| 00:00.30 | LemensTS | can u do a gotoif NUMBER =< NUMBER ? |
| 00:01.07 | *** join/#asterisk dysinger__ (n=tim@71-20-35-99.war.clearwire-wmx.net) |
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| 00:07.39 | p3nguin | So, about the comfort noise... the provider says it's a problem on my end rather than on the client address listed in the notice (which is their server). Any thoughts? |
| 00:19.20 | *** join/#asterisk sam555 (n=chatzill@udp153808uds.hawaiiantel.net) |
| 00:19.25 | sam555 | hello all! |
| 00:19.35 | sam555 | i'm testing out asterisknow |
| 00:19.51 | sam555 | i'm trying to make a voice mail recording and the recording sounds like it's in an out |
| 00:19.52 | sam555 | any ideas? |
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| 00:22.00 | Maximo | hi, what is the name of that store for ipphones? |
| 00:22.29 | phix | iptunes? |
| 00:22.57 | Maximo | something like iphones-warehouse something..... |
| 00:22.59 | Maximo | is it? |
| 00:24.38 | jaytee | www.telephonydepot.com or www.voipsupply.com |
| 00:25.00 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 00:25.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:25.17 | Maximo | jaytee: thansk alot |
| 00:25.20 | sam555 | this might be the answer to my question http://www.freepbx.org/support/documentation/faq/echo-info |
| 00:25.34 | Maximo | there's also another one, a nice one too |
| 00:27.38 | rollot | Could anyone tell me how to configure dialplan.xml for 79xx so that there is no delay after dialing *.. for feature codes? (Have timeout=0, tried multiple configurations but suspect it's the *=wildcard that's throwing it off) |
| 00:28.31 | el_critter | Hi, how can I know if a determinate patch is already applied on 1.6.1.6? |
| 00:30.57 | manxpower | each release has a Changelog file. I believe most of the bug numbers are listed with the patch |
| 00:31.25 | *** join/#asterisk kibombo (n=kibombo@unaffiliated/akafubu) |
| 00:31.42 | kibombo | hey you voice freaks, i need a good channel to discuss sip topics |
| 00:32.03 | kibombo | someone said that you can do some type of conference calling via sip |
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| 00:37.40 | el_critter | manxpower: thanks |
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| 00:45.44 | LemensTS | i launch an agi program in the dialplan, after that is done it comes back to the dialplan down to the next line, how do i see how many seconds the call has been up at that time? from what i read you can do it in ami with BillableSeconds: |
| 00:46.00 | tgunr | i have been at this for hours, sigh... Is there no way to see what pattern is being matched against on a connection? I have set core debug and verbose to highest level and all i see is "Call from '100' to extension '5306759045' rejected because extension not found." |
| 00:47.22 | LemensTS | tgunr: can you show us the line in extensions.conf that you think it should be going to |
| 00:47.35 | LemensTS | have you done an extensions reload |
| 00:47.40 | tgunr | [default] |
| 00:47.51 | tgunr | yes, several hundred times |
| 00:47.58 | LemensTS | lets see the line |
| 00:47.59 | tgunr | i can paste it |
| 00:48.01 | LemensTS | yea |
| 00:50.44 | xa0z | rtp.c:786 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.63.41.218 |
| 00:50.45 | *** join/#asterisk s14ck (n=s14ck@190-76-109-77.dyn.movilnet.com.ve) |
| 00:50.51 | xa0z | Any one have an idea as to how to stop this? |
| 00:51.41 | tgunr | http://paste.lisp.org/+1UYR |
| 00:53.26 | tgunr | all of my internal extension work fine, can't get any outbound calls to go to my provider |
| 00:53.34 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 00:53.37 | LemensTS | tgunr: u said it goes to default? |
| 00:54.10 | LemensTS | dont see anything in there that 5306759045 would go out on |
| 00:54.12 | tgunr | well, I'm not sure where it is going, doesn't seem to match anything |
| 00:55.27 | LemensTS | exten => _X.,1,Dial(SIP/sip-provider/${EXTEN}) |
| 00:55.45 | DaveCanoe | I've been dumping packets at several points on the network to diagnose a problem in a system where we have phone --> onsite asterisk --> datacenter asterisk --> SIP provider. The dump between the phone and the onsite asterisk has both audio streams. The dump at the datacenter shows one stream suddenly stops. The onsite asterisk kicks out a "Auto-congestion" message in the log and it seems like the onsite asterisk just stops passing that particular c |
| 00:55.46 | DaveCanoe | all (other ongoing calls not affected). |
| 00:55.46 | tgunr | shouldn't it match CallingRule_LocalAC |
| 00:56.10 | tgunr | at the bottom, that was created by the *GUI |
| 00:56.22 | LemensTS | gui? |
| 00:56.25 | LemensTS | is this freepbx |
| 00:56.57 | DaveCanoe | ... I'm not sure if the auto-congestion message is related, but I can't find anything pertinent in a several hour long google search. |
| 00:57.09 | tgunr | hmm, my UI shows #asterisk |
| 00:58.12 | LemensTS | this is asterisk. |
| 00:58.46 | LemensTS | no gui |
| 00:58.59 | DaveCanoe | the logic for the auto-congestion message seems to be not seeing a packet for 4x some interval. The G.711 traffic is 20ms per packet and there is a 80ms gap coming from the carrier --- which would seem to chuck out the message --- but this still doesn't explain why the onsite asterisk stops forwarding the RTP stream _and_ it never sends out the SIP BYE message, either. |
| 00:59.47 | tgunr | sorry, got dyslex for a moment, thought you meant i was on wrong channel :) The extension.conf was created by Asterisk-GUI but I have been editing it manually. |
| 01:00.00 | DaveCanoe | also... the stream that the onsite asterisk stops forwarding is the one from the phone out to the provider --- which is not the stream with the long packet interva. |
| 01:00.45 | LemensTS | tgunr are you using tribox or freepbx? |
| 01:01.22 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.226) |
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| 01:02.02 | LemensTS | either way you got to add the routing in the gui |
| 01:02.18 | LemensTS | cant edit extensions.conf by hand when you use a gui pbx |
| 01:02.19 | nyranger | I am trying to do some arithmatic in my dialplan... It worked in an older version of asterisk, but I cant seem to get it to work now... |
| 01:02.33 | nyranger | <PROTECTED> |
| 01:02.40 | tgunr | neither, Asterisk-GUI, ok will read some more |
| 01:03.01 | nyranger | <PROTECTED> |
| 01:04.17 | DaveCanoe | Any real experts on? |
| 01:04.34 | *** join/#asterisk propellerhead (n=yogurt2u@host42.190-231-29.telecom.net.ar) |
| 01:04.53 | LemensTS | nyranger: im trying to get call time in the dialplan too. how are you getting the DUR ? |
| 01:05.10 | *** part/#asterisk kibombo (n=kibombo@unaffiliated/akafubu) |
| 01:05.11 | LemensTS | can u pastebin your code? |
| 01:09.31 | nyranger | LemensTS: sure, but my code doesnt work |
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| 01:28.00 | nyranger | LemensTS: to get the call duration, you can use: ${CDR(duration)} |
| 01:28.40 | LemensTS | nyranger: i can do that at anytime in the dialplan and it will give the length? |
| 01:29.08 | nyranger | LemensTS: I believe so... According to http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
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| 01:34.04 | manxpower | also see /path/to/src/asterisk/doc/channelvariables.txt for the most up to date list. |
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| 01:54.16 | DetroitArmz | Hi folks, newby to this stuff, have a new system 1.4.26.2, couple of local IP Phones (7940's) and a SIP Trunk turned up. calls work between phones as well as inbound and outbound calling. Today I added (after some pain and suffering) an ATA-188 SIP. The exten works in same context as aother phones with same sip conf |
| 01:54.30 | DetroitArmz | but when I try outbound calling I get a fast busy |
| 01:55.11 | DetroitArmz | error in console says: WARNING[7425]: pbx.c:1833 pbx_extension_helper: No application '' for extension (phones, |
| 01:55.31 | DetroitArmz | and then the number I called (a 9,NANP number) |
| 01:56.36 | DetroitArmz | is there something special I have to do when adding ATA based phones that would impact routing? (ata and phones are in same context) |
| 01:58.04 | nyranger | exten => h,3,CURL(http://${ip}/ict/call-end.php,line=${line}&dur=${DUR-${line}}&stat=${STAT-${line}}&msg=${MSG-${line}}) |
| 01:58.23 | nyranger | that line just does not want to work. I keep getting an error: [Sep 8 20:52:58] WARNING[8846] pbx.c: No application '${CURL' for extension (out1con, h, 3) |
| 02:01.30 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 02:01.30 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:05.41 | DetroitArmz | doh! nevermind, forgot to add in the outbound callerid function. Now it's working. |
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| 02:30.31 | xpot | Question: anyone offer assistance on a2billing here? |
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| 03:32.06 | manxpower | nyranger: you have a syntax error. I suspect you have ${CURL) instead of ${CURL} |
| 03:32.29 | manxpower | oh, yes, if you paste just that one line I might he able to help |
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| 03:47.50 | kb3ien | i'm getting parse errors on my e164 NAPTR records, is there a good reference? or better yet a validator on the intertooobs? |
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| 03:56.57 | Zuchmir2 | anyone know what params to pass to configure for * on sheevaplug? tried: /configure --build=arm got the following error: http://pastebin.com/d29144d85 |
| 04:01.35 | kb3ien | reads rfc 2915 in all its glory. |
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| 04:45.36 | manxpower | ~answers |
| 04:45.37 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
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| 04:52.23 | kb3ien | is feeling better now enum dialing is working with multiple domains. |
| 04:58.00 | DaveCanoe | OMFG. It never ceases to amaze me who meticulously badly asterisk is written. |
| 04:59.28 | DaveCanoe | This type of cavalier disregard for not only lessons learned, but for even barely acceptable practices (good or best practices are just plain out-of-reach) is very special. |
| 04:59.38 | denon | submit a patch |
| 05:00.58 | DaveCanoe | hah! you fool! |
| 05:01.29 | denon | careful, cowboy |
| 05:01.46 | DaveCanoe | Asterisk doesn't take patches. Why do you think there are more than 20 forks of asterisk. It's literally infamous for not accepting patches... especially those that have nothing to do with digium hardware. |
| 05:01.56 | *** join/#asterisk martianixor (i=martianx@gateway/shell/blinkenshell.org/x-eybgwoxtnzzzynsf) |
| 05:02.10 | martianixor | hello again |
| 05:02.36 | martianixor | everyone |
| 05:03.01 | DaveCanoe | sigh. Just tired after debugging something that was discovered 4 or 8 months ago and still doesn't work (and was badly conceived in the first place). Actually thought I was typing in a friends channel... to tired to notice what tab is open. |
| 05:03.16 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 05:03.17 | TJNII | Uh-huh. |
| 05:03.41 | martianixor | guys so I've read that nat=yes helps only if the peer is behind nat and reaching an asterisk outside nat |
| 05:03.47 | martianixor | in my case |
| 05:03.57 | martianixor | both the peer and my asterisk are behind nat |
| 05:04.11 | martianixor | extensions unable to register |
| 05:04.11 | DaveCanoe | I asked earlier in the channel here about RTP oddities. How something triggering auto_congestion log entries causes asterisk to drop calls --- without even sending bye messages. |
| 05:04.23 | drmessano | martianixor: You need ports open |
| 05:04.26 | drmessano | ~sipnat |
| 05:04.27 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:04.35 | martianixor | drmessano: I have'em |
| 05:04.43 | drmessano | Follow the guide |
| 05:04.44 | martianixor | drmessano: all the needed ports actually, including RTP ones |
| 05:04.55 | DaveCanoe | ... and after ~4 hours of UTSL and Google it seems I've found a ports or two on the subject... |
| 05:05.00 | martianixor | drmessano: OK that means I'll retry |
| 05:05.11 | drmessano | peers and asterisk behind different NAT's with ports open is a workable solution |
| 05:05.32 | DaveCanoe | so... the hyperbole was meant for some friends (who might comiserate) ... but the criticism is valid. |
| 05:06.39 | martianixor | drmessano: thanks thought :-) |
| 05:06.44 | martianixor | s/thought/though/ |
| 05:06.57 | martianixor | w0w ! |
| 05:07.11 | denon | you frequently call your friends fools, eh? |
| 05:07.29 | denon | think it might be time to get a little sleep. :) |
| 05:07.43 | drmessano | DaveCanoe: Mistakes happen |
| 05:07.45 | martianixor | denon: a PM please ! |
| 05:08.14 | drmessano | s/Mistakes/Trollings/ |
| 05:08.19 | drmessano | :D |
| 05:08.20 | DaveCanoe | Well... I have a couple hundred lines of python before I knock off... this C hacking to find the asterisk bug was just a diversion demanded by other clients. |
| 05:08.22 | martianixor | :D |
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| 05:08.57 | DaveCanoe | heh. I don't normally troll. I have a few frends who are somewhat famous trolls... I'm actually famous for idling. |
| 05:09.11 | martianixor | denon: thanks anyway ! |
| 05:10.35 | DaveCanoe | I've been known to deride asterisk with most uncomplementary comparisons while subsequently admitting that it's the only object in that uncomplemantary class that we have at the momment. |
| 05:11.09 | DaveCanoe | as for friends and fools... there's no shame in being a fool from time-to-time. |
| 05:11.55 | *** join/#asterisk Tim_Toady (n=moi@adsl15-40.kln.forthnet.gr) |
| 05:11.59 | DaveCanoe | I'm sure the Bard has some quote or other on that topic... or if not him, then possible Chaucer. |
| 05:12.31 | drmessano | DaveCanoe: I've found that there has been much maturity brought to the Asterisk dev process over even the last year or so. Whether it's competition, necessity from growth, perhaps some lessons learned, or who knows what, things have changed.. Coming from me, thats kinda scary to hear. If you have a valid issue, I suggest hashing it out in Mantis and not just bitching about it on IRC |
| 05:14.32 | drmessano | DaveCanoe: I often reflect on the words of I. Drago in the fourth iteration of the "Rocky" motion picture series who said, "I must break you". |
| 05:14.43 | DaveCanoe | Mantis. Hrm. I've been all over the dev list and there some bug tracking stuff that appears in googling these things. Is "mantis" one of thoese? |
| 05:14.45 | drmessano | I find those words comforting and appropriate in times like these |
| 05:15.20 | drmessano | https://issues.asterisk.org/main_page.php <--- |
| 05:16.13 | DaveCanoe | I suppose "do timing right" isn't entirely a useful comment? Sigh. Effort. Probably has to wait for next week. |
| 05:17.22 | drmessano | DaveCanoe: Yes, it is always easier to complain than to submit and ticket which is actually the proper and useful beginning to the process of resolving most issues in any open source project |
| 05:17.39 | drmessano | s/and ticket/a ticket/ |
| 05:19.42 | Zuchmir2 | have full => notice,warning,error,debug,verbose in logger.conf, yet things like: VERBOSE[1797] logger.c: -- Executing... do NOT show up in /var/log/asterisk/full (only certain messages show up there) |
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| 05:22.24 | DaveCanoe | ... I would feel guilty if the previous researcher hadn't already submitted a patch with ample evidence that even if the patch weren't correct, the problem existed. |
| 05:27.27 | drmessano | https://issues.asterisk.org/view.php?id=14021 <-- This one? |
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| 06:18.13 | DaveCanoe | drmessano... that may be another problem that applies (unfortunately, the client has triggered a number of edge cases), but thank-you for that reference. |
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| 07:24.37 | martianixor | guys, if I'm trying to call my own extension "sip softphone supports 2 lines" and I'm getting extension not found, provided that I've added it to my extensions.conf and my softphone is registered with that extension, what could be the problem? |
| 07:30.02 | maour | martianixor: at least put your sip.conf and extension.conf somewhere |
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| 07:31.55 | martianixor | maour: I think It's a problem with the context |
| 07:32.13 | martianixor | maour: the context for the extension in my sip.conf file |
| 07:32.35 | martianixor | maour: CLI debuging mode is awesome |
| 07:32.43 | maour | martianixor: what do you expect of me ? i see nothing |
| 07:33.23 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:33.39 | martianixor | maour: the context of the extension I'm registered by and trying to call is normaluseraccess |
| 07:33.50 | martianixor | maour: and of course I don't have such context configured |
| 07:34.45 | martianixor | maour: I hope that's the problem though |
| 07:35.12 | martianixor | maour: yes sorry, I'll try to pastebin both |
| 07:35.22 | martianixor | maour: if the problem is persistent |
| 07:35.28 | maour | ok |
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| 07:38.59 | martianixor | maour: :D It's ringing |
| 07:39.12 | maour | :) |
| 07:39.13 | martianixor | maour: I like that twinkle client's sound |
| 07:39.33 | martianixor | maour: besides it has a CLI mode which is awesome over ssh for debugging |
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| 07:55.53 | hat_panda | hi, during a call my asterisk must re-register because the registration has expired. at this point asterisk locks up for 5 seconds until the registration has been sent and answered. When the registration tasks completes the rtp transmission is continued. |
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| 07:57.12 | hat_panda | anyone know anything about that issue? |
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| 08:20.59 | Jakobsen | Hi there. I have a strange issue with my Asterisk 1.6.0.9 running on CentOS 5.3. Every night just after midnight, all peers changes to "Lagged" and then 10 seconds later, they change to "Reachable" again. What can cause this behaviour? |
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| 08:35.02 | fiddur | Can I use callerid mapping for hint extensions, like exten => *11/123, hint, Custom:11_123 ? |
| 08:37.38 | fiddur | ...I guess not, since the phone doesn't send caller-id on a SUBSCRIPTION... :/ |
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| 09:02.20 | fiddur | Is there any way to uniquify hints based on the subscriber? E.g. all the phones use *11 for setting status to away, and I'd like to give the hint of *11 based on if that phone/user is away... ? |
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| 09:39.54 | tompaw | Hi Guys. I am having problems with NAT. What I did was: 1) I configured nat = yet in my users.conf, 2) I manually configured an rtp port range in rtp.conf and forwarded these ports on my router directly to my sip phone |
| 09:40.24 | tompaw | Now when I'm trying to call myself using a DID provider, I'm getting something like this: http://www.tompaw.pl/sip_nat.txt |
| 09:41.07 | tompaw | Why does Asterisk SDP my LAN IP to the remote party? |
| 09:50.07 | tompaw | Why doesn't Asterisk use its own ip? |
| 09:50.21 | kaldemar | tompaw: you need to define externip |
| 09:50.26 | kaldemar | ~sipnat |
| 09:50.27 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 09:51.50 | tompaw | thanks, will try now |
| 09:54.41 | tompaw | kaldemar: whoa, it did the trick! thank you :-) |
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| 10:21.30 | Zeeek | anyone using Sipgate in USA? |
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| 11:01.31 | buzzyd | Hi all |
| 11:02.36 | buzzyd | Does anyone know if a conference can be created that plays an 1 hour pre recorded message that people can dial into ? |
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| 11:11.16 | Druken | why would you want that? why not just have it play the whole message to each caller? |
| 11:12.03 | Druken | also, if i was looking for something like that, i wouldn't use a confrence, i'd use a context of music on hold and just send callers to MOH |
| 11:16.59 | buzzyd | So it appears like a live broadcast. |
| 11:17.13 | buzzyd | I've looking at auto dialout I think |
| 11:21.34 | Druken | if you want it to appear like a live broadcast then use MOH, as long as someone is listening it'll continue to play |
| 11:21.48 | Druken | otherwise it'll stop and continue on from where it left off |
| 11:23.23 | buzzyd | thanks for info. |
| 11:23.44 | Druken | if it's only one file, make sure you put it in the MOH file twice |
| 11:23.51 | Druken | otherwise it'll only play once |
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| 11:26.31 | *** join/#asterisk SparFux (n=raoul@e182020036.adsl.alicedsl.de) |
| 11:26.59 | SparFux | tzafrir_laptop: hey, cool patch zaphfc_no_owner :-) |
| 11:27.43 | tzafrir_laptop | actually, it has been excepted by upstream. I should update the zaphfc patch and drop it. |
| 11:29.08 | SparFux | anyway, this is nice. I was not into that faar enough to do the version checking :-) |
| 11:36.48 | tzafrir_laptop | well, that's how it was fixed elsewhere. I basically borrowed the fix from xpp |
| 11:37.15 | SparFux | ah, I see. |
| 11:38.06 | SparFux | but there seems to be no fix for 2.6.31-rc8 build right? Ahh, that's so tough. I need two modules extra, the dahdi and the nouveau. And it's so hard to get both into one machine in recent versions... |
| 11:41.53 | SparFux | I will try 2.6.31-rc8 now... |
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| 12:11.23 | grandpapadot | The new IAXModem (1.2) has an integrated jitter buffer and it works pretty good, neat. |
| 12:14.02 | iksik | morning |
| 12:14.38 | iksik | http://pastebin.com/m4e6e3598 - anyone can help me? |
| 12:15.58 | kaldemar | iksik: a peer is not an extension. extensions are numbers in the dialplan. |
| 12:16.57 | grandpapadot | In your DLPN... context, add an invalid handler "exten => i,1,DoSomethingUseful" or add an "exten => 7050,1,DoSomethingUseful" |
| 12:20.14 | Druken | iksik: consult who ever hosts your asterisk system... :) |
| 12:20.16 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:21.30 | kaldemar | Druken: you sound like microsoft help :D |
| 12:21.56 | Druken | Thank you for calling Microsoft, how man i ruin your day? |
| 12:23.50 | kaldemar | tell me why keeping a linux distro updated is so hard |
| 12:25.18 | Druken | Because it's built so stable it doesn't need daily updates like our products :) |
| 12:29.05 | kaldemar | Druken: quite the opposite of http://www.electronista.com/articles/09/09/07/best.buy.told.to.misrepresent.linux/ |
| 12:30.43 | grandpapadot | kaldemar: lol, don't update unless you have to, I don't understand the obsession with constantly feeling like you have to run the latest updates, as long as you're behind a well configured firewall, you should rarely have to update. |
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| 12:32.03 | kaldemar | grandpapadot: i don't have a problem with updates, i was just referring to that propaganda. :) |
| 12:32.46 | grandpapadot | kaldemar: yea, I was agreeing with you ... |
| 12:33.28 | grandpapadot | I subscribe to the "if it ain't broke, don't fix it" paradigm ... |
| 12:33.32 | kaldemar | but hey, it sure is easier to use a door if it's always open. :D |
| 12:36.02 | [TK]D-Fender | grandpapadot: If they release a fix... then it was broke :p |
| 12:38.11 | grandpapadot | [TK]D-Fender: I aggree that bugs need to be addressed but really, how many asterisk 1.2.8 systems are still out there running? At least hundreds (thousands?) ... I got bit hard one time with MS updates in a hug Citrix install... A single DLL thrown in there with the updates took our terminal server capacity from 60-80 users per box to like 3-8 per box and we had about 60 servers that *all* automatically updated. Last tim |
| 12:38.26 | grandpapadot | s/hug/huge |
| 12:38.53 | [TK]D-Fender | grandpapadot: Yeah and I remember all the SIP/IAX2 vulns that will let us ROOT them :) |
| 12:39.12 | grandpapadot | Maybe on the desktop it's a good idea .. but hey .. I have never ran anti-virus software since norton for DOS and I have *yet* to get a virus |
| 12:39.39 | grandpapadot | knocks on wood ... |
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| 12:44.03 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 12:44.09 | [TK]D-Fender | bludgeons grandpapadot |
| 12:44.15 | [TK]D-Fender | WOOD KNOCKS BACK!!! |
| 12:44.49 | Druken | sweet, i'm not gettin it today :) |
| 12:45.06 | riksta | Hi there, can someone kindly explain to me exactly how cache_record_files config directive works. |
| 12:45.50 | riksta | Does that mean that the recording is temporarily spooled into the /tmp dir for example, then once finished moved to the normal recording dir? |
| 12:50.08 | [TK]D-Fender | bludgeons Druken |
| 12:50.14 | [TK]D-Fender | PLENTY TO GO AROUND!!! |
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| 13:07.56 | Katty | mew. |
| 13:07.58 | Katty | pamples things |
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| 13:08.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:08.49 | beek | Good morning [TK]D-Fender , Katty |
| 13:09.02 | [TK]D-Fender | Katty: Mew. |
| 13:09.05 | [TK]D-Fender | beek: Mornin' |
| 13:09.16 | Katty | pamples beek |
| 13:09.27 | beek | purrs |
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| 13:15.15 | spck | well yesterday i had the joy of confirming that yes indeed queue member status events stopped being generated in 1.6.1.2 |
| 13:15.20 | spck | incase anyone was wondering |
| 13:15.24 | spck | :) |
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| 13:25.24 | afink | wow some people are so sensitive |
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| 13:35.20 | Druken | [TK]D-Fender: hey now... i didn't need the cluebat today... i managed to get my stuff working yesturday |
| 13:35.29 | Druken | only took me about 6 hours to do it |
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| 13:42.29 | merlin8282 | Hi. How could I uninstall bristuff in a simple way ? Is it even possible ? Or do I have to go through all makefiles and analyse them and act in consequence ? |
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| 13:53.42 | cuco | tzafrir_laptop: ping |
| 13:53.50 | tzafrir_laptop | cuco, pong |
| 13:53.59 | cuco | tzafrir_laptop: http://www.le-web.org/back-in-time/ |
| 13:54.10 | tzafrir_laptop | thanks |
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| 13:55.52 | cuco | tzafrir_laptop: it lacks the fancy 3d stuff apple adds to it's products, but hey, it works |
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| 14:02.34 | tgunr | what is the proper way to turn off loading of extensions.ael? I tried turning off autoload=no as described in the wiki but now it looks like my extensnions.con is not being loaded. |
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| 14:03.57 | ruben23 | hi |
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| 14:16.16 | [TK]D-Fender | tgunr: That's like completely disassembling your car when all you really want to do is remove the stereo |
| 14:16.25 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 14:16.26 | [TK]D-Fender | tgunr: noload => pbx_ael.so |
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| 14:23.13 | Kobaz | so i wrote a little perl script to periodically kill old postgres odbc connections that asterisk leaves dangling around |
| 14:25.25 | Naikrovek | Kobaz: does that work? |
| 14:25.55 | Kobaz | it looks for postgres processes for connections from localhost, to the pbx database, that were started more than 24 hours ago |
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| 14:26.15 | Kobaz | for some reason, asterisk forgets about those, and keeps spwaning new connections, until the server reaches it's limit |
| 14:26.40 | Naikrovek | file a bug? what version of asterisk are you using |
| 14:26.43 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 14:26.46 | Kobaz | yeah, filing a bug |
| 14:26.48 | Naikrovek | i can't believe you're the first person to come across this |
| 14:26.52 | Kobaz | 1.6.0.13 |
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| 14:28.09 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:28.12 | [TK]D-Fender | OLD :p |
| 14:28.38 | Kobaz | old? |
| 14:28.48 | Naikrovek | 1.6.1.15 is latest i think |
| 14:28.51 | Kobaz | oh 1.6.0.15 is out |
| 14:29.06 | Naikrovek | yeah that's it. |
| 14:29.13 | Naikrovek | 1.6.1.5 is newest of 1.6.1 |
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| 14:29.30 | Naikrovek | ah it's in the topic durr |
| 14:29.39 | dlynes | Naikrovek: could be he's one of a handful of people using the postgres extensions, too |
| 14:29.45 | Naikrovek | true |
| 14:29.54 | Kobaz | well it's odbc |
| 14:30.25 | dlynes | Kobaz: ah...well there's quite a few people using that I think, but most of them are probably using mysql, or sqlite |
| 14:31.02 | dlynes | But I would think that bug would affect all databases...not just postgresql |
| 14:31.09 | dlynes | if it's in odbc |
| 14:31.10 | Naikrovek | my retard coworkers use oracle for everything. then complain EVERY single time when they look at how much support costs |
| 14:31.42 | dlynes | For the most part, I don't understand why you need support for Oracle? |
| 14:32.03 | dlynes | Oracle works flawlessly for the most part, unless you're running a memory starved system |
| 14:32.22 | dlynes | Its installer's a piece of garbage though...requires twice as much memory as an installed system |
| 14:32.36 | Kobaz | heh |
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| 14:32.58 | dlynes | even then, it's still problematic |
| 14:33.05 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 14:33.27 | dlynes | but all in all, I'd have to say that Oracle's still my favorite database...i just can't afford it |
| 14:33.33 | tgunr | [TK]D-Fender: thanks |
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| 14:34.11 | Kobaz | i <3 postgres |
| 14:34.23 | tgunr | I finally have all warnings except this one taken care of, any clues as to what file it is referring to? |
| 14:34.24 | tgunr | [Sep 9 07:32:31] WARNING[27555]: acl.c:408 ast_get_ip_or_srv: Unable to lookup '' |
| 14:34.28 | ayeso | multiple instances of safe asterisk running is bad? |
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| 14:35.31 | Kobaz | probably |
| 14:35.42 | dlynes | ayeso: how do you know which one is going to answer your calls, unless they're all running on different ports? |
| 14:36.03 | ayeso | dlynes: safe_asterisk doesnt answer calls. |
| 14:36.16 | ayeso | dlynes: at least not to my knowledge |
| 14:36.26 | dlynes | ayeso: it spawns instances of asterisk |
| 14:36.48 | Kobaz | safe_asterisk is an asterisk respawner |
| 14:36.51 | dlynes | ayeso: it's a wrapper for asterisk...so if it's not spawning one currently, it will |
| 14:37.10 | ayeso | dlynes: well somhow i have 2 instances of safe_asterisk running but only 1 running process for asterisk itself |
| 14:37.21 | dlynes | ayeso: keep checking your process list |
| 14:37.36 | dlynes | ayeso: eventually, you'll probably see a second asterisk process respawning |
| 14:37.43 | Kobaz | ps auxf |
| 14:37.46 | Kobaz | maybe it's a child process |
| 14:38.02 | dlynes | or better yet (ps auxffww, on linux or ps -ef on bsd) |
| 14:38.23 | Kobaz | ff doesn't do anything more than f will |
| 14:38.48 | Kobaz | and if you have a wide terminal to begin with, ww won't show you more than w will |
| 14:38.52 | Kobaz | heh |
| 14:39.05 | Kobaz | or even without a w alltogether |
| 14:39.11 | merlin8282 | Seems there are more people, so I ask my question again: how could I uninstall bristuff in a simple way ? Is it even possible ? Or do I have to go through all makefiles and analyse them and act in consequence ? |
| 14:39.13 | dlynes | Kobaz: sure it does...it gives you the full command, instead of a truncated one |
| 14:39.25 | Kobaz | ps auxf will give you the full command as well |
| 14:39.38 | Kobaz | assuming your terminal is wide enough to fit it |
| 14:39.57 | dlynes | Kobaz: don't know...I tried your ps auxf and then tried my ps auxffww, but ps auxf was truncated; ps auxffww wasn't |
| 14:40.01 | *** join/#asterisk gego (n=gego@dyndsl-085-016-201-125.ewe-ip-backbone.de) |
| 14:40.03 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:40.11 | Kobaz | is your terminal wide enough to fit the whole command? |
| 14:40.25 | dlynes | Kobaz: no |
| 14:40.29 | Kobaz | well that's why |
| 14:40.33 | dlynes | Kobaz: ps auxffww makes it wrap |
| 14:40.33 | Kobaz | 10:38 <Kobaz> and if you have a wide terminal to begin with, ww won't show you more than w will |
| 14:40.37 | Skeeter- | icant update my asterisk, i get a lot of errors after the y |
| 14:40.44 | Kobaz | that's what i said right off the bat |
| 14:40.51 | dlynes | Kobaz: there's a limit on how wide your screen can be, though |
| 14:40.54 | Skeeter- | running transaction test |
| 14:40.56 | Kobaz | nope |
| 14:41.07 | Kobaz | i can have a 4000 column terminal if i want |
| 14:41.14 | Kobaz | if i span it across multile virtual desktops |
| 14:41.26 | Kobaz | s/multile/multiple |
| 14:41.29 | dlynes | Kobaz: really? and how do you do that? |
| 14:41.44 | Kobaz | i drag the window until it's mostly off the screen |
| 14:41.48 | Kobaz | and then make it wider |
| 14:41.57 | Kobaz | and then keep doing that until it's *really* wide |
| 14:42.10 | dlynes | Kobaz: how do you make it do that across multiple virtual desktops, though? |
| 14:42.17 | Kobaz | depends on your window manager |
| 14:42.22 | dlynes | Kobaz: gnome |
| 14:42.24 | Kobaz | i use afterstep. so the desktops are all linked |
| 14:42.38 | Kobaz | gnome has seperate desktops, they don't join each other |
| 14:42.41 | dlynes | Kobaz: ah...so i guess it's not possible in gnome or kde |
| 14:43.59 | Kobaz | i tried gnome for a little while, it's too restrictive for my tastes |
| 14:44.00 | Kobaz | heh |
| 14:47.28 | merlin8282 | I used Windowmaker for a while, but since Lenny it has bugs that make my console not show correctly, so I switched to gnome... :( |
| 14:47.30 | ayeso | im a windowmaker guy |
| 14:47.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:47.50 | Kobaz | afterstep is like a better windowmaker :P |
| 14:47.55 | ayeso | meh |
| 14:48.02 | merlin8282 | I mean, I don't know where the bugs are, but there are some. |
| 14:48.12 | Katty | :> |
| 14:48.19 | dlynes | Well, I was an OpenLook guy, then started using KDE, but since I switched to Debian, I just found it's easier to use Gnome |
| 14:48.20 | Katty | distributes hugs. |
| 14:48.28 | Kobaz | yays |
| 14:52.48 | Skeeter- | i cannot install anything using the yum install command |
| 14:53.30 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 14:53.31 | Kobaz | wouldn't that be a question for #redhat |
| 14:54.22 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:54.29 | dlynes | Skeeter-: cool |
| 14:55.11 | dlynes | Skeeter-: try #redhat, #fedora, or #centos for help with yum, though |
| 14:58.35 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:59.22 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 14:59.28 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 15:01.08 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:01.34 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 15:01.47 | geneticx | hello everyone |
| 15:03.52 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 15:05.33 | *** join/#asterisk jquintana (n=jquintan@c-98-195-194-57.hsd1.tx.comcast.net) |
| 15:06.25 | jquintana | can someone help me troubleshoot some permissions issues, please? |
| 15:06.29 | *** join/#asterisk errotan (n=errotan@62.201.122.72) |
| 15:06.33 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:06.51 | asterwiki | jquintana: what issues? |
| 15:06.53 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 15:07.13 | jquintana | I think my permissions for asterisk.ctl are wrong |
| 15:07.24 | jquintana | if I log in as root i am able to start asterisk |
| 15:07.31 | *** join/#asterisk tnt_ (n=tnt_@212-166-48-236.win.be) |
| 15:07.47 | jquintana | but my trixbox gui says asterisk is not running (using maint user) |
| 15:09.16 | [TK]D-Fender | ~trixbox |
| 15:09.17 | infobot | somebody said trixbox was a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 15:09.34 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 15:10.54 | jquintana | I will try the #trixbox channel -- thanks! |
| 15:12.36 | riksta | Hi there, can someone kindly explain to me exactly how cache_record_files config directive works. Does that mean that the recording is temporarily spooled into the /tmp dir for example, then once finished moved to the normal recording dir? |
| 15:13.01 | kn0x | [TK]D-Fender: ever seen this with followme confirm http://pastebin.ca/1559272 ? |
| 15:13.15 | kn0x | its just dropping the call when I press 1 to confir |
| 15:13.31 | tnt_ | Is it possible to make SIP forwarding works ? Here the phone responds with a 302 but then I get a "No channel type registered for 'Local'" error. |
| 15:13.44 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 15:14.13 | tgunr | How can I track this down? What file is it looking at? |
| 15:14.14 | tgunr | [Sep 9 08:12:50] WARNING[27555]: acl.c:408 ast_get_ip_or_srv: Unable to lookup '' |
| 15:14.36 | [TK]D-Fender | kn0x: You don't run IVR's inside of Macros. that is a shoot-on-sight offense |
| 15:14.46 | [TK]D-Fender | kn0x: Read() <- |
| 15:15.29 | [TK]D-Fender | tnt_: pastebin the entire call attempt |
| 15:16.05 | tnt_ | [TK]D-Fender: I just found a IRC log that described my problem, chan_local was not loaded and it's needed for SIP 302 handling to work properly. |
| 15:16.07 | kn0x | [TK]D-Fender: I didn't right the trixbox module |
| 15:16.12 | [TK]D-Fender | riksta: -cache_record_files = yes | no ; Cache record() files in another directory until completion |
| 15:16.19 | kn0x | Im just using the 'Confirm' option with Follow me |
| 15:16.28 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 15:16.29 | [TK]D-Fender | kn0x: And we don't support their crap here :) |
| 15:16.35 | CrashSys | Anyone ran into this error before: [Sep 9 11:15:09] WARNING[5469]: chan_sip.c:1788 __sip_xmit: sip_xmit of 0x8233458 (len 500) to 208.38.149.186:5060 returned -2: Bad file descriptor |
| 15:16.58 | kn0x | [TK]D-Fender: shit I know that... I was totally typing in to the wrong chan |
| 15:17.02 | riksta | [TK]D-Fender: yeah so to confirm it then copies them to the /var/spool/asterisk/monitor dir for example? |
| 15:17.11 | [TK]D-Fender | tnt_: You should always be loading chan_local. It is massively useful, and often needed |
| 15:17.17 | kn0x | and the first thing I thought was why is TK in here :P |
| 15:17.20 | riksta | [TK]D-Fender: the use of the word cache there is strange |
| 15:17.29 | [TK]D-Fender | riksta: Or wherever it was told to put them |
| 15:17.40 | riksta | [TK]D-Fender: yep |
| 15:17.40 | Skeeter- | ~recommended OS |
| 15:17.49 | dlynes | Skeeter-: Windows |
| 15:17.56 | Skeeter- | for asterisk |
| 15:17.57 | [TK]D-Fender | riksta: stops * from opening a file in a folder yuo may be monitoring to prevent locking with scripts, etc |
| 15:18.23 | riksta | [TK]D-Fender: yeah it's a good idea...it's what i was looking for :) thanks for the confirmation |
| 15:18.24 | dlynes | Skeeter-: Centos...just so you're compatible with whatever addons, drivers, ... you want to use |
| 15:18.36 | [TK]D-Fender | Skeeter-: whatever you can administer & install *'s dependencies on |
| 15:18.38 | riksta | [TK]D-Fender: it probably helps with NFS too |
| 15:18.41 | dlynes | Skeeter-: a lot of asterisk third party companies only officially support centos |
| 15:18.49 | [TK]D-Fender | riksta: I'd believe that |
| 15:19.18 | riksta | [TK]D-Fender: it will stop you truncating a currently recording wav to mp3 too which is a problem i had |
| 15:19.23 | riksta | cheers |
| 15:19.29 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 15:19.34 | ayeso | what does asterisk use mpg123 for? |
| 15:19.41 | [TK]D-Fender | riksta: * can't save in MP3 |
| 15:19.49 | [TK]D-Fender | ayeso: MoH |
| 15:19.56 | [TK]D-Fender | ayeso: If you tell it to. |
| 15:19.59 | dlynes | Skeeter-: however, that being said, if you're more comfortable with a given distribution, I'd use that until you have to switch to centos because something you need isn't supported on your distro |
| 15:20.11 | ayeso | [TK]D-Fender: thx |
| 15:20.19 | *** join/#asterisk HenrikBe (n=zapphir@d83-183-226-89.cust.tele2.se) |
| 15:21.04 | HenrikBe | Hi, I have a queue in asterisk and if a call in placed in the queue and an agent gets the call, all the agents hear a beep sound, am I doing something wrong? |
| 15:21.20 | Katty | twirls. |
| 15:21.33 | eppigy | :D |
| 15:21.39 | [TK]D-Fender | HenrikBe: Definitely. You aren't showing us the failed call and your configs |
| 15:21.41 | [TK]D-Fender | ~pb |
| 15:21.42 | 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 |
| 15:21.43 | Katty | twirls eppigy |
| 15:21.44 | [TK]D-Fender | ^^^^^^^^ |
| 15:21.45 | riksta | [TK]D-Fender: yes that's what i am saying, I run lame over all the *.wav files....but it is possible to truncate files that are being streamed to currently if you don't use the cache setting |
| 15:22.10 | [TK]D-Fender | riksta: Depending on how you trigger your conversions, I guess you've run into it |
| 15:22.12 | CrashSys | Anyone got any clues on what causes the following errors: http://pastebin.ca/1559869 |
| 15:22.26 | Skeeter- | http://pastebin.com/m18bdb84c |
| 15:23.04 | dlynes | CrashSys: it's just a warning, not an error |
| 15:23.10 | HenrikBe | TK: There is no specific call that goes wrong, all the calls connected to an agent results in a beep for all the other agents in the queue. |
| 15:23.50 | riksta | [TK]D-Fender: well i couldn't find a way to get bash to list all files which were not currently open by *.... out of interest do you know |
| 15:24.00 | [TK]D-Fender | Skeeter-: Nice bunch of repo conflicts on * core bits... |
| 15:24.05 | riksta | [TK]D-Fender: initially, i went with a find command to find all calls over XXX minutes |
| 15:24.12 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 15:24.16 | dlynes | Skeeter-: /join #trixbox |
| 15:24.21 | [TK]D-Fender | Skeeter-: and we've said it before, we'll say it again : Trixbox is NOT supported here. |
| 15:24.53 | [TK]D-Fender | HenrikBe: And you still haven't shown us anything |
| 15:25.04 | HenrikBe | tk: which file do you want to see? |
| 15:25.34 | bmoraca | what would cause a polycom 550 phone to freeze as it downloads its startup sound connected on a PoE switch and not freeze when powered by the AC adapter? |
| 15:25.37 | [TK]D-Fender | HenrikBe: Clearly your queues & CLI OUTPUT |
| 15:25.58 | [TK]D-Fender | bmoraca: What PoE Switch? |
| 15:26.26 | bmoraca | happens both with a Cisco 3550 PoE switch and with a Dell 3624 PoE switch |
| 15:26.28 | Skeeter- | didnt know that is was a trixbox issue, |
| 15:26.40 | [TK]D-Fender | [11:23]<[TK]D-Fender>Skeeter-: Nice bunch of repo conflicts on * core bits... |
| 15:26.59 | [TK]D-Fender | Skeeter-: YUM = distro issue. Sure as hell not * or FreePBX |
| 15:27.06 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:27.26 | Skeeter- | kk th |
| 15:28.11 | bmoraca | i got around the issue by disabling the start up sound, and now they start up normally, and all other associated sounds and files download just fine...it's just the startup sound that causes it to freeze |
| 15:29.26 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:29.35 | zerohalo | bmoraca: what bootrom are you using? There's known poe power issues with some. |
| 15:30.00 | bmoraca | 4.1.2 with 3.1.2 |
| 15:30.01 | Naikrovek | has anyone ever setup a dialplan to automatically add the 9 (for outside line) if a 7, 10, or 11 digit number is dialled |
| 15:30.24 | jaytee | Naikrovek, yeah, it's a piece of cake |
| 15:30.25 | bmoraca | Naikrovek: why would you want or need to do that? does your asterisk trunk to another PBX? |
| 15:30.31 | zerohalo | bmoraca: Interesting. Should be fine, then. |
| 15:30.58 | bmoraca | that's what i thought. it's really not a big deal, i was just curious why it was working that way |
| 15:31.04 | jaytee | Naikrovek, but bmoraca makes a valid point, why use 9? this isn't the 1970's |
| 15:31.07 | anonymouz666 | just curious: is it possible to play a busy tone (if it is) to the caller when you atxfer someone? A->B A atxfer C (but C is busy and should return busy tone to A and after connect back to B) |
| 15:31.28 | Naikrovek | well that's just how it's set up, it's how users know how to use the system |
| 15:31.29 | [TK]D-Fender | Naikrovek: Just make you're * dialplan smarter. Prefixes are so 1980 |
| 15:31.46 | Naikrovek | i suppose i could add an outgoing route to send the 7,10 and 11 digit numbers to my provider |
| 15:32.01 | Naikrovek | to make the 9 optional |
| 15:32.03 | Naikrovek | phase it out |
| 15:32.15 | Naikrovek | that's a better idea, and its why i asekd |
| 15:32.16 | Naikrovek | asked |
| 15:32.17 | jaytee | Naikrovek, when I moved my users from a Nortel pbx to * I dumped the 9 and just sent all the users to "reeducation camps" |
| 15:32.20 | [TK]D-Fender | Naikrovek: I haven't done prefixes in years... |
| 15:32.27 | bmoraca | Naikrovek: in that instance, you'd want to strip off the 9 from them. the provider doesn't want the 9, and in fact won't know what to do about the 9. the 9 is to tell the PBX that it's an outside line. the telco doesn't care. |
| 15:32.38 | Naikrovek | bmoraca: i know that |
| 15:32.46 | [TK]D-Fender | Naikrovek: I run 7-10-11 digit transparent dialing even bypassing normal mandatory 10 digit dialing here. |
| 15:32.48 | Naikrovek | but they're used to dialling 9 |
| 15:32.57 | Naikrovek | [TK]D-Fender: cool |
| 15:33.00 | Naikrovek | thats' what i'll do probably |
| 15:33.13 | Naikrovek | thanks |
| 15:34.01 | jaytee | Naikrovek, so put multiple pattern matches for 7,10 and 11 digit numbers that also have a 9 at the beginning and then just strip the first digit from the exten in that pattern match, that way if a user forgets the call still goes through |
| 15:34.06 | bmoraca | Naikrovek: so record a message that says "Dialing 9 is so 30 years ago...please redial your number in a manner which is suitable for the new millennium." and put it under the exten _9. in your outbound context |
| 15:34.13 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-3-211.w90-56.abo.wanadoo.fr) |
| 15:34.30 | jaytee | or try bmoraca's message approach |
| 15:34.41 | Katty | hugs on jaytee |
| 15:34.48 | bmoraca | might get you fired...but meh |
| 15:34.52 | jaytee | hugs on Katty |
| 15:34.53 | IBC_jkenney | ok i know this is not hylafax but there never anyone in there |
| 15:35.02 | IBC_jkenney | has anyone figured out FaxDispatch |
| 15:35.04 | Naikrovek | bmoraca: going to permit both, your option will get me in trouble if not just fired |
| 15:35.11 | jaytee | yeah, and the crickets are very, very loud |
| 15:35.29 | anonymouz666 | it sounds simple... but I can't find a way to play something (busy tone) to the caller when atxfer to someone that is busy |
| 15:35.31 | IBC_jkenney | i want to email faxes based on device |
| 15:35.34 | IBC_jkenney | to different e-mails |
| 15:36.07 | IBC_jkenney | I'll just create a IAXMODEM for each user who needs their own private fax number |
| 15:36.20 | bmoraca | IBC_jkenney: the way i usually do that is just have multiple iaxmodems set up...yeah, like that |
| 15:36.26 | Naikrovek | if i want to add areacode 321 to a 7 digit number, it's 321+NXXXXXX in the dialplan, correct? |
| 15:36.58 | [TK]D-Fender | anonymouz666: ATXFR isn't known until the hand-off actually happens. Its jsut a separate call until then. |
| 15:37.08 | [TK]D-Fender | anonymouz666: So YOU have to do BUSY on your extens. |
| 15:37.31 | [TK]D-Fender | Naikrovek: No. there is no such thing as "adding" |
| 15:37.31 | anonymouz666 | [TK]D-Fender: Did you try? I have a Busy(5) after Dial. |
| 15:37.45 | IBC_jkenney | I have tried what i found on the net in fax dispatch but it just defaults to the faxmaster account |
| 15:37.46 | [TK]D-Fender | anonymouz666: Show me something useful |
| 15:37.52 | IBC_jkenney | instead of sending to the correct e-mail |
| 15:38.10 | Naikrovek | [TK]D-Fender: okay well how do i add the area code to a 7 digit number (just assuming the local area code) |
| 15:38.26 | [TK]D-Fender | Naikrovek: You simply process that different pattern differently |
| 15:38.38 | anonymouz666 | [TK]D-Fender: Nothing useful. You transfer, you got Busy dialplan keeps executing... hits the Busy(5) application and the caller listen a very short sound of progress (the emulating one)... |
| 15:38.52 | anonymouz666 | and nothing from the application |
| 15:39.07 | Naikrovek | [TK]D-Fender: that's what i'm asking i guess. stand by while i compose the exact correct question that will solicit a reasonable response |
| 15:39.09 | [TK]D-Fender | Naikrovek: exten => _NXXXXXX,1,Dial(Zap/g0/putshitinfrontduh${EXTEN}) |
| 15:39.18 | [TK]D-Fender | ^^^ |
| 15:39.46 | [TK]D-Fender | anonymouz666: Show me the attempt |
| 15:40.05 | IBC_jkenney | I have the iax modems setup how do i specify it in Faxdispatch |
| 15:40.12 | IBC_jkenney | right now it all goes to one e-mail box |
| 15:40.22 | IBC_jkenney | I tried some stuff I found on the net |
| 15:40.37 | IBC_jkenney | and either i'm doing something wrong or its not correct |
| 15:43.03 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 15:43.38 | anonymouz666 | [TK]D-Fender: sure. I thought the same thing you told me but doesn't work that way. |
| 15:43.43 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 15:44.23 | anonymouz666 | http://pastebin.ca/1559892 |
| 15:44.26 | anonymouz666 | enjoy |
| 15:44.55 | anonymouz666 | at least in version 1.4 I don't see how this could be done |
| 15:45.41 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
| 15:46.08 | cjk | hi, does the qualify option (on or off) has an effect on packet retransmissions? |
| 15:46.39 | [TK]D-Fender | anonymouz666: Sounds like you had 5 sec to hear busy... |
| 15:46.43 | cjk | or better, is there a difference between qualify=9999 and qualify=no |
| 15:46.45 | *** join/#asterisk Nockian (i=hidden-u@unaffiliated/nockian) |
| 15:47.27 | anonymouz666 | [TK]D-Fender: Even 9999 won't have effect |
| 15:47.44 | [TK]D-Fender | cjk: Yes. 1 wll allow 9999 ms to respond to a qualify, and the other won't even bother |
| 15:47.51 | *** join/#asterisk DrCarumas (n=Carumas@adslfixo-b3-127-186.telepac.pt) |
| 15:47.58 | [TK]D-Fender | anonymouz666: Because you didn't ANSWER and it can responsd with a SIP BUSY |
| 15:48.02 | Nockian | i haven't changed anything, but my SIP phones no longer work properly. when i call them, they ring but when you answer you cannot hear anything, and the other person cannot hear you either. any suggestions as to where to start looking for things wrong? |
| 15:48.03 | DrCarumas | Hi! |
| 15:48.17 | [TK]D-Fender | anonymouz666: Welcome to "should have thought about this being carried OOB" |
| 15:48.17 | heedly | BAT problems |
| 15:48.19 | heedly | NAT |
| 15:48.43 | dustybin | The IP321 is now in stock, will order when i get home :D |
| 15:48.56 | [TK]D-Fender | reaches for his ClueBat (tm) and swats at heedly |
| 15:49.04 | cjk | [TK]D-Fender, well my question was more related if it has an effect on incoming calls. does asterisk gives up earlier if qualify is set? |
| 15:49.13 | dustybin | [TK]D-Fender: do you own a IP450? |
| 15:49.22 | [TK]D-Fender | cjk: this has nothing to do with incoming calls |
| 15:49.25 | [TK]D-Fender | dustybin: Nope |
| 15:49.30 | dustybin | 430? |
| 15:49.33 | Tim_Toady | im outside uk, i want a uk sip provider for cheap calls in london aera, any suggestions? |
| 15:49.46 | Tim_Toady | no need for did |
| 15:49.47 | [TK]D-Fender | dustybin: 301,430, 501, 600, 601, 320 |
| 15:49.54 | dustybin | aye ok |
| 15:49.58 | cjk | [TK]D-Fender, well setting qualify to a number too low will define the peer unreachable and as such incoming calls to that peer are not longer executed |
| 15:51.15 | dustybin | [TK]D-Fender: do you set them up via FTP |
| 15:51.18 | [TK]D-Fender | cjk: Incoming calls to peer. GRAMMAR FAIL |
| 15:51.32 | cjk | ok a user :) or friend |
| 15:51.42 | [TK]D-Fender | cjk: If you fail qualify, OUTGOING calls to that peer will not be attempted |
| 15:51.49 | iksik | mabe today someone can help me http://pastebin.com/m6f8c2926 ? |
| 15:52.02 | DrCarumas | hi guys, do you know if asterisk 1.6 is a stable version? |
| 15:52.13 | DrCarumas | or is it in beta? |
| 15:52.52 | [TK]D-Fender | ~asteriskversioning |
| 15:52.53 | infobot | i guess asteriskversioning is Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 15:53.03 | [TK]D-Fender | drmessano: 1.6.0 / 1.6.1 are stable |
| 15:53.15 | [TK]D-Fender | DrCarumas: 1.6.0 / 1.6.1 are stable |
| 15:53.23 | DrCarumas | thank you :) |
| 15:53.29 | *** join/#asterisk Whitor (n=Whitor@24.97.4.146) |
| 15:53.37 | [TK]D-Fender | DrCarumas: 1.6.0 is more mature naturallyt |
| 15:54.11 | DrCarumas | the thing is i have a production server making outbound calls only and i was thinking on updating to 1.6.x , do you thing 1.6.0 is rock solid as it it with 1.4.x versions? |
| 15:54.30 | iksik | it looks like peers from database are visible, but doesn't work :| |
| 15:54.58 | dustybin | [TK]D-Fender: i wrote some clear instructions on how to install asterisk as non-root user: http://www.thinkdebian.org/archives/828 |
| 15:55.36 | [TK]D-Fender | ~asterisk-non-root |
| 15:55.37 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 15:55.46 | [TK]D-Fender | dustybin: So did Leif... |
| 15:55.52 | [TK]D-Fender | dustybin: And the BOOK... |
| 15:56.04 | wcselby | dustybin - nice writeup |
| 15:56.04 | dustybin | those are not clear |
| 15:56.24 | [TK]D-Fender | dustybin: Its well established that YOU have "issues" :p |
| 15:56.25 | iksik | and centons o.O |
| 15:56.40 | dustybin | heh |
| 15:57.43 | wcselby | i think the more clean, clear documentation out there on how to do stuff with asterisk, the better. even if it's covered in THE BOOK |
| 15:59.04 | [TK]D-Fender | ~asterisk-non-root |
| 15:59.05 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 , and for Debian : http://www.thinkdebian.org/archives/828 |
| 15:59.08 | [TK]D-Fender | dustybin: There :) |
| 16:00.22 | wcselby | anyone seen the google homepage today? |
| 16:00.29 | wcselby | 09/09/09 at 09:09:09 |
| 16:00.39 | wcselby | think they've got something planned? |
| 16:02.54 | Qwell | wcselby: Google are geeks... |
| 16:03.05 | Qwell | numbers are geeky. today's date (and that time) is especially so |
| 16:03.14 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:03.17 | wcselby | isn't that a given (Google are geeks) |
| 16:04.21 | [TK]D-Fender | Chronology was sooo 5 minutes ago... |
| 16:05.17 | Qwell | [TK]D-Fender: Chronology shares the first four letters of Chrome! It's a sign! ZOMG |
| 16:05.42 | [TK]D-Fender | Qwell: Yeah... octagon... as in STOP :p |
| 16:06.56 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:09.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:09.48 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 16:10.34 | anonymouz666 | [TK]D-Fender: if you ANSWER the ATXFER then your transfer didn't fail. |
| 16:10.47 | anonymouz666 | so you listen the xfersound. |
| 16:10.52 | anonymouz666 | i.e.: beep. |
| 16:11.01 | [TK]D-Fender | anonymouz666: It isnt' a transfer until you actually hand it off |
| 16:11.03 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:12.40 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 16:12.41 | anonymouz666 | that's my case. I still didn't hand it off. |
| 16:13.06 | *** part/#asterisk justsomedood (n=justsome@mail.serverplus.com) |
| 16:13.19 | anonymouz666 | I am reading the dialplan through chan_local, and I can't see a way to play a busy tone to the caller when the dst is busy. |
| 16:13.31 | anonymouz666 | no matter what you do in the dialplan |
| 16:15.54 | anonymouz666 | it looks like a simples task... but it isn't... and it seems very basic stuff. |
| 16:16.05 | *** join/#asterisk HenrikBe (n=zapphir@d83-183-226-89.cust.tele2.se) |
| 16:16.38 | *** join/#asterisk Mango (n=Mango@96.49.69.137) |
| 16:18.47 | Nockian | it ended up being rtp ports blocked |
| 16:18.49 | *** part/#asterisk Nockian (i=hidden-u@unaffiliated/nockian) |
| 16:20.57 | HenrikBe | [TK]D-Fender: I have some log for my previous question about the beep that is played for all agents: http://www.pastie.org/611239 |
| 16:21.07 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:21.23 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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| 16:24.10 | Mango | What do I need to do to make voicemail work? I have the sample voicemail.conf file and I added load => app_voicemail.so to modules.conf. |
| 16:24.12 | *** join/#asterisk dysinger (n=tim@32.177.157.100) |
| 16:24.42 | Mango | When I make an extension that calls VoiceMail or VoiceMailMain, the console says "Spawn extension (devices, 206, 2) exited non-zero " |
| 16:25.42 | *** join/#asterisk MrNaz (n=mrnaz@118.208.238.5) |
| 16:27.26 | iksik | HenrikBe, how did You make it works? ;/ I've got realtime with postgresql, and weird behavior of asterisk ;/ |
| 16:28.02 | HenrikBe | iksik: it works well with mysql but it took some time configuring |
| 16:28.16 | HenrikBe | iksik: or do you mean the problem with the beep? |
| 16:28.21 | iksik | http://pastebin.com/m6f8c2926 |
| 16:28.23 | iksik | take a look |
| 16:30.08 | HenrikBe | iksik: can you paste you current extensions.conf? |
| 16:30.13 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 16:30.13 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:30.35 | iksik | extensions.conf is EXACTLY the same in both cases |
| 16:30.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:30.53 | iksik | only sip peers are going from pgsql |
| 16:32.41 | HenrikBe | iksik: hmm, looks strange, I have no good answer, I run all I can from mysql |
| 16:32.54 | iksik | hum... extensions to? |
| 16:32.58 | HenrikBe | iksik: yes |
| 16:33.05 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:33.42 | iksik | can You share with me, some rows with sample extensions? I can't find any good articles about that... ie. I don't understand how to create macros in it, etc.. |
| 16:33.46 | HenrikBe | iksik: take a look at my extensions.conf (part of the file: http://www.pastie.org/611253 |
| 16:34.02 | iksik | hmmmmmmmmm |
| 16:34.09 | iksik | I don't have anything like that |
| 16:34.10 | *** join/#asterisk jetsnoc (n=brian@72.22.242.66) |
| 16:34.42 | iksik | my extensions.conf is really big (mostly generated by GUI) |
| 16:34.44 | HenrikBe | iksik: well, this just tells asterisk that for the context in [], go look in the extension table in mysql |
| 16:35.03 | HenrikBe | iksik: okay :) |
| 16:37.00 | brad_mssw | [TK]D-Fender: not sure if you remember my issue from yesterday about a TDM400P not being able to dial out ... well, it resolved itself as soon as that line had an incoming call, I was then able to dial out ... upon further investigation, appears to be this bug: https://issues.asterisk.org/view.php?id=14577 |
| 16:38.12 | HenrikBe | iksik: have you read this? http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 16:38.44 | iksik | yes, i've setup my database and other stuff according to exactly this article |
| 16:39.03 | HenrikBe | iksik: I see |
| 16:40.12 | HenrikBe | iksik: perhaps you can turn on sql error reporting if you haven't already to see if there is some errors there |
| 16:40.15 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
| 16:40.36 | iksik | I have :) |
| 16:40.42 | iksik | and there is NO weird queries there |
| 16:42.03 | tgunr | I finally narrowed down the error 'Unable to lookup ''` which is coming from chan_sip.so which I assume means something in my sip.conf. I tried changing all the domain names to ip numbers and still get the error. Any clues would be appreciated. http://paste.lisp.org/display/86814#1 |
| 16:42.35 | *** join/#asterisk shyam (n=shyam@117.204.86.41) |
| 16:43.36 | shyam | i just wanted to make it sure the skype-asterisk thing isn't completely free software right? its just asterisk that is free software and the bridge is not am i right? |
| 16:44.30 | Qwell | shyam: Correct, Skype For Asterisk is not free. |
| 16:44.33 | shyam | means is that a new step towards for "free software" users to get to the skype protocol or something? |
| 16:44.43 | shyam | s/towards// |
| 16:45.02 | shyam | okay. |
| 16:45.04 | shyam | thanks Qwell |
| 16:46.47 | *** part/#asterisk shyam (n=shyam@117.204.86.41) |
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| 16:51.12 | hardwire | the 8x8 phones are kinda pretty |
| 16:53.47 | wcselby | skype for asterisk is 66 / channel, correct? |
| 16:54.24 | hardwire | si |
| 16:59.02 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:59.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:59.08 | Tim_Toady | and buggy as hell :P |
| 16:59.08 | *** join/#asterisk came0_ (n=came0@rrcs-71-42-53-177.se.biz.rr.com) |
| 16:59.29 | IBC_jkenney | Has anyone done AEX2400 cards and a PRI |
| 16:59.42 | IBC_jkenney | using the AEX2400 for analog break out to a modem pool |
| 17:00.00 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:00.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:00.05 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:00.18 | IBC_jkenney | i have this setup and i am having issues with the downloads stopping |
| 17:00.59 | *** join/#asterisk came0__ (n=came0@rrcs-71-42-53-177.se.biz.rr.com) |
| 17:01.53 | *** join/#asterisk came0___ (n=came0@71.42.53.177) |
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| 17:12.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:14.45 | *** join/#asterisk feder (n=feder@host.190.15.192.172.static.itcsa.net) |
| 17:15.10 | neurosys | Im trying to do a yum install for trixbox on CentOS 5. but its listing tb-appliance 1.0.0.0-5 in the group install. Isnt that old?? |
| 17:15.19 | Qwell | neurosys: #trixbox |
| 17:15.22 | *** join/#asterisk uski (n=uski@nor75-27-88-178-184-116.fbx.proxad.net) |
| 17:15.25 | neurosys | ah tru tru |
| 17:16.14 | feder | Is this the right place for asterisk/sip questions? |
| 17:16.21 | uski | hi; I'm trying to implement a call forwarding feature, but I'm a bit confused. I'd like the called party to be able to accept or deny the call by pressing a key (such as 1 or 2), so I cannot use Dial. I have to launch a separate call to the called person, to check his reply, and then to "connect" the two calls or to give a message to the caller. Any idea of the correct way to do that? |
| 17:16.25 | Qwell | feder: sure |
| 17:16.39 | Qwell | uski: Dial |
| 17:17.08 | uski | Qwell, I wasn't able to find a way to use Dial to allow the called party to accept or deny the call. Is that possible? (i'm reading the manual again right now) |
| 17:17.27 | Qwell | uski: see the M option |
| 17:18.26 | uski | oh I see. I also found G but I wasn't sure of how to connect the calls with G. As for M, if I call Hangup, who will be hang up? the caller? the called? How can I tell asterisk to connect the call? just leave the macro finish? |
| 17:18.27 | uski | (thanks)à|
| 17:19.45 | feder | Qwell: On incoming calls I receive the invite with "m=audio 57510 RTP/AVP 8 0 18 4 101" (alaw, ulaw, g729, g723......). My 200 OK says "m=audio 14818 RTP/AVP 18 0 8 101" (g729, alaw,ulaw......). I want the call to use G729, but sip show channels show alaw |
| 17:20.06 | Qwell | feder: disallow the other codecs on that peer then |
| 17:20.22 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-11.static.o1.com) |
| 17:20.30 | feder | Qwell: I can't afford failed calls |
| 17:20.37 | feder | I only have 10 linceses |
| 17:20.59 | feder | Qwell: I only have 10 licenses for g729 |
| 17:21.15 | [TK]D-Fender | feder: No dice |
| 17:21.36 | [TK]D-Fender | feder: If it CAN support g.729 it will use it. If you have run out of licenses it WILL FAIL |
| 17:21.53 | [TK]D-Fender | feder: So go buy a more appropriate # of licenses |
| 17:22.05 | feder | [TK]D-Fender: Ok, thx |
| 17:22.44 | wcselby | what's a quick way to see how many active calls I've got over my zap interfaces? |
| 17:23.00 | Qwell | wcselby: zap show channels |
| 17:23.23 | feder | [TK]D-Fender: do I need 2 licenses for every line (1 enconding 1 decoding)? |
| 17:23.28 | hardwire | if audio is bridged between two sip peers directly.. and you initiate a chanspy.. what exactly happens? |
| 17:23.30 | wcselby | Qwell - thanks, was afraid of that |
| 17:23.51 | [TK]D-Fender | feder: Each license includes 1 encode, 1 decode |
| 17:24.45 | feder | [TK]D-Fender: Great THX |
| 17:26.47 | *** part/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:27.20 | wcselby | time for some lunch I think |
| 17:27.21 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:27.21 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:27.34 | [TK]D-Fender | Gah |
| 17:28.55 | feder | [TK]D-Fender: it seems like the call starts in g729 (48391479-34 00101/00001 0x100 (g729)) and then it changes to alaw (48391479-34 00101/00001 0x8 (alaw)) |
| 17:29.01 | dustybin | Why can some VOIP phones handle different number of lines? Can't one setup asterisk to just use 1 line? |
| 17:29.15 | dustybin | And let asterisk re-route the same line? |
| 17:29.17 | feder | [TK]D-Fender: I have disallow=all / allow=g729 |
| 17:29.46 | [TK]D-Fender | dustybin: A phone may allow you to reg to a HUNDRED different servers at the same time. * does not have to give a crap about this |
| 17:30.09 | dustybin | ok |
| 17:30.11 | [TK]D-Fender | feder: That is not a complete picture |
| 17:30.57 | *** join/#asterisk came0_ (n=came0@rrcs-71-42-53-171.se.biz.rr.com) |
| 17:31.12 | [TK]D-Fender | dustybin: and I've told you this before : a phones meaning for LINES may not be what you think it is. It may mean "max supported simultaneous calls", or it may mean "number of unique registrations", or some mixture of the two |
| 17:31.35 | dustybin | ohhh |
| 17:31.37 | [TK]D-Fender | dustybin: What phone exactly? |
| 17:31.46 | dustybin | Polycom IP321 |
| 17:32.21 | dustybin | I need a phone what can handle my personal home phone line, and my business phone line |
| 17:32.29 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 17:32.33 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 17:32.36 | [TK]D-Fender | dustybin: that supports 2 REGS. It also has 2 line keys to divie amongst the number of regs you CHOOSE to use. |
| 17:32.46 | dustybin | ace :D |
| 17:32.50 | [TK]D-Fender | dustybin: Each LINE KEY can also support 1 ro 2 calls. |
| 17:33.02 | dustybin | excellent |
| 17:33.08 | [TK]D-Fender | dustybin: Meaning you could have a single reg supporting a total of 4 calls |
| 17:33.14 | dustybin | blimin ek |
| 17:33.19 | dustybin | ORDERS |
| 17:33.25 | raden_work | [TK]D-Fender, why would i get 14 * * * |
| 17:33.25 | raden_work | Unable to look up 64.2.142.15: Temporary failure in name resolution |
| 17:33.25 | raden_work | <PROTECTED> |
| 17:33.43 | raden_work | all with the same result |
| 17:33.55 | raden_work | traceroute inbound22.vitelity.net |
| 17:34.06 | raden_work | am i missing something |
| 17:34.08 | *** join/#asterisk came0____ (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 17:34.25 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 17:36.07 | youngproguru | inbound22.vitelity.net is up, but showing alot of loss at 64.2.142.1, the last hop before Vitelity. |
| 17:36.20 | youngproguru | 70% to 80% |
| 17:38.44 | PoWeRKiLL | hi |
| 17:38.59 | raden_work | youngproguru, yeah its up but i get that message alolot |
| 17:39.18 | PoWeRKiLL | I got some problem with 491 Request Pending from asterisk anyone here got also this problem ? |
| 17:39.20 | raden_work | how can you tell it showing allot of loss |
| 17:39.37 | youngproguru | mtr command, linux |
| 17:40.47 | Naikrovek | or pathping for windows |
| 17:42.10 | *** join/#asterisk came0___ (n=came0@rrcs-71-42-53-171.se.biz.rr.com) |
| 17:43.09 | *** join/#asterisk tuxx- (n=tuxx@nightshade.nl) |
| 17:43.14 | tuxx- | hello |
| 17:43.30 | tuxx- | Can anyone tell me how i can see if some agent from a queue is in wrapuptime? |
| 17:44.34 | PoWeRKiLL | mmichelson you there ? |
| 17:44.47 | putnopvut | PoWeRKiLL: I'm here! |
| 17:44.52 | putnopvut | PoWeRKiLL: who are you? |
| 17:45.14 | dustybin | 1 x Polycom SoundPoint IP 321 (IP321) = £73.30 1 x Polycom Power Supply for Soundpoint IP320 and IP330 = £10.10 |
| 17:45.17 | dustybin | ordered :D |
| 17:45.28 | Naikrovek | dustybin: good choice |
| 17:45.32 | Naikrovek | where'd you order from |
| 17:45.35 | dustybin | thanks |
| 17:45.41 | dustybin | https://www.voipon.co.uk |
| 17:45.42 | tuxx- | aastra > polycom tbh |
| 17:45.43 | tuxx- | :-O |
| 17:45.50 | Naikrovek | ~phones |
| 17:45.51 | 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, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 17:45.52 | PoWeRKiLL | I'm the guy about the 3 cdr and the 491 request pending |
| 17:46.12 | PoWeRKiLL | I try last 1.4 svn and still get the same problem |
| 17:46.17 | tuxx- | ^__^ |
| 17:46.22 | dustybin | Will people think im a bit geeky for using a VOIP as a phone at home? |
| 17:46.23 | tuxx- | pets infobot, good boi. |
| 17:46.31 | putnopvut | PoWeRKiLL: Hmm, I'm not familiar with that problem. Is there an issue open for it? |
| 17:46.33 | Nugget | dustybin: only if it sucks. |
| 17:46.44 | Naikrovek | dustybin: i do. it's tied in with my work system |
| 17:46.44 | PoWeRKiLL | https://issues.asterisk.org/view.php?id=14239 |
| 17:46.50 | dustybin | aye ok |
| 17:46.51 | Naikrovek | dustybin: i use voip at home i mean |
| 17:46.56 | dustybin | ace |
| 17:47.09 | Naikrovek | dustybin: and all those millions(?) of vonage customers are voip |
| 17:47.22 | putnopvut | PoWeRKiLL: that issue is closed. I don't understand. |
| 17:47.46 | dustybin | Naikrovek: I am also going to tie in my phone with my own business |
| 17:47.51 | *** join/#asterisk came0_______ (n=came0@rrcs-71-42-53-171.se.biz.rr.com) |
| 17:47.53 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 17:48.11 | PoWeRKiLL | I know but I still got the problem http://pastebin.com/d3a243106 |
| 17:48.14 | Naikrovek | dustybin: it's pretty easy |
| 17:48.20 | dustybin | aye indeed |
| 17:48.26 | [TK]D-Fender | [13:46]<dustybin>Will people think im a bit geeky for using a VOIP as a phone at home? <- I think you're asking in the wrong place... |
| 17:48.34 | dustybin | heh |
| 17:48.36 | Naikrovek | and the call quality from my desk at home to my desk at work is unreal.. sounds better when polycoms are on both ends of the call |
| 17:49.00 | uski | Qwell: thanks, Dial(...M...) works a treat :-) |
| 17:49.20 | PoWeRKiLL | putnopvut : xx.xx.124.238 is the asterisk box running 1.4.25 and now SVN 1.4 |
| 17:50.05 | putnopvut | PoWeRKiLL: I'm looking at the pastebin...give me a bit. |
| 17:50.14 | PoWeRKiLL | putnopvut no problem |
| 17:53.17 | Katty | sways. |
| 17:53.32 | uski | Does anyone know how I can cancel a call initiated with Dial from the macro called with the Dial(M) option? if I Hangup within the macro, the call is bridged, and instead of that I want the Dial to be cancelled |
| 17:53.39 | uski | I tried both MacroExit and Hangup |
| 17:54.04 | [TK]D-Fender | uski: "core show application dial" <- READ |
| 17:54.15 | Katty | PAGING FILE |
| 17:54.16 | uski | ok, thanks |
| 17:54.19 | Katty | PAGING FILE TO THE FRONT DESK |
| 17:54.50 | putnopvut | PoWeRKiLL: that situation is really really weird. |
| 17:54.52 | [TK]D-Fender | FILE NOT FOUND. THATS THE LEIF! |
| 17:54.57 | uski | omg [TK]D-Fender, I didn't' know about this help system, I only knew about http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial, thanks a lot |
| 17:55.10 | file | Katty: o.o |
| 17:55.11 | [TK]D-Fender | uski: WIKI = LAST place you should look |
| 17:55.19 | PoWeRKiLL | putnopvut I know I'm getting crazy for a week with the problem |
| 17:55.19 | Katty | file: you are past due on huggery. |
| 17:55.24 | Katty | file: PAST DUE I SAY |
| 17:55.29 | file | tickle hugs Katty twice |
| 17:55.36 | Katty | hugs file |
| 17:56.18 | Deeewayne | snaps a photo of the huggery |
| 17:56.25 | uski | [TK]D-Fender, ok |
| 17:56.28 | Katty | hugs Deeewayne too :> |
| 17:56.52 | Deeewayne | hugs Katty and file |
| 17:56.53 | file | hugs Deeewayne |
| 17:56.53 | raden_work | is there a way to unregister at console |
| 17:57.13 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:57.16 | Deeewayne | yay! its a hug fest |
| 17:57.26 | Deeewayne | hugs oej |
| 17:57.55 | raden_work | looks at everyone funny |
| 17:58.19 | Katty | don't worry. we're all adults here. |
| 17:58.35 | oej | evening |
| 17:58.43 | oej | hug fest? |
| 17:59.55 | Katty | mhmm |
| 18:01.56 | raden_work | omfg im currently not registered with vitelity and im still getting calls :( |
| 18:03.08 | youngproguru | I think that registration only tells Vitelity what your IP is,. If it hasn't changed I think you will still get calls. |
| 18:03.35 | *** join/#asterisk zerohalo_ (n=zerohalo@173.13.92.17) |
| 18:03.37 | *** join/#asterisk grEvenX (n=even@193.90.141.183) |
| 18:04.00 | *** part/#asterisk zerohalo (n=zerohalo@173.13.92.17) |
| 18:05.34 | *** join/#asterisk alecdavis (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 18:06.04 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 18:06.59 | tuxx- | Can anyone tell me how i can see if some agent from a queue is in wrapuptime? |
| 18:07.42 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 18:08.58 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 18:09.41 | Katty | i read that as Can anyone tell me how i can isee if an agent in queue is in bubble wrap |
| 18:09.54 | tuxx- | hehe |
| 18:09.59 | tuxx- | not what i meant |
| 18:09.59 | tuxx- | xD |
| 18:10.13 | tuxx- | but it might be fun, agents in bubblewrap |
| 18:10.25 | Katty | >.< |
| 18:10.27 | Katty | *hee* |
| 18:10.34 | tuxx- | 'Hello this is *pop* <name> speaking. *pop pop*' |
| 18:10.41 | tuxx- | :D |
| 18:11.01 | Katty | i'm going to start answering the phone: Hello! Katty Squeaking! |
| 18:11.09 | tuxx- | heh |
| 18:11.09 | tuxx- | :D |
| 18:11.15 | IBC_jkenney | i know this is not the hylafax room |
| 18:11.19 | IBC_jkenney | however i have been in there |
| 18:11.24 | IBC_jkenney | and its very frustrating |
| 18:11.32 | IBC_jkenney | i have been here and its been upsetting |
| 18:11.39 | IBC_jkenney | can anyone assist me |
| 18:11.49 | [TK]D-Fender | tuxx-: Probably not |
| 18:11.50 | tuxx- | <- no hylafax knowledge. sorry ;-( |
| 18:11.53 | Katty | IBC_jkenney: please keep in mind that everyone in here is a volunteer. |
| 18:12.05 | Katty | IBC_jkenney: if you require immediate help, I'd suggest finding a consultant to help you (= |
| 18:12.13 | tuxx- | [TK]D-Fender: its not possible to see if an agent is in wrapuptime? |
| 18:12.28 | [TK]D-Fender | tuxx-: Probably not |
| 18:12.32 | tuxx- | mkay |
| 18:12.33 | tuxx- | tnx :) |
| 18:12.34 | Katty | IBC_jkenney: otherwise, just be patience, continue asking your question, and hopefully someone will be able to help you. |
| 18:14.13 | Katty | eppigy: walk this way!!! |
| 18:14.31 | Katty | eppigy: just gimmie a hisssssssssssssssss >.< |
| 18:14.42 | Katty | has had too much caffeine this afternoon |
| 18:21.57 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:22.51 | *** join/#asterisk scalex000 (n=chatzill@180.120.88.200.f.sta.codetel.net.do) |
| 18:23.12 | scalex000 | hello |
| 18:23.27 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:23.36 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 18:23.39 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:24.13 | uski | any idea why this does not work? SayDigits(${CALLERID(num)}) i am able to print the caller id using the same syntax with a NoOp |
| 18:24.13 | scalex000 | I want to know how to dial and extention and pass another extention not from extension I dialed |
| 18:26.15 | [TK]D-Fender | uski: Where do we see your attempt & failure? |
| 18:26.36 | uski | [TK]D-Fender, I just made it work by adding " around the $ block |
| 18:26.37 | [TK]D-Fender | scalex000: Huh? |
| 18:28.23 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 18:28.49 | *** join/#asterisk came0_ (n=came0@rrcs-71-42-53-171.se.biz.rr.com) |
| 18:29.14 | *** join/#asterisk dysinger (n=tim@166.129.231.59) |
| 18:30.13 | exothermc | [TK]D-Fender: You get a chance to take a look at that polycom image problem yet? |
| 18:30.26 | [TK]D-Fender | exothermc: Nope |
| 18:30.37 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
| 18:31.04 | scalex000 | TK: how to make a call to exten but I need to indicate the exten I dial |
| 18:31.33 | *** join/#asterisk crevetor (n=crevetor@bureau.ubity.com) |
| 18:31.37 | crevetor | Hi |
| 18:32.01 | *** join/#asterisk errotan (n=errotan@62.201.122.72) |
| 18:32.22 | exothermc | [TK]D-Fender: Every single doc I follow doesn't produce any different results. Very confusing. |
| 18:33.31 | *** join/#asterisk gardo (n=gardo@110.55.63.220) |
| 18:34.37 | [TK]D-Fender | scalex000: Indicate what? To who? How? |
| 18:36.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:37.26 | exothermc | [TK]D-Fender: Those ID #s that are in line on key, specifically the one that is in the bitmaps section not the indicators section, is that referenced by anything else somewhere? |
| 18:38.28 | [TK]D-Fender | exothermc: Sorry, don't have the attention span right now... check with me later tonight :) |
| 18:38.40 | exothermc | lol ok |
| 18:39.08 | scalex000 | TK: Take a look on this. I have a linkys already setup and I use the analog line to plug to my BCM |
| 18:40.48 | [TK]D-Fender | scalex000: A Linksys WHAT? Your description is incredibly hollow |
| 18:41.08 | scalex000 | TK: Linksys PAP |
| 18:41.19 | scalex000 | TK: Ha ha ha |
| 18:42.00 | tgunr | if i restart as asterisk -c I see ast_get_ip_or_srv: Unable to lookup '' then if i try to make an outgoing call I see ast_request: No channel type registered for '' followed by Unable to create channel of type '' . Are these related? How do I track this down? |
| 18:42.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 18:42.20 | tgunr | internal extensions work fine |
| 18:42.41 | scalex000 | Tk: I'm hungry and I don't want to think in english |
| 18:42.57 | tgunr | auf deutsch? |
| 18:45.35 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 18:46.16 | [TK]D-Fender | scalex000: You had us at "don't want to think> |
| 18:46.24 | [TK]D-Fender | </jerrymcguire> |
| 18:46.28 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 18:46.46 | Mango | I have the sample voicemail.conf file and I added load => app_voicemail.so to modules.conf. What else do I need to do to make voicemail work? |
| 18:47.23 | [TK]D-Fender | Mango: the sample modules.conf autoloads this anyway. You need to USE voicemail in your DIALPLAN. |
| 18:48.00 | Mango | Like this? |
| 18:48.00 | Mango | exten => 206,1,Answer() |
| 18:48.01 | Mango | exten => 206,n,VoiceMailMain(1234@default) |
| 18:48.18 | [TK]D-Fender | Mango: That is one side of things, sure |
| 18:48.59 | *** join/#asterisk asif (n=chatzill@61.17.225.68) |
| 18:49.05 | asif | hello all! |
| 18:49.23 | Mango | Here's the tricky bit: when I call that extension, the console says "Spawn extension (devices, 206, 2) exited non-zero " |
| 18:49.37 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 18:50.04 | asif | Is it possible to have multiple MySQL CDR tables with Asterisk? |
| 18:50.32 | [TK]D-Fender | Mango: Did you define that box? |
| 18:51.58 | generalhan | im trying to find a way to shorten up 115 lines of code in my dialplan regarding hints, but pattern matching doesnt work, anyone have any ideas on how to get around having to put one hint line for each hint? lol. http://pastebin.com/d79ee4872 |
| 18:52.36 | Mango | [TK]D-Fender: Yes, it was in the sample voicemail.conf |
| 18:52.54 | Katty | dances with [TK]D-Fender |
| 18:53.24 | dustybin | watches and feels jealous |
| 18:53.25 | [TK]D-Fender | Mango: pastebin your call attempt at verbose 10, and include your voicemail.conf |
| 18:53.27 | [TK]D-Fender | ~pb |
| 18:53.28 | 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 |
| 18:53.28 | asterwiki | Mango:Use exten => 206,n,Voicemail(1234@default) instead of exten => 206,n,VoiceMailMain(1234@default) |
| 18:53.30 | [TK]D-Fender | ^^^^^ |
| 18:53.53 | [TK]D-Fender | asterwiki: Don't assume his goals |
| 18:54.00 | Mango | dialplan reload |
| 18:54.04 | Mango | oops wrong window :) |
| 18:54.19 | Katty | reloads Mango |
| 18:54.22 | dustybin | I really like the idea of having opening and closed hours |
| 18:54.32 | Mango | Same deal. I'll pastebin. |
| 18:54.41 | [TK]D-Fender | Mango: and you may have to leave a VM first to initialize the box |
| 18:54.49 | Katty | dustybin: what do you mean? gotoiftime functions? |
| 18:54.59 | dustybin | yep |
| 18:55.04 | Katty | it's loverly. |
| 18:55.30 | dustybin | i need to learn how to use macro- so everything looks nice and tidy |
| 18:56.09 | dustybin | what does the dialplan look like if asterisk is serving a _BUSY_ office with 200 people? O_O |
| 18:56.34 | Katty | about the same as any other server. |
| 18:56.40 | dustybin | eek |
| 18:56.42 | Katty | they all use the same extensions |
| 18:56.44 | Mango | http://pasteonce.com/44 |
| 18:56.48 | Mango | Thank you everyone. |
| 18:56.50 | Katty | same functions. |
| 18:56.53 | Katty | etc. |
| 18:56.57 | Katty | there's just more sip registrations |
| 18:56.58 | scalex000 | TK: I need to create a dialplan, but I don't know how to do it. I need to dial an exten and after another system answered dial an exten |
| 18:57.03 | dustybin | ok |
| 18:57.12 | eppigy | Katty: wha |
| 18:57.18 | Katty | eppigy: ohaider |
| 18:57.18 | Mango | Voicemail(1234@default) allows me to leave a message and VoiceMailMain(1234@default) allows me to hear the message, correct? |
| 18:57.28 | eppigy | hiya |
| 18:57.33 | [TK]D-Fender | scalex000: Go read the book and start making a dialplan |
| 18:57.33 | eppigy | it is nap time |
| 18:57.44 | Katty | Mango: i don't think so. |
| 18:57.46 | [TK]D-Fender | Mango: Yes. |
| 18:58.06 | dustybin | if somebody rings me at night > "i am sleeping right now, please leave a message after the beep.." |
| 18:58.09 | Katty | wouldn't that log you into that particular mailbox? |
| 18:58.19 | Katty | or prompt you for that mailbox's password? |
| 18:58.22 | Katty | or is that something different. |
| 18:58.35 | Katty | eppigy: go nap :> |
| 18:58.39 | [TK]D-Fender | Katty: prompt for that specific mailbox |
| 18:58.42 | leifmadsen | dustybin: like... GotoIfTime? :) |
| 18:58.51 | dustybin | yep, i need to learn how to use it |
| 18:58.58 | Katty | dustybin: i'll see if my blog is up. |
| 18:59.05 | dustybin | aye ok :D |
| 18:59.07 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 18:59.38 | Katty | dustybin: http://angela.sleekgeek.org/2008/11/11/blacklisting-numbers/ |
| 18:59.51 | dustybin | excellent :D |
| 18:59.53 | Katty | dustybin: it might be about blacklisting numbers, but there's a gotoiftime in there as an example. |
| 19:00.17 | dustybin | exten => s,n,GotoIfTime(7:30-17:00|mon-fri|*|*?BusinessHours,s,1) |
| 19:00.22 | dustybin | perfect |
| 19:00.36 | dustybin | the amount of things you can do is endless.. |
| 19:00.47 | doolittlework | could some please tell me if the following is possible with *? |
| 19:00.51 | Katty | dustybin: i eject the cdrom drive with extension 66. |
| 19:01.06 | doolittlework | i make a call and record the call |
| 19:01.08 | Katty | dustybin: manual backup scripts on 67 |
| 19:01.11 | dustybin | LOL |
| 19:01.21 | Katty | still trying to figure out how to adjust the temperature. |
| 19:01.24 | Katty | but i'll get there. |
| 19:01.26 | doolittlework | after i hangup the call i need to route it to a ivr |
| 19:02.17 | doolittlework | if prees one save the recording of press 2 delete the recording |
| 19:04.00 | [TK]D-Fender | doolittlework: If you hung up, what are you pressing "1" on? |
| 19:04.37 | supers | hi there, no matter what order i put ulaw in the allow codecs section, it always announces it's self as the first codec, if i replace it with alaw it works in the correct order |
| 19:04.40 | Kobaz | anyone know how to increase the volume on the 'call waiting' tone on polycom phones |
| 19:04.50 | doolittlework | if i call someone and the other side ends the call, i want an option to route the call to an ivr |
| 19:05.08 | bpgoldsb | I'm executing a Dial from the context all my SIP devices are in, named 'outgoing'. When I dial a call to an external line, it's going over one of my DAHDI channels. Is there any way to set variables on that DAHDI channel that gets created? |
| 19:05.46 | Katty | omegle is weird. |
| 19:06.23 | [TK]D-Fender | bpgoldsb: Channels created by Dial follow the basic inheritance rules as documented |
| 19:06.48 | hardwire | snom 870 eh. |
| 19:06.51 | hardwire | it's so pretty |
| 19:07.02 | hardwire | somebody buy me one |
| 19:07.06 | bpgoldsb | [TK]D-Fender, ah, thanks. |
| 19:08.25 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
| 19:08.37 | *** join/#asterisk ska (n=ska@cpe-70-124-73-96.austin.res.rr.com) |
| 19:08.56 | doolittlework | [TK]D-Fender: if i call someone and the other side ends the call, i want an option to route the call to an ivr |
| 19:09.14 | leifmadsen | continue after the Dial() then |
| 19:09.17 | leifmadsen | there is a flag for that |
| 19:09.24 | [TK]D-Fender | doolittlework: You ARE the call at that point. Dial() G <- option |
| 19:09.36 | leifmadsen | then do a check against ${DIALSTATUS} to make sure you should route to IVR or not |
| 19:09.53 | [TK]D-Fender | doolittlework: And you can check the DIALSTATUS to see that the call was indeed ended by the callee |
| 19:10.34 | leifmadsen | [TK]D-Fender: jinx! you owe me a beer |
| 19:11.01 | [TK]D-Fender | leifmadsen: deduct it from the pool you already owe me ;) |
| 19:11.10 | leifmadsen | I owe you nothing! |
| 19:11.11 | leifmadsen | NOTHING! |
| 19:11.28 | dustybin | anybody here bought a license for: http://www.cepstral.com/ |
| 19:11.37 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 19:11.46 | doolittlework | thx [TK]D-Fender will look into this |
| 19:14.37 | Katty | is having a random discussion about politics with a stranger on Omegle. weird. |
| 19:16.48 | Kobaz | it would be cool if you could change settings on polycom phones without having to reboot |
| 19:17.14 | Mango | Hm |
| 19:17.15 | [TK]D-Fender | Kobaz: Do it right the first time and you don't have to reboot :p |
| 19:17.29 | Mango | When I set autoload=yes, then Voicemail works. So, I must be missing some crucial module. |
| 19:17.32 | Kobaz | what if you're playing with settings trying to get 'just-the-right-one'(tm) |
| 19:18.13 | jaytee | brb |
| 19:18.14 | [TK]D-Fender | Manggenerally you SHOULD let * autoload and specifically exlude things you want to avoid |
| 19:18.21 | [TK]D-Fender | Mango: generally you SHOULD let * autoload and specifically exlude things you want to avoid |
| 19:18.52 | Katty | BYE JAYTEE |
| 19:18.59 | [TK]D-Fender | Kobaz: in Soviet Russia settings play YOU |
| 19:19.20 | Mango | Ah-ha! |
| 19:19.24 | Mango | res_adsi.so!!!! |
| 19:19.33 | dustybin | What is your favourite TTS program? |
| 19:19.37 | wcselby | i can't seem to access my google calendar....this is annoying. |
| 19:20.01 | retentiveboy | Got an all-data T1 for Internet traffic. Want to manage bandwidth to support off-site SIP extensions. Must I apply traffic shaping rules on both ends of that T1 (i.e. my router and the ISPs router) to manage traffic in both directions? |
| 19:20.47 | generalhan | anyone have any insight on how to shorten up the hints section of my code ... or know that it just isnt possible ? |
| 19:21.11 | eppigy | Katty: mebbe in 1.5 hours |
| 19:22.20 | Kobaz | ack |
| 19:22.25 | Kobaz | i just crashed a polycom |
| 19:22.36 | wcselby | Kobaz - crashed how? |
| 19:22.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:22.56 | wcselby | Kobaz - are you sure you didn't just uncover an undocumented feature? |
| 19:22.58 | Kobaz | it locked up |
| 19:23.05 | Kobaz | sent the phone too many calls |
| 19:24.24 | Kobaz | mm |
| 19:24.27 | Kobaz | just crashed another one |
| 19:24.45 | [TK]D-Fender | retentiveboy: You can't QoS your INBOUND |
| 19:25.19 | Kobaz | you can |
| 19:25.23 | [TK]D-Fender | generalhan: Hard to say... we can't SEE IT |
| 19:25.27 | retentiveboy | [TK]D-Fender: so, ISP would have to do it? |
| 19:25.35 | doolittlework | is there a way to increase volume on a iax trunk or does it rely on the sip phone? |
| 19:25.36 | Kobaz | [TK]D-Fender: it just doesn't always work as intended |
| 19:25.41 | generalhan | [TK]D-Fender:i was re-asking something i already posted. sorry |
| 19:25.46 | generalhan | http://pastebin.com/d79ee4872 |
| 19:26.04 | Kobaz | [TK]D-Fender: i qos my inbound so people can't download crap and max out the pipe |
| 19:26.16 | Kobaz | [TK]D-Fender: basically it drops packets until the rate slows |
| 19:26.29 | *** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) |
| 19:26.30 | [TK]D-Fender | Kobaz: Vile.... |
| 19:26.37 | Kobaz | it works quite well |
| 19:26.40 | LtScarr | hey |
| 19:26.52 | retentiveboy | [TK]D-Fender: yo doing this on your end of the circuit or getting the carrier to do it? |
| 19:27.00 | [TK]D-Fender | generalhan: Is what it is... |
| 19:27.02 | *** join/#asterisk jkroon (n=jkroon@dsl-240-169-69.telkomadsl.co.za) |
| 19:27.13 | Kobaz | it also is what it isn't |
| 19:27.15 | generalhan | [TK]D-Fender: damn, i was affraid of that |
| 19:27.28 | [TK]D-Fender | retentiveboy: I don't do QoS, or VoIP over internet anywhere I need to care about it |
| 19:27.36 | LtScarr | i have a question regarding behaviour of the queue command |
| 19:28.34 | retentiveboy | [TK]D-Fender: Understood. |
| 19:28.38 | LtScarr | if you add Queue(default|r) to your extensions.conf |
| 19:28.52 | LtScarr | does that keep announcements from playing? |
| 19:29.27 | [TK]D-Fender | LtScarr: I would suspect so |
| 19:29.59 | [TK]D-Fender | lsAlternative : make a really long recording of ringing and use it as a MoH class" |
| 19:30.12 | [TK]D-Fender | LtScarr: Alternative : make a really long recording of ringing and use it as a MoH class" |
| 19:30.27 | Naikrovek | doolittlework: yes the phones manipulate volume |
| 19:30.57 | Naikrovek | doolittlework: don't think you can modify volume up or down over a trunk, though if your transcoding that might be an option |
| 19:31.05 | LtScarr | [TK]D-Fender: alright thanks :) |
| 19:32.08 | Naikrovek | i made a recording of myself at a club once, music, lots of talking and pretended like i couldn't hear. made that my voicemail greeting for about 6 months |
| 19:32.11 | Naikrovek | it was awesoem |
| 19:32.36 | LtScarr | :D |
| 19:33.30 | [TK]D-Fender | Naikrovek: I just do "Hi uhh..... one sec..... Yeah (few seconds of silence)..... OH you thought I was actually here. No sorry please leave a message BEEP (ME doing beep)" |
| 19:33.54 | Naikrovek | awesome |
| 19:33.55 | [TK]D-Fender | and then more awkward silence before the REAL end of message beep |
| 19:34.15 | Naikrovek | my dad had a clever one: "you know what it does and when it does it. so when it does it, say it." |
| 19:34.26 | Kobaz | whoops |
| 19:34.28 | Kobaz | intellasoft-demo1*CLI> module unload res_musiconhold.so == Destroying musiconhold processes |
| 19:34.33 | Kobaz | Disconnected from Asterisk server |
| 19:34.35 | Naikrovek | my stupid friends always said "it! AHAHAHAHA" |
| 19:34.42 | Kobaz | it dieded |
| 19:34.44 | Naikrovek | Kobaz: lol |
| 19:34.48 | Kobaz | i keep breaking asterisk |
| 19:34.53 | Kobaz | this is the week of asterisk bugs for me |
| 19:34.55 | Naikrovek | Kobaz: wtf are you doing |
| 19:35.22 | Kobaz | i'm just coding some dialplan |
| 19:39.42 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:39.58 | Kobaz | *CLI> Segmentation fault (core dumped) |
| 19:39.59 | Kobaz | there you go |
| 19:40.03 | Kobaz | i founds it! |
| 19:41.12 | [TK]D-Fender | Kobaz: Cool how many segments? I'm aiming for "confetti". |
| 19:41.26 | Kobaz | heh |
| 19:47.47 | Kobaz | heh, i have four outstanding bug reports now... three crashes, and one resource leak |
| 19:50.24 | dustybin | is there a way to stop somebody brute forcing a numeric password on your dialplan? |
| 19:52.08 | Kobaz | no |
| 19:52.21 | Kobaz | well |
| 19:52.26 | Kobaz | you could kill them i guess |
| 19:52.30 | dustybin | LOL |
| 19:52.58 | Kobaz | if you want to make it harder for someone to brute force your password, you would have to rate limit the auth requests |
| 19:52.59 | Naikrovek | dustybin: how many digits is the dialplan |
| 19:53.00 | dustybin | i want to setup a IVR menu, and have a PRIVATE AREA section what is password protected |
| 19:53.10 | Naikrovek | um |
| 19:53.15 | Naikrovek | how many digits is the password |
| 19:53.32 | *** join/#asterisk PanicMan (i=SS7@122.102.33.80) |
| 19:53.37 | dustybin | inside the private area, i want to be able to turn on or off my zoneminder CCTV system, and also turn on and off X10 appliances |
| 19:53.50 | dustybin | Naikrovek: i dont mind, i havent set it up yet |
| 19:53.58 | PanicMan | hello, I'm new in asterisk, can anyone tell me, whats the difference between EXTEN:2 & EXTEN:3 |
| 19:54.27 | dustybin | PanicMan: if i was you, i would start 'Panicing', man :D |
| 19:54.29 | Kobaz | one has a 2, and one has a 3 |
| 19:54.45 | Naikrovek | dustybin: wardialers can make quick work of a short password |
| 19:55.03 | Kobaz | PanicMan: ${VAR:<number>} means... strip <number> of characters from the beginning of the string |
| 19:55.07 | dustybin | Naikrovek: well, even if somebody did get in, its no big deal |
| 19:55.20 | dustybin | Naikrovek: I will setup a 8 digit password |
| 19:55.23 | Naikrovek | dustybin: i'd go for 8 chars |
| 19:55.25 | Naikrovek | yeah |
| 19:55.26 | dustybin | 3ok |
| 19:55.39 | Naikrovek | 8, not 3 |
| 19:55.44 | dustybin | I will use AGI to execute BASH scripts |
| 19:55.49 | Kobaz | ${EXTEN:2} means.. remove 2 chars from the start of EXTEN |
| 19:55.50 | dustybin | yes 8 |
| 19:55.59 | [TK]D-Fender | dustybin: Make the option hidden, long to get to, and authed with a time/CID lockout |
| 19:56.00 | Kobaz | and give you back the result |
| 19:56.05 | PanicMan | thanks a lot |
| 19:56.07 | Kobaz | 1234 becomes 34 |
| 19:56.10 | dustybin | [TK]D-Fender: ok thanks |
| 19:56.13 | Naikrovek | [TK]D-Fender: can you do a caller-id check too? |
| 19:56.25 | dustybin | [TK]D-Fender: good idea |
| 19:56.46 | Naikrovek | learns to read |
| 19:56.48 | PanicMan | any body worked on SS7 ? |
| 19:56.50 | dustybin | yes caller id check is a good idea |
| 19:56.51 | Naikrovek | you said caller id |
| 19:56.59 | [TK]D-Fender | Naikrovek: Its dialplan. do whatever the hell you want. Base it ont he WEATHER if you feel like |
| 19:57.37 | Kobaz | phase of the moon is where it's at |
| 19:57.55 | [TK]D-Fender | Kobaz: More weighted factors... |
| 19:58.03 | dustybin | I like the idea of setting up TTS local weather, however, i was not impressed with festivals voice, yuk |
| 19:58.08 | *** part/#asterisk PanicMan (i=SS7@122.102.33.80) |
| 19:59.26 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 19:59.26 | dustybin | 1) VOICE MAIL 2) CONFERENCE HIDDEN = 3) PRIVATE AREA |
| 19:59.53 | Kobaz | private area should be like 1233894982734 |
| 20:00.02 | Kobaz | or at least three digits |
| 20:00.49 | dustybin | i could really do with a 3rd option, i think TTS weather report, anybody else got any ideas? |
| 20:00.56 | *** join/#asterisk Pan3D (n=Pan3D@streaming.sensoryresearch.net) |
| 20:00.56 | [TK]D-Fender | 8675309! |
| 20:01.11 | Kobaz | perfect |
| 20:01.32 | dustybin | idea!! |
| 20:01.36 | dustybin | TTS traffic report!!! |
| 20:01.36 | [TK]D-Fender | dustybin: E-mail checker. Last caller ID # |
| 20:01.38 | [TK]D-Fender | etc |
| 20:02.04 | dustybin | hmm, email checker could get messy, there are lots of crappy characters in email |
| 20:02.08 | dustybin | hyper links etc |
| 20:02.15 | [TK]D-Fender | dustybin: I never said to READ THEM |
| 20:02.21 | dustybin | ohhhh :D |
| 20:02.25 | dustybin | good idea !! |
| 20:02.29 | Naikrovek | lol |
| 20:02.38 | dustybin | [TK]D-Fender: that will need to go in the private area |
| 20:02.38 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:02.59 | dustybin | [TK]D-Fender: read the subject lines only? |
| 20:03.13 | [TK]D-Fender | dustybin: If even more than raw count |
| 20:03.21 | dustybin | ok |
| 20:03.29 | Naikrovek | the previous admin here had a bunch of text-only games like Zork set up with Festival to read you the line and some speech recognition to accept your commands |
| 20:03.35 | dustybin | traffic and weather will be perfect |
| 20:03.36 | Naikrovek | he played me a recording of it - epic |
| 20:03.40 | dustybin | even my friends might use that |
| 20:04.30 | dustybin | 1..VOICEMAIL 2..CONFERENCE 3..TRAFFIC REPORT 4..WEATHER REPORT HIDDEN 5..PRIVATE AREA |
| 20:04.34 | dustybin | perfect :D |
| 20:04.46 | Kobaz | wooooah |
| 20:04.54 | Kobaz | ctrl-c'd out of asterisk and got a craaaazy crash |
| 20:04.57 | [TK]D-Fender | dustybin: Call-back to save on LD / cell-time |
| 20:05.06 | Naikrovek | yes callback |
| 20:05.10 | dustybin | good idea |
| 20:05.22 | Naikrovek | this has all been done before, btw |
| 20:05.26 | Naikrovek | nothing new yet |
| 20:05.53 | dustybin | how about 6..LAUNCH NUCLEAR MISSILE ? |
| 20:06.14 | dustybin | [FAIL] |
| 20:06.54 | dustybin | i really like the idea of my server ringing me if something bad happens |
| 20:07.06 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 20:07.37 | Naikrovek | dustybin: get a smart phone and you can do all of this |
| 20:07.39 | dustybin | server rings --- > msg "WARNING: /var is 95% full" |
| 20:08.00 | Naikrovek | that could be cool, but if you're not at a computer anyway it's no better than a text message |
| 20:08.23 | Naikrovek | and one can send SMS messages for free from anywhere via the web |
| 20:08.25 | dustybin | server rings --- > msg "WARNING: your home is on fire" |
| 20:08.42 | dustybin | Naikrovek: how? |
| 20:09.01 | dustybin | there is a special tool designed for sms, qsms ? |
| 20:09.15 | dustybin | smsq |
| 20:09.39 | Naikrovek | just about every provider has a "send sms" form on their site somewhere |
| 20:09.45 | *** join/#asterisk f0ner00t (i=f0ner00t@99-21-148-71.lightspeed.frokca.sbcglobal.net) |
| 20:09.49 | dustybin | mine charges me 5p per text |
| 20:10.04 | Naikrovek | dustybin: how often does your /var get to 95% |
| 20:10.08 | dustybin | Naikrovek: never |
| 20:10.13 | dustybin | just incase :D |
| 20:11.50 | dustybin | i cannot wait to get my phone, hopefully it will arrive in a couple of days :D |
| 20:12.12 | dustybin | goes and makes a cup of tea and fetches a chocolate doughnut |
| 20:12.25 | raden_work | does * need anything special for faxing hooking a spa2102 upto it |
| 20:13.59 | [TK]D-Fender | raden_work: T.38 if you know whats good for you (which you clearly don't because it even approached VoIP) |
| 20:19.23 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 20:21.36 | [TK]D-Fender | checkout time, BBL |
| 20:22.44 | raden_work | which you clearly don't because it even approached VoIP ??????? |
| 20:22.47 | raden_work | what does that mean |
| 20:23.26 | Mango | He means you will likely become very frustrated getting faxing to work with VoIP properly. |
| 20:23.56 | raden_work | Mango, I KNOW :) |
| 20:24.00 | raden_work | wont it be exciting |
| 20:24.01 | *** join/#asterisk asif (n=chatzill@61.17.225.68) |
| 20:24.04 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:24.17 | Mango | LOL |
| 20:24.30 | Mango | Eh, I gave up. |
| 20:24.39 | Mango | on T.38. |
| 20:24.44 | ariel_ | afternoon folks |
| 20:24.57 | Mango | Even hired a company that did T.38-to-email, but their success rate was too low. |
| 20:26.01 | Naikrovek | http://store.digium.com/productview.php?product_code=804-00007 |
| 20:26.29 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 20:26.57 | Naikrovek | i need to get a good mic and build a recording room to record my IVR and prompts. |
| 20:27.05 | Naikrovek | i'm debating doing it ALL in my own voice |
| 20:27.07 | raden_work | sweet |
| 20:27.11 | Naikrovek | but i'm sure i'll regret it if i do it |
| 20:27.20 | raden_work | LOL |
| 20:27.39 | Naikrovek | thinking of completely replacing the standard voice |
| 20:27.44 | Naikrovek | that allison chick drives me nuts |
| 20:27.47 | Naikrovek | her voice anyway |
| 20:27.57 | Naikrovek | her inflections are not good |
| 20:27.57 | ariel_ | well I can give you a hint. I have a fairly good Plantronics headset/mic setup and I take the laptop to the car and do them there, they sound really good. |
| 20:28.09 | Chainsaw | Naikrovek: Yes, we paid a British girl to redo them. |
| 20:28.14 | Naikrovek | ooh car is a good idea |
| 20:28.35 | Naikrovek | i have a shitton of sound absorbing foam and lots of spare rooms |
| 20:28.44 | Naikrovek | just gotta make sure i can handle vibration |
| 20:28.45 | ariel_ | there is a set of recordings already done in the UK for asterisk |
| 20:29.01 | Naikrovek | i could listen to british girls talk all day every day for the rest of my life i think |
| 20:29.02 | youngproguru | Are the UK recordings free? |
| 20:29.22 | Naikrovek | youngproguru: there are lots of recording sets that are free, not sure about UK though |
| 20:29.29 | Chainsaw | Naikrovek: Irish girls are one step up, but yes :) |
| 20:29.47 | Naikrovek | Chainsaw: i lived in Australia for 2 years. I cannot stand that accent anymore |
| 20:29.51 | Naikrovek | i loved it when i moved there |
| 20:29.54 | youngproguru | Really? More then just Alison? Where can I find them? |
| 20:29.55 | ariel_ | oh how about a french girl doing english... |
| 20:30.10 | Chainsaw | ariel_: I approve of that, yes. |
| 20:30.16 | Naikrovek | yeah |
| 20:30.17 | Chainsaw | ariel_: Hard to find a french girl that's willing to speak english though. |
| 20:30.21 | Naikrovek | haha |
| 20:30.28 | Naikrovek | if you speak to her in french i'm sure she'll return the favor |
| 20:30.33 | ariel_ | knows of one.... |
| 20:30.36 | Naikrovek | speak in english im ean |
| 20:30.39 | Naikrovek | nevermidn.. |
| 20:30.41 | Naikrovek | can't type |
| 20:30.48 | Naikrovek | migrane comign on it hink |
| 20:30.55 | Chainsaw | ariel_: I can honestly say that's the best accent out there :) |
| 20:32.10 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 20:32.20 | ariel_ | sits in an office full of cubes and people talking all over the place... some with latina acents, french and one from down under. So yes the French one has it over the others.... |
| 20:32.41 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 20:32.50 | Naikrovek | lol http://damncoolpics.blogspot.com/2009/09/funny-talking-animals.html |
| 20:33.12 | Chainsaw | ariel_: Something to be said for a welsh accent, but that's just because it reminds me of Eve Myles I suppose :) |
| 20:34.20 | ariel_ | wow digium has a 100% Satisfied Guarantee on there web site now... |
| 20:34.30 | Naikrovek | they're serious |
| 20:34.31 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 20:34.41 | IBC_jkenney | I am having a problem with hylafax and fax dispatch i want to setup multiple inbound fax numbers and have them go to different e-mail accounts |
| 20:35.04 | Naikrovek | ariel_: G729 licenses are exempt from that guarantee though |
| 20:35.10 | IBC_jkenney | however it does not appear to work right or i am doing something wrong i would really appreciate a leg up |
| 20:35.32 | IBC_jkenney | if anyone has the time or inclination |
| 20:36.08 | Naikrovek | IBC_jkenney: i'd love to help but i know nothing of hylafax. i worked at a company that used it but that was 5 years ago and all my memories of that software have left me |
| 20:38.58 | asif | Hello all. |
| 20:39.07 | asif | Is it possible to have multiple MySQL CDR tables with Asterisk? |
| 20:39.24 | Naikrovek | why do you need multiple |
| 20:39.42 | Naikrovek | ceo doesn't want anyone seeing his calls or something? |
| 20:40.23 | asif | hehe... |
| 20:40.58 | asif | the cdr_custom.conf file says you can have multiple CDR CSVs with custom mappings. Wondering whether you can do the same with MySQL CDR tables? |
| 20:41.55 | asif | Like, one CDR table for a particular context in the dialplan and another for a different context ... |
| 20:42.36 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 20:42.39 | wcselby | IBC_jkenney - my FaxDispatch file has this line in it - SENDTO=$CIDNAME@mydomain.com |
| 20:43.01 | wcselby | then all I have to do is setup an email address at mydomain that matches the CIDNAME variable |
| 20:43.09 | Naikrovek | asif: have you read the asterisk book? |
| 20:43.11 | Naikrovek | ~book |
| 20:43.12 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:43.26 | eppigy | SNAP |
| 20:43.36 | Naikrovek | no i'm not like that, i am just asking |
| 20:43.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:43.57 | eppigy | SNAP |
| 20:44.07 | Naikrovek | asif: it has some good odbc stuff in there that you should be able to use. i'm pretty sure you can do what you want to do |
| 20:44.07 | wcselby | lol "like that" - you mean like [TK]D-Fender? |
| 20:44.09 | wcselby | <PROTECTED> |
| 20:44.15 | Naikrovek | wcselby: yes |
| 20:44.16 | asif | Naikrovek: Not the full book, I've read it in parts. |
| 20:44.16 | Naikrovek | :D |
| 20:44.22 | wcselby | ;) |
| 20:44.44 | IBC_jkenney | I want to route to the email based on the modem the fax comes in on |
| 20:44.46 | wcselby | anyone here ever use broadvox GO!xxx SIP trunking? |
| 20:45.04 | wcselby | IBC_jkenney - ahh, sorry don't know how to handle that. I use a t1 modem in my hylafax server |
| 20:45.04 | IBC_jkenney | I just did interop testing |
| 20:45.07 | IBC_jkenney | it seems |
| 20:45.17 | IBC_jkenney | to work well |
| 20:46.26 | wcselby | IBC_jkenney - which product did you test? |
| 20:46.35 | IBC_jkenney | sip trunk for carrier |
| 20:46.35 | asif | I hope somebody would here would have met with the same requirement... |
| 20:47.06 | riddlebox | man, I had an error in extensions conf, and my phones went straight to vm for a week lol good thing this wasnt a system for work |
| 20:49.10 | wcselby | asif - from what I'm seeing in my book (~book), it only covers setting up one table for all CDR records...but there is a context field in the table. Of course, we don't use CDR's here, so I'm not sure I can be of much help |
| 20:51.01 | IBC_jkenney | I use mysql for my cdr's |
| 20:51.48 | *** join/#asterisk mumtazah (n=mumtazah@203.82.79.101) |
| 20:51.49 | dustybin | i need to find a way of ripping the scrolly text at the top of this website, and convert it into text for TTS for asterisk http://www.highways.gov.uk/traffic/ |
| 20:52.11 | riddlebox | dustybin, I hate screen scraping |
| 20:52.11 | asif | hello kenney |
| 20:52.35 | dustybin | riddlebox: are there better ways of doing it? |
| 20:52.36 | riddlebox | I tried to do it for a bit, I think beautiful soup will help you though |
| 20:52.45 | asif | hey thanks wcselby |
| 20:53.32 | asif | yeah all documentation i came across dealt with setting up a single table for MySQL CDRs. |
| 20:53.33 | riddlebox | dustybin, I dont think so, I think it is the only way to do it |
| 20:53.34 | dustybin | riddlebox: what about ripping this: http://www.highways.gov.uk/rssfeed/rss.xml |
| 20:54.10 | asif | kenney: Do you use multiple CDR tables? |
| 20:54.29 | riddlebox | dustybin, that might be easier, but I wasnt to successful in my attempts, but I didnt give it a real good effort either |
| 20:54.36 | dustybin | ok |
| 20:54.45 | dustybin | riddlebox: maybe your box is 'riddled' |
| 20:55.01 | riddlebox | dustybin, that could be, I found a riddle I couldnt solve |
| 20:55.07 | dustybin | :D |
| 20:55.20 | dustybin | ok i have found a nice clean XML feed ready to pass |
| 20:55.30 | dustybin | now i need a nice clean XML weather report |
| 20:55.31 | riddlebox | dustybin, if you want I can try to help, cause there are things I would like to do as well using screenscraping |
| 20:55.41 | asif | i'm looking whether it's possible because i want to test out CDR stuff without touching my in-production CDR table. |
| 20:58.46 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 21:00.13 | iksik | is it possible to include a context in realtime? |
| 21:02.48 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:03.59 | *** part/#asterisk gego (n=gego@dyndsl-085-016-201-125.ewe-ip-backbone.de) |
| 21:05.57 | Naikrovek | how can i tell asterisk to use 729 internally, like with a meetme conference? my phone is set to 729 only and asterisk rejects the call entirely even though 'core show translation' shows 729 support |
| 21:07.08 | raden_work | alll i can say these days is wow vitelity sucks |
| 21:07.31 | raden_work | Naikrovek, u have lic per channel i will presume |
| 21:07.40 | Naikrovek | raden_work: of course |
| 21:08.11 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 21:08.26 | raden_work | it rejects the call what sip debug look like ? |
| 21:09.00 | Naikrovek | sip debug boils down to "SIP/2.0 488 Not acceptable here" |
| 21:09.23 | Naikrovek | need to allow it in sip.conf i think... |
| 21:16.04 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 21:17.30 | raden_work | do i need todo anything to asterisk for t.38 ? |
| 21:17.36 | raden_work | something just not adding up |
| 21:18.24 | drmessano | raden_work: Yes and know.. go over your setup.. How are you generating your faxes? |
| 21:19.04 | raden_work | have a fax machine set at 9600 baud connected to SPA2102 registering with asterisk using vitelity.net |
| 21:19.29 | raden_work | went over SPA2102 setup that a freind of mine uses successfully |
| 21:19.38 | raden_work | trying to fax to a vfax |
| 21:22.06 | raden_work | wishes he had a pots line |
| 21:22.46 | Chainsaw | raden_work: I must say, I did fax from an analog machine behind a Patton 4118 out over ISDN BRI, BT network, back into an analog line and then a TDM-400P. |
| 21:22.54 | Chainsaw | raden_work: And that worked. |
| 21:23.37 | Chainsaw | But agreed, it is nice to have a plain analog line for fax testing. Just to rule out the machine itself as the cause. |
| 21:23.49 | Chainsaw | raden_work: (Is it a Brother? They have a compatibility setting in the menu) |
| 21:24.49 | raden_work | HP L7590 |
| 21:25.23 | Chainsaw | Hm, okay. My experience is limited to Xerox & Brother, sorry. |
| 21:26.11 | raden_work | last fax error i got was a 388 There is too much noise on the line for a valid connection between two fax machines to complete. The noise problem might go away over time. |
| 21:26.21 | raden_work | how would i correct that |
| 21:26.32 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:27.35 | Chainsaw | raden_work: Normally it would try to negotiate for simpler and simpler protocols and eventually they'd understand each other at 9600bps. |
| 21:27.55 | raden_work | i have fax set at 9600 |
| 21:27.56 | Chainsaw | raden_work: You may be able to set the maximum protocol version it tries, or you might be able to enable "overseas" mode. |
| 21:28.10 | raden_work | lemee log into fax machine quick |
| 21:28.11 | Chainsaw | Which basically is "don't try anything fancy", which is exactly what you want over VoIP. |
| 21:28.19 | generalhan | anyone know if something special needs to be done to set the date or time in the globals context ? does seem to work the same as when i set it in any other context |
| 21:28.30 | generalhan | s/does/doesnt/ |
| 21:28.44 | generalhan | woah, forgot that infobot did that. |
| 21:28.52 | Chainsaw | Cool isn't it? :) |
| 21:29.04 | Chainsaw | raden_work: You can try my fax server here in the UK if you want. |
| 21:29.25 | raden_work | LOL i dont think thatd work :P |
| 21:29.30 | raden_work | much appreciated though |
| 21:29.38 | Chainsaw | raden_work: Faxing out towards it, I mean. |
| 21:29.53 | Chainsaw | raden_work: That won't try to negotiate anything above 9600. |
| 21:30.20 | generalhan | http://pastebin.com/d69f93be6 <--- this is what i mean. i can set the date via STRFTIME in a random context, but when i try in [globals] i get a strange result. rather, an unformatted result. |
| 21:30.58 | raden_work | someone told me putting a DSL line filter between SPA2102 and the fax machine helps |
| 21:31.54 | Chainsaw | The codec already takes out those high frequencies, it would never sample them. Not sure how that would help. |
| 21:35.27 | Chainsaw | Unless the SPA2102 has a particularly awful DAC I suppose. |
| 21:39.00 | raden_work | how can i tell if t.38 being used |
| 21:39.51 | bpgoldsb | If I GoSub to a FOO, is there a way to get the context/exten I came from inside FOO? |
| 21:40.03 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 21:40.27 | Chainsaw | raden_work: It should be an option under 'fax codec'. |
| 21:40.46 | raden_work | but how can i tell if its using it or just falling back to ulaw ? |
| 21:41.19 | Chainsaw | Hm, good question. On a Patton I'd start firing off debug incantations until it gave up the goods. |
| 21:41.30 | raden_work | holy crap i turned error correction off on my fax machine and i can now fax LOL |
| 21:41.39 | Chainsaw | raden_work: Go you :) |
| 21:43.00 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:44.01 | raden_work | 64.2.142.17 1920968931 6d2a71822af1866 0x4 (ulaw) No Tx: ACK |
| 21:44.01 | raden_work | 192.168.1.130 130 91b86afa-8fac9f 0x4 (ulaw) No Rx: ACK |
| 21:44.04 | raden_work | thats my fax |
| 21:44.21 | raden_work | wouldnt the channel say t.38 |
| 21:46.12 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 21:47.49 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:48.28 | Chainsaw | raden_work: Well, the call should establish as G711 and once the fax tone has been detected, it'll flip over to T.38 |
| 21:48.32 | Chainsaw | raden_work: You haven't disabled G711 by chance? |
| 21:48.40 | raden_work | no forced it |
| 21:48.49 | raden_work | i had 1 successful fax out of maybe10 |
| 21:49.00 | Chainsaw | Yeah, that sounds like fax over ulaw. |
| 21:49.16 | raden_work | :( |
| 21:49.24 | raden_work | spa2102 set to t.38 |
| 21:49.27 | Chainsaw | Face of the moon has to be right, and you need to stand in the same pose each time you send one. |
| 21:49.33 | raden_work | LOL |
| 21:49.35 | raden_work | ROFL |
| 21:49.39 | raden_work | good one |
| 21:50.12 | Chainsaw | Can you get that trace on a continuous basis? To see whether it ever establishes as G711? |
| 21:50.51 | Chainsaw | Well, whether it switches I mean. |
| 21:50.57 | raden_work | ill set debug on |
| 21:50.58 | raden_work | brb |
| 21:51.00 | Chainsaw | Okay. |
| 21:51.37 | Chainsaw | One other point: I see a user here reporting that it worked on firmware 3.3.5a, but that with firmware 5.1.9 "the success rate went down the drain". |
| 21:52.43 | *** join/#asterisk SLCarrijo (n=scarrijo@200.174.126.3) |
| 21:52.52 | Chainsaw | Other then that it's making sure the other end can do T.38; a SIP trace should show whether T.38 is attempted and whether the suggestion is accepted or denied. |
| 21:57.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:58.22 | raden_work | Chainsaw, working on it console dont hold enough line lost the first part |
| 21:58.31 | Chainsaw | *nod* |
| 21:59.17 | raden_work | and if people would stop using damn phones LOL |
| 21:59.41 | Chainsaw | Oh yeah, I hate that. Whenever I have to restart Asterisk, there's always someone on a long call. |
| 22:00.10 | Chainsaw | If you don't notice in time it gets in that half-shutdown state, which is awesome. No incoming calls, and your colleague won't hang up... |
| 22:02.25 | raden_work | http://pastebin.com/m557a1cc8 |
| 22:02.28 | raden_work | LOL |
| 22:02.40 | *** join/#asterisk came0__ (n=came0@rrcs-71-42-53-171.se.biz.rr.com) |
| 22:03.51 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 22:04.09 | Chainsaw | raden_work: Other end doesn't support t38: "SIP/2.0 488 Not acceptable here" |
| 22:04.20 | Chainsaw | raden_work: It does offer it. |
| 22:04.30 | Chainsaw | raden_work: It just doesn't get taken up on that and switches back. |
| 22:04.39 | raden_work | so ? |
| 22:04.50 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 22:05.23 | Chainsaw | raden_work: This looks like a SIP provider with termination, yes? |
| 22:05.44 | raden_work | yeah sending to there vfax |
| 22:06.19 | Chainsaw | I can give you the SIP address of a fax machine in the office. |
| 22:06.29 | Chainsaw | I'm not actually there, but that should negotiate T38 through the gateway. |
| 22:07.14 | Chainsaw | (Can't do it directly through Asterisk as I'm still on 1.2.35 atm) |
| 22:07.50 | *** join/#asterisk errotan (n=errotan@62.201.122.72) |
| 22:07.57 | Chainsaw | wb errotan |
| 22:08.00 | raden_work | Chainsaw, that might help |
| 22:10.50 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 22:12.19 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 22:12.52 | ruben23 | hi..any how to load balance asterisk server.. |
| 22:13.51 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-5-158.unitymediagroup.de) |
| 22:13.56 | Maliuta | canterlever? |
| 22:14.05 | Maliuta | :) |
| 22:15.47 | ArtemMakhutov | does somebody knows what OTASP numbers are? |
| 22:17.17 | Chainsaw | ArtemMakhutov: Over-The-Air Service Provisioning? |
| 22:17.43 | ArtemMakhutov | yes, and what does it does? |
| 22:18.54 | SLCarrijo | does anybody knows why I'm having this problem with hylafax ? |
| 22:19.03 | Chainsaw | ArtemMakhutov: You can use it to register CDMA 3G modems, for one. |
| 22:19.14 | SLCarrijo | [Sep 9 18:08:21] NOTICE[13447]: chan_iax2.c:9219 socket_process: Rejected connect attempt from 127.0.0.1, request '034217006@fax-out' does not exist |
| 22:20.41 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 22:28.46 | ArtemMakhutov | ok now I got it, OTASP number are special numbers you can call in order to make the GSM provider to send configuration to you and so on |
| 22:28.55 | Maliuta | SLCarrijo: looks like your extension doesn't exist |
| 22:29.08 | Chainsaw | ArtemMakhutov: Indeed. |
| 22:29.19 | raden_work | I dont get how when i had vonage like 5 years ago i could fax just fine but i cannot get it to work now |
| 22:29.24 | raden_work | can someone explain that |
| 22:29.45 | Chainsaw | raden_work: There was other hardware in the path that time. |
| 22:29.55 | Chainsaw | raden_work: (And software, of course) |
| 22:30.10 | Chainsaw | Likely Vonage operates proper T.38 end-to-end. |
| 22:30.12 | raden_work | i bet i could still use vonage without a hitch i was using a RTP300 |
| 22:30.21 | raden_work | Chainsaw, that is very true |
| 22:30.50 | Chainsaw | Whereas now, you are properly configured for T.38, offer it to the remote end which says "huh? don't know that. Stick to basics!" |
| 22:31.09 | Chainsaw | Which is what 488 really means in this context. |
| 22:32.21 | *** join/#asterisk came0 (n=came0@160.148.188.72.cfl.res.rr.com) |
| 22:33.50 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 22:35.13 | raden_work | faxing over VOIP SUCKS :( |
| 22:35.57 | ArtemMakhutov | has anybody tried to send faxes over GSM? |
| 22:37.11 | Chainsaw | raden_work: Yes, can't say I'm a fan. (Mostly because it involves Patton gateways. Configuring these makes you lose the will to live.) |
| 22:39.11 | drmessano | raden_work: Which asterisk version/ |
| 22:40.43 | Chainsaw | drmessano: 1.6.0.10 |
| 22:44.25 | ihatepeople | anybody been able to use a USB CDMA/GSM cell as a voice modem? |
| 22:49.46 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 22:49.53 | manxpower | ~answers |
| 22:49.53 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 22:56.16 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 22:57.35 | raden_work | drmessano, 1.6 |
| 22:58.00 | raden_work | manxpower, are u the one that gave me a contact for a wholesale dude at vitelity ? |
| 23:05.57 | Kobaz | hmm |
| 23:10.23 | hardwire | anybody have good experience with chan_mobile? |
| 23:10.23 | hardwire | thinking of using it instead of a GSM gateway. |
| 23:27.19 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 23:27.25 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:28.51 | Kobaz | awwwwwww |
| 23:29.05 | Kobaz | the new Originiate() dialplan app doesn't fork the call into the background |
| 23:29.21 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 23:29.55 | citywok | I have a visual voicemail XML application i wrote, but when i delete a voicemail from it (nuke the file from the voicemail folder), it doesnt tell the phone there are no new messages anymore. how would i tell the phoen to stop blinking the voicemail light? |
| 23:30.11 | manxpower | raden_work: I never gave you any contact at Vitelity, someone else did |
| 23:30.34 | raden_work | yeah i dont have logs :( |
| 23:30.50 | raden_work | mic or mac was the dudes name that gave it to me |
| 23:30.53 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:31.12 | manxpower | citywok: do you also nuke the .txt file for the message? |
| 23:31.47 | generalhan | citywok: when your app deletes the VM does it just delete the actual voicemail file? or do you also delete the txt file for that message as well ? |
| 23:31.49 | citywok | there's a text file? where does that hide? it's not in the voicemail/default folder |
| 23:31.51 | generalhan | oops |
| 23:31.59 | generalhan | sorry manxpower, didnt mean to step on your toes ! |
| 23:32.00 | manxpower | I keep waiting for a tech to call me back. Maybe if I get in my truck and drive to my cabin he'll call. |
| 23:32.19 | raden_work | manxpower, what yeah need a tech for ? |
| 23:32.27 | manxpower | generalhan: just unload chan_psychic.so and you'll be fine. |
| 23:32.32 | manxpower | raden_work: to move a T-1 cable |
| 23:32.33 | generalhan | citywok: should be in the same voicemail directory that the file was in. |
| 23:33.04 | citywok | oh, why yes, yes i did wipe the txt file |
| 23:33.09 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:33.12 | manxpower | citywok: for every msgxxxx.wav there is a .txt version of the file with the metainfo |
| 23:33.18 | citywok | lol i wrote this 2 weeks ago and already forgot i had done that, sheesh |
| 23:33.27 | citywok | just now getting around to figuring out how to get rid of the blinky light |
| 23:33.54 | generalhan | typically that WMI light is triggered by the msg0001.txt file as soon as its placed in the voicemail directory |
| 23:34.00 | manxpower | citywok: that should work automatically. IIRC you need to enable mailboxpolling if you want to modify mailboxes outside of Asterisk. |
| 23:34.16 | manxpower | I don't recall the EXACT option name, but you can find it in voicemail.conf.sample |
| 23:34.54 | citywok | you are correct. pollmailboxes is defaulted to no |
| 23:35.10 | manxpower | <-- smarter than he looks |
| 23:35.35 | citywok | yes, each time i ask a question you are fantastic at helping. do you live in the seattle/redmond area? if so i owe you a few beers |
| 23:35.49 | manxpower | ~manxpower |
| 23:35.50 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 23:36.03 | manxpower | that doesn't have my location! |
| 23:36.10 | manxpower | <-- huntsville, al |
| 23:36.21 | citywok | hah, nice |
| 23:36.40 | bmoraca | you work in AL for a company based in NYC? how's that work? |
| 23:36.53 | manxpower | bmoraca: very well so far |
| 23:37.09 | bmoraca | do they provide service in AL? |
| 23:37.19 | manxpower | I have a polycom on my desk registered to a server in NY, |
| 23:37.34 | manxpower | I don't deal with people in person. That's for level 1 techs 8-) |
| 23:37.40 | SaiSoma | need any hands on the ground in south alabama? |
| 23:37.43 | SaiSoma | ;) |
| 23:37.46 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 23:37.51 | raden_work | manxpower, do you have VOIP service better than vitelity :D ? |
| 23:37.52 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 23:38.12 | manxpower | raden_work: they only do VoIP over their own network |
| 23:38.16 | citywok | manxpower: if you are using mysql/realtime asterisk, how do you configure those options? lol |
| 23:38.27 | citywok | wait, i think i was on the wrong server, nm. |
| 23:38.30 | raden_work | manxpower, Smart people |
| 23:38.32 | manxpower | citywok: I don't use realyime |
| 23:39.05 | raden_work | how hard is it to setup a ITSP ? |
| 23:39.16 | bmoraca | raden_work: not hard, just expensive |
| 23:39.24 | manxpower | My employer designed their network (almost) exactly as I would have set it up. |
| 23:39.25 | raden_work | bmoraca, howso ? |
| 23:39.40 | manxpower | being a registered CLEC is like a half a million $ a year |
| 23:39.46 | raden_work | CLEC ? |
| 23:40.02 | bmoraca | in California, I don't have to register with the PUC as a CLEC |
| 23:40.08 | manxpower | phone companies other than the original BELL company |
| 23:40.16 | SaiSoma | clec = common? local exchange carrier |
| 23:40.28 | citywok | ~clec |
| 23:40.29 | infobot | it has been said that clec is Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's Public Utilities Commission. In exchange for the time and money spent t |
| 23:40.29 | bmoraca | competitive |
| 23:40.31 | Kobaz | competative |
| 23:40.33 | manxpower | competitive |
| 23:40.34 | raden_work | competitive |
| 23:40.37 | SaiSoma | i'm too lazy to look at it, thanks:) |
| 23:40.44 | Kobaz | i win! |
| 23:40.48 | Kobaz | wait no |
| 23:40.49 | Kobaz | aww |
| 23:40.52 | bmoraca | lol |
| 23:41.18 | raden_work | do you have to be a clec ? |
| 23:41.20 | citywok | manxpower: realtime voicemail doesnt let you set pollfreq, lol |
| 23:41.27 | raden_work | im talking small town 5,000 people |
| 23:41.29 | manxpower | citywok: it sucks to be you |
| 23:41.38 | raden_work | are there companies with reseller plans ? |
| 23:41.58 | bmoraca | raden_work: depends on your local state. like I said, California does NOT require VoIP providers to register in any way and does not regulate them at all |
| 23:41.59 | citywok | yea, if you modify voicemail.conf while using realtime it stops the realtime. lol |
| 23:42.10 | manxpower | they have dedicated SIP interconnect with Verizon and dedicated bandwidth to several of the wholesale carriers they use. |
| 23:42.18 | raden_work | bmoraca, where do i find out im in wisconsin |
| 23:42.26 | manxpower | my employer does POTS, DSL, T-1, PRI, etc |
| 23:42.50 | bmoraca | look at the wisconsin's public utilities commission webpage or talk to a lawyer familiar with the business |
| 23:43.14 | manxpower | raden_work: being a CLEC means you are a "real telephone company" with all the requirements and abilities of a telephone company |
| 23:43.39 | bmoraca | we used to do T1s, but we don't anymore. I just do VoIP and hosted PBXes now. though I do provide all of the trunking |
| 23:43.45 | manxpower | A VoIP provider does not usually have to be a CLEC, but it doesn't hurt. |
| 23:44.01 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 23:44.23 | *** join/#asterisk Druken (n=jdumais@70.54.242.169) |
| 23:44.44 | manxpower | bmoraca: most of the Ts I deal with are data with QoS voice on them |
| 23:44.52 | raden_work | im going to find a place i can resell for with residuals |
| 23:44.57 | raden_work | do setups etc.... |
| 23:45.06 | raden_work | once i get all my own issues worked out |
| 23:45.15 | raden_work | and im staying far away from freaking faxes |
| 23:45.25 | thehar | fax is bad mkay |
| 23:45.43 | Maliuta | why can't people just email? |
| 23:45.46 | bmoraca | manxpower: if only I could convince the bosses of the necessity of that...oh well |
| 23:45.48 | raden_work | no shit !!!!! |
| 23:46.22 | manxpower | we take all fax lines and run them over....can you guess?....what's the only reliable fax? That's right, boys and girls! POTS! |
| 23:46.37 | thehar | Being a CLEC lets you deal with all sorts of fun taxes n junk. |
| 23:46.40 | raden_work | yup dont blame u |
| 23:46.49 | Maliuta | manxpower: what about pans? |
| 23:46.54 | raden_work | lol |
| 23:46.56 | raden_work | im outta here guys |
| 23:47.02 | raden_work | have to go meet the woman for dinner |
| 23:47.03 | manxpower | thehar: *nod* and not allowed to do things like route calls weird to avoid costs. |
| 23:47.16 | Maliuta | passes manxpower some pots and pans |
| 23:47.17 | thehar | manxpower: << aware |
| 23:47.48 | raden_work | later |
| 23:47.50 | bmoraca | thehar: depending on your state's PUC, you have to pay USF taxes and 911 taxes whether you're registered as a CLEC or not...thankfully, California is not one of those states |
| 23:48.05 | thehar | In utah you must pay lots of fun taxes. |
| 23:48.35 | *** join/#asterisk s14ck (n=s14ck@190-76-119-238.dyn.movilnet.com.ve) |
| 23:48.36 | thehar | lots of "grey" areas as well |
| 23:48.42 | thehar | so you just pay them to ensure you don't get raped |
| 23:49.12 | Druken | isn't it in utah where you can have like 4 wives? |
| 23:49.25 | thehar | it isn't legal but some people down south have them |
| 23:50.11 | manxpower | You'd think they would learn after just one wife. |
| 23:50.25 | Druken | you'd think... |
| 23:50.29 | manxpower | I did. |
| 23:50.36 | thehar | but the choices |
| 23:50.49 | bmoraca | lol |
| 23:51.08 | Druken | manxpower: Mines still ok.. but i'm sure the day will come |
| 23:51.08 | manxpower | I would rather just have 4 friends w/benefits |
| 23:51.25 | manxpower | Druken: just wait until she has a psychotic break. |
| 23:51.38 | thehar | then you'll ditch the bitch and make the switch |
| 23:51.43 | Druken | you mean that didn't happen the day of?? hehehe |
| 23:55.02 | *** join/#asterisk mistiq (n=Dmitry@93.88.10.2) |
| 23:55.06 | *** join/#asterisk theGoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
| 23:55.39 | theGoat | i am in the process of setting up asterisk gui, connected to it with a web browser, but it seems to be stuck on 'Checking write permissions on gui folder' any ideas? |
| 23:55.51 | thehar | use asterisknow |
| 23:55.57 | manxpower | theGoat: My idea is ask on the #asteriskgui channel |
| 23:55.57 | [TK]D-Fender | theGoat: #asteriskgui <- Idea |
| 23:56.06 | grandpapadot | in an iax2 peer, is jitterbuffer=yes sufficient to activate the generic jitterbuffer: SIP->asterisk->IAX2 |
| 23:56.06 | [TK]D-Fender | theGoat: GUI's are not supported in this channel; |
| 23:56.22 | manxpower | Digium should just sue the makers of Trixbox and FreePBX to make them stop saying it's Asterisk |
| 23:56.24 | bmoraca | are gooeys supported, though? :P |
| 23:56.27 | theGoat | my bad. |
| 23:57.14 | manxpower | it's like calling a macy's parade float a "truck" |
| 23:57.34 | manxpower | yes, if you remove all the crap you might be able to find a truck under there somehere. |
| 23:57.59 | KyleK | I've got 2 phones each on two servers, can I link the two servers together with a password protected SIP connection but have correct caller id at the same time? |
| 23:57.59 | grandpapadot | .. or should I force one? |