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00:26.59 | file | yawnnnnnnn |
00:47.14 | lvlinux | is there a way to record echo on a call??? |
00:48.09 | lvlinux | I have a client that has a system that is giving a bad echo on PSTN calls---I had them record with automixmon---sounded 100% fine to me, no echo at all. |
00:48.27 | lvlinux | but they said there was echo on the call that they recorded. |
00:49.21 | lvlinux | the fxo card is a Sangoma B600 if that makes any difference. |
00:49.30 | bibz2 | D-Fender, hi there :) |
00:49.55 | lvlinux | and I have OSLEC turned on (at least it says it's active :-) |
00:52.07 | lvlinux | ideas anyone? |
00:55.25 | WIMPy | WHERE has that echo been heard? |
00:56.18 | lvlinux | the Asterisk side is where they here it, not the remote parties. |
00:57.04 | lvlinux | and they are across the country so I can't go hear it myself unfortunately. |
00:57.23 | WIMPy | Then I don't see how that wouldn't be recorded. |
00:58.18 | lvlinux | that's what i thought---but they insist that it's there and very bad. |
00:58.31 | MRH2 | get them to record with their smartphone and send you the file |
00:58.31 | lvlinux | and they affirmed that they were hearing it on the recorded call. |
00:59.07 | MRH2 | if it is echo on speaker |
01:00.07 | bibz2 | when I call a number which is routed to the asterisk, the asterisk dials out to a mobile phone and I can establish a call and even talk to the person on the other end, but after a few seconds there is a weird noise and we can't hear each other anymore.. here is the call: http://pastebin.com/jLQY5ziu (full debug, sorry about that) |
01:02.29 | lvlinux | MRH2: i think I will try that---hard to diagnose when I can't hear it myself |
01:02.39 | MRH2 | is it the same people they call |
01:02.43 | MRH2 | could be the other end |
01:02.54 | lvlinux | no they say it's all calls |
01:03.11 | MRH2 | do they use handsfree speakerphone |
01:03.13 | MRH2 | ? |
01:03.17 | [TK]D-Fender | bibz2: pastebin your [1202] |
01:03.28 | [TK]D-Fender | bibz2: masking only the secret |
01:03.32 | bibz2 | 1202? the extension? |
01:03.39 | [TK]D-Fender | sip.conf section |
01:03.48 | lvlinux | MRH2: I don't think so---I believe the echo is there with the handset. |
01:03.51 | bibz2 | there is no 1202 |
01:03.52 | WIMPy | bibz2: Do you have ICE enabled or something? |
01:03.54 | [TK]D-Fender | or whatever uses that as the authuser |
01:03.57 | bibz2 | I'll paste the whole |
01:04.13 | MRH2 | they have it the right way round? |
01:04.37 | [TK]D-Fender | bibz2: SIP/2.0 200 OK From: "1202" <sip:1202@192.168.3.217>;tag=e66b05bae3c71d55 CSeq: 43728 REGISTER |
01:04.42 | lvlinux | you mean the receiver? haha i sure hope so :) |
01:04.51 | [TK]D-Fender | bibz2: Your server sure seems to think there is a user with that name |
01:05.17 | MRH2 | could be high volume plus latency |
01:05.20 | bibz2 | I have a table which is called sipusers.. there are some users inside |
01:05.46 | MRH2 | like if you call a mobile and have the volumes right up |
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01:05.53 | lvlinux | i backed the txgain off 6db earlier, so maybe that will make a difference? |
01:06.18 | lvlinux | I thought it could be the volume (I've tried lots of echocanceller settings with no change) |
01:06.57 | bibz2 | weird.. there is no 1202 user.. |
01:07.18 | MRH2 | try making an extension that records them and plays it back |
01:07.48 | lvlinux | what do you mean? |
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01:08.01 | MRH2 | even if they leave a voicemail |
01:08.04 | MRH2 | for themselves |
01:08.06 | bibz2 | ok, here is my sip.conf: http://pastebin.com/xvQrmg1P |
01:08.23 | MRH2 | assume this is ok |
01:08.39 | MRH2 | voip is ok |
01:08.42 | MRH2 | is just to pstn |
01:08.48 | lvlinux | yes I believe there is no echo within the office, just when calling out the FXO |
01:09.12 | bibz2 | this issue is driving me crazy. we have 50% lost calls in the last 2 weeks :( |
01:09.36 | bibz2 | long time I thought its because of NAT so we removed the NAT |
01:10.13 | bibz2 | we thought it may be the big queries that are being executed on the SQL server on every call so we upgraded the server .. didn't help |
01:10.25 | bibz2 | new internet provider, didn't help |
01:10.54 | bibz2 | I can't even find in the debug anything helpful. There is no error when the audio interrupts.. nothing |
01:11.04 | [TK]D-Fender | bibz2: You are masking anything that could possibly match that entry I clearly showed you. |
01:11.15 | [TK]D-Fender | bibz2: Go really look... because you aren't showing us. |
01:11.20 | bibz2 | ok |
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01:11.56 | bibz2 | the user 1202 is not existent in the sip.conf. and also not in the sipusers table |
01:12.05 | bibz2 | what are some other places to look? |
01:12.41 | [TK]D-Fender | Look at the IP it's registering from |
01:13.00 | MRH2 | i guess recording is helpful (and proof) - maybe try without the hardware echo cancellation if possible. |
01:13.30 | lvlinux | the card doesn't have any hardware EC, only software. |
01:14.26 | lvlinux | i had it on MG at first and when they complained the first time I switched to OSLEC---they said there was no change in the echo. |
01:15.08 | WIMPy | What kind of phone are they using? |
01:19.18 | lvlinux | Yealink T46G |
01:19.34 | lvlinux | but I had them plug in a Polycom 430 and they said it was the same echo on it. |
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01:23.07 | MRH2 | does the phone do echo cancellation as well? |
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01:23.28 | MRH2 | cause i don;t u think u want both |
01:26.58 | bibz2 | but D-Fender.. I don't understand what the peer 1202 has to do with the call I initiated |
01:27.26 | bibz2 | I have tried to log into that IP, seems to be a Grandstream testing device we have in our office |
01:27.40 | bibz2 | I can't imagine how it could influence other calls |
01:31.44 | [TK]D-Fender | ... |
01:31.55 | [TK]D-Fender | that call you actually placed in there did NOT say NAT |
01:32.28 | [TK]D-Fender | The only thing that did was a REGISTER |
01:32.41 | [TK]D-Fender | You are clearly not even looking at the output you are providing. |
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01:33.34 | bibz2 | but my call-id is 6943857619bf0bcb5b15bcd8219f802b ... |
01:33.58 | [TK]D-Fender | [19:41]bibz2my asterisk isn't really behind a NAT but there are a lot of packets which state (Transmitting NAT) in the header. is that normal? |
01:33.59 | bibz2 | sorry if I explained it wrong, or if I just don't understand it |
01:34.01 | [TK]D-Fender | You asked this earlier |
01:34.25 | [TK]D-Fender | "is tht normal?" <- |
01:34.31 | bibz2 | I have removed the nat=force_rport,comedia line which was "hidden" on top of sip.conf |
01:34.40 | bibz2 | I thought that should've done it.. |
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01:35.02 | [TK]D-Fender | Doesn't matter where it is.... it matters what you TELL it about itself and other peers that may try to connect |
01:35.21 | bibz2 | so its still a NAT problem, even if the packet corresponding to my call states NO NAT? |
01:35.50 | [TK]D-Fender | No, we are talking about 2 different things |
01:36.11 | [TK]D-Fender | you have changed topic and still not realized they are SEPARATE points |
01:36.12 | bibz2 | only that call (1202) seems to be natted.. shouldn't affect my call.. or should it? |
01:36.25 | bibz2 | well that wasn't my intention |
01:36.30 | [TK]D-Fender | You asked the question |
01:36.36 | [TK]D-Fender | and provided the debug,. |
01:36.40 | bibz2 | seems like I'm confused now |
01:37.09 | [TK]D-Fender | at elast in as much as remembering the questions you asked. |
01:37.16 | [TK]D-Fender | least* |
01:37.31 | [TK]D-Fender | As for your audio... I see several peers with no reinvite rules. |
01:37.47 | [TK]D-Fender | You should just prevent them explicitly in all your peers and under [general] |
01:37.49 | bibz2 | ok, I try it one more time to describe the problem. without taking the first question about NAT in conclusion |
01:38.04 | [TK]D-Fender | [pbx3] <- not set |
01:38.33 | [TK]D-Fender | [sipgate-3] <- missing the "nat=no" they certainly require |
01:38.54 | [TK]D-Fender | You have done little pieces of your job spread across all these entires and have not been thorough or consistent |
01:39.01 | [TK]D-Fender | Go clean all of them up |
01:39.18 | bibz2 | I should either work on my discipline or sleep more |
01:39.24 | bibz2 | thanks |
01:44.04 | bibz2 | ok, set every peer to nat=no. |
01:45.34 | bibz2 | reinvite rules should be set? seems like I have commented it out for some reason.. |
01:45.45 | bibz2 | probably while experimenting around in the beginning.. |
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01:48.53 | [TK]D-Fender | directmedia=no |
01:49.06 | [TK]D-Fender | \"canreinvite" is long been deprecated |
01:50.19 | bibz2 | oh you were right. seems like pbx3-out has this set, but pbx3 not... |
01:56.17 | bibz2 | damnit, now its too late to test anytihng since there aren't any agents signed in anymore. should better change my working hours |
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03:06.20 | arosen1 | Can you have one phone number and load balance that over several asterisk boxes? I've been googling around and it seems like you need to use: OpenSIPS/OpenSER/Kamillo but all the things i'm finding are ~2009 time peroid |
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04:13.21 | darkdrgn2k3 | whats the latest polycom firmware for asterisk |
04:41.30 | [TK]D-Fender | Thre is no such thing as "polycom firmware for asterisk" |
04:41.42 | [TK]D-Fender | polycom firmware is for polycom |
04:41.53 | [TK]D-Fender | And the latest version... depends on the model. |
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05:21.18 | darkdrgn2k3 | [TK]D-Fender: Latest firmware has a note that its for LYNC only |
05:21.50 | darkdrgn2k3 | "Polycom UC Software 4.1.1 is for use with Microsoft® LyncTM Server and using a single registered line only." |
05:27.34 | [TK]D-Fender | Polycom Unified Communications (UC) Software 5.1.1 Rev B is a general release for all open SIP platforms including Microsoft Lync Server 2010 and Microsoft Lync Server 2013 |
05:27.41 | [TK]D-Fender | Tha implies open as well. |
05:28.57 | [TK]D-Fender | And clearly much newer than your ver you quoted |
05:34.47 | darkdrgn2k3 | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
05:34.53 | darkdrgn2k3 | matrix doesnt go that high !?!? |
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05:35.32 | [TK]D-Fender | You aren't looking very hard or being very specific |
05:36.24 | darkdrgn2k3 | i have a IP 335 |
05:36.33 | darkdrgn2k3 | the support site doesnt specify 5.x either |
05:36.50 | [TK]D-Fender | Context matters |
05:37.01 | [TK]D-Fender | because ther was a link of that page for OTHER Polycom models |
05:37.18 | [TK]D-Fender | Which DOES go higher and support standa along with Lync |
05:37.50 | [TK]D-Fender | You asked about firmware and waited this long before telling us what model you were looking for. |
05:38.13 | darkdrgn2k3 | yes you are right |
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05:46.36 | [TK]D-Fender | 3.3.53.3.5 as listed there... which is what I run on mine... |
05:46.40 | [TK]D-Fender | 3.3.5* |
05:46.48 | [TK]D-Fender | heads to bed |
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06:26.33 | spicyramen_ | anyone using HD video endpoints registered to asterisk 720p+? |
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06:45.13 | dsr | hi |
06:45.25 | dsr | what is the recommended memory requirement for asterisk 12.4? |
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07:33.42 | wdoekes | dsr: those questions are impossible to answer. it depends on the kind of load you expect |
07:36.31 | wdoekes | and even then it's far from trivial. our office pbx (does practially nothing), uses (together with OS/daemons) about 100MB on a 1GB system |
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10:34.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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11:21.22 | WHiZZi | Is it normal that Asterisk 11.10 doesn't actually reload sip when the sip.conf-file is untouched and only the include files are changed ? |
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12:06.00 | wdoekes | WHiZZi: https://issues.asterisk.org/jira/browse/ASTERISK-23683 fixed in 11.11 |
12:06.09 | WHiZZi | ok, tnx :) |
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12:36.20 | paolo_ | [TK]-D-Fender: Hi, so you remember our last conversation. It was about directrtpsetup. I gave you a link to pastebin (debug and sip.conf). Did you have the time to take a look at it ? |
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12:44.43 | marceloamorim | guys, I had an issue with my asterisk, I just tested right now about this, I make a call and the side B answered but the audio wait like 2 or 3 secs to pass through |
12:44.57 | marceloamorim | the call loke is at the same second, but it isn`t |
12:45.20 | marceloamorim | "[2014-08-15 09:41:37] -- DAHDI/2-1 answered SIP/gerente_fantasminha-0000025c" -> " [2014-08-15 09:41:37] > 0x2416f40 -- Probation passed - setting RTP source address to 192.168.2.101:10002 " |
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12:57.17 | marceloamorim | anyone knows how to fix? |
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13:06.40 | omenSP | hey c: who |
13:06.48 | omenSP | is up this morning*] |
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13:10.41 | bibz2 | any freelancers in here? |
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13:10.48 | omenSP | um |
13:10.56 | omenSP | I'm not |
13:11.23 | [TK]D-Fender | lances are expensive these days... |
13:11.51 | [TK]D-Fender | And the time it takes to break a horse? |
13:12.02 | marceloamorim | [TK]D-Fender: can I ask you something? |
13:12.02 | [TK]D-Fender | You can't be serious... |
13:12.10 | [TK]D-Fender | SURELY YOU JOUST?! |
13:12.16 | [TK]D-Fender | :) |
13:12.18 | omenSP | ohey d-fender! |
13:12.24 | omenSP | you helped me out like a month ago. |
13:12.48 | marceloamorim | "[2014-08-15 09:41:37] -- DAHDI/2-1 answered SIP/gerente_fantasminha-0000025c" -> " [2014-08-15 09:41:37] > 0x2416f40 -- Probation passed - setting RTP source address to 192.168.2.101:10002 " |
13:13.16 | [TK]D-Fender | marceloamorim: Stop thinking that single lines like that hold the story.... |
13:13.33 | [TK]D-Fender | marceloamorim: SIP is involved and you aren't showing a full call with matching debug |
13:13.53 | *** join/#asterisk TSM2 (~the_softw@fw-lon2.wenn.com) |
13:14.45 | marceloamorim | this log on 09:41:37 but this isn`t true, after I get those DAHDI/2-1 answered the audio isn`t pass through yet |
13:16.09 | marceloamorim | it takes like 2 or 3 seconds to have tx and rx audio |
13:16.26 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
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13:20.06 | omenSP | Is there an example that I can base my outbound calling routes and trunks off of? I bought the 8-port FXO card and plugged all the trunks into it, then set it up as 8 trunks in the system. Upon restarting Asterisk, outside calls work for like 15 minutes and then just give a constant dial tone. |
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13:23.20 | *** mode/#asterisk [+o file] by ChanServ |
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13:25.34 | omenSP | Is there an example that I can base my outbound calling routes and trunks off of? I bought the 8-port FXO card and plugged all the trunks into it, then set it up as 8 trunks in the system. Upon restarting Asterisk, outside calls work for like 15 minutes and then just give a constant dial tone. |
13:26.59 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
13:29.03 | omenSP | What are DAHDI Channels? |
13:30.58 | WIMPy | I think you should start with the |
13:31.03 | WIMPy | ~book |
13:31.03 | 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 |
13:31.30 | omenSP | I'm using FreePBX though. |
13:32.22 | WIMPy | GUIs aren't supported here. Ask about that in #freepbx . |
13:32.28 | omenSP | Thank you |
13:32.30 | *** part/#asterisk omenSP (32490489@gateway/web/freenode/ip.50.73.4.137) |
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13:38.37 | marceloamorim | [TK]D-Fender: http://pastebin.com/rWrstmSg |
13:38.53 | marceloamorim | I restart the asterisk to get the full.log |
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13:49.36 | BlackDex | Hello there |
13:49.44 | BlackDex | we have an Digum TE133 PCI-E card |
13:50.19 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
13:50.22 | BlackDex | and when we configure the card with dahdi we always get 24 channels, so it seems that it uses T mode.. But i need E mode |
13:50.42 | BlackDex | i can't seem to get it to the E mode, don't know where to look |
13:51.24 | *** join/#asterisk yano (~yano@freenode/staff/yano) |
13:52.09 | hexanol | I've found something a little strange on 2 different asterisk while looking for a "no sound" problem |
13:52.16 | hexanol | one is a 11.9.0, the other a 11.11.0 |
13:52.35 | hexanol | the usage count of the "res_rtp_asterisk" module is high, 811 on one, 6 on the other |
13:52.47 | [TK]D-Fender | BlackDex: Got set the channels in your configs |
13:52.51 | hexanol | and there's lots of UDP socket (bind on a port in the RTP range) that are open |
13:53.01 | hexanol | but there's no active channels |
13:53.20 | hexanol | like, on the one with a 811 use count, there's around ~1600 open UDP socket in the RTP range |
13:53.33 | hexanol | and on the one with a 6 use count, there's 12 open UDP socket |
13:53.44 | hexanol | someone else has seen this kind of things on similar asterisk version ? |
13:55.00 | hexanol | I've not found yet on how to reproduce the problem; it's happening, but I don't know how |
13:58.15 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
13:59.38 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-yunnfimgllpmzmer) |
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14:28.00 | marceloamorim | I`ll test later about the problem I had with audio, I set canreinvite=yes on the sip.conf |
14:28.56 | BlackDex | [TK]D-Fender: I think i have found something.. Could be the some settings i need to change modprobe |
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15:11.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
15:12.22 | Katty | my state is embarassing. |
15:13.28 | [TK]D-Fender | Its a homonym.... |
15:15.33 | Katty | [TK]D-Fender: you're a homonym! |
15:16.08 | [TK]D-Fender | UR MOM ...? ;) |
15:16.21 | Katty | your face! |
15:17.08 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
15:17.51 | puzzled | hi |
15:17.59 | Katty | hey puzzled (= |
15:18.08 | puzzled | hi Katty |
15:18.12 | Katty | how're you dear? |
15:18.22 | puzzled | looking forward to the weekend :) |
15:18.27 | Katty | meeee too! |
15:18.30 | Katty | big plans? |
15:18.47 | puzzled | nope just relaxing |
15:19.01 | Katty | sometimes that's the best kind of weekend! |
15:23.44 | puzzled | Anyone have any experience with didlogic.com for DIDs? |
15:25.40 | *** join/#asterisk sgriepentrog (~sgriepent@107-220-72-223.lightspeed.brhmal.sbcglobal.net) |
15:28.09 | Nugget | puzzled: I tried to sign up for service with them last week for some australian DIDs but got thwarted because their online signup process was too manual |
15:29.01 | Nugget | they wanted me to scan the credit card and upload images of it and whatnot and that was enough of a deterrent for me to seek another provider |
15:29.41 | puzzled | Nugget: thanks, that sounds like 1995 wants their process back :) I'll pass then |
15:29.49 | Nugget | oh, and my ID and a copy of a utility bill |
15:30.00 | Nugget | I decided they didn't really want business customers |
15:32.10 | puzzled | that sounds insane |
15:32.33 | file | Nugget, well you ARE a troublemaker... |
15:32.57 | puzzled | oh and they are located in Hong Kong and I doubt my consumer rights I enjoy in the EU applies there |
15:40.19 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-93-37.user.veloxzone.com.br) |
15:40.24 | anonymouz666 | I am back on! |
15:41.16 | anonymouz666 | sends a complete /var/log/asterisk/full to [TK]D-Fender and also a sip debug. |
15:41.23 | *** join/#asterisk af_ (~af@static-82-85-142-163.clienti.tiscali.it) |
15:43.05 | anonymouz666 | the new SIP channel driver is something that I'd love to test. |
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16:06.05 | bibz2 | any freelancers in here? |
16:06.52 | [TK]D-Fender | Told you earlier.. lances are expensive... don't expect them to be free... |
16:07.18 | [TK]D-Fender | Perhaps you should start with what you are actually trying to accomplish before people start lining up over things they can't help you with. |
16:09.00 | bibz2 | I still have calls which don't go through.. and I don't get it what the problem could be |
16:09.20 | bibz2 | the thing about is, that I'm trying to mantain/repair a pbx which is already 2 years old and quite big |
16:09.37 | bibz2 | I resolved all the NAT Problems I guess so |
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16:11.48 | [TK]D-Fender | Show your failed calls. |
16:11.53 | [TK]D-Fender | "guessing" is bad... |
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16:25.11 | Katty | [TK]D-Fender: here's my pastebin of a failed call. What did I do wrong???? http://pastebin.com/AjGXF7yV |
16:25.23 | *** join/#asterisk dundel (~daniel@190.98.82.213) |
16:25.46 | [TK]D-Fender | Well you finally broke double-digits with that one.... |
16:26.00 | [TK]D-Fender | You keep trying to get a laugh out of these, but no pun in ten did ;) |
16:26.15 | Katty | rolls eyes |
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16:39.08 | zamba | i'm seeing lots of these messages in my system logs: Aug 15 17:26:23 tahiti asterisk[30498]: NOTICE[30522]: chan_sip.c:22546 in handle_request_invite: Sending fake auth rejection for device 100<sip:100@ip>;tag=6d247154 |
16:39.17 | zamba | what do they mean? can they be ignored/removed? |
16:41.45 | [TK]D-Fender | it means they are getting chanllenged,. |
16:41.52 | [TK]D-Fender | Doesn't mean they don' get accepted |
16:41.55 | [TK]D-Fender | Go look for that |
16:42.47 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-56-241.om.om.cox.net) |
16:42.58 | zamba | is there a way to get rid of these messages? |
16:43.35 | [TK]D-Fender | Disable all "notice" in your logging |
16:47.33 | stevePearPear | Hi, whenever i reset Asterisk, it will automatically attempt to register all my peer with the siptrunk provider automatically |
16:47.45 | stevePearPear | can I only register when a call is attempted? |
16:48.09 | [TK]D-Fender | no |
16:49.13 | stevePearPear | [TK]D-Fender are you replying to me for the ânoâ ? |
16:50.38 | [TK]D-Fender | I am |
16:51.55 | stevePearPear | I am allowing my users to update the credentials for the siptrunk, so when too many of my users has invalid credential, the siptrunk provider block me as it thought that I am trying to brute force |
16:51.59 | stevePearPear | is there a way out of this? |
17:14.32 | anonymouz666 | surface 3 is such a good product |
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17:22.47 | bibz2 | my asterisk fills the cdr table with double entries |
17:22.52 | bibz2 | why could that be |
17:23.07 | bibz2 | for every call there are 2 entries in the cdr table |
17:24.23 | [TK]D-Fender | exactly identical? |
17:25.12 | pabelanger | anybody using sipml5? Specifically, are you getting native (webrtc client side) ringback on dial? |
17:25.24 | pabelanger | I know asterisk and generate it, but want to see if the client side works |
17:25.30 | bibz2 | yes exactly identical |
17:25.42 | [TK]D-Fender | bibz2: probably a double-entry in CDR configs |
17:26.09 | bibz2 | even the uniqueid is identical |
17:26.09 | bibz2 | ok |
17:34.41 | navaismo | pabelanger: you mean hear the ring tone when is dialing? |
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17:38.05 | navaismo | I use https://github.com/navaismo/Elastix-Agent_Console_WebRTC/blob/master/modules/themes/default/js/ml5.js#L331 and https://github.com/navaismo/Elastix-Agent_Console_WebRTC/blob/master/modules/themes/default/js/ml5.js#L152 |
17:38.15 | navaismo | based on the doubango's example |
17:39.48 | bibz2 | no double entry in the cdr configs. checked cdr_odbc also.. |
17:39.51 | bibz2 | what the heck |
17:44.27 | [TK]D-Fender | check your backend status |
17:45.05 | mjordan | bibz2: which version of Asterisk are you using? |
17:47.22 | pabelanger | navaismo, yes, from sipml5 dial a number. Do you hear local ringback? |
17:48.15 | bibz2 | mjordan: 11.6-cert2 |
17:48.35 | navaismo | pabelanger: Yes, but i need to call the function and the audiofile must exist |
17:48.38 | bibz2 | the problem occured yesterday, but I don't know why. I have made some alternations but didn't change anything in the CDR config |
17:48.42 | mjordan | k. You probably do have a configuration issue somewhere then. What all CDR backends do you have loaded? |
17:48.44 | WIMPy | Are IAX timestamps somehow related to realtime? |
17:49.21 | anonymouz666 | pabelanger: are you using ARI heavily? |
17:49.41 | bibz2 | mjordan: http://pastebin.com/tS9akCKg |
17:51.23 | mjordan | you have both odbc and adaptive odbc loaded. |
17:51.29 | mjordan | Just sayin. |
17:51.34 | mjordan | You may want to double check that. |
17:52.43 | doobeh | I know credit card machines have issues with the compression algo's-- if I have 8 FXO ports and 16 FXS ports on my channel bank-- will I still run into compression problems, since it's not going up to a sip trunk? |
17:52.55 | bibz2 | how can I unload the adaptive one? or better: which one should I unload |
17:53.15 | pabelanger | navaismo, perfect, that is the issue. We are not invoking that function |
17:53.17 | pabelanger | changing it now |
17:53.30 | pabelanger | anonymouz666, Well, we are using it. Mostly bridges an playback |
17:54.07 | WIMPy | doobeh: They don't only have issues with reducing codecs they also have issues with jitter. But as long as there's no IP involved, everything should wrok fine. |
17:56.13 | pabelanger | navaismo, thanks again. Owe you a beer |
17:56.37 | bibz2 | funny thing: the cdr_adaptive_odbc is empty |
17:57.35 | doobeh | WIMPy: thanks, I was confused when the compression started, so if I was just FXS#1 --> FXO#4 it doesn't bother, but if I FXS#1 --> WAN it'll kick in? |
17:58.12 | navaismo | no problem |
17:59.03 | WIMPy | doobeh: Exactely. |
18:01.29 | anonymouz666 | navaismo: I'll test some of your software specifically spy agents |
18:02.00 | navaismo | good, so far like nobody test that LOL |
18:02.21 | anonymouz666 | not even the author? heh... |
18:03.48 | navaismo | I do, but that doesnt count |
18:04.12 | navaismo | there are some videos about |
18:04.20 | navaismo | in my youtube account |
18:04.36 | anonymouz666 | yeah I am watch the plain asterisk |
18:04.38 | anonymouz666 | I don't use the elastix |
18:05.02 | doobeh | WIMPy: thanks :) |
18:07.16 | bibz2 | I don't get it why I get doubled CDR entries |
18:08.51 | anonymouz666 | navaismo: I understood the native video language!!! |
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18:10.07 | bibz2 | I'm also getting the warning, that the application delimiter is now the comma.. and the question if I did forget to convert my dialplan |
18:11.24 | navaismo | anonymouz666: super! |
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18:22.44 | usnavi | Hi. Trying to send email messages from *. I set serveremail = vm@mydomain.com but postfix is showing the FROM line as asteriskuser@machinename.machinedomain.com |
18:22.56 | usnavi | Postfix doesn't like this and spits it out ofcourse. |
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18:29.37 | [TK]D-Fender | always used to read that as "Severe Mail" and though the tone was a bit harsh... |
18:38.10 | bibz2 | Currect auth, but based on stale nonce received from XXXX |
18:38.15 | bibz2 | what does this mean? |
18:38.35 | bibz2 | getting a 401 unauthorized answer |
18:42.33 | usnavi | So how does asterisk know to send "from" a certain address? |
18:43.55 | usnavi | does general's "serveremail" apply to [default] entries ? |
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18:57.26 | usnavi | So serveremail = vm@domain.com, got that set in [general] of /etc/asterisk/voicemail.conf .... what else do I need to do here? |
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18:59.03 | usnavi | Aug 15 11:53:33 SNCA00-PH00 postfix/pickup[29173]: 196021422CA: uid=500 from=<asteriskpbx> |
18:59.03 | usnavi | <PROTECTED> |
18:59.21 | usnavi | Does the pickup mean I can't set the from address? |
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19:15.05 | Katty | a dial tone the line may be quite continental, but sip channels are a girl's best friend! |
19:19.25 | file | Katty, here we are in the center of the first world, it's laid out before us, who are we to break down? |
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19:20.32 | Katty | ^_^ |
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21:46.21 | usnavi2 | I'm looking for a refresher but fairly complete tutorial/documentation on the asterisk dialplan syntax and idioms |
21:46.45 | usnavi2 | https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Dialplan would be workable? |
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21:51.51 | ipengineer | Is there an AMI action that will return active queue calls? QueueStatus does not show calls in progress for the given queue |
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22:37.27 | [TK]D-Fender | usnavi2: > |
22:37.31 | [TK]D-Fender | ~book |
22:37.31 | 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 |
22:37.35 | [TK]D-Fender | ~asteriskwiki |
22:37.35 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
22:37.39 | [TK]D-Fender | ^^^ |
22:37.49 | usnavi2 | cool. |
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