| 00:00.04 | Ariel_ | SwK, but it still does not mean there being replaced. as far as I read the n+101 is still there and I don't see it being removed. |
| 00:00.05 | SwK_ | there was a post about it on the dev list or the bug tracker the other day |
| 00:00.42 | Ariel_ | SwK, if it's removed it would be very dumm move due to everyone that uses it will have to re-write all there macro's. |
| 00:01.06 | Ariel_ | This is a step backwards |
| 00:01.12 | SwK_ | Ariel_: aat some point in the future it might get removed, I know theres a pending patch that will start throwing warnings about it, then at some point they change the configs to turn it off by defualt unless you throw a config option to turn it back on (or atleast thats the way I read the notice) |
| 00:02.27 | SwK_ | actually it makes writing macros and complex dialplans much easier imho |
| 00:03.13 | Ariel_ | SwK, yes in the future but allot of people have systems with complex macro's working now. It stupid to make this a change instead of a addon |
| 00:03.17 | Nugget | the n+101 jumps are horrible to deal with. |
| 00:03.40 | SwK_ | Ariel_: that why they arent completely killing it |
| 00:03.49 | datagen24 | is any body around here who can look at my .conf files and see where i went wrong? |
| 00:03.51 | datagen24 | http://pastebin.ca/18667 |
| 00:03.51 | Ariel_ | Nugget, I agree but why would I want to spend weeks working on a dial plan that is working. |
| 00:04.43 | Nugget | because dialplans which use jumps are difficult to maintain, and migrating to a saner system will produce a less indirect, less error prone, and easier to maintain system for the long term |
| 00:04.46 | SwK_ | i'm trying to find the exact post |
| 00:04.59 | *** join/#asterisk flynux (127sdmz@pingou.in) |
| 00:05.52 | SwK_ | try adding 1 priority into a 40 priority extension with 1,2,3....40,141,142,143...154,155,156 is a pita at the very least |
| 00:06.28 | Nugget | or try using two applications which rely on n+101 exceptions handling near one another in the same sequence. |
| 00:07.04 | Nugget | you end up having to put goto placeholders at the respective +101 spots because there's not enough room to do anything on the exception side |
| 00:07.10 | Ariel_ | I have many rollover macro's and other things taken that already into account. |
| 00:07.29 | Nugget | sure, there are plenty of ways to work around it. |
| 00:07.38 | SwK_ | Ariel_: it was in an email to the -dev list from kpf on Jul26 |
| 00:07.52 | Nugget | but I'd argue that all of those techniques yield a dialplan which is more complicated than it ought to be |
| 00:08.22 | SwK_ | http://pastebin.ca/18668 |
| 00:08.31 | *** join/#asterisk Tiveron (~someone@66.146.140.5) |
| 00:09.21 | SwK_ | for easy to find I just cut/paste the body of his msg to that pastebin |
| 00:09.40 | hardwire | snom had some neat ideas |
| 00:09.44 | hardwire | the shared line |
| 00:09.49 | hardwire | kinda fun |
| 00:09.57 | hardwire | suppose thats quite hackable.. using meetme |
| 00:10.24 | hardwire | and meetme doesn't grok DTMF |
| 00:10.56 | *** join/#asterisk jartar45 (~root@node942.fsip.execulink.com) |
| 00:11.17 | jartar45 | ? |
| 00:11.25 | *** join/#asterisk kabewm (~kabewm@216.31.139.108) |
| 00:11.34 | *** join/#asterisk Coriantum (~asdfkle@67-41-182-243.slkc.qwest.net) |
| 00:12.55 | *** part/#asterisk kabewm (~kabewm@216.31.139.108) |
| 00:14.27 | jartar45 | does anyone here know if you need a soundcard in order to install Asterisk properly? I'm having trouble modprobe(ing) my x100p card right now. I'm also recieving errors when i try to instal the Zapetal tarbal |
| 00:14.52 | Nugget | You do not need a sound card, and in fact having extraneous cards can make zaptel a royal pain in the ass. |
| 00:14.55 | Ariel_ | SwK, thanks for the post. But it's like it said the option is going to be set via global settings. Which is why I don't use head until the get all these things working correctly. |
| 00:15.45 | harryvv | jar do a lspci |
| 00:15.46 | Ariel_ | jartar45, have you configured the zaptel.conf and the zapatal.conf files? |
| 00:16.05 | harryvv | Ariel_ he cannot modprobe his card |
| 00:16.06 | jartar45 | do you have any idea why im getting some error when trying to install zapetal? i think the reason why i cant modprobe my card is because of the errors i get when installing Zapetal. I can connect to Asterisk fine |
| 00:16.14 | *** part/#asterisk _deg_ (~deg@200.146.0.254) |
| 00:16.42 | harryvv | jartar45 try doing lspci and see if the card is showing up |
| 00:16.45 | Ariel_ | jartar45, yes what is the error your getting? |
| 00:16.55 | jartar45 | one sec, i'll post them |
| 00:17.09 | MicC_ | and cat /proc/interrupts to see if its sharing any |
| 00:17.09 | Ariel_ | harryvv, you can configure zaptel without a card installed |
| 00:17.41 | harryvv | Ariel_ okay was not sure about that. But thats a good idea if a card is late in the mail or backordered. |
| 00:17.48 | *** join/#asterisk sig- (sig@gnook.org) |
| 00:17.57 | eville83 | he haryvv |
| 00:18.01 | eville83 | hey i mean :) |
| 00:18.05 | Ariel_ | harryvv, you can always use ztdummy |
| 00:18.10 | harryvv | hi eville83 |
| 00:18.18 | eville83 | i'm jartar45 too. (on linux) |
| 00:18.25 | harryvv | yea dont know that much about ztdummy. |
| 00:18.32 | harryvv | ohh |
| 00:18.43 | jartar45 | the error i get with zaptel are here: www.remotetech.com/test2.txt |
| 00:18.47 | Nugget | building zaptel requires that the kernel source matching your kernel exist on your system where it expects to find it. |
| 00:19.14 | Nugget | you'll have problems if the kernel sources are absent or if your linux distro does nutty things with them. (fedora comes to mind here) |
| 00:19.19 | harryvv | nugget, yea I was talking to somone yesterday about that. |
| 00:19.22 | jartar45 | Nuggest: i installed redhat 9 i386 from cd. then i downloaded the updates to make my kernal i686 |
| 00:19.37 | Ariel_ | jartar45, you need to make linux26 |
| 00:19.39 | jartar45 | harryvv: it was me :) |
| 00:19.39 | Nugget | did you download the kernel sources, or did you just download a binary kernel package? |
| 00:19.41 | *** join/#asterisk phil0u (~philou@81.56.194.193) |
| 00:19.44 | harryvv | jar okay yes. |
| 00:20.06 | Nugget | you'll need to download the kernel sources package that corresponds to that i686 kernel package you installed, probably |
| 00:20.07 | phil0u | 'lo |
| 00:20.27 | Nugget | I thought that "make linux26" was legacy and no longer relevant. |
| 00:20.30 | *** join/#asterisk dasenjo (~dasenjo@63.245.86.33) |
| 00:20.37 | Nugget | and who said anything about 2.6? |
| 00:20.53 | jartar45 | i donwloaded kernel-smp-2.4.20-31.9.i686.rpm and it says its already installed |
| 00:21.01 | Nugget | that's the kernel |
| 00:21.04 | Nugget | not the source code |
| 00:21.20 | Nugget | find the kernel sources package which corresponds to that package |
| 00:21.24 | Ariel_ | the error's I saw was something to due with udev which is in 2.6 |
| 00:21.52 | Ariel_ | jartar45, which linux distro are you using? |
| 00:22.03 | Nugget | 19:19 <jartar45> Nuggest: i installed redhat 9 i386 from cd. then i downloaded the updates to make my kernal i686 |
| 00:22.23 | jartar45 | im using redhat |
| 00:22.28 | jartar45 | redhat 9 |
| 00:22.34 | sangee | Someone please help me, I want to execute a agi script when caller or callee hangup, it will jump to n+101 when caller or callee hangup? |
| 00:22.40 | SwK_ | Ariel_: yeah i know what you are saying, but at some point in the future people are only going to support priority jumping as a "legacy" mode... it wouldnt be a big leap from there to complete removal of that support for cleaner code base... |
| 00:22.56 | Ariel_ | jartar45, then if you have configured yum do yum install kernel-source |
| 00:23.56 | jartar45 | i havent configured yum, im searching google right now to find out how.... |
| 00:24.12 | Ariel_ | SwK, I am not saying it is not good for the add-on of better dialing rules. But you just can't remove people that have systems up and running fine for years either. |
| 00:24.35 | Ariel_ | jartar45, is this a new freash setup? |
| 00:25.01 | Ariel_ | fresh |
| 00:25.02 | jartar45 | Ariel_: i went through all the steps on the asterisk website. so it is partially set up |
| 00:25.24 | jartar45 | Ariel_: i just received the errors with the zaptel instalation and know i can't modprobe my card |
| 00:25.42 | jartar45 | harryvv was trying to help me install the source last night |
| 00:25.54 | Ariel_ | jartar45, ok look at the fedora legacy site for the settings for the rh 9 yum and updates. |
| 00:25.55 | jartar45 | but i guess in order to do that i have to configure yum then install it |
| 00:26.08 | jartar45 | Ariel_: ok |
| 00:26.23 | Ariel_ | all the sources for rh 9 is now kept there |
| 00:26.46 | jartar45 | i guess this is a good reference: http://www.fedoralegacy.org/docs/yum-rh9.php |
| 00:27.32 | Ariel_ | jartar45, that is a good start yes |
| 00:27.45 | jartar45 | thanx |
| 00:28.39 | SwK_ | if anything can be said about asterisk up to this point is kram is very fixed on maintaining backwards configs compatibility to keep people from having to completely re-write their dialplans etc.. with that in mind and knowing that just because something is depreciated, it doesnt mean its completely going away... so your old system is safe... however for future work, I would avoid it like the plague... its the same reason the unless theres a |
| 00:28.49 | Ariel_ | jartar45, if you want to start with a good one disk server setup look at CentOS 3.4 server. It's RH EL 3 without the name RedHat |
| 00:29.40 | Nugget | why does every question, no matter what it's about, always yield at least one guy who tells you to completely switch to a different distro? |
| 00:29.46 | Ariel_ | SwK, yes your correct. I fully agree. But since I don't use head yet. I will have to read more and look into it later. |
| 00:29.59 | Nugget | "Hey, how can I use colorls?" "You should be running ubuntu, dude!" |
| 00:30.10 | SwK_ | hah |
| 00:30.13 | harryvv | nugget, i get alot of that |
| 00:30.29 | harryvv | i could care less about the distro as long as i can use it and its stable :) |
| 00:30.33 | Ariel_ | Nugget, it's not a different it just an update rh 9 which very good is no longer available with updates. Unless you go the legacy route |
| 00:30.38 | SwK_ | just run ${YOUR_FAVORITE_PLATFORM} |
| 00:30.44 | SwK_ | its always the best |
| 00:30.48 | Nugget | naturally |
| 00:30.51 | jartar45 | Ariel_: i'm running a webserver as well on this server. i'm looking for something really stable that will work well with Cpanel/apache |
| 00:32.09 | harryvv | We are now getting the summer heat up here. |
| 00:32.50 | Ariel_ | jartar45, your running webservers on the system along with the Asterisk PBX.. humm well it does work. But is it wise unless it's for support of the asterisk box. (Just my view). |
| 00:34.03 | harryvv | since its your phones |
| 00:34.07 | harryvv | :) |
| 00:34.16 | harryvv | I agree |
| 00:34.35 | harryvv | kiss the asterisk box and expect running reliable system. |
| 00:37.30 | *** join/#asterisk alt (~donovan@relay.gwsn.com) |
| 00:38.32 | alt | I don't know if anyone is keeping track of this stuff, but I've got a Quad 3.2GHz Xeon with 2GB of RAM doing 54 active SIP channels (SIP to SIP calls) and the system is 99% idle. |
| 00:38.46 | alt | sorry |
| 00:38.52 | alt | 90% idle |
| 00:39.04 | _DAW | alt - any transcoding going on there? |
| 00:39.05 | alt | and there's a couple of transcoding sessions in there too. |
| 00:39.07 | alt | yes |
| 00:39.15 | _DAW | how many? |
| 00:39.17 | alt | 1 right at the moment |
| 00:39.19 | alt | but it varies |
| 00:39.33 | alt | we're trying to _not_ transcode |
| 00:39.43 | alt | but unfortunately, we do have to proxy all the RTP streams. |
| 00:40.24 | alt | if I had the know-how, I'd design a DSP card for transcoding instead of doing it in the CPU :\ |
| 00:40.51 | jartar45 | Arel_: at this point i can not afford to run 2 servers. I'm only running one line for Asterisk so im not expecting it to go to crazy. plus my server is pretty quick and should be able to handle it |
| 00:40.54 | alt | _DAW: nah. just been there before. |
| 00:41.19 | *** join/#asterisk paulankaster (Paul@201.239.125.148) |
| 00:41.27 | alt | okay, 52 SIP channels with 2 transcode sessions is 86% idle |
| 00:41.36 | alt | make that 3 transcode |
| 00:41.39 | alt | make that 3 transcode |
| 00:41.49 | alt | our goal is a DS3 :) |
| 00:42.12 | _DAW | you have a few channels to go :) |
| 00:42.40 | alt | yes indeedy |
| 00:42.54 | alt | a DS3 is 644 channels (assuming 23 channel PRIs) |
| 00:43.01 | Ariel_ | jartar45, no problem. how many users will you have on the system? |
| 00:43.28 | _DAW | anyh idea how the digium t3 card is coming? |
| 00:43.38 | alt | no idea. |
| 00:43.43 | alt | we're doing all SIP here. |
| 00:43.48 | alt | well, SIP and H.323. |
| 00:43.56 | paulankaster | I am having problems with my first install |
| 00:43.56 | alt | but we put the H.323 through a GNU Gatekeeper |
| 00:44.09 | alt | paulankaster: and this makes you different from everyone else how? ;-) |
| 00:44.26 | jartar45 | Ariel_: just myself and one other person for right now. |
| 00:44.28 | Ariel_ | paulankaster, is there a question you want to ask us aobut? |
| 00:44.38 | paulankaster | offcourse |
| 00:44.44 | paulankaster | the think is ... |
| 00:44.45 | Ariel_ | jartar45, like I said no problems then. |
| 00:44.53 | alt | 66 and 4 = 83% idle :) |
| 00:44.56 | paulankaster | I have installed asterisk at home 1.3 |
| 00:45.04 | paulankaster | it start all process ok |
| 00:45.24 | paulankaster | but when I try to connect with x-ten |
| 00:45.41 | paulankaster | it doesn't reply |
| 00:46.01 | *** join/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 00:46.10 | blitzrage | evening all |
| 00:46.14 | paulankaster | I run ngrep and only see packets from the windows box |
| 00:46.26 | paulankaster | nothing comming back from the asterisk box |
| 00:46.54 | alt | paulankaster: can you ping? |
| 00:47.06 | paulankaster | yes ping an ssh work great |
| 00:47.17 | Ariel_ | paulankaster, lets see you setup the user or extenion via the amp interphase, are they behind a nat firewall? |
| 00:47.25 | blitzrage | iptables -L -v |
| 00:47.37 | paulankaster | nop directly attached |
| 00:47.50 | paulankaster | via an crossover UTP cable |
| 00:48.06 | paulankaster | the user is an extension |
| 00:48.13 | SoloFlyer | does your box have multiple ip address on any of its interfaces? |
| 00:48.15 | paulankaster | I only created the 200 ext |
| 00:48.18 | Ariel_ | xlite is on a pc and the asterisk box on another one? |
| 00:48.24 | paulankaster | yes it does have 2 IPS |
| 00:48.28 | paulankaster | 2 nics |
| 00:48.39 | SoloFlyer | but not on 1 nic |
| 00:48.40 | paulankaster | yes 2 boxes |
| 00:48.50 | paulankaster | 1 win98 1 linux |
| 00:49.21 | *** join/#asterisk gamicalguy (and@c-24-99-71-88.hsd1.ga.comcast.net) |
| 00:49.28 | paulankaster | I have modified the *.conf in /etc/asterisk |
| 00:49.49 | paulankaster | to bind the IP of the nic that is directly attached to the windows box |
| 00:49.53 | bkw_ | iax is sucking |
| 00:50.11 | gamicalguy | im running a asterisk live cd and it says es3210.c not card found |
| 00:50.15 | *** join/#asterisk Saaib (~nabudocon@ns1.ensenada.gob.mx) |
| 00:50.31 | bkw_ | gamicalguy, um try using google |
| 00:50.32 | bkw_ | :P |
| 00:50.36 | bkw_ | I think thats an OS issue |
| 00:50.39 | bkw_ | and not an asterisk related one |
| 00:50.44 | *** join/#asterisk hermie (~nick@24-236-167-53.dhcp.bycy.mi.charter.com) |
| 00:51.36 | gamicalguy | oh, i didnt know that, i just have a live cd for aterisk and it gave me that error. ive run stuff like knoppix before and it works fine |
| 00:52.06 | gamicalguy | not really sure how to fix it and not really sure what to search for on google |
| 00:52.39 | sig- | gamicalguy then try use gogle.. |
| 00:52.42 | dudes | gamicalguy - the error for starters |
| 00:52.44 | sig- | or.. |
| 00:52.48 | sig- | ~mailinglist |
| 00:52.48 | jbot | mailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 00:54.06 | gamicalguy | so no one knows off the top of their head a solution? |
| 00:54.17 | dudes | gamicalguy - not here |
| 00:54.18 | *** join/#asterisk znoG (~gs@200.115.216.109) |
| 00:54.42 | sig- | why the ppl are so lazy. |
| 00:55.15 | dudes | sig - I've often asked that same question |
| 00:55.15 | gamicalguy | i tried to search google for my error and it didnt turn up with anytihg useful... |
| 00:55.52 | dudes | try search for bits and pieces of the error |
| 00:56.11 | dudes | that may aid in fixing the issue. or don't use a live CD for asterisk ? |
| 00:56.30 | gamicalguy | im just playing around with asterisk, i dont wnat to sign up for a mailing list to get thousands of emails a day, i was just triyng to get a direct answer, geez |
| 00:56.50 | dudes | search the mailing list |
| 00:56.57 | dudes | ~mailinglist |
| 00:56.57 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 00:57.02 | sangee | Hi, someone help me, how do i catch the calling party hangup in extensions.conf? |
| 00:57.05 | gamicalguy | i heard you the first time |
| 00:57.07 | dudes | read what jbot wrote |
| 00:57.14 | gamicalguy | this is dumb, no one is useful here |
| 00:57.39 | sig- | =) |
| 00:57.44 | sig- | funny guy. |
| 00:57.47 | dudes | he didn't read that |
| 00:58.11 | paulankaster | question |
| 00:58.24 | dudes | ask ... |
| 00:58.35 | paulankaster | how log take to process a subscription lo lists.digium.... |
| 00:58.38 | dudes | you don't need to raise your hand =0 |
| 01:00.00 | SoloFlyer | i dont know Paul but it took about 30mins for bugs.digium.com for me |
| 01:00.01 | dudes | you don't have to subscribe. You can just search it. |
| 01:00.02 | paulankaster | I have aplly over the web this noon , and posted a msg but nothing get to my inbox |
| 01:00.23 | dudes | no one has responded ? |
| 01:00.31 | paulankaster | I know but my intention is to post |
| 01:00.38 | paulankaster | not to search |
| 01:00.56 | SoloFlyer | what did u post.. |
| 01:01.11 | dudes | Well if a subscription to the mailing list takes as long as getting the licenses ... I'd say roughly 24hr's |
| 01:01.23 | paulankaster | I haven't receive neither the confirmation from the subscription |
| 01:01.23 | dudes | g729 licenses that is |
| 01:01.27 | ManxPower | ~docs |
| 01:01.27 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 01:01.29 | ManxPower | ~mailinglist |
| 01:01.29 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 01:04.45 | Darwin35 | man I love this new iax phone |
| 01:05.03 | brookshire | and what is it? |
| 01:07.12 | Darwin35 | netweb 401 |
| 01:07.27 | Darwin35 | aka x401 |
| 01:07.44 | Darwin35 | iax2 based |
| 01:07.47 | brookshire | hmm |
| 01:07.56 | Darwin35 | no more sip for me |
| 01:08.00 | SoloFlyer | i loved the GXP2000 when i got them |
| 01:08.21 | Darwin35 | this one has opensrc for the firmware |
| 01:08.28 | Darwin35 | I can learn it and dev it |
| 01:08.32 | ManxPower | SoloFlyer: I guess you've never used the Polycom 500 phone then |
| 01:08.56 | SoloFlyer | ManxPower i said when i got them i didnt say what i thought of them now :) |
| 01:09.35 | SoloFlyer | nahh they arent that bad |
| 01:09.47 | SoloFlyer | but speakerphone should not be on the features list |
| 01:09.53 | Darwin35 | grandstream is a great starter /low cost phone |
| 01:09.59 | Darwin35 | but I am over sip |
| 01:10.02 | DarthClue | Compared to a GS BT, the 2000 is very nice. Compared to a Polycom IP501, the 2000 is what I would expect to find in a slum lawyers office. |
| 01:10.27 | Darwin35 | no thats the cisco 7940 |
| 01:10.32 | Darwin35 | lol |
| 01:10.52 | SoloFlyer | other than speakerphone 2000 has proved to be very very good |
| 01:11.21 | SoloFlyer | but... Speakerphone is of major importance to me |
| 01:11.48 | SoloFlyer | if the open sourced it i could probally fix it but alas |
| 01:12.05 | Darwin35 | I just wish most sip/iax/h323 phones would get thier act together and put head phone jacks |
| 01:12.18 | SoloFlyer | 2000 has headphone jacks |
| 01:12.20 | ManxPower | SoloFlyer: LOL! Grandstream has got some of the most horrible firmware out there. |
| 01:12.34 | SoloFlyer | yeah i know |
| 01:12.41 | Darwin35 | using the radio shack addon that works off 2 aaa batt and only last abour 25 talk hours |
| 01:12.47 | ManxPower | Darwin35: Um, what phones DON'T have a headset port? |
| 01:13.07 | Darwin35 | like headphone jacks |
| 01:13.13 | Darwin35 | most dont |
| 01:13.23 | Darwin35 | they have speaker phones |
| 01:13.29 | ManxPower | The polycoms all support RJ-12 headsets. |
| 01:13.45 | ManxPower | The SIPura SPA-841 has a 2.5mm headset jack. |
| 01:13.51 | Darwin35 | and what are like rj jacks for headset plugi ons |
| 01:14.54 | ManxPower | The Cisco 7905 I have doesn't seem to have any headset/headphone jacks |
| 01:15.09 | hardwire | I am so not working |
| 01:15.12 | hardwire | and I should be |
| 01:15.23 | ManxPower | Darwin35: Huh? |
| 01:15.25 | Darwin35 | the 7920 and 7940 have rj12 jacks |
| 01:15.32 | hardwire | contimplating a way to make a caller id magically turn into a google search that will appear on a registered computer.. is a waste of time |
| 01:15.43 | ManxPower | the chinnese import IAX2 phone doesn't seem to have a headset jack. |
| 01:15.54 | Darwin35 | which one |
| 01:16.00 | ManxPower | I dunno. |
| 01:16.02 | Darwin35 | there is a imported iax phone |
| 01:16.15 | ManxPower | It blew up when I tried upgrading the firmware remotely and I've not had the interest to fix it. |
| 01:16.24 | ManxPower | It's so ugly I can't deploy it in production |
| 01:16.26 | *** join/#asterisk meppl (~mephisto@87.193.6.226) |
| 01:16.27 | SoloFlyer | actually if we developed a custom firmware for gxp2000 im sure it wouldnt take long for the opensource firmware to be better than grandstreams i means its a great phone with good potential its just that the firmware sucks ( thats what causes the speakerphone not to work for example) |
| 01:16.42 | Darwin35 | is it the pa168 chip ? |
| 01:16.58 | ManxPower | SoloFlyer: Pretty much ALL speakerphone suck, except for the high end like Cisco and Polycom |
| 01:16.59 | Darwin35 | its easy to fix with windows laptop and a crossover cable |
| 01:17.11 | ManxPower | Darwin35: When the phone won't even come out of POST? |
| 01:17.19 | DarthClue | hardwire: you could do that with firefox chrome. |
| 01:17.24 | SoloFlyer | manx there is a difference between suck and usable :) |
| 01:17.40 | ManxPower | I've been told you can press * or something when you power on the phone, but I've not bothered to try it. |
| 01:17.42 | Darwin35 | yes |
| 01:17.45 | hardwire | DarthClue: yeh.. first thought was to use google phonenumbers search engine to give me a map and so forth of the caller ID |
| 01:17.46 | ManxPower | I need to use the phones my users use. |
| 01:17.49 | Darwin35 | if its the pa168 chip |
| 01:17.53 | hardwire | I think I will just have to bite the bullet on this one |
| 01:18.09 | DarthClue | SoloFlyer: i too have considered creating firmware for the gxp2000, but it will have to wait a while at this rate. |
| 01:18.21 | DarthClue | hardwire: it can be done. |
| 01:18.27 | hardwire | DarthClue: it shouldn't be |
| 01:18.30 | hardwire | not on my time :) |
| 01:19.33 | hardwire | So3kris: any relation to www.soekris.com ? |
| 01:19.34 | SoloFlyer | lol |
| 01:19.34 | *** join/#asterisk zotz (~zotz@24.231.36.100) |
| 01:19.46 | hardwire | http://www.flickr.com/photos/telstar/29012853/ |
| 01:19.53 | Darwin35 | get a x401 and help dev it |
| 01:20.05 | DarthClue | ManxPower: um, it's for educational reasons, i doubt it would go very far, especially if we claim to be heavily christian but moderately muslim |
| 01:20.23 | SoloFlyer | darwin itsa little late for that i already have 32 granstream gxp2000's |
| 01:20.37 | hardwire | send me one |
| 01:20.37 | Darwin35 | ahh wow |
| 01:20.45 | SoloFlyer | lol |
| 01:20.47 | hardwire | I will send you a few snom 360's |
| 01:20.48 | *** join/#asterisk iq (~iq@207-224-100-81.omah.qwest.net) |
| 01:20.48 | hardwire | heh |
| 01:20.50 | hardwire | in fact |
| 01:20.52 | hardwire | I have 32 snoms |
| 01:20.54 | Darwin35 | I have 4 gs102 and 2 x401 |
| 01:20.54 | hardwire | wanna trade? |
| 01:21.11 | Darwin35 | and 1 main board from a gs100 |
| 01:21.23 | SoloFlyer | why hardwire? |
| 01:21.26 | iq | hi |
| 01:21.32 | Darwin35 | wich is becoming a paging module |
| 01:21.34 | hardwire | SoloFlyer: I want some grandstreams :) |
| 01:21.46 | Sedorox | get a Bt100 |
| 01:21.47 | Sedorox | :p |
| 01:21.49 | *** join/#asterisk Inv_arp (junya@adsl-156-144-76.mia.bellsouth.net) |
| 01:22.03 | Error_X | hardwire: Got ztdummy installed.. But how do I get it to work with asterisk? :) |
| 01:22.07 | SoloFlyer | i just wish gs would open theire firmware :( |
| 01:22.12 | hardwire | it just does (tm) |
| 01:22.13 | Darwin35 | you have to load it |
| 01:22.14 | SwK_ | hardwire: I doubt that soren (mr. Soekris.com) uses irc |
| 01:22.20 | Darwin35 | and then run ztcfg |
| 01:22.24 | Error_X | k |
| 01:22.31 | hardwire | Darwin35: is ztdummy modprobed? |
| 01:22.38 | SoloFlyer | i mean their support sucks in order to get 1 email from them i have to send the 5 |
| 01:22.45 | SwK_ | yes you can modprobe xtdummy |
| 01:22.47 | *** join/#asterisk Beccara (~Tristram@210-86-111-87.jetstream.xtra.co.nz) |
| 01:22.54 | hardwire | do you have a /proc/zaptel dir |
| 01:23.00 | SwK_ | and if you are using ztdummy you dont have to run ztcfg |
| 01:23.10 | *** join/#asterisk santiago (~santiago@63.245.86.222) |
| 01:23.23 | Error_X | it is loaded |
| 01:23.35 | SwK_ | the just start asterisk and make sure chan_zap loads |
| 01:23.55 | SwK_ | the do zap show channels you should see 1 pseudo chan |
| 01:24.01 | Error_X | It works :) |
| 01:24.02 | Darwin35 | but on linux why use ztdummy when you have rtc |
| 01:24.04 | Error_X | thanks alot :) |
| 01:24.23 | SwK_ | Darwin35: the RTC doesnt have fine enuff timers |
| 01:24.52 | Darwin35 | ahh ok |
| 01:25.39 | SwK_ | thats why in 2.4 kernels it needs specific USB chip... in 2.6 the kernel has high resolution timers |
| 01:26.31 | SoloFlyer | can u do 1 to many in asterisk? |
| 01:26.47 | SoloFlyer | one way only... |
| 01:26.50 | DarthClue | SoloFlyer: define 1 to many? |
| 01:27.17 | SoloFlyer | i rin 1 number 32 phones automaticlly pick up the call |
| 01:27.18 | Darwin35 | yes its called a confrence call/page |
| 01:27.46 | SoloFlyer | yeah but 1 way only |
| 01:27.47 | Darwin35 | you have a agi call all the extensions and put them in meetme then you join |
| 01:28.02 | DarthClue | i don't think that's what he wants. |
| 01:28.05 | Darwin35 | make your announcement and hangup it disconnects them |
| 01:28.31 | SoloFlyer | yeah thats what i want... |
| 01:28.35 | *** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net) |
| 01:28.47 | DarthClue | ok, then you would have to have phones that can autoanswer. |
| 01:28.58 | SoloFlyer | gxp2000 can |
| 01:28.59 | SwK_ | SoloFlyer: theres a script for that on the wiki |
| 01:29.08 | SwK_ | it uses meetme |
| 01:29.15 | SoloFlyer | voip-info.org wiki... |
| 01:29.22 | SwK_ | note: if this is for a real world business deployment most people hate |
| 01:29.45 | SwK_ | search the wiki for polycom intercom or paging |
| 01:30.11 | SwK_ | you're best bet is to use a paging amp some overhead speakers and a paging amp heh |
| 01:30.14 | Qwell | patch a single phone into the speaker system...much funner |
| 01:30.16 | Darwin35 | the cisco and grandstreams have auto answer |
| 01:30.19 | Qwell | SwK_++ |
| 01:30.44 | SoloFlyer | mind u the gs autoanswer is a Hack! |
| 01:30.55 | SoloFlyer | u have to set up a second user |
| 01:30.57 | Darwin35 | its in the firmware |
| 01:31.03 | Darwin35 | ahh ok |
| 01:31.07 | SwK_ | the polycom autoanswer is done with sip alerts on the invites |
| 01:31.07 | Darwin35 | did not know that |
| 01:31.11 | SoloFlyer | yeah |
| 01:31.15 | SoloFlyer | u have normal user |
| 01:31.16 | Darwin35 | wow |
| 01:31.25 | SoloFlyer | plus a second user which auto answers |
| 01:31.32 | SoloFlyer | its a little dodgy but it works |
| 01:31.41 | SoloFlyer | just another firmward bug :/ |
| 01:31.42 | Darwin35 | soem one needs to doc it |
| 01:31.52 | Error_X | rtpstart=10000, rtpend=20000 is this a port range? |
| 01:32.18 | SwK_ | with the polycom you just set the _SIP_ALERT (i think thats right) channel variable to autoanswer or whatever and its send in the INVITE, then you config the phones to look for it... its much fun to play with |
| 01:32.27 | SwK_ | Error_X: yes |
| 01:32.31 | Error_X | Error_X: k |
| 01:32.43 | Error_X | TCP or UDP? |
| 01:32.52 | SwK_ | UDP |
| 01:33.10 | Darwin35 | well I have to put that on the list for firmware for the pa168 chip |
| 01:33.35 | SwK_ | do you really want the retransmitting of packets by TCP when you are talking... you'll get crap like "HiHiHi this is B B B Bob |
| 01:33.39 | SoloFlyer | Darwin35 - http://voip-info.org/tiki-index.php?page=GXP-2000 <--- Still no real intercom system... I have been able to work around this by setting up a seperate user account on line4 which has auto answer enabled :) |
| 01:34.11 | SoloFlyer | i think i added that i cant remember... |
| 01:34.47 | SoloFlyer | yeah i did |
| 01:36.28 | SoloFlyer | i also added that Warning at the top lol |
| 01:39.32 | Darwin35 | oook now I need a video phone for the front door |
| 01:43.41 | *** part/#asterisk paulankaster (Paul@201.239.125.148) |
| 01:44.16 | *** join/#asterisk Monnok (~Monnok@67-41-182-243.slkc.qwest.net) |
| 01:45.54 | Coriantum | Hi Monnok |
| 01:46.13 | Monnok | Hey |
| 01:48.08 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca) |
| 01:48.14 | SarahEmm | hihi |
| 01:50.05 | DarthClue | how are you this evening SarahEmm? |
| 01:50.21 | sivana | hey |
| 01:50.30 | sivana | SarahEmm: did you try it again? |
| 01:51.00 | Coriantum | Is there a way to execute multiple exention.conf files? |
| 01:51.12 | Qwell | Coriantum: You can include another file |
| 01:51.25 | Coriantum | Qwell: cool! thanks |
| 01:51.32 | Qwell | #include I believe |
| 01:53.36 | SarahEmm | sivana: trying again |
| 01:53.41 | SarahEmm | DarthClue: umm.... i dunno |
| 01:53.47 | SarahEmm | DarthClue: not great but surviving |
| 01:54.46 | wulfy814 | why do I have to run ztcfg prior to starting asterisk everytime |
| 01:55.13 | SarahEmm | wulfy814: you should only have to right after loading the modules |
| 01:55.15 | ManxPower | wulfy814: because your /etc/modules.conf is wrong |
| 01:55.50 | ManxPower | If the /etc/modules.conf is correct, ztcfg will be automagically run when the card driver (or zaptel?) is loaded. |
| 01:56.16 | SarahEmm | sivana: testing now. works OK from my local phone, so i know it's registering right |
| 01:56.17 | *** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net) |
| 01:56.19 | SarahEmm | testing from the US now |
| 01:56.59 | wulfy814 | /etc/modules.conf has zaptel, wctdm - should there be anything else? |
| 01:57.47 | sivana | ok |
| 01:57.57 | ManxPower | you need a post-install line too, of course. |
| 01:58.02 | SarahEmm | <PROTECTED> |
| 01:58.03 | SarahEmm | ^-- sivana |
| 01:58.04 | SarahEmm | still |
| 01:58.14 | ManxPower | Granted, "make install" in the zaptel source directory would have set that up for you. |
| 01:58.27 | SarahEmm | sivana: that's from SBC |
| 01:58.32 | sivana | hrm.. they tested it from NY today |
| 01:58.34 | SarahEmm | cingular does the same |
| 01:58.37 | SarahEmm | so did i sivana |
| 01:58.40 | SarahEmm | from Rochester it doesn't work |
| 01:58.43 | SarahEmm | same message |
| 01:58.52 | wulfy814 | ManxPower: did make install... this is Ubuntu install |
| 01:59.01 | wulfy814 | I have it working fine on a machine with just ztdummy |
| 01:59.15 | ManxPower | wulfy814: Huh? Download and install from source or we can't help you. |
| 01:59.16 | sivana | SarahEmm: can you try from a US landline? |
| 01:59.25 | ManxPower | wulfy814: 2.4 kernel or 2.6 kernel? |
| 01:59.30 | wulfy814 | ManxPower: I did install from source 2.6 kernel |
| 01:59.37 | wulfy814 | just letting you know what distro |
| 01:59.46 | ManxPower | wulfy814: Ah. 2.6 does modules totally differently |
| 01:59.46 | SarahEmm | sivana: err, i have been. SBC and pacbell and whatever is in Rochester |
| 01:59.52 | SarahEmm | sivana: and also cingular but that's not landline |
| 02:00.41 | sivana | hrm... I'll dbl check with them again |
| 02:00.52 | SarahEmm | k |
| 02:00.52 | sivana | PacBell isn't in Rochester though :) |
| 02:00.53 | SarahEmm | thanks |
| 02:00.55 | SarahEmm | i know |
| 02:00.56 | Coriantum | Does #include work in AEL? |
| 02:01.06 | SarahEmm | i was saying pacbell, SBC, *and* whatever's in Rochester (didn't ask) :) |
| 02:01.12 | sivana | hehe |
| 02:01.18 | SarahEmm | i'm having like 4 people around the country test this each time |
| 02:01.22 | SarahEmm | just to make sure it's not just one provider |
| 02:01.36 | sivana | ok |
| 02:03.07 | mmlj4 | hey ManxPower :-) |
| 02:03.55 | Error_X | allow=all = use alot of bandwith? |
| 02:03.55 | ManxPower | mmlj4: Hiya |
| 02:03.57 | mmlj4 | got my screwy voicemail? |
| 02:04.08 | ManxPower | Error_X: No. Allow=all means "screw up my calls" |
| 02:04.17 | Error_X | aha |
| 02:04.27 | ManxPower | mmlj4: I listeded to the first 10 seconds of it. 8-) Figured I'd listen to the rest of it before I call you back |
| 02:04.45 | Error_X | when I call my meetme room its chopping (both machines are on LAN) |
| 02:05.33 | mmlj4 | well, my plan is more expensive and takes 4 days, better just ignore it :-) |
| 02:06.33 | mmlj4 | but i do hope to be in anaheim in october |
| 02:07.51 | *** join/#asterisk NormAst (HydraIRC@CPE000800c0c891-CM0012c90d3496.cpe.net.cable.rogers.com) |
| 02:09.40 | ManxPower | mmlj4: maybe you can see me speak |
| 02:10.30 | sivana | SarahEmm: ping |
| 02:11.04 | mmlj4 | maybe... what track or topic will you be doing? |
| 02:12.39 | SarahEmm | pong |
| 02:12.41 | *** join/#asterisk kingtux (~susekid@pool-141-157-9-215.balt.east.verizon.net) |
| 02:12.54 | kingtux | Can anyone give me some help with amp and asterisk |
| 02:12.56 | kingtux | ?? |
| 02:13.04 | ManxPower | kingtux: Try #amp |
| 02:13.12 | kingtux | ok |
| 02:13.16 | kingtux | thanks |
| 02:14.00 | ManxPower | Asking us to help you with AMP is like going into a BMW repair shop and asking for help on the 3rd party stereo system you had a non-BMW dealer install. |
| 02:14.21 | SarahEmm | hehee |
| 02:14.27 | SarahEmm | good acronym ManxPower |
| 02:14.48 | kingtux | yeah pretty funny |
| 02:15.17 | mmlj4 | s/acronym/analogy/ |
| 02:15.39 | SarahEmm | err |
| 02:15.39 | SarahEmm | yeah |
| 02:15.41 | SarahEmm | thanks mmlj4 |
| 02:15.43 | SarahEmm | s'what i meant |
| 02:15.47 | SarahEmm | kitriches are having issues with communication today |
| 02:15.48 | mmlj4 | i know :-) |
| 02:16.06 | ManxPower | mmlj4: Did you install those wall jacks in Mandeville that don't have any labels on them? |
| 02:16.37 | mmlj4 | um, which? |
| 02:16.59 | ManxPower | There are wall jacks/plates in use that have no numbers on them |
| 02:17.08 | mmlj4 | labels? i don't need no steeking labels |
| 02:17.30 | ManxPower | Well if you were the one, don't admit it. If I find the person I'll have Guido pay them a visit. |
| 02:17.40 | mmlj4 | well, have you see my diagram? each jack is basically sequentially-numbered |
| 02:18.19 | mmlj4 | you start by entering a room, looking to the left, and the jacks are in sequence around the room |
| 02:19.00 | mmlj4 | the map will tell you which block of ports on which patch panel belong to which room |
| 02:20.03 | ManxPower | Maps can be lost. We don't lose patch pannels or wall jacks. |
| 02:20.29 | ManxPower | I think John had to tone back every one of the Platinum ports. |
| 02:20.49 | mmlj4 | um, hmm... |
| 02:21.24 | ManxPower | Maybe he lost the map 8-) |
| 02:21.30 | mmlj4 | yeah, that set of rooms are a little messed up... i miscalculated when punching those down |
| 02:21.45 | SoloFlyer | lol |
| 02:22.01 | mmlj4 | no, the principle is the same, you start in the back room and work towards the reception area |
| 02:22.26 | ManxPower | mmlj4: I'll ask John if we can pay you to come in and make sure they are all labeled correctly for any that we have questions about. |
| 02:22.32 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:22.50 | JunK-Y | mooo |
| 02:22.57 | mmlj4 | well, i don't know if I need to be paid for that, that falls more under the idea of support, i think |
| 02:23.08 | SoloFlyer | mmlj4 just like i miscalculated when i used a 15pound sledge hammer to crack an egg :) |
| 02:23.27 | ManxPower | mmlj4: Having it done will save hours of work in the future. |
| 02:23.34 | mmlj4 | you find a bad jack, i go f1x0r it |
| 02:23.52 | SoloFlyer | lol |
| 02:24.03 | mmlj4 | well, the map idea for the main part of the building is a reasonably sound idea |
| 02:24.10 | mmlj4 | we can photocopy that map |
| 02:24.15 | ManxPower | regardless, I'm very happy that we have a wiremonkey again. |
| 02:24.30 | ManxPower | mmlj4: The thing is that maps get out of date very quickly. |
| 02:24.32 | mmlj4 | hey, no biggie :-) |
| 02:24.58 | mmlj4 | well, those rooms are not going to relocate themselves, and I doubt those patch panels will get redone |
| 02:25.19 | ManxPower | mmlj4: We thought the same thing about the Covington NOC 8-) |
| 02:26.00 | mmlj4 | if/when the tenant portion gets wired, i'll be sure to actually label them; ditto any future construction work |
| 02:26.46 | ManxPower | mmlj4: If you spent as many hours toning/tracking back wall jacks you would be paranoid about the issue too 8-) |
| 02:26.56 | mmlj4 | aye |
| 02:27.47 | mmlj4 | but seriously, i can pinpoint any jack in the building, give or take 1 panel port, by referring to the map |
| 02:28.01 | ManxPower | mmlj4: then go around and put labels on them. 8-) |
| 02:28.04 | mmlj4 | hehe |
| 02:28.14 | *** join/#asterisk file (~jcolp@mctn1-6719.nb.aliant.net) |
| 02:29.10 | mmlj4 | again, point taken, all future jobs will be labelled |
| 02:29.13 | ManxPower | mmlj4: We will be putting the entire office on 3 VLANs next week. |
| 02:29.19 | mmlj4 | cool |
| 02:29.57 | ManxPower | mmlj4: Also, I ask users what the label on the wall jack is when they report problems, and since I'm not on-site..... |
| 02:30.27 | mmlj4 | makes sense |
| 02:31.57 | *** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122) |
| 02:33.16 | harryvv | manx, using cisco switches? |
| 02:33.47 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-6719.nb.aliant.net) |
| 02:36.45 | ManxPower | harryvv: Cisco 5509 |
| 02:36.58 | harryvv | I programed thise in cisco acadamy |
| 02:37.08 | harryvv | not to hard to configure. |
| 02:38.05 | ManxPower | harryvv: Somewhat harder when the PCs are plugged into Polycom phones and we need the PC and the polycom on different VLANS |
| 02:38.27 | Qwell | ManxPower: scary |
| 02:38.46 | harryvv | putting them on seperate vlans is a good idea |
| 02:38.57 | irv999 | lalalal |
| 02:39.26 | ManxPower | harryvv: we will have different security policies for the phones/corporate PCs .vs. the agent PC's .vs. the printers |
| 02:39.37 | harryvv | yea :) |
| 02:39.45 | harryvv | and peace of mind |
| 02:40.31 | *** join/#asterisk mrproper_ (~psynode@CPE-60-225-51-188.nsw.bigpond.net.au) |
| 02:40.37 | mrproper_ | anyone here using oh323? |
| 02:41.32 | harryvv | called my cable company, want to charge 80 dollars for a second fixed ip address. |
| 02:41.51 | harryvv | I am sure any dsl provider can beat that price. |
| 02:43.24 | ManxPower | harryvv: Um, actually "creating a better image for MIS" is the reason. The agents are unhappy about the restrictions we put on their internet access. |
| 02:43.40 | ManxPower | And since they pay about US$38,000/year to work for the company..... |
| 02:43.49 | ManxPower | Yes, real estate is a fucked up industry. |
| 02:44.11 | opus_ | dude we can get 100 ip for $5 |
| 02:44.40 | harryvv | manx, thay pay your company 38k? |
| 02:45.10 | harryvv | opus, yea...our cable provider is ripping us off..saying thay dont have enough ip's to shelv out. |
| 02:45.11 | ManxPower | harryvv: Yes, AND they have to give the company part of the comissions. Not MY company, my largest customer. I'm an indie consultant |
| 02:45.40 | harryvv | so you provide IT services to a realestate agency? |
| 02:45.56 | ManxPower | The better agents, the ones that sell high end homes, can get something like $30,000 just in commissions from selling 1 house. |
| 02:46.12 | harryvv | I was thinking that one of the obiosly companies to sell my ipphones and a asterisk, or ser box would be realestate offices. |
| 02:46.30 | ManxPower | harryvv: In order of my revenue: The Real Estate Company, Health Clinic, Law Offices. |
| 02:46.44 | harryvv | homes here have reached over the million dollars mark...because of the up and comming 2010 Olympics |
| 02:46.50 | sivana | we're doing well with insurance brokers |
| 02:47.16 | sivana | 10+ lines for the agents |
| 02:47.28 | ManxPower | harryvv: Real Estate agents have the technical know-how of a turnip. |
| 02:47.32 | harryvv | what do you guys charge per seat per month? |
| 02:47.35 | harryvv | hehehe |
| 02:48.02 | ManxPower | One of them told me they didn't want voicemail notification via a texst message to her cell phone because "figureing out how to read the messages is too hard" |
| 02:48.12 | sivana | funny |
| 02:48.13 | ManxPower | I didn't bother to offer the service to any other agents after that. |
| 02:48.42 | SwK_ | i love looking at patches and seeing shit like this "void pri_message(struct pri *pri, char *fmt, ...)" |
| 02:48.47 | SwK_ | wrong line |
| 02:49.07 | SwK_ | char *stuff |
| 02:49.10 | ManxPower | Oddly enough, the top producing agents either understand technology or have assistants that understand it (the second is the more common) |
| 02:49.45 | harryvv | manx, i have it sent to my email...then read it on my cell phone internet account to save the cost of making calls to see if i have vm |
| 02:50.08 | ManxPower | harryvv: Our agents would take months of training in order to be able to do that. |
| 02:50.14 | harryvv | heheh |
| 02:50.29 | ManxPower | I'm starting that "people skills" and "technical skills" are mutually exclusive. |
| 02:51.02 | harryvv | I figure out things quickly. I was a Sharp/Fiery conssultant and would whip though the push button menues quickly like it was some kind of vidio game to fix it :) |
| 02:52.41 | sivana | SwK_: ya, there's some nice code lines char *this, char *that |
| 02:53.07 | harryvv | manx, would like to get past this firewall issue with sip. I guess the cheapest way and still robust is ser? |
| 02:53.42 | dudes | harryvv - what type of firewall issue ... |
| 02:53.43 | ManxPower | harryvv: what firewall problem? |
| 02:53.59 | drray | I wonder if a bounty would help motivate people into getting the FXS ports working on those linksys routers under openWRT |
| 02:54.46 | harryvv | sip not passing though it. have port forwaring 10000-20000 5060 pointing to the asterisk box ip. But obviosly the nat will not work with it properly. |
| 02:54.47 | *** join/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg) |
| 02:55.05 | harryvv | rtp.conf is configured properly |
| 02:55.26 | dudes | harryvv - what type of router? |
| 02:55.28 | harryvv | sip.conf with externip and intern is set |
| 02:55.31 | harryvv | ipcop |
| 02:56.02 | harryvv | one guy here sugested putting asterisk on the firewall. that way its behind the nat. |
| 02:56.05 | ManxPower | harryvv: So your Asterisk server is behind NAT? Are the SIP clients behind NAT? Will the SIP clients be calling each other often or mostly SIP<->PSTN? |
| 02:56.22 | harryvv | yea, sip clients are on the remote end of the internet. |
| 02:56.48 | dudes | harryvv - what does sip do when you try to make a call? |
| 02:56.49 | harryvv | the idea..is to demo this ip500 at some office..with luck there firwall will not have issues but cannot bet on that. :) |
| 02:57.08 | dudes | harryvv - no sound? Or just no register? |
| 02:57.08 | harryvv | dudes typical one way call..thay hear me but i cannot hear them. |
| 02:57.14 | Nugget | Ring ring ring ring ring ring ring Banana phone |
| 02:57.18 | harryvv | yea |
| 02:57.20 | dudes | harryvv - do you have nat=yes ? |
| 02:57.22 | harryvv | yes |
| 02:57.30 | dudes | take it out if you have the port forwarded |
| 02:57.30 | harryvv | on both ata and ip500 in si[ |
| 02:57.35 | harryvv | sip.conf |
| 02:57.50 | ManxPower | harryvv: I do that sort of stuff with no need for SER |
| 02:57.54 | dudes | I had a guy with ipcop and he had the same issue ... took it out after reading sip debug (no nat avablible) fixed |
| 02:57.55 | harryvv | take out nat=yes? |
| 02:58.00 | dudes | yes |
| 02:58.04 | dudes | do it! |
| 02:58.05 | harryvv | okay |
| 02:58.08 | harryvv | hold on. |
| 02:58.08 | ManxPower | I set the SIP device to use ports starting at 16384 (default on Cisco). |
| 02:58.53 | ManxPower | Then I port forward ports 16384 - 16394 (small number of devices) and port 5060. Set up rtp.conf for the same port range. You can expand the range, of course. |
| 02:59.26 | ManxPower | then I use externip (NOT HOST NAME), and localnet in sip.conf and nat=yes in sip.conf for each sip stanza and pretty much NOTHING else special for NAT |
| 02:59.31 | harryvv | darn wife is on phone. |
| 02:59.43 | dudes | harryvv - If you forward the right ports you shouldn't have to set nat=yes ... at least from my experience. So forward port 5060 and 5036 <--- inbound sip port I think? |
| 03:00.06 | harryvv | why not 10000-20000 as required? |
| 03:00.21 | dudes | I forward fewer rtp's than that |
| 03:00.24 | ManxPower | You should NOT port forward on the NAT router that the SIP client is behind, no should you set any nat options on the SIP device. This will mess things up. |
| 03:00.42 | harryvv | i see |
| 03:01.10 | ManxPower | harryvv: At the time I standardized on 16384 on up we were going to use Ciscos, and they default to that. |
| 03:01.15 | dudes | harryvv - is VOIP fun (not) |
| 03:01.20 | Darwin35 | ok to dnd to the db at the cli its database put DUN dnd |
| 03:01.24 | Darwin35 | DND |
| 03:01.26 | harryvv | mmm mostly sip is a headache |
| 03:01.41 | harryvv | never has worked across a firewall for me. |
| 03:02.28 | harryvv | manx, well the idea is to use my ip500 to log into my asterisk box on somone elses connection. So about the rtp range |
| 03:03.02 | harryvv | ohh fun |
| 03:03.04 | harryvv | he leaves |
| 03:03.10 | Darwin35 | grrr |
| 03:03.38 | Qwell | harryvv: It should be reassuring that it his connection was reset. :p |
| 03:06.16 | opus_ | hi |
| 03:06.40 | Darwin35 | bed tiime |
| 03:07.39 | dudes | Anyone have much experience with redirect? |
| 03:09.53 | *** join/#asterisk jr352k_ (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 03:11.28 | *** join/#asterisk Barza (~galellope@205.240.200.117) |
| 03:11.31 | Barza | hi |
| 03:12.00 | jr352k_ | hello |
| 03:12.12 | Barza | maybe you can help me |
| 03:12.17 | jr352k_ | we got split? |
| 03:12.25 | jr352k_ | what's up? |
| 03:12.27 | Barza | i got a iaxy |
| 03:12.51 | Barza | my asterisk is in the internet and my iaxy in home |
| 03:13.17 | Barza | but for some razon is unable to connect to my asterisk |
| 03:13.32 | Qwell | Does your asterisk box have the IAX2 port open? |
| 03:13.39 | Barza | mmmmmmm |
| 03:13.45 | Barza | wich one? |
| 03:13.50 | Qwell | I forget |
| 03:14.14 | dudes | 4569 |
| 03:14.24 | Barza | the coriuse thing.... i probe this aix in my office and connect!! |
| 03:15.08 | Barza | my server is complete open wrigth now |
| 03:15.14 | Barza | just for the test |
| 03:16.27 | Barza | where i can post a configuration? |
| 03:16.51 | Barza | i provisioning my iaxy with to servers |
| 03:16.54 | Barza | one for lan |
| 03:17.03 | Barza | and one for wan |
| 03:17.48 | Barza | but i dont know if i have to use some special configuration in the asterisk |
| 03:17.57 | jr352k_ | http://www.pastebin.ca |
| 03:18.10 | Barza | thanks |
| 03:18.39 | jr352k_ | then giva us the link |
| 03:19.38 | dudes | with redirect ... lets say instead of using chan_agent (can I parse "show agents") then use the channel provided and redirect to that upon a answered call? |
| 03:20.12 | dudes | use chan_agents to log in and instead of sending to Agent/X send the to the provided channel that is. |
| 03:20.18 | Barza | <PROTECTED> |
| 03:22.54 | *** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net) |
| 03:23.06 | irv999 | does anyone do this in ny full time? |
| 03:23.17 | Barza | and this is my asterisk iax.conf http://pastebin.ca/18679 |
| 03:23.20 | jr352k_ | barza, how about the iax.conf? |
| 03:23.25 | dudes | do what in ny full time? |
| 03:24.07 | irv999 | program asterisk systems |
| 03:24.42 | NormAst | irv999: If you box is hooked up to the internet.. anyone can help you. |
| 03:24.55 | jr352k_ | true!!!! |
| 03:24.57 | Barza | jr352k_, i provisioning wright? |
| 03:26.22 | dudes | irv999 - my friend and I do asterisk work for folk all over the world |
| 03:26.28 | *** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM] |
| 03:26.28 | Barza | in my home i got a lynksys dsl router with all open |
| 03:26.45 | irv999 | dudes: need face time.. someone local.. |
| 03:27.00 | irv999 | dudes: I have programmers |
| 03:28.16 | sivana | irv999: might want to send a note out -biz list |
| 03:28.27 | Barza | jr352k_, do you see it? |
| 03:28.27 | NormAst | irv999: you could always use screen... |
| 03:28.42 | NormAst | Anyone know what happened to www.voipforcanada.com |
| 03:29.27 | sivana | nope |
| 03:29.29 | jr352k_ | <PROTECTED> |
| 03:30.10 | Barza | oki |
| 03:30.25 | irv999 | norm: I need someone who can sell / install / and program asterisk while paying me commisions I dont want to sell this anymore |
| 03:31.57 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 03:32.03 | blitzrage | NormAst: hey! |
| 03:32.21 | NormAst | ivr999: Everyone needs that... :) |
| 03:32.29 | NormAst | Hay blitzrage!! |
| 03:32.41 | blitzrage | NormAst: haven't seen you online in a while, figured you died or something :) |
| 03:33.04 | NormAst | I've been really busy... When the next Toronto asterisk meeting? Web site? |
| 03:33.14 | sivana | hey guys, is there anymore meet ups? |
| 03:33.19 | sivana | ya |
| 03:33.20 | sivana | hehe |
| 03:33.20 | blitzrage | NormAst: one tomorrow actually - http://www.taug.ca |
| 03:33.23 | *** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net) |
| 03:33.27 | blitzrage | NormAst: its at Tobies at 7:30 |
| 03:33.41 | irv999 | norm: to many issues and to many 3rd parties that I can't deal anymore |
| 03:34.03 | blitzrage | jbot: taug is The Toronto Asterisk Users group. The website can be found at http://www.taug.ca |
| 03:34.04 | jbot | okay, blitzrage |
| 03:34.13 | blitzrage | ~meetup |
| 03:34.38 | SarahEmm | taug tomorrow? |
| 03:34.40 | SarahEmm | crap! |
| 03:34.43 | SarahEmm | afternoon shift AGAIN! |
| 03:34.47 | SarahEmm | people need to move these meetings :( |
| 03:34.54 | SarahEmm | i wanna go to one but every time there is one i'm on afternoon shift :P |
| 03:35.14 | *** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM] |
| 03:35.17 | *** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM] |
| 03:35.20 | blitzrage | jbot: meetup is A formally wicked place to setup any sory of Meet Ups (usually Users Groups of some sort). It used to be free, but is now unfortunately $19/mth. See also, ~taug |
| 03:35.20 | jbot | blitzrage: okay |
| 03:35.49 | NormAst | Oh |
| 03:35.57 | blitzrage | SarahEmm: I won't be there because I'm going camping for the long weekend. We need to move it to the 3rd Thursday of each month instead of the 4th. Always hits a long weekend. |
| 03:36.03 | sivana | ~meetup |
| 03:36.03 | jbot | [meetup] A formally wicked place to setup any sory of Meet Ups (usually Users Groups of some sort). It used to be free, but is now unfortunately $19/mth. See also, ~taug |
| 03:36.32 | SarahEmm | formally wicked? |
| 03:36.57 | blitzrage | hrmmm... wicked as in good |
| 03:37.03 | SarahEmm | formally? |
| 03:37.08 | blitzrage | well, its pay now :) |
| 03:37.09 | NormAst | Blitzrage: $20.00 bucks.... |
| 03:37.10 | *** join/#asterisk iswm (iswm@iswm.user) |
| 03:37.17 | SarahEmm | but.. how does that make it formal? |
| 03:37.18 | blitzrage | a month |
| 03:37.25 | blitzrage | I suppose not formal :) |
| 03:37.32 | SarahEmm | hence my confuzzlingness |
| 03:37.39 | blitzrage | jbot: forget meetup |
| 03:37.39 | jbot | blitzrage: i forgot meetup |
| 03:37.44 | SarahEmm | lol |
| 03:38.22 | blitzrage | jbot: meetup is a place where people can create online groups to organize physical meetings with people. Its at http:/www.meetup.com. It used to be free, but is now $19/mth. See also, ~taug |
| 03:38.22 | jbot | blitzrage: okay |
| 03:38.27 | Barza | jr352k_, do you see it? |
| 03:39.08 | NormAst | blitzrange: That must be for the cost of the beer right? :) |
| 03:40.35 | blitzrage | NormAst: no no, the meetup site costs money (that's why we're not using it anymore) - there are still no "dues" for TAUG. |
| 03:40.49 | NormAst | Oh. |
| 03:41.09 | blitzrage | jbot: forget meetup |
| 03:41.09 | jbot | blitzrage: i forgot meetup |
| 03:41.14 | syle2 | 19 a month hmmm |
| 03:41.15 | NormAst | Directions on the website and locations would be nice. |
| 03:41.20 | blitzrage | jbot meetup is a place where people can create online groups to organize physical meetings with people. Its at http://www.meetup.com. It used to be free, but is now $19/mth. |
| 03:41.20 | jbot | okay, blitzrage |
| 03:41.25 | syle2 | how many people go and who gets the 19 a month :) |
| 03:41.27 | blitzrage | damn typo's :) |
| 03:41.52 | blitzrage | syle2: the people who run meetup.com - hence why we don't use it. Way too expensive just to manage some meetings. Costs nothing to go to TAUG |
| 03:42.44 | syle2 | makes sense |
| 03:43.20 | blitzrage | taug is not associated with meetup.comf |
| 03:43.23 | blitzrage | LOL |
| 03:43.46 | blitzrage | been typing .conf too much - fingers wanted the extra f |
| 03:43.56 | jontow | question.. i note that editline includes some vague form of vi-style (and emacs stle) command line editing; anyone played with that support on the CLI? |
| 03:43.57 | dudes | How can one bridge two active channels? |
| 03:44.01 | blitzrage | ~seen jerjer |
| 03:44.02 | jbot | jerjer <~JerJer@jerjer.bronze.supporter.pdpc> was last seen on IRC in channel #asterisk, 11h 27m ago, saying: 'Blackthorn: registering to the proxy is how you get thru nat/firewall'. |
| 03:44.16 | JunK-Y | dudes: changrab ? |
| 03:44.35 | blitzrage | jontow: !vim /etc/asterisk/extensions.conf :) |
| 03:44.59 | blitzrage | jontow: that must be why some commands accept the ability to filter with regex's |
| 03:45.04 | jontow | *sigh*.. no ;) |
| 03:45.16 | blitzrage | jontow: hrmmm, twas a guess :) |
| 03:45.34 | jontow | i mean, at a shell (ksh or zsh for instance); you can type 'set -o vi' and it'll throw you into a permanent (for that session) 'vi' style line-editing syntax |
| 03:45.41 | blitzrage | JunK-Y: show dialplan <context> like <regex> <-- that'd be nice |
| 03:46.31 | jontow | ie. if you hit escape while typing a command at the shell prompt, you'll have the ability to use 'w' and 'b' to move forward or back a word (WITHOUT ERASING THEM;)) or 'x' to delete the current character, or whatever standard vi commands you want to use, including 'i', and 'a' to get back to insert mode |
| 03:46.35 | jontow | all without leaving the CLI ever |
| 03:47.01 | jontow | im not talking about editing files.. im talking about editing lines :) |
| 03:47.28 | blitzrage | jontow: that sounds like crazy talk! |
| 03:47.30 | JunK-Y | blitzrage: pay me a beer and maybe ;) |
| 03:47.32 | blitzrage | :) |
| 03:47.38 | blitzrage | JunK-Y: done! |
| 03:47.48 | blitzrage | JunK-Y: heck, I'll give you two :) |
| 03:47.49 | jontow | well.. the support is there; but i've yet to figure out how to turn it on.. :) |
| 03:47.53 | dudes | JunK-Y - that's too new =( |
| 03:47.57 | blitzrage | jontow: innnnnnnteresting :) |
| 03:48.10 | blitzrage | jontow: let me know if you figure it out |
| 03:53.06 | jontow | will do.. no idea if its built out at all in asterisk.. but the library sitting there in our source tree has the capability :) |
| 03:53.48 | nounoursfr | I have a question about my waiting queue:when I try to use the hold touch on my cisco 7960 |
| 03:54.13 | nounoursfr | , my correspondant doesn't heard my hold music |
| 03:54.51 | nounoursfr | Has someone an idea ? |
| 03:55.19 | bkw_ | ok its offical IAX2 has a major bug that effects audio |
| 03:55.35 | bkw_ | guess we'll try to fix that one tommorow |
| 03:55.57 | JunK-Y | bkw_: more details baby? |
| 03:55.58 | Qwell | bkw_: mind a really quick PM? got two issues/questions for ya |
| 03:56.18 | bkw_ | Qwell, shoot |
| 03:56.26 | bkw_ | IAX with or without jitter buffer. |
| 03:56.33 | blitzrage | bkw_: doesn't matter? |
| 03:56.37 | bkw_ | you'll get blocks of audio missing |
| 03:56.46 | blitzrage | bkw_: let me know about testing tomorrow - I'll help. |
| 03:57.55 | blitzrage | bkw_: patches, etc.. |
| 03:57.55 | bkw_ | i'm going to be collecting info tonight |
| 03:57.55 | dudes | nounoursfr - lookup music on hold mpg123 ... or re encode your on hold music as native asterisk format |
| 03:57.55 | bkw_ | to see if we can figure a way to attack this and fix it |
| 03:57.55 | blitzrage | bkw_: let me know if there is anything I can do to help |
| 03:57.55 | bkw_ | blitzrage, will do |
| 03:57.55 | blitzrage | bkw_: I've got like 3 boxes I can use |
| 03:57.55 | JunK-Y | we means ? |
| 03:57.55 | blitzrage | bkw_: actually, 4 now that I think about it. |
| 03:57.55 | bkw_ | JunK-Y, tony and I |
| 03:58.28 | JunK-Y | k, i'll join the conf tomorrow on my lunch break to get more info on all that. |
| 03:58.29 | blitzrage | bkw_: QC1 (final draft) comes Aug. 1st. |
| 03:59.18 | nounoursfr | dudes-My music doesn't have any problem to play during the begin of the waiting queue : the problem is just after, when I put my correspondant on hold |
| 04:00.18 | dudes | nounoursfr - if you have a asterisk version after May they are know to have audio issues (not sure if MOH/Queue is effected or not though.) |
| 04:00.22 | blitzrage | ok... who has gotten arguments passed from AGI() to a PHP script? |
| 04:00.32 | blitzrage | on stable (1.0.9) - works on HEAD, not on stable |
| 04:00.36 | syle2 | this pap2-na seems to loose connection to the server after while all the time, when dude resets his router it seems to work again, my question is is this because the ip address could be changing , and asterisk is not detecting it? |
| 04:02.15 | blitzrage | nevermind... its just that one box |
| 04:02.34 | JunK-Y | god, im listening nirvana. i know why i was listening that shit 10 years ago ! |
| 04:02.48 | blitzrage | JunK-Y: its still good |
| 04:03.02 | JunK-Y | ya. |
| 04:03.44 | NormAst | syle2: Try adding qualify=5000 to the config. |
| 04:03.53 | dudes | JunK-Y - why does app_queue and chan_agents suck so bad |
| 04:03.54 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-6719.nb.aliant.net) |
| 04:03.55 | nounoursfr | dudes - Do you know if I have to compile my audio drivers with a 1.0.6-BRIstuffed |
| 04:05.01 | syle2 | i have it set to just yes right now whatever that value is |
| 04:05.22 | JunK-Y | dudes: dunno man. |
| 04:05.30 | NormAst | syle2: yes is 2000ms |
| 04:05.32 | JunK-Y | why my grand-ma is so old? |
| 04:05.42 | dudes | Cause she had your mama |
| 04:05.45 | dudes | hehe |
| 04:05.55 | syle2 | k i'll try that thx, let you know tommorrow :) |
| 04:06.13 | dudes | nounoursfr - I'm not sure |
| 04:08.00 | dudes | Is there anyway to bridge two active channels ... being instead of sending a call to Agent/X ... how could one just send it to their channel? |
| 04:08.21 | JunK-Y | dudes: see app_changrab.c |
| 04:08.25 | dudes | that won't work |
| 04:08.44 | JunK-Y | ? |
| 04:09.25 | dudes | When we try to use redirect it logs the agent out when we send it to their channel (i.e SIP/talkingsip-5ca8) |
| 04:10.49 | bkw_ | blitzrage, killer |
| 04:10.52 | *** join/#asterisk zippp (~zip@63.98.170.221) |
| 04:11.23 | dudes | so sip/talkingsip-xxxx (Agent/X) and the called party sip/talkingsip-5ca8 ... when sent to sip/talkingsip-xxxx (instead of Agent/X) it logs the agent out and drops the call. |
| 04:12.28 | dudes | Lets say we want to just bridge those two channels together instead of having to Redirect to Agent/X (you can send to the channel) |
| 04:13.53 | *** join/#asterisk Paskifr (~Paskifr@stardust.noc.frontier.fr) |
| 04:14.08 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 04:15.32 | sivana | I'm an idiot... I partition a win2k primary partition to 1.5 GB instead of 15 GB |
| 04:15.47 | dudes | WTG stupid =p |
| 04:15.48 | sivana | which you could resize it on the fly |
| 04:15.52 | sivana | hehe |
| 04:16.06 | *** join/#asterisk scud (~scud@12-214-190-139.client.mchsi.com) |
| 04:16.33 | sivana | I go home to continue setting up the box with RDC and can't do shit |
| 04:16.47 | sivana | and I don't feel like driving back to the office and restart the install... hehe |
| 04:17.45 | scud | i ported asterisk to java and have it running on my toaster |
| 04:19.38 | dudes | Asterisk from C to Java ... fun |
| 04:19.55 | *** join/#asterisk yaaar (~chatzilla@12-216-231-118.client.mchsi.com) |
| 04:20.01 | yaaar | word |
| 04:20.21 | drumkilla | we are, actually, rewriting the dsp code in java. |
| 04:25.07 | *** join/#asterisk gushi (danm@prime.gushi.org) |
| 04:25.25 | gushi | Hey all, can someone help me with a stupid asterisk problem? I am pretty sure I've done something wrong. |
| 04:25.47 | scud | gushi: remember just to ask instead of asking to ask. |
| 04:26.03 | gushi | I am using AMP and asterisk is telling me "returned from dialparties with no extensions to call" |
| 04:26.12 | gushi | I've got both extensions configured and registered. |
| 04:26.21 | gushi | when I try to call from 200->201 |
| 04:28.11 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 04:29.01 | FuriousGeorge | just wondering if anyone has employed a presence server to go along w/ their * box |
| 04:31.00 | FuriousGeorge | could it be done with say, jabber? |
| 04:31.39 | yaaar | that's kind of a neat idea George... |
| 04:31.52 | FuriousGeorge | jabber? |
| 04:31.57 | yaaar | i guess it probably could... |
| 04:32.12 | yaaar | jabber can push most any kind of data really |
| 04:33.04 | FuriousGeorge | how, um, "esoteric" would it be for me to go and lear to run a jabber server? got any experience with that |
| 04:33.11 | yaaar | nope |
| 04:33.11 | FuriousGeorge | learn* |
| 04:33.24 | FuriousGeorge | jabber.org it is, for now |
| 04:37.36 | FuriousGeorge | is there a commercial or otherwise sip/iax client with any IM support |
| 04:38.00 | FuriousGeorge | i know eyebeam supports "presence" but i dont know of any popular presence servers for linux |
| 04:41.19 | *** join/#asterisk Cresl1n (~Cresl1n@24.96.136.141) |
| 04:41.29 | *** join/#asterisk PMantis (~pmantis@cpe-69-204-25-153.rochester.res.rr.com) |
| 04:41.52 | PMantis | Does anyone know of a way to check for dialed digits in a macro? |
| 04:41.53 | Qwell | FuriousGeorge: There is phonegaim. Not sure if that does what you want |
| 04:42.19 | Cresl1n | hey |
| 04:42.22 | Cresl1n | I'm here finally |
| 04:42.27 | Cresl1n | how's the party going? |
| 04:42.43 | Qwell | Cresl1n: just getting started now |
| 04:42.54 | *** join/#asterisk Inv_arp (junya@adsl-156-144-105.mia.bellsouth.net) |
| 04:43.42 | FuriousGeorge | Qwell: i know eyebeam supports a presence server, and i hear jabber can be that, but i dont know if messages would get from client a to client b, or if * must support it |
| 04:43.48 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 04:43.54 | PMantis | I'm expanding "macro-stdexten", after a s-BUSY status, I want to prompt caller for input. However, * looks for exten match in a prior context, not the macro. |
| 04:43.55 | file[laptop] | asterisk needs to support it, and it will very very soon |
| 04:44.15 | FuriousGeorge | file[laptop]: did it get picked up by the google summer of code thing |
| 04:45.10 | dudes | file - Do you know much about the manager interface : Redirect? |
| 04:45.20 | file[laptop] | FuriousGeorge: yes, I'm doing it |
| 04:45.37 | file[laptop] | dudes: I'm guessing it redirects the channel to a new extension and context |
| 04:45.40 | FuriousGeorge | how cool, i think i remember asking you a few months back and you were already working on it |
| 04:45.55 | file[laptop] | FuriousGeorge: yeah, but thanks to SoC it'll probably get accepted into CVS |
| 04:46.44 | PMantis | What's this Jabber thing do? |
| 04:47.30 | FuriousGeorge | file[laptop]: congrats again, and thanks a bunch |
| 04:47.35 | dudes | file - Yes. Mainly I was wondering if you might have a general idea why when we redirect a call to Agent/X channel instead of using Agent/X it hangs up both channels |
| 04:48.39 | mishehu | bah. |
| 04:48.52 | FuriousGeorge | PMantis: it does a bunch of IM oriented stuff, check out www.jabber.org |
| 04:49.01 | file[laptop] | dudes: I'd have to look at it |
| 04:49.33 | dudes | When we send calls to Agent/X it deadlocks |
| 04:49.40 | *** join/#asterisk zoo (nobody@ip-36-16.travedsl.de) |
| 04:50.57 | PMantis | FuriousGeorge, Heh, I run a jabber server. I'm wondering what this Jabber + Asterisk thing does. |
| 04:51.19 | FuriousGeorge | there is no such thing as far as i can tell |
| 04:51.49 | FuriousGeorge | i know eyebeam supports presence but i dont know of any sip client that does |
| 04:52.18 | FuriousGeorge | i mean: ...but i dont know of any sip/iax client that supports jabber |
| 05:01.14 | yaaar | so, anybody using realtime around here, and can answer a quick question about the table setups? the ones in the wiki for sip and iax seem to be setup oddly different to me, and i was wondering why? |
| 05:01.17 | *** part/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 05:01.39 | FuriousGeorge | does gizmo allow for simultaneous connection to multiple networks? i could connect to asterisk with one and have all the users log in with their own gizmo account to IM eachother |
| 05:01.42 | *** join/#asterisk IgorG (~gia@195.162.32.126) |
| 05:02.00 | FuriousGeorge | nice temporary solution |
| 05:02.19 | yaaar | the iax one has CREATE UNIQUE INDEX iax_buddies_username_idx ON iax_buddies(username); at the end, but then says something about why it's indexed by 'name' (not 'username') |
| 05:03.40 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:03.54 | FuriousGeorge | yaaar: what you mean the iax one? |
| 05:04.39 | FuriousGeorge | the one that's forthcomming? |
| 05:04.44 | yaaar | http://voip-info.org/tiki-index.php?page=Asterisk+RealTime+IAX |
| 05:06.39 | yaaar | also, the wiki talks about family iaxfriends, and if you set that up the server complains that it's deprecated in favor of iaxpeers and iaxusers (like sip is) |
| 05:06.54 | gushi | Hrmmm, I guess nobody knows. |
| 05:07.56 | yaaar | gushi: if they do, they're keeping mum |
| 05:09.10 | FuriousGeorge | yaaar: i'm having trouble understanding what this "RealTime" thing is. as far as i can tell it is a way to store sip users in a database? |
| 05:09.20 | yaaar | yep |
| 05:09.47 | FuriousGeorge | i was wondering about the gizmo account for the presence and IM it might could provide |
| 05:09.49 | yaaar | more than that, it'll also store your iax users, queues, dialplan, and voicemail configs |
| 05:10.08 | FuriousGeorge | yaaar: ... does it help with billing? |
| 05:10.25 | FuriousGeorge | keeping track of who calls where and for how long? |
| 05:10.36 | *** part/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 05:10.42 | Qwell | FuriousGeorge: cdr |
| 05:10.43 | yaaar | no, that's seperate support....the call detail records database support is done in cdr_mysql.conf |
| 05:10.50 | yaaar | or cdr_postgres |
| 05:10.52 | yaaar | or whatever |
| 05:10.54 | Qwell | can be stored in multiple places |
| 05:11.10 | FuriousGeorge | gotcha, i was familiar with cdr, but this is an unrelated database |
| 05:11.39 | yaaar | i'm using both cdr_mysql and res_config_mysql, so i hvaeboth call detail and config db's |
| 05:12.05 | Qwell | yaaar: odbc_voicemail is nice too |
| 05:12.16 | Qwell | odbcstorage actually |
| 05:12.26 | gushi | Is asterisk realtime also distributed separately because of the mysql licensing issues? |
| 05:12.33 | yaaar | does it do more than the voicemail functionality provided by realtime? |
| 05:12.47 | Qwell | yaaar: it stores your VMs in a db |
| 05:12.54 | yaaar | gushi: i dunno, but i got it from cvs under asterisk-addons |
| 05:12.55 | Qwell | instead of files |
| 05:13.02 | gushi | so then, yes |
| 05:13.19 | yaaar | Qwell: oh....hmmm. maybe i just don't see it....why do i want that? |
| 05:13.32 | fearnor | database = scalabol |
| 05:13.39 | fearnor | shareabol |
| 05:13.40 | Qwell | yaaar: makes it easy to have a seperate VM server. quite a few reasons, really |
| 05:13.50 | fearnor | etc. |
| 05:13.52 | yaaar | i see |
| 05:14.07 | Qwell | fearnor: exactly...minus the spelling |
| 05:14.19 | fearnor | of course, i wish half of asstricks code was rewritten as agi-ish stuff |
| 05:14.36 | fearnor | there's no reason app_voicemail has to be C app. |
| 05:14.38 | yaaar | so that database would also be distinct frmo the config, right? and so would sort of coexist with the realtime deal? |
| 05:14.48 | Qwell | yaaar: right |
| 05:15.09 | fearnor | for that matter, app_dial does a lot of things that really dont belong there |
| 05:15.18 | yaaar | well, i like everything going into a database....good place to store stuff. |
| 05:15.30 | fearnor | it would be so cool if we could just use a special language to connect channels like pipes |
| 05:16.22 | Cresl1n | fearnor: it's called C |
| 05:16.23 | Cresl1n | :-) |
| 05:16.25 | yaaar | you know what i'd like? if i could have multiple families of extensions....so i could have multiple customers with the same extension ranges than couldn't call each other and that i could route seperately |
| 05:16.35 | fearnor | cres: too many lower-level things to deal with |
| 05:16.50 | Cresl1n | not that bad |
| 05:16.51 | Cresl1n | :-) |
| 05:17.07 | fearnor | in words of larry wall "don't get me wrong, c is a nice language" :) |
| 05:17.21 | fearnor | yaar: welcome to [contexts] |
| 05:17.23 | fearnor | ktnx |
| 05:17.53 | *** part/#asterisk Cresl1n (~Cresl1n@24.96.136.141) |
| 05:18.00 | Qwell | yaaar: you could, heh |
| 05:18.31 | yaaar | fearnor: have i missed something? i didn't think i could duplicate my extension numbers, even in different contexts?! |
| 05:18.40 | fearnor | sure you can |
| 05:18.59 | yaaar | man i've gone through some weird gyrations over this.... |
| 05:19.21 | *** join/#asterisk newmember (user@S010600036d1139fb.cg.shawcable.net) |
| 05:19.43 | DarthClue | yaaar: whatever gave you the idea that you couldn't duplicate them? |
| 05:20.01 | yaaar | uh.... |
| 05:20.52 | DarthClue | i want to know who i need to kill for spreading that kind of mis-information. |
| 05:21.03 | fearnor | self-inflicted wound? ;) |
| 05:21.20 | yaaar | yeah i think so. probably just me horribly misreading something |
| 05:21.23 | Qwell | DarthClue: it was file. I totally caught him spreading mis-information the other day. :p |
| 05:21.25 | gushi | yaaar: If you want to do that, just run different asterisks on multiple ips |
| 05:21.30 | gushi | That's what we're doing. |
| 05:21.41 | yaaar | on the same hardware? |
| 05:21.45 | gushi | Yu |
| 05:21.47 | gushi | er yup |
| 05:21.48 | yaaar | why |
| 05:22.06 | fearnor | you'll run into some faggotry with sip if you do that and are not careful with externip= etc |
| 05:22.29 | Qwell | Why run two instances? Thats just silly |
| 05:22.57 | gushi | Qwuell, so everything can be configured totally separately. |
| 05:23.08 | Qwell | gushi: You could do that with one instance. |
| 05:24.08 | gushi | I tend to think it's easier to see each as its own |
| 05:24.15 | Qwell | waste of resources... |
| 05:24.39 | gushi | qwell: thanks for your opinion. |
| 05:25.10 | gushi | I realize two asterisk instances use more resources than one. |
| 05:25.47 | gushi | But it's far easier to simplify customers wanting to use their own voicemail, their own (possibly colliding) extensions, their own routing, their own providers, etc etc. |
| 05:26.05 | fearnor | you'll step on your own toes with sip if you aren't careufl |
| 05:26.06 | gushi | than worry about which line in which config file belongs to which customer. |
| 05:26.08 | Qwell | could still do all of that with one instance. however... |
| 05:26.18 | Qwell | you really should use a seperate box if the customers are large enough |
| 05:26.39 | gushi | They're not. |
| 05:26.52 | DarthClue | gushi: what company do you work for? i just want to know who to avoid should i ever leave The Empire. |
| 05:27.11 | yaaar | Qwell: but seperate boxes will make it tricky to oversubscribe al these PRIs! |
| 05:27.19 | yaaar | heheheh |
| 05:27.29 | *** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net) |
| 05:27.39 | gushi | who I work for has little bearing on it. |
| 05:27.40 | Qwell | yaaar: two instances can't share a single pri |
| 05:27.44 | gushi | I personally use ICH. |
| 05:27.48 | irv999 | has anyone had any problems with polycom phones not being able to set the time properly? |
| 05:27.51 | yaaar | Qwell: oh. |
| 05:28.02 | irv999 | i.e. not contacting NTP |
| 05:28.03 | Qwell | yaaar: at least, I'd hope they couldn't |
| 05:28.10 | fearnor | if you are terminating PRI on asterisk, i think you have already lost ;) |
| 05:28.14 | yaaar | hmm, yeah guess not |
| 05:28.19 | fearnor | just my opinion. |
| 05:28.24 | yaaar | fearnor: why? |
| 05:28.40 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 05:28.40 | *** mode/#asterisk [+o kram] by ChanServ |
| 05:28.51 | fearnor | i prefer to terminate it on proper-ish gear, convert to SIP, deal with it on asterisk |
| 05:28.51 | gushi | Does asterisk have a BRI termination module? |
| 05:28.57 | fearnor | gushi: bristuff |
| 05:29.17 | gushi | I mean hardware wise |
| 05:29.24 | gushi | or do you just take it in as two pots lines |
| 05:29.31 | yaaar | fearnor: what do you consider proper-ish? |
| 05:29.33 | fearnor | gushi: AVM fritz stuff |
| 05:29.38 | fearnor | yaaar: 5300 or TNT |
| 05:29.51 | gushi | but nothing digium |
| 05:30.22 | yaaar | fearnor: so this as/5300 i've got sitting in the rack in the datacenter (as a modem pool plugged into 7 pri's) can translate calls from the pri to sip?!?! |
| 05:30.37 | fearnor | i've had lots of wierd issues with timing on digium cards back in the day |
| 05:30.50 | *** join/#asterisk xeet2 (~xeet3@gw1.istx.net) |
| 05:30.55 | fearnor | basically, when you are running at full capacity, it was slipping frames, breaking faxes. |
| 05:31.07 | fearnor | yaar: if you have proper cards on it |
| 05:31.08 | irv999 | digium cards with POTS are really flaky |
| 05:31.09 | gushi | On the other side of the coin, is it possible to terminate modem calls on asterisk. |
| 05:31.18 | fearnor | gushi: no. |
| 05:31.26 | fearnor | gushi: there's no softmodem stack :( |
| 05:31.41 | fearnor | i've offered 20k$ bounty few years ago for full v.90 modem stack |
| 05:31.53 | gushi | That would be so beautiful |
| 05:31.58 | gushi | Prolly eat cpu like mad |
| 05:31.59 | yaaar | can the as/5300 tell the difference? Like, take modem calls to its own processing and voice calls send sip to *? |
| 05:32.04 | gushi | without good DSPs |
| 05:32.09 | xeet2 | can anyone tell me how I might go about grabbing a specific file revision from cvs? |
| 05:32.12 | fearnor | yes, it can do based on DNIS |
| 05:32.28 | yaaar | wow |
| 05:32.38 | irv999 | I have a digium t1 card right now and I think it is dropping calls but I can't tell if it is hardware or software.. |
| 05:32.54 | yaaar | although, it really seems it would add a lot of complexity to by setup... |
| 05:32.56 | fearnor | well, dropped calls are probably your software ;) |
| 05:32.57 | xeet2 | not the entire cvs head, just an older version of a specific file |
| 05:33.02 | essobi_ | hey irv999 |
| 05:33.10 | gushi | based on DNIS it can route the calls elsewhere, I assume |
| 05:33.12 | fearnor | xeet: cvs update -D"5 days ago" filename.c |
| 05:33.18 | gushi | Not answer and make modemy noises. |
| 05:33.22 | xeet2 | oh really, hmmm |
| 05:33.23 | xeet2 | thanks |
| 05:33.30 | fearnor | gush: you talking asterisk or cisco? |
| 05:33.36 | gushi | asterisk |
| 05:33.48 | fearnor | yeah. |
| 05:33.51 | gushi | I'd love to be able to hybridize a pri |
| 05:34.04 | gushi | So I also have a place to give myself 800 number access. |
| 05:34.05 | Qwell | gushi: I'd love to be able to get a PRI for testing purposes |
| 05:34.09 | gushi | into the office grid |
| 05:34.12 | yaaar | DarthClue: no kidding. the heat finally broke here (MO) ...was 105F with heat index to 118 a couple days back |
| 05:34.18 | gushi | Instead of paying the assrape hotels charge |
| 05:34.19 | brenda | DarthClue: you're a silly boy |
| 05:34.36 | fearnor | gushi: you can send it to analog modem via FXS line |
| 05:34.38 | gushi | or using my laughable GPRS connection |
| 05:34.40 | fearnor | that'd be funny :) |
| 05:35.06 | gushi | Now there's an option I hadn't considered. |
| 05:35.13 | DarthClue | yaaar: was 104 two days ago with a low of 81, hi was 85 today and it's down to 64 right now. i've got the window open and the attic fan running to make the house nice and cold for alot less than running the ac. |
| 05:35.14 | gushi | And I have an extra FXS card, too. |
| 05:35.37 | gushi | Might look a little silly to have an rj11 wire looping from one card to the other. |
| 05:35.49 | gushi | three cards to accept an analog call :) |
| 05:35.52 | DarthClue | brenda: if you managed to register for the thing that shall not be named, you will discover that i am really quite serious...sometimes. |
| 05:35.54 | citats | DarthClue: best weather here today than we've had in a long time |
| 05:35.57 | xeet2 | Qwell: where are you located? |
| 05:35.59 | yaaar | gushi: sometimes we all have to look silly in the name of progress |
| 05:36.10 | xeet2 | qwest is selling ld pri's for $120/mo |
| 05:36.12 | fearnor | it'd certainly qualify as ghetto fabulous |
| 05:36.16 | otmar | talking about weather: the heat wave is coming to Vienna/Austria now. |
| 05:36.23 | xeet2 | and that includes the local loop |
| 05:36.24 | fearnor | xeet: that's just the port. they make money on LD. |
| 05:36.30 | xeet2 | right |
| 05:36.36 | xeet2 | but its still cheap for a circuit =) |
| 05:36.38 | yaaar | xeet2: what's a local loop cost there? |
| 05:36.51 | xeet2 | normally about 4-500 in maryland |
| 05:36.58 | yaaar | xeet2: that port fee is on top of the loop, right? |
| 05:37.06 | fearnor | i doubt it they'll sell you with the loop for 120. |
| 05:37.07 | gushi | heh, I should take them up on that for my ski house in upstate NY |
| 05:37.07 | xeet2 | yaar: no, thats including the local loop |
| 05:37.12 | yaaar | 4-500!!!? per SAL? or for 2 legs? |
| 05:37.13 | xeet2 | fearnor: we have 4 |
| 05:37.13 | fearnor | i think they mean 120 at the qwest colo |
| 05:37.16 | citats | qwest prolly pays about 34 bucks for the loop to most COs |
| 05:37.19 | gushi | the telco there is so dated we had a party line 5 years ago |
| 05:37.19 | fearnor | not with the loop. |
| 05:37.23 | xeet2 | yes, with the loop |
| 05:37.33 | fearnor | as in, not with the 'type 2' loop. |
| 05:37.42 | fearnor | you may be in a qwest lit building |
| 05:37.49 | fearnor | and type 1 loop |
| 05:37.51 | gushi | it's not the loop TO the co, it's FROM the CO to your place that gets expensive |
| 05:37.52 | xeet2 | they have something like 5k t1 ports they *have* to sell by the end of the year |
| 05:38.04 | xeet2 | fearnor: our present location is not |
| 05:38.27 | yaaar | xeet2: it doesn't matter how many they have. they can't sell a t1 local loop at lower than the tarriff rate |
| 05:38.29 | xeet2 | if you want our sales guy's contact number I'll give it to you, I promise you thats including local loop, in a non-qwest-lit building |
| 05:38.42 | fearnor | that doesn't make sense. |
| 05:38.44 | xeet2 | they can if its ld only |
| 05:38.45 | *** join/#asterisk clive- (~pirch@rndf-146-30-75.telkomadsl.co.za) |
| 05:39.14 | yaaar | xeet2: no, that just means it's covered by the fcc or ctoc tarriff, instead of your state's psc tarriff |
| 05:39.30 | xeet2 | well, right, its a different tarriff |
| 05:39.41 | yaaar | anyway what do they charge for the ld on it? |
| 05:39.50 | fearnor | 2c-ish ;) |
| 05:39.54 | fearnor | but its quality termination. |
| 05:40.00 | yaaar | yeah |
| 05:40.05 | xeet2 | depends on what your commit is, under 50k minutes yeah its about 2c |
| 05:40.20 | fearnor | q is refusing to play ball on larger commits |
| 05:40.21 | xeet2 | over 1 mil they start doing per-lata rates |
| 05:40.32 | fearnor | one of my homies is saying q raised prices on him about 25% |
| 05:40.40 | fearnor | on a few mou commit |
| 05:40.43 | fearnor | er few mil mou |
| 05:41.04 | yaaar | damn sucka |
| 05:41.23 | yaaar | that's a dent in somebody's annual earning's report... |
| 05:41.35 | xeet2 | fearnor: even when his usage didn't go down? |
| 05:41.45 | fearnor | correct |
| 05:41.48 | xeet2 | mmm |
| 05:41.59 | xeet2 | I'm assuming he had per-lata rates? |
| 05:42.08 | fearnor | he had flat rate |
| 05:42.13 | fearnor | 1.3 or smth |
| 05:42.18 | fearnor | cause of his commit. |
| 05:42.18 | xeet2 | ah |
| 05:42.24 | fearnor | now its 1.6 ish |
| 05:42.30 | fearnor | anyways. |
| 05:43.07 | fearnor | i was kinda wonderink what do big voip providers use for inbound |
| 05:43.12 | xeet2 | regardless its still a good rate |
| 05:43.20 | xeet2 | fearnor: as in what clecs? |
| 05:43.22 | fearnor | i was of impression everyone is essentially a (3) customer |
| 05:43.33 | fearnor | or customer of a customer of an aggregator of a reseller |
| 05:43.35 | xeet2 | paetec is really big on that in this area |
| 05:43.43 | fearnor | for nationwide dids. |
| 05:43.52 | xeet2 | they have a ton of * servers colocated at their co in dc |
| 05:44.04 | fearnor | well |
| 05:44.18 | xeet2 | who they belong to, I haven't a clue |
| 05:44.23 | yaaar | fearnor: yeah it's really weird....seems like this whole game is an excercise in paying somebody to pay somebody to pay something and shaving your tenth of a cent off the top while you're in there |
| 05:44.29 | fearnor | every clec other than gx and l3 doesn't cover enough of country to give nationwide dids |
| 05:44.40 | xeet2 | right |
| 05:44.47 | xeet2 | and l3 just raised their minimum to 25k |
| 05:44.55 | fearnor | it was 10k, right? |
| 05:45.06 | xeet2 | yes |
| 05:45.11 | fearnor | fuck l3. i think they'll be fucked company soon. |
| 05:45.16 | xeet2 | agreed |
| 05:45.22 | xeet2 | they're being too picky with their customers |
| 05:45.26 | fearnor | they created the aggregator business and i think it'll bite them in the ass soon. |
| 05:45.41 | fearnor | when you have fewer large customers they have lots more power over your pricing and future. |
| 05:45.50 | fearnor | just my opinion. |
| 05:46.14 | twisted | xeet2, NO SHIT they're being too picky |
| 05:46.23 | fearnor | xoxo would do about ds3's worth of nationwide DIDs for ~15k$ |
| 05:46.24 | yaaar | so, is anybody up right now who's familiar with the realtime tables? |
| 05:46.29 | *** join/#asterisk grimse (~grimse@p5481E6E1.dip.t-dialin.net) |
| 05:46.42 | xeet2 | fearnor: yeah, but xo is horrible when it comes to number porting |
| 05:46.44 | DarthClue | yaaar: you need to be more specific there. |
| 05:46.47 | fearnor | i was wonderink if anyone wants to go in for that. |
| 05:46.55 | fearnor | xeet: as in, they won't port the numbers anymore. :) |
| 05:46.57 | yaaar | uh right....one sec, will get more specific |
| 05:47.03 | fearnor | cause freaking vonage abused xo to hell |
| 05:47.11 | xeet2 | vonage used xo? hehe |
| 05:47.14 | fearnor | basically turning XO into their outsourced LNP/ASR center. |
| 05:47.18 | fearnor | yes in certain latas |
| 05:47.29 | xeet2 | we're in the process of moving from xo to paetec for alot of our stuff |
| 05:47.39 | fearnor | and xo said 'we won't port numbers no mo' |
| 05:47.41 | brenda | DarthClue: we're no longer naming it? |
| 05:47.43 | xeet2 | they screwed up so many lnp requests we just got fed up |
| 05:48.02 | fearnor | xeet: strange. i've had about 10 lnp requests and they all went fine. |
| 05:48.13 | fearnor | anyway, i think at certain point, you gotta do your own LNP |
| 05:48.14 | xeet2 | what ilec? |
| 05:48.22 | fearnor | from vz to xoxo |
| 05:48.26 | xeet2 | < verizon MD |
| 05:48.33 | fearnor | vz ny |
| 05:48.36 | xeet2 | they had numbers down for weeks |
| 05:48.42 | DarthClue | brenda: i have no need to name it, everyone already knows what it is, and at this point, i'm just responding to inquiries here. i have other places to spam that don't result in interplanetary warfare. |
| 05:48.46 | fearnor | hadn't had a problem. |
| 05:48.47 | xeet2 | we finally just complained to one of their vps to get it fixed |
| 05:49.10 | xeet2 | yaar: what did you need to know about realtime now? |
| 05:49.11 | brenda | DarthClue: jerjer isn't here |
| 05:49.19 | fearnor | i have love/hate relationship with xo. |
| 05:49.20 | fearnor | heh |
| 05:49.24 | DarthClue | brenda: i know. |
| 05:49.33 | yaaar | i'm curious as to why the (iax peers/users) table shown in http://voip-info.org/tiki-index.php?page=Asterisk+RealTime+IAX is so differently setup than the (sip) one at http://voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip |
| 05:49.34 | brenda | fearnor: how can you love xo???!?!?! |
| 05:49.35 | xeet2 | fearnor: ditto, mostly hate lately |
| 05:49.43 | fearnor | mostly love lately for me |
| 05:49.49 | fearnor | the big thing is to find non-sucky sales rep |
| 05:49.49 | brenda | you lie! |
| 05:49.51 | *** join/#asterisk jayk- (jayk@vapid.reprehensible.net) |
| 05:49.53 | fearnor | i love my current guy |
| 05:49.59 | fearnor | he gets shit DONE and doesn't lie to me |
| 05:50.05 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:50.09 | xeet2 | seems to be a common issue |
| 05:50.11 | twisted | fearnor, sounds like a good man |
| 05:50.12 | yaaar | in particular, the line at the end about indexing 'username' seems odd to me, especially in light of the note afterward talking about how it's indexed by 'name' |
| 05:50.13 | fearnor | like "eh, i suggest you dont order this shit. it wont work" |
| 05:50.14 | xeet2 | our last rep said he would open tickets |
| 05:50.16 | twisted | fearnor, you should marry him |
| 05:50.17 | fearnor | highly recommended. |
| 05:50.21 | xeet2 | and do all kinds of things |
| 05:50.23 | xeet2 | and never did them |
| 05:50.24 | jayk- | when i get an outside call, asterisk reports the caller id as "From asterisk". Is there a way I can change this to say it just displays "No caller ID available" or "CallerID Unknown?" |
| 05:50.29 | xeet2 | he's been fired |
| 05:50.30 | brenda | fearnor: just hope you don't have to call their support! |
| 05:50.34 | fearnor | my previous two reps were the bottomfeeding scum |
| 05:50.38 | DarthClue | yaaar: welcome to the world of realtime. it hasn't been standardized yet so some things are a little wacky. |
| 05:50.43 | fearnor | who'd lie to get the contract signed and then disappeared |
| 05:51.04 | xeet2 | fearnor: yep. common xo theme |
| 05:51.12 | fearnor | nod. *lots* of xo reps like that |
| 05:51.23 | fearnor | but there are good ones. |
| 05:51.25 | fearnor | anyway |
| 05:51.33 | fearnor | xo for local services and for voice services has been good to me |
| 05:51.49 | fearnor | i have dark fiber from them, in/out pris, point-point t1s |
| 05:51.54 | fearnor | that shit once it works, it works. |
| 05:52.03 | xeet2 | their ptp's are expensive |
| 05:52.07 | fearnor | not really |
| 05:52.16 | xeet2 | verizon was about 50% less in most cases for us at least |
| 05:52.18 | fearnor | with algx acquisition, they have lots more 'lit' co's |
| 05:52.21 | fearnor | that's strange |
| 05:52.35 | xeet2 | yeah |
| 05:52.46 | fearnor | most of cos in nyc are xo-lit, and things are fairly cheap |
| 05:52.57 | xeet2 | algx was why we sent to xo in the first place, we used them for low cost stuff before when they were intermedia |
| 05:53.01 | xeet2 | er, went |
| 05:53.39 | xeet2 | ah well, I'm just glad there is paetec |
| 05:54.05 | jayk- | anybody have any ideas? |
| 05:54.38 | xeet2 | jayk: what kind of phone/interface? |
| 05:54.41 | DarthClue | jayk-: are you using amp or a gui or just pure asterisk? |
| 05:54.43 | xeet2 | sip? analog? |
| 05:55.07 | jayk- | cisco 7960 |
| 05:55.12 | xeet2 | on sip you can change the agent string in sip.conf |
| 05:55.15 | jayk- | using a te110p digital t1 |
| 05:55.26 | *** join/#asterisk zoo (nobody@ip-132-16.travedsl.de) |
| 05:55.36 | jayk- | what's the agent string? |
| 05:55.40 | DarthClue | jayk-: are you using amp or a gui or just pure asterisk and is it head or stable? |
| 05:55.43 | xeet2 | you could also set the callerid with the setcallerid app, but if you are all sip, just change it in sip.conf |
| 05:55.58 | jayk- | DarthClue: plain asterisk, no gui. using 1.0.9 |
| 05:56.03 | xeet2 | is this for *all* calls or just some? |
| 05:56.21 | jayk- | callerid works fine from phone to phone, from inside that is |
| 05:56.35 | jayk- | outside calls coming in through the voice t1 show up as "From asterisk" |
| 05:56.40 | xeet2 | ok, soits all inbound calls on your t1? |
| 05:56.44 | jayk- | yeah |
| 05:56.46 | jayk- | right. |
| 05:56.57 | xeet2 | is it a pri or a cas trunk? |
| 05:57.00 | jayk- | apparnetly you can't get callerid on a t1 |
| 05:57.07 | fearnor | sure you can |
| 05:57.08 | jayk- | not a pri, just a voice t1 |
| 05:57.15 | DarthClue | jayk-: if your provider supports it, you can. |
| 05:57.20 | xeet2 | yes you can get caller id, did your provider say you can't? |
| 05:57.22 | jayk- | my provider doesn't support digital callerid |
| 05:57.25 | xeet2 | ah |
| 05:57.32 | fearnor | get a different provider |
| 05:57.33 | jayk- | only analog, on a pri |
| 05:57.35 | jayk- | nah |
| 05:57.36 | jayk- | not worth it |
| 05:57.40 | xeet2 | ok well you're not going to get caller id then |
| 05:57.44 | jayk- | yeah i know |
| 05:57.45 | fearnor | i really doubt it tho |
| 05:57.48 | jayk- | i just dont want it to say "From asterisk" |
| 05:57.49 | fearnor | who is teh providar |
| 05:57.50 | xeet2 | change the agent string in sip.conf =) |
| 05:57.53 | jayk- | people have no idea what the hell that is or means. |
| 05:57.58 | jayk- | integra telecom |
| 05:58.04 | jayk- | whats the agent string? |
| 05:58.08 | fearnor | do s,1,SetCallerIDName("FooBar") |
| 05:58.11 | fearnor | in your inbound context |
| 05:58.12 | xeet2 | or that too |
| 05:58.30 | *** join/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 05:58.35 | jayk- | k |
| 05:58.38 | jayk- | let me try that |
| 06:00.12 | xeet2 | yay timestamp skews are finally fixed |
| 06:04.16 | *** join/#asterisk joerg (~joerg@p548896CA.dip0.t-ipconnect.de) |
| 06:07.29 | jayk- | that didnt work |
| 06:07.30 | jayk- | hrmm |
| 06:08.03 | Qwell | fearnor: no quotes btw |
| 06:08.30 | jayk- | should i have removed teh quotes? |
| 06:08.40 | Qwell | jayk-: it would still work |
| 06:08.44 | jayk- | oh. hrm |
| 06:08.45 | Qwell | it would just have quotes |
| 06:09.03 | jayk- | ah |
| 06:09.11 | jayk- | oh well. |
| 06:14.41 | yaaar | hey wait...if it says 'from asterisk' then your box *has* to be providing digital callerid, and your provider is bullshitting you...otherwise, how the hell is it getting set to that? |
| 06:16.23 | jayk- | what would it say otherwise? wouldn't it just default to 'from asterisk'? |
| 06:16.28 | jayk- | i think i found part of the problem. |
| 06:16.33 | jayk- | SetCallerIDName doesn't exist? |
| 06:16.39 | Qwell | setcidname |
| 06:16.44 | jayk- | gotcha |
| 06:16.56 | Qwell | might help, yeah |
| 06:17.19 | jayk- | that fixed it |
| 06:17.24 | jayk- | however |
| 06:17.25 | jayk- | it says |
| 06:17.33 | jayk- | From FooBar |
| 06:17.34 | jayk- | <PROTECTED> |
| 06:17.36 | jayk- | <PROTECTED> |
| 06:17.43 | jayk- | why does it say asterisk below the callerid name? |
| 06:17.46 | Qwell | on what? |
| 06:17.51 | jayk- | the 7960 cisco phone |
| 06:18.14 | Qwell | thats the user agent, isn't it? |
| 06:18.31 | jayk- | in sip.conf? |
| 06:18.38 | Qwell | I guess. Whatever xeet2 said earlier |
| 06:18.55 | jayk- | i'm confused about that. |
| 06:19.42 | jayk- | do you know how i could change that or remove that second line that says asterisk? |
| 06:19.46 | Qwell | got me |
| 06:20.51 | loud | what would you put ? |
| 06:20.58 | loud | instead of * i mean |
| 06:21.03 | jayk- | nothing. :) |
| 06:21.20 | jayk- | i'd like it to just say "Caller ID Not Available." |
| 06:21.30 | Qwell | god, cnn and the like bug me |
| 06:21.38 | Qwell | "I was gay, then I went to this camp, and now I'm cured!" |
| 06:22.02 | loud | heh |
| 06:22.08 | Qwell | I just want to tell him "Hey pal...you're still gay..." |
| 06:22.10 | loud | like that book, how to stop being gya |
| 06:22.23 | loud | best seller and all |
| 06:22.38 | Qwell | loud: purchased by worried fathers I assume |
| 06:23.05 | loud | probably |
| 06:23.10 | blitzrage | where's the documentatin on dialplan functions in CVS again? |
| 06:24.30 | yaaar | so am i to understand from http://voip-info.org/tiki-index.php?page=Asterisk+RealTime+Extensions that even though i've got dynamic database-driven dialplans, i still have to add each new context to extensions.conf so i can put in the switch statement to call in realtime? if so, doesn't that mean i have to reload the whole config each time i add another context? which is to say, what's the point? |
| 06:24.33 | Qwell | oh god |
| 06:24.48 | Qwell | "That...uhh...lust...is still there, but its slowly subsiding. I don't think it'll ever go away." |
| 06:25.16 | loud | r e al t i m e, you add an extension and you wont have to reload or sip reload or extensions reload |
| 06:26.38 | yaaar | loud: right, that's what it's claiming....but at the url above it shows that the extensions database is implemented via a switch statement from a context in extensions.conf......doesn't that mean that you have to add each context to extensions.conf so as to have a switch statement pointing to the appropriate extensions database? |
| 06:26.41 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:27.12 | loud | my opinion, same thing heh, doing a sip reload or extensions reload wont kill a phone call. |
| 06:28.12 | *** join/#asterisk kabewm (~kabewm@24-180-28-208.pas-mres.charterpipeline.net) |
| 06:28.49 | Piranha- | has anyone gotten the zaptel interface working with asterisk on freebsd |
| 06:29.00 | jayk- | i figured out how to change the second line |
| 06:29.09 | Piranha- | ive used the port of zaptel, but keeps giving me a not configured error |
| 06:29.14 | jayk- | that's SetCIDNum |
| 06:30.49 | yaaar | hey i'll catch you guys tomorrow (or, well, later this morning) |
| 06:49.00 | *** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo) |
| 06:52.32 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 06:58.40 | *** join/#asterisk jiro5281 (~anton281@203.177.242.192) |
| 07:04.11 | *** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it) |
| 07:05.34 | *** join/#asterisk bjohnson_ (~bjohnson@i216-58-63-97.igs.net) |
| 07:12.28 | *** join/#asterisk ptiggerdine (~ptiggerdi@c210-49-98-194.rochd1.qld.optusnet.com.au) |
| 07:13.15 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 07:31.32 | *** join/#asterisk Paski_fr (~Paski_fr@stardust.noc.frontier.fr) |
| 07:33.29 | *** join/#asterisk eddi3 (eddi3@bb203-169-115-219.singnet.com.sg) |
| 07:34.05 | Qwell | You guys might know... How can I convert a variable bitrate mp3 into something usable? |
| 07:34.25 | kabewm | winamp w/ output to wav |
| 07:34.26 | eddi3 | hi does asterisk support those analog office phones with a LCD display that can show alot of info? |
| 07:34.36 | Qwell | kabewm: something in Linux |
| 07:34.40 | DarthClue | Qwell: lame |
| 07:34.59 | Qwell | DarthClue: ok, cool, I'll look at that. sox made a pile of crap :) |
| 07:35.07 | kabewm | rofl |
| 07:35.13 | kabewm | i misinterpreted that for a sec |
| 07:35.30 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
| 07:35.31 | eddi3 | not to mention several extra buttons like forward, conference, etc |
| 07:35.35 | DarthClue | kabewm: i expected to get a similar response out of everyone else. |
| 07:35.54 | Qwell | DarthClue: any suggested options? |
| 07:35.57 | DarthClue | eddi3: probably not, but it does depend on the phone. you would have to do some configuration to make them work though. |
| 07:37.07 | eddi3 | DarthClue then how does asterisk for eg, forward a call to another user? |
| 07:37.10 | DarthClue | Qwell: i've only done it on a windows box and i don't exactly recall which options i used. i remember i had to experiment with it. |
| 07:37.31 | DarthClue | eddi3: you should read the wiki. it's all about extensions and your dialplan configuration. |
| 07:38.30 | Qwell | sox created a 1.7gb wav file...crazy |
| 07:39.10 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 07:39.32 | DarthClue | Qwell: i know that i lamed mine then used sox to bump them to gsm. |
| 07:40.01 | Qwell | DarthClue: it isn't really asterisk related. I asked here, because I know you guys do conversions sometimes (such as to gsm) |
| 07:40.27 | DarthClue | Qwell: yeah, we convert lots of things. |
| 07:41.18 | Qwell | ETA is over 1 million hours. thats sweet |
| 07:41.49 | Qwell | Why do people even do vbr? Thats so stupid. |
| 07:42.31 | DarthClue | Qwell: that's like asking why people don't read the wiki first. |
| 07:42.43 | Qwell | touche |
| 07:51.32 | tuxinator_linuxM | I have a non asterisk problem. Anyone want to help me figure out SSL with apache? |
| 07:51.55 | DarthClue | tuxinator_linuxM: what's the problem? |
| 07:53.15 | tuxinator_linuxM | I setup the public cert and private key, and set up the .conf to reflect that, but I still can't access it. Let me pastebin some stuff. |
| 07:54.00 | DarthClue | go for it. i'll take a look at it. |
| 07:55.23 | fearnor | eddi: ADSI you mean? |
| 07:55.55 | tuxinator_linuxM | DarthClue: http://pastebin.ca/18688 |
| 07:56.48 | tuxinator_linuxM | DarthClue: I'm first trying to do a self sign. I'm using CentOS 4.0 |
| 07:57.52 | tuxinator_linuxM | DarthClue: I didn't have a problem on my RHEL 2.1 server's. I just can't figure it out on these ones. Driving me nuts. |
| 07:58.57 | tuxinator_linuxM | DarthClue: The funny thing is that I was able to get a https on my laptop (also running CentOS 4.0) |
| 07:59.38 | DarthClue | apache 1 or 2? |
| 08:00.05 | tuxinator_linuxM | 2 |
| 08:00.22 | DarthClue | try adding this in there... |
| 08:00.22 | DarthClue | SSLEngine on |
| 08:00.22 | DarthClue | SSLCipherSuite ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL |
| 08:00.27 | *** join/#asterisk darkskiez (~darkskiez@194.247.78.146) |
| 08:00.52 | DarthClue | you might also check the ssl.conf file and verify that it is configured properly |
| 08:02.05 | tuxinator_linuxM | it is stock, I haven't changed anything |
| 08:02.14 | tuxinator_linuxM | let me take a look again |
| 08:05.24 | tuxinator_linuxM | http://pastebin.ca/18690 |
| 08:06.28 | CoaxD | I wonder how practical a "Get your network from wireless, and SERVE turnkey wireless" application would be |
| 08:07.01 | CoaxD | ...i also wonder exactly how many hops one could get without network latency getting rediculously dumb, etc |
| 08:07.27 | DarthClue | tuxinator_linuxM: how are you starting apache? i use apachectl -D SSL -k start so that SSL is actually used. |
| 08:11.07 | tuxinator_linuxM | service httpd start |
| 08:11.14 | tuxinator_linuxM | let me check that |
| 08:13.27 | tuxinator_linuxM | DarthClue: Sorry, no difference |
| 08:14.01 | *** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no) |
| 08:14.11 | DarthClue | are you connecting via an ip or the domain name? if the name, try via the ip and see if it changes. also, anything in the logs? |
| 08:14.30 | tuxinator_linuxM | Type IP and domain, let me check the logs again |
| 08:16.13 | tuxinator_linuxM | DarthClue: error_log http://pastebin.ca/18691 |
| 08:16.59 | tuxinator_linuxM | ssl_error_log: [Thu Jul 28 01:12:43 2005] [warn] RSA server certificate CommonName (CN) `xxx.xxx.xxx' does NOT match server name!? |
| 08:17.29 | DarthClue | that shouldn't matter, it is just a warning. can you get any kind of ssl connection out of it? |
| 08:17.30 | tuxinator_linuxM | nothing in ssl_access_log or ssl_request_log |
| 08:17.55 | DarthClue | try connecting via localhost? |
| 08:18.00 | tuxinator_linuxM | I am not able to get it working on my Fedora Core 3 box either |
| 08:20.01 | tuxinator_linuxM | DarthClue: I only have SSH access, it's at the office. I elinks (test browser) and it also doesn't work |
| 08:20.11 | tuxinator_linuxM | text browser |
| 08:20.59 | DarthClue | is it on a public ip? tried doing an nmap to see if it's bound to the port? |
| 08:21.28 | tuxinator_linuxM | wait, it did work |
| 08:21.41 | tuxinator_linuxM | let me give it to you in private |
| 08:22.10 | DarthClue | yeah, don't post it here or you'll get the mini-slashdot effect when people start reading the logs. |
| 08:25.06 | *** join/#asterisk w0w0 (~w0w0@14.Red-81-39-84.pooles.rima-tde.net) |
| 08:27.06 | *** join/#asterisk pietro (~pietro@nat.xsec.it) |
| 08:27.17 | pietro | <PROTECTED> |
| 08:27.30 | pietro | i have the syslog with this |
| 08:27.48 | pietro | with a zaphfc cad |
| 08:27.50 | pietro | card |
| 08:28.08 | *** join/#asterisk Zeeek (~icechat5@Zeeek.active.supporter.pdpc) |
| 08:30.52 | *** join/#asterisk dacleric (~dacleric@p5482A04E.dip0.t-ipconnect.de) |
| 08:31.26 | mrtwister | who tested ooh323? |
| 08:33.28 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 08:35.18 | So3kris | hardwire: i got a 4521 |
| 08:42.41 | So3kris | Why is using people centos ? |
| 08:42.49 | So3kris | s/is/are |
| 08:44.05 | DarthClue | So3kris: why not? it is a logical choice for the Redhat crowd that can't afford the cost of RHEL |
| 08:44.12 | DarthClue | and that don't need the support. |
| 08:45.24 | So3kris | oh i use debian is i must run linux. Normaly i use FreeBSD the best os in the world. |
| 08:47.51 | kabewm | So3kris, everyone knows that the best os in the world is OS X |
| 08:49.08 | kabewm | Mac OSX, because it was easier to make *NIX user friendly than fix Windows . . . |
| 08:50.02 | tzafrir | It has a wierd name. Xserver has no relation to X server :-( |
| 08:53.40 | RaYmAn-Bx | Who needs a bot to start a distro fight? ;) |
| 08:56.32 | *** join/#asterisk postel (~zz@postel.user) |
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| 08:58.05 | *** part/#asterisk kaunietis (~kaunietis@81.7.64.227) |
| 09:03.40 | pietro | i have this error when i try to dialout over a Zap channel |
| 09:03.42 | pietro | http://pastebin.com/323412 |
| 09:03.46 | pietro | any suggestion ? |
| 09:05.10 | *** part/#asterisk kabewm (~kabewm@24-180-28-208.pas-mres.charterpipeline.net) |
| 09:05.16 | So3kris | DarthClue: kabewm yes OSX is great but i like it more than a desktop at server level FreeBSD an securty server OpenBSD :D |
| 09:06.40 | *** join/#asterisk xylome (~asterisk@hg-msq-hol.levigo.de) |
| 09:13.24 | *** join/#asterisk jiro5281 (~anton281@203.177.242.192) |
| 09:13.56 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 09:16.55 | Zeeek | TRSDOS rules! |
| 09:21.45 | *** join/#asterisk |nix (~inix@218.208.24.248) |
| 09:21.56 | tuxinator_linuxM | Zeeek: oh |
| 09:23.04 | *** join/#asterisk Fabe (~spamhere@217.91.11.247) |
| 09:23.20 | tuxinator_linuxM | Zeeek: http://en.wikipedia.org/wiki/TRS-DOS |
| 09:23.45 | Zeeek | Hey man, with a full 48k, it rocked! Not |
| 09:24.45 | *** join/#asterisk dwmw2 (~dwmw2@nat-pool-stn.redhat.com) |
| 09:25.35 | tzafrir | RaYmAn-Bx, no. It is just that this bot helps make it look pathetic |
| 09:25.57 | tuxinator_linuxM | Zeeek: It came out about the time I was born |
| 09:26.17 | tuxinator_linuxM | I was between version 1.3 and 2.3 |
| 09:27.04 | tuxinator_linuxM | Zeeek: Why the sudden interested in trsdos? |
| 09:27.29 | tzafrir | dr-dos is still live and developed (and actually even sold) |
| 09:27.46 | tuxinator_linuxM | in a third world country? |
| 09:27.54 | Zeeek | Because that was the first I ever heard of an OS :) |
| 09:27.55 | tuxinator_linuxM | oh, dr-dos |
| 09:27.57 | tzafrir | worse: on embedded systems |
| 09:28.09 | tuxinator_linuxM | Dr dos works fine |
| 09:28.12 | tuxinator_linuxM | used it before |
| 09:28.36 | tuxinator_linuxM | you have to admit, dos works better than windows |
| 09:28.36 | Zeeek | TRSDOS sucked so bad that someone made a replacement! I wrote an article about that one |
| 09:29.51 | tzafrir | tuxinator_linux, dos is not exactly an OS. |
| 09:30.05 | tzafrir | And "windows" refers to quite a few different programs. |
| 09:30.18 | tuxinator_linuxM | I know, I just being simplistic |
| 09:30.28 | tuxinator_linuxM | http://personal.nbnet.nb.ca/mclays/trsmod1.html <-- ya baby |
| 09:30.46 | newl | tzafrir: So Disk Operating System was meant to be what? :) |
| 09:32.39 | tuxinator_linuxM | I think it time for bed or food |
| 09:32.45 | tuxinator_linuxM | take it easy guys |
| 09:33.04 | Zeeek | I had one of those! |
| 09:33.16 | Zeeek | with the 48k expansions! |
| 09:33.31 | Zeeek | a whopping 64k floppy |
| 09:42.24 | InfraRed | 10:37 <@EnemySpy> http://www.ananova.com/news/story/sm_1478984.html <-- ROFL |
| 09:42.24 | InfraRed | 10:37 <@EnemySpy> "A Thai woman cut off her husband's penis after he asked her to make love one more time before he |
| 09:42.27 | InfraRed | <PROTECTED> |
| 09:43.24 | Zeeek | better watch out |
| 09:45.05 | InfraRed | better not cry |
| 09:45.19 | Zeeek | better not route |
| 09:45.29 | Zeeek | I'l tellin you why |
| 09:45.34 | Zeeek | SIP won't work |
| 09:45.42 | InfraRed | heh |
| 09:45.44 | Zeeek | NEXT! |
| 09:46.03 | InfraRed | fine |
| 09:55.57 | *** join/#asterisk mrtwister (~Andrius@cable-1-32.cgates.lt) |
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| 10:02.04 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 10:03.52 | limbique | hi |
| 10:06.28 | *** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org) |
| 10:08.50 | *** join/#asterisk forkqueue (~sam@2001:4bd0:2024:0:0:0:0:10) |
| 10:09.00 | forkqueue | Hey ppl |
| 10:09.23 | forkqueue | Anyone know if an E1 crossover cable is the same as a T1 crossover cable? |
| 10:09.31 | *** join/#asterisk amran (~amran@host213-120-96-247.in-addr.btopenworld.com) |
| 10:11.35 | Strom_C | a pair for transmit, a pair for receive |
| 10:11.43 | Strom_C | sounds the same to me |
| 10:12.07 | forkqueue | Strom_C: That's what I thought, but when it comes to telephony I've learnt not to trust logic :) |
| 10:12.55 | Strom_C | your head just isn't bell-shaped enough |
| 10:13.38 | clive- | how do I check if truning is working or not? |
| 10:13.39 | oej | Asterisk 1.2 presentation now published on http://www.astricon.net/asterisk1-2/ |
| 10:13.41 | clive- | trunking |
| 10:13.44 | forkqueue | Also, any ideas on how I can connect two BRI devices? Am I right in thinking there's not such thing as a BRI crossover cable? |
| 10:15.25 | *** join/#asterisk tango1 (~murgs@dsl-084-059-148-089.arcor-ip.net) |
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| 10:34.58 | amran | how sensible is it to start using hte ael language to write the 'dialplan'? is it still being heavily worked on, or will 1.2 push for usage of ael? |
| 10:37.28 | tango1 | newbie :( look for a windows software to connect asterisk testserver |
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| 10:42.51 | *** part/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
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| 10:50.13 | InfraRed | i'm expecting a big phonebill this month :/ |
| 10:51.47 | darkskiez | shiit |
| 10:51.56 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 10:52.06 | darkskiez | why not gprs? |
| 10:52.50 | *** join/#asterisk aiks (~aiks@159.148.227.104) |
| 10:54.26 | aiks | anyone have had some experience with asterisk BE on FC3 with wildcard? |
| 10:54.42 | aiks | oh, hi, btw |
| 11:00.38 | InfraRed | asterisk be? |
| 11:00.47 | aiks | business edition |
| 11:00.58 | aiks | the precompiled one from digium |
| 11:01.53 | aiks | i have a problem with TE110P getting to work |
| 11:02.05 | aiks | most likely this is zaptels problem. |
| 11:02.20 | aiks | i already made needed changes to udev |
| 11:02.30 | aiks | ran udevstart |
| 11:02.42 | *** join/#asterisk kapejod (~kapejod@e178001061.adsl.alicedsl.de) |
| 11:04.36 | aiks | modeprobed wcte11xp |
| 11:04.53 | aiks | ztcfg still gives ZT_CHANCONFIG failed on channel 25: No such device or address (6) |
| 11:05.42 | kapejod | set the jumper for e1 |
| 11:05.50 | kapejod | or use the t1e1override option |
| 11:05.54 | aiks | on the card? |
| 11:05.59 | kapejod | yes |
| 11:08.25 | *** join/#asterisk jiro5281 (~anton281@203.177.242.192) |
| 11:09.22 | aiks | :) |
| 11:09.31 | aiks | yup, would suite just fine right now |
| 11:09.40 | aiks | or "large trout" |
| 11:10.49 | *** join/#asterisk buzzyd (~buzzyd@82-35-241-21.cable.ubr01.enfi.blueyonder.co.uk) |
| 11:11.23 | buzzyd | Hi, Does anyone here use sipgate and asterisk for in and out calls? |
| 11:12.22 | aiks | okay, got the card in front of me: - i see one jumper (T1 strap off /e1 strap on) |
| 11:13.02 | aiks | the jumper is set on only one of the two pins |
| 11:13.07 | aiks | should it cover both? |
| 11:13.14 | kapejod | yes |
| 11:13.39 | aiks | okay, will try to boot up one more time |
| 11:14.03 | kapejod | or use the big hammer on the jumpers so they are closed... |
| 11:15.10 | aiks | :) |
| 11:15.14 | aiks | booting up |
| 11:15.40 | aiks | i just hope this layout wont burn the chip down |
| 11:19.18 | aiks | wow :)))))) |
| 11:19.32 | aiks | kapejod many, may, many thanks |
| 11:19.51 | aiks | at least ztcfg didnt show any problems |
| 11:20.50 | kapejod | you're welcome. |
| 11:21.30 | aiks | yet one thing makes me a bit nervous, i apologise for the spam already |
| 11:21.44 | aiks | /var/log/messages |
| 11:21.45 | aiks | Jul 28 14:18:31 asterisk kernel: TE110P: Setting up global serial parameters for E1 FALC V1.2 |
| 11:21.46 | aiks | Jul 28 14:18:31 asterisk kernel: TE110P: Successfully initialized serial bus for card |
| 11:21.46 | aiks | Jul 28 14:18:31 asterisk kernel: Found a Wildcard: Digium Wildcard TE110P T1/E1 |
| 11:21.46 | aiks | Jul 28 14:18:31 asterisk kernel: Registered tone zone 3 (Netherlands) |
| 11:21.46 | aiks | Jul 28 14:18:31 asterisk kernel: TE110P: Span configured for CCS/HDB3/CRC4 |
| 11:21.47 | aiks | Jul 28 14:18:31 asterisk kernel: Calling startup (flags is 4099) |
| 11:21.49 | aiks | Jul 28 14:18:31 asterisk kernel: wcte1xxp: Setting yellow alarm |
| 11:21.51 | aiks | and then |
| 11:21.58 | aiks | Jul 28 14:18:36 asterisk wait_for_sysfs[3412]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug-devel@lists.sourceforge.net> |
| 11:22.20 | aiks | is tha ok? |
| 11:22.30 | aiks | that |
| 11:23.27 | kapejod | does it work? |
| 11:23.41 | aiks | i dont have PRI cable yet |
| 11:24.01 | aiks | pulsing red light on the back of the card |
| 11:24.55 | aiks | okay, found newsgroup posting about the latter problem |
| 11:24.59 | aiks | nevermind |
| 11:26.49 | aiks | just needed to update udev |
| 11:27.57 | aiks | now works perfectly - thanks you all so very much. i mean it ;) |
| 11:28.25 | forkqueue | buzzyd: I've used sipgate for calls in the past |
| 11:31.52 | *** join/#asterisk meppl (~mephisto@84.245.164.79) |
| 11:32.29 | zoo | I am looking for a documentation of SIP DID routing, but i dont find any. |
| 11:32.39 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 11:32.45 | zoo | I already appended /1234 to register |
| 11:33.08 | zoo | but how do i write the DID rules in [default] of extensions.conf? |
| 11:33.34 | zoo | just 1234,PRIO,COMMAND()? |
| 11:33.44 | zoo | just exten => 1234,PRIO,COMMAND()? |
| 11:35.49 | limbique | hi anyone :) |
| 11:36.31 | limbique | does anybody knows why i get a Response: Error Message: Ivalid/unknown command on this command? Action: Originate |
| 11:36.51 | InfraRed | in sip.conf what other settings can i put to send username? |
| 11:37.17 | InfraRed | i have : username, user, and fromuser |
| 11:37.24 | InfraRed | but in sip debug i get this |
| 11:37.59 | InfraRed | Proxy-Authorization: Digest username="", realm="XXXXX", algorithm=MD5, uri="sip:44XXX@XXX", nonce="42e8c3bbefda5f81bda1cfaccfc87d7cc33322dc", response="377e1534cd5ca8a1c31376237da6e7ca", opaque="" |
| 11:38.18 | InfraRed | username is blank |
| 11:38.52 | zoo | InfraRed: is your register-statement correct? |
| 11:38.58 | InfraRed | yes |
| 11:39.04 | InfraRed | it shows me as registered |
| 11:39.18 | InfraRed | sip show registery |
| 11:40.02 | zoo | i dont know |
| 11:40.10 | InfraRed | :/ |
| 11:40.25 | zoo | but i am a newbie, still |
| 11:41.14 | limbique | does anybody knows why i get a Response: Error Message: Ivalid/unknown command on this command? Action: Originate |
| 11:41.50 | zoo | limbique: no |
| 11:42.21 | limbique | hmm :*( |
| 11:44.19 | zoo | InfraRed: it says Proxy-Authorization |
| 11:44.37 | zoo | here it only says "Authorization: Digest username="xxx"... |
| 11:44.55 | *** join/#asterisk Henguei (~Henguei@196.203.53.45) |
| 11:45.03 | *** join/#asterisk mrproper_ (~psynode@CPE-60-225-51-188.nsw.bigpond.net.au) |
| 11:45.26 | mrproper_ | anyone know why i would be getting no audio on incoming oh323 calls but outgoing oh323 calls are fine? |
| 11:45.44 | zoo | sounds like a firewall/port issue |
| 11:46.02 | aiks | nat |
| 11:46.09 | mrproper_ | no nat, no firewall |
| 11:46.11 | eldu | mrproper_: hi |
| 11:46.15 | mrproper_ | eldu: hi |
| 11:46.28 | aiks | hmm, not even local firewall |
| 11:46.38 | eldu | nat/pat prob ? |
| 11:46.43 | aiks | like one on the asterisk itself |
| 11:46.49 | aiks | asterisk box i mean |
| 11:47.06 | zoo | maybe you neet to bind to an ip-address |
| 11:47.12 | eldu | what kind of codec and phone r u using ? |
| 11:47.16 | InfraRed | fucken unbelievable |
| 11:47.16 | mrproper_ | zoo: hmm ill try that |
| 11:47.18 | InfraRed | had to rename the tag |
| 11:47.20 | InfraRed | to [hostname.domain.com] |
| 11:47.24 | mrproper_ | eldu: im using alaw on the sip phone and oh323 |
| 11:47.49 | eldu | hard or soft phone ?. |
| 11:48.01 | zoo | mrproper_: i had problems binding to 0.0.0.0, too. With an IP it worked here. But that was with SIP |
| 11:48.25 | mrproper_ | eldu: if i dial from an h323 endpoint to voicemail, i cant hear any audio |
| 11:48.34 | eldu | coz i experienced some one way chat on hardphone with a bad G729 codec config |
| 11:48.36 | mrproper_ | eldu: so i doubt its phone related |
| 11:49.05 | eldu | ok ok |
| 11:49.17 | mrproper_ | i can see on the asterisk console when i call voice mail its 'playing' the sound files but i hear nothing on the phone |
| 11:49.32 | mrproper_ | but sip to voicemail for example works fine |
| 11:49.59 | Henguei | hello ! what is bindaddr=192.168.0.1 ? have i to change it , cause my ip adress is 192.168.1.4 |
| 11:50.10 | mrproper_ | if i do a oh323 show channels, it shows the channel and format says unknown |
| 11:50.11 | *** join/#asterisk grimse (~grimse@p5481BFA8.dip.t-dialin.net) |
| 11:53.23 | mrproper_ | this is what the channel says on an oh323 incomming call:0 ip$172.16.100.31:1357/13933 RING NONE Remote 0/0 unknown |
| 11:54.28 | *** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net) |
| 11:56.40 | *** join/#asterisk jeh_work (~jeh@ext116.almare.com) |
| 11:58.08 | jeh_work | hi folks |
| 11:58.28 | mrproper_ | any ideas about the oh323 problem? |
| 11:59.07 | Henguei | please what is bindaddr=192.168.0.1 ? |
| 11:59.18 | jalsot | hi |
| 12:00.10 | kaldemar | Henguei: that is the address the service is listening to. change it yo your ip. |
| 12:00.45 | mrproper_ | Henguei: you should be able to leave it as bindaddr=0.0.0.0 |
| 12:01.24 | Henguei | my ip or bindaddr=0.0.0.0 ? |
| 12:01.29 | jalsot | does anybody know in which PCI-X slot is better to put TE110P? 100MHz vs. 133MHz |
| 12:01.55 | kaldemar | Henguei: in case of 0.0.0.0, it should listen to all local addresses. |
| 12:02.11 | Henguei | ok thks kaldemar |
| 12:03.01 | Henguei | and what is bindport=4569 ? |
| 12:03.18 | kaldemar | Henguei: the communications port the service uses. |
| 12:03.29 | kaldemar | Henguei: 4569 is the default port for IAX2 protocol. |
| 12:03.42 | aiks | i am not sure if PCI-X is suited for TE110P at all |
| 12:03.57 | aiks | i guess you should look for standart PCI |
| 12:04.04 | Henguei | ok |
| 12:04.05 | aiks | but i might be wrong |
| 12:04.10 | Henguei | :) |
| 12:09.34 | jeh_work | a newbie question. if i have calls in a queue and want to redirect them somewhere else, is the AMI Redirect action what i want? |
| 12:10.06 | kapejod | jeh_work: it might be |
| 12:10.23 | jeh_work | i have a simple java app that communicates using AMI, and so far no major problems |
| 12:10.58 | jeh_work | kapejod: ok, what would cause the "might" there? |
| 12:11.38 | jeh_work | i may be a bit lost here, new as i am to the whole world of asterisk... rtfm pointers are most welcome :) |
| 12:12.00 | *** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com) |
| 12:12.51 | kapejod | jeh_work: there is always a "might" ;) |
| 12:13.17 | jeh_work | kapejod: heh, true. but, Redirect isn't obviously the wrong track then |
| 12:13.35 | kapejod | it might not be the wrong track. ;) |
| 12:14.54 | jeh_work | i'll see where it leads me then |
| 12:15.21 | jeh_work | the params to the action aren't that well documented |
| 12:15.31 | kapejod | less manager.c |
| 12:15.57 | *** join/#asterisk zotz (~zotz@24.231.36.100) |
| 12:16.51 | jeh_work | is there something else that can be used to transfer/redirect a call? |
| 12:17.39 | kapejod | app_transfer |
| 12:19.31 | jeh_work | kapejod: ok, thanks for the help |
| 12:19.41 | kapejod | which help? ;-) |
| 12:20.46 | limbique | does anyone knows why to set some variables in the originate command? (var1=23|var2=24|var3=25) see : http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate |
| 12:20.51 | jeh_work | well, you said "you aren't necessarily that far off the correct track" |
| 12:21.43 | kapejod | a few centimeters on or off the track could mean life or death (when the train comes...) |
| 12:22.25 | limbique | lol |
| 12:25.27 | *** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985701.sympatico.ca) |
| 12:27.55 | jeh_work | kapejod: i'll take necessary blame transfer precautions |
| 12:28.06 | kapejod | lol |
| 12:29.53 | darkskiez | what does Transfer do that Dial doesnt? |
| 12:30.36 | kapejod | it deflects a call |
| 12:30.39 | darkskiez | ive never noticed the transfer app before |
| 12:30.45 | kapejod | if the channel supports it |
| 12:30.55 | kapejod | like sip redirects |
| 12:31.00 | darkskiez | how is that different to just doing Dial(blah) |
| 12:31.03 | kapejod | or isdn call deflection |
| 12:31.15 | darkskiez | OH |
| 12:31.16 | kapejod | transfer() gets the call from your box |
| 12:31.20 | darkskiez | Thats how you do that |
| 12:31.42 | kapejod | nice to build loadbalancing stuff |
| 12:32.31 | darkskiez | other than ISDN transfer, i dont understand the benefits over Dial |
| 12:32.48 | kapejod | sip redirects! |
| 12:33.31 | darkskiez | does that not happen when you do Dial(Sip/blah) ? |
| 12:33.54 | kapejod | of course not |
| 12:34.05 | darkskiez | is that not the reinvite stuff |
| 12:34.19 | kapejod | that is totally different |
| 12:34.35 | darkskiez | reinvite redirects RTP, transfer redirects whole call ? |
| 12:35.04 | kapejod | yes, sir. |
| 12:35.20 | aiks | aih aih cptn. |
| 12:35.40 | *** join/#asterisk canabis (~CaN@63.144.16.242) |
| 12:35.46 | darkskiez | if a sip phone does a transfer, can asterisk catch this in the dial plan, and handle it differently? |
| 12:36.18 | darkskiez | like with a call forward |
| 12:36.23 | kapejod | you could modify chan_sip for that. |
| 12:39.05 | InfraRed | http://news.bbc.co.uk/2/hi/technology/4718719.stm |
| 12:39.06 | InfraRed | lol |
| 12:40.18 | limbique | haha |
| 12:40.23 | limbique | open source beer :P |
| 12:40.37 | *** join/#asterisk newl (~newlook@203-59-217-50.dyn.iinet.net.au) |
| 12:41.34 | InfraRed | open source brain fuckup |
| 12:41.35 | InfraRed | :) |
| 12:41.41 | aiks | :) |
| 12:42.16 | *** join/#asterisk zshuke (q@66.194.40.30) |
| 12:42.19 | InfraRed | http://www.voresoel.dk/main.php?id=70 |
| 12:43.03 | *** part/#asterisk kapejod (~kapejod@e178001061.adsl.alicedsl.de) |
| 12:43.17 | RaYmAn-Bx | it actually reached bbc? Scary. |
| 12:43.32 | InfraRed | haha |
| 12:43.46 | zshuke | Hello. I have yesterday's CVS of *. If I call * from a SIP phone that loses power, the thread * spawns never exits. I assume this is a bug? |
| 12:43.53 | aiks | whereever you are, however you listen, this is free beer |
| 12:44.26 | RaYmAn-Bx | not that many people have the equipment to make beer at home though :P |
| 12:44.36 | InfraRed | like i said in another channel |
| 12:44.39 | InfraRed | it's pikey homebrew with academic label |
| 12:44.44 | aiks | well ... |
| 12:44.53 | aiks | U dont need much equipment really |
| 12:45.00 | gordonjcp | it's a pretty poor recipe |
| 12:45.04 | aiks | lots of sugar water and yeast |
| 12:45.16 | gordonjcp | RaYmAn-Bx: you need a plastic barrel, that's about it |
| 12:45.22 | RaYmAn-Bx | fair enough |
| 12:45.31 | aiks | and hidiho mr.hangover |
| 12:45.46 | gordonjcp | aiks: depends on what you brew, my homebrew isn't hangover-y at all |
| 12:45.46 | nDuff | zshuke, sounds like one. |
| 12:46.00 | gordonjcp | aiks: cheap shitty chemical pish lager is really bad for hangovers |
| 12:46.02 | gordonjcp | like Stella |
| 12:46.05 | aiks | gordonjcp, just kiddin |
| 12:46.37 | aiks | oh, havent tried that one, - i usually stick to our nation beers |
| 12:47.07 | zshuke | nDuff: I just find it hard to believe noone has tested this in the past :P I can send a SIG_KILL to a softphone and get the same reaction from * |
| 12:47.16 | zshuke | (the threads never exit) |
| 12:47.40 | jake1932 | ah crap I just missed him |
| 12:49.08 | darkskiez | thats depressing, our pri doesnt seem to support the Transfer call :( |
| 12:49.28 | buzzyd | I am wonder if someone can give me some advice I am trying to configure an extension that will allow 1 call in and if that is in goes straight to voicemail instead of ringing and then going |
| 12:49.44 | buzzyd | I'm using check/setgroup to limit |
| 12:49.53 | nDuff | zshuke, #asterisk-bugs is possibly a better place to be on this topic, btw. |
| 12:50.09 | zshuke | Oh, thanks |
| 12:50.11 | nDuff | zshuke, you can also search on bugs.digium.com to see if this has already been reported/filed. |
| 12:50.50 | zshuke | nDuff: Great, I'll do that. |
| 12:52.30 | darkskiez | buzzyd: I had fun trying to do that, set the OUTGOING_GROUP when calls are being directed to that phone, and set GROUP on calls comeing from that phone, outwards. |
| 12:52.40 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:52.56 | buzzyd | ah of course thanks :) |
| 12:53.26 | darkskiez | buzzyd: took me a whole night of reading crap to find the OUTGOING_GROUP thing. |
| 12:53.51 | darkskiez | buzzyd: except its OUTBOUND_GROUP |
| 12:54.11 | buzzyd | Its always the little things that end up taking the most time |
| 12:56.04 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 12:56.29 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 13:04.06 | doughecka_ | does adtran's channel banks detect distintive ring? |
| 13:19.40 | *** join/#asterisk lters_ (~lters@eg1.ekn.com) |
| 13:20.56 | *** join/#asterisk phil0u (~philou@81.56.194.193) |
| 13:20.57 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 13:21.01 | phil0u | 'lo |
| 13:26.33 | *** join/#asterisk lehel (~Lehel@82.79.20.17) |
| 13:27.00 | lehel | hello |
| 13:28.05 | *** join/#asterisk DonX (don@tool.sparkhosting.net) |
| 13:32.51 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 13:33.21 | phil0u | would by any chance someone here would be using asterisk in France ? or Europe maybe ? |
| 13:34.02 | phil0u | i'm getting mad with supervision disconnection with 2 X101P and 2 PSTN lines |
| 13:35.39 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 13:35.43 | phil0u | basically everything works fine, except that when i try SIP1 => PSTN1 => PSTN2 => SIP2 and when i hang up on SIP1, then PSTN2 doesn't see that the first line has hung up and then switch to voicemail, which will by default record for ever |
| 13:36.46 | *** part/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 13:38.30 | phil0u | by looking at voip-info.org, and google and source code reading, i found out that i need to pass "opermode=1" to wcfxo modules, and so i think everything's correct. I've put a normal phone + answering machine on the line, and when i call it directly through SIP1 and hang up, the line seems to hang up pretty quickly as well ( maybe it's the phone's busydetect function) |
| 13:38.46 | *** join/#asterisk inspired (mikael@213.197.167.61) |
| 13:39.37 | phil0u | i can't imagine that "France Telecom", the historical phone carrier wouldn't provide "Disconnect Supervision" |
| 13:39.40 | Darwin35 | ok man fixing broken dial plans suck |
| 13:41.41 | phil0u | of course signalling is set correctly yo kewlstart in corresponding configuration files |
| 13:41.43 | *** join/#asterisk zaptel (~just@216.194.173.2) |
| 13:41.50 | phil0u | s/yo/to/ |
| 13:42.02 | [TK]D-Fender | Darwin35 : How bad could it be? |
| 13:42.04 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 13:42.36 | [TK]D-Fender | Pastebin it for our entertainment.... or would that be pity? |
| 13:49.07 | *** join/#asterisk sangee (~rkuru@207.188.77.86) |
| 13:50.28 | sangee | If calling party is hangup, it's goto n+1 or n+101 in dialpeer (extensions.conf)? |
| 13:51.27 | jake1932 | sangee: if the calling party hangs up, dialplan processing stops |
| 13:51.45 | sangee | how do i catch that |
| 13:52.05 | jake1932 | you can catch it by using asterisk manager |
| 13:52.50 | sangee | but i want to write the duration into my database? how do i do that? |
| 13:52.55 | pif | bongiorno |
| 13:53.33 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 13:53.44 | jake1932 | sangee: or try the "h" extension (never used that) |
| 13:54.05 | sangee | i don't understand |
| 13:54.12 | sangee | can you give me an example? |
| 13:54.42 | jake1932 | exten => h,1,System(write to db cmd) |
| 13:54.51 | jake1932 | exten => h,2,Hangup |
| 13:55.09 | sangee | ok, i will try now |
| 13:56.53 | Darwin35 | man as I fix things I like this iax2 fone more and more |
| 13:57.15 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
| 13:58.02 | jake1932 | sangee: according to http://www.voip-info.org/tiki-index.php?page=Asterisk+h+extension - h is not reliable. If it doesn't work for you, you might want to use the cdr that is automatically created - it has duration |
| 13:58.24 | *** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 13:59.45 | *** part/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 14:00.52 | sangee | Hi Jake1932, how do i use the deadAgi? |
| 14:01.16 | sangee | it's going to that extension,but complaining to use deadAgi? |
| 14:01.31 | jake1932 | are you simply tracking call durations? |
| 14:01.53 | lters_ | iax2 fone? |
| 14:02.03 | sangee | i want to write cdr info to database |
| 14:02.24 | jake1932 | sangee: just use cdr db |
| 14:02.37 | jake1932 | sangee: it'd be much easier |
| 14:02.50 | sangee | ok |
| 14:02.56 | sangee | i will try that |
| 14:02.58 | lters_ | Darwin35, what fone is that? |
| 14:03.07 | sangee | thx for your help |
| 14:03.11 | jake1932 | np |
| 14:04.57 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:05.01 | Ariel_ | hello everyone |
| 14:05.54 | *** join/#asterisk Johann000 (~nobody@12.44.215.11) |
| 14:06.15 | clive- | is there a way to see if trunking os working? |
| 14:06.41 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 14:06.43 | yaaar | word |
| 14:06.45 | jake1932 | clive-: are you in the US? |
| 14:06.56 | zaptel | hello everybody, what's the default jumper setup for the TE110P card? |
| 14:07.06 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 14:07.13 | astoria | zaptel: are you on a T1 or E1? |
| 14:07.14 | jake1932 | zaptel: on |
| 14:07.43 | *** join/#asterisk Tili (~Tili@202-133-65-124-dialup.sat.net.pk) |
| 14:07.55 | wizhippo | has anyone used fxotune? I run it but fxotune.conf has nothing but 0's in it. I find that strange. |
| 14:08.13 | jake1932 | zaptel: E1 is when the jumper is on |
| 14:08.17 | zaptel | the problem is that the card is in another buiding, i neet it to be on T1 but i don't know if thats the default |
| 14:08.34 | astoria | zaptel: it wasn't for me. I had to remove the jumper. |
| 14:08.37 | jake1932 | wizhippo: just used it yesterday |
| 14:08.37 | yaaar | ok, so i'm a bit confused. I'd like to have the same extension in a few different contexts, using a mysql realtime config database. but the wiki seems to indicate that the 'name' field must be unique, and should correspond with the extension name? i'm just taking that from the table where it shows how you would put the 'foo' config into the db |
| 14:08.53 | zaptel | thanks jake1932 |
| 14:08.57 | jake1932 | np |
| 14:08.59 | astoria | zaptel: some people receive it in E1 default, some in T1 defult |
| 14:09.00 | wizhippo | jake1932: did you get the same result or did actualy do something for you? |
| 14:09.03 | Tili | does asterisk support tcp for IAX2 |
| 14:09.20 | jake1932 | wizhippo: i got one number then a bunch of zeros |
| 14:09.25 | astoria | Tili: why would you want to use tcp? |
| 14:09.49 | zaptel | astoria: oh ok so i can not just assume a default setup, huh? |
| 14:09.59 | *** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au) |
| 14:10.04 | astoria | zaptel: yeah, you're going to have to look at it yourself. it varies |
| 14:10.11 | jake1932 | wizhippo: 4=11,0,0,0,0,0,0,0,0 |
| 14:10.21 | zaptel | ok thank you very much guys |
| 14:10.40 | Tili | astoria: with tcp I can solve NAT issue related to inbound calls. Where client is behind NAT. just like Skype does |
| 14:11.03 | Jimme | quick question, using a decent quality codec, how many voice calls can you run over 100mbit ethernet with QoS ? |
| 14:11.05 | astoria | Tili: you can traverse NATs with UDP too. |
| 14:11.06 | jake1932 | does tcp make sense for VOIP? |
| 14:11.24 | astoria | jake1932: no, there is too much overhead. |
| 14:11.32 | jake1932 | didn't think so |
| 14:11.40 | nDuff | jake1932, also, TCP doesn't let you discard packets that are too old to be useful |
| 14:11.52 | webman | using iax2, does a peer have a context, or is that defined by the other end anyway?? |
| 14:11.56 | jake1932 | Jimme: it depends - are we talking g729? |
| 14:12.09 | *** join/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu) |
| 14:12.22 | jake1932 | Jimme: http://www.voipproviderslist.com/voice-over-ip-calculator/ |
| 14:12.25 | Jimme | yea, g729 |
| 14:12.25 | Jimme | thanks |
| 14:12.29 | Jimme | thats what i needed :) |
| 14:12.30 | Tili | astoria: paranoid Firewalls sometimes have UDP restriction. I learned thru skype that it makes tcp connection to SkypeNode and then uses that for incoming calls. |
| 14:12.30 | Tili | kind of an idle connection |
| 14:12.32 | Tili | i am talking about both NAT and firewall issues at sametime. |
| 14:12.35 | webman | jimme: is a decent codec one that sounds good (g711) or a small one (ilbc or something)? |
| 14:12.40 | *** join/#asterisk kingtux (~susekid@pool-141-157-9-215.balt.east.verizon.net) |
| 14:12.43 | Tili | otherwise I will have to write a TCP tunneling app. |
| 14:12.46 | kingtux | Hello ALL |
| 14:13.06 | astoria | Yeah, you're probably going to have to, I don't think you can change any settings to do that. |
| 14:13.08 | kingtux | Has anyone implemented a calling card system with * |
| 14:13.24 | astoria | kingtux: yeah, a lot of people have |
| 14:13.37 | jake1932 | kingtux: http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications |
| 14:13.54 | *** join/#asterisk ctjctj (ctjctj@192.55.203.130) |
| 14:13.59 | kingtux | I"m pondering the idea of setting something like this up |
| 14:14.25 | kingtux | But really not sure what is all I need to have this work |
| 14:14.44 | webman | I have an IAX connection to my VOIP provider, if I do Dial(IAX2/username:secret@hostname/${EXTEN} it works, but if I define a peer with same user/pass and call DIAL(IAX2/username/${EXTEN}) I get Call rejected by 202.92.33.172: No such context/extension |
| 14:15.11 | Jimme | decent = sound good :) |
| 14:15.56 | jake1932 | webman: sounds like your peer entry is not set up properly |
| 14:16.10 | webman | anyone know some trick to getting that to work by dialling without the user/pass in the extensions.conf ? |
| 14:16.15 | ctjctj | Hello again. I'm still trying to get the festival app to work and having a hard time of it. The festival app exists, it parases festival.conf correctly. It connects to the festival server. It returns a WV. I've had it cache that wave form. It then plays that wave form and it sounds like a very short, very soft white noise. Festival is producing correct wave forms when run by hand. |
| 14:16.31 | kingtux | Is there a doc ou there that shows what will be needed to setup a calling card system |
| 14:16.38 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 14:16.39 | webman | jake1932: I was just copying the config from the wiki.... |
| 14:16.47 | astoria | kingtux: it's not exactly 1-2-3 |
| 14:17.08 | ManxPower | ~docs |
| 14:17.08 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 14:17.10 | ManxPower | ~mailinglist |
| 14:17.10 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 14:17.12 | jake1932 | kingtux:have you read the page I gave you? |
| 14:17.25 | kingtux | yes read it last night |
| 14:17.53 | astoria | do people still use calling cards? |
| 14:17.59 | jake1932 | kingtux: msg shido6 - he'll be able to help you get it up and running |
| 14:18.11 | kingtux | Yes in the hood |
| 14:18.23 | kingtux | I'm looking to start a calling card biz |
| 14:18.50 | kingtux | I live in an area where thier are alot of ethnic backgrouds |
| 14:19.13 | kingtux | latin, african mostly |
| 14:19.38 | kingtux | I'm looking to set somthing up to target thos markets |
| 14:20.02 | ctjctj | ManxPower: were the docs and mailing list references for me? I've been searching google and the wiki for a couple of days now. |
| 14:20.24 | astoria | DarthClue: are you around? |
| 14:22.22 | *** join/#asterisk wunderkin (kev@24.137.147.163) |
| 14:22.45 | kingtux | shido6 u around |
| 14:22.46 | *** join/#asterisk dasenjo (~dasenjo@208.195.214.9) |
| 14:22.57 | webman | BTW, for those that are interested, I was missing the peercontext=context in my peer definition.... this made all the difference !!!! |
| 14:23.53 | jake1932 | kingtux: greg at nufone.net |
| 14:24.08 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E92A.versanet.de) |
| 14:24.09 | Nebukadneza | hi |
| 14:24.25 | *** join/#asterisk Aze` (~aze@85.18.136.114) |
| 14:24.38 | *** join/#asterisk santiago (~santiago@63.245.86.141) |
| 14:25.19 | kingtux | jake1932 ??? |
| 14:25.27 | jake1932 | i'm here |
| 14:25.30 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:25.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:26.19 | riemensc | everybudddy speask german? |
| 14:26.47 | mut | only the germans |
| 14:26.47 | *** part/#asterisk santiago (~santiago@63.245.86.141) |
| 14:27.27 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:27.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:29.19 | clive- | athm? |
| 14:29.25 | clive- | anthm... |
| 14:29.37 | *** join/#asterisk Nix (~Nix@81.214.255.57) |
| 14:29.39 | anthm | yes? |
| 14:29.44 | bkw_ | YAY ITS NIX!!! |
| 14:29.54 | Darwin35 | hey bkw and anthm |
| 14:29.57 | Darwin35 | morning |
| 14:30.02 | Nix | whoop there it is! |
| 14:30.13 | bkw_ | Nix, congrats on the flight.. see ya soon eh? |
| 14:30.31 | Darwin35 | run as far as you can when you get where your going I will be there waiitng on you |
| 14:30.46 | bkw_ | oh for the love of god.. NOOOOOOOO |
| 14:30.47 | bkw_ | :P |
| 14:30.49 | riemensc | I cannot lead outgoing telephone calls over iax! |
| 14:30.53 | Nix | yep. |
| 14:30.55 | riemensc | can you help me |
| 14:31.01 | bkw_ | lead? |
| 14:31.24 | Nix | I will arrive in Chicago either at middaty or 2pm |
| 14:31.38 | bkw_ | Nix, seen that.. everyone is arriving about the same time |
| 14:31.48 | Nix | depending if I fly istanbul - munich - chicago or istanbul - frankfurt - chicago |
| 14:31.51 | bkw_ | ken, file, brc all arrive within 20 min of each other |
| 14:31.58 | astoria | Is anyone at cluecon planning on going downtown for drinks? |
| 14:32.18 | astoria | after the day's events, that is.. |
| 14:32.25 | bkw_ | astoria, hell ya boi.. no so much to drink than to see the town |
| 14:32.40 | astoria | i love chicago, i'm driving in from Detroit |
| 14:32.45 | Darwin35 | no we are going back to BKW's room for drinks and party faviors |
| 14:32.51 | bkw_ | um NO |
| 14:32.54 | bkw_ | NEXT!!! |
| 14:32.58 | astoria | uh oh, is this going to be like defcon? |
| 14:33.08 | bkw_ | ok I spent some time trying to track down this IAX issue |
| 14:33.14 | bkw_ | check the timestamps with ethereal |
| 14:33.17 | bkw_ | all looks fine |
| 14:33.27 | bkw_ | I can't get my box to fuck up nor jitter the audio like Strom_C can |
| 14:33.31 | bkw_ | very freakin weird |
| 14:33.48 | *** part/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 14:34.01 | *** join/#asterisk Tili_ (~Tili@202-133-67-52-dialup.sat.net.pk) |
| 14:34.03 | anthm | careful you might go "off topic" and get chewed out by one of the guards |
| 14:34.36 | mut | uckly duckly doo |
| 14:34.57 | Nix | So, Craig is bringing me some bottles of black label rum from Australia so I hope we can find a place to drink them within minutes of landing |
| 14:34.59 | Nix | :-D |
| 14:35.00 | Darwin35 | only issue I am having with my new iax phone is dialing it internally |
| 14:35.23 | gambolputty | asterisk.conf initialize crypto at startup? |
| 14:35.28 | Darwin35 | I can call out on it . but when I dial exten 1004 I get a busy tone insted of going to vm |
| 14:35.32 | gambolputty | there some new crypto feature? |
| 14:36.13 | Nebukadneza | hm |
| 14:36.19 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 14:36.46 | *** part/#asterisk zaptel (~just@216.194.173.2) |
| 14:37.02 | Nebukadneza | i got a really strange problem with sipgate (german sip/voip provider) ... when i make a call (or get a incoming) i can hear the other dude for about 40 seconds, and he can hear me ... but after those 40secs it seems he cannot hear me anymore |
| 14:37.06 | Nebukadneza | what could that be |
| 14:37.11 | Blackthorn | Hi all :) Hey i asked this over on the fedora channel and just not getting any responce. off topic here but thought i would give a shot here... What do I monotor for memory and processor usage? |
| 14:37.21 | Nebukadneza | theres nothing interesting (besides zap/1-1 answered etc.) in the cli |
| 14:38.15 | Ariel_ | Nebukadneza, sometimes this happens if your using sip with canreinvite=yes or iax2 with notransfer=no (Change it to canreinvite=no in sip) and notransfer=yes in iax2.conf |
| 14:38.34 | Nebukadneza | hm - i dont use iax2 in this setup, but okay |
| 14:38.37 | Nebukadneza | ill try |
| 14:38.38 | *** join/#asterisk bikokola (~root@202.67.82.37) |
| 14:38.40 | Nebukadneza | what about caninvite=? |
| 14:38.51 | Ariel_ | Nebukadneza, do you use sip setup? |
| 14:38.58 | Nebukadneza | sip setup? |
| 14:40.18 | bikokola | hey guys |
| 14:40.56 | bikokola | i was wondering, after make install of asterisk, where can i go to configure the iax and extension's files, where are they located |
| 14:41.11 | Nebukadneza | bikokola: /etc/asterisk |
| 14:41.13 | Ariel_ | then try canreinvite=no for that sip setting. |
| 14:41.14 | astoria | bikokola: /etc/asterisk |
| 14:41.22 | bikokola | thanks mate |
| 14:41.57 | Nix | Blackthorn: try using top, tload, sar, iostat and vmstat |
| 14:41.59 | Nix | ;-) |
| 14:42.04 | Nebukadneza | Ariel_: still waiting for a friend to call |
| 14:42.05 | bikokola | etc/asterisk is empty |
| 14:42.12 | bikokola | o items |
| 14:42.12 | Nebukadneza | do you use any voip provider? |
| 14:42.33 | Ariel_ | Nebukadneza, I use voip providers yes. |
| 14:42.44 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 14:42.52 | Nebukadneza | Ariel_: which one? (maybe with a deal with sipgate) |
| 14:42.58 | Nebukadneza | if yes it would be nice if you could test :P |
| 14:43.05 | astoria | bikokola: make samples |
| 14:43.09 | Ariel_ | bikokola, if this is a new setup do at the /usr/src/asterisk make samples |
| 14:43.32 | bikokola | so i just edit the sample files |
| 14:43.37 | Ariel_ | Nebukadneza, I use voipjet, nufone, race.com and voicepulse |
| 14:43.43 | bikokola | or copy and paste them into the etc/asterisk |
| 14:43.53 | Nebukadneza | Ariel_: ill check :P |
| 14:43.53 | Ariel_ | bikokola, yes to either |
| 14:44.06 | bikokola | wait, do i have to rename them |
| 14:44.47 | bikokola | all i want to use is iax and extensions, should i rename from iaxsample.conf to iax.conf and same for extensions |
| 14:44.55 | Ariel_ | bikokola, no you don't after you make samples you then go to the directory /etc/asterisk and start editing them |
| 14:45.14 | *** join/#asterisk dacleric (~dacleric@p5482A1EF.dip0.t-ipconnect.de) |
| 14:45.27 | bikokola | what exactly is involved in this "make samples" thing |
| 14:45.51 | bikokola | do i just go into the samples directory and make |
| 14:45.52 | astoria | bikokola: read the README file in /usr/src/asterisk |
| 14:46.19 | Ariel_ | bikokola, just go to the directory /usr/src/asterisk in that directory you can make samples |
| 14:46.48 | *** join/#asterisk brettnem (~Brett@207.90.232.34) |
| 14:47.08 | brettnem | hello all |
| 14:47.11 | Nebukadneza | Ariel_: argh - damned nothing supported by sipgate |
| 14:47.44 | brettnem | hey, anyone remember the story about ISPs killing voip connections to providers other than that ISP?? |
| 14:47.48 | Nebukadneza | sometimes i hate sipgate |
| 14:47.55 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 14:47.57 | Nebukadneza | btw: canreinvite=no had no effect :( |
| 14:48.02 | astoria | brettnem: yeah, there's been alot of stories about that.. |
| 14:48.08 | astoria | brettnem: the fcc made a ruling about that |
| 14:48.17 | brettnem | astoria: do you have a link to the ruling?? |
| 14:48.18 | bikokola | under my usr/src i only have one more directory called redhat, no asterisk |
| 14:48.26 | Ariel_ | yes the fcc says you can't block them here in the US |
| 14:48.40 | Ariel_ | bikokola, how did you setup asterisk? |
| 14:48.49 | brettnem | It appears that time warner may be corrupting some of my SIP messages.. |
| 14:48.49 | *** join/#asterisk grimse (~grimse@p5481BFA8.dip.t-dialin.net) |
| 14:48.56 | yaaar | can somebody confirm/deny for me that i do indeed still have to add each context to extensions.conf (so as to add the switch statement) even if i'm using realtime extensions? |
| 14:49.10 | Darwin35 | ok we have a iax issue |
| 14:49.10 | Ariel_ | Nebukadneza, during the call do sip debug and see what the cli says about the call when it drops. |
| 14:49.12 | Darwin35 | grrr |
| 14:49.18 | Nebukadneza | hm |
| 14:49.18 | Nebukadneza | kay |
| 14:49.20 | Darwin35 | its not ringing my phone |
| 14:49.25 | bikokola | i downloaded it to desktop, extracted it, went into console, typed make in the asterisk dir, followed by a make install |
| 14:49.26 | Nebukadneza | i totally forgot about sip debug |
| 14:49.28 | brettnem | does anyone have a link to that FCC ruling?? |
| 14:49.30 | Darwin35 | I can dial out from the iax phone |
| 14:49.43 | Darwin35 | but when I call it from other exten i get nothing |
| 14:49.47 | astoria | brettnem: i'm looking for it right now |
| 14:49.53 | brettnem | astoria: ah thanks! |
| 14:50.41 | Nix | http://www.voxgratia.net/blog/archives/2005/07/voip_security.html <-- excellent article |
| 14:50.47 | Ariel_ | bikokola, so your not folling any writen instructions. |
| 14:51.15 | Ariel_ | folling/following |
| 14:51.21 | astoria | brettnem: i can't find it.. it was a cable company on the east coast.. |
| 14:51.29 | brettnem | doh |
| 14:51.52 | bikokola | ariel_, i guess not |
| 14:51.56 | Darwin35 | exten => *1004,1,Dial(IAX2/1004|20|Tr) goes to fallthrew and does not ring the phone |
| 14:52.04 | astoria | brettnem: it's gotta be here somewhere http://www.fcc.gov/headlines.html |
| 14:52.24 | ManxPower | Only terrorists use the "r" option to Dial. |
| 14:52.26 | brettnem | thanks.. I'll check it out. |
| 14:52.32 | brettnem | ManxPower: heh |
| 14:52.37 | Ariel_ | ManxPower, wow |
| 14:52.38 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.active.supporter.pdpc) |
| 14:52.58 | gres | ManxPower, :) |
| 14:53.17 | *** join/#asterisk scud (~scud@12-214-190-139.client.mchsi.com) |
| 14:53.23 | Ariel_ | ~docs |
| 14:53.24 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 14:53.24 | Darwin35 | but why is it not ringing the phone |
| 14:53.36 | Ariel_ | bikokola, look at the doc's |
| 14:53.47 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 14:54.21 | bikokola | Ariel_, why, did i do something wrong the way i described my install to you |
| 14:54.24 | Ariel_ | Darwin35, your phone is not ringing or you don't hear a ring on the phone your calling from? |
| 14:54.29 | *** join/#asterisk mkrufky (~mk@68.160.103.77) |
| 14:54.36 | Darwin35 | no its not ringing the phone |
| 14:54.45 | Darwin35 | it drops to the fallthrew |
| 14:54.53 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 14:55.07 | astoria | brettnem: found it |
| 14:55.13 | Ariel_ | bikokola, desktop is where you placed the file? Well in my view yes. Since it's a server based program which should not be used with xwindows |
| 14:55.38 | astoria | http://hraunfoss.fcc.gov/edocs_public/attachmatch/DA-05-543A2.pdf |
| 14:55.47 | Darwin35 | if I dial *1004 i get a busy tone and no one is on it |
| 14:56.01 | Blackthorn | What is python ? And should I run * in any higher priorty than normal? |
| 14:56.03 | Ariel_ | Darwin35, is the phone registered? |
| 14:56.14 | Darwin35 | but i can dial out from it |
| 14:56.30 | bikokola | ye, but many of the doc's talk some trash bout cvs |
| 14:56.30 | Ariel_ | Darwin35, dialing is different the getting calls |
| 14:56.39 | bikokola | not everyone install's that way |
| 14:57.33 | Ariel_ | bikokola, you can use cvs stable as well it -rv1 if you don't use the -rv1 it gets the develelopment version which we need people to test it anyway. |
| 14:57.39 | Darwin35 | promethus*CLI> iax2 show peers |
| 14:57.39 | Darwin35 | Name/Username Host Mask Port Status |
| 14:57.39 | Darwin35 | 1004/1004 192.168.1.100 (D) 255.255.255.255 4569 Unmonitored |
| 14:58.02 | *** join/#asterisk syle (~blah@wnpgmb06dc1-167-98.dynamic.mts.net) |
| 14:58.50 | Ariel_ | Darwin35, what is it another asterisk box? or a iax softphone |
| 14:58.56 | lehel | can i simulate CAPI? |
| 14:58.59 | Nebukadneza | Ariel_: nothing besides the registering messages to sipgate in the cli with sip debug |
| 14:59.05 | Darwin35 | its a iax hard phone |
| 14:59.08 | Darwin35 | x401 |
| 14:59.15 | Darwin35 | aka netweb401\ |
| 14:59.30 | bikokola | ok, simply, has anyone here installed Asterisk and got it working, by downloading and extracting using xwindows |
| 14:59.33 | Darwin35 | connected to a asterisk box |
| 14:59.54 | astoria | what does xwindows have to do with *?? |
| 14:59.58 | Darwin35 | why would you have xwindows on a asterisk box |
| 15:00.12 | brettnem | ?!? |
| 15:00.15 | Ariel_ | Darwin35, put qualify=120 or yes |
| 15:00.20 | astoria | brettnem: did you see the link I gave you? |
| 15:00.29 | brettnem | to fcc headlines, yes.. |
| 15:00.32 | bikokola | because im running fedora, i guess im switching between console and xwindow |
| 15:00.41 | astoria | brettnem: no, i posted a link to the article. |
| 15:00.44 | brettnem | I actually have our counsel digging somethig up.. |
| 15:00.51 | brettnem | oh, I didn't see that.. |
| 15:00.58 | astoria | brettnem: http://hraunfoss.fcc.gov/edocs_public/attachmatch/DA-05-543A2.pdf |
| 15:00.58 | brettnem | ah ha |
| 15:01.08 | astoria | brettnem: thats the ruling |
| 15:01.14 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
| 15:01.17 | brettnem | cool.. thanks.. |
| 15:01.40 | brettnem | we noticed Time Warner is chaning the sip header of SIP 2.0 to something like: S23927IP 2.0 |
| 15:01.40 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 15:01.48 | Ariel_ | bikokola, ok but still you need to do more then just that. Like most of us put he files in /usr/src and also use zaptel drivers and addons and sounds files... |
| 15:01.52 | brettnem | when we change the port from 5060 to something else it works flawlessly |
| 15:02.18 | astoria | brettnem: i'm not surprised.. |
| 15:02.36 | bikokola | ye i know, but i dont want any sound files, and not using any hardware which i ned zaptel for |
| 15:02.54 | Darwin35 | still not rining it |
| 15:02.55 | bikokola | so should i extract it in usr/src |
| 15:03.07 | bikokola | then make install in usr/src |
| 15:03.08 | astoria | brettnem: the fcc ruling says that you cannot block SIP.. it says nothing about playing with latency or altering packets. |
| 15:03.12 | Darwin35 | this pisses me off |
| 15:03.29 | astoria | brettnem: i mean you cannot block VOIP |
| 15:03.33 | brettnem | astoria: isn't messng up the headers, essentially blocking it? |
| 15:03.41 | astoria | brettnem: tehcnically, no |
| 15:03.49 | Ariel_ | Darwin35, now that you put qualify what does iax2 show peers say? |
| 15:03.49 | brettnem | well it makes the packet useless |
| 15:03.58 | brettnem | in fact, the corruption actually crashes asterisk.. |
| 15:04.08 | astoria | brettnem: oh well. you're still getting the packets.. |
| 15:04.10 | brettnem | which is a bug.. |
| 15:04.19 | *** join/#asterisk file (~jcolp@mctn1-6719.nb.aliant.net) |
| 15:04.40 | brettnem | it says, "Madison River shall not block ports used for VoIP applications or otherwise prevent |
| 15:04.40 | brettnem | customers from using VoIP applications." |
| 15:05.12 | astoria | brettnem: it doesn't say that they have to gaurantee anything though.. |
| 15:05.21 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 15:05.27 | astoria | brettnem: comcast around her puts horribly latency on voip packets that aren't theirs. |
| 15:05.33 | brettnem | yes, but they are directly doing something to prevent customers from using VoIP applications.. |
| 15:05.43 | brettnem | all subject to interpretation of couse.. |
| 15:05.52 | brettnem | this looks more like a settlement than a ruling. |
| 15:05.58 | *** join/#asterisk abatista (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:07.09 | bikokola | Ariel_:should i download and extract in usr/src, then make intsall in there |
| 15:07.15 | bikokola | ok, then |
| 15:07.18 | bikokola | any1 |
| 15:07.34 | bikokola | Ariel_:should i download and extract in usr/src, then make intsall in there |
| 15:08.27 | ariel_ | bikokola, in my view you should follow the doc's and make your directorys and download the zaptel asterisk asterisk-addons asterisk-sounds into the /usr/src then work from there. |
| 15:08.34 | ariel_ | network split. |
| 15:08.59 | ChkDigit | Now, that was a lot of quits! |
| 15:09.06 | ChkDigit | Was it something I said? |
| 15:09.17 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 15:09.52 | [TK]D-Fender | b00m |
| 15:10.17 | [TK]D-Fender | All ph34r the net-split |
| 15:10.33 | lehel | anyone knows that i could use my ISDN Fritz! as a modem? |
| 15:10.42 | bikokola | its probably 53 guys, sharing the same comp |
| 15:10.53 | *** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no) [NETSPLIT VICTIM] |
| 15:11.13 | *** part/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 15:11.45 | *** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net) [NETSPLIT VICTIM] |
| 15:11.46 | ariel_ | bikokola, if you set the asterisk up like the doc's then more of us can help you with problems due to it's installed like most of us use it. |
| 15:11.56 | Darwin35 | wow that was fun |
| 15:12.07 | Qwell | Darwin35: Thats what you get |
| 15:12.08 | bikokola | kk, will do |
| 15:13.08 | *** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au) [NETSPLIT VICTIM] |
| 15:13.13 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM] |
| 15:13.13 | *** mode/#asterisk [+o twisted] by ChanServ |
| 15:13.18 | Darwin35 | what would make it unreachable |
| 15:13.31 | *** join/#asterisk Delvar (~irc@host-83-146-53-34.bulldogdsl.com) [NETSPLIT VICTIM] |
| 15:13.50 | *** join/#asterisk ReVoLvA (~revolva@host217-44-6-88.range217-44.btcentralplus.com) |
| 15:14.00 | ariel_ | Darwin35, the phone could have dnd on or many other reasons. |
| 15:14.16 | *** join/#asterisk santiago (~santiago@63.245.86.141) |
| 15:14.25 | Darwin35 | I did not set it |
| 15:14.27 | *** part/#asterisk Henguei (~Henguei@196.203.53.45) |
| 15:14.42 | *** join/#asterisk xylome (~asterisk@hg-msq-hol.levigo.de) [NETSPLIT VICTIM] |
| 15:14.51 | ReVoLvA | Does anyone have SIP firmware for cisco 7940 or 7960 phones ? i've spent weeks looking for it :( |
| 15:15.13 | Hmmhesays | weeks? |
| 15:15.15 | Hmmhesays | geebus dude |
| 15:15.19 | ariel_ | ReVoLvA, you need to buy it via the smartnet |
| 15:15.29 | ReVoLvA | i was hoping to avoid it |
| 15:15.33 | *** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net) |
| 15:15.37 | *** join/#asterisk Corydon76-home (eleven@Corydon76-home.sustaining.supporter.pdpc) [NETSPLIT VICTIM] |
| 15:15.38 | Hmmhesays | one word p2p |
| 15:15.39 | ReVoLvA | but it looks like i'm gonna have to pay for smartnet |
| 15:15.48 | *** join/#asterisk nDuff (~cduffy@fwext1-ext.isgenesis.com) [NETSPLIT VICTIM] |
| 15:15.49 | lehel | isdn fritz as a modem?.. |
| 15:16.03 | Hmmhesays | You sound like a cop trying to bait one of us ReVoLvA |
| 15:16.08 | ReVoLvA | lol |
| 15:16.09 | *** join/#asterisk Tili (~Tili@202-133-65-9-dialup.sat.net.pk) |
| 15:16.10 | ReVoLvA | relax |
| 15:16.14 | bsdfreak | heh |
| 15:16.16 | astoria | try not buying ciscos next time. |
| 15:16.18 | ReVoLvA | just gimmie the firmware and i be gone :) |
| 15:16.23 | brettnem | it's the cisco police |
| 15:16.24 | Qwell | ReVoLvA: That is exactly the case. You have to pay for it |
| 15:16.29 | ReVoLvA | guu :( |
| 15:16.32 | *** join/#asterisk jonathh (~asd@host217-46-145-65.in-addr.btopenworld.com) [NETSPLIT VICTIM] |
| 15:16.37 | brettnem | how much is that firmware going for these days?? |
| 15:16.44 | Hmmhesays | 5 bucks |
| 15:16.48 | Qwell | brettnem: $6.50 |
| 15:16.50 | brettnem | nice |
| 15:16.50 | ReVoLvA | eh ? |
| 15:16.59 | DarthClue | ReVoLvA: we can't help you with that. but if you found us, you can find the firmware. |
| 15:17.00 | ReVoLvA | i been quoted 55 buck |
| 15:17.04 | brettnem | do I hear $4.50?? $4.50 anyone? |
| 15:17.06 | jalsot | hi |
| 15:17.08 | Hmmhesays | if you are going from sccp it takes two updates to make it to the latest firmware, IIRC |
| 15:17.19 | ReVoLvA | yeah its sccp |
| 15:17.30 | Hmmhesays | have you not tried p2p? |
| 15:17.33 | brettnem | why does thunderbird sometimes suck |
| 15:17.46 | Zeeek | funny I just nopw upgraded it to 1.0.6 |
| 15:17.47 | ariel_ | last I saw the lisc for sip was something like 99 dollars and smartnet 9 |
| 15:17.48 | ReVoLvA | yeah, i'm not big on p2p.. tried limewire |
| 15:17.54 | ReVoLvA | but nothing |
| 15:17.57 | Hmmhesays | emule. |
| 15:18.04 | *** join/#asterisk gravemind (~omgwtfbbq@GXTi.developer.freenode) [NETSPLIT VICTIM] |
| 15:18.14 | jalsot | is it a problem when 2 TE110P cards are getting the same IRQ? |
| 15:18.17 | brettnem | I think I'm runing 1.0 |
| 15:18.23 | ReVoLvA | ok Hmmhesays will give it a go |
| 15:18.29 | Hmmhesays | and you will succeed |
| 15:18.32 | Zeeek | some progress made since then |
| 15:18.35 | ariel_ | jalsot, yes |
| 15:18.40 | brettnem | entrapment... |
| 15:18.48 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) [NETSPLIT VICTIM] |
| 15:18.53 | jalsot | ariel_: how could I fix that? I cannot find anything in BIOS |
| 15:19.24 | Hmmhesays | v5 then v7 unless you want to pull your hair out |
| 15:19.25 | ariel_ | jalsot, well move cards around to different slots. Disable things you don't need in bios like apic |
| 15:19.40 | Qwell | Hmmhesays: I did 3.2 > 7.3 |
| 15:19.50 | Qwell | universal loader...oh yeah |
| 15:19.58 | Hmmhesays | heh, nice |
| 15:20.01 | brettnem | try buying a card that isn't such a resource hog |
| 15:20.08 | brettnem | oops.. did I say that out loud? |
| 15:20.14 | Hmmhesays | lol |
| 15:20.27 | brettnem | I'm done with digium.. I think.. |
| 15:20.52 | jalsot | ariel_: I tryed in another slot, things which are not needed are already disabled |
| 15:20.55 | Hmmhesays | hopefully I'll have a sangoma card on the way soon, I want to try them out |
| 15:21.05 | brettnem | Hmmhesays: I hear good things.. |
| 15:21.12 | Hmmhesays | as do I |
| 15:21.40 | jalsot | supermicro X6DH8-XB motherboard.. |
| 15:21.53 | ariel_ | jalsot, well try it out see how it works. Unless you get a different system to put it on. Call Supermicro then and ask them? |
| 15:22.05 | brettnem | what was bkw_'s very non-PC description of the digium cards??? I think, "They are like a fat kid with a bar of chocolate" |
| 15:22.28 | jalsot | ariel_: thanks for suggestions! I'll call them ;) |
| 15:22.38 | Hmmhesays | none pc? in here? |
| 15:22.41 | Hmmhesays | ha! |
| 15:22.41 | *** join/#asterisk klictel (~klictel@207.107.208.140) |
| 15:22.45 | klictel | hi all |
| 15:23.09 | DarthClue | Hmmhesays: maybe you'll get lucky and win one at the thing that shall not be named. |
| 15:23.13 | brettnem | so what's the secret to upgrading the digium firmware for the zap cards? |
| 15:23.26 | zoo | we have go a problem: -- Extension 's' in context 'calls' from '30690116' does not exist. |
| 15:23.27 | Hmmhesays | you get my dinero DarthClue |
| 15:23.35 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) [NETSPLIT VICTIM] |
| 15:23.39 | zoo | where 30690116 is a MSN on an isdn BRI |
| 15:23.46 | Qwell | DarthClue: any idea if bkw_ got a response on his ticket to Qwest about my DID? ;/ |
| 15:23.53 | bikokola | ok, i finally installed, without changing anything, how do i call 1000 to test |
| 15:23.54 | zoo | we just want to dial out trough iax |
| 15:23.55 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 15:24.02 | *** join/#asterisk essobi (kstone@75.137.26.216.host.teledvance.com) |
| 15:24.15 | zoo | why do we need a "s"? |
| 15:24.36 | DarthClue | Qwell: not sure, are we porting it over to us? if so, who from? |
| 15:24.36 | zoo | [calls] is our outgoing context |
| 15:24.57 | Qwell | DarthClue: no port, just b0rked |
| 15:25.23 | Qwell | I can wait...I'm just anxious to test |
| 15:25.31 | brettnem | hey, is the "new digium firmware" for their zaptel cards just part of the zaptel drivers? or is there a firmware upgrade procedure.. anyone? |
| 15:25.37 | DarthClue | Qwell: just pm him, he'll respond if he can. |
| 15:25.52 | brettnem | DarthClue: I thought you were a bot? ;) |
| 15:25.59 | Qwell | DarthClue: I'll be at work. It wasn't that important. Saw you, thought I'd ask really quick |
| 15:26.05 | DarthClue | brettnem: i think the new cards have flashable firmware, but i don't think anything has been said about how to do it. |
| 15:26.14 | brettnem | oh fantastic.. |
| 15:26.24 | DarthClue | brettnem: i am, do you have a problem with bots? |
| 15:26.34 | lathos42 | bots are people too |
| 15:26.45 | DarthClue | Qwell: so pm him, tell him you are headed out and he can just update you when he gets a chance. |
| 15:26.49 | ReVoLvA | Hmmhesays: emule you beauty... got it :D |
| 15:26.57 | ReVoLvA | cheers |
| 15:26.58 | brettnem | no.. in fact, I have lots of friends who are bots.. |
| 15:27.04 | Qwell | DarthClue: I'll ask him tonight. It'll be a short day |
| 15:27.07 | Hmmhesays | ReVoLvA: yes it has everything |
| 15:27.09 | Qwell | no worries |
| 15:28.02 | DarthClue | lathos42: actually, bots are just a few thousand lines of code designed to emulate a responsive person. of course, my code is corrupt and i have an attitude problem when i get attacked for answering questions. ph34r the bot. |
| 15:28.41 | brettnem | I don't know DarthClue, you don't seem very botty to me |
| 15:28.51 | bikokola | guys, how can i dial a number to test my asterisk box |
| 15:29.04 | brettnem | hey, there's a loaded question |
| 15:29.07 | jalsot | ariel_: can it be a problem if those cards are on different PCI buses? this mobo has 5 PCI-X buses [2xGb LAN on one and 4 buses for 6 PCI slots] |
| 15:29.07 | bikokola | do i need a softphone |
| 15:29.14 | DarthClue | bikokola: define dial a number? you want to call into it or out of it or what? and have you read the wiki? |
| 15:29.27 | bikokola | ye, i read it, and completed my install |
| 15:29.37 | lathos42 | DarthClue: Well, I hope you dont flip out and start killing everyone at that thing that shall not be named :) |
| 15:30.00 | bikokola | there is a line telling me to test if it alls correct by dialling 1000 the defaulkt test, i want to know how to dial |
| 15:30.03 | ariel_ | jalsot, some pci-x are dual type of bus. But that question should be asked of the tech's at supermicro |
| 15:30.04 | DarthClue | brettnem: i have an advanced ai engine that allows me to simulate and underpaid, overworked member of The Empire |
| 15:30.12 | *** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) |
| 15:30.18 | DarthClue | bikokola: then you'll need a phone |
| 15:30.21 | Darwin35 | ok for iaxphones do you have to have 2 settings in the iax.conf |
| 15:30.36 | bikokola | can i use softphon |
| 15:30.44 | *** join/#asterisk Pfhorge (~user@rrcs-24-172-161-65.central.biz.rr.com) |
| 15:30.50 | ariel_ | Darwin35, no you can use type=friend |
| 15:30.54 | *** join/#asterisk loud (~roots@cypher.punk.net) [NETSPLIT VICTIM] |
| 15:30.59 | jalsot | ariel_: ok, so not a known case... |
| 15:31.00 | ariel_ | bikokola, yes like xlite |
| 15:31.03 | Darwin35 | ok then somethign is wrong |
| 15:31.08 | jalsot | thanks |
| 15:31.08 | DarthClue | lathos42: no, might remove a few limbs if people don't read the wiki but other than that things should be fine. |
| 15:31.33 | DarthClue | bikokola: or iaxcomm, or diax, or...well something |
| 15:31.38 | bikokola | this may sound really dumb, how do i know my host ip |
| 15:31.42 | bikokola | like my own ip |
| 15:31.45 | bikokola | to dial into |
| 15:31.56 | Qwell | oh boy |
| 15:31.58 | Pfhorge | Is there a way to make my hold music quieter? I changed quietmp3->mp3, but that didn't make enough difference |
| 15:32.35 | brettnem | grr SBC |
| 15:32.47 | lathos42 | DarthClue: Ok, I'd hate for someone to try to explain to my wife how I was killed by a IRC bot with an attitude when I attend a certain unnamed event next week |
| 15:32.57 | bikokola | lol, i shouldve keep my mouth shut |
| 15:33.00 | Qwell | lathos42: You mean cluecon? |
| 15:33.02 | Qwell | ~cluecon |
| 15:33.02 | jbot | it has been said that cluecon is http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Mark Spencer, Greg Boehnlein, Ken Rice, Brian West, Vikrant Mathur, Craig Southeren, David Sugar, Bob Andreasen, Joshua Colp, Greg Merriweather, Brian Fertig, Peter Nixon, Marc Olivier Chouinard, and Anthony Minessale II. |
| 15:33.14 | *** join/#asterisk Bile_One (~bile_one@adsl-208-191-252-109.dsl.ltrkar.swbell.net) |
| 15:33.33 | Zeeek | did anyone ever solve the problem abou calling GSM cellphones where Dial assumes an immediate answer? |
| 15:33.43 | DarthClue | lathos42: we'll send someone in uniform. no explanation will be needed. |
| 15:34.10 | DarthClue | Zeeek: i don't think that a resolution has been found. |
| 15:34.27 | lathos42 | DarthClue: Ok, thanks.. I just wanted to make sure everything was in order |
| 15:34.36 | brettnem | I'm interested in hearing about what David Sugar has to say at the conference.. any ideas?? |
| 15:34.41 | Pfhorge | anybody know how to make hold music quieter? Mine is clipping like crazy |
| 15:34.41 | bikokola | guys, how do i find my ip, i dont want to read anymore guides , and would rather a reply |
| 15:34.44 | DarthClue | lathos42: yeah, we'll have emts standing by and riot police as well. |
| 15:35.04 | DarthClue | bikokola: go to the box that * is running on and type ifconfig |
| 15:35.10 | brettnem | wow |
| 15:35.18 | bikokola | thx |
| 15:35.29 | bikokola | u can all stop laughin now |
| 15:35.31 | bsdfreak | booo |
| 15:35.31 | DarthClue | brettnem: we can't actually reveal anything. you'll have be there to find out. |
| 15:35.44 | brettnem | wish I could go. |
| 15:35.52 | Zeeek | DarthClue maybe it resides in some kind if (human_answer() ) ??? |
| 15:36.29 | ariel_ | DarthClue, so I am going to be missing allot. argh can't afford the plane ticket nor the entry fee's. |
| 15:36.31 | Pfhorge | hold music volume? anybody? |
| 15:37.03 | Zeeek | too bad we can't go back and tell the people that designed the phone system all we know now! Like "all answering machines should emit a certain frequency tone before beginning" |
| 15:37.05 | DarthClue | Zeeek: it is probably on the other end, i know that when i make calls to land lines when going thru another pbx it does the same thing. |
| 15:37.12 | MikeJ[Laptop] | brettnem, Mr Sugar replied: "I would be happy to come and speak at this event. Actually it also fits in well with the schedule for introducing a second generation (bayonne2) server." |
| 15:37.12 | brettnem | Pfhorge: /etc/asterisk/musiconhold.conf |
| 15:37.22 | brettnem | MikeJ[Laptop]: right... |
| 15:37.34 | MikeJ[Laptop] | about that ^^ |
| 15:37.37 | Zeeek | DarthClue makes me wo,der, are we paying for the ring time? I'll have to check |
| 15:37.49 | Zeeek | because it's like 40c/min here |
| 15:37.53 | Pfhorge | brettnem: got that far. I tried changing quietmp3->mp3 for default, but that didn't help enough |
| 15:38.12 | Zeeek | Pfhorge put quieter music files |
| 15:38.20 | *** join/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu) |
| 15:38.32 | Zeeek | Pfhorge you have windows available by any chance? |
| 15:38.33 | anthm | resample them all to .raw at the exact vol you want and use native moh the you will give your cpu a break too |
| 15:38.41 | DarthClue | Zeeek: could be. depends on where the answer is actually happening. if the telco has a box dedicated to making gsm calls and all gsm calls are terminated to that box, then you may be paying from the moment that the telco box picks up. |
| 15:38.42 | ariel_ | Pfhorge, use sox to resample them |
| 15:38.45 | Zeeek | there are several free mp3 level changers |
| 15:38.48 | Pfhorge | I was afraid of that. You're ruining my lazy streak |
| 15:39.01 | Pfhorge | mm, sox. That I can do |
| 15:39.06 | DarthClue | Pfhorge: The Empire expects hard working bots. Now get to it. |
| 15:39.12 | Pfhorge | oh noes! |
| 15:39.28 | *** join/#asterisk loud (~roots@cypher.punk.net) [NETSPLIT VICTIM] |
| 15:39.28 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) [NETSPLIT VICTIM] |
| 15:39.57 | Darwin35 | http://pastebin.ca/18706 there is my aix debug |
| 15:40.01 | Darwin35 | iax |
| 15:40.06 | anthm | most mp3 are like 50 times better quality than what can be played to the channel so you need to use the cpu to downsample it live over and over |
| 15:40.26 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 15:40.32 | lehel | does anyone knows how to check if i have the right isdn-line plugged into my Fritz? |
| 15:40.44 | *** join/#asterisk Dovid (~dovi5988@pool-138-89-154-217.mad.east.verizon.net) |
| 15:40.50 | anthm | when you could decode it to a .raw once at the exact quailty you need and it would just be a matter of passiing audio from the disk to the pipe |
| 15:41.15 | Pfhorge | I'm just using the mp3s that come with * to test with atm |
| 15:41.32 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 15:42.04 | ariel_ | Darwin35, your phone is sending this back to the asterisk box. We are busy |
| 15:42.59 | Darwin35 | no one is on that phone |
| 15:42.59 | brettnem | anthm: are there any devices that use raw mp3 format for both the channel and the endpoint? |
| 15:43.30 | anthm | elaborate, i'm not sure what you mean |
| 15:43.43 | brettnem | I have a friend who wants to setup asterisk as a home intercom system (like the old fashioned ones) but wants to be able to listen to radio and mp3s through a room speaker at high quality.. |
| 15:44.13 | brettnem | ie: > 4khz |
| 15:44.22 | *** part/#asterisk Pfhorge (~user@rrcs-24-172-161-65.central.biz.rr.com) |
| 15:44.48 | ManxPower | brettnem: he would have to find a phone that supported wideband codecs |
| 15:45.06 | RaYmAn-Bx | why bother sending it over asterisk at all then? Why not just use either access over nfs or whatever or broadcast with some program? |
| 15:45.09 | bkw_ | you mean 8khz |
| 15:45.13 | anthm | so like play music and bust in via phone to talk like virtual k-mart? |
| 15:45.26 | bkw_ | haha |
| 15:45.46 | brettnem | bkw_: actually the audio is 4khz, by nyquest says you have to sample that at 2x the freq |
| 15:45.52 | *** join/#asterisk Saaib (~nabudocon@200.76.231.14) |
| 15:46.08 | ariel_ | Darwin35, do does it have do not disturb setting on it any where? |
| 15:46.19 | brettnem | anthm: actually, no real phone capabilities are really needed.. He wants to hit buttons and listen to muzak |
| 15:46.20 | Darwin35 | nope |
| 15:46.30 | Darwin35 | nothing is set its in very basic mode |
| 15:46.31 | brettnem | like the old fashioned home intercoms that played raidio.. |
| 15:46.40 | brettnem | heh |
| 15:47.32 | brettnem | actually.. to be more accurate, I think voice audio is actually 3.1khz, but rounded up to 4khz by convention.. but I'm rusty there.. |
| 15:48.29 | bkw_ | brettnem, just use shoutcast |
| 15:48.48 | brettnem | bkw_: but still need an endpoint and a channel that supports mp3 |
| 15:49.04 | ManxPower | brettnem: It's a hopeless project. |
| 15:49.21 | ManxPower | since no channels support wideband |
| 15:49.21 | brettnem | or some other technology..shoutcast would require a PC at each location.. we're hoping for a wall mounted display.. or maybe a PC with like 15 independent sound outputs. |
| 15:49.27 | DarthClue | brettnem: just send anthm a blank check, he'll get what you need. |
| 15:49.27 | brettnem | Noooooooooo! |
| 15:49.38 | brettnem | mmmm spam |
| 15:49.39 | bkw_ | embeded players |
| 15:49.43 | *** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org) |
| 15:49.48 | essobi | Ayup. |
| 15:49.49 | bkw_ | it don't take much to play an mp3 |
| 15:49.55 | essobi | 15 gumsticks.. |
| 15:50.00 | Darwin35 | like mp3play |
| 15:50.01 | bkw_ | and sure as hell takes less to do a network connection |
| 15:50.06 | brettnem | bkw_: you know of some cheap ethernet based embedded players? |
| 15:50.08 | essobi | Can a gumstick even play an mp3? |
| 15:50.20 | bkw_ | brettnem, i'm sure If I tried I could find one |
| 15:50.26 | essobi | *COUGH*RESEARCHGUMSTICKS*COUGH* |
| 15:50.35 | brettnem | oh hell I'm fantasticaly off topic sorry guys.. ;) |
| 15:50.45 | opus_ | hi |
| 15:50.57 | essobi | seriously.. intel embedded linux with network stack |
| 15:51.03 | essobi | gumstick |
| 15:51.05 | essobi | go read it |
| 15:51.13 | brettnem | hmm ok, I'll check it out.. |
| 15:51.30 | brettnem | embeded soundcard too?? :) |
| 15:51.49 | essobi | I think it's usb |
| 15:51.56 | essobi | and you can get USB sound cards. |
| 15:51.59 | essobi | ifnot.. |
| 15:52.05 | brettnem | bad ass... |
| 15:52.08 | essobi | look at the mini-itx formfactors |
| 15:52.10 | brettnem | http://www.linuxdevices.com/news/NS3112296807.html |
| 15:52.15 | essobi | like 80 bucks per machine |
| 15:52.26 | brettnem | damn, very cool |
| 15:52.29 | essobi | with sound and video 4x5" or so.. |
| 15:52.36 | essobi | but gumsticks are REALLY small. |
| 15:52.55 | *** join/#asterisk kingtux (~susekid@pool-141-157-9-215.balt.east.verizon.net) |
| 15:53.01 | brettnem | hmmm.. we'd need some sort of input device.. |
| 15:53.20 | Darwin35 | 1004/1004 192.168.1.100 (S) 255.255.255.255 4569 UNREACHABLE |
| 15:53.40 | Qwell | bkw_: y0 |
| 15:54.05 | essobi | brettnem Input for what? |
| 15:54.19 | brettnem | to select a song and such need like a LCD and some buttons |
| 15:54.27 | essobi | Oh. |
| 15:54.28 | Darwin35 | logon failed now |
| 15:54.32 | MikeJ[Laptop] | jhonny 5 say: innnppuuuutttt |
| 15:54.40 | brettnem | exactally |
| 15:54.44 | essobi | INNNNNNNNNNNPUUUUUUUUUUUT |
| 15:54.45 | DarthClue | input you freak, input! |
| 15:54.54 | essobi | Uhh.. dell makes a touch screen |
| 15:54.59 | essobi | kinda nice lcd |
| 15:55.12 | brettnem | need cheap input |
| 15:55.17 | essobi | think it just equates to an usb mouse |
| 15:55.18 | essobi | hah |
| 15:55.22 | essobi | touch and cheap? |
| 15:55.25 | essobi | good luck. |
| 15:55.28 | brettnem | button and cheap |
| 15:55.32 | MikeJ[Laptop] | I'm cheap |
| 15:55.38 | essobi | You need like a 4x20 LCD USB driven |
| 15:55.44 | brettnem | ok MikeJ[Laptop]: I'll take 15 then |
| 15:55.44 | essobi | and a keypad |
| 15:55.54 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc0fi.dialup.mindspring.com) |
| 15:56.00 | yaaar | what's the difference between 'nreinvite' and 'canreinvite' ? sip.conf wiki page makes no mention of the first one, but i see it in some example configs... |
| 15:56.01 | essobi | and a hacked up mounting box made for all of them. |
| 15:56.02 | MikeJ[Laptop] | no.. I mean I'm cheap.. I don't like to spend money |
| 15:56.14 | brettnem | nreinvite is a typo |
| 15:56.22 | essobi | Did I meantion my engineering fee is $65 an hour? :) |
| 15:56.22 | yaaar | ah |
| 15:56.23 | MikeJ[Laptop] | yaaar. the ca at the begining |
| 15:56.27 | brettnem | x10 site is kinda flashy eh? |
| 15:56.33 | yaaar | MikeJ[Laptop]: hardehar |
| 15:56.46 | Qwell | brettnem: Do you not remember their flashy popup ads for 2 years? |
| 15:56.47 | MikeJ[Laptop] | its missing on nreinvite |
| 15:56.54 | brettnem | yesssss |
| 15:56.55 | MikeJ[Laptop] | no.. really |
| 15:57.01 | brettnem | that was so annoying |
| 16:01.06 | *** part/#asterisk lehel (~Lehel@82.79.20.17) |
| 16:02.28 | SuPrSluG | hello |
| 16:03.51 | Bile_One | anyone have any experience with a Leadtek vba 8055? |
| 16:04.28 | SuPrSluG | i have 2 x100p's in a server. 1 is ok, the other is in red alarm state, although i can receive calls on this card. so, why can i receive calls but not send? |
| 16:05.14 | gtigene | Does anyone know if the new Asterisk book (Asterisk: the future of telephony) talks about AMP? |
| 16:06.32 | Darwin35 | Jul 28 11:06:02 NOTICE[92646]: chan_iax2.c:7610 iax2_poke_noanswer: Peer '1004' is now UNREACHABLE! Time: 0 |
| 16:08.59 | syle | Jul 27 22:20:48 NOTICE[32671]: chan_sip.c:8343 handle_response_peerpoke: Peer '2044803382' is now REACHABLE! (3499ms / 5000ms) |
| 16:08.59 | *** join/#asterisk juice (~juice@mo-67-77-176-124.dyn.sprint-hsd.net) |
| 16:08.59 | syle | Jul 27 23:20:58 NOTICE[32671]: chan_sip.c:9900 sip_poke_noanswer: Peer '2044803382' is now UNREACHABLE! Last qualify: 85 |
| 16:09.00 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 16:09.09 | syle | this happens every hour it seems |
| 16:09.36 | astoria | syle: this means that you are using nat=yes |
| 16:09.43 | syle | yes |
| 16:10.04 | astoria | and your polling for it. |
| 16:10.25 | astoria | do you have qualify=yes too? |
| 16:10.35 | *** join/#asterisk lehel (~Lehel@82.79.20.17) |
| 16:10.37 | syle | yes look at what i pasted qualify=5000 |
| 16:10.41 | astoria | see: http://www.voip-info.org/tiki-index.php?page=asterisk+sip+qualify |
| 16:11.43 | syle | well i understand that, my question is how come is says unreachable every hour on the dot at times |
| 16:11.56 | astoria | beats me, just sounds like a coincidence |
| 16:12.08 | Zeeek | putting qualify=n sometimes makes iaxphones unreachable |
| 16:12.10 | astoria | does your network have some kind of cron thing that ties stuff up? |
| 16:12.19 | Zeeek | is this a phone? |
| 16:12.28 | syle | its a pap2-na yeah |
| 16:12.49 | Zeeek | well several phones I tested on LAN became unreachable as soon as qualify was used |
| 16:13.21 | syle | why would you use qualify on a lan |
| 16:13.27 | syle | theres no nay |
| 16:13.29 | syle | nat |
| 16:13.46 | Zeeek | curiousity |
| 16:13.56 | Zeeek | I expecetd to see 1ms or something |
| 16:14.25 | syle | you set nat to yes then if using qualify i hope hehe |
| 16:15.29 | opus_ | weird. try setting quailify to a large amount |
| 16:15.34 | opus_ | like 5000 |
| 16:16.04 | syle | lol |
| 16:16.05 | opus_ | what version of asterisk are you using Zeeek? |
| 16:16.45 | syle | i;m wondering if i set qualify to something insane like 10 seconds if all problems will be solved |
| 16:17.16 | Zeeek | 1.0.6 |
| 16:17.32 | ManxPower | Zeeek: Qualify measures the response to a SIP OPTIONS packet (or the IAX2 equiv), NOT round trip travel time of packets |
| 16:17.39 | Zeeek | no it's the phones. They don't like qualify. Some providers don't either, for example ICH (SIP) |
| 16:17.57 | ManxPower | So if a device does not respond to OPTIONS or the device is busy and doesn't respond quickly..... |
| 16:17.57 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:18.01 | opus_ | Zeeek what type of network gear do you have |
| 16:18.04 | Zeeek | ManxPower, whatever, thepoint is, in some situations it breaks connectivity |
| 16:18.12 | ManxPower | Zeeek: *nod* |
| 16:18.21 | Zeeek | if the device or provider doesn't respond correctly |
| 16:18.43 | Zeeek | opus it's more a matter of thephones, Chinese IAX phones |
| 16:18.43 | opus_ | i'd try cvs stable as well |
| 16:19.03 | opus_ | zeeek - it could be because of your network gear |
| 16:19.09 | Zeeek | I'll wait and get 1.2 when it's ready. If I'm gonna go thru the pain of an upgrade, may as well make it worth it :) |
| 16:19.35 | Zeeek | opus most everything works except these phones and that one provider, ICH |
| 16:19.47 | ManxPower | My new .sig: Only terrorists use the "r" option to Dial. |
| 16:19.49 | Zeeek | when I can I use qualify=300 or 500 depending on the distance |
| 16:20.15 | Zeeek | My main interest in 1.2 will be the freedom for # tyranny |
| 16:20.36 | Zeeek | "password?" nnnn# "transfer!" |
| 16:20.38 | Darwin35 | Jul 28 11:19:59 NOTICE[93533]: chan_iax2.c:7610 iax2_poke_noanswer: Peer '1004' is now UNREACHABLE! Time: 0 this is all I get when the phone tries to register |
| 16:20.51 | Zeeek | what phone? |
| 16:21.38 | Darwin35 | netweb401 |
| 16:21.39 | Darwin35 | iax2 |
| 16:21.39 | Zeeek | I'M TELLIN YA it won't work! |
| 16:21.39 | Darwin35 | ? |
| 16:21.39 | Zeeek | it doesn't answer the OPTIONS message properly |
| 16:21.39 | Zeeek | I've been saying that for a few minutes now |
| 16:21.56 | ManxPower | Darwin35: Chineese IAX2 phone? |
| 16:21.56 | bkw_ | well current CVS-HEAD leaks memory like mad boys and girls |
| 16:22.14 | Darwin35 | no the eezeephne |
| 16:22.21 | ManxPower | The Chineese IAX2 phones don't support qualify=, last I heard. |
| 16:22.24 | Zeeek | that is a cheap cjinese phone |
| 16:22.27 | *** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 16:22.33 | ManxPower | I thought eezephone is a Chinese IAX2 phone. |
| 16:22.39 | brettnem | bkw_: I'm tackling this file descriptor leak arghhhh |
| 16:22.40 | Zeeek | eezephone is exacly the ones I have |
| 16:22.47 | Zeeek | who's on first? |
| 16:22.55 | Craziman2 | Any ideas why music on hold works if a call comes in sip but not if it comes in IAX? |
| 16:23.04 | Zeeek | eezeephone are ALL cheap chinese phones :) |
| 16:23.08 | bkw_ | brettnem, in what? |
| 16:23.17 | Darwin35 | so explain what has to be fixed in the firmware so i can report it |
| 16:23.22 | brettnem | bkw_: malformed SIP headers leave sip channels hung |
| 16:23.28 | bkw_ | loverly |
| 16:23.33 | ManxPower | Darwin35: IAX2 or SIP? |
| 16:23.36 | brettnem | bkw_: they just build up until the system is out of fds |
| 16:23.37 | Darwin35 | iax2 |
| 16:23.52 | ManxPower | I *think* it's the IAX2 POKE command |
| 16:23.58 | Darwin35 | for the netweb401 to work right' |
| 16:24.07 | brettnem | bkw_: seems that time warner is modifing SIP headers from "SIP 2.0" to something like "S2637267IP 2.0" |
| 16:24.16 | Darwin35 | what does it have to return |
| 16:24.16 | ManxPower | I don't use statically configured IAX2 peers anymore, so I can't test it, but iax2 debug should give you a good idea. |
| 16:24.20 | opus_ | i can hear F1 race cars, 7 miles away |
| 16:24.39 | Darwin35 | I posted a debug earlier not alot of info |
| 16:24.54 | Zeeek | Darwin35 just remove the qualify and it works |
| 16:24.56 | tclark | Zeeek: do you have any of the pa168v based ata's ? |
| 16:25.02 | ManxPower | brettnem: Sure it's not your firewall? |
| 16:25.04 | Zeeek | no, only three phones |
| 16:25.22 | brettnem | ManxPower: it's just one of those home firewall things.. when I change the port from 5060 to something else, it works fine |
| 16:25.43 | tclark | i think the iax firmware is the same, do you have any issue with the iax based phones ? |
| 16:25.44 | ManxPower | brettnem: *nod* Could be the "home firewall thing". |
| 16:25.48 | Zeeek | Darwin35 by the way if you hgave access to Windows, the tool they provide can unlock the phone if it locks up |
| 16:26.09 | brettnem | ManxPower: why would that change the message content only 1 out of every 6 messages only on that port?? seems weird.. |
| 16:26.14 | Zeeek | they cleverly made it revert to 192.168.1.100 after two resets |
| 16:26.18 | brettnem | but we should change out that firewall and test.. |
| 16:26.51 | ManxPower | brettnem: because I've never seen a SIP aware NAT router that did things right. |
| 16:26.53 | Zeeek | I had one of theose newtweb phones at astricon Madrid. Worked as soon as I plugged it in |
| 16:26.56 | Hmmhesays | ooh323 is irritating me |
| 16:27.02 | ManxPower | I always disable the SIP awareness on the router. |
| 16:27.11 | brettnem | ManxPower: right, but to alter the header is a weird thing.. it's NOT sip aware.. |
| 16:27.21 | Zeeek | tclark other than the qualify, there are a few quirks |
| 16:27.25 | Darwin35 | the problem is I can dial out from it |
| 16:27.27 | ManxPower | brettnem: It COULD be the ISP, but they would be very unusual. |
| 16:27.35 | tclark | what other specfic quirks ? |
| 16:27.37 | Darwin35 | but its not ringing when people call in |
| 16:27.45 | Zeeek | Darwin35 confrim that you removed the qualify ? |
| 16:27.56 | Darwin35 | I did |
| 16:28.06 | Zeeek | and it still is unreachable? |
| 16:28.11 | brettnem | ManxPower: right.. seems weird to me as well.. although we heard in a hearing time warner mention that they were definately going to block ports on their internet service.. |
| 16:28.12 | Darwin35 | 1004 192.168.1.100 (D) 255.255.255.255 4569 Unmonitored |
| 16:28.22 | Zeeek | and is it registered, |
| 16:28.24 | Zeeek | ? |
| 16:28.40 | Darwin35 | I can make calls |
| 16:28.57 | Zeeek | tclark such as? |
| 16:29.00 | Darwin35 | but under iax2 show registry it doe not how is |
| 16:29.26 | alt | ManxPower: well. in light of the fact that where I live both major ISPs are somehow involved in telecom (Telus _is_ the ILEC and Shaw Cable is rolling out phone services), I wouldn't be at all surprised for it to happen where I like (time to SSL those SIP Sessions ppl) |
| 16:29.26 | tclark | for the pa168 pstn/iax switching & callerid from the iax calls |
| 16:29.39 | ManxPower | Darwin35: iax2 show registry only shows device ASTERISK is registered TO. |
| 16:29.47 | alt | ManxPower: as well, Telus is blocking a pro-union website right now (the union is locked out by Telus) |
| 16:29.52 | tclark | Zeeek: but if you have some other specfic issue please tell me about them |
| 16:30.08 | xheliox | I have an TDM04B (4 fxo's) and whenever it dials out, it connects for 5 to 30 seconds and then hangs up without explaination. Anyone had an experience like that? |
| 16:30.16 | Zeeek | well... there is some noise at the moment of call establishment |
| 16:30.22 | *** part/#asterisk brettnem (~Brett@207.90.232.34) |
| 16:30.43 | Zeeek | I'm not at the office right now where I'm burnin testing a couple of these |
| 16:31.04 | Zeeek | I also change the digitmap file |
| 16:31.17 | *** join/#asterisk RoyK (~roy@host217-45-210-53.in-addr.btopenworld.com) |
| 16:31.22 | ManxPower | alt: I said it's not common, not that it NEVER happens. |
| 16:31.22 | tclark | do you have 1.44 on those so we are comparing apples to apples ? |
| 16:31.25 | Darwin35 | ok I just setup a exten *1004 it dials iax2/1004 and its dropping right to the unavaible vm and not ringing the phone |
| 16:31.35 | Zeeek | I think I prolly have .42 ? |
| 16:42.11 | ManxPower | Darwin35: Turn your brain back on. You know how to debug these issues. |
| 16:42.11 | tclark | o man way ols & a numbe of issues |
| 16:42.12 | gambolputty | can the contents of an ael file be put in a database? |
| 16:42.12 | tclark | be better if you could get 1.44.022 then let talk about issues |
| 16:42.12 | Zeeek | one of my phones has a problem with the volume buttons |
| 16:42.12 | Darwin35 | I have been but nothing has worked |
| 16:42.12 | Darwin35 | everything but inbound is working |
| 16:42.12 | Darwin35 | to the phone |
| 16:42.12 | ManxPower | Well what does the console show? |
| 16:42.13 | Zeeek | Darwin35 you have other IAX devices on the network? |
| 16:42.13 | Zeeek | because that too can be a problem sometimes |
| 16:42.13 | Zeeek | wacky port choices/conflicts |
| 16:42.13 | Darwin35 | I have 2 of the netweb 401 and everything else is sip |
| 16:42.13 | Zeeek | sometimes you need to make sure Port field is blank |
| 16:42.13 | *** join/#asterisk coppice (~chatzilla@63.196.17.210.dyn.pacific.net.hk) |
| 16:42.13 | *** join/#asterisk Henguei (~Henguei@196.203.53.45) |
| 16:42.14 | Zeeek | I went thru some shit getting thes eto work and they're not happy when there are more thazn one on the same LAN |
| 16:42.14 | *** join/#asterisk darkskiez (~darkskiez@194.247.78.146) |
| 16:42.14 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 16:42.14 | Darwin35 | hmmmmm |
| 16:42.14 | Zeeek | in fact looking at my box right now, both LAN phones have dropped out of register! |
| 16:42.14 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 16:42.14 | Zeeek | who knows how long ago |
| 16:42.14 | NewSole | bkw you alive |
| 16:42.15 | Zeeek | I have an ip500 still waiting to be reshipped at my son's house |
| 16:42.15 | Darwin35 | gawd I was hoping these where the answer to getting way from grandstreams and sip phones |
| 16:42.15 | NewSole | I need some info on transcoding |
| 16:42.15 | NewSole | can anyone help |
| 16:42.15 | alt | hey. who was using a SIP aware NAT router? and what is it? |
| 16:42.15 | Zeeek | Darwin35 we all thought that |
| 16:42.15 | alt | model/make |
| 16:42.15 | Zeeek | and we're gonna become eBay whores as a result :) |
| 16:42.15 | Darwin35 | heheh |
| 16:42.15 | Darwin35 | well I can get the sip flash if I have to or mgcp |
| 16:42.15 | Zeeek | no my phone works fine on other networks |
| 16:42.15 | yaaar | brb...restarting firefox to get a new extension... |
| 16:42.15 | Zeeek | that one is still fine |
| 16:42.15 | DarthClue | newsole: what do you need? |
| 16:42.17 | Darwin35 | grrrr |
| 16:42.23 | riemensc | i´ve got the problem with bristuff -- Extension 's' in context 'calls' from 'MY ISDN TELEPHONE NR' does not exist. Rejecting call on channel 0/2, span 1 |
| 16:42.23 | Darwin35 | exten => 501,1,Dial(IAX2/guest@70.186.117.198/*1004) |
| 16:42.23 | Darwin35 | some one try that |
| 16:42.23 | doughecka_ | how do I configure an adtran channel bank? |
| 16:42.23 | doughecka_ | is there an ip address? |
| 16:42.23 | ManxPower | riemensc: set immediate=no in zapata.conf. |
| 16:42.23 | doughecka_ | default |
| 16:42.23 | ManxPower | doughecka_: I've always done it via the console port,. |
| 16:42.23 | doughecka_ | ah |
| 16:42.23 | riemensc | i´ve set immediate=no |
| 16:42.31 | ManxPower | doughecka_: They CAN be configured via IP, I just have never done so. |
| 16:42.35 | _DAW | Darwin35 - Call rejected by 70.186.117.198: No such context/extension |
| 16:42.35 | NewSole | to prevent asterisk from using a license on g729..... right now we are going from iax2 => sip via g729 and its using 1 license.... |
| 16:42.38 | RoyKilt | jbot: that's a 11 minute lag, you know |
| 16:42.42 | ManxPower | riemensc: and stopped and started Asterisk? |
| 16:42.51 | Darwin35 | hmm |
| 16:42.53 | Darwin35 | hold |
| 16:42.57 | xheliox | I have an TDM400P (4 fxo's) and whenever it dials out, it connects for 5 to 30 seconds and then hangs up without explaination -- every time. Anyone had an experience like that? I've adjusted the callprogress and busydetect, per some articles I found on Google, but that didn't mean to make a difference. I have the exact same setup working across town, but this one is making me batty. |
| 16:43.02 | doughecka_ | ManxPower: is that ntwk? |
| 16:43.07 | gambolputty | anyone know ael here? |
| 16:43.21 | ManxPower | xheliox: don't use busydetect or callprogress. set them to no |
| 16:43.28 | xheliox | ManxPower: I did. |
| 16:43.28 | NewSole | our dial uses "tTg" if I just remove the "tT" and just keep the "g" will that do |
| 16:43.35 | ManxPower | doughecka_: Labeled LAN |
| 16:43.40 | doughecka_ | hmm |
| 16:43.43 | doughecka_ | isnt that... LAN? |
| 16:43.50 | riemensc | i´ve stop and start asterisk |
| 16:43.59 | riemensc | the echo test go correctly |
| 16:44.03 | ManxPower | NewSole: T/t enable DTMF transfers. |
| 16:44.15 | riemensc | i use nt mode for isdn |
| 16:44.21 | ManxPower | doughecka_: WELL, LAN would be IP, right? |
| 16:44.27 | _DAW | Darwin35 - try exten => 501,1,Dial(IAX2/guest@70.186.117.198/*1004@yourcontext) |
| 16:44.59 | Darwin35 | retry |
| 16:45.02 | ManxPower | doughecka_: I think it's admin or management or something like that. The adtrans we have are 100 miles away |
| 16:45.06 | doughecka_ | truw |
| 16:45.29 | doughecka_ | ah, missed your config via lan comment :) |
| 16:45.32 | Darwin35 | it worked now |
| 16:46.12 | NewSole | ManxPower... but would that stop the asterisk from using a license and transcoding when we remove the "tT" |
| 16:46.31 | ManxPower | NewSole: in theory |
| 16:46.44 | ManxPower | assuming all legs of the call are G729. |
| 16:46.55 | ManxPower | T/t makes one leg of the call be converted to SLIN to detect DTMF |
| 16:47.03 | NewSole | they are... but can we keep the "g" |
| 16:47.16 | ManxPower | "g" does nothing to the audio, so yes. |
| 16:47.18 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 16:48.05 | ManxPower | NewSole: Just remember, Asterisk can't convert between IAX and any other protocol without decoding the audio, |
| 16:48.12 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 16:49.03 | NewSole | just wondering about CPU usage on per call basis |
| 16:49.24 | _DAW | Darwin35 - luck? |
| 16:49.30 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 16:49.48 | Zeeek | 46331490what is the site of voiptalk again? |
| 16:50.01 | Darwin35 | nope |
| 16:50.06 | Zeeek | oops let my bank pin go again |
| 16:50.34 | _DAW | Darwin35 - your problem is no mwi? |
| 16:53.30 | Darwin35 | but I got your vm but n it ssems the wmi is not there |
| 16:53.30 | dalabera | Hello Everybody! Anyone here using the old T400P Quadspad T1 card? |
| 16:53.30 | Darwin35 | but atleast I get inbound calls now |
| 16:53.30 | Darwin35 | thats the next step |
| 16:53.30 | _DAW | Darwin35 - What type of phone are you using? |
| 16:53.30 | NewSole | ManxPower... I got one more question if you dont mind..... |
| 16:53.30 | Darwin35 | netweb 401 |
| 16:53.30 | Darwin35 | with the pa168 chip |
| 16:53.30 | Zeeek | there is NO MWI on those |
| 16:53.30 | Darwin35 | then have to fix that |
| 16:53.31 | Zeeek | unless the new firmware fixed it |
| 16:53.33 | NewSole | when audio comes off TE410 card... can we select codec we want it to come out as |
| 16:54.10 | Darwin35 | I have a repor going with sashi at iarea and letting him know what needs be done in the fimware |
| 16:54.38 | _DAW | Dawrin32 - if it does support MWI in new firmware make sure you have mailbox=phone@context in your sip.conf |
| 16:55.04 | ManxPower | mailbox=voicemailbox@voicemailcontext |
| 16:55.06 | ManxPower | not |
| 16:55.12 | ManxPower | mailbox=voicemailbox@extensioncontext |
| 16:55.26 | _DAW | right |
| 16:56.23 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 16:56.25 | yaaar | word |
| 16:57.32 | Darwin35 | sorry its aseshu and he is out the office today |
| 16:57.35 | Darwin35 | grrr |
| 16:57.45 | *** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET) |
| 16:58.06 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 16:58.15 | Darwin35 | some of the buttons are missed used on the phone |
| 16:58.32 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 16:59.03 | bkw_ | leaky deaky |
| 16:59.22 | Darwin35 | ok but ies all working otherwise now |
| 16:59.38 | outtolunc | omg, thursday again |
| 16:59.55 | Darwin35 | yeah tomarrow is payday |
| 17:00.02 | outtolunc | (today for me) |
| 17:00.06 | bkw_ | yes oh yes they both reached for the gun, the gun |
| 17:00.15 | bkw_ | :P |
| 17:00.28 | Zeeek | you think you have problems? I have a siements DECT phone that won't register on the base which is connected to the IAXy! |
| 17:01.13 | bkw_ | Qwell, my man |
| 17:01.18 | bkw_ | go to #asterlink please |
| 17:02.07 | Lee__ | I'm trying to forward SIP requests from Asterisk to SER. The requests are coming back after the initial sanity checks, which are reporting "too many hops" and "max length too big". Why would Asterisk be sending a Max-forward header greater than 10 for a server on the same LAN? |
| 17:02.23 | DarthClue | Qwell!!!!!!!!!!!!!!!!! |
| 17:03.02 | *** join/#asterisk bankrupt (~bank@pcp08695394pcs.500ash01.tn.comcast.net) |
| 17:03.12 | Hmmhesays | so head leaks a little right now huh? how bad? |
| 17:04.00 | DarthClue | Hmmhesays: it may not be limited to head. |
| 17:04.11 | anthm | depends on if you have the 2 patches it took me all morning to produce or not |
| 17:04.30 | Hmmhesays | you, all morning? they must be good |
| 17:04.55 | anthm | leaks are hard to find you gotta put soap on it and dunk it under water |
| 17:05.30 | alt | anthm: and then there's the patch kit |
| 17:05.34 | Hmmhesays | its really bad when they are on your bcd and your vents are leaking |
| 17:05.34 | alt | PITA |
| 17:05.43 | Hmmhesays | hard to stay neutral |
| 17:06.34 | opus_ | yeah in realtime |
| 17:09.21 | *** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com) |
| 17:09.24 | _-Jon-_ | Hey everything |
| 17:09.24 | Hmmhesays | what were you patching anthm? |
| 17:09.42 | anthm | stay tuned for mantis |
| 17:09.47 | essobi | Hey.. will valgrind find leaks in applications too? NEver used it.. |
| 17:10.13 | _-Jon-_ | Is it possible for Asterisk to somehow mess up incoming CID? The reason I ask is cause if when I call my Teliax toll-free number I get a random number in my area code, but sometimes the CID is "Asterisk" |
| 17:10.41 | ManxPower | _-Jon-_: I've never seen that happen |
| 17:10.58 | alt | _-Jon-_: are you getting the INFO digits sent to you? |
| 17:11.02 | alt | from the telco? |
| 17:11.11 | ManxPower | alt: re-read his question. |
| 17:11.11 | nDuff | essobi, that's what it's there for. |
| 17:11.20 | alt | ManxPower: I did read his question |
| 17:11.26 | _-Jon-_ | alt, INFO digits? what do you mean? |
| 17:11.33 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 17:11.44 | essobi | nDuff I mean literally asterisk appications.. like app_dial, and etc. |
| 17:11.51 | nDuff | ahh. |
| 17:11.57 | nDuff | dunno, there. |
| 17:12.02 | alt | depending on how your telco sets up your line, they may send you two digits prepended to the number to indicate the billing area. |
| 17:12.04 | nDuff | should be possible. |
| 17:12.07 | ManxPower | alt: Well, he's not going to know what Teliax gets from the Telco. There is no such thing as "INFO" packets in IAX2, and if he's running SIP, he should be using RFC2833, and NEITHER of those things has anything to do with callerid |
| 17:12.21 | alt | ManxPower: not info packets, info DIGITS. |
| 17:12.37 | `Sauron | Aha! |
| 17:12.38 | alt | anyhow. |
| 17:12.43 | ManxPower | _-Jon-_: Send a trouble report to teliax |
| 17:12.45 | `Sauron | nDuff, ;) |
| 17:13.09 | nDuff | `Sauron, howdy. |
| 17:13.13 | `Sauron | hola |
| 17:13.13 | alt | it's possible (but I've never seen it happen) that someone (or asterisk) is stuffing the info digits into the area code. |
| 17:13.16 | yaaar | can anybody tell me what the 'mask' field in the realtime sip table is for? sip.conf's page groups permit,deny,mask all in one page, but never mentions mask, and both the permit and deny statements include the subnet mask, so i'm confused |
| 17:13.39 | _-Jon-_ | ManxPower, so basically it isn't a problem with my Asterisk configuration? |
| 17:13.56 | ManxPower | _-Jon-_: Unless you are fiddleing with CLID in your dialplan. |
| 17:14.14 | _-Jon-_ | ManxPower, nope, not at all |
| 17:14.20 | Darwin35 | now to go find the src code for the phone firmware |
| 17:14.24 | ManxPower | iax2 debug should show you what Asterisk is getting from Teliax if you are using IAX2 |
| 17:14.39 | _-Jon-_ | Let me run iax2 debug and take a look |
| 17:17.05 | yaaar | also, is it ok to have two sip friends both with the same username? |
| 17:17.19 | ManxPower | yaaar: never. |
| 17:17.23 | yaaar | k |
| 17:17.56 | ManxPower | I set my SIP devices and Asterisk to use the MAC address of the device as a username with -a -b -c, etc appended for the line appearance. |
| 17:18.15 | yaaar | not a bad plan |
| 17:18.30 | alt | I use the extension as the username. |
| 17:18.37 | yaaar | i was going to use '<context>_<ext>' |
| 17:18.51 | ManxPower | alt: We found that to be too confusing. |
| 17:18.59 | yaaar | alt: i'm planning on having some extensions that overlap in different contexts |
| 17:19.12 | ManxPower | We ended up having totally different usernames for each line appearance |
| 17:19.27 | alt | ManxPower: it makes it easier for me as I have a couple of configuration files that use the username to do call routing. |
| 17:19.38 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
| 17:20.01 | alt | actually, it's not the config files, but the config file script generator that uses it to create the config file. |
| 17:20.06 | alt | easier automation. |
| 17:20.33 | yaaar | so nobody knows what the 'mask' field in the realtime sip table is for? |
| 17:20.34 | ManxPower | So if you have 2 phones with the same extension, what do you do? |
| 17:20.49 | alt | we don't. we define a meta-extension that calls each phone. |
| 17:20.51 | alt | erm |
| 17:20.54 | alt | calls both phones |
| 17:20.54 | NewSole | ok whats this |
| 17:20.55 | NewSole | frame.c:138 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 17:21.11 | hans | for a small office (~ 20 people) with an incoming T1 (split with data), am I right that you'd want a wildcard te[24]*p? |
| 17:21.38 | ManxPower | Whereas, we know what phone is making the request, regardless of the extension, by looking at the userid, which is the MAC address. |
| 17:21.47 | ManxPower | i.e. we enforce the fact that an extension is NOT a device. |
| 17:21.53 | ManxPower | other people try to hide that. |
| 17:22.08 | alt | and we just don't need that as a feature anyhow. we have one DID that calls all the phones in the office and that's only so our receptionist across the street can call all the phones if needed. |
| 17:22.09 | *** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985439.sympatico.ca) |
| 17:22.16 | dalabera | join #asterisk-bugs |
| 17:22.20 | jeffik | anyody using voxee? |
| 17:22.30 | ManxPower | alt: small number of users, I assume. |
| 17:22.36 | alt | yeah. 6 |
| 17:22.39 | nDuff | hans, yup. |
| 17:22.43 | ManxPower | I have 60 extensions |
| 17:22.45 | alt | ah |
| 17:23.04 | hans | nDuff: one port for the t1, one (or in the future, more) for the channel bank inside, right? |
| 17:23.05 | ManxPower | we found that doing it the traditional way just didn't scale for us from a management viewpoint |
| 17:23.10 | *** join/#asterisk exonic (~exonic@209.172.11.54) |
| 17:23.12 | exonic | Hey all |
| 17:23.23 | Hmmhesays | i hate it when I ask people how many endpoints they need to have, and they say something like 500 |
| 17:23.38 | alt | I'd be interested to see how you've set it up. |
| 17:23.41 | nDuff | hans, that's what we do. |
| 17:23.49 | exonic | I'm trying to create a better IAX load balancing solution than doing db(get|put) in the dialplan. |
| 17:24.05 | nDuff | hans, though we use IP phones for most users, we've got a channel bank for fax machines and such. |
| 17:24.42 | Hmmhesays | seriously, who starts off with 500 endpoints |
| 17:25.04 | alt | ManxPower: we're actually using our Asterisk server to do URI rewriting more than anything. one customer wanted the info digits stripped from the CLID and another customer wanted the ANI stuffed (why? we don't know....) |
| 17:25.18 | alt | I shouldn't call it URI rewriting.... |
| 17:25.20 | alt | but anyhow. |
| 17:25.21 | mut | 500.. |
| 17:25.27 | mut | venture capital |
| 17:25.32 | alt | call centres. |
| 17:25.52 | nDuff | is there any convention for passing extension numbers via caller-id? |
| 17:26.00 | Hmmhesays | 3 dual xeon 3ghz machines should take care of it |
| 17:26.26 | ManxPower | nDuff: what do you mean? |
| 17:26.26 | alt | nDuff: how do you mean? like mapping extensions to CLIDs that are valid DIDs? |
| 17:28.32 | nDuff | we have more users than DIDs, so for a bunch of folks it's <primary number>, plus <extension> (at the menu). I already rewrite caller-id to show <primary number> in those cases -- it'd be nice to expose the <extension> part as well. |
| 17:28.43 | exonic | Anyone aware of a way to load balance outbound calls across two servers each w/ a zap interface? |
| 17:29.13 | alt | nDuff: maybe put the extension in the callerID "name" field? |
| 17:29.50 | nDuff | alt, that's what I was pondering. |
| 17:29.59 | alt | that's the only thing I can think of. |
| 17:30.03 | *** join/#asterisk SwK[Work] (~SwK@64.89.118.139) |
| 17:30.13 | alt | I don't believe you should put it in the number field. |
| 17:35.02 | alt | or you could be like a collection agency that called me years ago (I was bad with student loans :( ) and stuffed the extension number into the ANI :P |
| 17:35.02 | alt | okay. snack time. |
| 17:35.02 | *** join/#asterisk cire-- (~e@adsl-215-65-134.mia.bellsouth.net) |
| 17:35.54 | Zeeek | exit |
| 17:36.28 | Darwin35 | daw you around |
| 17:37.03 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.active.supporter.pdpc) |
| 17:38.36 | *** join/#asterisk fugitivo (~ajf@201.255.100.39) |
| 17:38.38 | fugitivo | hello |
| 17:40.40 | syle | whats a LATA code used for? |
| 17:41.26 | *** join/#asterisk file[aboot] (~jcolp@66.199.241.90) |
| 17:41.28 | exonic | Does is seem that DUNDI and ENUM/E.164 are simliar protocols? |
| 17:41.31 | file[aboot] | meep |
| 17:42.19 | *** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 17:42.31 | Nugget | DUNDI seems more for termination providers to announce their capabilities, whereas enum/e164 seems tailored for end users to announce better ways to reach them. |
| 17:43.12 | file[aboot] | indeed |
| 17:43.13 | Nugget | using enumlookups is a great way to avoid termination charges if you want to call *me*. using DUNDI is a great way to minimize termination charges if you want to call *Austin Texas* |
| 17:43.27 | file[aboot] | yay ~aUA |
| 17:43.28 | blitzrage | DUNDi = de-centralized, ENUM = centralized |
| 17:43.28 | *** join/#asterisk jdg (~jdg@CA03F319.adsl.mana.pf) |
| 17:43.29 | file[aboot] | yay Austin Texas |
| 17:43.34 | file[aboot] | boo stupi terminal |
| 17:43.38 | file[aboot] | blitzrage: LEIF!!! |
| 17:43.47 | blitzrage | Leif's not here man |
| 17:43.51 | syle | do you know how to take a LATA and determine the NPA/NXX for Tiers 1-6 ? |
| 17:44.00 | file[aboot] | blitzrage: darn |
| 17:45.40 | blitzrage | aboot! |
| 17:45.43 | *** join/#asterisk jake1932 (~jake1932@pool-68-236-16-157.phil.east.verizon.net) |
| 17:45.45 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 17:45.55 | *** join/#asterisk doolph (doolph@201.226.146.178) |
| 17:46.30 | doolph | what's the way that I need to connect ip phones behind a nat to asterisk behind another nat |
| 17:46.51 | file[aboot] | b;e |
| 17:46.53 | file[aboot] | bleh I'm gone |
| 17:47.22 | *** join/#asterisk pa (~Paolo@pa.user) |
| 17:48.46 | Hmmhesays | if you're going with redhat enterprise, is there a preferred version to run asterisk on? |
| 17:50.27 | Hmmhesays | dell seems to prefer it for their poweredge hardware |
| 17:50.42 | blitzrage | Hmmhesays: why don't you just use CentOS? (I know thats not really your question :)) |
| 17:50.56 | bkw_ | oh Hmmhesays |
| 17:51.00 | bkw_ | where art thou |
| 17:51.06 | Hmmhesays | off in the clouds |
| 17:52.03 | mkrufky | anybody see kram lately? |
| 17:52.34 | mkrufky | ...thought I might find him here... I don't want to call his phone, I always end up catching him at the wrong time |
| 17:53.58 | drumkilla | mkrufky: what do you need? |
| 17:54.01 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E92A.versanet.de) |
| 17:54.04 | drumkilla | I sit right next to him :) |
| 17:54.22 | *** join/#asterisk dsfr (~dsfr@dsfr.digium.sponsor.pdpc) |
| 17:54.33 | mkrufky | ah... tell mark to call me with his flight info |
| 17:54.41 | mkrufky | he knows who mkrufky is |
| 17:54.43 | mkrufky | :-) |
| 17:55.16 | yaaar | can anybody tell me what the 'mask' field in the realtime sip table is for? sip.conf's page groups permit,deny,mask all in one page, but never mentions mask, and both the permit and deny statements include the subnet mask, so i'm confused |
| 17:56.15 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:56.29 | blitzrage | yaaar: mask isn't actually used anywhere if I remember correctly |
| 17:56.43 | blitzrage | yaaar: I asked the same question when I was documenting the sip.conf file |
| 17:56.57 | yaaar | ok....i was wondering, cause i couldn't see any use for it.. |
| 17:57.01 | blitzrage | yaaar: I think its a relic from the original implementation for Asterisk behind NAT (which of course is different now) |
| 17:57.02 | yaaar | thanks |
| 17:57.35 | *** join/#asterisk licued (~licued@ool-182cf211.dyn.optonline.net) |
| 17:57.42 | licued | heyy how is everyone? :) |
| 17:58.08 | licued | is anyone in the NY/NYC area? |
| 17:58.37 | Nugget | about 22 million people. |
| 17:58.44 | licued | oh yea!??! |
| 17:58.47 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
| 17:58.48 | licued | smart answer |
| 17:59.06 | Nugget | you've got questions -- we've got answers |
| 17:59.15 | licued | yea, whatver |
| 17:59.27 | licued | anyhow, is anyone in the NYC area and would like an Asterisk project to work no? |
| 17:59.29 | licued | on* |
| 17:59.32 | Nugget | may I please have your zip code for our database? |
| 17:59.44 | licued | no? |
| 17:59.44 | Hmmhesays | 1500 miles close enough? |
| 17:59.52 | licued | if you want to fly here is it |
| 17:59.56 | licued | it is* err |
| 18:00.04 | licued | it's a medium size project |
| 18:00.16 | syle | how are lata codes used in the US? |
| 18:00.27 | ManxPower | syle: I don't understand your question. |
| 18:00.47 | syle | do you know how to take a LATA and determine the NPA/NXX for Tiers 1-6 ? |
| 18:01.03 | ManxPower | syle: there really isn't a direct correlation for that anymore. |
| 18:02.05 | yaaar | syle: badly |
| 18:02.52 | xheliox | Stupid question (as always) -- are the two outer pins of each port on the TDM400P used for anything? |
| 18:03.00 | ManxPower | xheliox: nothing |
| 18:03.15 | xheliox | Good. |
| 18:03.24 | xheliox | Gracias. # |
| 18:03.30 | ManxPower | syle: For example near me is the "New Orleans LATA". |
| 18:03.58 | ManxPower | You SHOULD be able to map an NPA/NXX to a Rate Center |
| 18:04.32 | licued | well if anyone in the NYC/CT area wants a very good paying job with Asterisk/VoIP let me know |
| 18:04.57 | mut | syle.. |
| 18:05.09 | mut | http://members.dandy.net/~czg |
| 18:05.40 | mut | local calling guide |
| 18:05.45 | mut | might help? |
| 18:06.54 | ManxPower | mut: He could take a LATA, get the list of rate centers, and then the list of NPA/NXXs for each rate center. That would work |
| 18:07.41 | mut | that site has come in handy for me a few time |
| 18:07.41 | mut | s |
| 18:08.06 | mut | script to dump local calling areas XO numbers across the us |
| 18:08.10 | syle | dandy.net is horribly updated |
| 18:08.13 | syle | but seems to be the best one |
| 18:08.48 | mut | well chances are whatever you're doing isn't going to require up to the minute updates... |
| 18:09.19 | ManxPower | Unless you want to spend the money for a LERG subscription |
| 18:09.31 | mut | i prefer the free option |
| 18:09.53 | syle | i have a buddy here trying to be a local carrier, he got permission to hookup to the tandum switches here in august, and gonna sell alot of termination |
| 18:10.13 | ManxPower | We have a pretty simple rate center with our CLEC. Calls to any number in Louisiana and Mississippi are free. 8-) |
| 18:10.14 | harryvv | nice |
| 18:10.20 | Johann000 | what could cause a "Red Alarm" to be triggered? |
| 18:10.34 | mut | ya, i got that for michigan |
| 18:10.42 | harryvv | Johann000 check your line connection on the back of the server |
| 18:10.59 | harryvv | phone line or pri disconected |
| 18:11.06 | ManxPower | Johann000: loss of physical layer of the T-1. If it was working and is no longer working, call your telco and say "I have a RED alarm on Circuit ID XXXXXXX, can you loop to the smart jack?" |
| 18:11.39 | ManxPower | if they can loop to the smartjack then you have a cable problem between the smartjack and Asterisk |
| 18:13.33 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 18:13.38 | ManxPower | If they CAN'T loop the smart jack then they know there is a problem and you don't have to spend a day convinving them |
| 18:13.43 | ManxPower | convince, too |
| 18:13.45 | *** join/#asterisk Assid (assid@PPP-219.65.11.218.mum1.dialup.vsnl.net.in) |
| 18:13.46 | Blackthorn | Does anyone know how many calls you can send to nufone at the same time? |
| 18:14.07 | ManxPower | Blackthorn: If you pay per/min, most carriers allow as many calls as you want. |
| 18:14.10 | Assid | . /usr/bin/ld: cannot find -lssl <-- but i have openssl and libssl installed |
| 18:14.20 | Assid | any clue whats up? |
| 18:14.23 | Assid | heya manx |
| 18:14.41 | jayk- | probably a revision problem. |
| 18:14.45 | riemensc | everbudy user voipbuster |
| 18:14.46 | riemensc | via iax |
| 18:14.54 | Blackthorn | thansk mp. |
| 18:15.01 | jayk- | did you upgrade openssl, assid? |
| 18:15.25 | *** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985282.sympatico.ca) |
| 18:15.37 | Assid | am using debian.. does it on its own |
| 18:15.45 | Assid | fresh install on this particular box |
| 18:15.49 | jayk- | oh |
| 18:16.09 | riemensc | who is the error code == No one is available to answer at this time |
| 18:16.11 | jayk- | you might need to run ldd to fnd out what rev it is looking for |
| 18:16.12 | riemensc | by iax |
| 18:16.18 | jayk- | and then link that rev to the existing rev |
| 18:16.42 | jayk- | or reboot if you just updated openssl, so that the ld cache is updated |
| 18:18.24 | Assid | nvm.. it worked |
| 18:18.28 | Assid | needed libssl-dev |
| 18:19.46 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 18:19.52 | Blackthorn | I am getting lots of reports over the last few days of calls goign to fast busy... was monitoring the console looks like there goign to nufone but getting hung up on |
| 18:20.04 | Craziman2 | Any one have any idead why Music on Hold would work with Sip Channels but not IAX Channels? |
| 18:20.22 | ManxPower | Blackthorn: NuFone has some network issue |
| 18:21.01 | ManxPower | Blackthorn: I think they were minor problems and only affected people on one of their servers. |
| 18:21.06 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 18:21.54 | *** join/#asterisk mistral (~mistral@jstevenson.plus.com) |
| 18:22.11 | riemensc | <PROTECTED> |
| 18:22.18 | riemensc | <PROTECTED> |
| 18:22.18 | riemensc | <riemensc> and can not do outgoing calls |
| 18:22.25 | riemensc | and recieve the error message |
| 18:22.30 | riemensc | <PROTECTED> |
| 18:22.43 | riemensc | please help me |
| 18:22.47 | ManxPower | iax2 show registry shows servers ASTERISK is registered to |
| 18:23.07 | riemensc | the serv ip username registered |
| 18:23.20 | riemensc | 213.61.187.146:4569 riemensc 83.169.155.92:4569 60 Registered |
| 18:23.34 | ManxPower | also for calls from Asterisk to a provider does NOT require registration |
| 18:24.07 | riemensc | i think this provider require a registration |
| 18:24.30 | *** join/#asterisk PaulTech2 (~PaulTech@66.195.243.254) |
| 18:25.14 | harryvv | manx, you said something about my sip problem yesterday |
| 18:25.25 | harryvv | running ipcop |
| 18:25.37 | ManxPower | riemensc: it would be the first provider in the world to do so. |
| 18:26.36 | PaulTech2 | Anyidea what dtmfmode i should use for incoming calls? |
| 18:26.42 | *** join/#asterisk darby_t (~tom@dns232.neoplus.adsl.tpnet.pl) |
| 18:26.56 | ManxPower | PaulTech2: always RFC2833 |
| 18:27.17 | PaulTech2 | Thanks |
| 18:27.17 | licued | hey you guys are all very great. Gonna help my aunt move out of her apt, talk to you all later ;) |
| 18:27.20 | mistral | how scalable is * ? |
| 18:27.49 | harryvv | mistral from my understanding up to 450 calls for a opteron system. |
| 18:28.02 | PaulTech2 | Works with iax correct? |
| 18:28.05 | ManxPower | mistik1: A1: very scalable A2: not very scalable. What do you want to do with Asterisk? |
| 18:28.07 | harryvv | actually 550 calls |
| 18:28.10 | mistral | opteron ? |
| 18:28.14 | harryvv | yes |
| 18:28.18 | harryvv | amd opteron |
| 18:28.18 | ManxPower | PaulTech2: IAX2 only have one DTMF mode and is always set and cannot be changed. |
| 18:28.25 | anthm | as scalable as the walls of alcatraz |
| 18:28.29 | *** join/#asterisk damin_ (~damin@nucleus.nacs.net) |
| 18:28.41 | PaulTech2 | ManxPower, ah... |
| 18:28.49 | PaulTech2 | trying to figure out why its doing this |
| 18:28.55 | PaulTech2 | my phones use info |
| 18:28.58 | ManxPower | If you want to do lots of G279<->SpeeX calls then it's not going to scale very well. |
| 18:29.01 | PaulTech2 | and I use IAX for Incoming |
| 18:29.02 | harryvv | if you want industrial streagth scalable go ser but other then that done know more about it. |
| 18:29.13 | ManxPower | PaulTech2: INFO is a SIP thing |
| 18:29.22 | PaulTech2 | Yea |
| 18:29.25 | PaulTech2 | I use SIP Phones for internal |
| 18:29.28 | PaulTech2 | and they send in info |
| 18:29.32 | PaulTech2 | and my office phones work |
| 18:29.36 | ManxPower | PaulTech2: try rfc2833 then |
| 18:29.39 | PaulTech2 | but some cell phones dont let you press the menu choices |
| 18:29.44 | PaulTech2 | We got alot of compliants |
| 18:29.47 | PaulTech2 | internal phones work great |
| 18:29.50 | PaulTech2 | using info |
| 18:29.55 | ManxPower | force both the phones and Asterisk to be rfc2833 |
| 18:30.01 | ManxPower | PaulTech2: how are you connecting to the PSTN? |
| 18:30.13 | PaulTech2 | IAX > IP > IAX > PSTN At remote |
| 18:30.30 | PaulTech2 | We use a third party Provider till our T1 gets in place |
| 18:30.37 | PaulTech2 | to provide the PSTN access over IAX |
| 18:30.41 | PaulTech2 | and Im having all kinds of problems |
| 18:30.43 | *** join/#asterisk xarmiex (~armie@arm.enter.net) |
| 18:30.52 | *** join/#asterisk scratchrf (~scratchrf@63-226-200-214.tukw.qwest.net) |
| 18:31.20 | ManxPower | PaulTech2: Yeah, since you can't control the provider's PSTN setup |
| 18:31.27 | PaulTech2 | Right |
| 18:31.42 | PaulTech2 | Dont suppose voicepulse rings a horrible bell with thesse problems |
| 18:31.47 | ManxPower | One of the MANY reasons I try, whenever I can, to use PSTN PRI that I control. |
| 18:32.19 | PaulTech2 | Cards are expensive |
| 18:32.19 | PaulTech2 | heh |
| 18:32.30 | PaulTech2 | We can afford but we're trying to save money going VoIP and I guess thats not the case |
| 18:32.37 | ManxPower | PaulTech2: so is having your PSTN access down because of a internet problem. |
| 18:33.48 | PaulTech2 | ManxPower, We run a datacenter |
| 18:33.56 | PaulTech2 | If we have internet problems, There is a huge problem ;) |
| 18:34.30 | astoria | PaulTech2: do you offer colos? |
| 18:34.43 | ManxPower | What about your ITSP's internet problems, or internet problems between you and the ITSP |
| 18:35.03 | PaulTech2 | astoria, very much so |
| 18:35.19 | astoria | PaulTech2: where are you located? |
| 18:35.25 | astoria | PaulTech2: in a carrier hotel, i presume? |
| 18:35.26 | PaulTech2 | ManxPower, I would have them with a local PSTN too |
| 18:35.36 | PaulTech2 | astoria, Orlando FL and no we own two of our own |
| 18:35.43 | *** join/#asterisk juanjoc (~juanjoc@200.73.189.82) |
| 18:35.44 | PaulTech2 | 5k and 14k sqft |
| 18:35.51 | astoria | PaulTech2: who would i contact about a quote? |
| 18:35.54 | PaulTech2 | With a pressence in Atlanta at TelX |
| 18:35.58 | PaulTech2 | astoria, I can provide that |
| 18:36.35 | astoria | i'm still trying to get some customers, but i'm trying to find an inexpensive data center/colo where i can terminate voip traffic. |
| 18:36.45 | juanjoc | What version of spandsp is recommended to send faxes over IP? |
| 18:36.54 | ManxPower | juanjoc: none. |
| 18:37.03 | astoria | i'm in the middle of issuing quotes to potential customers and whatnot. |
| 18:37.09 | ManxPower | juanjoc: sending data converted to voice converted to data is not a good idea. |
| 18:37.29 | harryvv | astoria, heard of a pri for less then 200 dollars per month in texas |
| 18:37.33 | PaulTech2 | astoria, Get at me with what you'll be needing and I can get you a quote right away |
| 18:37.42 | astoria | PaulTech2: drop me your email addy |
| 18:37.44 | juanjoc | I know but I have no other option with Asterisk right now. |
| 18:37.49 | astoria | harryvv: i can get a pri here for 150 a month |
| 18:37.59 | harryvv | astoria, where |
| 18:37.59 | yaaar | if i put a line like 'exten => 4345748877,1,Goto(customer,s,1)' into my incoming context, that'll just make the call get processed according to the dialplan within [customer], right? |
| 18:38.06 | yaaar | (just trying to make sure i'm not crazy) |
| 18:38.07 | Assid | <PROTECTED> |
| 18:38.22 | juanjoc | ManxPower: do you know of something else that works over IP with Asterisk. |
| 18:38.46 | ManxPower | juanjoc: no such thing exists for Asterisk |
| 18:38.48 | juanjoc | ManxPower: I've seen no T.38 implementation for Asterisk yet. |
| 18:38.53 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 18:39.12 | ManxPower | juanjoc: correct. |
| 18:40.16 | *** join/#asterisk pbxbart (~pbx111@p54B031F9.dip0.t-ipconnect.de) |
| 18:40.33 | xarmiex | hmm, we cant seem to be able to get rid of our echo problem here, we have a te405p with 1 pri on it now, if someone calls from a landline the person answwering with the cisco(sip) phone can hear himself all the time, does anyone know where we should be looking |
| 18:40.46 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
| 18:41.25 | pbxbart | hi |
| 18:41.26 | mut | try messin with the gain |
| 18:41.31 | mut | on the pri |
| 18:41.38 | pbxbart | i have a few problems with realtime |
| 18:41.39 | ManxPower | xarmiex: fiddle with the txgain= on the pri |
| 18:41.45 | ManxPower | Usually negative numbers. |
| 18:42.06 | mut | -1, -2 has usually worked well for me |
| 18:42.11 | xarmiex | we werent sure if the gain was for this card as well, we'll try that thx |
| 18:42.42 | pbxbart | anyone using realtime with rtcachefriends |
| 18:43.12 | pbxbart | this seams to be broken... |
| 18:44.19 | *** join/#asterisk Coriantum (~asdfkle@67-41-182-243.slkc.qwest.net) |
| 18:44.53 | pbxbart | sip show peer xxxx load does not load the ip |
| 18:45.14 | *** join/#asterisk |Barcode (~barcode@h-68-165-204-41.chcgilgm.covad.net) |
| 18:45.19 | *** join/#asterisk ecto (~ectospasm@69.85.202.242) |
| 18:45.19 | xarmiex | sip show peer doesnt work with realtime does it ? |
| 18:45.33 | pbxbart | it did. with rtcachefriends |
| 18:45.48 | pbxbart | it is stillworking but it does not load the ip |
| 18:46.03 | anthm | sip show peer foo load |
| 18:46.16 | pbxbart | not even with rtignoreregexpire=yes |
| 18:46.30 | pbxbart | I do the following |
| 18:46.40 | pbxbart | sip prune realtime peer 10001 |
| 18:46.45 | pbxbart | for cleanup |
| 18:46.54 | pbxbart | sip show peer 100011 |
| 18:46.54 | pbxbart | Peer 100011 not found. |
| 18:46.57 | pbxbart | then |
| 18:47.22 | pbxbart | it shows the peer but without the ip |
| 18:47.51 | pbxbart | if i do realtime load SIPpeers name 100011 it shows the ip |
| 18:47.59 | pbxbart | ?? |
| 18:48.13 | pbxbart | should i create a bug at mantis? |
| 18:48.27 | ecto | I'm having trouble getting external callers to any of our extensions (8XXX), apparently only the 8 is registering. It worked fine yesterday. |
| 18:49.07 | ecto | I'm getting a bunch of these errors: NOTICE[9122]: channel.c:1464 ast_read: Dropping incompatible voice frame on IAX2/voicepulse-wgw001@voicepulse-wgw001-1 of format slin since our native format has changed to ulaw |
| 18:49.17 | anthm | sip show peer foo "load" |
| 18:49.45 | pbxbart | @ anthm this is what im executing |
| 18:49.45 | anthm | if you like realtime sip you may also want the patch in bug 4832 |
| 18:50.22 | anthm | I made it along with 3 other curitial fixes this week, when I took time off from all the evil I supposedly do |
| 18:51.32 | pbxbart | 4832 looks good |
| 18:51.37 | pbxbart | i will try it |
| 18:51.45 | pbxbart | but does this solve my load issue |
| 18:51.53 | astoria | DarthClue: The cluecon hotel will provide free internet, correct? |
| 18:51.58 | PaulTech2 | Hmm |
| 18:52.05 | *** join/#asterisk santiago (~santiago@63.245.86.141) |
| 18:52.49 | DarthClue | astoria: bkw_ should have all the details on that. but i am pretty sure that there will be internet access available. |
| 18:53.15 | anthm | i'd update to this minute's CVS apply that patch and find out |
| 18:53.38 | astoria | Ok. Best Western's usually do. |
| 18:53.39 | pbxbart | i ll try |
| 18:53.43 | Blackthorn | hehe.. last week i checked into this hotel that advertised free internet.. well it would allow you to go to like cnn.com, fox.com, msn.com... if you wanted to serach for anyting or go anywhere else you had to pay. man that pissed me off. |
| 18:53.44 | anthm | if not let me know I made all of those options so I can probably tell you easier |
| 18:53.54 | astoria | DarthClue: is there going to be some kind of announcement about that or anything? |
| 18:54.22 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
| 18:54.51 | DarthClue | about the intarweb? um...i'll see if it's on the propoganda list. |
| 18:56.42 | *** join/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
| 18:57.16 | *** join/#asterisk dacleric (~dacleric@p5482A1EF.dip0.t-ipconnect.de) |
| 18:57.47 | astoria | DarthClue: anything? |
| 18:58.45 | Assid | <PROTECTED> |
| 18:58.47 | Assid | weird |
| 18:58.49 | Assid | why? |
| 18:59.07 | *** part/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 18:59.09 | *** join/#asterisk iswm (iswm@iswm.user) |
| 18:59.58 | Assid | im using CVS |
| 19:00.16 | anthm | make clean in zap pri and asterisk and rebuild |
| 19:00.40 | DarthClue | astoria: what kind of announcement are you looking for? we have been jumped for actually telling people what is going on, so the propaganda department is trying to limit what gets plastered. |
| 19:00.41 | pbxbart | anthm: make is running |
| 19:00.47 | PaulTech2 | DarthClue, I broke my shit agian |
| 19:00.48 | Assid | zaptel first? |
| 19:00.49 | PaulTech2 | again* |
| 19:00.51 | *** join/#asterisk clive- (~pirch@rrba-146-100-51.telkomadsl.co.za) |
| 19:00.52 | Assid | or libpri first? |
| 19:00.57 | anthm | that was for Assid |
| 19:01.39 | astoria | DarthClue: ha ha. beats me. i was just wondering if i can tell my employer i'll be able to login from offsite in case of emergency. |
| 19:01.40 | Assid | anthm: do i make libpri first or zaptel first? |
| 19:01.46 | anthm | pri |
| 19:01.52 | DarthClue | PaulTech2: i didn't get your last payment...but i can try to give ya some help at double the normal rate... ;) |
| 19:02.20 | DarthClue | astoria: one way or another, we can make that happen. although we may not actually put out a formal announcement. |
| 19:02.33 | astoria | DarthClue: okay, thats fine. thanks! |
| 19:02.47 | jeffik | anybody help with an aah question? |
| 19:03.04 | pbxbart | anthm: it works |
| 19:03.17 | anthm | woohoo ! |
| 19:03.24 | DarthClue | jeffik: although many of us do use asterisk in our homes, we don't use aah, but if you ask the question, you might get an answer. |
| 19:03.31 | anthm | tell kram he's doing patches |
| 19:03.44 | *** part/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
| 19:04.42 | Hmmhesays | I tinking with aah |
| 19:04.46 | Hmmhesays | *tinker even |
| 19:04.55 | *** join/#asterisk Paul[NOC] (~paul@66.195.243.254) |
| 19:05.26 | pbxbart | @ anthm: wait. there something strange |
| 19:06.01 | pbxbart | i recive a lot of chan_sip.c:945 __sip_xmit: sip_xmit of 0x8197860 (len 762) to xxx.xxx.xxx.xxx returned -1: Invalid argument |
| 19:06.27 | anthm | that should have nothing to do with the patch |
| 19:07.03 | pbxbart | do you know to which update this is related to? the call is still not going through |
| 19:07.35 | *** join/#asterisk jfonsecausa (~jfonseca@12.42.141.34) |
| 19:08.09 | Paul[NOC] | Hmm |
| 19:08.18 | Paul[NOC] | DarthClue, Recall that IVR we did? |
| 19:08.26 | DarthClue | Paul[NOC]: yep. |
| 19:08.32 | Assid | thanks anthm: that did it |
| 19:08.35 | anthm | does it really say xxx ? |
| 19:09.02 | pbxbart | no.. it says the ip of my phone |
| 19:09.05 | Paul[NOC] | DarthClue, any ideas why it would suddendly stop working |
| 19:09.12 | anthm | that is a network issue |
| 19:09.25 | Paul[NOC] | you press a button and it does nothing |
| 19:09.27 | anthm | it means it tried to send a udp packet that didnt work |
| 19:09.39 | DarthClue | Paul[NOC]: stop working completely? or is it back to the delay? |
| 19:09.41 | *** join/#asterisk konfuzed (~kvirc@H129.C72.B0.tor.eicat.ca) |
| 19:09.45 | anthm | do you have nat sip ? |
| 19:09.48 | Paul[NOC] | DarthClue, It just stays at the menu |
| 19:09.51 | Paul[NOC] | No matter what you press |
| 19:09.54 | pbxbart | yes |
| 19:10.00 | pbxbart | for that client |
| 19:10.16 | DarthClue | Paul[NOC]: on all phones or on just one phone? sounds like the dtmf isn't getting thru. |
| 19:10.28 | pbxbart | registraions are no longer working as well |
| 19:10.38 | Paul[NOC] | DarthClue, tried from phones outside the PBX |
| 19:10.38 | anthm | it's a low level socket call |
| 19:10.40 | Paul[NOC] | old PSTN Phones |
| 19:10.41 | Paul[NOC] | Cell phones |
| 19:10.43 | anthm | that is not working |
| 19:10.45 | pbxbart | hm |
| 19:10.51 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 19:10.55 | anthm | you should check your network settings and sip config |
| 19:11.11 | pbxbart | sip is not working at all on this maschine. no calls no regs |
| 19:11.13 | pbxbart | no debug |
| 19:11.15 | pbxbart | strange |
| 19:11.24 | fugitivo | Paul[NOC]: maybe a dtmf problem |
| 19:11.29 | anthm | review the entire sip.conf |
| 19:11.36 | anthm | also make sure there is no ip filter crap |
| 19:11.39 | pbxbart | it has not change |
| 19:11.40 | pbxbart | d |
| 19:11.59 | anthm | did it stop working recently ? |
| 19:12.11 | Paul[NOC] | I dont know why it would stop working |
| 19:12.15 | DarthClue | Paul[NOC]: has anything changed on the box? |
| 19:12.22 | Paul[NOC] | DarthClue, negative |
| 19:12.26 | pbxbart | sip stoped after any cvs update to todays head + applying your patch |
| 19:12.31 | Paul[NOC] | I'm the only person with access |
| 19:12.34 | anthm | did you make clean ? |
| 19:12.39 | pbxbart | sure |
| 19:12.50 | pbxbart | i'll revert you patch and see if it is related |
| 19:13.05 | Paul[NOC] | My config backups have problems too |
| 19:13.07 | Paul[NOC] | I dont know what could of changed |
| 19:13.13 | DarthClue | Paul[NOC]: turn on debug ... modify logger.conf ... and see if the dtmf is making it thru. |
| 19:13.21 | konfuzed | I really can't figure out this connection problem, I think it my be on the service provider side. wheather I try to connect via iax.conf or sip.conf I get the same message - NOTICE[23944]: chan_sip.c:8024 sip_poke_noanswer: Peer 'from-eicat' is now UNREACHABLE! - what is the cause of this poke no answer message ? |
| 19:13.38 | Paul[NOC] | sip debug? |
| 19:13.44 | anthm | oh wait |
| 19:13.57 | anthm | did you catch |
| 19:13.58 | anthm | *NOTE* make sure you add the fullcontact column to your realtime sip table if you are using a database. |
| 19:14.09 | anthm | perchance ? |
| 19:14.13 | astoria | konfuzed: you have qualify=yes in your sip.conf |
| 19:14.17 | DarthClue | Paul[NOC]: no, full blown debug. if you modify logger.conf, the debug will get output to the cli. it should show the dtmf digits as they are processed. |
| 19:15.17 | konfuzed | astoria I have qalify=1000 |
| 19:15.27 | Paul[NOC] | DarthClue, set debug ? |
| 19:15.30 | Paul[NOC] | I modified logger.conf |
| 19:15.31 | Paul[NOC] | to have debug |
| 19:15.35 | Paul[NOC] | on console |
| 19:15.42 | konfuzed | just because that's what was with the sip context I used as a base reference |
| 19:16.02 | astoria | konfuzed: is there something going on with NAT? |
| 19:16.07 | DarthClue | yeah, then set debug 99, and try it, see if the dtmf comes thru and then pastebin the output for me to look at. |
| 19:16.14 | astoria | konfuzed: usually you'd use that with a client, such as a phone |
| 19:16.30 | anthm | pbxbart did you see my last msg? |
| 19:16.39 | pbxbart | yes. if added the colum |
| 19:16.40 | pbxbart | soory |
| 19:16.52 | pbxbart | the compile window was in front of me |
| 19:17.04 | konfuzed | i dont think so other than both sides may not be matched correctly, ive tried both nat=yes and nat=no |
| 19:17.18 | hardwire | blah |
| 19:17.24 | hardwire | you think your confused |
| 19:17.25 | astoria | konfuzed: try qualify=no |
| 19:17.31 | hardwire | I have a joke |
| 19:17.38 | lathos42 | DarthClue: Are the times listed on the ClueCon website in CST or EST? |
| 19:17.39 | hardwire | anybody for hearing it? |
| 19:17.41 | astoria | konfuzed: that will probably solve your problems if your saying what I think you're saying |
| 19:17.56 | DarthClue | lathos42: GMT ... er ... CMT |
| 19:18.15 | DarthClue | lathos42: er CDT ... whatever time it is in chicago. |
| 19:18.20 | lathos42 | DarthClue: Ok :) |
| 19:18.29 | konfuzed | so just to note. this sip connection is for an incoming 10digit phone number from my service provider. |
| 19:18.38 | Paul[NOC] | Hmm DarthClue, I dont think they are |
| 19:18.40 | konfuzed | who is running asterisk |
| 19:18.44 | Paul[NOC] | Its kinda impossible to follow that very fast scrolling |
| 19:19.05 | DarthClue | Paul[NOC]: pastebin it and i'll have a look. |
| 19:19.18 | Paul[NOC] | DarthClue, Its just a bunch of lines |
| 19:19.32 | Paul[NOC] | DarthClue, Would you like to login? |
| 19:19.36 | DarthClue | konfuzed: asterisk doesn't actually run, we just have lots of fantasies that it does. |
| 19:19.53 | DarthClue | Paul[NOC]: i can do that if you want. |
| 19:20.24 | lathos42 | DarthClue: I would suspect that being a bot you synchronize to an ntp server, right? :) |
| 19:21.19 | *** join/#asterisk pbxbart (~pbx111@p54B031F9.dip0.t-ipconnect.de) |
| 19:21.21 | pbxbart | hi |
| 19:21.21 | Paul[NOC] | DarthClue,SEnt you a message |
| 19:21.32 | pbxbart | i'm back |
| 19:21.38 | hardwire | pbxbart? |
| 19:21.40 | hardwire | you rule |
| 19:21.42 | anthm | DarthClue is an example of the innovation of Asterlink he's the first bot that can ssh to your box and fix asterisk for you. |
| 19:21.53 | DarthClue | Paul[NOC]: got it, will be there in a moment. |
| 19:21.57 | hardwire | so |
| 19:22.01 | DarthClue | lathos42: ntp is so yesterday. |
| 19:22.09 | hardwire | what is the preferred drink when trying to come upw ith the company wide dial-plan? |
| 19:22.12 | *** part/#asterisk lehel (~Lehel@82.79.20.17) |
| 19:22.21 | ManxPower | hardwire: Chai |
| 19:22.23 | pbxbart | m windows mashine crasched @anthm did you wrote anything after add that column |
| 19:22.25 | hardwire | so far.. everything that I would want a PBX for.. nobody here wants |
| 19:22.26 | hardwire | like |
| 19:22.28 | lathos42 | DarthClue: I apologize if I've insulted you, please dont dismember me like a person who hasnt read the wiki |
| 19:22.29 | hardwire | they want it to ring |
| 19:22.30 | hardwire | say hi |
| 19:22.34 | anthm | no |
| 19:22.34 | DarthClue | hardwire: something strong, very, very strong, like battery acid. |
| 19:22.38 | hardwire | then say if you know an extension.. dial it now |
| 19:22.46 | hardwire | otherwise.. this linda person gets the phone |
| 19:22.50 | hardwire | or.. it goes to voicemail |
| 19:22.59 | hardwire | that doesn't solve the 1 problem we are trying to avoid.. |
| 19:23.04 | hardwire | lack of a response.. and being out of touch |
| 19:23.13 | hardwire | my boss is so limited |
| 19:23.20 | hardwire | can I tell him that |
| 19:23.28 | hardwire | You.. you are the most limited person I know |
| 19:23.32 | konfuzed | ok well I did not receive the previous message but I am back to this message chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 50a501a0010dd8bd341060d51f455aa7@66.96.30.25 for seqno 102 (Critical Response) |
| 19:23.35 | hardwire | I wish you would think more. |
| 19:23.54 | ManxPower | konfuzed: sounds like the SIP client is behind NAT |
| 19:24.09 | mut | who are you talking to hardwire? |
| 19:24.16 | hardwire | so I am going to apply for a NOC job soon |
| 19:24.22 | hardwire | and then take mut with me |
| 19:24.29 | hardwire | mut: I am talking to jesus |
| 19:24.38 | mut | yes my son, i am listening |
| 19:24.40 | anthm | fdisk /dev/hda would be less time consuming |
| 19:24.54 | ManxPower | or cfdisk -z /dev/hda 8-) |
| 19:25.04 | hardwire | dd if=/dev/zero of=/dev/hda bs=1024 count=10 |
| 19:25.05 | hardwire | that should do it |
| 19:25.20 | hardwire | 1024k :) |
| 19:25.23 | konfuzed | what's this maximum retires execeed from sip |
| 19:25.31 | ManxPower | konfuzed: sounds like the SIP client is behind NAT |
| 19:25.32 | konfuzed | so turn nat to nat=yes |
| 19:25.40 | hardwire | I think dmesg should spit out the entire partition map on bootup |
| 19:25.47 | hardwire | so that when something like that happens.. you can restore it |
| 19:25.49 | hardwire | yay |
| 19:25.52 | hardwire | meeting is back on |
| 19:25.56 | ManxPower | konfuzed: add qualify-yes |
| 19:26.00 | astoria | konfuzed: you can turn qualify back on too. |
| 19:26.12 | Paul[NOC] | Everyone is rock |
| 19:26.12 | astoria | konfuzed: i didn't think you were behind aNAT |
| 19:26.15 | Paul[NOC] | Everyone is wrong |
| 19:26.19 | mut | i am paper! |
| 19:26.24 | Paul[NOC] | Drive is on /dev/sda ;) |
| 19:26.28 | mut | i beat rock |
| 19:26.42 | konfuzed | ok when I use nat=yes I get WARNING[24051]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6e17f25d392ce32f0b6a99964f28988b@192.168.1.136 for seqno 102 (Non-critical Request) |
| 19:26.58 | mut | konfuzed: deal with it |
| 19:27.17 | *** join/#asterisk xxi (foobar@cpe-70-113-47-137.austin.res.rr.com) |
| 19:27.30 | *** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net) |
| 19:27.31 | konfuzed | ive set host=externalIP |
| 19:27.39 | DarthClue | fdisk /dev/sda |
| 19:27.44 | mut | it's going to happen, deal with it |
| 19:27.44 | DarthClue | oops ... wrong window. |
| 19:27.54 | Derkommissar | afther 260 calls sip stops reciving calls..... and i get this error when i turn the logger on |
| 19:27.55 | pbxbart | @ anthm. something seams to be strang with todays head |
| 19:27.56 | Derkommissar | sipsock_read: Failed to grab lock, trying again... |
| 19:27.58 | pbxbart | i |
| 19:28.03 | Derkommissar | wat is this suposed to mean? |
| 19:28.07 | pbxbart | 'll try the 25. |
| 19:28.08 | Derkommissar | is the nic phreaking out ? |
| 19:28.12 | konfuzed | but I get this message on the 136 |
| 19:28.21 | anthm | we should make a sip_with_nat.pl that asks you all the questions and generates the shit |
| 19:28.37 | anthm | ? |
| 19:28.37 | mut | heh |
| 19:28.46 | mut | probly wouldn't solve much |
| 19:28.54 | konfuzed | I'm trying to figure out how to deal with and so far im a little Konfuzed as to what the problem is |
| 19:28.58 | astoria | people would still come an dask. |
| 19:29.04 | astoria | and ask. |
| 19:29.11 | mut | deal with it = ignore it |
| 19:29.15 | astoria | konfuzed: are your calls going through? |
| 19:29.18 | Paul[NOC] | Hey I'm a newbie and I got 15 phones behind one nat and 5 behind other |
| 19:29.20 | konfuzed | no |
| 19:29.22 | Paul[NOC] | and that works great |
| 19:29.22 | Paul[NOC] | :D |
| 19:29.32 | Paul[NOC] | I cheated thou |
| 19:29.48 | konfuzed | back in ten |
| 19:30.33 | anthm | Darth you should make a web-based nat-sip configuator complements of evlicon |
| 19:30.44 | pbxbart | it there a way to finded all fixes appley to the Head via mantis? |
| 19:31.21 | anthm | I will update mine and tell you if sip works or not |
| 19:31.44 | pbxbart | @antm that would be nice |
| 19:32.54 | hardwire | wow |
| 19:33.01 | hardwire | explaining how a pbx works to this guy is like |
| 19:33.03 | hardwire | I don't know really |
| 19:33.11 | hardwire | you guys can probably fill it in yourself |
| 19:33.26 | hardwire | like trying to put a square peg in a round hole |
| 19:33.36 | *** join/#asterisk dasenjo (~dasenjo@208.195.214.28) |
| 19:33.38 | hardwire | like trying to convince a monkey to jump off a cliff |
| 19:33.44 | hardwire | like trying to talk to a boss |
| 19:34.05 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 19:34.10 | hardwire | like trying to convince your mind not to look down womens shirts. |
| 19:34.22 | pbxbart | @hardwire Be sure that i know how a pbx is working. |
| 19:34.31 | hardwire | ? |
| 19:34.40 | Blackthorn | I started asterisk with -rvvvv which said it was in level 4 logging/debug mode i guess. I'm finished looking at all the console invo. how do i get it back to the normal? |
| 19:34.51 | Paul[NOC] | set debug |
| 19:34.54 | hardwire | set verbose 0 |
| 19:34.57 | Paul[NOC] | set verbose 0 |
| 19:34.59 | Paul[NOC] | set debug 0 |
| 19:35.14 | hardwire | set boss -10 |
| 19:35.14 | Paul[NOC] | hardwire, man I was happy I knew the answer t oone |
| 19:35.29 | Blackthorn | thanks! |
| 19:35.33 | Blackthorn | that was too easy :P |
| 19:35.46 | *** join/#asterisk Tili (~Tili@202-133-67-172-dialup.sat.net.pk) |
| 19:37.19 | Nugget | http://joelonsoftware.com/items/2005/07/28.html <-- kickass |
| 19:39.30 | anthm | I'm on the phone on sip with the patch and up to the minute cvs head |
| 19:40.00 | harryvv | anyone care to test my sip connection? |
| 19:40.39 | DarthClue | Paul[NOC]: it's not your system, but it may be dtmf from voicepulse. what's your extension and i'll call ya and tell what i did. |
| 19:40.46 | harryvv | just see now if sip can pass though the firewall |
| 19:40.54 | Paul[NOC] | 210 is my extension |
| 19:41.07 | pbxbart | hm.. strange. |
| 19:41.14 | pbxbart | i'll reboot the mashine |
| 19:42.06 | anthm | reboot the phone too for good measure |
| 19:42.32 | mistral | somebody be able to help me with a problem ? |
| 19:43.06 | mistral | in sip.cponf i have a softphone as [1001] it registers etc... |
| 19:43.53 | mistral | but if i us it from entensions.confg like exten => 2000,1,Dial(SIP/1001),20,tr) |
| 19:44.00 | mistral | i get WARNING[12560]: chan_sip.c:1401 create_addr: No such host: 1001) |
| 19:44.10 | harryvv | testing my fw need somone to call my phone and sip takers? |
| 19:44.23 | DarthClue | harryvv: pm me the info, i'll have bashbot call you. |
| 19:44.27 | Paul[NOC] | DarthClue, by all means |
| 19:44.40 | Paul[NOC] | Im on the phone to VoicePulse |
| 19:45.18 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmu9.dialup.mindspring.com) |
| 19:45.32 | Paul[NOC] | They suggest I use sip insteed of iax |
| 19:46.20 | DarthClue | Paul[NOC]: not surprised. they are probably using a newer iax and it could be causing issues with the dtmf transmit. |
| 19:46.33 | DarthClue | Paul[NOC]: just verify with them how they send dtmf |
| 19:46.41 | Paul[NOC] | Sure |
| 19:46.45 | Paul[NOC] | Would you recommend sip? |
| 19:47.07 | DarthClue | since you aren't using the latest HEAD, yes. you'll probably have better luck. |
| 19:47.24 | Paul[NOC] | Ok let me register it with sip then |
| 19:47.31 | DarthClue | we've had a few clients that have done better with sip because of iax conflicts. |
| 19:48.15 | harryvv | Darth, info sent your way :) |
| 19:48.32 | *** join/#asterisk infotek (8096@shell.datasync.com) |
| 19:48.51 | infotek | anyone use Authenticate() sucessfuly? |
| 19:49.26 | ManxPower | infotek: every single day |
| 19:49.40 | Paul[NOC] | lol |
| 19:49.43 | Paul[NOC] | They admitted it was them |
| 19:49.49 | Paul[NOC] | They changed something with dtmf |
| 19:49.49 | ManxPower | exten => 11,1,Authenticate(1234) |
| 19:49.50 | ManxPower | exten => 11,2,DBPut(queue-main/night=yes) |
| 19:49.50 | Paul[NOC] | Thats good |
| 19:50.06 | Paul[NOC] | Now to get this thing to register |
| 19:50.20 | DarthClue | harryvv: k, one sec. |
| 19:50.26 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
| 19:50.28 | harryvv | k |
| 19:50.30 | DarthClue | Paul[NOC]: figures |
| 19:51.09 | infotek | ManxPower: i use "exten => _NXXNXXXXXX,2,Authenticate(666)" but it fails to authenticate |
| 19:51.35 | Paul[NOC] | DarthClue, now getting it to register should be fun |
| 19:51.42 | ManxPower | infotek: I don't know what the problem is then |
| 19:52.11 | *** join/#asterisk ataraxis (~ataraxis@p54AC1766.dip0.t-ipconnect.de) |
| 19:52.12 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 19:52.32 | pbxbart | @anthm ok, it is working now.. strange |
| 19:52.39 | infotek | has anyone had a problem with authenticate not correctly authenticating? |
| 19:52.58 | anthm | well that's good news then |
| 19:53.21 | ataraxis | Hi, can i use asterisk to redirect incoming data calls to pppd, the calls come from a device that is directly connected to a hfc card in nt mode |
| 19:53.53 | pbxbart | can anyone pont me to some info what the difference between addr and Defaddr is? |
| 19:54.10 | Paul[NOC] | Pl |
| 19:54.12 | Paul[NOC] | Ok |
| 19:54.16 | Paul[NOC] | Doesnt register at all |
| 19:55.31 | *** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr) |
| 19:55.43 | jhiver | good evening all |
| 19:58.15 | Hmmhesays | yo |
| 20:00.08 | pbxbart | @anthm: it is not working completely.. calls from client -> asterisk are working fine. But asterisk -> client are not working. i see these xis_xmit warnings again |
| 20:00.27 | harryvv | anyone here care to test my firewall by making a sip call? |
| 20:00.38 | anthm | is it filling in the fromaddr col |
| 20:00.42 | anthm | in the db |
| 20:00.49 | anthm | you said it's realtime right? |
| 20:00.50 | pbxbart | yes |
| 20:00.53 | pbxbart | it is filled |
| 20:01.01 | anthm | that is all my patch does |
| 20:01.07 | anthm | makes it use the db |
| 20:01.10 | pbxbart | ok |
| 20:01.10 | mistral | WARNING[12813]: chan_sip.c:1401 create_addr: No such host: 1002) |
| 20:01.21 | mistral | why do i get that ? |
| 20:01.29 | pbxbart | yes, it does |
| 20:01.46 | bkw_ | the peer/user name is 1002 right? |
| 20:01.54 | mistral | yes |
| 20:04.05 | DarthClue | harryvv: what auth method you using? |
| 20:04.40 | mistral | bkw_: still not sure why i am getting it |
| 20:04.41 | dudes | mistral - have you reloaded sip ? set the username=1002 ? |
| 20:05.00 | harryvv | darth, bjohnson just said i should add externip and localnet ip under [general] which I did |
| 20:05.09 | *** join/#asterisk trimi` (Pharrel@62.162.242.197) |
| 20:05.16 | mistral | dudes: in sip.conf i have a [1002] with a username=1002 in it |
| 20:05.28 | harryvv | authenicace method as md5 in the extentions? |
| 20:05.37 | hardwire | hmm |
| 20:05.40 | hardwire | playboy sent me an email |
| 20:05.49 | Paul[NOC] | forward paul@hostdime.com |
| 20:05.50 | hardwire | I should turn off auto-read while I have a laptop in a meeting |
| 20:06.00 | Hmmhesays | mkay as far as I can tell the small business poweredge 2800 and the medium business poweredge are set up identical in the base configuration, but one is 2K more |
| 20:06.06 | trimi` | Hello, i need some help cuz im a begginer. Can i make an outgoing call if i connect my phone to a FXO port directly (x100p card ) or FXO its only for ansering incoming calls |
| 20:06.07 | *** join/#asterisk jackfiber (~jack@66.96.209.21) |
| 20:06.19 | trimi` | ??? |
| 20:06.21 | hardwire | Paul[NOC]: ok |
| 20:06.27 | jackfiber | hiall, soft-switch.org is down anyone knows where can I get spandsp 2 |
| 20:06.32 | hardwire | that was quite a shock to have happen while in a meeting |
| 20:06.47 | Hmmhesays | fxo ports are for phone lines buddy |
| 20:07.04 | trimi` | ok how can i call with a headset |
| 20:07.05 | Hmmhesays | you plug phones into fxs ports |
| 20:07.07 | trimi` | using a sound card ? |
| 20:07.11 | Hmmhesays | lick it |
| 20:07.17 | Hmmhesays | feel the tingle |
| 20:07.23 | Hmmhesays | trimi`: a softphone |
| 20:09.04 | jackfiber | hey spandsp site is done is there any mirror? |
| 20:09.05 | trimi` | which is the best softphone for linux ? |
| 20:09.06 | hardwire | Paul[NOC]: you get it? |
| 20:09.13 | hardwire | or are you being scolded by the NOC lord? |
| 20:09.38 | harryvv | DarthClue ? |
| 20:10.45 | DarthClue | harryvv: set it up so that your sip guest will let me into the right context. |
| 20:11.19 | harryvv | it has |
| 20:11.51 | harryvv | i see you are atempting to log in |
| 20:12.57 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 20:12.59 | harryvv | yea let me look at something DarthClue |
| 20:13.00 | Paul[NOC] | DarthClue, I got SIP |
| 20:13.07 | Paul[NOC] | But now it doesnt play anything when you call |
| 20:13.17 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:13.23 | trimi` | any1 got a list of good iax providers ? |
| 20:15.49 | DarthClue | Paul[NOC]: one sec, i've been called upon by The Empire. |
| 20:16.15 | xarmiex | i use realtime odbc, is there anyway to get a list of who is registered at the time? sip show peers etc doesnt work anymore... |
| 20:16.22 | mishehu | The Evil Empire or Empire Chicken? |
| 20:16.29 | mishehu | hehehehe |
| 20:17.20 | hardwire | what is the "ht" cpu_flag? |
| 20:18.05 | mishehu | hardwire: hyperthreading or hypertransport perhaps? |
| 20:18.19 | hardwire | not hyperthreading |
| 20:18.26 | hardwire | its a p4.. but the HT isn't on this chip |
| 20:18.33 | hardwire | HT is only on certain p4's AFAIK |
| 20:18.39 | hardwire | its definatly not on the 2.4ghz ones Ihave |
| 20:18.41 | hardwire | or this 1.5ghz |
| 20:18.48 | hardwire | but it is on this 3.0ghz one |
| 20:19.05 | hardwire | I just wish there were a way to check |
| 20:19.15 | hardwire | easily |
| 20:19.18 | hardwire | via google :) |
| 20:19.47 | clive- | does anyone know a way to see if iax2 trunking is working? |
| 20:19.54 | hardwire | iax2 trunk debug |
| 20:20.00 | hardwire | while making a call over the supposed trunk interface |
| 20:20.02 | hardwire | make two calls |
| 20:20.13 | hardwire | then type that in.. it should say 2 active calls.. 2 trunked |
| 20:20.22 | clive- | hardwire I tried that....but no extra output to see.... |
| 20:20.22 | hardwire | also check your bandwidth usage :) |
| 20:20.31 | hardwire | clive-: then its not enabled |
| 20:20.33 | rvhi | hi, trying to find a conference solution in * |
| 20:20.43 | hardwire | meetme |
| 20:20.49 | rvhi | so many ways, meetme, meetme2, app_conference |
| 20:20.55 | clive- | ok,,,so I type iax2 trunk debug while calls are going... |
| 20:20.55 | rvhi | which one is the best? |
| 20:20.56 | harryvv | DarthClue try the other one. |
| 20:21.03 | rvhi | i am using stable |
| 20:21.06 | hardwire | rvhi: are you just trying to hold a conference ? |
| 20:21.11 | hardwire | use meetme. |
| 20:21.36 | rvhi | hardwire: yes just conference for a handful of people |
| 20:21.53 | clive- | hardwire is just says: IAX2 Trunk Debug Requested |
| 20:22.06 | hardwire | make sure there are calls |
| 20:22.10 | hardwire | anyhoot |
| 20:22.12 | hardwire | I am in a meeting :) |
| 20:22.20 | hardwire | and not a meetme one |
| 20:22.36 | hardwire | ugh.. my boss just went to A.D.D. land |
| 20:22.50 | DarthClue | harryvv: no go. can you pastebin your sip.conf ... just the general and guest sections. |
| 20:23.03 | harryvv | sire |
| 20:23.08 | harryvv | sure |
| 20:23.12 | pbxbart | @xarmiex: look at rtcachefriends |
| 20:23.40 | clive- | hmm, 3 calls going on,,,all with the exact same jitter stats,,,,something must be happenning in that trunking stuff.... |
| 20:24.37 | Paul[NOC] | DarthClue, went to sip. it would just be slient then beeping |
| 20:24.41 | Paul[NOC] | went back to iax |
| 20:24.42 | Paul[NOC] | nothing |
| 20:25.53 | *** join/#asterisk Assid (~assid@203.115.64.60) |
| 20:25.56 | DarthClue | Paul[NOC]: yeah, i just tried and got nothing on my end as well. gimme a bit and i'll try to get back in the box and see if i can figure out what is going on. |
| 20:26.32 | Paul[NOC] | Thanks |
| 20:26.50 | harryvv | DarthClue sent it to you |
| 20:27.26 | *** join/#asterisk Connor_ (~billy@198-144-174-5.knx.tn.nxs.net) |
| 20:27.47 | *** join/#asterisk Andrezo (~www@217.129.208.124) |
| 20:27.54 | Andrezo | hi all |
| 20:27.55 | Paul[NOC] | DarthClue, Thanks and I am upgrading the code base too |
| 20:27.59 | Connor_ | Hey, anyone know of a online query/database that I can use to put 2 numbers into and it tell me if if they're local call from 1 to the other? |
| 20:28.10 | Assid | umm... |
| 20:28.16 | Assid | i have a * box behind a nat |
| 20:28.20 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 20:28.34 | Assid | i am trying to connect to it.. and it does.. |
| 20:28.39 | Assid | but when i make /receive a call |
| 20:28.51 | Assid | i can hear the other person.. byt they cant hear me |
| 20:29.02 | Assid | i put myself nat=yes in my sip user account |
| 20:29.06 | pbxbart | bye bye |
| 20:29.08 | *** part/#asterisk pbxbart (~pbx111@p54B031F9.dip0.t-ipconnect.de) |
| 20:29.11 | harryvv | nat problem assid |
| 20:29.18 | ManxPower | ~fxofxs |
| 20:29.19 | jbot | fxofxs is probably An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 20:29.19 | harryvv | im working on it also. |
| 20:29.23 | ManxPower | ~mailinglist |
| 20:29.23 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:29.29 | Andrezo | inside a context, asterisk first tries the extens and then tries the include, altough i have the include BEFORE the extens, why it behaves like that? |
| 20:29.40 | Andrezo | i want * to try the first stuff |
| 20:29.46 | Andrezo | to respect the order |
| 20:30.53 | jackfiber | anyone knows any mirror of spandsp? |
| 20:32.55 | *** join/#asterisk Umaro (~umaro@c-24-20-122-106.hsd1.or.comcast.net) |
| 20:33.09 | *** join/#asterisk Twister (~jason@216.30.232.106) |
| 20:34.35 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-77-210.dsl.pltn13.pacbell.net) |
| 20:36.17 | Twister | hey all..having a bit of a problem getting calls to route by cid..i have this line in a context called ext-did |
| 20:36.19 | Twister | exten => 3045905796,1,SetVar(FROM_DID=3045905796) ; |
| 20:36.19 | Twister | exten => 3045905796,2,Dial(SIP/200&SIP/201&SIP/202&SIP/203) ; |
| 20:36.33 | *** join/#asterisk Mike (~mike@201.135.48.172) |
| 20:36.34 | *** join/#asterisk santiago (~santiago@63.245.86.141) |
| 20:36.38 | Mike | Jul 28 16:27:39 WARNING[4656]: app_db.c:148 put_exec: This application has been deprecated, please use the ${DB(family/key)} function instead. |
| 20:36.38 | Mike | <PROTECTED> |
| 20:36.43 | Mike | any idea whats the new command |
| 20:36.44 | Mike | ? |
| 20:36.48 | Twister | then in my start context that answers all calls i have include => ext-did |
| 20:37.20 | MikeJ[Laptop] | hey Mike, |
| 20:37.27 | MikeJ[Laptop] | give my name back ;) |
| 20:37.28 | Twister | 2005-07-28 16:29:21 SIP/303557... 3045905796 "Charleston WV" <3045905796> s ANSWERED |
| 20:37.32 | Mike | :) |
| 20:37.38 | Twister | thats what i have in my call log |
| 20:37.54 | Mike | anyone knows what dbput is now in -HEAD? |
| 20:38.26 | *** join/#asterisk ginvent (~joseph@adsl-67-121-208-105.dsl.sndg02.pacbell.net) |
| 20:38.34 | ginvent | anyone use sipphone with asterisk? |
| 20:38.38 | RaYmAn-Bx | Mike: Haven't read the UPGRADE.txt, eh? :P |
| 20:39.17 | Andrezo | ginvent, me |
| 20:39.22 | twisted[asteria] | Hey everyone |
| 20:39.28 | twisted[asteria] | bug 4832 needs testers and input |
| 20:39.33 | twisted[asteria] | http://bugs.digium.com |
| 20:40.32 | Connor_ | Hey, anyone know of a online query/database that I can use to put 2 numbers into and it tell me if if they're local call from 1 to the other? |
| 20:42.38 | *** part/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:43.30 | *** join/#asterisk ellvis (~evills@adsl-data-148.84-47-83.telecom.sk) |
| 20:43.34 | ellvis | hi people |
| 20:43.40 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:44.13 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 20:44.27 | xarmiex | http://members.dandy.net/~czg/lca_prefix.php |
| 20:44.32 | xarmiex | connor |
| 20:44.34 | xarmiex | go there |
| 20:45.04 | xarmiex | just put in the npa and nxx and hit submit |
| 20:45.04 | harryvv | anyone here care to send a test sip to my asterisk, need to test fw |
| 20:45.41 | ellvis | i am having troubles to call to BRI line with DDI numbering while calling from that line using asterisk is working. anyone have experience with DDI lines, please? |
| 20:45.56 | FarrisG | is there anything similar to IPSwitchBoard that runs on linux? |
| 20:46.04 | xarmiex | then click the NPA-NXX |
| 20:46.22 | Connor_ | xarmiex, Yea.. I've used them for sometime.. Just trying to figure out how I can make a script that allows my customers to put in their phone number and it spits out which # is local for them to call into. |
| 20:47.00 | ginvent | Anyone use sipphone.com service? |
| 20:47.21 | ellvis | ginvent, me |
| 20:47.37 | xarmiex | conner : use his xml interface |
| 20:47.45 | xarmiex | then you make your own scrip |
| 20:47.45 | wunderkin | what file type do you think would have the smallest file size? mp3? |
| 20:48.02 | xarmiex | we use his xml interface for our customers |
| 20:48.10 | xarmiex | with a php script |
| 20:48.23 | Connor_ | can you shoot me a copy of the php script? |
| 20:48.42 | wunderkin | as far as audio goes of course |
| 20:48.52 | wunderkin | stored in raw format now |
| 20:49.35 | wunderkin | think ill just mix them and put it into mp3 |
| 20:50.21 | Ahewes | Anyone know if the latest zaptel stuff require gcc 3.4? |
| 20:50.25 | *** join/#asterisk dasuberdavid (~david@207.111.174.1) |
| 20:51.40 | ginvent | Anyone use sipphone.com service with incoming calls working? |
| 20:54.41 | Andrezo | ginvent, just paste your extensions.conf file on http://pastebin.ca/ |
| 20:54.50 | Andrezo | i'll already told you what you have to do |
| 20:55.07 | Andrezo | you have to explain more |
| 20:55.13 | Andrezo | or show more |
| 20:55.31 | Darwin35 | this is a hold up give me all your voip phones and the rights to asterisk and no one gets hurt |
| 20:56.09 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:56.36 | Andrezo | i think he knows how to hide the passwords :) |
| 20:59.31 | Corydon-w | Ahewes: nah, it requires gcc 4.2 |
| 21:01.23 | Andrezo | with alan cox patches made with diff version 1.2,3.6 pre-release |
| 21:01.25 | harryvv | anyone here care to send a test sip to my asterisk, need to test fw |
| 21:02.10 | *** part/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com) |
| 21:04.32 | Ahewes | O.K. My problem was with this new ubuntu install. Why do distros install multiple versions of gcc? Anyway, zaptel builds fine with either gcc. |
| 21:04.34 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 21:07.42 | Nugget | Because Linux is a corpulent, poorly-managed pile of poo. |
| 21:07.57 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 21:08.53 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 21:09.56 | harryvv | anyone here care to send a test sip to my asterisk, need to test fw |
| 21:10.06 | harryvv | test my firewall that is |
| 21:11.57 | *** join/#asterisk raptorrat (~ucs_rat@ab1-1-26.shsu.edu) |
| 21:15.10 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 21:16.38 | *** join/#asterisk sd-tux (sd@2001:4ca0:0:fe00:0:0:a96:3f18) |
| 21:16.54 | harryvv | tired of this second hand pot wafting in from the outside. |
| 21:21.46 | *** join/#asterisk pdracevich (~bob@210.54.249.228) |
| 21:21.55 | pdracevich | Hello, all. |
| 21:22.05 | doughecka_ | can I tell a T1 to not pickup a particular line? |
| 21:22.14 | doughecka_ | its a fax line, but I still want to be able to dial out it |
| 21:22.21 | pdracevich | I need some help seetinh up the confieration files for a TE110P card for New Zealand |
| 21:22.38 | *** join/#asterisk L|NUX (linux@202.5.146.154) |
| 21:25.53 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 21:27.50 | Bile_One | Anyone able to help me understand why I can send a fax but not recieve one? |
| 21:28.04 | Katty | mew |
| 21:29.59 | Bile_One | I can see that when the fax detect is good it sends the fax call to the correct extension, and I have the extension on a Leadtek bva 8055, it answers the call, and says it is connecting, but from CLI I see that the hang-up is occuring? |
| 21:30.03 | Nugget | because faxing over voip is unreliable. |
| 21:31.04 | Bile_One | yes but I can send, why not recieve? I would think if I could not do either? |
| 21:31.27 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 21:33.00 | *** part/#asterisk raptorrat (~ucs_rat@ab1-1-26.shsu.edu) |
| 21:33.05 | harryvv | DarthClue whats next to get this going? |
| 21:37.51 | *** join/#asterisk hermie (~nick@24-236-167-53.dhcp.bycy.mi.charter.com) |
| 21:38.00 | DarthClue | harryvv: i need to see your configs so i can try and determine what is happening. can you give me access to the box? if so, msg me and i'll get in and take a look. |
| 21:42.43 | *** join/#asterisk zotz (~zotz@24.231.36.100) |
| 21:45.59 | *** join/#asterisk emrah (~emrah@aslann.aslann.net) |
| 21:46.01 | implicit | hows it going |
| 21:46.42 | emrah | Hello |
| 21:47.27 | *** join/#asterisk shadeboboo (~shadebob@212.217.71.121) |
| 21:47.41 | emrah | I'm having a strange problem with chan_local. Anyone can help me with this message? |
| 21:47.44 | emrah | Jul 28 23:43:37 NOTICE[12202]: chan_local.c:455 local_alloc: No such extension/context callingcardlocal@default creating local channel |
| 21:48.18 | emrah | I want to dial LOCAL/local/${EXTEN} |
| 21:48.34 | shadeboboo | Hi, I have a te110p and I have to connect to a rs232 connection (E1). Is someone have a wiring scheme? |
| 21:49.18 | doughecka_ | whats it pluggin into? |
| 21:50.10 | shadeboboo | Scheme is : sagem dcn pri <rs232>-------------<rj45> te110p |
| 21:50.30 | shadeboboo | i don't known how i can made this cable :( |
| 21:53.32 | jake1932 | shadeboboo: do you have the manual to the sagem dcn pri? |
| 21:54.34 | shadeboboo | no it's the problem :s |
| 21:55.39 | jake1932 | shadeboboo: your best bet might be to contact sagem tech support and get a wiring diagram |
| 21:55.58 | jake1932 | shadeboboo: the te110p is pretty standard |
| 21:57.47 | *** join/#asterisk th (~th@montana.hbsn.de) |
| 21:58.26 | fman | guys, I had a problem when I compile asterisk it dosn't build chan_zap |
| 21:58.43 | Qwell | fman: Did you build zaptel? |
| 21:58.45 | fman | I've got zaptel installed in the kernel and running |
| 21:58.51 | fman | it registers the cards |
| 21:59.02 | fman | I'm going to build asterisk again to confirm |
| 22:00.19 | *** part/#asterisk mkrufky (~mk@68.160.103.77) |
| 22:00.51 | shadeboboo | jake19232 : yes i known te110p wiring is : 1.2. rx 4.5. tx but i don't known how I can known sagem wiring scheme. It's 10pm here ;) |
| 22:01.22 | *** part/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 22:02.50 | ellvis | i am having troubles to call to BRI line with DDI numbering while calling from that line using asterisk is working. anyone have experience with DDI lines, please? |
| 22:07.14 | *** join/#asterisk moy (~kvirc@201.135.113.46) |
| 22:08.36 | *** join/#asterisk shadeboboo (~shadebob@212.217.71.121) |
| 22:10.02 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 22:10.05 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 22:10.45 | Veto | The 7960 doesn't support g.726? |
| 22:10.52 | *** part/#asterisk Bile_One (~bile_one@adsl-208-191-252-109.dsl.ltrkar.swbell.net) |
| 22:12.18 | shido6 | ulaw and g729 i thought |
| 22:12.33 | emrah | No one can just help me? |
| 22:12.41 | shido6 | emrah, whats wrong? |
| 22:12.43 | emrah | I want to dial LOCAL/local/${EXTEN} |
| 22:12.49 | emrah | and I'm having a strange message |
| 22:12.56 | shido6 | local local? |
| 22:12.58 | shido6 | really? |
| 22:13.20 | Veto | I think you are right...was hoping for g.726 for something less than ulaw (bandwidth), but not a very low bandwidth codec. |
| 22:14.03 | pdracevich | \ |
| 22:14.03 | emrah | sorry, I want to dial callingcard. local/callingcard, but it tries to dial local/callingcard@default. |
| 22:14.04 | pdracevich | exit |
| 22:14.48 | Twister | what do i need to do in order to enable fax detection on sip channels? |
| 22:15.43 | *** part/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu) |
| 22:16.30 | nDuff | Twister, there's an application for that -- something like NvFaxDetect |
| 22:16.41 | nDuff | Twister, and a NvBackgroundDetect version |
| 22:16.46 | Twister | ya |
| 22:16.49 | Twister | i emailed the creater |
| 22:16.52 | Twister | just a few moments |
| 22:16.59 | fman | hmm |
| 22:17.01 | nDuff | Twister, but that said, I've never had luck with faxing over IP. |
| 22:17.04 | fman | looks like it built it that time |
| 22:17.08 | fman | but asterisk isn't starting |
| 22:17.23 | nDuff | Twister, actually bought a T1 card and a channel bank for the fax lines. |
| 22:17.30 | fman | <PROTECTED> |
| 22:17.30 | fman | <PROTECTED> |
| 22:17.30 | fman | <PROTECTED> |
| 22:17.30 | fman | <PROTECTED> |
| 22:17.42 | fman | so, definatly tried to load the zap module |
| 22:17.51 | fman | but bitching about channel type |
| 22:22.35 | fman | hmm, so close :) |
| 22:24.43 | blitzrage | ~seen junk-y |
| 22:24.46 | jbot | junk-y is currently on #asterisk. Has said a total of 14 messages. Is idling for 18h 16m 2s |
| 22:24.51 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 22:24.54 | blitzrage | JunK-Y: hey! |
| 22:25.03 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:25.03 | JunK-Y | y0 |
| 22:25.17 | blitzrage | how goes? |
| 22:26.01 | ellvis | fman, and you run 'ztcfg' before starting asterisk? |
| 22:26.01 | JunK-Y | nothing with my bro |
| 22:26.13 | JunK-Y | mouhahaa a co-worker of him was sleeping at his job |
| 22:26.15 | JunK-Y | mouaa |
| 22:26.28 | blitzrage | JunK-Y: that's the second time I've heard of someone doing that today! |
| 22:26.45 | JunK-Y | mouhaha |
| 22:26.47 | blitzrage | JunK-Y: suppose you didn't get anywhere with 'show dialplan <context> like <regex>' ? |
| 22:26.48 | JunK-Y | its so funny. |
| 22:26.57 | blitzrage | yah, tis funny for sure :) |
| 22:27.11 | blitzrage | my buddy Kev is ALLOWED to do it at his job on night shifts |
| 22:27.22 | blitzrage | as long as he wakes up to answer the phones when they ring |
| 22:27.24 | *** join/#asterisk dysan (~ack@202.37.224.27) |
| 22:27.27 | JunK-Y | ya, but he works day shifts. |
| 22:27.34 | blitzrage | yah, thats totally funny :) |
| 22:27.54 | Qwell | man...I totally nodded off at work today too |
| 22:27.56 | dysan | im trying to get voicemail sent to my email address, how do i tell asterisk to use a different mailserver on the netwrok? |
| 22:28.09 | Qwell | only for a few seconds, but damn |
| 22:28.42 | Qwell | tip: don't watch a 4gb `cp -vr` |
| 22:29.00 | blitzrage | lol |
| 22:29.12 | blitzrage | dysan: look at the voicemail.conf.sample file - it'll show you |
| 22:29.50 | blitzrage | dysan: unless Asterisk will only use a local server - honestly not too sure |
| 22:30.09 | blitzrage | anyone? Bueller? |
| 22:30.22 | Qwell | it has a mailcmd in the conf, doesn't it? |
| 22:30.43 | shido6 | brb |
| 22:31.03 | dysan | yeah it has mailcmd commented out |
| 22:31.13 | dysan | points at /usr/sbin/sendmail -t |
| 22:34.28 | Assid | how do disable voicemail from emailing the vm |
| 22:34.32 | Assid | at all |
| 22:34.37 | Assid | i dont want any notification or anything |
| 22:36.11 | fearnor | remove email addy |
| 22:36.14 | _DAW | Assid - read comments in voicemail.conf |
| 22:39.20 | *** join/#asterisk lters (~lters@mrtcdsl-034.mis.net) |
| 22:41.33 | fman | ok, guys where do I get the zaptel tools source from |
| 22:41.45 | fman | looks like this debian package isn't installing ztcfg |
| 22:42.16 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 22:42.45 | Twister | ... |
| 22:42.49 | Twister | www.asterisk.org |
| 22:42.52 | Twister | click downloads |
| 22:43.20 | jskcr | hy all anyone using the odbc extended storage patch? |
| 22:43.28 | *** join/#asterisk dasenjo (~dasenjo@63.245.87.180) |
| 22:44.01 | fman | is that the tools and the kernel module? |
| 22:44.08 | Coriantum | Can I put a || in a gotoif? |
| 22:45.02 | blitzrage | Coriantum: what do you mean? |
| 22:45.05 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:45.21 | blitzrage | GotoIf($[ <condition> ]?true:false) <-- the format |
| 22:45.24 | Coriantum | I want to evaluate two expressions inside a GotoIf |
| 22:45.36 | blitzrage | Coriantum: ahhhh, I think you can do that |
| 22:45.57 | blitzrage | Coriantum: check the ./docs dir in your Asterisk source, it'll tell you about condition statements |
| 22:46.08 | Coriantum | GotoIf($[ <condition1> ] || $[ <condition2> ]?true:false) ? |
| 22:46.38 | Coriantum | great idea ;) |
| 22:48.36 | *** join/#asterisk santiago (~santiago@208.195.214.92) |
| 22:51.22 | blitzrage | Coriantum: I think that'd be the format, but double check witht he README.variables document (I believe) |
| 22:52.42 | fman | ok buiulding the zaptel stuff |
| 22:52.53 | fman | but it complains that I don't have the kernel sources for the currnt kernel installed |
| 22:52.58 | Ariel_ | hello very one |
| 22:53.20 | fman | /usr/src/linux is pointing towards /use/src/kernel-source-2.6.11/ |
| 22:53.20 | Ariel_ | very/every |
| 22:53.21 | blitzrage | Coriantum: I think it all needs to go inside a single $[ ] (which means condition) |
| 22:53.25 | fman | this is anoying |
| 22:53.39 | blitzrage | fman: is 2.6.11 your current kernel? (uname -a) |
| 22:53.55 | fman | 2.6.11-powerpc |
| 22:54.20 | hardwire | this dialplan is incredibly complicated so far ;) |
| 22:54.27 | hardwire | 26 people.. 200 extensions |
| 22:54.28 | hardwire | heh |
| 22:54.31 | fman | I have a link pointing 2.6.11-powerpc towards kernel-source-2.6.11 |
| 22:54.56 | fman | 2.6.11-powerpc -> kernel-source-2.6.11/ |
| 22:57.42 | fman | renamed kernel-source-2.6.11 |
| 22:57.47 | fman | to 2.6.11-powerpc |
| 22:57.50 | fman | still no luck |
| 22:58.34 | *** join/#asterisk laserfox (~jimbob@81-179-127-14.dsl.pipex.com) |
| 22:59.12 | Twister | wtf have you got 200 extensions for 26 peeps? |
| 22:59.50 | *** join/#asterisk marv (~ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) |
| 23:01.28 | hardwire | man the new grandstreams are pretty |
| 23:01.38 | hardwire | did they make a sidecar for it? |
| 23:02.31 | hardwire | thinking about getting some budgettones for a hotel |
| 23:02.35 | hardwire | anybody want to advise against it? |
| 23:02.38 | hardwire | need 25 |
| 23:03.13 | jontow | anyone built cvs HEAD on NetBSD 2.0.2 recently? |
| 23:03.37 | doughecka_ | hardwire: get the sipura phones |
| 23:03.40 | doughecka_ | alot nicer |
| 23:03.43 | hardwire | oh yeah |
| 23:03.49 | DarthClue | hardwire: get polycom ip301s or ip501s. you'll be happier and so will they. |
| 23:03.55 | hardwire | $85 as a reseller |
| 23:04.16 | doughecka_ | sipura's are just about as cheap |
| 23:04.21 | doughecka_ | i'd buy one just to see if it will work |
| 23:04.43 | hardwire | hardwire: thats what the sipuras cost |
| 23:04.50 | hardwire | I like the budgettone |
| 23:04.51 | hardwire | its simple |
| 23:04.53 | doughecka_ | ah |
| 23:04.54 | hardwire | big MWI |
| 23:04.56 | hardwire | no lines |
| 23:04.56 | doughecka_ | true |
| 23:05.03 | hardwire | no stupid people pushing buttons |
| 23:05.18 | harryvv | bugetone for small simple offices |
| 23:05.23 | hardwire | no reason to give people a reason to think "these cheap bastards didn't even use all these extra lines" |
| 23:05.24 | jontow | i think the budgetones definitely have the market of "unable to play with shit." |
| 23:05.25 | jontow | :) |
| 23:05.29 | hardwire | harryvv: or a hotel room :) |
| 23:05.41 | harryvv | good idea |
| 23:05.46 | *** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET) |
| 23:05.47 | jontow | the BT10x's i have would work well in a hotel room.. i think so, anyway :) |
| 23:06.11 | doughecka_ | and if someone gets bored in a hotel room.... |
| 23:06.18 | DarthClue | hardwire: ip301s will make you happier. the budgetones are a pain. |
| 23:06.30 | hardwire | link me yo |
| 23:06.51 | hardwire | nm voip-supply had a link |
| 23:06.56 | hardwire | http://www.voipsupply.com/product_info.php?products_id=757 |
| 23:06.57 | hardwire | see |
| 23:07.00 | DarthClue | tritechcoa.com - 115ish |
| 23:07.01 | hardwire | those look less sturdy |
| 23:07.02 | hardwire | cheap |
| 23:07.34 | hardwire | the zultys zip2 |
| 23:07.38 | hardwire | now thats in the price range |
| 23:08.23 | *** join/#asterisk doughecka_ (~Miranda@doughecka.user) |
| 23:08.32 | DarthClue | the 301 looks cheap? i think you may need to get a better look. |
| 23:08.37 | colinm_ | imo, the polys look way better in person than in the photos |
| 23:08.56 | hardwire | yeh |
| 23:09.08 | hardwire | we have 50 rooms to install into |
| 23:09.13 | hardwire | I migh tjust daisy chain it :) |
| 23:09.20 | hardwire | quack |
| 23:09.43 | DarthClue | hardwire: http://www.tritechcoa.com/product/791436.html |
| 23:10.09 | DarthClue | they don't have a picture there, but it's the same phone from voipsupply. the 301 will make the hotel think they are getting very nice phones. |
| 23:10.15 | Darwin35 | it dials 1 exten that dials all the phones |
| 23:10.29 | Darwin35 | and I just used the base pc board |
| 23:10.30 | hardwire | Darwin35: hey.. I need to make an auto answer intercom phone as well |
| 23:10.35 | hardwire | s/hey/yeh/ |
| 23:10.58 | *** join/#asterisk anthm (~anthm@000-419-125.area4.spcsdns.net) |
| 23:10.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:11.00 | DarthClue | hardwire: and for 50 phones, you won't pay shipping if they still offer that, and you can probably get them down a couple bucks per phone. |
| 23:11.30 | hardwire | yeh |
| 23:11.33 | hardwire | I usually go through abp |
| 23:11.43 | hardwire | trying to maintain that so I can become a better reseller through them |
| 23:11.54 | hardwire | also trying to get these dingotel adapters as their own channel |
| 23:11.58 | hardwire | or as a hacked alsa channel |
| 23:12.16 | hardwire | I need to incorporate the USB HID for issuing a PTT trigger |
| 23:12.32 | hardwire | I might just use dtmf detection to turn it on and off |
| 23:12.41 | hardwire | esp if its all out of band |
| 23:14.06 | yaaar | catch you guys tomorrow... |
| 23:21.43 | *** join/#asterisk IronHelix (~irc@ool-45785cfe.dyn.optonline.net) |
| 23:22.18 | *** join/#asterisk thal (~thalunil@walledcity.de) |
| 23:22.43 | _DAW | a |
| 23:22.44 | hardwire | somebody pop my back |
| 23:24.33 | harryvv | hardwire, thats 30 dollars a pop |
| 23:24.52 | harryvv | Thats what my chiropractor charges me :) |
| 23:26.07 | DarthClue | $30 per pop, average pop is 5 minutes, i'm in the wrong business. |
| 23:29.35 | hardwire | harryvv: yeh.. weird huh |
| 23:31.31 | *** join/#asterisk fugitivo (~ajf@201.255.100.39) |
| 23:31.34 | fugitivo | buenas |
| 23:31.37 | fugitivo | oops, hello |
| 23:32.38 | hardwire | DarthClue: http://www.tritechcoa.com/product/563961.html |
| 23:32.43 | hardwire | those are my new favorite PS |
| 23:33.28 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
| 23:34.32 | hardwire | DarthClue: they are techdata distributers |
| 23:34.39 | hardwire | we are techdata distributers as well |
| 23:35.08 | syle | hmmm |
| 23:35.11 | syle | how are the 301;s |
| 23:35.15 | syle | think i;ll get one |
| 23:35.33 | hardwire | the antec phantom 500 |
| 23:35.33 | DarthClue | they are better then anything from grandstream. and i have 2 grandstreams. |
| 23:35.35 | hardwire | just plain amazing |
| 23:35.51 | hardwire | techdata pricing for polycom 301 110 |
| 23:35.53 | hardwire | 43 available |
| 23:36.19 | syle | good price |
| 23:36.41 | syle | think that 20 bucks cheaper that voipsupply |
| 23:36.56 | jontow | yep.. don't understand, heheh |
| 23:38.08 | syle | Please Note: Polycom phones are not supported under Asterisk Open Source PBX. |
| 23:38.17 | syle | got to be kidding me |
| 23:39.01 | syle | they probably mean not a supported partner |
| 23:39.06 | syle | http://www.voipsupply.com/product_info.php?products_id=817 |
| 23:39.25 | colinm_ | right. polycom won't help you. |
| 23:40.13 | *** join/#asterisk MasterYoda (~mnicholso@MasterYoda.sustaining.supporter.pdpc) |
| 23:40.38 | MasterYoda | it seems that asterisk is not sending SIGHUP to my AGI scripts on hangup |
| 23:41.05 | Delta34 | anybody here has any iax2 shared dial plan knowledge? |
| 23:41.18 | Delta34 | need a little help setting this up |
| 23:41.27 | MasterYoda | anyone have any experience with this? |
| 23:41.40 | MasterYoda | Delta34: what do you mean by IAX2 shared dialplan? |
| 23:42.04 | syle | delta34 : http://www.voip-info.org/tiki-index.php?page=Asterisk%20-%20dual%20servers |
| 23:42.09 | syle | everything you need is there |
| 23:42.33 | jontow | woohoo.. got chan_zap.c to build on NetBSD 2.0.2 |
| 23:42.42 | jontow | suppose i oughtta submit a patch for that one.. real simple fix |
| 23:42.52 | jontow | sys/signal.h is included where pthread.h and signal.h oughtta be instead |
| 23:43.07 | Qwell | ~asstricks |
| 23:43.07 | jbot | rumour has it, asstricks is #asstricks, the underground gay Asterisk channel. Be afraid, very afraid |
| 23:43.24 | syle | why don;t you throw away netbsd and go with a real OS like freebsd |
| 23:43.35 | jontow | what've you got against netbsd? |
| 23:43.42 | syle | ports collection |
| 23:43.51 | jontow | so pkgsrc is bad? |
| 23:44.01 | syle | well count how many ports it has |
| 23:44.47 | Delta34 | syle, so which example do u recommend? the switch object approach looks cool |
| 23:45.16 | jontow | so.. its real difficult to get thousands of applications to build on 20+ archs simultaneously |
| 23:45.19 | jontow | and they STILL do a good job |
| 23:45.28 | syle | delta34: i went with example 2 personally |
| 23:45.43 | syle | i think user should actually be username though |
| 23:45.44 | jontow | im running it on an embedded box, one meant to run linux.. but it runs netbsd quite well |
| 23:46.19 | Delta34 | syle, so all your 7xxx extensions are on server A and all 8xxx extensions are on server B |
| 23:46.27 | syle | yep |
| 23:46.39 | syle | my setup was ADSL nat at home to public static ip address |
| 23:46.41 | jontow | its my NAT router and firewall as well.. i like ipfilter/ipnat a hell of a lot |
| 23:46.49 | syle | so that example was perfect |
| 23:47.06 | syle | ipfilter is on fbsd to |
| 23:47.38 | syle | yes i agree i hate linux iptables, ipfilter so much more clean |
| 23:47.49 | jontow | that im well aware of.. i've been using BSD for 7 years |
| 23:47.57 | Delta34 | so only one iax server has a register statement, or those both need to register with one another? |
| 23:48.02 | jontow | i use them both.. but netbsd for a lot of reasons is quite well suited to the box im running it on |
| 23:49.24 | opus_ | anyone have polycom dtmf problems? |
| 23:49.28 | syle | i just find more people work on fbsd, so if something comes out it will get in the ports collection alot sooner |
| 23:49.37 | opus_ | in CVS head? |
| 23:50.13 | MasterYoda | opus_: what kind of problems are you having? |
| 23:50.15 | jontow | yeah, ports still has a lot of issues, too; pkgsrc is designed clean enough to also work on freebsd, irix, solaris, macos X, etc; i've got no problems with uniformity when it makes things easier |
| 23:50.24 | syle | to you i guess that don;t really matter since you seem to be compiling shit from .tar.gz balls anyway |
| 23:50.37 | jontow | really. |
| 23:50.50 | jontow | and ports doesn't do that? |
| 23:51.24 | syle | you didn;t seriously ask me that |
| 23:51.27 | jontow | can i remind you that my desktop is freebsd, 80% of the servers i maintain are freebsd, yet my laptop, router, DNS servers all run netbsd |
| 23:51.28 | syle | i;ll ignore that |
| 23:51.32 | SwK_ | ph33r th3 b33r |
| 23:52.03 | jontow | do what ya gotta, but i've got no problems with freebsd.. im just saying that discounting netbsd as worthless is just.. stupid :) |
| 23:52.25 | syle | ok i didn;t realize you were running fbsd servers |
| 23:52.34 | syle | well then, good job on the port |
| 23:52.35 | jontow | well over 30 of them |
| 23:52.53 | jontow | by trade i'm a freebsd admin |
| 23:52.58 | syle | large website ==vlanned? |
| 23:53.12 | jontow | blind prejudice bothers me, though |
| 23:53.15 | opus_ | master : DTMF tones are not being sent out to my provider correctly. everything is dtmf=rfc |
| 23:53.18 | opus_ | rfc... |
| 23:53.18 | jontow | and i'll go to great extent to laugh at it :) |
| 23:53.48 | opus_ | master - There is a bug reported on bugs.digium.com but its been sidetrack and is very old, like 1.0.7 |
| 23:54.36 | syle | jontow: what is running 30 servers? |
| 23:54.39 | syle | large website? |
| 23:54.42 | opus_ | http://bugs.digium.com/view.php?id=3847 |
| 23:54.43 | hardwire | I am so opposed to 3 digit extensions |
| 23:54.53 | jontow | it isn't a single application |
| 23:55.06 | jontow | or service.. its all services for a mid-sized ISP |
| 23:55.09 | syle | i;m opposed to 3 digit account codes |
| 23:55.12 | syle | 4 is good |
| 23:55.14 | opus_ | how can I download CVS stable? |
| 23:56.04 | syle | cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 23:56.05 | hardwire | syle: well |
| 23:56.11 | hardwire | I ran out |
| 23:56.14 | hardwire | I need 4 digit |
| 23:56.17 | hardwire | 1 for location |
| 23:56.20 | hardwire | 2 for function |
| 23:56.28 | hardwire | 3-4 for person |
| 23:56.31 | hardwire | it just makes sense |
| 23:56.41 | laserfox | hi all, :) |
| 23:56.44 | hardwire | because we have 5 functions all of which are in different locations |
| 23:56.51 | hardwire | I think I just need 4 digit extensions.. |
| 23:57.06 | hardwire | the problem is.. we can't then afford to get mmatching DID's |
| 23:59.20 | opus_ | bugs bugs bugs |
| 23:59.30 | opus_ | is cvs stable pretty good? |
| 23:59.58 | syle | well maybe a separate field in database might help with that |