| 00:00.11 | R-MAN | this is going to sound real stupid but my putty isnt copying lol |
| 00:00.16 | R-MAN | Control and INS isnt it |
| 00:00.20 | Bryanstein | anyone here willing to make me a test user so I can test my phone registration..something just isn't going right over here? |
| 00:00.29 | drmessano | ok |
| 00:04.46 | R-MAN | http://www.pastebin.ca/945530 |
| 00:04.48 | R-MAN | sorry I got it |
| 00:05.22 | drmessano | externip |
| 00:05.23 | *** join/#asterisk cardiff (n=island@76-10-147-182.dsl.teksavvy.com) |
| 00:05.27 | drmessano | not externalip |
| 00:05.35 | drmessano | and you can take the entry out of the hosts file.. not needed |
| 00:05.58 | R-MAN | ok |
| 00:06.05 | R-MAN | but still doesnt work |
| 00:06.05 | EvilDeshi | hmm i get this when trying to load the module now [Mar 16 19:01:41] WARNING[30567]: loader.c:376 load_dynamic_module: Module 'app_flite' did not register itself during load |
| 00:06.18 | drmessano | You just changed externalip to externip and reloaded? |
| 00:06.39 | R-MAN | nope ill try that now |
| 00:06.49 | R-MAN | in the sip_nat right? |
| 00:07.05 | drmessano | what you just pasted is FAIL.. there is no "EXTERNALIP" |
| 00:07.08 | drmessano | Its EXTERNIP |
| 00:07.29 | R-MAN | ok let me change it |
| 00:08.35 | R-MAN | WORKS! |
| 00:08.39 | R-MAN | thank you sir |
| 00:08.44 | R-MAN | as easy as that |
| 00:08.44 | drmessano | yw |
| 00:09.01 | *** join/#asterisk marl (n=marl@84.13.17.134) |
| 00:09.02 | R-MAN | and you said no need for the entry in hosts right? |
| 00:09.07 | drmessano | Nope |
| 00:09.10 | drmessano | No need for it |
| 00:09.14 | R-MAN | k ill take that out |
| 00:09.23 | R-MAN | thank you sir much appreciated |
| 00:09.27 | drmessano | anytime |
| 00:09.49 | R-MAN | and thank you guys for letting me interupt the asterisk channel with freepbx stuff much appeciated.. |
| 00:09.52 | Bryanstein | anyone in here using the old pingtel phones...if so please let me know |
| 00:09.56 | drmessano | lol |
| 00:10.21 | drmessano | Well, it wasnt a freepbx problem.. so whatever... I'm not a GUI nazi like some |
| 00:10.35 | *** join/#asterisk CCFL_Man2 (i=efa37c07@pool-70-105-204-253.scr.east.verizon.net) |
| 00:10.37 | Bryanstein | need help with the pingtels...looking bad over here |
| 00:11.02 | drmessano | Now to go find some dinner.. brb |
| 00:11.29 | CCFL_Man2 | are there any T1 pci cards that use 5v pci? |
| 00:11.39 | JT | yes |
| 00:12.12 | CCFL_Man2 | my sun netra has a 5v pci slot |
| 00:12.20 | CCFL_Man2 | digium clones? |
| 00:12.59 | JT | i thought digium made some |
| 00:13.11 | JT | but if not, sangoma cards don't care if it's 5v or 3.3v pci |
| 00:13.38 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 00:13.48 | CCFL_Man2 | i was looking at a te210p |
| 00:14.10 | CCFL_Man2 | but that card will fry |
| 00:14.18 | FuriousGeorge | should i upgrade to 1.4.19-rc2 or 1.4.18 for a production server? |
| 00:15.41 | marl | hi folks, i have just installed the latest svn of * and astgui, but any urls i try to access on port 8088, i get url not found on this server from asterisk, anyone any pointers on were to start looking for the problem? |
| 00:17.28 | Bryanstein | FuriousGeorge, I don't know much about asterisk, but you never upgrade a production box without reason to. Is there a bug that you are fixing or a feature you need in rc2? If not I wouldn't touch it. |
| 00:18.06 | FuriousGeorge | Bryanstein: well there's the issue with inbound calls getting 'hung' and needing a softhangup |
| 00:18.23 | FuriousGeorge | else, serve becomes inoperable |
| 00:18.32 | drmessano | What version are you on? |
| 00:18.38 | *** join/#asterisk cardiff (n=island@76-10-147-182.dsl.teksavvy.com) |
| 00:18.38 | FuriousGeorge | 1.4.17 |
| 00:19.49 | _ShrikE | FuriousGeorge: Read the changelogs to see what version fixes your issue.. |
| 00:20.15 | drmessano | I believe 1.4.18 fixed some of that |
| 00:21.04 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:21.11 | FuriousGeorge | _ShrikE: ive always had this issue, i dont believe any will :) , but since im determined to find a solution, rather than just restart asterisk nightly, upgrading to the correct version is important when filing bug reports |
| 00:21.31 | FuriousGeorge | s/'find a solution'/'identify the problem' |
| 00:22.03 | FuriousGeorge | drmessano: that's the one ill try then |
| 00:29.27 | *** join/#asterisk PepOSX (n=angeldav@190.72.149.252) |
| 00:41.32 | billytwowilly | hey, anyone familiar with how complete the qsig support is in asterisk? |
| 00:42.47 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4ce7b7635df60659) |
| 00:51.29 | ectospasm | billytwowilly: I'm not 100% sure, but I think it's not 100% complete... |
| 00:51.50 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:51.50 | ectospasm | some of the devs might be able to answer better |
| 00:53.08 | ectospasm | I've got a call file, that I'm copying to /var/spool/asterisk/outgoing, and the call is not being initiated unless I stop asterisk, and restart... |
| 00:53.12 | billytwowilly | ectospasm: got any links to progress? I would eventually like to hook an asterisk box up to some proprietary trading phone hardware |
| 00:53.35 | ectospasm | billytwowilly: nope, sorry |
| 00:54.31 | billytwowilly | ectospasm: who is a devel, or is there a better place to ask to get this answered, or atleast a time frame on development? |
| 00:54.49 | ectospasm | billytwowilly: #asterisk-dev |
| 00:57.28 | billytwowilly | Ok, thanks. |
| 00:58.05 | file | there's basic qsig support but nobody is really working on expanding it... will it work with your stuff? maybe. |
| 01:02.37 | BobLutz | takes off his roller skates |
| 01:13.02 | *** join/#asterisk shido6 (n=shido6@74-130-50-233.dhcp.insightbb.com) |
| 01:23.55 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:32.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582425.dsl.bell.ca) |
| 01:32.42 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 01:32.51 | *** part/#asterisk a1fa (n=fiddy@unaffiliated/a1fa) |
| 01:44.44 | FuriousGeorge | what is the best of the hundred possible asterisk guis to allow my users to do some occasional 'soft hanging up' for themselves? |
| 01:47.52 | drmessano | Still having issues? |
| 01:53.15 | *** join/#asterisk hohum (n=dcorbe@68-26-29-54.area1.spcsdns.net) |
| 01:56.36 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 02:00.58 | FuriousGeorge | drmessano: time will tell, but i have a feeling i will |
| 02:02.57 | *** join/#asterisk shido6 (n=shido6@74-130-50-233.dhcp.insightbb.com) |
| 02:18.58 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8edd940535ed2480) |
| 02:21.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:26.33 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 02:28.07 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 02:33.58 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 02:38.41 | Yourname` | Hi. http://pastebin.ca/945662 -> That error is being sent back by Asterisk.. what did I do wrong again? :() |
| 02:38.59 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 02:39.31 | fujin | What does asterisk say? |
| 02:39.39 | fujin | the node at 65..91 |
| 02:39.53 | Yourname` | hey fujin |
| 02:39.55 | Yourname` | Nothing at all |
| 02:40.05 | tuxd00d | Asterisk says: The weasels have eaten our phone system |
| 02:40.06 | Yourname` | There was a call in progress on 65..91 |
| 02:51.33 | Yourname` | Is there a way to Pause an agent in AgentLogin()? |
| 02:51.48 | fujin | use aqm/rqm/pqm/upqm |
| 02:51.48 | fujin | ;> |
| 02:52.13 | Yourname` | With lazy agents that we have, AQM is a distant reality. :( |
| 02:52.22 | fujin | beat them more |
| 02:52.33 | Yourname` | We fire them! |
| 02:52.46 | fujin | ugh, I'v ebeen battling with QueueMetrics and the *QM stuff so that the agents can use the clicky buttons to login/logout/pause/unpause |
| 02:53.43 | Yourname` | Which reminds me, when a caller goes through the IVR and before there is a handoff of the call from one Asterisk to the Asterisk where there are queue members logged on (using sip extensions like 54321@IP), if the caller hangs up.. why does the call sometimes still make it to the agent and come up as deadair? |
| 02:53.58 | fujin | AgentLogin is awesome like that |
| 02:54.18 | Yourname` | Is it the caller hanging up enroute to the Agent, or does someone in here think it could be the handoff from one Asterisk outbound dialer to another Asterisk inbound queue? |
| 02:54.27 | Yourname` | fujin: Has been happening since the days of AQM man :( |
| 03:03.20 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.77.37) |
| 03:04.18 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 03:07.43 | FuriousGeorge | ~seen russellb |
| 03:07.46 | jbot | FuriousGeorge: i haven't seen 'russellb' |
| 03:07.55 | FuriousGeorge | ~seen russelb |
| 03:07.56 | jbot | FuriousGeorge: i haven't seen 'russelb' |
| 03:08.21 | jer | !seen russellb |
| 03:08.27 | jer | err |
| 03:08.40 | jer | oh wait, you tried that already |
| 03:08.40 | Yourname` | jbot: Have you seen russellb? |
| 03:08.44 | FuriousGeorge | ~seen russellb |
| 03:08.45 | jbot | i haven't seen 'russellb', FuriousGeorge |
| 03:08.53 | Yourname` | Please answer me. It's a question of self-esteem now. |
| 03:08.58 | jer | he last spoke at 1:15 pm EDT |
| 03:09.53 | plik | EDT? |
| 03:10.04 | jer | eastern daylight time (UTC-0500) |
| 03:10.05 | plik | already? |
| 03:10.15 | jer | nod |
| 03:10.27 | plik | did that change this week? |
| 03:10.32 | jer | last week |
| 03:10.36 | FuriousGeorge | jer: i've been away for a while, what happened to the other half of your nick |
| 03:10.37 | plik | blimey |
| 03:10.48 | jer | FuriousGeorge, i'm not JerJer, different person |
| 03:11.10 | FuriousGeorge | i think i'll call you 'slim' |
| 03:11.14 | FuriousGeorge | j/k |
| 03:11.27 | jer | call me whatever you want, just not after 11pm =] |
| 03:12.14 | FuriousGeorge | shirley, you're joking :) |
| 03:12.34 | drmessano | FAIL |
| 03:12.46 | drmessano | Shirley you can't be serious |
| 03:12.50 | FuriousGeorge | lol |
| 03:12.54 | drmessano | I am serious, and don't call me Shirley |
| 03:13.02 | jer | =] |
| 03:13.04 | drmessano | At least google before you quote |
| 03:13.14 | FuriousGeorge | airplane_test('shirley, you're joking') returns false |
| 03:18.43 | Yourname` | No but seriously, how can an agent make an outbound call while logged in using AgentLogin()? |
| 03:21.40 | fujin | I don't think it's possible; |
| 03:21.44 | fujin | Pick up another channel? |
| 03:22.50 | Yourname` | I guess so |
| 03:22.58 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 03:36.08 | *** join/#asterisk shital (n=shital@122.167.100.109) |
| 03:36.29 | shital | Good Morning ALL |
| 03:37.00 | cmantito | loh |
| 03:37.30 | shital | i wanted to Buy Digium card, just to make calls, and receive calls |
| 03:37.44 | shital | which model no i should go for |
| 03:38.05 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582453.dsl.bell.ca) |
| 03:38.18 | shital | plz any body can help me? |
| 03:42.25 | cmantito | a single port card would probably do? |
| 03:43.23 | x86 | shital: POTS? TDM? |
| 03:43.27 | x86 | SIP? |
| 03:45.48 | shital | i just wanted to use analog phones for extensions and one incoming calls |
| 03:47.14 | drmessano | Hmmm |
| 03:48.42 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 03:50.24 | *** part/#asterisk sacitec (n=tobi@201.144.211.82) |
| 03:53.31 | drmessano | Hmmm |
| 03:53.45 | drmessano | I need to figure out, if possible, how to initiate a 3-way call with Chan_mobile |
| 04:00.31 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:00.44 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:12.50 | Mavvie | silly asterisk box just hung itself up. |
| 04:13.01 | Mavvie | properly |
| 04:14.03 | drmessano | hmm |
| 04:18.23 | fujin | drmessano: dump them into app_conference or app_meetme? |
| 04:18.46 | drmessano | No |
| 04:19.01 | drmessano | I want to be able to able to call in, and then bridge the call back out |
| 04:19.12 | x86 | DISA |
| 04:19.28 | drmessano | yeah.. great idea.. Can't without 3-way |
| 04:19.35 | drmessano | If 3-way worked I could do 100 things |
| 04:19.52 | Mavvie | drmessano: use an AGI script which dials out for you? |
| 04:19.59 | drmessano | LOL |
| 04:20.05 | drmessano | You don't get it |
| 04:20.07 | fujin | drmessano: Use a macro and features.conf |
| 04:20.13 | drmessano | I call my cell phone |
| 04:20.22 | drmessano | It rings into * |
| 04:20.33 | drmessano | I want Asterisk to direct call back out via the same phone |
| 04:20.37 | drmessano | Which requires 3 way |
| 04:20.50 | Mavvie | to call back out via the same phone? |
| 04:20.54 | drmessano | yes |
| 04:21.04 | fujin | You're doing it wrong. |
| 04:21.10 | Mavvie | to call back out *to* the same phone you mean. |
| 04:21.14 | drmessano | No |
| 04:21.36 | fujin | Draw a picture |
| 04:21.41 | fujin | You're not very good at explaining what you want. |
| 04:21.52 | drmessano | No, you're not good at understanding what I want |
| 04:21.57 | drmessano | I explained it well |
| 04:22.02 | drmessano | I call my cell phone |
| 04:22.11 | drmessano | Its connected with Chan_mobile to my * box |
| 04:22.32 | drmessano | * gets the call and says "Oh, this needs to go out via the cell phone since mobile <> mobile is free" |
| 04:22.35 | Mavvie | oh, it's our fault. |
| 04:22.38 | drmessano | So It directs the call back out |
| 04:22.42 | drmessano | Via the cell |
| 04:22.44 | drmessano | Very simple |
| 04:23.03 | drmessano | I could do it with 2 cell phones |
| 04:23.08 | drmessano | or 3-way on one |
| 04:23.18 | drmessano | 3-way would just bridge the call |
| 04:24.11 | fujin | So why not do 3way with app_conference or meetme? |
| 04:24.18 | fujin | (like any other *-connected channel) |
| 04:24.38 | Mavvie | I must be missing something. |
| 04:24.43 | drmessano | It doesnt matter HOW I do it.. I cannot make a call out AND a call in via the SAME phone unless I have 3 way |
| 04:25.04 | fujin | Stop doing it wrong |
| 04:25.18 | drmessano | Ok, what is the correct way then |
| 04:25.39 | fujin | rm -rf chan_mobile |
| 04:25.44 | drmessano | 1. The call MUST come in via my cell phone |
| 04:25.56 | drmessano | 2. It MUST go out via my cell phone |
| 04:25.57 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 04:26.04 | drmessano | No |
| 04:26.06 | fujin | uh, right |
| 04:26.08 | fujin | so use an AGI? |
| 04:26.12 | fujin | or a macro/feature |
| 04:26.20 | drmessano | Still won't do it.. |
| 04:26.41 | fujin | get a sip client on your cellphone |
| 04:27.07 | Mavvie | oh dear, $boss comes with the request again: |
| 04:27.09 | drmessano | I don't care what happens when the call gets to Asterisk.. if chan_mobile won't flash and an initiate a 3-way call, it won't work |
| 04:27.16 | drmessano | Which was my original statement |
| 04:27.24 | fujin | Mavvie: lol :) |
| 04:27.25 | Mavvie | in case an asterisk server hangs, all calls need to be moved transparently. |
| 04:27.41 | fujin | impossible without frontend proxying |
| 04:27.48 | Mavvie | puts it in the pile next to faster than light network path recovery. |
| 04:27.50 | cmantito | in which case the proxy could still fail lol |
| 04:28.09 | fujin | no, prox(ies) would be l4+ loadbalanced |
| 04:28.12 | Mavvie | fujin: we're mostly doing PSTN <-> SIP. |
| 04:28.12 | drmessano | SIP client on the cellphone would be a better answer.. but Blackberry's suck |
| 04:28.16 | fujin | Mavvie: PRI? |
| 04:28.19 | Mavvie | yes |
| 04:28.26 | fujin | some of the PRI termination gear has active-failover of channels |
| 04:28.28 | fujin | but you're talking fat $$ |
| 04:28.39 | fujin | drmessano: buy a WM6 phone then? |
| 04:28.42 | cmantito | I wonder what it would take to get AT&T to deliver calls to my cellphone number via SIP... |
| 04:28.51 | drmessano | I don't have a choice which phone I use |
| 04:28.58 | fujin | definitely doing it wrong |
| 04:29.08 | fujin | so who makes all the decicions at $workplace? |
| 04:29.30 | drmessano | We all use Blackberry devices |
| 04:29.32 | fujin | (and second of all, what kind of stupid prick boss suggests using a mobile phone in any kind of business-grade telephony?) |
| 04:29.35 | drmessano | It is what it is |
| 04:29.45 | *** join/#asterisk shital (n=shital@122.167.74.169) |
| 04:30.05 | fujin | ah, I think I'm starting to get the picture. You're a cheapass. |
| 04:30.12 | fujin | that was the free calling between mobiles thing |
| 04:30.15 | drmessano | How so? |
| 04:30.18 | drmessano | No |
| 04:30.19 | cmantito | thinks fujin is being subhelpful. |
| 04:30.23 | fujin | cares |
| 04:30.26 | fujin | I'm enlightening, get out |
| 04:30.30 | drmessano | No, hes being a prick |
| 04:30.32 | drmessano | As always |
| 04:30.38 | cmantito | mm. |
| 04:30.39 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 04:30.44 | cmantito | drmessano: what's your goal again? |
| 04:30.45 | fujin | So why would you want to call between mobiles if not to save money? |
| 04:30.50 | fujin | why not use an upstream SIP peer? |
| 04:30.56 | fujin | (or IAX2) |
| 04:31.14 | fujin | ~cheap |
| 04:31.14 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 04:31.14 | fujin | eof. |
| 04:31.29 | fujin | detaches |
| 04:31.31 | *** join/#asterisk Alpha_AI (n=Ben@iti142937-1.gw.connect.com.au) |
| 04:31.35 | Alpha_AI | Hello |
| 04:31.38 | cmantito | thank fuck for some small favours. |
| 04:31.45 | cmantito | lo Alpha_AI |
| 04:31.48 | drmessano | I have TWO cellphones.. personal and work.. Im not carrying two of them anymore... but I get a few calls on my personal phone.. I wanted to leave my personal phone here as a home phone, let all the calls go into asterisk.. If its during the day, have the calls redirect out to my work cell.. if at night, ring the house phones |
| 04:32.11 | cmantito | but it has to call out from your personal cell? |
| 04:32.31 | drmessano | If it doesnt, I burn minutes on my work phone for personal, and I cant do that |
| 04:32.36 | cmantito | right |
| 04:32.40 | cmantito | that makes sense. |
| 04:32.46 | Alpha_AI | i have registered two extensions on freepbx, and have to softphones working on my host pc. i need to add another pc on the otherside of the world as an extension. that person is using x-lite how do i do that? |
| 04:32.51 | cmantito | and what problem are you having, drmessano? |
| 04:32.51 | drmessano | If I use the same network (verizon), they dont care |
| 04:32.58 | cmantito | eww verizon |
| 04:33.00 | cmantito | but oh well lol |
| 04:33.09 | Alpha_AI | *two |
| 04:33.11 | drmessano | Chan_mobile just doesnt seems to work with 3-way calling |
| 04:33.21 | cmantito | ~freepbx |
| 04:33.21 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 04:33.25 | cmantito | Alpha_AI: ^^^^ |
| 04:33.34 | cmantito | drmessano: ah. that stinks. |
| 04:33.39 | cmantito | cause you've got a good idea. |
| 04:33.59 | cmantito | here's a thought, drmessano |
| 04:34.16 | drmessano | I could do it with two phones... but god, that seems like a BFD to get around carrying another phone |
| 04:34.29 | cmantito | get a DID, *72<DID> from the VZ phone, call comes in to ast, then divert it out the cell as necesasry |
| 04:35.15 | cmantito | so have verizon deliver the call via SIP (kinda sorta) and then send it by chan_mobile over VZ's cdma net to your work phone ;p |
| 04:35.47 | drmessano | Forward the call on the phone... which sends it to Asterisk.. then send it out via the idle phone |
| 04:35.49 | drmessano | Hmmm |
| 04:35.52 | cmantito | espcially cause you can get a cheap DID if you don't care where the arecode is and in this case, you probably wouldn't. |
| 04:36.28 | cmantito | it's rather obtuse |
| 04:36.32 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:36.37 | cmantito | but it ...sorta resembles my mess of call forwards I've got setup. |
| 04:36.39 | drmessano | Thats almost perfect... if nothing else, it does take the burden of it not being free minutes off the personal > work leg of the call |
| 04:36.44 | cmantito | yeah |
| 04:37.04 | cmantito | I mean, for around a dollar a month you can get a DID in the middle of fucking nowhere with like 5000 minutes |
| 04:37.57 | drmessano | I could get $4 a month DID from les.net for that matter |
| 04:38.09 | drmessano | Or even use an IPKALL number, which really hasnt failed me yet |
| 04:38.18 | cmantito | see I have this problem where my AT&T number is printed on my biz cards, but I don't have reception at my desk, so before Ig et to my desk, I forward my calls to my DID, but when I'm away from my desk, I forward calls from my desk phone to my cellphone, which means when I come or go I have to set/unset two forwards |
| 04:38.21 | cmantito | aghh! |
| 04:38.23 | cmantito | I did not hit bold. |
| 04:38.25 | cmantito | sorry. |
| 04:38.32 | drmessano | No bold |
| 04:38.37 | drmessano | It stripped it |
| 04:38.41 | cmantito | ah, +c stripped it |
| 04:38.45 | cmantito | thank fuck |
| 04:38.46 | cmantito | I ahte colours |
| 04:38.48 | drmessano | lol |
| 04:38.52 | cmantito | s/ahte/hate |
| 04:39.14 | drmessano | So how about this.. |
| 04:39.23 | drmessano | Well, nm |
| 04:39.33 | drmessano | No.. |
| 04:39.50 | drmessano | Cron job to create a call via that turns forwarding on |
| 04:39.53 | drmessano | and then back off |
| 04:39.57 | drmessano | call file* |
| 04:40.14 | cmantito | I wonder if I could do bluetooth presence detection |
| 04:40.34 | cmantito | and have it work like, when I get in range of my computer, an app on my PC unforwards my deskphone, and an app on my cellphone forwards my cellphone |
| 04:40.36 | cmantito | and vice versa |
| 04:40.43 | cmantito | I have a smartphone, but I'm not really skilled at Symbian coding. |
| 04:40.55 | drmessano | hmm |
| 04:41.12 | cmantito | I've thought about getting a second SIM, permaforward my calls to my DID on the SIM I have, USE the second SIM with another phone number, and just set/unset my IP phone's forward as I come and go lol |
| 04:41.23 | cmantito | but that means I'm shovelling more money at AT&T for the second "line" |
| 04:41.38 | drmessano | yeah |
| 04:41.50 | cmantito | for around a hundred bucks I could LNP my cell phone number to a SIP DID |
| 04:42.27 | cmantito | but I'm not a fan of that idea either |
| 04:42.51 | drmessano | Multiline cell phones with very obvious line distinction are the way to go |
| 04:42.58 | cmantito | well GSM supports it |
| 04:43.03 | drmessano | We have 4 button desk phones.. have for years.. why not Cell |
| 04:43.03 | cmantito | but AT&T refuses to implement it |
| 04:43.07 | drmessano | yeah |
| 04:43.32 | drmessano | The SIP client on the phone would be the killer app |
| 04:43.34 | cmantito | in fact, I can get a multisimcard adaptor and allow the use of multiple SIM cards as different lines on the phones |
| 04:43.39 | drmessano | But damn.. blackberry |
| 04:43.44 | cmantito | I have a SIP client on mine, but it's quite crap |
| 04:43.50 | cmantito | and I don't have 3G coverage in the US |
| 04:43.52 | cmantito | only outside of. |
| 04:44.03 | cmantito | (imported phone) |
| 04:44.06 | drmessano | I downloaded the Gizmo app... lol |
| 04:44.10 | cmantito | I use .. Fring |
| 04:44.14 | drmessano | Found it they bridge calls |
| 04:44.22 | drmessano | Not a "real" client |
| 04:44.27 | cmantito | heh |
| 04:45.03 | cmantito | I exist in two countries, the US and the UK. I have a Symbian smartphone which is 3G happy in the UK on my US contract |
| 04:45.14 | cmantito | or in the UK on my UK paygo sim (which I've had for like 3 years) |
| 04:45.18 | drmessano | ah |
| 04:45.22 | cmantito | but it won't do 3G in this region of the US |
| 04:45.28 | cmantito | because AT&T kinda skipped UMTS |
| 04:45.57 | cmantito | even though it wouldn't cost them much at all (relatively speaking) to implement HSDPA AND UMTS at the same time, since they're supposed to be backwards comaptible |
| 04:46.11 | cmantito | unless they're using non-standard stuff I believe it's just another thing they've gone and opted not to implement |
| 04:46.45 | Alpha_AI | how do i edit sip_nat.conf |
| 04:47.12 | drmessano | Well, one can only hope Google solves it for us all |
| 04:47.32 | cmantito | heh |
| 04:47.46 | cmantito | well, i'm just afraid that if they get this chunk of airspace they've been trying for |
| 04:47.52 | cmantito | they'll create yet another standard |
| 04:47.54 | cmantito | which won't be global |
| 04:48.00 | cmantito | and therefore doesn't help me. |
| 04:48.13 | cmantito | that's why I use GSM/LTE and not CDMA/EV-DO |
| 04:48.30 | cmantito | cause I'm a standards freak AND I travel a lot |
| 04:48.56 | drmessano | Maybe we'll luck out and they'll just supplement GSM |
| 04:49.16 | cmantito | I'm hoping so |
| 04:49.27 | cmantito | but not in the 700mhz band since that's only available here |
| 04:49.51 | cmantito | GSM is split up enough as it is |
| 04:49.52 | cmantito | ;p |
| 04:49.59 | drmessano | heh yeah |
| 04:52.04 | drmessano | HAHAH |
| 04:52.12 | drmessano | Gizmo5 supports Asterix |
| 04:52.25 | drmessano | no, no.. "asterix" |
| 04:52.35 | cmantito | huh? |
| 04:52.38 | cmantito | are you serious? |
| 04:52.41 | cmantito | "asterix" |
| 04:52.41 | cmantito | ? |
| 04:52.45 | drmessano | http://gizmo5.com/pc/network/ |
| 04:52.47 | drmessano | scroll |
| 04:53.09 | drmessano | Follow the link |
| 04:53.16 | drmessano | The headline is the same |
| 04:53.51 | cmantito | oh god |
| 04:53.57 | cmantito | that's terrible |
| 04:54.16 | drmessano | Oh wait til you see their asterix config.. |
| 04:54.36 | drmessano | It's a CF of "jam this setting into a peer and see if it starts working" |
| 04:54.41 | drmessano | I mean... bad |
| 04:54.57 | cmantito | where is thaT? |
| 04:54.59 | cmantito | *that? |
| 04:56.00 | cmantito | found it |
| 04:56.14 | cmantito | owchies |
| 04:56.42 | drmessano | yeah |
| 04:56.43 | cmantito | why on earth would GIZMO need insecure=very? |
| 04:57.09 | drmessano | I found that to be necessary, actually |
| 04:57.17 | cmantito | *why*? |
| 04:57.26 | cmantito | they can't send from the right damned ports? ;p |
| 04:57.51 | drmessano | the invites were hitting from some funky Ips |
| 04:57.52 | drmessano | IPs |
| 04:58.19 | drmessano | They are using Yate, apparently |
| 04:58.57 | cmantito | heh |
| 04:58.57 | cmantito | wow |
| 04:59.12 | cmantito | btw, do you know of a softphone that supports BLFs? |
| 04:59.15 | drmessano | If you check the forums, they REALLLY have no clue |
| 04:59.24 | drmessano | No, I sure don't |
| 04:59.36 | cmantito | damn, oh well lol |
| 04:59.53 | drmessano | jblack and I spent some time working out the asterisk config for them.. trying to decipher the various forum posts.. |
| 05:00.03 | drmessano | and even posts from their own staff were inconsistent |
| 05:00.21 | cmantito | heh |
| 05:00.21 | drmessano | It was.. humorous |
| 05:00.26 | cmantito | that's ridiculous |
| 05:00.54 | drmessano | fromuser, fromdomain, authuser, etc, etc, etc |
| 05:00.59 | drmessano | WHYYYY OH WHYYY |
| 05:01.04 | Mavvie | ENOCLUE. |
| 05:01.07 | Mavvie | happens everywhere. |
| 05:01.17 | drmessano | very true |
| 05:01.19 | [TK]D-Fender | cmantito, eyeBeam |
| 05:01.24 | Mavvie | company I work for has the same problem. |
| 05:01.28 | cmantito | [TK]D-Fender: tahnks :) |
| 05:01.45 | Mavvie | despite having very clear work instructions not to use mail.domain, one person keeps refering to it. |
| 05:02.14 | drmessano | lol |
| 05:02.47 | drmessano | Alright.. time for me to go sleepy night time.. |
| 05:02.48 | drmessano | l8r |
| 05:03.12 | cmantito | doesn't get to sleep for just under 24 hours |
| 05:03.18 | tengulre | where have very cheap embedded device for asterisk? |
| 05:03.54 | cmantito | tengulre: check voipdw.com |
| 05:03.56 | cmantito | I buy lots from them |
| 05:05.11 | tengulre | anybody here successfule running asterisk on Linksys? |
| 05:05.19 | tengulre | cmantito: OK ! tks! |
| 05:05.37 | tengulre | but that 's very hard to buy from china! |
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| 05:32.14 | eric2 | how do you remove an asterisk build/installation from source? |
| 05:32.37 | eric2 | originally I did ./configure, make, make install but now I want to remove everything |
| 05:33.49 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
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| 05:42.46 | efort | where can I get asterisk-extra-sounds in wav format? the link I have to the file at digium is a sgml file despite ending in .tar.gz |
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| 05:55.36 | *** join/#asterisk ScoobyDooby (n=Me@unaffiliated/andshaggytooby) |
| 05:55.43 | ScoobyDooby | wow a lot of people in here |
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| 05:56.32 | BBHoss_Laptop | ScoobyDooby, all dead, since the netsplit of 08 |
| 06:00.05 | ScoobyDooby | I have a super lame newbie question. Just to confirm before I buy a tdm400p clone |
| 06:00.24 | ScoobyDooby | BBHoss_Laptop: do you have a firm graps of FXO and FXS jargon |
| 06:00.40 | BBHoss_Laptop | ScoobyDooby, somewhat, go ahead and ask |
| 06:01.00 | ScoobyDooby | I need to verify for sure that I can buy a tdm400 and slap on a fxo module and make calls on my landline |
| 06:01.22 | ScoobyDooby | those ports on the card look a lot like rj45 ports |
| 06:05.49 | ScoobyDooby | I see 3 fxs and one fxo's and it just boggles the hell out of me... |
| 06:05.58 | ScoobyDooby | can anyone explain this for me really quick |
| 06:06.33 | *** join/#asterisk ceres (n=ceres@ariane.digicomp.ch) [NETSPLIT VICTIM] |
| 06:06.35 | ScoobyDooby | I almost under stand what each device does, but there is still a little grey area |
| 06:06.55 | BBHoss_Laptop | fxs is what you plug your standard phone into, fxo is where you plug the phone line in from the telephone company |
| 06:07.44 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) [NETSPLIT VICTIM] |
| 06:07.47 | BBHoss_Laptop | an fxo port will allow you to make calls, but you must have someway to make them, which is usually and IP phone or some sort of FXS device (usually PAP2 or SPA8000etc) |
| 06:07.55 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 06:07.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 06:08.41 | BBHoss_Laptop | ScoobyDooby, FXS ports are usually not worth the space inside the PC, especially when you can get them much cheaper by buying atas and running the calls over sip |
| 06:08.57 | BBHoss_Laptop | understand? |
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| 06:10.04 | cmantito | ..what's the thing in ast that requires a custom version of perl to be built? |
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| 06:11.07 | ScoobyDooby | ok BBHoss_Laptop, I thought fxs was for generating ring and blah blah blah...why do people buy them then ? |
| 06:11.17 | *** join/#asterisk mamacito62 (n=XD@190.72.235.68) |
| 06:11.31 | BBHoss_Laptop | ScoobyDooby, it DOES generate ring and all that jazz |
| 06:11.47 | BBHoss_Laptop | ScoobyDooby, but so does a PAP2 or an SPA8000 etc |
| 06:12.02 | BBHoss_Laptop | which are usually cheaper/port, at least with digium hardware |
| 06:12.11 | BBHoss_Laptop | dunno about clones |
| 06:12.15 | ScoobyDooby | the doc I read had me thinking that you had to hook a phone into fxs so if you had 3 fxs that is 3 phones ? |
| 06:13.29 | BBHoss_Laptop | well, you can split and FXS line and have 10 phones on one fxs port, but they will all ring at the same time. If you want individual extensions, you need separate ports for each extension, and in larger deployments, many people use channel banks |
| 06:13.29 | cmantito | yes, but a PAP2 is also an FXS. |
| 06:13.51 | ScoobyDooby | ahhhh |
| 06:14.10 | ScoobyDooby | thanks...clearing things up...I just needed to confirm I was somewhat correct |
| 06:14.24 | ScoobyDooby | so those ports on the cards are not rj45 but rj11 after all ? |
| 06:14.45 | ScoobyDooby | or are they both, lol can fit both ? |
| 06:14.47 | BBHoss_Laptop | they should just be regular rj11, do you have a link |
| 06:15.06 | ScoobyDooby | well yes I can produce a link ...1 sec |
| 06:15.46 | ScoobyDooby | http://www.alwaysintouch.com/product.php?productid=1&js=n |
| 06:15.55 | ScoobyDooby | just some random site |
| 06:17.02 | BBHoss_Laptop | looks like the digium card fits both |
| 06:17.12 | ScoobyDooby | hey I finally found a site with specs... |
| 06:17.21 | BBHoss_Laptop | especially apparent with this photo http://www2.kev009.com:8081/wp/wp-content/uploads/2007/03/digium-tdm400p.jpg |
| 06:17.25 | ScoobyDooby | most looks like it BBHoss_Laptop ...what cards do you use |
| 06:17.59 | BBHoss_Laptop | im and ip guy really, but i have heard good things about rhino cards + their support is apparently great |
| 06:18.17 | BBHoss_Laptop | but you shouldnt have any trouble from a digium card |
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| 07:59.24 | lnx | can you tell me please what kind of string is it? : 'M\242\364\267\200\210\365\267\240\253\352\267Y^D' i have found it in CLI histrory. |
| 08:00.13 | jql | that's interesting |
| 08:00.59 | lnx | yes it is |
| 08:01.26 | jql | the ^D is what kills it, really |
| 08:01.56 | jql | might've been worth testing for utf8ness, if but for that |
| 08:02.00 | lnx | it looks like an injection |
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| 09:22.43 | Kamilion | Howdy, I tried to get freepbx installed on my ubuntu 8 box, but I ran into some problems. I had installed asterisk and mysql/php5/apache2 and then installed freepbx according to the ubuntu server 7.04 instructions, asterisk *was* running, after a reboot it refuses to start and I can't seem to find anything glaringly wrong in the conf files or /var/log/asterisk/full. |
| 09:22.55 | Kamilion | trying to run asterisk -c, the last line of output is "Already have an application 'VMAuthenticate' ". |
| 09:23.06 | Kamilion | asterisk -dc gives me slightly more than I see in the full log, but I still don't see any reason for it to halt with no clear error message. |
| 09:23.16 | Kamilion | Anything glaringly obvious I'm missing? |
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| 09:29.00 | bjartis | hmm. I lost all sound in asterisk all of the sudden. No changes done. Restart didn't work. Asterisk 1.2.17... Don't see any special errors in the logs either.. where should I start looking? |
| 09:30.58 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:37.48 | bjartis | conferencing audio works.. but not SIP -> SIP |
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| 09:53.38 | zeeesh | i hv made 3 peers for inbound. working fine .. now i want to give option to my peers to call tranfer to each other .. how to do it ? |
| 09:56.09 | zeeesh | i mean where to do it .. i could not find any option in queues.conf ... |
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| 10:01.48 | zeeesh | how can peers call transfer to another peer when getting inbound calls ... ? |
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| 10:18.41 | krdian_ | hello |
| 10:18.58 | yang | I am wondering to enable TAPI access to manager.conf i have these lines |
| 10:18.59 | yang | read = system,call,log,verbose,command,agent,user |
| 10:18.59 | yang | write = call,log,verbose,command,agent,user |
| 10:19.20 | yang | can write permisions anyhow break sip.conf or extensions.conf ? |
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| 10:25.06 | defswork | 2.2 firmware for aastras is out! |
| 10:25.59 | JT | anything interesting? |
| 10:26.08 | defswork | hopefully fixes all my problems |
| 10:32.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:32.49 | defswork | http://www.trixbox.org/forums/vendor-moderated-forums/aastra-endpoints/aastra-releases-firmware-version-2-2-0 |
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| 10:59.44 | madduck | this is kinda annoying: softphones can't hang up calls anymore before they're answered |
| 10:59.50 | madduck | if they click on hangup |
| 10:59.57 | madduck | then the remote phone just keeps ringing |
| 11:00.31 | madduck | asterisk says to that on the console: |
| 11:00.31 | madduck | Mar 17 12:00:23 ERROR[14728]: chan_sip.c:11416 sipsock_read: We could NOT get the channel lock for SIP/piper-0816da90! |
| 11:00.35 | madduck | Mar 17 12:00:23 ERROR[14728]: chan_sip.c:11417 sipsock_read: SIP MESSAGE JUST IGNORED: ACK |
| 11:00.38 | madduck | Mar 17 12:00:23 ERROR[14728]: chan_sip.c:11418 sipsock_read: BAD! BAD! BAD! |
| 11:00.40 | bjartis | do you use x-lite? |
| 11:01.06 | madduck | me? as softphone? no; twinkle... |
| 11:04.03 | lirakis | is away (leaving..."the internets" are safe ... for now) |
| 11:05.05 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 11:06.13 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 11:07.08 | tzafrir | madduck, what version of Asterisk? |
| 11:07.22 | *** part/#asterisk bjartis (n=test@195.1.73.1) |
| 11:08.31 | madduck | tzafrir: Debian etch |
| 11:08.40 | madduck | 1.2.13~dfsg-2etch2 |
| 11:08.44 | madduck | it used to work though |
| 11:08.51 | madduck | this is new and there was no update of asterisk |
| 11:08.58 | madduck | i should try another softphone |
| 11:09.04 | madduck | if only there were usable ones... |
| 11:10.18 | madduck | installs 86Mb of ekiga+half of gnome. pleasure. |
| 11:11.06 | *** part/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 11:11.12 | tzafrir | Have you tried twinkle? |
| 11:11.35 | madduck | yeah, i am using twinkle |
| 11:11.39 | madduck | this is how i get the problems... |
| 11:11.49 | madduck | let's hope it's the client |
| 11:13.09 | *** join/#asterisk mattman99 (n=matt@ppp59-167-154-210.lns4.mel6.internode.on.net) |
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| 11:23.59 | zeeesh | s |
| 11:24.02 | mvanbaak | it's not my bday anymore ;) |
| 11:27.06 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 11:27.22 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
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| 11:33.36 | mace | pri related question - would anyone expect a four channel digium pri card configured with 3 euro isdn spans and one t1 span to only show up in asterisk as 3 euroisdn spans and the fourth missing? |
| 11:35.13 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:36.45 | DrAk0 | whats better, a fxs gateway for 24 ports or a channel bank? |
| 11:37.52 | madduck | tzafrir: ekiga has the same problem. |
| 11:39.41 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:40.50 | tzafrir | mace, how exactly have you configured it? |
| 11:43.40 | *** join/#asterisk bjartis (n=test@195.1.73.1) |
| 11:43.43 | mace | tzafrir: spans 1,2 and 3 are euroisdn. span 1 is unused, span 2 is network, span 3 is cpe, span 4 is missing (from pri show spans) |
| 11:44.00 | *** join/#asterisk whitehat (n=whitehat@unaffiliated/whitehat) |
| 11:44.11 | mace | zttool lists all four spans, with span one in red alarm state as expected, 2,3 and 4 OK |
| 11:45.25 | tzafrir | mace, I think you need to configure some channels from it |
| 11:45.43 | DrAk0 | anyone with a rhino channel bank? |
| 11:47.55 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:48.14 | mace | DrAk0: me, trying to make it work :) |
| 11:48.34 | mace | tzafrir: i've done that, to the best of my understanding |
| 11:50.13 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 11:52.02 | JT | DrAk0: usually a sip gateway is more economical and less complex |
| 11:52.24 | JT | mace: i assume you mean E1 ETSI |
| 11:53.12 | mace | JT: that's correct, yes |
| 11:53.24 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 11:53.41 | mace | ponders whether the cb24 requires a data channel; the configuration examples don't list one.. |
| 11:53.55 | JT | mace: not sure you can have t1 and e1 on the same digium cardf |
| 11:53.57 | JT | -f |
| 11:55.18 | mace | JT: the physical card seems to support it, and given digium tell you how to configure each port for t1 or e1 via kernel params, i imagine that again confirms the same |
| 11:55.36 | mace | whether asterisk makes assumptions though is another matter :/ |
| 11:55.48 | JT | so it doesn't have a t1/e1 jumper? |
| 11:56.03 | *** join/#asterisk LucasAznarDcap (n=vircuser@host122.190-138-72.telecom.net.ar) |
| 11:56.05 | JT | pastebin your zapata.conf and zaptel.conf |
| 11:56.14 | mace | it does, but didn't want to dig around in the server at the time |
| 11:56.16 | mace | will do |
| 11:56.19 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 11:57.54 | tzafrir | mace, also, please pastebin the output of: cat /proc/zaptel/* |
| 11:58.35 | mace | rgr |
| 11:59.45 | DrAk0 | JT, i need connect over 60 analog phones |
| 12:00.31 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:00.32 | JT | DrAk0: ok |
| 12:01.50 | DrAk0 | JT, my first though was gateway sips, but then lots of ppl says is really unstable and stuff |
| 12:02.18 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:02.20 | mace | jt/tzafrir: http://pastebin.com/d50ae0acd |
| 12:03.26 | tzafrir | The fourth span is not PRI |
| 12:03.46 | JT | DrAk0: who said that? |
| 12:04.36 | LucasAznarDcap | DrAkO this depends on the GW manufacturer |
| 12:04.40 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:04.50 | LucasAznarDcap | try Audiocodes rock solid |
| 12:04.52 | tzafrir | mace, so it won't show up in the 'pri show spans', I guess |
| 12:05.18 | JT | mace: you trying to connect fxs or fxo? |
| 12:05.19 | mace | tzafrir: could you expand on that? the rhino channel bank is connected via pri to port 4 on the digium card, and shows up as span 4? |
| 12:05.27 | mace | JT: fxs |
| 12:05.47 | JT | mace: you are using the wrong signalling then |
| 12:05.52 | tzafrir | mace, it's connected as CAS, right? |
| 12:06.08 | tzafrir | not as PRI |
| 12:06.31 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:06.32 | tzafrir | 24 channels with no channel dedicated to signalling |
| 12:06.44 | tzafrir | No separate D channel |
| 12:07.05 | mace | tzafrir: re separate D channel, i did think the lack of that was strange, but it's definitely not described in the manual |
| 12:07.26 | mace | tzafrir: not sure what you mean re CAS |
| 12:07.53 | tzafrir | mace, did the manual also mention it has to show on 'pri show spans' ? ;-) |
| 12:07.57 | mace | ah i see |
| 12:08.09 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:08.14 | tzafrir | I might be slightly mixing technical terms here, though, but this is definetly not PRI |
| 12:08.17 | mace | tzafrir: no, didn't say that, but i'm taking the lack of dialtone as an indicator that somethings wrong :) |
| 12:08.25 | tzafrir | mace, look at 'zap show status' |
| 12:08.37 | mace | T4XXP (PCI) Card 0 Span 4 OK 0 0 0 |
| 12:08.44 | mace | aye, it's showing up there fine |
| 12:08.49 | JT | mace: as opposed to PRI, Channel Associated Signalling (CAS) / Robbed Bit Signalling (RBS) |
| 12:08.58 | JT | mace: there is no D channel for connecting to a channel bank |
| 12:09.10 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 12:10.05 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:10.10 | JT | mace: it steals the eighth bit of every sixth frame for basic signalling purposes |
| 12:10.14 | mace | nods |
| 12:10.44 | mace | so, am i misconfiguring either zaptel or asterisk, from what you can see? |
| 12:10.54 | JT | yes |
| 12:11.03 | JT | you said they are fxs ports |
| 12:11.12 | JT | you must use fxo signalling |
| 12:11.22 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:12.50 | mace | ahh, basic 101 error |
| 12:13.00 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
| 12:14.20 | mace | will have to wait for the system to become idle before i can roll the changes out |
| 12:15.21 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:15.59 | mace | cheerz jt and tzafrir |
| 12:18.47 | hmmhesays | fun |
| 12:25.30 | *** join/#asterisk dworkin (n=asdf@87-194-169-5.bethere.co.uk) |
| 12:25.56 | JT | np |
| 12:26.57 | DrAk0 | LucasAznarDcap, you have tried them with 24 fxs? |
| 12:28.48 | *** join/#asterisk cardiff (n=island@76-10-147-182.dsl.teksavvy.com) |
| 12:29.29 | LucasAznarDcap | yes |
| 12:30.03 | LucasAznarDcap | you can use a Digium 2400 , cheaper but les solid |
| 12:30.39 | LucasAznarDcap | If you need a carrier class solution for 24fxo use a gateway |
| 12:30.43 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
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| 12:31.55 | *** part/#asterisk Llama (n=bogdan@mm-143-246-209-82.adsl.mgts.by) |
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| 12:32.53 | LucasAznarDcap | we have 500 ports in a 24 7 faciliti in production with Audiocodes GWs , and is rock solid, only dont use if you has outdoor cable out of a city |
| 12:34.40 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:35.29 | LucasAznarDcap | the ports burn before the line filters on a lightning |
| 12:36.21 | LucasAznarDcap | we pay this experience with a lot of our money :( |
| 12:36.21 | *** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
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| 12:40.44 | DrAk0 | LucasAznarDcap, no, its inside an officce |
| 12:41.01 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
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| 12:41.52 | LucasAznarDcap | inside an ofice are rock solid |
| 12:43.07 | LucasAznarDcap | DrAko with more work , time and configuration you can has a Digium 2400 24Fxs rock solid , and its cheaper |
| 12:43.37 | LucasAznarDcap | with more money and less time go to Audiocodes Gw |
| 12:44.43 | LucasAznarDcap | If you go with Digium chose a server motherboard, best memory, and if you can pci express board |
| 12:44.59 | LucasAznarDcap | avoid IRQ problems |
| 12:45.56 | BobLutz | Is there a developer present? |
| 12:46.11 | BobLutz | I have a style question in regards to a module I am working on. |
| 12:50.33 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:51.10 | shido6 | how do you strip trailing digits? |
| 12:51.22 | BobLutz | shido6: In regards to what? |
| 12:51.23 | shido6 | ${EXTEN:-4} for example does what? |
| 12:51.35 | shido6 | if i send 1234567# and i "get" the pound |
| 12:51.39 | shido6 | i want to STRIP off the pound |
| 12:51.40 | BobLutz | Takes 4 digits starting from on the right side |
| 12:51.48 | BobLutz | ${EXTEN:-1} |
| 12:51.54 | shido6 | ah! |
| 12:52.07 | BobLutz | shido6: I think thats it..Ive read how to do it but never actually done it |
| 12:52.25 | shido6 | thats not quite what i had in mind...... |
| 12:52.27 | shido6 | heheh |
| 12:52.36 | shido6 | 555# with -1 just gives me "#" |
| 12:52.43 | shido6 | i want to take off the "#" keeping 555 |
| 12:53.22 | BobLutz | shido6: Yea, I know what you mean..its slipping my mind right now (Monday morning) |
| 12:53.45 | shido6 | i know..... thats why i came in here :) |
| 12:53.54 | *** part/#asterisk CaT[t2] (n=cat@CPE-124-189-48-61.nsw.bigpond.net.au) |
| 12:55.21 | BobLutz | shido6: Try ${EXTEN:0:-1} |
| 12:56.01 | shido6 | strip lsd? |
| 12:56.15 | shido6 | page 283 |
| 12:56.15 | BobLutz | lsd? |
| 12:56.23 | BobLutz | left his book in the car |
| 12:56.23 | shido6 | least significant digit |
| 12:56.28 | shido6 | :) |
| 12:56.29 | shido6 | thanks |
| 12:56.38 | BobLutz | shido6: that worked? |
| 12:56.39 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:56.39 | shido6 | you just said the same thing... thanks boblutz! |
| 12:56.42 | shido6 | um...... |
| 12:56.44 | shido6 | trying now |
| 12:57.21 | shido6 | that worked ${EXTEN:0:-1} |
| 12:57.46 | shasta | hm, is o'reilly's book the most up-to-date asterisk documenatation available? |
| 12:58.09 | BobLutz | shasta: The /doc folder in Asterisk source probably is more up-to-date |
| 12:58.53 | *** join/#asterisk jbigB (n=JbigB@216.207.245.1) |
| 13:00.10 | shasta | BobLutz, well, doc/ doesn't contain MeetMe() reference for example :) |
| 13:00.40 | BobLutz | shasta: Then the book would be a good starting point, unless a google search finds you a recent blog posting |
| 13:01.09 | shasta | that's what I thought :) |
| 13:04.45 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 13:04.45 | yang | I am wondering can my manager.conf login spoil the sip.conf or extensions.conf ? if i have read and write permissions defined? |
| 13:08.47 | [TK]D-Fender | yang: "spoil" how? |
| 13:09.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.41 | tzafrir | shasta, core show application meetme |
| 13:10.18 | BobLutz | tzafrir: Are you a developer? |
| 13:11.30 | *** join/#asterisk bidossessi (n=bidosses@41.209.158.163) |
| 13:11.36 | bidossessi | hi folks |
| 13:12.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.40 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137) |
| 13:13.30 | bidossessi | i just installed asterisk-gui from svn, but i have a strange problem. the setup phase will not move past step 2. it keeps looping back to step one. i'm running opensuse 10.3 with / asterisk-1.4.17-3.17 |
| 13:14.22 | [TK]D-Fender | bidossessi: ... |
| 13:14.24 | [TK]D-Fender | ~asterisknow |
| 13:14.25 | jbot | it has been said that asterisknow is based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 13:15.32 | BobLutz | [TK]D-Fender: Do you develop modules? |
| 13:15.41 | [TK]D-Fender | BobLutz: Nope. |
| 13:16.59 | bidossessi | [TK]D-Fender, my bad |
| 13:17.52 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 13:18.09 | tzafrir | [TK]D-Fender, Last time I checked, "opensuse 10.3 with / asterisk-1.4.17-3.17" is not AsteriskNow |
| 13:18.13 | yang | [TK]D-Fender: spoil, becouse with the current TAPI setup every user in the firm has access to manager.conf login, I don't know if they could aswell mess the sip.conf and extensions.con with such permissions: read = system,call,log,verbose,command,agent,user write = call,log,verbose,command,agent,user |
| 13:18.39 | tzafrir | bidossessi, the relevant deserted channel for you may actually be #asterisk-gui |
| 13:18.59 | shasta | tzafrir, brilliant, thanks |
| 13:19.00 | [TK]D-Fender | tzafrir : My bad on that misread... should be : |
| 13:19.03 | tzafrir | Be sure to use the version from branches/asterisknow rather than from trunk |
| 13:19.06 | [TK]D-Fender | ~asteriskgui |
| 13:19.09 | [TK]D-Fender | ~asterisk-gui |
| 13:19.28 | [TK]D-Fender | Hrm... no bot trainings, but equally unsupported here... |
| 13:19.47 | *** join/#asterisk ManxPower (n=manxpowe@137.sub-75-201-8.myvzw.com) |
| 13:20.10 | bidossessi | i like the "deserted" addition. well, guess i'll try another gui, which one is accepted as up to asterisk-standard |
| 13:20.17 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 13:20.19 | bidossessi | if it's not offtopic to ask |
| 13:20.31 | [TK]D-Fender | yang: Yes, write priveledge is dangerous. Its unencrypted, and can mess up your running dialplan. Should directly put sip config at risk though. |
| 13:20.42 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 13:20.43 | ManxPower | bidossessi: There are none. |
| 13:20.44 | tzafrir | bidossessi, hmm... it is quite up to the Asterisk standard |
| 13:20.46 | tzafrir | ducks |
| 13:21.14 | [TK]D-Fender | bidossessi: Check out the WIKI for a list. FreePBX is the most common. There are 1-2 other free workable ones, and the rest cost.. |
| 13:21.39 | ManxPower | I never liked WIMPS interfaces. |
| 13:21.41 | tzafrir | asterisk-gui is the least intrusive of the GUIs |
| 13:22.06 | tzafrir | That's the one thing I like about it. |
| 13:23.03 | bidossessi | well, that settles it, then. thank yall for your precious time. |
| 13:23.10 | hmmhesays | can I do a regexp replace in the asterisk dialplan? |
| 13:23.23 | *** part/#asterisk bidossessi (n=bidosses@41.209.158.163) |
| 13:24.15 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 13:24.28 | ManxPower | hmmhesays: sort of. see channelvariables.txt |
| 13:25.40 | hmmhesays | I know you can do a match |
| 13:27.42 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
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| 13:33.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:40.31 | shido6 | if i set a variable does the channel keep that variable until its hung up? Im losing it when i jump to another context somehow |
| 13:40.45 | *** join/#asterisk rpyne (n=richard@69.77.169.14) |
| 13:41.01 | shido6 | or is there a variable set that i can reference like the voicemail user thats logged in |
| 13:42.06 | *** join/#asterisk SteveUK (n=djdraevy@194.70.46.110) |
| 13:43.41 | [TK]D-Fender | shido6: what do "voicemail user" , "logged in", and "channel variable" have to do with each other? |
| 13:43.48 | shido6 | i need variable inheritance? |
| 13:44.04 | coppice | i need a large inheritance |
| 13:44.07 | shido6 | if 1 user logs into voicemail i need to know who that is |
| 13:44.44 | coppice | with a big enough I inheritance I won't give a damn who logs in |
| 13:45.09 | glaz | shido6: ${CALLERIDNUM} |
| 13:45.28 | shido6 | i need that to bet set to the mailbox user |
| 13:45.46 | shido6 | so i used setcallerid(${EXTEN}) |
| 13:45.53 | rpyne | Have you ever had the issue in Asterisk 1.4 where it is not reloading when the reload or dialplan reload cmd is used. Sip reload works fine though |
| 13:47.18 | shido6 | for example... when I get 4 digits i pass it to realtime and pull the mailbox and mailorg/context then i setcallerid to the mailbox and i get to voicemailmain, when i go to advanced options to place a call to dial out to a distribution list i need the envelope information to have the callerid of the original sender |
| 13:48.04 | shido6 | as soon as i go out to the other context i loose all the variables i set |
| 13:48.27 | shido6 | like ${mailbox} which was set in a previous context |
| 13:48.40 | [TK]D-Fender | shido6: IF you want to know who's logged in, then you'll have to make your own login. Regular one won't do. Use VM Authenticate, and then call Voicemail without authing. |
| 13:49.03 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:49.09 | Katty | hai |
| 13:49.17 | shido6 | thanks. |
| 13:49.30 | _ShrikE | heya Katty |
| 13:50.28 | shido6 | i'll give "{$_mailbox} a try |
| 13:50.33 | shido6 | err |
| 13:50.38 | shido6 | ${_mailbox} |
| 13:51.14 | rpyne | Have you ever had the issue in Asterisk 1.4 where it is not reloading when the reload or dialplan reload cmd is used. Sip reload works fine though |
| 13:53.31 | [TK]D-Fender | Katty: Mew. |
| 13:54.16 | anonymouz666 | calls to Quebec. |
| 13:54.17 | hmmhesays | Yep |
| 13:54.21 | anonymouz666 | [TK]D-Fender pickup the phone |
| 13:54.55 | bjartis | I can't get this to work. Sound when dial from Sip to SIP client stopped working all of the sudden.. I've tried upgrading kernel. asterisk. zaptel.. no change. :( |
| 13:55.06 | bjartis | No sound when* |
| 13:56.39 | bjartis | When i dial into meetme i get music on hold... so is it perhaps a codecs issue of some sort? |
| 13:57.28 | LucasAznarDcap | rpyne are you using Unicall |
| 13:57.42 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 13:59.20 | asteriskmonkey | anyone seen odd behaviour in 1.4 where extensions/dialplan reload dosnt work? |
| 14:01.20 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
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| 14:05.44 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 14:06.14 | Katty | [TK]D-Fender: mew (= |
| 14:07.27 | *** part/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 14:08.38 | BobLutz | AST_DECLARE_APP_ARGS() is declared in app.h ? |
| 14:11.30 | *** join/#asterisk shtoom (n=godson@123.176.41.46) |
| 14:13.58 | shtoom | Hi I am using asterisk 1.4.18 while recording calls using MixMonitor the voices are overlapping i mean not in sync |
| 14:14.35 | shtoom | and I am getting this warning on the screen all the time rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 14:16.14 | *** join/#asterisk ProofPack (n=no@208.255.225.67) |
| 14:20.17 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:25.01 | flush | yo mayday |
| 14:25.08 | flush | all of sudden asterisk doesnt work anymore |
| 14:25.11 | flush | it says " Ring/Off-hook in strange state 6 on channel 1" |
| 14:25.14 | flush | what does it mean?? |
| 14:28.07 | *** join/#asterisk blaylock (n=blaylock@c-24-30-249-234.hsd1.va.comcast.net) |
| 14:28.54 | blaylock | anyone know how to route calls to another extension while listening to a voice mail greeting |
| 14:29.23 | blaylock | someone said * and 0 are the only numbers allowed but what would the actual dial syntax look like? |
| 14:30.26 | *** join/#asterisk steliosk (n=Stelios@athedsl-4396544.home.otenet.gr) |
| 14:30.33 | [TK]D-Fender | blaylock: "*" and "0". "*' will jump to the "a" exten in the current context, "o" will jump to the "o" exten if their box is defined to allow it. |
| 14:30.49 | blaylock | o? |
| 14:30.56 | blaylock | how would one dial o? |
| 14:31.09 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:31.13 | blaylock | oh never mind |
| 14:31.29 | blaylock | you meant 0 will jump to the o extension |
| 14:31.45 | blaylock | so like exten => o,1,Goto(whatever,1) |
| 14:31.46 | blaylock | like that |
| 14:31.55 | [TK]D-Fender | blaylock: pretty much |
| 14:31.56 | blaylock | and likewise for the 'a' one |
| 14:32.11 | [TK]D-Fender | yes, I mean "0" will jump to "o" (operator) |
| 14:32.23 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-166-219.dsl.irvnca.pacbell.net) |
| 14:32.48 | blaylock | so If i put that in the same context as the phone extensions are in it would work for all extensions and voicemail greetings? |
| 14:32.50 | [TK]D-Fender | blaylock: "*" is hard-coded. "0" has to be enabled in voicemail.conf either gloablly or per box. Read the samples for this. |
| 14:33.01 | blaylock | ahh |
| 14:33.46 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 14:33.57 | __freedom__lover | hi! |
| 14:34.07 | dworkin | what is the usual cause for one way breaks of silence during a sip phone call? (one party heard the other one ok, the other party had 2 second long silence periods possibly in regular intervals. abundant bandwidth.) |
| 14:34.19 | __freedom__lover | i'm having trouble with iax trunking mode on freebsd. can anyone help me? |
| 14:34.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:34.44 | Yourname`` | Sweet home alabama. |
| 14:36.57 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:36.57 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:37.08 | Yourname`` | Mornin' Mr. Madsen. |
| 14:43.29 | *** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net) |
| 14:46.04 | l2trace99 | anyone know if there is a performance improvement with using sound files in sln format over gsm format ? |
| 14:47.19 | __freedom__lover | l2trace99: consult the translation table using 'core show translation'. in there, you will see the cost os translations from and for codecs supported |
| 14:47.51 | *** part/#asterisk dwayne (n=dwayne@76.29.245.9) |
| 14:49.10 | Lsodi | MixMonitor on zap channels are out of sync? v1.4.18 according to following link it should be fixed? http://lists.digium.com/pipermail/asterisk-bugs/2008-February/011882.html |
| 14:50.20 | [TK]D-Fender | __freedom__lover: you need to be running a zaptel timing source for that. Doable for hardware timing sources, but not sure about ZTDUMMY support of FBSD |
| 14:51.01 | [TK]D-Fender | l2trace99: SLIN isn't any better really AFAIK. You should use the closest codec to the one you most commonly use at the device level. |
| 14:52.27 | l2trace99 | thanks i think i got what i need to know |
| 14:54.06 | __freedom__lover | [TK]D-Fender: i've installed zaptel with ztdummy and loaded it into kernel |
| 14:54.28 | __freedom__lover | but the asterisk returns me: Unable to support trunking on user 'julia' without zaptel timing |
| 14:54.54 | [TK]D-Fender | __freedom__lover: Ok, well you'll also need to recompile * once you're sure the module is running so that * gets compiled with zaptel support |
| 14:55.12 | __freedom__lover | the asterisk was built after zaptel.. |
| 14:55.25 | [TK]D-Fender | __freedom__lover: then something is wrong with your zaptel setup. |
| 14:55.33 | __freedom__lover | :( |
| 14:55.43 | [TK]D-Fender | __freedom__lover: try this : "module load chan_zap.so" |
| 14:55.52 | __freedom__lover | i've installed from freebsd ports |
| 14:56.01 | __freedom__lover | my zap modules is loaded.. |
| 14:56.20 | [TK]D-Fender | __freedom__lover: I'd double-check that driver load then. |
| 14:56.41 | lnx | [TK]D-Fender: can you tell me please what kind of string is it? : 'M\242\364\267\200\210\365\267\240\253\352\267Y^D' i have found it in CLI histrory. |
| 14:57.25 | ManxPower | __freedom__lover: We don't really support packages/ports here. Talk to the ports/packager or compile from source. |
| 14:57.27 | [TK]D-Fender | lnx: No idea, and please refrain for targeting people for questions like that out fo the blue... |
| 14:57.56 | ManxPower | __freedom__lover: there are two types of Zap modules, Zap kernel modules, and Asterisk channel modules. |
| 14:58.18 | lnx | [TK]D-Fender: i do, no answer :/ |
| 14:58.28 | lnx | [TK]D-Fender: but thank you |
| 14:58.33 | BobLutz | chuckles |
| 14:58.50 | __freedom__lover | [TK]D-Fender: ok friend... thanks for your help |
| 14:58.57 | __freedom__lover | i'll check again |
| 15:00.29 | flush | [TK]D-Fender |
| 15:00.38 | flush | man im lost |
| 15:00.57 | [TK]D-Fender | flush: Try to remember where you put yourself last.... |
| 15:01.13 | flush | i call bell canada, i say "remove the phone box from my account" they do so, now when a call comes in asterisk says Ring/Off-hook in strange state 6 on channel 1 and wont work anymore |
| 15:01.17 | flush | right |
| 15:01.27 | BobLutz | bkruse: Qwell: I am reading the coding guidelines on www.asteriskpbx.com , and the page tells me to use ast_app_separate_args() -- Is this deprecated? |
| 15:01.27 | mcfloppy | i have a server with a usb isdn card. i can see all calls on the s0 line in the /var/log/messages |
| 15:01.33 | flush | channel 1 is my fxo module |
| 15:01.36 | [TK]D-Fender | flush: "remove the phone box"? HUH? |
| 15:01.42 | flush | sorry for my english |
| 15:01.50 | flush | for messages |
| 15:02.04 | mcfloppy | can i simply run asterisk to log all calls? |
| 15:02.21 | flush | do you have any idea why it wont work anymore since they changed some options in my account |
| 15:02.38 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 15:02.39 | [TK]D-Fender | mcfloppy: the effective appearance of * to the user may be small, but you'll have to have * pass off the calls in that way. |
| 15:02.41 | *** join/#asterisk antonyo14 (n=amanager@ip-64-32-239-224.chi.megapath.net) |
| 15:02.50 | *** join/#asterisk doolph (n=doolph@201.218.103.171) |
| 15:02.53 | doolph | hello |
| 15:02.57 | [TK]D-Fender | flush: What hardware? |
| 15:03.04 | flush | tdm400p |
| 15:03.13 | antonyo14 | i am having issues in which the caller can not hear the callee..., can anyone help? |
| 15:03.17 | [TK]D-Fender | flush: Try rebotting your system |
| 15:03.24 | flush | i did so.. same thing |
| 15:03.31 | mcfloppy | i dont know how to start |
| 15:03.48 | flush | http://pastebin.com/d4960c611 |
| 15:03.55 | [TK]D-Fender | mcfloppy: Strat by installing * and reading the book. |
| 15:03.57 | [TK]D-Fender | ~book |
| 15:03.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:03.59 | flush | this is what happens when a call comes in, but it wont work |
| 15:04.00 | [TK]D-Fender | start* |
| 15:04.01 | ManxPower | antonyo14: You have audio issues with a TDM card |
| 15:04.03 | ManxPower | ? |
| 15:04.08 | flush | doesnt really play Background(prompt04) |
| 15:04.15 | flush | all it does is ring in the phone, but not here |
| 15:04.43 | [TK]D-Fender | flush: plug a normal phone into the line and test the call. |
| 15:04.45 | antonyo14 | ManxPower: is the issue with hardware or software? |
| 15:04.49 | mcfloppy | [TK]D-Fender i have the book, but there is nothing about conecing to the isdn card |
| 15:04.56 | flush | it works, i tried |
| 15:05.05 | flush | when its plugged in asterisk it says the error on channel 1 |
| 15:05.07 | bjartis | Can anyone help me on why i can connect sip calls, but get no sound when talking. Both ways? I didn't change anything. But over the weekend something broke... :| |
| 15:05.12 | ManxPower | antonyo14: You did not answer my question. |
| 15:05.13 | [TK]D-Fender | mcfloppy: Go google up instructions for setting up your card then. Check the WIKI as well. |
| 15:05.54 | flush | what do you think [TK]D-Fender.. its their fault ? |
| 15:06.08 | [TK]D-Fender | flush: No idea. Try your server on another line |
| 15:06.09 | mcfloppy | ok |
| 15:06.32 | flush | right |
| 15:06.33 | flush | ill try |
| 15:06.51 | antonyo14 | Manx: I thought you were telling me that.. sorry. I have not had any issues with audio in the past, the problem is both on local and outside calls |
| 15:07.19 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:08.00 | mcfloppy | ~wiki |
| 15:08.06 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:08.08 | antonyo14 | ManxPower: what is weird is that if I call an extension and dont pickup I can hear voicemail prompts fine |
| 15:08.31 | mcfloppy | where is the asterisk wiki? i dont found |
| 15:08.42 | mcfloppy | ah okay |
| 15:08.45 | mcfloppy | i have it |
| 15:10.39 | mvanbaak | ~wiki |
| 15:10.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:10.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:10.54 | [TK]D-Fender | antonyo14: Sounds like a networking issue (typically NAT related) |
| 15:10.59 | [TK]D-Fender | ~wikis |
| 15:11.00 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:11.01 | [TK]D-Fender | ^^^ |
| 15:11.04 | mvanbaak | thanks |
| 15:11.11 | mvanbaak | I got a nice help message ;) |
| 15:11.20 | mcfloppy | i give up....... to much stuff for that little idea... |
| 15:12.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:12.59 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 15:13.36 | antonyo14 | [TK]D-Fender: thanks |
| 15:15.07 | *** join/#asterisk bantu (n=Miranda@p54A33F84.dip0.t-ipconnect.de) |
| 15:21.59 | *** join/#asterisk svenna_ (n=svenna@p548D332E.dip0.t-ipconnect.de) |
| 15:23.59 | *** join/#asterisk al123 (n=al123@190.40.92.209) |
| 15:24.23 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:24.38 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 15:24.59 | grandpapadot | Hi all. On the Polycom phones, I've been looking through the admin manual trying to find how to change my download FTP server via the config XML file with no luck. Does anyone know the setting? |
| 15:25.31 | Katty | meesha. |
| 15:25.34 | Katty | i'm starving! |
| 15:25.47 | [TK]D-Fender | grandpapadot: You don't. You do this via either a hard-coded entry in the bootrom, or via server specified via DHCP. |
| 15:26.05 | Seldon75 | hi all; my dialplan was set up by a 3rd party. When we dial 300 it goes straight to vm and now I want to change this to instead go to a specific user's phone. I found the following in extensions_users.conf: DN260 = Local/300@virtual_users can someone explain what that does? |
| 15:26.14 | grandpapadot | Thanks, TK! Ahh. What DHCP server option? And will it override a manually entered FTP server? |
| 15:27.04 | grandpapadot | Will TFTP option 66 work for FTP if FTP is set as the download protocol? |
| 15:27.25 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 15:27.36 | [TK]D-Fender | grandpapadot: No it will not override a manually entered one. |
| 15:27.44 | [TK]D-Fender | grandpapadot: Opt66. |
| 15:28.23 | [TK]D-Fender | Seldon75: Does noting but set a global. |
| 15:28.40 | grandpapadot | TK: tnx |
| 15:33.23 | al123 | hello, anyone can tell me pros/cons of internal or external fxo? like digium/fxo gateway? |
| 15:33.25 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:33.51 | al123 | i guess external fxo is better since is less load to server |
| 15:34.33 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:36.21 | [TK]D-Fender | al123: External : redundancy capable, remotely installable, simpler * server install, reduced load, no PIC / slot / case size issues |
| 15:37.43 | al123 | so you agree its better |
| 15:38.05 | *** join/#asterisk w3pog (n=pgrace@aeneas.fierymoon.com) |
| 15:39.20 | *** part/#asterisk whitehat (n=whitehat@unaffiliated/whitehat) |
| 15:39.58 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:42.51 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
| 15:43.02 | rickross | anyone had any issues with .WAV files failing to play under * 1.6.0 beta5 ? |
| 15:43.21 | rickross | mine have completely stopped playing, but .slin files play normally |
| 15:43.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:44.54 | *** topic/#asterisk by file -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta5 (2008/03/05), Asterisk 1.4.19-rc2 (2008/03/11), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
| 15:45.04 | Qwell | ~zombies |
| 15:45.06 | al123 | it is posible to play wav file with CLI? |
| 15:45.08 | Qwell | ~zombie |
| 15:45.09 | jbot | Library and server for developing networked apps/games.. URL: http://www.infa.abo.fi/~chakie/zombie/ |
| 15:45.25 | al123 | in CLI? |
| 15:45.35 | rickross | al123: I dunno how to do that - could you give me a CLI command for it? |
| 15:45.36 | km- | Qwell: have you ever tried load-balancing multiple asterisk boxen? |
| 15:46.25 | al123 | i dont know |
| 15:46.48 | al123 | when i play wav from freepbx i get i have no permission |
| 15:47.53 | rickross | I'm just doing a complete rebuild to be sure something didn't fail in the build process |
| 15:48.04 | rickross | format_wav.so looks like it is normal and loaded |
| 15:48.30 | rickross | make install |
| 15:48.36 | rickross | oops - sorry :( |
| 15:48.42 | BobLutz | lol |
| 15:49.59 | MatBoy | yeah, my digium cards arrived :) |
| 15:50.29 | BobLutz | MatBoy: mine too! |
| 15:50.33 | BobLutz | hi-fives |
| 15:50.49 | MatBoy | BobLutz, kewl ! I have some older ones... I'm going to test it with them because they were cheap :) |
| 15:50.55 | km- | anyone here play with biocluster? |
| 15:51.04 | jameswf | does that mean your going to buy us all green beer to celebrate |
| 15:51.37 | MatBoy | jameswf, I prefer dark beer |
| 15:51.50 | MatBoy | BobLutz, what kind of cards did you order ? |
| 15:52.10 | MatBoy | jameswf, maybe we can make a deal about that beer... I will buy it if you pay half |
| 15:53.35 | BobLutz | tdm402b |
| 15:53.49 | MatBoy | no pri cards I think ? |
| 15:53.54 | BobLutz | its a big step up - i know |
| 15:53.56 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-100-180.red.bezeqint.net) |
| 15:55.12 | MatBoy | BobLutz, nice... |
| 15:55.21 | BobLutz | heh |
| 15:55.30 | MatBoy | in what part of the connection will Echo occure if you have it ? |
| 15:56.07 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:56.07 | BobLutz | Echo() or echo on the line ? |
| 15:56.17 | MatBoy | yep |
| 15:56.36 | Qwell | yes, one of those? |
| 15:56.36 | MatBoy | it seems my telco is doing that already |
| 15:56.49 | MatBoy | one the line |
| 15:56.50 | MatBoy | :) |
| 15:57.41 | MatBoy | the cancelation |
| 15:57.48 | wwalker | I'm running Meetme with "MwsaA" everything associated with those flags works, BUT, there are no enter/leave sounds associated with people joining leving the call. It should be beeping if we don't have q as an option. anyone have any idea why it's not making noise? |
| 15:57.54 | MatBoy | I always can addon the addional cards |
| 15:58.06 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:58.11 | flujan | hi all. |
| 15:58.46 | flujan | guys, I have a small office with 3 peers outside my site. |
| 15:58.57 | flujan | This guys is registering on the pbx with the same IP. |
| 15:59.08 | flujan | they are the peers 19700 and 19701... |
| 15:59.30 | flujan | because they have the same IP when I call the peer 19700 the two devices rings... |
| 15:59.37 | flujan | how can I solve it? |
| 16:00.02 | wwalker | why would they have the same IP? |
| 16:00.56 | flujan | wwalver they are in the same lan and I have a firewall doing nat there... |
| 16:01.27 | al123 | hello, anyone can tell me pros/cons of internal or external fxo? like digium/fxo gateway? |
| 16:01.44 | Qwell | al123: didn't you already get an answer? |
| 16:01.47 | wwalker | flujan: how is traffic getting back to them? |
| 16:02.06 | al123 | i wnat to hear more oppinions |
| 16:02.12 | wwalker | do you have special tunnels set up? |
| 16:03.26 | *** join/#asterisk antonyo14 (n=amanager@ip-64-32-239-224.chi.megapath.net) |
| 16:06.22 | flujan | wwalker, now... They have a link to internet and I have a valid IP address on my server. |
| 16:07.39 | *** join/#asterisk shtoom (n=godson@123.176.41.46) |
| 16:10.34 | wwalker | flujan: the only way I've gotten that to work reliably is to use a VPN so that you have separate IP addresses for the machines behind that firewall. |
| 16:10.58 | wwalker | anyone else can jump in with another solution though. |
| 16:11.50 | Lsodi | any one having trouble with MixMonitor, channels on soundfile are out of sync, |
| 16:11.56 | wwalker | Total cost of the VPN solution, $150 and two hours of my time. I replaced the client's firewalls with WRT54GLs running OpenWRT and set up OpenVPN on them. |
| 16:12.26 | *** join/#asterisk VaNNi (n=VaNNi@lgb-static-216.70.165.200.mpowercom.net) |
| 16:12.48 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-123-234.bltmmd.east.verizon.net) |
| 16:14.14 | BobLutz | Asterisk module programming is mad tricky! |
| 16:14.44 | wwalker | flujan, OR, you can set up the different devices on different port #s and forward traffic through the firewall to them on a single IP that way |
| 16:15.13 | wwalker | still not sure if the RTP stream (which runs on an arbitrary port) will get through the firewall that way though |
| 16:23.45 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 16:24.30 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 16:32.04 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 16:32.39 | Lsodi | ok installed v1.4.17 incoming calls still are out of sync on zap cahnnel when using MixMonitor |
| 16:33.30 | Lsodi | but outgoing are good on zap channel |
| 16:33.48 | Lsodi | any one had similar problem? |
| 16:36.23 | *** join/#asterisk zobia (n=laurashr@222.212.65.250) |
| 16:36.34 | zobia | good morning everyone |
| 16:36.56 | zobia | any good man can share his spec file for asterisk for me to make asterisk 1.4.8 to rpm? thanks in advanced |
| 16:37.02 | anonymouz666 | Lsodi: I think there's an issue opened for that |
| 16:37.29 | file | there was, it has been fixed in SVN... and I'm pretty sure the code that caused it was never in 1.4.17 |
| 16:37.59 | BobLutz | chan->pbx->rtimeout - What type of timeout is this? (digit, absolute, etc) |
| 16:38.03 | anonymouz666 | so he means .18 |
| 16:38.29 | file | BobLutz: Timeout for response (seconds) |
| 16:38.47 | *** join/#asterisk xenonex (n=xenonex@89.218.236.55) |
| 16:38.49 | BobLutz | file: Thank you |
| 16:38.53 | Lsodi | I looked that issue, and its closed on 12.03 and buil is 13.03, I havent tryed downloading svn |
| 16:39.28 | file | we do not posess the power to go back in time and fix releases ;) |
| 16:39.45 | Qwell | or...do we? |
| 16:39.50 | Yourname`` | How can I change register timeouts? |
| 16:40.18 | Lsodi | ok, I'll try to build from svn |
| 16:40.22 | BobLutz | Yourname``: Set(TIMEOUT([digit|absolute])=n) |
| 16:40.40 | BobLutz | Yourname``: I think there is one more type of TIMEOUT but you get the idea |
| 16:41.06 | Yourname`` | I'm talking about the register asterisk does with the provider.. |
| 16:41.13 | BobLutz | bah |
| 16:41.21 | BobLutz | sucks at helping people |
| 16:41.23 | BobLutz | sorry |
| 16:43.39 | Lsodi | svn address is http://svn.digium.com/svn/asterisk/branches/1.4/ ? |
| 16:45.18 | floppybaer | need i a working capi? i have a usb hcfs isdn box |
| 16:45.41 | floppybaer | buerots:/usr/src/linux# capiinfo |
| 16:45.41 | floppybaer | capi not installed - No such device or address (6) |
| 16:45.41 | floppybaer | buerots:/usr/src/linux# |
| 16:46.01 | floppybaer | but incomming calls are listet in /var/log/messages |
| 16:46.59 | zobia | anyone knows how to make 1.4.18 to rpm? |
| 16:49.23 | *** join/#asterisk draygon-w (n=draygon@208.76.99.5) |
| 16:50.01 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:50.19 | jblack | whistles |
| 16:50.36 | jblack | Someone on bloomberg tv just used "It's a Wonderful Life" as a reference |
| 16:51.20 | Yourname`` | Is there a way to transfer a call when an agent gets a call via AgentLogin? |
| 16:51.41 | Dovid | Yourname: where do u want it sent to ? |
| 16:52.08 | jblack | Yourname``: Have them try flash-newnumber-ring-hangup |
| 16:52.36 | Dovid | i am too tired. i read it wrong |
| 16:52.43 | Yourname`` | They used to transfer by pressing xfer 6 xfer (6 was the extension where they had to transfer) |
| 16:53.12 | Yourname`` | jblack: So they'd do flash-6-ring-hangup on eyeBeam? |
| 16:53.17 | Dovid | Yourname: Do you want when ever they get a call to have it moved over or one by one if they want to xfer it ? |
| 16:53.44 | Yourname`` | Dovid: If they want to xfer it. |
| 16:54.13 | Dovid | when u try a reg. xfer what happens ? |
| 16:54.15 | floppybaer | cant you help me? |
| 16:54.24 | Dovid | nm. i was thinking of agentlogin |
| 16:54.32 | Dovid | where it calls them whenthey have a call. i need sleeeeeeep |
| 16:54.41 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 16:54.47 | Dovid | floppybaer: What do you need help with ? |
| 16:55.20 | floppybaer | Dovid i cant activate my capi |
| 16:55.24 | floppybaer | buerots:/usr/src/linux# capiinfo |
| 16:55.26 | Dovid | dont know capi :( |
| 16:55.30 | floppybaer | capi not installed - No such device or address (6) |
| 16:56.24 | km- | how many 4 port TDM boards can I reliably fit in a box? |
| 16:56.35 | km- | is 4 the going standard? |
| 16:56.43 | Qwell | km-: there are also 8 and 24 port |
| 16:56.49 | Qwell | depends on the size of the box, I guess |
| 16:56.50 | km- | I mean for T1's |
| 16:56.58 | floppybaer | hmm mustent i have capi runing? |
| 16:57.00 | km- | 4 port T1/E1's |
| 16:57.03 | Qwell | oh |
| 16:57.11 | Qwell | yeah, 4 |
| 16:57.16 | km- | so, 4x Te410p's or something of that nature. |
| 16:58.38 | dandre | hello |
| 16:59.02 | dandre | Where can I find doc on ${HINT} variable? |
| 16:59.31 | BobLutz | dandre: /doc/channelvariables.txt in the Asterisk source tree |
| 16:59.45 | [TK]D-Fender | km-: A108d <- |
| 17:00.06 | dandre | ok |
| 17:00.20 | Lsodi | in svn mixmonitor is fixed thank you good people. |
| 17:02.35 | km- | [TK]D-Fender: a108d? |
| 17:02.41 | [TK]D-Fender | km-: Yes. |
| 17:03.49 | km- | [TK]D-Fender: how many a108d's have you run simultaneously? |
| 17:04.00 | km- | I'm going to need to drop 1200 simultaneous calls into a RAD multiplexer. |
| 17:04.08 | km- | it's all a question of how many asterisk boxes I want in there. |
| 17:04.39 | [TK]D-Fender | km-: I wouldn't do more than 2/ server. So 16 T1 |
| 17:05.17 | km- | hmmm, ok |
| 17:05.23 | km- | damn, gotta go to a meeting, back in a bit. |
| 17:05.53 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 17:06.44 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 17:08.00 | *** join/#asterisk michelem (n=michelem@mac-dkr04.inf.ethz.ch) |
| 17:09.53 | *** join/#asterisk BBHoss_Laptop (n=r00t@71-14-23-142.static.dctr.al.charter.com) |
| 17:13.52 | *** join/#asterisk linuxmaniac (n=maniac@80.30.198.23) |
| 17:16.45 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:17.05 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 17:17.19 | dandre | I have setup something like http://pastebin.com/m6e1040ba but I have problems |
| 17:17.40 | Yourname`` | My friend [TK]D-Fender, is there any way I can transfer a call when an agent is logged in via AgentLogin()? |
| 17:17.46 | flujan | hi guys... I am with the following problem... I log in a ip phone with ip 192.168.1.115 and peer 29700 on asterisk |
| 17:18.00 | dandre | the inbound channel seems to be hungup between the two dials |
| 17:18.03 | flujan | I call 29700 and it works... ringing the 29700 peer. |
| 17:18.19 | dandre | the stdexten macro ends with macroexit |
| 17:18.22 | flujan | I switch the peer on the softphone configuration to 29701. |
| 17:18.39 | [TK]D-Fender | dandre: Why do you have 3 x "s,1"? |
| 17:18.43 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 17:18.53 | [TK]D-Fender | dandre: taht = no bright |
| 17:18.54 | flujan | So when i call 29701 all goes well and the 29701 peer rings... |
| 17:18.55 | flujan | but |
| 17:18.56 | [TK]D-Fender | that* |
| 17:19.10 | [TK]D-Fender | Yourname``: Yes. |
| 17:19.17 | flujan | when i call the 29700 peer ITS ALSO RING THE 29701 ip phone... |
| 17:19.25 | Yourname`` | [TK]D-Fender: How, good sir? |
| 17:19.51 | flujan | asterisk is not updating the information and keep the 29700 logged in with the same ip and port of the 29701 peer... |
| 17:19.59 | dandre | oups sorry, http://pastebin.com/m451a6a93 |
| 17:20.04 | flujan | how can I make asterisk expire it on let me say 60 seconds??? |
| 17:20.43 | [TK]D-Fender | Yourname``: "core show application queue" |
| 17:21.11 | [TK]D-Fender | dandre: And you are failing to show us the problem.... |
| 17:21.28 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 17:21.35 | *** join/#asterisk xenonex (n=xenonex@89.218.236.55) |
| 17:21.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:21.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:22.11 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 17:22.16 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 17:25.03 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 17:25.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:25.43 | *** join/#asterisk mipster (n=mipster@75.131.201.166) |
| 17:26.57 | *** join/#asterisk adjohn (n=adjohn@p7221-ipad87marunouchi.tokyo.ocn.ne.jp) |
| 17:27.18 | Seldon75 | we have an occasional problem with our Sangoama card where the analog input ports stop working, and the only way we have found to get them to work again is to restart the Asterisk box. Is there a command to reset the Zaptel interface I could try? |
| 17:27.33 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 17:27.44 | Seldon75 | zap restart from the Asterisk console I guess |
| 17:27.55 | Seldon75 | any other suggestiuons to diagnose the issue? |
| 17:35.42 | *** join/#asterisk ManxPower (n=manxpowe@137.sub-75-201-8.myvzw.com) |
| 17:40.57 | dandre | sorry, I have to go |
| 17:41.00 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:41.16 | *** join/#asterisk Navion (n=billp@75-107-96-17.cust.wildblue.net) |
| 17:41.49 | Navion | Anyone have experience with IAX2 through a pfsense firewall? |
| 17:42.08 | mipster | I'm running IAX2 to voicepulse thru pfsense |
| 17:42.14 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:42.48 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 17:43.07 | Navion | <mipster> did you have any trouble setting up the NAT hrough the firewall? |
| 17:44.40 | nny_1 | setting up a SIP failover (ha) here for testing.. scenario is incoming to local telco line (if busy) telco forwards to 1-888 number. 1-888 number is SIP from outside provider. If person is unavailable, forwards back out (on SIP, if local lines are still busy) to cell phone. |
| 17:44.47 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:44.50 | nny_1 | trying to discern the best way to handle that |
| 17:45.16 | Navion | I'm trying to set up an Asterisk to Asterisk IAX2 link and it looks like pfsense wants to change the reply port so it can map it back. |
| 17:47.07 | *** join/#asterisk codec (n=codec@iglu.paranoid-penguin.de) |
| 17:47.13 | codec | hi there |
| 17:47.17 | gutz|work | im using MixMonitor on my server to record calls and when i listen to the recorded calls the audio is totally out of sync |
| 17:47.23 | codec | anyone running asterisk 1.4.17 with bristuff on ubuntu 7.04? |
| 17:47.25 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 17:47.25 | gutz|work | i see nothing in the wiki about how to deal with this |
| 17:47.31 | gutz|work | is there a knows fix for this? |
| 17:47.33 | *** join/#asterisk xenonex (n=xenonex@89.218.236.55) |
| 17:48.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:48.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:48.09 | codec | i'm using a duoBRI card and if i try to load chan_zap all I get is: |
| 17:48.10 | codec | zt_open: Unable to specify channel 1: No such device or address |
| 17:48.49 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 17:49.00 | gutz|work | codec: whats ztcfg -vv say? |
| 17:49.33 | codec | gutz|work: http://rafb.net/p/gU8HmB24.html |
| 17:50.32 | gutz|work | well, i dont know much about this stuff but i would say that if ztcfg -vv isnt clean then you have a hardware or zaptel.conf configuration problem |
| 17:51.01 | codec | well, it worked fine on debian |
| 17:51.59 | codec | d'oh |
| 17:52.19 | codec | just removed span 3 and 4 |
| 17:52.23 | codec | seems to work (wtf?) |
| 17:53.06 | *** join/#asterisk AfricAnne (n=AfricAnn@41.221.58.4) |
| 17:53.09 | gutz|work | you have a 2 head card? |
| 17:53.13 | codec | yep |
| 17:53.16 | gutz|work | why do you have 4 spans? |
| 17:54.09 | codec | i'm wondering why this did work a few hours ago |
| 17:54.15 | codec | strane. |
| 17:54.17 | codec | *strange |
| 17:55.04 | gutz|work | so noone has run into issues in which Mix Monitor records out fo sync? |
| 17:55.38 | *** join/#asterisk joe (n=njsauer@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 17:56.55 | codec | now it stopped working again. |
| 17:57.33 | AfricAnne | Does one still require licensing to use thee G729 codec? |
| 17:57.39 | Qwell | AfricAnne: yes |
| 17:58.06 | AfricAnne | Qwell: Thank you, I was just curious... |
| 17:58.32 | AfricAnne | I'm having a look at the wiki right now for hardware dimensioning :-) |
| 17:58.57 | codec | http://rafb.net/p/Zr5jn792.html |
| 17:59.37 | gutz|work | codec, what is the output of zap show status? |
| 17:59.37 | nny_1 | anyone know of some cookbook code to setup SIP channel failover |
| 17:59.45 | gutz|work | codec, and ztcfg -vv is clean now"? |
| 18:00.12 | nny_1 | nm found it |
| 18:00.27 | codec | gutz|work: http://rafb.net/p/i2AVni78.html |
| 18:01.19 | gutz|work | im not sure if you need libpri for a BRI card |
| 18:01.28 | gutz|work | does pri show spans give you any output? |
| 18:02.05 | codec | oh, wait a sec |
| 18:02.10 | *** join/#asterisk michelem (n=f00@mac-dkr04.inf.ethz.ch) |
| 18:02.16 | michelem | hello folks |
| 18:02.30 | michelem | I have a very coarse question about asterisk |
| 18:02.52 | codec | gutz|work: looks like it chooses randomly which of the 2 ports is span 1 |
| 18:02.54 | michelem | I needed to dig to find that it requires FSX hardware |
| 18:03.03 | codec | gutz|work: just plugged the cable into the 2nd span and it works (detected as span1) |
| 18:03.13 | gutz|work | codec: it should be the same port every time |
| 18:03.26 | gutz|work | codec, then the second port should always be span 1 |
| 18:03.28 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 18:03.47 | codec | gutz|work: i'll do one more reboot to check that |
| 18:03.51 | michelem | but I have only found "modules" for these. What is the "minimal" hardware required to connect asterisk to a pstn? |
| 18:04.01 | *** part/#asterisk AfricAnne (n=AfricAnn@41.221.58.4) |
| 18:04.21 | nny_1 | michelem: each card is different |
| 18:04.29 | gutz|work | michelem, a te100p clone card |
| 18:04.29 | nny_1 | michelem: some require "modules" some don't etc |
| 18:04.34 | nny_1 | eww |
| 18:04.37 | ManxPower | michelem: Do you need analog or digital support? |
| 18:04.45 | gutz|work | thats the minimum to connect asterisk to a single POTS line |
| 18:04.49 | nny_1 | ~100p |
| 18:04.50 | michelem | wow :) |
| 18:05.00 | nny_1 | true true |
| 18:05.08 | michelem | ok so |
| 18:05.14 | gutz|work | whats wrong with 100p cards? |
| 18:05.25 | michelem | I don't known what features I can get from analog or digital |
| 18:05.25 | gutz|work | they are good, cheep and do what you need them to do |
| 18:05.26 | ManxPower | gutz|work: You are an idiot. A TE100P does NOT support analog. |
| 18:05.31 | nny_1 | lol |
| 18:05.41 | nny_1 | i was waiting for a senior channel member to pipe in |
| 18:05.43 | gutz|work | erm |
| 18:05.46 | gutz|work | my bad |
| 18:05.47 | michelem | basically I need calls to come in and the voice being forwarded to a SIP address |
| 18:05.48 | gutz|work | X100p! |
| 18:05.51 | nny_1 | yah i use em for testing and for FXO ojnly |
| 18:06.03 | ManxPower | The X100P has not been made in several years |
| 18:06.09 | gutz|work | ManxPower, watch the name calling |
| 18:06.21 | mipster | michelem: TDM410P with a single FXO module is a popular option |
| 18:06.30 | gutz|work | http://iaxtalk.com/index.php?main_page=product_info&products_id=6 |
| 18:06.33 | ManxPower | gutz|work: then don't give out wrong information |
| 18:06.43 | gutz|work | that is the 'minimum' needed to connect to a single POTS line |
| 18:06.43 | codec | gutz|work: it DOES change on reboot |
| 18:06.44 | michelem | mipster: thanks, actually I expected to hear about "popular options" |
| 18:07.01 | gutz|work | codec, hmmmmm, im not sure how to fix that. sorry. |
| 18:07.22 | mipster | michelem: I use a TDM400P (older card) with 1 FXS and 1 FXO. Rock solid. |
| 18:07.37 | mipster | michelem: not the cheapest option. But popular :) |
| 18:07.38 | codec | gutz|work: no problem. at least it works somehow :) |
| 18:08.07 | ManxPower | You can try to find a cheap X100P "clone card" which is not supported here, in fact it's not supported anywhere. |
| 18:08.14 | *** join/#asterisk xenonex (n=xenonex@89.218.236.55) |
| 18:08.44 | michelem | the iax one is 15$ and the TDM400P is 100$ |
| 18:08.53 | nny_1 | i think i have my SIP failover (ouotbound) right can someone look at http://pastebin.com/m6885945a and tell me how much I suck |
| 18:08.53 | michelem | what is the difference? hardware reliability? |
| 18:09.09 | mipster | michelem: one thing that might confuse the newcomer. There are model numbers for bundles: TDM410P + modules. |
| 18:09.35 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 18:09.35 | rkeene | <PROTECTED> |
| 18:09.46 | mipster | michelem: example -> TDM410p with 1 FXS and 1 FXO can be called TDM411BF |
| 18:09.55 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:10.44 | michelem | one thing that confuses me as a newcomer is the huge amount of terminology for all this :) |
| 18:10.44 | nny_1 | there is a variation on the CALLERID part, gonna fix that btw |
| 18:11.11 | michelem | I always needed personal calls to come in and being "forwarded" to a SIP address in the beginning :) |
| 18:11.39 | mipster | michelem: btw, the TDM410P can take 4 modules (any combination of FXO and FXS) |
| 18:12.21 | michelem | s/always/only |
| 18:12.28 | michelem | ok, let me check |
| 18:12.40 | codec | gutz|work: is it possible that it takes some time until the channel gets detected as up? |
| 18:13.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:13.48 | gutz|work | codec, shouldnt take very long... at least thats me experience with my PRI line. use the output of zap show status to see if it sees any alarms |
| 18:14.45 | codec | gutz|work: 1 port, 1 span - right? |
| 18:15.08 | michelem | so FXO is the "WAN" line in packet-switched network terminology and FXS is the LAN line |
| 18:15.18 | gutz|work | codec, tahts the way mine is |
| 18:15.20 | michelem | in which case I only need a FXO |
| 18:15.47 | mipster | michelem - FXO (Foreign Exchange Office) connects to the office (telco), FXS (FX Station) connects to a handset (station) |
| 18:16.05 | mipster | michelem, so FXO to connect to the telco, FXS to connect to a handset |
| 18:16.33 | michelem | yes, but I expect I can stay without the handset |
| 18:16.44 | mipster | michelem, although, somewhat confusingly, when you configure the FXO under linux, you'll specify FXS signalling :) |
| 18:16.46 | michelem | meaning my "backoffice" is all voip |
| 18:16.53 | mipster | michelem, and vice-versa |
| 18:18.09 | michelem | that's good, my next problem was indeed figuring out how much the hardware will cost. And having 15$ vs 100$ + modules make me indeed wonder what is the difference |
| 18:18.19 | mipster | michelem, so, if you need JUST an FXO, and its just for playing at the moment, there are cheaper options |
| 18:18.39 | mipster | michelem, but I'll let someone who knows the other options better comment... |
| 18:19.29 | shido6 | what costs $15? |
| 18:20.14 | codec | wow, this just seems to be broken |
| 18:20.16 | *** join/#asterisk stoffell (n=stoffell@d51A4D79B.access.telenet.be) |
| 18:20.22 | codec | i made asterisk segfault with unplugging the pri-line :> |
| 18:20.55 | michelem | okay, thanks |
| 18:21.44 | SteveTotaro | codec: yeah, that seems pretty borken |
| 18:22.10 | codec | sucks |
| 18:22.13 | gutz|work | i did that last week |
| 18:22.16 | gutz|work | :-/ |
| 18:22.25 | SteveTotaro | got a core dump? |
| 18:22.33 | gutz|work | nope, should have saved it |
| 18:22.39 | codec | i'll check for it |
| 18:22.42 | codec | just rebooted the machine |
| 18:22.47 | michelem | so there are those 15$ or some "x100p" originals for 29$, which is now 18€ :) |
| 18:22.48 | nny_1 | lol |
| 18:22.48 | gutz|work | where would it be? |
| 18:23.53 | michelem | last question |
| 18:24.18 | michelem | we all used to use good old fashioned modems for connecting to the internet years ago |
| 18:24.29 | nny_1 | so i treid http://pastebin.com/m6885945a and got http://pastebin.ca/946431 |
| 18:24.39 | nny_1 | can someone tell me why the chan status is congestion? |
| 18:24.40 | SteveTotaro | only if you can find one with the right 3200 chipset |
| 18:24.46 | michelem | I still have one. What is the difference with old serial ones and the PCI "FXO"s? |
| 18:24.54 | nny_1 | sorry about the mix matched pastebins, the second one was spam filtered lol |
| 18:25.33 | SteveTotaro | the driver is written specifically for that modem with 3200 chipset (going off old memory of chipset) |
| 18:25.49 | mipster | nny_1, if I'm not mistaken (Congestion) winds up being a catchall channel status? |
| 18:25.54 | nny_1 | mipster: yah |
| 18:26.01 | nny_1 | mipster: was hoping to get CHANUNAVAIL lol |
| 18:26.43 | SteveTotaro | chipset intel 537 or md3200 |
| 18:27.10 | SteveTotaro | that is just from a quick google, i never heard of the 537 though |
| 18:27.24 | michelem | ok |
| 18:27.51 | rkeene | All my incoming calls are directed to "s@from-pstn" -- is this normal ? |
| 18:27.56 | SteveTotaro | if you write drivers for another modem, i am sure you could probably make it work, just not worth the trouble |
| 18:28.18 | SteveTotaro | keene: it is normal if the arriving call does not have a destination supplied |
| 18:28.19 | michelem | great |
| 18:28.27 | mipster | nny_1, couldn't you just follow the Dial(Zap) with a Sial(SIP) and let the dialplan fall thru if the Zap is busy or unavail? |
| 18:28.43 | nny_1 | mipster: yah i could lol |
| 18:28.49 | michelem | if asterisk is running is it still possible to use the modem, say, with hylafax for sending a fax, or the access to the device is exclusive? |
| 18:29.06 | rkeene | SteveTotaro, Why wouldn't the call have a destination ? (How am I supposed to route it ? ... this is on a PRI) |
| 18:29.07 | nny_1 | but that wouldn't be nearly goldebergesque enough lol |
| 18:29.11 | SteveTotaro | you got me there, i do not think so though |
| 18:29.21 | JayTee52 | anyone here use PoE for VOIP phones? |
| 18:29.23 | mipster | rkeene, all calls from an FXO port will dial the "s" extension. The "from-pstn" is the context which is set in the channel config file. |
| 18:29.37 | SteveTotaro | rkeene: turn on pri debugging |
| 18:29.42 | *** join/#asterisk jetsnoc (n=brian@pdpc/supporter/active/jets) |
| 18:29.58 | mipster | rkeene, is the call coming from an FXO line or PRI? |
| 18:30.01 | rkeene | mipster, PRI |
| 18:30.03 | stoffell | JayTee52, we all do :-0 |
| 18:30.43 | SteveTotaro | you should be receiving however many digits you requested when the PRI was turned up |
| 18:30.53 | mipster | rkeene, oops. Then "s" would mean there is no called number info in the inbound PRI setup msg?? |
| 18:30.59 | JayTee52 | stoffell, so no problems with using it on VOIP phones? that's good to know. |
| 18:31.02 | *** join/#asterisk mirrorcolor (n=iunixan@196.218.222.13) |
| 18:31.16 | jetsnoc | any von attendees? |
| 18:31.17 | rkeene | mipster, So maybe my upstream isn't sending it ? |
| 18:31.30 | JayTee52 | we want to use PoE at our location so I'm trying to find more info on it. |
| 18:31.38 | mipster | rkeene, usually when you order a PRI you tell the provider how many digits to send |
| 18:31.43 | SteveTotaro | rkeene: if you turn on pri debugging on the console, you can see what is being sent |
| 18:32.08 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:32.08 | rkeene | SteveTotaro, I have it on, but I don't see anything particularly interesting |
| 18:32.10 | SteveTotaro | then you have ammunition when you call tier 1 |
| 18:32.14 | SteveTotaro | dial in |
| 18:32.34 | SteveTotaro | how many DIDs do you have? |
| 18:32.37 | stoffell | JayTee52, well, just make sure you use compatible hardware.. but most recent stuff is good (cisco, linksys, polycom, snom, ...) |
| 18:33.37 | SteveTotaro | do pri intense debug span 1 |
| 18:33.48 | SteveTotaro | call in and pastebin the output |
| 18:33.58 | rkeene | Hang on |
| 18:37.55 | rkeene | http://www.rkeene.org/viewer/tmp/pri-debug.txt.htm |
| 18:38.46 | codec | damn it |
| 18:39.09 | codec | gutz|work: thanks for the help. but something seems to be definitely fucked up. i'll check that tomorrow. gotta go home now or i'll go insane. :) |
| 18:39.12 | codec | bye folks |
| 18:39.26 | gutz|work | codec, i know what feeling, have a good one! |
| 18:40.08 | BBHoss_Laptop | anybody have a source for polycom firmwares? |
| 18:40.52 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 18:40.58 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D79B.access.telenet.be) |
| 18:40.58 | rkeene | BBHoss_Laptop, http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 18:41.18 | _ShrikE | BBHoss: your reseller should be able to provide you with the latest version |
| 18:41.24 | SteveTotaro | try unknown in zapata.conf for your dialplan entries |
| 18:41.48 | EmleyMoor | Are there any significant services for toll-free calls, other than North American via sipbroker and those which are available through FWD? How about one for French numeros verts? |
| 18:42.53 | rkeene | SteveTotaro, Hmm ? |
| 18:43.09 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 18:43.12 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D79B.access.telenet.be) |
| 18:43.21 | SteveTotaro | Location: Private network serving the local user (1) |
| 18:43.38 | SteveTotaro | pastebin your zapata.conf |
| 18:43.43 | rkeene | BBHoss_Laptop, http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip550.html has links to them, not sure why the page I sent you doesn't give you the REAL links |
| 18:44.27 | gutz|work | OK, this is driving me CRAZY. what could cause calls in mixmonitor to be out of sync? |
| 18:44.48 | EmleyMoor | gutz|work: How far out of sync? |
| 18:44.52 | BBHoss_Laptop | yeah i hate polycoms firmware strategy, doesn't make a bit of sense |
| 18:44.54 | gutz|work | few seconds |
| 18:44.57 | gutz|work | and its constant |
| 18:45.05 | rkeene | BBHoss_Laptop, Get the second URL I sent you ? |
| 18:45.06 | gutz|work | same duration out of sync from the beginning of the call to the end |
| 18:45.09 | gutz|work | at least as far as i can tell |
| 18:45.12 | file | gutz|work: fixed in SVN... |
| 18:45.14 | rkeene | SteveTotaro, http://www.rkeene.org/tmp/zapata.conf |
| 18:45.21 | UnixDog | the 550 are nice |
| 18:45.22 | BBHoss_Laptop | rkeene, yeah |
| 18:45.25 | UnixDog | I have one |
| 18:45.30 | UnixDog | it works great |
| 18:45.47 | BBHoss_Laptop | there was a site in NZ that used tohave the latest firmware, like up to 4.0 but it looks like its down now |
| 18:45.50 | JayTee52 | stoffell, sorry I had to step away from my desk for a moment. Thank you very much for your input. |
| 18:45.51 | gutz|work | file: really? sweet! problem is that SVN isnt exactly considered stable is it? |
| 18:45.54 | UnixDog | you can get the plantronics headset for it on ebay for 19.95 |
| 18:46.12 | file | gutz|work: it is what the next release candidate will be made out of... and eventually the next release |
| 18:46.35 | rkeene | SteveTotaro, How do I turn off PRI debugging now ? :-P |
| 18:46.45 | rkeene | Nevermind, found it |
| 18:46.52 | gutz|work | arnt i at a significant risk of checking out something with conflicts and brokenness though? |
| 18:47.43 | file | conflicts? no brokenness? it is possible |
| 18:48.05 | SteveTotaro | pri no debug span N |
| 18:48.16 | rkeene | Yeah, I found it |
| 18:50.51 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 18:51.37 | SteveTotaro | ok, look at the sample zapata.conf in your source, if you installed from source |
| 18:52.08 | rkeene | Okay, looking at it |
| 18:53.17 | rkeene | "pridialplan=unknown" ? |
| 18:53.26 | SteveTotaro | yes, i think that might do it |
| 18:53.34 | SteveTotaro | otherwise they may not be sending |
| 18:53.38 | rkeene | Okay |
| 18:53.40 | rkeene | Thanks! |
| 18:54.10 | *** join/#asterisk antxoneti (n=antonio@30.127-200-80.adsl-dyn.isp.belgacom.be) |
| 18:54.24 | SteveTotaro | just FYI http://lists.digium.com/pipermail/asterisk-users/2005-January/073887.html |
| 18:54.56 | Yourname`` | If asterisk crashes, does it do a dump somewhere? |
| 18:55.16 | SteveTotaro | prilocaldialplan=unkown |
| 18:55.16 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:58.16 | rkeene | SteveTotaro, Thanks... no change... must be NASA, grrbl. |
| 18:58.50 | SteveTotaro | can i see the output? |
| 19:02.00 | rkeene | http://www.rkeene.org/viewer/tmp/pri-debug-2.txt.htm |
| 19:03.29 | SteveTotaro | rkeene: did you stop asterisk and do ztcfg -s, ztcfg -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 19:03.57 | rkeene | No.. I thought ztcfg only looked at /etc/zaptel.conf (I did stop and restart asterisk, though) |
| 19:04.12 | SteveTotaro | Yourname``: i think it depends on what version of asterisk and how you start it |
| 19:04.20 | UnixDog | ok 1.6 beta 5 cores on bsd |
| 19:04.21 | Yourname`` | Running asterisk on a dual cpu duocore box which shows 4 cores on htop, and for some reason I only see the 4th core being used upto 95% and the rest of them are almost idle! |
| 19:04.23 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:04.26 | Yourname`` | SteveTotaro: 1.4 and safe_asterisk |
| 19:04.30 | UnixDog | ]it runs fine for 2 calls then cores |
| 19:04.34 | UnixDog | this bites |
| 19:05.00 | SteveTotaro | not sure about the safe_asterisk script or 1.4 |
| 19:05.11 | SteveTotaro | i think you need to compile it to give core dump |
| 19:05.17 | SteveTotaro | maybe wrong though |
| 19:05.52 | Yourname`` | Probably wrong. |
| 19:06.07 | SteveTotaro | i know with 1.2 if you start it with the c or g switch it will leave a dump |
| 19:06.14 | SteveTotaro | c or g not sure which |
| 19:06.24 | Yourname`` | yeah asterisk -cvvvvvvv |
| 19:06.31 | Yourname`` | or something |
| 19:06.50 | SteveTotaro | v is just verbose, i guess g give you the console |
| 19:08.16 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 19:09.07 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-6d63e94e64ebec9e) |
| 19:09.44 | SteveTotaro | Last reload: 15 weeks, 6 days, 56 minutes, 47 seconds |
| 19:09.55 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 19:10.09 | SteveTotaro | that is why i like 1.2.X |
| 19:10.19 | BBHoss_Laptop | anybody know what "Config file error 0x0" means on a polycom? |
| 19:11.58 | SteveTotaro | http://track.sipfoundry.org/browse/XCF-860 |
| 19:12.59 | SteveTotaro | BBHoss_Laptop: did you mess with bootrom? |
| 19:14.10 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 19:14.15 | BBHoss_Laptop | SteveTotaro, maybe, could it be a wrong bootrom version |
| 19:14.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 19:15.18 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 19:15.19 | puzzled | hi |
| 19:16.20 | SteveTotaro | http://lists.digium.com/pipermail/asterisk-users/2006-October/170190.html |
| 19:16.41 | SteveTotaro | BBHoss_Laptop: check that thread |
| 19:18.08 | *** part/#asterisk Navion (n=billp@75-107-96-17.cust.wildblue.net) |
| 19:24.53 | SteveTotaro | gold hit $1,030.80 today |
| 19:25.01 | SteveTotaro | oops, wrong window |
| 19:25.03 | Qwell | pawns his ring |
| 19:25.14 | Yourname`` | Ok guys, asterisk is crashing, |
| 19:25.17 | Yourname`` | And going the debug thing. |
| 19:25.29 | Yourname`` | I just rebooted the box, and I'm going to start it.. |
| 19:25.38 | Yourname`` | And put it back into produciton the way it was.. |
| 19:25.50 | Yourname`` | Now, is there a special way I need to start it so I can log as much as possible to submit? |
| 19:26.01 | SteveTotaro | ouch, how much production are we talking about? |
| 19:26.17 | SteveTotaro | you need a core dump |
| 19:26.29 | Yourname`` | I do have a core dump right now from a previous crash. |
| 19:26.43 | Yourname`` | But the wiki says I'd need to do more like strace and all, and I'm a newb when it comes to that. |
| 19:27.06 | SteveTotaro | run gdb on it |
| 19:27.17 | [TK]D-Fender | BBHoss_Laptop: Means you've corrupted your provisioning files |
| 19:27.40 | SteveTotaro | it should show you what code caused the core dump |
| 19:27.44 | BBHoss_Laptop | [TK]D-Fender, yeah i'm redoing everything now |
| 19:28.29 | generalhan | can some one point in me in the right direction here pleae? i used a lot of priority jumping in 1.2 and im getting my 1.4 box ready to roll out and im looking for good alternatives to replace the jumping. |
| 19:28.39 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 19:29.08 | [TK]D-Fender | generalhan: "upgrade.txt" in 1.2 & 1.4 |
| 19:29.19 | SteveTotaro | do you mean N+101? |
| 19:29.27 | generalhan | [TK]D-Fender: it shows alternatives ? |
| 19:29.34 | [TK]D-Fender | generalhan: And read the instructions for all the apps you are using priority jumping with. |
| 19:29.51 | *** join/#asterisk tsabi (n=tsabi@pool-2186.adsl.interware.hu) |
| 19:30.02 | SteveTotaro | is there a fix for fastagi failing? |
| 19:30.15 | SteveTotaro | without priority jumping patch like in 1.2? |
| 19:33.10 | *** join/#asterisk implicit (n=implicit@200.12.227.181) |
| 19:36.35 | TJNII | Uh.... Is it bad when nws.noaa.gov lists "Unknown Precipitation" as the weather? |
| 19:37.35 | rkeene | Having just got off the phone with someone at NOAA trying to configure a switch... nope! |
| 19:37.36 | bkruse | TJNII: That does not sound good |
| 19:37.38 | rkeene | :-P |
| 19:40.52 | *** join/#asterisk mintee (n=mintee@72-165-177-94.dia.static.qwest.net) |
| 19:40.58 | gutz|work | should the latest release version of asterisk-addons be compatible with the SVN version of asterisk? |
| 19:41.10 | gutz|work | the SVN version of asterisk-addons seems to be broken |
| 19:41.19 | mintee | Hi all, is there a stable fix for iax modems on asterisk 1.4? |
| 19:41.28 | mintee | they all seem to die after a few days |
| 19:41.54 | gutz|work | mintee, you are referring to the IAXmodem project by Lee Howard? |
| 19:46.39 | gutz|work | noone knows if the SVN version of asterisk will work with the release version of asterisk-addons? |
| 19:47.12 | bkruse | gutz|work: svn as in the 1.4 branch, or trunk |
| 19:47.24 | gutz|work | trunk |
| 19:48.21 | SteveTotaro | will zaptel in the 1.4 branch work with asterisk 1.2? |
| 19:48.31 | mintee | gutz|work, i've got a box up and running with iaxmodem and hylafax for listening in for faxes, but the iax module in 1.4.(18) dies after a few days |
| 19:49.19 | gutz|work | mintee, i tried that for a while and never found iaxmodem to be stable enough or have enough functions to work properly. If you wish to keep trying i would get in touch with Lee Howard directly as he is a great guy who is more then willing to help |
| 19:49.37 | mintee | i noticed a few weeks ago that it's a known problem, and just wanted to see if anyone had an update or not, as I failed to bookmark the threads... |
| 19:50.30 | SteveTotaro | cronjob at midnight to restart gracefully |
| 19:50.32 | mintee | gutz|work, cool, maybe i'll do that. I've found it to work fine... I'm recieving faxes and having them emailed as pdf's to me with out a problem... |
| 19:50.49 | mintee | heh, until of course asterisk misses sync with them, and considers them gone |
| 19:52.25 | SteveTotaro | why didn't he write sipmodem? it would work better i bet |
| 19:52.52 | SteveTotaro | or even skinnymodem |
| 19:53.10 | rkeene | According to the FAQ it's because IAX was esier. |
| 19:53.27 | *** join/#asterisk F-eeks (n=F-eeks@brc29-1-82-245-193-9.fbx.proxad.net) |
| 19:53.28 | SteveTotaro | now what is the lesson learned here? |
| 19:53.46 | F-eeks | Rico29: pouet. |
| 19:53.57 | SteveTotaro | "don't take the easy way out" |
| 19:54.01 | Rico29 | :] |
| 19:54.17 | F-eeks | viens sur le salon archlinux-fr c'est plis la classe |
| 19:54.17 | rkeene | SteveTotaro, I'd think it'd be: "Don't write buggy code" :-P |
| 19:54.21 | SteveTotaro | pinoy? |
| 19:54.41 | SteveTotaro | i thought code by it's very definition was buggy |
| 19:55.30 | SteveTotaro | mon crayon est tres grand |
| 19:57.29 | mintee | are you guys talking about iaxmodem still? |
| 19:57.50 | mintee | cause it's not iaxmodem that's buggy, it's the iax asterisk module that's buggy |
| 19:57.54 | mintee | worked fine in 1.2 |
| 19:59.43 | BobLutz | F-eeks: You love teh Archlinux? |
| 20:00.00 | BobLutz | F-eeks: Vous adorez l'Archlinux ? |
| 20:00.45 | F-eeks | yes |
| 20:00.46 | F-eeks | oui |
| 20:00.55 | SteveTotaro | tu adore l'Archlinux ? |
| 20:01.01 | F-eeks | lol |
| 20:01.09 | SteveTotaro | which is more correct? |
| 20:01.10 | BobLutz | frowns at poor French skillz |
| 20:01.13 | BobLutz | vous |
| 20:01.16 | F-eeks | i liebe das Archlinux |
| 20:01.20 | SteveTotaro | vous is many |
| 20:01.22 | F-eeks | ich liebe das archlinux |
| 20:01.25 | F-eeks | no |
| 20:01.25 | SteveTotaro | plural |
| 20:01.27 | F-eeks | vous is you |
| 20:01.32 | SteveTotaro | tu is you |
| 20:01.34 | BobLutz | vous is a polite "you" |
| 20:01.37 | F-eeks | yes |
| 20:01.38 | SteveTotaro | singular |
| 20:01.41 | F-eeks | or many you |
| 20:01.56 | BobLutz | if you were talking to a lil kid, you would say "tu", if they were talking to you, they would say "vous" as a sign of seniority |
| 20:02.13 | BobLutz | SteveTotaro: You would say "tu" to me, and I would say "vous" to you |
| 20:02.16 | BobLutz | ;) |
| 20:02.32 | SteveTotaro | let F-eeks elaborate |
| 20:02.41 | F-eeks | thats it BobLutz |
| 20:02.45 | BobLutz | bam |
| 20:02.47 | SteveTotaro | ok |
| 20:02.48 | F-eeks | but |
| 20:02.58 | SteveTotaro | always a "but" |
| 20:03.02 | F-eeks | vous is also when you speak to severals guys |
| 20:03.10 | BobLutz | o yea, true |
| 20:03.17 | SteveTotaro | so we are both correct? |
| 20:03.26 | F-eeks | i say "Tu... BobLutz " and "Vous... BobLutz and SteveTotaro " |
| 20:03.34 | SteveTotaro | bam |
| 20:03.36 | F-eeks | yep |
| 20:03.42 | F-eeks | you are both correct |
| 20:03.47 | BobLutz | wammy |
| 20:04.11 | SteveTotaro | what about mon crayon est tres grand, do the women like? |
| 20:04.19 | BobLutz | LOL |
| 20:04.38 | F-eeks | hmm... maybe if you are Hugh Grant... |
| 20:04.55 | BobLutz | burn! |
| 20:05.14 | SteveTotaro | ok lisse moi |
| 20:05.57 | F-eeks | lisse ? |
| 20:06.11 | F-eeks | smooth me ? |
| 20:06.22 | SteveTotaro | http://www.osiris.sn/article1636.html all me baby |
| 20:07.19 | SteveTotaro | j'etudie francais pour quatre ans, mais mon francais est tres mal |
| 20:07.41 | F-eeks | non non c'est bien |
| 20:08.17 | BobLutz | le francais, c'est.... |
| 20:08.25 | F-eeks | francais. |
| 20:09.30 | SteveTotaro | <---- est mal de tete |
| 20:11.46 | bkruse | .... #asterisk-fr |
| 20:11.56 | SteveTotaro | guingol, guingol, bravo guingnol, voila guingol!!! |
| 20:12.07 | F-eeks | lool |
| 20:13.26 | JunK-Y | SteveTotaro:" a mal a la tet" sounds much better! |
| 20:13.29 | JunK-Y | tete |
| 20:18.28 | [TK]D-Fender | F-eeks: Alors pour le bien conjuger en etre "direct" et "personel" : va t'ens tabarnac ;p |
| 20:18.53 | ice_croft | [TK]D-Fender> u know french |
| 20:18.56 | ice_croft | damn |
| 20:20.19 | SteveTotaro | one thing i really hate are french keyboards |
| 20:20.23 | SteveTotaro | i mean why |
| 20:20.26 | SteveTotaro | ? |
| 20:20.56 | *** join/#asterisk jhiver (n=jhiver@161-242.206-83.static-ip.oleane.fr) |
| 20:20.58 | jhiver | hi all |
| 20:21.17 | SteveTotaro | i could see Chinese or languages with different letters |
| 20:21.24 | BobLutz | accent agu |
| 20:21.34 | jhiver | i was wondering if it's possible to defined random variables such as foo=bar in under the sip.conf peer config and then use these variables in the dialplan in extensions.conf? |
| 20:21.37 | SteveTotaro | but the layout is different |
| 20:22.23 | jhiver | right now i have to use a different context in the dialplan just to set one variable, it kind of blows... |
| 20:22.32 | jhiver | one context per peer that is |
| 20:22.40 | [TK]D-Fender | BobLutz: "egu" |
| 20:22.44 | SteveTotaro | Cedilla |
| 20:23.17 | [TK]D-Fender | jhiver: "set=myvar1=12345" |
| 20:23.26 | jhiver | cool |
| 20:23.27 | SteveTotaro | just make that a function key |
| 20:23.29 | jhiver | thx |
| 20:23.35 | [TK]D-Fender | jhiver: "setvar=myvar1=12345" <- I believe its setvar actually.. |
| 20:24.39 | jhiver | ok found it :) |
| 20:24.41 | jhiver | <PROTECTED> |
| 20:24.53 | jhiver | great! |
| 20:24.55 | jhiver | cheers |
| 20:25.11 | BobLutz | bkruse : would altering (i.e - removing) any of the API calls to indications.c effect the DTMF detection in say app_read.c ? |
| 20:25.25 | [TK]D-Fender | jhiver: Bienvenue. |
| 20:25.36 | bkruse | BobLutz: search for the API call in indiciations, not sure though |
| 20:26.06 | BobLutz | Looking at it right now and not sure either, thanks and sorry for the random specific API question haha |
| 20:27.08 | bkruse | BobLutz: it is ok :] |
| 20:29.01 | *** join/#asterisk kombi_ (n=kombi@port-213-160-14-18.static.km-it.de) |
| 20:29.41 | [TK]D-Fender | BBIAB |
| 20:30.07 | kombi_ | something I never dared to ask: with a sip softphone, can you - in theory - increase sound quality to like 22k 16bit? |
| 20:30.28 | kombi_ | Hi Fender! |
| 20:31.21 | BobLutz | he gone |
| 20:31.42 | kombi_ | good old Fender Precision..;) |
| 20:35.59 | BobLutz | Does tone_zone refer to "us" or "uk" etc in indications.conf ? |
| 20:36.25 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:38.58 | tzanger | kombi_: yes, it's called wideband codecss |
| 20:39.03 | tzanger | I don't think asterisk supports them at this time |
| 20:39.37 | kombi_ | tzanger: I just checked, G711 seems to be the best one in there still |
| 20:39.56 | fujin | detachesheh |
| 20:39.58 | fujin | speex? |
| 20:40.14 | fujin | g711 is the best |
| 20:40.31 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 20:40.41 | kombi_ | a free codec for fee speech, I like that! |
| 20:41.07 | fujin | well, the PSTN uses g711, there's no reason to use anything else unless you're skimping on bandwidth |
| 20:41.38 | kombi_ | fujin: that's right, it is a very special case though |
| 20:42.44 | kombi_ | whoops, there something on voip-info about it.. |
| 20:46.15 | gutz|work | does anyone know why the SVN head of asterisk-addons is broken |
| 20:46.22 | gutz|work | erm |
| 20:46.34 | gutz|work | SVN trunk |
| 20:47.32 | Bryanstein | gutz|work, what link are you using? I don't know why but maybe the link is broken |
| 20:47.38 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 20:47.47 | gutz|work | svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons |
| 20:47.49 | gutz|work | it checks out |
| 20:47.53 | gutz|work | but wownt compile |
| 20:47.57 | gutz|work | the release version will compile |
| 20:49.20 | file | do you have trunk Asterisk installed? |
| 20:50.03 | flush | when i enter the vocemail service |
| 20:50.10 | gutz|work | file: yes |
| 20:50.11 | flush | what does she say??? i really dont understand.. |
| 20:50.21 | flush | vm-login |
| 20:50.23 | file | gutz|work: pastebin the output |
| 20:50.28 | Nugget | I compile my asterisk from the pre-svn copies I scrape out of /home/file/src/asterisk_work |
| 20:50.34 | gutz|work | let me rephrase that |
| 20:50.43 | gutz|work | i have the release asterisk isntalled, i have the trunk asterisk compiled |
| 20:50.45 | BobLutz | lol |
| 20:50.57 | file | that is vastly different |
| 20:51.04 | file | addons builds against the Asterisk you have installed |
| 20:51.08 | gutz|work | well, i wish to upgrade both at once |
| 20:51.17 | gutz|work | or must i install asterisk trunk then compile asterisk-addons |
| 20:51.23 | gutz|work | ahhhh |
| 20:51.26 | gutz|work | that makes sense then |
| 20:51.31 | file | unless you pass the path to configure |
| 20:51.42 | file | like so: ./configure --with-asterisk=<path to Asterisk source code for trunk> |
| 20:53.09 | Qwell | ^^ not gonna work :D |
| 20:53.19 | gutz|work | why not? |
| 20:54.10 | file | Qwell: eh? EH? |
| 20:54.25 | Qwell | it'll try to install the modules there |
| 20:54.30 | Qwell | err, maybe |
| 20:54.34 | Qwell | it'll do funky things :p |
| 20:54.52 | lirakis | is away (leaving..."the internets" are safe ... for now) |
| 20:54.58 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D79B.access.telenet.be) |
| 20:54.59 | file | worked for me :D |
| 20:55.01 | Qwell | ...maybe |
| 20:55.22 | file | I think you have just gone crazy again |
| 20:55.40 | Qwell | probably |
| 20:55.45 | Qwell | blames Corydon76-dig's brownies. |
| 20:55.58 | Qwell | (it was awesome, btw) |
| 20:56.32 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 20:57.42 | flush | vm-login, what is said?? |
| 20:59.14 | file | flujan: Comedian Mail. Mailbox? |
| 20:59.24 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:59.33 | flujan | file: ? |
| 20:59.58 | BobLutz | If you base a module off app_skel.c, but write the rest yourself, can you submit that to the Asterisk project? (and get street cred) |
| 21:00.23 | file | BobLutz: provided you wrote it, yes |
| 21:00.27 | BobLutz | giggles |
| 21:01.10 | BobLutz | Modifying app_swift isnt cutting the cheese...I realized I had the swift header file, so I might just re-write the swift integration myself |
| 21:01.18 | *** join/#asterisk russellb (n=mobile@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:01.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:01.25 | rpyne | exit |
| 21:01.36 | Qwell | O.o |
| 21:01.38 | BobLutz | rpyne: try /exit |
| 21:01.45 | *** part/#asterisk rpyne (n=richard@69.77.169.14) |
| 21:02.15 | *** join/#asterisk russellb[mobile] (n=mobile@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:02.15 | *** mode/#asterisk [+o russellb[mobile]] by ChanServ |
| 21:02.26 | Qwell | russellb[mobile]: welcome back |
| 21:02.43 | BobLutz | Is app_read.c the only basic app that accepts a "#"-delimited DTMF string ? |
| 21:02.46 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 21:02.48 | russellb[mobile] | thanks |
| 21:02.58 | file | russellb[mobile]: hi! |
| 21:03.34 | russellb[mobile] | status? |
| 21:03.54 | file | russellb[mobile]: we missed you today! <3 |
| 21:04.10 | russellb[mobile] | awww, thanks :) |
| 21:04.36 | BobLutz | anthm: Did you write app_cepstral ? |
| 21:05.28 | russellb[mobile] | these people next to me are acting like they have never heard an intercom ... WHERE IS THAT GUY? |
| 21:05.38 | fujin | rofl |
| 21:05.41 | fujin | Intercom is so fun |
| 21:05.48 | fujin | I wish it was possible to playback a song into app_page |
| 21:05.56 | russellb[mobile] | it is |
| 21:05.57 | fujin | I'm always app_paging my office and saying "Make your time gentleman" |
| 21:06.00 | fujin | "Main screen turn on" |
| 21:06.07 | fujin | "Move all zig for great justice" |
| 21:06.18 | fujin | Need to work out how to play back Rick Astley through app_page. |
| 21:06.26 | russellb[mobile] | originate a call to page with playback on the other end |
| 21:06.34 | Qwell | intercom rickroll...is that even fair? |
| 21:06.40 | russellb[mobile] | using a local channel |
| 21:06.46 | Qwell | I'm pretty sure there's a rule or something |
| 21:06.54 | russellb[mobile] | hehe |
| 21:07.09 | file | rules are meant to be broken. |
| 21:07.10 | fujin | meh! |
| 21:07.11 | fujin | breaks rules |
| 21:07.25 | fujin | Heh, we were streamign the cricket out from here the other day |
| 21:07.27 | fujin | with VLC |
| 21:07.28 | russellb[mobile] | I'm going |
| 21:07.34 | fujin | and after an over, we'd toss a rickroll in there |
| 21:07.40 | fujin | (doom rickroll, family guy.. etc.) |
| 21:07.41 | fujin | such win |
| 21:07.43 | russellb[mobile] | to page flea market at our office |
| 21:07.52 | fujin | haha |
| 21:07.52 | gutz|work | im getting segmentation faults when i try and start asterisk |
| 21:07.57 | file | russellb[mobile]: oh no you don't! |
| 21:08.00 | gutz|work | what can i use to get more detail about what is happening? |
| 21:08.01 | Yourname`` | russellb[mobile]: You missed it.. I got a coredump today! :D |
| 21:08.14 | russellb[mobile] | oh noes |
| 21:08.15 | fujin | gutz|work: DONT_OPTIMIZE, -g |
| 21:08.16 | Yourname`` | M12242 |
| 21:08.24 | *** join/#asterisk BryanFE4 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net) |
| 21:08.48 | gutz|work | fujin, it made no difference. im starting asterisk with 'asterisk -gvvvvv' |
| 21:08.55 | fujin | -g will drop a core dude :) |
| 21:09.01 | russellb[mobile] | I'll look tomorrow ... |
| 21:09.13 | gutz|work | ahhh |
| 21:09.14 | BryanFE4 | if I have a ".call" file which connects to a phone number and then executes an extension - is there a command I can put into the extension dialplan which says "hang up and re-execute this .call file from the start"? |
| 21:09.19 | gutz|work | well, where can i look for this dump now? |
| 21:09.46 | fujin | gutz|work: in /tmp iirc |
| 21:09.56 | gutz|work | ahh, yes |
| 21:09.57 | fujin | you'll have to gdb the core file, get a backtrace and some other stuff; there's a tutorial in the docs directory iirc |
| 21:09.57 | gutz|work | there it is |
| 21:10.35 | fujin | yeah, docs/backtrace.txt |
| 21:10.41 | Yourname`` | russellb[mobile]: Thanks.. |
| 21:11.07 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 21:11.16 | Yourname`` | russellb[mobile]: I'm trying to set affinity as well, bkruse recommends that.. |
| 21:11.22 | Yourname`` | I hope it solves my problemo! lol |
| 21:11.23 | russellb[mobile] | 'tis my job... |
| 21:11.41 | Yourname`` | It is a production box too.. |
| 21:11.48 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 21:11.49 | Yourname`` | So I'm hoping it doesn't happen again. |
| 21:12.01 | Yourname`` | But before I started it up this time, I disabled autofill. |
| 21:12.26 | Yourname`` | russellb[mobile]: Atleast for the rest of day till the bug is updated I'm prayin for it to work, lol |
| 21:12.37 | adeel | in the beginning of all my calls (zap -> sip, sip -> sip) there's a 2-3 second audio delay....is there anyway to get around that? or reduce it at the least? |
| 21:14.41 | *** join/#asterisk Alpha_AI (n=Ben@iti142937-1.gw.connect.com.au) |
| 21:15.53 | Alpha_AI | Hello, has anyone got a softphone that shows you which extensions are logged into the softphone and which extensions are currently on calls? |
| 21:16.23 | Alpha_AI | i meant has anyone got a softphone that shows you which extensions are logged into the pbx and which extensions are currently on calls? |
| 21:20.22 | [TK]D-Fender | Alpha_AI, eyeBeam |
| 21:20.40 | Alpha_AI | does eyebeam allow it? |
| 21:20.44 | Alpha_AI | ive installed x-lite |
| 21:20.49 | Alpha_AI | isnt that the same thing? |
| 21:20.50 | Katty | what's a nice ftp server? |
| 21:20.57 | *** part/#asterisk BryanFE4 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net) |
| 21:22.00 | [TK]D-Fender | Katty, vsftpd |
| 21:22.01 | Bryanstein | Katty, what distro are you using...must be M$ ? |
| 21:22.48 | Bryanstein | proftp is much better |
| 21:23.00 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:23.00 | Bryanstein | that is if you want a little security |
| 21:23.05 | Bryanstein | tls |
| 21:23.27 | nny_1 | so what is the best way to take a file (mp3/wav etc) and convert it with sox to something MOH likes? |
| 21:23.35 | nny_1 | i keep trying and failing lol |
| 21:23.59 | nny_1 | says frequency 4400 unexpected now, first it complained about stereo (ok good reason) |
| 21:25.38 | Katty | [TK]D-Fender: is that very hard to configure? |
| 21:25.43 | Katty | [TK]D-Fender: i've never done ftp in linux before. |
| 21:25.55 | [TK]D-Fender | Katty, No, pretty easy |
| 21:26.00 | nny_1 | check_header: Unexpected freqency 44100 heh code typo! |
| 21:26.15 | nny_1 | trying to figure out what* frequency MOH will enjoy |
| 21:26.50 | Katty | [TK]D-Fender: cheers (= |
| 21:27.37 | nny_1 | so i am assuming sox something foo.wav foobar.wav is what i need |
| 21:28.05 | nny_1 | is there a definitive do this with sox and MOH will always be happy command somewhere? |
| 21:28.46 | nny_1 | i see 8000 on the wiki |
| 21:28.47 | [TK]D-Fender | nny_1, You should know that all wav's should be 8khz ULAW.... |
| 21:28.55 | russellb[mobile] | make sure you tell sox to resample the output file to 8khz |
| 21:28.55 | nny_1 | oh yeah |
| 21:29.03 | gutz|work | ugh |
| 21:29.09 | nny_1 | [face palm] |
| 21:29.13 | nny_1 | thanks lol |
| 21:29.21 | nny_1 | just remember reading that chapter |
| 21:29.22 | gutz|work | how long before the next version of asterisk is out? trying to run trunk is giving me big problems |
| 21:29.26 | russellb[mobile] | Wav != ulaw |
| 21:29.40 | gutz|work | how could they release a version of * that has a broken mixmonitor |
| 21:30.22 | russellb[mobile] | guts .... it is not released, now is it :) |
| 21:30.25 | russellb[mobile] | report your bugs, please |
| 21:30.59 | russellb[mobile] | bugs.digium.com |
| 21:31.54 | file | and the issue in question has been fixed across all the trees, not just trunk... |
| 21:31.59 | Yourname`` | I just bugged one. |
| 21:32.02 | nny_1 | ty for jarring the memory [TK]D-Fender russellb[mobile] |
| 21:32.03 | Yourname`` | I mean reported a bug! |
| 21:32.10 | nny_1 | buggered what? |
| 21:32.12 | Yourname`` | And putnopvut says it actually is a bug! |
| 21:32.20 | russellb[mobile] | file: you rock |
| 21:32.29 | Yourname`` | Yes, file, you rock. |
| 21:32.35 | Yourname`` | Just read your interview. |
| 21:32.43 | russellb[mobile] | heh |
| 21:32.45 | Yourname`` | Actually both of yours interviews. |
| 21:32.54 | russellb[mobile] | I bet that is oldddd |
| 21:33.07 | russellb[mobile] | time to board ... |
| 21:33.18 | Yourname`` | And now I have a face to you guys. So everytime russellb[mobile] helps me out, I'd know that he's not being impatient with me! :D |
| 21:34.11 | bkruse | +b's Yourname`` |
| 21:34.14 | bkruse | :P |
| 21:35.05 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:35.10 | file | I prefer my facebook pic to the one on that interview |
| 21:35.11 | Yourname`` | lol, bkruse if you had an interview, it'd be helpful! :P |
| 21:35.30 | fujin | blows up Yourname`` |
| 21:36.10 | Yourname`` | jumps on facebook |
| 21:36.16 | Yourname`` | wtf! Everyone is damaging me left and right today. |
| 21:36.27 | Yourname`` | +bs, bombs, coredumps.. |
| 21:36.55 | bkruse | file: agreed |
| 21:37.38 | file | bkruse: I don't have you on facebook! |
| 21:37.53 | bkruse | file: I need to add you |
| 21:39.03 | Yourname`` | HAHA, if I'm looking at the right one, that seems like one classic pic. |
| 21:39.58 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-157-226.telkomadsl.co.za) |
| 21:40.15 | Bryanstein | hey any of you guys heard of the NXA400P tdm400 clone from nxtvox.com? I am buying a card today or tomorrow and I would appreciate a little feedback |
| 21:40.45 | xacatecas | hi guys, with DUNDi, is it possible to "re-use" an existing iax2 trunk that does not make use of the dbsecret=dundi/secret ? |
| 21:40.50 | Bryanstein | if know one knows then I'll spend 60 more for a tdm400 |
| 21:41.02 | xacatecas | i need to be able to accomodate clients that support DUNDi, as well as some that don't. |
| 21:41.06 | [TK]D-Fender | Bryanstein, Nope, and you'll find very few people here willing to attempt to buy a clone card like that... |
| 21:41.25 | Bryanstein | lol [TK]D-Fender why is that ? |
| 21:41.25 | file | i'M NOT HERE. |
| 21:41.28 | file | whoops |
| 21:41.41 | file | runs off |
| 21:41.42 | Bryanstein | thanks for replying too [TK]D-Fender |
| 21:42.25 | Bryanstein | It is touted as a native asterisk card...doesn't even need the zaptel drivers to run..yadda yadda yadda |
| 21:43.01 | Yourname`` | BS |
| 21:43.16 | [TK]D-Fender | Bryanstein, If its a TDM400P clone then it obviously requires zaptel. |
| 21:43.17 | xacatecas | was about to say ... how can you not need drivers? |
| 21:43.29 | Qwell | ~cheap |
| 21:43.30 | jbot | cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 21:43.36 | Bryanstein | well the site says so lol |
| 21:43.46 | [TK]D-Fender | Bryanstein, unless you're going to say they did something rediculous like write an enteri * channel driver for the interface. |
| 21:43.59 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:44.03 | [TK]D-Fender | Bryanstein, Which given the fact you said it was a "clone" is pretty near impossible |
| 21:44.11 | jetsnoc | Buy digium cards. Feed our friends. |
| 21:44.41 | jhiver | buy a SIP / PRI gateway (audiocodes, vegastream). Save yourself some trouble |
| 21:44.45 | [TK]D-Fender | Bryanstein, If you believe everything you read then I've got some ocean-front property in Nevada to sell you cheap.... |
| 21:45.23 | Bryanstein | Ocean front in Nevada...does it come with fishing poles |
| 21:45.39 | Bryanstein | Well I never pretend to know things that I don't [TK]D-Fender that is why I ask |
| 21:46.07 | Bryanstein | I'm fairly new to the whole pbx scene |
| 21:48.25 | jhiver | Hear me out: a decent SIP gw is your friend, you'll save yourself a lot of hassle... I've got a mediant and it's been running 24/7 for the last 3 years, I never even had to reboot the damn thing |
| 21:48.39 | bkruse | jhiver: running what? |
| 21:49.11 | jhiver | running 4 E1s and turning them into a nice and easy to use SIP trunk |
| 21:49.50 | jhiver | doing things like transcoding and jitter buffer properly |
| 21:50.10 | jhiver | supporting T.38 protocol |
| 21:50.13 | jhiver | stuff like that :) |
| 21:50.31 | [TK]D-Fender | Bryanstein, What do you want to do with * exactly? |
| 21:50.55 | jhiver | loves his mediant 2000 |
| 21:51.50 | Bryanstein | I have 4 sip phones(pingtels and avaya 8410d's) and I would like to answer my pots line and do a little freaky sip stuff |
| 21:52.07 | jhiver | huh |
| 21:52.14 | Bryanstein | huh me? |
| 21:52.29 | jhiver | 4 SIP phones, pehaps a big iron SIP gw is overkill |
| 21:52.44 | jhiver | there's some other analog models though |
| 21:52.45 | jhiver | but |
| 21:52.50 | jhiver | analog is so bad |
| 21:53.12 | Bryanstein | This is for the home...not a production mission critical system |
| 21:53.35 | Bryanstein | I'd like for the phones to always ring of course, but nothing serious |
| 21:55.01 | xacatecas | Bryanstein, based on personal experience I can recommend the TDM800 cards, with echo-can + quad-fxo module, but I suspect you only have a single line. |
| 21:55.03 | [TK]D-Fender | Bryanstein, How many lines? |
| 21:55.21 | Bryanstein | yes one line |
| 21:55.40 | xacatecas | then a tdm800 is probably overkill and overly expensive :p |
| 21:55.52 | [TK]D-Fender | Bryanstein, http://www.telephonydepot.com/product_p/105-054-312.htm |
| 21:55.59 | xacatecas | the echo-can module does make a huge difference though. |
| 21:56.02 | Bryanstein | lol I really could use a x100 but I heard they suck when it comes to caller id |
| 21:56.02 | [TK]D-Fender | Bryanstein, End of story. |
| 21:56.30 | jhiver | i had a play with one of those ages ago |
| 21:56.39 | jhiver | i think [TK]D-Fender is right |
| 21:56.48 | xacatecas | thirds that |
| 21:57.10 | jhiver | you're be a lot better off with one of the SPA's (Ex SIPURA if memory serves) |
| 21:58.04 | xacatecas | jip, your memory is correct. |
| 21:58.53 | Bryanstein | Thanks for the information guys |
| 22:04.19 | xacatecas | hmm, i'm only starting to play with DUNDi, so i'm full of questions, is there a descent DUNDi document available somewhere? |
| 22:13.02 | xacatecas | ok, question 1. Say I've got two clients. i've also got an "upstream", be that sip/iax2 shouldn't matter. Now, basically I've got some numbers locally (well, the numbers of my clients, these are routed to them), and I've got numbers from my upstream. |
| 22:13.37 | xacatecas | if either i own a number, or my upstream owns a number, then I'd like the client to call via the IAX trunk configured between myself and the client. |
| 22:14.05 | xacatecas | otherwise it's dependant on distances etc ... those are sorted out separately. |
| 22:14.45 | xacatecas | now, from what I've been reading DUNDi provides the caller with a username+password to use, but in my case I've already got a trunk between the two ... how exactly does that work? |
| 22:16.50 | rkeene | [Mar 17 17:14:18] WARNING[14851]: chan_sip.c:3669 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 22:16.59 | rkeene | What is this all about ? |
| 22:17.33 | jblack | xacatecas: Well, you setup your dialplan to check for a number through dundi... |
| 22:17.56 | jblack | such as: ;exten => s,n,Dial(Local/${ARG1}@dundi-e164-lookup) |
| 22:18.02 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:18.13 | jblack | Dundi does the lookup, returns that, and dial does it's magic. |
| 22:18.36 | xacatecas | and if dundi returns nothing, it just skips on to the next record in the dialplan? |
| 22:18.51 | rkeene | Ah.. it's just an annoyance, excellent. |
| 22:19.04 | jblack | Yah. |
| 22:19.18 | xacatecas | ok, i'm not even at the dialplan yet, i still need to make dundi work. |
| 22:19.22 | jblack | So, you can have another dial on the next priority up, and it'll use that as a backup. |
| 22:19.43 | xacatecas | and all the howto's i find is just "do this blah blah blah" with no explanation of why things are needed. or a very bad why. |
| 22:19.44 | jblack | It's your time, and your plan, but I wouldn't bother with dundi for anything but internal use |
| 22:20.05 | xacatecas | hehe, well, it's not entirely just internal. |
| 22:20.23 | jblack | Ok, well, public dundis always evetnually collapse due to tragedy of the commons. |
| 22:21.07 | xacatecas | hehe, this isn't entirely public, it's for my clients sake so that even though they're within "local" (cheap) rates of each other, can use VoIP. |
| 22:21.25 | jblack | Ok, that's part of what I'd call internal. |
| 22:21.38 | xacatecas | however, I don't want to have to log in to every client every time more clients get on. |
| 22:21.43 | jblack | Check out messinet.com/dundi |
| 22:22.01 | xacatecas | looking, thanks. |
| 22:22.11 | xacatecas | 404 ? |
| 22:22.14 | jblack | You'll have to configure each node upon original installation, but once you've done that, they'll work. |
| 22:22.26 | jblack | pardon. http://messinet.com/project/dundi |
| 22:23.00 | *** join/#asterisk sergey (n=chatzill@sergey.iks.ru) |
| 22:23.19 | xacatecas | what does E.164 stand for? |
| 22:23.31 | xacatecas | or rather, what does it mean? |
| 22:23.39 | defsdoor | it's a food additive |
| 22:24.08 | jhiver | actually that's a good question, i have no clue what it "means", i just know it's a numbering format where you dial |
| 22:24.14 | jhiver | countrycode + area code + SN |
| 22:24.22 | jhiver | without any leading zeroes |
| 22:24.25 | jhiver | or anything like that |
| 22:25.45 | _ShrikE | Is is possible to set alert-info on calls returning from a parking timeout? |
| 22:25.48 | *** join/#asterisk dlynes (n=dlynes@mail.247communications.com) |
| 22:26.03 | xacatecas | eg, 27121234567 |
| 22:27.55 | xacatecas | so mr james, i'd be willing to assist with that novel setup, but I've got to warn you that latencies between america and SA is sometimes rather horrible. |
| 22:28.09 | xacatecas | (>700ms at times) |
| 22:28.35 | jhiver | 700 ms is OK as long as the ping is stable (i.e no packet loss & little jitter) actually |
| 22:28.42 | jhiver | hell, even 900 ms is sort of OK |
| 22:28.58 | jblack | I'm not part of that network anymore. I'm just referring you to it so that you have documentation to look at |
| 22:29.01 | xacatecas | darn, i always thought you wanted <300ms for voice. |
| 22:29.13 | jhiver | you do |
| 22:29.20 | xacatecas | ok. |
| 22:29.20 | jhiver | but jitter and packet loss |
| 22:29.27 | jhiver | are much more important |
| 22:29.32 | dlynes | packet loss is intolerable |
| 22:29.50 | jhiver | give me a stable 900ms over a jittery packet lossy 200 ms |
| 22:29.52 | jhiver | anytime |
| 22:29.52 | xacatecas | ok well, let's say we want to give it a shot, does someone have an IP I can ping as a test? |
| 22:30.21 | jhiver | what do you want to test? |
| 22:30.32 | jhiver | an IP with high ping but stable connection? :) |
| 22:30.38 | jhiver | ping 83.206.114.90 |
| 22:30.42 | xacatecas | well, any potential dundi peer |
| 22:30.46 | jhiver | it'll be high :) |
| 22:30.47 | jhiver | ah |
| 22:30.48 | jhiver | dundi? |
| 22:30.50 | jhiver | eeewwww |
| 22:31.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:31.36 | xacatecas | rtt min/avg/max/mdev = 545.881/546.733/550.544/0.923 ms, pipe 6 |
| 22:31.38 | xacatecas | not bad. |
| 22:32.06 | dlynes | Is there anything special I need to do with Debian Etch R3, in order to get it build clean on zaptel 1.4.9.2? |
| 22:34.36 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 22:37.28 | matt_ | hello, I have a script that places a file in the outgoing directory for asterisk to call, and when this call is answered to call another number, this all works but when asterisk calls the number in the outgoing file and the line is answered there isn't any ringing tone played down the line, how can i play a ringing tone ? |
| 22:38.00 | *** join/#asterisk Bonix-BR (i=500@189.52.0.234) |
| 22:38.22 | matt_ | if i send them to an extension that has a Playtones() line will that work? |
| 22:44.02 | *** join/#asterisk RoyK (n=roy@ip-194-22-149-91.dialup.ice.no) |
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| 22:48.10 | jblack | can macros call macros? |
| 22:48.39 | jblack | never mind. won't work anyways. |
| 22:51.57 | xacatecas | hehe, must I use rsa keys for DUNDi? |
| 22:52.13 | jblack | why wouldn't you want to? |
| 22:57.59 | xacatecas | testing? |
| 22:58.26 | *** join/#asterisk BobLutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 22:58.26 | jblack | So.... make testing rsa keys. |
| 22:59.47 | BobLutz | Should I build Asterisk SVN code in my /home directory? |
| 22:59.57 | BobLutz | Is that problematic? |
| 23:00.01 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 23:02.13 | dlynes | BobLutz: no |
| 23:02.35 | dlynes | BobLutz: it's just fine...build it wherever makes the most sense for you |
| 23:03.09 | BobLutz | dlynes, I have only built stable releases, and that was in /usr/src/ as root, I did not know if it would make more sense to build SVN code as my user |
| 23:03.24 | BobLutz | this sucks |
| 23:03.56 | xacatecas | BobLutz, you should not ever need to compile as root. not ever. |
| 23:04.00 | dlynes | BobLutz: if you don't want to clobber your existing installs, I believe there's an override you can provide to configure or to make to tell it you want to use a staging directory |
| 23:04.42 | BobLutz | xacatecas, Other system was just a dev box, I never bothered to create other users |
| 23:04.45 | dlynes | xacatecas: some system level stuff will compile differently if you're not root user, but your statement is valid for 99.9% of the code out there |
| 23:05.21 | xacatecas | dlynes, please do give me one example where it compiles (with a valid reason) differently. |
| 23:05.39 | dlynes | xacatecas: well, not differently...just won't compile, period |
| 23:05.45 | xacatecas | example? |
| 23:05.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:06.14 | dlynes | xacatecas: if it needs to invoke a third-party utility that needs to run as root user, to extract some information only the root user has access to, in order to provide configuration information for the compile |
| 23:06.17 | BobLutz | Yea, I thought certain things during the build process could only be done by root... such as creating /etc/asterisk |
| 23:06.35 | dlynes | BobLutz: that's during make install, not normal run of make |
| 23:06.42 | BobLutz | teehee |
| 23:06.46 | xacatecas | dlynes, that's a valid reason. but i haven't yet seen such software. |
| 23:06.57 | xacatecas | BobLutz, thats part of make install. |
| 23:07.06 | BobLutz | yea wasnt thinking |
| 23:07.13 | xacatecas | argh, the lag. |
| 23:07.32 | dlynes | xacatecas: I think there was some earlier versions of nmap that required that, and I've come across some other low-level utilities that required it, too...but as I said...it's very uncommon |
| 23:07.46 | xacatecas | :) |
| 23:08.42 | BobLutz | Yea..soo I will make this as user bobbyb on my system |
| 23:11.41 | riddlebox | if you have grandstream gxp2000 phones, onetop it has line 1 - line 4, I know they are really accounts but is there a way to have them subscribe to asterisk-fx0-1 thru fxo4? |
| 23:12.21 | BobLutz | makemenuselect == denied! |
| 23:12.31 | BobLutz | err make menuselect |
| 23:13.00 | JT | riddlebox: are they really FXOs? |
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| 23:14.10 | riddlebox | JT, yeah the system will have 4 fxo ports, I know that they are really just accounts like line 1 connects to my asterisk system, and line 2 could connect to another provider.... |
| 23:15.06 | *** join/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net) |
| 23:18.18 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
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| 23:24.08 | really_phukt | What kind of hammer do u guys recommend to use if I can't get my asterisk running?.. |
| 23:25.26 | jetsnoc | LOL do you mean the hammer on a gun? :) |
| 23:26.10 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:27.45 | really_phukt | I like the hammer idea or maybe a crowbar... They wreak more damage... :D |
| 23:29.24 | *** join/#asterisk dansaz (n=dan@c-68-58-81-102.hsd1.in.comcast.net) |
| 23:30.29 | dansaz | Can asterisk work with any standard 18 button office phones? |
| 23:30.48 | JT | there are "standard" 18 button office phones? |
| 23:31.01 | dansaz | Avaya Euro 18D are the ones I'm interested in |
| 23:31.23 | *** join/#asterisk machpo (n=machpo@ketone.jawfish.com) |
| 23:31.51 | JT | they are not standard |
| 23:31.55 | JT | they would be proprietary |
| 23:33.37 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:34.08 | Alpha_AI | how do i find out what linx distro i am using |
| 23:34.13 | Alpha_AI | what can i type in to find out |
| 23:34.13 | *** join/#asterisk znoG (n=gs@host18.190-139-223.telecom.net.ar) |
| 23:36.03 | *** join/#asterisk tainted_ (n=tainted@adsl-75-34-145-25.dsl.irvnca.sbcglobal.net) |
| 23:36.17 | tainted_ | anyone have dtmf issues using rfc2833 through some smartphones |
| 23:36.26 | tainted_ | smartphones/mobile carriers |
| 23:36.40 | riddlebox | dansaz, no |
| 23:37.04 | [TK]D-Fender | Alpha_AI, how is it you don't know what you're using? |
| 23:37.37 | dansaz | is there a recommended hard phone for use with Asterisk? |
| 23:37.53 | [TK]D-Fender | dansaz, Polycom SIP phones are the most often recommended. |
| 23:38.12 | riddlebox | dansaz, telephonydepot.com, voipsupply.com, and I am sure there are more sites |
| 23:38.25 | dansaz | tyvm! |
| 23:39.46 | xacatecas | dundi show peers -> 00:15:17:62:d7:90 196.34.133.147 (S) Outbound Unavail OK (10 ms) |
| 23:39.54 | xacatecas | what does that mean? |
| 23:40.10 | xacatecas | tainted_, yea, if they also send the dtmf via sip info. |
| 23:40.27 | tainted_ | xacatecas what's the workaround? |
| 23:40.37 | xacatecas | disable one of the two. |
| 23:40.55 | tainted_ | how do i disable dtmf via sip info |
| 23:41.33 | dlynes | dansaz: there's a proprietary answer to your proprietary problem: Citel's EXTender series of products: http://www.citel.com/ |
| 23:42.09 | dlynes | dansaz: Their stuff interoperates at a minimum with Avaya and Nortel, but some products also interoperate with others keysystems and PBXs |
| 23:42.18 | tainted_ | i have dtmfmode=rfc2833 |
| 23:42.20 | [TK]D-Fender | dlynes, Not compatible. |
| 23:42.25 | [TK]D-Fender | dlynes, I checked :) |
| 23:42.28 | dlynes | [TK]D-Fender: not compatible with what? |
| 23:42.36 | [TK]D-Fender | dlynes, with the 18D handsets. |
| 23:42.41 | dlynes | [TK]D-Fender: ah |
| 23:43.21 | dlynes | [TK]D-Fender: but it works with T7308's and T7316's, rights? |
| 23:43.46 | dlynes | or was it T7208 and T7216....can't remember |
| 23:43.54 | dlynes | The basic nortel handsets, anyways |
| 23:44.19 | dansaz | can I use asterisk for all internal extensions and VM, but have copper POTS lines out? |
| 23:44.30 | JT | copper pots is poop |
| 23:44.32 | JT | consider PRI |
| 23:44.35 | jameswf | ummm yes |
| 23:44.39 | xacatecas | tainted_, on the phone, they usually have a web-interface. |
| 23:44.47 | jameswf | ~zao |
| 23:44.49 | xacatecas | at least, every single phone i've worked with so far. |
| 23:44.50 | jameswf | ~zap |
| 23:44.50 | jbot | ACTION zaps jameswf in the head with an overvoltaged tazer |
| 23:45.01 | jameswf | don't taze me bro |
| 23:45.09 | [TK]D-Fender | dlynes, correct, basc nortel, more modern Avaya's, etc |
| 23:45.11 | xacatecas | dansaz, yes, you can. |
| 23:45.28 | dansaz | what equipment would I need to interface my punchdown block with my asterisk server? |
| 23:45.41 | jameswf | I was in IKEA doing the minimall song :)) |
| 23:45.51 | [TK]D-Fender | dansaz, You would use * for EVERYTHING. You might use SIP (VoIP) for say jsut your phones, and you'd use a special interface card to allow you to plug in your copper lines. |
| 23:46.00 | [TK]D-Fender | dansaz, but * would control everything |
| 23:46.12 | jameswf | dansaz: a Zap card specifics depend on line count |
| 23:46.57 | dansaz | what sort of card is used to accept copper lines? |
| 23:47.03 | jameswf | how many |
| 23:47.07 | dansaz | 13 |
| 23:47.38 | jameswf | dansaz: http://rhinoequipment.com/R24FXXcard.html |
| 23:48.08 | [TK]D-Fender | dansaz, that IS a lot of copper... you should investigate getting a partial PRI... |
| 23:48.30 | jameswf | agrees around 13 is time to look at t1 |
| 23:48.42 | dansaz | We have an existing PRI from a CLEC. Qwest has made it much cheaper to just get copper |
| 23:48.44 | JT | once you go pri |
| 23:48.49 | JT | you'll never go back ;) |
| 23:48.55 | [TK]D-Fender | jameswf, I seriouly start at about 8.. |
| 23:48.58 | dansaz | about $200 per month cheaper |
| 23:49.17 | JT | dansaz: than a partial pri? |
| 23:49.21 | dansaz | yes |
| 23:49.22 | JT | not a full 23 channels |
| 23:49.23 | [TK]D-Fender | dansaz, Well if you realy want to live with the difference in service, its your call. |
| 23:49.33 | jameswf | yeah if your at like $50+/ea 8 is good |
| 23:49.54 | dansaz | what is the benefit of T1 over copper? |
| 23:50.03 | JT | unlimited DIDs |
| 23:50.08 | JT | proper call signalling |
| 23:50.23 | jameswf | dansaz: speed, control, realestate |
| 23:50.24 | JT | no issues with channels accidentally left off hook |
| 23:50.32 | jameswf | usualy cost |
| 23:50.37 | JT | this is for PRI btw |
| 23:50.42 | JT | CAS T1 is not as good |
| 23:51.08 | dansaz | for a relative telecom noob like me, it seems physical copper lines would be easier to manage |
| 23:51.13 | JT | no way |
| 23:51.18 | JT | one or 2 pairs |
| 23:51.23 | JT | vs 13 pairs |
| 23:51.26 | jameswf | manage 26 wires or 4 wires |
| 23:52.00 | dansaz | after installation, it seems easier to manage |
| 23:52.10 | JT | you can't easily do multiple DIDs with POTS |
| 23:52.18 | jameswf | move/add/change from anywhere in the world including @home in your chonies |
| 23:52.23 | JT | that hasn't been my experience either :) |
| 23:52.27 | dansaz | sorry, what is a DID? |
| 23:52.40 | JT | also, digital has no echo, in your leg (there may be echo from far end analogue) |
| 23:52.44 | jameswf | ~did |
| 23:52.45 | jbot | extra, extra, read all about it, did is Direct Inward Dialing, or just a phone number |
| 23:52.45 | JT | direct inward dialling |
| 23:53.05 | JT | you can have hundreds or more DIDs that will work on any free channel on a pri |
| 23:53.26 | JT | all phone calls have a from and a destination number in the pri signalling |
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| 23:53.43 | dansaz | so with 13 copper lines, I'm limited to 13 DIDS |
| 23:53.47 | JT | not even |
| 23:53.53 | jameswf | yes |
| 23:53.54 | JT | if you have a hunt group |
| 23:53.58 | jameswf | well np |
| 23:54.02 | jameswf | *no |
| 23:54.05 | JT | you will need one pool for the hunter number |
| 23:54.10 | JT | and the rest dedicated to a did each |
| 23:54.12 | jameswf | you can do distinctive ring |
| 23:54.23 | jameswf | and have ghost numbers |
| 23:54.26 | JT | and you can do dtmf after ringing |
| 23:54.32 | JT | but all major painful hacks |
| 23:54.49 | *** join/#asterisk mitcheloc (n=mitchelo@216.207.245.1) |
| 23:55.09 | JT | s/hunter/hunt/ |
| 23:55.48 | SteveTotaro | e1 would be so nice for the US |
| 23:55.48 | JT | imho analogue is only good for an emergency line, faxes and modems |
| 23:55.48 | tainted_ | it's not unlimited dids |
| 23:55.52 | JT | i try to avoid them |
| 23:55.58 | JT | tainted_: ? |
| 23:56.11 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:56.36 | tainted_ | most carriers are thrifty to give out dids, some even charge |
| 23:57.01 | tainted_ | it's unlimited in theory, or if u have deep pockets |
| 23:57.02 | lanning | att is 5$/month for batch of 20 DIDs. |
| 23:57.04 | JT | tainted_: yes, they usually charge in 100 number blocks |
| 23:57.04 | *** join/#asterisk pakipenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 23:57.04 | SteveTotaro | e1 and metric system I am all for adopting |
| 23:57.17 | JT | tainted_: however, techincally there is no limitations to DIDs |
| 23:57.39 | tainted_ | anyone use pacwest? |
| 23:57.41 | SteveTotaro | for just 10 DIDs (toll free) I pay a buck a month each |
| 23:57.50 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 23:57.58 | JT | it's much cheaper and simpler than running a new phone line for every single number... |
| 23:57.59 | lanning | JT, 3*3*4 (for US dialing...) :) |
| 23:58.15 | JT | lanning: heh |
| 23:58.17 | dansaz | if you are using hunt groups, are all the rollover lines assigned static DIDS or do they change? |
| 23:58.19 | SteveTotaro | but the t1 is only $150 so it's pretty nice |
| 23:58.42 | JT | dansaz: there is no such thing as a hunt group on a single pri or partial pri |
| 23:58.43 | dansaz | or does it even matter with a t! |
| 23:58.58 | SteveTotaro | a trunk group but not a hunt group |
| 23:59.05 | JT | dansaz: the call is just allocated the next allocated channel, it's all transparent to the user |
| 23:59.15 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 23:59.18 | bkruse | SteveTotaro: full voice t1 for 150? |
| 23:59.26 | JT | s/allocated channel/available channel/ |
| 23:59.49 | SteveTotaro | bkruse: yes but they are LD circuits |
| 23:59.50 | JT | it's just a tdm timeslot really |
