| 00:01.41 | [TK]D-Fender | ~ask |
| 00:01.42 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:02.39 | iPod-nano | Alright, I can access my Asterisk box via the local network but not from the internet. I have forwarded the proper ports on my router already. What could still be causing this? |
| 00:03.49 | iPod-nano | I'm running it on Debian Linux. |
| 00:05.22 | [TK]D-Fender | iPod-nano, you say "proper" yet I have no intrinsic trust on your assumptions of "proper" nor any backup of your firewall setup on either the router or your box. Then there is the though that whatever exact "sercice" you are connecting to is even LISTENING for connections let alone those outside its local subnet |
| 00:05.46 | [TK]D-Fender | iPod-nano, So basically... you've told us nothing of value and shown even less :) |
| 00:06.50 | iPod-nano | As far as I know there is no firewall running on the box itself, but I need someone who knows more than I do to help me make sure. I have ports 5060 and 5061 forwarded to the machine. |
| 00:07.05 | iPod-nano | Asterisk is running. |
| 00:07.27 | hypa7ia | iPod-nano: you aslso need to forwars all the rane of high ports that SIP uses |
| 00:07.40 | iPod-nano | What would those be? |
| 00:08.11 | iPod-nano | This is why I'm in here. I need information like that. |
| 00:08.17 | hypa7ia | look it up, i don't know off the top of my head :) |
| 00:10.03 | iPod-nano | Actually, wait a minute. I have the computer DMZ'd. |
| 00:10.09 | hypa7ia | hrm |
| 00:10.14 | iPod-nano | Which would mean every port. |
| 00:10.17 | [TK]D-Fender | iPod-nano, Oh wow, you've actually mentioned PORTS! |
| 00:10.29 | [TK]D-Fender | iPod-nano, Go read : |
| 00:10.31 | [TK]D-Fender | ~sipnat |
| 00:10.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:11.35 | iPod-nano | Given that this is Asterisk, I'm surprised I can't actually use the * wildcard in the configuration files. |
| 00:11.53 | hypa7ia | haha |
| 00:12.08 | [TK]D-Fender | iPod-nano, Sure you can... be SPECIFIC as to where.. |
| 00:12.29 | [TK]D-Fender | iPod-nano, "exten => *,1,NoOp(Yippi-kai-yay I CAN use an * in my config files!) |
| 00:18.16 | iPod-nano | Alright, how do I tell it that the external IP could be anything? |
| 00:18.28 | [TK]D-Fender | iPod-nano, you CAN'T |
| 00:18.35 | [TK]D-Fender | iPod-nano, you have to tell it what it IS. |
| 00:18.58 | iPod-nano | Wait, does that mean MY IP address? |
| 00:19.05 | hypa7ia | hehehe |
| 00:19.05 | [TK]D-Fender | iPod-nano, if you're on a dynamic IP then you'd better get a DynDNS service or invent a script to check for it and update your settings or you're DOA |
| 00:19.34 | iPod-nano | Are we talking about the IP my provider gives me? |
| 00:19.40 | hypa7ia | yes |
| 00:19.46 | iPod-nano | Oh, my mistake. |
| 00:19.54 | iPod-nano | Thought we were talking about the client's IP. |
| 00:19.56 | [TK]D-Fender | iPod-nano, I was... I can't be too sure about what YOU'RE talking about naturally.... |
| 00:20.08 | [TK]D-Fender | iPod-nano, You're all over the place :) |
| 00:20.13 | [TK]D-Fender | iPod-nano, Scatter-brain! |
| 00:22.15 | iPod-nano | Could you explain the fromdomain parameter? |
| 00:22.35 | hypa7ia | iPod-nano: it's if you need to modify the sip headers |
| 00:22.42 | hypa7ia | you probably don't need to set it |
| 00:22.45 | [TK]D-Fender | iPod-nano, before you go flying WAY off topic, can you explain why you're flying off on this new tangent? :) |
| 00:22.56 | iPod-nano | Hah |
| 00:23.14 | iPod-nano | When it stops giving me warning messages about something, I move on to something else. |
| 00:23.30 | [TK]D-Fender | iPod-nano, Stop looking for trouble, you might succeed... |
| 00:23.49 | iPod-nano | Now I need my friend to get back to his computer so he can test this stupid thing. |
| 00:24.45 | iPod-nano | Now, does anybody know why when I dial my hardware phone from a softphone it works, but when I dial the softphone from the hardware phone it doesn't? |
| 00:25.16 | fujin | are they being sent into the correct context, both devices? |
| 00:25.31 | iPod-nano | Yes, and both are being sent to the same context. |
| 00:25.53 | [TK]D-Fender | iPod-nano, Since you haven't describe exactly what hardware is being used, nor shown you * configurate, nor any CLI oupt, or described the network path, or ANYTHING...... |
| 00:25.58 | [TK]D-Fender | iPod-nano, **NO** |
| 00:26.18 | [TK]D-Fender | iPod-nano, We are only psychic on TUESDAYS |
| 00:26.26 | iPod-nano | Both are able to dial monkeys. |
| 00:28.06 | hypa7ia | have you looked at the console while you're placing calls? |
| 00:28.15 | iPod-nano | Yeah, nothing happens. |
| 00:28.35 | [TK]D-Fender | iPod-nano, have you considered enabling SIP DEBUG? |
| 00:29.02 | iPod-nano | What file do I find that in? |
| 00:30.11 | [TK]D-Fender | in * CLI |
| 00:30.23 | [TK]D-Fender | iPod-nano, "*CLI> sip debug on" |
| 00:31.58 | [TK]D-Fender | iPod-nano, then again give that you've just stated enough that all faith in your NAT settings has reached NULL I'm not sure we're even at this stage yet. |
| 00:32.04 | [TK]D-Fender | iPod-nano, PASTEBIN is your friend |
| 00:32.05 | [TK]D-Fender | ~pb |
| 00:32.06 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:32.52 | fujin | sounds like you haven't configured the device correctly if nothign happens in the console when you dial |
| 00:32.52 | fujin | I trust you've 'set verbose >3' |
| 00:34.01 | iPod-nano | Interesting. |
| 00:34.05 | iPod-nano | Absolutely nothing happens. |
| 00:34.18 | iPod-nano | I dial the extension for my softphone, and nothing. |
| 00:36.05 | iPod-nano | I see lots of output for everything else. |
| 00:36.29 | [TK]D-Fender | iPod-nano, good odds your HARDPHONE'S dialplan is ignoring what you are dialing and not passing it to * |
| 00:36.53 | iPod-nano | But why on Earth would it ignore the one extension yet work for everything else? |
| 00:37.15 | iPod-nano | I created another extension to play the tt-monkeys sound file. |
| 00:37.18 | iPod-nano | That works. |
| 00:37.25 | iPod-nano | From either phone. |
| 00:37.44 | rob0 | They have been carried away by monkeys. |
| 00:38.01 | rob0 | Oh, that's tt-monkeys-intro IIRC |
| 00:42.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:42.57 | iPod-nano | Oh dear. |
| 00:43.05 | iPod-nano | I think I may have just figured out the problem. |
| 00:43.20 | fujin | OH REALLY? |
| 00:43.28 | fujin | And what is that? |
| 00:43.48 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 00:43.58 | iPod-nano | mm hmm. |
| 00:44.14 | iPod-nano | My hardware adapter actually WAS set to treat that one extension differently. |
| 00:44.45 | fujin | awesomness |
| 00:44.55 | iPod-nano | And now it works. |
| 00:45.10 | iPod-nano | They can now call each other. |
| 00:45.29 | fujin | that's incredible, I commend your ability to do the simplest task. |
| 00:45.43 | rob0 | But, do they have anything to say to one another? |
| 00:47.11 | iPod-nano | OK, that leaves me with just one last hurdle to get over. |
| 00:47.36 | [TK]D-Fender | load chan_conversation.so |
| 00:47.50 | wishes | oh man, the more i play with it the more i like it, i just cant leave it alone :O |
| 00:47.56 | wishes | good thing im getting paid for it :D |
| 00:52.41 | CunningPike | wishes: Congratulations - you just made bash.org ;) |
| 00:56.12 | fujin | What are you playing with? |
| 00:56.15 | [TK]D-Fender | TMI!!!!!!!!!!!!!! |
| 00:56.43 | russellb | CunningPike: lol, nice |
| 00:59.05 | *** join/#asterisk thekidrio (n=dbzkai@24-205-77-42.dhcp.psdn.ca.charter.com) |
| 00:59.17 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 00:59.29 | rob0 | :) |
| 01:00.08 | thekidrio | Hello, anyone here know much about aastra phones? I have a phone that says Entered Web Recovery mode. I have navigated to that ip, I am at the recovery page. I just can't seem to get it to find the file on the tftp server I have set up |
| 01:01.00 | thekidrio | The tftp server works fine to config the other phones, oh its a 9112i btw. |
| 01:03.00 | hypa7ia | russellb: i've got a quick sip-tcp question for ya if you don't mind |
| 01:03.17 | WilliamK | dunno, my 9133i is sitting in the box still |
| 01:03.18 | WilliamK | :) |
| 01:03.27 | russellb | hypa7ia: sure |
| 01:04.51 | russellb | doesn't mean i'll have the answer, but feel free to ask |
| 01:06.01 | hypa7ia | :) |
| 01:06.16 | hypa7ia | is there a way to do a DIAL with transport=tcp set? |
| 01:06.23 | [TK]D-Fender | hypa7ia, pro-actively, * doesn't support SIP over TCP. |
| 01:06.30 | JT | asterisk does tcp sip? |
| 01:06.33 | russellb | [TK]D-Fender: she is running a developer branch that has it |
| 01:06.55 | [TK]D-Fender | hypa7ia, Ok, fine, sure! |
| 01:06.55 | russellb | hypa7ia: yeah, if you define a peer in sip.conf that has the line transport=tcp, then you can Dial(SIP/thatpeer) and it should work |
| 01:07.04 | russellb | JT: working on it ... |
| 01:07.11 | fujin | do you mean control the transport via dialplan, hypa7ia? |
| 01:07.25 | hypa7ia | [TK]D-Fender: but thanks for trying :) |
| 01:07.43 | hypa7ia | russellb: yeah, that's what i did, and it still insists on using udp |
| 01:08.00 | russellb | hypa7ia: oh :( |
| 01:08.15 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:08.19 | russellb | sounds like a bug, then |
| 01:08.34 | hypa7ia | yeah i think so |
| 01:08.43 | russellb | sorry :/ |
| 01:08.53 | hypa7ia | crummy |
| 01:09.11 | hypa7ia | so close! |
| 01:09.27 | russellb | perhaps i can look at it this week |
| 01:09.45 | russellb | don't really feel up to it tonight ... |
| 01:09.59 | russellb | i don't think it will be a complicated fix |
| 01:10.00 | hypa7ia | oh no worries |
| 01:10.04 | russellb | because there is code that should handle that |
| 01:10.05 | hypa7ia | yeah, me either |
| 01:10.15 | russellb | silly code |
| 01:10.18 | hypa7ia | indeed |
| 01:10.48 | russellb | that's life on the bleeding edge |
| 01:11.56 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 01:11.59 | drfreeze | hello |
| 01:12.24 | drfreeze | we're setting up a new system and are having problems with ACL |
| 01:12.25 | elixer | shouldn't you be in arkham asylum?? |
| 01:12.37 | elixer | or was that mr. freeze |
| 01:13.03 | drfreeze | 2001/2001 192.168.1.55 D N 5060 OK (72 ms) |
| 01:14.09 | elixer | damn. it was mr. and not dr. |
| 01:14.11 | elixer | ah well. |
| 01:20.33 | iPod-nano | Cool, my Asterisk server works over the internet. |
| 01:20.45 | *** join/#asterisk roberted (n=roberted@c-76-17-219-22.hsd1.mn.comcast.net) |
| 01:21.26 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 01:26.39 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:27.13 | tengulre | hi,all |
| 01:29.58 | wishes | CunningPike: eh lol, im a chick! its not like that! |
| 01:30.07 | wishes | ack, filthy males :( |
| 01:30.13 | elixer | this just keeps getting hotter... |
| 01:30.14 | elixer | heh |
| 01:30.27 | wishes | it would be hotter if i was lesbian :) |
| 01:30.38 | wishes | well .. to watch anyway |
| 01:31.08 | wishes | lesbian computer geek chicks hunched over a keyboard getting excited by asterisk configuration files franticly bashing in numbers on her softphone |
| 01:34.44 | elixer | you had me at "lesbian" ;-) |
| 01:35.21 | drfreeze | Anyone have problems with ACL and AsteriskNow |
| 01:36.02 | [TK]D-Fender | drfreeze, Sopt shooting blind and show some debug for your problem./ |
| 01:36.07 | [TK]D-Fender | stop* |
| 01:36.25 | drfreeze | :) |
| 01:37.20 | drfreeze | [TK]D-Fender: will some debug output do? |
| 01:37.41 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:37.53 | [TK]D-Fender | drfreeze, Would help to SEE a problem rather than guess based on nothign. |
| 01:38.44 | [TK]D-Fender | drfreeze, PASTEBIN is your friend. |
| 01:38.44 | [TK]D-Fender | ~pb |
| 01:38.45 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:39.26 | drfreeze | [TK]D-Fender: http://pastie.textmate.org/97800 |
| 01:40.21 | [TK]D-Fender | <PROTECTED> |
| 01:40.32 | [TK]D-Fender | drfreeze, Clearly your hostname is junk |
| 01:41.00 | [TK]D-Fender | drfreeze, Guess you'll have to show us what your register. mask only the password. |
| 01:41.50 | drfreeze | [TK]D-Fender: the sip file? |
| 01:42.07 | [TK]D-Fender | drfreeze, Clearly. |
| 01:43.19 | *** join/#asterisk Echinos (n=brian@142.46.83.12) |
| 01:43.25 | drfreeze | [TK]D-Fender: http://pastie.textmate.org/97802 |
| 01:44.00 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-814c601ab2bb7d47) |
| 01:44.34 | [TK]D-Fender | drfreeze, you problem is with REGISTERING. WEe need to see the REGISTER that is targeting your system or originating from it. |
| 01:45.44 | drfreeze | [TK]D-Fender: we are not registering. Just providing an ip address to the voip provider |
| 01:46.23 | [TK]D-Fender | drfreeze, those errors you showed are register attempts. |
| 01:46.28 | drfreeze | [TK]D-Fender: the router is mapping the internal ip to a public address |
| 01:46.37 | drfreeze | tengulre: hrmmm |
| 01:46.57 | [TK]D-Fender | registersip = yes <------------ |
| 01:47.16 | [TK]D-Fender | drfreeze, So you're saying this is for an ITSP? |
| 01:47.41 | [TK]D-Fender | host = dynamic <---- what its trying to register against |
| 01:47.46 | [TK]D-Fender | ==== BAD |
| 01:48.17 | drfreeze | [TK]D-Fender: what is ITSP? |
| 01:48.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 01:49.02 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 01:51.58 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:53.06 | [TK]D-Fender | ~itsp |
| 01:53.06 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 01:53.55 | [TK]D-Fender | drfreeze> [TK]D-Fender: we are not registering. Just providing an ip address to the voip provider <------ |
| 01:54.09 | [TK]D-Fender | drfreeze, And yes you have very clearly told it to register. |
| 01:54.58 | drfreeze | [TK]D-Fender: so, comment out the line with registersip=yes ? |
| 02:00.39 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 02:06.16 | [TK]D-Fender | drfreeze, What is that SIP entry supposed to be for? |
| 02:06.35 | drfreeze | I don't know. Was put there by AsteriskNow |
| 02:07.43 | [TK]D-Fender | drfreeze, Asterisk now INVENTED [2001] without you ASKING it to? |
| 02:08.43 | drfreeze | [TK]D-Fender: the new web interface called AsteriskNow for 1.4.9 |
| 02:09.13 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 02:10.03 | [TK]D-Fender | drfreeze, You seem to have some serious communications issues.... I asked if you believe the AsteriskNOW **INVENTED** [2001] wiout you asking for it... |
| 02:10.58 | Echinos | ~asterisk |
| 02:10.59 | jbot | rumour has it, asterisk is the best free PBX in the world |
| 02:11.12 | Echinos | what a disappointment! :/ |
| 02:11.26 | Echinos | I was expecting something more elaborate |
| 02:11.31 | drfreeze | [TK]D-Fender: I cannot parse your sentence |
| 02:11.40 | drfreeze | [Sep 16 21:10:11] NOTICE[6018]: manager.c:964 authenticate: 127.0.0.1 failed to authenticate as 'admin' |
| 02:11.52 | JT | Echinos: like? |
| 02:12.01 | [TK]D-Fender | drfreeze, this is NOT the AsteriskNOW support channel, please read the channel topic. |
| 02:12.05 | drfreeze | after removing host=dynamic and registersip=yes |
| 02:12.31 | Echinos | JT: I dunno, just something bigger |
| 02:12.41 | drfreeze | [TK]D-Fender: this is the interface written by digium |
| 02:12.49 | Echinos | JT: You want me to think something up? |
| 02:12.55 | JT | Echinos: ah ok |
| 02:13.44 | [TK]D-Fender | drfreeze, I'm fully aware, and the channel topic stateing that this isn't its support channel was set by Digium staff. |
| 02:13.48 | Cyon | Echinos: Such a very insightful and beneficial list of additional features that you think would be appropriate as as to allow asterisk to be of better service to those with your vision. Constructive critisism such as yours is always wonderful to receive. |
| 02:13.52 | Echinos | Although... being that this is the * channel, I would imagine asking what it is in here is kinda redundant |
| 02:14.17 | drfreeze | [TK]D-Fender: sure |
| 02:14.45 | Echinos | Wow. I can cut that sarcasm with a dull spoon! |
| 02:15.23 | Cyon | Crap, I didn't do a good enough job, I wanted to it to hit you on the head like a falling tree. |
| 02:15.26 | [TK]D-Fender | Echinos, Made with authentic Jell-o |
| 02:15.32 | Echinos | hehe |
| 02:16.31 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:16.39 | Echinos | I just had high hopes for the bot's desc of *. No biggie. |
| 02:17.39 | CunningPike | Echinos: Brevity is the soul of wit |
| 02:18.10 | Echinos | How... concise. :) |
| 02:18.15 | [TK]D-Fender | Echinos, And sarcasm it's poor cousin. You're not ready to meet the extended family :P |
| 02:19.04 | CunningPike | [TK]D-Fender: It's it's if you mean it is, otherwise it's its |
| 02:19.13 | CunningPike | ;) |
| 02:19.44 | [TK]D-Fender | CunningPike, :p |
| 02:19.44 | Echinos | I'm sure you just gave someone a headache |
| 02:19.47 | [TK]D-Fender | CunningPike, indeed... |
| 02:20.11 | [TK]D-Fender | Echinos, No, CunningPike is grammatically correct and I was distracted enough to deserve that :) |
| 02:20.39 | Echinos | Oh, I know he's correct, but I'm imagining a lurker staring closely at their screen |
| 02:21.36 | [TK]D-Fender | Echinos, "creepy" is an assumed trait around here |
| 02:21.42 | Echinos | Heh |
| 02:22.15 | elixer | err |
| 02:22.59 | *** join/#asterisk Corydon76-dig (i=yellow@pdpc/supporter/sustaining/Corydon76-home) |
| 02:22.59 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 02:23.06 | [TK]D-Fender | Echinos, "creepy" is an assumed trait around here |
| 02:23.37 | elixer | ohhh... "assumed" |
| 02:24.32 | *** join/#asterisk Corydon76-home (i=peuce@pdpc/supporter/sustaining/Corydon76-home) |
| 02:24.32 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 02:25.16 | [TK]D-Fender | elixer, you know what happens when you make an assumption.... |
| 02:26.15 | elixer | [TK]D-Fender: you make an ass out of u and me? |
| 02:26.41 | [TK]D-Fender | elixer, No, you make an ass out of "u" and "umption"! |
| 02:27.05 | elixer | "mption" |
| 02:30.13 | elixer | later folks |
| 02:31.21 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:32.03 | *** join/#asterisk jmacz (n=jmacz@190.24.98.133) |
| 02:35.12 | [TK]D-Fender | elixer, http://koti.mbnet.fi/badbee/wavs/assume.wav |
| 02:37.00 | *** join/#asterisk sniper[FOO] (i=Snip3r@217.27.214.111) |
| 02:40.07 | TJNII | Is there a pattern match rule that will match a single digit or nothing? Like for 9-1-(Area code)-(Number) or 9-(Area code)-(Number) |
| 02:41.11 | [TK]D-Fender | TJNII, those samples are not single digits... |
| 02:41.23 | [TK]D-Fender | TJNII, try rephrasing or providing samples that match :) |
| 02:41.31 | TJNII | Ignore the dashes |
| 02:41.49 | [TK]D-Fender | TJNII, those are all MULTI DIGIT patterns. |
| 02:42.04 | [TK]D-Fender | TJNII, so whats thei "single digit" business you're talking about then? |
| 02:42.59 | [TK]D-Fender | TJNII, an no, you can't math 2 fixed lengths like that. |
| 02:43.08 | [TK]D-Fender | TJNII, in a single pattern. |
| 02:43.15 | TJNII | Okay. Note the lack of a one in the second example. I am wondering if there is a way to do that in a single rule, combining 91XXXXXXXXXX and 9XXXXXXXXXX |
| 02:43.26 | *** join/#asterisk IgorG (n=FeedomPa@host-87-103-181-21.pppoe.omsknet.ru) |
| 02:43.27 | [TK]D-Fender | TJNII, noep |
| 02:43.38 | TJNII | mmkay. That's what I thought. |
| 02:43.50 | TJNII | Just thought I'd ask before I wrote a bunch of new extens. |
| 02:44.25 | [TK]D-Fender | TJNII, Oh, and having "9" as a dial-out prefix is SOOO 80's..... ditch that archaic concept :) |
| 02:45.14 | [TK]D-Fender | TJNII, My city adopted mandatory 10-digit local dialing... I then proceeded to have my * system add the assumed area code to 7-digit dialing while I was at it to thumb my nose at them :) |
| 02:45.18 | TJNII | Pfft. What's the point of having a home PBX if you can just get an outside line? |
| 02:45.38 | [TK]D-Fender | TJNII, the ide of needing a stupid 9 in front is ridiculous :) |
| 02:45.43 | TJNII | [TK]D-Fender: Yea, I'm doing the same thing for the local area code. |
| 02:45.55 | TJNII | [TK]D-Fender: I think it fits. |
| 02:46.23 | [TK]D-Fender | prefix adds nothing to your life and extra time lost dialing a crazy digit. |
| 02:47.01 | TJNII | Though I will admit, when I set it up I had just assumed I needed a dial out digit. Now that you've got me thinking about it there is no real reason for it. |
| 02:47.07 | [TK]D-Fender | TJNII, other downside : if you miss a call on a phone with CID and try to REDIAL the number as it appears, it won't have the 9 unless you mangle it on receipt so therefor you can't conveniently dial out from a missed calls list. |
| 02:47.51 | TJNII | True. |
| 02:48.23 | [TK]D-Fender | TJNII, My work setup even checks if you 10-digit an area code that isn't legitimately local and adds the 1 for you so you can "call-back" pretty much anyone without having to think about it, |
| 02:49.46 | TJNII | Well, what kind of sucks is that my VOIP provider didn't have any phone numbers in my area code, so I'm in 515 and my number is 712. So all my dialed numbers have to be changed into 10 digits. |
| 02:52.18 | TJNII | Though one thing the nine does give me is if you dial 9 it goes through BroadVoice, if you dial 8 it goes through Ekiga, and if you dial 2 it goes through a friends * box, all which can call out. |
| 02:52.50 | [TK]D-Fender | TJNII, when would you choose to use a secondary option? |
| 02:53.14 | TJNII | I don't know. The point is I can. :) |
| 02:54.55 | [TK]D-Fender | TJNII, I can do the same through my numerous clients. for THOSE I add a prefix though. |
| 02:56.55 | [TK]D-Fender | TJNII, a *XX pattern so I can dial anything I need on the other side |
| 03:00.00 | *** join/#asterisk Corydon76-dig (i=two@pdpc/supporter/sustaining/Corydon76-home) |
| 03:00.09 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 03:11.09 | magic_hat | anyone know if there's a way to stop asterisk from the shell (not the CLI)? I'm trying to configure monit. |
| 03:13.33 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 03:14.20 | [TK]D-Fender | magic_hat, 'asterisk -rx "stop now"' |
| 03:15.02 | magic_hat | is there a question you don't have the answer for? :) |
| 03:16.45 | [TK]D-Fender | 42 <---------- |
| 03:16.48 | [TK]D-Fender | nope! |
| 03:17.00 | magic_hat | lol, brilliant! |
| 03:18.00 | [TK]D-Fender | <- smarter than the average bear |
| 03:19.07 | magic_hat | yes, I've asked nearby bears a bunch of questions about *, but I've never received a satisfactory answer. |
| 03:21.10 | [TK]D-Fender | magic_hat, I'm a foreign bear :) |
| 03:21.40 | magic_hat | ah ha. that explains it |
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| 03:22.58 | dan__t | HI. |
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| 04:07.45 | magic_hat | anyone know why an /etc/event.d/asterisk file wouldn't start * on boot in ubuntu? |
| 04:08.33 | fujin | because you need /etc/init.d/asterisk |
| 04:08.40 | fujin | and then you need to add the relevant symlinks with update-rc.d |
| 04:09.07 | magic_hat | eek. what goes in /etc/init.d/asterisk? |
| 04:09.39 | fujin | uh |
| 04:09.50 | fujin | a process control script which responds to start|stop|restart |
| 04:10.07 | magic_hat | okay, so that file exists.... it's the default config that came with the distro. |
| 04:10.15 | fujin | yep |
| 04:10.58 | magic_hat | so what do I need to change in there? |
| 04:11.07 | fujin | whaddya mean? |
| 04:11.13 | fujin | does /etc/init.d/asterisk start not work? |
| 04:11.38 | magic_hat | nah, I get 'asterisk not yet configured' |
| 04:11.47 | fujin | vi /etc/default/asterisk |
| 04:11.54 | fujin | read the file |
| 04:11.57 | fujin | there is an option you have to change |
| 04:13.21 | magic_hat | ahhh. |
| 04:13.52 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 04:14.11 | magic_hat | and I want runastsafe=yes, right? |
| 04:14.55 | fujin | yup |
| 04:15.23 | nDuff | What could result in a "unable to request channel" error when doing a "originate Zap/26/1234567", given that the given zap channel exists and is part of a span that's green? |
| 04:15.39 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 04:15.55 | magic_hat | bad fd number? |
| 04:16.54 | nDuff | magic_hat: was that directed at me? |
| 04:17.14 | magic_hat | lol sorry... at fujin. or anyone else who knows what to do about that |
| 04:18.33 | nDuff | magic_hat: that's pretty general -- what exactly has a bad fd? |
| 04:19.10 | nDuff | magic_hat: (fd == file descriptor; every single place in asterisk where it has a file or device open has an fd, so there are lots of places that could happen). |
| 04:19.43 | magic_hat | okay, I think I got it... the distro I got has a 1-line bug. |
| 04:21.29 | magic_hat | is there any sense to running monit on asterisk when it's already set up to run at boot with /etc/event.d/asterisk? |
| 04:22.40 | nDuff | I'm not familiar with monit. |
| 04:22.54 | nDuff | personally, I don't use my distribution's init scripts for asterisk at all; I use runit. |
| 04:23.00 | styelz | change #!/bin/sh to bash. |
| 04:23.22 | styelz | solved my fd prob |
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| 04:24.50 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 04:25.33 | styelz | ...in ubuntu |
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| 04:34.22 | fujin | magic_hat, you need to alias /bin/sh to /bin/bash instead of /bin/ash |
| 04:34.30 | fujin | stupid Ubuntu nublets changed the link. |
| 04:34.48 | fujin | lrwxrwxrwx 1 root root 9 2007-08-09 16:13 /bin/sh -> /bin/bash |
| 04:35.14 | fujin | ln -sfv /bin/bash /bin/sh |
| 04:35.23 | styelz | heh |
| 04:36.12 | styelz | ergh |
| 04:45.51 | nDuff | hrm. the PRI bridging my two asterisk servers is doing nothing but "Sending Set Asynchronous Balanced Mode Extended" on both ends. |
| 04:48.00 | JT | nDuff: ok, i've missed your previous stuff, what have you done so far to make it work? |
| 04:50.56 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 04:51.24 | l0verb0y | hey |
| 04:52.19 | *** join/#asterisk bigmatt (n=bigmatt@cpe-75-83-221-164.socal.res.rr.com) |
| 04:53.16 | nDuff | JT: lessee... set one end to pri_net and the other to pri_cpe with matching bchan/dchan configuration on both ends... |
| 04:53.36 | bigmatt | excuse my noobness but why would anybody want to run asterisk in their home? |
| 04:54.01 | fujin | so that they don't have to have a phoneline? |
| 04:54.04 | fujin | they like SIP phones? |
| 04:54.14 | rob0 | Because? |
| 04:54.21 | fujin | BWC :) |
| 04:54.27 | JT | nDuff: with what cable? |
| 04:54.33 | nDuff | bigmatt: they want to do their own phone screening logic? they want to route calls between roommates based on a menu? |
| 04:54.44 | nDuff | JT: a T1 crossover cable. it shows up as green on both ends in the wanrouter logs. |
| 04:54.45 | bigmatt | hehe ya no offense i'm just wondering :) |
| 04:54.56 | bigmatt | i'm new to the whole voip thing |
| 04:55.01 | JT | nDuff: pastebin.ca zaptel.conf and zapata.conf |
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| 04:55.12 | rob0 | One GOOD reason is to learn how to work with it, and be able to put it on a resume. |
| 04:56.06 | bigmatt | how would somebody retain their phone number if they don't have a phone line :/ |
| 04:56.14 | bigmatt | is the only thing troubling my mind |
| 04:56.15 | fujin | number portability |
| 04:56.18 | nDuff | JT: this is Sangoma hardware on both ends, so not all the configuration is in zaptel.conf; if you need wanrouter* as well, feel free to ask. |
| 04:56.43 | fujin | bigmatt: or they could use a SIP provider which terminates phone numbers over SIP |
| 04:56.52 | JT | nDuff: sure why not |
| 04:57.33 | bigmatt | I see thx |
| 04:58.01 | bigmatt | so using it as a home user i would cancel my phone line keep my net and get a sip provider? |
| 04:58.10 | fujin | if you wanted to, yeah |
| 04:58.37 | JT | but i'd recommend against cancelling your POTS line |
| 04:58.43 | nDuff | JT: http://pastebin.ca/700337, excluding the wanrouter stuff; that to follow. |
| 04:58.43 | JT | you should have at least one |
| 04:58.49 | fujin | mm, you can use your POTS line in asterisk |
| 04:59.00 | fujin | with a digium fxo/fxs card |
| 04:59.00 | fujin | etc |
| 04:59.03 | nDuff | bigmatt: if you want to do faxing, you pretty much need a POTS line. |
| 04:59.12 | bigmatt | I r noob, pots line? |
| 04:59.13 | bigmatt | :) |
| 04:59.22 | nDuff | bigmatt: Plain Old Telephone System |
| 04:59.22 | bigmatt | well ya i hear faxing = no no |
| 04:59.24 | bigmatt | ooooh hehe |
| 04:59.52 | fujin | PSTN is what it is otherwise known as |
| 04:59.54 | l0verb0y | is there anyway to control how much time goes by before an IVR picks up a call? |
| 05:00.12 | fujin | yes |
| 05:00.15 | fujin | Ringing(seconds); |
| 05:00.18 | fujin | Answer(); |
| 05:00.24 | l0verb0y | thanks |
| 05:00.30 | fujin | show application Ringing |
| 05:01.00 | l0verb0y | thanks thanks |
| 05:01.26 | nDuff | JT: http://pastebin.ca/700337 now includes the wanrouter files. |
| 05:01.37 | JT | nDuff: don't know if it matters with sangoma, but: span=2,2,1,esf,b8zs should be 2,0,0,esf,b8zs |
| 05:01.59 | JT | nDuff: you cannot update pastebins |
| 05:02.18 | nDuff | ooh, right, it's now http://pastebin.ca/700339 |
| 05:02.32 | JT | nDuff: also, all spans should have 0 as their third number |
| 05:02.54 | nDuff | ...hrm. on server 2, I want to pull timing from span 1 (which is my connection to the telco) and feed it to span 2. |
| 05:04.05 | JT | right, so 1 will take timing with the span as primary clok source |
| 05:05.42 | nDuff | okay, right, the LBO should indeed be zero -- the systems are quite close -- though I believe that one's ignored in zaptel by wanrouter... |
| 05:06.02 | nDuff | oh, right, timing should be 0. |
| 05:06.14 | nDuff | okay, seeing if that helps... |
| 05:07.04 | JT | on the server with pri_net, it should be 0 |
| 05:08.05 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 05:08.22 | nDuff | yes. |
| 05:08.55 | JT | so it looks like it should all work |
| 05:08.59 | JT | what's actually wrong? |
| 05:10.52 | nDuff | JT: it's working now |
| 05:10.59 | nDuff | JT: took a minute to restart everything and test it. |
| 05:11.01 | nDuff | JT: many thanks! |
| 05:11.41 | JT | sweet |
| 05:11.43 | JT | no probs |
| 05:13.59 | bigmatt | anybody remember the name of those px100 cards? lol |
| 05:14.02 | bigmatt | i'm off by a little |
| 05:14.11 | nDuff | ...well, it worked for a minute. |
| 05:14.50 | nDuff | from server1, when trying to initiate a call to server2: "pri_find_dchan: No D-channels available! Using Primary channel 48 as D-channel anyway!" |
| 05:15.12 | nDuff | ...and server2 is constantly spamming "== Primary D-Channel on span 1 up" |
| 05:15.35 | nDuff | (which seems like pretty good google fodder, at least) |
| 05:16.03 | JT | make sure your cable doesn't suck |
| 05:16.11 | JT | make sure zttest results are okay |
| 05:17.03 | nDuff | zttest is continuous 100.000 (dual-core system, pci-x card) |
| 05:17.19 | JT | i think you mean pci-e |
| 05:17.23 | nDuff | er, yes. |
| 05:17.28 | JT | but 100 is pretty good i guess |
| 05:20.52 | nDuff | ...hrm. doing "pri debug" on both ends, looks like server1 is continuously sending SABME, and server2 is continuously receiving it and sending an acknowledgement... which server1 never gets. |
| 05:21.26 | JT | did you make the cable yourself? |
| 05:22.43 | nDuff | JT: my facilities guy did; I suck at cables. He typically doesn't. (Moreover, I'm used to Sangoma's toolage telling me when a cable is bad... though I suppose there could be some cases it doesn't handle). |
| 05:23.39 | JT | it sounds like the signals aren't getting through in one direction on a full time basis |
| 05:23.43 | nDuff | *nod*. |
| 05:24.15 | nDuff | it's odd that the one initial test call I did appeared to get through... *shrug*. I'll look and see if the wanrouter toolkit has something to try to diagnose this. |
| 05:26.51 | *** join/#asterisk GandalfX (n=123@190.65.109.26) |
| 05:27.07 | GandalfX | hola |
| 05:27.13 | GandalfX | gurus del asterisk |
| 05:27.14 | GandalfX | jejeje |
| 05:27.35 | GandalfX | buenas noches |
| 05:29.09 | nDuff | hola. ?habla ingles? no hablamos espanol |
| 05:30.57 | fujin | aren't PRI's just 2 pairs? |
| 05:31.13 | JT | right, PRIs are 2 pairs |
| 05:32.46 | GandalfX | yo no hablo ingles |
| 05:32.47 | GandalfX | jajajaa |
| 05:32.54 | GandalfX | pero hare lo posible |
| 05:34.15 | Corydon76-dig | no hablo espanol |
| 05:34.59 | Snake-eyes | Does the latest zaptel init script load fxotune -s ? I see no mention of fxotune in the init script |
| 05:35.29 | Corydon76-dig | Snake-eyes: nothing is preventing you from putting it in there |
| 05:35.41 | Snake-eyes | ok |
| 05:35.52 | Corydon76-dig | but the reason it's not in there by default is that it's only useful to those running analog cards |
| 05:38.26 | Snake-eyes | ah, i was wondering cause i saw this line in "Latest Zaptel init script should do that if /etc/fxotune.conf is present. " at voip-info |
| 05:47.06 | *** join/#asterisk [s]Animat (n=info@d220-238-210-46.dsl.vic.optusnet.com.au) |
| 05:47.57 | [s]Animat | Anybody know what to do when asterisk says "Unable to create channel of type 'SIP'" ? |
| 05:55.28 | *** join/#asterisk sAdEyES (n=Love@58.186.215.30) |
| 05:55.57 | sAdEyES | help me |
| 05:56.24 | [TK]D-Fender | sAdEyES, #drphil |
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| 06:12.44 | Snake-eyes | I noticed that if i setup asterisk to dial say zap/1 then zap/2, that if the line is unplugged in zap/1 it still tries it and sits there instead of going to zap/2 (tdm400 analog) |
| 06:13.12 | Snake-eyes | any way to get it to move onto zap/2 if line isn't plugged in |
| 06:18.21 | nDuff | heh. |
| 06:22.40 | nDuff | JT: found the issue. |
| 06:22.46 | nDuff | JT: the dchan wasn't configured in wanrouter. |
| 06:23.11 | JT | i see |
| 06:24.15 | luke-jr | anyone interested in a PAP2/SPA module for *? :p |
| 06:24.52 | Strom_M | Snake-eyes: under what conditions might you conceivably have a PBX configured to use circuits that aren't plugged in? |
| 06:31.23 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 06:35.44 | Snake-eyes | Strom_M, more like some one doesn't plug it incorrectly or line gets pull out by mistake |
| 06:36.34 | Strom_M | well, it should be tested before being put into production |
| 06:37.00 | Strom_M | and you should ensure that only people who are not going to pull the line out by mistake have access to the server |
| 06:37.23 | JT | indeed |
| 06:38.54 | Snake-eyes | .... |
| 06:39.07 | JT | no need to be dotty |
| 06:39.12 | JT | plain and simple logic |
| 06:39.21 | JT | asterisk is not a system to protect against morons |
| 06:39.27 | Snake-eyes | but not very helpful |
| 06:39.37 | JT | keeping the morons away from asterisk is your job |
| 06:39.42 | JT | it's perfectly helpful |
| 06:39.46 | JT | it's called a reality check |
| 06:41.33 | Snake-eyes | I don't want to get into argument over this but I rather have yes this can be done or can't, not "I don't like the problem debate" |
| 06:41.56 | JT | it's not an argument |
| 06:42.04 | JT | it's a statement of the facts |
| 06:42.16 | JT | your pbx's phone lines should not be easy to pull out |
| 06:42.35 | JT | and it's hard enough to detect stages of an analogue phone call properly |
| 06:42.52 | JT | let alone developing weird edge case technology like line failure detection for analogue |
| 06:47.35 | yang | hm how do i exit out of CLI (bye,logout,exit,quit has no effect) |
| 06:47.50 | JT | ctrl + c |
| 06:48.34 | Strom_M | "stop now" |
| 06:48.48 | Strom_M | you probably started asterisk with -c rather than -r |
| 06:49.33 | JT | he might not want to stop asterisk itself |
| 06:49.35 | *** join/#asterisk epaulin (n=epaulin@219.135.248.88) |
| 07:00.51 | Snake-eyes | JT, so you're saying its hard and silly to test for a current? |
| 07:01.42 | JT | it's definitely silly, but it may also be hard, depending on the internal design of the cards |
| 07:01.56 | JT | normal PBXes don't check to see if an analogue line is active |
| 07:02.16 | JT | pretty sure ATAs don't either |
| 07:02.45 | Snake-eyes | but analogue handset does, in a way |
| 07:03.06 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:03.07 | JT | what way, your ear listens for a dialtone? |
| 07:04.18 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:04.30 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:07.24 | Snake-eyes | true, but having something that detects current (like any standard volt meter) would bloat the card ? |
| 07:07.57 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 07:08.13 | JT | it's simply a waste of development effort |
| 07:08.21 | JT | considering all the other areas that could be improved |
| 07:08.26 | Snake-eyes | hmm |
| 07:08.35 | JT | normal PBXes don't do it, there's no need |
| 07:10.33 | Snake-eyes | i get what you saying but i still partly disagree with your 'fact' statement (not everyone is setting up asterisk for fortune 500 companies) |
| 07:10.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:11.24 | JT | the cheapest smalles PBXes are the ones least likely to have analogue line disconnection detection |
| 07:11.40 | JT | high availability is an enterprise/"fortune 500 company" feature |
| 07:11.45 | JT | not a small business feature |
| 07:12.28 | Snake-eyes | so you're saying small business shouldn't run asterisk ? |
| 07:13.06 | JT | no, i'm saying it's a storm in the teacup, i think very few businesses have issues with asterisk not detecting that an analogue line is inopperative |
| 07:15.24 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
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| 07:16.00 | Snake-eyes | hmm |
| 07:16.25 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:18.49 | flenders | I know for a fact that NECs and Nortels PBXes don't detect inoperable lines |
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| 07:37.11 | axscode | [Sep 17 23:34:16] WARNING[7205]: config.c:1235 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 07:37.17 | axscode | can anyone help me please.. |
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| 07:51.44 | nDuff | Snake-eyes: if it's important enough to need disconnection detection, it's important enough to not be using analog lines |
| 07:51.58 | nDuff | Snake-eyes: ...and detecting when your PRI goes dead is much easier. |
| 07:54.04 | nDuff | Snake-eyes: I'm generally in agreement with JT here -- if your wiring is randomly being broken, you have bigger problems. People at your site don't expect servers to keep running if cables are randomly yanked, right? |
| 07:57.04 | axscode | nDuff: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available <-- can you help me with this? |
| 07:57.37 | nDuff | axscode: If I could, I would have volunteered to help the first time you asked. |
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| 07:59.49 | nDuff | axscode: that said, is res_config_mysql loaded? |
| 08:00.39 | nDuff | axscode: it's part of asterisk-addons, not asterisk proper, so you may simply not have it installed. |
| 08:01.14 | nDuff | axscode: ...but then, you should have already set up res_mysql.conf, which doesn't exist as part of the base install either, so you should know by now if it's completely not there. |
| 08:01.19 | axscode | ive installed it |
| 08:01.43 | nDuff | axscode: dunno. All I can offer is common sense, because -- as previously discussed -- I don't use realtime. |
| 08:01.51 | axscode | i did the asterisk-addons; ./configure; make; make install |
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| 08:02.15 | axscode | nDuff: any file to check please? |
| 08:03.00 | nDuff | axscode: RTFM. Where is the engine supposed to be enabled? which module provides it? make sure that the module in question is loaded and that its configuration file is in place. |
| 08:03.37 | JT | jeebus |
| 08:03.43 | axscode | im having all of that. |
| 08:03.52 | JT | axscode: can you read, nDuff DOES NOT USE realtime, stop hassling him |
| 08:04.10 | yang | IAXTEL website down? Where can I register my IAX number? |
| 08:05.53 | InHisName | I have Innomedia ATA for Voiceline n2p. It works fine. Now try to setup * to register, I got as far as "invalid username/PIN" error. I AM using the correct username & PIN. |
| 08:07.28 | axscode | thanks |
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| 08:14.49 | arien | I'm running into trouble moving an Asterisk server behind a NAT. |
| 08:15.37 | arien | I've changed the server's network configuration (IP, gateway, netmask), forwarded SIP, IAX and RTP ports on the NAT to the server. |
| 08:16.14 | JT | and set externip= and localnet= in sip.conf, of course? |
| 08:16.24 | arien | In sip.conf I've added entries for externip and localnet. |
| 08:16.38 | JT | so the problem is? |
| 08:16.38 | arien | JT: right. |
| 08:17.23 | arien | when I call in from the outside the ringing doesn't stop when I pick up the phone. |
| 08:17.38 | JT | using what phone? |
| 08:17.38 | arien | from the inside it doesn't either IIRC |
| 08:17.50 | arien | mixture of phones from Cisco and Linksys |
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| 08:18.45 | yang | IAXTEL website down? Where can I register my free IAX username for testing? |
| 08:18.52 | arien | JT: the changes I've mentioned should be all that is required for this to work? From what I've found online it seems so... |
| 08:20.55 | arien | btw, the configuration as it is now (using a public IP for the server) is working fine |
| 08:22.06 | JT | what exactly did you port forwarD? |
| 08:22.54 | arien | SIP 5060, IAX 4569 (UDP), RTP 8000-30000, HTTP and HTTPS |
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| 08:33.34 | enioh | yo |
| 08:36.55 | JT | arien: you only told me what protocol you forwarded for iax |
| 08:37.53 | arien | JT: I'm not following? |
| 08:38.02 | JT | IAX 4569 (UDP) |
| 08:38.05 | JT | UDP |
| 08:38.06 | JT | ... |
| 08:38.25 | arien | ah, right... the rest is UDP and TCP, apat from HTTP and HTTPS |
| 08:38.41 | JT | don't see how http will help :P |
| 08:38.48 | JT | or tcp sip or rtp |
| 08:39.33 | arien | Right, HTTP(S) is only for the webserver on the same box :) |
| 08:39.48 | JT | weird to put a webserver on the same box, but okay |
| 08:40.23 | arien | about tcp sip and rtp, I saw that mentioned on a page documenting this situation so tried it to see if it would help |
| 08:40.54 | JT | they were obviously nubs |
| 08:40.59 | JT | tcp rtp doesn't existed |
| 08:41.01 | arien | (the webserver is only for easy access to some monitoring tools, they won't let me kill that) |
| 08:41.02 | JT | exist |
| 08:41.05 | JT | and wouldn't work |
| 08:41.15 | JT | and tcp sip is not supported by asterisk |
| 08:41.47 | arien | okay, I'll remove those |
| 08:42.30 | *** join/#asterisk Shadowfire__ (n=jeff@fl-204-215-37-142.sta.embarqhsd.net) |
| 08:44.20 | arien | JT, another thing: the phones have private IPs already. This means I'll only have to set nat=no for all local phones when moving the server behind the NAT, right? |
| 08:44.23 | Shadowfire__ | I gonna ahead and loaded fedora 7 and * 1.2. I have done a lsmod |grep zaptel and get |
| 08:44.47 | Shadowfire__ | zaptel 210928 1 wctdm |
| 08:44.50 | JT | i don't think it matters, but okay |
| 08:45.05 | Shadowfire__ | crc_ccitt 6337 1 zaptel |
| 08:45.38 | Shadowfire__ | but for some reason I do not have active channels |
| 08:45.56 | Shadowfire__ | I did a ztcfg -vvv |
| 08:46.12 | JT | Shadowfire__: you have configured zaptel.conf and zapata.conf correctly? |
| 08:46.17 | Shadowfire__ | and it sez that ch01 - 04 are configured |
| 08:46.27 | Shadowfire__ | I thought I did... |
| 08:46.30 | JT | sez? please, english |
| 08:46.42 | Shadowfire__ | I was able to do ext to ext |
| 08:46.51 | Shadowfire__ | but no out going or in coming... |
| 08:46.58 | arien | JT: Oh, okay... Does that mean that I should go and look at the NAT for a solution to my problems? |
| 08:47.17 | JT | arien: dunno, maybe you should packet dump |
| 08:47.38 | JT | Shadowfire__: sounds like a dialplan issue then |
| 08:47.43 | Shadowfire__ | hmmm |
| 08:47.53 | Shadowfire__ | probably |
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| 08:48.08 | arien | JT: hmm, yeah... but then I wouldn't know what to look for ;) |
| 08:48.26 | JT | arien: packets going to the wrong place. |
| 08:48.33 | Shadowfire__ | what command allows user.conf to be made? or is that made automatically>/ |
| 08:48.36 | Shadowfire__ | ? |
| 08:49.09 | JT | users.conf is a 1.4 oddity |
| 08:49.17 | Shadowfire__ | hmmm |
| 08:49.19 | Shadowfire__ | oops |
| 08:49.25 | Shadowfire__ | I have 1.2 |
| 08:49.27 | JT | you can not use it if you just configure sip.conf or iax.conf directly |
| 08:49.33 | Shadowfire__ | nod |
| 08:49.46 | Shadowfire__ | okay... I did that.. |
| 08:50.08 | JT | you really haven't explained your setup at all |
| 08:50.09 | Shadowfire__ | long story short... I was on *Now 1.4 |
| 08:50.23 | Shadowfire__ | and was looking back at config files to figure somethings out... |
| 08:50.28 | JT | i see |
| 08:50.32 | arien | JT: right |
| 08:50.35 | Shadowfire__ | apparently this was a BAD idea |
| 08:50.52 | Shadowfire__ | so I am on 1.2 |
| 08:50.54 | Shadowfire__ | now |
| 08:52.19 | Shadowfire__ | I think you are right... I am confused at the order of the sip context and how it links up with the dailplan.. |
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| 08:57.58 | axscode | question: mysql_config is generated by asterisk-addon or by mysql itself |
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| 09:22.09 | svensk_neutrino | hello boys |
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| 09:32.29 | mace | i'm having some trouble with garbled audio at the start of calls |
| 09:32.38 | E-bola | guys i need my aasterisk server to play sound over a loudspeaker |
| 09:32.39 | mace | for about 0.5 seconds or so |
| 09:33.42 | yang | How do I enable SIP forwarding from my asterisk to use another SIP ? |
| 09:36.08 | yang | Asterisk as a SIP client outside nat, connecting to outside SIP proxies |
| 09:37.00 | mace | most of the SIP proxies provide recipes for doing that |
| 09:37.09 | mace | eg. voip talk does |
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| 09:38.08 | yang | yes |
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| 09:38.26 | yang | I got all the settings, but do i have to enter a SIP proxy into sip.conf |
| 09:38.30 | yang | or channels.conf?P |
| 09:38.56 | mace | mine are all in sip.conf |
| 09:41.27 | *** join/#asterisk nain (n=nain@203.148.77.53) |
| 09:41.31 | yang | can you paste me your example to #flood please |
| 09:42.10 | yang | becouse my asterisk i thinkm it doesnt use proxy |
| 09:42.46 | yang | I am able to connect to it with SIP client |
| 09:42.59 | yang | but i cannot use proxy and make external calls |
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| 09:44.55 | tzafrir | is there a standard way to advertize asterisk via zeroconf? |
| 09:45.22 | tzafrir | There seems to be some elusive res_zeroconf which is not even availble for download |
| 09:45.36 | tzafrir | But I just want a services entry for avahi-daemon |
| 09:45.57 | tzafrir | Just something to allow "decent client" (which?) to pick it up |
| 09:47.29 | yang | mace: ? |
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| 09:48.34 | nain | Hi Every body |
| 09:51.09 | E-bola | Can anybody recomend a soundcard for use with linux/asterisk? |
| 09:51.21 | E-bola | i got a 1u server, so maybe an usb soundcard would be the easiest |
| 09:51.30 | E-bola | its just for playing audio to an external speaker |
| 09:51.37 | E-bola | so quality doesnt matter much |
| 09:51.39 | nain | is it possible to dial another asterisk box after answering the call, Calls to internal sip extensions work fine but when i tried to send same call to another asterisk box i got congestion and call got failed |
| 09:51.41 | E-bola | it just needs to be properly supported |
| 09:52.36 | yang | i get Got SIP response 481 "Call/Transaction Does Not Exist" |
| 09:52.53 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 09:58.36 | mace | yang: sorry was afk |
| 10:01.09 | nain | Can any one see where am i wrong with config while calling to another asterisk box after answering the call from zap channel! http://pastebin.ca/700527 |
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| 10:04.09 | mace | yang: i have two sections |
| 10:04.29 | mace | yang: a register => [accountnumber]:[password]@voiptalk.org/[accountnumber] |
| 10:04.30 | tzafrir | nain, your problem isn't clear |
| 10:04.55 | tzafrir | can you provide a trace (e.g: from the CLI) of a failed incoming call? |
| 10:04.57 | yang | mace: i have this section entered |
| 10:05.25 | mace | yang: and a [voiptalk] section, type=friend with username/fromuser set to [accountnumber], secret set to [password], nat=yes and insecure=port,invite |
| 10:05.47 | mace | yang: your firewall may need to be configured to route SIP packets to your asterisk box from outside |
| 10:06.05 | mace | depending on whether it understands SIP |
| 10:06.07 | yang | the VOIP talk is the login of your proxy or the login of the user on your asterisk? |
| 10:06.34 | mace | login of proxy |
| 10:06.49 | yang | hm |
| 10:07.28 | *** join/#asterisk webman (n=chatzill@52.87.233.220.exetel.com.au) |
| 10:07.38 | yang | but isnt this defined allready in register==> |
| 10:08.09 | nain | Tzafrir: infacti i am out of box now, unable to provide trace of failed call, However, i want to clear my concept that can we orignate answer call to another asterisk box of sip server or there is a concept of call forwarding |
| 10:08.34 | yang | mace: hm i dont run the firewall |
| 10:08.54 | mace | yang: not quite; the register is simply there to establish a connection to the proxy from behind the firewall; the friend section is so that asterisk knows about the peer |
| 10:09.41 | mace | yang: ok, in which case you want nat=no probably |
| 10:09.48 | mace | yang: how far is it getting? |
| 10:09.49 | nain | Tzafrir: As i have two asterisk box, I want to orignate call from first box, answered in 2nd box, and based on user selection of extension call routed back to first box where it is terminated to fxo line! |
| 10:10.49 | yang | mace: well i am not NATED i have a public ip |
| 10:11.06 | yang | mace: i was also able to dial 500 number |
| 10:11.14 | mace | yang: and you got audio? |
| 10:11.15 | yang | Dial("SIP/snom-0819eb60", "IAX2/guest@misery.digium.com/s@default") in new stack |
| 10:11.20 | yang | yes |
| 10:11.47 | mace | are you seeing any errors on the asterisk console? |
| 10:12.01 | mace | perhaps relating to username/password issues? or connection problems |
| 10:12.49 | yang | i ll paste you |
| 10:13.45 | tzafrir | nain, sure you can do that |
| 10:14.04 | tzafrir | nain, but maybe the word you miss is "sip user" |
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| 10:14.33 | mace | E-bola: is this for a local console? |
| 10:15.05 | mace | E-bola: any of the ac97 ones basically just work |
| 10:15.22 | E-bola | its a hp 1u rack server |
| 10:15.25 | yang | http://paste.debian.net/37333 |
| 10:15.26 | E-bola | there is no onboard sound |
| 10:15.55 | E-bola | its to connect a speaker to the main incoming phone line, so instaid of 1 or many phones ringing you hear a wave file playing |
| 10:16.01 | E-bola | and you can pickup the call at ur phone |
| 10:16.46 | nain | tzafrir: I have tested conectivity between both asterisk defined with sip peer and sip user vise versa and it works fine, without Answer() it's work fine and i am able to dial any internal sip user but not another asterisk box sip user |
| 10:18.58 | creativx | anyone here know of a sip softphone that only lets you use 1 line at the time |
| 10:24.50 | nain | where call will go to another asterisk box with this extension: exten => 1,1,Dial(SIP/foo) ; where foo is peer defined in sip.conf? |
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| 10:28.55 | GingerDog | hi, does anyone have any opinions as to the 'best' linux kernel version to run with asterisk? |
| 10:31.07 | mace | E-bola: i've found the ensoniq ES1371 based cards to work well in the past |
| 10:31.43 | mace | yang: nothing jumps out at me from that paste log unfortunately |
| 10:32.42 | mace | E-bola: basically any of the soundblaster cards |
| 10:35.56 | E-bola | im looking at the xmod things |
| 10:36.02 | E-bola | they reportedly works |
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| 11:02.47 | nain | tzafrir: Hi i have put my configration here! please see what's wrong with config! http://pastebin.ca/700562 |
| 11:03.22 | tzafrir | nain, sorry, busy right now |
| 11:06.55 | nain | tzafrir: no problem :) |
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| 11:13.46 | nain | Any one free to check the config! http://pastebin.ca/700562 for error? |
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| 11:39.24 | yang | mace: did you define the stun server in your asterisk...or how is it able to connect to voipbuster? |
| 11:40.18 | DrukenLPY | stun is for clients behind a nat to figure out what their ip is... |
| 11:40.25 | yang | <PROTECTED> |
| 11:40.47 | yang | i have this entry in sip, isit enought to use voipbuster as SIP proxy? |
| 11:41.51 | yang | do i need to make a separate entry for "friend" with these username and password in sip.conf or why isnt it working? |
| 11:42.41 | ai-a | yang: register (as the documentation states) is to tell the server where and who you are, and thats it... |
| 11:43.18 | ai-a | to make calls you need a sip entry for the server... voipbuster.com should explain what details are required for asterisk... if you read it. |
| 11:46.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:46.23 | DrukenLPY | sounds like someone struggling with the common asterisk learning curve.. er well... i guess the client/server learning curve |
| 11:49.24 | *** join/#asterisk stmaher (n=stmaher@mail.voxpilot.com) |
| 11:49.53 | stmaher | Hello everyone.. Im having a little trouble with terminated by silence.. |
| 11:50.08 | stmaher | I see that asterisk is still developing VAD support.. can anyone please tell me that status of the development? |
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| 11:56.57 | BockBilbo | hello |
| 11:57.24 | BockBilbo | im configuring the voicemail on my asterisk server, and i get this error: [Sep 17 13:51:49] WARNING[32035]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format |
| 11:57.35 | BockBilbo | i guess that it's because it cant find the audio file for saying afternoon |
| 11:57.44 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 11:57.59 | BockBilbo | where are those audio files supposed to be? |
| 11:58.35 | mace | cd /var/lib/asterisk/sounds usually |
| 11:58.41 | mace | -cd |
| 11:58.59 | BockBilbo | right |
| 11:59.23 | McDouglas | i need some help with this dialplan: http://pastebin.com/d58c9f8d4 i want to restirct the usage of lines in context [outgoing-gsm] for only 4 SIP clients |
| 11:59.35 | McDouglas | i'm not sure what to modify |
| 11:59.39 | BockBilbo | so i guess that if according to that error, a sound file is missing, it should be a be a bug on ubuntu, right? |
| 11:59.53 | BockBilbo | i think im going to do a bug report on it :) |
| 12:00.11 | mace | i'm not sure tbh, don't use precompiled packages myself |
| 12:00.24 | BockBilbo | :) |
| 12:00.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:00.35 | BockBilbo | thanks mace |
| 12:01.35 | mace | unrelated; i've set up call parking, and dialing the park extension correctly parks the call, and everything seems to be working.. except the announce of which park extension: asterisk console thinks it's announcing the number (701) but no audio is heard - any ideas? |
| 12:01.52 | mace | BockBilbo: you could maybe download the sound files by hand from digium? |
| 12:02.12 | BockBilbo | mace, the missing sound file isnt from digium |
| 12:02.18 | BockBilbo | is a translation on spanish |
| 12:02.22 | mace | ah right |
| 12:02.26 | BockBilbo | im going to try to download it |
| 12:02.32 | BockBilbo | and see if i still get the error |
| 12:02.41 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:02.45 | BockBilbo | if i dont, it'll be obvious that the file is missing in the ubuntu package |
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| 12:09.39 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:12.35 | BockBilbo | mace, supposly every foreign language files, should be inside a folder name by the language code at /var/lib/asterisk/sounds, right |
| 12:12.55 | BockBilbo | /var/lib/asterisk/sounds/es |
| 12:12.55 | BockBilbo | or |
| 12:12.57 | BockBilbo | /usr/share/asterisk/sounds/digits/es |
| 12:13.00 | BockBilbo | for instance? |
| 12:13.39 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:14.07 | mace | looks like it yeah |
| 12:14.18 | BockBilbo | ok |
| 12:14.19 | mace | in mine it's sounds/es |
| 12:14.30 | BockBilbo | and what about the digits? |
| 12:15.25 | mace | unfortunately i've only got the english samples installed; there are no directories in digits |
| 12:15.45 | mace | so whether that means they should be in sounds/es/digits/* or sounds/digits/es/* i'm not sure |
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| 12:17.07 | BockBilbo | dont worry |
| 12:17.10 | BockBilbo | ive just checked it out |
| 12:17.10 | BockBilbo | :) |
| 12:17.15 | mace | cool |
| 12:17.20 | BockBilbo | i need to make some symlinks |
| 12:21.04 | axscode | [Sep 18 04:18:08] ERROR[15618]: res_config_mysql.c:627 mysql_reconnect: MySQL RealTime: Failed to connect database server astbill on localhost (err 2002). Check debug for more info. |
| 12:21.10 | axscode | can anyone help me please. |
| 12:21.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:21.29 | ai-a | axscode: and the debug shows ? |
| 12:21.42 | axscode | ai-a: how am i going to show the debug? |
| 12:22.14 | ai-a | if an error says "You have an error, check debug" that message will be useless.. it stataes the debug needs checking... |
| 12:22.40 | axscode | yes, whats the command to show the debug? |
| 12:23.32 | ai-a | well a google search (as i dont use mysql in asterisk) states ... as you could research and search yourself without requiring people to use google for you... http://www.google.com .. goes to a page saying "if you turn debug on in logger.conf you'll see the actual error." |
| 12:24.33 | BockBilbo | mace |
| 12:25.00 | BockBilbo | could you tell me if you have a file called afernoon.gsm at /var/lib/asterisk/sounds/digits/ |
| 12:26.27 | jer | i'm having issues where when calling some numbers, it rings fine, then the other person picks up; the caller can talk for about 2 seconds then the line gets cut off. the other party isn't hanging up, and it doesn't happen to local pstn calls or even some calls out. i've got a debug log here: http://www.pastebin.ca/700628 and i'm really not sure why this is happening. any assistance would be great. |
| 12:27.07 | jer | and by some calls out i mean some long distance calls out (which go over an interconnect to another provider) |
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| 12:29.20 | mace | BockBilbo: i don't have that file |
| 12:29.26 | BockBilbo | interesting |
| 12:29.47 | BockBilbo | the file isnt either in the spanish language package ive just downloaded |
| 12:30.07 | BockBilbo | though, ive tried addeding an afternoon.gsm file in digits, and i dont get the error |
| 12:30.40 | mace | i don't have afternoon/morning/evening |
| 12:30.47 | mace | but i do have tomorrow/yesterday/today |
| 12:31.08 | mace | ah hang on |
| 12:31.16 | mace | afternoon is in the sounds directory |
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| 12:32.32 | BockBilbo | ah |
| 12:32.32 | BockBilbo | ok |
| 12:33.01 | BockBilbo | let me check the spanish package |
| 12:33.12 | BockBilbo | there it is |
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| 12:33.37 | BockBilbo | so i guess that is a problem of the asterisk build |
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| 12:51.43 | yang | What I need is a good SIP client for windows vista that also has DTMF support....I think that I tried 7 VOIP phones this far... |
| 12:52.04 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 12:52.26 | ai-a | yang: X-Lite |
| 12:52.41 | sniper[FOO] | yang: SJPhone? |
| 12:52.47 | McDouglas | i need some help with this dialplan: http://pastebin.com/d58c9f8d4 i want to restirct the usage of lines in context [outgoing-gsm] for only 4 SIP clients |
| 12:53.05 | *** join/#asterisk nacer (n=nacer@l.alcolo.a.mpl.pastIX.net) |
| 12:53.06 | ai-a | yakkop: i would like to know your list of sip phones not supporting dtmf. |
| 12:53.13 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 12:53.19 | nacer | hi |
| 12:53.31 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.220.191) |
| 12:53.49 | nacer | how can y re-launch the first setup of asterisk-now ? |
| 12:54.11 | nacer | the one in 7 step |
| 12:54.17 | hi365 | what are the changes to the language file layout in asterisk 1.4? |
| 12:54.33 | yang | ai-a: express phone, and i got a bunch which are binded to the VOIP provider only |
| 12:54.41 | nacer | i have reboot before finish because my difium card was not detected |
| 12:54.43 | nacer | digium |
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| 12:56.07 | yang | sniper[FOO]: i dont think sjphone works on vista? |
| 12:56.29 | yang | its not specified on the download site |
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| 13:00.26 | [TK]D-Fender | McDouglas: make a new context for the phones you want to restrict that has everything you want them to have and none of what you don't |
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| 13:06.56 | awk | somebody please give me some advicec |
| 13:07.12 | awk | hmm,I need to pipe everything coming from telnet ip 5038 (asterisk ami) to a file |
| 13:07.12 | Nugget | telnet is eeeeeeevil! |
| 13:07.20 | awk | any ideas |
| 13:07.42 | awk | it would be easy if i never needed to pass an action of login to the ami |
| 13:08.33 | JunK-Y | thats not so hard to pass a login action too, no? |
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| 13:08.53 | awk | JunK-Y hey? |
| 13:09.05 | awk | action: login; username: ... secret.... |
| 13:09.08 | awk | thats not what im after |
| 13:09.21 | awk | I want to push everything from ami to a log fileso I can give some guys the data stream |
| 13:09.28 | awk | thing is you cant just telnet ip port > blah.txt |
| 13:09.41 | awk | or else it wont give you the ability to pass the action |
| 13:10.23 | thewiizle | hi |
| 13:10.36 | JunK-Y | just login and listen everything thats comes in the socket and dump it in a file? |
| 13:10.44 | thewiizle | Can someone link me to a [from-internal] context |
| 13:10.56 | McDouglas | [TK]D-Fender: yes, thats the idea, but how do i put the phones into that context? |
| 13:10.57 | thewiizle | one that will allow internal dialling and then also outbound calls |
| 13:11.00 | awk | yes I can login, but i'm not piping when I do that |
| 13:11.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:11.04 | awk | im just viewing |
| 13:11.13 | awk | so must I copy and paste page by page to a txt file? |
| 13:11.18 | awk | thats jst crazy |
| 13:12.49 | awk | so ? |
| 13:14.00 | JunK-Y | i do that solution on a really busy system and works pretty well. |
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| 13:16.26 | [TK]D-Fender | McDouglas: You go into sip.conf, you look at where you clearly set their context the first time and you CHANGE IT |
| 13:17.11 | yang | must dtmf signaling be enabled in sip.conf -> dtmfmode = rfc2833 |
| 13:19.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:22.24 | GromiTM | Hi, sorry about my question, but, what is a trunk? |
| 13:22.59 | [TK]D-Fender | GromiTM: http://en.wikipedia.org/wiki/Trunk |
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| 13:23.12 | [TK]D-Fender | GromiTM: A term you're probably best not using or asking about. |
| 13:23.18 | GromiTM | Hmm ... always simple as that. :) Sorry.^^ |
| 13:23.44 | [TK]D-Fender | GromiTM: Its a messy thing to answer and rarely leads to anyone feeling any better about things. |
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| 13:26.06 | appelza | <PROTECTED> |
| 13:26.51 | [TK]D-Fender | appelza: There is no such thing |
| 13:27.04 | [TK]D-Fender | appelza: your dialplan does whatever you coded it to do. |
| 13:27.13 | GromiTM | [TK]D-Fender: ok ... |
| 13:27.36 | McDouglas | [TK]D-Fender: can i put a phone in two context? |
| 13:28.03 | [TK]D-Fender | McDouglas: No, and I'm quite sure I walked you through all this last week for some reason.... |
| 13:28.28 | [TK]D-Fender | McDouglas: You made a THIRD context that INCLUDES the other 2 and give them THAT as their context |
| 13:28.36 | ai-a | appelza: i think the samples has voicemail as 8500 |
| 13:29.12 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:31.06 | appelza | thanks ai-a |
| 13:31.45 | [TK]D-Fender | The Sample extensions.conf should never be used or even modded into place. It is psychotic mess showing tons of junk you really don't need to start and leaves you more work than building from scratch. Its best used as inpiration, or as a WARNING of what not to do. |
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| 13:32.25 | appelza | Thanks |
| 13:32.51 | appelza | I know, it also doesn't help me learn anything really. I just wanted to see voicemail in action. ;) |
| 13:33.30 | [TK]D-Fender | appelza: Exten => 12345,1,VoicemailMain(@default); <- yay, 1 line |
| 13:33.35 | McDouglas | [TK]D-Fender: hmm, but if i put the restricted phones in a separate context, how can they call each other? (restricted -> unrestricted) |
| 13:33.57 | [TK]D-Fender | Mclike I said make this new context INCLUDE all the others you want it to ahve access to. |
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| 13:37.04 | McDouglas | [TK]D-Fender: so it should look like this: http://pastebin.com/d33082009 ? |
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| 13:37.14 | JerJer | McDouglas: i call the concept that [TK]D-Fender just described as a landing context |
| 13:37.41 | JerJer | all of your devices land in a specific context that very simply includes the other contexts that actually provide the functionality |
| 13:38.13 | JerJer | McDouglas: it could be as simple as |
| 13:38.17 | JerJer | [staff] |
| 13:38.25 | JerJer | include => unrestricted-phone |
| 13:38.33 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 13:38.33 | JerJer | include => outgoing-gsm |
| 13:38.59 | JerJer | however, i put all of my devices within one context |
| 13:39.02 | [TK]D-Fender | McDouglas: See one thing you have realized is that 10-30 deserve their own CONTEXT |
| 13:39.10 | [TK]D-Fender | McDouglas: And THAT should be included in both. |
| 13:39.12 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:39.18 | lesouvage | I have this message in the cli. Does this indicate a serious problem. "15:26:06 WARNING[7390]: channel.c:780 channel_find_locked: Avoided deadlock for '0xb7209a98', 9 retries!" |
| 13:39.52 | [TK]D-Fender | McDouglas: And that you should have a Restricted that includes ALMOST everything, and an Unrestricted that includes Restricted PLUS whatever ADDITION stuff it deserves. |
| 13:39.59 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:40.11 | lesouvage | This message comes back once every 15 seconds with a couple of simaler lines at once. |
| 13:40.18 | [TK]D-Fender | JerJer: Put. Down. The. Cra..... oh forget it! :p |
| 13:41.35 | JerJer | McDouglas: http://pastebin.ca/700677 |
| 13:41.59 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 13:42.21 | JerJer | stations would account for all 3 digit extensions for this pbx (sip.conf would have corrosponding 3 digit [names]) |
| 13:42.36 | JerJer | then staff could call local, longdistance and other staff |
| 13:42.44 | JerJer | while public phones only can call local and staff phones |
| 13:42.51 | JerJer | er stations (to be correct) |
| 13:44.04 | McDouglas | and how do you make a difference in sip.conf between restricted and unrestricted stations? |
| 13:44.19 | JerJer | context=staff versus context=public-phones |
| 13:44.35 | [TK]D-Fender | McDouglas: Where as mine here would reduce the includes by a line and show a linear access heiracy so you can relate your 2 "ueser" contexts. http://pastebin.ca/700683 |
| 13:45.01 | [TK]D-Fender | user* |
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| 13:45.55 | JerJer | your staff cannot call stations |
| 13:46.08 | BockBilbo | is it possible to allow the users to access their own voicemails without using a password? Like, for instance, hanging up the phone, and wating some seconds before the messages can be heard? |
| 13:46.57 | shido6 | you can listen to the person recording the message in voicemail while its happening and sure you can access vmail without a password |
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| 13:48.05 | shido6 | difficulty level 0.....6...10 [6] "6" meaning you have more than 5 minutes of reading to do before implementing |
| 13:48.11 | thewiizle | is there a replacement for ${DIALEDPEERNUMBER} |
| 13:48.20 | shido6 | but less than an hr :) |
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| 13:48.52 | BockBilbo | shido6, but do you know if its posible to play the vmail after X seconds since the phone has been hanged up? |
| 13:49.05 | shido6 | play the voicemail where? :) |
| 13:49.15 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 13:49.28 | BockBilbo | on a phone connected to a an ATA |
| 13:49.59 | shido6 | you want the phone to call u and play the message? |
| 13:50.07 | BockBilbo | no |
| 13:50.22 | shido6 | you want to be on the phone already and have it beep you while you're on the phone wiht someone else? |
| 13:50.27 | BockBilbo | i want to play the messages on a phone without calling an extension |
| 13:50.34 | BockBilbo | no |
| 13:50.35 | BockBilbo | xD |
| 13:50.40 | thewiizle | i want the ability to dial numbers internally |
| 13:50.45 | shido6 | you want to pick up the phone and have it play a message? |
| 13:50.48 | BockBilbo | right |
| 13:50.48 | thewiizle | do i need a macro for that? |
| 13:50.49 | BockBilbo | xDD |
| 13:51.01 | shido6 | do you want to know how to do that? |
| 13:51.02 | JerJer | like a hotline phone |
| 13:51.07 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 13:51.17 | BockBilbo | shido6 that's it, i want to pick up the phone and have it play a message after 40 seconds |
| 13:51.18 | JerJer | the 'red' phone |
| 13:51.29 | shido6 | um |
| 13:51.31 | shido6 | 40 seconds? |
| 13:51.38 | BockBilbo | well.. 40 or 30 xD |
| 13:51.40 | shido6 | you must have a lot of time on your hands :) |
| 13:51.42 | BockBilbo | it doesnt matter |
| 13:51.44 | shido6 | thats a long wait |
| 13:51.44 | BockBilbo | xDD |
| 13:51.45 | BockBilbo | hehe |
| 13:51.47 | BockBilbo | so 10 |
| 13:51.47 | BockBilbo | xD |
| 13:51.49 | shido6 | how about immediately |
| 13:51.53 | BockBilbo | lmmm no |
| 13:51.54 | shido6 | or 3 seconds |
| 13:51.59 | BockBilbo | right |
| 13:52.00 | BockBilbo | xD |
| 13:52.03 | BockBilbo | whatever |
| 13:52.07 | BockBilbo | hehe... |
| 13:52.15 | BockBilbo | is it possible to do such thing? |
| 13:52.18 | shido6 | pick a number |
| 13:52.25 | JerJer | provided the sip phone can call a specific URL when you pick it up, you can wait as long as you want within the dialplan |
| 13:52.36 | JerJer | or if this is zaptel it is very trivial |
| 13:52.58 | BockBilbo | its for a lynksis ATA |
| 13:53.13 | BockBilbo | shido, i've already created an extension for accessing the voicemail |
| 13:53.19 | BockBilbo | its the 5555 number |
| 13:53.25 | shido6 | you may need to make another |
| 13:53.32 | shido6 | what does that do when u dial 5555 |
| 13:53.32 | BockBilbo | what for? |
| 13:53.41 | shido6 | does it ask you for a mailbox? |
| 13:53.46 | JerJer | your logic to wait |
| 13:53.56 | BockBilbo | right, and the password, |
| 13:54.04 | shido6 | but you dont want it to ask you for a mailbox |
| 13:54.07 | shido6 | or a password |
| 13:54.12 | shido6 | so guess what :) |
| 13:54.15 | BockBilbo | i actually dont want to call the number |
| 13:54.33 | shido6 | its going to call some # :) but the user picking up the phone wont know |
| 13:54.40 | BockBilbo | i want asterisk to call the extension automatically 10 seconds after i pick up the phone |
| 13:54.48 | BockBilbo | hehe |
| 13:54.48 | BockBilbo | xD |
| 13:54.53 | shido6 | so then you will "click over" |
| 13:55.04 | shido6 | if the line is already in use or off hook when you pick up the phone |
| 13:55.08 | shido6 | when asteirsk calls you |
| 13:55.11 | shido6 | you will have to click over |
| 13:55.11 | thewiizle | can anyone give me some ideas as to why i can not dial internal extensions |
| 13:55.15 | JerJer | BockBilbo: provided the linksys can do this - there will be a setting somewhere to call a specific URL |
| 13:55.23 | JerJer | but i do not recall seeing that feature |
| 13:55.27 | shido6 | thewiizle we need some kind of debug info.... |
| 13:55.33 | BockBilbo | mm i see |
| 13:55.37 | thewiizle | tell me what you need ;) |
| 13:55.40 | *** join/#asterisk jcanfield (n=jcanfiel@68.109.242.162) |
| 13:55.48 | JerJer | thewiizle: you have to tell us |
| 13:55.49 | thewiizle | im getting a 603 first of all |
| 13:55.58 | waverly360 | Has anyone here had much luck configuring AudioCodes devices? I'm having some disconnect supervision issues that I can't seem to resolve. |
| 13:55.58 | JerJer | turn on the appropriate debug and look through it |
| 13:56.04 | thewiizle | i can dial an echo test '*43' |
| 13:56.12 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:56.14 | thewiizle | just not other extensions in the same context |
| 13:56.33 | shido6 | the audiocodes will take about 30 minutes to configure or more the settings are there but there are so many friggin settings in teh audiocodes it will take you a while to tweak it |
| 13:56.38 | JerJer | 603 smells like a codec problem |
| 13:56.46 | thewiizle | i have ulaw&alaw |
| 13:57.04 | thewiizle | i can dial an echo test from both extensions |
| 13:57.10 | thewiizle | but extensions can not dial each other |
| 13:57.51 | thewiizle | i think its my dialplan for [from-internal] |
| 13:58.07 | thewiizle | if i make a dialplan like 77,Dial(SIP/102),30 |
| 13:58.08 | thewiizle | that works |
| 13:58.14 | thewiizle | when i dial 77 it works that is |
| 13:58.40 | thewiizle | if i have _.,Dial(SIP/${ARG1}),30 will that dial whatever i send to asterisk? |
| 13:58.55 | thewiizle | Its been a while since ive played with this :S |
| 13:59.18 | JerJer | ${ARG1} is a macro thing |
| 13:59.21 | JerJer | are you in a macro? |
| 13:59.28 | BockBilbo | shido6 what command do i have to use in order to do what i want? |
| 13:59.32 | thewiizle | im between a macro and a context |
| 13:59.38 | jer | i'm having issues where when calling some numbers, it rings fine, then the other person picks up; the caller can talk for about 2 seconds then the line gets cut off. the other party isn't hanging up, and it doesn't happen to local pstn calls or even some calls out the interconnect. Our asterisk seems to think the number exists locally, but I can't find reference to it anywhere. i've got a debug log here: http://www.pastebin.ca/700628 and i'm really not s |
| 13:59.48 | thewiizle | i was going to knock a macro together that will sort out my internal dialling |
| 14:00.12 | thewiizle | and then include that macro into my context but im not sure if thats a really arse about face way of doing it |
| 14:00.13 | JerJer | ${ARG1} only exists within the macro |
| 14:00.36 | appelza | you only exist within the macro |
| 14:00.37 | appelza | :D |
| 14:00.51 | thewiizle | ok, so how do i arrange for extension to extension dialling through a contexty |
| 14:00.54 | thewiizle | *context |
| 14:01.21 | JerJer | see my previous example |
| 14:01.22 | JerJer | http://pastebin.ca/700677 |
| 14:02.11 | thewiizle | same shizzle |
| 14:02.25 | JerJer | stations could be setup to call a standard macro - to incorporate voicemail or follow me |
| 14:02.53 | waverly360 | shido6: I've been playing with it for awhile now..and I can get it setup to make and receive calls just fine. It seems to be a rather dumb device when it comes to d/c supervision though. I've tried ALL of the options within the d/c supervision section..both indifidually..and different combinations. All of the research I've done thus far has shown me that others seem to have the same issues. |
| 14:03.28 | thewiizle | still a 603 jerjer |
| 14:03.29 | JerJer | waverly360: are you actually getting the disc supervision on the line ? |
| 14:03.46 | JerJer | thewiizle: then you have lower level problems than dialplan |
| 14:03.52 | JerJer | like SIP itself |
| 14:03.52 | thewiizle | hmm |
| 14:04.02 | JerJer | or the codec selection |
| 14:04.12 | shido6 | ok... the immediate dial thing is going to suck - u need to press SOME number on the PAP2T-NA's and it will contact your ast box - but to lift the phone and have anything other than a dial tone exist im not sure how to do - if this was a zap interface it would be a lot simpler and I could say YES you can do that. |
| 14:04.15 | [TK]D-Fender | JerJer>your staff cannot call stations <- Oh yes they CAN.... |
| 14:04.39 | JerJer | oh via public-phones |
| 14:04.44 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 14:04.44 | JerJer | i don't like that way |
| 14:04.45 | jcanfield | Is there a quick conary one liner to install LDAPget on *now? |
| 14:04.46 | shido6 | you could try to use S0 and have nothing in the dialplan on the pap2-tna |
| 14:04.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:04.48 | [TK]D-Fender | JerJer: Read the fine print :p http://pastebin.ca/700683 |
| 14:05.00 | shido6 | for immeidate dial but i ahve never tried it |
| 14:05.02 | axscode | how to load the zap interface again? |
| 14:05.07 | [TK]D-Fender | JerJer: And I don't like brussels sprouts! |
| 14:05.09 | waverly360 | JerJer: Here's the scenario: [softphone] - [asterisk] - [AudioCodes] - [Analog Line] |
| 14:05.40 | waverly360 | JerJer: If I dial into the Analog line, the call auto dials the softphone's extension through asterisk. |
| 14:05.40 | appelza | is there a way for volume boosting to happen? |
| 14:05.50 | [TK]D-Fender | JerJer: I try not to reinvent a context that has everything I want when I can simply ADD 10% to an existing one who is in a naturaly heirachy :) |
| 14:05.54 | axscode | hi |
| 14:06.08 | [TK]D-Fender | -y |
| 14:06.09 | waverly360 | JerJer: when I hangup the analog line, the audiocodes device keeps the connection to asterisk/softphone open. |
| 14:06.11 | JerJer | [TK]D-Fender: i think its easier for the non-iniated to have discreet landing contexts that include the other contexts that provide the required functionality |
| 14:06.39 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 14:06.46 | shido6 | BockBilbo - checkout https://www.myvoipstore.com.au/ModCoreFrontEnd/template.asp?pageid=112 or google pap2t-na immediate dial or pap2 immediate dial |
| 14:06.48 | JerJer | waverly360: find a phone that has a keypad that is lighted - by the phone line itself, it shouldn't take any batteries |
| 14:07.01 | shido6 | then u have a 1/2 hr of sip debugging and s,1 dialplan fun ahead of you |
| 14:07.14 | JerJer | plug it in to the audiocodes and have the far end hang up -- if the light blinks you have supervision on the line |
| 14:07.21 | [TK]D-Fender | JerJer: Wheich when you look at it... is EXACTLY what I'm doing. Mine jsut natually expands the "superior" user as the lOWER one gains more functionality as well. So in the big scheme mine makes long-term maintenance outside of over-specialization easier to manage :) |
| 14:07.22 | BockBilbo | thanks shido6 |
| 14:07.37 | shido6 | oh im not not... |
| 14:07.40 | shido6 | done |
| 14:08.05 | JerJer | [TK]D-Fender: do you geal with GUI users? We do |
| 14:08.19 | [TK]D-Fender | JerJer: Nope! Sanity FTW! |
| 14:08.22 | waverly360 | JerJer: Hmm...I don't have one...might have something else..will look around. |
| 14:09.06 | _ShrikE | waverly360: I had the same issue. I resolved it by setting the CurrentDisconnectDuration to 500 |
| 14:09.35 | JerJer | from a gui point of view its more logical to have discreet contexts that provide specific functionality - then specific landing context providing the desired suite of functionality |
| 14:09.46 | thewiizle | JerJer, if i had a codec issue surely i wouldn't be able to dial anything, including the echo test? |
| 14:10.06 | JerJer | not necessarily |
| 14:10.06 | thewiizle | i can make the phones dial each other if use 77,Dial(SIP/102),30 |
| 14:10.15 | thewiizle | i've enabled canreinvite |
| 14:10.26 | JerJer | what about the other phone ? perhaps its configured to only accept G.729 |
| 14:10.51 | JerJer | without actually seeing the CLI i can only guess |
| 14:10.58 | JerJer | but the CLI will tell you the problems |
| 14:11.10 | thewiizle | hmmm both have ulaw enabled |
| 14:11.13 | thewiizle | lemme enable sip debug |
| 14:11.14 | thewiizle | 2 secs |
| 14:11.25 | [TK]D-Fender | JerJer: Yeah I agree that if you're thinking from a GUI point of view separating a user context in ONLY access-level contexts is definately more readable. For text files you can save a nifty amount by not doing that. Not a huge difference, but I agree on your GUI approach |
| 14:11.27 | JerJer | you just have to turn on the appropriate info |
| 14:11.45 | JerJer | [TK]D-Fender: the power of asterisk :) |
| 14:12.09 | thewiizle | Just declined |
| 14:12.13 | [TK]D-Fender | JerJer: Yeah, the power for us to both be right given a certain frame of reference :) |
| 14:12.14 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 14:12.29 | JerJer | i hate guies all the way, but people want them, so i deal with it |
| 14:12.40 | *** join/#asterisk dhill (n=dhill@fog.mindcry.org) |
| 14:13.13 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:13.19 | dhill | hello, i am using asterisk 1.4.11. I have debug and verbose set to 5. I want to see how the extensions/dialplan is executed like in 1.2 via the CLI. How can I do this in 1.4? |
| 14:13.36 | coppice | if its good and its gui, its probably toffee |
| 14:14.23 | shido6 | Activating SpoonFeed 3k for 'BockBilbo' 9-17-07 |
| 14:14.32 | thewiizle | its gotta be my dialplan for [from-internal] |
| 14:14.35 | shido6 | Churning... done http://pastebin.ca/700708 |
| 14:14.37 | *** join/#asterisk ManxPower (n=manxpowe@175.sub-75-201-234.myvzw.com) |
| 14:15.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:15.30 | shido6 | crap |
| 14:15.38 | shido6 | and in voicemail.conf just have no password |
| 14:16.08 | _x86_ | hey every like 3 minutes (well, not precisely, but randomly, and close to roughly every 3 minutes), asterisk out of the blue just dials one of my SIP phones |
| 14:16.11 | *** join/#asterisk lolscorruptionof (i=aaron@nat/digium/x-d1184c8549e6a01d) |
| 14:16.25 | shido6 | or |
| 14:16.28 | _x86_ | and leaving dialtone voicemails if you don't answer it |
| 14:16.33 | _x86_ | what could be causing that? |
| 14:16.39 | [TK]D-Fender | _x86_: Provide some backup for these events please... |
| 14:16.41 | shido6 | get crazy with it and use the cid or the sipheader info for the user |
| 14:17.11 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 14:17.12 | shido6 | like if you're using Sylantro and dont get the right info from caller id but the header info is good |
| 14:17.14 | BockBilbo | oh! thanks a lot shido6 !!!:) im going to try it now :) |
| 14:17.29 | thewiizle | ok |
| 14:17.30 | thewiizle | right |
| 14:17.48 | *** join/#asterisk saftsack (n=saftsack@pD9E05C35.dip.t-dialin.net) |
| 14:18.00 | thewiizle | ive just put 101,1,Dial(SIP/101),30 in my [from-internal] |
| 14:18.04 | shido6 | yeah the spoonfeed 3k is a wonderful little guy who sits on my desk churning out dialplan info for green M&Ms and Blue Bawls |
| 14:18.08 | thewiizle | now it dials |
| 14:18.27 | shido6 | somtimes cheese |
| 14:18.33 | saftsack | if i have 2 phones on an S0 bus can i dial each other internal? |
| 14:18.35 | shido6 | but he smells after a few minutes from the cheese |
| 14:18.43 | tzanger | shido6: interesting |
| 14:18.46 | shido6 | the magic puffs |
| 14:18.48 | shido6 | well |
| 14:18.49 | shido6 | nevermind |
| 14:18.52 | tzanger | my dialplans are never magical |
| 14:19.06 | shido6 | they will after you smell one of these magic puffs |
| 14:19.41 | tzanger | I generate enough emissions on my own :-) |
| 14:19.48 | shido6 | it smells like a fart ... i should call it the attitude changer instead of magic puff |
| 14:20.26 | _x86_ | [TK]D-Fender: well I've got (2) T1's on a Sangoma A102D-x, 1 T1 going to a rhino FXS channel bank for my stations, and 1 T1 going to the PSTN |
| 14:20.37 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 14:21.00 | _x86_ | [TK]D-Fender: the T1 to the PSTN is outbound only, but when it happens, every available channel on the PSTN T1 calls the SIP phone |
| 14:21.15 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 14:21.17 | _x86_ | I have no clue why |
| 14:21.48 | _x86_ | when I came in this morning, that extension had over 100 voicemails of nothing but dialtone |
| 14:21.54 | _x86_ | from over the weekend |
| 14:22.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:22.29 | shido6 | FXO user? |
| 14:22.35 | stmaher | Hello everyone.. How do I enable sip call transfer capabilities into asterisk? |
| 14:22.39 | shido6 | _x86_ u using an FXO? |
| 14:23.18 | stmaher | Say i have an ivr.. the user makes a sip call to the asterisk box.. which is forwarded to the IVR.. i want to be able to use transfer capabilities.. |
| 14:23.39 | stmaher | IN a wireshark it states asterisk declines with a 603 |
| 14:23.58 | _x86_ | shido6: both T1's are b8zs, esf. PSTN T1 is using e&m wink signalling, channel bank T1 is using kewlstart signalling |
| 14:24.10 | shido6 | E&M |
| 14:24.25 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 14:24.33 | bobbytux | lo |
| 14:24.42 | shido6 | can you use something other than E&M ? |
| 14:24.52 | ManxPower | stmaher: you press the transfer button on your SIP phone. |
| 14:24.58 | shido6 | lol |
| 14:25.03 | _x86_ | shido6: why? |
| 14:25.07 | ManxPower | shido6: if you can't do PRI, I'm a fan of E&M/Wink. |
| 14:25.24 | shido6 | i had a very bad experience with e&m and skype |
| 14:25.34 | _x86_ | shido6: and no I can't, it's the standard CAS signalling for the major LEC we're using (AT&T) |
| 14:25.37 | shido6 | maybe you should chat with MP |
| 14:25.47 | stmaher | ManxPower: its not that simple.. the IVR does the transfer of the call.. Im testing a blind transfer first.. |
| 14:26.04 | stmaher | eg: if we have 3 devices.. a sip phone, asterisk and an ivr |
| 14:26.05 | ManxPower | shido6: it's been almost a year since I've done E&M/Wink. We converted the last of our lines to PRI. |
| 14:26.07 | shido6 | how are you transfering stmaher? |
| 14:26.12 | shido6 | LOL |
| 14:26.33 | ManxPower | stmaher: it does not matter, the transfer button should still work. |
| 14:26.52 | stmaher | The ivr picks up the call and plays a prompt.. it then makes a transfer request to asterisk to transfer the call to another locaiton |
| 14:28.12 | stmaher | Does this make sense? would you like me to explain it in another way? |
| 14:28.38 | stmaher | Sorry the IVR is not the one that comes with asterisk.. |
| 14:28.46 | ManxPower | stmaher: it does not make a "transfer request" |
| 14:28.48 | stmaher | Its a third party IVR |
| 14:29.01 | stmaher | Yes.. to which the asterisk box states 603 declined |
| 14:29.08 | ManxPower | It simply Dials another destination. It is not called a Transfer |
| 14:29.11 | shido6 | SEP 17 10:28:23 WARNING[76542] stomach.c: Context 'hunger' tries to includes nonexistent context 'steak' |
| 14:29.57 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:30.05 | ManxPower | stmaher: you need to find out why Asterisk is responding with a 603 Declined. |
| 14:30.19 | shido6 | SEP 17 10:28:23 WARNING[76542] stomach.c: Unable to register food item |
| 14:30.47 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:31.41 | shido6 | SEP 17 10:29:09 NOTICE [28510] stomac.c: wake mind find food NOW |
| 14:33.28 | ManxPower | stmaher: you will have to do a sip debug and try to figure out why the 603 is being generated. Most of these moronic SIP PBXs don't ever even send the auth info to Asterisk. |
| 14:34.03 | ManxPower | The Nortel BCM is one of the moronic SIP PBXs we can't figure out how to configure it to send authentication informtation. |
| 14:37.56 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:38.05 | BockBilbo | could you guys tell me about a good cordless ip phone to buy? |
| 14:38.26 | [TK]D-Fender | ~wifisip |
| 14:38.27 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 14:38.29 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 14:38.38 | ManxPower | BockBilbo: every single one of the sucks. As you would already know if you looked at the mailing list archives. |
| 14:38.56 | JerJer | acutally the few WIFI phones that support WMM really sound good |
| 14:38.57 | ManxPower | The best option is to use a good cordless phone with a SIP ATA. |
| 14:39.01 | BockBilbo | i havent check the mailing list |
| 14:39.18 | JerJer | just kinda sucks from the talk time point of view - which will get fixed as tech improves |
| 14:39.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:39.43 | JerJer | ManxPower: DECT baby |
| 14:40.03 | BockBilbo | so, in your opinion, the best choice is o use a normal cordless phone with a sip ata?... |
| 14:40.05 | ManxPower | The problems people seem to have with these phones are: low talk time, low standby time, poor sound quality, WEP not working, roaming between access points. |
| 14:40.23 | BockBilbo | wow.. |
| 14:40.24 | JerJer | enter UMA |
| 14:40.32 | BockBilbo | if wep doesnt work, neither wpa2.... :S |
| 14:40.36 | ManxPower | JerJer: I think I bought the worst DECT phone on the planet. |
| 14:40.41 | tzanger | my current love is the SIP DECT phones from Aastra |
| 14:40.48 | BockBilbo | what about a dect supporting sip using ethernet? |
| 14:40.55 | ManxPower | JerJer: expand your acronyms |
| 14:40.58 | McDouglas | is there a way to see the caller id in asterisk console? |
| 14:41.06 | McDouglas | i can only see the channel names |
| 14:41.25 | ManxPower | McDouglas: sip show channel X |
| 14:41.28 | shido6 | ut oh |
| 14:41.39 | ManxPower | where the channel name is from "sip show channels" |
| 14:41.48 | tzanger | heh not bad, 2100 messages on my mailing list account |
| 14:41.48 | JerJer | Unlicensed Mobile Access |
| 14:41.53 | shido6 | dont drink expired immune assisting products... like yogurt drinks... |
| 14:42.03 | shido6 | <PROTECTED> |
| 14:42.19 | shido6 | you would think is O kay... not true |
| 14:42.23 | JerJer | aka the T-mobile d-link / wrt device that works with like 3 handsets |
| 14:42.28 | McDouglas | ManxPower: but to do it automaticaly? |
| 14:42.50 | ManxPower | McDouglas: you mean like you look at the screen and the caller ID automagically appears? |
| 14:42.57 | McDouglas | yes |
| 14:43.23 | ManxPower | I think app_telepathy is slated for Asterisk 2.6 to be released sometime around 2045. |
| 14:44.01 | yang | Hrm I am having difficulties, I have ekiga runing and i can hear the ringing sounds....but when the conetion is established i see OUT:PCMU IN:PCMU and i hear no sound |
| 14:44.30 | McDouglas | ManxPower: no i meant, when i call asterisk from a cell phone i can see this in the console: http://pastebin.com/d1475c9e3 |
| 14:44.49 | McDouglas | ManxPower: but i want to see the caller id too |
| 14:44.50 | [TK]D-Fender | McDouglas: No, there is no "automatic". |
| 14:45.05 | [TK]D-Fender | McDouglas: then NoOp it YOURSELF in your dialplan |
| 14:45.13 | *** join/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 14:45.53 | ManxPower | McDouglas: Just put a Noop(CALLERID(all) is ${CALLERID(all)}) in the dialplan where the call comes in. |
| 14:47.03 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 14:47.18 | elixer | [TK]D-Fender: listened to that .wav this morning. is that from 'shaft?' |
| 14:47.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:47.24 | ManxPower | yang: sounds like a NAT problem to me. |
| 14:47.31 | [TK]D-Fender | elixer: "Long Kiss Goodnight" |
| 14:47.40 | elixer | [TK]D-Fender: ah |
| 14:48.28 | ManxPower | McDouglas: why are you allowing outside callers to transfer themselves out of your PBX? |
| 14:48.57 | McDouglas | ManxPower: i do? |
| 14:49.13 | ManxPower | McDouglas: the Tt on the Dial line does that. |
| 14:49.13 | Aeudian | Good Afternoon, We are going to be setting up asterisk servers in multiple locations but need the ability to do cross dialing between systems. We have decided to use a 7, followed by a company location code, lets say 01, followed by the extension, xxx. So the stream would look like 701xxx. we wanted the ability for the end users to just dial xxx and have a database check occur, which would also find the company code, 01, and dial the number automatica |
| 14:49.25 | ManxPower | Why else would you use those horrid evil options |
| 14:50.41 | ManxPower | Aeudian: We just make sure we do not have overlapping dialplans at our various offices. |
| 14:51.00 | McDouglas | MaliutaWrk: well, i wanted to allow the internal extensions to transfer |
| 14:51.32 | [TK]D-Fender | Aeudian: exten => _701XXX,1,Dial(SIP/peertocompany01/${EXTEN:3})) |
| 14:51.54 | [TK]D-Fender | McDouglas: pay attention to WHO gets to do the transferring... |
| 14:52.40 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:52.41 | Aeudian | ManxPower: we want to just assign a number to x company, say x company has ext range of 500-599, but that range can be located in baltimore, washington, denver. we want the database to be able to find out which location that extension lives in and dials an invite to the correct asterisk system |
| 14:52.46 | ManxPower | McDouglas: why not just use the transfer button on the phone? |
| 14:53.21 | ManxPower | Aeudian: seems like a lot of work to avoid having to have a good unified dialplan |
| 14:53.43 | McDouglas | ManxPower: there is no transfer button on this shitty phone |
| 14:53.49 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 14:53.50 | [TK]D-Fender | Aeudian: Then you'd need to figure out what database you want to do this in and get coding your dialplan to look it up. |
| 14:54.31 | ManxPower | On my system, the last 4 digits of the DID is the same as the extension. 35xx is Causeway, 31xx is cororpoate, 34xx is relocation, 38xx is Ormond, 32xx is Gulfport, etc. |
| 14:54.43 | [TK]D-Fender | Aeudian: At which point you'd likely do : exten => _7XXXXX,1,NoOp(Time to parse the shit outta this exten and spend 20+ lines to retrieve DB info CONSTANTLY....) |
| 14:54.45 | ManxPower | then we just route based on the first 2 digits of the dialed extension. |
| 14:54.48 | Aeudian | ManxPower: ya, it does, i was thinking of doing 100-199 per company and having the users dial 711100 where 7 would be dial out, 1 would be company code, 1 be location code, and 100 being extension, but that requires users to know where they are calling |
| 14:54.52 | ManxPower | This is how traditional PBXs do it. |
| 14:55.07 | Aeudian | [TK]D-Fender: so you think its a bad idea? |
| 14:55.11 | ManxPower | McDouglas: sounds like you need a better phone. |
| 14:55.48 | ManxPower | McDouglas: regardless, you can do a "show application dial" to see what the t and T options do and which ONE you need. |
| 14:55.57 | [TK]D-Fender | Aeudian: a DB entry for EACH? Yeah. You're going to code a dynamic system, and for what? You'll have to code their extens, and THEN a DB to parallel it? A lot of work for no real reason. |
| 14:56.07 | [TK]D-Fender | Aeudian: Just throw the call over and be done with it. |
| 14:56.23 | [TK]D-Fender | McDouglas: Which phone? |
| 14:56.31 | Aeudian | [TK]D-Fender: ya, i agree, thanks |
| 14:56.42 | McDouglas | [TK]D-Fender: Dlink DPH-300S |
| 14:57.47 | [TK]D-Fender | McDouglas: Any idea why the only pages I can find for it are in RUSSIAN? |
| 14:58.01 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.35) |
| 14:58.20 | [TK]D-Fender | McDouglas: And I would bet it has a transfer function, and you simply haven't found it yet |
| 14:58.21 | McDouglas | [TK]D-Fender: as i said... shitty phone |
| 14:58.46 | [TK]D-Fender | Oooooh... POLISH now... |
| 14:58.51 | ManxPower | McDouglas: Really, your lack of foresight in picking your phone really is your problem, not ours. |
| 14:59.04 | [TK]D-Fender | Yup... |
| 14:59.06 | [TK]D-Fender | ~wglwat |
| 14:59.07 | jbot | somebody said wglwat was well, good luck with all that |
| 14:59.35 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:59.57 | ManxPower | Phones that work well with asterisk are well known and many people use them. |
| 15:00.01 | ManxPower | ~phones |
| 15:00.01 | jbot | rumour has it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 15:00.20 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 15:00.34 | [TK]D-Fender | ManxPower: Hindsight is 20/20 |
| 15:01.08 | [TK]D-Fender | ManxPower: Makes me glad I've only got 2 Uniden UIP-200's :) |
| 15:01.09 | ManxPower | [TK]D-Fender: maybe so, but that doesn't mean we have to support bad phones. |
| 15:01.18 | [TK]D-Fender | ManxPower: Indeed , we don't :) |
| 15:01.33 | ManxPower | [TK]D-Fender: we bought about 5 different SIP phones before we picked Polycom as the standard. |
| 15:02.35 | goupil | bobbytux: hi |
| 15:04.01 | bobbytux | lo goupil |
| 15:05.50 | BockBilbo | is there anyway i could know the vmail id of a sip user by its callerid? |
| 15:06.23 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 15:06.23 | ManxPower | BockBilbo: there is no way to know the voice mailbox number based on anything. |
| 15:06.31 | ManxPower | unless you design it that was in the first place. |
| 15:06.50 | BockBilbo | what do you mean? |
| 15:07.51 | [TK]D-Fender | BockBilbo: If my callerid is 1234 how is * supposed to know that his box is 456? |
| 15:08.38 | BockBilbo | well, my callerdid is related to a sip user, which has a mailbox variable configured on it. |
| 15:09.17 | BockBilbo | so if i know the callerdid of the caller, i could know which is his mailbox and the context |
| 15:09.22 | BockBilbo | can't I? |
| 15:09.23 | [TK]D-Fender | BockBilbo: And is the box # the SAME as the callerid number? |
| 15:09.33 | BockBilbo | yes |
| 15:09.50 | [TK]D-Fender | BockBilbo: Then in that case you already kNOW the box. it IS his callerID # |
| 15:10.01 | [TK]D-Fender | BockBilbo: So what about it? |
| 15:10.20 | BockBilbo | i wanted to know how to get that variable |
| 15:10.33 | [TK]D-Fender | BockBilbo: ${CALLERID(num)} |
| 15:10.44 | BockBilbo | ah |
| 15:10.46 | BockBilbo | thanks |
| 15:10.46 | BockBilbo | :) |
| 15:10.47 | [TK]D-Fender | BockBilbo: Go read up on "asterisk variables" and "asterisk functions" on the WIKI. |
| 15:10.51 | BockBilbo | ok |
| 15:11.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:12.19 | lesouvage | I have this message repeatedly in the cli. Does this indicate a serious problem. "15:26:06 WARNING[7390]: channel.c:780 channel_find_locked: Avoided deadlock for '0xb7209a98', 9 retries!" |
| 15:14.23 | ManxPower | lesouvage: make sure you are running the latest version of Asterisk/Zaptel of the major release you are using. |
| 15:14.55 | ManxPower | they are fixing deadlock issues all the time. |
| 15:16.12 | lesouvage | ManxPower: but is this a missage to be really worry about? |
| 15:16.28 | lesouvage | missage is message |
| 15:17.40 | styelz | can anyone tell my why evrything would work. with sip calls.. but not music on hold. i get no audio, on external sip calls |
| 15:17.48 | styelz | i get normal audio though |
| 15:18.05 | styelz | MOH works locally |
| 15:18.41 | lesouvage | ManxPower: version 1.2.24 is running on this asterisk box |
| 15:21.05 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:21.10 | Aeudian | Can anyone provide a link or explain how "switch =>" works |
| 15:21.47 | BockBilbo | thanks a lot |
| 15:21.52 | BockBilbo | i have to go now |
| 15:21.53 | BockBilbo | bye! |
| 15:21.54 | shido6 | :) |
| 15:21.57 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:22.04 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 15:22.05 | kippi | hey# |
| 15:22.26 | [TK]D-Fender | Aeudian: ... |
| 15:22.28 | [TK]D-Fender | ~wikis |
| 15:22.28 | jbot | from memory, wikis is http://www.voip-info.org |
| 15:22.30 | kippi | how easy is it to listen into calls from one handset to another |
| 15:23.10 | kippi | sip pref |
| 15:23.36 | *** join/#asterisk McDouglas (n=mcd@wxdna0kdgg.adsl.datanet.hu) |
| 15:23.55 | ManxPower | kippi: is the audio going thru asterisk, or is it going direct between the two end points? |
| 15:24.14 | kippi | thru asterisk |
| 15:24.17 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 15:24.26 | kippi | can I use somthing like ZapBarge, but for sip? |
| 15:24.27 | ManxPower | kippi: "show application chanspy" |
| 15:24.52 | McDouglas | [TK]D-Fender: sorry, i had to go home. Anway that telephone you was asking about: same as: dph-c160s |
| 15:24.59 | nny | um.. i am getting unable to open asterisk-database errors in asterisk/messages .. but I dont use mysql or postgres to log CDR data... why would this suddenly start throwing errors? |
| 15:25.16 | McDouglas | [TK]D-Fender: which is australian only, and the dph-300s is east european only |
| 15:25.17 | nny | (i dont even run mysql or postgres on this box) |
| 15:25.45 | ManxPower | nny: chances are you have the dev libs and headers installed when you built asterisk |
| 15:25.54 | ManxPower | (for the databases, that is) |
| 15:26.27 | *** join/#asterisk |omni| (n=rob@c-67-185-70-220.hsd1.wa.comcast.net) |
| 15:26.40 | nny | ManxPower: is there a way to unfsck it or should I just reinstall? |
| 15:26.42 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 15:26.58 | nny | ManxPower: or, is there a way to tell asterisk to not look for a DB? |
| 15:27.08 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-254.lsanca.dsl-w.verizon.net) |
| 15:27.36 | ManxPower | nny: you didn't actually tell us what the messages are, but I just ignore them -- they are harmless |
| 15:27.44 | *** join/#asterisk pHnz (n=david@rhesus.net) |
| 15:27.47 | thansen|laptop | what permissions does asterisk need on /dev/zap/* in order to use MeetMe? |
| 15:28.15 | ManxPower | nny: you can always noload => whatever.so in /etc/asterisk/modules.conf |
| 15:28.16 | nny | ManxPower: FWIW I had to reinstall with 1.2.24 asterisk as we were diagnosing some issues (turned out to not be asterisk related) |
| 15:29.33 | ManxPower | thansen|laptop: normally root, I would imagine. |
| 15:29.34 | [TK]D-Fender | McDouglas: Manual is useless.... |
| 15:29.36 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 15:29.45 | McDouglas | [TK]D-Fender: yes |
| 15:29.50 | ManxPower | McDouglas: does the phone have a FLASH or RECALL button? |
| 15:29.51 | nny | ManxPower: but yeah the error is Sep 17 11:27:58 WARNING[4240] db.c: Unable to open Asterisk database |
| 15:29.56 | McDouglas | ManxPower: nope |
| 15:29.59 | nny | and the system is in distress ATM |
| 15:30.10 | thansen|laptop | ManxPower: you mean all 3 rwx? |
| 15:30.12 | nny | and thats the only pertinent error I could see |
| 15:30.12 | ManxPower | nny: that is not a postgress or mysql message. |
| 15:30.18 | nny | never said it was |
| 15:30.26 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 15:30.32 | ManxPower | nny: "make datafiles" in the asterisk source. |
| 15:30.32 | nny | nny: um.. i am getting unable to open asterisk-database errors in asterisk/messages |
| 15:30.58 | ManxPower | nny: your lack of English skills confused me, sorry. |
| 15:31.06 | nny | ManxPower: charming arent you |
| 15:31.39 | coppice | English skills? you mean moaning about the weather? |
| 15:31.43 | ManxPower | nny: if I was speaking your native language it would me much worse. |
| 15:31.53 | shido6 | if code had feelings things would compile a lot longer or not compile at all |
| 15:32.01 | nny | ManxPower: was that english? |
| 15:32.10 | ManxPower | did "make datafiles" fix your problem or not? |
| 15:32.30 | *** join/#asterisk Mw3_ (i=mw3@ip59934bd1.rubicom.hu) |
| 15:32.32 | styelz | er, MOH recordiong was too low |
| 15:32.40 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:32.50 | nny | ManxPower: I am recompiling, why isn't that step listed anywhere in both voip-info.org howto for compiling or in Asterisk O Reilly book? |
| 15:32.53 | McDouglas | is there a way to see what channels are currently active in an asterisk system? Like a brief list of every conversation |
| 15:32.57 | nny | in the* |
| 15:33.10 | shido6 | a lot of stuff isnt in the book |
| 15:33.12 | styelz | sip show channels ? |
| 15:33.13 | ManxPower | nny: I guess they assume you will never remove the datafiles out from under Asterisk. |
| 15:33.14 | shido6 | or on voip-info |
| 15:33.29 | ManxPower | McDouglas: "show channels" |
| 15:33.37 | nny | I wouldn't have.. look I am not trying to be an ass, I just want to know what step broke the process |
| 15:33.55 | nny | how would I have removed those files? I don't believe I did. |
| 15:33.57 | ManxPower | nny: the data files are not there, or are not readable by the asterisk user. |
| 15:34.31 | nny | by the files is this the log files in /var/log/asterisk/cdr-custom or another set of files? |
| 15:34.36 | ManxPower | how they got into that state is not something I can really speculate about. |
| 15:34.42 | ManxPower | nny: no. |
| 15:35.53 | ManxPower | those are LOG files. The default location for the data files would be /var/lib/asterisk |
| 15:36.29 | *** join/#asterisk Trionnis (i=lordkuri@s233-51-251.nap.wideopenwest.com) |
| 15:36.47 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 15:37.44 | Trionnis | 9/11 was an inside job... Elvis was an alien... Tinfoil keeps them from reading your mind... |
| 15:37.53 | Trionnis | wait... not that kinda speculation? |
| 15:37.59 | nny | ManxPower: I am going to be reverting back to 1.4.5 as the move to 1.2 was purely for diagnostic reasons on a completely unrelated issue, so I will make sure to do that step int he process of compiling. BTW, /var/log/asterisk/astdb is read only by the asterisk user |
| 15:38.02 | nny | owned by root |
| 15:38.25 | nny | this file should be r/w by asterisk I presume/ |
| 15:38.26 | nny | ? |
| 15:38.41 | ManxPower | nny: it should be owned by the user Asterisk runs as. |
| 15:38.45 | [TK]D-Fender | Trionnis: Warning, if you use a collander then the little holes may allow them some level of influence. Ensure that your tin-foil hat is solid... |
| 15:38.48 | *** join/#asterisk Corydon76-lap (i=Corydon7@pdpc/supporter/sustaining/Corydon76-home) |
| 15:38.48 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 15:38.50 | nny | asterisk is sun as asterisk |
| 15:38.52 | nny | run* |
| 15:38.54 | Trionnis | GASP |
| 15:38.56 | ManxPower | Generally Asterisk defaults to running as root. |
| 15:38.57 | nny | user |
| 15:39.16 | nny | what about the remaining files in /var/lib/asterisk ? |
| 15:39.29 | nny | should they be full perms for asterisk user or read/execute |
| 15:39.30 | nny | ? |
| 15:39.39 | alrs | ouch: https://support.rhinoequipment.com/blog/2007/09/14/r4t1-ec-hardware-upgrade/ |
| 15:39.46 | ManxPower | nny: I would assume so. |
| 15:39.54 | pHnz | Wich cisco phone are recommanded for Asterisk ? |
| 15:39.54 | ManxPower | I never mess with running Asterisk as non-root. |
| 15:39.57 | nny | ManxPower: than that appears to be the issue |
| 15:40.08 | ManxPower | I don't have machines that connect to the internet. |
| 15:40.14 | nny | ManxPower: gotcha |
| 15:40.53 | [TK]D-Fender | alrs: Whats with the "ouch"? Not a difficult looking process. |
| 15:41.36 | Trionnis | anyone happen to have some info for getting * to function as a straight unauthenticated SIP proxy? similar to OpenSER? |
| 15:41.52 | alrs | [TK]D-Fender: obviously not difficult, just fucking retarded |
| 15:41.52 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:42.01 | Trionnis | I've got a box here that needs to make calls out through it and it does not appear to be following the rfc's worth a damn... |
| 15:42.04 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:42.26 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:42.47 | ManxPower | Trionnis: Asterisk is not a SIP PROXY and nothing you can do will make it a SIP proxy. If you need a SIP proxy, then use a SIP proxy. Asterisk is a B2BUA in "SIP terms" |
| 15:43.05 | [TK]D-Fender | alrs: Really, what do you say that? |
| 15:43.32 | Trionnis | ok, sorry for the techincalities, that's what I really need |
| 15:43.37 | [TK]D-Fender | Trionnis: * is NOT a PROXY, it is a B2BUA, nothing more, nor will it ever likely be. |
| 15:43.47 | Trionnis | have to take incoming sip, and send it out an iax channel |
| 15:44.14 | alrs | [TK]D-Fender: Zero concept of package management. Rebooting the machine. |
| 15:44.15 | [TK]D-Fender | Trionnis: Well you DEFINATELY can't proxy that... |
| 15:44.17 | Trionnis | the problem is that VoiceGenie isn't following the standards, doesn't register properly, and does not appear to be sending some of the headers right |
| 15:44.44 | ManxPower | Trionnis: and you know this how? |
| 15:44.50 | Trionnis | wait... you're telling me that I can't have one side of a call come in via sip, and go out the other side via iax? |
| 15:44.53 | [TK]D-Fender | alrs: http://icanhascheezburger.com/2007/01/16/go-cry-emo-kid/ |
| 15:45.03 | alrs | [TK]D-Fender ? |
| 15:45.05 | ManxPower | i.e. how do you know it is not a config issue and rather a crappy device/gateway/provider issue. |
| 15:45.15 | Trionnis | because everything else works |
| 15:45.23 | nny | [TK]D-Fender: lol |
| 15:45.24 | Trionnis | softphones, hardphones, etc |
| 15:45.38 | Trionnis | the only time it becomes an issue is when VG tries to send a call out |
| 15:45.49 | ManxPower | Trionnis: sure you can one side of a call come in via SIP and the other side be IAX. THAT IS NOT A SIP PROXY |
| 15:46.25 | Trionnis | as I said the first time, I used the wrong term |
| 15:46.42 | Trionnis | yelling won't solve anything mate :) |
| 15:46.48 | ManxPower | A SIP proxy only talks SIP, and just routes SIP messages between SIP devices. |
| 15:46.57 | nny | ManxPower: Yeah FWIW I am still trying to sort out the best way to handle reinstalls as I had to hand remove some of 1.4.5's libs etc. when I instaled 1.2.24. It seems make-clean, well doesn't. |
| 15:46.58 | tzanger | it's ManxPower... yelliung does help :-) |
| 15:47.13 | ManxPower | Trionnis: So, lets start over with your question/issue without using the wrong terms. |
| 15:47.13 | kippi | ok, using ChanSpy I only seem to get getting the voice from my phone and not the other callers |
| 15:47.18 | Trionnis | debug from an attempted outcall: http://pastebin.ca/700803 |
| 15:48.01 | Trionnis | the From header is incorrect for starters, and the config on the VG box doesn't follow the manual for setting up registration... I've already tried to get them to address it with no luck |
| 15:48.23 | ManxPower | Trionnis: is the VG box on a static IP? |
| 15:48.26 | Trionnis | yep |
| 15:48.33 | Trionnis | same lan as the * box |
| 15:48.35 | ManxPower | then you do not need to have it register to Asterisk. |
| 15:48.52 | Trionnis | no idea why it's using NAT either, as it's set off sip.conf |
| 15:48.55 | [TK]D-Fender | Trionnis: Though you were only stuck with that lack of audio hand-off we discovered at the end... |
| 15:49.14 | Trionnis | well |
| 15:49.14 | ManxPower | The ONLY thing registration is for is to notify the remote side what your current IP is for that user/password. It does nothing else and is not needed for devices on a static IP. |
| 15:49.15 | Trionnis | I was |
| 15:49.22 | Trionnis | but then it started acting up again |
| 15:49.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:49.27 | Trionnis | and now it won't register |
| 15:49.34 | Trionnis | er |
| 15:49.36 | Trionnis | pass calls |
| 15:49.37 | [TK]D-Fender | :/ |
| 15:50.09 | ManxPower | Trionnis: Please put down the booze and step away from the keyboard. |
| 15:50.25 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:50.56 | McDouglas | is there any way i could make the sound of the phones better? what i mean is, i can hear a not too loud, but constant sizzling sound |
| 15:51.03 | ManxPower | I don't really see anything in your pastebin that would indicate a problem. |
| 15:51.22 | ManxPower | McDouglas: I'm sure there is, but I've never had an issue with audio quality. |
| 15:52.15 | McDouglas | then probably my "cool" dlink phone :P |
| 15:52.22 | McDouglas | is at fault |
| 15:52.29 | [TK]D-Fender | McDouglas: Set it to simmer and add a little evaporated milk. |
| 15:52.51 | ManxPower | McDouglas: perhaps your phone is sending 30ms audio packets instead of the 20ms asterisk is expecting. |
| 15:52.51 | McDouglas | yep, that would i help i guess |
| 15:53.09 | ManxPower | Linksys phones are known to do this. |
| 15:53.31 | McDouglas | can i change asterisk? (since there is almost no settings on the phone :P) |
| 15:54.08 | Trionnis | ManxPower: the problem is that it keeps throwing a 407 at voicegenie when it attempts to send an invite |
| 15:54.09 | ManxPower | McDouglas: you might be able to in 1.4, I don't know. |
| 15:54.30 | Trionnis | even though I have insecure set, allowguest set, and everything else |
| 15:54.58 | ManxPower | Trionnis: that is how ALL SIP calls start. Send an invite without auth, if it is rejected with auth required, then resend with the auth info. |
| 15:55.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:55.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:55.30 | ManxPower | Trionnis: is insecure=yes or insecure=very ? |
| 15:55.32 | Trionnis | and VG doesn't send auth |
| 15:55.39 | Trionnis | insecure=port,invite |
| 15:55.44 | Trionnis | this is 1.4.11 |
| 15:55.52 | pHnz | Cisco 7960G phone are 100 % compatible with Asterisk ? |
| 15:56.16 | ManxPower | Trionnis: so your REAL issue is that Asterisk is requiring auth even when it should not. |
| 15:56.31 | ManxPower | pHnz: Short answer: Yes, Long Answer: not really. |
| 15:56.44 | Trionnis | mostly, yes |
| 15:57.12 | pHnz | ManxPower wich one are u suggesting (in cisco brand) |
| 15:57.18 | ManxPower | pHnz: The phone, if it has the SIP firmware, works just fine -- but you won't have access to any of the really cool features that are provided by Cisco Call Manager (CCM) |
| 15:57.33 | ManxPower | pHnz: I recommend Polycom |
| 15:58.01 | pHnz | thx |
| 15:58.04 | nny | HA |
| 15:58.14 | *** join/#asterisk plantseeker (n=chatzill@77.240.56.18) |
| 15:58.16 | ManxPower | Cisco has some of the most expensive and hard to understand licenses for their SIP firmware. |
| 15:58.30 | *** join/#asterisk shinao1 (n=shinao1@196-220-29-237.netcomng.com) |
| 15:58.38 | pHnz | duhh |
| 15:58.41 | pHnz | Bull shit =p |
| 16:01.07 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:01.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:01.21 | ManxPower | pHnz: $100 for a legal copy of the SIP firmware. |
| 16:01.36 | nny | got an RFP this morning for a major university.. seems they want a device to add a way to broadcast messages (voice) and events (911 is called, broadcasts to 1100 phones) or when (?) a camera has ane vent? |
| 16:02.13 | [TK]D-Fender | pHnz: You'll forget Cisco if you know whats good for you |
| 16:02.17 | nny | funny thing is, it's for Cisco phones.. the only response we have is replace the pbx.. they seem to think some magic box is going to integrate with the cisco garbage and handle that |
| 16:02.22 | ManxPower | If you sell the phone, you cannot transfer the license for the firmware. |
| 16:03.22 | nny | so basically they bought cisco because some PHB get's blown by their PR department, and now they have to pay someone to add the functionality they need as a magic box.. yeah.. um, one big bucket of FAIL for you |
| 16:03.49 | ManxPower | Oh, and the phones don't come with a power supply by default -- you have to buy one -- the retail is $45 for a power supply. |
| 16:03.51 | pHnz | okay thank's. |
| 16:04.05 | pHnz | possibility of POE ? |
| 16:04.11 | pHnz | because i have a cisco switch with POE |
| 16:04.17 | ManxPower | There's nothing wrong with the PHONES, only with everything else. |
| 16:04.24 | pHnz | oki. |
| 16:04.31 | ManxPower | pHnz: I believe the G versions of the phones support IEEE PoE |
| 16:04.48 | pHnz | so thx i think i'll buy 10 polycom SoundPoint 430 |
| 16:05.04 | ManxPower | pHnz: no, but ONE and see if the phone meets your needs. |
| 16:05.08 | nny | real question is, whats the fun factor in using cisco client phones with asterisk |
| 16:05.18 | nny | ha we think this RFP is a response to the VA shootings |
| 16:05.23 | ManxPower | we tested 6 SIP phones before we decided on Polycom. |
| 16:05.34 | jcanfield | I know I'm not supposed to ask * now questions in here, and while I have found most of the GUI useless, i really like the fact that I have a working asterisk appliance quickly. Can someone recommend a good way to add the LDAPget module? Considering it's an rpath install, I was hoping there was a conary repository out there. |
| 16:05.43 | nny | ManxPower: you're saying phones good, but rest of cisco is shite? |
| 16:05.58 | ManxPower | nny: yes. |
| 16:06.03 | nny | k |
| 16:06.10 | ManxPower | well the Cisco POLICIES at least. |
| 16:06.12 | pHnz | wich model ManxPower of your Polycom |
| 16:07.01 | ManxPower | pHnz: We have 300, 500, 301, 501, 602, and 430s |
| 16:07.10 | ManxPower | ..er. 601, not 602. |
| 16:07.33 | nny | ha yeah this RFP is a response to the VA shootings, they just released a study that said a system wide response mechanism with a way to broadcast alerts what have helped... now we have an RFP to add that functionality to a Cisco system. Mwwahaha go propreitary go! |
| 16:07.40 | hypa7ia | jcanfield: have you tried asking that in #asterisk-gui? |
| 16:07.41 | pHnz | kewl thx =) |
| 16:08.00 | pHnz | easy to merge with asterisk ? |
| 16:08.10 | hypa7ia | lol |
| 16:08.16 | ManxPower | pHnz: all ip phones are hard to learn to configure. |
| 16:08.26 | Trionnis | http://pastebin.ca/700828 <-- any idea why that config would keep throwing a 407 at vg2 when it tries to send a call? |
| 16:08.28 | ManxPower | there |
| 16:08.43 | pHnz | that's true. =) |
| 16:08.49 | pHnz | So thank you very much for your times. |
| 16:08.53 | pHnz | Have a nice day. |
| 16:09.47 | ManxPower | Trionnis: No idea. On my systems if a call comes in without auth info asterisk 1.2 takes the stuff from [general] and accepts the call, |
| 16:10.16 | yang | w |
| 16:10.26 | Uatec | hi |
| 16:10.29 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-254.lsanca.dsl-w.verizon.net) |
| 16:10.31 | Uatec | i have a queue, and agents logging in to it |
| 16:10.37 | Uatec | and the enter their DDI so that it.... |
| 16:10.39 | Uatec | yes |
| 16:10.46 | Uatec | they enter their DDI so that it directs the call to them |
| 16:10.54 | Uatec | how can i specify what context it tries to dial the DDI in? |
| 16:11.03 | Uatec | s/DDI/extensions |
| 16:11.05 | ManxPower | hypa7ia: there is no excuse for using AsteriskNow and being on this channel. |
| 16:11.42 | yang | on linuxvoipbuster.com edit entered in sip.conf, but i am unable to make outbound calls over it |
| 16:12.20 | McDouglas | can i see what codec my phone is using? |
| 16:12.41 | ManxPower | McDouglas: "sip show channels" during an active call. |
| 16:12.51 | ManxPower | You can also look at the info in the SIP packets |
| 16:13.00 | nny | anyone have a sure fire method for cleaning out an old asterisk install and xaptel? |
| 16:13.04 | nny | zaptel* |
| 16:13.14 | hypa7ia | ManxPower: at least that user was polite about it :/ |
| 16:13.31 | ManxPower | nny: install over the old stuff. |
| 16:14.06 | ManxPower | hypa7ia: we dump the rude ones into the canal as aligator food. |
| 16:15.10 | hypa7ia | ManxPower: an excellent approach |
| 16:16.23 | cpm | that's the best approach /me thinks |
| 16:16.51 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:17.30 | Echinos | and throw in some telemarketers for dessert |
| 16:22.38 | [TK]D-Fender | Echinos: Ew... you don't want to POISON them now do you? |
| 16:22.46 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 16:22.51 | Sci_05 | morning all |
| 16:22.58 | Sci_05 | errr I mean afternoon |
| 16:30.21 | hi365 | what comands can i exececute using #exec ? |
| 16:30.34 | Sci_05 | I am having a bit of a strange problem, when my server goes to call out h323 I get the following debug error "rtp.c: RTP Transmission error of packet 26997 to IP.ADDRESS:12304: Invalid argument" Got any ideas where I can start to look for that? |
| 16:31.00 | hi365 | can it be any system command? do i need to enlose the command in quotes or something? |
| 16:34.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:36.18 | *** join/#asterisk juxhi (n=juxhi@nathan.epi.usf.edu) |
| 16:37.37 | *** join/#asterisk seldon75 (n=chatzill@69.77.161.2) |
| 16:39.09 | ManxPower | hi365: what does the sample config files say? |
| 16:39.30 | seldon75 | hello,we have Asterisk 1.5 running through a TDM2400E to 2x IP trunks and 4xCopper trunks |
| 16:39.44 | ManxPower | seldon75: there is no such thing as Asterisk 1.5 |
| 16:39.50 | seldon75 | oh |
| 16:39.52 | seldon75 | 1.4 |
| 16:39.56 | seldon75 | recent |
| 16:39.58 | ManxPower | are you sure about that? |
| 16:40.05 | seldon75 | like a month ago build |
| 16:40.09 | ManxPower | perhaps a "show version" would be helpful. |
| 16:40.34 | ManxPower | seldon75: that tells us nothing. |
| 16:40.49 | seldon75 | dont you know what the official build a month ago was? |
| 16:40.49 | ManxPower | paste the output of your "show version" from the Asterisk CLI. |
| 16:40.53 | seldon75 | how often does it change? |
| 16:40.55 | seldon75 | ok |
| 16:40.57 | seldon75 | one sec |
| 16:41.35 | ManxPower | seldon75: RELEASED versions are released and then they are done. SVN (which does not have versions like that) and changes 10 - 100 times per day. |
| 16:41.56 | seldon75 | ok it's 1.4.5 |
| 16:41.59 | seldon75 | sorry |
| 16:42.19 | seldon75 | anyway, occasionally, say every few days our box gets into a state where when we try to make an O/G call, we just get loud static hissing |
| 16:42.27 | ManxPower | perhaps you should consider upgrading to 1.4.11? |
| 16:42.30 | seldon75 | and I have to restart asterisk for it to go back to normal |
| 16:42.46 | ManxPower | you should try upgrading to 1.4.11 |
| 16:42.59 | Corydon76-lap | seldon75, what card are you using? |
| 16:43.07 | seldon75 | Digium TDM2400E |
| 16:43.15 | seldon75 | it has been replaced (including teh modules) |
| 16:43.26 | seldon75 | we thought it was a hardware thing |
| 16:43.32 | seldon75 | but all the hardware has been replaced |
| 16:43.35 | seldon75 | and the problem remains |
| 16:43.41 | ManxPower | I didn't know Digium had a "E" model, only the "P" model. Do you know what the difference between the E and the P models are? |
| 16:44.00 | seldon75 | the number of FXO/FXS quad modules |
| 16:44.09 | seldon75 | oh |
| 16:44.13 | seldon75 | no it's echo cancelling |
| 16:44.25 | seldon75 | E means hardware echo cancelling |
| 16:44.46 | Corydon76-lap | seldon75, reload your wctdm24xxp driver with the kernel module option nativebridge=0 |
| 16:45.12 | seldon75 | ah could you tell me where to find the commands to do that> |
| 16:45.33 | Corydon76-lap | rmmod wctdm24xxp ; modprobe wctdm24xxp nativebridge=0 |
| 16:45.45 | seldon75 | cool, ill try that |
| 16:45.50 | seldon75 | thx |
| 16:46.51 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:48.07 | hi365 | ManxPower: which config file? |
| 16:48.41 | *** join/#asterisk jtexter3 (n=jtexter3@12.159.220.114) |
| 16:51.03 | jtexter3 | Has anyone experienced an issue with 1.4.10 and Polycom 501's (remote worker over VPN), where when they dial, the ringback sounds really choppy, but call audio quality is fine? |
| 16:51.28 | ManxPower | hi365: any of them that use #exec, I think most of them can. The sample configs are in /path/to/src/asterisk/configs |
| 16:52.01 | ManxPower | jtexter3: that should never happen unless you are doing somethng stupid like Answer or Background before the Dial |
| 16:52.18 | ManxPower | the ringback is supposed to be generated locally by the SIP phone. |
| 16:52.47 | ManxPower | unless the line has been answered first. |
| 16:53.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:53.06 | jtexter3 | ManxPower: I need to check this, but what if progressinband=yes in sip.conf? |
| 16:53.14 | ManxPower | you should never do that either |
| 16:53.27 | ManxPower | at least with polycoms. |
| 16:53.31 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-90d89fd095cd4451) |
| 16:53.31 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:53.55 | jtexter3 | ManxPower: I think I had to do that when testing with an Audiocodes gateway, otherwise you never hear ringback. I was hoping to get confirmation as to which side provided the ringback, so I appreciate your help |
| 16:54.09 | hi365 | ManxPower: i dont think i installed the sample config's. this file doesnt show any #exec usage :( |
| 16:54.10 | hi365 | http://svn.digium.com/view/asterisk/branches/1.4/configs/extensions.conf.sample?revision=80047&view=markup |
| 16:54.34 | bkruse | was #exec introduced only in trunk ManxPower? or has it always been |
| 16:54.41 | ManxPower | hi365: the sample configs won't install unless you run "make samples" in that case it will overwrite your configs |
| 16:54.54 | ManxPower | bkruse: I thought it was pre-1.2 |
| 16:55.04 | bkruse | it could have been |
| 16:55.10 | bkruse | im thinking of the following #include |
| 16:55.14 | bkruse | my apologies |
| 16:55.19 | hi365 | isnt the sample config for extensiosn the same as the link i posted? |
| 16:56.01 | ManxPower | apparently an example of #exec usage is not in 1.2, but in the 1.2 CHANGES it mentions adding support for it. |
| 16:56.56 | hi365 | "added #exec syntax for including output from a command" |
| 16:56.59 | ManxPower | hi365: I doubt it as I use 1.2 |
| 16:57.40 | hi365 | i think my question is still not addresed: can it be any system command? do i need to enlose the command in quotes or something if it has switches? |
| 16:57.55 | hi365 | (well i guess it can be any system command) |
| 16:58.19 | ManxPower | hi365: #exec takes the output of the command an inserts it into the config file at the point where #exec is located. It does NOT care what command you are using. |
| 16:58.26 | ManxPower | Why don't you TRY it and see. |
| 16:58.31 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-254.lsanca.dsl-w.verizon.net) |
| 16:58.53 | ManxPower | since it is an undocumented option, your best option is play with it. |
| 16:59.29 | hi365 | i did. if i use #exec /myscripthere it runs fine. but if i insert a system command it apperently in SOT executed |
| 17:00.02 | ManxPower | and that command would be...... |
| 17:00.12 | hi365 | lynx -dump -nolist http://docs.google.com/Doc?id=dhc7h3tr_17cg2vxn| sed '1,6d' | sed -e :a -e '$d;N;2,4ba' -e 'P;D' | sed 's/^[ \t]*//' |
| 17:00.59 | ManxPower | well I would expect you would have to put that in double quotes |
| 17:01.16 | hi365 | doesnt work: |
| 17:01.16 | hi365 | #exec "lynx -dump -nolist http://docs.google.com/Doc?id=dhc7h3tr_17cg2vxn| sed '1,6d' | sed -e :a -e '$d;N;2,4ba' -e 'P;D' | sed 's/^[ \t]*//'" |
| 17:02.19 | *** part/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com) |
| 17:02.23 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-254.lsanca.dsl-w.verizon.net) |
| 17:04.18 | Aeudian | Instead of database lookups referencing a parsing file or sql server, how would dundi compare? It seems to be a good direction for multiple asterisk servers but i just worry about system impact and performance |
| 17:04.21 | ManxPower | that command spits out [blankdot.gif] at the end, which asterisk will see as a context and screw everything up. |
| 17:04.26 | *** join/#asterisk pjz (n=pj@66.219.59.183) |
| 17:04.34 | ManxPower | hi365: does it work if you put that command in a shell script. |
| 17:04.46 | ManxPower | Also remember that in Asterisk , is converted to a | |
| 17:04.46 | pjz | is it possible to use a skype phone as an asterisk extension? |
| 17:04.59 | ManxPower | pjz: yes and no. Google is your friend. |
| 17:05.02 | ManxPower | ~mailinglist |
| 17:05.03 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 17:05.36 | hi365 | hmm, actualy the sed is supposed to remove everything other than the asterisk contaxt |
| 17:06.05 | ManxPower | not when I ran it |
| 17:06.23 | nny | anyone here have a favorite product for protecting phone lines from power spikes etc? |
| 17:06.41 | ThoMe | hello |
| 17:06.54 | [TK]D-Fender | nny: Yeah, a PoE switch on a UPS |
| 17:07.19 | ThoMe | i search a isdn-proxy.. for dial from a windows pc (like capi?) about asterisk? |
| 17:07.22 | [TK]D-Fender | nny: And power-bars with surge supressors |
| 17:07.25 | ThoMe | is it posible? |
| 17:07.25 | ManxPower | sandman.com is always a good place to look. |
| 17:07.49 | ThoMe | ManxPower: me? |
| 17:07.55 | hi365 | ManxPower: can you post the output? here is what i get: http://pastebin.ca/700885 |
| 17:08.39 | hi365 | everything other than the asterisk context is filtered out |
| 17:08.47 | nny | [TK]D-Fender: we have a power surge protector properly grounded for the pbx, this is specifcally for the lines from the telco |
| 17:08.57 | ManxPower | ThoMe: I have no idea what you said, so I ignored ot. |
| 17:09.15 | nny | [TK]D-Fender: by power surge protector I mean an APC with voltage regulation |
| 17:09.26 | ManxPower | hi365: I could, but my mouse is screwing up and I can't copy and paste |
| 17:09.29 | nny | just wondering if there are any standalone boxes for protecting lines |
| 17:09.48 | hi365 | ctrl+c, ctrl+v if you on windows |
| 17:10.04 | hi365 | just tellll me this: is it different than my output? |
| 17:10.54 | hmmhesays | bah what is this crap 1 packet/second per each 1000 bits/second |
| 17:11.06 | hmmhesays | I don't even know how to measure that |
| 17:12.23 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 17:14.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:15.03 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 17:15.12 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:17.54 | *** join/#asterisk saftsack (n=saftsack@pD9E05C35.dip.t-dialin.net) |
| 17:18.41 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 17:21.35 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 17:22.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:25.20 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:32.59 | *** join/#asterisk dislikework (n=davidh@196.211.34.2) |
| 17:33.09 | dislikework | hi there |
| 17:33.24 | *** part/#asterisk dislikework (n=davidh@196.211.34.2) |
| 17:34.52 | *** join/#asterisk ManxPower (n=manxpowe@98.sub-70-221-75.myvzw.com) |
| 17:34.58 | *** part/#asterisk verto (n=ercardos@201.57.47.2) |
| 17:35.17 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:35.31 | ManxPower | that might be better |
| 17:36.24 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:37.57 | hi365 | ManxPower: im wondering if asterisk has a problem hannnndaling the commands after the file to exec |
| 17:38.18 | hi365 | do you understand c? here is the source for exec: http://bugs.digium.com/file_download.php?file_id=4645&type=bug |
| 17:38.26 | *** join/#asterisk Tall-guy (n=noway@207-195-103-110.regn.static.sasknet.sk.ca) |
| 17:38.53 | Tall-guy | Hello channel, long time no see (been years I think). |
| 17:38.57 | *** join/#asterisk pzn (i=foobar@201-13-41-24.dsl.telesp.net.br) |
| 17:40.10 | pzn | Hi. I'm new to asterisk. I'm developing a hardware for voip and I need to develop the driver for it to work in asterisk. can you point me to some information about how to develop a driver for asterisk? |
| 17:45.24 | tzafrir | pzn, does it connect to Asterisk through VoIP? What protocol? SIP? |
| 17:46.14 | tzafrir | ~seen Tall-guy |
| 17:46.17 | jbot | tall-guy is currently on #asterisk (7m 51s). Has said a total of 1 messages. Is idling for 7m 24s, last said: 'Hello channel, long time no see (been years I think).'. |
| 17:46.30 | tzafrir | too late to ask now :-( |
| 17:46.54 | Tall-guy | :) |
| 17:47.04 | Tall-guy | yeah, you'll just have to take my word for it I guess :) |
| 17:47.24 | Tall-guy | ~tallguy2 |
| 17:47.30 | Tall-guy | ~seen tallguy2 |
| 17:47.31 | jbot | i haven't seen 'tallguy2', Tall-guy |
| 17:47.40 | Tall-guy | hmm, never here under that moniker...worth a shot |
| 17:48.37 | nny | grrr.. why would would a digium card just hang up on answer? |
| 17:48.47 | nny | call, rings once, dies |
| 17:48.58 | Strom_M | nny: what does the console say? |
| 17:49.19 | [TK]D-Fender | nny: I've heard of countries where on answer they do a line cut which may be interpreted as CDS |
| 17:49.28 | [TK]D-Fender | nny: Where are you located? |
| 17:49.33 | nny | US |
| 17:49.57 | [TK]D-Fender | nny: ok, scratch taht basically.. |
| 17:50.05 | *** join/#asterisk semur (n=semushin@83.167.96.120) |
| 17:50.13 | Strom_M | nny: pastebin some asterisk console output |
| 17:50.41 | nny | ok |
| 17:53.14 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 17:55.28 | magic_hat | hi all... i have a weird situatiion hoping you can help with. My * box is running fine with a bunch of Xlites running on Mac, Win, and Linux all connected wirelessly. One of my Win machines works only when connected via ethernet, though. Any ideas on why that might be and how I can get it working wirelessly? |
| 17:55.55 | nny | Strom_M: I have core verbosity set to 5, should I set debug level as well? |
| 17:56.27 | Tall-guy | magic_hat: windows firewall enabled on wireless? |
| 17:56.28 | nny | ? |
| 17:57.21 | nny | Strom_M: I get hung up on, but the system tries to play vm etc |
| 17:58.00 | nny | http://pastebin.com/m36d4933 |
| 17:58.12 | nny | zap status says ok, no alarms |
| 17:58.18 | nny | dmesg shows two modules loaded |
| 17:59.04 | *** join/#asterisk Servergod (n=nunya@70.97.159.120) |
| 17:59.06 | Tall-guy | magic:hat???? |
| 17:59.10 | magic_hat | Tall-guy: just tried taking the firewall down, didn't make a difference. |
| 17:59.49 | Tall-guy | magic_hat |
| 17:59.58 | Tall-guy | magic_hat: what are ip address of LAN and wireless set to on this box? |
| 18:00.12 | Servergod | to be able to have users have to auth to be able to point to our boxes, we need a registrar server right? |
| 18:00.20 | magic_hat | the router is 192.168.1.1 |
| 18:00.30 | magic_hat | asterisk is 192.168.1.103 |
| 18:01.08 | Servergod | er, let me rephrase that |
| 18:01.09 | Tall-guy | magic: windows box ips? |
| 18:01.52 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 18:02.19 | magic_hat | Tall-guy: not sure. |
| 18:02.26 | Servergod | i have two gateways, one pointed to a carrier (360networks) and another towards the PSTN (4 pri's) |
| 18:02.53 | Servergod | i also have three asterisk boxes that point towards those, with dial plans for ld and local |
| 18:03.24 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:03.25 | Tall-guy | magic_hat: theory1: windows box is mixfuzed as to which nic to send data to/from...change the desktop LAN ip to something else (like 10.10.10.10) and/or disable it... |
| 18:03.28 | Servergod | to have other people trunk their asterisk to us, for security, we need a registrar server? |
| 18:03.59 | Tall-guy | magic_hat: theory2: windows firewall, or norton firewall, zonealarm, or some other 3rd party firewall may be blocking access out your Wireless NIC. |
| 18:04.46 | sniper[FOO] | hi all |
| 18:04.57 | sniper[FOO] | prepared for a lame question? |
| 18:05.21 | Tall-guy | magic_hat: theory3: your SIP software is reaching the asterisk box wirelessly (ie: registering), but sending asterisk the IP for the LAN nic.... |
| 18:05.28 | nny | Strom_M: any thoughts? |
| 18:05.55 | sniper[FOO] | StripMSD disappeared from 1.4, so how could I remove a prefix from a dialed extension? |
| 18:06.37 | *** join/#asterisk point (n=point@213.27.44.55) |
| 18:06.53 | jwh | something like {EXTEN:x} or -x? |
| 18:06.54 | ai-a | sniper[FOO]: RTFM - http://www.voip-info.org/wiki-Asterisk+variables |
| 18:08.07 | brodiem | Can anyone here report if they've had good/bad results using a router w/ built-in SPI firewall for SIP? |
| 18:08.36 | Sci_05 | brodiem: I have used the linksys ones with little problems |
| 18:08.49 | magic_hat | Tall-guy: okay, i'll check it out. thanks. |
| 18:08.58 | Tall-guy | magic_hat: yup. |
| 18:09.15 | ai-a | or.. he might not be meaning to strip from a variable :) |
| 18:09.19 | Servergod | Sniper use the | operator to cut. faxes for our t38 use 6 as an out bound dial. so the dial plan is 6., the trunk is set for 6|213+NXXXXXX we need to send 10 digits out so that is the + 213 |
| 18:09.19 | brodiem | Sci_05: so you did have problems using it or didn't? :) |
| 18:09.43 | Sci_05 | no I had no problems |
| 18:09.44 | sniper[FOO] | Servergod, thanks |
| 18:09.49 | Servergod | np |
| 18:10.09 | nny | so I can get out an outward call, but when I call the system asterisk hangs up.. I didn't change anything in the dial plan -_- |
| 18:10.18 | brodiem | Sci_05: do you know what model it was? I've got a LS wrv200 here that isn't behaving right |
| 18:10.38 | [TK]D-Fender | one should NOT use CUT to strip a fixed length prefix off a variable..... thats ridiculous. |
| 18:11.19 | Servergod | sip_nat.conf extern ip=xxx.xxx.xxx.xxx local net=xxx.xxx.xxx.xxx/xx nat=yes |
| 18:11.39 | Servergod | you have that set up nny? |
| 18:11.45 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:11.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:11.49 | *** part/#asterisk point (n=point@213.27.44.55) |
| 18:12.55 | Sci_05 | brodiem: I used the rtr and wls-g |
| 18:13.18 | Sci_05 | rtr300 wrtp54g |
| 18:14.36 | brodiem | Sci_05: ahh those have fxs ports right? Were you only using analog phones plugged into it or also using IP phones? |
| 18:14.43 | *** join/#asterisk ManxPower (n=manxpowe@155.sub-75-201-181.myvzw.com) |
| 18:15.05 | Sci_05 | nope thoes have 2 fxs ports on them (line1, line2) that you can setup to connect to any server |
| 18:15.24 | nny | Servergod:not using any NAT atm |
| 18:15.36 | magic_hat | Tall-guy: one thing -- i'm not getting any traffic on the asterisk CLI from that win box... does that narrow it down? |
| 18:15.38 | nny | Servergod: our dev says hardware related |
| 18:15.43 | brodiem | Sci_05: ok that would make sense that you didn't have probs with SPI if you were using the built-in FXS ports |
| 18:15.48 | nny | Servergod: but would a failing card allow calls out but not in??? |
| 18:16.55 | Tall-guy | magic_hat: can you ping it? |
| 18:17.45 | Servergod | nny, do a cat /proc/interupt and see if any irq are the same across multiple devices |
| 18:17.55 | ManxPower | nny: what is the specific issue? |
| 18:18.43 | nny | 193: 2425506 IO-APIC-level wctdm |
| 18:18.44 | nny | no dupes |
| 18:18.56 | nny | ManxPower: I can call out, but calling in just hangs up on caller |
| 18:19.04 | nny | ManxPower: reaaal basic dialplan |
| 18:19.14 | ManxPower | nny: and sip debug of a failed inbound call is located on what pastebin? |
| 18:19.47 | ManxPower | I assume you don't get any CLI info on inbound/failed calls. |
| 18:20.02 | nny | no |
| 18:20.15 | ManxPower | I'm still waiting for the pastebin info |
| 18:20.59 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:22.13 | ManxPower | nny: So a call comes in from the outside (using what technology?) and is supposed to hit what exten line in which context in Asterisk? |
| 18:22.23 | ManxPower | then the call is supposed to be sent where? |
| 18:23.09 | ManxPower | If you are not going to respond I'll trying to help. |
| 18:23.36 | nny | ManxPower: http://pastebin.com/m36d4933 and the incoming context is http://pastebin.com/m36d4933 |
| 18:23.40 | nny | mah |
| 18:23.44 | nny | second link is wrong |
| 18:23.49 | nny | exten => s,1,Goto(people,100,1) |
| 18:23.53 | nny | freaking c/p from term |
| 18:24.14 | nny | ManxPower: sorry fielding calls here as well. |
| 18:24.28 | ManxPower | that is not a real simply dialplan. |
| 18:24.44 | Tall-guy | :) |
| 18:24.59 | ManxPower | I see macros, voicemail, etc. That is NOT a simple dialplan |
| 18:25.03 | nny | well the funny thing is, when I called that time, it hung up on me, but ran through the normal steps |
| 18:25.11 | nny | well yeah, but it is supposed to just ring to exten 100 |
| 18:25.34 | ManxPower | heck, it looks from that pastebin that it all worked. |
| 18:25.37 | nny | but it shites and heads too VM (as designed) but the caller gets hung up on after first ring, before * even picks up |
| 18:25.40 | nny | hehe yeah |
| 18:25.59 | nny | so would a faulty card cause the incoming call to get dropped, but outgoing to work normally? |
| 18:26.05 | ManxPower | nny: you are not doing something STUPID like using callprogress=yes or busydetect=yes, are you? |
| 18:26.22 | nny | let me check, i dont believe so |
| 18:26.25 | brodiem | lol |
| 18:26.31 | ManxPower | nny: I see no indication that is it a bad card. |
| 18:26.32 | bkruse | ManxPower: oh so true. |
| 18:26.36 | Tall-guy | manx: I take it you don't like those options? :) |
| 18:26.45 | ManxPower | Tall-guy: they randomly disconnect calls. |
| 18:26.51 | Tall-guy | manx: sounds like my receptionist :) |
| 18:26.52 | ManxPower | so, no, I do not like those options. |
| 18:27.00 | nny | ManxPower: what conf files would those be in? zaptel or zapata |
| 18:27.01 | nny | ? |
| 18:27.07 | ManxPower | /etc/asterisk/zapata.conf |
| 18:27.33 | nny | k looking |
| 18:27.37 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:29.32 | ManxPower | take out the damn macro, it makes it virtually impossible to read |
| 18:29.41 | ManxPower | s,1,Dial(SIP/100) or whatever. |
| 18:29.46 | Tall-guy | heheh, man I can't read macros either... |
| 18:29.54 | bkruse | its quite easy, but ok :] |
| 18:30.12 | bkruse | Writing the macro yourself also helps.... |
| 18:30.14 | ManxPower | bkruse: I would be happy to read the macros, but not for free. |
| 18:30.15 | nny | I have: echotraining=800 rxgain=6.0 txgain=0.0 immediate=no relaxdtmf=yes echocancelwhenbridged=no echocancel=yes but callprogress is commented out as well as busydetect |
| 18:30.21 | bkruse | ManxPower: haha, so true :] |
| 18:30.34 | ManxPower | nny: turn off relaxdtmf but this is not your current problem. |
| 18:30.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:30.59 | ManxPower | nny: now, do a test call without macros and without other calls happening on the system. |
| 18:31.30 | nny | so just let it ring to exten 100? |
| 18:31.36 | nny | or just Answer() |
| 18:31.37 | bkruse | well, whatever exten it is.... |
| 18:31.48 | ManxPower | no. I told you what to put, that did not have an answer on it. |
| 18:32.17 | ManxPower | (1:29:25 PM) ManxPower: s,1,Dial(SIP/100) or whatever. |
| 18:32.19 | nny | ManxPower: sorry didn't see what you posted, |
| 18:33.28 | ManxPower | well, we are out of time today, I'll be around tomorrow |
| 18:34.58 | jsidhu2 | whats the default root password int he 2.3.0.2 beta (Second Beta for 2.4) of trixbox? |
| 18:35.03 | jsidhu2 | err.. wrong channel |
| 18:35.07 | bkruse | jsidhu2: got that right |
| 18:35.09 | bkruse | lol |
| 18:35.11 | nny | ok so um |
| 18:35.20 | jsidhu2 | heeh |
| 18:35.27 | nny | did what he suggested and it still hangs up |
| 18:36.50 | nny | no output nothing |
| 18:37.03 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:37.04 | bkruse | core set verbose 10 |
| 18:37.07 | bkruse | core set debug 10 |
| 18:37.11 | bkruse | I tried to tab complete debug :/ |
| 18:37.24 | nny | meh WTF |
| 18:37.27 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 18:37.56 | nny | Verbosity was 0 and is now 10 |
| 18:37.57 | nny | <PROTECTED> |
| 18:37.57 | nny | [Sep 17 14:37:18] WARNING[5020]: chan_zap.c:6496 ss_thread: CallerID returned with error on channel 'Zap/1-1' |
| 18:37.57 | nny | <PROTECTED> |
| 18:37.57 | nny | <PROTECTED> |
| 18:37.58 | nny | <PROTECTED> |
| 18:38.00 | nny | <PROTECTED> |
| 18:38.02 | nny | <PROTECTED> |
| 18:38.04 | nny | shit sorry |
| 18:38.06 | nny | damn cp |
| 18:38.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:38.41 | nny | but it hangs up.. meh |
| 18:38.50 | nny | will have to try new card |
| 18:39.44 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:40.22 | [TK]D-Fender | nny: thats you calling in? |
| 18:46.45 | nny | [TK]D-Fender: heh calling in who? |
| 18:46.59 | nny | [TK]D-Fender: you work with e4? |
| 18:47.28 | [TK]D-Fender | nny: What is placing that call to Zap/1? |
| 18:47.34 | nny | oh lol |
| 18:47.37 | nny | [TK]D-Fender: me |
| 18:47.54 | nny | [TK]D-Fender: it hangs up right away, but * rolls through the dialplan |
| 18:48.01 | [TK]D-Fender | nny: Gee thanks for the glorious detail :p |
| 18:48.16 | nny | [TK]D-Fender: lol what info do you need? |
| 18:48.28 | [TK]D-Fender | nny: Right away? As in you don't hear * starting ringing? |
| 18:48.39 | [TK]D-Fender | nny: Instant dead-silnece? |
| 18:48.56 | nny | I get 1 ring, which I always get prior to * hitting the line, but than the call disconnects |
| 18:49.09 | nny | so 1 ring.. silence, call disconnected |
| 18:50.36 | nny | I was gonna head over with our card we use here and swap em out.. but was wondering if the issue would present itself in console or somehow |
| 18:50.59 | pzn | tzafrir: sorry... I went to have lunch. my hardware is a sound hardware (but not a sound card). I need to develop a driver for asterisk, similar to the driver that asterisk use to access the /dev/dsp. |
| 18:51.39 | pzn | tzafrir: it does not understand voip/sip/... it is a sound device via a serial port. |
| 18:52.42 | *** join/#asterisk DaveCanoe (n=Dave@vgateway.libertyrms.info) |
| 18:52.49 | nny | [TK]D-Fender: anything else that would be available for diagnostics via terminal or should I just try the hardware swap? |
| 18:53.14 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:53.17 | [TK]D-Fender | nny: If you plug a normal analog phone onto the line, does the same thing happen? |
| 18:53.23 | nny | [TK]D-Fender: no it works |
| 18:53.39 | [TK]D-Fender | nny: have you tried a different port on your card? |
| 18:53.51 | nny | [TK]D-Fender: yes I have tried swapping the modules as well as the lines |
| 18:54.02 | nny | (2 FXO modules) |
| 18:54.11 | [TK]D-Fender | nny: ok, pastebin your zapata.conf |
| 18:54.17 | nny | [TK]D-Fender: k |
| 18:54.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-177-39.red.bezeqint.net) |
| 18:54.36 | bkruse | ManxPower leaves and [TK]D-Fender fender takes over :] |
| 18:54.44 | bkruse | [TK]D-Fender: you have helped me numerous times, ++ to you |
| 18:55.04 | [TK]D-Fender | bkruse: Scary considering you're employed for this stuff :) |
| 18:55.30 | [TK]D-Fender | bkruse: Maybe I should apply for a remote Community Relations position... |
| 18:55.38 | bkruse | [TK]D-Fender: exactly! |
| 18:55.53 | bkruse | I end up asking general questions for random config options before I look them up, and you can rattle off answers |
| 18:55.59 | bkruse | and mostly just watching you help people, its awesome :] |
| 18:56.01 | [TK]D-Fender | bkruse: I did get to mee Jared Smith this weekend on his bootcamp trip |
| 18:56.10 | bkruse | [TK]D-Fender: ohrly? hes a very cool guy |
| 18:56.17 | bkruse | helped me a lot with the gui developers documentation |
| 18:56.23 | nny | http://pastebin.com/m1ebfa619 |
| 18:56.30 | [TK]D-Fender | bkruse: Yup, got to visit with JunK-Y and file here as well. |
| 18:56.34 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 18:57.14 | [TK]D-Fender | nny: permanently remove all commented lines and repastebin. the clutter isn't doing anyone any good... |
| 18:57.21 | file | we had chinse, it was good |
| 18:57.29 | file | and chinese |
| 18:57.30 | nny | ok |
| 18:58.34 | [TK]D-Fender | file: And to think.... that was the tame stuff. Wouldn't want to send you into culinery shock ;) |
| 18:58.54 | file | or.... would you |
| 18:59.22 | bkruse | file: he would |
| 19:00.10 | nny | [TK]D-Fender: http://pastebin.com/m258e3fbb |
| 19:00.41 | [TK]D-Fender | file: You've got to expand your palatte though.... I mean, no sea-food? How gauche is that? Spicy=good! |
| 19:01.06 | [TK]D-Fender | nny: I'll spare you getting chewed out for excessive skipped lines... FOR NOW :p |
| 19:01.48 | bkruse | nny: cat blah.conf | grep -v ';' > blah.conf.bak; mv blah.conf.bak blah.conf |
| 19:02.02 | [TK]D-Fender | nny: Try switching from fxs_ks to fxs_ls (and don't forget to adjust zaptel.conf as well. |
| 19:02.29 | nny | [TK]D-Fender: :) |
| 19:02.35 | *** part/#asterisk pjz (n=pj@66.219.59.183) |
| 19:02.44 | nny | bkruse: lol yeah that would have worked as well, my script fu ist weak |
| 19:03.11 | bkruse | nny: itll get better :] |
| 19:03.41 | [TK]D-Fender | there is no cat....... |
| 19:03.55 | [TK]D-Fender | bkruse: Qwell would pwn you for the wasted pipe ;) |
| 19:04.01 | *** join/#asterisk bdheeman (n=bsd@122.162.3.30) |
| 19:04.09 | bkruse | [TK]D-Fender: ahh good point! |
| 19:04.10 | Qwell | ~uuoc |
| 19:04.11 | jbot | i guess uuoc is Useless Use of Cat Award. Given out for years by Randal Schwartz on the newsgroup comp.unix.shell. Basically, most constructions that look like "cat filename | grep pattern" can be more easily written as "grep pattern filename". Works for grep and most other Unix utilities. Easier to type and marginally more efficient. |
| 19:04.23 | Qwell | :D |
| 19:04.26 | bkruse | I think I showed Qwell the grep -v, then it was b0rked for awhile, then it came back : |
| 19:04.28 | [TK]D-Fender | bkruse: ...... pwned :D |
| 19:04.31 | bkruse | totally |
| 19:04.38 | bkruse | I do get a lot of script foo from Qwell, however |
| 19:04.44 | Qwell | heh, actually... |
| 19:04.50 | nny | [TK]D-Fender: so in zaptel.conf (as well as what I changed in zapata.conf) I would change fxsks=1,2 to fxsls=1,2 ? |
| 19:04.55 | Qwell | there was a problem I was running into on the appliance at one point with grep -v |
| 19:05.09 | Qwell | it would crash the unit unless you used cat.. |
| 19:05.10 | *** part/#asterisk bdheeman (n=bsd@122.162.3.30) |
| 19:05.20 | bkruse | Qwell: thats what I was talking about, when it b0rked |
| 19:05.23 | Qwell | :D |
| 19:05.29 | Qwell | that was seriously unfun |
| 19:05.33 | bkruse | I am not sure, its grep -r that is not included now |
| 19:05.39 | bkruse | Qwell: I remember that, and agreed. |
| 19:06.00 | nny | that makes me feel better.. I always wondered why people would say cat filename |grep text.. I would just use grep text filename and thought I was doing it ass backward |
| 19:06.10 | [TK]D-Fender | nny: Correct |
| 19:06.16 | nny | [TK]D-Fender: ok ty |
| 19:07.01 | nny | ok reloaded zaptel etc. |
| 19:07.09 | nny | i shall test |
| 19:07.25 | nny | pfft hung up on me |
| 19:08.41 | nny | but * keeps rolling along like the line is kept open |
| 19:08.45 | [TK]D-Fender | nny: You COULD simply have a defective card.... |
| 19:08.54 | nny | [TK]D-Fender: lol yeah thats the next step |
| 19:09.25 | [TK]D-Fender | nny: last shot : PB up dmesg |
| 19:09.28 | nny | [TK]D-Fender: heading over now to swap cards.. looks like I have a new card to order if thats the case |
| 19:09.31 | nny | [TK]D-Fender: ok |
| 19:10.03 | sniper[FOO] | hey all |
| 19:10.17 | sniper[FOO] | lame question again |
| 19:10.38 | [TK]D-Fender | 3 cheers for truth in advertising.... |
| 19:10.55 | sniper[FOO] | which app is going to return 503 to the outbound caller? |
| 19:12.02 | nny | http://pastebin.com/m7411938d |
| 19:14.15 | [TK]D-Fender | nny: Final silly question : these are the RED modules, correct? |
| 19:14.23 | nny | [TK]D-Fender: lol yeah |
| 19:14.37 | [TK]D-Fender | nny: Ok, well we've just run the full gamut from my POV. |
| 19:14.54 | nny | [TK]D-Fender: gotcha, I am gonna take a card over, swap, test, and RMA if needed |
| 19:15.54 | nny | [TK]D-Fender: thanks for the help, appreciate the time you took to look into it |
| 19:17.58 | sniper[FOO] | oh yes, congestion |
| 19:18.00 | sniper[FOO] | thanks |
| 19:23.09 | wishes | gmorning |
| 19:23.50 | ManxPower | and a gnome morning to you too. |
| 19:24.07 | ManxPower | or is it GNU morning? I can never remember. |
| 19:26.49 | Yourname`` | Hi, so recently I did an 'svn update' on my Asterisk 1.4, and then recompiled.. all well. But I just noticed that it's restarting itself occasionally. Not that my previous svn version was buggy at all, I just like updating things. So, question 1) How can I revert to that old version? 2) How can I do it if I don't remember what version it was? :S |
| 19:27.25 | ManxPower | Yourname``: paste one of the restarting messages. |
| 19:27.52 | ManxPower | or do you mean a real restarting and not just a "Restarting channel 1/0 on span 1" |
| 19:28.16 | Yourname`` | A real restarting, ManxPower. |
| 19:28.24 | ManxPower | Yourname``: Yourname``sounds like a seg fault to me. |
| 19:28.29 | Yourname`` | I was on CLI when it simply said "Disconnected from asterisk server".. |
| 19:28.40 | Yourname`` | ManxPower: That's what I'm thinking too. |
| 19:28.48 | ManxPower | I seem to recall a segfault with regards to res_features being talked about on -dev. |
| 19:29.39 | ManxPower | Yourname``: what branch of 1.4 did you update? |
| 19:29.43 | Yourname`` | So, is there an avn trick that can help me revert back to the old version I performed the update on? |
| 19:29.45 | ManxPower | there are like 30 of them. |
| 19:30.04 | ManxPower | Yourname``: not if you don't know the rxxxxx version number |
| 19:30.30 | ManxPower | Yourname``: you might want to try -dev, then seem to know quite a bit about svn |
| 19:30.31 | Yourname`` | ManxPower: Asterisk SVN-branch-1.4-r82444M <= this is what I'm at right now, which was before the update. |
| 19:31.18 | ManxPower | you should be able to use r82444 (I don't know what the M means) to specify the revision number. |
| 19:31.32 | ManxPower | but why not update again, as there are fixes posted to 1.4 every day. |
| 19:32.05 | Yourname`` | I could try doing that. |
| 19:32.22 | Yourname`` | Because all it is is going into /usr/src/asterisk and doing an svn update, and recompiling, lol |
| 19:32.52 | Yourname`` | Hmm, I think you're right.. res |
| 19:32.56 | ManxPower | I never run SVN, as those are developer versions and not release versions. The developer versions break all the time. |
| 19:33.01 | Yourname`` | res_features.c could have been broken. |
| 19:33.11 | Yourname`` | As I see there's an update to that and a couple others. |
| 19:35.02 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 19:35.18 | *** part/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:35.19 | southtel | Has anyone ever set up a TE110p with a channel bank? |
| 19:35.51 | [TK]D-Fender | southtel: Nope, never |
| 19:36.32 | ManxPower | southtel: yes. nothing special |
| 19:36.51 | ManxPower | have not done it in years, however. |
| 19:37.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:38.10 | Yourname`` | hah funny. |
| 19:38.13 | Yourname`` | I did the update. |
| 19:38.31 | Yourname`` | Then restart asterisk, yet I don't see the update being posted. |
| 19:38.34 | southtel | ManxPower: I think we're close... |
| 19:39.00 | southtel | But when we try to dial out, the T1 isn't seeing the outbound call... |
| 19:39.59 | southtel | We just get Zap/1-1 ringing. |
| 19:40.05 | [TK]D-Fender | southtel: pastebin is your friend..... |
| 19:40.07 | [TK]D-Fender | ~pb |
| 19:40.08 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:40.44 | ManxPower | [TK]D-Fender: My bets are on the signalling settings |
| 19:41.05 | [TK]D-Fender | ManxPower: Chips are on the board :) |
| 19:42.07 | southtel | http://pastebin.com/m5cc0a81e |
| 19:43.18 | southtel | I'm about to go crazy with this... |
| 19:43.52 | Yourname`` | I think after issuing a "restart gracefully" command, Asterisk should still let us run basic commands. |
| 19:47.13 | southtel | Any thoughs anyone? |
| 19:47.17 | rene- | Hey, where did OriginateResponse went in 1.4.x? |
| 19:47.22 | rene- | i cant find it |
| 19:55.35 | outtolunc | <PROTECTED> |
| 19:55.45 | outtolunc | same place it should be in manager.c |
| 19:57.09 | [TK]D-Fender | southtel: is the a channel bank for taking in LINES, or PHONES? |
| 19:58.44 | southtel | Lines. |
| 19:58.49 | southtel | From the provider. |
| 19:59.14 | [TK]D-Fender | southtel: signalling=fxo_ks should be signalling=fxs_ks in zapata, and fxsks in zaptel |
| 19:59.40 | [TK]D-Fender | southtel: also be sue to add "immediate=no" before your channel declaration in zapata |
| 20:01.07 | southtel | D-Fender: I thought we tried that...but I will again. |
| 20:05.20 | *** join/#asterisk Kwakwa (n=kwa@spc2-ward2-0-0-cust610.bagu.broadband.ntl.com) |
| 20:08.24 | southtel | Progress! Now the calls are ringing, but hangning up immediately... |
| 20:08.37 | southtel | Immediately upon the other user picking up, that is. |
| 20:09.01 | jsidhu2 | from the commandline, how can i check the status of a sip trunk? Like if I specify the wrong address, password etc? |
| 20:10.16 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 20:10.39 | *** join/#asterisk drbrown (n=drbrown@206.51.177.167) |
| 20:11.35 | s34n | does anybody have intercom working on a snom360? I'm struggling with the hints provided on the wiki. |
| 20:13.19 | *** join/#asterisk rrittenhouse (n=tad@unaffiliated/rrittenhouse) |
| 20:14.07 | rrittenhouse | I have a (probably stupid) question about a polycom phone if anybody can help. |
| 20:14.24 | s34n | rrittenhouse: just ask |
| 20:15.06 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 20:15.08 | rrittenhouse | Right now we have comdial impact phones and as it stands with our system if you want to pick up one of the 7 analog lines you can either pick up the phone and get one or you can hit one of the lines that are labeled 1-7 on the phone |
| 20:15.28 | rrittenhouse | I wasnt sure with the polycom soundpoint 320 would let me do this or how this differs |
| 20:16.07 | Strom_M | rrittenhouse: you're trying to make a PBX behave like a key system |
| 20:16.13 | rrittenhouse | They love to do all "all call pages" for people that arent in their office to pick up a specified line |
| 20:16.26 | rrittenhouse | and I wasnt sure if it could operate like this or if theres a better way to do it |
| 20:16.27 | southtel | Has anyone ever had an outgoing channel bank hangup immediately when the called party answers? |
| 20:18.31 | [TK]D-Fender | rrittenhouse: Forget about the concept of "lines" with *. |
| 20:18.58 | [TK]D-Fender | southtel: try switching from ks to ls for your ports |
| 20:19.23 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 20:20.00 | [T]ank | i have been looking at tons of SIP cordless phones. Can anyone recommend an affordable product that is good quality? I do not want an ATA. |
| 20:20.04 | rrittenhouse | So what if somebody calls in on line1 and the receptionist can do a page for this person and he happens to be in another room. Can he pick up that conversation if hes in another room with somebody elses extention? |
| 20:20.18 | southtel | In zapta.conf and zaptel.conf? |
| 20:20.34 | McDouglas | is there any decent program to display some statistics from the cdr logs? I dont want a billing system, jsut some usage/load data. Tried asterisk-stat and it doesnt seem to work well |
| 20:21.53 | [TK]D-Fender | [T]ank: ... |
| 20:21.56 | [TK]D-Fender | ~wifisip |
| 20:21.56 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 20:21.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:22.06 | [TK]D-Fender | southtel: both |
| 20:22.28 | [TK]D-Fender | McDouglas: Areski |
| 20:22.50 | [TK]D-Fender | rrittenhouse: Go lookup "call parking" on the WIKI |
| 20:22.51 | jsidhu2 | how about the aastra 480i-ct? |
| 20:23.08 | McDouglas | [TK]D-Fender: thats what i tried, but it doesnt seem to generate any graphs |
| 20:23.19 | [T]ank | [TK]D-Fender: wifi sip is not really what I am looking for. Rather something with a ethernet base and a cordless handset. |
| 20:23.23 | [TK]D-Fender | McDouglas: then you must be doing something wrong. |
| 20:23.27 | rrittenhouse | [TK]D-Fender: thx |
| 20:23.32 | [T]ank | sorry was not specific |
| 20:23.50 | McDouglas | i followed the "lenghty" instructions it came with |
| 20:23.51 | [TK]D-Fender | McDouglas: Take a look at the Seimens SIP DECT phones. They might be worthwhile. |
| 20:24.07 | [TK]D-Fender | [T]ank: rather |
| 20:24.25 | [TK]D-Fender | McDouglas: since it works for everyone else... it IS just you... |
| 20:24.32 | southtel | D-Fender, no love, not getting to the channel that way. |
| 20:24.51 | McDouglas | i can see in apache's error log that in refers to files in the wrong place |
| 20:25.17 | Corydon76-lap | [T]ank, you're looking for SIP DECT, probably, then |
| 20:25.32 | [T]ank | what is DECT? |
| 20:25.53 | Corydon76-lap | Yes |
| 20:26.10 | Corydon76-lap | ~dect |
| 20:26.11 | jbot | Digital European Cordless Telecommunications (telecommunication) [source: V.E.R.A.] |
| 20:26.16 | [T]ank | i see a ton of stuff for skype. will that work with asterisk? |
| 20:26.24 | Nivex | ~skype |
| 20:26.25 | jbot | skype is, like, evil. see gizmoproject.com |
| 20:26.25 | Corydon76-lap | Hell, no |
| 20:26.26 | [TK]D-Fender | [T]ank: `LOL. NO |
| 20:26.34 | [T]ank | thought i would ask ;-) |
| 20:26.41 | [TK]D-Fender | heading home, BBIAB |
| 20:26.42 | [T]ank | *flamed |
| 20:27.10 | Corydon76-lap | See http://asterisk.drunkcoder.com/hacks/ats-config/ for a cheap SIP DECT phone |
| 20:27.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:28.43 | McDouglas | is there any wireless headset which i could use with a soft sip phone, and also having the option to pick up the call with the headset (button) |
| 20:29.30 | McDouglas | well, or rather a recommended one |
| 20:33.30 | rrittenhouse | One last question.. is there a way to do some sort of a "page" so if somebody isnt in their office they can be told what extension to dial (where there call was parked at)? |
| 20:33.40 | *** join/#asterisk clive- (n=pirch@dsl-242-175-223.telkomadsl.co.za) |
| 20:34.24 | hypa7ia | rrittenhouse: there is a call parking app |
| 20:35.08 | rrittenhouse | I just learned that our receptionist could park the call but if the person is somewhere around the building and are around any phone It would be nice to announce what extension for them to dial to pick up their parked call |
| 20:35.16 | rrittenhouse | Im just used to these comdial impact phones (bleh) |
| 20:36.44 | *** join/#asterisk snk00sj (n=gnelisse@apollo.digitalbase.be) |
| 20:37.09 | s34n | setting VXML_URL=intercom=true doesn' |
| 20:37.31 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 20:37.32 | s34n | setting VXML_URL=intercom=true doesn't seem to make the snoms intercom work |
| 20:38.57 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-f4ac6ca07cd247cc) |
| 20:38.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:39.19 | McDouglas | if someone blind transfers a call to a different extension and nobody picks it up, is there a way to connect the caller back to the one initiated the transfer? |
| 20:39.34 | *** join/#asterisk kaje (n=kjkeefe@12.170.196.178) |
| 20:40.44 | kaje | If I'm only going to use asterisk for AsterFax to send faxes, do I need any special hardware besides a working fax modem? |
| 20:42.37 | s34n | kaje: you need a way to connect the fax to the * server |
| 20:43.24 | kaje | asterfax uses an external fax machine to actually send the faxes? |
| 20:44.04 | s34n | kaje: is your modem an internal modem? |
| 20:44.08 | kaje | yeah |
| 20:44.28 | s34n | kaje: does it work in general under linux? |
| 20:44.51 | kaje | dunno, yet... but if it doesn't I'll buy a modem that does... |
| 20:45.07 | s34n | k |
| 20:45.18 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 20:45.22 | kaje | but, is that the only special hardware you need to run an asterfax server? |
| 20:45.39 | s34n | kaje: also, please don't let any old anyone send email to your asterfax server |
| 20:46.06 | kaje | no, I won't... it will be firewalled appropriately |
| 20:46.19 | Yourname`` | Asterisk ended with exit status 139 |
| 20:46.19 | Yourname`` | Asterisk exited on signal 11. |
| 20:46.19 | Yourname`` | Automatically restarting Asterisk. |
| 20:46.27 | kaje | is asterfax pretty mature? I've only learned about it recently... |
| 20:46.35 | Yourname`` | ../usr/sbin/safe_asterisk: line 57: 12698 Segmentation fault (core dumped) ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} |
| 20:47.22 | nDuff | kaje: I strongly prefer asterisk+iaxmodem+hylafax. |
| 20:47.52 | nDuff | kaje: hylafax is extremely mature; iaxmodem only rarely has issues, and it's quite well-supported. |
| 20:47.57 | hmmhesays | these guys in the red bull air race are insane |
| 20:48.10 | *** join/#asterisk saftsack (n=saftsack@pD9E05C35.dip.t-dialin.net) |
| 20:48.30 | bkruse | hmmhesays: is that on now? |
| 20:48.56 | kaje | nDuff: what does hylafax provide that asterfax doesn't? |
| 20:49.11 | s34n | kaje: I hadn't heard of asterfax before. the docs look a bit skimpy |
| 20:49.34 | s34n | kaje: is faximg the only thing you will use asterisk for? |
| 20:49.35 | hmmhesays | bkruse yep |
| 20:49.51 | bkruse | hmmhesays: dang, i need to go home nad watch |
| 20:50.01 | hmmhesays | I bet you can get a torrent of it |
| 20:50.33 | bkruse | for some reason no one at digium can no longer get a response from anywhere outside the network without a 3 second delay :P ill wait for home :] |
| 20:50.41 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 20:50.49 | kaje | s34n: at this point yes, we only need it for sending out faxes using email |
| 20:51.05 | kaje | what is the most commonly used tool for this? hylafax? |
| 20:51.10 | s34n | hmmhesays: the news this morning reported a moment of silence at the air races in Reno for 3 dead pilots |
| 20:51.31 | s34n | kaje: probably hylafax, yes. |
| 20:51.32 | bkruse | 3 pilots died in the contest? |
| 20:51.34 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
| 20:51.36 | b11d|bbl | hello all |
| 20:51.43 | bkruse | b11d: hey |
| 20:51.52 | b11d | just got back from Amsterdam :) |
| 20:51.57 | bkruse | s34n: 3 pilots died in the contest? or just well respected pilots |
| 20:52.15 | s34n | bkruse: the news wasn't clear on that. Said they did cheer for one guy who lived through a crash. |
| 20:52.39 | b11d | I've got a Polycom 550 which crashes (and then, several mins later, reboots) when I dial a number that doesnt match a pattern found in the "digitmap" of the 550, or isnt found in the dialplan.. |
| 20:52.46 | bkruse | s34n: geez, if I was redbull, I would feel kind of lame for causing 3 deaths |
| 20:52.49 | b11d | should I be handlign this problem with a corrected digitmap, or in my dialplan somehow? |
| 20:52.52 | kaje | can someone explain the relationship of trixbox to asterisk? (sorry these are such noobie questions) |
| 20:53.01 | bkruse | kaje: none really |
| 20:53.10 | b11d | trixbox is built on asterisk |
| 20:53.12 | b11d | isnt it?? |
| 20:53.14 | bkruse | trixbox is just an OS with asterisk on top of it, with their little changes no one really knows about |
| 20:53.16 | Qwell | kaje: trixbox is a distro which is just a collection of packages |
| 20:53.19 | Qwell | it's nothing special |
| 20:53.19 | b11d | oh |
| 20:53.20 | b11d | weak |
| 20:53.20 | b11d | :) |
| 20:53.24 | s34n | bkruse: that isn't red bull. The Reno races are the Nationals |
| 20:53.30 | Qwell | and it's certainly not worth the money Andrew got paid for it |
| 20:53.31 | bkruse | s34n: oh, ok |
| 20:53.41 | kaje | would you recommend a new person install trixbox to get their asterisk server working? |
| 20:53.41 | bkruse | Qwell: geez........your telling me |
| 20:53.48 | Qwell | (and that Rob DIDN'T get paid) |
| 20:53.50 | bkruse | kaje: I would recommend http://asterisknow.org |
| 20:54.00 | b11d | anyone? digitmap or dialplan? |
| 20:54.14 | s34n | bkruse: planes are the only things crashing: http://www.airrace.org/indexJS.php |
| 20:55.13 | s34n | bkruse: looks like the deaths were during the Nationals competition |
| 20:55.20 | kaje | bkruse: it's in beta... is that like a google beta or is it a real (unstable) beta? |
| 20:55.26 | bkruse | s34n: thats horrible |
| 20:55.32 | bkruse | kaje: it is, but its fine |
| 20:55.37 | kaje | kk |
| 20:55.38 | bkruse | and it gives you a premade platform to start from |
| 20:55.47 | bkruse | like a good beta, well put |
| 20:55.54 | bkruse | and you can get the latest code once its installed anyways |
| 20:55.56 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 20:55.57 | s34n | s/planes are the only thing/planes are _not_ the only thing/ |
| 20:56.05 | nny | gah |
| 20:56.13 | *** join/#asterisk sureshot5151 (n=keito@cpc2-leed2-0-0-cust11.leed.cable.ntl.com) |
| 20:56.29 | kaje | thanks for all the help guys... I'm sure I'll have more questions in a few... |
| 20:56.41 | bkruse | kaje: np, check out #asterisknow and #asterisk-gui also :] |
| 20:56.42 | bkruse | gl! |
| 20:57.17 | sureshot5151 | hi I'd like to know if I can ditch skype and use asterisk for all my voip needs? it's a little intimidating looking at the site as it seems asterisk is more catered towards an exchange application |
| 20:57.33 | Qwell | sureshot5151: well, asterisk can't do skype |
| 20:57.46 | s34n | I need a snom360 to autoanswer when it is paged, and ring normally in all other cases. |
| 20:57.48 | Qwell | it can use real protocols though... |
| 20:57.49 | sureshot5151 | thats fine but what *can* it do? |
| 20:58.00 | Qwell | sureshot5151: it can do pretty much anything you can think of... |
| 20:58.08 | s34n | But I can only get it to always auto answer, or always ring normally |
| 20:58.20 | sureshot5151 | if I wanted to ring a landline can it do that? |
| 20:58.27 | Qwell | people use asterisk for *TONS* of things. one of the most recent examples is having a plant call you when it needs to be watered |
| 20:58.37 | sureshot5151 | lol |
| 20:58.43 | Qwell | sureshot5151: sure, you'd either need an ITSP, or some hardware to connect to your phone line |
| 20:58.46 | Qwell | ~itsp |
| 20:58.47 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 20:58.50 | sureshot5151 | with the correct sensors then I can see that being possible lol |
| 20:58.52 | sureshot5151 | 80) |
| 20:59.09 | s34n | sureshot5151: skype and asterisk aren't trying to do the same thing |
| 20:59.13 | nny | OK this is just a fun fact for all us noobs out here |
| 20:59.16 | sureshot5151 | so can i ring a landline? |
| 20:59.28 | Qwell | sureshot5151: see above |
| 20:59.42 | nny | no matter *what* you think may be right with the telco's landline, they can still f*ck things up royally |
| 20:59.47 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:00.03 | nny | [TK]D-Fender: ha |
| 21:00.18 | sureshot5151 | I know little to nothing about what asterisk does hence my being here |
| 21:00.22 | nny | [TK]D-Fender: second time I have wasted my time dealing with our telco's "service |
| 21:00.25 | Qwell | ~asterisk |
| 21:00.26 | jbot | hmm... asterisk is the best free PBX in the world |
| 21:00.42 | s34n | sureshot5151: are you looking for a way to make ip phone calls, or for a pbx that just happens to handle ip traffic very handily? |
| 21:00.54 | *** part/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 21:00.59 | sureshot5151 | ip calls |
| 21:01.06 | *** part/#asterisk clive- (n=pirch@dsl-242-175-223.telkomadsl.co.za) |
| 21:01.10 | [TK]D-Fender | sureshot5151, Know anything about ancient Mesopotamia? Move along and go to a university! |
| 21:01.15 | s34n | sureshot5151: asterisk is almost certainly not what you are looking for |
| 21:01.22 | Strom_C | sureshot5151: you may wish to start off by reading the following documents |
| 21:01.26 | Strom_C | ~thebook |
| 21:01.26 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:01.27 | Strom_C | ~101 |
| 21:01.27 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 21:01.28 | sureshot5151 | I have ekiga installed at the moment and skype but am looking at other options |
| 21:01.41 | nny | [TK]D-Fender: they turned on Centrex based VM two weeks ago, damn thing would pick up right when * VM did. We figured that out aftr hours of cursing, today, for no reason the hang up issue is from them. I brought the PBX in house, hooked it up, and it works great.Something on their line is hanging up and causing the it to crap the bed |
| 21:01.59 | s34n | sureshot5151: do you know what a pbx is? |
| 21:01.59 | sureshot5151 | [TK]D-Fender: gee thanks for the kick in the teeth I have been to uni cheers |
| 21:02.07 | b11d | TK... |
| 21:02.09 | b11d | whats up? |
| 21:02.11 | sureshot5151 | s34n: no thats why I'm here |
| 21:02.17 | [TK]D-Fender | nny, Good. the MYSTERY is solved, now its jsut a matter of time to resolution. |
| 21:02.19 | sureshot5151 | though i sense I've come to the wrong place |
| 21:02.21 | Qwell | sureshot5151: see the above URLs |
| 21:02.21 | [TK]D-Fender | b11d, y0 |
| 21:02.27 | b11d | Hey TK.. you might know.. I've got a Polycom 550 which crashes (and then, several mins later, reboots) when I dial a number that doesnt match a pattern found in the "digitmap" of the 550, or isnt found in the dialplan.. |
| 21:02.44 | nny | [TK]D-Fender: lol yeah. Shoot the hostage |
| 21:02.46 | sureshot5151 | Qwell: thanks I will |
| 21:02.46 | s34n | a pbx is like a business phone system that handles multiple phone lines and all sorts of extensions throughout a business |
| 21:02.57 | b11d | should I be modifying my digitmap, or is it better to handle it in the dialplan? |
| 21:02.58 | sureshot5151 | s34n: an exchange system yes? |
| 21:02.59 | nny | [TK]D-Fender: seriously, the client never even requested telco side VM |
| 21:03.02 | [TK]D-Fender | nny, woah :) |
| 21:03.13 | nny | [TK]D-Fender: :) |
| 21:03.26 | [TK]D-Fender | b11d, Dunno if I trust your firmware or your configuration of it :) |
| 21:03.42 | [TK]D-Fender | ~book |
| 21:03.43 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 21:03.44 | b11d | im running SIP 2.2.0 now |
| 21:03.52 | [TK]D-Fender | ^^^^^^^ mirror link for THE BOOK |
| 21:04.00 | b11d | I dont need the book :) |
| 21:04.02 | [TK]D-Fender | b11d, yee-haw |
| 21:04.15 | s34n | sureshot5151: you know that really fancy phone system at work? The one that lets you dial extensions within the building, and dial 9 to get an outside line? |
| 21:04.16 | [TK]D-Fender | b11d, Stop assuming I'm here exclusively for YOU :p |
| 21:04.17 | b11d | i can handle this problem in two places.. I figured the dialplan was the logical place to fix it, but thought i'd ask around for commentary |
| 21:04.22 | b11d | I dont assume that. |
| 21:04.39 | b11d | Didnt realize I came across like that.. sorry |
| 21:04.41 | nny | so |
| 21:04.44 | [TK]D-Fender | b11d, My dialplan forego those issues and am running 2.2.0 happily at home myself |
| 21:04.47 | brodiem | Can anyone here report if they've had good/bad results using a router w/ built-in SPI firewall for SIP? |
| 21:05.20 | [TK]D-Fender | brodiem, SIP aware firewalls are almost invariably too smart for YOUR good. |
| 21:05.30 | nny | my thought. We have a vonage box here. Iw ant to build something that I can take to a site, and test the fxo modules by generating a dial tone like and calling in. We figured Vonage + number + FXS port on vonage box = test equipment. Anyone have a fancier or better way? |
| 21:05.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:05.48 | brodiem | [TK]D-Fender: I said SPI (Stateful Packet Inspection) |
| 21:06.14 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 21:06.16 | [TK]D-Fender | brodiem, Isn't pretty much every router out there these days do that? |
| 21:06.46 | brodiem | [TK]D-Fender: a lot of them do, but a lot of them disable it by default too |
| 21:06.54 | s34n | brodiem: Every SPI firewall I've seen (not that many) mangles traffic |
| 21:07.34 | s34n | brodiem: especially sonicwall |
| 21:07.36 | brodiem | s34n: mangling SIP as unusable? |
| 21:08.09 | nny | brodiem: as long as by SIP router you dont mean something that handles the ports automagically, and needs configuration, I use DDWRT or OPenWRT on misc hardware. You can pretty much watch the packets fly by with tcpdump if you wish, or send them to a logger etc. |
| 21:08.10 | s34n | brodiem: mangling regular ip traffic |
| 21:08.21 | mcab | it's a rare device that leaves SIP being usable after mangling it... |
| 21:08.45 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 21:08.45 | brodiem | nny: SPI not SIP |
| 21:08.49 | BockBilbo | hello again |
| 21:08.50 | BockBilbo | :) |
| 21:08.56 | nny | brodiem: ever tried openWRT or DDWRT? |
| 21:09.04 | sureshot5151 | s34n: yes I understand, hence why I'm asking if asterisk allows me to make LANDLINE CALLS? nevermind though as I will go "read the book" I can do without the slating as I've had enough of that for one day on IRC I'm just glad people like [TK]D-Fender aren't teching in uni's as no-one would ever learn *anything* |
| 21:09.06 | brodiem | i'm not dyslexic :) |
| 21:09.17 | sureshot5151 | teaching/ |
| 21:09.28 | s34n | sureshot5151: you can make landline calls with asterisk |
| 21:09.34 | sureshot5151 | ty |
| 21:09.39 | nny | brodiem: that makes one of us |
| 21:09.48 | brodiem | lol |
| 21:09.51 | [TK]D-Fender | sureshot5151, Start over. What EXACTLY do you want to do and using what kind of resources & services? |
| 21:09.57 | sureshot5151 | w00h00 now we is getting somewhere ;0) |
| 21:10.07 | sureshot5151 | I want to learn about asterisk |
| 21:10.07 | brodiem | nny: I'vev not tried either, it was more of a general question about SPI-enabled routers |
| 21:10.11 | nny | DDWRT has an SPI firewall and some logging ability, although for massive traffic or long term logging, Id dump it to another server |
| 21:10.11 | sureshot5151 | not do anything |
| 21:10.12 | s34n | sureshot5151: but you need hardware to connect an analog phone to your asterisk system |
| 21:10.20 | sureshot5151 | coll |
| 21:10.21 | sureshot5151 | cool |
| 21:10.23 | nny | I use an SPI router here, and have tested SIP across VPN |
| 21:10.28 | sureshot5151 | 80)~ |
| 21:10.30 | BockBilbo | one question... which of these two ata's is better, the linksys PAP2 or SPA1001? |
| 21:10.41 | sureshot5151 | one of those links led to a 404 fyi |
| 21:10.44 | nny | not with holes punched for SIP though.. Don't trust the auth enough, nor my experience with it |
| 21:11.08 | s34n | sureshot5151: of course, you can use asterisk to make landline calls using an ip phone, too |
| 21:11.23 | sureshot5151 | [TK]D-Fender: sorry for the olwblow its just I was over at winxp room and they were about as much use a chocolate fireguard |
| 21:11.40 | [TK]D-Fender | sureshot5151, * is a telephony toolkit that can allow you to build a PBX, a call-out system, and a bunch of other stuff. |
| 21:11.42 | sureshot5151 | s34n: I have an ip phone |
| 21:11.44 | brodiem | nny: I seem to have some odd behaviour with a couple of SPI routers.. one keeps killing registrations intermittently, as well as of course causing both inbound/outbound call failures. Another prevents INVITEs from getting to the internal IP phones, but only if more than one simultaneous invites are sent.. |
| 21:12.12 | s34n | sureshot5151: for "land line" calls, you will need a "land line" |
| 21:12.36 | sureshot5151 | [TK]D-Fender: I've had my eye on it for a while after it being mention in an eipsode of "the borken" but only recently moved to linux you see |
| 21:12.38 | brodiem | nny: that's why I'm trying to figure out if SPI just doesn't play well with SIP in general or if it's a coincidence with the ones I've been trying |
| 21:12.46 | s34n | and it will need a way of connecting to the asterisk system |
| 21:12.47 | [TK]D-Fender | sureshot5151, As you have an IP phone, you can use * to conenct to hardware lines you may have, and to a variety of ITSP's for PSTN access. * can can provider intelligent call routing, Voicmeil, and tons of other stuff like voice menus, etc. |
| 21:13.01 | sureshot5151 | s34n: is there another term for landline that is more appropriate for * |
| 21:13.15 | nny | brodiem: hrrm... like I said, never actually used it with SIP opne on the router, or NAT transversing, I just popped an Ekiga SIP client on my laptop and tested over the VPN... sorry wish I could help more.. |
| 21:13.32 | s34n | sureshot5151: POTS or PSTN |
| 21:13.35 | nny | sureshot5151: POTS |
| 21:13.38 | brodiem | nny np, thanks |
| 21:13.39 | nny | oops |
| 21:13.41 | nny | sry |
| 21:13.43 | sureshot5151 | [TK]D-Fender: I have no landline here so I guess ip calls only |
| 21:13.47 | nny | brodiem: np good luck |
| 21:14.19 | sureshot5151 | s34n & nny cheers, much appreciated |
| 21:14.24 | [TK]D-Fender | sureshot5151, Great. So for MOST people * is used as a PBX between their phones and the telco (however they choose to get there, be it land lines, or TISP's, etc) |
| 21:14.31 | [TK]D-Fender | sureshot5151, that help clarify things a bit? |
| 21:14.42 | sureshot5151 | sure did |
| 21:14.46 | sureshot5151 | legends |
| 21:15.00 | yang | how can i reload extensions.conf |
| 21:15.11 | s34n | yang: extenstions reload |
| 21:15.15 | Corydon76-lap | "dialplan reload" |
| 21:15.19 | sureshot5151 | well now I know that little more I shall go do some reading, muchos gracias |
| 21:15.24 | nny | sureshot5151: there is a great tutorial in the asterisk book that shows how to test * with a FWD account and you can use any SIP client, essentially making the entire process low cost |
| 21:15.41 | [TK]D-Fender | sureshot5151, * offers you easy to implement ways of doing voice menu's, large conference rooms, voicemail, recording calls, and a pile of other useful stuff limited by your time and willingness to code if needed. |
| 21:15.47 | sureshot5151 | that was the link that 404'd so I'll go google |
| 21:15.52 | Corydon76-lap | or more generically, "reload pbx_config.so" |
| 21:16.01 | nny | ~book |
| 21:16.02 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 21:16.13 | nny | sureshot5151: 2nd one works |
| 21:16.20 | Corydon76-lap | That IS the old version, now |
| 21:16.23 | sureshot5151 | sw33t |
| 21:16.28 | nny | sureshot5151: i read that for a bit and then bought the oreilly version |
| 21:16.34 | Corydon76-lap | The 2nd edition is available in your local bookstore |
| 21:16.41 | nny | :( |
| 21:16.46 | nny | i think i have the 1st one still |
| 21:16.53 | Corydon76-lap | and is about twice as thick |
| 21:17.01 | sureshot5151 | $,$,$.....$ |
| 21:17.14 | nny | yah I have 1st edition |
| 21:17.16 | nny | bastards! |
| 21:17.22 | sureshot5151 | lol |
| 21:17.28 | Corydon76-lap | or buy from oreilly.com... |
| 21:17.38 | sureshot5151 | right I'm off danke |
| 21:18.24 | [TK]D-Fender | Corydon76-lap, TFOT 2nd ED should be released in PDF to the general public following Astricon. |
| 21:18.32 | nny | ok so im out. I have officially learned that the first rule of asterisk is if it's not working, it's hardley ever asterisks fault. I can blame someone else almost always and sound confident ;) |
| 21:18.40 | nny | hardly* |
| 21:18.57 | [TK]D-Fender | nny, GIGO > 99% of users |
| 21:18.59 | Corydon76-lap | [TK]D-Fender, that soon? |
| 21:19.04 | *** join/#asterisk mistermocha (n=chef@adsl-75-40-192-38.dsl.irvnca.sbcglobal.net) |
| 21:19.06 | [TK]D-Fender | Corydon76-lap, Yes. |
| 21:19.09 | brodiem | Corydon76-lap: sep 26 i believe |
| 21:19.10 | mistermocha | hi all... dumb question |
| 21:19.33 | Corydon76-lap | I'd much rather have the dead tree version, though |
| 21:19.41 | brodiem | that week at least |
| 21:19.49 | nny | [TK]D-Fender: yup, I believe it. |
| 21:19.51 | *** part/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 21:20.05 | mistermocha | or not, maybe... if a queue strategy is "random", does the penalty or priority have any weight? |
| 21:20.09 | [TK]D-Fender | Corydon76-lap, You're completely entitled to it and I encourage those who would to follow through. For the merely curious though I believe in information for the masses. |
| 21:20.44 | s34n | where's a good site to read ip phone comparisons? |
| 21:20.46 | mistermocha | no documentation anywhere seems to clarify this |
| 21:20.50 | putnopvut | mistermocha: I believe so. |
| 21:20.51 | [TK]D-Fender | mistermocha, it would be random across a given penalty. |
| 21:21.34 | [TK]D-Fender | mistermocha, AKA if multiple available agents chared a penalty then THEY would be the group chosen from at random, and not the complete set |
| 21:21.39 | mistermocha | so... if in a queue, I have three users of high priority and three users of low priority, it would ring a random agent from the high pri and another agent from the low? |
| 21:21.40 | Corydon76-lap | [TK]D-Fender, I believe in encouraging authors to write through royalty checks. ;-) |
| 21:21.40 | putnopvut | ...just checked, and [TK]D-Fender is 100% correct. |
| 21:22.14 | mistermocha | s/and another agent/then another agent/ |
| 21:22.16 | [TK]D-Fender | Corydon76-lap, I support the authors with beer & transport :) |
| 21:22.46 | [TK]D-Fender | Corydon76-lap, And of course in advertiseing the book :) |
| 21:22.50 | putnopvut | mistermocha: it would attempt to ring each of the agents of the higher priority and then each of the agents in the low priority. |
| 21:23.04 | mistermocha | awesome-ness |
| 21:23.11 | mistermocha | thanks a bunch all! |
| 21:23.40 | mistermocha | putnopvut: curiously... where did you find the info? is it documented somewherE? |
| 21:23.40 | putnopvut | mistermocha: keep in mind though that penalty is actually sort of the opposite of priority. You probably knew that already though. |
| 21:23.47 | putnopvut | I looked at the code :) |
| 21:23.50 | mistermocha | putnopvut: yeah, I did... |
| 21:23.57 | mistermocha | supadupa sweet |
| 21:24.37 | *** join/#asterisk cremedios (n=cremedio@82.155.230.99) |
| 21:25.10 | cremedios | Good night |
| 21:25.31 | cremedios | can anyone help me with a digium te220b install, please? |
| 21:30.30 | *** join/#asterisk Zyl0ne (n=zylone@206.40.234.218) |
| 21:30.46 | [TK]D-Fender | cremedios, ... |
| 21:30.47 | [TK]D-Fender | ~ask |
| 21:30.47 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:32.34 | Zyl0ne | Anyone know if I can use an old nortel meridian phone with asterisk? |
| 21:32.39 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:33.00 | [TK]D-Fender | Zyl0ne, there are a few expensive gateways that allow you to do that. About $3000 / 24 ports |
| 21:33.17 | Zyl0ne | eww |
| 21:33.18 | [TK]D-Fender | Zyl0ne, seriously not worth it in the majority of cases |
| 21:33.37 | Zyl0ne | yeah, just curious because I have some of them phones, and need some for my house |
| 21:33.38 | Zyl0ne | heh |
| 21:33.41 | ZaVoid | does anyone have anyidea why rtptimeout in sip.conf only applies to peers? |
| 21:33.45 | Zyl0ne | in that case I will just get something else |
| 21:35.24 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:36.07 | cremedios | i'm sorry, it's just that i am running out of ideas |
| 21:37.49 | cremedios | when i do a cat /proc/interrupts, my card shows up with 0. zttool says the configuration is ok. Any ideas ? |
| 21:39.48 | ZaVoid | hey [TK]D-Fender any idea how i can get any kind of rtptimeout to work ona non type=peer so type=friend or user.... |
| 21:41.01 | [TK]D-Fender | ZaVoid, what * version. |
| 21:41.08 | ZaVoid | 1.4.9 |
| 21:41.19 | ZaVoid | how ya doing this week too? |
| 21:41.22 | [TK]D-Fender | ZaVoid, In case you didn't know, friend effectively no longer exists in 1.4 |
| 21:41.30 | ZaVoid | effectively? |
| 21:41.42 | ZaVoid | hmm one sec |
| 21:41.43 | [TK]D-Fender | ZaVoid, EVERYTHING may as well be "peer" and you are advised to do as such |
| 21:42.05 | [TK]D-Fender | ZaVoid, Set your phones that way, you'll see. They're undoing the old SIP model. |
| 21:42.22 | ZaVoid | its depricated? |
| 21:42.36 | ZaVoid | i tried setting my account using REALTIME to peer and it stopped working |
| 21:42.46 | [TK]D-Fender | ZaVoid, I don't know all the particulars but I think thats the approach. |
| 21:42.52 | ZaVoid | hmm |
| 21:42.58 | ZaVoid | let me dblc hecksomthin 1 sec |
| 21:44.06 | ZaVoid | tyeah its type=friend in my DB right now |
| 21:46.04 | ZaVoid | testing |
| 21:46.40 | *** join/#asterisk NoCarrier (n=John@unaffiliated/badpacket) |
| 21:49.10 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-87.usadatanet.com) |
| 21:49.13 | ZaVoid | hmm |
| 21:49.27 | ZaVoid | the last time i changed an account to peer in my DB i couldn't getit to register |
| 21:49.38 | ZaVoid | did i have ot restart sasterisk for realtime update perhaps? |
| 21:49.56 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 22:09.32 | yang | could someone ring me on my SIP phone, i would like to test connection |
| 22:09.51 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:12.53 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 22:13.22 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:20.13 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 22:21.13 | PioneerVM2 | hi, i am currently using VOIP on an asterisk box, but may want to bring in a physical line for a business purpose (reliability). The boxes are 1U servers with no extra card slots, are there USB based devices i can plug in that can handle physical copper lines into asterisk? |
| 22:21.34 | PioneerVM2 | (or are my only options to use a card) |
| 22:23.26 | defsdoor | PioneerVM2: a channel bank ? |
| 22:24.25 | PioneerVM2 | im not too familiar with the terms for handling copper phonelines in asterisk |
| 22:24.32 | PioneerVM2 | right now im using all VOIP stuff over ethernet |
| 22:24.48 | defsdoor | PioneerVM2: you install a card usually |
| 22:24.53 | defsdoor | PioneerVM2: but if you cant..... |
| 22:25.18 | defsdoor | you could simply get an spa3102 if you just want a single analog line |
| 22:25.28 | defsdoor | 50quid from the right place |
| 22:25.33 | PioneerVM2 | how does that interface |
| 22:25.36 | PioneerVM2 | to the asterisk box |
| 22:25.37 | defsdoor | tcp |
| 22:25.46 | PioneerVM2 | ok so it goes on my local network |
| 22:25.53 | defsdoor | aye |
| 22:25.59 | PioneerVM2 | so its like a reverse PAP2T box |
| 22:26.21 | defsdoor | it works both ways |
| 22:26.49 | defsdoor | you get analog line out, analog line phone and sip |
| 22:27.07 | PioneerVM2 | oh duh, i guess i could just use my linksys pap2t box |
| 22:27.11 | defsdoor | I have one here but my asterisk box died :) |
| 22:27.16 | defsdoor | PioneerVM2: yes |
| 22:27.19 | PioneerVM2 | its just analog to ethernet, i use it right now for getting calls at home |
| 22:27.25 | defsdoor | exactly |
| 22:27.25 | PioneerVM2 | i never really thought about it in the reverse way |
| 22:27.41 | PioneerVM2 | sweet thanks |
| 22:27.53 | PioneerVM2 | i want to do this to make it more reliable for a biz line |
| 22:28.09 | defsdoor | I got a laptop with a broken screen |
| 22:28.16 | PioneerVM2 | i guess theonly catch is i can only handle one call at a time |
| 22:28.24 | defsdoor | stuck trixbox on it and the spa3102 |
| 22:29.16 | PioneerVM2 | i suppose the way to go would be get two copper lines with rollover |
| 22:29.24 | PioneerVM2 | copper A -> Copper B -> VOIP |
| 22:29.36 | PioneerVM2 | so i can usually handle on two coppers but if i get too busy i have voip as backup |
| 22:30.27 | *** join/#asterisk tripps (n=ss@m945f36d0.tmodns.net) |
| 22:30.30 | PioneerVM2 | oh wait if i have an external line into a PAP2t |
| 22:30.36 | PioneerVM2 | rather than a phone handset, can it pickup calls incoming? |
| 22:30.51 | PioneerVM2 | thats the only difference |
| 22:30.56 | PioneerVM2 | right now i pickup handset and dial on analog side |
| 22:31.04 | PioneerVM2 | with this setup the call would come in on analog line |
| 22:32.13 | hypa7ia | PioneerVM2: you need an FXO device to plug a copper line into |
| 22:32.27 | hypa7ia | versus an FXS that you plug a phone into |
| 22:32.33 | PioneerVM2 | ok |
| 22:32.41 | PioneerVM2 | it works the same though thru IP? |
| 22:32.49 | PioneerVM2 | can i get one that goes on USB or my IP network? |
| 22:33.03 | hypa7ia | you can get ATA's (which connect to your asterisk via SIP or IAX) with either FXO's or FXS's or both |
| 22:33.09 | yang | could someone ring me on my SIP phone, i would like to test connection |
| 22:33.11 | PioneerVM2 | ok cool |
| 22:33.15 | hypa7ia | PioneerVM2: i don't think USB ones exist |
| 22:33.21 | PioneerVM2 | Ethernet is fine |
| 22:33.26 | hypa7ia | good :) |
| 22:33.35 | PioneerVM2 | do you plug regular copper lines in |
| 22:33.42 | PioneerVM2 | or do i have to get some kind of special line |
| 22:34.00 | PioneerVM2 | rj45 i think its called? |
| 22:34.15 | hypa7ia | rj45 is an ethernet cable |
| 22:34.17 | hypa7ia | :) |
| 22:34.26 | PioneerVM2 | rj11 sorry |
| 22:34.28 | PioneerVM2 | wasnt thinking |
| 22:34.32 | hypa7ia | that's just a phone line |
| 22:34.40 | PioneerVM2 | yes so i plug rj11 into these boxes |
| 22:34.45 | hypa7ia | most ATA's take a regular phone line and an ethernet line |
| 22:34.48 | PioneerVM2 | ok |
| 22:34.53 | PioneerVM2 | now what if i want more lines |
| 22:34.59 | PioneerVM2 | i hear about these things that can take up to 20 lines |
| 22:35.06 | PioneerVM2 | like cards etc. |
| 22:35.43 | nDuff | PioneerVM2: if you want to do lots of lines, you'll probably want to get a PRI coming in -- basically, a T1 but with voice channels instead of data. |
| 22:35.52 | PioneerVM2 | thats the term i couldnt remember |
| 22:35.56 | hypa7ia | for that you need a PCI line card in your server |
| 22:36.03 | PioneerVM2 | do they make external boxes for that |
| 22:36.09 | PioneerVM2 | i cant put any more cards in |
| 22:36.46 | PioneerVM2 | and i guess i buy PRIs from my local phone company? |
| 22:36.52 | nDuff | PioneerVM2: external boxes exist, but you don't want to go that route. |
| 22:37.00 | jsidhu2 | ..how do you transfer calls? |
| 22:37.01 | nDuff | PioneerVM2: they'll cost more than a server with open PCI slots. |
| 22:37.07 | PioneerVM2 | ok |
| 22:37.12 | nDuff | PioneerVM2: yes, go through your telco. |
| 22:37.23 | McDouglas | [TK]D-Fender: you was right (big suprise :P) about the areski. When i imported the csv data to mysql, something got messed up. However now that its working i do miss something. Is there a way to check how many BRI ports are in use in a given time? |
| 22:37.35 | PioneerVM2 | do you know what a PRI roughly costs? |
| 22:37.37 | hypa7ia | jsidhu2: look on the wiki |
| 22:37.42 | nDuff | PioneerVM2: depends dramatically on where you're at. |
| 22:37.46 | hypa7ia | PioneerVM2: it varies dramatically based on your area |
| 22:37.51 | hypa7ia | nDuff: jinx |
| 22:37.55 | jsidhu2 | hypa7ia: link me please... |
| 22:37.59 | PioneerVM2 | lol |
| 22:38.27 | PioneerVM2 | can you get them with less lines, like a "1 line PRI" |
| 22:38.33 | hypa7ia | jsidhu2: you can google it as well as i can |
| 22:38.34 | PioneerVM2 | and expand later? or does that depend as well |
| 22:38.37 | jsidhu2 | voip-info wiki?> |
| 22:38.45 | hypa7ia | jsidhu2: yes |
| 22:38.51 | jsidhu2 | thank you |
| 22:38.59 | hypa7ia | PioneerVM2: yes, starting at about 6 lines iirc |
| 22:39.00 | PioneerVM2 | and does it work like "channels" so i get 20 channels that can be all one phone # or 20 different phone #s |
| 22:39.17 | hypa7ia | you get to set up whatever ring groups and DID's you want |
| 22:39.23 | PioneerVM2 | nice |
| 22:39.24 | hypa7ia | the price varies of course |
| 22:39.44 | PioneerVM2 | have you seen generally that they are cheaper per "channel" than getting 20 regular phone lines |
| 22:39.50 | PioneerVM2 | or do they charge you more because its a special app |
| 22:39.51 | hypa7ia | i think so |
| 22:39.54 | hypa7ia | cheaper |
| 22:39.57 | nDuff | PioneerVM2: it's cheaper, at least here. |
| 22:40.00 | PioneerVM2 | ok thats good, phone lines in general are a fortune |
| 22:40.05 | hypa7ia | indeed |
| 22:40.42 | PioneerVM2 | ok cool. I dont need to do this immediately but if everything works out ill want to make it more reliable and go this route |
| 22:41.00 | PioneerVM2 | thx for all the info -- nice to not get an ego on answers here :) |
| 22:41.22 | PioneerVM2 | gotta run feed my son, thx again |
| 22:41.39 | [T]ank | need a second pair of eyes to see why i am getting lost in my dial plan when a call is hung up: http://pastebin.ca/701231 |
| 22:42.01 | [T]ank | when the call is hung up it then calls my s-noanswer dial status. instead of just hanging up. |
| 22:42.15 | hypa7ia | PioneerVM2: you're welcome :) |
| 22:42.26 | [TK]D-Fender | -- User hung up <------------ |
| 22:42.38 | hypa7ia | PioneerVM2: i do also recommend reading up on the voip-info wiki about PRI |
| 22:42.48 | *** join/#asterisk minkus (n=minkus@pool-72-84-53-31.clrkwv.east.verizon.net) |
| 22:43.22 | [T]ank | oh, i am just misunderstanding the out put then i guess. it is just exiting the s-NOANSWER? |
| 22:43.27 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:49.15 | [TK]D-Fender | [T]ank, They hit voicemail and HANGUP themselves |
| 22:49.51 | [TK]D-Fender | PioneerVM2, T1-PRI allows you from 1-23 channels. You can have ANY number of DID's landing on your PRI up to the point where you have no more free channels. |
| 22:52.08 | [T]ank | [TK]D-Fender: thanks, was misunderstanding the output. |
| 22:55.31 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.12.183) |
| 22:56.26 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 23:02.47 | PioneerVM2 | DF thanks |
| 23:03.14 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.108) |
| 23:03.34 | dijungal | what is it with all these HDLC erros on the TE450P card :| |
| 23:04.11 | dijungal | i configured port 1 & 2 some time back....... |
| 23:04.17 | Qwell | dijungal: You should probably call support |
| 23:04.19 | dijungal | was working fine |
| 23:04.29 | Qwell | you get free install support, afterall |
| 23:04.38 | dijungal | now i configure port 3 & 4 i'm getting tons of HDLC errors :( |
| 23:04.47 | dijungal | oooh |
| 23:04.50 | dijungal | didn't know that |
| 23:05.40 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 23:08.01 | *** join/#asterisk fadey (n=fadey@89.140.51.39.static.user.ono.com) |
| 23:10.27 | fadey | hello, everyone. I'm desperate with setting a higher IRQ priority for my digium card. Is changing a PCI slot is the only way to do it? |
| 23:10.49 | JT | higher priority? huh |
| 23:11.28 | nDuff | fadey: mostly. |
| 23:11.38 | fadey | :-( |
| 23:11.46 | nDuff | fadey: depends on your motherboard. |
| 23:11.51 | JT | you mean different irq |
| 23:11.55 | fadey | yes |
| 23:11.56 | JT | not higher priority? |
| 23:12.54 | fadey | well, the page I've read says those are correlated. The lower the IRQ num - the higher priority device gets |
| 23:14.23 | nDuff | fadey: Granted, you've already bought Digium hardware, but I'm something of a fan of Sangoma's PCI express cards -- never had any lost-interrupt issues with those yet. |
| 23:16.36 | fadey | Yes, I've got a Digium card. zttest showed 99.9755. People on digium forum say that's kind of bad, but could work |
| 23:17.00 | elixer | fadey: what about digium support, what did they say? |
| 23:17.06 | nDuff | fadey: that's well beyond "kind of bad", especially if you want reliable faxing. |
| 23:17.49 | JT | 99.975 should be ok |
| 23:17.54 | JT | anything below that is not ok |
| 23:18.00 | fadey | Have no idea if I have their support. I'm from Spain, and I bought from a reseller |
| 23:18.09 | Qwell | fadey: doesn't matter - you still get free support |
| 23:18.20 | Qwell | call or email them |
| 23:18.22 | nDuff | 99.98 is quoted in some docs, but yeah, 99.975 might be OK. |
| 23:18.36 | fadey | Wow. Didn't know that. Thanks |
| 23:20.24 | datachomper | Can asterisk signal the upstream provider to forward a call to a different number? |
| 23:21.13 | JT | nDuff: i'm saying what is actually ok |
| 23:21.22 | jsidhu2 | so im messing around with xml applications for the aastra 480i and one of the samples is directory listing. it gets the data from sip_additional.conf. But instead of the name, it shows "device". and its getting it from the "caller id=" field in sip_additional.conf. I dont see a place from the web freepbx extensions config to put a name to this extension.. is this normal? Where is the "Dispaly Name" stored for an extension? |
| 23:21.24 | JT | 99.975 is obviously on the border |
| 23:21.37 | JT | jsidhu2: wrong channel |
| 23:21.39 | JT | ~freepbx |
| 23:21.40 | jbot | well, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:21.42 | jsidhu2 | :-/ |
| 23:21.44 | jsidhu2 | ok thanks |
| 23:22.02 | elixer | :-\ |
| 23:22.21 | jsidhu2 | can someone tell me what is asterisk and what is freepbx? |
| 23:22.33 | Qwell | ~asterisk |
| 23:22.33 | jbot | methinks asterisk is the best free PBX in the world |
| 23:22.39 | Qwell | and freepbx is not a free pbx |
| 23:22.40 | Qwell | ... |
| 23:23.05 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 23:24.06 | JT | freepbx is an aftermarket web based gui interface that uses asterisk |
| 23:24.17 | wishes | i have my asterisk set up with a database (mysql), are there any known problems with using a remote database on the same internal LAN ? |
| 23:24.35 | elixer | wishes: i have no problems with it |
| 23:24.58 | jsidhu2 | ahh |
| 23:24.59 | jsidhu2 | thanks |
| 23:25.04 | jsidhu2 | i see that now :-/ |
| 23:25.11 | wishes | cool, it would help me imensely to be using the CRM so we can do even more funky stuff with it |
| 23:25.50 | elixer | wishes: we do some automagic integration with our CRM here to track recordings and such. nothing special, but its 100% reliable so far. |
| 23:25.57 | elixer | wishes: via mysql, sorry. |
| 23:26.20 | wishes | oh we use a php script to call that kinda stuff (inserting it remotly using php) |
| 23:26.52 | wishes | thats for caller id and so customer service have the users account automagicly open up if the incoming call is mapped to a client |
| 23:27.03 | elixer | right |
| 23:27.36 | elixer | here we do asterisk -> fastagi -> .net windows service -> mysql db |
| 23:27.39 | wishes | but i want the db that maps exten to users and mobiles etc all in the same place so we can set statuses of each user based on their schedule etc |
| 23:28.07 | wishes | so book a meeting on intranet, set phone status to redirect to mobile or mark me as busy and go to xxx answerphone etc kinda thing |
| 23:28.19 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 23:28.23 | elixer | ohhhh, that would be cool |
| 23:28.25 | wishes | it all works, just want to be using the same database for the lot rather than 2 seperate ones |
| 23:28.30 | elixer | yeah |
| 23:28.59 | *** join/#asterisk perd (i=[U2FsdGV@207.44.158.6) |
| 23:29.20 | wishes | so users have like 5 custom states, available, on_phone, in a meeting/busy/away from desk ,away sick, redirect to mobile, custom recorded one |
| 23:29.36 | perd | hello dudes. if i had a system with 30 analog phones and 2 FXOs to the real world, would it be possible to show line status on those analog phones assuming they supported it? |
| 23:29.38 | wishes | some of course are automated - ie being on the phone, etc |
| 23:30.04 | [TK]D-Fender | perd, No. Analog phones don't support that in the first place. |
| 23:30.18 | perd | i thought as much |
| 23:30.31 | perd | the best i could do is set up a recording to tell people 'all circuits busy' ... |
| 23:31.14 | perd | is this a supported feature for sip phones? |
| 23:31.40 | [TK]D-Fender | perd, Yes, many SIP phones support the ability to report on the status of other devices. |
| 23:33.18 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:34.53 | perd | it's a shame they cost so much more than analog pos' |
| 23:36.22 | hypa7ia | perd: not really |
| 23:36.34 | hypa7ia | you can get decent sip phones for well under $150 |
| 23:37.47 | perd | well, when you're looking for 30 phones the difference between 20 and 100 dollars is pretty large ;/ especially since the place is a monestary |
| 23:37.56 | perd | they really dont need all the cool features |
| 23:38.18 | hypa7ia | well they have to connect to the pbx somehow |
| 23:38.32 | hypa7ia | so even analog phones will still need ATA's at 60-100$ a pop |
| 23:38.38 | perd | they're using an old TIE system at the moment, with a dead cmos battery so every time the power goes out they lose all their settings heh |
| 23:38.48 | hypa7ia | ouch, that sucks |
| 23:39.15 | perd | yeah i suppose that's true, factoring in the costs of TDM2400s .. but if i ran voip phones i would need to rewire the whole place |
| 23:39.33 | perd | they've already got telephone wire running to all the rooms/houses |
| 23:39.37 | dijungal | hi guys, is there anyway to disable a span on the TE410P ? |
| 23:39.54 | perd | the cost of cabling and time would offset the value of voip phones imo |
| 23:39.54 | dijungal | can i just comment out the span in zaptel.con? |
| 23:40.52 | hypa7ia | hmm |
| 23:41.06 | hypa7ia | perd: you could give them fast busy if all lines were busy |
| 23:41.07 | hypa7ia | on analog |
| 23:41.38 | perd | yeah it's easy enough to make a recording that says 'sorry priest dude, your lines are in use by extension 452 and 233' |
| 23:41.51 | hypa7ia | lol, that works too |
| 23:41.54 | [TK]D-Fender | hypa7ia, Sure you can, PLENTY less |
| 23:42.22 | perd | put on pantera for the hold music |
| 23:42.28 | hypa7ia | lorf |
| 23:42.39 | hypa7ia | [TK]D-Fender: what comment are you responding to? |
| 23:43.52 | [TK]D-Fender | hypa7ia, no good phones < 150$ |
| 23:45.23 | hypa7ia | [TK]D-Fender: you misread me |
| 23:45.40 | hypa7ia | <PROTECTED> |
| 23:45.47 | perd | now.. if there were some 802.11 voip phones that were cheap... but they'd have to be under 100 bucks and i doubt that will happen :) |
| 23:45.55 | hypa7ia | haha no, definitely not |
| 23:46.01 | [TK]D-Fender | hypa7ia, Indeed, oops |
| 23:46.17 | hypa7ia | perd: i've been pretty happy with an IAXy and a dual-handset cordless |
| 23:46.43 | JT | mobile 802.11 sucks for voip |
| 23:46.57 | hypa7ia | yup |
| 23:48.49 | perd | what's an iaxy, pardon my ignorance |
| 23:48.56 | Mavvie | everything with non-predictable characteristics sucks for any realtime protocol. |
| 23:49.03 | perd | oh nm i see it on voip-info |
| 23:49.09 | perd | that is one ugly sob. |
| 23:49.16 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 23:49.28 | hypa7ia | haha the old one was pretty ugly |
| 23:49.33 | perd | it looks like a punky brewster inspired old belkin device |
| 23:50.39 | perd | what is more cpu intensive, a analog phones with the DSP on board or voip using ulaw? |
| 23:51.07 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 23:51.27 | fujin | Which CPU? |
| 23:51.28 | fujin | :) |
| 23:51.45 | perd | oh i dont know, an amd x2 64 2.4ghz |
| 23:51.58 | fujin | and where is this cpu plugged in? |
| 23:52.03 | perd | to my FACE |
| 23:52.03 | fujin | your question wasn't very specific.. |
| 23:52.10 | fujin | cool. |
| 23:52.27 | hypa7ia | sounds painful |
| 23:52.33 | fujin | lol |
| 23:52.34 | perd | oh, well i'm thinking two TDM2400Ps with the noise cancellation on the board verses sip with ulaw |
| 23:52.43 | fujin | 939 pins into your head |
| 23:52.50 | perd | i can take it. |
| 23:53.07 | perd | it would increase my iq to triple digits i bet.. |
| 23:53.25 | fujin | perd: your question doesn't make sense, are the phones registered to a server with the amd in it, running asterisk? |
| 23:53.33 | fujin | are you doing packet2packet briding, or are endpoints talking to eachother |
| 23:54.29 | JT | perd: your question doesn't make sense |
| 23:54.33 | perd | well then |
| 23:54.36 | perd | forget about it. |
| 23:54.38 | JT | both don't involve compressed codecs |
| 23:54.52 | perd | i understand that, but there still must be some cpu overhead? |
| 23:54.59 | JT | sure |
| 23:55.07 | JT | you can assume it's identical |
| 23:55.23 | JT | but maybe sip is fractions of a hair less cpu usage intensive |
| 23:55.41 | perd | and that answers my question. thank you sir |
| 23:55.42 | hypa7ia | but who cares, if you're just running 20 phones :) |
| 23:55.50 | JT | conversion from slin to g.711 |
| 23:55.51 | perd | i know, i was just curious |
| 23:55.56 | JT | considering ec is handled by the card |
| 23:56.34 | perd | and of those 30 phones only two can actually be in use for external calls, the rest could listen to vm at most... |
| 23:57.03 | JT | make sure you store vm in an uncompressed codec if you're that worried about cpu then ;) |
| 23:57.17 | perd | oh i'm not worried about cpu, i was just curious. |
| 23:57.27 | perd | im sure a 500mhz box could handle the load for this setup |
| 23:57.33 | JT | right |
| 23:58.38 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 23:59.14 | fujin | heh, I have all of the vm stuff transcoded to ALAW and I still get weird clicky chop when people play voicemail |
| 23:59.19 | fujin | I've given up trying to diagnose it |
| 23:59.23 | fujin | no load on the asterisk box |
| 23:59.29 | fujin | only thign I can think of is that it's the phones |
| 23:59.35 | JT | packet loss? |
| 23:59.39 | perd | i have that with voip too, using x-lite, cisco phones, anything |
| 23:59.40 | fujin | nope :\ |
| 23:59.52 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 23:59.57 | perd | jumpy weird audio |