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| 01:06.08 | Thedr | Hi all, I'm getting a lot of these messages in my debug DEBUG[3271] chan_sip.c: = No match Their Call ID: 3fa056aa-f2b57081@10.0.1.195 Their Tag a3202d8efdc937dd Our tag: as131785b5 any idea what they mean? |
| 01:08.23 | WiretapWork_ | Thedr, post the rest of the debug |
| 01:09.00 | Thedr | there are 100 or so lines of that |
| 01:09.02 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 01:09.38 | WiretapWork_ | ~pastebin |
| 01:09.38 | infobot | [~pastebin] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 01:10.47 | Thedr | http://pastebin.com/Gc6yntaR |
| 01:11.12 | WiretapWork_ | Thedr, output of sip show channels |
| 01:11.31 | WiretapWork_ | I suspect you're being attacked, but I need to see that output to know |
| 01:11.34 | sawgood | When using 'core show translation' (in a nutshell) what do all the numbers really mean? |
| 01:12.52 | WiretapWork_ | sawgood, Translation times between formats (in microseconds) for one second of data |
| 01:12.52 | WiretapWork_ | <PROTECTED> |
| 01:13.11 | Thedr | http://pastebin.com/gniZPDs4 |
| 01:13.11 | sawgood | WiretapWork_: thank you! |
| 01:13.25 | WiretapWork_ | sawgood, that is the top two lines of core show translation :P |
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| 01:13.49 | WiretapWork_ | Thedr, are there phones on those addresses that show (none) |
| 01:14.15 | sawgood | I have a '2' on the column and row intersecting ulaw to alaw |
| 01:14.18 | Thedr | yes |
| 01:14.42 | WiretapWork_ | Thedr, hmm, odd |
| 01:15.24 | sawgood | oh, ok so to go from 'ulaw' to 'alaw' takes 1 microsecond ... |
| 01:15.36 | Thedr | yeah, I am trying to figure out why a phone is getting caller ID on the caller end when for all purposes that feature shouldn't exist on the current system |
| 01:15.40 | WiretapWork_ | do note that none of those devices on the addresses that its barfing at are not registered |
| 01:34.42 | sawgood | WiretapWork_: why is it on a 'slow P4 machine' I have a 2 for ulaw to gsm and a 2400 on a dual core 2 system (for core show translation) |
| 01:35.16 | WiretapWork_ | no idea |
| 01:35.21 | WiretapWork_ | I have 4001 on my machine |
| 01:36.23 | sawgood | what kind of CPU? |
| 01:36.58 | sawgood | I see on the P4 it says 'milliseconds' ... on the dual CPU it says microseconds |
| 01:37.04 | sawgood | different versions of Asterisk ... |
| 01:37.19 | WiretapWork_ | model name : Intel(R) Pentium(R) 4 CPU 2.40GHz |
| 01:37.30 | WiretapWork_ | I'd say its a 533 |
| 01:38.12 | sawgood | Asterisk 1.4 uses milliseconds to measure ... Asterisk 1.8 uses microseconds |
| 01:38.26 | WiretapWork_ | ast 1.4 is EOS |
| 01:38.47 | sawgood | End of Service? |
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| 01:42.02 | WiretapWork_ | end of support |
| 01:42.49 | sawgood | when is that going to happen? |
| 01:43.40 | justdave | what permissions does the ShowDialplan command require in the manager API in 1.8? |
| 01:43.41 | sawgood | says on Digium site EOL for 1.4 is 04/21/12 |
| 01:43.55 | justdave | the doc on it on voip-info suggests config + reporting, and the user in question has both of those |
| 01:44.03 | justdave | but it's still getting "Permission denied" |
| 01:45.55 | justdave | "manager show command" on the cli doesn't seem to list required permissions either |
| 01:45.58 | WiretapWork_ | justdave, manager show command ShowDialplan |
| 01:46.04 | justdave | see above |
| 01:46.25 | WiretapWork_ | odd, its meant to O_o |
| 01:46.28 | WiretapWork_ | doesn't on my box either |
| 01:47.19 | justdave | oh, just "manager show commands" lists the privs in a column |
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| 01:47.46 | justdave | <PROTECTED> |
| 01:47.53 | justdave | whadaya know, the two it already has |
| 01:48.22 | justdave | looks truncated, actually |
| 01:49.01 | justdave | is "all" actually a specific privilege, or is that a shortcut for everything? |
| 01:49.23 | WiretapWork_ | its a shortcut |
| 01:49.49 | justdave | hmm, doesn't seem to matter, granting "all" still gets a permission denied |
| 01:50.18 | WiretapWork_ | you using ShowDialPlan or ShowDialplan |
| 01:50.22 | WiretapWork_ | the latter will not work |
| 01:50.32 | WiretapWork_ | and I assume you're logging in and using the correct parameters? |
| 01:53.15 | justdave | I'm doing "manager reload" between editing the config file |
| 01:53.28 | justdave | cli shows the user logged in |
| 01:53.45 | justdave | it's using the Asterisk::AMI perl module |
| 01:53.51 | justdave | I do have ShowDialPlan in there |
| 02:06.45 | justdave | ok, ran a packet trace on the socket to see what was going on... it's returning two separate error responses with the same ActionID... |
| 02:06.53 | justdave | the library I'm using only sees the first one of course |
| 02:06.59 | justdave | which is the permission denied. |
| 02:07.03 | justdave | the second one shows: |
| 02:07.03 | justdave | Message: Invalid/unknown command: ShowDialPlan. Use Action: ListCommands to show available commands. |
| 02:07.04 | WiretapWork_ | lol |
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| 02:13.14 | justdave | so I now have that user configured with "read=all" and "write=all" |
| 02:13.29 | justdave | restarted asterisk instead of just manager reload, just to make sure |
| 02:13.51 | justdave | log in with that user and running ListCommand lists a LOT fewer commands than what "manager show commands" from the CLI does |
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| 02:18.03 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 02:18.10 | WiretapWork_ | odd |
| 02:19.13 | justdave | meh, okay, no idea what I did, suddenly it's listing them all |
| 02:20.57 | WiretapWork_ | XD |
| 02:30.34 | justdave | oho.... |
| 02:30.36 | justdave | found it |
| 02:30.48 | justdave | it needs *write* access to "reporting" |
| 02:31.13 | justdave | read=config,reporting write=reporting <- that config set lets ShowDialPlan work |
| 02:31.14 | WiretapWork_ | lol |
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| 04:33.27 | rue_mohr | what distro does asterisknow use? |
| 04:35.37 | WiretapWork_ | centos |
| 04:36.56 | rue_mohr | hmm I'v heard of it |
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| 04:54.49 | ChannelZ | man fail2ban's docs suck |
| 04:58.34 | ectospasm | with a name like fail2ban, sounds like it was the love child of a 4Chan idiot. |
| 04:58.54 | ChannelZ | f4il2b4n |
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| 05:32.52 | rue_mohr | what did I miss in the make commands to get it to set up asterisk to start on boot for me? |
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| 05:45.19 | rue_mohr | the other isntall I did did it automatically, the only thing I know what different was that I had the dahdi stuff on there |
| 05:46.56 | rue_mohr | ok, I found whats supposed to be isntalled |
| 05:47.02 | rue_mohr | now what was supposed to isntall it |
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| 06:13.34 | atan | Okay I'm bloody stumped. I guess I'm new to linux so it's to be expected. I've installed Asterisk 1.8.4.1 but how do I get it to start when I reboot the box? I foolishly assume I am to create a file /etc/init.d/asterisk and add to rc.d, but I do not know what to put in there. Within the contrib folder they kindly include examples but it seems like it requires me to edit the file so it will |
| 06:13.34 | atan | work. I don't know what to put in any of the fields. Can anyone point me at a guide for this? |
| 06:14.57 | atan | Little things like DAEMON=__ASTERISK_SBIN_DIR__/asterisk |
| 06:15.32 | atan | I'm just going to guess /usr/sbin for now =\ |
| 06:15.41 | atan | Then what is ASTVARRUNDIR? :S |
| 06:17.17 | atan | Hmm. Okay, seem to have gotten it. |
| 06:17.22 | irroot | mmm |
| 06:17.26 | atan | Thanks anyway though :D |
| 06:17.30 | irroot | edit rc.locak |
| 06:17.34 | irroot | local |
| 06:17.40 | irroot | add safe_asterisk ;) |
| 06:17.54 | atan | What's safe_asterisk? :-) |
| 06:18.08 | irroot | its a wrapper / startup script for asterisk |
| 06:18.35 | irroot | it does not load dahdi / fxotune |
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| 06:45.06 | iulhk | hi all |
| 06:53.12 | iulhk | asterisk-1.4 installed, i hv users' user100 and user200 , my extensions are,, exten => _X.,1,Wait(50) ,, exten => _X.,2,Dial(SIP/${EXTEN}@${EXTEN}) ,,, exten => _X.,3,Hangup ,,, my question if user100 calling to user200 first he will wait 50 second , if user100 disconnect the call after 10 second or 20 second what asterisk dialstatus will be appear, is it cancel, is it busy, is it what |
| 06:53.12 | iulhk | ? bcoz at this point i need actual dialstatus , will anybody guide pls ? |
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| 07:10.33 | atan | Okay so since I swapped boxes my CDR no longer records calls. |
| 07:10.53 | atan | I had it inserting into a mysql database, but clearly I went wrong somewhere with this move |
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| 07:22.56 | WIMPy | iulhk: None. You only get a DIALSTATUS when you do a Dial(). |
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| 07:25.10 | iulhk | <WIMPy>: great, but if on wait caller hangup the call what status or even asterisk generate ? |
| 07:25.52 | WIMPy | None. It's not Asterisk that terminates the call. |
| 07:26.06 | WIMPy | At least if I understood your question. |
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| 07:43.49 | ectospasm | ${DIALSTATUS} will be populated when Asterisk detects a hangup, no matter which side initiates it... |
| 07:44.47 | irroot | hangupcause can be more usefull |
| 07:44.57 | irroot | esp with PRI |
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| 07:49.24 | ectospasm | irroot: true enough, I suppose SIP has similar cause codes. |
| 07:55.51 | WIMPy | But tehre's no Dial(). |
| 07:56.30 | WIMPy | And HANGUPCAUSE would be even more useful if tehre also was a HANGUPLOCATION :-) |
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| 07:58.46 | ectospasm | WIMPy: well, with PRI and SIP you can tell which side issues the DISCONNECT or CANCEL/BYE |
| 08:01.20 | WIMPy | There's more detail available. |
| 08:01.44 | WIMPy | And it can make quite a difference, where something happened. |
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| 08:07.33 | atan | I have box1 which has/had 10 people connected. I setup box2 and changed my dns over. TTL was set to 5 minutes for the A record. Anyway, users are still connected to the old box. Is there a way to disconnect them so they'll try to reconnect (and I hope do a dns lookup!) |
| 08:08.08 | atan | It has been well over 5 minutes :-) DNS has had the tll set to 5 minutes for a couple of days now |
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| 08:15.54 | ilj | hi guys, what context is used if an [entity] in sip.conf doesn't have it specified explicitly? [default]? |
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| 08:18.02 | ilj | I mean [general], not [default] perhaps |
| 08:18.31 | doolittlework | hi there can someone please explain :sip.conf--->canreinvite=yes please i dont grasp the concept |
| 08:20.05 | Insonic | hi there, short question...in theory, should it be possible to use an asterisk as "router" between an S2M connection (input) (with a 2-Port S2M Card) and 4 or 8 normal normal BRI-S0 Ports to connect an non-S2M-cappable Agfeo AS100 for example ? |
| 08:20.56 | cneb3000 | doolittlework: what is your understanding of it so far? |
| 08:22.31 | nix8n82 | Anyone know of a good ec2 image to use to build an asterisk server with? |
| 08:22.37 | nix8n82 | 64 bit |
| 08:22.47 | doolittlework | cneb3000: kick me if i am wrong but I think it tels to accept reregiters from a sip client should the connection to the phone drop? |
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| 08:26.20 | doolittlework | cneb3000: do not worry found a proper link for info |
| 08:26.30 | cneb3000 | doolittlework: sort of! it's real purpose is to alter media in the middle of a call |
| 08:26.32 | cneb3000 | ..oh ok |
| 08:26.35 | cneb3000 | walks away :) |
| 08:26.47 | doolittlework | sorry dude |
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| 08:27.29 | irroot | doolittlework when a call is made there is a invite latter on one of the parties may wish to change the call setup often the endpoints, audio codecs or mode ie [T.38] |
| 08:28.21 | irroot | so when the call is answered it can negotiate a direct path for audio get asterisk out the middle |
| 08:29.46 | irroot | when a fax arrives on a ATA [CNG detected] it may change from G723/9/GSM/lowbw to G711[AU] as the fax tones cant be used in narrow band latter a further reinvite may switch to T.38 when [CED tone] detected. |
| 08:29.51 | irroot | hope this helps |
| 08:32.25 | doolittlework | ok i grasp it now. so when asterisk monitors a call it holds on to those channels and stays in the middel, all other codec=>codec it passes it on if the link is there and the codec coresponds(i think this is where transcoding comes into play?) |
| 08:34.02 | doolittlework | sorry for all the dumb questions i am working through the sip.conf setting to better understand what every setting does |
| 08:34.17 | cneb3000 | ^^ better than most :D |
| 08:35.56 | doolittlework | another one if i may, cancallforward=yes?? i have 2 sip phones and i dont have this in my sip.conf but i can still forward the call to another extension, is this not used? |
| 08:44.21 | ChannelZ | that's a DAHDI directive, not SIP AFAIK |
| 09:06.16 | ectospasm | doolittlework: yeah, call forwarding in SIP happens at the endpoint. I call SIP/100 who is forwarded to SIP/202, and when I get a call from Asterisk at SIP/100, I respond with a 3xx message Moved Temporarily, which tells Asterisk to direct the call to SIP/202 |
| 09:06.57 | ectospasm | SIP/202 doesn't necessarily have to be on the same network, it could be an external number (e.g., PSTN) |
| 09:07.23 | ectospasm | I've done that before walking around the building, forwarding my desk Polycom to my mobile phone |
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| 09:08.10 | ectospasm | works nicely even if my desk phone is in a queue |
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| 09:51.45 | nix8n82 | Does anyone here use Amazon EC2 service to run asterisk on? |
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| 10:06.12 | irroot | installing asterisk [fdisk/format/install in < 15min] |
| 10:12.08 | irroot | ok clock stops at 12min |
| 10:12.17 | irroot | i love it |
| 10:12.20 | cneb3000 | hands irroot a gold medal |
| 10:12.45 | cneb3000 | irroot: i bet I could ruin an Asterisk installation quicker than you |
| 10:12.47 | cneb3000 | ok? |
| 10:12.47 | cneb3000 | go! |
| 10:12.51 | irroot | lol |
| 10:13.11 | irroot | that was 12min for os/dahdi/asterisk/..... |
| 10:13.24 | cneb3000 | that is pretty fast..! |
| 10:13.42 | irroot | also includes doom 1 / quake 1 and duke nukem :P |
| 10:16.13 | cneb3000 | like an audio based version? 8 moves forward, 2 back 4 left and 6 right. when you hear a monster you push 5 to shoot. |
| 10:16.29 | cneb3000 | can we perhaps integrate video phones into some sort of asterisk doom FPS? |
| 10:16.35 | cneb3000 | that'd be cool |
| 10:16.42 | cneb3000 | i'll add that to the asterisk wish list |
| 10:17.52 | irroot | lol |
| 10:24.08 | cneb3000 | how about gesture based telephony? so if I give a video phone the middle finger it hangs up? :) |
| 10:24.47 | irroot | hehe kinect much ?? |
| 10:32.31 | cneb3000 | there's a gap in the market there? ;) |
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| 10:46.16 | irroot | nice thing with a "quick install" is auto provisioning |
| 10:47.14 | irroot | send it with 4phones and 2port FXO or BRI on usb its tiny atom box smaller than netbook |
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| 10:54.13 | Azrael808 | Hello peeps, any good suggestions for a conference phone that works well with asterisk? |
| 10:55.28 | cneb3000 | Azrael808: You'll find that as long as it's a SIP phone it will most likely work. But I specifically use the polycom IP5000 and it's never steered me wrong :) |
| 10:55.54 | Azrael808 | cneb3000: thanks again! :) |
| 10:56.36 | cneb3000 | azrael808: no prob! |
| 11:13.26 | k3asd` | hi |
| 11:15.27 | cneb3000 | afternoon k3asd |
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| 11:59.39 | AlHafoudh | hi all |
| 12:00.06 | AlHafoudh | how to limit channels on sip trunk? I dont understand call-limit, limitonpeer(s), counteronpeer and etc. |
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| 12:04.01 | irroot | dont use those |
| 12:04.09 | irroot | rather set a channel group |
| 12:04.17 | irroot | then count the channel group b4 dial |
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| 12:09.31 | Tribbers | I have set up a service which listens to the AMI. When we run the service we see the message saying we have logged in and the message "Authentication accepted", then the sip peer entries appear (we have two at the moment so it appears correctly). |
| 12:09.36 | Tribbers | After that though whenever we make a phone call we do not see any events being fired in the service. Have I missed something simple as I am getting the messages form the ami just not any to do with the phone?? |
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| 12:27.03 | evilbit | Hi all, wondering if anyone is using a sangoma usbfxo with asterisk18 yum repo? |
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| 12:30.27 | irroot | not yum but i use it U-100 |
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| 12:30.56 | irroot | evilbit also use it with 2.6.38 [unsupported by sangoma still] |
| 12:31.10 | irroot | lo there malcolmd |
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| 12:31.37 | malcolmd | howdy howdy |
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| 12:41.59 | evilbit | well, the drivers seems to want /usr/src/dahdi but I can't find that in the yum repo |
| 12:42.18 | evilbit | the wanpipe drivers that is |
| 12:42.40 | evilbit | and what is 2.6.38? |
| 12:43.00 | evilbit | the latest version of wanpipe seems to be 3.5.20 |
| 12:43.38 | evilbit | ah, I see... you mean dahdi version? |
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| 12:48.25 | irroot | linux 2.6.38 + 3.5.20 |
| 12:48.32 | irroot | wanrouter |
| 12:48.40 | irroot | its not official supported |
| 12:49.43 | irroot | dahdi 2.4.1 |
| 12:50.10 | evilbit | gotcha |
| 13:04.16 | cneb3000 | hmm.. |
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| 13:04.32 | cneb3000 | is there a way of combining Background() with something like Read()?.. |
| 13:05.13 | cneb3000 | while I'd normally go Background() > Waitexten() i'm having to use background > read > gotoif as a result of using Thirdlane Asterisk |
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| 14:26.08 | AlHafoudh | irroot: thanks |
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| 14:26.53 | irroot | pleasure |
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| 14:32.24 | psilikon | is there an up to date guide for installing SpanDSP on Asterisk 1.4.x? I can't seem to find a working link for app_txfax and app_rxfax. |
| 14:32.59 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 14:33.17 | irroot | :P asterisk 1.4 is not upto date psilikon what up ?? |
| 14:33.46 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 14:35.06 | psilikon | irroot, Good point. I guess I'll have to move to 1.8. Maybe then I can just follow the steps in the Definitive Guide. |
| 14:35.48 | irroot | should be ok with spandsp-0.0.6 |
| 14:35.58 | irroot | install it |
| 14:37.00 | psilikon | irroot, What should be ok with 0.0.6? Asterisk 1.4? I did install it 0.0.6 for asterisk 1.4 but I can't seem to locate the links for the rx and tx apps nor can I find a link for the patch. |
| 14:37.09 | irroot | 1.4 |
| 14:37.50 | irroot | add SPANDSP=@PBX_SPANDSP@ to build_tools/menuselect-deps.in |
| 14:38.10 | irroot | AST_EXT_LIB_SETUP([SPANDSP], [spandsp Library], [spandsp]) to configure.ac |
| 14:38.33 | irroot | AST_EXT_LIB_CHECK([SPANDSP], [spandsp], [fax_init], [spandsp.h], [-ltiff]) |
| 14:38.42 | irroot | also to conf.....ac |
| 14:38.49 | irroot | run ./bootstrap.sh |
| 14:39.16 | psilikon | irroot, where did you find that information??? |
| 14:39.29 | irroot | from my 1.4 patch :P |
| 14:39.49 | psilikon | irroot, where did you get the patch from? |
| 14:40.01 | irroot | built over time |
| 14:40.36 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 14:40.37 | psilikon | irroot, Is the above documented anywhere? |
| 14:40.58 | irroot | not sure |
| 14:41.07 | irroot | can send you something |
| 14:42.27 | psilikon | ok |
| 14:43.10 | psilikon | irroot, I wonder if it would just be easier to move to 1.8. |
| 14:43.27 | irroot | id recomend it |
| 14:44.11 | psilikon | I think I'll just upgrade then. |
| 14:44.16 | psilikon | Thanks for the help |
| 14:44.27 | cneb3000 | psilikon: I had a similar problem. Moved to 1.8.. much easier. |
| 14:44.48 | psilikon | cneb3000, ahh good to know. Thank you. |
| 14:45.18 | psilikon | cneb3000, Were you then able to just follow the simple steps in the Definitive Guide? |
| 14:45.24 | irroot | if you want a patch buzz me |
| 14:45.42 | AlHafoudh | irroot: and how to limit h323 calls? |
| 14:45.52 | irroot | same way :P |
| 14:46.25 | irroot | do it as much as possible in the dialplan |
| 14:46.33 | cneb3000 | psilikon: yes it was MUCH easier :) |
| 14:47.02 | psilikon | cneb3000, nice |
| 14:51.48 | irroot | psilikon cneb3000 the 1.4 patch set i maintained is at about 9000 lines |
| 14:51.59 | irroot | all of it is in 1.6/1.8 |
| 14:52.58 | irroot | psilikon may also like the fact i have a working T.38 gateway for 1.8 that will be in 1.10 likely |
| 14:53.24 | cneb3000 | irroot: I'm waiting on my offices test suite to become free for that. |
| 14:57.02 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:00.12 | asilva | is there a way to check if the destination extension is set to forward the call ? |
| 15:02.34 | *** join/#asterisk senator (lebbeous@nox.esilibrary.com) |
| 15:03.50 | AlHafoudh | irroot: and if I dont want to do it with groups? |
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| 15:04.54 | irroot | AlHafoudh see if the h323 dirver supports limits |
| 15:05.54 | irroot | ok bang head on wall wash rinse repeat ... 12 DAHDI channels in a ringall group on a 24 port card i think this is not within manufactures recomended operataions .... |
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| 15:11.04 | asilva | irroot, hey |
| 15:11.21 | asilva | irroot, how do i not use res_timing_pthread and use res_timing_dahdi ? just not load the pthred module ? |
| 15:12.01 | malcolmd | res_timing_dahdi should be preferred to res_timing_pthread, so as long as dahdi exists on the system and dahdi is loaded, res_timing_dahdi should do its thing |
| 15:12.26 | asilva | understood, but should the counter increses in module show ? |
| 15:12.59 | *** join/#asterisk bbryant (~brett@74.222.117.158) |
| 15:14.32 | irroot | noload => res_timing_timerfd.so |
| 15:14.34 | irroot | noload => res_timing_pthread.so |
| 15:14.50 | irroot | in asterik/modules.conf |
| 15:14.55 | asilva | ok, just to check if i was doing it right |
| 15:16.05 | irroot | im really keen to see if it makes the problem go away you having its most odd |
| 15:17.53 | asilva | well i tried here just leaving timing_dahdi but the problem still there |
| 15:18.24 | irroot | please put updated locks/backtrace on the bug report |
| 15:19.12 | asilva | without modules ? |
| 15:19.15 | asilva | okay let me get it |
| 15:20.19 | malcolmd | the usecount should be 1 if the module is in use, and 0 if the module is not in use. |
| 15:20.56 | asilva | that's odd than, after removing both timing leaving only dahdi the dahdi count didn't increase at all |
| 15:21.02 | asilva | then* |
| 15:21.33 | irroot | [options] |
| 15:21.35 | irroot | internal_timing = yes |
| 15:21.39 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 15:21.43 | irroot | in asterisk/asterisk.conf |
| 15:21.50 | irroot | and dahdi_dummy loaded ?? |
| 15:22.29 | asilva | wasn't.... dman mistype on modprobe :( let me try again |
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| 15:23.58 | irroot | on way home l8r |
| 15:24.03 | asilva | l8r |
| 15:24.04 | asilva | tkz |
| 15:24.38 | asilva | malcolmd, now the counter incresed!! :D |
| 15:24.44 | asilva | but the problem is still there :/ |
| 15:27.29 | asilva | updated the post.. out to lunch now !! |
| 15:28.30 | *** join/#asterisk xxiao (~xxiao@li41-126.members.linode.com) |
| 15:32.14 | xxiao | freepbx, freeswitch, asterisk, tribox,...what are the differences |
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| 15:32.59 | Qwell | trixbox uses freepbx(*) uses Asterisk |
| 15:33.05 | Qwell | freeswitch is something completely different |
| 15:33.21 | carrar | payswitch! |
| 15:33.34 | xxiao | oh there are opensips, kamailio,and more |
| 15:33.45 | carrar | opensips ftw |
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| 15:34.11 | cneb3000 | xxiao: OpenSIPS/Kamailio are just registrars |
| 15:34.14 | carrar | sipswitch |
| 15:34.39 | xxiao | cneb3000: if you read their site, they're calling themselves do-it-all platform nowadays, for audio/video |
| 15:34.55 | cneb3000 | xxiao: They deal with media now? :| |
| 15:35.05 | cneb3000 | is way behind |
| 15:35.19 | carrar | RTP Proxy |
| 15:36.17 | xxiao | if i intend to do multi-party video calls, which one could potentially work? asterisk* family still is pretty much ippbx for voice |
| 15:36.51 | Qwell | xxiao: None o fthe above. |
| 15:36.53 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 15:37.00 | Qwell | Unless "multi" = 2 |
| 15:37.21 | Qwell | In which case, obviously you would use Asterisk, as it's the only one capable of being a PBX. |
| 15:38.18 | xxiao | trying to compare all these stuff, dizzy, too many options and naming, video is a must though |
| 15:38.27 | Qwell | Do you want a PBX? |
| 15:38.45 | xxiao | optional |
| 15:38.57 | Qwell | Figure out your requirements, then you can figure out what you need. |
| 15:39.02 | xxiao | more of a multi-way video call over internet |
| 15:39.18 | xxiao | support up to 4 parties video call |
| 15:39.22 | Qwell | if you just want video chat, use skype |
| 15:39.28 | xxiao | that's 1:1? |
| 15:39.31 | Qwell | or that new thing from AOL (which is actually pretty damn slick) |
| 15:39.46 | Qwell | http://www.aim.com/av/ |
| 15:39.49 | fullstop | I set up Kamailio once.. it was a strange experience. |
| 15:40.01 | cneb3000 | fullstop: how so? :) |
| 15:40.04 | xxiao | fullstop: define strange |
| 15:40.09 | Qwell | xxiao: None of those you mentioned can support more than 2 video streams in a call. |
| 15:40.20 | fullstop | cneb3000: Finding documentation was kind of difficult. |
| 15:40.35 | Cadey | Hi guys, we just had a number of phoens loose calls and in the messages i see it reloading a laod of modules but dont no why it would do this or if that would be the cause of the dropped line. Either way is there a trick I can use to monitor the server more closely that you can suggest so when a call dies I can debug it? http://pastebin.com/Ji4y65pv |
| 15:40.44 | fullstop | That is, the documentation was very terse and explained all of the options, but not how to actually do anything. |
| 15:40.54 | Qwell | Cadey: It likely crashed. |
| 15:40.55 | xxiao | fullstop: opensips seems to have quite a lot documents, and paid training |
| 15:40.56 | Qwell | ~backtrace |
| 15:40.56 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 15:41.10 | fish-bulb | Cadey: Asterisk is probably restarting for some reason |
| 15:41.16 | Cadey | <@Qwell> : how can i work out the reason? |
| 15:41.21 | cneb3000 | fullstop: i chose opensips instead of kam.. i revisited kam the other day and felt i made the wrong choice. I feel the Kam documentation it much easier to follow. |
| 15:41.30 | Qwell | Cadey: See above. |
| 15:41.38 | xxiao | does not quite know 'diff opensips kamailio' |
| 15:41.59 | fullstop | cneb3000: I was going against the grain a little bit.. I had postgres instead of mysql. |
| 15:42.01 | Cadey | so run asterisk with -g |
| 15:43.08 | cneb3000 | fullstop: renegade ;) |
| 15:43.36 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:43.50 | Cadey | oh also, It didnt drop all the calls which could be consistant with a crash ? |
| 15:44.13 | Cadey | it seemed to drop only about 5 out of about 15/20 active calls |
| 15:44.53 | fullstop | cneb3000: anyway, just getting it up and running with postgres felt wrong. The default config specifies a database of "openser" still. |
| 15:45.18 | cneb3000 | fullstop: haha. how long ago was that? |
| 15:45.25 | fullstop | cneb3000: about 2 weeks ago. |
| 15:45.34 | cneb3000 | ooo err |
| 15:45.53 | fullstop | And, perhaps this is just the way it is, but it felt like a giant black box. The debug logging was intended for a kamailio developer, and not auser. |
| 15:46.24 | cneb3000 | hmm I see. Perhaps I made the right choice with OpenSIPs then? |
| 15:46.53 | fullstop | When something didn't happen as expected, there was no log. I had to add logging to the config file to see anything. |
| 15:47.11 | *** join/#asterisk angryuser_laptop (~angryuser@80.214.4.3) |
| 15:47.14 | fullstop | I'm sure that once you get the hang of it, that it's easy and makes sense.. but it was a fairly steep hill. |
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| 15:47.58 | Qwell | Cadey: depends on your definition of "drop". If the media was out of band, the call could have continued. |
| 15:48.04 | cneb3000 | it was fairly steep with OpenSIPS to, I felt. I think the 'user friendly..ish' culture of asterisk certainly isn't present in those eh? |
| 15:48.06 | Qwell | err, not out of band. not through Asterisk |
| 15:49.02 | fullstop | cneb3000: exactly. Asterisk is fairly user-friendly when it comes to seeing what is going on inside of the toybox. |
| 15:51.36 | senator | hi everyone. i'm hoping for a little help with this situation: using asterisk 1.4. have a digium te121 connected via t1 crossover to an NEC PBX, which is supposed to be providing channels with dialtone with E&M wink start signalling (this is not a PRI). |
| 15:51.42 | cneb3000 | xxiao: have you seen this 'appconference' add on for asterisk? or anyone else reading this for that matter. |
| 15:51.43 | senator | i want to make outbound calls only with asterisk. i'm close |
| 15:52.00 | senator | i have a sip trunk set up for outbound as well, and that works, but i'm not all the way there with dahdi. |
| 15:52.13 | senator | here are the logs and conf i think you'd be interested in: http://www.pastebin.ca/2073740 |
| 15:53.05 | senator | the interface lights are all green and dahdi_tool is happy; i think there are no issues at that layer |
| 15:53.13 | senator | (well, 2 irq misses, but that's not a showstopper right?) |
| 15:55.38 | xxiao | cneb3000: no, relatively new to asterisk |
| 15:55.42 | fullstop | senator: Have you tried DAHDI/G1 instead of DAHDE/1 ? |
| 15:55.56 | senator | fullstop: will try now |
| 15:56.15 | xxiao | is downloading pbx in a flash iso |
| 15:56.24 | Qwell | why? |
| 15:56.31 | Qwell | xxiao: asterisknow.org, you'll thank me later. |
| 15:56.52 | cneb3000 | xxiao: I've never used it... but google says this 'AppConference' is an option for asterisk and 'multi user video conf' |
| 15:57.00 | senator | fullstop: test complete, no difference :-( |
| 15:57.06 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:57.17 | xxiao | cneb3000: check out that now |
| 15:57.26 | cneb3000 | xxiao: it doesn't actually display all the users in a grid fashion. it just switches video to the lead speaker. |
| 15:57.36 | Qwell | cneb3000: which is almost entirely useless |
| 15:57.41 | xxiao | it's said newer asterisk release broke some syntax which is unnecessary? why |
| 15:57.52 | cneb3000 | qwell: how is that useless? |
| 15:58.25 | cneb3000 | qwell: i take that back. it's not great. |
| 15:59.02 | Qwell | cneb3000: I said almost entirely. :) |
| 15:59.15 | xxiao | cneb3000: you mean appconference addon lacks? |
| 16:00.11 | *** join/#asterisk irroot (~irroot@41.53.225.228) |
| 16:00.24 | cneb3000 | xxiao: not really.. it's just.. what happens when more than one person is speaking? could get complicated. then I suppose that's not a good conference call. |
| 16:01.10 | xxiao | hmm...guess that's why video conference companies are doing so well these days |
| 16:01.32 | Qwell | xxiao: It's a very difficult problem to solve. |
| 16:01.33 | cneb3000 | xxiao: yea... i think like Qwell said.. none of the above realllllyy support multi user video conferences.. |
| 16:01.52 | cneb3000 | xxiao: so the best you'll get is hack jobs done by members of the community, which do an OK job. |
| 16:01.52 | Qwell | and expensive.. it requires a lot of transcoding, which isn't cheap |
| 16:02.50 | xxiao | encoding/decoding/transcoding... |
| 16:03.18 | cneb3000 | xxiao: if you find a solution - let us know ;) |
| 16:03.35 | xxiao | sure :) |
| 16:04.16 | xxiao | i recall the sip negotation: audio codec, video codec, everything in sync, then the talk can start |
| 16:04.56 | xxiao | then you need a mcu(multi-conference-unit?) to co-ordinate multiple streams in real time, |
| 16:05.40 | xxiao | 's head spins |
| 16:07.32 | Cadey | when asterisk starts as a service should I be looking in /usr/sbin/ for the core file if it crashes (under default install condtions)? |
| 16:08.44 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 16:09.50 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:09.53 | bmoraca_work | join #cisco |
| 16:10.00 | bmoraca_work | er |
| 16:10.10 | cneb3000 | traitor! |
| 16:10.33 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 16:12.35 | saisoma | hey guys, I have two asterisk boxes, both 1.8.4 with sip trunks (config: http://pastebin.com/EsFRFgf1). I'm getting severe local echo on 10.10.0.33 when calling phones on 10.10.0.54. Any ideas? |
| 16:18.38 | *** join/#asterisk vfabi (~fabi@host-static-109-185-192-110.moldtelecom.md) |
| 16:19.35 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:19.35 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:21.18 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 16:24.35 | psilikon | I have an asterisk 1.4 system running on Debian 6.0 using an Atom D510 (1.66, Dual core w/ HT) and when I noticed that when I ssh into that box and do a 'ls' in a large directory my music on hold starts getting choppy. Would it be worth trowing in one of those $60 Amfeltech mini pcie timers? |
| 16:27.06 | irroot | psilikon dont think so |
| 16:27.11 | irroot | run top |
| 16:27.15 | *** join/#asterisk tehrabbitt (~tehrabbit@pool-71-172-235-4.nwrknj.fios.verizon.net) |
| 16:27.31 | tehrabbitt | hey, can someone take a look at this and give me some insight on why it' won't call the 2nd number? |
| 16:27.31 | tehrabbitt | http://pastebin.com/zvyRLe6K |
| 16:27.35 | irroot | look at wait state its heavy on IO listing a big dir |
| 16:27.46 | psilikon | irroot, top shows almost no load. |
| 16:27.53 | irroot | use directory hashing |
| 16:28.03 | irroot | not the load the waitstate |
| 16:29.13 | irroot | tehrabbitt look at the g option in app_dial |
| 16:29.44 | tehrabbitt | irroot, where is app_dial? |
| 16:29.59 | irroot | "show application dial" |
| 16:30.13 | psilikon | irroot, they stay at 0.0% |
| 16:30.24 | irroot | while ls is run ?? |
| 16:30.26 | tehrabbitt | irroot, "No Such Command" |
| 16:31.00 | irroot | core show .... |
| 16:31.02 | psilikon | irroot, yeah, even with 'ls -R' run in the root dir |
| 16:31.17 | *** join/#asterisk angryuser_laptop (~angryuser@80.214.4.10) |
| 16:31.43 | irroot | what timing you running at the moment dummy ?? |
| 16:31.59 | psilikon | irroot, yep |
| 16:32.01 | tehrabbitt | irroot, now how do I set option g? |
| 16:32.32 | irroot | Dial(....,<timeout>,<opts>g) |
| 16:32.33 | *** join/#asterisk pdtpatrick___ (~pdtpatric@mainstwan.farheap.com) |
| 16:34.06 | psilikon | irroot, With dahdi-linux-complete-2.4.0+2.4.0 I think an lsmod will just return 'dahdi' and there will be no actual 'dahdi_dummy'. Is that correct? |
| 16:34.57 | irroot | no there is a option [not default] to build and install dahdi_dummy |
| 16:36.14 | psilikon | irroot, meetme rooms are working. Why would they work without a dummy timer? |
| 16:36.32 | irroot | module show like timing |
| 16:36.40 | irroot | will show what one you use |
| 16:37.01 | irroot | only one will be used |
| 16:37.46 | psilikon | irroot, nothing is returned |
| 16:38.03 | irroot | hehe may be my spelling ;) |
| 16:40.28 | psilikon | irroot, your spelling is correct. |
| 16:41.03 | Qwell | irroot: iirc, dahdi_dummy, as a separate module, is completely gone now |
| 16:41.08 | irroot | thats a first :) |
| 16:41.16 | Qwell | and if it's not gone, it's a noop |
| 16:41.51 | irroot | ah the hidden mysteries of the universe :) |
| 16:42.04 | irroot | know its been on the to kill list for a while thx qwell |
| 16:43.03 | psilikon | So how do I verify that I have a timing source? I compiled and installed dahdi-linux-complete-2.4.0+2.4.0 |
| 16:43.12 | *** join/#asterisk vfabi (~fabi@host-static-109-185-192-110.moldtelecom.md) |
| 16:43.28 | Qwell | psilikon: It'll just work |
| 16:44.33 | psilikon | Qwell, And it does... so I guess I am good. Would you agree that a hardware timer is not necessary in my case? |
| 16:45.52 | irroot | as long as you running a softwarwe timer [confirmed] if you not running SW should be fine |
| 16:45.55 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 16:50.58 | psilikon | I ordered a mini pci-e timer anyway. I'll experiment with it and see if I can improve the moh quality. |
| 16:51.28 | Qwell | what? |
| 16:51.37 | Qwell | No such thing exists... |
| 16:52.54 | irroot | if there is such be most usefull for T.38 gw |
| 16:54.23 | psilikon | Qwell, http://www.amfeltec.com/products/minipcie-timer.php |
| 16:54.41 | Qwell | umm |
| 16:55.05 | Qwell | Please tell me that's only like $20 |
| 16:55.25 | psilikon | $50 before shipping from Canada |
| 16:55.29 | Qwell | blinks |
| 16:55.38 | fullstop | Qwell: If you have to request a quote, it's more than $20 |
| 16:56.00 | psilikon | Qwell, So you think it is snake oil or what? |
| 16:56.14 | Qwell | psilikon: well, I could sell you a VoIP NIC if you'd like |
| 16:56.44 | psilikon | Qwell, Sangoma sells something similiar that is USB. |
| 16:56.50 | irroot | how much does the sangoma usb cost ?? |
| 16:56.54 | Qwell | also useless. :) |
| 16:57.33 | irroot | Qwell thx for the insight the reseller here keeps trying to unload em on me and cant see the point |
| 16:57.46 | psilikon | These hw timers came recommended by a consulting firm for use in Vicidial dialers. That's how I learned about them. |
| 16:58.05 | paulc | psilikon: got a link to a product page for them? |
| 16:58.50 | *** join/#asterisk pdtpatrick____ (~pdtpatric@mainstwan.farheap.com) |
| 16:59.10 | psilikon | The logic I got from them was that dahdi timing can slip when a system is under load so by using a hardware timer you can keep timing consistent and not screw up things that depend on timing like meetme. |
| 16:59.20 | psilikon | paulc, for which? |
| 16:59.35 | Qwell | meetme doesn't depend on dahdi for timing |
| 16:59.50 | Qwell | That would be fallacy #2 |
| 16:59.54 | psilikon | http://wiki.sangoma.com/sangoma-wanpipe-voicetime |
| 16:59.59 | irroot | it used to heavily ie 1.2 / 14 |
| 17:00.11 | irroot | that is why vici is a candidate |
| 17:00.12 | Qwell | irroot: No, it uses dahdi for *mixing*. |
| 17:00.52 | irroot | the ONLY time i have problems is under heavy IO load |
| 17:01.11 | Qwell | irroot: a timer wouldn't fix that. |
| 17:01.13 | fullstop | Well, sir, there's nothing on earth like a genuine, electrified, six-car monorail! |
| 17:01.27 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
| 17:01.41 | irroot | running PgSQL realtime / cdr / ..... if i do a vacum with or without HW timer its broke |
| 17:01.44 | Qwell | fullstop: what's it called? |
| 17:01.51 | Qwell | that's right, monorail! |
| 17:01.54 | irroot | tis what qwell said |
| 17:02.58 | fullstop | Such an awesome episode. |
| 17:07.08 | psilikon | Qwell, Can you suggest a way to fix the choppy music on hold? |
| 17:07.22 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:09.08 | irroot | psilikon qwell are you transcoding it heavily perhaps ?? |
| 17:09.26 | saisoma | hey guys, I have two asterisk boxes, both 1.8.4 with sip trunks (config: http://pastebin.com/EsFRFgf1). I'm getting severe local echo on 10.10.0.33 when calling phones on 10.10.0.54. Any ideas? |
| 17:09.35 | psilikon | irroot, g711u |
| 17:10.45 | irroot | saisoma not looked yet but rule re echo ... it is caused by a hybrid/balun on analogue line |
| 17:10.57 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 17:11.32 | irroot | can also be a hardware design flaw feedback betweeen mic/speaker |
| 17:12.40 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 17:13.54 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
| 17:18.23 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 17:26.44 | tehrabbitt | does anyone here have a good sample AVR example I can use to help set up my dialplan? |
| 17:28.19 | tehrabbitt | erm i mean IVR sorry |
| 17:28.38 | irroot | what you want to do with the IVR |
| 17:29.21 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 17:29.50 | irroot | what you ask is not easy |
| 17:30.53 | tehrabbitt | irroot, I've had it set up before, but I forget the syntax for it sadly (never saved an old copy of what I had...) Basically just answer the incoming call, and then just "press 1 to be connected to a technician, press 2 to go to voicemail" |
| 17:32.06 | *** join/#asterisk vfabi (~fabi@host-static-188-237-244-195.moldtelecom.md) |
| 17:32.19 | tehrabbitt | irroot, eventually, i'm thinking of getting a different DID for "internal use" where I can dial *into* the PBX and place calls through it after entering lets say a 6 digit pin (and can only be accessed from a specific caller ID record... aka my cell phone) |
| 17:32.46 | irroot | [ivr-0400-ah] |
| 17:32.48 | irroot | exten => 1,1,Goto(userout,520,1) |
| 17:32.50 | irroot | exten => 2,1,Goto(userout,510,1) |
| 17:32.52 | irroot | exten => 3,1,Goto(userout,0442,1) |
| 17:32.56 | irroot | exten => i,1,PlayBack(custom/0010) |
| 17:32.58 | irroot | exten => i,n,Hangup() |
| 17:33.02 | irroot | exten => t,1,PlayBack(custom/0010) |
| 17:33.04 | irroot | exten => t,n,Hangup() |
| 17:33.13 | tehrabbitt | Ah, thanks irroot |
| 17:34.03 | irroot | use Background to play audio and get the digit |
| 17:34.22 | tehrabbitt | irroot, how would I specify background? |
| 17:34.32 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:34.32 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:35.31 | tehrabbitt | irroot, with just Background()? |
| 17:35.44 | irroot | nope needs options |
| 17:35.53 | irroot | "core show application background" |
| 17:35.56 | tehrabbitt | alright |
| 17:38.14 | tehrabbitt | irroot, now if I use Background() to output a message to the user, it'd go before the first Goto() command above, right? |
| 17:39.28 | irroot | yeah i call the "context" |
| 17:40.40 | tehrabbitt | wait should i be using exten => s,1 etc or exten=> 1,1 etc |
| 17:43.47 | *** join/#asterisk ipstatic (~ipstatic@ip67-90-138-68.z138-90-67.customer.algx.net) |
| 17:44.04 | irroot | BackGround(filename1[&filename2[&...]][,options[,langoverride[,contexet]]]) |
| 17:44.06 | irroot | exten => XXXX,BackGround(audiofile,m,,ivr-0400-ah) |
| 17:44.13 | ipstatic | anyone having problems installing Asterisk 1.8 from the asterisk RPM repo on Centos 5.6? |
| 17:44.15 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:beae:c5ff:fe3e:b315) |
| 17:44.16 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:44.19 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 17:44.22 | irroot | oops _X....,n,.... |
| 17:44.35 | irroot | im outta here bed time |
| 17:44.44 | ipstatic | I am getting this error when trying to install: dahdi-linux-2.4.1.2-1_centos5.x86_64 from asterisk-current has depsolving problems |
| 17:44.44 | ipstatic | <PROTECTED> |
| 17:45.48 | tehrabbitt | later irroot |
| 17:45.49 | tehrabbitt | thanks again |
| 17:51.13 | tehrabbitt | can anyone here explain the difference between having 1,1 or s,1 or n,1? |
| 17:53.45 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:53.45 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:54.14 | fish-bulb | tehrabbitt: that syntax is incorrect |
| 17:54.31 | fish-bulb | tehrabbitt: check out https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities |
| 17:54.59 | fish-bulb | well, that syntax isn't incorrect I guess, but I'm sure it is not what you meant |
| 17:55.17 | fish-bulb | it goes extension,priority,application |
| 17:55.44 | fish-bulb | so in that you have 3 extensions; 1, n, and s |
| 17:56.10 | fish-bulb | all with only 1 priority |
| 17:57.12 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:58.31 | *** join/#asterisk pdtpatrick___ (~pdtpatric@mainstwan.farheap.com) |
| 17:59.00 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:59.19 | pdtpatrick___ | Question for you smart folks.. can someone please explain to me what does dev mean in this case? |
| 17:59.20 | pdtpatrick___ | http://pastebin.com/eBprzJWF |
| 17:59.51 | fish-bulb | anyways, dialplan execution will proceed through the priorities sequentially. Numbering them explicitly will work, but isn't very manageable after a few stack up. In that case you can use the "n" priority for next. You can also use "s" for same, which means the same priority number as the one above it. |
| 18:01.37 | *** join/#asterisk trapa (~trapa@74.114.209.89) |
| 18:01.51 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 18:02.14 | trapa | How would i find out more information about this error [2011-06-02 10:52:19] WARNING[3062]: chan_sip.c:13450 check_auth: username mismatch, have <102>, digest has <1-pstn> |
| 18:02.14 | trapa | [2011-06-02 10:52:19] NOTICE[3062]: chan_sip.c:21256 handle_request_invite: Failed to authenticate device Bob Smith <sip:102@192.168.2.10>;tag=b78f8ac2f13503cfo1 |
| 18:02.43 | fish-bulb | pdtpatrick___: I don't really read AEL, but that looks like it would be the second ARG passed to the macro application |
| 18:03.28 | fish-bulb | trapa: that is about all the information you will get. It means that username in the From header does not match what is in the Authentication Digest |
| 18:03.44 | trapa | i have a phone (102) which is and has been working perfectly .... the error i'm almost positive is coming from a SPA3000 deivce i'm trying to configure at ip 192.168.2.108 |
| 18:03.59 | trapa | fish: Is it possible because it's coming through a second port on a polycom phone |
| 18:04.00 | fish-bulb | usually means another SIP device is between the endpoint and Asterisk |
| 18:04.08 | *** join/#asterisk bbryant (~brett@74.222.117.158) |
| 18:04.21 | fish-bulb | eh, I guess so, not sure that I have seen that specifically |
| 18:04.34 | fish-bulb | I wouldn't think that Polycom would mess with any SIP headers |
| 18:05.15 | fish-bulb | you can set "insecure = invite" sip.conf for that device |
| 18:05.17 | pdtpatrick___ | how does one call a macro ? |
| 18:05.28 | trapa | let me try runnign antoehr cable just in case |
| 18:06.41 | fish-bulb | trapa: it may also be that the auth username doesn't match the username set on the phone. Do you have "1-pstn" set anywhere in the phone? |
| 18:07.07 | trapa | Not in the phone, only on the SPA3000 |
| 18:07.14 | fish-bulb | er, I meant the ATA |
| 18:07.54 | fish-bulb | ah ok, so I guess there is no occurrence of 102 in the ATA config then? |
| 18:08.17 | fish-bulb | pdtpatrick___: it is an dialplan application, so like anything else |
| 18:08.46 | Insonic | hi there, short question...in theory, should it be possible to use an asterisk as "router" between a S2M connection (input) (with a 2-Port S2M Card) and 4 or 8 normal normal BRI-S0 Ports to connect a non-S2M-cappable Agfeo AS100 for example ? |
| 18:11.27 | trapa | Yeah there's only a occurance of 102 in the polycom ... still working on punching a new jack .. the one i punched doesn't work. :P |
| 18:11.58 | *** join/#asterisk rhollan_ (~rhollan@208.146.43.5) |
| 18:13.37 | rhollan_ | any way to combine nat=yes on a sip extension with directmedia=yes? That is, if the media paths are symmetric, use the same outbound and inbound UDP port, and A* figures out the mapping (nat=yes) and ignores what's in SDP messages, it can issue proper reINVITEs? I find nat=yes makes directmedia=yes not have effect |
| 18:13.39 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-aokrvnreyxlyfofg) |
| 18:14.12 | *** join/#asterisk eerie_ (hoax@gateway/shell/bshellz.net/x-bklrbcqvdnurcgff) |
| 18:14.43 | fish-bulb | trapa: Ok, sounds like it is changing up the sip headers then |
| 18:15.08 | fish-bulb | rhollan_: I would have to look, but I think directmedia has a "nonat" option |
| 18:15.37 | rhollan_ | FWIW, my A* is ona public IP address, but SIP client extentions are NATed (either 3G cell smart phones or connected via 802.11g WLAN) |
| 18:17.06 | fish-bulb | so you specify your nat settings like normal and then when Asterisk sees something that is no nat'ed it will reinvite |
| 18:18.26 | rhollan_ | yeah, trouble is the SIP client, whether on WLAN or 3G is NATted. I was hoping that whatever A* uses to figure out the media UDP ports with nat=yes could be communicated in the reINVITEs it sends. |
| 18:19.21 | rhollan_ | I guess I could use STUN on the SIP clients, but that tends to eat battery on the smartphone |
| 18:21.01 | fish-bulb | ah, I guess I thought you wanted it disabled for the NAT'ed devices |
| 18:22.08 | rhollan_ | Good guess, but no. I was hoping once A* knew the ports with nat=yes (ignoring the SDP messages), it could communicate that in reINVITEs so the media didn't have to go through A* |
| 18:22.13 | rhollan_ | Good guess, but no. I was hoping once A* knew the ports with nat=yes (ignoring the SDP messages), it could communicate that in reINVITEs so the media didn't have to go through A* |
| 18:24.06 | tehrabbitt | thanks fish-bulb for the link before |
| 18:24.15 | fish-bulb | tehrabbitt: np |
| 18:25.17 | fish-bulb | rhollan_: Asterisk has not private address in your setup, correct? |
| 18:25.49 | fish-bulb | s/not/no |
| 18:25.57 | pdtpatrick___ | Is there a way to pass from one asterisk box to another .. context over DUNDi .. right now im using something like this -- Dial(66700@dundi-jn) .. but can i do Goto(onp_main@dundi-jn,s,1) ... is this possible ? |
| 18:26.15 | rhollan_ | No, A* is on the public internet... one of the advantages of a business class account with Comcast: they give you a /30: broadcast, net, NATTED IP, and "true static" IP |
| 18:26.52 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 18:27.11 | trapa | fish-bulb: Thanks for your help, i've moved the conversation over to #freepbx .. don't want to waste two people's time but i'll contact here again should i continue to have issues |
| 18:27.44 | *** part/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 18:27.50 | fish-bulb | rhollan_: hmm, ok. Are you using 1.8? |
| 18:27.56 | rhollan_ | On the LAN side of the Comcast router one sees NATted traffic AND traffic on the "true static" IP address. Basically, a /whatever private network and one address of two on the public network (the other one being on the WAN side and natted to the private one). |
| 18:28.29 | rhollan_ | 1.8? I think so. I got the latest Ubuntu packages, though I could build from source if necesasry. |
| 18:28.48 | fish-bulb | if you have access to the Asterisk CLI run "core show version" |
| 18:29.14 | pdtpatrick___ | can anyone please help with that? |
| 18:29.15 | rhollan_ | not easy from here. lemme see if I can get "in" |
| 18:30.48 | fish-bulb | pdtpatrick___: I don't understand your question |
| 18:30.58 | *** join/#asterisk `mx (~mikey@216.17.86.224) |
| 18:31.29 | rhollan_ | fish_bulb: Asterisk 1.6.2.9-2ubuntu2. So, I guess it's derived from 1.6 |
| 18:31.47 | rhollan_ | thanks himself for starting sshd last night |
| 18:32.12 | rhollan_ | er, that was for fish-bulb |
| 18:32.33 | pdtpatrick___ | @fish-bulb.. in extensions.conf i have something like this: exten => _189988877666,n,Dial(Local/66700@dundi-jn/n) |
| 18:32.54 | fish-bulb | k, I think there were some more options added to 1.8 regarding this, but it may just be I didn't notice them before. Either way, are you using the localnet externaddr/externhost/externip setting? |
| 18:33.07 | fish-bulb | pdtpatrick___: what are you trying to accomplish though? |
| 18:33.16 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 18:33.26 | pdtpatrick___ | i was to be able to use a context on another box |
| 18:33.28 | pdtpatrick___ | using dundi |
| 18:33.31 | rhollan_ | externip, though I think I don't need it because A* is not NATted on the only NIC it is on. |
| 18:33.50 | pdtpatrick___ | sorta like how i did above using Local/ext@dundi-jn |
| 18:34.00 | `mx | where is a good location for a someone that has never used asterisk go to learn how to configure a small pbx? |
| 18:34.23 | rhollan_ | @`mx: lots of tutorials. Google is your friend. |
| 18:34.26 | fish-bulb | pdtpatrick___: mm... I would say use a switch instead |
| 18:34.51 | fish-bulb | `mx: Digium has an online course that is very good for starting out |
| 18:35.11 | pdtpatrick___ | fish-bulb can u give me an example |
| 18:35.14 | rhollan_ | brb. Caffiene interrupt |
| 18:35.22 | `mx | fish-bulb: thanks |
| 18:36.04 | fish-bulb | `mx: np, here is the link http://www.digium.com/en/training/courses/#essentials . Also, if you just want some good documentation then check out thebook |
| 18:36.34 | rhollan_ | Ahh! |
| 18:36.37 | fish-bulb | Asterisk: The Definitive Guide |
| 18:37.11 | rhollan_ | notes it for his "to buy" list: got $100 for winning a chili cookoff recently |
| 18:37.30 | rhollan_ | in B&N gift cert. |
| 18:38.26 | fish-bulb | pdtpatrick___: it works like "switch => IAX2/user:password@bigserver/local", or "switch => DUNDi/e164" |
| 18:38.43 | fish-bulb | pdtpatrick___: the extensions.conf sample has more examples |
| 18:39.09 | fish-bulb | rhollan_: very nice! That is a great read, very useful |
| 18:40.37 | rhollan_ | Thanks, fish-bulb. Googling 1.6 vs. 1.8 differences now |
| 18:40.53 | pdtpatrick___ | fish-bulb so then local = context right? Thanks again |
| 18:41.55 | fish-bulb | rhollan_: np. Let me see if I understand this right, you are using nat = yes on the Asterisk side since the phones are behind NAT, but that implicitly disables directmedia? |
| 18:42.07 | fish-bulb | pdtpatrick___: righto |
| 18:42.55 | rhollan_ | fish-bulb: to be precise, I'm using nat=yes in the device definitions in sip.conf, but NOT in the [general] section for A* itself: A* isn't NATted, the phones are. |
| 18:43.12 | fish-bulb | gotcha, that's what I was meaning |
| 18:43.36 | fish-bulb | though I see why it would be confusing in how I said it |
| 18:44.11 | rhollan_ | Oh, and it doesn't disable directmedia: I get the usual "one way audio only" typical of NATted RTP. So, it looks like A* sends out unaltered reINVITEs without using the wisdom of knowing what the peer ports really are. |
| 18:45.31 | fish-bulb | oh hmm.. |
| 18:45.36 | rhollan_ | kicks himself for not keepint the wireshark traces from last night. |
| 18:46.15 | rhollan_ | Was I expecting too much from A* to share it's knowledge of endpoints when sending reINVITEs? |
| 18:46.56 | fish-bulb | I can't say for certain on that, I haven't messed with directmedia much at all in practice, just have a general understanding of how it should work and how Asterisk works |
| 18:47.17 | rhollan_ | O.K. then. thanks. I guess I will have to look at the traces closer. |
| 18:47.50 | fish-bulb | sure thing |
| 18:47.51 | rhollan_ | It was WAY COOL, when I got a reinvite working when going through a pbxes.org PBX |
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| 18:48.47 | rhollan_ | of course, at that point, I was proxying the cell phone through a broadvoice DID on a pbxes.org trunk |
| 18:49.06 | rhollan_ | So, I saw PC softphone to pbxes.org for the first few RTP packets, then reINVITed to the BV proxy. |
| 18:49.16 | fish-bulb | hmm.. I would be interested to see what the captures show |
| 18:49.42 | fish-bulb | you may be able to get around all of this using a SIP ALG |
| 18:49.57 | rhollan_ | in my present setup, I have a SIP client in the phone, and have it register with my own A* (so, pfffffft to pbxes.org, as useful as they have been). |
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| 18:51.01 | fish-bulb | because that way you don't need to worry if it lands on the same ports |
| 18:51.20 | fish-bulb | and the phone should renegotiate with no problem |
| 18:51.40 | rhollan_ | fish-bulb: odd that you say that: there IS a SIP ALG setting on the netgear WNDR3700 WLAN router, and I've seen what it changes (IP and port) but the recommendation is to DISABLE it for VoiP. In any case, that would not help calls coming in from the public net that were NATed by the tmobile IP gateway |
| 18:51.59 | rhollan_ | But, it suggests rolling my own router. |
| 18:52.39 | rhollan_ | Then again, I wonder if I could run a SIP proxy elsewhere in my LAN, or even on the A* box itself. |
| 18:52.54 | fish-bulb | SIP ALG's cause all kinds of problems when they are not configured to play nice, but they can be a big help at times if they are configured for it |
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| 18:53.44 | fish-bulb | you could definitely use Asterisk to route calls like that, but it won't act as a proxy. It's a back to back useragent |
| 18:53.47 | rhollan_ | Yeah, I caught the traffic with and without the SIP ALG enabled on the WLAN router, and saw that it "did the right thing". But, again, that would help for WLAN connections only. |
| 18:54.39 | rhollan_ | fish-bulb: no, I was thinking of having A* listen on, say 5061, the SIP proxy on 5060, and have it rewrite SDP payloads. |
| 18:54.51 | rhollan_ | The proxy then sending the rewritten packets to 5060 |
| 18:54.58 | rhollan_ | er, 5061, and thence to A*. |
| 18:55.22 | rhollan_ | What's frustrating, is that A* is ALREADY smart enough to ignore SDP ports and IPs with nat=yes. |
| 18:56.13 | bobb_WU | hey everyone! can i take a quick survey of how the most experienced Asterisk'ers read through their call logs? |
| 18:56.22 | bobb_WU | i put a post on the forum but never got a real response |
| 18:59.30 | fish-bulb | rhollan_: ah gotcha, I misread your previous comment |
| 19:03.24 | Qwell | bobb_WU: less |
| 19:03.34 | rhollan_ | All the docs for siproxd talk about phones behind NAT, and running sipproxd on the NATting gateway, but I see no reason why I can't run it between A* and the public net: move it to the PBX instead of to the phone. So, UDP <publicIP:5060>->sipproxd->A*(lo:5060) |
| 19:03.55 | rhollan_ | IOW, put A* BEHIND siproxd instead of the phones. |
| 19:04.24 | rhollan_ | (heck, I could NAT A* then, and I have good reasons to do that, but it isn't necessary). |
| 19:05.35 | bobb_WU | less isn't going to do it for me... i will soon have 10 active phone nodes and a relay server |
| 19:06.11 | bobb_WU | and my intended users are technical but not experienced at linux in the least. our log-reading solution must be web accessable |
| 19:06.30 | rhollan_ | Hmm, no sipproxd has to see the public net. Still, I think this will solve my problems. |
| 19:07.54 | rhollan_ | Of course, siproxd between A* and a NATting gateway, with siproxd made smart enough to use STUN would solve THAT problem, but strikes me as a horrible complication for no good reason. |
| 19:08.46 | rhollan_ | Thanks, fish-bulb, for the input. |
| 19:09.21 | fish-bulb | rhollan_AFK: np, sorry I couldn't help more |
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| 19:24.30 | ipstatic | Qwell: Should I add another repository other than asterisk-current? |
| 19:27.36 | *** join/#asterisk angryuser_laptop (~angryuser@80.214.4.6) |
| 19:28.29 | Qwell | ipstatic: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29 |
| 19:30.28 | ipstatic | Qwell: thank you so much. I cannot believe I missed that wiki page. |
| 19:30.42 | Qwell | how did you get the info then? |
| 19:32.17 | ipstatic | Qwell: https://wiki.asterisk.org/wiki/display/AST/Alternate+Install+Methods |
| 19:32.50 | ipstatic | it just links to the repo and not the wiki page you showed me |
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| 20:16.31 | atan | Okay so I'm not sure what config file I should edit here. When I dial out from a SIP phone I see it connecting to my provider SIP/voipmsPremiumChicago-00000001 is making progress passing it to SIP/1133-00000000 but if I hang up on the device it continues to ring the p hone |
| 20:16.33 | atan | phone* |
| 20:16.53 | atan | Is that an issue with the device perhaps, or something I need to set in Asterisk? |
| 20:17.33 | asilva | is there away to keep the call online if asterisk process crash?! |
| 20:20.07 | wdoekes2 | asilva: only if the media is sent directly (directmedia/canreinvite) |
| 20:20.36 | asilva | directmedia=yes and canreinvite=no ? |
| 20:20.51 | wdoekes2 | (and then there are things like session-timers and having to hang up the call from both ends) |
| 20:21.19 | wdoekes2 | canreinvite is superseded by directmedia option |
| 20:21.43 | asilva | ok let me try |
| 20:22.08 | wdoekes2 | rtp set debug on <-- if you see nothing, the rtp (audio) is sent directly |
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| 20:33.12 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:33.13 | [sr] | howdy |
| 20:33.23 | [sr] | how can i have 2 extensions ringing at the same time? |
| 20:33.40 | [sr] | (if thats possible) |
| 20:33.40 | Gugge | you cant |
| 20:33.49 | Gugge | you can dial 2 devices at the same time though |
| 20:34.01 | [sr] | hum |
| 20:34.04 | Gugge | dial(SIP/xxx&SIP/yyy) |
| 20:34.51 | [sr] | the thing is, i need to make a a ring bell sound when a extension rings |
| 20:35.18 | [sr] | so i was thinking on having one extra extension with a TA for the bell to have a FXS port |
| 20:35.38 | jkroon | hi guys, I need to figure out the whole host vs username based authentication thing with SIP. |
| 20:35.51 | jkroon | at the moment it's a bit of a black art for me. |
| 20:36.36 | jkroon | situation is that I've got a SIP provider and I need to register to -multiple SIP accounts on that server, so far I'm unable to get the return INVITEs working properly on all of the accounts at the same time. |
| 20:38.03 | [sr] | hi WIMPy |
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| 20:47.55 | jkroon | ok, it seems the concept of a SIP type=user/friend isn't being used at all any more, so now let's say I have [user1] host=a.b.c.d authuser=user1 and [user2] host=a.b.c.d with authuser=user2 - and incoming call now comes in which I'd like to match on host ip/port - to which of the two peers will that call go? |
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| 20:49.17 | Gugge | jkroon: depends if you use realtime or not, and if you cache the realtime users or not .. but generally, the first one :) |
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| 20:49.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 20:50.08 | GreatSUN | re |
| 20:50.58 | jkroon | not using realtime at all. |
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| 21:07.49 | WIMPy | Hi [sr]! |
| 21:07.53 | [sr] | :) |
| 21:07.58 | [sr] | how have u been? |
| 21:08.22 | WIMPy | Oh, ok, just hat to concentrate on less interesting things. |
| 21:08.51 | WIMPy | Insonic: That's possible in practice as well. |
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| 21:10.43 | jkroon | Gugge, do you know whether IAX/2 behaves similarly or whether that is still totally separate for peers and users? |
| 21:11.41 | Freeaqingme | I think that with 1.8 there's no more difference between users and peers (internally)? |
| 21:14.12 | [sr] | WIMPy: i see! |
| 21:16.02 | ChannelZ | it's all how they match the peer, by IP or by user |
| 21:16.25 | ChannelZ | I thought that was still true of iax but perhaps not |
| 21:19.15 | jkroon | why can't they just behave the same though? |
| 21:19.39 | jkroon | i prefer the distinct user and peer way but that might be because I misunderstand something else. |
| 21:19.47 | Insonic | WIMPy: yeah, just playing around with some ideas...buy a new AS200 for s2m or buying a "gateway" for s2m to bri |
| 21:19.56 | jkroon | now I need insecure=invite on my SIP peers to get inbound working ... |
| 21:21.01 | WIMPy | Insonic: I would suggest to think about hardware that can be used with misdn as well. Gives you more implementation choices. |
| 21:22.53 | *** join/#asterisk caveat- (hoax@newshell1.bshellz.net) |
| 21:24.58 | Insonic | WIMPy: which 1 or 2port S2M card would you suggest ? |
| 21:25.50 | WIMPy | Insonic: Generally I'd try a HFC-E1 based one, ie Junghanns, beronet,... |
| 21:26.16 | WIMPy | But if you want to use VOIP as well, you might want something with HWEC. |
| 21:26.31 | *** part/#asterisk senator (lebbeous@nox.esilibrary.com) |
| 21:29.24 | Insonic | Voip ist not a target in the first place...primary target is to decide if we build a gateway for the old telephone hardeware (agfeo as100, not s2m cappable) or buy a new one , which could handle it |
| 21:30.02 | WIMPy | Insonic: How many phones do you have on it? |
| 21:30.11 | Insonic | ~50 |
| 21:30.33 | WIMPy | Ok, makes sense to keep them, I guess. |
| 21:31.39 | WIMPy | And if you only use a PC as interface converter, it can't do much wrong. |
| 21:32.02 | Insonic | maybe just 40...not sure about the exact numbers....it was not really administrated so far...a "grown" system |
| 21:32.43 | WIMPy | I gouess you should think about future growth before deciding anything. |
| 21:34.28 | Insonic | yeah..that was the plan....trying to find a some ways....my favorite would be an asterisk as interface AND for further expansions (voip-phones, or softphones ..) |
| 21:39.53 | jkroon | ok, so is it possible to (using iax/2) perform host/ip based authentication? |
| 21:40.20 | jkroon | ie, is it possible to create a user that will match purely on the incoming IP (and port) instead of based on the username? |
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| 21:53.35 | jkroon | jip, but you need to get the call initiator to not send a USERNAME informational element. |
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| 21:54.12 | jkroon | OK, I note that when using IAX/2 the caller can specify a custom CONTEXT value - is it possible to suppress this - ie, to NOT allow the caller to specify it's own context? |
| 21:54.53 | jkroon | nm, not a problem, the caller can only use a context for which there exists a context= line in the appropriate section being utilized. |
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| 22:12.02 | justdave | I remember when I was running Asterisk 1.2 ages ago that a "extensions reload" would hang asterisk every so often so it was best to wait till after hours to do it (and restart rather than reload) |
| 22:12.43 | justdave | ran 1.4 for a few years with no problems and did reloads all the time without issue |
| 22:12.57 | justdave | just had 1.8 hang doing a reload :| |
| 22:14.05 | justdave | (reloads are done constantly every time desktop support assigns someone a new extension number or edits a conference room config) |
| 22:14.13 | justdave | been running 1.8 for a couple weeks, first time it's happened so far. |
| 22:15.05 | jkroon | has * crash on him about 5 to 10 times daily at the moment. |
| 22:15.51 | jkroon | 1.6.2.17.3 ... looks like something in chan_local.so ... |
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| 23:10.23 | pdtpatrick__ | Question for you smart folks. . how can I allow users to search by extension or name? Is this possible in Asterisk ? |
| 23:11.25 | pdtpatrick__ | actually already have it to allow dialing extensions. How about searching for names? |
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| 23:20.17 | ectospasm | pdtpatrick__: you mean like the directory? |
| 23:20.30 | pdtpatrick__ | right |
| 23:20.34 | WiretapWork | yep |
| 23:20.37 | WiretapWork | directory function |
| 23:20.37 | ectospasm | pdtpatrick__: the directory is tied to voicemail |
| 23:20.54 | WiretapWork | its really funny being in someone's IVR and going '#' to skip a prompt |
| 23:21.00 | ectospasm | I can't remember if it's a function or an application, though I think the latter |
| 23:21.01 | WiretapWork | and instead getting the directory XD |
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