| 00:00.06 | niekvlessert | It's getting interesting, but it's 01:00 AM overhere :) |
| 00:00.09 | niekvlessert | but nevermind |
| 00:00.15 | niekvlessert | ok |
| 00:00.21 | [TK]D-Fender | niekvlessert: As-is its meant for updating the display of a caller as they progress through the dialplan. |
| 00:01.07 | niekvlessert | which patch are we talking about here? |
| 00:01.24 | CrashSys | You could use SIPSAK running on the server and an AGI script to call it to update the phone |
| 00:01.29 | niekvlessert | excuses me 8824 |
| 00:01.36 | CrashSys | that's what i've done before with ViciDial |
| 00:01.48 | niekvlessert | SIPSAK?? |
| 00:01.57 | CrashSys | SIP Swiss Army Knife |
| 00:02.05 | CrashSys | aka SIPSAK (the project name) |
| 00:02.06 | niekvlessert | http://sipsak.org/ |
| 00:02.36 | niekvlessert | clever |
| 00:02.46 | CrashSys | You'd have to do some sort of lookup to find the info you wanted, then you can pass it to that and blast whatever the hell you wanted to the remote phone |
| 00:02.59 | niekvlessert | can be done |
| 00:03.01 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:03.06 | niekvlessert | we have a DB with a lot of stuff in there |
| 00:03.13 | niekvlessert | so....... |
| 00:03.25 | niekvlessert | before i do the actual dial command |
| 00:03.35 | niekvlessert | I do this sipsak agi thing |
| 00:03.40 | CrashSys | Not quite sure how you will handle NAT traversal with it, but it's what i've used before to send display info to an originating party's phone |
| 00:03.40 | niekvlessert | and update the phone display |
| 00:03.54 | niekvlessert | we don't need NAT traversal |
| 00:03.57 | CrashSys | That would be the general idea |
| 00:04.07 | niekvlessert | mmmmmm |
| 00:04.10 | niekvlessert | mmmmmmm |
| 00:04.14 | niekvlessert | interesting |
| 00:04.23 | CrashSys | then it should work fine. If you had NAT traversal you would probably have to look up the external port that the NAT opened. |
| 00:04.25 | niekvlessert | So I could also fix the pickup thing with this |
| 00:04.43 | CrashSys | No clue, what is the pick-up thing? |
| 00:05.01 | niekvlessert | someone calls my collague |
| 00:05.03 | *** join/#asterisk jakent (~john@c-98-233-13-157.hsd1.va.comcast.net) |
| 00:05.19 | niekvlessert | I use *8 (pickup inside the group) |
| 00:05.26 | niekvlessert | i Get the someone on the phone |
| 00:05.29 | CrashSys | Try using #8 |
| 00:05.37 | niekvlessert | but the phone still displays *8 |
| 00:05.40 | CrashSys | *8 gets funky on most SIP phones as a dialing character for some reason |
| 00:05.47 | niekvlessert | Interesting! |
| 00:05.48 | CrashSys | well, polycoms at least |
| 00:06.00 | niekvlessert | *8 works completely fine |
| 00:06.07 | niekvlessert | but wrong number on the display |
| 00:06.17 | CrashSys | ok |
| 00:06.27 | niekvlessert | I could do the AGI thing |
| 00:06.36 | CrashSys | Then you can use SIPSAK to update the phone ID if you want |
| 00:06.43 | niekvlessert | I guess I could |
| 00:06.47 | niekvlessert | I'm really happy now :) |
| 00:07.02 | niekvlessert | worth a try |
| 00:07.17 | niekvlessert | and I could also fix the attended transfer thing with this I guess |
| 00:07.21 | niekvlessert | little more difficult |
| 00:07.28 | niekvlessert | because the phone makes the bridge |
| 00:07.38 | niekvlessert | i can't execute agi right away |
| 00:07.42 | niekvlessert | I'd have to listen to AMI |
| 00:07.49 | niekvlessert | but it's doable |
| 00:07.50 | niekvlessert | right? |
| 00:08.37 | CrashSys | Probably want to use the AMI so you can look up the other caller's CID info |
| 00:09.07 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 00:09.09 | niekvlessert | yeah, and also that the actual transfer is happening, because that's not happening in the dialplan but on the phone |
| 00:10.50 | niekvlessert | damn my buffer was full but i have most of the conversation :) |
| 00:11.14 | niekvlessert | anyone in here has experience with CDR's? |
| 00:11.26 | niekvlessert | that's also a topic we're having some trouble with :) |
| 00:12.05 | [TK]D-Fender | niekvlessert: You *8 could be solved by the patch I mentioned |
| 00:12.14 | CrashSys | CDRs suck |
| 00:12.15 | [TK]D-Fender | your |
| 00:12.35 | niekvlessert | Fender: you mean the 8824 patch? |
| 00:12.50 | niekvlessert | yeah I see it now |
| 00:13.02 | niekvlessert | that was the patch we tried |
| 00:13.09 | niekvlessert | It compiled |
| 00:13.12 | niekvlessert | but that wat it :) |
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| 00:13.16 | niekvlessert | *was |
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| 00:14.02 | niekvlessert | the first direction we were going in |
| 00:14.11 | niekvlessert | - Call Junghanns |
| 00:14.37 | niekvlessert | - He said if we could apply the patch on 1.4.XXX and it would work he could port it to 1.4.26.2 |
| 00:14.47 | niekvlessert | - We never got it working :) |
| 00:15.14 | niekvlessert | CrashSys: CDRs suck indeed |
| 00:15.50 | niekvlessert | We've come a long way... we've added fields to the mysql cdr table |
| 00:16.04 | niekvlessert | but queue logging... that's something else |
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| 00:18.42 | CrashSys | I don't use Asterisk for CDRs, I generate those from OpenSIPS event logs |
| 00:19.27 | niekvlessert | crashsys: very entertainig, the stuff you say |
| 00:19.28 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 00:19.37 | niekvlessert | today I learned that might be a good idea |
| 00:19.46 | niekvlessert | but what about queue logging? |
| 00:19.52 | niekvlessert | that will happen in Asterisk right? |
| 00:19.59 | CrashSys | CDRs have nothing to do with queues |
| 00:20.09 | niekvlessert | agree |
| 00:20.18 | CrashSys | I don't use Asterisk Queues |
| 00:20.20 | niekvlessert | QDRs :) |
| 00:20.50 | niekvlessert | you probably use queues from another call center like pbx system? |
| 00:20.54 | CrashSys | Just buy Queue Metrics if you want something to look at queues |
| 00:21.23 | p3nguin | Okay, so the problem with my phone NOT liking type=peer is that it is not even bothering to try authenticating with username. |
| 00:22.06 | niekvlessert | ok, that's something I can do |
| 00:22.13 | p3nguin | 7912G on the same LAN... type=peer is just fine. |
| 00:22.15 | niekvlessert | but what system do you use for queues then? |
| 00:22.40 | niekvlessert | you all should try the Aastra, XML push features are very very nice BTW. And LLDP & VLAN support and stuff |
| 00:23.24 | niekvlessert | although the LLDP is not finished yet... but at least it's not CDP :) |
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| 00:24.02 | CrashSys | niekvlessert: I use our own queuing-type of system called ViciDial. |
| 00:24.16 | CrashSys | But unless you are running a call center it won't help you much |
| 00:24.40 | drmessano | What about if you have teenagers? |
| 00:24.46 | CrashSys | shoot them |
| 00:24.51 | CrashSys | I mean, talk to them |
| 00:24.55 | niekvlessert | well, sometimes the area between a queue and callcenter is sometimes very gray |
| 00:25.11 | niekvlessert | forget the first sometimes :) |
| 00:25.29 | drmessano | niekvlessert: Wouldn't that be the parking lot? |
| 00:25.32 | CrashSys | niekvlessert: The solution is to learn the queue log if you want to reinvent the wheel and write your own stuff |
| 00:25.42 | drmessano | and wouldnt that depend on time of day, weather, season, etc? |
| 00:25.49 | *** join/#asterisk rpm- (~rpm@S01060014c25cd036.cg.shawcable.net) |
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| 00:26.44 | rpm- | to playback a mp3 in an asterisk context if i have the format_mp3 addon loaded (asterisk 1.6) i should be able to use the Playback application and specify the basename of the mp3 to listen to it right? |
| 00:27.51 | niekvlessert | CrashSys: ok, I will take that in account. But I'm very interested in OpenSips |
| 00:28.03 | niekvlessert | Because Asterisk does some reinvite stuff, but it's not a proxy |
| 00:28.32 | niekvlessert | and I want to use an audiocodes or whatever gateway to convert ISDN to SIP |
| 00:28.37 | niekvlessert | and use G729 |
| 00:28.48 | CrashSys | you just want all the buzzwords :D |
| 00:28.52 | drmessano | lol |
| 00:29.10 | niekvlessert | :) |
| 00:29.10 | drmessano | No kidding, you forgot Skype, IAX, and Bluetooth |
| 00:29.19 | CrashSys | chan_mobile to the rescue! |
| 00:29.24 | niekvlessert | Oh yeah we do want skype |
| 00:29.25 | niekvlessert | ;) |
| 00:29.34 | niekvlessert | Bluetooth is also on the latest Aastra phone |
| 00:29.46 | niekvlessert | it has gbit network, touch screen, usb and bluetooth |
| 00:29.53 | voipmonk | does chan_mobile work on the iphone in asterisk? |
| 00:30.01 | drmessano | Gbit network, ha |
| 00:30.14 | niekvlessert | for the PC's behind it... |
| 00:30.25 | niekvlessert | I know, it's useless, but ppl think they need it |
| 00:33.59 | *** join/#asterisk pawz (~pawz@ppp118-208-106-155.lns20.bne4.internode.on.net) |
| 00:34.45 | niekvlessert | rpm-: core show application MP3Player |
| 00:35.49 | niekvlessert | Ok you guys, thank you very much for your help, I have new hope that we will be able to fix this |
| 00:36.18 | niekvlessert | I will be back! |
| 00:36.27 | drmessano | Is that a threat? |
| 00:36.32 | niekvlessert | CrashSys: thanks a logt |
| 00:36.43 | niekvlessert | drmessano: depends on how nice you are to me |
| 00:36.45 | *** part/#asterisk niekvlessert (~niekniekn@5ED25657.cable.ziggo.nl) |
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| 00:38.19 | drmessano | lol |
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| 00:43.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 00:44.27 | Sparky-UK | Can someone help me with a very simple problem, I want to be able to dial a number and run a dialplan from extensions_custom.conf, I think I am just missing one key step to get it working? |
| 00:46.39 | CrashSys | sounds like freepbx |
| 00:47.01 | Sparky-UK | yes, actually the trixbox varient |
| 00:47.23 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 00:48.08 | CrashSys | ~freepbx |
| 00:48.09 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 00:48.22 | Sparky-UK | thank you |
| 00:48.23 | CrashSys | Hmm, iy eotkrf |
| 00:48.27 | drmessano | Hmm |
| 00:48.29 | CrashSys | err it worked |
| 00:48.47 | CrashSys | You know that's bad, when you type one-key off on your keyboard and you are so geeky that you understand it anyways |
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| 01:04.16 | ecrane | How come these phone fraud people always wanna call Cuba? |
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| 01:24.23 | p3nguin | ecrane: Bored, and think Cubans might spice things up a bit? |
| 01:35.04 | ecrane | p3nguin: heh.. could be. |
| 01:35.25 | p3nguin | Maybe it's some type of Cuban sex hotline. |
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| 01:51.25 | BeeBuu | if i want run SS7 with libss7, am i must configure it using file ss7.conf ? |
| 01:53.52 | titter | Can someone explain why the CLI closes after executing this dialplan? The dialplan works, and I get the fax to my e-mail ... but I get some T30 errors, and the CLI cloeses at the end. Already tried 1.6.1.14. I went back to 1.6.1.9 after reading a bug report, and it doesn't do it. |
| 01:53.56 | titter | http://pastebin.com/d7e72ab4b |
| 01:54.46 | *** join/#asterisk jakent (~john@c-98-233-13-157.hsd1.va.comcast.net) |
| 01:56.50 | leifmadsen | titter: the disconnection is likely due to a segfault, which means you likely either have the same bug as something already reported, or you need to follow the instructions in doc/backtrace.txt of your asterisk sources to report a bug |
| 01:58.07 | titter | will do -- thanks. this was what happened when it executed the first time after 14 was installed -- http://pastebin.com/d529c31a |
| 02:02.25 | *** join/#asterisk moy (~moy@74.12.123.169) |
| 02:05.35 | *** join/#asterisk DrkShadow (~andrew@host-72-175-240-62.static.bresnan.net) |
| 02:11.56 | DrkShadow | By default, Voicemail says "Please leave your message after the tone." ... that's if I don't put a busy or unavailable flag to the voicemail app. How can I change that to a customized greeting? seems the user can only change busy and unavailable... |
| 02:12.01 | *** join/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 02:13.05 | LemensTS | i got a PAP2T and when i pick up the phone it will do a few quick beeps letting me know its off hook for like a second or 2, than it will go to an error tone. If i dial during the first quick beeps i can call out, if not i cant. This a PAP2T setting? |
| 02:13.35 | voipmonk | DrkShadow: - do you want to remove it or replace it? |
| 02:13.45 | DrkShadow | replace it. |
| 02:14.07 | p3nguin | When you replace the sound files, it will be global. |
| 02:14.41 | voipmonk | go for it |
| 02:14.47 | voipmonk | find the name of the file |
| 02:14.52 | voipmonk | by watching the cli |
| 02:14.56 | DrkShadow | I want a _personalized_ version, not a global version. |
| 02:14.59 | voipmonk | and replace it :) |
| 02:15.02 | voipmonk | ahh |
| 02:15.11 | voipmonk | then so in addition to their personal greeting |
| 02:15.20 | DrkShadow | how do I get their personal greeting? |
| 02:15.31 | DrkShadow | they can record unavailable and busy. Unavailable doesn't play at the start. |
| 02:15.43 | voipmonk | u can either put one in their directory or mysql table or u can have them record it |
| 02:17.27 | p3nguin | Maybe the "temporary greeting" can be of some use. |
| 02:17.52 | p3nguin | From the mailbox, press 0 for mailbox options, then I think it's 4 for temporary greetings. |
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| 02:23.31 | LemensTS | interdigit short value |
| 02:23.39 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 02:23.41 | [TK]D-Fender | DrkShadow: "core show application voicemail" <-------- |
| 02:24.05 | [TK]D-Fender | DrkShadow: And no.. its not the "temporary greeting" |
| 02:24.09 | DrkShadow | I was thinking of temporary greeting.. but.. wtf. |
| 02:25.07 | DrkShadow | as far as I can tell, there is no way to personalize the vm-intro on a per-mailbox basis aside from using u or b with the voicemail app. Lame. |
| 02:25.16 | voipmonk | yes there is |
| 02:25.37 | DrkShadow | voipmonk: the temp greeting? uh huh. |
| 02:25.40 | voipmonk | no |
| 02:25.58 | DrkShadow | putting the files on the server manually? I could do that.. but, lame.. |
| 02:26.11 | voipmonk | Playback works beautiful BEFORE the voicemail app kicks in, and set the default greeting to a silence |
| 02:26.13 | DrkShadow | actually I'm not even sure that would work. I can't find any documentation on the file path. |
| 02:26.18 | DrkShadow | heh. |
| 02:26.21 | voipmonk | now push the staples button |
| 02:26.24 | voipmonk | "That was easy!" |
| 02:26.34 | DrkShadow | that was a consideration, too, but no way for them to record/change it themselves. |
| 02:26.40 | voipmonk | yes there is |
| 02:26.45 | voipmonk | ok.... |
| 02:27.12 | [TK]D-Fender | DrkShadow: they have a greeting. Thta is what they personalize. |
| 02:27.29 | DrkShadow | Fender: they don't have a greeting. The greeting can not be personalized. |
| 02:27.37 | voipmonk | I feel like you're cheating... |
| 02:28.03 | DrkShadow | voipmonk: by trying to manage voicemail greetings through the voicemail app as can be done with busy and unavailable? |
| 02:28.11 | voipmonk | its right there in front of you - watch the cli, asterisk spits out the answer right there for you to see |
| 02:28.26 | voipmonk | then core show application Voicemail |
| 02:28.31 | DrkShadow | I assure you, it doesn't. vm-intro is a global setting |
| 02:28.36 | DrkShadow | er, global soundfile. |
| 02:29.09 | DrkShadow | and core show app voicemail shows nothing that isn't in 5 documents I've already read. |
| 02:29.12 | DrkShadow | my god |
| 02:29.22 | DrkShadow | gpl software sucks, and the "support" is a great deal worse. |
| 02:29.32 | titter | ... |
| 02:29.50 | [TK]D-Fender | [21:28]<DrkShadow>I assure you, it doesn't. vm-intro is a global setting <--- |
| 02:29.54 | [TK]D-Fender | Yup.. BRILLIANT |
| 02:30.23 | titter | Now I understand why I get the "are you a fucking moron" talk when I ask something semi stupid in here. |
| 02:30.25 | [TK]D-Fender | Dumb-fuck couldn't see that this wasn't a GREETING it was the INSTRUCTIONS that the "s" option lets them DISABLE. |
| 02:30.53 | [TK]D-Fender | Winners never quit, and quitters never win |
| 02:31.06 | [TK]D-Fender | ...... hoever those that never win, and never quit are IDIOTS |
| 02:32.18 | rnp | does anyone here have Vici Dial setup where I could see the Agent interface? |
| 02:32.35 | jaytee | wow, I just came back and missed the wery, wery, wee-todd-id guy |
| 02:33.23 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 02:34.07 | [TK]D-Fender | jaytee: Sofa king wheat hearted. |
| 02:34.20 | jaytee | hehehee |
| 02:34.39 | jaytee | ever notice Ann Coulter has an Adam's apple? hmmmmm |
| 02:35.16 | [TK]D-Fender | feels dirty |
| 02:35.16 | titter | Ok ... there is a 3lb roll on my TV ... it can feed 12 people. Who the hell invents this crap? |
| 02:35.26 | titter | cinnamon roll* |
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| 02:36.03 | bluregard | hi all |
| 02:36.24 | *** join/#asterisk jmcdowell (~nooe@173-112-87-255.pools.spcsdns.net) |
| 02:36.25 | jaytee | I'm just waiting for when Wendy's comes out with a quadruple pattie burger |
| 02:36.39 | jmcdowell | hello all.. |
| 02:36.41 | file | who says you can't order one? |
| 02:36.58 | jmcdowell | I am seeking some sip related advise, not specific to asterisk but to polycom. |
| 02:37.05 | jaytee | again? |
| 02:37.14 | jmcdowell | Basically, I am getting ready to update the firmware to 3.x from 2.2, should I do this? |
| 02:37.23 | bluregard | does anyone know where I might find some case studies on VoIP service providers? |
| 02:37.29 | jmcdowell | Does anyone know any reason I shouldn't do so? |
| 02:37.36 | jaytee | lack of experience? |
| 02:37.50 | bluregard | I've dug around but all I can find are voip implementation studies |
| 02:37.53 | titter | jmcdowell: read the notes released by Polycom ... it can be pretty helpful, and informative |
| 02:38.19 | jmcdowell | titter: I am, and realizing that the firmware can be changed back and forth on the fly. |
| 02:38.22 | jmcdowell | Pretty interesting. |
| 02:38.40 | bluregard | I'm trying to put together some real world numbers on call volume, circuit usage, bandwidth use etc for a SIP provider |
| 02:38.40 | rnp | does anyone here run their own call center? |
| 02:38.59 | jmcdowell | not yet, but I have 5 kids, so it's coming soon. :D |
| 02:39.00 | bluregard | rnp: we run a call center |
| 02:39.05 | rnp | lol jmc |
| 02:39.05 | jaytee | no, but I run my own call perimeter |
| 02:39.14 | rnp | what is a call perimeter? |
| 02:39.22 | titter | jmcdowell: do you have more than one model of Polycom's you are upgrading? If they are all the same ... it's pretty simple. I personally use http in the tftp dhcp option to upgrade the firmware, and bootrom, as well as load cfg files. |
| 02:39.38 | rnp | blue, is your call center for your own proprietary calls or do you take on all campaigns? |
| 02:39.40 | jmcdowell | titter, I am still reading.. |
| 02:39.44 | jmcdowell | they are all the same though.. |
| 02:39.46 | [TK]D-Fender | jaytee: Perimeter? PI r not squared. PI r round! |
| 02:39.47 | jaytee | it's like a decentralized call center. |
| 02:39.58 | bluregard | rnp: just ours |
| 02:40.18 | rnp | blue: what is the goal of your calls if you mind my asking, set appointment, make a sale? |
| 02:41.01 | rnp | jaytee: how do you run your call perimeter? where are you in relation to your workers? |
| 02:41.18 | bluregard | rnp: usually to make an appointment or at least establish a relationship |
| 02:41.19 | jmcdowell | I thought the bootrom came with the sip firmware |
| 02:41.34 | jaytee | mp, pay no attention to the man behind the curtain. He has nothing to do with the all powerful Oz |
| 02:41.35 | jmcdowell | these polycoms are such a pain.. |
| 02:41.36 | titter | jmcdowell: they are seperate |
| 02:41.47 | paulc | I love my polycoms |
| 02:41.53 | titter | I also love mine |
| 02:41.55 | bluregard | rnp: but that's the very small sales portion of the call center. Our support call center is much bigger. |
| 02:41.59 | paulc | although their default XML provisioning files could be better laid out |
| 02:42.02 | jaytee | I <3 my Polycoms |
| 02:42.15 | rnp | blue: so most of your calls are inbound for support? |
| 02:42.21 | rnp | blue: how many support staff do you have? |
| 02:42.23 | bluregard | rnp: yeah |
| 02:42.35 | jaytee | "Mrs. Gump, yoah boy's different" |
| 02:42.39 | titter | paulc: true, but you can customize them into your own personal xml files that make more sense to you |
| 02:42.45 | [TK]D-Fender | jmcdowell: BootROM and Application are separate |
| 02:42.45 | jmcdowell | I love the sound quality |
| 02:42.54 | bluregard | rnp: all together I'd say ~100 |
| 02:42.55 | jmcdowell | I am just saying, pain to get up and running. |
| 02:43.08 | [TK]D-Fender | jmcdowell: And there is no actual pain to Polycoms. They work as advertised, and the docs work |
| 02:43.09 | rnp | blure: do you own the company? |
| 02:43.11 | paulc | Love/Hate: Love Polycoms, Asterisk, playing with that stuff, jelli.net Hate: current job, not winning the lottery, not having enough time to play with the fun stuff |
| 02:43.17 | bluregard | rnp: hell no |
| 02:43.28 | rnp | blure: so what's your position? |
| 02:43.39 | bluregard | sr consultant |
| 02:43.41 | paulc | titter: yeah - I did that recently for a "not my problem" assist to our tech team.. laid out a nice simple "one parameter per line" XML file so they could copy'n'paste.. saved the day.. RESULT :) |
| 02:43.41 | jmcdowell | Well.. I stand by my position that they are grear, just poor documentation |
| 02:43.46 | jmcdowell | great |
| 02:43.53 | jmcdowell | anyway, I have to get to reading.. |
| 02:43.54 | jmcdowell | bbl |
| 02:44.00 | rnp | blure: got it |
| 02:44.00 | jaytee | I never had a problem with the documentation |
| 02:44.07 | titter | jmcdowell: the docs are actually very good |
| 02:44.11 | rnp | jaytee: tell me more about your call center operations if you don't mind |
| 02:44.21 | jaytee | reading is fun-da-mental |
| 02:44.42 | jaytee | mp, do you have access to a dictionary? |
| 02:44.51 | bluregard | so is anyone in here a voice service provider? |
| 02:45.05 | titter | paulc: I did the same, just used a little ASP.net as cruel as it may sound. |
| 02:45.09 | paulc | I wasn't clear on polycom behind NAT - there's the keep alive timer, but does it do anything if you don't set the NAT address/port parameters? Or are you better off with a fairly low registration refresh timer? |
| 02:45.10 | jaytee | look up words like sarcasm, snarky, wiseass |
| 02:45.20 | paulc | titter: glutton for punishment? ;) |
| 02:45.23 | paulc | prefers PHP |
| 02:45.32 | paulc | but that's cos I'm a script kiddy, not an OOP guy |
| 02:45.35 | paulc | I'm fine with that :) |
| 02:45.36 | jmcdowell | It's a do or die night for me, I am getting these things working, or getting rid of them. |
| 02:45.49 | paulc | jmcdowell: what's your biggest problem with them? |
| 02:46.00 | jmcdowell | They keep cycling after "loading app" |
| 02:46.02 | titter | paulc: PHP/ASP.net C# all the same to my eyes ... I just like how quickly I can build a XML file in ASP.net lol |
| 02:46.20 | paulc | titter: ah yes, I have no doubt it's probably quicker in ASP.net - fair play |
| 02:46.22 | jmcdowell | I have something wrong in the provisioning, I figured I would get them as up to date as I can and then see what happens. |
| 02:46.53 | jaytee | jmcdowell, get them to work before you start upgrading firmware |
| 02:46.55 | paulc | jmcdowell: can you try with a brand new fresh install of the latest boot/sip firmware, configure a single account on a single phone, and see if it works? |
| 02:47.09 | rnp | paulc: what do you do? |
| 02:47.14 | jmcdowell | I know it works, I can make it work if I set it up manually |
| 02:47.20 | jmcdowell | but I want the provisioning to work |
| 02:47.22 | jaytee | 2.2 is fine, 3.x has more features but they aren't absolutely necessary |
| 02:47.27 | jmcdowell | and I am working with a single phone @ the moment. |
| 02:47.35 | titter | paulc: it would probably be just as easy in PHP, but, I just happened to have Visual Studio open, and was working on an Asterisk project ... so I figured why not |
| 02:47.39 | jmcdowell | Wow a 30 meg boot rom |
| 02:47.59 | [TK]D-Fender | jmcdowell: What model of phone? |
| 02:48.07 | jmcdowell | 601 |
| 02:48.22 | [TK]D-Fender | jmcdowell: 3.1 is the newest they support IIRC |
| 02:48.29 | jmcdowell | New BOOTROM? |
| 02:48.45 | jmcdowell | Or firmware? Because I think I got the 3.1 from the matrix |
| 02:48.52 | jmcdowell | the bootrom wasn't quite as clear |
| 02:49.03 | jaytee | jmcdowell, sometimes if you have a messed up config you might need to go into the Admin settings on the phone and from the Reset to defaults menu choose Format File System. |
| 02:49.25 | jmcdowell | I just downloaded 3.1.6 from their website. |
| 02:49.31 | paulc | rnp: kind of a developer, kind of an architect (but it's a wank title/term), also a problem solver, shit fixer, bit of this, bit of that.. all depends on the day |
| 02:49.32 | jmcdowell | That's the firmware version |
| 02:49.37 | paulc | rnp: how about you? |
| 02:50.12 | rnp | paulc: pinhead who outsources his call center work and is ready to take control of his business! |
| 02:50.49 | rnp | paulc: so you integrate call center solutions for businesses? |
| 02:51.02 | bluregard | rnp: how many agents are you looking to have? |
| 02:51.11 | jaytee | paulc, lately I've been using the title Technomage or Technomancer depending on my mood |
| 02:51.20 | rnp | blure: not looking for agents, I have about 20 |
| 02:51.33 | paulc | rnp: I've been known to do that sort of thing, outside the scope of my day job ;-) |
| 02:51.41 | bluregard | rnp: I thought you outsource it? |
| 02:51.41 | rnp | blure: but my agents are in the philippines and I'm in the USA, and basically I just leave them to their devices and don't control what they do and that needs to change. |
| 02:51.52 | paulc | jaytee: That's cool and funky.. I like it more than I like "Chief Scientist" - a job title a guy who used to work here adopted a while back |
| 02:52.09 | jmcdowell | So it says it's for the Soundpoint 600, does that cover the 601? |
| 02:52.17 | titter | jmcdowell: the bootrom itself is 13mb, however if you use the model specific bootrom it is much smaller |
| 02:52.22 | [TK]D-Fender | jmcdowell: Do not touch the BootROM unless you have to |
| 02:52.26 | bluregard | rnp: so what exactly are you looking to do? |
| 02:52.33 | jmcdowell | I have to if I want to upgrade the firmware |
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| 02:52.50 | jmcdowell | my firmware is at like 2.2x |
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| 02:52.59 | [TK]D-Fender | jmcdowell: BR & application are both "firmware" |
| 02:53.13 | rnp | blure: what I'm looking to do is instead of having my call centers just manually dial and manually pull leads, I want them to be able to just login and know what businesses to call |
| 02:53.14 | [TK]D-Fender | jmcdowell: You want to upgrade SIP. You probablky have no need to touch the BR |
| 02:53.17 | jaytee | outsourcing has it's drawbacks. in the final analysis you can often achieve the same level of customer dissatisfaction with 1 local employee whose native language is English than 20 Indian call center employees that sound like Apu from the Kwik-E Mart on the Simpsons |
| 02:53.22 | paulc | I got the latest boot + firmware from the Polycom site and had no problem with installing it.. that said, they were newer phones, not 601s.. |
| 02:53.28 | *** part/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 02:53.28 | rnp | blure: know which call backs to make, etc, this way they are not wasting leads, and I know what is actually going on. |
| 02:53.31 | paulc | rnp: have you looked at vicidial? |
| 02:53.39 | bluregard | rnp: so do you have a database that houses all of this information? |
| 02:53.51 | titter | [TK]D-Fender: depends on implementation ... I like 4.0 and new BR so I can use the model specific app files |
| 02:53.53 | jmcdowell | The notes say that if I want to use 3.1.x firnware I have to have 4.1.x firmware |
| 02:54.05 | rnp | paulc: yes, I'm looking at vicidial + asterisk |
| 02:54.08 | jmcdowell | Why shouldn't i update the bootrom? |
| 02:54.25 | paulc | rnp: I've never played with vicidial but I know it's got a fairly good reputation...I think? |
| 02:54.39 | rnp | blure: my database houses just the account information, for instance the areas to call and the # of appointments that need to be set. |
| 02:55.07 | [TK]D-Fender | jmcdowell: both the SIP load and you can always refresh it. Botch the BR and you're bricked |
| 02:55.20 | rnp | paulc: well I know I want to have my own system that I can regulate and manage but it needs to be inexpensive like Vici. I outsource my work and they are paid commission, thus I can't have high overheads for their labor, because they already have their own voip setup. |
| 02:56.06 | jaytee | jmcdowell, I'll say it again. you should focus on getting your phone configs with provisioning in 2.2 working before you start "playing" with a bootrom upgrade. |
| 02:56.16 | jmcdowell | [TK]D-Fender : Does this happen often? I mean, I have to update the bootrom to update the firmware. |
| 02:56.24 | rnp | and I want to be able to crowscource my operations, esentially let anyone sign up and do work for us around the world and have it all automated. |
| 02:56.28 | jmcdowell | I suspect the problems I am having are related to the said firmware. |
| 02:56.30 | [TK]D-Fender | jmcdowell: Highly doubt ir |
| 02:56.31 | [TK]D-Fender | it |
| 02:56.44 | jmcdowell | Ok... |
| 02:56.46 | jmcdowell | Hmmm |
| 02:56.57 | jmcdowell | Is it common to "brick" a polycom? |
| 02:57.03 | [TK]D-Fender | jmcdowell: No. |
| 02:57.04 | jaytee | I run 2.2 just fine on over 100 Polycom 330's and a half dozen 550's |
| 02:57.16 | [TK]D-Fender | jmcdowell: Power failure during BR install would do it |
| 02:57.16 | jmcdowell | I mean, according to the doc, you simply provision it out the same way I have been. |
| 02:57.22 | jmcdowell | ohh |
| 02:57.26 | jmcdowell | no worrys there.. |
| 02:57.27 | jmcdowell | LOL |
| 02:57.31 | paulc | rnp: sounds interesting.. might not be completely "inexpensive" to get up and running, but once you're there - kerching, watch the cash some in |
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| 02:58.16 | [TK]D-Fender | jmcdowell: SoundPoint IP 601 3.1.0 or greater <--- BR req for SIP 3.1.6 |
| 02:58.19 | bluregard | rnp: something where an agent logs in, a backend server dials a number for them then provides the agent the information they need on their PC? |
| 02:58.35 | rnp | paulc: well i've been doing this for 2 years now with absolutely no control of operations since I just send them their work and wait for the appointments. I'm ready |
| 02:58.43 | paulc | waits patiently for a Visual Source Safe share'n'branch to finish, so he can go home (love: subversion, hate: VSS) |
| 02:58.52 | paulc | rnp: so how much playing have you done with Asterisk, Vicidial, etc? |
| 02:58.54 | rnp | blure: exactly, that's what I want, and then I want it integrated with my custom online crm so my clients are notified |
| 02:59.21 | rnp | paulc: I'm not sure, I suppose just what needs to be done to get things working. |
| 02:59.51 | bluregard | rnp: that isn't unreasonable. |
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| 03:00.17 | bluregard | rnp: you said your agents have their own VoIP setup? |
| 03:00.46 | rnp | blure: yeah, I mean my agents have voip on their computers, some of my centers even have auto dialers |
| 03:01.04 | rnp | blure: however, my two main centers have 5 agents each and they just manual dial |
| 03:01.13 | rnp | blure: they are in the philippines |
| 03:01.45 | bluregard | rnp: I guess what I mean is where do you want the calls to originate? From your system or from your agents' system? |
| 03:02.27 | rnp | blure: it doesn't matter to me where they originate, and I would prefer not to actually pay for the voip |
| 03:02.44 | bluregard | rnp: if your agents are using soft phones then I'd imagine you would be providing the actual voice service. |
| 03:02.52 | rnp | blure: however, I want to control their work flow and the nubmers they dial. and since they all have different voip setups over there, it's impossible without paying for the voip i think |
| 03:03.09 | rnp | blure: well when we move over to vici, i will be providing the voip, now I don't |
| 03:03.46 | bluregard | rnp: you're already set on moving to vici? |
| 03:03.54 | jmcdowell | If I just DROP from the IRC then you know the power failed while flashing because god hates me. |
| 03:03.56 | jmcdowell | lol |
| 03:04.56 | jmcdowell | that would be my luck |
| 03:05.05 | rnp | blure: well I started designing my own custom dialer that would be competely proprietary, but it was getting too complicated and it wasn't going to allow for auto dialing. Thus I've made the decision to go with some software that can do all this for me and that isn't a hosted solution because the expense of hosted is way out of my reach. |
| 03:06.27 | jaytee | jmcdowell, I talked to God earlier in the evening. He said He doesn't hate you but He's still a little pissed at you about that book you never returned to the library that you checked out back in 7th grade. |
| 03:06.57 | rnp | I like what Five9 is doing but they want $150 a month per seat, I have 20 agents, even if 10 of them dial at a time that's $1500 a month, tha'ts unreasonable |
| 03:07.18 | rnp | my workers make around $200 a month |
| 03:07.25 | rnp | makes no sense to pay $150 for their dialer |
| 03:07.47 | jmcdowell | 200 a month? |
| 03:07.54 | jmcdowell | Where are they from? A warehouse in china? |
| 03:07.55 | paulc | offshore = cheap workers |
| 03:07.59 | paulc | philipines |
| 03:08.01 | bluregard | rnp: what kind of budget do you have for this provided it's a capital expense? |
| 03:08.01 | rnp | they are in the philippines |
| 03:08.10 | jmcdowell | damn |
| 03:08.20 | jmcdowell | i aint askin you for a part time job |
| 03:08.25 | rnp | blure: honestly, I hadn't really thought it through. |
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| 03:08.58 | rnp | blure: I spent the better part of a year and a half creating my custom crm software to manage my business |
| 03:09.18 | rnp | blure: but it hasn't stopped all of the problems. the problem still persists that my agents don't know what to work on and they have no way of managing all the data on their own |
| 03:09.30 | rnp | blure: they need something to keep track of calls, call backs, no answers, etc, etc. |
| 03:10.06 | rnp | blure: and then the additional problem of relaying to my clients that we've run out of leads to call |
| 03:10.21 | rnp | blure: i can't tell a client that we've run out of leads and we need a bigger area because I never know when we actually do run out of leads |
| 03:10.31 | bluregard | rnp: what backend is your CRM on? |
| 03:10.40 | rnp | mysql |
| 03:10.54 | rnp | it's all ajax + mysql |
| 03:11.18 | bluregard | well thats a damn good start |
| 03:11.30 | paulc | I'm outta here - VSS be damned, I'm going home! |
| 03:11.40 | rnp | see you paullyc |
| 03:11.49 | rnp | good thing your names not paullyd |
| 03:11.50 | rnp | ;p |
| 03:11.54 | paulc | ..becauase? |
| 03:12.07 | rnp | mtv reference to the new hit show jersey shore |
| 03:12.07 | rnp | lol |
| 03:12.08 | paulc | (and just "Paul" is fine - I'm not keen on "Pauly C") |
| 03:12.10 | paulc | oh |
| 03:12.11 | paulc | LOL |
| 03:12.29 | paulc | that show is trainwreck trash TV.. yet vaguely addictive.. potentialyl |
| 03:12.38 | rnp | lol |
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| 03:12.50 | paulc | (and I'm from Jersey.. as in the island.. www.jersey.com - home of pretty moo cows) |
| 03:12.52 | rnp | it's so ridiculous |
| 03:13.01 | paulc | but now in Canada - Olympic city represent! |
| 03:13.10 | rnp | ah ;) |
| 03:13.25 | paulc | And on that note.. I'm off home.. back in a bit :) |
| 03:13.26 | titter | rnp: what type of business |
| 03:13.46 | rnp | titter: we make appointments for small vending operators in the usa |
| 03:14.19 | rnp | titter: what do you do? |
| 03:14.47 | titter | rnp: long term care insurance, reverse mortgages, and some other insurance types |
| 03:15.14 | titter | rnp: we generate all our own leads, either web, direct mail, or telemarketing |
| 03:15.27 | rnp | titter: very nice, what's your position in the business? |
| 03:15.36 | titter | rnp: we run two larger call centers in the phillippines, manila, and abucay |
| 03:15.45 | titter | rnp: senior sys admin |
| 03:15.56 | rnp | titter: got it, so the centers are your own? |
| 03:16.10 | titter | rnp: yup, I take care of them ... as well as all our other offices |
| 03:16.24 | rnp | titter: very nice, and you run all this on asterisk? |
| 03:17.05 | titter | rnp: all asterisk, minus one acquisition that had a dialer, and actually had a spare from an office the closed ... we use these in two offices in the u.s. |
| 03:17.25 | jmcdowell | Can someone look at this.. http://pastebin.ca/1776613 |
| 03:17.32 | rnp | titter: i see, so if you mind my asking, do you know what your average agents are making in the phlippines? |
| 03:17.59 | bluregard | jmcdowell, what about it? |
| 03:18.06 | titter | rnp: we are paying ours around 2.50-4/hr if I remember correctly. |
| 03:18.22 | rnp | titter: wow, that seems like a lot. |
| 03:18.42 | jmcdowell | that is confusing |
| 03:18.55 | titter | rnp: they are in charge of some more responsibility, not just telemarketing ... they call insurance carriers, and do some data entry for our insurance agents |
| 03:19.07 | rnp | titter: I see, got it |
| 03:19.21 | rnp | titter: so what would you recommend for me. asterisk + vici? |
| 03:19.46 | rnp | voipmonk made it sound fairly easy to integrate those with my mysql database |
| 03:20.00 | bluregard | certainly is. is this the polycom sip firmware compatibility |
| 03:20.18 | voipmonk | ZZZz |
| 03:20.22 | voipmonk | what time is it? |
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| 03:21.38 | titter | rnp: there are some different gpl vici avaiable, asterisk can integrate with many types of databases |
| 03:21.52 | titter | rnp: we use oracle in our org for all databases |
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| 03:22.47 | rnp | titter: I see, can you rec any other GPL? |
| 03:23.34 | titter | rnp: are you familar with asterisk, or new to it? |
| 03:23.46 | rnp | titter: brand new |
| 03:24.08 | rnp | titter: one of my Filipino's said it was good so that's how I got onto it |
| 03:24.15 | titter | rnp: check out http://www.vicidialnow.org/ |
| 03:25.01 | titter | rnp: asterisk is very powerful, depending on development skills, or access to developers, you can create your own dialer fairly easily. |
| 03:25.37 | rnp | titter: i sketched out what I want my system to look like, care to take a browse of my images? |
| 03:25.53 | rnp | titter: this is before I knew about asterisk or anything else |
| 03:26.22 | rnp | by the way, that ViCi Dial Now looks very attractive |
| 03:26.22 | titter | rnp: sure |
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| 03:26.47 | titter | rnp: if you are new to asterisk, and are looking for something now that is under a GPL and probable pretty easy to tinker with I would try it |
| 03:28.35 | rnp | titter: those images produced are my idea for my own custom dialer for our online crm |
| 03:29.12 | titter | rnp: resend, needed to reconfigure my dcc |
| 03:31.40 | rnp | titter: thoughts? |
| 03:32.19 | titter | rnp: looks interesting, looks similar to the system we acquired -- http://www.spitfiredialers.com/products.php |
| 03:33.43 | rnp | titter: hmmm |
| 03:34.51 | titter | rnp: cost wise I can't asnwer, as I am not in charge of those offices for that dept ... I just make sure things function. I do know they have a good support staff, and can assist with the software with the telemarketing managers making my job easier |
| 03:34.53 | rnp | titter: tech question, for me, we have a client that say has two different orders, both in the same area, however, one order may specific SIC codes and one order may be able to use all SIC codes for businesses. However, we don't want leads to overlap, so if we call a business for order 1 we don't want it called again for order 2 |
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| 03:35.36 | rnp | how would you go about that problem |
| 03:36.30 | rnp | so we have 1 client and 2 orders, some leads over lap, but not all leads. if lead1 is called for order1 we don't want lead1 called for order2 |
| 03:37.28 | titter | rnp: need to set some sort of algorithm ... honestly can't asnwer that one ... I try to stay out of the database grunt work |
| 03:37.42 | rnp | gottcha ;) |
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| 03:48.08 | jmcdowell | wth |
| 03:48.09 | jmcdowell | transport error: Curl Error strings have been compiled out. |
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| 04:06.37 | jmcdowell | I thought that the bootrom would only update once.. |
| 04:06.41 | jmcdowell | it's updating over and over and over. |
| 04:10.36 | [TK]D-Fender | jmcdowell: seems like twice often... |
| 04:13.27 | voipmonk | http://www.dialapplet.com/ |
| 04:13.43 | voipmonk | rnp - here's another |
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| 04:21.07 | jmcdowell | Well |
| 04:21.20 | jmcdowell | I had to remove the bootrom.ld or it just keeps updating |
| 04:21.25 | b14ck | Hi everyone. |
| 04:21.31 | jmcdowell | but it did update, it did provision with blank settings |
| 04:21.31 | b14ck | I'm back on my primary nick, yey ^^ |
| 04:21.47 | jmcdowell | now I just have to figure out those crazy config files. |
| 04:21.47 | jmcdowell | :D |
| 04:22.49 | b14ck | Fixing some polycom stuff? |
| 04:23.06 | b14ck | Hrm, your nick is really familiar. Do you use the trixbox forums by any chance? |
| 04:24.48 | *** join/#asterisk pawz (~pawz@ppp118-208-187-51.lns20.bne4.internode.on.net) |
| 04:24.58 | jmcdowell | I have in the past |
| 04:25.09 | jmcdowell | and yes I am "trying to figure out" the polycom provisioning stuff.. |
| 04:25.23 | jmcdowell | I just got the firmware and bootrom upgraded on one phone. |
| 04:25.33 | jmcdowell | now I am reading about the config files setup |
| 04:25.50 | b14ck | Ah cool. |
| 04:27.52 | jmcdowell | omfg these config files suck |
| 04:28.17 | b14ck | heh |
| 04:28.20 | b14ck | They aren't too bad =p |
| 04:28.36 | jmcdowell | what ever |
| 04:30.23 | jmcdowell | so is there some way I can get the " |
| 04:30.28 | jmcdowell | things I want to change |
| 04:30.39 | b14ck | Just use searching in your text editor. |
| 04:30.45 | b14ck | I usually use vim |
| 04:30.47 | jmcdowell | generated into a file, rather than hack my way through that nasty xml config |
| 04:30.51 | b14ck | Just type ESC + /<search term> |
| 04:30.57 | jmcdowell | right but I only need a few thing |
| 04:31.04 | jmcdowell | I just pluck then right out huh? |
| 04:31.06 | b14ck | ya, gonna have to change it manually :x |
| 04:31.16 | jmcdowell | So.... |
| 04:31.19 | jmcdowell | lets see |
| 04:31.53 | jmcdowell | server voIpProt.server.1.address="" <-- I can just drop that into the file without the xml formatting? |
| 04:32.04 | jmcdowell | or do I have to format the xml stuff along with it ? |
| 04:32.13 | b14ck | have to format the xml stuff |
| 04:32.23 | b14ck | XML standards dont allow for non-correct syntax |
| 04:32.27 | *** join/#asterisk aidinb (~Aidin@32.172.150.85) |
| 04:32.38 | jmcdowell | <server voIpProt.server.1.address="xxx.com" /> |
| 04:32.41 | jmcdowell | like that? |
| 04:32.52 | b14ck | ya but it has to be in the right place too |
| 04:33.19 | jmcdowell | the right place? |
| 04:34.13 | jmcdowell | like this <server voIpProt.server.1.address="" /> |
| 04:34.14 | jmcdowell | ? |
| 04:34.26 | b14ck | no i mean |
| 04:34.29 | b14ck | in the file |
| 04:34.32 | b14ck | in your config file |
| 04:34.37 | b14ck | It has to be in the correct section |
| 04:34.43 | b14ck | Don't you have a config file already? oO |
| 04:34.47 | jmcdowell | So I am under the impression that I only put what I wanat to change in the file. |
| 04:34.49 | b14ck | That you can search/replace? |
| 04:34.57 | jmcdowell | I have the 000000000000.cfg file |
| 04:35.02 | jmcdowell | I am making this from scratch |
| 04:35.03 | b14ck | ya |
| 04:35.09 | b14ck | don't make it from scratch |
| 04:35.16 | b14ck | just copy the 0000 one and then make changes where necessary |
| 04:35.21 | jmcdowell | Right, but every phone can't be provisioned the same way |
| 04:35.28 | b14ck | correct |
| 04:35.33 | jmcdowell | I think I got it.. |
| 04:35.35 | jmcdowell | hang on. |
| 04:39.34 | jmcdowell | is there a reference anywhere as to what all this sh** means? |
| 04:42.29 | rnp | anyone know of any companies that offer good xen virtual servers? |
| 04:44.20 | b14ck | rnp, slicehost |
| 04:44.23 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:44.31 | b14ck | jmcdowell, the polycom guide? I'm not sure. I just learned it by looking :x |
| 04:44.40 | rnp | i'll check it out thanks |
| 04:45.19 | b14ck | rnp, if you're looking for something cheap (production) check out mediatemple, they are also pretty good |
| 04:46.12 | rossand | transfer and call park take over # for me right now. What do people normally do when they want # - say for external voicemail, conference systems? Just curious if there's a convention for this already? Thanks. |
| 04:46.57 | voipmonk | u can setup asterisk to be your external voicemail system |
| 04:46.59 | voipmonk | or conference system |
| 04:47.05 | voipmonk | sure |
| 04:47.17 | b14ck | sup voipmonk |
| 04:47.23 | voipmonk | hay lo |
| 04:48.50 | rossand | voipmonk: what I'm curious about is what do people do when they have no choice but to dial into webex or such? If I enable transfer and call park, then # takes priority for them. I could solve this a few ways but I'm curious what others do. Thanks again. |
| 04:49.16 | voipmonk | what is # ? |
| 04:49.31 | voipmonk | # takes priority for them? |
| 04:49.45 | rossand | for example, to dial into my company's conference bridge system, you dial the toll free number, then enter the passcode and then #. |
| 04:49.52 | voipmonk | ok |
| 04:50.00 | rossand | If I enable transfer and call park, hitting # sends me into transfer. |
| 04:50.09 | rossand | Did I explain it well enough? |
| 04:50.11 | voipmonk | so disable that |
| 04:50.14 | voipmonk | :) |
| 04:50.24 | rossand | :-) That's what I've done. |
| 04:50.30 | voipmonk | ok.. |
| 04:50.34 | voipmonk | next question |
| 04:50.36 | voipmonk | :) |
| 04:50.48 | rossand | technically correct answer, but unhelpful ;-) |
| 04:51.21 | voipmonk | its unhelpful for me to agree with what you've done? |
| 04:51.29 | rossand | I suppose the best solution may be to tweak my dialplan to use a Dial rule without the options. |
| 04:51.33 | voipmonk | Im starting to see a patern for asterisk questions |
| 04:52.03 | rnp | b14ck: do you think slicehost is good, it's a bit cheaper for the same offering. |
| 04:52.38 | b14ck | rnp, slicehost is great in my experience |
| 04:52.46 | b14ck | They are also really easy to setup (and automatic) |
| 04:52.51 | b14ck | Only takes a minute to get started. |
| 04:53.05 | rossand | voipmonk: let me say this another way. Is there a common way others have kept transfer enabled, yet not interfered with being able to hit # when dealing with external vm, conferencing, etc? |
| 04:53.05 | b14ck | You can format / re-provision your slice in a minute or so, really handy. |
| 04:53.47 | rnp | nice |
| 04:54.17 | [TK]D-Fender | rossand: what kind of phones are you using? |
| 04:54.38 | rossand | [TK]D-Fender: various - twinkle, xlite, snom, etc. |
| 04:55.10 | rossand | [TK]D-Fender: Thanks again for the dial() tip earlier. Got recording working perfect. The Mrs. is very pleased as she can do work from home. |
| 04:55.16 | [TK]D-Fender | rossand: With the execption of X-Lite, the rest have native transfer capabilities, etc.... o stop using douchebag DTMF call features :p |
| 04:57.00 | rnp | anyone have any idea what kind of sys specs i need for Vici on a virtual server? |
| 04:57.16 | [TK]D-Fender | rnp: Details might help |
| 04:57.32 | rossand | [TK]D-Fender: OK, fair enough. Thanks again. |
| 04:59.35 | rnp | Fender: sorry, what details do you need? |
| 04:59.42 | rnp | Vici with maybe 20 callers on it at one time max |
| 05:00.20 | b14ck | you'll need some decent bandwidth and stuff for that |
| 05:00.29 | b14ck | decent cpu too |
| 05:01.33 | rnp | well for $38 a month I can get 512 ram 20gb hd and 200gb bandwith |
| 05:01.41 | rnp | i can't imagine using up more bandwith than that |
| 05:03.18 | [TK]D-Fender | 20 calls isn't an issue. What would be an issue is transcoding & call recording |
| 05:03.22 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-galxjzzanhreqivl) |
| 05:03.43 | *** join/#asterisk [netman] (~netman@147.Red-83-63-246.staticIP.rima-tde.net) |
| 05:03.49 | rnp | well all call recording is done on the agent's pc |
| 05:03.49 | *** join/#asterisk brian (~brian@unaffiliated/brian) |
| 05:03.54 | rnp | and my guess transcoding too |
| 05:04.01 | rnp | before uploading to the server |
| 05:05.16 | rnp | so do you think a 256meg + 10gb would be good enough? |
| 05:06.21 | [TK]D-Fender | rnp: No, transcoding is done on the server if needed |
| 05:06.53 | rnp | Fender: I see |
| 05:07.06 | [TK]D-Fender | rnp : ram is fine, HD as long as it fits. |
| 05:07.22 | *** join/#asterisk helloritesh (~hellorite@121.243.172.188) |
| 05:09.42 | rnp | Fender: so what do you do? |
| 05:09.52 | [TK]D-Fender | rnp: in terms of? |
| 05:11.36 | rnp | Fender: your business, do you run a call center or own one or? |
| 05:12.33 | [TK]D-Fender | rnp: Full time IT position, and freelance * consultant |
| 05:12.56 | rnp | Fender: nice, what type of freelance do you do? |
| 05:13.20 | [TK]D-Fender | SMB * deployment, custom IVR work, etc |
| 05:14.42 | rnp | Can you tell me what that means in layman's terms? |
| 05:14.56 | [TK]D-Fender | rnp: ... those ARE laymans terms... |
| 05:15.13 | rnp | haha, ah, I see |
| 05:15.19 | [TK]D-Fender | ~ivr |
| 05:15.20 | infobot | ivr is probably Interactive Voice Response |
| 05:15.28 | rnp | i looked up IVR |
| 05:15.31 | rnp | but couldn't find SMB |
| 05:15.37 | [TK]D-Fender | SMB : small/medium sized Business |
| 05:15.40 | rnp | ah |
| 05:16.05 | jmcdowell | does anyone know the command or what I would look for to reverse ssh in such a way that I can access the web interface for a freepbx box? |
| 05:16.08 | rnp | ok wait, so this IVR, could be used for voicecast? |
| 05:16.51 | *** join/#asterisk jakent (~john@c-98-233-13-157.hsd1.va.comcast.net) |
| 05:17.49 | [TK]D-Fender | rnp: No, I'm talking voice menus. "Press 1 for, etc". Including things like DB lookups, sever maintenance menues, funky bits. |
| 05:18.23 | [TK]D-Fender | [00:18]<coopbot>porttrick: to forward a port on your local machine to a remote system's port use ssh -p 22 -N -f -L 80:127.0.0.1:80 root@host (example would make your localhost port 80 show the remote system's web server without exposing it to the world) |
| 05:18.26 | [TK]D-Fender | jmcdowell: ^^^6 |
| 05:18.44 | rnp | Fender: Ok, I have a question for you. |
| 05:18.55 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 05:19.08 | [TK]D-Fender | 42 |
| 05:19.39 | p3nguin | This is ridiculous that a Cisco 7940G won't cooperate with type=peer, but every other phone I can get my hands on will. |
| 05:19.51 | rnp | Could you setup an automated voice cast to businesses, that called a business and asked them "Would you be willing to donate space for a candy machine that supports XYZ Charity? If so Press 1." IF YES - Please say your first and last name. For drop off in the morning Press 1 or for evening Press 2 |
| 05:19.58 | rnp | can you do stuff like that? |
| 05:20.29 | [TK]D-Fender | rnp: Yes.. that is an automated call-out polling setup. |
| 05:20.41 | [TK]D-Fender | rnp: The kind of things most people despise being called by |
| 05:21.32 | rnp | yeah |
| 05:21.34 | p3nguin | Just call them normally instead. |
| 05:21.38 | rnp | but it is basically free to make the calls |
| 05:21.47 | [TK]D-Fender | rnp: Pardon? |
| 05:21.53 | rnp | Fender: do you have any idea what kind of response rate you would get? |
| 05:21.59 | [TK]D-Fender | rnp: No clue |
| 05:22.05 | p3nguin | Found no matching peer or user for '192.168.192.26:50977' |
| 05:22.07 | rnp | Fender: waht I mean by basically free is that you don't have to pay staff to make the calls |
| 05:22.17 | [TK]D-Fender | rnp: True |
| 05:22.17 | p3nguin | no kidding, since my peer is named '201' |
| 05:22.49 | [TK]D-Fender | p3nguin: No, that isn't showing you the name it's ID'ing as |
| 05:23.10 | [TK]D-Fender | p3nguin: p3nguin its just showing a trace of where the attempt came from |
| 05:23.33 | rnp | Fender: how much would it cost to set something like that up? |
| 05:23.35 | p3nguin | Every other phone or ITSP matches the correct peer name that is within the [] in sip.conf. |
| 05:24.39 | p3nguin | For example: |
| 05:24.41 | p3nguin | Found peer 'sipgate' |
| 05:25.02 | [TK]D-Fender | p3nguin: Can't say 201 since it can't ID as 201 |
| 05:25.15 | [TK]D-Fender | rnp: Depends on the complexity |
| 05:25.15 | p3nguin | Found peer 'ipkall0' |
| 05:26.01 | p3nguin | Found peer '211' |
| 05:26.16 | p3nguin | Every other ITSP or phone... but not this 7940G for some reason. |
| 05:26.56 | p3nguin | The solution is to leave it as type=friend or if I want to make it a peer, I have to set insecure=port. |
| 05:27.26 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 05:27.36 | p3nguin | It makes calls as type=user, too, but of course that doesn't allow for it to receive calls... so that's out. |
| 05:28.32 | p3nguin | And since it's a "peer" the username= field is IRRELEVANT. |
| 05:28.36 | *** join/#asterisk bsdbandit (~csh11@wsip-24-249-123-207.hr.hr.cox.net) |
| 05:28.48 | bsdbandit | has anyone compiled asterisk on solaris 10 x86 |
| 05:29.17 | p3nguin | bsdbandit: Maybe for novelty purposes. |
| 05:32.32 | [TK]D-Fender | ok, its getting late... off to bed. Back tomorrow |
| 05:32.53 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 05:32.54 | *** mode/#asterisk [+o file] by ChanServ |
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| 05:33.28 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 05:44.12 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 05:57.23 | dlynes_laptop | p3nguin, so did you ever figure out what's changed for 'type=friend' in asterisk 1.6? |
| 06:01.40 | carrar | google it |
| 06:01.44 | carrar | You'll find the answer |
| 06:05.56 | *** join/#asterisk lordmortis (~lordmorti@203.206.51.27) |
| 06:07.24 | timahvo1 | anyone have any expereience running asterisk as a xen server guest ? |
| 06:07.59 | timahvo1 | asterisknow to be more accurate |
| 06:08.23 | carrar | should work fine |
| 06:08.32 | *** join/#asterisk helloritesh (~hellorite@121.243.172.188) |
| 06:09.37 | timahvo1 | well am using a digium card and I can't seem to detect it, top it all off lspci returns nothing , wondering what I might be doing wrong |
| 06:09.58 | carrar | works better with just SIP |
| 06:10.33 | carrar | use a dedicated serer for T1 to SIP |
| 06:10.43 | carrar | or media gw |
| 06:12.34 | timahvo1 | ok |
| 06:16.35 | dlynes_laptop | timahvo1, don't try using hardware in a VM |
| 06:16.47 | dlynes_laptop | timahvo1, especially not tdm hardware |
| 06:17.07 | dlynes_laptop | timahvo1, it's just asking for something to not work, or not work correctly |
| 06:19.49 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 06:28.08 | *** join/#asterisk oej (~olle@ns.webway.se) |
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| 06:36.05 | p3nguin | dlynes_laptop: Yeah, I thought I mentioned what I could find about it. I'm not sure which version it has changed in, but they are changing friend from being two objects in memory to being only a single object. |
| 06:36.52 | dlynes_laptop | p3nguin, so how does that affect the behaviour from a software perspective? |
| 06:37.09 | dlynes_laptop | p3nguin, you've told me how it's changed from a low level perspective |
| 06:37.14 | p3nguin | And from memory, I think I read something about friend was going to be the same as peer (for compatibility reasons), but that could be way off. |
| 06:38.00 | dlynes_laptop | kinda makes you wonder why they even introduced type=friend to begin with, then |
| 06:38.05 | p3nguin | I'm not a dev and I rarely read the lists, but I know I read about it. |
| 06:38.08 | dlynes_laptop | type=friend never used to exist |
| 06:40.05 | ChannelZ | The asterisk book talks about it, mentions that friend is supposed to go away.. perhaps that's what you remember reading |
| 06:40.15 | p3nguin | friend was a good idea, I think. It combined the username matching on inbound calling (aspect of user) with the ability to have outbound calls from dialplan (aspect of peer). |
| 06:40.45 | p3nguin | There has also been talk of type=service for service providers. |
| 06:41.43 | dlynes_laptop | p3nguin, now that's just getting ridiculous |
| 06:41.57 | dlynes_laptop | p3nguin, that would create lots of confusion |
| 06:43.33 | carrar | line 23761 of chan_sip.c still shows it 3 types |
| 06:43.42 | carrar | 1.6.2.2 |
| 06:44.01 | dlynes_laptop | carrar, p3nguin's not saying it's not still there |
| 06:44.08 | dlynes_laptop | carrar, just that it's changed behaviour |
| 06:44.43 | p3nguin | I don't have dates on the stuff, so it could be an old plan that is in effect and we don't even realize it. |
| 06:45.20 | oej | Any issues with the types? |
| 06:45.25 | oej | Just dived in |
| 06:45.27 | oej | :-) |
| 06:45.43 | p3nguin | Heh, I'm looking at something you wrote about it a year ago. |
| 06:46.00 | p3nguin | oej: What's the status on type=friend currently? |
| 06:46.06 | oej | A big mess |
| 06:46.27 | oej | We had two user objects in memory and I cleaned that up and have only one object type in the code |
| 06:46.41 | p3nguin | Is it now basically a duplicate of peer? |
| 06:46.51 | oej | So the allocation of objects is very different, the function should be exactly the same as before |
| 06:47.11 | oej | Type=friend is a user and a peer |
| 06:47.20 | oej | A user matches on the name on incoming calls |
| 06:47.21 | dlynes_laptop | p3nguin, there ya go...like i said...just because it's one way at a low level, has absolutely no bearing on what it is at a high level |
| 06:47.26 | p3nguin | Okay, so that really hasn't changed, then. |
| 06:47.35 | oej | A peer matches on ip/port on incoming calls |
| 06:47.35 | oej | a friend matches first on name, then on peer/ip as before |
| 06:47.57 | oej | I am not saying that it hasn't changed, but if it has, it's a bug ;-) |
| 06:48.00 | dlynes_laptop | oej, and so that basically means that blf should work much more predictably on type=friend now? |
| 06:48.11 | oej | Yes, absolutely |
| 06:48.19 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 06:48.27 | oej | type=user was never used before for subscriptions |
| 06:48.31 | oej | ONly type=peer |
| 06:48.32 | p3nguin | I guess I correctly recalled the memory aspect of it, but I certainly could not remember how it changed configuration of things. |
| 06:48.40 | oej | That's why I had to hack that "limitonpeer" stuff |
| 06:48.54 | oej | So that's not needed any more |
| 06:49.00 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 06:49.09 | oej | We have three new types coming up to replace this mess |
| 06:49.13 | carrar | no one reads the changelog anymore :) |
| 06:49.18 | oej | type=phone for basic phones |
| 06:49.25 | oej | ...which is much like friend |
| 06:49.42 | oej | type=service when Asterisk emulates a phone and registers externally for a service |
| 06:49.51 | oej | which is the combination of register= and a peer |
| 06:50.10 | oej | type=trunk for when we exchange traffic mutually based on DNS or IP |
| 06:50.12 | p3nguin | Will that do away with register statements? |
| 06:50.18 | oej | which is what people have been using type=peer |
| 06:50.35 | oej | Yes, the register statements will not be needed for service |
| 06:50.38 | dlynes_laptop | oej, yeah...that limitonpeer was pretty nasty, and didn't solve the issues 100% of the times, either |
| 06:50.51 | oej | they're overloaded, messed up and not very useful today. Too many poor hacks. |
| 06:51.09 | oej | That's strange - I haven't got that feedback on limitonpeer |
| 06:51.32 | p3nguin | Will any of these changes make their way down to 1.4, or only in new 1.6 and trunk? |
| 06:51.34 | dlynes_laptop | oej, well, my problems could have been caused by buggy firmware on the phones, too |
| 06:51.53 | dlynes_laptop | oej, I haven't attempted to do any blf or sla for about 3 years now |
| 06:52.01 | oej | I try to backport as much as possible to 1.4 - you guys propably have seen that |
| 06:52.08 | oej | I don't care much for 1.6.x |
| 06:52.14 | dlynes_laptop | oej, why's that? |
| 06:52.14 | oej | So possibly 1.4 and trunk |
| 06:52.17 | carrar | blf works fine for all my clients |
| 06:52.34 | dlynes_laptop | oej, I find 1.6.1 is extremely stable, compared to previous versions |
| 06:52.55 | oej | I work mostly with large scale installations. The 1.6.x release policy was too messy and chan_sip was messed up. |
| 06:53.15 | oej | Well, 1.6.x won't be maintained as long as we've maintained 1.4 |
| 06:53.29 | oej | So it's going to be hard to put it in production. |
| 06:53.33 | dlynes_laptop | oej, ah...so 1.8 will have a better release policy then? |
| 06:53.38 | drmessano | 1.8 LTS ! |
| 06:53.46 | carrar | FTW |
| 06:53.47 | oej | Yes, 1.8 and 1.4 is now classified as long term support releases |
| 06:54.03 | dlynes_laptop | drmessano, you sound like a ubuntu evangelist :) |
| 06:54.12 | p3nguin | I'm pretty happy with 1.4. |
| 06:54.25 | drmessano | lol |
| 06:54.28 | oej | I hope to be able to get funding to fix the TCP/TLS support for 1.8 so we really can use it. |
| 06:54.41 | dlynes_laptop | p3nguin, well, i was until i was forced to support faxing |
| 06:54.44 | oej | Digium messed up, but refuse to fix it. |
| 06:54.47 | dlynes_laptop | p3nguin, then i dove into 1.6 |
| 06:54.56 | oej | There's no 1.6 |
| 06:54.59 | p3nguin | dlynes_laptop: I'm doing okay with FFA. |
| 06:55.09 | dlynes_laptop | p3nguin, on 1.4? |
| 06:55.12 | p3nguin | yes |
| 06:55.13 | oej | We have 1.4, 1.6.0, 1.6.1, 1.6.2, trunk |
| 06:55.19 | dlynes_laptop | p3nguin, but you're not using t.38, right? |
| 06:55.19 | oej | They are all very different |
| 06:55.26 | oej | As a developer, I have to support all these versions |
| 06:55.30 | p3nguin | dlynes_laptop: Right, no t.38. |
| 06:55.42 | dlynes_laptop | oej, yeah...by 1.6, I meant 1.6.0/1.6.1/1.6.2 |
| 06:56.09 | p3nguin | dlynes_laptop: SIP and ulaw, ReceiveFAX() ... seems to work fine. |
| 06:56.11 | oej | Yes, but you can't compare "1.6" with "1.4" |
| 06:56.13 | dlynes_laptop | oej, i haven't touched 1.6.0, and i"ve heard it's pretty much a crap shoot as to which minor revisions will crash and which ones won't |
| 06:56.28 | oej | 1.6.0 is a nightmare |
| 06:57.04 | dlynes_laptop | p3nguin, yeah...I need to use t.38; that's why i'm using 1.6.1 |
| 06:57.42 | dlynes_laptop | p3nguin, way too much jitter and latency to use straight ulaw, and tdm/pri is not an option |
| 06:57.55 | dlynes_laptop | p3nguin, we're using a VOIP PRI |
| 06:58.12 | carrar | VOIP PRI? |
| 06:58.14 | p3nguin | How's that work? |
| 06:58.18 | carrar | You mean a sip provider |
| 06:58.25 | dlynes_laptop | Yeah..that's what they call their voip service |
| 06:58.26 | p3nguin | Virtual PRI? |
| 06:58.45 | carrar | I love these terms :) |
| 06:58.47 | dlynes_laptop | It's basically 12 to 24 channels, on a dedicated leased network |
| 06:59.03 | dlynes_laptop | So almost zero jitter |
| 06:59.27 | carrar | SIP on a pl |
| 06:59.29 | p3nguin | Why do you have to use T.38? |
| 06:59.31 | dlynes_laptop | pl? |
| 06:59.32 | carrar | private line |
| 06:59.34 | dlynes_laptop | yes |
| 06:59.39 | carrar | clear channel T1 |
| 06:59.51 | dlynes_laptop | carrar, Not really |
| 06:59.55 | dlynes_laptop | carrar, E10 |
| 07:00.18 | carrar | they should call it as it is :) |
| 07:00.47 | carrar | Magic VoIP |
| 07:00.50 | carrar | Now on sale |
| 07:00.56 | dlynes_laptop | p3nguin, because we're sending out fax calls to our upstream that they know are fax, and so they send out long distance calls on t.38 |
| 07:01.11 | dlynes_laptop | p3nguin, all incoming voip calls that are fax come in as t.38 as well |
| 07:01.57 | carrar | What is a E10 anyways |
| 07:02.01 | p3nguin | I don't get it. If I can send and receive faxes without T.38, you should be able to also. |
| 07:02.07 | dlynes_laptop | p3nguin, all of our sip 'trunks' to them actually terminate to PRI's on a Metaswitch for local calls |
| 07:02.20 | dlynes_laptop | carrar, 10Mbit ethernet |
| 07:02.39 | carrar | they do QoS over that? |
| 07:02.47 | dlynes_laptop | carrar, yes |
| 07:02.58 | carrar | nice |
| 07:02.59 | dlynes_laptop | carrar, some facilities use Cisco, some use Juniper |
| 07:03.14 | carrar | both are excellent vendors |
| 07:03.41 | dlynes_laptop | carrar, our equipment all runs inside of their colo, so there's no inter-network jitter for local calls |
| 07:04.23 | dlynes_laptop | p3nguin, i don't get how you're able to do pure ulaw for faxing, and not have any problems |
| 07:04.25 | *** join/#asterisk Tim_Toady (~moi@77.49.162.92.dsl.dyn.forthnet.gr) |
| 07:04.41 | dlynes_laptop | p3nguin, i've never been able to get pure ulaw working properly for faxing.....ever |
| 07:05.23 | p3nguin | dlynes_laptop: I don't need to fax much, so I guess the failure rate hasn't caught up to me yet. |
| 07:05.30 | carrar | p3nguin, does ulaw faxing in a lab only |
| 07:05.34 | carrar | heh |
| 07:05.49 | carrar | with nothing else on the network and 1 page faxes |
| 07:05.56 | dlynes_laptop | p3nguin, even doing two test faxes, I had one fail 3/4's of the way through the first page, and the other one went fine |
| 07:06.09 | dlynes_laptop | p3nguin, then i tried three more faxes, and they all fail on the first page |
| 07:06.48 | dlynes_laptop | p3nguin, I get about a 0.5% success ratio for 1 page faxes, and 0% success ratio for anything more than 1 page |
| 07:07.02 | p3nguin | dlynes_laptop: I can't ever get anyone to just send me a stack of pages, so I don't really know what type of failure rate to expect. |
| 07:07.33 | dlynes_laptop | p3nguin, that being said, I get about 80% success rate for faxing over PRI |
| 07:07.43 | p3nguin | I've received at least three good pages since I started using FFA a few days ago. |
| 07:07.53 | dlynes_laptop | p3nguin, so, based on that, I can't see you ever getting 100% success rate on SIP |
| 07:08.16 | dlynes_laptop | p3nguin, ffa uses t.38 |
| 07:08.18 | p3nguin | It's at 100% so far. Send me 100 pages and I'll tell you how many of them are good. |
| 07:08.26 | dlynes_laptop | p3nguin, or wait...you're on 1.4, so it doesn't |
| 07:08.29 | p3nguin | right |
| 07:09.16 | dlynes_laptop | p3nguin, try doing a multiple page transmission (say 8 pages in one transmission) |
| 07:09.35 | p3nguin | I don't really have anyone to send eight pages to. |
| 07:09.46 | dlynes_laptop | p3nguin, send 8 pages to yourself |
| 07:10.06 | p3nguin | I ... don't think that'll work. |
| 07:10.13 | carrar | heh |
| 07:10.23 | p3nguin | I have a free FFA single channel license. |
| 07:10.26 | dlynes_laptop | p3nguin, what service are you using? |
| 07:10.31 | p3nguin | sipgate |
| 07:10.33 | dlynes_laptop | for your sip line, that is? |
| 07:10.42 | dlynes_laptop | oh....do they have a faxing interface? |
| 07:10.50 | p3nguin | Yes, but I'm using SIP instead. |
| 07:11.02 | Madkiss | shut his asterisk up. fullstrike. |
| 07:11.07 | dlynes_laptop | No...I meant a web form where you can upload a file to fax |
| 07:11.11 | dlynes_laptop | And it'll fax it for you |
| 07:11.23 | p3nguin | Maybe. I'll look. |
| 07:11.31 | dlynes_laptop | I know vitelity has that |
| 07:15.56 | p3nguin | Looks like they have it. |
| 07:16.12 | p3nguin | You think a single PDF with multiple pages would be good enough? |
| 07:21.42 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rsvwbrsaqtlmtnxx) |
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| 07:24.07 | dlynes_laptop | p3nguin, yep |
| 07:24.45 | dlynes_laptop | p3nguin, if you want to test a multipage fax from your fax machine, you can try faxing to 604-628-0026 |
| 07:25.05 | p3nguin | It's either busted or not compatible with konqueror, 'cause it won't update my 7-page PDF. |
| 07:25.06 | dlynes_laptop | p3nguin, i.e. from your fax on box |
| 07:25.15 | dlynes_laptop | update your 7 page pdf? |
| 07:25.27 | p3nguin | upload |
| 07:25.37 | dlynes_laptop | ah |
| 07:25.46 | p3nguin | finger-brain disconnection |
| 07:25.54 | dlynes_laptop | a lot of these website designers only test with internet exploder |
| 07:26.04 | dlynes_laptop | and some of them don't even test, period |
| 07:27.30 | p3nguin | sipgate.com crashes epiphany-webkit instantly, so I'm basically SOL. |
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| 07:28.43 | p3nguin | For my outbound faxes using SendFAX(), I send them over IAX2 using ulaw. |
| 07:33.30 | dlynes_laptop | oh...so you're just sending them to hylafax using iaxmodem? |
| 07:34.09 | p3nguin | negative |
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| 07:34.40 | dlynes_laptop | oh...you talk to sipgate using iax2? |
| 07:34.47 | p3nguin | negative |
| 07:35.11 | p3nguin | sipgate gives me SIP/ulaw and a DID, where I receive faxes. |
| 07:35.38 | p3nguin | I use VoIP.ms for termination services. |
| 07:36.36 | p3nguin | I talk to VoIP.ms with IAX2, and that includes outgoing fax calls with SendFAX(). |
| 07:38.30 | p3nguin | If I fax single pages to HP's faxback system, they get it and fax me back. Now I can't attest to the quality of the page they got from me, but I believe that it has to be received entirely in order for them to return a fax. |
| 07:41.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
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| 07:43.22 | joobie | hey guys.. i have a sip peer setup that i want to only route 1 call through at any given time.. i currently use 'call-limit=1' in the sip.conf.. it works fine, but when in the asterisk console i see an ERROR notification coming up when the line is in use already, saying "Call to peer rejected due to usage limit of 1" ... which is good, but just wondering if there's a better way to do this, so that i don't get an ERROR event per say? |
| 07:44.51 | dlynes_laptop | p3nguin, correct |
| 07:44.53 | p3nguin | dlynes_laptop: Well, I can't fax that document to myself... Trial 3: Credit insufficient |
| 07:44.59 | dlynes_laptop | p3nguin, heh |
| 07:45.23 | p3nguin | Firefox uploaded it, though, so I guess they just suck at writing web sites. |
| 07:48.16 | p3nguin | If you've got about 10 pages of useless documentation that you can fax to me, that would be cool. |
| 07:52.32 | kaldemar | joobie: group functions. core show functions like GROUP. call-limit is deprecated and will be removed. |
| 07:59.47 | dlynes_laptop | does anyone if lang/hebrew.ods is available in english? |
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| 08:13.38 | benngard | i have been struggling with an avaya 9650 sip phone, everyhing works except name of caller on display, i just see the number, i am pretty sure the phone supports it so i wonder if any have an idea what can be wrong, if i register for example a x-lite to the same extension i get name of caller... |
| 08:14.35 | ChannelZ | You can set the caller ID to anything you want in sip.conf |
| 08:14.51 | benngard | i know |
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| 08:15.31 | dlynes_laptop | benngard, maybe avaya 9650 sip phone doesn't support caller id name |
| 08:15.58 | *** join/#asterisk icyValk77 (~icyValk77@cl-670.lon-02.gb.sixxs.net) |
| 08:16.01 | benngard | dlynes_laptop: i have been thinking about it, but the documentations claims it does |
| 08:16.26 | dlynes_laptop | benngard, then it's probably a setting you've got set on the phone that's overriding it |
| 08:17.17 | dlynes_laptop | anyways...g'night |
| 08:17.19 | dlynes_laptop | need sleep |
| 08:18.21 | benngard | when the phoen starts up, it reads a file 46xxsettings.txt which contains "SET DISPLAY_NAME_NUMBER 1" ## If it is set to 1 then phone will display name and number of incoming call. |
| 08:19.31 | ChannelZ | Are you sure the call it's receiving has both? |
| 08:22.23 | benngard | if i unplug the avaya and connect a x-lite to the same extension, i see both name and number on x-lite display |
| 08:23.59 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 08:25.32 | benngard | which sip-message is it that triggers the display, gonna do tcpdump and examine the package with wireshark, but unsure whick sip-message i gonna look for |
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| 08:35.50 | benngard | i am looking at the sip session, is it i have name and number in both 0317998985 calls 975: SIP/2.0 100 Trying |
| 08:36.04 | benngard | From: "Cecilia Benngard" <sip:0317998985@inputinterior.se>;tag=882285896 |
| 08:36.04 | benngard | To: <sip:975@inputinterior.se;user=phone |
| 08:36.26 | benngard | but the display on 975 shows only 0317998985 |
| 08:37.08 | benngard | or what sip message should i look for?r |
| 08:37.43 | kaldemar | it will have the same From-header for the whole dialog. your phone is getting the information, it's just not showing it. |
| 08:39.25 | kaldemar | btw, asterisk has sip debug to print all messages, you don't need to use tcpdump or wireshark. help sip set debug in the CLI. |
| 08:39.56 | *** join/#asterisk pawz (~pawz@ppp118-208-65-148.lns20.bne4.internode.on.net) |
| 08:40.18 | benngard | kaldemar: i know, but sometimes i prefer to watch it in wireshark |
| 08:41.50 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.196.125) |
| 08:42.48 | benngard | hmm... INVITE sip:0317998975@90.230.92.67:46474;avaya-sc-enabled;transport=udp SIP/2.0, avaya-sc-enabled wonder if the phone think it talks to an avaya sip server and do some strange things |
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| 10:23.02 | aeperezt | helo |
| 10:23.17 | aeperezt | any tip agains this issue chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
| 10:27.22 | benngard | yay manage to get name and number of caller on my avaya :) |
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| 10:28.20 | benngard | and connected line information works to |
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| 10:29.49 | benngard | rpid_update=yes, sendrpid=pai, trustrpid=yes did the trick for me! :) |
| 10:31.39 | cusco_ | hi |
| 10:33.02 | cusco_ | astmanproxy replies: missing action in request... |
| 10:33.09 | cusco_ | anyone knows? |
| 10:35.15 | *** join/#asterisk Korolev (BlasRoca@c-75-74-122-15.hsd1.fl.comcast.net) |
| 10:35.21 | TommyBotten | Using asterisk 1.6.2 - is there a way to make the 'h' extension make an outgoing call, even if the channel is dead? I'd prefer not using deadagi/AGI |
| 10:38.02 | nix8n82 | if the channel is dead, what would you connect the call to? |
| 10:41.52 | *** join/#asterisk sergey (~sergey@ua0zeh.iks.ru) |
| 10:43.57 | TommyBotten | nix8n82: A playback ... using a local channel perhaps? |
| 10:47.12 | ktwilight[m] | am looking into migrating our current pbx system to asterisk. however, the lack of documentation on our current pbx is quite a hindrance. does anyone have any advise in what to look for to move forward with it? |
| 10:51.17 | *** join/#asterisk MAbbas (Jinbaba@115.186.15.50) |
| 10:51.29 | kaldemar | TommyBotten: core show application originate |
| 10:52.27 | MAbbas | Hi everyone, I am getting AMI connection refuse error .. here is my manager.conf http://pastebin.com/m13f37c65 |
| 10:53.13 | MAbbas | root@muhammad-desktop:/etc/asterisk# telnet 192.168.153.129 5038 |
| 10:53.13 | MAbbas | Trying 192.168.153.129... |
| 10:53.13 | MAbbas | telnet: Unable to connect to remote host: Connection refused |
| 10:55.30 | MAbbas | <kaldemar> ? |
| 10:56.09 | TommyBotten | kaldemar: It's not available... Do you know where I can get it? |
| 10:56.28 | TommyBotten | kaldemar: As in - which module? |
| 10:59.53 | MAbbas | TommyBotten: try manager show command Originate |
| 11:00.01 | MAbbas | "manager show command Originate" |
| 11:00.39 | TommyBotten | MAbbas: Thanks :-) |
| 11:00.48 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 11:04.47 | MAbbas | <MAbbas> Hi everyone, I am getting AMI connection refuse error .. here is my manager.conf http://pastebin.com/m13f37c65 <- anybody? |
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| 11:10.43 | kaldemar | TommyBotten: app_originate.so. it is in 1.6.2. |
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| 11:18.17 | *** join/#asterisk Akiraa (~Akira@92.83.172.105) |
| 11:20.35 | Akiraa | Is there a working scheme for real-time fax over IP? So far, experience with the service has been underwhelming... |
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| 11:24.32 | m0t3jl | Akiraa, as they should be, because you're trying to encode an analogue signal into digital signal and back using lossy compression ;) |
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| 11:27.33 | m0t3jl | Akiraa, I tend to use these fax2email email2fax solutions some SIP providers offer together with their SIP service. |
| 11:28.28 | m0t3jl | Akiraa, it seems even cheaper to use their solution since in many cases (especially for orders and stuff) you don't even have to print those faxes. |
| 11:28.34 | Akiraa | m0t3jl: so far I would only recommend some form of multi-function fax with LAN connectivity |
| 11:28.46 | Akiraa | which sends faxes as e-mail |
| 11:29.15 | m0t3jl | Akiraa, yes, but you have to buy that thing ;) If you do not send faxes written by hand, you do not even have to buy it ;) |
| 11:29.32 | Akiraa | or if possible, a faxmodem extension card for the asterisk PC |
| 11:30.53 | Akiraa | hand written documents have to be occasionally sent |
| 11:31.26 | m0t3jl | Akiraa, yes, that was our first FAX PC (without Asterisk). It was a Debian with two faxmodem cards capable of receiving and sending faxes. Nevertheless when we realized we were just sending faxes from our IS and receiving faxes most of which did not have to be print out, we simply threw the PC away and went with our SIP providers fax2email email2fax solution ;). |
| 11:36.55 | m0t3jl | Akiraa, anyway, your questions was whether there is any working scheme for real-time fax over IP. IMHO the only working solution is either to keep some PSTN lines dedicated for your faxes, or go with the fax2email email2fax solutions ;). |
| 11:38.33 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 11:38.45 | k-man | anyone know of a good sip client for the iphone? |
| 11:40.47 | m0t3jl | Anyone has any experience with the Skype for Asterisk or any other (not officially supported) solution for SIP to Skype calls? |
| 11:44.08 | m0t3jl | What I would like to know is whether it is possible to somehow allow my users to use the ordinary Skype client on their desktops for chat and to only direct calls to my Asterisk PBX. |
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| 11:46.17 | *** join/#asterisk basty (~basty@212.218.65.195) |
| 11:46.19 | basty | Hi |
| 11:46.40 | m0t3jl | Hi |
| 11:46.43 | m0t3jl | : |
| 11:46.45 | m0t3jl | ) |
| 11:47.08 | k-man | hi |
| 11:47.17 | *** join/#asterisk devgru (~devgru@94.25.220.54) |
| 11:47.30 | basty | I am trying to install Zaptel 1.4.12 on an ubuntu 64bit system. As soon as I try to run a make i get "make[2]: *** [_module_/usr/src/zaptel-1.4.12.1/kernel] Error 2 - make[2]: Leaving directory `/usr/src/linux-source-2.6.31' - make[1]: *** [modules] Error 2 - make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1' - make: *** [all] Error 2" errors. Anyone knows why ? ;-) |
| 11:48.04 | m0t3jl | basty, seriously there's something missing from that error message ;) |
| 11:48.32 | basty | m0t3jl: lemme pastebin it...sec. |
| 11:48.33 | m0t3jl | basty, we know there were two errors, but you have not pasted any of them ;) |
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| 11:48.58 | basty | m0t3jl: see http://pastebin.com/d1a15d4ba ;-) |
| 11:49.57 | m0t3jl | basty, I hate these errors, since to me they look like an error in the sources ;) |
| 11:50.19 | m0t3jl | basty, but it's possible there is some kind of problem with some library etc. |
| 11:50.45 | basty | yeah...strange thing that..i used to install zaptel 1.4.12.1 on a debian 5.0 system - and it worked fine...so it seems that it must be something on that ubuntu system :-/ |
| 11:50.56 | tzafrir_laptop | basty, why bother with zaptel when you can install DAHDI? |
| 11:51.18 | Chainsaw | Hrm. Has anyone seen Cisco 7960G on SIP firmware P0S3-08-11-00 fail transfers? |
| 11:51.19 | tzafrir_laptop | those errors have been resolved in the SVN of Zaptel, if you really must |
| 11:51.21 | Chainsaw | -- SIP/45018-00000025 answered SIP/45007-00000024 |
| 11:51.22 | Chainsaw | <PROTECTED> |
| 11:51.22 | Chainsaw | <PROTECTED> |
| 11:51.27 | basty | tzafrir_laptop: will it work with asterisk 1.4.29 ? |
| 11:51.29 | Chainsaw | I can transfer once, just not twice. |
| 11:51.41 | tzafrir_laptop | basty, both will |
| 11:51.58 | tzafrir_laptop | basty, it's a matter of the newer kernel |
| 11:52.17 | basty | tzafrir_laptop: hrm....and i can really use dahdi in a productiv system ? we need to run 2x T1 |
| 11:52.55 | devgru | Hi, I need help to dig into RTCP. I'm getting Receiver Report with 32-bit "interarrival jitter" field, and I can't understand what units its measured in. In RFC 3550: "jitter ... measured in timestamp units", but timestamp units are 1/90000 of second for this stream, and jitter values are 10...40. |
| 11:53.02 | tzafrir_laptop | our clients sometime have problems running 4 E1-s |
| 11:53.11 | tzafrir_laptop | (4 quad E1s, that is) |
| 11:53.16 | basty | mhh okay... |
| 11:53.23 | basty | i will just check the zaptel svn then ;-) |
| 11:53.38 | tzafrir_laptop | yeah, DAHDI is the one to use. Consider DAHDI the latest version of DAHDI |
| 11:53.44 | tzafrir_laptop | of Zaptel, that is |
| 11:53.53 | basty | ok |
| 11:53.54 | m0t3jl | :) |
| 11:54.07 | basty | thanks for the help tzafrir_laptop and m0t3jl ;-) |
| 11:55.44 | basty | bye |
| 12:02.36 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 12:03.34 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 12:04.45 | m0t3jl | Anyone has any experience with the Skype for Asterisk or any other (not officially supported) solution for SIP to Skype calls? |
| 12:04.45 | m0t3jl | <m0t3jl> What I would like to know is whether it is possible to somehow allow my users to use the ordinary Skype client on their desktops for chat and to only direct calls to my Asterisk PBX. |
| 12:04.57 | m0t3jl | Thanks a lot |
| 12:14.09 | *** join/#asterisk basty (~basty@212.218.65.195) |
| 12:14.14 | basty | hi - me again ! ;-) |
| 12:14.57 | *** join/#asterisk grayhame (~miller@74-94-250-169-Nashville.hfc.comcastbusiness.net) |
| 12:16.14 | basty | okay...i just had problems with the zaptel 1.4.12 - after I tried the latest zaptel svn - the compile finally worked. Now just another little and dumb question. I need to run 2x T1 on my machine. What would you guys recommend to use for a productiv system? Zaptel SVN or the latest dahdi 2.2.1 ? |
| 12:18.45 | m0t3jl | basty, don't ask me I have no idea, but tzafrir_laptop AFAIK said you should use Dahdi, so I'd go with his idea ;) |
| 12:19.08 | kaldemar | basty: zaptel was renamed to DAHDI, so it would be better to use DAHDI. what version of asterisk are you using? |
| 12:21.29 | kaldemar | oh, 1.4.29. choose DAHDI. |
| 12:21.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:24.19 | basty | kaldemar: yep...1.4.29 |
| 12:24.26 | basty | okay..then i will try the dahdi...thanks a lot |
| 12:28.48 | *** join/#asterisk qjb (~qjb@2001:470:1f15:5ad:21f:5bff:fef1:f005) |
| 12:28.52 | basty | bye for now ;-) |
| 12:32.53 | *** join/#asterisk Footman (~Footman@gwdev.creape.unilim.fr) |
| 12:32.58 | Footman | hello |
| 12:33.05 | *** join/#asterisk maverick (~maverick@83.240.229.38) |
| 12:33.09 | *** part/#asterisk devgru (~devgru@94.25.220.54) |
| 12:34.06 | maverick | hello having some problems with dtmf on analogue card |
| 12:34.17 | maverick | dtmf doesn't work |
| 12:36.28 | [TK]D-Fender | maverick: Details & debug would help |
| 12:37.46 | maverick | got one line connected to one analogue port |
| 12:37.53 | Footman | I installed Asterisk 1.6.0.17 via tarball on Debian Lenny but there is no dialplan. In fact, extensions.conf and/or extensions.ael doesn't parsed while a reload. It's certainly simple but I don't find why it doesn't work... Do you have an idea ? |
| 12:38.01 | maverick | when calling to the line number |
| 12:38.13 | maverick | it goes to an ivr |
| 12:38.28 | maverick | then when trying to press a key |
| 12:38.37 | maverick | it doesn't work |
| 12:38.45 | maverick | how can i enable dtmf debug ? |
| 12:40.44 | [TK]D-Fender | maverick: Show us your dialplan and the CLI output of your failed attempt in a PASTEBIN |
| 12:40.46 | [TK]D-Fender | ~pb |
| 12:40.47 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 12:41.12 | [TK]D-Fender | Footman: check the permissions on it |
| 12:42.44 | Footman | [TK]D-Fender: permissions are OK for me |
| 12:42.58 | tzafrir_laptop | Chainsaw, have you just managed to update the Mantis? |
| 12:43.26 | [TK]D-Fender | Footman: wrong anser. SHOW us what they are. |
| 12:43.39 | [TK]D-Fender | answer* |
| 12:43.49 | tzafrir_laptop | maverick, enable logging of 'dtmf' to 'console' in /etc/asterisk/logger.conf |
| 12:43.58 | tzafrir_laptop | and run 'logger reload' |
| 12:44.30 | tzafrir_laptop | Do you get a dialtone on the phone when you pick it up? |
| 12:45.26 | Chainsaw | tzafrir_laptop: Hrm? |
| 12:45.30 | [TK]D-Fender | tzafrir_laptop: Not FXS <- |
| 12:46.18 | Footman | [TK]D-Fender: so, wrong question ;) |
| 12:46.25 | Footman | # ls -ail /etc/asterisk/extensions.* |
| 12:46.26 | Footman | 302704 -rw-r--r-- 1 root root 12243 fév 3 14:05 /etc/asterisk/extensions.ael |
| 12:46.26 | Footman | 302705 -rw-r--r-- 1 root root 26867 fév 3 14:05 /etc/asterisk/extensions.conf |
| 12:46.36 | [TK]D-Fender | Footman: are you running * as root? |
| 12:46.37 | Footman | # ps -ef | grep asterisk |
| 12:46.37 | Footman | root 2021 1 0 Feb02 ? 00:01:26 /usr/sbin/asterisk |
| 12:47.00 | [TK]D-Fender | Footman: I'll take that as a "yes'. What is your modules.conf like/ |
| 12:47.54 | maverick | like this console => dtmf,notice,warning,error |
| 12:48.22 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 12:48.41 | maverick | tzafrir: like this ..... console => dtmf,notice,warning,error |
| 12:48.56 | Footman | [TK]D-Fender: this is modules.conf : http://pastebin.com/d353346a4 |
| 12:49.24 | mallchin | hi guys, I am having an issue with E1 lines in my asterisk box dropping every few days with a yellow alarm, wondering if anyone has any suggestions please? |
| 12:49.27 | mallchin | Feb 2 06:52:34 avs03 kernel: wct4xxp: Setting yellow alarm on span 2 |
| 12:49.30 | mallchin | Feb 2 06:52:39 avs03 kernel: wct4xxp: Clearing yellow alarm on span 2 |
| 12:50.12 | mallchin | It happens on 3 lines from 3 providers |
| 12:50.47 | [TK]D-Fender | Footman: do "module reload pbx_config.so" |
| 12:50.56 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:51.22 | etfonhomey | [TK]D-Fender, do you have Group Call Pickup implemented with your Polycom's and Asterisk? |
| 12:51.32 | [TK]D-Fender | etfonhomey: Nope |
| 12:51.42 | [TK]D-Fender | etfonhomey: never used pickup at all |
| 12:52.30 | etfonhomey | [TK]D-Fender, I have it working using an EFK, just was wondering if there was an easier way. |
| 12:52.36 | Footman | [TK]D-Fender: CLI> module reload pbx_config.so |
| 12:52.36 | Footman | [Feb 3 14:51:35] NOTICE[12363]: loader.c:660 ast_module_reload: The module 'pbx_config.so' was not properly initialized. Before reloading the module, you must run "module load pbx_config.so" and fix whatever is preventing the module from being initialized. |
| 12:52.53 | Footman | so I do "module load pbx_config.so" and it works |
| 12:53.06 | Footman | it parsed my extensions.conf |
| 12:53.17 | [TK]D-Fender | Footman: Curious it didn't seem to have started on its own |
| 12:53.34 | [TK]D-Fender | etfonhomey: EFK? |
| 12:54.03 | [TK]D-Fender | etfonhomey: Reading up on now... |
| 12:54.45 | etfonhomey | [TK]D-Fender, Polycom's enhanced feature key, http://pastebin.com/d568ffbec |
| 12:55.35 | Footman | [TK]D-Fender: after restart Asterisk, it's still good so it's OK. I keep your command if the problem occurs again. Thanks for help. :) |
| 12:58.26 | etfonhomey | [TK]D-Fender, I have a small 4 person CPA office with 4 x SPIP 550's. 2 CPA's, 1 bookkeeper, 1 receptionist. CPA's don't want their phones ringing on incoming calls but wanted to be able to pickup if the receptionist was at lunch. |
| 12:59.46 | Chainsaw | tzafrir_laptop: I update that bug for every new release, rebasing if necessary. Still waiting for that lab test. |
| 13:01.11 | Akiraa | m0t3jl: did you have success in routing calls to or from Skype and Asterisk? |
| 13:06.26 | *** part/#asterisk maverick (~maverick@83.240.229.38) |
| 13:10.56 | *** join/#asterisk coppice (~chatzilla@172.169.232.220.dyn.pacific.net.hk) |
| 13:14.33 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:14.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:16.00 | *** join/#asterisk Skeeter- (Skeeter@190-141.cgocable.ca) |
| 13:29.33 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 13:30.20 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:32.43 | *** join/#asterisk roni (~roni@190.196.71.206) |
| 13:34.34 | roni | hi everybody... just a little question , how i can return to the asterisk cmd line , if asterisk is already running .. can i do that without restart (no asterisk -r) |
| 13:35.23 | ManxPower-work | roni: asterisk -r is not a restart |
| 13:35.29 | ManxPower-work | roni: how did you start Asterisk |
| 13:35.36 | roni | just asterisk |
| 13:35.50 | ManxPower-work | then asterisk -r will connect you to the existing asterisk process |
| 13:36.10 | [TK]D-Fender | roni: And if you quit the console that just called "asterisk" it will kill off * entirely |
| 13:36.12 | roni | thanks .. that is what i need to know .. thank you very much |
| 13:38.20 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:38.25 | krion | hi guys |
| 13:38.38 | [TK]D-Fender | roni: You need to start * via either safe_asterisk, or under an init script that runs it as a daemon |
| 13:38.49 | [TK]D-Fender | roni: then you connect to it with "-r" |
| 13:39.13 | [TK]D-Fender | roni: the only reason to run * live at CLI like that is to see why it fails to load as a daemon |
| 13:39.33 | roni | Thanks .. i gonna do that .. |
| 13:44.32 | krion | is there people here how do audio quality test ? |
| 13:44.58 | krion | i've plenty of complain from some customer about mediocre audio quality |
| 13:45.09 | krion | i want to apply automatic test check |
| 13:45.37 | krion | i've a look at sevana product but don't know if it's able to test using a sip account |
| 13:47.36 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:48.13 | *** join/#asterisk TheDavidFactor (~chatzilla@fw1.safedataisp.net) |
| 13:49.02 | krion | i want the mos score in fact |
| 13:55.21 | *** join/#asterisk garymc (~garymc@81.138.225.161) |
| 13:58.35 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 13:59.31 | aeperezt | any tip agains this issue chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
| 14:01.19 | ManxPower-work | aeperezt: That is one of the hardest errors to fix. |
| 14:02.08 | ManxPower-work | aeperezt: The issue is either 1) line errors (contact telco) or 2) interrupt latency issues (very hard to fix) |
| 14:03.06 | aeperezt | ManxPower-work: yes I know, well telco check it out and they said it works fine, 2 option is what i'm looking for tips on |
| 14:03.56 | [TK]D-Fender | aeperezt: pastebin "cat /proc/interrupts' and tell us what card you're using. |
| 14:03.58 | [TK]D-Fender | ~pb |
| 14:03.59 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:03.59 | ManxPower-work | aeperezt: The only way I have solve 2) is replace the server |
| 14:04.34 | aeperezt | the problem is that it works fine for like a month then one day start showing this and part of the pri goes down then it works back again |
| 14:05.24 | benngard | how can a subroutine (when it excecutes Return) be sent into invalid extension? i mean thats why u use gosub or? |
| 14:05.26 | ManxPower-work | aeperezt: virtually anything from excessive network traffic to cron job reports could trigger it. |
| 14:05.45 | aeperezt | I read of that solution, seems to works best |
| 14:05.51 | ManxPower-work | benngard: it can't unless you screwed up the return stack, and I doubt you've done that. |
| 14:05.57 | [TK]D-Fender | benngard: SHOW US |
| 14:06.18 | benngard | plz look here: will do |
| 14:06.29 | benngard | <PROTECTED> |
| 14:06.29 | benngard | -- Executing [s@set-connected-line:8] Return("SIP/0317998975-000000a3", "") in new stack |
| 14:06.40 | benngard | [Feb 3 14:53:58] WARNING[3966]: pbx.c:4737 __ast_pbx_run: Channel 'SIP/0317998975-000000a3' sent into invalid extension '90510' in context 'inputinterior.se', but no invalid handler |
| 14:07.15 | ManxPower-work | benngard: 1) don't flood the channel 2) there are 7 missing CLI outout lines |
| 14:07.17 | aeperezt | ManxPower-work: the funny part is that i see it when the server is doing nothing like at night then when there is activity it stop and comes back later |
| 14:07.36 | [TK]D-Fender | benngard: PASTEBIN including the entire dialplan |
| 14:07.37 | benngard | the i try to use pastebin, never used it before |
| 14:07.42 | benngard | then* |
| 14:07.54 | ManxPower-work | benngard: if you want help here, learn to use pastebin |
| 14:08.00 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 14:09.31 | benngard | http://pastebin.com/m7d7c997a |
| 14:11.07 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:11.21 | ManxPower-work | benngard: you should have a space before the < in the CONNECTED LINE and CALLERID sets |
| 14:11.29 | benngard | did miss the cli output http://pastebin.com/m1a9b5aca |
| 14:11.48 | benngard | ManxPower-work: oki, thx |
| 14:12.14 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:12.23 | ManxPower-work | benngard: other than that it looks good to me |
| 14:12.53 | benngard | ManxPower-work: yupp, that why i asked, couldnt se any errors |
| 14:13.07 | benngard | i am changing atm add a space i mean |
| 14:14.16 | benngard | same error |
| 14:15.14 | benngard | exten => _0X.,1,Gosub(set-connected-line,s,1) |
| 14:15.15 | benngard | exten => _0X.,n,Dial(OOH323/0${EXTEN}@Avaya,,i) <= that works as a clock, with connected line updates |
| 14:15.36 | benngard | it must be that i mess around with dnid number :( |
| 14:17.35 | *** join/#asterisk skymeyer (~skymeyer@91.183.54.9) |
| 14:19.26 | [TK]D-Fender | benngard: no, that has nothing to do with anything |
| 14:20.30 | [TK]D-Fender | benngard: Now show us the actual dilplan that is being using and the complete call attempt |
| 14:21.12 | benngard | i did send it for some minutes ago |
| 14:21.25 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:21.31 | benngard | http://pastebin.com/m7d7c997a and http://pastebin.com/m1a9b5aca |
| 14:22.18 | benngard | http://pastebin.com/d62c268a when i dial with area code, that works |
| 14:24.27 | *** join/#asterisk smooth_penguin (~smoove@59.95.30.99) |
| 14:24.34 | kaldemar | benngard: you're missing /_031X in the last pasted example, compared to earlier. |
| 14:24.40 | [TK]D-Fender | benngard: Executing [s@set-connected-line:6] ExecIf("SIP/0317998975-000000a3", "1?Set(CONNECTEDLINE(all)=Froken Ur<90510>)") in new stack |
| 14:24.52 | [TK]D-Fender | benngard: Yuo changed the CALLerID and your CID matching on that exten fails |
| 14:25.07 | [TK]D-Fender | benngard: _031X. <--- DOA |
| 14:25.24 | [TK]D-Fender | benngard: exten => _[1-9]X./_031X.,2,Gosub(set-connected-line,s,1) <- Not going to match |
| 14:25.38 | [TK]D-Fender | benngard: Your approach to this should be changed |
| 14:26.55 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 14:29.59 | benngard | [TK]D-Fender: i am not with u, but i am trying to understand... |
| 14:30.54 | [TK]D-Fender | benngard: that line matches based on CALLER ID. You are CHANGING the CALLER ID in your amcro and have no place to return because _031X. will not match 90510 |
| 14:31.41 | benngard | but the same macro works as i showed u in http://pastebin.com/d62c268a |
| 14:32.20 | benngard | the only difference is that i am not messing around with dnid |
| 14:33.50 | [TK]D-Fender | benngard: When did CONNECTEDLINE get introduced? |
| 14:34.25 | benngard | dont know, was reading source code and found it |
| 14:34.33 | [TK]D-Fender | benngard: what ver? |
| 14:34.39 | benngard | sec |
| 14:34.50 | kaldemar | benngard: that's because in the latter example your extension is not limited to caller id's that start with 031. |
| 14:35.07 | [TK]D-Fender | kaldemar: I SNAFU'd.. wasn't CID being set |
| 14:35.24 | [TK]D-Fender | kaldemar: CONNECTEDLINE(all) <- or... whatever this is & does |
| 14:36.04 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 14:36.10 | kaldemar | [TK]D-Fender: connectedline is a feature of OOH323 channel that lets the user know who he/she is calling. that will come to SIP driver too, iirc. |
| 14:36.15 | benngard | i know i am running a trunk version, SVN-trunk-r243986M, need stuff from the latest ooh323 channel |
| 14:36.35 | benngard | i have it in the sip version i am running |
| 14:36.46 | [TK]D-Fender | kaldemar: Ah, so this is pre-release CPID? |
| 14:37.01 | kaldemar | [TK]D-Fender: and yes, you already told him about the caller id, seems that it didn't stick. |
| 14:37.04 | [TK]D-Fender | kaldemar: And currently OOH323 specific? |
| 14:37.10 | jaytee | leifmadsen, I love the new FB profile pic! :-) |
| 14:37.11 | [TK]D-Fender | kaldemar: Correct. |
| 14:37.28 | *** join/#asterisk yangpp (~yang@freenode/sponsor/cacert.assurer.yang) |
| 14:37.41 | kaldemar | [TK]D-Fender: called party id, and yes, only for OOH323 afaik. |
| 14:37.55 | benngard | when i dial for example 971 from my sip phone the display changes to Stefan Andersson 971 |
| 14:38.02 | file | SIP supports it too. |
| 14:38.05 | file | chan_sip rather |
| 14:38.11 | benngard | yupp |
| 14:38.26 | kaldemar | file: released 1.6.2 already? |
| 14:38.29 | benngard | i did notice that for some nights ago so i did start testing it |
| 14:39.01 | benngard | but it is very example so it was some tries before i figure out howto use it |
| 14:39.16 | leifmadsen | jaytee: :) |
| 14:39.32 | benngard | very few* |
| 14:40.16 | kaldemar | doesn't seem to be in 1.6.2.1. |
| 14:41.54 | [TK]D-Fender | benngard: -- Executing [90510@inputinterior.se:2] Gosub("SIP/0317998975-000000a3", "set-connected-line,s,1") in new stack <-- this line.... |
| 14:43.07 | [TK]D-Fender | benngard: Nevermind... the samples you give don't match up.. |
| 14:43.15 | eppigy | SHE A NYMPHO |
| 14:43.23 | [TK]D-Fender | benngard: I can go comparing your pastebins because of it.. |
| 14:43.43 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 14:44.06 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 14:44.15 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:44.45 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 14:48.31 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:51.19 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:53.51 | benngard | yay now it works, both with and without areacode and my display updates :) |
| 14:55.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:55.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.32 | ManxPower-work | Well, 1.4.23.2 is....interesting...in the same way a train wreck is "interesting" |
| 14:57.37 | benngard | very simple when i found out what i was doing: http://pastebin.com/d4408b63 |
| 14:57.38 | *** join/#asterisk traxx (~traxx@91-64-130-76-dynip.superkabel.de) |
| 14:58.56 | benngard | time for me to add all my external friens to my "set-connected-line" subroutine :) |
| 14:59.14 | benngard | friends* i need a new keyboard :( |
| 14:59.21 | *** join/#asterisk moy (~moy@74.12.123.169) |
| 15:01.04 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 15:12.00 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:15.35 | *** join/#asterisk Cherebrum (jgarland@209.9.237.93) |
| 15:16.14 | Cherebrum | Anyone know if I can use the TDM400P card without the power connector if I am using only one FXS module? From what I understand the power is needed to power FXO modules |
| 15:16.59 | ManxPower-work | Cherebrum: you have it backwards |
| 15:17.02 | smooth_penguin | I thought it was for the FXS |
| 15:17.17 | ManxPower-work | the power connector is needed to provide ring voltage to the phones. |
| 15:17.20 | smooth_penguin | why would you want to power FXO :P |
| 15:17.22 | Cherebrum | yea.. that's what I meant |
| 15:17.36 | Gido-E | Cherebrum if you connect phones to it, you need the power |
| 15:17.43 | ManxPower-work | Cherebrum: then say what you mean. In telecom, not being accurate can cost you thousands of dollars |
| 15:17.45 | Cherebrum | because I'm installing in a DL360 G4p server and it has no power connectors for such things |
| 15:17.58 | ManxPower-work | ALSO the PCIe cards apparently need power regardless |
| 15:18.10 | ManxPower-work | Cherebrum: then your card will be pretty useless. |
| 15:19.04 | Gido-E | ManxPower-work if it doesn't cost thousands of dallars, it can't be annything i need :-) |
| 15:19.32 | Cherebrum | no wait.. this is a TDM400 It's PCI |
| 15:19.33 | Gido-E | s/ dal/dol/ |
| 15:19.48 | ManxPower-work | Gido-E: For example if you plug phone lines into FXS ports on Digium cards, the port will blow the first time the phone rings. Not being accurate can cost you thousands of dollars. |
| 15:19.56 | Cherebrum | I pulled it out of my box of free TDM cards I received at conferences. :p |
| 15:20.11 | Cherebrum | Might even be a Nextvox card |
| 15:20.15 | Cherebrum | not sure |
| 15:20.25 | Cherebrum | ok.. I'll give it a try with just the FXO module in it and no power |
| 15:20.26 | Gido-E | ManxPower-work i did note that in my Asterisk book. (dutch language only) |
| 15:20.43 | *** join/#asterisk HenrikJott (~info@d83-183-134-141.cust.tele2.se) |
| 15:20.54 | Cherebrum | then I'll see if I can make it work with freeswitch |
| 15:21.07 | Cherebrum | that'll be fun |
| 15:21.22 | *** part/#asterisk Cherebrum (jgarland@209.9.237.93) |
| 15:21.33 | *** join/#asterisk igorg (~igorg@net220.186.188-215.dynamic.omsk.ertelecom.ru) |
| 15:22.35 | HenrikJott | Hi all! I have a problem with asterisk not handling some of my call files. It seems like it puts call files on pause when it does a srv_lookup and then when the lookup is done it does not start over with the call file but just leaves it there. Maybe it would have handled the call eventually, but is there any way to tell asterisk to do all the calls in /var/spool/asterisk/outgoing/ right now? |
| 15:23.01 | ManxPower-work | HenrikJott: do you need srvlookups? I usually disable them |
| 15:23.39 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 15:25.51 | garymc | anyone knwo what i need to do to configure my analogue ports on my Sangoma A200 to work with my A101D ? |
| 15:26.57 | tzafrir_laptop | ManxPower-work, are you sure (about exploding)? |
| 15:27.35 | HenrikJott | ManxPower-work, yes unfortunatly i do, my itsp has two servers for redundancy and asterisk switches between them all the time... |
| 15:27.39 | tzafrir_laptop | doesn't make sense |
| 15:29.39 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:30.03 | ManxPower-work | tzafrir_laptop: unless they changed their design with regards to that issue, yes. |
| 15:30.32 | HenrikJott | ManxPower-manx: or could i skip srv_lookup and still use both servers from my itsp? |
| 15:30.36 | tzafrir_laptop | We had one such card and I never tried being careful about this |
| 15:31.01 | ManxPower-work | tzafrir_laptop: it's a fairly well known issue |
| 15:33.03 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 15:34.44 | ManxPower-work | tzafrir_laptop: try plugging in a PSTN line into an FXS port. Call it. Let us know what happens. 8-) |
| 15:35.01 | ManxPower-work | (digium analog card, obviously) |
| 15:38.16 | tzafrir_laptop | Just did, with a TDM410 (newer one, that is). No smoke detected. |
| 15:38.56 | ManxPower-work | tzafrir_laptop: does the port still work? |
| 15:40.38 | *** join/#asterisk tengulre (~tengulre@222.174.116.210) |
| 15:41.28 | *** join/#asterisk RichardLynch (~RichardLy@74.222.60.226) |
| 15:44.20 | RichardLynch | I'm trying to get a handle on dial plans... All the docs I read make perfect sense, and seem straightforward enough. But when I go to "branch out" on my own, I seem to always hit a brick wall and spend hours/days/weeks spinning my wheels. I suspect I'm missing a "big picture" diagram somewhere of how all the conf files fit together. I've read several books and online docs... Is there a diagram picture flow-chart form that relates all the conf f |
| 15:45.14 | ManxPower-work | RichardLynch: are you using a GUI for Asterisk? |
| 15:45.41 | tzafrir_laptop | ManxPower-work, yes |
| 15:45.54 | RichardLynch | ManxPower-work: I'm flexible. We have freepbx, and I'm developing a custom module, but I'm personally more comfy with vi and CLI. |
| 15:46.24 | RichardLynch | I had a similar experience previously, with no GUI in the way, fwiw. :-) |
| 15:46.52 | ManxPower-work | If you are trying to work with FreePBX's config files -- well I wish you the best of luck. |
| 15:46.57 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
| 15:47.04 | [TK]D-Fender | RichardLynch: No, there is no miracle map. There isn't a clear line of association between most config files. Each deals with their own thing for the most-part |
| 15:47.06 | RichardLynch | I "get" all the individual "trees" when I read the docs and go to put in _NXXNXXXXXXX or whatever. I'm missing the "forest" though, I think... |
| 15:47.48 | ManxPower-work | RichardLynch: one forrest people many times don't understand is that the context= in sip.conf and zapata/dahdi specifies where a call from that device lands in the dialplan |
| 15:48.00 | [TK]D-Fender | RichardLynch: that is dialplan (extensions.conf) |
| 15:48.03 | RichardLynch | But those things are inter-related... For example, my latest stumbling block was a dial plan to tie into OCS that took hours to figure out was we needed an outbound trunk. |
| 15:48.41 | [TK]D-Fender | RichardLynch: And that term "outbound trunk" is GUI crap and non-indicative as to the actual Asterisk configs it modifies |
| 15:48.48 | ManxPower-work | Asterisk doesn't have such a thing as an "outbound trunk" |
| 15:49.20 | RichardLynch | My phone guy called it an "outbound trunk" as we were editing the .conf files by hand... Go figure. What *is* it, then? |
| 15:49.53 | ManxPower-work | RichardLynch: it's just another Dial line. |
| 15:49.57 | RichardLynch | I feel like I'm in the middle of a railroad yard flipping switches, with no operator map showing me where the lines go. |
| 15:50.18 | ManxPower-work | Asterisk doesn't really know or care if the device is a phone or a connection to the PSTN |
| 15:50.19 | benngard | when i originate a call from ami ${CALLERID(dnid)} is not set :( |
| 15:50.47 | RichardLynch | But it does care that it can route it somewhere somehow... |
| 15:51.08 | ManxPower-work | RichardLynch: yes, but that has nothing to do with a "trunk". |
| 15:51.08 | p3nguin | Not really, since Asterisk doesn't route. |
| 15:51.40 | RichardLynch | I'm not sure how he even intuited that we were missing that device/route/trunk/gerbil... |
| 15:51.49 | Ad-Hoc | hi ppl |
| 15:51.53 | ManxPower-work | RichardLynch: Because FreePBX uses the terms |
| 15:52.34 | RichardLynch | We were staring at .conf files, and CLI debug tail log and /var/log/asterisk/full tail -f at the time... |
| 15:52.36 | *** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 15:52.42 | ManxPower-work | Dial(SIP/12345@9999) Is the sip.conf entry [9999] a "trunk" or a "phone" We have no way of telling.. |
| 15:52.53 | *** join/#asterisk reptiles (jim@198.96.210.227) |
| 15:52.59 | leifmadsen | a better term is "connection" |
| 15:53.02 | ManxPower-work | RichardLynch: practically useless when you are using FreePBX configs. way, way, way too much noise. |
| 15:53.15 | p3nguin | People say "trunk" when they are talking about the connnection to/from your ITSP. |
| 15:53.16 | leifmadsen | I've had to start using FreePBX for a customer -- never again |
| 15:53.24 | Geminizer | Hello all. I am assuming that anytime Dial(...) is invoked in a dialplan, another channel is consumed ? |
| 15:53.35 | leifmadsen | Geminizer: that's the point of Dial :) |
| 15:53.38 | ManxPower-work | leifmadsen: *nod* We hope to stop selling FreePBX systems by the end of the year. |
| 15:53.39 | RichardLynch | ManxPower-work: We did have to turn off a lot of other "trunks" on this box to reduce log noise, yes. |
| 15:53.46 | thehar | makes a joke that freepbx.org is failed right now |
| 15:54.53 | RichardLynch | Even now, we don't have what we actually want... Perhaps I should start with that instead of this general wish for a "big picture" map... |
| 15:54.55 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:54.55 | reptiles | my boss gave me the simple task of "find a ip phone or related vendor that is "loop" something or something "loop". |
| 15:54.57 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:55.08 | reptiles | google, is oobviously useless. |
| 15:55.26 | reptiles | anyone heard of a phone or vendor of ip phones with "loop" in its name? |
| 15:55.38 | ManxPower-work | reptiles: your boss is too. Find a better question and you'll find better answers. |
| 15:55.50 | ManxPower-work | reptiles: "loop" is a generic telecom term used in thousands of products |
| 15:55.55 | p3nguin | The term "trunk" really only applies to IAX2 devices which have trunking enabled (trunk=yes). |
| 15:55.56 | RichardLynch | reptiles: Ask him where he heard of it. |
| 15:55.58 | Naikrovek | yeah just ask the boss again |
| 15:56.01 | reptiles | yeah, i tried that route, didn't get me more details.\ |
| 15:56.05 | ManxPower-work | ~trunk |
| 15:56.06 | infobot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 15:56.32 | ManxPower-work | reptiles: buy polycom phones. I'm sure somewhere in their docs they use the term "loop" |
| 15:56.40 | reptiles | heh. |
| 15:56.46 | Naikrovek | polycom > * |
| 15:56.50 | ManxPower-work | ~phones |
| 15:56.51 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
| 15:56.51 | RichardLynch | reptiles: Don't ask about the phone itself. Ask where he heard it. Then go to the horse's mouth to find out what they were talking about. |
| 15:56.51 | reptiles | is quite aware of the terminology. |
| 15:57.27 | reptiles | i'm sure if i came in here and asked "is ther a phone vendor with "poly" or "nom" in its name, i'd get some responses. |
| 15:57.47 | reptiles | so, i'm simply asking the same question with "loop" instead. |
| 15:57.55 | RichardLynch | reptiles: I don't think any of them can recall any phone/vendor with "loop" in the name... |
| 15:58.06 | Naikrovek | well "loop" is a term that is used a lot, and phones don't do a lot of looping, unless they're stuck in a reboot loop |
| 15:58.27 | reptiles | yep. |
| 15:58.46 | Naikrovek | loop has 3 letters, and polycom has all of those three in its name |
| 15:58.47 | ManxPower-work | reptiles: the answer to your question is "No, there are no well known IP phones with the word "loop" in their model name. |
| 15:58.59 | RichardLynch | I think reptiles knows all that. He's been tasked by an idiot to find a company or product with "loop" in the name. |
| 15:59.04 | reptiles | manx: thanx, that's moreso the answer i was looking for. |
| 15:59.34 | RichardLynch | Go buy a Polycom and slap a WLUP sticker on it. :-) (local radio station) |
| 15:59.41 | Naikrovek | heh |
| 15:59.42 | RichardLynch | "The Loop" nickname. |
| 15:59.55 | reptiles | we are bidding on a procurement contract for a telco. |
| 15:59.59 | *** join/#asterisk Warp4 (~Robert@firewall-a.buf.ny.i-evolve.net) |
| 16:00.03 | reptiles | we are pushing grandstream. |
| 16:00.29 | reptiles | boss heard that one of the other companies was pushing csiomething called loop, or a vendor called loop. |
| 16:00.38 | reptiles | wanted me to see if i could find more info. |
| 16:01.29 | reptiles | also, one of the other vendors is pushing that their phone supports the "RT codec" |
| 16:01.31 | RichardLynch | Dollars to donuts you have a case of the game "telephone" and there's no "loop" in any name at all... |
| 16:01.38 | reptiles | which i found kinda entertaining. |
| 16:05.04 | p3nguin | ~grandstream |
| 16:05.05 | infobot | [grandstream] the Yugo of VoIP hardware. Run. Run away now.. Though therealcircut says that they're not that bad |
| 16:05.11 | p3nguin | ~gs |
| 16:05.12 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:05.43 | reptiles | i'm sure the channel has numerous opinions on any vendor. |
| 16:06.32 | Qwell | reptiles: no, just a few opinions on other vendors |
| 16:06.39 | Qwell | many on grandstream though, yes |
| 16:06.44 | Qwell | ...for good reason |
| 16:07.10 | reptiles | any recommendations, at/around the same or better pricepoints? |
| 16:07.24 | Naikrovek | polycom phones are amazing and well priced |
| 16:07.28 | Qwell | ^ |
| 16:07.29 | Naikrovek | easy to configure in bulk |
| 16:07.30 | p3nguin | You get what you pay for. |
| 16:07.43 | reptiles | nods to p3nguin |
| 16:08.06 | reptiles | unfortunately, the telco will be making their decision 90% based on price. |
| 16:08.12 | p3nguin | The most common Polycoms I see people using are priced right around $100 retail. |
| 16:08.27 | Naikrovek | the 321 is $80 |
| 17:56.18 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:56.18 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.2 (2010/02/02), Asterisk 1.6.1.14 (2010/02/02), 1.6.0.22 (2010/02/02), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.1 (2010/01/20), dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- |
| 17:57.59 | ecrane | Sometimes I use Reynwood for LD. |
| 17:58.23 | ecrane | But I don't know rates ^^. |
| 18:02.11 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 18:02.59 | citywok | flowroute sounds pretty good. Mo was helpful, answered my porting questions pretty quick, the only downside is you are trapped with amazon payments |
| 18:05.25 | titter | span_message: WARNING T.30 ECM carrier not found -- any ideas? |
| 18:07.54 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:11.26 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 18:13.15 | p3nguin | citywok: VoIP.ms allows me to pay with PayPal, AmEx, VISA, MC, and Discover. |
| 18:13.57 | *** join/#asterisk neurosys (~neurosys@173.200.201.73) |
| 18:14.10 | p3nguin | They also allow bank wire, Western Union, and Xoom.com, but there are some stipulations associated with those. |
| 18:14.58 | citywok | wire is always a pain and costs another $35-45 normally |
| 18:16.49 | *** join/#asterisk stope (~nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 18:16.52 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:16.59 | p3nguin | Yeah, they have a $500 minimum on bank wires. :/ |
| 18:17.05 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.12 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 18:21.29 | *** join/#asterisk ttl- (~patrick@d5153A420.access.telenet.be) |
| 18:23.16 | *** join/#asterisk evilbit (~evilbit@n1-24-57.dhcp.drexel.edu) |
| 18:23.19 | orangepower | anyone in San Francisco? |
| 18:23.47 | p3nguin | Will you judge those who say yes? |
| 18:23.51 | evilbit | hi, wondering if anyone is using digium's yum repositories and has mysql working with asterisk-1.6? |
| 18:23.59 | jaytee | no, but if you're going be sure to wear a flower in your hair |
| 18:24.56 | evilbit | <PROTECTED> |
| 18:25.07 | orangepower | p3nguin: naw, i'm just looking to hire someone in SF to do some asterisk work |
| 18:25.23 | p3nguin | It needs to be on-site? |
| 18:25.26 | jaytee | i've been using mysql with * 1.6 but I compiled asterisk and the add-ons, I didn't use the RPMs |
| 18:25.50 | orangepower | p3nguin: lot of problems, might be ok offsite i guess, but onsite is preferred |
| 18:26.12 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:26.23 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:27.52 | beek | Hi jaytee |
| 18:29.58 | titter | span_message: WARNING T.30 ECM carrier not found -- any ideas? |
| 18:31.45 | jaytee | afternoon beek |
| 18:33.43 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-042-222-050.hsi3.kabel-badenwuerttemberg.de) |
| 18:34.18 | *** join/#asterisk mnick86 (~mnick86@95-90-159-200-dynip.superkabel.de) |
| 18:34.48 | raden_work | eppigy, how goes it bro ? |
| 18:36.51 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:40.47 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-042-222-050.hsi3.kabel-badenwuerttemberg.de) |
| 18:41.24 | Footman | bye ! |
| 18:43.37 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.36.189) |
| 18:48.59 | ManxPower-work | titter: As I understand it that means no ECM carrier (fax tone) was found. |
| 18:49.29 | titter | ManxPower-work: faxes are being received, just get this error repeated twice, every 3 seconds of the transmission |
| 18:49.58 | titter | ManxPower-work: thinking about start fresh with spandsp, and asterisk again |
| 18:50.24 | ManxPower-work | titter: that sounds like you are trying to send fax over voice over up |
| 18:50.33 | ManxPower-work | fax over voice over ip |
| 18:51.08 | titter | It's coming in via DAHDI on a new PRI |
| 18:51.27 | ManxPower-work | titter: Weird. |
| 18:51.53 | titter | The fax is being sent from analog lines at my location, but also other fax machines give this error. |
| 18:52.51 | titter | I think I will just start fresh, and see if I get the same error. On my test machine, it worked flawless. |
| 18:56.52 | Naikrovek | amazon ec2 is so wicked |
| 18:57.09 | Naikrovek | is impressed by the price of the spot instances. $4/week |
| 18:59.12 | benngard | is trying to make up his mind, either pay Avaya about $40000 for upgrading our CM or start moving users to an almost working * |
| 18:59.41 | Naikrovek | what's not working with * |
| 19:00.35 | benngard | we have a switch-board application named "Svara" that's gonna be a hell to get to work with * |
| 19:00.43 | benngard | uther than that nothing! |
| 19:00.58 | benngard | some users have alrady moved to * |
| 19:01.03 | benngard | :) |
| 19:01.13 | Naikrovek | they don't know it, probably, huh? |
| 19:01.25 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 19:01.31 | benngard | yes they do |
| 19:01.47 | Naikrovek | what are the requirements of that Svara app. just basic receptionist stuff or special stuff |
| 19:01.55 | benngard | but i could actually move some more and they shouldnt notice |
| 19:02.16 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 19:02.28 | benngard | that "Svara" is talking tsapi with our avaya and i dont know a shit about tsapi |
| 19:06.13 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 19:07.10 | benngard | Naikrovek: just std stuff, answer, status of why a call is diverted and shit like that |
| 19:07.42 | Naikrovek | well Asterisk has AMI, which you can use to get status of extensions, calls, etc |
| 19:07.52 | Naikrovek | conferences, queues, others i'm sure |
| 19:08.13 | benngard | i know i use it for us that have moved, manipulating queues and so |
| 19:09.28 | benngard | and i really like it, i can do almost everything there that i couldnt do with our avaya |
| 19:09.50 | benngard | originate calls from a small web app for example |
| 19:10.28 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 19:10.31 | citywok | how do you make * tag packets for QoS? |
| 19:10.44 | benngard | just select who u wanna dial and press [Dial] wait for your phone to ringa, pickup and w8 for other end to answer |
| 19:10.49 | p3nguin | citywok: Use tc. |
| 19:10.50 | ManxPower-work | citywok: set the option in sip.conf |
| 19:11.18 | ManxPower-work | assuming you are running Asterisk as root or using the libcap library to allow Asterisk to set the ToS/DSCP when not running as root. |
| 19:11.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:11.58 | citywok | perfect, testing now :) |
| 19:12.05 | titter | ManxPower-work: that did it ... still curious to why that error was occuring |
| 19:12.17 | mnick86 | does anybody know a barebone / mainboard with intel atom and 1x PCI, 2x PCI-Express ? |
| 19:13.48 | orangepower | anyone ever setup a Polycom VX 1500? |
| 19:14.00 | orangepower | it won't register with asterisk |
| 19:14.38 | citywok | ManxPower-work: do i need to stop/restart or will a simple reload do the trick? |
| 19:14.38 | ManxPower-work | orangepower: it works like any other Polycom phone with regards to REGISTRATION to Asterisk. |
| 19:14.50 | ManxPower-work | citywok: sip reload will do it. |
| 19:14.57 | orangepower | ManxPower-work: which means.. it won't? |
| 19:15.10 | ManxPower-work | orangepower: all polycoms register just fine to Asterisk. |
| 19:15.34 | Naikrovek | just fine. if it doesn't you got a config problem somewhere |
| 19:15.36 | orangepower | ManxPower-work: i can register fine with xmeeting and such.. but the polycom won't work for some reason |
| 19:15.48 | orangepower | obviously my config isn't right.. :( |
| 19:15.53 | Naikrovek | it happens, just look through the phone log or * log to see if you can determine what's up |
| 19:16.17 | orangepower | polycom's have a log? |
| 19:16.22 | Naikrovek | yeah |
| 19:16.34 | Naikrovek | they upload it to the (t)ftp server they're provisioned with |
| 19:16.37 | ManxPower-work | orangepower: are you using a provisioning server or the web gui? |
| 19:16.44 | orangepower | webgui |
| 19:16.46 | Naikrovek | ah |
| 19:16.55 | orangepower | we only have one phone for the time being |
| 19:17.00 | Naikrovek | i have all kinds of problems with that damn gui |
| 19:17.05 | ManxPower-work | orangepower: standby |
| 19:17.36 | orangepower | Naikrovek: yea, it refused to save for a while, finally got it half-way working |
| 19:18.09 | orangepower | Naikrovek: would it be... horrible if I screenshotted my settings and asked you guys to take a look at both screens? (sip and line1) |
| 19:18.19 | pigpen | I think I know how extension priority works, but I have read a few things, so I want to clairify. |
| 19:18.44 | p3nguin | pigpen: What are your concerns? |
| 19:18.46 | ManxPower-work | orangepower: I'm rebooting my polycom now to enable the web GUI. I'll give you the settings |
| 19:19.00 | pigpen | if I have exten => 5555 AND exten => 5XXX, will the 5XXX ever be hit? |
| 19:19.03 | Naikrovek | orangepower: it would be far easier for me to help you with the automatic provisioning |
| 19:19.11 | orangepower | ManxPower-work: you rock man thanks |
| 19:19.12 | Naikrovek | orangepower: ManxPower-work will take care of you |
| 19:19.37 | p3nguin | pigpen: When you dial something that is four numbers, starts with 5 and isn't 5555. |
| 19:19.39 | ManxPower-work | pigpen: tnose are not priorities. |
| 19:19.54 | p3nguin | pigpen: Except you have to use _ for matching, i.e. _5XXX |
| 19:19.55 | pigpen | sorry, pattern matching. |
| 19:20.13 | pigpen | yeah, once again, forgot to mention _5XXX |
| 19:20.14 | p3nguin | pigpen: Those are just regular extensions. |
| 19:20.37 | pigpen | but since one overlaps the other, won't the most specific me used? |
| 19:20.44 | ManxPower-work | orangepower: Address: and Authentication ID should be your SIP userID. the part in [here] in sop.conf. Authentication password should be your sip.conf [here] secret. |
| 19:20.56 | p3nguin | pigpen: If you have 5555 and _5XXX, calling 5555 will match 5555 ... calling 5556 won't, but it will match _5XXX. |
| 19:20.57 | ManxPower-work | THEY ARE NOT REGULAR EXPRESSIONS |
| 19:21.16 | ManxPower-work | pigpen: all this assumes the 2 extensions are in the same context. |
| 19:21.26 | pigpen | yes, in the same. |
| 19:21.31 | orangepower | ManxPower-work: ok so Address and Auth should be "14" (for ext 14) and secret.. ok got it |
| 19:21.32 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 19:21.49 | ManxPower-work | orangepower: The "address" always confused me. |
| 19:21.54 | orangepower | ManxPower-work: and server settings? |
| 19:22.00 | orangepower | ManxPower-work: yea.. but i had it right.. |
| 19:22.02 | pigpen | I have been having some odd "loops" of did's that are not mapped, it will then proceed to span both of my PRI's due to the fact they were not "falling through" do a catch all ext. |
| 19:22.19 | orangepower | ManxPower-work: private or shared |
| 19:22.21 | pigpen | And I don't want to foobar anything during business hours. |
| 19:22.39 | pigpen | Thanks agian both of you. |
| 19:22.45 | ManxPower-work | orangepower: make sure you have Register: 1 orangepower: leave everything else except the server ip the default |
| 19:22.57 | orangepower | already had that :( |
| 19:23.05 | p3nguin | pigpen: You could hard code the extensions matching the DID numbers if you wanted. |
| 19:23.23 | pigpen | p3nguin, yeah, I thought about it, but I have around 500 |
| 19:23.32 | pigpen | and some of the blocks are chopped up. |
| 19:23.35 | pigpen | call it lazy. |
| 19:23.50 | p3nguin | pigpen: How many are in use, how many unused? |
| 19:24.11 | pigpen | about 250 are allocated. |
| 19:27.04 | p3nguin | pigpen: Are all of those 250 already being directed to hard-coded extens? |
| 19:27.12 | pigpen | yeah. |
| 19:27.16 | eppigy | raden_work: pretty good |
| 19:28.05 | p3nguin | pigpen: If you are SURE that every other DID that you are supposed to be using is hard-coded and matching correctly, you could try using _X. as a catch-all exten for every other DID that will be directed into the given context. |
| 19:28.31 | pigpen | yeah, I tried that....I had an oddity.... |
| 19:28.32 | p3nguin | pigpen: The exact matches will be "grabbed" before patterns are checked. |
| 19:28.45 | raden_work | eppigy, good deal |
| 19:28.51 | raden_work | about to mess with my 1710 lol |
| 19:28.56 | eppigy | word |
| 19:29.03 | eppigy | it will be fun |
| 19:30.09 | benngard | p3nguin: id that hard-coded and have a catcher in mys case _0X. but the idea is the same |
| 19:30.10 | pigpen | I did however did where lets say _5555 goes here, and many more, I also made a _55XX which plated the ss-noservice as you noted, then hangup |
| 19:30.31 | pigpen | and I made one of these for each did block of 100 |
| 19:30.39 | benngard | i did hard-code like 600 numbers and it is working |
| 19:31.17 | ManxPower-work | I hardcode all the DIDs, then use a wildcard match to catch anything else and play an error message |
| 19:31.49 | ManxPower-work | I'm starting to hate ODBC |
| 19:31.51 | p3nguin | If every DID does something different, that's really the only choice. |
| 19:31.54 | pigpen | yeah, same here...I map out what needs to be mapped, then with this catch all (per block). |
| 19:32.11 | p3nguin | pigpen: Is that not working for you? |
| 19:32.20 | pigpen | no, that is working, just tested. |
| 19:32.24 | ManxPower-work | p3nguin: I'm not aware of anyone routing all their DIDs to the same extension |
| 19:32.27 | orangepower | ManxPower-work: no matter what i do.. it won't register |
| 19:32.28 | orangepower | :( |
| 19:32.31 | ManxPower-work | I'm sure there are ones out there. |
| 19:32.34 | orangepower | god i hate polycom |
| 19:32.36 | ManxPower-work | orangepower: "service iptables stop" |
| 19:32.36 | pigpen | I just don't want to piss off any customers at this point. |
| 19:32.49 | ManxPower-work | orangepower: they are the best phones on the market |
| 19:32.52 | ManxPower-work | ~phones |
| 19:32.53 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
| 19:33.11 | pigpen | polycom's have gone the distance for me. |
| 19:33.31 | orangepower | ManxPower-work: did that, reboot the phone now? |
| 19:33.50 | ManxPower-work | orangepower: you're not running in a browser that blocks scripts, are you? |
| 19:33.56 | orangepower | no |
| 19:33.56 | benngard | ManxPower-work: after my sucess with Avaya 9650 earlier today, maybe we shoul put them ay the boot |
| 19:34.02 | ManxPower-work | orangepower: if the change is applied the phone will reboot itself |
| 19:34.02 | orangepower | ManxPower-work: the settings save |
| 19:34.04 | benngard | should* |
| 19:34.11 | orangepower | yea, it does |
| 19:34.19 | orangepower | i even tried editing it on the phone interface |
| 19:34.28 | ManxPower-work | editting what? |
| 19:34.37 | orangepower | sip/line settinga |
| 19:34.38 | orangepower | s |
| 19:35.11 | ManxPower-work | orangepower: Almost everyone with polycoms use a provisioning server -- they are really designed for that. The web GUI appears to be an afterthought. |
| 19:35.14 | garymc | where does "pri-dialplan=local" go? what files? |
| 19:35.23 | ManxPower-work | garymc: there is no such option |
| 19:35.28 | orangepower | ah ok |
| 19:35.33 | *** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net) |
| 19:35.38 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:35.42 | ManxPower-work | garymc: are you meaning pridialplan=? |
| 19:35.56 | ManxPower-work | and if you are, did you see the note that says almost nobody ever has to set that option? |
| 19:36.06 | garymc | yes |
| 19:36.09 | p3nguin | pigpen: You've made a backup of extensions.conf before editing it, so it won't be hard to "cp -iv extensions.conf-20100103.bak extensions.conf" in the case of your edits are bad. |
| 19:36.14 | garymc | I do have to set it |
| 19:36.16 | garymc | I know I do |
| 19:36.22 | garymc | my telco require it |
| 19:36.29 | ManxPower-work | orangepower: look here: http://www.fnords.org/~eric/polycom-config-examples/ |
| 19:36.40 | orangepower | ManxPower-work: would the firmware be too old? 3.1.2.0306 is what's on the phone |
| 19:36.45 | ManxPower-work | garymc: if you set that you won't be able to make toll or international calls. |
| 19:36.52 | ManxPower-work | garymc: you set it in chan_dahdi.conf |
| 19:37.17 | garymc | I hve to have it |
| 19:37.17 | titter | How would you guys load test a * fax server? I only have 4 machines ... but want to hit it with a lot more |
| 19:37.38 | ManxPower-work | orangepower: upgrading a VVX to anything higher requires you to fillow a whitepaper or rick brickig the phone. |
| 19:37.42 | p3nguin | pigpen: Once you have made your changes and are confident in the matches for those unused DIDs, "sip reload" and see how it goes. |
| 19:37.57 | orangepower | ... |
| 19:38.50 | pigpen | p3nguin, yeah, it works fine. |
| 19:39.14 | pigpen | I have been running many systems for years, but mine is the only one that has this many blocks, and I have never taken the time to address it. |
| 19:40.06 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:42.04 | FinboySlick | I'm trying to figure out if a phone is already in a conversation before I Dial() to it. I ChanIsAvail(SIP/555), then I check ${AVAILSTATUS}... It always seems to be zero. |
| 19:42.59 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 19:43.09 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 19:43.11 | FinboySlick | I was sort of expecting 2 if it's in use. |
| 19:43.54 | ManxPower-work | FinboySlick: what version of Asterisk |
| 19:43.57 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-042-222-050.hsi3.kabel-badenwuerttemberg.de) |
| 19:44.31 | FinboySlick | ManxPower-work: Still on my outdated 1.4.22. I promise I'll upgrade soon ;) |
| 19:44.36 | [TK]D-Fender | FinboySlick: You're calling it wrong, and Your phones are likely set up wrong |
| 19:45.03 | ManxPower-work | FinboySlick: there is a calllimit (or similar named option) that must be set for Asterisk to track device states. |
| 19:45.15 | ManxPower-work | you can set it to 99 for all Asterisk cares, as long as it's set to something. |
| 19:45.35 | FinboySlick | ManxPower-work: Sounds good, I'll look into that then. |
| 19:45.54 | ManxPower-work | FinboySlick: see sip.conf.sample in the Asterisk source code directory |
| 19:46.07 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
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| 19:57.41 | *** part/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 19:58.37 | garymc | Sangoma has just messed with my config. I now cant recieve any calls inwards |
| 19:58.38 | evilbit | hmm, res_config_mysql.c keep ERROR'ing out on me. I have /etc/asterisk/cdr_mysql.conf but running asterisk in debug doesn't seem to show asterisk using the info from that file |
| 19:59.35 | garymc | [TK]D-Fender : you was right earlier I need pridialplan=unknown in a config file. But sangoma was messing with my settings and now incoming calls dont happen |
| 19:59.47 | garymc | outgoing work fine now |
| 20:00.41 | garymc | any help or advice here would be great as my head is soooo burnt |
| 20:01.10 | p3nguin | Go fix whatever got changed. |
| 20:01.24 | garymc | he said he chnaged nothing and I cant find it |
| 20:01.39 | p3nguin | diff it against your backup. |
| 20:02.15 | garymc | again he didnt do backup i asked him too he forgot |
| 20:02.23 | garymc | i wasnt sure on how too |
| 20:03.10 | garymc | here is a pri debug of incoming call. The external caller hears no sound. http://pastebin.ca/1786229 |
| 20:04.23 | p3nguin | Making backups is easy. For extensions.conf, for example, cp -iv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf-$(date +%Y%m%d%H%M).bak would make a suitable backup. |
| 20:05.36 | garymc | anyone see anything in the pastebin that would give a clue as to whats going on? |
| 20:05.54 | moy | garymc: I thought you said incoming did not happen ... so, the call does not happen at all or there is just no audio? |
| 20:06.03 | *** join/#asterisk m0t3jl (~m0t3jl@ip-41.galance.net) |
| 20:06.18 | moy | and who did you speak to in Sangoma? |
| 20:07.41 | garymc | no phones in offic ering and no audio for the caller either |
| 20:08.07 | *** join/#asterisk Ujjain (dennis00@unaffiliated/ujjain) |
| 20:08.22 | Ujjain | Does anybody here have a bluetooth headset? Or preferably Toshiba even. |
| 20:08.24 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:09.22 | garymc | moy sangoma tech Shaunt |
| 20:09.27 | moy | garymc: well, the call arrives just fine, but number 876699 does not seem routable, does the number look chopped to you? what is at the other end of the link? |
| 20:09.29 | garymc | I was on the phone for 1 hour |
| 20:09.44 | garymc | it is chopped |
| 20:09.59 | garymc | the full number is 4876699 |
| 20:11.25 | garymc | with area code its 01514876699 |
| 20:12.20 | orangepower | what ports do I open for sip based asterisk? i'm just making sure i have them all |
| 20:12.51 | p3nguin | 5060 for SIP, 10000-20000 for RTP (or whatever range you configured) |
| 20:13.02 | orangepower | ok |
| 20:13.24 | p3nguin | Those are UDP, by the way. |
| 20:14.09 | garymc | anyone at all help me get this fixed? |
| 20:14.22 | moy | garymc: from the PRI link already comes chopped, [70 07 81 38 37 36 36 39 39] .... |
| 20:14.28 | grayhame | has anyone ever seen asterisk 1.6 lose all of it's contexts and extensions after a reload? (at which point another reload restored them) |
| 20:14.28 | voipmonk | gary you're still at it? |
| 20:14.43 | m0t3jl | Anyone has any experience with the Skype for Asterisk or any other (not officially supported) solution for SIP to Skype calls? What I would like to know is whether it is possible to somehow allow my users to use the ordinary Skype client on their desktops for chat and to only direct calls to my Asterisk PBX. Thanks a lot |
| 20:14.46 | *** join/#asterisk bmg505 (~leon@196-209-77-216-rndf-esr-5.dynamic.isadsl.co.za) |
| 20:15.18 | moy | garymc: meaning the other end is sending 876699 |
| 20:15.36 | moy | and asterisk can't route it |
| 20:15.48 | garymc | why would that be happening now? |
| 20:15.52 | paulc | garymc: 0151 = Liverpool? ;-) |
| 20:15.56 | garymc | yes |
| 20:16.23 | p3nguin | That sounds like a terrible name for a city. |
| 20:16.30 | paulc | laughs at how many UK area codes he knows, randomly |
| 20:16.30 | moy | garymc: that I don't know, with the information you have given, that's all I can say |
| 20:16.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 20:16.42 | p3nguin | Reminds me of something like Kidneyriver or Spleenlake. |
| 20:16.46 | paulc | p3nguin: the locals can be a bit rough too ;-) |
| 20:17.08 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.14 | paulc | garymc: so did the telco change the number of DDI digits they're sending you? or was it always 6? |
| 20:17.20 | Ujjain | LIVERPOOL! |
| 20:18.09 | *** join/#asterisk qjb (~qjb@95.99.82.7) |
| 20:18.15 | *** join/#asterisk sebbl (~Momofu@HSI-KBW-078-042-222-050.hsi3.kabel-badenwuerttemberg.de) |
| 20:18.24 | garymc | it was always 6 for local |
| 20:18.58 | bmoraca_work | garymc: so what, exactly, is the problem? |
| 20:19.26 | garymc | no calls are coming in. the caller hear nothing |
| 20:20.00 | paulc | garymc: Does the call hit your box though? You see it in the dialplan, activity in the console etc? |
| 20:20.10 | bmoraca_work | garymc: when you say "no calls are coming in", do you mean that Asterisk is rejecting it or does it mean you have no activity on your PRI? |
| 20:20.19 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 20:20.50 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 20:21.03 | garymc | asterisk is rejecting i think |
| 20:21.16 | bmoraca_work | garymc: well, in that case, pastebin a failed call log |
| 20:21.21 | garymc | http://pastebin.ca/1786229 check out this pastebin of an incoming call. no phones in office ring |
| 20:21.27 | garymc | and caller hears nothing |
| 20:22.16 | bmoraca_work | garymc: shame on your for asking for help with freepbx...but, your problem is that you don't have an inbound route to phone number 876699. |
| 20:22.34 | garymc | i do |
| 20:22.42 | bmoraca_work | garymc: no, you don't. |
| 20:22.52 | bmoraca_work | garymc: oh. balls. |
| 20:23.11 | paulc | decides now might be a good time to skip out for lunch... |
| 20:23.14 | bmoraca_work | garymc: the context your PRI is pointing to is wrong. |
| 20:23.32 | bmoraca_work | garymc: "876699@from-internal" <==== that's not right. |
| 20:23.46 | garymc | what should it be |
| 20:23.47 | bmoraca_work | garymc: go fix your dahdi-channels.conf file |
| 20:23.49 | garymc | and how do i fix it |
| 20:24.04 | bmoraca_work | garymc: it should be "from-trunk". |
| 20:25.43 | garymc | ok i changed them to from trunk. should i amportal restart? |
| 20:25.59 | ManxPower-work | freepbx support on #asterisk. |
| 20:26.01 | garymc | or wanrouter? |
| 20:26.04 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 20:26.12 | p3nguin | dahdi restart? |
| 20:26.18 | bmoraca_work | i suppose you could...you don't need to. all you need to do is restart asterisk |
| 20:27.43 | benngard | ~phones |
| 20:27.44 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
| 20:29.56 | garymc | bmoraca_work that fixed it thanks |
| 20:30.08 | garymc | one thing is now my Zap extension cant dial out |
| 20:30.29 | bmoraca_work | garymc: without a failed call log, there's nothing i can do. |
| 20:30.50 | garymc | ok |
| 20:32.49 | bmoraca_work | ok... |
| 20:34.25 | *** join/#asterisk Orbixx (Orbixx@office.exoware.net) |
| 20:34.31 | Orbixx | WARNING[18761]: app_meetme.c:1097 build_conf: Unable to open pseudo device |
| 20:34.41 | Orbixx | Cause, solution? |
| 20:34.53 | Orbixx | dahdi is installed |
| 20:35.10 | Orbixx | Using a trunk for my incoming calls. |
| 20:35.14 | Orbixx | No physical line. |
| 20:39.09 | [TK]D-Fender | Orbixx: Happens if DAHDI isn't initialized or if * wasn't compiled AFTER your DAhDI timer was installed |
| 20:39.31 | Orbixx | [TK]D-Fender: How may I check if dahdi is initialized? |
| 20:40.06 | [TK]D-Fender | Orbixx: Did you do it or not? |
| 20:40.28 | Orbixx | [TK]D-Fender: I compiled and installed dahdi and dahdi-tools, then compiled and installed the new 1.6.2.2 * |
| 20:41.02 | p3nguin | Did you load the kernel module? |
| 20:41.04 | [TK]D-Fender | Orbixx: Stop *. "dahdi_cfg -vvvv", Start *. |
| 20:41.26 | Orbixx | line 0: Unable to open master device '/dev/dahdi/ctl' |
| 20:42.28 | Orbixx | doh |
| 20:42.31 | Orbixx | forgot to modprobe |
| 20:42.43 | p3nguin | Damn, I wish I had thought of that. |
| 20:43.02 | Orbixx | Hmm. |
| 20:43.08 | Orbixx | Oddly I'm still getting the same initial issue. |
| 20:43.14 | Orbixx | Althoufh dahdi_cfg works now. |
| 20:43.29 | Orbixx | Or rather, it doesn't error... |
| 20:43.36 | Orbixx | Is "0 channels to configure." bad? |
| 20:43.44 | [TK]D-Fender | no |
| 20:44.02 | evilbit | hmm, so I found my answer for mysql. but is there a difference btwn asterisk and asterisk realtime? |
| 20:44.32 | Orbixx | Still getting: WARNING[19262]: app_meetme.c:1097 build_conf: Unable to open pseudo device |
| 20:44.55 | Chainsaw | Orbixx: So provide it with a timing source. |
| 20:45.10 | Chainsaw | Orbixx: Either with some Digium hardware or with zt/dahdi_dummy. |
| 20:45.35 | Orbixx | Chainsaw: I believe I need dummy. |
| 20:45.48 | Chainsaw | Orbixx: If you don't have hardware for it, indeed. |
| 20:45.58 | [TK]D-Fender | Orbixx: What did you you just modprobe? |
| 20:46.03 | p3nguin | lol |
| 20:46.06 | Orbixx | [TK]D-Fender: modprobe dahdi |
| 20:46.15 | [TK]D-Fender | Orbixx: dahdi_dummy <- |
| 20:46.33 | p3nguin | He loaded dahdi, dummy! |
| 20:46.57 | Orbixx | SUCCESS! |
| 20:47.11 | Orbixx | Many thanks all. |
| 20:47.21 | Chainsaw | raises a thumb |
| 20:48.02 | [TK]D-Fender | chops it off |
| 20:48.27 | [TK]D-Fender | thumbs up to War Amps! |
| 20:50.49 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 20:50.54 | *** join/#asterisk engrxyz (~jkjkjk@host81-143-50-92.in-addr.btopenworld.com) |
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| 20:58.50 | raden_work | on our main menu we have press 1 for sales press 2 for tech support etc.... is there a way to make a web interface to control what extension 1 or 2 goes to etc ????? |
| 20:59.54 | Orbixx | How does one declare the start time and the finish time of a conference? |
| 21:01.44 | *** join/#asterisk fofware (~chatzilla@190.30.113.171) |
| 21:01.52 | [TK]D-Fender | raden_work: Yes.... its called "programming" |
| 21:02.06 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 21:02.07 | [TK]D-Fender | raden_work: Millions have tried it and lived happy lives |
| 21:02.13 | Warp4 | LOL |
| 21:02.51 | Warp4 | not sure if anyone can help or not, but is it possible to initiate a call over the AMI? |
| 21:03.22 | raden_work | <PROTECTED> |
| 21:03.38 | [TK]D-Fender | \o/ |
| 21:04.23 | [TK]D-Fender | raden_work: Well why WOULDN'T it be possibly to make a web interface to changing how an IVR works? You know... like all the other GUI's out there already already do? |
| 21:04.55 | raden_work | you have a GUI recomendation |
| 21:05.06 | voipmonk | freepbx |
| 21:05.24 | raden_work | [TK]D-Fender, I apologize, the idiots here think i can wave my magic mouse and change the whole phone system :( |
| 21:05.38 | raden_work | voipmonk, i have asterisk, thats what i have to work with |
| 21:05.39 | Warp4 | lol |
| 21:05.53 | [TK]D-Fender | raden_work: You do.... its just takes about 5,000,000 passes to accomplish ;) |
| 21:05.59 | raden_work | lol |
| 21:06.23 | raden_work | proper implementation takes proper planning, Im never given the plans just told what to implement step by step |
| 21:06.40 | [TK]D-Fender | raden_work: You can make some easier pages yourself for limited functionality. for more free-form, thats another matter. |
| 21:07.23 | raden_work | well I know php & mysql just need to have a list of lets say option one and a scroll box next to it of extensions |
| 21:07.28 | raden_work | if that sort of thing is possible |
| 21:07.31 | raden_work | just one simple pag4e |
| 21:07.46 | voipmonk | raden_work: did you ask for a gui recommendation then tell me after I gave you one that asterisk is what you have to work with? |
| 21:07.46 | [TK]D-Fender | raden_work: This would be pretty easy to do. |
| 21:08.34 | raden_work | [TK]D-Fender, guess my confussion is where are the options stored ? do i use mysql or do i need to write to asterisk and pull info from asterisk |
| 21:09.13 | [TK]D-Fender | raden_work: As a recommendation I would store your choices in an external structure and build * configs the way FreePBX does for example |
| 21:09.26 | [TK]D-Fender | raden_work: Somewhat of a cut&paste engine |
| 21:09.59 | raden_work | so make it so php edits a file and swaps into place ? |
| 21:10.47 | [TK]D-Fender | raden_work: I'd say "rebuilds" an #include -able conf file |
| 21:11.20 | *** join/#asterisk Blue_Day (~Kern@71.105.22.252) |
| 21:11.31 | raden_work | i get where your going with that |
| 21:15.21 | [TK]D-Fender | raden_work: give them a few common targets that map to your extension structure. Maybe let it parse out things like sip.conf to pick valid targets based on an assumed extension dialplan structure, etc |
| 21:15.50 | raden_work | yea, I'll get it |
| 21:15.57 | [TK]D-Fender | raden_work: not too hard... depends how much you want to phase in at a time. |
| 21:16.00 | raden_work | i was under the assumption i could just use a database somehow |
| 21:16.08 | [TK]D-Fender | raden_work: You can... |
| 21:16.26 | bmoraca_work | raden_work: can always use realtime extensions... |
| 21:16.27 | raden_work | just need IVR 1,2,3,4,5,6 to be able to select 1 of 6 extensions each |
| 21:16.40 | raden_work | really ? |
| 21:16.58 | [TK]D-Fender | raden_work: thats one place to store the "structure" that you use to generate the dialplan. Or you could reun the whole thing in an AGI and that pulls the actions right out of the DB. |
| 21:17.10 | bmoraca_work | yep...it's a pain in the ass, but you could do it without having to modify any config files or reloading asterisk at all |
| 21:17.17 | [TK]D-Fender | raden_work: how many menus are you looking to give them control over? |
| 21:17.26 | [TK]D-Fender | raden_work: Just one? |
| 21:18.17 | *** join/#asterisk basty (~basty@212.218.65.195) |
| 21:18.18 | basty | Hi |
| 21:18.48 | raden_work | yea just one menu |
| 21:18.52 | raden_work | that has 6 options |
| 21:19.07 | basty | i just updated to asterisk 1.4.29 including the init scripts. After I reboot I noticed, that Asterisk is eating 100% CPU. When I stop it and run "asterisk -vvvv" everything is normal. There must be something wrong in the init scripts...anyone knows what ? |
| 21:19.13 | raden_work | only 6 extensions well 8 extensions ( 2 cellphones that we redirect to ) |
| 21:19.31 | raden_work | basty, why not just use 1.6 ? |
| 21:19.44 | [TK]D-Fender | raden_work: then I would do something entirely different |
| 21:19.53 | p3nguin | I'm using 1.4.29 and there's nothing wrong with it. |
| 21:19.54 | [TK]D-Fender | raden_work: this is an el-cheapo job. |
| 21:19.56 | basty | raden_work: because I need it for a big productiv system...I would prefer using asterisk 1.2. tho ;-) |
| 21:20.00 | black | Well, this weekend my fiancee is buying a wedding dress. |
| 21:20.00 | raden_work | what would you suggests ? |
| 21:20.05 | black | She's like 'give me your credit card' |
| 21:20.06 | black | =p |
| 21:20.13 | [TK]D-Fender | raden_work: use AstDB to store the "target" for each option and use an IVR to PROGRAM this other one. |
| 21:20.13 | raden_work | just needs to be so when boss is gone etc..... or im at home i can switch things |
| 21:20.20 | [TK]D-Fender | radethey'll jsut DTMF in targets, etc./ |
| 21:20.21 | raden_work | or if somone sont show up i can change things |
| 21:20.50 | raden_work | ok |
| 21:20.50 | [TK]D-Fender | raden_work: this way you can reprogram from whatever phone... without even need of a computer |
| 21:21.04 | [TK]D-Fender | raden_work: Snow day? Phone in and redirect things |
| 21:21.14 | [TK]D-Fender | raden_work: Pure dialplan |
| 21:21.18 | raden_work | boss dude wants it on a browser :( |
| 21:21.23 | [TK]D-Fender | raden_work: Make the solution scale to the need |
| 21:21.28 | raden_work | i agree |
| 21:21.35 | raden_work | he wants a call log on browser too |
| 21:21.36 | raden_work | etc... |
| 21:21.42 | [TK]D-Fender | raden_work: easy enough to make browser-able as well. |
| 21:21.43 | raden_work | so there a few small things i need to get done |
| 21:21.49 | raden_work | i like that idea much better simple .... |
| 21:22.05 | raden_work | i will go with that solution do a lil reading and implement |
| 21:22.06 | raden_work | thanks |
| 21:22.40 | [TK]D-Fender | raden_work: Idea : make other things like Queue's easily codable extensions. thatway your IVR's are Goto targets 100% |
| 21:22.47 | *** join/#asterisk Skeeter- (Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 21:22.51 | [TK]D-Fender | raden_work: You can template this to make it easy |
| 21:22.58 | raden_work | template ? |
| 21:24.10 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:24.47 | [TK]D-Fender | raden_work: make extens to go to queues, etc. That way you enter "targets" for IVR choices taht can all be numeric and thus programmable by phone |
| 21:31.49 | [TK]D-Fender | checkout time... later |
| 21:32.43 | raden_work | later |
| 21:39.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:49.50 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 21:51.49 | *** join/#asterisk sulan (ksjoberg@89.253.95.143) |
| 21:53.32 | Katty | hungry :< |
| 21:53.38 | eppigy | :< |
| 21:53.46 | eppigy | huggles Katty |
| 21:54.10 | Katty | hugs eppigy |
| 21:58.59 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 22:00.37 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
| 22:01.06 | niekvlessert | hello people! |
| 22:01.20 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
| 22:01.22 | niekvlessert | i'm trying to grasp the concepts of sip proxies |
| 22:01.47 | niekvlessert | anyone has any experience using asterisk in conjunction with a centralized sip proxy? |
| 22:02.03 | Katty | conjunction junction |
| 22:02.06 | Katty | what's your function!!! |
| 22:02.31 | niekvlessert | I'm dutch, did I mispell? :) |
| 22:02.44 | niekvlessert | spelled wrong, how do you say.... |
| 22:02.51 | Katty | it's a song. |
| 22:02.58 | Katty | http://www.youtube.com/watch?v=mkO87mkgcNo |
| 22:03.11 | niekvlessert | ah ok :) |
| 22:03.25 | Katty | it's intended for young Grammar students. |
| 22:04.04 | niekvlessert | ok well i guess i'm on the level of a young grammar student in English :) |
| 22:04.27 | niekvlessert | so I should watch |
| 22:04.33 | Katty | sure |
| 22:04.49 | niekvlessert | but first I would like to know some more about sip proxies if that's ok with you. |
| 22:05.31 | *** join/#asterisk fofware (~chatzilla@host171.190-30-113.telecom.net.ar) |
| 22:05.33 | niekvlessert | ah i'm sorry, here we go again... grammar is math |
| 22:05.42 | Katty | but = conjunction! |
| 22:06.48 | Katty | eppigy: what's fur dinner...i'm starving |
| 22:07.01 | niekvlessert | grammatica (f) |
| 22:07.02 | niekvlessert | spraakkunst, grammatica { http://www.babylon.com/definition/grammar/Dutch } |
| 22:07.14 | niekvlessert | ah now I'm sure, it's language |
| 22:07.27 | eppigy | Katty: steak and baked potato :D |
| 22:07.46 | niekvlessert | well, what about the proxy people? |
| 22:08.04 | niekvlessert | anyone knows what happens if a sip reinvite cannot be done when using opensips? |
| 22:08.10 | Katty | niekvlessert: patience is a virtue. |
| 22:08.25 | Katty | eppigy: that sounds gooooood. |
| 22:08.35 | Katty | eppigy: i am ready to gnaw my arm |
| 22:08.42 | *** join/#asterisk brezular (~brezular@adsl-dyn162.78-99-111.t-com.sk) |
| 22:08.54 | eppigy | Katty: dont do that :< |
| 22:09.00 | eppigy | the consequences will be bad |
| 22:09.05 | eppigy | and far reaching |
| 22:09.07 | Katty | yes probably |
| 22:09.25 | Katty | gnaws on blackberry phone |
| 22:09.34 | eppigy | yesh |
| 22:09.36 | niekvlessert | what is gnaws |
| 22:09.41 | niekvlessert | like chewing? |
| 22:09.43 | eppigy | I must leave to go home now |
| 22:09.46 | Katty | yesh. |
| 22:09.47 | Katty | kbai |
| 22:09.57 | eppigy | <33 |
| 22:10.13 | niekvlessert | you guys know brocoli overthere? |
| 22:11.10 | Katty | green |
| 22:11.13 | Katty | good with cheese? |
| 22:11.24 | niekvlessert | ah you do :) |
| 22:12.02 | Katty | mhmm |
| 22:12.05 | Katty | i do likes me some broccoli |
| 22:12.28 | niekvlessert | well I just gave the last to the rabbit so no go... |
| 22:13.47 | Katty | :< |
| 22:13.47 | Katty | rabbit needs to eat too, i suppose |
| 22:14.04 | niekvlessert | yeah, although he's not doing a lot |
| 22:14.23 | niekvlessert | but it's -5 degrees celcius at nights overhere |
| 22:14.30 | niekvlessert | so he deserves some nice food |
| 22:14.33 | Katty | what kind of rabit? |
| 22:14.51 | niekvlessert | a giant dwarf rabbit |
| 22:15.11 | niekvlessert | he's way to big for his race :) |
| 22:15.21 | niekvlessert | lots of fur |
| 22:15.22 | niekvlessert | gray |
| 22:15.25 | lanning | is that like a jackalope? :) |
| 22:15.42 | Katty | i bet he's pretty |
| 22:16.09 | niekvlessert | :) |
| 22:16.20 | niekvlessert | well he's ok |
| 22:16.24 | niekvlessert | little grumpy though |
| 22:16.29 | niekvlessert | he can actually growl |
| 22:16.33 | niekvlessert | no kidding! |
| 22:16.55 | Katty | :> |
| 22:17.04 | Katty | i demand more lettuces! *grrrr* |
| 22:19.47 | niekvlessert | yeah true |
| 22:20.11 | niekvlessert | when you go in and get his food tray |
| 22:20.18 | niekvlessert | he'll try to destroy you |
| 22:20.29 | *** join/#asterisk cweagans (~cweagans@66-193-105-126.static.twtelecom.net) |
| 22:20.33 | niekvlessert | although you are just getting it to fill it with food... |
| 22:20.46 | niekvlessert | so he's not that bright |
| 22:20.48 | cweagans | do I need to open a port in my firewall for incoming SIP calls for Asterisk? |
| 22:20.53 | cweagans | if so, which port? |
| 22:20.59 | Katty | infobot: nat? |
| 22:21.00 | infobot | from memory, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:21.05 | Katty | infobot: natting? |
| 22:21.28 | niekvlessert | cweagans: 5060 UDP for SIP. But for RTP its a lot of extra ports |
| 22:21.45 | cweagans | ok. what's RTP? Do i need it? |
| 22:21.51 | cweagans | is a phone system noob. |
| 22:21.52 | niekvlessert | wow you're a newbie :) |
| 22:21.53 | Katty | heh |
| 22:21.54 | cweagans | :) |
| 22:21.59 | Katty | there's more than ports, too, but hey |
| 22:22.00 | cweagans | just started yesterday |
| 22:22.05 | Katty | refer to your rtp.conf |
| 22:22.21 | cweagans | well, I just want to make sure my network isn't going to cause me issues at this point |
| 22:22.36 | cweagans | the PBX has to be behind a firewall =/ |
| 22:22.56 | niekvlessert | sip is not happy with a lot of nat |
| 22:23.10 | niekvlessert | do you have asterisk behind a nat firewall or a bridged firewall? |
| 22:23.20 | niekvlessert | http://www.siptutorial.net/SIP/index.html |
| 22:23.29 | cweagans | will be behind a nat firewall |
| 22:23.38 | Katty | there's a how to guide on the wiki |
| 22:24.04 | niekvlessert | first try to connect your phone to asterisk in your lan |
| 22:24.15 | niekvlessert | will give you enough headache to start with... |
| 22:24.27 | Katty | let me see if i can find it |
| 22:24.31 | Katty | oh |
| 22:24.39 | Katty | you've not connected local devices yet? |
| 22:24.48 | *** join/#asterisk lernest (~ircap@c-66-229-203-18.hsd1.fl.comcast.net) |
| 22:25.01 | cweagans | I don't even have the devices available yet. they won't be in till next week |
| 22:25.16 | niekvlessert | lol |
| 22:25.19 | niekvlessert | get a softphone |
| 22:25.21 | niekvlessert | like zoiper |
| 22:25.26 | niekvlessert | try to register it with asterisk |
| 22:25.40 | Katty | backs away slowly |
| 22:25.50 | niekvlessert | tip for zoiper: open advanced options and disable stun |
| 22:25.52 | cweagans | doesn't even have the asterisk server available yet. |
| 22:25.54 | niekvlessert | took me 15 minuts..... |
| 22:26.01 | Katty | LOL |
| 22:26.02 | cweagans | not till next week |
| 22:26.03 | Katty | nice. |
| 22:26.08 | niekvlessert | well plz leave the channel then and start installing |
| 22:26.11 | cweagans | haha |
| 22:26.24 | cweagans | port 5060, does that sound right? |
| 22:26.30 | niekvlessert | udp yeah.. |
| 22:26.45 | cweagans | oh right, you said that. I just found it in another tutorial =P |
| 22:26.57 | lernest | any help with outbound call ? |
| 22:27.16 | cweagans | alright, thanks niekvlessert and Katty and voipmonk :) |
| 22:27.24 | *** join/#asterisk ktwilight_ (~keliew@158.92-136-217.adsl-dyn.isp.belgacom.be) |
| 22:27.47 | niekvlessert | my question again, what will happen if two asterisks are connected with a opensips in the middle, and the two asterisks cannot communicate directly? |
| 22:28.02 | niekvlessert | what will happen with a conversation between the two asterisks? |
| 22:28.17 | niekvlessert | will opensips function as a RTP proxy? guess not right? |
| 22:28.34 | niekvlessert | will i need http://mediaproxy.ag-projects.com/ or something? |
| 22:29.03 | *** join/#asterisk beters (~beters@c-68-54-100-46.hsd1.ms.comcast.net) |
| 22:29.31 | niekvlessert | well i guess all the people don't want me to use opensips :) |
| 22:29.47 | beters | can someone help me out with something in my dial plan? |
| 22:29.54 | niekvlessert | maybe... |
| 22:29.56 | beters | im using a call to a php script |
| 22:29.58 | beters | using lynx |
| 22:30.27 | beters | but the address that lynx needs to dump has to have quotation marks around it |
| 22:30.36 | beters | for the system command |
| 22:30.48 | beters | i tried escaping it with \" |
| 22:30.54 | beters | but it just breaks it |
| 22:31.10 | niekvlessert | what do you mean by needs to dump? |
| 22:31.21 | *** join/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 22:31.24 | beters | well i call it just by dumping it |
| 22:31.26 | beters | dont have to |
| 22:31.28 | beters | but i do |
| 22:31.30 | beters | either way |
| 22:31.40 | beters | it has php variables attached to it |
| 22:32.02 | niekvlessert | i don't get it |
| 22:32.04 | beters | so if it doesnt have quotes around the address, lynx doesnt work |
| 22:32.13 | niekvlessert | so u are in the dir where you want a file to be? |
| 22:32.24 | niekvlessert | and then you use lynx with a php script to get that file in there? |
| 22:32.34 | beters | no i dont need it to actually do anything but call the url |
| 22:32.40 | beters | to pass the php variables |
| 22:32.44 | niekvlessert | yeah |
| 22:32.45 | beters | to put them in a database |
| 22:32.47 | niekvlessert | so what's you url |
| 22:35.16 | beters | http://paste-it.net/public/x00e01a/ |
| 22:35.26 | beters | thers the line in the dial plan |
| 22:35.59 | niekvlessert | ah |
| 22:36.02 | beters | if i take the quotes off of the link that lynx is calling, the page called, but none of the variables at the end are passed |
| 22:36.07 | niekvlessert | well don't know why the " don't get in there |
| 22:36.13 | niekvlessert | but you can always use agi... |
| 22:36.42 | beters | just like if you run it from the command line without them |
| 22:36.43 | *** join/#asterisk juanjoc_ (~juanjoc@200.69.219.113) |
| 22:37.19 | beters | this is just a simple way to pass variables, without having to reinvent the wheel for me |
| 22:37.39 | niekvlessert | shouldn't it be System(' blabla ')? |
| 22:37.47 | niekvlessert | but i still would use agi |
| 22:38.28 | juanjoc_ | Hi. Has anybody experienced TDM PCI Master aborts with a Digium TDM400P card and Asterisk 1.6.0.2/DAHDI 2.2.0.2? |
| 22:39.36 | beters | well crap, guess ill have to just figure it out then |
| 22:39.37 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
| 22:40.19 | niekvlessert | what about the System(' blabla ')?? |
| 22:40.30 | beters | what do you mean? |
| 22:40.51 | Katty | System(echo foo and ${STUFF} | thing) |
| 22:41.06 | beters | it is in a System() |
| 22:41.43 | Katty | hungryhungryhungry |
| 22:44.55 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 22:46.18 | niekvlessert | exten => 900,53,System('/usr/bin/lynx "http://www.blahblah.com/alltest/eagletest/index.php?callerid=${CALLERID(num)}"\&"zip=${APCODEE}"\&"calleridname=${CALLERID(num)}"\&"loadno=${APCODEPRO}"\&"temp=${APCODETEMP}"') |
| 22:46.53 | niekvlessert | but why not execute an agi script, give it the variables and do the stuff in php?? |
| 22:47.39 | Katty | maybe he doesn't know php |
| 22:47.47 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 22:48.01 | niekvlessert | i guess |
| 22:48.13 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
| 22:48.20 | niekvlessert | he might say that he doesn't know what I mean? |
| 22:48.26 | niekvlessert | if that's the case... |
| 22:48.32 | Katty | doesn't matter. |
| 22:48.39 | niekvlessert | right... |
| 22:48.52 | niekvlessert | is your house empty katty? |
| 22:48.57 | niekvlessert | no food in there? |
| 22:49.01 | Katty | i'm at work |
| 22:49.05 | niekvlessert | ah |
| 22:49.09 | Katty | 10min till i go home and raid the fridge |
| 22:49.21 | *** part/#asterisk FinboySlick (~shark@74.117.40.10) |
| 22:49.53 | niekvlessert | sounds good |
| 22:50.03 | niekvlessert | btw i've got my answer already at the opensips channel. :) |
| 22:50.33 | Katty | super |
| 22:51.06 | niekvlessert | yeah pretty nice |
| 22:51.08 | niekvlessert | fyi |
| 22:51.10 | Deeewayne | ~thebook |
| 22:51.11 | infobot | extra, extra, read all about it, thebook is 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://astbook.asteriskdocs.org |
| 22:51.16 | Deeewayne | lernest, ^^ |
| 22:51.34 | niekvlessert | opensips has per default a mediaproxy in there since 1.6 |
| 22:52.14 | Katty | hi Deeewayne |
| 22:52.17 | Katty | hugs on Deeewayne |
| 22:53.26 | Deeewayne | hugs Katty while twirling |
| 22:53.35 | beek | hugs on Katty |
| 22:53.54 | Katty | hi beekster. |
| 22:53.55 | mcrownover | i imagine the answer is if it aint broke why fix it - but should i upgrade from asterisk 1.4.23 to 1.6?? |
| 22:53.56 | Deeewayne | file: you should hug Katty too :-( |
| 22:53.56 | Katty | hugs beters |
| 22:54.17 | beek | Katty: I'm sure that beters enjoyed the hug! |
| 22:54.29 | Katty | oh |
| 22:54.32 | Katty | hugs beek |
| 22:54.32 | Katty | sorry |
| 22:54.35 | beek | :D |
| 22:55.49 | Katty | 5 minutes >.< |
| 22:56.05 | niekvlessert | mcrownover: if it ain't broke don't fix it |
| 22:56.30 | Katty | that's a poor atitude. |
| 22:56.43 | Katty | if everyone had that attitude, nothing would get better. |
| 22:57.04 | Deeewayne | adds some food to Katty's squirrel feeder |
| 22:57.13 | Katty | Deeewayne: they almost out? |
| 22:57.14 | mcrownover | niekvlessert: - but i need to break something today... :-) |
| 22:57.15 | Deeewayne | Katty, feeder improved! |
| 22:57.18 | Katty | oh |
| 22:57.20 | Katty | yes |
| 22:57.32 | Katty | feeder is always improved by the precense of visitors |
| 22:57.42 | Deeewayne | prepares to visit |
| 22:57.45 | Katty | checks video feed |
| 22:58.02 | Katty | my |
| 22:58.08 | Katty | i filled everything up this morning |
| 22:58.11 | Katty | and the lighthouse is nearly empty! |
| 22:58.16 | Deeewayne | Katty, snow! |
| 22:58.25 | Katty | it's mostly melted now, but we had 8 inches (= |
| 22:58.28 | Deeewayne | I haven't looked in a couple weeks |
| 22:58.47 | mcrownover | so where is everyone from?? i'm in south alabama |
| 22:58.55 | niekvlessert | holland |
| 22:59.20 | Deeewayne | from NJ, living in Huntsville AL |
| 22:59.53 | beters | got it working |
| 22:59.56 | beters | finaly |
| 23:00.02 | beters | hattiesburg ms |
| 23:00.05 | beters | not far from you |
| 23:00.07 | mcrownover | Deeewayne - u work for digium?? |
| 23:00.26 | niekvlessert | good work beters |
| 23:00.35 | niekvlessert | in holland beter is the same as better |
| 23:00.45 | Deeewayne | mcrownover, no |
| 23:00.45 | niekvlessert | so beters beters |
| 23:00.55 | beters | lol |
| 23:01.17 | beters | this is how i had to escape that character |
| 23:01.23 | beters | "\"" |
| 23:01.35 | beters | just to keep the " |
| 23:01.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:01.52 | mcrownover | Deeewayne - thought u might since they are located up there |
| 23:02.21 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754708.dsl.bell.ca) |
| 23:02.37 | niekvlessert | ah that's pretty regular |
| 23:02.45 | niekvlessert | in most programming languages it's like that |
| 23:02.55 | niekvlessert | but I STILL would use agi ;) |
| 23:05.29 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 23:05.48 | niekvlessert | you should try sed or awk |
| 23:06.12 | niekvlessert | if you want to get '"\"'"''"'\\" mysteries going on |
| 23:06.28 | beters | lol |
| 23:06.36 | beters | no thanks, im getting enough now |
| 23:10.27 | niekvlessert | anyone tried sipsak? |
| 23:10.34 | niekvlessert | error: didn't received '200 OK' on register (see above). aborting |
| 23:13.55 | niekvlessert | Using Sipsak to reboot a Snom phone |
| 23:14.02 | niekvlessert | <PROTECTED> |
| 23:14.11 | niekvlessert | :) |
| 23:14.35 | niekvlessert | well i'm going to bed |
| 23:14.38 | niekvlessert | see you later |
| 23:15.25 | *** join/#asterisk garymc (~chatzilla@host86-158-86-203.range86-158.btcentralplus.com) |
| 23:17.11 | garymc | Ive had some trouble with settings in my dahdi.conf etc today. managed to get all my incoming and outgoing calls working. Ive got one problem left to fix. If I assign a DID to a certain extension it doesnt display it anymore when i call out on that extension or group etc. anyone know wht i need to do to diagnose this? |
| 23:19.16 | bmoraca_work | garymc: without a better description of what's failing and the failed call logs, there's no way anyone can help you |
| 23:19.54 | garymc | hey bmoraca_work ; thanks for the help earlier you fixed my problem. What could i show you for a better description? |
| 23:20.23 | garymc | hmmm it seems to be working now |
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