| 00:00.04 | jql | (corn oil & canola oil, of course) |
| 00:00.07 | mohsen | dc |
| 00:00.59 | cybertooth | Qwell if it comes down to that, I don't think the ILECs will be up and running for us to connect to.... |
| 00:01.00 | jql | I wonder if there actually are multiple suppliers of california-legal fuel diesel? |
| 00:01.15 | Qwell | cybertooth: sat |
| 00:01.26 | Qwell | jql: There has to be |
| 00:01.39 | Qwell | (by law) |
| 00:02.09 | Qwell | besides, there are a bunch of refineries in CA |
| 00:02.46 | *** part/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 00:02.59 | jql | yeah, I don't doubt it's *possible*, but short of importing it from europe? heh |
| 00:03.24 | Qwell | nah, there are at least 3 places in the US where it can be drilled/refined |
| 00:12.18 | voiper1 | http://pastebin.com/m3a8f300 anyone seen a error like that? |
| 00:14.00 | tzanger | damn coppice isn't around :-( |
| 00:18.30 | *** join/#asterisk Guimaraes_Br (n=IceChat7@201.58.137.49) |
| 00:19.46 | *** part/#asterisk Guimaraes_Br (n=IceChat7@201.58.137.49) |
| 00:19.46 | *** join/#asterisk Guimaraes_Br (n=IceChat7@201.58.137.49) |
| 00:20.38 | cybertooth | Yes. I've seen this when there are routing shifts and packets are delayed. |
| 00:20.57 | Nugget | shift to the left, shift to the right, push up, pop down, byte byte byte! |
| 00:21.21 | cybertooth | They are often accompanied by a Peer lagging message - if you check for such things. |
| 00:21.23 | Blue_Ice | Nugget: <o/, \o>, <o>, \o/ |
| 00:21.39 | cybertooth | Nugget++ |
| 00:21.48 | Nugget | \o/ ^o^ /o_ /o\ |
| 00:21.53 | Nugget | It's fun to stay at the |
| 00:21.55 | Nugget | \o/ ^o^ /o_ /o\ |
| 00:22.08 | Blue_Ice | lol |
| 00:25.28 | voiper1 | cybertooth: okay that makes sense |
| 00:26.11 | tzanger | All a hacker needs is a tight PUSHJ, a loose pair of UUOs, and a warm place to shift. |
| 00:27.06 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 00:28.59 | craigk | does anybody know if Asterisk has plans to move to GPLv3 licensing or will it stay with GPLv2 ? |
| 00:29.21 | Qwell | craigk: there isn't much reason for it to |
| 00:29.23 | jql | given Digium's proclivity to license patented codec, that seems unlikely |
| 00:29.44 | craigk | thanks |
| 00:29.57 | craigk | i will shift off to reading a web page now :) |
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| 00:42.36 | [TK]D-Fender | Nugget, PRICELESS |
| 00:44.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:46.56 | Nugget | :D |
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| 00:48.04 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 00:55.39 | Nugget | yay |
| 01:02.19 | cybertooth | Oh ho! |
| 01:14.09 | *** part/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
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| 01:39.53 | Nugget | Recent statistics released by W3Counter reveal that the market share of Windows 98 fell from 1.44 percent to 1.34 percent in August, reducing it to the same level of popularity as the open source Linux operating system, which saw its market share increase from 1.33 to 1.34 in the same period. |
| 01:39.59 | Nugget | oof |
| 01:40.01 | Nugget | http://arstechnica.com/news.ars/post/20070903-linux-marketshare-set-to-surpass-windows-98.html |
| 01:41.55 | elixer | great. linux is almost as popular as one of the worst versions of windows other than 'me' |
| 01:42.14 | ptiggerdine_ | 98 SE was ok. |
| 01:42.31 | ptiggerdine_ | as MS OS's go. |
| 01:42.32 | elixer | ptiggerdine_: i s'pose |
| 01:47.35 | dijungal | has anyone successfully configured two TE100P cards in an asterisk box? i'm trying and i'm getting missed interrupts |
| 01:49.51 | Corydon76-dig | There is no such card as a TE100 |
| 01:50.36 | Corydon76-dig | There are T100, E100, TE110, and TE120 cards |
| 01:52.07 | dijungal | sorry TE110P |
| 01:52.15 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 01:52.24 | Corydon76-dig | Did you turn off shared interrupts? |
| 01:52.27 | *** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com) |
| 01:52.41 | Corydon76-dig | Is there any other device sharing the interrupt with the cards? |
| 01:52.49 | watchy | in 1.4 whats the best fax to email solution? |
| 01:52.59 | dijungal | nope |
| 01:53.07 | Corydon76-dig | watchy: for speed or what? |
| 01:53.11 | Qwell | watchy: a separate dedicated fax line |
| 01:53.22 | watchy | qwell: software wise |
| 01:53.24 | Qwell | oh, to email...meh |
| 01:53.46 | Corydon76-dig | I'd personally recommend a channel bank connected to a fax card running with hyperfax |
| 01:54.00 | dijungal | both cards are on IRQ5 |
| 01:54.19 | Corydon76-dig | dijungal: I'd try to get them on separate interrupts, and shared with nothing else |
| 01:54.33 | watchy | hyperfax pretty good? |
| 01:54.39 | Corydon76-dig | dijungal: that will probably require swapping one of the cards to another slot |
| 01:54.40 | dijungal | but how do i do that |
| 01:54.41 | dijungal | ? |
| 01:54.46 | dijungal | hmmm |
| 01:54.47 | dijungal | k |
| 01:54.58 | Corydon76-dig | watchy: Yes, you'll get 14400 fax speed using hyperfax |
| 01:55.11 | Corydon76-dig | watchy: you can only get 9600 fax with software modems |
| 01:55.29 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 01:55.33 | watchy | i just got hired on with a company that sells VOIP shit and they are having major issues |
| 01:55.45 | watchy | i've never delt with fax2email before though |
| 01:56.30 | Corydon76-dig | watchy: any of the programs receive a TIFF file. Then it's just a matter of compressing it into another format (like PDF) and attaching the file to an email |
| 01:57.01 | watchy | ah |
| 01:57.21 | watchy | i'll check out hyperfax. any special hardware I need for it come into the phone line with? |
| 01:57.43 | Corydon76-dig | You can usually do a good job by using a combination of ghostscript and a Perl module to create the attachment |
| 01:58.42 | Corydon76-dig | Mail::Send is a good one |
| 01:59.03 | watchy | i'm not sure what the guy is currently trying to use |
| 02:00.42 | watchy | i know hes selling rhino boxes for some reason |
| 02:01.19 | Corydon76-dig | Oh, there's one I was thinking of... MIME::Lite |
| 02:01.51 | *** join/#asterisk d-tech (n=d-dtech@72.245.233.107) |
| 02:02.14 | Corydon76-dig | Dead simple: http://search.cpan.org/~rjbs/MIME-Lite-3.020/lib/MIME/Lite.pm |
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| 02:08.48 | *** mode/#asterisk [+o mog] by ChanServ |
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| 02:28.54 | fujin | anyone have a module that polls devices for DEVSTATE and then puts them into pause/unpause? I'd rather not write one, but will, if necesarry. |
| 02:29.21 | [TK]D-Fender | fujin: "them"? |
| 02:30.39 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:31.04 | fujin | uh doh |
| 02:31.09 | fujin | that didn't really make sense |
| 02:31.14 | fujin | I have had a funny request; |
| 02:31.29 | fujin | actually forget it |
| 02:31.32 | fujin | it's ludicrous |
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| 02:35.02 | fujin | anyone have any cdr_mysql parsing software, preferably PHP? |
| 02:39.49 | tengulre | hi,all I need install a call center with asterisk, but I don't like agent. is it possible? |
| 02:40.07 | tengulre | member =>iax2/20001; etc, |
| 02:40.28 | tengulre | is that can auto distrube to iax2 client when inbound? |
| 02:42.38 | DrAk0 | tengulre, like what? |
| 02:43.01 | DrAk0 | tengulre, ur talking about ringing in many iax clients at once? if is that , yes. |
| 02:43.17 | Nugget | yes direct IAX or SIP targets can join a queue just like an agent can. |
| 02:44.10 | tengulre | DrAk0, Nugget: thanks, I don't like agent mode, |
| 02:46.21 | fujin | use aqm/rqm! |
| 02:46.31 | tengulre | ?? |
| 02:47.15 | tengulre | I use 2E1 lines + asterisk box(3.0GHz/2GB)+IAX2 phone to building call center. |
| 02:48.21 | tengulre | customer --->dailing in -->2E1--->Queue-->IAX2 phone. is that right? |
| 02:50.00 | [TK]D-Fender | tengulre: yes you can add direct devices as members to queues. |
| 02:50.10 | [TK]D-Fender | tengulre: jsut like your sample shoed |
| 02:50.13 | [TK]D-Fender | showed |
| 02:50.51 | tengulre | how to record voice in this case? |
| 02:51.10 | tengulre | I want record all talking history. |
| 02:51.15 | Nugget | the same way you'd record voice in any other case. |
| 02:51.56 | Nugget | respectfully, tengulre, your questions are starting to border dangerously on "set this all up for me" and I'd suggest that you'd be well-served by spending some more time with the documentation. |
| 02:52.19 | Nugget | once you have a better understanding of the fundamentals this sort of stuff will all be clear |
| 02:53.53 | [TK]D-Fender | tengulre: this is ALL in the sample QUEUES.CONF that Asterisk COMES WITH. Go read it for crying out loud... |
| 02:56.10 | tengulre | OK |
| 02:59.53 | fujin | die in a fire, too |
| 03:01.07 | JT | Nugget: repectfully, STARTING?! |
| 03:01.37 | JT | as in started many months ago |
| 03:01.39 | JT | ;) |
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| 03:04.49 | tengulre | <PROTECTED> |
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| 03:06.33 | Nivex | everybody was kung fu fighting? |
| 03:07.06 | mmlj4 | man, those cats were fast as lightning |
| 03:07.36 | jql | it was a little bit frightening |
| 03:09.45 | fujin | p00p |
| 03:09.53 | Nivex | dung |
| 03:10.07 | [TK]D-Fender | But they fought with expert timing |
| 03:10.25 | Nivex | [TK]D-Fender: which apparently you don't have tonight :) |
| 03:10.51 | [TK]D-Fender | Nivex: says YOU :p |
| 03:11.10 | Nivex | [TK]D-Fender: *bzzzt* I'm sorry, the correct response was "Damn lag!" |
| 03:11.45 | [TK]D-Fender | Nivex: *bzzzzt* I'm sorry, you forgot to phrase that in the form of a question! |
| 03:11.58 | Nivex | [TK]D-Fender: "Suck it, Trebek!" |
| 03:12.06 | J4k3 | ~gs |
| 03:12.06 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 03:15.53 | J4k3 | (I had to show that to my dad, who says "these grandstream phones work fine!") |
| 03:18.42 | fujin | anyone suggest a good method for diagnosing call crackle, internal, over a lan? |
| 03:18.47 | fujin | I have transocded *everything* to alaw |
| 03:18.52 | fujin | but still experience crackle occasionally |
| 03:20.18 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 03:20.43 | [TK]D-Fender | fujin: DETAILS <----------- |
| 03:20.56 | fujin | The audio crackles |
| 03:21.01 | fujin | What details do you want? |
| 03:21.55 | fujin | linksys spa942's, asterisk 1.4.10.1 |
| 03:21.56 | J4k3 | when I had crackle, it was loss between my * box and the phone (the phone reported the packets dropped) due to a bad ethernet switch |
| 03:22.07 | fujin | mm, I'm not seeing any packets dropped |
| 03:22.12 | J4k3 | happened with all protocols, worse with ulaw/alaw |
| 03:22.18 | J4k3 | funky |
| 03:22.22 | J4k3 | tried a softphone> |
| 03:22.27 | fujin | not yet |
| 03:22.31 | watchy | hey sexy |
| 03:22.40 | fujin | what phones, J4k3? |
| 03:22.49 | J4k3 | fujin: grandsuck in my case |
| 03:22.56 | fujin | ah |
| 03:23.04 | J4k3 | the switch it was attached to wasn't detecting the port's non-full-duplexity properly |
| 03:23.08 | watchy | hey tk: do you do fax2email? |
| 03:23.09 | J4k3 | duplexity is a fun word to say out loud |
| 03:23.19 | fujin | how were you seeing the packets dropping? |
| 03:23.29 | J4k3 | the web interface has stats on it |
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| 03:30.12 | fujin | hrm, network jitter. |
| 03:30.17 | fujin | could jitter cause crackle? |
| 03:30.47 | fujin | or rather, network jitter adaptive options |
| 03:35.52 | watchy | snap crackle and pop |
| 03:35.54 | watchy | bitches |
| 03:37.01 | Mavvie | http://www.itconversations.com/shows/detail1874.html <- FreeNum: The Phone Numbers of the Future |
| 03:37.52 | [TK]D-Fender | fujin: spa-942 on BOTH ends of the call and local lan to *? |
| 03:38.51 | fujin | [TK]D-Fender: haven't experienced it going spa942->spa942 |
| 03:38.57 | fujin | but spa942->voicemail even has crackle. |
| 03:39.12 | raidenz | Is their a way to emulate a modem using a digidum PRI card and a PRI line? ZapRas only sets up a PPP session but doesn't create a modem data session. |
| 03:39.15 | raidenz | ? |
| 03:39.22 | raidenz | err digium |
| 03:39.29 | [TK]D-Fender | fujin: crackle listening to just the prompts? |
| 03:39.29 | fujin | I'm wondering if it's fixed by r80166-80167, patching and rebuilding now |
| 03:39.33 | fujin | [TK]D-Fender: correct |
| 03:50.58 | raidenz | ? |
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| 04:21.15 | [TK]D-Fender | raidenz: go look up IAXModem on the WIKI |
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| 05:27.19 | fujin | anyway I can configure the asterisk console to display a timestamp at the begining of every line? |
| 05:28.38 | styelz | you could tail /var/log/asterisk/debug instead.. it has timestamps |
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| 05:34.20 | [TK]D-Fender | fujin: You have the source, get coding...... |
| 05:34.20 | styelz | heh |
| 05:40.01 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 05:51.50 | pkunkra | Has anyone tried out digium's IAXy? |
| 05:57.44 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 05:58.34 | pkunkra | the IAXy is this little guy: |
| 05:58.40 | pkunkra | http://www.digium.com/en/products/hardware/s101i.php |
| 05:59.52 | *** join/#asterisk BFAH (i=Shaun@cblmdm72-241-21-108.buckeyecom.net) |
| 06:00.09 | BFAH | hello |
| 06:00.44 | pkunkra | hi BFAH |
| 06:02.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:02.30 | BFAH | anyone ever use vicidial or gnudialer? |
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| 06:07.52 | WildPikachu | ok, i think i'm going to give up ... I've tried everything and just cannot fix this echo/distortion problem |
| 06:08.01 | WildPikachu | can anyone point me in the right direction? |
| 06:08.09 | JerJer | turn left |
| 06:09.01 | WildPikachu | sip to sip is fine, but sip to PRI sometimes is fine, but sometimes gives bad (repeat) echo and sometiems not |
| 06:10.22 | WildPikachu | i checked out all the guides on voip-info for clues |
| 06:10.30 | pkunkra | wildpikachu, i have echo problems in my headset. |
| 06:10.37 | pkunkra | might not be the software. |
| 06:10.50 | WildPikachu | i got a grandstream phone, one gxp2000 and one budgetone 101 |
| 06:11.07 | BFAH | what kind of wildcard are you using? |
| 06:11.42 | WildPikachu | BFAH, tried with my PRI card, now trying with a BRI card (i got multiple lines) |
| 06:11.54 | WildPikachu | my BRI card is currently plugged in, its an HFC-S |
| 06:12.22 | BFAH | what's BRI? |
| 06:13.23 | WildPikachu | basic rate ISDN |
| 06:13.29 | WildPikachu | as apposed to primary rate ISDN (PRI) |
| 06:13.40 | WildPikachu | i'm too scared to try my TDM card for outgoing calls |
| 06:14.58 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
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| 06:17.26 | BFAH | why's that? |
| 06:18.41 | BFAH | one of my clients want me to build them a predictive dialer. they're a small operation and have 4 POTS lines they want to use for 2 operators |
| 06:19.30 | BFAH | I was thinking about getting a 4 port fxo card and using vicidial or gnudialer |
| 06:19.45 | BFAH | do you think the quality would be crap? |
| 06:20.05 | pkunkra | i hear analog is not so good now. |
| 06:20.23 | pkunkra | digital lines like isdn are better |
| 06:20.50 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d07b274330418e8b) |
| 06:20.54 | BFAH | I was looking at digium cards but it looks like rhino is the only one that has fxo cards with hardware echo cancelation |
| 06:22.59 | BFAH | they have a t1 with 24 lines hooked up to a norstar pbx for everyone else |
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| 06:28.00 | WildPikachu | pkunkra, would be very nice if i can get my damn echo fixed ... heheeh |
| 06:32.46 | Raneth | I have problems with my Digium te120p card in asterisk ... |
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| 06:33.24 | pkunkra | wildpikachu: its usually pots stuff |
| 06:33.42 | pkunkra | or handsets |
| 06:34.14 | WildPikachu | well ... its a ISDN line ... seemed to work fine on our old asterisk installation |
| 06:34.49 | WildPikachu | and grandstream phones :( |
| 06:35.18 | Raneth | WildPikachu You use any EC cards? |
| 06:35.19 | pkunkra | what changed? |
| 06:35.38 | WildPikachu | i got a HFC-S card in now, plus a 4 port TDM fxo |
| 06:35.55 | WildPikachu | my pri card gives the same results |
| 06:36.02 | WildPikachu | (not tried the tdm yet) |
| 06:36.14 | WildPikachu | pri gives same results as the bri card that is |
| 06:36.17 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:36.25 | pkunkra | hmmm |
| 06:36.36 | Strom_M | WildPikachu: if it's not consistent, then it's far-end echo |
| 06:37.28 | WildPikachu | on some calls my staff report no echo, on others its very bad (then i modified some of the echo cancel settings and now you get slight crackling every few seconds) |
| 06:38.50 | mbit | which card? |
| 06:39.49 | WildPikachu | I have my HFS-C ISDN BRI card in atm |
| 06:40.05 | mbit | is that chan_capi? |
| 06:40.13 | WildPikachu | nope, zaphfc |
| 06:40.36 | mbit | might be worth testing the octasic echo canceller |
| 06:40.49 | WildPikachu | ooo ... how would I change which one is used? |
| 06:40.56 | *** join/#asterisk AJayMN (n=contact@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 06:41.16 | mbit | umm when you install octasic there instructions on how to change it |
| 06:41.29 | mbit | i assume your using asterisk from source |
| 06:42.11 | mbit | or you can try oslec |
| 06:42.24 | mbit | or the digium one |
| 06:42.28 | mbit | hpec |
| 06:43.54 | WildPikachu | whats the diff between the default echo canceller and octasic? |
| 06:44.13 | mbit | it is just far more advanced |
| 06:44.36 | mbit | the echo algorithm does alot more to get rid of more types of echo |
| 06:45.11 | WildPikachu | heh, but costs |
| 06:46.03 | *** join/#asterisk henkoegema (n=henkoege@d54C552E4.access.telenet.be) |
| 06:46.51 | henkoegema | q |
| 06:46.53 | mbit | yeah but can be cheap if it fixes the problems |
| 06:47.47 | *** join/#asterisk lbow (n=lbow@dsl-241-38-187.telkomadsl.co.za) |
| 06:50.58 | henkoegema | who has experience with the Portech MV-370 GSM Gateway? |
| 06:51.55 | henkoegema | i'm using one with asterisk. it works ok, except the DISA function |
| 06:52.35 | henkoegema | asterisk doesn't recognize the DTMF tones via the gateway |
| 06:52.36 | mbit | i just setup the portechs as one stage dialling in both ways |
| 06:53.06 | henkoegema | i'm using one stage dialling |
| 06:53.11 | mbit | have you set the portech to rfc2833 on the portech? |
| 06:53.44 | henkoegema | no to inbandf. rfc2833 didnt work. neither is inband |
| 06:55.29 | mbit | did you set your dtmfmode in asterisk when you set it to rfc2833 |
| 06:55.55 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:56.01 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:56.42 | henkoegema | i have to check what I have done. i have dtmfmode=inband (in asterisk) but I have to chech the gateway i think |
| 07:00.32 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 07:02.09 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4a406e96ff1a9ad8) |
| 07:03.20 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:05.09 | *** join/#asterisk saftsack (n=oliver@p54A7F43E.dip.t-dialin.net) |
| 07:05.50 | Strom_M | henkoegema: inband isn't going to work with GSM... |
| 07:07.07 | deegan | 1 |
| 07:12.42 | Qapf | would anyone happen to know how in the voicemail system to not have the asterisk voice say "please leave a message after the beep" and instead just play my greeting and then beep at people? |
| 07:17.32 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.128) |
| 07:18.44 | Strom_M | Qapf: the "s" option |
| 07:26.10 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 07:26.20 | Qapf | Strom_M, thanks, ill try to find out exactly how to stick that in within trixbox, at least i know where to look now |
| 07:26.52 | *** join/#asterisk saftsack (n=oliver@p54A7E126.dip.t-dialin.net) |
| 07:29.21 | pkunkra | has anyone tried out Digium's IAXy? |
| 07:29.29 | pkunkra | like it? hate it? |
| 07:29.42 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:30.42 | CCFL_Man2 | i hate it when boinc just work work on fbsd |
| 07:37.28 | *** part/#asterisk BFAH (i=Shaun@cblmdm72-241-21-108.buckeyecom.net) |
| 07:38.06 | Raneth | I have install Digium TE120P wildcard in asterisk server, but system cannot load module and asterisk dies.. hungs. |
| 07:38.27 | mbit | what module are you loading for the 120? |
| 07:38.47 | Raneth | wcte12xp |
| 07:39.04 | mbit | should be fine |
| 07:39.09 | mbit | zaptel compiled ok |
| 07:39.09 | mbit | ? |
| 07:39.14 | Raneth | yeah.. |
| 07:39.16 | Raneth | no problems.. |
| 07:40.07 | Raneth | modprobe wcte12xp and then system hungs or something tehn i hit ctrl z then it sayes that "Digium TE120P wildcard found" |
| 07:40.09 | Raneth | So strange :S |
| 07:40.33 | mbit | what kernel are you running? |
| 07:40.57 | Raneth | 2.6.9-34.0.2 |
| 07:41.08 | mbit | trixbox |
| 07:41.16 | Raneth | Yeah |
| 07:41.29 | mbit | what version of asterisk are you running |
| 07:41.35 | Qapf | Strom_M, ive tried sticking the option "s" into my voicemail line, but im still getting the computer voice giving instructions to the user. is there anything else i need to do? |
| 07:41.37 | Raneth | 1.2.18 |
| 07:42.23 | mbit | are you using the card as e1 or t1 |
| 07:42.29 | Raneth | e1 |
| 07:42.37 | mbit | have you changed the jumper on the card |
| 07:43.17 | Raneth | jumper is on, so its e1 mode |
| 07:43.57 | mbit | are you running the base version of asterisk with trixbox or a compiled version? |
| 07:44.03 | tzafrir | any bored vim fans out there? |
| 07:44.22 | Raneth | Im using base version of astersik |
| 07:44.25 | Raneth | asterisk* |
| 07:44.30 | tzafrir | http://vimperator.mozdev.org/ |
| 07:44.41 | tzafrir | First there was a Navigator, then there was an Explorer. Later it was time for a Konqueror. Now it's time for an Imperator, the VIMperator :) |
| 07:45.00 | many | heh |
| 07:45.25 | many | i really had to think hard what navigator supposed to be |
| 07:45.34 | many | now that tells you something about market shares |
| 07:45.37 | tzafrir | zaptel modules shouldn't be related to Asterisk version |
| 07:45.53 | tzafrir | Raneth, first thing to do is to look at kernel messages |
| 07:46.08 | tzafrir | for you it would be /var/log/messages |
| 07:46.17 | Raneth | one moment |
| 07:46.39 | tzafrir | What version of Zaptel do you have? |
| 07:46.46 | tzafrir | modinfo zaptel | grep ^version |
| 07:47.07 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 07:47.57 | Raneth | I think I found a error, Ill be right back |
| 08:00.00 | knerd2 | hello room |
| 08:00.48 | mvanbaak | Qapf: what asterisk version ? |
| 08:01.38 | mvanbaak | Qapf: 1.2 needs: VoiceMail(s<voicemailbox>@<context>) |
| 08:02.01 | mvanbaak | Qapf: 1.4 needs: VoiceMail(<voicemailbox>@<context>|s) |
| 08:05.19 | *** join/#asterisk RsaMan (n=aa@196.210.155.2) |
| 08:05.22 | RsaMan | greetigns |
| 08:05.35 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:06.25 | RsaMan | i would like to setup my caller ID correctly for my iax clients , i am set my calledid = my users names but would like to show have a called id number as well |
| 08:06.34 | RsaMan | what is the field in iax that does this ? |
| 08:08.56 | tzafrir | hi knerd2 |
| 08:10.17 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:10.32 | andyd | does srvlookup=yes by default in modern asterisks ? |
| 08:12.53 | pkunkra | wish there was a gsmplay program. |
| 08:13.36 | pkunkra | or at least something that will play it. |
| 08:13.53 | pkunkra | ... its the one time vlc has failed me. :-( |
| 08:14.11 | mvanbaak | RsaMan: callerid = "My name" <nr> |
| 08:16.28 | Qapf | mvanbaak, i managed to figure it out, it was asterisk version 1.2, but im an idiot and cant read and the option was on the freepbx page in plain text to supress alison's voice. sorry to bother everyone |
| 08:16.38 | tzafrir | pkunkra, try "play" (comes with sox) |
| 08:16.55 | tzafrir | if complied with libgsm support, it will play gsm |
| 08:17.54 | tzafrir | pkunkra, generally try play/sox first, as it is a very capable program. |
| 08:18.16 | tzafrir | as is display/convert (of ImageMagick) for images |
| 08:19.17 | pkunkra | tzfir |
| 08:19.18 | pkunkra | oh |
| 08:19.23 | pkunkra | i'll give that a shot |
| 08:19.27 | *** join/#asterisk RsaMan2 (n=aa@196.210.154.3) |
| 08:19.27 | RsaMan2 | hi |
| 08:19.32 | RsaMan2 | caller id question |
| 08:19.33 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:20.00 | RsaMan2 | using zoiper, as an iax client and the caller id only shows the person name if i set the caller ID number |
| 08:20.01 | RsaMan2 | in the client |
| 08:20.07 | RsaMan2 | how do i set this in the dialplan ? |
| 08:22.06 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:22.54 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 08:22.55 | pkunkra | hmmm |
| 08:30.58 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:31.29 | mvanbaak | RsaMan2: Set(CALLERID(num)=0123456789) |
| 08:31.54 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 08:32.47 | Uatec_ | how can i force disconnect a SIP client from the CLI? |
| 08:33.30 | implicit | what do you mean by disconnect |
| 08:33.41 | implicit | unregister? cut off in progres call? |
| 08:34.21 | Uatec_ | yes, unregister |
| 08:34.23 | Uatec_ | sorry |
| 08:34.27 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 08:34.59 | implicit | are you trying to prevent the client from receiving or making calls though? |
| 08:40.01 | *** join/#asterisk Berra (n=qwerty@portia.csbnet.se) |
| 08:41.37 | Berra | Is it possible to remove voicemail from the commandline? |
| 08:42.28 | Uatec_ | no i'm not |
| 08:42.45 | Uatec_ | it's just i have two clients that are connected with the same sip account (i changed the account details about) |
| 08:42.56 | Uatec_ | the first client is no longer connected |
| 08:43.04 | Uatec_ | but asterisk still lists the first client in "sip show peers" |
| 08:43.07 | Uatec_ | which is rather confusing |
| 08:46.43 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 08:51.34 | *** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 08:52.07 | SA007 | i'm trying to get dial out to work, it looks like it's almost working |
| 08:52.52 | SA007 | i've got an outgoing sip line (budgetphone) and an sip phone connected to 1234 |
| 08:53.42 | mvanbaak | Berra: it is. you have to remove the files in /var/spool/asterisk/voicemail/<context>/<voicemailbox>/INBOX/ |
| 08:53.46 | SA007 | i get this output in asterisk when i try to dail anything: http://pastebin.com/d5777570a |
| 08:54.48 | *** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk) |
| 08:55.01 | mvanbaak | SA007: like I told you yesterday: after I fixed the 'Loop detected' issue with budgetphone I ran into 'circuit-busy' trouble |
| 08:55.09 | mvanbaak | looks like you hit the same walls as I did |
| 08:55.29 | SA007 | don't start with the 'i told you so' stuff :P |
| 08:55.31 | Berra | mvanbaak: it's that simple? |
| 08:56.42 | Uatec_ | Berra, he's right. |
| 08:56.48 | *** join/#asterisk cheGGo (n=snafu__@gate.goobernetworks.com) |
| 08:56.49 | Uatec_ | there's a text file and a wav file |
| 08:56.50 | mvanbaak | Berra: I just did it on my home system |
| 08:56.55 | Uatec_ | if they're both gone, they're just gone |
| 08:56.55 | cheGGo | hi there |
| 08:57.02 | SA007 | the problem is that the normal phoneline here is dead (thanks kpn :S) and i've got this voip account and i don't see why it shouldn't be working as normal |
| 08:57.22 | mvanbaak | SA007: call their support |
| 08:57.24 | Berra | mvanbaak: nice, thank you |
| 08:57.35 | SA007 | lol, with what phone :D |
| 08:57.41 | mvanbaak | mobile ? |
| 08:57.50 | SA007 | that's very expensive |
| 08:58.08 | SA007 | mvanbaak: dit you get it fixed? |
| 08:58.11 | SA007 | did* |
| 08:58.18 | mvanbaak | I did |
| 08:58.58 | mvanbaak | but I cant remember what I had to do to fix it |
| 08:58.58 | SA007 | with their helpdesk? |
| 08:58.58 | cheGGo | mh, maybe anyone knows my problem |
| 08:58.58 | Wonka | disadvantageous plan, i'd say :) |
| 08:58.59 | cheGGo | i had realized a callback solution via callfiles through asterisk |
| 08:59.05 | Wonka | if mobile is too expensive |
| 08:59.10 | mvanbaak | SA007: no, their support did not help with asterisk back then |
| 08:59.25 | cheGGo | and i'm using the canreinvite option to give the rtp stream away |
| 08:59.25 | mvanbaak | but it's over a year ago |
| 08:59.56 | cheGGo | so, if the call is established and bridged.. the rtp stream is gone |
| 08:59.57 | cheGGo | but |
| 09:00.03 | Renacor | hmm I got extension s in my incoming but incoming calls never go through s, any reason? |
| 09:00.06 | cheGGo | if one of both callers hangup |
| 09:00.26 | cheGGo | asterisk tries to return the rtp stream to asterisk |
| 09:00.32 | cheGGo | with an reinvite |
| 09:01.04 | cheGGo | but on my pstn both calls are dead, so he cant reinvite anything and i get "request terminated" back from my pstn |
| 09:01.54 | cheGGo | did anyone know how to disable the last reinvite when one of both callers do hangup? |
| 09:01.54 | SA007 | mm, apparantly i get 404 user not found... |
| 09:03.18 | cheGGo | :( |
| 09:04.25 | *** join/#asterisk toot (n=tokeit@68.Red-83-37-17.dynamicIP.rima-tde.net) |
| 09:05.07 | Renacor | is there a reason asterisk would not go to the "s" extension in a context if no other extensions existed? |
| 09:05.18 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 09:05.56 | *** join/#asterisk ganga (n=ganga@59.95.246.210) |
| 09:07.59 | SA007 | looks like i keep getting SIP/2.0 407 Proxy Authentication Required |
| 09:14.24 | Renacor | anybody? |
| 09:14.50 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 09:15.16 | cheGGo | Renacor, u should use the s extensions for Macros only |
| 09:15.50 | cheGGo | or jumping with goto to an existing s priority in a context |
| 09:16.21 | cheGGo | not usable for direct context calls through any channel |
| 09:16.26 | Renacor | yeah |
| 09:17.11 | Renacor | http://pastebin.ca/680860 |
| 09:17.14 | Renacor | thats what Im trying to do |
| 09:17.32 | kaldemar | s matches to s, nothing else. |
| 09:18.06 | kaldemar | if a channel is defined as immediate, it will also look for s in the context. |
| 09:18.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:18.58 | Renacor | what I pastebinned should work |
| 09:19.05 | Renacor | I have done the same on an older asterisk server |
| 09:19.43 | kaldemar | what is your problem? |
| 09:19.51 | kaldemar | what does it do when you dial in? |
| 09:19.54 | Renacor | it's not executing s |
| 09:20.21 | Renacor | it rejects the call |
| 09:20.29 | Renacor | complaining the extension does not exist |
| 09:20.56 | kaldemar | what you pastebinned works only if the incoming channel is defined as immediate and the context is right. |
| 09:21.29 | kaldemar | pastebin the cli output for a call. |
| 09:21.39 | Renacor | kaldemar: my incoming context is what is attached to my pri line |
| 09:22.15 | Renacor | Extension '98700' in context 'incoming' from '+123123123123' does not exist. Rejecting call on channel 0/6, span 1 |
| 09:22.27 | Renacor | 98700 is the last 5 digits of the phone number |
| 09:22.29 | fujin_ | why do you have the space |
| 09:22.41 | kaldemar | it's looking for 98700, not s. |
| 09:22.45 | fujin_ | dude |
| 09:22.47 | fujin_ | read your shit |
| 09:22.49 | fujin_ | you have massive typos |
| 09:22.58 | Renacor | right but if 98700 does not exist it should go to s no? |
| 09:22.58 | fujin_ | mismatched curly brace |
| 09:23.10 | kaldemar | Renacor: no. only s matches to s. |
| 09:23.18 | Renacor | oops your right thanks fujin |
| 09:23.21 | fujin_ | http://pastebin.ca/680861 |
| 09:23.50 | fujin_ | and you probably want $["${DNID}" = "98700"] |
| 09:23.57 | fujin_ | dunno if you can match like that |
| 09:23.57 | Renacor | kaldemar: so how can i tell it to execute a dialplan even if it doesn't match the extension |
| 09:24.06 | fujin_ | Renacor, use the 'i' handler |
| 09:24.12 | cheGGo | sigh |
| 09:24.29 | kaldemar | Renacor: match it. |
| 09:24.32 | fujin_ | or that |
| 09:24.47 | cheGGo | anyone knows, why asterisk tries to get back the rtp stream after a hangup from one of both dialog partners? |
| 09:24.48 | fujin_ | doh, yeah; i handler won't work |
| 09:25.55 | cheGGo | had set up a callback through callfiles |
| 09:26.17 | cheGGo | if the connection is bridged, asterisk do a reinvite... thats exactly what i want |
| 09:26.29 | cheGGo | but, if one of them does a hangup |
| 09:26.49 | cheGGo | asterisk tried to get the rtp stream back (with another re-invite) |
| 09:27.06 | cheGGo | but at this moment both calls are dead |
| 09:27.23 | cheGGo | and i get a 487 response back (Request Terminated) |
| 09:27.35 | cheGGo | is it possible to avoid this behaviour? |
| 09:27.47 | Renacor | i doesn't match either |
| 09:28.01 | Renacor | i don't get it on my other phone server s matches just fine |
| 09:28.02 | fujin_ | do a regex match then |
| 09:28.34 | cheGGo | can anyone help meeeee? :( |
| 09:28.41 | fujin_ | depends on how calls are being sent to the context, Renacor |
| 09:29.45 | cheGGo | it must be possible to avoid this anyway :((( |
| 09:30.20 | fujin_ | You're doing it wrong. |
| 09:30.26 | cheGGo | me? |
| 09:30.35 | fujin_ | yep |
| 09:30.41 | cheGGo | how u mean? |
| 09:30.53 | fujin_ | think outside the square |
| 09:30.57 | fujin_ | it is not the spoon that bends |
| 09:31.09 | cheGGo | sigh |
| 09:31.37 | cheGGo | if i would understand it, i dont need to ask ;( |
| 09:32.06 | cheGGo | the callfile calls the first channel, if this is be answered |
| 09:32.18 | cheGGo | he jumped into the defined context and makes the second call |
| 09:32.29 | cheGGo | if this gets answered, he bridged |
| 09:32.35 | fujin_ | oh really? |
| 09:32.37 | cheGGo | where is my mistake? |
| 09:33.00 | fujin_ | somewhere in your brain, I'd say |
| 09:33.07 | cheGGo | omg |
| 09:33.21 | cheGGo | stfu if u couldnt help anyway |
| 09:33.57 | fujin_ | die in a fire |
| 09:35.38 | cheGGo | so, i understand, you IQ isnt high enough to me help |
| 09:35.46 | cheGGo | help me* ;) |
| 09:37.11 | fujin_ | no, you're doing something stupid and I refuse to help you |
| 09:37.14 | fujin_ | and therefore make fun at you |
| 09:37.21 | fujin_ | to try and provoke some kind of smart thinking patterns |
| 09:41.06 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 09:41.50 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 09:44.29 | JT | cheGGo: have a cry |
| 09:44.48 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 09:46.01 | cheGGo | fujin_, ok, but if i had any other idea, i dont need to ask |
| 09:46.27 | cheGGo | so, where is my error in reasoning |
| 09:49.53 | JT | draw a timing diagram or something |
| 09:50.09 | JT | it's actually a bit difficult to follow the broken english explanation |
| 09:51.18 | cheGGo | ok... indeed... my english isn't very well ;o) |
| 09:52.26 | cheGGo | i try to elaborate it cleary in a diagram |
| 10:08.32 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-10cd91627ef45402) |
| 10:08.53 | *** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 10:09.07 | SA007 | mvanbaak: you there? |
| 10:12.24 | *** join/#asterisk konqi_ (n=konqi@217.193.163.2) |
| 10:12.54 | konqi_ | Hello! Still looking for a way to pass an isdn-connection through asterisk... anybody who can help? |
| 10:14.46 | codejunky | konqi_: If you ask a concrete question then I think somebody may help you. |
| 10:16.31 | Renacor | what app can play gsm files in linux |
| 10:17.30 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 10:18.10 | konqi_ | i have a pc with an isdn-card and i want to make a data connection though asterisk. Configuration is ISDN-Card -> OldPBX -> Asterisk (with Digium Te420) -> Telco. Basically i believe i need help to set up a dialplan. |
| 10:25.11 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:29.11 | *** join/#asterisk saftsack (n=oliver@p54A7E8C2.dip.t-dialin.net) |
| 10:32.37 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 10:35.04 | michael-i | Hi everyone, I'm wondering if the TDM400p needs 5V or 12V for proper operation. I've only found references to it needing a "standard pc connector" and cannot tell on the pci board itself if is connecting to both positive supplies. |
| 10:36.34 | JT | does it matter? |
| 10:36.40 | JT | just plug it in :) |
| 10:37.22 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 10:38.11 | michael-i | I'm wiring up a custom connection on an embedded board and have access to a 5V source but not a 12V without doing some soldering :) |
| 10:38.16 | michael-i | just wondering if I can be lazy |
| 10:38.32 | JT | my guess would be 12V |
| 10:45.51 | *** join/#asterisk duxy786 (n=duxy786@comxodatchet2.plus.com) |
| 10:46.13 | duxy786 | Hi People |
| 10:47.12 | duxy786 | Aug 31 14:11:23 SM5-AST1 kernel: Uhhuh. NMI received for unknown reason 30 on CPU 0. |
| 10:47.12 | duxy786 | Aug 31 14:11:23 SM5-AST1 kernel: Do you have a strange power saving mode enabled? |
| 10:47.12 | duxy786 | Aug 31 14:11:23 SM5-AST1 kernel: Dazed and confused, but trying to continue |
| 10:47.35 | duxy786 | am getting the above errors and system is crashing, anyone know reasons to this? |
| 10:49.26 | *** join/#asterisk yassaccan (n=yassacca@admin186.hgo.se) |
| 10:54.01 | cheGGo | JT, right there? |
| 10:55.00 | cheGGo | http://www.nopaste.org/p/adFA0zkeF (my reinvite problem) |
| 10:55.11 | cheGGo | anyone can help me? |
| 11:01.00 | J4k3 | just say no to reinvitation? |
| 11:01.00 | J4k3 | :D |
| 11:02.15 | Renacor | anybody know how to play gsm files in linux? |
| 11:02.58 | cheGGo | J4k3, no, as i said, i want a reinvite on the bridge |
| 11:03.12 | cheGGo | no need to handle the media stream |
| 11:03.32 | duxy786 | has anyone come accross this error: |
| 11:03.37 | cheGGo | to avoid unnecessary traffic |
| 11:03.43 | duxy786 | Uhhuh. NMI received for unknown reason 30 on CPU 0 |
| 11:03.55 | cheGGo | duxy786, no sry, |
| 11:04.09 | cheGGo | Renacor, try audacity |
| 11:04.15 | Renacor | thanks |
| 11:04.39 | cheGGo | not sure if that can handle gsm files |
| 11:13.40 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:20.50 | *** join/#asterisk toot (n=tokeit@68.Red-83-37-17.dynamicIP.rima-tde.net) |
| 11:24.24 | *** join/#asterisk kkn088 (n=kikoun@84.4.216.243) |
| 11:26.55 | Renacor | how can I put an OR in an ExecIf ? |
| 11:31.56 | Renacor | exten => s,102,ExecIf($[${LANGOPT} != 1 & ${LANGOPT} != 2]| (Background(invalid), Goto(s,100))) <--- would that work? |
| 11:32.38 | kaldemar | why don't you try it? |
| 11:36.00 | Renacor | yeah not so much |
| 11:38.33 | *** join/#asterisk Galeras (n=Galeras@190.84.206.174) |
| 11:40.26 | konqi_ | how can i steal an incoming call from a callgroup or an extension? |
| 11:40.26 | cheGGo | kaldemar, may u can help me? http://www.nopaste.org/p/adFA0zkeF |
| 11:40.47 | cheGGo | re-invite issues |
| 11:41.30 | duxy786 | anyone out therE? |
| 11:41.43 | cheGGo | 4sure |
| 11:41.45 | Wonka | no |
| 11:42.00 | kaldemar | cheGGo: sorry, can't help you with that one. |
| 11:42.33 | cheGGo | real pity ;( |
| 11:42.46 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:51.19 | *** join/#asterisk Cyorxamp (i=Cyorxamp@212.57.229.111) |
| 11:52.35 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.182) |
| 11:57.36 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 11:57.45 | lirakis | morning everyone |
| 11:59.26 | *** join/#asterisk saftsack (n=oliver@p54A7BC07.dip.t-dialin.net) |
| 12:01.04 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 12:02.23 | *** join/#asterisk kkn088 (n=kikoun@84.4.216.243) |
| 12:05.39 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 12:05.52 | duxy786 | Kongi_, we have done this using software we have developed in house. |
| 12:06.27 | duxy786 | if you need more info.let know |
| 12:06.39 | duxy786 | let me know |
| 12:07.20 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 12:10.29 | *** join/#asterisk ManxPower (n=manxpowe@234.sub-70-216-152.myvzw.com) |
| 12:11.36 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:13.08 | defswork | are they any other prerecordings available ? the US accent one don't agree with some of my users |
| 12:18.49 | fujin_ | defswork, make a macro to record |
| 12:18.54 | fujin_ | get your receptionist to record everything |
| 12:18.56 | fujin_ | worked fine here |
| 12:19.13 | fujin_ | well, not receptionist here |
| 12:19.16 | fujin_ | but programmer |
| 12:20.29 | *** join/#asterisk skrusty (i=muad@xdev.net) |
| 12:22.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:22.37 | skrusty | quick question:anyone know how to set the source ip of an outbound sip connection (to a peer). I want to send calls to the same peer, but on different source ip's |
| 12:23.34 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 12:29.45 | Renacor | how can you match * ? |
| 12:29.49 | Renacor | im doing a read() |
| 12:29.59 | Renacor | and want to match if the variable read was a * |
| 12:30.51 | fujin_ | uh |
| 12:31.10 | fujin_ | if ($["${variable}" == "*"]) |
| 12:31.13 | fujin_ | (AEL) |
| 12:31.45 | fujin_ | same way you match any variable |
| 12:32.15 | Renacor | hmm that doesn't work |
| 12:32.21 | Renacor | not using AEL |
| 12:32.57 | Renacor | exten => s,104,ExecIf($[${MENUOPT} = "*"]|Goto|pls_menu,s,103) <--- that makes it freak out |
| 12:33.29 | JT | president bush just landed here |
| 12:33.37 | JT | insanse security |
| 12:33.50 | fujin_ | JT, kill him |
| 12:33.52 | fujin_ | do the world a favor |
| 12:33.54 | JT | haha |
| 12:33.59 | JT | you terrorist ;) |
| 12:34.04 | Renacor | Sep 4 17:38:21 WARNING[10072]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_MULT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 12:34.05 | fujin_ | Renacor, you do realise you can use both, simultaneously, right? |
| 12:34.17 | fujin_ | you have to put your shit into ""'s |
| 12:34.22 | Renacor | fujin: nope but thats good to know, thanks |
| 12:34.30 | fujin_ | like ExecIf($["${MENUOPT}" = "*"] |
| 12:34.37 | fujin_ | learn2$[] |
| 12:35.27 | Renacor | no worky |
| 12:35.45 | Renacor | exten => s,104,ExecIf($["${MENUOPT}" == "*"]|Goto|pls_menu,s,103 |
| 12:36.06 | Renacor | with the ) at the end of course |
| 12:36.55 | fujin_ | i dunno how execif works |
| 12:36.59 | fujin_ | is tha tlike gotoif? |
| 12:37.42 | Renacor | yeah |
| 12:40.03 | Renacor | meh, works with a GotoIf |
| 12:40.29 | *** join/#asterisk Jabeeds (n=jabeeds@117.210.dsl.mel.iprimus.net.au) |
| 12:40.56 | Jabeeds | Can someone please tell my why the following incoming context does not work: |
| 12:40.59 | Jabeeds | exten=> s, 1,Dial(SIP/${EXTEN}) |
| 12:41.08 | fujin_ | that's not an incoming context |
| 12:41.26 | *** join/#asterisk Zylkron (i=idjit@mempertahankan.agama.islam.org.ru) |
| 12:41.42 | Jabeeds | what is it then? |
| 12:41.58 | fujin_ | it's an exten line |
| 12:42.05 | fujin_ | not a context definition |
| 12:42.12 | cheGGo | fujin_, u may help me with my re-invite problem? sry for reacting so bad in the past, but i could get crazy with that re-invite issue |
| 12:42.18 | fujin_ | now, if that was inside a context, I could understand |
| 12:42.21 | fujin_ | cheGGo, no, sorry |
| 12:42.26 | Jabeeds | it is |
| 12:42.33 | Jabeeds | [incoming] |
| 12:42.34 | Jabeeds | exten=> s, 1,Dial(SIP/${EXTEN}) |
| 12:42.49 | fujin_ | so, what's it not doing? |
| 12:43.00 | fujin_ | not dialing? |
| 12:43.02 | fujin_ | not getting to 's'? |
| 12:43.07 | Jabeeds | trying to dial Sip/ |
| 12:43.12 | Jabeeds | nothing |
| 12:43.17 | fujin_ | well that's because ${EXTEN} doesn'twork |
| 12:43.22 | fujin_ | on the 's' handler |
| 12:43.40 | fujin_ | that'd be dumb |
| 12:43.45 | fujin_ | you probably want to do a wildcard match |
| 12:43.51 | fujin_ | *that* will work. |
| 12:44.01 | Jabeeds | any idea what i can use to read the CLID then? |
| 12:44.12 | fujin_ | uh |
| 12:44.17 | fujin_ | ${CALLERID(num)}? |
| 12:44.20 | fujin_ | like everyone else? :P |
| 12:44.32 | cheGGo | fujin_, cuz of our difference a short while ago? |
| 12:44.37 | Jabeeds | sorry i mean the destination not source |
| 12:44.40 | Zylkron | I fail registration it says no matching peers |
| 12:44.46 | Zylkron | can anyone help pls :P |
| 12:44.51 | fujin_ | Jabeeds, well, you have to make a place for it to go |
| 12:45.04 | fujin_ | like a wildcard match |
| 12:45.26 | fujin_ | exten => ._,1,Noop(inbound call to: ${EXTEN}) |
| 12:45.44 | fujin_ | Zylkron, is there a matching peer? |
| 12:46.24 | Jabeeds | thanks ill read up on that |
| 12:47.04 | fujin_ | well, not much to read up on apart from that ^^ |
| 12:47.24 | Zylkron | should I define a section named [mysipprovider] and set the type to peer? |
| 12:48.12 | Jabeeds | Ok, but how would that then dial the extension? |
| 12:48.29 | fujin_ | uh |
| 12:48.31 | fujin_ | with Dial |
| 12:48.47 | fujin_ | I'm not going to spoon feed you much more than that. |
| 12:49.31 | Zylkron | cuz I have problem configuring sip.conf :P |
| 12:50.09 | fujin_ | Zylkron, yes, you need a matching peer section |
| 12:50.11 | fujin_ | and a register line |
| 12:50.15 | fujin_ | to register to a remote peer |
| 12:50.36 | Jabeeds | I mean: Call comes in to 123456 from sip trunk, goes to incoming context. From the Noop command, how do I make it Dial Sip Ext 123456. Keeping in mind there are other indials coming over this trunk, so i cand just go Dial(SIP/123456) |
| 12:50.39 | Zylkron | what if its a localhost peer? |
| 12:50.56 | fujin_ | why would you have a localhost peer? |
| 12:51.11 | fujin_ | Jabeeds, use your brain |
| 12:51.11 | Zylkron | sorry Im really, like, really really new to this :P |
| 12:51.19 | fujin_ | Noop line just prints to console |
| 12:51.22 | fujin_ | replace noop with dial |
| 12:51.34 | fujin_ | Zylkron, yeah, but why would you have a localhost peer? I don't understand |
| 12:51.40 | ManxPower | Jabeeds: if a call comes in for an 123456 then exten => 123456,1,whatever will be executed. |
| 12:52.05 | fujin_ | Jabeeds, it's generally better to match all *expected* extensions, than do a wildcard match |
| 12:52.07 | Zylkron | fujin_: so I setup asterisk on my server, and installed xlite and when I try to register it says registration failed because of no matching peer |
| 12:52.10 | ManxPower | Jabeeds: there is no such thing as a "sip trunk" in Asterisk |
| 12:52.29 | fujin_ | Zylkron, you probably want a 'friend' then, not a peer |
| 12:52.34 | ManxPower | Zylkron: then you have no matching [whatever] section in sip.conf |
| 12:52.41 | fujin_ | peer is only one-way and requires asterisk->peer registration |
| 12:52.51 | ManxPower | peers do not REQUIRE registration. |
| 12:52.56 | fujin_ | oh |
| 12:53.03 | fujin_ | yeah |
| 12:53.03 | Jabeeds | Well, what would you call a peer that is passed many DIDs. |
| 12:53.08 | Zylkron | okay well, allright I didnt register peer anyway |
| 12:53.11 | Zylkron | but |
| 12:53.13 | ManxPower | Jabeeds: no. |
| 12:53.19 | Jabeeds | ?? |
| 12:53.22 | fujin_ | forgot, I've used a no-registration peer for SIPp testing |
| 12:53.35 | fujin_ | Jabeeds, a sip connection? |
| 12:53.39 | ManxPower | the peer/friend/user is passed the dialed number. |
| 12:53.42 | Zylkron | I defined 2 section |
| 12:53.56 | Zylkron | zylkron, type = friend |
| 12:53.57 | ManxPower | Zylkron: the "sip trunk" is just another SIP device, like a phone. |
| 12:53.58 | Zylkron | and |
| 12:54.06 | fujin_ | ManxPower, 'sip trunk' is unfortunately a widely accepted term now, as much as we hate it |
| 12:54.21 | JT | accepted by who? |
| 12:54.27 | fujin_ | sales men |
| 12:54.36 | ManxPower | fujin: So is "hacker" to mean "bad geek", but that does not have to mean we accept it. |
| 12:54.37 | Zylkron | and mysipserver , type is set to peer |
| 12:54.49 | fujin_ | telco operators who apply telco concepts to networks |
| 12:55.01 | fujin_ | don't get me wrong, I cringe whenever I read/hear sip trunk |
| 12:55.02 | ManxPower | So what. We do not use that term in Asterisk. |
| 12:55.07 | Zylkron | sacrebleu I spent hours workin on this shit :p |
| 12:55.20 | ManxPower | Calling it a "sip trunk" makes it look like you are using a GUI. |
| 12:55.25 | fujin_ | not really |
| 12:55.30 | fujin_ | it's a valid description of what is happening |
| 12:55.35 | fujin_ | multiple calls are being pumped down a peer |
| 12:55.38 | fujin_ | "trunking" |
| 12:55.40 | fujin_ | by definition |
| 12:55.59 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 12:56.03 | ManxPower | fujin: it also confuses people. |
| 12:56.08 | JT | it fails the description |
| 12:56.12 | ManxPower | as iax2 trunking is totally different. |
| 12:56.17 | ManxPower | so are telco "trunking" |
| 12:56.19 | Zylkron | anyone have a quix sample of a working sip.conf :P |
| 12:56.25 | fujin_ | yes, but not vlan trunking |
| 12:56.33 | ManxPower | Zylkron: pretty much every one on the web. |
| 12:56.33 | fujin_ | which is probably more what the concept is borrowed from |
| 12:56.44 | Jabeeds | exactally where i got it from |
| 12:56.50 | ManxPower | Zylkron: what is the specific issue you are having problems with? |
| 12:56.54 | JT | don't justify the unjustifiable :) |
| 12:57.09 | Zylkron | ManxPower: I think my sip.conf configuration is wrong |
| 12:57.15 | fujin_ | Zylkron, configs/sip.conf.sample |
| 12:57.24 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:57.26 | ManxPower | Zylkron: you can either answer my questions or have someone else help you. |
| 12:57.44 | Zylkron | the asterisk server is up, I configured xlite, tried to register, and it failed cuz of no matching peers |
| 12:57.49 | fujin_ | right, I'm outta here |
| 12:57.50 | fujin_ | bai |
| 12:57.50 | ManxPower | Now what is the specific issue? Calls come in and get rejected? Calls never arrive? Calls are sent to the wrong phone? |
| 12:58.05 | ManxPower | Zylkron: PASTE the error message! |
| 12:58.05 | Zylkron | cant even register, sir :P |
| 12:58.11 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:58.17 | Zylkron | one sec |
| 12:59.06 | ManxPower | I have 5 mins before I have to leave for work, so you have that much time. |
| 12:59.06 | Zylkron | good enough :) |
| 13:00.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:00.52 | Zylkron | too much multi tasking , really lagged :P |
| 13:02.45 | Jabeeds | When I put "exten => ._,1,Noop(inbound call to: ${EXTEN})" into my incoming context, the output is "h". What exactally is that and why is the actual indial number not output? |
| 13:02.51 | Zylkron | <PROTECTED> |
| 13:02.54 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:03.14 | ManxPower | Do you have a [100] section in sip.conf? |
| 13:03.20 | Zylkron | yup |
| 13:03.47 | ManxPower | Jabeeds: _. matches TWICE for each call, once for the real exten, once when the call gets hung up and exten h is run. |
| 13:03.48 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 13:03.56 | ManxPower | you are missing the first match, Jabeeds |
| 13:04.12 | ManxPower | Zylkron: put the [100] section on pastebin.ca |
| 13:05.50 | ManxPower | OK, time is up. |
| 13:05.52 | ManxPower | have a nice life. |
| 13:06.00 | Jabeeds | Thanks Manx |
| 13:06.10 | Zylkron | darn |
| 13:07.45 | Jabeeds | exit |
| 13:07.54 | Uatec | from my sip phone (SIP/sparedesk) i'm dialing (SIP/mytrunk/123) |
| 13:08.07 | Uatec | but the other asterisk box on the receiving end of the trunk is sending back this message: |
| 13:08.08 | Uatec | Sep 4 14:07:18 WARNING[6936]: chan_sip.c:9856 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"sparedesk" <sip:sparedesk@10.20.20.251>;tag=as6f929788' |
| 13:08.18 | Uatec | it's not supposed to be authenticating as "sparedesk" |
| 13:08.22 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:08.27 | Uatec | the local asterisk box is supposed to be authenticating |
| 13:08.57 | etfonhomey | ManxPower, Have you setup * for a client with a "dynamic T1"? |
| 13:10.41 | Uatec | how can i make the receiving asterisk box not require a password from the originating device? |
| 13:10.59 | Uatec | i've already got: insecure=invite,port in the right section of the sip.conf |
| 13:11.57 | cheGGo | Hi, there, anyone knows, why asterisk is re-inviting again when one of both dialog partners hangup? |
| 13:12.08 | cheGGo | http://www.nopaste.org/p/adFA0zkeF |
| 13:12.12 | cheGGo | i'm getting crazy |
| 13:16.23 | _x86_ | pbx.c:1700 pbx_extension_helper: No application 'Dial' for extension |
| 13:16.26 | _x86_ | what's this mean? |
| 13:16.37 | _x86_ | this morning some how i no longer have a Dial application |
| 13:17.38 | _x86_ | any idea why? |
| 13:21.21 | _x86_ | and I load app_dial.so, and still have no Dial application?! |
| 13:32.14 | _x86_ | http://pastebin.ca/680996 |
| 13:32.18 | _x86_ | this is not good... |
| 13:32.25 | _x86_ | channel not implemented? |
| 13:32.39 | Uatec | that's weird |
| 13:32.45 | Uatec | i have a snom and a linksys |
| 13:32.52 | Uatec | the linksys will work over the SIP trunk |
| 13:32.56 | Uatec | the snom wont |
| 13:33.05 | JT | it's not a trunk |
| 13:33.06 | Uatec | they're connected EXACTLY the same |
| 13:33.09 | Uatec | sorry |
| 13:33.13 | Uatec | sip channel |
| 13:33.32 | Uatec | everybody else in my office who talks about sip is in marketting, so they call it a sip trunk |
| 13:33.36 | Uatec | i've given up correcting them |
| 13:33.49 | Uatec | the point is |
| 13:33.52 | Uatec | the snom doesn't work |
| 13:33.55 | Uatec | and the linksys does |
| 13:33.58 | Uatec | which is all very strange |
| 13:34.12 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 13:34.59 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 13:35.03 | ZaVoid | morning guys |
| 13:35.19 | [TK]D-Fender | _x86_: rIGHT NOW i'D SUSPECT YOU MIGHT HAVE SCREWED UP YOUR MODULES.CONF. |
| 13:35.26 | ZaVoid | is there a way to show my license file for g.729 in use on an asterisk? show g729 only shows the number of licenses not the license code.. |
| 13:35.37 | [TK]D-Fender | _x86_: those are 2 important modules that should ahve gotten autoloaded. |
| 13:36.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:36.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:36.49 | cheGGo | hi anthm |
| 13:36.58 | anthm | hello. |
| 13:37.26 | cheGGo | how r u? |
| 13:37.40 | awk | hmm where can I get nice support on asterisk manager api |
| 13:37.55 | anthm | ok |
| 13:37.59 | anthm | yourself? |
| 13:39.00 | *** join/#asterisk Galeras (n=Galeras@190.84.206.174) |
| 13:40.03 | cheGGo | fine... thanks, but getting crazy with asterisk %) |
| 13:40.16 | *** join/#asterisk korihor (n=humberto@190.75.38.113) |
| 13:40.58 | cheGGo | to tear my hair :-( |
| 13:41.04 | ZaVoid | lol i been there cheGGo |
| 13:41.09 | ZaVoid | why you making your hair come out? |
| 13:41.42 | cheGGo | cuz of my re-invite issue with callback ;/ |
| 13:41.50 | ZaVoid | what happens? |
| 13:42.10 | cheGGo | asterisk tried to make a re-invite after one of both call legs hangup |
| 13:42.24 | cheGGo | i initiate the callback via callfiles |
| 13:42.45 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-a2a163d34205890a) |
| 13:42.55 | cheGGo | first, calling the channel directly via SIP/0049xxx@peer |
| 13:43.11 | cheGGo | if answered, jumping to an definied context |
| 13:43.19 | cheGGo | and dials the second number |
| 13:43.35 | cheGGo | if answered, asterisk makes a bridge and a re-invite |
| 13:43.47 | cheGGo | thats exactly what i want |
| 13:44.01 | *** join/#asterisk ESCulapio_ (n=elvyn@66.44.88.200.l.sta.codetel.net.do) |
| 13:44.13 | cheGGo | but, if one of the opposites hang up the call |
| 13:44.33 | cheGGo | asterisk send another re-invite |
| 13:45.20 | cheGGo | to fetch back the rtp stream for the channel who had not hangup |
| 13:46.02 | cheGGo | <PROTECTED> |
| 13:46.04 | ESCulapio_ | Hi, I have a problem when compiling asterisk-addons. I have the following error |
| 13:46.14 | cheGGo | ZaVoid, any ideas? |
| 13:46.23 | ESCulapio_ | cdr_addon_mysql.c:292: error: too few arguments to function ‘ast_config_load’ |
| 13:46.59 | ZaVoid | nope sorry :( |
| 13:47.04 | ZaVoid | but i understand the hiar pull out |
| 13:47.21 | ESCulapio_ | somebody can help with the following error when compiling asterisk-addons me |
| 13:47.48 | cheGGo | pity ;( |
| 13:47.52 | ESCulapio_ | cdr_addon_mysql.c:292: error: too few arguments to function ‘ast_config_load’ |
| 13:47.52 | ESCulapio_ | make[1]: *** [cdr_addon_mysql.o] Error 1 |
| 13:48.05 | JunK-Y | ESCulapio_: wrong version of * installed? |
| 13:48.21 | ESCulapio_ | the version trunk svn |
| 13:48.24 | cheGGo | anthm, may u know whats wrong with my asterisk? |
| 13:48.31 | cheGGo | http://www.nopaste.org/p/adFA0zkeF |
| 13:48.35 | JunK-Y | and which version of addons? |
| 13:48.38 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.56.120) |
| 13:48.43 | ESCulapio_ | JunK-Y, svn trunk |
| 13:49.29 | ESCulapio_ | JunK-Y, but already it tries with other versions of addons 1.4.2, 1.4.1 |
| 13:49.46 | ESCulapio_ | and continuous the same error |
| 13:50.00 | *** join/#asterisk alejandro (n=asanchez@kde/developer/alejandro) |
| 13:50.01 | _x86_ | strange |
| 13:50.14 | _x86_ | restarting the entire server fixed the problem where i could not use zap channels |
| 13:50.18 | anthm | what if you add another priority to cb-calee after DIAL that is hangup ? |
| 13:50.21 | JunK-Y | thats normal, ast_config_load has changed in trunk. |
| 13:50.29 | JunK-Y | hiya anthm |
| 13:50.35 | anthm | hey JunK-Y |
| 13:51.02 | justdave | got an IP501... it pulls the .bmp file off the tftp server, but it just shows a blank spot on the screen where it's supposed to be |
| 13:51.39 | datachomper | justdave, I've got custom images in my polycom 501's |
| 13:52.05 | JunK-Y | ESCulapio_: i just tried both trunk, it works great here. |
| 13:52.17 | cheGGo | anthm, same issues :( |
| 13:52.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:52.21 | datachomper | Is your image in the right format? Correct size for your model and 16bpp ? |
| 13:52.56 | cheGGo | before i changed it, there were another priorities after the DIAL command |
| 13:52.56 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 13:53.11 | cheGGo | i thought that could be the reason, so i changed it |
| 13:53.23 | justdave | file says: PC bitmap data, Windows 3.x format, 114 x 51 x 4 |
| 13:53.34 | cheGGo | but unfortunately the same error |
| 13:54.00 | cheGGo | i think asterisk want to fetch back the rtpstream of the second call |
| 13:54.14 | datachomper | It should be 112x52 |
| 13:54.36 | ESCulapio_ | JunK-Y, but I have the same error with all the versions |
| 13:54.45 | anthm | does it matter which side hangs up? |
| 13:54.56 | ESCulapio_ | that I can do? |
| 13:55.39 | cheGGo | anthm, no it doesnt matter... on both sides same behaviour |
| 13:56.19 | anthm | so the hangup of one is not propagating across to the other |
| 13:56.29 | anthm | because the call is not answered |
| 13:56.38 | JunK-Y | ESCulapio_: make sure u run both trunk. |
| 13:57.12 | cheGGo | anthm, no... the other side hangs up too |
| 13:58.18 | cheGGo | actually i get a BYE from one caller |
| 13:58.26 | cheGGo | then Asterisk sends an OK |
| 13:58.37 | cheGGo | and directly another invite |
| 13:59.04 | cheGGo | (which is tagged as re-invite in sip debugging mode) |
| 13:59.09 | anthm | as soon as it gets the bye on the 1 leg it should cancel the other call |
| 13:59.28 | cheGGo | yes, indeed! thats what i thought |
| 13:59.32 | anthm | the callflow is different depending on which side hangs up first but the effect is the same |
| 13:59.39 | cheGGo | but, asterisk tried to get back the rtp stream for call leg 2 |
| 13:59.56 | justdave | we had another image we used to be using that worked, and that one was 114x51, so I just kept the size. That's interesting that the old one worked if the size was wrong |
| 13:59.57 | anthm | if A leg hangs up, it should send CANCEL to B leg |
| 14:00.12 | anthm | if B hangs up it should send BYE to A |
| 14:00.13 | justdave | but yes, indeed, changing that size fixed it |
| 14:00.16 | cheGGo | thats not happened |
| 14:00.31 | anthm | it's not findiong out fast enough probably |
| 14:00.36 | cheGGo | i got everytime a BYE from my PSTN Gateway |
| 14:00.39 | justdave | thanks |
| 14:00.42 | cheGGo | if one call is hang up |
| 14:01.06 | cheGGo | the call is going to my PSTN gateway via SIP |
| 14:01.25 | anthm | there are several codepaths in asterisk where it does some prolonged actions and is not aware it has to react to the state of 2 calls not just 1 |
| 14:01.29 | cheGGo | and if anyone hangup, i got a BYE from them |
| 14:01.45 | cheGGo | sure |
| 14:02.01 | anthm | your problem is probably in app_dial during the code that establishes a call |
| 14:02.33 | cheGGo | thats what i dont know |
| 14:02.38 | anthm | if not it's in chan_sip itself where the reinvite code is a blocking function that is not checking for the call being cancelled along the way |
| 14:02.59 | cheGGo | aeh |
| 14:03.00 | cheGGo | ah |
| 14:03.04 | cheGGo | i forget |
| 14:03.15 | cheGGo | after the re-invite |
| 14:03.29 | *** join/#asterisk Seyr (n=Seyr@c-98-194-30-143.hsd1.tx.comcast.net) |
| 14:03.35 | cheGGo | asterisk terminate the call itself |
| 14:03.45 | cheGGo | by sending a BYE paket to our pstn |
| 14:04.12 | cheGGo | than, i got 487 Request Terminated back from our PSTN |
| 14:04.20 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:04.24 | Seyr | I have a Peer defined and "sip show peers" shows it, but when I call in from it, Asterisk says "Found no matching peer or user for" |
| 14:04.26 | Seyr | any idea? |
| 14:05.15 | cheGGo | anthm, i thought that asterisk terminates both call legs on the BYE from our PSTN |
| 14:05.35 | [TK]D-Fender | Seyr: I think a PASTEBIN of the CLI output of your failed attempt with SIP debug enabled and a copy of your sip.conf would be an IDEA |
| 14:05.50 | cheGGo | but, sends another re-invite and THAN terminating the last call |
| 14:06.04 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 14:06.19 | Lucky7 | What would cause 1) the phone to cut off the second it hits a voicemail |
| 14:06.23 | anthm | you should look at a trace |
| 14:06.28 | Zeeek | IDEAs are cheap |
| 14:06.29 | Lucky7 | and 2) some lagtime, on a T1 line |
| 14:06.42 | anthm | your best bet is to collect as much data as you can and file it in a bug report |
| 14:06.48 | anthm | fill sip trace |
| 14:06.57 | anthm | wireshark pcap |
| 14:07.07 | anthm | the logs with the debug cranked |
| 14:07.16 | *** join/#asterisk Vanisher (n=Vanisher@s55916da7.adsl.wanadoo.nl) |
| 14:07.31 | Vanisher | Hi folks, anyone running asterisk within a VE (OpenVZ)? |
| 14:09.06 | Seyr | [TK]D-Fender: I typed the exact error |
| 14:09.30 | Seyr | Peer is defined, shows in "show peers" ... but when call comes in it says "Found no matching peer" |
| 14:09.31 | [TK]D-Fender | Seyr: Sorry, but clearly I want to see the EXACT CALL that causes the error. |
| 14:09.45 | [TK]D-Fender | Seyr: and no, I don't trust your setup. Period |
| 14:09.47 | *** join/#asterisk mog (i=mog@nat/digium/x-7ec2769cd702ee09) |
| 14:09.47 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:09.58 | Qwell | mog: ! |
| 14:10.11 | [TK]D-Fender | Seyr: Why should I when its not working? :) |
| 14:10.14 | Seyr | [TK]D-Fender: I havent been here in 6 months.. nice to know your attitude is the same |
| 14:10.54 | [TK]D-Fender | Seyr: All part of the service :p You should still know better than to paste the error without backup as to what iriginates your error. |
| 14:11.16 | mog | Qwell: ! |
| 14:11.27 | Qwell | mog: has you connection at home been sucking lately? |
| 14:11.42 | Qwell | your* |
| 14:11.49 | mog | ? |
| 14:11.54 | Qwell | comcast |
| 14:11.55 | mog | a little slow |
| 14:11.58 | mog | but not bad |
| 14:12.07 | mog | sat morning actually now that i think about it was bad |
| 14:12.11 | Seyr | [TK]D-Fender: Ah, sorry.. this will better help I think: http://bugs.digium.com/view.php?id=6069&nbn=1 |
| 14:12.17 | Qwell | until around...11? |
| 14:12.25 | mog | no around 9 |
| 14:12.26 | Seyr | I have the exact same problem.. for the most part, but "insecure=port" doesnt help |
| 14:12.29 | mog | i was gone by 10 |
| 14:12.39 | Qwell | did you get online much at night this weekend? |
| 14:12.42 | mog | i didnt spend much of hte weekend at home |
| 14:12.49 | mog | is it still sucky for you |
| 14:12.54 | Qwell | yeah |
| 14:13.45 | [TK]D-Fender | Seyr: that is an ancient bug, and long listed as closed. Could you please just provide the info I asked for.... |
| 14:14.30 | Lucky7 | hm |
| 14:14.32 | Lucky7 | I've got a PBX |
| 14:14.36 | *** join/#asterisk cayorde (n=flexable@87.19.166.253) |
| 14:14.43 | Lucky7 | 1% Load |
| 14:14.54 | Lucky7 | 40% of memory useage |
| 14:15.14 | Lucky7 | And i'm getting alot of sales people complaining of "lagtime" between the two |
| 14:15.17 | cheGGo | anthm, than contact asterisk-dev mailinglingst? |
| 14:15.26 | Lucky7 | that he'll talk, there'll be a long pause, and then he'll get the responce |
| 14:15.31 | anthm | no the bug tracker? |
| 14:15.34 | cheGGo | ah |
| 14:15.36 | cheGGo | sure :) |
| 14:15.43 | anthm | that's what it's for |
| 14:15.52 | anthm | you collect evidence |
| 14:15.55 | anthm | and put it on there |
| 14:16.05 | anthm | and then ppl can look to it as a file on the issue |
| 14:16.11 | Qwell | be sure to assign it to anthm |
| 14:16.17 | Qwell | (I kid, I kid) |
| 14:16.31 | anthm | i'm a little rusty but you neva know |
| 14:16.46 | Qwell | anthm: You're more than welcome to help with bugs :) |
| 14:17.20 | cheGGo | ok, so u think thats a bug, and not a feature? =) |
| 14:17.30 | Lucky7 | will a /etc/init.d/asterisk reload drop calls? |
| 14:17.38 | cheGGo | Lucky7, 4sure |
| 14:17.39 | Qwell | Lucky7: depends on your init script |
| 14:17.49 | anthm | it's an unwanted behaviour |
| 14:17.51 | Lucky7 | great. |
| 14:17.51 | cheGGo | on the default init |
| 14:17.53 | anthm | i assume |
| 14:17.54 | Qwell | if it just does `asterisk -rx "reload"`, then no |
| 14:18.05 | Lucky7 | Ok, thanks qwell |
| 14:18.13 | anthm | if you say you see both calls terminate |
| 14:18.14 | cheGGo | oh |
| 14:18.20 | Lucky7 | lame |
| 14:18.24 | anthm | and then re-invite after that |
| 14:18.29 | anthm | i guess that is a bug |
| 14:18.44 | anthm | so my best guess w/o actually looking at it |
| 14:18.50 | cheGGo | yeah, but as i said, first asterisk tries to make a re-invite |
| 14:19.03 | cheGGo | than he sends a BYE packet by itself |
| 14:19.05 | anthm | is in the re-invite code you need to check that the call is not terminated or cancled |
| 14:19.13 | anthm | well |
| 14:19.16 | anthm | if it sends it before |
| 14:19.25 | anthm | then it has no idea it's doing anything wrong |
| 14:19.33 | anthm | then in that case |
| 14:20.11 | anthm | your problem is when one leg terminates the other doesn't realize it fast enough |
| 14:20.18 | cheGGo | indeed! |
| 14:20.19 | cheGGo | yes! |
| 14:20.33 | anthm | is it all really fast ? |
| 14:20.43 | anthm | like the invite and bye are withing a few ms? |
| 14:21.21 | Seyr | [TK]D-Fender: got it, just set "insecure=yes" |
| 14:21.33 | cheGGo | nope |
| 14:21.42 | anthm | far apart ? |
| 14:22.06 | cheGGo | nor |
| 14:22.10 | cheGGo | just 1 sec |
| 14:22.19 | cheGGo | but minimum 1 sec |
| 14:22.21 | Seyr | [TK]D-Fender: I had tried what was posted in that old bug.. insecure=port,invite and it did not work. So I changed it to just "insecure=yes" :-) works perfect |
| 14:22.22 | cheGGo | no ms |
| 14:22.30 | cheGGo | so |
| 14:22.47 | anthm | ok you see bye or cancel of the one side? |
| 14:22.54 | anthm | then 1 second goes by? |
| 14:22.59 | anthm | then it sends an invite? |
| 14:23.09 | cheGGo | i see the BYE of one side |
| 14:23.17 | *** part/#asterisk Seyr (n=Seyr@c-98-194-30-143.hsd1.tx.comcast.net) |
| 14:23.20 | cheGGo | then asterisk sends immediatly |
| 14:23.24 | cheGGo | an OK |
| 14:23.40 | cheGGo | and immediatly the re-invite |
| 14:24.04 | cheGGo | than after 1 sec came the BYE |
| 14:24.27 | cheGGo | mom |
| 14:24.37 | anthm | so probably |
| 14:24.40 | codejunky | Hello, if I want to connect multiple dect phones with asterisk, and want to give every phone a number which hardware do you recommend? |
| 14:24.53 | anthm | the reinvite code doesn't know that the other leg is dead |
| 14:25.02 | codejunky | I want them callable via sip. :) |
| 14:25.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:25.21 | anthm | and the hungup leg does not do something to terminate the other leg |
| 14:25.33 | anthm | so the invite getting rejected is what hangs up the other leg |
| 14:25.45 | anthm | not the fact that the opposite leg was terminated |
| 14:26.35 | anthm | you need to find out if the code is still in app_dial or has moved on to res_features or the core bridge stuff |
| 14:26.37 | Zeeek | codejunky how many is multiple? more than 4? |
| 14:26.43 | anthm | or whatever it does now |
| 14:26.51 | anthm | cos I am out of touch on any code past 1.2 |
| 14:26.55 | *** join/#asterisk jfitzgibbon (n=NADT@64.72.237.130) |
| 14:27.01 | codejunky | Zeeek: Which one would you recommend for 4 and which for more than 4? :) |
| 14:27.07 | cheGGo | anthm, ok |
| 14:27.22 | anthm | bbl |
| 14:27.29 | Zeeek | for up to 4 you could use a TDM400 (pricey but works) |
| 14:27.46 | Zeeek | or four IAXy |
| 14:27.56 | [TK]D-Fender | EW |
| 14:28.14 | Zeeek | there are other mfrs but I don't know their specific products |
| 14:28.34 | [TK]D-Fender | codejunky: These phones have a BASE included? |
| 14:28.34 | Zeeek | four IAXy lined up would be cute AND reduce your home heating bill |
| 14:28.41 | codejunky | [TK]D-Fender: No. |
| 14:28.42 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 14:28.43 | [TK]D-Fender | IAXY = suck |
| 14:28.51 | kombi | Hi fender! |
| 14:29.02 | Zeeek | my little IAXy has been working perfectly for three years |
| 14:29.12 | [TK]D-Fender | codejunky: Find one with a SIP base. Seimens makes a few. |
| 14:29.24 | kombi | when doing reload in cli while in operation, are calls lost? |
| 14:29.53 | codejunky | [TK]D-Fender: I want that people can bring there dect phone and I can put it in my phone network. :) |
| 14:29.57 | [TK]D-Fender | Zeeek: Low on features, only usable with *, no web interface, icky provisioning, fugly, and not COST EFFECTIVE EITHER. IAXY = suck. |
| 14:30.00 | Sweeper | kombi: shouldn't be |
| 14:30.07 | kombi | thanks sweeper! |
| 14:30.24 | Aeudian | Has anybody been able to configure the Linksys SPA400 (POTS Gateway) with asterisk. I can make inbound/outbound phone calls, but registration fails telling my password is FORBIDDEN. |
| 14:30.33 | Zeeek | [TK]D-Fender I'll give you the cost effective point, but not the rest |
| 14:30.58 | Zeeek | rthere are multi port SIP ATA from Sipura though now I think on it |
| 14:31.00 | [TK]D-Fender | Zeeek: Has a functional web interface? Works with other PBX's than *? |
| 14:31.03 | kombi | and the other thing, are exten => foo,1,Answer() and foo/bar,1,Answer() treated as completely separate extensions? |
| 14:31.16 | [TK]D-Fender | Zeeek: SPA-8000 <-. TDM400 = trouble |
| 14:31.24 | Zeeek | no, I dismiss your argument about only working with * because that is within the constraints of our topic |
| 14:31.41 | kombi | ..got to block a caller while the pbx is running.. |
| 14:31.46 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:31.47 | [TK]D-Fender | kombi: yes. If the CID matches "bar" then the latter will get executed |
| 14:32.14 | Zeeek | as for web interface, muhahaha better none than the Polycom 3 minute booter with reboot for any single change |
| 14:32.25 | [TK]D-Fender | Zeeek: Doesn't gorgo it being a dead-end solution. Going "short-term" is a great way to keep paying your whole life. |
| 14:32.31 | kombi | fender: should exten => foo/bar,1,Answer be first or does order not matter? |
| 14:32.40 | Zeeek | gorgo? |
| 14:32.54 | Zeeek | lmeets godzilla |
| 14:32.58 | [TK]D-Fender | Zeeek: Thing with Polycom is you don't go changing it every minute, and at elast its comprehensive and quality. |
| 14:33.08 | Zeeek | gorgo meets godzilla, great film |
| 14:33.14 | Aeudian | Is there a guide explaining how to configure asterisk to pickup a phone call on hold on a specific phone, like a remote hold pickup |
| 14:33.25 | Zeeek | Aeudian parking |
| 14:33.50 | Aeudian | zeeek, you mean like a parking lot? |
| 14:34.01 | Zeeek | yeah but that'zs calls in slots, not phones |
| 14:34.09 | Zeeek | so maybe not what you want |
| 14:34.48 | Aeudian | zeeek: but is there a way say in PhoneA has a call on hold, and user goes to say PhoneB in warehouse, can the user pickup PhoneA from the warehouse? |
| 14:34.54 | Zeeek | [TK]D-Fender all joking aside, and I was joking about four IAXy, the unit is very handy in certain situations |
| 14:35.21 | Zeeek | Aeudian I don't think so, but I've been know to be wrong more often than right ;) |
| 14:36.07 | Aeudian | zeeek: blah i hate call parking lol |
| 14:36.11 | Zeeek | as a matter of fact, when phone A puts a call on hold, AFAIK ONLY phone A can recover it |
| 14:36.50 | *** join/#asterisk ManxPower (n=manxpowe@41.sub-70-218-15.myvzw.com) |
| 14:36.53 | Zeeek | but someone may jump in and prove me wrong |
| 14:37.30 | Zeeek | call parking is great when you have like 200 calls at once :) |
| 14:42.49 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 14:43.13 | WildPikachu | how would I group say 5 channels into a trunk for outgoing calls?\ |
| 14:43.37 | WildPikachu | would it be callgroup? |
| 14:43.46 | elixer | i think its just group |
| 14:44.11 | [TK]D-Fender | Aeudian: Go lookup "call parking" ont he WIKI |
| 14:44.19 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:44.29 | [TK]D-Fender | WildPikachu: what KIND of channels? |
| 14:44.49 | WildPikachu | 10 pri's channels and 4 fx0 |
| 14:45.02 | WildPikachu | they not in sequence |
| 14:45.17 | *** join/#asterisk implicit_ (n=implicit@vc240146.vpn.uci.edu) |
| 14:45.48 | *** join/#asterisk cybertooth (n=cybertoo@cpe-075-182-111-118.nc.res.rr.com) |
| 14:45.49 | [TK]D-Fender | WildPikachu: yes, you can group your zap channels together in several different combinations |
| 14:46.20 | WildPikachu | do i use group= or callgroup=? i'm currently using Zap/g1 to dial outgoing, but i need only certain channels to be used |
| 14:46.32 | WildPikachu | g1 i suspect is my pri |
| 14:47.40 | cybertooth | switchtype = national |
| 14:47.40 | cybertooth | signalling = pri_cpe |
| 14:47.40 | cybertooth | group = 2 |
| 14:47.40 | cybertooth | context = From_LVL3 |
| 14:47.40 | cybertooth | channel => 73-95 |
| 14:47.52 | cybertooth | switchtype = national |
| 14:47.52 | cybertooth | signalling = pri_cpe |
| 14:47.52 | cybertooth | group = 1 |
| 14:47.52 | cybertooth | context = From_PSTN |
| 14:47.52 | cybertooth | channel => 49-71 |
| 14:47.58 | Zeeek | oh oh |
| 14:48.07 | [TK]D-Fender | WildPikachu: "group" |
| 14:48.14 | cybertooth | "group =" |
| 14:48.16 | WildPikachu | what is callgroup used for? |
| 14:48.21 | [TK]D-Fender | cybertooth: please don't spam in here |
| 14:48.35 | [TK]D-Fender | WildPikachu: Go look it up on the WIKI and READ |
| 14:48.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 14:48.46 | WildPikachu | :) |
| 14:49.11 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-47-114.pskn.east.verizon.net) |
| 14:49.55 | WildPikachu | spamassassin < cybertooth |
| 14:50.36 | elixer | cybertooth: pasting more than a couple (i.e. 2) lines is considered spam. for anything more, use a pastebin. |
| 14:50.38 | [TK]D-Fender | WildPikachu: If you only need certain channels, then you shouldn't have specified that group for ALL of them. |
| 14:50.47 | WildPikachu | yep, thanks ... got it |
| 14:51.12 | cybertooth | elixer, Danke. |
| 14:51.26 | elixer | cybertooth: de nada |
| 14:55.32 | *** join/#asterisk ToyMan (n=Stuart@64.241.37.140) |
| 14:56.40 | Zeeek | it's been fun |
| 14:56.43 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:57.07 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 15:03.25 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 15:07.55 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 15:08.49 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:09.23 | Lucky7 | hm |
| 15:09.35 | Lucky7 | if it was an IRQ issue with the wcte12xp |
| 15:09.49 | Lucky7 | a SIP to SIP call (internal) wouldn't have any of those problems correct? |
| 15:10.11 | WildPikachu | hrmmmm, i'm reading on parking a call .... why would one want to park a call btw? |
| 15:11.51 | [TK]D-Fender | WildPikachu: To pick it up from another phone clearly. |
| 15:11.56 | Lucky7 | yea |
| 15:12.24 | WildPikachu | ah, so u park it as 720, then go to another phone and pick it up, using a Pickup()? |
| 15:14.35 | Lucky7 | we're using Softphones |
| 15:14.50 | [TK]D-Fender | WildPikachu: No. Go read the INSTRUCTIONS. |
| 15:14.50 | Lucky7 | Softphones, on a Wired GigaBit Network |
| 15:15.02 | WildPikachu | i am reading |
| 15:15.10 | WildPikachu | [TK]D-Fender, chill a bit man, jee |
| 15:15.38 | [TK]D-Fender | Lucky7: Which softphone? I remember a few CAUSING the lag. |
| 15:15.44 | creativx | [TK]D-Fender is missing the word "chill" in the dictionary WildPikachu. |
| 15:15.51 | WildPikachu | :) |
| 15:15.58 | creativx | instead he has 2 entries for "crazy" |
| 15:16.00 | creativx | :] |
| 15:17.11 | *** join/#asterisk MedozasSVR (n=MedozasS@p549B9617.dip0.t-ipconnect.de) |
| 15:17.49 | Lucky7 | zoiper |
| 15:17.51 | Lucky7 | on IAX |
| 15:18.15 | [TK]D-Fender | Lucky7: Haven't seen it with Zoiper, but give X-lite a test to confirm. |
| 15:18.38 | Lucky7 | hm. |
| 15:18.38 | [TK]D-Fender | Lucky7: I believe I did a while back in an old IDEFisk release |
| 15:18.48 | Lucky7 | yea, its IDEFISK |
| 15:18.52 | Lucky7 | its the "new version" |
| 15:19.59 | [TK]D-Fender | Lucky7: I know, I'm saying that I seem to recall this problem with the older version, but not noticed in the new. |
| 15:20.27 | [TK]D-Fender | Lucky7: test another soft-phone to see if tis a software issue (which I have seen int he past) |
| 15:20.46 | *** join/#asterisk Shido6 (n=shido6@204.126.120.132) |
| 15:20.58 | *** part/#asterisk dg (i=dgl@otherwize.co.uk) |
| 15:21.00 | Vanisher | hm just installed asterisk on centos 5. Setup a sip account.. when i try to call the demo (1000) i get call failed: The person you are calling is unavailable |
| 15:21.18 | Lucky7 | Vanisher : and 1000 is in the dialplan |
| 15:21.33 | Vanisher | Lucky7, yes, installed the sample configs |
| 15:22.06 | Vanisher | Lucky7, exten => 1000,1,Goto(default,s,1) |
| 15:22.27 | jfitzgibbon | Vanisher: and the context that 1000 appears in is the context that you are placing the call from? |
| 15:22.30 | Vanisher | Lucky7, and 500 is also not working: exten => 500,1,Playback(demo-abouttotry); |
| 15:22.31 | [TK]D-Fender | Vanisher: pastbin your configs, and the CLI output of your failed call at verbose 10 and sip debug enabled |
| 15:22.39 | [TK]D-Fender | ~pb |
| 15:22.39 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:22.41 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:24.02 | *** join/#asterisk waltj (n=walt@216.179.31.170) |
| 15:24.10 | Shido6 | any CUT experts |
| 15:24.25 | Vanisher | jfitzgibbon, that was it, the sip account was not in the same context |
| 15:24.27 | Vanisher | brb |
| 15:26.02 | *** join/#asterisk plla (n=nekomimi@200.31.103.86) |
| 15:26.19 | plla | Greetings. |
| 15:27.26 | plla | I would like some help setting up my IAX channel through nat. |
| 15:28.39 | plla | I have my 4569 port open and in the Asterisk box it says the registration message arrived but it doesn't seem to be able to reply it. |
| 15:29.40 | plla | I see several "Tx-Frame Retry" in the console. |
| 15:32.12 | plla | The phone just timeouts after trying to register. |
| 15:32.55 | Shido6 | if im getting <sip:5025155594@myip.com> in a sip header and I want to use "cut" |
| 15:32.56 | Shido6 | http://pastebin.ca/681105 |
| 15:33.57 | *** join/#asterisk toombaloomba (n=hola@89.216.197.140) |
| 15:35.16 | cybertooth | Shido6, what do you want to CUT |
| 15:35.17 | MedozasSVR | hi guys ... i have one asterisk with realtime in backend running nice so far ... but one main question: i would like to add the field subscribecontext to my sip table as column, and im quite unsure if it gets read by asterisk as parameter (sip.conf) ... does anyone have an idea if when adding this field its really gets read by realtime? |
| 15:35.57 | Shido6 | I want to cut everything after the @ and everything before the sip: in the header |
| 15:36.11 | Shido6 | <PROTECTED> |
| 15:36.23 | Shido6 | so everything before the sip: and everything after the @my.ip.com> |
| 15:36.34 | Shido6 | err.... I just want the number |
| 15:37.09 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-829-100.area5.spcsdns.net) |
| 15:37.20 | cybertooth | cut -f2 -d: |cut -f1 -d@ |
| 15:37.24 | Shido6 | so Im doing this Set(Vmail_CID=${SIP_HEADER(FROM):5}) and Set(Vmail_CID=${CUT(Vmail_CID,@,1)}) |
| 15:37.53 | cybertooth | Is that not working? |
| 15:38.01 | Shido6 | which gives me VMTest <sip:5025155593} |
| 15:38.29 | Shido6 | from Greg VMTest <sip:5025155593@my.ip.com> |
| 15:38.51 | [TK]D-Fender | Shido6: You are cutting the QUOTED NAME which is part of the string with your :5 |
| 15:39.02 | [TK]D-Fender | Shido6: not removing through "<sip:" |
| 15:39.03 | Shido6 | yeah its going to be a variable width |
| 15:39.15 | Shido6 | so my 5 doesnt really make sense |
| 15:39.16 | [TK]D-Fender | Shido6: You'll need 2 cuts. |
| 15:39.20 | plla | ${CUT(${CUT(CALLERID(name),:,1)}, @, 0) ? |
| 15:39.38 | [TK]D-Fender | Shido6: cut through ":" on one side, "@" on the other. |
| 15:40.12 | Shido6 | mmm K |
| 15:40.51 | Lucky7 | anyone here who have done a successful Softphone installation of more then 30 phones |
| 15:41.50 | plla | hmm, anyone about the iax question? |
| 15:41.52 | [TK]D-Fender | Lucky7: have you tried the test I suggested? |
| 15:42.01 | Lucky7 | the XLite? yes |
| 15:42.04 | [TK]D-Fender | ~softphone |
| 15:42.05 | jbot | something that should be drug out into the street and shot |
| 15:42.08 | [TK]D-Fender | Lucky7: and? |
| 15:42.13 | Lucky7 | XLite, Express Talk, and Zoiper |
| 15:42.25 | Shido6 | SWEEET |
| 15:42.26 | Shido6 | thanks |
| 15:42.31 | [TK]D-Fender | Lucky7: On Windows? What system spec? |
| 15:42.49 | Lucky7 | p4's 2.2ghz, 512mb ram, XP home |
| 15:43.21 | Lucky7 | or better, some people have intel Core duo, 1gb ram, and XP pro |
| 15:43.21 | plla | It's more than enough for any softphone. |
| 15:43.41 | [TK]D-Fender | Lucky7: all on a local lan to *? |
| 15:43.55 | Lucky7 | yes |
| 15:44.01 | Lucky7 | Gigabit Local LAN |
| 15:45.05 | *** join/#asterisk davixx (n=davixx@82.253.174.106) |
| 15:45.07 | [TK]D-Fender | Lucky7: And do all of the soft-phones lag? |
| 15:45.18 | Lucky7 | no. |
| 15:45.33 | Lucky7 | 1/2 of the softphones work, 1/2 of them lag. |
| 15:45.41 | MedozasSVR | anyone any idea of asterisk realtime? |
| 15:46.14 | [TK]D-Fender | Lucky7: is the anything largely consistant about thoe ones that fail? What codec are you using, etc? |
| 15:46.26 | Lucky7 | all GSM |
| 15:46.41 | [TK]D-Fender | Lucky7: is it always the same ones? (computer / softphone model, etc) |
| 15:46.44 | Lucky7 | and no, some systems that are running the CORE processors, some are runing the 2.2ghz P4's |
| 15:47.00 | Lucky7 | no, it comes an goes |
| 15:47.07 | Lucky7 | sometimes is sounds beautiful |
| 15:47.12 | Lucky7 | and sometimes it lags to crap |
| 15:47.17 | Lucky7 | on any phone. |
| 15:47.30 | Lucky7 | and it does the same thing when its low load, vs higher load. |
| 15:47.32 | *** join/#asterisk rodent|S (n=astrutt@foster.stonedcoder.org) |
| 15:47.32 | [TK]D-Fender | Lucky7: checked your server load? |
| 15:47.43 | plla | Try posgresql Mendoza, res_pgsql.so |
| 15:47.59 | Lucky7 | yea, sometimes i'll get latency, i'll check the load, and i'm the only caller |
| 15:48.13 | Lucky7 | and then i'll get latency, check the load, and there's 5-6 open calls |
| 15:48.16 | plla | Check the documentation of extconfig.conf |
| 15:48.23 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:48.28 | Lucky7 | the "load" never actually changes, its always less then 1% CPU |
| 15:48.49 | Lucky7 | (this system is a Intel Pentium 4 2.8 ghz dualcore, with 2048GB of ram) |
| 15:49.00 | Lucky7 | 2* GM of ram |
| 15:49.04 | MedozasSVR | @plla: well thats actually not what i need - i have already one great realtime running already... im just unsure if asterisk would automatically use new colums i specify, such as "subscribecontext" |
| 15:49.05 | Lucky7 | ..... ugh. 2 GB. |
| 15:49.12 | [TK]D-Fender | Lucky7: Ok, Seems there is little to suggest at this point... |
| 15:49.36 | Lucky7 | Yes. At this point, it'd be migrate to Hardphones |
| 15:50.11 | [TK]D-Fender | ~softphone |
| 15:50.12 | jbot | something that should be drug out into the street and shot |
| 15:50.13 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 15:50.15 | plla | I think it does, I added some columns to it when I required and it worked without complains. |
| 15:50.31 | plla | Like call-limit which isn't in the documentation. |
| 15:50.48 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:50.54 | MedozasSVR | ah i found already ... thx http://www.asterisk.org/doxygen/1.2/AstARA.html |
| 15:53.05 | plla | Lucky7: you are doing something wrong, I have setup Asterisk on a pentium III with 1GHz and 256mb with more than 30 sip clients. |
| 15:53.10 | plla | All p3 with less than 1ghz and 256mb ram |
| 15:53.30 | *** join/#asterisk Vanisher (n=Vanisher@s55916da7.adsl.wanadoo.nl) |
| 15:53.34 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 15:53.36 | cybertooth | Lucky7, what kind of core switches do you have? |
| 15:53.39 | plla | Try using packet sniffers it seems there is a high packet drop somewhere. |
| 15:53.44 | *** join/#asterisk michael-i (n=michael-@W9bc5.w.pppool.de) |
| 15:54.16 | cybertooth | I've seen Dell switches eat VoIP traffic - chew it up and partially swallow it. |
| 15:54.46 | michael-i | Afternoon/morning, everyone. What does one have to reload to apply changes made to rtp.conf? is a complete restart of asterisk required? |
| 15:55.19 | [TK]D-Fender | michael-i: Quite likely |
| 15:56.12 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 15:56.39 | JunK-Y | michael-i: reload will do the job. |
| 15:58.09 | plla | and back to my own dilemma with IAX2, my connections drop. I have the exact same configuration behind a real firewall and not a router and IAX works like a charm. |
| 15:58.20 | plla | what does my firewall have that my router doesn't have? |
| 15:59.06 | michael-i | [TK]D-Fender, that's what I'm thinking... |
| 15:59.09 | plla | qualify=yes doesn't keep the connection open. |
| 15:59.31 | plla | I believe is the udp nat timeout |
| 15:59.53 | michael-i | JunK-Y, do you know what specifically needs to be reloaded? I don't think rtp is handled by a specific module |
| 16:00.12 | plla | But even with a qualifyfreqok less than 30 seconds the connection drops. |
| 16:00.38 | plla | Has anyone experienced the same problem? |
| 16:00.57 | JunK-Y | michael-i: just do restart now then. |
| 16:02.26 | JunK-Y | its driven by the core itself, that might be good to have an rtp reload. |
| 16:02.28 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 16:03.50 | michael-i | JunK-Y, my goal in asking is to reload as little as possible. If I can simply reload a module instead of the whole system I'd be much happier. :) |
| 16:03.51 | dijungal | I have two TE110P cards in an asterisk box, that are experiencing IRQ misses. I believe this is causing bad audio. Both cards are on the same IRQ (4), is there anyway in linux to set them on different IRQ? |
| 16:04.06 | Qwell | dijungal: I would highly recommend calling Digium support |
| 16:04.51 | dijungal | Qwell... how's the programming coming along... |
| 16:04.51 | [TK]D-Fender | dijungal: go check your BIOS as well |
| 16:04.51 | dijungal | lol |
| 16:04.56 | dijungal | i'll consider that |
| 16:05.39 | dijungal | i checked the BIOS and set the PCIs to different IRQs manually, but same thing.. the cards get the same IRQ |
| 16:05.48 | *** part/#asterisk MedozasSVR (n=MedozasS@p549B9617.dip0.t-ipconnect.de) |
| 16:06.01 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 16:06.06 | [TK]D-Fender | dijungal: Tried changing slots? |
| 16:06.20 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 16:06.48 | brodiem | In Ast1.4, how can I tell if a current fax session is being handled with T38? |
| 16:07.27 | dijungal | that's my last resort |
| 16:07.43 | dijungal | but i can only do it after 10 pm |
| 16:07.48 | Qwell | first resort would be calling support |
| 16:07.53 | dijungal | the box is currently being used |
| 16:08.14 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:08.18 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:15.22 | brodiem | anyone? |
| 16:16.06 | brodiem | console indicates ulaw is being used but it's working so well I want to know if t38 is working |
| 16:24.23 | plla | hmm, does anyone know what this warning means? |
| 16:24.28 | plla | WARNING[2139]: chan_iax2.c:8016 socket_process: Received mini frame before first full voice frame |
| 16:24.56 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 16:30.25 | JerJer | plla: means asterisk does not know what codec is being used yet - so its just blindly passing the frame |
| 16:30.49 | JerJer | that message has turned into a debug message in thedevelopment version of asterisk |
| 16:32.34 | plla | I see, it's the consequence of dropping the udp connection. |
| 16:32.36 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:33.21 | plla | Is there a workaround for routers which have a nat udp timeout of 30 seconds for IAX? |
| 16:33.22 | exvito | hi, I'm looking for feedback with the new PCIexpress (TE220 / TE420) cards from Digium... how do they compare to the PCI versions + IRQ sharing behaviour, etc ? better, worse, the same ? in short, if PCIe is a requirement is it safe to go Digium or should one go Sangoma ? |
| 16:34.16 | JerJer | personally i avoid sangoma, but that's me |
| 16:35.55 | duxy786 | hi all, getting th following error, any idea's: kernel: Uhhuh. NMI received for unknown reason 20 on CPU 0. |
| 16:41.53 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 16:46.37 | *** join/#asterisk dasuberdavid (i=david@nat/digium/x-41dad248b54454be) |
| 16:47.26 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:49.44 | *** join/#asterisk Abedegno (n=test@87-194-176-39.bethere.co.uk) |
| 16:51.03 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 16:52.09 | hmmhesays | hello folks |
| 16:52.40 | Abedegno | hi |
| 16:52.56 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:53.19 | Keltus | howdy |
| 16:54.14 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-183.net.novis.pt) |
| 16:56.28 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 16:56.45 | *** part/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 16:57.21 | JerJer | duxy786: have you checked your RAM with memtest86 ? |
| 16:57.38 | JerJer | everytime i've seen NMI messages i've had a bad stick of ram |
| 16:57.45 | JerJer | but YMMV |
| 16:57.59 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 16:59.03 | *** join/#asterisk prudhvi (n=prudhvi@pdpc/student/Prudhvi) |
| 16:59.24 | *** join/#asterisk GlobeTrotter (n=eric@196.40.26.98) |
| 17:00.09 | kiscokid | I need two SIP phones to answer tha same extension. Any easy way to do that? |
| 17:00.23 | GlobeTrotter | hi, i get this error on asterisk 1.4.1.1 ::translate.c:163 framein: no samples for g729tolin |
| 17:01.57 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:02.06 | JerJer | kiscokid: register your phones to openser |
| 17:02.14 | JerJer | using the same info |
| 17:02.19 | JerJer | then send calls from asterisk to openser |
| 17:03.08 | kiscokid | what is openser? |
| 17:03.08 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:03.08 | JerJer | hehehe over your head |
| 17:03.21 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied. -=- Going to AstriCon? Join us in #astricon! |
| 17:03.26 | elixer | ~openser |
| 17:03.27 | jbot | openser is probably an open source GPL project that aims to develop a robust and scalable SIP server. It is spawned from FhG FOKUS SIP Express Router (SER) and it promotes a development strategy open for contributors and contributions. From project's website http://www.voip-info.org/wiki/view/About+OpenSER |
| 17:04.09 | kiscokid | I don't want to install another sip server? |
| 17:04.39 | JerJer | asterisk is not a sip proxy |
| 17:04.44 | JerJer | so good luck |
| 17:05.15 | Abedegno | kiscokid, if it's a particular inbound number you want the SIP phones to answer, you can do that with a ring group |
| 17:05.24 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 17:05.50 | MrMister2 | Hi. I've read the TFOT book but must be doing something wrong. I want to do something very basic, just register a SIP trunk, when a call comes in that trunk just play a message and hangup. anyone willing to help? |
| 17:05.55 | Abedegno | otherwise, like JerJer said you need to install a SIP proxy in front of Asterisk and register the phones with that |
| 17:11.31 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 17:12.14 | *** join/#asterisk MdeP (n=mdep@204-87-22-190.adsl.tie.cl) |
| 17:12.17 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:12.38 | wwalker | anyone had good or bad experience with purchasing from ipphone-warehouse.com? PM me with responses if you like. |
| 17:13.33 | MrMister2 | http://pastebin.ca/681199 - Can anyone take a look and see what's wrong? |
| 17:14.03 | MrMister2 | I receive the call on * but the message doesn't play :( |
| 17:16.19 | MrMister2 | I'm trying not to use freepbx or trixbox and go with pure * but it seems it's a no go :( |
| 17:17.44 | [TK]D-Fender | MrMister2: Looking for 351305501057 in incoming (domain 62.193.231.116) |
| 17:17.52 | [TK]D-Fender | MrMister2: SIP/2.0 404 Not Found |
| 17:18.10 | [TK]D-Fender | MrMister2: sure doesn't look good to me. |
| 17:18.21 | MrMister2 | [TK]D-Fender: mmm... What could cause that? |
| 17:18.26 | GlobeTrotter | anyone know what this means?? translate.c:163 framein: no samples for g729tolin |
| 17:18.47 | [TK]D-Fender | MrMister2: Umm... DUH, you have no exten => 351305501057,1,..... in [incoming] |
| 17:19.35 | generalhan | [TK]D-Fender: i ordered the HWEC for the TE card, is there any setup involved with that? or is it just a plug in, and it works kinda deal ? |
| 17:20.16 | generalhan | i know i will prolly have to (should |
| 17:20.26 | generalhan | ) remove the SWEC lines that i have in my setup, |
| 17:20.39 | hmmhesays | are there any sip related gain settings in asterisk? |
| 17:21.02 | *** join/#asterisk kaigoh (n=kaigoh@82.133.70.150) |
| 17:21.07 | kaigoh | hi there guys |
| 17:21.35 | MrMister2 | [TK]D-Fender: AH! I thought it would receive _all_ calls on the [incoming] on extensions.conf |
| 17:22.15 | kaigoh | can anyone tell me how to get a value out of VM_MSGNUM? I am trying to get a message waiting type thing working |
| 17:22.16 | MrMister2 | [TK]D-Fender: How can I get it to receive all calls on that context independent of trunk? |
| 17:23.44 | elixer | MrMister2: use _X. as your extension |
| 17:24.03 | elixer | MrMister2: exten => _X.,1,.... |
| 17:24.17 | [TK]D-Fender | MrMister2: How many extens are you planning on having land on that context? |
| 17:24.26 | [TK]D-Fender | wildcards like that = ick |
| 17:24.45 | [TK]D-Fender | hmmhesays: nope |
| 17:24.53 | [TK]D-Fender | generalhan: pretty much |
| 17:25.38 | generalhan | [TK]D-Fender: awsome, thanks ! i just didnt know if somewhere in zaptel, or zapata i had to configure the card to use the EC ! |
| 17:26.12 | elixer | it may be icky... but if he has to ask... |
| 17:26.29 | [TK]D-Fender | generalhan: its an option when you do wancfg, and you just need "echocancel=yes" in zapata where you normally put it |
| 17:26.43 | elixer | MrMister2: or you could do _NXXNXXXXXX,1,.... if you wanted to be more concise |
| 17:26.53 | [TK]D-Fender | elixer: Sure, go give him some MORE over-generalized ideas, like he doesn't have enough already! ;) |
| 17:27.09 | elixer | [TK]D-Fender: i'm trying to fix it... gimme a sec |
| 17:27.09 | elixer | :) |
| 17:27.16 | [TK]D-Fender | MrMister2: You should probably HARD-NUMBER themn. |
| 17:27.22 | elixer | ugh |
| 17:27.23 | elixer | heh |
| 17:27.46 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 17:28.15 | elixer | or not. |
| 17:28.17 | MrMister2 | [TK]D-Fender: Well, right now it will be only 1 or 2 but it might grow to a dozen or so. |
| 17:28.43 | MrMister2 | Right now I just want to get the message to test that it _is_ working, after that I'm going to make it run a script |
| 17:29.02 | [TK]D-Fender | MrMister2: ... hard-number them |
| 17:29.30 | kaigoh | can anyone tell me how to get a value out of VM_MSGNUM? I am trying to get a message waiting type thing working |
| 17:29.57 | elixer | MrMister2: or use a pattern match |
| 17:29.59 | elixer | (teehee) |
| 17:31.11 | hmmhesays | no sip related gain stuff eh? |
| 17:31.12 | hmmhesays | thats no good |
| 17:31.15 | MrMister2 | I'm using _XXXXXXXXXXXX right now to make sure it runs :) |
| 17:31.17 | MrMister2 | [Sep 4 18:29:57] WARNING[13445]: pbx.c:1779 pbx_extension_helper: No application '' for extension (incoming, 351305501057, 1) |
| 17:31.17 | MrMister2 | <PROTECTED> |
| 17:31.25 | MrMister2 | oops |
| 17:31.27 | hmmhesays | i'm getting some low volume on my 601 |
| 17:31.29 | hmmhesays | poly 601 |
| 17:32.50 | elixer | MrMister2: you should replace the '...' in the example exten we gave you with an application, e.g. NoOp(Got a call on ${EXTEN}) |
| 17:34.36 | *** join/#asterisk marc7 (n=marc@128.189.199.30) |
| 17:34.41 | [TK]D-Fender | elixer: I think he sorta realizes he should have it do something :p |
| 17:35.21 | elixer | [TK]D-Fender: well just in case... |
| 17:35.58 | MrMister2 | LOL. Yes, I did :) |
| 17:36.06 | MrMister2 | thanks you both anyway. |
| 17:36.25 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:36.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:37.31 | *** join/#asterisk gardo (n=gardo@121.97.242.3) |
| 17:37.51 | MrMister2 | I have the call coming in and I get the "playing welcome" on the console but I don't hear anything :( Any ideas? |
| 17:38.25 | MrMister2 | this is a "vannila" * so I might be missing something? I _do_ have a welcome.gsm file in sounds |
| 17:38.32 | Abedegno | These guys 'www.ic-talk.co.uk' appear to be violating the GPL |
| 17:38.43 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:39.35 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 17:40.14 | *** join/#asterisk t3rror (n=harrison@gateway.sscgp.com) |
| 17:40.19 | MrMister2 | I don't get any errors on the full log, only "logger.c: -- <SIP/351305501057-085a28a0> Playing 'welcome' (language 'en')" so I have no idea why I don't get any sound |
| 17:40.47 | *** join/#asterisk roxy_ (n=roxy_@4.249.97-84.rev.gaoland.net) |
| 17:41.57 | elixer | MrMister2: have you ever heard any sound out of your sip phone? |
| 17:42.02 | [TK]D-Fender | MrMister2: First guess : Your * server behind NAT? |
| 17:43.35 | elixer | Abedegno: how's that? |
| 17:44.25 | Abedegno | elixer: They're selling a modified Asterisk@home system which packages Asterisk and AMP amongst other GPL products |
| 17:44.48 | Abedegno | I sent them an email and they said "The system is based on Asterisk but we've practically rewritten it" |
| 17:44.58 | MrMister2 | [TK]D-Fender: No, it's a physical server with 2 fixed public IPs |
| 17:44.59 | elixer | Abedegno: wow. yeah. |
| 17:45.08 | Abedegno | I've asked for a copy of the source code and they're ignoring me |
| 17:45.29 | Abedegno | look at the screenshots, it's Asterisk@Home/AMP with their branding added |
| 17:45.31 | *** join/#asterisk sacitec (n=tobi@189.149.101.160) |
| 17:45.43 | sacitec | hi |
| 17:45.45 | Abedegno | AND |
| 17:45.55 | [TK]D-Fender | MrMister2: Have you tested the recordings yourself? Have you issued an ANSWER before attempting playback? |
| 17:46.03 | denon | Abedegno: haha, their idea of rewriting it is probably building a new extensions.conf |
| 17:46.22 | Abedegno | They've modified AMP so you can only add 16 extensions, you have to pay them $1000 for a "key" to be able to add 32 |
| 17:46.28 | elixer | Abedegno: i only see the one screenshot and its pretty small. but i'll take your word for it. |
| 17:46.47 | Abedegno | Wish I'd thought of that :-) |
| 17:47.04 | *** join/#asterisk RsaMan (n=aa@196.210.155.2) |
| 17:47.07 | RsaMan | hello guys |
| 17:47.29 | Abedegno | http://www.provu.co.uk/protalk_screenshot2.html |
| 17:47.31 | MrMister2 | [TK]D-Fender: well, I have a Answer, Playback(welcome) and Hangup. Let me do a pastebin, it's only 3 lines but what the hell :) |
| 17:47.33 | Abedegno | Looks like AMP to me :D |
| 17:47.40 | elixer | Abedegno: you should notify Digium. not sure via #asterisk is the best way. |
| 17:47.54 | [TK]D-Fender | Abedegno: Well by all means, tear them a new one... |
| 17:47.54 | RsaMan | what function/sound should i use if a caller is not available , i dont want voicemail |
| 17:47.55 | elixer | Abedegno: wow, ok, yeah. that is pretty obvious. |
| 17:48.08 | Abedegno | hehehe |
| 17:48.10 | elixer | RsaMan: Busy() |
| 17:48.11 | elixer | ? |
| 17:48.27 | [TK]D-Fender | RsaMan: I like "hangup" personally |
| 17:48.53 | [TK]D-Fender | RsaMan: though I usually do "Congestion(5)" first |
| 17:49.06 | elixer | sigh |
| 17:49.27 | RsaMan | thanks |
| 17:49.44 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:50.13 | MrMister2 | [TK]D-Fender: http://pastebin.ca/681233 |
| 17:50.46 | *** join/#asterisk beighto (n=chatzill@12.176.156.130) |
| 17:51.15 | [TK]D-Fender | MrMister2: And you've confirmed this file is ok? its AWEFULLY small..... |
| 17:51.21 | MrMister2 | :) |
| 17:51.41 | [TK]D-Fender | MrMister2: Looks like a "blip" of audio before a hangup on first glance |
| 17:52.09 | [TK]D-Fender | MrMister2: is taht an * std one? maybe try to exand it a bit. |
| 17:52.13 | MrMister2 | I know, just installed *, copied extensions.conf to the side, edited the file, deleted everything with a ; before it |
| 17:52.28 | MrMister2 | [TK]D-Fender: yes, perfectly vannila *, no changes to it |
| 17:52.31 | [TK]D-Fender | MrMister2: pastebin the complete call attempt with SIP debug. |
| 17:54.40 | MrMister2 | http://pastebin.ca/681240 |
| 17:55.02 | roxy_ | does someone knows when ghenry usually logs in ? |
| 17:56.11 | [TK]D-Fender | Retransmitting #1 (NAT) to 82.94.244.100:5060: |
| 17:56.12 | [TK]D-Fender | BYE sip:351305503503@192.168.120.100 SIP/2.0 |
| 17:56.14 | [TK]D-Fender | Via: SIP/2.0/UDP 62.193.231.116:5060;branch=z9hG4bK7ae7b6a5;rport |
| 17:56.15 | [TK]D-Fender | ummm... |
| 17:56.17 | [TK]D-Fender | NAT? |
| 17:56.21 | [TK]D-Fender | Private IP? |
| 17:56.23 | [TK]D-Fender | WTF? |
| 17:57.05 | beighto | I am having a problem with some Polycom IP 430 phones connected to my * server. They reboot randomly during calls. I updated the firmware to the latest release for non-resellers as well as the latest bootrom and the problem only got worse. I changed it back to an older version and now they only reboot for about 1 in every 50 calls. Any ideas? |
| 17:59.39 | MrMister2 | [TK]D-Fender: Sorry? I'm doing the call from XLite that is connected to a * server behind a router, so NAT, yes (My PBX * is 192.168.120.100, 62.193.231.116 is the * server that I'm trying to get working). |
| 17:59.45 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 18:00.16 | MrMister2 | [TK]D-Fender: The * server that I'm trying to get working is 62.193.231.116 so a public IP |
| 18:00.57 | hmmhesays | [TK]D-Fender, are there any gain settings on the config webpage of the poly 601? |
| 18:01.19 | GlobeTrotter | [TK]D-Fender: do you know what this error means:: translate.c:163 framein: no samples for g729tolin |
| 18:01.24 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:01.49 | cybertooth | GlobeTrotter, I think that means that you don't have a g729 codec license. |
| 18:02.18 | [TK]D-Fender | MrMister2: try the most direct tests first. Right now we have 2 servers and a softphone in the way |
| 18:02.24 | cybertooth | It can't do any stats on the translation because it cant do G729 |
| 18:02.25 | GlobeTrotter | i do,, when i do a show g729 it show 40 lincene installed |
| 18:02.27 | [TK]D-Fender | hmmhesays: Extremeley unlikely |
| 18:02.43 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 18:02.43 | [TK]D-Fender | hmmhesays: You you should likely be shot for even hinting at using the web admin for that :p |
| 18:02.48 | GlobeTrotter | and i can send and receive calls using the codec,, but i get this error on incoming calls |
| 18:02.54 | MrMister2 | [TK]D-Fender: K, will try it with a mobile phone and pastebin the sip debug info |
| 18:03.17 | [TK]D-Fender | MrMister2: try this : Wait(5) before your playback, and play it 3-4 time after waiting |
| 18:03.23 | cybertooth | What codec are the incoming calls using (where you get the error)? |
| 18:03.32 | hmmhesays | [TK]D-Fender, i've never actually touched on in my life |
| 18:03.44 | GlobeTrotter | g729 at both ends |
| 18:03.44 | [TK]D-Fender | hmmhesays: What, a Polycom? |
| 18:04.03 | [TK]D-Fender | GlobeTrotter: Pastebin all of your backup |
| 18:04.07 | roxy_ | what is the way to ask a last-seen question to the bot on this channel ? |
| 18:04.11 | mcab | beighto: what version did you upgrade the 430s to? |
| 18:04.24 | mcab | beighto: I know there were a few 430 fixes that went into 2.1.2 |
| 18:04.36 | cybertooth | @seen GlobeTrotter |
| 18:04.39 | cybertooth | nope. |
| 18:04.52 | GlobeTrotter | D-Fender,, what do you mean my backup? |
| 18:04.52 | cybertooth | ~seen GlobeTrotter |
| 18:04.57 | jbot | globetrotter is currently on #asterisk (1h 5m 33s). Has said a total of 7 messages. Is idling for 5s, last said: 'D-Fender,, what do you mean my backup?'. |
| 18:04.59 | [TK]D-Fender | beighto: Go right ahead and upgrade to 2.2.0 |
| 18:05.06 | cybertooth | There. |
| 18:05.07 | *** join/#asterisk crsNeil (n=crsNeil@75.146.5.126) |
| 18:05.28 | beighto | mcab: I think it was either 2.1.1 or 2.1.2 |
| 18:05.32 | [TK]D-Fender | GlobeTrotter: Pastbin everything supporting your problem so we can see where the problem is |
| 18:05.49 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 18:05.50 | roxy_ | cybertooth: thanks |
| 18:05.57 | roxy_ | ~seen ghenry |
| 18:05.58 | jbot | ghenry <n=ghenry@212.159.59.85> was last seen on IRC in channel #asterisk, 39d 3h 41m 51s ago, saying: 'Polycom ip501 a safe bet?'. |
| 18:05.59 | beighto | [TK]D-Fender, mcab: Was this a known issue with them? |
| 18:06.08 | mcab | beighto: I know there were some 430 fixes between 2.1.1 and 2.1.2 - especially if you were seeing "DSP assert" errors in your logs |
| 18:06.11 | crsNeil | Greets all. Can anyone point me to the correct info for using a stream in MOH on 1.4.5 or better? |
| 18:06.18 | [TK]D-Fender | beighto: Not that I know of. Check the changelogs |
| 18:06.35 | hmmhesays | [TK]D-Fender, yeah |
| 18:06.39 | anonymouz666 | file: how it works the func_volume? can I use that on SIP<->SIP? |
| 18:07.04 | *** join/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net) |
| 18:07.45 | MrMister2 | [TK]D-Fender: From a mobile phone: http://pastebin.ca/681250 |
| 18:08.10 | crsNeil | Everything I can find online for streaming in MOH uses mpg123, which steadfastly refuses to install on CentOS. |
| 18:08.16 | roxy_ | anyone could help built and load this module : http://bugs.digium.com/view.php?id=5768 ? do I have to built asterisk or can I include the module only ? |
| 18:09.02 | [TK]D-Fender | MrMister2: Retransmitting #3 (NAT) to 82.94.244.100:5060: <-------- NOT a good sign |
| 18:09.11 | MrMister2 | crsNeil: Doesn't trixbox use mpg123 to play MOH? they use CentOS as the OS |
| 18:09.17 | file | anonymouz666: sure. |
| 18:09.53 | crsNeil | MrMister2: Not using trixbox. Using elastix - nevertheless, can't get mpg123 to install. The compile fails. |
| 18:10.22 | beighto | [TK]D-Fender, mcab: Would either of you care to share the latest firmware and bootrom as I am not a certified reseller yet? |
| 18:10.23 | MrMister2 | [TK]D-Fender: mmmm.... any ideas on what could be happening there? It's a Fedora Core 3 server, very light load and it's a physical server with a 10mb/s connection so no idea on what could be happening |
| 18:10.34 | GlobeTrotter | http://pastebin.com/d2a3f31a5 |
| 18:11.01 | [TK]D-Fender | MrMister2: NAT settings where you tell me you aren't supposed to HAVE any. |
| 18:11.10 | [TK]D-Fender | MrMister2: go clean up your sip.conf |
| 18:11.26 | crsNeil | MrMister2: mpg123 has a whole pile of targets, but the closest they get to CentOS is either linux-i386 or "generic", neither one of which will compile on CentOS 5. |
| 18:11.28 | [TK]D-Fender | beighto: You don't have to be.... just CALL one. |
| 18:11.42 | anonymouz666 | file: that is present in trunk? |
| 18:11.46 | file | anonymouz666: yes. |
| 18:12.01 | anonymouz666 | I need to use that for 1.2, is it possible? |
| 18:12.16 | file | easily? no |
| 18:12.28 | file | it's only code so the answer is yes, you could backport it... |
| 18:13.06 | datachomper | I've got a phone on nat=yes and qualify=yes, asterisk keeps looking for the phone at an old ip address and won't let the phone regregister, i believe. |
| 18:13.21 | datachomper | Can I manually refresh this? I tried reload chan_sip, but nothing ... |
| 18:13.37 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 18:14.06 | beighto | [TK]D-Fender: Call what? |
| 18:14.48 | [TK]D-Fender | beighto: a reseller |
| 18:16.00 | beighto | [TK]D-Fender: Last time I tried that they wouldn't hook me up. I know someone else to try though... thanks. |
| 18:16.08 | crsNeil | Is it possible that streaming just doesn't work with a CentOS box? |
| 18:16.26 | roxy_ | when I have a file: res_config_ldap.c , do I just need to compile into configLdap.ko and load it into the kernel ? or do I need to load it into asterisk ? any pointer to doc would be appreciated . |
| 18:18.35 | sparq | Hey -- Are there any BroadVoice users here? I'm wondering if anyone knows how to get incoming calls to work when you're behind a NAT (BroadVoice doesn't do STUN). |
| 18:20.37 | sparq | They will peer with Asterisk, so the solution seems to be to run Asterisk and stund on my own hardware, but I was wondering if there is a simpler solution before I plunge in. |
| 18:21.24 | _x86_ | eh |
| 18:21.33 | _x86_ | you dont need stun if you register to them ;) |
| 18:22.43 | [TK]D-Fender | sparq: Go read this, now : |
| 18:22.45 | [TK]D-Fender | ~sipnat |
| 18:22.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:22.47 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 18:23.30 | GlobeTrotter | D-Fender,, did you check out my pastebin ? |
| 18:23.36 | sparq | [TK]D-Fender: Thanks -- I figured someone would drop the FAQ on me ^_^ |
| 18:23.42 | roxy_ | ~ldap |
| 18:23.42 | jbot | LDAP is the Lightweight Directory Access Protocol, and is a protocol used to access "Directory Servers". The Directory is a special kind of database that holds information in a tree structure. |
| 18:24.54 | _x86_ | now there's an idea... dynamic extensions stored in LDAP |
| 18:25.03 | _x86_ | has that been done yet? |
| 18:25.14 | roxy_ | _x86_: yes |
| 18:25.39 | roxy_ | _x86_: but I am desesperatly trying to find someone help me install it |
| 18:25.50 | _x86_ | ah |
| 18:26.01 | roxy_ | _x86_: using realtime: http://bugs.digium.com/view.php?id=5768 |
| 18:26.04 | _x86_ | let me know when you get it done -- i'd be interested to know how it works out |
| 18:26.14 | [TK]D-Fender | GlobeTrotter: I am expecting to see a "sip show channels", sip debug, and "show translation" in there as well you know.... |
| 18:26.20 | roxy_ | _x86_: have a look it is supposed to work out |
| 18:26.42 | `Sean | do you guys use any cards with IP phones or switches? |
| 18:26.45 | [TK]D-Fender | file: MUFFINS! I demand..... MUFFINS! |
| 18:26.54 | file | [TK]D-Fender: fresh out |
| 18:27.26 | roxy_ | my kingdom (full of muffins) for help on how to install a module. |
| 18:27.52 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 18:27.59 | GlobeTrotter | ok great |
| 18:28.02 | GlobeTrotter | doing that now |
| 18:28.27 | sparq | [TK]D-Fender: Hmm... According those docs, I should be able to have incoming calls since I'm registered to their SIP proxy (yes?) |
| 18:28.46 | [TK]D-Fender | sparq: No. |
| 18:28.49 | [TK]D-Fender | ~sipregister |
| 18:28.50 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 18:29.14 | sparq | Ah ha. |
| 18:29.33 | duxy786 | hi all, getting th following error, any idea's: kernel: Uhhuh. NMI received for unknown reason 20 on CPU 0. |
| 18:29.38 | [TK]D-Fender | sparq: Roesn't mean they don't force you to AUTH incoming calls, and screwed up NAT settings = DEATH |
| 18:29.51 | [TK]D-Fender | duxy786: Looks like time for Digium support |
| 18:30.36 | duxy786 | how good are they at replying back? |
| 18:31.33 | [TK]D-Fender | duxy786: YMMV. Go call and add to the statistic! |
| 18:32.16 | sparq | [TK]D-Fender: So, setting up my own Asterisk/STUN server outside the NAT and peering it with BroadVoice is likely to be the best way to proceed? |
| 18:33.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:33.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:33.38 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 18:33.53 | Vanisher | pff, itÅ› not that easy :) trying to get a ringing softphone while connecten my asterisk to a sip provider |
| 18:36.01 | GlobeTrotter | http://pastebin.com/d25552dc |
| 18:36.15 | GlobeTrotter | D-Fender,, let me know if you get this |
| 18:41.01 | [TK]D-Fender | sparq: * neither needs nor supports stun. |
| 18:41.18 | [TK]D-Fender | sparq: You are NAT'd TWICE before getting to BV? |
| 18:43.05 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 18:44.31 | wishes | can somebody help me with a problem im having. voicemail, is there any way i can cusomize the messages a user gets without having to use voicemail() for it ? |
| 18:45.32 | putnopvut | wishes: can you elaborate? |
| 18:45.35 | wishes | basicly, we have several status messages, if the users on the phone, they are away from desk, etc. how can i get it to use the personalized messages |
| 18:46.01 | putnopvut | Why don't you want to use voicemail() for that? |
| 18:47.06 | wishes | we do want to use voicemail to take the message, but its at the moment defaulting to using digits from the extension rather than the voice message recorded, thats my main problem |
| 18:47.37 | wishes | so when you call you get 'im sorry the person from extension 815 isnt available ..' rather than 'hi this ix <name> im away from my desk at the moment..' |
| 18:48.47 | wishes | now i realize that you can record messages and they be in /var/spool/asterisk/voicemail/default , but im trying to find a way of saying that the actual recorded message is elsewhere |
| 18:49.03 | wishes | am i making sense? :O |
| 18:50.12 | wishes | i want to change the default place of the voicemail messages i guess |
| 18:50.18 | *** join/#asterisk [hC] (n=hardcore@c-67-183-213-132.hsd1.wa.comcast.net) |
| 18:50.40 | Aeudian | Anyone have any experience with the Pickup2/Steal2 modules? before i install i would like user feedback since its not created by asterisk team |
| 18:50.41 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 18:50.44 | *** join/#asterisk stack_ (n=sgerstac@198.30.100.203) |
| 18:50.48 | [TK]D-Fender | wishes: "show application voicemail" <---------- |
| 18:51.16 | [TK]D-Fender | wishes: And why would the actual recordings be elsewhere? |
| 18:51.34 | [TK]D-Fender | wishes: You could always symlink them if there was really a need, but i can't think of too many needs... |
| 18:53.57 | stack_ | I have a credit card terminal that I'd like to use through our Asterisk system. It works about 30% of the time through an ATA box (CC terminals have sensitive modems). We currently use a GrandStream ATA box. Would a different box work better? What about the IAXy? How about using an FXS module on one of our cards? |
| 18:54.03 | elixer | so what does this karma on bugs.digium.com buy me? can i score a free t-shirt or something? :) |
| 18:54.40 | russellb | elixer: cool points |
| 18:54.58 | russellb | elixer: if you got enough karma, i'm sure we could work out a t-shirt :) |
| 18:55.15 | [TK]D-Fender | elixer: My karma ran over your dogma. |
| 18:55.36 | [TK]D-Fender | stack_: Get it its own line that * doesn't get within 10' of. |
| 18:55.37 | elixer | russellb: ok, i'll add color support to menuselect now. ;-) |
| 18:55.37 | russellb | there are plenty of people that i would mail a shirt to if they asked because they have contributed so muc |
| 18:55.40 | elixer | kidding. |
| 18:55.46 | russellb | ha, go for it |
| 18:56.00 | elixer | was thinking of a libnewt version, actually. |
| 18:56.05 | elixer | because i'm a massochist |
| 18:56.14 | russellb | elixer: lol, i looked into doing that briefly |
| 18:56.23 | elixer | s/masso/maso/ |
| 18:56.25 | [TK]D-Fender | elixer: Yes, and while you're at it, can you make my BSOD's a "corn-flower" blue instead of "royal"? |
| 18:56.27 | russellb | elixer: it would certainly look cool ... if you can figure out libnewt |
| 18:56.33 | stack_ | [TK]D-Fender: yeah, we were kind of thinking that, but since the fax machine was working fine through *, we thought we might be able to make a terminal go through |
| 18:56.47 | [TK]D-Fender | stack_: key word : TERMINAL |
| 18:57.25 | russellb | elixer: join us in #asterisk-dev if you have any questions |
| 18:57.27 | elixer | russellb: alrighty. i'm a pretty smart guy, just ask [TK]D-Fender. |
| 18:57.31 | elixer | russellb: will do |
| 18:57.52 | Shido6 | http://pastebin.ca/681310 |
| 18:57.55 | russellb | elixer: did you see my gtk frontend? it's pretty ugly. |
| 18:58.13 | [TK]D-Fender | Shido6: Didn't we just go through this? |
| 18:58.21 | elixer | russellb: i saw that existed, but i didn't fire it up. |
| 18:58.37 | russellb | gotcha .. |
| 18:58.41 | Shido6 | I thought so but I cant seem to get asterisk to understand my thoughts |
| 18:58.52 | Shido6 | I keep feeding it pb&j |
| 18:58.59 | Shido6 | but it keeps getting slower |
| 18:59.40 | elixer | russellb: my only linux access is CLI, so never needed to/been able to. |
| 19:01.15 | wishes | [TK]D-Fender: basicly the boss wants each user to have their own personalized 'on the phone' 'away from desk' 'away sick' 'outside business hours' |
| 19:01.47 | [TK]D-Fender | wishes: Then do a playback before entering Voicemail. |
| 19:02.00 | wishes | the previous employee set it up the long hard way using database and status, when in fact only one needs to be set |
| 19:02.11 | [TK]D-Fender | Shido6: its the PB... its making everything sticky |
| 19:02.18 | wishes | so now you can manually set your self to be 'on the phone' etc which is kinda silly |
| 19:02.29 | wishes | mm i tried that but it doesnt appear to be working |
| 19:02.52 | [TK]D-Fender | wishes: Of course you can put a Playback before you call Voicemail. And yes it works. |
| 19:03.54 | wishes | yeah i know you can do it, and that it apparently works, but what im saying is that its not working and its not really giving me any error why |
| 19:04.13 | wishes | mostly the lack of error is the frustrating part :) |
| 19:04.16 | wishes | but ill keep at it |
| 19:06.50 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
| 19:06.51 | [TK]D-Fender | wishes: Well you haven't shown us what you're doing, and while we ARE normally psychic on tuesdays, there is a week-shift due to Labor Day. |
| 19:06.59 | [TK]D-Fender | wishes: PASTEBIN is your friend. |
| 19:07.01 | [TK]D-Fender | ~pb |
| 19:07.01 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:07.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:07.21 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 19:08.21 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-59-195.pskn.east.verizon.net) |
| 19:10.15 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 19:10.23 | wishes | [TK]D-Fender: well its wednesday here today :D |
| 19:10.29 | wishes | i think ive figured it out |
| 19:10.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:12.24 | wishes | hmm how do i get the variable username in the config? |
| 19:12.42 | Shido6 | http://pastebin.ca/681330 |
| 19:13.55 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.25.53) |
| 19:15.46 | [TK]D-Fender | Shido6: esten => _711XXXXXXX,9,Set(Vmail_CID=${CUT(Vmail_CID,,:,1)}) |
| 19:15.53 | wishes | mm nm figured that out |
| 19:15.59 | [TK]D-Fender | Shido6: "S"?! |
| 19:16.09 | Shido6 | ? |
| 19:16.11 | [TK]D-Fender | Shido6: and CUT only takes *3* parameters |
| 19:16.19 | [TK]D-Fender | Shido6: "eSten" |
| 19:16.24 | Shido6 | LOL |
| 19:16.28 | _x86_ | hahahaha |
| 19:16.29 | Shido6 | spanish? |
| 19:16.30 | _x86_ | nub |
| 19:16.31 | Shido6 | :) |
| 19:16.32 | Shido6 | ok |
| 19:16.36 | _x86_ | :p |
| 19:16.57 | [TK]D-Fender | Shido6: fix 10 while you're at it, and the REST :p |
| 19:18.07 | Shido6 | exten => _711XXXXXXX,9,Set(Vmail_CID=${CUT(Vmail_CID,:,1)}) |
| 19:18.19 | Shido6 | exten => _711XXXXXXX,10,Set(Vmail_CID=${CUT(Vmail_CID,@,1)}) |
| 19:18.58 | [TK]D-Fender | Shido6: Fix all the bugs, then confirm you have APPLIED them, then pastebin the full CLI output, not jsut 1 line so we can see how you fail to progress. |
| 19:19.00 | Shido6 | gives me Vmail_CID= VMTest <sip:5025155593 |
| 19:19.42 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 19:20.54 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579178.dsl.bell.ca) |
| 19:21.32 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 19:23.12 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579178.dsl.bell.ca) |
| 19:23.13 | _x86_ | [TK]D-Fender: fail to progress... haha |
| 19:23.20 | _x86_ | classic |
| 19:23.23 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) [NETSPLIT VICTIM] |
| 19:23.54 | _x86_ | Shido6: just so you know, we're laughing at you, not with you... |
| 19:23.58 | _x86_ | oh wait, the other way around |
| 19:24.11 | [TK]D-Fender | :D |
| 19:24.29 | _x86_ | :p |
| 19:24.35 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:25.40 | Shido6 | I dont care, if u help me I'll drop you a few paypal digits |
| 19:26.13 | wishes | like 0.01 digits? |
| 19:26.20 | _x86_ | rofl |
| 19:26.21 | wishes | hehe :D |
| 19:26.41 | Shido6 | http://pastebin.ca/681345 |
| 19:26.41 | wishes | is there any alternative to ${DIALEDPEERNAME} since that doesnt work |
| 19:26.42 | _x86_ | Shido6: [TK]D-Fender has been helping you |
| 19:26.56 | _x86_ | Shido6: so you already owe him money, then ;) |
| 19:27.16 | wishes | i have ${ARG2} which is SIP/username , but i want just the username |
| 19:27.48 | JerJer | anyone happen to know how one would add the MySQL libraries to Asterisk BE (which is rpath linux) ?? |
| 19:27.54 | _x86_ | ${ARG2:4} |
| 19:28.07 | wishes | ohhh i never knew you could do that :D |
| 19:28.21 | _x86_ | in-place manipulation ;) |
| 19:28.26 | _x86_ | substrings, anyway |
| 19:28.40 | [TK]D-Fender | Shido6: how about NoOping the FROM you're mangling BEFORE you start? :) |
| 19:28.58 | wishes | mm doesnt work :/ |
| 19:29.15 | JerJer | _x86_: have you ever parsed out a variable into more than one variable? like split on a comma |
| 19:30.15 | wishes | oh yes baby |
| 19:30.17 | wishes | now its working |
| 19:30.35 | _x86_ | wishes: im not responsible for typos ;) |
| 19:30.38 | wishes | you fucking own :D |
| 19:30.48 | _x86_ | pay me! :P |
| 19:31.10 | wishes | 0.01c? |
| 19:31.13 | _x86_ | see, that's where the praise always stops ;) |
| 19:31.17 | wishes | do you take western union? |
| 19:31.24 | wishes | from nigeria .. |
| 19:31.48 | [TK]D-Fender | wishes: he gave you his .02c worth and all you do is pay him HALF?! ;) |
| 19:31.57 | wishes | io have this friend whos the friend of some diplomat who needs to urgently transfer like lots of billions offshotre before 15th April 07 .. |
| 19:32.58 | _x86_ | [TK]D-Fender: that's right! :P |
| 19:33.10 | _x86_ | wishes: i'll take it! |
| 19:33.14 | wishes | haha |
| 19:33.17 | _x86_ | ;) |
| 19:33.22 | _x86_ | 419 baby! |
| 19:33.23 | generalhan | Anyone ever have an issue with a Cisco 7960, where the phone would just reboot itself for no reason? i cant get to the bottom of this issue. |
| 19:33.26 | wishes | hows that? |
| 19:33.41 | wishes | generalhan: power supply? overheating? sounds hardware like |
| 19:33.48 | _x86_ | wishes: hehe, accepted :P |
| 19:33.53 | _x86_ | well accepted at that :P |
| 19:34.08 | generalhan | wishes: chaqnged the brick 3 times. even changed the phone itself for this user a few times ... same issue |
| 19:34.29 | _x86_ | generalhan: cisco phones are generally very crappy, and some (especially like the 7912) can be very unstable with a SIP firmware loaded on them |
| 19:34.35 | wishes | changed the power cable? plug? |
| 19:34.42 | generalhan | then i thought it was maybe the firmware, but this is only happening on 2 extensions of 15 |
| 19:34.50 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 19:34.54 | _x86_ | generalhan: power spikes? |
| 19:34.56 | wishes | i dislike anything that involves several k to do a course to understand something that should be basic |
| 19:35.00 | _x86_ | generalhan: use PoE |
| 19:35.06 | wishes | ergo, Cisco can go out the door |
| 19:35.18 | generalhan | _x86_: not an option ATM |
| 19:35.33 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.208) |
| 19:37.27 | wishes | score! i can make wengaphone shit itself on demand :D |
| 19:37.39 | _x86_ | wishes: cisco should stick to networking... polycom owns the phone market |
| 19:38.02 | _x86_ | polycom++ |
| 19:38.49 | wishes | arg |
| 19:39.54 | *** join/#asterisk TokyoMoD (n=mod@softbank060081070010.bbtec.net) |
| 19:41.31 | wishes | ok, can i set up answerphones for users with softphones who are not logged in. |
| 19:41.45 | wishes | atm im getting "Sep 5 07:38:50 NOTICE[9686]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 19:42.11 | wishes | because the user isnt signed in, idealy what i want is that it takes a message for that user until they sign in |
| 19:43.19 | TokyoMoD | have you added Voicemail function in the routine? |
| 19:43.38 | TokyoMoD | I have users who aren't logged in and it goes to voicemail directly. |
| 19:43.54 | _x86_ | TokyoMoD: you're doing voicemail with queues? *gasp* |
| 19:44.01 | _x86_ | that's usually a very bad thing |
| 19:44.29 | TokyoMoD | yeah. It's only 5 users. |
| 19:44.30 | _x86_ | well, unless you have an AGI script to see if anyone is in the queue at all, otherwise direct the call to voicemail... |
| 19:44.49 | _x86_ | otherwise, you'll always have voicemail picking up the call |
| 19:46.52 | *** join/#asterisk pat2man (n=ptescher@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 19:50.04 | _x86_ | is there a way to get asterisk to log status changes on a zaptel interface? |
| 19:50.27 | _x86_ | i've got a PSTN T1 that keeps losing sync |
| 19:53.24 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 19:58.14 | wishes | welp, i think ive cracked it |
| 19:58.41 | *** part/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579178.dsl.bell.ca) |
| 19:59.10 | wishes | time for breakfast |
| 19:59.23 | wishes | hopefully i havnt fucked up all the phones forever too badley :) |
| 20:01.04 | _x86_ | :P |
| 20:03.28 | chemikk | hm here is 10:03 PM |
| 20:03.54 | wishes | so now we have groups (cust, sales, dev, etc) and each person has their status messages going, and can set them for some silly reason to fake not being here, and users can listen to crap comedy skits whilst on hold |
| 20:04.12 | wishes | and users can press numerous buttons and go round and round the mulberry bush |
| 20:04.24 | wishes | chemikk: its 8:03am here |
| 20:04.26 | _x86_ | lol |
| 20:04.30 | _x86_ | Tue Sep 4 15:04:30 CDT 2007 |
| 20:04.40 | _x86_ | wishes: where are you? |
| 20:04.41 | wishes | Wed Sep 5 08:04:21 NZST 2007 |
| 20:04.45 | wishes | NZ :) |
| 20:04.51 | _x86_ | ah |
| 20:04.53 | wishes | speaking of which, thats the next thing i have to fix |
| 20:05.00 | wishes | silly daylight savings has changed this year |
| 20:05.01 | _x86_ | here mulberries are on trees, not bushes :) |
| 20:05.12 | wishes | gotta fix all the servers before that happens |
| 20:07.25 | wishes | oh power surge |
| 20:07.29 | wishes | that means time to make coffee :D |
| 20:07.44 | wishes | thanks for all the help, most indebted to yall :) |
| 20:08.06 | wishes | mulberries bush was in reference to the childs poem/song thing "here we go round the mulberry bush.." |
| 20:08.31 | Shido6 | thank you |
| 20:08.35 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 20:12.41 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3f652d2e044a2530) |
| 20:12.55 | bkruse | how was the weekend everyone? |
| 20:14.02 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 20:14.02 | _x86_ | great here thanks |
| 20:14.04 | Qwell | meh |
| 20:14.05 | coldsteal | hello |
| 20:14.12 | Qwell | bkruse: it sucked :p |
| 20:14.31 | coldsteal | is there a way to call a # and have it hangup after xamount of sec? |
| 20:14.43 | Qwell | coldsteal: check out the L() option to Dial |
| 20:16.03 | bkruse | Qwell: same, i was in ohio all week, ugh |
| 20:16.13 | Qwell | eh, I had no internet |
| 20:16.17 | Qwell | no internet trumps Ohio |
| 20:17.11 | De_Mon | I sat around and played on my DS all weekend, that and my cousin had her baby girl saturday so we went and visited with them a little. |
| 20:18.22 | *** join/#asterisk askarel (n=frederic@88.147.8.72) |
| 20:19.17 | De_Mon | I'm looking for a web-meetme developers email address... |
| 20:21.13 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 20:21.18 | *** join/#asterisk iBuMp (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:21.30 | *** join/#asterisk askarel (n=frederic@2001:6f8:374:0:202:6fff:fe34:96b2) |
| 20:21.38 | *** join/#asterisk ReDNeQ- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:21.47 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:22.18 | *** join/#asterisk iBuMp- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:25.52 | *** join/#asterisk the_Goat (n=chatzill@h-67-103-23-130.phlapafg.covad.net) |
| 20:26.30 | the_Goat | are there any docs out there that can tell me how to tie together the cdr fields. i want to write some custom reports in cf, and not sure how everything fits together |
| 20:27.09 | *** join/#asterisk pabelanger (n=pb@bas4-ottawa23-1088827152.dsl.bell.ca) |
| 20:28.18 | codefreeze | the_Goat: you mean in a single row? What the fields mean? Or, across multiple rows? |
| 20:28.57 | askarel | hello |
| 20:29.12 | the_Goat | across multiple rows. i want to see the session from the beginning from the call till the hangup |
| 20:29.32 | codefreeze | What version of asterisk, the_Goat? |
| 20:29.41 | the_Goat | 1.4.something |
| 20:30.36 | codefreeze | OK. Well, to be honest, at the current moment, there may be no way to link the pieces of what could have been a single session. |
| 20:31.07 | pabelanger | anybody know the differences between --prefix and DESTDIR when building asterisk? |
| 20:31.19 | the_Goat | darn, i wanted to see what happens when a call comes in,if it gets transferred, parked, etc. |
| 20:31.22 | Qwell | pabelanger: DESTDIR is used for temporary installation paths. |
| 20:31.48 | bkruse | Qwell: no internet!? |
| 20:31.55 | Qwell | so, for example, in the chroot example.. You don't want /mnt/blah to be stored anywhere. All of the files in /mnt/blah can easily be moved to /, and it would work fine |
| 20:32.07 | *** join/#asterisk LukinoVoip (i=LukinoVo@151.82.2.161) |
| 20:32.08 | askarel | I'm a newbie with Asterisk and i don't really know where to begin... I try to find a good tutorial to get started... Anybody with a good link ?? |
| 20:32.21 | Qwell | whereas with --prefix (and the others, like --sysconfdir), if you want to install to /usr/local/, you *do* want things to search in /usr/local/etc/asterisk |
| 20:32.22 | pabelanger | Qwell: That is what I thought... |
| 20:32.39 | Qwell | does that answer your question? |
| 20:32.45 | codefreeze | the_Goat: you CAN make out some of that from the CDR's.... maybe enven most; but all, I think not. |
| 20:33.36 | pabelanger | A little, still have a slight problem tho |
| 20:35.22 | the_Goat | i am looking at the cdr data and it looks like the dstchannel and channel fields reference each other sometimes |
| 20:37.06 | coldsteal | does anyone here use voipjet? |
| 20:38.39 | the_Goat | well hopefully someday they will be able to tag a call from start to finish. i see a uniqueid field that has something to the effect of xxxxxxxxxx.xx digits in it, and the first group before the . is the same sometimes, but not very often |
| 20:42.18 | *** join/#asterisk pabelanger_ (n=pb@bas4-ottawa23-1088827152.dsl.bell.ca) |
| 20:42.43 | pabelanger_ | sorry about this, internet access at coffee shop expired, and my other session has not expired. |
| 20:44.50 | pabelanger_ | without spamming the channel, could somebody send me the last few lines of the channel, since my last question (if it even made it in). |
| 20:44.55 | Qwell | <pabelanger> A little, still have a slight problem tho |
| 20:45.14 | pabelanger_ | ok, thanks... |
| 20:47.13 | pabelanger_ | I'm testing asterisk, non-root, using /tmp/asterisk directory. Since I set DESTDIR=/tmp/asterisk all files get installed here. And because asterisk.conf is build dynamicly, it currently does not get updated with my new paths. Is this where the --prefix option would be used too? |
| 20:47.23 | Qwell | yes |
| 20:47.28 | Qwell | if you want to run it from /tmp/asterisk |
| 20:48.05 | *** join/#asterisk ReDNeQ- (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:48.26 | *** join/#asterisk chemikk (i=abap@real.wilbury.sk) |
| 20:49.37 | *** join/#asterisk iBuMp (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:49.44 | pabelanger_ | great, thanks... testing now |
| 20:50.07 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-f36ddaf2d35bd6d5) |
| 20:52.11 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:55.24 | *** join/#asterisk kkn088 (n=kikoun@84.4.216.243) |
| 21:01.12 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 21:02.10 | Lucky7 | Real quick |
| 21:02.10 | Lucky7 | http://rafb.net/p/MuW41a18.html |
| 21:02.10 | Lucky7 | That means the T1 card is got its own IRQ, Correct? |
| 21:02.17 | Qwell | Lucky7: looks like it |
| 21:02.58 | [TK]D-Fender | Lucky7, pastebin "dmesg" |
| 21:06.42 | pabelanger | Hmm, must be screwing something up, can't get this to work |
| 21:07.25 | fujin | damn, forgot to do my upgrade to * in the middle of the night. |
| 21:08.48 | *** join/#asterisk Xarion (n=xarion@c1-34-6.rndf.isadsl.co.za) |
| 21:09.03 | Xarion | OMG i am going to put a hammer thru this spa3000 |
| 21:09.13 | Xarion | can someone help me before i do so :P |
| 21:11.56 | Lucky7 | op |
| 21:12.01 | Lucky7 | sorry, meeting, lemme ssh back in |
| 21:12.46 | Lucky7 | holy crap |
| 21:12.52 | *** join/#asterisk amilcar_ (n=amilcar@201.34.202.17) |
| 21:12.56 | [TK]D-Fender | Xarion, www.voxilla.com <- go check out the forums |
| 21:13.13 | Xarion | sheesh yeah i've been thru like 5 howtos |
| 21:13.18 | Xarion | and i still can't come right |
| 21:13.20 | Xarion | argh |
| 21:13.27 | Xarion | I'm beginning to doubt myself! |
| 21:14.10 | Lucky7 | gimme a sec to paste crap into nopaste |
| 21:14.19 | Xarion | =) |
| 21:15.37 | Lucky7 | http://rafb.net/p/Ukord534.html |
| 21:17.02 | Lucky7 | http://rafb.net/p/hT78L146.html |
| 21:17.10 | Lucky7 | thats the full, i might have missed a few lines |
| 21:19.58 | [TK]D-Fender | Lucky7, http://lists.digium.com/pipermail/asterisk-dev/2004-March/003448.html |
| 21:20.47 | [TK]D-Fender | Lucky7, in case you are still having issues. Also check your chipset against the Digium compatability lists |
| 21:21.17 | Lucky7 | we're not having crashing at all |
| 21:21.22 | Qwell | Lucky7: what is your issue? |
| 21:21.35 | Lucky7 | little bit of latency on the line |
| 21:21.52 | Qwell | how much latency are we talking about? |
| 21:21.55 | Lucky7 | IE, if i do an "echo test" where i ask the person on the other end, to repeat the number to me when they hear it |
| 21:22.00 | Lucky7 | i count to 10 |
| 21:22.17 | Lucky7 | sometimes its fine, and i hear 1, as i get to 2, which is about normal for cellphones |
| 21:22.31 | Lucky7 | but sometimes its not untill i get to 5-6 before i hear 1 |
| 21:22.41 | Lucky7 | which is super perplexing |
| 21:22.53 | Qwell | you have a human on the other end repeating the digits back to you? |
| 21:23.06 | Lucky7 | yes, so there is a little bit of extra time |
| 21:23.18 | Lucky7 | the human-lag time |
| 21:23.32 | Qwell | and it's all going over the PSTN? |
| 21:23.55 | Lucky7 | yes. |
| 21:24.02 | Lucky7 | we have no VOIP or anything like that |
| 21:24.08 | Lucky7 | its strictly a T1 box |
| 21:24.13 | Qwell | have you discussed with your carrier? |
| 21:24.27 | Qwell | that doesn't sound like a hardware issue to me |
| 21:24.39 | Lucky7 | thats what I like to hear. |
| 21:24.48 | Lucky7 | The two big issues we have right now |
| 21:24.50 | Lucky7 | are softphones |
| 21:25.01 | Lucky7 | ( Polycom IP301's being overnighted ) |
| 21:25.04 | Qwell | wait, this is happening when you call in from a softphone? |
| 21:25.12 | Lucky7 | yea |
| 21:25.15 | Lucky7 | we're fixing that now |
| 21:25.17 | Qwell | well, there's your problem |
| 21:25.18 | [TK]D-Fender | Lucky7, softphones usually add a littel extra delay of their own which can make EC a pain |
| 21:25.25 | [TK]D-Fender | Lucky7, 301? EW |
| 21:25.37 | [TK]D-Fender | Lucky7, wish I'd have seen you before that order |
| 21:25.41 | Lucky7 | ? |
| 21:25.44 | [TK]D-Fender | Lucky7, 301 = dead-end |
| 21:25.48 | Lucky7 | ? |
| 21:25.53 | [TK]D-Fender | Lucky7, IP 320 = cheaper & better |
| 21:26.17 | Lucky7 | and minus the switch |
| 21:26.22 | Qwell | 330 has a switch |
| 21:26.26 | [TK]D-Fender | Lucky7, 330 then |
| 21:28.07 | *** join/#asterisk dasenjo (n=dasenjo@190.5.197.254) |
| 21:28.13 | dasenjo | Hi! |
| 21:30.24 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:30.52 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.193) |
| 21:31.41 | dasenjo | I have an old cdr, with integer codes in the disposition field. I know that 1 is for not answered, 2 is for failed, 3 is for busy and 4 is for Answered. But right now I've found a entrie with an 8. ¿Can someone help me and sayme what does it mean? |
| 21:32.31 | *** part/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:35.05 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:35.50 | codefreeze | dasenjo: 1.4? |
| 21:36.10 | dasenjo | Maybe 1.0 even |
| 21:36.32 | dasenjo | no, it is sure 1.2 |
| 21:37.35 | fujin | howdy |
| 21:37.40 | fujin | anyone know if asterisk has g729a support? |
| 21:37.54 | Qwell | fujin: It can. |
| 21:37.56 | codefreeze | dasenjo: see include/asterisk/cdr.h; AST_CDR_*, (*== FAILED, BUSY, NOANSWER, ANSWERED) |
| 21:38.09 | Qwell | You can do passthrough just fine, but in order to transcode to/from, you'll need to buy licenses from Digium |
| 21:38.55 | fujin | even for g729a? |
| 21:38.59 | fujin | not g729 |
| 21:39.05 | fujin | I thought 'a' was the free one |
| 21:39.07 | russellb | no .. |
| 21:39.09 | Qwell | there is no free one |
| 21:39.16 | dasenjo | codefreeze, opening it .. |
| 21:39.20 | Qwell | but, g729a is what asterisk can support |
| 21:39.26 | Qwell | it can't do b, iirc |
| 21:40.07 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 21:40.11 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:40.24 | codefreeze | dasenjo: 1=FAILED; 2=BUSY; 4=NOANSWER; 8=ANSWERED |
| 21:40.29 | *** join/#asterisk RoyK (n=roy@ti211310a080-1578.bb.online.no) |
| 21:40.46 | codefreeze | I guess previously, it never said a call was answered |
| 21:40.55 | fujin | ah, righto |
| 21:40.56 | fujin | ta |
| 21:41.02 | RoyK | localtime(); |
| 21:41.21 | fujin | getting weird crackle on alaw, even phone->voicemail, dunno what I can do to diagnose it |
| 21:41.28 | fujin | i've transcoded all the voicemails tuff to alaw |
| 21:41.28 | dasenjo | codefreeze, thanks a lot! I was comparing with the half of the number .. |
| 21:41.31 | fujin | not seeing any loss on the links |
| 21:44.02 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 21:47.27 | dasenjo | bye everybody! |
| 21:47.36 | dasenjo | thanks again codefreeze |
| 21:50.50 | fujin | anyone know if I can set up DSCP for sip/rtp on Asterisk? |
| 21:50.57 | fujin | or should i do it with like outgoing mangle rules in iptables |
| 21:55.01 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 21:56.30 | *** join/#asterisk AJayMN (n=contact@h460c0cce.area2.spcsdns.net) |
| 21:58.52 | *** part/#asterisk RoyK (n=roy@ti211310a080-1578.bb.online.no) |
| 22:06.54 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 22:13.30 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 22:14.47 | *** join/#asterisk bob198125 (n=chatzill@216.230.150.13) |
| 22:18.46 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:20.11 | *** join/#asterisk gmfm (n=hithere@216.161.142.20) |
| 22:26.14 | *** join/#asterisk Nichtwirklich (n=guess@88.134.54.113) |
| 22:26.19 | Nichtwirklich | hello all |
| 22:27.17 | gmfm | anyone know how to use distinctive ringing for inbound PSTN calls on spa3k boxes? |
| 22:27.20 | Nichtwirklich | can anybody recommend an us-american sip provider? it must provide a normal phone number, reachable from a classic phone, outgoing calls are not necessary |
| 22:27.33 | JT | us-american? |
| 22:28.20 | Nichtwirklich | america is bigger than the usa, I think :) |
| 22:28.51 | JT | that is the weirdest term ever, us-american |
| 22:29.36 | Nichtwirklich | okay, but I dont want a canadian number |
| 22:29.48 | Nichtwirklich | so far this would be an american number too ;) |
| 22:31.10 | generalhan | anyone in here ever used an Aastra 480i with * ? |
| 22:32.12 | Nichtwirklich | no sip provider in the whole us? btw a landline number in new england would be prefered |
| 22:32.39 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 22:32.59 | Sweeper | Nichtwirklich: les.net |
| 22:33.03 | Abedegno | Nichtwirklich, use Gradwell and get a landline in old england :-) |
| 22:33.17 | Sweeper | technically, they're in canada, but they have plenty of US DID's |
| 22:33.33 | Nichtwirklich | Abedegno: this won't help, I have german number |
| 22:33.54 | Sweeper | also, they say they don't do CC's via paypal, but you can do it that way if you don't mind getting charged the paypal fee |
| 22:34.18 | Nichtwirklich | the problem with american customers is, they cannot beleave that anybody outside North America has electricity at all or is doing any business |
| 22:34.20 | gmfm | Nichtwirklich: nufone.net and broadvoice provide USA DIDs |
| 22:34.36 | Abedegno | Nichtwirklich, how unfortunate for you :-) |
| 22:34.39 | Abedegno | Schwarzwälder Kirschtorte |
| 22:34.57 | Sweeper | da! spreken ze deusch? |
| 22:35.14 | Abedegno | non, je parle francais bien |
| 22:35.16 | Sweeper | je'mappel teddy! |
| 22:35.30 | Sweeper | moi maison et petite! |
| 22:35.58 | Nichtwirklich | thanks folks, will check your links |
| 22:36.02 | Abedegno | vous couchez avec les poisson |
| 22:38.19 | *** join/#asterisk philippel (n=philippe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 22:39.07 | Strom_M | gardez-vous les rouges pour la fin? |
| 22:39.40 | Strom_M | je compose mauvais numero avec votre lait homo |
| 22:39.46 | JT | Nichtwirklich: when people say "american" they mean the usa |
| 22:40.16 | Strom_M | je ne me souviens sexe du chat |
| 22:40.21 | Strom_M | fin |
| 22:40.49 | Nichtwirklich | JT: usually here, but if you want to make sure it's a us whatever, than we say us or us-american, anyway, you got the point |
| 22:40.53 | Nichtwirklich | too |
| 22:43.53 | Sweeper | when people say "amerikkan" they mean democrats |
| 22:44.30 | *** join/#asterisk Techie-Micheal (n=Techie-M@phpbb/leader/Techie-Micheal) |
| 22:44.39 | Sweeper | gardez-vous les rouges pour la fin? <-- you save the reds for last? |
| 22:44.49 | Techie-Micheal | How can I adjust the bitrate on the zap channel for conference calls? |
| 22:45.15 | Strom_M | Sweeper: yes |
| 22:45.16 | Sweeper | Techie-Micheal: uh, don't you ahve to do 8khz? |
| 22:45.31 | Strom_M | Sweeper: Canadian packet of Smarties |
| 22:45.56 | Sweeper | I mean, call me silly, but I think the pstn might object if you started sending them an 4khz signal... |
| 22:45.59 | Sweeper | Strom_M: ahhhhh |
| 22:46.09 | Techie-Micheal | Sweeper: No pstn :) |
| 22:46.37 | Sweeper | Techie-Micheal: ok, so s/pstn/pots hardware/ |
| 22:46.47 | Qwell | Techie-Micheal: what, precisely, are you trying to accomplish? |
| 22:47.39 | Techie-Micheal | Qwell: Improve sound quality. |
| 22:47.53 | Techie-Micheal | Sweeper: Not POTS/PSTN was harmed in the making of this conference call. :P |
| 22:48.08 | Sweeper | Techie-Micheal: then wth are you doing with a Zap channel? |
| 22:48.39 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 22:48.57 | Techie-Micheal | I can do conference calls without Zap? Everything I had read said I needed ztdummy and a zap channel for conferencing. |
| 22:48.58 | Nichtwirklich | I dont think you can adjust the sample rate there |
| 22:49.54 | wishes | ok, another q. if background($user/$user_on_phone) doesnt exist, can i have it do the generic "$EXTEN is busy" ? |
| 22:50.53 | fujin | background will go to the next priority |
| 22:50.58 | fujin | after playback or not |
| 22:51.11 | fujin | Busy(); takes care of what you need |
| 22:51.45 | wishes | mm |
| 22:52.16 | fujin | are you building custom voicemail? |
| 22:52.27 | Sweeper | Techie-Micheal: ah. well, if you're doing really high numbers of people in a conf (such that the zap channel's processing is causing trouble), you might want to move the conf calls to freeswitch or something |
| 22:52.41 | wishes | yeah |
| 22:53.05 | Sweeper | if it's a bandwidth problem, you should be adjusting the SIP/IAX bitrates |
| 22:53.37 | wishes | i have a macro defined for busy-options, where they hear $user say 'im on the phone atm, press 1 to leave a message or 2 to be transfered to somebody else in my team' kinda thing |
| 22:53.38 | Techie-Micheal | Okay, how do I do that? :P |
| 22:53.39 | *** join/#asterisk umanghc (n=umanghc@ool-182fface.dyn.optonline.net) |
| 22:54.05 | wishes | so say for customer service, if its non urgent they can leave a message, if they *must* talk to somebody then they will get put through to somebody else in the group |
| 22:54.30 | wishes | many groups defined (sales,cust,dev,noc,marketing etc) |
| 22:54.41 | Sweeper | Techie-Micheal: change audio codecs ;) |
| 22:54.50 | Sweeper | g.729 for preference |
| 22:54.55 | Techie-Micheal | And how do I do that? :P |
| 22:55.04 | wishes | its all working, just having hiccups if somebody hasnt recorded their voices :) |
| 22:55.06 | Sweeper | set the allowed codecs in sip.conf and iax.conf |
| 22:55.18 | *** join/#asterisk linagee_ (n=linagee@about/linux/staff/linagee) |
| 22:55.33 | fujin | wishes, show application Background |
| 22:55.39 | fujin | should tell you what happens when it plays back |
| 22:55.40 | fujin | or doesn't |
| 22:57.29 | wishes | fujin: i can see what the problem is, its how do i get around the problem :/ |
| 22:58.02 | fujin | not being very specific as to what the problem is ;) |
| 22:58.10 | fujin | background should skip to the next priority, doesn't matter if it plays back or not |
| 22:58.20 | wishes | problem is that there are the odd person who is new that doesnt have a recorded message, and it just goes dead because it cant play that message |
| 22:58.39 | wishes | i need it to play $user/$user_message or play a default one if that one doesnt exist |
| 22:59.00 | wishes | currently it plays $user/$user_message , and if that doesnt exist it sits quietly |
| 22:59.15 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 22:59.15 | fujin | shouldn't sit quietly, there's no reason for it to do that unless you've got a waitexten in there. |
| 22:59.39 | fujin | I'm not sure if there is a level of abstraction (variable/function) which will allow you to check if a file exists. |
| 22:59.46 | wishes | mmm |
| 22:59.47 | fujin | you may want to write a module to take care of it |
| 22:59.53 | fujin | app_wishes_voicemail.c ;][ |
| 22:59.58 | wishes | haha |
| 23:00.24 | fujin | err |
| 23:00.25 | fujin | cancel that |
| 23:00.28 | wishes | app_i_0wn3d_j00_all.c ? |
| 23:00.33 | fujin | ${STAT()} will do what you want |
| 23:00.39 | wishes | ohhh, nice |
| 23:01.16 | wishes | app_fujin_0wn3d_j00_all.c |
| 23:01.38 | fujin | Noop(${STAT(e,/path/to/file)}); |
| 23:01.55 | wishes | just googling it now :D |
| 23:02.02 | fujin | show function STAT :P |
| 23:02.23 | fujin | how's nzlinux? |
| 23:02.30 | fujin | <djfu |
| 23:02.50 | wishes | No function by that name registered. |
| 23:02.59 | fujin | lol |
| 23:03.01 | fujin | show version? |
| 23:03.17 | fujin | you're running 1.2, aren't you -_- |
| 23:03.21 | wishes | probably |
| 23:03.31 | fujin | well, that's not going to work. |
| 23:03.59 | wishes | im dealing with a previous admins bullocks |
| 23:04.00 | fujin | I guess you could use System() |
| 23:04.10 | wishes | Connected to Asterisk 1.2.18 |
| 23:04.13 | wishes | hah |
| 23:04.46 | wishes | time to upgrade? |
| 23:04.58 | fujin | Noop(System("stat poo")); |
| 23:05.07 | Techie-Micheal | wishes: Just slightly. :P |
| 23:05.12 | fujin | yeah, 1.4 will give you ${STAT} |
| 23:05.22 | fujin | and numerous other awesome things |
| 23:05.24 | wishes | but will it break anything - config wise? |
| 23:05.25 | fujin | like AEL |
| 23:05.29 | fujin | probably :) |
| 23:05.48 | wishes | :/ |
| 23:05.53 | fujin | dev box |
| 23:05.54 | wishes | i cant afford to break shit |
| 23:06.37 | fujin | I didn't have any issues going from 1.2 to 1.4 |
| 23:06.42 | fujin | but then again, I rebuilt my entire dialplan in AEL |
| 23:06.54 | wishes | yeah but how many users are on your system? |
| 23:07.10 | fujin | 50, Maxnet's entire callcentre. |
| 23:07.11 | wishes | and if it breaks will the company go bust ? :) |
| 23:07.22 | fujin | no, because I have development platforms, dual asterisk boxes |
| 23:07.29 | wishes | oh i wish |
| 23:07.38 | wishes | im just lucky to not have a xen server pabx box |
| 23:07.48 | fujin | vmware |
| 23:07.48 | codefreeze | ... and murf will fix any problems reported with AEL as quick as possible. |
| 23:07.54 | wishes | oh no, i lie, i think it might be a xen server |
| 23:07.58 | Qwell | codefreeze: cool guy, that murf |
| 23:08.01 | wishes | wtf you doing at maxnet anyway |
| 23:08.12 | wishes | bloody bunch of christians :D |
| 23:08.17 | Sweeper | hey now ;) |
| 23:08.20 | codefreeze | Qwell: feel free to elaborate and expound!! ;) |
| 23:08.33 | *** join/#asterisk saftsack (n=saftsack@p57A764C1.dip.t-dialin.net) |
| 23:09.36 | wishes | not that theres anything wrong with christians ... :) |
| 23:10.32 | fujin | wishes, I haven't really been affected by the apparent christianity |
| 23:10.50 | fujin | I'm a systems engineer here |
| 23:10.59 | wishes | i moved departments |
| 23:11.08 | fujin | oh? |
| 23:11.19 | wishes | i was doing systems/network years ago, but got sick of it so went developer, did that here for 4 years |
| 23:11.28 | wishes | now ive just moved back to systems/network since the last guy quit |
| 23:11.33 | wishes | so im network manager now |
| 23:12.05 | wishes | i got sick of dealing with customer service and bugs - 4 years of development will get you ready with a chainsaw :D |
| 23:12.26 | wishes | but the last network guy we had was ok, just had a weird way of doing stuff i think |
| 23:12.52 | fujin | lol |
| 23:12.59 | fujin | i'm in a junior position |
| 23:13.04 | fujin | overskilled and underpaid unfortunately. |
| 23:13.08 | fujin | overworked, too |
| 23:13.15 | fujin | I probably shouldn't have volunteered my skeelz |
| 23:13.19 | wishes | lol |
| 23:13.35 | wishes | yeah underpaid is right |
| 23:13.49 | wishes | ive been underpaid for years |
| 23:13.58 | wishes | but its worth it if the company is a good company |
| 23:14.18 | wishes | i think if you work for pricks and assholes then its time to get asshole tolerance pay or move on |
| 23:14.45 | wishes | the hubby is now looking for another job - his asshole tolerance pay isnt enough any more i dont think :D |
| 23:17.16 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 23:17.21 | n0n4m3 | evenin/mornin |
| 23:17.21 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:17.46 | *** join/#asterisk elixer (n=seanbrig@c-68-55-114-113.hsd1.md.comcast.net) |
| 23:17.55 | n0n4m3 | Asterisk 1.4.4 |
| 23:18.08 | n0n4m3 | what is the diff with 1.4.11? |
| 23:18.15 | n0n4m3 | is sip any better? |
| 23:21.33 | fujin | check the changelog ya muppet |
| 23:24.37 | *** join/#asterisk Kwakwa (n=kwa@spc2-ward2-0-0-cust610.bagu.broadband.ntl.com) |
| 23:26.12 | Kwakwa | Is it possible to play a wav without the cdr recognising the call as being answered? `Queue(queue-test,rn,,,20) / Playback(asterisk-recording) / Queue(queue-test,,,,)` sets the call as being answered when I don't want it to. |
| 23:26.37 | Kwakwa | Its not a periodic announce I'm after because I want it to queue for 20 seconds, play the wav, then queue with hold music. |
| 23:26.41 | Qwell | Playback(file|noanswer) |
| 23:26.53 | Kwakwa | ahh, thanks a lot |
| 23:27.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:29.12 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585253.dsl.bell.ca) |
| 23:39.06 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 23:40.14 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:43.12 | n0n4m3 | gah! i can't seem to convince asterisk to actually REGISTER with a remote SIP server... it only sends the OPTIONS command and no REGISTER :S |
| 23:43.55 | russellb | n0n4m3: did you add a register => line in sip.conf ? |
| 23:44.45 | n0n4m3 | ofcourse |
| 23:46.43 | ltdwk | Can someone tell me, what happens when you dial multiple SIP channels, and one is not registered/unavailable? |
| 23:47.11 | n0n4m3 | [Sep 5 01:46:48] NOTICE[19516]: chan_sip.c:12331 handle_response_peerpoke: Peer 'detel-outgoing' is now Reachable. (17ms / 2000ms) |
| 23:49.45 | n0n4m3 | could i somehow 'force' asterisk to send just REGISTER and no OPTIONS? |
| 23:58.30 | n0n4m3 | the server in question doesn't seem to return REGISTER as a valid option :S |
| 23:59.17 | GlobeTrotter | hello,, im getting this error on my box for incoming calls.. chan_sip.c:3625 sip_write: Asked to transmit frame type 256, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8) |
| 23:59.38 | GlobeTrotter | can someone tell me what that means? |