| 00:00.05 | brookshire | i would upgrade |
| 00:00.07 | brookshire | i don't think it will help |
| 00:00.12 | brookshire | maybe even try trunk |
| 00:00.17 | brookshire | buit definate reboot first |
| 00:00.23 | opus_ | i think trunk is broken right now |
| 00:00.29 | brookshire | boo! |
| 00:00.35 | brookshire | silly pisa coders |
| 00:00.55 | kink0 | opus_ ok, I can try a 1.2.5 I have installed also, at least compiled ready to install |
| 00:01.17 | FuriousGeorge | let me ask this way, lets say i have a dialtone come in from the NID, then i run it to another NID which is still hooked up to the phone company, but with no line. is that NID gonna kill my dialtone beyond that? |
| 00:01.30 | kink0 | but 2 days ago, there was 1.2.7 and was cdr reported as BUSY |
| 00:01.38 | brookshire | furious: so you have two lines of fxs? |
| 00:01.46 | brookshire | and you want to hook them together? |
| 00:01.53 | kink0 | and I have stoped and restart Asterisk, so I don't suspect about 1.2.7.1 |
| 00:02.27 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:02.30 | FuriousGeorge | brookshire: the wiring in this building is strange and mysterious. apparently in fixing it the phone company guy hooked the line up on a NID that belonds to the other side of the warehouse (another unit) |
| 00:02.38 | brookshire | kink0: if zaptel is locking somehow, it won't clear up without a reboot |
| 00:02.48 | brookshire | sometimes reloads fix it |
| 00:02.51 | brookshire | but not always |
| 00:02.52 | FuriousGeorge | so i cant send the dialtone back over, and i /THINK/ it muyst be because the old NID which is now dead is still on the line |
| 00:03.18 | FuriousGeorge | and thats "sucking up" my dialtone |
| 00:03.40 | FuriousGeorge | i cant see why the tone from my tonetester would make it across, but not the dialtone |
| 00:03.44 | kink0 | brookshire, I know, but Asterisk in CLI reports as must be busy, only in CDR is reported as failed instead busy |
| 00:03.54 | brookshire | if you hook two phonelines together, bad things happen ;) |
| 00:04.35 | brookshire | maybe busy is failed in cdr |
| 00:04.36 | brookshire | ?? |
| 00:05.08 | FuriousGeorge | brookshire: it appears that on our side of the second NID everything is fine, but that second NID must be sitting between me and the alarm unit messing up the other side of the bulding |
| 00:05.13 | dahunter3 | I'm having trouble with a remote SIP phone. I can place calls on it just fine, check voicemail etc. However, I cannot call the SIP phone, it immediately says busy and returns instead of ringing it? Is this a firewall issue? |
| 00:05.48 | FuriousGeorge | it now goes OUR NID --> OUR ALARM -> splice > their NID --> their alarm |
| 00:06.32 | FuriousGeorge | thats the only reason i can see that the test tone makes it from splice to alarm, but not the dialtone |
| 00:07.44 | FuriousGeorge | i can hear the test tone on the other side but i dont even get talk battery |
| 00:08.28 | brookshire | dahunter: could be, do you have your phone setup as type=friend? |
| 00:09.15 | brookshire | dahunter: and is it registering? |
| 00:09.45 | FuriousGeorge | can anyone think of a situation where the tone from a tone tester would make it from point a to point b, but not the dialtone? |
| 00:10.13 | brookshire | disconnected line |
| 00:10.27 | brookshire | bad wiring? |
| 00:10.42 | FuriousGeorge | im implying the dialtone exists on one side, so its not disconnected |
| 00:11.02 | FuriousGeorge | and if the wiring was shorted or not complete i would still get talkbat in the former case, and no tone in the second case |
| 00:11.16 | FuriousGeorge | no test tone in the second case |
| 00:11.32 | dahunter3 | brookshire: so if it is type=friend and it is set to register it wouldn't call? |
| 00:11.35 | danilom | with a Fxs can i connect two lines? |
| 00:12.13 | danilom | i mean, can i communicate two lines? |
| 00:12.20 | FuriousGeorge | danilom: yes, you get like 5 REN from a tdm400p, and most modern phones use less than 1 ren |
| 00:13.03 | danilom | furiousgeorge_: ok can i connect to a PBX? panasonic 616? |
| 00:13.29 | FuriousGeorge | sure, the other pbx will just think * is a phone company line |
| 00:13.46 | brookshire | dahunter: maybe, i don't know |
| 00:15.21 | dahunter3 | brookshire: It is set to friend, I'm checking on the registration required now |
| 00:16.32 | danilom | FuriousGeorge_: with a FXS linksys sipura spa 2100 works to connect two lines? |
| 00:17.54 | FuriousGeorge | danilom: i assume at least two but see how many REN it provides. you see old phones can use up to 1.5 REN but new ones are around .5, so you see one phone can equal three other ones |
| 00:17.58 | dahunter3 | brookshire: It is a sipura 841, and Register is set to yes, "Make call without Reg," is set to "no," and "Ans Call without Reg," is no. |
| 00:18.48 | brookshire | dahunter: so is it registering? |
| 00:19.09 | ManxPower | Remember SUPURA's G726 codec does not work by default with Asterisk. |
| 00:19.12 | brookshire | does 'sip show peers' say anything? |
| 00:19.13 | danilom | FuriousGeorge_: the server is good, excelent? |
| 00:19.41 | brookshire | manx: he says it is coming back busy |
| 00:19.43 | danilom | FuriousGeorge_: i mean, service, im sorry |
| 00:19.53 | brookshire | would that cause that? |
| 00:20.01 | ManxPower | brookshire, What would "not acceptable here" come back as. |
| 00:20.26 | FuriousGeorge | danilom: im not sure what you mean, but you will know if you have hooked up too many phones to one fxs cuz they wont ring or DTMF wont work |
| 00:20.39 | brookshire | no idea, what? |
| 00:20.40 | brookshire | :D |
| 00:21.25 | ManxPower | Easy to test. In sip.conf: disallow=all and allow=ulaw and no other allow lines |
| 00:21.32 | dahunter3 | 251/251 71.XX.XX.XX D N 50012 OK (76 ms) |
| 00:21.43 | dahunter3 | brookshire: I assume that means it is registered. |
| 00:21.45 | brookshire | dahunter: do what manxpower said ;) |
| 00:21.57 | danilom | FuriousGeorge_: ok im just want to connect two offices with one line each one, can i use a fxs for this? |
| 00:22.35 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 00:23.06 | brookshire | danilon: fxs drives a phone... fxo connects to the phone company |
| 00:23.14 | ManxPower | ~fxofxs |
| 00:23.16 | jbot | fxofxs is, like, An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 00:23.42 | dahunter3 | brookshire: Which was? I didn't see anything with a "dahunter," in the message. |
| 00:23.51 | FuriousGeorge | he wants to hook multiple devices to one fxs' loop |
| 00:23.53 | brookshire | in your sip.conf |
| 00:23.53 | ManxPower | OK, boys and girls, it's time for the pop quiz! What kind of port does a PHONE have? |
| 00:24.03 | brookshire | add 'disallow=all' |
| 00:24.06 | brookshire | allow=ulaw |
| 00:24.07 | ManxPower | dahunter3, Easy to test. In sip.conf: disallow=all and allow=ulaw and no other allow lines |
| 00:24.19 | FuriousGeorge | danilom: i dont know how much REN your fxs provides, and i dont know how much REN your devices draw, so I cant answer that |
| 00:24.30 | FuriousGeorge | my advice is to spend the money and get another FXS |
| 00:24.34 | ManxPower | *smacks dahunter3 on the knuckles with a ruler* Pay attention! |
| 00:24.49 | *** join/#asterisk esculapio___ (n=ESCulapi@adsl-200-136.tricom.net) |
| 00:25.04 | brookshire | ~ren |
| 00:25.09 | brookshire | boo! |
| 00:25.28 | danilom | FuriousGeorge_: ok its a older phone 1.5 REN |
| 00:25.46 | ManxPower | The problem with connecting Asterisk with a PBX using analog is that most PBX FXS lines do NOT drop battery on disconenect and so Asterisk has a hard time knowing when the PBX line hangs up. |
| 00:25.50 | danilom | FuriousGeorge_: can i connect them? |
| 00:25.57 | FuriousGeorge | to what? |
| 00:26.18 | ManxPower | I do not know if ASTERISK's FXS ports drop battery (I doubt it) on hangup and so the PBX can have trouble detecting a hangup |
| 00:26.36 | danilom | FuriousGeorge_: each other, two lines |
| 00:26.38 | brookshire | 1 ren = 1 old school 2500 set |
| 00:27.19 | ManxPower | All my phones have a -1.0 REN, so they ring 1 few seconds before the far end start dialing. |
| 00:27.29 | FuriousGeorge | danilom: if you plug one phone into the other, nothing will happen |
| 00:27.39 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 00:28.19 | brookshire | and on the other hand.. you plug two phone lines from the phone company together, bad things happen |
| 00:28.43 | dahunter3 | manxpower: <G> Checking that now. |
| 00:29.18 | danilom | FuriousGeorge_: ok, but i want to use a FXS linksys sipura to connect two lines, 1: is a PBX and the another one too! |
| 00:29.40 | ManxPower | brookshire, I've been tempted to plug a mains (power) feed into a T-1 a time or two. Figured that a blown up card would get the telco to replace it, rather than keep using the flaky card they had. Never actually did it, however. |
| 00:29.53 | ManWithYellowGun | fuck a bat. |
| 00:29.57 | FuriousGeorge | danilom: i dont know how else to tell you i have no idea how much REN your spirua fxs provides |
| 00:30.06 | FuriousGeorge | ahhhhhhh |
| 00:30.13 | ManxPower | danalien, Read THE BOOK |
| 00:30.15 | ManxPower | ~thebook |
| 00:30.16 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 00:30.34 | brookshire | haha.. "thebook" |
| 00:30.51 | danilom | FuriousGeorge_: I TOLD YOU 1.5 REN!!!!!! |
| 00:30.51 | ManxPower | ~ECFO |
| 00:30.52 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 00:30.53 | brookshire | or buy it! |
| 00:30.55 | FuriousGeorge | r_evolution: i think "ManWithYellowGat" would have been better |
| 00:31.00 | brookshire | it's better when you kill trees |
| 00:31.01 | brookshire | :D |
| 00:31.12 | r_evolution | duly noted. |
| 00:31.25 | r_evolution | you know danilom... yelling at people is NEVER a good way to get help |
| 00:31.36 | FuriousGeorge | danilom: you said it was the phone that USED 1.5 REN not the fxs PROVIDED 1.5 REN |
| 00:31.39 | ManxPower | r_evolution, REALLY?!?!?! |
| 00:31.43 | FuriousGeorge | (20:30:36) danilom: FuriousGeorge_: ok its a older phone 1.5 REN |
| 00:31.46 | r_evolution | haha manx. |
| 00:32.04 | FuriousGeorge | so to answer your question |
| 00:32.05 | FuriousGeorge | NO |
| 00:32.12 | brookshire | hmm.. an iaxy has a REN of 5 |
| 00:32.24 | r_evolution | ~FuriousGeorge |
| 00:32.25 | jbot | it has been said that furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
| 00:32.25 | r_evolution | :( |
| 00:32.26 | dahunter3 | Okay, I put in disallow all, allow ulaw, etc. reloaded asterisk, dialed the extension in questions and got: -- SIP/251-8375 is circuit-busy |
| 00:32.30 | dahunter3 | <PROTECTED> |
| 00:32.34 | r_evolution | sweet. |
| 00:32.47 | danilom | FuriousGeorge_: ok its another thing |
| 00:32.53 | dahunter3 | Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from XX.XX.XX.XX |
| 00:33.09 | FuriousGeorge | you see 2 phones X 1.5 ren > 1.5 ren provided by sipura according to you |
| 00:33.16 | ManxPower | danalien, All PBXs have multiple line/port types. What type of line/port do you want to use on your PBX? |
| 00:33.57 | *** part/#asterisk danilom (n=danilom@201.209.43.241) |
| 00:33.59 | FuriousGeorge | danilom: why would you want both offices on the same loop? |
| 00:34.08 | ManxPower | FXO port, Analog FXS port, digital phone port, T-1/E-1 port, etc. |
| 00:34.30 | *** join/#asterisk op3r (n=op3r@202.124.153.4) |
| 00:34.31 | FuriousGeorge | who will give me back the time i just wasted on him |
| 00:34.35 | ManxPower | FuriousGeorge, I think he wants PBX -> SIPURA -> INTERNET -> SIPURA -> PBX |
| 00:34.55 | brookshire | furious: i have trade for more of your time! |
| 00:35.04 | ManxPower | FuriousGeorge, Now you know why I've not been here most of the day |
| 00:35.24 | brookshire | FuriousGeorge: http://video.google.com/videoplay?docid=-2139555376132383479 |
| 00:35.29 | r_evolution | feel better? |
| 00:35.29 | op3r | is it possible to set up in sip.conf that the extensions can only use g711 and dont try to take g729? |
| 00:35.37 | r_evolution | yes op3r |
| 00:35.38 | ManxPower | FuriousGeorge, any time I want to say "who will give me back the time i just wasted on him", I put the person on /ignore |
| 00:35.51 | ManxPower | op3r, if you read The Book, you would know. |
| 00:35.52 | FuriousGeorge | lol |
| 00:36.02 | ManxPower | Hell, if you read the sip.conf.sample you would know. |
| 00:36.04 | brookshire | op3r: disallow=all and allow=ulaw |
| 00:36.18 | ManxPower | op3r, Are you using AMP/FreePBX/Asterisk@Home? |
| 00:37.43 | r_evolution | no kidding manx |
| 00:37.50 | r_evolution | whats up with lately? |
| 00:38.28 | ManxPower | r_evolution, Finished running the underground conduit. Pulled the first run of cables the other day. Will pull more runs when the weather dries out. |
| 00:39.00 | r_evolution | actually i was referring to the obvious questions that come through the channel llately |
| 00:39.05 | r_evolution | but that's good that you finished :-D |
| 00:39.22 | ManxPower | r_evolution, Ah. I dunno. Too many newbies. |
| 00:39.41 | r_evolution | i have no problem with new people... because im still pretty new... |
| 00:39.43 | r_evolution | but dammit... READ. |
| 00:39.45 | ManxPower | rather than screaming at them I usually just /part |
| 00:40.00 | op3r | ManxPower: no amp this time man |
| 00:40.15 | ManxPower | op3r, go read sip.conf.sample, then come back |
| 00:40.22 | r_evolution | ~thebook |
| 00:40.23 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 00:40.23 | r_evolution | :-\ |
| 00:40.28 | FuriousGeorge | brookshire: this is pretty funny: |
| 00:40.34 | FuriousGeorge | ive seen one to punchout too |
| 00:40.36 | op3r | brookshire: is it possible todo that on some several extensions only? |
| 00:40.40 | FuriousGeorge | done in china |
| 00:40.51 | r_evolution | again. yes. op3r |
| 00:40.52 | r_evolution | go |
| 00:40.52 | r_evolution | read |
| 00:40.54 | r_evolution | ~the |
| 00:40.55 | jbot | i guess the is full of rotten mash potatoes |
| 00:40.56 | r_evolution | ~thebook |
| 00:40.58 | jbot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 00:40.58 | r_evolution | ewps. |
| 00:41.11 | op3r | ok |
| 00:41.37 | r_evolution | :) it'll seriously help a lot... great starting point it was for me |
| 00:41.43 | ManxPower | op3r, you can do it PER sip.conf entry |
| 00:42.18 | ManxPower | Using the "r" option to dial is like walking around with a sign on your back that says "dork" |
| 00:42.32 | r_evolution | haha |
| 00:42.34 | ManxPower | Reading The Book helps you avoid that |
| 00:42.44 | op3r | ok |
| 00:42.44 | r_evolution | that actually made me chuckle manx |
| 00:44.17 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 00:44.55 | FuriousGeorge | brookshire: along the same lines: http://www.starterupsteve.com/swf/mdm3.html |
| 00:44.58 | iceyp | hey guys, anyone know how to implement 729 codec? want to test it at home... its for bsd, i downloaded http://kvin.lv/pub/Linux/Asterisk/freebsd/codec_g729-gcc-pentium4.so to /usr/local/lib/asterisk/modules/ |
| 00:45.11 | iceyp | but i still get the error : |
| 00:45.13 | iceyp | May 11 12:31:11 WARNING[37526]: translate.c:132 ast_translator_build_path: No translator path from g723 to alaw |
| 00:45.13 | iceyp | May 11 12:31:11 WARNING[37526]: chan_sip.c:2536 sip_write: Asked to transmit frame type 2, while native formats is 256 (read/write = 256/2) |
| 00:45.21 | r_evolution | O_o |
| 00:45.40 | r_evolution | register it? :) |
| 00:45.43 | brookshire | op3r: you mean on all extensions? |
| 00:45.52 | FuriousGeorge | later all |
| 00:45.59 | brookshire | you can make it default by placing those lines above all the other contexts |
| 00:46.01 | r_evolution | don't get caught George |
| 00:46.13 | FuriousGeorge | r_evolution: i aint scared |
| 00:46.17 | iceyp | r_evolution what u mean register it? |
| 00:46.38 | r_evolution | well... i mean pay the simple $10 to digium for the free software you're using it with |
| 00:46.38 | brookshire | iceyp: g729 != g723 |
| 00:46.54 | iceyp | i using it to test |
| 00:47.09 | iceyp | brookshire when did i mention 723? |
| 00:47.17 | r_evolution | <PROTECTED> |
| 00:47.23 | SplasPood | [20:45] iceypMay 11 12:31:11 WARNING[37526]: translate.c:132 ast_translator_build_path: No translator path from g723 to alaw |
| 00:47.27 | iceyp | ohh |
| 00:47.28 | iceyp | mmm |
| 00:47.32 | r_evolution | good catch brook... i wasnt even paying attention to that ;x |
| 00:47.49 | iceyp | cisco 7912's have 729 codecs dont they |
| 00:47.59 | iceyp | didnt even notice the translator either |
| 00:48.21 | iceyp | disallow=all |
| 00:48.24 | iceyp | allow=g729 |
| 00:48.36 | SplasPood | maybe the phone is set to use g723 |
| 00:48.43 | iceyp | maybe because i called my voicemail |
| 00:48.47 | iceyp | let me try a real number |
| 00:49.06 | rpm | is anyone in calgary and good with asterisk/development. |
| 00:49.14 | r_evolution | possible Splas... the UT Starcom ATAs we were testing here (for distrib to customers) used 723 as the preferred no matter WHAT you set it as |
| 00:49.24 | r_evolution | until i disallowed everything but ulaw/729 |
| 00:50.41 | brookshire | rpm: if you can find anyone one the channel... you can try here: http://forums.digium.com/viewforum.php?f=6&sid=5a1670365bc884590b160bf1d08d67fc |
| 00:50.52 | iceyp | <PROTECTED> |
| 00:51.09 | iceyp | mm but it went through nefone, maybe because no one else accepted 729 |
| 00:51.14 | ManxPower | My policy is disallow=all and allow=onlythecodeciwantforthatdevice |
| 00:51.15 | sevard | http://www.smh.com.au/news/breaking/old-computers-harm-office-morale/2006/05/10/1146940596174.html |
| 00:51.19 | sevard | I just sent that to my boss |
| 00:51.19 | iceyp | so it was doing passthrough |
| 00:51.22 | sevard | Subject: Subtle Hint |
| 00:51.34 | r_evolution | haha sevard. |
| 00:51.44 | r_evolution | Subject: Not-So-Subtle Hint |
| 00:51.48 | r_evolution | unless he's hella dense |
| 00:51.51 | sevard | i hope i don't get fired |
| 00:52.06 | r_evolution | kinda what i'm doing here Manx... but we need both 729/ulaw for outgoing |
| 00:52.15 | r_evolution | so customers can fax... for what thats worth |
| 00:52.42 | *** join/#asterisk eastwolf (i=eastwolf@221.217.217.243) |
| 00:52.43 | sevard | r_evolution: you fax over sip? |
| 00:52.49 | sevard | so many issues with that |
| 00:53.06 | r_evolution | no choice sevard |
| 00:53.07 | *** join/#asterisk project_2501 (n=project-@S010600502c06f081.br.shawcable.net) |
| 00:53.09 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 00:53.22 | sevard | how do your customers like resending a fax 4 times :| |
| 00:53.25 | r_evolution | the customers of the ITSP had an absolute FIT |
| 00:53.27 | gandhijee | anyone here have the latest firmware for the SP601's? |
| 00:53.36 | r_evolution | actually it usually goes through surprisingly enough |
| 00:53.46 | r_evolution | p.s. level3 doesn't do T.38 :( |
| 00:53.57 | ManxPower | gandhijee, the stuff on freedomfones is not recent enough? |
| 00:54.05 | r_evolution | the fax itself usually go through... it just looks ugly. |
| 00:54.21 | gandhijee | lemme check again |
| 00:54.25 | r_evolution | but hey it gets upper-management off my ass... because God knows they wont actually LISTEN to my advice |
| 00:54.27 | ManxPower | Ah. We don't do FaxOverVoiceOverIPOverInternet |
| 00:54.35 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 00:54.35 | gandhijee | last time i checked, it wasn't up there |
| 00:54.49 | eastwolf | hello everyone |
| 00:54.59 | iceyp | how does one import a .so codec into asterisk? |
| 00:55.08 | r_evolution | hey manx... they're also wanting cred card processing over ulaw O_o |
| 00:55.17 | MikeJ[Laptop] | iceyp, import? |
| 00:55.26 | iceyp | or install |
| 00:55.35 | r_evolution | i think he's wanting * to use some random codec he DL'd from somewhere |
| 00:55.37 | iceyp | i just dropped it into the modules folder |
| 00:55.38 | MikeJ[Laptop] | as in an out of tree or in tree? |
| 00:55.38 | r_evolution | well |
| 00:55.42 | r_evolution | not so random |
| 00:55.45 | r_evolution | but 729... |
| 00:55.50 | ManxPower | r_evolution, I sometimes feel sorry for service providers. I have it lucky, I'm in a corporate enviroment. PRI and Analog at each office. |
| 00:55.51 | r_evolution | make sure the permissions are set icey :) |
| 00:56.00 | MikeJ[Laptop] | is it actually an asterisk module? |
| 00:56.02 | iceyp | MikeJ[Laptop] i have the e729.so file |
| 00:56.07 | ManxPower | iceyp, the codec has to be ported to work with Asterisk. |
| 00:56.11 | iceyp | http://kvin.lv/pub/Linux/Asterisk/freebsd/ |
| 00:56.14 | MikeJ[Laptop] | that's nice, I have no idea what that is |
| 00:56.18 | iceyp | its for asterisk |
| 00:56.31 | r_evolution | heh... i'm replacing the AGED pbx here... |
| 00:56.34 | ManxPower | iceyp, the contact the vendor, since there is NO legal and free G729 codec |
| 00:56.37 | r_evolution | sometime this/next week |
| 00:56.38 | MikeJ[Laptop] | looks like it has somthing to do w/ 729, but probably nothing to do with the asterisk codec interface |
| 00:57.09 | ManxPower | r_evolution, I learned LONG ago to ALWAYS have at LEAST 1 analog line at each location. |
| 00:57.10 | MikeJ[Laptop] | ManxPower, that's not entirely true... but probably correct in this case... |
| 00:57.23 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 00:57.33 | r_evolution | Again... I keep pushing for such things... |
| 00:57.39 | MikeJ[Laptop] | such what? |
| 00:57.39 | r_evolution | but who-ever says my advice is taken? |
| 00:57.57 | r_evolution | I think they may keep one T1 here... but even that remains to be seen |
| 00:58.01 | ManxPower | r_evolution, as a consultant, if they don't take my advice, I don't do the project. |
| 00:58.19 | MikeJ[Laptop] | as a consultant, I make recomendations in writing, |
| 00:58.24 | r_evolution | im not a consultant... I work here -_- |
| 00:58.32 | MikeJ[Laptop] | then if they tell me to do somthing else in writing.. I do it and get paid |
| 00:58.47 | MikeJ[Laptop] | I tend to charge more when they don't follow my advice. |
| 00:59.00 | ManxPower | Like when they bought a different model of motherboard than the one I spec'd for Asterisk. |
| 00:59.04 | MikeJ[Laptop] | and will only do per hour support on it |
| 00:59.11 | littleball | hello, how many current SIP calls can be supported by a dual-cpu(3.0GHz) and 2G memory server? the voice traffic go through the server also. any referecence number ? |
| 00:59.12 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:59.13 | eastwolf | ä½ å¥½ |
| 00:59.26 | MikeJ[Laptop] | littleball, depends |
| 00:59.46 | ManxPower | I have three customers currently. Two of them have been custoemrs for more than 10 years. |
| 01:00.10 | ManxPower | littleball, if you were reading the mailing lists over the past month you would know. |
| 01:00.13 | littleball | MikeJ{Laptop], just a referecen number . no need to be accurate |
| 01:00.19 | MikeJ[Laptop] | 1 |
| 01:00.24 | littleball | ManxPower, how many? |
| 01:00.26 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 01:00.26 | r_evolution | hgahahahaha |
| 01:00.36 | MikeJ[Laptop] | well he said no need to be accurate |
| 01:00.50 | ManxPower | littleball, for the system and requirements specified on the thread, I think the number was 250 |
| 01:00.51 | r_evolution | nice manx... i work with one company who likes to do the opposite of everything i say :) |
| 01:00.52 | littleball | ex. 100 - 1000? |
| 01:01.16 | littleball | ManxPower, 250 is using ZAP hardware |
| 01:01.17 | r_evolution | i told them they needed more than a T1 to support 300 customers |
| 01:01.18 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:01.21 | MikeJ[Laptop] | calls per seccond is probably a better number to be concerned with than concurrent calls |
| 01:01.24 | r_evolution | they needed a DS3 in fact |
| 01:01.28 | r_evolution | you know what they did? |
| 01:01.30 | r_evolution | ignored me. |
| 01:01.35 | gandhijee | how can i find out what version my polycom is currently running? |
| 01:01.38 | ManxPower | littleball, I have no idea. I think it was SIP |
| 01:01.39 | r_evolution | here we are... some 3 months later... |
| 01:01.46 | ManxPower | gandhijee, there's a menu option |
| 01:01.47 | r_evolution | and we're having quality issues ;x |
| 01:01.49 | r_evolution | i wonder why ;x |
| 01:02.05 | MikeJ[Laptop] | cuz |
| 01:02.08 | gandhijee | through the phone, or web interface? |
| 01:02.10 | littleball | I have E1 lines connected . but now want to scale the system. E1 line is not the best i think . so i am considering SIP. |
| 01:02.11 | r_evolution | im out. |
| 01:02.33 | MikeJ[Laptop] | littleball, E1's are great... better quality than sip any day |
| 01:02.47 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 01:03.01 | MikeJ[Laptop] | I'll take a tdm connection over voip any chance I can |
| 01:03.08 | MikeJ[Laptop] | all other things equal of course |
| 01:04.21 | littleball | to scale the system, i am planing to use big server(more cpus and more memory) with SIP. so this makes maintainence easy |
| 01:04.22 | gandhijee | i've heard about some people getting unchannalized T1's over here and connecting directly to the telco's POP |
| 01:04.58 | MikeJ[Laptop] | gandhijee, running data I assume you mean? |
| 01:05.11 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:05.12 | MikeJ[Laptop] | littleball, clusters are good... |
| 01:05.21 | ManxPower | be back in a few, need to do a circuit turnup |
| 01:05.22 | MikeJ[Laptop] | much easier to scale on multiple boxes |
| 01:05.31 | littleball | clusters. how ? any basic idea? |
| 01:05.33 | MikeJ[Laptop] | turnip :D |
| 01:05.36 | MikeJ[Laptop] | ummm |
| 01:05.43 | MikeJ[Laptop] | asterisk and ser comes to mind |
| 01:05.44 | gandhijee | well if you wanna call the voice codec data, then yes |
| 01:06.07 | gandhijee | i've been discussing it some with the telco in the state i live in |
| 01:06.20 | gandhijee | tryin to get a test setup to see exactly what is goin on |
| 01:06.26 | MikeJ[Laptop] | gandhijee, well... it just wasn't clear.. you mean internet T1's, or point to point data? |
| 01:07.08 | littleball | ser can be used to redirect sip call. i canot see its usage in purely PSTN system |
| 01:07.09 | gandhijee | i guess it would be a point to point data, stright to the telco's point of presence |
| 01:07.24 | MikeJ[Laptop] | littleball, you said you were going sip.... |
| 01:07.34 | MikeJ[Laptop] | gandhijee, to what? |
| 01:07.36 | littleball | i use sip to connect to my provider |
| 01:07.46 | littleball | not connect to my clients |
| 01:07.57 | MikeJ[Laptop] | what does telcos pop mean.... |
| 01:08.13 | MikeJ[Laptop] | cuz a wire going to a building means very little |
| 01:08.38 | gandhijee | the telcos pop, ie where all the telco equipment is |
| 01:08.43 | MikeJ[Laptop] | yees |
| 01:08.48 | MikeJ[Laptop] | I understand |
| 01:09.02 | MikeJ[Laptop] | you said an unchanelized t1... |
| 01:09.07 | MikeJ[Laptop] | so it's not a voice circuit |
| 01:09.16 | MikeJ[Laptop] | so it's hooked up to the internet? |
| 01:09.27 | MikeJ[Laptop] | I am saying that your statement makes no sense |
| 01:09.28 | gandhijee | you can say that, |
| 01:09.34 | gandhijee | yeah it does |
| 01:09.50 | MikeJ[Laptop] | so you are talking about an internet t1? |
| 01:09.54 | MikeJ[Laptop] | or a voice t1? |
| 01:09.56 | gandhijee | caviler in my state when they roll out a T hooks it in to thier network |
| 01:10.01 | gandhijee | its a DATA T1 |
| 01:10.03 | gandhijee | no voice |
| 01:10.06 | MikeJ[Laptop] | ok.. |
| 01:10.18 | MikeJ[Laptop] | and what is it talking to? |
| 01:10.25 | gandhijee | i dunno all the details yet |
| 01:10.43 | gandhijee | thats why i am tryin to get a test setup to figure out exactly how it works |
| 01:10.45 | MikeJ[Laptop] | like I said.. makes no sense |
| 01:10.56 | MikeJ[Laptop] | it works to do what? |
| 01:11.19 | gandhijee | wow |
| 01:11.37 | gandhijee | i said i am trying to get a test setup so i can test it and figure out exactly how it works. |
| 01:11.51 | gandhijee | so when i discuss it in the future i can explain it totally. |
| 01:11.57 | MikeJ[Laptop] | ok. |
| 01:12.00 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.61) |
| 01:12.00 | littleball | gandhijee, i think data T1 can be used to pass voice data in any format only if the other party accept it. |
| 01:12.11 | gandhijee | yes it can |
| 01:12.25 | littleball | that is to say, you can compress as what you want to optimize the bandwidth |
| 01:12.25 | MoutaPT | Hi does any one could tell me how to do Supervised Call transfer with Asterisk? |
| 01:12.31 | gandhijee | i *think* caviler would hook it in to the softswitches |
| 01:13.14 | littleball | my local teleco also provide such package. but the sales guy cannot explain clearly. |
| 01:13.16 | iceyp | how do i know if my g729 is installed properly? |
| 01:13.29 | MikeJ[Laptop] | gandhijee, you realize that you just for the first time mentioned that this had to do w/ voice... |
| 01:13.50 | MikeJ[Laptop] | iceyp, scroll back to my response 10 minutes ago |
| 01:13.51 | iceyp | all i did to install it was download it and rename it |
| 01:13.52 | iceyp | cp /usr/local/lib/asterisk/modules/codec_g729-gcc-pentium4.so /usr/local/lib/asterisk/modules/codec_g729a.so |
| 01:14.01 | iceyp | MikeJ[Laptop] ahh ok may have been away |
| 01:14.03 | iceyp | let me look |
| 01:14.06 | gandhijee | well this is the asterisk room.... |
| 01:14.20 | gandhijee | i really have no reason to discuss pushing data through a * box.... |
| 01:15.01 | MikeJ[Laptop] | you randomly brought up that you can get unchanelized t1's to the telco.... |
| 01:15.14 | iceyp | MikeJ[Laptop] cant find what u said |
| 01:15.17 | MikeJ[Laptop] | well.. whatever.. I don't care |
| 01:15.19 | gandhijee | in regards to what littleball was saying |
| 01:15.19 | iceyp | can u repeat please |
| 01:15.33 | gandhijee | about not being able to scale w/ E1.... |
| 01:15.57 | MikeJ[Laptop] | iceyp, you were talking about g729e.so, I asked if it was just a g729 library, or actutally an asterisk codec module |
| 01:16.00 | gandhijee | ie being fixed at the 32 lines in the E |
| 01:16.12 | *** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 01:16.37 | MikeJ[Laptop] | gandhijee, products like that (basically telco playing itsp) are still channel limited |
| 01:16.51 | iceyp | its the codec |
| 01:16.58 | iceyp | asterisk codec |
| 01:17.15 | iceyp | do i need to download lib's too ? |
| 01:17.22 | MikeJ[Laptop] | I have no idea... |
| 01:17.27 | gandhijee | whatever, i'll just keep my mouth shut till i get my test setup and figure out exactly what is goin on |
| 01:17.37 | MikeJ[Laptop] | I know for digiums codec, you need to install the license too |
| 01:17.57 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 01:17.58 | iceyp | MikeJ[Laptop] this is a free version for testing |
| 01:18.04 | ManxPower | damn storm |
| 01:18.06 | MikeJ[Laptop] | oh... |
| 01:18.08 | MikeJ[Laptop] | hmmm |
| 01:18.31 | MikeJ[Laptop] | so you compile the module against your source tree and install to the modules dir |
| 01:18.50 | littleball | gandhijee, any suggestions to scale E1 connections? too many E1 makes maintainence difficult. I prefer to use big box with more cpus and memory to do SIP. bindwidth is not a problem |
| 01:18.58 | littleball | doon't care bindwidth first |
| 01:19.27 | MikeJ[Laptop] | littleball, and I am saying, regardless of sip or e1, asterisk scales better as a group of boxes, than doing tons of calls on 1 box |
| 01:19.32 | littleball | this is the reason i ask how many sip calls can be supported |
| 01:19.53 | Qwell | iceyp: That the intel one? |
| 01:19.53 | iceyp | so i need to recompile asterisk for it to work? |
| 01:19.55 | MikeJ[Laptop] | and how many really depends on MANY factors |
| 01:20.01 | iceyp | qwell i beleive so |
| 01:20.09 | Qwell | iceyp: Do you live in the US? |
| 01:20.11 | iceyp | found it from this post http://lists.digium.com/pipermail/asterisk-bsd/2005-May/000846.html |
| 01:20.12 | MikeJ[Laptop] | iceyp, I said compile the module |
| 01:20.14 | iceyp | qwell nz |
| 01:20.14 | littleball | Qwell, any reference number for current sip calls on one box? |
| 01:20.29 | iceyp | mikeJ how do u compile a module? |
| 01:20.29 | Qwell | littleball: anywhere from 0 to 30000 |
| 01:20.36 | MikeJ[Laptop] | I don't know any specifics of that |
| 01:20.43 | iceyp | k |
| 01:20.47 | MikeJ[Laptop] | Qwell, 30,000, w media? |
| 01:20.51 | MikeJ[Laptop] | that's a lot :P |
| 01:21.00 | littleball | 30000? in my system, the voice traffic also pass the box. 30000? |
| 01:21.00 | Qwell | MikeJ[Laptop]: Think that's what he said |
| 01:21.30 | MikeJ[Laptop] | it's really no where near 30000 |
| 01:21.42 | Qwell | littleball: What hardware? |
| 01:21.48 | MikeJ[Laptop] | I have heard of people getting thousands, but I would never size a box like that... |
| 01:21.56 | MikeJ[Laptop] | the rest totally depends on what you are doingn |
| 01:22.19 | MikeJ[Laptop] | so what exactly are you doing? |
| 01:22.41 | Qwell | in reality, the max number of calls you can get per box is like 32k |
| 01:22.51 | MikeJ[Laptop] | Qwell, sush |
| 01:22.59 | littleball | Qwell, i need to scale my system. currently, i use two-cpu (3GHZ) boxes with E1 line connecting to teleco. now i want to use SIP instead of E1 to make my maintainence easy. (no need to login different boxes)... |
| 01:23.23 | MikeJ[Laptop] | littleball, trust me.. don't do it on one box |
| 01:24.08 | MikeJ[Laptop] | Qwell, you get the impression that he doesn't like my answer, so he is ignoring me? |
| 01:24.11 | littleball | MikeJ[Laptop], of course, i will not use on one box. ex., i have 10 small box,(40 E1). if i can put them on two bigger box with SIP, it make me happy |
| 01:24.31 | Qwell | 20 E1 per box? ha! |
| 01:24.35 | MikeJ[Laptop] | so what exactly are these boxes doing? |
| 01:24.38 | littleball | two or threee boxes. make me happ. |
| 01:24.43 | MikeJ[Laptop] | just bridging calls? |
| 01:24.51 | littleball | 10 box, each box 4 e1 |
| 01:25.24 | Qwell | heh, that's like a full DS3 |
| 01:25.31 | littleball | so, instead of connecting by e1, i want to connect to operator by SIP |
| 01:25.33 | MikeJ[Laptop] | just about... |
| 01:25.38 | Qwell | per box, that is |
| 01:26.02 | MikeJ[Laptop] | yeah.. that's probably not going to happen |
| 01:26.12 | Qwell | You'll be much happier with 4-8 boxes |
| 01:26.24 | MikeJ[Laptop] | then again.. what he is describing is probably exactly what proxies are for |
| 01:26.42 | Qwell | probably exactly? :p |
| 01:26.47 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 01:27.20 | littleball | one box also make application logic easy.... |
| 01:27.39 | littleball | i have no idea about how to cluster asterisk. any documents? |
| 01:27.39 | MikeJ[Laptop] | ahh.. applications.. |
| 01:27.57 | MikeJ[Laptop] | hire somone who knows proxies |
| 01:27.58 | littleball | yes. i have voice applications used by customers |
| 01:28.05 | MikeJ[Laptop] | your going to need one. |
| 01:28.22 | littleball | it is difficult to hire good person here. |
| 01:28.29 | MikeJ[Laptop] | yeah.. no way you are going to want to do 600 bridged calls on one box with other stuff going on |
| 01:28.36 | MikeJ[Laptop] | here... |
| 01:28.44 | MikeJ[Laptop] | it's a global marketplace ;) |
| 01:28.48 | littleball | no, my application server is runing on seperate server |
| 01:29.04 | MikeJ[Laptop] | honestly.. |
| 01:29.09 | MikeJ[Laptop] | you already have the 10 boxes... |
| 01:29.16 | MikeJ[Laptop] | I would stick with that approach |
| 01:29.31 | MikeJ[Laptop] | stick a proxy out front for registration |
| 01:29.59 | littleball | the clients/consumers are not SIP users. |
| 01:30.09 | MikeJ[Laptop] | what are they? |
| 01:30.16 | littleball | callback system |
| 01:30.21 | littleball | PSTN callback ssytem |
| 01:30.30 | MikeJ[Laptop] | so it's all sip to sip then... |
| 01:30.55 | MikeJ[Laptop] | you should be able to redirected audio to your provider.... |
| 01:31.12 | MikeJ[Laptop] | and just stay in the signaling path |
| 01:31.25 | MikeJ[Laptop] | and bill off the proxy. |
| 01:31.46 | MikeJ[Laptop] | save you a huge ammount of bandwidth too |
| 01:32.47 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 01:33.12 | tainted- | yea but u have to say in audio path to do stuff like dtmf and notification |
| 01:33.20 | tainted- | s/say/stay/ |
| 01:33.54 | littleball | MikeJ[Laptop], i will consider your suggeestions to keep Signalling on my system... |
| 01:36.24 | MikeJ[Laptop] | tainted-, only if you want to give the users the option to do stuff other than hang up once they are in the call.. |
| 01:37.14 | MikeJ[Laptop] | if you are in the signaling... you can still cut off the call when their credit runs out and all |
| 01:38.18 | tainted- | hmm.. and handle dtmf w/ rfc2833? |
| 01:38.28 | MikeJ[Laptop] | yah |
| 01:38.42 | MikeJ[Laptop] | so you redirect your rtp.. and it's all in the media stream.. |
| 01:38.46 | tainted- | it's just nice to send an audio cue Dial(L()) when times running low |
| 01:38.58 | tainted- | or doing Monitor() |
| 01:39.00 | MikeJ[Laptop] | screw it.. just hang up on em... :P |
| 01:39.17 | tainted- | lol |
| 01:40.01 | MikeJ[Laptop] | or... use info for dtmf, and then it's in the signaling... |
| 01:40.15 | MikeJ[Laptop] | and you can just send them tones when the call is going to end |
| 01:40.30 | tainted- | lol |
| 01:41.09 | tainted- | *stop pushing your *beep* telephone buttons* *beep* *i'm not* *beep* NO CARRIER |
| 01:41.18 | MikeJ[Laptop] | then you could conceivabally grab the call back w/ reinvite if you needed to be in the stream.. |
| 01:41.27 | tainted- | true |
| 01:41.41 | MikeJ[Laptop] | and you could do it based on user pressing digits... |
| 01:41.59 | MikeJ[Laptop] | now... I am not sure how much of this asterisk can do right now... |
| 01:42.02 | tainted- | that would have to be done at the proxy wouldn't it? |
| 01:42.07 | MikeJ[Laptop] | nope |
| 01:42.08 | tainted- | b/c i don't know if asterisk can do that |
| 01:42.15 | tainted- | the reinvite/fwd |
| 01:42.30 | MikeJ[Laptop] | a ua can do that conceivabally... |
| 01:42.47 | MikeJ[Laptop] | not sure... asterisk sip has had a ton of changes lately.. |
| 01:43.03 | tainted- | i can't even get at SIP responses from asterisk |
| 01:43.16 | MikeJ[Laptop] | 100 trying... |
| 01:43.18 | MikeJ[Laptop] | there ya go |
| 01:43.20 | tainted- | it's still dumbed down to ${DIALSTATUS} |
| 01:43.36 | tainted- | unless i'm missing something |
| 01:43.54 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 01:43.55 | MikeJ[Laptop] | you mean on a failure? |
| 01:44.03 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 01:44.03 | tainted- | yea or whatever |
| 01:44.04 | MikeJ[Laptop] | that should be a trivial patch |
| 01:44.11 | tainted- | so i can trap for responses |
| 01:44.12 | MikeJ[Laptop] | just add it |
| 01:44.20 | MikeJ[Laptop] | actually... |
| 01:44.24 | MikeJ[Laptop] | it may be there.. |
| 01:44.33 | MikeJ[Laptop] | in those sip functions that were added.. |
| 01:44.36 | MikeJ[Laptop] | I don't recall. |
| 01:44.50 | MikeJ[Laptop] | but would be trivial to store the last response in the pvt. |
| 01:45.10 | MikeJ[Laptop] | gotta smoke. |
| 01:46.54 | *** part/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 01:46.58 | *** join/#asterisk themikester60 (n=Owner@66-100-35-23-static.dsl.oplink.net) |
| 01:48.09 | themikester60 | Can anyone think of a reason why my asterisk server spits out "no compatible codecs found" when I'm trying to use g726 as the codec? At first I thought perhaps it was the phones that were screwed up, but I've tried 3 or 4 different models and none have worked. This is over sip by the way. |
| 01:48.57 | Qwell | themikester60: it isn't called g726 in asterisk |
| 01:49.12 | Qwell | or, maybe it is |
| 01:49.47 | themikester60 | Qwell, at voip-info.org they claimed it to be g726, but its entirely possible I have the name wrong, I just havn't seen anything indicating that yet though |
| 01:53.09 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 01:55.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:01.46 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 02:06.20 | *** join/#asterisk rickB|server (n=rickb@cpe-71-66-110-248.neo.res.rr.com) |
| 02:07.12 | rickB|server | Hello. I have been reading tutorials for the last week or so. And cant make a softphone test, I am using the Ekiga softphone for linux and attempting to use SIP to gain access to the Vmail I set up.. Any ideas? |
| 02:07.26 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:14.57 | *** join/#asterisk austinnichols10 (i=austinni@dsl-10-169.cofs.net) |
| 02:17.43 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 02:21.31 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 02:23.49 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 02:26.55 | *** part/#asterisk blebleble (n=ble@d149-67-206-171.col.wideopenwest.com) |
| 02:27.22 | *** join/#asterisk EastWolf (i=eastwolf@221.217.217.243) |
| 02:31.50 | EastWolf | hell everyone |
| 02:32.13 | *** part/#asterisk themikester60 (n=Owner@66-100-35-23-static.dsl.oplink.net) |
| 02:33.17 | austinnichols10 | o |
| 02:34.53 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 02:35.48 | ManxPower | BTW, has ANYONE gotten any form of DID to work in the USA into a Digium TDM400P? |
| 02:37.22 | *** join/#asterisk budmang (i=budman@12.206.134.162) |
| 02:37.27 | budmang | Anyone in? |
| 02:38.32 | budmang | root@102:/usr/src/zaptel-1.2# make clean |
| 02:38.32 | budmang | /bin/sh: line 1: modprobe: command not found |
| 02:38.32 | budmang | rm -f torisatool makefw tor2fw.h radfw.h |
| 02:40.07 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 02:51.37 | litage | hey guys, i've got a bit of a quandary happening. when i call North America on my GXP-2000 with Asterisk and one particular voip provider, as soon as the callee picks up, i'm disconnected. however, if i remove Asterisk from the equation, the problem disappears. any thoughts?: http://rafb.net/paste/results/z4KjoT23.html |
| 02:52.28 | *** join/#asterisk adker (n=adker@67-136-210-63.dsl1.glv.ny.frontiernet.net) |
| 02:52.28 | Strom_C | litage: you're using SIP, I assume? |
| 02:52.37 | litage | Strom_C: yes |
| 02:53.04 | Strom_C | litage: set canreinvite=no in the SIP configurations of both the Grandstream and the SIP provider |
| 02:53.57 | ManxPower | Not having any kind of decent internet access makes me feel like I live on an uncharted island at times |
| 02:56.42 | X-Rob | litage, that's a codec mismatch. |
| 02:56.53 | X-Rob | probably trying to use g729 and you don't have a g729 codec. |
| 03:04.37 | EastWolf | what's the password Asterisk@Home 2.7 |
| 03:10.26 | EastWolf | what's the password Asterisk@Home 2.7 WMware . who can tell me |
| 03:11.29 | budmang | test |
| 03:14.13 | litage | Strom_C: thanks!! that fixed the problem! |
| 03:14.50 | Strom_C | you're welcome! please deposit three dollars for the first one minute |
| 03:19.36 | litage | now that i know that the reinvites were causing the problem, i'm curious as to why reinvites were issued only when using grandstream products. if i remove "canreinvite=no" and make a call with a linksys spa-941, asterisk doesn't send reinvites |
| 03:21.53 | Strom_C | litage: is canreinvite=no set in the linksys's config? |
| 03:22.19 | litage | Strom_C: "in the linksys's config" as in within the device, or within the * configs? |
| 03:22.23 | Strom_C | asterisk |
| 03:22.35 | *** join/#asterisk codebreaker (n=codebrea@xserver.flexserv.de) |
| 03:23.19 | litage | Strom_C: no, "canreinvite=no" is not set within the * configs for the linksys devices |
| 03:24.14 | EastWolf | what's the password Asterisk@Home 2.7 WMware . who can tell me |
| 03:24.30 | Strom_C | EastWolf: please google for the documentation |
| 03:24.56 | Strom_C | litage: who knows. the linksys might not be set to reinvite, and the grandstream is likely on crack anyway |
| 03:24.58 | opus_ | anyone here recommend a VOIP headset/ gamer headset/ music headphones I can use? |
| 03:25.04 | opus_ | should I look for bluetooth ones? |
| 03:25.44 | gandhijee | whatever, i'll just keep my mouth shut till i get my test setup and figure out exactly what is goin onq |
| 03:25.51 | gandhijee | whoops |
| 03:26.28 | litage | opus_: i quite like the Sennheiser PC110 and PC120, and they're quite inexpensive |
| 03:26.39 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 03:28.28 | litage | Strom_C: i've done packet captures when making calls without "canreinvite=no" using the spa-941 and gxp-2000. both phones send only one invite to asterisk |
| 03:29.07 | litage | Strom_C: in the packet captures, the first 9 packets are exactly the same for both the spa-941 and gxp-2000 |
| 03:30.25 | litage | Strom_C: however, when using the spa-941, packet #10 is an ACK from the spa-941 to asterisk, while packet #10 using the gxp-2000 is a reinvite from asterisk to the voip provider |
| 03:31.09 | litage | Strom_C: considering packets 1-9 were the "same", i can't figure out why asterisk only sends the reinvite if the gxp-2000 is making the call |
| 03:32.39 | *** join/#asterisk Abydos313 (n=abydos31@adsl-71-129-61-88.dsl.irvnca.pacbell.net) |
| 03:33.14 | opus_ | litage you don't have a global canreinvite=no setting set or each of your peers don't have dontreinvite |
| 03:34.35 | litage | opus_: what i described above occured when i didn't have "canreinvite=no" anywhere in my configs |
| 03:35.09 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 03:35.18 | opus_ | liteage: Yes. re-think about it. You are getting correct behavior |
| 03:36.02 | EastWolf | ok. thank u Strom_C |
| 03:36.16 | litage | opus_: i'm afraid i don't understand why reinvites would occur for the gxp-2000 but not for the spa-941 |
| 03:39.41 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 03:40.32 | iceyp | guys i trying to make a call from Cisco 7912 -----g729---- asterisk ---- ulaw ----- voip provider .... but i keep getting this error |
| 03:40.37 | iceyp | May 11 15:39:49 WARNING[59198]: channel.c:2719 ast_channel_make_compatible: No path to translate from SIP/akl.italk.co.nz-4508 () to SIP/099742910-78b9 (ulaw,g729) |
| 03:40.51 | Strom_C | iceyp: do you have a g729 license on your asterisk box? |
| 03:40.51 | iceyp | any ideas? |
| 03:40.55 | iceyp | yes 2 |
| 03:41.02 | iceyp | melissa*CLI> show g729 |
| 03:41.03 | iceyp | 0/0 encoders/decoders of 2 licensed channels are currently in use |
| 03:41.08 | iceyp | digium ones |
| 03:41.14 | iceyp | it works fine if i call my voicemail |
| 03:41.27 | iceyp | just doesnt translate 729 to ulaw |
| 03:42.04 | Strom_C | iceyp: odd. do you have canreinvite=no in both SIP configs? |
| 03:42.18 | iceyp | umm, dont think i do |
| 03:42.33 | iceyp | just doing a quick upgrade to see if that fixs it |
| 03:42.34 | Strom_C | try adding that, doing a SIP reload, and try the call again |
| 03:43.20 | iceyp | why canreinvite no and not yes? |
| 03:43.37 | Strom_C | just try it and see if it works |
| 03:43.40 | iceyp | kk |
| 03:44.09 | Strom_C | my theory is that it's not working because asterisk wants to make the two channels talk the same codec if it anticipates reinvites |
| 03:44.26 | Strom_C | besides, why would you want to talk on g729 anyway? |
| 03:44.32 | iceyp | same |
| 03:44.33 | iceyp | May 11 15:44:14 WARNING[59198]: channel.c:2719 ast_channel_make_compatible: No path to translate from SIP/akl.italk.co.nz-a0ec () to SIP/099742910-b14a (ulaw,g729) |
| 03:44.33 | iceyp | <PROTECTED> |
| 03:44.39 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:44.41 | litage | Strom_C: what's wrong with g729? |
| 03:44.46 | Strom_C | it sounds like ass |
| 03:44.51 | Strom_C | is what's wrong with it :) |
| 03:44.54 | *** join/#asterisk DeaLER123 (i=Tester@toronto-HSE-ppp4227443.sympatico.ca) |
| 03:45.02 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 03:45.12 | fjean | hi guys |
| 03:45.13 | DeaLER123 | is there any good docs on what the difference between vpm revision A and B are? |
| 03:45.37 | DeaLER123 | for TDM2400P echo cancellation modules |
| 03:45.47 | fjean | anybody had the kernerl "spinlock.h" bug issue while compiling zaptel ? |
| 03:45.53 | DeaLER123 | no |
| 03:46.09 | [TK]D-Fender | fjean, Well known CentOS bug |
| 03:46.10 | fjean | s/kernerl/kernel/ |
| 03:46.12 | [TK]D-Fender | ~centosbug |
| 03:46.14 | jbot | [centosbug] a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 03:46.46 | litage | while examining the INVITEs that my Grandstream GXP-2000 sends, i've noticed that it puts ";user=phone" after various sip addresses. what does that mean? eg: Request-Line: INVITE sip:001127216742229@202.168.41.242;user=phone SIP/2.0 |
| 03:47.19 | fjean | d-fender: right, I did this part, but I still get the error, is there any step I should do before trying to compile ? |
| 03:47.50 | Strom_C | litage: sounds like you should curl up with a fire, a cup of warm cocoa, and a copy of the RFC :) |
| 03:48.35 | [TK]D-Fender | fjean : In running into it myself, you just follow the instructions, and thats it. I did it from clean install |
| 03:48.39 | DeaLER123 | does anyone know what changes were made to digiums new echo cancellation chips? |
| 03:49.01 | fjean | d-fender: ok |
| 03:49.02 | litage | Strom_C: checking the rfc... |
| 03:49.04 | *** part/#asterisk opus_ (n=opus@68.216.187.60) |
| 03:50.21 | Strom_C | DeaLER123: yes, they had to add special anti-echo electrons to the silicon substrate |
| 03:50.27 | Strom_C | </if phone people were audiophiles> |
| 03:51.16 | catch23 | ooo anti-echo electrons... although, i prefer the cancellation set protons |
| 03:51.36 | *** join/#asterisk bmg505 (n=leon@c1-130-4.rndf.isadsl.co.za) |
| 03:53.10 | [TK]D-Fender | No, protons would cause an inverse cascade in the ODN relays causing a feedback-loop in its duotronic core! Clearly not acceptable! |
| 03:53.24 | fjean | d-fender: got it, it worked ;-) |
| 03:54.18 | DeaLER123 | the first revision of the echo cancel was he same thing as MARk2 Aggressor |
| 03:54.31 | DeaLER123 | thats what it sounded like |
| 03:55.00 | DeaLER123 | i took out the echo cancel chip and used software with default algorithm and it sounded bette |
| 03:56.26 | DeaLER123 | so i emailed digium and they are RMA'ing the chip and sending me a revision b |
| 03:56.44 | DeaLER123 | im curious as to what kind of changes the chip will have |
| 03:57.49 | fjean | d-fender: you played with the udev files created by make install or you just left them as is ? |
| 03:58.45 | litage | Strom_C: alrighty, i've read through several sections of the sip rfc that deal with URIs and whatnot. i understand that adding ";user=phone" to the end of a URI is valid. however, i don't fully understand what the point of doing so is. is it just to pass additional info on to the other parties in the call? |
| 03:58.59 | [TK]D-Fender | fjean : I only worked with RH based systems infrequently. Never had any issues except the obvious bug I just pointed you to, and knew about it before |
| 03:59.11 | fjean | ok, thanks |
| 04:00.18 | *** join/#asterisk Smi|k (n=ling@cpe-66-74-216-66.san.res.rr.com) |
| 04:00.24 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
| 04:00.56 | Smi|k | I'm struggling with a polycom phone... we moved an 8 phone office and after the move 2 of the phones didnt seem to work... I assumed corrupt config files from power cycles... |
| 04:01.35 | Smi|k | so I am at home and have been playing with this 301 for about 3 hours and finnaly got it to boot up, but it does not see the network... it is plugged directly into the cable modem, but the cable modems green link light is not on, and the orange activity light is not lit |
| 04:02.00 | Smi|k | on the polycom phone in the network stats area it says "Lan Port: Inactive" any idea how to turn it back on or diagnose it? |
| 04:02.31 | litage | Smi|k: "inactive" is usually different from "disabled" |
| 04:03.52 | [TK]D-Fender | Smi|k : My guess, got it plugged into the "PC" port? |
| 04:04.01 | Smi|k | if I boot with DHCP client enabled (as I should be at home, no static IP) the phone will not boot at all, it just fails to connect to dhcp server.... if I tell it the IP address it currently has and try, it boots saying "network link is down" once booted |
| 04:04.22 | Smi|k | currently PC is not linked to phone at all.... and this is a phone that powers from the network cable too, so I know it must be in the lan plug as its the only one that would ever fit into it |
| 04:04.28 | ManxPower | It looks like SIP/akl.italk.co.nz-a0ec doesn't like the codecs you are accepting |
| 04:04.56 | [TK]D-Fender | Smi|k : If could simply go dead. I RMA'd one like that 2 months ago that refused to see the network at all... |
| 04:05.30 | Smi|k | the green light on the cable modem is deff not on.... any idea where in the boot process that should go green? |
| 04:05.42 | Smi|k | the weird thing is two of the phones are doing this, a 201 and a 301 |
| 04:06.06 | Smi|k | its as if their network ports just fried or something |
| 04:06.12 | ManxPower | Smi|k, confim the network cable is in the LAN port of the phone and not the PC port. |
| 04:06.23 | Smi|k | its deff in the lan port or it would not power on at all |
| 04:06.30 | Smi|k | for kicks I just looked at it too :) |
| 04:06.45 | *** join/#asterisk nortex (n=nortex@host-69-59-87-157.nctv.com) |
| 04:06.49 | ManxPower | Um, the polycom 30x use an external power supply unless you have a PoE cable |
| 04:07.06 | [TK]D-Fender | Smi|k : It happens. |
| 04:07.43 | Smi|k | these 301's have no spot for a power cord to plug directly into back of phone, it plugs into the network cable itself |
| 04:07.45 | nortex | Does anyone have ooH323c workin on a CentOS 4.x setup? |
| 04:08.10 | ManxPower | Smi|k, Must be another not well known change from the 300 to the 301. |
| 04:08.23 | Smi|k | what causes the green network light to turn on for cable modems and routers etc..etc.. typically? does that have to do with com at all, or if there is no green light the cables are not making a complete connection |
| 04:08.27 | ManxPower | all the 300s have a power port on the phone. |
| 04:08.38 | ManxPower | Smi|k, no light means no signal |
| 04:08.41 | Strom_C | Smi|k: it varies from manufacturer to manufacturer |
| 04:08.46 | ManxPower | sure you are not using a crossover cable somewhere? |
| 04:09.03 | Smi|k | oops, this is a 501 :) |
| 04:09.31 | Smi|k | I always think 3 line instances = 301 and 6 line instances = 601 |
| 04:09.39 | Smi|k | but no, it cant make that much sense |
| 04:09.54 | Smi|k | any idea what causes green on the 501 -> cable modem? because no light is on |
| 04:10.05 | Strom_C | wait wait |
| 04:10.09 | Smi|k | when I plug cable modem to laptop the light is instantly green |
| 04:10.14 | Strom_C | you're plugging the 501 DIRECTLY INTO THE MODEM? |
| 04:10.25 | Smi|k | yes |
| 04:10.34 | Strom_C | *blink* |
| 04:10.39 | Strom_C | *shakes head* |
| 04:10.43 | Strom_C | *blinks again* |
| 04:10.49 | Strom_C | um, ok then |
| 04:11.08 | Smi|k | router was used at work, now that I'm home its just testing |
| 04:11.23 | Strom_C | Smi|k: perhaps you need a crossover cable |
| 04:11.40 | Smi|k | that sould not be the case.... |
| 04:11.44 | Strom_C | or a straight-through, if you're using a crossover right now |
| 04:12.23 | Smi|k | the cable I'm using is the PoE cable, so I cant change it |
| 04:12.38 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 04:12.40 | Smi|k | so I assume it must be the correct type to go from router/modem to the phone |
| 04:12.46 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:13.01 | Strom_C | assumptions will kill you |
| 04:13.09 | Smi|k | plus it worked before and now doesnt |
| 04:13.29 | Smi|k | i.e. there is no way to plug another cable into this phone, it must use the PoE cable |
| 04:13.30 | Strom_C | it worked before when plugged directly into the cable modem? |
| 04:13.32 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:13.39 | Smi|k | it worked before when plugged into the router |
| 04:13.50 | Strom_C | yes, but it's not working when plugged into the cable modem |
| 04:13.57 | Smi|k | or the router |
| 04:14.06 | Smi|k | my cable co has a router on the other end of this cable modem |
| 04:14.15 | Strom_C | maybe you fucked the PoE cable up |
| 04:14.16 | Smi|k | it provisions DHCP etc..etc.. just as my router does |
| 04:14.20 | Smi|k | so same type of calbe.... |
| 04:14.25 | Strom_C | Smi|k: thats not the point |
| 04:14.25 | Smi|k | I just want to know what cuases the link to turn green |
| 04:14.34 | Strom_C | the point is the type of ethernet interface on teh modem |
| 04:14.39 | Strom_C | and on the phone |
| 04:14.44 | Smi|k | if it is simply grounding of all cables, or if it is chips on the network hardware sending a quick com to eachother and then it lights green |
| 04:14.57 | Smi|k | because I think with the laptop the light is green even before the OS boots |
| 04:15.15 | Smi|k | phone uses same type of port as laptop uses |
| 04:15.22 | Smi|k | and laptop works plugged directly into cable modem |
| 04:15.25 | Strom_C | Smi|k: listen to me. the laotop's ethernet interface is probably autosensing and can switch between regular and crossover type ports |
| 04:15.34 | Smi|k | ooohhhhh |
| 04:15.35 | Strom_C | whereas the phone probably isn't autosensing |
| 04:15.50 | Smi|k | so if I plug it into a router right now it might work again? |
| 04:15.54 | *** join/#asterisk galasura (n=eddie@60.51.204.178) |
| 04:15.54 | Strom_C | so will you please try a crossover cable between the injector and the modem? |
| 04:16.06 | galasura | hi all |
| 04:16.23 | Smi|k | is there any little dongle that will convert the two? |
| 04:16.34 | Smi|k | as I cannot plug anything other than the polycom poe cable into my phone |
| 04:16.45 | Strom_C | Smi|k: your cable modem does poe? |
| 04:16.49 | Smi|k | thanks Strom, sorry I'm not realizing the auto-switching thing |
| 04:17.15 | Smi|k | Strom_C, no, the lan cable plugged into the polycom has a plug for DC adapter on it near the cable modem, adapter to the wall plugs into the PoE cable |
| 04:17.26 | Smi|k | other part of cable ot cable modem, end goes to phone |
| 04:17.32 | galasura | i need some advice & solution, can anyone help? |
| 04:17.39 | Strom_C | Smi|k: just put a switch between the two |
| 04:17.48 | Strom_C | galasura: just ask your question and someone will answer |
| 04:18.02 | Smi|k | okay, thanks, I'll try all this |
| 04:18.10 | Strom_C | I've got to go |
| 04:18.15 | Smi|k | no way to switch the phone to use crossover mode right? |
| 04:18.29 | Strom_C | Smi|k: I doubt it |
| 04:18.57 | nortex | can I get some help on ooH323c |
| 04:19.24 | *** join/#asterisk vo (n=saigon@pdpc/supporter/basic/vo) |
| 04:19.47 | galasura | <PROTECTED> |
| 04:20.07 | galasura | does anyone had this problem before? |
| 04:22.08 | galasura | anybody had problem where u cannot leave voicemail? |
| 04:22.11 | vo | your log file is a wav? |
| 04:22.39 | *** part/#asterisk vo (n=saigon@pdpc/supporter/basic/vo) |
| 04:26.20 | nortex | what is the easiest way to setup a H323 trunk in asterisk |
| 04:34.07 | *** join/#asterisk galasura (n=eddie@60.51.204.178) |
| 04:35.03 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-110-89.aubnin.fios.verizon.net) |
| 04:36.09 | CunningPike | galasura: what does 'ls -l /var/spool/asterisk/voicemail/default/1000/INBOX/' say? |
| 04:37.34 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 04:38.13 | CunningPike | Gotta switch networks - brb |
| 04:39.06 | galasura | CunningPike: there's only msg0000.txt |
| 04:39.20 | galasura | no msg0000.wav |
| 04:39.35 | CunningPike | galasura: Can you post your error message again? |
| 04:41.09 | CunningPike | galasura: If you delete the msg0000.txt file, all will be well |
| 04:41.32 | galasura | -- Recording the message |
| 04:41.32 | galasura | <PROTECTED> |
| 04:41.32 | galasura | /var/spool/asterisk/voicemail/default/8086/INBOX/msg0005 format: wav49, |
| 04:41.32 | galasura | (nil) |
| 04:41.41 | galasura | May 11 10:52:07 WARNING[21267]: |
| 04:41.41 | galasura | app.c:706 ast_play_and_record: Error creating writestream |
| 04:41.42 | galasura | '/var/spool/asterisk/voicemail/default/8086/INBOX/msg0005', format |
| 04:41.42 | galasura | 'wav49' |
| 04:41.51 | galasura | May 11 10:52:07 WARNING[21267]: |
| 04:41.51 | galasura | app_voicemail.c:787 base_encode: Failed to open log file: |
| 04:41.51 | galasura | /var/spool/asterisk/voicemail/default/8086/INBOX/msg0005.WAV: No such |
| 04:41.51 | galasura | file or directory |
| 04:43.12 | galasura | CunningPike: I've tried deleting msg0000.txt ... still won't solve the problem |
| 04:43.27 | *** join/#asterisk terrapen_ (n=cjs@chrissnell.dsl.xmission.com) |
| 04:43.33 | CunningPike | galasura: ~pb |
| 04:43.37 | terrapen_ | anybody receiving faxes over their PRI? |
| 04:43.42 | CunningPike | ~pb |
| 04:43.44 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 04:45.05 | CunningPike | galasura: That last error message was for a different mailbox - what about the original mailbox - is that working now that you've deleted the file? |
| 04:45.21 | galasura | ok .. i've paste it |
| 04:46.05 | galasura | CunningPike: no, all available voicemailbox face the same problem |
| 04:46.26 | CunningPike | galasura: Do they all have orphaned text files in them? |
| 04:46.39 | CunningPike | galasura: Just ruling that out..... |
| 04:47.22 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 04:47.25 | galasura | CunningPike: Yes. Only text files. No wav files. |
| 04:47.41 | CunningPike | OK - clear all the orphaned text files |
| 04:48.28 | CunningPike | Leave another message and delete it and see if the text file gets left behind |
| 04:48.53 | galasura | CunningPike: hang on :) |
| 04:49.10 | CunningPike | galasura: I got nothing but time ;) |
| 04:50.17 | terrapen_ | im not sure faxes can be sent via TDMoE and PRI |
| 04:50.23 | terrapen_ | but im gonna try |
| 04:54.26 | galasura | CunningPike: there's only msg0000.txt is in the folder when I leave a new voicemail |
| 04:54.38 | CunningPike | gala |
| 04:54.51 | CunningPike | galasura: OK - that narrows it down a bit |
| 04:55.09 | CunningPike | And the error message you get when trying to record a new message is? |
| 04:55.19 | CunningPike | Use pastebin if it's more than 3 lines |
| 04:57.29 | *** join/#asterisk isamar (n=isamar@202.95.220.92) |
| 04:57.38 | isamar | hi folks.. |
| 04:58.05 | isamar | trying to unlock Linksys WRTP54G.. anyone did that already? |
| 05:00.31 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 05:01.55 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 05:03.48 | *** part/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 05:03.57 | galasura | CunningPike: ok. I've paste it. |
| 05:04.20 | CunningPike | galasura: Care to provide the link? :D |
| 05:05.21 | galasura | CunningPike: sorry about that .. http://pastebin.com/710826 |
| 05:08.07 | CunningPike | galasura: Are you running asterisk as root? |
| 05:09.06 | CunningPike | galasura: Also, what formats are you loading in modules.conf? |
| 05:09.49 | galasura | CunningPike: yes as root |
| 05:10.26 | CunningPike | OK - what is the format= line from voicemail.conf, and what format_ modules are you loading in modules.conf |
| 05:10.33 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 05:10.54 | *** join/#asterisk bigmac4029 (n=mtur2848@CPE-60-231-112-137.qld.bigpond.net.au) |
| 05:11.24 | lme | hi all |
| 05:11.30 | bigmac4029 | hiya lme |
| 05:11.59 | galasura | CunningPike: http://pastebin.com/710835 |
| 05:12.08 | bigmac4029 | got a digium g729 question... |
| 05:13.06 | brookshire | bigmac: okay, what's up? |
| 05:13.20 | Qwell | brookshire: ! |
| 05:13.25 | bigmac4029 | licensing.. i license = 1 connected call at a time via server? |
| 05:13.29 | brookshire | qwell, omg! |
| 05:13.31 | bigmac4029 | 1 license* |
| 05:13.34 | Qwell | bigmac4029: one transcoded channel at a time |
| 05:13.51 | brookshire | most calls take 2 channels |
| 05:13.56 | brookshire | one in and one out |
| 05:14.01 | bigmac4029 | so if I want 30 simultanious calls... |
| 05:14.13 | Qwell | You'd need to figure out how much transcoding needs to be done |
| 05:14.22 | Qwell | It's not as simple as it seems |
| 05:14.36 | brookshire | like.. you can have 2 t1s which is 46 lines |
| 05:14.47 | brookshire | but they do not need g729 |
| 05:14.47 | lme | anybody having experience with E1 in israel and * here ? |
| 05:14.52 | CunningPike | galasura: Well, nothing strikes me there...... |
| 05:14.57 | bigmac4029 | already bought and installed 1 license. PSTN gateway connections are fine now |
| 05:15.04 | brookshire | you only need g729 on the voip side, if you are running the that codec |
| 05:15.10 | CunningPike | galasura: It's usually permissions or a format mis-match |
| 05:15.13 | bigmac4029 | we will have about 100,000 customers |
| 05:15.18 | bigmac4029 | lol |
| 05:15.35 | bigmac4029 | not on one server of course |
| 05:16.06 | *** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca) |
| 05:16.11 | bigmac4029 | ok, i'll purchase 30 licenses to begin with? |
| 05:16.42 | brookshire | if you know all 30 are going to be active and running g729 that's a safe bet |
| 05:16.45 | galasura | CunningPike: I've changed every mailbox to permission 777 |
| 05:16.56 | bigmac4029 | yep, thx heaps brook =) |
| 05:17.03 | CunningPike | galasura: No need if you're running as root |
| 05:17.16 | lme | :( |
| 05:17.18 | CunningPike | galasura: I'm out of ideas....... |
| 05:18.47 | galasura | CunningPike: It's ok. I'll try to recompile asterisk. |
| 05:18.56 | CunningPike | galasura: |
| 05:19.02 | CunningPike | galasura: OK |
| 05:19.09 | galasura | CunningPike: Thanks for your help |
| 05:19.23 | CunningPike | galasura: It was nothing |
| 05:19.28 | CunningPike | galasura: Literally ;) |
| 05:19.59 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:24.13 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 05:27.05 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:33.28 | *** join/#asterisk drray (n=drray@c-67-183-123-24.hsd1.wa.comcast.net) |
| 05:39.42 | lme | nobody for E1 in israel with bezeq operator ? :( |
| 05:40.00 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:43.10 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 05:44.07 | jsaunders | Anyone familiar w/ 'tos=0x18' in sip.conf or iax.conf? As in does it ever help? |
| 05:44.38 | dlynes | jsaunders: you need a qos router/firewall that understands it |
| 05:44.55 | dlynes | jsaunders: if you're not using such a beast, it won't do anything |
| 05:45.24 | jsaunders | So my isp won't honor it. |
| 05:45.35 | dlynes | jsaunders: don't see why they would |
| 05:45.48 | jsaunders | Makes sense, just thought I'd inquire. |
| 05:45.51 | dlynes | jsaunders: unless they're going to benefit by it |
| 05:46.01 | dlynes | jsaunders: you might want to try emailing them about it though |
| 05:46.16 | dlynes | jsaunders: but if they offer voip as a service, i doubt they'd be willing to cooperate |
| 05:46.45 | dlynes | galasura: are you still having that problem? |
| 05:51.03 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:51.33 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 05:54.07 | *** join/#asterisk lorinc (n=ang@caracas-3289.adsl.interware.hu) |
| 05:56.06 | galasura | dlynes: Yes, I am |
| 05:56.29 | galasura | dlynes: any ideas? |
| 05:56.39 | dlynes | galasura: I take it Windows is your usual operating system? |
| 05:57.20 | galasura | dlynes: No. I use both Windows and Linux |
| 05:57.31 | dlynes | galasura: but predominantly windows, right? |
| 05:57.46 | dlynes | i.e. you're kinda new to linux? |
| 05:58.07 | galasura | dlynes: Yes |
| 05:58.16 | dlynes | galasura: ok, I thought so |
| 05:58.24 | galasura | dlynes: my current pc is SuSe10 |
| 05:58.28 | dlynes | galasura: the reason i was asking wasn't to make you feel uncomfortable |
| 05:58.47 | dlynes | galasura: it was to determine if my suspicion about what the problem was might have any merit |
| 05:59.13 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 05:59.14 | galasura | dlynes: what do u have in mind |
| 05:59.23 | dlynes | galasura: the error you were getting was: galasura app_voicemail.c:787 base_encode: Failed to open log file: |
| 05:59.23 | dlynes | galasura /var/spool/asterisk/voicemail/default/8086/INBOX/msg0005.WAV: No such |
| 05:59.23 | dlynes | galasura file or directory |
| 05:59.30 | dlynes | right? |
| 05:59.41 | galasura | dlynes: right |
| 06:00.08 | dlynes | Now, could you paste the output of 'ls -l /var/spool/asterisk/voicemail/default/8086 for me? |
| 06:01.10 | dlynes | Just paste it to the channel window...it should only be one line |
| 06:02.15 | galasura | dlynes: it's ok now. I've recompile asterisk, which solved the voicemail problem |
| 06:02.32 | dlynes | so what was the problem, then? |
| 06:03.19 | galasura | dlynes: i'm not sure. it was working fine before. out of a sudden, we just can't leave voicemails |
| 06:03.36 | dlynes | galasura: ah |
| 06:04.01 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:04.06 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 06:04.24 | dlynes | galasura: anyways, my suspicion was that INBOX was actually inbox or Inbox, or that one of the directories along that path did not have 755 privilege |
| 06:04.51 | galasura | dlynes: I did change the permission to 777 |
| 06:04.59 | dlynes | galasura: Linux and UNIX pathnames are case-sensitive, whereas Windows pathnames are case preservative |
| 06:05.03 | sjobeck | hi all, can someone point me to where the voicemail recording volume booster is enabled? |
| 06:05.10 | dlynes | galasura: on all directories? |
| 06:05.23 | galasura | dlynes: yes |
| 06:05.26 | dlynes | galasura: and you should never use 777, unless you want your box cracked |
| 06:06.00 | dlynes | galasura: 777 means give everyone full permission to that directory/file |
| 06:06.04 | galasura | dlynes: i changed it for testing and trouble shooting purpose only |
| 06:06.10 | dlynes | galasura: ah |
| 06:06.25 | harryvv | Anyone seen this before. I call my personal extention from the ivr it looks like it rings the other extention which at this time is disconected and my phone does not ring. it then goes to voice mai. This is from the cli |
| 06:06.27 | dlynes | sjobeck: there isn't one |
| 06:06.28 | harryvv | <PROTECTED> |
| 06:06.28 | harryvv | <PROTECTED> |
| 06:06.38 | harryvv | What does this mean exactly:? |
| 06:06.55 | galasura | dlynes: thanks for your help |
| 06:07.02 | dlynes | harryvv: someone's got call forwarding turned on, for that phone, more than likely |
| 06:07.30 | harryvv | nope. I checked it and reset it again |
| 06:07.32 | dlynes | galasura: yeah, anyways...it was probably that the case for the letters in INBOX wasn't correct, then |
| 06:07.56 | galasura | dlynes: yeah, it might be that |
| 06:08.16 | dlynes | galasura: actually nope...that wouldn't be it...INBOX is all in caps |
| 06:08.33 | harryvv | some kind of error in the db so it cannot kill the call forward. |
| 06:08.34 | dlynes | galasura: but the voicemail shouldn't have WAV all in caps lock |
| 06:08.48 | dlynes | harryvv: no...i don't mean in the asterisk macro database |
| 06:08.56 | dlynes | harryvv: i mean on the actual sip phone, itself |
| 06:09.13 | harryvv | http://pastebin.ca/55005. |
| 06:09.15 | harryvv | look at that |
| 06:09.45 | dlynes | beauty |
| 06:09.58 | harryvv | sooo |
| 06:10.12 | harryvv | I guess i have to go into the db manually and delete that or change it? |
| 06:10.12 | dlynes | Does it give you any extra information with higher verbosity? |
| 06:10.15 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 06:10.43 | harryvv | What are the steps to increase verbolsity |
| 06:10.48 | harryvv | verbosity |
| 06:10.49 | harryvv | :) |
| 06:10.53 | dlynes | set verbose 10 |
| 06:11.00 | dlynes | or whatever value you want to use |
| 06:11.22 | harryvv | okay will try again |
| 06:13.44 | harryvv | http://pastebin.ca/55009 |
| 06:13.54 | harryvv | look at past below window below verbose 10 |
| 06:16.25 | kaldemar | harryvv: have you checked the settings in the phone? |
| 06:16.33 | *** join/#asterisk Eggplants (n=none@dsl-216-155-215-185.cascadeaccess.com) |
| 06:17.14 | kaldemar | that SIP 302 is coming from the ip phone, it's not generated by asterisk. |
| 06:17.16 | harryvv | polycome ip500. this is asterisk@home that uses dtmf preset keys to change the behavior of asterisk such as fall forwarding. |
| 06:18.11 | harryvv | yes, the phone cannot do this |
| 06:19.15 | dlynes | harryvv: the highest useful value for verbosity looks like it's 6 |
| 06:19.26 | harryvv | usefull |
| 06:19.29 | harryvv | not 10 |
| 06:19.30 | harryvv | ? |
| 06:19.37 | dlynes | 10 is useful |
| 06:19.42 | dlynes | it's higher than 5 also |
| 06:19.45 | harryvv | thats what its set to |
| 06:20.22 | dlynes | harryvv: what does your full.log file look like for that period of time? |
| 06:20.44 | dlynes | kaldemar: I already mentioned that to him |
| 06:21.06 | dlynes | kaldemar: however, he's got other issues where the asterisk database key is not being deleted for some specific key |
| 06:21.15 | kaldemar | dlynes: i saw you mention it, and thought that he maybe missed it because he keeps digging at asterisk's side. |
| 06:21.32 | harryvv | yea |
| 06:21.49 | dlynes | harryvv: but, harryvv the sip 302 is actually generated by the phone, not by asterisk |
| 06:21.57 | harryvv | ohh really |
| 06:21.58 | harryvv | mmm |
| 06:22.00 | dlynes | harryvv: i'm just trying to solve your database problem |
| 06:22.05 | harryvv | mabey its a xml configuration? |
| 06:22.17 | dlynes | harryvv: does your phone get configured via xml? |
| 06:22.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:22.24 | harryvv | about a week ago I made some minor changes but it has worked since then. |
| 06:22.28 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 06:22.51 | harryvv | dlynes you know xml in the ip500s? |
| 06:22.58 | dlynes | harryvv: nope...never used them |
| 06:23.03 | harryvv | k |
| 06:23.16 | dlynes | kaldemar might have, though |
| 06:23.33 | kaldemar | ip500's manual says it is capable in call forwarding. |
| 06:23.53 | kaldemar | sorry, i've never used polycom phones. |
| 06:23.54 | dlynes | i think every phone that costs more than $10 is capable of call forwarding :) |
| 06:24.18 | dlynes | it's a pretty basic sip phone feature |
| 06:24.38 | kaldemar | dlynes: they are, i just had to be sure bacuse harryvv said that the phone can't do it. :) |
| 06:24.53 | dlynes | yep |
| 06:25.02 | dlynes | i don't always believe everything people tell me though :) |
| 06:25.35 | harryvv | okay I never seen any options in the phone that it could. |
| 06:25.40 | dlynes | i don't need to get the polycom to do it |
| 06:25.49 | dlynes | so no point me wasting my time figuring out how |
| 06:26.20 | dlynes | harryvv: have you looked in your full log file yet for the database problems? |
| 06:26.39 | harryvv | no |
| 06:26.44 | harryvv | which one should I look at |
| 06:26.52 | dlynes | Your full log file |
| 06:27.06 | dlynes | Check your logger.conf file to find out what the name of it would be, and where it is |
| 06:27.28 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 06:27.39 | harryvv | its not in /var/log/asterisk |
| 06:28.16 | harryvv | k found it |
| 06:29.03 | watchy | if i say have 2 4 port fxo cards can i tell asterisk to send calls incoming on certain channels to certain extensions? |
| 06:29.16 | dlynes | watchy: yes |
| 06:29.36 | harryvv | only have this |
| 06:29.38 | harryvv | full => notice,warning,error,debug,verbose |
| 06:29.47 | harryvv | everything else is commented out |
| 06:29.55 | dlynes | watchy: define different channel groups, and define different incomgin contexts for each channel group |
| 06:30.04 | watchy | i got 2 offices, i wanna dedicate port 1-4 on card 1 to a certain extension and 1-4 on card2 on another extension |
| 06:30.08 | dlynes | harryvv: so /var/log/asterisk/full would be your file then |
| 06:30.21 | watchy | dlynes: sweet |
| 06:30.33 | watchy | i dunno how to do that but now that i know its possible |
| 06:30.35 | watchy | ill figure it out |
| 06:31.03 | dlynes | watchy: so [channels] group = 1 ; channel = 1-4 ; group = 2 ; channel = 5-8 |
| 06:31.16 | harryvv | okay going to test this again and read the log |
| 06:31.16 | dlynes | watchy: of course you need to put all the rest of your stuff in there, too |
| 06:31.30 | watchy | yea |
| 06:31.39 | watchy | ah sweet dlynes |
| 06:31.55 | watchy | i just put in a 28 phone asterisk system spanning a shitload of locations |
| 06:31.56 | dlynes | watchy: the directive might be channels...I can't remember if it's channel or channels before the equals sign |
| 06:32.06 | watchy | i dont know alot about asterisk but i can make phones work and figure it out |
| 06:32.18 | watchy | im about to put like 10 atas into service |
| 06:32.31 | dlynes | watchy: good luck with that |
| 06:32.45 | watchy | dlynes: i think i may trade being a net admin in for doing voip |
| 06:32.51 | *** join/#asterisk CKGL (n=Cglob@202.8.86.162) |
| 06:32.55 | dlynes | watchy: heh |
| 06:33.00 | watchy | it just intrest me |
| 06:33.10 | dlynes | watchy: i'd tread the waters a little more before jumping in head first :) |
| 06:33.17 | watchy | yea i agree |
| 06:33.29 | watchy | i sold this phone systems pretty much for cost |
| 06:33.32 | dlynes | watchy: you might not like the headaches associated with voip :) |
| 06:33.37 | watchy | so they wuold let me implemnet it and learn it for em |
| 06:33.57 | watchy | they are bunkers where they make bombs and have to have phones in them |
| 06:34.21 | watchy | but didnt wanna run lines to each one. so wep ut up a tower and broadcasted 802.11b across the area |
| 06:34.25 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:34.32 | watchy | put in like bridges and hubs and hooking up atas in them |
| 06:34.49 | orlok | hmm |
| 06:34.57 | orlok | bombs and 802.11b and voip |
| 06:34.57 | dlynes | watchy: 802.11b's more reliable than 802.11g? |
| 06:34.58 | orlok | good mix! |
| 06:35.07 | orlok | :) |
| 06:35.09 | watchy | actually i think we did g |
| 06:35.16 | watchy | yea haha they make all kinds of bombs |
| 06:35.25 | brookshire | dlynes: g drops to b if i can't get a solid connection |
| 06:35.28 | dlynes | watchy: ah...just noticed you mentioned 802.11b :) |
| 06:35.31 | watchy | i live in a small arkansas town that makes like 50% of all weaponry |
| 06:35.36 | dlynes | brookshire: yeah, i know |
| 06:35.38 | Qwell | brookshire: does b drop to a?! |
| 06:35.40 | watchy | we make missles and shit here |
| 06:35.50 | dlynes | Qwell: no...a is incompatible with b and g |
| 06:35.57 | orlok | Qwell: different freq. i think |
| 06:36.03 | brookshire | qwell: b covers 1,2,5, and 11mbit i believe |
| 06:36.03 | watchy | b/g is like 2.4ghz |
| 06:36.07 | Qwell | dude |
| 06:36.07 | watchy | a is like 5.2 |
| 06:36.09 | Qwell | nubs |
| 06:36.09 | watchy | or something |
| 06:36.12 | Qwell | sarcasm :P |
| 06:36.16 | dlynes | qwell's a nub |
| 06:36.24 | watchy | qwells my boyfriend |
| 06:36.34 | Qwell | wasim: eh? |
| 06:36.35 | watchy | this fucking fxotune takes to long |
| 06:36.39 | Qwell | erm, watchy |
| 06:36.42 | *** join/#asterisk Dr^ZigMan (n=DrZigMan@87.146.204.68.cfl.res.rr.com) |
| 06:36.44 | watchy | i hope its worth it |
| 06:36.51 | dlynes | ah...thought i'd seen you two down there in the bunker together...*wink* *wink* |
| 06:36.56 | watchy | haha |
| 06:37.12 | Qwell | please, if I was gonna go gay... |
| 06:37.13 | Qwell | nevermind |
| 06:37.19 | Qwell | let's not go there... |
| 06:37.21 | dlynes | lol |
| 06:37.36 | dlynes | besides |
| 06:37.36 | brookshire | you had your chance at astricon! |
| 06:37.39 | brookshire | lol |
| 06:37.41 | dlynes | who said watchy was a guy? |
| 06:37.48 | watchy | oh i am a dude |
| 06:37.52 | Qwell | brookshire: did I? |
| 06:38.00 | brookshire | haha |
| 06:38.01 | watchy | whens astricon |
| 06:38.03 | Qwell | kaldemar: astricon is tame |
| 06:38.49 | kaldemar | Qwell: no passion? |
| 06:39.36 | brookshire | oh.. you know you want that |
| 06:39.57 | dlynes | dooed....wtf? where did all this gayness come from? |
| 06:40.04 | Qwell | brookshire: Kristian has a better story, but I sat next to crunch at dinner...ugh |
| 06:40.12 | Qwell | awful, I tell ya |
| 06:40.22 | brookshire | haha.. i heard! |
| 06:40.27 | watchy | if i goto astricon can i hang out with you guys |
| 06:40.38 | watchy | ill buy your alcohol |
| 06:40.53 | brookshire | oh.. we're there |
| 06:41.00 | harryvv | dlynes here you go |
| 06:41.00 | Qwell | brookshire: I think zoa was pissed... |
| 06:41.04 | harryvv | http://pastebin.ca/55016 |
| 06:41.04 | brookshire | at? |
| 06:41.10 | Qwell | brookshire: he was sitting on the other side of me... |
| 06:41.12 | watchy | are there any hot asterisk girls |
| 06:41.18 | Qwell | I left early that night, so he got stuck next to him. haha |
| 06:41.23 | brookshire | watchy: allison! |
| 06:41.30 | watchy | but is she hot |
| 06:41.34 | harryvv | Qwell my asterisk is not killing the call forwarding in the database...its stuck. |
| 06:41.45 | brookshire | yes! |
| 06:41.53 | brookshire | and she wears geek shirts! |
| 06:42.08 | Qwell | ooo, I have the best picture ever |
| 06:42.15 | watchy | sweet |
| 06:42.17 | brookshire | of allison? |
| 06:42.23 | Qwell | well, somebody does |
| 06:42.35 | dlynes | harryvv: You've got an error in your dialplan |
| 06:42.45 | dlynes | harryvv: near the "!=" |
| 06:42.52 | harryvv | the dialplan has never changed. |
| 06:43.03 | watchy | Could not fill input buffer |
| 06:43.03 | watchy | Tuning module 4Failure! |
| 06:43.07 | watchy | no line hooked to it? |
| 06:43.13 | dlynes | harryvv: Also, the database is working just fine...you're trying to remove an entry "201" from the database that's not in the database |
| 06:43.19 | brookshire | ugh.. file needs a new server |
| 06:43.28 | brookshire | his makes sineapps slooow |
| 06:43.35 | dlynes | harryvv: if the dialplan's never changed, it's always been buggy then |
| 06:43.52 | harryvv | this is a new issue as a few days ago out of one year its been working. |
| 06:44.08 | dlynes | harryvv: it might very well be working |
| 06:44.23 | dlynes | harryvv: but you might have some unintended side effects because of the bug in your dial plan |
| 06:44.46 | dlynes | harryvv: if you read the log file, you'll see it's telling you there's a syntax error in your dialplan near the '!=" |
| 06:44.54 | harryvv | there is no bug in the dialplan. it has worked all the time. I have not made any changes for probebly 6 months. |
| 06:44.58 | dlynes | # |
| 06:44.58 | dlynes | May 10 23:33:51 WARNING[1316]: ast_yyerror(): syntax error: syntax error; Input: |
| 06:44.58 | dlynes | # |
| 06:44.58 | dlynes | <PROTECTED> |
| 06:45.01 | dlynes | Then what's that? |
| 06:45.07 | harryvv | no idea |
| 06:45.26 | harryvv | Let me do a seach with vi of the dial plan for != to confirm your suspitions. |
| 06:46.57 | brookshire | http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album02 |
| 06:46.59 | *** join/#asterisk Tusker (n=tusker@203.117.94.152) |
| 06:47.02 | brookshire | astricon pictures! |
| 06:47.12 | harryvv | found only three examples in the extentions.conf with =! care to see them? |
| 06:47.26 | harryvv | != I mean |
| 06:47.28 | dlynes | harryvv: sure...pastebin them |
| 06:48.04 | kaldemar | comparing an empty variable gives that particular error. |
| 06:48.11 | harryvv | http://pastebin.ca/55017 |
| 06:48.33 | harryvv | Those have always been a part of the dial plan. This is asterisk@home and that is never messed with. |
| 06:49.49 | dlynes | Yeah...no idea what RGPREFIX is, but obviously it's empty |
| 06:50.15 | *** join/#asterisk bartlebee (n=largo@202.5.145.13) |
| 06:50.43 | dlynes | harryvv: but anyways...as i see it, you've only got one major problem |
| 06:51.09 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
| 06:51.10 | dlynes | harryvv: you've got a phone that's got call forwarding turned on on the phone itself...it's call forwarding to extension #201 |
| 06:51.27 | Tusker | I have a query... I have a DID with a locked Motorola VT1005, which I can connect to directly with a SIP uri... I currently have the FXO port on a SPA-3000 connected to the FXS port on the VT1005V, but the voice quality is quite bad... is there any way to connect to a native RTP stream on the VT1005 ? (does this even make sense?) |
| 06:52.04 | dlynes | harryvv: the phone in question that's doing the call forward has an ip address of 192.168.10.30 |
| 06:52.13 | dlynes | s/\.30/.40 |
| 06:52.38 | harryvv | ohh mmmm |
| 06:52.40 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 06:52.47 | dlynes | Tusker: what's a motorola vt1005? |
| 06:53.14 | Tusker | dlynes: it's an ATA with FXS lines... same device that vonage used to ship... |
| 06:53.32 | harryvv | cannot ping 30 |
| 06:53.43 | harryvv | thats probebly the ATA thats disconected at this moment. |
| 06:54.11 | harryvv | But the phone I am having problems with going to call forwarding is the ip500 at .50 address |
| 06:54.15 | Tusker | dlynes: I have the SIP proxy IP that the ATA communicates to... but I doubt I'll be able to use the auth channel with a different ATA |
| 06:54.30 | dlynes | harryvv: is the ip500 extension #201? |
| 06:56.26 | dlynes | Tusker: none of what you're doing makes any sense |
| 06:56.32 | viLeR | somebody knows if is possible to set the some field of the CDR ? |
| 06:56.40 | Tusker | dlynes: thought as much... |
| 06:56.47 | dlynes | Tusker: i don't know why you would want to hook up a sipura 3000 to an fxs port |
| 06:57.00 | harryvv | dlynes no thats the ATA |
| 06:57.10 | dlynes | Tusker: it sounds like you're using a sipura 3000 as an overpriced extension cord |
| 06:57.15 | harryvv | The extention 200 is the ip500 |
| 06:57.31 | Tusker | dlynes: well, the motorola is locked to the provider, and I don't have the SIP login/password |
| 06:57.47 | Tusker | I use the SPA3000 to connect to * |
| 06:58.02 | dlynes | harryvv: well, .30 and extension 201 are not the same device |
| 06:58.25 | dlynes | Tusker: why not have the motorola connect to asterisk directly? |
| 06:58.26 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:58.35 | dlynes | Tusker: or does the motorola not have a network port? |
| 06:58.47 | KeX_WorX | hi |
| 06:58.51 | Tusker | dlynes: yeah, it does have a network port... but I can't touch its config |
| 06:59.05 | KeX_WorX | is it possible to disable logging in asterisk ? |
| 06:59.12 | dlynes | Tusker: ah...so someone else installed it? |
| 06:59.15 | Qwell | KeX_WorX: logger.conf |
| 06:59.17 | Tusker | dlynes: I can only see the SIP traffic between it and the SIP proxy while sniffing |
| 06:59.26 | Tusker | dlynes: it's a device from the provider |
| 06:59.42 | harryvv | dlynes the ata with that address is currently disconected for painting that room. |
| 06:59.52 | KeX_WorX | Qwell, so it is possible to completely disable logging ? |
| 07:00.01 | Qwell | yes |
| 07:00.02 | dlynes | harryvv: yeah, but there's something else on that ip address, at the moment |
| 07:00.14 | harryvv | i ping .30 and its not replying |
| 07:00.25 | dlynes | harryvv: there's something replying on udp port 5060 at that address, though |
| 07:00.25 | KeX_WorX | k, thanks |
| 07:00.30 | dlynes | harryvv: it might be blocking pings |
| 07:01.03 | viLeR | it is possible to set some field of the CDR ? |
| 07:01.06 | Tusker | dlynes: the only way I can communicate to the motorola is via the PSTN number, the FXS port on it, or via the SIP URI of the device itself |
| 07:01.07 | harryvv | its a local net and it has never been a issue. its only 4 devises with ips. |
| 07:01.10 | dlynes | harryvv: May 10 23:33:51 VERBOSE[1316]: -- Got SIP response 302 "Moved Temporarily" back from 192.168.10.40 |
| 07:01.22 | harryvv | .40? |
| 07:01.24 | dlynes | yeah |
| 07:01.31 | harryvv | From 192.168.10.2 icmp_seq=44 Destination Host Unreacha |
| 07:01.32 | dlynes | i think that's what i said originally |
| 07:01.37 | harryvv | hold on |
| 07:01.40 | harryvv | thats wierd |
| 07:01.43 | dlynes | and then you kept saying 30, so then i started saying 30 i guess |
| 07:01.48 | *** part/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:01.58 | harryvv | this is off |
| 07:02.00 | harryvv | odd |
| 07:02.02 | harryvv | PING 192.168.10.30 (192.168.10.30) 56(84) bytes of data. |
| 07:02.02 | harryvv | From 192.168.10.2 icmp_seq=0 Destination Host Unreachable |
| 07:02.08 | dlynes | 40, not 30 |
| 07:02.30 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 07:02.39 | harryvv | PING 192.168.10.40 (192.168.10.40) 56(84) bytes of data. |
| 07:02.39 | harryvv | From 192.168.10.2 icmp_seq=0 Destination Host Unreachable |
| 07:02.51 | harryvv | why is it directing the pings to .2 |
| 07:03.00 | harryvv | .2 is the asterisk box. |
| 07:03.08 | dlynes | try nmap -v -sS -O -P0 192.168.10.40 |
| 07:03.22 | dlynes | You're doing the pinging from the asterisk box |
| 07:03.40 | dlynes | The asterisk box is reporting back to you that all the ping packets it's sending out to the other host are not returning |
| 07:03.53 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 07:04.00 | harryvv | nmap not installed |
| 07:04.07 | dlynes | harryvv: install it |
| 07:04.43 | harryvv | dont have to much experaince with centos mostly fedora |
| 07:04.59 | dlynes | harryvv: nmap, ethereal, iptraf, and tcpdump are all pretty useful for resolving voip and network issues |
| 07:05.11 | dlynes | fedora, centos, redhat...waht's the difference? |
| 07:05.21 | dlynes | aren't they all pretty much the same crappy base? |
| 07:05.22 | harryvv | apt get not installed on centos |
| 07:05.23 | harryvv | ;) |
| 07:05.35 | dlynes | aptget is debian, not fedora :) |
| 07:06.21 | harryvv | doing yum install |
| 07:06.24 | dlynes | Tusker: is there any particular way the voice quality is degrading? |
| 07:06.25 | Tusker | dlynes: I'm guessing there is no way to hijack the RTP stream ? |
| 07:06.45 | dlynes | Tusker: no idea...i've never tried doing anything like that |
| 07:07.09 | Tusker | dlynes: well, i'm guessing it's because it is going from RTP -> Analog -> RTP that it is quite bad... |
| 07:07.23 | Tusker | the volume is very low, and it is a bit choppy |
| 07:07.33 | dlynes | Tusker: are you using a timing device? |
| 07:07.40 | Tusker | dlynes: timing device ? |
| 07:07.49 | dlynes | Tusker: yeah...ztdummy, or something? |
| 07:08.09 | KeX_WorX | Qwell, do you know how to disable logging? cause I can't find it : / |
| 07:08.15 | Tusker | never tried... |
| 07:08.22 | Qwell | KeX_WorX: Just remove all the things from logger.conf |
| 07:08.24 | harryvv | doing a nmap scan |
| 07:09.17 | Tusker | dlynes: my config is asterisk <-> SPA-3000 <-> Motorola |
| 07:09.17 | harryvv | Starting nmap V. 3.00 ( www.insecure.org/nmap/ ) |
| 07:09.17 | harryvv | Host (192.168.10.40) appears to be up ... good. |
| 07:09.18 | Tusker | dlynes: will ztdummy still help ? |
| 07:09.22 | dlynes | Tusker: yeah...try loading ztdummy |
| 07:09.27 | dlynes | Tusker: then restart asterisk |
| 07:09.42 | dlynes | Tusker: It might help with the stuttering a bit |
| 07:09.44 | KeX_WorX | Qwell, oh, k.thx |
| 07:09.50 | Tusker | dlynes: ok, let me check it out |
| 07:09.50 | dlynes | Tusker: don't think it'll help with the low volume though |
| 07:09.59 | dlynes | Tusker: asterisk will use a timing device if it's available |
| 07:10.04 | Tusker | dlynes: well, I can make the SPA-3000 turn up the gain |
| 07:10.24 | dlynes | Tusker: also, the sipura 3000 has a jitter adjustment too |
| 07:10.46 | dlynes | Tusker: if you're using the old firmware, it's got low medium and high; if you're using the new firmware, it has 5 or 6 settings |
| 07:10.58 | Tusker | dlynes: yeah, i'm using 3.1.10... |
| 07:11.10 | dlynes | Tusker: i think there's a newer version than that |
| 07:11.59 | Tusker | dlynes: not according to sipura.com :) |
| 07:12.04 | dlynes | Tusker: oops |
| 07:12.12 | dlynes | was thinking the newest was 3.1.5 |
| 07:12.18 | dlynes | when did 3.1.10 come out? |
| 07:12.38 | dlynes | when i checked a couple weeks ago, 3.1.5 was the newest |
| 07:12.59 | Tusker | sometime in April |
| 07:13.14 | dlynes | Tusker: do they have a new website to download firmware from then? |
| 07:13.23 | dlynes | Tusker: i was looking on sipura.com |
| 07:13.34 | harryvv | dlynes whats next |
| 07:13.48 | dlynes | nvm |
| 07:13.48 | Tusker | http://www.sipura.com/support/index.htm |
| 07:13.52 | dlynes | yeah...i see |
| 07:13.59 | dlynes | weird that they'd jump 5 version numbers |
| 07:15.11 | harryvv | wife is complaining and i need to head off to bed. |
| 07:15.12 | dlynes | harryvv: try nmap -v -sU -O -P0 192.168.10.40 |
| 07:15.35 | harryvv | Starting nmap V. 3.00 ( www.insecure.org/nmap/ ) |
| 07:15.54 | harryvv | Host (192.168.10.40) appears to be up ... good. |
| 07:15.54 | harryvv | Initiating UDP Scan against (192.168.10.40) |
| 07:16.03 | dlynes | harryvv: it might take a while |
| 07:16.10 | dlynes | harryvv: possibly an hour or so |
| 07:16.22 | harryvv | thats fine |
| 07:16.23 | dlynes | harryvv: just paste me the results in a /msg |
| 07:16.55 | harryvv | ill get back to you tomarro. |
| 07:17.54 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:23.13 | *** join/#asterisk Cresl1n (n=matt@myskin.iet.unipi.it) |
| 07:23.58 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:24.39 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 07:26.28 | Tusker | dlynes: that might be a bit difficult... asterisk is running within a domu Xen VPS |
| 07:27.58 | litage | alrighty, i've read through several sections of the sip rfc that deal with URIs and whatnot. i understand that adding ";user=phone" to the end of a URI is valid (eg: sip:john@sip.domain.com;user=phone). however, i don't fully understand what the point of doing so is. is it just to pass additional info on to the other parties in the call? |
| 07:29.16 | dlynes | Tusker: ah...if it's in a vps, you might never get the sound quality to be good |
| 07:29.30 | dlynes | Tusker: the weirder your configuration, the less likely you're going to have something working |
| 07:30.59 | Tusker | dlynes: ok, understood |
| 07:38.44 | *** part/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 07:44.55 | *** join/#asterisk DarthClue_ (n=DarthClu@adsl-69-152-247-72.dsl.snantx.swbell.net) |
| 07:45.53 | harlequin516 | Anyone use virtualphoneline.com? |
| 07:49.20 | gandhijee | do i have to remove the res_config_odbc and res_odbc modules if i plan to use MySQL? |
| 07:49.41 | Tusker | harlequin516: how much does it cost ? |
| 07:51.17 | gandhijee | anyone have any idea? |
| 07:52.42 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 07:54.37 | Tusker | sorry, have to run, cya guys |
| 07:58.01 | dlynes | hrm |
| 07:58.11 | dlynes | who was it that was looking for recording gain for voicemail? |
| 07:58.46 | many | gandhijee: why dont you use odbc to adress the mysql DB? |
| 07:58.48 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 07:59.24 | many | gandhijee: install unixodbc, edit /etc/odbc.ini, configure res_*odbc to use the odbc dsn |
| 08:01.13 | dlynes | gandhijee: yes |
| 08:01.25 | dlynes | gandhijee: assuming you don't want to use odbc |
| 08:01.37 | many | i wouldnt bother running res_*mysql directly, as its not supported. |
| 08:01.43 | many | (but thats just MVHO.) |
| 08:01.43 | gandhijee | umm this is my first venture in to asterisk + a DB |
| 08:01.52 | dlynes | gandhijee: odbc is better...it's in regular development; mysql isn't kept in regular development |
| 08:01.57 | many | gandhijee: but you know what odbc is/means? |
| 08:02.01 | gandhijee | ok |
| 08:02.11 | gandhijee | open DB connection/client |
| 08:02.12 | gandhijee | ? |
| 08:02.24 | many | its a generic "wrapper" |
| 08:02.27 | dlynes | gandhijee: besides...everyone knows postgresql is better than mysql, anyways ;) |
| 08:03.07 | gandhijee | ok |
| 08:03.17 | many | gandhijee: asterisk addresses the generic wrapper "odbc", which in turn can use any database in the backend |
| 08:03.53 | gandhijee | many: the last time i did any DB stuff was 5 years ago, and it was w/ ColdFusion and wackass MSSQL |
| 08:04.00 | gandhijee | which we later moved to MySQL |
| 08:05.13 | many | gandhijee: well, under windows AFAIK (iam no windows expert at all), uses odbc for everything. it just doesnt matter what DB you have in the backend, it can even be flatfiles. you adress odbc and odbc wraps it up in calls which the backend db supports. |
| 08:05.16 | gandhijee | many: you have i a guide or something i can look at? |
| 08:05.40 | many | http://www.voip-info.org/wiki/view/Asterisk+res_config |
| 08:05.57 | gandhijee | and is there a place i can find a description of every module asterisk loads? |
| 08:06.13 | gandhijee | and what purpose they serve? |
| 08:06.38 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 08:07.24 | gandhijee | also can i use the odbc to store my CDR data? |
| 08:07.34 | many | gandhijee: or maybe better use this one: http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 08:08.04 | many | it uses pgsql, but you can easily adapt it |
| 08:08.13 | gandhijee | crap and i just spent all this time installed mysql & php |
| 08:08.16 | gandhijee | ok |
| 08:08.51 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:09.14 | gandhijee | many: you know a place i can go find out exactly what the purpose of each module is? |
| 08:10.08 | Agrajag- | gday. my musiconhold is very stuttery, i can't make out what it's play at all. i'm just using the sample musiconhold.conf, and the mpg123 that make mpg123 gave me. what could be the problem? |
| 08:10.16 | Agrajag- | err s/play/playing |
| 08:10.31 | gandhijee | like exactly what chan_agent does and wtf Agent Proxy Channel is/means |
| 08:10.33 | darkskiez | Agrajar: what about the Milliwatt() test? |
| 08:10.45 | darkskiez | gandhijee: agent.c |
| 08:11.11 | gandhijee | umm, that means i have to translate the code into english |
| 08:11.21 | gandhijee | its not something i am very apt at doing.... |
| 08:11.33 | gandhijee | hence me asking if there is a place to find that |
| 08:11.46 | Agrajag- | darkskiez: err give me a second to look that up, i have noi dea what that is |
| 08:12.05 | darkskiez | google, asterisk users mailing list, digium support, voip-info.org |
| 08:12.19 | Agrajag- | ta |
| 08:12.34 | darkskiez | Agrajag-: theres also an asterisk book |
| 08:13.04 | gandhijee | \\\ |
| 08:13.05 | gandhijee | <PROTECTED> |
| 08:13.05 | gandhijee | <PROTECTED> |
| 08:13.11 | gandhijee | mean anything to anyone? |
| 08:13.50 | darkskiez | gandhijee: yes |
| 08:14.04 | gandhijee | then can you enlighten me |
| 08:14.38 | Agrajag- | darkskiez: yeah that gives me a stuttered sound |
| 08:14.40 | darkskiez | gandhijee: if you need to know _exactly_ how something works, u look at the source code, otherwise look at voip-info.org, google, the mailing list, ask questions there, or specific questions here. |
| 08:14.50 | darkskiez | Agrajag-: then its not your music on hold, its packet loss on your network. |
| 08:15.03 | Agrajag- | hmm it's a direct lan connection to the box |
| 08:15.13 | gandhijee | well i just said i am not apt at translating source to english |
| 08:15.16 | darkskiez | Agrajag-: if its not that, then perhaps the machine is overloaded, bad codec perhaps |
| 08:15.29 | gandhijee | if i was i wouldn't be asking |
| 08:15.31 | darkskiez | gandhijee: well ask specific questions on the mailing list, or here. |
| 08:15.37 | Agrajag- | darkskiez: ekiga tells me it's GSM |
| 08:16.15 | gandhijee | well then exactly what does chan_agent do? |
| 08:16.36 | darkskiez | Agrajag-: could be a problem with ekiga / local machine, does it have a packet loss meter/network stats panel? |
| 08:16.54 | darkskiez | Agrajag-: if you're not losing packets then its probably an ekiga problem |
| 08:17.26 | Agrajag- | darkskiez: ok, thanks a lot, i'll look into that |
| 08:18.18 | darkskiez | gandhijee: its for telephone operators handling queues in a call center. |
| 08:19.25 | gandhijee | ok |
| 08:19.36 | gandhijee | so what does it do? |
| 08:20.42 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 08:23.44 | *** join/#asterisk Etop (n=Etop@80.72.33.130) |
| 08:23.49 | gandhijee | darkskiez: i went to voip-info, there is the same post there that i am asking here. for a description of what the modules do |
| 08:23.54 | Etop | hiya |
| 08:23.59 | gandhijee | i've actually been there for over a year |
| 08:24.16 | Etop | i've got a problem with my asterisk and g729... its too damn quiet... anybody knows the solution ?? |
| 08:25.13 | *** join/#asterisk HotaruT (i=massar@bratwurst.unix-ag.uni-kl.de) |
| 08:25.23 | darkskiez | gandhijee: if you send a call to an agent channel it will go to whatever device that agent has logged in under, thats what it does. |
| 08:25.43 | Etop | anyone... ?? |
| 08:25.48 | gandhijee | thats cool, but i'm sure you don't want to explain the other 131 modules |
| 08:26.05 | gandhijee | i know what some of them do, but not what everyone does, which is something that i do need to know |
| 08:26.14 | darkskiez | gandhijee: why? |
| 08:26.32 | gandhijee | so i know exactly what the heck i am removing |
| 08:26.51 | darkskiez | gandhijee: you making a cut down asterisk for embeedded devices or soemthing? |
| 08:26.53 | gandhijee | or it might have an interesting feature that i would like to use |
| 08:27.02 | gandhijee | darkskiez: yeah |
| 08:27.06 | gandhijee | you can say that |
| 08:27.36 | darkskiez | gandhijee: i'd go for the opposite approach, remove everything, and add it as you need it. |
| 08:28.07 | gandhijee | ok, but i'd still like to know exactly what everything does, it just the way i am |
| 08:28.31 | darkskiez | gandhijee: well if you are doing that sort of project, you really should be able to read the source code, thats simply the best way to find out what things do. |
| 08:28.45 | gandhijee | kinda like surgery, they don't just gut the dude and put stuff back in until he stops working |
| 08:29.09 | darkskiez | gandhijee: hahaha, it depends if you were trying to make a person fit in a small box or not. |
| 08:29.13 | gandhijee | darkskiez: i can read some source, but it doesn't mean i understand what everyones source does |
| 08:29.19 | darkskiez | gandhijee: otherwise thats a bad analogy. |
| 08:29.26 | gandhijee | specially if some of its not commented |
| 08:29.43 | darkskiez | file a commenting bug on the bug tracker. |
| 08:30.08 | gandhijee | i think its a moot point now, the project is 7 years along |
| 08:30.35 | Etop | guys guys guysssss can you help me with those damn codecs please? |
| 08:30.40 | gandhijee | it'd be easier if there was just a list of the modules purpose/function |
| 08:31.06 | *** join/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
| 08:32.28 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 08:32.49 | gandhijee | to give you an example, i am going through the pbx_config.c right now, there is a variable RESULT_SHOWUSAGE, i know what it means, but there is no were it says where it gets its value from. |
| 08:33.06 | *** join/#asterisk alphaque (n=alphaque@myskin.iet.unipi.it) |
| 08:33.36 | russellb | gandhijee: RESULT_SHOWUSAGE is used in CLI commands |
| 08:33.56 | russellb | it tells the core that the CLI command was not used with correct syntax, and to show the registered help text for the command to the user |
| 08:34.18 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:34.21 | gandhijee | i know the purpose. i was just giving an example |
| 08:34.37 | gandhijee | actually maybe you would know russellb |
| 08:34.50 | gandhijee | is there a place i can find a description of the purpose of each modules |
| 08:34.55 | gandhijee | i mean module |
| 08:35.09 | russellb | in the trunk? |
| 08:35.36 | darkskiez | Etop: how do you know its the codec? |
| 08:35.41 | russellb | every module has to register a short description ... they must implement a description function that returns it |
| 08:35.46 | gandhijee | like when i do show modules, i get a limited description |
| 08:36.03 | Etop | darkskiez it started to sound crappy since i changed it from g711 |
| 08:36.09 | gandhijee | i want something a little more detailed.... |
| 08:36.33 | russellb | well, the details wouldn't belong there |
| 08:36.41 | russellb | if the module implements an application, it's show application foo |
| 08:36.42 | Etop | i know it requires more cpu power etc... but why is it so quiet ?? |
| 08:36.48 | gandhijee | O |
| 08:37.04 | russellb | if it implements an API, such as ODBC connectivity, it's documentation in header files for programmers using it |
| 08:37.35 | gandhijee | is there a way to show the relations between the applications and the modules they depend on? |
| 08:37.38 | russellb | there really isn't more than the brief description you see that would be appropriate for the entire module |
| 08:37.59 | russellb | doxygen generates graphs for a lot of that .... http://www.asterisk.org/doxygen/ |
| 08:38.18 | gandhijee | sweet. |
| 08:38.26 | gandhijee | i think this is exactly what i was looking for |
| 08:39.04 | Etop | darkskiez... is it a problem with my asterisk and its codec or my cisco ???? |
| 08:39.20 | russellb | that's a bit out of date, i'll update it now though |
| 08:39.33 | Etop | its driving me nuts, everything seems to be fine on g711, but i have to switch to g729 |
| 08:39.36 | russellb | it's supposed to update nightly, but it appears to not be working :) |
| 08:39.39 | gandhijee | the doxygen dpcs? |
| 08:39.41 | gandhijee | lol |
| 08:40.01 | russellb | it's about ... 10000 revisions behind |
| 08:40.14 | russellb | and i'm not making that up :) |
| 08:40.23 | gandhijee | jeez |
| 08:40.25 | darkskiez | Etop: you'll have to contact digium support who you got the licences off, not many folk have g729 codecs here. |
| 08:40.35 | russellb | we're now at revision 26720 |
| 08:40.44 | gandhijee | woozers |
| 08:40.54 | gandhijee | wowzers |
| 08:41.11 | Etop | g729 doesnt seem to be licensed in where i am |
| 08:41.17 | Etop | what do you guys use then darkskiez ? |
| 08:41.35 | gandhijee | hey russellb, you guys been talking to any intel people lately? |
| 08:41.59 | russellb | i haven't been involved in that |
| 08:42.18 | darkskiez | Etop: I just use g711, thats why I cant really help. |
| 08:42.21 | gandhijee | so i'll take that as a yes?? |
| 08:42.26 | Etop | thanks anyway |
| 08:42.34 | dlynes | Etop: g729 is fine...there's no low voice quality issues with it that i've experienced |
| 08:42.41 | russellb | gandhijee: i have no idea what has been going on |
| 08:42.45 | gandhijee | O |
| 08:42.49 | gandhijee | got ya |
| 08:42.53 | Etop | dlynes... but what can i do to make it sound a bit ... LOUDER |
| 08:42.54 | darkskiez | Etop: if you are using the third party g729 codec, you have to talk to them. |
| 08:43.06 | dlynes | Etop: it's not the codec...it's something with your configuration |
| 08:43.12 | dlynes | Etop: ulaw and g729 are the same volume |
| 08:43.20 | Etop | alaw in my case |
| 08:43.22 | dlynes | Etop: neither one alters the volume |
| 08:43.28 | dlynes | alaw, ulaw...not much difference |
| 08:43.33 | dlynes | one's signed, one's unsigned |
| 08:43.35 | darkskiez | Etop: press the volume button on the phone. |
| 08:43.37 | Etop | so what can be wrong with my configuration |
| 08:43.43 | Etop | :) i tried that darkskiez |
| 08:43.57 | darkskiez | Etop: try a different phone firmware, I beleve there are volume differences in those. |
| 08:44.00 | dlynes | Etop: volume on your phone, gain control on your ata's, gain control on your zaptel channels, ... |
| 08:44.37 | dlynes | Etop: i'm just using the default codecs.conf |
| 08:44.37 | Etop | i use sip only |
| 08:45.09 | Etop | me too |
| 08:45.22 | darkskiez | talking to yourself? |
| 08:45.41 | Etop | nah |
| 08:45.43 | Etop | to just anyone |
| 08:45.52 | darkskiez | I like cheese |
| 08:45.53 | darkskiez | me too |
| 08:46.11 | Etop | me too |
| 08:46.16 | dlynes | darkskiez: those were two replies to two different statements...you would have noticed that if you were paying attention :) |
| 08:46.33 | darkskiez | dlynes: damn, i suck. coffee time. |
| 08:46.44 | gandhijee | what time is it where you guys are? |
| 08:46.51 | darkskiez | 9:45am |
| 08:46.51 | Etop | 10:47 am |
| 08:46.58 | dlynes | almost 2am |
| 08:46.58 | Etop | darkskiez uk ? |
| 08:47.01 | darkskiez | yarr |
| 08:47.05 | gandhijee | def not in the stats |
| 08:47.09 | Etop | dlynes east coast us ? |
| 08:47.15 | dlynes | west coast canada |
| 08:47.19 | gandhijee | no, he is west cost |
| 08:47.21 | Etop | my maths suck |
| 08:47.26 | gandhijee | east cost is almost 5 AM |
| 08:47.33 | gandhijee | which is where i am |
| 08:47.36 | Etop | my ex was from there |
| 08:47.46 | gandhijee | east coast? |
| 08:47.49 | Etop | uhhuh |
| 08:48.06 | russellb | gandhijee: docs are up to date now |
| 08:48.07 | gandhijee | west coast, or canada.... take your pick |
| 08:48.14 | dlynes | west coast is the only coast that matters :) |
| 08:48.16 | gandhijee | cool, thanks russell |
| 08:48.21 | russellb | you're welcome |
| 08:48.23 | *** join/#asterisk littlejohn (n=little@host219-78.pool8711.interbusiness.it) |
| 08:48.24 | gandhijee | yah, west cost is better |
| 08:48.25 | Etop | actually |
| 08:48.26 | azeteg | anyone here with experience setting up asterisk to talk to a h323 gatekeeper |
| 08:48.36 | Etop | europe is the only place in world that matters |
| 08:48.37 | azeteg | ? |
| 08:48.38 | Etop | A-Tuin |
| 08:48.41 | Etop | sorry |
| 08:48.42 | Etop | azeteg |
| 08:48.44 | dlynes | azeteg: i've got experience with swearing at pwlib, if that's any help :) |
| 08:48.45 | azeteg | yes? |
| 08:48.46 | Etop | i have a h323 gk |
| 08:48.50 | Etop | but its not asterisk based |
| 08:48.51 | azeteg | ah |
| 08:48.54 | azeteg | same here |
| 08:48.58 | azeteg | my isp has one |
| 08:49.03 | Etop | oh323 is crap |
| 08:49.21 | dlynes | Etop: what else is there, to connect to an h323 gatekeeper, using asterisk? |
| 08:49.28 | azeteg | my isp also has a messed up network, and a horrible pbx |
| 08:49.42 | Etop | i used yates dlynes |
| 08:49.45 | Etop | worked for some time |
| 08:49.46 | azeteg | I installed asterisk-oh323 channel module |
| 08:50.08 | dlynes | Etop: and yates has its own h323 stack that doesn't have umpteen million different dependencies? |
| 08:50.16 | azeteg | currently our voip traffic goes through our isp network |
| 08:50.21 | azeteg | no pbx here |
| 08:50.22 | Etop | dlynes... what can i say |
| 08:50.24 | Etop | it worked for me |
| 08:50.32 | azeteg | and they run h323 |
| 08:50.36 | Etop | i still have the gk but i dont use it anymore |
| 08:50.39 | dlynes | Etop: so yates doesn't use oh323 then, right? |
| 08:50.51 | azeteg | I setup asterisk with SIP here |
| 08:50.55 | Etop | to be honest, i have no idea what yates uses |
| 08:51.05 | Etop | maybe i messed something up while compiling it on my asterisk |
| 08:51.06 | azeteg | in an attempt to get cisco 7940/7960 talk to h323 GK |
| 08:51.16 | dlynes | ah...but it does connect to an h323 gatekeeper |
| 08:51.26 | Etop | yeh |
| 08:51.32 | Etop | it does but sounds awful |
| 08:51.38 | azeteg | hm? |
| 08:51.52 | Etop | azeteg what h323 gk do you use |
| 08:52.09 | azeteg | I don't know what they use |
| 08:52.24 | Etop | ok... i have gnugk and its based on openh323 |
| 08:52.46 | dlynes | ah...it uses oh323 and pwlib, also :) |
| 08:52.53 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
| 08:52.56 | azeteg | yeah |
| 08:52.59 | Etop | uhhuh |
| 08:53.03 | azeteg | this is major isp installation gk |
| 08:53.17 | dlynes | gk? garbage krap? |
| 08:53.33 | Etop | gk, good krap |
| 08:53.51 | Etop | somehow ive never had any problems with gnugk |
| 08:54.01 | dlynes | oh...gatekeeper |
| 08:54.04 | *** join/#asterisk Djeli (n=Djeli@203-158-46-73.dyn.iinet.net.au) |
| 08:54.07 | Etop | anyways... |
| 08:54.08 | azeteg | TeliaSonera here in Sweden |
| 08:54.21 | Etop | what codecs do you guys use ??? besides 711 and 729 |
| 08:54.30 | azeteg | g711a |
| 08:54.38 | azeteg | mp3 |
| 08:54.39 | dlynes | g711u/a, g726, g729, gsm, ilbc |
| 08:54.54 | azeteg | divx |
| 08:54.57 | Etop | :) |
| 08:55.08 | dlynes | mp3 and divx aren't codecs |
| 08:55.10 | Etop | divx for asterisk... not a bad idea :) |
| 08:55.12 | dlynes | they're file formats |
| 08:55.18 | Etop | lol |
| 08:55.23 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:55.26 | Djeli | umm... yeah. |
| 08:55.30 | azeteg | Etop, you know how to configure asterisk for use with h323 gk? |
| 08:55.30 | dlynes | well |
| 08:55.41 | Etop | azeteg, i tried |
| 08:55.42 | dlynes | mp3 is a codec, too...just a really bad one |
| 08:55.49 | Etop | but it didnt work good enough |
| 08:55.56 | Etop | i could log on to the other gk with my asterisk |
| 08:56.02 | Etop | but it would drop all my calls |
| 08:56.06 | azeteg | hmm |
| 08:56.25 | Djeli | I've done H.323 without a gatekeeper. |
| 08:58.14 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
| 08:59.29 | *** part/#asterisk Djeli (n=Djeli@203-158-46-73.dyn.iinet.net.au) |
| 09:00.40 | azeteg | why? |
| 09:00.53 | azeteg | its just a big protocol with inherent problems |
| 09:00.59 | Etop | h323... isdn |
| 09:01.03 | Etop | isdn...h323 |
| 09:01.11 | Etop | almost no difference |
| 09:01.16 | *** join/#asterisk Shark_y (n=paoloc@host181-54.pool8288.interbusiness.it) |
| 09:01.40 | azeteg | do all voip protocols embed source/dest ip addresses ? |
| 09:01.46 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:01.54 | Shark_y | Hi guys |
| 09:02.04 | CKGL | anyone knows of E1 channel banks reported to work with *? |
| 09:02.10 | azeteg | (like h323 does with h225/h245) |
| 09:02.22 | Shark_y | I'm starting with asterisk... can anybody suggest me a resource where to find how to configure isdn cards (HCF-PCI) with '*' |
| 09:02.51 | gandhijee | voip-info.org prolly has some stuff |
| 09:03.27 | dlynes | bleh...fscking hell |
| 09:03.34 | dlynes | everything for h323 seems to use oh323 |
| 09:04.03 | dlynes | Etop: same thing with pri |
| 09:04.15 | Etop | yeh |
| 09:04.26 | Etop | anyone of you ever used eicon pri cards ?? |
| 09:04.33 | [x86] | dlynes: h323 is yuck ;) |
| 09:04.40 | dlynes | [x86]: why? |
| 09:04.48 | Etop | i have a whole box of them and im trying to find a use for them |
| 09:04.57 | Etop | diva server pri something |
| 09:05.16 | dlynes | [x86]: it gives you much better error reporting than sip, it traverses firewalls better than sip...i can think of more reasons why h323 is good and sip is bad than i can find for why sip is good and h323 is bad |
| 09:05.39 | azeteg | traverses firewalls better: explain?P |
| 09:05.51 | Etop | h323 is more nat-friendly i guess |
| 09:06.02 | azeteg | I've just had a major session with h323 and binat |
| 09:06.15 | azeteg | made me decide h323+nat is horror |
| 09:06.19 | dlynes | azeteg: if you do port mappings for h323, h323 has no problems...if you do port mappings for sip, it just plain refuses to work properly |
| 09:06.21 | Etop | the biggest disadvantage of h323 is the price of endpoint gateways... |
| 09:06.57 | Etop | linksys pap2 is about $50 while patton smartnode with 2fxs is around $300 ? |
| 09:06.59 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:07.01 | [x86] | dlynes: what's wrong with IAX? |
| 09:07.12 | dlynes | [x86]: nothing...i didn't mention iax |
| 09:07.17 | azeteg | dlynes: nat with h323 was complete horror - works with some implementations of h323 protocol, but not all |
| 09:07.18 | Etop | [x86] almost no hardware ? |
| 09:07.27 | dlynes | [x86]: the discussion was comparing sip and h323, not iax |
| 09:07.29 | [x86] | Etop: IAXy ;) |
| 09:07.35 | Etop | :) |
| 09:07.45 | [x86] | dlynes: i just caught the tail-end, where you were bashing oh323 ;) |
| 09:08.00 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 09:08.08 | dlynes | [x86]: Why should i need to install openldap for something that i'm not even going to use ldap? |
| 09:08.09 | azeteg | have a cisco 7301 router that is supposed to handle the binat |
| 09:08.15 | azeteg | but it only works with cisco phones |
| 09:08.19 | [x86] | dlynes: eh? |
| 09:08.21 | *** join/#asterisk Raynor23 (n=chatzill@87.237.104.253) |
| 09:08.26 | dlynes | [x86]: or for that matter something that requires X libraries, when I don't have X libraries installed? |
| 09:08.34 | Etop | we should get back to using analog phones... |
| 09:08.39 | dlynes | [x86]: i'm never planning to run a gui on that machine |
| 09:08.39 | [x86] | dlynes: what are you talking about? oh323? |
| 09:08.51 | Etop | or transfer isdn over ip |
| 09:08.51 | dlynes | [x86]: all that crap is required by pwlib |
| 09:08.55 | Etop | :P |
| 09:08.56 | dlynes | [x86]: oh323 requires pwlib |
| 09:09.00 | [x86] | right |
| 09:09.20 | [x86] | h323 also supports video, which might require graphics libraries ;) |
| 09:09.26 | dlynes | it seems to me that oh323 is a security hole waiting to happen |
| 09:09.27 | Etop | it does |
| 09:09.33 | [x86] | oh323 didnt require openldap on my box though |
| 09:09.40 | [x86] | dlynes: probably ;) |
| 09:09.48 | dlynes | [x86]: did you download a binary, or source code? |
| 09:09.56 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:09.57 | [x86] | dlynes: source with patches (gentoo) |
| 09:10.06 | KeX_WorX | hi |
| 09:10.25 | dlynes | [x86]: yeah...maybe gentoo applied some patches to get rid of openldap and crap then |
| 09:10.27 | KeX_WorX | i'm trying to turn off logging into Master.cvs in asterisk, but withou any success |
| 09:10.37 | dlynes | [x86]: i'm getting some other errors trying to compile pwlib too |
| 09:10.37 | KeX_WorX | is it possible to turn that off? |
| 09:10.47 | dlynes | [x86]: it doesn't like my lack of updated C++ headers |
| 09:10.54 | KeX_WorX | and how? I've played around with logger.conf |
| 09:11.00 | dlynes | [x86]: for whatever reason it doesn't like strstream.h |
| 09:11.01 | [x86] | dlynes: in Gentoo there are whats called USE flags, where you can fine-tune the features you want stuff to support... i dont have ldap in there on my asterisk box |
| 09:11.29 | dlynes | KeX_WorX: you mean Master.csv? |
| 09:11.44 | KeX_WorX | jep. typo : / |
| 09:11.46 | dlynes | KeX_WorX: that's not logging...that's cdr |
| 09:11.52 | [x86] | dlynes: so it could be either one of the patches it is applying, or it could be a flag to the configure script of pwlib... not sure |
| 09:11.54 | KeX_WorX | ahhh, correct |
| 09:12.07 | KeX_WorX | so, is it possible to turn off cdr completely? |
| 09:12.08 | Shark_y | Do I require a sound card to run asterisk? It is mandatory? |
| 09:12.13 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.15 | [x86] | Shark_y: nope |
| 09:12.20 | [x86] | KeX_WorX: sure |
| 09:12.23 | dlynes | KeX_WorX: noload => app_cdr.so |
| 09:12.43 | [x86] | KeX_WorX: put dlynes's config line in /etc/asterisk/modules.conf |
| 09:12.44 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 09:12.50 | dlynes | Shark_y: only for paging, or softphone from the asterisk cli |
| 09:13.23 | KeX_WorX | dlynes, x86, or just set enable = no in cdr.conf |
| 09:13.34 | [x86] | dlynes: you mean that CLI dial command can actually use a soundcard and utilize the voice path? |
| 09:13.39 | [x86] | dlynes: i didnt know that ;) |
| 09:13.53 | [x86] | KeX_WorX: why load the module if you're not going to use it though? |
| 09:13.57 | dlynes | [x86]: yeah...using chan_alsa or chan_oss |
| 09:14.08 | [x86] | dlynes: that's cool :) |
| 09:14.09 | dlynes | [x86]: it's a pretty stupid feature if you ask me |
| 09:14.19 | Shark_y | but if I don't use a soft-phone on the machine that runs * I don't need to have this configured, true? |
| 09:14.20 | [x86] | kinda, but nifty for playing around with :P |
| 09:14.27 | dlynes | [x86]: i don't some clown running a softphone on my server :) |
| 09:14.30 | [x86] | dlynes: i just use x-lite on another box ;P |
| 09:14.35 | dlynes | s/some/want some |
| 09:14.46 | [x86] | dlynes: hehe... indeed |
| 09:14.54 | [x86] | Shark_y: correct |
| 09:15.03 | dlynes | Shark_y: chan_alsa/chan_oss gives you a softphone that's built into asterisk |
| 09:15.16 | dlynes | Shark_y: it's not a separate program |
| 09:15.55 | dlynes | Shark_y: anyways...the only reason you'd need a soundcard is for chan_alsa.so/chan_oss.so, or for app_page.so(?) |
| 09:16.33 | KeX_WorX | x86, you are right. I'll do it you way |
| 09:16.34 | dlynes | app_page might also use chan_alsa or chan_oss for all I know |
| 09:17.08 | dlynes | I've never actually tried using hte soundcard in asterisk yet |
| 09:17.23 | dlynes | I don't get customers that need a paging system very often |
| 09:17.38 | dlynes | last one we had was about 6 months ago |
| 09:17.44 | dlynes | last one before that was over a year ago |
| 09:18.23 | RoyK | anyone that knows a good BRI card for asterisk? i know junghanns had some, but I haven't heard a word from him in ages..... |
| 09:18.25 | [x86] | why does paging require a sound card? |
| 09:18.47 | dlynes | [x86]: to output through the line out or speaker out on the soundcard? |
| 09:18.50 | [x86] | RoyK: digium cards no good? |
| 09:18.59 | RoyK | [x86]: do they have BRI cards now? |
| 09:19.12 | dlynes | digium doesn't sell bri cards |
| 09:19.26 | key2 | could someone tell me why when I add a user to a queue(), it doesn't reinvite |
| 09:19.27 | [x86] | dlynes: oh... i thought by "page" you were talking about one phone talking to the speakerphone of all registered phones :P |
| 09:19.41 | dlynes | [x86]: nope...afaik, that's not even possible in asterisk |
| 09:19.44 | Shark_y | ok, I can bother on that later... what modules should I configure to start to place call considering that I have a hfc-pci card for Isdn and a atcom at320 phone? |
| 09:19.47 | dlynes | [x86]: no...like paging over a horn |
| 09:19.51 | [x86] | key2: do you have nat=yes for the queue member? |
| 09:19.56 | [x86] | dlynes: yeah i understand now |
| 09:20.42 | dlynes | Shark_y: probably chan_capi (for the isdn), and chan_sip (for the atcom at320 phone...i think it's a sip phone) |
| 09:20.54 | [x86] | key2: nat=yes or nat=always or nat=rfc... set reinvite=no |
| 09:21.10 | Shark_y | Yep it's IAX2 phone |
| 09:21.19 | dlynes | [x86]: all those values for nat? |
| 09:21.19 | [x86] | key2: since you can't reinvite behind a nat ;P |
| 09:21.25 | dlynes | Shark_y: then chan_iax2, not chan_sip |
| 09:21.42 | [x86] | dlynes: any of them will set reinvite=no, even if you explicitly say reinvite=yes |
| 09:21.44 | dlynes | [x86]: only ones i've ever used are nat=1/0 and nat=yes/no |
| 09:22.01 | [x86] | dlynes: welcome to asterisk 1.2-land ;) |
| 09:22.10 | dlynes | [x86]: what does nat=rfc do? |
| 09:22.24 | [x86] | there is a number too... forget what it is though... |
| 09:22.40 | dlynes | nat=8675309 |
| 09:24.23 | *** part/#asterisk Raynor23 (n=chatzill@87.237.104.253) |
| 09:24.23 | dlynes | Shark_y: It might be chan_misdn instead of chan_capi |
| 09:24.30 | dlynes | Shark_y: I've seen both used |
| 09:24.37 | dlynes | Shark_y: i don't know what hte difference is, personally |
| 09:25.04 | [x86] | dlynes: actually, asterisk 1.0 had yes/no/never/route for nat= options |
| 09:25.12 | key2 | x86: I'm on a local network |
| 09:25.19 | dlynes | ah...so what's route and rfc do? |
| 09:25.29 | [x86] | dlynes: where no means to use rfc3581 techniques |
| 09:25.50 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 09:26.01 | dlynes | ok, whatever that means :) |
| 09:26.15 | [x86] | dlynes: this article i'm getting this from (on voip-info) is somewhat outdated, and doesnt show the rfc or auto options |
| 09:26.38 | dlynes | yeah...most of the crap on voip-info is kinda outdated |
| 09:26.45 | dlynes | Which reminds me |
| 09:26.54 | *** join/#asterisk Raynor23 (n=chatzill@87.237.104.253) |
| 09:26.55 | Shark_y | what sounds to you this: "Unable to open IAX timing interface: Permission denied" I've sucessfully loaded ztdummy |
| 09:26.57 | dlynes | I need to finish writing my article for BLF/hinting |
| 09:27.11 | dlynes | Shark_y: are you running asterisk as root? |
| 09:27.19 | [x86] | dlynes: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+nat |
| 09:28.03 | *** part/#asterisk Raynor23 (n=chatzill@87.237.104.253) |
| 09:28.04 | Shark_y | I'm on a gentoo box and using the default /etc/init.d/asterisk script |
| 09:28.25 | dlynes | aaaandd....are you running asterisk as root, or a regular user? |
| 09:28.46 | dlynes | telling me you're running gentoo is not any help unless i'm actually running gentoo also |
| 09:28.52 | *** join/#asterisk parag7732 (n=root@de2-b1976.alshamil.net.ae) |
| 09:29.57 | Shark_y | it's running as "asterisk:asterisk" |
| 09:30.29 | russellb | dlynes: you should submit that article to asteriskdocs.org |
| 09:30.32 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 09:30.43 | russellb | not the wiki :) |
| 09:30.45 | dlynes | russellb: oh |
| 09:30.52 | russellb | well, you can of course |
| 09:31.01 | russellb | but I just like to encourage people to help that project ... |
| 09:31.24 | dlynes | russellb: ah...didn't know you could contribute to it...figured it was just some docs that some company put up there |
| 09:31.39 | russellb | nope, it's a group of community members |
| 09:32.07 | russellb | with a mailing list, even! asterisk-doc@lists.digium.com |
| 09:32.08 | dlynes | So there's more documentation there than just the slideshow for asterisk 1.2? |
| 09:32.16 | russellb | well, yeah |
| 09:32.25 | russellb | they are the ones that wrote the O'Reilly book |
| 09:32.28 | dlynes | I know about the mailing list, i'm even subscribed to it |
| 09:32.34 | russellb | but they are working on documentation in a new online format ... |
| 09:33.01 | dlynes | I've been archiving all the mailing lists for close to a year now |
| 09:33.05 | russellb | ~cool |
| 09:33.08 | jbot | from memory, cool is there, there, everything is going to fix out |
| 09:33.21 | russellb | stupid type ... and stupid bot |
| 09:33.26 | russellb | typo* :( |
| 09:33.26 | dlynes | heh |
| 09:33.42 | dlynes | and the stupid bot doesn't respond to s/regex/replace anymore, either |
| 09:34.42 | *** join/#asterisk eds0n (n=edson@c-3569e353.03-53-6f736c2.cust.bredband.no) |
| 09:34.44 | dlynes | s/stupid/smart |
| 09:34.50 | dlynes | see? |
| 09:34.51 | dlynes | heh |
| 09:34.55 | russellb | laaaame |
| 09:34.57 | Shark_y | dlynes: if I run "asterisk -vvvvvc" as root I don't get that message... on which device shuld I set permissions? |
| 09:35.18 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-37-dsl.end.estpak.ee) |
| 09:35.20 | dlynes | Shark_y: trying to remember...gimme a sec |
| 09:35.38 | eds0n | hey, could anyone give me ahand with astertest compilation? I seem to get error msgs about app_securax_serverload.c |
| 09:36.06 | dlynes | Shark_y: /dev/zap/ctl i believe |
| 09:36.06 | shiznatix | does anyone have any expierence with using ghost script to convert a multipage PDF to a tiff then faxing it? for some reason only the last page is being faxed right now |
| 09:36.40 | dlynes | shiznatix: how are you converting it? |
| 09:37.31 | *** join/#asterisk samourai1 (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 09:37.50 | Shark_y | eds0n: about zaptel compilation I warn anybody not to waste time running on kernels later than 2.6.15, I've lived on my skin |
| 09:38.19 | dlynes | Shark_y: 2.6.15.5 is fine |
| 09:39.44 | samourai1 | hi, plz i want to know if it's possible to record automaticly without pressing "*#" a call with Dial() function |
| 09:39.53 | eds0n | Shark_y: I'm using 2.6.16-ck |
| 09:40.01 | dlynes | russellb: btw...thought you told me to not use C++? then you're saying go for it...you're a very confusing individual :) |
| 09:40.07 | eds0n | I'm going to use a laptop as orignating server for testing purposes |
| 09:40.24 | russellb | dlynes: i was just kidding, heh ... i stand by my original statement |
| 09:42.04 | samourai1 | hi, plz i want to know if it's possible to record automaticly a call without pressing "*#" with Dial() function |
| 09:42.26 | Shark_y | eds0n: to downgrade to 2.6.15 as told by dlynes is my suggestion |
| 09:42.33 | russellb | samourai1: yes, you use the MixMonitor application before Dial |
| 09:42.39 | shiznatix | dlynes, gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -dNOPAUSE -sOutputFile='.$fileName.' -- /'.$file |
| 09:42.48 | shiznatix | dlynes, that is how i am converting my pdf |
| 09:43.22 | dlynes | shiznatix: how about pdf2ps? |
| 09:43.53 | dlynes | shiznatix: then do a gs on the ps? |
| 09:44.12 | Shark_y | eds0n: don't waste time with 2.6.16 kernels, it's too early because they have a different realtime clock handling |
| 09:44.34 | dlynes | Shark_y: how has it changed? |
| 09:45.36 | key2 | x86: so If am not natted |
| 09:45.48 | samourai1 | yes i tried but the problem is that when i saw the file recorded i didn't find the exact time |
| 09:45.55 | key2 | x86: if A calls B, it reinvite, but if A is queued and then got into B, it doesnt reinvite |
| 09:46.19 | Shark_y | dlynes: It's ok!! thanks I have now a 0666 on /dev/zap/ctl and it's running |
| 09:46.20 | parag7732 | May 11 13:44:54 DEBUG[1253] chan_sip.c: Oooh, format changed to 1 |
| 09:46.21 | parag7732 | May 11 13:44:57 DEBUG[2858] chan_sip.c: Auto destroying call '7369dab46c1037b901be047e703fc586@127.0.0.1' |
| 09:46.21 | parag7732 | May 11 13:45:03 DEBUG[2858] chan_sip.c: Acked pending invite 102 |
| 09:46.21 | parag7732 | May 11 13:45:03 DEBUG[2858] chan_sip.c: Stopping retransmission on '3b5dc0897a3359c43014aa2d40edb199@64.86.198.130' of Request 102: Match Found |
| 09:46.21 | parag7732 | May 11 13:45:03 WARNING[2858] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"1500" <sip:124984416@64.86.198.130>;tag=as7f9c2ccd' |
| 09:46.22 | samourai1 | therefore automon gives the exact time |
| 09:46.23 | parag7732 | May 11 13:45:03 VERBOSE[1253] logger.c: -- SIP/minifone-8e8d is circuit-busy |
| 09:47.01 | dlynes | It's pretty obvious what's wrong, parag |
| 09:47.05 | samourai1 | <PROTECTED> |
| 09:47.06 | dlynes | you've got the wrong password |
| 09:47.09 | parag7732 | noo |
| 09:47.24 | parag7732 | thats the problem...the same pin and password is working perfectly |
| 09:47.33 | parag7732 | in linksys pap2 |
| 09:47.52 | parag7732 | but when i m configureing a trunk into the asterisk |
| 09:48.05 | parag7732 | its giving the above |
| 09:48.07 | parag7732 | error |
| 09:49.38 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 09:49.41 | dlynes | have you retyped your password into the sipura unit to test your theory? |
| 09:49.52 | parag7732 | yaa |
| 09:50.06 | parag7732 | i know my password |
| 09:50.10 | parag7732 | and its working |
| 09:50.16 | parag7732 | in linksys pap2 |
| 09:50.32 | dlynes | parag7732: is the pap2 and your asterisk box both behind the firewall? |
| 09:50.38 | parag7732 | I have already tried with all the codec also |
| 09:50.47 | parag7732 | noo |
| 09:50.51 | dlynes | it's not a codec issue |
| 09:51.08 | dlynes | they're both fully exposed to the internet, then? |
| 09:51.12 | parag7732 | yaa |
| 09:51.14 | parag7732 | they are |
| 09:51.19 | parag7732 | there is no problem at all |
| 09:51.26 | parag7732 | my asterisk server is on DMZ |
| 09:52.41 | parag7732 | minifone 64.86.198.130 5060 OK (310 ms) |
| 09:52.50 | parag7732 | this is the status of sip show peers |
| 09:54.15 | parag7732 | what do you think dlynes |
| 09:54.17 | parag7732 | ?? |
| 09:55.06 | dlynes | parag7732: i would try sniffing the connection with ethereal if you can |
| 09:56.35 | parag7732 | okk |
| 09:56.43 | parag7732 | so what details exectly you want |
| 09:57.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 09:58.41 | dlynes | I don't...i'm just suggesting you use ethereal to see what's happening |
| 09:59.03 | dlynes | You should have a good idea after using ethereal as to what's happening |
| 09:59.19 | dlynes | if you don't, then do the same thing on your pap2, and compare notes with how asterisk was doing it |
| 09:59.33 | dlynes | see what the difference between the two devices was |
| 09:59.34 | dlynes | anyways |
| 09:59.36 | dlynes | i've gotta run |
| 09:59.38 | dlynes | need sleep |
| 10:01.29 | parag7732 | thanks dlynes |
| 10:02.19 | eds0n | Shark_y: I'm running 2.6.15 on the server, I still get the same error when trying to recompile asterisk with the new modules |
| 10:02.33 | *** join/#asterisk Tusker (n=tusker@203.117.94.152) |
| 10:03.30 | eds0n | the server is running ubuntu-server kernel 2.6.15-21-686 #1 SMP PREEMPT |
| 10:03.32 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 10:05.38 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 10:07.09 | Tusker | heya guys... I've got a strange problem with a particular sip provider (bbpglobal) I can receive calls no problem, though I can't seem to make calls through it... I've configured it as type=peer, but I always get "Forbidden - wrong password on authentication for INVITE"... any ideas ? |
| 10:13.37 | Jacke | sip debug shows what? |
| 10:14.03 | Shark_y | eds0n: couldn't help anymore... I've just experienced that it's too early to run on 2.6.16 kernels |
| 10:14.14 | Tusker | sorry to be a pain in the arse... I have "solved it" in that I have to set it as a peer, and then create a type=user for it |
| 10:14.48 | Faithful | my caller id is dropping the first digit incoming |
| 10:16.38 | drray | congrats Tusker |
| 10:18.00 | Tusker | cheers... is there a reason why adding the type=user fixes the problem? |
| 10:18.45 | eds0n | Shark_y: ok, I just posted a msg in the astertests forums.. hope someone can help |
| 10:19.03 | eds0n | Shark_y: although, it should work on the ubuntu server which is running 2.6.15 kernel :S |
| 10:19.33 | samourai1 | hi it's me again :), what is the asterisk cmd that return the current time plz |
| 10:20.55 | *** join/#asterisk adorah (n=FreePBX2@87.68.158.11.cable.012.net.il) |
| 10:21.11 | PoWeRKiLL | someone use ser with presence ? |
| 10:22.31 | mut | Tusker: use type=friend |
| 10:22.35 | mut | then you can make and recieve calls |
| 10:23.55 | Tusker | mut: I've tried that, and it doesn't seem to like it... it was set as peer previously... |
| 10:24.28 | Tusker | i mean, set as friend previously... |
| 10:24.38 | mut | hm |
| 10:24.42 | Tusker | incoming calls worked, but then outgoing had an invite problem |
| 10:24.58 | mut | so if you set it as a peer outgoing calls work? |
| 10:25.14 | Tusker | well, if I set as peer, as well as create a type=user for it |
| 10:26.02 | mut | weird |
| 10:26.07 | mut | friend should do both |
| 10:26.08 | Tusker | agreed :) |
| 10:26.14 | mut | ya sure you reloaded and all that fancy stuff? |
| 10:26.32 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 10:26.35 | Tusker | yah |
| 10:26.37 | Tusker | very sure |
| 10:27.01 | mut | very weird |
| 10:27.13 | Faithful | my caller id is dropping the first digit incoming, how do I fix that |
| 10:27.47 | mut | how many digits is the number total? |
| 10:28.00 | Faithful | 10 |
| 10:28.14 | mut | so in asterisk ya only see 9 digits? |
| 10:28.20 | mut | or on the caller id machine? |
| 10:28.22 | Faithful | yup |
| 10:28.42 | Faithful | asterisk is dropping the first digit |
| 10:28.45 | mut | ya sure the telco isn't messin with ya? |
| 10:28.50 | Faithful | nope |
| 10:28.54 | mut | cause |
| 10:29.05 | mut | actually |
| 10:29.06 | mut | hm |
| 10:29.23 | mut | how is the CID coming in? |
| 10:29.29 | mut | via sip or zap or iax? |
| 10:30.56 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 10:32.13 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 10:32.48 | azeteg | where do I setup outgoing trunks? |
| 10:34.47 | mut | w00t |
| 10:35.02 | mut | another 2 months or so and we'll officially be a CLEC |
| 10:35.02 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:35.06 | azeteg | I have an asterisk setup with SIP phones |
| 10:35.19 | azeteg | I have now isntalled asterisk-oh323 driver |
| 10:35.49 | azeteg | <PROTECTED> |
| 10:36.08 | azeteg | (say 00 prefix should go via my ISPs H323 gatekeeper) |
| 10:36.52 | mut | h323 is so 1930's |
| 10:37.21 | mut | check out the wiki tho |
| 10:37.21 | mut | http://www.voip-info.org/wiki/view/Asterisk+oh323+channels |
| 10:39.32 | azeteg | let me see |
| 10:41.22 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 10:42.51 | *** join/#asterisk Djeli (n=Djeli@203-158-46-73.dyn.iinet.net.au) |
| 10:43.19 | azeteg | so |
| 10:43.26 | azeteg | in extensions.conf |
| 10:43.34 | azeteg | I would setup extensions like... |
| 10:43.38 | *** part/#asterisk Djeli (n=Djeli@203-158-46-73.dyn.iinet.net.au) |
| 10:43.46 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 10:43.57 | azeteg | exten => WHAAAAAAAAAAAAAT |
| 10:44.06 | azeteg | to dial using the h323 gatekeeper? |
| 10:46.32 | RoyK | dial(H323/blah) |
| 10:46.36 | RoyK | rtfm :) |
| 10:47.30 | azeteg | yeah |
| 10:47.37 | azeteg | oh323/blah ;) |
| 10:47.42 | azeteg | but I mean the trunk setup |
| 10:48.39 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 10:48.45 | Dr-Linux | [root@LHR-PBX root]# asterisk -r |
| 10:48.46 | Dr-Linux | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 10:48.53 | Dr-Linux | my asterisk service is working fine |
| 10:48.56 | Dr-Linux | how can i get to CLI ? |
| 10:49.25 | zigman | does /var/run/asterisk.ctl exist? |
| 10:49.46 | RoyK | Dr-Linux: asterisk prolly sn't running |
| 10:49.48 | RoyK | isn't |
| 10:49.52 | RoyK | Dr-Linux: asterisk -gvvvvc |
| 10:51.02 | Dr-Linux | RoyK: then how i can make calls and take calls? |
| 10:51.16 | RoyK | read the docs |
| 10:51.24 | RoyK | ~docs |
| 10:51.28 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:52.06 | zigman | ~rtfm |
| 10:52.07 | jbot | rumour has it, rtfm is Read The F*cking Manual (TM) |
| 10:52.12 | RoyK | then, after reading through the documentation a few times, testing and learning, then ask if you need to know something specific |
| 10:52.14 | Dr-Linux | RoyK: i just wanna understand the logic, i know if i kill the service then, restart it will work fine |
| 10:52.19 | Dr-Linux | but i came to get it's logic |
| 10:52.34 | RoyK | it's all in the docs |
| 10:52.43 | Dr-Linux | RoyK: asterisk -gvvvvc << this command is also not working |
| 10:53.02 | RoyK | ~pb |
| 10:53.04 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:53.10 | Dr-Linux | May 11 16:01:43 WARNING[5906]: manager.c:1747 init_manager: Unable to bind socket: Address already in use |
| 10:53.40 | RoyK | killall asterisk ; sleep 3 ; killall -9 asterisk ; sleep 1 ; asterisk -gvvvvc |
| 10:53.41 | RoyK | :P |
| 10:53.57 | zigman | you forgot reboot ;) |
| 10:54.01 | zigman | and rm -rf /* |
| 10:56.01 | azeteg | hmm |
| 10:56.55 | RoyK | zigman: dd if=/dev/zero of=`mount | grep -w /| awk '{ print $1 }'` |
| 10:57.01 | azeteg | docs do not help me much |
| 10:57.19 | RoyK | azeteg: then rtfs :P |
| 10:59.10 | azeteg | I think its a bit unclear, howto configure oh323.conf |
| 10:59.18 | azeteg | especially the register part |
| 10:59.33 | SheriF_WorK | dose asterisk come with STUN server? |
| 11:01.52 | drray | no |
| 11:01.58 | qdk | azeteg: better to just take out the actual device. |
| 11:03.06 | azeteg | how do I set extensions.conf to route all calls with prefix 00 through this oh323 gatekeepeer? |
| 11:03.22 | Faithful | how do I route calls based on calleridname ? |
| 11:05.03 | kaldemar | Faithful: for example with GotoIf and ${CALLERIDNAME}. |
| 11:06.00 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 11:07.12 | SheriF_WorK | is there any Manula / howto to enable STUN in asterisk SIP server? |
| 11:07.24 | SheriF_WorK | i have some public STUNs server would like to enable one |
| 11:07.25 | azeteg | I get this response from asterisk oh323 debug: |
| 11:08.05 | azeteg | H.323 call '$iplocalhost/8011-99ca706e' cleared, reason 10 (Gatekeeper cleared call) |
| 11:08.37 | azeteg | Hungup 'OH323/031834171@212.181.217.129-99ca706e' |
| 11:08.48 | Faithful | kaldemar: we match extensions with s/callerid,1,Set(FROM_DID=s/callerid) but how to do it with calleridname? |
| 11:08.55 | azeteg | Everyone is busy/congested at this time (1:0/0/1) |
| 11:09.49 | hwt | is the new voicemailmain() in asterisk 1.4 CVS? |
| 11:12.15 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:16.23 | kaldemar | Faithful: GotoIf($[${CALLERIDNAME} = "jakethesnake"]?subscribers,jake,1:subscribers,operator,1) |
| 11:16.32 | kaldemar | Faithful: something like that for example. |
| 11:22.19 | *** join/#asterisk motu (n=motu@192.165.166.181) |
| 11:22.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:25.03 | RoyK | kaldemar: don't use space in there |
| 11:25.28 | RoyK | GotoIf($["${CALLERIDNAME}"="jakethesnake"]?subscribers,jake,1:subscribers,operator,1) |
| 11:25.31 | RoyK | there |
| 11:25.48 | RoyK | methinks |
| 11:26.10 | kaldemar | wellwell.. |
| 11:27.13 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 11:28.04 | kaldemar | RoyK: my 1.0.5 says parse error without the spaces... |
| 11:28.46 | kaldemar | so take your hose and stick it up the newer versions. :P |
| 11:28.48 | *** join/#asterisk lazy_et (n=root@82-45-0-54.cable.ubr08.azte.blueyonder.co.uk) |
| 11:31.18 | kaldemar | svn trunk seems to like it without the spaces too. |
| 11:33.35 | RoyK | kaldemar: use 1.2, not 1.0 :P |
| 11:33.59 | Dr-Linux | RoyK: what exactly this mean? |
| 11:34.00 | Dr-Linux | May 11 12:31:27 WARNING[22023] manager.c: Unable to bind socket: Address already in use |
| 11:35.42 | *** join/#asterisk YellowFruit (n=ms@195.182.4.66) |
| 11:36.46 | YellowFruit | Hello! Do the TDM400P and the TE110P /really/ have the same PCIIDs? And is there a way to use one of each in the same machine? |
| 11:37.20 | Etop | can someone explain me this : app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 11:37.36 | Etop | whats wrong with the dialplan ? |
| 11:39.45 | RoyK | Dr-Linux: it means something is already listening to whatever port asterisk is trying to bind to |
| 11:39.46 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 11:40.01 | RoyK | Dr-Linux: usually it means asterisk is running already |
| 11:41.07 | Agrajag- | does anyone know if any special settings ekiga users need when connecting to asterisk with sip? mine gets lots of stuttering with Milliwat() and any gsm files i play - other sip clients from the same box work fine though |
| 11:41.10 | parag7732 | how to off sip debug ip |
| 11:41.49 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 11:42.03 | parag7732 | how to disable sip debug ip |
| 11:44.19 | Faithful | This doc-n-talk is so noise it cannot receive dtmf |
| 11:44.31 | mut | OMG |
| 11:44.37 | mut | http://intellicast.com/IcastPage/LoadPage.aspx?loc=usmict22229&seg=LocalWeather&prodgrp=Forecasts&product=Forecast&prodnav=none |
| 11:44.53 | mut | 9 day flood |
| 11:45.40 | *** part/#asterisk Etop (n=Etop@80.72.33.130) |
| 11:46.16 | mut | http://intellicast.com/IcastPage/LoadPage.aspx?loc=usmict22229&seg=LocalWeather&prodgrp=RadarImagery&product=RadarLoop&prodnav=none |
| 11:46.20 | SheriF_WorK | i have problem with CF .. i use asterisk CVS-v1-0-03/29/05 i know it's old but i can't upgrade at the moment... where i can find a good examples for CF menus ? |
| 11:46.25 | mut | low right over top of michigan just circling like a vulture |
| 11:47.35 | becks | parag7732: sip no debug |
| 11:47.59 | parag7732 | thanks becks |
| 11:49.57 | parag7732 | becks, do you have any idea about this problem "wrong password on authentication for INVITE to '"1500" <sip:1500@192.168.1.70>;tag=as2dd2ac87'" |
| 11:50.12 | parag7732 | the same trunk is working perfectly with linksys pap2 |
| 11:56.27 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 12:02.46 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 12:05.11 | azeteg | darn |
| 12:05.22 | azeteg | I wonder how I can talk to my h323 gatekeeper |
| 12:08.19 | kaldemar | azeteg: if you're using gnugk, netcat port 7000 to the address it's listening on and say hello. saying help will be more useful, though. ;) |
| 12:11.37 | *** join/#asterisk ErSeba (n=Sebastia@ip-5-112.sn2.eutelia.it) |
| 12:12.27 | ErSeba | hi, is it possible to hide my own number when calling from asterisk? i tryied setting Outbound Caller ID in the trunk but i always see my own number |
| 12:13.06 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:13.56 | SheriF_WorK | i have problem with CF .. i use asterisk CVS-v1-0-03/29/05 i know it's old but i can't upgrade at the moment... where i can find a good examples for CF menus ? |
| 12:15.00 | *** join/#asterisk jahani (n=k@41.250.35.187) |
| 12:15.20 | drray | ErSeba what kind of telephone line do you have? |
| 12:15.48 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:15.51 | parag7732 | ErSeba , same thing also want to know |
| 12:16.02 | parag7732 | Drray, i m having Pstn |
| 12:16.04 | parag7732 | line |
| 12:16.23 | parag7732 | In call back i don't want to show my no. |
| 12:16.27 | ErSeba | with a SIP provider |
| 12:16.38 | ErSeba | should i ask my provider to hide th ID ? |
| 12:17.30 | azeteg | kaldemar: I'm connecting to my ISPs H323 gatekeeper |
| 12:20.35 | ErSeba | any hint? |
| 12:22.42 | *** join/#asterisk ptblank (n=MURDER1@68-169-176-165.lmdaca.adelphia.net) |
| 12:23.23 | Dr-Linux | any codecs expert around? |
| 12:27.37 | drray | ErSeba - I don't know that a PSTN provider will hide it for you |
| 12:27.50 | drray | a place like voipjet.com would let you set a caller ID |
| 12:28.05 | ErSeba | drray, it's not a pstn, is a SIP number |
| 12:28.27 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 12:28.57 | drray | ask your provider to change it then? tell him you want it set to your cell phone number and give him a bogus one |
| 12:30.04 | Dr-Linux | my all users works fine with any of codec, but one of my location, works only with iLBC codec, if i set anyother codec for them, then there is too much noise. what could be the reason? |
| 12:30.12 | SheriF_WorK | <PROTECTED> |
| 12:30.13 | SheriF_WorK | <PROTECTED> |
| 12:30.13 | SheriF_WorK | <PROTECTED> |
| 12:30.16 | ErSeba | i don't get why setting "Outbound Caller ID" is the same as leaving it empty :( |
| 12:30.20 | SheriF_WorK | but the call forwording still not working :( what i'm missing |
| 12:31.15 | [TK]D-Fender | SheriF_WorK : That only set a DB value. It offers no prrof that your system is properly set up to USE it for anything useful after... |
| 12:32.18 | [TK]D-Fender | SheriF_WorK : Pastebin your dailing macro/dialplan snippet and CLI showing its retreival and being discarded. |
| 12:32.24 | SheriF_WorK | [TK]D-Fender: sorry i didn't understand .. what should i check ? |
| 12:32.30 | SheriF_WorK | [TK]D-Fender: okay 1 min |
| 12:33.12 | [TK]D-Fender | ~pb |
| 12:33.20 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 12:34.30 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:34.40 | SheriF_WorK | [TK]D-Fender: ok i got it ! but i don't know how to fix wait 1 min :) i'm pasting what u need |
| 12:35.22 | SheriF_WorK | [TK]D-Fender: http://pastebin.com/711319 |
| 12:37.53 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 12:38.35 | MRH2 | hey anyone familiar with func_cdr |
| 12:38.57 | MRH2 | is it supposd to be limited to only account code and userfield? |
| 12:43.20 | [TK]D-Fender | SheriF_WorK : Ok, well you macro doesn't look up the DB value you set. What more do you want? |
| 12:44.45 | satlan32 | hi |
| 12:45.17 | satlan32 | is it possible to invite someone to a conference from within the conference room, without him knowing about it? |
| 12:46.53 | MRH2 | satlan 3 way call using the actual phone itself |
| 12:47.25 | SheriF_WorK | [TK]D-Fender: i'm trying to fix it |
| 12:47.28 | *** join/#asterisk apardo (n=apardo@87.217.145.14) |
| 12:48.22 | [TK]D-Fender | SheriF_WorK : Well there is nothing to fix right now, becaue ther is nothing in there trying to check for a forwarding value. |
| 12:49.28 | SheriF_WorK | [TK]D-Fender: yes i'm trying to add the DBget into the database |
| 12:49.57 | [TK]D-Fender | SheriF_WorK : Yes, and that seems to work just fine. Its that your are DOING ANYTHING WITH IT AFTER. |
| 12:50.18 | [TK]D-Fender | SheriF_WorK : And are you on 1.2.x? |
| 12:53.07 | SheriF_WorK | [TK]D-Fender: no 1.0 |
| 12:53.52 | [TK]D-Fender | SheriF_WorK : Really should update yourself... its been well over a year |
| 12:54.09 | [TK]D-Fender | If I'm not imstaken |
| 12:54.27 | *** join/#asterisk Becky75 (n=pirch@dsl-165-220-194.telkomadsl.co.za) |
| 12:54.32 | Becky75 | hey all. |
| 12:54.35 | SheriF_WorK | [TK]D-Fender: yes more than year :) but i didn't install it and i'm still not that good to understand the full dial plan.. so i cant upgrade now |
| 12:54.38 | [TK]D-Fender | SheriF_WorK : Either way when you've wroked a bit more at integrating your forwarding I'll be happy to take a look for you |
| 12:54.50 | Becky75 | how can i get rif of 1 to 2 seconds of dead air in a call on asterisk |
| 12:55.03 | Becky75 | it happens at random in the middle of a call |
| 12:55.05 | SheriF_WorK | [TK]D-Fender: yes working on it give me some mints ;-) |
| 12:55.35 | Becky75 | err rif = rid |
| 12:57.10 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 12:58.04 | Becky75 | how can i get rid of 1 to 2 seconds of dead air in a call on asterisk. it happens at random between calls but its long enough to let you lose what the person on the other side says |
| 12:59.01 | [TK]D-Fender | Becky75 : Perhaps you can eloborate as to what technologies and circunstances are related to both ends of this call..... |
| 12:59.19 | SheriF_WorK | [TK]D-Fender: thx :) |
| 12:59.26 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 12:59.58 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 12:59.58 | MikeJ[Laptop] | Becky75, sip? do the phones have VAD enabled? |
| 13:01.59 | *** join/#asterisk Cresl1n (n=matt@myskin.iet.unipi.it) |
| 13:02.04 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 13:03.45 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:04.56 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 13:05.33 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 13:06.59 | SheriF_WorK | [TK]D-Fender: u think it will work this way ? http://pastebin.com/711384 |
| 13:07.38 | Becky75 | [TK]D-Fender : asterisk 1.2.7 junghanns BRI card with BRIstuff drivers.. this only happens on out going calls |
| 13:07.46 | Becky75 | and in comming calls |
| 13:08.16 | Becky75 | MikeJ[Laptop] nope i have turned VAd off had some bad experiances with VAD in the past |
| 13:08.45 | Becky75 | between extentions the calls are fine |
| 13:08.50 | Becky75 | eben between 2 branches |
| 13:08.59 | Becky75 | eben = even |
| 13:11.21 | *** join/#asterisk assaf (n=Drake@192.118.54.150) |
| 13:12.18 | RoyK | ~seen zoa |
| 13:12.32 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 12d 20h 39m 28s ago, saying: 'you need to type it fast enough'. |
| 13:13.18 | Becky75 | ~seen a answer to my asterisk problem? |
| 13:13.20 | jbot | i haven't seen 'a answer to my asterisk problem', Becky75 |
| 13:14.20 | hwt | what sip.conf options do i need to be able to send the following string in the INVITE? |
| 13:14.23 | hwt | Remote-Party-ID: <sip:21674055@82.134.95.94;user=phone>;party=calling;screen=yes;privacy=full |
| 13:14.50 | hwt | i have enabled it using sendrpid. |
| 13:15.05 | hwt | but i can't find how to enable party=calling, screen=yes and privacy=full. |
| 13:15.58 | assaf | i heard there is a tool in asterisk@home that allwos the user to edit the *.conf files via a PHP interface , but i forgot the name of the tool ,can anyone help me in this ? |
| 13:17.31 | [TK]D-Fender | SheriF_WorK : Nowhere near working. You aren't retreiving the same FAMILY of values (CFIM vs CFBM), and if the value isn't set your dial = DEAD |
| 13:18.03 | [TK]D-Fender | SheriF_WorK : You need to first try to get the value. Then if it HAS a value use it INSTEAD, but use the requested # |
| 13:18.29 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 13:18.59 | Faithful | assaf yes ... Config EditMaintenance1.0toolEnabled |
| 13:19.10 | Faithful | assaf: under modules |
| 13:19.32 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:20.07 | Faithful | Doc-N-Talk are not so good folk... just be warned |
| 13:20.31 | Faithful | I cannot do dtmf into my * box with one. |
| 13:20.55 | SheriF_WorK | [TK]D-Fender: yes noticed and fixed this issue :D |
| 13:21.07 | SheriF_WorK | now i have wrong dial syntax |
| 13:21.09 | assaf | i tried to look in the sources and couldn't find it |
| 13:21.19 | Faithful | assaf: ... |
| 13:21.24 | SheriF_WorK | [TK]D-Fender: -- DBget: varname=temp, family=CFIM, key=108 |
| 13:21.24 | SheriF_WorK | <PROTECTED> |
| 13:21.24 | SheriF_WorK | <PROTECTED> |
| 13:21.24 | SheriF_WorK | May 11 16:28:16 WARNING[8618]: app_dial.c:698 dial_exec: Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout) |
| 13:21.31 | Faithful | are you running AMP ? |
| 13:21.39 | Faithful | or freePBX |
| 13:21.48 | SheriF_WorK | Faithful: join #freePBX |
| 13:21.56 | MikeJ[Laptop] | assaf, phpedit |
| 13:22.12 | MikeJ[Laptop] | it's out in asterisk addons too I think |
| 13:22.17 | Faithful | SheriF_WorK: chill he is asking the questions |
| 13:22.51 | assaf | the new A#home comes with freebpx |
| 13:22.57 | SheriF_WorK | [TK]D-Fender: done :D |
| 13:22.58 | SheriF_WorK | ;-) |
| 13:23.09 | MikeJ[Laptop] | assaf, did that answer your question? |
| 13:23.15 | Faithful | SheriF_WorK: I am already in freePBX |
| 13:23.43 | mut | my stress and job security went thru the roof yesterday |
| 13:23.48 | Faithful | assaf: join #freepbx |
| 13:24.08 | mut | we bought out telnetww's CO locations in mio, rose city and west branch, CLEC status soon to be ours! |
| 13:24.08 | Nugget | telnet is eeeeeeevil! |
| 13:24.15 | assaf | i am on freebpx |
| 13:24.35 | SplasPood | is there some trick to using labels within AEL macro definitions? |
| 13:27.17 | *** join/#asterisk shaZwaz (n=chatzill@203.81.196.166) |
| 13:27.47 | shaZwaz | hi |
| 13:28.34 | SplasPood | I simply do a :labelhere on a line by its own, but the parser seems to screw that up |
| 13:32.13 | [TK]D-Fender | SheriF_WorK : pastebin your final version |
| 13:33.10 | shaZwaz | anyone has call transfer problems with GS Budgetones with latest ver of * ? |
| 13:33.12 | SheriF_WorK | [TK]D-Fender: ok |
| 13:35.38 | SheriF_WorK | [TK]D-Fender: http://pastebin.com/711384 |
| 13:35.39 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 13:36.10 | RoyK | is there a way to restrict an iax2 peer to a given set of callerids? |
| 13:36.34 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 13:36.49 | mut | http://www.weather.com/outlook/health/allergies/local/49931?lswe=49931&lwsa=Weather36HourAllergiesCommand&from=whatwhere |
| 13:36.51 | mut | SNOW! |
| 13:36.56 | [TK]D-Fender | SheriF_WorK : You still have the CFIM / CFBS problem in there... |
| 13:37.07 | SheriF_WorK | [TK]D-Fender: but the voice maiul still not working now. |
| 13:37.22 | SheriF_WorK | [TK]D-Fender: why i want it when CFIM only and i tried it's working .. |
| 13:37.41 | shaZwaz | RoyK: you can change the CLid on calls |
| 13:38.02 | [TK]D-Fender | SheriF_WorK : You forgot to account for 2 sets of priority jumping.... |
| 13:38.07 | RoyK | shaZwaz: i'm aware of that, but I want to restrict a customer to - say - 50 CLIDs |
| 13:38.21 | RoyK | mut: lol. the snow's even gone up here, oslo, norway |
| 13:38.40 | shaZwaz | well you can make a check of his CLIDs |
| 13:38.55 | mut | how do ya staple 81 pages together? |
| 13:39.01 | [TK]D-Fender | RoyK : Sure, make sure every call they make validates their user vs CID agains a database before processing |
| 13:39.01 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:39.04 | mut | need a hole puncher and binder probly eh? |
| 13:39.04 | RoyK | shaZwaz: dozens of GotoIf? |
| 13:39.27 | RoyK | [TK]D-Fender: hm. with AGI, yes..... :) |
| 13:39.31 | [TK]D-Fender | mut : I've got a jumbo stapler for things like that |
| 13:39.36 | shaZwaz | or one AGI |
| 13:40.20 | [TK]D-Fender | RoyK : So make your DB key by the tech part of the channel name. |
| 13:41.01 | RoyK | [TK]D-Fender: yep. guess i'll be using mysql here as well. should work |
| 13:41.24 | shaZwaz | RoyK: there must be something already out there ....a little googling will help you out |
| 13:42.23 | shaZwaz | somthing similar to call screeing....AGI |
| 13:42.30 | [TK]D-Fender | RoyK : If you don't already use it for something else I'd suggest SQLite and ODBC at most for this |
| 13:43.07 | [TK]D-Fender | shaZwaz : No, what he'd want would be too custom, and not at all hard to do for anyone with minor programming skills |
| 13:43.51 | SheriF_WorK | [TK]D-Fender: how come ? |
| 13:43.52 | SheriF_WorK | [TK]D-Fender:ahhh wait |
| 13:43.56 | azeteg | someone who cares to help me with a little oh323 problem? |
| 13:43.58 | SheriF_WorK | [TK]D-Fender: i forgot the put ,20 |
| 13:44.16 | [TK]D-Fender | SheriF_WorK : There are a few ways to do it. |
| 13:44.46 | shaZwaz | RoyK: how many peers are you talking about here ? |
| 13:44.56 | [x86] | SheriF_WorK: :P |
| 13:44.59 | SheriF_WorK | [TK]D-Fender: as long as it works for the moment it's okay :) |
| 13:45.08 | SheriF_WorK | [x86]: i'm not weak :P or for the moment i done it and i'm very happy |
| 13:45.15 | [TK]D-Fender | SheriF_WorK : Question. If ext 100 wants to forward to 200, who's VM should it fall into? :) since you are really calling 100, should it only count as RINGING 200, but still VM in 100? |
| 13:45.19 | SheriF_WorK | [x86]: http://pastebin.com/711319 |
| 13:45.26 | shaZwaz | I think Ast DB too will do a great job ...if you have a managable number |
| 13:45.33 | [TK]D-Fender | SheriF_WorK : It only PARTIALLY works. If you are on the phone it will BREAK. |
| 13:45.39 | SheriF_WorK | [TK]D-Fender: now my brain hurts |
| 13:45.49 | [TK]D-Fender | SUCCESS! |
| 13:46.00 | SheriF_WorK | [TK]D-Fender: why will brake ? |
| 13:46.12 | [x86] | SheriF_WorK: this is the same thing you just showed me? |
| 13:46.23 | SheriF_WorK | [x86]: nop this one is working :P |
| 13:46.29 | SheriF_WorK | [x86]: i'm using same link |
| 13:46.38 | shaZwaz | my GS BT doesn't recognize the # key for call transfer |
| 13:47.04 | [x86] | SheriF_WorK: you must have given me the wrong link.... it's still not showing a DBGet anywhere :P |
| 13:47.22 | [x86] | shaZwaz: log into the web interface and turn call features off |
| 13:47.29 | *** join/#asterisk ToyMan (n=stuq@74-32-62-42.dsl1.mdl.ny.frontiernet.net) |
| 13:47.50 | SheriF_WorK | [x86]: http://pastebin.com/711416 <-- check this refresh :P |
| 13:48.10 | [TK]D-Fender | SheriF_WorK : If "s,102" is busy it will jump to "s,203" for VM which doesn't exist :) |
| 13:48.13 | [x86] | SheriF_WorK: much better ;) |
| 13:48.51 | [TK]D-Fender | SheriF_WorK : MUCH SMARTER! I'm impressed you got that idea on your own :) |
| 13:49.08 | [x86] | [TK]D-Fender: watch out, he's running 1.0 and refuses to upgrade ;) |
| 13:49.38 | [TK]D-Fender | SheriF_WorK : HOWEVER <- remove the "20" timeout from local.... Let the macro do what it has to or it might break later :) |
| 13:49.45 | shaZwaz | x86: thats already done |
| 13:49.56 | [TK]D-Fender | [x86] : I am well aware |
| 13:50.29 | [x86] | [TK]D-Fender: i told him I'd give him my MySQL-base CFIM/CFBS AGI scripts... but he wants to use the clunky astdb instead too heh |
| 13:50.40 | [x86] | s/-base/-based/; |
| 13:50.56 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 13:50.57 | [TK]D-Fender | [x86] : ASTDB is just great for this, no need for that amount of overkill. |
| 13:51.16 | [x86] | [TK]D-Fender: well my setup is a little more complex too... |
| 13:51.32 | [TK]D-Fender | [x86] : You'd freak if you saw what I built for ASTDB ;) |
| 13:51.48 | [x86] | [TK]D-Fender: for example, i let users store as many numbers as they like for CFBS and CFIM, and mark one active or inactive whenever they way |
| 13:51.50 | [x86] | want* |
| 13:52.42 | [x86] | so if they have a bunch of numbers they forward to commonly, they dont have to type them in each time :P |
| 13:53.03 | [TK]D-Fender | [x86] : I did mine in ALL ASTDB. I have like 10 values / peer :) and a massive IVR structure to let them choose what kind of forwarding to do (immed/busy/unavail) and where to send it (VM, intern, external, none) |
| 13:53.05 | [x86] | and they can put descriptions on each one |
| 13:53.14 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:53.29 | [x86] | nice ;) |
| 13:53.40 | [TK]D-Fender | [x86] : I had 300 lines of dialplan to support mine... I may forward it to you some time |
| 13:53.51 | [TK]D-Fender | SheriF_WorK : Get fixing your macro! |
| 13:53.52 | [x86] | i didnt think astdb supported more than a simple key=value pair.. |
| 13:54.10 | [TK]D-Fender | SheriF_WorK : You're almost there, then I'll help you with a last little idea on it. |
| 13:54.38 | [TK]D-Fender | x86 : it doesn't, ita all about careful CONSTRUCTION of them to ensure their validity, and sanity for retreival and management |
| 13:55.05 | [x86] | [TK]D-Fender: both of my AGI scripts, that section of my dialplan, AND my database schema dont even come close to 300 lines heh |
| 13:55.16 | SplasPood | anyone have a working AEL example containing labels/goto within a macro? |
| 13:55.38 | SheriF_WorK | [TK]D-Fender: i didn't get it what is wrong with my Macro :D? |
| 13:55.43 | [TK]D-Fender | [x86] : here : http://pastebin.com/711444 |
| 13:56.06 | SheriF_WorK | hehe [TK]D-Fender [x86] to asterisk gurus saying i'm smart :P mawhahahhahahhahhahahahh |
| 13:56.10 | [TK]D-Fender | SheriF_WorK : if the number isn't forwarded and is busy it will jump to 203 which doesn't exist and you won't be able to leave a VM! |
| 13:56.29 | SheriF_WorK | [TK]D-Fender: hum ok 1 min |
| 13:56.57 | shaZwaz | Q. will asterisk run on this ? http://www.storlinksemi.com/support-sl3316j.htm |
| 13:57.10 | [x86] | [TK]D-Fender: yeah... 346 lines just in dialplan... mine is way more efficient ;) |
| 13:57.36 | SheriF_WorK | [TK]D-Fender: exten => s.203,Voicemail(b${ARG1}) <-- thats it ? |
| 13:58.08 | [TK]D-Fender | SheriF_WorK : Think on it for a few minutes, and present me what you think will do the job. |
| 13:58.34 | [TK]D-Fender | x86 : Yeah, with tons of added app, security and a MySQL config file in exchange ;) |
| 13:58.36 | SheriF_WorK | [TK]D-Fender: i think this will fix it .. but i'm thinking but i have to leave work in 10 min.. max |
| 13:58.36 | [x86] | SheriF_WorK: we expect a full 10 page report on your findings... within the next 10 minutes! |
| 13:58.42 | [TK]D-Fender | x86 : not to mention LOAD :) |
| 13:58.49 | SheriF_WorK | [x86]: kiss my A$$ :P |
| 13:59.22 | [x86] | [TK]D-Fender: ALL of my code is well under 150 lines |
| 13:59.32 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:59.36 | [x86] | [TK]D-Fender: including 2 AGI scripts, dialplan, and mysql schema ;) |
| 13:59.50 | [TK]D-Fender | SheriF_WorK : Also given the number of parameters I'm setting I'm not sure it could shortened really. |
| 13:59.57 | [TK]D-Fender | err [x86] rather |
| 14:00.08 | [x86] | i handle all of this as well ;) |
| 14:00.25 | [x86] | well, except internal forwarding... |
| 14:00.30 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.98.Dial1.SanJose1.Level3.net) |
| 14:00.43 | [x86] | cfim, cfbs, cfim->voicemail I handle |
| 14:00.44 | [TK]D-Fender | [x86] : I doub't it takes less lines, but could be more flexible for the difference and definately more scalable |
| 14:00.53 | SheriF_WorK | [TK]D-Fender: http://pastebin.com/711434 thats all i have in mind now :D |
| 14:01.44 | [TK]D-Fender | SheriF_WorK : Your 102 dial doesn't account for BUSY |
| 14:02.21 | [TK]D-Fender | SheriF_WorK : and your 2 Dial call 103 with an unavailable message (cheap hack, but uninformative) |
| 14:02.26 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-188.z143-154-67.customer.algx.net) |
| 14:02.36 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.98.Dial1.SanJose1.Level3.net) |
| 14:02.37 | SheriF_WorK | [TK]D-Fender: hummmmmmmm |
| 14:03.11 | SheriF_WorK | [TK]D-Fender: i can't think of it now i have to go now i'll check it on sunday :-s |
| 14:03.19 | SheriF_WorK | see u later guyes from home.. |
| 14:03.20 | [TK]D-Fender | SheriF_WorK : Keep an eye out for me |
| 14:03.29 | SheriF_WorK | [TK]D-Fender: [x86] : thx alot guys ;-) |
| 14:03.58 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 14:04.01 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 14:05.30 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:05.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:05.56 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 14:06.55 | Shark_y | a little help with an hcf-pci card please, anybody? |
| 14:08.17 | RoyK | hfc-pci? |
| 14:08.23 | Shark_y | span=1,1,3,ccs,ami |
| 14:08.23 | Shark_y | bchan=1-2 |
| 14:08.24 | Shark_y | dchan=3 |
| 14:08.24 | Shark_y | yep |
| 14:08.28 | RoyK | use either visdn or bristuff |
| 14:08.43 | RoyK | ~pb |
| 14:08.44 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:09.23 | Shark_y | ztcfg -vvvv |
| 14:09.37 | Shark_y | gives out: |
| 14:09.51 | Shark_y | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 14:10.30 | RoyK | Shark_y: bristuff? |
| 14:10.32 | Shark_y | should I use isdn4linux? |
| 14:10.39 | RoyK | Shark_y: NO YOU SHOULD NOT |
| 14:11.06 | coppice | why? he might be a masochist? |
| 14:11.15 | RoyK | indeed... |
| 14:11.26 | Shark_y | recompile kernel with no isdn4linux support? |
| 14:11.43 | RoyK | might be a good start |
| 14:12.30 | Shark_y | which module will be my heelper? |
| 14:13.20 | Shark_y | capi 2.0 will be even helpful? |
| 14:14.12 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 14:14.12 | *** part/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:14.19 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:14.44 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 14:15.14 | Shark_y | coppice: I'm gentooist... ehehe not masochist!! ;) ;) |
| 14:15.41 | Blackthorn | Hello, when calling from spa-2000 through * to my local pri I get a ring then halfway through the second reing you here it ringing even louder. Basicly double or echoed rings. Do you know how to change this? |
| 14:15.55 | RoyK | I've heard something about using capi with hfc-pci, but have no idea how that's supposed to work. |
| 14:15.59 | Nivex | "Ubuntu: Ancient African word for ''I'm sick of compiling Gentoo''" -- Jeff Waugh |
| 14:16.01 | RoyK | Shark_y: use bristuff or rvisdn |
| 14:16.07 | *** join/#asterisk marlowe (n=mmarlowe@pdpc/supporter/active/marlowe) |
| 14:16.14 | RoyK | Nivex: lol |
| 14:16.55 | coppice | Shark_y: can you explain the difference? its too subtle for me to see |
| 14:17.00 | *** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca) |
| 14:17.15 | Shark_y | Nivex: distcc is the word!!! |
| 14:17.48 | DeeJay[2] | Is it possible with linksys pap2 to auto dial a number... |
| 14:17.50 | Shark_y | I can agree in some way.... ;) |
| 14:18.07 | DeeJay[2] | We have no dial pad phones in elevators and would like to get in a context as soon as the phone hang up.. |
| 14:18.10 | DeeJay[2] | since it cannot dial anything.. |
| 14:18.13 | DeeJay[2] | it is connected to a pap2 |
| 14:18.28 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 14:18.30 | Katty | hi lads. |
| 14:18.51 | Hmmhesays | Katty |
| 14:18.56 | Hmmhesays | you check out that software? |
| 14:19.02 | Katty | not yet. |
| 14:19.11 | Katty | was pretty busy last night |
| 14:19.23 | Katty | i might have time next week :P |
| 14:20.30 | Shark_y | but a gentoo box is like a girl: nobody has gone so deep in it like you and nobody has done something to her for you, although it takes long time to get her |
| 14:20.41 | Hmmhesays | I got busy last night too |
| 14:20.50 | Shark_y | lol |
| 14:20.53 | Nivex | Shark_y: I like having a Linux box ready to go in under 30 minutes. |
| 14:21.09 | azeteg | when I trunk to h323 gatekeeper, I send my calling number twise as id, I don't see why it sends it twice - like NUMBERe:NUMBER |
| 14:21.21 | azeteg | where can I modify this? |
| 14:21.23 | scanna | hi |
| 14:21.30 | Katty | Hmmhesays: yeah? |
| 14:21.38 | *** join/#asterisk Rick_Hunter (n=rhunter@10-134.008.popsite.net) |
| 14:21.45 | Hmmhesays | Katty: indeed |
| 14:21.49 | scanna | can someone explain me what is a bridged call? |
| 14:22.10 | wasim | scanna: its when you carry string across an open span of water |
| 14:23.22 | scanna | mmm |
| 14:23.23 | jake1932 | scanna: asterisk refers to calls as bridged when two endpoints are "bridged" together |
| 14:24.17 | coppice | wasim: are there flyovered calls, or are they all bundled under the term bridged? |
| 14:24.42 | scanna | jake1932: but if I call from a local phone through my carrier the call is always bridged? |
| 14:24.58 | Shark_y | Nivex: I don't disagree for a workstation and forr that I like ubuntu... but you don't know how the dual xeon 64bit compiled performs with gentoo bootstrapped from scratch |
| 14:25.39 | jake1932 | scanna: from a local phone? |
| 14:25.43 | scanna | I am trying to understand the difference between echocancel and echocancelwhenbridged for hardware configuration |
| 14:25.56 | scanna | jake: a local extension of my pbx |
| 14:26.21 | Shark_y | should leave, nice talking to you guys! THX!!! |
| 14:27.31 | shaZwaz | see ya later |
| 14:28.00 | *** join/#asterisk MikeB (n=chatzill@175.54.187.81.in-addr.arpa) |
| 14:28.00 | Hmmhesays | google is like a drug |
| 14:28.55 | MikeB | Hi All, Have interesting problem to give you somthing to think about |
| 14:28.56 | jake1932 | scanna: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf |
| 14:29.03 | ManxPower | scanna, normally you don't need echocancel when both legs of the call are Zap aka TDM, since you won't hear the echo because of the very low latency of that configuration |
| 14:30.03 | MikeB | Connecting an asterisk box between an Lucent Index and a BT PRI error "cant find extension '' in context isdn2. cant trap with => s,1 or t,1 or i,1 or _.,1 Any ideas? |
| 14:30.37 | MikeB | Yes I know that it should be sending a called number but it is not |
| 14:30.42 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 14:30.52 | azeteg | How do I setup an extensions.conf rule that takes outbound phone number from the phone, like: exten => _SOMETHING,1,Dial(OH323/-whattoputhere-@gatekeeper) |
| 14:31.48 | MikeB | exten => _X.,1,dial(OH323/${EXTEN}.... |
| 14:32.05 | MikeB | stupid emoticon = > |
| 14:33.27 | azeteg | thanks |
| 14:33.30 | [TK]D-Fender | MikeB : not for us it wasn't. |
| 14:33.42 | wasim | coppice: same, 'cept for tunneled calls |
| 14:33.59 | MikeB | [TK]D-Fender: Weird, Chatzilla. |
| 14:34.17 | coppice | wasim: don't tunneled calls requiring training? |
| 14:34.20 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:35.20 | wasim | coppice: only if they haven't been yet |
| 14:36.12 | MikeB | So anyone help me then, I have been trying this all day, works fine between an asterisk server and a BT PRI but not between an Lucent Index and BT PRI, the Lucent seems to be trying to dial a blank extension number which asterisk does not like and does not match to exten = s,1 |
| 14:36.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:36.47 | [TK]D-Fender | MikeB : Think you may be DOA on that.. |
| 14:37.21 | MikeB | DOA ? |
| 14:37.59 | [TK]D-Fender | MikeB : Dead On Arrival. PRI expects a DID and I'm guessing can't deal with not having one |
| 14:38.12 | [TK]D-Fender | MikeB : Chan_zap anyways.... |
| 14:38.24 | MikeB | It looks like the BT Line is happy to accept the call with no called number then they provide a dialtone and the digits get sent inband. |
| 14:38.25 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:38.51 | *** join/#asterisk kimosabe (n=kimosabe@dsl-200-78-71-61.prod-infinitum.com.mx) |
| 14:39.03 | kimosabe | can some one help me get x-lite working |
| 14:39.06 | azeteg | ok - I finally got asterisk to work with the h323 gatekeeper |
| 14:39.12 | azeteg | now I have another problem - |
| 14:39.20 | kimosabe | it says registered but when i dial it says call not aproved |
| 14:39.25 | azeteg | audio only works from asterisk to h323 gk |
| 14:39.51 | azeteg | in the other direction, audio work for 0.5 seconds |
| 14:39.57 | azeteg | then its quiet |
| 14:40.03 | azeteg | any idea where I could start looking? |
| 14:40.40 | azeteg | rtp packets arrive |
| 14:40.45 | azeteg | but they contain nothing at al |
| 14:41.15 | MikeB | [TK]D-Fender there is suff in the latest zaptel.conf about priindication = outofband |
| 14:42.15 | [TK]D-Fender | MikeB : sounds more like for progress indication. |
| 14:42.42 | *** join/#asterisk apardo (n=apardo@176.Red-83-54-133.dynamicIP.rima-tde.net) |
| 14:42.58 | kimosabe | any one my x-lite says registered but when i enter the number it and dial it says call not aproved |
| 14:43.44 | C4T3l | are they in your dialplan |
| 14:44.49 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 14:44.53 | mercestes | morning. |
| 14:44.56 | chiardon | bkw? |
| 14:45.03 | C4T3l | murnin |
| 14:45.07 | kimosabe | c4t31 im using sip provider |
| 14:45.20 | C4T3l | whom? |
| 14:46.19 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:47.01 | *** join/#asterisk robtro (n=rob@unaffiliated/robtro) |
| 14:48.06 | kimosabe | selectfone |
| 14:48.31 | kimosabe | i have many acounts with them but not with xlite all with sipuras |
| 14:48.52 | C4T3l | are yu using asterisk in your configuration? Iax?? |
| 14:49.13 | Blackthorn | Hello, when calling from spa-2000 through * to my local pri I get a ring then halfway through the second reing you here it ringing even louder. Basicly double or echoed rings. Do you know how to change this? |
| 14:49.34 | kimosabe | c431 im just sending xlite directlly to selecfone |
| 14:49.50 | kimosabe | it sasys registered but i cant make calls |
| 14:50.39 | [TK]D-Fender | Blackthorn : You are setting too many extra indications on your PRI |
| 14:51.24 | MikeB | THere must be a way to trap extension '' in context blah ? anyone? |
| 14:51.33 | MikeB | PRI ZAP |
| 14:52.01 | C4T3l | kimosabe: perhaps you should contact them. this area is really more for questions about asterisk. sorry. |
| 14:52.08 | Blackthorn | feder: thank you for your responce, could you give me an example of an indicator that could be set? Should i go to zaptel and turn off echo and echo when bridged? |
| 14:52.37 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:53.21 | ManxPower | MikeB, extension "s" is run when there is no destination number., |
| 14:53.43 | ManxPower | However, if there is no destination number on an incoming call on a PRI, your config or provider is SERIOUSLY fucked. |
| 14:53.45 | file | lame lame lame |
| 14:54.04 | kimosabe | somo one know how to hack into the pap2 in order 2 iliminate the device being locked |
| 14:54.18 | *** join/#asterisk jhava (n=icechat5@200.58.26.21) |
| 14:54.21 | mercestes | have you plugged it into the internet? |
| 14:54.35 | mercestes | Kimo? |
| 14:54.39 | jake1932 | http://voip-info.org/wiki/view/MG3000-R - an american PR person is in order |
| 14:54.40 | kimosabe | yes |
| 14:54.55 | kimosabe | i screwed it up huh |
| 14:55.01 | mercestes | then if I rmeember correctly....your screwed, Kimo. |
| 14:55.17 | mercestes | it updates upon Internet connection and cannot be unlocked after that point. |
| 14:55.45 | kimosabe | ok no prob it is my friends device |
| 14:55.52 | kimosabe | i just wanted 2 help him out |
| 14:59.29 | *** part/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 15:00.16 | *** join/#asterisk AsteriskAddict (n=speedy@r172h230.dixie-net.com) |
| 15:01.15 | hwt | what is "hidden caller-id" called in asterisk? |
| 15:01.25 | hwt | anonymous call? |
| 15:01.50 | *** join/#asterisk MacDeath (n=davidn@196.202.248.34) |
| 15:03.19 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 15:03.43 | MacDeath | haai |
| 15:03.55 | MacDeath | as usual, i only come here if i have a question |
| 15:04.20 | MacDeath | i have a * machine on a private network with the IP 192.168.0.1 |
| 15:04.34 | MacDeath | i have 6 other private networks connected to it |
| 15:04.44 | MacDeath | and there is only routing (no nat) between them all |
| 15:04.52 | MacDeath | everything works great |
| 15:05.21 | azeteg | what could be the cause of one-way audio when talking through an oh323 channel? |
| 15:05.28 | MacDeath | I also have 2 incoming internet connections which NATs to the private range |
| 15:05.30 | azeteg | I have audio in one direction working fine |
| 15:05.39 | azeteg | in the other - I get audio for 0.5secs, then silence |
| 15:05.45 | azeteg | (incoming audio) |
| 15:05.50 | azeteg | people hear what I say |
| 15:06.02 | MacDeath | outgoing calls work fine, incoming doesnt |
| 15:06.03 | hwt | ugh, h323. |
| 15:06.11 | jake1932 | almost sounds like reinvite - but you're not using sip |
| 15:06.23 | azeteg | I try to analyze the traffic... |
| 15:06.27 | azeteg | and rtp packets arrive |
| 15:06.31 | azeteg | they just look... weird |
| 15:07.02 | mercestes | one way audio often is some sort of firewall debauchery. Check your firewall and tinkering with your nat settings sometimes fixes it. |
| 15:07.05 | coppice | there's nothing weirder than an RTP packet |
| 15:07.27 | *** join/#asterisk momelod (n=momelod@67.71.84.45) |
| 15:07.35 | momelod | greetings peoples of * |
| 15:07.39 | azeteg | hmm |
| 15:07.49 | sevard | anyown know any wml? ;/ |
| 15:07.51 | azeteg | well, I just think its weird it sounds for half a sec |
| 15:07.53 | azeteg | then silence |
| 15:08.08 | MacDeath | if i then add the IP range to nat.conf, incoming / outgoing calls to the internet work fine, but the phones on the other private ranges only have 1 way audio... |
| 15:08.12 | mercestes | IT is weird, I agree....but I come to appreciate weird. weird == job security. |
| 15:08.29 | azeteg | hehe |
| 15:08.53 | azeteg | want to see a .cap of the traffic? |
| 15:09.13 | momelod | i have a questions about default permissions for voicemail files. how do i make them readable by my httpd process? |
| 15:09.14 | mercestes | MacDeath: Any chance for port forwarding in the routers? |
| 15:09.18 | momelod | and writable too.. |
| 15:11.01 | MacDeath | mercestes : yeah, there is |
| 15:11.01 | MacDeath | but i didnt think i would need to |
| 15:11.06 | MacDeath | as the whole 192.168.0.0/21 is just routed |
| 15:11.28 | triple-e | anyone used the iptables tc functions for qos ? |
| 15:11.32 | MikeB | Manxpower sorry too many windows and phones |
| 15:12.15 | azeteg | jej |
| 15:12.17 | azeteg | heh |
| 15:12.24 | MikeB | Swithc providor is telling me there is no way to stop it doing what it does. |
| 15:13.58 | azeteg | http://194.218.216.4/dump/asterisk-dump-4.cap |
| 15:14.05 | azeteg | thats a dump from the one-way traffic |
| 15:14.28 | *** join/#asterisk shammash (n=shammash@host29-107.pool8252.interbusiness.it) |
| 15:16.32 | shammash | hi * |
| 15:17.10 | MikeB | ManxPower: Lucnet at tellling me that asterisk is broken they may have a point. |
| 15:17.56 | ManxPower | MikeB, I have several Asterisk servers with PRI. No problems |
| 15:18.07 | MikeB | Me too, |
| 15:18.08 | *** join/#asterisk watchy (n=watchy@70.238.57.237) |
| 15:18.09 | ManxPower | what switchtype is the line set to |
| 15:18.23 | MikeB | The issue here is that asterisk does not cope with a blank called number. |
| 15:18.37 | MikeB | manxpower euroisdn |
| 15:18.38 | ManxPower | MikeB, the exten => s did not work? |
| 15:18.43 | watchy | how do i make a polycom 501/601 ring when someones on the line instead of the calling waiting type beep? |
| 15:19.39 | ManxPower | watchy, I would read the admin manual |
| 15:19.48 | chiardon | Hewllo |
| 15:20.03 | watchy | i;m about to dload that but i was looking for a quick answer while i did that |
| 15:20.03 | triple-e | anyone used tc for qos |
| 15:20.12 | ManxPower | then I would read it again, and then read it again. The admin manual is not exactly easy to figure out. |
| 15:21.28 | chiardon | what's the meaning of this messege: "unable to get index, and nullok is not asserted" |
| 15:21.41 | *** part/#asterisk parag7732 (n=root@de2-b1976.alshamil.net.ae) |
| 15:23.00 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:23.30 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 15:23.34 | *** join/#asterisk bruce1266 (n=dflkjsdf@HSE-Toronto-ppp3490056.sympatico.ca) |
| 15:23.37 | MikeB | maxnpower no manual, :( |
| 15:24.04 | chiardon | "unable to get index, and nullok is not asserted" used to apper after call break |
| 15:24.40 | MikeB | ManxPower: No I tried s,t,i, _x. all no go |
| 15:24.53 | MikeB | I think asteris should cope with this, |
| 15:25.16 | mercestes | Watchy: Random answer: You could code in the same sip peer on all the lines. |
| 15:25.24 | mercestes | watchy: Results my vary. |
| 15:25.29 | bruce1266 | hello everyone ;;; I have a problem with oh323 installation ;;; I followed everything in the Add-on package and I got install successfull msg ;; but now there is oh323.conf file or any "oh323 show channel" in asterisk -r ;;; so how can I check if it is installed or not? |
| 15:25.54 | watchy | mer: i did that for 2 lines. it actually only rings on the earpiece of the phone |
| 15:26.03 | watchy | but i do see line 2 blinking. i actually want the phone to ring though |
| 15:26.24 | *** join/#asterisk Tili (i=Tili@202.133.65.44) |
| 15:26.51 | mercestes | watchy: Oh....I'd go to the manual...though another random answer...callsperline="1" again, results may vary. |
| 15:26.52 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 15:26.56 | dyn | hi |
| 15:26.57 | *** join/#asterisk triple-e (n=piaergnj@adsl-70-128-78-22.dsl.stlsmo.swbell.net) |
| 15:27.19 | dyn | is there any way to shorten the softphone configuration entries in sip.conf? can I use macros there somehow for example? |
| 15:27.32 | snitt | hmm |
| 15:28.28 | bruce1266 | Any help with oh323 please? |
| 15:28.52 | xai | manx: hey manx, whats up? |
| 15:28.53 | dyn | i mean, for configuring an Ekiga client, I seem to need about 15 lines in sip.conf....... now this for 20+ users seems a bit overkill :) |
| 15:29.03 | dyn | is there any config trick to shorten this? |
| 15:29.27 | mercestes | dyn: Copy paste always works for me. |
| 15:29.47 | dyn | mercestes: it should work but I prefer the DRY pattern (dont repeat yourself :) |
| 15:30.01 | mercestes | dyn: I would use VI (that's "Edit" in klingon.) It has a nice block copy paste function. |
| 15:30.11 | dyn | mercestes: and also, would it work for eg 300 softphones as well? |
| 15:30.17 | dyn | mercestes: (prefer vim myself ;) |
| 15:30.35 | bruce1266 | Has anyone tried the Add-Ons package for Asterisk@Home ;;; plz plz plz respond and it is driving me crazy !!!! |
| 15:31.05 | mercestes | Bruce1266: Read the damn topic. |
| 15:31.17 | MikeB | ManxPower: Do you think anyone here could issue a patch to set extension = 's' where extension='' ? |
| 15:31.19 | bruce1266 | what topic? |
| 15:31.40 | mercestes | Bruce1266: Channel #freepbx will be glad to help you with all your a@h questions. |
| 15:31.51 | dyn | bruce1266: channel topic :) |
| 15:31.56 | mercestes | Topic: Asterisk: The Open Source PBX -=- Asterisk 1.2.7.1 Released! (April 13, 2006), Upgrade from 1.2.7 only necessary if you use app_page -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 15:31.56 | bruce1266 | so it is a question about a@h |
| 15:32.10 | *** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 15:32.13 | mercestes | yes.....a@h is asterisk at home. |
| 15:32.25 | bruce1266 | perfecto ;) |
| 15:32.47 | bruce1266 | Thanks :) |
| 15:33.34 | mercestes | NP. |
| 15:33.37 | mercestes | :) |
| 15:36.43 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 15:36.47 | *** join/#asterisk TheoC (n=theochao@c-67-166-183-193.hsd1.tx.comcast.net) |
| 15:37.32 | TheoC | Is there a way to alert a user as to what number was dialed via asterisk? |
| 15:37.48 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 15:37.51 | Hmmhesays | there is always a way |
| 15:37.58 | TheoC | What might that way be? |
| 15:38.23 | Hmmhesays | money can buy you a lot of things |
| 15:38.27 | Hmmhesays | including answers |
| 15:39.02 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 15:39.40 | dyn | is there any way to shorten the softphone configuration entries in sip.conf? can I use macros there somehow for example? 15 lines of config for each softphone looks like a lot for me.. |
| 15:40.05 | TheoC | assuming i'm not willing to pay money for this particular answer, what ways are there? |
| 15:40.38 | Hmmhesays | Start reading |
| 15:40.49 | Hmmhesays | familiarize yourself with the dialplan and its commands |
| 15:40.54 | paryl | hi all... i'm hitting an issue where say, 2% of our incoming calls, when answered, the user hears their own voice and the caller hears themselves, but they can't hear each other. it's only a recent issue. i've reset the asterisk server, but i don't know what else to look for. any ideas? |
| 15:41.13 | TheoC | Any commands in particular? |
| 15:41.22 | chiardon | Hi |
| 15:41.27 | MrChimpy | can someone explain zap channel naming in logs? ideally I need to be able to know which E1 on which card a call has come in on.... |
| 15:42.16 | mercestes | paryl: Changed internet setttingts, changed IP's, funky router, funky switch. Mostly networking debauchery/changes. |
| 15:42.58 | *** join/#asterisk Cresl1n (n=matt@host56-92.pool8838.interbusiness.it) |
| 15:43.12 | ManxPower | MikeB, What IS your problem? put in exten => s,1,Goto(whateverextensionyouwant,1) and be done with it. |
| 15:43.15 | chiardon | Hi |
| 15:43.35 | chiardon | someon has som advice with how to manage this:"unable to get index, and nullok is not asserted" |
| 15:43.35 | paryl | mercestes: but nothing has really changed on my network at ALL... could just high bandwidth usage cause it? |
| 15:44.03 | chiardon | or . . . I can forget it |
| 15:44.16 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 15:45.40 | bruce1266 | Guys ;;; freepbx is not as expert as you guys are ;;; so has anyone installed H.323 support for Asterisk? |
| 15:46.42 | MikeB | ManxPower: Yes, I did that but is does not work! |
| 15:46.47 | ManxPower | MrChimpy, channels as numbnered sequentially |
| 15:46.48 | [TK]D-Fender | ManxPower : His PBX is sending calls via PRI to * but WITHOUT the DID, so Exten = "". "s" never gets called apparently. |
| 15:46.53 | ManxPower | MikeB, then you are doing something wrong. |
| 15:46.53 | MikeB | It says extension '' does not exist in context |
| 15:47.01 | ManxPower | [TK]D-Fender, it should be called. |
| 15:47.05 | ManxPower | MikeB, which context? |
| 15:47.19 | MikeB | isdn2 |
| 15:47.24 | GerbilWrk | alright, anyone have an example for listening into a call that another user is on? |
| 15:47.32 | ManxPower | MikeB, put your /etc/asterisk/zapata.conf and extensions.conf on pastebin.ca and make is fast if you want my help. |
| 15:48.09 | MikeB | for some reason earlier people were getting the demo message sometimes under a lot of load but the zapata.conf definatly specifies context=isdn2 |
| 15:48.12 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 15:48.21 | stack_ | morning everyone... |
| 15:49.09 | ManxPower | MikeB, I should be able to fix it for you in less than 5 mins, once those files are on pastebin.ca |
| 15:49.11 | *** join/#asterisk momelod (n=momelod@67.71.84.45) |
| 15:49.26 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 15:49.28 | ManxPower | But I have cable to pull and so don't have much more time to spend helping you. |
| 15:49.32 | Blackthorn | I am having an issue of echo rings (i guess for lack of better understanding) on outgoing calls through pri. A person here said "you are setting to many indicators on the pri" Can somone explain futher? I have turned off echo and echo when bridged on the pri and tried that. no change. |
| 15:50.09 | MikeB | ManxPower: Done, |
| 15:50.18 | ManxPower | MikeB, and the link? |
| 15:50.28 | MikeB | Manx would rather pull cables or teeth then this ; ) |
| 15:50.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.40) |
| 15:50.35 | momelod | i keep getting a warning about "DigitTimeout is deprecated", where can i see an example of the "Set(TIMEOUT(digit)=timeout)", im just not sure about the syntax.. |
| 15:50.59 | MikeB | ManxPower: sorry dont know pastbin. where do I look for the link? |
| 15:51.09 | dyn | laters.. |
| 15:51.09 | *** join/#asterisk lorinc (n=ang@caracas-3845.adsl.interware.hu) |
| 15:51.13 | many | www.pastebin.com |
| 15:51.16 | MikeB | http://pastebin.ca/55135 |
| 15:51.19 | MikeB | got it sorry |
| 15:51.20 | ManxPower | there |
| 15:51.23 | Hmmhesays | she did it, hells yeah |
| 15:51.33 | many | or better yet :P |
| 15:51.39 | [TK]D-Fender | momelod : Set(TIMEOUT(digit)=3) |
| 15:51.56 | MrChimpy | manxpower: sequentially as in Zap[1-120] on first card and Zap[120-240] on second TE411? |
| 15:52.11 | unixgeek | Anyone out there ever have a Digium T1 card not accept external clocking? |
| 15:52.17 | MrChimpy | and do they split on E1 channels, or are they numbered as they come in? |
| 15:52.22 | momelod | [TK]D-Fender: ty |
| 15:52.23 | stack_ | I have one user that is getting hung up on. She is using a Polycom 301 and dialing out through a TE110p to a PRI. The other end put her on hold and the call was dropped. The console had: "-- Channel 0/23, span 2 got hangup request" and "-- Hungup 'Zap/27-1'" This is only happening to one person |
| 15:52.29 | MrChimpy | come in/calls are recieved |
| 15:52.38 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 15:52.45 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 15:53.31 | ManxPower | MrChimpy, correct |
| 15:53.32 | Qwell | stack_: she's the only one complaining. |
| 15:53.42 | ManxPower | MikeB, paste the ACTUAL error message now |
| 15:53.45 | Qwell | stack_: she pisses people off, who say they're gonna put her on hold, and "accidently" hangup |
| 15:53.51 | MikeB | ManxPower: The error is : "-- Extension '' in context 'isdn2' from '0161xxxxx' does not exist. Rejecting call on channel 0/12 span 2 |
| 15:54.11 | azeteg | hmm |
| 15:54.20 | stack_ | Qwell: no she was in good relations with the people... it's happened to her 3 times in the past two days |
| 15:54.20 | ManxPower | stack_, remove busydetect=yes and remove callprogress=yes |
| 15:54.28 | Ariel_ | hello everyone |
| 15:54.30 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 15:54.31 | Ariel_ | ~ilbc |
| 15:54.33 | jbot | extra, extra, read all about it, ilbc is at http://www.ilbcfreeware.org |
| 15:54.35 | ManxPower | both options have problems with people with higher pitched voices |
| 15:54.51 | ManxPower | MikeB, did you do a "reload" in the asterisk CLI after making changes to your files? |
| 15:54.57 | azeteg | I still have this problem with audio in only one direction |
| 15:55.02 | MikeB | yes |
| 15:55.10 | MrChimpy | manx: but which is correct?! |
| 15:55.25 | azeteg | I have analyzed the rtp stream - and the correct audio exists in the rtp stream |
| 15:55.25 | MikeB | Will try again when everyone goes home in 45 minutes, (UK) |
| 15:55.34 | azeteg | but the phone won't play it back |
| 15:55.46 | azeteg | phone plays "dialing tone" only |
| 15:55.57 | azeteg | when speak audio is supposed to start |
| 15:55.59 | MrChimpy | or whatever it is |
| 15:55.59 | ManxPower | MikeB, you are missing a [general] and a [globals] sections in extensions.conf You should have them in there, even if they are empty |
| 15:56.01 | azeteg | it just goes silent |
| 15:56.20 | ManxPower | MrChimpy, MrChimpy manxpower: sequentially as in Zap[1-120] on first card and Zap[120-240] on second TE411? |
| 15:56.20 | MikeB | ManxPower, It seems to no like '' at all Im mor than happy to write a look to grab the number as dtmf yuck! but first I need to accept the call. |
| 15:56.21 | ManxPower | correct |
| 15:56.38 | MrChimpy | ok, thanks. and they're statically assigned? |
| 15:56.42 | MikeB | Manx sorry it was a snip of the relevent but |
| 15:56.56 | MrChimpy | in terms of the lines on the channels. |
| 15:57.02 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 15:57.17 | stack_ | ManxPower: neither of those options were set |
| 15:57.23 | MikeB | ManxPower: There is a general and globals section , sorry only send the relevent contects |
| 15:57.43 | Nivex | voicepulse hands a call to PBX A. caller dials extension, which hands call to PBX B. PBX B calls Authenticate(), but does not appear to be recieving digits. All call legs are IAX2. Suggestions to debug. |
| 15:57.49 | MrChimpy | makes sense - sorry - difficult working out what out switch is doing and what * is up to :) |
| 15:58.05 | ManxPower | MikeB, Asterisk is VERY picky about the format of the files. do a "show dialplan" Do you see an "isdn2" context |
| 15:59.39 | MikeB | ManxPower [ Context 'isdn2' created by 'pbx_config' ] |
| 15:59.40 | ManxPower | MrChimpy, "ztcfg -vvv" will show you all the channels, but will kill any active calls |
| 15:59.54 | *** join/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 16:00.01 | VoicePulse | Nivex: Is the dtmfmode same across the board? |
| 16:00.12 | imperfect- | Howdy. Anyone know why when I ring my sip phones when I answer it just dies? |
| 16:00.21 | Nivex | VoicePulse: it was my impression that dtmfmode was a moot point with IAX. |
| 16:00.39 | MikeB | ManxPower: This worked with Asterisk - Asterisk - BT PRI, The issue is that the Lucent Index does not send a number at all in the call setup and exten = s,1 doesn not cope. |
| 16:00.43 | ManxPower | IAX2 ONLY does out of band DTMF |
| 16:00.43 | *** join/#asterisk Funkmaster (n=spider@s55932490.adsl.wanadoo.nl) |
| 16:00.44 | *** join/#asterisk Malthus (n=root@uslec-66-255-127-2.cust.uslec.net) |
| 16:00.53 | Malthus | hi all |
| 16:01.06 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 16:01.17 | Funkmaster | hi every1 :D |
| 16:01.18 | VoicePulse | Nivex: Ah, right. Then we're back to your original question... |
| 16:01.32 | Malthus | I seem to be missing some signalling on a T1 e&m wink line |
| 16:02.04 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 16:02.04 | Malthus | sometimes when calls come in pressing digits doesn't work the IVRs |
| 16:02.16 | TripleFFFFFFFFFF | http://pastebin.ca/55138 |
| 16:02.26 | TripleFFFFFFFFFF | can one tll me why asterisk not talking the caller id ? 1.2.7.1 |
| 16:02.40 | Nivex | I just ran the call with "iax2 debug" and I am seeing DTMF packets come in. |
| 16:02.45 | ManxPower | MikeB, It SHOULD be working. try setting your switchtype=5ess just in case they lied to you. |
| 16:02.48 | TripleFFFFFFFFFF | Malthus also try elax dtmf |
| 16:02.54 | ManxPower | RESTART asterisk, don't reload |
| 16:03.33 | TripleFFFFFFFFFF | seems my callerid is cropped... |
| 16:03.49 | TripleFFFFFFFFFF | asterisk should remove \n \r etc from the srting no ? |
| 16:04.12 | TripleFFFFFFFFFF | anyone ? |
| 16:04.15 | Malthus | TripleFFFFFFFFFF thanks |
| 16:04.19 | Malthus | trying now |
| 16:04.22 | ManxPower | TripleFFFFFFFFFF, why? |
| 16:04.26 | MikeB | ManxPower = But I "know" that the BT line is euroisdn |
| 16:04.30 | TripleFFFFFFFFFF | look at paste bin |
| 16:04.35 | TripleFFFFFFFFFF | <PROTECTED> |
| 16:04.57 | TripleFFFFFFFFFF | i call cname to get cname from #.. it returns MONTREAL QC from a shell |
| 16:05.01 | TripleFFFFFFFFFF | no prob here |
| 16:05.05 | Nivex | My machine is running 1.2.7. I suppose I can try going to 1.2.7.1, but according to the changelog it was a minor bugfix release |
| 16:05.20 | TripleFFFFFFFFFF | but once i send the asnwer back to asterisk it SET CALLERID "MONTREAL QC |
| 16:05.31 | TripleFFFFFFFFFF | then nothing ... like killed by something |
| 16:05.38 | ManxPower | MikeB, And I know that the world should worship me as a god, but that's not happening either. Try changing the switch type. |
| 16:05.50 | TripleFFFFFFFFFF | <PROTECTED> |
| 16:05.54 | TripleFFFFFFFFFF | is my C line |
| 16:05.56 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 16:06.05 | Malthus | apart from missing the dtmf, it also misses the DNIS digits |
| 16:06.08 | Malthus | sometimes |
| 16:06.09 | TripleFFFFFFFFFF | maybe the second quote ? |
| 16:06.22 | TripleFFFFFFFFFF | malthus ... you know how to test ?????? |
| 16:06.24 | TripleFFFFFFFFFF | call fedex |
| 16:06.29 | *** join/#asterisk tld (n=tld@343006.ds.nac.net) |
| 16:06.31 | TripleFFFFFFFFFF | neve r works for me |
| 16:06.41 | Malthus | say what? |
| 16:06.46 | TripleFFFFFFFFFF | dtmf fedex |
| 16:06.50 | azeteg | hmmm |
| 16:06.51 | Malthus | oh |
| 16:06.55 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 16:06.58 | Malthus | well outgoing seems to be fine |
| 16:07.00 | boch | hello |
| 16:07.02 | TripleFFFFFFFFFF | press 1 for crap.. oh |
| 16:07.03 | TripleFFFFFFFFFF | ok |
| 16:07.08 | tld | Could anyone give me a hand understanding US billing and rate centers? Is the system such that calls in the same rate center are free, calls withing the state are cheap, and inter-state calls cost more? |
| 16:07.09 | Malthus | its my IVR that doesn't work |
| 16:07.17 | azeteg | why the hell could RTP traffic be sent from gatekeeper ok, but asterisk not forwarding it to the phone correctly? |
| 16:07.19 | Malthus | and when it works, all the digits work |
| 16:07.29 | mercestes | I think it is the second quote triple...I don't see it in yoru pastebin. |
| 16:07.33 | Malthus | when it doesn't work, the digits never work |
| 16:07.40 | ManxPower | TripleFFFFFFFFFF, DON'T PUT QUOTES IN CALLERID |
| 16:07.43 | TripleFFFFFFFFFF | so if line as a \n or other eol char .. in snprintf(cmd,200,"SET CALLERID \"%s\" <%s>", line, agi.callerid);.. that will fuck my callerid.. how can i ? |
| 16:07.45 | TripleFFFFFFFFFF | what ? |
| 16:07.48 | TripleFFFFFFFFFF | i need... |
| 16:07.59 | TripleFFFFFFFFFF | SET CALLERID "TEST"<numebr> |
| 16:08.00 | Malthus | its not like some of the digits work, but like the digits work on some of the calls |
| 16:08.01 | TripleFFFFFFFFFF | no ? |
| 16:08.12 | MikeB | ManxPower: Will try when everyone goes home thanks for your help. |
| 16:08.16 | mercestes | tld : Intrastate calls are more expensive than interstate calls. |
| 16:08.22 | ManxPower | NO, SET CaLLERID Robert Dobbs <666> |
| 16:08.33 | TripleFFFFFFFFFF | tryuing |
| 16:08.33 | ManxPower | TripleFFFFFFFFFF, Callerid does not contain quotes |
| 16:08.49 | tld | mercestes, huh? calls withing the state costs more than calls to another state? |
| 16:09.00 | Malthus | there is no 'zap debug'? |
| 16:09.11 | TripleFFFFFFFFFF | hmmm |
| 16:09.16 | TripleFFFFFFFFFF | nope |
| 16:09.27 | VoicePulse | Nivex: You did iax2 debug on A or B? |
| 16:09.28 | TripleFFFFFFFFFF | unless.. nope.. |
| 16:09.40 | Nivex | VoicePulse: B |
| 16:10.01 | TripleFFFFFFFFFF | ${CALLERID} should display all the info right ? |
| 16:10.05 | Nivex | The tones are making it all the way to the destination PBX. It's as though Authenticate() is not parsing |
| 16:10.51 | *** join/#asterisk wolfson (n=wolfson@65.167.193.79) |
| 16:11.03 | TripleFFFFFFFFFF | <PROTECTED> |
| 16:11.52 | VoicePulse | Nivex: Are you Answer()ing the call before doing Authenticate() on B? |
| 16:12.29 | Nivex | VoicePulse: yep. |
| 16:13.08 | VoicePulse | Nivex: Same version of * on both? |
| 16:13.41 | TripleFFFFFFFFFF | found it.. .manpower |
| 16:13.41 | Nivex | VoicePulse: he's running 1.2.6 and I'm on 1.2.7, but as I said, I can see the IAX DTMF packets reaching me. |
| 16:14.22 | TripleFFFFFFFFFF | http://pastebin.ca/55141 |
| 16:14.23 | TripleFFFFFFFFFF | see ? |
| 16:14.30 | TripleFFFFFFFFFF | in the top .. |
| 16:14.47 | TripleFFFFFFFFFF | *****************RESULT FROM SHELL***********MONTREAL QC |
| 16:14.47 | TripleFFFFFFFFFF | *************END OF RESULT |
| 16:14.53 | Malthus | TripleFFFFFFFFFF: http://kb.digium.com/entry/1/113/ |
| 16:14.55 | TripleFFFFFFFFFF | theres a line feed... |
| 16:15.04 | VoicePulse | Nivex: A is 1.2.7 and B is 1.2.6? |
| 16:15.07 | Malthus | I might have the wrong e&m signalling |
| 16:15.14 | Nivex | VoicePulse: reverse that |
| 16:15.14 | Malthus | I'll check that out |
| 16:15.24 | Malthus | thanks for the relax dtmf tip |
| 16:15.25 | Nivex | actually B is 1.2.7.1 as of about 1 minute ago |
| 16:15.50 | TripleFFFFFFFFFF | nahi mean.. its just that it has a new line invisible in the string... so it crops the rest... cant set an invalid callerid format..so |
| 16:15.58 | TripleFFFFFFFFFF | how can i strip it down ? |
| 16:17.13 | MikeB | manxpower have fun pulling your cables in. |
| 16:17.43 | VoicePulse | Nivex: Ok, test again... Otherwise, we're running out of ideas. Maybe pastebin the relevant info and someone here might see something we've missed. |
| 16:20.00 | SplasPood | Does anyone have a DID that always returns CONGESTION when called? need to test something |
| 16:20.17 | SplasPood | ideally an always BUSY too |
| 16:20.29 | Malthus | give me 10 dolla and I do dat on Tuesday |
| 16:20.41 | TripleFFFFFFFFFF | <PROTECTED> |
| 16:20.56 | Nivex | http://pastebin.ca/55143 |
| 16:21.02 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 16:21.19 | TripleFFFFFFFFFF | hmm.. not quite.. but getting somewhere.. |
| 16:21.19 | TripleFFFFFFFFFF | 0ET CALLERID Montreal QC<41831234556> |
| 16:21.21 | TripleFFFFFFFFFF | ;) |
| 16:21.32 | SplasPood | s/bases/bashes |
| 16:21.38 | TripleFFFFFFFFFF | can one tell me what wrong with that line of code... |
| 16:21.42 | TripleFFFFFFFFFF | line containts the string. |
| 16:21.43 | TripleFFFFFFFFFF | <PROTECTED> |
| 16:21.54 | TripleFFFFFFFFFF | or shoudlit be sizeof |
| 16:22.26 | rpm | nothing is wrong with it if you are trying to pad the second to last character with a 0. |
| 16:22.41 | TripleFFFFFFFFFF | im trying to remove the last |
| 16:22.43 | rpm | is line a pointer or char array? |
| 16:22.53 | TripleFFFFFFFFFF | <PROTECTED> |
| 16:23.02 | TripleFFFFFFFFFF | while ( fgets( line, sizeof line, fp)) { |
| 16:23.03 | TripleFFFFFFFFFF | ;) |
| 16:23.47 | TripleFFFFFFFFFF | i know theres a stupid new line char or something in that string.. i need to remove it for asterisk agi to correctly return the function set callerid |
| 16:24.05 | TripleFFFFFFFFFF | rpm also it paded the first character |
| 16:24.05 | rpm | char line[300]; char *p = line; while (fgets(p, sizeof(line), fp)) { p[strlen(p)-1]='\0'; } |
| 16:24.22 | rpm | and then just reset the pointer every loop |
| 16:24.24 | sevard | you're a hwhole lot of F |
| 16:24.39 | TripleFFFFFFFFFF | ;) thanks |
| 16:25.32 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 16:25.33 | sevard | ./hump --eyebrow file |
| 16:25.39 | TripleFFFFFFFFFF | lets try |
| 16:25.41 | file | eeeeeeeeep |
| 16:26.26 | gonzo- | anybody with VegaStream devices experience? |
| 16:26.38 | sevard | I've heard about that |
| 16:26.38 | TripleFFFFFFFFFF | nope |
| 16:26.41 | sevard | Do I qualify |
| 16:26.42 | TripleFFFFFFFFFF | its croping first char |
| 16:27.58 | *** join/#asterisk mog_home (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 16:28.47 | rpm | TripleFFFFFFFFFF: *p=0; will pad the end of the line with a null character also. |
| 16:28.52 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:28.59 | nahirean | SplasPood, the nufone support number always rings busy if you want to use it for testing |
| 16:29.07 | TripleFFFFFFFFFF | i need to remove that i think |
| 16:29.25 | Hmmhesays | wiki wiki wild wild |
| 16:29.31 | mog_home | omg its malcomd |
| 16:30.02 | TripleFFFFFFFFFF | k got it |
| 16:30.35 | TripleFFFFFFFFFF | nope. crashes |
| 16:32.13 | *** join/#asterisk Cresl1n (n=matt@host56-92.pool8838.interbusiness.it) |
| 16:32.40 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 16:33.10 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 16:33.53 | TripleFFFFFFFFFF | darn |
| 16:34.21 | TripleFFFFFFFFFF | ok then other approach |
| 16:34.28 | TripleFFFFFFFFFF | in bash how can i stip last char ? |
| 16:34.31 | TripleFFFFFFFFFF | or null crap |
| 16:34.44 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:35.29 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 16:36.06 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 16:37.47 | TripleFFFFFFFFFF | i think popen adds that |
| 16:39.00 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:40.05 | TripleFFFFFFFFFF | darn |
| 16:40.17 | *** join/#asterisk Lino` (n=Lino@87.123.243.255) |
| 16:40.59 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:42.02 | TripleFFFFFFFFFF | ok asterisk bug |
| 16:42.29 | TripleFFFFFFFFFF | AGI Script cname completed, returning 0CALLERID BOB DURELL<4185551212> |
| 16:42.30 | TripleFFFFFFFFFF | -- Executing noop("SIP/inmte-df61", ""418655000" <418655000> |
| 16:42.51 | TripleFFFFFFFFFF | so if i set callerid ? how come is the noop showing before the agi ? |
| 16:44.18 | SplasPood | nahirean: hahahaha |
| 16:45.44 | TripleFFFFFFFFFF | but no one told me #1 callerid is now ${CALLERID(name)} |
| 16:47.35 | nahirean | =] |
| 16:53.54 | myiagy | when i'm in the middle of a call, and another call comes in, i hear a "beep".. is it my phone that does the beep or is it asterisk? |
| 16:54.32 | myiagy | i'm guessing it's the phone? |
| 16:54.49 | coppice | analogue phone, or IP phone? |
| 16:54.53 | myiagy | IP |
| 16:56.15 | TripleFFFFFFFFFF | both |
| 16:56.23 | TripleFFFFFFFFFF | asterisk |
| 16:56.27 | TripleFFFFFFFFFF | but both hear it |
| 16:57.09 | myiagy | i see.. can it be turned off? |
| 16:58.07 | SplasPood | depends on the phone, or you can use asterisk to insure only one call is routed at any given time |
| 16:58.14 | [TK]D-Fender | myiagy : actually if its your phone, only you should hear it and as for disabling it, that dependson your phone. |
| 16:58.23 | [TK]D-Fender | myiagy: What make/model? |
| 16:59.08 | SplasPood | [TK]D-Fender: You don't happen to have any experience setting up labels within macros using AEL, do you? |
| 16:59.38 | *** part/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 16:59.50 | Blackthorn | I am having an issue of echo rings (i guess for lack of better understanding) on outgoing calls through pri. A person here said "you are setting to many indicators on the pri" Can somone explain futher? I have turned off echo and echo when bridged on the pri and tried that. no change. |
| 17:00.03 | myiagy | [TK]D-Fender i dunno.. gotta check |
| 17:02.04 | *** join/#asterisk Splat (n=Splat@220-253-98-106.TAS.netspace.net.au) |
| 17:02.16 | myiagy | [TK]D-Fender oh, it seems it's not an IP phone.. |
| 17:02.21 | myiagy | it's a linksys PAP2 |
| 17:04.04 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 17:06.19 | myiagy | SplasPood in order to do that i have to set incominglimit=1 right? |
| 17:07.20 | MikeB | manxpower: are you still here? |
| 17:07.44 | MikeB | manxpower: the issue seems to be that the number is sent as an information element and not in the call setup, |
| 17:08.36 | SplasPood | myiagy: I believe so.. never done it |
| 17:08.37 | [TK]D-Fender | SplasPood : I don't touch AEL, serves no needed purpose in my eyes |
| 17:08.54 | SplasPood | [TK]D-Fender: case... case statements are nice :) |
| 17:09.00 | myiagy | ok thanks |
| 17:09.00 | [TK]D-Fender | myiagy : You can only DISABLE call-waiting period, not just the "beep". |
| 17:09.20 | [TK]D-Fender | SplasPood : Nice, but rarely of consequence. |
| 17:09.38 | [TK]D-Fender | myiagy : You'd do that in your PAP2 web-config |
| 17:10.02 | myiagy | [TK]D-Fender oh, that's fine.. i won't need call-waiting anyways.. :) |
| 17:10.05 | myiagy | thank ypu |
| 17:10.08 | myiagy | you* |
| 17:10.17 | [TK]D-Fender | myiagy : np |
| 17:10.27 | [TK]D-Fender | myiagy : it is nice for several things though. |
| 17:11.25 | myiagy | [TK]D-Fender i know.. it's not for me though.. client told me "i don't want that beep.. in fact, i don't want more than 1 call at a time".. so, gotta do it :P |
| 17:11.53 | MikeB | Help!, Im having a lot of trouble getting an asterisk server to talk to a lucent INDEX, the problem is that the lucent creates a call setup without a called number and then sends the called number in an Information element, is there are way to make asterisk cope with this? |
| 17:12.00 | [TK]D-Fender | myiagy : Best reason to do it :) |
| 17:12.11 | myiagy | just out of curiosity, wont disabling call waiting affect transfer? |
| 17:12.35 | [TK]D-Fender | myiagy : No, basically when the ATA receives the 2nd call it will reject it outright. |
| 17:12.57 | [TK]D-Fender | myiagy : Transfer is something YOU initiate by way of the ATA and can be disabled independantly |
| 17:13.19 | myiagy | oh, ok.. i'll disable the waiting then :) |
| 17:13.57 | [TK]D-Fender | myiagy : Think on 3-way calling while you're at it. |
| 17:14.53 | Ariel_ | MikeB, how are you connected to the lucent system |
| 17:15.03 | MikeB | E-PRI |
| 17:15.26 | MikeB | euroisdn to a digium dual PRI card |
| 17:15.52 | myiagy | [TK]D-Fender what you mean? conferences won't work? (sorry, english is not my main language) |
| 17:16.02 | MikeB | Ariel_ Lucent - Asterisk - BT PRI |
| 17:16.13 | Ariel_ | MikeB, I would suggest you give a call to digium if it's there board they can help you with it. But most pri sends the info on the dchannel |
| 17:16.37 | MikeB | Ariel_ Calls from PRI - Asterisk - Lucent are fine but Lucent - Asterisk fail because called number is blank on call setup. |
| 17:17.04 | MikeB | Ariel_: It does send it on the d channel but as an information element and not in the call setup. |
| 17:17.20 | justinu | info elements are how everything is sent in q931 |
| 17:17.48 | MikeB | justinu: I maybe confussed then, could you look at a debug dump? |
| 17:17.56 | justinu | ok, pastebin |
| 17:19.58 | MikeB | justinu http://pastebin.ca/55146 |
| 17:20.00 | MikeB | thanks. |
| 17:20.33 | MikeB | justinu, had to trick asterisk with and exten = ,1,DISA to get this far. |
| 17:21.03 | justinu | you're right. I don't see any Called Party IE in that setup. |
| 17:21.22 | justinu | in your PBX, what type of trunk have you configured? Tie trunk? |
| 17:21.47 | jake1932 | justinu - you would happen to have an example of a working Call Route Selection to a VDN? |
| 17:22.00 | jake1932 | s/would/wouldn't |
| 17:22.10 | justinu | heh, sorry |
| 17:22.32 | jake1932 | guess it was worth a try :) |
| 17:22.37 | MikeB | justinu I simply unplugged the Lucent from the Telco and used a crossover cable to connect it to the asterisk box |
| 17:22.58 | justinu | MikeB: bizzare, I can't imagine how that could have ever worked! |
| 17:23.15 | MikeB | It does the Telco cope some how! |
| 17:23.38 | MikeB | Im told that this is how all lucent indexes behave. |
| 17:24.30 | MikeB | Im pretty sure that this isn't just me Im fairly sure I need someone to confirm what asterisk is expected to do in this case, cant do much more debugging on my own. |
| 17:24.52 | MikeB | but agree that the lucent is being weird but it is none the less working without the asterisk box. |
| 17:24.54 | [TK]D-Fender | myiagy : No, they will work until you disable them. Just that since you are limiting all sorts of features I wouldn't want you to overlook thatone :) |
| 17:25.41 | MikeB | Anyone else help ? Should asterisk cope with an pri_net call with no Called Party ID set ? |
| 17:25.43 | justinu | yeah - there's no way that could work in my opinion. if the setup message contains no called party IE, there's no way to tell what the destination of the call is. |
| 17:25.58 | myiagy | [TK]D-Fender right.. i think that pretty much all she needs is make and receive calls so, probably won't have much trouble with advanced features.. |
| 17:26.09 | MikeB | The telco would give a dialtone and wait for the information elements that follow. |
| 17:26.10 | justinu | ohhhh |
| 17:26.12 | justinu | wait a second |
| 17:26.21 | justinu | Overlapped Dialing! |
| 17:26.25 | MikeB | yep, |
| 17:26.28 | justinu | that must be it... |
| 17:26.48 | Blackthorn | Is there anyway when sending a call out a pri to tell * not to send a ring ? |
| 17:26.49 | MikeB | s/works/words |
| 17:26.53 | Blackthorn | or play a ring sound? |
| 17:27.15 | justinu | MikeB, i see... your switch is sending the called party IE in an "INFORMATION" message, not in the setup message. |
| 17:27.26 | MikeB | justinu Yep |
| 17:27.31 | justinu | mikeb: did you make sure to set switchtype=euro in your zapata.conf? |
| 17:27.35 | jake1932 | MikeB - called party is in there! |
| 17:27.45 | jake1932 | [70 0b 80 30 38 30 30 35 30 30 30 30 35] |
| 17:27.49 | jake1932 | that's it |
| 17:28.06 | MikeB | not in the setup though? |
| 17:29.04 | jake1932 | right |
| 17:29.28 | *** join/#asterisk pbx1 (n=pbx1@124.106.209.150) |
| 17:29.52 | MikeB | jake1932: I think that asterisk is not able to use the information element I had to add a blank exten to the content to stop it looping. |
| 17:30.01 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 17:30.13 | justinu | MikeB: pastebin your zapata.conf |
| 17:30.15 | jake1932 | agreed - it should be in the setup |
| 17:31.59 | MikeB | justinu http://pastebin.ca/55147 |
| 17:32.25 | justinu | MikeB: try adding the following to your zapata.conf |
| 17:32.32 | justinu | MikeB: overlapdial=yes |
| 17:32.59 | justinu | restart asterisk |
| 17:33.12 | *** part/#asterisk codebreaker (n=codebrea@xserver.flexserv.de) |
| 17:34.13 | MikeB | seems the same will change extensions. |
| 17:34.39 | *** join/#asterisk RevK (n=RevK@excalibur.bracknell.aaisp.net.uk) |
| 17:34.46 | MikeB | Hi Revk |
| 17:34.50 | RevK | Hi |
| 17:35.26 | pbx1 | Hello |
| 17:35.44 | MikeB | justinu I added overlapdial=yes but seems the same, I think I may need to do something in extensions.conf. |
| 17:35.55 | MikeB | I think your on the right track though. |
| 17:36.02 | pbx1 | I have a problem with called agents not being able to hear callers. can any body help out? |
| 17:37.05 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:39.38 | MikeB | justinu: http://pastebin.ca/55149 |
| 17:39.57 | Nivex | Need some help on a Authenticate() problem: http://pastebin.ca/55143 |
| 17:40.29 | justinu | what are we looking at? |
| 17:40.40 | justinu | looks like a call teardown, but not sure why thats significant |
| 17:40.49 | *** join/#asterisk ^unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 17:40.56 | MikeB | justinu does not seem any better with overlapdial=yes |
| 17:41.26 | justinu | what country you in? |
| 17:41.38 | MikeB | justinu: what sould I have in extenstions.conf I simple want to put this call out to ZAP/g1 |
| 17:41.47 | MikeB | justinu: UK england |
| 17:41.52 | justinu | Dial(ZAP/G1/${EXTEN}) |
| 17:41.58 | justinu | something like this |
| 17:42.11 | MikeB | Yeah, I think that the problem because EXTEN='' |
| 17:42.14 | justinu | exten => _X.,1,Dial(ZAP/G1/${EXTEN}) |
| 17:43.18 | MikeB | I need exten= ,1,dial or it goes nuts with extension '' not in context... |
| 17:43.35 | *** part/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 17:43.38 | justinu | you tried _X. after setting overlapdial? |
| 17:44.01 | Nivex | Got it. My problem appears to be related to bug 6011 |
| 17:44.12 | MikeB | justinu: no will try now |
| 17:46.09 | MikeB | justinu: It gets in a big loop because extension '' does not exist in context. |
| 17:46.16 | harryvv | Last night I was troubleshooting my asterisk box as to why it was first, forwarding calls to another extention and why I cannot kill the call forwarding. Looked into the log files and there seems to be a error generated when asterisk is reaching into the database to change the call forwarding feature. I have this in paste bin for anyone who cares to look at it. |
| 17:46.32 | harryvv | http://pastebin.ca/55151 |
| 17:46.42 | harryvv | I will brb wife is naging me to do something. |
| 17:47.10 | *** join/#asterisk Teeli (i=Tili@202.133.67.131) |
| 17:48.19 | VoicePulse | Nivex: Good to know... let us know if that fixes it. |
| 17:48.35 | Nivex | VoicePulse: It does. As soon as I turn off the jitterbuffer, DTMF works fine |
| 17:49.11 | justinu | mikeB: why a loop tho? it should just give up the first time, and reject the call |
| 17:49.13 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 17:49.37 | MikeB | I think the index is looking for a free channel. |
| 17:50.11 | MikeB | justinu, some progress, Accepting overlap call from '01614xxxxxx' to '<unspecified>' on channel 0/12, span 2 |
| 17:50.22 | justinu | ok |
| 17:50.27 | justinu | that looks better maybe |
| 17:50.53 | justinu | mikeb: can you understand C code? |
| 17:51.03 | MikeB | a little, |
| 17:51.08 | MikeB | tiny |
| 17:51.12 | MikeB | tryme |
| 17:51.15 | justinu | well, that probably won't help, because the code that makes this all work is a real mess |
| 17:51.21 | BadPacket | anyone seen pudgy? |
| 17:51.22 | justinu | but I was gonna say, take a look |
| 17:51.30 | justinu | libpri, and chan_zap.c |
| 17:51.33 | BadPacket | I'm still having a problem w/ nufone |
| 17:52.14 | MikeB | justinu: Dont think I understand it enough, but I would like to confirm that asterisk "should" do it and if so is my config correct |
| 17:52.30 | harryvv | anyone here ever delt with what is unresponsive call forwarding feature in asterisk? |
| 17:52.31 | justinu | well, i could only do that via reviewing the code |
| 17:52.32 | wolfson | badpacket: nufone incomming is still down |
| 17:52.37 | justinu | and that would be time consuming |
| 17:52.46 | BadPacket | wolfson: that would explain my problem |
| 17:52.48 | justinu | lol |
| 17:54.49 | justinu | mikeb: i'm pretty sure your config is correct, question is whether the asterisk PRI stack can understand what your PABX is saying to it. |
| 17:54.53 | harryvv | What does this mean? |
| 17:54.55 | MikeB | justinu, Thanks for your help, I undersand, its timeconsuming Its beyond me I think and this is very frustraiting, I have been working on this all day yesterday and all day today its 18:55 and I started at 06:00, |
| 17:54.55 | harryvv | # |
| 17:54.55 | harryvv | May 11 10:40:42 VERBOSE[1316]: -- Got SIP response 302 "Moved Temporarily" back from 192.168.10.50 |
| 17:54.55 | harryvv | # |
| 17:54.56 | harryvv | May 11 10:40:42 DEBUG[1316]: Found 302 Redirect to extension '201' |
| 17:55.06 | MikeB | justinu: does that help you feel sorry for me? |
| 17:55.12 | justinu | not really :P |
| 17:55.21 | justinu | been there done that |
| 17:56.07 | jake1932 | looks like your call is being forwarded to extension 201 |
| 17:56.09 | MikeB | justinu Do you thinking digium express support can help? |
| 17:56.12 | ManxPower | harryvv, it means they enabled call forwarding on their SIP phone |
| 17:56.21 | justinu | MikeB: i would hope so! |
| 17:56.24 | justinu | they created this stuff |
| 17:56.38 | ManxPower | MikeB, try asking on the mailing list and if that doesn't work, post a bug on bugs.digium.com |
| 17:56.42 | MikeB | yes be there is a lot of water in the pond ;) |
| 17:56.52 | justinu | MikeB: i would suggest turning off overlap dialing in the PABX if you can. |
| 17:57.18 | MikeB | ManxPower: really need to get this cracked quickly, will pay if I have to, I think that overlapdialing=yes has helped |
| 17:57.34 | ManxPower | MikeB, overlapdial is only normally used for OUTGOING calls. |
| 17:57.48 | MikeB | justinu: would love to change index, infact would rather rip it out and replace with soms; but no budget |
| 17:57.54 | ManxPower | and only if you have a . in your outgoing extension pattern |
| 17:58.09 | justinu | MikeB: you can't modify it's config at all? |
| 17:58.37 | MikeB | ManxPower: Right but I thin justin is on the right lines but maybe noone has thought of it for incomming? |
| 17:59.02 | justinu | nah, it's gotta work in both directions |
| 17:59.25 | MikeB | justinu dont know where to start I do have a password for it , Any lucent expersts in the channel? Im told it Can't do it but not am not sure |
| 17:59.37 | justinu | no, i've never seen that system before |
| 17:59.51 | ManxPower | MikeB, there are thousands of people using EuroISDN with Asterisk and do NOT have your prpblem |
| 17:59.53 | MikeB | Its quite popular but old I beleive. |
| 18:00.08 | ManxPower | MikeB, are you in the UK? |
| 18:00.18 | MikeB | ManxPower: I know I too use it and the asterisk-Telco PRI is fine! |
| 18:00.29 | MikeB | ManxPower: yep UK, BT Line |
| 18:00.56 | chiardon | ManxPower . . but plenty of another problems like me |
| 18:01.06 | ManxPower | I didn't know you were trying Asterisk <-> internal PBX connection./ |
| 18:01.24 | MikeB | ManxPower: the issue is that span 2 is pri_net and the switch is sending the called number in the information element not the call setup |
| 18:01.27 | harryvv | Manx, whats strange is I dont recall using call forwarding on my phone but I use the proper dtmf command set to disable it in extension 200 which is a polycom ip500. I hear a verbal confirmation from alison that is is disabled and looked at it in cli to verify. I make another call to my asterisk box and it still calls the other extension. The vm is saved in extension 200 which is what it should do. |
| 18:01.45 | harryvv | Manx I just submited this to list.digium.com a few min ago. |
| 18:01.53 | justinu | MikeB: perhaps setting "immediate=no"? |
| 18:02.00 | justinu | in zapata.conf |
| 18:02.09 | MikeB | ManxPower: >> BT PRI - asterisk - lucent is fine but lucent - asterisk - BT PRI is not . |
| 18:02.13 | ManxPower | justinu, immediate=yes should never ever ever be used unless you know what you are doing. |
| 18:02.23 | ManxPower | immediate=yes would cause these problems |
| 18:02.48 | jake1932 | harryvv: is .50 your phone? |
| 18:02.52 | justinu | he doesn't have anything set... i assume immediate=no is default |
| 18:02.59 | MikeB | shall I add immediate=no? |
| 18:03.07 | justinu | might as well try |
| 18:03.14 | chiardon | inmediate=yes is for a hot line |
| 18:03.16 | harryvv | yes. .50 is the polycom ip 500 extention 200 I am having problems with |
| 18:03.36 | jake1932 | harryvv: looks like your phone itself is forwarding - not asterisk |
| 18:03.47 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 18:03.49 | ManxPower | harryvv, I did tell you what the cause is |
| 18:04.00 | harryvv | jake, well I dont recall how i ever used such a feature on the phone. |
| 18:04.19 | jake1932 | ok |
| 18:04.25 | ManxPower | harryvv, the phone is either set up to forward calls or maybe DND is enabled |
| 18:04.49 | *** join/#asterisk rnovotny22 (n=Bob@198.57.19.126) |
| 18:04.51 | harryvv | DND would show it as enabled on the display..its not |
| 18:05.09 | chiardon | justinu . . Hi . . this message "got pulse digit 16 on TE4/0/3/4" ? |
| 18:05.11 | ManxPower | harryvv, I have 80 polycoms. this is how polycoms forward calls |
| 18:05.23 | MikeB | manxpower thats much better now what exactly should I have in extensions.conf? |
| 18:05.41 | harryvv | Can by chance the xml be setup or changes to call forward. A week to a week and a half ago made some minor changes but nothing about call forwaring to extention 201 |
| 18:05.53 | MikeB | manxpower do I still need exten = ,1,noop? |
| 18:06.21 | harryvv | ManxPower how do thay forward calls? |
| 18:07.02 | ManxPower | harryvv, they send back a moved temoporarily message to Asterisk |
| 18:07.35 | harryvv | I see |
| 18:07.55 | MikeB | manxpower, You the man! |
| 18:08.00 | MikeB | justin you too! |
| 18:08.03 | harryvv | That still does not resolve my dilema. I dont see any call forwarding option on the phone. |
| 18:08.15 | MikeB | manxpower any way to remove the delay? |
| 18:08.36 | justinu | ? |
| 18:08.39 | justinu | what happened? |
| 18:08.56 | *** join/#asterisk jahani (n=k@41.250.40.245) |
| 18:09.07 | MikeB | justinu imediate=no fixed it |
| 18:09.15 | MikeB | http://pastebin.ca/55155 |
| 18:09.21 | chiardon | justinu . . . what's the meaning of the message . . something is working bad? |
| 18:09.28 | justinu | mikeB: good deal |
| 18:09.46 | justinu | where's the delay? |
| 18:10.00 | MikeB | Thanks guys, would just like to make it a little quicker now, there seems to be a delay. |
| 18:10.32 | harryvv | the Ip500 has the status as online vs busy,brb or away. So it must be a asterisk then |
| 18:10.40 | justinu | i think you'd have to change the PBX to en-bloc dialing to fix that |
| 18:10.42 | jake1932 | it's not asterisk |
| 18:10.52 | jake1932 | the 302 is coming from your phone |
| 18:11.05 | MikeB | justinu seems to be 4 seconds between Called g1/0800500005 and Zap/1-1 is proceeding passing it to Zap/43-1 |
| 18:11.08 | harryvv | my hands are up in the air. What would cause it |
| 18:11.15 | jake1932 | check the xml file |
| 18:11.43 | harryvv | okay |
| 18:11.55 | jake1932 | manx is the resident polycom expert |
| 18:12.02 | justinu | mikeB: hmm, maybe stop using overlapdial on the span that goes to your telco? |
| 18:12.07 | justinu | not sure |
| 18:12.23 | MikeB | justinu: Im never happy, its not too bad at all. |
| 18:12.35 | MikeB | thanks for you help guys! |
| 18:12.41 | harryvv | I suspect it is the phone..just not the phone its self but its xml configuration. |
| 18:12.43 | justinu | sure, good luck |
| 18:13.15 | jake1932 | http://threebit.net/mail-archive/asterisk-users/msg01285.html |
| 18:13.25 | jake1932 | <PROTECTED> |
| 18:13.30 | chiardon | justinu . . what's PBX to en-bloc dialing? |
| 18:13.46 | justinu | that was for mikeb |
| 18:14.16 | chiardon | OK . . . but the overcome must be . . ? |
| 18:15.40 | chiardon | justinu . . . if he will made this change . . .wich the overcome? |
| 18:15.53 | justinu | oh, the delay |
| 18:16.03 | chiardon | haaaaaappppp . . .thanks |
| 18:16.09 | justinu | en-block sends all digits at once, overllap sends them one at a time |
| 18:16.57 | chiardon | clear . .thanks |
| 18:20.52 | stoffell | very off-topic.. is it forbidden/bad practice to log the irc channel and keep logs for future reference? |
| 18:21.03 | stoffell | (or even place them online?) |
| 18:21.05 | carrar | yes |
| 18:21.07 | harryvv | okay justinu I made some changes to phone1.cfg where the divert was changed |
| 18:21.09 | carrar | RFC1318 |
| 18:21.13 | carrar | states no logging |
| 18:21.49 | carrar | err RFC1459 |
| 18:21.51 | carrar | heh |
| 18:22.03 | stoffell | uh, carrar, oke :) but it would be handy, wouldn't it ? |
| 18:22.11 | MikeB | justinu would exten = XXXXXXXXXX,1,dial(zap/g1/${EXTEN}) be quicker than exten = _X.,1,dial.... ? |
| 18:22.12 | harryvv | mmm |
| 18:22.27 | justinu | mike, not sure |
| 18:22.29 | harryvv | could not contact boot server using existing configuration on my ip500 display. |
| 18:22.36 | carrar | stoffell, there is nothing prohibiting that |
| 18:22.37 | [TK]D-Fender | ManxPower, harryvv : Just FYI that SIP 1.6.6 (out now) solves the presence limit and the 601 is now glorious :) |
| 18:22.38 | carrar | heh |
| 18:22.40 | *** join/#asterisk littlejohn (n=little@host208-13.pool8257.interbusiness.it) |
| 18:23.08 | stoffell | carrar, okay, it would even prove useful i think :) |
| 18:23.22 | justinu | chiardon: sorry, not right now |
| 18:23.29 | chiardon | OK |
| 18:23.46 | harryvv | justinu I guess one good way to confirm that it would be diverting the call to the other extension is just unplug this phone and the other extension should not ring. |
| 18:25.03 | pbx1 | Is there anybody there that can help? |
| 18:25.28 | stoffell | pbx1, approx 319 people, not including yourself :) |
| 18:25.36 | jpabuyer | hahaha |
| 18:25.59 | pbx1 | cool, are you up for it stoffell? |
| 18:26.11 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 18:26.12 | carrar | stoffell, -1 |
| 18:26.17 | carrar | he is not helping himself |
| 18:26.21 | pbx1 | heh |
| 18:26.26 | stoffell | lol carrar |
| 18:26.29 | *** join/#asterisk dyegoe (i=dyego@200.157.224.69) |
| 18:26.49 | stoffell | pbx1, i'm up for beer and women, but ask your question, maybe somebody can help you |
| 18:27.17 | pbx1 | if we meet up, you I can arrange the beer at least |
| 18:27.38 | pbx1 | ok, well I'm having problems with one-way audio on my queue calls |
| 18:27.50 | harryvv | yoru right it is the phone. The phone is disconected and its going right to vm and not being call forwarded to the other extension. I did make some minor changes to the xml phone1.cfg file and now for some reason the phone does not see the boot server. |
| 18:27.51 | pbx1 | agents can hear the caller, but not the other way around |
| 18:27.54 | stoffell | deal ;) only on queue calls? |
| 18:28.04 | pbx1 | yes, just queue calls |
| 18:28.30 | [TK]D-Fender | harryvv : Phone based forwarding only works when the phone is running :) |
| 18:28.32 | stoffell | can you dial the agents directly without a problem? |
| 18:28.35 | pbx1 | i have 2 agents taking calls, it works fine for 2 hours then all of a sudden all calls are one way |
| 18:28.50 | pbx1 | yes agents can be dialed no problem |
| 18:29.08 | justinu | polycom phones? |
| 18:29.10 | stoffell | pbx1.. ah, hmm... no firewalls or whatever in between? (and there;s 2-way audio *anytime* when calling the agents?) |
| 18:29.15 | harryvv | TK, thing is..ive never used call based call forwarding only asterisk based call forwarding. |
| 18:29.18 | pbx1 | no firewall |
| 18:29.44 | stoffell | pbx1, does it happen with calls from sip<->sip, or zap<->sip ? and what hardware ? |
| 18:29.45 | stack_ | I have a need for an ATA for our one manual fax because some people here still need it... any suggestions? |
| 18:30.01 | pbx1 | it's happning from sip provider to zap channels |
| 18:30.13 | [TK]D-Fender | harryvv : * based ensures that it happens... certain things don't work as expected on phones. Like DND does not get reflected in Presence properly, reject reasons may not feel natural, though somewhat indicitive. |
| 18:30.15 | jake1932 | an ata for fax? does that work? |
| 18:30.39 | websae | yes it works |
| 18:30.39 | justinu | it works if it's on a LAN, usually |
| 18:31.37 | stoffell | pbx1, what phones are you using? |
| 18:31.46 | harryvv | TK i see. BTW phone now cannot find the boot server. |
| 18:31.50 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 18:31.51 | pbx1 | analog phones |
| 18:31.55 | harryvv | vsftp is running |
| 18:32.20 | stoffell | harryvv, do they get an IP adres? (from your dhcp server) |
| 18:32.46 | stoffell | pbx1, and how do you 'fix'it ? restart asterisk ? |
| 18:33.11 | harryvv | stoff everythign works. No changes except to one xml file. |
| 18:33.28 | pbx1 | yeah restart |
| 18:33.31 | harryvv | everything is static |
| 18:33.32 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 18:33.35 | pbx1 | it fixes it for a while then starts again |
| 18:33.43 | MikeB | justinu, Think the delay in the index, thanks again. |
| 18:33.54 | stoffell | harryvv, i have the same problem when the dhcp server screws up.. |
| 18:34.00 | stoffell | pbx1, what * version |
| 18:34.07 | pbx1 | I have a te110p with a carrier access bank using g729 |
| 18:34.25 | pbx1 | asterisk head from last may 2005 |
| 18:34.26 | harryvv | stoffell I dont have a dhcp server |
| 18:34.50 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:35.14 | stoffell | harryvv, if i have problems, i erase my polycom's (4,6,8,*) and let them get all from the ftp again.. |
| 18:35.31 | pbx1 | do you think upgrading asterisk to lastest head could do it? |
| 18:35.33 | [TK]D-Fender | harryvv : VSFTP can be an annoying POS. Thats why I still prefer ProFTPD |
| 18:36.15 | stoffell | pbx1, could be. i'm not running head, but 1.2.7.1. but using latest head is worth a try (astridevcon just finished :p) |
| 18:36.31 | harryvv | So you are saying that this will erase the memory of the settings in the phone |
| 18:36.34 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 18:37.01 | stoffell | harryvv, yes, it will 'format' your phone, and leave you with a bootrom, eagerly searching for an ftp server with the sip firmware :) |
| 18:37.20 | harryvv | okay |
| 18:37.41 | pbx1 | any thing else I could try in the mean time? |
| 18:37.51 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) |
| 18:38.02 | harryvv | okay bot rom deleted |
| 18:38.05 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 18:38.08 | pbx1 | what usually causes one way audio? |
| 18:38.12 | TripleFFFFFFFFFF | NAT |
| 18:38.12 | harryvv | its restarting |
| 18:38.15 | stoffell | pbx1, yes, try setting logging and debug higher.. |
| 18:38.18 | [TK]D-Fender | harryvv : BootROM gone would be BAD :) |
| 18:38.22 | TripleFFFFFFFFFF | pbx1 nat |
| 18:38.27 | stoffell | lol [TK]D-Fender :) |
| 18:38.54 | pbx1 | yes I'm behind a nat, but I have a dedicated ip that port forwards to the asterisk behind the nat |
| 18:39.22 | pbx1 | I only get one way audio with agents on queue |
| 18:39.23 | jake1932 | which was is the audio going? |
| 18:39.28 | pbx1 | calls to ext are fine |
| 18:39.36 | TripleFFFFFFFFFF | hmm |
| 18:39.40 | jake1932 | calls to ext both ways are fine? |
| 18:39.41 | pbx1 | agents can hear the caller, callers cannot hear the agent |
| 18:39.45 | jake1932 | ok |
| 18:39.48 | TripleFFFFFFFFFF | no idea.. sip debug ook for 192.XX ec in headers |
| 18:39.58 | pbx1 | yes calls out to sip from zap is fine, calls in to zap ext is fine |
| 18:40.34 | jake1932 | the agents phone must not be sending to the right rtp port or there are forwarding issues |
| 18:40.48 | pbx1 | just with queues, and usually when both agents are on a call and more than 1 perso is on hold |
| 18:40.49 | jake1932 | does asterisk see the rtp traffic from the devices (rtp debug)? |
| 18:41.12 | pbx1 | how would I check the rtp traffic? |
| 18:41.20 | jake1932 | rtp debug |
| 18:41.41 | harryvv | man its been so long i cannot recall the ftp user/pass |
| 18:41.42 | harryvv | :) |
| 18:41.54 | pbx1 | ok trying that rtp debug |
| 18:42.08 | jake1932 | the 'sip debug' messages should show which port it's trying to send packets to |
| 18:42.42 | pbx1 | ok, I'll let you kow what I see soon. |
| 18:42.55 | pbx1 | whoa that's a lot of output |
| 18:43.16 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:43.17 | jake1932 | do you see traffic in both directions? |
| 18:43.55 | pbx1 | rtp port 19196 |
| 18:44.03 | pbx1 | sent and got |
| 18:45.47 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 18:46.14 | jake1932 | pbx1: is the got a public ip of an agent? |
| 18:46.37 | robin_sz | meep? |
| 18:46.46 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 18:47.04 | pbx1 | it's hard to tell what's going on with so much output |
| 18:48.07 | pbx1 | it's printing out Sent RTP packet to 168.215.247.75:19296 (type 18, seq 14941, ts 435352, len 20) |
| 18:48.10 | pbx1 | over and over |
| 18:48.19 | pbx1 | and my agent couldn't hear the caller |
| 18:48.34 | jake1932 | right because it's only sending packets |
| 18:48.48 | pbx1 | any suggestions? |
| 18:48.57 | jake1932 | your not getting any packets - is your firewall routing rtp correctly? |
| 18:49.10 | jake1932 | do you have any logging? |
| 18:49.20 | pbx1 | not on my firewall |
| 18:49.30 | jake1932 | also, is your phone setup properly for nat? |
| 18:49.37 | pbx1 | would it make sense that this only happens some times? |
| 18:49.49 | jake1932 | if your phone is sending a private addy, asterisk won't know how to get to it |
| 18:50.11 | jake1932 | asterisk needs a public address if your agent is not on the LAN |
| 18:50.31 | pbx1 | it's all on the lan |
| 18:50.34 | jake1932 | your can see messages from your phone by doing a 'sip debug' |
| 18:50.43 | pbx1 | it's a zap phone |
| 18:50.53 | harryvv | same thing could not contact boot server |
| 18:50.55 | pbx1 | connected to a channel bank |
| 18:51.09 | jake1932 | if your agent and asterisk are both on the LAN, there are not NAT issues. |
| 18:51.23 | harryvv | Well, back to square one |
| 18:51.25 | pbx1 | agent and asterisk are both on the lan |
| 18:51.31 | pbx1 | heh, thanks though |
| 18:51.43 | stoffell | pbx1, if the problem occurs, look at the rtp debug and maybe try a tcpdump, but also don't forget to update to latest head, maybe there's a bug in queues.. |
| 18:51.48 | pbx1 | I'm a bit of a noobie, but I've got it this far |
| 18:51.52 | pbx1 | thanks for dealing with me |
| 18:52.52 | jake1932 | if asterisk and your agent's phone are on the same lan, turn off nat everywhere and it should work |
| 18:53.26 | pbx1 | could it be my sip provider? |
| 18:53.46 | jake1932 | though you said you have a channel bank and zap |
| 18:53.50 | pbx1 | yes |
| 18:54.03 | jake1932 | how does the sip provider fit into all this? |
| 18:54.21 | harryvv | I wonder if kerbos is preventing the phone from logging in. I just tryed to login a ftp client on windows and got this when entering in only, the user. KERBEROS_V4 rejected as an authentication type |
| 18:55.03 | pbx1 | could the sip provider be sendin just one channel? is that possible? |
| 18:55.30 | jake1932 | what is the config? sip phone -> asterisk -> sip provider? |
| 18:55.35 | stoffell | harryvv, so your problem is probably just the ftp server ? |
| 18:55.57 | pbx1 | no sip phone |
| 18:56.26 | pbx1 | sip provider to my dsl modem -> asterisk box -> te110p -> carrier access bank -> analog phon |
| 18:56.44 | jake1932 | ok - channels bank for - fxs |
| 18:57.14 | jake1932 | does it work phone to phone? |
| 18:57.40 | harryvv | stoffell no, the entire problem is my call forwarding cannot be disabled. Been talking about it since late last night. Today, in the last 20 min since erasing the rom data of the phone and have it log into the ftp server is says it cannot upload the image it cannot find it. I have the ftp servers ip set correctly on the phone as well as the user/pass. But I dont know if these phones can log into a kerbos enabled ftp server but, it has f |
| 18:58.08 | pbx1 | I've had this set up for about a year. It's been working fine, phone to phone, phone to sip, sip to phone with very minimal calls |
| 18:58.29 | jake1932 | does it work now phone to phone? |
| 18:58.30 | pbx1 | recently, yesterday actually, I made a queue live with 2 agents |
| 18:58.34 | pbx1 | that is when the problems started |
| 18:58.47 | pbx1 | yes phone to phone is fine |
| 18:58.57 | stoffell | harryvv, if you can login with a standard linux or windows (dos) FTP client, the polycom should be able to log in also |
| 18:58.57 | pbx1 | ext 10 can call ext 12 and so on |
| 18:59.20 | jake1932 | does your sip provider support nat? |
| 18:59.23 | harryvv | for some reason i cannot. I get this when logging in with dos ftp client. |
| 18:59.25 | harryvv | 530 Please login with USER and PASS. |
| 18:59.25 | harryvv | KERBEROS_V4 rejected as an authentication type |
| 18:59.25 | harryvv | Name (192.168.10.2:root): phone001 |
| 18:59.25 | harryvv | 530 Permission denied. |
| 18:59.26 | harryvv | Login failed. |
| 18:59.36 | stoffell | what ftp server is it ? try disabling kerberos |
| 18:59.43 | harryvv | vsftpd |
| 18:59.43 | pbx1 | yes, sip provider supports it, it's been fine for a year |
| 18:59.58 | pbx1 | now that there are so many calls coming in on the queue, the problems have started |
| 19:00.03 | dlynes_ | harryvv: Did you get any success from last night? |
| 19:00.30 | dlynes_ | harryvv: i.e. for 192.168.10.40? |
| 19:00.32 | jake1932 | pbx1: can you pastebin a sip debug? |
| 19:00.44 | pbx1 | ok let me try to get that |
| 19:01.03 | pbx1 | this is all that is printing out though from what I see |
| 19:01.09 | pbx1 | Sent RTP packet to 168.215.247.75:16886 (type 18, seq 42889, ts 3726320, len 20) |
| 19:01.17 | pbx1 | I'll try to get a full debug too |
| 19:01.20 | harryvv | .40 is timming out. its not used |
| 19:01.25 | jake1932 | sip debug (not rtp debug) |
| 19:01.33 | pbx1 | opps sorry |
| 19:01.36 | dlynes_ | harryvv: if it wasn't being used, it wouldn't be sending you a response back |
| 19:03.20 | harryvv | I have 4 devises with four ips set. the gateway,firewall,ip500,ata. thay are .1 .2 .50 .30 you tell me. |
| 19:03.50 | harryvv | I dont know about that issue. |
| 19:04.10 | harryvv | Anyway this phone is taking to much time so i need to go. will work on it tonight. |
| 19:04.18 | stoffell | harryvv, i'd suggest focusing on your ftp problem, so you can use plain ol' ftp provisioning... |
| 19:05.19 | harryvv | yea |
| 19:05.36 | harryvv | and vsftpd has worked for well over a year now. phone would always retrive the files. |
| 19:07.51 | harryvv | I changed the chmod on the phone1.cfg so the edited changes I made could be made to the ftp server would occure now the phone does not want to log into it. |
| 19:10.09 | pbx1 | hey jake1932, I had the dsl modem reboot and the it started to work again |
| 19:10.27 | jake1932 | ah yes - the ole reboot |
| 19:10.35 | pbx1 | so you guys were right the first time |
| 19:10.41 | stoffell | harryvv, still sounds more like an ftp issue to me, if the phone can't retrieve the configs, if 's surely not good.. |
| 19:10.44 | pbx1 | it was nat, but I guess i have a crappy modem |
| 19:10.55 | pbx1 | thanks for leading me to the right place |
| 19:11.25 | pbx1 | so I'll keep monitoring it and make sure that's what it is. thanks again |
| 19:12.09 | *** join/#asterisk mog_home (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 19:12.09 | stoffell | pbx1, you've got a focus now, so that's good, good luck |
| 19:12.09 | pbx1 | thanks to you too stoffell |
| 19:12.21 | pbx1 | that debug stuff helps, but it's a nightmare to see come out, man |
| 19:12.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:12.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:12.56 | stoffell | nightmares are dreams coming true ? ;) |
| 19:13.00 | file | lame lame lame |
| 19:13.00 | pbx1 | hahaha |
| 19:16.44 | harryvv | okay got the ftp server to work right. |
| 19:17.28 | azeteg | What hardware is optimal for an asterisk server running about 300 connections? |
| 19:17.52 | [TK]D-Fender | azeteg : Depends on what kind of connections obviously. |
| 19:18.06 | harryvv | it did not work!!!! |
| 19:18.10 | harryvv | arggg |
| 19:18.33 | [TK]D-Fender | harryvv : ProFTPD ;) |
| 19:18.33 | azeteg | SIP clients |
| 19:18.34 | *** join/#asterisk docelmo (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 19:18.35 | jake1932 | transcoding? |
| 19:18.50 | stack_ | we run a call center and we would like a manager to turn on call monitoring for someone's extension and leave it on all day so that that person's calls can be reviewed... is there a way to do this? |
| 19:18.57 | harryvv | TK, i fixed the ftp issue. the ip500 is coping the files but the xml changes have no effect on the call forwarding issue. |
| 19:19.00 | dlynes_ | stack_: chanspy |
| 19:19.01 | harryvv | I need to go |
| 19:19.02 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 19:19.02 | azeteg | no transcoding |
| 19:19.11 | azeteg | just g711a |
| 19:19.14 | harryvv | will be back around 6 |
| 19:19.21 | dlynes_ | stack_: i think that's what it was, anyways |
| 19:19.36 | azeteg | just wondering what kind of processor etc I need |
| 19:19.39 | dlynes_ | stack_: if not, mixmonitor should work for sure |
| 19:19.43 | harryvv | take a look at this error in list.digium.com it will be in the next one. I submited it about 2 hours ago. |
| 19:20.04 | *** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.184) |
| 19:20.05 | dlynes_ | azeteg: how many clients? |
| 19:20.26 | azeteg | about 300 SIP clients |
| 19:20.32 | stack_ | dlynes_, I'd like to use MixMonitor, but I need turn it on only when the manager wants to and only on specific channels, we'd be monitoring both inbound and outbound calls |
| 19:20.36 | dlynes_ | azeteg: any pstn hardware? |
| 19:20.41 | azeteg | yeah |
| 19:21.05 | azeteg | looking at sangoma cards |
| 19:21.05 | *** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 19:21.31 | dlynes_ | azeteg: i'm pretty sure you can find some good statistics on such a configuration on the asterisk wiki then, under the performance heading, or something like that |
| 19:21.41 | azeteg | ok, I'll take a look |
| 19:22.29 | russellb | u/wi3 |
| 19:22.44 | justinu | May 11 12:18:05 WARNING[18847] chan_sip.c: Maximum retries exceeded on transmission STLMGC0120060511191751042895@xxx.xxx.xxx.xxx for seqno 102 (Non-critical Request) |
| 19:22.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:22.53 | justinu | why oh why are you ignoring my ACKs, level3? |
| 19:22.56 | dlynes_ | stack_: sounds like you might need to use an agi script |
| 19:23.21 | azeteg | any current machine sounds like a good bet |
| 19:23.42 | RoyK | anyone ever used asterfax? |
| 19:23.50 | dlynes_ | azeteg: well, i wouldn't try doing that on a Nehemiah C3 1GHz, personally :) |
| 19:23.51 | *** join/#asterisk moreece (n=moreece@s1-gw-223.dial-up.net) |
| 19:24.29 | dlynes_ | RoyK: what is it? |
| 19:25.29 | azeteg | my current lab asterisk server is celeron533 128MB and centos |
| 19:25.32 | azeteg | ;) |
| 19:25.37 | chiardon | asterfax was deprecated - -didn't it? |
| 19:26.16 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 19:27.02 | RoyK | dlynes_: email/fax gateway |
| 19:27.16 | RoyK | chiardon: their roadmap says 1.0 release in mid may |
| 19:27.48 | chiardon | really? . . .sorry |
| 19:27.52 | dlynes_ | RoyK: ah...yeah...i'd found a few others, and ultimately decided to do my own, when I realized they depended on applications I didn't have installed |
| 19:27.53 | *** join/#asterisk volkerli (n=volkerli@port-212-202-198-104.dynamic.qsc.de) |
| 19:28.26 | *** part/#asterisk volkerli (n=volkerli@port-212-202-198-104.dynamic.qsc.de) |
| 19:28.39 | RoyK | I'm kinda looking for a system that supports smtp auth so I can bill customers for outgoing faxes |
| 19:28.51 | gandhijee | hey the pages for polycom 601's have to be in XHTML, right? |
| 19:29.35 | chiardon | got pulse digit 14 on te 4/0/3/14 . . .what's that? |
| 19:29.50 | *** join/#asterisk ToTo (n=ToTo@host186-167.pool872.interbusiness.it) |
| 19:30.12 | *** join/#asterisk yo2lux (n=joe@82.79.81.6) |
| 19:30.28 | yo2lux | Hi |
| 19:30.35 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 19:31.19 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:31.48 | yo2lux | Exist an option in sip.conf to increase the sound quality ? |
| 19:32.19 | jake1932 | sort of - ulaw is probably the best you'll get |
| 19:32.45 | jake1932 | disallow=all |
| 19:32.48 | jake1932 | allow=ulaw |
| 19:32.50 | yo2lux | I use G.711 ulaw but the sound quality is not better than Google Talk. I hear a little noise in background |
| 19:33.22 | yo2lux | I have this issue in Intranet, the sound with Free World Dialup user is good. |
| 19:33.42 | yo2lux | both clients use x-lite softphone |
| 19:33.48 | yo2lux | any idea? |
| 19:34.23 | jake1932 | never had any luck with softphones |
| 19:34.33 | *** join/#asterisk adorah (n=FreePBX6@84.94.127.155.cable.012.net.il) |
| 19:34.41 | jake1932 | the hardphones quality was much better |
| 19:35.16 | yo2lux | thanks ! |
| 19:35.36 | *** join/#asterisk Malthus (n=herman@uslec-66-255-127-2.cust.uslec.net) |
| 19:35.56 | RoyK | jake1932, yo2lux: we have our customer support centre running with lots of staff using softphones, and they work better than some hardphones |
| 19:36.07 | azeteg | what ethernet cards would you guys recommend? |
| 19:36.09 | RoyK | not good hardphones, though, but they are usually expensive |
| 19:36.18 | Malthus | hi |
| 19:36.19 | RoyK | azeteg: anything that works, until traffic is high |
| 19:36.42 | Malthus | the dtmf on my t1 zap channels are screwed |
| 19:36.47 | jake1932 | ok - i was going to say - the call quality i get is pretty good with a hardphone - but it's a little expensive |
| 19:36.50 | Malthus | (incoming) |
| 19:36.51 | azeteg | I've used intel pro1000/mt before |
| 19:36.59 | yo2lux | exist another option to set voice quality in sip.conf ? not the codec.. |
| 19:37.06 | Malthus | they work for a while, then asterisk starts to ignore them |
| 19:37.17 | xai | ManxPower: how's work and life? |
| 19:37.21 | gandhijee | the 3c905B is still a reliable card |
| 19:37.22 | RoyK | azeteg: e1000 is one of my preferred choices |
| 19:37.28 | gandhijee | if u can still find them |
| 19:37.32 | Malthus | both an e&m t1 line and a channel bank |
| 19:37.38 | jake1932 | makevoicequalitygood=yes |
| 19:37.44 | RoyK | azeteg: since intel writes the driver and it's open in kernel, and it works |
| 19:37.49 | azeteg | yeah |
| 19:38.08 | azeteg | I use two e1000 in an openbsd bridge |
| 19:38.24 | azeteg | fw/shaper |
| 19:38.29 | azeteg | no problems yet |
| 19:38.42 | RoyK | broadcoms are good as well |
| 19:38.52 | RoyK | realteks are not |
| 19:42.35 | znoG | do a lot of you use gigabit in your networks? |
| 19:42.40 | znoG | we're still at 100 :( |
| 19:43.23 | RoyK | onboard nVidia NICs are the spawn of satan and other various bad characters |
| 19:44.04 | *** join/#asterisk clive- (n=pirch@dsl-145-2-216.telkomadsl.co.za) |
| 19:45.15 | znoG | anyone manage a lucent PBX from Linux? |
| 19:45.31 | znoG | just wondering whether there is an alternative to the ECS software for Linux |
| 19:45.52 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 19:46.55 | yo2lux | azeteg I use asterisk on openbsd |
| 19:47.12 | gandhijee | do webpagesfor the polycom 601 have to be in XHTML? |
| 19:49.50 | *** join/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 19:50.50 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:51.01 | RoyK | znoG: Can't you just hook it up to asterisk with some PRIs? |
| 19:51.01 | qseek | hello all |
| 19:52.04 | *** part/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 19:53.17 | yo2lux | the sound quality depend by sound card ? |
| 19:55.51 | RoyK | yo2lux: no |
| 19:56.29 | znoG | RoyK: that wouldn't fix the fact that I want to admin it from Linux :) |
| 19:57.27 | RoyK | nah.... |
| 19:57.57 | RoyK | it's probably sufficient to purchase an API from them for a few hundred k |
| 19:57.59 | RoyK | :P |
| 20:00.37 | sevard | does anyone use the queue-out feature in queues.conf and actually have it work? |
| 20:01.25 | [TK]D-Fender | gandhijee : Yes, XHTML |
| 20:01.49 | [TK]D-Fender | sevard : Queue-out? |
| 20:02.50 | sevard | queue-out is a feature in queues.conf and it works like this, when a user is in queue and wants to screw holding and go to voicemail, you press a digit and it kicks you to the queue-out context |
| 20:03.00 | sevard | supposed to be simple, but i set the context and it doesn't work |
| 20:03.55 | justinu | asterisk seems to have some kind of issue with it's scheduler now and then |
| 20:04.21 | justinu | everyonce and a while, it sends out a bunch of 200 OKs in rapid succession, without waiting for the replies |
| 20:06.00 | *** join/#asterisk Cresl1n (n=matt@host56-92.pool8838.interbusiness.it) |
| 20:06.03 | [TK]D-Fender | sevard : Yeah, I do that. |
| 20:06.03 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 20:06.29 | sevard | [TK]D-Fender: yeah it doesn't work |
| 20:06.45 | [TK]D-Fender | sevard : It works *BUT* I believe I've discovered a bug that while an announcement is playing, the first DTMF aborts it, goes back to MoH, THEN your DTMF will cause an exit like expected. |
| 20:06.53 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 20:07.08 | [TK]D-Fender | So you'd have to press twice |
| 20:07.39 | sevard | [TK]D-Fender: for a while I was noticing it would dial my agents over and over if i pressed any DTMF, now it doesn't do jack |
| 20:07.59 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.40) |
| 20:08.02 | [TK]D-Fender | sev, pastebin your queue def and the exit context |
| 20:08.34 | DarKnesS_WolF | hi [TK]D-Fender |
| 20:08.44 | DarKnesS_WolF | but from home :P |
| 20:08.56 | [TK]D-Fender | DarKnesS_WolF : y0 |
| 20:09.24 | DarKnesS_WolF | i have a very nasty nating problem seems i have to install STUN server :(.... i'm behind a nat trying to call the asterisk box .. it has 2 NIC one private to the phones and on real and i can't reach the phones. |
| 20:09.34 | sevard | [TK]D-Fender: http://pastebin.ca/55184 |
| 20:09.49 | Malthus | is there any way to see stats on a digium t1 card? |
| 20:10.00 | Malthus | errors, frame slips |
| 20:10.06 | Malthus | that kind of thing |
| 20:11.48 | justinu | heh, no |
| 20:12.12 | Malthus | great |
| 20:12.33 | mog_home | frame slips are logged in /var/log/asterisk/messages |
| 20:12.35 | Malthus | asterisk keeps on ignoring dtmf on incoming zap channels after a while |
| 20:12.38 | mog_home | along with everything |
| 20:13.05 | Malthus | and its also missing digits on incoming dnis sometimes |
| 20:13.44 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 20:13.58 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 20:14.32 | FlyboySR22 | Quick question for everyone that is a VOIP provider using Asterisk...What are you using for a billing package...? |
| 20:14.52 | [TK]D-Fender | sevard : and the exit context from extensions.conf? |
| 20:14.57 | FlyboySR22 | Seem to be a lot of options, just looking for suggestions as to which one works best for people. |
| 20:15.54 | Malthus | is 'file.c: failed to write frame' a t1 frame slip message? |
| 20:16.09 | Malthus | or is that a filesystem thing |
| 20:16.19 | sevard | [TK]D-Fender: look at the snipplet below the queues.conf, it's on the same pastebin page. |
| 20:16.42 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 20:16.46 | Assid | heya |
| 20:17.06 | justinu | May 11 12:45:26 DEBUG[18840] channel.c: Avoiding initial deadlock for 'SIP/peer-b720e828' |
| 20:17.10 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 20:17.22 | justinu | everytime I see that, asterisk freaks out and sends a bunch of retransmitted SIP responses |
| 20:17.29 | justinu | what the hell is it's problem? |
| 20:17.30 | *** join/#asterisk distortion (i=distorti@junipero.3sheep.com) |
| 20:17.46 | sevard | justinu: you say "it's" a lot when you mean its |
| 20:18.17 | [TK]D-Fender | sevard : no "s" ! Only DTMF! |
| 20:18.18 | justinu | i didn't realize you were my grade school grammar teacher |
| 20:18.25 | justinu | wtf? |
| 20:18.32 | [TK]D-Fender | sevard : Single digit |
| 20:18.48 | sevard | [TK]D-Fender: say what?! |
| 20:18.52 | Assid | man.. playing xbox can be dangerous for health] |
| 20:19.03 | sevard | [TK]D-Fender: OH OH |
| 20:20.47 | sevard | [TK]D-Fender: SWEET |
| 20:21.10 | justinu | anyone gonna help, or just correct my spelling? |
| 20:22.41 | jpabuyer | "going to" |
| 20:22.52 | mercestes | actually, your spelling was correct, sevard was creaging your grammar. |
| 20:22.59 | mercestes | correcting... |
| 20:23.06 | mercestes | gah...that's what I get for reading and typing at the same time.:P |
| 20:23.26 | justinu | useless |
| 20:24.14 | *** join/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 20:24.22 | chino[server] | which voip provider do you guys suggest ? |
| 20:27.37 | *** join/#asterisk woolbeo (n=woolbeo@toby.stoneflytech.com) |
| 20:28.50 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 20:29.15 | cytrak | is there a way to reload only the extension.conf file without doing a full reload ? |
| 20:29.45 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:30.06 | russellb | cytrak: extenstions reload |
| 20:30.12 | russellb | extensions reload |
| 20:31.13 | cytrak | thanks |
| 20:32.50 | [TK]D-Fender | sevard : So, got it figured out and functional now? :) |
| 20:34.34 | distortion | russellb: you have any advice on how to handle codec negotiation, it looks like theres a couple patches and a branch by oej |
| 20:35.04 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:35.56 | *** join/#asterisk mogorman (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 20:37.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:37.33 | sevard | [TK]D-Fender: word brother, i thought it kicked any digit pressed to that context, i didnt' think it carried over the digit |
| 20:38.07 | sevard | which i guess is much nicer |
| 20:38.48 | sevard | i just wish the documentation project was a bit more indepth :| |
| 20:38.55 | woolbeo | anyone had the g# option for voicemail crash *? |
| 20:39.25 | Ariel_ | hello I have a quick question about fixing an inbound callerID problem. I need to get the caller ID from a sip account that when I get the call it comes with a callerID as sip/4151234567-4eooug I need to cut the end off and the sip/ off then put he callID back correctly. |
| 20:39.27 | CunningPike | woolbeo: Sure hope not - use it all the time |
| 20:39.34 | *** join/#asterisk tuxd00d (n=tuxinato@69-169-11-49.lmdaca.adelphia.net) |
| 20:39.49 | [TK]D-Fender | sevard : Wiki page seemed 100% clear to me :) |
| 20:40.00 | CunningPike | Ariel_: Sounds like you're getting the channel? |
| 20:40.06 | woolbeo | CunningPike, I just tried to use it on my phonesystem at my office, and it crashes * hard... |
| 20:40.07 | sevard | Ariel_: pipe it out to a perl script, do some regular expression kung fu, pipe it back |
| 20:40.12 | CunningPike | woolbeo: Ouch |
| 20:40.19 | CunningPike | woolbeo: What version? |
| 20:40.25 | Ariel_ | CunningPike, yes it's comes in via a sip provider that just sends the call to an ip address. |
| 20:40.41 | Ariel_ | sevard, I am not a programer |
| 20:40.46 | woolbeo | CunningPike, 1.2.4 |
| 20:40.48 | CunningPike | Ariel_: Ah - use ${CUT} |
| 20:41.01 | sevard | Ariel_: neither am i, regular expressions are easy though |
| 20:41.07 | Ariel_ | I was trying to make a macro and use cut but I read the wiki and it made me really nuts. |
| 20:41.33 | sevard | Ariel_: wiki-inspired insanity, welcome to the club :) |
| 20:41.34 | CunningPike | Ariel_: I parse extension from channel all the time - I'll pm you what we do |
| 20:41.35 | woolbeo | CunningPike, Its an old RH9 system, we have a new server with 1.2.7 setup on. We're going to switch over any day now... |
| 20:41.47 | Ariel_ | CunningPike, great thanks |
| 20:42.43 | *** join/#asterisk ph|ber (i=phiber@slackwaresupport.com) |
| 20:42.43 | SplasPood | I'm getting the following error, and CONGESTION randomly when making outgoing calls terminated with one of my SIP providers (all others are fine). 2128124190 is my CID for that call... Seems to me like this might be a misconfiguration on my side.. anyone have any thoughts? May 11 15:46:13 WARNING[9799] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"2128124190" <sip:2128124190@voice1.lga1.us.voxel.net>;tag=as3a422c15' |
| 20:43.16 | ph|ber | can someone help me with a sangomas card. |
| 20:43.22 | ph|ber | May 11 16:40:27 ERROR[27965] chan_zap.c: Signalling requested on channel 1 is PRI Signalling but line is in FXS Kewlstart signalling |
| 20:43.25 | ph|ber | May 11 16:40:27 ERROR[27965] chan_zap.c: Unable to register channel '1-23' |
| 20:43.28 | ph|ber | May 11 16:40:27 WARNING[27965] loader.c: chan_zap.so: load_module failed, returning -1 |
| 20:43.37 | Zodiacal | anyone know if asterisk has events.. e.g. i wanta run somthing when someone parks a call |
| 20:43.49 | [TK]D-Fender | ph|ber : thats not a sangoma problem, you set your modes to a mismatch between zapata.conf and zaptel.conf. |
| 20:44.08 | ph|ber | modes mismatch? |
| 20:44.31 | [TK]D-Fender | ph|ber : 1 files says its an analog line, and the other says its a Digital PRI circuit. |
| 20:44.55 | dlynes_ | ph|ber: one is declaring a pri, the other is declaring an fxs kewlstart |
| 20:45.06 | justinu | brilliant: May 11 13:36:14 DEBUG[18847] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 32 ms (t1 16 ms (Retrans id #1199242)) |
| 20:45.08 | ph|ber | k. so..... where would that be? |
| 20:45.15 | CunningPike | Hey, dlynes_ |
| 20:45.17 | justinu | waiting a whole 32ms for a response... |
| 20:45.20 | dlynes_ | Hey cp |
| 20:45.32 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 20:45.39 | dlynes_ | CunningPike: I get to play with the perils of h323 |
| 20:45.40 | dlynes_ | yay! |
| 20:45.48 | CunningPike | Just fixed a lovely memory leak on our * box |
| 20:45.52 | dlynes_ | cool |
| 20:45.56 | CunningPike | h232, eh? Have fun :D |
| 20:46.08 | dlynes_ | Yeha....pwlib is a huge POS |
| 20:46.26 | ph|ber | so that would be the signalling? |
| 20:46.26 | dlynes_ | I don't understand why i need openldap, guilib, .... all just to do h323 |
| 20:46.30 | CunningPike | dlynes_: Remember I said that we kept our voicemail on a samba-mounted Windows share? |
| 20:46.36 | dlynes_ | CunningPike: yeah? |
| 20:46.48 | CunningPike | dlynes_: Because the ITU invented it |
| 20:47.11 | cytrak | anyone here using idefisk ? |
| 20:47.17 | *** part/#asterisk woolbeo (n=woolbeo@toby.stoneflytech.com) |
| 20:47.19 | CunningPike | dlynes_: Well, don't - I think smbfs leaks like crazy due to the file locking |
| 20:48.05 | dlynes_ | CunningPike: i'm not planning to use it for voicemail |
| 20:48.12 | *** part/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 20:48.13 | dlynes_ | CunningPike: only moh and autoattendant files |
| 20:48.19 | dlynes_ | CunningPike: so basically 2 files |
| 20:48.24 | cytrak | I'm setting up my users with idefisk soon and I've been using it with a small group of testers .. I'm kind of wondering why when I make outcalls my history doesn't show the called number but instead it shows my callerID !!! |
| 20:48.36 | CunningPike | dlynes_: That'll be fine - we tested that and it works fine - only voicemail causes the leaks |
| 20:48.43 | dlynes_ | CunningPike: ah |
| 20:48.58 | dlynes_ | CunningPike: i have no reason to put voicemail on a fileshare |
| 20:48.59 | CunningPike | dlynes_: So, we're using nfs to a linux server instead |
| 20:49.09 | dlynes_ | CunningPike: Can't see why you would need voicemail on a fileshare, either :) |
| 20:49.26 | dlynes_ | CunningPike: why do the end users need to be able to read the voicemail tree? |
| 20:49.27 | CunningPike | dlynes_: We do - it's so our redundant hot standby server can serve up voicemail |
| 20:49.45 | CunningPike | dlynes_: Both servers point to the same share |
| 20:49.50 | dlynes_ | CunningPike: ah...I would just use nfs or something |
| 20:49.58 | CunningPike | dlynes_: We are now........ ;) |
| 20:50.01 | dlynes_ | CunningPike: then you don't have to worry about clueless windows users screwing with it |
| 20:50.13 | CunningPike | dlynes_: Is there any other kind? |
| 20:50.19 | dlynes_ | don't think so :) |
| 20:50.42 | ph|ber | signalling = pri_cpe |
| 20:50.48 | ph|ber | thats my signal. |
| 20:51.28 | dlynes_ | CunningPike: mind if i open a /msg window with you for a sec? |
| 20:51.45 | dlynes_ | It's regarding some possible work |
| 20:51.54 | [TK]D-Fender | sure |
| 20:52.26 | CunningPike | dlynes_: Certainly :D |
| 20:53.45 | Assid | is there a way to reduce the time before it jumps to the next priority ? |
| 20:54.21 | ph|ber | i only have a zapata.conf |
| 20:54.23 | ph|ber | no zaptel.conf |
| 20:54.39 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 20:54.40 | sevard | dang |
| 20:54.51 | sevard | i need better hold music than this stock stuff |
| 20:55.03 | Dr-Linux | Hi all |
| 20:55.45 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 20:55.49 | ph|ber | can anyone help me??? |
| 20:56.17 | CunningPike | ph|ber: What's the problem? |
| 20:57.48 | ph|ber | May 11 16:56:23 ERROR[30820] chan_zap.c: Unable to register channel '1-23' |
| 20:57.48 | ph|ber | May 11 16:56:23 WARNING[30820] loader.c: chan_zap.so: load_module failed, returning -1 |
| 20:58.04 | Assid | sometimes.. it doesnt roll over to the next priority in time for me to jump to the next provider |
| 20:58.20 | Assid | like sometimes insufficient funds in 1.. and then it doesnt roll over |
| 20:58.56 | *** join/#asterisk nowork (i=quickmon@216.254.141.97) |
| 20:58.58 | ph|ber | CunningPike: any ideas? |
| 20:59.11 | sevard | if anyone wants free moh wget --mirror http://www2.signate.com/free-moh/ |
| 21:00.56 | nowork | hi, my redhat server time keep going slower and slower, do you know any webpaye/website has instruction of the procedure to do time synchronization?thanks |
| 21:01.03 | [TK]D-Fender | ph|ber : YOU AREN"T LISTENING. |
| 21:01.11 | ph|ber | i did listen. |
| 21:01.16 | ph|ber | i do not have a zaptel.conf |
| 21:01.20 | ph|ber | just zapata |
| 21:01.41 | [TK]D-Fender | ph|ber : In zapata.conf you are telling * your card is one kind of thing and in zaptel.conf you are telling * its csomthing COMPLETELY DIFFERENT. Not compatible! |
| 21:01.58 | [TK]D-Fender | ph|ber : Yes you do. Its in /etc/ |
| 21:02.07 | [TK]D-Fender | ph|ber : You seriously need to read THEBOOK. |
| 21:02.10 | distortion | nowork this isnt #linuxhelp please ask those kinds of questions there, but just cause im nice: http://www.siliconvalleyccie.com/linux-hn/ntp.htm |
| 21:02.12 | [TK]D-Fender | !thebook |
| 21:02.25 | Dr-Linux | nowork: hhm.. i had same problem with my red hat, i made a script for synchronization, but i'm at home right now. |
| 21:02.27 | [TK]D-Fender | ~THEBOOK |
| 21:02.28 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 21:02.48 | Dr-Linux | nowork: it looks like your hardware clock issue |
| 21:03.02 | *** join/#asterisk razu (n=razu@80-235-90-146-dsl.prn.estpak.ee) |
| 21:03.07 | b00mer_ | nowork: is the machine really old? |
| 21:03.08 | ph|ber | [TK]D-Fender: i have no zaptel.conf in /etc either. |
| 21:03.21 | ph|ber | er scratch that. |
| 21:03.22 | ph|ber | sorry |
| 21:03.23 | nowork | not old ..it's Dell |
| 21:03.23 | b00mer_ | nowork: I had some really bad clock drift on older machines... |
| 21:03.37 | ph|ber | fxsks=1 |
| 21:03.37 | ph|ber | loadzone = us |
| 21:03.37 | ph|ber | defaultzone = us |
| 21:03.44 | b00mer_ | it's dell :) cause dell just started making machines last week |
| 21:03.50 | nowork | :) |
| 21:03.52 | *** join/#asterisk raggi (n=ra66i@host-84-9-127-182.bulldogdsl.com) |
| 21:04.12 | b00mer_ | like over 5 years old? |
| 21:04.39 | b00mer_ | I would install ntp if your concerned |
| 21:04.43 | ph|ber | http://pastebin.com/712297 |
| 21:04.46 | ph|ber | zapata.conf |
| 21:04.59 | ph|ber | to some extent.. |
| 21:05.04 | raggi | can anyone tell me if it's possible to add a number to the end of a varialbe in a Dial() configuration, eg exten=> _nxxxx,2,Dial(${EXTEN}#) |
| 21:05.58 | distortion | raggi: try using Set? |
| 21:06.56 | distortion | exten => _nxxxx,2,Set(${EXTEN}#) |
| 21:06.56 | glm2k | raggi: maybe you want this? exten=> _nxxxx,2,Dial(${EXTEN}${MYNUM}#) |
| 21:07.12 | CunningPike | ph|ber: What hardware do you have in your server? FSX, FXO, PRI? |
| 21:07.20 | raggi | will try, thx glm2k & distortion |
| 21:07.24 | ph|ber | its PRI lines |
| 21:07.27 | ph|ber | a sagnoma |
| 21:07.31 | distortion | exten => _nxxxx,2,Set(${EXTEN}=${EXTEN}#) rather |
| 21:07.46 | distortion | err, somthing close to that |
| 21:07.47 | ph|ber | er |
| 21:07.49 | ph|ber | sangoma |
| 21:09.49 | *** join/#asterisk p1p (i=p1p@mail.comp911.com) |
| 21:11.09 | raggi | yeah, that's got us on the right track i think :) |
| 21:14.39 | Assid | anyone her using sip broker? |
| 21:19.08 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 21:19.21 | *** part/#asterisk dyegoe (i=dyego@200.157.224.69) |
| 21:20.50 | [TK]D-Fender | ph|ber : your zaptel.conf line is set up as if you had a single ANALOG phone line. Your zapata.conf is set up as though you has 2 PRI ports. |
| 21:21.11 | [TK]D-Fender | ph|ber : And NEITHER file is "complete" for the kind of technology they are talking about. |
| 21:23.29 | dlynes_ | Hrm...Guess CunningPike is having connection problems or something |
| 21:23.44 | *** join/#asterisk lorinc (n=ang@caracas-3845.adsl.interware.hu) |
| 21:24.37 | *** join/#asterisk seb- (n=seb@cpe-72-132-240-243.san.res.rr.com) |
| 21:24.53 | seb- | can someone please recommend a good headset/microphone that works with linux? |
| 21:25.30 | dlynes_ | seb-: you could always use a decent bluetooth headset with a bluetooth usb dongle |
| 21:27.07 | seb- | dlynes: thanks....1. i'm not sure usb drivers exist 2. iirc those are like multi-$100s 3. i was hoping for a sub-$100 solution ? |
| 21:27.21 | Hmmhesays | shake that ass for me, shake that ass for me |
| 21:27.22 | seb- | dlynes: possible? |
| 21:29.26 | dlynes_ | seb-: my usb dongle cost me $10Cdn, and my bluetooth headset cost me $35Cdn |
| 21:29.32 | dlynes_ | seb-: that's multi $100s? |
| 21:30.14 | dlynes_ | seb-: if you're using a recent kernel version, there's support for bluetooth usb devices |
| 21:30.45 | seb- | dlynes: how do Cdn (Canadian dollars?) translate to US$? |
| 21:30.53 | dlynes_ | seb-: now, or the end of the summer? |
| 21:31.01 | Qwell[] | seb-: about $500csn-$1usd |
| 21:31.02 | seb- | now :) |
| 21:31.02 | dlynes_ | seb-: the end of the summer they'll probably be equal :) |
| 21:31.04 | Qwell[] | cdn* |
| 21:31.15 | dlynes_ | seb-: because the usd is so weak right now |
| 21:31.44 | seb- | weak? |
| 21:32.13 | seb- | oh usd..ok |
| 21:32.25 | [TK]D-Fender | Blame Bush :) |
| 21:32.43 | many | 1 USD ~ 0.78EUR |
| 21:32.48 | many | heh. :) |
| 21:32.51 | dlynes_ | seb-: atm, $1USD = $1.10Cdn |
| 21:33.04 | dlynes_ | seb-: two years ago it was more like $1.26Cdn |
| 21:33.11 | dlynes_ | Blame Bush! |
| 21:33.23 | dlynes_ | That should be a new south park slogan :) |
| 21:33.53 | sevard | does anyone have any issues with a SIP 2002 not getting CID name? when the sipura calls the hop-1002 it passes cid name but the other way around the ata only sees cid number |
| 21:34.20 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 21:38.08 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) [NETSPLIT VICTIM] |
| 21:38.08 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) [NETSPLIT VICTIM] |
| 21:43.40 | *** join/#asterisk saftsack (n=saftsack@p54A7E08B.dip.t-dialin.net) |
| 21:45.54 | TripleFFFFFFFFFF | can one please try to call (800) 854-8957 i wanna see if my fax woroking |
| 21:45.55 | TripleFFFFFFFFFF | please |
| 21:46.00 | TripleFFFFFFFFFF | from usa |
| 21:46.39 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 21:46.43 | gandhijee | u want a fax along w/ that? |
| 21:46.52 | nowork | dlynes, two years ago was 2004 1.3 as i remember, 2003 is 1.45 |
| 21:46.53 | TripleFFFFFFFFFF | bah if you want |
| 21:46.58 | TripleFFFFFFFFFF | but i just want a tone |
| 21:47.21 | s34n | is there a way for a meetme conference to dial out? |
| 21:47.36 | dlynes_ | nowork: 1.45? I think you're dreaming :) |
| 21:47.49 | dlynes_ | nowork: 1.3, maybe...but definitely not 1.45 |
| 21:48.19 | gandhijee | i just called it |
| 21:48.27 | TripleFFFFFFFFFF | whathappeend |
| 21:48.33 | TripleFFFFFFFFFF | seems i get 404 |
| 21:48.34 | TripleFFFFFFFFFF | hmmm |
| 21:48.38 | TripleFFFFFFFFFF | SIP/2.0 404 Not Found |
| 21:49.06 | *** join/#asterisk mog_home (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 21:49.21 | TripleFFFFFFFFFF | you cal that 800 ? |
| 21:49.25 | gandhijee | yeah |
| 21:49.30 | gandhijee | now i get a busy |
| 21:50.20 | TripleFFFFFFFFFF | can you redial |
| 21:50.21 | TripleFFFFFFFFFF | ;) |
| 21:50.27 | TripleFFFFFFFFFF | just moded upsting |
| 21:50.29 | TripleFFFFFFFFFF | darn |
| 21:50.59 | TripleFFFFFFFFFF | ah |
| 21:51.00 | gandhijee | now its fast busy |
| 21:51.00 | s34n | how can a user have * call a third party, then connect them to a meetme? |
| 21:51.03 | dlynes_ | nowork: woah...holy crap...it was actually up to 1.575 in that calendar year |
| 21:51.04 | TripleFFFFFFFFFF | i see you |
| 21:51.13 | gandhijee | =þ |
| 21:51.31 | TripleFFFFFFFFFF | you got a dial tone now ? |
| 21:51.56 | gandhijee | no, still fast busy |
| 21:52.02 | TripleFFFFFFFFFF | shit |
| 21:52.06 | nowork | dlynes: FYI 1.5x http://www.bankofcanada.ca/cgi-bin/famecgi_fdps |
| 21:52.06 | TripleFFFFFFFFFF | its sendinf to s@ |
| 21:52.18 | gandhijee | anyone ever get 2 blocks after every href line in any of the polycom pages? |
| 21:52.30 | TripleFFFFFFFFFF | let me add dnis to the sip |
| 21:52.40 | s34n | twisted[asteria]: ? |
| 21:52.40 | dlynes_ | nowork: No cgi or cgidebug field was detected in the args |
| 21:52.41 | dlynes_ | passed to famecgi (either by form submission or |
| 21:52.41 | dlynes_ | query-string). Check your html form or href URL |
| 21:52.41 | dlynes_ | and try again. |
| 21:52.41 | nowork | dlynes: begining of yr 2003, 1.5x |
| 21:52.49 | TripleFFFFFFFFFF | how bout now ? |
| 21:53.01 | dlynes_ | nowork: anyways...i just went to sauder school of business |
| 21:53.12 | dlynes_ | nowork: it was 1.575 January 1, 2003 |
| 21:53.14 | nowork | dlynes:http://www.bankofcanada.ca/en/rates/exchange-look.html |
| 21:53.36 | gandhijee | slow busy now |
| 21:53.42 | TripleFFFFFFFFFF | hey |
| 21:53.50 | TripleFFFFFFFFFF | you guys are 2 doing at once |
| 21:53.54 | ph|ber | anyone have a zaptel.conf for a sangoma a102? |
| 21:53.56 | dlynes_ | nowork: and as high as 1.61 in 2002 |
| 21:54.10 | TripleFFFFFFFFFF | ok |
| 21:54.11 | TripleFFFFFFFFFF | i see |
| 21:54.14 | TripleFFFFFFFFFF | o application 'ODBCget' for |
| 21:54.15 | dlynes_ | ph|ber: should be the same as a te110p |
| 21:54.16 | TripleFFFFFFFFFF | hmm |
| 21:54.21 | TripleFFFFFFFFFF | i need odbc brbr |
| 21:54.22 | nowork | dlynes, i convered lot of from line of credit at 1.59:( to USD |
| 21:54.24 | gandhijee | ph|ber: you setup the wanpipe stuff yet? |
| 21:54.27 | dlynes_ | ph|ber: erm te210p i mean |
| 21:54.40 | dlynes_ | nowork: ah |
| 21:54.50 | gandhijee | after that, you just treat them like zap channels |
| 21:54.51 | dlynes_ | nowork: oh...you're in BC? |
| 21:54.58 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 21:55.10 | nowork | dlynes:in ON. |
| 21:55.15 | dlynes_ | nowork: Ah...you just seem to be on a Group telecom IP address |
| 21:55.23 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) [NETSPLIT VICTIM] |
| 21:55.23 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) [NETSPLIT VICTIM] |
| 21:55.52 | De_Mon | where can I find documentation on func_odbc? Specifically, using it with a multirow result set in the dialplan |
| 21:57.02 | dlynes_ | ah...i guess primus must be a reseller of either bell or group telecom |
| 21:57.46 | TripleFFFFFFFFFF | retry |
| 21:57.48 | TripleFFFFFFFFFF | ;) |
| 21:59.02 | TripleFFFFFFFFFF | fast busy i assume |
| 21:59.11 | TripleFFFFFFFFFF | man.. my provider is fucked |
| 21:59.55 | nahirean | you using nufone or something? |
| 21:59.59 | TripleFFFFFFFFFF | no |
| 22:00.09 | TripleFFFFFFFFFF | hol d on |
| 22:00.11 | TripleFFFFFFFFFF | let me try to send to a SIP instea of fax |
| 22:00.26 | TripleFFFFFFFFFF | now try |
| 22:01.12 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 22:01.37 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 22:02.06 | TripleFFFFFFFFFF | again ;) |
| 22:02.11 | TripleFFFFFFFFFF | sorry to be a pain |
| 22:02.14 | MRH2 | hi using current 1.2svn installation of zaptel completely rewrites modprobe.conf (removing all the non zaptel stuff along with it) |
| 22:02.16 | TripleFFFFFFFFFF | i just cant dial that sucker from ehre |
| 22:02.39 | TripleFFFFFFFFFF | unless.. |
| 22:02.44 | MRH2 | must be something that changed in the last week |
| 22:03.44 | *** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 22:04.01 | docelm0 | w00t! |
| 22:04.39 | docelm0 | or not |
| 22:05.10 | TripleFFFFFFFFFF | ok this makes no sense |
| 22:06.21 | TripleFFFFFFFFFF | ok taht rang |
| 22:06.24 | TripleFFFFFFFFFF | my cisco |
| 22:06.31 | TripleFFFFFFFFFF | now i push to fax lets see... |
| 22:06.36 | TripleFFFFFFFFFF | coz my other number got a tone |
| 22:07.15 | *** part/#asterisk nowork (i=quickmon@216.254.141.97) |
| 22:07.22 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 22:10.02 | TripleFFFFFFFFFF | ok one last time |
| 22:10.08 | TripleFFFFFFFFFF | then im changing providers lol |
| 22:11.22 | TripleFFFFFFFFFF | i see its coming to s: |
| 22:11.29 | TripleFFFFFFFFFF | To: <sip:s@209.17XXX |
| 22:11.44 | TripleFFFFFFFFFF | oh |
| 22:14.52 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.254) |
| 22:15.46 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 22:16.16 | KranZ | why are conference phones so damn expensive |
| 22:17.12 | Assid | cause you can conference them ;) |
| 22:17.30 | KranZ | meh |
| 22:17.42 | KranZ | anyone find some for <$500? |
| 22:19.03 | MRH2 | KranZ polycom Ip 500 works ok if ur gathered around it |
| 22:20.10 | KranZ | heh |
| 22:20.13 | KranZ | like around a fire? |
| 22:20.13 | websae | what's a good multi line business phone...i know polycom 301 is good, but it only has 2 line functionality...any other suggestions? |
| 22:20.26 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:20.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:20.48 | KranZ | websae, the spa-941 does 2 extensions out of box, but you can put more lines on the same extension |
| 22:20.55 | TripleFFFFFFFFFF | ok i think its all good now |
| 22:20.59 | TripleFFFFFFFFFF | Redial ;) |
| 22:21.06 | KranZ | so configure it for 1 extension, and enable the other 3 "lines" to use that extension |
| 22:21.16 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
| 22:21.16 | TripleFFFFFFFFFF | and tell me if it worked since i dont jknoiw |
| 22:21.28 | websae | KranZ: i had the SPA 841---was terrible....is the 941 better? |
| 22:21.49 | KranZ | i havent had any problems |
| 22:21.57 | websae | how's the speaker phone? |
| 22:22.00 | KranZ | the 841 was pre cisco+linksys+sipura |
| 22:22.11 | KranZ | works ok |
| 22:22.21 | KranZ | what problems did you have before? |
| 22:22.22 | achandra | hello...generic question when using pcap_play in sipp ( i made sure to compile it), it throws an error cannot exec...does anyone know how to fix it? |
| 22:22.23 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 22:22.37 | websae | no one could ever hear me |
| 22:22.37 | TripleFFFFFFFFFF | ?? |
| 22:22.45 | websae | I DMZ the phone |
| 22:22.52 | websae | I did everything |
| 22:22.54 | znoG | tried a later firmware websae? |
| 22:23.02 | websae | yep |
| 22:23.05 | KranZ | could have been the silence supression |
| 22:23.51 | websae | yeah....I put it in a box and put it in the closet |
| 22:24.53 | TripleFFFFFFFFFF | can someone dial 800 854 8957 and tell me if tone or not.. this is dringving me nuts |
| 22:25.27 | CunningPike | ph|ber: Did you get sorted out? Sorry - I was dragged away to a meeting |
| 22:25.41 | ph|ber | no.. |
| 22:25.42 | ph|ber | i didnt. |
| 22:25.47 | *** part/#asterisk seb- (n=seb@cpe-72-132-240-243.san.res.rr.com) |
| 22:26.08 | CunningPike | So, you have the info that your Sangoma is like a TE210P |
| 22:26.08 | ph|ber | http://pastebin.com/712484 |
| 22:26.10 | ph|ber | zaptel.conf |
| 22:26.31 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 22:27.17 | ph|ber | May 11 18:27:08 VERBOSE[4462] logger.c: -- Unregistered channel 23 |
| 22:27.17 | ph|ber | May 11 18:27:08 WARNING[4462] loader.c: Loading module chan_zap.so failed! |
| 22:28.29 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 22:29.07 | ph|ber | CunningPike: can i pm? |
| 22:29.28 | CunningPike | ph|ber: Sure |
| 22:29.33 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 22:35.01 | achandra | anybody got pcapplay function working with sipp? and if so which version did yyou use? |
| 22:35.28 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 22:36.24 | *** join/#asterisk p1p (i=p1p@mail.comp911.com) |
| 22:38.52 | Ariel_ | Hello I am trying to use the CUT command to get my callerID correct Here is what I have but it's not working any one have any suggestions: http://pastebin.ca/55210 |
| 22:39.26 | CunningPike | Ariel_: Didn't work for you, eh? |
| 22:39.55 | Ariel_ | CunningPike, no actuall it crased with some really nasty messages. |
| 22:40.06 | Ariel_ | crased/crashed |
| 22:40.06 | *** join/#asterisk MoRpHeUz (n=artur@tux14.ltc.ic.unicamp.br) |
| 22:41.52 | CunningPike | Ariel_: Really? Wow |
| 22:42.37 | MoRpHeUz | anybody here knows if I can do the following: I want to have an asterisk server that will create a conference room. 3 users will conect to this server and enter this conference room using VoIP. Then the Asterisk server must dial to a phone number so another guy can join this room. (note that the guy cannot dial to the asterisk server, the asterisk server must dial him)... any ideas! |
| 22:43.22 | websae | CLI |
| 22:43.25 | iCEBrkr | Callfiles? |
| 22:43.36 | websae | bridge him in with a sip phone |
| 22:43.36 | dlynes_ | Does anyone know what might cause lost packets on iax2, and if lost packets could cause echo on iax2? |
| 22:43.38 | *** join/#asterisk abatista (n=Ariel@70.46.87.158) |
| 22:43.54 | MoRpHeUz | websae: but he does not have access to internet... |
| 22:44.05 | websae | you do though |
| 22:44.06 | MoRpHeUz | websae: this is why the asterisk server must call him |
| 22:44.09 | websae | if you're in the conference |
| 22:44.15 | websae | you can bridge him in with your SIP phone |
| 22:44.22 | CunningPike | Ariel_: Your incoming callerid is "3055741495;phonecontext=+" right? |
| 22:44.26 | MoRpHeUz | websae: hmm.. |
| 22:44.29 | MoRpHeUz | didnt know that..... |
| 22:44.44 | dlynes_ | MoRpHeUz: Use the transfer button on your sip phone |
| 22:44.52 | websae | yep |
| 22:44.53 | websae | exactly |
| 22:45.00 | CunningPike | Ariel_: Then you want CUT(FIXED2=${FIXED},;,1) |
| 22:45.05 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:45.08 | MoRpHeUz | any of you know if any softphone do that ? |
| 22:45.12 | websae | as long as your in the MeetMe conference....then just dial his number |
| 22:45.15 | websae | yeah |
| 22:45.17 | ariel_ | CunningPike, yes |
| 22:45.18 | websae | any of them |
| 22:45.26 | MoRpHeUz | sjphone for example ? =) |
| 22:45.27 | websae | with multiple line appearance |
| 22:45.29 | dlynes_ | MoRpHeUz: it's a pretty basic feature on most softphones |
| 22:45.33 | websae | grandstream gxp 2000 |
| 22:45.36 | websae | andy sipura phone |
| 22:45.40 | websae | any polycom phone |
| 22:45.41 | ariel_ | CunningPike, I tried with 1 instead of 2 and gives me the same messages |
| 22:45.43 | MoRpHeUz | Ow sorry... |
| 22:45.50 | MoRpHeUz | I meant software phone... |
| 22:45.53 | websae | sure |
| 22:45.54 | *** join/#asterisk Ax0n (n=morcant@207.149.230.1) |
| 22:45.55 | websae | Xlite |
| 22:46.00 | dlynes_ | MoRpHeUz: i think i might have only run across 1 or 2 softphones that don't support transfer |
| 22:46.00 | MoRpHeUz | hmmmm.... |
| 22:46.01 | MoRpHeUz | =) |
| 22:46.05 | CunningPike | ariel_ - but change your delimter to ; |
| 22:46.10 | MoRpHeUz | I'll take a look at that! |
| 22:46.12 | MoRpHeUz | =) |
| 22:46.15 | MoRpHeUz | you saved my day! =) |
| 22:48.37 | MoRpHeUz | bye thanks !! |
| 22:49.29 | Ax0n | I have *89 setup to be handled by VoiceMailMain, however asterisk segfaults when I dial that extension... gdb isn't much help it just says it's in __strtol_internal from app_voicemail.so. |
| 22:49.45 | Ax0n | The last line in console is -- Executing VoiceMailMain("SIP/353-d2ee", "") in new stack |
| 22:49.51 | dlynes_ | Ax0n: which version of asterisk? |
| 22:50.14 | Ax0n | Gentoos asterisk-1.2.7_p1-r1 |
| 22:51.06 | cytrak | damn asterisk keeps hangin up on me when I make calls over the PSTN line .. not sure if it is asterisk or the free version of idefisk |
| 22:51.09 | dlynes_ | Ax0n: pastebin your backtrace? |
| 22:51.14 | Ax0n | k |
| 22:52.00 | dlynes_ | cytrak: define 'hang'? |
| 22:52.48 | Ax0n | dlynes_: http://pastebin.com/712531 |
| 22:53.19 | cytrak | the call is hung up |
| 22:53.43 | ariel_ | CunningPike, I did but it gives me this: May 11 15:47:09 WARNING[28096]: pbx_config.c:1727 pbx_load_module: No closing parenthesis found? 'CUT(FIXED2=${FIXED},-,' |
| 22:53.54 | dlynes_ | Ax0n: If you recompile asterisk with debugging info enabled, does it still happen? |
| 22:54.02 | cytrak | -- Accepting AUTHENTICATED call from 192.168.120.212: |
| 22:54.03 | cytrak | <PROTECTED> |
| 22:54.03 | cytrak | <PROTECTED> |
| 22:54.03 | cytrak | <PROTECTED> |
| 22:54.03 | cytrak | <PROTECTED> |
| 22:54.03 | cytrak | <PROTECTED> |
| 22:54.05 | cytrak | <PROTECTED> |
| 22:54.07 | cytrak | <PROTECTED> |
| 22:54.09 | cytrak | <PROTECTED> |
| 22:54.09 | dlynes_ | ~pb |
| 22:54.11 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:54.11 | cytrak | <PROTECTED> |
| 22:54.13 | cytrak | <PROTECTED> |
| 22:54.15 | cytrak | <PROTECTED> |
| 22:54.17 | cytrak | <PROTECTED> |
| 22:54.19 | cytrak | <PROTECTED> |
| 22:54.21 | cytrak | <PROTECTED> |
| 22:54.24 | CunningPike | ariel_: Paste the syntax you're using (just that one line) |
| 22:54.26 | cytrak | sorry I should have done so |
| 22:54.45 | Ax0n | dlynes_: I suppose to should have tried that first, thought this might be known thing. Let me try. |
| 22:54.55 | ariel_ | exten => _X.,2,CUT(FIXED2=${FIXED},-,;) |
| 22:55.06 | dlynes_ | Ax0n: Well, i'm currently working on fixing up the app_voicemail.c code |
| 22:55.23 | dlynes_ | Ax0n: So if you're able to give me a bit more debugging info, I might be able to help you with the problem |
| 22:55.25 | Corydon-w | Cut(FIXED2=FIXED,-,\\\;) |
| 22:55.36 | Ax0n | dlynes_: k, it's recompiling now |
| 22:55.40 | Corydon-w | Actually, what's the ; for? |
| 22:56.02 | Corydon-w | Shouldn't that be Cut(FIXED2=FIXED,-,1) ? |
| 22:56.24 | ariel_ | Corydon-w, it give a message not to use Cut but to use CUT |
| 22:56.36 | Corydon-w | ariel_: Yeah, that's not an app |
| 22:56.53 | Corydon-w | ariel_: Set(FIXED2=${CUT(FIXED,-,1)}) |
| 22:57.36 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 22:58.16 | ariel_ | Corydon-w, should it not be ; instead 1? http://pastebin.ca/55210 |
| 22:58.26 | ariel_ | has the message I need to change the callerID on. |
| 22:59.00 | ariel_ | that one errors out |
| 22:59.06 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:59.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:59.08 | Corydon-w | ariel_: you can collapse the first two lines... |
| 22:59.38 | Corydon-w | Set(FIXED2=${CUT(CALLERID(num),-,2)}) |
| 22:59.46 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 23:00.38 | Corydon-w | And, you don't need to number the priorities 1,2,3,4,5 anymore... you can just do 1,n,n,n,n |
| 23:01.10 | cytrak | could it be hanging up if it doesn't hear any sound going on ? |
| 23:01.22 | cytrak | cause sometimes no one is talking |
| 23:01.32 | ariel_ | Corydon-w, yes on the number just a bad habbit. |
| 23:01.57 | nextime | can i use unicall on a recent svn trunk? |
| 23:02.09 | Ax0n | dlynes_: same backtrace, let me try compiling app_voicemail.so with -ggdb by hand |
| 23:02.24 | nextime | ( where recent == SVN-trunk-r16533 ) |
| 23:03.53 | nextime | i'm able to find chan_unicall only for asterisk 1.0.x and 1.1.x on libunicall web site |
| 23:03.59 | dlynes_ | Ax0n: -g should be sufficient |
| 23:05.00 | *** part/#asterisk MoRpHeUz (n=artur@tux14.ltc.ic.unicamp.br) |
| 23:05.17 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 23:06.02 | Ax0n | dlynes_: It looks like it's getting compiled with -fomit-frame-pointer |
| 23:06.07 | Ax0n | so I removed that |
| 23:06.10 | Ax0n | will let you know in a sec |
| 23:06.14 | Corydon-w | nextime: there was never a 1.1 |
| 23:06.16 | dlynes_ | ok |
| 23:07.48 | De_Mon | ariel_ numbering gets important when you're using goto's |
| 23:08.25 | ariel_ | De_Mon, yes that is what I am going to be doing once I get the correct callerID |
| 23:09.03 | Ax0n | dlynes_: http://pastebin.com/712531 |
| 23:09.11 | De_Mon | I usually do 1,n,n,n,5,n,n etc where 5 is a goto point |
| 23:09.18 | nextime | Corydon-w : you have to say this at people on soft-switch.org |
| 23:09.38 | nextime | they distribute 2 versions of chan_unicall, one is for "asterisk-1.1.x" |
| 23:10.18 | dlynes_ | Ax0n: That's the same pastebin as before :) |
| 23:10.40 | Ax0n | dlynes: Same URL, I updated it? |
| 23:11.05 | Ax0n | it should say something about adsi now |
| 23:11.05 | De_Mon | what's the benefit to using pre-connect=yes in res_odbc.conf? |
| 23:11.05 | dlynes_ | ok....rhm |
| 23:11.05 | *** part/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 23:11.18 | Corydon-w | De_Mon: some databases take time to connect to |
| 23:11.22 | dlynes_ | nope....no mention about adsi |
| 23:11.35 | dlynes_ | i even forced a refresh on it |
| 23:11.36 | Ax0n | ok, let me do a new one |
| 23:12.12 | nextime | anyway, i need to use a recent asterisk, SVN-trunk-r16533 is working good for me, and i have a new PRI which is r2, so i need unicall, i think... |
| 23:12.13 | Ax0n | http://pastebin.com/712560 |
| 23:12.18 | Ax0n | try that one |
| 23:12.50 | De_Mon | Corydon-w it leaves a connection open indefinately then? |
| 23:13.00 | dlynes_ | Ax0n: that's a different problem than your previous problem |
| 23:13.01 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 23:13.06 | De_Mon | is there a 'show' command that will tell me about res_odbc? |
| 23:13.08 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 23:13.14 | nagl | hi |
| 23:13.18 | Ax0n | dlynes_: That's what I get with debuggin on? |
| 23:13.23 | Corydon-w | "odbc show" |
| 23:13.27 | dlynes_ | Ax0n: maybe....lemme check something |
| 23:14.18 | Ax0n | dlynes: Hmm |
| 23:14.31 | ariel_ | Corydon-w, that did not work. |
| 23:14.51 | Corydon-w | ariel_: it should've |
| 23:15.01 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 23:15.05 | Ax0n | commenting out noload => res_adsi.so seams to allow voicemail to work now |
| 23:15.10 | De_Mon | ah hah |
| 23:15.35 | Ax0n | dlynes: I guess res_adsi.so is required for voicemail? |
| 23:15.51 | dlynes_ | Ax0n: res_adsi.so is required for some things |
| 23:15.58 | dlynes_ | Ax0n: I can't recall what offhand, though |
| 23:16.16 | dlynes_ | Ax0n: but it's required by some modules that don't even have a need for adsi |
| 23:16.18 | *** join/#asterisk apardo (n=apardo@87.217.145.14) |
| 23:16.53 | Ax0n | dlynes: hmm |
| 23:17.01 | Ax0n | anything else I should put back then? :> |
| 23:17.16 | dlynes_ | Ax0n: but that still shouldn't make app_voicemail.c segfault, either |
| 23:17.40 | dlynes_ | Ax0n: or did you unload res_adsi.so after asterisk was running? |
| 23:17.47 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:18.00 | Ax0n | dlynes: nope, just added noload => res_adsi.so in modules.conf |
| 23:18.09 | dlynes_ | ok |
| 23:18.30 | dlynes_ | and in debug mode, it's not crashing at all, any longer? |
| 23:18.41 | Ax0n | even without debug it's not crashing |
| 23:18.48 | dlynes_ | erm i mean with debugging symbols |
| 23:18.54 | CunningPike | ariel_: Paste the syntax now |
| 23:18.59 | dlynes_ | ah...weird |
| 23:19.00 | distortion | load => app_voicemail.so ; Requires res_adsi.so |
| 23:19.15 | Ax0n | Yeah, with the stock install it's working with res_adsi.so now |
| 23:19.19 | dlynes_ | distortion: yeah, but what i was getting is that it doesn't make any sense |
| 23:19.26 | dlynes_ | distortion: adsi is not required for voicemail |
| 23:19.38 | dlynes_ | distortion: hell, if you're not using analog, adsi shouldn't be required |
| 23:19.51 | ariel_ | CunningPike, http://pastebin.ca/55215 |
| 23:20.16 | dlynes_ | distortion: it should only be needed both the adsi module, and nothing else |
| 23:20.21 | dlynes_ | s/both/by |
| 23:20.37 | CunningPike | ariel_: Replace the - with a ; |
| 23:20.37 | ariel_ | did that too. |
| 23:20.38 | CunningPike | And? |
| 23:20.42 | ariel_ | let me post the message I get with that. |
| 23:20.46 | CunningPike | OK |
| 23:21.33 | s34n | I need the ability for a meetme conference to dialout to include a party |
| 23:21.52 | ariel_ | CunningPike, http://pastebin.ca/55216 |
| 23:22.25 | dlynes_ | Ax0n: Yeah...I think when I rework that, I'll move the adsi crap into its own app, or move it into the adsi app |
| 23:23.03 | CunningPike | ariel_: Try quoting the ; - it thinks it's a comment marke |
| 23:23.10 | CunningPike | s/marke/marker/ |
| 23:23.35 | ariel_ | CunningPike, like "" or () |
| 23:23.45 | CunningPike | Try "" |
| 23:24.45 | ariel_ | CunningPike, no workie |
| 23:25.02 | ariel_ | ok then how do I well get the for 10 or 11 numbers? |
| 23:25.31 | elvisthedj | Hey all.. is video supported on any of the 79xx phones? (via SIP or SCCP?) |
| 23:25.40 | CunningPike | ariel_: Try this instead ${CALLERID(number):10} |
| 23:25.51 | dlynes_ | TripleFFFFFFFFFF: type /join #freebsd |
| 23:26.18 | ariel_ | That removes the first 10 numbers |
| 23:26.31 | Ax0n | dlynes: okie |
| 23:26.37 | Ax0n | Thanks for your help |
| 23:26.44 | *** part/#asterisk Ax0n (n=morcant@207.149.230.1) |
| 23:27.40 | elvisthedj | cisco 7085 = ~3,000 bucks :/ |
| 23:27.47 | elvisthedj | err.. 7985 |
| 23:29.08 | MRH2 | <-- thinks the 7985 it is a tandberg in cisco clothing |
| 23:29.23 | CunningPike | ariel_: Set(FIXED2=${CUT(CALLERID(num),";",1)}) doesn't work? |
| 23:30.45 | elvisthedj | tandberg 1000 "buy it now" 1,200 ..getting better :) |
| 23:31.10 | ariel_ | CunningPike, no it does not work |
| 23:31.16 | CunningPike | ar |
| 23:31.23 | CunningPike | ariel_: Hmm |
| 23:31.41 | elvisthedj | is there a list (if any) of asterisk supported video enabled sip devices? |
| 23:32.24 | TripleFFFFFFFFFF | can ok then how do i jump to a new exten in realtime |
| 23:32.37 | TripleFFFFFFFFFF | like .. i need to jump out of that table to another |
| 23:32.55 | TripleFFFFFFFFFF | tried exten|context"prio |
| 23:32.58 | TripleFFFFFFFFFF | no" |
| 23:33.00 | *** part/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 23:35.02 | *** part/#asterisk mog_italy (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 23:38.28 | ariel_ | CunningPike, exten => _X.,2,Set(FIXED2=$[${CALLERID(number):0:10}]) works it gives me the first 10 |
| 23:38.52 | CunningPike | OK - I was trying to avoid trying you to 10 digitrs |
| 23:39.04 | CunningPike | s/digitrs/digits/ |
| 23:39.23 | distortion | ariel_ are you trying to strip everything but the first 10 digits on ani? |
| 23:39.47 | ariel_ | CunningPike, same here the main reason I need this is due to the provider actually gives me 11 digits sometimes and it's got a zero in front of it. but I got it figured out now. Thanks |
| 23:39.59 | CunningPike | OK |
| 23:40.17 | ariel_ | distortion, actually yes but I now need to do some routing on 10 digits and 11 digits |
| 23:40.26 | *** join/#asterisk Sephen (n=Sephen@proxy5.med-web.com) |
| 23:41.12 | ariel_ | CunningPike, and Corydon-w thanks for all your help. I got if working. But not using the CUT command |
| 23:41.41 | CunningPike | ariel_: We just need to figure out how to stop * from interpreting the semi-colon |
| 23:41.49 | CunningPike | ariel_: Then you'd be all set |
| 23:41.59 | distortion | airiel: I used Set(CALLERID(num)=${CALLERID(num):0:10}) in the past |
| 23:42.14 | MRH2 | some users are starting to wear out the keys on the phones anyone got any ninja like recommendations to combat this? clear nail polish maybe? |
| 23:42.21 | ariel_ | CunningPike, yes but I have been at this for about 5 hours and I am getting very mad about this... |
| 23:42.33 | CunningPike | ariel_: I feel your pain :) |
| 23:42.56 | CunningPike | MRH2: I'm not sure how the color of the user's nail polish will help you |
| 23:43.04 | distortion | sounds like you got it working anyway, guess i was late ;) |
| 23:43.05 | MRH2 | lol |
| 23:43.14 | ariel_ | distortion, I had something similar with the old cut when I had it with asterisk 1.0.9 working But we upgraded the server and it broke. |
| 23:43.30 | MRH2 | i was hoping some might rub off on the keys |
| 23:43.31 | ariel_ | upgraded to 1.2.5.71 |
| 23:43.47 | ariel_ | argh can't type any more today |
| 23:44.03 | MRH2 | ruby red just doesn't go well with polycom |
| 23:44.20 | Sephen | Is there a simple way to strip characters before pattern matching? Say I wanted to strip 9s and 1s that are prefixed before I start a pattern match, so the user can use any of the following and I only need to have a single match for the NPA-NXX-XXXX, and not for every combination? 9-1-222-333-4444, 1-222-333-4444, 9-222-333-4444, and 222-333-4444? I have approximately 590 of these I need to add, and I didn't want to have 2300+ records. :) |
| 23:44.52 | ariel_ | Sephen, yes you can |
| 23:48.31 | Sephen | ariel_: Would you care to share how? :) |
| 23:50.17 | ariel_ | Sephen, let me finish a problem I just got a solution for that has got me pulling my hair out for 5 hours. And Yes.. your solutions is fairly easy it's just exten => _X.,1,Dial,zap/g1,${EXTEN:1} |
| 23:50.54 | ariel_ | you can actually change the _X. for 1 or 9 what ever you want to remove. |
| 23:51.06 | znoG | anyone here experienced with Lucent Definity PBX? |
| 23:51.23 | distortion | stephen: exten => _9.,1,Goto(${EXTEN:1) then exten => _1NXXNXXXXXX,1,Dial(blah) |
| 23:51.56 | Sephen | ugh. That'll be changing quite a bit of my existing dialplan. |
| 23:53.17 | distortion | why, just add that goto command once |
| 23:56.45 | Sephen | distortion: I have a PRI in my office, and a PRI in another city which we operate in. VPN between two asterisk boxes. We're a local CLEC as well, so our LD is extremely cheap here. What I was setting up was a pattern match of NPA-NXXs that were local to the PRI we have. Some of the NXXs with the same NPA are not local, and I want those to go out of my PRI, not the remote. So, if I pattern match every NPA-NXX match that I know of that is local, thats |
| 23:56.45 | Sephen | <PROTECTED> |
| 23:57.50 | Sephen | So I was looking for a way to strip the 9 or 1 (if present) and just pass the 10D to the context that does the pattern matching. |
| 23:58.46 | distortion | i use a database to do that, makes it very simple to create/recreate/modify patterns, you could easily do it in access |
| 23:59.16 | distortion | I manage about 30k-50k nxx routes in a include file that way |
| 23:59.34 | CunningPike | Sephen: Unless I'm missing something, we do exten => _9X.,1,Goto(${CONTEXT},${EXTEN:1},1) |