| 00:00.08 | jblack | I used to have a boss that would answer yes to very either/or question. I _despised him_ |
| 00:00.49 | [TK]D-Fender | jblack: thing is that wasn't an either or, and the options not mutually exclusive |
| 00:01.21 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 00:01.35 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:02.20 | jblack | [TK]D-Fender: No worries. I was reminiscing. I'm not pointing the finger of guilt at you. |
| 00:03.21 | jblack | Even though I'm legally obligated to blame you for everything due to your country of origin. |
| 00:04.25 | [TK]D-Fender | jblack: Remember 1812 ;) |
| 00:04.59 | jblack | can't say that i can remember it. I wasn't around back then. |
| 00:06.59 | jblack | Huh. So canada took... detroit? |
| 00:07.27 | [TK]D-Fender | jblack: Kicked your lilly asses :p |
| 00:08.08 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 00:09.32 | jblack | Out of all the places that could have been conquered... you guys went after Detroit? -DETROIT-?? |
| 00:10.07 | jblack | plays with anachronism |
| 00:11.53 | suma | How to disconnect the caller in the middle of the call and connect to another new call? |
| 00:12.19 | jblack | That sounds like a call transfer |
| 00:12.48 | suma | yes the right term :) thanks jblack |
| 00:13.36 | jblack | Polycom phones, there is a transfer button. Most soft clients have a transfer button. If you have an analog phone, try flash - new number - flash |
| 00:13.36 | k-man | currently i run the debian packages of asterisk - what would be the procedure for switching to running from compiled source? |
| 00:14.02 | [TK]D-Fender | k-man: Remove packages. Download source, compile, install |
| 00:14.03 | jblack | k-man: Sure you want to do that? |
| 00:14.15 | k-man | jblack: no, not sure |
| 00:14.38 | jblack | k-man: If you don't need to deviate from your distro's offerings, don't do it. |
| 00:14.40 | k-man | [TK]D-Fender: you think that the config files and sound files and stuff would all still be in the right place? |
| 00:14.50 | suma | jblack: with a keypress from the caller or callee, it need to transfer the call |
| 00:15.02 | k-man | jblack: i'd quite like to try 1.6 out and theres no packages of 1.6 yet |
| 00:15.03 | [TK]D-Fender | k-man: probably |
| 00:15.24 | suma | jblack: using asterisk |
| 00:15.30 | jblack | suma: Look at features. the config file is features.conf, but you'll want to google for good instructions on enabling features. And don't let [TK]D-Fender find out you're doing it. |
| 00:15.56 | k-man | jblack: yeah, i should probably just be a little more patient |
| 00:16.08 | [TK]D-Fender | jblack: I might like an actual description of the PARTICIPANTS of this call. |
| 00:16.32 | suma | jblack: thanks |
| 00:17.26 | docid | just found out our T1 comes off the oldest operation tel switch in the world.....this might be why im haveing problems, they said the 'carbon blocks' are getting old...umm...woah... |
| 00:17.58 | k-man | cool! |
| 00:18.06 | k-man | hehe - what do they use the carbon blocks for? |
| 00:19.01 | *** join/#asterisk rift0r (i=rift@420nugs.info) |
| 00:19.17 | docid | im not sure really...... its long before my time |
| 00:19.31 | docid | getting a moden t1 card to work with it is umm..challengeing |
| 00:19.39 | docid | modern |
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| 00:31.52 | Merlin | what does switchvox use for their fax solution? 3rd party software or something open source? anyone besides tk-dfender know? |
| 00:33.31 | rift0r | i have 2 questions.... what would cause a delay in voice to be broadcast on the remote line? I have a sip phone routed to my pbx which connects to my sip provider... i have rtp opened on the pbx and specified in the rtp.conf and my pings are sub 100 but when i call someone and they answer i dont always here the first hi |
| 00:34.23 | rift0r | 2nd question is sometimes very rarely, as soon as I answer a call it will hang up immediately... seems like it happens when call id is blocked and calls show up as anonymous |
| 00:34.37 | rift0r | what would cause that |
| 00:34.59 | k-man | is it possible to stream windows media or real media into asterisk somehow? |
| 00:35.05 | [TK]D-Fender | Merlin: Uses SpanDSP which is 3rd party OSS. Just like EVERYTHING ELSE. |
| 00:35.10 | [TK]D-Fender | Merlin: BOTH |
| 00:36.02 | [TK]D-Fender | k-man: soft-phone using the stream as a recording source. |
| 00:36.34 | Merlin | [tk]d-fender: i've heard that iaxmodem is a much better solution |
| 00:37.10 | k-man | [TK]D-Fender: so what I'd like to do is set up an extension that I can dial when I want to listen to an internet radio station |
| 00:37.15 | [TK]D-Fender | Supposedly more reliable, but more complex to configure. Different solutions for different needs |
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| 00:40.33 | Merlin | [tk]d-fender: gotcha |
| 00:41.49 | [TK]D-Fender | Merlin: You'd use Hylafax if you needed the rest of what it offers, like queued office outbound faxing, etc. For silly little inbound stuff, SpanDSP + rxFax usually works fine. |
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| 00:48.26 | [TK]D-Fender | BBIAB |
| 00:48.57 | ThoMe | hello |
| 00:49.00 | ThoMe | my log said: |
| 00:49.01 | ThoMe | [Mar 12 01:48:41] WARNING[21395]: chan_sip.c:12913 handle_response: Forbidden - maybe wrong password on authentication for NOTIFY |
| 00:49.04 | ThoMe | [Mar 12 01:48:41] WARNING[21395]: chan_sip.c:12412 handle_response_invite: Received response: "Forbidden" from '"01786323765" <sip:01786323765@10.0.10.1>;tag=as542143c2' |
| 00:49.05 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 00:49.14 | ThoMe | i have a snom+vpn. can connect to teh server, works good |
| 00:49.34 | ThoMe | but when i try with my cell phone call to my asterisk and redirect to my sip then i have a forbidden. why? |
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| 00:52.41 | nemik | how can i see the things being outputted with ast_debug(1,....)? |
| 00:53.03 | nemik | from console i mean. set sip debug and set core debug didn't do it |
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| 01:15.40 | ThoMe | hello |
| 01:15.41 | ThoMe | [Mar 12 02:15:30] WARNING[21395]: chan_sip.c:12412 handle_response_invite: Received response: "Forbidden" from '"01786323765" <sip:01786323765@10.0.10.1>;tag=as251698df' |
| 01:15.45 | ThoMe | can i disable that? |
| 01:16.54 | uluatu | is this possible to use monitor instead of mixmonitor when recording agent calls? |
| 01:17.40 | uluatu | i need do decrease the cpu cycles used by each calls that comes to my call center. |
| 01:17.55 | uluatu | is this a good aproach? |
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| 01:35.11 | Chotaire | dudes, I present you a (probably stupid) question... where do I find ztxen? is it part of the distribution? |
| 01:36.02 | Chotaire | I've been trying the zaptel-source that comes with hardy (ubuntu) and had the compiled module running for a while (unused). for whatever reason the box crashed and took dom0 with it. |
| 01:36.27 | Chotaire | the oopsie log doesn't really give me much clue, so I assume that ztdummy is responsible. |
| 01:37.29 | harry_v | no such ztxen that I have ever heard of |
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| 01:37.59 | Chotaire | google says different. |
| 01:38.39 | harry_v | been using asterisk for years never heard of it. |
| 01:38.52 | harry_v | Let see what goo says |
| 01:40.19 | harry_v | something I have not used. |
| 01:40.53 | Chotaire | how would it be possible to run meetme (with sip only) without the use of ztdummy? is there any solution you guys are aware of? |
| 01:41.09 | ThoMe | when i try call in then i have an error: |
| 01:41.10 | ThoMe | [Mar 12 02:40:56] WARNI |
| 01:41.13 | ThoMe | [Mar 12 02:40:56] WARNING[21395]: chan_sip.c:12412 handle_response_invite: Received response: "Forbidden" from '"01786323765" <sip:01786323765@10.0.10.1>;tag=as172de3cd' |
| 01:41.26 | Chotaire | obviously ztdummy has a flaw... it's not really possible to crash a dom0 from within userspace, so that's why I believe ztdummy has something to do with it, being the only module loaded on domU |
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| 01:46.57 | sin | is it impossible to do something like TEMP_VAR=${VAR}; in ael? |
| 01:47.14 | sin | ael2 |
| 01:50.20 | harry_v | I dont see why not |
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| 01:50.50 | sin | it never works for me. it just comes up as 0 |
| 01:51.18 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-78-138.nycmny.east.verizon.net) |
| 01:51.25 | Steve_J-obs | hello guys!!! |
| 01:54.07 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 01:54.09 | Steve_J-obs | guys, someone requested me an application for 10 thousand concurrent calls, and I have a question about how do you think the dids handle that? |
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| 01:55.23 | bougyman | Steve_J-obs: you aren't going to do 10k concurrent calls with media. |
| 01:55.25 | bougyman | The End. |
| 01:55.43 | Steve_J-obs | with media? |
| 01:56.15 | Steve_J-obs | this is pure voice |
| 02:00.16 | Steve_J-obs | your answer doesn't make sense |
| 02:01.37 | chigambamukoko | Greetings to all in the name of the creator |
| 02:01.47 | chigambamukoko | Yo master Mog, are you home? |
| 02:01.57 | Mog | ya |
| 02:01.58 | Steve_J-obs | allelluyah! |
| 02:02.02 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:05.37 | lanning | Steve_J-obs: 10,000 CONCURRENT calls is over half a gigabit of voice data. |
| 02:06.39 | lanning | That's over 434 PRI circuits (of course, you wouldn't use PRI at that load) |
| 02:06.50 | Steve_J-obs | lanning: yes sir, they say 1 DS3 handles 640 calls, so I calculate that it wold take over 10 DS3s! |
| 02:07.21 | Steve_J-obs | it is for an application that will run in a stadium...really cool |
| 02:08.20 | lanning | I am guessing that there will not be 10,000 phones... |
| 02:09.06 | Steve_J-obs | yes they will, in a stadium with 60,000 people, I would expect 10,000 of them with cell phones |
| 02:09.52 | Steve_J-obs | my question is this: will anyone think this can theoritically run with ONE DID??... meaning, suppose it takes 10 providers with one DS3 each... |
| 02:10.21 | lanning | no you get one provider with 10 DS3s |
| 02:11.02 | Steve_J-obs | ...and then, after all the channels of the first DS3 gets filled, when it gets to the last channel, can we make it point to the next DS3 in another ip, and so on? |
| 02:11.29 | apeiron | 10k concurrent? wow. |
| 02:11.34 | lanning | the provider can do that. |
| 02:12.48 | Steve_J-obs | lanning: I am asking this question because I've never seen it done, and I dont want to look like I dont know what I am talking about when they ask me |
| 02:13.45 | lanning | you can get 10,000 calls to the premises, that isn't really that hard (carrier does all the work). |
| 02:13.51 | Steve_J-obs | question is: can we make the incoming calls jump from one ds3 to the next, as they get filled |
| 02:14.13 | lanning | the hard part is routing it IN YOUR network to balance the load across the app servers. |
| 02:14.37 | lanning | you don't do that, you request the carrier to do that. |
| 02:14.59 | Steve_J-obs | the reason I would want to use several carriers is to save money... in real life one carrier will charge a whole month worth of service for that |
| 02:15.23 | lanning | different concurrent carriers means different phone numbers |
| 02:15.43 | lanning | period |
| 02:16.39 | Steve_J-obs | well... my idea is that I can program the asterisk servers to handle the incoming calls in such a way that when channels are full, they start sending the calls to the next DS3, everything on the same DID |
| 02:16.54 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 02:17.01 | Steve_J-obs | remember, all this calls are for the same DID |
| 02:17.02 | lanning | that's only if you are on the OTHER side of the DS3s |
| 02:17.26 | Steve_J-obs | yes, I have the servers |
| 02:17.33 | lanning | wrong! |
| 02:17.39 | lanning | the OTHER side |
| 02:17.44 | lanning | origination side |
| 02:18.25 | lanning | once the call hits your server, you are now dealing with that call, period. no rerouting. |
| 02:18.31 | Steve_J-obs | I see |
| 02:19.40 | lanning | if there are no channels left in the DS3, the carrier must route the call to the next DS3. This is all BEFORE you see the call. |
| 02:20.00 | Steve_J-obs | let me ask you a dumb question: are there a lot of carriers that can provide you with 10 DS3s capacity, or can that only be lthe level-3s and XOs?? |
| 02:20.34 | Steve_J-obs | I mean 10 DS3s for a one day event |
| 02:21.19 | lanning | Voice or data? |
| 02:21.28 | Steve_J-obs | voice 100% |
| 02:21.33 | lanning | k |
| 02:22.07 | lanning | most any big carrier could do it. XO and the likes would be grabbing AT&T |
| 02:23.05 | bougyman | Steve_J-obs: i'd be looking towards a discount provider like Tel-West for that. |
| 02:23.13 | lanning | just remember that there will be BIG $$$$ for 10 DS3s for only a day. And this must be planed about 2 years in advanced. |
| 02:23.27 | bougyman | they've got a lot of dark fiber they can turn up for such uses. |
| 02:23.27 | pdmmm | 10 DS3s?! |
| 02:23.42 | pdmmm | its not the fiber tho |
| 02:23.46 | pdmmm | its the thing to light it up |
| 02:23.48 | bougyman | Steve_J-obs: it shouldn'e be more than 3k per DS3. |
| 02:23.49 | pdmmm | that shit aint cheap |
| 02:24.04 | bougyman | but you'd need at least 45 days notice for turnup |
| 02:24.10 | bougyman | more likely 60 days. |
| 02:24.46 | Steve_J-obs | thats another issue, I am calcualting that if they charge one hal cent per minute, the 10 thousand calls wiill add up to $500 |
| 02:24.59 | ThoMe | is it posible set the caller id from the remote call? |
| 02:24.59 | bougyman | that's really expensive. |
| 02:25.07 | ThoMe | examole: call id from remtoe: 089123456 |
| 02:25.09 | bougyman | i'm getting .1c/minute currently. |
| 02:25.14 | ThoMe | i would like: 0089123456 |
| 02:25.16 | ThoMe | is it posible? |
| 02:25.22 | bougyman | but that's voip, you're talking TDM, right? |
| 02:25.30 | bougyman | our TDM is 1.8c/minute. |
| 02:25.34 | Steve_J-obs | I am talking about voip |
| 02:25.36 | bougyman | and we do about 1M minutes/month. |
| 02:26.01 | bougyman | oh, VOIP you can just hook up with iCall/Checkbox/etc. to get originations under 1/2 cent per minute. |
| 02:26.04 | Steve_J-obs | you pay 1.8 per min for 1M/ a month?? |
| 02:26.05 | ThoMe | can i set this: ${CALLERID(num)} |
| 02:26.08 | ThoMe | to this: 0${CALLERID(num)} ? |
| 02:26.10 | ThoMe | when yes, how? |
| 02:26.14 | bougyman | those are both < .2c/minute |
| 02:26.33 | bougyman | Steve_J-obs: on TDM, yes. |
| 02:26.34 | ThoMe | can you help me? please? |
| 02:26.37 | ThoMe | bougyman: ? |
| 02:26.42 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:26.43 | pdmmm | when i worked @ a local isp/clec the cheapest we got was .004 |
| 02:27.05 | bougyman | .004c? |
| 02:27.11 | bougyman | that's ridiculous cheap. |
| 02:27.11 | pdmmm | ya |
| 02:27.22 | bougyman | that's cheaper than a CNAM lookup. |
| 02:27.23 | Steve_J-obs | yes, you can find at .004 if it is local |
| 02:27.31 | pdmmm | but thats also filing as clec |
| 02:27.34 | bougyman | oh, i do TDM local so it's unlimited. |
| 02:27.47 | pdmmm | wholesale t1's were about $90 |
| 02:27.52 | pdmmm | per month |
| 02:27.55 | bougyman | crap |
| 02:27.59 | pdmmm | but jesus, at&t treated you like *shit* |
| 02:28.02 | bougyman | i pay $430/per |
| 02:28.05 | pdmmm | jesus |
| 02:28.07 | bougyman | Qwest rates. |
| 02:28.12 | pdmmm | even XO will give better rate than that |
| 02:28.19 | bougyman | no, XO was my former provider. |
| 02:28.30 | bougyman | it was 210 local loop + 260 access. |
| 02:28.34 | pdmmm | my friend has a provider - i got a t1 in my apartment that he gives me free |
| 02:28.35 | bougyman | 470 > 430 |
| 02:28.39 | ThoMe | exten=> 284141405,n,Set(CALLERID(num)=${CALLERID(num)}) |
| 02:28.41 | ThoMe | is it correct? |
| 02:29.04 | ThoMe | yes, |
| 02:29.08 | NovceGuru | test it |
| 02:29.33 | ThoMe | NovceGuru: works :) |
| 02:29.38 | NovceGuru | :D |
| 02:29.40 | ThoMe | exten=> 284141405,n,Set(CALLERID(num)=0${CALLERID(num)}) |
| 02:29.42 | ThoMe | :-) |
| 02:30.03 | pdmmm | bougyman: why u leave XO? |
| 02:30.28 | bougyman | pdmmm: because they sucked the big one. |
| 02:30.38 | bougyman | a single T1 DIA was $1700/month. |
| 02:30.51 | bougyman | i'm getting DS3 DIAs for $3500/month through tel-west. |
| 02:30.53 | pdmmm | dude |
| 02:31.17 | pdmmm | thats terrible |
| 02:31.22 | bougyman | and XO techs were clueless. |
| 02:31.22 | pdmmm | :) |
| 02:31.38 | bougyman | what's terrible? |
| 02:32.08 | pdmmm | the T1 DIA :) |
| 02:32.12 | *** join/#asterisk orbi (n=orbi@c-69-137-124-219.hsd1.tn.comcast.net) |
| 02:32.17 | bougyman | yeah, i know. |
| 02:32.22 | orbi | pokes Corydon76-dig and runs |
| 02:32.23 | *** part/#asterisk orbi (n=orbi@c-69-137-124-219.hsd1.tn.comcast.net) |
| 02:32.31 | pdmmm | DIA T1-1.5Mbps |
| 02:32.31 | pdmmm | 1 |
| 02:32.31 | pdmmm | 1 YR |
| 02:32.31 | pdmmm | $120.00 |
| 02:32.32 | pdmmm | $ 0.00 |
| 02:32.34 | pdmmm | $ 0.00 |
| 02:32.36 | pdmmm | DIA Network Access |
| 02:32.38 | pdmmm | 1 |
| 02:32.40 | pdmmm | 1 YR |
| 02:32.42 | pdmmm | $121.00 |
| 02:32.44 | pdmmm | $121.00 |
| 02:32.46 | pdmmm | $ 250.00 |
| 02:32.48 | pdmmm | $ 0.00 |
| 02:32.50 | pdmmm | thats my T1 in my apartment ;) |
| 02:32.52 | pdmmm | from XO |
| 02:33.02 | bougyman | in DFW? |
| 02:33.06 | bougyman | that's about right. |
| 02:33.07 | pdmmm | Chicago |
| 02:33.22 | bougyman | residential service is always a ton cheaper than commercial. |
| 02:33.26 | VJFROMGT | hi guys, trying to figure out what file to put r in for fake ringback |
| 02:33.42 | bougyman | my senior engineer gets 50/20 FIOS for like $50/month across the street. |
| 02:33.45 | pdmmm | its no different |
| 02:33.56 | pdmmm | wholesale T1 from XO |
| 02:34.01 | bougyman | verizon wants $2,800/month for 20/20 fibre from us. |
| 02:34.12 | pdmmm | and the ilec still does the last mile |
| 02:34.14 | bougyman | same goddamned backbone. |
| 02:36.55 | pdmmm | where are you @? |
| 02:38.52 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 02:41.46 | lmadsen | VJFROMGT: which file? uhhh... none... that's a Dial() option |
| 02:42.24 | VJFROMGT | i mean, what is the format to write dialplan? |
| 02:43.26 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 02:47.15 | bougyman | pdmmm: DFW |
| 02:47.20 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 02:48.58 | blitzrage | VJFROMGT: you have some serious amounts of documentation to read then |
| 02:49.07 | VJFROMGT | i know |
| 02:49.09 | NovceGuru | bougyman: we are paying $600/mo for 5x5 fiber from TW |
| 02:49.17 | blitzrage | at least chapter 5-6 of The Future of Telephony |
| 02:49.19 | NovceGuru | not enogh lube in the world :( |
| 02:49.22 | blitzrage | !book |
| 02:49.24 | VJFROMGT | i have dialplan up and running but dont know how to add ringback feature |
| 02:49.29 | blitzrage | ~book |
| 02:49.30 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:49.30 | bougyman | that's about the same, novce |
| 02:49.49 | uluatu | guys im wodering if it is possible to only Monitor calls to agents and not MixMonitor them. |
| 02:49.55 | NovceGuru | bougyman: it's not terrible, other then being TW |
| 02:50.02 | blitzrage | VJFROMGT: 1) you probably don't need it, 2) read the application appendix to see the format for Dial() |
| 02:50.02 | NovceGuru | and sla excludes power outages |
| 02:50.20 | blitzrage | uluatu: just use.... Monitor()? |
| 02:50.22 | uluatu | i need to optimize the load of my server... |
| 02:50.36 | blitzrage | otherwise, re-state your question |
| 02:50.46 | bougyman | 5/5@600 !> 20/20@ $2800, really. |
| 02:50.58 | bougyman | i guess it's $400 better. |
| 02:51.06 | NovceGuru | < |
| 02:51.07 | NovceGuru | yeah |
| 02:51.07 | uluatu | blitzrage: but when configuring agents to be recorded in agents,conf I cant. Is this possible to change this behaviour through variables? |
| 02:51.20 | *** kick/#asterisk [docelmo!n=twisted@router.asteriasgi.com] by twisted (Niq flood (3 nicks in 24secs of 30secs)) |
| 02:51.38 | *** join/#asterisk docelmo (n=vircuser@pool-70-110-112-204.lyn.east.verizon.net) |
| 02:51.46 | NovceGuru | is it normal for fiber to exclude power outages from the SLA? |
| 02:52.21 | NovceGuru | I asked how long I have on their end when there's a wide area power outage and they said "until the batteries in our nodes die" |
| 02:52.22 | docelmo | yes.. Verizon told me to get a generator cause the telcom battery setup was only good for 8 hours |
| 02:52.32 | uluatu | blitzrage: What I want is to improve the performance to handle as many recorded calls I can |
| 02:52.48 | NovceGuru | my facility is battery/generator backed |
| 02:52.55 | NovceGuru | tier 4 specs |
| 02:53.01 | NovceGuru | self proclaimed :P |
| 02:53.03 | docelmo | Most CO's are |
| 02:53.17 | blitzrage | uluatu: so don't use the recording in agent.conf -- just call Monitor() before calling the agent |
| 02:53.20 | VJFROMGT | exten => 321xxxxxxx,1,r would this be valid? |
| 02:53.54 | docelmo | VJFROMGT add a _ in front of the 321 when you try to call orlando |
| 02:53.55 | Kobaz | you need _321... |
| 02:54.13 | blitzrage | VJFROMGT: you need to learn how to understand the dialplan |
| 02:54.28 | VJFROMGT | i know |
| 02:54.34 | uluatu | blitzrage: Can I do that when sending my inbound calls to the queues app? In this case queue will do that for me, right? |
| 02:54.37 | blitzrage | exten => _321NXXXXXX,1,Dial(SIP/my_itsp/${EXTEN},30,r) |
| 02:55.28 | VJFROMGT | ok ,, will try |
| 02:55.49 | blitzrage | VJFROMGT: you seriously need to stop working on the dialplan, and go read first |
| 02:56.02 | blitzrage | if you don't know where to put the 'r' option, you are missing some seriously fundamental knowledge. |
| 02:56.14 | VJFROMGT | please entry ure password followed by # key,,, is message i got |
| 02:56.59 | blitzrage | o.O |
| 03:00.09 | uluatu | blitzrage: Could you give me an example how to call Monitor before Queue send the call to an Agent channel? |
| 03:00.53 | blitzrage | just do it before the Dial() |
| 03:00.57 | uluatu | Im using agentlogin to connect my agents |
| 03:01.12 | uluatu | I dont dial the agent. Queue do that for me. |
| 03:01.33 | blitzrage | so Monitor() before Queue() |
| 03:01.46 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:02.12 | Juggie | anyone have a good NPA db w/ proper state/province & country names? |
| 03:02.44 | uluatu | blitzrage: hmmmm, when doing that I will not record the entire hold process of the calle, I will only record the agent bridged call, right? |
| 03:03.10 | *** join/#asterisk alerer (i=markus@AEP-ONE-FIFTY-NINE.MIT.EDU) |
| 03:03.10 | alerer | hi |
| 03:04.28 | alerer | I want to quickly try out an instance of asterisk |
| 03:04.54 | *** join/#asterisk jcoffi1 (n=jcoffi@208.87.0.146) |
| 03:06.17 | *** join/#asterisk brunner (n=chris@66.35.172.123) |
| 03:06.59 | brunner | Who, other than AudioCodes, makes VoIP gateways with T3 interfaces? |
| 03:07.08 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 03:08.57 | pdmmm | didnt lucent? |
| 03:09.02 | pdmmm | max tnt or some crap |
| 03:09.23 | pdmmm | http://cgi.ebay.com/Lucent-Max-TNT-576-Port-DSP-T3-SIP-VoIP-Retail-50-000_W0QQitemZ140288219978QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item140288219978&_trksid=p3286.c0.m14&_trkparms=72%3A1205|66%3A2|65%3A12|39%3A1|240%3A1318|301%3A1|293%3A1|294%3A50 |
| 03:10.07 | pdmmm | i dunno how good it is |
| 03:10.45 | brunner | Holy shit!! $3K!? |
| 03:11.34 | brunner | who cares how good it is |
| 03:11.35 | brunner | lol |
| 03:11.38 | brunner | it's $k |
| 03:11.41 | brunner | $3k* |
| 03:13.08 | pdmmm | haha |
| 03:13.11 | bougyman | brunner: the sangoma 301D is only $1500 |
| 03:13.50 | pdmmm | brunner: too funny |
| 03:13.53 | pdmmm | wut u gunna do with it |
| 03:14.29 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 03:14.31 | brunner | pdmmm: push 10 million minutes |
| 03:15.48 | pdmmm | nice |
| 03:17.15 | brunner | bougyman: I don't see that anywhere on their site |
| 03:17.32 | bougyman | maybe it's D301 |
| 03:17.39 | bougyman | i have 3 of em, i promise they exist. |
| 03:18.27 | pdmmm | i wanna mess with asterisk on solaris i think |
| 03:18.30 | bougyman | weird that they show up as 05:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 03:18.42 | bougyman | when it's a ds3 card. |
| 03:19.00 | brunner | bougyman: you mean A301? |
| 03:19.08 | bougyman | yeah. |
| 03:19.08 | brunner | and aren't they data-only? |
| 03:19.10 | bougyman | that's prolly it. |
| 03:19.13 | bougyman | wanpipe1 | N/A | A300 | 24 | 1 | 1 | N/A | 0 | |
| 03:19.16 | bougyman | Wanrouter Status: |
| 03:19.19 | bougyman | Device name | Protocol | Station | Status | |
| 03:19.21 | bougyman | wanpipe1 | AFT TE3 | N/A | Connected | |
| 03:19.26 | bougyman | i thought you said voip |
| 03:19.39 | brunner | *gateway* |
| 03:19.43 | brunner | as in, TDM to SIP |
| 03:19.51 | bougyman | oh, we're using FS for that. |
| 03:20.10 | brunner | FS? |
| 03:20.17 | bougyman | FreeSWITCH |
| 03:20.34 | brunner | yes, but what hardware? PRI's, right? |
| 03:20.44 | bougyman | yessir, sangoma 8 ports. |
| 03:21.00 | bougyman | they're 4k with echo cancel, < $2k without. |
| 03:21.09 | brunner | I don't want to have to try to connect and configure more than 20 PRIs |
| 03:21.23 | bougyman | it takes minutes, in *. |
| 03:21.33 | bougyman | we've got 16 pris on a few * boxen. |
| 03:21.52 | brunner | but if that Lucent T3 gateway works, |
| 03:22.05 | bougyman | that'd be nice. |
| 03:22.15 | bougyman | <PROTECTED> |
| 03:22.21 | pdmmm | the ascend tnt is cheap |
| 03:22.27 | bougyman | there are a few channelized DS3 cards in prototype. |
| 03:22.35 | bougyman | but none available for purchase yet. |
| 03:22.44 | bougyman | digium announced one, but I never saw it released. |
| 03:22.54 | bougyman | sangoma is telling me they'll have a prototype next month. |
| 03:23.03 | bougyman | they've been telling me that for like 9 months. |
| 03:23.05 | brunner | hmm, nice |
| 03:23.10 | brunner | oh |
| 03:23.41 | *** join/#asterisk Chotaire (i=chotaire@to.to.to) |
| 03:23.53 | docelmo | DS3 card in a PCI infrastructure would eat it alive. The amount of switching that would take place would be insane.. I will believe one will be out when I see it |
| 03:24.03 | bougyman | PCI-E? |
| 03:24.10 | bougyman | there shouldn't be a problem. |
| 03:24.15 | docelmo | I dont know |
| 03:24.32 | docelmo | The amount of contacts would make things interesting for the machine running it |
| 03:24.44 | brunner | god, I wish my telco would get on the ball and finish their SIP setup |
| 03:25.07 | docelmo | Personally on the DS3 level I use AS5400HPX or AS5850 |
| 03:25.10 | bougyman | i decided to screw TDM and just get solid bandwitdh for SIP. |
| 03:25.22 | pdmmm | i think u run into issues w/ interrupts and the kernel |
| 03:25.40 | docelmo | I am going that way now.. I will still have 4 T1's purly for backup and nothing more |
| 03:25.55 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 03:26.01 | docelmo | I admin a 200 seat call center network and right now we have 17 PRI's |
| 03:26.11 | docelmo | We are at the point where VOIP is just cheaper.. |
| 03:26.28 | docelmo | as well some of our clients want Tie lines to their phone systems |
| 03:26.42 | brunner | well I'm racking up at the telco, they just don't offer SIP yet |
| 03:27.03 | docelmo | brunner you a clec? |
| 03:27.09 | brunner | nope, but they are |
| 03:27.13 | docelmo | ahh |
| 03:27.16 | brunner | well, an ILEC, actually |
| 03:27.57 | docelmo | nice.. Im in podunk.. there isnt shit here like that |
| 03:28.48 | brunner | dude, ILECs are only in podunk places |
| 03:28.59 | brunner | this one is in rural PA |
| 03:29.24 | brunner | I wonder if I should take my chances on this Lucent Max TNT |
| 03:29.56 | docelmo | brunner where in PA? |
| 03:30.01 | docelmo | Im from Uniontown |
| 03:30.07 | brunner | I don't even know, honestly |
| 03:30.22 | brunner | I'll know when they give me the ship to address and I'll Google Map it up |
| 03:30.33 | brunner | I wouldn't really care if they were on Mars |
| 03:30.53 | brunner | in fact, the access fees would probably be much higher if they were =D |
| 03:31.16 | brunner | then we could rape the IXCs for dollars per minute instead of cents =p |
| 03:31.18 | docelmo | Mars and Moon is right outside of pittsburg |
| 03:31.23 | docelmo | pittsburgh |
| 03:31.29 | *** join/#asterisk alerer (i=markus@AEP-ONE-FIFTY-NINE.MIT.EDU) |
| 03:35.22 | pdmmm | fuck |
| 03:35.32 | pdmmm | i got like 200 servers to upgrade tomorrow |
| 03:35.59 | brunner | have fun |
| 03:36.43 | pdmmm | shit |
| 03:36.44 | pdmmm | cfengine mang |
| 03:36.56 | pdmmm | makes short work that |
| 03:36.58 | brunner | I think I'm going to pay a little more and go with the cisco |
| 03:45.33 | VJFROMGT | can anyone tell me why the following is not issueing a fake ringback exten => _321xxxxxxx,1,Macro(dialout-trunk,2,${EXTEN},,r) |
| 03:46.03 | Juggie | docelmo, do you have a NPA list w/ good detail, eg State/Province,Country. |
| 03:55.52 | *** join/#asterisk [netman] (n=netman@24.Red-88-22-68.staticIP.rima-tde.net) |
| 04:00.03 | *** join/#asterisk docelm0 (n=vircuser@pool-70-110-117-185.lyn.east.verizon.net) |
| 04:08.00 | brunner | I do, for the US |
| 04:09.41 | apeiron | I wonder, has anyone ever hooked up sphinx, festival, asterisk, and eliza? |
| 04:19.27 | Mog | not all together |
| 04:19.35 | Mog | sphinx isnt that good i thought |
| 04:19.43 | apeiron | All the better! |
| 04:19.48 | Mog | heh\ |
| 04:20.14 | apeiron | This is the sort of thing I'd do for fun, of course, purely because I *can*. |
| 04:24.32 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 04:25.22 | phix | hey |
| 04:26.08 | phix | Say I have an incoming call but I am already using the phone, if I hang up then pick up I get a dial tone (since I am on a seperate line), any way to pickup on the currently incoming call? |
| 04:26.20 | phix | or do i need to setup a queue first? |
| 04:27.51 | *** join/#asterisk thelordmortis (n=lordmort@203.8.160.250) |
| 04:33.36 | k-man | if i set up a music on hold source as an audio stream, does asterisk only stream if it someone is on hold or trying to listen to it? |
| 04:33.57 | *** join/#asterisk Gopaul (n=chatzill@61.17.185.118) |
| 04:45.49 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 04:53.41 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:54.50 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:54.50 | *** mode/#asterisk [+o Mog] by ChanServ |
| 04:55.06 | brunner | "The load is extremely depandant upon what you are doing with it. For example, a simple IVR/Zap-T1-channels-only system can handle 10 times the number of consecutive calls of a SIP&Zap conference call system (at least in my experience)." |
| 04:55.23 | brunner | can anyone verify that the above is roughly the case? |
| 04:58.12 | brunner | and if that's true... why is that the case? |
| 04:59.55 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 05:00.54 | mchou | holy shit |
| 05:01.00 | mchou | google voice |
| 05:01.10 | mchou | unbelievable! |
| 05:01.23 | Qwell | mchou: eh? |
| 05:01.37 | ricko73 | voice.google.com |
| 05:01.41 | ricko73 | nothing there yet |
| 05:01.58 | ricko73 | from their acquisition of Grand Central a few years ago |
| 05:02.02 | Qwell | oh, my G1 has had that for weeks |
| 05:02.07 | Qwell | voice search? |
| 05:02.23 | ricko73 | no it's a phone service |
| 05:02.29 | drmessano | About time |
| 05:02.41 | mchou | no. GC is folding into Google Voice |
| 05:02.41 | ricko73 | Google Features... http://tinyurl.com/aewckq |
| 05:03.10 | Qwell | oh, bleh |
| 05:03.17 | Merlin | brunner: that doesn't seem right to me |
| 05:03.19 | ricko73 | yeah, my thoughts exactly |
| 05:03.43 | brunner | Merlin: no? |
| 05:03.54 | Merlin | brunner: the only thing on the SIP side that could cause higher load is if the SIP peer used some kind of compression |
| 05:04.18 | brunner | hmmm, that's what I thought, but I figured there must be something going on that I wasn't aware of |
| 05:04.29 | Merlin | brunner: on the Zap-T1 side, the IRQ locking issues would give it a disadvantage in my opinion |
| 05:04.33 | *** join/#asterisk MMirkov (n=eraser@85.217.192.18) |
| 05:05.22 | Merlin | brunner: there are so many factors that contribute to load on a conference call |
| 05:05.39 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 05:05.57 | Merlin | brunner: if you have local SIP registrations for handsets or softphones, they might be using different codecs, or different codecs from the SIP trunk, which force asterisk to do transcoding, which is very expensive |
| 05:05.59 | brunner | Merlin: well do you agree with the rough estimate of ~22 meetme users per xeon core? |
| 05:06.23 | brunner | yeah, I won't be doing any transcoding at all |
| 05:06.26 | Merlin | brunner: you'll hit an asterisk limit before you hit a CPU limit |
| 05:06.39 | brunner | Merlin: what sort of asterisk limit? |
| 05:06.42 | Merlin | context locking |
| 05:06.44 | *** join/#asterisk columbo (n=columbo@pool-173-67-102-193.lsanca.dsl-w.verizon.net) |
| 05:07.18 | brunner | Merlin: where can I read about that? |
| 05:07.29 | Merlin | good question :) |
| 05:07.32 | Merlin | i'm not entirely sure |
| 05:07.41 | brunner | I can't find anything about it online |
| 05:07.45 | Merlin | brunner: what version of asterisk are you using? |
| 05:07.52 | brunner | 1.4 |
| 05:07.57 | brunner | but I'm flexible |
| 05:08.07 | brunner | I don't mind switching to avoid issues |
| 05:08.12 | phix | k-man: correct |
| 05:09.11 | phix | hey so anyone going to answer my question above? |
| 05:09.23 | Merlin | brunner: 1.4 is much better than 1.2, and 1.6 is better than 1.4 |
| 05:09.33 | Merlin | brunner: you'll get more users in a meetme with 1.6 |
| 05:09.36 | brunner | Merlin: would you briefly describe what context locking is for me? |
| 05:10.00 | Merlin | brunner: i'm sorry, i meant context switching |
| 05:10.10 | brunner | brb |
| 05:13.18 | Merlin | phix: it sounds like you are using an ATA |
| 05:13.28 | Merlin | phix: you should not use an ATA |
| 05:15.35 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d013f7c7e04e4b51) |
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| 05:33.14 | RichardLynch | I'm looking for that page on the voip-info wiki that lists contacts for asterisk pros in the US. (and failing to find it, though I did find an international list...) |
| 05:38.21 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128544556.dsl.bell.ca) |
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| 05:44.15 | *** part/#asterisk Magicblaze007 (n=sony@fl-67-233-204-182.dhcp.embarqhsd.net) |
| 05:45.40 | SunnyDP | quick recommendation ? i am starting asterisk at school, what distro should i use ??? |
| 05:50.11 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
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| 06:08.29 | jsgoecke | CentOS |
| 06:08.43 | jsgoecke | And if you want it really easy, http://asterisknow.org, which is based on CentOS |
| 06:08.54 | jsgoecke | Or Ubuntu is fine |
| 06:11.43 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 06:22.58 | Steve_J-obs | RichardLynch: do you want an asterisk pro? |
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| 06:45.16 | nbags | hi *'ers, i have a mobile device (an iphone) as a sip client that can either connect via 3G or WiFi. I would like to set it up so that when on WiFi it uses G.711 codec, whilst when on 3G it will use GSM codec. The client software doesn't support this. Can anyone suggest a way to do this on the server (asterisk) side? My 3G IP address is static so I can detect whether it is a 3G connection or not. |
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| 08:26.05 | tokozedg | hi, how can i add sip number in queue? |
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| 09:02.31 | czindy | Hello. Could somebody help please to find out why asterisk tell me te following: ERROR[15790]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. |
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| 09:22.39 | tokozedg | czindy, try to set correct database in res_mysql.conf and in cdr_mysql.conf |
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| 10:03.09 | czindy | Hello again. (Sorry I dosconnected) Could somebody help please to find out why asterisk tell me te following: ERROR[15790]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. |
| 10:04.47 | czindy | I tested the odbc connection on Debian unixodbc/freetds and it is working |
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| 10:30.19 | czindy | Is here anybody who can help me on configure odbc cdr connection? |
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| 10:46.18 | ThoMe | hello |
| 10:46.22 | ThoMe | exten => s,n(spy),ChanSpy(,wSIP/${sipid}) |
| 10:46.27 | ThoMe | is hit s ok for whisper mode? |
| 10:46.30 | ThoMe | or how i can set this? |
| 10:46.40 | ThoMe | exten => s,n(spy),ChanSpy(,w,SIP/${sipid}) |
| 10:46.40 | ThoMe | ? |
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| 11:09.51 | kaldemar | ThoMe: neither is ok. see core show application ChanSpy |
| 11:20.29 | nbags | i have a iphone with a sip client that can either connect via 3G or WiFi. I would like to set it up so that when on WiFi it uses G.711 codec, whilst when on 3G it will use GSM codec. The client software doesn't support this. Can anyone suggest a way to do this on the server (asterisk) side? My 3G IP address is static so I can detect whether it is a 3G connection or not. |
| 11:21.36 | kaldemar | nbags: what version of asterisk are you using? |
| 11:21.56 | nbags | 1.4.22 |
| 11:24.05 | kaldemar | you could try setting variable SIP_CODEC in the dialplan based on the client's ip address. find out ip address with function SIPPEER (core show function SIPPEER in asterisk's CLI). |
| 11:25.48 | czindy | Could somebody help please to find out why asterisk tell me te following: ERROR[15790]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. |
| 11:27.44 | dpryo | Check your connection settings, auth etc |
| 11:28.26 | nbags | kaldemar: Set(clientip=${SIPPEER(peername:ip)}) and use an ExecIf and and another Set() to set the codec? |
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| 11:29.44 | nbags | kaldemar: or i can probably do that in 1 line |
| 11:29.52 | nbags | kaldemar: i will try |
| 11:30.28 | kaldemar | you can do it with a Set that has IF in it. |
| 11:30.56 | nbags | ah, ok. i've never used functions before |
| 11:31.05 | nbags | i will try to figure it out |
| 11:32.16 | kaldemar | Set(SIP_CODEC=${IF($[${SIPPEER(peername:ip)} = "1.2.3.4"]?alaw:gsm)}) <-- something like that |
| 11:32.43 | Chainsaw | orkid: Glad to hear you've solved it. I only saw your message just now from the log. |
| 11:33.28 | dpryo | Do you have any form of string matching? |
| 11:33.31 | nbags | kaldemar: fingers crossed |
| 11:33.57 | dpryo | matching against an exact ipadress would probably not work on 3g, since many operators use dynamic adresses |
| 11:34.13 | nbags | dpryo: mine doesn't ;) |
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| 11:35.36 | kaldemar | dpryo: he has a static LAN address. matching to that will do, if anything else should be gsm. |
| 11:36.47 | nbags | no its actually the opposite, the 3g is static and i want that gsm and everything else ulaw. but thats ok i switched your if statement around |
| 11:41.34 | kaldemar | ok, same case anyway. :) |
| 11:41.39 | nbags | kaldemar: that worked. i get 'Changing codec to 'gsm' for this call because of ...' but now my calls (3g only) are dropping. wonder why... |
| 11:42.27 | kaldemar | pastebin a cli output of a dropped call with sip debug. |
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| 11:51.27 | dr_gogeta86 | good morning |
| 11:54.50 | nbags | kaldemar: sorry it took a while i had to obfuscate it |
| 11:54.57 | nbags | kaldemar: http://pastebin.ca/1358947 |
| 11:55.32 | nbags | kaldemar: i think its probably somewhere between 'SIP/sipppeer-0994d680 is ringing' and 'Spawn extension (default, 9999, 1) exited non-zero' |
| 11:56.00 | nbags | kaldemar: the wifi calls still work |
| 11:59.19 | nbags | now its working ... and i didn't change anything |
| 11:59.58 | nbags | and now its not working again |
| 12:00.04 | nbags | must have gotten lucky |
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| 12:02.03 | kaldemar | nbags: your iphone seems to send BYE right after asterisk has sent a new invite for re-invite purposes. you could try disabling re-invites. |
| 12:03.26 | nbags | kaldemar: yes disabling reinvites works |
| 12:03.44 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 12:04.26 | Stese | hey all |
| 12:04.39 | Stese | Has anyone recently got MixMonitor working? |
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| 12:04.53 | nbags | weird, that it always worked fine with either codec until i put the if statement in |
| 12:05.07 | nbags | but i got the result i was after, so thanks kaldemar |
| 12:05.14 | kaldemar | np |
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| 12:15.59 | dpryo | Anyone with experience regarding crashes/segfaults in channel.c when running standard compiled asterisk? ..which isn't reproduced when running a compiled asterisk without compiler optimizations? |
| 12:20.07 | czindy | Could somebody help on the following error please: ERROR[10665]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. |
| 12:24.40 | czindy | I have a working odbc connection under Debian unixodbc/freetds configured. |
| 12:25.13 | Stese | Newbie Question.... Why would "exten=_2033013836,1,Goto(default|6000|1)" not work if the exten = priority is another other than 1 (ie 2 or n) |
| 12:25.24 | czindy | I configured the cdr_odbc correctly. Could anybody suggest how can I check why asterisk cannot handle this please. |
| 12:25.49 | Stese | Is the table available in your DB? |
| 12:26.13 | czindy | under MSSQL the cdr table is availavle |
| 12:26.55 | Stese | and field names? |
| 12:28.01 | czindy | please wait I'll pastebin it |
| 12:28.48 | Stese | kk, i'm no expert, just suggesting things I'd check... |
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| 12:32.38 | *** join/#asterisk zeljkoMON (n=bum@93.86.99.129) |
| 12:34.20 | kaldemar | Stese: first priority must always be 1. and, don't use _ in front of the extension if it's not a pattern. |
| 12:34.50 | tompaw | Morning! |
| 12:36.44 | tompaw | if I upgrade from 1.6.0.1 to 1.6.0.6 - is there any changelog that would state which config files have to be reviewed? |
| 12:36.48 | tompaw | or are they all compatible? |
| 12:38.01 | angryuser | is there any way in asterisk to do a vide conference more than for 2 participants ? ie 5 users |
| 12:38.07 | angryuser | video* |
| 12:38.17 | mvanbaak | nope |
| 12:38.54 | angryuser | hm, is there any soft i can use with asterisk (or hard) to implement such a feature ? even biz is ok |
| 12:39.17 | russellb | tompaw: when you upgrade within a release like that, config should always be compatible. If there ever was an exception, it would be noted in the UPGRADE.txt file at a minimum |
| 12:40.26 | mvanbaak | angryuser: I have no idea |
| 12:41.13 | zeljkoMON | is there a way to match incoing call to extension (is it coming from another extension or BRI isd)? |
| 12:41.16 | czindy | Stese: thank you btw. Here is my configuration: http://pastebin.com/d5a3d1dc3 |
| 12:41.36 | tompaw | russellb: thanks for an answers. |
| 12:42.13 | russellb | you're welcome |
| 12:42.18 | angryuser | zeljkoMON, try to explain better what do you want to achieve |
| 12:42.53 | zeljkoMON | recepcionist picks up a call, transfers it to extension |
| 12:43.12 | zeljkoMON | if extension doesnt pick up to reroute it back to recpetionst |
| 12:43.44 | tompaw | What is the recommended way for managing * network? I mean - having a few boxes interconnected with each other. Of course I can manage dialplans manually on all of them, but is there some solution for centralized management? |
| 12:43.49 | zeljkoMON | bit prob is that receptionist are lazy and use transfer button on the phone ehicj oni dials extension transferd to |
| 12:44.23 | *** part/#asterisk Merlin (i=merlin@omni.gcinfotech.com) |
| 12:44.23 | Stese | czindy > Sorry, but I can't see anythign in there to stop it working... I'm still struggling with MixMonitor myself! |
| 12:45.05 | angryuser | zeljkoMON, use the timeout in Dial() core show application Dial and assing action you need |
| 12:45.53 | Stese | kaldemar > the first is my Mixmonitor line, and i've removed the _, and it still fails to find the extension |
| 12:45.58 | zeljkoMON | hmm, but shouldnt that reroute all not answerd calls to receptionist? |
| 12:46.07 | angryuser | zeljkoMON, there are dozend of ways doing it, read all options about diall application |
| 12:46.32 | [TK]D-Fender | zeljkoMON: You have to invent this. |
| 12:46.53 | angryuser | zeljkoMON, nope lets say you can create another extension with the timeout and another one withot |
| 12:47.31 | czindy | Somebody check this odbc error please: http://pastebin.com/d5a3d1dc3 |
| 12:47.37 | angryuser | zeljkoMON, so the DID direct ring without imeout one, and the transfer with... |
| 12:47.45 | angryuser | timeout* |
| 12:47.49 | zeljkoMON | angryuser thx for help |
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| 12:52.06 | [TK]D-Fender | Stese: pastebin your dialplan and the debug of your failed call (whatever it comes in on) |
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| 12:55.40 | czindy | sry I disconnected again. [TK]D-Fender do you have any idea regarding this: http://pastebin.com/d5a3d1dc3 |
| 12:59.20 | [TK]D-Fender | czindy: uncomment the table line, and please provide more backup |
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| 13:01.18 | czindy | I uncommented, and the same problem. What do you mean under backup? |
| 13:01.44 | [TK]D-Fender | czindy: odb configs, clis showing the login & table contexts & structure, etc. |
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| 13:01.50 | [TK]D-Fender | czindy: EVERYTHING. |
| 13:05.13 | mvanbaak | tompaw: cfengine can do that for you |
| 13:06.20 | czindy | [TK]D-Fender: I collected more info from environment: http://pastebin.com/d7b270668 |
| 13:06.55 | *** join/#asterisk clintc (n=clintc@n128-227-159-41.xlate.ufl.edu) |
| 13:07.25 | clintc | we do survey research for people... we have a potential client who thinks a phone number is personally identifiable information.. they want us to manually dial the number and have no trace of it on our asterisk pbx... is this even possible? |
| 13:08.17 | mosty | how many phone numbers are you talking about? |
| 13:08.17 | [TK]D-Fender | clintc: What right does anyone have to ask you not to log YOUR CALLS? |
| 13:08.38 | [TK]D-Fender | clintc: And yes you can tell * to keep no logs (CDR) if you really want to. |
| 13:09.00 | clintc | [TK]D-Fender: well... he who pays the piper calls the tune I suppose |
| 13:09.02 | [TK]D-Fender | clintc: tahts what NoCDR() is for as a more selective app |
| 13:09.13 | mosty | how will they give you the number to dial if you're not supposed to see it? |
| 13:09.15 | clintc | [TK]D-Fender: right, but isn't the number is other logs |
| 13:09.29 | [TK]D-Fender | clintc: only log is CDR <- |
| 13:09.39 | clintc | mosty we will log into a web site with a code that gives us a name and number |
| 13:10.17 | mosty | offer to disabled CDR logging, see if that's acceptable |
| 13:10.18 | [TK]D-Fender | clintc: Sounds like you shouldn't even have your rep see the number but rather have * dial it for them so it remains hidden from even them. |
| 13:10.52 | clintc | [TK]D-Fender: it seems to me I have looked at other logs like /var/log/asterisk/messages where there are phone numbers.. maybe not.. maybe just on the asterisk console |
| 13:11.14 | [TK]D-Fender | clintc: by default the only logging is CDR |
| 13:11.20 | tompaw | mvanbaak: sounds like a hardcore way. |
| 13:12.09 | mvanbaak | tompaw: you can also use something like rsync or subversion |
| 13:12.36 | tompaw | mvanbaak: so you're suggesting propagating the same dialplan file to all the servers, yeah? |
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| 13:13.14 | clintc | [TK]D-Fender: thanks, I'll have a look at noCDR and see if we can make that work |
| 13:13.20 | mvanbaak | if they have to do the same task, yes |
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| 13:29.58 | jplank | has anyone every hooked up * with a viking rc-2a controller? |
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| 13:35.21 | jsgoecke | Anyone out there have an Asterisk connected to an Avaya SES? http://groups.google.com/group/adhearsion/t/50aa978dbce6e65c |
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| 13:48.42 | dandre | Hello, |
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| 13:49.15 | dandre | I don't understand this message in asterisk console: |
| 13:49.16 | dandre | <PROTECTED> |
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| 13:49.27 | dandre | where does it come from? |
| 13:49.38 | dandre | why do I have this message? |
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| 13:55.46 | jsgoecke | How did I get here? |
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| 13:57.11 | LuisTorres | Hi all |
| 13:57.53 | Kate-o | Hello |
| 13:58.09 | LuisTorres | Hi Kate |
| 13:58.27 | LuisTorres | if anyone could help me with a realtime question |
| 13:58.37 | LuisTorres | I set realtime mysql sip config |
| 13:59.09 | LuisTorres | but when I add a new sip extension I always need to reload it on the Cli..., doesnt update automaticly |
| 13:59.18 | LuisTorres | any ideas where I can look? |
| 14:00.10 | tompaw | if canreinvite is used, does asterisk act like a switch then? |
| 14:00.22 | tompaw | I mean, regarding just the rtp |
| 14:00.33 | *** join/#asterisk davevg (n=davevg__@74.94.3.214) |
| 14:00.35 | LuisTorres | yep |
| 14:01.38 | tompaw | so, am I correct here? [voip_switch: rtp proxy: no, sip proxy: no] [ast_reinvite: rtp proxy: no, sip proxy: yes] [ast_noreinvite: rtp proxy: yes, sip proxy: yes] |
| 14:02.28 | kaldemar | tompaw: asterisk is not a sip proxy in any situation. |
| 14:02.45 | kaldemar | if reinvites are used, asterisk doesn't stay on the media path. |
| 14:03.02 | LuisTorres | lol |
| 14:03.19 | LuisTorres | srry I was thinking that was to me |
| 14:03.24 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
| 14:04.25 | tompaw | kaldemar: sorry, by "proxy" I mean it sets up an intependent connection with both sides. so when I call someone through asterisk, there is a sip call between me and asterisk and between asterisk and my destination. |
| 14:04.46 | kaldemar | or to be more exact, if reinvites are used, asterisk doesn't force itself on the media path. it may still be on it though, depending on the scenario. |
| 14:05.40 | kaldemar | tompaw: yes, and that's called a back to back user agent (B2BUA). |
| 14:06.06 | kaldemar | proxy is different. |
| 14:06.20 | LuisTorres | any ideas why asterisk is not updating in realtime? |
| 14:08.54 | tompaw | kaldemar: ok, I meant b2bua then ;-) |
| 14:11.29 | *** join/#asterisk af_ (n=getsmart@88-149-230-72.dynamic.ngi.it) |
| 14:15.13 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 14:15.13 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05) |
| 14:15.13 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk viq (n=viq@unaffiliated/viq) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk dpryo (n=hn@donatello.ondskap.net) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk bullium (n=will@216.68.250.30) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM] |
| 14:15.13 | *** join/#asterisk unpaidbill (n=bill@alteredbeastiality.org) |
| 14:18.40 | ingenius | Split ... |
| 14:19.49 | tompaw | ... is a town in Croatia. |
| 14:20.47 | *** join/#asterisk SirWhit (n=sirjames@blk-222-38-6.eastlink.ca) |
| 14:21.50 | Katty | hi |
| 14:22.00 | Katty | how to use asterisk pls? |
| 14:22.05 | stintel | lol |
| 14:22.13 | ingenius | :P |
| 14:22.21 | Kobaz | you must first be initiated |
| 14:22.35 | Katty | hun, i was initatited 5 years ago |
| 14:22.38 | Kobaz | licks Katty |
| 14:22.44 | Katty | pats Kobaz |
| 14:22.54 | Katty | oh wait, no, 6 |
| 14:23.17 | SirWhit | has anyone used the new confbridge app yet? |
| 14:23.20 | dandre | I don't understand this message in asterisk console: |
| 14:23.20 | dandre | -- Local/43@auto-offhook-fa32,1 requested special control 20, passing it to mISDN/tmp0-u32 |
| 14:23.20 | dandre | where does it come from? why do I have this message? |
| 14:23.42 | Kobaz | is it any better than meetme? |
| 14:23.54 | Katty | sadly i've not upgraded yet |
| 14:23.58 | Kobaz | speaking of meetme, i have a patch to meetme i should submit |
| 14:24.04 | Katty | due to some API changes with support of isymphony |
| 14:24.08 | SirWhit | it uses the new bridging API.. |
| 14:24.14 | Kobaz | ah |
| 14:24.19 | Kobaz | the arbitrary bridging functions |
| 14:24.23 | Kobaz | those are definitly a nice add |
| 14:24.25 | SirWhit | but I am trying to figure out if there are any huge benefits to using it |
| 14:24.32 | Katty | as soon as isymphony rolls out support for 1.6 i'm going to be all over that. |
| 14:24.35 | SirWhit | besides not requiring a zaptel timing source |
| 14:24.45 | Kobaz | yeah, that's probably the biggest thing |
| 14:24.45 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:24.51 | Katty | hi jason |
| 14:25.00 | jasonwoot | ahoy hoy |
| 14:25.36 | Katty | jasonwoot: my asterisk is broke. |
| 14:25.39 | Katty | jasonwoot: pls to fix. |
| 14:25.43 | Katty | jasonwoot: i give you cookie |
| 14:26.28 | jasonwoot | press star 723 charlie |
| 14:26.39 | Katty | horay! it's fixed! |
| 14:26.42 | Katty | gives jasonwoot cookie |
| 14:26.57 | jasonwoot | you know, a cookie with one bite out of it looks like a C |
| 14:27.14 | jasonwoot | remembers every episode of sesame street |
| 14:27.29 | Katty | do you remember when maria had to go to the hospital |
| 14:27.47 | *** join/#asterisk [Rated-R] (n=b0red@bookit-dev.com) |
| 14:28.09 | jasonwoot | a dark day on the street |
| 14:28.59 | Katty | do you remember the yipyips? |
| 14:29.18 | jasonwoot | actually, those things were a wee bit creepy |
| 14:29.27 | Katty | REF: http://www.youtube.com/watch?v=Z4VNMERVsC4 |
| 14:29.50 | Katty | wtb earthbook |
| 14:30.10 | [TK]D-Fender | Katty: as a kid I was terrified of them :) |
| 14:31.36 | jasonwoot | I've got my kids watching fraggle rock now, but I think they see it as punishment |
| 14:31.57 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 14:32.12 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 14:32.13 | Katty | meh |
| 14:32.18 | Katty | letter people were more fun |
| 14:32.58 | Deeewayne | I always liked the tweedle bugs that lived in ernie and bert's plant |
| 14:33.18 | Deeewayne | nobody ever remembers them |
| 14:33.20 | jasonwoot | my ernie impression is spot on... see it at astricon 09 |
| 14:33.37 | ayeso | When purchasing T1s to carry g711 calls, how many concurrent calls should be estimated for each T1? |
| 14:33.48 | ayeso | 24? |
| 14:34.15 | Katty | 23 |
| 14:34.24 | Katty | 24th channel is used for data |
| 14:34.30 | Katty | such as callerid, and signalling |
| 14:34.43 | Katty | or maybe that's a pri. |
| 14:34.43 | ayeso | Katty: Im talking about SIP though, |
| 14:35.02 | ayeso | Katty: yea thats a pri |
| 14:35.06 | coppice | about 15 using SIP |
| 14:35.23 | jasonwoot | ayeso, as in, how many concurrent SIP calls can you carry across a 1.5 mbit symettrical circuit? |
| 14:35.52 | ayeso | jasonwoot: well yes, more of whats best practice for network architecture. |
| 14:36.02 | Kobaz | wide_awake: does foo have any sequences in use? |
| 14:36.04 | Kobaz | er |
| 14:36.06 | jasonwoot | ayeso, what preferred codec? |
| 14:36.27 | ayeso | jasonwoot: I need to make 2 estimates, one for 711u and 729 |
| 14:36.33 | SirWhit | don't forget about any data traffic too... (if your network is not only using SIP) |
| 14:36.49 | ayeso | SirWhit: this will be dedicated to voip traffic only |
| 14:38.03 | SirWhit | take a look at http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 14:38.14 | SirWhit | that is a good tool for estimating the number of lines.. |
| 14:38.22 | coppice | 15 for G.711, and about 50 for G.729 |
| 14:38.28 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:38.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:38.29 | SirWhit | taking packet encapsulation into account |
| 14:40.15 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:40.15 | SirWhit | where it really gets interesting are customers that want to jam as many lines on a cable modem or some other hybrid internet connection |
| 14:40.28 | Katty | we actually have a cable modem with about 20 sip trunks on it |
| 14:40.32 | Katty | but they're not all in use at the same time. |
| 14:40.37 | Katty | mnostly, they're just did numbers |
| 14:40.47 | Katty | well one of our clients, i should say |
| 14:41.08 | SirWhit | I meant.. concurrent calls.. ;) |
| 14:41.24 | Katty | i would say they have no more than 10 calls at any given time |
| 14:42.55 | *** part/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 14:42.57 | jasonwoot | ayeso, we use a 10mbit fiber to support about 50 concurrent inbound and perhaps 20 concurrent outbound, ulaw |
| 14:44.32 | Stese | Can anyone advise me on this... I know it's simple to most, but i've poured over all the info i have (forums, oReillys book and Wiki) and I can't work it out... :( |
| 14:44.33 | Stese | http://pastebin.com/m3a23ff48 |
| 14:44.42 | *** join/#asterisk Imo (n=Imo@brln-4db82b76.pool.einsundeins.de) |
| 14:44.46 | Imo | hello |
| 14:45.14 | Imo | i have Asterisk 1.2.13 and i dont found my sounds ? |
| 14:45.33 | Imo | where can i found this or in this conf i found the path to the sounds ? |
| 14:45.51 | ayeso | jasonwoot: I need to figure out what I need for 864 concurrent calls, and account for redundancy.. Im thinking 3 dS3s in a PPP multilink, but its overkill |
| 14:45.57 | mosty | imo: how did you install asterisk? |
| 14:46.14 | Imo | yes with apt-get install asterisk |
| 14:46.18 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 14:46.30 | Imo | i looked in /var/lib/asterisk |
| 14:46.36 | Imo | but there no sounds |
| 14:46.39 | Katty | Stese: what purpose does the / in the middle of the SIP extension serve? |
| 14:46.51 | Katty | Stese: REF: SIP/84415307/07872376824 |
| 14:46.53 | mosty | Imo, then you should have asterisk-sounds-main installed, which provides the sound files |
| 14:47.11 | Stese | exten=2033013835,n,Dial(SIP/84415307/07872376824) ? |
| 14:47.17 | Imo | mosty: what ? |
| 14:47.18 | Katty | Stese: yes. |
| 14:47.25 | Katty | Stese: what is the purpose of the / between 7 and 0 |
| 14:47.26 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:47.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:47.34 | Katty | lmadsen: mew. |
| 14:47.43 | [TK]D-Fender | Katty: peer/exten |
| 14:47.44 | lmadsen | Katty: berk |
| 14:47.52 | Stese | it the number to pass to the VSP to dial... apologies the last line isn't relevent... I'm calling the other ext! |
| 14:47.56 | lmadsen | anyone know how to stop a Monitor() on a channel beyond a SIP transfer? |
| 14:48.00 | [TK]D-Fender | Stese: And you aren't looking at the SIP debug of the failed call... |
| 14:48.01 | Imo | mosty: i want do change my soundfiles |
| 14:48.04 | neurosys | all: Ive been doing my newbie trial and playing on 1.4. After having a blast trashing the filesystem with you dont want to know, I'm going to start fresh with a new install. Should i be using 1.6? I ask because i noticed 1.4 compain a lot about future deprected commands in the CLI. |
| 14:48.04 | mosty | Imo, the asterisk debian package requires that another package called asterisk-sounds-main is installed. this package provides the sounds- you are probably just looking in the wrong place |
| 14:48.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:48.08 | Imo | from english to german |
| 14:48.22 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 14:48.28 | [TK]D-Fender | neurosys: Kiss them goodbye in 1.6 then |
| 14:48.50 | neurosys | [TK]D-Fender: Then I should be a big boy and dive in eh? :) |
| 14:48.54 | Imo | asterisk-sounds-main also installed |
| 14:49.10 | lmadsen | i.e. caller --> Monitor() --> callee. Then callee --> SIP transfers (attended) --> another_person. The Monitor() is associated with the 'caller', and beyond the transfer the call recording continues for both conversations -- I need the file broken up between the transfers, or, at least the 2nd portion chopped off and scrapped. |
| 14:49.17 | [TK]D-Fender | neurosys: Get your hands off your nuts an SEIZE THE DAY! |
| 14:49.19 | mosty | Imo, asterisk-sounds-main only provides the english sounds, as far as i know |
| 14:49.28 | neurosys | [TK]D-Fender: Yes sir! :) |
| 14:49.44 | Imo | mosty: i know i have german files but i must found the folder |
| 14:49.54 | Imo | mosty: where are my sounds ;) ? |
| 14:50.02 | Stese | [TK]D-Fender > I had assumed that the issue was dialplan related, not SIP... |
| 14:50.04 | mosty | imo: how did you install the german sounds? |
| 14:50.29 | Katty | Imo: i'm going to say that your apt-get installation did not include everything. most of us compile from tarballs. |
| 14:50.36 | JayTee52 | Katty, mornin! |
| 14:50.45 | [TK]D-Fender | neurosys: Honestly though, get your stuff up to 1.4 / 1.6 hybrid spec minimum. no more "|", deprecated stuff in 1.4 then look at when 1.6 offers you something you want that 1.4 doesn't and you'll be more ready to convert |
| 14:50.45 | Katty | JayTee52: oh my |
| 14:50.48 | Imo | mosty: /etc/lib/asterisk/sounds |
| 14:50.49 | Katty | JayTee52: did you have a birthday? |
| 14:50.59 | [TK]D-Fender | Stese: You can't see what CONTEXT its looking in. |
| 14:50.59 | Imo | Katty: there is include |
| 14:51.01 | JayTee52 | not yet |
| 14:51.06 | [TK]D-Fender | Stese: SIP debug would reveal that. |
| 14:51.08 | mosty | imo: ok but how did you install the german sounds? |
| 14:51.13 | Katty | JayTee52: for what porpoise did you add numbers to your name |
| 14:51.18 | neurosys | [TK]D-Fender: Good advice. Thank you. |
| 14:51.21 | [TK]D-Fender | stevetotaro: Nor what peer was matched, etc. |
| 14:51.23 | Imo | how ? |
| 14:51.33 | Imo | wget ........... tar xvfz |
| 14:51.36 | JayTee52 | oh, I must have my other nic logged in at the same time, this is a secondary |
| 14:51.56 | JayTee52 | yep, I'm still logged in from home |
| 14:52.08 | Katty | oh, right |
| 14:52.08 | Imo | but on this path didnt had a folder sounds |
| 14:52.14 | Imo | <PROTECTED> |
| 14:52.23 | Imo | but i dont found the path |
| 14:52.31 | Katty | jbot: basic installation guide? |
| 14:52.36 | Katty | jbot: getting started? |
| 14:52.41 | Imo | please say me the path to the foulder ;) ? |
| 14:52.42 | Katty | jbot: installation? |
| 14:52.50 | Katty | jbot: :< |
| 14:52.51 | jbot | methinks < is redirection of stdin to a program |
| 14:52.52 | mosty | imo: the tar command should have shown you where it was installing the files |
| 14:52.58 | jaytee_work | ~installation |
| 14:53.07 | Katty | jbot: setup |
| 14:53.08 | jbot | Graphical installer for Unix applications based on GTK and XML. URL: http://www.lokigames.com/development/setup.php3 |
| 14:53.08 | jasonwoot | ayeso, is fiber an option? |
| 14:53.15 | Katty | jbot: gah! |
| 14:53.16 | jbot | methinks gah is one of the top favourite words |
| 14:53.21 | jaytee_work | hehe |
| 14:53.23 | jasonwoot | no way I would do thta much over copper unless necessary |
| 14:53.39 | Imo | mosty: yes i think i have to install the sounds to a other path |
| 14:53.52 | Imo | but i dont found |
| 14:54.24 | ayeso | jasonwoot: It is, but I need to have redundant circuits, so that would probably be way overkill |
| 14:54.28 | mosty | run tar tvzf on the tarball, to see the file names |
| 14:54.42 | mosty | then run find / -type f -name <put the name here> |
| 14:54.46 | Imo | mosty: the path is wrong |
| 14:55.09 | [TK]D-Fender | Stese: Nor what peer was matched, etc. |
| 14:55.47 | jasonwoot | ayeso, I use 3 voice T1s to backup the fiber SIP trunks, because although they come from the same provider, they route through different pedestals and COs...., |
| 14:56.11 | jasonwoot | ayeso, if I got data T1s and fiber, chances are if one were down the other would be too |
| 14:56.27 | Imo | mosty: dosnt work |
| 14:56.37 | jasonwoot | ayeso, sucks supporting zap for this reason, but oh well |
| 14:56.43 | Imo | mosty: in this conf i can look the path ```?? ? |
| 14:56.57 | ayeso | jasonwoot: I bet |
| 14:57.02 | mosty | Imo, what doesn't work? |
| 14:57.09 | Imo | find ..... |
| 14:57.11 | Katty | mosty: you don't work |
| 14:57.27 | Katty | mosty: i think he is broken. |
| 14:57.30 | mosty | imo: show me the exact command you ran |
| 14:57.31 | ayeso | jasonwoot: were actually going to specify diverse path for these circuits. |
| 14:57.32 | Imo | Katty: yesyes ;) i need the path |
| 14:57.49 | Imo | i have searched but i dont found anyone |
| 14:58.28 | *** join/#asterisk flujan (n=flujan@189-039-010-068.static.spo.ctbc.com.br) |
| 14:58.34 | flujan | ping putnopvut |
| 14:58.36 | Katty | Imo: locate tt-monkeys |
| 14:58.47 | putnopvut | flujan: pong |
| 14:59.26 | Imo | katty: why you dont say me the conf from asterisk ??? there stand the path ??? |
| 14:59.40 | flujan | putnopvut: Hey put, the AUDIOHOOK_INHERIT works great on the attended transfers... but there is a issue I wanna to disculls. |
| 14:59.42 | *** join/#asterisk mazpe (n=me@adsl-074-173-020-013.sip.mia.bellsouth.net) |
| 14:59.43 | Katty | Imo: that does not parse. please try again. |
| 14:59.48 | flujan | putnopvut: *discuss. |
| 14:59.55 | putnopvut | flujan: What is the issue? |
| 15:00.04 | flujan | putnopvut: It is not working on queues using the mixmonitor |
| 15:00.11 | Katty | where is file |
| 15:00.18 | mazpe | how can i set a voicemail to use the custom recorded unavailable message? instead of the standard one |
| 15:00.24 | *** part/#asterisk cptcrash|away (n=jonmoore@70.159.118.86) |
| 15:00.24 | putnopvut | flujan: you mean if you set the monitor-type=mixmonitor in queues.conf? |
| 15:00.25 | Katty | file: muffins. |
| 15:00.27 | [TK]D-Fender | Katty: Have you searched folder? |
| 15:00.29 | Katty | file: MUFFINS |
| 15:00.37 | Imo | I NEED ONLY THE PATH TO MY NORMAL ASTERISK SOUNDS |
| 15:00.39 | flujan | putnopvut: yes |
| 15:00.40 | mosty | imo: you put the files in the wrong place. the asterisk config doesn't magically know where you put them |
| 15:00.41 | putnopvut | flujan: That kind of makes sense since the mixmonitor would be on the member's channel instead of the caller's. |
| 15:00.52 | Imo | mosty: i know |
| 15:00.54 | mazpe | i tried "exten => 999,1,VoiceMail(u999@mrh)" |
| 15:00.59 | putnopvut | flujan: as a result, you'd have to set the AUDIOHOOK_INHERIT function on the member's channel, possibly using a macro. |
| 15:01.11 | Stese | [TK]D-Fender > I've updated the Pastebin with some SIP Debug info (http://pastebin.com/d618f5c85) and for reference, i'm calling the number ending in 836 |
| 15:01.11 | Imo | <PROTECTED> |
| 15:01.17 | mazpe | but i get the following error: app_voicemail.c:4188 leave_voicemail: No entry in voicemail config file for 'u999' |
| 15:01.26 | Katty | Imo: did you locate tt-monkey? |
| 15:01.30 | flujan | putnopvut: I am considering to hack the code and enable it to everything... what do you say? |
| 15:01.31 | [TK]D-Fender | mazpe: "core show application voicemail" <- |
| 15:01.34 | Kobaz | mazpe: find more up-to-date documentation |
| 15:01.41 | Imo | Katty: what is tt-monkey ? |
| 15:01.42 | mosty | imo: ahh i see. in debian it's /usr/share/asterisk/sounds/ |
| 15:01.45 | putnopvut | flujan: Yeah, you could do that if you want. |
| 15:01.47 | thehar | tt-monkey is monkies! |
| 15:01.50 | Katty | Imo: it is an audio file. |
| 15:01.55 | Katty | Imo: located in the sound directory. |
| 15:01.55 | thehar | tt-weasels > tt-monkey |
| 15:02.02 | [TK]D-Fender | mazpe: Its pretty clear its looking for the "u" as part of the box #. |
| 15:02.02 | flujan | putnopvut: I can change the behavior of the mixmonitor app, but i dunno if it will work. The app mix_monitor is the right place to change? |
| 15:02.03 | Katty | Imo: locate your file, and you locate your directory. |
| 15:02.13 | [TK]D-Fender | mazpe: That should tell you your formatting is wrong. |
| 15:02.21 | Imo | mosty: thank you |
| 15:02.23 | Imo | ;) |
| 15:02.25 | Katty | thehar: i agree. |
| 15:02.36 | thehar | Weasels have eaten our phone system! |
| 15:02.51 | Katty | we should redirect the blacklisted numbers to the weasels. |
| 15:02.53 | putnopvut | flujan: hmmm, I don't think you'll be able to make the appropriate changes there... |
| 15:03.02 | thehar | that's a fantastic idea Katty ! |
| 15:03.05 | Kobaz | should play that through Page() to all phones with forced auto-answer |
| 15:03.11 | flujan | putnopvut: probably not... lol |
| 15:03.20 | Katty | OR even better, we can reroute them to an IVR, which APPEARS to be a sex chat. |
| 15:03.27 | Katty | Please hold for the first available honey. |
| 15:03.29 | thehar | haha |
| 15:03.34 | mazpe | [TK]D-Fender: for it... ,b options |
| 15:03.39 | mazpe | [TK]D-Fender: thank you |
| 15:03.43 | jasonwoot | ayeso, PM pls |
| 15:03.44 | Katty | We take mastercard and visa. |
| 15:03.55 | [TK]D-Fender | Katty: And AB- :p |
| 15:03.57 | Katty | Please note you are currently being billed. |
| 15:04.02 | thehar | haha |
| 15:04.03 | thehar | yes! |
| 15:04.22 | Katty | Your wait time is now, 15 minutes. |
| 15:04.50 | Katty | or maybe Next In Line would be more appropriate. |
| 15:05.23 | jaytee_work | "please continue to hold as your call is very unimportant to us" |
| 15:05.30 | Imo | how i can copy a folder with the console ? |
| 15:05.32 | tzafrir_laptop | "you are not the next in line" |
| 15:05.38 | thehar | has to stop playing around and do actual work. :( |
| 15:05.38 | tzafrir_laptop | Imo, cp -a |
| 15:05.42 | flujan | putnopvut: well, it is part of the learning process... i will check it out. :) thanks for the tips putnopvut |
| 15:05.54 | putnopvut | flujan: all right. good luck. If you have problems, let me know. |
| 15:05.56 | Imo | thanks |
| 15:06.05 | flujan | putnopvut: ok thank you |
| 15:06.58 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 15:07.46 | Katty | jaytee_work: i have a random talkswitch box i should connect to my home phone with an IVR setup like that on it. |
| 15:08.04 | *** join/#asterisk sasargen (n=chatzill@72-58-184-175.pools.spcsdns.net) |
| 15:08.25 | Stese | Has anyone ever seen letters appear in CLID's before? |
| 15:08.37 | Katty | jaytee_work: it's a shame i can't get a static IP address at home |
| 15:08.51 | Katty | jaytee_work: i could do all sorts of fun rerouting |
| 15:09.22 | Katty | Stese: is the name field or the number field? |
| 15:09.29 | Stese | it's the number field... |
| 15:09.54 | Katty | if you put a letter in the number fiend, and then call a sprint number off a local switch... |
| 15:10.01 | Katty | provided it's routing through AT&T |
| 15:10.05 | Stese | something my commercial manager just asked me...we had one on our incoming CDR |
| 15:10.11 | Katty | you can get letters in the number field |
| 15:10.34 | Katty | Stese: mostly i would expect it to be a database lookup error at the telco |
| 15:10.40 | Stese | Hmm, I doubt it's AT&T, they aren't a provider in the UK :) |
| 15:10.50 | Katty | Stese: or someone who's outputing something funky off a PRI where they have permission to overwrite the callerid |
| 15:11.03 | Stese | I'm just going to double check the master.csv |
| 15:11.07 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:11.50 | bmoraca | hexadecimal phone numbers! |
| 15:12.00 | Imo | where can i found german asterisk sounds file for asterisk 1.2 ?? or can i use the sounds from 1.4 ? |
| 15:12.04 | Qwell | bmoraca: Why not alpha-numeric? |
| 15:12.15 | Stese | Katty > DYNDNS service helpful in this situation? |
| 15:12.40 | Stese | katty > (Dynamic IP Addressing) |
| 15:12.59 | LuisTorres | Hi .., just found that is Im using rtcachefriend=yes , I must to do a sip reload everytime I add some new sip ext..., anyone knows if it the normal behavior? |
| 15:12.59 | bmoraca | Qwell: cause putting 36 keys on a telephone is a lot more inconvenient than 15 keys? |
| 15:13.04 | brutuz | can someone point me to implementing HA on asterisk? |
| 15:13.46 | bmoraca | er, 16 keys |
| 15:13.48 | brutuz | or something close.. since moving T1 lines cant be automatic if a hardware failure occurs |
| 15:13.53 | Qwell | bmoraca: says who? |
| 15:14.46 | bmoraca | i know i certainly don't want 36 buttons on my damn telephone, that's for sure. and i don't fancy having to use predictive text or any other kind of letter entry mechanism to dial a phone number |
| 15:15.12 | Stese | is it going to be worse than putting an email address? |
| 15:15.17 | Qwell | You realize numbers would be shorter and...like dns...words |
| 15:15.21 | Qwell | right? |
| 15:18.16 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 15:18.33 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:19.07 | Imo | how can i update asterisk 1.2 to asterisk 1.4 on debian minimal ?? |
| 15:19.55 | bmoraca | Qwell: still not sure i like that idea. most of the time when that happens, it gets over-engineered...look at IPv6. |
| 15:20.20 | *** join/#asterisk tokozedg (n=rock@89.232.24.53) |
| 15:21.08 | bmoraca | brutuz: have your trunkgroup split up between multiple boxes so that if one goes down, the other automatically picks it up. |
| 15:21.36 | bmoraca | brutuz: that's more a function of your telco than your asterisk boxes. |
| 15:22.38 | bmoraca | brutuz: and then on the other side, use a load-balancing appliance such as an F5 or Barracuda, so that clients are seemless. |
| 15:23.00 | tokozedg | hi, how can i record call after user pucks up? |
| 15:23.16 | Stese | Can anyone have a look at this, and tell me why it might not be working... i've now added SIP debug info to my pastebin |
| 15:23.49 | Stese | tokozedg > I'm trying to do that as well... what is your configuration? |
| 15:24.23 | mosty | Imo, you might want to just update to the new stable release of debian- it has asterisk 1.4 |
| 15:24.48 | Imo | mosty: i have a vserver ;) |
| 15:24.51 | Imo | i cant update |
| 15:24.55 | tokozedg | i dont have it right now, but i set it and it was recording everything include ringing, Monitor was before Dial so, is there another way? |
| 15:25.45 | mosty | imo: you can update vservers |
| 15:25.55 | Imo | ????? |
| 15:26.10 | *** join/#asterisk tdonahue (n=tdonahue@vonmail.vonworldwide.com) |
| 15:26.15 | Imo | how ? ? |
| 15:26.17 | tdonahue | hi all |
| 15:26.50 | Stese | tokozedg > Ok, i'm not personally bothered by ringing, so my issue isn't particularly relevent |
| 15:28.13 | Stese | my issue > http://pastebin.com/d618f5c85 |
| 15:29.40 | *** join/#asterisk adnc (n=adnc@unaffiliated/adnc) |
| 15:30.01 | tdonahue | i'm trying to build asterisk 1.6.0.6 with dahdi-linux-complete-2.1.0.4+2.1.0.2 installed on a CentOS 5.2 server, but the build is failling at app_dahdiras.c |
| 15:30.07 | tdonahue | the log is at http://pastebin.ca/1359089 |
| 15:30.19 | tdonahue | can anyone recommend how to fix this problem? |
| 15:30.26 | Chainsaw | tdonahue: Yes, there's an existing patch for that. |
| 15:30.28 | brutuz | bmoraca: can you suggest something to read on this one? online or books.. |
| 15:30.29 | adnc | hello, does someone know if there a sort of VoiceXML to asterisk-configuration conversion application? |
| 15:30.30 | Chainsaw | tdonahue: Let me find it for you. |
| 15:31.13 | Chainsaw | tdonahue: It is Digium bug #14480, #14516, #14620 or #14626 |
| 15:31.25 | Imo | tdonahue: use dahdi complete |
| 15:31.34 | Chainsaw | tdonahue: Those are the "will be in 1.6.0.7" patches that I had to apply to make 1.6.0.6 actually compile on my kit. |
| 15:31.36 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:31.49 | tokozedg | Stese, exten=2033013836,1,Goto(default|6000|1) |
| 15:31.57 | tdonahue | Imo, it is dahdi complete |
| 15:32.03 | tokozedg | here you have this and in default, exten = 6050,1 |
| 15:32.15 | Chainsaw | tdonahue: Yes, it's a specific change. The patch will fix it. |
| 15:32.15 | [TK]D-Fender | Stese: there is no exten to match in that context, just like it says |
| 15:32.22 | tdonahue | Chainsaw, i guess that means it was a really popular one, huh? :) |
| 15:32.30 | Imo | tdonahue: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz |
| 15:32.46 | Imo | tdonahue: this version ?? |
| 15:32.52 | tokozedg | exten = 6000,1,VoiceMailMain(${CALLERID(num)}@default), try this one in default |
| 15:33.04 | [TK]D-Fender | Stese: Looking for 2033013836 in DID_84415307 (domain 80.68.42.146). [DID_84415307] only has "s" |
| 15:33.22 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:33.22 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:34.22 | tdonahue | Imo, same sha1 as the one I downloaded |
| 15:35.01 | Imo | tdonahue: hmmm i used that and i dont had problems ;) sorry i dont know |
| 15:35.06 | tdonahue | i prefer to have the version numbers in the tarball to make it easier to figure out what version i have downloaded :) |
| 15:35.24 | Chainsaw | Imo: I do know, it's one of the Digium bug numbers I just gave. |
| 15:36.57 | tdonahue | Chainsaw, it looks like #14516 |
| 15:37.22 | *** join/#asterisk Curus (n=Curus@92.62.204.2) |
| 15:38.06 | tdonahue | figures they wouldn't put the patch into the bug... |
| 15:38.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:39.03 | Chainsaw | tdonahue: I probably have it separate. Let me see. |
| 15:39.31 | *** join/#asterisk ming_zym (n=ming_zym@220.181.35.152) |
| 15:39.33 | tdonahue | i found it in the SVN, no big deal |
| 15:39.40 | Chainsaw | tdonahue: Okay. |
| 15:40.30 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:41.18 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 15:41.25 | tompaw | Hmm.. |
| 15:41.39 | bmoraca | brutuz: not really...are your PRIs all from the same telco provider? |
| 15:41.48 | brutuz | yes |
| 15:41.52 | tompaw | why does my asterisk try to authenticate an invite even though it's set to insecure = invite? |
| 15:41.55 | tompaw | http://pastebin.com/m4503da93 |
| 15:41.55 | brutuz | bmoraca: yrd |
| 15:42.21 | brutuz | bmoraca: yes |
| 15:42.27 | bmoraca | brutuz: what happens, or should be happening, anyway, is that your PRIs are part of a trunk group, not treated as separate trunks |
| 15:43.01 | Curus | tompaw: Is there a peer with the same name? |
| 15:43.07 | bmoraca | brutuz: so what'll happen is when a call comes in for your trunk, it'll choose the lowest-ordered available trunk (usually, this configuration can be changed) |
| 15:43.27 | bmoraca | brutuz: so, if trunk 1 is not available (full or down), it'll roll to trunk 2, etc |
| 15:43.41 | tdonahue | perfect, it compiled with that patch |
| 15:43.50 | tompaw | Curus: peers are identified by IP, not names, aren't they? |
| 15:43.51 | tdonahue | Chainsaw, thank you for the link to that bug |
| 15:43.57 | Curus | tompaw: Nope |
| 15:44.05 | Curus | asterisk first tries by name, then by IP |
| 15:44.06 | bmoraca | brutuz: so by increasing the number of trunks you have (8 trunks if you accomodate for 96 total calls) and devide them between two systems, you will give yourself failover |
| 15:44.11 | bmoraca | brutuz: it's not cheap, though |
| 15:44.12 | Chainsaw | tdonahue: Any time. I packaged 1.6.0.6 for Gentoo and I ran into it myself. |
| 15:44.12 | RypPn | um, might not be related, but shouldn't it be insecure = port,invite ? |
| 15:44.29 | brutuz | bmorca: that's what im about to say... |
| 15:44.34 | brutuz | bmorca: $$$$$ |
| 15:44.36 | Curus | RypPn: Only if the port might not match between the register and the actual call |
| 15:44.42 | bmoraca | brutuz: what I would do at that point, though, is use a third box as your proxy/gateway |
| 15:45.20 | bmoraca | brutuz: so, now what happens is you can use the capacity from all four needed trunks without wasting any, and calls will be completed even if an entire box fails |
| 15:45.45 | bmoraca | brutuz: HA on a box that trunks via sip is easier than one with physical trunks...so you can allocate a failover box for the gateway |
| 15:45.58 | bmoraca | brutuz: either by loadbalacing or using Linux HA |
| 15:46.08 | tompaw | Curus: but there is no peer by that name. |
| 15:46.09 | ayeso | I asked this previously, but there seems to be more people here now.. what is the best practice for how many simultaneous SIP g711 calls you can have on 1 T1? |
| 15:46.11 | bmoraca | brutuz: then you lose the need to duplicate your required number of trunks |
| 15:46.29 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:47.49 | bmoraca | brutuz: not sure how a loadbalancer would work with SIP, although I imagine it'd be fine...most of them work a lot like NAT. obviously, failover would be stateless, so active calls would be dropped, but there would be no service interrutions |
| 15:47.50 | brutuz | bmorca: do have a sample infra diagram for that 2nd suggestion? |
| 15:47.53 | lowtek | Hi all. Is there a function that will check to see if an extension is valid for a context in 1.4.x? I need to use a pattern _NXXNXXXXXX to handle some logic then I want to see if the extension dialed that matches that pattern actually exists in that context and if it doesn't provide congestion. |
| 15:48.19 | bmoraca | brutuz: no...but i could probably figure one up in paint real quick |
| 15:48.54 | brutuz | bmorca: can you place it somewhere in the net.. i just want to have an idea.. |
| 15:49.46 | bmoraca | damn, forgot to install visio...give me a sec |
| 15:51.16 | Katty | where do i want lunch at |
| 15:51.35 | tompaw | why does asterisk ignore my sip.conf peer? |
| 15:51.40 | apeiron | wonders how Visio diagrams actually help people |
| 15:51.43 | tompaw | I mean |
| 15:51.54 | tompaw | if it doesn't find a matching peer by name it should look for it by ip, shouldn't it? |
| 15:52.40 | tompaw | I think I know. |
| 15:53.00 | Stese | [TK]D-Fender > Sorry for slow reply... yes that extension doesn't exist in that context, but surely it should follow the catchall 's' to _default? |
| 15:53.23 | tompaw | forget my question ;) |
| 15:53.36 | Katty | [TK]D-Fender: what do i want for lunch. |
| 15:53.56 | Gido-E | s extension is not a catchall extension. |
| 15:54.32 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:55.29 | [TK]D-Fender | Katty: have you considered eating? Many mortals enjoy that I hear |
| 15:56.16 | Stese | Katty > how about some lightyly fried Newbie.... my arm is quite tender |
| 15:57.24 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 16:00.35 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:00.36 | mort_gib | You call the local telco, Uhm I can't ping address XXX from Address YYY, could you check your routing please |
| 16:01.07 | mort_gib | I can ping address XXX from ANY address in the world (tried Denmark, Spain, Gibraltar UK) |
| 16:01.22 | brutuz | bmorca: hmm.. so i need a F5 (LTM) w/c will go between client and asterisk |
| 16:01.39 | mort_gib | Telco replies, give me the config from router running XXX I think there is an ACL issue on that router... |
| 16:01.51 | mort_gib | %$£)(*&!!! |
| 16:01.57 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:02.18 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 16:02.18 | mort_gib | Not having a nice time with the local telco these days!!! |
| 16:02.25 | lowtek | Is there a way to check if an extension is valid in a context before sending a call to that extension? |
| 16:02.26 | *** join/#asterisk dni (n=one@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 16:03.32 | lowtek | an explitic extension, not a wildcard/pattern |
| 16:04.01 | dni | hello all,. can someone verify if "just for interoffice/local" calls i would not need a digium card ? |
| 16:05.12 | bmoraca | brutuz: check PM |
| 16:05.37 | mosty | dni, correct. depending on what you mean by "local" |
| 16:05.53 | mosty | just need a bunch of sip phones and an asterisk server |
| 16:06.00 | Chainsaw | dni: If you're all SIP and don't need conferencing, then indeed, you can do without. |
| 16:06.36 | Chainsaw | dni: Conferencing features will require a stable timing source, as will devices that only support analog connections (that old fax machine in the corner, etc). |
| 16:06.45 | Chainsaw | That didn't come out right. |
| 16:06.56 | Chainsaw | Conferencing needs a stable time source, which Digium hardware can provide. |
| 16:07.29 | Chainsaw | Also, that will provide FXS ports to connect old-style analog phones/faxes and perhaps FXO to have a backup POTS line coming in. |
| 16:07.39 | bmoraca | brutuz: the point of the loadbalancer is that it uses NAT to make multiple machines have a single outside pressence. in the context of SIP, it would be in terms of failover not loadbalancing in the conventional sense. |
| 16:07.41 | Chainsaw | It depends on how your calls come in really. |
| 16:08.01 | *** join/#asterisk Shadad (n=Shadow@S01060040f4fac494.vf.shawcable.net) |
| 16:08.26 | lowtek | dni, you don't need hardware for a timeing source, you can use zaptel by itself without hardware, it has a psuedo channel that will handle timeing for meetme. |
| 16:08.36 | *** join/#asterisk Ashetic (n=Ashetic@89.119.206.193) |
| 16:08.40 | bmoraca | brutuz: my drawing was wrong...you would only need one, but the idea is the same. multiple SIP trunks from each gateway box, and lots of fault tolerance built into your asterisk dialplan |
| 16:08.45 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 16:09.15 | Stese | ztdummy i think is a valid timing source |
| 16:09.25 | lowtek | yup, that's it, ztdummy |
| 16:09.40 | bmoraca | what would really be cool is if there was stateful failover with asterisk...but the overhead would probably be huge |
| 16:10.12 | Shadad | Hello everyone, I am looking for some info on how to accept "*" in a GotoIf statement. Cannot find any information on this in the wiki and im not sure if its treated like "#". |
| 16:10.33 | lowtek | Shadad: exten => *,1,do_something |
| 16:11.53 | Shadad | lowtek: The * I want to "read" and be a selection in an ivr, when a user enters * I get "unexpected '*', expecting $end; Input: *= 6" with the program exiting |
| 16:12.16 | mosty | Shadad, pastebin the dialplan code |
| 16:12.22 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 16:13.48 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 16:14.26 | Shadad | http://pastebin.com/m56658172 - Fails at 888,15 with the "unexpected '*', expecting $end; Input: *= 6" error |
| 16:15.06 | mosty | Shadad, try quotes around ${MENU} ? |
| 16:16.05 | DavidR2008 | does anyone have any experience with the applicationmap section of features.conf on * 1.4? If I could get a very basic example working I should be able to build on that |
| 16:16.56 | mosty | DavidR2008, there's one here: http://www.voip-info.org/wiki-Asterisk+config+features.conf |
| 16:17.04 | Shadad | mosty: You sir are a genius :) That did it! thank you |
| 16:17.43 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
| 16:18.59 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 16:19.45 | DavidR2008 | mosty: the example there (which I have been reading and trying) seem to be 1.2 and either doesn't work, or works differently on 1.4 |
| 16:20.42 | *** join/#asterisk t_corr (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:20.43 | t_corr | phew |
| 16:20.51 | Imo | i have the next problem ;) i dont listen the music from musiconhold where can i test the music ?? |
| 16:21.56 | DavidR2008 | using the monkey example I put this in features.conf: monkey => #9,peer,Playback,tt-monkeys |
| 16:21.58 | DavidR2008 | and after reloading res_features.so I see this on the console: |
| 16:22.00 | DavidR2008 | == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9' |
| 16:22.02 | DavidR2008 | What do I need to put in my dialplan to actually be able to use this? |
| 16:22.08 | mosty | DavidR2008, there should be an up to date version in the sample configs from whatever version of asterisk you're using |
| 16:22.17 | brutuz | bmorca: im familiar with network level but not in telephony.. it should work right? |
| 16:22.48 | Imo | DavidR2008: hmm but i want test the musiconhold ??? |
| 16:23.02 | Imo | i listen the other sounds ? |
| 16:23.15 | brutuz | bmoraca: PM'd u |
| 16:23.32 | mosty | DavidR2008, i don't think you need anything special in your dialplan |
| 16:23.39 | *** part/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 16:23.47 | suma | Is it possible to have the featured for a certain call and other calls will not ? |
| 16:23.51 | suma | *features |
| 16:24.02 | suma | The features in features.conf |
| 16:25.09 | mosty | suma, you can disable transfers and one touch recording with Dial options and channel variables i believe |
| 16:25.27 | Ashetic | hallo.. i have some throuble with asterisk. My inbound registratons doesn't seem to hit the inbound context. i have configured it in "register => blablabla/inbound", in the [sipproxy] context=inbound |
| 16:25.35 | Ashetic | what can it be? |
| 16:25.42 | Imo | i dont listen and i dont get errors |
| 16:28.18 | Imo | i get this error |
| 16:28.19 | Imo | Mar 12 16:28:04 NOTICE[7221]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 16:28.42 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 16:29.33 | Imo | what i have to do ? |
| 16:30.49 | kaldemar | Ashetic: show the configuration. what you just said doesn't make sense. |
| 16:30.54 | Ashetic | yes |
| 16:31.03 | Ashetic | sip.conf is ok? |
| 16:31.18 | kaldemar | yes, for starters |
| 16:32.22 | Ashetic | http://rafb.net/p/LrsTp016.html |
| 16:32.49 | kaldemar | the inbound you have in the register statements is ment for extensions, not contexts. |
| 16:33.19 | Ashetic | this is the last try i made, the running config actually doesn't have it and still doesn't work |
| 16:34.27 | kaldemar | what is your actual problem? incoming calls don't land in [inbound]? |
| 16:34.40 | Ashetic | exact |
| 16:34.55 | Ashetic | incoming call actually doesn't land at all |
| 16:35.04 | Ashetic | i tried with verbose and debug |
| 16:35.07 | Stese | where does the call come from? |
| 16:35.18 | kaldemar | do you have a cli output of a failed call with sip debug? |
| 16:35.27 | Ashetic | having just "=== SIP ToS ..." |
| 16:35.30 | Ashetic | yes |
| 16:35.45 | Ashetic | http://rafb.net/p/BrH2QC19.html <--- failed incoming call |
| 16:36.09 | Imo | when i registar my sip account into my fritz box to my asterisk. and then i restart my asterisk i cant call to my sip account from fritz box ??? |
| 16:36.28 | bkw_ | Ashetic: I think the / is contact not context |
| 16:37.00 | Ashetic | bkw_, what line are you referring to? (Thanks for support) |
| 16:37.06 | kaldemar | Ashetic: not debug on, but sip debug on so the SIP trace shows |
| 16:38.04 | kaldemar | Ashetic: he means the same /inbound i was saying about earlier |
| 16:39.01 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 16:39.24 | bkw_ | Ashetic: on the register. its blahblah/contact |
| 16:39.24 | Ashetic | ah ok |
| 16:39.34 | Ashetic | ok ok... this is not the problem :D |
| 16:39.35 | bkw_ | the inbound call is matched against a user/peer entry in the sip.conf |
| 16:39.52 | bkw_ | not the register line... but that could be different since I don't really use asterisk anymore |
| 16:40.11 | *** join/#asterisk rashed2020 (n=shabati@67.205.245.208) |
| 16:40.28 | rashed2020 | Anyone know if I can interface the SPA3000 and * over the internet using dyndns |
| 16:40.39 | kaldemar | bkw_: the register line can be responsible for telling the other end where this particular asterisk is. we haven't seen an incoming call yet. |
| 16:41.11 | bkw_ | kaldemar: I know all about that part |
| 16:41.24 | Ashetic | http://rafb.net/p/JxnS5M22.html <--- failed call with sip debug |
| 16:41.25 | kaldemar | bkw_: i'm sure you do. :) |
| 16:41.28 | [TK]D-Fender | rashed2020: Its an device like any other. |
| 16:41.33 | *** join/#asterisk pawpro (n=Miranda@213.166.12.34) |
| 16:42.18 | rashed2020 | [TK]D-Fender: What I was wondering was if I can use a URL instead of an IP address in the SPA. |
| 16:42.23 | rashed2020 | I don't have one yet so I can't check |
| 16:42.27 | kaldemar | oh, a cisco gw.. |
| 16:43.16 | Ashetic | kaldemar, talking to me? one of my proxyes use a sip gw |
| 16:43.28 | Ashetic | is it something i should concern of? |
| 16:43.39 | mosty | rashed2020, of course you can |
| 16:43.49 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 16:43.54 | kaldemar | Ashetic: no concers. |
| 16:44.02 | pawpro | Hi everybody. I need help. I'm sipping from Xlite to *1.6 the call falls into extension "sipout" exten => _X.,1,Dial(SIP/${EXTEN}@secondasterisk) now in sip.conf i've got [secondasterisk]. Xlite authenticates to first asterisk with username xlite. The second asterisk reject the call from first asterisk because of unknown peer xlite. But it's a peer of first asterisk. what am i doing wrong? |
| 16:44.04 | Stese | rashed2020 > you can, just ensure you can resolve the name correctly to the right IP address |
| 16:44.06 | RoPBX | hello all |
| 16:45.02 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 16:45.02 | RoPBX | please, someone knows how to share a sip trunk in two asterisk boxes via IAX ? |
| 16:45.30 | rashed2020 | Thanks, guys. |
| 16:46.06 | rashed2020 | Oh one more thing, does the SPA need any ports open on the WAN? |
| 16:46.28 | kaldemar | Ashetic: the INVITE comes in, your asterisk responds with "401 Unauthorized", but the other end doesn't send a new invite with a challenge response. |
| 16:46.48 | Ashetic | ...so? |
| 16:46.53 | Stese | rashed2020 > Port 5060 for SIP and 10,000 to 20,000 for RTP |
| 16:46.53 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 16:46.57 | Ashetic | what am i missing? :( |
| 16:47.07 | kaldemar | Ashetic: the context is not your problem, the gw not authenticating its requests is. |
| 16:47.30 | *** join/#asterisk pbrunnen (n=pbrunnen@67.151.65.231) |
| 16:47.37 | *** part/#asterisk pbrunnen (n=pbrunnen@67.151.65.231) |
| 16:47.56 | rashed2020 | Great, thank you. |
| 16:48.04 | RoPBX | somebody knows about IAX ? |
| 16:48.29 | kaldemar | Ashetic: it even matches to peer voip.eutelia.it. remove the /inbound from your register lines and start resolving the authentication issue either by making the other end authenticate or disable authentication on your end (which is bad). |
| 16:48.45 | Ashetic | try |
| 16:49.29 | Ashetic | tnx |
| 16:49.58 | mosty | pawpro, pastebin the two sip.conf's? |
| 16:50.00 | kaldemar | RoPBX: http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 16:50.06 | pawpro | mosty: sure |
| 16:51.10 | RoPBX | thanks kaldemar |
| 16:51.27 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 16:59.24 | Ashetic | still doesn't work |
| 16:59.27 | Ashetic | uff |
| 17:00.20 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 17:00.26 | Ashetic | http://rafb.net/p/BGcft447.html <--- new sip debug on call |
| 17:01.35 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 17:01.38 | pawpro | mosty: First asterisk : http://pastebin.com/d3925fce second asterisk: http://pastebin.com/d5a42a26b |
| 17:02.30 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:04.08 | *** join/#asterisk mtryfoss (n=mtryfoss@80.239.93.22) |
| 17:04.22 | pawpro | mosty: so the first asterisk is just a pbx and the second asterisk is a sip trunk both should authenticate in a secure matter. second asterisk heas realtime that i'm using for clients but now i'm trying to put testing machine outside and i cant make it work. Both asterisks 1.6 |
| 17:04.34 | [TK]D-Fender | Ashetic: They are sending you an un-authed call. Do "insecure=port,invite" for their peer entry |
| 17:05.40 | *** join/#asterisk prpa1982 (i=4e00f8f0@gateway/web/ajax/mibbit.com/x-3a7198a8e1afaa42) |
| 17:06.07 | kaldemar | nice function from the gateway. it ACK's the 401 and just sends a new invite with a new call-id. now that's a dumb network device. |
| 17:06.22 | mtryfoss | I have some strange problems when dialing local channels. Now and then when the calls is answered I get this error: ast_do_masquerade: Fixup failed on channel IAX2/pstn6-2643<MASQ>, strange things may happen. and the server crashes. |
| 17:06.25 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 17:06.26 | mosty | pawpro, can you also post the dialplan on the first server, and the error logs? |
| 17:06.28 | mtryfoss | anybody experienced the same thing ? |
| 17:06.39 | Ashetic | trying. Thanks [TK]D-Fender |
| 17:06.42 | pawpro | mosty: one sec |
| 17:06.54 | prpa1982 | hi all.... |
| 17:07.06 | prpa1982 | can i ask You something about Asterisk X100P(B2) FXO |
| 17:07.23 | SuPrSluG | i'm trying to redirect a number from one asterisk box to another using Transfer cmd, but it fails. If I use the Dial command it works. What does transfer need to complete? |
| 17:07.44 | Ashetic | [TK]D-Fender, which line tells you that are sending unauthed calls? (tnx) |
| 17:07.53 | *** join/#asterisk kerx (n=kerx@adsl-69-104-17-222.dsl.irvnca.pacbell.net) |
| 17:07.57 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 17:09.17 | SuPrSluG | insecure= |
| 17:09.20 | prpa1982 | i was wondering of You guys know would Asterisk X100P(B2) FXO would OOB with LinuxMCE? |
| 17:09.55 | Ashetic | [TK]D-Fender, wow!!! it works now! Thanks a lot! |
| 17:10.57 | tompaw | the Big Task: configure SPA3102 to direct incoming PSTN Calls to a local extension passing the caller id. |
| 17:11.02 | kaldemar | cisco's gateways can authenticate and do work with asterisk though. |
| 17:11.08 | tzafrir_laptop | prpa1982, the X100P is a PCI device. Asterisk is a software (that can also use it) |
| 17:11.11 | *** join/#asterisk areay (n=areay@93-97-161-123.zone5.bethere.co.uk) |
| 17:11.28 | tzafrir_laptop | LinuxMCE is an operating system (linux distribution, even) |
| 17:11.44 | tzafrir_laptop | Now could you please ask a more specific question? |
| 17:11.52 | prpa1982 | tzafrir_laptop: ....thx for the anwser...i know that, LinuxMCE uses Asterisk for its telephony system |
| 17:12.37 | prpa1982 | i was just wondering if anybody here knew something about Asterisk X100P(B2) FXO....and would it work on LinuxMCE OOB?...its a Kubuntu distribution |
| 17:13.08 | prpa1982 | tzafrir_laptop: ....since i only have POTS telephones in my home |
| 17:13.28 | [TK]D-Fender | Ashetic: You're welcome |
| 17:13.41 | prpa1982 | i want to use this card and install it in my LMCE machine so i can hook up my POTS telephones on LMCE... |
| 17:13.55 | pawpro | mosty: extensions.conf http://pastebin.com/d46e6624b , debug from first asterisk http://pastebin.com/d5b15b6b2 , debug from second asterisk http://pastebin.com/d27d9afc1 |
| 17:13.55 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 17:13.58 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 17:14.00 | [TK]D-Fender | prpa1982: Yes it will work. if your system came with ZAPTEL compiled is another matter |
| 17:14.11 | mikealeonetti | I do love coming in here |
| 17:14.17 | [TK]D-Fender | prpa1982: Zaptel (or DAHDI) is required for that hardware interface. |
| 17:14.22 | Imo | how i can configurate the voicemail ??? |
| 17:14.34 | *** join/#asterisk lyll (n=lYlL@wikipedia/lylvlyl) |
| 17:14.37 | adnc | is there a voicexml interpreter for asterisk? |
| 17:14.43 | prpa1982 | [TK]D-Fender....yes, it came with ZAPTEL compiled... |
| 17:14.43 | [TK]D-Fender | Imo: "vi /etc/asterisk/voicemail.conf" |
| 17:14.53 | [TK]D-Fender | prpa1982: then you should be fine |
| 17:15.05 | Imo | i mean the voicemail menu ? |
| 17:15.15 | [TK]D-Fender | adnc: I've head of one or two on the WIKI. go search it |
| 17:15.16 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:15.26 | prpa1982 | [TK]D-Fender...thx a lot...yust cant figure out do i need FXO or FXS card |
| 17:15.28 | [TK]D-Fender | Imo: You don't configure the menu. the options are fixed |
| 17:15.35 | adnc | [TK]D-Fender: are they named as such? |
| 17:15.38 | tzafrir_laptop | prpa1982, that card can hook to a PSTN line. It does not allow you to hook an analog phone to Asterisk |
| 17:15.42 | [TK]D-Fender | prpa1982: What do you want to do? |
| 17:15.47 | tzafrir_laptop | ~fxsfxo |
| 17:15.49 | jbot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 17:15.58 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:16.02 | mikealeonetti | A lot of the users in my office are having problems that when they dial out to people that have other auto attendants, the other system cannot "hear" the tones. The cisco phones we use a fine, though. Any option I'm missing? |
| 17:16.13 | Imo | when i have a unavail massage |
| 17:16.33 | Imo | after that plays asterisk vm-intro again |
| 17:16.35 | Imo | its shit |
| 17:16.44 | prpa1982 | jbot....so i would need fxs/fxo card? |
| 17:17.02 | mort_gib | mikealeonetti: Look at your DTMF setting in sip.conf |
| 17:17.05 | [TK]D-Fender | Imo: It plays the prerecorded instructions because you did not tell it to do otherwise |
| 17:17.15 | tzafrir_laptop | prpa1982, jbot's a bot |
| 17:17.18 | prpa1982 | [TK]D-Fender...i want to connect my POTS telephone to Asterisk.... |
| 17:17.19 | [TK]D-Fender | prpa1982: Stop talking to the bot... |
| 17:17.25 | prpa1982 | lol:) |
| 17:17.31 | prpa1982 | figuered after i typed it:) |
| 17:17.36 | [TK]D-Fender | prpa1982: then that card is not what you need. That is for plugging you HOME LINE into |
| 17:17.39 | mikealeonetti | jbot: I love you |
| 17:17.39 | jbot | You love you? |
| 17:17.50 | prpa1982 | damn, havent used irc for loooooooooooooonng time:) |
| 17:18.04 | prpa1982 | [TK]D-Fender....then which card do i have to use? |
| 17:18.12 | mikealeonetti | mort_gib: it's rfc2833 |
| 17:18.14 | [TK]D-Fender | prpa1982: http://www.telephonydepot.com/Catalog/Linksys-Analog-Adapters/Linksys-PAP2T-NA |
| 17:18.23 | *** part/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 17:18.38 | prpa1982 | [TK]D-Fender....thanx a lot for the link |
| 17:18.49 | [TK]D-Fender | prpa1982: tons of choices, this is the best value at entry and will probably serve all of your needs quite well |
| 17:19.20 | prpa1982 | [TK]D-Fender....thanks again for your suggestion...i will look it up... |
| 17:19.41 | Imo | [TK]D-Fender: i have overwrite my message |
| 17:19.58 | prpa1982 | [TK]D-Fender...all i need is to hook up my phone to Asterisk...thats all |
| 17:20.08 | prpa1982 | maybe i will upgrade my phones with a few Cisco 7970 |
| 17:20.25 | [TK]D-Fender | prpa1982: overkill and waste of $ IMO |
| 17:20.50 | prpa1982 | cisco 7970? |
| 17:20.51 | tzafrir_laptop | prpa1982, just get yourself a grandstream |
| 17:20.55 | [TK]D-Fender | prpa1982: yup |
| 17:20.58 | tzafrir_laptop | expects TK autoresponder |
| 17:21.03 | [TK]D-Fender | tzafrir_laptop: EWWWWWWWW!!!!!!!!!!!! |
| 17:21.14 | [TK]D-Fender | tzafrir_laptop: First the personalized scream! |
| 17:21.16 | [TK]D-Fender | then... |
| 17:21.18 | [TK]D-Fender | ~gs |
| 17:21.19 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 17:21.19 | ricko73 | pick up an open box Polycom |
| 17:21.20 | [TK]D-Fender | ~grandstream |
| 17:21.21 | jbot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 17:21.28 | pawpro | mosty: any hope for me? |
| 17:21.35 | mikealeonetti | all of the Cisco phones have no trouble dialing out, though |
| 17:21.43 | mikealeonetti | rather being "heard" |
| 17:21.48 | mikealeonetti | everybody can dial out |
| 17:21.49 | Corydon76-dig | [TK]D-Fender: I think we may want to avoid those characterizations |
| 17:21.52 | prpa1982 | [TK]D-Fender....i would use Cisco 79701 phones for Orbiters in LMCE system |
| 17:21.57 | *** join/#asterisk SparFux (n=raoul@e182025114.adsl.alicedsl.de) |
| 17:21.59 | prpa1982 | hehehe, YUGO |
| 17:22.01 | mazpe | whend dialing out ${EXTEN} is the number that is been called... is there a variable that defines that extension that is making the call? |
| 17:22.15 | prpa1982 | i am from CRoatia guys..former Yugoslavia republic |
| 17:22.20 | [TK]D-Fender | prpa1982: Avoid Cisco period. Overpriced and trouble both legally and configuration wise |
| 17:22.35 | Corydon76-dig | Grandstream makes phones that people love to hate, but most of the characterizations are based upon their earliest phones, which are no longer produced |
| 17:22.43 | tzafrir_laptop | prpa1982, if you happen to have a cisco phone: do use it. They make good phones. But probably not wirth their price for a home system |
| 17:23.10 | kaldemar | Corydon76-dig: don't forget handytones. oh so handy. |
| 17:23.13 | [TK]D-Fender | mazpe: Stop calling a "device" as an "extension". And you can look at the CHANNEL name to get that, etc |
| 17:23.28 | [TK]D-Fender | mazpe: go read the CHANNElVARIABLES documentation for all of this |
| 17:23.32 | Corydon76-dig | Grandstream's videophones aren't that bad |
| 17:23.42 | n3hxs | prpa1982, We won't hold that against you as long as you didn't work in the YUGO factory. |
| 17:24.48 | mazpe | [TK]D-Fender: got it |
| 17:24.56 | Corydon76-dig | ~cisco |
| 17:24.57 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 17:25.32 | *** join/#asterisk prpa1982 (i=4e00f8f0@gateway/web/ajax/mibbit.com/x-6291c4daf52d5f00) |
| 17:25.57 | prpa1982 | [TK]D-Fender: ...sorry, got booted |
| 17:26.25 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05) |
| 17:27.19 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 17:29.38 | dan__t | hrm I thought Grand Central was dead in all forms |
| 17:29.40 | dan__t | Interesting. |
| 17:30.02 | SparFux | Hello. When trying to connect my iaxcomm to asterisk, I get the error: "chan_iax2.c:7565 socket_process: Rejected connect attempt from 192.168.118.8, who was trying to reach '2663@res'" I have set up iax.conf and iaxcomm configuration appropriately, I'd say. http://pastebin.com/d76a8e6dc What is wrong here? |
| 17:30.06 | *** join/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 17:30.30 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 17:31.40 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 17:33.23 | *** join/#asterisk kc2tnk (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 17:35.59 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk aenaus (n=hdgfghf@91.140.106.239) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk dude7064 (n=dude7064@78-86-79-212.zone2.bethere.co.uk) [NETSPLIT VICTIM] |
| 17:35.59 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-41-171.dhcp.mrqt.mi.charter.com) [NETSPLIT VICTIM] |
| 17:36.14 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 17:36.55 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 17:37.38 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:41.18 | rbd | hi guys...I'm registered for some AMI events from my asterisk server...when a user is in a meetme room and hangs up, it seems that the Hangup event is happening before the MeetMeLeave event...is this normal, and is there any way to switch this order around? |
| 17:41.28 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 17:42.46 | *** part/#asterisk Imo (n=Imo@brln-4db82b76.pool.einsundeins.de) |
| 17:44.29 | SuPrSluG | SparFux, there's no username= |
| 17:45.45 | mosty | rbd, how does asterisk know the channel as left the conference before it hangs up? |
| 17:48.57 | SuPrSluG | ~polycom |
| 17:48.58 | jbot | it has been said that polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 17:53.06 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 17:57.00 | *** join/#asterisk stabler (n=seedbox@rrcs-70-60-8-130.central.biz.rr.com) |
| 17:59.51 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:00.18 | SuPrSluG | i'm trying to redirect a number from one asterisk box to another using Transfer cmd, but it fails. If I use the Dial command it works. What does transfer need to complete? |
| 18:01.08 | [TK]D-Fender | SuPrSluG: Transfer just throughs the call with no auth attached and the phone tries to auth itself to the other side |
| 18:01.14 | *** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 18:01.36 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 18:01.57 | ZX81 | hey all, anyone know what this means: __sip_xmit: sip_xmit of 0x8d56ef8 (len 504) to 192.168.7.250:5066 returned -1: Operation not permitted |
| 18:01.59 | rpm | any mediatrix gurus in here? i'm using an 1104 and trying to make it not place the Route: header in my sip dialog/messages. |
| 18:02.04 | ZX81 | run by root |
| 18:02.20 | ZX81 | also got the same result on a ping as root last night |
| 18:03.39 | SuPrSluG | [TK]D-Fender, thanks its a telephone number i'm trying to forward. Transfer(NXXNXXXXXX@xxx.xxx.xxx.xxx) , like that, but it never hits the other box. with dial it will complete. |
| 18:04.00 | [TK]D-Fender | SuPrSluG: Show me. |
| 18:04.25 | [TK]D-Fender | ZX81: Depends on the operation. |
| 18:05.24 | ZX81 | [TK]D-Fender: idle box - 7:05am here |
| 18:05.53 | [TK]D-Fender | ZX81: you'd have to see what that is in response to. |
| 18:06.12 | ZX81 | lsof -p 19096 |wc -l shows 260 |
| 18:06.35 | ZX81 | I got the same thing last night pinging the network gateway |
| 18:07.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:07.26 | ZX81 | ping: sendmsg: Operation not permitted |
| 18:07.51 | SuPrSluG | [TK]D-Fender, http://pastebin.com/m11493c4d |
| 18:08.18 | ZX81 | Google just has links to people who can't configure their network and get that for every packet |
| 18:09.07 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 18:09.11 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 18:10.28 | mazpe | anyone having an issue polycom 301 (or any other) transfering a call to any extension starting with 10? trying to transfer to extension 101, the phone with out waiting dials 10. |
| 18:10.35 | mazpe | yet extension 201 works fine |
| 18:10.53 | [TK]D-Fender | SuPrSluG: I don't see a failed attempt in that. |
| 18:11.16 | SuPrSluG | i'll call it and post |
| 18:11.20 | [TK]D-Fender | ZX81: how does "ping" turn into a SIP REsPONSE? |
| 18:11.44 | [TK]D-Fender | mazpe: becase you did not configure the dialplan on the phone. |
| 18:12.42 | *** join/#asterisk mmlj4-play (n=jkelly@209.16.86.78) |
| 18:13.46 | ZX81 | :) it doesn't - just interesting I got the same error from ping as from asterisk - both permission denied - both local lan - both as root |
| 18:14.44 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 18:16.23 | pdmmm | selinux? |
| 18:18.01 | SuPrSluG | [TK]D-Fender, http://pastebin.com/m101e0823 |
| 18:18.25 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 18:18.32 | ZX81 | SELINUX=disabled |
| 18:21.41 | SuPrSluG | not much to go on,eh |
| 18:23.34 | ZX81 | wow murf gone too now? |
| 18:23.42 | ZX81 | how many people laid off at digium? |
| 18:23.51 | ZX81 | will we ever have working cdr? :) |
| 18:24.23 | pdmmm | eh? |
| 18:24.25 | pdmmm | wuts wrong with cdr |
| 18:24.35 | ZX81 | :) |
| 18:24.36 | ZX81 | heh |
| 18:24.40 | ZX81 | what's not |
| 18:24.41 | ZX81 | :) |
| 18:25.30 | ZX81 | I might try and do an interview - who should I ask? Greg Vance/Mark/Kevin/xxx? |
| 18:26.38 | Corydon76-dig | An interview for what? |
| 18:27.12 | ZX81 | regarding layoffs - how many - how are things generally going etc |
| 18:27.40 | ZX81 | Dwayne and Murf seemed pretty good assets - maybe I just don't fully understand |
| 18:27.59 | Corydon76-dig | For media contacts, Leslie Conway is the person to ask |
| 18:28.18 | ZX81 | but their posts of layoffs have been to the asterisk-biz list and I wonder if it might be an idea to write a post to calm the waters |
| 18:28.24 | ZX81 | ok |
| 18:28.31 | Qwell | ZX81: See msg for contact information. |
| 18:28.34 | ZX81 | ty |
| 18:28.44 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:28.50 | madgeek | it might be a good idea to mind your own damn business |
| 18:29.00 | ZX81 | heh |
| 18:29.01 | *** join/#asterisk vncsnvs (n=vncsnvs_@201.86.135.130.dynamic.adsl.gvt.net.br) |
| 18:29.05 | [TK]D-Fender | SuPrSluG: ......... |
| 18:29.08 | ZX81 | it is my business |
| 18:29.10 | [TK]D-Fender | SuPrSluG: no comment. |
| 18:29.12 | ZX81 | ~adn |
| 18:29.13 | jbot | well, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 18:29.13 | SuPrSluG | yes |
| 18:29.37 | SuPrSluG | ? |
| 18:29.51 | madgeek | your looking mighty lonely on my ignore list |
| 18:29.57 | madgeek | you're* |
| 18:30.09 | *** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca) |
| 18:30.56 | Qwell | madgeek: For asking a question he feels is important? |
| 18:31.08 | SuPrSluG | [TK]D-Fender, no ideas? |
| 18:31.20 | prashant_jois | The Time on the asterisk console is off by one hour (ahead) and I can't figure out why. The system time is correct and the timezone on the system time is correct. Any ideas why? Google did not turn up anything useful. |
| 18:31.20 | [TK]D-Fender | Qwell: Hearing that from a non-Digium person.. kinda funny |
| 18:31.21 | *** join/#asterisk JohnAds (n=chatzill@200.169.18.174) |
| 18:31.29 | *** part/#asterisk JohnAds (n=chatzill@200.169.18.174) |
| 18:31.39 | [TK]D-Fender | SuPrSluG: why the hell don't I see SIP DEBUG ion there? |
| 18:31.40 | Qwell | [TK]D-Fender: I'm a bit confused by it, honestly. |
| 18:31.56 | *** join/#asterisk JafoJ (i=40506c37@gateway/web/ajax/mibbit.com/x-abb094489d0249bb) |
| 18:32.19 | [TK]D-Fender | prashant_jois: because time zone data CHANGED last year perhaps.... |
| 18:33.54 | Corydon76-dig | prashant_jois: restart Asterisk |
| 18:34.14 | prashant_jois | [TK]D-Fender: I don't understand what you mean by that. Is Asterisk not up to date on the new time zone data? |
| 18:34.34 | prashant_jois | [TK]D-Fender: I'm using 1.4.20 |
| 18:34.41 | Corydon76-dig | prashant_jois: Asterisk caches the timezone file |
| 18:34.48 | Qwell | Corydon76-dig: Is there some callback we can trigger to get notified when that changes? |
| 18:35.01 | Corydon76-dig | Qwell: yes, starting in 1.6.2 |
| 18:35.04 | Qwell | ahh, sweet |
| 18:35.11 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 18:35.15 | Corydon76-dig | That was the inotify stuff |
| 18:35.20 | KavanS | anyone have a suggestion to make certain area codes dial out via certain trunk? |
| 18:35.24 | *** part/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 18:35.41 | prashant_jois | Corydon76-dig: How do I get it to the correct time then? |
| 18:35.42 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:35.48 | Corydon76-dig | KavanS: it's called LCR |
| 18:35.53 | prashant_jois | Corydon76-dig: Is there a way without stopping and restarting? |
| 18:35.55 | Corydon76-dig | prashant_jois: restart Asterisk |
| 18:35.59 | Corydon76-dig | prashant_jois: nope |
| 18:36.07 | prashant_jois | Corydon76-dig: Okay thanks |
| 18:36.28 | Qwell | Corydon76-dig: I had wondered what inotify was for... |
| 18:36.33 | [TK]D-Fender | KavanS: its your dialplan, you have to put the separation logic in yourself. |
| 18:36.37 | Qwell | s/ for/ |
| 18:37.17 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 18:37.27 | Corydon76-dig | prashant_jois: actually, there is a way, but it's going to be a lot simpler simply to restart |
| 18:38.04 | prashant_jois | Corydon76-dig: Yeah I'll just bite the bullet and restart it |
| 18:38.10 | prashant_jois | Corydon76-dig: Thanks for your help |
| 18:53.51 | putnopvut | Qwell: the inotify stuff that Corydon76-dig has been writing is pretty cool. It seems really useful in pbx_spool |
| 18:54.25 | *** join/#asterisk BlargMaN00 (n=blargman@12.234.16.130) |
| 18:54.40 | Corydon76-dig | You mean like for quitting the polling of a directory status? |
| 18:54.56 | Corydon76-dig | should save a ton of cpu time |
| 18:55.05 | putnopvut | Yes, that's what seems really attractive about it. |
| 18:55.42 | Qwell | so it just keeps track of certain directories? |
| 18:55.59 | Corydon76-dig | No, the kernel does |
| 18:56.35 | Corydon76-dig | Kernel sends an event when the directory changes |
| 18:57.20 | Corydon76-dig | and only events that we specify |
| 19:00.50 | *** join/#asterisk chandoo (n=chandoo@ool-4353bb46.dyn.optonline.net) |
| 19:00.54 | chandoo | hi :) |
| 19:01.16 | *** join/#asterisk f00bar80 (n=f00bar80@41.234.161.82) |
| 19:01.36 | chandoo | I am looking for voip softphone, can some one advice one |
| 19:01.50 | chandoo | looking for best free voip softphone |
| 19:02.12 | ZX81 | huntsville = 2:01pm? |
| 19:02.19 | Qwell | ZX81: yes |
| 19:02.24 | ZX81 | sweet |
| 19:02.27 | *** join/#asterisk maszlo (n=reckenro@65.223.240.146) |
| 19:02.32 | ZX81 | chandoo: Zoiper |
| 19:02.35 | ZX81 | ~zoiper |
| 19:02.36 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 19:03.22 | Corydon76-dig | not to be confused with lutefisk |
| 19:03.23 | ZX81 | :) |
| 19:03.25 | bougyman | wow, i thought zoiper was the worst. |
| 19:03.34 | Corydon76-dig | which is the best type of fish preserved with lye |
| 19:03.35 | vncsnvs | !faltam |
| 19:03.38 | bougyman | i tried every open source and demo there was. |
| 19:03.39 | ZX81 | Corydon76-dig: yeah inotify rocks - we've started using it instead of cron - well incron |
| 19:03.45 | [TK]D-Fender | chandoo: Ekiga |
| 19:03.47 | [TK]D-Fender | ~ekiga |
| 19:03.48 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 19:03.48 | bougyman | (of soft phones, taht is) |
| 19:03.53 | f00bar80 | anyone aware with magicjack, i have difficulty for the other side to hear me clearly when placing a call, i tried to switch to other phone and adsl router instead of usb adsl modem, i know maybe i'm off-topic but i'm asking someone to help me or guide me to where i can get a solution for this problem as i already contacted the support and they couldn't help |
| 19:04.01 | bougyman | ekiga crashed like mad on me. |
| 19:04.09 | ZX81 | same |
| 19:04.12 | bougyman | wengophone was the only non-oss that seemed ok. |
| 19:04.19 | bougyman | but it's really a mess in its release cycle now. |
| 19:04.28 | ZX81 | zoiper crashes once every few days - with a huge list of accounts |
| 19:04.33 | bougyman | the x-lite one is ok. |
| 19:04.34 | ZX81 | eyebeam/xten is slow |
| 19:04.37 | ZX81 | :) |
| 19:04.40 | bougyman | twinkle has been really great for me. |
| 19:04.43 | bougyman | but it's linux only. |
| 19:04.44 | ZX81 | way too slow for me |
| 19:04.45 | [TK]D-Fender | bougyman: I'm sensing a common trait amongst your problems. |
| 19:04.48 | f00bar80 | i'm on windows xp prof and 256K ADSL connection , effective speed 212kbps/48kbps |
| 19:05.02 | ZX81 | 48kbps? |
| 19:05.02 | bougyman | i finally moved FS with portaudio controlled by StumpWM |
| 19:05.05 | bougyman | and i'm in heaven. |
| 19:05.07 | ZX81 | with which codec? |
| 19:05.10 | f00bar80 | any comment ? |
| 19:05.16 | bougyman | er moved to FS with portaudio and StumpWM controlling it. |
| 19:05.23 | ZX81 | f00bar80: what codec you using? |
| 19:05.26 | chandoo | downloaidng ekiga |
| 19:05.28 | f00bar80 | ZX81, what do you mean by which codec ? |
| 19:05.29 | [TK]D-Fender | f00bar80: waht does "difficulty" mean? |
| 19:05.38 | chandoo | looking at zoiper website |
| 19:05.43 | bougyman | [TK]D-Fender: what is that common trait? |
| 19:05.51 | [TK]D-Fender | bougyman: YOU :p |
| 19:05.51 | ZX81 | :) |
| 19:06.06 | f00bar80 | [TK]D-Fender, the other side can't hear me clearly |
| 19:06.06 | bougyman | oh, yeah, i'm picky. |
| 19:06.12 | ZX81 | f00bar80: ok |
| 19:06.13 | bougyman | i want it to be perfect, what's the matter with that? |
| 19:06.14 | ZX81 | look |
| 19:06.18 | ZX81 | you have 48kbps |
| 19:06.20 | ZX81 | to send with |
| 19:06.24 | ZX81 | 212 to receive |
| 19:06.26 | chandoo | thanks guyz |
| 19:06.37 | ZX81 | ulaw/alaw = ~80kbps |
| 19:06.42 | [TK]D-Fender | f00bar80: clearly means poor network conditions/bandwidth, generally. |
| 19:06.43 | ZX81 | so receive == fine |
| 19:06.45 | ZX81 | send == not |
| 19:06.47 | chandoo | I bought allvoi service, i downloded their softphone |
| 19:06.48 | ZX81 | but |
| 19:06.53 | ZX81 | you can use other codecs |
| 19:06.55 | [TK]D-Fender | f00bar80: Describe the call flow in DETAIL |
| 19:06.57 | chandoo | i am want to try something else too |
| 19:06.58 | ZX81 | ~codec |
| 19:07.06 | ZX81 | hmm |
| 19:07.09 | maszlo | I have been working on our system on the caller id.. or the lack there of. I was wondering If anyone could give me any pointers on where I should look, I have setup 'caller id superfecta' and if I goto the page directly with the same settings I applied in the callerid lookup source it works. The callerid source has be set for the inbound route, yet when I test it to the phone it shows phonenumber@server ip |
| 19:07.32 | ZX81 | jbot: codec is A codec is a device or program capable of encoding and/or decoding a digital data stream or signal. The word codec may be a combination of any of the following: 'compressor-decompressor', 'coder-decoder', or 'compression/decompression algorithm'. |
| 19:07.32 | jbot | ...but codec is already something else... |
| 19:07.44 | ZX81 | ~codec |
| 19:07.52 | ZX81 | ffs stupid bot |
| 19:07.58 | bougyman | it's just another StumpWM binding to me now, completely ubiquitous. when a call comes in my music pauses, same as when I make a call. when i hang up the music starts back up again. If i'm at my desk (via bluetooth presence on my Treo) it'll auto-answer, away from my desk it'll Follow me. handy all around. |
| 19:07.59 | ZX81 | :) |
| 19:08.13 | ZX81 | jbot: so what is codec then? |
| 19:08.15 | jbot | ZX81: what are you talking about? |
| 19:08.15 | ZX81 | :) |
| 19:08.18 | f00bar80 | ZX81, still can't get what do you mean codec ? |
| 19:08.38 | f00bar80 | ZX81, also what do you mean by ulaw/alaw ? |
| 19:08.46 | ZX81 | when you send data over a network you compress and uncompress it |
| 19:09.01 | ZX81 | your microphone and speakers normally talk 44.1/16 bit audio |
| 19:09.02 | [TK]D-Fender | f00bar80: voice data is ENCODED. |
| 19:09.04 | Corydon76-dig | jbot: codec? |
| 19:09.10 | ZX81 | this takes too much data |
| 19:09.10 | maszlo | the phones are cisco 7940 not sure if that could be the issue or not. is there a log for the caller id source? I don't see anything in /var/log/asterisk/full |
| 19:09.10 | [TK]D-Fender | f00bar80: like WAV & MP3. they are FORMATS |
| 19:09.28 | ZX81 | so it gets compressed (well and companded etc) |
| 19:09.43 | Corydon76-dig | jbot: no, codec is <reply>A codec is a device or program capable of encoding and/or decoding a digital data stream or signal. The word codec may be a combination of any of the following: 'compressor-decompressor', 'coder-decoder', or 'compression/decompression algorithm'. |
| 19:09.43 | jbot | okay, Corydon76-dig |
| 19:09.49 | Corydon76-dig | ~codec |
| 19:09.50 | jbot | A codec is a device or program capable of encoding and/or decoding a digital data stream or signal. The word codec may be a combination of any of the following: 'compressor-decompressor', 'coder-decoder', or 'compression/decompression algorithm'. |
| 19:09.54 | ZX81 | sweet |
| 19:09.55 | ZX81 | :) |
| 19:09.56 | Corydon76-dig | ZX81: better? |
| 19:10.00 | ZX81 | yaha |
| 19:10.01 | ZX81 | :) |
| 19:10.19 | Corydon76-dig | You just have to know how to talk to the bot |
| 19:10.44 | ZX81 | yeah, just wasn't sure about replacing something that wasn't showing up |
| 19:10.50 | ZX81 | ~segfault |
| 19:10.50 | jbot | extra, extra, read all about it, segfault is what asterisk does when it's not given the correct configuration! redo your config! |
| 19:10.50 | ZX81 | :) |
| 19:11.17 | ZX81 | hah |
| 19:11.21 | ZX81 | that's kinda funny |
| 19:11.25 | ZX81 | blame the config |
| 19:11.26 | ZX81 | :D |
| 19:11.34 | Corydon76-dig | ~botsnack |
| 19:11.34 | jbot | :), Corydon76-dig |
| 19:11.41 | ZX81 | :) |
| 19:11.57 | [TK]D-Fender | ~areyouadog ? |
| 19:11.58 | jbot | Bark! Bark! |
| 19:12.02 | [TK]D-Fender | jbot: Good boy! |
| 19:12.02 | jbot | :), [TK]D-Fender |
| 19:12.06 | RobH | ~fire |
| 19:12.07 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 19:12.22 | ZX81 | :D |
| 19:12.30 | ZX81 | heh: [asterisk-users] log to cdr each dialpan action, not only one record for each call |
| 19:12.36 | ZX81 | might take a while without murf |
| 19:12.37 | ZX81 | :) |
| 19:12.58 | *** join/#asterisk pbrunnen (n=pbrunnen@mail.aycanus.com) |
| 19:13.16 | *** part/#asterisk pbrunnen (n=pbrunnen@mail.aycanus.com) |
| 19:13.28 | f00bar80 | ZX81, :) still can't get what is problem , is it the upload rate ? why then it's already working fine on other same speed |
| 19:13.42 | f00bar80 | ZX81, but different machine |
| 19:13.52 | ZX81 | cpu then? |
| 19:13.52 | RobH | ~lobotomy ZX81 |
| 19:13.53 | jbot | ACTION pulls out a rusty saw to perform a lobotomy on ZX81 |
| 19:13.54 | ZX81 | microphone |
| 19:13.56 | ZX81 | :D |
| 19:13.57 | RobH | i amuse me |
| 19:14.07 | ZX81 | :D |
| 19:14.09 | chandoo | ekiga crashes on windows and makes windows hang |
| 19:14.13 | ZX81 | nice |
| 19:14.14 | ZX81 | :D |
| 19:14.23 | f00bar80 | ZX81, :( |
| 19:14.26 | ZX81 | good process isolation |
| 19:14.40 | ZX81 | f00bar80: try to use sound recorder maybe |
| 19:14.51 | f00bar80 | ZX81, then ? |
| 19:16.18 | f00bar80 | ZX81, could it be a sound Card problem or still we are talking about a network/Connection Speed problem ? |
| 19:16.23 | KavanS | who uses least cost routing here? |
| 19:16.29 | KavanS | i.e. has experience with it? |
| 19:16.56 | bougyman | < uses it |
| 19:17.09 | Corydon76-dig | I don't use it, but I do have experience with it |
| 19:17.20 | Corydon76-dig | 2 different things |
| 19:17.34 | [TK]D-Fender | KavanS: I look at the area code and pick a source based on it. Woo---hoo.... |
| 19:17.35 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:18.01 | KavanS | [TK]D-Fender: sweet! |
| 19:18.06 | Corydon76-dig | You can do LCR with func_odbc, for example |
| 19:18.13 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 19:18.25 | [TK]D-Fender | KavanS: But you see I've already answered this question for you earlier |
| 19:18.29 | bougyman | we look at the area code, find the cheapest rate area code we have available to make that call, set the caller_id to that area code, then make the call via the cheapest provider for that. |
| 19:18.35 | [TK]D-Fender | KavanS: YOU have to code this login YOURSELF in your DIALPLAN |
| 19:18.51 | [TK]D-Fender | s/login/logic/ |
| 19:19.26 | Corydon76-dig | and what solution you choose will depend upon your needs |
| 19:19.33 | f00bar80 | ZX81, any comment ? |
| 19:19.43 | Corydon76-dig | KavanS: you have any programmers who work with you? |
| 19:19.53 | perlypoo | agi with a cobol program to determine the route? |
| 19:20.08 | Corydon76-dig | KavanS: A programmer can whip up LCR in no time at all |
| 19:20.09 | KavanS | Corydon76-dig: lol are you saying I should forward this task to them? |
| 19:20.15 | KavanS | ahh ok... |
| 19:20.50 | Corydon76-dig | KavanS: and ask him for a database-oriented solution |
| 19:21.05 | *** join/#asterisk alerios (n=alerios@190.144.75.22) |
| 19:21.07 | Corydon76-dig | because a database is only way you're going to get this to scale |
| 19:21.24 | perlypoo | each possible provider has a list of area codes and rates. take area code, extract providers and rates, pick lowest one. |
| 19:21.36 | Corydon76-dig | A data analyst/programmer, even better |
| 19:22.08 | Corydon76-dig | but some area codes are different lengths than others |
| 19:22.17 | Corydon76-dig | You want the longest area code that is a match |
| 19:22.32 | Corydon76-dig | from each provider |
| 19:22.35 | KavanS | yeah, I was looking into LCDial...but from the sound of it, maybe I will ask our guys to look into something custom |
| 19:22.39 | perlypoo | wut? |
| 19:22.43 | Corydon76-dig | otherwise, you may be getting the wrong rate |
| 19:22.58 | Corydon76-dig | perlypoo: international calls |
| 19:23.07 | bougyman | oh weird. |
| 19:23.18 | perlypoo | hmm. i didn't think of that. |
| 19:23.19 | bougyman | i didn't know * didn't have an lcr module. |
| 19:23.36 | perlypoo | the expect of my "lcr" is just forwarding 800 numbers thru a free termination service, so heh. |
| 19:23.36 | Corydon76-dig | bougyman: it doesn't, because there are many different needs |
| 19:23.57 | bougyman | Corydon76-dig: that seems counterintuitive. |
| 19:24.08 | Corydon76-dig | I've written several different LCR solutions |
| 19:24.10 | bougyman | if there are many needs for it i'd suppose it'd be a needed component. |
| 19:24.19 | perlypoo | really it would take a programmer like 2 hours to whip something up |
| 19:24.25 | Corydon76-dig | Not all LCR works the same way |
| 19:24.29 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-23-21.w86-215.abo.wanadoo.fr) |
| 19:24.43 | Corydon76-dig | The most common is a table of prefixes and rates |
| 19:24.48 | bougyman | Corydon76-dig: not all _any feature_ works the same way. |
| 19:25.00 | bougyman | it can be flexible. |
| 19:25.02 | perlypoo | opportunity cost |
| 19:25.04 | Corydon76-dig | But some LCR is run with an OCR indirection table |
| 19:25.18 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-23-21.w86-215.abo.wanadoo.fr) |
| 19:25.48 | bougyman | so long as you can customize the db query it seems the user could choose from any sort algo or query path. |
| 19:25.56 | *** join/#asterisk leo66 (n=chatzill@201.47.187.177) |
| 19:26.00 | bougyman | that's how the lcr I use works. |
| 19:26.15 | bougyman | it has the standard table format but you can use custom sql if you desire. |
| 19:26.27 | perlypoo | what db? do you require one? sqlite? odbc? |
| 19:26.46 | [TK]D-Fender | bougyman: And thats "you". Others may require more specialized decisions based on complete #'s, time of day, etc |
| 19:26.55 | bougyman | [TK]D-Fender: it supports all of that already. |
| 19:26.57 | Qwell | bougyman: You could probably write an LCR thing in like 10 lines of Ruby.. |
| 19:27.02 | Corydon76-dig | and one even requested an override table |
| 19:27.11 | perlypoo | serio, or like 5000 lines of C++ |
| 19:27.22 | bougyman | [TK]D-Fender: and of course custom sql would allow for taht. |
| 19:27.27 | Corydon76-dig | Nah, 100 lines of C |
| 19:27.40 | perlypoo | Corydon76-dig: C would be nicer |
| 19:27.54 | perlypoo | don't forget your LCREntryFactoryFactory |
| 19:27.54 | [TK]D-Fender | Qwell: EMACS could do it in 1 line ;) |
| 19:27.57 | Corydon76-dig | The issue isn't the main code. It's all the exceptions to the rules, according to business logic |
| 19:28.16 | [TK]D-Fender | Qwell: http://xkcd.com/378/ |
| 19:28.23 | perlypoo | [TK]D-Fender: I'll bet my soul EMACS *already* has an LCR .el somewhere in the tubes |
| 19:28.23 | bougyman | that's what you put in the users hands, Corydon76-dig. |
| 19:28.31 | Corydon76-dig | The exceptions are an order of magnitude larger than the main LCR |
| 19:28.34 | bougyman | not an all-in-one solution, a pliable one. |
| 19:28.57 | Qwell | perlypoo: FactoryFactory? java? |
| 19:29.17 | Corydon76-dig | bougyman: ask yourself why in an open source project, nobody has contributed that solution yet |
| 19:29.24 | perlypoo | so once the lcr module is finished it will be so generic as to be effectively useless. to cover all the bases, the allow enough customization, it would basically just be a LCR() function that called a module with your own logic - so the same thing we have now |
| 19:29.32 | bougyman | Corydon76-dig: they have on the open source project i use for LCR. |
| 19:29.34 | *** join/#asterisk Forai (i=45f81b38@gateway/web/ajax/mibbit.com/x-e910f7783d4274b7) |
| 19:29.52 | jjshoe | Qwell 10 whole lines of ruby? I thought everything in ruby was four or less? |
| 19:30.03 | KavanS | lol |
| 19:30.04 | bougyman | which makes it more weird that * doesn't have it, Corydon76-dig. |
| 19:30.08 | Corydon76-dig | bougyman: so it's large enough to need its own project, to encapsulate all the needs? |
| 19:30.10 | Forai | Pardon my ignorance, where do I go to ask dumb trixbox questions? |
| 19:30.13 | KavanS | yeah I think I'll have our guys look into it... |
| 19:30.19 | Corydon76-dig | Forai: #trixbox |
| 19:30.24 | Forai | thank you |
| 19:30.31 | perlypoo | "Dialing from the Emacs BBDB address book with least-cost routing "http://www.math.ucdavis.edu/~mkoeppe/bbdb-isdnlog-estic/lcr.html |
| 19:30.34 | bougyman | Corydon76-dig: nossir, it's a different foss telephony platform. |
| 19:30.40 | perlypoo | hahas |
| 19:30.46 | Qwell | perlypoo: ... |
| 19:30.49 | bougyman | one with nowhere near the userbase or contributors of *, of course. |
| 19:30.52 | Qwell | You win the Internet. |
| 19:31.22 | bougyman | probably easier because of core support for ODBC and db's |
| 19:31.37 | Corydon76-dig | probably |
| 19:31.38 | *** join/#asterisk javb (n=javb@tdev212-139.codetel.net.do) |
| 19:31.55 | perlypoo | speaking of features i wish asterisk had |
| 19:32.13 | perlypoo | i wish there were a way to specify a filter for a vmail file before attaching it to an email |
| 19:32.23 | f00bar80 | [TK]D-Fender, any comment ? what could be the problem or what i have to do ? |
| 19:32.26 | perlypoo | so i can convert it to mp3 |
| 19:32.28 | javb | I have two PBXs connected via a SIP TRUNK, the call between them is ok, but it hangs up automatically exactly 20 seconds after the chan is up. Any idea? |
| 19:32.41 | jjshoe | perlypoo write your own vmail app :) |
| 19:32.52 | perlypoo | i have given it some thought |
| 19:33.10 | perlypoo | the main thing holding me back is that i'd have to record custom prompts, and that makes me ugh |
| 19:33.12 | jjshoe | perlypoo at one point I had an app_voicemail doing calls to notify you of new vm. dunno what I ever did with the source though. |
| 19:33.32 | Qwell | jjshoe: app_voicemail can call a script when a new message arrives |
| 19:33.35 | jjshoe | perlypoo why you? alison is a cheap date. |
| 19:33.55 | perlypoo | it can? |
| 19:34.01 | jjshoe | Qwell yeah I've seen some dial plan call back solutions... they where quite interesting |
| 19:34.03 | perlypoo | thinks |
| 19:34.12 | [TK]D-Fender | f00bar80: You seem to have comprehension issues with such basics as to what a CODEC is. Sorry, but I'm not currently up to fighting for this right now. |
| 19:34.18 | perlypoo | that would be nice, just a python script and convert and send it off. |
| 19:34.22 | Qwell | perlypoo: externnotify= |
| 19:34.29 | Qwell | see voicemail.conf.sample |
| 19:34.57 | f00bar80 | [TK]D-Fender, is it a codec problem ? |
| 19:35.02 | Forai | actually, does anyone have a good explanation to trunking? |
| 19:35.07 | Qwell | ~trunk |
| 19:35.08 | jbot | [trunk] a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 19:35.31 | [TK]D-Fender | f00bar80: You have not provided any useful information to aid in debugging your issue |
| 19:35.31 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 19:35.52 | perlypoo | that is awesome. |
| 19:35.58 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 19:36.05 | perlypoo | yay another project for me |
| 19:36.08 | Forai | ok then |
| 19:36.42 | perlypoo | ; in a car, a trunk carries bodies |
| 19:37.22 | Titanous | I've got a remote SIP phone on a network connected via openvpn to asterisk, SIP works great, but no audio is transmitted. a rtp debug shows no packets going to the remote phone, just 'Sent RTP P2P packet to 0.0.0.0:42502 (type 00, len 000160)' |
| 19:37.33 | Titanous | HOw do I fix this? |
| 19:38.11 | perlypoo | the 0.0.0.0 looks suspect to me |
| 19:38.21 | f00bar80 | [TK]D-Fender, like ? |
| 19:39.11 | Titanous | perlypoo: that's what I think too, but what's the cause? |
| 19:39.32 | [TK]D-Fender | f00bar80: no troubled call with SIP/other debug enabled, no configs, no description of other settings attempted. No comparison to other services in the sam network scenario. |
| 19:42.43 | f00bar80 | [TK]D-Fender, configs and settings for what , magicjack is just a usb which's plugged and that's it , do you know about it ?> |
| 19:43.24 | [TK]D-Fender | f00bar80: What does this have to do with *? |
| 19:45.37 | perlypoo | hmm |
| 19:47.05 | perlypoo | all of a sudden call quality on my tdm400p has degraded - calls into it seem quiet and there seem to be a lot of dropping out, almost like the caller were on a cell phone going thru a tunnel, but they aren't |
| 19:47.09 | f00bar80 | [TK]D-Fender, i'm using magicjack to place calls , but don't know how it works , that's regarding your talking about configs and settings |
| 19:47.10 | perlypoo | what could cause this? |
| 19:47.27 | perlypoo | (nothing regarding the configuration of this has changed) |
| 19:47.40 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 19:47.44 | areay | i'm using a sip trunk with asterisk... incoming calls work perfectly, but i can't dial out... i checked the sip debug and it says "Found no matching peer or user for '192.168.1.100:5078'". the ip in question is the client (ekiga softphone)... in my users.conf, under [6000] (the extension in question), i have specified the 'host' and 'type' attributes as the ip address and "user" respectively, but still I'm getting the same error... how |
| 19:47.44 | areay | do i setup a peer or user using users.conf? |
| 19:47.45 | [TK]D-Fender | f00bar80: This isn't #magicjack , and it is not supported here. |
| 19:48.23 | perlypoo | no |
| 19:48.50 | perlypoo | type=friend should be under [6000] |
| 19:49.46 | areay | perlypoo, ah ok i'll try that... is that all i need to do to get rid of that error? |
| 19:50.17 | chandoo | need some help in configuring sip software, i have allvoi account |
| 19:50.25 | chandoo | i downloaded ivm software |
| 19:50.52 | chandoo | it is asking for Server(Proxy and Domain) |
| 19:51.03 | chandoo | is it SIP Server Realm |
| 19:51.17 | chandoo | i can see settings in allVOI softphone |
| 19:52.21 | leo66 | can someone tell me if its possible to set up a digivoice card in a asterisknow installation? |
| 19:52.33 | *** part/#asterisk alerios (n=alerios@190.144.75.22) |
| 19:53.16 | perlypoo | see voip-info's page on sip.conf or sip.conf.sample |
| 19:53.41 | chandoo | perlypoo: is it for me |
| 19:54.22 | Titanous | Has anyone ever seen 'Sent RTP P2P packet to 0.0.0.0'? |
| 19:54.49 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:55.40 | areay | perlypoo, i changed it to "friend" and i'm still getting "Found no matching peer or user for '192.168.1.100:5078'" in the sip debug... there are a couple other errors too, should i pastebin it? |
| 19:58.29 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 19:59.56 | chandoo | what is STUN server |
| 20:00.01 | perlypoo | the problem is most likely due to a misconfiguration in ekiga |
| 20:00.10 | chandoo | it is failing or setup incorrectly |
| 20:00.29 | chandoo | can some guru help me in setting up this piece of software |
| 20:00.32 | f00bar80 | [TK]D-Fender, so point me to where i have to get a help, at the time i mentioned at the start of my talk , that the support couldn't help, and i'm here as magicjack is using voip , that's it and sorry again for being off-topic, but i can't get help anywhere else than here as you do have a good voip experience |
| 20:01.37 | perlypoo | f00bar80: this channel is so totally the wrong place to ask about this |
| 20:02.00 | madgeek | "i can't get help anywhere else" doesn't really justify hijacking some unrelated channel |
| 20:02.30 | f00bar80 | perlypoo, so which channel do you suggest ? |
| 20:02.53 | madgeek | as far as most ppl here are concerned, that's the equivalent of coming into their home and fucking their wife in the ass RIGHT IN FRONT OF THEM |
| 20:03.01 | areay | perlypoo, there's only like 4 or 5 settings to change in ekiga... i looked further into the sip debug and it's saying 404 not found... |
| 20:03.11 | areay | i'm pretty sure ekiga's setup right |
| 20:03.53 | apeiron | f00bar80, Are you the same f00bar80 who comes on #perl asking for help with things that indicate you don't read error messages given to you/ |
| 20:05.03 | madgeek | oh snap, it's on |
| 20:05.07 | perlypoo | f00bar80: the world does not owe you anything. you've been in here quite a while and still i've not seen any information regarding your problem. just "blah blah blah, what sounds like an amusement park name, blah blah, doesn't work, amusement park name" |
| 20:05.25 | areay | lol |
| 20:05.31 | madgeek | someone done got told |
| 20:06.15 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 20:07.07 | nullable_type | Hey guys is there any reason why ${DIALSTATUS} will not work? I am trying to timestamp when a call is anwered.... |
| 20:07.46 | [TK]D-Fender | f00bar80: If audio is bad, then your network conditions are bad or their service sucks. |
| 20:08.11 | [TK]D-Fender | nullable_type: Show us the problem and maybe we can do something about it. |
| 20:08.52 | chandoo | can some one tell me how to find out voip settings for AllVoi.com |
| 20:09.33 | nullable_type | exten => s,1,Dial(SIP/777777@gateway) |
| 20:09.34 | nullable_type | exten => s,n,Goto(s-${DIALSTATUS},1} |
| 20:09.34 | nullable_type | exten => s-NOANSWER,1,NoOp(NoAnswer for Call) |
| 20:09.34 | nullable_type | exten => _s-.,1,NoOp(Some Other DialStatus: ${DIALSTATUS}) |
| 20:09.35 | chandoo | i am triying to use other voip software other than Allvoi softphone |
| 20:09.53 | nullable_type | D-Fender >> The above messag is for you. The problem with DIALSTATUS not working |
| 20:09.54 | f00bar80 | [TK]D-Fender, may even tell me how this magicjack depends on the network conditions , from the point of view of voip ? |
| 20:10.20 | perlypoo | i believe that s,n, will only be reached if the call fails |
| 20:10.40 | nullable_type | oh ya |
| 20:11.19 | nullable_type | so is it possible to catch it when the destination answeres? |
| 20:11.22 | perlypoo | you can try h => but i dont think DIALSTATUS is available in that context |
| 20:11.23 | [TK]D-Fender | f00bar80: Bandwidth, Jitter, and Packet-loss |
| 20:11.37 | [TK]D-Fender | nullable_type: PASteBIN |
| 20:11.50 | [TK]D-Fender | nullable_type: Show the code AND the CLI output of the failure |
| 20:11.52 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 20:11.57 | f00bar80 | [TK]D-Fender, some more brief details ? |
| 20:12.07 | [TK]D-Fender | f00bar80: there are no more details. |
| 20:12.09 | nullable_type | peryl >> I have h too but i want exact TimeStamp when destination Answered |
| 20:12.36 | perlypoo | Hmm. |
| 20:12.54 | perlypoo | Well, I believe the CDR information would tell you that. What are you doing with this information? |
| 20:12.59 | [TK]D-Fender | nullable_type: Ah.. I see a problem... |
| 20:13.21 | [TK]D-Fender | nullable_type: the variable works fine. your next priority never gets EXECUTED if the call is answered |
| 20:13.34 | nullable_type | peryl >> Just for statistics, I am recording in a remote database. |
| 20:13.51 | [TK]D-Fender | nullable_type: You need to sue a combination of the "g" dial option, and the "h" standard extension. |
| 20:14.10 | nullable_type | ok i will look wiki for g extension |
| 20:14.10 | [TK]D-Fender | nullable_type: lAnd indeed you are reinventing the whhel. This is what CDR is for. |
| 20:14.33 | nullable_type | i c |
| 20:14.43 | nullable_type | thanks guys |
| 20:14.54 | perlypoo | < is a piece of poo, not a guy |
| 20:14.56 | perlypoo | but yw |
| 20:15.48 | nullable_type | lol, alrity |
| 20:17.36 | Titanous | I've got a remote SIP phone on a network connected via openvpn to asterisk, SIP works great, but no audio is transmitted or received. a rtp debug shows no packets going to the remote phone, just 'Sent RTP P2P packet to 0.0.0.0:42502 (type 00, len 000160)' |
| 20:17.47 | Titanous | What is the problem? |
| 20:18.56 | bougyman | the RTP ip address is wrong. |
| 20:19.04 | bougyman | i bet it's going back out asterisk's non-vpn interface. |
| 20:19.21 | bougyman | tcpdump your interfaces, i bet sip is going over your tap or tun device but the rtp is using another. |
| 20:19.45 | bougyman | s/tcpdump/your_favorite_sniffer/ |
| 20:21.19 | *** part/#asterisk Chex (i=chex@random.supermario.org) |
| 20:22.04 | perlypoo | sounds like the signature of a stalker on an anonymouse valentine |
| 20:22.14 | perlypoo | -e |
| 20:22.52 | madgeek | ha |
| 20:28.31 | *** join/#asterisk rashed2020_ (n=shabati@67.205.245.208) |
| 20:28.32 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 20:28.46 | rashed2020_ | What are devices like the SPA3000 called? |
| 20:28.56 | rashed2020_ | FXO gateways? |
| 20:29.46 | perlypoo | the documentation of externnotify is kinda lacking. the wiki says it gets called with options but the voicemail.conf.sample does not say? |
| 20:29.56 | *** part/#asterisk vncsnvs (n=vncsnvs_@201.86.135.130.dynamic.adsl.gvt.net.br) |
| 20:30.38 | [TK]D-Fender | BBIAb |
| 20:31.18 | *** part/#asterisk maszlo (n=reckenro@65.223.240.146) |
| 20:33.19 | *** join/#asterisk pwnedulongtime (n=dresdean@rrcs-72-43-106-2.nyc.biz.rr.com) |
| 20:33.26 | pwnedulongtime | hello? |
| 20:34.14 | Kate-o | Hello everyone |
| 20:34.38 | pwnedulongtime | I'm using the function Dial with option W, anyone know the default location of the recording file? |
| 20:38.04 | kannan | hello, i am using asterisk manager api to to an originate to call an internal phone first and then a number thru zap. sometimes there is no one to answer the internal phone. I want to still call the zap chanel and stream a file. I am not sure hw to do this , any ideas please? |
| 20:43.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 20:44.08 | Kate-o | I'm having problems getting the modem to be on hook, currently it stays off hook until we unplug it and then plug it back in. |
| 20:44.45 | Kate-o | Is that something we can change in configuration? I'm not quite sure why the default would be set to always be off hook |
| 20:45.36 | *** join/#asterisk HoverHell (n=hell@91.146.50.221) |
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| 21:09.55 | areay | i can receive incoming calls through asterisk, but i can't make outgoing ones... here's my sip debug info: http://paste.ubuntu.com/130337/ |
| 21:14.44 | *** join/#asterisk aenaus (n=hdgfghf@91.140.106.239) |
| 21:16.24 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
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| 21:19.39 | areay | why would i be able to *receive* calls, but not make them? i'm using the same sip trunk for both and my dialplans are set up right |
| 21:21.46 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
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| 21:29.21 | dverzolla | has anyone using asterisk in solaris with CMT? |
| 21:29.25 | dverzolla | SUN with CMT! |
| 21:33.05 | pdmmm | does it compile ? |
| 21:38.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:38.40 | *** join/#asterisk SparFux (n=raoul@e182025114.adsl.alicedsl.de) |
| 21:43.30 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 21:43.48 | edibrac | is /var/log/asterisk/messages typically owned by root or by the asterisk user? |
| 21:44.16 | edibrac | by default my logrotate has been thinking it's asterisk, but i haven't created that user yet. I'm guessing non-root is "best practice" |
| 21:44.21 | edibrac | similar to apache |
| 21:45.09 | thehar | hasn't Monitor() been replaced with MixMonitor() ? |
| 21:46.13 | edibrac | then again, should the asterisk process be running under the root user? or do most people create a separate user for it? |
| 21:47.26 | pwnedulongtime | I'm using the function Dial with option W, anyone know the default location of the recording file? |
| 21:47.53 | mmlj4 | in /etc/asterisk, probably |
| 21:48.03 | mmlj4 | or wait, recording file? |
| 21:48.06 | pwnedulongtime | actually, I don't think anything is being recorded anyway |
| 21:48.26 | pwnedulongtime | Dial("SIP", "222","15","HgWD(*1)"); |
| 21:48.31 | pwnedulongtime | should that work? |
| 21:49.36 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 21:51.41 | edibrac | hye pwnedulongtime can u do me a favor? |
| 21:51.47 | edibrac | what user does your asterisk run as? |
| 21:51.54 | edibrac | just ps aux | grep asterisk |
| 21:52.49 | pwnedulongtime | k one sec |
| 21:53.18 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 21:53.27 | Titanous | I'm having some really weird issues with RTP over vpn, full details here: http://pastie.org/414649 (description, tcpdump, SIP Debug) |
| 21:53.45 | *** join/#asterisk neverblue (n=jezus@unaffiliated/neverblue) |
| 21:54.01 | pwnedulongtime | edibrac: root 17044 0.0 0.2 5032 1036 ? S Mar10 0:00 /bin/sh /usr/sbin/safe_asterisk |
| 21:54.02 | pwnedulongtime | asterisk 17045 0.4 1.8 26748 9392 ? Sl Mar10 12:30 /usr/sbin/asterisk -v -g -p -U asterisk -G asterisk |
| 21:55.47 | neverblue | i recently purchased Polycom Soundpoint 330 phones, I am having issues with dialing numbers. For example: when I attempt to dial a number, with 10 digits, I need to add a '9', for an ouside line, which means I press 11 digits. I have to enter the number, then lift the receiver. |
| 21:56.04 | Qwell | neverblue: the phone has a dialplan as well |
| 21:56.12 | neverblue | is there something in the dialplan I have to change to not have this happen or within the phone? |
| 21:57.53 | ayeso | how many SIP g711u calls can a T1 support? |
| 21:58.01 | neverblue | so I have to adjust the current dial-plan(which is setup for Grandstream phones), to adjust for the Polycom ? |
| 21:58.08 | Mog | 20ish ayeso |
| 21:58.17 | Mog | its 80kbps per call |
| 21:58.27 | Titanous | Can anyone help me? |
| 21:58.29 | Mog | you have 1.54 megabits |
| 22:00.08 | neverblue | so dialplans are specific to which phone type is used? |
| 22:00.22 | neverblue | is there examples of dialplans for each 'type' of phone? |
| 22:00.33 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 22:02.02 | KavanS | ohh snap! |
| 22:02.50 | pwnedulongtime | edibrac? |
| 22:03.26 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 22:03.40 | edibrac | oh that was for my question to you, not to really help your problem :) |
| 22:04.17 | edibrac | i'm not starting my asterisk with safe_asterisk - so my /var/log/asterisk/messages is owned by root and that breaks my logrotate |
| 22:06.06 | *** join/#asterisk luca`gervasi (n=ashura@host29-123-dynamic.59-82-r.retail.telecomitalia.it) |
| 22:06.08 | luca`gervasi | hallo |
| 22:06.11 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 22:06.43 | Ashetic1 | I have my asterisk behind nat, do you know the ports i need to forward to it ? |
| 22:07.25 | pwnedulongtime | is Set(DYNAMIC_FEATURES=automon) set by default? |
| 22:08.55 | jjshoe | Qwell ping |
| 22:09.02 | jjshoe | Qwell fonality is now charging for tbpro SE |
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| 22:19.11 | Kate-o | [TK]D-Fender: Hey, you around anywhere? |
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| 22:23.42 | fbnts | Hi, We're having a NAT issue (well thats what I presume it is) we have asterisk running, not behind nat. We have a SIP Phone which is behind a nat router. For that sip peer i have set nat=yes and qualify=yes |
| 22:24.16 | fbnts | The phone can call out with audio working both ways but calls to the sip peer fail with circuit-busy |
| 22:27.32 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:27.32 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:39.48 | DavidR2008 | is it possible for the callee to execute a macro during the call? if so is there a very basic example? |
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| 23:30.57 | riddlebox | pdmmm, you around? |
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| 23:31.40 | riddlebox | I was at a client today and they had a cisco adaptor for the phone, so I have tried to do this, but the upgrade seems to fail |
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| 23:37.53 | riddlebox | hrmm has anyone got any experience upgrading the firmware on cisco phones to SIP? |
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| 23:51.49 | alerios | Hi all. I can't make DISA recognize my passwords file.. it says Mailbox is null, but it seems to me that the file is ok: http://paste.debian.net/30469/ |
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| 23:55.51 | Merlin | i just realized that iaxmodem actually uses spandsp. is there any opensource analog/digital fax conversion software that doesn't use spandsp? |
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| 23:56.46 | coppice | nope |
| 23:57.07 | jaytee | can't think of one but since you have a Wizard's nickname can't you just conjure something up? |
| 23:57.09 | coppice | why would you want something that doesn't use spandsp? |
| 23:57.27 | jaytee | yeah! what's wrong with spandsp? :-) |
| 23:58.17 | docid | so heres a question....when i pick up a line, and dial through asterisk, same number dialed, same line dialing from, about 30% of the time it just hangs up, the rest of the time it routes it correctly,.....ummm....is this per chance a known issue? |
| 23:59.27 | Merlin | because spandsp sucks |
| 23:59.58 | Merlin | i should rewrite it |