baby | http://pastebin.ca/270468 |
| 12:01.03 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 12:01.27 | EmleyMoor | Do Zap channels support caller ID boxes? |
| 12:02.45 | MrChimpy | yes |
| 12:02.59 | MrChimpy | in that i can make a call out on a zaptel port and see clid on my phone |
| 12:03.25 | MrChimpy | may depend on your interface and provider, and also config |
| 12:03.28 | EmleyMoor | How about on calls bridged to Zaptel ports? |
| 12:05.00 | EmleyMoor | (e.g. an incoming call - can a caller ID box on the Zaptel port show caller ID, if the dialplan is programmed appropriately) |
| 12:05.02 | *** join/#asterisk Skarmeth (n=Skarmeth@201009008073.user.veloxzone.com.br) |
| 12:06.30 | MrChimpy | yep afaik |
| 12:06.56 | MrChimpy | i've done various bits of bridging and it passes CLID as it should without extra stuff |
| 12:07.01 | EmleyMoor | Any examples of how, assuming UK incoming line and UK equipment? |
| 12:07.13 | MrChimpy | it'lll just work |
| 12:07.22 | MrChimpy | you using analog equipment? |
| 12:07.31 | EmleyMoor | Yes |
| 12:07.48 | MrChimpy | ah. probably your zaptel/zapata config then |
| 12:07.53 | MrChimpy | i'm using E1s |
| 12:08.08 | MrChimpy | do you have CLID working at any level? |
| 12:08.22 | MrChimpy | do you see the variables set correctly in dialplan when people call in? |
| 12:08.49 | EmleyMoor | Well, it gets passed to softphones, and it's presented on the incoming line. I can't answer the last point until my replacement FXO module arrives |
| 12:09.06 | MrChimpy | try #asterisk-uk for advice on this |
| 12:09.20 | MrChimpy | may well be provider stopping you setting a CLID |
| 12:09.31 | EmleyMoor | I don't want to set it |
| 12:09.36 | MrChimpy | ok |
| 12:09.55 | MrChimpy | so your problem is CLID just not appearing on outbound analog calls? |
| 12:10.08 | EmleyMoor | No, nothing to do with outbound |
| 12:10.44 | *** join/#asterisk phonetalk (n=phonetal@host210-2-164-29.isb.dancom.net.pk) |
| 12:10.53 | EmleyMoor | My query is about how I can handle incoming CLID and how I can display it on British equipment designed to do so but plugged in on the inside |
| 12:11.44 | MrChimpy | oh. I'd expect it can. |
| 12:11.53 | MrChimpy | best mailing list it. |
| 12:15.55 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.26) |
| 12:17.48 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 12:20.31 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-146-63.dyn.embarqhsd.net) |
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| 12:23.05 | EmleyMoor | My FXO module has arrived |
| 12:33.50 | EmleyMoor | What a pain to fit! |
| 12:35.45 | *** part/#asterisk andresmujica (n=andresmu@201.244.244.26) |
| 12:38.52 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 12:39.13 | EmleyMoor | Well, the permissions fixed the "asterisk doesn't run at startup" |
| 12:52.26 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 12:55.06 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 12:55.18 | joelsolanki | I have installed sangoma's wanpipe. installation is sucessful. but how do i configure the sangoma card. |
| 12:55.27 | joelsolanki | <PROTECTED> |
| 12:55.39 | joelsolanki | wancfg: Error in File: menu_hardware_probe.cpp, Function: run(), Line: 174. Text: |
| 12:55.48 | joelsolanki | Failed to get Card Type from selected card line!! line: AFT-A200-SH SLOT=8 BUS=2 IRQ=193 CPU=A PORT=PRI |
| 12:55.57 | joelsolanki | any help plz |
| 12:56.00 | tRSS | I am having a hard time getting my TE110P to work with the channel bank. i had the card and the channel bank working on another machine before I moved it onto this machine. |
| 12:56.04 | RoyK | A104 and so |
| 12:56.39 | joelsolanki | A200 sangoma with 8 analog ports |
| 12:58.33 | *** join/#asterisk Modcuts (n=Moducts@88-110-122-48.dynamic.dsl.as9105.com) |
| 13:01.44 | *** part/#asterisk oej (n=olle@apollo.webway.se) |
| 13:05.06 | *** join/#asterisk jorge__ (n=te_lo_me@adsl-149-57-198.mia.bellsouth.net) |
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| 13:12.29 | *** join/#asterisk acidoverflow (n=acidover@p3EE29B40.dip0.t-ipconnect.de) |
| 13:12.34 | acidoverflow | hi |
| 13:12.53 | acidoverflow | does anybody know what to do against increasing lags i get now and then? |
| 13:13.50 | acidoverflow | like this IAX2/101-35 192.168.0.19 101 00035/26986 00063/00061 lag:17220ms 0025ms 0086ms ulaw |
| 13:13.59 | *** join/#asterisk groogs___ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
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| 13:19.10 | *** join/#asterisk groogs____ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
| 13:19.48 | zoa | acidoverflow: disable the iax2 jitter buffer |
| 13:20.14 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:20.14 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 13:20.55 | acidoverflow | zoa i had it disabled, just enabled it yesterday as probably solution ... but it didn't help |
| 13:22.38 | zoa | aha |
| 13:22.42 | zoa | hey ho coppice |
| 13:22.44 | acidoverflow | when the lag in iax2 show channels is around 17000ms and i do a normal ping from the server to the client or vice versa the ping is under <1ms |
| 13:23.43 | coppice | hi |
| 13:24.00 | coppice | anyone use mediatrix boxes? |
| 13:24.14 | acidoverflow | it normaly happens 1 hour after the client is bootet and from then on every hour the lag increases and after 5 minutes when starting nxt call its back to normal |
| 13:25.00 | *** join/#asterisk kimmimy (n=jkl@ppp-58.10.155.185.revip2.asianet.co.th) |
| 13:29.36 | javar | hi coppice |
| 13:29.37 | *** join/#asterisk TypMic1004 (n=chrislro@outland.cmf.nrl.navy.mil) |
| 13:29.37 | javar | i have a mediatrix 1204 |
| 13:29.37 | javar | <PROTECTED> |
| 13:29.37 | coppice | the 1204's T.38 looks like total crap. is the whole box that bad? |
| 13:29.37 | zoa | i think it is |
| 13:29.37 | zoa | i hate them |
| 13:29.37 | zoa | i had some |
| 13:29.37 | zoa | and started to really kick them |
| 13:29.37 | zoa | they used to be snmp only to configure |
| 13:29.38 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:29.38 | zoa | i gave up after a week |
| 13:29.38 | zoa | hey ho |
| 13:29.38 | zoa | olle |
| 13:29.38 | oej | hey ho |
| 13:29.39 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 13:29.39 | coppice | most of the weird stuff we've had so far with T.38 has been with mediatrix boxes. I'm just starting to work through the issues |
| 13:29.57 | DeeJayTwo | I just insatlled 1.4b3 but I need some fax detection on it lik nvdetect did |
| 13:31.24 | kimmimy | hello , If I buy many sip accounts and register to asterisk server , and then lets user call pass this asterisk , how its called this way ? registra server ? |
| 13:31.24 | DeeJayTwo | the nvdetect stuff doesn'T seem to compile on 1.4 unfortunately... |
| 13:31.24 | *** part/#asterisk TypMic1004 (n=chrislro@outland.cmf.nrl.navy.mil) |
| 13:33.26 | kimmimy | hello , If I buy many sip accounts and register to asterisk server , and then lets users call pass this asterisk , how its called ? registra server ? |
| 13:35.59 | EmleyMoor | What do I type to see the details presented on my Zap channel? |
| 13:40.06 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 13:41.48 | *** join/#asterisk Ebola (n=Ebola@host86-134-167-28.range86-134.btcentralplus.com) |
| 13:44.25 | *** join/#asterisk puk_jp (n=ukris@p4224-ipad81marunouchi.tokyo.ocn.ne.jp) |
| 13:48.57 | EmleyMoor | How do I see if asterisk can see incoming CLID? |
| 13:49.08 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 13:49.41 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 13:50.44 | in-pt | I got problem in starting asterisk-1.4.0-beta3..its like this |
| 13:50.48 | in-pt | [Dec 7 13:44:47] NOTICE[29428]: cdr_radius.c:258 load_module: Cannot load radiusclient-ng configuration file /etc/radiusclient-ng/radiusclient.conf |
| 13:51.04 | in-pt | though i had disabled cdr loggings in cdr.conf filew |
| 13:51.16 | in-pt | *file |
| 13:51.30 | in-pt | can anyone give any suggestion??? |
| 13:52.29 | in-pt | i dont want to do anything with radiusclient-ng with asterisk |
| 13:53.09 | darkskiez | whats the problem ? |
| 13:53.21 | darkskiez | noload the module |
| 13:53.31 | darkskiez | or delete it |
| 13:53.38 | darkskiez | (cdr_radius) |
| 13:53.47 | in-pt | ok |
| 13:54.36 | in-pt | thanks darkskiez its working now |
| 13:55.43 | jmesquita | Anyone has problems with ticket 7765? |
| 13:57.17 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.12) |
| 13:58.33 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 14:01.27 | *** join/#asterisk ComPuTeR (n=PHiLiPS@88.240.42.50) |
| 14:04.20 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 14:06.43 | *** join/#asterisk acidoverflow (n=acidover@p3EE2888F.dip0.t-ipconnect.de) |
| 14:13.17 | *** join/#asterisk heison (n=heison@216.126.67.39) |
| 14:15.11 | *** join/#asterisk beyond (n=Evandro@200-155-185-1.static.spo.ifx.net.br) |
| 14:16.52 | EmleyMoor | I'd like to see if asterisk can get the caller ID on incoming Zap calls - what do I need to do to see it? |
| 14:20.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.29 | *** join/#asterisk groogs____ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
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| 14:26.40 | *** join/#asterisk spyda (n=scott@hera.copi-rite.com) |
| 14:26.45 | *** join/#asterisk leoncamel (n=leoncame@219.142.137.2) |
| 14:30.57 | joelsolanki | can anybody help with installing sangoma ? |
| 14:31.03 | joelsolanki | i have A200 sangoma card |
| 14:31.18 | joelsolanki | i have installed it correctly. wanpipe driver is installed. |
| 14:31.52 | *** join/#asterisk SnAzZpOp (n=snazzpop@15-48-231-201.fibertel.com.ar) |
| 14:32.03 | joelsolanki | wanroute hwprobe gives me output for AFT-200--SH |
| 14:32.10 | joelsolanki | now how do i configure ? |
| 14:32.18 | joelsolanki | any docs / hints |
| 14:33.12 | *** join/#asterisk alerios (n=alerios@190.24.97.151) |
| 14:36.43 | *** part/#asterisk SnAzZpOp (n=snazzpop@15-48-231-201.fibertel.com.ar) |
| 14:39.38 | *** join/#asterisk \lart (n=nunya@arwen.jasons.org) |
| 14:39.45 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:39.49 | Katty | morning! |
| 14:40.18 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:42.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:42.22 | *** join/#asterisk Agimat2K4 (i=0@124.107.2.147) |
| 14:43.02 | *** join/#asterisk stresscool (n=vayrette@LNeuilly-152-22-102-237.w193-251.abo.wanadoo.fr) |
| 14:43.05 | stresscool | hi |
| 14:43.11 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:43.25 | Katty | gosh so quiet :< |
| 14:43.36 | mercestes | Hi, Katty |
| 14:43.41 | Katty | ello, mercestes |
| 14:43.52 | stresscool | i try to record a message for my queue, but when i do that it's happen this mistake |
| 14:44.03 | stresscool | “Unable to add /var/lib/asterisk/sounds/custom/test.wav |
| 14:44.05 | mercestes | I missed you. |
| 14:44.09 | Katty | did you? |
| 14:44.13 | mercestes | I did. |
| 14:44.15 | Katty | mercestes: i still have no clue who you are :P |
| 14:44.17 | Agimat2K4 | hi all |
| 14:44.31 | Katty | Agimat2K4: hihi |
| 14:44.34 | mercestes | stresscool: did you check for the existance of that folder and permissions? |
| 14:44.36 | Katty | stresscool: did you hug it? |
| 14:44.44 | Agimat2K4 | hi katty |
| 14:44.45 | mercestes | katty: You remember that cute guy? at that place? you know, the one you were hitting on. |
| 14:44.51 | Katty | mercestes: mew? |
| 14:45.04 | Katty | mercestes: i love everyone equally. |
| 14:45.07 | mercestes | katty: that blonde guy...oh come on, you remember. |
| 14:45.10 | Agimat2K4 | katty wer from? |
| 14:45.17 | Katty | mercestes: i dun like blondes :< |
| 14:45.31 | mercestes | katty: well I was the guy *next* to him..you know, the one you were hitting on. |
| 14:45.36 | stresscool | i use trixbox on sme7 with php4.3.9 freepbx 2.1.1 |
| 14:45.37 | mercestes | katty: with dark hair. |
| 14:45.48 | Katty | mercestes: are you a canadian?! |
| 14:45.48 | mercestes | stresscool: try #freepbx |
| 14:46.01 | mercestes | katty: eh? |
| 14:46.07 | Katty | mercestes: umm, umm.. |
| 14:46.08 | stresscool | thank |
| 14:46.11 | stresscool | you |
| 14:46.12 | Katty | mercestes: i need more hints, apparently. |
| 14:46.36 | mercestes | lol |
| 14:46.55 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 14:48.14 | hmmhesays | walker texas ranger |
| 14:48.22 | hmmhesays | is on right now |
| 14:49.27 | Katty | hmmhesays: mew. |
| 14:49.36 | Katty | hmmhesays: you recorded any new stuff lately? |
| 14:49.58 | hmmhesays | Katty: not as of late |
| 14:50.04 | hmmhesays | I could though |
| 14:50.43 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:50.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:50.48 | Katty | morning anthony (= |
| 14:50.57 | Katty | hmmhesays: you should. |
| 14:51.08 | Katty | hmmhesays: it'd give ya somethin to do, other than chase girlies around. |
| 14:51.21 | xnon_ | asterisk en español? #asterisk-ve |
| 14:52.58 | hmmhesays | Katty: haha |
| 14:54.59 | Katty | hmmhesays: in other mews, do you know of a terminal based yahoo client that will work with asterisk? |
| 14:55.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:55.36 | hmmhesays | doesn't really have to work 'with' asterisk cause you can do a system call with asterisk |
| 14:56.05 | Katty | true. |
| 14:56.13 | Katty | but i still need one just the same. |
| 14:56.20 | hmmhesays | whats google tell ou? |
| 14:56.21 | Katty | cause, clearly, i need to be spammed everytime the phone rings. |
| 14:56.27 | hmmhesays | of course |
| 14:57.09 | Katty | i found centericq. |
| 14:57.26 | hmmhesays | http://freshmeat.net/projects/arisyahooclient/ |
| 14:57.32 | hmmhesays | that maybe |
| 14:58.10 | Katty | butbut, centericq has more client support. |
| 14:58.15 | *** join/#asterisk bigjb (n=nnbigjb@195.60.10.114) |
| 14:58.45 | DeeJayTwo | how can I detect fax with asterisk 1.4 on iax channels? |
| 14:59.25 | hmmhesays | Katty: I dunno I just pulled that off google |
| 14:59.35 | mercestes | DeeJayTwo: Must you *detect* fax or can you use dedicated IAX #'s? |
| 14:59.36 | Katty | hmmhesays: kay. |
| 15:00.57 | *** join/#asterisk celosdedma (n=celosded@124.107.2.148) |
| 15:02.37 | hmmhesays | I think i'm going to make a pizza |
| 15:04.51 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.236.Dial1.SanJose1.Level3.net) |
| 15:04.54 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 15:05.15 | *** part/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 15:05.26 | xnon_ | anybody can help me! i need some help i was probe all tutorials and i can finish nothing! i have a problem with de polarity detecction! whe a people make a call near movil or other PSTN line and latter hang up Asterisk cant recognized this action |
| 15:05.38 | xnon_ | and continue the call! |
| 15:06.24 | Katty | hmmhesays: i want some. |
| 15:06.53 | tim27 | any consultant here can help me with a DID inroute problem with a sip provider... i use freePBX and i have paypal to pay |
| 15:07.33 | mercestes | I like his attitude...lol. |
| 15:08.20 | mercestes | What's your inroute problem? |
| 15:10.05 | hmmhesays | yeah people should be willing to pay more |
| 15:10.45 | mercestes | Indeed...but now that I'm offering him help.....he's not talking to me..:/ |
| 15:11.00 | hmmhesays | oh i'd help him too |
| 15:11.01 | mercestes | maybe he read the topic and went away. |
| 15:11.04 | hmmhesays | cause i'm poor |
| 15:11.11 | tim27 | mercestes: my provider sent the DID info in the TO: field of the sip headere |
| 15:11.33 | tim27 | they alway sent the principal DID number in the INVITE field |
| 15:12.11 | mercestes | tim27: By DID, you mean ..the number they dialed? |
| 15:12.18 | tim27 | they say they use the INVITE field for the account number who it also the pricipal number |
| 15:12.40 | tim27 | By DID i'm mean my 3 numbers on my sip account |
| 15:13.03 | tim27 | if i dial 18198502523 they will sent like this... |
| 15:13.28 | tim27 | INVITE: SIP 18198502520@192.168.1.101 TO: 18198502523@192.168.1.101 |
| 15:13.55 | mercestes | tim27: That looks right to me. |
| 15:14.00 | tim27 | and 18198502520 is the principal number |
| 15:14.33 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:14.39 | mercestes | tim27: here is a question, if *I* dial 18198502523.... |
| 15:15.03 | mercestes | tim27: Will you still get INVITE: SIP 18198502520@192.168.1.101 TO: 18198502523@192.168.1.101? |
| 15:15.38 | mercestes | tim27: Or are you getting that because you are dialing from 18198502520 or your CID is set to 18198502520. |
| 15:17.39 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:18.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:18.44 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 15:20.32 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:21.25 | mercestes | tim27: Maybe you should post yoru extensions.conf to pastebin.ca and then give us an example call, and an explanation of what yoru Sip/UId's are. |
| 15:23.25 | puzzled | hi |
| 15:23.49 | mercestes | hi, puzzled |
| 15:25.00 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 15:25.11 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:25.21 | *** join/#asterisk aa (n=SpShah@59.184.23.180) |
| 15:27.41 | hmmhesays | mercestes: thats not going to help much, if he's using freepbx, gonna be a mess |
| 15:28.53 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 15:29.45 | *** join/#asterisk Soul (n=Soul@82.102.1.42) |
| 15:30.03 | mercestes | hmmhesays: Yea, I know, but I hate to trust #freepbx to know everything. |
| 15:30.05 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 15:30.15 | *** join/#asterisk Tili (n=tili@202.133.67.157) |
| 15:30.31 | badcfe | hello, is there some way on detecting that the RTP has disapeared and then destroy the bridge and both call legs? |
| 15:31.14 | mercestes | badcfe: I am going to surmise that you are complaining about a "hung" sip call. |
| 15:31.28 | badcfe | mercestes: you can name it so |
| 15:31.33 | mercestes | badcfe: I'm also going to guess that if a sip call is "hung" then the RTP stream would still be present, but not transmitting anything worthwhile. |
| 15:32.09 | badcfe | mercestes: nope. i guess the RTP is not present -- as i physically breaks my eth cable. |
| 15:32.13 | mercestes | badcfe: Try canreinvite=yes. That should allow the phones to establish a direct RTP stream from peer to peer and "drop" once they are no longer recieving RTP. |
| 15:32.26 | mercestes | badcfe: you what? |
| 15:32.53 | badcfe | mercestes: to test i detach the ethernet cable. the sip call is still there, in asterisk. |
| 15:33.11 | badcfe | mercestes: since there is no BYE ofcourse. and i cant do re-INVITE here. |
| 15:33.30 | mercestes | badcfe: so you set up a call...and then unplugged the phone? |
| 15:33.33 | badcfe | mercestes: do you say that having * in between the peers induces this issue? |
| 15:34.06 | badcfe | mercestes: exactly. i unplug just to see if * detects RTP is gone and kills it. * doesnt. |
| 15:34.08 | mercestes | badcfe: no, I said that by removing * from between the peers you could resolve the issue by offloading it, I never said * was causing it. |
| 15:34.53 | mercestes | badcfe: *stares* What version of * are you running anyways? |
| 15:34.56 | badcfe | mercestes: * continues sending RTP to my sip friend even if it doesnt receive any. |
| 15:35.05 | badcfe | mercestes: * v 1.2 |
| 15:35.14 | mercestes | badcfe: 1.2.? |
| 15:35.18 | mercestes | badcfe: or BE? |
| 15:35.34 | badcfe | mercestes: not BE. Asterisk 1.2.13 |
| 15:35.38 | mercestes | .... |
| 15:35.41 | mercestes | Same version I'm running. |
| 15:35.45 | mercestes | I have *GOT* to see this happen. |
| 15:35.47 | mercestes | one sec. |
| 15:37.24 | tim27 | mercestes: http://pastebin.ca/270620 |
| 15:37.48 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 15:37.54 | mercestes | Well I'll be a monkey's uncle. |
| 15:37.56 | tim27 | my principal number is 8198502500... this is what happend on the server when i call the phone number 8198502523 |
| 15:38.34 | badcfe | mercestes: you get to provoke the same behaviour? |
| 15:39.00 | mercestes | badcfe: Yea, I did. Call still shows in show channels. |
| 15:39.03 | *** join/#asterisk lilalinux_ (n=plasma@80.69.41.2) |
| 15:39.06 | mercestes | badcfe: I'd bug report that one. |
| 15:39.15 | badcfe | mercestes: and your sip phone. what does it tell ya? |
| 15:39.31 | mercestes | badcfe: Nothing. Just silence. |
| 15:39.44 | mercestes | however, when the "other" end hangs up, it does destroy the channel |
| 15:39.55 | *** join/#asterisk spyda (n=sryan@hera.copi-rite.com) |
| 15:39.58 | mercestes | I am assuming that a full lock condition would only occure if both phones were unplugged. |
| 15:40.30 | mercestes | still bug report worthy if there isn't another bug similar to that one. |
| 15:40.51 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 15:41.27 | mercestes | Asterisk should be handling that for you in that case. |
| 15:41.34 | ESCulapio__ | I have problem with h323 out call |
| 15:41.50 | ESCulapio__ | Called 2090@192.168.1.18 |
| 15:41.50 | ESCulapio__ | Dec 7 11:40:31 ERROR[18785]: ast_h323.cxx:169 void PAssertFunc(const char*): Assertion fail: Null pointer reference, file h225_1.cxx, line 431, Error=115 |
| 15:41.50 | ESCulapio__ | Dec 7 11:40:31 ERROR[18785]: ast_h323.cxx:169 void PAssertFunc(const char*): Assertion fail: Invalid cast to non-descendant class, file h225_1.cxx, line 431, Error=115 |
| 15:41.50 | ESCulapio__ | Fallo de segmentación |
| 15:41.52 | mercestes | hmm...I recreated on 1.0.9 though. |
| 15:42.33 | mercestes | my 1.2.13 isn't online yet but I trust your seeing the same behavior. |
| 15:42.40 | De_Mon | What does this SIP traffic tell you? |
| 15:42.41 | De_Mon | http://pastebin.ca/270614 |
| 15:42.41 | De_Mon | http://pastebin.ca/270623 |
| 15:43.33 | De_Mon | Either asterisk, openser or Exchange is having issues and I can't really tell |
| 15:43.43 | bkw_ | I bet on asterisk |
| 15:45.24 | tim27 | mercestes: you have a clue |
| 15:45.59 | mercestes | tim27: Yes. Exactly what errant behavior are you seeing? What is getting screwed up? |
| 15:46.49 | tim27 | i'm not able to route call to did |
| 15:50.48 | *** join/#asterisk frenzy (n=frenzy@196.46.104.95) |
| 15:50.57 | DeeJayTwo | How can I detect incoming fax on a iax channel with asterisk 1.4? |
| 15:51.41 | mercestes | DeeJayTwo: I so asked you a question a long time ago...and even found a WIKI with an example on IAX2. |
| 15:51.56 | frenzy | using IAX2 wont receive a call unless I set the string in the context to s,1,Dial(SIP/123) |
| 15:52.14 | frenzy | I want it to go to 123,1,Dial(SIP/123) |
| 15:53.22 | DeeJayTwo | oh sorry didn't see the answer.. |
| 15:53.29 | frenzy | ? |
| 15:53.52 | mercestes | DeeJayTwo: Google asterisk cmd faxdetect and it's the WIKI hit. It even has an IAX example. |
| 15:54.06 | *** join/#asterisk mkrufky (n=mk@unaffiliated/mkrufky) |
| 15:55.13 | De_Mon | I have asterisk setup in a callcenter and want supervisors to have the ability to monitor calls, and to break into established SIP calls. What's the best way to make that happen? |
| 15:55.30 | mkrufky | is atcom harvesting email addresses from the asterisk / digium mailing lists ? |
| 15:55.52 | mkrufky | i got a solicitation from them about an X100P clone |
| 15:56.21 | zoa | i didnt get one yet |
| 15:56.25 | mercestes | De_Mon: try the operator Flash Panel. rewrite it..make it open source. Share with the rest of us. |
| 15:56.30 | mkrufky | i'll fwd it to you if you want to see |
| 15:56.53 | mkrufky | but i cant think of any way they would have gotten my email other than harvesting from the mailing lists |
| 15:57.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:57.57 | De_Mon | uh |
| 16:01.03 | mercestes | De_Mon: Oh, sorry, you said "best" not "easiest." Look up zapbarge and..um...sip barge I think. |
| 16:03.47 | *** join/#asterisk xnon (n=xnon@200.8.5.123) |
| 16:04.07 | mercestes | De_Mon: zapbarge and Chanspy. |
| 16:04.12 | mercestes | http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy |
| 16:05.08 | *** join/#asterisk grantm (n=grantm@207.88.78.2) |
| 16:06.04 | De_Mon | mercestes chanspy doesnt let you break into the call, just listen |
| 16:06.04 | DeeJayTwo | mercestes : it talks about the NVFaxDetect which I already use on 1.2 |
| 16:06.09 | DeeJayTwo | but it doesn't compile on 1.4 |
| 16:06.12 | DeeJayTwo | that's the problem.. |
| 16:07.37 | badcfe | mercestes: do you think that "RTP dead=>call hung" issue is address till release of * 1.4? |
| 16:08.51 | mercestes | badcfe: I don't know, I'd never heard of it before. Search the bug reports. |
| 16:10.09 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 16:10.54 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 16:14.03 | *** join/#asterisk jusse (n=jusse@200.106.89.112) |
| 16:14.55 | vader-- | does anyone know if it's possible to speed up or slow down the audio of a voicemail? or is there a way to enable that option? We have an old avaya system which would allow you to adjust the playback rate of the voicemail so you could catch what people were saying. |
| 16:17.02 | mercestes | vader--: that would require you to have that functionality in the audio player handling voicemail, which I believe is internal to *. |
| 16:17.37 | mercestes | vader--: So it's *possible* but not without altering internal code. |
| 16:18.35 | *** join/#asterisk queuetwo (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) |
| 16:18.41 | mercestes | vader--: What you would need is to have asterisk use an external player with that functionality either via commandline or API and then create system commands in extensions.conf to pass the audio player the required interaction to adjust it's playback speed. |
| 16:19.31 | mercestes | vader--: Or you could just get an Fxs card and make it interface with your avaya system voicemail and just use it. |
| 16:19.36 | *** part/#asterisk frenzy (n=frenzy@196.46.104.95) |
| 16:19.43 | mercestes | <PROTECTED> |
| 16:21.30 | vader-- | ya |
| 16:21.35 | vader-- | ok ill just tell them no then |
| 16:21.35 | vader-- | hehe |
| 16:21.36 | vader-- | :) |
| 16:21.59 | mercestes | lol |
| 16:23.43 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
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| 16:24.38 | jeremy_g | vader--:cool |
| 16:25.03 | vader-- | ? |
| 16:25.44 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 16:27.13 | *** join/#asterisk xnon (i=xnon@200.8.5.123) |
| 16:27.40 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 16:27.49 | Xen^ | mog : arroun ? |
| 16:28.13 | *** join/#asterisk Modcuts (n=Moducts@88-110-122-48.dynamic.dsl.as9105.com) |
| 16:29.40 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
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| 16:35.12 | *** join/#asterisk frk2 (n=none@202.70.153.211) |
| 16:35.15 | frk2 | hey people |
| 16:35.37 | frk2 | anyone willing to shed some light on my random E1 issues? |
| 16:36.03 | frk2 | and why on this particular PRI i keep on getting a Yellow alarm back? |
| 16:36.07 | *** part/#asterisk jm|work (n=jamiem@sentry.flags.co.uk) |
| 16:36.43 | frk2 | it runs PERFECTLY and then i get yellow alarms on all channels and the whole PRI resets itself |
| 16:37.01 | mercestes | frk2: Does it run perfectly after that?? |
| 16:37.04 | frk2 | yup |
| 16:37.09 | frk2 | runs perfectly after that as well |
| 16:37.13 | mercestes | frk2: Are you sure that's not normal? |
| 16:37.33 | frk2 | dude- all the calls drop out.. so its not normal |
| 16:37.38 | mercestes | frk2: there is an issue with PRi that sometimes results in hung channels and the "fix" for it is to call a full reset of the PRI. |
| 16:37.49 | mercestes | frk2: oh, well the dropping of calls is *not* normal. |
| 16:37.51 | frk2 | and besides... i have 4 other PRIs working.. just this one keeps on throwing the random yellow alarm |
| 16:38.11 | frk2 | dude this happens every 5 minutes |
| 16:38.14 | frk2 | sometimes 15 |
| 16:38.25 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:38.26 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 16:38.27 | mercestes | Ok ok ok, I just said it wasn't normal..:P |
| 16:38.45 | frk2 | so am i the only one with this issue? |
| 16:38.57 | mercestes | Is it possible something is wrong with the PRI itself? What happens if you move it to another card? |
| 16:39.17 | frk2 | I dont have another card |
| 16:39.35 | mercestes | frk2: I thought you said you had 4 other working PRis? |
| 16:39.47 | frk2 | yes at different clients |
| 16:39.48 | frk2 | :) |
| 16:39.53 | mercestes | frk2: ... |
| 16:39.56 | mercestes | frk2: oh. |
| 16:39.57 | frk2 | same telco though |
| 16:40.10 | mercestes | Switch out the card and see what happens. |
| 16:40.19 | frk2 | you think its a bad card? |
| 16:40.25 | mercestes | no. |
| 16:40.27 | frk2 | could it be a bad connection to the mdoem? |
| 16:40.35 | mercestes | It could. |
| 16:40.50 | mercestes | It could be monkeys peeing on your E1 line at this point, I don't know. |
| 16:41.00 | frk2 | hahah |
| 16:41.06 | mercestes | I am going to assume your configs are correct...because you ahve 4 other workign E1s |
| 16:41.10 | frk2 | i thought about little piranhas chewing away at the telco though |
| 16:41.16 | mercestes | and I'm going to assume the Telco is saying there is nothign wrong with their E1. |
| 16:41.34 | frk2 | yes, thats what the telco always says though |
| 16:41.48 | mercestes | so my next troubleshooting step would be to switch out the card. If it works, buy yourself a drink. If not, then I would reject the E1 with the Telco |
| 16:42.23 | frk2 | I was thinking of installing a Hardware PBX like panasonic just to show the telco |
| 16:42.24 | mercestes | frk2: I would not dismiss a possible error in your configs, but I'm also guessing that you probably copy pasted from one installation to another just to be safe. |
| 16:42.51 | frk2 | yes.. but im wondering if different exchanges of the same telco are running different configs |
| 16:43.49 | mercestes | frk2: Not if it's the same E1 setup |
| 16:44.09 | *** join/#asterisk danbrwn (n=danny@216.77.58.40) |
| 16:44.10 | mercestes | frk2: different on their end I'd wager. Not likely too different on yours |
| 16:44.55 | mercestes | frk2: I've seen T1's occassionaly just drop carrier for no damned reason and the Telco swear it was a clean connectino otherwise....until I got an engineer giong "holy crap! Your entire T1 just went down for a full 48 seconds! WTF? how long has it been doing that?" |
| 16:46.22 | *** join/#asterisk Modcuts- (n=Moducts@88-110-122-48.dynamic.dsl.as9105.com) |
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| 16:52.07 | *** part/#asterisk dorel__ (n=liran@80.179.31.43.static.012.net.il) |
| 16:52.42 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:58.57 | danbrwn | anyone with experience with aastra 480i sip |
| 17:02.16 | *** join/#asterisk yebo (n=private@cpe-76-168-249-132.socal.res.rr.com) |
| 17:05.21 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-176f334e43ec875f) |
| 17:08.32 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 17:09.29 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 17:10.19 | tRSS | I unplugged and replugged my TDM400P w/ 4 FXO and now I am getting a 'Freshmaker failed register test' error? Anyone knows about this problem? |
| 17:11.20 | mog | tRSS, it is possible sign of a bad card |
| 17:12.22 | tRSS | but it was working just fine. I took the card out, cleaned the connectors and put it back in |
| 17:13.34 | tRSS | i am going to unplug the card, since, I really don't need it anymore. |
| 17:13.43 | mog | you might try powering off and making sure there isnt a short or other issue |
| 17:16.08 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
| 17:16.24 | tRSS | i am going to do that now... but thanks for the help mog. really appreciate it. |
| 17:16.48 | Dr-Linux|work | how can i change voicemail application messages? |
| 17:18.04 | frk2 | sorry got distracted |
| 17:18.07 | frk2 | okay some more info |
| 17:18.27 | frk2 | This PC which has the T1 card has two other TDM-400ps attached to it |
| 17:18.39 | frk2 | its i take out the TDM 400ps, the E1 works a whole lot better |
| 17:18.45 | frk2 | the HDLC messages disappear |
| 17:18.56 | frk2 | ANd the yellow alarms are less frequent |
| 17:20.12 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:20.20 | *** join/#asterisk jorge__ (n=te_lo_me@adsl-149-57-198.mia.bellsouth.net) |
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| 17:42.26 | tim27 | danbrwn |
| 17:42.30 | tim27 | yout here |
| 17:44.29 | tim27 | <PROTECTED> |
| 17:48.39 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:49.28 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:50.56 | danp | what's the problem? |
| 17:52.06 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 17:53.14 | tim27 | danp: check this |
| 17:53.14 | tim27 | http://pastebin.ca/270733 |
| 17:53.38 | *** join/#asterisk lilalinux (n=plasma@dslb-084-058-201-099.pools.arcor-ip.net) |
| 17:53.46 | tim27 | when i compose 8198502523, they invite my server as 8198502500 , because they say its my principal number.... and they say they pass the DID infos in the TO: fields |
| 17:54.15 | *** join/#asterisk BSD_Tech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 17:55.06 | danp | hmm |
| 17:55.43 | danp | how does that mess things up? |
| 17:56.01 | tim27 | freepbx look in the invite field to route DID |
| 17:56.47 | tim27 | so i will need someone to do a special context that will look in the TO: field ... of the sip header to match the DID ... and route all call based on the DID that was composed... |
| 17:56.55 | danp | ahh, i see. is there a way to look inside the SIP headers? |
| 17:57.39 | danp | ${SIP_HEADER(TO):5:11} perhaps |
| 17:57.55 | tim27 | what the 5:11 mean |
| 17:58.45 | danp | that should give you back the prefix and extension of the number (like 5551212) |
| 17:58.54 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:59.51 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 18:00.50 | tim27 | to it possible to do it |
| 18:00.52 | danp | that's not right...you'd want 5:15 to get the whole number |
| 18:00.56 | danp | yeah |
| 18:01.28 | tim27 | danp: do you know free pbx |
| 18:02.13 | *** join/#asterisk Robyn (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 18:02.42 | danp | i'm familiar with asterisk |
| 18:04.26 | *** join/#asterisk conico (n=chatzill@85.107.210.206) |
| 18:06.55 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:08.14 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:08.19 | *** join/#asterisk OneBinary (n=d@64.161.217.18) |
| 18:08.29 | tim27 | so you are also familiar with freepbx... its just a GUI that make diaplan |
| 18:08.37 | tzanger | asterisk doesn't support sRTP yet does it? |
| 18:08.47 | OneBinary | this may be beyond the scope of this chan, but anybody know of a good place i can get royalty free music for MOH? |
| 18:08.56 | tzanger | freeplay.com? or org? |
| 18:09.06 | danp | i'm not familiar with freepbx specifically, no. it uses extensions.conf/ael, though, right? |
| 18:09.27 | EmleyMoor | Are there any notes on using a caller ID display unit on a Zap channel? |
| 18:09.43 | naftali5 | tim27, u need help w/ freepbx? |
| 18:10.14 | tim27 | yes |
| 18:10.16 | danp | OneBinary: http://www.google.com/search?q=royalty+free+hold+music maybe |
| 18:10.20 | naftali5 | sup |
| 18:10.43 | tim27 | i need routing DID from a sip provider based on the TO: field and the the INVITE field in a special context |
| 18:10.50 | OneBinary | danp: 99% of those sites you have to pay for the royalty free music (kind of an oxymoron) |
| 18:12.00 | naftali5 | tim27, your provider doesnt give you DIDs? |
| 18:12.13 | tim27 | yes |
| 18:12.19 | tim27 | but they call this virtual number |
| 18:12.31 | naftali5 | you tried DID inbound routing? |
| 18:12.36 | tim27 | but the prob is that they link all this to the main account |
| 18:12.39 | tim27 | http://pastebin.ca/270733 |
| 18:12.45 | tim27 | check this and you will understand |
| 18:13.16 | danp | OneBinary: i've used stuff from http://freeplaymusic.com before but it's pretty cheesy |
| 18:13.28 | tim27 | when i compose 8198502523 ... they send the invite 8198502500 ... because they say it my main account... |
| 18:13.30 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 18:13.44 | tim27 | they say they send DID infos in the TO: fields |
| 18:15.17 | naftali5 | one trunk set up for all your dids? |
| 18:15.22 | danp | http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels, see Crossed Incoming SIP Lines |
| 18:16.11 | tim27 | yes one trunk for all the dids |
| 18:16.16 | *** part/#asterisk ruinous (n=ruinous@ip24-251-205-253.ph.ph.cox.net) |
| 18:17.19 | *** join/#asterisk gJon (n=gjon@ein.cr.aptaculo.us) |
| 18:17.26 | tim27 | danp ... i think we are near a solution |
| 18:17.33 | naftali5 | tim27, i was going to suggest what danp said. It should work. give me a minute and i'll rewrite it for freepbx |
| 18:18.36 | tim27 | they are a very sucking provider... :((( |
| 18:18.42 | *** join/#asterisk Digivoice (n=ronaldo_@200.206.211.121) |
| 18:20.13 | tim27 | danp... that exactly what i need. to do... put my trunk it this special context... |
| 18:20.22 | tim27 | it will check the TO field... |
| 18:20.32 | tim27 | and route the did from the TO field... |
| 18:20.56 | danp | yeah |
| 18:21.40 | frk2 | tzanger asterisk DOES support srtp |
| 18:21.44 | frk2 | through libsrtp |
| 18:21.57 | frk2 | never got it to work myself though |
| 18:22.01 | frk2 | need a srtp phone |
| 18:23.56 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.mn.comcast.net) |
| 18:24.20 | tzanger | frk2: oh I did not know that |
| 18:24.26 | tzanger | do the polycoms have sRTP support? |
| 18:24.28 | frk2 | yeah man |
| 18:24.34 | frk2 | i think the 601 does |
| 18:24.41 | frk2 | snoms got it |
| 18:25.01 | tzanger | nice |
| 18:25.43 | frk2 | http://www.voip-info.org/wiki/view/Asterisk+encryption |
| 18:26.01 | tzanger | thank you |
| 18:26.41 | naftali5 | tim27, if you didnt get it set up yet |
| 18:26.45 | frk2 | actually |
| 18:26.50 | frk2 | this is the shit you really wanna see |
| 18:26.51 | frk2 | http://www.e164.org/wiki/AsteriskSRTP |
| 18:26.58 | naftali5 | [from-babytel] exten => s,1,Set(FROM_DID= ${SIP_HEADER(TO):5:11}) exten => s,n,Goto(from-trunk,s,1) |
| 18:27.52 | tzanger | there were some references before to recording a LOT of simultaneous calls with asterisk |
| 18:30.07 | tzanger | I thought it was ManxPower |
| 18:31.52 | tim27 | naftali5: doing it like this will not overwrite... all the FROM_DID... |
| 18:32.21 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 18:32.27 | naftali5 | no the start of [from-babytel] means make this new context in extensions-custom |
| 18:32.37 | naftali5 | then put that as the incomin context for that trunk |
| 18:32.57 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.246.telnor.net) |
| 18:32.57 | tim27 | ha ok |
| 18:33.03 | tim27 | so all things will work |
| 18:33.54 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:40.47 | hmmhesays | you figure out your problem tim27? |
| 18:47.31 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-215-87.cablep.bezeqint.net) |
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| 18:50.39 | Katty | hmmhesays: i'm getting ready to start on my project! |
| 18:53.00 | mercestes | Katty: Good luck with that! :) |
| 18:53.10 | SheriF_SpacE | Katty: if i may ask what kind of project ? |
| 18:55.32 | Katty | SheriF_SpacE: the kind of project that makes people ask questions. ;) |
| 18:55.53 | mercestes | She's trying to take over the world. |
| 18:55.58 | Katty | naturally. |
| 18:56.59 | Katty | SheriF_SpacE: it has to do with centericq |
| 18:57.11 | mercestes | It involves the evolution of a new language based loosely upon english called "Kitty" in which the consonant noun combination "mew" is gratuitiously integrated into everyday words. |
| 18:57.14 | rob0 | I thought the world was already overtaken? |
| 18:57.27 | mercestes | It's a mew standard of communication. |
| 18:57.38 | SheriF_SpacE | Katty: lol i can see best luck :-) |
| 18:57.41 | mercestes | Hey katty, comewnication. add it to the dictionary..:) |
| 18:57.54 | rob0 | mewsic |
| 18:57.59 | Katty | rob0: hush, dear. |
| 18:58.01 | *** join/#asterisk KD-Misafir270 (n=KD-Misaf@85.102.82.179) |
| 18:58.20 | mercestes | mew-hahaha...It's an evolution. |
| 18:58.48 | mercestes | mew cannot resist. |
| 18:59.05 | mut | mew 2! |
| 18:59.48 | mercestes | mewcestes. |
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| 19:00.31 | *** join/#asterisk beyond (n=beyond@200-155-185-1.static.spo.ifx.net.br) |
| 19:02.06 | *** join/#asterisk evandro (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
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| 19:02.34 | BSDTech | ok in a dial plan |
| 19:03.14 | BSDTech | how do I make a exten check a id and pwd from a conf file with out a agi ? |
| 19:03.28 | *** join/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 19:03.42 | BSDTech | I am writing a custom chanspy |
| 19:03.46 | BSDTech | app |
| 19:03.59 | BSDTech | and it needs to check user id and pwd to allow |
| 19:04.12 | De_Mon | BSDTech you can do that in the dialplan |
| 19:04.15 | BSDTech | and I want to store them in chanspy.conf |
| 19:04.46 | De_Mon | BSDTech er, it would be easier to use a database |
| 19:04.50 | BSDTech | how ? |
| 19:05.04 | De_Mon | or use agent pins/passwords |
| 19:05.08 | hoobastoob1 | I am doing a ps aux | grep asterisk and I am seeing something strange.... http://pastebin.ca/270816 |
| 19:05.27 | hoobastoob1 | what causes these additional processes to start /usr/sbin/asterisk -vvvg -c |
| 19:05.41 | BSDTech | but not all agents get chanspy |
| 19:05.43 | hoobastoob1 | asterisk is running as 3723 |
| 19:05.49 | BSDTech | I want it for managers only |
| 19:06.23 | De_Mon | then make agents with an agentiD 1XXX is a manager or something simple |
| 19:06.42 | mercestes | hoobastoob1: Asterisk is running as root... |
| 19:07.05 | *** join/#asterisk evandro (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 19:07.22 | mercestes | hoobastoob1: What is the strangeness, other than your running asterisk as Root? |
| 19:07.26 | De_Mon | pbx*CLI> *** glibc detected *** double free or corruption (!prev): 0x081866a8 ** |
| 19:07.29 | De_Mon | * |
| 19:07.40 | naftali5 | hoob, the easiest way to run as asterisk is amportal restart |
| 19:07.55 | hoobastoob1 | the multiple processes are what confuse me. should there not only be one? |
| 19:08.17 | mercestes | hoobastoob1: Depends on your load. If * gets overburdeneed it seems to manage to chain off new processes of itself. |
| 19:08.41 | mercestes | hoobastoob1: Somewhere around 10-20 you start running into problems. |
| 19:08.59 | BSDTech | I want it to be like disa |
| 19:09.12 | mercestes | s/start/can start |
| 19:09.23 | Katty | hmmhesays: http://centericq.de/docs/readme.php?mode=1&chapter=9.2.1 :>>> |
| 19:10.28 | BSDTech | http://pastebin.ca/270825 |
| 19:10.36 | hoobastoob1 | mercestes: do you know, is that CPU or ram that gets overburdened? |
| 19:10.49 | BSDTech | this is what I am doing now but want to cut out the perl agi |
| 19:11.29 | mercestes | hoobastoob1: Depends on where yoru bottlenecks go. I'm not 100% sure why * chains off new processes. It could be whenever it fails to respond, it could be just in response to load, it could be threading. I don't know. |
| 19:12.15 | hoobastoob1 | interesting... but i should be able to kill all processes but the original one and be ok right? |
| 19:12.42 | mercestes | hoobastoob1: ...erm. Well, attaching to one asterisk process via CLI will give you output from all * processes, so they are linked together. |
| 19:12.58 | hoobastoob1 | interesting... |
| 19:13.02 | mercestes | hoobastoob1: I'm not certain killing any one of them would be wise...they should self terminate as load decreases. |
| 19:13.13 | hoobastoob1 | ok |
| 19:13.17 | hoobastoob1 | i will see if it does, thank you |
| 19:13.22 | mercestes | hoobastoob1: NP..let us know. |
| 19:13.24 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 19:13.47 | BSDTech | http://pastebin.ca/270827 |
| 19:14.00 | BSDTech | sorry had to fix a few things |
| 19:14.06 | De_Mon | BSDTech yave you looked at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Authenticate |
| 19:14.11 | BSDTech | but how would I rm the agi |
| 19:15.27 | BSDTech | ok but 2 parts have to be stored the id and the pwd |
| 19:15.54 | mercestes | BSDTech: why? |
| 19:16.16 | BSDTech | managers are given id's and pins |
| 19:16.22 | mercestes | why? |
| 19:16.25 | BSDTech | how the company wants it |
| 19:16.43 | mercestes | Can't just use one pin for all managers? Not like your gonig to do any logging of who listens to what channel. |
| 19:16.55 | BSDTech | nope |
| 19:17.09 | BSDTech | they want every manager to have his own id and pin |
| 19:17.10 | mercestes | So there is no reason to have multiple ID's and PIN's. But you can still do tha tin exten.conf |
| 19:17.14 | De_Mon | BSDTech do your managers have a way to join a call in progress? |
| 19:17.35 | mercestes | BSDTech. have an extension background a (please enter your PIN number). |
| 19:17.41 | hmmhesays | manager redirect into a meetme conference |
| 19:17.59 | mercestes | BSDTech: then exten => ID_NUMBER,1,AUTHENTICATE(PIN_NUMBER) |
| 19:18.09 | De_Mon | I have it setup so managers phones are able to chanspy, managers just have to keep their phones secure. |
| 19:18.12 | mercestes | BSDTech: Basically use the "ID" as an extension. |
| 19:18.30 | BSDTech | they just want chanspy for now and told me how they want it and I am trying to code it if you look at the pastebin |
| 19:18.42 | BSDTech | cant use exten |
| 19:18.46 | mercestes | why not? |
| 19:19.01 | BSDTech | managers have to be able to loginfrom any phone and enter thier id and pwd |
| 19:19.11 | mercestes | Right. |
| 19:19.27 | mercestes | sec, let me code it in pastebin so you can see. |
| 19:19.44 | BSDTech | edit what I have done to understand |
| 19:19.49 | De_Mon | exten => Read(${MANAGERID} ... s,n1,GOTO(${MANAGERID}) |
| 19:20.01 | *** join/#asterisk jpeeler (n=jpeeler@130-127-45-101.chouse.resnet.clemson.edu) |
| 19:20.01 | De_Mon | oh man I jumbled that up |
| 19:20.30 | De_Mon | exten => s,1,Read(${MANAGERID} ... exten => s,1,GOTO(${MANAGERID},1) |
| 19:20.33 | BSDTech | I want to get away from agi |
| 19:21.04 | BSDTech | did you look at what I have working |
| 19:21.09 | De_Mon | exten => <managerid>,1,Authenticate(PIN_NUMBER) |
| 19:21.24 | mercestes | http://pastebin.ca/index.php |
| 19:21.28 | mercestes | damnit |
| 19:21.31 | De_Mon | are you paying any attention to what we are suggesting |
| 19:21.46 | mercestes | http://pastebin.ca/270835 |
| 19:21.49 | BSDTech | yes but its not making sense from what I have learned |
| 19:21.52 | De_Mon | I like the authenticate(/etc/asterisk/${MANAGERID} method myself |
| 19:22.15 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 19:22.24 | mercestes | 1111, 2222, 3333 are your manager "IDs" |
| 19:22.34 | mercestes | 1234, 4567, and 7890 are your PIN numbers. |
| 19:22.58 | *** join/#asterisk alerios (n=alerios@190.24.97.151) |
| 19:23.35 | BSDTech | I guess the way I do it will have to do |
| 19:23.43 | BSDTech | with the agi |
| 19:23.44 | mercestes | .... |
| 19:23.52 | mercestes | something is wrong with your brain. |
| 19:23.57 | BSDTech | adding loads of exten is not the answer |
| 19:24.18 | BSDTech | I want it to read the id and pwd from a file |
| 19:24.21 | De_Mon | no, writing modules to do something the dialplan is perfictly capiable of doing is much better... |
| 19:24.23 | mercestes | oh. |
| 19:24.25 | BSDTech | its the proper way |
| 19:25.47 | BSDTech | let me show you more |
| 19:25.50 | BSDTech | hold on |
| 19:27.08 | BSDTech | http://pastebin.ca/270843 this is what I modeled it after |
| 19:27.17 | De_Mon | http://pastebin.ca/270844 |
| 19:27.20 | De_Mon | how about that |
| 19:28.42 | De_Mon | if your managers all have agent ids that follow a specific patern you can test if the ID matches that patern and then authenticate using the agentID pin |
| 19:30.04 | BSDTech | I guess I will have to rewrite my agi in php . my disa module works perfectly allowing staff to work from home when neede |
| 19:30.21 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 19:30.24 | BSDTech | it checks the voicemail.conf for exten and pbd |
| 19:31.12 | BSDTech | ow ell thanks |
| 19:31.47 | BSDTech | and your free to use the disa module if you like |
| 19:33.32 | *** join/#asterisk ComPuTeR (n=ELif__@88.224.166.157) |
| 19:33.57 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 19:34.14 | BSDTech | exten => s,n,Authenticate(/etc/asterisk/voicemail.conf|${id_num}|${id_pwd}) whold this not work |
| 19:34.42 | BSDTech | but change voicemail to chanspy ? |
| 19:34.43 | *** join/#asterisk Ashura (n=Ashura@adsl-ull-70-221.49-151.net24.it) |
| 19:34.56 | Ashura | hello |
| 19:36.53 | BSDTech | wout that auth line work ? |
| 19:36.54 | SheriF_SpacE | hello |
| 19:37.20 | BSDTech | exten => s,n,Authenticate(/etc/asterisk/chanspy.conf|${id_num}|${id_pwd}) whold this not work |
| 19:38.18 | *** join/#asterisk atSquiGgs (n=jwaters@brtransport.com) |
| 19:38.24 | atSquiGgs | hey guys |
| 19:38.52 | BSDTech | no responce O well |
| 19:39.35 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:40.51 | atSquiGgs | can someone explain how you get over users wanting to have 20 buttons they can push to dial other people in the company when you switch to voip phones? Just use the directory or memorize the extentions? |
| 19:41.28 | awannabe | atSquiGgs: yeah or use a phone with programmable keys. people are lazy and want speed dials! |
| 19:41.59 | atSquiGgs | lol. The polycom 601 has the expansion module, but that is 500 bucks for a phone a dock. Are there any better solutions? |
| 19:42.07 | awannabe | snom 360 |
| 19:42.11 | atSquiGgs | do you think it will hold back people accepting the technology? |
| 19:42.11 | awannabe | 200 for phone, sidecar is 100 |
| 19:42.40 | atSquiGgs | for a polycom 601? |
| 19:42.55 | awannabe | no no, a snom 360 |
| 19:43.47 | naftali5 | grandstream gxp2000 has sidecar option, cheap |
| 19:44.07 | awannabe | yeah, IMHO the grandsstream look like a cheap POS |
| 19:44.21 | atSquiGgs | grandstream phones suck! |
| 19:44.25 | atSquiGgs | well I think so anyway |
| 19:44.43 | naftali5 | true, but some ppl like the old ksu look&feel |
| 19:46.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:46.11 | BSDTech | but you are limited to 3 sidecars |
| 19:46.15 | BSDTech | right now |
| 19:46.27 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 19:46.31 | BSDTech | and 2 with the grandstream gxp2000 |
| 19:46.52 | atSquiGgs | they are coming from a partner system so it's very different |
| 19:47.27 | atSquiGgs | the partner 18d |
| 19:47.38 | atSquiGgs | do you guys like the polycom 501 and 601 phones? |
| 19:48.11 | awannabe | ahh |
| 19:48.34 | *** part/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 19:48.46 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.18) |
| 19:49.59 | atSquiGgs | so what do these large companies do? just make everyone remember the extensions? |
| 19:50.37 | BSDTech | I like them alot |
| 19:50.48 | BSDTech | 301/501/601/651 |
| 19:50.53 | BSDTech | all great phones |
| 19:51.26 | atSquiGgs | whats the point of having mulitple lines? don't you just make them all the same extension anyway? |
| 19:51.51 | atSquiGgs | Like why buy the 601 over the 501, just for extra lines? and you can use the sidecar? |
| 19:52.04 | naftali5 | yes sometimes, but it makes it easier to juggle calls |
| 19:52.26 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 19:52.33 | atSquiGgs | can you explain that? I think I'm just missing something here |
| 19:54.31 | naftali5 | get a call, next call rings on the next key. you can place first on hold take second |
| 19:55.20 | atSquiGgs | what about incoming/outgoing lines? I have two customers that I'm thinking about this for. One has 3 lines now, that would be using a Digium TDM04B(fxo) card right? What about a customer that has 12 lines? They are coming from the T1, what card would I buy for that? 3 of the TDM cards? or can I get one card that will handle it all? |
| 19:55.31 | naftali5 | see the first still on hold on the first key, hit transfer, then a key on your sidecar to xfer to another exten, get back to key 1 |
| 19:55.38 | *** join/#asterisk marlow (n=marlow@87.198.132.2) |
| 19:55.53 | atSquiGgs | thank you naftali5 |
| 19:56.04 | naftali5 | digium has T1 cards |
| 19:56.13 | marlow | howdy folkgs |
| 19:56.18 | atSquiGgs | howdy |
| 19:57.08 | atSquiGgs | what card would you recommend? or anyone for that matter |
| 19:57.37 | naftali5 | http://www.digium.com/en/products/hardware/digitalcards.php |
| 19:58.09 | naftali5 | or if you already have a channel bank for the T1, you may want to get TDM2400 |
| 19:58.32 | naftali5 | http://www.digium.com/en/products/hardware/tdm2400p.php handles 24 analog lines |
| 19:58.45 | atSquiGgs | We already have a channel bank |
| 19:58.47 | *** join/#asterisk marlow (n=marlow@87.198.132.2) |
| 19:59.18 | atSquiGgs | so that beast just plugs right into our t1 channel bank? |
| 19:59.18 | *** join/#asterisk ming_zym (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 20:00.00 | naftali5 | if it works well, you may want to just go with the tdm2400p which can handle 24 analog lines (from channel bank or other) |
| 20:01.33 | atSquiGgs | I think that is what I would use for the T1 setup |
| 20:01.59 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 20:02.11 | *** join/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 20:02.26 | Qwell[] | only problem there, is if you ever add more T1s, it's much harder to stick two of those cards in the box |
| 20:02.38 | hmmhesays | use external gateways |
| 20:02.39 | Qwell[] | can get much higher density with T1 cards |
| 20:02.41 | atSquiGgs | so I could buy one of those cards and get this http://www.voipsupply.com/product_info.php?products_id=1164 and then plug whatever into it? |
| 20:03.04 | atSquiGgs | Qwell I see what you are saying |
| 20:03.22 | hmmhesays | 2500 bucks for a 24 port fxs fxo sip gateway |
| 20:03.44 | atSquiGgs | hmmhesays - do you mean for the card alone? |
| 20:04.08 | hmmhesays | i'm saying instead of a card in ther asterisk box |
| 20:04.14 | afrosheen | wouldn't it be cheaper to go Sangoma A101dm to a PRI box locally? |
| 20:04.31 | afrosheen | but that requires a card.. |
| 20:04.48 | atSquiGgs | hmmhesays - i'm not sure what you are talking about, can you link something? |
| 20:05.14 | hmmhesays | weren't you looking for a way to get 24 analog lines to your asterisk box? |
| 20:05.34 | atSquiGgs | yeah, I just didn't understand what you mean by a sip gateway |
| 20:05.40 | atSquiGgs | isn't that an asterisk box |
| 20:05.49 | atSquiGgs | as you can see I'm new to this |
| 20:06.32 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 20:07.01 | afrosheen | atSquiGgs, there would be a standalone box that the 24 analog lines go into..then asterisk connects to that via a special card |
| 20:07.16 | afrosheen | that's one way to do it at lease |
| 20:07.20 | atSquiGgs | oh ok |
| 20:07.21 | afrosheen | s /lease/least |
| 20:07.29 | hads | Gateway usually refers to PSTN in Ethernet out. |
| 20:07.32 | atSquiGgs | right now they only have 12 lines, so it would take awhile to get to 24 |
| 20:07.45 | atSquiGgs | this is what you are talking about right, http://www.voipsupply.com/product_info.php?products_id=1116 |
| 20:08.04 | hmmhesays | i was talking about a stand alone sip gateway what handles the pstn connection, then it is sip between that box and asterisk |
| 20:08.15 | hads | http://www.voipsupply.com/index.php?cPath=94_286 |
| 20:09.43 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:10.11 | *** part/#asterisk alerios (n=alerios@190.24.97.151) |
| 20:10.25 | BSDTech | exten => s,n,Authenticate(/etc/asterisk/chanspy.conf|${id_num}|${id_pwd}) whold this not work |
| 20:10.42 | BSDTech | will it not read into the fill and match the info ? |
| 20:10.46 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 20:10.55 | [hC] | anyone experienced a problem w polycom where the phone starts to get really slow and sluggish, for example when dialing, then soon after reboots/crashes? |
| 20:10.58 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 20:11.19 | BSDTech | HC what ver of firmware what model poly |
| 20:11.23 | afrosheen | [hC], yeah details |
| 20:11.45 | [hC] | 1.6.7 and an ip601 w/ 3 expansion modules |
| 20:12.01 | Katty | hmmhesays: i got it to work :> |
| 20:12.09 | Katty | hmmhesays: asterisk is spamming my yahoo client ^_^ |
| 20:12.16 | hmmhesays | Katty: great |
| 20:12.22 | Katty | hmmhesays: it's hotttt. |
| 20:12.25 | Katty | hmmhesays: in a very slow way. |
| 20:12.31 | hmmhesays | will you pet my brain? |
| 20:12.47 | [hC] | BSDTech/afrosheen: any ideas why or how to fix? |
| 20:12.47 | BSDTech | sounds like te firmware might have a bug |
| 20:13.07 | BSDTech | I use 2.0.1 firmware |
| 20:13.10 | [hC] | ahh |
| 20:13.26 | [hC] | has the 2.x stuff been stable enough? |
| 20:13.26 | BSDTech | and looking for 2.03 wich is out |
| 20:13.28 | afrosheen | expansion modules? are you talking about the sidetrain thing for secretaries? |
| 20:13.34 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 20:13.46 | [hC] | afrosheen: yep |
| 20:13.46 | BSDTech | they are called sidecars |
| 20:13.56 | Katty | now if only i knew what protocol the microsoft IM server thingy used. |
| 20:13.57 | afrosheen | sidecar, sidetrain, sidetrack, whatever |
| 20:13.59 | hmmhesays | awesome |
| 20:14.10 | hmmhesays | Katty, sip capable i believe |
| 20:14.23 | mercestes | From what I've read, it is sip |
| 20:14.23 | afrosheen | have you done the vulcan neck pinch to clear the flash memory yet? |
| 20:14.26 | Katty | hmmhesays: mew? |
| 20:14.36 | hmmhesays | Katty: google msn sip |
| 20:14.39 | Katty | hmmhesays: i just want to send ims that says ${callerid} blahblahblah etc |
| 20:14.48 | hmmhesays | you'll find some guides on using msn messenger with asterisk |
| 20:14.58 | Katty | hmmhesays: but..but...that's slowwwwwwwwww. |
| 20:15.05 | hmmhesays | slow? |
| 20:15.06 | Katty | hmmhesays: it'll take forever to get through the msn network. |
| 20:15.13 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 20:15.14 | hmmhesays | what? |
| 20:15.14 | Katty | hmmhesays: i want it nownownow! |
| 20:15.25 | hmmhesays | it should be near instant |
| 20:15.27 | Katty | it takes like 10 seconds to get it from asterisk to the yahoo client |
| 20:16.00 | Katty | i'm doing an echo "foobar" | centericq asdkjflaksjdf |
| 20:16.00 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 20:16.07 | De_Mon | BSDTech thats what the docs say it will do |
| 20:16.15 | Katty | is that not a good way to do it? |
| 20:16.17 | afrosheen | [hC], that question about the neck pinch was for you |
| 20:16.28 | BSDTech | ook |
| 20:16.44 | Katty | hmmhesays: so i figured if i could setup an internal IM server... |
| 20:16.44 | [hC] | afrosheen: hah no havent done that |
| 20:16.45 | De_Mon | BSDTech authenticate just reads a list of valid pins from <file> |
| 20:16.47 | Katty | hmmhesays: it would go faster. |
| 20:16.54 | Katty | hmmhesays: and not spaz out if our internet was down, etc. |
| 20:17.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-37-212.red.bezeqint.net) |
| 20:17.14 | *** join/#asterisk Simplix_ (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 20:17.15 | De_Mon | BSDTech it doesnt match it with user names, so 1 file = userid and many valid pins |
| 20:17.34 | BSDTech | crap ok |
| 20:17.34 | Katty | hmmhesays: irc is too complicated for our receptionist |
| 20:18.07 | Un1x | Dec 8 03:21:10 WARNING[10421]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'inbound' |
| 20:18.10 | De_Mon | BSDTech thats why I was using /etc/asterisk/managers/${ext_num} |
| 20:18.12 | Un1x | why do i keep getting that :( |
| 20:18.39 | rob0 | IRC is too receptive to complications! |
| 20:19.09 | afrosheen | [hC], if you can figure it out, give that a shot, it's the best reset for the phone |
| 20:19.21 | hmmhesays | Katty: set up a jabber server |
| 20:19.24 | hmmhesays | internal |
| 20:19.25 | [hC] | afrosheen: will do. i may do that then upgrade to 2.x |
| 20:21.41 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 20:24.09 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
| 20:25.06 | *** join/#asterisk Blu3 (n=david@208.55.122.7) |
| 20:25.13 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-115-171.vic.westnet.com.au) |
| 20:25.29 | *** join/#asterisk ToTo (n=ToTo@host154-166-dynamic.0-87-r.retail.telecomitalia.it) |
| 20:27.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:28.44 | mercestes | katty: how come you never talk to me on yahoo anymore? |
| 20:31.53 | hads | Un1x: Because there was a timeout but there was no t extension perhaps? |
| 20:32.20 | Un1x | w00t |
| 20:32.35 | Un1x | i finished up cleaning my extensions.conf and got disa to work as well with a perfectly well context :) |
| 20:32.56 | hmmhesays | disa is cool |
| 20:33.22 | Un1x | hey is there any free Hello msgs from disa by default |
| 20:33.32 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 20:34.28 | hmmhesays | Un1x: use cmd playback before disa |
| 20:34.59 | *** join/#asterisk dgergo (i=applet@host-87-242-9-196.prtelecom.hu) |
| 20:35.06 | dgergo | helloo |
| 20:35.46 | *** part/#asterisk dgergo (i=applet@host-87-242-9-196.prtelecom.hu) |
| 20:35.56 | *** part/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 20:36.00 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 20:36.13 | Un1x | hmmhesays, i know that im asking about the soundfiles lol |
| 20:36.58 | hmmhesays | go to the wiki and look at the sound files additional page |
| 20:38.06 | Un1x | link please ;P? |
| 20:38.17 | rob0 | ~wikis |
| 20:38.19 | jbot | [wikis] http://www.voip-info.org |
| 20:38.22 | hmmhesays | google sound files additional |
| 20:38.34 | *** part/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 20:39.32 | *** join/#asterisk klasstek_ (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 20:39.35 | Un1x | heh there is nothing good |
| 20:40.48 | afrosheen | what about the super high quality sound files..where did those go |
| 20:41.17 | *** join/#asterisk klasstek__ (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 20:41.35 | Un1x | where are those |
| 20:41.58 | atSquiGgs | what a good phone for a conference room? |
| 20:42.59 | Un1x | Polycom Soundstation |
| 20:43.14 | atSquiGgs | the 4000? or 100? |
| 20:43.33 | hmmhesays | why do you need a super high quality file when it gets transcoded down anyway |
| 20:45.39 | ManxPower | Me: "So where is the other end of the Frac Data T-1 going?". Telco: "You don't have a Frac Data T-1, you have 6 pots lines and 384k of internet." Me: *sigh* |
| 20:46.11 | atSquiGgs | lol |
| 20:46.57 | ManxPower | Maybe next time the office admin guy will listen to me when I say "run all telecom contracts by me before signing them" |
| 20:48.03 | Un1x | i dont know wich one is analogue one wichever one is analogue is the one i'll get because |
| 20:48.06 | Un1x | i dont have a digital card |
| 20:48.09 | Un1x | i got the TDM400P |
| 20:53.15 | *** join/#asterisk ComPuTeR (n=WoLLe@88.240.230.116) |
| 20:57.31 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:57.51 | brian | hey is there anyway to make the DTMF in a meetme be quieter |
| 20:59.06 | afrosheen | meetme seems like a forgotten module at this point :( |
| 20:59.23 | ManxPower | brian: I didn't know that DTMF was supposed to be transported in MeetMe |
| 20:59.23 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 21:02.40 | brian | well it is |
| 21:03.21 | brian | there has to be a way to silence the channels |
| 21:03.35 | *** join/#asterisk tRSS (n=tRSS@124.29.254.12) |
| 21:03.37 | blitzrage | just mute them (listen only) |
| 21:04.02 | brian | its kind of hard to mute them BEFORE they press the DTMF |
| 21:04.23 | brian | the module shouldn't really broadcast it to all the participants :( |
| 21:04.37 | ManxPower | are you sure it's not just a DTMF mode problem. |
| 21:04.46 | ManxPower | i.e. Asterisk is set for rfc2833 and the device is inband? |
| 21:05.05 | ManxPower | that would explain meetme hearing dtmf |
| 21:05.26 | brian | it might be |
| 21:05.48 | brian | but i also have it set that if a user presses a certain button it exits the meetme conference |
| 21:05.56 | brian | and then returns them back to it |
| 21:06.05 | brian | like to get user count |
| 21:06.16 | ManxPower | brian: does that work? |
| 21:06.25 | brian | yeah |
| 21:06.28 | brian | but it broadcast the DTMF |
| 21:06.38 | brian | how do I change the DTMF mode though its probably that |
| 21:06.53 | ManxPower | change it on the phone and set dtmfmode=rfc2833 in sip.conf |
| 21:07.41 | brian | it is already rfc2833 |
| 21:07.55 | brian | u sure i don't need inband |
| 21:09.26 | brian | ManxPower: i'm not calling using a sip phone |
| 21:14.19 | linlin | what is an MGCP gateway ? |
| 21:14.22 | *** join/#asterisk bloch (n=lol@67.170.7.94) |
| 21:16.48 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-50-134.w90-5.abo.wanadoo.fr) |
| 21:17.49 | brian | I try to change to inband and nothing works at all. |
| 21:18.28 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:19.22 | *** join/#asterisk hads_ (n=hads@mail.nice.net.nz) |
| 21:21.07 | *** join/#asterisk mhz121 (n=mhz121@h-72-245-152-12.nycmny83.covad.net) |
| 21:23.37 | ManxPower | brian: well what tech are you using? |
| 21:24.10 | ambriento | ~pastebin |
| 21:24.12 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:24.12 | ManxPower | brian: Asterisk and phones do not magically figure out what dtmf mode to use. You need to set it on the device and on the server. |
| 21:26.21 | danp | might i also recommend pastie: http://pastie.caboo.se |
| 21:27.40 | blitzrage | no way, .ca is better! :) |
| 21:27.47 | *** join/#asterisk ManxPowe1 (n=manxpowe@68-114-99-147.dhcp.slid.la.charter.com) |
| 21:28.17 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 21:28.44 | zoa | tsss |
| 21:28.46 | zoa | wussie |
| 21:28.47 | mercestes | .ca eh? |
| 21:29.22 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 21:29.26 | afrosheen | canada FTW |
| 21:29.35 | mercestes | canada is silly |
| 21:30.49 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 21:30.53 | mercestes | http://s2.photobucket.com/albums/y25/Mercestes/?action=view¤t=1138536287343.jpg |
| 21:31.17 | mercestes | oops |
| 21:31.18 | mercestes | http://i2.photobucket.com/albums/y25/Mercestes/1138536287343.jpg?t=1165527040 |
| 21:31.20 | mercestes | There :) |
| 21:34.04 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:34.05 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:34.59 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 21:35.05 | StyleWarz | Hey guys |
| 21:36.18 | StyleWarz | is there any hardware for asterisk which allows me to attach isdn phones directly to my asterisk pbx? it looks kinda stupid if i sell my customer an asterisk solution where he needs a new ntba (dunno what that is in english) for every two phones :) |
| 21:36.25 | zoa | yeah |
| 21:36.27 | zoa | there are |
| 21:36.39 | zoa | for example the digium b410p |
| 21:37.01 | StyleWarz | aaaah |
| 21:37.05 | zoa | mr gayanalfisting |
| 21:37.06 | StyleWarz | i almost forgout about digium *headbang* |
| 21:37.16 | StyleWarz | hrhr ;) |
| 21:37.16 | zoa | there are more cards |
| 21:37.16 | zoa | depends on how much you want to spend |
| 21:37.34 | zoa | all HFC cards should work |
| 21:37.41 | zoa | the cheapest one is some USB thing |
| 21:37.45 | zoa | gazelle or so |
| 21:37.48 | *** join/#asterisk bageddy (n=chatzill@82.246.145.238) |
| 21:38.12 | StyleWarz | zoa: well my customer is quite in the mood to spend good money for a good solution |
| 21:38.21 | StyleWarz | zoa: but digium only has 4-port cards :( |
| 21:38.25 | zoa | Gazel 128 USB modem |
| 21:38.27 | zoa | is the only 1 port |
| 21:38.37 | zoa | the best one by far is the digium |
| 21:38.40 | atSquiGgs | see ya guys |
| 21:38.46 | zoa | because it has the on board octasic |
| 21:38.57 | StyleWarz | zoa: i want to connect 6 isdn phones, 1 isdn to analog terminator, and one for outgoing calls over PTP |
| 21:39.03 | zoa | aha |
| 21:39.09 | *** part/#asterisk atSquiGgs (n=jwaters@brtransport.com) |
| 21:39.16 | zoa | check the big beronet cards then |
| 21:39.20 | StyleWarz | zoa: so i would need 2x b410p, right? |
| 21:39.21 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 21:39.23 | StyleWarz | aaah |
| 21:39.25 | zoa | yeah |
| 21:40.07 | StyleWarz | zoa: with the beronet one's i won't need an NTBA? |
| 21:40.42 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 21:40.43 | zoa | i have no clue what an NTBA is |
| 21:41.03 | StyleWarz | hrhr |
| 21:41.28 | StyleWarz | etwork termination for basic access |
| 21:41.45 | StyleWarz | network even |
| 21:42.14 | zoa | NT i know |
| 21:42.18 | zoa | they can do NT |
| 21:42.22 | zoa | all hfc cards can do NT |
| 21:42.27 | StyleWarz | yeh i know that =) |
| 21:42.28 | zoa | but i dont know the BA :) |
| 21:42.46 | StyleWarz | it is the little box you put inbetween your multipointlink which makes your s0 bus |
| 21:42.53 | zoa | aaaah |
| 21:43.01 | zoa | i think you will need one |
| 21:43.05 | StyleWarz | hm |
| 21:43.14 | StyleWarz | well then i can also stick with the junghanns octobri |
| 21:43.26 | zoa | yes |
| 21:43.30 | StyleWarz | zoa: so there is no way to connect s0 phones directly to my asterisk? :( |
| 21:43.32 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 21:43.34 | zoa | nopez |
| 21:44.12 | StyleWarz | Hrmpf, this is ugly because my solution looks like "self-made" ;) |
| 21:44.24 | StyleWarz | but well |
| 21:44.29 | StyleWarz | with those beronet i won't need them |
| 21:44.38 | StyleWarz | i can enable the 100 ohms for each port |
| 21:44.46 | StyleWarz | so i could leave it out |
| 21:44.50 | zoa | beronet or junghanns should be pretty much the same |
| 21:47.46 | *** join/#asterisk s0lid (n=jlq@210.213.199.232) |
| 21:50.18 | mut | what're polycom phones default passwords? |
| 21:50.27 | mut | & username |
| 21:55.06 | ManxPowe1 | mut: Depends on which one you are asking about. |
| 21:55.11 | ManxPowe1 | 456 is one of the default passwords |
| 21:55.47 | mut | default user? |
| 21:55.55 | linlin | i need someone with experence with MGCP |
| 21:56.29 | ManxPowe1 | mut: no default user for the phone interface via the keypad. What SPECIFIC default user/pass do you need? |
| 21:56.42 | mut | web admin |
| 21:56.49 | ManxPowe1 | Polycom/456 |
| 21:56.58 | ManxPowe1 | This is in the Admin Guide |
| 21:57.05 | ManxPowe1 | As is everything else you want to know. |
| 21:57.07 | mut | that works |
| 21:57.11 | mut | yay |
| 21:57.17 | mut | i dun have the admin guide |
| 21:57.27 | mut | i just have a user trying to login with an invalid username and password |
| 21:57.30 | mut | every 60 seconds |
| 21:57.31 | ManxPowe1 | Well get it! |
| 21:57.37 | mut | so i'm killing their config |
| 22:00.52 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 22:02.57 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 22:03.04 | zoa | hey ho manxy |
| 22:03.54 | mercestes | try googling polycom admin guide |
| 22:04.08 | *** join/#asterisk remmo (n=chatzill@202.172.106.161) |
| 22:04.21 | *** part/#asterisk remmo (n=chatzill@202.172.106.161) |
| 22:04.33 | *** join/#asterisk M_at (n=matt@dsl092-214-175.atl1.dsl.speakeasy.net) |
| 22:06.48 | *** join/#asterisk jorge_ (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 22:09.38 | M_at | Quickie: If I have a PRI card and nothing plugged into it should that be anough to cause * to stop loading at chan_zap ? |
| 22:09.55 | tzanger | nope |
| 22:10.39 | M_at | OK - I have a bigger zaptel config problem then :) |
| 22:11.11 | M_at | if I don't have the "channel => 25-47" line in place it doesn't complain but of course I have no PRI channels |
| 22:11.59 | *** join/#asterisk jorge_ (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 22:13.43 | tzanger | M_at: did you run ztcfg? does /etc/zaptel.conf match? |
| 22:13.52 | M_at | Yup |
| 22:14.57 | M_at | Can someone suggest a pastebin.ca alternative - it's down currently |
| 22:15.07 | linlin | help me out here |
| 22:15.08 | linlin | Dec 7 16:14:52 NOTICE[621]: chan_mgcp.c:1654 find_subchannel_and_lock: Gateway '192.168.1.103' (and thus its endpoint 'aaln/1') does not exist |
| 22:15.14 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:15.15 | linlin | what would you do if you got that error |
| 22:15.28 | linlin | 192.168.1.103 = my asterisk box by the way |
| 22:15.37 | mercestes | <jbot> rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:15.37 | mercestes | <danp> might i also recommend pastie: http://pastie.caboo.se |
| 22:16.18 | mercestes | linlin: I would google that error and see what it meant and how to fix it. |
| 22:16.28 | M_at | ztcfg -vvvvv output : http://pastie.caboo.se/26454 |
| 22:17.19 | M_at | zaptel.conf : http://pastie.caboo.se/26455 |
| 22:18.10 | M_at | zapata-channels.conf : http://pastie.caboo.se/26457 |
| 22:18.58 | *** join/#asterisk alexmontoanelli (n=alexm@alexmm.unetvale.com.br) |
| 22:19.26 | *** join/#asterisk smp|null (n=sageone@mh2.netfirms.com) |
| 22:20.00 | M_at | ./var/log/asterisk/full : http://pastie.caboo.se/26458 |
| 22:20.19 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:24.34 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.mn.comcast.net) |
| 22:25.29 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:27.35 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-b82c11586a6368d6) |
| 22:28.14 | *** join/#asterisk crochat (i=crochat@84-74-145-139.dclient.hispeed.ch) |
| 22:33.32 | *** join/#asterisk Weezey (n=ohno@206.210.109.230) |
| 22:34.22 | Weezey | What's the best way to make your sets stay online in the event of a hardware failure? |
| 22:35.04 | Supaplex | reboot |
| 22:36.27 | Weezey | Do I run two asterisk boxes with Linux-HA? Or does something like SER allow for a cluster of front-ends? |
| 22:37.50 | Supaplex | oh that kind of hardware failure. :p |
| 22:38.03 | Supaplex | I thought you were speaking of the handsets |
| 22:38.23 | Weezey | heh, nah, if a handset is down, it's down. I'm talking network-wide. |
| 22:46.16 | rpm | how do i execute more commands in my dialplan after the a meetme conference is over? |
| 22:46.42 | CunningPike | Weezey: Take a look at Asterisk-at-large in The Wiki |
| 22:46.47 | CunningPike | ~thewiki |
| 22:46.49 | jbot | somebody said thewiki was at http://www.voip-info.org/wiki-Asterisk |
| 22:49.31 | *** join/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 22:51.26 | Katty | bye bye (= |
| 22:51.35 | mercestes | Aww. |
| 22:56.00 | *** join/#asterisk FWP (n=FWP@unaffiliated/fwp) |
| 22:56.08 | *** part/#asterisk mkrufky (n=mk@unaffiliated/mkrufky) |
| 22:57.08 | hoobastoob1 | i have one server (180) trying to dial another server (189). I have in the general section of the iax.conf of 180 a register => line and at the bottom I have a [user] section. on 189 i have the [user] section with all of the matching user name and secret stuff... am i thinking of this correctly? |
| 23:01.56 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-110-113.dsl.tulsok.swbell.net) |
| 23:03.28 | hoobastoob1 | i dial from one server to the other which shows registerd, but i get the error: |
| 23:03.28 | hoobastoob1 | Call rejected by 10.0.1.189: No authority found |
| 23:04.00 | hoobastoob1 | on the server i am dialing to i get: Rejected connect attempt from 10.0.1.180, who was trying to reach 's@' |
| 23:05.14 | hoobastoob1 | Call rejected by 10.0.1.189: No authority found |
| 23:05.36 | *** part/#asterisk dasenjo (n=dasenjo@190.24.178.18) |
| 23:12.14 | *** join/#asterisk runderwood (n=runderwo@200.91.181.154) |
| 23:13.04 | runderwood | Hello every body |
| 23:13.08 | *** join/#asterisk jorge_ (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 23:13.24 | runderwood | I'm kind of new in asterisk |
| 23:13.50 | runderwood | and I was wondering if some one can give a direction with a problem that I have |
| 23:14.16 | linlin | -- Resetting interface aaln/4@192.168.1.115 |
| 23:14.16 | linlin | -- No command found on [192.168.1.115] for transaction 19. Ignoring... |
| 23:14.22 | linlin | what would something like that mean |
| 23:14.42 | runderwood | I have my box setup with Asterisk 1.2, with AMP and g729 as default codec |
| 23:15.01 | runderwood | I can ear fine the people that calls me but they don't |
| 23:15.50 | De_Mon | runderwood tell us about your setup, sip/iax nat pstn etc |
| 23:16.03 | runderwood | is fully sip |
| 23:16.08 | runderwood | I don't have any pstn |
| 23:16.23 | runderwood | I have a DSL modem and small network at my place |
| 23:16.49 | runderwood | I have a free service from gizmo |
| 23:17.16 | runderwood | so the people is trying to call me from their pstn phone |
| 23:17.36 | runderwood | the box is running under RH 4 |
| 23:17.41 | runderwood | RH E$ |
| 23:17.48 | runderwood | sorry RH E4 |
| 23:20.27 | runderwood | at the begining I thought it was the bandwith but why I can listen so fine |
| 23:20.32 | fx0 | codec mismatch ? |
| 23:21.28 | runderwood | maybe, I ran asterisk -rvvv and then the comand show g279 and it shows 1/1 encoders/decoders of 1 licensed channels are currently in use |
| 23:22.25 | runderwood | so I'm kind of loss |
| 23:23.26 | runderwood | maybe some one had have the same problem in the pass |
| 23:24.06 | runderwood | and can give a tip or something that guide me a little, because I have try everything that I know |
| 23:24.23 | linlin | sounds like a NAT issue |
| 23:24.35 | fx0 | that too |
| 23:24.44 | runderwood | you think |
| 23:25.18 | runderwood | I don't have a public IP address |
| 23:26.09 | hoobastoob1 | so I think that my server is registered to the one i am trying to dial, but i get an error: on the server i am dialing from i get: Call rejected by 10.0.1.189: No authority found and on the one i am trying to dial to i get: Rejected connect attempt from 10.0.1.180, who was trying to reach 's@' |
| 23:26.48 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 23:27.33 | runderwood | does the asterisk log show something about it? |
| 23:27.43 | *** join/#asterisk pt105 (n=npc@c-69-251-187-28.hsd1.md.comcast.net) |
| 23:27.43 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 23:27.46 | *** join/#asterisk Stelios (n=toggy@host-81-191-169-198.bluecom.no) |
| 23:28.00 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.18) |
| 23:28.37 | hoobastoob1 | i do an iax2 show registry from the server i am dialing from and it shows registerd |
| 23:29.05 | hoobastoob1 | but i get no authority found when i try to dial that other server |
| 23:29.39 | robin__sz | mmm .. nice new phone |
| 23:29.47 | *** part/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 23:30.05 | *** join/#asterisk npc105 (n=npc@c-69-251-187-28.hsd1.md.comcast.net) |
| 23:30.20 | robin__sz | just picked up some Elmeg 290's ... basically Snom 190s made by some other non-snom firm |
| 23:30.40 | robin__sz | using the original Snom tooling |
| 23:30.56 | robin__sz | very nice .. and cheap too. £57.50 inc vat and postage :) |
| 23:31.50 | JT | 60 pounds doesn't sound the cheap for a clone :P |
| 23:31.58 | JT | s/the/that/ |
| 23:32.07 | robin__sz | how do you mean "clone"? |
| 23:32.10 | robin__sz | as in copy? |
| 23:32.21 | JT | how is it not a clone? |
| 23:32.24 | npc105 | Anyone ever had a problem with MONITOR_EXEC not being recognized by Monitor()? |
| 23:32.27 | JT | from what you're described |
| 23:33.05 | robin__sz | because Snom no longer make it ... they sold the whole design and tooling ... it is even loaded with the original Snom firmware |
| 23:33.28 | JT | so will updates be coming out, support too? |
| 23:33.34 | npc105 | I am trying to get sox to join the two legs and compress the single file into a ogg, however it never calls the MONITOR_EXEC commmand |
| 23:34.09 | robin__sz | a clone woudl be if someone copied it .. its not a copy, its the original thing, with a different badge |
| 23:34.20 | JT | sure |
| 23:34.26 | JT | so how are updates and support? |
| 23:35.11 | robin__sz | its a mature product, so I dont really expect any updates, it had stabilised and the last couple of updates were stunningly minor |
| 23:35.47 | JT | well sometimes people find issues or security flaws in something years after it's released |
| 23:36.04 | robin__sz | as for support? just how much supprt do you need ona basic office phone? I already have a dozen or so Snom originals and never really felt the need to phone Snom |
| 23:36.10 | *** join/#asterisk Soul (n=Soul@89-180-137-32.net.novis.pt) |
| 23:36.16 | robin__sz | true .. that could happen |
| 23:38.44 | robin__sz | they run Linux anyway, so perhaps openhardphone.org will eventually release a GPL phoen application to run on it |
| 23:39.14 | robin__sz | now, that would be WAY kewl :) |
| 23:39.34 | JT | i'm not sure if there's all that much point in buying new at retail prices if you get no updates, support or warranty |
| 23:39.37 | JT | may aswell go to ebay |
| 23:40.06 | robin__sz | sorry, please explain why I get no updates, support or warranty? |
| 23:40.17 | JT | you seemed ti imply such |
| 23:40.25 | JT | or at least tell me they were unnecessary |
| 23:41.08 | robin__sz | oh they are unnecessarry IMHO, but Elmeg are a proper real company .. I expect they are there shoudl I wnat them .. certainly the warranty is there |
| 23:41.19 | JT | right |
| 23:41.52 | robin__sz | I was just pleased to find decent hardware at a very good price |
| 23:42.19 | robin__sz | makes me kinda happy afetr my grandstream disaster |
| 23:43.21 | *** join/#asterisk The_Ball (n=alex@203.27.180.111) |
| 23:43.47 | The_Ball | im reading about a voip provider and they state as a featuer: "no flag falls" what does that mean? |
| 23:45.58 | JT | it means there's no flagfalls |
| 23:46.03 | JT | <PROTECTED> |
| 23:46.05 | JT | :) |
| 23:46.18 | The_Ball | oh, marketing term i bet |
| 23:46.27 | JT | shrug, it's quite common here |
| 23:46.52 | Un1x | hey JT know of a player i can use to play .gsm files on windows? |
| 23:47.03 | JT | and you are in australia i see, The_Ball |
| 23:47.08 | JT | it's a common term here :) |
| 23:47.34 | JT | if you've ever spent any time shopping around any telcos whatsoever |
| 23:47.56 | JT | most of the traditional PSTN providers have flagfalls on timed calls |
| 23:48.04 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 23:48.04 | The_Ball | im from norway though ;) |
| 23:48.08 | JT | the term really only applies to a timed call |
| 23:48.10 | JT | i see |
| 23:48.51 | The_Ball | so which voip provider do you use= |
| 23:52.28 | *** join/#asterisk jorge_ (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 23:52.58 | *** join/#asterisk onta1 (n=root@clnet-p03-090.ikbnet.co.at) |
| 23:55.31 | JT | The_Ball: using engin a bit at the moment |
| 23:55.36 | JT | trialling a few others too |
| 23:57.53 | brian | engin? |
| 23:58.14 | The_Ball | i do a heap of calls to my family in norway, would i be better getting a voip provider there, and a different voip provider here in australia for the other calls? |
| 23:58.22 | hoobastoob1 | so if server 10.0.1.180 is registered to server 10.0.1.189 with iax2, why when dialing from 10.0.1.180 to 10.0.1.189 give me the error: Call rejected by 10.0.1.189: No authority found |
| 23:58.31 | JT | brian: google.com |
| 23:59.06 | JT | The_Ball: will you be calling norwegian landlines? |
| 23:59.11 | The_Ball | yes |
| 23:59.23 | JT | get a voip account in norway then |
| 23:59.55 | JT | well, check the prices first and compare with voip rates from an australian provider to norway |
| 23:59.55 | The_Ball | then i can get an incoming norwegian number for them to call me on as well i guess |
