| 00:09.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 00:18.39 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 01:22.57 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 01:23.32 | saint_ | hi all ... how can i find out if a call is using g729 or g711 ..? is there a command line to enter in the console for that ? |
| 01:24.47 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 01:28.51 | carrar | ore show channel blah |
| 01:28.55 | carrar | core show channel blah |
| 01:30.55 | carrar | maybe even: asterisk -rx "sip show channels" | grep ACK |
| 01:36.32 | saint_ | I'm making a test.. if I have one dialplan that is _1xxxx and another one that is _12345 ...... if I dial 12345 , will my call take the 1st route or the 2nd route ? It looks like it goes to the 1st one with multicards all the time .. |
| 01:37.54 | WIMPy | It will use the more specific match. |
| 01:38.11 | WIMPy | Note that there's no need for an underscore if there's no pattern. |
| 01:38.16 | saint_ | WIMPy: never mind, I forgot the priority between the _xxxx and Dial ... |
| 01:38.29 | saint_ | it s working now, thanks |
| 01:45.18 | saint_ | if I have a peer in sip.conf with allow=ulaw and allow=g729 , or do i prioritize g729 ? |
| 01:46.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 02:03.54 | saint_ | i have disallow=all / allow=g739 / allow=ulaw ... and it goes to ulaw all the time .. |
| 02:04.01 | saint_ | s/739/729 |
| 02:09.18 | WIMPy | Does the peer support G.729? What's it's preference? |
| 02:13.33 | saint_ | yes |
| 02:13.42 | saint_ | if i remove ulaw, the call goes in g729 |
| 02:14.31 | WIMPy | Then the peer obviousely prefers ulaw. |
| 02:15.02 | saint_ | when i do tcpdump traces , is the order the media attribute is presented in the SDP , the order of priority ? |
| 02:16.33 | saint_ | i guess I'll have to buy another g729 license. damn it. |
| 02:25.01 | DBordello | I am trying to get a Polycom phone to pull configuration from a central tftp server 192.168.1.50. What should I use for my DHCP-option 66? "tftp://192.168.1.50"? |
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| 05:31.17 | techman97 | hey all - anyone dealt with a 2-nic multi-homed system (*1.8) with two 10.x.x.x/24 networks - RTP not working properly? |
| 05:31.24 | techman97 | one-way audio. |
| 05:31.28 | techman97 | no NAT involved |
| 05:34.07 | *** join/#asterisk hystryfe (~hystryfe@unaffiliated/hystryfe) |
| 05:34.59 | hystryfe | hey! is anyone intimately familiar with Q.931? |
| 05:35.11 | hystryfe | i just have a quick question on the standard |
| 05:35.46 | gnudna | techman97, do you have 2 different gateways? |
| 05:36.23 | gnudna | had similar issue a while back which snat was able to fix on iptables |
| 05:36.35 | techman97 | yeah, two diff. gateways. |
| 05:36.49 | techman97 | one is a regular inet cnx, other is private SIP pipe. |
| 05:36.54 | gnudna | aka make sure the traffic is going out from the same ip it came in |
| 05:37.15 | techman97 | I didn't think of using snat in iptables... |
| 05:37.22 | techman97 | hmmmm |
| 05:37.33 | gnudna | just an idea |
| 05:38.24 | gnudna | we had pub ip -> fw -> asterisk |
| 05:38.49 | gnudna | but goin out it went asterisk -> fw -> primary public interface |
| 05:39.07 | gnudna | aka not the same one traffic for asterisk was coming in from |
| 05:39.41 | gnudna | well i best get going late and i need sleep hopefully reference works for you |
| 05:40.04 | techman97 | sounds good man |
| 05:40.05 | techman97 | thanks! |
| 05:40.16 | gnudna | later |
| 05:40.20 | *** part/#asterisk gnudna (~sklav@unaffiliated/sklav) |
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| 05:42.36 | hystryfe | anyone familiar with D channel signalling? |
| 05:47.07 | hystryfe | i just want to know if the ALERTING message is mandatory |
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| 06:18.56 | din3sh | hystryfe: what ALERTING? |
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| 07:20.01 | gdeeble | Can anyone tell me, how to you have a dial plan, call an extension that is not a device? I have for example, *1234 as an internal extension for the server to restart asterisk, is there a way so I can have my dial plan from 1 extension to call *1234? |
| 07:21.52 | gdeeble | Main thing I'm looking at is, when I'm away and the phones are working exactly right for my family, I call the phone, press a key combo before it begins calling the phones to drop me to a menu, where I dial in *1234 for example, and it transfers me to that dial plan and will allow me to proceed with the reboot/reload asterisk. |
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| 07:34.40 | gdeeble | Seems I figured out it needs to be Dial(Local/Extension@Context), but now internal phones can't dial that extension? |
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| 08:50.51 | xkln | hi guys |
| 08:51.16 | xkln | in our office we need to 'authenticate' before we can receive calls |
| 08:51.39 | xkln | this is done by calling *11, and putting in the extension/password on the dialpad |
| 08:51.43 | xkln | anyone know what this is called? |
| 08:51.54 | ChannelZ | retarded |
| 08:52.00 | xkln | thanks |
| 08:52.07 | xkln | anyone who isnt a dipshit know? |
| 08:52.34 | ChannelZ | Oh ok be like that |
| 08:52.39 | ChannelZ | good luck |
| 08:52.55 | xkln | well what good was your answer.. |
| 08:53.14 | ChannelZ | It's called humor. |
| 08:53.48 | xkln | calling shit retarded is called humor? |
| 08:54.28 | xkln | anyway i didnt come here to argue about bullshit |
| 08:54.31 | ChannelZ | I believe you're talking about hotdesking, but hey, I'm just a dipshit |
| 08:54.38 | xkln | indeed i am |
| 08:55.31 | xkln | is using the phone the only way to auth? can it be done with a script? |
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| 09:25.42 | Rahail | hi there wha tis teh exact command to limit concurent call under each sip user |
| 09:25.44 | Rahail | i am using Asterisk 1.8.21.0 |
| 09:34.38 | ChannelZ | I believe the new way is to use callcounter in sip.conf but then you have to enforce things yourself in the dialplan using some functions |
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| 10:11.33 | din3sh | hey all |
| 10:11.50 | din3sh | Asterisk hangs after a series of such messages: WARNING[9210] channel.c: Exceptionally long voice queue length queuing to Local/ |
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| 10:57.04 | WIMPy | hystryfe: No, it's optional in any case. |
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| 12:04.26 | din3sh | how to set a call duration limit on an inbound call? |
| 12:04.59 | WIMPy | TIMEOUT(absolute) or option L to Dial. |
| 12:15.54 | din3sh | ok |
| 12:15.56 | din3sh | :) |
| 12:16.39 | din3sh | want to set it in freepbx rather |
| 12:16.43 | din3sh | but got no answer there |
| 12:16.44 | din3sh | lol |
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| 13:05.25 | mrothe | so yesterday I asked if I could use asterisk to be reachable by sip on my domain. turns out I can't get it working. |
| 13:05.39 | mrothe | my very minimal config is this: https://gist.github.com/anonymous/964c7cddcfc7601ed19b |
| 13:06.38 | mrothe | logging in as markus does not work on all devices. linphone for example never reaches the server ('redirected' by SRV record). |
| 13:07.08 | mrothe | using telepathy I can log in, but dialing hello@unixforces.net results in this: |
| 13:09.29 | mrothe | https://pics.mrothe.de/asterisk1.txt |
| 13:10.08 | WIMPy | cannot fetch that URL. |
| 13:10.19 | mrothe | line 434 sais SIP/2.0 401 Unauthorized |
| 13:10.38 | WIMPy | Wrong username/password. |
| 13:10.39 | mrothe | can curl that domain |
| 13:10.54 | mrothe | s/domain/url |
| 13:11.22 | mrothe | WIMPy: try as http? |
| 13:11.27 | mrothe | (fetching) |
| 13:11.46 | mrothe | WIMPy: how can it be wrong username/password if I can successfully log in? |
| 13:12.03 | WIMPy | There is no "log in". |
| 13:12.21 | mrothe | sorry, I don't understand... |
| 13:12.52 | WIMPy | SIP is both connection- and stateless, so no login. |
| 13:14.36 | mrothe | WIMPy: okay, but the authorization does not fail? |
| 13:14.54 | WIMPy | It works a bit like http, you try a request, get rejected and retry including authentication. |
| 13:15.24 | mrothe | WIMPy: okay, so that 434 line is no indication for the problem? |
| 13:15.47 | WIMPy | Sorry, but that's too much SIP for me. |
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| 13:30.45 | mrothe | okay. seems to be NAT problem. using linphone and IPv6 it works. |
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| 14:45.31 | pickleheadjones | Hello, Can you have multiple hold music set in Asteriak. In other words. Running 2 separate businesses on one server and each business wants separate hold music. (Hosted PBX HERE) |
| 14:46.12 | WIMPy | yes |
| 14:46.32 | pickleheadjones | I have tried and do not know how |
| 14:46.37 | pickleheadjones | any help? |
| 14:46.51 | pickleheadjones | System seems to default to the hold music that is on the server |
| 14:47.14 | WIMPy | CHANNEL(musicclass) |
| 14:47.42 | pickleheadjones | Ok I will give this a look see. Thank You. |
| 14:47.49 | pickleheadjones | Now
Here is another tricky one |
| 14:48.54 | pickleheadjones | "Parking Lot" Parking lot starts at 71 - W/E. Each business doesn't need access to the same parking lot. I would like to segment the parking lots and have one business in 71-79 and the next 81-89 then ETC... |
| 14:49.48 | pickleheadjones | Cant seem to separate the parking lot. It runs on a matrix and will place callers on hold in the next available slot. |
| 14:50.02 | WIMPy | See PARKINGLOT and CHANNEL(parkinglot) |
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| 14:50.38 | WIMPy | Looks like they are like contexts. But I've never tried parking. |
| 14:50.41 | pickleheadjones | can you be a little more specific |
| 14:51.19 | WIMPy | See PARKINGLOT <- variable and CHANNEL(parkinglot) <- function |
| 14:51.40 | *** join/#asterisk navaismo (~navaismo@189.241.10.217) |
| 14:51.43 | WIMPy | Or see the parameters of the Park* applications. |
| 14:52.57 | pickleheadjones | Yeah we have played with this and have had no luck. Just seeing if anyone had a solution here |
| 14:53.54 | pickleheadjones | we decided to have each client on their own VM running asterisk to fix this prob. Only prob is
Running multiple VM takes a machine with a lot of Ram. So
We purchased a dell with 72 g |
| 14:56.16 | pickleheadjones | Thanks |
| 15:02.25 | [TK]D-Fender | [10:52]pickleheadjonesYeah we have played with this and have had no luck. Just seeing if anyone had a solution here <- it works. If you've had no luck then you are probably doing something incorrectly. |
| 15:02.45 | [TK]D-Fender | pickleheadjones: And if you would like help with that then you should show us what you're doing, and what's actually ahppening. |
| 15:02.47 | [TK]D-Fender | ~pb |
| 15:02.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:02.50 | [TK]D-Fender | ^^^ your friend |
| 15:45.22 | *** join/#asterisk Zitter (~danilo@host52-231-static.242-95-b.business.telecomitalia.it) |
| 15:48.50 | Zitter | hi, I would like to try Asterisk as VM (Virtualbox). Is there any suggested ISO to download? |
| 15:55.59 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
| 15:56.28 | [TK]D-Fender | Depends what you actually want to do |
| 16:03.25 | Zitter | I would like to try it, a "prepared" ISO will be the ideal solution |
| 16:04.25 | *** join/#asterisk imox (~imox@24.134.18.71) |
| 16:04.46 | [TK]D-Fender | If you plan on actually learning Asteriskand simply want to save 10 minutes on installing it yourself, then AsteriskNOW 3.0 w/o FreePBX. |
| 16:05.05 | [TK]D-Fender | Or if you simply want a ready-made toaster : FreePBX's latest ISO |
| 16:06.08 | Zitter | thanks a lot |
| 16:07.26 | Zitter | 4.2 or 3.2? |
| 16:08.19 | [TK]D-Fender | I'd go with the latest stable |
| 16:21.23 | DBordello | Zitter, I was completely new to Asterisk. I suggest you compile from source, and ditch the GUIs |
| 16:22.44 | [TK]D-Fender | again it depends on your objectives. |
| 16:23.00 | DBordello | Agreed |
| 16:26.24 | [TK]D-Fender | Actually want ot learn asterisk and want a fast install as you don't really know linux much at all : AsteriskNOW w/o FreePBX. Actually have a clue about linux and want to actually learn *? Compile it on whatever base OS you want. Want to add it to a server you already have? Almost certainly compile it yourself. Actually just want a "cool VoIP Toy" (tm) and don't care about learning... |
| 16:26.26 | [TK]D-Fender | ...anything and figure you'll live with a cookie-cutter world : Any distro with FreePBX. |
| 16:38.15 | DBordello | :) |
| 16:38.39 | DBordello | I setup Asterisk and a Polycom 501 for the first time last night. |
| 16:38.47 | DBordello | My initial impression is that it is a bit hacky :) |
| 16:38.52 | DBordello | But works. |
| 16:39.25 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 16:40.24 | DBordello | It seems like it requires a lot of scripts to make it all work, and different technologies |
| 16:40.54 | [TK]D-Fender | scripts? |
| 16:41.41 | DBordello | Well, for starters it requires a lot of configuration of phone specific files |
| 16:41.51 | DBordello | And to get the time, I need an NTP server |
| 16:42.05 | DBordello | To get a directory on the phone, I need a script to generate a phone-specific XML file. |
| 16:42.07 | DBordello | Etc. |
| 16:42.24 | DBordello | And I bet if I add another type of phone, I have to start over |
| 16:42.50 | [TK]D-Fender | Well there is no generic phone directory... so every phone has their own way |
| 16:43.03 | [TK]D-Fender | Polycom's can use LDAP for that with new firmwares |
| 16:43.09 | DBordello | That is nice. |
| 16:43.26 | [TK]D-Fender | And that directory can start from a file, but it's phone-based so you can just put it right into the phone |
| 16:43.26 | DBordello | I can certainly see the flexability |
| 16:43.30 | DBordello | Just my observation, not a complaint |
| 16:43.55 | [TK]D-Fender | As for needing an NTP server... you don't have to run your own, you could simply point it out to a public internet one |
| 16:44.07 | [TK]D-Fender | eg: pool.ntp.org |
| 16:44.34 | DBordello | That is what I ended up doing |
| 16:44.43 | DBordello | Still just playing, having fun doing it |
| 16:45.04 | [TK]D-Fender | if you already started by provisioning it then you're off to a good start. That's where you gt all the power options. |
| 16:45.12 | [TK]D-Fender | What firmware are you running o it currently? |
| 16:45.21 | DBordello | SIP 3.1.8 |
| 16:45.29 | DBordello | That is the lastest I believe that was released for the 501 |
| 16:45.45 | DBordello | Yes, I spent a few hours frustrated trying to provision it before I actually got it to do anything :) |
| 16:46.22 | [TK]D-Fender | Good pain :) |
| 16:46.26 | DBordello | Yes it was :) |
| 16:46.31 | DBordello | It wanted sip.ld, not something else :) |
| 16:46.40 | [TK]D-Fender | lots of fun to be had on it with the MicroBrowser,etc |
| 16:47.11 | DBordello | not sure I have done anything with that |
| 16:47.15 | DBordello | well, I know I haven't |
| 16:47.16 | [TK]D-Fender | it picks the ld according to the <mac>.cfg file for it |
| 16:47.22 | DBordello | and the applications button returns an error :) |
| 16:47.49 | DBordello | It was a good learning experience. Watching the tftp logs to see what it was asking for |
| 16:47.59 | [TK]D-Fender | So you could specify the model-specific one for it and not the grouped sip.ld |
| 16:48.28 | DBordello | That makes sense |
| 16:48.40 | [TK]D-Fender | Does load a little faster... for as long as that matters. |
| 16:48.53 | DBordello | It doesn't. |
| 16:48.54 | [TK]D-Fender | because one you've got them right they don't tend to crash and need rebooting. |
| 16:49.11 | DBordello | That is good |
| 16:49.12 | [TK]D-Fender | Some people actually nag on that point... I never understood it personally... |
| 16:49.37 | DBordello | Now that I have you here, I do have on thing that seems silly. |
| 16:49.41 | DBordello | I am using 00000000000.cfg |
| 16:49.53 | DBordello | which points to phone1.cfg and sip.cfg |
| 16:50.05 | DBordello | But these files require SIP credentials for Asterisk. |
| 16:50.15 | [TK]D-Fender | things I've down the with MicroBrowser : live call queue stats for 4 agents & 2 queues while "idle", night-mode status, web view of call routings, etc |
| 16:50.31 | DBordello | If I want a really generic deployment, without needing files for each phone, could I do some sort of MAC authentication? |
| 16:50.37 | [TK]D-Fender | mac a mac file for your specific phone and not using the 00000000 |
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| 16:51.13 | [TK]D-Fender | can copy the phone1.cfg to something more specific to the SIP device it registers as like phone100.cfg for readability |
| 16:51.25 | [TK]D-Fender | amd make sure your mac.cf points to those proper files. |
| 16:53.04 | [TK]D-Fender | [12:50]DBordelloIf I want a really generic deployment, without needing files for each phone, could I do some sort of MAC authentication? <- this would require some hackery.... |
| 16:54.54 | DBordello | [TK]D-Fender, damn. This is for a home, so each phone is essentially identical |
| 16:55.04 | DBordello | I was trying to make all the configuration on the Asterisk side |
| 16:55.38 | [TK]D-Fender | the phone still needs to have the right registration info... |
| 16:56.02 | [TK]D-Fender | for this you jsut need to make a mac.cfg for each phone, and point to a differnt phoneXXX.cf type file like I described |
| 16:56.13 | [TK]D-Fender | You could search/replace a template you made for this. |
| 16:56.24 | [TK]D-Fender | really a 1/2 minute job |
| 16:56.48 | [TK]D-Fender | that's what it takes for me from opening the box , assemling the phone and getting to load new configs and firmware. |
| 16:56.52 | [TK]D-Fender | 1 minute + boot time :) |
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| 20:17.38 | mrothe | are there arguments against allowing all codecs? i.e. allow=all in sip.conf? |
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| 21:02.13 | Gr3mlin | morning all, question, setting up asterisk as a Voip Gateway, is it easy? |
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| 21:36.21 | newtonr | mrothe: better to always know exactly what will be negotiated if you can. Certainly makes for easier troubleshooting when you see something weird going on. I'd try to argue it the other way. Why do you *need* to allow=all ? |
| 21:40.08 | newtonr | If you can't find a really good reason for allowing all, then I'd stick with defining what you want to allow. knowns are better than unknowns. |
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| 23:27.26 | igcewieling | DBordello: most of our phone specific config files only have a few lines |