| 00:00.34 | WIMPy | NFI about a2b. And we have not really seen a question so far. |
| 00:02.00 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:02.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-igucirkubtoyngwx) |
| 00:03.08 | filo1234 | pdtpatrick1: jitsi |
| 00:03.49 | filo1234 | pdtpatrick1: I remember that was multi protocol |
| 00:05.28 | fiesch | I'll head off, need to get the legacy pbx wired back up for tomorrow.. thanks again WIMPy and have a good one.. |
| 00:07.36 | *** part/#asterisk fiesch (d95c43dc@gateway/web/freenode/ip.217.92.67.220) |
| 00:09.21 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 00:10.40 | KavanS | please share suggestions: How to obtain a quality australian DID from the U.S. via an ITSP, possible? |
| 00:14.35 | SeRi | p3nguin: you in? |
| 00:14.37 | *** join/#asterisk jpsharp (~jsharp@74-95-145-82-Naples.hfc.comcastbusiness.net) |
| 00:18.39 | *** join/#asterisk timahvo1 (~rogue@41.90.67.41) |
| 00:18.48 | pdtpatrick1 | Question .. has anyone successfully been able to send message from Jabber to asterisk? mine always causes asterisk to crash. |
| 00:19.11 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 00:23.52 | chigambamukoko | KavanS: you looking for just 1 DID from DownUnder? |
| 00:25.57 | KavanS | chigambamukoko, yes sir |
| 00:26.00 | KavanS | just need one |
| 00:26.21 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 00:31.27 | SeRi | iptables -L ADAPTIVE_BAN_CHAIN |
| 00:31.31 | SeRi | damn |
| 00:31.33 | SeRi | lol |
| 00:32.29 | jpsharp | You're supposed to "Wrong Window" your credit card info. |
| 00:32.45 | SeRi | Wrong Window :P |
| 00:32.55 | SeRi | lol |
| 00:33.25 | WIMPy | root on one line and some funny word on the next is ok as well. |
| 00:35.35 | SeRi | lol |
| 00:37.11 | s[X] | hey SeRi |
| 00:38.14 | s[X] | How long did IANA take to get back to you |
| 00:40.10 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 00:42.05 | SeRi | s[X]: It takes a few days |
| 00:42.23 | s[X] | hah you were the first one on my new growl notification |
| 00:42.27 | s[X] | Neato :P |
| 00:42.49 | s[X] | Its only been 2 days |
| 00:43.03 | s[X] | ill give it a bit longer :P |
| 00:44.01 | SeRi | yes it will take some time. |
| 00:44.22 | SeRi | I think the whole process eincluding the mistake that freenum made took about 1 1/2 weeks |
| 00:44.25 | s[X] | I contacted samsung, they have yet to test it lol |
| 00:44.39 | SeRi | about? |
| 00:44.55 | s[X] | ITAD |
| 00:45.23 | SeRi | oooo lol |
| 00:45.54 | s[X] | They said they were unaware of anyone using it yet, but would look into trialing it if i were to require it |
| 00:46.17 | SeRi | what type of pbx do you have digital? |
| 00:46.23 | s[X] | Samsung 7200s |
| 00:46.28 | s[X] | all IP Phones |
| 00:46.49 | s[X] | 8 Channel's for SIP and 2 PSTN Backup lines |
| 00:46.52 | SeRi | o cool... Thats hybrid right? |
| 00:47.01 | s[X] | yeah |
| 00:47.03 | s[X] | its a piece of shit |
| 00:47.07 | SeRi | lmao! |
| 00:47.22 | s[X] | The stupid thing doesnt quite know what it is |
| 00:48.27 | s[X] | Ive been talking to irroot and im thinking of putting in a * box on my ESXi server (Vmware) and have the 7200s connect to the * box and let the * box handle the sip stuff |
| 00:49.09 | SeRi | Sounds like a plan to me! |
| 00:49.28 | s[X] | It does handle sip relatively well, its just not perfect |
| 00:49.43 | s[X] | Every now and again itll go tits up and ill loose inbound audio on incomming calls |
| 00:49.55 | s[X] | Wiresharked the hell outta it for weeks |
| 00:50.42 | SeRi | :/ |
| 00:50.43 | s[X] | ITSP captures show that it hits our WAN router, my captures show we dont receive it |
| 00:50.52 | jpsharp | * on ESXi works well. I'm running it right now. |
| 00:50.58 | SeRi | qos? |
| 00:51.14 | s[X] | Tried 4 diffrent routers, DMZ's etc etc etc |
| 00:51.37 | s[X] | irroot says its a common problem on that model |
| 00:51.43 | s[X] | it doest NAT Properly |
| 00:51.44 | billy_ran_away | Does asterisk 10 support asterisk-gui? |
| 00:51.45 | s[X] | doesnt* |
| 00:52.00 | s[X] | jpsharp: im a huge fan of esxi |
| 00:53.01 | jpsharp | I wish I had learned about it before I did. I couldn't have crunched a rack full of machines into one. |
| 00:53.09 | jpsharp | er, I could have. |
| 00:53.23 | pdtpatrick1 | Is there a way to see all installed apps that start with jabber? or related to jabber? |
| 00:53.49 | pdtpatrick1 | show applications i guess would be it |
| 00:56.13 | s[X] | im still running esxi 4.1 |
| 00:56.18 | s[X] | but its faultless |
| 00:56.45 | s[X] | My only wish is a native OSX Vmware Viclient |
| 00:57.47 | SeRi | how can I add a time condition? I want to make my phone ring only between 7AM and 10PM. |
| 00:58.05 | SeRi | mhhhh I think I got it |
| 01:00.38 | jpsharp | I have a windows VM that serves no other purpose than to to manage the ESXi installation. Had to borrow a windows machine to set the system up to begin with, though. |
| 01:00.54 | s[X] | I run a Virtual Win7 under Vmware Fusion |
| 01:01.06 | s[X] | which has ViClient & few other Windows bits n pieces |
| 01:01.21 | s[X] | jpsharp, What happens if ur Windows VM crashes ? |
| 01:01.22 | pdtpatrick1 | can i do something like this? |
| 01:01.23 | pdtpatrick1 | exten => s, n, NoOp(${JABBER_RECEIVE(asterisk,bob@domain.com)}) |
| 01:01.30 | pdtpatrick1 | based on .. https://wiki.asterisk.org/wiki/display/AST/Function_JABBER_RECEIVE |
| 01:02.32 | jpsharp | s[X]: Power cycle the VM via the ESXi ssh shell. If that doesn't work, then I beg the wife to let me use her Windows box to poke around. |
| 01:03.13 | jpsharp | Fortunately, when Windows eats itself and dies, it seems to be restarting on its own. |
| 01:03.51 | s[X] | you on OSX aswell ? |
| 01:04.14 | billy_ran_away | What's the best way to manage users? |
| 01:04.23 | jpsharp | Linux mostly. |
| 01:04.28 | billy_ran_away | Is there a way to just do local users accounts? |
| 01:04.32 | s[X] | Considered Virtual Box ? |
| 01:06.20 | jpsharp | had, but the need hasn't come up yet. Though the VM does seem to crash regularly because my RDP client seems to tickle some sort of bug, I haven't had to do any major disaster recovery. |
| 01:08.09 | s[X] | Ive got mine with custom snapshot backup script |
| 01:11.57 | SeRi | ok I thought I had it but I dont. |
| 01:12.24 | SeRi | How can I dail thre ext at the same time but one only between 7AM and 10PM? |
| 01:12.38 | SeRi | s/dail/ring/ |
| 01:14.28 | s[X] | jpsharp, do you backup ur vm's ? |
| 01:15.27 | jpsharp | Not yet. I have a LTO2 drive in the works. |
| 01:16.47 | s[X] | with ESXi theres a script called GhettoVCB |
| 01:16.53 | s[X] | Which is awesome |
| 01:17.54 | jpsharp | looks...makes notes. |
| 01:18.15 | s[X] | If ur running the free version of ESXi theres no way to do hot backups |
| 01:18.31 | s[X] | unless u script it to snapshot the volume |
| 01:18.37 | s[X] | thats what GhettoVCB does |
| 01:19.18 | SeRi | any body? :) |
| 01:21.27 | jpsharp | Use the time-based contexts? |
| 01:22.07 | jpsharp | time-based context includes, rather. |
| 01:22.55 | SeRi | jpsharp: I tried that and that works for a whole group of ext |
| 01:23.00 | [TK]D-Fender | "core show application gotoiftime" |
| 01:23.22 | SeRi | not to ring 3 extensions at the same time excluding one at surten hours |
| 01:23.52 | SeRi | Thanks TK |
| 01:24.52 | SeRi | perfect |
| 01:24.58 | SeRi | what I was looking for :) |
| 01:26.24 | florz | but beneatheleven? |
| 01:32.47 | billy_ran_away | No matching peer found... |
| 01:34.35 | billy_ran_away | What's the difference between entries in sip.conf and users.conf? |
| 01:35.07 | SeRi | [TK]D-Fender: http://pastebin.com/bzwQ3aNm |
| 01:35.14 | SeRi | Like that? |
| 01:35.20 | SeRi | would that work? |
| 01:35.25 | SeRi | well I think it would. |
| 01:35.28 | SeRi | :) |
| 01:35.56 | [TK]D-Fender | nope |
| 01:36.01 | billy_ran_away | Test? |
| 01:36.07 | [TK]D-Fender | read your parms & targets again real close |
| 01:36.36 | SeRi | ooo lol |
| 01:36.43 | SeRi | for got a few things like Dial :P |
| 01:37.06 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:39.25 | SeRi | [TK]D-Fender: Thanks I fixed the .s and add it Dial |
| 01:39.35 | SeRi | s/.s/,s/ |
| 01:40.44 | billy_ran_away | Can someone please point me to a link or something that explains the differences between users.conf and sip.conf? |
| 01:40.52 | billy_ran_away | It seems like sip.conf is for users... |
| 01:41.03 | billy_ran_away | But why does Asterisk-GUI create them in users.conf? |
| 01:41.54 | [TK]D-Fender | billy_ran_away: users.conf was an attempt to mash extensions.conf, sip/iax/etc w/e voicemail configs all into one file. It is only really used by the GUI and does everything in a very linear limited way |
| 01:42.11 | [TK]D-Fender | billy_ran_away: There is virtually no point in using it unless you're using Asterisk GUI |
| 01:42.31 | billy_ran_away | So if I am shouldn't I expect the users in users.conf to be able to log in? |
| 01:42.41 | [TK]D-Fender | billy_ran_away: Think of it as a lesser sum of its parts |
| 01:42.48 | billy_ran_away | [TK]D-Fender: I really appreciate the response! |
| 01:43.03 | [TK]D-Fender | billy_ran_away: don't think of it as "users". Its' jsut device configs w/ VM sprinkled in |
| 01:43.25 | billy_ran_away | [TK]D-Fender: Okay, so it's an attempt by the creators of to wrap a extensions, sip, and other bs into one entry? |
| 01:43.27 | [TK]D-Fender | It also generates live dialplan when loaded depending on directives issued |
| 01:43.32 | [TK]D-Fender | billy_ran_away: Yes |
| 01:43.48 | billy_ran_away | [TK]D-Fender: Well I have the asterisk-gui installed… why am I getting no matching peer found? |
| 01:43.54 | billy_ran_away | I had type=peer... |
| 01:44.01 | billy_ran_away | so I need peer in extensions or sip? |
| 01:44.06 | [TK]D-Fender | billy_ran_away: The only key nice thing about it is thte GUI knows how to parse it out so you can do direct mods to the text file and the GUI will pick them up |
| 01:44.22 | billy_ran_away | that is nice |
| 01:44.29 | [TK]D-Fender | billy_ran_away: Are you running *GUI now? What do you want? This is key... |
| 01:44.48 | billy_ran_away | I want to be able to login and authenticate with a softphone... |
| 01:45.11 | [TK]D-Fender | Sounds like normal registering of a SIP device. |
| 01:45.15 | [TK]D-Fender | Anything will do... |
| 01:45.21 | [TK]D-Fender | What about the longer term? |
| 01:45.39 | billy_ran_away | Longer term I want to have users using soft phones to do multi-video conferencing |
| 01:45.39 | [TK]D-Fender | 6 lines in a config file can give you peer you can register to |
| 01:45.56 | billy_ran_away | I guess I'm confused why I'm getting no matching peer found... |
| 01:46.03 | [TK]D-Fender | * won't mix video. I did answer that for you earlier |
| 01:46.21 | billy_ran_away | Won't? According to Asterisk 10 there is support... |
| 01:46.42 | [TK]D-Fender | No, not "mixing". "Follow the speaker" in app_conference". I told you that specifically as well. |
| 01:47.04 | SeRi | [TK]D-Fender: Thanks. I fix it all. it now works |
| 01:47.08 | billy_ran_away | Yea follow the speaker will work |
| 01:47.11 | [TK]D-Fender | not simultaneous multi-cast |
| 01:47.18 | [TK]D-Fender | SeRi: You're welcome |
| 01:47.33 | billy_ran_away | Do you know of better solution for self hosted multi video chat? |
| 01:47.39 | billy_ran_away | Supporting screen sharing is a plus... |
| 01:47.46 | [TK]D-Fender | billy_ran_away: Ok, well you know the rules on that. I would recommend ditching the GUI and starting from scratch with a very basic config |
| 01:48.00 | billy_ran_away | XMPP/Jabber doesn't seem to support that... |
| 01:48.14 | jpsharp | realtime! |
| 01:48.28 | billy_ran_away | jpsharp: You have a url? |
| 01:49.04 | jpsharp | No, realtime is the database driven configuration for asterisk, not a multiparty conference. |
| 01:49.07 | jpsharp | Sorry for the confusion. |
| 01:51.27 | billy_ran_away | Well thanks for your help [TK]D-Fender |
| 01:51.37 | billy_ran_away | Really do appreciate it |
| 01:54.55 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 01:56.56 | vader-- | do any of you know of any free SIP trunks i could use for testing? |
| 01:57.36 | [TK]D-Fender | For testing what? |
| 01:57.51 | vader-- | inbound/outbound, etc |
| 01:58.05 | vader-- | dialplans, ringroups, etc |
| 01:58.06 | [TK]D-Fender | Invalid "test" |
| 01:58.36 | [TK]D-Fender | You could fake whatever you want and see that it issues the dial you think it should. |
| 01:59.09 | [TK]D-Fender | Testing inbound is a matter of setting the context for any device like a SIP phone to your inbound context |
| 02:14.59 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 02:21.23 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-knctwxwzqakdrdtg) |
| 02:21.23 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 02:21.59 | *** join/#asterisk master_of_master (~master_of@p57B544AF.dip.t-dialin.net) |
| 02:23.58 | Maliuta | anyone able to take a look at http://pastebin.com/TexN72As for me ... for some reason the voicemail isn't working |
| 02:24.08 | *** join/#asterisk TheCops (~mdb@2001:470:1d:5b3:20c:29ff:fee7:997e) |
| 02:25.09 | Maliuta | not sure why, in other extensions the messages get played back, but from that context it just comes up with a "Voicemail off. Security tone please." message |
| 02:26.50 | Maliuta | on * 1.8.7.1 |
| 02:32.41 | [TK]D-Fender | Where's the failure? |
| 02:33.35 | SeRi | It also looks like he is using a remote vmail... maybe another asterisk doing voicemail for him... |
| 02:34.03 | SeRi | ops never mind wrong window |
| 02:34.30 | SeRi | :/ need to get use to irssi :P |
| 02:35.11 | *** join/#asterisk mintos (~mvaliyav@117.206.20.197) |
| 02:36.23 | dijib | quit |
| 02:39.12 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 02:39.59 | Maliuta | [TK]D-Fender: in the s and 61731363065 contexts. The gotoiftime works fine, passes things through so the Dial() works ... but if there is no answer the voicemail() call doesn't send things to the mailbox |
| 02:40.33 | Maliuta | SeRi: no, the vmail is on the same box |
| 02:40.48 | SeRi | Maliuta: Wrong window. |
| 02:40.55 | SeRi | that was meant for somebody else |
| 02:40.56 | SeRi | <PROTECTED> |
| 02:40.59 | SeRi | sorry! |
| 02:41.12 | Maliuta | just clarifying incase someone mistook my situation |
| 02:41.32 | SeRi | coincidence though :/ |
| 02:42.13 | Maliuta | I have seen something like this before ... where if I takeaway any of the branching statements things work as they should, but with them the voicemail system doesn't work as advertised |
| 02:42.36 | jpsharp | Just a quick look, but maybe the whole == versus = thing? |
| 02:44.55 | Maliuta | jpsharp: hand't thought of that, testing now |
| 02:45.12 | Maliuta | and fail |
| 02:45.49 | Maliuta | getting the same msg "Answering Machine off. Security tone please. (then a series of beeps)" |
| 02:47.30 | jpsharp | Are you using custom audio messages? That's not an Asterisk message I recognize. Are you sure the calls are actually landing on *? |
| 02:47.47 | jpsharp | just checking the obvious before delving into the obscure. |
| 02:50.23 | [TK]D-Fender | Maliuta: I mean... where is the failure for us to SEE? |
| 02:50.24 | Maliuta | No custom messages, standard comedian mail. And yes the calls are coming in on the asterisk box ... there is no other termination |
| 02:51.25 | jpsharp | pastebin the * console outbut as you try to place a call through. |
| 02:51.32 | jpsharp | and the output, too. |
| 02:57.05 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 02:59.30 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 03:00.04 | Maliuta | [TK]D-Fender: I have a sip debug of one of the calls, but it's no help |
| 03:00.59 | Maliuta | [TK]D-Fender: how do you suggest I get some visible output of the error? since it seems to be invoking the vm system but no reffering the call to the correct mailbox? |
| 03:01.58 | SeRi | waz up p3nguin |
| 03:02.52 | Maliuta | my voicemail.conf is at http://pastebin.com/9CNhpbuh |
| 03:03.38 | [TK]D-Fender | Maliuta: Show us the failed call |
| 03:21.47 | Maliuta | http://pastebin.com/yDzYBwYA |
| 03:22.04 | Maliuta | would have been faster to do if I could have simply uploaded a file :( |
| 03:24.14 | [TK]D-Fender | That is only SIP debug. You are trying to fix your dialplan and that is the one thing you are not showing in there |
| 03:24.38 | [TK]D-Fender | SIP isn't the issue here if its hitting your dialplan and simply not processing the way you'd like |
| 03:24.44 | [TK]D-Fender | New call please.... |
| 03:27.06 | SeRi | me I take a number please? |
| 03:27.24 | SeRi | s/me I/may I/ |
| 03:27.38 | SeRi | :P |
| 03:28.26 | Maliuta | [TK]D-Fender: I have core debug set to 9, and dialplan debug seems to be useless |
| 03:28.38 | Maliuta | [TK]D-Fender: what do you recommend I do? |
| 03:28.55 | Maliuta | (to get a proper debug output) |
| 03:29.49 | SeRi | Maliuta: core set verbose 3 shows what your dial plan is doing. |
| 03:30.05 | SeRi | I think thats what [TK]D-Fender is looking for. not debug. |
| 03:30.41 | [TK]D-Fender | VERBOSE |
| 03:30.46 | [TK]D-Fender | 10 <- |
| 03:30.54 | [TK]D-Fender | core & SIP are not needed |
| 03:31.41 | SeRi | I see. |
| 03:33.06 | SeRi | [TK]D-Fender: is it possible to call another phone sort of like an intercom? I want to call the kids room and activate the speaker phone... is that possible? |
| 03:33.47 | SeRi | They have a el chepo granstream |
| 03:33.52 | [TK]D-Fender | Many SIP phones have auto-answer features precisely for this sort of thing |
| 03:33.55 | Maliuta | [TK]D-Fender: http://pastebin.com/9ZZNq8Vt |
| 03:34.07 | SeRi | [TK]D-Fender: Thanks |
| 03:35.15 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 03:35.44 | [TK]D-Fender | Maliuta: Nothing wrong with dialplan or processing. Everything is happning perfectly normal. No voicemail should be called. |
| 03:37.54 | s[X] | SeRi: is that so you can tell them to shut the hell up or else |
| 03:38.57 | SeRi | lol. So I can wake them up in the morning, called them for dinner, etc.... |
| 03:39.53 | SeRi | What I see is that the auto answer does just that :P auto answers so I can not send the main line calls to that phone because if I do they will get answered :) |
| 03:40.31 | SeRi | Mhhhhhhhh maibe ill just dont have it ring for the main line |
| 03:40.46 | Maliuta | gah! fecking stupid base station for the handset ... even with the bloody answering machine off it pick up somewhere between 25 and 30 seconds |
| 03:41.25 | [TK]D-Fender | malNice when you notice what it is that is actually answering your calls :) |
| 03:41.32 | SeRi | Maliuta: rofl! lol! did it burn? :P |
| 03:41.42 | Maliuta | I had set the Dial()to go for 30 seconds to give me time to get to a handset, look like I'll have to live with 25 |
| 03:41.50 | Maliuta | SeRi: it will. it will |
| 03:42.17 | SeRi | Maliuta: been there :) happen to me with a stupid panasonic phone. |
| 03:42.47 | Maliuta | SeRi: if you hear that parts of suburban Brisbane went up in flames ... that was probably me destroying the P.O.S. with a little too much vigour |
| 03:42.58 | SeRi | hahahaha |
| 03:43.28 | s[X] | Brisbane hey Mailuta |
| 03:43.33 | Maliuta | I make a mean anti-personnel napalm ... this might call for a batch of nitro though |
| 03:43.34 | s[X] | Mailuta* |
| 03:43.44 | s[X] | son of a cunt |
| 03:43.48 | s[X] | Maliuta* |
| 03:43.53 | Maliuta | s[X]: yes I live in Brisneyland |
| 03:44.09 | s[X] | I myself live on the Sunshine Coast |
| 03:44.31 | Maliuta | and you're on internode ethernet? which DC is that in? |
| 03:44.55 | s[X] | Who said i was on Internode Ethernet ? |
| 03:45.25 | Maliuta | your whois |
| 03:45.37 | s[X] | my whois says something about eth it also says something about adsl |
| 03:46.07 | Maliuta | meh, I just had them change the reverse dns on mine :P |
| 03:46.19 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:46.21 | s[X] | They are useless |
| 03:47.00 | Maliuta | I actually like 'node. they have good support |
| 03:47.27 | s[X] | Yeah i liked them untill they told me my exchange was overcapacity and well tough shit.... I know its not their fault but still i wouldve liked a bit more action |
| 03:48.50 | *** join/#asterisk jpsharp (~jsharp@74-95-145-82-Naples.hfc.comcastbusiness.net) |
| 03:49.03 | Maliuta | I don't have an agile DSLAM near me, and the Heltsra line for my naked has a fault 4km from here ... my speeds are shite but there is nothing 'node can do because they lease the copper from Optarse, who lease from Hellstra |
| 03:49.17 | Maliuta | awaits the NBN at his place |
| 03:49.27 | s[X] | awaits the NBN for his unborn children |
| 03:50.26 | s[X] | I cant even get Naked at mine and im on a RIM *Winning* |
| 03:50.43 | SeRi | well the kids phone does not work! LOL |
| 03:50.45 | Maliuta | I could just move my servers to my uncles place a Dutton Park. Since the knocking down of the old South Brisbane exchange they have FTTH and for what I pay now I could be getting 100Mbs synchro |
| 03:51.11 | SeRi | any recomendations for a cheap sip phone with speaker? |
| 03:52.06 | Maliuta | SeRi: I only use an old cisco 7942, you can get them pretty cheap these days |
| 03:52.37 | SeRi | Thats to much for a 4,9, and 10 year old |
| 03:53.48 | Maliuta | SeRi: but you might be able to get it to auto-answer |
| 03:53.59 | SeRi | True. |
| 03:54.37 | *** join/#asterisk tekoholic (~tekoholic@97-118-239-237.hlrn.qwest.net) |
| 03:55.10 | Maliuta | I need to find a replacement for my cordless. thinking something DECT->SIP might be nice |
| 03:55.42 | s[X] | Maliuta, You have many servers at home ? |
| 03:55.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:56.28 | Maliuta | s[X]: does my mythtv frontend count as a server? |
| 03:56.41 | s[X] | I think so |
| 03:56.49 | s[X] | Ill let it count |
| 03:57.25 | Maliuta | s[X]: In that case it's 3 servers one desktop and a netbook for a total of 12 cores |
| 03:57.35 | *** join/#asterisk irroot (~gregory@197.106.4.99) |
| 03:57.42 | s[X] | hey Irroot |
| 03:57.53 | Maliuta | that's just my stuff, my house mate runs his desktop which is another 4 cores |
| 03:58.02 | irroot | hi there |
| 03:58.04 | s[X] | Maliuta: Ive got a Clipsal 45RU in my garage |
| 03:58.33 | Maliuta | I have a 1/3 rack sitting here waiting for me to do up the patch panel |
| 03:58.34 | SeRi | irroot: good morning |
| 03:58.48 | tekoholic | Can anybody help debug this error? Google has very little related, and none of the solutions work in my environment... "ERROR[27684]: res_jabber.c:3123 aji_initialize: JABBER ERROR: No DNS for client to talk.google.com" |
| 03:58.59 | irroot | indeed 6am counts as morning |
| 03:59.07 | SeRi | :) |
| 03:59.15 | s[X] | Dayum, im still very much asleep at that time |
| 03:59.30 | irroot | good day to you folks whateva da time :P |
| 03:59.35 | Maliuta | tekoholic: can you get dns resolution from that host? i.e. with 'host talk.google.com' |
| 03:59.37 | s[X] | 2pm for me |
| 03:59.56 | Maliuta | s[X]: in 8 seconds |
| 04:00.03 | s[X] | wise ass |
| 04:00.18 | Maliuta | s[X]: use ntp and atomic clocks :P |
| 04:00.23 | tekoholic | Maliuta: Yes, I can, from the very router that asterisk runs upon, as well as all hosts it NAT's for... |
| 04:00.35 | s[X] | I dont have seconds in osx |
| 04:01.05 | Maliuta | s[X]: you're running Fisher Price My First *nix? |
| 04:01.08 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 04:01.10 | tekoholic | Maliuta: I even changed the talk.google.com reference in jabber_users.conf to the IP address, still same error |
| 04:01.21 | Maliuta | s[X]: what linux distro fo you use? My First Debian? |
| 04:01.35 | irroot | ROFLMA @ Maliuta |
| 04:02.03 | s[X] | CentOS & Ubuntu mainly |
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| 04:02.17 | irroot | is a regular luke skywalker and uses the source |
| 04:02.26 | Maliuta | irroot: apparently one of my friends is using the "My First Debian" line now ... although i told him he needs to give me credit :) |
| 04:02.54 | Maliuta | s[X]: so wannabe DeadRat and My First Debian then :) |
| 04:03.03 | Maliuta | <-- Debian snob |
| 04:03.05 | s[X] | You south african ? |
| 04:03.26 | Maliuta | but I do my own packaging |
| 04:03.34 | s[X] | Congratulations ? |
| 04:03.35 | SeRi | s[X]: Ubuntu is an African word for "I am scared of slackware" |
| 04:03.40 | Maliuta | no. Born at the Mater Mothers |
| 04:04.00 | irroot | Ubuntu is "brotherhood" or "comunity" |
| 04:04.15 | Maliuta | SeRi: I thought it was "I'm too pussy for Debian - but I'll steal from them anyhow" |
| 04:04.15 | SeRi | irroot: I know. you messed the joke up! |
| 04:04.23 | SeRi | Maliuta: ROFL! |
| 04:04.26 | irroot | and mark grants brother is a real party animal and good drinking bud |
| 04:04.31 | tekoholic | Maliuta: I can't be sure, but I don't think it's really a DNS error, in the sense that we're looking at it. Wondering if DNS has a different meaning, in / with asterisk? |
| 04:04.45 | irroot | mark is a bit of a prick but never said that :P |
| 04:04.59 | SeRi | Ok I found a Polycom 321 for 70 dollars free shipping. |
| 04:05.03 | SeRi | I think taht would do. |
| 04:05.04 | s[X] | I thought it was Mark Shuttleworth |
| 04:05.06 | Maliuta | tekoholic: not normally. I don't use jabber, so I doubt I can be of much use |
| 04:05.48 | irroot | s[x] thats the one his brother mark lived 2 houses over the drunkard :P |
| 04:05.50 | tekoholic | Maliuta: OK, thanx anyway. Anyone else? |
| 04:06.59 | s[X] | Maliuta, what do you for a living ? |
| 04:10.56 | ChannelZ | tekoholic: can you resolve talk.google.com yourself? |
| 04:11.13 | irroot | ChannelZ yo yo what up |
| 04:11.28 | ChannelZ | hey irroot, same ol same ol |
| 04:12.30 | tekoholic | ChannelZ: Yes, from all hosts on my network. Asterisk runs on OpenWRT Attitude Adjustment, on a wndr3700v2 router, with public IP, and dns servers to spare |
| 04:12.44 | ChannelZ | tekoholic: nevermind I see that you can. Make sure you don't have some invisible character or something on your serverhost= line |
| 04:14.22 | tekoholic | ChannelZ: I have, and there are none. I've completely uninstalled and reinstalled, and use the luci pbx app to config, and it just stops working with this error until I reflash the router. |
| 04:15.22 | irroot | things like , instead of . have caused many to reinstall bind |
| 04:15.38 | ChannelZ | oh actually looking at the source, you have something missing somewhere perhaps. |
| 04:16.06 | ChannelZ | ast_log(LOG_ERROR, "JABBER ERROR: No DNS %s for client to %s\n", client->name, S_OR(client->serverhost, client->jid->server)); |
| 04:16.37 | ChannelZ | client->name is empty, whatever that is.. hmm |
| 04:17.16 | tekoholic | name isn't empty in the error, at least not on mine, I just removed it from the paste job... |
| 04:17.39 | ChannelZ | oh.. that was useful |
| 04:19.07 | ChannelZ | in any event it's an error from iksemel, not sure what the exact significance of it is.. the library unable to resolve one would assume, but why, couldn't say |
| 04:19.22 | tekoholic | It's typically the gmail address, and sits as ...NO DNS gtalk-username_gmail_com for client... |
| 04:20.02 | Maliuta | s[X]: right now I'm a professional patient (on disability). But when I'm working I'm a *nix Sysadmin who dabbles in System architecture |
| 04:20.26 | s[X] | Ah ok |
| 04:20.58 | tekoholic | ChannelZ: OK, so why might it work, until suddenly it doesn't? It fails at random, seemingly... Usually on reboot |
| 04:21.13 | ChannelZ | no idea |
| 04:21.29 | s[X] | You guys getting by much rain in brissy today ? |
| 04:21.32 | s[X] | getting hit* |
| 04:22.55 | *** join/#asterisk leroux (~leroux_@108-79-20-223.lightspeed.hstntx.sbcglobal.net) |
| 04:24.41 | leroux | Is it possible to call 'exec agi test2.agi' from test1.agi which has been called from extensions.conf and have a variable from test2.agi returned to test1.agi for use. |
| 04:24.42 | leroux | ? |
| 04:25.07 | leroux | I'm not even sure if 'exec agi <agi script> works, I am just wondering if anything of this type is possible. |
| 04:25.37 | JunK-Y | why not call your functions from test2.agi directly from test1.agi ? |
| 04:25.40 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:25.42 | [TK]D-Fender | leroux: Don't see why not. And all * vars are equally available |
| 04:25.54 | [TK]D-Fender | leroux: Though JunK-Y |
| 04:25.55 | irroot | leroux possibly would work but dont think it be a good idea |
| 04:26.10 | [TK]D-Fender | leroux: Though JunK-Y's point about directly calling it is a good one |
| 04:26.22 | leroux | If I call test2.agi from test1.agi, can test2.agi communicate with the Asterisk server? |
| 04:26.26 | irroot | im using php and have files i "include" with functions in |
| 04:26.54 | leroux | Well, what I am doing is building a C framework that should be able to call an AGI script of any lang. |
| 04:26.59 | Maliuta | s[X]: not here |
| 04:27.05 | leroux | Through the use of #!/bin/env <lang> mostly. |
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| 04:27.40 | Maliuta | s[X]: looks like some is heading my way though |
| 04:27.48 | leroux | And we'd rather not be dependant on 'set extension' for this. |
| 04:27.57 | leroux | [TK]D-Fender, any comments? |
| 04:28.02 | s[X] | Maliuta: yeah we are supposed to be getting a rather big storm this arov aswell nothing yet though |
| 04:28.11 | leroux | JunK-Y, you? |
| 04:28.36 | [TK]D-Fender | leroux: Each agi can access vars just the same |
| 04:29.03 | leroux | I don't understand how that would work. I'll explain what I'm trying to do and how. |
| 04:29.14 | JunK-Y | leroux: not too sure to understand your "framework", since agi can be any language natively with * |
| 04:29.24 | leroux | Yes. |
| 04:29.27 | leroux | I understand that. |
| 04:29.34 | Maliuta | s[X]: from the look of the current radar image within an hour I'll have a bit of rain and they will probably have to call play at the 'Gabba for the day |
| 04:29.37 | leroux | Look. |
| 04:30.17 | leroux | The thing is, if I call an *.agi script from an initial agi script, will the output from the agi script being called contact asterisk? |
| 04:30.48 | JunK-Y | test1.agi and test2.agi are both in the same language? |
| 04:30.56 | leroux | No. C and PHP. |
| 04:31.01 | leroux | respectively. |
| 04:31.04 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 04:31.23 | leroux | So I want a var from test2.agi to be returned to test1.agi. |
| 04:31.43 | JunK-Y | so, you want AGI(test1.agi) and in that C agi call test2.agi, which is written in php, right? |
| 04:31.52 | leroux | Yes. |
| 04:32.14 | JunK-Y | i would say its your "framework" to manage this. |
| 04:32.51 | leroux | Well, the framework is called from extensions.conf as AGI(frmk); |
| 04:32.58 | leroux | frmk is test1.agi. |
| 04:33.01 | leroux | Assume this. |
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| 04:34.46 | JunK-Y | leroux: i dont know, that should be pretty easy to test with your 2 AGIs |
| 04:35.07 | leroux | Alright, thanks. |
| 04:35.18 | JunK-Y | from test2.agi, just do a set variable and see if you have a return from test1 when calling GET variable |
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| 04:35.35 | JunK-Y | leroux: let me know ur results in pv, im curious |
| 04:35.41 | leroux | Yeah, will do. |
| 04:36.02 | leroux | If you want to come help us out, I'll send you the irc server and site link. |
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| 05:11.07 | s[X] | hey irroot u still there bub ? |
| 05:11.10 | s[X] | lol |
| 05:11.11 | s[X] | bud* |
| 05:11.26 | irroot | im here |
| 05:11.33 | s[X] | Was curious as to the 7200s -> * box with regards to extensions on the * box |
| 05:11.35 | irroot | will be leaving in bit |
| 05:11.54 | s[X] | ie: Extension on the 7200s dialing an extension on the * box |
| 05:12.19 | s[X] | and vice versa |
| 05:12.34 | irroot | s[X] that is little tricky easiest way is to add a new prefix for that range |
| 05:13.30 | irroot | there other methods on samsung too |
| 05:13.53 | irroot | if you dial 0 for a trunk line you can dial 0<EXTEN> to get * |
| 05:13.53 | s[X] | we Dial 0 get the LCR |
| 05:14.11 | irroot | * -> 7200 is in extensions.conf |
| 05:14.32 | irroot | Dial/..../<exten>,....) |
| 05:14.56 | s[X] | ah ok im with ya |
| 05:14.57 | irroot | s[x] you can create a LCR access of anything to do this |
| 05:15.31 | irroot | its also possible to add a extension forward or map a exten onto a LCR table |
| 05:15.56 | s[X] | We just might a few warehouse phones onto the * box |
| 05:15.59 | s[X] | add* |
| 05:16.22 | irroot | lol thats how it starts ...then in 3 months you turn the samsung off |
| 05:17.09 | s[X] | I dont doubt that for a second its just we have 12 IP Handsets |
| 05:18.34 | irroot | s[x] well they expensive thats all :P nothing too fancy about them easy to replace |
| 05:20.01 | s[X] | yeah im sure the boss will love the fact we have to get rid of them |
| 05:21.11 | irroot | s[x] indeed i can see how that discussion goes |
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| 05:46.16 | psilikon | I am looking for the easiest way possible to concatenate the files used in the sayunixtime application for the current unixtime. Can I just pass a time to asterisk somehow and get a list of the files it would play? |
| 05:49.56 | p3nguin | If you increase the core verbose, it will show you what files are playing as they play. |
| 05:52.12 | psilikon | p3nguin, i know and I could just get my agi script to grab those return results and add them to an array it would be awesome |
| 05:53.05 | psilikon | I = if |
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| 06:03.21 | kikohnl | Wow, what a day, migragted our company from a 15 year old NEC PBX to Asterisk 10-rc2! |
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| 06:50.25 | s[X] | Whats the deal with "Free" g729 codecs |
| 06:50.26 | s[X] | lol |
| 06:51.04 | ChannelZ | well the deal is that they're 1. illegal and 2. probably non-functional |
| 06:51.17 | s[X] | Figured as much |
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| 06:53.25 | ChannelZ | YAY! New Canada's Worst Driver! |
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| 08:21.24 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 08:23.24 | ChannelZ | Crisco! |
| 08:26.40 | ollii | heyho |
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| 08:52.03 | s[X] | yo all |
| 08:53.14 | zkn | Hi, i've more of a general question, and I know this depends on many things, yet i'm curious what do "Asterisk consultants" get paid for their services these days, say by the hour? |
| 08:53.18 | *** join/#asterisk irroot (~gregory@197.172.126.33) |
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| 09:04.18 | zkn | what would be a fair pay to ask, should some company turn to you requesting to build them a typical Asterisk based PBX system from ground up that includes configuring Asterisk, writing the dial plan for ACD, voicemail, CFWD, CDR, IVR, call monitoring-recording and web interfacing capabilities with AMI |
| 09:05.12 | zkn | i know the above are vague terms and can require different amount of work and effort based on how to solve it according to the needs of the customer |
| 09:06.07 | gordonjcp | zkn: well, start with a decent hourly rate, and work out how long it's going to take you |
| 09:06.31 | zkn | ok, but is there some general rate ? |
| 09:07.47 | zkn | i would work freelance on this |
| 09:09.09 | zkn | so, I wonder what others have charged / would charge for this |
| 09:09.26 | zkn | i'm sure someone has experience |
| 09:09.52 | angryuser | Someone from sangoma here ? |
| 09:09.54 | *** join/#asterisk cuco (~cuco@mail.bio-guard.net) |
| 09:10.00 | cuco | tzafrir: ping |
| 09:10.39 | tzafrir | cuco, pong |
| 09:11.14 | cuco | tzafrir: wazzzzzzzzzzzzzzzzzzzzzzzaaaaaaaaaaaaaaaaaaaaaaaaaaappppppppppppp |
| 09:11.31 | *** join/#asterisk fazool (~ada_has@182.185.185.85) |
| 09:11.31 | tzafrir | All's well, basically |
| 09:11.37 | tzafrir | fazool, hi |
| 09:11.42 | s[X] | hey mofo's |
| 09:11.53 | cuco | tzafrir: non related question, are you using vtiger there? |
| 09:12.01 | tzafrir | yes |
| 09:12.08 | fazool | hello |
| 09:12.20 | fazool | <PROTECTED> |
| 09:12.22 | cuco | tzafrir: can i ask you some questions, in #vtiger? (off topic here) |
| 09:12.36 | fazool | i want to tranfser call withing agents |
| 09:12.56 | s[X] | What is everyone up 2 |
| 09:12.57 | singler | angryuser: just ask your question, maybe someone will answer, sangoma's support is rechable in #sangoma from 9am to 5pm EST |
| 09:13.19 | angryuser | singler, sure |
| 09:13.36 | fazool | but i am still getting agent call |
| 09:14.51 | angryuser | Ubuntu server 10, wanpipe driver do not detect any missing libs, but compilation fails at the end (tools & api) log here http://pastebin.com/3C6NEeLc |
| 09:14.56 | fazool | anyone help me? |
| 09:15.11 | angryuser | Same for debian |
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| 09:20.05 | like_a_horse | greets, anyone know how to do the following: person calls certain extension (say 2000) then it rings for 5 rings and then diverts to a different phone (say 3000) |
| 09:20.16 | like_a_horse | heard of ring groups and call groups |
| 09:20.38 | like_a_horse | is this something thats straight forward to do? |
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| 09:27.08 | singler | angryuser: can you pastebin output before the one you pastebined? |
| 09:27.32 | kaldemar | like_a_horse: set a timeout in seconds in the first dial command to 2000 and put another dial command in the following priority. |
| 09:28.00 | like_a_horse | kaldemar, aha...thanks |
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| 09:41.48 | gavimobile | hope im not off of the topic here, but I cannot get my softphone app to register with my sip server when behind my router. all other phones are working with the public ip address both in front and behind my router. my softphone app will register only if I point it to the local servers address or if I use the public address on a different network.. where might my problem be? |
| 09:43.01 | angryuser | singler, pasting all the process in a min |
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| 09:48.09 | kaldemar | gavimobile: sip server as in asterisk or something else? |
| 09:48.58 | angryuser | singler, here it is http://pastebin.com/XKsA2Gkn all the process |
| 09:49.29 | angryuser | singler, all is going well till the end :( the feature frozen driver 3.4.9 crashes so i use the latest |
| 09:50.09 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 09:50.25 | angryuser | gavimobile, maybe a port forwarded somewhere elso |
| 09:51.05 | gavimobile | angryusers, ports 5060 is forwarding to my sip server so it 10000-20000 |
| 09:51.38 | gavimobile | but I see in the logs of the sip application that its trying to use 9083 for some reason |
| 09:51.50 | *** join/#asterisk screenn (~screenn@178.251.111.71) |
| 09:52.19 | angryuser | singler, /usr/include/wanpipe has all the headers |
| 09:53.07 | singler | angryuser: can you check if is there somewhere any compile log? |
| 09:54.26 | angryuser | singler, yes, i will paste the part with error |
| 09:55.33 | angryuser | singler, but in 2 words, entering the directory « /usr/src/wanpipe-3.5.24/api/libsangoma » |
| 09:56.02 | angryuser | singler, unable to find makefile, stop |
| 09:56.47 | angryuser | singler, so it quits the directory and throwing Error 2 |
| 09:57.01 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 09:57.42 | singler | oh, I guess sangoma didn't bundle something with new release, you could try to modify Makefile to not traverse there, because you do not need libsangoma for dahdi operation |
| 09:58.57 | singler | in about 4 hours sangoma's support will be online, they will be able to assist you better with this problem |
| 09:59.17 | angryuser | singler, i will try to compile the tools alone |
| 09:59.30 | angryuser | singler, there is a readmo, who knows ;) |
| 09:59.44 | angryuser | *readme |
| 10:00.19 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 10:00.36 | kaldemar | gavimobile: sip server as in asterisk or something else? |
| 10:01.46 | gavimobile | kaldemar: asterisks with freepbx |
| 10:02.09 | kaldemar | ~freepbx |
| 10:02.09 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 10:02.42 | angryuser | singler, i think i found hte lacking package, autoconf |
| 10:03.07 | gavimobile | kaldemar: freepbx should have anything to do with this networking issue. shall I take it to #networking? |
| 10:03.21 | kaldemar | gavimobile: we can tell you how to configure sip with NAT involved in asterisk, but freepbx will most likely just override the changes. |
| 10:03.22 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 10:03.58 | kaldemar | gavimobile: nothing so fat suggests that you have a networking issue. |
| 10:04.01 | gavimobile | kaldemar: so you don't believe this is a networking issue? |
| 10:04.04 | angryuser | singler, setup assumes that it exists, and do not check it, will try to recompile |
| 10:04.09 | kaldemar | s/fat/far/ |
| 10:04.33 | gavimobile | kaldemar: ok! thanks for the help. |
| 10:04.36 | *** join/#asterisk schmick1 (~schmick1@60-242-87-233.static.tpgi.com.au) |
| 10:04.39 | gavimobile | I will take it up with freepbx |
| 10:05.37 | *** part/#asterisk schmick1 (~schmick1@60-242-87-233.static.tpgi.com.au) |
| 10:05.51 | gavimobile | kaldemar: could you also tell me how to configure sip with NAT involved in asterisks? I think your explination would be enough for me to figure it out with freepbx |
| 10:05.58 | gavimobile | unfortunetly freepbx is sleeping now |
| 10:07.26 | kaldemar | ~sipnat |
| 10:07.27 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
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| 10:08.29 | gavimobile | kaldemar: thanks buddy! |
| 10:08.47 | angryuser | singler, yep, it compiles now |
| 10:09.36 | kaldemar | gavimobile: you're welcome |
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| 10:18.34 | *** join/#asterisk ik_5 (~ik@109.226.17.43) |
| 10:19.27 | ik_5 | hello, i'm with asterisk 1.8, and sometimes on an "h" extension the AGI command sends sighup to my agi, eventhough the channel already disconnected. what might be wrong here ? |
| 10:21.50 | *** part/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 10:23.12 | tzafrir | ik_5, hi |
| 10:23.31 | tzafrir | When eaxctly is sighup sent? After hangup? |
| 10:24.28 | ik_5 | tzafrir, the call is already supposed to be disconnected, because it entered the "h" extension |
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| 10:33.23 | ik_5 | tzafrir, the weird part, is that it does not enter the full log (that is set to debug as well). I can't see it at all when it happens |
| 10:33.47 | tzafrir | Maybe it got delayed for some odd reason? |
| 10:35.17 | ik_5 | tzafrir, the hangup signal ? |
| 10:35.46 | tzafrir | yes |
| 10:36.18 | ik_5 | tzafrir, so how did it arrive to the "h" extension ? |
| 10:37.12 | tzafrir | The trace should show you that, I guess |
| 10:37.23 | tzafrir | Any chance it actually got there with a GoTo? |
| 10:37.40 | ik_5 | tzafrir, no way |
| 10:46.24 | ik_5 | found it on the full log, but it seems that the hangup happens before the call to the agi, and after the call to the agi as well, i don't get it |
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| 11:03.42 | ik_5 | i think that the problem is the bridge, the first channel is zombie |
| 11:09.56 | angryuser | Anyone from sangoma support here ? |
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| 11:52.24 | [sr] | damn it's cold |
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| 12:25.43 | beek | angryuser: try #sangoma |
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| 13:01.39 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 13:02.20 | jacc0 | hi all |
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| 13:34.25 | ollii | jacc0: o/ |
| 13:44.39 | jacc0 | :) |
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| 14:21.23 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
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| 14:22.46 | anonymouz666 | p3nguin: why you say that all the time? |
| 14:24.25 | [TK]D-Fender | scripting of course... |
| 14:24.31 | [TK]D-Fender | Silly rabbit... |
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| 14:26.00 | *** join/#asterisk _omer (~omer@119.152.118.96) |
| 14:26.16 | _omer | hello .... any help ?? http://pastebin.com/Y9NQpsKP ?? |
| 14:27.09 | [TK]D-Fender | Broken pipe errors are usually AGI's generating noise when they shouldn't. |
| 14:27.31 | [TK]D-Fender | And that post describes as much |
| 14:27.58 | _omer | yes but I couldn't understand the post ,, I mean how to do that in code ... |
| 14:28.14 | _omer | I am not "PRINT"ing anything in Perl |
| 14:28.39 | [TK]D-Fender | We don't see your code. |
| 14:28.52 | _omer | ok let me paste it . |
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| 14:35.29 | _omer | http://pastebin.com/GMmU5j81 <---- when call hangs up .. then Script delete data from database [&deletedata() function] .... there I receive this error ... but I dont get this error when CPU load is low .. |
| 14:36.02 | _omer | for example 50 concurrent calls.....I get Error ....... if 10 concurrent calls.... No error |
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| 14:41.50 | _omer | [TK]D-Fender : any help ??? |
| 14:49.49 | SeRi | sipbri what a bunch of clowns. |
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| 14:50.03 | puzzled | morning |
| 14:50.11 | SeRi | g/m |
| 14:53.14 | tompaw | I know it's kinda OT, but... is there anyone here with some AS5300 experience - in particular using trunk-groups with PRI controllers? |
| 14:54.07 | WIMPy | tompaw: I used a bunch of them to sit on when we were short of chairs in the office. |
| 14:54.43 | tompaw | WIMPy: did you use trunk-group on their PRI controllers to make them more comfortable to sit on? |
| 14:55.42 | _omer | I also used them not for sitting but shitting purposes |
| 14:55.55 | tompaw | That's interesting. |
| 14:56.03 | WIMPy | I'm pretty sure they were configured for trunk groups (if you talking NFAS), but that did not make them more comfortable. |
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| 14:56.15 | tompaw | :( |
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| 15:04.29 | jacc0 | @cisco guy: I'm still waiting for a security patch for the huge security hole in the cisco SPA phone; it has been months sinds it was reported and cisco is no longer replying to my mail |
| 15:04.54 | jacc0 | It's about time I'll release an exploit |
| 15:05.23 | tompaw | what hole? sip related? |
| 15:06.18 | jacc0 | you can push an XML to the phone using the default username password sdkapps:sdkapps |
| 15:06.24 | jacc0 | the password can not be changed |
| 15:06.44 | jacc0 | you have total controle over the phone by pushing the XML |
| 15:07.14 | [TK]D-Fender | Polycom > All |
| 15:07.16 | [TK]D-Fender | :) |
| 15:07.17 | jacc0 | you can make a key-macro that can emulate any key-press; you could even reset the admnin password |
| 15:07.29 | jacc0 | maybe they will fix it sooner |
| 15:08.15 | WIMPy | Probably not. That would break backwards compatibility. They will just document it. |
| 15:08.31 | jacc0 | if you would look into it more you will notice that it isn't even checking login credentials: you could use any username and password |
| 15:09.08 | jacc0 | just push the xml to : http://SPA-IP/CGI/Execute |
| 15:09.36 | tompaw | sexy, but they're probably going to assume that spa is to be used in corporate environment, and the network protections should take care of that (as in - prevent anyone from accessing :80 on spa) |
| 15:09.48 | jacc0 | it wil not |
| 15:09.56 | jacc0 | because of other exploits |
| 15:10.11 | jacc0 | like xss using the caller id name |
| 15:10.24 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:10.28 | jacc0 | when opening the call-history list in the webinterface it will get executed |
| 15:10.42 | tompaw | It has G729 and "24" ringtone! What else do you expect!! |
| 15:11.03 | jacc0 | so if you are able to send a malisous callerid you can controle the phone |
| 15:11.11 | jacc0 | :) |
| 15:11.16 | jacc0 | now it's public |
| 15:11.17 | CVirus | I get a segmentation fault when trying to connecto asterisk using asterisk -rvvvvv http://pastie.org/2949885 |
| 15:11.25 | jacc0 | I hope cisco will come with an update soon |
| 15:11.43 | jacc0 | tell it to any criminal you know!!! |
| 15:12.04 | jacc0 | this is a zero-dya |
| 15:12.08 | jacc0 | *zero-day |
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| 15:13.49 | tekoholic | jacc0: What are you talking about, zero-day and criminals? |
| 15:14.50 | jacc0 | ciminals could make a lot of money |
| 15:15.37 | jacc0 | it seems that it is unpatched |
| 15:15.49 | jacc0 | cisco know about it for months |
| 15:16.02 | jacc0 | so maybe not a zero day |
| 15:16.25 | tekoholic | Oh, on the callerid bug? Well, I'm glad I don't run Cisco! |
| 15:16.49 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:17.13 | jacc0 | on being able to push a key-macro xml to the spa phone |
| 15:17.42 | jacc0 | no usename or password needed |
| 15:17.56 | jacc0 | on by default |
| 15:18.48 | jacc0 | or better: every username and password will do |
| 15:18.56 | jacc0 | just push the xml to : http://SPA-IP/CGI/Execute |
| 15:19.22 | jacc0 | XML=<CiscoIPPhoneExecute><ExecuteItem Priority=\"0\" URL=\"Softkey:Exit\"/><ExecuteItem Priority=\"0\" URL=\"Softkey:Exit\"/></CiscoIPPhoneExecute> |
| 15:19.33 | jacc0 | that is an example of how to push the exit button |
| 15:20.12 | jacc0 | you can make the xml push any button you like |
| 15:20.20 | jacc0 | you can even reset passwords |
| 15:20.22 | WIMPy | So you send a few keypresses to transfer callers to your 900 number? |
| 15:20.33 | jacc0 | yes |
| 15:20.35 | jacc0 | for sure |
| 15:20.41 | *** join/#asterisk radic (~radic@dslb-094-216-250-020.pools.arcor-ip.net) |
| 15:20.44 | WIMPy | Nice |
| 15:20.48 | jacc0 | try it |
| 15:21.05 | jacc0 | I've been testing it on a Cisco SPA525g |
| 15:21.15 | jacc0 | have funn all |
| 15:21.18 | jacc0 | :P |
| 15:21.25 | jacc0 | is always pushing for a change |
| 15:21.32 | WIMPy | So if you find an Asterisk thet won;t do that for you, yu can still hope to find phones that will. |
| 15:22.01 | jacc0 | I've made an asterisk application to do it for you |
| 15:22.10 | jacc0 | app_cisco_spa_push.c |
| 15:22.20 | jacc0 | I'll be releasing it soon |
| 15:22.44 | jacc0 | it's been over 6 months sinds the bug was reported |
| 15:23.06 | WIMPy | If I have access to the Asterisk it's connected to, why would I need access tot he phone? |
| 15:23.39 | jacc0 | I use it to show data on the screen when a call comes in |
| 15:24.28 | jacc0 | nice graphics :) |
| 15:24.44 | jacc0 | you could als display video stream and more |
| 15:25.56 | jacc0 | change ringtone ( to a spoken announcement) |
| 15:26.03 | jacc0 | or other funny things |
| 15:26.17 | jacc0 | :) |
| 15:26.41 | jacc0 | you can also make the phone auto pickup the call |
| 15:27.40 | jacc0 | use your imagination |
| 15:27.42 | *** join/#asterisk psharmor (~tekoholic@97-118-192-128.hlrn.qwest.net) |
| 15:27.54 | leifmadsen | sounds like a feature |
| 15:27.59 | jacc0 | hehehehe |
| 15:28.26 | jacc0 | yeah it is; the only thing is it is missing the feature to configure your own username and password |
| 15:28.35 | jacc0 | :P |
| 15:28.54 | jacc0 | to prevent others from using this nice feature |
| 15:29.11 | jacc0 | lol |
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| 15:32.51 | WIMPy | Don't be so selfish. |
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| 15:33.47 | WIMPy | The video thing has been done on the Snom370. |
| 15:35.11 | jacc0 | hehehe |
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| 15:40.38 | jacc0 | hi irroot |
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| 15:42.00 | dijib | court today boys wish me luck |
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| 15:42.27 | gordonjcp | dijib: heh, good luck |
| 15:42.32 | WIMPy | What did you do? |
| 15:42.52 | dijib | :) |
| 15:42.56 | gordonjcp | has only been in court once, relating to a protest over bridge tolls |
| 15:42.59 | gordonjcp | got a Not Proven |
| 15:43.02 | dijib | thats the funny thing, i didnt do anything |
| 15:43.07 | dijib | wanted my privacy. |
| 15:43.28 | WIMPy | Privacy? So you're a terrorist? |
| 15:43.37 | dijib | yes |
| 15:43.49 | WIMPy | too |
| 15:45.52 | Qwell | I went to a courthouse once. They had metal detectors, and wouldn't let me bring in my scissors. I said screw that and went home. |
| 15:46.34 | Qwell | (I used to carry scissors. Because, why not?) |
| 15:46.38 | tzanger | ... why did you bring scissors to a courthouse? |
| 15:46.42 | WIMPy | Did you have to leave those scissors before you were allowed to go? |
| 15:46.54 | Qwell | WIMPy: I would have, had I opted to go in. |
| 15:47.02 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:47.16 | WIMPy | But not to get home? |
| 15:47.27 | Qwell | tzanger: You'd be surprised how useful scissors can be when you carry them all the time. |
| 15:47.38 | Qwell | WIMPy: I left when they said I couldn't take them in. |
| 15:47.56 | gordonjcp | Qwell: I went to the local court to drop off some paperwork, and didn't realise I didn't actually need to go into the metal-detectored bit |
| 15:48.01 | Qwell | tzanger: it's like if I ever quit smoking, there's no chance in hell that I'd not continue carrying a lighter. |
| 15:48.01 | WIMPy | In Denmark, carrying a cardboard cutter will end you in jail, unless you can give a good reason for carrying one. |
| 15:48.23 | WIMPy | But they have accepted that that was way over limit. |
| 15:48.24 | gordonjcp | Qwell: metal detector went ape, possibly related to my steel-toecap, steel-soleplate, steel-heelguard boots |
| 15:48.48 | gordonjcp | my work boots weigh about 3kg each ;-) |
| 15:48.49 | Qwell | gordonjcp: ugh.. I always forget about my belt at the airport. |
| 15:49.06 | gordonjcp | heh, I've flown with these in my bag |
| 15:49.17 | *** join/#asterisk AdamN (~AdamN@63.230.70.254) |
| 15:49.26 | gordonjcp | only ever had one comment, "f-ing h-, those are some f-ing serious boots, what the f- do you do?" |
| 15:51.02 | jacc0 | I once got arrested for having a lock-pick-set in my car |
| 15:51.29 | jacc0 | :P |
| 15:51.57 | tzanger | Qwell: sure, they're handy, but I can't be bothered to pack a scissors and lighter with me whereever I go. |
| 15:52.11 | Qwell | You sir, are missing out. |
| 15:52.15 | tzanger | Qwell: I keep reducing the number of things I carry |
| 15:52.37 | tzanger | my wallet is down to a single-fold with exactly 4 cards in it. my keys are car+house only, and my phone. that's about it |
| 15:52.44 | WIMPy | I used to always carry a Leatherman until it got nicked. |
| 15:53.14 | Qwell | WIMPy: Leatherman would be extremely useful too. The problem with those is the stupid belt clip thing. |
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| 15:53.34 | *** join/#asterisk timahvo1 (~rogue@41.80.84.105) |
| 15:53.42 | Qwell | same with phones... if you wear your phone on your belt, I quietly judge you. |
| 15:53.42 | tzanger | Qwell: how do you carry a scissors without getting jabbed? do you have a carry case for it or something? |
| 15:53.51 | Qwell | tzanger: very carefully. |
| 15:54.12 | Qwell | You pretty much nailed the reason that I no longer carry them. :p |
| 15:54.17 | tzanger | Qwell: haha |
| 15:54.38 | tzanger | (and I keep typing wq<tab> and wondering why it's not completing. do you mind changing your nick to wqwell? :-) |
| 15:54.58 | wq | aww |
| 15:55.04 | Qwell | no : in nicks |
| 15:55.08 | tzanger | qwell: hahahah now I'm tying qw<tab> to complet |
| 15:55.11 | tzanger | it's a losing battle |
| 15:55.44 | WIMPy | Far too many keys on those keyboards. |
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| 16:00.53 | Qwell | tzanger: apparently some stupid clients will tab complete backwards. |
| 16:01.15 | Qwell | People type a sentence like "well, today I ...foo", and it'll complete the "well" to my nick. O.o |
| 16:08.50 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:14.15 | *** join/#asterisk jozza (~chatzilla@BSN-77-186-224.dsl.siol.net) |
| 16:14.34 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 16:15.06 | jozza | hi, what does asterisk take as did number in the incoming invite message? |
| 16:15.52 | jozza | my to: looks like to: "12345 6789"<sip:123@bla.com> |
| 16:16.20 | jozza | it always says its looking for s in my-context |
| 16:16.31 | p3nguin | It doesn't take the DID number... unless the DID number is the extension the call is sent to. |
| 16:16.37 | jozza | why is it not looking for 123? |
| 16:17.08 | jozza | i cant seem to find a variable that contains 123 |
| 16:17.22 | p3nguin | The extension is apparently 's' in your case. Pastebin the sip debug of the entire call. |
| 16:17.27 | p3nguin | There's no variable. |
| 16:17.31 | jozza | ok |
| 16:18.45 | jozza | http://pastebin.com/ZPTNFqg8 |
| 16:19.59 | p3nguin | Increase you core verbose to 3 and do it again. I need to see the extensions running as well. |
| 16:20.08 | jozza | oh i can see the problem now |
| 16:20.26 | jozza | invite has s as extension |
| 16:20.46 | jozza | so this came from provider in this form right? |
| 16:20.53 | p3nguin | Yeah, first line... but why is the To: different? |
| 16:21.01 | jozza | i i have no idea |
| 16:21.07 | p3nguin | Yes, that came from the provider. |
| 16:21.12 | jozza | ok |
| 16:23.13 | jozza | but can i get the did from the to header somehow |
| 16:23.22 | jozza | until the provider fixes the invite message? |
| 16:23.35 | Qwell | You're telling the provider to use s. |
| 16:23.38 | Qwell | Fix your register => line |
| 16:23.53 | jozza | how do i do that? |
| 16:24.05 | p3nguin | sip.conf register statement |
| 16:24.09 | jozza | i have a format: usename:password@domain.com |
| 16:24.19 | Qwell | the /extension at the end |
| 16:24.21 | [TK]D-Fender | usename:password@domain.com/whatyouwantinsteadofs |
| 16:24.26 | jozza | i can try |
| 16:24.48 | p3nguin | This is pretty useless when you have more than one phone number, though. |
| 16:25.25 | *** join/#asterisk irroot (~gregory@197.170.148.167) |
| 16:26.15 | jozza | i have many |
| 16:26.25 | [TK]D-Fender | jozza, http://www.voipcoop.org/viewtopic.php?p=209#209 |
| 16:26.30 | jozza | hod do i access the To: header? |
| 16:26.40 | [TK]D-Fender | jozza, Easy mod to that link |
| 16:26.52 | p3nguin | hmm |
| 16:26.55 | Qwell | [TK]D-Fender: what...have you done? |
| 16:27.10 | p3nguin | Try using exten => s,1,Goto(${SIP_HEADER(TO):5:10},1) |
| 16:27.27 | jozza | yes |
| 16:27.34 | [TK]D-Fender | Qwell, What now ...? |
| 16:27.43 | Qwell | You linked n3glv. >.> |
| 16:27.47 | [TK]D-Fender | oops |
| 16:27.48 | p3nguin | Put it in the same context as your extensions/DIDs. |
| 16:27.50 | Qwell | as...the solution for something. |
| 16:27.55 | Qwell | ...a valid one, even. |
| 16:28.23 | *** join/#asterisk radic (~radic@dslb-094-216-250-020.pools.arcor-ip.net) |
| 16:28.41 | [TK]D-Fender | There used to be a good link.. that was CID mod. wasted junk |
| 16:28.49 | [TK]D-Fender | Qwell, LOL :p |
| 16:29.04 | [TK]D-Fender | Qwell, There was a good link in that chan's bot for this... this wasn't it though |
| 16:29.08 | jozza | yes, no problem there, just need to srape the number from the header somehow |
| 16:29.14 | jozza | until the provider fixes this |
| 16:29.20 | p3nguin | (1027.06) <p3nguin> Try using exten => s,1,Goto(${SIP_HEADER(TO):5:10},1) |
| 16:29.26 | [TK]D-Fender | ^^ |
| 16:29.31 | jozza | i'll see now what comes out when i register with my leading extension |
| 16:29.52 | jozza | gotta sit a the lunch table... |
| 16:29.55 | p3nguin | They will either send every call to that number, or they will ignore it and keep sending to s. |
| 16:30.09 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 16:30.44 | jeffspeff | what's the trick to using ExtenSpy() to spy on specific channels? |
| 16:30.56 | p3nguin | ExtenSpy() spies on extensions. |
| 16:30.57 | jeffspeff | excuse me, extensions |
| 16:31.05 | p3nguin | For channels, use ChanSpy. |
| 16:31.09 | ChannelZ | you had it right the first time |
| 16:31.10 | jeffspeff | p3nguin, you caught it before it did |
| 16:31.18 | p3nguin | ExtenSpy(some-extension@context) |
| 16:31.20 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 16:31.29 | Qwell | core show application extenspy ? |
| 16:31.50 | jeffspeff | p3nguin, i tried that, and it seems to grab random extens in my context |
| 16:31.57 | p3nguin | Show me. |
| 16:32.23 | p3nguin | ExtenSpy(a-very-explicit-extension@context) |
| 16:32.34 | p3nguin | It shouldn't randomly choose something else. |
| 16:32.58 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:33.43 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 16:36.34 | jeffspeff | p3nguin, http://pastebin.com/i00XsynW |
| 16:38.29 | [TK]D-Fender | jeffspeff, Please show us a channel list ebfore you attempt that |
| 16:38.35 | jeffspeff | p3nguin, exten 1000 is a valid exten, and that made me spy on kmurphy... exten 963 is not a valid exten, and that made me spy on kmurphy. kmurphy was not the only one in a call |
| 16:39.21 | klynn | sup fender |
| 16:39.42 | jeffspeff | [TK]D-Fender, the channels change as people get on and off the phone calls. it won't be accurate if i run the command now |
| 16:40.10 | [TK]D-Fender | then do a new test. |
| 16:40.16 | [TK]D-Fender | list + spy |
| 16:40.25 | [TK]D-Fender | I never expected to keep that old PB |
| 16:40.36 | jeffspeff | ok, just a min |
| 16:41.57 | p3nguin | Also, without defining the context, ExtenSpy is using default. Why would default have all these calls in it? |
| 16:42.41 | jeffspeff | p3nguin, when i tried specifying the context, it wouldn't work at all |
| 16:42.57 | p3nguin | You have to define the context where the extensions are. Remember, phones are not extensions. |
| 16:43.22 | p3nguin | ChanSpy spies on the phones, ExtenSpy spies on extensions that phones have called. |
| 16:43.58 | jeffspeff | so, it looks at my outbound context? |
| 16:44.04 | jeffspeff | for outbound calls |
| 16:44.05 | p3nguin | ExtenSpy takes the extension and turns it into the channel which has called the extension on which you are spying. |
| 16:44.26 | p3nguin | If you are trying to spy on the extensions using for outbound calls, that's the correct context. |
| 16:44.45 | p3nguin | You'd then spy on the number the phone has called, such as 3145551212. |
| 16:45.07 | Nugget | telnet miku.acm.uiuc.edu |
| 16:45.17 | p3nguin | Or, if you want to spy on the phone rather than the extension, use ChanSpy. |
| 16:45.25 | Qwell | Nugget: Telnet is eeeeeevvil! |
| 16:45.26 | jozza | p3nguin: you are right, they are sending all numbers to the leading one now, hehe |
| 16:45.29 | Nugget | hee |
| 16:45.35 | Nugget | in this case, doubly do |
| 16:45.37 | Nugget | er so |
| 16:46.09 | p3nguin | jozza: Change it back so the calls go to s. Then use my extension s that I pasted. |
| 16:47.06 | jozza | thats, exactly what i'll do, thx |
| 16:48.05 | p3nguin | And then talk the ITSP into fixing it so calls go to the number instead of s. |
| 16:48.19 | *** join/#asterisk coppice (~chatzilla@host86-136-94-225.range86-136.btcentralplus.com) |
| 16:50.12 | jozza | i will, but its Serbia, they are still learning asterisk |
| 16:50.17 | jozza | :)) |
| 16:51.06 | [TK]D-Fender | jozza, Your ITSP is using * ? |
| 16:51.34 | jozza | i dunno, why would be a s extension in the invite then? |
| 16:51.48 | [TK]D-Fender | jozza, because your register told them to <- |
| 16:52.10 | jozza | ok |
| 16:53.01 | jozza | my register actualy told them nothing, but if this is the case in sip protocol then i couldnt say if they are really using ast |
| 16:53.14 | p3nguin | You can't tell them to send calls to multiple numbers in the register statement. |
| 16:53.22 | jozza | sure |
| 16:53.35 | [TK]D-Fender | <jozza> i have a format: usename:password@domain.com |
| 16:53.39 | p3nguin | You can either accept them to whatever they send them to, or you can register using a single number. |
| 16:53.46 | [TK]D-Fender | <[TK]D-Fender> usename:password@domain.com/whatyouwantinsteadofs |
| 16:53.46 | jozza | so it's correct from me to register without the extension |
| 16:53.59 | jozza | yes |
| 16:54.01 | [TK]D-Fender | jozza, Put nothing on the end and * will tell them "s" |
| 16:54.05 | p3nguin | [tk]d-fender: Useless for multiple DIDs. |
| 16:54.13 | jozza | i did now |
| 16:54.17 | [TK]D-Fender | p3nguin, dependson the provider. |
| 16:54.27 | jozza | and i'll be cutting the to header as it has the correct information |
| 16:54.29 | p3nguin | Registering with a single phone number sends all calls to that single number, even when they are FOR another number. |
| 16:54.41 | [TK]D-Fender | p3nguin, Some providers are just useless, ignore whatever you send them and jsut slap it in the "to:" :) But we've just been through that |
| 16:55.16 | p3nguin | He tested it using a single phone number, and all calls to other DIDs went to the number he registered with. |
| 16:55.27 | p3nguin | So s is actually better. |
| 16:55.32 | Kobaz | time to call tech support for the provider |
| 16:55.35 | p3nguin | Then parse the header. |
| 16:56.10 | p3nguin | Every once in a while someone pops in with a provider doing this crap. |
| 16:56.24 | [TK]D-Fender | p3nguin, Certainly stands out when the call comes in. |
| 16:57.09 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 16:57.52 | p3nguin | VoIP.ms has a setting in the portal for the customer to choose what kind of equipment he uses. You can choose between ATA or IP PBX. If you choose ATA, all calls to all DIDs are sent to s. If you use Asterisk/IP PBX, all calls are sent to the extension which is the DID. |
| 16:58.22 | [TK]D-Fender | p3nguin, They sound pretty smart... |
| 16:58.43 | p3nguin | I thought it was a good idea. |
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| 17:00.37 | jozza | man, but how do i cut the string? |
| 17:00.42 | jozza | i need to search it |
| 17:00.49 | jozza | should i use system? |
| 17:01.26 | p3nguin | I already showed you. Twice. |
| 17:01.34 | jozza | there are 3 different cases |
| 17:01.38 | p3nguin | How? |
| 17:01.43 | p3nguin | s is s is s is s. |
| 17:01.59 | jozza | one is "number number"<sip:number@123.com |
| 17:02.16 | jozza | one is "name"<tel.:number@123.com |
| 17:02.21 | [TK]D-Fender | jozza, "core show function CUT" |
| 17:02.27 | jozza | ok |
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| 17:02.51 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:03.40 | jozza | i am going to cut the number before the at sign and after the colon |
| 17:03.47 | jozza | then all cases should work out |
| 17:05.59 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:15.09 | MrTelephone | Does anyone know what this means? [2011-11-30 16:26:41] WARNING[17750] sig_pri.c: Span 1: Digit '2' may be ignored by peer. (Call level:3) |
| 17:16.22 | WIMPy | Why and how is it sent and what's the call state? |
| 17:17.21 | MrTelephone | Me? |
| 17:17.39 | WIMPy | yes |
| 17:17.57 | MrTelephone | I don't even know how to check. Why couldn't the log specify a channel? |
| 17:18.32 | WIMPy | So you have no idea about the circumstances? |
| 17:18.57 | MrTelephone | It really shouldn't do that? I can check the CDR to see what device may have sent digit 2 |
| 17:19.29 | WIMPy | I haven't seen that message so far. |
| 17:19.41 | MrTelephone | I know it's wierd. has something to do with overlap dial |
| 17:20.01 | MrTelephone | I have a lot of DTMF issues so I'm trying to dig dipper |
| 17:20.39 | WIMPy | Could it be DTMF being forwarded out of band? |
| 17:22.53 | WIMPy | If so it will probably not do what you want. |
| 17:23.04 | MrTelephone | not sure. It's certain phone systems we call. They don't recognize the dtmf sometimes. When they don't sometimes sending the dtmf 2 or 3 more times will work |
| 17:23.08 | WIMPy | Just as the message suggests. |
| 17:23.34 | WIMPy | What are you sending from? |
| 17:23.48 | MrTelephone | any sip endpoint outgoing on the pri |
| 17:24.41 | WIMPy | YOu need to send DTMF inband to the PSTN it you want it to reach the peer. What DTMF mode do you use on the SIP side? |
| 17:24.52 | MrTelephone | I might have to take a raw audio sample off the pri to see wtf is going on. |
| 17:25.27 | MrTelephone | rfc2833 on the inside. chan_dahdi should be forwarding that inband. Is there a setting I should double check? |
| 17:26.10 | WIMPy | Yes, that's how it should work. I'm not sure I've seen a setting for that in chan_dahdi. |
| 17:27.01 | MrTelephone | yeah it should work. I mean it does on most phone systems. I find it's the worst using old nortel PBX's. Calling them or calling us is not reliable |
| 17:27.08 | WIMPy | But that message suggests it's sending out of band. OTOH, if it works sometimes, maybe it's sending in-band as well. |
| 17:27.35 | MrTelephone | I didn't even know PRI supports out of band dtmf. I should see if there is a setting to disable it |
| 17:27.58 | WIMPy | I don't see any DTMF setting. |
| 17:28.10 | WIMPy | Yes, you can send out of band. |
| 17:28.17 | beek | Does anyone here know of instances where Shoretel and Asterisk have coexisted? Can Shoretel peer with Asterisk? I have the latter and am told we're going to install the former. |
| 17:28.28 | Qwell | it's just SIP, right? |
| 17:28.32 | WIMPy | But that will only get to the switch. These messages are not forwarded to the other party. |
| 17:28.57 | beek | Qwell: That's what I'd like to think but I know absolutely nothing about Shoretel. |
| 17:29.12 | MrTelephone | Neither do I. I see priindication but that is only for busy/congestion. |
| 17:29.22 | MrTelephone | something is crapping the bed here. :P |
| 17:30.16 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 17:30.42 | MrTelephone | could it be possibel that a cell carrier is sending inband + out of band? The CDR shows a call from a cell phone at that time |
| 17:31.29 | WIMPy | I'm pretty sure, you will never receive keypad infos. |
| 17:36.15 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:39.51 | MrTelephone | My vvx1500 is bombshelling my asterisk server with subscribe messages.. my god |
| 17:40.49 | pabelanger | disable them? |
| 17:41.28 | *** part/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 17:41.59 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 17:42.38 | MrTelephone | Yeah I'm trying to find the setting for that |
| 17:46.03 | MrTelephone | Server based dnd is interesting but what is the point? |
| 17:47.49 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 17:48.27 | tzanger | MrTelephone: I use it since I want to DND all phones in the house, not specific phones (usually) |
| 17:48.46 | tzanger | i.e. DECT cordless phones are one ext, my desk phone's another, route to cell, etc. |
| 17:49.02 | Qwell | MrTelephone: less traffic, which means less time |
| 17:49.07 | Qwell | rather than take the few hundred ms to setup the call, ring the phone, it can just respond |
| 17:49.07 | tzanger | I have DND for unknown CID, tollfree CID, etc. |
| 17:49.39 | MrTelephone | Gotchya |
| 17:53.48 | *** join/#asterisk Netgeeks-laptop (~chris@gw1.netgeeks.net) |
| 17:57.11 | jozza | later... |
| 18:01.48 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 18:01.57 | elliot98 | gives a friendly wave to all |
| 18:01.59 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 18:02.54 | elliot98 | wondering, if the settings for DTMF in channel.c are changed, would this require more than only the channel.c module to be recompile? |
| 18:03.06 | Qwell | elliot98: it depends |
| 18:03.33 | elliot98 | do tell |
| 18:03.47 | Qwell | it just depends on what you have to change |
| 18:04.00 | Qwell | just run make, and it will build whats needed |
| 18:04.47 | elliot98 | trying to improve the recognition of DTMF, and there are a few values that can be tinkered with in channel.c |
| 18:05.01 | elliot98 | there is the out-duration, in-duration, and gap-duration |
| 18:05.57 | MrTelephone | elliot98, what is wrong with your dtmf? |
| 18:06.17 | elliot98 | hard to know, but seems that it's not always being recognized |
| 18:06.40 | elliot98 | Qwell: ok, good, so just copy the modules that are recompiled |
| 18:06.44 | elliot98 | thanks |
| 18:06.54 | elliot98 | MrTelephone: if there is a way to make DTMF more reliable, that'd be great |
| 18:07.05 | elliot98 | I've set relaxdtmf |
| 18:09.16 | MrTelephone | do you use a PRI? |
| 18:09.17 | coppice | making random changes probably results in a random outcome. looking at what it wrong with the signal is more likely to produce a positive outcome |
| 18:10.19 | elliot98 | MrTelephone: PRI is used for some calls |
| 18:10.28 | elliot98 | coppice: yes, but the problem is also random :) |
| 18:11.22 | coppice | most problems appear random until you try to get to grips with the underlying cause |
| 18:12.21 | tzanger | coppice: then you realize that they really are random. :-) |
| 18:25.45 | MrTelephone | I need to keep up on changes. I just found out callwaiting is not a setting in sip.conf anymore? |
| 18:26.25 | MrTelephone | or was that always just for zapata.conf? |
| 18:26.28 | MrTelephone | wierd |
| 18:31.24 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 18:31.58 | [TK]D-Fender | Was never part of SIP |
| 18:32.17 | elliot98 | does libss7 work with *1.4? |
| 18:32.40 | elliot98 | and besides the libss7 library, dahdi, and asterisk, is there anything that's needed? |
| 18:35.29 | [TK]D-Fender | Yes, and no, respectively |
| 18:37.04 | elliot98 | ok, good, that makes yes and no both positive answers |
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| 18:39.24 | *** join/#asterisk jkroon (~jkroon@dsl-241-252-251.telkomadsl.co.za) |
| 18:41.22 | joesuffceren | I have a Dadhi/libPRI problem: When an incoming call comes in over the PRI, and I attempt to forward it back out using a SIP redirect on my endpoint, the call is dropped by my carrier. The carrier believes this is because Asterisk is sending the oubound call to them as Plan/NPI and Type/TON as "Unknown" instead of "ISDN/National" as I would expect. Here is PRI debug for a failed call... |
| 18:41.24 | joesuffceren | ...example. Line 315 shows the unexpected "unknown" behavior: http://pastebin.com/HCEy3nS0 |
| 18:42.53 | WIMPy | I can't fetch that pastebin. |
| 18:43.34 | *** join/#asterisk orn (~orn@178.19.49.66) |
| 18:44.04 | joesuffceren | Here is some PRI debug from the provider's IAD. lines 12-15 are analogous to line 315 in the previous PB. Lines 39-42 are from the PBX in my branch office performing the same operation successfully: http://pastebin.com/wph9bLzT |
| 18:44.18 | joesuffceren | WIMPy: I'm getting a "pb is under heavy load right now" error |
| 18:44.39 | WIMPy | That's what I get now as well. |
| 18:44.53 | WIMPy | Did you say IAD? Bad situation. |
| 18:47.15 | WIMPy | What kindo of connectivity do you have with an PRI IAD? |
| 18:52.55 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 18:52.59 | sereal-work | what is fromuser for? |
| 18:53.14 | sereal-work | does it have anything to do with call transfering? |
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| 19:02.44 | [TK]D-Fender | sereal-work, No, it is used in forcing an outbound identity when you call out to a peer. Don't do this for phone devices.... |
| 19:03.40 | sereal-work | Humm I really don't know why call transfers aren't working then. |
| 19:04.13 | *** join/#asterisk irroot (~gregory@197.175.131.141) |
| 19:07.49 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
| 19:07.49 | akrohn | so. GoSub() and Macro() differ in that macro passes some system arguments and returns automatically. gosub doesn't have the default arguments (but accepts passed args) and requires the return() function. in real life dial plans, what is the difference between them? what situations would you use one vs the other in? |
| 19:08.21 | joesuffceren | WIMPy: perhaps I'm using the term incorrectly. That's just what the telco guy said. We have a T1/PRI connection between our asterisk box and the telco's Cisco equipment onsite |
| 19:08.49 | WIMPy | Cisco? |
| 19:10.16 | joesuffceren | WIMPy: Metro-E Circuit Terminates into a telco-provided cisco device that has an Ethernet port for my data and a PRI port for my voice |
| 19:11.07 | WIMPy | Sounds like an IAD. |
| 19:11.18 | tzanger | better than an IED I guess |
| 19:11.41 | WIMPy | Got your trace, BTW, but an intense version would be easier to read. |
| 19:11.53 | joesuffceren | WIMPy: :-) Every PRI voice circuit I've ever seen has been like this. what's the alternative? |
| 19:12.15 | joesuffceren | I just want to make sure I'm not using terms incorrectly and making it seem more complicated than it is |
| 19:12.46 | WIMPy | A simple NT. Either from an UK2 or today usually from SHDSL. |
| 19:16.15 | WIMPy | There are a few calls in there. Which one is the one with the issue? |
| 19:16.15 | joesuffceren | gotcha. I've only ever dealt with PRIs from this one telco, come to think of it, so this must just be the way they do it. can't figure out the syntax for intense pri debug with 1.8. can you help me out with that? |
| 19:16.47 | joesuffceren | whi pb are you looking at? |
| 19:16.51 | WIMPy | pri set debug 2 span 1 |
| 19:17.03 | WIMPy | Your 1st. |
| 19:18.07 | joesuffceren | well, there's an incoming call that is then attempted to be forwarded back out to an external number, so that's two different calls as far as asterisk is concerned, if that's what you mean. |
| 19:18.13 | [TK]D-Fender | sereal-work, pastebin a failed call with SIP DEBUG enabled for us to look at and maybe we'll be able to tell you. |
| 19:18.17 | [TK]D-Fender | ~pb |
| 19:18.17 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:18.18 | [TK]D-Fender | ^^^ |
| 19:18.59 | joesuffceren | line 315 shows the beginning of the part that fails (attempted outbound call/forward to the 2929391 number) |
| 19:19.19 | WIMPy | joesuffceren: I see 5 calls in there. |
| 19:19.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:20.29 | WIMPy | Ok, call 11378 |
| 19:21.22 | WIMPy | It ends with a timer expiry. |
| 19:21.40 | elliot98 | is there a way to change the toneduration in asterisk settings or do I need to recompile it? |
| 19:22.04 | elliot98 | for DTMFs that is |
| 19:22.29 | WIMPy | Is that one of those interesting destinations I read anout here, that reuire DTMF input befor accepting the call? |
| 19:23.12 | *** join/#asterisk ttyandroid (~tty665@189-015-118-103.xd-dynamic.ctbcnetsuper.com.br) |
| 19:23.43 | elliot98 | just for general DTMF recognition |
| 19:24.11 | elliot98 | want improved DTMF somehow on the server |
| 19:24.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:24.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:25.31 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
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| 19:25.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:25.56 | elliot98 | and do different countries have different frequencies for DTMF? |
| 19:26.33 | *** join/#asterisk d00gster (~dt@94.97.83.60) |
| 19:28.00 | coppice | no, but they may use different thresholds |
| 19:28.58 | elliot98 | thresholds? |
| 19:29.24 | joesuffceren | WIMPy: here's the intense debug: http://pastebin.com/piuwCsCL |
| 19:29.51 | coppice | thresholds for twist and purity |
| 19:30.02 | *** part/#asterisk ttyandroid (~tty665@189-015-118-103.xd-dynamic.ctbcnetsuper.com.br) |
| 19:32.08 | orn | elliot98: Are you having problems detecting the tones? |
| 19:33.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:37.20 | MrTelephone | do theresholds consider frequency or amplitude? |
| 19:37.38 | WIMPy | joesuffceren: I have no clue as to why you get an timout. |
| 19:37.42 | joesuffceren | WIMPy: Do you have any idea why the TON and NPI would be 'Unknown' on line 517 of the intense debug pb I just posted? You mentioned the timer expiry earlier, but I'm not sure what to make of that. Can you elaborate |
| 19:37.57 | WIMPy | But your redirecting number is certainly invalid. |
| 19:38.38 | WIMPy | Unless you know what you're doing, "unknown" is the safe choice. |
| 19:39.08 | joesuffceren | WIMPy: ah, ok. it's just not what the PBX in my branch office sends out during this same forwarding scenario, so I assumed it was related to the problem |
| 19:39.21 | WIMPy | Same goes for the connected number in the next call. |
| 19:40.33 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-raqgxfdqbffcxnru) |
| 19:40.34 | joesuffceren | nothing at all on my PBX has changed since this worked (the 23rd). we just switched from a T1 to a Metro E with our telco, but the PRI was supposed to stay the same. They can't figure out the problem and are pointing the finger at that redirect number you mentioned. I just don't understand why it worked before / why it works in my branch office with the same config |
| 19:40.35 | WIMPy | It's hard to guess why there's a timeout. But as it's happening elsewhere, there isn't much we can do find out, what's going on. |
| 19:40.37 | elliot98 | orn: seems so |
| 19:40.57 | tzanger | hmm, wonder if I can get an arbitrary string into the manager events for sip reg/dereg |
| 19:41.06 | elliot98 | coppice: so what's available for tinkering with thresholds? |
| 19:41.14 | elliot98 | orn: DTMF issues |
| 19:41.19 | orn | elliot98: Do you have relaxdtmf=yes ? |
| 19:41.42 | WIMPy | joesuffceren: Invalid numbers can cause calls to be dropped, but a timeout is a different thing. |
| 19:41.48 | elliot98 | orn: interestingly enough, when that is set, voices can trigger DTMFs |
| 19:42.06 | WIMPy | So it may be a good idea to fix it, but I don't think it's the cause for this issue. |
| 19:42.15 | coppice | elliot98: if you haven't worked out your problem, how can you search for a solution |
| 19:42.37 | orn | elliot98: That's not uncommon -- happens sometimes too when it's not relaxed. |
| 19:42.59 | joesuffceren | WIMPy: ah, I see. may have to get the telco to file a ticket with Cisco, then... ugh. thanks for your help! |
| 19:43.14 | orn | elliot98: So you're having problems with just regular users using DTMF, or are you working with some burglar alarm/fire alarm systems or something? |
| 19:44.09 | WIMPy | joesuffceren: Unfortunately I have no idea about that equipment. Usually IAD implys some sort of media gateway. If that's the case, it's the IAD that's the issue. |
| 19:45.21 | WIMPy | Hmm |
| 19:45.45 | WIMPy | I just wonder if the whole message is ok as it is.. |
| 19:45.53 | elliot98 | orn: good ol' IVRs |
| 19:46.51 | joesuffceren | WIMPy: "whole message" at me? If so, I'm not sure I follow |
| 19:47.48 | orn | elliot98: and you're positive it's not just mis-matching dtmf settings? |
| 19:48.13 | elliot98 | sometimes creates doubles |
| 19:48.42 | WIMPy | joesuffceren: The redirecting number is not part of a setup. |
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| 19:51.40 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-aycisdmafphqxqle) |
| 19:53.47 | elliot98 | orn: should the gap be changed for DTMFs? |
| 19:53.57 | joesuffceren | WIMPy: it's just there for informational purposes? |
| 19:56.45 | elliot98 | if DTMF is creating doubles, what should be changed |
| 19:56.47 | elliot98 | ? |
| 20:00.42 | WIMPy | joesuffceren: Interestingly enough I found a reference to redirecting numbers not being forwarded without explicit number type. |
| 20:01.02 | WIMPy | That's still no explanation for a timeout, but interesting. |
| 20:01.30 | WIMPy | Unfortunatly there's no information on the equipment. |
| 20:01.36 | joesuffceren | WIMPy: interesting. Is there any way to / Any reason we can't explicity set the number type in asterisk? |
| 20:01.57 | WIMPy | You can. |
| 20:02.11 | WIMPy | Either in chan_dahdi.conf or in the caller ID itself. |
| 20:02.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:02.28 | WIMPy | The later is explained in the chan_dahdi.conf.samlpe as well. |
| 20:02.51 | joesuffceren | you said earlier that shouldn't be done unless one know what one is doing. What are the pitfalls? |
| 20:02.52 | *** join/#asterisk chasing`Sol (~cS@41.206.152.243) |
| 20:03.25 | WIMPy | If you set an explicit TON, the number has tobe formatted correctly. |
| 20:03.42 | WIMPy | If you set it to unknown, the switch will find out itself. |
| 20:04.18 | WIMPy | For the called party number, you would restrict outgoing calls. |
| 20:06.26 | WIMPy | Setting up a second call to do forwarding isn't ideal, anyway. |
| 20:07.03 | joesuffceren | I know that the IAD will accept calls as 7, 10, or 11 digits, so I should still be able to use any of those, correct? Perhaps the timeout occurs when the switch is trying to determine the number type and cannot for some reason? |
| 20:07.06 | WIMPy | Well, "forwarding" is incorrect anyway. |
| 20:07.28 | joesuffceren | I hear you on the 2nd call is not ideal bit. unfortunately, that's what these endpoints do, and I can't afford to replace them. :-/ |
| 20:07.45 | WIMPy | Usually yu get a status message if there's somethign the switch doesn't like. |
| 20:08.54 | WIMPy | I have to admit, I didn't look at the dialplan part at all. So it's a terminal based diversion? |
| 20:11.24 | joesuffceren | yeah, a call forward button on the endpoint |
| 20:11.45 | WIMPy | Which isn't forwarding, either. |
| 20:12.24 | joesuffceren | well, it looks that way to the user. "SIP diversion by generating additional call" wouldn't fit on the little button, I suppose |
| 20:12.52 | WIMPy | "diversion" would be correct as a generic term. |
| 20:14.04 | WIMPy | I haven't experimented with that part, but dahdi can do transfers. Not sure if it will do forwards as well. |
| 20:14.16 | WIMPy | I remember bristuff did. |
| 20:14.21 | WIMPy | So maybe... |
| 20:16.38 | WIMPy | I can only find transfer and facilityenable, which seem to be for transfers only. But who can be sure? |
| 20:19.51 | cj | hey folks |
| 20:20.05 | cj | I'm having fail when asterisk bridges the call locally |
| 20:20.17 | cj | what might cause this? |
| 20:21.04 | [TK]D-Fender | cj, Show us the failure |
| 20:21.05 | [TK]D-Fender | ~pb |
| 20:21.05 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:21.07 | [TK]D-Fender | ^ |
| 20:21.25 | cj | [TK]D-Fender: The failure is a lack of audio |
| 20:21.30 | cj | hard to paste that |
| 20:21.44 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 20:23.24 | SeRi | waz up guys |
| 20:23.45 | kikohnl | on our old NEC pbx, we had a feature called "Virtual Extensions" basically if a virt exten was dialed, it would turn around an call a preconfigured outside number, how do I do that in asterisk? |
| 20:25.16 | WIMPy | kikohnl: Asterisk doesn't care, you can dial anything you want. |
| 20:26.24 | [TK]D-Fender | cj, probably not. Also probably not so hard to tell us what you're dealing with. |
| 20:27.03 | [TK]D-Fender | kikohnl, dial and outside number instead of some fixed device |
| 20:27.50 | WIMPy | I think, we need a dictionary. |
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| 20:32.38 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 20:33.07 | anny__ | hi all, I want to set the __DYNAMIC_FEATURES variable to multiple values how can i do that? |
| 20:33.34 | [TK]D-Fender | anny__, feature#feature#feature |
| 20:35.01 | anny__ | it's not working, it's taking only the first value and the others it's ignoring them |
| 20:37.31 | [TK]D-Fender | shwo us your features.conf and your call attempt |
| 20:37.33 | [TK]D-Fender | ~pb |
| 20:37.34 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:37.36 | [TK]D-Fender | ^^^ |
| 20:37.52 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 20:40.40 | anny__ | here's my configuration: http://pastebin.com/Dsq14ddy |
| 20:41.52 | [TK]D-Fender | anny__, "__" <- don't do this in front of the var |
| 20:42.30 | cj | [TK]D-Fender: http://paste2.org/p/1804747 |
| 20:42.47 | cj | line 12 is the output I get when it bridges the two channels |
| 20:42.55 | cj | however, I get no audio from either of the phones |
| 20:43.46 | anny__ | [TK]D-Fender: This is from Asterisk docs: The two leading underscores allow these feature settings to be set on the outbound channels, as well. |
| 20:45.55 | [TK]D-Fender | anny__, Yeah I do see that for when you want to give the callee access.. |
| 20:48.14 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 20:49.12 | anny__ | thank you it's solved |
| 20:49.50 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 20:51.10 | [TK]D-Fender | anny__, What did you do? |
| 20:51.55 | [TK]D-Fender | cj, Now try that call again, this time with SIP DEBUG enabled. |
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| 20:54.36 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:54.48 | anny__ | turn out it's entering a wrong context in the call attempts |
| 20:54.57 | [TK]D-Fender | :) |
| 20:55.22 | [TK]D-Fender | anny__, That's why I ask to see the failure.... You can show me all the dialplan you want ... and watch it never get used :) |
| 20:57.32 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 21:01.50 | cj | [TK]D-Fender: http://proxy01.colliertech.org/~cjac/tmp/call.log |
| 21:03.09 | anny__ | [TK]D-Fender: lol |
| 21:10.06 | SeRi | p3nguin: you avail? |
| 21:10.08 | vader-- | any of you guys have an idea why i have a TDM400P card with 4 FXS channels, that shows up ok in the CLI but not in Asterisk? |
| 21:10.14 | vader-- | i also don't get a dialtone |
| 21:10.19 | p3nguin | ring-a-ding |
| 21:10.26 | SeRi | lol |
| 21:10.42 | SeRi | p3nguin: I fix that issue I had with my phone. |
| 21:10.49 | p3nguin | Which one? |
| 21:11.10 | SeRi | the parking issue. where it would kill the audio one way |
| 21:11.28 | p3nguin | Did you figure out the problem, or did it magically stop happening? |
| 21:11.30 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 21:11.52 | SeRi | p3nguin: I fix the problem |
| 21:11.53 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-ehoayznikutijioi) |
| 21:11.53 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 21:12.01 | Qwell | p3nguin: very nice |
| 21:12.41 | SeRi | the RTP port was screwed up along with the jitter settings and the buffer setting |
| 21:13.13 | SeRi | put it all back the way "I had it before" and it all work again. |
| 21:15.57 | [TK]D-Fender | vader--, Stick to #freepbx . And you didn't configure the channels, that's why |
| 21:17.10 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 21:17.12 | wcselby | o/ |
| 21:18.08 | p3nguin | http://www.speedtest.net/result/1624231624.png No wonder shit feels slow. |
| 21:18.43 | [TK]D-Fender | cj, Ok, SIP debug looks OK, check your firewalls on the * side to ensure your RTP ports are clear. |
| 21:19.11 | SeRi | p3nguin: damn |
| 21:19.17 | SeRi | waz up wcselby |
| 21:19.27 | SeRi | I am out with shingles :( |
| 21:19.27 | wcselby | howdy |
| 21:19.32 | wcselby | :/ that sucks |
| 21:19.39 | SeRi | sure does. |
| 21:19.57 | SeRi | hurts like a mofo |
| 21:20.15 | orn | SeRi: My wife has shingles for the first time and is experiencing some weird ass shit |
| 21:21.11 | SeRi | orn: shingles them self are some weird ass shit.... I hope she gest better...I know it sucks |
| 21:21.17 | wcselby | SeRi if it makes you feel any better (probably won't), i've been fighting a chest cold for about a month? |
| 21:21.33 | *** join/#asterisk willzzz (ce0a99a2@gateway/web/freenode/ip.206.10.153.162) |
| 21:21.41 | orn | SeRi: It's a disease that's always dormant, right (after you contract it)? |
| 21:21.41 | SeRi | wcselby: sorry to hear that. can I trade you? |
| 21:21.49 | wcselby | nah I'm good |
| 21:21.50 | willzzz | i have a question regarding the asterisk real-time architecture versus the current CDR logging... |
| 21:21.50 | wcselby | :) |
| 21:21.55 | SeRi | wcselby: LOL |
| 21:22.03 | willzzz | Right now my macro for encodequeue encodes everything into the db which my php app queries at the end of a call |
| 21:22.09 | wcselby | willzzz realtime and cdr don't have much to do with each other |
| 21:22.14 | wcselby | other than you can store cdr's in a database |
| 21:22.15 | SeRi | orn: Yes. Its basically shicken pox |
| 21:22.50 | willzzz | wcselby, is it possible for the caller # and id and the rest i have defined in cdr_mysql.conf to be written to the db as the call is still happening? |
| 21:22.51 | orn | crap |
| 21:22.59 | SeRi | orn: Usually is trigerd by stress... IN my case at least it was... |
| 21:23.27 | wcselby | willzzz you mean, the table aliases you have defined in cdr_mysql.conf? |
| 21:23.28 | orn | SeRi: Does it have any effect on you apart from the pain (which I understand feels like it's in the skin?) |
| 21:23.34 | SeRi | orn: Not contagious though. Only when somebody who has never had chickenpox touches the rash which I am sure nobody would like too :/ |
| 21:23.38 | wcselby | willzzz what is it you have defined in cdr_mysql.conf? |
| 21:23.39 | willzzz | wcselby, yesl |
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| 21:24.04 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
| 21:24.04 | willzzz | wcselby, start, clid, src, dst,basically e verything |
| 21:24.09 | orn | SeRi: Yeah, she works in a kindergarten, probably contracted it from one of the little bastards |
| 21:24.11 | SeRi | orn: The pain is insane! and is not just on the skin! it is on your nerves too |
| 21:24.20 | SeRi | shingles attack your nerve syste, |
| 21:24.21 | willzzz | really i only need the clid in the call-log and src and dst |
| 21:24.24 | wcselby | willzzz can you reask the question, I'm not sure I understand what you're wanting to do? |
| 21:24.37 | SeRi | system* |
| 21:24.49 | willzzz | i want the src dst, and a few other aliases written into the mysql cdr db before the call is finished. |
| 21:25.06 | willzzz | so the person answering the call with the customer can look at our call log and see oh that clid# |
| 21:25.12 | willzzz | and match it into our internal order syste, |
| 21:25.15 | orn | SeRi: She's been having some weirdness in her heart, palpitations, heaviness, rapid heart rate -- has that ever happened to you? |
| 21:25.21 | wcselby | willzzz at what point of the call do you want them recorded? |
| 21:25.25 | SeRi | orn: I doubt it was from another kid. but you never know. It is said that shingles can only give you once.... A myth.... I am on my second round. |
| 21:25.32 | orn | SeRi: We went to the hospital, just in case it was a heart attack |
| 21:25.41 | willzzz | wcselby, everything. but the recording can happen after the call has hung up (I understand that part) |
| 21:26.02 | willzzz | basically i want a split cdr entry, some written when the call is connected, finishes when its disconnected (recording). |
| 21:26.14 | SeRi | orn: If she is on steroirds than yes thats normal. but if she is not... You best keep an eye on her. |
| 21:26.36 | SeRi | steroids tend to cause what you described. |
| 21:26.45 | [TK]D-Fender | checkout time, BBIAB |
| 21:26.52 | SeRi | cya! |
| 21:26.56 | orn | SeRi: You mean in general, or when you have shingles? |
| 21:27.36 | p3nguin | My router is up 225 days. Maybe it could stand a reboot? |
| 21:27.47 | SeRi | orn: if her shingles sort of brake out on a bad rash doctors prescribe steroids to help the medecine take its course.... |
| 21:27.52 | SeRi | p3nguin: Never! |
| 21:27.53 | p3nguin | <PROTECTED> |
| 21:28.01 | p3nguin | It's not really loaded up at all. |
| 21:28.08 | wcselby | willzzz i think that's how it already works, but I'm not sure. But what it really sounds like is you just need to create a separate database that stores all the same info, but you can then control when stuff goes into it using AGI or MYSQL calls or whatever |
| 21:28.09 | SeRi | let it be :) |
| 21:28.13 | p3nguin | Maybe it's the proxy. |
| 21:28.20 | SeRi | +1 |
| 21:28.22 | orn | SeRi: Yeah, I got that, but I meant if steroids generally would cause that, or just steroids when prescribed to deal with shingles |
| 21:28.33 | s[X] | hey all |
| 21:28.44 | SeRi | orn: generally would cause that |
| 21:28.48 | orn | i see |
| 21:28.50 | SeRi | steroids in general |
| 21:28.52 | willzzz | wcselby, i can have a entry in our extensions.conf |
| 21:29.01 | willzzz | that tells MySQL to write the clid of the CDR correct? |
| 21:29.10 | willzzz | and the other relevant call data that is static once the call is connected? |
| 21:29.14 | p3nguin | http proxy: disabled |
| 21:29.16 | p3nguin | So that's not it. |
| 21:29.22 | willzzz | but the regular cdr, will it be overwritten? |
| 21:29.26 | willzzz | basically i want a split cdr |
| 21:29.58 | SeRi | p3nguin: :/ |
| 21:30.07 | SeRi | do an iperf test |
| 21:30.20 | SeRi | p3nguin: ^^ |
| 21:30.44 | p3nguin | Don't need to. |
| 21:30.57 | p3nguin | curl http://mirror.ece.vt.edu/archlinux/community/os/x86_64/alienarena-data-7.52-1-any.pkg.tar.xz >/dev/null |
| 21:31.11 | SeRi | well if you think the router is the problem and iperf test TO the router would tell you |
| 21:31.36 | wcselby | willzzz you can use these variables to access from the dialplan the values in the CDR -> https://wiki.asterisk.org/wiki/display/AST/CDR+Variables |
| 21:31.38 | p3nguin | I'm getting an average of 56616 download. |
| 21:32.24 | SeRi | p3nguin: yes but you are going over http.... my idea was more or less to stress the router it self and see if is not handeling load very well... |
| 21:32.25 | MrTelephone | 56616 gigabits? |
| 21:32.37 | p3nguin | bytes/second |
| 21:32.41 | wcselby | hey willzzz you need to go through some of the links on this page -> https://wiki.asterisk.org/wiki/dosearchsite.action?where=AST&tooltip=Type+ALL%3A+in+your+query+to+search+all+of+Confluence&spaceSearch=true&queryString=cdr |
| 21:32.46 | SeRi | TF |
| 21:33.09 | p3nguin | curl http://mirror.rit.edu/archlinux/community/os/x86_64/alienarena-data-7.52-1-any.pkg.tar.xz >/dev/null |
| 21:33.26 | MrTelephone | I lost 2 motorola canopy radio power supplies in 1 day? I think there is a solar flare coming |
| 21:33.28 | p3nguin | This mirror is giving me over 2000kB/s from another system. |
| 21:33.54 | p3nguin | 300kB/s on the one giving me trouble. |
| 21:34.16 | SeRi | looks like aserver side issue nothing to worry about I guess |
| 21:34.45 | SeRi | MrTelephone: hu? the end of the world? OMFG! |
| 21:35.07 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 21:35.24 | p3nguin | 2249kB/s via one router/modem, but 300kB/s via another is a SERVER SIDE PROBLEM? Are you kidding me? |
| 21:35.47 | SeRi | o I thought you where talking about the same. |
| 21:36.25 | p3nguin | The one giving me problems is supposed to be the fast one! |
| 21:36.36 | p3nguin | I can typically download 6MB/s. |
| 21:36.43 | p3nguin | or more. |
| 21:36.48 | SeRi | I see. Is that the same one you had issues with *? |
| 21:36.51 | p3nguin | no |
| 21:37.00 | p3nguin | That one is giving me great speed. |
| 21:37.04 | MrTelephone | mr. poppers |
| 21:37.14 | SeRi | p3nguin: I see. |
| 21:38.52 | p3nguin | From my computer to the router, I get 95.7 Mbits/sec (57.0 MBytes) |
| 21:39.10 | wcselby | anyone use any kind of tabbed remote desktop / putty app that works on windows 7? |
| 21:42.13 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 21:42.14 | joesuffceren | wcselby: I use visionapp Remote Desktop (does VNC, RDP, putty, etc. and can save creds/connections to a local db or to SQL db so you can share with others). I absolutely adore it! |
| 21:42.51 | p3nguin | I'm running that same curl download directly on the router, and it gave me 2251kB/s on average for the file. |
| 21:42.56 | p3nguin | So it's the router. |
| 21:44.26 | p3nguin | I've noticed for the past few weeks that surfing feels very sluggish. I'm going to reboot the router, I guess. |
| 21:44.46 | p3nguin | I hate to, but I think I need to. |
| 21:45.08 | wcselby | joesuffceren that's the only recommendation I've received so far. did you pay the 99 license fee for it, or is there another version without the associated cost? |
| 21:45.45 | wcselby | joesuffceren I mean, I've received that recommendation from some other admin friends |
| 21:45.53 | wcselby | everyone that has it loves it |
| 21:46.00 | wcselby | i may have to talk my work into spending a little cash on it |
| 21:46.51 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 21:48.16 | joesuffceren | wcselby: I paid. There's a free version that is limited to 3 concurrent connections and cannot export/backup more than 15 items. |
| 21:48.24 | wcselby | ahh |
| 21:48.25 | wcselby | thanks |
| 21:51.55 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 21:52.03 | *** join/#asterisk asteriskn00b (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 21:52.23 | asteriskn00b | anyone seen this before? [Dec 1 15:42:38] WARNING[24947]: chan_sip.c:3196 sip_call: No audio format found to offer. Cancelling call to sip502_dipstorrens |
| 21:55.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:55.47 | p3nguin | I guess you didn't offer any codecs. |
| 21:55.59 | *** join/#asterisk mindCrime (~chatzilla@24.106.207.82) |
| 21:56.53 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 21:59.35 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:04.39 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:08.57 | *** join/#asterisk canp (~resu@pool-173-67-137-75.hrbgpa.fios.verizon.net) |
| 22:09.20 | canp | hello :) |
| 22:09.26 | joesuffceren | asteriskn00b: order of allow and disallow matters, so allow=gsm before disallow=all has a net effect of disallow=all iirc. |
| 22:10.39 | canp | does the asterisk chat have any problem with idlers? |
| 22:10.47 | canp | i know some chats do |
| 22:11.47 | canp | finally got my copy of asterisk: the future of telephony |
| 22:11.56 | canp | i cant put it down :D |
| 22:12.12 | asteriskn00b | looks like 729 is missing |
| 22:12.28 | asteriskn00b | fs16*CLI> show modules like codec_ |
| 22:12.28 | asteriskn00b | codec_a_mu.so codec_adpcm.so codec_alaw.so codec_g726.so codec_gsm.so |
| 22:12.28 | asteriskn00b | codec_lpc10.so codec_ulaw.so |
| 22:16.07 | *** join/#asterisk coppice (~chatzilla@host86-136-94-225.range86-136.btcentralplus.com) |
| 22:16.46 | p3nguin | If you didn't buy the g.729 license, why would you expect it to be present? |
| 22:17.46 | asteriskn00b | didnt know it wasnt loaded by default, got lucky finding that command online and checking against my other servers |
| 22:17.51 | wcselby | can't you use g729 in passthrough mode without a license? |
| 22:17.59 | asteriskn00b | heh and that but |
| 22:21.09 | p3nguin | And the correct command is "module show like ..." so I'm kind of surprised it worked at all. |
| 22:21.52 | p3nguin | The license is required whenever you have to transcode. |
| 22:24.15 | p3nguin | seri: Now I feel silly. I rebooted the router and it didn't help. And now I checked throughput on my laptop, and it's well over 2000kB/s through the same router/modem. |
| 22:27.25 | wcselby | p3nguin I think there's an alias for module show to show modules |
| 22:28.04 | [TK]D-Fender | wcselby: Yes, but you won't see a module for it |
| 22:28.22 | [TK]D-Fender | No module means no transcode. |
| 22:28.53 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 22:30.03 | p3nguin | <PROTECTED> |
| 22:30.03 | p3nguin | <PROTECTED> |
| 22:30.10 | p3nguin | So that isn't the problem, either. |
| 22:32.36 | asteriskn00b | how can i check how many licenses i have on the other servers? |
| 22:33.02 | p3nguin | g729 show licenses? |
| 22:33.16 | p3nguin | I can't remember if that's right. |
| 22:33.25 | asteriskn00b | it is |
| 22:33.27 | asteriskn00b | thank you |
| 22:34.22 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 22:34.30 | IsUp | hi |
| 22:34.33 | SeRi | p3nguin: you still at it? |
| 22:34.51 | *** join/#asterisk hajekd (~hajekd@2a00:1268:15:1800:908a:5c9f:307b:e7f8) |
| 22:39.07 | SeRi | elo |
| 22:39.44 | p3nguin | Yeah, I don't get it. There isn't very much traffic on that computer, iperf to the router shows 100Mbit, http through the router is slow as hell, and http from my laptop through the router is normal (fairly fast). |
| 22:40.07 | IsUp | p3nguin: still problem with calls? |
| 22:40.44 | p3nguin | Not today. Can you send that info again via /msg? I didn't get it before. |
| 22:41.11 | p3nguin | Right now I'm trying to figure out another issue. |
| 22:42.52 | WIMPy | Damn, I hate Linux. |
| 22:43.00 | SeRi | lol |
| 22:43.03 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yzwvdlpknjivpjdb) |
| 22:43.16 | WIMPy | Trying netinstall and everythime NFS hangs at the same package. |
| 22:43.25 | SeRi | Linux r0x0r y0ur b0xr! |
| 22:43.31 | SeRi | lol |
| 22:43.50 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:44.13 | WIMPy | wonders if it could be the bridge. |
| 22:44.30 | WIMPy | That didn't exist last time I did it. |
| 22:45.47 | SeRi | p3nguin: is the routing tables ok on the router? is there any http zombie sessions? |
| 22:45.59 | p3nguin | I just rebooted it. |
| 22:46.03 | SeRi | good |
| 22:46.09 | SeRi | That was my next advice |
| 22:46.10 | p3nguin | It didn't help. |
| 22:46.14 | SeRi | well fuck |
| 22:46.15 | SeRi | lol |
| 22:46.26 | SeRi | ISP? |
| 22:46.29 | p3nguin | Speed test is fast on my laptop. |
| 22:46.37 | SeRi | fuck twice |
| 22:46.38 | p3nguin | It's just my desktop. |
| 22:46.48 | SeRi | :/ |
| 22:46.48 | p3nguin | iperf from my desktop to the router is fast. |
| 22:47.02 | p3nguin | does not make sense! |
| 22:47.40 | SeRi | so if your laptop is ok. than somethign is up on your desktop... did you check your nic? is it flapping? any drop packets on the nic? |
| 22:48.11 | SeRi | did you try ftp on the desktop? |
| 22:48.19 | p3nguin | <PROTECTED> |
| 22:48.19 | p3nguin | <PROTECTED> |
| 22:48.22 | SeRi | is it only with http? |
| 22:48.29 | p3nguin | I'll check ftp now. |
| 22:48.36 | SeRi | That looks ok. |
| 22:49.11 | p3nguin | ftp is just as slow. |
| 22:49.27 | SeRi | uptime? |
| 22:49.30 | p3nguin | <PROTECTED> |
| 22:49.30 | p3nguin | <PROTECTED> |
| 22:49.44 | p3nguin | 117 days |
| 22:49.55 | SeRi | Mhhhh.... |
| 22:50.22 | SeRi | restart your nic |
| 22:51.42 | SeRi | also you can try to unload and load the module and see if anything got fubar with it... If is built in to the kernel that you shouldnt have to worry about that... |
| 22:51.58 | SeRi | s/that/than/ |
| 22:53.10 | p3nguin | Still wrong! |
| 22:53.15 | p3nguin | "then" |
| 22:53.31 | SeRi | Sr Yes Sr! |
| 22:53.50 | SeRi | make notes of it.... |
| 22:55.32 | p3nguin | 8069kB/s |
| 22:55.36 | p3nguin | wtf |
| 22:55.43 | p3nguin | 9399kB/s |
| 22:55.57 | p3nguin | That's a fast friggin' mirror. |
| 22:56.10 | p3nguin | 10.0MB/s |
| 22:56.16 | ChannelZ | My mirror works at the speed of light |
| 22:56.28 | p3nguin | Downloaded the entire 327MB file in 45 seconds. |
| 22:56.42 | p3nguin | Averaged 7151kB/s |
| 22:57.02 | p3nguin | I should have restarted the network weeks ago! |
| 22:57.20 | p3nguin | I didn't know I could download that fast. |
| 22:57.44 | SeRi | LOL |
| 22:58.10 | p3nguin | On ftp, it's giving me 10.3MB/s. |
| 22:58.11 | SeRi | is it working? |
| 22:58.27 | SeRi | Nice! |
| 22:58.31 | p3nguin | I really didn't know my internet was that fast. |
| 22:58.42 | wcselby | lol |
| 22:59.00 | ChannelZ | turbo boost |
| 22:59.00 | p3nguin | on ftp, it averaged 9367kB/s. Downloaded the 327MB file in 35 seconds. |
| 22:59.06 | coppice | that isn't fast. 327M in 5s is a respectable speed |
| 22:59.21 | p3nguin | Over cable internet, it's pretty good. |
| 22:59.46 | IsUp | p3nguin: my download speed is 930 kb/s |
| 22:59.52 | IsUp | p3nguin: you are lucky. |
| 22:59.59 | p3nguin | 930 kbits? |
| 23:00.23 | IsUp | p3nguin: exactly. i have ADSL, 8192 downstream and 782 upstream |
| 23:00.39 | p3nguin | I'm measuring a file transfer in Bytes per second. |
| 23:01.28 | SeRi | p3nguin: so you got it now? what did you do? |
| 23:01.42 | p3nguin | /etc/rc.d/network restart |
| 23:02.01 | SeRi | ah cool! :) |
| 23:02.51 | SeRi | That happen before to me and I restarted the nic. but I had some errors.... Yours didnt but I thought I throw it out there any ways |
| 23:03.09 | WIMPy | At least I'm not alone. |
| 23:03.29 | p3nguin | So I rebooted the router and restarted networking on the desktop, and got another 13 Mbits on my internet speed. |
| 23:03.32 | SeRi | the net scalers are down at work... :( no mail. |
| 23:03.34 | p3nguin | I should do that more often. |
| 23:04.17 | SeRi | you shouldnt have too :/ |
| 23:04.41 | p3nguin | But if I get 13 more every time, I could do it a few more times and break 100 Mbit. |
| 23:05.30 | SeRi | lmao |
| 23:07.31 | IsUp | gotta go |
| 23:07.34 | IsUp | night all |
| 23:15.46 | ChannelZ | What speed are you paying for? I have 12/2 at home and 20/5 at work. I forget how much the turbo boost is for the first 15 seconds or whatever it is |
| 23:18.56 | joesuffceren | WIMPy: telco says the issue is between their IAD and their feature server. I'm not sure why it happens only with my "forwarded" calls and not my normal outbound ones, but it seems there's definitely no issue on my end in their opinion, now. |
| 23:19.01 | joesuffceren | thanks again for your help |
| 23:19.33 | canp | has anyone here ran asterisk on fedora? |
| 23:19.42 | WIMPy | "feature server"? |
| 23:19.54 | [TK]D-Fender | canp: tons of people. Its RH like everything else |
| 23:20.30 | joesuffceren | WIMPy: those were his words. I guess it's the SIP server on their network that their IAD proxies calls to |
| 23:20.47 | canp | [TK]D-Fender: awesome |
| 23:21.37 | WIMPy | I've not seen a SIP gateway yet, that I would consinder as working. |
| 23:23.47 | *** join/#asterisk orn (~orn@178.19.49.66) |
| 23:26.13 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:31.34 | *** join/#asterisk woleium (~woleium@email.tecglobal.net) |
| 23:34.56 | woleium | Is asterisk 10 considered stable enough for production? |
| 23:35.49 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:38.41 | s[X] | Hey all |
| 23:40.12 | *** join/#asterisk mirco (~mirco@p5B282D9C.dip.t-dialin.net) |
| 23:42.12 | puzzled | woleium: if it works for you, sure. other than that, it's called a release candidate for a reason. and even after that it is still a point release which is reason enough for some to avoid it yet others will rejoice and roll it out in a jiffy |
| 23:42.13 | WIMPy | woleium: That probably depends on what features you're using. |
| 23:42.18 | WIMPy | It's been working well for me. |
| 23:42.27 | mirco | Hi all, I've setup Asterisk on voyage-linux, and I can't fix my web-manager prob: "== Unable to write (Permission denied)" or "The GUI does not have necessary privileges. |
| 23:42.27 | mirco | Please check the manager permissions for the user !" |
| 23:44.09 | WIMPy | mirco: Try to ask in the channel for the GUI you're using. |
| 23:44.59 | mirco | it's configured in http.conf so I expected it to be default... |
| 23:45.07 | woleium | Hmm, thanks puzzled, WIMPy. How hard is it to upgrade, if I go with stable for now? i.e. is there any manual configuration change required between the verisons? |
| 23:45.52 | WIMPy | woleium: Usually not. |
| 23:46.22 | puzzled | woleium: there are a bunch of UPGRADE<something>.txt files in the source. read each of those |
| 23:46.48 | woleium | goes off to read - thanks guys |
| 23:48.40 | *** join/#asterisk _N1X_ (~z03r0c00l@41.34.221.51) |
| 23:49.07 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:49.18 | _N1X_ | hello every one |
| 23:49.36 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
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| 23:50.10 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 23:53.57 | _N1X_ | how to receive calls in DID ? |
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| 23:55.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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