| 00:00.10 | martyn-dev | I have a question about it... |
| 00:01.04 | NovceGuru | man i've really been missing out |
| 00:01.18 | NovceGuru | I never realized how badass blackberry's email system was |
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| 00:16.11 | Magicblaze007 | In our company, we have more than 1k phones. I am thinking of adding an ATA adaptor to every phone and connect to asterisk. Is this a bad idea? (I need minimal functionality on each phone, just make and receive calls, nothing more for starters). |
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| 00:20.10 | denon | Magicblaze007: that is a very bad idea |
| 00:20.38 | denon | you'd either want to use a channelbank to hook up analog phones, replace them with sip phones, or using a multi-port ata at very least |
| 00:20.46 | denon | one ata per phone would be a management nightmare |
| 00:21.06 | denon | if you really prefer to keep the analog phones, you'd like want to use channelbanks |
| 00:21.20 | denon | likely |
| 00:22.04 | Magicblaze007 | The problem is that I can not get new wiring done in these buildings. |
| 00:22.22 | denon | they dont have computers? you can get phones with an ethernet passthrough |
| 00:22.48 | denon | (so you only need 1 drop per office) |
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| 00:23.17 | Magicblaze007 | I can assume that they have ethernet ports. |
| 00:23.25 | denon | right |
| 00:23.30 | denon | if you dont have ethernet, you can do channelbanks |
| 00:25.51 | Qwell | if you have rj11 |
| 00:26.53 | denon | yeah, if they have no computers/ethernet at all .. and just one rj11 .. just hook up channel banks |
| 00:27.05 | Magicblaze007 | I just have ethernet ports in rooms, nothing else. |
| 00:27.10 | denon | unless you want data, then you can do cobbled up dsl-like stuff, like hotels do |
| 00:27.16 | denon | oh, that's easy then |
| 00:27.23 | denon | do a sip phone with a second ethernet port in the back |
| 00:27.42 | Magicblaze007 | That's why I was thinking of ATAs. just an ata in every room. |
| 00:27.51 | Magicblaze007 | But then it's a management nightmare probably. |
| 00:27.53 | denon | that'd be a nightmare to manage |
| 00:28.09 | Magicblaze007 | denon: I completely agree. |
| 00:28.15 | denon | no budget for sip phones? |
| 00:28.21 | Magicblaze007 | I think I'll try to see if they will give me cheap ip phones. |
| 00:28.42 | denon | well .. dont go too cheap |
| 00:28.48 | denon | a thousand budgetones would be worse :) |
| 00:28.56 | Magicblaze007 | lol, yes indeed. |
| 00:29.03 | denon | but like a polycomm or snom or somethin |
| 00:29.13 | Magicblaze007 | any good cheap but good ip phone recommendations? |
| 00:29.14 | denon | maaaybe a linksys spa941 or something |
| 00:29.26 | Magicblaze007 | I don't need any fancy functionality at all, call and receive that's it. |
| 00:29.28 | denon | that linksys would be as low as I go |
| 00:29.48 | Magicblaze007 | I saw that phone earlier... |
| 00:30.31 | EmleyMoor | I'm thinking 942s for home |
| 00:30.31 | x86 | I know this probably isn't the best place to ask, but i'm playing with FreePBX and I can't get Asterisk to see any of the SIP peers (phones) i've added |
| 00:30.35 | denon | a few bucks more and you can get POE and a passthrough and stuff |
| 00:30.53 | denon | x86: try #freepbx, that's in fact the best place to go |
| 00:31.02 | x86 | it doesn't look like asterisk is setup for realtime, and none of the DB tables look like a realtime-compatible table for pulling SIP peers into asterisk |
| 00:31.07 | x86 | denon: rock on |
| 00:31.10 | Magicblaze007 | denon: any other recommendations? I think spa941 is a good choice. |
| 00:32.44 | EmleyMoor | Still, I also like the fact that you can use traditional phones - I am using one dating from 1971 |
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| 00:35.29 | Magicblaze007 | Why is a ATA harder to maintain than the IP Phone? Isn't it almost the same thing? Just wanted to know the reasons... |
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| 00:39.15 | denon | Magicblaze007: well, with a sip phone you have a screen, when you call the user and ask them to give you info .. |
| 00:39.22 | denon | instead of messing with #123 kind of commands |
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| 00:39.25 | Magicblaze007 | has anyone here bought 1k voip phones at a time? What is a realistic discount one can expect? |
| 00:39.50 | EmleyMoor | With an ATA all you can really ask is "does the phone have dialtone?" |
| 00:39.51 | denon | Magicblaze007: we've sold large numbers, discount depends heavily on the model, terms you're paying with, etc |
| 00:40.39 | Magicblaze007 | denon: With ATAs, they can just tell me the IP. I can even put a static ip per room and don't even have to ask them...what am I missing. |
| 00:40.45 | denon | Magicblaze007: sip phone also gives native hold/transfer/conference/etc, which I know you're not overly interested in .. but .. |
| 00:41.00 | denon | Magicblaze007: it's when it's acting up .. |
| 00:41.03 | denon | and forgets it's IP |
| 00:41.06 | denon | and you can't get at it |
| 00:41.12 | denon | or whatever other weird thing happens |
| 00:41.15 | Magicblaze007 | I see. |
| 00:41.28 | denon | it's also more clutter, but that might not matter |
| 00:41.52 | denon | with a sip phone you can do POE and have pretty minimal amount of cables/boxes/etc |
| 00:42.04 | Magicblaze007 | denon: how much is the margin in ip phones? with 1k phones, you think one can get a 10% reduction in price? 50%? any clues ? Something like linksys SPA941? |
| 00:42.18 | denon | Magicblaze007: 10-15% maybe |
| 00:42.30 | Magicblaze007 | Thanks. |
| 00:42.31 | Magicblaze007 | That helps. |
| 00:42.34 | denon | I could look up numbers for you, but not at the moment (we're a linksys reseller, among other things) |
| 00:42.44 | Magicblaze007 | I've to go, thanks for all the help. |
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| 00:43.07 | denon | sure, have a good one |
| 00:46.24 | Dovid | hi |
| 00:46.44 | Dovid | can I do Set(foo=bad,1=a) or they all need to be on seperate lines ? |
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| 01:07.55 | CapriCoRN^80 | hi |
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| 01:27.39 | CapriCoRN^80 | I am working on Mv-370. I need some help from LAN to Mobile call |
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| 01:28.35 | CapriCoRN^80 | its working when i use static URL and static call num settings in MV 372 |
| 01:28.41 | CapriCoRN^80 | its working when i use static URL and static call num settings in MV 370 |
| 01:29.28 | CapriCoRN^80 | but when i enter * and # in URL and num and i dial it give me busy tone |
| 01:32.30 | CapriCoRN^80 | need some help in that |
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| 01:46.31 | CapriCoRN^80 | hi again |
| 01:46.58 | CapriCoRN^80 | its working when i use static URL and static call num settings in MV 370 , but when i enter * and # in URL and num and i dial it give me busy tone |
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| 01:50.15 | kakashka | hi! =) sorry for lame question. can i make calls to jabber client via asterisk? e.g. can it replace xlite? |
| 01:51.01 | kakashka | or there is only notifications feature? :> |
| 01:52.17 | CapriCoRN^80 | xlite is softphone. you can use it with asterisk |
| 01:52.19 | mmlj4 | kakashka: yes, probably |
| 01:52.39 | mmlj4 | you want chan_gtalk or somesuch |
| 01:53.20 | kakashka | yup.. but ... can it used for custom xmpp server(openfire for example)? or only gtalk? |
| 01:53.41 | mmlj4 | gtalk uses jabber, so... |
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| 01:54.15 | kakashka | ok. lets try ^) |
| 01:57.48 | kakashka | flexo*CLI> gtalk reload |
| 01:57.50 | kakashka | IT DOES WORK! |
| 01:57.50 | kakashka | <PROTECTED> |
| 01:57.51 | kakashka | lawl) |
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| 02:54.31 | shmaltz | hi everyone |
| 02:54.35 | shmaltz | ~132 |
| 02:56.06 | shmaltz | ~132 is the most symmetric binary when using ones 10 fingers. Since which ever way one puts their hands it will always be the same 2 fingers up. |
| 02:56.07 | jbot | okay, shmaltz |
| 02:56.40 | shmaltz | ~132 |
| 02:56.41 | jbot | [132] the most symmetric binary when using ones 10 fingers. Since which ever way one puts their hands it will always be the same 2 fingers up. |
| 02:57.09 | shmaltz | ~132 is the 2 middle fingers |
| 02:57.09 | jbot | ...but 132 is already something else... |
| 02:57.16 | shmaltz | ~132 |
| 02:57.17 | jbot | extra, extra, read all about it, 132 is the most symmetric binary when using ones 10 fingers. Since which ever way one puts their hands it will always be the same 2 fingers up. |
| 02:57.41 | shmaltz | ~132 is also it's the 2 middle fingers of both hands |
| 02:57.42 | jbot | okay, shmaltz |
| 02:57.48 | shmaltz | ~132 |
| 02:57.48 | jbot | 132 is probably the most symmetric binary when using ones 10 fingers. Since which ever way one puts their hands it will always be the same 2 fingers up. it's the 2 middle fingers of both hands |
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| 04:07.34 | joobie | guys i have got a sip peer setup.. how can i restart just that one sip peer from console? |
| 04:07.42 | joobie | i want to get it to disconnect / reconnect |
| 04:07.48 | thehar | send a notify to the phone |
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| 04:08.54 | joobie | can it be done from console? |
| 04:09.00 | joobie | it's actually my sip provider btw (pennytel) |
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| 05:09.53 | YoMama | anyone bored enough to help me figure out what's wrong with my config? :-P |
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| 05:14.07 | timeshell | Greetings |
| 05:14.29 | timeshell | Does anyone know if the Polycom SoundPoint 500 supports http provisioning? |
| 05:14.43 | leifmadsen | it does |
| 05:14.53 | timeshell | Hey leifmadsen |
| 05:15.00 | YoMama | anyone using gizmo with their asterisk? |
| 05:15.03 | leifmadsen | wait... is the soundpoint different than the IP500? |
| 05:15.14 | leifmadsen | timeshell: howdy :) |
| 05:15.17 | timeshell | I think it's the same |
| 05:15.46 | leifmadsen | ya, that's what I tohught |
| 05:15.48 | timeshell | But, I don't see an option in the startup that allows me to select HTTP.... only FTP and Trivial FTP |
| 05:15.57 | leifmadsen | my 500 and 501 are provisional via http and https |
| 05:16.08 | leifmadsen | really? maybe you need to update the firmware? |
| 05:16.30 | timeshell | I have 2.1.3 on the 500 now.... far as I can see, that's the highest SIP it supports |
| 05:16.42 | timeshell | I just put 4.1.2B BootROM on |
| 05:18.00 | timeshell | Bought a bunch of IP500's on ebay for $50 each |
| 05:18.02 | timeshell | :D |
| 05:18.28 | YoMama | nice |
| 05:18.47 | timeshell | Couldn't pass that up. |
| 05:18.55 | YoMama | i had a grandstream piece o' crap ip phone, but it broke after a few months |
| 05:19.11 | YoMama | i should just suck it up and buy a real ip phone |
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| 05:19.50 | timeshell | Either the my phones don't support the http prov, or I've forgotten something in the config somewhere :| |
| 05:20.06 | timeshell | I wish some of this stuff was a little more straightforward... or at least had better documentation |
| 05:20.10 | bobsaccamano | hi..how can i set different ring cadences for two sip channels |
| 05:20.11 | bobsaccamano | ? |
| 05:22.31 | timeshell | And I did miss something. :p I forgot to set the correct port |
| 05:23.41 | timeshell | Trying again |
| 05:25.24 | timeshell | Anyone ever have an issue with an IP500 where some of the keys stopped working? |
| 05:25.38 | timeshell | 7 keys on one of mine don't respond |
| 05:25.41 | YoMama | ok...i'll figure this out later...it must be a firewall problem |
| 05:25.50 | YoMama | i can't even get asterisk to show that my line is ringing |
| 05:25.53 | timeshell | What kind of fw? |
| 05:26.02 | YoMama | it's a dumb 2wire device |
| 05:26.12 | YoMama | AT&T U-Verse |
| 05:26.37 | YoMama | 3800HGV-B Gateway |
| 05:27.05 | YoMama | I opened up 5060 (udp), 10000-20000 udp |
| 05:27.18 | YoMama | and forwarded it to my asterisk server |
| 05:27.30 | YoMama | i do a sip show registry and it shows i've registered |
| 05:27.37 | YoMama | but then i call my # and no ringy |
| 05:27.53 | timeshell | Set the extern IP on the asterisk server? |
| 05:28.02 | YoMama | hmm |
| 05:28.17 | YoMama | does asterisk support stun? i think my ip is dynamic |
| 05:34.20 | drmessano | ~sipnat |
| 05:34.21 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 05:50.55 | timeshell | leifmadsen : What version of SIP do you run on your IP500's? |
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| 06:59.54 | timeshell | leifmadsen : If you truly are doing http provisioning on SPIP500's, you gotta tell me how you do it. Everything I've found so far suggests that it's not supported on the IP500. |
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| 07:01.04 | bobsaccamano | Hi..ive defined a SIP cahnnel for 911 calling...ive used Dial function as the entry point, but i do not get a 180 Ringing message |
| 07:01.13 | bobsaccamano | any idea how to get it working? |
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| 07:43.25 | kaldemar | bobsaccamano: be more specific. how have you defined and what? where are you dialing? |
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| 07:47.14 | bobsaccamano | kaldemar, problem solved...just had to put a Ringing and Wait extensions |
| 07:47.54 | bobsaccamano | but on a different plane...is there a way to make asterisk play some music when a call is put on hold? |
| 07:47.58 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 07:48.14 | kaldemar | yes, that's called music on hold. |
| 07:48.50 | bobsaccamano | so this is enough : exten => 123,n,MusicOnHold(default) ? |
| 07:49.02 | bobsaccamano | i want the music to play when Flash is hooked |
| 07:49.47 | jplank | anyone ever play with video and *? |
| 07:49.55 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 07:49.59 | EmleyMoor | bobsaccamano: That is for testing |
| 07:50.51 | bobsaccamano | EmleyMoor, so how is it done? |
| 07:51.31 | EmleyMoor | You need to set up mouic on hold content and set the musiconhold in the appropriate conf file for the connection |
| 07:51.41 | *** join/#asterisk mchou_ (n=mchou@unaffiliated/mchou) |
| 07:51.52 | EmleyMoor | (musiconhold=default is assumed) |
| 07:52.58 | bobsaccamano | but default would also play a tune right? |
| 07:53.16 | bobsaccamano | I dont want any customizations.. |
| 07:53.27 | EmleyMoor | bobsaccamano: If you've got a default musiconhold class set up |
| 07:54.11 | bobsaccamano | EmleyMoor, Okay i have no idea how to do that...can you give some pointers? |
| 07:54.14 | EmleyMoor | (it would play whatever you could hear if you called the MusicOnHold app with the given class...) |
| 07:55.10 | EmleyMoor | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 08:09.12 | fcois93 | good morning! |
| 08:11.56 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 08:19.18 | *** join/#asterisk stoffell (n=kristof@55.210-136-217.adsl-dyn.isp.belgacom.be) |
| 08:26.09 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 08:29.06 | *** join/#asterisk pimpwell (n=domin8@pool-72-80-199-124.nycmny.east.verizon.net) |
| 08:29.38 | pimpwell | for the past 10 years there has been this cookie cutter chat line... DefCon used to use it and they use it for all the local number phone chats... the format goes like this: |
| 08:30.12 | pimpwell | Listen to long disclaimer or press 7 to by pass, join the lobby, have 6 rooms to choose from, have private rooms and a voice board.. |
| 08:30.26 | pimpwell | what system is this? It's always the same guys voice too. |
| 08:31.38 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 08:36.12 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:39.17 | SunnyDP | pimpwell: do you hav ethe number ? |
| 08:41.51 | nfi|ermes | i use externnotify parameter in voicemail.conf; is there a way to be notified on ly when a new message is arrived, and not when someone log in tho the voicemail ??? |
| 08:43.08 | pimpwell | Its called Talkee |
| 08:43.12 | pimpwell | http://www.talkee.com/diagram.gif |
| 08:43.20 | pimpwell | they run party lines all over the USA for many years now |
| 08:52.55 | pimpwell | the pervs on those lines are siiiick |
| 08:52.57 | pimpwell | careful |
| 08:53.11 | pimpwell | these are the guys from 1980s who never learned the internet, bad bad |
| 08:56.25 | ghenry | 64bit asterisk or 32bit more stable? |
| 08:56.28 | ghenry | thanks |
| 08:56.41 | ghenry | with 1.4 |
| 08:56.50 | ghenry | I think 1.4.22 |
| 09:01.10 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 09:02.08 | p1mrx | I finished up my google voice click2call script today: http://www.pmarks.net/posted_links/google-voice-dialout.agi |
| 09:02.29 | p1mrx | that said, it's still a dirty hack |
| 09:03.54 | p1mrx | and, for some reason DTMF doesn't seem to work |
| 09:19.58 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 09:27.47 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:37.25 | lanning | I have an IVR with exten = s,3,Background(Resilience/main_message) |
| 09:37.55 | lanning | the file /var/lib/asterisk/sounds/Resilience/main_message.gsm exists |
| 09:38.02 | lanning | but I get: |
| 09:38.17 | lanning | [Mar 20 02:20:31] WARNING[2883]: channel.c:2734 set_format: Unable to find a codec translation path from g723 to gsm |
| 09:38.17 | lanning | [Mar 20 02:20:31] WARNING[2883]: file.c:868 ast_streamfile: Unable to open Resilience/main_message (format 0x1 (g723)): No such file or directory |
| 09:38.29 | ectospasm | lanning: what codec is the channel? G.723? |
| 09:38.36 | lanning | ulaw |
| 09:38.58 | lanning | there is also |
| 09:39.00 | ectospasm | looks like the channel is in G.723, not ulaw. |
| 09:39.06 | lanning | [Mar 20 02:20:30] WARNING[2883]: channel.c:2734 set_format: Unable to find a codec translation path from g723 to ulaw |
| 09:40.24 | lanning | let me go disallow g723 |
| 09:40.41 | ectospasm | I'd disallow=all, and only allow the codecs you want. |
| 09:42.22 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 09:42.58 | lanning | @!%%!%%%%%! works now. |
| 09:43.31 | lanning | it had been working before. I think the ITSP changed their codec priorities |
| 09:44.31 | lanning | it was just weird timing, as it all stopped working after I put in new prompts. |
| 09:44.50 | lanning | I thought it was something I had changed. |
| 09:45.21 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 09:45.23 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 09:46.05 | shyam_k | will it work if i try to connect the two ekigas one on my desktop and laptop to connect eachother? |
| 09:46.19 | ectospasm | shyam_k: will what work? |
| 09:46.31 | shyam_k | i mean some logical error? like i can't connect between the ekiga and asterisk on the same computer.. |
| 09:46.34 | ectospasm | If their both registered to the Asterisk system? |
| 09:47.04 | ectospasm | are the ekigas showing up as registered? |
| 09:47.05 | shyam_k | hmm i am doing it like, connecting the ekiga in desktop to laptop and connecting the ekiga in laptop to desktop!:) crossover..:) |
| 09:47.32 | shyam_k | hmm sip show peers at desktop shows both asterisk and ekiga online but the other one is showing only asterisk online |
| 09:48.35 | jblack | set qualify=yes in your sip or iax conf files |
| 09:49.04 | *** join/#asterisk joobie (n=joobie@203-217-82-215.dyn.iinet.net.au) |
| 09:49.17 | ectospasm | wait, so you've got asterisk loaded on both the desktop and the laptop? |
| 09:49.31 | shyam_k | yeah |
| 09:53.03 | shyam_k | i get [[http://www.pastebin.ca/1366060][this]] when i try to call from the "unconnected" ekiga and ekiga says remote user rejected the call.. may be i should give qualify=yes on both sip.conf? i |
| 09:53.03 | ectospasm | why? |
| 09:53.19 | shyam_k | to study connecting and calling asterisks.. |
| 09:53.22 | ectospasm | no, qualify has nothing to do with that. |
| 09:53.49 | ectospasm | shyam_k: you have an IAX peer and user on both ends? |
| 09:54.00 | shyam_k | yeah sip actually. |
| 09:54.18 | shyam_k | (i think) |
| 09:54.37 | ectospasm | it may be easier to do it with IAX |
| 09:55.00 | shyam_k | ic.. i'll try that then.. The holy biible have sip at first;-) |
| 09:55.17 | ectospasm | bible, you mean TFOT? |
| 09:55.24 | shyam_k | yeah;-) |
| 09:55.26 | *** join/#asterisk ravib123 (n=newbie@c-24-20-206-103.hsd1.wa.comcast.net) |
| 09:55.37 | shyam_k | any other of the same status? |
| 09:55.43 | ectospasm | I always found it to be easier to set up IAX, but ymmv |
| 09:55.49 | ectospasm | shyam_k: nope |
| 09:55.59 | shyam_k | which sip phone do you use? |
| 09:56.12 | ravib123 | wow this one has some live folks |
| 09:56.18 | ravib123 | hows it going tonight? |
| 09:56.27 | ectospasm | I use Polycom at work, and a cheap cordless AT&T connected to an IAXy at home. |
| 09:56.47 | shyam_k | ravib123: its just a fine morning here..and am gonna rock it with asteRISK:) |
| 09:56.55 | ravib123 | hehe cool cool |
| 09:57.26 | ravib123 | Being a consumate noobie I was having some odd problems, any chance you could share some of the rocking knowledge? |
| 09:57.28 | ravib123 | :) |
| 09:57.38 | ectospasm | It's waay to late for me to be up, but I've got a milestone coming in about 40min... |
| 09:57.52 | ectospasm | And, I don't have to be at work until 4pm |
| 09:58.04 | shyam_k | which free(As in freedom and not as in free beer like that one linked on the "BIble") softphone work the best with asterisk? |
| 09:58.20 | ectospasm | they're all about the same |
| 09:58.21 | shyam_k | ekiga? wengophone? or some other less appealing command line tool? |
| 09:58.24 | shyam_k | okay. |
| 09:58.28 | ectospasm | You just gotta figure out which one works best for you. |
| 09:58.49 | jblack | they each have a different fatal flaw. :) |
| 09:58.54 | ectospasm | The nice thing about freedom is that what works for me won't necessarily be right for you. |
| 09:59.12 | shyam_k | hmm.. would be better if bible says about these things instead of the screenshot of that ** THING |
| 09:59.15 | ectospasm | I tend to eschew softphones in favor of hard phones. |
| 09:59.28 | shyam_k | ah sure freedom is costly:) |
| 10:00.01 | ectospasm | while a softphone can definitely be more economical than a hard phone. |
| 10:00.03 | shyam_k | okay i get it.. and now am paying that:) |
| 10:00.20 | jblack | sure, but it's a one time cost. |
| 10:00.21 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9c97e8a0f1352b15) |
| 10:00.31 | jblack | a low end polycom is what? maybe 150 bucks? |
| 10:00.44 | ravib123 | So I have been working on my 1st *now server, kinda been a fun project thusfar. Anyway, it works well for internal calls (between extensions). However when I make outgoing I get no sound, and no incomming at all. ... the server is DMZ on the router however it is not responding to any external ports at all |
| 10:00.45 | ectospasm | shyam_k: you might want to try getting both softphones to register to the same asterisk system... maybe trying to set up two is a bit ambitious? |
| 10:00.47 | jblack | You'll save that in electricity within a year or two. |
| 10:01.20 | ectospasm | ravib123: what technology are you using to make the outbound calls? |
| 10:01.27 | ravib123 | sip |
| 10:01.40 | ravib123 | it does actually ring through the trunk just fine |
| 10:01.55 | ectospasm | SIP is notorious for not dealing with NAT very well. To get it working you usually have to set up a STUN server, which is a major pain. |
| 10:01.58 | ravib123 | just nothing is getting back to the *now server because it wont respond to external ports |
| 10:02.12 | ravib123 | not even port 80 |
| 10:02.14 | ectospasm | right, classic problem with SIP |
| 10:02.23 | ravib123 | which made me think firewall problem |
| 10:02.28 | ravib123 | but it isn't on |
| 10:02.45 | shyam_k | jblack: also you have to go for softphones when you are in a lill town barely sells even a wifi router.. and when you have a gsoc dream coming next week:) |
| 10:03.47 | jblack | Some little town without a wifi router... If only there were some sort of international network, where you could purchase equipment from hundreds.. nay.. thousands of retailers. |
| 10:04.00 | ectospasm | hehehehh |
| 10:04.17 | jblack | some sort of... inter.. net. |
| 10:04.49 | jblack | loaded with a google different places to buy. |
| 10:04.52 | ectospasm | A network of networks? By jove, I think you're on to something! We should get together and sell the crap out of it! |
| 10:05.10 | Frogzoo | ectospasm: it will never sell |
| 10:05.23 | ravib123 | So my thought is if it was a nat problem wouldn't being DMZ on the router remove that issue? |
| 10:05.24 | jblack | Sure it will! We just need to add xml to it. |
| 10:05.27 | ectospasm | Oh, yeah? That's what they said about the PC, too. |
| 10:05.30 | jblack | Everyone buys crap with xml. |
| 10:06.12 | ectospasm | I pirate all of my XML |
| 10:06.46 | jblack | That's the way it should be. :) |
| 10:07.08 | jblack | speaking of which, wil someone please convert the * book to mobibook? |
| 10:07.12 | ectospasm | ravib123: no, because you're still going through a NAT |
| 10:07.53 | ectospasm | ravib123: I'm telling you, you'll have far less problems if you use IAX. |
| 10:07.55 | jblack | ravib123: That solves half the problem. The other part is that the protocols embed the natted ip, so you have to convince them to lie. That's where things like stun and nat=yes comes in. |
| 10:08.17 | ravib123 | gotchya |
| 10:08.37 | jblack | I'm not sure, but I think iax has powerful nat magic. |
| 10:08.51 | ravib123 | so, it's a pain to setup a stun server? |
| 10:09.02 | ectospasm | IAX was designed with NAT in mind, whereas SIP was not. |
| 10:09.07 | jblack | heh. you don't set one up. You use one of the zillion out there. |
| 10:09.16 | ravib123 | I see |
| 10:09.25 | jblack | I've set iax on natted machines, and they Just Work. Freaky. |
| 10:09.45 | ravib123 | craziness |
| 10:09.55 | ectospasm | jblack: that wasn't an accident (-; |
| 10:10.30 | jblack | it's a little creepy, all the same, to find that my firewall is wearing no clothes. |
| 10:11.10 | ectospasm | mine doesn't either. I got used to it a long time ago. |
| 10:11.46 | jblack | btw, be careful of promoting iax here. There's a handful of mostly-daytimers that despise iax2. |
| 10:12.27 | jblack | closest I've ever figured out is that iax2 used (as in years) to have some serious problem with large deployments. |
| 10:12.33 | ectospasm | One thing about IAX wrt to Asterisk and SIP is I like SIP's Asterisk debug better than IAX. |
| 10:12.49 | jblack | yeah. iax's debug isn't very good. |
| 10:13.10 | bobsaccamano | hi..im trying to enable distinctive ringing for sip phones...now essentially i want the sip phone to ring differently based on the caller id...and I want to configure this on the asterisk server |
| 10:13.12 | ectospasm | ...but that's because I see it much more on a daily basis |
| 10:13.36 | ectospasm | bobsaccamano: what kind of endpoint? |
| 10:13.55 | jblack | I rarely need it. By the time I'm debugging iax2, there's usually something drastically broken going on in the network. |
| 10:14.02 | bobsaccamano | its an analog phone connected to a WiMAX CPE with an inbuilt SIP Stack |
| 10:14.02 | ravib123 | well, doesn't it seem odd that I can't even get to the freepbx web panel remotely? |
| 10:14.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:14.39 | bobsaccamano | ectospasm, i just need to send an ALERT_INFO in the header |
| 10:14.40 | ectospasm | bobsaccamano: best you'll get is by setting SIPAddHeader() with the right magic to get the analog phone to ring with a distinctive ring. |
| 10:14.41 | jblack | we couldn't tell you anything about freepbx here. There's a channel for it here on freenode, #freepbx |
| 10:14.47 | ravib123 | I had a different machine running sourceforge DMZd running for months no problem for port 80 and more |
| 10:14.59 | ectospasm | bobsaccamano: there ya go, then |
| 10:15.01 | ravib123 | well *now comes with it by default :P |
| 10:16.04 | bobsaccamano | ectospasm, is this good for the control statement ;exten=>4444,n,GotoIf($["${CALLERID(num)}" = "5555"]?ring5:ringdefault) ? |
| 10:16.33 | bobsaccamano | where ring and ringdefault are extensions that add the SIP Header info |
| 10:16.42 | ectospasm | that should work, then. |
| 10:17.02 | ectospasm | I admit I'm not a SIP guru. |
| 10:17.07 | bobsaccamano | ectospasm, thanks..ill give it a shot |
| 10:17.35 | bobsaccamano | ectospasm, btw can i set the ring type in asterisk for the channel? |
| 10:17.46 | bobsaccamano | like in indications.conf or something |
| 10:17.46 | bobsaccamano | ? |
| 10:18.00 | ectospasm | I'm not sure what you mean. |
| 10:18.48 | bobsaccamano | well indications.conf has a list of available ring cadences..that you can specify, so i was wondering if I could use those to make my endpoints ring differently |
| 10:18.55 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 10:19.11 | ectospasm | I thought that was only for analog channels. |
| 10:19.21 | *** part/#asterisk ravib123 (n=newbie@c-24-20-206-103.hsd1.wa.comcast.net) |
| 10:19.30 | *** join/#asterisk Arkaos` (n=root@66.71.241.142) |
| 10:19.34 | bobsaccamano | coz im clueless on what ring types my endpoints support |
| 10:19.54 | ectospasm | Unless your SIP frontend can accept ring cadences, I dunno how it will work. |
| 10:20.10 | bobsaccamano | frontend? |
| 10:20.58 | ectospasm | you said yourself you've got a machine between your analog phone and Asterisk, right? |
| 10:21.41 | Arkaos` | Hi guys, i have had a look at asterisk docs and cant find out how to keep the originating caller id after an attended transfer. Can anyone point me uin the right direction? |
| 10:22.02 | bobsaccamano | ectospasm, yeah..ohk |
| 10:22.07 | ectospasm | Arkaos`: try the 'o' option to Dial |
| 10:22.10 | bobsaccamano | ectospasm, thanks again |
| 10:22.22 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:23.05 | Arkaos` | ectospasm: what conf file would that be in? Apologies I have little experience of this and dropped in at the deep end |
| 10:23.18 | ectospasm | it will be in yur dialplan, extensions.conf |
| 10:23.32 | Arkaos` | brilliant thanks |
| 10:23.57 | ectospasm | Arkaos`: I'm not 100% sure it will work, but that's where I'd look first. |
| 10:24.25 | ectospasm | You'll also want to look for the Dial() application calls that have the 't' and 'T' options (for transferring) |
| 10:24.39 | ectospasm | consult the book if you get lost |
| 10:24.49 | ectospasm | http://asteriskdocs.org |
| 10:25.37 | shyam_k | grr i have the ekiga process killed but asterisk server on the other side still says its online.. should i reload the conf or something? |
| 10:26.21 | shyam_k | i just have the very default three line config in sip.conf for the softphone.. |
| 10:26.50 | ectospasm | shyam_k: how did you try to stop asterisk? |
| 10:27.29 | *** join/#asterisk Dekken (n=dekken@ip-78-137-139-140.mobile.digiweb.ie) |
| 10:28.16 | shyam_k | ectospasm: i didn't stop asterisk.. its running.. i just closed the ekiga which was online.. and the asterisk still shows its online.. while wengophone works fine.. asterisk shows its unregistered when wengophone goes offline.. |
| 10:28.46 | Dekken | is there rules I should know about before asking questions? |
| 10:28.55 | ectospasm | do you have qualify=yes for the ekiga? |
| 10:29.01 | Dekken | please disregard my previous question if that is the case ^_^ |
| 10:29.03 | ectospasm | Dekken: be polite, and don't ask to ask. |
| 10:29.10 | Dekken | thanks |
| 10:29.17 | shyam_k | ectospasm: ops not yet.. i'll try that.. |
| 10:29.31 | shyam_k | now i am working on a single asterisk anyway.. |
| 10:30.27 | ectospasm | first things first, baby steps, and all that. |
| 10:30.45 | shyam_k | yeah sure.. |
| 10:33.04 | ectospasm | for setting up Asterisk to Asterisk, normally you'll have to set one to be the peer, and the other to be the user, for one direction of call flow... then vice versa for the other direction. |
| 10:34.51 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 10:35.00 | shyam_k | oh ic.. but then bible goes to that straightly and it seemed plain other than that i have problem with these softphones. i could understand the connection between two asterisks.. |
| 10:40.56 | ectospasm | and with that, I must retire. Catch y'all l8r! |
| 10:41.35 | shyam_k | ectospasm: thanks a lottt! and tc |
| 10:50.29 | Arkaos` | ectospasm: i think my nose is about to bleed now ;P |
| 10:51.36 | jblack | Arkaos`: You can always sweat cash instead of blood..... :) |
| 10:51.59 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 10:56.17 | joobie | guys is there a dialplan function that can be used to grab /var/spool/moo from /var/spool/moo/bah.txt in the dialplan? |
| 10:56.28 | joobie | playing with cut, but cut seems to remove the / if i use / as the delimiter |
| 10:56.31 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 10:59.22 | shyam_k | how to clear the unreachable users? |
| 11:01.39 | jblack | <PROTECTED> |
| 11:03.16 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.220.215) |
| 11:03.27 | DelphiWorld | hello my friends (All Asterisk Users) |
| 11:03.43 | shyam_k | hi here too..:) |
| 11:03.45 | DelphiWorld | please any JAVA or Flash based softphone to use for C2C using asterisk ? |
| 11:04.45 | *** join/#asterisk aksyn (n=aksyn@212.183.134.129) |
| 11:06.14 | joobie | <PROTECTED> |
| 11:08.56 | joobie | hmm which function in the dialplan can run a system command? |
| 11:09.12 | joobie | like say i want to run a script on the filesystem |
| 11:10.34 | shyam_k | joobie: ! command? |
| 11:10.44 | shyam_k | i mean the command "!" |
| 11:12.33 | joobie | huh? |
| 11:12.55 | DelphiWorld | please any java web dialer ? |
| 11:13.13 | shyam_k | joobie: nevermind.. |
| 11:13.49 | joobie | i want to like run a binary on the asterisk server, using the dialplan |
| 11:26.46 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 11:31.50 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:39.01 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 11:39.54 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 11:44.18 | *** join/#asterisk contrabanda (n=contr@79.99.248.8) |
| 11:44.28 | *** join/#asterisk destructure (n=de@67-23-12-32.static.slicehost.net) |
| 11:46.59 | DelphiWorld | please any java Web softphone (C2C) ? |
| 11:55.25 | leifmadsen | timeshell: hmmmm.... well I could be wrong then.... I had an IP500 a while ago, and now I have an IP501, but I really thought it had http provisioning -- apparently I was way wrong |
| 11:59.14 | *** join/#asterisk ZeNN (n=abc@ip224-160-173-82.adsl2.static.versatel.nl) |
| 12:00.10 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 12:01.28 | timeshell | cries |
| 12:03.11 | SuPrSluG | ip 500's have web provisioning |
| 12:04.42 | joobie | guys how do u run a shell script from a dialplan |
| 12:04.45 | joobie | what function can u use? |
| 12:05.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 12:06.35 | timeshell | SuPrSluG : Please support your statement |
| 12:07.05 | timeshell | SuPrSluG : I have been monitoring traffic from it. It makes no HTTP request. |
| 12:07.20 | timeshell | No setting that defines http as the boot server |
| 12:08.09 | timeshell | And I have found a webpage that says it was specifically supported in ROM that doesn't work on the IP500 |
| 12:10.21 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 12:12.58 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 12:13.12 | *** join/#asterisk propellerhead (n=yogurt2u@host108.190-30-184.telecom.net.ar) |
| 12:15.30 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 12:17.04 | *** join/#asterisk mizerydearia (n=mizeryde@CPE-65-27-107-31.new.res.rr.com) |
| 12:32.29 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 12:33.58 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 12:34.30 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:34.44 | leifmadsen | joobie: how about SHELL()? |
| 12:34.47 | leifmadsen | (in 1.6.x) |
| 12:34.59 | leifmadsen | or you can try the more limited 'System()' command in 1.4 |
| 12:38.38 | joobie | i have 1.4 |
| 12:39.06 | joobie | thanks leifmadsen exactly what i was after :) |
| 12:54.40 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 12:54.49 | *** join/#asterisk xloafx (n=Administ@rrcs-24-103-201-115.nys.biz.rr.com) |
| 12:55.01 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-771993edf42fc78d) |
| 12:55.10 | *** join/#asterisk bartpbx (n=bartpbx@p5099e196.dip0.t-ipconnect.de) |
| 12:55.15 | bartpbx | hello |
| 12:55.43 | bartpbx | anyone knows a patch to implement rtsavesysname for iax clients? |
| 12:55.45 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:00.52 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:06.17 | Katty | allos |
| 13:06.48 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:07.08 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:07.19 | Katty | tummy hurts today )= |
| 13:07.56 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 13:08.06 | Katty | hi |
| 13:09.18 | awk_r | hands Katty some Pepto. |
| 13:09.47 | Katty | i dun think that's going to fix me |
| 13:10.19 | awk_r | hands Katty some morphine. |
| 13:10.25 | awk_r | better? |
| 13:10.34 | Katty | >.< |
| 13:10.39 | awk_r | :-) |
| 13:10.59 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) |
| 13:11.40 | Katty | i will feels better next week. |
| 13:12.09 | Katty | until then, i will be a bitch )=< RAWR |
| 13:13.39 | Katty | and then sob )_= |
| 13:13.50 | Katty | <PROTECTED> |
| 13:22.05 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 13:22.37 | destructure | wow |
| 13:22.53 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.24.187) |
| 13:25.16 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 13:26.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:26.47 | gambler1 | Hi, is there anyway * could "load balance" outgoing calls to two or more E1/T1? |
| 13:27.27 | awk_r | gambler1, yes |
| 13:27.53 | gambler1 | awk_r: Can you please point me to right direction? |
| 13:27.55 | [TK]D-Fender | gambler1: You are the one choosing what channels to send calls to... its your job and yes you can do this in dialplan. |
| 13:28.25 | awk_r | redirects gambler1 to [TK]D-Fender's answer. |
| 13:28.35 | gambler1 | [TK]D-Fender: oh, I tought something like grouping multiple PRI's to one.. |
| 13:29.12 | [TK]D-Fender | gambler1: there is no "load balance on a group", it has to be you balancing against multiple groups |
| 13:29.22 | [TK]D-Fender | gambler1: Otherwise you're just sequentially spanning. |
| 13:29.54 | gambler1 | [TK]D-Fender: multiple groups? You mean per PRI? |
| 13:30.38 | [TK]D-Fender | gambler1: Each PRI as a group, and you doing the determination of which to use for the next call |
| 13:31.44 | gambler1 | [TK]D-Fender: hmmmm.. ok, then I'll do some dialplan magic.. I want just to be sure, that there is no other elegant soultion |
| 13:32.03 | gambler1 | [TK]D-Fender: thank you for your support |
| 13:32.51 | [TK]D-Fender | gambler1: You're welcome. |
| 13:32.56 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:32.59 | gambler1 | [TK]D-Fender: just one more question please |
| 13:33.30 | gambler1 | [TK]D-Fender: is there snmp support for TE420 cards? |
| 13:33.51 | [TK]D-Fender | gambler1: ummmm... can't make much sense of that question... |
| 13:34.17 | [TK]D-Fender | gambler1: Its not the card that offers this, and I don't know SNMP software |
| 13:35.14 | gambler1 | [TK]D-Fender: sorry :) I want to read some statistics from those cards (yes I know it's not the card feature but, you still need MIB in order to know what to read) |
| 13:35.27 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 13:35.29 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 13:35.45 | [TK]D-Fender | gambler1: Unfortunately I have experience in that arena |
| 13:36.14 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:37.12 | gambler1 | [TK]D-Fender: Thank you, I will try to find out is there anyway I could find some mib's from Digium |
| 13:37.13 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:37.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:38.44 | *** join/#asterisk jeff_phillips (n=ceramics@66-112-49-13.stat.centurytel.net) |
| 13:38.46 | jeff_phillips | good morning |
| 13:39.05 | [TK]D-Fender | jeff_phillips: No. |
| 13:39.07 | [TK]D-Fender | :p |
| 13:39.24 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 13:39.29 | jeff_phillips | aww, sorry you're having a bad morning TK-Defender |
| 13:40.05 | *** join/#asterisk BlargMaN00 (n=blargman@12.234.16.130) |
| 13:45.36 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:46.37 | [TK]D-Fender | jeff_phillips: No, this is in answer to your "can I turn chan_alsa into a multi-track paging system" :) |
| 13:46.58 | [TK]D-Fender | touts his private beta copy of res_psychic.so |
| 13:47.48 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 13:48.21 | jeff_phillips | well actually -- I still only intend to use one track at a time |
| 13:48.45 | jeff_phillips | i bet it can be done if I waste enough time with it |
| 13:48.56 | [TK]D-Fender | jeff_phillips: So you want to try to do "split zones" without being simultaneous? |
| 13:49.00 | jeff_phillips | right |
| 13:49.32 | jeff_phillips | the easy solution (in my expectation) would be to trigger something to just shift the soundcard's left/right balance when initating the call |
| 13:49.44 | [TK]D-Fender | jeff_phillips: Possible if of course you massively rewrite stuff, everything else is a hardware cost |
| 13:50.15 | [TK]D-Fender | jeff_phillips: Yeah, balance could do it perhaps as a hack.. you'd have to personalize it to that system perhaps, but yeah, maybe... |
| 13:51.18 | *** join/#asterisk _theHub (n=_theHub@firewall.cierant.com) |
| 13:51.27 | DavidR2008 | need some help compiling asterisk 1.4.23.1 I already had it installed, now I want to add some zap channels. I installed zaptel-1.4.12.1 and libpri-1.4.9 and reinstalled asterisk: ./configure make make install. however asterisk didn't have cli zap commands so I checked menuconfig and it doesn't have a chan_zap. what should I do? |
| 13:55.59 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 13:58.21 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 13:59.29 | [TK]D-Fender | DaviDAHDI replaced Zaptel as of 1.4.22. use that instead |
| 13:59.39 | kaptengu | when I forward some to en external number, my pbx number is showing up at the external number instead of the caller's, how can I change this? |
| 13:59.48 | [TK]D-Fender | DavidR2008: DAHDI replaced Zaptel as of 1.4.22. use that instead |
| 14:00.16 | [TK]D-Fender | kaptengu: Depends what you are dialing out of and how its configured |
| 14:00.19 | DavidR2008 | [TK]D-Fender: I'm trying to install sangoma and the stable drivers only support Zaptel |
| 14:00.48 | [TK]D-Fender | DavidR2008: then don't. Their "beta" drivers are rather stable typically |
| 14:01.03 | DavidR2008 | k, thx |
| 14:01.11 | kaptengu | [TK]D-Fender: it's a SIP-trunk, what do you mean how it's configure? |
| 14:01.41 | [TK]D-Fender | kaptengu: I mean that certain settings will prevent you from passing on the information in a way your ITSP will process. |
| 14:01.52 | [TK]D-Fender | kaptengu: And it also depends if they even permit you to change it |
| 14:02.20 | kaptengu | [TK]D-Fender: can I do something about it or do I have to talk to my SIP service provider? |
| 14:02.45 | [TK]D-Fender | kaptengu: My answer said it all... depends if you are forcing on your side, and if they permit it |
| 14:02.57 | [TK]D-Fender | kaptengu: The 2nd you obviously have to talk to them about |
| 14:03.07 | kaptengu | [TK]D-Fender: so, how can I force it? |
| 14:03.25 | [TK]D-Fender | kaptengu: pastebin your config masking only passwords |
| 14:05.42 | pdmmm | man |
| 14:05.44 | pdmmm | i rock |
| 14:05.47 | pdmmm | i got h323 working |
| 14:05.49 | pdmmm | what a pile |
| 14:06.28 | kaptengu | [TK]D-Fender: which file do you need to see? |
| 14:06.42 | [TK]D-Fender | kaptengu: sip.conf |
| 14:06.44 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 14:07.45 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.67) |
| 14:08.40 | anonymouz666 | SELECT COUNT(*) from [TK]D-Fender where file = 'sip.conf'; |
| 14:09.02 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 14:09.07 | anonymouz666 | Segmentation fault |
| 14:09.30 | [TK]D-Fender | grabs his katana and segments anonymouz666 |
| 14:09.49 | anonymouz666 | heh |
| 14:09.52 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) |
| 14:11.56 | DavidR2008 | which is worse: a segmentation fault or a segmentation? |
| 14:12.53 | kaptengu | [TK]D-Fender: http://pastebin.com/d5544117 |
| 14:13.06 | timeshell | Can't asterisk do phoneprov with ftp instead of http? |
| 14:14.22 | [TK]D-Fender | kaptengu: "fromuser=105110228" <- this usually forces taht to be the CID #. Try adding "sendrpid=yes" , "trustrpid=yes" to that peer and test. If that fails then either th fromuser is interfereing or your provider is blocking you |
| 14:14.29 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:14.49 | [TK]D-Fender | timeshell : * is not an FTP server. * shouldn't even HAVE phoneprov |
| 14:14.55 | kaptengu | [TK]D-Fender: ok, thank you very much |
| 14:15.52 | *** join/#asterisk InfoNutz (n=what@204.50.209.225) |
| 14:16.02 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 14:16.06 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:20.29 | timeshell | [TK]D-Fender You're wrong |
| 14:20.50 | timeshell | EVERY phone server should be able to provision. |
| 14:21.33 | timeshell | A hard core programmer may not like it, but the fact is, you'll have a lot less support for your product if you don't support the needs of those who want to use it. |
| 14:21.35 | [TK]D-Fender | timeshell : |
| 14:21.50 | [TK]D-Fender | timeshell : "phone server". Lol... * is a toolkit, not an assembled 747. |
| 14:22.21 | timeshell | Don't go into literals. You know what I mean. |
| 14:22.35 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:22.58 | timeshell | A large number of real IT people HATE MS. However MS has the market because they give the rest of the world what they need. |
| 14:23.15 | [TK]D-Fender | timeshell : Drag your sorry ass out of GUI-Dream-Land where 1 app does your web, ftp, mail, sip, routing, HTPC all in one miracle package named "Asterisk" |
| 14:23.31 | timeshell | :) |
| 14:23.32 | timeshell | No |
| 14:23.32 | DavidR2008 | if you consider the "phone server" as the physical box, then yes it can do all that install the tftp, ftp or http server of your choice and presto! you have a "phone server" that can "phoneprov" |
| 14:23.35 | Chainsaw | timeshell: So you're saying there's a market for Ultimate Asterisk. |
| 14:23.36 | [TK]D-Fender | timeshellAnd those that try to do everything end up doing nothing particularly well. |
| 14:23.45 | Chainsaw | timeshell: Go ahead, package it up. |
| 14:24.13 | [TK]D-Fender | timeshell : Yup clearly time to "segment" * into a minimum of 10 flavours and introduce a pricing scheme. Oh and of course close the source. |
| 14:24.17 | DavidR2008 | oh, don't forget DHCP, you'll probably want that too! |
| 14:24.23 | timeshell | Chainsaw : Why should I re-invent the wheel? Asterisk-gui, Trixbox, FreePBX are all headed in that direction. |
| 14:24.26 | [TK]D-Fender | DavidR2008: YEAH! |
| 14:24.38 | Chainsaw | timeshell: So run off and use those? |
| 14:24.44 | timeshell | sighs |
| 14:24.49 | timeshell | bye |
| 14:24.59 | [TK]D-Fender | Chainsaw: You say that like he isn't the majority speaker in at least on GUI channel already ;) |
| 14:25.13 | [TK]D-Fender | one* |
| 14:31.22 | awk_r | doesn't astlinux try to do everything? |
| 14:31.28 | jeff_phillips | Will T38 calls be okay without QoS? |
| 14:31.39 | jeff_phillips | since it handles it differently |
| 14:31.54 | *** join/#asterisk gradgrind (n=Gradgrin@216.68.101.253) |
| 14:33.14 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 14:38.42 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:40.50 | *** join/#asterisk gradgrind (n=Gradgrin@216.68.101.253) |
| 14:42.45 | jaytee | yawns |
| 14:43.39 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 14:55.40 | Katty | michelle obama is putting in a garden. |
| 14:55.47 | Katty | in the south lawn! |
| 14:56.12 | Katty | hugs yawny jaytee |
| 14:56.26 | jaytee | hugs Katty back |
| 14:56.33 | Gido-E | Katty ? |
| 14:56.40 | Katty | Gido-E: yes? |
| 14:56.46 | mmlj4 | thanks for telling us, i don't know if I would have made it through the day, not knowing what the ugly woman was up to |
| 14:57.21 | Katty | mmlj4: if you're going to be grumpy and anal, please do so quietly (= |
| 14:57.26 | jaytee | ugly? man, you've got some serious issues |
| 14:57.34 | *** part/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 14:57.39 | mmlj4 | and you need glasses, apparently |
| 14:57.49 | kaptengu | is it possible to let the caller enter some dtmf digits when entering a queue, to later pass it on as a CID name prefix? |
| 14:58.03 | jaytee | and you've probably got a car up on block in your front yard right in front of the trailer |
| 14:58.10 | Katty | giggles |
| 14:58.16 | Katty | okay come guys |
| 14:58.24 | mmlj4 | at least mine runs |
| 14:58.29 | Katty | let's at least TRY to be positive |
| 14:58.29 | Gido-E | Katty on your face? :-) |
| 14:58.42 | Katty | Gido-E: you do not parse. please try again. |
| 14:59.04 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 14:59.34 | jaytee | wow, SciFi is running a full day's marathon of BSG leading up to the show's finale tonight |
| 14:59.45 | jsmith | kaptengu: Sure... use the Read() application to gather those digits |
| 15:00.04 | jsmith | kaptengu: Then use the Set(CALLERID(num)=${read_digits}) to overwrite the caller-id number |
| 15:00.13 | jsmith | kaptengu: Then send the call to the queue with the Queue() application |
| 15:00.39 | kaptengu | jsmith: that's great, thank you! |
| 15:00.43 | jaytee | Ohayo goziamasu, Sensei!!! |
| 15:00.59 | jsmith | jaytee: How long you been speakin' Japanese? |
| 15:01.08 | jaytee | bout a week :-) |
| 15:02.35 | jaytee | jsmith, are you going to teach any of those overseas classes that were in the email I got yesterday? |
| 15:03.25 | jsmith | jaytee: It's possible I might do the one in India, but most of the international classes are taught by our training partners (in the local language, no less) |
| 15:03.44 | *** join/#asterisk xloafx (n=Administ@rrcs-24-103-201-118.nys.biz.rr.com) |
| 15:03.48 | [TK]D-Fender | Most of my Japanese is restricted to wrods involving cutting people down with sharp metal objects :) |
| 15:03.50 | jsmith | jaytee: There are always exceptions, however... like in Belgium, we've got classes in both English and French scheduled |
| 15:03.58 | jaytee | ah, that's a pity. I'd be boning up on spanish with Rosetta Stone if I had a chance to teach in Barcelona |
| 15:04.20 | jaytee | Belgium, mmmmm, that'd be a nice road trip for waffles :-) |
| 15:04.24 | [TK]D-Fender | jaytee: Boning Rosa indeed ;) |
| 15:05.04 | jaytee | hehe |
| 15:05.22 | jsmith | jaytee: I'm quiet good at Spanish, actually... I wouldn't say I'm fluent anymore, but I can hold my own |
| 15:05.47 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:05.49 | jsmith | jaytee: And yes, I'd *love* to get a chance to spend some time in Spain |
| 15:08.55 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 15:09.21 | ayeso | in comedian mail, if someone changes their mailbox greeting. Where is the audio file kept? |
| 15:10.42 | leifmadsen | anyone used Rosetta Stone? |
| 15:11.58 | jaytee | my buddy used it for French and liked it alot |
| 15:12.07 | leifmadsen | interesting... |
| 15:12.07 | jaytee | it has the best rep |
| 15:12.10 | leifmadsen | ya |
| 15:12.24 | leifmadsen | I'm really interested in finally learning french... |
| 15:12.33 | leifmadsen | and maybe someday will learn danish |
| 15:12.42 | jaytee | plus they use it at the Monterey language center where they teach foreign languages to members of the armed forces. |
| 15:12.53 | jaytee | it's expensive though |
| 15:13.15 | jsmith | ayeso: /var/spool/asterisk/voicemail/[voicemail context]/[mailbox number]/greet.* |
| 15:13.24 | ayeso | jsmith: thanks |
| 15:13.42 | jsmith | leifmadsen: You might also check out the Michel Thomas CDs... they're good, and less expensive than Rosetta Stone |
| 15:14.15 | jaytee | I've used Berlitz myself and found their books and "tapes" of decent quality (<---- look everyone! a dinosaur! ) |
| 15:14.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 15:14.54 | coppice | linguaphone recently died. they've been around a long long time |
| 15:16.12 | [TK]D-Fender | ayeso: In the voicemail folder for their box |
| 15:16.25 | [TK]D-Fender | ayeso: asterisk.conf <- look for a clue |
| 15:16.36 | [TK]D-Fender | jsmith: Darn you jsut handed it right over! :p |
| 15:17.02 | jsmith | [TK]D-Fender: Well, you know... I've given up on teaching people to fish... now I just throw them a plate of sushi... |
| 15:17.16 | [TK]D-Fender | jsmith: At least its uncooked |
| 15:18.11 | jsmith | [TK]D-Fender: Like I have the time to cook things for people :-) |
| 15:18.15 | awk_r | jsmith, if you give them bad sushi they probably won't ask for fish again...preventing the same problem from happening again!!! |
| 15:18.30 | awk_r | (i jest of course) |
| 15:18.35 | jsmith | (and to those who are wonder, I'm just joking... I'll never give up on trying to teach people to fish!) |
| 15:18.41 | jsmith | awk_r: Of course! |
| 15:19.01 | coppice | if you have bad sushi give it to an AIG director |
| 15:19.34 | jsmith | coppice: But those receiving the bonuses are supposed the good guys... uh huh.... right.... |
| 15:19.36 | ayeso | How can I run a system command after someone has checked their voice mail? and after someone leaves a voice mail? Is this even possible? |
| 15:19.38 | [TK]D-Fender | awk_r: I'll make your cut of fugo extra lean ;) |
| 15:19.38 | awk_r | coppice, they don't deserve bad sushi |
| 15:19.55 | leifmadsen | jsmith: ya, I a Michel Thomas CD actually, and like it quite a bit |
| 15:19.57 | coppice | teach a man to fish, and he'll undercut you and drive you out of business |
| 15:20.02 | [TK]D-Fender | ayeso: Read the sample voicemail.conf |
| 15:20.03 | leifmadsen | s/I a/I have a/ |
| 15:20.20 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 15:20.40 | jsmith | coppice: Then why do you try (in vain, mostly, I know) to teach people about faxing? :-p |
| 15:21.09 | jaytee | hahaha |
| 15:21.18 | awk_r | ayeso, [TK]D-Fender has a point. A lot of the basic questions about features can usually be answered in a sample config files. :-) |
| 15:21.26 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
| 15:22.01 | coppice | Its strange. when I talk about something I know a fair amount about things seem to go OK. when I talk about something I'm a genuine expert in I seem to collect a bunch of weirdos :-\ |
| 15:22.28 | awk_r | s/(questions) about (feature)(?=s)/\2 \1/ |
| 15:22.37 | awk_r | heh, figured |
| 15:22.54 | jsmith | coppice: That's unfortunately the way it works :-( |
| 15:23.57 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:24.58 | *** join/#asterisk zenfox (n=Steven@host-64-179-125-210.mil.choiceone.net) |
| 15:25.15 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:29.30 | DavidR2008 | coppice: do you teach people about faxing? if so, where do I sign up? |
| 15:30.23 | coppice | 1: insert sheet |
| 15:30.24 | coppice | 2: dial number |
| 15:30.26 | coppice | 3: profit |
| 15:30.45 | DavidR2008 | aw dang, I was missing step three :-) |
| 15:30.52 | jsmith | DavidR2008: http://soft-switch.org/foip.html and http://soft-switch.org/t38/index.html |
| 15:30.59 | ayeso | [TK]D-Fender: OK, I see that if I uncomment "externnotify" I can run an external script, but how can I pass an asterisk variable to that app? I need to pass some of the callerid info to it. |
| 15:32.02 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
| 15:32.34 | DavidR2008 | in actuality I need to run a fax from an analog port on one asterisk server to a PRI channel on another asterisk server in the same room. I would like to do this in the least complicated way possible. |
| 15:32.48 | ayeso | [TK]D-Fender: actaully i could just pass the current mailbox number rather than callerid, but I cant see how to do that either. |
| 15:35.18 | DavidR2008 | I have a really dumb question: I have an analog card in my asterisk server with FXO ports. the channels show up in asterisk. I plug a standard phone in to one of the ports and lift the handset, nothing. no messages in asterisk either. Where should I start looking? and/or what am I doing wrong? (if it's simple) |
| 15:36.08 | coppice | aren't FXS ports rather better for plugging phones into? |
| 15:37.02 | DavidR2008 | I got it backward?! dang, that's dumb ... reference the first line :-S |
| 15:37.28 | jaytee | DavidR2008, just buy a single port Grandsuck Handytone 286 ATA and run the fax off that to the server the PRI is on. |
| 15:37.35 | coppice | someone needs spanking for that stupid naming |
| 15:37.59 | jaytee | coppice, that convention has been around long before VOIP was even an idea |
| 15:38.02 | iCEBrkr | Why does SMS have to be so $$$ |
| 15:38.07 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 15:38.30 | [TK]D-Fender | iCEBrkr: So the telco still has was to gouge you. |
| 15:38.30 | coppice | jaytee: and that makes the names less stupid? they were just as stupid in the 80s |
| 15:38.47 | iCEBrkr | [TK]D-Fender: I know. I know.. But it's such a simple protocol/technology. |
| 15:38.58 | iCEBrkr | [TK]D-Fender: I wish I could get SMS into my Asterisk box... |
| 15:38.59 | coppice | iCEBrkr: is the price holding back sales? |
| 15:39.29 | iCEBrkr | coppice: pfft, I'm just wanting to built a SMS toy.. and it's too much to spend on something that's a toy. |
| 15:40.12 | jaytee | coppice, true but all you have to do is think, FXS=station=phone FXO=office=line. Anyone who isn't a mouth breather with a room temperature I.Q. should be able to get their head around that. |
| 15:41.03 | [TK]D-Fender | jaytee: Yeah, its their kind that give the other .02% a bad name.... |
| 15:41.05 | coppice | yeah, but then people make it complex by saying they put fxo signaling with fxs hardware and vice versa |
| 15:41.18 | ayeso | Where can I get the source code for comedian mail? |
| 15:41.37 | iCEBrkr | ayeso: download asterisk source |
| 15:41.53 | [TK]D-Fender | ayeso: Same place as the rest of * |
| 15:41.55 | ayeso | iCEBrkr: thanks, you know what language its written in? |
| 15:42.05 | [TK]D-Fender | ayeso: C |
| 15:42.06 | coppice | the FXS port is actually the office, and the FXO is actually the station |
| 15:42.17 | jaytee | the signalling nomenclature IS assbackwards or counter-intuitive as far as the zaptel/zapata/dahdi configs are concerned but hey, just blame it on russellb :-) |
| 15:42.29 | ayeso | [TK]D-Fender: thanks. |
| 15:43.20 | DavidR2008 | jaytee: have to used the HT 286 ATA with faxing before? as long as it's on the same switch as the asterisk server with the PRI, should be stable enough to work? |
| 15:43.26 | DavidR2008 | you* |
| 15:43.32 | jeff_phillips | coppice: I thought FXS = forign exchange "station" and FXO = forign exchange "operator", thus operator = the telco & station = the phone |
| 15:43.51 | coppice | there's the "thing wat puts out the volts" and there's the "thing wat takes in the volts" |
| 15:43.52 | jaytee | DavidR2008, yeah, with T38 passthrough it works fine |
| 15:44.14 | coppice | FXO is actually foreign exchange office, as in central office |
| 15:44.24 | jaytee | correct |
| 15:44.35 | Katty | hmm. hungry. |
| 15:44.37 | jaytee | which means to me "line side" |
| 15:44.43 | coppice | and the FXO port does *not* go in the central office |
| 15:44.49 | jaytee | hands Katty a sammich |
| 15:45.24 | coppice | and what's so bloody foreign about them? they might be made in China, but its to a local spec :-\ |
| 15:45.37 | jeff_phillips | coppice: me confused. what does "office" in "forign exchange office" refer to then? |
| 15:45.57 | coppice | the central office, i.e. the exchange building |
| 15:46.21 | jeff_phillips | then fxo=telco's office, and fxs=the phone... like I thought |
| 15:46.44 | Katty | jaytee: i was actually plotting mexican |
| 15:46.45 | jeff_phillips | i don't see what's backwards about the zaptel then |
| 15:46.51 | tompaw | HEllo |
| 15:46.56 | jaytee | coppice, trying to undo 30 some odd years of bad naming conventions is impractical. when I was younger and having to debug some old poorly documented COBOL code I thought of going back in time to murder the guy who invented the GOTO but a friend said, "Nah, someone else would just invent the COMEFROM". I surrendered to the inevitable. |
| 15:47.02 | tompaw | is there a way to set CallerID for DISA? |
| 15:47.05 | coppice | except the ports called FXO are the station, and the ports called FXS go in the central office |
| 15:48.40 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:48.55 | jaytee | what? |
| 15:48.56 | coppice | jaytee: outside the US we didn't have this bloody stupid naming until VoIP came along. how come the dumb names are the ones to become universal? |
| 15:48.58 | jeff_phillips | I think it is correct as it is |
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| 15:49.29 | jaytee | coppice, so you're saying an analog line to the telco is supposed to go into an FXS port? WRONG! |
| 15:49.41 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:49.53 | jeff_phillips | I think the confusion of naming conventions is the "forign exchange" part of the name. |
| 15:49.59 | coppice | no. the port at the exchange is the FXS port, but its not the station, its the office |
| 15:50.33 | deadpigeon | if I'm just changing echotraining and echocancel settings, can I do a dahdi restart to test, or do I have to restart the server? |
| 15:50.41 | coppice | I think it was named by some sadistic bastard the day he was laid off |
| 15:51.27 | jeff_phillips | coppice: I think it was named this way because they have FXS ports for each customer's "station" on the local exchange, and FXO ports for each connection to a "forgin exchange operator" |
| 15:51.35 | jaytee | coppice, or deliberately obfuscated by AT&T in a blatant attempt to control their monopoly and create a kind of "guild" knowledge base where information was cloistered. |
| 15:51.51 | jeff_phillips | In our private PBX, we have FXS for each end user's "station" and FXO for the "forign" telephone company operator's lines coming into our system |
| 15:52.18 | deadpigeon | which is correct. |
| 15:52.20 | jaytee | jeff_phillips, makes perfect sense to me, but then I'm a dinosaur :-) |
| 15:52.20 | jeff_phillips | think of your PBX as BEING a telco exchange |
| 15:52.39 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:52.42 | jeff_phillips | the telco's excahnge is "forign" to our local PBX exchange |
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| 15:53.08 | coppice | "damn foreigners took our jobs. I'll show 'em" |
| 15:53.15 | deadpigeon | so..... can I just do a dahdi restart to test echo settings? or does the server have to be reboot? |
| 15:53.19 | tompaw | "they took our jobs!" |
| 15:53.38 | jaytee | hell, I remember working in a RCA telco facility on Elmendorf AFB in Anchorage back in the 70's that had the old electromechanical rotary switches. You needed earplugs just to work. |
| 15:54.05 | coppice | and they lasted a lot longer than any electronic exchange |
| 15:54.12 | jaytee | it's so hard to find a good anglo-saxon gardener in America these days! |
| 15:54.20 | jeff_phillips | jaytee: Verizon bought out the old GTE Michigan exchanges and until around 1996 or so we still had that in our local exchange! |
| 15:54.33 | jeff_phillips | They had some equipment that decoded the DTMF tones and converted it back to rotary dial |
| 15:54.45 | jeff_phillips | if you walked in there you could hear all the melchanical switches working |
| 15:55.16 | jeff_phillips | I want to say around 1989 or 1990 was when we got private lines. I remember a party line growing up in the 80's |
| 15:55.36 | coppice | In 1996 Indian Telephone Industries was still making that stuff |
| 15:55.37 | deadpigeon | our central office here is relatively loud just with a dozen or so clec's equipment in the general area. |
| 15:56.15 | DavidR2008 | I'm a little confused by the statement "T.38 passthrough" the problem may be that I don't understand T.38 well enough, but the asterisk server is connect to a POTS PRI so it seems like the asterisk server would have to convert T.38 back in to the "normal" fax on the PRI voice channel |
| 15:56.17 | jaytee | remember the original Bell "Princess" phone? it had to have a second pair to the jack just to light the dial and good old Ma Bell charged extra for it. Then they finally came out with a phone with an A/C adapter for the light. |
| 15:56.40 | jaytee | WTF is a POTS PRI? |
| 15:56.41 | deadpigeon | pots pri? heheh. |
| 15:57.34 | deadpigeon | pri signalling is digital. t.38 is old at&t signalling right? |
| 15:57.35 | jeff_phillips | When you'd dial an adjacent rate center, which was a call completed over local trunks to the adjacent CO -- whenever those trunks were busy the call would route from Verizon's CO to a Michigan Bell & later Ameritech operator who would say "Number please?" and was expecting you to say YOUR OWN number, not the number you are calling... so they'd know who to bill the call to |
| 15:57.39 | deadpigeon | pots is typically fxs or gr303 |
| 15:57.41 | deadpigeon | these days. |
| 15:57.48 | DavidR2008 | a standard PRI. I'm not sending VoIP over the PRI it's standard channelized T1 |
| 15:57.51 | jeff_phillips | because the mechancial switch would lose the information of who was the dialing party when it reached the overflow routes |
| 15:58.00 | DavidR2008 | maybe I shouldn't have said POTS |
| 15:58.17 | deadpigeon | no you shouldnt have. is it just a channelized T1 loop? or is it actually a pri? |
| 15:58.44 | DavidR2008 | actually a PRI |
| 15:58.56 | coppice | DavidR2008: most people still consider them POTS if they have none of the clever stuff enabled |
| 15:59.02 | jaytee | in laymen's terms do you have 24 voice channels or 23 voice channels and a D channel? |
| 15:59.09 | jeff_phillips | my friend Nick who grew up just down the street from me had CenturyTel, where private lines & touch tone service seemed to always exist as far back as he could remember. when he tried to call his mom from my house one time and had an operator cut in asking "number please?" when we were teenagers, it threw him for a loop |
| 15:59.10 | DavidR2008 | 23+D |
| 15:59.22 | deadpigeon | right. so wheres the issue david? fax troubles? |
| 15:59.25 | jeff_phillips | he said his own number -- the number he was trying to dail. The operator said "no it's not" and hung up on him |
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| 15:59.51 | DavidR2008 | maybe no issue, I'm trying to set it up and make sure I purchase the right equipment |
| 16:00.23 | deadpigeon | i never cared much for fax over pri's via asterisk, although that was years ago when I had issues with asterisk and faxing. |
| 16:00.26 | jaytee | David, the ATA adapter treats the fax on the FXS port as SIP device. * handles the channel bridging between SIP (VOIP) and PRI. T38 passthrough means * doesn't handle any of the T38 stuff, it just routes the packets. |
| 16:00.29 | deadpigeon | im sure it's had some work since. |
| 16:01.11 | DavidR2008 | ok, great! so it should just work (tm) :-) |
| 16:01.35 | deadpigeon | still wondering if I can test echo without rebooting =/ |
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| 16:02.32 | jeff_phillips | My solution to the fax problem was to just get an ATA device that supported T.38 and configure it to talk directly to the SIP termination provider and forget about sending it through asterisk |
| 16:02.36 | jeff_phillips | works just dandy |
| 16:03.04 | DavidR2008 | well I am the SIP termination provider :-) I have voice PRIs from the telco |
| 16:03.07 | jaytee | DavidR2008, you need to have t38_udptl=yes in the general section of your sip.conf and set the ATA to enable T38 passthrough. |
| 16:03.10 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 16:03.26 | DavidR2008 | jaytee: thx! |
| 16:03.36 | jeff_phillips | DavidR2008: Ah, I'm stuck with a couple of POTS lines and the rest of everything is over IP in some flavor. |
| 16:03.37 | jaytee | DavidR2008, yw! |
| 16:03.53 | jaytee | jeff_phillips, hopefully chocolate |
| 16:04.04 | beek | morning jaytee |
| 16:04.15 | jaytee | morning beek |
| 16:04.26 | jeff_phillips | only issue I have now is we are paying 1.6 cents a minute to terminate the occasional local fax that could have been free over the POTS. But when we had it plugged into the POTS we were paying like 7 cents a minute for long distance, and most of our faxes are long distance. So the boss is happy as the bill is a lot lower now overall |
| 16:04.38 | beek | jaytee: How's the Nortel removal coming? |
| 16:04.51 | jeff_phillips | we did introduce one new issue though that I'm telling the office secretaries to just deal with |
| 16:05.01 | jaytee | beek, it's off the wall and sold for peanuts already |
| 16:05.12 | DavidR2008 | now to throw a curve ball, can I have * take the T.38 and send the fax out a POTS line (true POTS: DS0)? |
| 16:05.16 | beek | So you get to admire that empty space. |
| 16:05.26 | jaytee | beek, yup |
| 16:05.50 | jaytee | DavidR2008, you'd need an FXO line in the * box to route the call to the POTS line |
| 16:05.52 | jeff_phillips | DavidR2008: I'm sure you can, the only reason I didn't was because I was tired of fussing with the gateway I had the building extension wiring hooked up to |
| 16:06.09 | jaytee | and a way to handle it specifically for that fax in your dialplan |
| 16:06.27 | DavidR2008 | ok, I've got a sangoma A200 with four FXO lines |
| 16:06.30 | jeff_phillips | use a custom context so calls from that extension go to that channel |
| 16:07.42 | jeff_phillips | our secretaries apparently had the habbit of sending a fax to ourselves by dialing the number of the fax to e-mail service we ported our fax over to a long time ago. When I spoofed that same # on the outbound caller ID sent on our SIP trunk which we're using to send faxes now, they discovered they can't send a fax to themselves anymore |
| 16:08.18 | jeff_phillips | when the fax-to-email provider receives a call from the same # that they provide it goes to some voice mail pin number prompt which they say is ran by the same software that does their fax service. |
| 16:08.36 | jaytee | DavidR2008, so make sure that the SIP account for the ATA the fax is hooked to has no other outbound route in it's context for routing outbound calls other than the zap channel the POTS line is plugged into. |
| 16:08.45 | jeff_phillips | I asked the secretaries why they dont' just use the "scan" button---all that faxing it to ourselves did was get a crappy scan to their e-mail. |
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| 16:09.03 | jeff_phillips | They can't give me a straight answer but want me to fix it so they can use the "fax" button to scan stuff to the computer right next to the machine |
| 16:09.27 | *** join/#asterisk nima0102 (n=nima@91.98.195.126) |
| 16:09.34 | deadpigeon | i'd just slap them upside the head. |
| 16:09.43 | jeff_phillips | i'd like to |
| 16:09.44 | DavidR2008 | jaytee: what I need to do is route a normal faxes out over the PRI (using T.38 pass through) and any 800# faxes out over the POTS line (I can't dial an 800# on the PRI) |
| 16:09.53 | jaytee | jeff_phillips, when you've worked in IT as long as I have you never underestimate the stupidity or the laziness of the average computer user. |
| 16:10.13 | jaytee | most of them are totally perplexed just trying to create a shortcut on their desktops. |
| 16:10.36 | jeff_phillips | the best answer they've come up with is that the scan button produces "nicer" results and therefore it doesn't "match" the poor quality of the other faxes they have saved in the same folder |
| 16:10.56 | deadpigeon | nicer results, that's unacceptable. |
| 16:11.08 | jeff_phillips | yeah, lol! that's their complaint! |
| 16:11.18 | jaytee | DavidR2008, so just do a pattern match for standard numbers to dial out the PRI and a pattern match for 18XXNXXXXXX numbers to go out the FXO zap channel number |
| 16:11.33 | jeff_phillips | and as such they've actually put a ticket in for me to 'fix' this 'problem' |
| 16:11.51 | DavidR2008 | I'll be picking up a HT 286 ATA and testing this out post haste |
| 16:12.01 | jaytee | jeff_phillips, can't you just close the ticket with a PEBKAC or a RUTOK? |
| 16:13.00 | deadpigeon | jeff_phillips: In situations like that, I tell people "That isn't possible, because it will cause I D ten T errors on our equipment." |
| 16:13.03 | deadpigeon | heh =/ |
| 16:13.15 | jaytee | hehehe |
| 16:13.29 | jeff_phillips | Well they've pressured me for details as to why it's a problem now and it wasn't a problem when we just had the fax line plugged directly into the POTS line |
| 16:13.41 | jaytee | "that's a Layer 8 switching problem" |
| 16:13.54 | deadpigeon | It's an ID10T error! |
| 16:13.59 | jeff_phillips | so I tried to explain it to them that when we did it that way the outbound caller ID was incorrect and showed our phone number, not our fax number. |
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| 16:14.19 | jeff_phillips | So now that I have the fax's outbound caller ID correct, they want me to change it to not be correct anymore so that they can continue sending faxes to themselves instead of pushing scan |
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| 16:15.18 | jaytee | jeff_phillips, sounds like one of those "stay with Charlie Babbitt or go back to Wallbrook" issues. |
| 16:17.37 | sacitec | hello everyone, i was doing research about wireless IP phone 7925G(http://www.cisco.com/en/US/products/ps9900/index.html) that uses sccp protocol. I see in voip-info.org that this model is supported. Does anyone have a success case with this protocol and this ip phone model ? thanks in advance |
| 16:18.31 | DavidR2008 | jeff_phillips: is this in *? if so couldn't you munge the the outbound caller id if they're dialing their own number? (Not really suggesting continuing to support their bad habbits) |
| 16:22.10 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
| 16:27.38 | jeff_phillips | No I just put an ATA device and went straight to the SIP provider for a cheap fix rather than having to fuss with setting everything up properly |
| 16:28.03 | jeff_phillips | So yes, I could "fix" it so they can fax themselves, i suppose... but I'd rather they just learn to push the right button |
| 16:28.18 | jeff_phillips | in other words, I'd rather they learn to do things right even when I don't. haha |
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| 16:30.06 | Ritzerisk | any of you gotten t38modem to work or even messed with it yet |
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| 16:30.29 | DavidR2008 | jeff_phillips: that's always best! :-D |
| 16:30.42 | jeff_phillips | lol |
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| 16:33.50 | jeff_phillips | One thing I'm wondering about -- does T38 get crummy without QoS on the IP connection? Because they have crappy IP connectivity in the building I threw that ATA in for their lousy fax machine. The phone extensions over there I have hard wired to a VoIP gateway in this building (different IP connection) |
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| 16:34.49 | hjb_256 | I'm using a softphone with ABE on the local LAN. The HOLD button on the softphone will place a call on hold but will not pick the call back up. Soft phones that traverse my InGate work fine. Hard phone on the LAN also work fine. Anyone have any suggestions? |
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| 16:45.07 | hjb_256 | nobody has any clues? |
| 16:45.17 | *** join/#asterisk martyn-dev2 (n=admin@190.24.134.154) |
| 16:45.19 | martyn-dev2 | Hi |
| 16:45.28 | martyn-dev2 | I need your help.. look it. |
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| 16:45.51 | CapriCoRN^80 | i am using Mv 370 gateway with asterisk. I have configured it and my sip users call to outside using sim. i just want to put a sound when i user call to MV sip account after that a sound is play saying enter your mobile no |
| 16:45.56 | awk_r | looks it. |
| 16:45.59 | CapriCoRN^80 | how can i accomplish that ? |
| 16:46.32 | martyn-dev2 | i have some grandstream bt100 and some bt200 phones on my networks companny. the bt200 registered ok but all the bt100 is not registered. :( |
| 16:46.40 | DavidR2008 | awk_r: you definitely look it! |
| 16:46.43 | martyn-dev2 | Yesterday was works :( but today no. |
| 16:47.25 | martyn-dev2 | my asterisk server is on network2 and my phones are on network1 |
| 16:48.13 | rue_mohr | and the address of your gateway is? |
| 16:48.41 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 16:49.51 | *** join/#asterisk codebanshee (n=chris@81.171.245.107) |
| 16:49.51 | awk_r | martyn-dev2, sounds like you need to replace your flux capacitor. |
| 16:49.51 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 16:49.54 | awk_r | (was just kidding). check your Asterisk CLI to see if it is receiving the SIP registers or not |
| 16:50.31 | martyn-dev2 | awk_r: not recive that. i try with ngrep -d any port 5060 | grep some_bt100ip, but not request is there :( |
| 16:50.58 | awk_r | martyn-dev2, then it looks like a network issue? talk to your network guru |
| 16:51.26 | martyn-dev2 | no, really not |
| 16:51.55 | awk_r | martyn-dev2, have you changed your phone configs recently? |
| 16:51.56 | jaytee | martyn-dev2, are the bt100's on the same net as the bt200's? |
| 16:52.01 | awk_r | or the IP of Asterisk? |
| 16:57.13 | zeeesh | how and where can i identify either both legs answered or not... i hv try to find in /var/log/asterisk/cdr-csv... but if peer B receiving ring from peer A , and if peer a will not even answer the call ... and peer B hangup ... it shows in Master.csv like this ... 17,17,"ANSWERED","DOCUMENTATION" ... how can i know either both legs open or not ? |
| 16:58.25 | CapriCoRN^80 | can i get some help ? ;) |
| 16:58.45 | *** join/#asterisk quintana (n=sylvain@aghnar.doowan.net) [NETSPLIT VICTIM] |
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| 17:00.18 | martyn-dev2 | awk_r: jaytee yes the network is the same |
| 17:01.17 | awk_r | martyn-dev2, and you've verified that the bt100 phones are attempting to register to the correction IP/domain |
| 17:02.13 | jaytee | martyn-dev2, yes, verify that the bt100's are set to register to the correct address |
| 17:02.55 | awk_r | s/correction/correct/ |
| 17:03.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:03.07 | *** join/#asterisk af_ (n=getsmart@88-149-240-30.dynamic.ngi.it) |
| 17:07.21 | Ritzerisk | anyone messed with getting t38modem to work |
| 17:07.29 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:08.46 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 17:12.41 | DavidR2008 | hey all, I've got a POTS line from the phone company and * isn't detecting hangup on it, any suggestions? |
| 17:15.55 | Chainsaw | DavidR2008: Normally means you need kewlstart signalling (if you're currently on loopstart, that is). |
| 17:16.07 | Chainsaw | DavidR2008: Also, make sure you're on the correct country setting for your telephony adapter. |
| 17:16.18 | DavidR2008 | that would be FXO_ks right? that's what it is now |
| 17:16.19 | Qwell | DavidR2008: What hardware? |
| 17:16.27 | DavidR2008 | Sangoma A200 |
| 17:18.34 | jaytee | DavidR2008, no you need to use fsx_ks signalling for FXO ports. I know that sounds backwards but that's just "the way it is" in happy Asterisk land. |
| 17:18.56 | jaytee | oops, that's fxs_ks not fsx_ks |
| 17:19.18 | DavidR2008 | sorry!!!! I started a whole firestorm a few minutes ago by saying the wrong port. it is fxs_ks |
| 17:19.44 | Chainsaw | DavidR2008: Alright, so you're on kewlstart already. Is your country set correctly? |
| 17:19.49 | *** part/#asterisk martyn-dev2 (n=admin@190.24.134.154) |
| 17:20.01 | *** join/#asterisk elred (i=sauron@fucksheep.org) |
| 17:20.14 | Qwell | elred: interesting...err...hostname |
| 17:20.23 | Qwell | the .org is a nice touch |
| 17:20.26 | jaytee | hahaha |
| 17:20.32 | elred | :D |
| 17:21.05 | Qwell | Are you 503(c)? |
| 17:21.07 | DavidR2008 | where is country set? |
| 17:21.15 | jaytee | bet if I did a whois on the domain it would either come up Wyoming or Australia (ducks and runs away) |
| 17:21.15 | Qwell | err, 501* |
| 17:21.26 | elred | I am trying to automate the detection of number of available channel (FXO) on a given zaptel-compatible card. Any idee how to do ? |
| 17:21.26 | Chainsaw | DavidR2008: Generally in the same config where you select fxs_ks signalling. |
| 17:21.26 | Qwell | both. whatever. |
| 17:22.17 | *** join/#asterisk martyn-dev2 (n=admin@190.24.134.154) |
| 17:22.23 | jaytee | elred, checkout the Groupcount dialplan function |
| 17:22.25 | martyn-dev2 | hi awk_r |
| 17:22.51 | martyn-dev2 | i'vebeen talked with network guru. and the request frombt100 is none across firewall :( |
| 17:23.07 | elred | Also, I am coding an AGI to allow outbound call thru Zap channels, but I don't know which one is in use and which one is free. I first thought of doing Dial(Zap/1/${number}&Zap/2/${number}&etc) but what is the right way to do it ? I am afraid it use all channel available trying all to call the same number, and if it doesn't reply within 30 seconde for example, all my ZAP channel will be "busy" when only one would have been enough. Any idee? |
| 17:23.15 | elred | jaytee : thx let's look at that |
| 17:23.50 | Qwell | elred: Zaptel has groups you can use... |
| 17:23.53 | DavidR2008 | I'm using dahdi and I can't find anything about setting a country in span_dahdi.conf |
| 17:24.03 | DavidR2008 | chan_dahdi.conf* |
| 17:24.04 | Qwell | So instead of dialing on a specific channel like Zap/1, you use Zap/g1, and it'll use any line in that group |
| 17:24.10 | Qwell | (any available line, I should say) |
| 17:24.19 | ayeso | Im trying to create a conference, i have setup the room in meetme.conf, when I try to join i get this error: app_meetme.c:800 build_conf: Unable to open pseudo device anyone know why? |
| 17:24.30 | Qwell | that's all built-in - you don't need any fancy logic for it |
| 17:24.36 | jaytee | DavidR2008, it's in /etc/dahdi/system.conf and it's called loadzone="countrycode" |
| 17:24.39 | elred | Qwell : oh I see, thanks |
| 17:24.40 | Qwell | ayeso: got ztdummy loaded? |
| 17:24.51 | *** join/#asterisk bob_slacker (n=alberthf@201.22.137.27) |
| 17:24.59 | ayeso | Qwell: probably not, what is it? |
| 17:26.01 | Gido-E | ayeso do you use dahdi? |
| 17:26.45 | ayeso | Gido-E: I dont know, I always install * from a repo, and I only use SIP, no cards |
| 17:27.00 | jaytee | ugh |
| 17:28.59 | Gido-E | ayeso is there a zaptel script in your /etc/init.d? |
| 17:29.40 | ayeso | Gido-E: no there is not |
| 17:30.16 | DavidR2008 | jaytee: I didn't have a system.conf I created on and put the line in it, but no change. |
| 17:30.23 | DavidR2008 | one* |
| 17:30.28 | *** join/#asterisk filo1234 (n=filo@unaffiliated/filo1234) |
| 17:30.36 | Chainsaw | DavidR2008: You want loadzone & defaultzone. |
| 17:30.41 | ayeso | Gido-E: core show version: Asterisk 1.4.23.1 |
| 17:30.42 | jaytee | DavidR2008, are you running dahdi or zaptel? |
| 17:30.46 | Chainsaw | DavidR2008: Afterwards, unload & reload your dahdi modules to apply the settings. |
| 17:30.53 | Gido-E | ayeso, you need a pseudo timer to get meetme working. That is your problem. You can get it with dahdi or zaptel. |
| 17:30.54 | DavidR2008 | dahdi |
| 17:31.24 | Chainsaw | DavidR2008: You may also need it on the module options itself. Before I got it right: |
| 17:31.25 | Chainsaw | Port 1: Installed -- AUTO FXO (FCC mode) |
| 17:31.26 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:31.29 | jaytee | DavidR2008, and these ports have been working? without a system.conf file? |
| 17:31.35 | Chainsaw | And after I fixed that up: |
| 17:31.36 | Chainsaw | Port 1: Installed -- AUTO FXO (UK mode) |
| 17:32.00 | ayeso | Gido-E: know of any URL that can get me started? I do have dahdi in my init.d |
| 17:32.04 | DavidR2008 | well, sort of. I can dial out and dial in, the * server just don't recognize when the far end hangs up |
| 17:32.05 | filo1234 | hi i have installed asterisknow, and i have connect 2 analogic phones on my card, but i don't know how to test it |
| 17:32.24 | DavidR2008 | I just installed the card about 30 minutes ago |
| 17:32.25 | [TK]D-Fender | filo1234: place calls <- |
| 17:32.53 | jaytee | I wouldn't think that dahdi would even work without a system.conf file |
| 17:32.54 | jaytee | odd |
| 17:32.55 | Gido-E | ayeso, i sould try /etc/init.d/dahdi start . then restart asterisk |
| 17:33.23 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 17:33.33 | hardwire | anybody using broadvoice.. please plug your ears |
| 17:33.35 | hardwire | &(%$()#*)(#*$)(*)(*#)($*% |
| 17:33.39 | hardwire | ok it's over. |
| 17:33.52 | hardwire | rather.. anybody here working for them.. :P |
| 17:33.52 | Gido-E | ayeso : check with lsmod if the modules are loaded: dahdi, dahdi_dummy, dahdi_transcode |
| 17:34.07 | filo1234 | [TK]D-Fender: where? |
| 17:34.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:34.30 | [TK]D-Fender | filo1234: Configure your devices & your dialplan and place calls. |
| 17:34.44 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 17:36.47 | ayeso | Gido-E: I get a bunch of failures when trying to start dahdi.... should i reinstall it? |
| 17:37.10 | jeff_phillips | [TK]D-Fender: a bit embarassed to admit but in wiring up this new amplifer i discovered that using the 70 volt speaker output to use speakers with transformers requires the entire amp be put in mono mode instead of stereo. so much for trying to seperate the left / right * console audio output as seperate extensions |
| 17:37.31 | Gido-E | ayeso hmm, i dont know. Did you upgrade the kernel? |
| 17:38.30 | ayeso | Gido-E: i probably have since the initial installation of * |
| 17:38.35 | [TK]D-Fender | jeff_phillips: SMRT |
| 17:38.43 | Gido-E | ayeso then i would give it a try. |
| 17:38.54 | ayeso | Gido-E: ill give a shot |
| 17:39.03 | filo1234 | [TK]D-Fender: sorry but i don't know how configure my devices |
| 17:39.05 | [TK]D-Fender | jeff_phillips: Still a few more corners to paint yourself into.... keep at it DaVinci |
| 17:39.10 | [TK]D-Fender | ~book |
| 17:39.11 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:39.13 | [TK]D-Fender | filo1234: ^^^^ |
| 17:39.30 | filo1234 | :( |
| 17:39.50 | jeff_phillips | [TK]D-Fender: Working on it. believe it or not I got 3 buckets of paint sitting behind me & am intending on finshing painting into the corner of my office next week hopefully |
| 17:41.26 | DavidR2008 | I can't find a system.conf sample file. I tried running make samples in my * src directory |
| 17:42.04 | Ritzerisk | anyone messed with getting t38modem to work |
| 17:44.15 | [TK]D-Fender | DavidR2008: maybe because it isn't part of ASTERISK |
| 17:45.35 | DavidR2008 | well chan_dadhi is in * menuconfig so I thought it was worth a try |
| 17:45.44 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 17:45.54 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 17:46.19 | CapriCoRN^80 | [TK]D-Fender: i am using Mv 370 gateway with asterisk. I have configured it and my sip users call to outside using sim. i just want to put a sound when i user call to MV sip account after that a sound is play saying enter your mobile no |
| 17:46.55 | [TK]D-Fender | CapriCoRN^80: "core show application playback" |
| 17:47.13 | CapriCoRN^80 | [TK]D-Fender: any site containing example of such scanrio |
| 17:47.29 | *** part/#asterisk codebanshee (n=chris@81.171.245.107) |
| 17:47.31 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 17:47.33 | [TK]D-Fender | CapriCoRN^80: Scenario? Its a &^%#$ing single PLAYBACK |
| 17:47.43 | [TK]D-Fender | SHEESH!! :p |
| 17:47.48 | DavidR2008 | ok, I continue to stumble through this. :-S I didn't run dahdi_cfg which creates system.conf |
| 17:49.03 | jeff_phillips | where to i turn up the volume level on the console sound card output? |
| 17:50.17 | [TK]D-Fender | jeff_phillips: alsamixer |
| 17:50.54 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.66) |
| 17:51.45 | DavidR2008 | I'm not thinking clearly. I'll come back after lunch and take a shot at this again. thanks everyone for the help |
| 17:51.51 | jeff_phillips | thanks. :) |
| 17:54.34 | jeff_phillips | Ok, I get the master, PCM, CD, line in, etc etc... But what's this volume slider for an item named "IEC958 Playback AC97-SPSA"? |
| 17:54.51 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 17:55.35 | raden_work | we have 22 phone lines total between office and apartments is there a way to get like wholesale pricing on SIP ? |
| 17:55.56 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 17:56.07 | CapriCoRN^80 | [TK]D-Fender: you are right but as i told you in my case that i am little confused. As when sip user dial exten 200x which is my MV 370. it will connect to it and i get a different dial tone and i enter any mobile no to dial |
| 17:56.29 | jeff_phillips | raden_work: there are tons of sip termination providers. where do most of your calls go? |
| 17:56.36 | [TK]D-Fender | CapriCoRN^80: Go read your device's manual |
| 17:56.39 | CapriCoRN^80 | [TK]D-Fender: i placed playback after dial but it didnt work |
| 17:57.05 | raden_work | jeff, mostly inbound i figured switch apartments over save tenants money verizon 66 mo for unlimited long distance up here |
| 17:57.06 | [TK]D-Fender | CapriCoRN^80: Nevermind the "playback", that was before you properly explained what you wanted |
| 17:57.16 | CapriCoRN^80 | [TK]D-Fender: why device manual , why not * ? |
| 17:57.22 | raden_work | i mean i want to be down to like $ 20 per line |
| 17:57.36 | raden_work | jeff_phillips, ^ |
| 17:58.03 | [TK]D-Fender | CapriCoRN^80: because if your device gives a dialtone to * then IT has already answered the call and is in control. |
| 17:58.07 | jeff_phillips | raden_work: I was looking at didforsale.com It's a worldcom reseller $8.99/did |
| 17:58.20 | jeff_phillips | 20 channels unmetered, and 20% of the channels can be shared |
| 17:58.38 | raden_work | what you mean shared ? |
| 17:58.42 | CapriCoRN^80 | [TK]D-Fender: ok |
| 17:59.28 | jeff_phillips | raden_work: from what they tell me, each DID has 20 channels. If you need more it's $2 per channel, but if some DIDs are only using a couple of channels while you're going over 20 channels on just one or two of them, you can "share" 20% of the other DID's unused channels to pool with the high-traffic DID |
| 17:59.30 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-9c42d130b8ef67f6) |
| 17:59.47 | jeff_phillips | I'm using Gafachi.com for outbound right now. Seems to work well. |
| 17:59.50 | [TK]D-Fender | jeff_phillips: DID's don't have "channels" |
| 17:59.59 | raden_work | yeah im confussed |
| 18:01.25 | raden_work | [TK]D-Fender, well i see why he said that checkout this http://www.didforsale.com/ |
| 18:02.10 | raden_work | what do they mean 20 channels on one number im lost |
| 18:02.43 | *** join/#asterisk jeff_phillips (n=ceramics@66-112-49-13.stat.centurytel.net) |
| 18:02.50 | raden_work | 5. Each DID comes with Up to 20 Simultaneous channels. << on there website |
| 18:03.06 | jeff_phillips | back sorry, DSL seems to be crappy todya |
| 18:03.26 | filo1234 | [TK]D-Fender: can i configure all from webpage? |
| 18:03.59 | jeff_phillips | Yes, each DID includes 20 channels, $2/channel for additional. OR if you have multiple DIDs on the same account you can share 20% of the included 20 channels on the lightly used DIDs with the channels you need over 20 in the high traffic DIDs |
| 18:04.02 | [TK]D-Fender | filo1234: GUI's are not supported in this channel, please read the topic for a list of other related palces |
| 18:04.35 | filo1234 | sorry |
| 18:05.28 | raden_work | whats the difference in a DID vs a channel I'm confused |
| 18:05.37 | jeff_phillips | DID = a phone number |
| 18:05.43 | raden_work | yes |
| 18:05.49 | jeff_phillips | # of channels = number of simultaneous phone calls |
| 18:06.31 | jeff_phillips | You pay $8.99 and you can receive 20 simultaneous phone calls on that phone number. Additional simultaneous calls = $2/month or you do the 20% sharing trick |
| 18:07.01 | jeff_phillips | it's the cheapest provider I've found for high-volume inbound calls |
| 18:07.03 | *** join/#asterisk BBMitch (n=Miranda@66.199.170.249) |
| 18:07.08 | jeff_phillips | for low volume you can do better |
| 18:07.31 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:07.32 | raden_work | how manyoutbound can i have |
| 18:07.34 | raden_work | ???????? |
| 18:07.43 | jeff_phillips | None from them. They only sell inbound. |
| 18:07.44 | *** part/#asterisk filo1234 (n=filo@unaffiliated/filo1234) |
| 18:07.51 | raden_work | im confussed |
| 18:07.57 | raden_work | i need inbound & outbound |
| 18:08.10 | jeff_phillips | Just setup a different provider with low rates for SIP termination for your outbound |
| 18:08.22 | *** join/#asterisk HouseMD (n=nandateb@unaffiliated/geek) |
| 18:08.28 | jeff_phillips | I'm using gafachi.com for my outbound, but you can setup multiple SIP providers to do least cost routing depending on where you are dialing |
| 18:09.15 | jeff_phillips | The package deals that include both inbound & outbound usually cost more than if you break it down into specific services |
| 18:09.21 | raden_work | sip termination is outbound i take it origination is inbound ? |
| 18:09.23 | BBMitch | Hi guys - I have a weird RTP issue - I captured it with wireshark - an older phone (linksys SPA942 with firmware 4.1.18) starts a malformed rtp as identified by wireshark - thish causes asterisk to start a stream of about 10000 packets per second - when a few of these occur at once the switch flips into blocking mode... is this a known issue? Even though the firmware should be updated, it does see |
| 18:09.23 | BBMitch | m like a vulnerability - what should I do to help - I have packet traces |
| 18:09.36 | jeff_phillips | yes, termination = outbound, origination = inbound |
| 18:10.57 | raden_work | whgats a rate center for origination ? |
| 18:11.20 | jeff_phillips | rate center is the city the phone number is in that people dial to reach you on. |
| 18:11.21 | bmoraca | raden_work: doesn't usually matter. rates are billed based on source and destination ANIs |
| 18:11.31 | BBMitch | raden: rate centers are the local region around a calling area... |
| 18:11.54 | DavidR2008 | ok, I finally have everything sorted out. Here are my two config files: http://pastebin.com/d529695bd the problem is that * doesn't recognize hangups from the POTS line |
| 18:11.57 | BBMitch | raden: lnp (local number portability) is only possible WITHIN a rate center for example |
| 18:12.13 | raden_work | ok that makes sense |
| 18:12.17 | jeff_phillips | Good point, you probably want to keep your existing phone numbers, right? |
| 18:12.44 | raden_work | dont really matter but its nice |
| 18:12.55 | *** join/#asterisk roy_hobbs (n=roy_hobb@pool-96-242-209-249.nwrknj.fios.verizon.net) |
| 18:13.03 | raden_work | i just want $20 per phone number but i want to have extra outgoing if needed |
| 18:13.07 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176207886.dsl.bell.ca) |
| 18:13.24 | raden_work | $20 unlimited incoming / outgoing nationwide or less |
| 18:13.44 | jeff_phillips | raden_work: I was unable to port our local POTS # to any DID provider because of limited CLEC competition in our rate center |
| 18:13.57 | BBMitch | Does anyone know the prefered method for submitting a bug report - I don't know for sure, but this RTP issue could be a possible denial of service attack on asterisk |
| 18:14.05 | *** part/#asterisk roy_hobbs (n=roy_hobb@pool-96-242-209-249.nwrknj.fios.verizon.net) |
| 18:17.50 | [TK]D-Fender | DavidR2008: You need to ask your telco to provide CDS. |
| 18:17.52 | [TK]D-Fender | ~cds |
| 18:17.52 | jbot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 18:18.27 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
| 18:18.29 | DavidR2008 | [TK]D-Fender: thanks |
| 18:22.15 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
| 18:23.07 | BBMitch | I see the mantis tool - I guess I'll post some information there - but I don't think I can attach the packet trace - it's only 1 minute long but it's over 100MB |
| 18:25.16 | Qwell | BBMitch: not much the bug tracker can do to help with a DoS.. |
| 18:25.30 | BBMitch | It's not a DOS though.... |
| 18:25.39 | BBMitch | I have it happening twice. |
| 18:25.51 | Qwell | you just said you thought it might be |
| 18:26.06 | BBMitch | Said it could be turned into one... First the phone sends an RTP packet which wireshark reports is "malformed" |
| 18:26.26 | BBMitch | Then asterisk responds with about 10000pps of rtp traffic until the port is blocked |
| 18:26.48 | Qwell | hmm |
| 18:26.50 | BBMitch | I have the packet capture, the asterisk logs... |
| 18:27.31 | BBMitch | It only happens with a phone with older firmware, so I can fix MY problem by updating the phone, but if a phone can trigger that effect I figure something else could to... |
| 18:28.02 | Ritzerisk | anyone gotten any t38 working .. .. |
| 18:28.29 | raden_work | i know this is problaly a dumb question but my phones are G.711 is there a way to run g.729 to my provider and g.711 internally ? |
| 18:28.52 | BBMitch | it seems to happen around the time the phone call is put on hold - then he puts a second call on hold to return to the first, but these rtp streams saturate the update storm limit on the switch triggering blocking mode |
| 18:29.26 | BBMitch | We've seen it happen a few times, but this time we had wireshark running (we leave it running all the time now ;-) |
| 18:30.10 | Qwell | BBMitch: well, if you can show the trace leading up to it, that might be most useful |
| 18:31.08 | [TK]D-Fender | raden_work: Of course, but you'll need to purchase G.729 licenses for * to be able to transcode to it |
| 18:31.10 | BBMitch | I have asterisk full output and the full packet capture - there's no "gdb" output or similar though. Is that enough? |
| 18:33.04 | raden_work | so trascode at my server i can run phones 711 on network and 729 to provider from server |
| 18:34.06 | [TK]D-Fender | raden_work: Sure |
| 18:36.34 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:43.40 | *** join/#asterisk stoffell (n=kristof@d51A4D629.access.telenet.be) |
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| 18:52.54 | *** join/#asterisk specialist1 (n=research@119.160.105.172) |
| 18:53.09 | *** join/#asterisk [netman] (n=netman@142.Red-83-50-80.dynamicIP.rima-tde.net) |
| 18:57.05 | Shadad | Trying to reslove a problem installing asterisk-addons on FreeBSD 7. ./configure works fine but running gmake I get "error: asterisk.h: No such file or directory". I have tried ./configure --with-asterisk= pointing it to my src directory and adding +CFLAGS+=-I to the makefile with no success. Any suggestions? |
| 18:59.01 | mmlj4 | Shadad: stick your asterisk source in /usr/include and recompile that, then work on the addons |
| 18:59.19 | mmlj4 | /usr/include/asterisk, i mean |
| 18:59.49 | Shadad | Thanks mmlj4, ill give it a try |
| 19:00.14 | *** join/#asterisk stoffell (n=kristof@d51A4D629.access.telenet.be) |
| 19:02.16 | rue_mohr | I need a variable name that holds the extension number of the person who is on the phone |
| 19:02.22 | rue_mohr | ? |
| 19:03.44 | EmleyMoor | I have a macro that is called on answer by using the M option of the Dial app... is there any way in 1.4 to make this a Gosub instead? |
| 19:04.29 | rue_mohr | hmm could you show us what you want to change using a pastebin site? |
| 19:04.31 | rue_mohr | ~pb |
| 19:04.32 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 19:04.52 | Katty | mmm, bww |
| 19:05.11 | Lyma | http://paste.lymas.com.br |
| 19:05.12 | Lyma | :D |
| 19:05.35 | Shadad | rue_mohr: ${EXTEN} is a predined variable that tells you the current extension. |
| 19:05.46 | rue_mohr | no it tells you what the user dialed |
| 19:05.52 | rue_mohr | not what their phone is |
| 19:05.56 | [TK]D-Fender | rue_mohr: No, it doesn't |
| 19:06.05 | rue_mohr | yea, I been using it |
| 19:06.10 | [TK]D-Fender | [15:05]<Shadad>rue_mohr: ${EXTEN} is a predined variable that tells you the current extension. <- true |
| 19:06.28 | rue_mohr | then why does it only contain what they dialed? |
| 19:06.40 | rue_mohr | I want to kow who they are not what they dialed |
| 19:07.11 | rue_mohr | I think I worded my question wrong to be fair |
| 19:07.21 | rue_mohr | I want to know the call origionator |
| 19:07.49 | rue_mohr | but I think that would be a problem if I didn't have the sip accounts with the same number as their extension |
| 19:07.57 | [TK]D-Fender | rue_mohr: Look at the channel name |
| 19:08.10 | rue_mohr | ok, ${CHAN}? |
| 19:08.30 | [TK]D-Fender | rue_mohr: Go rad the CHANNELVARIABLES texts in your DOCS folder |
| 19:08.49 | rue_mohr | hah, silly me, I had my head burried in the book |
| 19:10.27 | EmleyMoor | http://paste.debian.net/31089/ shows the macro and why I think it currently has to be one... |
| 19:11.06 | rue_mohr | sweet |
| 19:12.39 | *** join/#asterisk Snublefot (n=kjell@c23B100C3.dhcp.bluecom.no) |
| 19:16.05 | *** part/#asterisk surfdue (n=anthonyu@unaffiliated/surfdue) |
| 19:17.33 | EmleyMoor | I have migrated all my other "no parameters" macros... |
| 19:19.54 | EmleyMoor | I have to wait for 1.6 to migrate the others |
| 19:22.10 | *** join/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) |
| 19:27.24 | leifmadsen | Asterisk 1.6.0.7-rc2, 1.6.1.0-rc3, 1.6.2.0-beta1 & Asterisk-Addons 1.6.0.2-rc1, 1.6.1.0-rc3 Now Available! |
| 19:27.34 | leifmadsen | See http://www.asterisk.org |
| 19:27.46 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 19:28.16 | jplank | w00t rc2! |
| 19:28.32 | EmleyMoor | I'm likely to be sticking with 1.4 for some time |
| 19:29.33 | jplank | I wont be upgrading existing clients, but I'm looking forward to start working with it in my lab |
| 19:29.46 | jplank | I'm looking forward to the better video codec negotiation |
| 19:31.34 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 19:31.43 | RoPBX | hello all |
| 19:32.05 | RoPBX | please how do I activate MySQL CDR on asterisk? |
| 19:32.29 | Katty | so i'm thinking of taking the rest of the day off |
| 19:33.22 | shyam_k | after configuring the s extension, to make it working, how to call? i just call "s" from the softphone? |
| 19:33.50 | Katty | 1111111 => Goto(context,s,1) |
| 19:34.01 | Katty | or something like that |
| 19:35.08 | Katty | exten => 11111111,1,Goto(context,s,1) |
| 19:35.18 | Katty | being braindead today, apparently |
| 19:35.39 | shyam_k | hmm.. k.. |
| 19:36.12 | rue_mohr | anyone know reasons why read wouldn't work? |
| 19:36.30 | Katty | because you're not giving it anything to read. |
| 19:36.36 | rue_mohr | aka, you sit there pushing buttons like mad and it times out with "no user input" or the like |
| 19:37.01 | Katty | i think i have a read example on my blog. |
| 19:37.10 | rue_mohr | url? |
| 19:37.35 | Katty | http://angela.sleekgeek.org/2008/03/18/passing-variables-from-asterisk-to-email/ |
| 19:37.39 | Katty | you can dig the read bits out |
| 19:37.41 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 19:38.15 | rue_mohr | huh, thats no different thatn what I did |
| 19:38.27 | rue_mohr | accept my variable wasn't all uppercase |
| 19:38.40 | mchou | Call 1 Round Trip Delay:34218843 ms |
| 19:38.43 | Katty | then as fender says |
| 19:38.46 | Katty | pastebin |
| 19:38.48 | mchou | wtf?? |
| 19:38.51 | Katty | cause we don't read minds :P |
| 19:39.06 | rue_mohr | yep, just asking if there was a common gotcha |
| 19:39.28 | shyam_k | the TFOT says "when calls enter a context without a specific destination extension( for example, a ringing FXO line), they are passed to the extension." i didn't get what that situatio n is.. |
| 19:39.35 | Katty | http://angela.sleekgeek.org/2008/03/13/survey/ <- more read |
| 19:40.42 | Katty | i have another read example in some call barging stuff |
| 19:40.55 | Katty | but it's not user input |
| 19:41.26 | Katty | hi moggy |
| 19:41.43 | Mog | hi katty |
| 19:41.51 | Mog | i need to figure out why it keeps doing that |
| 19:48.53 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 19:49.39 | Katty | hehe peas out |
| 19:51.19 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 19:58.43 | hardwire | does anybody have a firmware pusher for an older spa941? |
| 19:58.53 | hardwire | it's not looking on my servers via the provisioning options on it's web ui |
| 19:58.58 | hardwire | and ignoring dhcp |
| 19:59.08 | rue_mohr | sweet, I have an extension that reads you your incomming extension number |
| 20:00.03 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 20:00.20 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 20:01.54 | rue_mohr | if I make a global thats set to ${CHANNEL:4:2} will it be reevaluated properly for each call? |
| 20:01.58 | hardwire | it's completely ignoring provisioning.. even directly using the UI |
| 20:02.34 | rue_mohr | I dont know when the globals are evaluated, if its when * starts of if its done realtime aka substitution |
| 20:03.10 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-8aaca014932f5438) |
| 20:05.00 | [TK]D-Fender | rue_mohr: No, you cannot |
| 20:05.09 | rue_mohr | ok, thanks |
| 20:05.25 | rue_mohr | so their not macros |
| 20:12.38 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 20:14.14 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:14.38 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:15.05 | mcargile | does the TDMoE driver provide any kind of a timing signal or do I need to use like ztdummy or a real tdm card with it? |
| 20:17.43 | jplank | grrr my polycom distributor's support sucks. I call to find out if the new vvx1500 can make a SIP video call to their HDX or VSX series, his first answer in a snotty attitude was "sir video is built into the SIP standard" then I reply saying I understand that, but I'm asking if the HDX/VSX support video over the SIP standard, he then replies with the same thing with the same attitude "video is built into the SIP standard so of cour |
| 20:18.26 | [TK]D-Fender | jplank: Lets summarize then : "YES" |
| 20:18.53 | jplank | actually the answer is NO, not until the end of the year |
| 20:19.00 | *** join/#asterisk neurosys (n=neurosys@173.9.159.182) |
| 20:19.32 | jplank | They are pushing this vvx so hard, yet it doesn't even interop with their other video conferencing products |
| 20:20.28 | jplank | so now I have a VVX1500 on order, and nothing to do with it until the end of the year |
| 20:20.39 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 20:20.49 | jplank | anyone have a video phone and looking for someone to call :) |
| 20:21.30 | jplank | fender, maybe I could show you how well I've gotten my g chords :P |
| 20:21.52 | telnettech | i am wanting to setup a small 5 user test system with baisc functionality that can ater be expanded as I need to. Which version of asterisk would be stable enough to run as a home PBX. |
| 20:22.05 | jplank | any |
| 20:22.55 | [TK]D-Fender | telnettech: any |
| 20:22.55 | Nugget | telnet is eeeeeeevil! |
| 20:24.29 | jplank | I once had a customer tell me he didn't want me to enable telnet on a router I was installing because its so much easier to brute force telnet then ssh....I went to explain to him why that wasn't true..stopped...and just agreed |
| 20:25.44 | [TK]D-Fender | checkout time, BBIAB |
| 20:28.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:30.51 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 20:31.05 | ayeso | how can I tell if I have zaptel or dahdi installed? |
| 20:31.06 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 20:31.29 | jplank | did you install it? |
| 20:32.02 | ayeso | jplank: Im looking for help. |
| 20:32.29 | jplank | if your using zap the command is zap channel status |
| 20:32.44 | jplank | I dont know the command for dahdi, but I'm sure its similar |
| 20:33.41 | jplank | zap show status** |
| 20:34.34 | ayeso | jplank: Well let me explain what Im trying to accomplish. I use asterisk stictly for SIP traffic, no cards are installed. I am trynig to get the meetme app to work, but when I try to join a meeting it fails bitching about "unable to open pseudo device" I was advied that I needed zaptel or dahdi. |
| 20:35.06 | jplank | theres ztdummy for that if you don't have a zaptel card |
| 20:35.21 | jplank | zaptel/dahdi is for analog or T1 devices |
| 20:35.37 | jplank | do a zap show channels |
| 20:35.43 | jplank | do you see somthing along the lines of |
| 20:35.49 | jplank | pseudo default en default |
| 20:36.40 | *** part/#asterisk bob_slacker (n=alberthf@201.22.137.27) |
| 20:36.41 | ayeso | jplank: yes but nothing appears below, other than some verbiage advising to use dahdi instead |
| 20:37.39 | ayeso | core show version: Asterisk 1.4.23.1 |
| 20:39.21 | mcargile | then you do not have a timing source loaded. |
| 20:41.00 | mcargile | try running "modprobe ztdummy" on the command line. If that fails try running "modprobe dahdi_dummy". and if that fails install either of them and try again |
| 20:43.29 | ayeso | mcargile: odd, when I do modprobe ztdummy, I get an error... but when I do, modprobe dahdi_dummy, there is no output at all... I did a locate and found: /lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_dummy.ko |
| 20:43.47 | mcargile | then you have dahdi compiles |
| 20:43.53 | mcargile | *compiled |
| 20:44.02 | mcargile | try meetme now |
| 20:44.28 | ayeso | mcargile: I get the error that no pseudo device an be found |
| 20:44.41 | mcargile | you might need to restart asterisk |
| 20:45.25 | ayeso | mcargile: I have.... unfortunately no luck. Should there be a dahdi daemon running? |
| 20:45.58 | mcargile | nope... sorry I dont use dahdi so I am at the end of my knowledge |
| 20:46.25 | ayeso | mcargile: Thanks anyway... |
| 20:46.28 | mcargile | np |
| 20:48.05 | jsmith | ayeso: What kernel are you currently running (type "uname -a") |
| 20:48.36 | ayeso | 2.6.18-92.1.22.el5 |
| 20:48.54 | ayeso | ll /etc/init.d/ | grep -i dahdi = nothing |
| 20:50.01 | ayeso | ll /etc/init.d/ | grep -i zap = nothing |
| 20:50.01 | jaytee | ayeso, did you run make config after running make with dahdi? |
| 20:50.01 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:50.01 | jaytee | and chkconfig dahdi on? |
| 20:50.56 | jsmith | ayeso: It looks like you've got DAHDI compiled (at least the dahdi_dummy driver) for your kernel, but don't have it loading with an initscript |
| 20:50.56 | ayeso | jaytee: well I actually installed with YUM from the atrpms repo.. I uninstaled it earlier today after talking to you guys and complied from source, but had the exact same issues, so I have now reinstalled via YUM because I prefer to use a package manager whenever possible. |
| 20:51.21 | jaytee | ~wglwat |
| 20:51.22 | jbot | from memory, wglwat is well, good luck with all that |
| 20:52.23 | jaytee | ayeso, try running dahdi_cfg -vvv from a terminal and see what the output says |
| 20:52.35 | ayeso | jaytee: 1 sec |
| 20:52.42 | jaytee | that's assuming your "package" contains the dahdi tools |
| 20:54.16 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 20:54.30 | ayeso | dahdi_cfg not in the path anywhere.. but located it in /usr/sbin... I get the following output http://pastebin.ca/1366509 |
| 20:54.42 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:55.01 | pigpen | hi all, I setup a test server with Asterisk 1.6.0.6, with a single fxo module (digium). I am getting: chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 4 (No such device) |
| 20:55.27 | pigpen | Of course I am getting a huge echo on calls via this port. |
| 20:55.36 | rue_mohr | zaptel? |
| 20:55.38 | rue_mohr | dahdi? |
| 20:55.40 | pigpen | But it is only my side that is geting echo. |
| 20:55.48 | jaytee | dahdi, since it's 1.6 |
| 20:55.52 | pigpen | dahdi, zap is gone in 1.6 |
| 20:55.57 | rue_mohr | you didn't but the echo canceler card did you? |
| 20:56.14 | jaytee | pigpen, pastebin your system.conf and chan_dahdi.conf files |
| 20:56.22 | pigpen | jaytee, k. |
| 20:56.46 | rue_mohr | your gonna need it add $300 to budget and done pass go |
| 20:57.14 | rue_mohr | you can try getting the hplec to work, good luck |
| 20:57.42 | rue_mohr | I couldn't on 1.4, the dahdi drivers and hplec dont seem to be compatible with each other* *yet |
| 20:57.50 | jaytee | rue_mohr, why? I'm using the TDM410 card I got in class with one FXS and one FXO and I use software EC with mg2. Works fine for me. |
| 20:58.23 | rue_mohr | all your guys help with software echo here didn't do a thing, you using it with pots lines? |
| 20:58.30 | rue_mohr | yea |
| 20:58.33 | rue_mohr | k, good |
| 20:58.54 | jaytee | no, I'm using it with kettle lines |
| 20:59.23 | rue_mohr | my experiance is that 1.4.22 with dahdi cannot work with oslec |
| 20:59.35 | rue_mohr | if there was a channel history you could learn more |
| 20:59.53 | pigpen | jaytee, well, my config is pretty bare, standard loadzone = us, fxoks=1, fxoks=2, fxoks=3, fxsks=4 |
| 21:00.06 | pigpen | and echocanceller=mg2,1-4 |
| 21:00.10 | jaytee | rue_mohr, I'm sure if I searched the chat archives on rikers.org I'd fine all kinds of cool stuff with your nick attached |
| 21:00.26 | rue_mohr | I wouldn't say cool, but ok |
| 21:00.27 | jaytee | pigpen, you only have one module you said? |
| 21:00.33 | pigpen | well, one fxo. |
| 21:00.35 | pigpen | :) |
| 21:01.06 | jaytee | ok, so 3 fxs mods (green) and 1 fxo (red) |
| 21:01.24 | jaytee | is this Digium card? |
| 21:01.30 | pigpen | I have been running since 1.2.3, been on bleeding edge through 1.4.x |
| 21:01.31 | pigpen | yeah. |
| 21:01.44 | pigpen | but, just haven't taken the time to get familiar with 1.6 |
| 21:02.34 | jaytee | and you have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes in chan_dahdi.conf? |
| 21:03.20 | pigpen | echocancel=yes, yes. bridged = no, training = 800 |
| 21:03.49 | pigpen | one thing to note, when I run dahdi_cfg -d -5 -f -v I get: |
| 21:04.13 | pigpen | All is fine but: |
| 21:04.13 | pigpen | Setting echocan for channel 1 to mg2 |
| 21:04.13 | pigpen | DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) |
| 21:05.01 | pigpen | hmm..maybe the echocanceller port range does not like the range....I'll do one line for each. |
| 21:05.54 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 21:06.23 | pigpen | nope. |
| 21:06.42 | pigpen | I'll set the echo settings in chan_dahdi to your reccomendations. |
| 21:06.47 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 21:08.15 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:09.01 | pigpen | chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 4 (No such device) |
| 21:09.07 | pigpen | ^^upon a call. |
| 21:10.06 | jaytee | the syntax should be correct for your system.conf echocanceller=mg2,1-4 |
| 21:10.28 | pigpen | yeah, it didn't help splitting them out. |
| 21:10.32 | rue_mohr | ah thats it, the dahdi drivers bailed every time I turned on the software echo can |
| 21:11.55 | pigpen | I'll get it running some how. For that matter, I have had more issues with echo canceler hardware than they are worth. |
| 21:12.02 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:12.42 | jaytee | pigpen, do a locate dahdi_echocan_mg2.ko from the command line |
| 21:13.06 | pigpen | /lib/modules/2.6.26-hardened-r9/dahdi/dahdi_echocan_mg2.ko |
| 21:13.14 | pigpen | maybe it doesn't like mg2 |
| 21:14.33 | jaytee | so try kb1, sec or sec2? |
| 21:15.30 | pigpen | noe. |
| 21:15.32 | pigpen | nope. |
| 21:16.20 | pigpen | http://markmail.org/message/66wqebwbgzdpqju7 |
| 21:18.12 | jaytee | well, that's oslec, not mg2 or kb1 |
| 21:19.53 | pigpen | yeah |
| 21:20.17 | jaytee | pigpen, what happens if you just run dahdi_cfg -vvv instead of -d -5 -f -v? |
| 21:20.56 | *** join/#asterisk s14ck (i=c82302dc@gateway/web/ajax/mibbit.com/x-fd3de9a5e20e84c2) |
| 21:21.38 | pigpen | Setting echocan for channel 1 to kb1 |
| 21:21.38 | pigpen | DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) |
| 21:21.47 | pigpen | ^^^less output, but same resule. |
| 21:22.00 | pigpen | s/resule/result |
| 21:22.25 | jaytee | is the echocanceller="something" the last line in system.conf? |
| 21:22.32 | pigpen | I bet the version of dahdi has an incomplete echo package. |
| 21:22.44 | pigpen | yes, last line |
| 21:22.47 | s14ck | moy: hi, i install SVN-moy-mfcr2-r182170 but i can't load chan_dahdi.so |
| 21:22.50 | jaytee | what version? |
| 21:23.11 | *** join/#asterisk ta^3 (n=tacvbo@189.146.176.251) |
| 21:23.44 | pigpen | dahdi - 2.1.0.4 |
| 21:23.51 | pigpen | dahdi-tools 2.1.0.2 |
| 21:24.15 | jaytee | try going backwards. use 2.0.0 |
| 21:24.33 | jaytee | and download the dahdi complete package |
| 21:24.53 | pigpen | yeah, kinda what I was thinking. |
| 21:24.59 | s14ck | i did try with 2.1.0.3 |
| 21:25.06 | pigpen | we build everything from source (gentoo) |
| 21:25.21 | s14ck | yes, me too on debian |
| 21:25.22 | jaytee | but I'm pretty sure you'll need to recompile and reinstall * afterwards |
| 21:26.00 | pigpen | oh, yeah. |
| 21:26.19 | s14ck | but the drivers load ok |
| 21:26.26 | pigpen | actually I'll have my business partner do it. He is a kernel dev for gentoo, along with strongswan and others. |
| 21:26.40 | pigpen | one spooky bastard. |
| 21:27.02 | s14ck | *CLI> module load chan_dahdi.so |
| 21:27.10 | s14ck | <PROTECTED> |
| 21:27.13 | s14ck | <PROTECTED> |
| 21:27.51 | s14ck | and when i do *CLI> module reload chan_dahdi.so |
| 21:28.00 | s14ck | nothing happend |
| 21:29.06 | hardwire | there needs to be some sort of "Original-Destination" standard. |
| 21:29.21 | s14ck | i readed about the --prefix=/usr parameter |
| 21:30.23 | s14ck | # file /usr/lib/libopenr2.so |
| 21:30.48 | s14ck | \/usr/lib/libopenr2.so: symbolic link to `libopenr2.so.1.0.1' |
| 21:31.17 | s14ck | some idea? |
| 21:33.27 | *** join/#asterisk gaetronik (n=gaetan@n07-036.lp.newplanet.cl) |
| 21:33.27 | moy | s14ck: ldd /usr/lib/asterisk/modules/chan_dahdi.so |
| 21:33.39 | moy | read the guide in google code, and follow the steps there |
| 21:34.00 | moy | if the output of ldd does not show libopenr2.so then you did not installed asterisk/openr2 correctly |
| 21:34.10 | s14ck | moy: yes, i do step by step the guide |
| 21:35.34 | s14ck | # ldd /usr/lib/asterisk/modules/chan_dahdi.so |
| 21:36.43 | gaetronik | Hi there |
| 21:36.43 | s14ck | linux-gate.so.1 => (0xb7fd3000) libtonezone.so.2.0 => /usr/lib/libtonezone.so.2.0 (0xb7f56000) libopenr2.so.1 => /usr/lib/libopenr2.so.1 (0xb7f41000) libpthread.so.0 => /lib/i686/cmov/libpthread.so.0 (0xb7f27000) libc.so.6 => /lib/i686/cmov/libc.so.6 (0xb7dcc000) libm.so.6 => /lib/i686/cmov/libm.so.6 (0xb7da6000)lib/ld-linux.so.2 (0xb7fd4000) |
| 21:37.15 | s14ck | moy: libopenr2.so.1 => /usr/lib/libopenr2.so.1 |
| 21:37.23 | moy | enable debugging in /etc/asterisk/logger.conf and try to load the module, pastebin.com the output of module load chan_dahdi.so |
| 21:37.33 | s14ck | ok |
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| 21:40.36 | s14ck | moy: http://pastebin.com/m6b875bee |
| 21:41.45 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:42.21 | moy | you don't have dahdi devices, the guide you said you read clearly say you need to have dahdi devices, that is /dev/dahdi/1, /dev/dahdi/2 ... etc |
| 21:43.21 | *** join/#asterisk _theHub (n=_theHub@ool-ad02788c.dyn.optonline.net) |
| 21:43.47 | moy | also you have parameters commented with # and that is not accepted by Asterisk, accepted comments in chan_dahdi.conf start with semicolon ; |
| 21:44.04 | s14ck | moy http://pastebin.com/m696a9ac1 |
| 21:44.33 | gaetronik | hi |
| 21:44.45 | moy | s14ck: which branch is this? |
| 21:44.53 | gaetronik | is the atxfer ami action present in the 1.6.0 branch? |
| 21:45.43 | s14ck | moy digium TE121 |
| 21:47.00 | moy | ?? I asked which branch, where did you get the code from? |
| 21:47.58 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.66) |
| 21:50.01 | s14ck | jajaja |
| 21:50.38 | s14ck | moy: http://svn.digium.com/svn/asterisk/team/moy/mfcr2 |
| 21:51.19 | s14ck | moy: http://pastebin.com/m7f672d47 |
| 21:51.30 | moy | ok that branch is kind of deprecated, I need to update the guide, 2 or 3 days ago the branch was merged with asterisk trunk, that is http://svn.digium.com/svn/asterisk/trunk now has R2 code in there |
| 21:52.29 | s14ck | moy the same error |
| 21:52.50 | Qwell | moy: and 1.6.2! |
| 21:53.08 | s14ck | moy: http://pastebin.com/m293e4ef9 |
| 21:53.13 | moy | Qwell: oh right :) |
| 21:53.15 | Qwell | 1.6.2 was basically merged immediately after r2 went into trunk. |
| 21:53.22 | Qwell | erm, branched |
| 21:53.34 | moy | s14ck: try 1.6.2 |
| 21:53.48 | s14ck | moy: http://svn.digium.com/svn/asterisk/branches/1.6.2 |
| 21:53.58 | moy | yep |
| 21:54.01 | moy | in any case the problem seems to be chan_dahdi does not see the devices, try upgrading, if it does not work send an e-mail to asterisk-r2 mailing list with your findings, I will check it as soon as I can |
| 21:54.02 | s14ck | moy i did it, and the same error |
| 21:57.53 | *** join/#asterisk techie (n=root@164.sub-75-212-249.myvzw.com) |
| 21:57.57 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 21:58.03 | moy | s14ck: did you fixed the comment issues btw? |
| 21:58.34 | s14ck | moy: yes, i do |
| 22:00.30 | moy | second thought, it seems it finds the device, open()n succeeds, but DAHDI_SPECIFY fails |
| 22:02.21 | moy | Qwell: any idea of why that could happen? open() on /dev/dahdi/channel goes well but DAHDI_SPECIFY fails with "No such device or address", even though ls -la /dev/dahdi shows device 1 there |
| 22:03.24 | moy | s14ck: what dahdi version u have? |
| 22:05.54 | s14ck | moy: http://pastebin.com/m6b6f041c |
| 22:06.22 | s14ck | # /etc/init.d/dahdi stop Unloading DAHDI hardware modules: ERROR: Module dahdi is in use |
| 22:07.44 | moy | s14ck: try using a dahdi release and make sure you restart the dahdi service (which I expect to unload and load the kernel modules) when asterisk is stopped, otherwise the old modules are still in use |
| 22:08.39 | s14ck | ok |
| 22:08.41 | s14ck | i will try it |
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| 22:13.10 | *** join/#asterisk gsiener (n=gsiener@206.48.2.97) |
| 22:14.53 | gsiener | Hi all. I'm using Voicepulse with Asterisk 1.4.21-2. I'm recently having an issue where I'll often dial and the call won't go through. There's just a long pause after "-- Called HoD95nTb93/17323395100" and eventually nothing. Sometimes I have to redial 3 or 4 times just to get through. I have no idea where to start troubleshooting, any thoughts? |
| 22:15.24 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 22:16.42 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:21.40 | *** part/#asterisk roy_hobbs (n=roy_hobb@pool-96-242-209-249.nwrknj.fios.verizon.net) |
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| 23:05.17 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:10.22 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-34.dyn.centurytel.net) |
| 23:10.25 | jeff_phillips | hello |
| 23:18.43 | jeff_phillips | we've got a DSL line that seems to ... well it's hard to describe but some days it's rock solid, other days it goes a lot slower and is kinda iffy |
| 23:18.58 | jeff_phillips | I also have a server at a dedicated hosting facility with tons of bandwidth. |
| 23:20.02 | jeff_phillips | What I'm wondering is if I can have a DID # ring into the asterisk box that's behind the DSL line for at least the first couple of calls, and any remaining calls beyond what the DSL line seems to be able to handle that day I want to have answered on a hold queue on the dedicated server and passed through when there is less congestion on the DSL line... |
| 23:20.05 | jeff_phillips | is this feasible? |
| 23:23.48 | hardwire | So.. apparently.. when running a calling card business it's common practice to make sure the first call will be able to use 100% of it's minutes for a domestic call.. and after that first use you can just charge whatever? |
| 23:24.26 | hardwire | take it from 0.8/min to 1.2/min if you wanted to. no big deal. no legal hairy funky? |
| 23:24.26 | jeff_phillips | hardwire: Whaaaa?? if I bought such a calling card I'd be ticked |
| 23:24.38 | hardwire | jeff_phillips: of course.. that's how I'd feel too. |
| 23:24.52 | hardwire | I'm just trying to figure out more information on how calling card ethics work. |
| 23:24.58 | hardwire | personally I'd only sell clean cards. |
| 23:25.15 | jeff_phillips | Back in the day I'd use the Sam's Club version of the AT&T calling cards a lot |
| 23:25.16 | hardwire | but that's because I like to think before I spend money.. that's not our target audience apparently. |
| 23:25.24 | jeff_phillips | the ones that were like 3.9 or 4.3 cents a minute |
| 23:25.28 | hardwire | me as well.. I knew there were connect charges, etc.. |
| 23:25.35 | hardwire | but they weren't horrible. |
| 23:25.41 | jeff_phillips | what ticked me off was one day the charge for calling from a pay phone suddenly jumped from being "35 cents" to all the sudden "35 UNITS" |
| 23:25.51 | hardwire | heh |
| 23:25.57 | hardwire | 25 * 3.5 |
| 23:26.01 | hardwire | err |
| 23:26.05 | hardwire | 35 * 3.9 |
| 23:26.05 | hardwire | yeh |
| 23:26.08 | jeff_phillips | You realize 35 UNITS at 4.9 cents a minute or whatever they had hiked the rate to at the time ... |
| 23:26.21 | jeff_phillips | It was like $1.70 just to dial the # |
| 23:26.22 | hardwire | well.. I'm a bit miffed |
| 23:26.31 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 23:26.36 | jeff_phillips | and this is what really ticked me off... You can press *** to terminate the call and go back to the menu to make another call without hanging up |
| 23:26.42 | hardwire | I need to program in all sorts of logic to handle all calls AFTER the first call differently |
| 23:26.52 | jeff_phillips | Well they'd charge the $1.70 AGAIN for having called from a pay phone! But it was the same originating call!!! |
| 23:27.09 | hardwire | and make sure that when a card is sold it uses a specific static rate deck.. and then use a completely different rate deck after it's first use. |
| 23:27.35 | hardwire | Thankfully we aren't selling in "minutes" |
| 23:27.38 | jeff_phillips | so in other words you want to scam people |
| 23:27.52 | hardwire | except he took the liberty to say "calling cambodia? this card can reach 22 minutes there" |
| 23:28.02 | hardwire | bah |
| 23:28.08 | hardwire | I'd love to have a legit system |
| 23:28.25 | hardwire | I'm still trying to get this guy to "make sense" of his rate decks he magically finagled together. |
| 23:28.38 | jeff_phillips | Well I'm working on setting up a calling card system... except I'm going to be such a cheap bastard I won't even provide a toll free # as the access number |
| 23:28.46 | hardwire | heh |
| 23:28.50 | hardwire | what backend software? |
| 23:29.16 | hardwire | I'm using a2billing at the moment.. there hasn't been a commit to trunk since 2008 sometime. |
| 23:29.19 | jeff_phillips | I'm evaluating stuff right now -- the idea came to me last week |
| 23:29.27 | jeff_phillips | how is a2billing workin out for ya? |
| 23:29.32 | hardwire | meh |
| 23:29.36 | hardwire | it could be less frustrating. |
| 23:29.47 | hardwire | I'm not a huge fan of how the db's are arranged. |
| 23:30.00 | hardwire | it makes it nearly impossible to make dynamic rate decks |
| 23:30.03 | jeff_phillips | I am using over a dozen different SIP termination providers that all have different rates that keep changing to different countries, some with "expiring" pools of minutes |
| 23:30.20 | jeff_phillips | like one gives me x number of minutes per week at one rate on x number of channels after which it switches to a different rate |
| 23:30.37 | hardwire | the problem there is garanteing the first call on the card reaches the total time advertised |
| 23:30.37 | jeff_phillips | I'm trying to find a program that would make it easy to download in all the rate tables and let it figure out the least cost routing |
| 23:30.55 | hardwire | so if I buy a $5 card from you and it says I have 1000 minutes on the iVR.. I get to use that all up. |
| 23:31.07 | hardwire | so you need to compute your disconnect charge and connect charge into the initial amount. |
| 23:31.18 | jeff_phillips | isn't that the whole idea? |
| 23:31.19 | hardwire | also |
| 23:31.20 | hardwire | wtf |
| 23:31.24 | hardwire | disconnect charge? |
| 23:31.27 | hardwire | I hate this line of work. |
| 23:31.33 | jeff_phillips | what the frig a disconnect charge?! |
| 23:31.43 | hardwire | it's a charge that you get when you hang up the phone :0 |
| 23:31.50 | jeff_phillips | Oh for godsakes |
| 23:31.52 | hardwire | jeff_phillips: you get charged for hanging up.. basically. |
| 23:32.10 | hardwire | but it's also a good method of charging the card AFTER the first call.. and for every call thereafter |
| 23:32.33 | jeff_phillips | I'm going to keep it simple. a unit is a unit is a unit. Some destinations cost 1 unit per minute, others 2 units per minute, and so on. No connect or disconnect fees |
| 23:33.23 | jeff_phillips | I would like to add voice activated dialing with a speed dial list... not sure how to do that yet |
| 23:33.29 | hardwire | apparently you can't make money that way.. |
| 23:33.36 | hardwire | I'm interested to prove otherwise |
| 23:33.43 | jeff_phillips | How can you not make money that way? |
| 23:33.44 | hardwire | higher cents/minute vs funky charges. |
| 23:33.54 | hardwire | well you have to make and distribute the cards if they are physical |
| 23:33.59 | hardwire | then of course your time is added in |
| 23:34.05 | hardwire | jeff_phillips: lets put it this way |
| 23:34.09 | jeff_phillips | I can call england for 9 tenths of a cent and charge people 10 cents a minute to call from a US cell phone which direct dialing costs 26 cents with a $4 monthly fee, or $1.69 without the monthly fee |
| 23:34.18 | hardwire | I have a server in cali that could terminate a million minutes a month if I just had the audience to use it. |
| 23:34.35 | jeff_phillips | oh see, I'm just e-mailing people their pin # and dialing instructions. If they want the "card" they can print it out themselves |
| 23:34.43 | hardwire | yeh |
| 23:34.54 | bougyman | hardwire: i'll take 600,000 of them. |
| 23:35.02 | hardwire | bougyman: whats your anti? |
| 23:35.04 | bougyman | will you do $0.008 flat? |
| 23:35.06 | hardwire | pulls out an ace. |
| 23:41.47 | deadpigeon | gah |
| 23:42.04 | deadpigeon | most long distance carrier's require a million minutes a month just to do business. |
| 23:42.14 | deadpigeon | and thats on the clec level. |
| 23:51.53 | *** part/#asterisk mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 23:53.42 | Kyosh | then comes the headaches of the fusf |
| 23:56.15 | jeff_phillips | oh god, I don't even want to think of the FUSF tonight |
| 23:58.44 | *** join/#asterisk infinity1 (n=brendon@li6-32.members.linode.com) |
| 23:59.01 | infinity1 | whats the list of recomendded sip providers? |
| 23:59.07 | infinity1 | i know the bot has one. how do i get it? |
