| 00:02.52 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
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| 00:08.15 | Katty | hihi |
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| 00:50.09 | Ryushin | What format are the gsm sound files in? I'm trying to play them back using alsaplayer. |
| 00:50.31 | JT | gsm format? |
| 00:50.42 | JT | gsm is a codec, as well as a mobile telephone system |
| 00:52.22 | Ryushin | I want to play the files in /var/lib/asterisk/sounds |
| 00:52.47 | *** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net) |
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| 00:54.15 | JT | Ryushin: sox can probably help |
| 00:54.42 | Ryushin | Thanks JT. |
| 00:55.24 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
| 00:57.26 | awe6 | Outside phones cannot ring into Asterisk connected to Sipura3000 POTS line. |
| 00:57.46 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 00:57.55 | awe6 | Anyone familiar enough with Sipura to give advice on getting calls to "ring through"? |
| 00:58.03 | [hC] | man.. im having an interesting time moving over to a number of #include's |
| 00:58.10 | [hC] | seems asterisk stops loading them after some point |
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| 01:24.13 | Ryushin | I can't figure out how to download the extra sounds. Are they not available anymore? |
| 01:24.26 | Ryushin | I've installed svn asterisk-sounds, but there aren't any. |
| 01:26.08 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
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| 01:34.38 | asteriskmonkey | anyone have any examples of long distance code implementation in asterisk? |
| 01:35.53 | teknoprep | 011. |
| 01:36.14 | teknoprep | anyone here get the spa-1001 to work with call waiting? |
| 01:36.15 | asteriskmonkey | i mean more like where a security code has to be entered to dial out ld |
| 01:36.26 | asteriskmonkey | I was looking for some example scripts... |
| 01:36.49 | asteriskmonkey | Else ill have to be un-lazy and make an sql database and dial plan logic :P |
| 01:37.30 | [shodan] | when I run voicemail() is there a way that if the caller press the # to indicate he is done , that instead of hanging up it would go back to (s,1) ? |
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| 01:38.37 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
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| 01:48.58 | Corydon76-home | [shodan]: how about if you add a Goto in the dialplan? |
| 02:01.42 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:02.10 | shmaltz | anybody here using an AMD based system with redundant PS, and SATA Based RAID? |
| 02:03.13 | [shodan] | Corydon-w, yes I thought it was the voicemail application hanging up ;) oops |
| 02:04.31 | bkw_ | muhahahaha |
| 02:07.04 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.53 | [shodan] | shmaltz, what do you want to know ? |
| 02:08.56 | shmaltz | shodan, what hardware |
| 02:09.39 | shmaltz | shodan, I'm trying to write a quote for someone with the above specs, however I don't want to underquote |
| 02:10.55 | asteriskmonkey | file you awake .. prod prod |
| 02:11.05 | file | no. |
| 02:11.12 | asteriskmonkey | lol |
| 02:12.14 | asteriskmonkey | is it possible to define an array of numbers in asterisk and have call logic return a 0/1 based on if a number is in that array? |
| 02:12.38 | file | why don't you just use astdb? |
| 02:13.16 | asteriskmonkey | mmmm ... never thought of using it to be honest.. its for a bunch of users who all have long distance codes... |
| 02:13.24 | asteriskmonkey | i could use astdb for that right? |
| 02:13.38 | file | sure... it's a file based database |
| 02:13.51 | file | with a set of applications/dialplan functions to read/write/whatever |
| 02:14.17 | asteriskmonkey | ah cool ive always been using mysql :P and agi's |
| 02:16.26 | asteriskmonkey | where are the astdb flat files kept by default? |
| 02:16.38 | file | in a place you shouldn't touch directly :P |
| 02:16.49 | file | it's not meant to be messed with outside of Asterisk |
| 02:16.53 | [shodan] | shmaltz, the only thing I'm not sure is redudant psu , how do you want to implement it ? a board with two atx connector will seriously limit your options |
| 02:17.16 | [shodan] | the rest is commodity hardware now |
| 02:17.17 | shmaltz | shodan, why is that? |
| 02:17.18 | asteriskmonkey | ah ok so i have to write my values in manually through asterisk |
| 02:17.34 | shmaltz | shodan, any good one stop supplier that you know of? |
| 02:17.39 | [shodan] | shmaltz, it's not common |
| 02:18.08 | shmaltz | shodan, I see, thanks, so I'll have to go with the PS that gives just one ATX output |
| 02:18.11 | [shodan] | I use eprom.com (in canada) but they have no board that have redundant psu management onboard |
| 02:19.40 | shmaltz | shodan, are the redundant psu that just give one atx ps connector as good as the boards that have 2? |
| 02:20.22 | *** join/#asterisk The_TiK (n=jeff@cpe-70-114-47-78.satx.res.rr.com) |
| 02:20.50 | asteriskmonkey | are astdb values maintained after a reboot? |
| 02:20.53 | [shodan] | I'd trust more , two normal psu with a device that connects it all together |
| 02:21.18 | [shodan] | btw, I love this case for server , 12 bay , 2 psu bay and not too expensive http://www.coolermaster.com/index.php?LT=english&Language_s=2&url_place=product&p_serial=RC-820&other_title=+RC-820+CM-Stacker% |
| 02:21.26 | [shodan] | (155$cad w/o psu) |
| 02:30.34 | *** join/#asterisk EzWayz (n=ez@c207.134.50-113.clta.globetrotter.net) |
| 02:31.36 | asteriskmonkey | file: astdb is primary key based |
| 02:32.11 | asteriskmonkey | file : i cannot have more than 1 of the same key type.. i have to use mysql.. unless you know another way of having multiple values for a single key in astdb |
| 02:33.30 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
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| 02:37.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:37.32 | JT | what the hell |
| 02:37.42 | bkw_ | what? |
| 02:37.49 | JT | who makes dual ATX connector motherboards for the purposes of redundancy? |
| 02:37.52 | JT | that's crazy talk |
| 02:37.57 | bkw_ | oh /me has been working on g722 |
| 02:38.08 | JT | just get a redundant power supply and be done with it, shmaltz, [shodan] |
| 02:39.50 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
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| 02:47.08 | [Outcast] | has anyone seen the user list? |
| 02:47.17 | JT | ? |
| 02:48.53 | hads | I was wondering about that myself. |
| 02:49.01 | bkw_ | wondering about what? |
| 02:49.17 | sumasuma | JT: anyone received the G729 and G723 code from the mailing list ! |
| 02:49.29 | JT | no? |
| 02:49.45 | sumasuma | i just received the email from mailing lists |
| 02:49.46 | *** join/#asterisk BrianHV (n=bhv1@copland.brianhv.org) |
| 02:49.55 | sumasuma | the g729 and g723 code sent from digium |
| 02:50.00 | sumasuma | authored by mark spencer |
| 02:50.18 | sumasuma | Thanks to digium once again |
| 02:50.57 | hads | sumasuma: That email wasn't sent by Digium, it was sent by asteriskspy@yahoo.com |
| 02:51.20 | sumasuma | hads: yes, but the author of the code is Marks Spencer ! |
| 02:51.25 | sumasuma | is it not from them ? |
| 02:51.34 | JT | everything says it's authored by mark spencer |
| 02:51.37 | JT | even when it's not |
| 02:51.42 | bkw_ | that is true |
| 02:52.30 | sumasuma | ha ha, what a joke ! |
| 02:52.57 | JT | don't you know how to check mail headers to see who sent an email? |
| 02:52.58 | BrianHV | pardon me for being newbieish, but I'm interested in a very simple system that allows me to use a computer to make and receive plain phone line calls. is asterisk appropriate for that? |
| 02:52.59 | sumasuma | guys around the world work hard to make mark spencer great ! what a cheat ! |
| 02:53.17 | BrianHV | I just have a single phone line |
| 02:53.57 | *** part/#asterisk The_TiK (n=jeff@cpe-70-114-47-78.satx.res.rr.com) |
| 02:54.02 | sumasuma | JT: I'm not talking about a code from Linux Operating System, or Code for Libxml. I'm talking about the code that works well with asterisk, then I confirm Digium must have distributed across the files |
| 02:54.25 | sumasuma | JT: i never bothered about who sent the email, I'm on the content ! |
| 02:54.40 | hads | BrianHV: Yes, Asterisk is suitable for that. You will need an FXO device to hook up a phone line. |
| 02:54.43 | JT | the content can be affected by who sends it |
| 02:55.28 | sumasuma | JT: I will appoint you as a judge ok, leave me free. I'm not here for law ! |
| 02:55.46 | file | w t h... |
| 02:56.00 | hads | What he said |
| 02:56.07 | file | I go away for 15 minutes, and look what happens! |
| 02:56.36 | BrianHV | hads: thanks, that gives me some more info to go on |
| 02:56.50 | JT | sumasuma: i have no idea what you are talking about |
| 02:56.57 | sumasuma | JT: me too |
| 02:57.18 | hads | BrianHV: There are basically two types, PCI cards such as the Digium TDM400 and external boxes like the Linksys SPA3102 |
| 02:57.19 | JT | well if you care about the content of an email |
| 02:57.27 | JT | why wouldn't you care about who sent it? |
| 02:58.00 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 02:58.37 | BrianHV | hm. this seems like a larger investment than I think this is worth to me. ;) |
| 02:58.43 | sumasuma | ha ha, it is not about my personal pain, it is about technology, I just look into the technology and whether it is useful for me or not, i don't care from which part of world the author is from |
| 02:59.19 | sumasuma | i don't even mind whether mark or who wrote the asterisk |
| 02:59.30 | sumasuma | i just look into whether asterisk is useful for me or not |
| 03:00.01 | sumasuma | if works fine, i appreciate him, if not just through it away and look for something |
| 03:03.05 | [Outcast] | http://digg.com/software/g723_and_g729_codecs_source_for_asterisk_leaked |
| 03:03.26 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 03:03.29 | _DAW | is there an ser channel? |
| 03:06.41 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 03:07.03 | bkw_ | yo h3x ltns |
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| 03:17.12 | tengulre | This channel is sleeping?? |
| 03:17.33 | bkw_ | I gues so |
| 03:18.27 | blitzrage | sleep? great idea! |
| 03:18.38 | tengulre | :) |
| 03:19.01 | hads | Sleeping at 3pm is silly |
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| 03:32.33 | *** join/#asterisk ComputerWarm (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 03:32.35 | ComputerWarm | hello all |
| 03:46.41 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 03:55.59 | Strom_C | zing |
| 03:56.03 | bkw_ | zag |
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| 04:05.51 | docelmo | Did anyone see that post on the release of the digium codecs? |
| 04:06.12 | carrar | heh |
| 04:06.12 | file | docelmo: yes |
| 04:06.24 | file | now never speak of it again |
| 04:06.29 | file | :D |
| 04:06.40 | [Outcast] | http://digg.com/software/g723_and_g729_codecs_source_for_asterisk_leaked |
| 04:06.51 | docelmo | what? its mark's GPL code |
| 04:09.41 | *** join/#asterisk sconasq (n=aaaazz@pool-71-245-225-219.bstnma.fios.verizon.net) |
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| 04:16.57 | [shodan] | is there a way I can have every phone call made or received by a particular sip phone, recorded and sent by email the phone user's inbox for future reference ? |
| 04:17.14 | [shodan] | (I have it working for voicemails) |
| 04:20.12 | orlock | i am sure there is, but.. _email_? |
| 04:20.25 | orlock | freepbx has an option to save them to disk, but not email |
| 04:21.52 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 04:22.32 | sconasq | what ver of asterisk should i install |
| 04:22.56 | bkw_ | 1.2.xx |
| 04:23.31 | sconasq | k |
| 04:23.41 | sconasq | i believe theres a bug with g729 passthru |
| 04:24.07 | file | howso? |
| 04:24.47 | sconasq | i forget the error.. but i recall this patch is supposed to fix it: http://bugs.digium.com/view.php?id=4825 |
| 04:29.39 | *** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) |
| 04:29.39 | *** join/#asterisk zepmantra (i=dsadsads@125.212.110.115) |
| 04:30.42 | nvzn | hi, using trixbox for the first time, with a sip provider, im getting "the number you are trying to call is not in service" Outgoing calls work well. Im running the trixbox behind an openbsd pf firewall |
| 04:31.13 | nvzn | is this a nat issue, and if so wheres the problem? sip.conf? |
| 04:31.33 | nvzn | s/solution/problem/ |
| 04:31.41 | sconasq | nvzn, are you registered? |
| 04:31.51 | nvzn | how to tell? |
| 04:31.56 | sconasq | sip show registry |
| 04:31.57 | nvzn | i know outgoing calls work |
| 04:32.28 | nvzn | yep |
| 04:32.30 | nvzn | registered |
| 04:32.46 | sconasq | try turning on debug info |
| 04:32.49 | sconasq | sip debug |
| 04:33.07 | sconasq | call again and see if anything pops up in the log |
| 04:33.11 | nvzn | ok |
| 04:33.45 | nvzn | what am i looking for? |
| 04:34.21 | nvzn | UDP write: fd 12 |
| 04:34.21 | nvzn | Destroying call '0bba98830440172913d7f8126dd70caa@66.49.255.38' |
| 04:35.05 | sconasq | try set verbose 5 |
| 04:35.20 | sconasq | it should tell you why the call wasn't routed |
| 04:35.24 | nvzn | ok |
| 04:35.30 | nvzn | lemme try again |
| 04:38.51 | nvzn | are these logs written to a file? |
| 04:39.26 | nvzn | they keep moving :P |
| 04:42.55 | sconasq | disable debug for now and just check the verbose msgs.. sip no debug |
| 04:44.11 | nvzn | cool |
| 04:45.19 | *** join/#asterisk Piston (n=what@206M28.oasis.mediatti.net) |
| 04:47.36 | nvzn | so i guess its my incoming route |
| 04:52.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:53.00 | sconasq | yeah.. did u see a failure in the log? |
| 04:53.40 | *** join/#asterisk Cloneman (i=nicky@22-80-252-216-dsl.enter-net.com) |
| 04:53.43 | nvzn | playing 'ss-noservice' |
| 04:54.08 | nvzn | something about unknown peers |
| 04:54.12 | nvzn | peer |
| 04:54.22 | Cloneman | I have I quick question before I finish RTFM |
| 04:54.25 | nvzn | and s|1 |
| 04:54.48 | nvzn | executing NoOp |
| 04:55.11 | *** join/#asterisk linlin (i=linlin@c-67-184-154-143.hsd1.il.comcast.net) |
| 04:55.26 | Cloneman | what special things can a zaptel card do for me that a SIpura adapter wont, in the case of using analog phones with my pbx? |
| 04:56.09 | *** join/#asterisk linlin (i=linlin@c-67-184-154-143.hsd1.il.comcast.net) |
| 04:56.52 | sconasq | nvzn, sounds like the peer isn't setup |
| 04:57.10 | nvzn | hmm |
| 04:57.22 | sconasq | check that the ip/hostname of the inbound peer is defined in sip.conf |
| 04:57.36 | nvzn | ok |
| 05:01.16 | *** join/#asterisk SkramX (n=Mark@hermes.sentiensystems.com) |
| 05:01.41 | SkramX | anyone gotten a 900 (inbound) service with asterisk? |
| 05:01.47 | SkramX | like they can pipe it to SIP? |
| 05:02.03 | sconasq | sure why not |
| 05:02.14 | SkramX | any places that offer this? |
| 05:03.10 | sconasq | any 900 number will work |
| 05:03.18 | sconasq | u tell them to fwd it to a local number xxx |
| 05:03.23 | SkramX | right |
| 05:03.28 | SkramX | but providers of the 900 |
| 05:03.33 | SkramX | :) |
| 05:04.01 | sconasq | www.advancedtele.com is one |
| 05:04.49 | SkramX | thanks |
| 05:10.46 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 05:12.28 | salaud | Anyone know what the best way to tell a user that there time is about to run out in a "prepaid" type application? |
| 05:13.04 | salaud | Assuming that they are currently involved in a call? |
| 05:15.54 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.70) |
| 05:17.05 | sconasq | this looks like an interesting asterisk billing app: http://cs.sisnema.com.br/EducCS/blogs/asterisk/archive/2006/08/14/12928.aspx |
| 05:17.16 | sconasq | Warn the caller about the call interrupt X seconds before the call gets interrupted |
| 05:17.53 | salaud | sconasq: you suggesting looking at the source code there? |
| 05:18.01 | sconasq | anyway there are tons of solutions here: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications |
| 05:18.17 | salaud | sconasq: I don't really need a prepaid solution |
| 05:18.46 | salaud | I'm looking for more for a general way that one might step in the middle of a call and play some audio? |
| 05:19.59 | *** part/#asterisk SkramX (n=Mark@hermes.sentiensystems.com) |
| 05:20.50 | salaud | sconasq: I just need to integrate a little bit of "prepaid" logic into an entirely different kind of application |
| 05:21.00 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-54.range217-44.btcentralplus.com) |
| 05:21.14 | sconasq | not sure how u can play a sound in the middle of a call |
| 05:21.36 | salaud | right... and not sure how the asterisk2billing app would do that either |
| 05:24.03 | salaud | I'm thinking maybe set one timeout |
| 05:24.09 | salaud | then play a message |
| 05:24.13 | salaud | and then play another? |
| 05:24.38 | salaud | I mean and then let the call continue? .. but that doesn't make any sense |
| 05:25.14 | sconasq | yeah |
| 05:25.23 | sconasq | that sounds good |
| 05:25.23 | [shodan] | what's the first word of vm-login.gsm ? comedianmail ? |
| 05:25.48 | sconasq | yeah [shodan] |
| 05:26.04 | [shodan] | what's that , the name of the voicemail module ? |
| 05:27.04 | sconasq | maybe if u did TIMEOUT=foo then in the t extension played a file salaud |
| 05:27.22 | salaud | sconasq: That's what I was thinking.... |
| 05:27.35 | salaud | but... does it make sense when the call is connected? |
| 05:27.48 | salaud | where would the dialplan go after that? |
| 05:27.57 | nvzn | sconasq: thanks for the tips, im going to try again tommorrow :) |
| 05:28.13 | *** join/#asterisk lehel (n=t@82.77.41.206) |
| 05:28.28 | salaud | because it starts out by doing a Dial... |
| 05:28.29 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 05:28.42 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 05:28.45 | salaud | if the Dial command timesout... I imagine it breaks the call legs |
| 05:31.23 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 05:32.23 | sconasq | yeah dunno how it works |
| 05:32.58 | salaud | sconasq: appreciate the response though... I'll keep pushing it around for a while... |
| 05:34.57 | sconasq | maybe the agi could issue a playback command X seconds after Dial |
| 05:35.41 | salaud | sconasq: From what the docs say, if the AGI issues a Dial it is disconnected from the call but is free to clean up only in background |
| 05:36.20 | salaud | One of the weakest links in asterisk seems to be the callout stuff |
| 05:36.34 | sconasq | i c |
| 05:38.13 | sconasq | theres a bounty open on allowing ChanSpy to send audio to one side of the call |
| 05:41.19 | salaud | I see that it WILL be in asterisk 1.4... I'll hold my breathe until then... |
| 05:41.21 | *** join/#asterisk kimo_sabe (n=nlopez@70-58-49-185.tcsn.qwest.net) |
| 05:41.35 | salaud | but... it would need to work with files and not live channels |
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| 05:50.48 | Rahail | <PROTECTED> |
| 05:51.31 | salaud | Rahail: Did you look on the Wiki? http://www.voip-info.org/wiki-Asterisk |
| 05:52.14 | Rahail | yeah did they all good however pirce is about.03 |
| 05:52.36 | salaud | Rahail: I can say that I use JunctionNetworks and Asterlink |
| 05:53.02 | salaud | but... You can only get better pricing at higher levels of call volume... we are fairly small time... |
| 05:53.15 | salaud | So we pay what everyone pays... |
| 05:53.29 | salaud | about 2.9 cents a minutes |
| 05:53.45 | salaud | but asterlink seems to be cheaper for 800 |
| 05:54.07 | Rahail | how much they charge for 800 |
| 05:54.38 | salaud | I can't remember... but it's not 4.9cents |
| 05:54.49 | salaud | or 4.6 or whateverr |
| 05:55.04 | salaud | I think it's like asterlink.com or something |
| 05:55.31 | Rahail | thanx for the info |
| 05:55.39 | salaud | Rahail: np |
| 05:55.42 | Rahail | I think I am better with other provider |
| 05:55.48 | Rahail | i get about 1.5 |
| 05:55.54 | salaud | Rahail: who do you use? |
| 05:55.54 | Rahail | for incoming and outgoing :) |
| 05:56.11 | Rahail | one of the level3 master reseller |
| 05:56.25 | Rahail | they need huge commitment tho... |
| 05:56.32 | salaud | Rahail: which one? You have large volume? |
| 05:56.39 | Rahail | if i want keep that rate i have to send lot of minute |
| 05:56.58 | Rahail | not realy so they sent me letter if dont sent 500 dollar worth minute they will cancel my account |
| 05:57.00 | Rahail | :( |
| 05:57.21 | salaud | Rahail: Can you say which one it is? |
| 05:57.26 | Rahail | broadband |
| 05:57.27 | salaud | bummer.. by the way |
| 05:57.31 | salaud | oh.. ok |
| 05:57.49 | salaud | they get their money... |
| 05:58.15 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 05:58.29 | Rahail | i guess i got no choice i have to give up this company |
| 05:58.43 | salaud | Yeah... if you want that rate |
| 05:58.54 | [hC] | Anyone here fairly familiar with using #include throughout extensions.conf? Its seeming like if i use it inside of an existing context, things are not imported into that context, they are treated as unique |
| 05:59.46 | Rahail | i am just gone give all my friend a acount and have they pay me so that way i can have that account ... i gueess that only choice for me to keep that rate ... |
| 06:00.10 | Rahail | any on here used nufon |
| 06:00.13 | Rahail | nufone ? |
| 06:00.43 | sconasq | rahail.. pm me your email. i might have a contact u can go with |
| 06:01.38 | [shodan] | can I call a phone from a bash script ? like when I receive a fax , from my faxrcvd I would like to run call-ext101.sh and that script would call extention 101 then playback(misc-faxrcvd) , then hangup , how would I do that ? |
| 06:01.39 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 06:02.21 | [shodan] | asterisk -x goto(faxrcvd,s,1) ? |
| 06:02.37 | *** join/#asterisk NoRemorse (n=bah@eth2462.vic.adsl.internode.on.net) |
| 06:02.42 | salaud | [shodan]: you can do that with manager or dialout |
| 06:03.01 | NoRemorse | hi all, how does one set DIAL timeout and transfer options for asterisk realtime please? |
| 06:03.39 | salaud | [shodan]: from a bash script I would use a callout file |
| 06:03.49 | salaud | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 06:04.25 | salaud | with -x you can only execute commands that you would use from the command line inside the manager interface |
| 06:04.30 | [shodan] | k , I'll check that out |
| 06:04.48 | [shodan] | ah ok I thought you could place calls that way |
| 06:05.17 | Rahail | i need help |
| 06:05.26 | Rahail | how much bandwith ulaw use and how much bandwiht gsm use |
| 06:05.33 | salaud | [shodan]: You can only do 'sip show users' etc... also, if you use perl.. (or maybe php, etc.) there are libraries to create call files and use the manager |
| 06:05.36 | Rahail | and which one is better if you guiess can hint me... |
| 06:05.42 | Rahail | I am kind of limit on bandwith |
| 06:05.55 | salaud | gsm is much |
| 06:05.55 | salaud | less |
| 06:06.32 | Strom_C | ulaw is 64kbps plus overhead |
| 06:06.38 | Strom_C | gsm is 13kbps plus overhead |
| 06:07.11 | Rahail | 13+maby another 12k for overhead = 25kbps |
| 06:07.14 | Rahail | is enough |
| 06:07.23 | salaud | but... gsm is variable though |
| 06:07.37 | salaud | ulaw, I believe is constant |
| 06:07.40 | Rahail | like what |
| 06:07.54 | [hC] | any of you guys using #include ? |
| 06:07.59 | Rahail | i have like 256 upload |
| 06:08.20 | Rahail | i guess sound quality is better on gsm or ulaw |
| 06:08.32 | salaud | ulaw is better quality |
| 06:08.37 | Rahail | I know this question been answerd alread |
| 06:08.39 | Strom_C | gsm isnt variable IIRC |
| 06:08.48 | Rahail | what about g729 ? |
| 06:08.52 | salaud | [hC]: sorry don't use it. |
| 06:08.55 | Strom_C | g729 is 8kbps |
| 06:08.59 | Strom_C | plus overhead |
| 06:09.05 | Strom_C | [hC]: I use it |
| 06:09.12 | Rahail | so after all ualw is better ... |
| 06:09.18 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-54.range217-44.btcentralplus.com) |
| 06:09.27 | salaud | Strom_C: I thought that gsm was by it's very nature variable.. as it takes more data to represent certain types of audio then others.. |
| 06:09.54 | salaud | Strom_C: It just looks like the rates change on a gsm channel... but, not on a ulaw ... perhaps? |
| 06:10.18 | [hC] | Strom_C: It seems as though if i do something like.. [somecontext] then inside that an #include contents.txt - it wont actually insert the contents of the file into the context. it treats it as a new entry, (which is different than how a regular context-include works) is this correct or am i missing something? |
| 06:10.27 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 06:10.30 | Strom_C | no, as far as I know, the frame sizes are statically defines |
| 06:10.39 | Strom_C | er, defined |
| 06:10.41 | [shodan] | salaud, do I need a channel in the call file ? if I make a call file to just call one sip phone and playback a sound file , do I put that sip phone as the channel in my call file ? or do I put a "null' channel (if there is such a thing)in my call file , and then dial() the phone ? |
| 06:10.48 | Rahail | ok I am confused guiess... I am not that techie so which one do you guiess recomend |
| 06:10.50 | Strom_C | [hC]: show me some examples via pastebin |
| 06:10.54 | [hC] | mkay |
| 06:10.54 | Rahail | ulaw gsm or g729 |
| 06:11.01 | Strom_C | ulaw for call quality |
| 06:11.07 | Strom_C | g729 for low-bandwidth |
| 06:11.19 | Strom_C | gsm for free low-bandwidth |
| 06:11.39 | Rahail | cool any one have wiki link how to install gsm on asterisk |
| 06:11.47 | Strom_C | its already installed |
| 06:11.54 | Rahail | realy |
| 06:12.05 | salaud | [shodan]: Channel is the SIP channel you are calling... then you put a context, extension, and priority in there.. |
| 06:12.06 | Strom_C | maybe you should read the book |
| 06:12.13 | NoRemorse | has anyone here managed to get fax passtrhough working on cisco 827 cpe's? |
| 06:12.28 | salaud | [shodan]: when the channel picks up it is sent to that context, extension, and priority |
| 06:12.39 | Rahail | ok Strom_C one more what do you type to see which codec are installed |
| 06:12.46 | Strom_C | show translations |
| 06:12.50 | Strom_C | or show translation |
| 06:12.52 | Strom_C | one of the two |
| 06:13.17 | salaud | [shodan]: so your context/extension would have myexten,1 Playback() / myexten, 2, hangup |
| 06:13.41 | Rahail | <PROTECTED> |
| 06:14.01 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:14.04 | salaud | [shodan]: It basically calls out to a channel and connects the other end to the dialplan... as if they had called into the dialplan at that extension, only you called out to them |
| 06:14.47 | [shodan] | ok I get it ! thanks |
| 06:15.01 | salaud | [shodan]: np |
| 06:15.06 | [hC] | Strom_C: http://pastebin.ca/161267 |
| 06:15.15 | [hC] | Strom_C: is the second way i laid it out the correct way to do this? |
| 06:16.26 | Strom_C | honestly, i just do the includes in the main extensions.conf and put the context headings in the included files |
| 06:16.35 | [hC] | yeah.. okay |
| 06:16.46 | Strom_C | typically I have extensions_local, extensions_inbound, extensions_outbound, etc |
| 06:16.48 | [hC] | im pretty sure the second way i laid it out is how it has to work, it will explain all the weirdness ive come across so far. |
| 06:17.19 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 06:17.30 | [hC] | just kind of annoying i guess having to do an #include then an include right under it to have the parent context adopt it |
| 06:17.34 | Strom_C | splitting the config up into a few files makes management easier; using too many files is stupid |
| 06:17.38 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 06:18.01 | [hC] | Im not going too far but im building a large management system for config update rollouts for all my clients with overridable parts |
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| 06:22.29 | *** join/#asterisk juhas (i=juhas@hot.juhas.net) |
| 06:22.49 | juhas | hello. anybody here who could help debugging asterisk euroisdn connection to an AXE switch? |
| 06:24.25 | juhas | (or who has any pointers on how to debug it.. 2M data link is up but for some reason D-channel isn't getting a valid conversation; asterisk is only sending SABME but not seeing any replies) |
| 06:31.45 | *** part/#asterisk kimo_sabe (n=nlopez@70-58-49-185.tcsn.qwest.net) |
| 06:32.32 | salaud | Anyone know why Set(TIMEOUT(absolute) = seconds) doesn't seem to work in 1.2.10? |
| 06:32.55 | salaud | AbsoluteTimeout() does seem to work |
| 06:35.46 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:36.08 | Dico_ | salaud, may be it has been deprecated ? |
| 06:36.25 | salaud | Dico_: Set(TIMEOUT(absolute) = seconds) is the "New" way |
| 06:36.36 | Dico_ | lol, i see |
| 06:38.45 | juhas | i'm using Set(TIMEOUT(absolute)=1800) and it seems to work in 1.2.10 |
| 06:39.07 | salaud | juhas: you calling an AGI after it? |
| 06:39.28 | juhas | mm, no |
| 06:39.29 | sx-wks | heya guys. |
| 06:39.42 | sx-wks | would this be too big a setup ? http://rafb.net/paste/results/gAHfP576.html |
| 06:39.52 | salaud | I haven't gone through a lot of testing steps... but, I can clearly do set(TIMEOUT) and nothing happens... but, AbsoluteTimeout definitely works |
| 06:40.05 | sx-wks | I'm looking at having 1 full TE412P conferencing with each other |
| 06:40.10 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:40.16 | salaud | I'm in an AGI when the timeout should go off |
| 06:40.44 | litage | how many bytes is an average SIP packet? |
| 06:41.30 | sconasq | sip has very little bandwidth consumption |
| 06:41.59 | salaud | litage: perhaps you are more interested in the UDP packets that make up RTP? |
| 06:42.52 | sconasq | here's all the stats on RTP payloads: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 06:43.08 | litage | salaud: actually i'm specifically interested in SIP packets, not RTP |
| 06:43.19 | salaud | litage: cool... just checking |
| 06:43.21 | litage | sconasq: "very little" as in <1KB? |
| 06:43.36 | sconasq | oh yeah way under |
| 06:44.14 | sconasq | more like 250 - 500 bytes |
| 06:46.59 | litage | thanks |
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| 07:05.30 | Rahail | rgfb ' |
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| 07:52.23 | phearless | good morning #asterisk |
| 07:52.32 | Niklas- | Exten => 1234,1,Set(VOICEMAIL_ENABLED=${DB(FEAT/${EXTEN}/voicemail)}) - can anyone spot an error? |
| 07:53.46 | salaud | Niklas-: I can't see one... right off... but.. have you tried doing each part in a NoOp to see the results of each part? |
| 08:00.23 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 08:02.20 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 08:04.43 | Niklas- | salaud, NoOp? |
| 08:04.44 | phearless | exten => 0XXX,1,Background(tt-monkeysintro) |
| 08:04.47 | phearless | not working |
| 08:04.53 | phearless | exten => 0123,1,Background(tt-monkeysintro) |
| 08:04.59 | phearless | it is working |
| 08:05.31 | phearless | so why, with the first line,I can not dial for example 0421 and listen to the monkey sound ? |
| 08:05.33 | salaud | Niklas-: NoOp() is a command... You can do: NoOp(${DB(FEAT/${EXTEN}/voicemail)}) for instance |
| 08:05.35 | macTijn | phearless: maybe you already have an exten that starts with 0 and is 4 digits long |
| 08:05.45 | phearless | I had a look and did not see it |
| 08:05.47 | phearless | weird |
| 08:05.53 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 08:06.02 | salaud | phearless: You probably need an underscore '_' in front of that... right? |
| 08:06.26 | salaud | Niklas-: That will show you the result value in the asterisk CLI |
| 08:06.37 | phearless | why should I put an underscore ? |
| 08:07.08 | salaud | phearless: I'm pretty sure that's how you do pattern matching... and it looks like you have a pattern |
| 08:07.28 | phearless | okay, it is not explained in http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 08:07.30 | phearless | I will try |
| 08:07.36 | salaud | without a leading '_' I believe it is looking for an extension with real 'X' characters in it |
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| 08:08.14 | salaud | phearless: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 08:08.26 | salaud | It's on the config extensions page as a link |
| 08:08.52 | salaud | phearless: also: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
| 08:09.29 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:09.30 | phearless | you are right, _ is needed for patterns |
| 08:09.33 | phearless | cool. |
| 08:10.01 | macTijn | oh, right :) |
| 08:10.02 | voipmagic | have2have _ yes ! |
| 08:10.14 | phearless | thanks salaud |
| 08:10.20 | salaud | phearless: np |
| 08:10.24 | macTijn | <- not awake yet |
| 08:12.22 | Niklas- | ah thanks salaud |
| 08:12.54 | salaud | Niklas-: npaa |
| 08:14.34 | Niklas- | hmm |
| 08:15.04 | Niklas- | not sure on how to use it though :p |
| 08:16.18 | salaud | Niklas-: Stick a NoOp(${somevar_or_expression}) in your dialplan |
| 08:16.39 | Niklas- | ah |
| 08:16.45 | salaud | then make sure you are at something like verbose 3 |
| 08:16.51 | Niklas- | yup got it to 4 |
| 08:16.55 | salaud | and watch the asterisk CLI |
| 08:17.09 | salaud | you will see the result of the var or expression in the NoOp() |
| 08:17.11 | phearless | ${EXTEN:1} is ${EXTEN} without the first char ? |
| 08:17.16 | Niklas- | greay thanks |
| 08:17.22 | Niklas- | great* |
| 08:17.25 | salaud | phearless: that seems right |
| 08:17.27 | phearless | where is the doc for this kind of tricks ? |
| 08:18.04 | salaud | phearless: http://www.voip-info.org/wiki/view/Asterisk+variables |
| 08:18.17 | salaud | phearless: look at substringgs |
| 08:18.25 | salaud | s/substringgs/substrings |
| 08:18.33 | Niklas- | oh heh didn't see you writing an example before :p |
| 08:19.06 | phearless | thanks salaud |
| 08:19.22 | salaud | phearless: npaa |
| 08:19.29 | phearless | BTW salaud in french is an insult |
| 08:19.38 | e-ddie | anyone knows why people fall back to the queue, when the person taking the call hangs up? |
| 08:19.39 | salaud | phearless: je sais bien |
| 08:19.46 | e-ddie | and what to do, to make it not do this |
| 08:20.24 | phearless | hehe ok salaud |
| 08:22.14 | Niklas- | Okay, i have this extension: exten => *83,2,Set(DB(FEAT/${CALLERIDNUM}/voicemail=1)), and right after that i have a: exten => *83,3,NoOp(${DB(FEAT/${CALLERIDNUM}/voicemail)}) - shouldn't the last one return '1'? |
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| 08:22.31 | NoRemorse | hi all |
| 08:22.42 | Dico_ | e-ddie, what do you mean ? |
| 08:22.43 | *** join/#asterisk voipmagic (n=voipmagi@myw-stp-196-34-112-59.sentechsa.net) |
| 08:22.50 | NoRemorse | does anyone know if it is possible to configure a secondary sip server on a cisco sip-ua? |
| 08:23.24 | phearless | is it possible to modify the menus in the Cisco 7960 ? |
| 08:23.38 | postel | phearless: yes, with ccm |
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| 08:24.42 | salaud | Niklas-: I guess it depends on what is in your DB, right? |
| 08:24.49 | salaud | can you verify the DB? |
| 08:25.22 | salaud | Niklas-: it should return the value from the DB |
| 08:25.27 | salaud | afaik |
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| 08:27.11 | Niklas- | yea i think it should, since i just wrote it into the DB :/ |
| 08:27.29 | Niklas- | and the db is in some wierd format (db1), which i dont know how to read from :p |
| 08:30.24 | phearless | postel: ok i am googleing "ccm" ... |
| 08:31.18 | phearless | Cisco Call Manager ? |
| 08:31.25 | phearless | but i use SIP/asterisk |
| 08:32.51 | postel | you cant with * |
| 08:32.54 | NoRemorse | does anyone know if it is possible to configure a secondary sip server on a cisco sip-ua? |
| 08:35.07 | [Outcast] | NoRemorse, as in a second line? |
| 08:35.19 | [Outcast] | or a backup server? |
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| 08:38.20 | phearless | so, with asterisk I can't add any features to the 7960 phone :-( |
| 08:38.35 | phearless | I would like for example to add softkey "buttons" |
| 08:39.17 | phearless | like modify [NewCall] to [VoiceMail] for example |
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| 08:40.35 | sconasq | phearless, u can make star codes |
| 08:40.37 | NoRemorse | backup server |
| 08:40.42 | sconasq | *1 for VM fo example |
| 08:40.45 | NoRemorse | I have 2 identically configured * servers |
| 08:40.58 | NoRemorse | and if one dies I want the 827 cpe to send calls to the 2nd |
| 08:41.00 | phearless | yes I would like to display it on the phone screen |
| 08:41.10 | phearless | sconasq. |
| 08:41.35 | voipmagic | won't use *1 if i were you , it's the default for automon.... |
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| 08:43.00 | Niklas- | Is it possible to have 1 extension that handles both "global" numbers like "XXXXXXXX" and local numbers like "XXXX" - in one extension? |
| 08:44.31 | sconasq | yes use a pattern |
| 08:44.51 | sconasq | exten => _X.,1,Macro(something) |
| 08:50.51 | NoRemorse | if I press # to transfer a call, then dial the call, how do i speak to the person I am transfering to before the b party gets xferred? |
| 08:50.58 | phearless | exten => i,1,Answer() |
| 08:50.59 | phearless | exten => i,2,Playback(privacy-incorrect) |
| 08:50.59 | phearless | exten => i,3,Hangup |
| 08:51.01 | phearless | is it right ? |
| 08:51.22 | phearless | because I never hear privacy-incorrect when I dial a random number |
| 08:51.25 | Niklas- | hmm not sure how to make that sconasq - would you be able to provide an example? |
| 08:51.28 | phearless | I got just a tone |
| 08:52.19 | sconasq | Niklas-, what action does this extension do? |
| 08:52.26 | salaud | phearless: I doubt you need to Answer() again |
| 08:53.28 | Niklas- | Call a SIP unit - right now i have 2 extensions per SIP unit, one with the local number extension (four digits) and one with the global number (eight digits) |
| 08:53.45 | phearless | salaud: same thing |
| 08:53.48 | phearless | I tried : |
| 08:53.56 | phearless | exten => i,1,Playback(privacy-incorrect) |
| 08:53.58 | phearless | exten => i,2,Hangup |
| 08:54.13 | phearless | and I got /var/lib/asterisk/sounds/privacy-incorrect.gsm |
| 08:54.44 | salaud | phearless: do a exten => i,1, NoOp(HeyIGotHere!) |
| 08:55.04 | salaud | just to make sure you actually getting to the 'i' extension correctly |
| 08:55.14 | phearless | good idea |
| 08:55.23 | salaud | unless, you can already tell by looking at the CLI that that part is working |
| 08:55.28 | salaud | ok |
| 08:55.49 | phearless | I do not get here |
| 08:56.07 | salaud | well... at least that's a start point... where do you go? |
| 08:56.45 | phearless | http://paste-bin.com/291 |
| 08:56.52 | phearless | this is my ext config file |
| 08:57.43 | phearless | 333 should be invalid for example |
| 08:58.33 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 08:58.48 | *** join/#asterisk oej (n=oej@cC3012269.inet.catch.no) |
| 08:58.55 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 09:00.30 | NoRemorse | how can I recall a parked call? |
| 09:00.53 | salaud | phearless: I can't tell why it wouldn't work... one thing MIGHT be the autofallthroguh |
| 09:01.06 | salaud | s/autofallthroguh/autofallthrough |
| 09:01.10 | NoRemorse | oh come on |
| 09:01.32 | sconasq | Niklas-, exten => 1234,Macro(std-exten,1234,SIP/1234) |
| 09:01.46 | salaud | phearless: that's new... but, where do you actually go? Can you tell... nowhere? |
| 09:01.51 | sconasq | Niklas-, exten => 99991234,Macro(std-exten,1234,SIP/1234) |
| 09:01.57 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 09:02.16 | Niklas- | Great thanks :) |
| 09:02.20 | salaud | NoRemorse: features.conf has settings for what keys recall a parked call... |
| 09:02.26 | NoRemorse | thanks |
| 09:02.34 | phearless | I have set "autofallthrough=no" and same problem |
| 09:02.52 | phearless | where do you actually go? <-- how can I know ? |
| 09:03.10 | salaud | NoRemorse: They may have created a new config file though... something like parking? |
| 09:03.37 | salaud | phearless: when you press the DTMF what do you see in the CLI as to where it might go next? |
| 09:03.45 | salaud | does it just not respond at all? |
| 09:04.00 | *** join/#asterisk vlt (n=dm@p54B34133.dip0.t-ipconnect.de) |
| 09:04.11 | salaud | phearless: nevermind... |
| 09:04.27 | salaud | phearless: you aren't doing an IVR... you are dialing these extensions directly |
| 09:04.31 | phearless | Desktop Management Task Force ? |
| 09:04.57 | salaud | phearless: DTMF... Dialtone Multi-Frequency... as in.. buttons on the phone |
| 09:04.58 | phearless | Interactive Voice Response |
| 09:05.20 | phearless | so when I dial 333, I see : |
| 09:05.51 | NoRemorse | are the #1, *2, and *8 parked call and xfer commands some sort of rfc or standard? |
| 09:05.59 | phearless | I see nothing in the CLI |
| 09:06.12 | salaud | NoRemorse: not really... |
| 09:06.36 | NoRemorse | so it's just an asterisk thing? |
| 09:06.45 | salaud | phearless: What do see in the client? 404? |
| 09:06.52 | salaud | NoRemorse: more or less |
| 09:06.55 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:06.59 | phearless | ouistiti*CLI> |
| 09:07.09 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 09:07.33 | salaud | phearless: I mean in your SIP client... your phone app.. does it say extension can't be found? |
| 09:07.48 | phearless | my cisco 7960 say "Reorder" |
| 09:08.20 | Niklas- | sconasq, in my std-exten content, what should i specify for the extenension? |
| 09:09.22 | salaud | phearless: I got It! |
| 09:09.24 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 09:09.33 | sconasq | Niklas-, the default std-exten code doesn't need to be changed |
| 09:09.33 | salaud | phearless: http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension |
| 09:09.49 | salaud | phearless: You would think that Asterisk would automatically jump to the 'i' extension if the client dials a number that is not matched by any other extensions. But it doesn't. |
| 09:10.06 | phearless | ok great |
| 09:10.18 | salaud | phearless: check the "Alternative to i" |
| 09:11.14 | salaud | I believe the 'i' extension only works in cases where you are inside the dialplan already... like a WaitExten() or Goto() |
| 09:11.27 | salaud | or maybe Dial(Local/123) |
| 09:11.40 | salaud | but, not in the SIP message |
| 09:13.22 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6EB47.versanet.de) |
| 09:14.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:14.58 | Niklas- | sconasq, i dont have any std-exten in my config here, thats why i'm asking :d |
| 09:18.05 | sconasq | qanyone using business edition? |
| 09:18.26 | sconasq | Niklas-, try stdexten |
| 09:18.36 | sconasq | u should see a [macro-stdexten] in the default extensions.conf |
| 09:18.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:20.20 | phearless | exten => _.,1,Playback(privacy-incorrect) |
| 09:20.29 | phearless | I have put this at the END of extensions.conf |
| 09:20.40 | phearless | is SEEMS to work... thanks salaud |
| 09:20.48 | salaud | phearless: great! |
| 09:21.12 | phearless | :-] |
| 09:24.01 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 09:30.02 | phearless | I found mine on google |
| 09:30.56 | pif | what keywords? |
| 09:31.23 | phearless | which one are you looking for ? |
| 09:31.32 | phearless | I needed 4 "steps" to upgrade my fw |
| 09:31.46 | pif | the 8.x series |
| 09:31.46 | pif | I'm at 7.5 |
| 09:31.51 | pif | yeah, it's a pain |
| 09:32.30 | phearless | http://www.cisco.com/pcgi-bin/Software/Tablebuild/doftp.pl?ftpfile=pub/voice/ip-phone/sip-7960/P0S3-08-2-00.zip&swtype=FCS |
| 09:32.37 | phearless | check this first |
| 09:33.47 | phearless | you will need P0S3-08-2-00.loads and P0S3-08-2-00.sb2 in the zip file |
| 09:34.05 | pif | "auth required" |
| 09:34.05 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:34.19 | pif | maybe they closed that hole |
| 09:34.48 | pif | phearless : is 8.x multilingual? |
| 09:35.27 | phearless | <pif> "auth required" |
| 09:35.32 | phearless | the login/pass is written |
| 09:35.36 | phearless | it is anonymous something |
| 09:35.39 | phearless | just read the page |
| 09:35.55 | phearless | " To download files, click on the link below and enter user name as anonymous and password as your email address. " |
| 09:36.01 | pif | oh! by ftp there is no auth! |
| 09:36.12 | pif | thanks |
| 09:36.22 | phearless | <pif> phearless : is 8.x multilingual? <--- no fw are multilangual I think |
| 09:36.42 | pif | oki |
| 09:37.00 | phearless | but I got a "Localization>Language" menu |
| 09:37.05 | phearless | but only english |
| 09:37.06 | pif | is 8.2 the latest? |
| 09:37.12 | pif | yeah, same on 7.x |
| 09:37.14 | phearless | I think so |
| 09:37.25 | pif | thanks again |
| 09:37.35 | *** join/#asterisk lehel (n=t@82.77.41.206) |
| 09:37.37 | phearless | I heard about 8.3 but I head that it does not work |
| 09:37.41 | phearless | cf http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 09:38.01 | *** join/#asterisk netf (i=netf@81.15.165.154) |
| 09:38.16 | phearless | ok there is a v8.4 in fact |
| 09:39.26 | *** part/#asterisk voipmagic (n=voipmagi@myw-stp-196-34-112-59.sentechsa.net) |
| 09:41.49 | x86 | anyone in the US, Canada, or UK with a fax machine, and want to help me test my IAXmodem setup? |
| 09:41.50 | netf | hello there. After I've changed NIC in a asterisk box it gives me messages like : "Unable to find a path from GSM to G729A" which basicaly means that license doesn't work. Should I buy a new license then? Or there is some other way to sort it out |
| 09:42.15 | x86 | netf: certain hardware configurations require you to basically re-activate your license keys |
| 09:42.20 | x86 | netf: contact digium |
| 09:42.46 | netf | x86: ok thx |
| 09:42.48 | *** join/#asterisk _gabry (n=gabry@host162-9.pool80181.interbusiness.it) |
| 09:42.50 | *** join/#asterisk jixi (n=damien@tcts.fpms.ac.be) |
| 09:43.04 | _gabry | hi all... |
| 09:43.17 | _gabry | brief question about queues... |
| 09:43.41 | jixi | hi, is there any way to all an user to take a specific call in a queue instead of the oldest one? |
| 09:44.04 | x86 | jixi: your question was not understandable |
| 09:44.27 | jixi | s/to all/to allow |
| 09:44.29 | jixi | (sorry) |
| 09:44.31 | _gabry | how can i play .gsm files in my pc? |
| 09:45.01 | jixi | say you have 5 people waiting in a queue, and an agent can decide to take the 3rd call instead of the 1st one |
| 09:45.20 | _gabry | no, you can't do this |
| 09:45.33 | _gabry | u go against the queue priority |
| 09:45.53 | jixi | exactly, that's what I'd like to do |
| 09:45.58 | jixi | however I was sure it was not possible |
| 09:46.04 | jixi | :-/ |
| 09:46.16 | _gabry | u have to modify source code |
| 09:46.41 | _gabry | or creating an application for this... |
| 09:46.48 | _gabry | like "ChangePriority" |
| 09:46.57 | _gabry | it's not simple |
| 09:47.00 | _gabry | :-) |
| 09:47.04 | jixi | indeed ;-) |
| 09:47.09 | _gabry | so my question... |
| 09:47.10 | jixi | thanks for the tip |
| 09:47.22 | _gabry | how can i hear my .gsm files? |
| 09:47.27 | jixi | you can play a gsm file on a pc with audacity |
| 09:47.34 | _gabry | oh, ok... |
| 09:47.46 | _gabry | and queue announcements can be modified? |
| 09:47.49 | _gabry | for example... |
| 09:48.03 | _gabry | the position announcement can be customized? |
| 09:48.11 | _gabry | with a .wav file |
| 09:48.13 | _gabry | ? |
| 09:49.19 | jixi | I think so :) |
| 09:49.44 | _gabry | cause i'm italian and i want to have announcements in italian... |
| 09:49.55 | _gabry | don't you know the way to do this? |
| 09:50.00 | *** join/#asterisk stoffell (n=stoffell@pot.catsanddogs.com) |
| 09:50.08 | phearless | exten => _.,1,Playback(privacy-incorrect) |
| 09:50.12 | phearless | it sucks in fact |
| 09:50.33 | _gabry | ? |
| 09:50.34 | phearless | it plays privacy-incorrect after each call !!!!!!!!!!!!!!! |
| 09:50.37 | jixi | _gabry: look at queues.conf, you can redefine all messages |
| 09:50.51 | phearless | this damn "i" = invalid, is broken |
| 09:50.59 | *** join/#asterisk fulgas (n=fulgas@80.172.227.30) |
| 09:51.01 | fulgas | morning |
| 09:51.26 | _gabry | <jixi>: for example: "queue-thankyou=file.wav" ? |
| 09:52.15 | jixi | _gabry: taht's what I had in mind, yes |
| 09:52.36 | _gabry | are u sure? now i could not try this.. ;-( |
| 10:02.20 | [shodan] | is there a way to mark a voicemail "urgent" ? (I think I read something about that in the docs .. ?) like I have an option in my menu to leave a voicemail , then I ask "is it urgent press 9 if yes , 1 if no or stay on the line" then I give voicemail instructions and I voicemail(s101) |
| 10:04.00 | *** join/#asterisk kartik (n=kart_@dialpool-210-214-11-50.maa.sify.net) |
| 10:04.59 | Niklas- | How can i make an if to check if a variable exists/is empty? ${var} = "" gives an error |
| 10:07.26 | [shodan] | what's the error ? |
| 10:08.07 | [shodan] | btw, I suggest something like lenght(variable) > 0 (not actual function) |
| 10:09.27 | e-ddie | how come the person calling into the queue ends up in the queue again, after the agent hanging up? |
| 10:09.47 | e-ddie | how can i prevent this from happening? |
| 10:11.14 | RoyK | e-ddie: that shouldn't happen |
| 10:11.26 | [shodan] | how are you sending the caller to the queue ? |
| 10:11.27 | e-ddie | it does |
| 10:11.51 | e-ddie | Queue(queuename|t|||360) |
| 10:12.09 | e-ddie | followed by hangup |
| 10:16.49 | e-ddie | so, it's answer,ringing,wait(2),queue(queuename|t||||360),hangup |
| 10:17.23 | e-ddie | any suggestions? |
| 10:18.15 | [shodan] | looks at the * console with lots of verbose to see what's going on |
| 10:18.43 | kaldemar | Niklas-: try "${var}x" = "x" |
| 10:22.08 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 10:24.24 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 10:26.37 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
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| 10:28.49 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 10:29.42 | x86 | anyone in the US, Canada, or UK with a fax machine, and want to help me test my IAXmodem setup? |
| 10:29.46 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 10:32.15 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:32.34 | *** join/#asterisk oej (n=oej@cC3012269.inet.catch.no) |
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| 10:41.56 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
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| 10:45.10 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 10:48.41 | phearless | http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension |
| 10:48.57 | phearless | this "i" feature is 100% broken |
| 10:49.07 | phearless | ok I forget it |
| 10:49.26 | phearless | is there any desktop integration between linux and asterisk ? |
| 10:49.52 | phearless | not an admin GUI ! but a software that show when my phone rang etc |
| 10:54.32 | stoffell | phearless: flash operator panel |
| 10:54.43 | Dr-Linux|work | FOP .... |
| 10:54.47 | Dr-Linux|work | i don't like FOP |
| 10:54.55 | stoffell | me neither :) |
| 10:55.11 | phearless | I mean a desktop app |
| 10:55.14 | phearless | not a web-app |
| 10:55.26 | phearless | my browser is not always running |
| 10:55.28 | RoyK | phalacee: astman, perhaps |
| 10:55.36 | RoyK | gastman |
| 10:55.58 | Niklas- | oki kaldemar thanks |
| 11:02.40 | [shodan] | is there a way to specify the format that voicemail as saved in , and the format that are sent by email ? they are sent in gsm but I'd rather use ulaw |
| 11:04.32 | kaldemar | [shodan]: http://www.voip-info.org/wiki-Asterisk+VoiceMail |
| 11:05.20 | [shodan] | ah , it's the first format that is sent by email ! |
| 11:05.21 | [shodan] | thanks ! |
| 11:07.13 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 11:10.43 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:20.01 | *** join/#asterisk negativecreep (n=xaeem@host210-2-170-89.isb.dancom.net.pk) |
| 11:20.19 | negativecreep | anyone had a chance to use uptech's iSurf 1004 IAD? |
| 11:20.34 | negativecreep | i am able to register the sip users but no tone on the phone. |
| 11:20.39 | negativecreep | very very weird. |
| 11:20.57 | negativecreep | the device registers the sip users with my * machine but no tone on the phone ports. |
| 11:21.09 | negativecreep | does this has anything to do with dtmf? |
| 11:24.42 | Dr-Linux|work | negativecreep, what's your SIP account entries in sip.conf for this user? |
| 11:24.48 | Dr-Linux|work | brb |
| 11:29.52 | *** join/#asterisk nailbags (i=someone@c220-237-123-137.randw1.nsw.optusnet.com.au) |
| 11:31.58 | *** join/#asterisk hotroot (n=michael@pD9E96DF6.dip.t-dialin.net) |
| 11:32.54 | hotroot | can someone tell me why asterisk do not use a stuttered dialtone on isdn-phones if there are new voicemails in the mailbox specified in zapata.conf? |
| 11:33.21 | *** join/#asterisk apardo (n=apardo@87.217.144.114) |
| 11:34.51 | negativecreep | Dr-Linux|work: http://xim.pastebin.co.uk/723 |
| 11:36.34 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 11:44.26 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:45.48 | *** join/#asterisk enots (i=dimka@freelsd.net) |
| 11:45.53 | Dr-Linux|work | Nebukadneza, sip looks fine to me |
| 11:49.18 | *** join/#asterisk preto (n=klaus@host98-128.pool82104.interbusiness.it) |
| 11:51.31 | preto | hi there.. Could someone tell how can i get the peer who answers a channel? Thanks |
| 12:02.37 | negativecreep | Dr-Linux|work: any ideas about what could be the issue with my IAD? |
| 12:03.29 | Dr-Linux|work | IAD? |
| 12:03.32 | Dr-Linux|work | what's your phone? |
| 12:03.32 | Nebukadneza | Dr-Linux|work: ? |
| 12:03.37 | negativecreep | Internet Access Device.. |
| 12:03.46 | negativecreep | its an iSURF1004 from Uptech. |
| 12:03.51 | negativecreep | like linksys or sipura |
| 12:03.59 | negativecreep | provides 4 analog ports. |
| 12:04.10 | Dr-Linux|work | negativecreep, can you dial to this phone? |
| 12:04.15 | negativecreep | no |
| 12:04.17 | Dr-Linux|work | or it can dial other phone? |
| 12:04.18 | negativecreep | it has a web interface. |
| 12:04.27 | negativecreep | no..there is just no tone. |
| 12:04.29 | Dr-Linux|work | Nebukadneza, sorry |
| 12:04.39 | Nebukadneza | :P |
| 12:04.41 | Nebukadneza | no prob |
| 12:04.42 | Dr-Linux|work | but it's registered? |
| 12:04.46 | negativecreep | yes |
| 12:04.49 | Dr-Linux|work | Nebukadneza, change your nick :P |
| 12:05.15 | Nebukadneza | *g |
| 12:05.16 | Dr-Linux|work | negativecreep, well, i think you will have to play with this device setting .. |
| 12:06.03 | negativecreep | Dr-Linux|work: i did. |
| 12:06.05 | negativecreep | :( |
| 12:08.34 | Dr-Linux|work | negativecreep, hhmm... maybe someone will be able to help you who uses this device, i never even heard it's name |
| 12:08.51 | Dr-Linux|work | the rest of your sip account stuff looks fine to me |
| 12:09.08 | Dr-Linux|work | negativecreep, i suggest do a SIP debug and see what you can see |
| 12:10.32 | DrukenHME | morning Dr-Linux|work |
| 12:11.56 | negativecreep | Dr-Linux|work: let me.. |
| 12:12.04 | Dr-Linux|work | DrukenHME, hey there |
| 12:12.39 | Dr-Linux|work | DrukenHME, few days back i seen your nick in my channel on dalnet |
| 12:12.54 | DrukenHME | sounds about right,... |
| 12:13.09 | DrukenHME | ya looked kinda lonely.. hehehe |
| 12:13.13 | DarKnesS_WolF | is there anohter cool management web ineterface for asterisk calls ? like flashoperator panal? |
| 12:13.51 | DrukenHME | DarKnesS_WolF: google? |
| 12:13.58 | negativecreep | Dr-Linux|work: http://xim.pastebin.co.uk/725 |
| 12:14.00 | negativecreep | have a look |
| 12:14.28 | Dr-Linux|work | negativecreep, ok |
| 12:14.31 | DarKnesS_WolF | DrukenHME: googling and checking voip-info i just think may be someone know a nice too l:) |
| 12:14.56 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 12:16.00 | DrukenHME | DarKnesS_WolF: :) i don't know of anything else, however i haven't been around in a long while |
| 12:19.22 | DrukenHME | Dr-Linux|work: your title on dalnet is outdated.... |
| 12:20.08 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:21.30 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 12:25.11 | Niklas- | Hmm, www.dundi.com - shouldn't that have something to do with dundi? :o |
| 12:30.06 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:30.16 | DrukenHME | hahaha digium fucked up their namevirtualhosts |
| 12:32.07 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
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| 12:47.47 | EzWayz | which fxo product do you suggest me ? |
| 12:48.10 | eldu | digium is a good start :) |
| 12:48.47 | EzWayz | hehe good ;) |
| 12:54.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:55.29 | Niklas- | Hmm, i'm not sure i get DUNDi correctly - i do have to configurere every non-local extension to use dundi? |
| 13:00.12 | *** join/#asterisk fonzai (n=tosalora@kosh.hut.fi) |
| 13:00.35 | fonzai | hello and good day to everyone |
| 13:03.34 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 13:09.42 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:13.14 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 13:14.28 | gaspiz | hi, does anyone know why dialstatus can be read correctly from agi only on hangup and not after the dial attempt |
| 13:14.35 | gaspiz | ? |
| 13:16.33 | BjornRobertsson | I am wondering if I should move from A@H 2.8 to trixbox, voice quality is ok but echo/reverb is driving everyone mad |
| 13:18.50 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:23.49 | pablus | morning |
| 13:29.34 | hotroot | evening |
| 13:30.37 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 13:30.57 | adamowitz | can I get some recommendations on WiFi telephones for use with Asterisk? |
| 13:31.18 | *** join/#asterisk seicherlbob (n=peter@62-99-165-26.dynamic.adsl-line.inode.at) |
| 13:31.43 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:31.53 | coppice | adamowitz: if you can use a DECT IP phone instead you will be happier |
| 13:32.08 | adamowitz | pointer? |
| 13:32.19 | seicherlbob | hi there! i have a problem with a solo-sip-asterisk. i can connect to my voip-provider but can't call my asterisk. it seems it keeps failing answering the calls |
| 13:32.28 | coppice | There is this thing called Google |
| 13:33.23 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 13:33.31 | adamowitz | Do you have a pointer to Google? (jj) Thank you coppice. |
| 13:33.50 | Muck- | two eicon diva server bri 2m pci in one pc are incompatible |
| 13:33.55 | coppice | its <<<<< that way |
| 13:33.59 | Muck- | and eicon won't provide a patch :( |
| 13:34.32 | coppice | wow, they've picked up the dialogic spirit quickly :-) |
| 13:35.26 | Muck- | is it a problem, if i get one BRI over CAPI and the other one over ZAPHFC |
| 13:35.29 | Muck- | ? |
| 13:35.31 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 13:37.53 | MRH2 | hi - is the asterisk init.d script supposed to restart asterisk if it crashes? |
| 13:38.17 | seicherlbob | has anybody ever done a pbx with only SIP? cause mine wont work. i could need some help. pls |
| 13:40.52 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net) |
| 13:41.03 | seicherlbob | it looks like i can connect to my VoIP provider but the asterisk wont answer when calling. |
| 13:47.12 | *** join/#asterisk anthonyl (n=Default@c-71-57-41-193.hsd1.il.comcast.net) |
| 13:52.46 | *** join/#asterisk kumbalae (n=suma@cm53.omega182.maxonline.com.sg) |
| 13:53.09 | *** join/#asterisk parag_ast (n=root@dxb-b18678.alshamil.net.ae) |
| 13:53.45 | kumbalae | hi all |
| 13:53.47 | parag_ast | Can anybody help me to test sip on another port then |
| 13:53.52 | parag_ast | 5060 |
| 13:53.59 | parag_ast | on asterisk |
| 13:58.04 | *** join/#asterisk I-MOD (i=opticron@c-71-207-209-230.hsd1.al.comcast.net) |
| 13:58.23 | tzafrir | MRH2, no, this is not the job of the init.d script |
| 13:58.37 | tzafrir | An init.d script should exit right after invoking it |
| 14:02.31 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:06.12 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.43.Dial1.SanJose1.Level3.net) |
| 14:09.49 | *** join/#asterisk preto (n=klaus@host98-128.pool82104.interbusiness.it) |
| 14:12.29 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:14.06 | *** join/#asterisk seicherlbob (n=peter@62-99-165-26.dynamic.adsl-line.inode.at) |
| 14:14.47 | seicherlbob | i got some massive problems with starting an asterisk by reading through o'reiley. can someone look at my sip.conf and extension.conf? |
| 14:15.09 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 14:19.35 | *** join/#asterisk jlatwiline (n=woodhead@h-72-244-145-197.phlapafg.covad.net) |
| 14:21.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:21.50 | *** join/#asterisk Film11 (n=film11@host86-130-141-190.range86-130.btcentralplus.com) |
| 14:22.04 | Film11 | Hm. |
| 14:22.06 | Film11 | Goats. |
| 14:22.07 | Ebola | Stalker |
| 14:22.08 | *** part/#asterisk Film11 (n=film11@host86-130-141-190.range86-130.btcentralplus.com) |
| 14:22.32 | puzzled | morning |
| 14:24.48 | seicherlbob | i got some massive problems with starting an asterisk by reading through o'reiley. can someone look at my sip.conf and extension.conf? it seems my server cant answer incoming calls |
| 14:24.54 | preto | Could someone tell me how could i get the peer who picked up the phone on a multiple dial? |
| 14:25.32 | *** part/#asterisk parag_ast (n=root@dxb-b18678.alshamil.net.ae) |
| 14:26.55 | preto | i'm still googling with no luck :( |
| 14:27.30 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 14:27.31 | *** join/#asterisk Egonis (n=chultay@207.245.14.10) |
| 14:27.59 | Egonis | I am trying to block all outgoing calls to 1-900 numbers, I am trying this: exten => 1900.,1,Playback(invalid) -- how do I make this work? |
| 14:28.20 | *** join/#asterisk jcaceres (n=jcaceres@200.48.115.191) |
| 14:28.21 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:28.31 | smackus | ok, I am trying to pass a PRI through asterisk into another phone system. On the incoming pri is it pri_cpe and out going pri_net or is it the other way around? |
| 14:28.43 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 14:28.48 | DasTech | morning |
| 14:29.33 | DasTech | how do I setup a *XX to voicemail and have it not require a password and have it match the exten |
| 14:29.40 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 14:29.43 | hmmhesays | morning folks |
| 14:30.17 | Egonis | how do I block an outgoing wildcard? i.e. 1900.......? |
| 14:31.40 | hmmhesays | um don't put that match in your dialplan? |
| 14:31.52 | Egonis | hmmhesays: I currently use _9. |
| 14:31.54 | hmmhesays | or send 1900 calls to an extension that plays an invalid message |
| 14:32.16 | Egonis | hmmhesays: And what command would I use to do that? I am trying exten => 1900NNNNNNN,1,Playback(invalid) |
| 14:32.20 | Egonis | but it doesn't work |
| 14:32.22 | hmmhesays | _9. would not match a _1900. call |
| 14:32.37 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) |
| 14:32.41 | hmmhesays | and 1900NNNNNNN is not valid |
| 14:32.42 | Madkiss | hi all. |
| 14:32.47 | Egonis | hmmhesays: So I should exten => _91900.,1,Playback(invalid)? |
| 14:32.52 | Madkiss | FreeBSD gives me the possibility to install asterisk or asterisk-bristuff. What's the preferred variant? |
| 14:33.08 | hmmhesays | Egonis I bet if your read the section on pattern matching that would be answered for you |
| 14:33.21 | Egonis | hmmhesays: Will do, ty |
| 14:33.31 | hmmhesays | You're pretty close |
| 14:33.44 | preto | Could someone tell me how could i get the peer who picked up the phone on a multiple dial? |
| 14:33.48 | hmmhesays | look up at the dp examples in the upper parts of extensions.conf |
| 14:33.59 | hmmhesays | preto you need to be more specific |
| 14:34.28 | *** join/#asterisk paryl (n=chatzill@www.admiralexpress.com) |
| 14:34.32 | paryl | hi guys |
| 14:35.05 | preto | well.. i need to get into a variable the sip peer who picked up the phone when dialing (for example) to SIP/1000&SIP/1001&SIP/1002 |
| 14:35.13 | preto | sorry but i'm not a native speaker |
| 14:35.32 | preto | i'll try my best to explain.. |
| 14:35.33 | hmmhesays | preto, try setvar in your sip.conf |
| 14:36.08 | preto | yes but i don't know how to identify the peer who answered |
| 14:36.08 | paryl | i've got a little issue with a remote system... they have 3 pots lines. calling out on those lines works, and calling in works fine, but they can't call their own local numbers... it just rings with no ring event in the logs |
| 14:36.38 | hmmhesays | depends on what you have the dialplan do for their own local numbers paryl |
| 14:36.59 | paryl | hmmhesays: they're just set up to dial out of Zap/g1 |
| 14:37.12 | paryl | like i said, it works fine dialling local numbers |
| 14:37.15 | paryl | just not their own |
| 14:37.17 | hmmhesays | set verbose 5 |
| 14:37.25 | paryl | i set verbose 50 :) |
| 14:37.32 | hmmhesays | and you get nothing? |
| 14:37.35 | paryl | the call goes out, gets put on the interface |
| 14:37.38 | *** join/#asterisk Bambr (n=Bambr@213-35-237-23-dsl.end.estpak.ee) |
| 14:37.41 | paryl | but no incoming ring event |
| 14:37.48 | paryl | they hear ringing |
| 14:37.51 | paryl | but that's it |
| 14:38.03 | hmmhesays | paryl: plug a regular telephone into one of the pots lines and try to dial one of their numbers |
| 14:38.41 | paryl | hrmm... i'm not physically there, and i'm pretty sure they don't have a regular telephone to plug in :\ |
| 14:38.45 | paryl | i'll try that though |
| 14:39.04 | *** join/#asterisk saftsack (n=saftsack@p54A7E0D2.dip.t-dialin.net) |
| 14:39.26 | hmmhesays | might be a telco dialing pattern issue, are they 7 or 10 digit dialing? |
| 14:39.26 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 14:40.57 | seicherlbob | i got some massive problems with starting an asterisk by reading through o'reiley. can someone look at my sip.conf and extension.conf? it seems my server cant answer incoming calls |
| 14:41.03 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:41.04 | hmmhesays | preto you can you ${CHANNEL} do I identify the current channel name |
| 14:41.27 | paryl | hmmhesays: 7 digit |
| 14:41.37 | preto | but if i do so i get the sip who initiated the call not the one who receives |
| 14:41.55 | preto | am i wrong? |
| 14:42.31 | hmmhesays | preto, you are probably right |
| 14:43.06 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:43.12 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 14:43.33 | hmmhesays | you could create a new variable in dial.c that identifies the answered party |
| 14:44.03 | preto | unmmm.. so i have to code uh? |
| 14:44.12 | hmmhesays | preto: also take a look at the source and README.variables |
| 14:44.31 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 14:44.55 | preto | ok i'll see. thanks. |
| 14:45.03 | hmmhesays | preto: yw |
| 14:45.10 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:45.15 | hmmhesays | paryl: and other locals have no problem calling into them? |
| 14:45.38 | *** join/#asterisk hwt (n=hwt@195.139.204.157) |
| 14:45.43 | paryl | hmmhesays: nope. they've actually been running just fine for about a year now. they just tried to call themselves :) |
| 14:45.56 | hmmhesays | paryl you want a quick fix? |
| 14:46.19 | hmmhesays | just change the dialplan so calls to themselves stay on the ip network |
| 14:46.25 | hmmhesays | simple and effective |
| 14:46.36 | paryl | true... i thought of that... |
| 14:46.51 | hmmhesays | any time you can keep the call off the pstn network you should anyway |
| 14:46.53 | paryl | i guess i just wondered if that was an indicator that something bigger was wrong |
| 14:46.55 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 14:47.04 | hmmhesays | keeps a line free |
| 14:47.05 | hwt | this is really strange. |
| 14:47.16 | hwt | after between 6 and 8 minutes during a call i get this: http://pastebin.ca/161515 |
| 14:47.24 | hwt | when i trace with tethereal |
| 14:47.33 | hwt | it's and asterisk peering with a Nortel MCS5000. |
| 14:47.49 | hmmhesays | someone trying to send dtmf via INFO? |
| 14:47.56 | hwt | apparently a sip info message causes the asterisk to hang up. |
| 14:48.03 | hmmhesays | i've had the problem before |
| 14:48.11 | hwt | nah, i've traced with ethereal and no dtmf. |
| 14:48.17 | hwt | and, i do dtmf inband. |
| 14:48.25 | hmmhesays | with audiocodes, audiocodes was to send dtmf via the an INFO message |
| 14:48.29 | hmmhesays | and asterisk would hang up |
| 14:48.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.154) |
| 14:49.00 | Assid | err whats RTT again? in iax2 show netstats? |
| 14:49.25 | Assid | round trip time right? |
| 14:49.25 | hwt | round trip time? |
| 14:49.29 | hwt | yeah. |
| 14:49.36 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:49.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:50.14 | hwt | hmmhesays: but no, there are no dtmf signaling. |
| 14:50.26 | hwt | hmmhesays: i have control of both the phones, and i don't touch them. |
| 14:50.28 | hmmhesays | find out why that nortel is sending out info messages then |
| 14:50.46 | hmmhesays | and crank your logging up in asterisk |
| 14:51.07 | hmmhesays | in logger.conf |
| 14:51.31 | kumbalae | hello, i would like to g729 works as listed in the lists ? |
| 14:51.56 | hmmhesays | it works as listed on the wiki |
| 14:52.16 | *** join/#asterisk markQing (n=mark@82-197-202-242.dsl.cambrium.nl) |
| 14:52.20 | kumbalae | oh it went to wiki too ? |
| 14:52.42 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 14:52.52 | hmmhesays | ~docs |
| 14:52.54 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 14:53.24 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:54.02 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 14:54.22 | hwt | hmmhesays: i have it all in the ethereal dump. |
| 14:54.25 | hwt | hmmhesays: http://pastebin.ca/161523 |
| 14:55.21 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 14:55.23 | hwt | astmasters looks a lot like assmasters. |
| 14:55.24 | hwt | :) |
| 14:56.40 | hwt | anyone? |
| 15:00.35 | hmmhesays | this isn't paid support hwt, be pushy and you'll get ignored |
| 15:00.56 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:01.21 | hmmhesays | set your logger.conf to display debug information |
| 15:04.00 | hwt | hmmhesays: yeah, sorry, i don't mean to be push. i'll check my logger settings. |
| 15:04.18 | *** join/#asterisk Tim__P (n=nospam@80.168.59.31) |
| 15:06.23 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 15:07.39 | Tim__P | Hi all. I've managed to get asterisk to automatically callout using a .call file, but the person it calls out to is not able to send DTMF back - ie - .call file calls a phone, and connects them to a meetme room, but the person is unable to enter pin numbers for press # to accept recordings etc. Is there something which needs enabling to allow called handsets to send tones? Thanks. (Gentoo Linux/ 1.2.9.1) |
| 15:07.58 | Tim__P | *or press |
| 15:08.51 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:09.43 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:11.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:12.02 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:15.37 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:16.13 | hmmhesays | Tim__P: your dtmf settings in sip.conf have to match what you have on your phone |
| 15:16.25 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 15:16.58 | preto | just a stupid question... the "callee" is whom receives the call? |
| 15:17.22 | Tim__P | if i dial in to my asterisk box, i can send DTMF fine - its only if asterisk initiates the call using /var/spool.... |
| 15:18.38 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:20.09 | hmmhesays | so when you have your call file Channel: SIP/FOO; Exten: My-Meetme it doesn't work? |
| 15:21.04 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:21.05 | Tim__P | it connects the called phone to the meetme room no problem, but that called phone is unable to enter a pin |
| 15:21.18 | Tim__P | asterisk is not receiving/ignoring the dtmf |
| 15:21.21 | [TK]D-Fender | Yay, Polycom SIP 2.0.1 released.... |
| 15:21.46 | hmmhesays | what are you using for dtmf in meetme? |
| 15:21.57 | Tim__P | if the phone dials into the meetme room itself, ie asterisk does NOT initiate the call, dtmf is fine |
| 15:22.27 | hmmhesays | initiate a call that sends your phone to a voice menu after answering |
| 15:22.35 | hmmhesays | see if it is a meetme specific problem |
| 15:22.52 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:23.01 | hmmhesays | make sure you are using the same extension to test a call to meetme as the callfile is using |
| 15:23.49 | Tim__P | ok just done that... same issue - dtmf not understood. meetme aint the problem |
| 15:24.09 | hmmhesays | how did you do that? |
| 15:24.11 | RoyK | "Donald Rumsfeld briefed the President this morning. He told Bush that 3 Brazilian solders were killed in Iraq. To everyone's amazement, all the color drained from Bush's face. Then he collapsed onto his desk, head in hands, visibly shaken, almost in tears. Finally, he composed himself and asked Rumsfeld, "Just exactly how many is a Brazilian?"" |
| 15:24.25 | Tim__P | commented out the meetme statement and just put an authenticate command in |
| 15:24.33 | tzanger | RoyK: hahaha yes I've heard that one before... |
| 15:24.36 | tzanger | awesome |
| 15:24.54 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:24.57 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:25.46 | hmmhesays | Tim__P: just for the hell of it set your dtmf settings in the general section of sip.conf the same as you have set for your phone |
| 15:26.18 | *** join/#asterisk lordbaron (n=redbaron@63.113.144.67) |
| 15:27.11 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:28.28 | lordbaron | I have a live server that has a span defined of channels 1-72. I am having a problem with a t1, channels 25-49. I want to stop any new calls on these channels. How can I do this? |
| 15:28.40 | lordbaron | I have live calls on some of those, so I don't want to kill those live calls |
| 15:29.03 | hmmhesays | lordbaron: remove them from your span |
| 15:29.06 | Tim__P | hmmhesays: in [general] of sip.conf i put dtmfmode=rfc2833 - no change |
| 15:29.06 | phearless | =QUESTION= |
| 15:29.25 | phearless | can I use many wireless VOIP phones on the same AccessPoint ? |
| 15:29.36 | hmmhesays | Tim__P: set it in your phone entry too |
| 15:29.38 | lordbaron | hmmhesays: will a reload do this? I thought zaptel.conf did not work with a reload |
| 15:29.44 | hmmhesays | phearless: yes |
| 15:29.54 | hmmhesays | run ztconfig |
| 15:30.00 | phearless | hmmhesays: ok |
| 15:30.06 | phearless | =QUESTION #2= |
| 15:30.24 | phearless | which VoIP wireless phones should I buy ? |
| 15:30.26 | Tim__P | hmmhesays: erm...the phone i'm dialing is a mobile via a sip provider... how do you mean? |
| 15:30.40 | phearless | many wifi phones got a baaaaad quality |
| 15:30.50 | hmmhesays | you should have a sip.conf entry for your voip provider |
| 15:30.58 | hmmhesays | phearless: wip300 is ok |
| 15:31.31 | [TK]D-Fender | phearless: Unless you really need WiFI, you're much better off with ATA's & normal cordless phones |
| 15:31.34 | phearless | I will hqve a look |
| 15:31.43 | phearless | I need cordless yes |
| 15:31.45 | phearless | not wifi |
| 15:31.59 | [TK]D-Fender | phearless: Gete an ATA and a normal analog cordless phone then |
| 15:32.03 | phearless | just cordless/wireless |
| 15:32.06 | hmmhesays | cheaper |
| 15:32.09 | phearless | what is an ATA ? |
| 15:32.26 | [TK]D-Fender | phearless: Analog Terminal Adapter. Look for the SPA-2002 |
| 15:32.30 | lordbaron | hmmhesays: where is ztconfig? My system does not have it |
| 15:32.42 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:34.18 | phearless | [TK]D-Fender: I do not really understand what is the SPA-2002 |
| 15:34.23 | phearless | you plug all the phones on it ? |
| 15:34.29 | phearless | but I need to buy phones |
| 15:34.37 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-b47bba3a435225b9) |
| 15:34.41 | lordbaron | ok, found ztcfg. Does executing this cause a reload? |
| 15:34.45 | [TK]D-Fender | phearless: You plug any normal phone into it and it becomes a SIP device. |
| 15:34.45 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:35.05 | phearless | ah ok but I don't have normal phones |
| 15:35.09 | tzafrir | lordbaron, zaptel.conf is the configuration file of ztcfg . IT is "reloaded" by runinng ztcfg |
| 15:35.18 | phearless | I need cordless/wireless SIP phones |
| 15:35.18 | lordbaron | thx |
| 15:35.28 | hmmhesays | phearless you need to read a little more |
| 15:35.28 | tzafrir | lordbaron, why do you need to reload? |
| 15:35.38 | tzafrir | what did you change? |
| 15:35.39 | hmmhesays | to change what channels are in his span |
| 15:35.55 | lordbaron | yes, need to stop channels 25-49 from being used |
| 15:35.57 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 15:36.01 | tzafrir | That would require a restart of asterisk (in 1.2) |
| 15:36.05 | [TK]D-Fender | phearless: Perhaps you have a hearing disorder... I said go BUY an ATA & normal cordless phone for yoru wireless needs! |
| 15:36.23 | phearless | but it is better than just sip phones ? |
| 15:36.26 | tzafrir | to stop channels from being used, just destroy them from the CLI |
| 15:36.31 | *** join/#asterisk truz_`24 (n=truz_`24@74.129.166.232) |
| 15:36.37 | tzafrir | (without restarting anything) |
| 15:36.39 | hmmhesays | tzafrir: you can't just change channel => ? |
| 15:36.44 | truz_`24 | using asterisk, are you able to get real time status ? |
| 15:36.52 | hmmhesays | tzafrir: he wants to stop calls from going out those channels in the future |
| 15:37.01 | tzafrir | It is perfectly well to have a channel configured in zaptel.conf and not in zpata.conf |
| 15:37.07 | truz_`24 | How many lines are on hold and etc... so a gui status interface can be created using asterisk in the background? |
| 15:37.08 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:37.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:37.26 | tzafrir | so remove them from zapata.conf. No need to touch zaptel.conf |
| 15:37.38 | [TK]D-Fender | phearless: (still not sinking in) You want a wireless solution. If you wanted a wired SIP solution I'd have just suggested a Polycom IP phones. you need cordless, that leaves 2 real options. 1. WifI SIP phone. They all suck (really). or 2. Normal cordless phone using an ATA to become a SIP device. |
| 15:37.41 | lordbaron | I have about 4 calls on these channels. Group 1 is defiend in zapata.conf as channels 1-72 |
| 15:37.49 | lordbaron | I want to change the definition in zapata.conf |
| 15:37.53 | lordbaron | will a reload in asterisk work? |
| 15:38.02 | phearless | http://www.sipura.com/products/spa2002.htm is just for 2 phones, and I need ~12 phones |
| 15:38.11 | phearless | thanks for the explanation, [TK]D-Fender |
| 15:38.15 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:38.31 | phearless | so it will be a hard task to find a good solution |
| 15:38.44 | tzafrir | truz_`24, the manager interface is generally more useful |
| 15:38.47 | [TK]D-Fender | phearless: I jsut gave your the model # to get.... wheres the trouble? |
| 15:39.00 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:39.01 | hmmhesays | phearless buy a 12 port or 24 port ata |
| 15:39.10 | truz_`24 | tzafrir, I see. |
| 15:39.11 | phearless | okay |
| 15:39.12 | [TK]D-Fender | phearless: http://www.voipsupply.com/product_info.php?products_id=713 |
| 15:39.28 | RoyK | phearless: best would prolly be to get a T1 card and a channel bank |
| 15:39.34 | wunderkin | yey i just ordered a poly 430 :D |
| 15:39.39 | RoyK | phearless: well-proven stuff |
| 15:39.51 | tzafrir | lordbaron, no. A reload won't work. But if you want to avoid restarting asterisk, destroy the extra channels: zap destroy channel NNNN |
| 15:39.54 | rpm | does channel hinting not work with realtime in the asterisk 1.2.x branch? |
| 15:40.18 | [TK]D-Fender | phearless: well you could by 6 of those (2 ports each), or get a BIGGER device like :http://www.voipsupply.com/product_info.php?products_id=207 |
| 15:40.22 | tzafrir | lordbaron, on 1.4 you'll have 'zap restart' (but it disconnects all calls) |
| 15:40.34 | [TK]D-Fender | wunderkin: Congrats, great little phone |
| 15:40.53 | wunderkin | someday i will have a 501 and 601 |
| 15:41.07 | lordbaron | tzafrir: will redefining zapata.conf in 1.2.11 work without the restart? I want to redefine groups |
| 15:41.21 | phearless | $1,499.95 |
| 15:41.23 | phearless | !!!!!!!!!!!!!!!!!!!!!!!!!! |
| 15:41.29 | [TK]D-Fender | phearless: Avoid the channel bank approach. Can work, but costs more in the end for your needs, places a higer load on your server, and is far less versatile in your case. You would lose much sanity in the process. |
| 15:42.06 | *** part/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 15:42.23 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 15:42.24 | [TK]D-Fender | phearless: Thats for supporting 24 phones though, so $1500 isn't that bad. If you're just talking 12 I might be so cheap as to say just buy the 6 x SPA-2002's for 414$ they'll cost you total. |
| 15:43.01 | elriah | Hey guys, isn't mixmonitor (* 1.2.x) supposed to record both ingoing and outgoing streams? Mine is only recording the outgoing streams, nothing on the ZAP channel is making it into the recordings... |
| 15:43.17 | [TK]D-Fender | wunderkin: I own a 301, 430, and 501 personally at home. |
| 15:43.27 | [TK]D-Fender | wunderkin: And have 600/601's here at work. |
| 15:43.38 | hmmhesays | get a mediatrix 1124 if you need 24 phones |
| 15:43.41 | hmmhesays | you'll not regret it |
| 15:44.02 | wunderkin | cool |
| 15:44.12 | [TK]D-Fender | phearless: Indeed, as hmmhesays said that is a great model, though it may cost you more than the AudioCodes, its worth a bit more. |
| 15:44.33 | hmmhesays | very much so |
| 15:44.57 | rpm | i hate configuring those mediatrix 1124's via snmp.. and the web interface is slow and alot of configurable settings aren't in it. |
| 15:45.07 | RoyK | phearless: you can get channel banks cheap from ebay, and a single E1 or T1 from sangoma or digium isn't that expensive |
| 15:45.23 | hmmhesays | rpm: auto config |
| 15:45.26 | [TK]D-Fender | wunderkin: Though mind you the SPA's are still much cheaper and I must say ARE more versatile per-port that a big unit, just more to manage however. |
| 15:45.44 | elriah | Anyone? MixMonitor? |
| 15:46.07 | toerkeium | hello guys, does anyone know what or who "myvoiceline.com" is? |
| 15:46.08 | hmmhesays | rpm: they were more made to grab config files, you would never provision a group of 1124's by hand |
| 15:46.15 | RoyK | what's so cool about mixmonitor compared to the old monitor? |
| 15:46.28 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:46.30 | [TK]D-Fender | RoyK: Any channel bank scenario will end up costing him as much as a SIP gateway while placing higer load on * and requiring all sort of stupid Dial paramters to tell * how to bloody well let him handle the petty stuff like transferring.... |
| 15:46.35 | [TK]D-Fender | RoyK: *ICK* |
| 15:46.40 | elriah | It mixes both channels into one wave file so you can hear the entire call without shelling out to mix them afterwards ... (i.e., less cpu, etc) |
| 15:47.02 | RoyK | [TK]D-Fender: eh. ok |
| 15:47.02 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:47.11 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-25ee48d8c0354e32) |
| 15:47.11 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:47.15 | [TK]D-Fender | phearless: Avoid ANY Zaptel device for FXS at all cost. |
| 15:47.26 | RoyK | :) |
| 15:47.28 | phearless | ok ! |
| 15:47.36 | Cresl1n | file!!! |
| 15:47.39 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 15:47.42 | RoyK | [TK]D-Fender: it's that bad? |
| 15:47.44 | [TK]D-Fender | phearless: you don't want to be stuck with a million dial options to handle all your transfers, etc.... |
| 15:47.57 | fourcheeze | is there some secret to dialling h323 (chan_h323.so)? |
| 15:48.03 | fourcheeze | trying to talk to an avaya |
| 15:48.10 | fourcheeze | currently getting this: |
| 15:48.11 | fourcheeze | No translator path exists for channel type H323 (native 4) to 256 |
| 15:48.30 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:48.32 | [TK]D-Fender | RoyK: " to transfer? Dial(Zap/12,tTwW-ABCDE FUCKING-G) |
| 15:48.48 | [TK]D-Fender | RoyK: ICK!!!! |
| 15:48.58 | RoyK | :) |
| 15:49.21 | fourcheeze | exten => 123,3,Dial(H323/some.ip.num.ber/200) |
| 15:49.27 | [TK]D-Fender | RoyK: All SIP gateways do all the dirty work and do local conferencing, and do it all with hook-flash etc and don't require you to be in T1 wiring range of your server and offer failovers, etc.... |
| 15:49.55 | RoyK | [TK]D-Fender: but SIP gateways are for cowards! |
| 15:50.06 | [TK]D-Fender | RoyK: Oh you mean sane & smart people! ;) |
| 15:50.19 | RoyK | hehe |
| 15:50.45 | [TK]D-Fender | RoyK: Not to mention cost-consious. The only Zaptel is preferrable is native bridge for analog faxes which is something Digium dissavows anywyas.... |
| 15:50.53 | [TK]D-Fender | time* |
| 15:51.14 | phearless | brimstone: I cant understand what is tdm2400e |
| 15:51.25 | RoyK | ...and with openpbx now supporting t.38 endpoints..... |
| 15:51.31 | elriah | Ahh.. figured it out... |
| 15:51.33 | [TK]D-Fender | phearless: Save yourself the trouble and just go with the SPA-2002.... really... |
| 15:51.45 | brimstone | phearless, one of the digium cards, 24 analog ports |
| 15:51.49 | phearless | I am afraid by "adapters" |
| 15:52.07 | phearless | it seems that it will make the things more complex |
| 15:52.12 | RoyK | phearless: I would guess you should listen to mr [TK]D-Fender |
| 15:52.20 | [TK]D-Fender | phearless: I've deployed all of the kinds of units discussed here including channel banks.... |
| 15:52.35 | phearless | ok I will continue to read about this |
| 15:52.45 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:53.08 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:53.33 | [TK]D-Fender | phearless: I've handled the AudioCodes & Meditrix gateways, SPA's all over the plave and my customers have been plagued by their WifI phones.... you want a small pile of cordless phones for your office? ATA + normal analog phones. does the job just great.... |
| 15:54.04 | phearless | what is SPA ? |
| 15:54.11 | RoyK | ~SPA |
| 15:54.20 | jbot | extra, extra, read all about it, spa is the Software Publishers Association |
| 15:54.20 | phearless | I do not know AudioCodes and Meditrix too |
| 15:54.20 | [TK]D-Fender | phearless: SPA-2002 and the rest of that product line from Linksys/Sipura |
| 15:54.24 | phearless | ah okay |
| 15:54.36 | [TK]D-Fender | place* |
| 15:54.48 | RoyK | jbot: SPA is also SPA-xxxx from linksys/sipura |
| 15:54.52 | jbot | okay, RoyK |
| 15:54.59 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 15:55.00 | phearless | ~SPA |
| 15:55.01 | jbot | [spa] the Software Publishers Association. SPA-xxxx from linksys/sipura |
| 15:55.01 | Tim__P | phearless: http://www.voiptalk.org/products/Linksys+SPA+2002 |
| 15:55.04 | Tim__P | read it |
| 15:55.14 | phearless | ok |
| 15:56.09 | RoyK | jbot: SPA is also Health And Beauty Treatment For Old And Tired Asterisk Hackers |
| 15:56.12 | jbot | okay, RoyK |
| 15:56.17 | RoyK | :) |
| 15:56.47 | Tim__P | right, i still need some advise. DTMF not getting through from from client to asterisk, but ONLY IF asterisk initiates the call using a .call file. If the client dials in manually, DTMF is interperated fine. Anyone seen this before? |
| 15:57.19 | E-bola | hi |
| 15:57.49 | *** join/#asterisk c4t3l (n=c4t3l@cpe-70-116-139-170.houston.res.rr.com) |
| 15:58.19 | fourcheeze | ok, I'm now officially annoyed with h323 |
| 15:58.34 | fourcheeze | is there some odd thing I need to do to make * dial out on it? |
| 15:58.53 | fourcheeze | chan_h323.so is loaded and incoming works |
| 15:58.55 | h3x | h323 sucks |
| 15:59.02 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 15:59.03 | fourcheeze | yeah it may suck but it's all avayas talk |
| 15:59.09 | MikeJ | h323 doesn't suck. |
| 15:59.21 | fourcheeze | MikeJ: so how can I dial out? |
| 15:59.22 | RoyK | h323 + asterisk sucks |
| 15:59.27 | eKo1 | hehehe |
| 15:59.41 | h3x | this is true |
| 15:59.45 | MikeJ | using h323, pretty similar to sip, but using h323 signalling |
| 16:00.01 | h3x | too bad openh323 sucks |
| 16:00.01 | RoyK | MikeJ: theory != practice |
| 16:00.07 | fourcheeze | well I have an avaya IP office sitting there listening on tcp:1720 |
| 16:00.17 | fourcheeze | and I have an asterisk server |
| 16:00.19 | hmmhesays | heh openh323 doesn't suck either |
| 16:00.23 | MikeJ | openh323 doesn't suck either.. the asterisk impelmentations do. |
| 16:00.29 | MikeJ | try woomera with chan_woomera |
| 16:00.30 | h3x | openh323 still sucks |
| 16:00.32 | h3x | its huge |
| 16:00.34 | h3x | it takes hours to compile |
| 16:00.34 | fourcheeze | what's the magic solution to get one to dial? |
| 16:00.35 | hmmhesays | if it wasn't for openh323 voip probably would not be where it is today |
| 16:00.52 | MikeJ | it is large, but that doesn't mean it sucks, it means it's large... |
| 16:00.55 | hmmhesays | h3x: heh, wow |
| 16:01.03 | hmmhesays | what MikeJ said |
| 16:01.08 | MikeJ | if it takes you |
| 16:01.15 | RoyK | h3x: it takes quite some time to build a full linux kernel with all drivers, but it doesn't make it suck |
| 16:01.30 | MikeJ | "hours" to compile, then you might want to get rid of that P II 400 |
| 16:01.33 | hmmhesays | just because it you don't know what it can do doesn't mean it sucks |
| 16:01.39 | *** join/#asterisk remiss (i=bofh@46.80-203-38.nextgentel.com) |
| 16:01.59 | MikeJ | I highly suggest woomera |
| 16:02.00 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 16:02.05 | RoyK | "unix sucks because it doesn't have windows" |
| 16:02.06 | RoyK | :) |
| 16:02.34 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:02.34 | h3x | it took 3 hours to compile on a p4 3ghz |
| 16:02.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:02.42 | h3x | 2 meg cache |
| 16:02.44 | RoyK | h3x: 128MB RAM? |
| 16:02.48 | h3x | 2 gb ram |
| 16:02.50 | carrar | Windows sucks because it does,'t have X |
| 16:02.54 | lordbaron | can anyone confirm that reloads on zapata.conf should be in 1.2.11? I am trying to redefine the channels within a group. This bug says that it is 1.2 and later http://bugs.digium.com/view.php?id=5508 |
| 16:02.55 | h3x | have you tried to do it :P |
| 16:02.56 | MikeJ | h3x, then your machine is badly broken.. |
| 16:03.06 | h3x | it probably just had optimizations cranked up |
| 16:03.12 | MikeJ | carrer, windows has an X in the upper right hand corner of every windows ! |
| 16:03.29 | carrar | heh |
| 16:03.40 | h3x | but i cant imagine running it without optimizations |
| 16:03.45 | Filar | dude 3 hours |
| 16:03.50 | MikeJ | and in fact, there is both commercial and free x as you were saying for windows |
| 16:04.00 | Filar | thats crazy |
| 16:04.09 | carrar | for windows, but it isn't windows |
| 16:04.15 | fourcheeze | what's woomera? |
| 16:04.28 | Tim__P | DTMF problem FIXED: cannot use RFC method. "inband" works fine. thanks all, thanks hmmhesays. |
| 16:04.46 | h3x | I can make world in freebsd faster than building h323 |
| 16:04.47 | hmmhesays | Tim__P: np |
| 16:04.51 | h3x | OpenH323 |
| 16:04.53 | lordbaron | h3x: yes |
| 16:04.55 | MikeJ | carrar, huh/ |
| 16:05.25 | MikeJ | fourcheeze, http://www.pbxfreeware.org/chan_woomera/ |
| 16:05.48 | h3x | woomera does 323? |
| 16:06.06 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 16:06.07 | MikeJ | openh323 has a woomera server |
| 16:06.17 | h3x | oh ok |
| 16:06.25 | fourcheeze | MikeJ: does this help me dial my avaya with h323? |
| 16:06.25 | h3x | well im sure thats a better way to do it |
| 16:06.26 | MikeJ | I think craig is finishing off the one for opal now, not sure if it is in tree yet |
| 16:06.38 | MikeJ | fourcheeze, probably |
| 16:06.51 | hmmhesays | if your avaya uses h323 |
| 16:07.13 | hmmhesays | you could try any of the various h323 channels |
| 16:07.25 | fourcheeze | what does "Sep 5 16:45:28 WARNING[10924]: channel.c:2541 ast_request: No translator path exists for channel type H323 (native 4) to 256" mean? |
| 16:07.45 | hmmhesays | fourcheeze no path from g729 to ulaw |
| 16:07.50 | fourcheeze | ahh |
| 16:07.55 | fourcheeze | ok now we're getting somewhere |
| 16:08.03 | fourcheeze | I want it to call with g729 |
| 16:08.20 | fourcheeze | so I don't mind the lack of translator |
| 16:08.38 | MikeJ | ok.... does that stupid warning still not translate the numbers to the codec names... |
| 16:08.39 | fourcheeze | can I just tell it to carry on and do its thing |
| 16:08.40 | MikeJ | dumb |
| 16:09.02 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:09.09 | MikeJ | fourcheeze, it's doing it's thing.. |
| 16:09.10 | RoyK | hi. can't compile trunk on os x: http://pastebin.ca/161569 |
| 16:09.55 | file | RoyK: it tells you what to do |
| 16:09.56 | fourcheeze | MikeJ: how do I tell it to use a particular codec when I call a particular IP number? |
| 16:09.58 | *** join/#asterisk ErickBono (n=adada@200.77.71.186) |
| 16:10.33 | file | RoyK: the easiest way is to remove menuselect.makeopts and have it be regenerated |
| 16:10.33 | hmmhesays | well if the ip number has a peer entry you can set codecs per peer |
| 16:10.44 | RoyK | file: trying.... |
| 16:11.14 | syzygyBSD | fourcheeze: a particular ip number... do you mean sip extension? |
| 16:11.22 | RoyK | file: i tried running make menuselect but it didn't want to, but removing that file helped |
| 16:11.26 | syzygyBSD | iax, whatever |
| 16:11.38 | fourcheeze | syzygyBSD: no this is an avaya ip office listening for h323 |
| 16:11.57 | syzygyBSD | fourcheeze: how are you calling it? |
| 16:12.18 | fourcheeze | well I was doing Dial(H323/ip.nu.mb.er/200) |
| 16:12.20 | *** part/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 16:12.55 | syzygyBSD | hmm, never seen that before, maybe someone else can help |
| 16:13.13 | fourcheeze | well it didn't work :-) |
| 16:14.45 | fourcheeze | hmm |
| 16:14.52 | fourcheeze | hmmhesays: so if I create a peer thus: |
| 16:14.57 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:14.57 | fourcheeze | host=200.0.2.215 |
| 16:14.57 | fourcheeze | disallow=all |
| 16:14.57 | fourcheeze | allow=g729 |
| 16:15.46 | fourcheeze | where I put my actual ip number in there |
| 16:15.49 | fourcheeze | that shold work? |
| 16:16.21 | RoyK | you can set it as dynamic, of course |
| 16:16.53 | fourcheeze | I've tried disallow=all and allow=g729 in my global h323 config but that doesn't seem to help |
| 16:18.02 | hmmhesays | I think you need to stop and start asterisk to reload chan_h323 |
| 16:18.13 | fourcheeze | ok |
| 16:18.26 | RoyK | unload chan_h323 |
| 16:18.30 | RoyK | load chan_h323 |
| 16:19.30 | phearless | what do people think about the Linksys SPA 942 ? (not-cordless SIP phone) |
| 16:20.24 | E-bola | very discontent with ours |
| 16:20.30 | phearless | discontent? |
| 16:20.34 | E-bola | im not certain if its the phones or our setup though |
| 16:20.38 | E-bola | ya there is alot of echo |
| 16:21.10 | E-bola | which seam to be alot worse when using the 942 comapred to a headset and a sooftphone |
| 16:21.14 | E-bola | actualy we have the 922 i think |
| 16:21.18 | E-bola | but its pretty much the same model |
| 16:21.54 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 16:22.02 | P4C0 | hello guys, I'm having a nat problem here :( |
| 16:22.07 | phearless | ok E-bola |
| 16:22.21 | hmmhesays | P4C0: fun |
| 16:22.26 | phearless | P4C0: try IPv6 ;-) |
| 16:22.59 | Dr-Linux|work | any sipura expert around? :) |
| 16:23.06 | fourcheeze | hmmhesays: RoyK: nope I've unloaded and reloaded h323 and restarted asterisk and neither time has it helped |
| 16:23.12 | P4C0 | :( but it's strange... I have all port forwarded in my firewall... (asterisk server is inside local network) but went a new call get's in I can't heard it... (but they hear me) |
| 16:23.27 | fourcheeze | so I now have an incoming g729 call |
| 16:23.32 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 16:23.33 | fourcheeze | on iax |
| 16:23.47 | P4C0 | rtp debug ip (server provider ip) doesn't show any recieved packages :( |
| 16:24.00 | fourcheeze | and still it complains I'm trying to convert to ulaw despite the fact that h323.conf has it disallowed |
| 16:24.31 | P4C0 | when the call is made from the outside it works fine..., so the problem is with outgoing calls... but not sure why! |
| 16:26.27 | P4C0 | does anyone know why this may be happening? seems that my firewall/router is not forwarding incoming rtp data |
| 16:26.38 | MikeJ | fourcheeze, which h323 module? |
| 16:26.46 | fourcheeze | chan_h323.so |
| 16:26.57 | MikeJ | oh.. dunno.. sorry. |
| 16:27.17 | MikeJ | actually, come to think of it, I don't recall of woomera will let you do passthrough properly either |
| 16:27.26 | MikeJ | you'll have to check |
| 16:27.36 | fourcheeze | ahh so I need a license? |
| 16:28.08 | MikeJ | there is no technical reason it's not possible |
| 16:28.19 | MikeJ | just not sure what supports it |
| 16:29.37 | P4C0 | does anyone here have similar setup than I? (asterisk server inside private network?) |
| 16:29.46 | *** join/#asterisk saftsack (n=saftsack@p54A7D51F.dip.t-dialin.net) |
| 16:35.17 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:36.15 | P4C0 | what can I do :( |
| 16:37.32 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:38.18 | P4C0 | is there a way to try to debug this or find possible reasons why it's not working? |
| 16:38.34 | Tim__P | P4C0: its because the audio and control stream are separate, and the audio part of that (your rtp) hates NAT - even if its fowarded |
| 16:38.51 | Tim__P | avoid NAT, otherwise look up how to use STUN servers |
| 16:39.17 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 16:39.51 | P4C0 | Tim__P, can I use stun with asterisk? (all client are inside nat and work fine calling between them) server provider is outside |
| 16:40.29 | Tim__P | P4C0: gtg. hope all works out. beleive me i had the same problem - and theres really not easy workaround. you will save yourself hours of time is you just bind your asterisk server to a publicly available IP, no NAT at-all |
| 16:40.38 | Tim__P | yes, asterisk supports STUN. |
| 16:41.07 | P4C0 | Tim__P, thanks, but I can't do that... :( so I'll try to find a way to use stun with asterisk... :S |
| 16:41.14 | P4C0 | let's see how it goes, thanks |
| 16:41.15 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 16:41.26 | Tim__P | ok. check out voip-users |
| 16:41.35 | Tim__P | some good reading about STUN and asterisk there |
| 16:41.42 | Tim__P | google it. |
| 16:41.46 | Tim__P | bye! |
| 16:42.03 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 16:42.14 | *** part/#asterisk jlatwiline (n=woodhead@h-72-244-145-197.phlapafg.covad.net) |
| 16:42.18 | Tim__P | yeah and thanks again for everyone elses help earlier with my prob. ta. |
| 16:44.02 | h3x | didnt you mean voip-info |
| 16:45.29 | Ryushin | It looks like Polycom has release new bootrom 3.2.2 and firmware 2.0.1. Does anyone have access to this right now? |
| 16:46.27 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 16:48.05 | Ciber311 | hmm |
| 16:48.14 | rpm | does anyone use realtime sip users and channel hinting in asterisk 1.2? |
| 16:48.35 | Ciber311 | gonna look for it now Ryushin |
| 16:48.53 | SplasPood | What would everyone recommend for a conference room phone.. Specifically one with a limited amount of cabling due to a non-drillable table surface |
| 16:49.10 | SplasPood | Ryushin: 2.0.1 ? hrm, lemme look |
| 16:49.23 | Ciber311 | SplasPood: one of the polycom ones? :P |
| 16:49.59 | SplasPood | Ciber311: heh, ok thats what I was thinking |
| 16:50.30 | Ciber311 | polycom is pissing me off though |
| 16:50.43 | Ciber311 | they need to release new phones with backlit screens already |
| 16:50.49 | *** part/#asterisk hotroot (n=michael@pD9E96DF6.dip.t-dialin.net) |
| 16:51.11 | Ciber311 | 60 dollar grandstreams are backlit for gods sakes |
| 16:52.15 | C6Vette | When a call comes in on our DIDs frmo our voip provider it comes in like:SIP/XXX.XXX.XXX.XXX-b78033d0 Is there a way I can change it to something like SIP/inbound-b78033d0? |
| 16:52.22 | SplasPood | Ryushin: Whats new in 2.0.1 ? |
| 16:52.37 | SplasPood | Ciber311: true |
| 16:52.37 | Ciber311 | atacomm's ftp seems to be screwed up |
| 16:53.02 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:53.46 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 16:54.28 | jmls | hey people |
| 16:54.43 | sconasq | C6Vette, not sure.. maybe if you use the dns name in sip.conf instead of the ip? |
| 16:54.50 | jmls | anyone using mixmonitor in anger after the latest round of trunk updates ? |
| 16:55.02 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:55.04 | jmls | there's been a lot fixed in it |
| 16:55.37 | Ryushin | Don't know. I'm just hoping it will fix my ip430's. |
| 16:55.38 | Ciber311 | who what when? |
| 16:56.17 | sconasq | is 1.4 good for production use? |
| 16:57.02 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:57.05 | [TK]D-Fender | Ryushin: Indeed, 2.0.1 and 3.2.2 are released |
| 16:57.34 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 16:57.44 | jmls | sconasq: we are using a *old* version of 1.4 (svn trunk) (r37613) in production since mid July. Only problems we've had has been with jabber and mixmonitor. Latest version of trunk should have fixed most (if not all) of the mixmonitor issues |
| 16:57.54 | [TK]D-Fender | Ryushin: Changelog is very "busy".... |
| 16:58.24 | jmls | currently testing r41990 :) |
| 16:58.39 | Ciber311 | can't even find a simple changelog on polycom's site |
| 16:59.25 | sconasq | cool jmls |
| 16:59.38 | Ryushin | [TK]-Fender, Yea, I expected it to be a major upgrade. But right now, I don't have much of a choice with my ip430's. They can't get any worse. |
| 16:59.55 | Ciber311 | i've stayed away from those heh |
| 17:00.27 | [TK]D-Fender | Ryushin: I expect to upgade mine tonight. |
| 17:01.41 | *** join/#asterisk alexis101 (n=as@MTRLPQ02-1177996380.sdsl.bell.ca) |
| 17:02.32 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 17:02.43 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.203) |
| 17:03.29 | alexis101 | Hi , I was wondering if there is a way to manually generate The QueueMemberStatus event in the manager? |
| 17:04.24 | adamowitz | anyone here using DECT IP phones with asterisk? do you like it? |
| 17:04.27 | adamowitz | from the little i've read on voip-info.org, it seems like it's more trouble than it's worth. |
| 17:04.30 | adamowitz | i'm having trouble finding any documentation for doing this. there seems to be little real content in the many links for DECT that i find on voip-info.org |
| 17:05.28 | alexis101 | I mean the event is automaticly generated when you send the command agentcallbacklogin or agentlogoff but i want to call it manually |
| 17:05.55 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 17:06.31 | Boggi | how do i specify what number to use for an outbound connection? |
| 17:07.03 | Ryushin | Does anyone know who updates http://www.freedomphones.net/polycom/files/ ? |
| 17:07.12 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 17:07.46 | jmls | alexis101: show manager command queuestatus |
| 17:08.59 | Ciber311 | Ryushin: forget about that place, they update really slow |
| 17:09.33 | Ciber311 | Ryushin: i usually get the updates from the atacomm ftp, but it's screwed up heh |
| 17:10.17 | *** part/#asterisk anthonyl (n=Default@c-71-57-41-193.hsd1.il.comcast.net) |
| 17:11.47 | Ciber311 | so i guess we get to wait until someone decides to "leak" the super top secret golden polycom software :P |
| 17:12.29 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:12.47 | *** join/#asterisk beuster (i=beu@freenode/developer/gentoo.developer.beu) |
| 17:14.03 | Ryushin | Yea, I'm not patient for this one though. My ip430's are acting like crap. |
| 17:14.14 | adamowitz | ping coppice |
| 17:14.40 | coppice | hi |
| 17:14.46 | adamowitz | Hi. |
| 17:14.56 | Boggi | in extensions.conf, how do i configure my sip phone to use a second number instead the primary one? |
| 17:14.56 | adamowitz | So, are you using the DECT IP phones with *? |
| 17:14.59 | *** join/#asterisk Stp1800 (n=Stp1800@atlsfl-bundle-69-167-93-164.atlsfl.adelphia.net) |
| 17:15.26 | coppice | no, but a number of people use them, and have far better results than WiFi phones |
| 17:15.41 | mut | anyone know when channelized t3/ds3 might come out? |
| 17:15.42 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 17:15.45 | Ciber311 | Ryushin: i guess that's what we get for purchasing products from such an uptight company :P |
| 17:16.33 | Ryushin | Ciber311: Yea, not kidding. |
| 17:16.36 | adamowitz | coppice: I asked about that in here a few minutes ago and got no replies. |
| 17:16.37 | [TK]D-Fender | Ciber311: No, thats what you get from not dealing with a very prompt reseller. |
| 17:16.40 | adamowitz | Although I've found several products on the web, I find no real documentation for using them with *. Do you know of any? |
| 17:17.04 | P4C0 | asterisk, as a sip client, behind a firewall, is there a way to make it use stun server? where can I get documentation about it? thanks... (sorry I got disconnected) |
| 17:19.38 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 17:19.53 | P4C0 | asterisk support stun ? |
| 17:20.14 | [TK]D-Fender | P4C0: No. |
| 17:20.18 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 17:20.34 | [TK]D-Fender | P4C0: * behind nat needs about 4 settings in SIP.conf and thats about it... |
| 17:20.42 | *** join/#asterisk trelane` (n=trelane@unaffiliated/trelane) |
| 17:20.43 | P4C0 | [TK]D-Fender, someone here just told me that it does :p |
| 17:21.02 | file | it doesn't support the stun you think it does |
| 17:21.10 | [TK]D-Fender | P4C0: Congratulations, no someone has just told you otherwise. |
| 17:21.13 | file | that is all. |
| 17:21.47 | P4C0 | [TK]D-Fender, I'm behind nat, acting as a client (to connect to my sip provider) it's working fine but when making calls, I can't hear the other end... but they heard me... incoming calls are fine |
| 17:21.48 | nortex | Hey file, do you know if Iaxtel is accepting new signups? |
| 17:21.49 | [TK]D-Fender | file: Yeah, print TFOT and slam an obnoxious troll upside the head with it and then you'll have "stun" :D |
| 17:21.59 | file | nortex: should be |
| 17:22.09 | file | actually let me rephrase |
| 17:22.14 | [TK]D-Fender | P4C0: Fix your sip.conf and put in the settings that are advertised all over the place that you need to set. |
| 17:22.23 | file | there should be no code or restriction not allowing people to signup if they find out how if it's not there |
| 17:22.35 | Ciber311 | [TK]D-Fender: maybe, but it's still way too much drama over a silly firmware update that can only be used on their products anyway |
| 17:22.54 | P4C0 | [TK]D-Fender, the problem seesm to be rtp... but just to be sure, exactly what settings are you talking about? |
| 17:22.56 | [TK]D-Fender | Ciber311: Its not drama, its anti-troll methodolgy. |
| 17:23.06 | Ciber311 | lol |
| 17:23.13 | nortex | file, Ok, wanted to be sure since I signed up but no emails yet :) |
| 17:23.28 | file | nortex: you should get one unless it's blocked |
| 17:23.32 | [TK]D-Fender | P4C0: "localnet, nat=yes,externip / externhost-externrefresh. |
| 17:24.06 | P4C0 | [TK]D-Fender, but this is in global sip.conf or inside the entry for my provider? |
| 17:24.09 | file | nortex: it was sent, I see it |
| 17:24.42 | Egonis | I want to allow users to press * in VoiceMail to return to the main incoming menu, how would I do this? |
| 17:25.29 | [TK]D-Fender | P4C0: Global |
| 17:25.42 | [TK]D-Fender | P4C0: Go read the sample sip.conf and pay attention to the parameters.... |
| 17:26.25 | P4C0 | thanks [TK]D-Fender |
| 17:26.40 | *** join/#asterisk Andretii (n=andretii@66.80.140.115) |
| 17:26.48 | [TK]D-Fender | Ciber311: Polycom doesn't want to deal with every 2-bit idiot on the plant using their products and have offloaded support to their reseller network. I have a good one so I get my updates nice and prompt (yes I have them all already, almost a dozen releases) |
| 17:27.25 | [TK]D-Fender | Egonis: Please read the "a" exten in the list of "standard extensions" |
| 17:28.03 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 17:28.17 | nortex | file thanks |
| 17:29.26 | mitcheloc | file, i'm trying to work here, thanks |
| 17:30.24 | Egonis | What must a user press to fast forward / rewind a message? |
| 17:30.54 | file | mitcheloc: you work? I never would have guessed |
| 17:31.11 | mitcheloc | file, pfft |
| 17:31.33 | mut | alright, question for the guys that know their big kid switches, i'm lookin at lucent, tekelec and santera class 4/5 switches, any recommendations? |
| 17:33.02 | eKo1 | 5ess? |
| 17:33.37 | [TK]D-Fender | Egonis: Read the big print... http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain |
| 17:35.05 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 17:35.42 | Egonis | [TK]D-Fender: lol, ty! |
| 17:36.04 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:36.51 | knobo | Is there a webpage that shows an overview of ISDN with asterisk technoloyes? |
| 17:37.01 | knobo | there is so many different consepts |
| 17:37.11 | *** join/#asterisk Luke-Jr_work (n=ubuntu@rrcs-74-62-64-204.west.biz.rr.com) |
| 17:37.21 | Luke-Jr_work | Do Debian packages exist for Sangoma cards? |
| 17:38.26 | *** join/#asterisk jpablo_ (n=jpablo@200.94.130.194) |
| 17:38.30 | benjk | knobo, which card/s do you have |
| 17:38.48 | knobo | benjk: None |
| 17:38.53 | jpablo_ | hey people, anyone knows if there's a way for someone in san antonio to access my sipphone number from his landline? |
| 17:39.08 | knobo | benjk: which card should I have? for example :) |
| 17:39.43 | benjk | then do yourself a favour, get HFC based cards (using the chips from Cologne Chip AG, Germany) and use BRIstuff |
| 17:39.51 | benjk | most straightforward |
| 17:40.13 | [TK]D-Fender | Luke-Jr_work: Just use the source..... |
| 17:40.54 | benjk | all the other BRI ISDN stuff is either outdated ISDN stacks or cumbersome to set up |
| 17:41.28 | benjk | for devl/test/lab/home/small-biz you can use the passive HFC PCI cards which are less than $50 |
| 17:41.45 | Luke-Jr_work | [TK]D-Fender: I'd rather not on a production box |
| 17:42.40 | benjk | D-Fender, ... if that doesn't work, use a bigger hammer ;) |
| 17:42.42 | [TK]D-Fender | luke-jr_ : If there are deb's it should be on their FTP. Their drivers are pretty damn stable anyways..... |
| 17:43.11 | [TK]D-Fender | benjk: There are few problems in this world that can't be solved with really REALLY BIG LASERS..... |
| 17:43.21 | benjk | heh |
| 17:43.27 | file | really REALLY big lasers |
| 17:44.01 | benjk | I thought all problems can be solved with a hammer |
| 17:44.51 | file | hammers are SO 19th century |
| 17:45.05 | benjk | I think they are a lot older than that |
| 17:45.42 | benjk | mind you, a tomahawk is a hammer too |
| 17:46.03 | adamowitz | could i get some recommendations (for or against) regarding wifi phones for use with *? |
| 17:46.08 | adamowitz | coppice thinks DECT IP phones might be better but I find precious little documentation for DECT and asterisk, so I'm still considering plain ole' wifi phones with *. Thoughts? |
| 17:46.09 | eKo1 | laser guided hammer.... |
| 17:46.11 | adamowitz | hi ben |
| 17:46.25 | benjk | adamowitz, coppice is right |
| 17:47.34 | benjk | eKo1, I was thinking about the original tomahawk though |
| 17:47.48 | adamowitz | benjk: could you point me to some documentation for * and DECT? I've searched voip-info.org and turned up many links but precious little content. |
| 17:48.06 | twisted[asteria] | all problems cannot be solved with a hammer. |
| 17:48.08 | benjk | adamowitz, if you can find a DECT IP phone, say, one that speaks SIP, then as far as * is concerned its just another SIP phone |
| 17:48.43 | twisted[asteria] | sometimes you need to use a nuke |
| 17:48.43 | benjk | twisted, you're kidding!!! |
| 17:48.43 | twisted[asteria] | am I? |
| 17:48.46 | benjk | if you can't use a hammer, use a sledge hammer, then |
| 17:48.49 | twisted[asteria] | guess I better be otherwise the feds might show up |
| 17:49.09 | file | twisted[asteria]: I'll hide you! |
| 17:49.28 | twisted[asteria] | file, haha |
| 17:49.37 | SwK[Work] | TERROR!!! |
| 17:49.44 | twisted[asteria] | no |
| 17:49.47 | twisted[asteria] | it's TERRA!! |
| 17:49.59 | twisted[asteria] | according to GW |
| 17:50.55 | [TK]D-Fender | twisted[asteria]: : I've got this ant problem that sounds like its right up your alley! |
| 17:50.55 | adamowitz | benjk: ok, that's good to know, but about the wireless with DECT? |
| 17:51.01 | adamowitz | It's apparently not 802.11x, right? What then? |
| 17:51.08 | adamowitz | I know plain ole cordless phones use the same frequencies as 802.11x. |
| 17:51.09 | adamowitz | Is there any interference problem between DECT and 802.11x (and is your opinion based on experience)? |
| 17:51.14 | benjk | DECT is not 802.11 |
| 17:51.37 | benjk | DECT is DECT, Digital European Cordless Telephone |
| 17:51.49 | adamowitz | Enhanced (sorry...) ;) |
| 17:51.52 | benjk | or maybe E is for Enhanced |
| 17:52.04 | adamowitz | was European tho |
| 17:52.09 | benjk | but it has absolutely nothing to do with 802.11 |
| 17:52.15 | benjk | not even the same frequency band |
| 17:52.24 | Qwell | benjk: it used to be European, now it's Enhanced, I believe |
| 17:52.29 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 17:52.35 | Qwell | and wikipedia confirms that |
| 17:52.42 | E-bola | why was it changed? |
| 17:52.43 | Qwell | DECT or Digital Enhanced (former European) Cordless Telecommunications is an ETSI standard for digital portable phones, commonly used for domestic or corporate use. DECT can also be used for wireless data transfers. |
| 17:52.45 | E-bola | americans felt offended? :) |
| 17:52.50 | benjk | Qwell, I think it may have been enhanced all the way |
| 17:52.57 | coppice | although they haven't enhanced it :-) |
| 17:53.03 | benjk | because the CT in DECT comes from its predecessor CT |
| 17:53.10 | coppice | like GSM was not global |
| 17:53.25 | benjk | remember the rabbit phones in the UK? |
| 17:53.29 | benjk | those were CT2 |
| 17:53.35 | benjk | after than came DECT |
| 17:54.25 | benjk | so rather than calling it CT2+ they called it DECT, and the E may as well have been "enhanced" |
| 17:54.48 | coppice | DECT was being developed at the same time as CT2 |
| 17:55.00 | MRH2 | anyone know if polycom sip 2.0.1 is floating around somewhere and if so could u point me in the direction ;) |
| 17:55.06 | benjk | yeah, well, those UK folks always have to do their own thing |
| 17:55.07 | Qwell | MRH2: ask your reseller |
| 17:57.20 | benjk | heh |
| 17:57.20 | coppice | the tail behind that is quite amusing |
| 17:57.20 | Qwell | tale? |
| 17:57.20 | Qwell | or like an echo tail? |
| 17:57.21 | benjk | a waggy tail |
| 17:57.21 | coppice | both, really |
| 17:57.21 | [TK]D-Fender | MRH2: extranet.polycom.com |
| 17:57.21 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 17:58.22 | benjk | a tell tale tail then |
| 17:58.41 | kratzers | I get 'Looking for +NPANXXXXXX in sip-in' when doing a sip debug... how can I deal with the +? |
| 17:58.47 | Qwell | coppice: please tell the tale of the tail |
| 17:59.21 | kratzers | where NPANXXXXXX is 10-digit did and sip-in is the context set for the sip friend |
| 17:59.41 | benjk | exten => _+...... |
| 18:00.17 | kratzers | Ah, wasn't sure if characters were legal in a pattern match; I'm dumb. |
| 18:00.29 | syzygyBSD | how many tries do I need to do with a fax system to assume it works? |
| 18:00.39 | benjk | 42 times |
| 18:00.43 | syzygyBSD | ok, thanks |
| 18:00.50 | Qwell | You don't have to try it to assume it works |
| 18:00.57 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:00.58 | Qwell | just say "eh...it probably works" |
| 18:01.00 | syzygyBSD | lol.. ok |
| 18:01.05 | benjk | no, its definitely 42 |
| 18:01.09 | Qwell | rephrase the question :) |
| 18:01.22 | benjk | doesn't matter what the question is, the answer is 42 |
| 18:01.34 | coppice | The PAT Centre is a big private R&D centre in the UK owned by one of the world's largest PR companies. They did a project for Ferranti the UK to develop a digital cordless phone. It turned out crazily expensive. To save their asses, the PR side of the business stepped in, and promoted use of this thing for wide area use to the government. That was CT2. |
| 18:01.39 | syzygyBSD | no, 42 is the answer to life, the universe, everything |
| 18:01.48 | [TK]D-Fender | jbot: What's benjk's IQ? |
| 18:01.53 | syzygyBSD | not the answer to every question |
| 18:01.58 | Qwell | ~iq benjk |
| 18:02.53 | Egonis | how do I get osp out of asterisk? chan_sip will not load! :O |
| 18:03.05 | syzygyBSD | how about this, why does the same fax have 5 different file sizes? |
| 18:03.21 | benjk | "There is a theory which states that if ever anyone discovers what the |
| 18:03.21 | benjk | universe is for and why it is here, it will instantly disappear and be |
| 18:03.21 | benjk | replaced by something even more bizarre and inexplicable.", Douglas Adams |
| 18:03.29 | Qwell | syzygyBSD: The same reason that encoding the same data into mp3 will be different |
| 18:03.42 | Qwell | well, I assume anyhow ;p |
| 18:04.36 | coppice | syzygyBSD: that rather depends on how the 5 files were created |
| 18:04.56 | Boggi | seriously, is this impossible: i got two sip phones, and two phone lines (with two different phone number), and im trying to make each sip phone have different number when im calling out. But right now each sip phone uses the same number out. |
| 18:05.05 | syzygyBSD | they were all created using rxfax, from the same tiff file being sent using txfax on the same machine |
| 18:05.08 | Egonis | I compiled asterisk with osp support, and it broke chan_sip.so, I recompiled w/out osp and it still won't run, any ideas? |
| 18:05.22 | syzygyBSD | i guess there are only 4 different sizes out of the 5 files |
| 18:06.16 | coppice | slightly different or a lot different? |
| 18:06.28 | *** join/#asterisk Qball (n=qball@ipd50a4125.speed.planet.nl) |
| 18:06.39 | adamowitz | Ok, so if a DECT phone is just a SIP phone to *, then what' s this I see here about DDI? http://www.voip-info.org/wiki/view/Gigaset+DECT+with+activation+of+Direct+Dial+In |
| 18:06.50 | adamowitz | Is that not necessarily a part of DECT? I get that impression. |
| 18:07.15 | adamowitz | The author of the link seems to have english as a second language tho, so I'm really not sure I understand what (s?)he's saying. |
| 18:08.14 | syzygyBSD | they range from 58K to 105K |
| 18:08.46 | knobo | benjk: thanx |
| 18:09.00 | syzygyBSD | ahh, but the 50K pages have transmission errors |
| 18:09.01 | benjk | adamowitz ... "I hear the talking of the DJ, can't understand just what does he say? I'm on a mexican radio ...." |
| 18:09.50 | coppice | ah, that sounds bad. small difference can be due to things like time stamps. you can compare the actual images with "tiffcmp -t <file a> <file b>" |
| 18:09.54 | syzygyBSD | ok, all but the 75K pages do... weird |
| 18:10.30 | adamowitz | benjk: not sure i follow... i'm sometimes slow with jokes... |
| 18:10.47 | aydiosmio | afternoon all |
| 18:10.47 | benjk | It's a classic song from a group called Wall of Voodoo |
| 18:10.56 | adamowitz | Ah. |
| 18:11.03 | adamowitz | I'll hafta check it out. |
| 18:11.14 | adamowitz | benjk: do you use DECT IP phones with *? If so, which one(s)? |
| 18:11.18 | *** join/#asterisk mtaht4 (n=m@adsl-75-10-213-145.dsl.pltn13.sbcglobal.net) |
| 18:11.28 | benjk | its got a very unusual sound and the text is rather funny |
| 18:11.31 | *** part/#asterisk mtaht4 (n=m@adsl-75-10-213-145.dsl.pltn13.sbcglobal.net) |
| 18:11.52 | benjk | like "I wish I was in Tijuana, eating barbequed iguana" and such |
| 18:12.11 | aydiosmio | heh |
| 18:12.22 | aydiosmio | The new zealand prompt package for * is great |
| 18:12.40 | aydiosmio | pretty in comprehensible |
| 18:12.51 | [TK]D-Fender | benjk: DJ at my poolhall plays that on request for a freak who frequents the place.... |
| 18:13.08 | adamowitz | do you know the name of the song, benjk? |
| 18:13.15 | benjk | adamowitz, I am in JP, no DECT here |
| 18:13.19 | aydiosmio | jbalcomb: you get anywhere with that mysql junk? |
| 18:13.22 | benjk | Mexican Radio |
| 18:13.27 | [TK]D-Fender | adamowitz: "Mexican Radio" |
| 18:13.34 | adamowitz | otay |
| 18:13.37 | adamowitz | thanks |
| 18:13.39 | Qwell | for the bonus points, name the artist who made that song |
| 18:13.45 | aydiosmio | did the FCC approve DECT in the US? |
| 18:14.02 | adamowitz | So, benjk, what's your recommendation of DECT based upon? Personal experience outside JP? Second-hand experience? |
| 18:14.04 | aydiosmio | I was pretty sure the 1.8ghz band was reserved for aircraft here |
| 18:14.13 | [TK]D-Fender | Qwell: Wall OF Voodoo |
| 18:14.18 | benjk | that's the group |
| 18:14.19 | adamowitz | Qwell: done already. |
| 18:14.24 | Qwell | [TK]D-Fender: You googled that :P |
| 18:14.41 | [TK]D-Fender | Qwell: ....And you have a point somewhere, right? ;) |
| 18:14.43 | Qwell | ;) |
| 18:15.05 | [TK]D-Fender | Qwell: I did know it previously because I asked him who was requesting that freakish shit.... |
| 18:15.32 | benjk | the artist is Stan Ridgeway |
| 18:15.37 | [TK]D-Fender | Qwell: Its an monotonous little piece. I hate anti-climatic music... |
| 18:16.08 | benjk | I like it for nostalgic reasons and also the electronic background doodle |
| 18:16.46 | Qwell | I quote enjoy that song... |
| 18:17.58 | *** join/#asterisk zekeonfire (n=zekeonfi@208.186.65.172) |
| 18:18.29 | benjk | http://en.wikipedia.org/wiki/Wall_of_Voodoo |
| 18:18.53 | Qball | bah, remiss was telling the truth |
| 18:21.35 | adamowitz | So, benjk, what's your recommendation of DECT based upon? Personal experience outside JP? Second-hand experience? |
| 18:21.42 | adamowitz | familiarity with the tech? |
| 18:21.44 | adamowitz | other? |
| 18:24.20 | adamowitz | benjk? |
| 18:24.34 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:25.30 | benjk | I have used DECT phones in the UK |
| 18:25.53 | benjk | and I have used a variety of WiFi phones in a variety of environments |
| 18:26.28 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 18:30.34 | *** join/#asterisk rnovotny22 (n=rnovonty@198.57.19.126) |
| 18:30.35 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 18:30.46 | *** join/#asterisk wglenncamp (n=wcampbel@208.203.43.127) |
| 18:30.58 | *** join/#asterisk pablus (n=nn@test.conama.cl) |
| 18:31.31 | wglenncamp | Why do my calls that go over an IP Trunk sound so High Pitched? Is there recommended codec that I should use? |
| 18:32.39 | wglenncamp | My IAX.conf says that I am using: allow=ulaw |
| 18:32.47 | wglenncamp | allow-gsm |
| 18:35.20 | *** join/#asterisk markus_nf (n=mark@63.250.108.202) |
| 18:36.08 | markus_nf | I'm looking for a little help configuring asterisk2billing |
| 18:36.55 | [TK]D-Fender | markus_nf: Just like all things passingly related to Trixbox, that is nearly forbidden around here. |
| 18:37.56 | Damin | lease digg: http://digg.com/linux_unix/Top_10_Things_You_Could_Be_Doing_Instead_of_Attending_Ohio_Linux_Fest |
| 18:38.00 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 18:38.19 | Damin | Please digg: http://digg.com/linux_unix/Top_10_Things_You_Could_Be_Doing_Instead_of_Attending_Ohio_Linux_Fest |
| 18:38.21 | Damin | Bahh.. |
| 18:38.22 | Damin | sorry.. |
| 18:38.26 | Damin | But do that anyway.. :) |
| 18:39.11 | nortex | I'm pretty sure of the answer, but is there any way to change the callerID number on an analog line? Special services from the telco or what not included? |
| 18:39.13 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:39.26 | bkw_ | dugg |
| 18:40.14 | *** part/#asterisk markus_nf (n=mark@63.250.108.202) |
| 18:40.35 | brodiem | I have a question about the T.38 passthrough patch. I realize that non-T.38 faxes coming in over a SIP trunk would be useless for a T38 patch. But, if a fax was coming in from PSTN to a Zaptel device, does the T.38 passthrough effectively pass the fax as T.38? |
| 18:40.36 | mog | Damin, !!! |
| 18:40.43 | mog | im sorry im not gonna get to go this year |
| 18:40.50 | mog | but we are sending some spiffy people out there |
| 18:40.50 | MikeJ | dugg |
| 18:41.06 | *** join/#asterisk Zikey (n=Cooler2@m41.net81-64-140.noos.fr) |
| 18:41.16 | vader-- | does anyone know if there is a way to build a gotoiftime where it's a file filled with dates? |
| 18:41.34 | vader-- | i want to fill a file with dates that we are not in the building |
| 18:41.35 | [TK]D-Fender | nortex: Normally, NO. There are probably services out there that you can dial into that will let you rig it and then bridge you're outgoing call.... |
| 18:41.44 | Zikey | Hi all, do you happen to know if you can raise the volume of native mp3 file playing music on hold ? |
| 18:42.42 | vader-- | defender should i fear anything if i upgrade asterisk to the latest version? |
| 18:42.53 | vader-- | im running 1.2.7.1 |
| 18:43.20 | Damin | mog: Cool.. have them DIGG that link too! ;) |
| 18:43.29 | mog | heh will do |
| 18:44.08 | nortex | [TK]D-Fender, Thanks, that is what I expected. Back to the drawing board to get around it :) |
| 18:44.09 | file | brodiem: no, that's not T.38 passthrough |
| 18:44.54 | brodiem | file, there currently is no way to originate T38 with * correct? |
| 18:45.18 | file | correct |
| 18:45.47 | brodiem | file, do you know if there are any SIP providers out there that can pass T38? |
| 18:46.11 | file | I'm not a VoIP provider directory :( |
| 18:46.44 | brodiem | well I'm just wondering if it's rare to be able to find |
| 18:46.50 | trelane` | file, then what data is stored in you? |
| 18:48.20 | file | trelane`: recipes for muffins |
| 18:48.31 | trelane` | I demand that you give me muffins! |
| 18:48.36 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 18:48.43 | file | never! |
| 18:49.29 | trelane` | you will pay for your insolence! today your insolence costs $1.32, paypal is happily accepted for insolence payments, thank you for your insolence. |
| 18:49.36 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 18:49.41 | Qball | hmm |
| 18:50.38 | *** part/#asterisk Zikey (n=Cooler2@m41.net81-64-140.noos.fr) |
| 18:52.14 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 18:54.06 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 18:54.40 | *** join/#asterisk anthonyl (n=Default@c-71-57-41-193.hsd1.il.comcast.net) |
| 18:56.12 | sivana | user = inbound, peer = outbound? |
| 18:57.11 | hmmhesays | yeah |
| 18:57.23 | sivana | ok |
| 18:57.29 | hmmhesays | user is someone that uses service. peer is someone's service you use |
| 18:57.50 | aydiosmio | lol friend |
| 18:58.11 | sivana | friend is both |
| 18:58.40 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:00.52 | Egonis | is there a simple command I can use which will 'busy' all zap channels? |
| 19:00.53 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 19:03.50 | *** join/#asterisk Avalone (n=Avalone_@83.239.191.16) |
| 19:07.35 | *** join/#asterisk s0lid (n=jlq@61.28.161.132) |
| 19:07.37 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 19:08.41 | saftsack | does * has support for ventrilo? |
| 19:09.01 | Juggie | no |
| 19:09.11 | eKo1 | wtf is ventrilo? |
| 19:09.23 | saftsack | a voice communication tool like teamspeak |
| 19:09.46 | aydiosmio | <PROTECTED> |
| 19:09.46 | aydiosmio | <PROTECTED> |
| 19:09.52 | aydiosmio | sweet. |
| 19:13.34 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:13.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.210) |
| 19:18.11 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 19:21.27 | *** join/#asterisk budairc (n=chatzill@200.215.57.173) |
| 19:27.53 | [TK]D-Fender | *crickets* |
| 19:28.13 | DarKnesS_WolF | [TK]D-Fender: missed ya mate ;-) |
| 19:28.32 | [TK]D-Fender | DarKnesS_WolF: But your aim is improving! |
| 19:29.07 | DarKnesS_WolF | aim ? |
| 19:29.09 | DarKnesS_WolF | what aim ? |
| 19:29.38 | [TK]D-Fender | DarKnesS_WolF: Joke.. if I have to explain it, it'll kill the fun... |
| 19:29.55 | DarKnesS_WolF | oh okay :-) |
| 19:31.29 | *** part/#asterisk Stp1800 (n=Stp1800@atlsfl-bundle-69-167-93-164.atlsfl.adelphia.net) |
| 19:32.26 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 19:34.36 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 19:34.38 | *** part/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 19:36.19 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 19:36.35 | Egonis | How would I go about 'busying' all zap channels to test a rollover configuration? |
| 19:37.06 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:38.03 | [TK]D-Fender | Egonis: Fill it up with calls from one channel to another. |
| 19:38.08 | smackus | has anyone ever built into their dial plan a way to log an agent out which has been logged in using addqueuemember, if they fail to answer? |
| 19:38.18 | Egonis | [TK]D-Fender: Is there any scripting method for this? |
| 19:38.28 | [TK]D-Fender | Egonis: TOYWY |
| 19:38.43 | Egonis | [TK]D-Fender: Guh? :) |
| 19:38.52 | [TK]D-Fender | ~toywy |
| 19:39.08 | [TK]D-Fender | "The One You Write Yourself" |
| 19:39.08 | *** join/#asterisk sarum4n (n=some@saruman.demon.nl) |
| 19:39.12 | Egonis | hah |
| 19:39.12 | Nivex | the bot, it fails. |
| 19:39.44 | [TK]D-Fender | jbot: toywy is The one you write yourself. |
| 19:39.46 | jbot | [TK]D-Fender: okay |
| 19:39.50 | [TK]D-Fender | ~toywy |
| 19:39.51 | jbot | well, toywy is The one you write yourself. |
| 19:39.58 | [TK]D-Fender | THERE |
| 19:41.45 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 19:41.50 | [TK]D-Fender | Egonis: Just grab a phone supporting a lot of calls and fill it up... |
| 19:41.50 | smackus | has no one done the logout thing for addqueuemember |
| 19:42.02 | [TK]D-Fender | smackus: What kind of device are you adding? |
| 19:42.19 | smackus | SIP |
| 19:42.35 | smackus | just doing addqueuemember SIP/extension |
| 19:42.37 | [TK]D-Fender | smackus: I do believe I gave you a solution a long time ago on this... |
| 19:42.55 | smackus | oh... thats right. |
| 19:42.59 | smackus | gotta go back to the logs. |
| 19:43.00 | smackus | thanks |
| 19:43.31 | P4C0 | how is the correct way to set a peer for outgoing calls that doesn't require registry? |
| 19:44.13 | [TK]D-Fender | P4C0: Peers don't require you to register.... |
| 19:44.16 | P4C0 | I mean I want to connecte my asterisk to my server provider, and it says they will filter my ip so no need to register |
| 19:44.17 | [TK]D-Fender | smackus: Use chan_local for your agents and add the removal in the dialplan it calls |
| 19:46.55 | P4C0 | [TK]D-Fender, problem is that this seems to be working really strange... I call my server, then I hang up my phone and I get an incoming call from asteriks... !? |
| 19:48.03 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:49.55 | hmmhesays | huh |
| 19:52.17 | P4C0 | I think I have a huge missconfiguration in sip.conf :( |
| 19:54.00 | P4C0 | for making and entity where I will send and recive call from (voip provider) and doesn't require to register I need to specify it as peer, user or friend? for my local voip phones that will connecte to asterisk? looking at the documentaion I supposed that my softphone should be user, and the voip provider a peer si that right? |
| 19:54.50 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 19:56.01 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:56.07 | [TK]D-Fender | P4C0: To RECEIVE calls that'd be "user", to PLACE calls that'd be "peer" and for both it'd be "friend" |
| 19:56.27 | [TK]D-Fender | P4C0: Your softphone should almost certainly be friend. |
| 19:57.02 | P4C0 | [TK]D-Fender, same as my voip provider... cause I want to use him to call others and to recive calls from him correct? |
| 19:57.28 | [TK]D-Fender | P4C0: Go ask him for a config sample. it depends on how they set themselves up. |
| 19:57.51 | [TK]D-Fender | P4C0: Some companies use 1 auth for incoming and a completely different one for outgoing. |
| 19:58.22 | P4C0 | [TK]D-Fender, humm I'll ask, but not sure if they will provide that information |
| 19:58.50 | *** join/#asterisk Stp1800 (n=Stp1800@atlsfl-bundle-69-167-93-164.atlsfl.adelphia.net) |
| 19:59.42 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:00.17 | aydiosmio | is it possible to get asterisk to trim silence from the end of a recording? |
| 20:00.41 | aydiosmio | or is that another utility? |
| 20:00.51 | Qwell | sox |
| 20:00.54 | [TK]D-Fender | aydiosmio: * is not an audio editing suite.... |
| 20:01.16 | aydiosmio | no kidding |
| 20:01.30 | aydiosmio | but I've seen references to * stopping a recording when it detects silence? |
| 20:01.58 | Qwell | aydiosmio: That's much different |
| 20:02.35 | aydiosmio | yeah I found it. trim is not used to describe it |
| 20:02.45 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 20:05.27 | *** join/#asterisk batphone (n=will@cpe-24-162-13-48.houston.res.rr.com) |
| 20:05.46 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.227) |
| 20:06.01 | batphone | hey fellas, im having a huge problem with a system losing voice all the sudden; polycoms connected to asterisk 1.2.10 |
| 20:06.28 | batphone | tcpdump shows a shitstorm of icmp port unreachable messages coming from the pbx on the RTP port |
| 20:06.38 | batphone | no firewall |
| 20:06.58 | batphone | any clues? |
| 20:07.03 | batphone | canreinvite=no |
| 20:07.06 | batphone | nat=no |
| 20:09.07 | batphone | its like asterisk forgets about the RTP handshake |
| 20:09.23 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:09.38 | P4C0 | I have a local area network, with one asterisk and several softphones registered to it, one connection to a voip provider, from outside if I call to one extension, and after a while I hang up, the extension never realize it and I receive a call again from the extension |
| 20:09.52 | *** join/#asterisk Mw3_ (i=mw3@national.t-error.hu) |
| 20:10.01 | justinu|laptop | batphone: happening in the middle of calls? |
| 20:10.24 | batphone | yes |
| 20:10.28 | batphone | killing me man |
| 20:10.38 | batphone | these people are gonna shoot me |
| 20:11.03 | justinu|laptop | icmp port unreachable means that the socket isn't open on the pbx end... if it just closes in the middle of a call, that's fubar'd |
| 20:11.14 | batphone | yep |
| 20:11.20 | batphone | what is fubar'ed |
| 20:11.25 | batphone | the pbx? |
| 20:11.26 | justinu|laptop | try downgrading? |
| 20:11.28 | batphone | asterisk? |
| 20:11.30 | batphone | ah |
| 20:11.31 | batphone | ok |
| 20:11.34 | justinu|laptop | yeah, sounds like some kind of rtp stack failure |
| 20:11.38 | batphone | bingo |
| 20:11.49 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:11.52 | batphone | ill put 1.0.8 on this mofo |
| 20:11.56 | batphone | itll make me feel better ;) |
| 20:12.04 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:12.20 | batphone | my employer doesnt pay me enough to use gdb ;) |
| 20:12.24 | justinu|laptop | hah |
| 20:12.26 | justinu|laptop | that's a big downgrade |
| 20:12.31 | justinu|laptop | maybe one of the earlier 1.2 releases |
| 20:13.15 | batphone | justinu|laptop: i have quite a few 1.2.10's out there w/o this problem |
| 20:13.16 | batphone | however |
| 20:13.23 | batphone | this particular box has rtc issues |
| 20:13.27 | hmmhesays | so should I go with a radeon mobility x1400 or a geforce go 7300 |
| 20:13.47 | justinu|laptop | rtc issues? you running any zaptel modules? |
| 20:13.55 | batphone | yes |
| 20:14.27 | batphone | i could rebuild them w/o rtc support |
| 20:14.33 | batphone | give it a shot before downgrading |
| 20:14.34 | justinu|laptop | worth a shot |
| 20:14.35 | justinu|laptop | yeah |
| 20:14.47 | justinu|laptop | SMP vs uniprocessor kernel? |
| 20:15.00 | batphone | this is SMP |
| 20:15.07 | batphone | im damn proud of my stage 1 build |
| 20:15.12 | batphone | NPTL |
| 20:15.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:15.31 | batphone | this image is the gw/fw for a thousand hosts |
| 20:15.33 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:15.33 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:15.55 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:15.58 | batphone | some of these beasts filter so much traffic they keep a load of around 1.5 or 2 |
| 20:16.22 | justinu|laptop | yeah, i'm just throwing things out that have troubled me in the past |
| 20:18.31 | *** join/#asterisk euphobot (n=don@adsl-64-170-69-114.dsl.lsan03.pacbell.net) |
| 20:19.42 | batphone | im gonna roll back AND remove rtc |
| 20:19.58 | batphone | production box, its gonna be bad when i restart asterisk anyway. might as well only restart it once' |
| 20:21.49 | P4C0 | anyone have a working asterisk behind a nat? |
| 20:22.18 | [TK]D-Fender | P4C0: Plenty of us, myself included |
| 20:22.55 | [TK]D-Fender | P4C0: Did you use those settings I told you to look at? Have you checked out the WIKI on them? How about the sample sip.conf file? |
| 20:23.53 | P4C0 | [TK]D-Fender, yes, I'm on it, but I'm having a problem right now with my provider... they just call me and I couldn't hear anything |
| 20:24.15 | hmmhesays | nat? |
| 20:24.15 | hmmhesays | fun |
| 20:24.46 | file | [TK]D-Fender: I don't want to know your name |
| 20:24.55 | P4C0 | [TK]D-Fender, now he called again it it works... problem is that some times it works some times no :( |
| 20:24.55 | [TK]D-Fender | P4C0: make sure you also have "canreinvite=no" global as well.... when you think eveything is ready pastebin your sip.conf |
| 20:25.12 | [TK]D-Fender | file : I just want.... |
| 20:25.14 | P4C0 | canreinvite=no as global? |
| 20:25.16 | [TK]D-Fender | file : ! ! ! |
| 20:25.20 | hmmhesays | bah playing melodic groups in scales with only alternate picking sucks |
| 20:25.21 | [TK]D-Fender | P4C0: Yes |
| 20:25.27 | P4C0 | [TK]D-Fender, ok, I'll |
| 20:25.33 | [TK]D-Fender | hmmhesays: like? |
| 20:26.30 | hmmhesays | i'm doing some metronome excercises, normally when I do string skipping I don't use true alternate picking |
| 20:26.34 | *** join/#asterisk jtoy (n=jtoy@cust-206-40-173-219.bos-static.gis.net) |
| 20:26.49 | jtoy | is there a site listing all opensource asterisk tools/add ons? |
| 20:26.55 | hmmhesays | something like A-B-C-D; B-C-D-E; C-D-E-F |
| 20:27.00 | bkw_ | www.pbxfreeware.org has sone |
| 20:27.47 | [TK]D-Fender | hmmhesays: Economy picking time! |
| 20:27.58 | hmmhesays | i'm trying to get around that though |
| 20:28.13 | hmmhesays | i got my metronome set at about 80bpm |
| 20:29.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 20:29.44 | jtoy | are there any asterisk add ons that you can control a phone call coming into your voip phone, but form the desktop? |
| 20:30.09 | [TK]D-Fender | hmmhesays: Here : http://www.geocities.com/tk_dfender/Untitled.mp3 |
| 20:30.13 | jtoy | so if I had a call coming in, a popup would appear on my desktop asking me to forward the call/send to voicemail/hangup/etc |
| 20:30.20 | batphone | every now and then during a call the polycom sends a packet to the PBX that is not destined for the same port as the RTP stream |
| 20:30.23 | batphone | what is this packet? |
| 20:30.25 | batphone | its 60 bytes |
| 20:30.27 | *** join/#asterisk oej (n=oej@x1-6-00-02-72-55-4c-5f.k693.webspeed.dk) |
| 20:30.28 | batphone | it LOOKS like voice |
| 20:30.30 | jtoy | so it is a supplement to the phone, but not an actual software phone client? |
| 20:30.32 | [TK]D-Fender | hmmhesays: 80bpm? Molassas! |
| 20:30.53 | batphone | every 10 seconds |
| 20:31.02 | batphone | and the pbx sends back ICMP unreachable |
| 20:31.05 | hmmhesays | [TK]D-Fender i've economy picked my entire life |
| 20:31.30 | hmmhesays | try my my technical playing a little beter |
| 20:31.32 | hmmhesays | *better |
| 20:31.35 | hmmhesays | nice mp3 |
| 20:31.36 | justinu|laptop | <PROTECTED> |
| 20:32.02 | [TK]D-Fender | hmmhesays: at 1:30 I interleave pick up 2+ octaves. |
| 20:32.07 | batphone | and asterisk is not liking this RTCP packet |
| 20:32.12 | batphone | for what reason |
| 20:32.36 | justinu|laptop | asterisk doesnt' speak RTCP |
| 20:32.49 | P4C0 | [TK]D-Fender, this is my sip.conf http://rafb.net/paste/results/MoQe5E71.html |
| 20:32.51 | jtoy | anyone got ideas? |
| 20:32.51 | batphone | hmmm |
| 20:33.03 | batphone | i wonder if this new polycom firmware is eating up the calls |
| 20:33.11 | batphone | by making heavy use of RTCP |
| 20:33.19 | batphone | thats gonna piss some customers off heh |
| 20:33.38 | justinu|laptop | batphone: ethereal should decode that RTCP packet for you |
| 20:33.45 | [TK]D-Fender | P4C0: sip.conf looks ok, have you forwarded all appropriate ports to your * box? |
| 20:33.47 | batphone | tcpdump is doing it |
| 20:33.49 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 20:33.52 | batphone | im seeing some stuff in it |
| 20:34.03 | [TK]D-Fender | batphone: What version? |
| 20:34.12 | justinu|laptop | it should have qos statistics, like jitter, and lost packets, etc. |
| 20:34.14 | batphone | ok tcpdump is not REALLY doing it |
| 20:34.17 | batphone | gotcha |
| 20:34.23 | batphone | [TK]D-Fender: 1.2.10 |
| 20:34.47 | [TK]D-Fender | batphone: Was askinf about your Polycom firmware |
| 20:35.11 | batphone | 1.6.7 |
| 20:35.23 | batphone | i think 1.6.8 is out.. |
| 20:35.24 | P4C0 | [TK]D-Fender, yup, in the firewall I have, forward udp port from 500 to 31000 comimng from ip of my server provider to my asterisk local ip |
| 20:35.37 | batphone | <PROTECTED> |
| 20:35.39 | P4C0 | from 5000 (sorry) |
| 20:35.42 | batphone | i see that all day long on some systems |
| 20:36.16 | [TK]D-Fender | batphone: 2.0.1 is out |
| 20:36.21 | batphone | oooh |
| 20:36.29 | batphone | used it? |
| 20:36.44 | justinu|laptop | batphone: i see that on transfers w/ polycom phones |
| 20:36.50 | justinu|laptop | seems fairly harmless |
| 20:37.09 | batphone | justinu|laptop: yeah i havent noticed it creating any real problems |
| 20:37.32 | P4C0 | [TK]D-Fender, that srvlookup=yes is ok? |
| 20:37.44 | [TK]D-Fender | P4C0: Never used.. try removing it |
| 20:38.48 | P4C0 | ok |
| 20:39.43 | P4C0 | [TK]D-Fender, all the rest is as yours? |
| 20:40.51 | [TK]D-Fender | P4C0: Pretty much |
| 20:41.00 | *** join/#asterisk s0lid (n=jlq@61.28.161.132) |
| 20:41.02 | P4C0 | nat=yes should set the same rtp port for sending and receiving data right? so if they heard me, I must heard them as well? |
| 20:41.24 | batphone | this would have to be due to the polycom firmware |
| 20:41.33 | batphone | but i havent noticed this kind of RTCP traffic on any other box |
| 20:41.46 | batphone | im looking at some right now and i dont see any RTCP packets coming from any polycoms |
| 20:42.23 | justinu|laptop | it's for sure RTCP? |
| 20:43.14 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 20:43.23 | rene- | hey |
| 20:43.52 | rene- | i was told that patlooptest only works with cards configured to T1 mode.. is that statement correct? |
| 20:44.42 | Ryushin | So if ztcfg shows a channel as "Channel 01: FXO Kewlstart (Default) (Slaves: 01)", does that mean I'm signalling FXO to a analog phone? |
| 20:45.37 | Ryushin | I'm trying to talk to a analog phone connected to my card. |
| 20:45.43 | *** join/#asterisk delta34ooo (n=delta34o@global-sf.keen.com) |
| 20:46.39 | P4C0 | now my server provider said that my asterisk is not responding :( |
| 20:46.50 | P4C0 | I can receive calls from him but can't place calls :( |
| 20:47.56 | batphone | justinu|laptop: no not yet |
| 20:48.01 | batphone | its a 60 byte packet |
| 20:48.16 | batphone | encoded it has the ip of the phone in it in plain text |
| 20:48.36 | justinu|laptop | batphone: try tethereal |
| 20:51.20 | *** join/#asterisk ComputerWarm (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 20:51.34 | ComputerWarm | hello |
| 20:51.35 | P4C0 | this is really strange... if I take away the localnet parameter from my sip.conf I can make calls througt my server provider... if no server provider sais my asterisk timedout... wtf |
| 20:51.40 | ComputerWarm | is the creator of a2b around? |
| 20:53.51 | ComputerWarm | anyone here using A2B in production? |
| 20:53.54 | delta34ooo | anybody have experience using the Cisco phone to display a name instead of a number from the caller side, for instance if I dial 0, the name Operator will show up on my cisco 7960 phone display |
| 20:54.19 | P4C0 | [TK]D-Fender, are u there? |
| 20:54.23 | *** join/#asterisk Op3r (n=op3r@125.212.36.242) |
| 20:54.27 | *** join/#asterisk prog (n=vdsoft@vdsoft.kh-net.cz) |
| 20:54.31 | prog | hello to all |
| 20:54.50 | [TK]D-Fender | P4C0: What kind of router are you using? |
| 20:55.07 | [TK]D-Fender | ComputerWarm: Don't expect much GUI help around here.... |
| 20:55.22 | P4C0 | [TK]D-Fender, a netgear one |
| 20:55.31 | *** join/#asterisk Gunter12 (n=aa@pool-71-104-125-65.lsanca.dsl-w.verizon.net) |
| 20:55.44 | [TK]D-Fender | P4C0: Could be a router issue... Cisco PIX for instance makes NAT a living hell... |
| 20:56.44 | prog | i have a question related with ASTDB. I put "exten => _*22*XXX,1,Set(DB(CFIM/${CALLERIDNUM})=${EXTEN:4})" into extensions.conf and when i analyse (via sip debug ) call forwarding, SIP says "Declined" ... could anyone of you give an advice ? thank you |
| 20:57.01 | P4C0 | [TK]D-Fender, humm maybe, but I'm not sure... if I remove localnet I can place calls to my provider... with localnet values server provider said that I timeout... so he gets the requests... |
| 20:57.32 | [TK]D-Fender | prog: AstDB has NOTHING to do with anything that you don't set in extensions.conf yourself. |
| 20:58.13 | [TK]D-Fender | P4C0: that makes no sense.... |
| 20:58.51 | P4C0 | [TK]D-Fender, I know... it's really strange but I tried 2 times now... same behaviewr.. |
| 21:01.12 | batphone | justinu|laptop: these 60 byte packets dont really look like RTCP packets |
| 21:01.17 | batphone | at least not to ethereal |
| 21:01.24 | batphone | they look like garbage |
| 21:01.30 | batphone | and are just labeled "UDP" |
| 21:03.18 | Ryushin | Shouldn't by Sangoma 200 FXS card give me a dial tone if I plug in a phone? |
| 21:03.40 | batphone | crap |
| 21:03.41 | batphone | justinu|laptop: |
| 21:03.49 | batphone | they are RTCP END packets |
| 21:03.51 | batphone | jesus... |
| 21:04.02 | batphone | so the phone is like "ok i dont wanna talk no more so im gonna end the call" |
| 21:04.15 | batphone | but asterisk thinks the call is still happening |
| 21:04.25 | batphone | ooh i love this shit |
| 21:05.50 | justinu|laptop | so there's no SIP BYE associated with that RTCP END? |
| 21:06.08 | *** join/#asterisk rrivas (n=rrivas@200.68.91.21) |
| 21:06.33 | *** part/#asterisk rrivas (n=rrivas@200.68.91.21) |
| 21:09.29 | batphone | justinu|laptop: none |
| 21:09.51 | batphone | the nearest sip packet in either direction is a re-registration from another phone |
| 21:10.30 | *** join/#asterisk brijn (n=bas@204.244.176.116.net-conex.com) |
| 21:10.35 | brijn | Hello all |
| 21:11.12 | *** part/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 21:11.19 | brijn | Does anybody know if the the latency as reported in "sip show peers" is already compensated with the latency you would see for a ping? |
| 21:13.09 | justinu|laptop | batphone: at this stage, I would consider reflashing the phones with a known good image |
| 21:13.18 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 21:13.19 | hmmhesays | qualify does use ping doesn't it? |
| 21:13.29 | justinu|laptop | nope |
| 21:13.34 | justinu|laptop | qualify sends a SIP OPTIONS packet |
| 21:13.47 | Gunter12 | I am having a problem with incoming calls, I have two phones numbers and two different contexts for each number, but when I call either number, it goes to the same context |
| 21:14.37 | brijn | justinu|laptop: So i would expected that ping RTT / 2 is always smaller then SIP "latency"? |
| 21:14.50 | Gunter12 | So Phone # A, should goto context A, which it does, but phone # B should goto Context B, but instead it goes to A |
| 21:15.12 | batphone | justinu|laptop: those 60 byte garbage packets highly resemble the 88 byte RTCP packets that ethereal DOES recognize |
| 21:15.19 | batphone | justinu|laptop: good idea |
| 21:16.10 | justinu|laptop | brijn: in my experience, that is the case... some IP phones (polycom 501) are notoriously slow to respond to an options msg |
| 21:16.49 | [TK]D-Fender | justinu|laptop: rECENT FIRMWARE RELEASES HAVE SUPPOSEDLY IMPROVED THAT A LOT. |
| 21:16.54 | brijn | justinu|laptop: I'm writing a small script to plot SIP latency and want to plot a line for network latency as well. |
| 21:17.18 | justinu|laptop | [TK]D-Fender: i am without doubt very far behind |
| 21:17.34 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 21:17.51 | [TK]D-Fender | justinu|laptop: 2.0.1 is out. I should be upgrading tonight. |
| 21:18.19 | brijn | justinu|laptop: It's between my * box and my provider.. I have a 501 and never,ever understood why the SIP latency was ~100ms while connected to the same switch.. Thate xplains! |
| 21:18.42 | Gunter12 | Anyone? |
| 21:19.12 | justinu|laptop | heh yeah... sometimes i've seen like 1.5 second response times when ping is <= 30ms |
| 21:19.16 | brijn | [TK]D-Fender: Are the fw's available for download? |
| 21:19.33 | brijn | justinu|laptop: Never seen that bad, but up to 300ms yes |
| 21:19.47 | brijn | Good to know that it's not an issue with my network |
| 21:20.14 | [TK]D-Fender | brijn: Yes |
| 21:20.44 | brijn | Any issues with newer firmware that I should be aware of (not that I fix the latency, but break 99 other things ;-) |
| 21:21.32 | [TK]D-Fender | brijn: Not that I know of. I was working pretty well of 2.0.0 beta. |
| 21:21.57 | brijn | [TK]D-Fender: Tx! Will download it tonight and give it a try |
| 21:22.22 | P4C0 | [TK]D-Fender, with ethereal I think I had found the problem... asterisk is not sending the re invite to the provider with the password... provider keeps asking for auth values... |
| 21:22.56 | [TK]D-Fender | P4C0: Well your GENERAL section it right... can't say the same for your provider setup. |
| 21:23.50 | P4C0 | [TK]D-Fender, i woudln't be amazed.. but if asterisk received a 407 auth required it should sent the invite again with the auth right? but it isn't :( |
| 21:24.08 | [TK]D-Fender | P4C0: Maybe your setup is just wrong... I don't know.... |
| 21:24.52 | P4C0 | [TK]D-Fender, maybe, but not sure where to change that... let me see what's packages I get when chaning the localnet |
| 21:28.42 | P4C0 | humm this is really strange... if I have the localnet value when provider ask for a invite with auth values it just keep sending the same invite without those values... when I remove the localnet asterisk do sent again the invite with the digest... why!? isn't localnet just to avid appying nat to peers inside that network? |
| 21:29.02 | wunderkin | anyone know of someplace cheap that sells polycoms and does not charge up the butt for shipping that i can still get an order shipped out today? my place fell through! ugh |
| 21:31.54 | eKo1 | wunderkin: might as well ask what the winning lotto numbers are |
| 21:32.28 | wunderkin | heh, i was ready to order at 8am pacific but i was stuck waiting all day for this place that said they could sell it for me and now they tell me they arent authorized to sell in my state |
| 21:32.40 | [TK]D-Fender | wunderkin: What place? |
| 21:32.48 | wunderkin | redorbit |
| 21:33.00 | [TK]D-Fender | wunderkin: Try www.telephonydepot.com |
| 21:34.56 | wunderkin | they are probably closed now |
| 21:38.04 | justinu|laptop | try voipconnection.com |
| 21:38.18 | justinu|laptop | i called them for some emergency crap at 6pm EST, and they shipped that day |
| 21:40.05 | marv[work] | Why does asterisk compile with -O6 by default, when anything over -O3 is treated as -O3? |
| 21:40.27 | aydiosmio | double your pleasure |
| 21:40.38 | wunderkin | cool, ill keep them in mind |
| 21:40.44 | Nugget | it keeps the gentoo users quiet. |
| 21:40.54 | aydiosmio | voipconnection is good people |
| 21:41.04 | justinu|laptop | aydiosmio: agreed |
| 21:41.15 | justinu|laptop | they don't have rock bottom prices, but they will go the extra mile for you |
| 21:41.22 | justinu|laptop | and they will work with you on price |
| 21:42.49 | *** part/#asterisk BrianHV (n=bhv1@copland.brianhv.org) |
| 21:43.29 | justinu|laptop | the real question is why does asterisk compile with debuging info turned on, but then with -O6, which prevents you from getting any backtraces |
| 21:43.53 | aydiosmio | for her pleasure |
| 21:45.49 | marv[work] | i always prefered -OTEXAS and -fomit-everything |
| 21:47.06 | aydiosmio | ah heck, just install the directories! |
| 21:53.09 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 21:56.18 | x86 | anyone ever used iaxmodem and had it constantly re-register non-stop? |
| 21:56.41 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 22:00.19 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:04.16 | P4C0 | any asterisk developers here :p |
| 22:04.34 | tessier__ | hmm....one of either transfer or blind transfer is implemented in asterisk and the other in the phone. I can never remember which. |
| 22:04.53 | file | P4C0: not one! |
| 22:05.04 | docelmo | P4C0 NOT two! |
| 22:05.10 | P4C0 | :) |
| 22:05.13 | Qwell | not four! |
| 22:05.19 | Qwell | wait, maybe 4 |
| 22:05.35 | docelmo | haha |
| 22:05.56 | Qwell | depends on your definition of asterisk developer, I guess |
| 22:06.04 | [TK]D-Fender | Qwell: Careful there... you're running out of fingers & toes ;) |
| 22:06.05 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-214.cybersurf.com) |
| 22:06.15 | P4C0 | I'm having problem will localnet parameter :( |
| 22:06.20 | Qwell | [TK]D-Fender: I don't have toes - way to troll. |
| 22:06.23 | P4C0 | and sip invite digest |
| 22:06.40 | Qwell | </troll target=[tk]d-fender> |
| 22:06.40 | [TK]D-Fender | ;) |
| 22:06.48 | Qwell | yes, I'm joking :P |
| 22:06.50 | P4C0 | insecure=yes means authteticate based on ip right? |
| 22:07.01 | docelmo | yes |
| 22:07.04 | [TK]D-Fender | ok, off for a few hours, back later. Later all. |
| 22:07.13 | docelmo | as long as host=xxx.xxx.xxx.xxx |
| 22:07.25 | docelmo | I prefer insecure=very personally |
| 22:07.45 | *** join/#asterisk Amilcar_ (n=xxxxx@201.34.202.17) |
| 22:08.23 | P4C0 | strange... I had insecure=yes and host=ipserverprovider and incoming call get's rejected cause callerid@ipserverprovider was not authorized :( |
| 22:09.14 | P4C0 | maybe I didn't get the concept... anyways, asterisk doesn't send sip digest when localnet is set :( |
| 22:09.23 | *** part/#asterisk knight_ (n=knight@209.85.11.98) |
| 22:13.30 | c4t3l | is there a command line tool that can be used as a SIP protocol communicator? |
| 22:13.42 | *** join/#asterisk ctrix (n=CtRiX@88-149-166-154.f5.ngi.it) |
| 22:13.55 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:13.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:14.12 | syzygyBSD | c4t3l: telnet? |
| 22:14.28 | syzygyBSD | wait.. telnet can't do UDP... |
| 22:15.10 | syzygyBSD | how about asterisk.. I use that on the command line |
| 22:15.24 | P4C0 | what does realm really means... in sip auth attribute? |
| 22:16.00 | Ciber311 | it means the world your character is in |
| 22:16.06 | Ciber311 | don't you play WoW? :P |
| 22:16.12 | [Outcast] | had to step away for a moment |
| 22:16.39 | Ciber311 | am i crazy or did the new firmware make my 501's speakerphone louder... |
| 22:16.50 | *** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net) |
| 22:18.05 | P4C0 | :) |
| 22:19.12 | *** join/#asterisk nesys (n=nesys@88-149-169-192.f5.ngi.it) |
| 22:19.47 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 22:19.52 | P4C0 | god why is asterisk ignoring auth, secret and username values for my service provider when localnet is set!?!?!?? pleaseee anyonee |
| 22:21.08 | nesys | hi folks ... how could I activate the voicemail after x rings, or x seconds, when unavailable? I've forgotten the option |
| 22:23.51 | C6Vette | DIAL(SIP/somephone,10) will ring the extension for 10 seconds then goto next line in dial plan |
| 22:23.56 | kratzers | specify a timeout as an argument to the dial application |
| 22:24.15 | nesys | thanks C6Vette |
| 22:25.47 | hmmhesays | i need good online blackjack site |
| 22:26.16 | P4C0 | guys noone is having my problems!?? :( I can't be the only one having this issue :'( |
| 22:26.25 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 22:26.40 | *** join/#asterisk linagee (n=na@cpe-66-75-142-207.san.res.rr.com) |
| 22:27.52 | RoyK | P4C0: pastebin your config and tell what's not working in detail, please |
| 22:27.55 | RoyK | ~pastebin |
| 22:28.01 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 22:28.12 | P4C0 | RoyK, moment |
| 22:28.29 | P4C0 | RoyK, http://rafb.net/paste/results/MoQe5E71.html there u go |
| 22:28.53 | RoyK | and the problem? |
| 22:28.56 | nesys | C6Vette just a question: there's an option like that for "Answer" and not only Dial? |
| 22:29.46 | nesys | C6Vette for something like that: http://pastebin.ca/161904 |
| 22:30.13 | nesys | the internal number, and mailbox, is 5001 |
| 22:30.29 | P4C0 | RoyK, my setup: private network (192.168.6.0) asterisk on 192.168.6.10, sipphones in .5 and .2 voipserver, voipserver doesn't support asterisk to register to it, it works fine when I comment localnet, if I put localnet I can't make calls... when I check the sip packages with etherreal, when localnet is set, astersik doesn't resend an invite with digest auth when voip server sends auth required |
| 22:31.19 | C6Vette | nesys: http://pastebin.ca/161905 is this what your looking for? |
| 22:32.01 | RoyK | P4C0: dunno then. sorry |
| 22:32.07 | nesys | C6Vette thank you very much :) |
| 22:32.13 | C6Vette | That will ring the phone for 10 seconds and goto voicemail if no answer |
| 22:32.28 | P4C0 | RoyK, this is really strange... I can't be only one... |
| 22:33.22 | C6Vette | nesys, http://pastebin.ca/161907 also includes the busy message |
| 22:34.15 | benjk | P4C0, which version is this? |
| 22:34.35 | P4C0 | benjk, moment |
| 22:34.46 | P4C0 | benjk, 1.2.6 |
| 22:35.20 | *** join/#asterisk jmacz (n=jmacz@201.244.168.55) |
| 22:36.57 | *** join/#asterisk kusznir (n=kusznir@bakken9.eecs.wsu.edu) |
| 22:37.01 | nesys | C6Vette mmm ... it doesn't work, it seems |
| 22:37.18 | nesys | check the debug |
| 22:37.47 | kusznir | Hi all: I'm having trouble placing IAX calls to some number. I've been doing an iax debug trace, and my session looks good until it gets a HANGUP packet with "CAUSE CODE : 16". I've been googling, but can't seem to figure out what cause code 16 means. |
| 22:38.19 | kusznir | could anyone here shed some light on it? |
| 22:38.33 | C6Vette | nesys, what happends. |
| 22:38.39 | nesys | C6Vette .... nothing ... have you got ideas? |
| 22:38.52 | nesys | C6Vette what could I do? |
| 22:39.00 | C6Vette | you dont need the _ in front of the 5001 |
| 22:39.20 | benjk | kusznir, cause codes are listed in causes.h in include/asterisk |
| 22:39.41 | kusznir | thanks, I'll go look there now. |
| 22:40.51 | benjk | btw, those cause codes follow Q.931, so if you have a copy of the Q.931 docs you can get more detailed information there |
| 22:42.53 | *** join/#asterisk budairc (n=chatzill@200.215.57.174) |
| 22:42.56 | P4C0 | humm I'm having a problem... rtp... service provider is sending rtp to port 8540 and in sdp they negotiate to 24968... !? |
| 22:43.23 | budairc | hi |
| 22:50.07 | P4C0 | is this possible?? that default asterisk just ignore the sdp media port from package? |
| 22:53.25 | batphone | P4C0: that could be due to some NAT wierdness |
| 22:53.57 | batphone | SIP + NAT = Alcoholism |
| 22:54.06 | RoyK | X-Rob_: ping |
| 22:54.17 | X-Rob_ | RoyK, pong |
| 22:54.28 | P4C0 | batphone, :'( but not from my service provider!? hi have nat active to me? |
| 22:54.36 | RoyK | X-Rob_: where're you from? .au where? |
| 22:54.43 | X-Rob_ | Queensland |
| 22:54.47 | RoyK | k |
| 22:54.56 | X-Rob_ | http://aussievoip.com/wiki/RobThomas |
| 22:55.00 | X-Rob_ | ^^ more info there |
| 22:55.19 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:55.29 | RoyK | X-Rob_: I was just speaking to an old friend from newcastle about .au telecom. it's still a state monopoly?? |
| 22:55.38 | X-Rob_ | pretty much so |
| 22:55.47 | bkw_ | RoyK, you raising hell again? |
| 22:55.58 | X-Rob_ | heya bkw_ |
| 22:56.01 | RoyK | bkw_: not really :) |
| 22:57.25 | X-Rob_ | any fallout over the g729 codec leak? |
| 22:57.53 | RoyK | hehe |
| 22:57.58 | P4C0 | if I have asterisk inside a private network, and my service provider uses asterisk as well, should he set a nat flag on me? |
| 22:58.16 | RoyK | P4C0: always set it. it won't hurt |
| 22:58.18 | bkw_ | X-Rob_, I heard about that today. |
| 22:58.49 | RoyK | bkw_: i was just answering an email. i never started the thread.... |
| 22:59.23 | bkw_ | RoyK, but you did a naughty thing I see... because I didn't believe it when I heard it.. so I was watching the thread via the archive since I'm not on the list anymore. |
| 22:59.26 | bkw_ | you posted the LINK again |
| 22:59.52 | RoyK | just the same link that was posted in the original post |
| 23:00.01 | RoyK | which isn't bad |
| 23:00.02 | bkw_ | but it was scrubbed from the archives |
| 23:00.03 | RoyK | just copying old stuff |
| 23:00.10 | X-Rob_ | It was scrubbed from the archives? |
| 23:00.11 | X-Rob_ | *BAHAHAHA* |
| 23:00.14 | bkw_ | yep |
| 23:00.16 | X-Rob_ | they're censoring the archives? |
| 23:00.20 | bkw_ | as it should be |
| 23:00.24 | RoyK | well. I don't read the archives |
| 23:00.27 | bkw_ | X-Rob_, yes it isn't the first time |
| 23:00.32 | RoyK | i just read my fscking email |
| 23:00.34 | benjk | OpenPBX is now officially several orders of magnitude faster in dialplan execution than Asterisk |
| 23:00.52 | bkw_ | I recall something in the past getting scrubbed also |
| 23:00.58 | P4C0 | RoyK, humm but what is hurting now... my client sent invite with media port to 49352, local asterisk reply with media port 28152, local asterisk send invite to voip provider with media port 24968 voip provider reply with media port 11444, comunication starts... rtp packages between client and local asterisk go the way they should... local asterisk sent rtp to voipprovider with src 24968 dest 11444 and voip provider reply with src 11444 and dest 8540 !?? where |
| 23:00.58 | P4C0 | <PROTECTED> |
| 23:01.01 | bkw_ | it linked to some naughty stuff. |
| 23:01.12 | benjk | whazzat? |
| 23:01.13 | bkw_ | P4C0, sounds like you have crack headed nat |
| 23:01.22 | RoyK | ~google lagavulin |
| 23:01.35 | benjk | steroids? |
| 23:01.41 | P4C0 | bbw_ what you mean by crack headed nat? |
| 23:02.03 | RoyK | benjk: one of the best single malts of scotland |
| 23:02.11 | benjk | ah |
| 23:02.27 | P4C0 | my firewall/nat/router is forwarding all upd packages from 6000 to 31000 from my voip provider to the local ip of my * server |
| 23:02.31 | benjk | sounds like steroids or tranquilizers |
| 23:02.54 | Ryushin | Okay, time to work on this stuff again. |
| 23:03.23 | Ryushin | Any ideas on how to find out what the current bootrom version is of a ip430 phone. |
| 23:03.57 | Ryushin | The phone won't boot all the way. |
| 23:03.58 | P4C0 | bkw_, what do you mean by crack headed nat? |
| 23:05.05 | bkw_ | nat isn't doing 1 to 1 mapping for ports |
| 23:05.49 | *** join/#asterisk jeebusmobile (n=jeebusmo@130.sub-75-214-86.myvzw.com) |
| 23:06.55 | P4C0 | bkw_, humm but it should! humm I'll log all ports in the nat/router/firewall... let see |
| 23:08.39 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 23:08.42 | DasTech | hey all |
| 23:08.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:09.05 | DasTech | anyone have issues with polycoms 501 and asterisk when dialing 800 numbers |
| 23:09.16 | DasTech | some of them just ring |
| 23:09.51 | benjk | bkw, are you feeling alright today? |
| 23:17.38 | MRH2 | file can i ask u a quick question? |
| 23:18.04 | X-Rob_ | CrackNAT is l33t. |
| 23:18.58 | X-Rob_ | jbot: cracknat is something that SpeedTouch modems do - randomly change ports numbers on traffic going through them. Give it up, throw them away. |
| 23:19.07 | jbot | okay, X-Rob_ |
| 23:19.07 | file | hrm? yes |
| 23:19.23 | MRH2 | is that the mixmonitor fix in 1.2? |
| 23:19.39 | file | it's in the 1.2 branch, not in a release yet |
| 23:19.52 | MRH2 | fabulous! |
| 23:20.15 | MRH2 | i'm on the list of people owe u a beer |
| 23:20.53 | file | I'm *hoping* it solves everyone's issues ... but I make no promises until people test it and give me feedback |
| 23:21.19 | MRH2 | i'll give it a good go - thanks for the backport |
| 23:21.22 | X-Rob_ | file, FYI - freepbx uses mixmonitor and has for a while, so I'll hear about any problems. |
| 23:22.19 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 23:23.42 | batphone | which one of you wrote this perl script to reboot polycoms? |
| 23:23.59 | batphone | i just rebooted 700 phones remotely in just a few minutes... |
| 23:24.13 | batphone | just wanted to mail you a 12 pack. |
| 23:25.02 | X-Rob_ | Ooh |
| 23:25.03 | X-Rob_ | It was me |
| 23:25.08 | X-Rob_ | me me! |
| 23:26.05 | orlock | bit early isnt it? :) |
| 23:26.07 | X-Rob_ | *vomit* |
| 23:26.13 | X-Rob_ | RoyK, geez, c'mon. |
| 23:26.18 | RoyK | :D |
| 23:36.52 | *** join/#asterisk deb_user (n=Hypnotis@70-59-108-105.albq.qwest.net) |
| 23:37.16 | deb_user | does caller ID on a zap interface for incoming calls have anything to do with my telco? |
| 23:37.41 | deb_user | I'm unable to detect the incoming caller's phone # |
| 23:38.03 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 23:38.14 | P4C0 | is it normal all those option and 404 sip messages? |
| 23:38.45 | x86 | i've got a really weird issue |
| 23:39.27 | x86 | I setup a new DID, and when you call it, you hear no ringing tone at all, even though the extension is actually ringing, and the CLI says the extension is ringing |
| 23:39.32 | x86 | but the caller hears no ring tone |
| 23:39.42 | x86 | none of my other DIDs act like this |
| 23:39.53 | x86 | and it does not seem to matter what I do with the Dial string |
| 23:39.54 | deb_user | x86: did you include the option r in the dialplan? |
| 23:40.10 | x86 | deb_user: with or without r in the dial string the behavior is the same |
| 23:40.50 | x86 | i also tried exten => s,1,Answer exten => s,n,Ringing exten => s,n,Dial(SIP/xxx|1000|t) |
| 23:40.56 | x86 | no ring tone at all |
| 23:41.02 | x86 | now here is the weird part ;) |
| 23:41.29 | x86 | if i Playback a file before the Dial string, after the file is done playing, the ring tones work perfectly fine |
| 23:41.55 | ctrix | x86, that happens on sip ? |
| 23:42.04 | x86 | what do you mean? |
| 23:42.13 | x86 | oh |
| 23:42.16 | x86 | lol, read it wrong |
| 23:42.19 | x86 | yeah, SIP |
| 23:43.05 | x86 | I know Playback before Dial makes it work, as it was just something I tried for grins and it happened to work, but other things like a Wait might do the trick also, not sure |
| 23:43.07 | ctrix | that's your provider which soessilence suppression |
| 23:43.17 | ctrix | and you have no zaptel cards for timing |
| 23:43.31 | ctrix | and you are not usinf ztdummy |
| 23:43.31 | x86 | ztdummy is no good? |
| 23:43.37 | x86 | i am |
| 23:43.43 | ctrix | it should but not always works |
| 23:43.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:43.49 | x86 | hmm |
| 23:44.06 | x86 | i see |
| 23:44.10 | ctrix | there a solution, anyway. |
| 23:44.17 | ctrix | (but without *) |
| 23:47.00 | tzanger | interesting |
| 23:47.12 | tzanger | I seem to have WORSE echo when turning on MMX on a P4 |
| 23:47.21 | tzanger | on a wct1xxp |
| 23:47.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:51.25 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 23:51.29 | Dr-Linux | hey all |
| 23:51.58 | tessier__ | If I add t to the Dial command options in the dialplan does that allow blind transfer or attended transfer? |
| 23:52.05 | *** part/#asterisk P4C0 (n=ash@200.124.22.34) |
| 23:52.35 | Dr-Linux | anybody is using spa3000? |
| 23:54.26 | Dr-Linux | :S |
| 23:56.34 | *** part/#asterisk Amilcar_ (n=xxxxx@201.34.202.17) |
| 23:58.05 | hmmhesays | tzanger record it |
| 23:58.34 | tzanger | hmmhesays: yeah I'm going to |
| 23:59.13 | tessier__ | It seems I have no clue how to do attended transfers. Been well over a year since I had to do one. |
| 23:59.22 | tessier__ | The transfer button on the phone itself does blind transfer. |