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| 00:06.12 | Docfxit | [TK]D-Fender: I'm guessing you mean the phone configs. |
| 00:07.36 | [TK]D-Fender | BOTH sides |
| 00:07.44 | [TK]D-Fender | Asterisk and your phones have to AGREE |
| 00:09.18 | Docfxit | [TK]D-Fender: This has been running for years. I haven't changed any configs. Something must have been distroyed when the hard drive filled up. |
| 00:10.03 | [TK]D-Fender | Well you have plenty fo config errors, and it doesn't stop working for nothing. |
| 00:10.10 | [TK]D-Fender | We are getting no details |
| 00:10.43 | [TK]D-Fender | Go validate your peers yourself |
| 00:12.59 | *** join/#asterisk [NC] (~nc@173.245.31.71) |
| 00:14.30 | Docfxit | [TK]D-Fender: I'm sorry I'm trying to give you as many details as I can. By validate your peers do you mean to look at the setup on each phone? |
| 00:14.45 | [TK]D-Fender | I mean check BOTH end thoroughly |
| 00:17.28 | Docfxit | [TK]D-Fender: I will go check everything. I will leave this chat now. It will take me a couple hours to do that. Thank you very much for all your help. |
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| 00:28.51 | Penguin | w. t. f. |
| 00:28.59 | Penguin | cpe-e650*CLI> core show uptime |
| 00:28.59 | Penguin | System uptime: 12 years, 7 weeks, 4 days, 7 minutes, 20 seconds |
| 00:28.59 | Penguin | Last reload: 12 years, 7 weeks, 4 days, 7 minutes, 20 seconds |
| 00:30.15 | WIMPy | What version? |
| 00:30.25 | Penguin | Asterisk 1.8.23.1 built by rob @ cpe-e650 on an i686 running Linux on 2013-10-09 17:45:26 UTC |
| 00:30.52 | [TK]D-Fender | GREAT SCOTT! |
| 00:30.53 | WIMPy | Ok, so that is obviousely not correct. |
| 00:31.49 | Penguin | It's wrong, no doubt about it. |
| 00:31.58 | Penguin | I have never seen that happen before. |
| 00:32.36 | WIMPy | Looks like it calculates uptime from the date/time. |
| 00:32.55 | Penguin | # date;hwclock |
| 00:32.55 | Penguin | Wed Feb 19 18:32:49 CST 2014 |
| 00:32.55 | Penguin | Wed 19 Feb 2014 06:32:50 PM CST -0.806175 seconds |
| 00:33.19 | WIMPy | And what was the date when Asterisk started? |
| 00:33.29 | Penguin | I wouldn't know how to find that out. |
| 00:33.36 | WIMPy | It's not a system without RTC, is it? |
| 00:33.59 | WIMPy | You could have luck and find something in the logs, if they still exist. |
| 00:38.39 | carrar | time travel |
| 00:38.47 | carrar | thats the *ONLY* answer |
| 00:40.53 | *** join/#asterisk darkdrgn2k3 (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 00:40.55 | darkdrgn2k3 | hi all, |
| 00:41.03 | darkdrgn2k3 | what happens if you run our of g729 licenses? |
| 00:41.07 | darkdrgn2k3 | does it just use g711 instead? |
| 00:51.24 | [TK]D-Fender | no, calls will flat-out fail |
| 00:51.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.136) |
| 00:53.45 | darkdrgn2k3 | even if g711 is set as backup? |
| 00:54.11 | [TK]D-Fender | It will agree on 729. then it will try to use it. Then it will fail. Then the call dies, horribly |
| 00:54.48 | [TK]D-Fender | any device you control that won't change on you will always negotiate the same way |
| 00:54.50 | darkdrgn2k3 | dam.. ok on a asterisk call is it 2 concurrent calls, one from phone to asterisk the other from asterisk to telco ? |
| 00:55.35 | WIMPy | yes |
| 00:55.54 | darkdrgn2k3 | and once a license is active it can never be moved to a differnt hardware right? |
| 00:56.51 | [TK]D-Fender | it can, a limited number of times |
| 00:57.11 | darkdrgn2k3 | thats good to know.. |
| 00:57.17 | [TK]D-Fender | somewhere between 3-5 IIRC |
| 00:57.23 | darkdrgn2k3 | any idea what hardware it indexs? |
| 00:57.23 | [TK]D-Fender | check with digium |
| 00:57.53 | [TK]D-Fender | NIC MAC amonsgts others IIRC |
| 00:57.58 | Penguin | It uses an ethernet MAC address, but it might use other keys too. |
| 00:58.09 | darkdrgn2k3 | i figured mac (standard) |
| 00:58.27 | Penguin | Maybe MAC and UUID/GUID? |
| 00:59.03 | darkdrgn2k3 | hey on a sip progress are the codecs listed in priorit? |
| 00:59.06 | darkdrgn2k3 | a=rtpmap:0 PCMU/8000 |
| 00:59.07 | darkdrgn2k3 | a=rtpmap:18 G729/8000\ |
| 00:59.15 | darkdrgn2k3 | does that mean g729 will be second |
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| 01:18.31 | darkdrgn2k3 | how do you set priority of the codecc? |
| 01:19.46 | WIMPy | The order in which you list them in your configuration. |
| 01:21.06 | darkdrgn2k3 | hmm |
| 01:21.13 | darkdrgn2k3 | cause i listed g729 and ulaw |
| 01:21.18 | darkdrgn2k3 | but the trunk always connects as ulaw |
| 01:21.24 | darkdrgn2k3 | i took out ulaw.. and g729 connects |
| 01:21.35 | darkdrgn2k3 | i have allow=g729&ulaw |
| 01:24.45 | darkdrgn2k3 | any ideas? |
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| 01:48.41 | paulc | darkdrgn2k3: Is it ok to have & in there? I thought you needed separate lines.. allow=ulaw, allow=g729, and the order of the lines defined preference |
| 01:49.09 | darkdrgn2k3 | freepbx automaticaly does that for seperate lines so i figured its good |
| 01:49.35 | paulc | Ah, ok.. |
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| 02:03.59 | darkdrgn2k3 | dam intenret |
| 02:04.00 | darkdrgn2k3 | lol |
| 02:04.06 | darkdrgn2k3 | yeh appernty & is cosher |
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| 02:35.16 | Kattyroo | fiddle sticks! |
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| 02:44.39 | maruen | Hi, does anybody knows why I'm getting the message " Getting IMSI number failed" when I try to start a device with chan dongle and asterisk? |
| 02:44.52 | maruen | and if there's a anyway to solve |
| 02:48.04 | Kattyroo | have you tried beer? |
| 02:49.28 | maruen | no, with beer I will only forget my problems friend |
| 02:49.32 | maruen | and not solve them |
| 02:52.38 | darkdrgn2k3 | when i use openVPN for asteirsk, do i just QOS openvpn traffic? |
| 02:54.08 | darkdrgn2k3 | i mean router---- VPN Router---- Phone on the router i QOS openvpn traffic right? |
| 03:02.10 | Penguin | kattyroo: I tried some beer recently. It was Miller Fortune. It's not bad... I have had worse. |
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| 03:05.16 | Kattyroo | Penguin: never heard of it |
| 03:05.41 | Penguin | It's new. I heard they weren't even supposed to launch it until March, but it's in stores now. |
| 03:05.55 | Penguin | It's undistilled and 6.9% ABV. |
| 03:06.12 | Penguin | It's a dark gold lager. |
| 03:10.59 | Kattyroo | interesting. |
| 03:11.05 | Kattyroo | i'll see if i can find it. |
| 03:11.19 | Kattyroo | i found a new beer this past weekend as well! Well's banana bread beer. pretty tasty :> |
| 03:11.28 | Kattyroo | anywho, bedtime. cheerio (= |
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| 04:37.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op |
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| 05:37.14 | zloifei | morning all! Need advice. When I open Elastix->PBX->Extensions->%some_extension% I can see in callgroup and pickupgroup fields the variable set as 1, but when I execute: sip show peer %some_extension% over PBX->console, in call and pickup groups.. the variables set as 2 or 3 ..or empty. Tell me pls, how to explain this fact? |
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| 06:12.37 | tulga | Hi. I need to connect to my box from external IP. but asterisk return error like "NOTICE[1941]: chan_sip.c:28003 handle_request_register: Registration from '<sip:101@192.168.4.110>' failed for 'public_ip:64116' - Wrong password". how to solve it? |
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| 06:34.24 | kaldemar | tulga: use a correct password... |
| 06:35.51 | tulga | kaldemar: password is correct. it works from local network, not working from external |
| 06:36.51 | kaldemar | then you probably have a matching issue. |
| 06:37.39 | kaldemar | what peer are you trying to register as? 101? |
| 06:38.49 | tulga | kaldemar: yes 101 |
| 06:39.59 | kaldemar | what does sip debug and verbosity say when you try to register? use pastebin to show. |
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| 06:40.22 | kaldemar | also, what version of asterisk are you using? |
| 06:42.44 | tulga | asterisk 11.7 |
| 06:43.49 | tulga | http://pastebin.com/aZCFNHxd |
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| 06:48.39 | kaldemar | and what does your sip.conf look like? mask secrets. |
| 06:49.13 | chiggins | Got a newbie question here. Trying to call from one extension to another, but I keep getting rejection messages? Here's what I'm working with currently: http://pastie.org/8751051 |
| 06:50.08 | tulga | kaldemar: I'm using freepbx. |
| 06:50.24 | kaldemar | chiggins: did you reload your dialplan? |
| 06:50.26 | tulga | kaldemar: "sip show settings" is ok? |
| 06:50.44 | chiggins | kaldemar: Yup |
| 06:50.58 | kaldemar | tulga: no, it only shows global settings. |
| 06:51.06 | chiggins | I've run both 'dialplan reload' / 'sip reload', along with just 'reload' |
| 06:51.52 | kaldemar | chiggins: dialplan show local-dial |
| 06:52.47 | chiggins | "There is no existence of 'local-dial' context" o_O |
| 06:53.08 | kaldemar | chiggins: what file did you add it to? |
| 06:53.36 | tulga | kaldemar: http://pastebin.com/wU9axQNS |
| 06:54.12 | zloifei | <PROTECTED> |
| 06:57.30 | kaldemar | tulga: i asked for the whole sip.conf in case there is something else that matches the request. what may happen is that the locally made call matches the peer you expect, but when you call from outside your network it matches something else. you should be seeing a "Found peer..." message in the verbosity/sip debug when you make the call. |
| 06:58.41 | tulga | kaldemar: freepbx changed sip.conf, then splitted some files like sip_general.conf etc |
| 06:59.10 | kaldemar | zloifei: this is not a support channel for elastix. |
| 07:00.14 | kaldemar | tulga: yes, that's why i'm not that eager to continue debugging this since i'm not that familiar with freepbx. there's also #freepbx if it happens to be an issue caused by freepbx-generated configs. |
| 07:01.11 | tulga | kaldemar: I set public IP in proxy field, and changed domain to internal IP. so it connected :) |
| 07:02.08 | kaldemar | tulga: ok, good to hear. |
| 07:02.21 | chiggins | kaldemar: I'm adding it all to the right files, I believe... |
| 07:02.36 | chiggins | Here's a little screen recording of what I'm seeing from my files https://asciinema.org/a/7756 |
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| 07:05.01 | D30 | hi all, what could be possiblly the RED flag means? > http://codepad.org/OE4FLAEn |
| 07:05.04 | tulga | kaldemar: thanks for your help |
| 07:05.34 | kaldemar | chiggins: that ascii cast is just making things harder, use a pastebin next time. |
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| 07:06.15 | kaldemar | D30: that the ports are not connected. |
| 07:06.52 | D30 | ahhh okay... now i get it.. i have only 1 pstn line connected on line 1 |
| 07:07.01 | D30 | its clear to me now :D thanks kaldemar |
| 07:08.35 | chiggins | kaldemar: My bad. It's been useful before heh |
| 07:08.36 | chiggins | http://paste.debian.net/83042/ |
| 07:08.38 | chiggins | http://paste.debian.net/83043/ |
| 07:08.46 | kaldemar | chiggins: are you on the right machine editing extensions.conf? |
| 07:09.12 | MaliuatLap | kaldemar: that's funny for so many reasons |
| 07:09.22 | chiggins | kaldemar: aye, it's my asterisk box |
| 07:09.26 | chiggins | 100% positive |
| 07:10.07 | MaliuatLap | kaldemar: my vim swap file once cost a company an estimate 4 to 5 million over a weekend :) |
| 07:10.12 | kaldemar | MaliuatLap: it's funny because it's true! really, wouldn't be the first time. :) |
| 07:10.31 | kaldemar | MaliuatLap: ouch! |
| 07:10.43 | MaliuatLap | kaldemar: yeah, but I'm normally looking at logs on the wrong machine |
| 07:11.19 | MaliuatLap | kaldemar: not even my boss new that was flowing over that machine - so even though I found the cause and owned up there were no reprecussions (for me) |
| 07:12.40 | MaliuatLap | I frequently have an issue where I watching a live tail for of a log and not seeing the activity ... because the ssh session is "thinking about it" |
| 07:26.52 | chiggins | So any reason that running dialplan reload doesn't read from /etc/asterisk/extensions.conf ? |
| 07:30.45 | MaliuatLap | can't remember what dialplan reload does. |
| 07:31.02 | MaliuatLap | I use AEL for my dialplan, so I just use ael reload |
| 07:32.02 | MaliuatLap | chiggins: just remembered that there is a conf thing that tells * where to get files like that from ... it might be set to look somewhere else |
| 07:32.43 | chiggins | MaliuatLap: Oh is that righ? |
| 07:32.47 | chiggins | Interesting |
| 07:37.48 | chiggins | Ah ha! |
| 07:38.03 | chiggins | I guess I just had to change the stuff in extensions.ael for it to all work |
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| 14:51.19 | Katty | general hannnnnnn |
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| 14:57.58 | Astermon | Having some issues registering my sip with asterisk |
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| 15:10.15 | *** join/#asterisk g011um (~g011um@LLagny-156-35-17-119.w80-14.abo.wanadoo.fr) |
| 15:10.23 | g011um | hi everybody |
| 15:10.58 | g011um | I have an issue on elastix... but chan #elastix seems snooring |
| 15:11.32 | g011um | is there anybody alive here ? |
| 15:12.53 | MaliuatLap | we're all zombies in here |
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| 15:16.37 | Qwell | g011um: It's not likely that your issue can be solved here, if it's with the Elastix GUI. |
| 15:21.48 | [TK]D-Fender | g011um: Everything "depends". Tell us your specific issue and we'll see what we can advise |
| 15:23.00 | g011um | ok |
| 15:23.11 | g011um | default language for sounds is en |
| 15:23.27 | g011um | i would like fr as default language |
| 15:25.32 | g011um | when you try to join an unregistered user, the default language is EN. I know how to change the language in the dialplan for meetme, sip, ect |
| 15:26.28 | g011um | but if i try to dial a bad extension (non existent extension) messages are EN |
| 15:27.34 | g011um | in our configuration, we have an ooh323 trunk between an avaya pbx and the elastix box |
| 15:28.20 | g011um | calls are routed from avaya to elastix through this trunk |
| 15:28.54 | g011um | on the elastix, language=fr is declared in ooh323.conf |
| 15:29.44 | g011um | we have declared virtual extension with voicemail on elastix |
| 15:30.47 | g011um | we already have two elastix box. And as workaround we have renamed sounds/fr in sounds/en |
| 15:31.11 | g011um | but it's not the good way, just the only one we have found |
| 15:32.10 | g011um | we have gold support for our elastix servers so we can ask the support for this issue. |
| 15:32.28 | g011um | but if you have an answer.. |
| 15:32.32 | workingcats | this may be a dumb question, but then why don't you ask them ;) |
| 15:33.29 | g011um | because i would like to understand why |
| 15:34.27 | workingcats | they should know that |
| 15:34.28 | g011um | with a why i could know if it's a bug or a configuration issue. |
| 15:34.41 | g011um | yes i think so |
| 15:35.38 | [TK]D-Fender | [10:25]g011umwhen you try to join an unregistered user, the default language is EN. I know how to change the language in the dialplan for meetme, sip, ect <- this is a general setting for defaults |
| 15:36.06 | [TK]D-Fender | g011um: Next you need to actually HAVE all the proper sound files present and they should be in the PROPEr folders |
| 15:36.16 | g011um | yes i have check in different context |
| 15:36.25 | [TK]D-Fender | g011um: Then you have to set the language not only in your extensions, but also your routes, etc |
| 15:36.41 | g011um | files have the good right at the good location |
| 15:37.23 | g011um | yes i think that it can be in the inbound route |
| 15:39.05 | [TK]D-Fender | remove all "can's" and SHOW us what you've done, and a call being passed through |
| 15:39.07 | [TK]D-Fender | ~pb |
| 15:39.07 | 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. |
| 15:39.09 | [TK]D-Fender | ^^^ |
| 15:39.21 | [TK]D-Fender | www.tinypic.com for screenshots (or similar site) |
| 15:39.35 | [TK]D-Fender | imgur.com , picpaste.com <- maybe best |
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| 15:43.14 | *** join/#asterisk brittel (~jack@542713e4.ftth.concepts.nl) |
| 15:43.17 | brittel | hi |
| 15:43.44 | brittel | Is there a way to get call status codes before the call is actually connected? |
| 15:44.03 | [TK]D-Fender | AAs in? |
| 15:44.22 | brittel | To determine the validity of the phonenumber |
| 15:44.31 | brittel | without actually connecting the call to both endpoints |
| 15:45.20 | brittel | I need ppl to fill in a "valid" phonenumber, i want to check that wihout actually calling them, but just use possible status codes |
| 15:45.49 | [TK]D-Fender | There is no such thing as "check without calling". |
| 15:45.57 | [TK]D-Fender | Proof requires acceptance |
| 15:46.30 | brittel | Let me rephrase |
| 15:46.50 | brittel | I place the call, but before it actually rings I want to disconnect the call |
| 15:47.08 | [TK]D-Fender | Doesn't work that way. |
| 15:47.41 | brittel | Hmm I have seen it working on ISDN |
| 15:47.49 | [TK]D-Fender | > CALL requested to 12345 |
| 15:47.57 | [TK]D-Fender | < Let me Try |
| 15:47.58 | brittel | Where the in band data status codes are used to determine the validity of calls |
| 15:48.03 | [TK]D-Fender | < RINGING |
| 15:48.14 | [TK]D-Fender | or you get < FAILURE X |
| 15:48.37 | [TK]D-Fender | that news arrives at the same time |
| 15:48.45 | brittel | These would be SIP status codes correct? |
| 15:48.55 | [TK]D-Fender | same concept |
| 15:49.00 | brittel | I know there things are used in call centers for example |
| 15:49.03 | brittel | these* |
| 15:49.14 | brittel | in Predictive dailers for example |
| 15:49.20 | *** join/#asterisk Blashyrkh (~michaeld@94.136.25.97) |
| 15:49.34 | [TK]D-Fender | those run off lists that are pre-determined. |
| 15:49.36 | Blashyrkh | after updating asterisk from 10.4 to 11.7 i have problems with my iaxmodems |
| 15:49.54 | [TK]D-Fender | And upon error thoses system REMOVE numbers as appropriate |
| 15:49.58 | Blashyrkh | they can no longer connect to asterisk and i get a registrations timed out when i try iaxmodem ttyIAX10 |
| 15:50.00 | traph | what could be the cause of a softphone not getting sound and other having no such problem?? |
| 15:50.01 | brittel | [TK]D-Fender: The hang up before it rings |
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| 15:50.29 | [TK]D-Fender | brittel: Predictive dialers... DIAL,, they don't "not dial" |
| 15:50.56 | brittel | I know, the do dial, it's the hang-up part |
| 15:50.59 | [TK]D-Fender | SIP has no abort spot that's safe to ensure "no ringing" |
| 15:51.03 | Blashyrkh | password, username are coorect, requirecalltoken set to no, but they are not peering |
| 15:51.15 | brittel | probably they are using regular trunk lines then |
| 15:52.04 | [TK]D-Fender | brittel: If there is some signalling in QSIG for this, Asterisk has no means of using it, so that's an automatic dead-end for you |
| 15:52.24 | brittel | [TK]D-Fender: Thanks for the information |
| 15:52.56 | brittel | i'll check the ISDN signalling protocol |
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| 15:53.03 | *** join/#asterisk dbazor (~ircap@159.Red-79-150-251.dynamicIP.rima-tde.net) |
| 15:53.48 | dbazor | hi people |
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| 15:59.00 | vNistelroot | hi people |
| 15:59.16 | brittel | [TK]D-Fender: there appears to be QSIG support in chan_capi |
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| 16:03.01 | [TK]D-Fender | brittel: Dial has no cut-off function <- |
| 16:03.32 | [TK]D-Fender | brittel: Regardless of certain signalling potential sequence (possibly region / telco specific), ASTERISK has no means of enacting it |
| 16:06.15 | brittel | [TK]D-Fender: You mean SIT? |
| 16:07.52 | [TK]D-Fender | SIT is a failure status. You get either failure... or progress. You don't get, "it's gonna work but not ringing yet" |
| 16:08.00 | dbazor | guys I have a little problem with an iax2 extension. It's well registered, when I use the "iax2 show peers" command I can see in the status that is "OK" but after a few minutes/hours, the status changes to "UNKNOWN" and I can only make it register again by rebooting the pbx. I've tried restarting the iaxmodem service, the hylafax service and using the "iax2 reload" command, but nothing worked. Any idea how can I solve this? Thank |
| 16:08.10 | brittel | [TK]D-Fender: check |
| 16:09.09 | brittel | [TK]D-Fender: another small question. If this functionality is embedded in QSIG, it appears I need a fysical PRI/E1/T1 connection in order to make use of tdm. is that correct? |
| 16:09.14 | [TK]D-Fender | "ringing", "busy", maybe "congestion" could be "valid", but you're not goign to know until you hit one. And you could just as easily get "ringing" as any failure |
| 16:09.21 | [TK]D-Fender | You don't get to stop them prior |
| 16:09.22 | brittel | (has been a while since i wondered in this world) |
| 16:10.14 | [TK]D-Fender | brittel: H.323 is supposed to be a functional equivalent to Q.SIG IIRC. That should have the same potential I think. SIP is NOT. |
| 16:11.06 | brittel | [TK]D-Fender: I'll investigate h323 (stumled on that one as well) |
| 16:17.55 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 16:17.58 | brittel | [TK]D-Fender: you happen to know a h.323 provider? |
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| 16:20.40 | WIMPy | If you want to do it over IP you could use SIGTRAN as well, or pay someone to give you some other interface to validate numbers. |
| 16:21.43 | brittel | WIMPy: "who" might i be looking for then? :) |
| 16:22.18 | workingcats | brittel, what are you trying to do, check for a list of phone numbers whether they're still valid? |
| 16:22.43 | WIMPy | I don;t even know where you are. But here there are services that will tell you which telco hosts a given number. That way you would also find out if it's valid at all. |
| 16:23.02 | brittel | workingcats: correct |
| 16:23.33 | workingcats | hm |
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| 16:24.23 | WIMPy | Or just do it the oldschool way. |
| 16:24.44 | [TK]D-Fender | [11:17]brittel[TK]D-Fender: you happen to know a h.323 provider? Nope, and it'd be a question of who actually passed the full spec. Even then, ASTERISK does not support this |
| 16:25.38 | brittel | i understand that Asterisk does not support it, the question just became more general :) |
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| 16:26.19 | brittel | WIMPy: It's a rather elaborate list of numbers, so we just can't go and call all of those :) |
| 16:27.03 | WIMPy | Why not? |
| 16:27.22 | [TK]D-Fender | That's why telemaketers pay for lists of verified numbers |
| 16:27.31 | brittel | We are not telemarketeers :) |
| 16:27.34 | WIMPy | Well, if you do it too fast, you're telco will kill you. How many do you have and how much time? |
| 16:27.46 | brittel | 500k+ |
| 16:27.57 | [TK]D-Fender | Even worse |
| 16:28.01 | brittel | time is not really a factor |
| 16:28.13 | brittel | (within certain bounds, it can be an ongoing process) |
| 16:28.38 | brittel | I understand the telco inclanation that they will kill us if we start doing this |
| 16:29.00 | WIMPy | At one per seond that would be 6 days. |
| 16:30.19 | brittel | WIMPy: True, but that also means that phones of ppl will be ringing; that would be rather annoying |
| 16:30.58 | WIMPy | No |
| 16:31.36 | brittel | WIMPy: I assume you are referring to SIGTRAN? |
| 16:31.39 | WIMPy | Just make data calls. |
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| 16:32.01 | brittel | WIMPy: How would that work? |
| 16:32.07 | WIMPy | It might give false results with some PBXs, however. |
| 16:32.48 | WIMPy | Same as any call, just with another bearer service. |
| 16:33.08 | Blashyrkh | i still have the problem with iaxmodem not registering to asterisk. In netstat i can see thaht asterisk is listening on 0000/4569, but the recv-q is getting higher |
| 16:33.14 | Blashyrkh | with each try to connect iaxmodem |
| 16:33.19 | brittel | Can you elaborate a bit more? (aka can i PM you? |
| 16:33.21 | Blashyrkh | can anyone make sense of this? |
| 16:33.33 | WIMPy | That's all there is about that. |
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| 16:42.22 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:42.23 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 16:42.24 | darkdrgn2k | hi all |
| 16:43.22 | darkdrgn2k | When graphed by VOIP MONITOR all RTP streams seem to have a declining angle when it comes to delay.. is there a reason for it? |
| 16:45.58 | Qwell | Over how much time? It could be a number of things. The first that comes to mind is jitter buffer |
| 16:46.57 | *** join/#asterisk troyt (~troyt@2601:7:6200:1382:44dd:acff:fe85:9c8e) |
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| 16:48.53 | darkdrgn2k | IE: |
| 16:49.02 | darkdrgn2k | http://imagebin.org/294660 |
| 16:49.15 | darkdrgn2k | left axis is labled 80ms 40ms 0ms |
| 16:52.40 | *** join/#asterisk Newbieeee (~Begnerrr@216-191-88-42.dedicated.allstream.net) |
| 16:52.45 | Newbieeee | i need help |
| 16:53.16 | Newbieeee | is it possible to forward a specific extension to one specific did on zap channel? |
| 16:53.42 | [TK]D-Fender | Thre is no such thing as "forwarding" |
| 16:53.49 | [TK]D-Fender | Your dialplan does whatever you tell it to. |
| 16:53.58 | WIMPy | Newbieeee: If you have a time machine. zap is dead. |
| 16:54.06 | [TK]D-Fender | That too |
| 16:54.10 | WIMPy | And yes, define "forward". |
| 16:55.46 | Newbieeee | ok thank you but how do we do it .. like a general idea? what I am required to do is : "IF WE A PERSON AN EXTENSION 1234 CALLS A LONG DISTANCE NUMBER, IT GOES TO ONE DID xxxxxxxx |
| 16:55.58 | Newbieeee | while others uses all other lines |
| 16:56.16 | Penguin | Dial() |
| 16:56.16 | darkdrgn2k | Qwell: is that jitter buffer - http://imagebin.org/294660 |
| 16:56.54 | Penguin | The extension needs to Dial() whatever you want. |
| 16:57.05 | WIMPy | Newbieeee: Your terminology is confusing. And your grammar as well. |
| 16:58.28 | Newbieeee | I apologise for my grammer and thank you for your cooperation. is there any article for a beginner make a dialplan like this |
| 16:58.34 | [TK]D-Fender | Newbieeee: You just described... one person dialing a LD number... and then dialing ANOTHER fixed number instead |
| 16:58.43 | [TK]D-Fender | Newbieeee: Your description is broken. |
| 16:58.52 | Penguin | ~book |
| 16:58.52 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:58.56 | [TK]D-Fender | ^^^ |
| 16:59.04 | Qwell | darkdrgn2k: It sounds like you're inventing problems, really. |
| 16:59.11 | Qwell | Are you having call quality issues? |
| 16:59.40 | darkdrgn2k | its more of a question then a solution to a problem |
| 16:59.59 | darkdrgn2k | I am having some dropped calls... but not necessarliy on that graph.. just trying to understand the data |
| 17:00.09 | darkdrgn2k | and im curiouse why the graphs all show a downward slope |
| 17:00.49 | Newbieeee | :S ok let me google it and get familiar with the terminology first |
| 17:00.56 | Newbieeee | thank you |
| 17:01.30 | Penguin | Extensions execute applications. The application used to send a call to another device is Dial(). |
| 17:01.39 | darkdrgn2k | last night i managed to get into the router and define priosity QOS for the openvpn traffic that holds the sip traffic. |
| 17:01.44 | [TK]D-Fender | Newbieeee: get your idea straight in you your head first. |
| 17:02.24 | Penguin | If you want extension 1234 to Dial() a "long distance" number, make it happen. It's not hard to accomplish. |
| 17:03.05 | Penguin | Determine the channel tech you are going to use, if a specific channel or peer needs to be used, and the number to call. |
| 17:04.56 | Newbieeee | thank you Penguin. |
| 17:05.43 | darkdrgn2k | Qwell: so what coudl cause that.. cause if i read the chart right it shows thea the call gets lower latency as time goes on? |
| 17:06.15 | Penguin | But the phrase, "if we a person an extension..." is rather hard to decipher. |
| 17:08.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.200) |
| 17:08.59 | Newbieeee | yes Penguin- let me re write it. Here is the actual problem: one of the employee in our office has an extension 1234. Now when he dials a long distance call through zap - the number that is displayed on the client end is 6473035226, while I want the displayed number to be 6473035111 |
| 17:09.14 | [TK]D-Fender | newthat is the CALLERID you want to set |
| 17:09.18 | [TK]D-Fender | Newbieeee: that is the CALLERID you want to set |
| 17:09.51 | [TK]D-Fender | Newbieeee: So Go make your extension match the callerid of your employee and then change the outbound callerid accordingly |
| 17:10.46 | [TK]D-Fender | Newbieeee: exten => _[dialed-pattern]/[callerid_ofcaller],priority,Application(data) |
| 17:11.12 | WIMPy | Or set the callerID of the device he's calling from. |
| 17:11.40 | [TK]D-Fender | Newbieeee: Or use a GotoIf and check it in the actual dialplan code for the general pattern. |
| 17:11.43 | *** join/#asterisk serafie (~erin@nat/digium/x-owdpfvniizxslclr) |
| 17:12.03 | WIMPy | Or put that device in to another context. |
| 17:12.46 | Newbieeee | thank TK so this is what I have done - using GUI web Interface(as I am a beginner)- I went to extensions - clicked on his extension and wrote "GISI"<6473035111> - applied the configurations but still no result |
| 17:14.06 | [TK]D-Fender | Newbieeee: Then trow out everything we just told you. YOU are not configuring your system ... your GUI is. |
| 17:14.09 | [TK]D-Fender | throw* |
| 17:14.30 | [TK]D-Fender | Newbieeee: And you are in the wrong channel. Go ask how to proerly do this in your GUI's support channel |
| 17:14.42 | Newbieeee | and what is that channel |
| 17:14.57 | Newbieeee | like the name of the channel - sorry guys |
| 17:14.58 | Penguin | What is your GUI? |
| 17:15.15 | Newbieeee | it is a web interface for asterisk |
| 17:15.20 | Penguin | Name? |
| 17:15.33 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 17:16.39 | *** join/#asterisk serafie (~erin@nat/digium/x-hyzxezflwuvztjhq) |
| 17:17.02 | Penguin | If it doesn't have a name, you're out of luck. |
| 17:17.12 | Newbieeee | Penguin I a totally newbie to asterisk |
| 17:17.18 | Penguin | I know. |
| 17:17.23 | Newbieeee | am* |
| 17:17.47 | Newbieeee | so apologies, I am trying to find out but I dont see a name anywhere |
| 17:17.55 | [TK]D-Fender | show us a SCREENSHOT <- |
| 17:18.02 | [TK]D-Fender | picpaste.com <- |
| 17:18.03 | Newbieeee | ok |
| 17:18.07 | Penguin | You've chosen to use a GUI instead of configuring asterisk. How did you select the GUI to use? |
| 17:19.48 | Newbieeee | http://picpaste.com/FreePBX_administration_-_Google_Chrome_2014-02-20_12-18-37-oCP12fUs.png |
| 17:19.57 | Penguin | hmm |
| 17:20.04 | Penguin | FreePBX administrations? |
| 17:20.17 | Qwell | a rather old one, at that |
| 17:20.18 | Penguin | And you can't figure out the name of the program is FreePBX? |
| 17:20.43 | [TK]D-Fender | That is clearly an ancient hack-job of FreePBX |
| 17:20.50 | Penguin | Trixbox? |
| 17:20.50 | [TK]D-Fender | It may be a fork... |
| 17:20.54 | [TK]D-Fender | Quite probably |
| 17:20.55 | Newbieeee | Penguin i am actually a virtualization specialist and I am totally zero on voip - this is just I am thrown by my top management to look into this problem as the voip specialist has left the company |
| 17:20.56 | Penguin | Elastix? |
| 17:21.17 | [TK]D-Fender | That looks like 2.8 or prior... quite old |
| 17:21.33 | Newbieeee | yes its v old |
| 17:21.46 | [TK]D-Fender | Penguin: No, Elastix doesn't use the main page format like that. They super-hack access to eaach modules config screens their own way |
| 17:21.58 | Newbieeee | in one of the small branch offices |
| 17:22.44 | [TK]D-Fender | Newbieeee: Use a CallerID match in your OUTBOUND ROUTE to match those specific callers and then set the outbound CallerID before it goes to your trunk. |
| 17:22.59 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 17:23.13 | Penguin | ~freepbx |
| 17:23.13 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:23.26 | Newbieeee | it gives me asterisk info 2.5.0.2 in teh modules section- thats how I think it is asterisk |
| 17:23.40 | [TK]D-Fender | that is a FREEPBX version, not an ASTERISK version |
| 17:23.46 | [TK]D-Fender | And that is positively ancient |
| 17:24.07 | Newbieeee | hm alright |
| 17:29.42 | [TK]D-Fender | Newbieeee: "48 bad destinations" says you broke a LOT of stuff.... |
| 17:40.52 | *** join/#asterisk serafie1 (~erin@nat/digium/x-abeovnfkefemthgw) |
| 17:41.01 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:48.34 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 17:49.02 | *** part/#asterisk Blashyrkh (~michaeld@94.136.25.97) |
| 17:50.22 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 17:57.39 | Katty | navaismo: time for more script re-arranging |
| 17:58.52 | Newbieeee | lol tk |
| 17:59.30 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 18:00.39 | Katty | rehi, navaismo |
| 18:00.46 | navaismo | o/ |
| 18:01.04 | Katty | navaismo: the database compiled. |
| 18:01.12 | navaismo | awesome |
| 18:01.21 | Katty | it's doing other fancy things now |
| 18:02.51 | navaismo | hehe |
| 18:04.27 | Katty | compiling compiling compiling. |
| 18:04.29 | Katty | it takes forever. |
| 18:05.28 | spillere | is there a way to list all codecs installed on a linux distro with asterisk? |
| 18:05.57 | Katty | core show codecs, perhaps? |
| 18:07.04 | spillere | oh, great! ty |
| 18:07.49 | spillere | i installed speex, have selected speex on linphone (iphone and mac) but it still says "no compatible codecs" |
| 18:11.17 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.200) |
| 18:11.25 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 18:12.07 | [TK]D-Fender | [13:05]spillereis there a way to list all codecs installed on a linux distro with asterisk? <- has nothing to do with distro. has everything to do with Linux. |
| 18:12.25 | [TK]D-Fender | spillere: They are compiled modules from Asterisk itself. |
| 18:12.32 | spillere | i mean that, sry |
| 18:13.25 | spillere | when i list 'core show codecs' it shows '100009 audio speex (SpeeX)'. I set speex codec on linphone and it say no compatibility |
| 18:13.28 | spillere | can i show you the log? |
| 18:14.14 | [TK]D-Fender | live CLI dump. verbose 10, core debug 0, sip debug on |
| 18:16.16 | *** part/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 18:18.19 | spillere | all those codes? |
| 18:19.03 | [TK]D-Fender | 3 things 1 place, know your syntax |
| 18:19.04 | Penguin | No codes, all requirements. |
| 18:19.58 | spillere | sorry im new to asterisk, trying to understand all :) |
| 18:20.21 | spillere | how to make a CLI dump to file? |
| 18:21.15 | [TK]D-Fender | Don't |
| 18:21.18 | [TK]D-Fender | just grab it live |
| 18:21.30 | spillere | let me try to call and save |
| 18:22.02 | *** join/#asterisk foamz (~foamy@pool-108-41-222-138.nycmny.fios.verizon.net) |
| 18:24.40 | spillere | [TK]D-Fender: http://pastie.org/private/zqfef1wshtz2s5lpsskmg |
| 18:26.16 | Penguin | I hate when people use pastie.org... it does not wrap the text. |
| 18:26.40 | [TK]D-Fender | Show us the peer |
| 18:26.44 | Penguin | Having to scroll side to side to read things is a real inconvenience. |
| 18:26.49 | [TK]D-Fender | Penguin: I always go RAW on those |
| 18:26.55 | spillere | what is a good pastie alternative? |
| 18:27.04 | Penguin | If it wraps, raw would be fine. |
| 18:27.32 | Penguin | Nope, it doesn't wrap either. |
| 18:27.44 | Penguin | pastebin.com |
| 18:27.44 | [TK]D-Fender | spillere: just paste it... |
| 18:29.52 | spillere | *CLI> [Feb 20 13:26:34] NOTICE[3514]: chan_sip.c:23522 handle_response_peerpoke: Peer 'user002' is now Reachable. (307ms / 2000ms) |
| 18:29.52 | spillere | [Feb 20 13:26:34] NOTICE[3514]: chan_sip.c:23522 handle_response_peerpoke: Peer 'user001' is now Reachable. (367ms / 2000ms) |
| 18:29.52 | spillere | <PROTECTED> |
| 18:29.54 | spillere | [Feb 20 13:27:28] NOTICE[3514][C-00000000]: chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer! |
| 18:30.15 | Penguin | That's not "the peer" that he said to show. |
| 18:30.26 | spillere | ahh, ops lol |
| 18:30.27 | MaliuatLap | ~pb |
| 18:30.28 | 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. |
| 18:30.30 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 18:30.33 | spillere | you mean sips.conf |
| 18:31.56 | spillere | http://paste.lisp.org/display/141310 |
| 18:32.45 | Penguin | The defaultuser lines do not belong in there. |
| 18:33.07 | spillere | do i remove from configs? |
| 18:33.10 | [TK]D-Fender | spillere: I see you also didn't bother to specifically SET your codecs |
| 18:33.13 | [TK]D-Fender | ^ |
| 18:33.27 | spillere | im using allow=all |
| 18:33.46 | Penguin | disallow=all then allow ONLY the ones you want to be available to the devices. |
| 18:33.48 | [TK]D-Fender | I don't trust that is in the right order in the right place... or much of anything els |
| 18:33.57 | [TK]D-Fender | eSet it PER DEVICE |
| 18:34.33 | spillere | i was thinking on allow all, and then configure on the device |
| 18:35.04 | Penguin | If you only want to use speex, disallow=all \ allow=speex |
| 18:35.16 | spillere | just did that |
| 18:35.20 | [TK]D-Fender | spillere: You always want to set every device explicitly. |
| 18:35.41 | spillere | so for user001, user002, i add te allow/disallow? |
| 18:35.52 | Penguin | disallow first, allow second. |
| 18:37.16 | spillere | http://piratepad.net/XLoQylAnpz |
| 18:37.39 | [TK]D-Fender | sure. Not sure if it's speex or speex16 |
| 18:37.42 | [TK]D-Fender | Give it a try |
| 18:37.44 | Penguin | This is the third paste site you've used. What's the aversion to pastebin.com? |
| 18:38.12 | spillere | pad you can edit |
| 18:38.30 | [TK]D-Fender | so does pastebin... right at the bottom |
| 18:38.52 | spillere | ok |
| 18:40.51 | BeachBall | anyone ever get astpp working on asterisk? |
| 18:41.01 | BeachBall | and is it worth installing, or old and useless? |
| 18:41.23 | spillere | changed to speex16 and enabled all kinds of speex on linphone and its working |
| 18:43.21 | Penguin | There's speex and speex16. |
| 18:43.51 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 18:47.17 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-100-80.user.veloxzone.com.br) |
| 18:49.21 | Nugget | I just wired an old speak-n-spell up to a clone x100p card I bought off ebay. It sounds better than speex. :) |
| 18:53.10 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.200) |
| 18:58.44 | anonymouz666 | moy: what's the recommended way to debug backwards "muted calls"? I am looking for something to debug the audio. the calls files are enabled and I can see the "clear forward" coming from telco. |
| 19:03.02 | Penguin | I wish someone would /mode +b vlad_starkov!*@*$##fix_your_connection |
| 19:09.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.200) |
| 19:09.51 | *** mode/#asterisk [+b *!*vlad_star@109.188.125.*] by Qwell |
| 19:09.51 | *** kick/#asterisk [vlad_starkov!~north@pdpc/sponsor/digium/Qwell] by Qwell (Message an op when you have a more stable connection.) |
| 19:10.02 | Penguin | <PROTECTED> |
| 19:10.08 | Qwell | slow |
| 19:10.29 | Penguin | He uses more than one netblock and the memo would be useful. |
| 19:11.11 | *** mode/#asterisk [+b vlad_starkov!*@*$##fix_your_connection] by Qwell |
| 19:11.13 | Qwell | shrugs |
| 19:11.18 | *** mode/#asterisk [-b *!*vlad_star@109.188.125.*] by Qwell |
| 19:16.38 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 19:18.33 | anonymouz666 | dahdi_pcap works with MFCR2 or ISDN? |
| 19:18.37 | anonymouz666 | ou just SS7? |
| 19:28.18 | *** join/#asterisk CeBe (~CeBe@port-92-206-64-148.dynamic.qsc.de) |
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| 19:44.04 | *** join/#asterisk spillere (~spillere@molus.co) |
| 19:47.02 | spillere | in asterisk, my connection from user1 to user2 passes trough the server. Which kind of configuration do I need to do in order to have the server only handling calls, and one it's placed, user1 and user2 are connected directly, making the delay smaller? |
| 19:50.05 | BeachBall | wants to see how the ##ban thingy works |
| 19:50.31 | MaliuatLap | spillere: look at canreinvite/directmedia http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
| 19:50.38 | spillere | thanks |
| 19:50.42 | BeachBall | @206.162.174.* |
| 19:50.45 | BeachBall | ;D |
| 19:50.48 | BeachBall | try it on me |
| 19:50.50 | BeachBall | Qwell |
| 19:52.24 | BeachBall | :/ |
| 20:06.50 | Penguin | When you get a ban on you, you don't get to speak in the channel. If you leave, you won't be able to come back in. |
| 20:07.00 | Penguin | The part after ## is just a memo so you can see it in the ban list. |
| 20:25.11 | *** join/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
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| 20:26.13 | volga629 | Hello Everyone, I have how to get right way value from db https://fpaste.networklab.ca/gDES/ |
| 20:26.55 | volga629 | this snippet to send message to users like 101&10101 |
| 20:30.51 | *** join/#asterisk xaristax (~xaristax@148.240.65.35) |
| 20:34.51 | [TK]D-Fender | use pastebin.com |
| 20:34.57 | [TK]D-Fender | This one gives a certificate issue |
| 20:40.02 | volga629 | ups |
| 20:40.39 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:40.48 | volga629 | http://fpaste.org/79047/39292876/ |
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| 20:50.26 | Kobaz | okay sooo |
| 20:50.31 | Kobaz | you polycom peoples |
| 20:50.52 | Kobaz | if on a polycom 550, you hit **, it replaces the dial string with a + |
| 20:51.02 | [TK]D-Fender | volga629: that is the right syntax for DB(). No go proves what is IN the DB |
| 20:51.04 | Kobaz | or you dial ** on a 330/etc it works fine |
| 20:51.04 | [TK]D-Fender | now* |
| 20:51.18 | Kobaz | anyone know how to make the 550 behave "normally" and just send asterisk a ** |
| 20:52.26 | volga629 | pbx01*CLI> database showkey DEVICE |
| 20:52.29 | volga629 | /AMPUSER/101/device : 101&10101 |
| 20:52.54 | [TK]D-Fender | <PROTECTED> |
| 20:53.11 | [TK]D-Fender | <PROTECTED> |
| 20:53.21 | [TK]D-Fender | You are dialing 10101, NOT 101 |
| 20:53.36 | [TK]D-Fender | There is no /AMPUSER/10101/device |
| 20:54.45 | volga629 | yes you right, I am trying pull devices assigned to 101 and send message to each device |
| 20:55.03 | MaliuatLap | well at least dialing a non-existent extension stops the spammers getting in ;) |
| 20:55.22 | [TK]D-Fender | You told it to look for something that isn't going to be there. |
| 20:56.04 | volga629 | this whole message sent http://fpaste.org/79053/39292972/ |
| 20:56.05 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 20:56.06 | MaliuatLap | This is why I find AEL easier to express my dialplans |
| 20:57.13 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 20:57.27 | [TK]D-Fender | meh.... |
| 20:57.57 | *** join/#asterisk generalhan_ (~generalha@about/windows/staff/generalhan) |
| 21:01.31 | xaristax | its recommended to put qualify i a user |
| 21:01.34 | xaristax | ? |
| 21:02.11 | xaristax | i ask this because i delete the qualify directive on one of the external phones and works like a charm but in what can affect me? |
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| 21:03.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 21:31.32 | volga629 | ok I have right now working properly, -- Executing [101@netlab-msg:7] Set("Message/ast_msg_queue", "PEER_GRP=101&10101") in new stack |
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| 21:32.18 | volga629 | yand when will be mobile ext 10101 I will look for first value in DB and then route, send message |
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| 22:16.19 | xaristax | can someone tell how to change the Reg. default duration please :D |
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| 22:25.04 | volga629 | how good way get from extension Set(PEER_IN=${SIPPEER(callerid_num)}) ? |
| 22:28.58 | Penguin | xaristax: That's set by the other side. |
| 22:29.55 | xaristax | Penguin: you mean the phone? |
| 22:30.07 | xaristax | im not so good with my english :P |
| 22:30.13 | gusto | hi Penguin |
| 22:30.18 | gusto | how are your mirrors? |
| 22:30.39 | Penguin | xaristax: Describe the problem, not the solution you are looking for. |
| 22:31.29 | gusto | xaristax: you are from mexico? |
| 22:31.53 | navaismo | i am |
| 22:32.06 | xaristax | Penguin i been suffering with some external phones they become UNREACHABLE and also dont let me make phone calls but if i use ignoreregexpire works like charm to put calls but when i try to reach the phone its UNREACHABLE |
| 22:32.54 | xaristax | so im guessing the problem is with the regtime i try to play a little bit with qualify without success and i put a registry every 10 seconds on the phone and also works but im affraid to load my PBX |
| 22:32.59 | gusto | you have a firewall in between? |
| 22:33.04 | Penguin | xaristax: Most phones have a registration interval setting. It might be too long. |
| 22:33.17 | gusto | did you try to do it without authentication? |
| 22:33.18 | vader- | any of you guys seen freepbx/asterisk run in Amazon AWS? |
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| 22:33.40 | gusto | lol, amazon |
| 22:33.59 | xaristax | penguin i set the interval time to 10 seconds and it works but i think is a very short period of time |
| 22:34.02 | Penguin | xaristax: You can enable qualify to provide NAT keepalive messages if you think NAT could be causing the loss of registration. |
| 22:34.16 | xaristax | gusto yep i have a firewall in between with the known ports open |
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| 22:34.47 | xaristax | i enable qualify but not success |
| 22:35.12 | xaristax | if i enable the qualify through me UNREACHABLE and if i set it to no put unmonitored |
| 22:36.12 | xaristax | i mean i already solve in someway the problem setting the registry every 10 seconds on the phone but i have 30 external phones i dont wanna make a mess with my pbx |
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| 22:38.46 | gusto | xaristax: is it NATted? |
| 22:39.00 | xaristax | gusto: yep |
| 22:39.09 | gusto | xaristax: AHA! |
| 22:39.13 | xaristax | you think that could be the problem? |
| 22:39.20 | gusto | of course |
| 22:39.30 | xaristax | how can i solve it :D |
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| 22:39.43 | gusto | you would not need qualify at all if there was no NAT |
| 22:39.49 | Penguin | If you set the reg interval to 300 seconds, how long before they go unreachable? |
| 22:40.10 | xaristax | i take time and it takes aprox 1 minute |
| 22:40.23 | xaristax | to become unreachable even if i delete qualify |
| 22:40.24 | gusto | that should not happen |
| 22:40.42 | xaristax | can i set i higher value to the reg interval? |
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| 22:40.56 | gusto | normally NAT routers should keep the connection in table for at least 150 seconds |
| 22:41.29 | Penguin | If you enable qualify, a default qualify frequency is 60 seconds. |
| 22:41.37 | xaristax | so its good to put a registry every 100 seconds to my phone? |
| 22:41.55 | xaristax | yep i already play with that value with no success |
| 22:41.59 | Penguin | Be very sure that you did not change the qualify frequency to something too high for your NAT. |
| 22:42.18 | xaristax | in fact i down that value in some tests that i did |
| 22:42.18 | Penguin | Default is 60 if you leave it commented. |
| 22:42.28 | Penguin | Leave it commented out or set to 60. |
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| 22:42.50 | Penguin | Set your qualify value to yes, not a number value. |
| 22:43.19 | xaristax | done that still the problem at least when i try to reach the phone |
| 22:43.41 | Penguin | The problem is with the NAT. |
| 22:43.58 | xaristax | you can recommend me a place to start diggin? |
| 22:44.05 | Penguin | I just did! |
| 22:44.15 | xaristax | i been breakin my head with this problem |
| 22:44.32 | gusto | but the phone registers, right? |
| 22:44.32 | Penguin | qualifyfreq and qualify <----- this is where you start, and usually finish. |
| 22:44.49 | JeffC_NN1 | would it help to use TCP instead? |
| 22:45.09 | xaristax | i play with these vales already but im going to go deeper on this two things that you recommend |
| 22:45.24 | gusto | not really |
| 22:45.59 | gusto | RTP still has to go over UDP so even if he gets that SIP connection right, he would be cut off diring a call if NAT router ignores data flow |
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