| 00:01.07 | voxter | Well, this is a first |
| 00:01.30 | voxter | Someone brute forced registering a sip device to the asterisk box i use for hosted sip clients |
| 00:01.44 | voxter | and started calling random 860 numbers blasting phishing scam recordings! |
| 00:02.54 | jtodd | voxter: Can you write up what you saw, the methods used, etc. and post to the asterisk-users list? Getting more horror stories out there is (in my opinion) a good thing so that people understand why they should choose strong passwords on their SIP clients. |
| 00:03.34 | voxter | jtodd: yeah, absolutely :) It was clearly a random brute force, because the sip secrets were dictionary words, and the guy connected is in finland some where. I already have the guys name and home phone number :P |
| 00:03.38 | jtodd | It's sometimes scary for people to see stories about voip security, but I think if you correctly decribe how things were attacked, that it will improve security overall. |
| 00:05.26 | jtodd | Make sure you say that it could have been prevented by choosing better passwords; posting something about security without including a "how to fix" is inviting Google-based FUD without context... |
| 00:05.57 | voxter | Yeah absolutely. It was all about sip secret strength here. |
| 00:06.42 | Uruguay | hello everyone I can not get caller ID in uruguay what is wrong I dont know I have TDMF signal |
| 00:06.46 | Uruguay | any idea |
| 00:07.51 | Uruguay | hello everyone I can not get caller ID in uruguay what is wrong I dont know I have DTMF |
| 00:09.22 | seanbright | Uruguay: didn't you find a bug report that described your problem? |
| 00:09.36 | jaytee | jeev? |
| 00:10.26 | drmessano | Anyone know much about the RTP300? |
| 00:10.48 | jaytee | never heard of it. Is that a new model from Chrysler? |
| 00:11.26 | drmessano | You're funny, jeevtee |
| 00:11.34 | jaytee | wasn't he the one that took over Kirk's body? oh, wait, no that's Sargon. |
| 00:11.52 | jaytee | drmessano, don't make me come over there and slap you :-) |
| 00:12.18 | drmessano | Got TFTP yourself |
| 00:12.34 | ManxPower | Don't trivialize the issue! |
| 00:12.37 | jaytee | ah, so it's a Linksys VoIP router |
| 00:12.43 | jaytee | hehee |
| 00:14.53 | drmessano | lol |
| 00:15.07 | jaytee | where'd jeev go? |
| 00:16.07 | drmessano | I found this trick with using sipsak to send a Sip NOTIFY to an ATA and setting a report rule on the device to have it spit it's parameters to the TFTP server |
| 00:16.27 | drmessano | I actually generated a full config for a SPA3102 like this yesterday |
| 00:16.39 | jaytee | sipsak? sounds like a kid's drink in a pouch |
| 00:16.50 | drmessano | I wanted to do it with a RTP300, but I think the internal ATA is behind the firewall |
| 00:16.55 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 00:17.08 | drmessano | and it doesn't seem to work from inside the firewall |
| 00:17.39 | irieKen | Does anyone recommend a specific receptionist/switchboard app?:) |
| 00:18.01 | jaytee | irieKen, mmmm, maybe FOP? |
| 00:18.30 | irieKen | jaytee: Have you used it much?:) |
| 00:18.34 | jaytee | nope |
| 00:18.55 | baliktad | I'd like to add a logrotate script to handle old CDR's. Do I need to run any command afterwards for Asterisk to handle this correctly (like "logger reload" for the other log files)? |
| 00:19.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-212a465e67aff6a2) |
| 00:19.25 | jaytee | irieKen, but I've heard other say they like it. http://www.voip-info.org/wiki-Asterisk+Flash+Operator+Panel |
| 00:19.49 | irieKen | jaytee: Hmm. It seems like a bit of a pain in the ass to set up though. |
| 00:19.56 | jaytee | it is |
| 00:20.31 | irieKen | :S |
| 00:21.48 | voxter | irieKen: i like isymphony. |
| 00:22.02 | irieKen | Voxter: Do you know what the price on that suite is? |
| 00:22.13 | voxter | irieKen: its up on the website at i9technologies.com |
| 00:23.29 | irieKen | Voxter: Ok:) What makes you like it more than the others? |
| 00:25.47 | voxter | irieKen: its easier to manage from a users perspective.. well, from an admin's perspective too. they have a demo available, download it and give it a try, see for yourself |
| 00:26.38 | irieKen | Voxter: Yeah, I'm looking at it now. |
| 00:28.21 | jaytee | pretty pricey app |
| 00:29.12 | voxter | its still the only one that i like using so far. |
| 00:29.32 | voxter | i have isymphony and fop deployed |
| 00:29.35 | voxter | fop works, but... meh. its fop. |
| 00:30.29 | voxter | is there a better way to read asterisk-users than by getting email? |
| 00:30.38 | voxter | I'd rather be able to read it in some sort of threaded rss viewer or something.. |
| 00:31.47 | codefreeze-lap | view by thread via http://lists.digium.com/pipermail/asterisk-users/ |
| 00:31.50 | irieKen | I'm surprised that Digium hasn't created their own switchboard app. |
| 00:32.25 | codefreeze-lap | Digium can be a surprising bunch of people |
| 00:33.05 | irieKen | hehe:) |
| 00:33.16 | voxter | codefreeze-lap: yeah, tis not collapsable :P im surprised nobody has made an rss feed for asterisk-users |
| 00:33.33 | voxter | or a good collapsable threaded rss viewer for mailing lists just like this. |
| 00:34.26 | codefreeze-lap | shhhh! one of the guys will read this, and go make it happen, if you're not careful! |
| 00:34.41 | arpu | hmm i found out i get this errors and send sms to smsc |
| 00:34.44 | arpu | rtp.c: Unknown RTP codec 100 received from '212.144.96.102' |
| 00:34.57 | arpu | anybody an idea ? |
| 00:36.18 | jaytee | I used to get that error when I was messing around with video and X-Lite phones |
| 00:37.47 | jaytee | but yours is related to sms not video |
| 00:38.14 | arpu | yes :-( |
| 00:38.30 | arpu | send fax with hylafax and asterisk -> sip works fine |
| 00:40.57 | jeev | does asterisk handle sms too? |
| 00:42.26 | riddlebox | jeev, http://www.google.com/search?q=asterisk+%2B+sms&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a |
| 00:44.24 | jeev | ok cool thanks |
| 00:48.24 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 00:50.27 | arpu | but i cannot getting working with a smsc in germany and send sms to handy |
| 00:51.54 | arpu | with sip ^^ |
| 00:52.25 | jtodd | what SMS provider do you use that supports SMS via SIP? |
| 00:58.48 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
| 01:00.33 | arpu | jtodd |
| 01:00.38 | arpu | i do not know |
| 01:00.39 | arpu | i test |
| 01:00.52 | arpu | http://groups.google.com/group/fa.hylafax/browse_thread/thread/2cfa839f1b61cd5f/c3f23ef5f1a9fe83?lnk=gst&q=0151#c3f23ef5f1a9fe83 |
| 01:01.12 | arpu | per alaw |
| 01:01.45 | jtodd | So this isn't SIP at all. |
| 01:01.47 | arpu | jtodd: you think this is not possible ? |
| 01:01.47 | jtodd | Right? |
| 01:02.05 | arpu | i use asterisk with sip and iaxmodem for hylafax |
| 01:02.18 | jtodd | I think it's possible, but it is an analog connection of some type that is doing the faxing. |
| 01:02.23 | *** join/#asterisk LakeSolon (n=blake@66-188-163-57.dhcp.stcd.mn.charter.com) |
| 01:02.39 | arpu | i can send fax over hylafax -> iaxmodem -> asterisk -> sip |
| 01:03.06 | arpu | this works greate |
| 01:03.12 | arpu | but not the sms thing |
| 01:04.06 | jtodd | OK, I don't have any direct experience with hylafax going to SIP, and also none with SMS via analog. |
| 01:04.39 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-109-62.nycap.res.rr.com) |
| 01:04.42 | jtodd | It may be possible, but I thought you were asking about SIP SIMPLE messaging, which is a different method. I don't think anyone is (yet) offering SIMPLE transmission of SMS, though anyone is welcome to correct me... |
| 01:05.08 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-109-62.nycap.res.rr.com) |
| 01:05.09 | jtodd | There are SMTP gateways for many SMS providers, and there are also SMS transmission methods from Asterisk to bluetooth phones... |
| 01:06.08 | arpu | jtodd: not in germany |
| 01:06.36 | jtodd | The bluetooth method works anywhere, but perhaps not the SMTP gateway. |
| 01:07.04 | jtodd | Another option is to buy some credits with clickatell, who have a fairly low-end product that has lots of interfaces and works pretty much anywhere. |
| 01:08.13 | jtodd | however, with the iaxmodem solution perhaps you might want to keep trying. If you succeed, write it up on voip-info.org and I'm sure others will be very interested in your results. |
| 01:12.56 | Qwell | ~lart jtodd |
| 01:12.56 | jbot | beats jtodd over the head with a microkernel |
| 01:13.06 | Qwell | jtodd: hey :D |
| 01:14.20 | arpu | hmm jtodd the strange is i get the same errors on faxing |
| 01:14.29 | arpu | but the fax comes to the fax |
| 01:14.54 | jtodd | clubs Qwell over the head with a trout. |
| 01:15.00 | Qwell | ~trout JT |
| 01:15.00 | jbot | ACTION slaps JT around a bit with a large trout! |
| 01:15.10 | jtodd | I do my own trout-clubbing instead of outsourcing it. |
| 01:15.13 | Qwell | jtodd: typo, but he probably deserved it. |
| 01:15.35 | hardwire | is it just me or is SS7 wonky? |
| 01:15.57 | jtodd | arpu: so you get errors, but faxing works but SMS doesn't with iaxmodem/hylafax? Sorry, no ideas there. I would possibly slow the modem speed down? |
| 01:17.18 | jeev | [Aug 11 18:17:11] ERROR[7202]: rtp.c:2530 ast_rtcp_write_rr: RTCP RR transmission error, rtcp halted: Operation not permitted |
| 01:17.19 | jeev | rawrawrawrawr |
| 01:19.21 | arpu | hmm yes i think it is a modem hylafax problem |
| 01:19.23 | arpu | No carrier detected |
| 01:21.33 | *** join/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 01:33.34 | chackal_sjc | does someone compiled iaxclient here???? |
| 01:34.19 | arpu | jtodd: |
| 01:34.22 | arpu | Aug 12 03:29:38.08: [ 3227]: DIAL 01712092522 |
| 01:34.22 | arpu | Aug 12 03:29:38.08: [ 3227]: <-- [16:ATDT01712092522\r] |
| 01:34.22 | arpu | Aug 12 03:30:20.46: [ 3227]: --> [10:NO CARRIER] |
| 01:34.22 | arpu | Aug 12 03:30:20.46: [ 3227]: PAGE FAILED: No carrier detected |
| 01:34.35 | arpu | any idea ? |
| 01:34.50 | jtodd | arpu: sorry, I'm not much help here. I haven't used Hylafax in the way you're using it, and it's been years since I've used it in the first place. |
| 01:35.21 | jtodd | is there any way to send the call out an analog port instead of over SIP to see if hylafax can do the right thing with an analog or ISDN call and then add the complexity of SIP after you've determined that it works? |
| 01:36.42 | arpu | jtodd this is the normal way |
| 01:36.58 | arpu | but i have no analog modem and i have no isdn/analog line |
| 01:37.52 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 01:38.03 | *** join/#asterisk Tobsn (n=luxus@freshmeat.de) |
| 01:38.06 | Tobsn | hey |
| 01:38.34 | *** join/#asterisk chendy (n=chatzill@219.134.30.227) |
| 01:40.51 | Tobsn | somebody know a good voip service that gives you a phone number too? |
| 01:41.27 | heedly | most let you purchase as many as you like. |
| 01:42.01 | chackal_sjc | does someone compiled iaxclient here???? |
| 01:42.21 | Tobsn | yeah i just tried voipbuster and thos retards gave me a number without even asking me what number etc. now i have a german number because i created the account a year ago in germany |
| 01:42.21 | Tobsn | ... |
| 01:43.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:49.19 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 01:49.44 | TJNII | ~itsplist-us |
| 01:49.45 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 01:49.50 | TJNII | Tobsn: ^^^^ |
| 01:50.10 | Tobsn | hmm |
| 01:50.21 | TJNII | Tobsn: The bot has some other itsp lists programmed into it, but I can never remember the phrases to get them. |
| 01:50.27 | Tobsn | i need one with a plan for free landline and cellphone calls including a area code number |
| 01:50.41 | TJNII | I use Broadvoice's unlimited in state plan |
| 01:50.52 | *** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 01:50.57 | TJNII | unlimied inbound calling and unlimited in state outbound calling |
| 01:51.04 | hoegaatit | can I get a link to the asterisk book again? |
| 01:51.06 | Tobsn | i would need unlimited to florida, illinois and michigan |
| 01:51.20 | Tobsn | TJNII can you chose your number? |
| 01:51.24 | TJNII | The have unlimited world for $10 more |
| 01:51.30 | TJNII | Tobsn: Yes. |
| 01:51.43 | *** join/#asterisk tyl3r_ (n=tyler@ip72-208-192-56.ph.ph.cox.net) |
| 01:51.58 | Tobsn | http://www.broadvoice.com/rateplans_unlimited_us.html |
| 01:52.02 | TJNII | It's limited, obviously, but you can choose the first 6 digits and select the last 4 from what's avaliable |
| 01:52.03 | Tobsn | see they never say if its cellphone too |
| 01:52.10 | tyl3r_ | Is it possible to have asterisk do call queuing but for a outside number? It would treat an outside number as an agent and always send the call to it? |
| 01:52.28 | TJNII | Tobsn: What do you mean by "cellphone" |
| 01:52.37 | TJNII | A number is a number, afaik... |
| 01:52.39 | Tobsn | calling a cellphone number |
| 01:52.51 | TJNII | I've never been billed differently.... |
| 01:52.55 | Tobsn | hmm |
| 01:52.59 | TJNII | I call my GF's cell all the time |
| 01:53.06 | Tobsn | ah okay so all is one |
| 01:55.12 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 01:55.48 | Tobsn | TJNII can you give me their sip gateway ip? |
| 01:55.51 | Tobsn | wanna ping it from here |
| 01:56.01 | TJNII | tyl3r_: Yes. I do it by having ab extension (1228) that calls the outside number. Then I have member => local/1228@public in queues.conf |
| 01:56.12 | TJNII | Tobsn: Check their website for a list. The have 8 or so |
| 01:56.32 | TJNII | Tobsn: I think it is in the asterisk instructions |
| 01:56.38 | *** join/#asterisk obnauticus_ (n=obnautic@about/windows/regular/obnauticus) |
| 01:57.09 | TJNII | Tobsn: I use 147.135.12.221 or 147.135.0.128 |
| 01:57.35 | Tobsn | 130ms |
| 01:57.36 | Tobsn | hmm |
| 01:57.51 | Tobsn | 110 on the other one |
| 01:58.11 | TJNII | Tobsn: You in Germany or the US? |
| 01:58.18 | Tobsn | germany at the moment |
| 01:58.22 | Tobsn | i need it here for 2-3 month |
| 01:58.27 | Tobsn | till i got back to usa |
| 01:58.37 | Tobsn | *go |
| 01:58.45 | tyl3r_ | I'll try that -- Thanks TJNII |
| 01:59.08 | TJNII | tyl3r_: I don't know if it is necessairy to go through a local extension, but it works. |
| 01:59.36 | tyl3r_ | member => local/1228@public |
| 01:59.39 | tyl3r_ | so it looks like that? |
| 01:59.41 | tyl3r_ | it tells me thats invalid. |
| 02:00.07 | TJNII | Well, you need an extension numbered 1228 in the public context for it to work |
| 02:00.24 | TJNII | That was copied out of my queues.conf from my setup. |
| 02:00.54 | TJNII | Tobsn: Yea, they offer calling to Germany but it looks like all their proxies are in the US. |
| 02:01.01 | TJNII | Tobsn: http://www.broadvoice.com/support_install_asterisk.html |
| 02:01.11 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 02:01.17 | TJNII | The list is under 3 /etc/hosts |
| 02:01.31 | Tobsn | nah i need free calls to usa ;) |
| 02:01.32 | tyl3r_ | strange, still says invalid |
| 02:01.32 | tyl3r_ | hmph |
| 02:01.49 | TJNII | Yea, this isn't free. Cheap, but not free. |
| 02:02.00 | Tobsn | uh |
| 02:02.00 | Tobsn | roxy.lax.broadvoice.com |
| 02:02.02 | Tobsn | has 30ms |
| 02:02.03 | *** join/#asterisk mallochig1 (n=Administ@63.135.231.241) |
| 02:02.24 | TJNII | ...really? That's better than the proxy I'm using! |
| 02:02.28 | TJNII | goes to re-ping |
| 02:02.31 | Tobsn | i just dont know what service i should use... i mean why not just voipbuster |
| 02:02.34 | Tobsn | ;) |
| 02:03.19 | TJNII | shrugs |
| 02:03.31 | TJNII | I was just making a suggestion based off of my esperiences. |
| 02:04.40 | mallochig1 | I can't get any incommong calls. My connection registers.. And I can receive incomming.. http://collegecomputernerd.com/error for my configs and debug output. |
| 02:05.37 | mallochig1 | grumbles about spelling and grammar. |
| 02:08.26 | mallochig1 | Its driving me nuts. Here my cry! |
| 02:08.39 | *** join/#asterisk Keizer (n=Keizer@208.50.100.60) |
| 02:09.38 | TJNII | So you can recieve incoming but not get incoming? |
| 02:12.37 | mallochig1 | seems so |
| 02:13.10 | mallochig1 | I see it dialing from my cell phone to my did (asterisk) but it looks like it drops. |
| 02:13.23 | mallochig1 | INVITE sip:#DID1#@63.135.231.241:6764 SIP/2.0 |
| 02:13.36 | TJNII | pastebin what the cli says with verbose >= 5 |
| 02:13.53 | mallochig1 | http://collegecomputernerd.com/error |
| 02:14.02 | mallochig1 | Or do you need another dump? |
| 02:14.30 | mallochig1 | by verbose >= 5 running asterisk -vvvvvvvvvvgc does the job correct? |
| 02:15.02 | TJNII | I want to see the asterisk messages, not the sip debug messages. |
| 02:16.17 | mallochig1 | http://collegecomputernerd.com/full |
| 02:16.28 | mallochig1 | its a sym link so your getting everything.. |
| 02:16.46 | tyl3r_ | hey TJNII -- Do you know a lot about asterisk's queuing? |
| 02:16.53 | tyl3r_ | I got it to work but have a few questions |
| 02:17.08 | mallochig1 | <- is also a tyler :) |
| 02:17.30 | tyl3r_ | :) |
| 02:17.35 | TJNII | tyl3r_: I'm not an expert. I got my queues working and called it good. |
| 02:17.45 | *** join/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
| 02:18.10 | tyl3r_ | well right now when i call it plays music, then tells the person they are next in line and rings the phone... I'd like it to keep playing music until the support person picks up |
| 02:18.14 | tyl3r_ | and if they dont pick up |
| 02:18.16 | tyl3r_ | keep playing music |
| 02:18.22 | tyl3r_ | if it starts to ring and the ext number doesnt pick up |
| 02:18.24 | tyl3r_ | they hang up the call |
| 02:18.42 | TJNII | mallochig1: Okay, I'm only interested in what * prints when you try and dial in. |
| 02:18.58 | mallochig1 | doesn't print out anything unless I have sip set debug on |
| 02:19.15 | *** join/#asterisk ariel_ (n=ariel_@adsl-074-186-013-209.sip.mia.bellsouth.net) |
| 02:19.16 | mallochig1 | also the did does not shouw up in the cdr log |
| 02:19.21 | trev | I need advice on selecting a VOIP provider. I want a 1-800 number, SIP inbound/outbound, what company is good (and cheap) ? |
| 02:20.10 | mallochig1 | refresh it and you will see that now.. |
| 02:22.19 | TJNII | tyl3r_: Does it behave properly without the external entry? Because I think that is how it is supposed to work. |
| 02:25.50 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 02:28.28 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 02:29.10 | mallochig1 | this is wierd |
| 02:29.21 | mallochig1 | I think it may be dialing my cell phone? |
| 02:30.35 | [TK]D-Fender | mallochig1: you shouldn't be phrasing statements as questions to yourself. |
| 02:31.34 | mallochig1 | Is it dialing my cell phone number? |
| 02:31.40 | mallochig1 | I have no idea.. |
| 02:31.48 | mallochig1 | http://collegecomputernerd.com/full |
| 02:31.51 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 02:31.52 | jeev | Fender |
| 02:32.02 | jeev | everything is ok, i guess.. |
| 02:32.12 | jeev | only crap part is bad sound quality, even on local call.. so i'll figure that out soon |
| 02:32.15 | mallochig1 | Its as if its calling the * box seeing the from field, looking it up, then forwarding it to my cell phone. |
| 02:33.15 | TJNII | You should probably try and debug those 404 and 401 errors |
| 02:34.14 | tyl3r_ | hmph |
| 02:34.29 | mallochig1 | yeah.. |
| 02:34.40 | mallochig1 | im not even sure what it was looking for though.. |
| 02:40.59 | mallochig1 | It works now |
| 02:40.59 | mallochig1 | thx |
| 02:42.14 | *** join/#asterisk stencil (n=stencil@76-10-136-205.dsl.teksavvy.com) |
| 02:42.58 | jeev | Fender, as far as dtmf, everything is ok now, infact.. everything is pretty much finished here. i do have an issue that's not required but the IVR is sounding really funky. it's choppy. |
| 02:44.49 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 02:50.55 | *** join/#asterisk Segnale007 (n=Segnale0@host37-252-dynamic.33-79-r.retail.telecomitalia.it) |
| 03:10.01 | mosty | jeev, upgrade your bandwidth |
| 03:13.22 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:17.50 | jeev | mosty, why assume it's a bandwidth issue ? |
| 03:18.43 | mosty | "choppiness" generally occurs because you're dropping network packets or some of those packets arrive too late |
| 03:19.04 | TJNII | Turn off the torrents and try again |
| 03:19.06 | mosty | or try a lower bandwidth codec |
| 03:21.12 | jeev | so did you ever think that it's on a LAN that's giving me choppiness? |
| 03:21.15 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 03:22.27 | mosty | upgrade the bandwidth of whatever network the audio is going over, LAN or otherwise |
| 03:22.39 | jeev | so 100mbit with 0 utilization is too little.. |
| 03:23.01 | trev | are these rates reasonable? http://pstn.junctionnetworks.com/pricing-signup |
| 03:23.17 | mosty | it's strange that you're getting dropped packets then. what latency do you have on the link? |
| 03:23.29 | [TK]D-Fender | trev: no |
| 03:23.31 | jeev | trev, what are your ideal goals ? |
| 03:23.46 | jeev | trev, flowroute.com seems cheap as hell. i am unsure of service from them. |
| 03:23.53 | heedly | trev: not bad for premium routes. There is cheaper though. |
| 03:23.58 | jeev | mosty, no latency what so ever. |
| 03:24.22 | mosty | jeev, how many ms? |
| 03:24.28 | trev | hm, I'd like something that works. it's important to have inbound/outbound SIP--I'm slightly new to this and it's unclear if most hosts offer this. It looks like a lot of them just do SIP to your PBX and end it there |
| 03:24.43 | jeev | 0ms.. its 10 feet away man, no congestion, no nothing. |
| 03:25.07 | [TK]D-Fender | trev: How many minutes/month are you looking to use? |
| 03:25.18 | mosty | jeev, what codec are you using? |
| 03:26.17 | JT | jeev: this using just sip? |
| 03:26.20 | trev | Fender: at this stage for prototyping, not many. like < 100min. Eventually I want something that can give good rates for the needs of a mid-size call center. |
| 03:26.37 | *** part/#asterisk stencil (n=stencil@76-10-136-205.dsl.teksavvy.com) |
| 03:26.37 | jeev | ulaw, yes JT |
| 03:26.57 | mosty | jeev, what format are the sounds stored as? |
| 03:27.20 | mchou | trev: where are you based? |
| 03:27.40 | trev | mchou: san francisco |
| 03:28.00 | [TK]D-Fender | trev: Shop through this list first |
| 03:28.02 | jeev | mosty: gsm |
| 03:28.02 | mchou | trev: for prototyping look at diamondcard.us |
| 03:28.07 | [TK]D-Fender | ~itsplist-us |
| 03:28.08 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 03:28.14 | mchou | trev: right in your back yard |
| 03:28.55 | mchou | trev: when you are ready to go production go ahead and transfer to ITSP eith "unlimited" monthly |
| 03:29.07 | mchou | trev: as your call volume increase |
| 03:29.27 | mchou | with* |
| 03:29.48 | trev | mchou: ok, I'll check out the rates now. |
| 03:30.18 | *** join/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
| 03:30.31 | mchou | trev: diamondcard is strictly pay as you go, decently quality at slightly below avg. prices |
| 03:30.35 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 03:31.20 | johnantypas | Good evening. Anyone here care for the strange Asterisk 1.4.21 question of the day? |
| 03:31.34 | jeev | johnantypas, can't get any stranger than my questions.. go ahead |
| 03:31.49 | johnantypas | I hope I'm an easy one :-) |
| 03:32.00 | trev | mchou: I'm not clear which service on diamondcard I want |
| 03:32.02 | jeev | you'll be easier than me, i'm the record holder. |
| 03:32.11 | trev | what's the lingo for SIP inbound/outbound ? |
| 03:32.18 | mchou | trev: what do you intend to do? |
| 03:32.24 | johnantypas | I've got an Asterisk box (1.4.21.2) running on a Fedora 8 box. The box is also the NAT router. The Asterisk box has a direct connection to the Internet. |
| 03:32.44 | mchou | trev: "DID" means you want an "incoming #" |
| 03:33.06 | johnantypas | I get 5-10sec audio dropouts. HOWEVER, if I provision a SIP phone to the SIP termination provider directly, no audio dropouts. ONLY when it goes through the Asterisk box. |
| 03:33.13 | mchou | trev: "termination" generally means outbout to PSTN |
| 03:33.23 | mchou | outbound* |
| 03:33.25 | johnantypas | No hardware messages, no logs from Asterisk. Everything seems fine, save for the dropouts. |
| 03:33.38 | jeev | johnantypas, you know.. we're almost doing the same thing hehe |
| 03:33.51 | [TK]D-Fender | johnantypas: make sure you have "canreinvite=no" under [general] and each peer in sip.conf Re-test |
| 03:33.55 | mchou | trev: so it sounds like you want DID and termination |
| 03:34.06 | trev | mchou: so I have my DID, call goes through to my asterisk sip-hub, which places calls through SIP to remote operators |
| 03:34.08 | johnantypas | Checking now.... |
| 03:34.09 | trev | is that termination? |
| 03:34.16 | jeev | Fender, is canreinvite required for each peer also ? |
| 03:34.38 | mchou | trev: so you have DID already? |
| 03:35.01 | trev | mchou: not yet, I'm hoping to get all of this in one package |
| 03:35.19 | trev | well, I'm supplying the asterisk box of course |
| 03:35.26 | mchou | trev: diamondcard has it all, afaik |
| 03:35.29 | trev | ok |
| 03:35.51 | TJNII | jeev: [TK]D-Fender can correct me if I'm wrong, but once a setting is in general putting it in the peer won't do anything unless the setting is different. |
| 03:35.55 | mchou | trev: this asterisk sip hub, you intend to run it, right? |
| 03:35.59 | trev | right |
| 03:36.18 | johnantypas | OK, All of my SIP phones have it (canreinvite=no), My SIP termination providers did not. |
| 03:36.28 | mchou | trev: then look into DID and termination |
| 03:36.37 | trev | however if I could find a provider that would do all of this on their end, cheaply, I would go for it |
| 03:36.52 | trev | but I think that gets into the hosted-call-center space which looks pricey |
| 03:36.59 | johnantypas | What else should I look for/at? My providers are claiming they see this with Askterisk occasionally, but without diagnostics.... |
| 03:37.00 | TJNII | johnantypas: Byt the providers have it now, correct? |
| 03:37.04 | jeev | trev, flowroute, fast.. cheap, just a $35 deposit in the beginning, once again.. i dont know their quality but it'll definitely be good for tests. |
| 03:37.04 | mchou | trev: how many different sip operators? |
| 03:37.07 | jeev | flowroute.com |
| 03:37.25 | johnantypas | Yes, providers now have canreinvite=no set. |
| 03:38.10 | mchou | trev: cause diamondcard can transfer calls to sip or iax (as well as PSTN) |
| 03:38.21 | jeev | TJNII, you ever see the default asterisk sounds come out with crappy quality even on LAN with no congestion and latency? |
| 03:38.27 | trev | yeah that's all I'm looking to do is cheap call-transfer |
| 03:38.32 | trev | is there a special word for that? |
| 03:38.36 | TJNII | jeev: Nope, can't say that I have |
| 03:38.44 | jeev | hm. |
| 03:38.48 | trev | mchou: cheap call transfer and the ability to modify the call-queue whenever I want |
| 03:39.02 | TJNII | jeev: I would check for network overload followed by processor overload. |
| 03:39.23 | mchou | trev: call transfer to sip/iax at diamondcard is 1/3 of 1 cent/min. :) |
| 03:39.39 | mchou | trev: you cant beat that |
| 03:39.40 | jeev | there is absolutely no issue when it comes to resources, i'm betting it's a misconfiguration issue again.. but i know it shouldn't be a firewall issue since it's pretty much on the LAN, nothing leaving it. |
| 03:39.43 | trev | mchou: do you own diamondcard? is there an #asterisk discount? =) |
| 03:40.06 | mchou | trev: nope, merely a reasonably satified customer |
| 03:40.08 | jeev | mchou, if you own it.. make the site a little more user friendly and less busy :D |
| 03:40.42 | trev | mchou: so what do you call what I just described? sip-based call transfer? |
| 03:40.47 | mchou | jeev: I dont own it. The web site looks like it's made by some kid in kindergarten, I grant you that :) |
| 03:40.54 | trev | the site is so poorly designed |
| 03:40.56 | jeev | yea seriously |
| 03:41.20 | mchou | but the iax service is legit and good |
| 03:42.06 | mchou | the only reason I know about it is thru ekiga (the softphone) |
| 03:42.23 | mchou | they are somehow affliated |
| 03:42.26 | trev | the ability to modify the call queue whenever I want is critical... |
| 03:42.31 | trev | maybe they can rig something up |
| 03:42.39 | mchou | some sort of partnership. |
| 03:44.00 | johnantypas | Jeev: WIth that change, as I test, anything else I can use as a diagnostic when dropouts occur given nothing appears in logs (-rvvvvvvvv set). |
| 03:44.21 | johnantypas | (I hate just saying its broken without providing some sort of diagnostics data) |
| 03:44.43 | jeev | so asterisk and nat/iptables on the same box ? |
| 03:44.56 | mchou | there are areas where ekiga.net beats diamondcard though. (like the ability to simultaneously register more than one phone with the same credentials) |
| 03:45.13 | johnantypas | Jeev: Asterisk 1.4.21.2, Shorewall. Asterisk has direct access. |
| 03:45.22 | johnantypas | Cable (12-16Mb down/1-2Mb up) |
| 03:45.33 | jeev | i am not sure what shorewall is.. so, i'm assuming it's some type of firewall hehe |
| 03:45.44 | johnantypas | Just a wrapper for IPtables. |
| 03:46.01 | johnantypas | (Oh, and using static IPs.... :-) ) |
| 03:46.01 | jeev | i'd still fill in the proper rules for 5060 and RTP stuff.. |
| 03:46.35 | TJNII | johnantypas: I had issues with shorewall and sip. I had to make it stop loading the sip module for * to work (IIRC) |
| 03:46.58 | johnantypas | I've got 5060 open. What is the port for RTP. |
| 03:47.14 | johnantypas | Not sure SIP module is even loded. |
| 03:47.18 | jeev | usually 10000-20000 depending on rtp.conf. |
| 03:48.25 | johnantypas | I stand corrected.... Conntrac SIP is loaded...... |
| 03:48.35 | johnantypas | (See what happens when you ask people who know what they're doing...) |
| 03:50.59 | jeev | mchou, aparently, i disabled all codec, enabled gsm, called my test *81, got clear ass sound. |
| 03:51.57 | *** join/#asterisk hi365_m (n=hi365@213.151.63.155) |
| 03:52.07 | JT | get rid of sip contract |
| 03:52.13 | JT | contrack |
| 03:53.40 | jeev | eh, looks like he got rid of his nic module :D |
| 03:54.00 | ReDNeQ | evening |
| 03:55.38 | mchou | jeev: what are you referring to? |
| 03:56.21 | mchou | jeev: meaning which itsp? |
| 03:56.55 | jeev | the network congestion you spoke of... was the wrong audio files in my share folder :D |
| 03:58.05 | mchou | mchou: huh? I spoke nothing re network congestion. you sure it was me? |
| 03:58.28 | jeev | crap, sorry! |
| 03:58.30 | jeev | it was mosty haha |
| 03:59.24 | mosty | perhaps i misunderstood what you meant by "choppy" |
| 04:00.38 | jeev | ahh it's ok |
| 04:00.40 | jeev | i'm happy now. |
| 04:06.45 | *** join/#asterisk astassistant (n=jtknapp@h-72-244-204-146.sttnwaho.dynamic.covad.net) |
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| 04:11.13 | *** join/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
| 04:11.37 | johnantypas | Thank you all. All changes made. I'll continue to test and see if these dropouts go away. |
| 04:12.09 | *** part/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
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| 04:16.27 | *** mode/#asterisk [+o denon] by ChanServ |
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| 04:17.25 | *** join/#asterisk elvisthedj (n=kris@72.24.236.195) |
| 04:18.59 | elvisthedj | Just upgraded to 1.6 beta and can't get sip inbound to work (not using rt).. I use broadvoice and can see the call on the cli, but i get my bv voicemail.. as if the call has no entry point in the dialplan |
| 04:19.43 | elvisthedj | anybody have any ideas.. google is not being my friend with this one.. outbound calling is fine |
| 04:20.31 | elvisthedj | cli shows ==Using SIP RTP TOS bits 184 \n ==Using SIP RTP CoS mark 5 |
| 04:21.12 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:22.48 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:28.48 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:28.48 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 04:31.51 | jeev | hi leif. |
| 04:36.38 | *** join/#asterisk jameswf-home (n=james@jfinstrom.broker.freenet6.net) |
| 04:37.11 | TJNII | elvisthedj: Guy by the name of mallochig1 had a similar problem a few hours ago. He solved it but didn't say how. Maybe try poking him. |
| 04:38.57 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 04:39.27 | ZX81 | hi all, has anyone else seen a situation where a polycom phone is on a call, but then when another call comes in it disconnects the first? |
| 04:40.28 | ZX81 | only solution is to use call-limit=1 but then they cant use the xfer button |
| 04:40.31 | ZX81 | latest firmware |
| 04:40.55 | elvisthedj | TJNII: I just figured out my problem i think :) |
| 04:44.15 | elvisthedj | TJNII: forgot about the whole lumenvox asterisk bridge (1.4 only).. so.. down i go |
| 04:50.05 | oilinki | is the webmin same as the asterisk-gui? |
| 04:51.52 | oilinki | or where should I find the asterisk-gui? |
| 04:52.13 | *** join/#asterisk infinity1 (i=brendon@saleen.netcal.com) |
| 04:52.26 | jeev | asteriskgui si called asteriskNOW |
| 04:52.30 | infinity1 | ~centos52bug |
| 04:52.31 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 04:55.09 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:55.09 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:57.12 | Tobsn | damn |
| 04:57.14 | Tobsn | packet8.net sucks. |
| 04:58.29 | Tobsn | i jus signed up - and i didnt knew about the whole send equiment and active over the gay free phone they give you story - so now i payed almost 90 bucks to get the gay phone i dont even need to my home address in chicago with what i have to activate my account while im sitting in munich for the next 3 month with my linksys WIP330... |
| 04:58.30 | Tobsn | gay. |
| 04:59.19 | Tobsn | sometimes i just hate my life. ;) |
| 04:59.21 | jeev | uh heh |
| 04:59.24 | jeev | Tobsn, wip 330's are gay |
| 04:59.28 | Tobsn | :P |
| 04:59.30 | Tobsn | youre gay too! |
| 04:59.32 | jeev | i was about to sue linksys.. |
| 04:59.39 | jeev | they sent me a brand new one.. then i sold it |
| 04:59.44 | mosty | tobsn: forget the packet8, go get a beer |
| 04:59.49 | jeev | lol |
| 05:00.47 | Tobsn | i dont even know if its gay because my neighbour who has the internet account and has a new wireless router has all my wireless stuff down there and his wifi signal is here around 40% strngh but thats not enought for the WIP330 - enough for my laptop but not for the phone. so i could use my wifi expander - but its in his apartment and hes gone for 2 month |
| 05:00.51 | Tobsn | everything works out perfect... |
| 05:01.10 | Tobsn | i think by now i invested 300 bucks into wifi VOIP and i got nothing |
| 05:01.11 | Tobsn | ;) |
| 05:01.17 | drmessano | Wifi phones suck in general |
| 05:01.23 | Tobsn | a phone that cant connect to the wifi and a voip account that i cant activate |
| 05:01.43 | Tobsn | dude if i could tell you it sucks i would be 100 times more happy ;) |
| 05:03.42 | jeev | as fender told me months back |
| 05:03.46 | jeev | 'if you get a wip330, you're a dilweed' |
| 05:03.52 | Tobsn | http://www.youtube.com/watch?v=efFgzOA6qfE |
| 05:03.52 | jeev | i got one and regreted it. |
| 05:04.08 | jeev | dood, i dont get it, do you like it or not, you're confusing me. |
| 05:04.11 | *** part/#asterisk ZX81 (n=matt@202.20.97.211) |
| 05:04.21 | Tobsn | mosty yep. |
| 05:04.22 | drmessano | Half duplex battery sucking voip sounds fun |
| 05:04.38 | Tobsn | jeev i cant tell if i like it |
| 05:04.42 | Tobsn | because i was never able to test it |
| 05:04.52 | Tobsn | (Tobsn): i think by now i invested 300 bucks into wifi VOIP and i got nothing |
| 05:04.56 | Tobsn | (Tobsn): a phone that cant connect to the wifi and a voip account that i cant activate |
| 05:05.29 | mosty | Tobsn, try fransiskaner dunkel |
| 05:05.36 | Tobsn | only with food |
| 05:05.37 | Tobsn | ;) |
| 05:05.40 | jeev | tobsn, the one i got was horrific.. sucked walls, then i got them to send me a new one, worked well.. but decided it was POS technology. |
| 05:06.00 | jeev | tobsn, if you spent the 300 bucks on a fat hooker with no teeth, it'd be more satisfying. |
| 05:06.14 | Tobsn | i would go and get a reuthberger but i only get that in my area... not in that crappy lower bavarian zone i ended up now |
| 05:06.17 | drmessano | 46/49 MHZ cordless and an ATA 4TW |
| 05:06.51 | drmessano | You put a 4.5 foot whip on the handset, bam Military COmms |
| 05:06.57 | Tobsn | yeah its like... i mean you can spend as much money as you want - you still get bullshit |
| 05:07.05 | drmessano | Paint it green, make your friends call you "Blue leader" |
| 05:07.14 | Tobsn | like i said - 300 bucks down the drain and i still dont have what i wanted - not even close |
| 05:07.45 | drmessano | You didn't get bullshit, you got a shitty phone you can't activate. That's much worse |
| 05:08.01 | Tobsn | yeah its just 8000 miles away |
| 05:08.01 | Tobsn | haha |
| 05:08.13 | jeev | what the hell is a reuthberger |
| 05:08.17 | Tobsn | i mailed the support and set up a support ticket on packet8 - we'll see how it turns out |
| 05:08.24 | Tobsn | jeev the best beer ever made |
| 05:08.28 | jeev | i dont drink |
| 05:08.30 | drmessano | If I were you, I wouldnt even be telling this story.. I would be writing out my suicide note and checking rafters for tensile strength |
| 05:08.55 | mchou | Tobsn: half duplex battery? What does that mean? |
| 05:08.58 | Tobsn | http://www.beerlabels.com/labels/labels.pl/3668/reutberger-export-hell-beer.html |
| 05:09.03 | Tobsn | what? no idea |
| 05:10.40 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-84-177.w86-199.abo.wanadoo.fr) |
| 05:12.17 | mchou | drmessano: what's a half duplex battery? |
| 05:14.00 | carrar | I prefer autosensing batterys |
| 05:15.47 | drmessano | There is no such thing as a half duplex battery |
| 05:15.55 | drmessano | That doesn't even make sense |
| 05:16.17 | mchou | [22:04] <drmessano> Half duplex battery sucking voip sounds fun |
| 05:16.41 | mchou | trying to decipher that :) |
| 05:16.45 | drmessano | <drmessano> Half duplex, battery sucking voip sounds fun <-- DUH |
| 05:16.53 | drmessano | <drmessano> Half duplex battery, sucking voip sounds fun <-- WTF |
| 05:16.59 | drmessano | Not that hard |
| 05:17.15 | mchou | ok |
| 05:27.25 | Tobsn | is there any free win software that shows my signal strength? |
| 05:32.30 | jeev | check freshmeat :P |
| 05:34.04 | Tobsn | for windows :P |
| 05:54.53 | *** join/#asterisk jameswf (n=james@jfinstrom.broker.freenet6.net) |
| 05:55.29 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 06:06.34 | baliktad | can someone tell me what's wrong with this line |
| 06:06.36 | baliktad | exten => _X.,n,Set(CALLERID(number)=${IF([${CALLERID(number):0:1}=+]?${CALLERID(number):1}:${CALLERID(number)})}) ;strip off leading + if present |
| 06:07.09 | baliktad | it's always stripping the first character |
| 06:07.55 | mosty | why not just use two lines, match for the + on the first line |
| 06:08.20 | baliktad | how do you mean |
| 06:08.56 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 06:09.07 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 06:10.15 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 06:11.38 | kaldemar | baliktad: looks like you're missing a $ in there. |
| 06:12.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:12.53 | baliktad | where |
| 06:12.55 | kaldemar | baliktad: IF([ => IF($[ |
| 06:13.26 | baliktad | arg |
| 06:13.29 | baliktad | dammit |
| 06:14.40 | baliktad | of course that was it |
| 06:15.03 | kaldemar | it happens. |
| 06:15.16 | baliktad | good eyes though, thanks for spotting that |
| 06:30.31 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 06:31.53 | *** join/#asterisk ShaunWing (n=chatzill@dsl-244-214-169.telkomadsl.co.za) |
| 06:33.27 | ShaunWing | Say, any idea how to setup Asterisk so that the IAX2/SIP calls are sent down two seperate network interfaces (connected to ADSL) and thereby allowing agregating the bandwidth |
| 06:33.59 | *** join/#asterisk the_5th_wheel (n=edd@196.212.110.130) |
| 06:34.25 | creativx | add static routes |
| 06:34.40 | creativx | if you have different ips for sip/iax2 itsp |
| 06:42.15 | ShaunWing | Problem is I have only one itsp |
| 06:42.54 | creativx | set up tunnels |
| 06:42.55 | creativx | i dunno |
| 06:42.56 | creativx | :) |
| 06:43.48 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 06:45.10 | jeev | how does the g729 codec work/ and other ones.. like licensing. |
| 06:45.37 | jeev | is it recommended? |
| 06:46.12 | drmessano | $10 per channel |
| 06:46.33 | mosty | g729 is recommended if you want a low bandwidth codec |
| 06:47.20 | jeev | is the license a one time fee per channel ? |
| 06:47.34 | drmessano | yeah |
| 06:48.28 | [netman] | the licence is bound to your network card MAC address |
| 06:48.35 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 06:49.41 | jeev | what if it's spoofed? |
| 06:51.41 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 06:55.53 | mosty | where does the switchvox switchboard get is presence information from? the screenshots have stuff like "do not disturb until 4pm" |
| 06:57.27 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 06:57.39 | drmessano | ----> #switchvox |
| 06:58.12 | jeev | dunno what that is |
| 07:00.53 | jeev | [netman], any idear? |
| 07:01.22 | drmessano | You're asking about violating the licensing terms |
| 07:02.44 | andrewy | he's asking if he has to repurchase everything should his NIC fail. having to do so significantly drives up the cost of licensing |
| 07:04.08 | jeev | well no, i read the agreement thing |
| 07:04.13 | jeev | i dont have a million channels.. |
| 07:04.25 | jeev | just curious, can't someone spoof the mac addy and put the license elsewhere? |
| 07:05.08 | andrewy | probably, but I'm sure there are other ways to get around the licensing if you want to do it that way |
| 07:06.32 | jeev | so how would it work, would you set up ulaw on the phone to asterisk box, then only allow g729 to ITSP? |
| 07:15.14 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 07:15.36 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 07:16.19 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 07:24.01 | gnorbert | Hi all. I have two softphones connected from two computers to an *. Both computers are connected, but I can speak only with one of them, the other can't be heard. My sip.conf looks like this: http://pastebin.com/d4d1501a2 Does somebody have an idea, what can be the problem? (If I try to connect from the wrong computer with an other softphone, on both computers can be heard only a little rare and short beeping) |
| 07:24.31 | gnorbert | And the other softphone writes, that it's registered. |
| 07:24.48 | *** join/#asterisk chendy (n=chatzill@58.61.40.100) |
| 07:28.42 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 07:28.59 | gnorbert | The computers are in one meetme conference, if that means something... |
| 07:29.14 | marc7 | has asterisk done away with the notion that there's a difference between IAX peers/users, or is one used for outbound (peer) and the other for inbound (users)? |
| 07:29.25 | marc7 | if we were expecting two way traffic between two servers, do each need a user/peer pair? |
| 07:32.13 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:32.26 | *** join/#asterisk nickdom (n=nickdom@83.244.78.241) |
| 07:32.35 | nickdom | hello |
| 07:32.37 | jeev | gnorbert, i think you chose a bad time, nobody is here |
| 07:32.39 | jeev | sleepy time |
| 07:33.04 | jeev | gnorbert, enable sip debug and see if you're having nat transmission issues |
| 07:33.05 | jeev | night night |
| 07:33.11 | nickdom | when i send calls from asterisk to an external gateway, all calls are returned "Answered" even when the call is not really answered |
| 07:33.51 | nickdom | when terminating calls in asterisk, in the cdr logs i get "No answer" when the call is not made |
| 07:33.58 | gnorbert | jeev: It's 9:33am here. :)) |
| 07:34.46 | gnorbert | But anyway, you are right, nobody is here right now. :) |
| 07:35.31 | Tobsn | http://www.youtube.com/watch?v=xgfOh_Xz7yw |
| 07:35.32 | nickdom | i tried to post my problem last night and no one relpied so i thight it was a bad time |
| 07:35.51 | nickdom | it's 10:35am here :S |
| 07:37.43 | gnorbert | nickdom: Try afternoon. :) Then usually more people are on, just I tried, maybe somebody can help already now. Who knows? :) |
| 07:39.08 | *** join/#asterisk JoZu (i=asdf@84.120.221.158.dyn.user.ono.com) |
| 07:39.45 | nickdom | gnorbert: thank u. i hope so |
| 07:44.25 | obnauticus | heh. Does anyone here know of any WiFi ATA's? |
| 07:46.49 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 07:47.53 | mosty | billion make a few |
| 07:50.10 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:52.34 | *** join/#asterisk jack_sparo (n=eddy@81-208-91-225.ip.fastwebnet.it) |
| 07:53.17 | obnauticus | captian jack_sparo ! |
| 07:55.38 | baliktad | any way to restrict certain sip clients to registration only from a defined netblock? |
| 07:57.00 | *** join/#asterisk h3x0r (n=hex@wsip-24-120-86-83.lv.lv.cox.net) |
| 07:57.16 | *** join/#asterisk jplank (n=GBove@ool-18bb0206.dyn.optonline.net) |
| 07:59.42 | obnauticus | baliktad: you can define host= :\ |
| 08:00.39 | jplank | I dont understand the use of hint in the dialplan |
| 08:00.59 | jplank | if I wanted to use it, I would have to use it as a speed dial? |
| 08:01.26 | jplank | hmmm |
| 08:02.06 | baliktad | I didn't think host= would restrict registration |
| 08:02.56 | mosty | jplank, hints are for presence monitoring |
| 08:04.01 | jplank | yea |
| 08:04.09 | jplank | that I understand |
| 08:04.29 | jplank | but the example on voipinfo uses something like exten => 200,hint,SIP/20 |
| 08:06.05 | *** join/#asterisk skyNomad (n=ralfe@196.212.110.130) |
| 08:06.30 | skyNomad | Does anyone know if there is a channel for PHP-AGI? |
| 08:06.48 | yang | skyNomad: you are trying to install billing? |
| 08:06.58 | mosty | jplank, yes that has nothing to do with dialing |
| 08:07.13 | skyNomad | yang: no |
| 08:07.23 | skyNomad | yang: I'm trying to get AGI help for PHP. |
| 08:08.03 | nickdom | i have a system that consists of: asterisk, gateway, radius and abilling system |
| 08:08.03 | skyNomad | yang: Basically, I'm writing some agi scripts in php, and have hit a snag and I'm trying to find a solutino. |
| 08:08.16 | nickdom | calls sent to asterisk are sent to the gateway |
| 08:08.18 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:08.33 | nickdom | the first authenticate through radius |
| 08:08.56 | nickdom | when the calls r not made, the billing system still charges the call |
| 08:09.19 | skyNomad | nickdom: What billing system are you using? |
| 08:09.20 | nickdom | it's because asterisk cdr logs return "Answered" for all calls |
| 08:09.43 | nickdom | a billing system built under tomcat |
| 08:10.30 | nickdom | skyNomad: when terminating calls in asterisk, calls that r not made return "No answer" |
| 08:11.02 | nickdom | when sent to another gateway, they all return "Answer" and that's why the billing system always charge |
| 08:11.03 | skyNomad | nickdom: Do you think there might be a bug in your billing software? |
| 08:11.21 | nickdom | i checked that, no bug in the billing |
| 08:11.30 | skyNomad | nickdom: Perhaps the second gateway is not returning the correct channel status code? |
| 08:11.34 | nickdom | asterisk always show calls as "Answered" |
| 08:11.58 | nickdom | actually i am just sending the call |
| 08:12.02 | skyNomad | nickdom: It is not about calls being answered, it is about the cannel status. |
| 08:12.11 | skyNomad | *channel status |
| 08:12.18 | nickdom | so i have to let the gateway send feedback? |
| 08:12.45 | skyNomad | nickdom: The billing software has to monitor the channel status to see if the call was connected to the other end or not. |
| 08:12.59 | nickdom | i though i should add sth in the "extensions.conf" |
| 08:13.09 | skyNomad | nickdom: You need to get that channel status from whatever additional gateways you are using to break out through. |
| 08:13.44 | nickdom | the billing system talks to asterisk through radius |
| 08:13.56 | nickdom | it's doesn't communicate with the gateway |
| 08:14.30 | skyNomad | nickdom: Hmm... I've never used asterisk with radius before. |
| 08:14.43 | nickdom | skynomad: what u say makes sense |
| 08:15.02 | nickdom | but anything easier, like modifying extensions.conf? |
| 08:15.03 | *** join/#asterisk nix8n82 (n=nate@63.162.28.92) |
| 08:15.17 | skyNomad | nickdom: But wouldn't the gateways send back some sort of response message to the radius server about the channel status, or the call status? |
| 08:15.49 | nickdom | i am only sending the calls, no feedback from the gateway |
| 08:16.02 | skyNomad | nickdom: I think that the issue is in one of two places. It is either with the billing software, or with the gateway not sending correct information to the radius system. |
| 08:16.35 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-45d89e93b8d4952b) |
| 08:16.44 | nickdom | since the billing doesn't deal with the gateway, the gateway should send back feedback |
| 08:16.59 | nickdom | to asterisk |
| 08:17.13 | skyNomad | nickdom: correct. |
| 08:18.11 | nickdom | i will try to see what can i do in extensions.conf first |
| 08:18.28 | nickdom | use wait or answer after bridge maybe |
| 08:18.48 | *** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 08:18.58 | E-bola | Can anybody recomend a german SIP provider? |
| 08:23.44 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 08:24.40 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:27.52 | gnorbert | Could somebody help, what can be wrong with this sip.conf file? http://pastebin.com/d749f4020 (Sorry for this form, I couldn't paste a wrong call, that computer had to be shut down for a few minutes) |
| 08:30.35 | *** join/#asterisk Paige (n=Paige@2001:470:1f05:531:21f:c6ff:fe48:8988) |
| 08:30.43 | Paige | hello |
| 08:30.57 | Paige | looking for help with mysql and cdr storage |
| 08:31.04 | oej | gnorbert: if you need help, you need to say what leads you to believe that somethings is wrong. |
| 08:31.27 | Paige | anyone have any expereince with triggers and or stored procedures? |
| 08:31.29 | gnorbert | oej: Sorry, my fault. :) |
| 08:32.02 | oej | No worries, just helping you to get help |
| 08:32.08 | gnorbert | oej: I can connect to the server from both 2 computers, but I can speak only from one of them in a meetme conference. |
| 08:33.23 | skyNomad | Paige: What's the problem? |
| 08:34.09 | Paige | skyNomad, hi. I am trying to get * to write cdr's with a rate applied to them based on a rates table |
| 08:34.38 | skyNomad | Paige: Alright, and you are wanting to do it with MySQL? |
| 08:35.09 | Paige | yes or with *, which ever is most efficient |
| 08:35.09 | skyNomad | brb |
| 08:35.19 | Paige | ok |
| 08:36.20 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:37.11 | skyNomad | back |
| 08:37.15 | *** join/#asterisk tengulre (n=tengulre@125.70.84.164) |
| 08:37.36 | skyNomad | Paige: So what exactly is your problem? |
| 08:38.15 | Paige | well i am trying to write a billing system |
| 08:39.09 | Paige | i figure that I will need triggers or stored procedures to work out the call charges |
| 08:39.11 | skyNomad | Paige: me too. But it is going to require a very funky stored procedure if you want MySQL to handle it. |
| 08:39.27 | skyNomad | Paige: What language are you writing the billing system in? |
| 08:39.33 | Paige | in php |
| 08:39.46 | skyNomad | Paige: hehe... I'm also busy writing a billing system in PHP. |
| 08:39.52 | Paige | nice |
| 08:40.07 | Paige | perhaps we could collaborate |
| 08:40.09 | Tobsn | why dont you just buy one? |
| 08:40.18 | skyNomad | Paige: I think it is much better to get PHP to calculate the rate, and then update the CDR table. |
| 08:40.39 | Paige | skyNomad, you mean like from a cron job? |
| 08:40.43 | skyNomad | Tobsn: Because I can do a better job than what is currently avaliable. Also, because I need spcialised features. |
| 08:40.46 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 08:40.56 | Tobsn | aha |
| 08:40.58 | Paige | Tobsn, same reasons |
| 08:40.59 | skyNomad | Paige: No, from the AGI script that handles the call. |
| 08:41.12 | Paige | interesting idea |
| 08:41.26 | skyNomad | Paige: Well, it depends on how you want to structure the billing I suppose. |
| 08:41.30 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 08:41.53 | skyNomad | Paige: If you want the billing to be done in real time, then the AGI should handle it. But if you want to do the billing in batches, you can use a cron job. |
| 08:41.59 | Paige | right now i only bill for international (outside of north america) |
| 08:43.54 | Paige | Tobsn, i have not found any decent php based billing systems. do you have any suggestions? |
| 08:44.27 | Paige | skyNomad, i am not using any agi scripts yet, i have yet to leabr how to use them |
| 08:44.34 | Paige | learn |
| 08:44.34 | skyNomad | Paige: What I do, is I get the callers to first enter in a PIN number, then I use that to authenticate and check their balance. Then I ask them to enter in the destination number. Then PHP calculates what trunk to use and the maximum amount of time they are allowed to call that destination for according to the billing rates. Then I execute the dial command, setting the max length duration. After the call is finished, the relevant amount is deducted from |
| 08:44.52 | Tobsn | dont know but im sure if its about pure billing and cart system there are existing solutions |
| 08:45.28 | skyNomad | There is a2billing , but that looks like it was programmed by a monkey with a hammer. But... it works...somehow. |
| 08:45.40 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
| 08:45.41 | Paige | lol |
| 08:45.50 | Tobsn | sounds like wordpress haha |
| 08:45.56 | skyNomad | only worse. |
| 08:46.04 | skyNomad | much much worse. |
| 08:46.11 | skyNomad | It really is shocking programming. |
| 08:46.14 | Paige | skyNomad, i would love to dfo a pin based billing system as well |
| 08:46.22 | Paige | and sell pre-paid cards |
| 08:46.32 | skyNomad | Paige: Check out http://phpagi.sourceforge.net |
| 08:46.48 | Paige | i have, it is a bit beyond me at this time |
| 08:46.59 | Paige | i have been out of the game too long |
| 08:47.00 | Sniper_VOIP | Hi all, I created a [general] peer in sip.conf to accept calls to a specific extension...The problem that when dialing the extension I got Service Unavailable with No route to destination |
| 08:47.08 | skyNomad | Writing AGI scripts in PHP is actually really simple. |
| 08:47.38 | mallochig1 | So, how is the asterisk type of business? |
| 08:47.40 | mallochig1 | lol |
| 08:48.07 | mallochig1 | I'm hearing bad programming and php and I have to wander what I am missing out on :) |
| 08:49.06 | skyNomad | mallochig1: Not in relation to Asterisk itself, but we (by we I mean myself) were insulting a PHP app to add billing functionality on to Asterisk. |
| 08:50.32 | mallochig1 | Who are you billing? |
| 08:50.53 | skyNomad | clients |
| 08:50.59 | skyNomad | callers |
| 08:51.37 | mallochig1 | Are they sip clients or are you doing a calling card type of system? |
| 08:51.41 | skyNomad | I'm using Asterisk as a callcard based system. |
| 08:52.36 | Paige | skyNomad, another thing that i wish to do is insert a city/state/country entry for each call |
| 08:52.58 | Paige | based on npa/nxx and country codes |
| 08:53.58 | mallochig1 | How much do you charge? People actually still use the ol' calling card aye? Are you doing this overseas or just within the us? |
| 08:54.13 | skyNomad | mallochig1: http://www.worldchat.co.za |
| 08:54.55 | skyNomad | Paige: We do that, we basically have a rates table with the destination country, the dialing prefix, and then the cost per billing block (ie 60 seconds). |
| 08:55.12 | skyNomad | Paige: Then it is easy to link up the call with the country, the cost etc... based on the dialing prefix. |
| 08:55.17 | Paige | cool. do you use e164 for thwat? |
| 08:56.02 | mallochig1 | interesting.. |
| 08:56.55 | skyNomad | Paige: Yup. |
| 08:58.38 | Paige | out of my league right now |
| 09:00.33 | *** join/#asterisk oilinki3 (n=oil@ppp-124-121-251-56.revip2.asianet.co.th) |
| 09:01.00 | skyNomad | Paige: It's actually not as difficult as it seems. You should check out that PHPAGI stuff, and maybe look at how A2Billing does it, just brace yourself for the complete lack of formatting and the really obscure use of objects. |
| 09:04.24 | *** join/#asterisk genin (n=phrame@ANice-252-1-10-118.w82-122.abo.wanadoo.fr) |
| 09:04.31 | genin | mornin folks |
| 09:06.10 | genin | say i have a call center full of empployees with indian accents and i want to reduce and modify the accent of them in real time. what are the best options? other than making them all go to language school. i was thinking if there is a professional type software i could put on each ones computer and then have them use it in conjunction with a softphone |
| 09:06.39 | Paige | genin, lol |
| 09:06.43 | genin | heh |
| 09:06.47 | skyNomad | hahahahaha |
| 09:06.49 | genin | i know it sounds funny |
| 09:06.55 | skyNomad | This sounds very 007 |
| 09:07.01 | genin | exactly |
| 09:07.04 | Paige | hire American |
| 09:07.12 | genin | i tried to tell my boss that even the CIA doesnt have anything like this |
| 09:07.13 | skyNomad | what about a "speech wobble stabalizer" sort of thign? |
| 09:07.15 | skyNomad | *thing |
| 09:07.23 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-141-153.neo.res.rr.com) |
| 09:07.35 | genin | im thinking it will all sound weirder than just the indian accent |
| 09:07.58 | skyNomad | Yes it will. |
| 09:08.08 | Paige | hire American, end of accent problem |
| 09:08.10 | genin | recording a voice and changing it after is one thing but in real time recognizing a voice and then changing it so it sounds normal |
| 09:08.13 | genin | yeah right |
| 09:08.21 | skyNomad | What about doing text-to-speech ? |
| 09:08.23 | the_5th_wheel | genin: Hire south african. Everyone understands our flavor of english |
| 09:08.30 | genin | :) |
| 09:08.33 | JoZu | genin: http://www.lobstertech.com/code/voicechanger/ |
| 09:08.37 | genin | text to speech |
| 09:08.44 | skyNomad | So the call center people just need to type really fast. |
| 09:08.53 | genin | HEH |
| 09:09.06 | genin | and have a robot lady voice talk to people? |
| 09:09.07 | genin | hrm |
| 09:09.08 | skyNomad | Then espeak or festival can synthesize the voice for you ;) |
| 09:09.13 | skyNomad | exactly. |
| 09:09.16 | genin | festival |
| 09:09.21 | genin | hrm is that better than indian |
| 09:09.22 | genin | heh |
| 09:09.34 | skyNomad | genin, or you could replace all your employees with a simple bash script. |
| 09:09.40 | genin | not to be racist sorry if there are any indians in here |
| 09:09.41 | Paige | great way to keep customers |
| 09:09.41 | genin | HEH |
| 09:09.45 | *** join/#asterisk Tili (n=tili@58-27-158-26.wateen.net) |
| 09:09.52 | genin | i can bash my employees? |
| 09:09.56 | skyNomad | yup |
| 09:10.08 | tzafrir_laptop | tcsh them, even |
| 09:10.14 | genin | sweet |
| 09:10.19 | genin | JoZu |
| 09:10.27 | genin | thanks for the voicechanger link |
| 09:10.32 | Tili | GotoIf($[${CALLEDSUBADDR}="US-ME-ME"]?6:7) when I use this asterisk tries minus operator in US-ME-ME |
| 09:11.21 | genin | cool thanks for the ideas ;) |
| 09:11.26 | tzafrir_laptop | once mistaked bash.org to be the homepage of gnu bash |
| 09:11.35 | tzafrir_laptop | which made no sense |
| 09:11.58 | skyNomad | genin: good luck. |
| 09:16.41 | JoZu | a question: its possible to put an url (http://127.0.0.1/vtigercrm2/asterisk/call_ident.php?callerid=20010&extension=SIP/20006) in queue? |
| 09:17.22 | JoZu | 1,1,queue(20014|t|URL)? |
| 09:18.50 | JoZu | i need to recive this url for determine who is callingme and see the client in vtiger |
| 09:22.13 | skyNomad | JoZu: Could you not just set the CallerID or set an additional variable in the dial plan? |
| 09:24.18 | *** part/#asterisk mallochig1 (n=Administ@63.135.231.241) |
| 09:28.03 | JoZu | in http://127.0.0.1/vtigercrm2/asterisk/call_ident.php?callerid=20010&extension=SIP/20006, 20010 is the caller and 20006 is agent |
| 09:29.12 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-9fc13d556f5402fb) |
| 09:31.47 | the_5th_wheel | can anyone tell me why the destination for a call in my cdr would be listed as 'h' ? |
| 09:32.07 | *** join/#asterisk LND (n=chatzill@host90-152-16-28.ipv4.regusnet.com) |
| 09:42.19 | wizzy_ | I am failing to build chan_zap.so on Ubuntu 8.04 from svn head. I *think* it is something to do with tonezone. When I use "make menuconfig" chan_zap is XXX'ed out - saying Depends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E), tonezone( |
| 09:43.17 | wizzy_ | what do the (M) and (E) s mean, and is tonezone missing, or is it just a too-long-line ? |
| 09:43.33 | Great_Anta_Baka | if i have two companies running off the same pabx how do i give each one seperate music on hold? |
| 09:43.57 | wizzy_ | I symlink'ed /usr/include/dahdi to zaptel |
| 09:44.42 | skyNomad | Great_Anta_Baka: You could probably do it by using different contexts for each company. |
| 09:45.53 | Great_Anta_Baka | each one has a different context |
| 09:46.23 | skyNomad | So, in each context, you set the music on hold. |
| 09:46.36 | Great_Anta_Baka | say context for company one is called conta and for company 2 its contb do i just make a context in music on hold called [conta] and [contb] |
| 09:47.01 | Great_Anta_Baka | and put the moh directory in there? |
| 09:47.05 | tzafrir_laptop | wizzy_, there's no chan_zap in SVN head. There's chan_dahdi |
| 09:47.27 | skyNomad | Great_Anta_Baka: You could try that. I think it might work. I haven't actually tried that myself. |
| 09:47.31 | tzafrir_laptop | In 1.4 it will build with either zaptel or dahdi. In 1.6.0 and trunk: only with dahdi |
| 09:47.34 | Sniper_VOIP | Hi all...I created a general peer to receive calls on sip.conf...When calling a specific extension from an Openser server I got No route to destination |
| 09:47.53 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 09:48.02 | Great_Anta_Baka | how would you do it skyNomad? |
| 09:48.03 | *** join/#asterisk xenonex (n=xenonex@89.218.237.48) |
| 09:48.25 | tzafrir_laptop | wizzy_, dahdi is different than zaptel in a number of small but subtle ways. If you try pretending that zaptel is dahdi it will probably eventually just won't work |
| 09:50.03 | Sniper_VOIP | Hi all...I created a general peer to receive calls on sip.conf...When calling a specific extension from an Openser server I got No route to destination |
| 09:52.01 | wizzy_ | tzafrir_laptop: thanks. Not sure how chan_zap got there. I need this for an astribank. do I use chan_dahdi ? "make menuconfig" still lists chan_zap (??) and chan_dahdi is also XXX'ed - Depends on: res_smdi(M), dahdi(E), tonezone(E) |
| 09:54.41 | wizzy_ | I did ./configure again - chan_zap is gone |
| 09:54.44 | oilinki3 | humh. it's more easy to setup dialplans etc. editing config files with vi, that using the gui |
| 09:54.55 | skyNomad | Great_Anta_Baka: I'm just getting a URL for you. |
| 09:55.15 | Great_Anta_Baka | nice .. thanks skyNomad |
| 09:55.39 | tzafrir_laptop | dahdi should come from dahdi-linux . libtonezone: from dahdi-tools |
| 09:56.03 | tzafrir_laptop | to see what is available: look at build_tools/menuselect-deps |
| 09:57.45 | skyNomad | Great_Anta_Baka: http://www.automated.it/asterisk/lah-3-6-05_6.html |
| 09:58.34 | skyNomad | Great_Anta_Baka: I was wrong, apparently, you set up different classes of Music On Hold in the musiconhold.conf file, and then in the extensions.conf file, you specify which class to play. |
| 09:59.11 | *** join/#asterisk blq (n=Bl@dslb-088-066-251-002.pools.arcor-ip.net) |
| 09:59.33 | *** join/#asterisk blq (n=Bl@dslb-088-066-251-002.pools.arcor-ip.net) |
| 10:01.09 | wizzy_ | tzafrir_laptop: thanks. Am getting dahdi-kernel and dahdi-tools from svn |
| 10:01.55 | tzafrir_laptop | What Astribank is it, BTW? |
| 10:03.53 | Great_Anta_Baka | thanks skyNomad |
| 10:04.12 | wizzy_ | tzafrir_laptop: usb:008/009 xpp_usb+ e4e4:1152 Astribank-multi FPGA-firmware |
| 10:04.43 | wizzy_ | 32 channel ? |
| 10:08.37 | wizzy_ | I presume I have to flush out all zaptel things from the kernel, startup, etc ? |
| 10:12.06 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 10:18.59 | Sniper_VOIP | Hi All...Can you help me in creation a new peer in order to send calls from an OpenSer server to Asterisk Server? |
| 10:19.38 | wizzy_ | dahdi is now available in make menuconfig |
| 10:27.08 | *** join/#asterisk rabby (n=rabby@p4FCE9BFB.dip0.t-ipconnect.de) |
| 10:27.14 | rabby | hi |
| 10:27.51 | rabby | i switched asterisk to verbose mode 10 and wonder why it does not show anything when there comes a call in :/ |
| 10:28.08 | rabby | capi info tells me: Contr1: 2 B channels total, 2 B channels free. |
| 10:28.13 | rabby | so what may be wrong? |
| 10:33.06 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 10:33.56 | hunmonk | does asterisk set any channel variables if a REALTIME() query fails? i'd like some way to test the status of the query in the dialplan |
| 10:39.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 10:39.58 | E-bola | Can anybody recomend a german SIP provider? |
| 11:55.29 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 11:55.29 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- Related channels: #asterisknow, #asterisk-gui, #switchvox, #freepbx, #asterisk-commits, #asterisk-bugs, #asterisk-dev |
| 11:55.59 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 11:56.40 | tzafrir_laptop | jbot, welcome |
| 11:56.41 | jbot | thx |
| 11:58.05 | jazzmann | Hi is there anyway I can use my mobile phone or telephone line with voice modem without any other pci card |
| 12:02.03 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:02.03 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:02.50 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 12:03.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.05 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:17.02 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 12:17.35 | [TK]D-Fender | Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs... |
| 12:17.36 | [TK]D-Fender | ...#asterisk-dev |
| 12:17.47 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 12:18.07 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 12:18.24 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 12:27.21 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:27.21 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:28.23 | oilinki3 | what would be an easy and practical way to handle user account/dial plans etc on asterisk? asterisknow or realtime with some own interface to the mysql-db? |
| 12:28.56 | oilinki3 | asterisknow is a bit confusing to use (and a bit slow when the server is on another continent) |
| 12:30.30 | Great_Anta_Baka | my ivr takes a long time to invoke reponses from the pabx when keys are being pressed on the phone, what could possible reasons be for this? |
| 12:31.09 | [TK]D-Fender | Great_Anta_Baka: dialplan overlap |
| 12:31.18 | [TK]D-Fender | oilinki3: VI <- |
| 12:32.04 | oilinki3 | [TK]D-Fender: VI ? |
| 12:32.05 | lmadsen | oilinki3: can you build the simple user interface to add/remove/change users? If so, the most flexible would be a realtime solution from a database using ODBC |
| 12:32.28 | [TK]D-Fender | oilinki3: emacs, mc, nano, pico, gedit, kwrite, or any of a million others. Take your pick |
| 12:32.44 | lmadsen | oilinki3: otherwise, you could use the asterisk GUI which modified the users.conf file, so that might be your next best step. The asterisk gui is independent of asterisknow, so you can utilize that with just stock asterisk |
| 12:32.49 | oilinki3 | lmadsen: I'm not familiar with ODBC, mysql + php a bit. |
| 12:33.01 | oilinki3 | [TK]D-Fender: ah vi :) |
| 12:33.34 | seanbright | emacs <3 |
| 12:34.04 | lmadsen | oilinki3: ahhh... ODBC is just an abstraction layer -- you can get information about how to use it in the TFoT book (www.asteriskdocs.org). If you can use mysql + php, you're well on your way |
| 12:34.09 | lmadsen | vi <3 |
| 12:34.16 | Sniper_VOIP | [TK]D-Fender, I have an OpenSer server that trying to send a call to an extension on asterisk server...This extension has been created on extensions_custom.conf in order to record voice message and playback this voice |
| 12:34.47 | [TK]D-Fender | Sniper_VOIP: extensions_custom.conf <- meaningless file. Says nothing (positive) |
| 12:34.50 | oilinki3 | lmadsen: ok thanks for the info. in that case I might go for realtime, mysql, php combination. |
| 12:34.51 | Sniper_VOIP | [TK]D-Fender, When trying to make a call to that specific extension I got No ROute to Destination on the Asterisk side |
| 12:35.07 | lmadsen | oilinki3: thats my favourite combo too |
| 12:35.16 | [TK]D-Fender | Sniper_VOIP: And per the laswt time you brought this up, the exten dialed was not in the context the call landed on. |
| 12:35.36 | oilinki3 | then again. I wish to get the user interface in the way that it's easy to use for non-technical people as well. |
| 12:35.37 | lmadsen | oilinki3: you'll also probably like func_odbc a lot too -- check that out in the book as well |
| 12:35.50 | oilinki3 | lmadsen: I will. |
| 12:36.42 | lmadsen | oilinki3: yep -- you should be able to build some simple pages for them as long as you know just enough php to be dangerous. I'm not a programmer, but I could probably build a simple user interface to make add/move/changes to the database from php in order to manage some simple aspects of the pbx system |
| 12:36.53 | Sniper_VOIP | [TK]D-Fender, Can you tell me where i can find the manual to do that? |
| 12:37.24 | [TK]D-Fender | Sniper_VOIP: manual? IN CLI with SIP debug it tells you to your face exactly what context its looking for that context in. |
| 12:37.26 | lmadsen | oilinki3: just don't even bother trying to create an interface to really modify the dialplan. You can use func_odbc so that your GUI just has to modify the database, then the dialplan will just look at the database, and you don't have to do anything crazy like have your gui modify the extensions.conf files, etc.... |
| 12:37.37 | oilinki3 | lmadsen: i'm in the same level. not an programmer, but can do a bit of scripting. |
| 12:37.51 | [TK]D-Fender | s/context in/extension in/ |
| 12:38.02 | marc7 | ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '!=', expecting $end; Input: |
| 12:38.02 | lmadsen | oilinki3: you should be fine then -- just check out the database route, and don't make your php scripts have to really modify the asterisk configuration files -- you'll be a lot better off |
| 12:38.05 | marc7 | what module am I missing? |
| 12:38.07 | lmadsen | asterisk --> database <-- GUI |
| 12:38.32 | [TK]D-Fender | marc7: Not missing a module, you have a DIALPLAN error |
| 12:38.47 | marc7 | exten => _NXXNXXXXXX,9,GotoIf($[${GID} != 0]?10:20) |
| 12:38.53 | marc7 | i suppose that's not correct syntax anymore |
| 12:38.57 | marc7 | *looks up what should be correct* |
| 12:39.09 | oilinki3 | lmadsen: simple is good for me. takes a bit more time to think, but then again, can undestand it better. |
| 12:39.15 | [TK]D-Fender | marc7: ${GID} was BLANK when you ran this. Yuo can't ha ve"balnk" and then an operator |
| 12:39.31 | marc7 | D-Fender: gotcha, thank you. |
| 12:39.48 | *** join/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-ab0f4d280b1fa09a) |
| 12:39.51 | [TK]D-Fender | marc7: $[ != 0 <- is what it evaluated to. |
| 12:40.01 | lmadsen | marc7: this is a good reason why putting double quotes around comparisons is a good idea |
| 12:40.33 | lmadsen | marc7: and better syntax would be to use priority labels and the 'n' priority |
| 12:40.56 | marc7 | lmadsen: migrating an old 1.2 dialplan right now, labels are definitely on the way |
| 12:41.05 | lmadsen | marc7: ahhh.. understood :) |
| 12:43.13 | [TK]D-Fender | labels not needed. |
| 12:43.48 | marc7 | there's a lot of unnecessary jumping around in this plan, would clean things up a bit |
| 12:45.57 | [TK]D-Fender | marc7: exten => _NXXNXXXXXX,9,GotoIf($["${GID}" = "0"]?20) <- better coding |
| 12:46.30 | marc7 | *nod* i was going to ask what lmadsen meant by double quotes, looked up GotoIf on voip-info.org and found that suggestion |
| 12:48.27 | Sniper_VOIP | [TK]D-Fender, If what you are saying is true then why If I make a call from another asterisk server to that extension I can record my voice? |
| 12:48.35 | [TK]D-Fender | marc7: *'s evaluator is a dumb as it gets. Purely literal |
| 12:49.11 | [TK]D-Fender | Sniper_VOIP: Because the call is processed by a peer that it DOES recognize and the extension is included in its path. |
| 12:49.40 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
| 12:49.52 | [TK]D-Fender | Sniper_VOIP: Because the call is processed by a peer that it DOES recognize and the extension is included in its path. |
| 12:51.50 | *** join/#asterisk NoxIn- (n=noxin@zorlit.org) |
| 12:52.00 | *** join/#asterisk nate187 (n=nate@gw.bigrivertel.net) |
| 12:52.08 | NoxIn- | Hello |
| 12:55.29 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 12:56.12 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 13:03.52 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:05.46 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 13:08.17 | *** join/#asterisk coppice (n=chatzill@61.157.17.210.dyn.pacific.net.hk) |
| 13:09.17 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:10.32 | *** join/#asterisk questprojects (n=mg@82-71-8-45.dsl.in-addr.zen.co.uk) |
| 13:11.12 | *** join/#asterisk cNILE (n=canabis@ip70-172-239-41.br.br.cox.net) |
| 13:11.48 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:12.40 | qp | anyone know if theres a working iaxclient channel? #iaxclient seems empty |
| 13:13.43 | [TK]D-Fender | qp: #asterisk-dev may be able to help |
| 13:14.04 | qp | ok cheers d-fender |
| 13:14.12 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 13:15.09 | cNILE | hey guys, newbie here, I just installed a new, Digium 8-port fxo card (AEX800), and all is well, but channels 3 & 4 show a "WCTDM, no module" in zapata.conf, however a "zap show channels" shows all 8 channels. Any suggestions? |
| 13:15.38 | *** join/#asterisk eliel (n=eliel@200.61.172.61) |
| 13:16.56 | seanbright | #asterisk-dev != #iaxclient |
| 13:17.23 | Sniper_VOIP | [TK]D-Fender, I did what you asked me to doo..I'm getting now "declined" from Asterisk server |
| 13:18.21 | [TK]D-Fender | seanbright: true, but there are real deveplopers there who because of involvement in coding the protocol might be able to help. I certainly can't think of anywhere better. |
| 13:18.35 | [TK]D-Fender | Sniper_VOIP: and what did I ask you to do? |
| 13:18.55 | Sniper_VOIP | [TK]D-Fender, add the extension to the context created |
| 13:18.56 | mvanbaak | [TK]D-Fender: their website states: |
| 13:18.59 | mvanbaak | "# You can also catch some of the developers online at irc.freenode.net, #iaxclient, or #asterisk" |
| 13:19.45 | [TK]D-Fender | <Sniper_VOIP>[TK]D-Fender, add the extension to the context created <- EXTENSIOSN have absolutely nothing to do with this SIP AUTH PROBLEM. |
| 13:20.07 | *** join/#asterisk MrNaz (n=naz@ppp59-167-169-142.lns1.mel4.internode.on.net) |
| 13:20.11 | Sniper_VOIP | [TK]D-Fender, Oh my god...Are you playing with me? |
| 13:20.26 | [TK]D-Fender | Sniper_VOIP: No, you are simply clueless. |
| 13:20.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:20.50 | Sniper_VOIP | [TK]D-Fender, fuck you and your mother bitch |
| 13:21.13 | [TK]D-Fender | \o/ |
| 13:21.27 | [TK]D-Fender | Another FreePBX schmuck bites the dust |
| 13:21.30 | qp | heh |
| 13:21.30 | brodiem | lol |
| 13:21.59 | [TK]D-Fender | Wonder if I should post my "private thoughts #asterisk shit-list" publicly :) |
| 13:22.03 | brodiem | wasn't he the guy trying to accept calls from openser |
| 13:22.08 | [TK]D-Fender | BroYup |
| 13:22.15 | [TK]D-Fender | brodiem: rather. |
| 13:22.24 | creativx | autocompletion not working there [TK]D-Fender? ;) |
| 13:22.35 | [TK]D-Fender | Dammit, why can't everyone auto-complete properly on 3 chars! |
| 13:22.43 | brodiem | heh |
| 13:22.57 | creativx | predictive autocompletion |
| 13:22.58 | [TK]D-Fender | brodiem: Your nick is problematic. |
| 13:23.25 | [TK]D-Fender | creativx: YES! Multiple possibly matches should pick the last one that SPOKE! |
| 13:24.20 | creativx | i know |
| 13:24.24 | creativx | ive wondered why it isnt so |
| 13:24.32 | creativx | lazy programmers |
| 13:24.36 | *** join/#asterisk nny_1 (n=nny_1@64.203.237.47) |
| 13:24.54 | brodiem | [TK]D-Fender: Must be conflicting with someone in another channel? |
| 13:24.58 | nny_1 | screams coffee deprived obscenities |
| 13:25.04 | [TK]D-Fender | brodiem: No, in this channel |
| 13:25.33 | nny_1 | seriously.. is dtmf compliance that god damn difficult |
| 13:25.33 | [TK]D-Fender | namely BrokenNoze and bronx |
| 13:25.39 | brodiem | ahh |
| 13:26.13 | nny_1 | think i have permanently lost my patience with vitelity |
| 13:27.00 | brodiem | nny_1: how come out of curiousity? I used them for a bit a year or so ago.. |
| 13:27.03 | nny_1 | runs off to snort the bottom of the coffee bag |
| 13:27.12 | [TK]D-Fender | nny_1: I think I heard something about them switching modes without telling people too well. |
| 13:27.21 | nny_1 | yeah.. they did |
| 13:28.05 | nny_1 | I dunno.. i am hoping it isn't their fault.. an exodus to another carrier will be painful and time consuming |
| 13:28.29 | nny_1 | although.. call to number A through pots/ cellular = dtmf fine. Call from vitelity.. boom |
| 13:29.23 | brodiem | nny_1: you use Ast1.2 or 1.4? |
| 13:29.29 | nny_1 | 1.4 |
| 13:29.35 | nny_1 | 1.4.21.2 to be exact |
| 13:30.03 | brodiem | ah ok, some carriers I find would never work properly with 1.2's broken RFC2833 support |
| 13:31.49 | *** join/#asterisk fish-bulb (n=cstewart@conference/asterisk/x-f53fe694cf7df231) |
| 13:33.14 | [TK]D-Fender | brodiem: And 1.2 has not been "supported" for a long time. Wake up and smell the toast burning :) |
| 13:34.09 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 13:36.19 | brodiem | Just a thought, I know it wasn't intended to be rewritten on 1.2 |
| 13:38.22 | [intra]lanman | how's the rfc2833 support in 1.4? |
| 13:38.42 | [intra]lanman | or 1.6 if anyone's using it |
| 13:39.46 | brodiem | [intra]lanman: No idea if it's fully standards-compliant but has never acted up on me before |
| 13:40.06 | [intra]lanman | brodiem: what carrier[s] are you having problems with? |
| 13:40.39 | nny_1 | I am trying to hit up another asterisk box on a didww number. The console on the other box reports 999 or 95 or 9 when trying to pass 995 to the beast.. I just tried it from my pots line and it worked flawlessly. I think it is safe to say its the sip outbound |
| 13:40.40 | brodiem | [intra]lanman: Bandwidth.com comes to mind with a 1.2 install I used to have |
| 13:40.48 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 13:40.58 | *** join/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 13:41.19 | nny_1 | i need another number to test my sip outbound dtmf on |
| 13:41.30 | ipstatic | Has anyone ran into problems with asterisk not being able to save new voicemail passwords to voicemail.conf? |
| 13:41.39 | ipstatic | I am now getting this on the cli: Unable to write /etc/asterisk/voicemail.conf (Operation not permitted) |
| 13:41.47 | [intra]lanman | brodiem: have you tried any other sip stacks? freeswitch or sipx or something to make sure it's not a bw.com problem? |
| 13:42.05 | ipstatic | however the file is group owned by asterisk, and it just chmoded it to 777 and it still doesn't work |
| 13:42.10 | [intra]lanman | although, i guess dtmf isn't technically sip related, but you get what i'm saying |
| 13:42.39 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:43.01 | [intra]lanman | ipstatic: i had some weird problems like that when selinux was enabled.... also when using an nfs share |
| 13:43.07 | [intra]lanman | ipstatic: you doing either of those? |
| 13:43.26 | ipstatic | [intra]lanman: I am using neither |
| 13:43.37 | ipstatic | [intra]lanman: local fs and no selinux |
| 13:43.44 | [intra]lanman | shrugs |
| 13:43.46 | brodiem | [intra]lanman: right, and no, they acknowledge DTMF is problematic with their gateways on 1.2 |
| 13:43.48 | *** join/#asterisk moy (n=moy@nat/ibm/x-2fcb37bd356a76f4) |
| 13:44.00 | nny_1 | ipstatic i did at one point |
| 13:44.22 | ipstatic | nny_1: what did you do to resovle it? |
| 13:44.28 | nny_1 | ipstatic permissions issue >< |
| 13:44.53 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:45.10 | ipstatic | nny_1: -rwxrwxrwx 1 root asterisk 12151 2008-08-11 12:40 voicemail.conf |
| 13:45.41 | ipstatic | nny_1: /usr/sbin/asterisk -U asterisk -G asterisk |
| 13:46.29 | nny_1 | ipstatic may want to check the formatting of the voicemail.conf file as well. |
| 13:46.30 | [intra]lanman | brodiem: you might consider putting freeswitch in between your asterisk box and the gateway... it can relay or decode/reencode dtmf |
| 13:46.43 | [TK]D-Fender | ipstatic: pastebin the full "ls -la" dump of that folder including the call. |
| 13:47.13 | nny_1 | ipstatic seem to remember someone had an extra [mailboxname] line in the file and it was trying to read the entries below it |
| 13:47.54 | nny_1 | ipstatic even though asterisk itself saw and registered the boxes properly.. |
| 13:48.11 | arpu | hello again |
| 13:48.16 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:48.26 | arpu | anyone an idea why i can not not see any tx rx |
| 13:48.29 | arpu | http://pastebin.com/m39658042 |
| 13:48.38 | brodiem | [intra]lanman: yeah it wasn't a big deal anyway since I was planning on upgrading it to 1.4 |
| 13:48.42 | arpu | smsm over smsc sip asterisk |
| 13:48.51 | nny_1 | brb aneurysm |
| 13:49.36 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:49.51 | ipstatic | [TK]D-Fender: http://pastebin.com/m5c47a751 |
| 13:50.27 | mgroman | Hello, Has anyone here used the new Cepstral TTS engine with Asterisk (version 5) and is there any issues compared to version 4? |
| 13:50.42 | brodiem | ipstatic: chmod 755 /etc/asterisk |
| 13:50.59 | nny_1_away | yar |
| 13:51.12 | brodiem | and chmod 644 your *.conf |
| 13:51.29 | brodiem | and chown asterisk. /etc/asterisk -R |
| 13:51.41 | [TK]D-Fender | Good idea |
| 13:52.03 | nny_1_away | or better yet chown asterisk:asterisk /etc/asterisk -R |
| 13:52.15 | brodiem | 08:50 < brodiem> and chown asterisk. /etc/asterisk -R |
| 13:52.22 | brodiem | trailing period includes its group.. |
| 13:52.25 | nny_1_away | ! |
| 13:52.28 | nny_1_away | i fail! |
| 13:52.35 | nny_1_away | (did not know that) thanks |
| 13:53.00 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:54.04 | ipstatic | thank you!! |
| 13:54.25 | ipstatic | now, why did this stop working all of the sudden? |
| 13:54.34 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:54.52 | nny_1_away | trudges off to read the man pages on chown harder. I'll get you next time brodiem.. NEXT TIME! |
| 13:57.35 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 13:58.34 | brodiem | ipstatic: probably when you set that restricted delete flag on the directory.. |
| 13:59.00 | brodiem | but otherwise your permission structure is pretty messed up regardless.. |
| 13:59.04 | brodiem | nny_1_away: lol |
| 14:00.05 | arpu | no one an idea ? |
| 14:00.12 | arpu | i get no debug outbut |
| 14:00.22 | ipstatic | brodiem: thanks, I followed this to setup the structure: http://www.voip-info.org/wiki-Asterisk+non-root |
| 14:00.36 | ipstatic | guess I should have checked it a little more closely |
| 14:01.07 | brodiem | ipstatic: hate to tell you but it was far from the instructions there :) |
| 14:05.05 | ipstatic | brodeim: thanks, maybe I should go off and read the man pages :) |
| 14:06.15 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 14:06.42 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 14:06.44 | thomas | hello |
| 14:06.59 | thomas | is it posible to use sip also over tcp instead udp? |
| 14:07.26 | *** join/#asterisk shido6 (n=shido6@74-132-94-24.dhcp.insightbb.com) |
| 14:09.23 | brodiem | thomas: IIRC it isn't officially supported until 1.6, although a patch exists for 1.4. |
| 14:09.38 | brodiem | my memory could be wrong there.. |
| 14:09.48 | x86 | I'm getting a shit ton of overruns on a T1 interface going to the telco (using a sangoma A102DE card, CAS T1 running E&M Wink signaling, receiving timing from the line) |
| 14:10.12 | x86 | any ideas why this might be happening? perhaps the telco is having issues with wiring? |
| 14:10.57 | x86 | brodiem: sip.conf has had TCP options since 1.2, although I'm not sure if it's ever worked or not, as I've never tried it |
| 14:11.49 | [TK]D-Fender | x86: There have been patch attempts, nothing really fully uable, and no back-port. wait for 1.6 |
| 14:12.08 | [intra]lanman | thomas: this is another place that you could drop in freeswitch or openser to act as a relay since both officially support tcp |
| 14:12.31 | [intra]lanman | that is... if you just "need" tcp to work now |
| 14:15.18 | NoxIn- | does anyone knows where I can find the description of the fields of the commande "core show channels concise", I tried the asterisk wiki but didn't found the info. In fact it's only the 2 fields before the duration that I don't find |
| 14:15.47 | *** join/#asterisk poller (n=poller@85.17.230.154) |
| 14:17.06 | poller | Hello. If I've registered the extension 1001 with exten => 1001 in context foo, and included context foo in context bar with include => foo. Shouldn't i be able to call extension 1001 from context bar? |
| 14:17.23 | poller | Either i misunderstood include, or i've done it wrong :) |
| 14:17.36 | lmadsen | poller: pastebin your dialplan -- more than likely you may have an overlap |
| 14:17.56 | lmadsen | if there is a 1000 in bar independent of the one in foo, then the one in bar will take precident |
| 14:18.06 | *** join/#asterisk MindTheGap (n=MindTheG@201.17.149.252) |
| 14:18.09 | x86 | [TK]D-Fender: I'm not the one wanting to use it ;) |
| 14:18.38 | lmadsen | poller: it will work though if you've executed correctly |
| 14:18.46 | lmadsen | Yourname`: ping! |
| 14:19.01 | [TK]D-Fender | poller: so far nothing tells us that your device is using the right context in the first place. |
| 14:19.19 | [TK]D-Fender | poller: PASTEBIN is your friend. |
| 14:19.21 | [TK]D-Fender | ~pb |
| 14:19.22 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:19.23 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:20.02 | poller | lmadsen, [TK]D-Fender: http://bin.bamze.net/1486 |
| 14:20.28 | poller | The device uses contect outgoing, i can do outgoing calls. |
| 14:20.41 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:20.42 | lmadsen | poller: right.... and 1000 will match that pattern |
| 14:20.52 | lmadsen | your pattern is too loose |
| 14:20.57 | poller | You're right! |
| 14:21.01 | lmadsen | of course I am :) |
| 14:21.04 | poller | :-] |
| 14:21.17 | lmadsen | add at least 5 X's before the . |
| 14:21.20 | lmadsen | and it'll work |
| 14:21.43 | poller | Or requre a 0 to reach external line. |
| 14:21.47 | lmadsen | also, replace those pipes with commas so you don't have to do it when you move to 1.6 |
| 14:21.57 | poller | How do I crop the first digit of ${EXTEN} ? :) |
| 14:22.01 | lmadsen | poller: ugh... I hate phone systems that make you dial something prior to dialing out |
| 14:22.10 | lmadsen | looks at file |
| 14:22.10 | poller | I love it |
| 14:22.14 | poller | Makes me feel powerful |
| 14:22.22 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 14:22.25 | lmadsen | makes me assume the admin doesn't know what they're doing :) |
| 14:22.31 | poller | I miss my avaya dialplan :) |
| 14:22.38 | poller | Darn =) |
| 14:22.46 | lmadsen | goes back to work! |
| 14:22.59 | poller | glhf :) |
| 14:23.04 | poller | Thanks for you help |
| 14:24.39 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:28.40 | *** join/#asterisk Robba (n=rob@mail.ampwest.com.au) |
| 14:32.58 | jaytee | [TK]D-Fender, which Polycom phone would you recommend for an agent phone for queued calls? |
| 14:33.34 | [TK]D-Fender | jaytee: All the same, depending on headset connectivity |
| 14:33.45 | [TK]D-Fender | jaytee: they're all jsut phone. |
| 14:36.02 | brodiem | I just learned that I could never use a non-backlit polycom as a daily-use phone, at least with the viewing angle I have on it.. makes me sick to my stomach after a while seeing doubles of the text at my view angle.. |
| 14:36.21 | jaytee | [TK]D-Fender, cool, I wasn't sure if I needed something more than a 320 or 330 like we've been buying. With the old Nortel ACD agent phones I always had to go with a higher end model with more line and function keys. |
| 14:36.31 | jaytee | [TK]D-Fender, thanks man! |
| 14:43.32 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:49.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:52.01 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:53.47 | *** join/#asterisk chackal_sjc (n=pet@200-168-20-217.dsl.telesp.net.br) |
| 15:01.23 | anonymouz666 | what the fuck is FAX NSE in the ATA configuration? |
| 15:02.26 | *** join/#asterisk M1s3ry (n=M1s3ry@conference/asterisk/x-0e17be04f11ab5f5) |
| 15:04.25 | *** join/#asterisk Corydon76-dig (i=silver@pdpc/supporter/bronze/Corydon76-home) |
| 15:04.25 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 15:05.08 | *** join/#asterisk ManxPower (n=manxpowe@68.sub-75-248-203.myvzw.com) |
| 15:06.25 | *** join/#asterisk Dr-Linux|work (n=Nothing@119.63.130.34) |
| 15:06.55 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:10.08 | chackal_sjc | guys |
| 15:10.20 | chackal_sjc | does anyone here compiled iaxclient???? |
| 15:11.12 | coppice | FAX NSE == something in RFC2833 |
| 15:11.54 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 15:12.11 | *** join/#asterisk SparFux (n=raoul@e182024251.adsl.alicedsl.de) |
| 15:12.13 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:12.32 | SparFux | Hi. Is there a cdr view for linux? So that I can comfortably view the asterisk cdr files? |
| 15:13.02 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 15:13.22 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 15:13.22 | [TK]D-Fender | SparFux: "man more" :) |
| 15:13.40 | SparFux | TK]D: LOL :-) |
| 15:13.49 | ManxPower | SparFux: feel free to write one |
| 15:14.20 | [TK]D-Fender | SparFux: depends on what you mean "for linux", and in what format CDR is stored, and what you want to be able to do with it. |
| 15:14.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:15.15 | [TK]D-Fender | SparFux: Go look at the WIKI GUI list. |
| 15:16.14 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.2.87) |
| 15:16.35 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:16.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:18.09 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 15:20.14 | *** join/#asterisk icel (n=dan@75.150.16.110) |
| 15:23.41 | Tebi | hi, does asterisk support sip diversion...diversion header? |
| 15:24.25 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 15:24.48 | [TK]D-Fender | Tebi: chan_sip.c is right there for you to see |
| 15:25.17 | *** join/#asterisk Stereo (n=stereo@78.141.155.243) |
| 15:25.24 | Stereo | hello |
| 15:25.50 | Stereo | I'm trying to connect an asterisk box from behind a nat using siproxd, as described on http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c7s5.html |
| 15:26.21 | [TK]D-Fender | Stereo: Don't bother, read the following : |
| 15:26.23 | [TK]D-Fender | ~sipnat |
| 15:26.24 | 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 |
| 15:26.25 | Stereo | when I dial, I get this error: *CLI> [Aug 12 17:26:04] WARNING[5858]: chan_sip.c:12347 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '68d7ddd8474596650434bf7974caa1bd@bellshare.com'. Giving up. |
| 15:26.26 | [TK]D-Fender | ^^^^^ |
| 15:26.59 | Stereo | thank you :) |
| 15:27.31 | anonymouz666 | [TK]D-Fender: do you already tried to fax with PAP2? |
| 15:27.37 | anonymouz666 | pass-through |
| 15:27.56 | [TK]D-Fender | anonymouz666: Never used a PAP2, not tried FoIP. |
| 15:28.02 | anonymouz666 | it's not FoIP |
| 15:28.06 | anonymouz666 | it's using T1 |
| 15:28.17 | anonymouz666 | just LAN is VoIP with g711u |
| 15:28.24 | icel | anyone have suggestions for loudspeaker setup (paging)? I don't really know what to use. |
| 15:28.44 | anonymouz666 | sometimes works, sometimes doesn't... |
| 15:28.59 | anonymouz666 | I hate FAX so much |
| 15:29.04 | [TK]D-Fender | anonymouz666: the PAP2 is IP no matter what. |
| 15:29.29 | coppice | sometimes works, sometimes doesn't == functioning to spec |
| 15:29.31 | Stereo | yeah, I'm in case #1 on the wiki, tried to configure a proxy, which fails with the error above |
| 15:29.32 | [TK]D-Fender | anonymouz666: I never said anything about with or without T.3X |
| 15:29.36 | ManxPower | I simply never ever have fax issues. |
| 15:29.51 | [TK]D-Fender | Stereo: the OTHER guide. and remove the proxy from the equation. |
| 15:30.07 | anonymouz666 | ManxPower: you probably don't use it :) |
| 15:30.34 | [TK]D-Fender | icel: What do you have, and what do you want to do? |
| 15:31.13 | Stereo | I'll try that |
| 15:31.13 | heedly | FAX generally doesn't work well with VOIP. |
| 15:31.20 | heedly | unless its t38 |
| 15:31.41 | icel | [TK]D-Fender: I have overhead speakers in the building from a previous paging system. I would like to dial an extension from * and send out a page |
| 15:31.44 | [TK]D-Fender | heedly: T.37 ... probably far more reliable ;) |
| 15:32.01 | coppice | definitely far more reliable |
| 15:32.12 | [TK]D-Fender | icel: Go look at getting a Viking analog paging module and hook it up with an ATA |
| 15:32.23 | icel | thx |
| 15:32.41 | ManxPower | anonymouz666: we just don't route faxes over voip or Asterisk |
| 15:33.16 | anonymouz666 | coppice: out of curiosity how was your tests with PAP2 T38? |
| 15:33.20 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 15:33.36 | coppice | PAP2 doesn't support T.38 |
| 15:33.49 | anonymouz666 | even the latest firmware? |
| 15:33.51 | brodiem | PAP2T does |
| 15:33.59 | coppice | PAP2T does not |
| 15:34.04 | *** join/#asterisk _fury (n=flolry@ool-4b7fd6a2.static.optonline.net) |
| 15:35.19 | brodiem | coppice: I thought certain firmware revisions did. |
| 15:35.49 | ManxPower | brodiem: if it suppoers T.38 then you should have T.38 settings in your ATA. |
| 15:35.54 | coppice | I have a PAP2T loaded with the latest firmware, and if you can show me the secret place where they hid T.38 I would be delighted |
| 15:36.10 | brodiem | ManxPower: Yeah, they do - the same options as their RTP3XX which supports T38 |
| 15:36.11 | ManxPower | Just remember that T.38 is a new protocol and did not even exist a few years ago. |
| 15:36.22 | brodiem | coppice: the latest doesn't from what I can remember |
| 15:37.11 | coppice | T.38 is now 9 years old |
| 15:37.24 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 15:37.24 | ManxPower | brodiem: so tell coppice where the option is. |
| 15:37.42 | ManxPower | in telecom "9 years" is "new" |
| 15:37.43 | Paige_ | morning |
| 15:38.01 | Paige_ | where is a good place to learn php agi scripting? |
| 15:38.03 | ManxPower | All the PAP2s support non-T.38 FoVoIP |
| 15:38.04 | Qwell | I still don't trust that new fangled "analog" telephony. |
| 15:38.14 | brodiem | ManxPower: I know he doesn't have it, I'm just saying I've definitely seen a PAP2T include the T38 options - I don't know if it was the firmware, certain HW revision, etc |
| 15:38.23 | ManxPower | But FoVoIP is not T.38. |
| 15:38.35 | [TK]D-Fender | Paige : the BOOK |
| 15:38.37 | [TK]D-Fender | ~book |
| 15:38.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:38.52 | [TK]D-Fender | Paige : 2 entire chapters on it |
| 15:39.11 | Paige_ | i think i have that book,,, |
| 15:39.20 | Paige_ | yes i have it |
| 15:39.27 | coppice | "the latest doesn't support T.38" should be a clue. another clue would be categorical statements from Linksys. :-) As far as I can tell there might have been a 3.something with provisional support. I haven't been able to find any firmware with T.38, though. I think people get confused, as Linksys muddle the documentation for the SPA3102 and SPA2102 with the PAP2 and PAP2T |
| 15:40.19 | brodiem | coppice: I just know I gave up on trying to get it to work a while ago, and I know it was negotiating T.38 from capturing packets. The RTP300 did work (marginally) for me at least.. |
| 15:40.43 | ManxPower | coppice: I think it's sort of funny/sad that people spend so much time and money trying to get FaxOverVoiceOverIP working when a simple solution already exists -- don't run fax over IP. |
| 15:41.13 | coppice | dunno about the RTP300. The SPA2102 and SPA3102 do support T.38, though I don't know how well. |
| 15:41.20 | *** part/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-ab0f4d280b1fa09a) |
| 15:41.23 | brodiem | ManxPower: It's only to satisfy the penny pinchers for me =/ |
| 15:41.38 | ManxPower | brodiem: you sure are spending a lot of money to be a penny pincher. |
| 15:41.44 | ManxPower | Unless you don't get paid for this stuff. |
| 15:43.08 | ManxPower | 4 hrs @ $100/hr = $400, fax line $50/month, take 8 months to even break even |
| 15:43.28 | ManxPower | I imagine it would take much more than 4 hrs to make it work. |
| 15:43.52 | mgroman | 4 hours is a lot in seconds |
| 15:43.54 | ManxPower | So the only one that wins with trying to use T.38 is the consultant, not the customer. |
| 15:44.16 | *** join/#asterisk jameswf-home (n=james@jfinstrom.broker.freenet6.net) |
| 15:45.14 | ManxPower | When I moved our faxes to analog PSTN telco lines the number of complaints about fax went from about 4 - 5/week to none in 15 months. |
| 15:45.20 | brodiem | ManxPower: hey I don't have any argument, but it's such a common complaint and nobody wants to keep a POTS line so I figured I'd just try it out.. and yeah it was a waste of time ;) |
| 15:45.33 | ManxPower | I say THAT saves the customer money. |
| 15:46.08 | ManxPower | a local fax line can be used as a 911 line if there are no other local lines. |
| 15:46.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:46.23 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 15:47.04 | coppice | depends on your volumes. if you are shaking things out for a big deployment it makes sense. if you are pissing around trying to get one fax line working it doesn't |
| 15:47.22 | ManxPower | I'm getting audio quality complaints from one of my sites. Turns out they are running VoIP over a 384K non-QoS frame relay connection. My response was "I told you it was a bad idea, now you see why." |
| 15:47.50 | NoxIn- | and a fax that don't manage to negociate with the fax on the other end is a valid communication from the carrier point of view, and most faxs retry automaticly every 4 minutes ... so it can cost a lot |
| 15:48.13 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:48.44 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
| 15:50.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 15:53.07 | _fury | can anybody reccommend a good DID provider that will work with asterisk? I need just a single channel with a single DID, most of the ones I've found have minimum order requirements. |
| 15:53.56 | brodiem | _fury: you'll find lots of pay as you go vendors on the wiki and the -biz mailing list |
| 15:54.09 | ManxPower | _fury: Vitelity and Teliax |
| 15:54.33 | [TK]D-Fender | _fury: Go check out the following : |
| 15:54.53 | [TK]D-Fender | ~itsplist-us |
| 15:54.55 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 15:55.20 | [TK]D-Fender | ^^^^^ |
| 15:55.20 | *** join/#asterisk bad_duck_ (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 15:55.21 | bad_duck_ | Hi |
| 15:55.21 | _fury | thanks :) |
| 15:55.30 | *** join/#asterisk cjk (n=cjk@vodsl-10830.vo.lu) |
| 15:55.53 | cjk | hi, is it possible to play a soundfile and continue in the dialplan while playing the soundfile ? |
| 15:55.55 | bad_duck_ | I am looking for an usb bluetooth adapter that works with asterisknow and linux debian + asterisk, do you have a idea ? |
| 15:56.28 | _fury | just to confirm, here's what I want to do, I'm just in the beginning research stages of this. We have software here that we use to send fax blasts. I want to set up a phone number where a customer can call in, and an automated system will offer to remove their fax number from our contact list. Can I use asterisk to do this? |
| 15:57.27 | [TK]D-Fender | _fury: FAX over IP? Oh God |
| 15:57.37 | _fury | no, we're not doing fax over ip |
| 15:57.39 | brodiem | _fury: scroll up :) |
| 15:57.47 | [TK]D-Fender | _fury: Wait... yeah I got it |
| 15:57.59 | ManxPower | _fury: yes, you should be able to do that, no it is not build into Asterisk, yes you would have to write it. |
| 15:58.06 | [TK]D-Fender | _fury: Yeah, just take your pick from the most cost-efficient |
| 15:58.24 | _fury | ya I can write it no problem |
| 16:00.07 | Stereo | heey, it's working, I've now moved on to another problem :) |
| 16:00.29 | Stereo | when I get an incoming call, it says this: NOTICE[5921]: chan_sip.c:14033 handle_request_invite: Call from 'kipperchen' to extension 'kipperchen' rejected because extension not found. |
| 16:00.39 | Stereo | obviously it's not routing the call to the proper extension |
| 16:00.53 | bijit | is there a way when I change the ip address of the dhcp server on vlan.. aastra phones It autodetects it? |
| 16:00.56 | Stereo | I'm sure the answer is very obvious, but I haven't found it by rtfm'ing or googling |
| 16:01.46 | Stereo | in my extensions.conf I've got [bell_in] exten => s,1,DIAL(SIP/1234,20);dial extension 1234 exten => s,2,Hangup(), so the bell_in context should take care of redirecting to my 1234 phone |
| 16:01.50 | *** join/#asterisk chrisM8 (n=chatzill@78-105-1-131.zone3.bethere.co.uk) |
| 16:02.00 | [TK]D-Fender | Stereo: its looking for an ALPHA extension. so exten => kipperchen,1,NoOp(Yippy-kai-yay) |
| 16:02.09 | [TK]D-Fender | Stereo: (in the proper context) |
| 16:02.15 | Stereo | aha |
| 16:02.36 | chrisM8 | hi, is there anyone from digium support team here? |
| 16:02.43 | Stereo | so in the bell_in context I assume |
| 16:03.12 | Stereo | tries |
| 16:03.30 | Qwell | chrisM8: no, but what seems to be the problem? |
| 16:04.36 | Stereo | I'm actually not sure how asterisk determines it should be looking for kipperchen locally |
| 16:05.02 | chrisM8 | bought TE220 card recently - and it is crashing my server. Sent an email around 40 hours ago to support@digium.com with all the details, they assigned ticket ID... and haven't heard since than. |
| 16:05.15 | Stereo | kipperchen is the outgoing username, so it implicitly becomes the one it's looking for internally, I assume |
| 16:05.29 | Qwell | chrisM8: Please PM me the ticket # |
| 16:05.54 | Stereo | so how do I tell asterisk to connect the call to 1234 instead of kipperchen? |
| 16:05.57 | chrisM8 | Qwell: GZE-774863 |
| 16:06.05 | Qwell | chrisM8: I meant privately. :) |
| 16:06.09 | Qwell | but that'll do |
| 16:06.11 | chrisM8 | oops |
| 16:06.24 | ManxPower | that's sort of what "PM" means. |
| 16:06.25 | bad_duck_ | nobody ? (for a name of a bt usb adaptor that works with asterisknow and debian linux ?) |
| 16:06.41 | *** join/#asterisk HichamH (n=h@41.248.14.139) |
| 16:12.18 | *** join/#asterisk cerbianguard (n=mark@c-71-232-59-18.hsd1.ma.comcast.net) |
| 16:12.52 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 16:17.21 | Stereo | I don't get it |
| 16:17.26 | Stereo | it tells me: Call from 'kipperchen' to extension 'kipperchen' rejected because extension not found. |
| 16:17.34 | Stereo | yet I've got a kipperchen extension |
| 16:17.59 | cerbianguard | Guys: I installed a 2nd sangoma fxo module into the sangoma pci card. I changed the zaptel.conf file to read "fxoks = 1-2\nfxoks = 3-4" . I no longer get service on the poe phones connected to the server. I suppose this means something is wrong? How would I fix it? I know nothing about asterisk but have some experience with the linux terminal. |
| 16:18.03 | Qwell | Stereo: in the correct context? |
| 16:18.03 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 16:19.05 | mltlnx | Hello, I am using a Adit 600 channel bank to provide analog extensions. Is it possible to provide stutter tone when there is a voicemail? |
| 16:19.30 | Stereo | Qwell: perhaps not |
| 16:19.36 | Stereo | tries |
| 16:19.59 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 16:22.53 | [TK]D-Fender | cerbianguard: PoE? your card has NOTHING to do with that. |
| 16:23.19 | [TK]D-Fender | cerbianguard: is * running at all? |
| 16:23.20 | jeev | Fender, the crap quality audio was fixed because apparently i didn't install all the sounds but only gsm.. and that's why, it was doing some crap stuff with gsm |
| 16:23.21 | *** join/#asterisk bmg505 (n=leon@196-209-76-115-tbnb-esr-2.dynamic.isadsl.co.za) |
| 16:23.32 | jeev | everything is finally finished, i'm curious about g729.. |
| 16:23.39 | [TK]D-Fender | jeev: GCC 4.2 = trouble with GSM |
| 16:24.04 | jeev | a cross compiler having trouble with it? hmm.. ok probably the codec |
| 16:24.04 | *** join/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 16:24.11 | jeev | yea, i had to make menuselect, install the ulaw format. |
| 16:24.18 | coppice | what is broken with that GSM implementation? |
| 16:24.52 | jeev | for mine? because i was only using ulaw, so i guess it was trying to convert it and coming out horribly choppy. |
| 16:24.55 | mgroman | Any comments on res_cepstral ? |
| 16:24.56 | jeev | the IVR. |
| 16:25.00 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:25.06 | cerbianguard | [TK]D-Fender: iLink seems to be up and running |
| 16:25.21 | jeev | Fender, is G729 recommended? i mean is sound quality better than ulaw? or is it just necessary in low bandwidth requirements |
| 16:25.23 | AiT | I've got a test asterisk system (TrixBox w/FreePBX). For a novice, what you guys suggest, AsteriskNow or installing CentOS and Asterisk, etc independently? Remember, a novice with only a 5 concurrent call test system running! |
| 16:25.32 | [TK]D-Fender | cerbianguard: iLink? And that is? |
| 16:25.43 | [TK]D-Fender | jeev: Only for low BW |
| 16:25.59 | jeev | AiT, i started with asteriskNOW but within a few days i figured it was pointless and just did it independently. |
| 16:26.00 | cerbianguard | [TK]D-Fender: the web interface to asterisk |
| 16:26.08 | [TK]D-Fender | AiT: Any normal distro you can manage + Asterisk yourself |
| 16:26.18 | jeev | ok Fender, so nothing really positive about having it. i was considering it, maybe it's useful but there's enough bandwidth i guess. |
| 16:26.29 | jeev | ~bandcalc |
| 16:26.31 | qp | my boss insisted on giving trixbox a go, but I'm not convinced |
| 16:26.33 | [TK]D-Fender | cerbianguard: All web interfaces are bolt-on solutions and I've never even heard of that one. |
| 16:26.56 | mgroman | qp: arch linux > rhel, no? |
| 16:26.58 | [TK]D-Fender | berAnd your description give me no indication that things are running properly |
| 16:27.03 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 16:27.14 | Stereo | hurrah! |
| 16:27.52 | jeev | is redhat not linux architecture? |
| 16:28.08 | Stereo | thank you [TK]D-Fender and Qwell, we made it :D |
| 16:28.50 | [TK]D-Fender | Stereo: Congratulations |
| 16:29.05 | [TK]D-Fender | jeev: huh? |
| 16:29.13 | mgroman | jeev: it is, arch linux is a distro in of itself |
| 16:29.16 | *** part/#asterisk Stereo (n=stereo@78.141.155.243) |
| 16:29.23 | mgroman | www.archlinux.org |
| 16:29.39 | jeev | yea, i've never been fond of redhat, since those initial 5000 vulnerabilities per release. |
| 16:29.50 | jeev | nevermind Fender, well.. i'm glad everything is finally done, now i can clean up my dial plan |
| 16:29.57 | mgroman | i dont like it either...despite deploying asterisk on redhat |
| 16:30.02 | [netman] | hi, anybody could help me with the following issue? http://readlist.com/lists/lists.digium.com/asterisk-users/13/67133.html |
| 16:30.28 | mgroman | Does anyone here use res_cepstral ? How does it differ from app_swift? |
| 16:31.02 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 16:31.32 | jeev | Fender, so i couldn't get it running as a hosted pbx.. i dont think multi-wan matters in this situation, i can't figure out how a reply from asterisk on an outside network through a router where the phones are is supposed to have nat send the proper packets to 5060.. |
| 16:31.36 | n3hxs | [netman], I would first look at line noise on your circuit, or actually the Telco circuit. |
| 16:32.23 | *** join/#asterisk jazzmann (n=chatzill@cpc1-lutn9-0-0-cust163.lutn.cable.ntl.com) |
| 16:33.00 | n3hxs | Here in the states, when the T1 starts slipping or getting alarms, we first ask the telco to check their lines. They never admit a problem, but it usually magically goes away after they check their equipement. |
| 16:33.11 | *** part/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 16:33.55 | jazzmann | hi can anyone guide me how to install chan_mobile from http://www.chan-mobile.org/?page_id=4.I installed asteriek 1.4 with addons from mandriva software manager |
| 16:33.55 | fskrotzki | ~book |
| 16:33.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:33.57 | jazzmann | thanks |
| 16:34.26 | qp | is it OK to run asterisk on VMware for voip only implementation? I heard about "timing issues" ... |
| 16:35.21 | jeev | wow, i wouldn't run this behind vmware. |
| 16:35.44 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:35.53 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 16:37.17 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.146) |
| 16:37.37 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 16:38.33 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 16:39.03 | Qwell | qp: people have been known to do so |
| 16:39.11 | jeev | i can't believe i failed at natd does. |
| 16:39.17 | Qwell | it works reasonably well, but yeah, vmware is notorious for poor timing |
| 16:39.25 | jeev | Qwell, can you take a look at this and tell me if i'm just an idiot or it wouldn't work anyway ? |
| 16:39.37 | Qwell | jeev: You know my answer already. |
| 16:39.40 | jeev | haha |
| 16:39.44 | cerbianguard | [TK]D-Fender: Please pardon my ignorance. I've tried restarting the asterisk server with safe_asterisk and get "Asterisk dies with code 127" |
| 16:39.47 | mkillebrew | it can't be both? |
| 16:39.52 | jeev | quite possibly |
| 16:41.03 | jeev | Qwell: so of course the dual wan situation could be it.. but i shouldn't have had issues with the phones inside a nat environment with asterisk not at the datacenter right? i dont know what i was doing wrong.. maybe the fact that i was using pf for nat but whenever something would be inbound, pf/nat wouldn't know where to send it. |
| 16:41.21 | jeev | since you have every phone sending and receiving on 5060... would the only choice be to change local ports? |
| 16:41.22 | Qwell | ~sipnat |
| 16:41.23 | 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 |
| 16:41.32 | jeev | i've gone over them 500 times. |
| 16:41.42 | Qwell | 501 might be the key. |
| 16:42.10 | mkillebrew | when I tried at oing asterisk behind NAT I ould only get one way communication, I gve it up and got it its own cable modem |
| 16:42.42 | [TK]D-Fender | cerbianguard: I'd contact your reseller because you're running some rare GUI interface that is likely configuring everything for you |
| 16:42.45 | jeev | nowhere does the wiki show what if asterisk is a client and server. |
| 16:42.48 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 16:42.49 | [TK]D-Fender | cerbianguard: For which we can't do too much |
| 16:42.59 | blebleble | can someone recommend the most stable / reliable sip trunk provider? |
| 16:43.10 | jeev | mkillebrew, one way audio is an easy fix i guess |
| 16:43.23 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 16:43.48 | jazzmann | hey guys please help I need to install chan_mobile .I could not find any good article |
| 16:43.59 | jazzmann | is anyone there |
| 16:44.18 | jeev | i should take two phones, change local port to 5061 and 5062.. and see what'll happen. |
| 16:44.59 | jameswf-home | vonage |
| 16:45.01 | jameswf-home | ducks |
| 16:45.13 | jeev | what vonage |
| 16:46.03 | cerbianguard | [TK]D-Fender: I would if i could, but they're out of business. Am I just screwed? OOC, what does 127 suggest? |
| 16:46.39 | [TK]D-Fender | cerbianguard: You just made changes to Zaptel related hardware. so zaptel settings are suspect. |
| 16:47.17 | cerbianguard | [TK]D-Fender: Thanks for your help...I guess I'll just try wading on from here. Not that I have any other optios |
| 16:47.23 | cerbianguard | *options |
| 16:51.37 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 16:54.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:55.25 | [TK]D-Fender | cerbianguard: You might do well to hire a consultant to convert your setup to a standalaone install. |
| 16:56.49 | *** join/#asterisk DarylVoip (n=daryl@c-76-124-247-86.hsd1.nj.comcast.net) |
| 16:58.11 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 16:58.20 | DarylVoip | Hey all. I'm having a problem getting postgres compiled on Centos 5.2 and asterisk 1.4.15. Keeps telling me my postgres install is broken. Anyone have any pointers for figuring this one out? |
| 17:02.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:02.36 | ManxPower | DarylVoip: you should start out with 1.4.21.2 |
| 17:03.06 | DarylVoip | No, I shoudln't. I pass 900k minutes of calls a day, and 1.4.15 is the last non-broken-for-me version. |
| 17:03.30 | DarylVoip | This is just the first 64-bit install I'm trying. |
| 17:03.37 | ManxPower | ah |
| 17:03.49 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 17:04.00 | DarylVoip | <--- Not a noob. |
| 17:04.19 | ManxPower | Then you are a rare person around here. 8-| |
| 17:04.23 | DarylVoip | lol |
| 17:04.26 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 17:04.27 | mgroman | I just compiled asterisk 1.4.17 on RHEL, I was able to start Asterisk before, but I am toying with res_cepstral and now when I try and start Asterisk (asterisk -cvvvvvvvv), It stops and says "Floating point exception", is it possible to find out what this is talking about? |
| 17:05.12 | ManxPower | would postgress library have to be 64-bit too? |
| 17:05.43 | jeev | DarylVoip, what do you mean by broken? what isn't working |
| 17:05.54 | DarylVoip | I have both 32 and 64 bit libs installed....standard Centos5.2 64 bit install. |
| 17:06.01 | DarylVoip | ./configure --with-postgres give me: |
| 17:06.04 | Qwell | eww.. |
| 17:06.07 | DarylVoip | checking for PQexec in -lpq... no |
| 17:06.14 | Wayhigh | got his free vonage rt31p2 unlocked and registered |
| 17:06.15 | Wayhigh | woot |
| 17:06.19 | DarylVoip | configure: *** The PostgreSQL installation on this system appears to be broken. |
| 17:06.23 | Qwell | how exactly did you install postgresql-devel? |
| 17:06.29 | jeev | hmm Wayhigh, i think i got that. did you perm unlock it ? |
| 17:06.31 | Qwell | I mean the exact command |
| 17:06.54 | Wayhigh | jeev: I dunno if it's perm unlocked.. I did turn off the provisioning for it so probably |
| 17:06.55 | jeev | if you set postfix with a prefix, you should probably point --with-postgres to the proper dir's. |
| 17:07.05 | jeev | wayhig, that's the big linksys router thingy look alike, right ? |
| 17:07.12 | Wayhigh | yeah.. |
| 17:07.20 | jeev | that thing is gonna crap out on you soon and go lock itself again. |
| 17:07.32 | DarylVoip | My standard install_dep script. Which is: |
| 17:07.32 | DarylVoip | yum install gcc automake autoconf gcc-c++ openssl-devel libtermcap libtermcap-de |
| 17:07.32 | DarylVoip | vel newt newt-devel ncurses ncurses-devel postgresql postgresql-devel postgresql |
| 17:07.33 | DarylVoip | -libs rpm-build php-pear php-devel php php-pear-Date php-pear-DB php-pgsql |
| 17:07.37 | jeev | there is someone on dslreports who does it, the REAL way.. but no reason to pay him for it right now. he charges like 20-30 bux. |
| 17:07.39 | mags2 | DarylVoip: did you install libpq-dev |
| 17:07.49 | Qwell | DarylVoip: you did it wrong then :D |
| 17:07.59 | Qwell | You need to explicitly specify .x86_64 |
| 17:08.07 | DarylVoip | Oh...guess that only worked on the other 11 32-bit boxes I have then :) |
| 17:08.09 | Qwell | CentOS 5.2 is...well...I have no further comment. |
| 17:08.34 | Wayhigh | jeev: if it annoys me enough I'll probably see about unpackin the firmware and looking at it with my disassembler |
| 17:08.40 | DarylVoip | Centos 5.2 is....welll....passing over 100 million minutes of calls so far this year. |
| 17:09.03 | Qwell | It's very very broken. yum does not properly install x86_64 dependencies |
| 17:09.22 | Qwell | be explicit and specify that you want the .x86_64 version of those -devel packages |
| 17:09.26 | mags2 | ew. glad we use debian here. |
| 17:09.39 | ManxPower | hugs his Mandriva |
| 17:09.58 | DarylVoip | Yeah....they are all there. With no fuss. Every postgres-* and libpqxx-*. |
| 17:10.09 | denon | real men just get the kernel and build their * box around that! |
| 17:10.16 | Qwell | yum install postgresql-devel.x86_64 |
| 17:10.23 | denon | or. . yeah, use debian |
| 17:10.35 | mgroman | coughs Arch linux coughs |
| 17:10.40 | Qwell | if it doesn't pick up the deps properly, you'll need to uninstall all of those packages, and reinstall with .x86_64 |
| 17:10.41 | DarylVoip | And those of us making money just use what works. I'm not here for a distro war. |
| 17:10.55 | mags2 | yes I often recode asterisk in 64-bit assembly from memory |
| 17:10.57 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 17:11.02 | DarylVoip | Qwell - I see matching i386 and x86_64 packages for everything relevant. |
| 17:11.09 | Qwell | DarylVoip: no you don't. |
| 17:11.22 | jeev | Wayhigh, when you figure it out.. let me know. |
| 17:11.33 | Qwell | again, the dep matching is very broken. It's not something you would "just see" |
| 17:11.46 | ManxPower | DarylVoip: Qwell is also not a noob. |
| 17:11.57 | DarylVoip | So you are saying the the RPM db is reporting inaccurately? |
| 17:12.08 | Qwell | No, I'm saying that yum is doing the wrong thing. |
| 17:12.25 | DarylVoip | OK...well, the RPM DB is showing me that I have what I asked yum to install on the box. |
| 17:12.35 | jeev | weird, linux is built on open source but people still install packages when they could be compiled. welcome to windows! |
| 17:12.43 | ManxPower | Qwell: Do you have a specific suggestion? |
| 17:12.45 | Qwell | DarylVoip: okay, you're right, I'm wrong. Ignore me. |
| 17:12.49 | Qwell | ManxPower: I already gave it. |
| 17:12.51 | *** join/#asterisk max_______ (n=mochoa@chi-pat.cashnetusa.com) |
| 17:12.52 | Qwell | twice. |
| 17:12.53 | Wayhigh | jeev: will do.. |
| 17:12.54 | DarylVoip | I'm trying to figure out what part is broken with yum, and if it had any effect on what I'm doing, because, unless the RPM DB is wrong, it's not the issue. |
| 17:13.01 | mags2 | you might have better luck doing postgres thru odbc |
| 17:13.08 | ManxPower | Qwell: you can't make him listen. |
| 17:13.24 | DarylVoip | I'm listening.....but I'm not getting answers that make sense. |
| 17:13.32 | Qwell | The PROBLEM doesn't make sense. |
| 17:13.32 | jeev | guys, be happy.. at least you dont have to listen to my stupid questions anymore |
| 17:13.35 | Qwell | It's BROKEN. |
| 17:13.42 | Qwell | I'm telling you how to fix it. |
| 17:13.52 | ManxPower | DarylVoip: you can follow his suggestions or not, but don't argue about it. |
| 17:14.10 | Qwell | DarylVoip: Would you like to see an example? |
| 17:14.19 | DarylVoip | I got it: yum is sometimes doing the wrong thing by not installing x86_64 deps. But I can't see that happening on this system. |
| 17:14.22 | Qwell | yum install curl-devel - don't actually install it. Look at the list |
| 17:14.24 | DarylVoip | Yes...I would love to se an example. |
| 17:14.29 | heedly | IT's possible centos doesn't handle 64bit libs properly. |
| 17:14.30 | Qwell | now do yum install curl-devel.x86_64 |
| 17:14.34 | DarylVoip | I'm not telling you you're wrong. I'm saying I can't see it. |
| 17:14.50 | ManxPower | DarylVoip: you don't have to see it. Trust him. |
| 17:14.52 | Qwell | curl-devel.x86_64 *requires* libidn-devel.x86_64, but unless you are explicit, it does the wrong thing |
| 17:15.01 | Paige_ | ok so I typed in the complete php example agi in the BOOK and added and extentsion, and it just falls through without playing anything |
| 17:15.07 | Paige_ | any ideas? |
| 17:15.09 | ManxPower | Qwell: what does it do instead? |
| 17:15.17 | Qwell | ManxPower: It thinks .i386 satisfies the dep |
| 17:15.22 | mags2 | zang |
| 17:15.24 | Qwell | it...obviously doesn't |
| 17:15.28 | ManxPower | Qwell: *nod* |
| 17:15.30 | mags2 | wtf redhat |
| 17:15.55 | DarylVoip | I see that. OK...so you are talking about other deps, not just the x86_64 versions of the 386 libs. |
| 17:16.09 | Qwell | packages that what you are installing depend on |
| 17:16.14 | DarylVoip | That's what I was trying to figoure out based on what you were saying. |
| 17:16.16 | Qwell | so, the package you are installing is broken |
| 17:16.20 | DarylVoip | OK...go it. |
| 17:16.31 | jeev | even understood it the first time Qwell said it. |
| 17:16.47 | ManxPower | So this is the fix "(12:10:15 PM) Qwell: yum install postgresql-devel.x86_64"? |
| 17:16.48 | DarylVoip | Goot for you. Do you want a cookie or a my little pony sticker? |
| 17:16.54 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 17:16.57 | mags2 | why on earth would the repo maintainers let this slide |
| 17:16.57 | Qwell | ManxPower: unfortunately, no |
| 17:17.02 | Qwell | well...probably not. maybe |
| 17:17.04 | jeev | cookie preferred. |
| 17:17.09 | ManxPower | Qwell: I'll dig thru my scrollback more then. |
| 17:17.25 | Qwell | I had to remove everything that was installed when I installed those... |
| 17:17.30 | Qwell | and then specify .x86_64 ;/ |
| 17:18.23 | ManxPower | seems like a fast and simple thing to do. |
| 17:18.26 | BrokenNoze | Hi, don't mean to keep going on but has ANYONE had a situation where there are available members in a queue, but the calls just sit there and don't get distributed? |
| 17:18.29 | Qwell | I ended up reinstalling. |
| 17:18.50 | ManxPower | BrokenNoze: only the reasons JerJer said the other day. |
| 17:19.05 | *** part/#asterisk bad_duck_ (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 17:19.07 | mags2 | nuke site from orbit often the best choice |
| 17:19.18 | jeev | hey guys, so nobody had flowroute as an ITSP ? |
| 17:19.44 | BrokenNoze | ManxPower: yeah, no the members are def available damn it, i'm gonna have to write a whole queuing system in PHP from the looks of it! |
| 17:19.55 | Qwell | DarylVoip: If you're lucky, you can just `yum install those_packages.x86_64`, but.. |
| 17:20.07 | Qwell | mostly the -devel packages |
| 17:20.22 | JerJer | sticks his head out of the bat cave |
| 17:20.23 | DarylVoip | I'm gonna remove them first...then try that. |
| 17:20.23 | DarylVoip | If not, this i just a base VM for cloning. I can go back to a snapshot before I added the deps |
| 17:20.35 | Qwell | that would be best |
| 17:21.05 | Qwell | there's more silliness with the "multiarch" stuff.. if you remove the .i386 package and leave the .x86_64 package, some of the files that both provide may be removed |
| 17:21.21 | DarylVoip | No surprise. |
| 17:21.25 | ManxPower | BrokenNoze: searching the mailing list archives was not helpful? |
| 17:21.25 | mags2 | good lord |
| 17:21.56 | DarylVoip | I'm not saying I like CentOS....but it's that or pay for RHEL5, which I'm not gonna do. I don't have the time to play with IPMI and Dell OMSA crap to make it work under Debian. |
| 17:21.59 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-089-140.pools.arcor-ip.net) |
| 17:22.33 | Qwell | nothing wrong with CentOS.. just the fact that nobody will really acknowledge that as a bug |
| 17:23.07 | Qwell | it won't get fixed in CentOS until after it gets fixed in RHEL, and it's never going to be fixed in RHEL, because they don't rely on yum. |
| 17:23.19 | Wayhigh | anyone know if the EFF's Switzerland tool can identify isp's interfering with SIP? |
| 17:23.28 | DarylVoip | I hate RPM distros...but whatever. I use what makes my life easiest. I've got too many boxes to maintain to play games. |
| 17:23.51 | BrokenNoze | ManxPower: all I get is URL not found on the ones i'm interested in |
| 17:24.24 | ManxPower | ~mailinglist |
| 17:24.25 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 17:24.32 | max_______ | Anyone have an idea when 1.6.0 will be released? Like, maybe at Astricon? |
| 17:24.46 | ManxPower | max_______: when it's ready? |
| 17:25.04 | mgroman | fck rhel and fck centos |
| 17:25.07 | *** part/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 17:25.19 | Juggie | Qwell, what bug is that? |
| 17:25.20 | jeev | ManxPower, where do you go to speak of ITSP's? |
| 17:25.37 | Qwell | Juggie: look at the difference between `yum install curl-devel` and `yum install curl-devel.x86_64` |
| 17:25.45 | Qwell | the former is badly broken |
| 17:26.35 | Qwell | Juggie: only on centos 5.2 |
| 17:26.47 | *** join/#asterisk rabby (n=rabby@p4FCE9BFB.dip0.t-ipconnect.de) |
| 17:26.49 | mags2 | jeev: dslreports mebbe? lots of noise but some signal. never used flowroute meself. |
| 17:26.56 | Juggie | Qwell, i dont have a system to check atm |
| 17:26.58 | Juggie | whats broken about it |
| 17:27.10 | Qwell | .i386 deps satisfy .x86_64 libs |
| 17:27.35 | Qwell | yum tries to install (only) libidn-devel.i386 but curl-devel.x86_64 |
| 17:27.45 | Qwell | so things won't compile |
| 17:27.57 | Juggie | would that be a yum bug or a rpm bug |
| 17:28.02 | Qwell | yum |
| 17:28.33 | Juggie | heh |
| 17:28.46 | Juggie | nice, my repo sync script for centos is nukeing 5.1 atm |
| 17:29.17 | Qwell | Juggie: it's weird though, because if you specifically say you want a .x86_64 package, it'll satisfy all the deps correctly |
| 17:29.41 | jeev | hmm ok. thanks mags2. |
| 17:30.49 | Juggie | on a x86_64 system, should doing yum install curl-devel install .i386 or x86_64 by default? |
| 17:30.58 | Qwell | Juggie: it installs both |
| 17:31.05 | Qwell | at least of curl-devel |
| 17:31.16 | Qwell | it *should* do both all the way down, but it doesn't |
| 17:31.22 | Juggie | hmmmmm i'm not sure thats usually the case |
| 17:32.12 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-175-210.w86-214.abo.wanadoo.fr) |
| 17:32.32 | mags2 | jeev: it's problematic because most anywhere people trying to discuss service gets clogged up with gits trying to sell their service e.g. asterisk-biz and voip-info |
| 17:33.36 | Juggie | my x86_64 systems certainly do not have both of everything |
| 17:34.15 | mags2 | wtf fwd is going to start charging? dammit I donated to them last year and they never sent me the coffee mug |
| 17:35.26 | ManxPower | mags2: First hit is free, dude. |
| 17:36.21 | *** join/#asterisk jameswf (n=james@jfinstrom.broker.freenet6.net) |
| 17:36.23 | mags2 | ManxPower: 12 years of free hits, why I didnt mind kicking them some bucks last year. but I want my damn coffee mug lolol |
| 17:36.29 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:39.05 | Paige_ | can anyone help me debug the example weather php agi script that is in the BOOK? |
| 17:39.27 | mltlnx | Is it possible that an analog extension connected to channel bank would not have mwi (stutter tone) when there is a voicemail due to the channel bank? |
| 17:39.51 | [TK]D-Fender | mltlnx: Could be the CB, your zap channel setup, etc |
| 17:40.26 | ManxPower | mltlnx: it is possible maybe, but I highly doubt it. The issue some CBs have is with ON-HOOK MWI. |
| 17:40.57 | ManxPower | what is the mailbox= line ? |
| 17:41.06 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 17:41.09 | hi365 | a client asked me the following question: can I set up recording that can be played to the caller when a member recives a call froma queue? |
| 17:41.10 | hi365 | (i.e. Thank you for calling XYZ, this is ABC speaking, how may I help you?) |
| 17:41.13 | jeev | ahh mags2, i haven't seen anyone selling their services here.. |
| 17:41.20 | mltlnx | mailbox=27805 |
| 17:41.31 | mltlnx | in the default context |
| 17:41.38 | ManxPower | put the context in |
| 17:41.43 | mltlnx | i tried that too |
| 17:41.49 | ManxPower | then leave it in |
| 17:41.57 | mltlnx | its an adit 600....anyone familiar? |
| 17:42.11 | mags2 | jeev good point ^_^ |
| 17:42.18 | jaytee | While Linux and Apache have managed to carve out a significant niche in the web server market and Asterisk has made some strides in the VOIP area for linux servers I believe Linux will never be more than a minor player and eventually a footnote in the history of computing due to the idiocy and distro specific zealotry that abounds in the linux community......oh and another thing...all you KDE users are infidels and must die! |
| 17:42.31 | jeev | hey jaytee. |
| 17:42.42 | mags2 | yay I love kde |
| 17:42.45 | Qwell | niche? |
| 17:42.48 | jaytee | :-) |
| 17:42.51 | Qwell | right, because > 50% is a niche.. |
| 17:43.03 | _fury | where do I set the default admin username/password for asterisk-gui? |
| 17:43.16 | jaytee | that's just for web Qwell, in which it obviously dominates |
| 17:43.24 | russellb | _fury: in /etc/asterisk/manager.conf |
| 17:43.27 | mvanbaak | infideeeeeeeeeeeeeeeeeeeel |
| 17:43.30 | jeev | hey russellb! |
| 17:43.36 | russellb | jeev: hey there |
| 17:43.39 | jeev | how you doin |
| 17:43.42 | mvanbaak | gotta Love achmed |
| 17:43.42 | russellb | doing well |
| 17:43.48 | russellb | writing teh codez |
| 17:43.51 | jeev | russellb, you can't message me for help anymore, i'm through giving it to you. |
| 17:43.55 | russellb | lol |
| 17:44.01 | russellb | damnit! :( |
| 17:44.03 | jaytee | "I keeeeeeeeeel you!!!" |
| 17:44.07 | _fury | russellb: thanks, what's the username though? there are a couple of things in there, closest thing I see to a password is the "secret" setting but no username setting |
| 17:44.18 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 17:44.19 | mags2 | username == context name |
| 17:44.20 | mvanbaak | :) |
| 17:44.26 | russellb | _fury: the secret is the pw. the thing in brackets in the username |
| 17:44.37 | _fury | ah heh |
| 17:44.38 | _fury | thanks |
| 17:44.40 | russellb | np |
| 17:44.52 | ManxPower | feels slightly ill after seeing that. |
| 17:45.08 | russellb | ManxPower: what, helping someone using a GUI? :) |
| 17:45.10 | mvanbaak | food, brb |
| 17:45.20 | ManxPower | russellb: no need to get vulgar 8-) |
| 17:45.22 | russellb | likes the GUI crowd, too |
| 17:45.23 | mvanbaak | vim is a wonderfull gui |
| 17:45.28 | russellb | mvanbaak: it sure is |
| 17:45.31 | russellb | it's the best IDE there is |
| 17:45.34 | mags2 | shouldnt the gui let you get access to change manager.conf oh wait lololol |
| 17:45.35 | mvanbaak | indeed |
| 17:45.48 | mvanbaak | and a good IRC client |
| 17:46.06 | mvanbaak | hhmm, wife is bringing dinner, brb |
| 17:46.30 | jeev | russellb, so eventually.. everything went well and my massive complaints about DTMF were noticed by my provider, i'll show you a pastebin of his config.. that makes dtmf work incoming and outgoing. |
| 17:46.54 | russellb | that's ok. |
| 17:47.28 | jeev | http://rus.pastebin.com/d2bc17cbd |
| 17:47.52 | russellb | pretends to look |
| 17:48.04 | jeev | heh |
| 17:48.45 | jeev | oh well, it's working! |
| 17:48.53 | jeev | so screw it, if it's a bug or not! |
| 17:50.09 | *** join/#asterisk blq (n=Bl@dslb-088-067-044-063.pools.arcor-ip.net) |
| 17:56.13 | *** join/#asterisk DarylVoip_ (n=daryl@c-68-36-66-30.hsd1.nj.comcast.net) |
| 17:57.11 | jaytee | everyone loves a GUI.............until their mouse breaks |
| 17:58.23 | *** join/#asterisk jpastore (n=jpastore@75.144.122.49) |
| 17:58.32 | *** part/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 17:58.38 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 18:00.11 | jeev | heh, i used asteriskNOW for about 3 days.. the beginning chapter. |
| 18:00.35 | jeev | i figured, i can't bother fender and stuff with asteriskNOW, so i solely changed to asterisk because it'd be bothering you guys! |
| 18:01.21 | jaytee | jeev, I want you to start wearing white socks and a bowtie from now on or it will bother me if you don't. I expect full compliance. |
| 18:01.33 | jeev | bowtie, lol |
| 18:01.40 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 18:01.46 | hsv-al | hello |
| 18:02.28 | jeev | just looked at his pflog and noticed bootp is being blocked. |
| 18:02.44 | hsv-al | u know that warm saliva feeling that wells up in the mouth |
| 18:03.02 | hsv-al | before one dry heaves? or pukes? well i constantly have that now, because i had "wasabi peanuts" as a snack earlier, that is the most disgusting item ive had in my life |
| 18:03.08 | jazzmann | can anyone help me install chan_mobile |
| 18:03.10 | DarylVoip_ | OK...thanks to all who helped. Centos5x86_64 stupidities have been worked around. |
| 18:03.18 | mags2 | wasabi peanuts aaa+++++ would eat again |
| 18:03.31 | hsv-al | mags, i never get weird over food |
| 18:03.40 | hsv-al | but this tops it all, disgusting.. |
| 18:03.42 | Qwell | DarylVoip_: report a bug - maybe they'll listen if they get inundated :p |
| 18:03.43 | _fury | well, I'm no power user. I want to set this up for one simple thing, and I don't want to have to read an entire book to do it. The GUI helps me do that very easily. nothing wrong with that. I'm more than capable of going through config files and doing everything manually, but there's a much smaller learning curve using the GUI. If there's anything I can't do with it, then I'll be sure to look it up and edit things manually. :) |
| 18:03.50 | mags2 | hsv-al: does cilantro taste like metal to you |
| 18:04.05 | DarylVoip_ | Somehow I doubt it. But I was planning on it anyway. |
| 18:04.21 | jazzmann | I need to connect my mobile to asrteisk |
| 18:04.29 | hsv-al | i actually have anxiety now because it disgusted me so much, |
| 18:04.32 | DarylVoip_ | Not like I can deploy this right now anyway, as VMWare broke esxi, and I'm not setting my time back a day on all of my production blades. |
| 18:04.42 | Qwell | DarylVoip_: yeah... |
| 18:04.50 | DarylVoip_ | It's been one f those days. So I'm just gonna bag it and head to the marina. |
| 18:04.51 | Qwell | that sucks so bad for them |
| 18:05.02 | DarylVoip_ | Thanks again. |
| 18:06.39 | *** join/#asterisk mgroman (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 18:08.37 | SwK | anyone know where to get a good npa-nxx to time zone database (and no telcodata and local calling guide arent it) |
| 18:09.19 | Qwell | SwK: well, with number port... |
| 18:10.14 | Qwell | or just cellphones in general, for that matter |
| 18:10.37 | Qwell | somebody with 256-658 could be in China for all you know |
| 18:10.50 | SwK | I know I just need home region tz |
| 18:10.52 | ManxPower | SwK: chances are you'll have to take the lat/long and compare it to a map database |
| 18:10.53 | SwK | for the exchange |
| 18:11.08 | SwK | ManxPower, yeah I know someones got to have already done it tho |
| 18:11.16 | SwK | its not like telemarketers dont use that data all the time |
| 18:12.25 | ManxPower | and they always share |
| 18:14.06 | mags2 | SwK: http://www.fraudlabs.com/areacodeworld.aspx ? not free |
| 18:14.43 | SwK | trying to find something that isnt 600+ USD |
| 18:14.44 | SwK | hah |
| 18:14.50 | mags2 | I hear that |
| 18:14.51 | SwK | telcodata has me spoiled |
| 18:15.16 | mags2 | sign up for some spammer support boards and make some friends |
| 18:15.35 | mags2 | prolly some torrents out there too |
| 18:19.21 | Kobaz | mmm |
| 18:19.27 | Kobaz | why does firefox love to crash so much |
| 18:21.28 | rob0 | Did you install the firefox random crash extension? |
| 18:21.44 | Kobaz | perhaps |
| 18:22.17 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:22.47 | jblack | [TK]D-Fender: Ping |
| 18:22.58 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:27.21 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 18:31.43 | *** join/#asterisk kd8ikt (n=kd8ikt@we.trekbeyond.org) |
| 18:31.50 | Kobaz | okay, firefox stopped crashing |
| 18:32.08 | Kobaz | i managed to close off some tabs before it crashed again... i'm down to 72 tabs from 112 |
| 18:32.20 | seanbright | firefox stopped crashing? must be a bug |
| 18:32.26 | seanbright | it's not supposed to stop crashing |
| 18:32.59 | [TK]D-Fender | jblack: Yes? |
| 18:33.08 | AlexTO | Someone knows if there is a command for CLI where it can show me the active calls with the connected time? |
| 18:34.03 | [TK]D-Fender | AlexTO: "core show channels concise" |
| 18:35.25 | AlexTO | Thanks, the numbers like SIP/redtelecom-4891 |
| 18:35.47 | AlexTO | Sorry, the number like 206:18:5 are in seconds? |
| 18:36.17 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:43.00 | jblack | [TK]D-Fender: Google is having troubles today. You ready to take over as backup? |
| 18:43.25 | [TK]D-Fender | AlexTO: Buy a watch, yo tell me. |
| 18:43.31 | [TK]D-Fender | jblack: ? |
| 18:43.56 | jblack | First gmail, then google finance. Next will be searching. The internet needs a proxy. =) |
| 18:44.00 | rob0 | is selling watches ... genuine Rolex, only ten bux US american |
| 18:44.00 | jblack | a caching proxy |
| 18:44.40 | *** join/#asterisk `paul (n=aldee@125.252.68.126) |
| 18:45.29 | [TK]D-Fender | jblack: I'll gladly proxy your finance lookups |
| 18:45.41 | jblack | Awesome. |
| 18:45.45 | [TK]D-Fender | jblack: just forward me your CC details :) |
| 18:46.02 | jblack | Oh, sure. I can trust you. |
| 18:46.07 | `paul | is roundrobin deprecated? |
| 18:46.38 | jblack | 5159-4233-4422-0432, exp 02/12 |
| 18:46.54 | jameswf-home | i think you depricated on my shoes |
| 18:46.58 | jblack | Listed name "Yeah Right" |
| 18:47.03 | Qwell | deprecated* |
| 18:47.19 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 18:47.38 | jameswf-home | windows regular? |
| 18:47.42 | jameswf-home | eek |
| 18:47.54 | Qwell | yeah, you might want to turn off that mask :p |
| 18:48.20 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 18:48.42 | jameswf-home | should make a mask asterisk/user/newb |
| 18:48.44 | Dovid | hi |
| 18:48.44 | rob0 | It's obnautixious! |
| 18:48.54 | Dovid | does anyone know of a Soft phone that has a VPN client built in ? |
| 18:49.10 | rob0 | KitchenSink.exe |
| 18:49.53 | *** join/#asterisk hi365_m (n=hi365@213.151.63.155) |
| 18:51.21 | [TK]D-Fender | SwissArmyVoIP.vbs |
| 18:51.31 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 18:51.51 | blebleble | any sip trunk providers offer unlimited channels? |
| 18:53.05 | ManxPower | blebleble: many of them do, but never on a flat rate plan. |
| 18:53.17 | jeev | awaits his 15mbit dsl at the office. |
| 18:53.21 | jeev | must be adsl2.. |
| 18:53.35 | blebleble | ManxPower: could you recommend any (for a call center) |
| 18:54.02 | kamanashisroy | Hi, A hazy question .. is there any reason to get "bad request"(400) from sip did provider if we move from 1.2 to 1.4 ?? |
| 18:54.10 | ManxPower | blebleble: vitelity and teliax are well known. |
| 18:54.14 | ManxPower | ~isplist-us |
| 18:55.15 | [TK]D-Fender | kamanashisroy: perhaps you should see what its a response TO. |
| 18:56.33 | twisted | is getting ready to release app_kamasutra.so |
| 19:00.42 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:02.39 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.49 | kamanashisroy | [TK]D-Fender: it is responding to REGISTER request .. my asterisk is registering to did provider .. it is connecting to proxy .. But I cannot see a way to set the realm for this registration :( |
| 19:06.37 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 19:06.44 | *** part/#asterisk tuxx- (n=tuxx@nightshade.nl) |
| 19:12.29 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:16.27 | jeev | cha cha cha |
| 19:18.09 | kamanashisroy | jeev: what that means ?? |
| 19:18.52 | [TK]D-Fender | Dance Monkey, Dance! |
| 19:19.01 | jeev | damn straight, i'm dancing |
| 19:19.04 | jeev | so happy |
| 19:19.20 | ManxPower | kamanashisroy: sip.conf.sample did not show any examples? |
| 19:20.50 | [TK]D-Fender | ManxPower: Realm is a global only thing unfortunately. |
| 19:20.50 | kamanashisroy | ManxPower: how could that show any problem ??? It does not have any register thing .. I just got an warning like the "insecure" is deprecated .. My brain is totally frozen (because I am in fever :( ) |
| 19:21.42 | [TK]D-Fender | ManxPower: "register" needs to stop being a single line directive and needs to be an entire section like peer/friend/user |
| 19:22.00 | ManxPower | It needs to do many things. |
| 19:22.23 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 19:22.35 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 19:23.44 | kamanashisroy | feels feverish .. it is nice feelings .. aha aha .. jeev does not know the actual sound .. |
| 19:31.44 | jeev | heh |
| 19:31.50 | jeev | is annoyed again, first DTMF key is ignored. |
| 19:31.53 | jeev | tone or whatever |
| 19:31.55 | kamanashisroy | jeev: you do not know at all |
| 19:32.04 | jeev | i dont know what you're talkin baout. |
| 19:32.08 | kamanashisroy | jeev: which proto ?? |
| 19:32.19 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 19:32.23 | *** join/#asterisk delrocas (n=IceChat7@168-103-31-91.spkn.qwest.net) |
| 19:32.28 | ManxPower | jeev: do an Answer + Wait(1) to let the RTP settle down |
| 19:32.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:32.50 | kamanashisroy | jeev: sure at lest 1 second |
| 19:33.05 | ManxPower | Chances are Wait(.25) will be just fine, but try 1 second first |
| 19:33.35 | jeev | ManxPower, on an outgoing call |
| 19:36.13 | ManxPower | jeev: It sucks to be you. |
| 19:36.59 | jeev | it had to be youuuuuuuuuuuu |
| 19:37.11 | jeev | frankly ManxPower, i dont give a damn, i'm cnotent with this |
| 19:40.21 | *** join/#asterisk km- (n=pgrace@fmservices.v6.fierymoon.com) |
| 19:40.45 | km- | manx: hey, you said you have AS5400's right? |
| 19:42.50 | *** join/#asterisk outtolunc (n=me@adsl-75-62-237-1.dsl.pltn13.sbcglobal.net) |
| 19:43.43 | max_______ | whee... just backported dundi dynamic weighting from 1.6.0b9 to 1.4.21.2 |
| 19:43.45 | *** part/#asterisk SparFux (n=raoul@e182024251.adsl.alicedsl.de) |
| 19:44.05 | russellb | max_______: heh, I think I did that a long time ago :-p |
| 19:44.13 | Qwell | russellb: mmhmm |
| 19:44.15 | russellb | http://svncommunity.digium.com/svn/russell/asterisk-1.4 |
| 19:44.46 | russellb | dundi_dynamic_weights.path.txt |
| 19:44.47 | russellb | :) |
| 19:44.52 | russellb | patch* |
| 19:45.00 | russellb | anyway, glad you find it useful, max_______ |
| 19:45.17 | max_______ | neat. Have to add SHELL func though. Yeah, it's really going to be handy. |
| 19:45.55 | Qwell | max_______: also a backport of that |
| 19:46.45 | lmadsen | I think I just copied the code and compiled |
| 19:47.01 | jeev | LEIF! |
| 19:47.02 | hi365_m | is there any way to playback a file to the CALLER after a queue member answers the call? |
| 19:48.05 | Paige_ | how would I manipulate CDR's using an AGI script once a call is completed? |
| 19:49.29 | ManxPower | km-: no, I never said I had an AS5400 |
| 19:50.20 | Paige_ | hello? |
| 19:50.47 | jeev | i dunno what AGI is... |
| 19:50.48 | jeev | ~agi |
| 19:50.49 | jbot | rumour has it, agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 19:50.50 | jeev | i haven't read that far |
| 19:50.53 | jeev | oh lol |
| 19:50.57 | jeev | hides |
| 19:51.14 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 19:51.57 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:52.01 | kamanashisroy | good night |
| 19:52.25 | jeev | bye |
| 19:55.40 | exothermc | Can you use ex-girlfriend logic in asterisk realtime? |
| 19:56.06 | seanbright | my ex-girlfriend didn't use logic... ever |
| 19:56.08 | jeev | heh |
| 19:56.23 | jeev | my ex girlfriend wanted a gucci bag for her bday 1 month after she was my girl. |
| 19:58.59 | `paul | how do i set the timeout before the call jumps to the next extension in a queue? |
| 19:59.13 | baliktad | can someone explain the difference between the timeout=xx line in queues.conf and the value I specify in the Queue() timeout parameter |
| 19:59.25 | baliktad | *blinx* |
| 19:59.26 | jeev | hmm? Timeout(1) ? |
| 19:59.39 | baliktad | how odd that you would ask a queue timeout question at the same instant as me |
| 19:59.55 | ManxPower | baliktad: happens all the time here. |
| 19:59.59 | jeev | sorry, Wait(1) maybe ? |
| 20:00.02 | [TK]D-Fender | baliktad: Queue() = total time the caller will sit around for. in queues.conf its the time and agent's phone willl ring |
| 20:00.04 | jeev | or replace.. |
| 20:00.04 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 20:00.12 | *** part/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 20:00.42 | baliktad | I have this line: exten => _X.,n,Queue(intphones,r,,,10) |
| 20:00.59 | baliktad | but callers will sit in the queue for 60 seconds before the next line in the dialplan executes |
| 20:01.00 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:01.09 | baliktad | (if no one is answering their phones that is) |
| 20:01.18 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 20:01.33 | mgroman | jeev: fight(1) |
| 20:01.34 | exothermc | [TK]D-Fender: You done much with realtime? |
| 20:01.55 | jeev | heh |
| 20:02.19 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:02.39 | anonymouz666 | is there any way to execute (through MANAGER) a extension/context WITHOUT calling the Channel:? |
| 20:02.45 | exothermc | [TK]D-Fender: I'm just wondering the best way to handle setting options on an extension basis for outbound calls from peers. Inbound is easy you just pass a macro some vars. outbound isn't so nice. The best hack I can come up with is do a db lookup to populate some vars, but I don't like how messy it is. |
| 20:02.49 | rwaite | why does asterisk have so many race conditions? |
| 20:03.19 | [TK]D-Fender | baliktad: Queue checks BETWEEN agent calls. |
| 20:03.32 | mgroman | Dovid: lol |
| 20:03.50 | *** join/#asterisk cyphase (n=cyphase@c-76-103-214-250.hsd1.ca.comcast.net) |
| 20:03.59 | [TK]D-Fender | baliktad: So if you want "X" total seconds, make both the same |
| 20:04.23 | exothermc | [TK]D-Fender: of course I can just make two new contexts for each option but that doesn't scale so nice. |
| 20:04.50 | [TK]D-Fender | exothermc: Nothing stopping you from using macros for outbound. I do. |
| 20:04.53 | baliktad | right now I have no timeout specified in my queues.conf |
| 20:05.24 | [TK]D-Fender | anonymouz666: No, but nothing you can't deal with easily. |
| 20:05.31 | [TK]D-Fender | baliktad: then go set one. |
| 20:05.47 | exothermc | [TK]D-Fender: Ya but how do you get your arguments? No real place to put them in sipbuddies. |
| 20:06.27 | `paul | so how do i set the timeout before a call jumps to the next entry in queue? |
| 20:06.42 | [TK]D-Fender | exothermc: This is DIALPLAN. And you can set variables in your peers just fine |
| 20:06.44 | `paul | lets say jump to the next extension in queue after 5 secs |
| 20:06.51 | Paige_ | [TK]D-Fender, have you messed around with custom ringbacks? |
| 20:07.06 | [TK]D-Fender | `paul: Go read the sample config |
| 20:07.20 | [TK]D-Fender | Paige_: ... huh? |
| 20:07.40 | jeev | Paige, fender is the master os disaster, he's messed with everything.. even your sister. |
| 20:07.47 | jeev | master of disaster |
| 20:07.49 | Paige_ | if someone dials a number in my extensions, instead of ringing, they get a custom mp3 |
| 20:08.19 | mgroman | Paige_: what is a custom mp3? |
| 20:08.47 | Paige_ | mgroman, a mp3 that a customer would upload to the server |
| 20:09.05 | mgroman | lol, playback()? |
| 20:09.05 | Paige_ | sort of like how at&t does with cellphones |
| 20:09.54 | Qwell | ringback tones |
| 20:09.59 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 20:10.04 | exothermc | [TK]D-Fender: I see what you're saying I just don't see a nice clean place to pull the data from the DB on an outbound call. I guess if I do multiple DB lookups per outbound call, but I was hoping to avoid that. |
| 20:10.15 | [TK]D-Fender | Paige_: "core show application dial" |
| 20:10.23 | russellb | mgroman: I get your nick confused with "mogorman" every time I see it. That is someone that works here (@digium) |
| 20:10.40 | mgroman | ah no kidding |
| 20:10.48 | exothermc | [TK]D-Fender: Well I guess since the peer is cached, I'm not going to get away without doing that lookup. |
| 20:10.58 | [TK]D-Fender | exothermc: You don't need to pull from the DB. When the device places a call the channel vars are set |
| 20:11.17 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 20:13.00 | baliktad | ok I added timeout=10 to my queue which makes sure my timeout functions correctly. thanks TK |
| 20:13.37 | exothermc | [TK]D-Fender: What I'm saying is that I want to pull from the DB. can the peer make international, dial 411 dial 911 etc etc. all independent of each other. Yes I can do that with contexts on each peer, but each option grows it by n! |
| 20:14.07 | Paige_ | [TK]D-Fender, that is not that much useful, do you know of any examples? |
| 20:14.10 | [TK]D-Fender | exothermc: What part of SETTING CHANNEL VARIABLES is not clear to you? |
| 20:14.44 | [TK]D-Fender | Paige_: Yes it is. There is an option to set the MoH to use instead of ringing. |
| 20:14.49 | exothermc | [TK]D-Fender: I can specify chan vars in the peer definition? |
| 20:15.07 | [TK]D-Fender | exothermc: unload chan_brokenrecord.so |
| 20:15.13 | ManxPower | exothermc: yes! see sip.conf.sample |
| 20:15.27 | exothermc | ahh ok thanks |
| 20:15.56 | Paige_ | [TK]D-Fender, i cannot find it in there |
| 20:16.03 | ManxPower | exothermc: who knows if you can do it using Realtime. I don't. |
| 20:16.16 | anonymouz666 | [TK]D-Fender: I know that I can do that through another way... I just wanted to be sure since I have a whole CLASS that uses the manager. call files can easily do that |
| 20:16.16 | [TK]D-Fender | Paige_: keep reading the instruction page over and over till your eyes bleed |
| 20:16.40 | mgroman | ouch |
| 20:16.57 | Paige_ | m([class]) ? |
| 20:17.18 | ManxPower | see, it was there all long. |
| 20:17.52 | [TK]D-Fender | Paige_: You're WONDERING if that's it? |
| 20:18.02 | Paige_ | yes i am |
| 20:18.20 | `paul | how do i implement roundrobin using rrmemory? |
| 20:18.22 | [TK]D-Fender | Paige_: thi is the "big print". Please don't miss it. |
| 20:18.33 | Paige_ | and i am wondering how i would implement it in extensions |
| 20:18.34 | [TK]D-Fender | `paul: strategy=rrmemory |
| 20:18.52 | [TK]D-Fender | Paige_: ...huh? |
| 20:18.55 | ManxPower | Paige_: you would implement it in Dial |
| 20:20.11 | `paul | Fender: i mean reset it so the evrytime it enters the queue the firts entry would ring... |
| 20:20.33 | Paige_ | my dial command is: SIP/12066047199@PSTN,,rt |
| 20:20.55 | [TK]D-Fender | Paige_: add the option on like all the rest |
| 20:20.55 | ManxPower | Paige_: we are not going to spoon feed you. "core show application dial". |
| 20:20.56 | *** join/#asterisk Katty (n=asterisk@hera.copi-rite.com) |
| 20:21.00 | Katty | herroes. |
| 20:21.04 | [TK]D-Fender | ManxPower: Already linked it to him. |
| 20:21.08 | [TK]D-Fender | Katty: Mew. |
| 20:21.09 | Katty | jbot: cnam |
| 20:21.11 | ManxPower | [TK]D-Fender: time to give up? |
| 20:21.16 | Katty | [TK]D-Fender: herro. |
| 20:21.19 | [TK]D-Fender | ManxPower: Probably |
| 20:21.26 | Katty | [TK]D-Fender: cnam? |
| 20:21.32 | ManxPower | [TK]D-Fender: I agree. |
| 20:21.53 | Katty | [TK]D-Fender: ref: outgoing callerid |
| 20:22.02 | Paige_ | i am not looking to be |
| 20:22.02 | Paige_ | i am not looking to be "spoonfed" |
| 20:22.10 | Paige_ | I am simply trying to find it |
| 20:22.16 | ManxPower | Katty: telco overrides any CID Name you set. |
| 20:22.23 | [TK]D-Fender | Katty: <trebek> I'm sorry you gorgot to write that in the form of a question. |
| 20:22.31 | Katty | [TK]D-Fender: the telco is telling me that because our callerid is being sent as 'cnam' that it is being rejected by other carriers. |
| 20:22.33 | ManxPower | If the call stays on the same telco it might be seen on the far end. |
| 20:22.49 | [TK]D-Fender | Paige_: r & t are options. you've already added them... |
| 20:23.03 | Paige_ | ok |
| 20:23.09 | exothermc | [TK]D-Fender: specifying multiple key value pairs separated by ';' is being deprecated for SET() so is that the same for setvar in sip.conf? |
| 20:23.16 | Katty | ManxPower: hrmm. so if they overwrite my callerid name, then why would it matter? or would that somehow effect how we call people? |
| 20:23.23 | `paul | how do i reset rrmemory so it would act like roundrobin? |
| 20:23.28 | Katty | thinks the telco is just nuts. |
| 20:23.34 | ManxPower | Katty: I said that before you explained further. |
| 20:23.47 | ManxPower | I suspect your outgoing callerid number is not being accepted, not the name. |
| 20:23.57 | Katty | oh! |
| 20:24.18 | Katty | that parses. |
| 20:24.21 | [TK]D-Fender | `paul: The old roundrobin no longer exists IIRC |
| 20:24.24 | ManxPower | make sure callerid is 10 digits only, no leading one or 0, no dots, dashes or commas or spaces or anything not number. |
| 20:25.08 | ManxPower | For a while our telco allowed us to send 4-digit callerid number. The people my users were calling were like "WTF?" when they looked at the callerid. |
| 20:25.20 | ManxPower | we sent 4-digits by mistake a couple of times. |
| 20:25.36 | Katty | ManxPower: hrmm. sounds suspicious |
| 20:25.53 | ManxPower | now all outgoing calls go thru a Caller*ID fixer upper script, that massages the CLID based on some rules and variables. |
| 20:25.57 | `paul | D-fender: so how do i implement it using rrmemory? i mean evry time a call enters queue the first entry would ring |
| 20:25.59 | exothermc | ManxPower: most backbones will take whatever you pass them. |
| 20:26.17 | ManxPower | exothermc: not in my experience |
| 20:26.28 | `paul | the problem with adding weights is the first entry should be engaged or sumthing for it to jump to the next |
| 20:26.40 | exothermc | ManxPower: Qwest, VZ, ATT and Level3 all do, which ones are you thinking of? |
| 20:27.08 | ManxPower | BellSouth, AT&T, XFone |
| 20:27.11 | Katty | ManxPower: http://pastebin.ca/1168594 <- i'm out putting 10 digits. |
| 20:27.22 | *** join/#asterisk hi365_m (n=hi365@213.151.63.155) |
| 20:27.23 | Katty | ManxPower: do you think it could be my callerid name, not matching the telco, that's causing the issue? |
| 20:27.27 | ManxPower | Plus seeing hundreds of people here have calls fail because of not valid clid info |
| 20:27.31 | Katty | ManxPower: on outgoing calls we overwrite the callerid number, but not the name |
| 20:27.34 | exothermc | isn't bellsouth ATT, and I know for a fact that ATT does on their high volume term circuits |
| 20:28.08 | ManxPower | Katty: I've never seen that happen before so I'd say "no", but it is possible, I guess, maybe. |
| 20:28.33 | Katty | ManxPower: can't hurt to test it i guess |
| 20:28.36 | ManxPower | Ah, also Expedius |
| 20:29.10 | [TK]D-Fender | `paul: You DON'T. that stategy was REMOVED. |
| 20:29.25 | [TK]D-Fender | Checkout time, heading home. BBL |
| 20:29.26 | ManxPower | Katty: Those are not failed calls. I don't see an issue in the pastebin. |
| 20:29.28 | exothermc | ManxPower: ATT AVOICS platform will let you pass anything you wish. |
| 20:29.43 | Katty | ManxPower: the first one says "your call cannot be completed at this time, please try the call again later" |
| 20:29.52 | Katty | ManxPower: the second call is instantly hung up |
| 20:29.52 | exothermc | wonders if he just violated some NDA |
| 20:29.59 | ManxPower | exothermc: Oh, I imagine most carriers would allow you to send whatever you want, if you request it. |
| 20:30.28 | Katty | ManxPower: the first one being a non asterisk audio file. |
| 20:30.41 | exothermc | ManxPower: Nah most don't care on the wholesale level, they just figure you better know what you are doing. |
| 20:30.43 | ManxPower | Katty: pastebin your /etc/zaptel.con and /etc/asterisk/zapata.conf and I'll continue to help you on #asterisk-cli, OK? |
| 20:30.56 | Katty | ManxPower: let's do it later. |
| 20:30.57 | ManxPower | exothermc: I was not talking about whilesale. |
| 20:31.04 | ManxPower | or wholesale either 8-) |
| 20:31.11 | exothermc | ManxPower: ahh ;) |
| 20:31.14 | ManxPower | Katty: sure thing |
| 20:32.03 | ManxPower | wholesale is a totally different thing. |
| 20:33.24 | *** part/#asterisk cyphase (n=cyphase@c-76-103-214-250.hsd1.ca.comcast.net) |
| 20:34.10 | exothermc | ManxPower: Ya well I'm not really aware of a way to connect to any of those providers other than wholesale. I guess you can get a PRI from them, but I've never used a PRI ;) |
| 20:34.24 | ManxPower | Katty: you have exra quotes in your CLID |
| 20:36.01 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 20:36.38 | `paul | can i set a timeout the the call will leave the queue? |
| 20:38.23 | seanbright | `paul: 'core show application Queue' |
| 20:38.41 | Katty | ManxPower: oh? |
| 20:38.47 | Katty | looks |
| 20:39.06 | ManxPower | Katty: If the telco IS upset by the name, the extra quotes could make them upset. |
| 20:39.20 | Katty | ManxPower: hmm. the name or the number cid? |
| 20:39.24 | Paige_ | manxpower does this look correct: SIP/12065551212@PSTN,90,m |
| 20:39.30 | ManxPower | Asterisk happily sends the name, but the terminating telco normally overwrites it |
| 20:39.42 | ManxPower | Paige_: yes |
| 20:39.57 | Katty | can't be the number. |
| 20:40.08 | Katty | i don't see any quotes around them in extensions.conf. i'll check sip.conf for name |
| 20:40.10 | ManxPower | Katty: I would look at the number last |
| 20:40.13 | Paige_ | so that will play music on hold instead of ringing? |
| 20:40.21 | ManxPower | after we try everything else. |
| 20:40.35 | seanbright | Paige_: 'core show application Dial' |
| 20:40.41 | ManxPower | Paige_: if you have /etc/asterisk/musiconhold.conf set up correctly, yes. |
| 20:40.54 | Katty | ManxPower: callerid="Office 203"<203> <- no quotes? |
| 20:41.05 | ManxPower | Katty: I see two quotes there. |
| 20:41.08 | Qwell | Katty: callerid=Office 203 <203> |
| 20:41.12 | Katty | oooooh |
| 20:41.12 | Katty | k |
| 20:41.23 | Qwell | though, I think chan_zap or whatever might remove the quotes |
| 20:41.29 | Qwell | don't count on it though |
| 20:41.32 | *** join/#asterisk h3x0r (n=hex@wsip-24-120-86-83.lv.lv.cox.net) |
| 20:41.33 | Paige_ | it is set up correctly, i can dial 500 and play the music on hold |
| 20:41.46 | exothermc | ManxPower: What is the purpose of having Katty pass the name to the telco? |
| 20:41.47 | Katty | well ijust changed mine and i'll do a lil callaroonie |
| 20:41.55 | Katty | exothermc: none. |
| 20:41.58 | Katty | exothermc: i don't care about it. |
| 20:42.11 | Katty | exothermc: but various numbers spaz out when i call them. and the telco says IT"S YOU OMGROFL |
| 20:42.13 | exothermc | Katty: well just pass the number and be done. |
| 20:42.18 | ManxPower | exothermc: fixing the quotes is easiar then figuring out how to tell asterisk to not send it. |
| 20:42.23 | seanbright | i do a Set(CALLERID(name)=) right before my outbound dial statements |
| 20:42.27 | exothermc | ManxPower: ahh ok |
| 20:42.36 | outtolunc | usecallerid=no <G> |
| 20:42.41 | Paige_ | seanbright, i have already done that |
| 20:42.54 | ManxPower | outtolunc: that would stop sending the number too |
| 20:42.59 | outtolunc | nods <G> |
| 20:43.04 | exothermc | Katty: sounds like you need a different telco |
| 20:43.04 | outtolunc | i was more of a joke |
| 20:43.28 | outtolunc | you can play around with usecallingpres also |
| 20:43.53 | seanbright | *or* you could just set CALLERID(name) to an empty string |
| 20:44.02 | seanbright | done and done |
| 20:44.07 | Katty | ManxPower: i removed quotes from sip.conf on mine, and did reload chan_sip. the cli still shows the quotes tho. |
| 20:44.27 | ManxPower | exothermc: With Asterisk, if you accept it's oddities you will live a happy life, if you fight it's oddities -- well, Asterisk ALWAYS wins those fights. |
| 20:44.30 | Katty | exothermc: this telco is a joke. |
| 20:44.39 | Katty | seanbright: i like that idea. |
| 20:44.44 | Katty | seanbright: i'm gonna DO EET |
| 20:44.48 | ManxPower | Katty: repastebin the new output |
| 20:44.51 | seanbright | you're a smart woman |
| 20:45.21 | rob0 | Why did the telco cross the road? |
| 20:45.22 | Paige_ | ManxPower, ok, it dials out, cli say that it started music on hold class 'default', but i still get ringing instead of music |
| 20:45.36 | rob0 | Which came first, the telco or the goose egg? |
| 20:45.42 | ManxPower | Paige_: do you have a correct moh config file? |
| 20:45.43 | seanbright | rob0: jesus |
| 20:46.00 | Katty | ManxPower: unomomento |
| 20:46.15 | Katty | blast it! |
| 20:46.16 | Paige_ | ManxPower, yes, it is correct. i have a test extension that plays it fine |
| 20:46.17 | Katty | grrr!!! |
| 20:46.20 | rob0 | WWJD? JW make jokes about the telco! |
| 20:46.30 | seanbright | precisely |
| 20:46.33 | seanbright | good talk. |
| 20:47.23 | jaytee | WWJD? What would Jesus Drive? Well, in one part of the New Testament Jesus speaks to his followers about keeping his own accord and them keeping to theirs so obviously Jesus drove a Honda. |
| 20:47.55 | Paige_ | omg, that is bad jaytee |
| 20:48.17 | jaytee | bad how? as in lame or as in blasphemous? |
| 20:48.22 | Paige_ | lame |
| 20:48.30 | jaytee | ok, I'll go with that |
| 20:48.33 | seanbright | drives a honda! just like jesus. |
| 20:48.37 | rob0 | They'd save more by riding together. Jesus saves, right? All the disciples and Him in no more than 3 Accords. 2 if they tried hard. |
| 20:48.56 | jaytee | Jesus saves, but Esposito scores for goal!!! |
| 20:49.22 | ManxPower | Jesus saves, but Buddha saves at Wal-Mart! |
| 20:49.23 | seanbright | jesus saves, moses invests. |
| 20:49.39 | *** join/#asterisk Arck-FR (n=Arck-FR@cvl92-2-82-228-145-232.fbx.proxad.net) |
| 20:49.55 | jaytee | they tried to crucify Buddha too but he was so fat they couldn't get him on a cross so they told him, "Just sit there!" |
| 20:50.08 | Katty | ManxPower: http://pastebin.ca/1168629 <- that's what it's doing now |
| 20:50.14 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:50.28 | Katty | [TK]D-Fender: mooooooo. |
| 20:50.33 | Katty | [TK]D-Fender: i'm starving. |
| 20:50.40 | Katty | [TK]D-Fender: would you like some celery and peanut butter? |
| 20:50.44 | [TK]D-Fender | Katty: I've got "moo" in the oven :) |
| 20:50.59 | Katty | sweet! i'll be over in an hour |
| 20:51.14 | infinity1 | screen -rd |
| 20:51.17 | jaytee | Jesus saved a bunch of money on his car insurance after switching to Geico. |
| 20:51.39 | jaytee | moo? |
| 20:52.13 | [TK]D-Fender | jaytee: COW <- |
| 20:52.15 | *** join/#asterisk dennisharrison (n=dennisha@71-15-46-99.dhcp.thbd.la.charter.com) |
| 20:52.18 | jaytee | who cooks moo in an oven. You need to charbroil that steak on a real grille |
| 20:52.30 | seanbright | use propane |
| 20:52.33 | dennisharrison | broiling a steak is a good way to go |
| 20:52.34 | seanbright | taste the meat, not the heat |
| 20:52.41 | seanbright | -- hank hill |
| 20:52.45 | jaytee | ugh, propane, you sound like Hank |
| 20:52.45 | [TK]D-Fender | jaytee: Blue & seared under the broiler on a cast-iron grill plate |
| 20:52.48 | dennisharrison | lol |
| 20:52.59 | Katty | [TK]D-Fender: i hate you. |
| 20:53.01 | Katty | [TK]D-Fender: feed me :< |
| 20:53.04 | Paige_ | crockpot moo |
| 20:53.28 | dennisharrison | howdy everyone by the way :) |
| 20:53.29 | Paige_ | cooks while at work |
| 20:53.37 | jaytee | howdy, dennisharrison |
| 20:53.54 | [TK]D-Fender | crockpot? EW. Slow cook = slow death. I like mine "fresh".... preferrably with the bell still attached :p |
| 20:54.09 | jaytee | I still love a heavily marbled roast beef cooked rare. |
| 20:54.12 | ManxPower | [TK]D-Fender: carnivore |
| 20:54.37 | ManxPower | bloodthirsty carnivore, even |
| 20:54.39 | dennisharrison | trying to setup an ivr menu that on (t) will forward the caller to <some number on an external providor> |
| 20:54.42 | seanbright | he just has a fever |
| 20:54.45 | seanbright | and the only cure |
| 20:54.45 | [TK]D-Fender | ManxPower: Omnivore... except the BS that flies around here is often hard to swallow |
| 20:54.47 | seanbright | is more cowbell |
| 20:54.49 | eXistenZ | hey [TK]D-Fender |
| 20:54.55 | jaytee | "I ate his liver with some fava beans and a nice chianti" |
| 20:55.10 | dennisharrison | and I can't see a way to easily do that |
| 20:55.12 | Paige_ | [TK]D-Fender, when at home, the only way to moo is to get charcoal good and hot and sear the bitch on both sides and serve |
| 20:55.12 | [TK]D-Fender | "I'm having a friend for lunch" |
| 20:55.17 | seanbright | FFFFffffffFFFffffFFffff |
| 20:55.20 | Katty | with bbq sauce? |
| 20:55.24 | Katty | wow, that's really white seanbright! |
| 20:55.34 | seanbright | hmm? |
| 20:55.38 | Katty | nevermind. |
| 20:55.47 | Paige_ | over his head katy |
| 20:55.51 | jaytee | I loved the part in Hannibal where Ray Liotta goes, "Mmmm, that smells good! what is that?" and it's part of his brain frying in a skillet. |
| 20:55.54 | Paige_ | katty sorry |
| 20:55.55 | seanbright | that actually rhymed though |
| 20:55.58 | Katty | Paige_: hehe |
| 20:56.08 | Katty | seanbright: FFFFFF = white |
| 20:56.13 | seanbright | ohhhh |
| 20:56.18 | Katty | seanbright: FF0000 = red |
| 20:56.18 | Katty | etc |
| 20:56.26 | dennisharrison | <PROTECTED> |
| 20:56.28 | [TK]D-Fender | reduces the aperture to F32 @ 1/4000 shutter and looks at seanbright .... briefly.... |
| 20:56.30 | seanbright | well, not exactly, but i get it |
| 20:56.52 | seanbright | FFFFFF = hexadecimal representation of a number |
| 20:56.53 | Katty | it was just a funny. |
| 20:56.59 | jaytee | hope you're using ASA 1600 at these light levels |
| 20:57.06 | seanbright | that in a browser is used as a color |
| 20:57.12 | seanbright | and yes, i am that guy. |
| 20:57.15 | mgroman | funny? |
| 20:57.17 | Katty | seanbright: indeed, representing RGB numbers. |
| 20:57.27 | Katty | seanbright: just in hex. |
| 20:57.43 | seanbright | Katty: FFFFFF does not *have* to be RGB numbers |
| 20:57.46 | *** join/#asterisk Great_Anta_Baka (i=c636caf6@gateway/web/ajax/mibbit.com/x-0423d59cdb15dd8d) |
| 20:57.50 | seanbright | Katty: but you're right, it was funny. |
| 20:57.54 | Katty | red 00, green 00, blue 00 (= |
| 20:58.12 | Katty | acutally |
| 20:58.14 | Katty | that'd be black |
| 20:58.18 | Katty | but yeah |
| 20:58.18 | [TK]D-Fender | Katty: Still living in a 24bit world> Silly! |
| 20:58.30 | Katty | [TK]D-Fender: you hush, or your getting celery |
| 20:58.37 | seanbright | you're. |
| 20:58.38 | mgroman | wtf |
| 20:58.40 | seanbright | stops now |
| 20:58.45 | Katty | seanbright: /bonk |
| 20:58.49 | [TK]D-Fender | Katty: Warning to vegetarians : YOU ARE WHAT YOU EAT |
| 20:59.01 | jeev | hahaha |
| 20:59.07 | Katty | [TK]D-Fender: how dare you call me cheap fast and easy! |
| 20:59.10 | Katty | [TK]D-Fender: i'll have your head! |
| 20:59.12 | seanbright | damn hippies |
| 20:59.29 | tzanger | [TK]D-Fender: I always preferred "vegetarians eat vegetables. Beware of humanitarians" |
| 20:59.34 | Katty | tzanger: haha |
| 20:59.52 | seanbright | "if fire fights right fire, and crime fighters fight crime... what do freedom fighters fight?" |
| 20:59.53 | [TK]D-Fender | tzanger: Comedy gold... I'm sure to recycle that shortly! |
| 20:59.53 | Great_Anta_Baka | hi all.. dont mean to interrupt but if i have a recording in an ivr thats 10 secs how do i make asterisk wait 10 seconds more AND listen for keypresses? |
| 21:00.00 | seanbright | YIKES |
| 21:00.09 | seanbright | i couldn't have butchered that any more |
| 21:00.16 | Katty | Great_Anta_Baka: WaitExten? |
| 21:00.24 | [TK]D-Fender | seanbright: Congress subpoenas ;) |
| 21:00.34 | Great_Anta_Baka | so WaitExten(10) ? |
| 21:00.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:00.41 | bkruse | [TK]D-Fender: Hey, how are you doing? |
| 21:01.02 | Katty | Great_Anta_Baka: exten => s,n,WaitExten(5) or 10 whatever |
| 21:01.09 | Great_Anta_Baka | nice |
| 21:01.11 | Great_Anta_Baka | thanks katty |
| 21:01.18 | tzanger | seanbright: that's like "vegetable oil is made of vegetables. What's baby oil made of?" |
| 21:01.22 | [TK]D-Fender | bkruse: Still breathing... |
| 21:01.31 | seanbright | tzanger: babies! |
| 21:01.33 | Katty | hugs [TK]D-Fender |
| 21:01.33 | bkruse | [TK]D-Fender: That is a good thing :) |
| 21:01.39 | Great_Anta_Baka | vegetable? |
| 21:01.43 | Katty | [TK]D-Fender: more martial arts! |
| 21:01.48 | Katty | [TK]D-Fender: keep yourself busy! |
| 21:01.53 | jaytee | what the heck is a Great_Anta_Baka? |
| 21:01.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:01.59 | Great_Anta_Baka | lol |
| 21:02.01 | Katty | jaytee: apparently a nice person. |
| 21:02.04 | Katty | jaytee: mind your manners! |
| 21:02.11 | Great_Anta_Baka | inspired by GTO |
| 21:02.18 | rwaite | ~centos52bug |
| 21:02.19 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 21:02.21 | seanbright | jaytee: it's an old wooden ship. |
| 21:02.32 | jaytee | oh, sorry. It's someone's nick! |
| 21:02.33 | Great_Anta_Baka | lol |
| 21:03.01 | Katty | jbot: Great_Anta_Baka |
| 21:03.03 | Katty | jbot: useless! |
| 21:03.04 | jbot | ACTION starts crying and hides from katty in the darkest corner of the room. :( |
| 21:03.14 | Katty | jbot: hug. |
| 21:03.14 | jbot | ACTION hugs katty |
| 21:03.19 | [TK]D-Fender | Katty: Working on that. Getting into photography as well now |
| 21:03.32 | Katty | [TK]D-Fender: oh? plants? |
| 21:03.36 | Katty | [TK]D-Fender: scenery? |
| 21:03.38 | [TK]D-Fender | Katty: All sorts |
| 21:03.39 | Katty | [TK]D-Fender: the neighbors? |
| 21:03.41 | seanbright | ~Great_Anta_Baka |
| 21:03.47 | Katty | [TK]D-Fender: ah. |
| 21:04.03 | Great_Anta_Baka | oi leave my name alone :/ |
| 21:04.12 | Katty | Great_Anta_Baka: i still love you. |
| 21:04.16 | seanbright | ~Great_Anta_Baka |
| 21:04.16 | jbot | extra, extra, read all about it, great_anta_baka is an old wooden ship.a |
| 21:04.21 | seanbright | sweeet |
| 21:04.21 | [TK]D-Fender | jaytee: Translates roughly as "total idiot" |
| 21:04.26 | seanbright | goes home now |
| 21:04.29 | Katty | seanbright: bai. |
| 21:04.55 | [TK]D-Fender | makes a toast to "Truth in Advertising" |
| 21:05.15 | Katty | i never understood a toast with wine. |
| 21:05.21 | Katty | if you're going to toast, use bread ;) |
| 21:07.24 | Great_Anta_Baka | well a more accurate translation would be you're a total idiot :P |
| 21:07.36 | jaytee | it's a cool sounding nick. it sounds kinda like some old Babylonian diety, one of the evil, wicked, mean and nasty ones. |
| 21:07.51 | jeev | rawr. |
| 21:08.11 | jaytee | anyways, it's quitting time. be back later |
| 21:08.11 | Great_Anta_Baka | hahaha |
| 21:08.27 | Great_Anta_Baka | cherz |
| 21:09.20 | Katty | cherz, what you sit on in the ghetto |
| 21:09.28 | Great_Anta_Baka | mmm anyone with suggestions on some neutral music on hold i can use |
| 21:09.37 | Great_Anta_Baka | na we dont have cherz here |
| 21:09.37 | Katty | i like pink floyd |
| 21:09.45 | Great_Anta_Baka | i like tom waits |
| 21:09.55 | russellb | well, a few files come with asterisk ... |
| 21:09.57 | Great_Anta_Baka | and senor floyd |
| 21:09.58 | Katty | you could do some george straight |
| 21:10.11 | Great_Anta_Baka | no idea |
| 21:10.13 | Great_Anta_Baka | who that is |
| 21:10.14 | russellb | pretty typical hold music |
| 21:10.18 | Qwell | slayer would make good hold music |
| 21:10.24 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 21:10.34 | Great_Anta_Baka | or slipnot |
| 21:10.38 | Katty | i'd hangup. |
| 21:10.41 | Great_Anta_Baka | thats should get those grannies moving |
| 21:10.42 | jayrod422 | does any know if asterisk can parse pri info of a digium card to see if a call origs from a payphone? |
| 21:10.43 | russellb | Katty: exactly |
| 21:10.44 | [TK]D-Fender | Qwell: Anthrax... great elevator music! |
| 21:10.48 | Katty | watches digium incorporate it immediately |
| 21:11.16 | russellb | the point of hold is to hope the person will hang up, isn't it? |
| 21:11.26 | Great_Anta_Baka | hahaha |
| 21:11.26 | Qwell | Do you get ANI II from PRI? |
| 21:11.27 | errr | I think more people would hangup if you used something like ICP |
| 21:11.52 | jayrod422 | i know i can get the ani |
| 21:11.54 | errr | the dating game song would go over well :) |
| 21:11.58 | Qwell | ANI II is different |
| 21:12.02 | jayrod422 | but i dont know about ani II |
| 21:12.05 | Qwell | I want to say it's on SS7 |
| 21:12.18 | Qwell | I don't know.. that's what you need though, is ANI II |
| 21:12.25 | jayrod422 | so what kind of circuit do you order? an inbound pri or a ss7 cir |
| 21:12.55 | file | Qwell: yes you can get it from PRI |
| 21:12.57 | *** part/#asterisk CrashHD` (n=crashhd@65.74.161.225) |
| 21:13.00 | Qwell | well then! |
| 21:14.29 | `paul | how do i set the filename of the voicelogs to include the Agent who answered the call in the queue..? |
| 21:15.23 | Great_Anta_Baka | if i use WaitExten(15) and a user enters the value in the 5th second does that mean i will still have to wait for the next 10 secs before it proceeds? or will DigitTimeout take care of this? |
| 21:16.09 | [TK]D-Fender | Filet Mignon came out PERFECT. yummmmmm |
| 21:16.17 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 21:16.31 | GhOnDiE | hi |
| 21:16.41 | GhOnDiE | ive just had the most odd thing happen |
| 21:17.10 | WhiteWolf | hm? |
| 21:17.25 | GhOnDiE | well i had somebody was in a confrence |
| 21:17.32 | GhOnDiE | he said that everything went dead |
| 21:17.44 | GhOnDiE | and now after doing a reboot of teh server nothing is working |
| 21:17.53 | GhOnDiE | i have no voice prompts or anything |
| 21:18.03 | Great_Anta_Baka | disk space? |
| 21:18.05 | GhOnDiE | i can dial and it shows up on the console |
| 21:18.14 | GhOnDiE | just no response |
| 21:18.50 | Great_Anta_Baka | just asking cos i had the same thing happen to me |
| 21:18.54 | Great_Anta_Baka | hdd was full :/ |
| 21:18.59 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 21:19.02 | GhOnDiE | hmm no dont think so |
| 21:19.10 | GhOnDiE | will delet a few things though |
| 21:19.20 | Great_Anta_Baka | just do a df -h |
| 21:19.59 | GhOnDiE | less than 30% |
| 21:20.33 | Great_Anta_Baka | meh.. i am useless from here on out lol.. thats about my only experience with stuff going just dead |
| 21:20.49 | GhOnDiE | its very odd |
| 21:20.50 | Great_Anta_Baka | can you make pstn calls? |
| 21:20.55 | GhOnDiE | nothing to explain it even |
| 21:21.19 | Great_Anta_Baka | sip calls? |
| 21:21.28 | Great_Anta_Baka | voice mail? |
| 21:21.33 | GhOnDiE | everything |
| 21:21.37 | GhOnDiE | none of it works |
| 21:21.42 | GhOnDiE | its via sip |
| 21:21.48 | GhOnDiE | even my iax inbound dont work |
| 21:22.11 | Great_Anta_Baka | stop asterisk and do an asterisk -cv |
| 21:22.27 | Great_Anta_Baka | see if anything goes funky there.. i dunno man i am useles |
| 21:23.11 | Great_Anta_Baka | sigh.. think i am just going to smoke and sleep |
| 21:23.20 | GhOnDiE | lol sounds good |
| 21:26.58 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:32.55 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 21:33.50 | Great_Anta_Baka | sweet this looks like its goinf to knock me out.. hehe.. night guys |
| 21:33.51 | Great_Anta_Baka | check you all tomorrow |
| 21:34.17 | GhOnDiE | yeah cheers night |
| 21:40.05 | eric_hill | Does the asterisk appliance allow SSH access to the console, or is it web-only? |
| 21:40.14 | Qwell | eric_hill: you can ssh |
| 21:40.36 | eric_hill | tx |
| 21:40.56 | exothermc | is there a good linux sip client out there? |
| 21:42.54 | exothermc | oh nm xlite for linux |
| 21:44.45 | eric_hill | @Qwell, do you know if the AA can be configured for use in Mexico (i.e. Spanish VM and Telmex ringing)? |
| 21:45.49 | Qwell | eric_hill: don't know |
| 21:46.24 | *** join/#asterisk edon (n=edon@albalug/edon) |
| 21:50.22 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
| 21:52.24 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 21:56.21 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 22:00.57 | *** join/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 22:03.13 | gaetronik | hi there |
| 22:04.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:06.46 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 22:09.53 | *** join/#asterisk DarnoQ (n=d@chello089076117062.chello.pl) |
| 22:10.00 | DarnoQ | hi! anyone home ? |
| 22:10.38 | Strom_C | no |
| 22:11.15 | mkillebrew | sucker, now he knows you're here |
| 22:11.19 | mkillebrew | crap. |
| 22:12.44 | DarnoQ | lol ;) |
| 22:13.48 | DarnoQ | i need a magic idea how to make mixmonitor work properly on current relase |
| 22:14.11 | Strom_C | magic, eh? |
| 22:14.13 | Strom_C | *poof* |
| 22:14.16 | Strom_C | nope, didn't work |
| 22:14.19 | Strom_C | sorry |
| 22:15.23 | mkillebrew | what's mixmonitor and why doesn't it work |
| 22:15.58 | fogo | DarnoQ: how isn't it working? |
| 22:16.20 | DarnoQ | huh i've already tried secret cli command known only to best mastas "core abrakadabra pls make mixmonitor work" but it also failed |
| 22:17.03 | fogo | DarnoQ: what is it doing/not doing? |
| 22:17.05 | DarnoQ | i am recording g729 call to one file but channels won't record properly, one starts before the other |
| 22:17.36 | fogo | DarnoQ: what version of * |
| 22:18.36 | DarnoQ | 1.4.18.1 |
| 22:19.17 | fogo | I'd look at http://bugs.digium.com/view.php?id=12296 |
| 22:19.23 | DarnoQ | thx |
| 22:19.49 | fogo | I don't recall having an issue with it on 1.4.18; although I could be wrong |
| 22:21.35 | *** join/#asterisk DarienWork (n=dan@209.17.173.13) |
| 22:23.08 | DarienWork | so I've been looking around trying to figure this out, but so far haven't come up with much... we're looking to upgrade our call centre from the crappy setup we have now to a proper one, with nice reporting and so on, but I can't find any open-source packages to handle this type of thing for us |
| 22:23.28 | DarienWork | can anyone suggest any open-source call centre management systems based on Asterisk that they have experience with or have heard good things about? |
| 22:23.59 | *** join/#asterisk VaNNi (n=VaNNi___@38.104.76.90) |
| 22:24.16 | fogo | DarienWork: like queue stats and such? |
| 22:24.26 | DarienWork | yeah, queue stats, time on call, dropped calls, short calls, etc. |
| 22:24.55 | fogo | DarienWork: We're using QueueMetrics - not open source, but works well |
| 22:25.11 | DarienWork | ooh |
| 22:25.49 | DarienWork | fogo: how do you set up queues and so on? does it do that, or is that done separately and this is just for queue stats? |
| 22:25.57 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 22:26.22 | fogo | DarienWork: We also looked at .. AMA?.. although it didn't work for us (http://www.intuitivecreations.com/contributions/AMS/ama.php) although I don't recall why |
| 22:26.45 | jasonwoot | ama looks nice |
| 22:27.00 | fogo | DarienWork: it's just queue stats - queue setup is all done in * |
| 22:27.07 | DarienWork | hmm |
| 22:27.18 | DarienWork | because I'd like to have a simple way to set up queues for our call centre manager |
| 22:27.32 | DarienWork | if we could set up call flow and/or handle triggering of external apps, that would be super keen |
| 22:27.47 | DarienWork | maybe we could add that in ourselves |
| 22:27.57 | jasonwoot | fogo PM |
| 22:28.16 | DarienWork | what would be great is getting our call centre people off of windows |
| 22:28.35 | DarienWork | ooh, AMS *does* look neat |
| 22:29.34 | jasonwoot | I exchanged a lot of emails with AMA guy, but it didn't seem practical for large volume |
| 22:29.36 | fogo | DarienWork: yeah, it does have a nice look, and is open source, but it wouldn't work for us - although I don't recall why ... I think it may not work with * 1.4?.. not sure |
| 22:30.02 | jasonwoot | has anyone created different button map classes for Polycom 501? |
| 22:31.50 | gaetronik | we use home made script to load queue_log and then pentaho |
| 22:32.05 | gaetronik | but it's not optimal. |
| 22:32.36 | gaetronik | maybe I'm OT but we plan to go to astricon |
| 22:32.55 | gaetronik | and we are looking for a cheaper hotel than the glendale spa whatever |
| 22:33.36 | gaetronik | is anyone have a cheaper solution? |
| 22:34.35 | russellb | yay astricon! |
| 22:34.48 | russellb | there are supposedly a lot of hotels around there ... |
| 22:35.30 | gaetronik | the job will be to find them |
| 22:36.01 | jeev | russellb, are you going to astricon! |
| 22:36.08 | jeev | ? |
| 22:36.09 | russellb | yes, i am |
| 22:36.14 | russellb | speaking a few times, actually .. |
| 22:36.25 | jeev | ok can i throw stuff at you if i come |
| 22:36.44 | gaetronik | jeev, for deadlocks issues? |
| 22:36.45 | russellb | sure, as long as stuff == money |
| 22:36.47 | russellb | paper money. |
| 22:37.14 | jeev | russellb, wow. is digium paying for my flight and room? i like first class.. and ritz. |
| 22:37.16 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 22:37.22 | jeev | i'm high maintenance. |
| 22:37.54 | *** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 22:37.58 | ZX81 | what's the deal with busy-level - does it exist? I've grep'd the configs/* and don't see it and yet http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level says it exists from 1.4.18 |
| 22:38.14 | ZX81 | suppose I could grep the src if noone knows |
| 22:38.26 | gaetronik | ZX81, use he source luke |
| 22:38.31 | ZX81 | wickliffe:~# grep busy-level /usr/src/asterisk/channels/* |
| 22:38.32 | ZX81 | wickliffe:~# |
| 22:38.37 | ZX81 | nope - not there |
| 22:38.37 | hoegaatit | I'm looking for a dedicated host to run asterisk on--needs to have enough punch to handle 300 callers--any recommendations? |
| 22:38.38 | ZX81 | :) |
| 22:38.41 | Qwell | ZX81: trunk/1.6 |
| 22:38.45 | ZX81 | ah ok |
| 22:38.50 | ZX81 | will update wiki |
| 22:38.56 | Qwell | BUT |
| 22:38.59 | Qwell | the option is busylevel |
| 22:40.16 | ZX81 | ok :) |
| 22:41.54 | DarienWork | fogo: do you recall how much QueueMetrics cost? |
| 22:42.02 | DarienWork | oh, here it is |
| 22:42.22 | DarienWork | mmm, kind of pricey |
| 22:42.58 | fogo | DarienWork: well, for us it was pony up the cash or deal with something that didn't work as well :) |
| 22:43.04 | gaetronik | hoegaatit, in the same time or 300 users |
| 22:43.05 | DarienWork | yeah, I suppose |
| 22:43.31 | DarienWork | but dropping â¬600 isn't what I'd prefer |
| 22:43.38 | DarienWork | then again, it's cheaper than the service we were going to go with |
| 22:44.26 | DarienWork | it also annoys me that AMS uses BDB |
| 22:44.29 | DarienWork | but oh well |
| 22:44.50 | hoegaatit | gaetronik: at the same time |
| 22:46.03 | jeev | russel! |
| 22:46.05 | gaetronik | hoegaatit, sorry i might not help you since i don't hav the experience but a reasonably powerfull server may do the job |
| 22:46.08 | fogo | DarienWork: I just searched back through my emails - AMA doesn't work with anything past 1.2 (yet) |
| 22:46.26 | fogo | DarienWork: although that may have changed in the last 6 months... |
| 22:46.37 | hoegaatit | gaetronik: no worries--I'm mostly looking for a cheap VPS provider |
| 22:46.56 | gaetronik | you will not host yourself the server |
| 22:46.59 | hoegaatit | I can figure out exactly how much cpu/ram I need after that |
| 22:47.03 | hoegaatit | no |
| 22:50.18 | gaetronik | hoegaatit, the must important is the bandwith |
| 22:50.38 | gaetronik | and the stability of it |
| 22:50.41 | russellb | jeev: yes? |
| 22:51.31 | eric_hill | Is it possible to have a SIP phone fallback to directly talking with a VoIP gateway if the SIP registrar falls off the network for some reason? |
| 22:52.00 | jeev | is astricon gonna have any girls? |
| 22:52.14 | jasonwoot | I thought you were all girls? |
| 22:52.19 | jeev | like playmates |
| 22:52.20 | jasonwoot | you guys are guys? |
| 22:52.24 | jasonwoot | I feel skeevy.. |
| 22:52.43 | jeev | yea |
| 22:52.59 | jasonwoot | skeevy goider stabbers |
| 22:53.36 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 22:57.34 | jblack | eric_hill: Some phones can register to more than one place at a time, yes. |
| 22:58.20 | eric_hill | jblack: I'm looking at deploying a dozen or so phones to a branch office that has a Cisco router with a few voice ports on it. |
| 22:58.40 | eric_hill | jblack: I want to make sure that if the WAN goes down, calls can still come and go from the building. |
| 22:59.05 | eric_hill | jblack: Not sure if the Cisco can be a SIP registrar.... I know it does skinny for Cisco phones, but... |
| 23:01.27 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 23:06.21 | *** join/#asterisk eliel (n=eliel@151-202-114-200.fibertel.com.ar) |
| 23:06.31 | *** join/#asterisk DrCron (n=rszasz@saxonco.com) |
| 23:14.52 | jameswf-home | heh http://www.thesmokinggun.com/archive/years/2008/0806082hammer1.html |
| 23:15.01 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 23:15.24 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 23:18.18 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 23:18.43 | *** join/#asterisk spartan949 (n=spartan7@wsip-70-169-241-66.oc.oc.cox.net) |
| 23:19.47 | spartan949 | Hello y'all. I am new to asterisk and was wondering what hardware requirement is needed to run 15 phones on asterisk |
| 23:19.47 | Paige_ | anyone here have the BOOK handy? |
| 23:19.56 | Paige_ | [TK]D-Fender, do you? |
| 23:22.20 | spartan949 | i understand that you do not need any phone cards? Can I just attach the ip phones to the network using an ethernet cord? Can I run 15 lines all on softphones? |
| 23:23.20 | spartan949 | i have a Pentium celeron with 256 ram. can that run 15 lines? |
| 23:23.45 | spartan949 | any help would be appreciated. Thank you |
| 23:24.01 | Paige_ | spartan949, should handle it fine |
| 23:26.39 | hoegaatit | when running asterisk on a VPS, does it perform better on a xen system or openvz? |
| 23:31.47 | spartan949 | Paige, can I run all soft phones? |
| 23:33.57 | zuez | Does asterisk support saving more than one user defined column these days in a CDR table? |
| 23:35.11 | zuez | I know in previous versions (and perhaps current as well) you're able to define one custom userfield with something like Set(CDR(userfield)=${foo}) |
| 23:38.22 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:39.57 | *** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net) |
| 23:40.20 | jeev | ~sipnat |
| 23:40.20 | 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 |
| 23:41.39 | AndyML | I'm looking or a way to recover the 'callback' channel for a caller. I'm aware of ${CHANNEL} (i.e. SIP/232-2342) but need just SIP/232 - is there a way to strip off the -2342 without using :5 - the number of digits change right? |
| 23:45.52 | AndyML | I guess I could be more specific - I need to incorporate it into dialplan logic |
| 23:50.41 | ZX81 | AndyML: have a look at the cut function |
| 23:50.53 | ZX81 | thing it defaults to cutting - |
| 23:50.58 | ZX81 | *think |
| 23:51.04 | AndyML | zx81 - will do, thanks for the recommendation |
| 23:51.08 | ZX81 | np |