| 00:01.52 | *** join/#asterisk FuriousGeorge (n=Magda@ool-43536ea8.dyn.optonline.net) |
| 00:02.05 | FuriousGeorge | anyone using asterlink for termination right now? |
| 00:05.03 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:05.46 | flenders | kimosabe: sorry was on the phone |
| 00:06.34 | flenders | kimosabe: it can be tricky, but mine worked following tutorials I found online |
| 00:07.15 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:17.59 | *** join/#asterisk koss- (i=koss@adsl-75-36-80-214.dsl.bcvloh.sbcglobal.net) |
| 00:18.09 | koss- | anyone know where i can get polycom 2.1.2 firmware? |
| 00:18.46 | saftsack | kimosabe: try to find out if you can set a timeout |
| 00:21.39 | *** join/#asterisk zutoo (n=j@74.94.57.214) |
| 00:22.13 | zutoo | Can anyone help me troubleshoot an incoming call issue? |
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| 00:39.44 | r0d3nt | viva DefCon bitchez. |
| 00:39.49 | r0d3nt | see you @ toorcon and HOPE. |
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| 00:52.11 | netvulture | has something changed with realtime sip in 1.4 branch?? I can't get a peer to register no matter what I have in the db row. Keeps saying 'No matching peer found', yet I can see the query is correct and does return 1 row |
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| 01:17.36 | zutoo | Just finished setting up a new box. Everything seems to be working right except i cant hear my ivr menus or moh... any suggestions? |
| 01:17.56 | *** join/#asterisk ukris (n=ukris@aa20060807547d355914.userreverse.dion.ne.jp) |
| 01:22.11 | chendy | hi folks |
| 01:23.39 | chendy | what can i do, to play a notice to the caller waited in the queue while the agent picked up the phone? |
| 01:24.18 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 01:26.47 | tengulre | how to config a CallCenter with asterisk box? an asterisk card + 20 users(IAX2/SIP). |
| 01:27.14 | JT | read the goddamned book |
| 01:27.16 | JT | ~thebook |
| 01:27.17 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:27.19 | JT | or hire a consultant |
| 01:27.21 | JT | ~hafc |
| 01:27.22 | jbot | somebody said hafc was hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 01:30.00 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:31.38 | phix | hi |
| 01:31.53 | phix | I am having issues, the volume of a conversion will dip down very low after a few minutes of conversation, what would cause this? |
| 01:32.01 | phix | Incomming calls are either from SIP or a POTS ZAP channel |
| 01:36.09 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 01:37.03 | phix | hi |
| 01:40.39 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 01:41.48 | phix | hmmmm, wake me up with some suggestions on what I can do to try and debug this problem |
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| 01:58.23 | tengulre | my mean is I don't like agent. I want PSTN ---> ZapChannel--->Queues---->IAX2 peers. is right? |
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| 02:01.04 | FuriousGeorge | when i take an inboud call and route it out to the pstn via voip it sounds terrible. no real way around that is there? |
| 02:02.05 | *** join/#asterisk minkus (n=minkus@pool-72-84-49-162.clrkwv.east.verizon.net) |
| 02:02.09 | FuriousGeorge | iow, pstn-->sip provider-->asterisk->iax provider->pstn |
| 02:02.53 | FuriousGeorge | from asterisk to pstn via iax2 sounds fine |
| 02:03.01 | FuriousGeorge | i think my sip provider is not so good |
| 02:08.24 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 02:15.28 | WilliamK | FG, what codec are you using on the SIP provider? |
| 02:16.47 | WilliamK | and is the sip provider one of the more reconized names or a tiny vendor? |
| 02:22.48 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 02:28.04 | zutoo | I have an interesting issue |
| 02:28.22 | zutoo | Anyone wanna help me try to figure it out? |
| 02:28.52 | zutoo | After hours of troubleshooting I finally know what the problem is, i am just not sure how to fix it |
| 02:33.00 | JT | ~ask |
| 02:33.00 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:34.41 | zutoo | If I allow ulaw, alaw, g729, g723, and gsm in sip.conf incoming works but not outgoing, if i take them out of i have the opposite problem |
| 02:35.44 | zutoo | any ideas?? |
| 02:36.09 | JT | you're connecting what to what? |
| 02:36.43 | zutoo | im using sip trunks and phones |
| 02:36.49 | *** join/#asterisk JunK-Y (n=junky@modemcable223.205-56-74.mc.videotron.ca) |
| 02:37.08 | zutoo | provider is vonage business plus |
| 02:37.18 | JT | :o |
| 02:37.48 | JT | btw sip calls aren't trunks |
| 02:38.30 | JT | in sip.conf, canreinvite=no, and don't allow g.723, and don't allow g.729 unless you have licenses for it or are doing pure passthrough |
| 02:39.17 | zutoo | u actually need licenses for them!? |
| 02:39.28 | JT | obviously, they're patented |
| 02:39.37 | JT | g.723 has no codec available for asterisk |
| 02:39.47 | JT | g.729 you can buy licenses from digium |
| 02:40.49 | *** join/#asterisk keith4_ (n=ladmin@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 02:41.10 | hmmhesays | you can passthru g.723 can't you? |
| 02:41.37 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:42.03 | keith4_ | me entire network is NAT'd, including my asterisk server, how can I connect SIP clients outside the NAT without forwarding 10,000 UDP ports through the NAT? |
| 02:42.12 | keith4_ | uh.... me=my |
| 02:42.22 | JT | if he has a g.723 endpoint and never uses playback or anything |
| 02:42.37 | JT | keith4_: just forward the ports |
| 02:42.44 | hmmhesays | that would be passthru yes |
| 02:43.08 | keith4_ | no other way? what about VPN? |
| 02:43.08 | JT | yes |
| 02:43.14 | JT | keith4_: why would you? |
| 02:43.20 | keith4_ | i dunno |
| 02:43.50 | JT | so you're wanting a non-standard setup for no good reason? |
| 02:44.14 | keith4_ | seems that way, yup |
| 02:44.17 | zutoo | ok... i added canreinvite=no and took out the allows in sip.conf |
| 02:44.24 | keith4_ | so let's start with the standard setup, then |
| 02:44.28 | keith4_ | what other ports do I need besides tcp 5060? |
| 02:44.33 | zutoo | now outgoing works, but not incoming |
| 02:44.49 | JT | keith4_: udp 5060, 10000-20000 |
| 02:45.05 | JT | zutoo: there are some allow= lines, right? |
| 02:45.13 | zutoo | incoming rings, but when i pickup i just get a fast busy signal |
| 02:45.44 | zutoo | not in the general section, only under the trunks |
| 02:45.59 | koss- | anyone -- any ideas why being behind a NAT router with no additional traffic would cause VoIP to be choppy? |
| 02:45.59 | JT | zutoo: doesn't sound like you're using asterisk |
| 02:46.36 | zutoo | why do u say that? |
| 02:46.46 | JT | there shouldn't be a trunks section |
| 02:47.26 | zutoo | doesnt each trunk get its own section? |
| 02:47.52 | JT | yes but they're not trunks |
| 02:48.04 | keith4_ | JT: so no tcp ports at all? |
| 02:48.09 | JT | keith4_: nope |
| 02:48.59 | zutoo | oh |
| 02:49.35 | zutoo | im pretty new to voip |
| 02:49.45 | JT | zutoo: what phones are you using? |
| 02:50.20 | zutoo | i use freepbx and they call them trunks, so i thought they were trunks |
| 02:50.41 | zutoo | gxp-2000's, bt-101's, and a pap2t |
| 02:50.50 | JT | ugh |
| 02:50.54 | JT | grandstream |
| 02:51.09 | JT | zutoo: like i suspected, you're not using asterisk, you're using freepbx |
| 02:51.16 | JT | ~freepbx |
| 02:51.17 | jbot | it has been said that freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:51.38 | zutoo | the gpx isnt bad. the bt-101's are garbage! |
| 02:51.55 | JT | the GXPs are still rubbish |
| 02:51.56 | JT | ~gs |
| 02:51.57 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 02:52.22 | zutoo | sorry, i was in #freepbx all day and was not able to get any help |
| 02:53.01 | zutoo | incoming and outgoing both seem to be working now, so you were extremely helpful |
| 02:53.04 | JT | that sounds normal |
| 02:53.25 | JT | they release a product that is hard to support, then they don't bother to support it themselves |
| 02:54.14 | zutoo | once its running right it is very easy to maintain, its just a pain setting up properly |
| 02:54.37 | JT | how did you get incoming and outgoing to work? |
| 02:54.53 | JT | so is asterisk, except asterisk is easier to get running properly |
| 02:55.21 | *** part/#asterisk FuriousGeorge (n=Magda@ool-43536ea8.dyn.optonline.net) |
| 02:55.48 | keith4_ | am I going to have trouble connecting a sip client from work through a firewall? |
| 02:56.01 | keith4_ | .... that sounded better in my head |
| 02:56.23 | zutoo | i had all the allows i mentioned above in the general section of sip.conf, but they were also listed under each "trunk" so i had to remove g729 and g723 from both places |
| 02:56.32 | keith4_ | the firewall nazis at work don't allow any inbound connections, is this going to break sip from my office to my home? |
| 02:56.44 | zutoo | btw: what is the proper term for what i keep calling a trunk? |
| 02:56.54 | JT | keith4_: it's an outbound connection |
| 02:57.01 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:57.20 | keith4_ | it doesn't require a reciprocal udp connection coming back? |
| 02:57.50 | JT | well yes, but i have no idea if the firewall will allow it |
| 02:58.00 | keith4_ | hmmm |
| 02:58.04 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 02:58.05 | keith4_ | well, i'll test it tomorrow |
| 03:00.27 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:00.30 | zutoo | JT: thank you.. allowing g729 and g723 was screwing up my irv menus and moh too... everything is working perfect now! |
| 03:00.46 | JT | right |
| 03:01.06 | JT | just try and avoid all the bad stuff ;) freepbx, grandstream, vonage |
| 03:01.20 | zutoo | lol |
| 03:01.43 | keith4_ | are there any national voip providers that are asterisk friendly? |
| 03:01.43 | zutoo | what phones would u reccommend for under $100? |
| 03:01.52 | JT | polycom |
| 03:02.02 | JT | what is a national voip provider? |
| 03:02.12 | keith4_ | well, vonage would be |
| 03:02.19 | keith4_ | i.e. not regional |
| 03:02.20 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 03:02.25 | JT | of what country? |
| 03:02.26 | fujin | hi, could anyone tell me what Aug 6 15:01:53 WARNING[10925]: pbx.c:1700 pbx_extension_helper: No application '' for extension (callerids, i, 1) |
| 03:02.29 | fujin | means? |
| 03:02.30 | keith4_ | US? |
| 03:02.39 | JT | so maybe you should state that, keith4_ |
| 03:02.41 | zutoo | are aastra any good? i was looking at thier corded cordless combo |
| 03:02.57 | fujin | doh was a typo |
| 03:02.57 | JT | zutoo: not as good as polycom, but not too bad |
| 03:02.58 | fujin | lol |
| 03:03.17 | keith4_ | JT: good point |
| 03:05.10 | keith4_ | so... are there any US voip providers that don't suck like vonage and are asterisk friendly? |
| 03:05.32 | fujin | and also, could anyone take a look at http://rafb.net/p/VYO3BO52.html please? 192.168.108.1 is my asterisk box, 192.168.108.210 is my cisco, which one of those devices is trying to register? |
| 03:06.09 | zutoo | keith4_: have you looked at VoicePulse? |
| 03:06.31 | keith4_ | zutoo: i haven't. should I? |
| 03:06.32 | fujin | voipjet.. IAX2 trunk |
| 03:06.38 | zutoo | http://connect.voicepulse.com |
| 03:07.40 | zutoo | they dont have plans like vonage does, but they have unlimited incoming cheap outgoing dids $11/month |
| 03:08.15 | keith4_ | fantastic |
| 03:09.12 | JT | ~phones |
| 03:09.13 | jbot | methinks phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 03:10.37 | nick125 | What about a phone to set on fire? They don't have a suggestion for that! |
| 03:11.12 | nick125 | </smarta**> |
| 03:11.44 | nick125 | I bet grandstreams will work for that. |
| 03:12.41 | JT | heh |
| 03:15.12 | fujin | anyone? http://rafb.net/p/VYO3BO52.html I need to work out if that is asterisk (192.168.108.1) trying to register, or the device at 192.168.108.210 |
| 03:17.45 | JT | fujin: what, you expect an answer and don't even have any sip debug output? |
| 03:17.55 | fujin | I figured that'd be enough information. |
| 03:18.02 | JT | weird |
| 03:18.06 | fujin | Just needed someone who could understand the cryptic asterisk error messages |
| 03:18.12 | fujin | NOTICE messages, rather |
| 03:18.17 | JT | if i had to guess i'd say it's the .210 |
| 03:18.21 | JT | not particularly crytpic |
| 03:19.54 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 03:20.08 | fujin | JT: http://rafb.net/p/oiAeLU87.html. |
| 03:20.11 | fujin | http://rafb.net/p/oiAeLU87.html rather |
| 03:20.12 | fujin | how about that? |
| 03:20.40 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-73-3.pskn.east.verizon.net) |
| 03:20.55 | JT | you only gave one sip message, there should be a whole dialogue of messages |
| 03:21.04 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 03:22.33 | fujin | doh |
| 03:22.35 | fujin | http://rafb.net/p/S9sC6x48.html |
| 03:22.38 | fujin | thought I selected all of it. |
| 03:22.40 | fujin | that better? |
| 03:23.19 | JT | 2 messages, pretty generous ;) but yes it's coming from the cisco, clearly |
| 03:24.07 | fujin | thoser are the only messages which arrive |
| 03:24.13 | fujin | probly cause I only did sip debug ip 192.168.108.210 |
| 03:24.16 | fujin | right, thanks. |
| 03:24.20 | fujin | now to work out *why* it's doing it. |
| 03:24.47 | JT | what is it? |
| 03:26.19 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 03:26.24 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 03:27.36 | jgoddess | you don't have any warnings in messages as to other errors going on? |
| 03:27.47 | jgoddess | is it a softphone? |
| 03:29.51 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:32.38 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 03:36.56 | fujin | JT: it's an as5400 |
| 03:37.13 | JT | look into its configuration then :) |
| 03:37.18 | fujin | that's the plan |
| 03:38.01 | fujin | Can't see any 1. |
| 03:38.04 | fujin | I wonder what it is. |
| 03:38.09 | fujin | will have to get the network dude to take a look |
| 03:38.25 | kimosabe | does anyone have a spa3000 working as fxo registered to asterisk tu use the pstn via other sip clients |
| 03:38.35 | fujin | the only 1.anything I can see is 'destination-pattern 1.T' |
| 03:40.48 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 03:41.20 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 03:41.43 | Nugget | Step 1: buy a polycom. |
| 03:45.30 | fujin | that's a pretty retarded comment to make |
| 03:45.35 | fujin | I didn't think Polycom made E1 termination devices. |
| 03:45.56 | kimosabe | can some one help me to get my device dialing i have dial tone i can dial to the spa3000 and i get the pstn dial tone but when i try to dial via the pstn i just keep etting dial tone wont dial |
| 03:46.00 | JT | though to be fair, these are alternatives to cisco :) |
| 03:46.26 | fujin | Definitely, but I wouldn't want to run one |
| 03:46.41 | JT | s/these/there/ |
| 03:46.42 | JT | why |
| 03:47.01 | fujin | I can't think of anything that'd offer the same level of support that they do |
| 03:47.06 | fujin | 2 hour onsite hardware replacement = win. |
| 03:47.33 | JT | haha, good support, only if you pay them big bucks |
| 03:47.36 | Sweeper | paying $5k a month for this benefit=fail |
| 03:47.39 | JT | if uptime is really critical, buy a spare |
| 03:47.50 | JT | you can't trust them to be there in your time of need |
| 03:48.23 | Sweeper | yea, company I worked with had a 4 hour mtf contract with bellsouth on their pbx |
| 03:48.28 | fujin | haven't had any issues so far |
| 03:48.40 | fujin | they've managed to fix everythign I've thrown at them. Really don't have the time to deal with retards either |
| 03:48.44 | fujin | I have big bucks to spend |
| 03:48.51 | JT | easy to say until you do have issues |
| 03:49.07 | fujin | we had one just the other day, regarding clid presentation |
| 03:49.10 | fujin | was resolved within the hour |
| 03:49.28 | JT | you needed them to code something? |
| 03:49.32 | Sweeper | let's just say, 4 hours after it went down, I had cell forwarding up with asterisk, and was discussing *'s benefits with the company's CFO, while it took the bell guy another 20 hours to get his shit fixed... |
| 03:49.58 | Sweeper | mmm that was a good day |
| 03:51.02 | fujin | JT: yeah, needed a dialplan modification |
| 03:51.05 | fujin | or whatever. I don't know |
| 03:51.08 | fujin | they fixed it, regardless |
| 03:51.19 | JT | fujin: and it wasn't user configurable? |
| 03:51.25 | fujin | of course it was |
| 03:51.30 | fujin | we just didn't have a user here who knew how to do it |
| 03:52.01 | JT | well that's not a real issue, i'm talking about hardware failures and firmware failures or deficiencies, not configuration support |
| 03:52.21 | fujin | err |
| 03:52.26 | *** join/#asterisk bmg505 (n=leon@196.209.176.19) |
| 03:52.48 | fujin | have you ever owned an as5400, with a premium service contract on it? |
| 03:53.05 | fujin | where premium is platinum |
| 03:53.17 | JT | no, and i fail to see how that's relevant. it was clearly a configuration issue you are refering to. |
| 03:53.23 | fujin | yeah, so? |
| 03:53.35 | fujin | you're hardly in a position to comment on their platinum grade support now are ya |
| 03:53.36 | JT | so has it failed yet? |
| 03:53.41 | fujin | no, it's been great |
| 03:53.46 | JT | i am, it's called the use of logic |
| 03:53.58 | JT | so you have yet to see if it will go to plan |
| 03:54.01 | JT | in the event of failure |
| 03:54.11 | fujin | I'm sure we will, if it comes to that |
| 03:54.15 | kimosabe | come on now room can some one here help me out |
| 03:54.23 | fujin | we've been thinking of ditching it and using another local provider for all of our calls, over SIP |
| 03:54.30 | JT | kimosabe: it's not a room, it's a channel |
| 03:54.42 | JT | over the Internet? |
| 03:54.45 | fujin | no |
| 03:54.50 | fujin | over a peering exchange |
| 03:54.52 | kimosabe | jt i have a problem with my spa3000 |
| 03:54.53 | JT | i wouldn't do it |
| 03:54.58 | fujin | what? |
| 03:54.58 | JT | kimosabe: yes i saw that. |
| 03:55.11 | JT | fujin: i would not eliminate TDM |
| 03:55.17 | JT | for reliability reasons |
| 03:55.20 | kimosabe | can u help me out please i just want to be able to dial over it |
| 03:55.32 | fujin | yeah |
| 03:55.33 | JT | kimosabe: no. i don't know how to help you. |
| 03:55.35 | fujin | thinking != doing |
| 03:55.49 | JT | sure |
| 03:56.07 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-52-56.pskn.east.verizon.net) |
| 03:57.19 | fujin | the price of an E1 here is ridiculous |
| 03:57.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:57.44 | JT | how much? |
| 03:57.53 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:57.58 | fujin | something like $18/channel, +ddi's +monthly rental |
| 03:58.18 | jer | which works out to be... ? |
| 03:58.19 | JT | in whose dollars, and where? |
| 03:58.22 | fujin | nzd |
| 03:58.33 | JT | per channel + monthly rental? |
| 03:58.36 | L|NUX | hello every one |
| 03:58.37 | fujin | yup |
| 03:58.41 | fujin | + tolls ontop of that |
| 03:58.49 | JT | how much is monthly rental |
| 03:59.03 | fujin | not sure, I don't pay the bills |
| 03:59.12 | Kumbang | hi guys, how can i connect * to ss7 network? |
| 03:59.16 | JT | i see |
| 03:59.33 | JT | well it's AUD$20/ch/mo for a pri with optus here |
| 03:59.35 | jer | so monthly rental could be $5 ? |
| 03:59.35 | L|NUX | can some one help me with Line Polarity Reversal, when we call to a number using sip -> asterisk -> zap -> pstn when it will ring the phone it start counting time how can i avoide this ? |
| 03:59.36 | jer | =] |
| 04:00.44 | JT | L|NUX: i assume you mean POTS |
| 04:00.57 | L|NUX | JT: yes |
| 04:01.10 | JT | fun |
| 04:01.31 | L|NUX | ummm |
| 04:01.51 | L|NUX | does it means there will be no help :) |
| 04:02.42 | JT | in theory asterisk might be able to detect polarity reversal upon answer |
| 04:03.16 | L|NUX | yes |
| 04:03.19 | L|NUX | i have read that |
| 04:03.29 | L|NUX | but it will doing when it ring the bell |
| 04:03.54 | JT | i have no idea what you just said |
| 04:04.45 | fujin | You just crit the english language for 1k |
| 04:04.46 | fujin | -_- |
| 04:06.01 | L|NUX | can i post my zapata.conf ? |
| 04:06.06 | JT | not here |
| 04:06.07 | JT | ~pb |
| 04:06.08 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:06.41 | L|NUX | yes i know JT |
| 04:08.11 | JT | fujin: can you find out what the monthly fee is? |
| 04:08.28 | JT | fujin: and is it the same regardless of location? |
| 04:08.38 | L|NUX | http://pastebin.ca/647104 |
| 04:09.09 | fujin | JT: I assume so, have sent an email to our finances manager |
| 04:09.24 | fujin | He's working on some costing at the moment for sip vs. e1 |
| 04:09.31 | fujin | will let you know when I do |
| 04:10.21 | JT | okay cool |
| 04:10.37 | JT | L|NUX: that is an extremely inefficient way to specify channels |
| 04:10.56 | fujin | JT: would you be looking to terminate here? |
| 04:11.11 | JT | maybe one day |
| 04:17.01 | *** join/#asterisk reaxion (n=reaxion@218.189.184.34) |
| 04:17.06 | reaxion | Hello |
| 04:17.18 | *** join/#asterisk dkg0 (n=dkg@lair.fifthhorseman.net) |
| 04:17.18 | L|NUX | JT : i did not |
| 04:17.32 | L|NUX | JT : its automatically generated from Sangoma Configuration |
| 04:17.44 | reaxion | Does anyone have a moment to help with a module registration issue please? |
| 04:17.45 | *** join/#asterisk Paul_UK (n=foo@87.224.50.101) |
| 04:18.05 | JT | L|NUX: i see. |
| 04:18.12 | JT | L|NUX: try adding answeronpolarityswitch=yes |
| 04:18.17 | L|NUX | okies |
| 04:19.07 | JT | of course i am assuming that your telco is sending polarity reversal upon answer |
| 04:19.21 | dkg0 | I'm having a problem with a TDM800P loaded with two X100M's (two POTS lines) and an S400M (four analog telephones). |
| 04:19.26 | reaxion | My issue seems to be due to the fact I'm using Asterisk 1.4.9 and trying to get a proprietary module (Aculab) to install. It has compiled correctly as far as I can tell. |
| 04:19.40 | dkg0 | the fourth port on the S400M doesn't seem capable of making the attached telephone ring |
| 04:19.59 | dkg0 | even though the other 3 S400M ports can make the same phone ring if attached to them. |
| 04:20.04 | L|NUX | JT : humm |
| 04:20.07 | reaxion | The error I receive is: WARNING[5507] loader.c: Module 'chan_aculab.so' did not register itself during load |
| 04:20.07 | L|NUX | JT : let me try again |
| 04:20.18 | JT | dkg0: is the molex power connector plugged in? |
| 04:20.25 | dkg0 | JT: yes, it is. |
| 04:20.27 | reaxion | I think it may be because of the fact the channel driver was originally written for 1.2 |
| 04:20.37 | reaxion | but I'd like to try porting it if it's not too tough |
| 04:20.39 | JT | reaxion: that may be an issue |
| 04:20.41 | L|NUX | JT: its same |
| 04:20.50 | dkg0 | JT: and the device can ring phones on ports 1-3 |
| 04:20.51 | JT | reaxion: do you have full source? |
| 04:20.54 | reaxion | Yes |
| 04:21.00 | JT | dkg0: yeah, i saw that |
| 04:21.09 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 04:21.17 | reaxion | JT: I do |
| 04:21.43 | reaxion | I've made some mods already for things like the ast_channel_alloc function change |
| 04:21.59 | dkg0 | i'm running debian etch with backported zaptel modules 1.4.4 |
| 04:22.01 | JT | reaxion: what does this aculab do? |
| 04:22.09 | dkg0 | and asterisk 1.2.13 |
| 04:22.15 | reaxion | It's a 4 port T1/E1 card with DSPs |
| 04:22.47 | JT | dkg0: so you've got an asterisk and zaptel version mismatch |
| 04:22.53 | reaxion | JT: It is functioning okay on the system with the Aculab software. Only part I can't get working is registration within Asterisk |
| 04:23.15 | JT | reaxion: doesn't sound like it's worth porting |
| 04:23.18 | dkg0 | JT: is that a serious problem? the zaptel modules in etch didn't have support for the TDM800P, from what i could tell. |
| 04:23.31 | dkg0 | Do you think i should upgrade asterisk? |
| 04:23.38 | reaxion | JT: I don't have a choice unfortunately. |
| 04:23.45 | JT | dkg0: it's probably advisable |
| 04:23.50 | JT | reaxion: how so? |
| 04:24.11 | reaxion | It's either that or I need to pay Aculab for a new support contract just to get it working. |
| 04:24.30 | JT | reaxion: and you "had" to upgrade asterisk? |
| 04:24.40 | JT | reaxion: i'm thinking it's easier to junk the aculab |
| 04:25.30 | reaxion | JT: I'm using AsteriskNOW at present |
| 04:25.50 | reaxion | Will *NOW work okay if I downgrade Asterisk to 1.2? |
| 04:25.57 | JT | reaxion: no |
| 04:26.10 | JT | maybe junk asteriskNOW ;) |
| 04:26.26 | reaxion | hahaha |
| 04:26.49 | reaxion | :-/ |
| 04:28.08 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:28.08 | reaxion | JT: I see the dev docs are down |
| 04:28.19 | JT | no idea on that front |
| 04:28.21 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:28.24 | reaxion | ok |
| 04:28.43 | reaxion | Was wondering how the registration is performed, that's all. |
| 04:29.07 | JT | why must you keep this old aculab pile of junk? :) |
| 04:29.38 | reaxion | Budget constraints |
| 04:29.57 | reaxion | Basically, if I get it working I can make enough cash to buy some new cards |
| 04:30.13 | JT | hrm |
| 04:42.05 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 04:47.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:47.52 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 04:59.08 | reaxion | Gah. |
| 04:59.27 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:01.15 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 05:04.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:06.15 | *** join/#asterisk Kenton (n=chatzill@static-71-241-245-6.washdc.fios.verizon.net) |
| 05:06.47 | Kenton | what happened to "show transcoders"? i'm trying to test my new tc400b |
| 05:07.25 | dkg0 | My other issue with this setup is that Asterisk doesn't seem to be able to hear incoming audio from either of the two POTS lines (the X100Ms) |
| 05:07.44 | dkg0 | it can link the X100M through the card (natively) to any of the ports handled by the S400M |
| 05:07.57 | dkg0 | but if it tries to accept DTMF tones, it doesn't do anything |
| 05:08.10 | dkg0 | and if you just route the incoming call directly to the Echo() routine |
| 05:08.20 | dkg0 | you hear nothing back on the line at all |
| 05:09.01 | dkg0 | the FXO ports are all capable of a functional Echo() |
| 05:09.14 | dkg0 | as well has handling DTMF input. |
| 05:10.46 | dkg0 | i've just upgraded to asterisk 1.4.9 now, and restarted the machine (to be on the safe side while debugging) |
| 05:11.06 | dkg0 | is asterisk 1.4.9 and zaptel 1.4.4 a mismatch? |
| 05:12.24 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 05:12.29 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 05:32.00 | *** join/#asterisk zeeesh (n=zeeesh@202.166.161.36) |
| 05:40.14 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 05:43.03 | zutoo | how can i change outgoing caller id? |
| 05:43.31 | zutoo | i changed it for each outgoing line in sip.conf, but its not working |
| 05:44.09 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.36) |
| 05:44.52 | Kenton | did you put it in the right format? |
| 05:45.00 | Kenton | "text" <number> |
| 05:46.06 | jgoddess | can you paste your extensions.conf and set your verbose higher and paste what happens on the CLI when you dial out |
| 05:46.09 | jgoddess | pastebin.ca |
| 05:47.58 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 05:48.07 | *** join/#asterisk saftsack (n=oliver@p54A7F0B7.dip.t-dialin.net) |
| 05:49.13 | *** join/#asterisk mosty (n=mostyn@202-161-9-60.dyn.iinet.net.au) |
| 05:50.06 | mosty | does the digium b410p provide a hardware timing device? |
| 05:50.24 | JT | it'd have to |
| 05:51.08 | tzafrir_laptop | mosty, there's a patch to provide zaptel timing from misdn devices |
| 05:51.09 | mosty | JT, i mean a timing device that can be used for iax trunking, meetme etc |
| 05:51.39 | mosty | tzafrir_laptop, ahh cool that's what i want |
| 05:51.43 | JT | yes |
| 05:51.45 | jgoddess | mosty 1.4 is already configured for timestamps |
| 05:51.51 | jgoddess | which in a sense would be timecode |
| 05:51.59 | JT | oh misdn, i forgot about that annoyance |
| 05:52.23 | JT | i suppose it's too late to say don't buy a b410p? :) |
| 05:52.27 | mosty | jgoddess, i can't use 1.4 on this particular machine |
| 05:52.33 | jgoddess | oh nice |
| 05:52.47 | *** part/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 05:52.54 | mosty | jt: yeah and you'd be talking to the wrong person, i didn't buy it |
| 05:53.02 | JT | heh |
| 05:53.45 | mosty | tzafrir_laptop, do you know where i can find the patch? |
| 05:53.54 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 05:55.10 | tzafrir_laptop | http://bugs.digium.com/8896 |
| 05:56.45 | mosty | thanks |
| 05:57.08 | Aces1Up | <PROTECTED> |
| 05:57.37 | JT | Aces1Up: is this an asterisk or a freeswitch question? |
| 05:58.31 | zutoo | jgoddess : I have the caller id in the right format... http://www.pastebin.ca/647168 |
| 05:58.49 | Aces1Up | both :) |
| 05:58.59 | Aces1Up | its a general question. |
| 05:59.05 | Aces1Up | didn't know where to put it. |
| 06:00.53 | mosty | Aces1Up, callerid and dnid are the only data you get from a call before it's answered |
| 06:01.03 | Aces1Up | bummer.. |
| 06:01.36 | JT | you also get call type (SPEECH, 3k1AUDIO) |
| 06:01.53 | JT | on some PRIs you get ANI and ANI2 as well |
| 06:02.21 | JT | you also usually get CPRES and CTON as well |
| 06:02.56 | JT | as well as sometimes info about if the call was originated from an isdn device or not |
| 06:06.52 | mosty | Aces1Up, what are you trying to do? |
| 06:08.41 | *** join/#asterisk implicit (n=implicit@210.16.50.57) |
| 06:10.42 | zutoo | jgoddess : here is the results of the cli... http://www.pastebin.ca/647178 |
| 06:18.51 | jgoddess | ok so it is using macro dialout trunk |
| 06:18.55 | *** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it) |
| 06:19.06 | jgoddess | and macro-outbound-callerid |
| 06:19.27 | jgoddess | you using a gui like freepbx or something? |
| 06:19.54 | jgoddess | zutoo |
| 06:20.24 | zutoo | yeah, freepbx |
| 06:20.52 | jgoddess | ok |
| 06:20.52 | jgoddess | well |
| 06:21.05 | jgoddess | I know if you go to the sip config and modify the actually sip device page |
| 06:21.25 | jgoddess | you can change the Outbound CID |
| 06:22.12 | jgoddess | and it should keep it to what you want otherwise you will need to modify the extensions.conf to manully change what you want listed as the outgoing CID info |
| 06:22.29 | jgoddess | I would join that channel and speak to them more if it doesn't work |
| 06:22.47 | *** join/#asterisk nimopal (n=nimopal@pool-72-93-207-95.bstnma.fios.verizon.net) |
| 06:22.56 | jgoddess | hhe that littel setting caused quite a problem when join the queues until i took it out |
| 06:23.06 | jgoddess | =P |
| 06:23.09 | jgoddess | :) |
| 06:23.11 | Aces1Up | mosty |
| 06:23.15 | Aces1Up | can i pm you? |
| 06:23.21 | mosty | i prefer to talk in the channel |
| 06:23.30 | nimopal | heyal |
| 06:24.03 | zutoo | so can i add callerid= in sip.conf for each outbound line? |
| 06:24.12 | Aces1Up | mosty i want to share 1 did with 2 channel amongst multiple users. i'm looking at possibly using 5 users per channel for 1 did for a total of 10 users per did. |
| 06:24.16 | jgoddess | no |
| 06:24.22 | jgoddess | that is for the device it self |
| 06:24.34 | jgoddess | don't modify the sip.conf |
| 06:24.40 | zutoo | ok |
| 06:24.41 | jgoddess | modify where you add sip devices |
| 06:24.41 | Aces1Up | so i want to be able to identify users via their caller id, but the caller id won't be reliable. |
| 06:24.43 | jgoddess | on the gui |
| 06:24.44 | mosty | Aces1Up, how do you want to map did's to local users? |
| 06:24.48 | jgoddess | and it has a line called Outbound CID |
| 06:25.05 | Aces1Up | mosty i was thinking of using virtual numbers. |
| 06:25.05 | zutoo | yeah, thats where i originally set it |
| 06:25.16 | Aces1Up | to when they call the did they are asked for their virtual caller id. |
| 06:25.25 | Aces1Up | same as a pin number really. |
| 06:25.44 | zutoo | i tried getting help in #freepbx, but lately they haven't been very responsive |
| 06:25.56 | mosty | Aces1Up, probably simplest to just play an IVR when calls come in. "to call alice press 1, to call bob press 2" |
| 06:25.59 | *** join/#asterisk mirco (n=mirco@p578b4a10.dip0.t-ipconnect.de) |
| 06:26.05 | zutoo | the more time i spend here, the more i am wishing i didnt go with freepbx |
| 06:26.20 | Aces1Up | mosty each user will have their own calling patterns. |
| 06:26.42 | mosty | Aces1Up, why don't just buy more did's? |
| 06:26.46 | jgoddess | does this sound like what is going on |
| 06:26.46 | jgoddess | http://www.freepbx.org/trac/ticket/847 |
| 06:27.48 | Daviey | Anybody else have problems with lcd alignment on linksys spa-94*? |
| 06:28.11 | Aces1Up | mosty have any suggestions for reliable did providers in mexico? |
| 06:28.39 | zutoo | its passing the caller id, but i want it to pass the main number no matter what line its dialing out on |
| 06:28.48 | mosty | Aces1Up, i'm not in mexico, so no |
| 06:29.19 | jgoddess | If you want it to pass the main number no matter what line it is dialing out then remove all Outbound CID info on all sip devices |
| 06:29.46 | JT | or you could do it the easy way, in the dialplan |
| 06:29.57 | JT | Set(CALLERID(num)=) |
| 06:30.27 | jgoddess | well righ but with freepbx if he does it manually and then reloads |
| 06:30.41 | jgoddess | it will not save his info and over right the files |
| 06:30.55 | JT | this isn't a freepbx support channel, but there are places to add such stuff in freepbx |
| 06:31.07 | jgoddess | there is I already advised him |
| 06:31.09 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:31.15 | jgoddess | but I actually ran into the problem before so ;) |
| 06:31.18 | jgoddess | sorry |
| 06:32.05 | zutoo | with freepbx there are sip.conf and sip_additional.conf (for example) anything i do in freepbx goes into sip_additional.conf, so i can do whatever i want in sip.conf without it getting overwritten |
| 06:33.06 | jgoddess | Ok well I am going to bed its 1:30 in tejas |
| 06:33.07 | nimopal | Daviey, my lcd alignment seems OK |
| 06:33.12 | zutoo | I know there is a freepbx support channel, but it sucks! |
| 06:33.15 | nimopal | Daviey, unless im not understanding |
| 06:33.24 | zutoo | jgoddess : thanks for your help |
| 06:33.27 | JT | zutoo: sort of like freepbx itself ;) |
| 06:33.35 | zutoo | lol |
| 06:33.46 | jgoddess | hehe |
| 06:34.14 | *** join/#asterisk saftsack (n=oliver@p54A7ECE3.dip.t-dialin.net) |
| 06:34.28 | jgoddess | I'm trying to remember that nasty gui but it has been banned from my memory |
| 06:34.32 | jgoddess | = |
| 06:34.36 | jgoddess | =P |
| 06:34.43 | jgoddess | ok nite me go |
| 06:34.53 | jgoddess | good luck |
| 06:34.57 | zutoo | thanks again |
| 06:36.12 | zutoo | i wish i would have talked to you guys before this weekend... i just upgraded our asterisk box and did a fresh install with freepbx |
| 06:36.32 | zutoo | i probably wouldn't have gone that route if i talked to you guys first |
| 06:37.07 | zutoo | well, im about to go to sleep too... |
| 06:37.18 | *** join/#asterisk kikoy (n=kikoy@124.106.83.9) |
| 06:37.29 | zutoo | thanks for your help jgoddess and JT |
| 06:38.48 | kikoy | hi guys... how do u capture "no route to destination errors" in your dialplan. i want my server to respond with a "destination unavailable" or with a busy tone, if a "no host to destination" occurs... |
| 06:38.59 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:39.00 | kikoy | any ideas? |
| 06:39.04 | kikoy | thanks in advance |
| 06:39.13 | bintut | hello all.. |
| 06:39.53 | bintut | can i ask fwd related problem here? or, does anybody knows the right channel to ask for fwd questions? |
| 06:41.47 | bintut | i already followed this link ==> http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 |
| 06:42.04 | bintut | yet, i still got a problem here ==> http://paste.debian.net/34093 |
| 06:42.18 | bintut | i'm running asterisk-1.4.9 here |
| 06:42.44 | adeel | anyone know if there's an a2billing channel on any network? there doesn't seem to be one on freenode |
| 06:45.42 | nimopal | kikoy, i dont know the answe to your question, but you could try, _X.,n,Congestion(20) |
| 06:45.46 | nimopal | not sure what that does exactly |
| 06:46.01 | nimopal | or maybe Busy(20) |
| 06:46.22 | nimopal | but wont it already give a "busy here" if there's a dial error.. not sure |
| 06:47.32 | Daviey | nimopal: After i upgraded the firmware and factory reset, the alignment shifted about 5 pixels left :( |
| 06:47.59 | nimopal | weird daviey |
| 06:48.41 | Daviey | tried dumping the version back to a previous one, no joy :( |
| 06:49.59 | *** part/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-fc7b5c7bd4c281fd) |
| 06:50.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:50.38 | L|NUX | JT: we are using H323 -> Asterisk -> Zap -> PSTN |
| 06:50.54 | nimopal | that sux Daviey.. i would return it |
| 06:51.02 | nimopal | geta new one 96x |
| 06:51.26 | L|NUX | when i dial it will ring the phone and cdr start charging the call even party did not picked up the phone |
| 06:51.47 | JT | L|NUX: i don't have experience with asterisk H.323, why are you directing your query at me? |
| 06:52.14 | L|NUX | JT : i just saw you active |
| 06:52.15 | L|NUX | :) |
| 06:52.21 | flenders | L|NUX: are you using a TDM400 card? |
| 06:52.35 | L|NUX | flenders : Sangoma A400D |
| 06:52.42 | JT | uhuh |
| 06:52.47 | L|NUX | flenders : similar to TDM400 |
| 06:53.22 | JT | L|NUX: get rid of analogue is the best suggestion there |
| 06:53.37 | flenders | I have the same problem with a TDM400 and it's because it's analogue |
| 06:53.40 | L|NUX | umm |
| 06:53.58 | flenders | you can't do peoper billing on analogue |
| 06:54.20 | L|NUX | ummm |
| 06:54.21 | flenders | s/peoper/proper |
| 06:54.29 | L|NUX | then i believe these cards are useless |
| 06:54.53 | flenders | for billing per channel, yes |
| 06:54.58 | JT | L|NUX: analogue is useless ;) |
| 06:55.03 | JT | get digital |
| 06:55.11 | flenders | I don't need to do billing here, so they're alrigh |
| 06:55.17 | L|NUX | in Bangladesh Voip is not allowed :P |
| 06:55.18 | flenders | though we're switching to PRI |
| 06:55.36 | flenders | L|NUX: digital doesn't mean voip |
| 06:55.45 | L|NUX | yeah |
| 06:55.46 | L|NUX | i know |
| 06:55.51 | L|NUX | i could not take PRI |
| 06:56.06 | JT | could not take? |
| 06:56.09 | flenders | why not? no PRIs in bangladesh? |
| 06:56.20 | L|NUX | well they want to know the purpose |
| 06:56.29 | L|NUX | and if they know i am doing VOIP then they will kill me :P |
| 06:57.02 | JT | just say it's for your business phone system |
| 06:57.32 | flenders | wow... you can get killed because you wanna use voip?? |
| 06:57.37 | flenders | better not try then |
| 06:57.53 | L|NUX | well its not that easy that you think it is |
| 06:59.32 | flenders | what is not that easy? getting killed or getting a PRI? |
| 06:59.44 | flenders | PRI is not voip, I don't see what the problem is |
| 07:01.04 | L|NUX | flenders : you can not even export voip card in BD |
| 07:01.42 | flenders | again, a PRI is not voip |
| 07:01.53 | flenders | it's a digital multi-channel line |
| 07:02.14 | tzafrir | flenders, there are actually some hacks to have proper billing with analog. |
| 07:02.27 | flenders | tzafrir: accurate? |
| 07:02.33 | tzafrir | If your telco supports them |
| 07:02.57 | JT | said hacks still don't tell you if a call was successful or not |
| 07:03.21 | tzafrir | But then again, I figure that PRI would leave you with a larger proportion of your hair connected |
| 07:03.27 | *** join/#asterisk AnimalMother (n=Krle@194.247.208.90) |
| 07:04.44 | tzafrir | well, for instance, the telco can tell you when the call is actually answered |
| 07:04.50 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 07:05.20 | JT | that's what i was refering to |
| 07:05.30 | JT | if billing wise it was a success or not |
| 07:05.36 | tzafrir | JT, with analog |
| 07:06.13 | JT | err, inband audio won't really tell you that in a way a computer can easily understand |
| 07:06.31 | tzafrir | no. check answeronpolarityswitch |
| 07:06.41 | JT | yes |
| 07:06.51 | JT | that tells you ringback has stopped |
| 07:07.00 | JT | no guarantee it means a far end success |
| 07:07.06 | tzafrir | Generally the telco flips the polarity of the line when the call is answered, and flips again when it is hung up |
| 07:07.28 | JT | i don't think L|NUX's telco does it at all |
| 07:07.55 | L|NUX | tzafrir : i believe it does not |
| 07:08.06 | tzafrir | It's flipped when the remote party has answered. But then again, not every telco does that |
| 07:08.28 | L|NUX | tzafrir : because if they did then it adding answeronpolarityswitch=yes must work |
| 07:08.35 | JT | i already told him to set answeronpolarityswitch |
| 07:08.54 | tzafrir | ah, ok |
| 07:09.24 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 07:09.38 | JT | what do you mean? |
| 07:10.35 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 07:10.39 | L|NUX | tzafrir : i asked and they said they support polarity |
| 07:10.53 | threat | hey |
| 07:11.36 | tzafrir | youcannotunderstandanythingwithoutspaces |
| 07:11.53 | JT | tzafrir: asteriskdoesntlikeparsingconfigfiles |
| 07:12.25 | L|NUX | lolz |
| 07:12.44 | tzafrir | L|NUX, what card do you have? |
| 07:12.55 | L|NUX | tzafrir : Sangoma A400D |
| 07:13.16 | tzafrir | ok, so it's not the X100P that does not support detecting polarity |
| 07:13.46 | L|NUX | humm |
| 07:13.59 | *** join/#asterisk ManxPower (n=manxpowe@032-401-363.area5.spcsdns.net) |
| 07:14.07 | tzafrir | I figure that the Sangoma card does report polarity events. You should see polarity events in the debug of asterisk |
| 07:14.33 | L|NUX | tzafrir : ok let me enable debug :) |
| 07:16.31 | threat | What would cause the volume of a call to lower 5 - 10 minutes during the call? |
| 07:16.56 | threat | the == a |
| 07:18.57 | L|NUX | tzafrir : http://pastebin.ca/647203 |
| 07:18.59 | L|NUX | not much |
| 07:19.19 | *** join/#asterisk kv0s (n=kv0s@p4FD22407.dip0.t-ipconnect.de) |
| 07:21.09 | Chris-NB | hi |
| 07:21.29 | tzafrir | L|NUX, I don't see anything there |
| 07:22.01 | L|NUX | tzafrir : can i pm you |
| 07:22.21 | *** join/#asterisk martin[ug] (n=martin@gewus.de) |
| 07:22.24 | Chris-NB | if asterisk gets a 404 or 403 in a register attempt (register with a SIP provier), asterisk stops trying to register. is it possible to configure asterisk to try to register again? |
| 07:22.27 | martin[ug] | hi all |
| 07:24.00 | L|NUX | tzafrir : can i pb my zapata.conf and zaptel.conf ? |
| 07:24.19 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 07:24.47 | tzafrir | yes |
| 07:25.02 | tzafrir | But I'm not sure how much I can help |
| 07:25.04 | martin[ug] | i have a problem with dial and Limit data |
| 07:25.17 | martin[ug] | i pasted the eytension and debug output here: http://zeus.crashmail.de/nopaste/?633 |
| 07:25.20 | kv0s | Good morning from Germany! ;-) |
| 07:25.44 | L|NUX | tzafrir : http://pastebin.ca/647212 |
| 07:25.49 | martin[ug] | the problem is, i can not hear the start_sound or warning message |
| 07:26.18 | martin[ug] | neither without specifiying a LIMIT_VAR i can not hear the macro "timeleft" |
| 07:26.29 | martin[ug] | just a simple "ring ring" until X is reached |
| 07:26.33 | martin[ug] | any ideas? |
| 07:26.44 | martin[ug] | hallo kv0s :) |
| 07:27.27 | tzafrir | L|NUX, one thing though: you may need to restart asterisk for this to take effect and a reload won't do (fixed on asterisk 1.4.9 / 1.2.23 or a bit before that) |
| 07:27.45 | L|NUX | tzafrir : i did that sir :) |
| 07:28.00 | tzafrir | I can't think of anything else |
| 07:28.15 | L|NUX | hummm |
| 07:28.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:29.53 | martin[ug] | i tested it on different channels, with and without SIP. call from outside... nothing |
| 07:30.19 | martin[ug] | i have no idea what that could be. MusicOnHold is played when i specify "m" in the dial options |
| 07:37.37 | Chris-NB | nobody knows if it is possible to configure asterisk to retry registration after a 403 or 403 response on register with a SIP Provider? |
| 07:38.26 | Sweeper | does there exist a very low-level sip switch, at the level where one does things like "connect sip1.tx to sip2.rx"? |
| 07:39.26 | Sweeper | s/sipswitch/softswitch/ |
| 07:40.24 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 07:40.34 | nimopal | connect? |
| 07:41.13 | nimopal | Dial(SIP/user-id@peer) |
| 07:41.47 | nimopal | or use openser |
| 07:41.52 | nimopal | as a stateless proxy |
| 07:42.15 | Sweeper | uhh |
| 07:42.25 | Sweeper | I mean at the audio level |
| 07:42.38 | *** join/#asterisk gardo (n=gardo@121.97.197.33) |
| 07:42.40 | nimopal | like an audio proxy? |
| 07:42.40 | Sweeper | so you could insert prompts and listen for dtmf and stuff |
| 07:42.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:42.45 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:42.53 | nimopal | asterisk can do that with Monitor |
| 07:43.04 | Sweeper | don't want asterisk, I want something better :P |
| 07:43.17 | nimopal | voxeo.com maybe |
| 07:43.41 | nimopal | but ast will do everything u need in this case |
| 07:43.55 | nimopal | Background, Play, ReadDtmf, it all works fine in asterisk |
| 07:44.15 | JT | Sweeper: yes yate and freeswitch can work at a lower level |
| 07:44.42 | Sweeper | except you get maybe 100 channels per box, and it's all implemented inconsistently, and the SIP stack is icky |
| 07:44.55 | Sweeper | JT: I'll look into yate again, I suppose~ |
| 07:45.07 | nimopal | yeah, I dont personally know any better open source IVR servers |
| 07:45.57 | nimopal | but I found asterisk works pretty OK for that if you have a stable setup and throw in some redundancy |
| 07:46.14 | nimopal | for instance, i think theplanet uses asterisk for some of their call center stuff |
| 07:46.28 | nimopal | plenty of companies use it for ivr |
| 07:46.50 | Sweeper | I'm very familiar with asterisk, nimopal. :) |
| 07:46.58 | JT | but it's not really answering his question |
| 07:47.05 | JT | he wanted lower level audio access and control |
| 07:47.26 | Sweeper | and higher performance in return, although I didn't mention that :D |
| 07:47.36 | JT | heh |
| 07:47.39 | nimopal | well reading dtmf and prompts isnt very low level |
| 07:47.44 | nimopal | of course you have the asterisk source |
| 07:47.51 | JT | yawn |
| 07:47.57 | Sweeper | ha |
| 07:48.39 | WilliamK | is there a way to setup buttons on an IP phone just like you would on a digital PBX to login/logout of a queue with the press of a button? |
| 07:48.50 | WilliamK | sorry; I've never delt with queues under asterisk prior |
| 07:49.10 | Sweeper | WilliamK: depends on the phone, but I don't see why not, with a polycom |
| 07:49.28 | WilliamK | planning on using the Aastra 9133i's |
| 07:49.33 | Sweeper | well |
| 07:49.34 | JT | :/ |
| 07:49.39 | Sweeper | it'd basically just be a speed dial |
| 07:49.46 | JT | go polycom if you can |
| 07:49.49 | WilliamK | JT - not my choice... client budget |
| 07:49.52 | JT | err |
| 07:49.58 | JT | how much is a 9133 |
| 07:50.05 | nimopal | WilliamK, you can add a vertical code to invoke RemoveQueueMember on the caller's extension |
| 07:50.31 | Sweeper | JT: more than a ip330 with brick ;) |
| 07:50.41 | JT | ;) |
| 07:50.50 | Sweeper | but it does have more lines and softkeys |
| 07:51.02 | *** join/#asterisk vgster (n=vgster@91.103.131.90) |
| 07:51.03 | kv0s | Has anyone some ideas where i can get free music on hold (germany - gema!)? |
| 07:51.04 | JT | WilliamK: polycom is cheaper |
| 07:51.09 | nimopal | i havent done that much IVR at all with asterisk |
| 07:51.14 | nimopal | so i have no clue Sweeper |
| 07:51.19 | JT | WilliamK: IP320 or IP330 |
| 07:51.24 | JT | much better too |
| 07:51.28 | nimopal | not sure what would be more low level.. is yate like that |
| 07:51.35 | JT | telephonydepot.com |
| 07:52.14 | Sweeper | nimopal: I don't specifically want ivr, I just want to replace the whole big clusterfuck with something tiny that I can slap a high-level library onto |
| 07:52.25 | nimopal | yah indeed |
| 07:53.05 | WilliamK | diff being the softkeys |
| 07:53.14 | WilliamK | and I definitely don't know howto write XML yet |
| 07:53.17 | Sweeper | how many do you need? |
| 07:53.20 | Sweeper | xml is easy |
| 07:53.28 | WilliamK | I need to setup 4 queues |
| 07:53.45 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.36) |
| 07:53.49 | Sweeper | does every phone need to log in and out of every queue? |
| 07:53.50 | WilliamK | tech, sales-residential, sales-commercial, and I think one other |
| 07:53.55 | martin[ug] | kv0s: check sounddogs.com |
| 07:53.56 | WilliamK | yep |
| 07:54.17 | WilliamK | that's why I was looking at the Aastra because it had alot of free buttons |
| 07:54.26 | *** join/#asterisk tc3driver (n=huh@dsl253-090-134.lax1.dsl.speakeasy.net) |
| 07:54.41 | WilliamK | if it could be accomplished using Poly's I'd go for that |
| 07:54.41 | Sweeper | WilliamK: well, with adhearsion, you could write an easy page for the inphone browser that would handle the login/logouts |
| 07:55.06 | WilliamK | Sweeper, how much it gonna cost me? :) |
| 07:55.08 | *** join/#asterisk saftsack (n=oliver@p54A7DAD6.dip.t-dialin.net) |
| 07:55.32 | threat | hmmmmm |
| 07:55.39 | nimopal | You can have a quick speed dial code that calls RemoveQueueMember |
| 07:55.40 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 07:55.47 | threat | so I take it no one has experience this problem before? |
| 07:55.51 | Sweeper | WilliamK: to pay me to do it? about $200. to do it yourself? 5-8 hours |
| 07:56.25 | WilliamK | 200.00 is well worth it :) |
| 07:56.32 | nimopal | here's an example WilliamK http://astrecipes.net/index.php?q=AstRecipes/AddQueueMember%20and%20the%20queue_log%20file |
| 07:57.08 | WilliamK | nimopal, didn't even know that site existed :) |
| 07:57.32 | nimopal | let Sweeper help u... he knows what's up |
| 07:58.02 | WilliamK | nimopal, I'm game for that |
| 07:58.10 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-27ec1855945a2cc7) |
| 07:58.11 | WilliamK | just need to get phones here |
| 07:58.27 | WilliamK | so what's the diff between the IP 320 and 330s? |
| 07:58.38 | JT | 1 ethernet port |
| 07:58.43 | JT | for pc piggyback |
| 07:59.11 | WilliamK | That ALL?!?! |
| 07:59.14 | Sweeper | yea |
| 07:59.15 | WilliamK | big diff in price for that |
| 07:59.24 | JT | yeah, it's worth it if you need it |
| 07:59.31 | Sweeper | whoa |
| 07:59.44 | Sweeper | voipsupply is selling the 320 for the price I got my 330.... |
| 08:00.13 | Sweeper | some crazy noname has it for $94 tho.... |
| 08:00.20 | JT | as if you'd use voipsupply |
| 08:00.36 | WilliamK | Sweeper, please priv msg me your contact info if you wanna set this up - I'd rather give you the 200 than me waste 5-8hrs :) |
| 08:00.54 | Sweeper | done |
| 08:01.07 | Sweeper | JT: they have pretty pictures and long descriptions! |
| 08:01.08 | Sweeper | :P |
| 08:01.14 | JT | heh |
| 08:02.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 08:04.35 | nick125 | Or maybe a IP 550, they are both pretty nice phones |
| 08:06.14 | threat | Does anyone have any suggestions? |
| 08:06.52 | nimopal | threat, what prob? |
| 08:07.17 | E-bola | What would you guys say where the next step up in quality fromt he polycom's? |
| 08:07.32 | JT | nothing |
| 08:07.38 | E-bola | cisco? |
| 08:07.41 | JT | no |
| 08:07.42 | JT | they're lower |
| 08:07.46 | Sweeper | E-bola: polycoms with gold leaf |
| 08:07.49 | nimopal | are the linksys 96x's better or worse? |
| 08:07.55 | JT | worse |
| 08:07.56 | E-bola | hmm i thought polycoms where only high middle |
| 08:08.13 | Sweeper | cisco can probably make you a more expensive phone |
| 08:08.22 | JT | cisco conference bridges are rebadged polycoms. |
| 08:08.34 | threat | nimopal, volume of a call decreases intermittently, it starts happening about 10 minutes into a call. |
| 08:08.37 | Sweeper | marked up \o\ |
| 08:08.41 | JT | linksys is fairly middle in the quality stakes |
| 08:08.47 | nimopal | threat, hmm |
| 08:08.59 | E-bola | we have a bunch of linksys 92x's and our snom 320's are MUCH better |
| 08:09.10 | JT | polycom is high, but they also have economical models too that maintain good voice quality |
| 08:09.11 | E-bola | so i'd definately consider snom better than linksys |
| 08:09.22 | JT | heh |
| 08:09.22 | nimopal | does the call get choppy threat? |
| 08:09.26 | threat | nimopal, also, at the start of some calls there is a loud booming noise, which dies off after a few seconds. |
| 08:09.59 | JT | E-bola: the polycoms easily surparse snom in audio quality and aesthetics |
| 08:10.03 | threat | nimopal, not that I have noticed, the volume just gets very low and I can hardy hear the other party |
| 08:10.23 | E-bola | JT: the snom's arent aesthetically pleasing thats for sure |
| 08:10.26 | nimopal | i havent run into that... does it only happen with a particular provider? |
| 08:10.27 | threat | nimopal, if it was getting choppy I couldn't tell since the volume was so low |
| 08:10.38 | JT | E-bola: they remind me of grandstream |
| 08:10.48 | E-bola | the linksys models looks much better, but when it comes down to it, 95% percent of satisfaction are from audio quality, not design |
| 08:10.59 | E-bola | atleast thats beent he case with my clients |
| 08:11.03 | threat | nimopal, it usually lasts for about 2 - 10 seconds, sometimes it doesn't come back up at all |
| 08:11.51 | nimopal | seems like a qos issue on one of the channels |
| 08:12.12 | nimopal | if you get hit with some latency suddenly you'll be losing packets |
| 08:12.27 | JT | threat: firstly, what's the setup? |
| 08:12.29 | JT | what channels? |
| 08:13.01 | threat | JT, TDM11B as well as GoTalk VoIP provider |
| 08:13.19 | JT | what phone are you using? |
| 08:13.26 | JT | and does it occur on both channels? |
| 08:13.27 | s0ck | [09:09:59] <JT> E-bola: the polycoms easily surparse snom in audio quality and aesthetics |
| 08:13.31 | E-bola | threat: does it happen when u talk internally too? |
| 08:13.34 | threat | JT, analog phone |
| 08:13.37 | s0ck | what is *the* polycom to get atm? |
| 08:13.40 | s0ck | 501/601? |
| 08:13.40 | JT | ugh |
| 08:13.53 | threat | E-bola, I only have one line so I couldn't know |
| 08:13.54 | JT | s0ck: depends on your needs |
| 08:14.34 | threat | JT, I am not sure if it occurs on both channels. |
| 08:14.51 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 08:15.25 | nimopal | one thing to improve qos on sip channels is to use canreinvite=yes |
| 08:15.44 | nimopal | that way the endpoints may be able to pass audio to each other directly, instead of proxying through ast |
| 08:15.53 | s0ck | needs... perfect audio quality and blf would be nice |
| 08:16.04 | threat | JT, how would I go about troubleshooting this? |
| 08:16.31 | s0ck | been playing with snom360s |
| 08:16.35 | s0ck | they're alright |
| 08:16.38 | s0ck | but nothing special |
| 08:17.24 | JT | s0ck: if you need a million bottons, 601+ expansion consoles, otherwise 320/330/501 is fine |
| 08:17.27 | JT | or maybe even 430 |
| 08:17.28 | nimopal | threat, hmm how consistent is the problem |
| 08:17.36 | nimopal | threat, i mean sounds like it could be the voip provider |
| 08:17.44 | JT | threat: try using a softphone instead of the FXS port. |
| 08:17.46 | nimopal | or your asterisk box is getting hit with some network fluctuations |
| 08:17.53 | JT | threat: that would be my first step |
| 08:18.00 | JT | sounds like an analogue problem |
| 08:18.09 | JT | possibly related to software echo cancellation |
| 08:18.18 | threat | JT, ok |
| 08:18.28 | threat | JT, would I need to tweak it? |
| 08:18.35 | JT | shrug |
| 08:19.01 | nimopal | yeah maybe if the PC gets a lot of load suddently, the card performance will suffer |
| 08:19.07 | s0ck | JT: heard scare stories about 430s |
| 08:19.09 | JT | i'm allergic to analogue |
| 08:19.28 | JT | nimopal: sounds like a digium ;) |
| 08:19.32 | JT | s0ck: like? |
| 08:19.41 | s0ck | calls dropping mid call |
| 08:19.43 | s0ck | random reboots |
| 08:19.52 | threat | nimopal, so you recommend I setup real time scheduling for asterisks / zap ? |
| 08:19.53 | JT | i've had no such issues |
| 08:19.53 | s0ck | allegedly now fixed in a firmware |
| 08:20.01 | JT | even with default fw |
| 08:20.13 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:20.13 | JT | i've had calls going for hours |
| 08:20.27 | JT | and i stress tested them putting 5 calls on hold for about 3 weeks |
| 08:20.44 | nimopal | that might be helpful threat. i would check w/digium support, they may have some solutions |
| 08:20.48 | JT | the timer does not display an interval longer than "hours" ;) |
| 08:20.55 | nimopal | now if your call is dropping completely, that may be a provider issue |
| 08:21.09 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 08:21.17 | s0ck | that's good to hear |
| 08:21.20 | threat | JT, lol |
| 08:22.18 | threat | nimopal, ok, well it is just a volume issue |
| 08:26.20 | threat | Locking over the logs again, it looks like the times it was playing up was when I was receiving a call from the PSTN Line |
| 08:26.49 | JT | hrm |
| 08:26.57 | JT | is the molex connector plugged in? |
| 08:29.03 | *** join/#asterisk gzero (n=gzero@81.175.82.2) |
| 08:30.40 | *** part/#asterisk gzero (n=gzero@81.175.82.2) |
| 08:32.04 | *** join/#asterisk saftsack (n=oliver@p54A7F41A.dip.t-dialin.net) |
| 08:38.04 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 08:41.16 | *** join/#asterisk nighty^ (n=nighty@p1146-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 08:42.22 | threat | JT, of corse |
| 08:43.23 | threat | JT, The problem is intermittent, usually after the first 10 minutes. It starts with a popping sound then the volume goes all soft. |
| 08:44.51 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:46.48 | JT | is it always during an incoming call? |
| 08:46.58 | *** join/#asterisk tuzhila (i=tuzhila@84.47.128.99) |
| 08:47.01 | tuzhila | hi |
| 08:47.13 | tuzhila | how i can get color in my asterisk console? |
| 08:47.27 | JT | start it with your termtime set to xterm |
| 08:47.45 | tuzhila | i dont understand |
| 08:49.03 | JT | term type, i meant to say |
| 08:49.12 | JT | your terminal must be set to xterm |
| 08:50.25 | tuzhila | no, it is not help |
| 08:50.49 | tuzhila | im using xterm, and i dont see color in my asterisk console |
| 08:51.30 | tuzhila | may be there is some command, like color |
| 08:51.47 | JT | try asterisk -cvvvvv |
| 08:52.00 | Daviey | tuzhila: does your normal shell show colours? |
| 08:52.06 | tuzhila | yes |
| 08:52.56 | tuzhila | i have started my asterisk, such asterisk -vvvv |
| 08:53.06 | tuzhila | then i use asterisk -r |
| 08:53.13 | tuzhila | and i havent color |
| 08:53.20 | creativx | asterisk -r kills my colors |
| 08:55.13 | kv0s | Mhm. It is possible to create several voicemailboxes for different types of extensions, such as ringgroups, queues and so on. I will handle different boxes for differente user groups (support calls, marketing, ...). All such user groups can share one voicemailbox. any ideas how i can configure it? |
| 08:59.19 | nimopal | you can create a voicemail entry in voicemail.conf |
| 08:59.26 | nimopal | and reference that it mailbox= |
| 08:59.29 | nimopal | in sip.conf |
| 08:59.35 | nimopal | and when calling VoiceMail() |
| 09:00.48 | Sweeper | JT: is AMI the only option for checking how many people are in a queue? |
| 09:01.18 | kv0s | nimopal: Mhm. Does it work with the freepbx addon too? |
| 09:01.49 | *** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 09:02.00 | nimopal | i assume kv0s, not sure |
| 09:02.47 | E-bola | whats AMI? |
| 09:03.16 | kv0s | nimopal: Which requirements does my linux box has to send voicemessages with e-mail? |
| 09:05.08 | nimopal | kv0s, u just need sendmail which is standard |
| 09:05.20 | nimopal | edit voicemail.conf to enable audio clip attachments |
| 09:07.14 | kv0s | nimopal: LOL - just checked. My linux box hasn't installed sendmail ... no wonder the sending of voicemails doesn't work ... hehe |
| 09:08.25 | nimopal | yeah, you should have something in /usr/lib/sendmail |
| 09:08.46 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:13.06 | kv0s | nimopal: Thanks. Now it works. Great. |
| 09:13.59 | nimopal | cool |
| 09:14.40 | kv0s | nimopal: one additional question. Is it possible, that i define some phonenumber for my voicemail which are called if someone leaves a message on my supportbox. so the voicemailsystem calls me every 10 minutes until it reaches someone and it accepts the call and hear the voicemessage?!? |
| 09:17.13 | Daviey | kv0s: If you can think it, it probably can be done with *nix/asterisk |
| 09:18.14 | nimopal | nimopal, asterisk can send u an sms i think |
| 09:18.36 | nimopal | i would create an alert@site.com and forward that to a bunch of sms phones |
| 09:20.41 | nimopal | pager-e-mail: E-mail address for the pager to which pager notifications should be sent. |
| 09:20.49 | kv0s | Daviey: Hehe. Thanks. Sometimes it is very helpfull to get some tipps which way is the easiest or best or ... ;-) |
| 09:21.02 | nimopal | pagerbody = New message in mailbox ${VM_MAILBOX} from ${VM_CALLERID}. |
| 09:21.23 | nimopal | so you set page-e-mail to alerts@mysite.com.. and make it a distribution list that sends it to a bunch of mobile phones |
| 09:21.55 | kv0s | nimopal: Thanks. But i don't need a pager message. Asterisk must CALL the cell phones. Because SMS not really safe and for support calls it is timecritical ... |
| 09:22.29 | nimopal | SMS is pretty reliable if the phone is on the network |
| 09:23.29 | nimopal | but, theres an app_vmoutcall that does what you want also |
| 09:23.35 | kv0s | nimopal: Mhm. But sometimes u don't check your phone for a sms - or don't hear it. ringing phones are much better to hear .. ;-) |
| 09:23.43 | kv0s | nimopal: for all supporters ... |
| 09:23.44 | kv0s | ;-) |
| 09:25.13 | nimopal | but also you could just invoke Dial(SIP/123@peer&SIP/456@peer) after the call to VoiceMail in the macro |
| 09:27.41 | JT | tuzhila: cool, so you didn't listen to what i said? |
| 09:27.46 | JT | let's try again |
| 09:27.56 | JT | tuzhila: 18:51 < JT> try asterisk -cvvvvv |
| 09:28.04 | JT | i didn't say use -r |
| 09:28.09 | JT | don't waste my time |
| 09:29.12 | JT | Sweeper: probably, not sure |
| 09:29.32 | Sweeper | JT: figures D: |
| 09:29.57 | Sweeper | sure, we can tell the CALLER how many people are in the queue, but tell the agent? nooo |
| 09:30.17 | JT | shrug, you can't work with AMI? |
| 09:33.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:34.53 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 09:35.04 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 09:35.58 | WilliamK | do the polycom 320s come with AC adapters? |
| 09:36.12 | JT | no |
| 09:36.21 | JT | they're PoE by default |
| 09:37.17 | WilliamK | any idea on cost of the bricks? |
| 09:37.20 | WilliamK | :) |
| 09:37.58 | *** join/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 09:38.28 | blackhole | <PROTECTED> |
| 09:38.28 | blackhole | have web callbacks already configured. |
| 09:38.32 | Zipper_32 | about $13-$15 |
| 09:38.34 | blackhole | Can some one help me? |
| 09:38.46 | kv0s | Mhm. Voicemail calls supporters ... Yes! http://www.voip-info.org/wiki/view/Asterisk+Voicemail+Notify+for+Support+Numbers i've tested the script, but getting a "line 97: syntax error: unexpected end of file |
| 09:38.53 | kv0s | Don't know why. Any ideas? |
| 09:39.11 | Zipper_32 | WilliamK: Polycom IP 330 power supply $ 15.50 |
| 09:39.22 | Sweeper | JT: I sure can, I just wish I didn't have to |
| 09:39.33 | Sweeper | and it seems a very silly oversite :P |
| 09:39.44 | WilliamK | from telephonydepot? |
| 09:40.04 | Zipper_32 | Just from my local provider, I'm sure that's a gneral price. |
| 09:40.08 | WilliamK | =) |
| 09:40.12 | WilliamK | ah |
| 09:40.15 | Zipper_32 | Or at least close to it |
| 09:40.39 | WilliamK | I was being a good person and took my meds to kill my pain, unfortunately it likes to put me to sleep |
| 09:41.00 | Zipper_32 | I prefer the POE Switch for the 320/330's. |
| 09:41.21 | blackhole | Can some one have some suggestions/ideas for me? |
| 09:41.48 | WilliamK | I just have a few people who will insist on carrying the phones home with them |
| 09:44.31 | *** part/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-27ec1855945a2cc7) |
| 09:44.40 | blackhole | Anyone please any ideas? |
| 09:44.52 | GromiTM | What is the meaning of the "s" extension in the extension.conf at asterisk 1.4.6? I thought, it will match all incoming numbers for which do not exist any rules in the extension.conf. |
| 09:45.01 | GromiTM | Am I wrong? |
| 09:45.38 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-9369c75844a0abce) |
| 09:48.05 | *** join/#asterisk meyma (n=meyma@lbck-4db2495a.pool.einsundeins.de) |
| 09:48.55 | meyma | Hey guys |
| 09:49.59 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 09:50.09 | meyma | I currently have a small problem with Asterisk: Does somebody know how to connect two incoming channels to one? |
| 09:50.35 | meyma | I have already searched google and the documentation, but I found nothing |
| 09:51.28 | ashd | meyma: you mean like a conference - or something differnet |
| 09:53.00 | meyma | A conference would be possible, but Its not very clean |
| 09:54.37 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:55.09 | meyma | The problem is, that person1 shouldn't recognize, that its not a direct connection to person2 |
| 09:55.53 | ashd | meyma: you mean like someone else monitoring the calls? |
| 09:56.15 | ashd | meyma: must be a way as they do it in call centers all the time |
| 09:56.31 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 09:57.29 | dg | anyone know if there's a way to display a message on a nokia phone? |
| 09:57.36 | dg | sip message doesn't seem to be supported :( |
| 09:58.02 | nimopal | an sms? |
| 09:58.27 | dg | i was hoping for something using sip, seems a bit silly to pay when there's a wifi connection.. |
| 09:58.36 | tuzhila | JT: i dont want to use -c, i want to use -r |
| 10:00.16 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 10:01.20 | nimopal | dg, if you use truphone i think u can send free sms to each other |
| 10:07.16 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 10:07.48 | dg | guess i'll just have to play some messages, was hoping for something a bit more inband |
| 10:08.20 | *** join/#asterisk pejo_ (n=peter@138.240.13.217.in-addr.dgcsystems.net) |
| 10:13.23 | threat | I also have another question / problem. When I dial a mobile phone and the end party presses cancel / hangs up call instead of picking up, my asterisks box will try redialing again via the PSTN line, is this normal behaviour? |
| 10:14.14 | threat | is there any way to distinguish with a person hanging up / busy and the VoIP line is not available? |
| 10:15.38 | JT | tuzhila: it was a test, but if you can't even be bothered to do the test, i can't help you |
| 10:15.58 | JT | threat: how are you calling them? |
| 10:16.02 | JT | over what technology |
| 10:16.32 | Uatec_ | hi there, i'm looking in to getting one of these Sanguma A500 jobbies. (http://www.voipon.co.uk/sangoma-a500-bri-card-pci-express-p-549.html?osCsid=bc158bdfea6767a3f9093adb85367d24) |
| 10:16.36 | creativx | threat: i have the same behaviour here with iax |
| 10:16.43 | Uatec_ | it says ฃ250 |
| 10:16.55 | Uatec_ | do you reckon that's just the card with no bri modules? |
| 10:17.05 | threat | JT, Analog phone to TDM11b to VoIP (GoTalK) |
| 10:17.05 | Uatec_ | or the card with 1 module? or what? |
| 10:17.12 | Uatec_ | it doesn't appear to specify |
| 10:17.18 | Uatec_ | cos the card on it's own would be pretty useless |
| 10:17.51 | tuzhila | JT: yes, with -c it works, i have colors with -c |
| 10:17.52 | JT | threat: i think you mean SIP |
| 10:18.01 | threat | JT, yes SIP |
| 10:18.24 | threat | sorry, used to talking to end-users :) I say VoIP when I mean SIP :) |
| 10:18.42 | JT | threat: ok, so what you do it look at the DIALSTATUS variable upon call termination |
| 10:18.56 | JT | you can then choose to make it not redial the number |
| 10:19.18 | meyma | Does somebody know a editor for Linux with syntax highlighting for asterisk config file? |
| 10:20.00 | tuzhila | JT: i want colors with asterisk -r |
| 10:20.01 | threat | JT, ok, so the DIALSTATUS variable should be some number if the call was busy / hangup and another if it was unable to carry the call through SIPs? |
| 10:20.18 | threat | should contain some number even |
| 10:20.57 | JT | i think dialstatus has a word |
| 10:21.16 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 10:21.38 | JT | a clause |
| 10:22.33 | threat | ok |
| 10:22.45 | threat | I guess I will read up on how to use that |
| 10:23.29 | JT | it would be nice if asterisk just used ITU ISDN cause codes the whole way :) |
| 10:24.44 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:26.03 | creativx | threat: goto(s-${DIALSTATUS}) |
| 10:26.13 | creativx | then you have extensions s-BUSY, s-CONGESTION etc |
| 10:26.32 | threat | ok |
| 10:26.39 | threat | and a default, catch all one as well? |
| 10:26.49 | threat | creativx, what docs would that be under? |
| 10:27.03 | creativx | "show application dial" |
| 10:27.06 | threat | hmm I have that Orielly asterisk ebook somewhere |
| 10:27.06 | creativx | in the CLI |
| 10:27.16 | creativx | <PROTECTED> |
| 10:27.46 | JT | ~thebook |
| 10:27.49 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:27.52 | JT | also, you may not want to use s- |
| 10:28.04 | JT | if you want to handle different extensions differently |
| 10:28.06 | threat | ok so CHANUNAVAIL will occur if asterisk is unable to reach my SIP provider? |
| 10:28.54 | threat | JT only one extension, the analog FXS line. |
| 10:29.20 | JT | no... different called numbers and what not |
| 10:34.55 | threat | hmm |
| 10:36.24 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 10:41.50 | blackhole | <PROTECTED> |
| 10:41.50 | blackhole | have web callbacks already configured. |
| 10:47.36 | adeel | i'm having trouble getting my extensions.conf to correctly work with a2billing |
| 10:47.46 | adeel | anyone have any experience and is willing to help out? |
| 10:49.03 | Sweeper | anyone know if it's possible to configure what the idle state of the softkeys on a polycom 330 do? |
| 10:52.51 | kv0s | Mhm. Just searched for asterisk and sugarcrm integration. But nothing really finished modules found. Any tipps? |
| 10:52.56 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 10:53.39 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 10:55.41 | *** part/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-9369c75844a0abce) |
| 11:00.09 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 11:05.14 | blackhole | adeel, What troubles tell me? |
| 11:06.09 | adeel | blackhole, i dial into the box, a2billling processes the call, but then returns with the fast-busy...it doesn't complete to the trunk |
| 11:06.13 | adeel | and i can't figure out why |
| 11:06.32 | blackhole | Hmm, adeel Have you configured anything related to sms callback? |
| 11:06.53 | adeel | no |
| 11:07.23 | blackhole | Any of your friend have done similar acts? |
| 11:16.20 | HarryR | blackhole, it depends on your SMS provider |
| 11:16.59 | HarryR | blackhole, most will just call a HTTP server of yours when an incoming SMS message comes in, all you have to do is drop a call file into asterisks spool dir after parsing the sms message |
| 11:22.47 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 11:30.04 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 11:31.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:38.19 | styelz | hello, i am having issues with my FXO card. it used to work but has stopped working. can anyone help? |
| 11:39.44 | styelz | i can see the call going to zap. |
| 11:39.48 | styelz | -- Called g0/######### |
| 11:39.53 | styelz | -- Zap/1-1 answered IAX2/melb_asterisk-4 |
| 11:40.25 | styelz | and in ztmonitor i can see the number being sent. but nothing happens. |
| 11:40.44 | styelz | if i dial into the ZAP/1 channel, i can hear a dial tone and dial out |
| 11:40.51 | styelz | that works fine |
| 11:41.00 | styelz | incomming calls work fine.. but i get no audio |
| 11:41.38 | styelz | using asterisk 1.4.9 and zaptel 1.4.4 |
| 11:41.55 | styelz | works fine on my 1.2 setup |
| 11:42.28 | styelz | i thought it could be my kernel ..2.6.22-9-generic |
| 11:42.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:43.13 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 11:44.30 | tzafrir | styelz, will it work if you connect a phone instead of your astribank (with its FXO card)? |
| 11:44.47 | styelz | yes |
| 11:47.08 | styelz | maybe it is the card.. |
| 11:47.14 | styelz | but odd how it works |
| 11:48.21 | *** join/#asterisk spunz_ (n=spunz@h081217155195.dyn.cm.kabsi.at) |
| 11:56.14 | *** join/#asterisk shinao1 (n=shinao1@41.205.188.86) |
| 11:57.29 | *** join/#asterisk kv0s (n=kv0s@p4FD22407.dip0.t-ipconnect.de) |
| 11:57.31 | kv0s | Hi! |
| 11:57.35 | kv0s | .. back ... ,-) |
| 11:57.49 | kv0s | Hopefully only with one small question ... |
| 11:58.02 | kv0s | ... all outgoing calls my cid is withheld. don't know why. |
| 11:58.11 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 11:58.47 | kv0s | at trunks, zapatel.conf, etc. all cid is configured. which options responsibility for cid issues? only sending my cid! incoming cid works well ... |
| 12:02.36 | michael-i | Does anyone know when changes to indications.conf are applied? Should I reload a pbx_* module or should asterisk be restarted? |
| 12:02.37 | Sci_05 | kv0s: you going out a provider? |
| 12:03.28 | kv0s | Sci_05: Telekom with european ISDN. |
| 12:03.55 | kv0s | Sci_05: It is possible, that the cid at the provider is withheld? |
| 12:04.56 | Sci_05 | very possible, might be able to ask them to allow that thru. Some plases here in the states with hold that info unless you ask for it to be send thru |
| 12:08.52 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 12:12.22 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:12.33 | lirakis | morning |
| 12:16.51 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:21.59 | kv0s | Sci_05: Thanks. |
| 12:23.33 | *** join/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 12:23.36 | kv0s | Sci_05: My Provider sucks ... actually they hold back my numbers .. :-( |
| 12:23.36 | kieran491 | hi |
| 12:24.07 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:24.23 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 12:24.27 | kieran491 | i get this error when trying to recive calls from FWD "[Aug 6 22:19:56] NOTICE[2228]: chan_iax2.c:7393 socket_process: Rejected connect attempt from 192.246.69.186, request '838764@incoming' does not exist" its claming that incoming dose not exist but in extension.conf the entry is there... along with another entry that works so i was wondering what could cause this error? |
| 12:26.03 | MrChimpy | did u reload#? |
| 12:26.39 | MrChimpy | and have you got exten 838764 defined in the incoming context/ |
| 12:26.57 | kieran491 | :| |
| 12:27.39 | kieran491 | ohh |
| 12:27.40 | kieran491 | i see |
| 12:29.56 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 12:30.35 | falz | morning. I've had nothing but issues with .lock files with voicemail (since upgrading from 1.0x to 1.2x, then to 1.4x). Any way to just disable them as opposed to just removing them via cron periodicly? |
| 12:32.14 | kimosabe | doeas anyhave a spa3000 registered to * for use of fxo ? if so i have some problem can i get a hand |
| 12:33.34 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 12:34.10 | styelz | hmm, my ZAP works fine if i turn off echo cancelation. |
| 12:34.16 | styelz | ookay |
| 12:34.26 | styelz | maybe its HPEC |
| 12:35.37 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 12:37.35 | blackhole | HarryR, Are you there..? |
| 12:37.42 | blackhole | HarryR, I need your help please.. |
| 12:37.48 | *** join/#asterisk pejo_ (n=peter@138.240.13.217.in-addr.dgcsystems.net) |
| 12:38.05 | blackhole | HarryR, I have SMS server configured using Kannel and modem |
| 12:38.23 | *** join/#asterisk pycsusz (n=istvan_p@pluto.euronetrt.hu) |
| 12:38.35 | blackhole | HarryR, all you have to do is drop a call file into asterisks spool dir after parsing the sms message -- How can i do this? |
| 12:39.58 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:40.43 | falz | http://www.voip-info.org/wiki-Asterisk+cmd+Sms |
| 12:40.45 | pycsusz | Hi All! I have got a question and I hope, that somebody know the answer. I used TE110P card, but now already I can buy just TE120P card. Is there any difference between the two cards? Thank you the answer! |
| 12:44.34 | *** join/#asterisk chenxueqin (n=robbie@235.65.66.218.broad.fz.fj.dynamic.163data.com.cn) |
| 12:45.43 | blackhole | falz, The link doesn't seems to help me....? |
| 12:45.50 | styelz | seems ok... hpec_license_check: License granted for 1 channels |
| 12:46.08 | falz | blackhole: that link mentions something about 'smsq' which talks about how to do it manually |
| 12:48.22 | *** join/#asterisk guillote_GNU (n=bancaria@host191.190-31-26.telecom.net.ar) |
| 12:48.39 | kieran491 | Hi i am reading the Asterisk book that is providied on there site i am reading the section where it comes to the part where you configure your machine to dial outbound calls but once you do this you loose the abilty to recive calls.... its to do with the FWD network i think the problem acures once you change type=user to type=peer |
| 12:48.52 | blackhole | falz, But it sends the message to desired number, i need to provide call back facility |
| 12:49.22 | blackhole | So that it provide a call to desired number |
| 12:49.23 | falz | blackhole: I don't know the exact answer, but the callerid variables have the # in them, so I'd assume you could just pass a variable to a script or whatever you're using to write to that spool directory |
| 12:49.36 | blackhole | Hmm! |
| 12:49.46 | blackhole | HarryR May have information! |
| 12:50.00 | falz | I'm sure you could figure it out without the help of whoever harry is |
| 12:50.15 | blackhole | Hmm! |
| 12:50.17 | blackhole | Let me check! |
| 12:50.23 | HarryR | uh? |
| 12:50.26 | shido6 | blackhole, you can do callback with .call files |
| 12:50.29 | kieran491 | eheh |
| 12:50.45 | blackhole | shido6, Can you guide me how i do that, i don't have idea how can i do that? |
| 12:51.21 | HarryR | blackhole, with the SMS server, I have no experience of sending or recieving sms's directly, so I cant really help there |
| 12:51.45 | blackhole | HarryR, No worries can you tell me how can i place call file into asterisk spool dire? |
| 12:51.57 | blackhole | Any command simialar to smsq |
| 12:52.43 | kv0s | Grml. |
| 12:52.46 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:52.48 | HarryR | blackhole, check out: http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out |
| 12:53.24 | kv0s | Just defined a ring-group which calls some mobile with "Confirmation" (freepbx). But if i answer the call at my mobile, asterisk hangs up and don't play the remote announcement ... |
| 12:53.44 | blackhole | Hmm, Okay Thanks! |
| 12:54.01 | [TK]D-Fender | kv0s: ... |
| 12:54.03 | [TK]D-Fender | ~freepbx |
| 12:54.03 | jbot | i guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:54.05 | HarryR | kv0s, are you sure the extension it's supposed to go into is working ok? |
| 12:54.16 | blackhole | HarryR, Can that me place call between two numbers? |
| 12:54.21 | kieran491 | wait never mind i found it out :) |
| 12:54.22 | HarryR | blackhole, yes |
| 12:54.33 | HarryR | blackhole, just have the extension it put the calls into do a Dial() |
| 12:54.38 | blackhole | HarryR, Okay gr8 seems it may help me to achieve ...! |
| 12:54.49 | *** part/#asterisk pycsusz (n=istvan_p@pluto.euronetrt.hu) |
| 12:54.52 | *** join/#asterisk pycsusz (n=istvan_p@pluto.euronetrt.hu) |
| 12:54.53 | kv0s | HarryR: Yes. The ring group calls out ... |
| 12:54.53 | *** part/#asterisk meyma (n=meyma@lbck-4db2495a.pool.einsundeins.de) |
| 12:55.01 | *** join/#asterisk bintut (n=bintut@cm94.gamma178.maxonline.com.sg) |
| 12:55.02 | zeenix | hello guys |
| 12:55.03 | *** part/#asterisk pycsusz (n=istvan_p@pluto.euronetrt.hu) |
| 12:55.05 | bintut | hello all.. |
| 12:55.25 | zeenix | i still don't fully understand the dialplan-fu :( |
| 12:55.39 | kv0s | If i answer no announcement is played and asterik hangups immedietly |
| 12:55.47 | kv0s | mhm wrong spelling .. ,-) |
| 12:55.53 | zeenix | is this a correct dialplan? |
| 12:55.54 | zeenix | exten => 911,1,Answer |
| 12:55.54 | zeenix | exten => 1,1,SayDigits(1) |
| 12:55.54 | zeenix | exten => 2,1,SayDigits(2) |
| 12:55.54 | zeenix | exten => 911,2,Hangup |
| 12:55.59 | bintut | can i ask FWD<-->IAX2<-->Asterisk-1.4.9 here? |
| 12:56.51 | bintut | anyone here able to peer with FWD through IAX2? |
| 12:58.02 | HarryR | zeenix, uh, I dont think that's correct, check out : http://www.paste2.org/p/5556 |
| 12:58.15 | bintut | i'm always getting a 'Registration Refused' error message here.. :( |
| 12:59.18 | zeenix | HarryR: yes! but i want to play '1' when 1 is entered (through dtmf) and the same for 2 |
| 13:00.05 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 13:00.24 | HarryR | zeenix, well create a new dialplan context called '911-menu' which contains your menu, then for the actual 911 bit, but them into the context |
| 13:00.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:00.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:01.10 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:01.32 | [TK]D-Fender | zeenix: No, that is not at all a proper IVR. Go read up on the "s" standard extension. |
| 13:03.23 | zeenix | [TK]D-Fender: ok |
| 13:03.46 | [TK]D-Fender | zeenix: You'll need to know how to setup timeouts, etc.\ |
| 13:04.31 | lmadsen | hint: check out the TIMEOUT() function |
| 13:05.55 | [TK]D-Fender | zeenix: And though I almost never advise it, the WAITEXTEN application. |
| 13:06.56 | [TK]D-Fender | lmadsen: exten => _.,1,WaiExten ; What does recursion mean again? |
| 13:07.18 | lmadsen | that won't work :) |
| 13:07.31 | lmadsen | (you have a typo :)) |
| 13:07.37 | [TK]D-Fender | lmadsen: exten => _.,1,WaitExten ; What does recursion mean again? |
| 13:07.40 | [TK]D-Fender | lmadsen: I saw that ;) |
| 13:07.44 | lmadsen | hehehehe |
| 13:07.57 | lmadsen | I use WaitExten() in my IVR menu after Background() |
| 13:08.13 | [TK]D-Fender | lmadsen: I use "s" the good 'ole fashioned way.... |
| 13:08.53 | lmadsen | all my stuff comes from SIP and I handle my dialplans a bit differently than most people |
| 13:09.24 | lmadsen | I start with 's' as well though, but it's really a GoSub() calling an IVR feature context from a routing context |
| 13:11.27 | [TK]D-Fender | lmadsen: I exten match, and always Goto. |
| 13:12.08 | lmadsen | ya, I use a lot of func_odbc and routing is based on the DID because I'm running a vPBX system |
| 13:12.10 | [TK]D-Fender | lmadsen: My typical setup does 3 NoOp's for CID / line info for debugging, then proceeds to processing. |
| 13:12.12 | lmadsen | there is 100 was to do the same thing though |
| 13:12.40 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 13:12.46 | HarryR | You know the Asterisk wikipedia article is probably going to get scheduled for deletion? |
| 13:12.51 | HarryR | the YATE page just got deleted. |
| 13:15.35 | lmadsen | that's... strange |
| 13:15.51 | kieran491 | just wondering is it possiable to play sound thought somthing like xine to asterisk over a call? |
| 13:16.28 | [TK]D-Fender | kieran491: Possible, but likely quite complicated. Why? |
| 13:16.37 | *** join/#asterisk Tili (n=tili@46.17.221.87.dynamic.jazztel.es) |
| 13:16.43 | kieran491 | ohh i am just intrested in doing it |
| 13:16.55 | kieran491 | just for learning/ FUN :) |
| 13:16.57 | [TK]D-Fender | kieran491: *DUH*, but what for? |
| 13:17.35 | kieran491 | i want xine or an application to load streamed music off an internet radio station |
| 13:17.39 | kieran491 | as an experiment |
| 13:20.55 | [TK]D-Fender | kieran491: There may be other tools better suited, but you could probably loop through your sound card for that at worst-case |
| 13:21.15 | kieran491 | the machine dose not have a sound card as far as i am aware |
| 13:21.55 | [TK]D-Fender | kieran491: that may prove to be a real problem... |
| 13:22.31 | kieran491 | ahh well cant hurt to try |
| 13:22.58 | [TK]D-Fender | kieran491: Actually... you'd be amazed ;) |
| 13:24.14 | kieran491 | could you give me any directions on where i can read up on this? |
| 13:24.45 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 13:25.15 | backblue | hi, anyone out there, with wanpipe-2.3.4-13? |
| 13:26.33 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:27.09 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-1aecf50b69cb0058) |
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| 13:28.20 | kimosabe | anyone here with spa3000 |
| 13:29.58 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 13:30.01 | kieran491 | TK what program in linux that can run from command consult could i use to play sounds in asterisk (in particuar Streamed Audio from Internet Radio) |
| 13:30.22 | twitchnln | is there an easy way to setup redundant servers with failover? |
| 13:31.58 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:32.15 | twitchnln | and if so, could someone point me toward the howto |
| 13:32.21 | kieran491 | wait i think i found an article |
| 13:32.51 | *** join/#asterisk qdk (n=qdk@193.164.155.10) |
| 13:32.55 | [TK]D-Fender | kieran491: there are shoutcast/icecast docs out there that may be relevant |
| 13:33.22 | puzzled | hi |
| 13:33.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:33.32 | [TK]D-Fender | twitchnln: No, there isn't. Time to learn about SER, PRI gateways, etc. |
| 13:35.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:39.14 | *** part/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 13:40.55 | GromiTM | Is there something wrong with |
| 13:40.55 | GromiTM | exten => 0987654321,2,Set(Var_TO=${SIP_HEADER(TO)}) |
| 13:40.56 | GromiTM | exten => 0987654321,3,GotoIf($["${Var_TO}" = "<sip:01234567@127.0.0.1;user=phone>"]?SIP108,s,1:4) |
| 13:40.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:41.32 | GromiTM | I get [2007-08-06 15:41:16] DEBUG[24234] pbx.c: Launching 'GotoIf' |
| 13:41.32 | GromiTM | [2007-08-06 15:41:16] DEBUG[24234] pbx.c: Not taking any branch |
| 13:41.42 | GromiTM | and I do not know why. |
| 13:41.45 | GromiTM | An ideas? |
| 13:43.50 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:45.28 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 13:46.30 | zeenix | HarryR: how about this one: http://www.paste2.org/p/5558 |
| 13:47.06 | HarryR | looks good |
| 13:49.00 | *** join/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 13:49.11 | [TK]D-Fender | GromiTM: pastebin the call attempt and NoOp your var before testing it. |
| 13:50.00 | [TK]D-Fender | ~stdextens |
| 13:50.01 | jbot | "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 13:50.25 | GromiTM | [TK]D-Fender: How do I NoOp a Var? I see only a Message like "launch NoOp()" but the string init. |
| 13:50.29 | GromiTM | in it even. |
| 13:50.54 | [TK]D-Fender | zeenix: You should change from 911 to "s", and use exten = 911,1,Goto(mycontext,s,1) to start your IVR from another context |
| 13:51.22 | [TK]D-Fender | GromiTM: NoOp(var holds "${avarhere}") |
| 13:51.29 | GromiTM | thnx |
| 13:51.42 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 13:52.19 | *** part/#asterisk NetNut404 (i=grants@nat/yahoo/x-239af5b5ffc87ade) |
| 13:53.27 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 13:54.18 | lmadsen | alternatively: Verbose(1|var holds "${avarhere}") |
| 13:55.05 | GromiTM | should < be escaped in strings? With \? |
| 13:55.37 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:55.46 | lmadsen | it's possible -- the ; too |
| 13:56.17 | lmadsen | the " " in dialplan logic does not mean the same thing as other programming languages (i.e. it is not required and does not mean 'this is a string') |
| 13:56.43 | GromiTM | lmadsen: ah ok. So, there are some things more clear..... :) |
| 13:57.10 | lmadsen | like [TK]D-Fender said... see what the variable actually contains so you know what to match on |
| 13:57.14 | *** join/#asterisk polerin (n=erin@c-71-228-222-87.hsd1.tn.comcast.net) |
| 13:57.16 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 14:01.44 | *** part/#asterisk chenxueqin (n=robbie@235.65.66.218.broad.fz.fj.dynamic.163data.com.cn) |
| 14:01.56 | *** join/#asterisk luke-jr (n=luke-jr@ip72-206-114-236.om.om.cox.net) |
| 14:02.22 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:04.31 | *** join/#asterisk rolanro (n=rolando@200.71.62.199) |
| 14:04.56 | martin[ug] | hi - i upgraded to 1.2.23 because of my cmd Dial Problem with the Limit Data for a call. but i still habe the problem that neither start_sound nor the timeleft macro is palyed to the caller - i posted some information here... http://zeus.crashmail.de/nopaste/?633 - thanks again |
| 14:04.59 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 14:06.03 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:09.07 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 14:09.17 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 14:09.34 | [TK]D-Fender | martin[ug]: Well in that call you also never answered... |
| 14:10.43 | Mercestes | That's just an example, [TK]D-Fender, demonstrating that he did everything correctly. A real call, of course, is in error and the fault must lie somewhere with Asterisk. |
| 14:11.04 | Mercestes | So if you could please fix your crap so it works. >.> |
| 14:11.15 | martin[ug] | what... |
| 14:11.24 | Mercestes | what? |
| 14:11.26 | GromiTM | lmadsen: I tried it but I get this: http://paste.debian.net/34112 |
| 14:11.42 | Mercestes | MORNING FENDER! |
| 14:11.44 | GromiTM | lmadsen: I think, there is a big failure in my thought.... |
| 14:12.01 | martin[ug] | [TK]D-Fender: you mean, you need more info? |
| 14:12.51 | Mercestes | martin[ug], I think he means.....you should answer your call. |
| 14:13.04 | [TK]D-Fender | martin[ug]: I mean you never answered your call so we can't see any proble with it executing as requested |
| 14:13.15 | martin[ug] | ah, okay |
| 14:14.28 | martin[ug] | but i understand correctly, that the start_sound is played when "Dial" starts and before the first ringtone signal? |
| 14:16.08 | [TK]D-Fender | martin[ug]: I doubt it. You seem to be dialing out an interface taht HAS proper call-progress.... you aren't in a call yet if its not answered. |
| 14:17.39 | martin[ug] | i tought that happens with the "Answer" |
| 14:17.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:17.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:17.54 | Mercestes | I didnt' see an "Answer" |
| 14:18.01 | Mercestes | Did you see an Answer, Fender? |
| 14:18.07 | twitchnln | hey, i broke my freepbx permissions, someone got a sec that they could help me get them straightened out? |
| 14:18.07 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:18.10 | Mercestes | oh, there it is |
| 14:18.13 | martin[ug] | Executing Answer("SIP/SPA962-L1-092a8480", "") in new stack isn't what you talking from? |
| 14:18.13 | [TK]D-Fender | Mercestes: I don't see the Answer, do YOU see the Answer? |
| 14:18.36 | [TK]D-Fender | martin[ug]: You missed the point... your DIAL is not considered ANSWERED. |
| 14:18.39 | Mercestes | martin[ug], I would answer the phone anyways. |
| 14:18.43 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 14:19.04 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 14:19.06 | Mercestes | I think you have a sip answer and not a zap answer. |
| 14:19.28 | *** part/#asterisk buleeahn (n=buleeahn@cpe-24-26-243-228.satx.res.rr.com) |
| 14:19.33 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 14:19.35 | [TK]D-Fender | martin[ug]: If you want to "beep' through the ringing, etc, then you can dial through a local channel with the limit attached and answer in there (hard), and the dial out your line. |
| 14:21.02 | *** part/#asterisk falz (n=falz@proxy.supranet.net) |
| 14:21.40 | *** join/#asterisk javb (n=javb@190.80.235.249) |
| 14:22.16 | martin[ug] | [TK]D-Fender: i'am a little bit confused, i understand what you said, but i have no idea how to put that in the dialplan ;) |
| 14:26.39 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 14:28.20 | javb | i have something weird here. im using the waitexten() cmd. When u enter the exten asterisk takes you to that exten. But it seems that asterisk is not following the dialplan because it takes me to the voicemailMAIL if nobody picks it up. |
| 14:28.32 | javb | But if u dial from inside, we the the correct voicemail normally. |
| 14:28.39 | javb | any ideas guys? |
| 14:29.03 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 14:29.36 | martin[ug] | [TK]D-Fender: did you mean sth. like Dial(local/xxx@intern) ? |
| 14:30.35 | [TK]D-Fender | martin[ug]: Yes, like that with your limits. |
| 14:31.02 | [TK]D-Fender | martin[ug]: and in the local exten you dial, you then ANSWER first, THEN DIAL out your actual link. |
| 14:31.36 | [TK]D-Fender | javb: I think your should PASTEBIN your dialplan for us to see.... |
| 14:31.37 | [TK]D-Fender | ~pb |
| 14:31.38 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:32.26 | javb | [TK]D-Fender <---- i know that pasting here is kind of flood.. what do you mean with "PASTEBIN" |
| 14:32.31 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:32.43 | *** join/#asterisk saftsack (n=oliver@p54A7CF99.dip.t-dialin.net) |
| 14:32.43 | [TK]D-Fender | javb: Look up for a taste of OBVIOUS |
| 14:33.09 | s0ck | what does everyone use as a b410p alternative...? |
| 14:34.25 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:34.40 | *** join/#asterisk marcan (i=1337@10.Red-88-24-250.staticIP.rima-tde.net) |
| 14:36.46 | javb | [TK]D-Fender: My DIALPLAN ---> http://www.geocities.com/valdezjoelandres/extensions.txt |
| 14:36.55 | *** join/#asterisk ewaldo (n=chatzill@206.80.91.195) |
| 14:36.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:37.06 | JT | s0ck: junghanns.net card, but the A500 from sangoma is coming out soon |
| 14:37.06 | *** part/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 14:37.19 | martin[ug] | [TK]D-Fender: i understand it like that: http://zeus.crashmail.de/nopaste/?635 |
| 14:38.03 | ewaldo | I've recently compiled and installed 1.4.9 and now when voicemails are left, it does not seem to be able to send it via email. I have manually sent email messages through the local sendmail and that works fine. Anybody have any ideas at what I might look at? |
| 14:38.18 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
| 14:38.59 | [TK]D-Fender | javb: well it seems like it should dial a bunch of people then fall through to VM. whats the problem? |
| 14:39.18 | [TK]D-Fender | ewaldo: pastebin your configs. |
| 14:39.30 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:39.53 | [TK]D-Fender | martin[ug]: Close : exten => 220,1,Dial(local/300@foobar) <---- THIS is where you need to put your LIMITS |
| 14:40.08 | pj_ | NO NO LIMIT |
| 14:40.16 | [TK]D-Fender | martin[ug]: because the LOACL channel WILL get answered immediately, THEN dial out. |
| 14:40.25 | javb | [TK]D-Fender ... if u are inside, u dial, if no one answers, perfect, VM, no prob. If u call FROM OUT SIDE, dial the exten, if no one answer, u dont get VM, u get VMmain.. asking your user and pass |
| 14:40.32 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 14:40.42 | *** join/#asterisk ricky (i=ricky@fedora/ricky) |
| 14:40.51 | ewaldo | [TK]D-Fender: That'd be time prohibitive. I actually took the latest trixbox and yum erased the existing asterisk and compiled and install the 1.4.9. All the configs work and the voicemail shows in the mailbox, but no email is generated and there doesn't seem to be anything in the log. Do you have any ideas on what I should look at as far as asterisk logs or though the commandline? |
| 14:41.15 | [TK]D-Fender | javb: pastebin and actual call please |
| 14:41.32 | Mercestes | <PROTECTED> |
| 14:41.46 | Mercestes | I mean, you've used one before, haven't you? |
| 14:41.58 | JT | ~trixbox |
| 14:41.58 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 14:42.36 | [TK]D-Fender | ewaldo: Sorry, unti you show us your configs we can't trust that things are sane. |
| 14:42.46 | ewaldo | Mercestes: I'll use it if ya'll seriously want to parse through all those configs |
| 14:42.58 | [TK]D-Fender | ewaldo: Show us something USEFUL. |
| 14:43.03 | JT | we seriously don't want to support freepbx |
| 14:43.09 | javb | [TK]D-Fender <--- im not able to do that now, its a custumer PBX and im on my way to see that problem.. im here getting some info to see i can solve this. |
| 14:43.28 | ewaldo | [TK]D-Fender: Well, I would expect asterisk to be a little bit more useful. That's what logging is for. |
| 14:43.39 | [TK]D-Fender | javb: What context is the call that doesn't work placed in? |
| 14:43.46 | Mercestes | ewaldo: But your not using Asterisk |
| 14:43.49 | JT | ewaldo: ah assumptions. leave them at the door. |
| 14:44.29 | ewaldo | JT: Well, I've been told that the only way ya'll can troubleshoot the problem is to paste in my configs, nobody has offered any suggestions on how to track it otherwise. |
| 14:44.39 | JT | ewaldo: standard practice |
| 14:44.51 | ewaldo | JT: KK, will set up. everybody beware! |
| 14:44.59 | JT | ewaldo: your assumption that we can magically know how your system is configures without seeing them is silly |
| 14:45.12 | JT | also you should consider ridding freepbx and going plain asterisk ;) |
| 14:45.17 | [TK]D-Fender | ewaldo: You are stilling here whining and showing us nothing. Are you expecting us to be PSYCHIC? That's only on TUESDAYS |
| 14:45.21 | JT | /sconfigures/configured/ |
| 14:45.51 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:45.55 | [TK]D-Fender | sitting* eek... dyslexic today... |
| 14:46.07 | Mercestes | s/\/sconfigures\/configured\/s\/sconfigures\/configured\// |
| 14:46.08 | JT | sssh, it's tuesday here |
| 14:46.12 | cpm | is monday dyslexia day? |
| 14:46.14 | Mercestes | yay for using sed to fix sed statements |
| 14:46.17 | javb | [TK]D-Fender: if u see my dialplan, waiexten is in [from-pstn] .. but the extens are in [from-internal] ... with "include => from-internal" in [from-pstn] is enough for the waitexten to find it ? |
| 14:46.23 | cpm | JT, that explains everthing |
| 14:46.35 | javb | i mean, does waitexten search for a syntaxt that match the exten entered? |
| 14:46.36 | cpm | Mercestes, that's sick |
| 14:46.57 | Mercestes | cpm: I know. |
| 14:47.05 | cpm | k |
| 14:47.08 | martin[ug] | [TK]D-Fender: great - works - now clear to me!! |
| 14:47.11 | Mercestes | I'm a sick person. |
| 14:47.17 | [TK]D-Fender | martin[ug]: You're welcome |
| 14:47.24 | martin[ug] | thank you |
| 14:47.42 | [TK]D-Fender | javb: You should be able to dail with that included context. Again, I'd ahve to see it in use.. |
| 14:48.13 | *** join/#asterisk heh_v_water (n=heh_v_wa@208-45-6-30.hlna.qwest.net) |
| 14:48.21 | javb | Perfect. i ll come back and pastebin it for u.. let me go overther |
| 14:51.14 | *** join/#asterisk DEac- (n=deac@Platin.DenKn.de) |
| 14:51.33 | DEac- | moin |
| 14:51.51 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 14:54.59 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 14:55.02 | masus | hi all |
| 14:55.27 | masus | i have installed sox from the source -13.0.0 version |
| 14:55.33 | masus | but there is no soxmix |
| 14:55.37 | masus | only sox |
| 14:55.41 | masus | can anybody help |
| 14:58.39 | tzafrir | hmmm.... I thought soxmix was a symlink to sox. Aparantly here they are even not of the same size |
| 14:59.07 | tzafrir | anyway, try making it a symlink. Might just work |
| 14:59.48 | ManxPower | tzafrir: on my system sox is just a shell script that eventually calls something like sox-bin or something silly like tjat |
| 15:01.44 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:01.48 | ewaldo | Alright, ya'll asked for it! Tried to only start off with the more important ones. |
| 15:01.53 | ewaldo | asterisk.conf http://pastebin.com/m729f0179 |
| 15:01.59 | ewaldo | extensions.conf http://pastebin.com/m48fff8a8 |
| 15:02.03 | ewaldo | voicemail.conf http://pastebin.com/m64ef9765 |
| 15:02.06 | ewaldo | vm_general.inc http://pastebin.com/mb4c1f93 |
| 15:02.09 | ewaldo | vm_email.inc http://pastebin.com/m44c30c23 |
| 15:02.12 | ewaldo | sip.conf http://pastebin.com/m6cdc1c12 |
| 15:02.15 | ewaldo | sip_additional.conf http://pastebin.com/m3946fbfe |
| 15:02.16 | masus | hmm |
| 15:02.29 | masus | tzafrir: thanks |
| 15:02.55 | [TK]D-Fender | ewaldo: And some proof that your MTA is running please... |
| 15:03.05 | [TK]D-Fender | ewaldo: And a queue dump, etc |
| 15:03.30 | *** join/#asterisk emergence (i=usji@CPE000c6e52557e-CM000e5c709370.cpe.net.cable.rogers.com) |
| 15:03.37 | ewaldo | [root@asterisk mail]# telnet localhost 25 |
| 15:03.37 | Nugget | telnet is eeeeeeevil! |
| 15:03.38 | ewaldo | Trying 127.0.0.1... |
| 15:03.40 | ewaldo | Connected to localhost (127.0.0.1). |
| 15:03.41 | ewaldo | Escape character is '^]'. |
| 15:03.43 | ewaldo | 220 asterisk.zns.com ESMTP Sendmail 8.13.1/8.13.1; Mon, 6 Aug 2007 10:03:28 -0500 |
| 15:04.44 | ewaldo | might need some guidance on how to get a "queue dump" |
| 15:05.03 | emergence | I'm trying to compile asterisk 1.4.9 on an NSLU2, everything was going fine... until a make error occurred: http://pastie.caboo.se/85275 |
| 15:08.22 | *** part/#asterisk ewaldo (n=chatzill@206.80.91.195) |
| 15:08.29 | *** join/#asterisk ewaldo (n=chatzill@206.80.91.195) |
| 15:09.50 | [TK]D-Fender | ewaldo: "man sendmail" its in there somewhere, just look for a min... |
| 15:10.24 | ewaldo | [root@asterisk asterisk]# sendmail -bp |
| 15:10.25 | ewaldo | /var/spool/mqueue is empty |
| 15:10.27 | ewaldo | <PROTECTED> |
| 15:11.27 | [TK]D-Fender | ewaldo: Well.... maybe they are being rejected elsewhere.... |
| 15:11.55 | ewaldo | Like I said, I manually used telnet to send a mail message through the sendmail server and it worked |
| 15:12.13 | ewaldo | I've set a relay on the destination MTA as well as set the smarthost in my sendmail.cf |
| 15:12.43 | ewaldo | So it's definately not a sendmail issue. That's why I was asking if there are any logs in asterisk that could show me the email attempt. |
| 15:14.41 | [TK]D-Fender | ewaldo: Not sure at this point, maybe someone else can help you more from here... |
| 15:14.53 | ewaldo | <PROTECTED> |
| 15:14.55 | ewaldo | <PROTECTED> |
| 15:14.56 | ewaldo | <PROTECTED> |
| 15:14.58 | ewaldo | <PROTECTED> |
| 15:14.59 | ewaldo | <PROTECTED> |
| 15:15.01 | ewaldo | <PROTECTED> |
| 15:15.03 | ewaldo | <PROTECTED> |
| 15:15.04 | ewaldo | That's what I get on the command line |
| 15:16.16 | ewaldo | Hold on, I had turned off emailing voicemails on that extension for a second, let me turn it back on and get the commandline |
| 15:18.06 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:18.17 | ewaldo | hrm, same result, mqueue still empty |
| 15:18.40 | Dr-Linux | can anyone remind the name of that program which we can use with asteirsk and rubyon rails: something like atherton or what? |
| 15:19.05 | *** join/#asterisk Nockian (i=nockian@unaffiliated/nockian) |
| 15:19.11 | lmadsen | Dr-Linux: adhearsion |
| 15:19.19 | Nockian | is anybody else having problems with voicepulse DIDs? |
| 15:19.24 | Dr-Linux | thanks |
| 15:21.34 | VoicePulse | Nockian, what kind of problem? |
| 15:22.27 | Nockian | VoicePulse: when a person dials our number, they get a 'this number has been disconnected' |
| 15:22.51 | ewaldo | So, does anybody else have any suggestions or recommendations on how to troubleshoot and/or resolve this voicemail emailing problem? |
| 15:26.14 | VoicePulse | Nockian, I'd suggest calling into our support number (732-339-5100) or emailing (contact@voicepulse.com) so someone can begin troubleshooting the issue with your number as quickly as possible. It's unlikely someone in #asterisk will be able to diagnose this issue for you unless we have narrowed it down to a configuration issue with your Asterisk server. |
| 15:26.14 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:26.37 | *** join/#asterisk masterisk (n=mascool@70.88.122.206) |
| 15:26.50 | drako | is there way to try free G.729? |
| 15:27.04 | Nockian | VoicePulse: we have a ticket opened already, i just thought someone else in here might be experiencing the same thing. thanks |
| 15:27.22 | masterisk | drako: try http://asterisk.hosting.lv/ |
| 15:28.17 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:28.21 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:29.07 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 15:29.18 | brodiem | anyone have any experience using Directory() with realtime voicemail.conf? Can't seem to get it to recognize a valid voicemail user. |
| 15:29.24 | drako | masterisk, ty |
| 15:36.13 | *** part/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 15:36.32 | *** join/#asterisk Tili (n=tili@21.17.221.87.dynamic.jazztel.es) |
| 15:37.02 | GromiTM | lmadsen,[TK]D-Fender thanks for the debug hints. It is working now. |
| 15:37.18 | [TK]D-Fender | GromiTM: Glad to hear |
| 15:41.05 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:42.47 | GromiTM | :-) |
| 15:43.16 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:49.58 | lirakis | finally! i got telasip to stop stamping my caller id so i can manipulate it now! YAY! |
| 15:50.19 | *** join/#asterisk skvidal (n=skvidal@fedora/skvidal) |
| 15:50.45 | styelz | sweet |
| 15:50.51 | skvidal | hi all, I'm using 1.4.9 with app_conference and I'm trying to get conference passwords working |
| 15:51.18 | skvidal | the example I found on voip-info.org doesn't appear to work b/c of a lot of missing apps/commands |
| 15:51.21 | *** join/#asterisk fadey (n=fadey@68.19.218.87.dynamic.jazztel.es) |
| 15:51.34 | skvidal | is there a complete command list I can get from my current ver of asterisk? |
| 15:52.08 | [TK]D-Fender | skvidal: "core show applications" |
| 15:52.19 | skvidal | ah ha, thanks |
| 15:55.11 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:55.18 | skvidal | [TK]D-Fender: and If I'm missing DBGet() from that list of commands - should I be looking for anything else? |
| 15:55.35 | zeenix | [TK]D-Fender: how would a sip client reach this 's' extension? i mean with '911' extension, the client dials 911 and the first '911' extension is executed |
| 15:55.38 | Corydon76-work | core show function DB |
| 15:55.52 | [TK]D-Fender | skvidal: You have net been keeping up with this. it was replaced by the DB FUNCTION |
| 15:56.11 | Corydon76-work | DBget was deprecated all through 1.2 |
| 15:56.18 | [TK]D-Fender | zeenix: Clarify who this "sip client is please... |
| 15:56.37 | skvidal | ah, I'm mostly following some docs on voip-info - apparently they're out of date, thanks |
| 15:56.49 | zeenix | [TK]D-Fender: a sip user calling from a sip client (application) |
| 15:56.50 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 15:56.56 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 15:57.08 | zeenix | [TK]D-Fender: asigned 'mycontext' |
| 15:57.35 | datachomper | If I dial a number and include a long string of digits after the initial 10 or 11 digits, will those be successfully transferred to my ivr? or does the phone system only recognize the first 10/11 digits? |
| 15:57.43 | [TK]D-Fender | zeenix: you would give them a numbered exten in another contectt they could dial that would GOTO your IVR context like I showed you earlier |
| 15:57.51 | zeenix | Re: you told me to replace '911' with 's' in: http://www.paste2.org/p/5558 |
| 15:58.27 | zeenix | maybe i missed something while i was away.. |
| 15:58.28 | *** join/#asterisk markgreene (n=markgree@209.12.142.2) |
| 15:58.58 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-73ae545c34348ba2) |
| 15:58.58 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:59.07 | zeenix | [TK]D-Fender: oh! ok! read you message now :) |
| 15:59.13 | [TK]D-Fender | [09:50]<[TK]D-Fender>zeenix: You should change from 911 to "s", and use exten = 911,1,Goto(mycontext,s,1) to start your IVR from another context |
| 15:59.38 | [TK]D-Fender | zeenix: You generally want to avoid trying to make one exten too smart for its own good. |
| 15:59.48 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:00.14 | [TK]D-Fender | zeenix: IVR's should normally be considered "abstract", and not linked to HOW you got there. |
| 16:01.07 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 16:03.38 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 16:04.13 | kv0s | Mhm. Just checked options at freepbx webinterface ... and wonder about the option "reports". What must be installed or configured to work? |
| 16:04.40 | kv0s | ... now i can choose any option, but no call is collected... |
| 16:05.25 | [TK]D-Fender | kv0s: Go look at that projects site. |
| 16:05.44 | *** part/#asterisk masus (n=tet@88.248.14.186) |
| 16:05.48 | [TK]D-Fender | kv0s: It requires that you do CDR/queue logging via mysql |
| 16:06.50 | kv0s | Mhm. |
| 16:07.00 | *** join/#asterisk ChicagoBud (n=Bud@adsl-76-203-90-142.dsl.chcgil.sbcglobal.net) |
| 16:07.25 | kv0s | Okay. Another question - actually i have call and i can hear my own echo. The last day it works perfect ... *grml* |
| 16:08.02 | ChicagoBud | hello. Does anyone have x-lite working through monowall to asterisk? |
| 16:09.39 | kv0s | [TK]D-Fender: Is there any documentation out to install the cdr logging? |
| 16:09.43 | *** join/#asterisk thansen|laptop (n=thansen@151.155.248.199) |
| 16:10.44 | [TK]D-Fender | kv0s: Go look on that projects page. |
| 16:10.54 | markgreene | Has anyone in here implamented extensions in mysql with realtime? I am having an issue getting it to work. |
| 16:10.56 | [TK]D-Fender | kv0s: and : |
| 16:10.57 | [TK]D-Fender | ~wikis |
| 16:10.58 | jbot | well, wikis is http://www.voip-info.org |
| 16:11.40 | GromiTM | Is fowarding calls allways be a thing of a telefon or is it configurable by the asterisk server? |
| 16:13.05 | *** join/#asterisk dos000_ (n=dos000@CPE000f66912f92-CM0018c0c6147e.cpe.net.cable.rogers.com) |
| 16:13.30 | dos000_ | howdy |
| 16:14.04 | markgreene | So nobody has setup extensions in mysql? |
| 16:14.53 | dos000_ | i have am trying to run an asterisk inside linux-vserver . It dies when i try to start it. can someone tell me where i should look to detect why it never gets started. any kind of logs cmd line args ... etc |
| 16:15.15 | dos000_ | i am following http://www.telephreak.org/papers/vpa/ |
| 16:15.36 | dos000_ | i am using * 1.2 |
| 16:15.36 | *** join/#asterisk xorb-1a37 (n=xorb@pool-151-203-62-112.bos.east.verizon.net) |
| 16:15.50 | dos000_ | on debian etch |
| 16:15.54 | markgreene | dos000_: look in /var/log/asterisk/.... |
| 16:16.12 | markgreene | dos000_: most programs running on linux will produce a log in /var/log |
| 16:16.39 | markgreene | dos000_: to look at the most reacent entry you can use a command called "tail" |
| 16:17.09 | markgreene | so in your case you can run tail /var/log/asterisk/messages or /var.../debug |
| 16:17.15 | brodiem | anyone have any experience using Directory() with realtime voicemail.conf? Can't seem to get it to recognize a valid voicemail user. |
| 16:17.29 | hmmhesays | Brodiem |
| 16:17.37 | hmmhesays | you have to tell it to list directory |
| 16:17.45 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 16:17.49 | markgreene | dos000_: you can also start asterisk with the command "asterisk -vvvvvvvvvvvvvvvv" and that will force asterisk to tell you everything it is doing starting up - until the crash |
| 16:17.51 | tzafrir | dos000_, linux-vserver is basically the same as a standard linux system for most userspace programs |
| 16:18.02 | dos000_ | markgreene, i get nothing in both places .. /var/log/messages and /var/log/asterisk are not getting anything |
| 16:18.07 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 16:18.26 | markgreene | dos000_: try starting asterisk by typing "asterisk -vvvvvvvvvvvvvv" |
| 16:18.31 | tzafrir | so you can generally follow standard docs to run asterisk |
| 16:18.34 | markgreene | dos000_: without the quotes ofcourse |
| 16:18.38 | brodiem | hmmhesays what do you mean? I thought the only thing to set was the correct VM context for Directory() |
| 16:18.49 | tzafrir | dos000_, do you run asterisk as root or as a non-root user? |
| 16:18.50 | *** join/#asterisk fadey (n=fadey@68.19.218.87.dynamic.jazztel.es) |
| 16:18.51 | dos000_ | ok ... |
| 16:20.34 | dos000_ | tzafrir, i start it from an init script which tries to run under a certain ast user |
| 16:21.59 | dos000_ | markgreene, heh ... its complaining about not being able to open a log ! i prolly forgot some chmods ! |
| 16:22.15 | *** join/#asterisk galeras (n=galeras@201.244.159.231) |
| 16:22.19 | brodiem | hmmhesays n/m, found a patch: http://bugs.digium.com/bug_view_page.php?bug_id=0002475 |
| 16:22.25 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 16:23.08 | tzafrir | ok. asterisk -U asterisk -vvv |
| 16:23.21 | tzafrir | (if the user is actually asterisk) |
| 16:23.59 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 16:25.20 | *** join/#asterisk shinao1 (n=shinao1@41.205.184.56) |
| 16:27.06 | markgreene | dos000_: if it's running as the user asterisk then I would reccomend running the following commands and see if that clears it up |
| 16:27.26 | markgreene | dos000_: "chown -R asterisk /etc/asterisk/" |
| 16:27.45 | markgreene | dos000_: "chown -R asterisk /sbin/asterisk" |
| 16:27.57 | dos000_ | markgreene, thanks man ! |
| 16:27.59 | markgreene | dos000_: "chown -R asterisk /var/log/asterisk/" |
| 16:28.50 | markgreene | dos000_: let me know what that gives you |
| 16:28.50 | dos000_ | ok ... |
| 16:28.57 | ewaldo | Okay, I have done an extensive configuration analysis and I am certain that my voicemail to email problem is directly within the VoiceMail() function. Is there any way to directly troubleshoot this asterisk command? |
| 16:29.14 | *** join/#asterisk zpertee (n=zach@oh-69-34-21-229.sta.embarqhsd.net) |
| 16:29.48 | zpertee | is a grandstream ata or a linksys ata better? |
| 16:30.22 | styelz | linksys is a sipura |
| 16:30.23 | styelz | heh |
| 16:30.32 | lirakis | how can i change the CDR to show some thing other than 1,2,3 etc. when i use an IVR?? |
| 16:31.01 | *** join/#asterisk Carlos_Tico (n=usmleste@c-76-30-253-15.hsd1.tx.comcast.net) |
| 16:31.07 | lirakis | i mean... how can i change the CDR dest.. |
| 16:31.08 | *** join/#asterisk GlubZor (n=yann@vig38-3-82-228-184-159.fbx.proxad.net) |
| 16:31.12 | GlubZor | Hi all |
| 16:31.58 | GlubZor | I looking for a working mgcp phone start log to add a phone in chan_mgcp.conf, is anyone have that ? |
| 16:32.31 | GlubZor | sorry, not .conf, in chan_mgcp.c |
| 16:33.20 | GlubZor | In fact, I looking for a sample of a mgcp phone connectiong on asterisk to change the init code |
| 16:33.42 | GlubZor | ( a sample of the log...) |
| 16:33.44 | markgreene | Is there anyone in here who has setup extensions in mysql and added them to the dialplan using the "Realtime => ..." switch? I am having trouble. |
| 16:33.46 | [TK]D-Fender | zpertee: ... |
| 16:33.48 | [TK]D-Fender | ~gs |
| 16:33.49 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:36.34 | lmadsen | markgreene: do you *really* want your extensions.conf in the database? func_odbc is a better solution... |
| 16:37.33 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 16:37.49 | kv0s | Is there really some difference between voiceprompts for asterisk 1.2 and 1.4? |
| 16:38.07 | Qwell[] | kv0s: They were re-recorded for 1.4. |
| 16:39.38 | GlubZor | Is anyone already use mgcp nortel phones ? |
| 16:41.41 | markgreene | lmadsen: I like mysql bc it's so flexible |
| 16:42.00 | markgreene | lmadsen: what does odbc write to? |
| 16:42.15 | kv0s | Qwell: So i can't use voiceprompts made for 1.4 with my asterisk 1.2? |
| 16:42.18 | x86 | anyone know a default username / password for the Polycom IP301's web interface? |
| 16:42.19 | hmmhesays | yes |
| 16:42.32 | x86 | what is it? |
| 16:42.39 | _DAW | Polycom.. 456 |
| 16:42.42 | dos000_ | markgreene, its complaining that libodbc.so.1 cannot be loaded ! 1) i already compiled odbc and put it in /usr/local/lib 2) i have LD_LIBRARY_PATH pointing to it ! |
| 16:42.48 | x86 | 456 for the username or password? |
| 16:43.01 | _DAW | Polycom = username 456=password |
| 16:43.03 | x86 | i know 456 will do it from the phone itself, but i'm talking about the web interface |
| 16:43.14 | x86 | ah cool |
| 16:43.16 | x86 | thanks :) |
| 16:43.22 | ewaldo | I've even added |mailcmd=touch /tmp/test to my extension under voicemail.conf and /tmp/test does not exist after I leave a voice message |
| 16:43.25 | _DAW | np |
| 16:43.30 | markgreene | dos000_: I have no idea. It's more than I have time to troubleshoot with you. |
| 16:43.52 | dos000_ | markgreene, thank you again |
| 16:44.06 | markgreene | dos000_: Thought it could be a permissions issue still |
| 16:44.22 | dos000_ | i think i have to put an export LD_LIBRARY_PATH=/usr/local/lib |
| 16:44.52 | markgreene | dos000_: I am not sure. I have never had to set paths when installing asterisk. It's never been an issue |
| 16:45.00 | ewaldo | And I've tried putting mailcmd=touch /tmp/test generally with the same result |
| 16:45.37 | markgreene | lmadsen: I will get your answer later. Gotta jet |
| 16:45.43 | dos000_ | yeeeeeeeeeha ! .. it owrks ! |
| 16:47.47 | *** part/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 16:48.42 | *** join/#asterisk Op3r (n=Op3r@121.97.211.236) |
| 16:49.05 | skvidal | does anyone have any good examples of using the While application to wait for a good conference number/pin? |
| 16:49.18 | skvidal | or, for using the While application at all |
| 16:49.44 | Op3r | anyone here knows any voip provider than can handle more than multiple channels? and can handle 800k minutes per month? |
| 16:50.10 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 16:50.15 | martin[ug] | [TK]D-Fender: damn, early joy ... now the warning message keep repeating during the call |
| 16:50.18 | dos000_ | Op3r, did you try voxbone ? |
| 16:50.33 | Op3r | dos000_: do they accept resellers at voxbone? are they any good? |
| 16:50.52 | dos000_ | i have not had issues with them .. so far and they like resellers |
| 16:51.07 | [TK]D-Fender | martin[ug] Isn't that exactly what that limit is supposed to do? play X after Y seconds and repeat? |
| 16:51.23 | *** part/#asterisk Carlos_Tico (n=usmleste@c-76-30-253-15.hsd1.tx.comcast.net) |
| 16:51.26 | Op3r | dos000_: they seems like for inbound only? |
| 16:51.39 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:51.41 | [TK]D-Fender | skvidal: you wouldn't sue "while" for a PW check. |
| 16:51.51 | skvidal | [TK]D-Fender: 'sue'? |
| 16:51.52 | martin[ug] | [TK]D-Fender: i tought it should do it only trough "building up the call" |
| 16:51.53 | dos000_ | Op3r, sorry .. i misunderstood as incoming channels |
| 16:52.03 | [TK]D-Fender | skvidal: What are you using to validate the input? |
| 16:52.05 | [TK]D-Fender | use* |
| 16:52.07 | Op3r | dos000_: ok |
| 16:52.11 | skvidal | [TK]D-Fender: the database |
| 16:52.22 | skvidal | database put conference foo pinnumber |
| 16:52.27 | skvidal | err conferences, sorry |
| 16:52.31 | [TK]D-Fender | skvidal: what is "the database"? |
| 16:52.35 | martin[ug] | [TK]D-Fender: i wanted to have a Dial, and while the other side didn't answered it should repeat the message |
| 16:52.35 | skvidal | the command |
| 16:52.37 | skvidal | database |
| 16:52.39 | skvidal | in the console |
| 16:52.43 | *** join/#asterisk guillote_GNU (n=guillote@host3.200-117-189.telecom.net.ar) |
| 16:52.49 | martin[ug] | [TK]D-Fender: not while the persons talking to each other |
| 16:52.50 | [TK]D-Fender | martin[ug]: AH, you are going about it completely backwards then |
| 16:53.01 | [TK]D-Fender | martin[ug]: Wrong tool for the job entirely |
| 16:53.01 | skvidal | [TK]D-Fender: database put conferences 3456 1234 |
| 16:53.19 | martin[ug] | [TK]D-Fender: i should have explained what i wanna do.. sorry |
| 16:53.25 | [TK]D-Fender | skvidal: Just do a DB retreive and GotoIf it doesn't match |
| 16:53.25 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 16:53.33 | [TK]D-Fender | martin[ug]: Correct.. |
| 16:53.45 | skvidal | [TK]D-Fender: which I did - but I want it to fail back to ask again in case the user fat-fingers it |
| 16:53.51 | martin[ug] | [TK]D-Fender: and there is a tool available or do i have to use sth like a macro |
| 16:53.52 | [TK]D-Fender | martin[ug]: your only real change is to use a MoH class w/ the "m" option. |
| 16:53.54 | [TK]D-Fender | chance* |
| 16:54.17 | [TK]D-Fender | skvidal: Well if it DOESn"T match go back to where you Aasked for the password |
| 16:55.15 | martin[ug] | [TK]D-Fender: and make a complete e.g 90sec gsm file with the whole thing, and always start moh from beginning |
| 16:56.57 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
| 16:57.00 | skvidal | [TK]D-Fender: okay - that works reasonably well. thanks |
| 16:57.05 | b11d|bbl | allo |
| 16:57.16 | [TK]D-Fender | martin[ug]: the MoH approach is pretty much it... |
| 16:57.21 | [TK]D-Fender | b11d: OMG, it lives! |
| 16:57.26 | b11d | :) |
| 16:57.27 | skvidal | [TK]D-Fender: are there examples for how to do things like this? |
| 16:57.34 | b11d | :) |
| 16:57.41 | skvidal | [TK]D-Fender: b/c I've not found a lot of ones that aren't for 1.2 |
| 16:57.42 | b11d | how have you been TK? |
| 16:57.43 | Daejeo1 | [TK]D-Fender: my friend |
| 16:57.49 | Daejeo1 | how r u |
| 16:57.51 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 16:58.00 | [TK]D-Fender | skvidal: its 1 single stupid line. there is no more to be said of it than "show application gotoif" |
| 16:58.10 | [TK]D-Fender | Daejeo1: Still breathing... |
| 16:58.37 | Daejeo1 | [TK]D-Fender: good to hear that |
| 16:59.21 | Daejeo1 | [TK]D-Fender; did you hear about www.ywh.com |
| 16:59.36 | skvidal | [TK]D-Fender: I was mostly trying to figure out why WaitExten() doesn't seem to accept > 4 digits nor a pound sign - I thought I could go look it up rather than bother people with the question |
| 16:59.45 | [TK]D-Fender | Daejeo1: ummm. what is that? |
| 17:00.03 | Daejeo1 | [TK]D-Fender: this guy sells IP phones |
| 17:00.20 | martin[ug] | [TK]D-Fender: "pretty much it" ... you mean it is the easiest or the only way? i am german, sorry ;) |
| 17:00.21 | Daejeo1 | hard ip phone iax2 |
| 17:00.36 | [TK]D-Fender | skvidal: it does if your dialplan accounts for it. |
| 17:00.37 | Daejeo1 | very cheap phone |
| 17:00.51 | Daejeo1 | i got the quote today |
| 17:00.59 | skvidal | [TK]D-Fender: is there a place I should look to read up on how to make it do that? |
| 17:01.02 | Daejeo1 | let me show you |
| 17:01.07 | [TK]D-Fender | martin[ug]: I just showed you how to get the instructions on this application. There is no "manual" for a single line fo dialplan. |
| 17:01.21 | [TK]D-Fender | skvidal: ... |
| 17:01.23 | [TK]D-Fender | ~book |
| 17:01.23 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:01.31 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 17:01.38 | [TK]D-Fender | Daejeo1: link me specifcaly to it |
| 17:02.09 | *** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk) |
| 17:02.45 | Op3r | can i do this to clock calls going to 011447? exten => _011447.,1,Busy |
| 17:02.45 | Op3r | exten => _011447.,n,Hangup |
| 17:02.58 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 17:03.54 | skvidal | [TK]D-Fender: care to tell me what the single line in the dialplan is that allows for it, then? b/c I'm not understanding |
| 17:04.08 | [TK]D-Fender | Op3r: Sure |
| 17:04.30 | Op3r | clocl block rather |
| 17:04.37 | Op3r | [TK]D-Fender: ok |
| 17:04.39 | Op3r | thanks |
| 17:05.22 | [TK]D-Fender | skvidal: yo are not making any sense. |
| 17:06.18 | skvidal | [TK]D-Fender: that makes two of us: I'm trying to figure out why WaitExten() does not seem to accept > 4 digit extensions. You responded with 'it does if your dialplan accounts for it' - I'm trying to figure out how to make the dialplan account for it |
| 17:06.30 | Op3r | [TK]D-Fender: can u recommend any other voip providers other than voicepulse? |
| 17:06.31 | skvidal | the options to WaitExten() don't |
| 17:07.11 | Daejeo1 | [TK]D-Fender : http://www.didjunction.com/elastix/ipphone.pdf |
| 17:07.29 | [TK]D-Fender | skvidal: Waitexten worksw jsut fine if you have extens to match that kind of #. |
| 17:07.34 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 17:07.56 | [TK]D-Fender | skvidal: Exten => 55555,1,NoOp(Yay, the user can dial 55555 to do this!) |
| 17:08.04 | Daejeo1 | [TK]D-Fender: did you see? |
| 17:08.29 | *** join/#asterisk mirco (n=mirco@p54B27481.dip.t-dialin.net) |
| 17:08.33 | [TK]D-Fender | Daejeo1: Oh god.. thats that Soyo hunk of shit... |
| 17:08.43 | Daejeo1 | not good? |
| 17:09.22 | [TK]D-Fender | Daejeo1: Ok, you are looking for any cheap piece of junk out there. I really can't (won't) help you. |
| 17:09.51 | Daejeo1 | well price does not matter |
| 17:10.25 | [TK]D-Fender | ~phones |
| 17:10.26 | jbot | somebody said phones was http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 17:11.34 | NOT_guru | ~Grandstream |
| 17:11.35 | jbot | [grandstream] the Yugo of VoIP hardware. Run. Run away now. |
| 17:11.42 | NOT_guru | i love that one |
| 17:12.17 | Daejeo1 | [TK]D-Fender: did you ever use GSM termination gateway with asterisk?> |
| 17:12.49 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:13.06 | [TK]D-Fender | Daejeo1: Nope |
| 17:13.38 | Daejeo1 | can you recommend one? |
| 17:13.42 | dg | i had a grandstream ata, the power adapter blew up |
| 17:14.22 | Daejeo1 | 2 0r 4 channels? |
| 17:15.34 | *** join/#asterisk Carlos_Tico (n=usmleste@c-76-30-253-15.hsd1.tx.comcast.net) |
| 17:15.36 | Carlos_Tico | what is PSTN Pass-through |
| 17:15.38 | [TK]D-Fender | Daejeo1: nope |
| 17:16.53 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 17:17.55 | kv0s | asteriskies have a nice day! bye bye |
| 17:21.42 | Carlos_Tico | what is PSTN Pass-through |
| 17:21.52 | pigpen | [TK]D-Fender, back at the grind? |
| 17:24.00 | [TK]D-Fender | pigpen: Yup... Monday... and life sucks... |
| 17:24.20 | pigpen | yes it does. |
| 17:25.16 | *** join/#asterisk Netgeeks (n=root@pbx5.netgeeks.net) |
| 17:25.40 | *** part/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 17:25.54 | *** join/#asterisk toot (n=toot@84.19.255.122) |
| 17:25.58 | [TK]D-Fender | Carlos_Tico: Try to be more specific and let us know exactly what it is you're talking about (what hardware, etc). that term is not that specific without knowing what device is offering it. |
| 17:26.46 | pigpen | [TK]D-Fender, have any time to think how to inject pre-recorded audio into a meetme (using page app) |
| 17:26.46 | pigpen | ? |
| 17:28.25 | [TK]D-Fender | pigpen: Originate a local channel that does a delay, then playback and connects that call to meetme with an auto-login. |
| 17:28.27 | Carlos_Tico | Looking for a device like the AU600 but to use with SIP ????? |
| 17:28.30 | [TK]D-Fender | pigpen: fairly easy |
| 17:28.44 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:29.04 | [TK]D-Fender | Carlos_Tico: Look at the Linksys SPA-3102 |
| 17:29.10 | pigpen | hmm..I will look into it...thanks. |
| 17:29.50 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 17:29.56 | Carlos_Tico | <[TK]D-Fender> Carlos_Tico: Look at the Linksys SPA-3102 ---> Can it toll by pass a sip call to pstn dial tone ? |
| 17:30.24 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 17:30.37 | *** join/#asterisk ricky (i=ricky@fedora/ricky) |
| 17:30.38 | [TK]D-Fender | Carlos_Tico: this is a device that will let you use an Analog Line via SIP, and seperately an Analog Phone as well. |
| 17:31.16 | Carlos_Tico | what i am looking is call to the device and get a pstn dial tone ... is that posible ? |
| 17:31.26 | [TK]D-Fender | Carlos_Tico: yes |
| 17:31.47 | Carlos_Tico | how about the AG188 ? |
| 17:32.35 | [TK]D-Fender | Carlos_Tico: That appears to be H323 and *'s support of it sucks. I would avoid that. |
| 17:32.38 | *** join/#asterisk mirco (n=mirco@p54B27481.dip.t-dialin.net) |
| 17:32.48 | [TK]D-Fender | Carlos_Tico: The SPA-3102 is the best option for a single line. |
| 17:33.38 | Carlos_Tico | how can i setup the SPA-3102 to get dial tone |
| 17:33.54 | [TK]D-Fender | Carlos_Tico: Go read the forums at www.voxilla.com |
| 17:33.54 | Carlos_Tico | do you know any good page ? |
| 17:34.13 | Carlos_Tico | how can i use *'s with spa 3102 |
| 17:35.28 | [TK]D-Fender | Carlos_Tico: Go read the guides. We are not here to hand hold you through the whole process. |
| 17:35.34 | [TK]D-Fender | ~wikis |
| 17:35.35 | jbot | well, wikis is http://www.voip-info.org |
| 17:35.36 | [TK]D-Fender | ~book |
| 17:35.36 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:38.50 | dos000_ | anyone here used digium g729 codec ? |
| 17:39.07 | dos000_ | i am trying to see the process |
| 17:40.55 | Carlos_Tico | Hey brother TK D -Fender ..... i am confused .... |
| 17:41.08 | Carlos_Tico | do i have to install *'s into the linksys ? |
| 17:41.14 | *** join/#asterisk sysreq` (n=sysreq@70.53.190.81) |
| 17:41.28 | shido6 | wow |
| 17:42.32 | [TK]D-Fender | Carlos_Tico: No, go read the BOOK to understand what * is. |
| 17:42.38 | *** join/#asterisk CoderCR (n=creyna@wsip-68-15-30-74.sd.sd.cox.net) |
| 17:42.41 | CoderCR | Hello all |
| 17:44.36 | CoderCR | I am trying to get some feedback on Rhino cards |
| 17:45.29 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM0019477f689a.cpe.net.cable.rogers.com) |
| 17:45.57 | [TK]D-Fender | CoderCR: Look to be ultra minimal Zaptel compatable with no HWEC, support for E1, etc at a very marginally lower cost. I wouldn't touch them with a 10' pole. |
| 17:46.06 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 17:47.14 | *** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com) |
| 17:52.35 | *** part/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 17:54.14 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:56.29 | CoderCR | Fender you would not use Rhino? |
| 17:56.46 | [TK]D-Fender | CoderCR: Not their cards, channel banks, sure |
| 17:56.52 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 17:56.59 | CoderCR | All their cards have EC |
| 17:58.27 | thx2000 | Has anyone had any luck getting siptapi to run in vista? |
| 17:58.35 | thx2000 | Or any other TAPI device for that matter? |
| 18:00.21 | [TK]D-Fender | CoderCR: I see a seperate module listed on their site, no listing on VoipSupply for this piece though. And the price is little different than Digium & Sangoma. Again, I would not sue. |
| 18:00.50 | coppice | its unamerican not to sue |
| 18:01.01 | [TK]D-Fender | use* |
| 18:01.04 | mvanbaak | thx2000: I didn't even got to the point I have vista working |
| 18:01.10 | mvanbaak | ;) |
| 18:01.11 | [TK]D-Fender | coppice: And I am NOT American :) |
| 18:01.21 | thx2000 | mvanbaak: hehe, u didn't try hard enough :P |
| 18:01.32 | coppice | oh. sorry for the inadvertent insult |
| 18:01.51 | coppice | but that would explain the tendency not to sue |
| 18:01.57 | mvanbaak | thx2000: all it did was tell me: "You are about to move the mousecursor. This action can harm you or the data on your pc. Are you sure you want to continue?" |
| 18:02.12 | mvanbaak | and 10.000+ questions like this |
| 18:02.22 | thx2000 | lol |
| 18:06.21 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 18:06.26 | [TK]D-Fender | CoderCR: I have jsut found a 4 port analog card with HWEC from them (R4FXO-EC) for $ 360 @ Atacomm. Only 1 user rating though.... could be interesting considering that IS a *very* aggressive price-point |
| 18:06.35 | [TK]D-Fender | CoderCR: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-58429846784.htm |
| 18:06.58 | CoderCR | Fender all their cards have EC |
| 18:07.01 | [TK]D-Fender | CoderCR: They are a veritable "unknown" for me as far as cards go. I'd be willing to test one. http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-58429846784.htm |
| 18:07.01 | CoderCR | for about 1 year now |
| 18:07.12 | CoderCR | they were the first to put EC on all cards |
| 18:07.34 | [TK]D-Fender | CoderCR: Show me ANOTHER that list this fact. Their own site lists HWEC avaialble through a seperate MODULE that I have not even seen for sale anywhere |
| 18:07.54 | CoderCR | Fender all -EC cards have it built in |
| 18:08.12 | thx2000 | Does Rhino, or any other manuf. offer free install support? |
| 18:08.13 | CoderCR | the seperate module thing was from 3 years ago |
| 18:08.19 | CoderCR | Rhino does |
| 18:08.28 | CoderCR | Rhino give free 5 year support |
| 18:08.31 | thx2000 | nice |
| 18:08.39 | coppice | if they all have EC built in, why does the front page of their web site refer to optional EC modules? |
| 18:08.41 | thx2000 | Rhino > Digium ... in terms of quality? |
| 18:08.46 | CoderCR | LOL |
| 18:08.54 | CoderCR | coppice they do not maintain it well |
| 18:08.55 | CoderCR | lol |
| 18:08.55 | BSD_Tech | <PROTECTED> |
| 18:08.58 | CoderCR | just like marketing |
| 18:09.09 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:09.10 | [TK]D-Fender | CoderCR: I see that option in the analog line, but not the digital one so far |
| 18:09.31 | coppice | I doubt that. Its right under the PCI-E cards, which are obviously recent, and have no EC |
| 18:09.53 | [TK]D-Fender | CoderCR: I'm keeping an open mind. Link me one please. |
| 18:10.02 | thx2000 | [TK]D-Fender: you got a good read for getting up to speed on digital telephony? PRI/T1, etc... |
| 18:10.38 | CoderCR | Want more information |
| 18:10.40 | CoderCR | Call Rhino |
| 18:10.40 | Mercestes | thx2000: google wiki PRI google wiki T1 google wiki VoIP |
| 18:10.44 | CoderCR | They will tell you all about it |
| 18:10.53 | Mercestes | CoderCR, You work for Rhino, don't you? |
| 18:10.55 | CoderCR | I am just trying to see what the people on channel thing of cards |
| 18:10.57 | CoderCR | No |
| 18:10.59 | CoderCR | I do not |
| 18:11.09 | CoderCR | I am looking to check some things for a project I am going to be doing |
| 18:11.13 | CoderCR | Which cards have more support |
| 18:11.18 | CoderCR | but I know a lot about their project |
| 18:11.27 | CoderCR | they are zaptel cards that are plug and play |
| 18:11.34 | CoderCR | Rhino have a great product |
| 18:11.37 | Mercestes | Well, you ask us for our opinion, and then tell us all about it. |
| 18:11.39 | CoderCR | I just dont know why they dont sell more |
| 18:11.47 | CoderCR | it just seems to be poor marketing and a poor web site |
| 18:11.51 | Mercestes | methinks you didn't want our opinion to begin with. |
| 18:11.58 | CoderCR | No |
| 18:12.02 | CoderCR | I want your opinion |
| 18:12.03 | Mercestes | seems like you're more trying to sell us on them. |
| 18:12.09 | CoderCR | I also want to clear false information |
| 18:12.13 | CoderCR | LOL |
| 18:12.26 | [TK]D-Fender | CoderCR: How many have you used personally? |
| 18:12.30 | Mercestes | Well, thanks for the information, Mr. Rhino sales person. |
| 18:12.32 | CoderCR | 0 |
| 18:12.39 | CoderCR | which is why i want to know about reliability |
| 18:12.42 | Mercestes | ..... |
| 18:12.43 | [TK]D-Fender | CoderCR: then you opinion seems well qualifiesd. |
| 18:12.49 | CoderCR | Not what the card has. |
| 18:13.08 | CoderCR | I have used Sangoma |
| 18:13.12 | CoderCR | great cards |
| 18:13.13 | CoderCR | reliable |
| 18:14.06 | CoderCR | annoying that you need to configure zaptel and wanpipe |
| 18:14.22 | CoderCR | Rhino are plug and play. I have had 1 person use them |
| 18:14.32 | CoderCR | you only need to patch zaptel for their cards |
| 18:14.35 | CoderCR | and compile |
| 18:14.39 | CoderCR | configure zaptel |
| 18:14.56 | mvanbaak | digium cards work without a patch to zaptel :) |
| 18:15.04 | CoderCR | true |
| 18:15.07 | CoderCR | They do |
| 18:15.26 | mvanbaak | that's more plug&play than a card that needs a patch to zaptel |
| 18:15.32 | CoderCR | But that is because you get the zaptel from digium |
| 18:15.36 | CoderCR | Well |
| 18:15.54 | BSD_Tech | well soon zaptel wont be needed |
| 18:16.08 | BSD_Tech | with the new projects |
| 18:16.18 | CoderCR | plug and play because if you put an analog card with 2fxo and 2fxs.. proc/zaptel tells you what channels have what on them |
| 18:16.21 | coppice | the Rhino site says EC will be be available for all their cards from August. I can see anything about it shipping as standard |
| 18:16.33 | CoderCR | August of last year.. lol |
| 18:16.35 | aptura | had a little tour of a ATT owned LEC the other night after finishing up our changeout. Amazing the amount of fiber and racks of switching gear that is installed. Mostly interesting is the fact that also other local CLEC owned equipment is renting floor space in the building. |
| 18:16.47 | CoderCR | way cool |
| 18:17.01 | tzafrir | CoderCR, there's actually a patch for wctdm for that... |
| 18:17.10 | [TK]D-Fender | CoderCR: I admit their lower density stuff looks appealing.... jsut not sure I'd be the guinea pig for it... |
| 18:17.27 | aptura | tzafrir Do you know what is the largest asterisk install known to date? |
| 18:17.34 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 18:18.33 | tzafrir | aptura, what do you mean? |
| 18:19.11 | coppice | BSD_Tech: what did you mean by that? |
| 18:19.14 | thx2000 | SHSU so far has moved 1,600 IP phones from Cisco CallManagers to Asterisk |
| 18:19.28 | [TK]D-Fender | CoderCR: Well, I'll bit... I'm going to DL docs and drill for reviews. |
| 18:19.30 | aptura | Ie the total traffic flow across all the servers. I know A university in texas is the larest so far. |
| 18:20.04 | thx2000 | Sam Houston State University |
| 18:20.12 | aptura | yes thats it. |
| 18:20.22 | thx2000 | http://www.networkworld.com/news/2006/091206-von-sam-houston.html |
| 18:21.06 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:21.20 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 18:22.01 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:22.54 | BSD_Tech | there is a new project going on to make a open lib and a open timer |
| 18:23.18 | generalhan | anyone have any issues using MixMonitor with 1.2.23 ?? or rather, anyone using it with good results ? |
| 18:23.31 | BSD_Tech | so ztdummy go byebye and zaptel would be less of a issue |
| 18:23.40 | BSD_Tech | its not out yet but we are close |
| 18:24.41 | CoderCR | ok |
| 18:24.44 | CoderCR | I was wrong |
| 18:24.55 | CoderCR | their EC module has been under works for 3 years now |
| 18:24.56 | CoderCR | lol |
| 18:25.02 | CoderCR | it is just not about to be released |
| 18:25.06 | CoderCR | they have all the info on it |
| 18:25.13 | CoderCR | all the analog cards do have EC on them |
| 18:25.23 | *** join/#asterisk ikey (i=ikey@219.64.139.201) |
| 18:25.25 | generalhan | When i try to use MixMonitor(${callfilename}.wav} i get an error saying that the file doesnt exist ... am i using this incorrectly ? i know the file doesnt exist, i am trying to create the recording with this ! http://generalhan.pastebin.ca/647761 anyone have any ideas ? |
| 18:25.29 | CoderCR | The new T1 cards been released will have EC on them also |
| 18:25.31 | CoderCR | So |
| 18:25.39 | CoderCR | just to let you know I had some bad info |
| 18:25.45 | *** part/#asterisk CoderCR (n=creyna@wsip-68-15-30-74.sd.sd.cox.net) |
| 18:25.50 | Mercestes | ... |
| 18:25.52 | Mercestes | retard |
| 18:26.07 | Mercestes | I hope he searches himself and finds the log. |
| 18:26.17 | Aces1Up | if i want to set a variable in asterisk to nil, what is the equivilent statement? |
| 18:26.19 | mvanbaak | food ! |
| 18:26.33 | aptura | Mercestes he is new at using these cards right? |
| 18:26.34 | generalhan | Mercestes: its funny when people just spout off and then leave ! |
| 18:26.49 | Mercestes | aptura, he never even used the cards! |
| 18:26.50 | Corydon76-work | generalhan: does the directory exist? |
| 18:26.57 | aptura | I see |
| 18:27.00 | Mercestes | he just saw it on a website and started spouting off about it. |
| 18:27.13 | lirakis | does the cdr always fork.. or split when an AGI script is called?? |
| 18:27.15 | Mercestes | That conversation started with, "I want your opinion on these cards.." and then he went into his sales pitch |
| 18:27.16 | *** join/#asterisk mog (i=mog@nat/digium/x-4ac198015050e00c) |
| 18:27.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:27.18 | Netgeeks | Aces1Up: try Set(VARNAME=) |
| 18:27.19 | aptura | really the only way to learn is just buy it and use it in the field. |
| 18:27.25 | Mercestes | so..no one should by rhino because they attract retards |
| 18:27.30 | Aces1Up | netgeeks will do. |
| 18:27.35 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:27.42 | generalhan | Corydon76-home: at the time of the FIRST call it doesnt exist .. i use Monitor() now and it creates the directory for me. but even if i use MixMonitor after the directory does exists i get the same error |
| 18:28.06 | Corydon76-work | generalhan: I see a possible typo in the path |
| 18:28.21 | Corydon76-work | "outoging" instead of "outgoing" |
| 18:28.59 | Netgeeks | hrm, outoging... sounds german |
| 18:29.04 | generalhan | Corydon76-home: but thats ok ... is it just that MixMonitor will not create a new directory for the files like Monitor() does ? |
| 18:29.15 | Corydon76-work | generalhan: correct |
| 18:29.20 | generalhan | eww |
| 18:29.33 | generalhan | so much for that ... so i have to keep using Monitor anyway :( |
| 18:29.49 | Corydon76-work | No, you could use System(mkdir ...) |
| 18:30.05 | Netgeeks | what does monitor not do that mix does that made you want to use mix? |
| 18:30.44 | Mercestes | monitor does not do mix, you have to do some outside scripting to mix them. |
| 18:31.15 | Corydon76-work | Mercestes: it's just a flag passed to Monitor |
| 18:31.19 | generalhan | Netgeeks: i was having bad experiences with Monitor on my 1.2.10 system ... so i wanted to try MixMonitor as a change ... but now Monitor seems to be working ok on 1.2.23, i just still wanted to give MixMonitor a try to see if it performed better |
| 18:31.29 | Netgeeks | ah |
| 18:31.54 | generalhan | <PROTECTED> |
| 18:32.07 | Mercestes | cronjob and sox. |
| 18:32.16 | Netgeeks | ewww |
| 18:32.22 | *** join/#asterisk kannan (n=kannan@121.246.24.85) |
| 18:32.23 | generalhan | Mercestes: you mean to mix the 2 files ?? i used to do that too |
| 18:32.51 | Netgeeks | why not use exten => h and System call to fix it after the call is over... *shurg* I guess both work |
| 18:33.21 | generalhan | but then this system i was using started to mess up left and right ... segmentation faults every other second and cronjob started to fail each night, so i had to get away from that |
| 18:33.24 | lirakis | generalhan: have you just tried doing Monitor(somechan,filebase,m) ? |
| 18:33.27 | *** join/#asterisk pagec (n=pagec@cpe-74-73-191-68.nyc.res.rr.com) |
| 18:33.33 | generalhan | lirakis: that is how i do it |
| 18:33.39 | pagec | the compile order is zaptel, libpri, asterisk right? |
| 18:33.48 | generalhan | Monitor(wav,${CALLFILENAME},m) |
| 18:33.48 | lirakis | generalhan: i thought you were using MixMonitor |
| 18:34.07 | generalhan | lirakis: no i am not using MixMonitor because it doesnt work the way i need it to |
| 18:34.20 | lirakis | generalhan: have you NoOp(${CALLFILENAME}) to see what it is? |
| 18:34.47 | lirakis | generalhan: that would show the contents of callfilename on the cli |
| 18:34.57 | generalhan | lirakis: yes... the issue is not with the filename itself ... it is with the fact that MixMonitor will not create the direcotry for the calls |
| 18:35.05 | Mercestes | Netgeeks: Because I can batch it after hours isntead of doing it on the fly and sucking up IOs in the middle of th eday |
| 18:35.17 | generalhan | lirakis: i am not an asterisk guru by any means ... but i am no noob |
| 18:35.32 | lirakis | generalhan: ah.. so .. you need dynamic directories.. |
| 18:35.33 | lirakis | hmm |
| 18:36.20 | Netgeeks | Mercestes: ah makes sense |
| 18:36.31 | generalhan | lirakis: I work for a law firm ... call recordings are priority #1, i need to be able to pull any employees call at any given time ... so labeling them in proper directories is very important |
| 18:37.11 | generalhan | so i label them by EXTEN/DATE/INCOMING(or OUTGOING)/CID.wav |
| 18:37.22 | lirakis | generalhan: i gotcha.. why dont you just have the filebase be different... like src-dst-date |
| 18:37.40 | Mercestes | generalhan, are you using linux? |
| 18:37.49 | lirakis | generalhan: or .. accountcode.. |
| 18:37.50 | generalhan | Mercestes: CentOS 4.5 |
| 18:37.54 | Netgeeks | if you wanted to use mixmonitor, then just preface the mixmonitor call with System(mkdir $[path name}) |
| 18:37.57 | Mercestes | generalhan, then why are you doing that? |
| 18:38.10 | Mercestes | generalhan, just make it a filename exten_date_incoming/outgoing_cid.wav |
| 18:38.29 | Mercestes | I mean, it's not like your using DOS. |
| 18:38.48 | lirakis | generalhan: yeah .. if you have all taht info .. i dont know why it must be split out by directory... |
| 18:39.25 | *** join/#asterisk sakic (n=sakic@adsl-1-172-32.clt.bellsouth.net) |
| 18:39.43 | BSD_Tech | cool |
| 18:39.44 | sakic | does anyone have a good instrucational page about provisioning a polycom phone? |
| 18:39.51 | Mercestes | I mean, you can ls *a*b*c*d*e**f*g.*.* |
| 18:39.51 | BSD_Tech | I just got the best news |
| 18:39.52 | generalhan | because ... lets say the attorney wants me to pull a certain persons calls for an entire day ... all i have to do is go into that directory ... if they were all in one directory i would have to do searches for certain criteria and go through MILLIONS of calls |
| 18:40.04 | Mercestes | ok ls *exten*date*.* |
| 18:40.07 | BSD_Tech | the new card will be release in the next 2 weeks |
| 18:40.19 | lirakis | generalhan: yeah.. what Mecestes said... |
| 18:40.24 | Mercestes | or even ls *exten*date*outgoing*.* or ls *exten*date*incoming*.* |
| 18:40.30 | lirakis | generalhan: greping for files is cake |
| 18:40.37 | BSD_Tech | a opensource 4port fxs/fxo jumper configurable |
| 18:40.47 | BSD_Tech | for 250 dollars |
| 18:41.00 | generalhan | nah ... SOO much easier to open WinSCP or any other graphical SCP/SSH tool and open their directory |
| 18:41.19 | Mercestes | generalhan, if it was so much easier you wouldn't be here right now asknig us for help |
| 18:41.24 | lirakis | generalhan: ... |
| 18:41.25 | lirakis | lol |
| 18:41.26 | Mercestes | get out of windows. linux is a very nice OS. |
| 18:41.36 | BSD_Tech | linux = bloat |
| 18:41.40 | lirakis | Mercestes: thats a bit extreme |
| 18:41.41 | BSD_Tech | use BSD |
| 18:41.55 | Mercestes | lirakis, I am extreme, baby! |
| 18:41.56 | Mercestes | :D |
| 18:41.57 | generalhan | Mercestes: all due respect ... im not asking for help at all ... i only asked if MixMonitor had issues creating directories ... just looking for all my options |
| 18:41.58 | *** join/#asterisk ricky (i=ricky@fedora/ricky) |
| 18:42.04 | lirakis | BSD_Tech: .. i ran FreeBSD for my desktop for a while.. but it just isnt as good a desktop as linux |
| 18:42.14 | Mercestes | yay, DISTROWARS! |
| 18:42.24 | lirakis | BSD_Tech: even with linux binary compatability.. browser plugins are a pain .. and just random stuff like that |
| 18:42.31 | Netgeeks | and if he stuck them all in one directory he would run out of inodes in that directory eventually |
| 18:42.44 | Mercestes | generalhan, heh, ok, sorry. ;) Just saying....I don't think what yoru trying to do is easier than ls. |
| 18:42.53 | generalhan | and i LOVE linux, i really do ... but in an environment where i have to teach people how to turn on their computers i would rather stick to a M$ OS that more people are familiar with |
| 18:43.05 | BSD_Tech | to many derivatives of linux and all claiming they are the best |
| 18:43.09 | lirakis | BSD_Tech: dont get me wrong though.. BSD is light and fast.. and way secure.. especially with a non executable stack |
| 18:43.13 | Mercestes | generalhan, however...if you write a script that does a test -r 'directory/you/want' in an if case and mkdir if it fails and do nothing otherwise...... |
| 18:43.28 | Mercestes | generalhan, then you can just system(yourscript) and proceed onwards with your dynamic directory structure. |
| 18:43.46 | Mercestes | lirakis, BSD is secure because it's poorly documented and no one can figure out how it works. |
| 18:43.58 | lirakis | Mercestes: you are foolish |
| 18:44.03 | Mercestes | lol |
| 18:44.06 | BSD_Tech | bullshit |
| 18:44.07 | generalhan | well the m flag for Monitor works ... and does create direcotories for me ... i just hope that after 3 years of running it doesnt start to mess up like the last server did !!! |
| 18:44.30 | lirakis | generalhan: the m flag just mixes the audio legs.. it has nothing to do with directories |
| 18:44.33 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 18:44.39 | BSD_Tech | its very wwell documneted and its verysecure because we have devs who bust thier asses to secure it |
| 18:44.46 | BSD_Tech | and work on bugs |
| 18:45.11 | generalhan | lirakis: lol ... you are KILLING me ... the m flag does what MixMonitor does without flags. the directory comment was in regards to Monitor being able to create them |
| 18:45.23 | lirakis | .. its funny how much an identity an OS can be |
| 18:45.59 | Mercestes | gentoo don't identify with my OS. |
| 18:46.00 | Mercestes | ... |
| 18:46.03 | Mercestes | s/gentoo/I/ |
| 18:46.49 | lirakis | Mercestes: ..? what? |
| 18:46.58 | Mercestes | nothing. |
| 18:47.16 | lirakis | im running gentoo here.. love it.. ;) |
| 18:47.34 | lirakis | okay.. yeah.. enough distro / os crap |
| 18:48.16 | *** join/#asterisk _Simon[work] (n=IRC@i216-58-40-193.cybersurf.com) |
| 18:48.17 | Mercestes | You said "how much an identity an os can be." so I replaced "I" with gentoo to be ironic |
| 18:48.45 | lirakis | ooo find replace .. sorry i get it |
| 18:49.11 | _Simon[work] | anyone know a good provider I can get voip softphone service from? just something cheap it won't be used much :) |
| 18:49.23 | Mercestes | Jbot was supposed to pick up from there and correct it for me. Would've been funnier that way |
| 18:49.25 | lirakis | im used to doing it in vim :%s/gentoo/I/g |
| 18:49.26 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 18:49.27 | _Simon[work] | would like an ottawa incoming # |
| 18:49.50 | generalhan | Mercestes: i was sondering about that when i saw it ... usually he does |
| 18:49.53 | Mercestes | ~ITSP |
| 18:49.53 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 18:50.02 | generalhan | s/sondering/wondering |
| 18:50.05 | Mercestes | generalhan, I've gotten so bad that jbot has /ignored me. |
| 18:50.12 | generalhan | hahaha |
| 18:50.15 | generalhan | lol |
| 18:50.18 | Mercestes | :D |
| 18:50.25 | generalhan | s/lol/LOL/g |
| 18:50.25 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 18:50.36 | lirakis | ok .. enough find replace.. |
| 18:50.37 | lirakis | haha |
| 18:50.39 | generalhan | hmm need the global option ? |
| 18:50.55 | generalhan | s/hmm/hmmmm/ |
| 18:51.06 | generalhan | Mercestes: wow ... he just really doesnt like you anymore !!! |
| 18:51.11 | lirakis | haha |
| 18:51.13 | Mercestes | Indeed. |
| 18:51.41 | Mercestes | ~8ball Will I get lucky with Katty? |
| 18:51.41 | jbot | Sure. Yeah, exactly. |
| 18:51.46 | Mercestes | ... |
| 18:51.50 | generalhan | lol |
| 18:51.51 | Mercestes | ~8ball Will I get lucky with Corydon? |
| 18:51.52 | jbot | I'm not sure. |
| 18:51.56 | Mercestes | YES! |
| 18:51.59 | lirakis | ha ha |
| 18:52.02 | lirakis | sigh.. |
| 18:52.52 | _Simon[work] | god those plans are expensive, I will barely be making any outgoing calls, want it mostly for voicemail lol |
| 18:53.05 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:53.33 | generalhan | well i just need to find a more stable way to do my queues so i dont have to worry about my phone system anymore ! |
| 18:53.38 | lirakis | _Simon[work]: its tough now to get a "cheap" provider.. most have stopped selling all you can eat plans... |
| 18:53.55 | _Simon[work] | well I don't want all you can eat really because I will be mostly just using voicemail |
| 18:53.57 | lirakis | _Simon[work]: i got telasip "back in the day" |
| 18:54.06 | lirakis | _Simon[work]: it was $9.95 /mo |
| 18:54.11 | _Simon[work] | so most of these plans offer so much calling etc |
| 18:54.16 | lirakis | _Simon[work]: i think its more now.. |
| 18:54.16 | _Simon[work] | I don't want it for calling really |
| 18:54.28 | lirakis | _Simon[work]: well inbound DID's are cheap |
| 18:54.30 | _Simon[work] | just an incoming # for voicemail |
| 18:54.54 | lirakis | _Simon[work]: you can often get those for free.. but free=unreliable.. and bound to disappear |
| 18:55.06 | lirakis | _Simon[work]: just look around for any SIP DID provider |
| 18:55.12 | _Simon[work] | well I don't mind paying a bit for it however a 20 dollar plan isn't useful for me |
| 18:55.21 | _Simon[work] | because I won't use it to its cost |
| 18:55.31 | _Simon[work] | just want an ottawa incoming # for voicemail |
| 18:55.48 | lirakis | _Simon[work]: you have a asterisk box? or you want this to go strait to a softphone? |
| 18:56.08 | _Simon[work] | to a soft phone, I would hate to have to setup all of asterisk just for this purpose lol |
| 18:56.18 | lirakis | _Simon[work]: then you will have to pay for a plan. |
| 18:56.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-101-116.red.bezeqint.net) |
| 18:56.46 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:57.06 | _Simon[work] | that sucks. need to find a cheap plan lol |
| 18:57.18 | lirakis | _Simon[work]: telasip still sells a $9.95 plan.. |
| 18:57.21 | lirakis | i just checked |
| 18:57.29 | wothinn | Is this just for you to dial in or for you to give other people to dial in? |
| 18:57.37 | sakic | does anyone have a good instrucational page about provisioning a polycom phone? |
| 18:57.41 | _Simon[work] | to give other people to dial in, to leave msgs |
| 18:58.10 | wothinn | Aah, well so much for my SIPBroker idea, then. :) |
| 18:58.20 | _Simon[work] | whats SIPBroker? |
| 18:58.29 | wothinn | http://www.voip-info.org/wiki/view/SipBroker |
| 18:58.42 | ewaldo | sakic: http://www.voip-info.org/wiki-Polycom%20Phones |
| 18:59.08 | _Simon[work] | interesting, but not what I need lol |
| 18:59.37 | hmmhesays | anyone know how callid field is calculated in sip? |
| 18:59.55 | sakic | do I need to allow my linux box to act as dhcp? |
| 19:00.18 | ewaldo | sakic: Do you currently have a DHCP server? |
| 19:00.27 | sakic | yes my netopia router does it |
| 19:00.44 | ewaldo | sakic: Does your netopia router support Option 66? |
| 19:00.47 | _Simon[work] | is it hard if I purchase a DID to hook it up to asterisk on my linux box? |
| 19:00.54 | sakic | no clue, never even heard of it :) |
| 19:01.09 | ewaldo | sakic: Look in to it and if it doesn't, switch to dhcpd |
| 19:01.29 | sakic | I can turn off dhcp |
| 19:01.40 | ewaldo | You'll need Option 66 to specify your tftp://IP, otherwise you'll have to go to every phone and configure it for a TFTP or FTP server manually. |
| 19:04.09 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 19:05.04 | lirakis | _Simon[work]: .. its not "hard" but if you have never done it.. expect a weekend project.. and the start of a new hobby |
| 19:05.05 | lirakis | lol |
| 19:05.15 | _Simon[work] | I had an asterisk server running before |
| 19:05.33 | _Simon[work] | I still have it, just would have to setup the DID to a username I have |
| 19:05.42 | lirakis | _Simon[work]: .. that .. is simple.. |
| 19:05.50 | DrAk0 | which softphone support g729 > |
| 19:05.52 | DrAk0 | ? |
| 19:05.53 | _Simon[work] | I don't know how to do DID's and point them to users though |
| 19:06.15 | ewaldo | DrAk0: xten eyebeam does |
| 19:06.25 | DrAk0 | ewaldo, and xlite? |
| 19:06.32 | lirakis | _Simon[work]: you would define a sip account in sip.conf for you DID provider.. in there you would specify the context ... context=myprovider_in |
| 19:06.46 | _Simon[work] | I was using iax I beleive |
| 19:06.48 | lirakis | .. then in extensions.conf you would have that context [myprovider_in] |
| 19:07.00 | ewaldo | DrAk0: Not sure, wouldn't make sense since xlite is free and there's licensing costs for G.729 |
| 19:07.11 | DrAk0 | yes |
| 19:07.15 | lirakis | _Simon[work]: .. good luck getting an IAX DID provider |
| 19:07.19 | DrAk0 | true |
| 19:07.29 | _Simon[work] | at the time I was using GloboTech |
| 19:07.36 | *** join/#asterisk sysreq` (n=sysreq@197.236-ppp.3menatwork.com) |
| 19:08.01 | lirakis | _Simon[work]: whatever.. it works very similar.. |
| 19:08.35 | lirakis | _Simon[work]: to finish... in [myprovider_in] ... you have exten => yourdid,1,DIAL(SIP/1234) |
| 19:09.15 | lirakis | so .. when your provider routes a call to you .. it will go to that context and dial extension 1234 |
| 19:12.58 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 19:18.46 | [TK]D-Fender | DrAk0: Do you really NEED G.729? |
| 19:19.19 | DrAk0 | [TK]D-Fender, yeah , we talk about that few days ao |
| 19:19.22 | DrAk0 | ago* |
| 19:19.34 | DrAk0 | ppl is being forced to use skype |
| 19:19.40 | cpm | being forced? |
| 19:19.45 | cpm | with a gun to their heads? |
| 19:19.45 | DrAk0 | cuz they can't call in the noon (bw problem) |
| 19:20.08 | [TK]D-Fender | DrAk0: if its that bad, you're probably screwed either way. |
| 19:20.20 | [TK]D-Fender | DrAk0: Go buy Eyebeam |
| 19:20.46 | Aces1Up | anyone know how i reset a channel variable so that they is nothing in it like a null? |
| 19:21.35 | *** join/#asterisk tomcontr3 (n=tomcontr@200.55.198.106) |
| 19:21.43 | tomcontr3 | hi, anyone here using MeetMe? |
| 19:22.53 | [TK]D-Fender | Aces1Up: Set(var=) |
| 19:23.13 | [TK]D-Fender | tomcontr3: wide open & silly question. DUH, of course there are |
| 19:24.09 | tomcontr3 | I just want to know if I can install MeetMe without affecting the FreePbx instalation? |
| 19:25.28 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 19:25.29 | [TK]D-Fender | tomcontr3: First, this isn't #freepbx, and you are clearly missing a clue |
| 19:25.45 | [TK]D-Fender | tomcontr3: MeetMe is just a dialplan app taht COMES WITH * |
| 19:25.55 | [TK]D-Fender | tomcontr3: you don't "install" ANYTHING |
| 19:25.59 | Dan0maN_Work | ~book |
| 19:26.00 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:26.03 | tomcontr3 | oh |
| 19:26.15 | tomcontr3 | I was confused with WebMeetMe then |
| 19:26.32 | [TK]D-Fender | Dan0maN_Work: Whats the point, he's on FreePBX, he clearly has no inclination to get his hands dirty and learn much... |
| 19:26.50 | [TK]D-Fender | tomcontr3: You are DEFINATELY in the wrong channel.. |
| 19:26.50 | Dan0maN_Work | D-Fender: sorry. was getting the link for a friend of mine |
| 19:27.58 | *** join/#asterisk rhiliam (n=user@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 19:28.50 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:29.29 | ewaldo | Okay, I have installed a fresh 1.4.9 installation with sample configs and tested out the Voicemail email feature and it works just fine. I had to upgrade from a trixbox 1.2.22 system to 1.4.9 and have preserved the original configs. Everything works fine except that the voicemail does not generate an email. After analyzing configuration, I can tell that it hands off to the VoiceMail() and... |
| 19:29.30 | ewaldo | ...from what I understand, as long as voicemail.conf is set up corrently, asterisk should directly generate an email through sendmail as long as attach=yes and shouldn't be dependent on any other configuration set by freePBX or anything else. My sendmail logs show nothing attempting to send mail through the server. I've even tried setting the mailcmd= to something like touch /tmp/test and... |
| 19:29.32 | ewaldo | ...that command is never executed (because /tmp/test doesn't exist after I leave a voice message). Is there any other configuration that would prevent an email from being generated directly from the VoiceMail function within asterisk? |
| 19:30.17 | rhiliam | I have an analog installation that is causing my client a bit of grief. When a person calls in, the ringing sound in 'normal' for the first ring, then gets really loud when asterisk takes over. Is this normal and/or anyway to correct this? |
| 19:31.11 | [TK]D-Fender | rhiliam: Sounds like the gain to your * system is hig. What are you using for PSTN access? |
| 19:31.34 | rhiliam | zap TDM card |
| 19:31.43 | rhiliam | FXO modules |
| 19:32.18 | rhiliam | the gain is currently set to whatever the default is - I haven't changed anything |
| 19:32.21 | tzafrir_laptop | ewaldo, you use a complex dialplan desinged for asterisk 1.2 and expect it to just work for 1.4? There are some changes needed. So are you sure that the dialplan is not causing you issues here? |
| 19:34.05 | ewaldo | tzafrir_laptop: from the asterisk console, I am certain that it executes the command VoiceMail("SIP/100-b77119d0", "100@default|su") |
| 19:35.15 | ewaldo | tzafrir_laptop: I actually didn't expect it to work for 1.4, but I tried it last week and thought I had tested everything after the upgrade and everything worked and I was pleasantly surprised. Unfortunately the one thing I didn't test was voicemail. Quite an oversight I must admit. |
| 19:35.54 | tzafrir_laptop | ok: does it leave a message in the mailbox? |
| 19:36.04 | ewaldo | tzafrir_laptop: yes it does |
| 19:36.28 | [TK]D-Fender | rhiliam: Well I guess you're going to have to start playing with rxgain & txgain in zapata.conf to normalize this. |
| 19:36.58 | rhiliam | k, so the issue you figure is with the gain settings? |
| 19:37.06 | rhiliam | anything else that may be the culprit |
| 19:37.16 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 19:37.16 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 19:37.30 | tzafrir_laptop | rx: what asterisk recieves, tx: what asterisk transmits (sends) |
| 19:37.50 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 19:38.27 | tzafrir_laptop | next: why are you sure that it is configured to attach messages? |
| 19:39.56 | ewaldo | tzafrir_laptop: because I have "100 => 0000,First Last Name,user@domain.tld,,attach=yes|saycid=no|envelope=no|delete=no" configured in voicemail.conf. If I change delete=no to delete=yes it removes the message so it is definately processing those flags. |
| 19:40.07 | tzafrir_laptop | can you move aside the current voicemail.conf and create a simple one that will attach? |
| 19:40.43 | ewaldo | tzafrir_laptop: I can definately try that, good idea. |
| 19:41.05 | tzafrir_laptop | also: is that really the entry for 100? try changing the password and see if it applies |
| 19:41.50 | tzafrir_laptop | and is it in the context [default] ? |
| 19:41.57 | ewaldo | tzafrir_laptop: I zero'd out the password before posting |
| 19:42.14 | rhiliam | k, another issue: My client would like to have the option of when a person leaves a voicemail, they press a number (whatever number) and then once pressed, * will dial his paging service. Any ideas of how to set this up? From my reading, it appears call f |
| 19:42.17 | ewaldo | tzafrir_laptop: yes, it is in [default] |
| 19:43.12 | rhiliam | sorry, call files, but wondering if there is a better way to do this? |
| 19:43.26 | tzafrir_laptop | ewaldo, what I mean is: edit the password in voicemail.conf, reload (what exactly?) and see if your change applies to mailbox 100 |
| 19:43.35 | [TK]D-Fender | rhiliam: Go read up on Asterisk Standard Extensions. This is the purpose of the "a" Standard Extension. |
| 19:43.39 | *** join/#asterisk guillote_GNU (n=bancaria@host191.190-31-26.telecom.net.ar) |
| 19:43.48 | rhiliam | thx |
| 19:44.19 | *** part/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 19:44.35 | Aces1Up | i am trying to clear a channel variable through agi. and i can't seem to do it. |
| 19:44.41 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:45.11 | Aces1Up | i don't want to set it to a 0, just want it to be clear like when the scripts first starts. |
| 19:45.36 | Aces1Up | the variable is getting set and then it is being misintrepted by my code. so i need to clear it somehow. |
| 19:45.41 | ewaldo | tzafrir_laptop: yes, changing the password in voicemail.conf required me to use the new password when I checked my voice mailbox |
| 19:46.10 | *** join/#asterisk ixela (i=ixela@nat/digium/x-315e735ac347fba6) |
| 19:48.42 | rhiliam | voip-info.org, apparently is having issues - the page about the a extension appears to be AWOL - can you suggest another location for this? |
| 19:50.23 | GlubZor | hop |
| 19:53.10 | ewaldo | tzafrir_laptop: I slimmed down voicemail.conf to just the context and my voice mailbox line and have the same results. |
| 19:54.59 | tzafrir_laptop | ewaldo, if you set attach=yes in the [general] section, does it change anything? |
| 19:55.42 | *** join/#asterisk CoderCR (n=creyna@wsip-68-15-30-74.sd.sd.cox.net) |
| 19:55.53 | ewaldo | tzafrir_laptop: it is set that way by default in an include file, but let me slim down my voicemail.conf again and add that in specifically |
| 19:56.50 | *** part/#asterisk CoderCR (n=creyna@wsip-68-15-30-74.sd.sd.cox.net) |
| 19:57.13 | ewaldo | tzafrir_laptop: Nope, didn't work and sendmail logs still show nothing. |
| 19:58.31 | *** join/#asterisk ManxPower (n=manxpowe@015-798-018.area5.spcsdns.net) |
| 19:58.44 | *** join/#asterisk icel (n=icel@63.78.162.19) |
| 19:59.58 | ewaldo | http://pastebin.com/d1bb0dd6e |
| 20:02.02 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 20:02.22 | tzafrir_laptop | ewaldo, I guess that the next step would be to check in app_voicemail.c for near-by debug messages |
| 20:03.28 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:04.15 | ewaldo | tzafrir_laptop: Well crap, I guess the debugs sent from core level 10 aren't sufficient for this particular problem? |
| 20:04.34 | *** join/#asterisk CrashHD (n=crashhd@70.96.98.65) |
| 20:04.49 | CrashHD | Hello |
| 20:04.57 | CrashHD | where can I find an asterisk voicemail tree guide? |
| 20:05.49 | *** join/#asterisk ChicagoBud (n=Bud@adsl-76-203-90-142.dsl.chcgil.sbcglobal.net) |
| 20:07.36 | javar | :O |
| 20:07.36 | Qwell[] | saw that one coming |
| 20:09.09 | lirakis | netsplit?ddos? |
| 20:09.58 | *** join/#asterisk sakic (n=sakic@adsl-227-157-225.clt.bellsouth.net) |
| 20:10.26 | Qwell[] | and here comes another |
| 20:10.28 | *** join/#asterisk marcan (i=1337@10.Red-88-24-250.staticIP.rima-tde.net) |
| 20:10.37 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) [NETSPLIT VICTIM] |
| 20:10.53 | sakic | I am trying to provision my polycom via the wiki, and it says unzip polycom files, what files???!!! |
| 20:11.29 | *** join/#asterisk SplasPood (n=jwb@69.9.183.138) [NETSPLIT VICTIM] |
| 20:11.32 | *** join/#asterisk [X-tp] (n=xtp@c-c19e70d5.015-136-6b736410.cust.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 20:11.33 | tzafrir_laptop | ewaldo, do you see the message: Attaching file '%s', format '%s', uservm is '%d', global is %d |
| 20:11.57 | *** join/#asterisk jsin (n=jason@adsl-68-77-81-142.dsl.ipltin.ameritech.net) |
| 20:12.13 | datachomper | anybody know a media player for linux that can play .ul or .gsm files? |
| 20:13.07 | [TK]D-Fender | sakic: The ones from the SIP firmware ZIP file. |
| 20:13.09 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) [NETSPLIT VICTIM] |
| 20:13.12 | *** join/#asterisk dg (i=dgl@otherwize.co.uk) |
| 20:13.27 | sakic | tk-d where are they? |
| 20:13.30 | *** join/#asterisk Sweeper (i=sweeper@208.64.37.147) [NETSPLIT VICTIM] |
| 20:13.40 | lirakis | datachomper: sox |
| 20:13.50 | sakic | http://www.freedomphones.net/polycom/files/ ?? |
| 20:13.58 | lirakis | datachomper: its not a "player" |
| 20:14.09 | lirakis | datachomper: but it can make things work for you |
| 20:14.24 | sakic | and which one.. there are so many :P |
| 20:14.49 | lirakis | datachomper: http://www.linuxcommand.org/man_pages/sox1.html |
| 20:15.54 | datachomper | Ah, "Play" is what I was looking for |
| 20:16.08 | *** join/#asterisk wothinn (n=Allfathe@209.91.190.26) [NETSPLIT VICTIM] |
| 20:16.20 | ewaldo | tzafrir_laptop: yeah I see it when option_debug > 2....does core debug 10 turn on option debugging 3 or better? |
| 20:16.57 | *** join/#asterisk Cardoe (n=cardoe@gentoo/developer/Cardoe) |
| 20:17.03 | lirakis | datachomper: FYI play is a front end to sox |
| 20:17.50 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) [NETSPLIT VICTIM] |
| 20:18.02 | *** join/#asterisk kimosabe (n=kimosabe@189.175.44.143) |
| 20:18.48 | kimosabe | room i rella need help from some one with codecs i canot dial via my sip becuase i only continuouslly get dial tone |
| 20:19.01 | Cardoe | I'm having an issue with my SIP setup I believe |
| 20:19.18 | Cardoe | basically you can't dial in for a small chunk of time and then it'll start working.. then it won't |
| 20:19.25 | *** join/#asterisk xipi (n=oliver@94.pool85-60-21.dynamic.orange.es) |
| 20:19.29 | xipi | hi |
| 20:19.32 | Cardoe | basically the SIP provider says "the customer you are trying to reach can't be reached" |
| 20:19.35 | *** join/#asterisk keith4 (n=keith@128.180.3.104) [NETSPLIT VICTIM] |
| 20:19.42 | Cardoe | and my console is being filled with this every 20 seconds.. |
| 20:19.45 | Cardoe | -- parse_srv: SRV mapped to host alpha1.callcentric.com, port 5060 |
| 20:19.50 | *** join/#asterisk Telamon (n=telamon@blk-222-160-100.eastlink.ca) |
| 20:19.58 | *** part/#asterisk Telamon (n=telamon@blk-222-160-100.eastlink.ca) |
| 20:20.01 | Cardoe | I'm behind a NAT but I've opened up port 5060, and the RTP ports to my asterisk box |
| 20:20.23 | xipi | how can i get the voicemail system to give the caller the choice to review the message, rerecord or accept it? |
| 20:20.33 | icel | Has anyone here done any high availability solutions? |
| 20:20.41 | xipi | i am using 1.4.9 |
| 20:20.51 | kimosabe | how can i make my codecs be ulaw or alaw on my sip devices |
| 20:21.05 | [TK]D-Fender | sakic: Thats one place, though you should be getting them from your reseller. |
| 20:21.11 | [TK]D-Fender | sakic: What phones do you have? |
| 20:23.16 | Cardoe | [TK]D-Fender: any suggestions to what I'm screwing up? |
| 20:23.35 | xipi | i tried to find a solution via google, but haven't found any and don't know, what else to search for |
| 20:23.43 | xipi | any suggestions? |
| 20:24.15 | [TK]D-Fender | Cardoe: Here : |
| 20:24.16 | [TK]D-Fender | ~sipnat |
| 20:24.17 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:24.28 | [TK]D-Fender | Cardoe: You have to do a fair bit more than just port forwarding |
| 20:24.48 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) [NETSPLIT VICTIM] |
| 20:24.58 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:25.14 | [TK]D-Fender | xipi: There is a complete list of the options in the sample voicemail.conf that * comes with. Go read. |
| 20:25.28 | kimosabe | anyone here how can i configure my spa 2000 to use the pstn from the spa3000 while registered to asterisk |
| 20:25.50 | xipi | [TK]D-Fender: thank you. i will have a look |
| 20:25.59 | xipi | [TK]D-Fender: hopefully this will help |
| 20:26.23 | xipi | [TK]D-Fender: up to now, the caller can only leave a message and the the system will hangup |
| 20:26.35 | xipi | [TK]D-Fender: i'll check it out |
| 20:27.44 | kimosabe | can i get one of the admins here to help me out please |
| 20:27.59 | Dan0maN_Work | does anyone have any definity pbx experience? i'm needed to find out what the most number of active calls we've had was. (i know it's not *, but there's alot of experience in this channel) |
| 20:28.07 | kimosabe | ive been at this problem for 8 days now and i havent been abke to get a clue to fix it |
| 20:28.15 | sakic | hmm it says cannot contact boot server on my polycom, any idea of problem? |
| 20:28.37 | tzafrir_laptop | ewaldo, do you log debug messages? (see logger.conf) |
| 20:28.47 | *** join/#asterisk PC_Clone (n=pc_clone@165.sub-75-204-96.myvzw.com) |
| 20:29.19 | *** part/#asterisk pagec (n=pagec@cpe-74-73-191-68.nyc.res.rr.com) |
| 20:29.54 | ewaldo | tzafrir_laptop: notice, warning, error, debug, verbose. Looking through the full log now for that Attaching file message as I type. |
| 20:30.25 | tzafrir_laptop | ewaldo, fine. So that specific part was not called |
| 20:31.07 | [TK]D-Fender | sakic: Yes, you have not pointed it to a valid boot server. |
| 20:31.21 | sakic | yeah I think I don't have ftp setup right |
| 20:31.55 | ewaldo | tzafrir_laptop: http://pastebin.com/d79ae4431 |
| 20:33.20 | ewaldo | tzafrir_laptop: If I manually type /usr/sbin/sendmail -t from the command-line, it shows up in the log. But when I leave a voicemail, nothing shows in the sendmail log. |
| 20:34.29 | [TK]D-Fender | heading home, later all. |
| 20:34.58 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 20:36.39 | Cardoe | [TK]D-Fender: is it ok to have parse_srv splitting out a line every 20 seconds? |
| 20:36.54 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:38.55 | ewaldo | tzafrir_laptop: Interestingly enough, if I look through the log, I see "[Aug 6 11:44:16] DEBUG[21571] app_voicemail.c: Sent mail to ewaldo@thnetwork.net with command 'touch /tmp/test'" When I had made mailcmd= to that. So I guess it comes down to the fact that asterisk is unable to because of system configuration or incapable of because of who knows what, to actually execute system commands. |
| 20:39.06 | xipi | it's working now :) |
| 20:39.30 | ewaldo | tzafrir_laptop: any ideas why it would be incapable of doing so? |
| 20:39.40 | thansen|laptop | any automon experts around? |
| 20:40.28 | errr | is there a way to show what time it is according to asterisk? |
| 20:41.26 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 20:41.55 | ewaldo | tzafrir_laptop: and before it's asked, yes SELinux is disabled :P |
| 20:43.14 | errr | when I run date the time is correct but the time stamps in my voicemails are like 4 hours off |
| 20:43.38 | *** join/#asterisk marcan (i=1337@10.Red-88-24-250.staticIP.rima-tde.net) |
| 20:46.13 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:46.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-101-116.red.bezeqint.net) [NETSPLIT VICTIM] |
| 20:46.31 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-206-171-23.perm.iinet.net.au) [NETSPLIT VICTIM] |
| 20:46.31 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) [NETSPLIT VICTIM] |
| 20:46.31 | *** join/#asterisk snuff-work (n=na@61.29.30.137) |
| 21:00.55 | kimosabe | is anyone here using spa3000 to pstn with asterisk |
| 21:01.19 | sakic | ack I get sip.ld is not compatible with phone |
| 21:02.02 | icel | has anyone used BioCluster? |
| 21:03.58 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 21:05.05 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:05.21 | sakic | anyone know why I get sip.ld is not compatible with phone |
| 21:06.44 | Qwell[] | sakic: My guess is because it isn't compatible with the phone |
| 21:07.52 | sakic | got that part |
| 21:08.24 | karleeto | kimosabe: whats the spa3000? |
| 21:09.22 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 21:09.52 | [TK]D-Fender | sakic, What other part is there? |
| 21:09.59 | [TK]D-Fender | sakic, And what model do you have? |
| 21:12.39 | mercestes | Hey, who wants an example of the WORST IVR EVER????? |
| 21:13.02 | mercestes | 1-888-704-3515 |
| 21:13.49 | zeenix | what is an IVR btw? |
| 21:14.10 | Cardoe | zeenix: interactive voice response |
| 21:15.36 | *** join/#asterisk b11d (n=no@134.29.200.234) |
| 21:16.33 | *** join/#asterisk PC_Clone (n=pc_clone@165.sub-75-204-96.myvzw.com) |
| 21:16.41 | [TK]D-Fender | ~ivr |
| 21:16.41 | jbot | extra, extra, read all about it, ivr is Interactive Voice Response |
| 21:17.08 | [TK]D-Fender | zeenix, menus in a phone system, etc |
| 21:19.28 | kimosabe | looking for some one to help me out please with spa3000 problem cant use pstn via asterisk |
| 21:21.55 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 21:23.55 | [TK]D-Fender | kimosabe, go pastebin your configs and host up some screen-shots of your SPA's config |
| 21:27.10 | Cardoe | [TK]D-Fender: I followed those NAT setup rules.. I had my system already configured like that.. the trick that seemed to fix it was srvlookup=no, however I realize that's a hack |
| 21:29.59 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:30.23 | Cardoe | [TK]D-Fender: is there anything else I'm overlooking? |
| 21:31.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-101-116.red.bezeqint.net) [NETSPLIT VICTIM] |
| 21:31.20 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-206-171-23.perm.iinet.net.au) [NETSPLIT VICTIM] |
| 21:31.20 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) [NETSPLIT VICTIM] |
| 21:31.20 | *** join/#asterisk snuff-work (n=na@61.29.30.137) |
| 21:33.00 | *** part/#asterisk ewaldo (n=chatzill@206.80.91.195) |
| 21:33.09 | *** join/#asterisk ewaldo (n=chatzill@206.80.91.195) |
| 21:33.40 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 21:35.37 | fujin | moringin asterisk! |
| 21:35.45 | b11d|bbl | GOOD MORNING! |
| 21:35.47 | b11d|bbl | :) |
| 21:35.49 | b11d|bbl | now.. good night! |
| 21:35.52 | fujin | night! |
| 21:47.22 | *** join/#asterisk ricky (i=ricky@fedora/ricky) |
| 21:49.19 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:50.59 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 21:55.11 | *** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 21:57.01 | [T]ank | i am having issues getting asterisk to work with T1s that I know work. Here are my configs, my test call and the output. I get channel congestion with everything I try http://pastebin.ca/648015 |
| 21:58.00 | [T]ank | any help would be greatly appreciated. Been working on this now for a couple of weeks. |
| 21:59.46 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 21:59.58 | hmmhesays | so what kind of pc hardware would I need to run a quad t1 card? |
| 22:01.06 | Qwell[] | something with a pci slot that can push about 100 calls.. nothing too fancy |
| 22:01.19 | Qwell[] | You could probably get away with a P3 fairly easily |
| 22:02.18 | [T]ank | anyone able to take a look at my issue? |
| 22:04.38 | kimosabe | t ank can i pm u |
| 22:04.40 | kimosabe | please |
| 22:05.56 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 22:06.01 | hmmhesays | hmm did my last question show up? |
| 22:06.34 | generalhan | hmmhesays: yes, and Qwell[] replied ... |
| 22:06.39 | generalhan | ~seen Qwell[] |
| 22:06.43 | jbot | qwell[] is currently on #asterisk (2h 29m 27s). Has said a total of 5 messages. Is idling for 5m 24s, last said: 'You could probably get away with a P3 fairly easily'. |
| 22:06.51 | hmmhesays | can you paste it again I got booted |
| 22:07.12 | generalhan | unless you asked something again after that |
| 22:07.30 | hmmhesays | I asked if you could get 2 quad pri cards into one box |
| 22:07.42 | generalhan | ohh ... then we did NOT get that one ! sorry ! |
| 22:07.59 | hmmhesays | i haven't used any zaptel hardware in forever so I don't know anymore |
| 22:08.10 | *** join/#asterisk xipi (n=oliver@94.pool85-60-21.dynamic.orange.es) |
| 22:08.33 | generalhan | hmmhesays: i wouldnt take my word on ANYTHING ... but i dont see why not, if the system can handle pushing 8 sets of lines |
| 22:08.44 | xipi | is there any way to record ivr messages by phone? |
| 22:09.13 | *** join/#asterisk sysreq` (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:09.18 | hmmhesays | i'm curious what kind of system requirements we'd need for 8 pri's |
| 22:09.47 | generalhan | hmmhesays: what kind of MONDO-call room you got going on over there ? lol |
| 22:10.02 | hmmhesays | a ds3 |
| 22:10.03 | Juggie | hmmhesays, if theres no transcoding then not a hell of alot |
| 22:10.06 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 22:10.19 | hmmhesays | sip ->tdm g729 the whole way |
| 22:10.29 | kimosabe | is asterisk 1.0.9 old |
| 22:10.33 | hmmhesays | yes |
| 22:10.34 | Juggie | kimosabe, ancient. |
| 22:10.57 | kimosabe | juggie how do i update the asterisk version on my xircom |
| 22:11.14 | Juggie | hmmhesays, you want to convert 192 channels from ulaw/alaw to g729 in realtime? :P |
| 22:11.45 | hmmhesays | yeah that seems like a bad idea |
| 22:12.20 | hmmhesays | i'm guessing that is going to take some serious processing power |
| 22:12.22 | Juggie | well for starters it will cost you 192 x 10$ for the g729 licenses :> |
| 22:12.27 | _DAW | TC400B? |
| 22:12.34 | Juggie | or you can get the processing card |
| 22:12.40 | Juggie | but i'm not sure how many channels that can do |
| 22:12.45 | hmmhesays | i'll look it up |
| 22:12.47 | Qwell[] | I think 4 E1's worth |
| 22:13.07 | _DAW | 96 g.729 |
| 22:13.12 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:13.23 | pigpen | So lets say I have a Polycom 600, won't power up with the brick or a known working POE connection. Any resolve other than Warranty or trash can? |
| 22:13.26 | Juggie | there you go, 2 quad spans and two processing cards :) |
| 22:13.41 | Juggie | but i'd do two seperate pc's |
| 22:14.09 | Netgeeks | for redundancy sake at least |
| 22:14.10 | Juggie | not because it wont work just because :) i'd feel safer. |
| 22:14.31 | Netgeeks | better to lose half your circuits than all of them |
| 22:14.33 | hmmhesays | yeah how much processing power do I need though |
| 22:15.35 | Juggie | hmmhesays, not very much.. |
| 22:15.46 | Netgeeks | never used the processing card, so I can't guess how much load it takes oss |
| 22:15.47 | Netgeeks | off |
| 22:15.50 | Juggie | i mean i woudnt skimp on it, but you arnt going to need a quad xeon or anything |
| 22:16.12 | kimosabe | is there a way to upgrade asterisk on xorcom to the latest version help would be apreciated |
| 22:16.19 | Netgeeks | but if it was PRI to ulaw equivalent, then I can fit 400 channels of pri to ulaw on 1 core 2 duo if I could fit the cards in there |
| 22:16.28 | Qwell[] | kimosabe: sounds like a question for Xorcom |
| 22:16.37 | Qwell[] | (or maybe tzafrir will pipe in if you're lucky) |
| 22:16.59 | Juggie | hmmhesays, the only way to find out is to try :) |
| 22:17.08 | hmmhesays | yeah its expensive to try though |
| 22:17.24 | hmmhesays | so you guys think I could get buy with a core 2 duo with 8 pri's? |
| 22:17.46 | Netgeeks | with or without the processing card? with, yes, without, prolly not |
| 22:17.46 | Qwell[] | easilt |
| 22:17.50 | Qwell[] | easily even |
| 22:18.12 | Qwell[] | oh, g729 |
| 22:18.21 | Juggie | hmmhesays, for redundancy sake, i'd split it into two machines. |
| 22:18.25 | Netgeeks | yeah, ulaw would be no problem at all |
| 22:18.40 | hmmhesays | Qwell[] what if I was using the g729 card? |
| 22:18.44 | Qwell[] | easily |
| 22:19.37 | hmmhesays | cool |
| 22:20.13 | hmmhesays | how much ram do I need for such a setup? |
| 22:20.23 | russellb | 1 TRILLION TERABYTES |
| 22:20.33 | Netgeeks | per channel |
| 22:20.36 | Juggie | russellb, see pm. |
| 22:20.44 | hmmhesays | heh that much huh? |
| 22:20.49 | Juggie | hmmhesays, we dont know the answers to this stuff, you just will have to try :) |
| 22:20.59 | hmmhesays | right |
| 22:21.12 | Netgeeks | I would think 1G would be fine give you don't run X on the system |
| 22:21.13 | hmmhesays | I'm guessing russellb and Qwell[] probably have a good idea |
| 22:21.16 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 22:21.21 | Qwell[] | we really don't |
| 22:21.33 | hmmhesays | you don't have systems out there with many pri's in them? |
| 22:21.34 | Qwell[] | there are so many variables, it isn't even funny |
| 22:21.34 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:21.34 | russellb | Juggie: i have no pm from you |
| 22:21.37 | Netgeeks | We run 400 ulaw calls on a core 2 duo based blade with 2G mem and have zero problems |
| 22:21.45 | Qwell[] | hmmhesays: us? no |
| 22:21.51 | Qwell[] | somebody at Digium has done testing, sure |
| 22:22.03 | russellb | hmmhesays: i've heard of people running 4 quad span cards per box |
| 22:22.08 | russellb | we generally recommend only 2 |
| 22:22.25 | generalhan | Netgeeks: what kind of blade ? |
| 22:22.27 | Juggie | hmmhesays, if you buy asterisk be, they will guarantee a certain amount of channels will work yes. |
| 22:22.28 | _DAW | Netgeeks: Out of curiosity, how many CPS can that system handle? |
| 22:22.37 | Netgeeks | I agree with russell and juggie on the really should use 2 systems for redundancy |
| 22:22.47 | Juggie | russellb, sent again |
| 22:22.56 | hmmhesays | Netgeeks: that would be one quad span per box then |
| 22:23.08 | russellb | Juggie: see your main server window, you're likely not registered and getting denied |
| 22:23.20 | Netgeeks | _DAW, I'm not really sure what it's limits are, we run a couple call distributors which spread the calls over 12 blades. |
| 22:23.38 | Juggie | russellb, bingo, i had changed my nick and back :) |
| 22:23.39 | Netgeeks | the call distribs can see 30-40 cps at peak times |
| 22:24.19 | _DAW | nice |
| 22:24.22 | irule | someone here using phpagi? |
| 22:25.10 | CrashHD | Hello |
| 22:25.52 | *** join/#asterisk luke-jr_ (n=luke-jr@ip72-206-114-236.om.om.cox.net) |
| 22:26.40 | Netgeeks | most of our calls are international calls, and last longer than your average us to us call would last, we see alot more static load (concurrent call volume) over dynamic (cps) volume |
| 22:30.27 | *** join/#asterisk metfan2007 (n=metfan20@189.136.86.34) |
| 22:31.01 | metfan2007 | hi all! have you seen Jared around here?ยก |
| 22:31.06 | Sweeper | what's the best way to log an agent into a queue without any prompts? |
| 22:31.13 | Sweeper | in 1.4, that is |
| 22:34.55 | styelz | wow this FWD stuff is neat |
| 22:35.36 | styelz | found a free call pstn to voip -> fwd for my country too. sweet |
| 22:36.27 | styelz | does anyone know if FWD have voip to PSTN accounts? |
| 22:37.08 | styelz | guess i go my local voip -> FWD provider |
| 22:37.19 | styelz | to pstn |
| 22:41.08 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:41.50 | CrashHD | does anyone know where I can find an asterisk voicemail tree guide? |
| 22:42.09 | CrashHD | like a graphic or pdf etc... |
| 22:42.23 | hmmhesays | so it is worth the extra thousand dollars for the echo can card? |
| 22:44.42 | Netgeeks | I would buy cards with hardware echo cancelling just to save my sanity |
| 22:45.43 | mercestes | <PROTECTED> |
| 22:49.50 | bkruse | anyone here fimiliar with multi dimentional arrays in bash? |
| 22:49.55 | bkruse | particularly how to make them global to functions? |
| 22:51.13 | kimosabe | is there any one here familiar with the sipura 3000 model |
| 22:51.26 | mercestes | bkruse, did you try #bash?? |
| 22:51.39 | kimosabe | is there somewhere where i can donate some paypal dolars for help that is admited via the asterisk project |
| 22:51.44 | Qwell[] | bkruse: you need the quantum branch of bash to do multi-dimensional |
| 22:52.08 | JT | hmmhesays: yes |
| 22:52.29 | mercestes | kimosabe, Use the official paypal donation site. mercestes@hotmail.com |
| 22:52.39 | bkruse | mercestes: im in there now :D |
| 22:52.40 | lirakis | mercestes: lol |
| 22:52.44 | bkruse | they are helpful, and im an op in #bash :P |
| 22:53.10 | kimosabe | mercestes can i pm u |
| 22:53.20 | bkruse | kimosabe: pm support@digium.com. |
| 22:53.20 | mercestes | Sure....what could it hurt? |
| 22:53.39 | Netgeeks | What exactly did the Application StripLSD do? (I don't have a 1.2 or older version handy to check from the cli). ooking at this dialplan I'm guessing that it works against the current extension and removes the n least significant digits Where n is provided as the argument for it? |
| 22:53.39 | kimosabe | which is digium suport |
| 22:54.11 | mercestes | Netgeeks, it removes hallucenogenic acids from your dialplan. |
| 22:54.26 | Netgeeks | shit, my dial plan will stop working then |
| 22:54.47 | mercestes | Netgeeks, Yea, probably. StripLSD(1) should remove the 1 from the front o fa dial string if I recall recorrectly. |
| 22:55.14 | Netgeeks | thanks |
| 22:55.18 | mercestes | np. |
| 22:55.22 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 22:56.01 | Netgeeks | bah, so to replace it I'm goint to have to write some ugly mish mash using :x:y and LEN()... bah on lost functionality in the conversion to functions! |
| 22:56.49 | mercestes | maybe, depends on what the LSD's are. |
| 22:57.26 | fujin | Sweeper: agentcallbacklogin |
| 22:59.01 | *** part/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 22:59.16 | tako-san | Any recommendations for an ATA device? Make/model appreciated. |
| 23:00.37 | mercestes | taco-san, what are you doing with the ATA? |
| 23:01.11 | tako-san | trying to interface an analog fax device into the phone system (asterix) over cat5 |
| 23:02.39 | mercestes | Use a quintum. They are larger, have a metal casing, and will penetrate your skull faster when you bludgeon yourself with it. |
| 23:02.51 | fujin | jesus |
| 23:02.57 | bkruse | i hate bash forever |
| 23:02.58 | fujin | analog fax device = cry |
| 23:03.06 | fujin | bkruse: what's wrong with bash? |
| 23:03.14 | fujin | If you can't do it with bash, you're doing it wrong *or* using the wrong language |
| 23:03.19 | tako-san | fujin: So what is the alternative? |
| 23:03.20 | bkruse | fujin: global multidimentional arrays |
| 23:03.26 | fujin | You're using the wrong language. |
| 23:03.32 | bkruse | fujin: i cant use ioctl to access zaptel? |
| 23:03.34 | bkruse | in bash? |
| 23:03.36 | bkruse | :[ |
| 23:03.37 | mercestes | tako-san, PRI. FXO card. |
| 23:03.47 | mercestes | tako-san, anything but FoIP |
| 23:03.55 | bkruse | fujin: but you can! :D |
| 23:03.56 | fujin | foip isn't smart, even with t38 |
| 23:04.03 | fujin | bkruse: wrong language |
| 23:04.09 | bkruse | fujin: bah |
| 23:04.11 | fujin | can != should |
| 23:04.15 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 23:04.18 | bkruse | fujin: totally agreed. |
| 23:04.20 | tako-san | Mercestes: IP is only internally. The phone lines are all FXO (soon to be PRI) |
| 23:04.31 | fujin | Patch a line directly to your fax |
| 23:04.33 | fujin | save time/hastle. |
| 23:04.40 | tako-san | Mercestes: However the location of the fax machine is remote from the PBX |
| 23:04.46 | tako-san | fujin: I would if I could |
| 23:05.00 | mercestes | tako-san, Your still asking for trouble. |
| 23:05.02 | fujin | ring your telco and get an analogue line provisioned, save alll of the problems |
| 23:05.12 | mercestes | Cisco 186a most likely though |
| 23:05.17 | tako-san | Ok, thanks for the input guys |
| 23:05.50 | bkruse | ahh, figured it out |
| 23:06.13 | mercestes | I still advise against it, tako-san. strongly. |
| 23:06.26 | mercestes | You'd be better off running a 500' phone cord to an FXS card |
| 23:07.02 | tako-san | Mercestes: I have taken your comments to heart. I will see what management has to say. Thanks. :) |
| 23:07.08 | E-bola | our fax runs "ok" on our pbx |
| 23:07.09 | mercestes | Good luck |
| 23:07.15 | E-bola | but its always best to get an analog line if possible |
| 23:09.48 | drwelby | The error message: chan_sip.c:1245 retrans_pkt: Hanging up call XXXXX - no reply to our critical packet = phone isn't ack-ing back? |
| 23:10.18 | JT | tako-san: what is "asterix"? and never try and fax with fxo ports |
| 23:10.53 | antimoof | asterix is the little viking dude, I think. |
| 23:11.23 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 23:11.25 | JT | tako-san: sarcasm, you said it not me |
| 23:14.43 | JT | asterix is some cartoon |
| 23:15.26 | styelz | yep |
| 23:15.31 | *** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net) |
| 23:15.40 | drwelby | Asterix was a Gaul, wasn't he? |
| 23:15.59 | styelz | Astrix and Oblix |
| 23:16.10 | tako-san | I would prefer to say *is* a Gaul. ;) |
| 23:16.27 | antimoof | and he was split into three parts! |
| 23:16.47 | *** join/#asterisk jmav8 (n=jmav@201.243.61.136) |
| 23:17.32 | tako-san | He is an asterisk because he is small and the star of the story. |
| 23:17.33 | drwelby | Asterix, Obelisk, Cacafonix, Getafix, Vitalstatistix |
| 23:18.04 | thansen|laptop | functionally what's the difference between agi and writing your own application? |
| 23:19.57 | drwelby | Anyway, I have a Cisco 7940 that likes to hang up 20secs into VoiceMailMain |
| 23:20.57 | drwelby | It's not using NAT, either |
| 23:21.19 | JT | "i'm a cisco, i'm starving for MONEY!" |
| 23:21.39 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 23:21.40 | drwelby | JT: tell me about it |
| 23:22.19 | drwelby | I've been watching TCPdump for the last hour and it's never hiccuped on OPTIONS |
| 23:22.29 | *** join/#asterisk mirco (n=mirco@p54B25AEF.dip.t-dialin.net) |
| 23:22.37 | hmmhesays | so the echo can cards are worth the extra money huh? |
| 23:26.40 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 23:27.02 | JT | hmmhesays: this has already been said, but yes |
| 23:27.09 | JT | if you plan to connect to the PSTN |
| 23:27.27 | hmmhesays | yeah |
| 23:27.45 | *** join/#asterisk riddlebox (n=james@75.132.225.75) |
| 23:32.25 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:39.18 | styelz | the HPEC addon for zaptel fixed my local echo issues over my zap channel |
| 23:39.24 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 23:39.29 | styelz | worth the $10 |
| 23:39.30 | remmo | yumcha |
| 23:40.17 | styelz | it recisters the HPEC licence to your lan card mac |
| 23:41.38 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 23:41.51 | JT | too bad if you've got lots of channels, the CPU load would be insane |
| 23:42.09 | bkruse | JT: yes, it is |
| 23:42.10 | styelz | yer |
| 23:42.52 | *** join/#asterisk markgreene (n=markgree@209.12.142.2) |
| 23:43.20 | markgreene | Anyone here compiled zaptel on centOS 5.0? |
| 23:44.04 | riddlebox | hrmm I should probably rewrite my extensions.conf file, its looking pretty ugly |
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| 23:53.36 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 23:53.36 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 23:59.47 | thx2000 | Anyone else been experiencing a ton of problems w/ teliax recently? |