00:08.19 | Gio__ | nobody ? |
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00:30.31 | Get_The_Fish | does anyone know of a web frontend to the AstDB? |
00:31.13 | bcrisp | gasps |
00:31.21 | bcrisp | ~roulette |
00:31.21 | infobot | ACTION watches bcrisp pull the trigger: Click! |
00:31.33 | Get_The_Fish | lol |
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00:31.55 | Get_The_Fish | I want users to be able to look at the entries in the blacklist, add/change/delete |
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00:40.28 | sahafeez | anyone have a recommendation for someone that can provide dailtone via iax in the uk and the usa |
00:40.38 | hardwire | teliax |
00:40.39 | hardwire | :P |
00:41.12 | sahafeez | numbers in the uk? |
00:41.36 | sahafeez | yah no |
00:41.52 | hardwire | you may have to split the load |
00:43.20 | drfreeze | I've got phones in two offices connected by a Pt2Pt (T1). |
00:43.35 | drfreeze | The phones in the main office have been setup for a month now and work fine |
00:43.50 | voipmonk | ok... |
00:43.51 | drfreeze | The pt2pt was connected today and the offices can call each other fine |
00:44.11 | drfreeze | But, the phones in the terminal office can't call out. I get the error: Channel 0/2, span 1 got hangup request, cause 41 |
00:44.41 | drfreeze | The phones in the far office use the same dialplan too |
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00:46.32 | gwav8or | neurosys you on? |
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00:47.52 | hardwire | heh |
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00:58.50 | ManxPower-work | Cause 41 is "Temporary Failure: The call was disconnected due to a network failure. The network is not functioning correctly and that the condition is not likely to last a long period of time; e.g. the user may wish to try another call attempt almost immediately. " |
01:00.55 | hardwire | we should train infobot on those |
01:01.14 | hardwire | infobot: hug |
01:01.14 | infobot | ACTION hugs hardwire |
01:01.17 | hardwire | !!! |
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01:14.55 | bahjons | does anyone know where I can find documentation on asterisk realtime? namely database structures |
01:24.20 | [TK]D-Fender | bahjons: in the tarball |
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01:25.10 | drmessano | IN TEH TARPITS |
01:25.12 | drmessano | MR BOND |
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01:26.54 | bahjons | [TK]D-Fender: yea the only thing that references the realtime is docs/realtime.txt, docs/extconfig.txt, configs/sip.conf, configs/extconfig.conf, but nothing has the database structure. |
01:28.28 | bahjons | [TK]D-Fender: I'm still working with this state_interface garbage, but can't get call-limit field in realtime to work. I've determined that is the cause of the problem. It's the only different in settings between realtime and conf. |
01:29.36 | [TK]D-Fender | bahjons: Should map fine if the name matches in your DB AFAICT |
01:30.49 | bahjons | <PROTECTED> |
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01:52.06 | steve745 | can anyone help me with getting sql data like agent status available or unavailable??? |
01:52.20 | steve745 | what db would i query and what field |
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02:20.23 | ManxPower-work | steve745: Are you running agents in Realtime? |
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02:48.35 | ChannelZ | This again? |
02:49.32 | jblack | I dare you to watch this: http://www.getonmyhorse.com/ |
02:49.46 | jblack | NSFW, NSFC, NSFS, NSFL |
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02:51.29 | jblack | Shut up woman, get on my horse! |
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02:54.53 | ChannelZ | Hmm. |
02:55.15 | ChannelZ | I like weebl and bob but that's just kind of too random to be funny |
02:56.55 | jblack | This is fantastic. http://www.weebls-stuff.com/toons/Meow/ |
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03:01.02 | kerx | Hi, I'm receiving lots of messages that look like this: "WARNING[20903]: chan_sip.c:1804 __sip_xmit: sip_xmit of 0xd3ee790 (len 959) to 192.168.1.141:5060 returned -1: Operation not permitted" |
03:01.50 | kerx | I'm not sure what to make of this, however I know the phones are not able to get called |
03:03.14 | loathsome | kerx: are you running asterisk as root? |
03:03.23 | kerx | loathsome, Yes |
03:03.28 | [TK]D-Fender | jblack: That was..... retarded |
03:03.30 | loathsome | ok, do you have selinux enabled? |
03:03.40 | ChannelZ | sexlinux |
03:03.59 | jblack | Which one? The really retarded one, or the incredibly retarded one? |
03:04.02 | kerx | selinuxenabled ; echo $? |
03:04.04 | kerx | I receive a 1 |
03:04.18 | kerx | i have my SELINUX=disabled |
03:05.09 | kerx | man page tells me selinuxenabled returns a 1 if disabled |
03:05.25 | kerx | -rw-r--r-- 1 root root 24576 Dec 28 18:57 astdb |
03:09.58 | kerx | Any suggestions? |
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03:27.44 | jesse098765 | Hi Everyone. Anyone know why my MP3Player won't play? The .wavs works fine, but there seems to be a problem. I'm new.... http://pastebin.ca/1730211 |
03:28.33 | Tech_Travis | Does * read the extensions.conf file sequentially? I have an existing dialplan with 714-555-1212 coming in with a couple of IVR options, 1 for tech-support going to queue1, 2 for sales etc. Now I need to add a second number 619-555-1212 but keep the same option numbers such as 1 for techs, however it needs to go to tech-support queue2. Will this approach work if I just add the new number and menu options after all of the origina |
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03:31.17 | ChannelZ | Tech_Travis: it's not sequential in so much as it will 'fall through' to a totally different extension |
03:32.23 | ChannelZ | but depending on your setup you could just set a channel variable with the name of the queue you want for each number, and centralize the main IVR into a different different extension that the other two jump to for instance |
03:32.51 | [TK]D-Fender | Tech_Travis: Thats what CONTEXTS are for. |
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03:36.48 | Tech_Travis | ChannelZ: I'm okay with changing around existing dialplan to centralize things, just having a hard time wrapping my mind around how best to set this up so I won't need to redo it each time a new number needs to be created. |
03:37.47 | Tech_Travis | [TK]D-Fender: I wasn't sure that contexts could be used before the call actually comes into the box, I mistakenly believed that the call needed to be picked up before I could use contexts for routing. |
03:38.22 | [TK]D-Fender | Tech_Travis: Every line in extensions.conf is an opportunity to go somewhere ELSE |
03:38.32 | [TK]D-Fender | Tech_Travis: and there is no such thing as "routing". |
03:38.50 | [TK]D-Fender | Tech_Travis: Another term you can trown in the dumpster |
03:38.54 | [TK]D-Fender | throw* |
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03:40.03 | Tech_Travis | [TK]D-Fender: dumpster noted. what is the correct term be for sending things elsewhere in the dialplan? |
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03:41.35 | [TK]D-Fender | Tech_Travis: No magic term.... however Goto() as an app is reference enough. its always a question of what command you use. |
03:41.55 | [TK]D-Fender | Tech_Travis: Then there is the concept of INCLUDE-ing contexts in another. |
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04:05.47 | drfreeze | Anyone have any luck with creating a call file that can call to a channel other than Sip? |
04:06.22 | drfreeze | If I send it to a Dahdi channel, I get: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) |
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04:10.06 | [TK]D-Fender | drfreeze: It works with anything you'd Dial() |
04:10.17 | [TK]D-Fender | drfreeze: So show us everything relevent and we'll tell you why |
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04:14.48 | hardwire | meh |
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04:18.10 | drfreeze | [TK]D-Fender: http://pastie.textmate.org/private/agyhfgmdsfiftnros8cqfg |
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04:23.51 | pagec | in ael is there a way to dial a number in another context with using include? |
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04:32.57 | [TK]D-Fender | drfreeze: Show my your configs, and your failed attempt |
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04:38.17 | [TK]D-Fender | pagec: There is this mircale app called Goto() you could try.... |
04:49.40 | drfreeze | [TK]D-Fender: http://pastie.textmate.org/private/ompn19p9mwgawxi81fvrg |
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04:57.45 | [TK]D-Fender | drfreeze: Cause No. 21 - call rejected. This cause indicates that the equipment sending this cause does not wish to accept this call. although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. The diagnostic... |
04:57.47 | [TK]D-Fender | ...field may contain additional information about the supplementary service and reason for rejection. |
04:57.59 | [TK]D-Fender | drfreeze: Seems pretty clear to me. Call file is fine |
05:01.43 | drfreeze | [TK]D-Fender: seems as unclear as ever to me |
05:01.55 | drfreeze | there is no reason that I can see why the call should be rejected |
05:02.03 | drfreeze | the internal context dials out all the time |
05:02.09 | [TK]D-Fender | drfreeze: Where are you that a 7 digit number is legal to dial? |
05:02.20 | drfreeze | DAHDI/g1/number work just fine, no rejection |
05:02.38 | [TK]D-Fender | drfreeze: Show me you calling that same number normally. |
05:02.49 | drfreeze | work for 5551212, 5125551212 and 15125551212 |
05:02.58 | drfreeze | but, all those cases fail in the call file |
05:03.14 | [TK]D-Fender | drfreeze: SHOW ME |
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05:05.26 | drfreeze | exten => _NXXXXXX,n,Dial(DAHDI/g2/${EXTEN}) |
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05:07.40 | [TK]D-Fender | moves on to more productive matters |
05:11.19 | drfreeze | [TK]D-Fender: not on site right now to make that call locally |
05:11.54 | drfreeze | not sure how to make a call from a sip extension |
05:12.13 | [TK]D-Fender | drfreeze: ..... You don't know how to dial a friggen number on a soft phone? |
05:12.24 | [TK]D-Fender | drfreeze: How long have you been using * now>? |
05:14.54 | drfreeze | don't have a softphone installed at the moment |
05:17.17 | [TK]D-Fender | drfreeze: Eitehr way, you're getting an ISDN rejection and aren't looking at PRI debug. Your number doesn't look legit in that jsut about ever NA PRI I've ever seen tend to demand 10-digit dialing. |
05:17.45 | [TK]D-Fender | drfreeze: Enable PRI debug and see if it tells you anything more |
05:19.31 | loathsome | in all my 7-digit dial sections I have to prefix a local area code. |
05:20.04 | loathsome | in fact, i have to prefix the 1 as well. It needs 11-digit dialling. |
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05:56.36 | drfreeze | [TK]D-Fender: I installed a sip phone |
05:57.14 | drfreeze | It is behaving the same as the call file. Somehow, the sip phone and the polycom phones are different, even tho both use the same dialplan |
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05:59.55 | [TK]D-Fender | drfreeze: Perhaps its the CALLERID that it doesn't like. Perhaps you should try actually providing one with your call file. |
06:00.09 | [TK]D-Fender | drfreeze: PRI's tend to expect you to announce yourself properly |
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06:06.14 | drfreeze | [TK]D-Fender: seems like adding the caller id fixed the problem |
06:06.54 | Get_The_Fish | can anyone point me to some documentation on the AMI MXML interface? Havent been able to find much on this... |
06:07.56 | [TK]D-Fender | drfreeze: Good to hear... PRI debug may have alluded to this if you looked at it when you started all of this, and is a factor that should never be overlooked |
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06:11.02 | drfreeze | ok, now I need to get that callerid set in the call file |
06:13.55 | Get_The_Fish | anyone know why I would get a "permission denied" immediately following a successful login using the mxml interface |
06:20.05 | drfreeze | [TK]D-Fender: added callerid to the call file |
06:20.14 | drfreeze | it makes the call but my phone never rings |
06:20.14 | drfreeze | http://pastie.textmate.org/private/jo5kic1fdcezrysjirjvq |
06:21.11 | Get_The_Fish | nevermind on that, I apparently cant read |
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06:22.34 | [TK]D-Fender | drfreeze: And All I see is you running out of dialplan to execute |
06:22.58 | drfreeze | [TK]D-Fender: ok, duh, figured it out |
06:23.09 | drfreeze | had the '1' in the number from a previous trial |
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06:39.03 | BugKhaM | Which variable in asterisk holds the client(either SIP or IAX)'s IP address? |
06:48.00 | Get_The_Fish | BugKham, function SIPPEER(peername,ip) |
06:48.36 | Get_The_Fish | does anyone know what the sipregs realtime family does? |
06:52.16 | BugKhaM | Get_The_Fish, does it work for agi? |
06:52.42 | Get_The_Fish | should, but I dont know... |
06:52.42 | Get_The_Fish | http://www.asterisk.org/docs/asterisk/trunk/functions/sippeer?type=functions&value=SIPPEER |
06:52.49 | Get_The_Fish | (I love that page) |
06:56.01 | BugKhaM | Get_The_Fish, thanks |
06:56.41 | Get_The_Fish | np |
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07:03.32 | shamelessn00b | channelz |
07:03.42 | shamelessn00b | Hi guys |
07:04.10 | shamelessn00b | hey regarding that asterisk server stress testing, thiis server is pissing me off, its handling about 300 SIP calls |
07:04.35 | shamelessn00b | and it shows 20 percent CPU utilization at upto 100 dhadi calls |
07:04.51 | shamelessn00b | but just as I reach 101-103 calls CPU shoots up to 100 percent |
07:05.02 | shamelessn00b | and all hell brakes lose |
07:05.16 | shamelessn00b | using asterisk 1.6.2.0 |
07:05.29 | shamelessn00b | ppl saying I should switch to 1.6.1.12 |
07:05.54 | shamelessn00b | dunno if its asterisk to blame or DAHDI/wanpipe |
07:11.25 | aiksa[LV] | morning |
07:11.26 | aiksa[LV] | :) |
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07:13.13 | shamelessn00b | morning :) |
07:14.31 | aiksa[LV] | shamelessn00b: could be related to some other bottleneck |
07:14.53 | aiksa[LV] | lets say - recording for example |
07:15.03 | shamelessn00b | Im landing on the same context |
07:15.11 | shamelessn00b | in SIP calls |
07:15.12 | aiksa[LV] | if at 100 calls you were at 80% of your disk max write speed |
07:15.13 | aiksa[LV] | .... |
07:15.15 | shamelessn00b | and DAHDI calls |
07:15.26 | shamelessn00b | so I doubt it |
07:15.40 | shamelessn00b | the system can manage around 250-300 SIP calls |
07:15.51 | shamelessn00b | lemme try that again |
07:16.03 | shamelessn00b | gimme like 4-5 mins |
07:16.26 | BugKhaM | shamelessn00b, I'm using asterisk 1.2 + zaptel on my P3 box and can handle around 80 calls thru ISDN PRI |
07:17.27 | aiksa[LV] | shamelessn00b: do you have sw echo canc on those dahdi callls? |
07:17.36 | aiksa[LV] | this could be another issue |
07:17.50 | aiksa[LV] | oslec is rather resource hungry if that is your choice |
07:17.52 | shamelessn00b | I was testing on P4 dual core 3 ghz with 1 gb ram |
07:18.01 | shamelessn00b | and xeon quad core 2 ghz |
07:18.05 | shamelessn00b | 2 gb ram |
07:18.15 | shamelessn00b | both crashing after exactly 100 calls |
07:18.33 | shamelessn00b | before 100 they show minimal loads |
07:18.39 | shamelessn00b | like 20 30 percent CPU |
07:18.52 | shamelessn00b | except for when the calls are actually being set up |
07:18.52 | aiksa[LV] | and 15 min load AVG of what? |
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07:19.08 | shamelessn00b | around the same |
07:19.16 | shamelessn00b | they are constant at 20 percent of CPU |
07:19.17 | aiksa[LV] | 0.2 that woudl be |
07:19.21 | shamelessn00b | yeah |
07:19.26 | aiksa[LV] | ok |
07:19.47 | shamelessn00b | no idea why this is happening just as I reach above 100 calls |
07:21.41 | shamelessn00b | 100 DAHDI calls |
07:21.43 | shamelessn00b | ** |
07:25.21 | tzafrir_laptop | shamelessn00b, system crashes? or just hangs? |
07:25.29 | shamelessn00b | hangs |
07:25.33 | shamelessn00b | indefinately |
07:25.38 | shamelessn00b | doesnt process any call |
07:25.42 | shamelessn00b | SIP or DAHDI |
07:25.47 | tzafrir_laptop | hangs: makes sense. No more CPU cycles available |
07:25.54 | shamelessn00b | and the DAHDI card start showing error messages |
07:26.03 | shamelessn00b | I frame out of sync |
07:26.03 | tzafrir_laptop | what DAHDI device? What echo canceller? |
07:26.11 | shamelessn00b | hardware |
07:26.15 | shamelessn00b | sangoma |
07:26.18 | ChannelZ | or your PCI bus is going titsup |
07:27.02 | aiksa[LV] | PCI bus should be able to handle more than 100 calls |
07:27.18 | aiksa[LV] | if its not maxed out by other things |
07:28.06 | tzafrir_laptop | shamelessn00b, start adding calls steadily, and keep an eye on 'top' while you do that |
07:28.17 | shamelessn00b | yeah I did that |
07:28.23 | shamelessn00b | tried with 1 call a sec |
07:28.32 | shamelessn00b | that waits for 600 secs |
07:28.38 | shamelessn00b | lands in the remote context |
07:28.46 | shamelessn00b | that runs some AGIs |
07:28.59 | shamelessn00b | 1 call/sec 100 times |
07:29.07 | shamelessn00b | (did it using a script and a call file) |
07:29.13 | tzafrir_laptop | Also note that the setup of the call typically takes more cpu time than an actual "steady state" |
07:29.20 | shamelessn00b | yeah |
07:29.22 | shamelessn00b | I mentioned |
07:29.23 | shamelessn00b | that |
07:29.31 | tzafrir_laptop | try leaving e.g. 3 seconds or more between new calls |
07:29.43 | shamelessn00b | its handling 1 call/sec |
07:29.48 | shamelessn00b | and 100 calls/sec |
07:29.58 | shamelessn00b | its not handling 101-103 |
07:30.01 | shamelessn00b | callsin total |
07:30.12 | shamelessn00b | which pisses me off |
07:30.39 | tzafrir_laptop | you typically hit a wall somewhere when you run out of free cpu cycles |
07:30.48 | shamelessn00b | well I didnt really try 100 calls /sec |
07:30.57 | shamelessn00b | I tried 10 calls/sec 10 times |
07:31.08 | shamelessn00b | somewhere around 12 calls/sec |
07:31.13 | shamelessn00b | it gets messy |
07:31.21 | shamelessn00b | but recovers |
07:31.31 | shamelessn00b | it doesnt recover if I hit more than 100 calls |
07:45.36 | aiksa[LV] | tzafrir_laptop: btw there is a nice alternative for top - htop |
07:46.13 | tzafrir_laptop | aiksa[LV], but htop takes more cpu cycles :-) |
07:47.47 | aiksa[LV] | tzafrir_laptop: yeah it does, but gives more insight |
07:48.09 | aiksa[LV] | whats happening across the cores etc. |
07:48.10 | tzafrir_laptop | for the global load? not really |
07:48.49 | tzafrir_laptop | press '1' in top to toggle multi-cpu view |
07:48.58 | aiksa[LV] | wow. thanks |
07:48.59 | ChannelZ | and 'I' |
07:49.40 | aiksa[LV] | I learened something new today |
07:50.07 | aiksa[LV] | and perhaps top also knows how to cascade processes and scroll? |
07:50.33 | shamelessn00b | im using htop :P |
07:51.10 | shamelessn00b | hey can anyone tell me how to set up sip calls on local system using call files |
07:51.14 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.69) |
07:52.08 | aiksa[LV] | just as DAHDI calls, just change channel specification to SIP/something |
07:52.44 | shamelessn00b | Channel: SIP/3002 |
07:52.46 | shamelessn00b | Extension: 1001 |
07:52.48 | shamelessn00b | Context: default |
07:52.49 | shamelessn00b | Priority: 1 |
07:53.08 | shamelessn00b | im on the same system and originating calls that land on the same system |
07:53.24 | shamelessn00b | 3002 is a SIP account I registered on that sysem |
07:53.26 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
07:53.34 | shamelessn00b | I just want to eliminate dahdi for a while |
07:55.13 | shamelessn00b | but not working |
07:55.15 | shamelessn00b | :F |
07:55.42 | aiksa[LV] | where does calls to SIP/3002 go? |
07:56.11 | aiksa[LV] | 3002 is an entry in sip.conf right? |
07:56.28 | aiksa[LV] | Does it have registered peer associated with it? |
07:57.04 | benngard | any knows who "may213" is, if he can be reached here or at mail |
07:58.12 | shamelessn00b | yes |
07:58.23 | shamelessn00b | how do I connect it to the default context |
07:58.25 | shamelessn00b | ? |
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08:02.55 | aiksa[LV] | well I would make two entries in my sip conf |
08:03.03 | aiksa[LV] | one for peer and one for user |
08:03.12 | aiksa[LV] | with the same credentials |
08:04.02 | aiksa[LV] | so when routing a call to peer it would be picked up user |
08:04.20 | aiksa[LV] | then you have to define a context for a receiving party |
08:04.37 | aiksa[LV] | and within that context simply add an s extension |
08:04.57 | aiksa[LV] | with a) Answer and b) Wait(THE_LENGHT_OF_CALL) |
08:05.53 | aiksa[LV] | c) Hangup |
08:06.18 | aiksa[LV] | smth. like that |
08:06.36 | aiksa[LV] | a loopback call to describe in more simple terms |
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09:08.21 | ChannelZ | hmm anyone know of any free faxback services off the top of their head so I can test a fax receive? |
09:12.18 | sawgood | I just saw one tonight ... give me a second |
09:14.25 | sawgood | I searched on voip-info.org for ENUM stuff, and in the list was a URL for free faxes to US and Canada |
09:14.32 | sawgood | I can't recall it right off the top of my head ... sorry |
09:14.38 | ChannelZ | I think I found one, http://faxzero.com/ |
09:16.12 | sawgood | Yeah, I have an account with them too ... |
09:16.27 | sawgood | very good for testing SIP and PSTN faxes at customers sites |
09:18.39 | ChannelZ | this is PSTN, I just got a bunch of fax voicemails and something wierd is happening in my fax detection in 1.6 |
09:19.43 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
09:20.17 | ChannelZ | well wtf |
09:22.34 | ChannelZ | -- Redirecting DAHDI/3-1 to fax extension |
09:22.40 | ChannelZ | -- Sent into invalid extension '?' in context 'incoming' on DAHDI/3-1 |
09:24.07 | *** join/#asterisk vally (n=kvirc@217.243.245.34) |
09:25.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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09:47.41 | ChannelZ | hmm ok anyone know what the frequency of the initiate tone on a sending fax machine uses? one article I found suggests 2100hz but that sounds too high |
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09:49.43 | shamelessn00b | ChannelZ: whats chan_woomera |
09:49.45 | shamelessn00b | ?? |
09:49.59 | shamelessn00b | also, Im suck withthis shitsux stress testing |
09:50.28 | shamelessn00b | my system is able to land `250 SIP calls on a context |
09:50.53 | shamelessn00b | but if I make dahdi calls to taht context load averages at 20 percent till 100 calls |
09:51.06 | shamelessn00b | then as I increase number of calls beyond 100 the cpu goes crazy |
09:51.19 | shamelessn00b | all cores on 100 and no calls are entertained |
09:57.09 | ChannelZ | I have no idea, chan _woomera ?? |
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10:03.39 | benngard | i have some aastra rfp32 running h.323, any1 knows if they can be converted to sip? |
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10:14.32 | *** join/#asterisk mykhyggz (n=col@evolone.org) |
10:16.04 | shamelessn00b | yeah chan_woomera |
10:16.11 | shamelessn00b | I read its some alternate for dahdi |
10:17.13 | ChannelZ | "The chan_woomera channel driver allows the Asterisk IP-PBX to act as a Woomera client." |
10:17.17 | ChannelZ | I've never heard of Woomera |
10:18.21 | Chainsaw | ChannelZ: Initiating tone can mean two things. "CallED" is 2100hz, "CalliNG" is 1100hz. |
10:18.35 | ChannelZ | Chainsaw: yeah it appears to be 1100hz |
10:18.56 | ChannelZ | fax detection is borked in >1.6.0.5 |
10:19.20 | Chainsaw | ChannelZ: Seems to work for me on a TDM400P in 1.6.1.12 |
10:20.07 | ChannelZ | hmm try it a few times. I got like 6 fax voicemails, and looking at the console get what I pasted awhile back, sending it to extension '?' (where ? is actually a strange non-ascii character) |
10:20.35 | ChannelZ | So I used one of those faxback things to test, and that actually went through. But all subsequent tests are failing |
10:21.43 | ChannelZ | cutting out the middleman just calling in and playing a 1100hz tone to trigger, and it keeps coming up with this wierd extension |
10:23.17 | Chainsaw | ChannelZ: Provided I patch the Asterisk core so the BT automated linetest doesn't knock my line out every night... faxing works forever. |
10:23.53 | Chainsaw | ChannelZ: You may find this document helpful, at any rate: http://telecom.tbi.net/fax-call.htm |
10:23.59 | *** join/#asterisk Dovid (i=David_M_@91.205.155.29) |
10:24.17 | ChannelZ | I think I just discovered the trigger |
10:25.03 | ChannelZ | If the tone comes in during a Background() it's barfing the extension. If the tone comes in before or after (in my case 'after' is a WaitExten()) it works |
10:25.23 | *** join/#asterisk ivan_paes (n=paes@187.7.160.188) |
10:26.12 | Chainsaw | ChannelZ: Yes, don't try to listen for DTMF when detecting a fax. |
10:26.59 | ChannelZ | well if I knew I was listening for a fax I wouldn't need fax detection |
10:27.59 | *** join/#asterisk DND (n=arabia@94.200.7.26) |
10:28.09 | DND | hi guys, what codec does x-lite use? |
10:28.15 | Chainsaw | DND: Whichever one you tell it to. |
10:28.56 | DND | hmm i just thought that x-lite only uses ulaw |
10:28.58 | Chainsaw | ChannelZ: The documentation I linked you to gives you the time you need for efficient fax detection. I'd say roughly 9 seconds to be safe. |
10:29.30 | Chainsaw | ChannelZ: You can do whatever DTMF-related thing you want to do afterwards, but you need to get the faxes shunted away to your fax backend first. |
10:30.15 | DND | Chainsaw, i dont see any dropdown to choose codec on x-lite |
10:30.20 | DND | this is the free version |
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10:30.45 | Chainsaw | DND: The cheapskate version is ulaw-only, yes. |
10:30.50 | Chainsaw | DND: The proper one is not. |
10:31.47 | DND | hmm so eyebeam is the one that has most codecs |
10:31.47 | DND | x-lite free is ulaw |
10:32.12 | Chainsaw | My interest in proprietary software is low. But yes, that sounds correct. |
10:32.14 | ChannelZ | Chainsaw: this is a bug.. the point is, * is detecting the 1100Hz tone just fine, even says it's redirecting, but then does so to an invalid extension |
10:32.38 | Chainsaw | ChannelZ: Because you're listening for DTMF at the same time, or too soon afterwards. |
10:32.46 | Chainsaw | ChannelZ: Shunt your fax *away* before you listen for DTMF. |
10:32.49 | ChannelZ | DND: Zoiper has a few more; GSM, ulaw, alaw, speex, something else.. |
10:33.15 | shamelessn00b | channelz do you have some time, I need help on something very basic |
10:33.37 | ChannelZ | Chainsaw: nevermind you're not getting what is going on |
10:33.54 | Chainsaw | ChannelZ: That must be it, yes. Good luck. |
10:34.18 | ChannelZ | shamelessn00b: I really need to get to bed but based on what you've asked earlier I can't help you anyway with this system overload issue |
10:34.43 | ChannelZ | Chainsaw: https://issues.asterisk.org/view.php?id=16050 |
10:35.41 | DND | Chainsaw, can you suggest a free or open source softphone that requires less bandwidth? i read that g711 requires 80kbit both ways? |
10:35.48 | shamelessn00b | well, I want to take dahdi out of the equation |
10:35.56 | shamelessn00b | and make SIP calls to taht context |
10:35.56 | Chainsaw | DND: Isn't there a Windows port of Ekiga these days? |
10:36.10 | shamelessn00b | Im using call files to generate the calls |
10:36.25 | shamelessn00b | I have 2 SIP extentions defined on the target system |
10:36.35 | shamelessn00b | and the context answers on extention 1001 |
10:36.42 | shamelessn00b | the SIP accounts are 3001 and 3002 |
10:37.08 | shamelessn00b | so I want to make a call from 3001 to 3002 that plays the context defined onextension 1001 |
10:38.20 | tzafrir_laptop | shamelessn00b, the simplest way to originate calls is with the 'originate' CLI command |
10:38.33 | shamelessn00b | 100calls |
10:38.35 | shamelessn00b | or more |
10:38.43 | shamelessn00b | thats why im going for a call file |
10:38.49 | tzafrir_laptop | It's a bit limited (vs. call files and the manager interface), but it's way simpler to script |
10:38.58 | drmessano | X-Lite has GSM, Speex, iLBC, and ULAW |
10:38.59 | shamelessn00b | I made a simple script |
10:39.07 | drmessano | GSM or Speex work fine |
10:39.21 | shamelessn00b | that renames copies the call file to /var/spool/asterisk/outgoing/ |
10:39.23 | tzafrir_laptop | originate SIP/whatever extension 123456@test-context |
10:40.02 | shamelessn00b | using call files |
10:40.04 | shamelessn00b | :/ |
10:40.10 | DND | drmessano, how will i define GSM on x-lite? or i'll just retrict it in extensions page? |
10:40.20 | shamelessn00b | call files have thier own syntax |
10:40.22 | drmessano | Go into the options and look man |
10:40.33 | drmessano | Enabled Codecs |
10:40.37 | shamelessn00b | http://www.the-asterisk-book.com/unstable/call-file.html |
10:40.38 | drmessano | Audio |
10:41.06 | DND | yup got it. i didnt know there was an advanced button underneath |
10:41.08 | DND | thanks |
10:42.21 | shamelessn00b | ChannelZ: I made this call file |
10:42.24 | shamelessn00b | Channel: SIP/3002 |
10:42.26 | shamelessn00b | Extension: 1001 |
10:42.28 | shamelessn00b | Context: default |
10:42.30 | shamelessn00b | Priority: 1 |
10:42.34 | *** join/#asterisk Omorika (n=omorika@193.198.31.85) |
10:43.07 | ChannelZ | and? |
10:43.59 | shamelessn00b | it gives me error |
10:44.06 | shamelessn00b | errors |
10:44.09 | ChannelZ | which says... |
10:44.10 | shamelessn00b | doesnt make any calls |
10:44.13 | shamelessn00b | wait |
10:45.40 | shamelessn00b | [Dec 29 10:44:29] NOTICE[10007]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
10:45.42 | shamelessn00b | [Dec 29 10:44:29] ERROR[9850]: app_queue.c:1101 device_state_cb: Received invalid event that had no device IE |
10:46.11 | shamelessn00b | im sitting on the same machine calling the samemachine |
10:46.55 | ChannelZ | well if I had to guess SIP/3002 isn't configged right or something |
10:47.17 | ChannelZ | or it's a single call device and you have it calling it's self and it's throwing an error or I-dont-know-what |
10:47.57 | shamelessn00b | <PROTECTED> |
10:47.58 | shamelessn00b | <PROTECTED> |
10:48.00 | shamelessn00b | [Dec 29 10:46:51] NOTICE[10028]: channel.c:3834 __ast_request_and_dial: Unable to request channel SIP/3001 |
10:48.02 | shamelessn00b | [Dec 29 10:46:51] ERROR[9850]: pbx.c:9264 device_state_cb: Received invalid event that had no device IE |
10:48.04 | shamelessn00b | [Dec 29 10:46:51] ERROR[9850]: app_queue.c:1101 device_state_cb: Received invalid event that had no device IE |
10:48.06 | shamelessn00b | [Dec 29 10:46:51] NOTICE[10028]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
10:48.07 | shamelessn00b | thats the whole thing |
10:48.25 | shamelessn00b | when I dial from a sip client like x lite |
10:48.32 | shamelessn00b | authenticated with 3001 |
10:48.36 | shamelessn00b | it works fine |
10:49.01 | ChannelZ | is x-lite still running when you put the call file in? |
10:49.06 | shamelessn00b | no |
10:49.17 | ChannelZ | well then SIP/3001 doesn't really 'exist' then does it |
10:49.35 | shamelessn00b | then I start getting calls on 3001 |
10:49.38 | shamelessn00b | on my ip phone |
10:49.40 | shamelessn00b | :/ |
10:49.49 | shamelessn00b | softphone rather |
10:49.53 | ChannelZ | well yea that's what dialing SIP/xxxx usually does.... |
10:49.58 | ChannelZ | calls things... |
10:50.20 | ChannelZ | You were expecting what to happen instead exactly? |
10:50.49 | shamelessn00b | I want 100 calls to be generated |
10:50.59 | shamelessn00b | dont really want to listen to them |
10:51.01 | ChannelZ | Well then you need 100 things to call. |
10:51.14 | shamelessn00b | 100 instances tof xlite |
10:51.16 | shamelessn00b | :/ |
10:51.44 | shamelessn00b | is there a workaround |
10:51.45 | shamelessn00b | ?? |
10:51.46 | ChannelZ | You can't call "nothing". You're not really testing anything at that point. |
10:52.13 | shamelessn00b | I can put the call on some extension instead |
10:52.15 | shamelessn00b | that just waits |
10:52.17 | shamelessn00b | can I do that |
10:52.19 | shamelessn00b | ?? |
10:52.30 | shamelessn00b | define some context xyz |
10:52.34 | shamelessn00b | wait(6000) |
10:52.39 | shamelessn00b | somethingsimilar |
10:52.39 | ChannelZ | Yes but you have to be calling FROM SOMETHING! |
10:53.53 | ChannelZ | maybe you can hack up something with Local channels, I dunno, you can go read about it: http://www.voip-info.org/wiki/view/Asterisk+local+channels |
10:54.03 | ChannelZ | Me I have to go to bed so I can wake back up in 4 hours and go to work :/ |
10:55.40 | shamelessn00b | I was up since 1 AM last night |
10:55.49 | shamelessn00b | went to sleep at 2 am today |
10:56.01 | shamelessn00b | back at work at about 9 am |
10:56.03 | shamelessn00b | epic |
10:57.56 | shamelessn00b | gnite |
11:05.03 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
11:08.59 | lesouvage | I'm calling an extension that is mend to list the status of all the devices in the cli with ",NoOP(The status of ${NR1002} attached to 1002 is ${DEVSTATE(SIP/${NR1002})})". The output for the device I use to call this extension is "NOT_INUSE" and that is definitly wrong. What do I have to do to get the proper dev status as output? |
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11:58.58 | ManxPower-work | I hate mornings |
11:59.42 | mort_gib | ManxPower-work: Yes, but working late into the night is just much worse! |
12:00.05 | ManxPower-work | mort_gib: Not for me. |
12:00.32 | ManxPower-work | <--- nocturnal |
12:00.50 | mort_gib | :- |
12:00.55 | mort_gib | :-) |
12:01.08 | mort_gib | I get enough of computer eventually! |
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12:57.55 | benngard | i must be stupid |
12:58.18 | benngard | i have a "test" line: |
12:58.20 | benngard | ExecIf($[${LEN(${BLINDTRANSFER})}>0]?NoOp(${BLINDTRANSFER})) |
12:58.46 | benngard | trying to use "CUT" so i just have the SIP/number left |
12:59.11 | benngard | ...?NoOp(SIP/0317998975-000001c0)") in new stack |
12:59.46 | ManxPower-work | I'm not seeing your CUT statement |
13:00.51 | ManxPower-work | ${CUT(${BLINDTRANSFER},-,1)} |
13:01.14 | ManxPower-work | sorry... |
13:01.30 | ManxPower-work | ${CUT(BLINDTRANSFER,-,1)} That should be correct. |
13:03.32 | benngard | lets try |
13:03.46 | ManxPower-work | next time paste the broken statement |
13:04.16 | benngard | ExecIf($[${LEN(${BLINDTRANSFER})}>0]?NoOp(${CUT(${BLINDTRANSFER},-,1)})) that didnt work so minimized it |
13:04.38 | ManxPower-work | as you can see that's not what I pasted. |
13:04.46 | benngard | no i see that |
13:05.08 | ManxPower-work | Cut wants a variables name BLINDTRANSFER not a string ${BLINDTRANSFER} It's an easy mistake to make |
13:07.10 | benngard | thx that did the trick |
13:07.33 | benngard | didnt know that cut wanted a variable |
13:07.43 | benngard | ExecIf($[${LEN(${BLINDTRANSFER})}>0]?NoOp(${CUT(BLINDTRANSFER,-,1)})) gave |
13:07.47 | ManxPower-work | benngard: *nod* It's sort of odd, but I'm sure there's a reason |
13:08.43 | benngard | the correct answer |
13:09.00 | benngard | just se whats happen when i replace noop with a dial then |
13:11.11 | *** join/#asterisk Thorn (n=Thorn@unaffiliated/thorn) |
13:11.27 | *** join/#asterisk alrs (n=lars@46.sub-70-213-201.myvzw.com) |
13:11.37 | Thorn | hello |
13:11.42 | *** part/#asterisk alrs (n=lars@46.sub-70-213-201.myvzw.com) |
13:12.34 | Thorn | how do I write an interval from 22:00 dec. 31st to 08:00 jan. 1st in GotoIfTime()? |
13:12.55 | ManxPower-work | Thorn: I don't think you can do that in one statemnet. |
13:13.25 | ManxPower-work | you would want 22:00 to 23:59 and 0:00 to 08:00 |
13:14.06 | Thorn | ManxPower-work: that was my initial idea too, thanks |
13:14.50 | *** join/#asterisk alrs (n=lars@46.sub-70-213-201.myvzw.com) |
13:16.42 | benngard | hmm, strange i did call 0317998975 from my cell phone, 0317998985 did transfer it to 0317998985 (didnt answer) got the call back and my cell phone "was talking" to 0317998975 again, perfect, but 0317998985 did continue to ring |
13:16.56 | benngard | exten => 0317998985-NOT_INUSE,1,Dial(SIP/0317998985,20,t) |
13:16.57 | benngard | exten => 0317998985-NOT_INUSE,2,ExecIf($[${LEN(${BLINDTRANSFER})}>0]?Dial(${CUT(BLINDTRANSFER,-,1)})) |
13:20.11 | *** part/#asterisk Omorika (n=omorika@193.198.31.85) |
13:20.44 | benngard | spli |
13:20.44 | benngard | split |
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13:30.41 | Dakon | I've just upgraded from 1.4 to 1.6 |
13:31.06 | Dakon | which is the supposed way to get app_rxfax and app_txfax again? |
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13:35.09 | ManxPower-work | Dakon: why not use app_fax? |
13:35.15 | Chainsaw | Dakon: Enabling spandsp support. |
13:36.43 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
13:36.49 | ManxPower-work | Dakon: have you read the UPGRADE*.txt files, which contain all important changes to Asterisk? |
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13:43.17 | voipmonk | yawns |
13:47.35 | ManxPower-work | I guess Dakon didn't really want help. |
13:48.33 | benngard | weird, 985 calls 976 who blind transfer to 975, 975 starts to ring but dont answer call gets back to 976, but no voice between 985 and 976 and 975 continues to ring |
13:49.00 | benngard | <PROTECTED> |
13:49.00 | benngard | <PROTECTED> |
13:49.23 | ManxPower-work | call doesn't "get back" to 976 unless you tell it to. |
13:49.37 | benngard | i do tell |
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13:49.58 | benngard | thats was what i struggled with before |
13:50.18 | Dakon | sorry, was distracted by work |
13:50.21 | benngard | exten => 0317998975-NOT_INUSE,2,ExecIf($[${LEN(${BLINDTRANSFER})}>0]?Dial(${CUT(BLINDTRANSFER,-,1)})) |
13:51.18 | ManxPower-work | dialplan snippets are pretty useless to us |
13:51.34 | ManxPower-work | Dakon: What are the answers to my two questions? |
13:51.38 | benngard | the get back works |
13:52.33 | ManxPower-work | Dakon: come back when you have time to focus on your question |
13:52.35 | Dakon | "was not aware of", yes |
13:52.56 | ManxPower-work | Read those UPGRADE*.txt files. |
13:57.02 | benngard | "Asked to transmit frame type slin" but i have hardcodec alaw everywhere... |
13:58.00 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
14:00.03 | Dakon | where do I find app_fax.c? core asterisk? |
14:00.32 | ManxPower-work | Dakon: yes. but if you don't have spandsp installed you won't be able to select it. Use menuselect. |
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14:00.37 | *** mode/#asterisk [+o angler] by irc.freenode.net |
14:03.36 | ManxPower-work | Now go read the UPGRADE.txt files |
14:04.09 | Dakon | spandsp 0.0.6_pre12 |
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14:04.21 | Dakon | just to note that the UPGRADE files say basically nothing about fax |
14:04.25 | ManxPower-work | read them all since something in, for example, in the UPGRADE12 file may say an application was depricated. It won't me mentioned again when it's removed in 1.6, for example |
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14:06.36 | benngard | ${BLINDTRANSFER} just works for sip, correct? |
14:07.35 | Dakon | how intuitive |
14:09.57 | ManxPower-work | benngard: what does channelvariables.tex tell you about BLINDTRANDFER? |
14:10.17 | ManxPower-work | I think the asterisk.pdf that's usually built with Asterisk will also have that information in it. |
14:12.53 | ManxPower-work | Dakon: I think the entire documentation system in Asterisk is crap. |
14:14.55 | coppice | O'Reilly might disagree :-) |
14:14.58 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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14:15.04 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
14:15.11 | _cgc | hi everyone |
14:15.37 | Dakon | is there any documentation on app_fax? Or can I just switch any app_[rt]xfax to app_fax and expect that working? |
14:16.16 | _cgc | does anyone know when you record a call in asterisk 1.6.1.11 it records it as 2 different files, 1 for each side of the conversation? |
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14:17.05 | _cgc | i want the recording to be as 1 file ${UNIQUEID}.wav, but instead i get ${UNIQUEID}-in.wav and ${UNIQUEID}-out.wav |
14:20.24 | ManxPower-work | _cgc: Use MixMonitor |
14:21.18 | _cgc | <ManxPower-work> Is this new then because I just used Monitor before? |
14:21.30 | ManxPower-work | _cgc: It was new in 1.2 |
14:21.41 | _cgc | ahh lol, ok cool, thanks |
14:21.48 | ManxPower-work | "core show application monitor" and "core show application mixmonitor" |
14:22.25 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
14:22.32 | Katty | guten morgan |
14:22.54 | benngard | must be blind :( i am reading channelvariables.tex... |
14:23.43 | benngard | Asterisk standard channel variables |
14:23.55 | benngard | ${BLINDTRANSFER} The name of the channel on the other side of a blind transfer |
14:24.10 | ManxPower-work | does it mention anything about being SIP specific? |
14:24.53 | Katty | vie heist DU |
14:24.54 | benngard | no but when i did try to transfer a call that came in on a h.323 channel, that parameter was empty |
14:25.18 | benngard | so i thought it was sip only |
14:25.33 | ManxPower-work | Chances are H323 is a special case. |
14:25.43 | Katty | hat jemand deutsch sprechen? |
14:25.46 | benngard | how should i know that? ;) |
14:26.03 | ManxPower-work | because H323 support in Asterisk sucks and nobody uses it if they have any choice in the matter? |
14:26.10 | Katty | mein telefon funktionert nicht |
14:26.43 | benngard | i know that h323 sucks, did got a patch for ooh323 gonna test and recompile later |
14:26.53 | Katty | was kannich tun? |
14:27.22 | Dakon | Stecker rein |
14:27.43 | Katty | hehehehe |
14:27.50 | Katty | plugggggggg it in plug it in! |
14:31.10 | _cgc | ManxPower-work> thanks, that worked :) |
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14:31.58 | tzafrir_laptop | _cgc, moved to sip? |
14:31.58 | ManxPower-work | Looks like we have our usual "at least 5 clients are down because of Verizon problems" again today |
14:32.02 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
14:32.11 | ManxPower-work | How many fsckin' cable cuts can they have in a fsckin' week? |
14:32.11 | Katty | Dakon: es nun, was soll ich tun? |
14:33.29 | ManxPower-work | _cgc: "core show applications" and "core show FUNCTIONS" are your friend. |
14:33.49 | shamelessn00b | HURRRR |
14:33.53 | shamelessn00b | found the fkin error |
14:33.55 | ManxPower-work | BTW, functions themselves are uppercase |
14:34.25 | Dakon | Katty: keine Ahnung |
14:34.44 | ManxPower-work | Sometimes I think Verizon dispatches a tech with an axe a couple of times a week to cause "massive outage" |
14:35.03 | Katty | Dakon: uber die lippen gebracht glanz |
14:35.30 | shamelessn00b | the database server had the connection limit set to 100 |
14:35.40 | shamelessn00b | thats why my processorfkinshot up to 100 percent |
14:35.49 | shamelessn00b | right after it reached the 100 call mark |
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14:35.52 | Katty | Dakon: how do you pronounce Ahnung? |
14:36.16 | ManxPower-work | Ahnung? Sounds like a porn star name. |
14:36.23 | x86 | lol |
14:36.58 | x86 | Katty: du spreche zie Deutsch? |
14:37.00 | Katty | ManxPower-work: http://gangstaname.com/porn_name.php |
14:37.06 | Katty | x86: just a little (= |
14:37.14 | x86 | Katty: nifty :) |
14:37.26 | Katty | probably not enough to get by without a proper translator |
14:37.39 | x86 | Katty: I've learned (a little) german from the guys over in #theplanet (here on this network) |
14:37.42 | Katty | i'd be telling everyone to slowwwwww down |
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14:37.55 | x86 | it's a native german-speaking channel.. |
14:38.02 | x86 | yeah |
14:38.21 | x86 | I can barely read it proper and less than barely write it |
14:38.34 | x86 | like, how the hell am I supposed to know that a tree is male? :) |
14:38.34 | Katty | ManxPower-work: hilarious, i put in Angela and it says my name is Kitty Jam |
14:38.44 | x86 | hahahaha |
14:39.11 | Katty | Angie is Asia Jizz ^_- |
14:39.20 | Katty | x86 is Sweat Darkholer LOL |
14:39.27 | x86 | ROFL |
14:39.34 | x86 | darkholer... wow |
14:39.40 | Katty | manxpower is corporal fuegobutt |
14:39.49 | Katty | ManxPower-work: i didn't know you were a corporal! |
14:42.08 | Katty | i really hate one of my senators. |
14:42.17 | Katty | he keeps voting no on all the important bits. |
14:43.05 | drmessano | <--- BJ Jiggler |
14:43.12 | drmessano | :( |
14:43.21 | Katty | ah ahahhaaha |
14:43.21 | drmessano | That hits too close to home for me\ |
14:43.55 | Dakon | ManxPower-work: I replaces rxfax(...) by ReceiveFAX(...) and txfax(...) by SendFAX(...) and it sort of works |
14:44.29 | Dakon | it crashes when receiving faxes and has strange problem calling my script, but at least the tiff file looks correct |
14:44.40 | drmessano | The PDFs for the Digium apps document the differences well |
14:44.45 | jaytee | I hate words and phrases like "sort of" and "kinda" |
14:45.00 | Katty | i sort of kinda do too |
14:45.05 | jaytee | :-) |
14:45.10 | Katty | cept, sorta kinda |
14:45.14 | drmessano | I'm kinda... slow on that one, thanks Katty |
14:45.55 | Dakon | cu |
14:45.55 | Katty | jaytee's porn star name...wow. |
14:46.00 | Katty | i'm not even gonna type that one here |
14:46.21 | drmessano | Lemme guess... |
14:46.30 | drmessano | Dolph Lundgren |
14:46.44 | benngard | ;) |
14:47.07 | drmessano | That would be jaytee's porn name |
14:47.44 | benngard | == Dick Maxim :) |
14:48.30 | ManxPower-work | Manx Power IS my porn name. |
14:49.53 | Katty | Minx Power. |
14:50.07 | drmessano | wow |
14:50.12 | drmessano | what a difference a space makes: |
14:50.18 | drmessano | leifmadsen -> Corporal Muffmuncher ..... leif madsen -> Humpy Jizz |
14:51.06 | drmessano | Proving once again that real life sucks worse than IRC |
14:52.35 | x86 | heh |
14:52.44 | x86 | corporal muffmuncher... rofl |
14:53.26 | Katty | so many critters in my yard this morning. |
14:53.34 | Katty | you'd think the apocolypse happened |
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14:54.26 | Katty | 5 squirrels so far |
14:54.31 | Katty | and it's only 9! |
14:54.54 | x86 | Katty: they are probably just gather food and stuff for when 12-21-12 rolls around... you know, then end of the world |
14:55.21 | Katty | they probably ARE gathering food for 12/21, but not because it's the end of the world :P |
14:55.30 | ManxPower-work | x86: I'm pretty sure they just know there's free food. |
14:55.42 | Katty | and who doesn't like free food? |
14:55.57 | Katty | infobot: forget Critter Cam |
14:55.57 | infobot | i forgot critter cam, Katty |
14:56.02 | Katty | infobot: forget crittercam |
14:56.02 | infobot | Katty: i forgot crittercam |
14:57.55 | Katty | infobot: critter cam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
14:57.56 | infobot | Katty: okay |
14:58.06 | Katty | infobot: crittercam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
14:58.07 | infobot | okay, Katty |
14:58.53 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
14:59.14 | Katty | a 6th just showed up |
15:00.41 | *** join/#asterisk gme30066 (n=gme@173.160.69.30) |
15:00.43 | Katty | 8 |
15:01.17 | Katty | it's like i'm supporting squirrely welfare. |
15:01.30 | Katty | food stamps for the children. |
15:02.22 | ManxPower-work | Don't you mean "rats with bushy tails welfare"? |
15:02.36 | Katty | rats are smarter than squirrels. |
15:02.57 | Katty | rats are cute. |
15:04.38 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
15:04.42 | Katty | hi gr0mit |
15:04.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
15:04.59 | Katty | hi _ShrikE |
15:05.06 | gr0mit | hi Katty |
15:05.10 | gr0mit | you ok? |
15:05.18 | Katty | well that's debatable. |
15:05.20 | *** part/#asterisk gme30066 (n=gme@173.160.69.30) |
15:05.24 | gr0mit | oh dear! |
15:05.31 | Katty | :P |
15:05.44 | gr0mit | looks at the pouring rain |
15:05.47 | *** join/#asterisk Akiraa (n=Akiraa@79.112.17.211) |
15:05.59 | gr0mit | and wishes he was somewhere other than .uk |
15:06.01 | Katty | gr0mit: :< |
15:06.13 | Katty | gr0mit: there's some sunshine here. i will bottle it up and ship some to you! |
15:06.21 | Katty | gr0mit: first sunshine in what feels like weeks. |
15:06.21 | gr0mit | where is .here? |
15:06.25 | leifmadsen | drmessano: o.O |
15:06.27 | Katty | central usa |
15:06.35 | gr0mit | any particular bit? |
15:06.39 | Katty | hi mister madsen |
15:07.02 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
15:07.09 | leifmadsen | ms. katty |
15:07.28 | gr0mit | central as in mid-west? |
15:08.04 | Katty | gr0mit: 36 degrees 18'33" N, 89 degree 32'47" W |
15:08.18 | x86 | heh |
15:08.18 | Katty | gr0mit: yes, midwest. southern missouri |
15:08.23 | x86 | http://icanhascheezburger.files.wordpress.com/2008/06/funny-pictures-dueling-lobsters.jpg |
15:08.27 | x86 | rofl |
15:08.30 | gr0mit | aah ok. |
15:08.42 | gr0mit | has never been there. spent a lot of time in chicago |
15:08.42 | Katty | x86: that's awesome. |
15:08.55 | Katty | gr0mit: there's mostly trees and cattle here. |
15:08.59 | Katty | gr0mit: oh, and corn. |
15:09.05 | Katty | gr0mit: fields and fields of corn. everywhere. |
15:09.19 | gr0mit | once you leave the suburbs of northern chicago, thats the same! |
15:09.23 | leifmadsen | Katty: but not corn syrup funny enough |
15:09.34 | Katty | leifmadsen: they probably turn it into corn syrup |
15:09.41 | Katty | leifmadsen: it's mostly field corn, afterall. |
15:09.42 | leifmadsen | :) |
15:09.51 | leifmadsen | guess I'm not funny this early in the morning |
15:09.54 | Katty | or livestock feed |
15:10.09 | x86 | Katty: http://icanhascheezburger.files.wordpress.com/2008/05/funny-pictures-technical-support-cat.jpg |
15:10.30 | Katty | ahahaa |
15:10.34 | x86 | yeah ;) |
15:10.43 | Katty | i'm totally goign to answer my phone like that today |
15:10.50 | gr0mit | wishes he was wearing his "No, I will not fix your daughter's laptop" T-shirt |
15:10.54 | x86 | i'm gonna print that out and tape it up somewhere in the helpdesk area at work |
15:11.25 | ManxPower-work | offers to let gr0mit borrow his "No I will not fix your computer" t-shirt |
15:11.51 | gr0mit | puts it on, realising it is too late. Vista re-install nearly complete |
15:12.14 | gr0mit | humph. 96 important updates. |
15:12.26 | Katty | i used to have some thinkgeek shirts. |
15:12.40 | Katty | had one that had PI on the front, and then there's no place like 127.0.0.1 |
15:12.47 | gr0mit | lol! |
15:12.53 | ManxPower-work | I have the Pi one. |
15:13.02 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
15:13.15 | Katty | the st. louis science center has a shirt that says PI on the front, and has a scoop of icecream on top of it |
15:13.32 | Katty | does it come in girl sizes tho :< |
15:13.45 | Katty | s/does/doesn't/ |
15:14.28 | gr0mit | struggles wuth a 900 metre wifi link |
15:14.34 | ManxPower-work | I used to have this one: http://www.t-shirthumor.com/Merchant2/graphics/fullsize/flam_lg2.gif |
15:14.58 | Katty | omg |
15:15.00 | Katty | eww lol |
15:15.03 | x86 | gr0mit: why struggle? get an amplifier ;) |
15:15.04 | ManxPower-work | LOL! |
15:15.17 | Katty | that's just WRONG! |
15:15.23 | Katty | retweets it |
15:15.36 | gr0mit | well, our telco is installing FTTC here |
15:15.43 | ManxPower-work | gr0mit: Have you looked at defactowireless.com I get all my long range gear |
15:15.46 | ManxPower-work | from them |
15:15.50 | gr0mit | but the rollout stops 900 metres from my house |
15:16.00 | gr0mit | so have installed a link to a friends house |
15:16.09 | ManxPower-work | They sell 600mw APs |
15:16.15 | gr0mit | and will get the VDSL installed there |
15:16.21 | ManxPower-work | 14dBi gain antennas, enclosures, etc |
15:17.13 | gr0mit | http://www.compex.com.sg/DownLoads/Manual/UM-MMJ543.pdf is wot i got |
15:17.16 | x86 | gr0mit: http://www.l-com.com/item.aspx?id=22137 |
15:17.50 | gr0mit | fears the RF police! |
15:17.59 | ManxPower-work | x86: lcom has some pretty cool too. |
15:18.00 | gr0mit | i am running at 5.5ish GHz |
15:18.12 | ManxPower-work | Hello Rain fade! |
15:18.14 | Katty | who was it that recommened i watch Marine 2 last night? |
15:18.32 | gr0mit | hmm its all ulaw-ish |
15:18.53 | gr0mit | 5.5 GHZ is not so prone to raing, right? |
15:18.58 | gr0mit | or wrong? |
15:19.29 | ManxPower-work | gr0mit: I'm not an expert. I think most all frequencies in the microwave range will have rain fade. |
15:20.36 | gr0mit | http://www.flickr.com/photos/13418468@N07/ |
15:20.48 | x86 | ManxPower-work: used to be hyperlink technologies back when I used them |
15:21.10 | Katty | is that a solar panel? |
15:21.13 | x86 | ManxPower-work: I used to manage a small WISP and we used all hyperlink antennas / amps / cabling / everything... loved them |
15:21.14 | Katty | or just a window |
15:21.27 | gr0mit | Velux roof window |
15:21.32 | Katty | k |
15:21.40 | gr0mit | with my 5Gz antenna mounted outside |
15:22.07 | Katty | sounds pretty schnazzy |
15:22.31 | gr0mit | it was really cheap, like £10 for the pair |
15:22.35 | ManxPower-work | I have something like this that I need to install http://shop.defactowireless.com/core/media/media.nl?id=18793&c=300197&h=140ccd8debf949f6e937&resizeid=-2&resizeh=340&resizew=240 |
15:22.35 | gr0mit | i mean £120 |
15:22.51 | shamelessn00b | oh yeah |
15:23.01 | shamelessn00b | handling 500+ calls |
15:23.03 | shamelessn00b | just awesome |
15:23.05 | ManxPower-work | x86: I think I bought some ethernet surge supressors from them. |
15:23.08 | shamelessn00b | all calls executing agis |
15:23.15 | shamelessn00b | herp derp |
15:23.44 | gr0mit | am planning to operate a small wisp |
15:23.45 | ManxPower-work | When you run 1000 ft of underground cable thru iron saturated soil on a mountain you need ethernet surge supressors. |
15:23.51 | gr0mit | to our street |
15:24.13 | gr0mit | got pppoe logins to a mikrotik box |
15:26.24 | Akiraa | What does a cable with a ring capacitor do? |
15:26.50 | gr0mit | in UK? |
15:27.14 | Akiraa | anywhere, but I find them on a UK distribuitor |
15:27.22 | gr0mit | ok, it is a UK thing |
15:27.41 | Akiraa | what is it used for? |
15:27.53 | gr0mit | http://www.wppltd.demon.co.uk/WPP/Wiring/UK_telephone/uk_telephone.html |
15:29.47 | gr0mit | was used on old pulse dial phones to stop bells on other extensions rattling when you dialled |
15:29.53 | x86 | ManxPower-work: we never worried about lightning on the ethernet side... we'd run fiber from our switches to the AP... we'd just use a converter on the AP side that didn't have a fiber connection |
15:30.19 | x86 | ManxPower-work: so if the AP got hit, that's as far as it could go (well, and the media converter) |
15:30.34 | gr0mit | Akiraa, you probably don't need whatever it is you are looking at! |
15:33.26 | Akiraa | gr0mit: thanks, just wondered what it was |
15:39.03 | coppice | The UK tried to made it hard for foreign suppliers by making their phone system incompatible with the rest of the world - a dumb and unsafe plug/socket and a stupid third wire arrangement |
15:39.40 | gr0mit | unsafe? |
15:40.39 | coppice | yep. it fails international safety standards |
15:40.55 | coppice | a child's finger can go in |
15:41.18 | coppice | the RJ11 is a little smaller, and passes that test |
15:41.40 | gr0mit | aah yes i recall when i worked at BABT |
15:41.48 | gr0mit | it was all a mess |
15:41.56 | coppice | for a long time, until they cooked the standards in the UK, new sockets couldn't get approval :-) |
15:42.30 | gr0mit | indeed so. They needed to be recessed by approx 3mm |
15:42.31 | Corydon76-dig | coppice: Dunno, maybe it's Darwin at work? |
15:42.45 | gr0mit | in order for a finger not to get access to the contacts |
15:45.08 | coppice | HK used that stupid system, too, which is odd, because HK mostly followed US telephony practice |
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15:46.18 | ManxPower-work | coppice: I thought the 3rd wire was a common ground so your tx path and rx path don't go on the same wire. |
15:46.36 | ManxPower-work | Which honestly sounds like an awesome idea from the standpoint of echo |
15:46.39 | gr0mit | ManxPower-work, nope |
15:47.02 | gr0mit | it just separates the 100v ring current |
15:47.03 | coppice | ManxPower-work: it might do, until you understand some engineering :-) |
15:47.11 | ManxPower-work | Too logical for the Brits? |
15:47.25 | gr0mit | 3rd wire does not go back to the exchange |
15:47.37 | gr0mit | only appears in the Master socket |
15:47.40 | coppice | the third wire in the UK only passes around the phones within a house |
15:48.01 | Katty | this tea tastes attrocious |
15:48.13 | gr0mit | hands Katty some Earl Gray |
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15:48.26 | Katty | gr0mit: iced tea. |
15:48.32 | benngard | prefer the old swedish system |
15:48.39 | Katty | i'm not sure how earl gray would taste cold ^_- |
15:48.59 | gr0mit | iced tea? |
15:49.02 | gr0mit | shudders |
15:49.14 | coppice | Katty: he has been dead for years so he's probably pretty cold |
15:49.44 | Katty | hehehe |
15:49.45 | Katty | nice. |
15:49.47 | Katty | applauds coppice |
15:50.06 | benngard | any1 know if u can find may213 on any irc channel? |
15:50.36 | ManxPower-work | benngard: type /whois may213 |
15:50.51 | Katty | you can also ask nickserv for info |
15:50.58 | Katty | it'll tell you the last time they were online |
15:51.04 | Katty | something like /msg nickserv info Katty |
15:53.07 | eppigy | schooches closer to Katty |
15:53.53 | Katty | hello deary! |
15:53.55 | Katty | hugs eppigy |
15:59.41 | eppigy | hiya |
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16:06.01 | *** join/#asterisk bpgoldsb (n=bpgoldsb@ip24-250-198-162.ga.at.cox.net) |
16:06.33 | bpgoldsb | Has anyone run Asterisk under Xen and can comment on the performance of it? |
16:10.41 | bmoraca | bpgoldsb, i prefer VMware ESXi |
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16:12.31 | murraytm | can someone tell me what compile time options were used for the 1.6.0.20 asterisk that's in the asterisk-current yum repo? |
16:13.46 | Chainsaw | murraytm: The .spec file, hopefully? |
16:14.34 | bmoraca | why worry about it? just download the source and compile it yourself. it's not exactly rocket surgery |
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16:15.35 | Katty | eppigy: you would not believe how difficult it is to find boots :< |
16:15.44 | Katty | eppigy: i think it's a conspiracy |
16:16.29 | jaytee | rocket surgery? like a gastric bypass on a Saturn 5? |
16:16.42 | murraytm | i'm basically trying to get a couple of modules to load in the one installed from yum and would rather not have to reinstall all of asterisk to do it |
16:17.02 | murraytm | where could i find the .spec file? |
16:19.17 | bmoraca | what modules? if they're not statically linked, you'll need to compile those modules anyway |
16:19.45 | murraytm | custom volume function and modified res_agi |
16:19.54 | ManxPower-work | Remmember almost nobody will help you if you compile from packages. |
16:20.34 | ManxPower-work | Katty: your local army/navy surplus may have a good selection of boots. |
16:20.34 | [TK]D-Fender | ManxPower-work: I don't recall mention of RPMS |
16:20.52 | ManxPower-work | [TK]D-Fender: sorry, I thought a yum-repo had RPMs. |
16:21.10 | murraytm | i'm trying to come up with a fast and reliable way to migrate a system running 1.4 and installing by yum, then dropping in some binaries seems like a quicker way than recompiling all of asterisk |
16:21.21 | bmoraca | murraytm, if they're statically linked, just drop the files in the modules directory and restart asterisk (or use the module load command) |
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16:21.36 | [TK]D-Fender | ManxPower-work: (source) RPM's.... |
16:22.06 | [TK]D-Fender | murraytm: Compiling * takes what... 5 minutes maybe? |
16:22.14 | bmoraca | murraytm, compiling really doesn't take that long. and, like i said, unless those binaries you have were statically linked, you won't be able to use them anyway |
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16:22.41 | murraytm | maybe i'm just being paranoid. seems like a lot can go wrong during compilation compared to just installing from the repo. |
16:23.50 | bmoraca | ./configure will tell you if you're missing anything. make menuselect lets you choose exactly what you want and won't let you choose what you don't have prereqs for...it's a fairly brainless process anymore |
16:24.13 | murraytm | it's not the compiling, it's the making sure the install works after it's compiled that i'm worried about |
16:24.18 | bmoraca | well, it will let you choose ilbc if you don't have prereqs...so, i'd advise caution around that one |
16:24.19 | Katty | ManxPower-work: no, i mean dress boots. |
16:24.23 | Katty | ManxPower-work: just plain NORMAL dress boots. |
16:24.31 | Katty | ManxPower-work: without all this extra buckle frill fur stuff |
16:24.32 | bmoraca | murraytm, make install puts everything where it needs to go |
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16:24.57 | murraytm | i did 1.4 from source so i'm familiar with the process, just was thinking there might be a better way |
16:25.21 | bmoraca | murraytm, if you need custom modules, there is no other way |
16:25.25 | eppigy | Katty: :< |
16:25.33 | murraytm | ok, well that settles it then. thanks. :) |
16:25.33 | eppigy | i have faith that you will find your boots |
16:25.38 | Holister | I was having problems with my DAHDI card (it made very wierd noises when a call was connected), so I rebooted. Now I can't get the card to initialize, and port 1 doesn't have a green light. Is the card done for? |
16:26.23 | Katty | eppigy: i will :> |
16:26.24 | Katty | eppigy: but i'm going to complain in the meantime! |
16:26.24 | bmoraca | murraytm, installing on CentOS? |
16:26.24 | ManxPower-work | Holister: can you power cycle the machine? |
16:26.24 | murraytm | yes |
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16:26.45 | Holister | ManxPower-work: would be difficult, but I could try. The problems started when I power cycled |
16:27.07 | benngard | avaya - ooh323 - asterisk is not the working :( |
16:27.10 | Holister | ManxPower-work: well, not all of the problems, but not being able to initialize the card, and the light goinbg off |
16:27.10 | bmoraca | murraytm, there's a fairly replete guide on voip-info for installing 1.6 on CentOS...at the very least, it'll give you all the prereqs you need |
16:27.14 | ManxPower-work | Holister: you can also contact tech support. This should be somethnig they will support. |
16:27.39 | Holister | ManxPower-work: the manufacturer of the card? |
16:27.42 | murraytm | ok, thanks, i'll have a look through that |
16:28.03 | ManxPower-work | Holister: correct. I assume it's Digium since you didn't tell use the make or model of the card. |
16:28.16 | Holister | ManxPower-work: it is TDM400 |
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16:28.52 | ManxPower-work | Contact Digium then. |
16:29.10 | ManxPower-work | Holister: also remember that if you upgraded your kernel then you need to recompile and reinstall DAHDI |
16:29.12 | Holister | thx |
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16:29.22 | Holister | hmmmm |
16:29.28 | Holister | well modprobe appears to work |
16:29.31 | torrancew | would anyone have a recommend a good, affordable, cordless IP phone? |
16:29.41 | ManxPower-work | torrancew: There are none. |
16:29.59 | ManxPower-work | torrancew: But you can easily use a SIP ATA + your favorite cordless phone. |
16:30.45 | torrancew | would the Linksys PAP2 suffice for that? |
16:30.50 | Akiraa | Are there licensing limitations in deploying IAX2 devices? |
16:31.01 | ManxPower-work | torrancew: yes |
16:31.23 | torrancew | and what if i were willing to compromise on the affordable end of cordless IP phone? |
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16:31.42 | ManxPower-work | Akiraa: no, but there are copyright issues with using things like Asterisk, Digium, etc in marketing. Digium has a policy on this. |
16:31.42 | torrancew | does that change the situation at all? |
16:31.47 | bmoraca | torrancew, i recommend the Philips CD1 phones...they support CID and have a MWI that works really well with the Linksys PAP2T and the SPA8000 |
16:31.52 | ManxPower-work | torrancew: Define "affordable" |
16:32.04 | torrancew | that's negotiable - it's for a business |
16:32.18 | ManxPower-work | ~phones |
16:32.19 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
16:32.21 | torrancew | reliable and relatively portable would be the concern |
16:32.28 | ManxPower-work | You'll notice there is no cordless model listed. |
16:32.38 | bmoraca | torrancew, the Polycom cordless phones are the only ones I'd trust and they're NOT cheap. Snom makes one that's more affordable, but doesn't work as well |
16:32.39 | torrancew | touche' |
16:32.48 | torrancew | thanks all |
16:32.52 | bcrisp | i want a wrist watch phone 007 style |
16:33.01 | torrancew | bcrisp: agreed |
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16:33.16 | Katty | bcrisp: LASER PEWPEWPEW |
16:33.16 | anonymouz666 | what's the problem with Cisco phones? |
16:33.19 | Chainsaw | *G* Neat. Cisco & Grandstream swapped? |
16:33.34 | Katty | anonymouz666: they're just very expensive |
16:33.43 | Katty | anonymouz666: but they do have a lot of very nice features. |
16:33.44 | Chainsaw | anonymouz666: Firmware hidden behind a paywall, bleak XML support on the SIP firmware... |
16:33.45 | anonymouz666 | Katty: indeed but works |
16:33.53 | Katty | anonymouz666: i'm particularly fond of cisco's blackberry call precense app |
16:33.57 | Katty | prescense |
16:33.59 | Katty | whatever |
16:34.02 | Katty | i can't spell that word |
16:34.05 | Chainsaw | pre-sense? |
16:34.08 | Katty | PRESENTS |
16:34.15 | Chainsaw | presence :) |
16:34.16 | ChannelZ | yay! |
16:34.26 | Katty | i also can't spell convience |
16:34.30 | Katty | con viene ence |
16:34.36 | torrancew | Katty: convenience |
16:34.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
16:34.39 | Katty | whatever. |
16:34.40 | torrancew | :-) |
16:34.49 | coppice | Katty: I'm particularly fond of Blackberry and Apple Pie |
16:34.54 | Katty | i should make them a password for something. |
16:35.02 | Corydon76-dig | Katty: phonetics FTW |
16:35.03 | Katty | coppice: blackberry cobbler's some mean stuff. |
16:35.23 | Katty | pie FTW. |
16:35.45 | Katty | there is something terribly wrong with spring fashion trends. |
16:36.01 | Katty | it all looks /retarded/ |
16:36.19 | Katty | it's all tunics, and fluffy frills. |
16:36.43 | ChannelZ | fashion is retarded |
16:37.32 | Katty | have you seen the shoes? |
16:37.33 | Corydon76-dig | Uh, you're talking to an audience of geeks. Fashion is not really something that any of us take into account. Jeans and tshirts, remember? |
16:38.05 | [TK]D-Fender | .. and the occasions Storm Trooper uniform |
16:38.05 | Katty | http://images.bakersshoes.com/images/products/1_102526_FS.JPG <- seriously? come on now. |
16:38.18 | Katty | Corydon76-dig: oh right. yeah. |
16:38.38 | Katty | there's an idea, i will just get a ton of star trek uniforms. |
16:38.40 | Corydon76-dig | Bonus if the Tshirt was free, because a vendor gave it to us |
16:38.55 | torrancew | ManxPower-work: how feasible would it be to run my own STUN server to help with using SIP extensions from off-site locations? |
16:38.59 | Corydon76-dig | Extra bonus if the Tshirt is clean |
16:39.06 | torrancew | (have an in-house asterisk server) |
16:40.19 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
16:40.44 | SuPrSluG | torrancew: there are plenty for free use. Why spin your own? |
16:40.54 | ManxPower-work | torrancew: seems like quite a bit of work for almost no gain. |
16:41.10 | torrancew | company doesn't want to use an external server |
16:41.28 | *** join/#asterisk jmacz (n=jmacz@186.80.77.231) |
16:41.29 | torrancew | though if someone here can put up a better argument i can take to The Man for why it's no big deal, i'm game |
16:42.05 | torrancew | the company in question is an IT consulting firm, specializes in Macs. |
16:42.33 | ManxPower-work | Why do you need a STUN server? |
16:42.49 | torrancew | they want to make SIP calls from off-site with their iPhones |
16:43.39 | ManxPower-work | torrancew: OK. You DO NOT NEED STUN WHEN USING NAT |
16:43.57 | bcrisp | ATT is supposed to open up for SIP on their network |
16:44.03 | torrancew | ah, i misunderstood then |
16:44.18 | torrancew | don't you need open ports for RTP though? |
16:45.13 | torrancew | at both networks, that is |
16:46.06 | ManxPower-work | torrancew: You only need to port forward on the NAT for the Asterisk box. No port forwarding needed (and in fact may break stuff) on the Phone NAT router. |
16:46.17 | torrancew | ah i see |
16:46.22 | SuPrSluG | the client shouldn't have to. you will need to port forward SIP 5060 & RTP 10000-20000 to the * box |
16:46.34 | torrancew | right |
16:46.47 | torrancew | ok |
16:46.52 | torrancew | i'll explore that a bit |
16:46.53 | torrancew | thanks |
16:48.57 | torrancew | one last question - going through a PAP, would I lose any of the key asterisk features? i'd assume most of them will be fine, since alot of that is handled by extensions and dial plans, but would like to hear from experienced minds |
16:51.21 | *** join/#asterisk thieums93 (n=mathieu@LPuteaux-156-16-101-43.w80-12.abo.wanadoo.fr) |
16:51.37 | SuPrSluG | not asterisk features, but phone features |
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16:52.34 | torrancew | SuPrSluG: such as? |
16:53.00 | thieums93 | Hi, do you know how to transmit voicemailuser to callback / dialout voicemail context ? |
16:53.11 | SuPrSluG | one touch dialing for features like voicemail or extensions |
16:53.40 | SuPrSluG | not deal breakers or anything |
16:54.09 | [TK]D-Fender | thieums93: huh? |
16:54.31 | thieums93 | not clear ? |
16:54.47 | [TK]D-Fender | thieums93: No. |
16:55.33 | torrancew | SuPrSluG: thanks |
16:55.47 | SuPrSluG | better codecs like hd voice too, that's nice |
16:57.00 | Katty | jeebus, the lighthouse feeder has gone down 50% since this morning |
16:57.08 | Katty | and i believe the suet block is officially GONE |
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17:00.53 | thieums93 | a user is authenticated on voicemailmain, press 3(advanced option) then 4(dialout), jump to the context defined by dialout= in [general] (voicemail.conf) . I want to get in my dialout context the authenticated mailbox account, is that possible ? |
17:01.28 | Katty | did you check to see if that's a variable? |
17:01.43 | thieums93 | yep, nothing on wiki |
17:01.47 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
17:02.52 | [TK]D-Fender | thieums93: set the "dialout=" in the box's definition. Most parms can be used in [general], and in the box definition itself |
17:03.18 | cosmicwombat | Anyone know of a common reason callers on hold get dropped after 30 seconds... everytime ? |
17:06.37 | Kobaz | cosmicwombat: misconfiguration |
17:07.11 | *** join/#asterisk michael-i (n=michael-@208.53.198.95) |
17:07.20 | Kobaz | cosmicwombat: it could be the calling party, it could be the phone, it could be dialplan |
17:07.21 | cosmicwombat | Kobaz: where should I look ? |
17:07.35 | Kobaz | are you using sip phones? |
17:07.36 | [TK]D-Fender | Kobaz: Umm... nope |
17:07.42 | Kobaz | paste the sip debug |
17:07.46 | Kobaz | [TK]D-Fender: nope what? |
17:07.51 | [TK]D-Fender | Kobaz: Not dialplan. |
17:08.08 | cosmicwombat | I think firewall is at play |
17:08.10 | Kobaz | [TK]D-Fender: if he's got some dialplan code in the background killing calls after 30 seconds, that could be it |
17:08.11 | [TK]D-Fender | cosmicwombat: Common cause for this is a bithy device complaining about not getting RTP while on hold |
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17:08.25 | ManxPower-work | thieums93: don't look at the Wiki, look at /path/to/src/asterisk/doc/tex/channelvariables.tex |
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17:08.48 | [TK]D-Fender | ManxPower-work: .... he's not talking about channel vars... |
17:08.49 | Kobaz | [TK]D-Fender: you're always the one that encourages people to not overlook stuff :P |
17:09.02 | [TK]D-Fender | Kobaz: Hey look... its ELVIS! |
17:09.08 | Kobaz | heh |
17:09.13 | [TK]D-Fender | points the other way then runs while Kobaz is distracted |
17:09.27 | cosmicwombat | Thank you very much , uh huh |
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17:09.36 | [TK]D-Fender | curls his lip |
17:10.22 | benngard | Dec 29 18:04:52 sip kernel: [2146494.253819] asterisk[14647]: segfault at 94 ip b7cc49a0 sp b62d3204 error 4 in libpthread-2.7.so[b7cbd000+15000] :( |
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17:11.44 | jblack | benngard: Not good at all. |
17:11.52 | benngard | nup |
17:11.55 | jblack | at least you have a good hint of where to start. |
17:12.09 | benngard | yupp |
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17:12.25 | benngard | did know what i did so i think i do a "rollback" |
17:12.40 | jblack | actually, maybe not. being pthread, it could be a lot of things. |
17:12.54 | jblack | pthread may just be catching it. |
17:13.15 | benngard | yes but i know what i changed so i "hope" i can go back |
17:13.29 | ManxPower-work | What did you change? |
17:13.57 | benngard | i went from h323 channel to ooh323 channel |
17:14.25 | benngard | i did some tests with may213, gave hime some logs |
17:15.24 | benngard | and ofc some dumb asshole at my office put some live traffic on my lab box so i have to do some reversed work later |
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17:17.53 | benngard | i have may213 in a private chat, gonna give him access to the box if he wants to have it |
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17:22.14 | cusco | hi... |
17:22.55 | cusco | [Dec 29 17:19:25] WARNING[13670] /home/murf/asterisk/1.6.1/main/ast_expr2.y: non-numeric argument |
17:23.41 | cusco | I can't find the why |
17:23.45 | cusco | what argument is it referring to |
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17:24.40 | ManxPower-work | cusco: make sure you have bison and yacc installed |
17:25.11 | _cgc | does anyone have any experience with liz for sugar? |
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17:25.59 | _cgc | but on the asterisk side obviously |
17:26.03 | ChannelZ | anyone bored and want to try a test for me? |
17:26.06 | ManxPower-work | _cgc: did you try asking on #SugarCRM? |
17:26.09 | cusco | ManxPower-work: bison is installed. yacc calls bison |
17:26.42 | TheDavidFactor-H | ChannelZ what do you need? |
17:26.50 | _cgc | ManxPower-work: its asterisk i need help with though, not sugar |
17:27.36 | ChannelZ | TheDavidFactor-H: Try playing a .gsm file while on a ulaw transport and see if it sounds right |
17:27.36 | _cgc | ManxPower-work: i had it working before, but for some reason it's stopped working with the configuration I had, and not too sure why... |
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17:28.02 | TheDavidFactor-H | what version of *? |
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17:28.29 | ChannelZ | TheDavidFactor-H: Any - what are you running? |
17:29.00 | TheDavidFactor-H | trunk, 1.6.1.6, 1.4.20, 1.4.23.1, and one or two others |
17:29.00 | _cgc | its related to dialing queues, when I change the queue from member => SIP/phone1 to member => Local/1@phones it says all the phones are invalid |
17:29.27 | ManxPower-work | Do you have an exten => 1 in the [phones] context in extensions.conf? |
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17:29.30 | ChannelZ | TheDavidFactor-H: I'd be interested in 1.6.1.6 and 1.4.20 |
17:30.04 | _cgc | yep exten => 1,1,Macro(callphone,phone1,${EXTEN},SIP) |
17:30.11 | _cgc | and in there is: |
17:30.14 | ChannelZ | TheDavidFactor-H: Something odd is happening to me on 1.6.1.12 and I reverted to 1.4.27.2 to test and it was doing it then too |
17:30.31 | ManxPower-work | _cgc: "dialplan show" and confirm that |
17:30.39 | ChannelZ | TheDavidFactor-H: But I know it worked once upon a time when I originally built the system (but I don't even remember what version of 1.4 that was under, couple years ago now) |
17:31.11 | _cgc | chatterbox*CLI> dialplan show phones |
17:31.11 | _cgc | [ Context 'phones' created by 'pbx_config' ] |
17:31.11 | _cgc | <PROTECTED> |
17:31.11 | _cgc | <PROTECTED> |
17:31.11 | _cgc | <PROTECTED> |
17:31.12 | _cgc | <PROTECTED> |
17:31.14 | _cgc | <PROTECTED> |
17:31.16 | _cgc | <PROTECTED> |
17:31.18 | _cgc | <PROTECTED> |
17:31.20 | bcrisp | stop flooding |
17:31.22 | _cgc | <PROTECTED> |
17:31.23 | ChannelZ | _cgc: USE PASTEBIN.CA |
17:31.24 | _cgc | <PROTECTED> |
17:31.26 | _cgc | <PROTECTED> |
17:31.28 | _cgc | <PROTECTED> |
17:31.30 | _cgc | <PROTECTED> |
17:31.31 | TheDavidFactor-H | ~pastebin |
17:31.32 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
17:31.32 | _cgc | <PROTECTED> |
17:31.34 | _cgc | <PROTECTED> |
17:31.34 | bcrisp | ... |
17:31.36 | _cgc | <PROTECTED> |
17:31.38 | _cgc | <PROTECTED> |
17:31.40 | _cgc | <PROTECTED> |
17:31.42 | _cgc | sorry |
17:31.43 | ChannelZ | grrrrrr |
17:32.09 | _cgc | http://pastebin.ca/1730876 |
17:33.02 | sektorNBA | ahh |
17:33.18 | _cgc | http://pastebin.ca/1730877 |
17:33.31 | Qwell | ChannelZ: https://issues.asterisk.org/view.php?id=16516 |
17:33.38 | Qwell | ChannelZ: if you have any more information to add, please do |
17:33.46 | Qwell | (unless that's your bug..) |
17:34.05 | ChannelZ | Qwell: That's me ;) |
17:34.18 | Qwell | yeah, kinda thought so |
17:34.23 | Qwell | hard to remember some of thenick mappings :p |
17:34.31 | ChannelZ | Just trying to figure out if it really _is_ a bug |
17:34.32 | ManxPower-work | _cgc: you could be banned from the channel for flooding |
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17:34.39 | ChannelZ | or just something screwy with me |
17:34.44 | _cgc | yes, sorry about that |
17:35.37 | Qwell | ChannelZ: would it be possible to record the bad audio from the phones perspective? |
17:36.00 | _cgc | it will not happen again :) |
17:36.19 | Qwell | oh. 8.04. what gcc version? |
17:36.23 | Qwell | ~gsmbug |
17:36.23 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
17:36.33 | ChannelZ | Qwell: Yeah I can do it through my audio card on a softphone for sure when I get home |
17:37.07 | ChannelZ | Qwell: ahh interesting.. (reading) |
17:37.08 | ManxPower-work | A few phones default to 30ms packets instead of 20ms (which is what Asterisk expects). |
17:37.48 | ChannelZ | looks like gcc 4.2.4 |
17:38.26 | Qwell | well, there's your problem. upgraded recently? |
17:38.37 | SuPrSluG | _cgc: are you trying to call 2 technologies at once? SIP and Local |
17:39.56 | TheDavidFactor-H | ChannelZ, I didn't have a problem with 1.6.1.6, but it looks like you found your problem? |
17:40.26 | Qwell | ChannelZ: post on that issue once you've recompiled with something besides 4.2, so we can close it or whatever |
17:40.44 | ChannelZ | Qwell: Yes and no, upgraded a long time ago but only recently found this as a problem when I upgraded to 1.6 (I don't use gsm much). I'll try a recompile with optimizations off but it looks like this is probably what is happening for me |
17:41.03 | ChannelZ | TheDavidFactor-H: Thanks for testing that! |
17:41.07 | ChannelZ | Qwell: I will |
17:41.11 | _cgc | SuPrSluG: ultimately its meant to dial some SIP phones, but liz requires you to use Local/${EXTEN}@phones so from the macro that gets called in the phones context i'm dialling the sip phone, is this wrong then? |
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17:43.24 | _cgc | the other alternative is rewriting liz but i'm not that good with php |
17:44.07 | voipmonk | local is not a bad solution |
17:44.13 | voipmonk | esp if it works |
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17:44.23 | voipmonk | get er done - move on to the next problem |
17:45.11 | _cgc | well it should according to the documentation but it seems to say all of the queue members are 'Invalid' when using Local |
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17:50.25 | KaneHau | Aloha... asterisk newbie here with a problem. When I do a 'ztcfg -vv' it identifies the channel properly but then says "1 channels to configure" rather than "1 channel configured" - any hints? |
17:50.50 | [TK]D-Fender | KaneHau: I don't see a problem there. |
17:51.04 | [TK]D-Fender | KaneHau: because ASTERISK configures it... not ztcfg |
17:51.26 | KaneHau | ah, ok... hmmm. does zapata.conf go in /etc or in /etc/asterisk? |
17:51.38 | [TK]D-Fender | KaneHau: Feel free to move on and show us where an actual problem presents itself :) |
17:51.47 | [TK]D-Fender | KaneHau: /etc/asterisk |
17:52.02 | SuPrSluG | _cgc: try turning up the verbose to see how it dials, for it to be invalid it must try to do something |
17:52.08 | [TK]D-Fender | KaneHau: You really should migrate to DAHDI... |
17:52.25 | KaneHau | well, I'm trying to test a very simple script - to answer the phone in Echo() mode. I have a Viking advanced line simulator which provides the dialtone. It rings, but the FXO is not picking up the line |
17:53.55 | [TK]D-Fender | KaneHau: Show us your actual attempt.... |
17:53.59 | [TK]D-Fender | ~pb |
17:53.59 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
17:54.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
17:54.11 | [TK]D-Fender | KaneHau: and PASTEBIN it.... do not flood the channel |
17:54.48 | KaneHau | zaptel.conf is very simple: fxsks=1 loadzone=us defaultzone=us |
17:55.20 | KaneHau | zapata.conf is also simple: context = incoming signaling = fxs_ks channel=>1 |
17:55.54 | KaneHau | extensions.conf has, under [incoming] exten => s,1,Answer() exten=>s,n,Echo() |
17:56.22 | [TK]D-Fender | KaneHau: You said it provides dialtone. What is there for * to ANSWER? |
17:56.38 | _cgc | SuPrSluG: http://pastebin.ca/1730907 |
17:57.14 | KaneHau | no, you misunderstand... I have a line simulator that actually gives the dial tone to the FXO device. Plugged into the other end of the simulator I have a real phone. When I dial a number on the real phone, the simulator rings the FXO line which I'm expecting to pick up, but it doesn't |
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17:57.44 | ManxPower-work | KaneHau: Less talk, More pastebin. |
17:58.17 | ManxPower-work | CLI output, specifically |
17:58.31 | [TK]D-Fender | KaneHau: And * has to be started once Zaptel is actually ready.. and must have been compiled AFTER Zaptel is installed to even have support for it |
17:59.01 | [TK]D-Fender | KaneHau: So go prove a few things and pastebin "zap show status" "zap show channels" and "zap show channel 1" along with your failed attempt |
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18:01.38 | raden_work | Naikrovek, heya bro |
18:03.11 | _cgc | http://pastebin.ca/1730907 |
18:04.20 | [TK]D-Fender | _cgc: Should be using /n at the end, and I don't see dialplan to match |
18:04.46 | ChannelZ | Qwell: Note added to https://issues.asterisk.org/view.php?id=16516 - this is the gcc bug! |
18:05.07 | ChannelZ | (or gcc 4.2-ism anyway) |
18:05.45 | ChannelZ | Thanks for your help - Dunno why I didn't find that one, but I was doing a few searches and guess I missed looking for closed bugs on one of them.. |
18:06.06 | KaneHau | btw, I'm on SUSE and can't seem to find libnewt so that I can create zttool, any suggestions? |
18:06.39 | KaneHau | all the libnewts seem to be debian distributions |
18:08.38 | _cgc | http://pastebin.ca/1730923 |
18:08.55 | tzafrir_laptop | KaneHau, newt-devel ? |
18:09.15 | KaneHau | I'll look, thanks |
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18:09.27 | tzafrir_laptop | KaneHau, also note that the latest version of Zaptel is DAHDI |
18:09.48 | KaneHau | oh, I see. So rather than doing the zaptel install, I should do the DAHDI install? |
18:10.13 | KaneHau | I'm following teh "Asterisk, the future of telephony" pdf (2nd edition) - is there a better reference I should be reading? |
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18:10.59 | ManxPower-work | KaneHau: it's the best Asterisk Book out there, but it is becoming outdated. Good thing there are lots of docs in the doc/ directory of the Asterisk souce code. |
18:11.00 | torrancew | in the same vein as KaneHau, is there a 3rd edition in the works? |
18:11.19 | KaneHau | thanks manx |
18:11.35 | _cgc | [TK]D-Fender: http://pastebin.ca/1730923 |
18:11.44 | torrancew | i found the book quite helpful when i was starting, but the DAHDI/Zaptel bit was frustrating, as well as some 2.6 stuff in general |
18:12.24 | [TK]D-Fender | KaneHau: And I'm not seeing any of what I asked for... |
18:12.35 | _cgc | the bottom bit that starts [levelone] is straight out of the queues.conf file |
18:12.42 | [TK]D-Fender | _cgc: add the /n and retry |
18:12.53 | _cgc | in queues.conf? |
18:13.08 | KaneHau | TK.. correct - I think I need to get rid of zaptel and go to DAHDI - so I want to reconfigure first |
18:13.25 | ManxPower-work | _cgc: read localchannel.tex in the Asterisk source. /n for for Local/ channels. |
18:13.39 | KaneHau | no sense having you help if my basic setup is wrong |
18:13.53 | _cgc | ok thanks, ill give it a try :) |
18:14.31 | [TK]D-Fender | KaneHau: What * are you on? |
18:14.52 | KaneHau | 1.6.2.0-rc7 |
18:15.18 | ManxPower-work | KaneHau: newbies should not be using unreleased code. |
18:15.20 | [TK]D-Fender | KaneHau: 1.6.x does not work with zaptel PERIOD. |
18:15.26 | ManxPower-work | I recommend you start with 1.6.1.x |
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18:15.31 | [TK]D-Fender | KaneHau: It requires DAHDI |
18:15.31 | KaneHau | mahalo nui loa... |
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18:16.22 | [TK]D-Fender | KaneHau: And 1.6.2.0 release is out... you shouldn't be on an RC at all if you even really want to be on that new a branch |
18:16.48 | KaneHau | I'm downloading 1.6.1.12 |
18:17.06 | angryuser_laptop | good day, i have some strange cpu lock's with dahdi dummy, i have removed it from asterisk use but still the system is hanging sometimes, maybe it is CPU problem, can someone look at it ? http://www.pastebin.ca/1730930 |
18:17.19 | KaneHau | my application is extremely simple... the device just needs to place outgoing calls - so I don't need 99% of the bangs and whistles |
18:19.22 | cusco | err... http://paste.debian.net/55193/ |
18:19.24 | bjifas | Hola, tengo una duda de compatibilidad |
18:19.28 | cusco | compiling asterisk addons errors out |
18:19.32 | bjifas | Hello, I have a question of compatibility |
18:19.41 | bjifas | libpri-1.2.8 and zaptel-1.2.27 are compatible with Asterisk 1.2.23? |
18:19.45 | Qwell | cusco: Are you using Asterisk 1.6.2.0? |
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18:19.53 | Qwell | (hint: you aren't) |
18:20.03 | cusco | I will be compiling it |
18:20.12 | cusco | do I need to compile asterisk first? |
18:20.18 | cusco | ok.. |
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18:25.43 | ariel_ | Hello everyone |
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18:36.00 | hardwire | bookawikawakachika |
18:36.05 | hardwire | does a little dance |
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18:40.15 | axelilly | Can someone one point me in the direction of a document that can should me how to test what day and time it is in extensions.ael? |
18:44.45 | [TK]D-Fender | axelilly: "core show application GotoIfTime" |
18:44.48 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
18:44.58 | [TK]D-Fender | axelilly: "core show function IFTIME" |
18:45.32 | axelilly | [TK]D-Fender: thanks! Looks like what I needed. cheers |
18:46.12 | Tim_Toady | and core show function STRFTIME or STRPTIME if u want to have a date string |
18:49.00 | *** join/#asterisk ticoit (n=ticoit@201.191.151.139) |
18:52.14 | *** join/#asterisk rare1980_ (n=rare@119.152.35.15) |
18:58.09 | axelilly | [TK]D-Fender: when I tried the gotoiftime I got this message: application call to GotoIfTime needs to be re-written using AEL if, while, goto, etc. keywords instead! |
18:58.27 | axelilly | Can I not use that application in AEL? |
18:58.39 | [TK]D-Fender | axelilly: I don't see how you TRIED to use it. |
18:59.18 | axelilly | [TK]D-Fender: ok, what I mean is that I did a ael reload and I got that message. |
19:00.12 | [TK]D-Fender | axelilly: What I mean is where's the damn code for me to look at? :) |
19:00.15 | [TK]D-Fender | ~pb |
19:00.15 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
19:01.35 | axelilly | [TK]D-Fender: yea, just realized, that'd help http://pastebin.com/d291f2123 |
19:02.19 | [TK]D-Fender | axelilly: try usinf IF() and IFTIME() |
19:02.23 | Tim_Toady | i think in ael its better to use iftime |
19:02.32 | axelilly | ok, will do |
19:02.47 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
19:03.21 | benngard | is happy! may213 did solve the "bug" by "borrow" my lab box |
19:03.30 | Qwell | benngard: woot |
19:03.58 | Qwell | it's unfortunate, but sometimes borrowing hardware really is the only way to fix a problem |
19:04.00 | benngard | pstn - avaya -ooh323 - asterisk is fucking working |
19:05.37 | benngard | he just "borrowed" some accounts ;) |
19:06.00 | benngard | and i did what he wanted me to do |
19:07.13 | Qwell | well, either way.. glad it's working for you now |
19:07.25 | benngard | Qwell: why ist it unfortunate, we are in the same boat, if we can help each other why not? |
19:07.44 | Qwell | benngard: I mean that it's unfortunate that some things can't be fixed without going that far |
19:08.13 | benngard | it was an avaya involved, what do u excpect ;) |
19:08.17 | Qwell | indeed |
19:08.25 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
19:09.16 | *** join/#asterisk asterisk1 (n=Jim@82-169-242-190.ip.telfort.nl) |
19:09.18 | benngard | but may213 have to finish my bugreport! i have not a clue what he did with my "source-tree"! |
19:09.21 | asterisk1 | hello all !! |
19:09.59 | asterisk1 | i have a very difficult question. When i forward a call to my cell phone i cant see the orginal CALLER ID |
19:10.09 | asterisk1 | using asterisk 1.4 |
19:10.21 | asterisk1 | i spend hours on this |
19:10.33 | asterisk1 | but always see the caller id of my asterisk box |
19:10.43 | asterisk1 | in stead of the orginal caller |
19:10.50 | asterisk1 | Can someone help me please ? |
19:10.57 | TheDavidFactor-H | how is your * connected to the POTS? |
19:11.20 | asterisk1 | i |
19:11.27 | asterisk1 | 'm using SIP |
19:11.30 | asterisk1 | dont have pots |
19:11.35 | asterisk1 | only a VOIP provider |
19:11.45 | asterisk1 | and a Grandstream GXP2000 phone |
19:12.50 | asterisk1 | when someone calls my and the call is being forwarded to my cell phone, |
19:12.58 | asterisk1 | i can't see who is calling |
19:13.04 | [TK]D-Fender | astMaybe your provider doesn't allow you to set your caller id |
19:13.08 | asterisk1 | and can't call back .. |
19:13.12 | [TK]D-Fender | asterisk1: Maybe your provider doesn't allow you to set your caller id |
19:13.26 | asterisk1 | that might be possible |
19:13.44 | TheDavidFactor-H | yea, double check the outbound callerid right before you dial your sip provider then call them |
19:13.51 | [TK]D-Fender | asterisk1: Then have your call out to your cell announce the callerID number before bridging. |
19:14.16 | asterisk1 | i'm going to check my extensions.conf |
19:14.26 | asterisk1 | and try your advise |
19:14.26 | ManxPower-work | asterisk1: find a provider that DOES let you set the callerid |
19:14.33 | asterisk1 | ok |
19:14.38 | asterisk1 | Thank you ! |
19:15.15 | benngard | is still happy, have done a lot of test calls all worked |
19:15.56 | benngard | gonna celebrate with a BIG BEER |
19:16.09 | benngard | is gone for a while |
19:17.57 | ManxPower-work | asterisk1: Remember callerid number NEVER EVER has a leading 1, quotes, dashes or other non-number chars. (the leading 1 is the toll prefix, not part of the callerid) |
19:18.02 | asterisk1 | it must be my asterisk box... When i connect my Grandstream phone directy |
19:18.16 | asterisk1 | to my provider, it works perfectly ! |
19:18.26 | [TK]D-Fender | asterisk1: huh? |
19:18.39 | asterisk1 | i did something wrong in extensions.conf |
19:18.45 | ManxPower-work | A Noop(CALLERID(num) is ${CALLERID(num)}) as the priority before your dial |
19:18.56 | ManxPower-work | asterisk1: what version of Asterisk? |
19:19.02 | asterisk1 | 1.4.24.1 |
19:19.19 | Katty | welp, i'm 100 bucks more broke |
19:19.23 | ManxPower-work | See the "o" option to Dial in "core show application dial" |
19:19.47 | asterisk1 | ok will have a look right now |
19:20.03 | asterisk1 | thank you for your advise !!! |
19:21.10 | asterisk1 | <PROTECTED> |
19:21.11 | asterisk1 | <PROTECTED> |
19:21.11 | asterisk1 | <PROTECTED> |
19:21.44 | ManxPower-work | Ugh! Apparently my new script is not actually smarter than a salesperson. |
19:22.05 | ManxPower-work | asterisk1: "core show applications" "core show functions" (function names are all UPPERCASE) |
19:22.14 | Superbartt | is it that dumb ManxPower-work? |
19:23.34 | ManxPower-work | Superbartt: I didn't think so, but I guess I was wrong. |
19:24.26 | asterisk1 | CALLERID(datatype[,<optional-CID>]) Gets or sets Caller*ID data on the channel. |
19:24.29 | axelilly | How do you do multiple time checks, like 8-8 M-F and 9-5 Sat? |
19:25.24 | [TK]D-Fender | axelilly: Do multiple IF's |
19:29.55 | ChannelZ | Katty: What'd you buy? |
19:32.00 | ManxPower-work | Does anyone know of a way to force Asterisk to ignore inband indications and just make Dial hangup? |
19:32.49 | ManxPower-work | (this is on a PRI, Asterisk 1.4.recent |
19:33.36 | Katty | ChannelZ: girly stuffs. |
19:33.41 | Katty | ChannelZ: unmentionables. |
19:34.12 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
19:35.24 | KaneHau | guys... ok, I installed DAHDI and reinstalled the non-beta asterisk - and THANK YOU VERY VERY MUCH - the FXO module now answers the ring and Echo() works for me |
19:35.29 | KaneHau | a BIG mahalo nui loa! |
19:36.50 | Geminizer | hello all. question -- when Dial(...) is called from a dialplan, it blocks. When Dial(...) is done blocking, the DIALSTATUS variable is immediately available for reading, correct? |
19:37.51 | ChannelZ | should be |
19:38.09 | ManxPower-work | Geminizer: Correct. |
19:38.12 | Katty | ChannelZ: black polka dots, and cherries, if you must know. |
19:38.32 | ManxPower-work | In fact DIALSTATUS is set even before the call terminates, you just can't usually access it since Dial blocks. |
19:38.45 | Katty | ChannelZ: oddly enough, i can't even find it on their website :< |
19:39.08 | Geminizer | but it is always guaranteed to store some value, regardless if the Dial(...) succeeded or not? |
19:39.14 | ChannelZ | That's OK I have a pretty good imagination |
19:39.19 | Katty | k |
19:40.45 | ChannelZ | Geminizer: I don't use words like 'guarantee' but it's at least supposed to always contain something interesting |
19:41.07 | Katty | ChannelZ: they were on sale, buy 1 get 1 50% off. |
19:41.12 | Katty | ChannelZ: regular price is 45 each |
19:41.19 | Geminizer | got it... thanks |
19:41.41 | Katty | ChannelZ: but then i got side tracked at american eagle. |
19:42.14 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
19:42.19 | *** join/#asterisk ocnarf (n=chatzill@114.108.194.153) |
19:42.24 | ChannelZ | meh. Why don't gun shops do 'buy 1 get 1 50% off' sales? |
19:42.27 | ChannelZ | pouts |
19:42.53 | Geminizer | they may offer that deal at blow outs |
19:43.35 | ocnarf | hi everyone, need help. i want callers not be able to join the queue when all agents are enage. is it possible? |
19:43.46 | ChannelZ | Anyone have OSX 10.6? |
19:43.48 | benngard | "hardcore" testing of ooh323 channel, my vife is using it atm ;) |
19:44.08 | [TK]D-Fender | ocnarf: Go read the sample queues.conf. there are rather clear looking options in there |
19:44.12 | benngard | calling mother iin law and vice versa |
19:45.20 | ocnarf | D-Fender: i tried joinempty=no but it doesnt work |
19:46.13 | ChannelZ | that's the opposite |
19:46.19 | Katty | ChannelZ: well. |
19:46.27 | Katty | ChannelZ: some have ammo discounts when you get a gun. |
19:46.39 | Katty | ChannelZ: i know a couple places around here ryan has gotten a gift card for 20 bucks when he purchased one. |
19:46.46 | Katty | course that was a rather pricey rifle |
19:46.53 | Katty | it's just sitting in the closet, collecting dust. |
19:47.12 | ChannelZ | ocnarf: look at maxlen maybe |
19:47.14 | Katty | there's ammo in it, but it's not the word.... |
19:47.18 | Katty | uhmmm |
19:47.23 | Katty | not loaded? |
19:47.27 | Katty | it's loaded |
19:47.37 | Katty | but..you'd have to pull that one thing back to get the ammo into the other thing |
19:47.46 | *** join/#asterisk mrbnet (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
19:47.48 | Katty | my description must be hilarious :P |
19:47.55 | ChannelZ | so it's loaded, but without one in the chamber |
19:48.17 | Katty | yeah. |
19:48.35 | Deeewayne | ~roulette |
19:48.36 | infobot | ACTION watches deeewayne pull the trigger: Click! |
19:48.38 | mrbnet | Can anyone recommend a sip provider in the UK? |
19:48.41 | [TK]D-Fender | ocnarf: there is another option and value for there... |
19:49.46 | ocnarf | D-FEnder: tried both joinempty =no and leavewhempty =yes, still doesnt work. |
19:50.03 | [TK]D-Fender | ocnarf: And you still have managed to skip things... |
19:50.24 | ocnarf | D-Fender: any thoughts? |
19:50.28 | gr0mit | mrbnet. i do sip in UK... |
19:50.52 | gr0mit | what are you lookkingfor? |
19:51.51 | [TK]D-Fender | ocnarf: READ IT AGAIN |
19:52.11 | raden_work | Naikrovek, heya bro |
19:52.53 | ChannelZ | wispers "hint: strict" |
19:53.44 | mrbnet | gr0mit: I have a customer opening an office there an my current provider does not offer numbers outside the US. I am looking for some UK numbers |
19:54.07 | gr0mit | no probs - pm me. i can do most UK areas |
19:59.17 | *** join/#asterisk batphone (n=will@rrcs-24-153-211-180.sw.biz.rr.com) |
19:59.31 | batphone | im looking at these INVITES from a customer |
19:59.50 | batphone | they show the To: field to be completely different than the INVITE URI |
20:00.16 | batphone | in fact, it says To: <sip:5551212@customer.ip.address> |
20:00.22 | batphone | rather than to my address |
20:01.01 | batphone | is this right? |
20:06.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
20:07.13 | *** join/#asterisk ticoit (n=ticoit@201.205.153.166) |
20:09.09 | ChannelZ | guess everyone is at lunch |
20:11.52 | jblack | Nah. I'm thinking about lunch. |
20:12.09 | jblack | I want a big burger. |
20:12.16 | ChannelZ | me too, I'm hungry |
20:12.21 | ChannelZ | Burrito, methinks |
20:12.42 | jblack | Ever been to panchero's ? |
20:12.49 | ChannelZ | never heard of it |
20:13.17 | jblack | It's teh r0x0rs |
20:13.50 | ChannelZ | hmm Grand Junction or Montrose. Not my neck of the woods |
20:14.37 | Kobaz | grand junction is fun |
20:14.39 | Kobaz | good mountain biking |
20:14.41 | ChannelZ | looks like another Chipotle but with more interesting plates |
20:14.46 | jblack | They use fresh stuff, and even make their tortillas on the fly |
20:14.53 | cusco | mrbnet: voipuser.org |
20:22.17 | *** part/#asterisk torrancew (n=torrance@ip70-186-186-21.br.br.cox.net) |
20:23.12 | shamelessn00b | ChannelZ: its working now |
20:23.14 | shamelessn00b | :D |
20:23.22 | shamelessn00b | dahdi calls |
20:23.25 | ChannelZ | it? |
20:23.29 | shamelessn00b | I tweaked 2 parameters |
20:23.33 | shamelessn00b | 240 calls |
20:23.36 | shamelessn00b | 240 agis |
20:24.01 | shamelessn00b | increasedthe chunk size to 80 |
20:24.20 | shamelessn00b | and changed the max connection limit on my DB server xD |
20:24.34 | shamelessn00b | wanpipe chunk size |
20:25.46 | ChannelZ | I don't know what that is but glad it worked |
20:29.16 | [TK]D-Fender | ChannelZ: Changes the interrupt load drastically |
20:29.50 | ChannelZ | wanpipe == sangoma's hardware? |
20:31.38 | ChannelZ | so it was a throwup of the PCI bus like I thought last night |
20:32.32 | [TK]D-Fender | ChannelZ: nothing says " I love you" like projectile vomit.... |
20:33.18 | ChannelZ | or being gagged while trying to, really.. |
20:33.20 | ChannelZ | :) |
20:35.03 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
20:37.14 | bmoraca | autoeroticasphyxiation...i think there's a support group for that |
20:45.09 | bmoraca | crazy german guy cracked GSM's encryption |
20:45.21 | voipmonk | crazy? |
20:45.39 | TSM | still need a lil bit of power todo it in realtime |
20:46.06 | TSM | but give 6months and with the next gen crop of cuda cards etc it will be much easier |
20:46.06 | KaneHau | Ok... I'm reading the asterisk pdf book... the examples work fine for me. However, my applciation needs to PLACE calls, never answer them. I don't see in the book how to actually initiate a brand new phone call to an outside number programtically (I'm writing in 'C') |
20:46.14 | ChannelZ | damnit, NOW how will I order my explosive underwear? |
20:46.39 | KaneHau | channel: don't bother - TSA's next decision will be to make us all "FLY NAKED" |
20:46.46 | KaneHau | should cut down on obesity :) |
20:47.40 | bmoraca | if you're having conversations that you don't want anyone else to overhear, you probably shouldn't have them over a phone anyway. i don't get what the fuss is |
20:48.23 | bmoraca | KaneHau, you can do that three ways: AMI, call files, and the "Originate" CLI command |
20:48.42 | KaneHau | bmoraca: thank you for the pointers |
20:48.43 | ChannelZ | I think he's actually writing an * application |
20:48.59 | ChannelZ | or no? |
20:48.59 | KaneHau | well, I have a C application that needs to make phone calls |
20:48.59 | shamelessn00b | bmoraca: A5 was cracked ages back |
20:49.08 | KaneHau | (needs to call scientists to report on alarm conditions) |
20:49.23 | ChannelZ | oh. I thought you meant you were writing an interneral * app. |
20:49.30 | bmoraca | KaneHau, your C application...is that WITHIN asterisk or does it just need to INTERFACE with asterisk? |
20:49.42 | shamelessn00b | there are embedded systems that can sniff GSM calls in realtime |
20:50.03 | shamelessn00b | k gais, me off |
20:50.05 | shamelessn00b | tc gnite |
20:50.05 | KaneHau | just needs to INTERFACE to asterisk to place the call, and monitor for DTMF tones |
20:50.34 | KaneHau | it places a call, speaks a custom message (created on the fly) and then awaits for DTMF tones to make a few decisions |
20:50.48 | ChannelZ | Robodialer spam?! |
20:50.49 | bmoraca | KaneHau, you won't be able to monitor for DTMF tones unless the audio path goes through your application. |
20:51.10 | KaneHau | the number it calls is unknown until the moment the call is placed |
20:51.10 | KaneHau | ok |
20:51.10 | KaneHau | so that means I'm writing a * application, right? |
20:51.32 | bmoraca | KaneHau, why bother with that when said applications already exist? |
20:51.58 | [TK]D-Fender | [15:46]<KaneHau>channel: don't bother - TSA's next decision will be to make us all "FLY NAKED" <- Not so... |
20:52.11 | [TK]D-Fender | spins up Pat Benatar's "Sex As A Weapon" |
20:52.24 | KaneHau | bmoraca: this has to interface into one of our huge telemetry systems - much easier if I custom design it |
20:52.24 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
20:52.33 | KaneHau | we alraedy have it for TAPI - just replacing TAPI with * |
20:52.41 | bmoraca | KaneHau, use AGI for that. |
20:52.50 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
20:53.36 | bmoraca | KaneHau, unless you want to create a custom SIP UA...but I would see stabbing myself in the eye with a marlinspike as more entertaining and a better use of time :) |
20:54.50 | KaneHau | yes, I looked at AGI, but the examples are all from a standpoint of answering a call... how does the AGI actually initiate a call? |
20:55.03 | bmoraca | KaneHau, the two are not related at all |
20:56.14 | bmoraca | KaneHau, you HAVE to have an external application initiate the calls...whether by sending SIP messages or by using Asterisk's built-in methods (spool, AMI, originate). once the call is setup, everything else happens within Asterisk...which is where AGI comes in if you need to integrate with another application (or func_odbc if just a database) |
20:56.46 | KaneHau | thank you for the pointers |
20:56.50 | KaneHau | much reading ahead |
20:56.55 | ChannelZ | it is definately the time of lunch. |
20:56.57 | bmoraca | KaneHau, the audio path, however, does NOT go through the application that initiates the call unless that application was created as a SIP UA |
20:57.13 | bcrisp | um |
20:57.22 | KaneHau | actually, the manual said audio was available on file descriptor 3 |
20:57.24 | bmoraca | KaneHau, in that case, Asterisk is nothing more than a SIP gateway. |
20:57.26 | KaneHau | is that not the case? |
20:58.09 | KaneHau | well, atcually I don't need the audio, I just need to know what DTMF tones were hit by the user |
20:58.51 | bmoraca | KaneHau, it may be, but that's only going to be available within Asterisk (if you plan to write this as an Asterisk module), but you still need an external program to initiate the calls |
20:59.10 | KaneHau | ok, thanks |
20:59.12 | bmoraca | KaneHau, DTMF is sent with the audio...or out of band, but still follows the audio path |
20:59.27 | bmoraca | generally |
21:01.24 | bmoraca | KaneHau, there are two ways, really, to do what you want...you can create a SIP UA that does all processing and call setup itself (including playing messages and capturing audio (DTMF)), or you can create an application that tells asterisk to setup a call and supplies certain variables over AMI and use Asterisk itself to actually play the audio and capture DTMF (then you can do anything you want with it via AGI or func_odbc or anything e |
21:01.24 | bmoraca | lse) |
21:01.31 | *** join/#asterisk alfa202 (n=svelluto@dhcp-0-9-e8-4a-96-80.cpe.quickclic.net) |
21:01.43 | bmoraca | option 2 would most likely be a lot easier to put together |
21:01.50 | KaneHau | I think the 2nd solution you gave is probably what I want |
21:02.16 | KaneHau | my program needs to create the script on the fly, then initiae the call and have astrisk then deal with everything (for the most part) |
21:03.11 | bmoraca | KaneHau, that shouldn't be too difficult. AMI is how you'll initiate the call, and AGI is likely where you'll do your processing (though you don't have to use AGI) |
21:03.25 | KaneHau | btw, unrelated... the VIKING Advanced Line Simulator I'm using rocks! Very nice for testing out this hardware |
21:03.26 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
21:03.40 | bcrisp | i like viking stoves |
21:03.43 | KaneHau | got it, thanks |
21:04.08 | KaneHau | this is the Model DLE-300 - has two phone ports on it and handles creating a dial tone on both ports. Can simulate 911, etc |
21:04.47 | bmoraca | i use an Adtran TA900 with another asterisk server (virtualized) to simulate the PSTN, generally |
21:05.17 | *** join/#asterisk Tim_Toady (n=moi@77.49.136.12.dsl.dyn.forthnet.gr) |
21:05.24 | KaneHau | probably would have done that if I was already * savy - this got me up and running without having to be * smart |
21:06.02 | [TK]D-Fender | KaneHau: I find a < $50 Linksys PAP2T-NA to be a more than adequate line simulator.... |
21:06.05 | *** join/#asterisk akira2014 (n=chatzill@220.Red-88-6-197.staticIP.rima-tde.net) |
21:06.16 | KaneHau | reads the AGI chapter |
21:06.38 | KaneHau | I had money to burn |
21:06.57 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
21:07.44 | *** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com) |
21:09.13 | akira2014 | can some one tell how to migrate a sip conf from asterisk 1.4 to asterisk 1.6 |
21:09.15 | akira2014 | http://pastebin.com/d16b4effc |
21:09.26 | akira2014 | this is part of my sip.conf |
21:09.31 | akira2014 | thk's in advance |
21:09.37 | Defraz | bmoraca: Do you have any configures for an Adtran 924. Trying to use it as a SIP gateway |
21:09.57 | Defraz | I have a PRI and an adtran 924 and trying to use that to sip it to my * server. |
21:10.13 | Defraz | Can do it on a cisco but can't figure out this Adtran. |
21:12.09 | *** join/#asterisk ticoit (n=ticoit@201.205.153.166) |
21:14.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:14.59 | KaneHau | bmorace: AMI looks to be what I'm looking for, thank you greatly |
21:20.42 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
21:23.24 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
21:25.52 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
21:28.01 | ManxPower-work | Defraz: standby |
21:29.13 | [TK]D-Fender | \o/ |
21:29.29 | [TK]D-Fender | Finally got asterisk.org 's downloads page looking right.... |
21:31.05 | ManxPower-work | Defraz: We use massive amounts of Adtran gear. I'm asked about the configs, but my boss is AFK |
21:32.04 | Qwell | [TK]D-Fender: what was wrong with it? |
21:32.38 | Akiraa | What's the latest edition of "Asterisk: The Future of Telephony" ? I have a pdf of Edition 2 (2007) |
21:32.39 | [TK]D-Fender | Qwell: Lack of direct downloads link. Section for sounds with no link to them either (which is highly desirable for non-internet site installs), etc |
21:33.05 | [TK]D-Fender | Qwell: Qwell And the earlier correctio for complete lack of link to Addons. |
21:33.38 | [TK]D-Fender | Akiraa: thats it |
21:34.02 | Akiraa | so no significant changes since 2007 |
21:35.02 | [TK]D-Fender | Akiraa: to the BOOK? Its paper :) |
21:35.24 | [TK]D-Fender | Akiraa: It isn't an "official" doc anyway. Thats what the tarballs are for |
21:35.39 | Akiraa | [TK]D-Fender: no, to Asterisk itself, 2 years is a long time for software |
21:36.09 | luckyaba | What is the best router in your opinion for VOIP? |
21:36.22 | [TK]D-Fender | Akiraa: the book was circa 1.4. We are at 1.6.2 branch now. Feel free to activate all those dormant neurons... |
21:36.30 | [TK]D-Fender | luckyaba: iptables |
21:37.06 | luckyaba | [TK]D-Fender, haha, fair enough. How about something that we can put in front of a clients network that is a bit easier to setup and manage? |
21:37.53 | [TK]D-Fender | luckyaba: * doesn't really care, but PIX & D-Link's = trouble |
21:38.09 | [TK]D-Fender | luckyaba: Your typical Linksys home router tends to work just fine |
21:38.16 | voipmonk | i can smack a dlink into submission |
21:38.19 | ManxPower-work | luckyaba: Cisco 2621XM |
21:38.20 | voipmonk | but why? |
21:38.21 | voipmonk | :) |
21:38.40 | ChannelZ | mmmm burritoooooo |
21:39.00 | luckyaba | you guys have good feedback on maybe using a Linksys or Buffalo with DD-WRT? |
21:39.11 | ManxPower-work | luckyaba: You said good router. |
21:39.22 | luckyaba | lol |
21:39.34 | luckyaba | touche |
21:39.41 | ManxPower-work | "best consumer grade" is what it looks like you are looking for. |
21:39.59 | Chainsaw | You're going to end up with a Billion or something similar. |
21:40.06 | ManxPower-work | I personally use an old Cisco 175x as my router. |
21:40.08 | Chainsaw | Zyxel used to be decent, but lost their way around the Zywall 2 plus. Avoid. |
21:40.21 | Chainsaw | ManxPower-work: Overpriced for what you get, but yes, reliable. |
21:40.25 | luckyaba | This is for a solution provided to clients ManxPower-work |
21:40.32 | luckyaba | so it has to be a product readily available |
21:40.35 | ManxPower-work | Chainsaw: like $50 on ebay |
21:40.52 | luckyaba | used? |
21:40.55 | luckyaba | warranty? |
21:40.59 | luckyaba | supported? |
21:41.05 | ManxPower-work | Yes. No warrenty. Buy two of them if you want that. |
21:41.15 | ManxPower-work | luckyaba: like you'd ever get an support from Linksys support. |
21:41.17 | luckyaba | client(s) |
21:41.25 | TSM | i like sonicwall units, the total secure units are fairly good TZ210 i think arement to be good, ive got PRO2040 HA units |
21:41.25 | luckyaba | we will need to be buying a lot more than 2 |
21:41.37 | luckyaba | Sonicwall is what we currently support |
21:41.43 | luckyaba | and they are put simply.... Garbage |
21:41.48 | Chainsaw | TSM: I don't know, it sounds like an air conditioning unit. |
21:42.09 | TSM | they are good, ive had a load of them over the years |
21:42.13 | TSM | no problems |
21:42.40 | TSM | the Total Secure units have full IPS/GAV etc protection plus warranty etc.. |
21:43.21 | TSM | most of them now come with HA function built in and you can get the second unit a a fraction of the cost of a normal one |
21:43.28 | *** join/#asterisk clyrrad (n=IceChat7@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
21:43.31 | Chainsaw | TSM: High Airflow? |
21:43.46 | luckyaba | TSM, They handle VOIP horribly |
21:43.55 | luckyaba | Sonicwall themselves will admit it |
21:44.00 | TSM | ive had no issues, but then im running my VOIP in DMZ |
21:44.37 | luckyaba | Sonicwall is making an effort to resolve the problems but its going to be a long time before that happens |
21:44.39 | TSM | transparent DMZ is easy to use |
21:44.44 | luckyaba | and we need a reliable solution for our clients |
21:45.03 | TSM | its reliable, i dont know why people have had issues, mabey yr talking about old software |
21:45.05 | luckyaba | That is a lot of money in extra setup time on the clients dime |
21:45.18 | ariel_ | firewall I like taking old pc with 2 nic's and putting Endian on it. |
21:45.19 | luckyaba | because then your talking about configuring iptables |
21:45.28 | luckyaba | so at that point we might as well throw the Sonicwall away |
21:46.09 | luckyaba | areay, personally I am with you on that but from a company's perspective I won't get them on board |
21:57.06 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
22:10.09 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
22:13.55 | Geminizer | why does Playback(exit_status_6-eng) work but not Playback(exit_status_6-${lang}) ? |
22:14.09 | Geminizer | where $lang = "eng" |
22:14.29 | Qwell | How did you set lang? |
22:15.23 | Geminizer | it's in a .call file as: Set: lang=eng |
22:15.36 | ManxPower-work | try setting lang to en not eng |
22:15.49 | ManxPower-work | Ah, you said it works. |
22:16.40 | Geminizer | right... it works when I use the full name. When I introduce a variable as part of the name, it fails.. |
22:16.52 | Qwell | Are you setting the var on the right channel? |
22:18.05 | ManxPower-work | Asterisk's language is "en", which is what I was thinking. Perhaps I'm caffeine deprived. 8-) (no |
22:18.14 | Geminizer | yes, along with all the other channel variables which work |
22:18.26 | Qwell | (tip: you need a channel in order to use a variable. calling Application: Playback from the callfile won't work) |
22:19.18 | ManxPower-work | Geminize: You set it in the call file as Set: lang=eng and not Set: lang="eng"? |
22:19.43 | Geminizer | correct.. the first way you mentioned is how I have it |
22:19.51 | ManxPower-work | Remember, most times quotes are literal in Asterisk |
22:20.32 | ManxPower-work | Geminizer: Put a Noop in the dialplan to show you the values of the variables you set. |
22:20.45 | ManxPower-work | If you're running Playback, I'm assuming you're in the dialplan. |
22:20.51 | Geminizer | ahh, I think I know what the problem is... one moment |
22:24.13 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
22:24.23 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
22:25.45 | Geminizer | got it... I hadn't realized the filenames were named differently... |
22:30.49 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
22:32.55 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
22:36.32 | ChannelZ | Speaking of filenames |
22:37.02 | ChannelZ | Why does * say that it's playing "blahblah.slin" when really it's playing a file called "blahblah.wav"? |
22:38.56 | bmoraca | Defraz, if you're still here and interested, the TA900s were not really meant to be used in that direction. if the problem is that the PRI is not coming up (remember, it has to be in port 3 or 4), it could be that you need to use a T1 crossover cable...depending on how your jack was wired. |
22:45.24 | *** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath) |
22:46.40 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
22:54.09 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
22:58.23 | citywok | ChannelZ: my guess is it decided to use the wav because it required no/less transcoding |
22:58.28 | citywok | and it made that decision after announcing it? |
22:59.02 | ChannelZ | what I mean is that physically on disk the file is called whatever.wav but when voicemail plays it it calls it whatever.slin |
22:59.27 | citywok | odd |
22:59.43 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
22:59.45 | citywok | sounds like a trivial bug, lol |
22:59.58 | *** join/#asterisk ManxPower (n=ewieling@216.186.151.147) |
23:00.31 | [TK]D-Fender | ChannelZ: perhaps you could pastebin the complete call from the call to Playback through the end, along with the dump of your sounds folder... |
23:02.55 | file | it's actually <name>.<format being fed to the channel> |
23:05.39 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
23:09.57 | ChannelZ | well it's not that it doesn't work, it's just that it looks funny |
23:10.04 | ChannelZ | -- <DAHDI/1-1> Playing '/var/spool/asterisk/voicemail/default/200/unavail.slin' (language 'en') |
23:25.08 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
23:43.39 | *** join/#asterisk johnyjj2 (n=mainacco@p9g117.traco.pl) |
23:44.14 | johnyjj2 | hello :) |
23:45.19 | johnyjj2 | can somebody help me with configuring Asterisk Win32? |
23:46.12 | [TK]D-Fender | johnyjj2: asteriskwin is not supported here. for general configuration, the common docs should still apply |
23:49.54 | johnyjj2 | [TK]D-Fender: thank you, anyway if somebody would be willing to have a look, I made some print-screens from my configuration here http://www.speedyshare.com/data/401343461/20015137/72160770/foto.rar i'm available either here or by mail johnyjj2@gmail.com I just wanted to check Asterisk configuration with X-lite, thanks |
23:52.52 | [TK]D-Fender | johnyjj2: Want to check it... USE IT |
23:52.58 | [TK]D-Fender | johnyjj2: And watch * CLI |
23:55.20 | johnyjj2 | That's the difficulty. I would prefer using Asterisk on Linux but I need to use in on Windows, this is why I downloaded Asterisk Win32. It created icon on desktop for WillVoice PBX Manager but this CLI all the time says "Unable to connect to remote asterisk". |
23:57.11 | [TK]D-Fender | johnyjj2: "need to use it on windows"? What insanity is the basis of this? |
23:57.59 | [TK]D-Fender | johnyjj2: And now you are bringing 3rd party management tools into this... lovely |
23:58.07 | johnyjj2 | this is not my server and the owner/admin of the server asked me to install asterisk on it and configure it for ivr/asr |
23:58.45 | TSM | tell him to foff |
23:58.55 | TSM | i mean the owner/admin |
23:58.56 | [TK]D-Fender | johnyjj2: Well this 3rd party tool seems to want to connect via AMI. Go set up your manager.conf |
23:59.51 | johnyjj2 | this asterisk 32win is porting asterisk to windows with cygwin, it installed this willvoice pbx manager (3rd party tool) so i guess that's the only available cli for asterisk win32 but i may be wrong |