| 00:00.04 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:04.46 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:07.07 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:14.02 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:14.39 | p3nguin | fofware: You think you can fix your stuff? Or at least take us off autojoin. You're filling the channel with useless join/quit crap. |
| 00:15.09 | *** join/#asterisk digilink (~digilink@irc.stephennet.net) |
| 00:21.23 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:22.00 | p3nguin | fofware: Or I guess I could ignore you until you piss off someone that matters and then they can solve it for you/me. |
| 00:23.06 | ferdna | what is Support SIP Instance ID? |
| 00:24.29 | *** join/#asterisk ShaunR (~shaun@freenode/sponsor/NDChost.com) |
| 00:24.38 | ShaunR | anybody know of a soft phone that supports SIP over TCP? |
| 00:26.37 | joeyjones | ShaunR: i want to say counterpath x-lite 4 does |
| 00:26.52 | joeyjones | bot sure, but i seem to remember it having that option |
| 00:26.56 | p3nguin | By default, or is it a configurable setting? |
| 00:26.57 | joeyjones | it did suck though |
| 00:27.01 | joeyjones | configurable iirc |
| 00:27.16 | p3nguin | I'm not totally sure why anyone would want to use TCP for SIP. |
| 00:27.57 | joeyjones | to get to it |
| 00:28.08 | p3nguin | Say what? |
| 00:28.13 | x86 | yeah x-lite does TCP |
| 00:28.14 | WIMPy | To work around shitty routers perhaps? |
| 00:28.38 | p3nguin | Even the shittiest of router should know how to route UDP just like any other part of IP. |
| 00:28.41 | joeyjones | softphone->account settings->transport->signalling transport |
| 00:28.45 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:28.47 | x86 | I've not had any problems with Internet-based TCP SIP clients... but YMMV |
| 00:28.49 | joeyjones | p3nguin: that was not at you... |
| 00:28.53 | p3nguin | oh |
| 00:29.14 | joeyjones | p3nguin: i was layoing out how to get to the setting in x-lite4 |
| 00:29.18 | ShaunR | ah, i'm running version 3 |
| 00:29.20 | joeyjones | *laying |
| 00:29.21 | WIMPy | p3nguin: Should. yes. some seem to have an alzheimers problem when it comes to udp. |
| 00:30.29 | *** join/#asterisk exothermc_ (~miles@74.85.89.146) |
| 00:30.35 | joeyjones | i'd have expected delays from tcp as well as extra overhead |
| 00:31.11 | WIMPy | It certainly can happen. |
| 00:31.16 | joeyjones | and iirc rtp.conf has a setting to prevent missed packets |
| 00:33.14 | cmnky | x86 ... joeyjones ... what up |
| 00:33.29 | joeyjones | yo |
| 00:34.48 | joeyjones | i'm eating; preparing to hibernate |
| 00:34.51 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:36.18 | *** join/#asterisk mountainm2k (~msturtz@192.sub-75-220-90.myvzw.com) |
| 00:38.09 | mountainm2k | Configuration: single PRI, configured as ZIP/g1 (yes, Zap, its old). Single analog extension, ZIP/25, goes to a fax machine. Problem: PRI is down. Question: When somebody tries to call outside, its hitting ZAP/25 -- any ideas why? |
| 00:38.38 | mountainm2k | Minor irritation I guess, office isn't that big, and everybody knows its down -- but I can't figure out WTF is happening. |
| 00:39.19 | WIMPy | You put both interfaces in to one group? |
| 00:39.42 | mountainm2k | I don't think so, but I wouldn't rule it out... |
| 00:39.57 | mountainm2k | [trunkgroups] |
| 00:40.06 | mountainm2k | trunkgroup => 1,24 |
| 00:40.15 | mountainm2k | no 25 there, I don't think :-) |
| 00:40.16 | WIMPy | no. group= |
| 00:40.35 | mountainm2k | under that is spanmap -> 1,1 -- I have no idea what that does |
| 00:40.44 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 00:40.50 | WIMPy | trunkgroups configure NFAS. That's got nothing to do with the groups Asterisk uses. |
| 00:40.52 | ShaunR | whahoo, tcp over xlite :) |
| 00:40.55 | ShaunR | thanks |
| 00:40.57 | *** part/#asterisk ShaunR (~shaun@freenode/sponsor/NDChost.com) |
| 00:41.10 | mountainm2k | R U sure its group= instead of trunkgroup= ??? Its been working for like 4 and a half years as-is :-) |
| 00:41.19 | mountainm2k | guess I could try it, nobody will care :-) |
| 00:41.21 | WIMPy | I am |
| 00:41.38 | mountainm2k | its a PRI, should it be 1,23 or 1,24 ? |
| 00:41.44 | mountainm2k | its really 23B + 1D |
| 00:41.53 | mountainm2k | but the group might want the D as well |
| 00:41.55 | WIMPy | Probably you never had all 24 channels in use so the 25th outbound call hit the fax. |
| 00:42.07 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:42.43 | mountainm2k | OK, changed to group => 1,24 -- same result... |
| 00:42.47 | WIMPy | With a single pri you don;t need to configure trunkgroups or spanmaps. Only needed for NFAS. |
| 00:43.21 | WIMPy | You should have the pri on group=1 and the fxs on group=2 or something. |
| 00:43.47 | mountainm2k | OK, I can do that... The FXS is used only as outbound, nothing calls ZAP/25 |
| 00:43.50 | WIMPy | i think the default is group=63 for no group. |
| 00:43.54 | mountainm2k | (its a fax machine) |
| 00:44.40 | mountainm2k | so the first one for the PRI should be a group, or no? |
| 00:45.14 | WIMPy | You definitely want a group for the pri, yes. |
| 00:46.02 | mountainm2k | OK, so in zapata.conf I have [trunkgroups] |
| 00:46.08 | mountainm2k | and then group => 1,24 |
| 00:46.13 | mountainm2k | I commented out the spanmap |
| 00:46.17 | mountainm2k | same behavior |
| 00:46.25 | mountainm2k | I'm doing "restart now" before trying it -- before you ask :-) |
| 00:46.57 | WIMPy | You have to put a group=63 before the channel=>25. |
| 00:47.05 | mountainm2k | Ideally it would produce a wave-off or a fast-busy |
| 00:47.26 | mountainm2k | Oh... Yeah, I didn't have a second group at all for the fxs |
| 00:47.27 | mountainm2k | one sec |
| 00:47.49 | WIMPy | Yes, so it gets into the one you have. |
| 00:48.10 | WIMPy | All settings are for all following channels. |
| 00:48.26 | WIMPy | Not just the next. |
| 00:48.31 | mountainm2k | yeah, which is odd... I'd call that a bug, heh :-) |
| 00:48.50 | mountainm2k | ok: [trunkgroups] |
| 00:49.00 | mountainm2k | group => 1,24 (24 or 23?) |
| 00:49.04 | WIMPy | It's probably not very intuitive, but it makes sense if you think about it. |
| 00:49.06 | mountainm2k | group = 63 |
| 00:49.13 | mountainm2k | channel => 25 |
| 00:49.14 | WIMPy | And know it :-) |
| 00:49.16 | mountainm2k | look right ? |
| 00:49.35 | WIMPy | yes |
| 00:49.40 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:49.56 | mountainm2k | (sorry for the copy and paste -- our whole service is down, so I'm on my laptop / verizon card, and my desktop computer connected to the pbx -- heh |
| 00:50.44 | mountainm2k | yeah, that's still doing the same thing... Do I need to completely restart zapata (kernel module, etc)? |
| 00:50.51 | mountainm2k | or just restarting asterisk enough? I would assume just * |
| 00:51.09 | WIMPy | No, only the asterisk channel. |
| 00:51.17 | *** join/#asterisk tasca (~tasca@189.73.91.88) |
| 00:51.59 | mountainm2k | I don't think I can pastebin my whole zapata.conf |
| 00:52.10 | WIMPy | Maybe you should pastebin your zapata.conf and the dial from your extensions.conf |
| 00:52.11 | mountainm2k | [trunkgroups] is at the top, followed by [channels] |
| 00:55.46 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 00:57.30 | mountainm2k | ok, sneakering those files to here so I can pastebin 'em |
| 00:57.34 | mountainm2k | thanks for hanging around :-) |
| 00:58.40 | carrar | sneaky |
| 00:58.41 | mountainm2k | zapata.conf: http://www.pastebin.ca/2021686 |
| 00:59.05 | WIMPy | the groups go under channels. |
| 00:59.20 | WIMPy | Your trunkgroups section should be empty. |
| 01:00.08 | WIMPy | the group=63 goes any place between the channel=>1-23 and the channel=>25 |
| 01:00.38 | mountainm2k | oh... Crap, OK |
| 01:00.42 | mountainm2k | Let me change that |
| 01:02.06 | mountainm2k | hey, wow, it works |
| 01:02.27 | WIMPy | And BTW: I don't think you want echocanellation on your fax line. |
| 01:02.30 | mountainm2k | Now hopefully it will *still* work once the PRI comes back up |
| 01:03.35 | mountainm2k | huh... I knew that -- its been working very well though. In fact, every employee has an extension, and a fax estension (ext + 100 = fax) which goes to iaxmodem / hylafax |
| 01:03.53 | mountainm2k | We only *just* got a network scanner -- before that people would fax things to themselves -- lol |
| 01:03.57 | mountainm2k | it got a lot of use |
| 01:08.54 | ferdna | guys what should i use for my vocoder? |
| 01:08.58 | ferdna | http://img94.imageshack.us/img94/3403/grandstreamdeviceconfig.jpg |
| 01:10.30 | WIMPy | Maybe someone should tell them what a vocoder is. |
| 01:10.49 | ferdna | WIMPy, is an encoder isnt? |
| 01:11.02 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:11.10 | ferdna | vocoder... is voice encoder |
| 01:11.49 | drmessano | Codec is a more appropriate term here |
| 01:12.09 | ferdna | my provider only supports G.711 |
| 01:12.19 | ferdna | but it is not listed in the comboboxes |
| 01:12.24 | ferdna | what should i use? |
| 01:12.37 | WIMPy | That's pcma |
| 01:12.54 | WIMPy | Or pcmu |
| 01:15.49 | *** join/#asterisk mountainm2k (~msturtz@192.sub-75-220-90.myvzw.com) |
| 01:16.08 | mountainm2k | Ugh, got kicked off... |
| 01:16.13 | mountainm2k | Thanks WIMPy |
| 01:16.19 | *** join/#asterisk AppLync (~AppLync@mail.mrdavidscarpet.com) |
| 01:16.36 | mountainm2k | what I have now: http://www.pastebin.ca/2021696 |
| 01:17.22 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:17.28 | WIMPy | Looks ok. But you don't need to specify group=1 twice. |
| 01:17.44 | AppLync | where can I enable explicit MWI? |
| 01:18.00 | AppLync | i have a few phones that arent getting MWI, but asterisk is never sending it out. |
| 01:18.06 | mountainm2k | whoops... OK, Deleted the second one. |
| 01:18.34 | *** join/#asterisk Howie69 (~Howie69@65.111.172.189) |
| 01:18.47 | WIMPy | AppLync: You need to configure the mailbox for the peers (where possible). |
| 01:18.51 | AppLync | even the MWI notification says there are no messages, but I can see the message in their mailbox. |
| 01:18.55 | AppLync | permissions problem? |
| 01:19.08 | WIMPy | In the peer definition to be precise. |
| 01:19.10 | AppLync | the peers are configured |
| 01:19.19 | AppLync | okay,let me look |
| 01:19.35 | WIMPy | Are the Vm contexts correct? |
| 01:19.50 | mountainm2k | (I'm going to follow along on this question :-) |
| 01:20.26 | AppLync | vm contexts are correct. all extensions ar ehte setup the same, but one extension works, and one doesnt.. |
| 01:25.03 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:25.36 | *** join/#asterisk KingDavidNYC (~Chris1232@ip68-110-242-224.dc.dc.cox.net) |
| 01:26.57 | KingDavidNYC | Hello everybody |
| 01:27.46 | KingDavidNYC | has anybody used Dialogic cards with asterisk? |
| 01:28.26 | WIMPy | Probably, but is that really what you want to know? |
| 01:28.48 | KingDavidNYC | Definitely totally what I really have to do |
| 01:29.20 | WIMPy | hmm |
| 01:29.32 | WIMPy | just wonders if chan_capi exists for 1.8 |
| 01:29.33 | KingDavidNYC | In fact, the idea is, to used the dialogic cards for incoming calls, and then connect to the agents on the ip phones |
| 01:29.40 | KingDavidNYC | why? |
| 01:29.49 | KingDavidNYC | is that a bad idea? |
| 01:30.08 | WIMPy | No, it most probably the only option you have. |
| 01:30.48 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:31.04 | KingDavidNYC | but, whats the info on the Asterisk+Dialogic? does it works?, is it doable?, have specifically worked with one? |
| 01:31.59 | WIMPy | It's like I just told you. The only way to interface the two will most probably be capi and I'm not sure it exists for Asterisk 1.8. |
| 01:32.22 | KingDavidNYC | i would be using 1.6 |
| 01:32.31 | KingDavidNYC | what is capi? |
| 01:33.03 | WIMPy | The oldest kind of ISDN drivers. Comes from DOS times. |
| 01:35.30 | KingDavidNYC | I can't find much dialogic/asterisk references in google |
| 01:35.41 | KingDavidNYC | tomorrow I am gonna be playing with one card |
| 01:36.42 | WIMPy | The common interface is CAPI. Beyound that the type of card doesn;t matter. |
| 01:37.01 | WIMPy | The CAPI itsels however determines what features are available. |
| 01:37.21 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:38.38 | KingDavidNYC | and you think that with capi I would able to pretty much use any dialogic model card (provided they are full duples)? |
| 01:38.40 | KingDavidNYC | x |
| 01:39.07 | WIMPy | How could they not be full duplex? |
| 01:39.49 | WIMPy | Or is it some old analog stuff? |
| 01:40.34 | WIMPy | No 1.8 support as it seems. |
| 01:40.36 | KingDavidNYC | I mean I am excluding anything half duplex in my comment |
| 01:41.25 | KingDavidNYC | I am finding something about something called chan_dialogic |
| 01:41.54 | WIMPy | Oh, yet another option I havent heard of before... |
| 01:42.07 | KingDavidNYC | If I am able to use the Dialogic cards as if they were Digium cards, that would be so awsome |
| 01:42.25 | WIMPy | No |
| 01:42.30 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 01:42.32 | WIMPy | But that shouldn't matter. |
| 01:44.11 | KingDavidNYC | will I be able to receive incoming calls on the pri, and transfer the calls to the SIP stations? |
| 01:44.39 | WIMPy | I would think so. |
| 01:45.35 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:45.42 | KingDavidNYC | thank you |
| 01:45.49 | WIMPy | can't find any mention of chan_dialogic less than 5 years ago. So that does not look like an option. |
| 01:46.19 | WIMPy | But you should have a capi for your card. That should do with chan_capi. |
| 01:47.00 | KingDavidNYC | chan_capi it would be |
| 01:48.52 | WIMPy | Actually capi has one advantage: You can have multiple applications attached to it. |
| 01:50.53 | KingDavidNYC | Have you used chan_capi? |
| 01:51.00 | WIMPy | yes |
| 01:51.12 | KingDavidNYC | in what? digium cards? |
| 01:51.15 | WIMPy | But that was some years ago. |
| 01:51.42 | WIMPy | That was with an external AVM T1-B. |
| 01:52.22 | AppLync | I am having a problem with my MWI not working. I can see there is a message using voicemail show users, but the MWI sip notify says "Messages-Waiting: no" |
| 01:52.25 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 01:52.38 | KingDavidNYC | mmm..sounds like it is gonna be fun |
| 01:52.49 | AppLync | Could that be a permissions problem on /var/spool/asterisk/voicemail perhaps? ort incorrect owner? |
| 01:52.49 | WIMPy | I still got it, but the problem is that the supplied capi won't work unless you insall a really old version of Linux. |
| 01:53.21 | WIMPy | KingDavidNYC: It's certainly easier with a cheaper card. |
| 01:53.33 | AppLync | MWI does work on some extension, but not others... |
| 01:53.57 | dwayne | KingDavidNYC, I wrote chan_dialogic like 5 years ago. I supported something like 23 different Dialogic cards |
| 01:54.10 | p3nguin | Anyone here ever use Netgear ProSafe switches? I have a GS108 that started behaving very oddly. With five of the eight ports connected and in use, the lights began flashing on and off together steadily and the switch was no longer operational. I started pulling cables and when I got down to only two, it acted normal again. I tried several combinations of connecting cables in an attempt to rule out the problem ... |
| 01:54.16 | p3nguin | ... being caused by just one or two ports, but there was no pattern to make it mess up -- it sometimes took three cables connected, sometimes four before it went into blink mode. I was wondering if anyone here has experienced this and if you found a solution. |
| 01:54.32 | KingDavidNYC | dwayne: where? please give me the links to all that stuff |
| 01:54.38 | AppLync | p3nguin: latest firmware? |
| 01:54.55 | p3nguin | It's a dumb, unmanaged switch. |
| 01:55.05 | p3nguin | I found a thread on the netgear forum that indicates it could be a blown capacitor. |
| 01:55.23 | AppLync | all the prosafe stuff has a lifetime warranty, FYI |
| 01:55.30 | AppLync | are you a powershift partner? |
| 01:55.42 | dwayne | KingDavidNYC, I can't provide links, Digium owns the code |
| 01:56.11 | WIMPy | Didn't I just read that it went to Eicon? |
| 01:56.13 | p3nguin | While it is several years old, it was brand new in the box, and I probably only had it in use for under a month before this happened. |
| 01:56.41 | dwayne | Intel sold Dialogic to Eicon and then Eicon renamed themselves Dialogic |
| 01:57.30 | KingDavidNYC | dwayne: you mean chan_dialogic would not have free downloads of chan_dialogic? |
| 01:57.39 | WIMPy | Ja, I just thought from taking a quick glimps that I read that the licence went that way as well. |
| 01:57.50 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 01:58.07 | p3nguin | applync: The warranty card in the box says five years on the switch, two years on the AC adapter. |
| 01:58.10 | shmaltz | hi everyone |
| 01:58.20 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 01:58.24 | dwayne | no, it was work done for Intel before Eicon bought Dialogic and since Digium owns that code it wouldn't matter anyway because Dialogic can't distribute it |
| 01:59.01 | WIMPy | Sonds like it's lost. |
| 01:59.40 | dwayne | a new, better channel driver could probably be written if someone funded it |
| 01:59.49 | KingDavidNYC | dwayne:ok, can you confirm to me that anyway that it is doable to use the Dialogic cards as if they were regular digium cards, same functionality, like being able to receive incoming calls on the card and transfer the calls to a sip phone? |
| 01:59.59 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 02:00.42 | dwayne | KingDavidNYC, yes |
| 02:01.03 | KingDavidNYC | This customer of mine wants to use his 9 dialogic PRI cards in one box with asterisk |
| 02:01.13 | KingDavidNYC | he makes 500,000 minutes per day |
| 02:01.17 | dwayne | but you wouldn't be really using Dialogic's DSPs, so it would be a very expensive 'digium-like' card |
| 02:01.39 | KingDavidNYC | dwayne: fine, I understand that |
| 02:02.03 | dwayne | I think the new Dialogic has channel drivers for both FreeSWITCH and Asterisk |
| 02:02.33 | KingDavidNYC | dwayne: but the customer already has his dialogic cards, so it is better than telling him that now he has to buy new digium cards |
| 02:03.17 | dwayne | they are either going to pay for new cards or pay for development, unless you find a channel driver |
| 02:03.22 | WIMPy | It's not as if they still cost anything. |
| 02:03.38 | dwayne | I wish they did, I have a closet full of them |
| 02:03.42 | KingDavidNYC | dwayne: any links or any pointers of what I have to do to get one dialogic cards working with asterisk? |
| 02:03.58 | dwayne | KingDavidNYC, can you write C code ? |
| 02:04.01 | KingDavidNYC | they are ok about paying for development |
| 02:04.14 | KingDavidNYC | dwayne: oh yes |
| 02:04.24 | WIMPy | dwayne: If you did it once and got a ton of the stuff, it's rally up to yourself to make then useful, isn;t it? |
| 02:05.02 | WIMPy | But the cards do come with a capi, don't they? |
| 02:05.09 | KingDavidNYC | dwayne: I am not going to write a channel driver, but I can write to the dialogic api |
| 02:06.06 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 02:06.07 | WIMPy | KingDavidNYC: It's not per chance a diva card? |
| 02:06.08 | dwayne | KingDavidNYC, channel drivers are not too difficult. Your hint on getting Dialogic cards to work is you would want to use setuio to feed audio frames between Asterisk and the Dialogic stack |
| 02:08.04 | KingDavidNYC | dwayne: I am a little bit not there yet |
| 02:08.46 | KingDavidNYC | WIMPy: no, these are older cards, I am seeing about the diva cards in google, no chance |
| 02:08.49 | dwayne | You are going to need a channel driver; even if you write something that takes audio frames from Asterisk and gives them to a Dialogic application; you would need a channel driver to trick Asterisk into thinking a channel is there |
| 02:09.16 | KingDavidNYC | dwayne: WIMPy says to use chan_capi |
| 02:09.56 | dwayne | that may work, I don't know. I've never used anything other than chan_dialogic for Asterisk/Dialogic integration |
| 02:10.04 | *** join/#asterisk fluppie (~fluppie@84.107.178.236) |
| 02:10.18 | dwayne | KingDavidNYC, what card(s) are you trying to use? |
| 02:11.09 | WIMPy | Actually the dialogic website has a pdf telling you to use chan_capi with Asterisk. |
| 02:11.40 | KingDavidNYC | dwayne: sorry, dont have that info with me, I am the new asterisk guy in this dialogic dialer company, they have plenty, but they have no clue about asterisk, sufice it to say that the cards are about 5-7 years old |
| 02:12.01 | KingDavidNYC | WIMPy: please give me the link |
| 02:12.07 | WIMPy | So it's simple: If you have a capi, use it. If you don't, buy something else. |
| 02:12.27 | *** join/#asterisk tyman (~tyler@adsl-065-015-255-051.sip.hsv.bellsouth.net) |
| 02:12.38 | KingDavidNYC | WIMPy: I agree |
| 02:12.48 | WIMPy | I just types 'asterisk' in to the search box. |
| 02:13.06 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 02:13.16 | KingDavidNYC | WIMPy: the search box of dialogic.com? |
| 02:13.20 | tyman | what are the equivalents of the sip tools like 'sip show ....' for iax? |
| 02:13.26 | WIMPy | yes |
| 02:13.43 | WIMPy | tyman: Take a guess |
| 02:13.55 | WIMPy | (and use your tab key) |
| 02:13.56 | tyman | iax show... |
| 02:14.02 | dwayne | tyman, use tab completion |
| 02:14.03 | tyman | not coming up |
| 02:14.05 | tyman | i am |
| 02:14.12 | WIMPy | almost. iax2 ... |
| 02:14.23 | KingDavidNYC | WIMPy: I can't find it |
| 02:14.55 | WIMPy | I already closed that tab. Was the 5th link or so. |
| 02:15.11 | tyman | WIMPy: it's not there... |
| 02:15.16 | *** join/#asterisk Godfather_ (~godfather@241.Red-81-35-89.dynamicIP.rima-tde.net) |
| 02:15.18 | Godfather_ | hi |
| 02:15.21 | tyman | that is why i was asking |
| 02:15.31 | WIMPy | tyman: Do you have cahn_iax loaded? |
| 02:15.54 | KingDavidNYC | WIMPy: oh man |
| 02:16.11 | p3nguin | So I opened up the switch to take a look, and exactly like the forum post said, the two green capacitors are swollen and it looks like they might be starting to leak. |
| 02:16.12 | AppLync | I am having a weird issue where some users have VM's in the voicemail box,m but the SIP MWI notify says 0, how can I trouble shoot this? |
| 02:16.34 | dwayne | tyman, "module unload chan_iax2.so" ; "module load chan_iax2.so" ; i<tab> |
| 02:17.20 | tyman | dwayne: unload failed, chan_iax2.so could not be found |
| 02:17.31 | Godfather_ | i upgraded to 1.8, but when i reload sip i get the message " Invalid address for externaddr", i also tried externip with the same result, and this works is 1.4 and 1.6, ? http://pastebin.com/8XSQ8WVB |
| 02:17.33 | dwayne | "module load chan_iax2.so" |
| 02:17.37 | tyman | is it a compile time option (and not default like i thought) |
| 02:17.41 | dwayne | tyman, ^^ |
| 02:18.18 | *** join/#asterisk moy (~moy@CPE0026f3968130-CM0026f396812d.cpe.net.cable.rogers.com) |
| 02:19.07 | tyman | dwayne: the module appears to be straight missing...wont load: Module 'chan_iax2.so' could not be loaded. |
| 02:19.48 | WIMPy | Turn up verbose and debug and try again. |
| 02:20.20 | WIMPy | It might just be an invalig configuration. |
| 02:20.46 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 02:20.54 | WIMPy | damn. two keys aside is getting bad :-( |
| 02:21.24 | dwayne | tyman, ls /usr/lib/asterisk/modules/ |
| 02:21.41 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 02:22.14 | tyman | dwayne: Looks like I should have seen this as a compile time error: http://pastebin.com/BTrRqkA2 |
| 02:22.22 | neurosys | soooo. which genius decided it was a good thing to leave allowgeuest-yes as the default? |
| 02:22.53 | tyman | dwayne: * 1.8.1 |
| 02:23.04 | WIMPy | neurosys: Who didn't realise it is a sample configuration not intended to be used? |
| 02:23.17 | neurosys | no samples compiled. |
| 02:23.52 | tyman | dwayne: iax2 module exists |
| 02:23.59 | neurosys | if it is NOT explicitly set... it is defaulted to YES |
| 02:24.18 | pabelanger | tyman: install openssl-dev |
| 02:24.21 | pabelanger | recompile asterisk |
| 02:24.35 | tyman | pabelanger: ok... |
| 02:24.46 | pabelanger | tyman: what version of Asterisk are you using? |
| 02:24.46 | WIMPy | neurosys: Ah, but IIRC that was changed. |
| 02:25.03 | pabelanger | tyman: $gcc -version |
| 02:25.11 | pabelanger | what's the output |
| 02:25.16 | dwayne | tyman, listen to pabelanger |
| 02:25.17 | neurosys | WIMPy: Wrong. As of 1.8... its ia defauted to YES. |
| 02:25.47 | WIMPy | neurosys: It was changed to default to yes? |
| 02:26.01 | tyman | pabelanger: * 1.8.1, gcc (GCC) 4.5.1 20100924 (Red Hat 4.5.1-4) |
| 02:26.21 | dwayne | KingDavidNYC, how many minutes / month are you dialing ? |
| 02:26.26 | neurosys | WIMPy: NO... It is DEFAULT YES. If you dont explicitly set in sip.conf 'allowguests=no', it will allow ANYONE TO CONNECT. |
| 02:26.27 | pabelanger | tyman: Hmm... might be a good idea to open an issue on the tracker too |
| 02:26.45 | pabelanger | neurosys: Its always be yes by default |
| 02:26.53 | pabelanger | s/be/been |
| 02:27.00 | neurosys | pabelanger: i realize this. but WHY is the question. |
| 02:27.17 | tyman | pabelanger: openssl-devel pkg was not installed...i'll install and rebuild |
| 02:27.27 | pabelanger | neurosys: refer to the asterisk-dev mailing list for previous discussions. |
| 02:27.55 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 02:28.00 | pabelanger | tyman: That will fix the issue, but chan_iax2 _should_ work without it. The issue has to do with the version of GCC you are using. |
| 02:28.01 | Godfather_ | i upgraded to 1.8, but when i reload sip i get the message " Invalid address for externaddr", i also tried externip with the same result, and this works is 1.4 and 1.6, ? http://pastebin.com/8XSQ8WVB |
| 02:28.58 | neurosys | pabelanger: I dont care what the reasons are. That's a nasty security issue. This should be NO by default. |
| 02:29.21 | tyman | pabelanger: ok, I was indeed a little perplexed as to why there was no dependency failures with configure or compile issues if it was required |
| 02:29.22 | pabelanger | so setup a firewall then |
| 02:30.05 | pabelanger | tyman: Ya, like I said, your best to open an issue on the tracker and we can see about fixing it |
| 02:30.32 | tyman | pabelanger: i'll do that |
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| 02:31.10 | pabelanger | & |
| 02:35.58 | dwayne | KingDavidNYC, ping |
| 02:36.48 | KingDavidNYC | hi twayne, the customer says he makes 500,000 minutes per day! |
| 02:37.22 | KingDavidNYC | twayne: they have 18 PRIs |
| 02:37.31 | dwayne | KingDavidNYC, see pm |
| 02:37.58 | WIMPy | You still get PRIs in those amounts? |
| 02:38.19 | KingDavidNYC | the box is about 7 years old |
| 02:38.44 | tyman | pabelanger: chan_iax2 works fine with openssl-devel now installed |
| 02:39.11 | tyman | pabelanger: thanks, i would've smoked a bunch of hours on that |
| 02:39.16 | WIMPy | has seen SDH connecions for less and longer ago. |
| 02:47.57 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 02:55.44 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
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| 03:02.42 | *** part/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 03:03.47 | tyman | This is probably a simple fix, but my google'ing is not turning up an answer that is working; http://pastebin.com/ak37g5q6 |
| 03:04.33 | tyman | damn...forgot to paste error in there... |
| 03:05.16 | *** part/#asterisk n1gak (~lawrence@189.162.186.156) |
| 03:05.32 | tyman | Here it is with iax call token error: http://pastebin.com/NJhhT4hw |
| 03:06.08 | WIMPy | tyman: That message already contains the instructions. |
| 03:06.39 | WIMPy | You can find more information on the subject in the UPGRADE*.txt files |
| 03:06.40 | tyman | WIMPy: set user=jnctn, and requirecalltoken=no as well...same message |
| 03:06.43 | tyman | ok |
| 03:07.10 | tyman | i'll read those...couldn't find the definitive docs on this |
| 03:08.04 | WIMPy | also in doc/IAX2-sexurity |
| 03:11.04 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 03:12.55 | *** join/#asterisk mightydoggy (~mightydog@c-71-200-206-143.hsd1.fl.comcast.net) |
| 03:17.34 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 03:18.10 | tyman | WIMPy: I read the docs...still same error, with setting user=jnctn, requirecalltoken=no (as described in the prior error output pastie) |
| 03:19.40 | tyman | debug output for looks good except for this error (from what i can tell)...looks this is the only problem. shall i paste the debug output? |
| 03:20.54 | WIMPy | you put 'requirecalltoken=no' under [jnctn] or calltokenoptional=66.227.100.30 under [general] |
| 03:23.15 | tyman | under the jnctn user stanza [onsip-gw] |
| 03:23.36 | tyman | is that a general only option? |
| 03:24.03 | WIMPy | both |
| 03:24.09 | atan | Sounds has 'blocked' but no 'unblocked' ? |
| 03:24.11 | *** part/#asterisk mightydoggy (~mightydog@c-71-200-206-143.hsd1.fl.comcast.net) |
| 03:24.15 | atan | Is there something similar in there? =) |
| 03:24.21 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 03:24.26 | WIMPy | But it says user jnctn. |
| 03:24.34 | atan | Your called id has been set to BLOCKED (I need a word to replace BLOCKED with that means unblocked =() |
| 03:25.17 | *** part/#asterisk fluppie (~fluppie@84.107.178.236) |
| 03:25.36 | p3nguin | atan: /var/lib/asterisk/sounds/privacy-not.wav says "not" |
| 03:25.45 | p3nguin | You could do "not blocked" |
| 03:28.23 | atan | Hmm. That will need to do. |
| 03:30.09 | tyman | WIMPy: I got it...i changed my "username=" directive, not the name of my 'user'. This is all clear now to me. Thanks so much for your help. |
| 03:32.15 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:37.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:52.59 | *** join/#asterisk bitplane (~bitplane@91.110.50.190) |
| 03:53.14 | bitplane | hello, world |
| 03:54.43 | bitplane | I have some completely dumb questions... I'd like to set up Asterisk at home and use it to block calls that aren't in my google address book, route outgoing international calls through skype and so on |
| 03:55.17 | bitplane | I'm pretty sure that I can handle all the software side of things, I'm not scared of hacking in almost any language, but I'd like a bit of advice on the hardware side |
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| 03:56.30 | bitplane | what's the smallest, least ugly, Asteroid-based unit-type thing I can buy that I can hack around with its Linux install? I need one input, one output and a LAN connection |
| 03:56.41 | bitplane | I don't really want a desktop in my hallway |
| 03:58.17 | p3nguin | mini ITX? |
| 03:58.31 | p3nguin | Anything Atom based should be okay. |
| 03:59.08 | bitplane | ooh that sounds good |
| 03:59.35 | bitplane | thanks |
| 03:59.51 | bitplane | second question: what sort of magic do I need to plug my analogue phone into it? |
| 04:00.06 | bitplane | I take it standard modems don't work |
| 04:00.52 | p3nguin | Typically they do not, but there used to be some that could. |
| 04:01.14 | p3nguin | For just a single home phone line, I'd go with a Linksys SPA-3102. |
| 04:01.46 | bitplane | cool thanks |
| 04:02.05 | p3nguin | That'll allow you to connect it to your line as well as a phone. |
| 04:02.15 | p3nguin | as well as Ethernet to Asterisk. |
| 04:02.41 | bitplane | Will I be able to write software to do this: http://www.reddit.com/r/linux/comments/emxmb/how_to_troll_telemarketers_with_asterisk_aka/c19d8pi |
| 04:02.54 | bitplane | using that kit |
| 04:03.50 | p3nguin | Without reading the entire thing, it looks like Asterisk is capable of doing all that. |
| 04:04.07 | p3nguin | The only concern is integration of the external phone book. |
| 04:04.42 | bitplane | I'm guessing that Asterisk can be programmed to run some shell / python scripts that will do that? |
| 04:05.15 | bitplane | I don't mind writing the whitelist app using Python and the GData API, and of course sharing the code with everyone else |
| 04:05.52 | p3nguin | You'll be using AGI, probably. |
| 04:06.44 | bitplane | cool, this is looking feasible |
| 04:06.52 | bitplane | I take it it's more than a weekend project? |
| 04:07.01 | *** join/#asterisk Siblor (~putthibon@58.136.24.11) |
| 04:07.11 | p3nguin | You'll probably spend the most time tuning your dial plan logic. |
| 04:07.17 | bitplane | I mean, just the programming part should be easy, but I have no idea about telephony networks |
| 04:07.34 | bitplane | Do I need to use that BASIC style dial-plan language? It looks very restrictive |
| 04:07.49 | atan | Err. sip reload wouldn't cause a sip client on a call to disconnect? |
| 04:08.10 | p3nguin | The standard extensions.conf is the only thing I use for my dial plan. |
| 04:08.45 | p3nguin | But if you've got other ideas, there's probably someone that could help you out. |
| 04:09.33 | *** join/#asterisk m_tadeu (~quassel@89.180.238.29) |
| 04:11.09 | bitplane | I guess I should get a virtualbox set up and mess about with some skype accounts or something, then when I've got the hang of it I can actually buy some hardware |
| 04:11.49 | atan | Sorry, let me change my wording a bit. Would 'sip reload' cause currently connected SIP clients on calls to disconnect? |
| 04:11.53 | *** join/#asterisk morphbot (~morphbot@p57B579B1.dip.t-dialin.net) |
| 04:11.55 | *** part/#asterisk morphbot (~morphbot@p57B579B1.dip.t-dialin.net) |
| 04:12.04 | p3nguin | You can use a software phone to do your testing. I prefer twinkle on Linux or zoiper on Windows. |
| 04:12.17 | bitplane | cool thanks |
| 04:12.26 | atan | bitplane, I've been using Asterisk on a VPS with great luck ^_^ |
| 04:13.16 | bitplane | like, run it on my hosted VPS online? |
| 04:13.31 | p3nguin | You certainly can do that if you want. |
| 04:13.59 | bitplane | hehe that would certainly eliminate the need for a bulky box in the hall |
| 04:14.15 | p3nguin | If you don't need it to be on-site, give it a try. |
| 04:14.45 | p3nguin | As long as your internet connection is good enough, it shouldn't be a problem. |
| 04:14.46 | bitplane | I will, thanks, I'll test on my local box and if it works well I'll shift it to the 'net |
| 04:14.53 | bitplane | yeah that's a risk I guess |
| 04:14.58 | bitplane | is the quality okay? |
| 04:15.31 | atan | I have a crappy small VPS running with ~15 clients connected two or three calls at a time. No issues here! |
| 04:15.33 | p3nguin | The concerns will be latency, available bandwidth, and codecs used. |
| 04:15.38 | atan | Running on debian. |
| 04:15.42 | bitplane | also how much RAM would it use? |
| 04:15.49 | p3nguin | 40 M |
| 04:15.49 | atan | Mine has 256 megs =| |
| 04:16.10 | atan | <3 * |
| 04:16.16 | p3nguin | I run Asterisk on a 933 MHz 512 MB box. |
| 04:16.20 | bitplane | 40M is good, I've got a crummy dreamhost account with a VPS that costs me $5 a month for 300MB... I use about ~150 |
| 04:17.17 | p3nguin | I'm currently using just under 35 MB RAM for asterisk. |
| 04:21.42 | bitplane | so an incoming call would go like this: external line -> SPA-3102 -> internet -> Asterisk on VPS -> my scripts -> Asterisk -> internet -> SPA-3102 -> internal phone line? |
| 04:23.18 | p3nguin | Well, Asterisk will run the scripts via AGI. |
| 04:23.28 | p3nguin | The rest looks like correct call flow, though. |
| 04:24.09 | bitplane | the Linksys VOIP thing acts as a relay in both directions, it's both client and server? |
| 04:24.50 | p3nguin | yes |
| 04:25.25 | p3nguin | It is also capable of pass-through, if the need arises. |
| 04:25.35 | bitplane | is it a bidirectional channel initiated by the VOIP gateway, or do I need to open specific ports? |
| 04:26.07 | *** part/#asterisk Howie69 (~Howie69@65.111.172.189) |
| 04:26.33 | bitplane | yeah the pass-through thing is a major selling point, hopefully I could run it locally in Linux and the phone would still work if I booted into Windows to play a game |
| 04:26.55 | p3nguin | I have one 3102 in a remote site behind a NAT router and I do not forward any ports for that device. |
| 04:27.04 | bitplane | sweet |
| 04:27.21 | bitplane | I'm getting all excited now! |
| 04:27.22 | p3nguin | Asterisk has NAT support for its peers. |
| 04:27.46 | bitplane | downloads virtualbox and asterisknow |
| 04:27.48 | p3nguin | Sometimes it doesn't work, and you have to do some unusual networking tricks, but for the most part it is fine. |
| 04:28.58 | p3nguin | Using AsteriskNOW is probably not what you'll want when you're ready to dive in and get your hands dirty in Asterisk. The fact that it uses a GUI for configuration imposes great limitations. |
| 04:29.30 | bitplane | oh |
| 04:30.10 | bitplane | just download Ubuntu server and apt-get asterisk? |
| 04:30.14 | p3nguin | It's a great system, but it is not as flexible as pure Asterisk when you're using the GUI. You can install it without GUI, though. |
| 04:30.34 | p3nguin | AsteriskNOW 1.8 has the no-gui install option. |
| 04:30.52 | p3nguin | FreePBX, Asterisk GUI, or no GUI |
| 04:31.16 | p3nguin | If you prefer Ubuntu over CentOS, go with Ubuntu and Asterisk. |
| 04:31.33 | p3nguin | If you want CentOS, AsteriskNOW with no GUI would be fine. |
| 04:32.15 | bitplane | is CentOS Red Hat based? |
| 04:32.27 | p3nguin | CentOS is a rebranded RHEL. |
| 04:33.24 | bitplane | ah, no preference then, at least all I know is yum is slightly nicer to use than aptitude |
| 04:33.45 | bitplane | though I've been using Debian based stuff at home for a couple of years now, probably best to go with that |
| 04:34.11 | p3nguin | I personally can't stand Debian or anything that resembles Debian, so I'd use CentOS if those were the only choices I had. |
| 04:34.22 | bitplane | hehe how come? |
| 04:34.59 | p3nguin | It's just not my thing. |
| 04:35.06 | bitplane | Ubuntu really shines as a no-nonsense desktop OS, took about half an hour to get it installed and configured for developing on this new PC |
| 04:35.14 | p3nguin | I run Asterisk on Arch Linux and FreeBSD. |
| 04:35.29 | bitplane | ah.. I see ;) |
| 04:36.09 | bitplane | The barbarian Debian hordes "make it go" unlike like the FreeBSD crowd who "make it right" |
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| 04:51.21 | *** join/#asterisk dalbaech (~dalbaech@host-173-45-208-58.tvmax.net) |
| 04:56.40 | Siblor | Hello. I try to originate to FastAGI instead of exten, AGI run answer() and then exec("Dial", "DAHDI/g0/##########"). After received the call, there was ringing on the other ends with '-- DAHDI/2-1 is proceeding passing it to DAHDI/1-1'. Could you please explain this? I quite not clear about it. |
| 04:59.49 | tyman | I'm not seeing MeetMe when doing a core show application MeetMe |
| 05:00.10 | tyman | any ideas as to why? |
| 05:01.32 | p3nguin | I guess you forgot to build and install it. |
| 05:01.54 | tyman | damn...crazy...thought that was a default...ok |
| 05:02.08 | p3nguin | I'm just taking a wild guess. |
| 05:02.47 | p3nguin | Do you have /usr/lib/asterisk/modules/app_meetme.so ? |
| 05:03.45 | p3nguin | If you do, then the other explanation is that you don't have a meetme.conf or you aren't using autoload. |
| 05:04.38 | tyman | ah....dahdi driver dependency in menuselect....dahdidummy driver for timing...i remember now. just running on a vm for testing...forgot about that |
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| 05:06.57 | drmessano | If you're using 1.6.2 or above ConfBridge is pretty slick |
| 05:08.47 | tyman | drmessano: that looks cool...checking it out |
| 05:08.56 | tyman | running 1.8.1 |
| 05:13.18 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 05:20.04 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 05:23.40 | atan | Anyone have an idea how to enable remote access to voicemail by calling an extension that forwards to voicemail? I'd like to let the user press * to be prompted for a password. |
| 05:24.38 | sawgood | When you say, remote access to voicemail, do you mean ... someone calling a DID number, which is answered by an Asterisk extension? |
| 05:24.46 | p3nguin | You can either call any extension that ends up at a user's voicemail, then press *, or create an extension that runs VoiceMailMain(). |
| 05:28.44 | atan | p3nguin, when I call in I hit a mailbox VoiceMail(1@1000,su), but * does not prompt for a password |
| 05:29.35 | p3nguin | Yeah? The s option skips instructions, so what did you expect? |
| 05:29.47 | WIMPy | You need to enable it in voicemail.conf and create an a extension with voicemailmain. |
| 05:31.50 | atan | p3nguin, voip.info.org says "s: The letter s, if present, causes the instructions ("Please leave your message after the tone. When done, hang up, or press the pound key.") to be skipped. " |
| 05:32.33 | *** join/#asterisk greezmunkey (~GM@173.241.161.62) |
| 05:36.12 | p3nguin | Oh yeah, the a extension. I totally forgot about that. |
| 05:37.03 | p3nguin | exten => a,n,VoiceMailMain(@default) |
| 05:40.17 | WIMPy | better make that 1 |
| 05:40.45 | p3nguin | My a,1 was something else not relevant, so I skipped pasting it. |
| 05:40.53 | atan | Woah. Woah woah woah. Come again? I add this to my context... |
| 05:41.15 | atan | 'I replace Voicemail(1@default) with this line? err, Voicemail->VoiceMailMain? |
| 05:41.18 | atan | tries this |
| 05:41.35 | p3nguin | I forgot that pressing * during the voicemail prompt runs the a extension. |
| 05:41.47 | WIMPy | No, you add the a extension. |
| 05:42.07 | p3nguin | So exten a needs to run VoiceMailMain(). |
| 05:42.31 | WIMPy | It might be handy to put the users mailbox to a variable that can be passed to voicemailmain. |
| 05:43.40 | p3nguin | And that is the source of my complaint from last night when someone brought up voice mail stuff. Now I can fix it. |
| 05:44.14 | WIMPy | complaint? |
| 05:45.08 | greezmunkey | With voicemail, how are the contexts used. I read through the voicemail.conf file, and attempted to use two seperate voicemail contexts, but it failed. In a nutshell how does the voicemail context tie into the dialplan? |
| 05:46.40 | p3nguin | VoiceMail(512@default) or VoiceMail(4321@office2) |
| 05:47.00 | p3nguin | It's just a context pertaining to voicemail. |
| 05:47.35 | *** join/#asterisk coppice (~chatzilla@116.84.115.211) |
| 05:47.50 | atan | p3nguin, could you help me add this in to my dialplan? http://pastie.org/private/ws748o4byt8w9iglhyrq I added it in there as exten =>a,n,VoiceMailMain(1@default) and exten =>123,n,VoiceMailMain(1@default) but neither of them accept the * press while playing the voicemail greeting |
| 05:48.11 | greezmunkey | p3nguin: well, you get another facepalm from me! I read _way_ too much into what I was reading in voicemail .conf. That makes too much sense... |
| 05:48.49 | WIMPy | atan: a,1,... |
| 05:49.38 | atan | tries this facepalm thing |
| 05:51.00 | greezmunkey | get used to it :) |
| 05:53.04 | atan | But now me adding in exten =>a,1,VoiceMailMain(${mailbox}) means when the user presses * anywhere this is going to run though? =S |
| 05:53.39 | p3nguin | Exten a is going to be run in the context where the call is. |
| 05:53.43 | dalbaech | greezmunkey,did you get the answer? |
| 05:53.49 | WIMPy | Anywhere in VoiceMail() |
| 05:53.57 | WIMPy | Or only in the greeting? |
| 05:54.38 | dalbaech | atan, are you trying to get it to go into VoiceMailMain when someone hits *? |
| 05:55.14 | atan | Well I have [inbound] as my context that handles inbound calls, matching phone numbers and calling extensions. I added exten =>a,1,VoiceMailMain(${mailbox}) to the top of it. So I assume if I want to try any fancy magic with asterisk that requires the user to hit * I best make a new context for it? |
| 05:55.41 | atan | dalbaech, yep. Got that part working though. My concern now is when the * will apply. For example, if they are in the IVR and hit * I'd hate to send the caller to voicemail. |
| 05:55.50 | atan | If they hit '0' more than once I think I'll play sexual noises though. |
| 05:56.06 | greezmunkey | dalbaech: yes, it works - I have some cleanup to do in my dialplan, but yes. I'm just glad I washed my hands before I asked that question. |
| 05:56.09 | dalbaech | only when the IVR is in the same context that the call to * is in... |
| 05:56.32 | atan | dalbaech, hmm. How do I seperate what context my calls go to? |
| 05:56.47 | atan | I have a sip trunk which has context=inbound set for it |
| 05:56.48 | p3nguin | a only works from inside voicemail. |
| 05:56.54 | p3nguin | 'a' that is. |
| 05:57.07 | atan | p3nguin, sweet. |
| 05:57.20 | p3nguin | If I am in my menu and hit *, it looks for exten * |
| 05:57.33 | dalbaech | what p3nguin said. :P |
| 05:57.45 | p3nguin | If I am in voicemail greeting and hit *, it runs exten a. |
| 05:58.27 | dalbaech | atan, http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail |
| 05:58.34 | dalbaech | '*' - the call jumps to extension 'a' in the current voicemail context. |
| 05:58.58 | dalbaech | IE: nifty for allowing people to login |
| 06:00.45 | dalbaech | greezmunkey, different voicemail contexts allow for a more organized structure of the voicemail and the directories if you have multiple groups or different companies on the same PBX instance... |
| 06:02.24 | greezmunkey | dalbaech: I need to dig onto this more, but can the voicemail contexts reside in their own databases? |
| 06:03.29 | atan | dalbaech, unless you were to allow sending voicemails to different mailboxes would there ever be a need to use more than one context? |
| 06:03.42 | dalbaech | greezmunkey, define "databases" |
| 06:04.12 | dalbaech | atan, the contexts inside the voicemail definition can be thought of as groups |
| 06:04.34 | greezmunkey | dalbaech: Voicemail(123@companyA) in a database called CompanyA, and so on... |
| 06:04.42 | dalbaech | greezmunkey, yep |
| 06:04.47 | dalbaech | CompanyA is the context |
| 06:05.02 | atan | I wonder if there is any way I can build the voicemail storage into my hosting setup so it does not save on my server, but offloads it to cloud storage |
| 06:05.27 | p3nguin | You're probably not going to be storing the voicemail recordings in a database, though. |
| 06:05.29 | dalbaech | greezmunkey, nifty for separating directory lookups for CompanyA and CompanyB |
| 06:06.01 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 06:06.05 | dalbaech | atan, what type of "cloud"? |
| 06:06.12 | greezmunkey | dalbaech: Effecively isolating them, I guess is what I'm getting at. Excellent, I'm back to the books. Yes, the directories are my primary concern. Thanks. |
| 06:06.13 | atan | dalbaech, mosso |
| 06:06.24 | p3nguin | If you know how to set up file systems, you can put the voicemail anywhere you want it. |
| 06:06.28 | dalbaech | how do you post files to it? |
| 06:06.33 | dalbaech | greezmunkey, yep |
| 06:06.40 | atan | dalbaech, right now I offload recorded calls and faxes to a cloud container for storage and so the user can pull them down via a little web portal |
| 06:06.47 | atan | Now if I could do the same with voicemail that would be too cool. |
| 06:06.59 | atan | I'm thinking about this imap option here... perhaps this is what I'm looking for. |
| 06:07.20 | dalbaech | maybe, but that's a beast I've stayed away from. |
| 06:07.46 | dalbaech | There's real no reason for me to use IMAP when local file storage works fine |
| 06:07.47 | atan | I think showing them a list on a website, letting them tag & add comments, download, forward, e-mail... blah blah would be really neat. |
| 06:07.57 | p3nguin | Just put /var/spool/asterisk/voicemail/ on another machine. |
| 06:07.59 | dalbaech | You can do that with file based storage |
| 06:08.08 | atan | How much space does a voicemail take though? |
| 06:08.31 | dalbaech | I rewrote vmail.cgi to work on realtime DB instead of using config files... |
| 06:08.43 | dalbaech | atan, what duration of recording, what format? |
| 06:08.52 | dalbaech | how many messages can they have at once |
| 06:08.54 | dalbaech | ... |
| 06:08.56 | atan | 10 minutes, default encoding |
| 06:09.04 | dalbaech | You're asking, how large is an audio recording. |
| 06:09.44 | dalbaech | atan, asterisk saves in multiple different formats for the voicemail app. Are you saving as WAV, wav, gsm, etc... Then figure out what the bit rate is, then do some math. |
| 06:10.55 | dalbaech | "format" parameter.... Examples on http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 06:11.39 | p3nguin | I could pull out random voicemail message and check the file size. |
| 06:11.58 | atan | is wav49 the same was WAV? |
| 06:12.08 | dalbaech | gsm: one minute = 469 KB |
| 06:12.13 | dalbaech | sorry... |
| 06:12.21 | dalbaech | wav at ulaw = 469 KB |
| 06:12.27 | atan | I recall using WAV for call recording and had really great file sizes =) |
| 06:12.32 | dalbaech | GSM = 103 KB |
| 06:12.50 | p3nguin | A random one... 43K on my mailbox. |
| 06:13.17 | atan | p3nguin, what format would that sucker be in, and any idea how long it is? =) |
| 06:13.27 | p3nguin | in WAV format, 27 seconds |
| 06:13.54 | dalbaech | I don't know the bitrate of wav49, but I found a guestimate of: 1 mbyte/10minutes |
| 06:14.30 | p3nguin | Another is 52K, WAV, 32 seconds. |
| 06:14.38 | greezmunkey | The wav format, I figured that at about 1Mb per minute. I need to look into this as well, thanks. |
| 06:15.01 | greezmunkey | wav49, I see. |
| 06:15.17 | tyman | p3nguin: what is the best way to tshoot call parking from the cli? I do have parkedcalls included in my dialing context btw... |
| 06:15.31 | p3nguin | Describe the problem. |
| 06:16.21 | tyman | transfer 700 (default within features.conf), just timesout...no output to cli under verbose 5 |
| 06:16.30 | dalbaech | tyman, if the feature context is the same as the extension, you don't really need it in the DP. |
| 06:17.31 | dalbaech | "context=defaultext" in my features.conf allows any extension that's registered in the defaultext context to just dial the feature extension for parking lot... |
| 06:17.53 | dalbaech | what do you have for parkext => and parkpos => in features.conf? |
| 06:17.56 | dalbaech | and context= |
| 06:18.00 | tyman | dalbaech: not sure i understand, I was under the impression (because i'm reading it right now), that i had to include => parkedcalls into my respective dialplan context |
| 06:18.25 | dalbaech | tyman, are they sip buddies? |
| 06:18.34 | p3nguin | It should be fine that you did even if it wasn't exactly necessary. |
| 06:19.20 | tyman | dalbaech: y, the two "phones" are actually just lines on my bria softphone (ext 6001, 6002) |
| 06:19.28 | tyman | both in same context |
| 06:20.36 | dalbaech | tyman, in the sip registration of them, what's the context set to? |
| 06:21.01 | tyman | both [users] |
| 06:21.47 | dalbaech | in sip.conf, is there a line for the SIP registration that says "context="? |
| 06:22.08 | dalbaech | or if using realtime, the context column |
| 06:23.11 | tyman | dalbaech: both sip configs context=users |
| 06:23.29 | dalbaech | ok... well, will any other contexts be using features? |
| 06:23.44 | dalbaech | doesn't really matter right now. |
| 06:23.51 | dalbaech | change context= in features.conf to be users |
| 06:24.16 | dalbaech | then in cli, features reload |
| 06:24.22 | dalbaech | and try calling 700 |
| 06:24.40 | dalbaech | and also you have the parkpos => setup |
| 06:25.01 | dalbaech | sorry; context => users |
| 06:25.16 | dalbaech | in features.conf under where you put parkext => and parkpos => |
| 06:25.22 | tyman | the parkpos is default as parkpos => 701-720 |
| 06:25.27 | dalbaech | k |
| 06:25.52 | tyman | dalbaech: stone stock features.conf (* 1.8.1) |
| 06:26.14 | dalbaech | ok, what's the context currently set to? |
| 06:26.28 | dalbaech | I don't have a copy of the defaults. :P |
| 06:26.43 | tyman | context => parkedcalls |
| 06:27.18 | tyman | include => parkedcalls within dialplan context [users], with matching context=users within sip.conf |
| 06:28.37 | greezmunkey | Thanks p3nguin and dalbaech have to go... |
| 06:29.05 | *** join/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
| 06:29.07 | dalbaech | then it should be able to call 700 if you're including parkedcalls context in the users context.... |
| 06:30.04 | dalbaech | are they listed in "dialplan show"? |
| 06:30.58 | tyman | yes...they call each other just fine |
| 06:31.09 | dalbaech | no, the features extensions |
| 06:31.14 | dalbaech | [ Context 'parkedcalls' created by 'features' ] |
| 06:31.35 | dalbaech | '700' => 1. Park() [features] |
| 06:32.00 | tyman | tma-linux*CLI> dialplan show 700@users |
| 06:32.01 | tyman | [ Included context 'parkedcalls' created by 'features' ] |
| 06:32.01 | tyman | <PROTECTED> |
| 06:32.31 | p3nguin | If you change context to context => users, that's going to change. |
| 06:32.36 | dalbaech | ok, and is 700 repeated in the users extension also? |
| 06:32.44 | dalbaech | extension=context |
| 06:32.46 | tyman | y |
| 06:32.46 | dalbaech | sorry; late night |
| 06:33.05 | tyman | extension=context? |
| 06:33.34 | tyman | 700 is only defined with features.conf |
| 06:33.45 | dalbaech | in my last statement, the word "extension" should have been "context" |
| 06:33.46 | dalbaech | heh |
| 06:33.53 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-visxdizoolbkuxuz) |
| 06:33.56 | tyman | ah...threw me there :-) |
| 06:34.11 | X-Raimo | hello I'm using asterisk 1.8 + TLS. TLS connection is up, echo test runs good and when I try to call to SIP number (he connects using TLS). I got this: http://paste.org/pastebin/view/26257 |
| 06:34.19 | tyman | you're getting tired like me prob? |
| 06:34.42 | dalbaech | tyman, this is really weird behavior... it should work the way it is setup... set verbose higher and see if it gives you anything |
| 06:35.42 | tyman | dalbaech: yeah...i'm stumped...nothing to it really. |
| 06:36.04 | dalbaech | X-Raimo, what is set for the variable tlscipher in sip.conf; and what type of device is it? |
| 06:36.40 | X-Raimo | dalbaech: none is set. Device is PortGo softphone |
| 06:38.15 | dalbaech | http://ibot.rikers.org/%23asterisk/20101016.html.gz |
| 06:38.24 | dalbaech | Timestamp 13:09.26 |
| 06:39.03 | dalbaech | Could that be it? |
| 06:39.09 | tyman | dalbaech: my bria client is on my mac, and my * is on my linux vm and we are bridged interfaces in a mother F'ing crappy motel wlan. see this http://pastebin.com/YkDNbbGB |
| 06:39.53 | dalbaech | tyman, you are a brave soul |
| 06:39.59 | tyman | restarted bria client with same issue on 1 line appearance |
| 06:40.30 | tyman | i'm in huntsville, al |
| 06:40.35 | tyman | dcap tomorrow |
| 06:41.27 | dalbaech | well, try disabling directmedia and directrtpsetup |
| 06:41.58 | dalbaech | in sip.conf (for now since you're probably having fun with NAT issues) |
| 06:42.12 | tyman | no nat on my vm...same net |
| 06:42.39 | dalbaech | does the call start at all or die right after Remotely bridging SIP/tma-polycom-0000000f and SIP/tma-xlite-00000010 ? |
| 06:43.14 | tyman | yeah...it's starts and appears to die after about 15-30secs |
| 06:43.36 | dalbaech | 32 seconds on that one |
| 06:43.37 | dalbaech | should've read |
| 06:43.38 | dalbaech | :P |
| 06:43.56 | tyman | when i xfer to 700 from 1 line to another, says "transferring to 700" .....snor, hangup |
| 06:44.27 | tyman | i'm going to chock this up to bs network for tonight and move on... |
| 06:44.51 | dalbaech | I feel ya |
| 06:44.51 | tyman | thanks dalbaech |
| 06:45.31 | dalbaech | I'm debating on copying my latest backup to my VM at home so that I can relaunch a community PBX I used to run. My hardware is currently in a legal dispute because the colo didn't pay their rent. |
| 06:45.37 | dalbaech | So I've pretty much called that one a loss |
| 06:45.38 | dalbaech | lmao |
| 06:46.29 | tyman | sounds like a hot mess |
| 06:47.29 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
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| 06:51.32 | dalbaech | tyman, indeed. |
| 06:51.33 | dalbaech | G'nite |
| 06:51.43 | tyman | nite |
| 06:51.47 | tyman | thx again |
| 06:53.54 | dalbaech | np, sorry I couldn't be more help. My brain is fried. |
| 06:53.55 | dalbaech | :) |
| 07:01.37 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 07:02.09 | atan | What's wrong with this password? secret=s~#$jmJa2*%(jo8ie5784ae2.mNsh34f& |
| 07:02.24 | atan | It's not my password anymore however the phone won't register with it. Did I use a special char in there or something? |
| 07:02.34 | atan | I'd like to know so I can avoid this trouble in the future. |
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| 07:15.33 | atan | Oh pfft. The trouble is this cisco. grr. |
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| 07:33.42 | atan | I am going to pitch this phone out the window here soon. Only a Cisco would refuse to let line #4 have a secret longer than 10 chars. For fuck sake! =( |
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| 07:34.03 | atan | Of course a wrong password on 1/6 extensions cause the other 5 to go offline. Of course. |
| 07:37.00 | *** join/#asterisk _cassini (~ckwall@c-76-27-9-24.hsd1.ut.comcast.net) |
| 07:37.50 | _cassini | whwnever i hang up a call after leaving a voicemail on my asterisk server, I am getting the following error:-- Got SIP response 603 "Decline" back from x.x.x.x |
| 07:37.54 | _cassini | what could be causing that? |
| 07:38.09 | _cassini | it happens after i leave a voicemail, then the voicemail is never emailed out. |
| 07:38.49 | _cassini | seems to have started after upgrading from 1.6.1.1 to 1.6.2.15 |
| 07:39.06 | _cassini | any help would be appreciated |
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| 07:43.35 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:43.38 | schmidts | good morning |
| 07:50.29 | atan | Morning =) |
| 07:51.01 | atan | What is check_auth: username mismatch, have <411>, digest has <s>? digest (if it means the sip.conf peer) has username=411... how is that ==<s>? |
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| 08:08.35 | atan | http://pastie.org/1384951 says "pbx.c:9595 pbx_builtin_setvar: Set requires one variable name/value pair. |
| 08:08.35 | atan | " even when it is set to just 0. |
| 08:10.52 | kaldemar | atan: the phone sends "s" and asterisk is expecting 411. |
| 08:11.48 | kaldemar | atan: your Set is not setting anything. the syntax is Set(var=value). |
| 08:12.31 | kaldemar | atan: to be more accuerate, the Set's with IF. |
| 08:12.38 | atan | Well wouldn't Set(CIDBlock=${DB(CIDBlock/${CHANNEL:4:4})}0) set it to 0 if ${DB(CIDBlock/${CHANNEL:4:4})} was blank? |
| 08:13.14 | atan | Hmm. There's an idea. |
| 08:13.17 | kaldemar | yes, that one is not your problem. |
| 08:14.03 | atan | ${IF($[${DB(CIDBlock/${CHANNEL:4:4})} != 1]?CIDBlock=${DB(CIDBlock/${CHANNEL:4:4})})} then perhaps |
| 08:14.03 | kaldemar | if CIDBlock is empty, all you have is Set(), which is not valid. |
| 08:14.30 | kaldemar | it doesn't have to be so overly complicated. |
| 08:15.25 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:15.35 | kaldemar | what do you want to do if ${CIDBlock} does not equal to 10? |
| 08:15.50 | atan | nothing =) |
| 08:16.51 | kaldemar | use app ExecIf then. ExecIf($[${CIDBlock} = 10]?Set(CALLERID(name)=anonymous)) for example. |
| 08:23.48 | atan | ExecIf($[${DB(CIDBlock/${CHANNEL:4:4})} = 1]?Set(CALLERID(number)=0000000000)) works great. Thanks! |
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| 08:31.13 | atan | Next issue is chan_sip.c:13413 check_auth: username mismatch, have <333>, digest has <s> |
| 08:31.51 | atan | The phone registers fine but doesn't send in the username when a call is made? =\ err |
| 08:34.25 | kaldemar | it sends a wrong username. check the phone configs. |
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| 08:36.52 | atan | Failed to authenticate device "333" <sip:333@64.120.22.242>. Why is it trying to register at my SIP provider? =\ |
| 08:37.15 | atan | The phone should be connecting to my asterisk box. I don't see how it should even see the 64.X server. |
| 08:37.21 | atan | That's my Asterisk SIP trunk =\ |
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| 08:47.20 | kaldemar | atan: pastebin a sip debug of a call and show your sip.conf for the peer, masking secret. |
| 08:47.38 | atan | I just added in insecure=port,invite and things look good again |
| 08:48.43 | kaldemar | and now anyone can make calls through your box as long as they use username 333. if your network allows anyone to connect, you better fix the real issue and not cover it. |
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| 08:51.22 | atan | Wouldn't they need the secret as well? =\ |
| 08:52.49 | schmidts | atan yes but it this secret good enough to not be hacked by an brute force attack? |
| 08:53.43 | atan | What I'm not following here is why my device (333) can make outbound calls properly but if it calls an 800 number which redirects back to my box I get check_auth: username mismatch, have <333>, digest has <s> everywhere |
| 08:53.58 | atan | schmidts, I hope so? The sip accounts are for peers that needs to call outbound :S |
| 08:55.06 | atan | adding the insecure=port,invite to my sip peer prevents the have <333> message |
| 08:56.29 | kaldemar | atan: no, insecure means that they don't have to authenticate. |
| 08:57.05 | atan | kaldemar, well then that's out of there!! |
| 08:57.11 | atan | strips that line out |
| 08:57.42 | atan | Of course we're back to check_auth: username mismatch, have <333>, digest has <s> now though |
| 08:58.15 | kaldemar | why are 800 numbers coming back to your box? you could already trap them in your dialplan if you want to handle them yourself and avoid redirections. |
| 08:58.25 | atan | I only get the check_auth: username mismatch, have <333>, digest has <s> message if the phone tries to call a number that routes back to the asterisk box |
| 08:58.37 | atan | kaldemar, the 800's are the inbound |
| 08:58.44 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 08:58.55 | atan | I have them inside my [inbound] context |
| 08:58.56 | kaldemar | everything is inbound from asterisk's point of view. |
| 08:59.31 | atan | One sec... WTFIYPB |
| 08:59.33 | kaldemar | put them elsewhere then and include in inbound and other contexts. |
| 09:01.52 | atan | http://pastebin.com/Sd77DdTg is SIP peer, http://pastebin.com/1Q1B8M3S is outbound context it's hitting on |
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| 09:03.16 | atan | Using this I can call an external number, like 19055551212 but any number that routes back to my box (like one of the 800's) gives me the bloody 333 error =\ |
| 09:03.44 | atan | Shows it as trying though... -- SIP/voipms-0000025c is making progress passing it to SIP/333-0000025b |
| 09:04.10 | atan | But when it tries to pass it back the errors show up check_auth: username mismatch, have <333>, digest has <s>, handle_request_invite: Failed to authenticate device "333" |
| 09:09.29 | atan | Are you guys sure using insecure=port,invite |
| 09:09.29 | atan | <PROTECTED> |
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| 09:14.17 | kaldemar | atan: see "insecure=invite" in http://svn.digium.com/svn/asterisk/tags/1.8.1.1/configs/sip.conf.sample |
| 09:15.48 | atan | kaldemar, but does that matter since it's in my outbound context anyway? |
| 09:16.50 | atan | voipms provided me this for my config to send calls to them |
| 09:16.50 | atan | http://pastebin.com/b6ei0KSf |
| 09:18.04 | kaldemar | atan: that's restricted to a host, but asterisk does not indeed require authentication for calls that come from that host and with that username. |
| 09:18.22 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:18.32 | kaldemar | or from that host, disregarding the username. |
| 09:18.43 | atan | Okay. Well going with that thought, since all the other phones seem to work bloddy fine without that insecure line... |
| 09:19.12 | atan | Still the issue of the phone getting Failed to authenticate device "333" <sip:333@64.120.22.242> |
| 09:19.34 | atan | I don't understand why the phone is even trying to contact the 64 ip, again, not mine =\ |
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| 09:23.44 | atan | Well it makes no sense to me. I added allow=all; just before disallow=all; and things are good now. |
| 09:23.52 | kaldemar | atan: because of their redirection. |
| 09:23.54 | atan | shakes head |
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| 09:27.02 | atan | Well this is just bloody odd. If I don't send callerid=Scooby-Doo<8005551212>; or send something bogus, like callerid=Scooby-Doo<123>; it fails with that auth message. |
| 09:27.08 | atan | callerid is required now? O_O |
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| 09:34.04 | kaldemar | atan: it probably affects the matching device when the call comes back to you. a sip debug will be helpful, as i said over 45 minutes ago. |
| 09:35.22 | kaldemar | when you set the caller id as something, the call won't match [333] when it comes back in. you don't have a matching device name then, so it matches by ip to [voipms] which does not require authentication, hence it seems to work. |
| 09:35.35 | atan | Sorry dude ^_^ should I let it rest now that it's working as expected so long as callerid is defined? Or does it warrant more time? |
| 09:35.48 | *** part/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
| 09:36.37 | atan | Woah. What's this about matching device name? Do you mean to tell me if I have a sip peer named [19055551212] and a call comes in to exten 19025551212 that sip peer will pick it up, not the extension? |
| 09:36.51 | atan | rethinks naming things with numbers if this is the case |
| 09:37.12 | atan | I did see something about that inside the readme-SERIOUSLY document but, err |
| 09:38.04 | kaldemar | atan: no. you need to look at the sip.conf.sample i pasted earlier and read the "Naming devices" section. it will clarify how incoming calls (which the redirection from voipms is) are matched to devices ([something]) in sip.conf. |
| 09:40.20 | kaldemar | using numbers as device names MAY cause unexpected behavior in some circumstances, yes. you're seeing some right now. |
| 09:41.42 | atan | Well isn't this neat. I wish I had read all these docs long ago. I honestly can't remember anything these days =( |
| 09:42.19 | atan | I suppose I could rename all my users to PeerN then to help avoid this |
| 09:46.10 | Diffen2 | Hello, is it possible to get a sip-phone to send information to the Asterisk? I mean if you press DND on a SNOM phone that the extension in Asterisk will be set to DND. Not just the phone. |
| 09:50.20 | kaldemar | Diffen2: an extension does not have state in that way. you'd need to make some solution yourself, for example an extension that sets a DND state into astdb and modify the dialing extension to check that state before dialing the device. |
| 09:52.10 | Diffen2 | ok nice thanks kaldemar. |
| 09:52.56 | atan | ExecIf($[${DB(CIDBlock/${CHANNEL:4:4})} = 1]?Set(CALLERID(name)=anonymous)) results in WARNING[31871]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 09:53.26 | atan | Input: = 1 ^ |
| 09:55.08 | kaldemar | ${DB(CIDBlock/${CHANNEL:4:4})} must be empty. throw in some "'s. |
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| 09:57.12 | atan | kaldemar, you are all kinds of special. Seriously. Dude you rock. |
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| 10:33.09 | uzz75 | need help for patton 4634 with 2 Provider... |
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| 10:39.05 | CareBear\ | Hello all! asterisk rocks. |
| 10:39.39 | CareBear\ | What exactly is the use case for chan_mobile in 1.6 addons? |
| 10:39.58 | CareBear\ | "Bluetooth Mobile Device Channel Driver" - well.. |
| 10:40.13 | kaldemar | CareBear\: for example using a mobile phone as a channel in asterisk. |
| 10:40.28 | CareBear\ | by making the PC look like a headset? |
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| 10:41.10 | kaldemar | CareBear\: no, a telephony channel. you can make calls from asterisk to PSTN using your mobile phone and use the phone to receive calls in asterisk. |
| 10:42.38 | CareBear\ | kaldemar : understood, but I'm thinking of what actually gets communicated over BT.. I'm reading the code and trying to figure out which protocols and/or BT profiles are used in which way with the mobile phone |
| 10:43.05 | CareBear\ | kaldemar : it is very nice functionality! I'm curious to understand the details. |
| 10:44.31 | CareBear\ | eh! |
| 10:44.40 | CareBear\ | "Send an SMS message" ?! |
| 10:44.44 | *** join/#asterisk tasca (~tasca@189.4.108.113) |
| 10:45.03 | CareBear\ | well, of course; asterisk does messaging too :) |
| 10:45.33 | CareBear\ | apropos that; are there any good ways to deliver SMS-like messages to a SIP client that has registered to asterisk? |
| 10:45.51 | CareBear\ | it depends on the client of course, but what are the options in general here? |
| 10:49.57 | uzz75 | there are someone expert on Patton device ? |
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| 10:51.04 | kaldemar | CareBear\: last i checked, asterisk didn't support SIP messages outside of a call, maybe XMPP does. |
| 10:56.48 | Chainsaw | kaldemar: So SIP NOTIFY doesn't actually exist? |
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| 10:58.16 | kaldemar | Chainsaw: i meant MESSAGE, not NOTIFY. yes, NOTIFY can probably be used. |
| 10:58.37 | kaldemar | should have written it in caps. |
| 10:59.03 | CareBear\ | kaldemar : ok - and what about messages *within* a call then? |
| 10:59.20 | CareBear\ | kaldemar : that would also work for what I want |
| 10:59.27 | kaldemar | CareBear\: iirc, SIP MESSAGE is supported within a call. |
| 10:59.42 | CareBear\ | ok |
| 10:59.52 | CareBear\ | do clients make sense of it? |
| 10:59.54 | CareBear\ | :) |
| 10:59.57 | kaldemar | but it's been a while since i took a look at that. |
| 11:00.40 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:01.20 | Diffen2 | Does anyone have any good suggestion on how to connect a CRM system with the Asterisk? I have two options I guess. First TAPI and second Asterisk Manager API. I guess if the users arent supposed to keep track on ip-address to their phones the Asterisk Manager API is the way to go right? |
| 11:03.22 | Chainsaw | Diffen2: It depends on how intricately you want to connect the two. |
| 11:03.39 | Chainsaw | Diffen2: You can also log CDR into a database of choice and process that information for CRM usage. |
| 11:04.14 | Diffen2 | Ahh Chainsaw the users want to be able to click on a number and then asterisk should call their phone. |
| 11:04.18 | Chainsaw | Diffen2: But yes, if you want immediate things like turning phone numbers into Name / Company on the display... you will need the manager API. |
| 11:04.30 | Chainsaw | Diffen2: Manager API also seems the best bet there, yes. |
| 11:05.00 | Chainsaw | Diffen2: There are some disgusting hacks you can do if it is a Cisco "SIP" phone like a 7960, but I wouldn't recommend their use. |
| 11:05.02 | CareBear\ | kaldemar : Many thanks for the pointers! |
| 11:05.08 | CareBear\ | have a nice weekend everyone |
| 11:05.09 | *** part/#asterisk CareBear\ (peter@stuge.se) |
| 11:05.39 | Diffen2 | Chainsaw sweet. thanks for you answer. So basically the CRM adds a login and password for the manager api and then every user adds their own extension and password, right? |
| 11:05.57 | Diffen2 | We are only using SNOM |
| 11:06.29 | Chainsaw | Diffen2: Well, the manager API is basically an easy way to send commands to Asterisk from a PHP script. |
| 11:06.52 | Chainsaw | Diffen2: It is then up to either you or the existing CRM code to do something useful with that access. |
| 11:07.16 | Diffen2 | good good. one thing that bugs me a little is that we have multiple tenants and i only want this manager account to see "their" own information |
| 11:09.11 | Chainsaw | Diffen2: I don't believe the manager subsystem supports multiple logins, but you can add this privilege separation in the code blocks. |
| 11:11.12 | Diffen2 | Chainsaw: hmm you mean that I can´t add more then one manager api user? I was planning on adding one extra manager api user for this customer and then let every user logon for them self. So their server have one login and the users uses their phone credentials to logon to receive the correct information |
| 11:13.54 | Chainsaw | Diffen2: No way that I can see, no. |
| 11:15.17 | Diffen2 | Hmm thats not good. Hmm How would I do then? Only one login for the customer at all and then let them see all the data? |
| 11:15.46 | Chainsaw | Diffen2: Don't give them a direct login. Have the CRM backend log into the manager interface and present the data that way. |
| 11:16.33 | Diffen2 | Chainsaw: yes but wouldnt all the data be sent to the CRM backend then? I mean all the calls? |
| 11:17.32 | Chainsaw | Diffen2: The CRM backend would be able to retrieve this information, yes. You can still apply filtering to what you retrieve? |
| 11:18.36 | Diffen2 | Chainsaw: On the CRM side I guess. It would be lovely to set that filtering on the Asterisk side :) |
| 11:19.31 | Chainsaw | Diffen2: Yes. But don't try to make a CRM out of Asterisk or a phone system out of your CRM... |
| 11:19.48 | Chainsaw | Diffen2: Interface the two bits of software, leave them their strong points. |
| 11:20.54 | Diffen2 | Chainsaw: Na the only thing I want to do is to send the CRM system the information that are nessesary for the CRM. |
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| 12:44.35 | m_tadeu | hi...I'm trying to connect my asterisk to a voip provider. I created the user in the database but, when I 'sip show peers' the status of the user is unknown |
| 12:45.19 | m_tadeu | my voip provider says that my asterisk has to register the user in their system. How do I do that? |
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| 12:47.10 | kaldemar | m_tadeu: with a register statement in sip.conf. and the status of such registrations is show with "sip show registry". sip show peers will show what others look like to your system. |
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| 13:02.49 | m_tadeu | kaldemar: thanx for the info...but it's not working yet.... |
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| 13:39.24 | goldroger | hi guys, is this the proper place to ask about dahdi's tor2 driver code ? |
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| 13:40.45 | goldroger | so basically, I was going through dahdi's ph_tor3_e1.c code, and noticed that in tor3_intr, the interrupt handler |
| 13:41.16 | goldroger | we do a transmit output straight before receive input |
| 13:41.38 | goldroger | and in both the cases, we read from tor->mem32, which is a virtual representation of the xilinx memory mapped area |
| 13:41.57 | goldroger | but, if we're writing first, and then reading back right after it, won't we read back just what we have written ? |
| 13:42.04 | goldroger | thanks |
| 13:43.10 | goldroger | or are the writechunk[i] of the channels and the readchunk[i] of the channel mutually exclusive |
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| 13:57.23 | tzafrir_laptop | goldroger, this is the right place. However this tor3 driver is an unsupported fork |
| 13:57.37 | tzafrir_laptop | I'm not sure if anybody maintains it |
| 13:57.55 | tzafrir_laptop | I wonder what it would take to make tor2.c support those devices |
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| 14:15.28 | WIMPy | There is tor3 support in dahdi? |
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| 14:17.16 | WIMPy | goldroger: Where did you find that? |
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| 14:39.28 | goldroger | tzafrir_laptop: I missed the last line you made after 'tor3 is an unsupported fork' |
| 14:39.32 | goldroger | can you please repeat |
| 14:39.32 | goldroger | thanks |
| 14:40.43 | WIMPy | goldroger: Where did you find that? |
| 14:41.41 | goldroger | WIMPy: tzafrir_laptop told me |
| 14:42.03 | goldroger | do you mean where I found the driver for ph_tor3_e1.c ? |
| 14:42.18 | goldroger | it is at the site for phonicEQ |
| 14:43.14 | WIMPy | Yes. I've got someon who uses a tor3, but I could only find some stonage zaptel version for it. |
| 14:43.40 | goldroger | oh |
| 14:43.45 | phix | hi |
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| 14:44.03 | goldroger | well I have the link : www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/dahdi-2.3.0/ph_tor3_e1.c |
| 14:44.19 | WIMPy | The card is being replaced by a digium one thins weekend, but it would be nice to make better use of the tor3 if only for testing. |
| 14:44.26 | goldroger | yes |
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| 14:44.43 | WIMPy | Thanks. Will take a look. |
| 14:45.03 | goldroger | WIMPy: Awould you be aware of tor2_intr - the code in tor2.c |
| 14:45.07 | goldroger | :D |
| 14:45.28 | yonahw | Hi, I am experiencing dropped calls in a blind transfer done over SIP using REFER. I found this to be a reported bug https://issues.asterisk.org/view.php?id=18185 which has been fixed. Checked out 1.8.1, installed and still experiencing the same issue. What should my next step be? |
| 14:45.56 | goldroger | tzafrir_laptop: even if tor3 is an unsupported fork, tor2 does the same thing - so can you please explain why we do a read from tor->mem32 right after writing to tor->mem32 ? |
| 14:46.08 | goldroger | won't we read back the same thing that we just wrote |
| 14:46.09 | goldroger | thanks |
| 14:46.34 | WIMPy | goldroger: Not yet, but I will probably take a look at it as soon as the card is replaced. |
| 14:46.43 | goldroger | oh |
| 14:47.00 | WIMPy | At least if it ends up with me :-) |
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| 14:47.05 | goldroger | hehe |
| 14:47.37 | goldroger | ok np I'll wait for max(tzafrir_laptop replying, you_reading_thecode_and_then helping me out hopefully) |
| 14:49.27 | goldroger | or min |
| 14:50.14 | phix | hi |
| 14:50.21 | goldroger | hi phix |
| 14:50.42 | tzafrir_laptop | goldroger: __iomem volatile unsigned int *mem32; /* Virtual representation of 32 bit Xilinx memory area */ |
| 14:50.56 | phix | what up Mr/Miss/Mrs/Ms/Shim goldroger ? |
| 14:51.16 | tzafrir_laptop | Xilinx is the FPGA chip on that card, I guess |
| 14:51.42 | goldroger | tzafrir_laptop: ok .. but won't writing to it overwrite any data on it that was waiting to be read ? |
| 14:52.00 | goldroger | I hope this is not a lame doubt |
| 14:52.01 | florz | goldroger: it's not memory, probably |
| 14:52.12 | WIMPy | goldroger: That is probably not memory but some hardware register. |
| 14:53.03 | goldroger | ok .. so what ? the overwriting thing still holds right ? |
| 14:53.24 | goldroger | we're both writing to and reading from the same register - even though it's mapped to a virtual memory area |
| 14:53.44 | florz | goldroger: potentially no |
| 14:53.55 | goldroger | why florz ? |
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| 14:54.29 | goldroger | are you saying that the card takes care of it ? |
| 14:54.35 | goldroger | since the registers are in the card ? |
| 14:54.44 | WIMPy | goldroger: It might have completely defferent functions when reading or writing. |
| 14:54.51 | WIMPy | That's it |
| 14:55.09 | goldroger | oh |
| 14:55.34 | goldroger | ok thanks a lot everyone :D |
| 14:55.38 | goldroger | clarifies stuff |
| 14:56.06 | florz | goldroger: I don't have any clue what the specifics in this case are, but writing and reading are completely separate transactions on PCI, and in any case a PCI card can do whatever it likes with data you write to it and reply with whatever data it likes to any single read request |
| 14:56.12 | WIMPy | It is common to have a read-only and a write-only function on the same address. |
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| 14:56.45 | goldroger | ah |
| 14:56.46 | goldroger | alright |
| 14:57.10 | goldroger | thanks again :D |
| 14:57.24 | WIMPy | But even if it was teh same function it might make sense to read a value to check if it has been accepted by the hardware. |
| 14:57.34 | WIMPy | There are quite some possibilities. |
| 14:57.57 | goldroger | hmm |
| 14:58.13 | florz | well, it also may be an actual memory location (as in "a real register") that just gets updated by the host as well as the PCI device itself |
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| 14:58.58 | WIMPy | Or the value might be completely useless and it's only about the fact that a read or write occurs that triggers some function. |
| 14:59.44 | florz | essentially, it's just not necessarily sensible to think of writes and reads to a PCI device as "accessing memory" |
| 15:00.12 | florz | it's just reads and writes as distinct actions that can cause the hardware to do certain things |
| 15:01.04 | florz | it's not about storing data |
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| 15:03.07 | yonahw | anybody have any ideas about my SIP REFER's getting dropped even in 1.8.1. I posted more details above but would be happy to repeat if anyone can shed some light here. |
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| 15:26.14 | leifmadsen | yonahw: use 1.8.2-rc1, not 1.8.1 |
| 15:26.27 | leifmadsen | 18185 was not resolved in 1.8.1-rc1 (which 1.8.1 was built from) |
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| 15:27.48 | tuxx- | hi guys, we have an analog phone connected on a asterisk box with a TDM8 card. We get the following error when the analog phone is calling, the call is silenced for a second, and then continues. Anyone have a clue what this is about? Google gives me nothing. If you need a dahdi debug log, we can supply you with that (have to make it though ;-) http://pastie.org/1382555 |
| 15:28.27 | goldroger | maybe echo cancellation or something ? not sure |
| 15:28.28 | yonahw | leifmadsen: thanks |
| 15:28.36 | goldroger | it might be training or something |
| 15:29.13 | tuxx- | hmz |
| 15:29.41 | goldroger | florz: thanks for that clarification - I was thinking of it as that exactly - as reads/writes into the memory which I shouldn't :) |
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| 15:42.43 | SirDekar | hi, I need help with agenda/calendar can somebody help me? |
| 15:43.23 | SirDekar | I just need to make lists of extensions which server will call to notify something |
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| 15:54.40 | ManWithNoName_ | Hello! Is anyone receiving this message: NOTICE[13179]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '10.1.21.127:4570' |
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| 16:22.39 | asilva | Can someone give me a help, my asterisk is calling to my sip users by itself.. And nothing appears on LOGS |
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| 16:24.19 | asilva | On the ip phone shows <<asterisk>> <asterisk> and when i answer stays mute! |
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| 16:28.46 | mort_gib | asilva: Let the goblinns out of your box :-) |
| 16:29.08 | asilva | that's what i thought so!! :( |
| 16:29.35 | asilva | I enabled sip debug so i'm waiting to call again to see if i catch anything 'cause on verbose and debug nothing shows! |
| 16:30.01 | mort_gib | What version are you on ?? |
| 16:32.20 | asilva | mostly servers 1.6.2.14 and .13 |
| 16:34.16 | mort_gib | And re-invite always off |
| 16:34.31 | asilva | let me check |
| 16:34.34 | asilva | allowguest=no |
| 16:35.30 | mort_gib | And you see NOTHING in the cli?? |
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| 16:37.08 | asilva | nothiig |
| 16:37.09 | asilva | no dials |
| 16:37.12 | asilva | no redirects |
| 16:37.17 | asilva | nothinggg |
| 16:37.31 | asilva | i try to match time/date nothing!!!! |
| 16:38.14 | mort_gib | call logs |
| 16:38.50 | asilva | full log |
| 16:39.07 | mort_gib | no call logs, not /var/log/asterisk/messages |
| 16:39.23 | asilva | you mean cdr ? |
| 16:39.28 | mort_gib | Yes |
| 16:39.31 | asilva | wait |
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| 16:42.34 | asilva | mort_gib, also nothing on CDR |
| 16:43.04 | asilva | I hopping to happen again so i can catch on sip debug |
| 16:43.52 | mort_gib | Yeah, well I'm out of ideas |
| 16:44.25 | mort_gib | Are you using the Nice Noname Chinese phones I stumbled over last week?? |
| 16:44.27 | asilva | yeah i'm stuck too |
| 16:44.43 | asilva | i use Yealink phones and some Linksys Spa942 |
| 16:44.49 | mort_gib | iftop.... |
| 16:44.53 | mort_gib | tcpdump |
| 16:45.19 | mort_gib | Ok... :-) I found a box of "IP phones" @ a clients |
| 16:45.28 | asilva | ? |
| 16:45.34 | mort_gib | Not even a trace of a name on them |
| 16:45.49 | asilva | damn |
| 16:45.50 | asilva | strange |
| 16:45.54 | mort_gib | They did very advanced stuff |
| 16:46.10 | mort_gib | Like knowing in advance when a conversation was really over |
| 16:46.17 | mort_gib | and aotumatically hang up |
| 16:46.43 | mort_gib | They also kept staff awake, by making small merry sounds from time to time |
| 16:46.55 | asilva | really? |
| 16:46.56 | mort_gib | -I HAD to try them out :-) |
| 16:47.12 | mort_gib | My client has some 45 Polycom phones on their system |
| 16:47.37 | mort_gib | They don't know when conversations are over, you have to hang up yourself |
| 16:47.55 | mort_gib | I'll writing to polycom about their shortcomings |
| 16:47.57 | mort_gib | :-) |
| 16:48.17 | mort_gib | I feel Polycom will want to know |
| 16:49.50 | mort_gib | Seriously though |
| 16:50.09 | mort_gib | I had those problems with my noname phones |
| 16:50.38 | mort_gib | Anyway, best of luck |
| 16:51.37 | asilva | Anyone has any ideas about what's happening with me?! |
| 16:51.52 | asilva | Asterisk calling to sip users and no log apears (anywhere) |
| 16:54.03 | cmnky | anyone know the bandwidth required (1 way x 2) for the major codecs in the US ? |
| 16:55.01 | path | any suggestions to grab DMTF stuff? |
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| 16:56.05 | AgentMadsen | path: what do you mean? |
| 16:56.14 | AgentMadsen | path: enable 'dtmf' logging in logger.conf |
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| 17:02.20 | path | thanks AgentMadsen :) |
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| 17:14.27 | Katty | gooooooooooooood morning! |
| 17:14.53 | thehar | Katty! |
| 17:14.54 | thehar | hugs |
| 17:14.58 | Katty | hugs thehar |
| 17:15.00 | thehar | yay |
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| 17:30.35 | carrar | HARRO |
| 17:34.14 | drudge` | hi |
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| 18:01.10 | mducharme-laptop | afternoon |
| 18:01.36 | mducharme-laptop | what are the possible causes of a handset failing to register, sip 403 forbidden error in the sip debug log |
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| 18:02.49 | syncer | hi |
| 18:02.59 | syncer | i have installed asterisk 1.6.2.15 |
| 18:03.19 | syncer | freepbx and also free fax for asterisk |
| 18:04.09 | syncer | all seems to be fine, but if asterisk detect cng it redirect call |
| 18:04.14 | syncer | <PROTECTED> |
| 18:04.15 | syncer | <PROTECTED> |
| 18:04.23 | syncer | and hangs up |
| 18:04.47 | syncer | can someone point me in correct direction how to solve this? |
| 18:09.22 | *** join/#asterisk [T]ank (~T]ank@206.71.78.158) |
| 18:10.07 | [T]ank | so, i upgraded from 1.6.11 to 1.6.2.15, but after upgrade, my voicemail does not send email anymore... i havent got a clue where to start looking... ideas? |
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| 18:34.17 | doctorray | Hello.. looking to experiment with beeping during recording (every 15 sec, etc).. I've played with the announcement method described in the wiki using a custom beep file I've created, etc.. it works, but for the duration of the beep, though short, it cuts out audio passing though, so words can be missed.. I'm looking for a way to bridge it into the call instead.. has anyone had the chance to play with this? |
| 18:40.09 | drmessano | I sure wish there was some way of working around the GV issue where if you're logged into GMAIL, you can't make outgoing calls. |
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| 18:43.26 | [T]ank | i upgraded from 1.6.11 to 1.6.2.15, but after upgrade, my voicemail does not send email anymore... i havent got a clue where to start looking... ideas? |
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| 18:49.11 | doctorray | [T]ank: have you tried sending mail from the command line outside asterisk? something else may have happened |
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| 18:49.58 | joel_oliveira | hello all |
| 18:49.59 | [T]ank | good idea... testing |
| 18:51.49 | joel_oliveira | I have installed the Asterisk 1.6.2.15 and now I am bumping into a problem regarding the qualifyfreq option. I changed it in sip.conf to 60 and other values but it still does a check in 12 seconds interval. doesn't matter how value I change it, it's always made a check on a client each 12 seconds |
| 18:52.23 | joel_oliveira | I checked on the peer by doing sip show peer XXX and the value that I define for qualify frequency is right |
| 18:52.38 | joel_oliveira | does anybody has any clue for what can be wrong? |
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| 19:20.16 | [T]ank | what email does asterisk use by default? sendmail? |
| 19:22.49 | citywok | [T]ank: voicemail.conf "mailcmd" |
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| 19:24.03 | [T]ank | should i see something in the cli> when it i leave a voicemail saying that an email is being sent? |
| 19:24.32 | citywok | no |
| 19:25.09 | citywok | at least i've never seen anything with verbosity cranked up, it may show in the debug log files if you enable them, but i've never needed to look. |
| 19:25.20 | [T]ank | im not seeing anything in /var/log/messages regarding email either. Where would i find soemthing that says "hey dumb-a, if failed" |
| 19:25.41 | citywok | you can try enabling debug in logger.conf and watch that log file to see more info. |
| 19:25.58 | [T]ank | what would be a good command line test? |
| 19:26.11 | citywok | where you would see it fail depends on your configuration. i use asterisk + postfix which works pretty well without any hassle. |
| 19:27.26 | [T]ank | odd thing is that this worked until i upgraded from 1.6.11 to 1.6.2.15 |
| 19:27.30 | [T]ank | then it seemingly stopped |
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| 19:43.36 | *** join/#asterisk fofware (~Fabian@host71.200-82-55.telecom.net.ar) |
| 19:44.18 | fofware | hello any one know if extensions.lua work on openwrt? |
| 19:44.39 | Qwell | if you built lua support, sure |
| 19:45.23 | fofware | Qwell: I suspect is a parameter in make file, right? |
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| 19:55.29 | [T]ank | i am getting no where on this voicemail to email thing. i am not super savy with the linux and getting flamed in the linux channels. anyone able to walk thorugh this with me? |
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| 20:06.07 | _Corey_ | [T]ank: You need to check to see what your server is running, could be sendmail or postfix or something else... /var/log/maillog is probably going to be helpful in seeing what's up |
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| 20:17.25 | fofware | How I know if my asterisk support extension.lua? |
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| 20:28.45 | Katty | hi |
| 20:28.48 | Katty | i had a nap |
| 20:29.39 | Qwell | my turn? |
| 20:29.46 | Katty | kay |
| 20:29.52 | Katty | hands Qwell pillow |
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| 20:33.11 | drmessano | steals said pillow |
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| 20:49.57 | bkruse | SCF steering committee has a guy named Ed Guy, who is from e-MC Software - is that emc.com ? |
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| 20:52.44 | Qwell | no |
| 20:52.58 | Qwell | EMC ~= vmware |
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| 21:16.54 | bkruse | Qwell: Darn, I was hoping to get a deal on some VMAXs :[ |
| 21:20.27 | Qwell | bkruse: I know a guy. Go to vmware.com, tell them Qwell sent you. Instant $0 off. |
| 21:22.06 | drmessano | Windows Virtual Server FTW |
| 21:22.28 | drmessano | Make sure it says "R2" in the box.. that's the new "Pro" |
| 21:22.47 | bkruse | Qwell: hahaha, exactly! Instant $0 off of 2 million dollars |
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| 21:34.06 | bkruse | Qwell: Those vmax boxes are _insane_ though |
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| 21:40.10 | dalbaech | yeo |
| 21:45.29 | CheBuzz_Home | I have an asterisk install on a public IP, and am trying to configure that with an ATA behind a NAT. Everything seems to register just fine, but when I try to make a call, it just falls through. Here is the sip http://pastebin.com/ghK9ycmP Any ideas what is going on? This one has me stumped. |
| 21:46.04 | dalbaech | CheBuzz_Home, want an easy way? |
| 21:46.11 | CheBuzz_Home | Sure |
| 21:46.14 | CheBuzz_Home | Easy is good. |
| 21:46.44 | dalbaech | setup port forwarding for whatever is setup on the * box in rtp.conf |
| 21:48.22 | CheBuzz_Home | I'm not sure I understand. Port forwarding on the public IP? Because that is un-firewalled. |
| 21:48.33 | dalbaech | no, on whatever router it's behind.... |
| 21:48.34 | kaldemar | or set nat=yes and qualify=yes for the peer in sip.conf |
| 21:48.44 | seanbright | ~sipnat |
| 21:48.44 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:48.48 | dalbaech | or what kaldemar said, but it doesn't always work |
| 21:49.01 | CheBuzz_Home | dalbaech, unfortunately that's not an option for me, it's a shared router that I don't control. |
| 21:49.12 | dalbaech | have fun if the sip NAT doesn't help. |
| 21:49.15 | dalbaech | Try using IAX |
| 21:49.16 | CheBuzz_Home | kaldemar: I have set both those options. |
| 21:49.22 | dalbaech | it's better with evil NAT situations |
| 21:49.36 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 21:50.29 | dalbaech | IAX uses the same connection for signaling and payload |
| 21:50.36 | dalbaech | so it theoretically works better |
| 21:50.49 | CheBuzz_Home | I love IAX, but this ATA doesn't support it. |
| 21:51.00 | CheBuzz_Home | The thing I don't get is why the call just kind of goes into the ether. I don't see an error or any indication of why it fails in the sip debug |
| 21:51.01 | dalbaech | sounds like you're going to have some fun. |
| 21:51.18 | CheBuzz_Home | :) Can't think of a better way to spend my day.... |
| 21:51.19 | dalbaech | it goes into the ether because the NAT router doesn't know what to do with itr |
| 21:51.30 | dalbaech | CheBuzz_Home, start drinking. |
| 21:51.31 | dalbaech | :P |
| 21:51.41 | CheBuzz_Home | Haha, that will surely help! |
| 21:52.05 | dalbaech | try STUN |
| 21:52.11 | CheBuzz_Home | So you are saying that the SIP handshake completes, and then when the RTP stream tries to start, it dies? |
| 21:52.49 | dalbaech | yep |
| 21:52.57 | dalbaech | because the router has no idea they're associated with each other |
| 21:53.03 | dalbaech | welcome to my world. |
| 21:53.09 | dalbaech | I work for a VOIP provider |
| 21:53.10 | dalbaech | :/ |
| 21:53.18 | CheBuzz_Home | I even tried going through OpenVPN to that box, but that was even worse! I couldn't even get it to register. |
| 21:53.28 | dalbaech | we use rtp proxy for some things |
| 21:53.32 | CheBuzz_Home | Does * provide STUN service? |
| 21:53.33 | dalbaech | but in general, nat is evil. |
| 21:53.37 | dalbaech | no |
| 21:53.47 | dalbaech | ~stun |
| 21:53.47 | infobot | it has been said that stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs, or a client side method to cater to crappy sip servers, or a phaser setting |
| 21:54.20 | CheBuzz_Home | I, for one, will be gratefully welcome our IPv6 overlords. |
| 21:54.44 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 21:57.22 | dalbaech | well, google for public STUN |
| 21:57.24 | dalbaech | then try using one |
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| 22:05.43 | *** join/#asterisk diemos (~diemos@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 22:06.03 | dalbaech | CheBuzz_Home, if that didn't work, try turning off directrtp/directmedia |
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| 22:06.04 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f2-69-94-238-26.totalprocess.net) |
| 22:06.18 | Letoric | Evenin folks. |
| 22:06.26 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 22:07.00 | Letoric | Does anybody have some good reference material for configuring call monitoring to allow somebody to opt out a specific call (inbbound or outbound) from the monitoring? |
| 22:07.03 | diemos | Evening, Letoric . |
| 22:07.24 | Letoric | We have, by default, all calls monitored, and the executives want the ability to push a code and turn it off even in the middle of a call |
| 22:08.19 | Letoric | I'm running 1.6.2.11, in case there are some version specific differences for this |
| 22:08.32 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 22:08.34 | dalbaech | Leddy, |
| 22:08.37 | dalbaech | sorry |
| 22:08.51 | dalbaech | Letoric, you might want to enable it, but then setup a feature code to disable it. |
| 22:09.12 | dalbaech | However, if you're in trading or anything like that, disabling the recording actually makes it look bad. |
| 22:09.19 | Letoric | I'm not aware of how to set up a feature code that they can push during a call |
| 22:09.27 | dalbaech | You might consider making DIDs or outbound that aren't recorded |
| 22:09.34 | Letoric | nah, not in trading, they just want some of their legal calls etc.. to be not monitored |
| 22:09.35 | dalbaech | ~automon |
| 22:09.39 | dalbaech | tried. lmao |
| 22:09.55 | dalbaech | well, I'd setup DIDs or access codes to not be recorded |
| 22:10.10 | dalbaech | for example, if it comes in on a IVR and then they dial it, if that has automon on.... |
| 22:10.14 | dalbaech | turn it off before Dial() |
| 22:10.37 | Letoric | yeah, I understand how to do it before the call comes in/goes out |
| 22:10.45 | Letoric | just trying to comply with their request of 'during' |
| 22:10.46 | dalbaech | can't really do it mid-stream unless you used automon in features.conf to record it |
| 22:11.03 | dalbaech | unless automon turns off mixmontor when its turned off |
| 22:11.06 | diemos | So, I'm new to Asterisk CLI. I've downloaded StarfishPBX and I've got SIP trunks setup on it which seem to be fine, however any time I try to make a call in/outbound, it immediately drops the call with a fastbusy. I've got this log from asterisk cli interface: http://pastebin.com/N6jq4TUF |
| 22:11.27 | Letoric | is automon an application, function, or 3rd party? |
| 22:11.47 | Letoric | nm |
| 22:12.15 | diemos | The call is also dropped when trying local extensions. |
| 22:12.36 | dalbaech | Letoric, find it? |
| 22:12.48 | dalbaech | I think the monitor in features might toggle mixmonitor |
| 22:12.50 | dalbaech | I'm not sure |
| 22:12.57 | dalbaech | if it doesn't, I can patch it and send the patch |
| 22:13.06 | dalbaech | I'm not putting it to bugtracker |
| 22:13.17 | dalbaech | because everything I've ever submitted, someone else took credit for. |
| 22:13.18 | dalbaech | :P |
| 22:13.30 | dalbaech | IE: realtime meetme support |
| 22:13.36 | Letoric | I found automon, on voip-info.org |
| 22:13.44 | Letoric | not sure I've found the 'how' yet, but at least a starting point |
| 22:13.45 | dalbaech | ok... look in features.conf |
| 22:13.56 | dalbaech | find the feature code for monitor |
| 22:13.59 | dalbaech | or mixmon |
| 22:14.02 | dalbaech | ... then enable it |
| 22:14.15 | dalbaech | see if a call will toggle mixmonitor off when it's pressed |
| 22:14.25 | dalbaech | you'll need the correct options in the Dial() |
| 22:14.27 | dalbaech | to enable monitor |
| 22:14.33 | dalbaech | to be able to toggle it |
| 22:14.39 | dalbaech | if it allows it anyway |
| 22:15.07 | Letoric | I've always done the mixmonitor stuff as a separate line, never used the options in Dial() for it |
| 22:15.11 | Letoric | will see what I can do ;) Thanks |
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| 22:15.57 | dalbaech | Letoric, for features, the dial() needs to be told it can use them |
| 22:16.31 | dalbaech | Ww |
| 22:16.34 | dalbaech | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 22:16.42 | dalbaech | # W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 22:16.42 | dalbaech | # w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 22:17.14 | dalbaech | My ATA won't register.... |
| 22:17.17 | dalbaech | but I don't have Internet. |
| 22:17.20 | dalbaech | (that's my day.) |
| 22:17.29 | dalbaech | eek, I need more vodka. |
| 22:17.36 | Letoric | ;) |
| 22:25.56 | dalbaech | My favorite one.... |
| 22:26.00 | dalbaech | I need to have electricity? |
| 22:26.04 | Letoric | Heh |
| 22:26.05 | dalbaech | (seriously.) |
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| 22:31.34 | diemos | can someone help me make sense of this? http://pastebin.com/N6jq4TUF |
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| 22:33.40 | cmnky | this $5 bluetooth headset works pretty good |
| 22:33.53 | Letoric | ok dalbaech - that worked....but not exactly ;) |
| 22:34.06 | cmnky | although i dont have any point of reference, since its my first one |
| 22:34.09 | Letoric | It did indeed start/stop a new instance of mixmonitor/monitor, but it didn't stop the original one |
| 22:34.42 | dalbaech | fuck |
| 22:34.43 | dalbaech | well |
| 22:34.49 | dalbaech | pardon my language. |
| 22:34.57 | bmoraca_work | how do you hang up a SIP channel from the CLI in 1.6.2? |
| 22:34.57 | Letoric | np |
| 22:35.00 | dalbaech | you might have to have specific dids that don't record. |
| 22:35.07 | dalbaech | or patch it |
| 22:35.11 | dalbaech | and I'm not in the mood. |
| 22:35.13 | dalbaech | :P |
| 22:35.14 | bmoraca_work | nm, got it |
| 22:35.32 | Letoric | ok, well thanks anyway! |
| 22:35.37 | dalbaech | well |
| 22:35.40 | dalbaech | know any C? |
| 22:35.44 | Letoric | nope |
| 22:35.50 | dalbaech | d'oh |
| 22:35.56 | dalbaech | check PM |
| 22:35.57 | Letoric | we have a programmer on staff, but I don't think they'll allocate him to this task |
| 22:36.31 | cmnky | they lock the C coders to the floor of their cubicle with shackles |
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| 22:56.21 | CheBuzz_Home | Ok, this is odd. I have quite a few extensions in extensions.conf, but dialplan show only lists s, NoOp. Why would this be? extensions.conf is owned by asterisk.asterisk, and it is 750 |
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| 23:21.56 | russellb | CheBuzz_Home: run "dialplan reload" at the CLI and see what errors you get. |
| 23:22.27 | recurs|ve | im having some odd things go on with all our phones (polycom ip560 and ip670). when any call is connected, there is up to a 5 second delay before each party can hear eachother, i can gist anything necessary to get this fixed, but im at a huge loss |
| 23:22.32 | thehar | [ERROR] chan_russell.so Failure |
| 23:23.28 | diemos | recurs|ve: I'd probably do a packet capture and see if there's anything odd in the SIP messages. |
| 23:24.06 | recurs|ve | diemos: im wondering if this would have anything to do with sharing the pri with data? |
| 23:24.32 | recurs|ve | but i will try your suggestion |
| 23:27.26 | diemos | That could be a QoS issue if it's over a PRI, any jitter/distrotion? |
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| 23:27.57 | recurs|ve | diemos: no |
| 23:29.20 | dalbaech | ok, I'm out of here for a virgil. http://blogs.houstonpress.com/hairballs/2010/12/aaron_scheerhorn_28_bayou_body.php |
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| 23:30.02 | recurs|ve | diemos: every minute to two we get dropped calls though, the odd thing is this never happened before until we updated the server and phone rom |
| 23:30.23 | recurs|ve | so i think it has to do with the phone rom, but i may just be guessing here |
| 23:31.08 | goldroger | hi, is there an rfc describing the tor spec, etc on which the tor2/3 cards etc are based ? |
| 23:31.33 | diemos | at least you can make calls -.-; my pbx keeps dropping mine and I can't figure out why |
| 23:31.41 | diemos | fastbusy all day D: |
| 23:32.09 | recurs|ve | diemos: well the odd thing is sometimes i have no problems at all, but this shit is whack |
| 23:33.36 | diemos | welcome to my world |
| 23:33.47 | diemos | boss throws a new PBX at me at least once a week |
| 23:37.18 | WIMPy | recurs|ve: When and where exactely does the problem arise? |
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