| 00:08.40 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 01:17.41 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 01:17.45 | jzaw | evenin all |
| 01:18.06 | WIMPy | good morning |
| 01:18.48 | jzaw | say when you set messages => security,notice,warning,error |
| 01:18.49 | jzaw | <PROTECTED> |
| 01:18.55 | jzaw | and similar for console |
| 01:19.13 | jzaw | is there a way to only have *security* show the failed and warning items |
| 01:19.19 | jzaw | not the successful stuff |
| 01:19.42 | jzaw | obivously the error/warning/fails will be hidden in all that *noise* |
| 01:20.12 | jzaw | the messages setting is for fail2ban ofc |
| 01:20.24 | jzaw | but i also always keep a console up on screen |
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| 01:37.33 | *** join/#asterisk KavanS (~MIRC@LINBIT/KavanS) |
| 01:38.23 | KavanS | looking to connect a polycom to a POTS landline in the most simple way possible...suggestions? |
| 01:39.13 | [TK]D-Fender | SOLDER! |
| 01:39.26 | [TK]D-Fender | No wait.. that requires heat... |
| 01:39.34 | [TK]D-Fender | GLUE! |
| 01:40.36 | KavanS | lol |
| 01:41.18 | KavanS | was thinking one of those linksys adapters might fit the bill...I've just never used one before |
| 01:42.15 | KavanS | SPA3000 |
| 01:42.54 | [TK]D-Fender | very old model |
| 01:43.46 | KavanS | hmm |
| 01:43.53 | KavanS | googling now |
| 01:44.24 | KavanS | any suggestions for a newer model? |
| 01:45.13 | [TK]D-Fender | There was the 3102, and I'm not sure what the current one is.... |
| 01:45.45 | KavanS | yeah the Cisco 3102 is the result that I'm finding on repeat |
| 02:01.05 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 02:01.12 | KavanS | [TK]D-Fender: ever use one of these U4EA Fusion 200 devices? |
| 02:02.25 | [TK]D-Fender | nope |
| 02:03.00 | *** join/#asterisk hardwire (~hardwire@222-83-237-24.gci.net) |
| 02:03.40 | [TK]D-Fender | Google search shows them pretty much exclusively plastered all over ebay..... which means cheap junk |
| 02:04.26 | KavanS | yep, couldn't find a review |
| 02:04.31 | KavanS | independently speaking... |
| 02:04.41 | [TK]D-Fender | And is that where you found them? |
| 02:04.58 | KavanS | googled a variety of search terms |
| 02:05.08 | KavanS | fxo voip gateway, voip gateway, pots voip gateway, etc. |
| 02:05.19 | [TK]D-Fender | better tip : see what RETAILERS are actually setting |
| 02:05.29 | KavanS | true that. |
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| 02:21.14 | KavanS | ok, looks like I found something. thanks for the pointers TK |
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| 02:22.38 | [TK]D-Fender | KavanS: NWhat did you pick? |
| 02:25.32 | KavanS | going to use an analog phone in this circumstance :) |
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| 03:51.00 | hebber | exit |
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| 10:09.45 | Groenleer | Anyone here experience with the combination of Asterisk on a Synology NAS and Follow me ? |
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| 11:39.21 | jaflong | any ideas what is wrong: ERROR[17656]: pjsip:0 <?>: icess0x7ff3a00 ..Error sending STUN request: Invalid argument |
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| 12:17.38 | teo_ | jaflong: no server configured? |
| 12:18.07 | teo_ | btw there's something in the buglist, STUN related |
| 12:22.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:24.11 | jaflong | <teo_> : stun server is running. I get the same result against a external stun server as well |
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| 12:26.50 | teo_ | jaflong: https://issues.asterisk.org/jira/browse/ASTERISK-22911 |
| 12:27.28 | teo_ | jaflong: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2010-December/012236.html |
| 12:28.19 | teo_ | jaflong: https://groups.google.com/forum/#!topic/jssip/1-08lrr8ar4 |
| 12:29.35 | jaflong | Thanks for the info, I am checking the links out. However I am wondering does STUN actually work in Asterisk 12 |
| 12:30.45 | teo_ | well, AFAIK they've replaced SIP stack with pjproject and STUN code needs debug so that can be a problem |
| 12:31.03 | file | a new channel driver exists which uses PJSIP, but chan_sip still exists |
| 12:31.08 | teo_ | I believe it worked fine before |
| 12:31.15 | file | the underlying code is common between both of them |
| 12:31.36 | file | and it's working for some people, or at leas twas |
| 12:33.36 | jaflong | I see the stun features are from chan_pjsip. Does chan_pjsip and chan_sip work together or is one to be used exclusivly |
| 12:34.34 | file | they aren't from chan_pjsip |
| 12:35.09 | file | PJSIP is a part of what is provided, but is standalone from the ICE/STUN/TURN support |
| 12:35.20 | jaflong | ok |
| 12:35.27 | file | and they are completely separate |
| 12:37.25 | file | the problem is likely environment specific |
| 12:38.55 | jaflong | in terms of config to use stun i have icesupport=yes in sip.conf and stunaddr=stun.stunprotocol.org in rtp.conf |
| 12:39.10 | jaflong | Are any other config needed |
| 12:40.37 | file | no |
| 12:42.52 | teo_ | jaflong: have you specified the transport? |
| 12:43.40 | jaflong | i dont think so. Where is that set? |
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| 13:02.57 | Groenleer | does anyone know why FollowMe would not work with Asterisk and AsteriskGUI ? |
| 13:03.20 | [TK]D-Fender | AsteriskGUI is dead and has not been supported in a long long time |
| 13:03.21 | Groenleer | It seems my extension is not triggering the macro for FollowMe, altough everything seems to be set. |
| 13:03.52 | Groenleer | I figured that out, but it seems to be the only Asterisk implementation currently available for Synology NAS. |
| 13:05.07 | Groenleer | And the package is not really updated by Synology either cause it is still having Asterisk 1.8... |
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| 13:15.57 | teo_ | jaflong: I think you need to specify everything including STUN server port and sip.conf shall contain binding instructions, including port value as well as mentioned here http://serverfault.com/questions/466049/not-able-to-register-via-sipclient-on-asterisk-server |
| 13:16.52 | teo_ | in particular, udpbindaddr=0.0.0.0:5060 and stunaddr=stun.l.google.com:19302 must be set |
| 13:24.00 | *** join/#asterisk hellc2 (~elio@100.Red-88-26-250.staticIP.rima-tde.net) |
| 13:24.12 | hellc2 | Hi all |
| 13:24.32 | hellc2 | !clear |
| 13:24.47 | Chainsaw | You resetting us all? |
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| 13:53.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:01.11 | jwr__ | I read that res_jabber was deprecated in favor of res_xmpp. Does anyone know if res_xmpp does jabber IM notifications? Everything I see on Google shows people using res_jabber for that. |
| 14:01.46 | Katty | fyi, google is ending support from xmpp. |
| 14:01.53 | Katty | s/from/for/ |
| 14:02.01 | Katty | infobot: thank you, dear. |
| 14:02.01 | infobot | de nada, Katty |
| 14:02.43 | jwr__ | they are ending support for xmpp for google voice. but not for IM, if i recall correctly. i'm only interested in sending an IM to someone when they receive a call. |
| 14:02.44 | file | res_xmpp exposes the same stuff as res_jabber, it's just internally different |
| 14:02.55 | file | heck it even accepts the same config file |
| 14:03.29 | Katty | i think i'd use res_jabber. |
| 14:03.36 | Katty | but that's just me. |
| 14:04.09 | Katty | file: how're you and ms zoe. |
| 14:04.20 | file | Katty, good! preparing for snow |
| 14:04.34 | Katty | file: oh? not a lot i hope? |
| 14:04.36 | file | Katty, and she's presently just dead weight in my lap |
| 14:04.47 | file | Katty, 30-40cm tomorrow and potentially another 30-40cm Sunday |
| 14:04.49 | Katty | excellent. i see she's doing her job. |
| 14:04.53 | jwr__ | res_xmpp uses the same config as res_jabber? so any directions that talk about using res_jabber should also work for res_xmpp, then? |
| 14:04.59 | Katty | file: yuck! keep it up there. |
| 14:05.04 | file | jwr__, should! |
| 14:05.18 | Katty | file: i trust you have french toast supplies? bread, milk, and eggs |
| 14:05.20 | jwr__ | file: awesome. that is what i need. thanks. |
| 14:05.56 | file | Katty, I do not! |
| 14:06.32 | *** join/#asterisk kW_ (~user@wlan249021.rz.uni-leipzig.de) |
| 14:06.41 | Katty | file: oh noes! |
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| 14:12.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:17.03 | *** join/#asterisk BeachBall (~eXcAliBuR@206.162.174.6) |
| 14:17.05 | BeachBall | http://pastebin.com/zyX2iUCm |
| 14:17.30 | BeachBall | i get that while trying to do a install |
| 14:17.54 | BeachBall | make doesn't finish |
| 14:18.02 | BeachBall | and make install doesn't work for make didn't finish |
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| 14:21.57 | Qwell | BeachBall: How did you download this? |
| 14:22.43 | Qwell | Also, give us the complete log of: make dist-clean && ./configure && make |
| 14:23.02 | BeachBall | wget using current |
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| 14:29.25 | BeachBall | takin for ever |
| 14:29.27 | BeachBall | might work |
| 14:29.28 | BeachBall | yay |
| 14:29.30 | BeachBall | it worked |
| 14:31.21 | Qwell | might want to check the disk space on that system, and make sure the time is correct. |
| 14:31.32 | BeachBall | yikes |
| 14:31.54 | pabelanger | or out of memory |
| 14:32.03 | BeachBall | it's a droplet |
| 14:32.07 | BeachBall | from digital ocean |
| 14:32.11 | BeachBall | the 512MB one |
| 14:32.14 | BeachBall | 20GB space |
| 14:32.15 | BeachBall | :{ |
| 14:32.25 | pabelanger | do you have a swap? |
| 14:32.27 | Qwell | I'm not saying that's the problem - it's just a few potential reasons. |
| 14:32.28 | pabelanger | swapon |
| 14:32.37 | BeachBall | i don't think it's got swap |
| 14:32.47 | pabelanger | I suspect gcc is getting killed |
| 14:33.10 | pabelanger | setup a 2gb swap |
| 14:33.37 | Qwell | Or get a properly sized machine, so you don't utterly tank performance... |
| 14:33.52 | BeachBall | it's only for 1 line |
| 14:33.55 | BeachBall | 1 call at a time |
| 14:33.57 | BeachBall | :} |
| 14:34.08 | BeachBall | plus my email and website |
| 14:35.01 | BeachBall | 16% used space |
| 14:35.06 | BeachBall | do i really need a swap |
| 14:35.11 | pabelanger | yes |
| 14:36.13 | BeachBall | how big? |
| 14:36.16 | BeachBall | oh |
| 14:36.17 | BeachBall | i c |
| 14:36.19 | BeachBall | 2 GB |
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| 14:39.52 | pabelanger | after you compile you can swapoff |
| 14:40.07 | pabelanger | but, if you run out of memory (512) you have no overflow |
| 14:40.13 | pabelanger | and things will just get killed |
| 14:45.02 | BeachBall | i made it |
| 14:45.06 | BeachBall | ;D |
| 14:45.37 | BeachBall | thanks |
| 14:45.41 | BeachBall | you guys are so smart |
| 14:46.49 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:52.16 | BeachBall | i copied all my config files from old server to new one |
| 14:52.22 | BeachBall | that won't cause problems? |
| 14:52.23 | BeachBall | right? |
| 14:52.34 | BeachBall | [Mar 25 10:52:25] ERROR[23333]: chan_sip.c:4343 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 14:52.44 | BeachBall | i updated the IP address in the sip.conf |
| 14:53.01 | BeachBall | registration is timing out |
| 14:53.21 | [TK]D-Fender | Time to actually look at the call... |
| 14:53.27 | [TK]D-Fender | (txn) |
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| 14:55.47 | [TK]D-Fender | [10:52]BeachBallthat won't cause problems? <- we know nothing of your previous system, module differences, networking difference, etc. |
| 14:56.02 | BeachBall | how do i get the debugger on |
| 14:56.08 | BeachBall | so i see everything |
| 14:56.51 | [TK]D-Fender | ... |
| 14:56.55 | [TK]D-Fender | "core set verbose 10" |
| 14:57.01 | [TK]D-Fender | "sip set debug on" |
| 14:57.06 | BeachBall | i have verbose at 10 |
| 15:04.52 | BeachBall | i had to disable the bing IP lines |
| 15:05.33 | BeachBall | thank you for keeping me calm while i paniced |
| 15:05.35 | BeachBall | :{ |
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| 15:45.35 | zapphir | Hi! I cannot get device_status for my x-lite software phone to work in asterisk. The device_status is always not_in_use no matter what I try. I am running Asterisk 11.5.1. Does anyone have any tips how to get device_status to work? |
| 15:49.29 | [TK]D-Fender | Show us your peer and your dialplan you're runing for it |
| 15:49.40 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:50.12 | WIMPy | Do you have call-counters enabled? |
| 15:50.15 | [TK]D-Fender | ~pb |
| 15:50.15 | 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. |
| 15:50.16 | [TK]D-Fender | ^^^ |
| 15:51.28 | zapphir | Wimpy: yes I have |
| 15:53.39 | WIMPy | counteronpeer? |
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| 16:01.09 | zapphir | WIMPy: I have tried that too |
| 16:02.43 | zapphir | Wimpy: would it be possible to set the device_state manually? I could use a custom variable, but it would be nice to use the built in functions for queue handling with regards to the device_state. |
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| 17:18.37 | FuriousGeorge | hey all |
| 17:19.00 | file | moo |
| 17:20.36 | FuriousGeorge | im having a heck of a time trying to provision my soundstation 500 with asterisk. im following this guide: http://community.polycom.com/t5/VoIP/FAQ-Can-I-register-my-Polycom-Phone-with-a-XYZ-SIP-Server/td-p/4223 |
| 17:21.13 | Chainsaw | FuriousGeorge: It's working talking to jkroon about this later. |
| 17:23.30 | [TK]D-Fender | FuriousGeorge: Model doesn't seem right... |
| 17:23.51 | FuriousGeorge | Chainsaw: I don't understand |
| 17:24.06 | FuriousGeorge | [TK]D-Fender: i meant to say "5000" not 500 |
| 17:24.54 | Chainsaw | [TK]D-Fender: Those board room table units, the older ones. |
| 17:25.36 | FuriousGeorge | Chainsaw: they are working and you will be talking to jkroom about it? i'm a bit confused |
| 17:25.48 | Chainsaw | FuriousGeorge: No, I recommend that you, FuriousGeorge, talk to jkroon when he next appears. |
| 17:25.59 | Chainsaw | FuriousGeorge: He's very recently done work to deploy them. |
| 17:26.15 | [TK]D-Fender | Chainsaw: IP 5000 isn't raelly "old" |
| 17:26.17 | FuriousGeorge | Chainsaw: i understand now, thanks |
| 17:26.25 | Chainsaw | [TK]D-Fender: It's not fantastically new either. |
| 17:26.35 | *** join/#asterisk jploh (~textual@121.54.44.94) |
| 17:27.10 | Chainsaw | FuriousGeorge: I autoprovision my units, at any rate. Over HTTP (using a DHCP option string). |
| 17:30.09 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:31.11 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 17:46.30 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 17:49.54 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 17:50.15 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 17:56.35 | marceloamorim | Hey guys, I used to set on my freebsd-asterisk, the modules dahdi on rc.conf > dahdi_modules="wtcdm24xxp" because I use digium wildcard tdm800p |
| 17:56.52 | marceloamorim | now I need to use tdm400p, which module I need to set up? |
| 17:56.57 | marceloamorim | anyone knows? |
| 17:57.45 | [TK]D-Fender | wctdm |
| 17:58.42 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 17:59.08 | marceloamorim | there is wctdm.ko and wct4xxp.ko at /usr/local/lib/dahdi/ |
| 17:59.52 | marceloamorim | ok, thx one more time [TK]D-Fender |
| 18:02.55 | marceloamorim | to complement http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation there is a good documentation about this digium wildcards |
| 18:04.17 | [TK]D-Fender | That page is ancient |
| 18:04.22 | [TK]D-Fender | Zaptel hasn't existed for years |
| 18:04.36 | [TK]D-Fender | It was replaced by DAHDI in 1.4.20 or 22 |
| 18:09.12 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:09.20 | marceloamorim | you right, but still good for which module you can use for some wildcards |
| 18:12.56 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:21.17 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:22.27 | *** join/#asterisk path (~lsanmarti@lsanmartin.uchile.cl) |
| 18:22.33 | path | hey boys |
| 18:22.37 | pabelanger | Anybody know the syntax in sox to setup a sln audio file (raw asterisk format)? |
| 18:23.01 | path | is there a way to edit the SIP header? I dont want the ';tag=' on it |
| 18:23.18 | path | went through chan_sip.c without any success |
| 18:23.35 | path | p->tag, p->theirtag, clid_tag, and so on |
| 18:23.39 | file | uhhhh, that would likely break SIP interop |
| 18:24.25 | path | how come |
| 18:24.46 | path | From: sipp <sip:sipp@127.0.1.1:5061>;tag=12834SIPpTag00110 I just want to remove the ';tag...' |
| 18:25.29 | file | perhaps a better question is this - where are you using the From and why? |
| 18:26.28 | path | the thing is I give this header to an sbc |
| 18:43.35 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 18:45.15 | *** join/#asterisk morfin (~morfin@smtp.city-call.ru) |
| 18:46.42 | *** join/#asterisk startledmarmot (~startledm@cpe-76-172-72-246.socal.res.rr.com) |
| 18:50.30 | *** join/#asterisk R1ppa (~R1ppa@mail.hrsg.ca) |
| 18:51.37 | R1ppa | trying to override callerid so that when calling SALES, at least the sales folks will see that a certain extension was dialed or even changing from to SALES would be nice, tried callerid(name) and callerid(num), error is function callerid not registered |
| 18:51.58 | R1ppa | doing show function callerid shows information, dont want to assume but thought that meant it is indeed registered |
| 18:53.32 | [TK]D-Fender | Show us |
| 18:53.36 | [TK]D-Fender | ~pb |
| 18:53.37 | 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. |
| 18:53.39 | [TK]D-Fender | ^^^ |
| 18:53.43 | R1ppa | k |
| 18:54.43 | R1ppa | [TK]D-Fender, http://pastebin.ca/2679180 |
| 18:54.53 | R1ppa | snippet of extensions.conf, or did you mean the error? |
| 18:55.25 | [TK]D-Fender | R1ppa: Functions are CASE SENSITIVE. |
| 18:55.46 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-kgkadzlsyfgcohrw) |
| 18:55.46 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:56.09 | R1ppa | [TK]D-Fender, so CALLERID(name) ? |
| 18:56.13 | [TK]D-Fender | yes |
| 18:57.08 | *** join/#asterisk loggiew (~logan@198.52.217.61) |
| 18:57.49 | loggiew | i have a server on a public IP (no nat), a DID, a softphone, and my cell with a voip app |
| 18:57.53 | loggiew | all of them get an echo test |
| 18:57.59 | loggiew | none of them will communicate with each other |
| 18:58.13 | loggiew | and im getting a little frustrated. Not sure where to even look at this point |
| 18:59.00 | loggiew | its almost like the device behind my home NAT is having port issues but Ive been chasing NAT for a while now and it does it over 4g also |
| 18:59.26 | R1ppa | [TK]D-Fender, thank you sir!! |
| 18:59.36 | loggiew | so i dont think its my home NAT per se, I think I have a misconfig. Im just not sure where. |
| 18:59.43 | loggiew | They all perform the echo test fine. |
| 19:00.00 | [TK]D-Fender | loggiew: You need to set each of your peers to match their scenario. |
| 19:00.04 | loggiew | the softphone is sipml5 |
| 19:00.14 | [TK]D-Fender | loggiew: and prevent reinvites (directmedia=no) |
| 19:00.19 | [TK]D-Fender | R1ppa: You're welcome. |
| 19:00.22 | loggiew | [TK]D-Fender: ya :/ Im fairly sure I did |
| 19:00.28 | loggiew | [TK]D-Fender: definitely have directmedia=no |
| 19:00.57 | [TK]D-Fender | R1ppa: and I'd advise you to try to conserve at least a bit of the orignal name rather than ovewriting all of it like that. |
| 19:00.58 | loggiew | and the DID config is correct (its cut and paste what I used previously when I had it working) |
| 19:01.21 | [TK]D-Fender | loggiew: Telling us it's right isn't confirmation that it is so. |
| 19:01.38 | loggiew | ok, would you prefer I pastebin or another method? |
| 19:01.46 | [TK]D-Fender | [14:58]loggiewnone of them will communicate with each other <- this could use a little clarification |
| 19:01.50 | [TK]D-Fender | ~pb |
| 19:01.50 | 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. |
| 19:01.51 | [TK]D-Fender | ^^^ |
| 19:03.44 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 19:04.07 | loggiew | I am using the sample configs so Im trying to paste the relevent sections and leave out the excess |
| 19:05.02 | [TK]D-Fender | Trash the excess. It will only lead to confusion and then errors when you miss enabling something you thought you had, or the reverse |
| 19:11.15 | loggiew | I know its probably ugly and I tried to make sure I included anything having to do with my changes |
| 19:11.18 | loggiew | http://pastebin.com/XghdxJ7L |
| 19:11.29 | loggiew | I also focused on one device |
| 19:11.40 | loggiew | and included the data for the SIP trunk |
| 19:11.50 | loggiew | because working between those two would be ideal |
| 19:12.16 | loggiew | It rings on phone A but phone B never actually rings to be answered |
| 19:13.56 | loggiew | I always get messages in sip debug saying "Retransmitting" etc |
| 19:14.10 | [TK]D-Fender | enable SIP debug, pastbein a call. |
| 19:14.20 | loggiew | ok, one moment |
| 19:14.30 | [TK]D-Fender | and you should be setting your peers explicitly and not trying to leave things to [general]. |
| 19:15.47 | loggiew | are the sip trunk and [100] not explicitely set? |
| 19:16.06 | [TK]D-Fender | your ITSP should be "nat=no", and your phone "nat=yes", and allow=all is a BAD idea. * could permit the first leg to something that a bridge will fail to transcode |
| 19:16.50 | loggiew | instead of allow=all is there a relatively safe standard I can try to start with? |
| 19:17.22 | [TK]D-Fender | pick your codec. disallow=all, allow=ONECODEC |
| 19:17.37 | [TK]D-Fender | Settle on the easiest lowest common denomicator |
| 19:17.52 | loggiew | ya, i dont know much about codecs so I wasnt sure |
| 19:18.14 | loggiew | let me start with those changes and then I'll list a pastebin of the debug if it still misbehaves. Thanks for the response |
| 19:19.06 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 19:21.21 | loggiew | omg dude |
| 19:21.27 | loggiew | you just got me further than a day of reading |
| 19:21.55 | loggiew | the phone finally rang but then immediately hung up so Ive gotta go figure that out now |
| 19:24.01 | [TK]D-Fender | Right questions + pastebin = opportunity |
| 19:24.12 | loggiew | right? |
| 19:24.16 | loggiew | I really appreciate it |
| 19:25.56 | [TK]D-Fender | pPB up a call and lets see what's happening... |
| 19:26.44 | loggiew | hm? |
| 19:27.00 | loggiew | oh |
| 19:27.01 | loggiew | pb |
| 19:27.24 | loggiew | is there a way to save the output to a file? |
| 19:28.10 | [TK]D-Fender | You could pipe it I guess... I just grab live. |
| 19:32.24 | loggiew | ya, im using `script` |
| 19:32.39 | loggiew | but now i have to copy it to a location i can easily highlite and paste :P |
| 19:32.42 | loggiew | virtual servers.. |
| 19:32.56 | [TK]D-Fender | Are you SSH'd to the server? |
| 19:33.08 | [TK]D-Fender | If so, from what platform? |
| 19:33.28 | loggiew | Im on windows and it is centos |
| 19:33.39 | loggiew | im using my brothers laptop. i normally run anything else |
| 19:35.24 | [TK]D-Fender | Use Putty. Right click. Copy All To buffer. Done |
| 19:36.15 | loggiew | http://pastebin.com/pJuwGgkP |
| 19:36.42 | loggiew | im in putty.. i dunno if it saves to the buffer properly like that when Im in screen |
| 19:37.04 | loggiew | i started apache and moved it to a public location and pulled it up |
| 19:38.20 | rrittgarn | loggiew: you can always turn on putty logging... i think that should capture all the output |
| 19:38.25 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 19:38.56 | loggiew | cool. Ill keep that in mind |
| 19:38.58 | loggiew | thanks |
| 19:39.46 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 19:40.11 | loggiew | those debug logs are really long |
| 19:40.15 | loggiew | tedious to read through |
| 19:40.26 | Kobaz | okay so, something new i'm getting into now |
| 19:40.39 | Kobaz | what does jitter or voip packet loss look like in a packet dump |
| 19:40.48 | Kobaz | i'm trying to prove to a customer that it's not the asterisk box that's the problem |
| 19:40.58 | Kobaz | the audio received is choppy on some calls |
| 19:41.09 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:42.24 | loggiew | [TK]D-Fender: was that PB of any benefit? |
| 19:44.44 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 19:46.16 | loggiew | Kobaz: Im guessing, obvious retransmits and latency on the connection? |
| 19:46.33 | *** join/#asterisk serafie (~erin@2601:2:8800:373:f5ce:2a66:9da:7b47) |
| 19:46.58 | Kobaz | well there's no retransmits on our sending-side |
| 19:48.07 | Kobaz | i'm looking for anything interesting on the packet dump at the moment |
| 19:49.00 | loggiew | it seems you would need a dump of both sides? Am I wrong? |
| 19:49.01 | [TK]D-Fender | [15:36]loggiewim in putty.. i dunno if it saves to the buffer properly like that when Im in screen <- it doesn't |
| 19:49.26 | loggiew | :) |
| 19:49.55 | loggiew | its time for a cigarette |
| 19:50.17 | Kobaz | i dont have access to the other side |
| 19:50.19 | Kobaz | another tech does |
| 19:50.27 | Kobaz | and he's not seeing anything |
| 19:51.00 | Kobaz | i'm thinking it's something on the carrier... like we're all sip from asterisk to an avaya (on the same switch), and then out to an ip box from lightpath |
| 19:53.04 | loggiew | but you aren't really noting TCP retransmissions? Nor SIP resending? |
| 19:53.30 | Kobaz | i have a tcpdump of everything |
| 19:53.40 | loggiew | ya. sorry. im just sitting here going hmm. |
| 19:54.14 | Kobaz | nothing obvious so far |
| 19:54.21 | loggiew | im wondering if it would be dropped packets? Don't notice anything except 1 less packet here or there? |
| 19:54.31 | loggiew | rather somethign that is there, its something thats not? |
| 19:55.44 | loggiew | should be able to use MTR and such to check latency and packet loss along the path |
| 19:59.44 | *** join/#asterisk dxd828_ (~dxd828@dsl-dynamic-77-44-45-38.interdsl.co.uk) |
| 20:01.50 | loggiew | why does it send CANCEL immediately when I pick up? |
| 20:02.28 | Kobaz | loggiew, i'm listening to call recording |
| 20:02.30 | Kobaz | and it's like |
| 20:02.49 | Kobaz | h-e-l-l-o t-h-e-r-e |
| 20:02.54 | Kobaz | and then resumes back to normal |
| 20:03.02 | loggiew | wild |
| 20:03.03 | Kobaz | like tiny little 10ms audio chunks are missing |
| 20:03.11 | Kobaz | and then it doesn't happen for like 20 minutes |
| 20:03.13 | Kobaz | and then happens again |
| 20:03.37 | loggiew | does it seem to go in regular intervals? And do other calls or other network services notice it? |
| 20:03.59 | Kobaz | doesnt seem to be regular |
| 20:04.05 | Kobaz | here's a sample real quick |
| 20:04.24 | Kobaz | www.kobaz.net/misc/sample.wav |
| 20:04.40 | Kobaz | so someone is like |
| 20:04.54 | Kobaz | "h--iii, thi---s is john...." |
| 20:04.55 | loggiew | hmm |
| 20:04.56 | Kobaz | at the beginning |
| 20:05.06 | Kobaz | and then it's back to normal |
| 20:05.17 | loggiew | does it always do it at the start of the call? |
| 20:05.21 | Kobaz | not always |
| 20:05.26 | loggiew | no, you said it does 20 min later too.. |
| 20:05.27 | loggiew | ya |
| 20:05.28 | loggiew | hm |
| 20:05.39 | Kobaz | some call starts are perfectly fine |
| 20:05.50 | Kobaz | see how it's cut up there |
| 20:05.56 | Kobaz | and then it's fine after that |
| 20:06.13 | loggiew | what is the load like on your network |
| 20:07.20 | Kobaz | no idea |
| 20:07.28 | Kobaz | it's not actually my network |
| 20:07.41 | loggiew | and, does it do it on the LAN also or only across the intarwebs? |
| 20:07.52 | Kobaz | but everything's on a cisco catalyst with like 600gigabit backplane |
| 20:08.09 | Kobaz | apparently there's no reported issues on exten->exten calls |
| 20:08.27 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:08.33 | loggiew | ya, those switches still get bad cards etc |
| 20:08.34 | loggiew | hm |
| 20:08.44 | Kobaz | and the other tech who does have access to the cisco is saying it's very very low usage |
| 20:09.05 | Kobaz | so like |
| 20:09.15 | Kobaz | i can try turning on a jitter buffer? |
| 20:09.32 | Kobaz | but i dont think that's gonna matter if i'm being passed broken up audio |
| 20:09.37 | loggiew | so it suggests its across the internet. Possibly a bogged down router even if the switch is fine? |
| 20:09.48 | Kobaz | well nothing is internet |
| 20:09.59 | Kobaz | dedicated fiber from lightpath, with lightpath voip coming in on it |
| 20:10.09 | loggiew | ah |
| 20:10.15 | Kobaz | going to an avaya, and then going to the asterisk for conf bridging |
| 20:10.33 | Kobaz | no reported issues on exten<->exten on the avaya |
| 20:12.12 | loggiew | im not familiar with the avaya devices. Possibly it's external port is fucked or something? |
| 20:14.57 | marceloamorim | oh boy, I`m so confused right now, I have an tdm400p with x100m ( FXO ) modules and my /dahdi/system.conf I set fxsks=1-4 but when I restart my dahdi the message DAHDI_CHANCONFIG failed on channel 1: invalid argument (22) keep apper |
| 20:15.21 | loggiew | omg, every time my phone rings and i answer/hang up, it rings again. I cant get it to actually accept the call or stop :P |
| 20:15.22 | marceloamorim | the manual said to do that |
| 20:18.40 | Kobaz | loggiew, no idea |
| 20:19.25 | loggiew | ok, it can tell i hung up the call. But it hangs up and calls back immediately when I answer |
| 20:19.28 | loggiew | wtf |
| 20:20.34 | loggiew | gooooooooooot it |
| 20:27.23 | *** join/#asterisk dxd828_ (~dxd828@dsl-dynamic-77-44-45-38.interdsl.co.uk) |
| 20:28.57 | *** join/#asterisk uosiu (~uosiu@hajdamowicz.info) |
| 20:31.27 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 20:33.26 | loggiew | yaaaaay |
| 20:33.30 | uosiu | Hi all. I'm trying to diagnose weird problem with asterisk & SIP. When VoIP phone (extension 20) calls 30 or 31, connection is established and both ends hears each other. When 30/31 calls 20 only data stream is passed and no voice stream is send. Asterisk says it's zombie |
| 20:34.35 | loggiew | is one of them behind a NAT and not configured properly? |
| 20:34.55 | uosiu | no, no NAT at all |
| 20:35.19 | uosiu | both phones run on stock, default setting and only username, password and SIP server is entered |
| 20:35.25 | uosiu | Yealink T42G |
| 20:35.47 | uosiu | asterisk and phones run in same local network |
| 20:36.05 | loggiew | ah |
| 20:36.08 | loggiew | hm |
| 20:36.53 | uosiu | http://wklej.to/j9jUp rasterisk at "core set verbose 99" |
| 20:38.00 | uosiu | one of example cases - attended forward. Incoming call from upstream SIP provider answered by agent 31 and 31 wants to transfer call to 20 |
| 20:39.17 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 20:39.20 | loggiew | i was having similar (not the same) problems earlier and I was surprised to find out that not having proper codecs setup contributed to a lot of that |
| 20:42.58 | loggiew | i was told being to general might have allowed it to default to the wrong one on one device |
| 20:45.43 | uosiu | hmmm, both has same enabled setup: pcmu, pcma, g729, g722 :/ |
| 20:51.13 | uosiu | OK, fresh, local call from 20 to queue "10" which contains both 31 and 30. 20 calls with silence. After while it responds "Timed out" and hangs connection. Asterisk still calls 30 and plays music on hold for 20, despite fact 20 hanged out :O |
| 20:52.16 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 20:54.44 | *** join/#asterisk wolrah (~wolrah@50.54.48.90) |
| 20:57.34 | Kobaz | fooooound it |
| 20:57.56 | Kobaz | loggiew: i found a thing in wireshark that'll show % of packets out of order 2.4% out of order from this one source so far |
| 20:57.58 | Kobaz | that'll do it |
| 20:58.09 | Kobaz | so it is a local problem, or at least... some of it is a local problem |
| 20:58.31 | Kobaz | so jitter buffer would fix that if that's the only issue |
| 20:58.53 | Kobaz | but i think they should qos the voice... this is a big customer and the tech said they don't qos, which was mind boggling |
| 20:59.01 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 21:00.35 | loggiew | Kobaz: niiice |
| 21:00.47 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:00.59 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-hdmkmknqbdaqhuhg) |
| 21:00.59 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:01.11 | loggiew | qos no doubt. do they run qos on the rest of their network? |
| 21:01.18 | loggiew | or is it a blank canvas? |
| 21:03.30 | *** join/#asterisk nny (~Scott@cpe-174-107-218-002.sc.res.rr.com) |
| 21:03.34 | marceloamorim | guys, I couldn`t fix this problem, anyone could help me? |
| 21:03.46 | loggiew | marceloamorim: Im not familiar enough with that, Im sorry. |
| 21:04.06 | nny | I have am trying to use a local phone to forward calls (bypass telco issue) but I get app_dial.c:901 do_forward: Not accepting call completion offers from call-forward recipient Local/4222394@sip-00000001;1 Do I have a setting off somewhere? |
| 21:04.43 | marceloamorim | don`t worry loggiew |
| 21:04.48 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:88a5:3889:2994:3ecf) |
| 21:05.29 | nny | http://pastebin.com/VSgbL9TP is full output of cli |
| 21:05.38 | Katty | OH WHERE IS MY HAIRBRUSH |
| 21:05.44 | nny | the 302 moved temp precedes it |
| 21:08.51 | nny | trying promiscredir= yes |
| 21:12.11 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 21:14.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:16.00 | Katty | fenderbender. |
| 21:16.36 | gusto | hey, is that a bad thing when i am planning to define nat=yes for only one peer? |
| 21:16.54 | loggiew | gusto: I think that's what I just had to do to get it working properly |
| 21:16.58 | gusto | because there is this security issue that an attacker could then find out that .. you know what |
| 21:17.00 | Katty | guess it depends on where your peer is |
| 21:17.03 | Katty | and if you want rtp to work properly |
| 21:17.23 | gusto | yes, the RTP is the problem behind NAT |
| 21:17.31 | nny | er ignore me |
| 21:17.42 | nny | it was working, person I was forwarding to had their cell phone muted >< |
| 21:17.50 | loggiew | :D lol |
| 21:17.54 | gusto | does someone know how it works exactly? does it wait until the other side starts sending RTP and it is sending back to the same port (comedia)? |
| 21:18.00 | loggiew | gotta love users |
| 21:19.00 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 21:19.37 | loggiew | gusto: Im relatively newish to asterisk but isn't that what stun/ice servers and such are for? |
| 21:19.44 | gusto | when i say that i want to use tcp (or tls) transport for one peer, is it exclusive, so do the udp auth reqests fail then? |
| 21:20.21 | loggiew | gusto: it's my understanding that you can define most things such as that on a per peer basis |
| 21:20.27 | gusto | stun only tells you where you are, but you also need something from the clients side to kick a port open where you want to send your RTP data |
| 21:21.06 | loggiew | gusto: I was under the belief that stun was the middle point where both hosts could communicate as an OOB option on what ports and IP's they are on |
| 21:21.22 | gusto | and also limiting the access range of IP addresses for a peer |
| 21:21.27 | loggiew | my phone is behind a NAT with no open ports |
| 21:21.52 | loggiew | i dunno, it's probably getting outside the area of what Im familiar with pretty quickly. |
| 21:22.23 | gusto | your phone maybe uses the same port for in and out RTP |
| 21:23.03 | loggiew | regardless, it has to open the port from the inside and indicate which port and IP to the server |
| 21:23.13 | loggiew | it uses comedia etc for 'detecting' things like that |
| 21:23.38 | gusto | yes |
| 21:23.59 | loggiew | so between comedia and stun you should be able to get accurate data on the machine behind the NAT using the stun server to clear up the confusion. |
| 21:24.07 | loggiew | which means any open port behind a nat |
| 21:24.09 | gusto | but theoretically one could also use two (or more) ports ... it would only need to kick that ports open first |
| 21:24.15 | loggiew | is open because the NAT'ed device opened it first |
| 21:24.31 | Kobaz | loggiew: i have no idea what they do in terms of qos |
| 21:25.02 | loggiew | the device registers across a port (mine appears to default to 5060 regardless, im wondering about multiple devices behind the NAT) |
| 21:25.14 | Kobaz | loggiew: even in a non-maxed out network packet prioritization is beneficial to guarantee latency.. people dont understand that |
| 21:25.20 | loggiew | once that port is open, you use things like 'qualify=yes' to keep it open |
| 21:25.28 | Kobaz | they think, oh my pipe is big enough for everything and then some, i dont need to qos |
| 21:25.48 | gusto | loggiew: when there are multiple devices on the same port, it should use another port |
| 21:25.52 | loggiew | Kobaz: until you drop 100x the load of web traffic on top of your tiny voip traffic |
| 21:26.21 | Kobaz | at random and unspecific intervals of course |
| 21:26.41 | loggiew | gusto: that's what I figured, but regardless the cell phone opens up 5060 on the NAT and maintains registration through that open port. It's the control layer |
| 21:28.23 | loggiew | the rest of the ports opening and closing should be easily detected #1 as it responds on the port it receives from or #2 through the stun server |
| 21:29.02 | loggiew | using comedia and force_rport to detect oddities it should acount for in the addressing and ports |
| 21:29.38 | loggiew | but the device behind the NAT, as far as Im aware, must be the one to initiate unless you have ports explicitely opened and forwarded |
| 21:30.36 | jzaw | if im accepting guest uri calls to default context ... and have say just one exten in there ... which dials a single internal extension |
| 21:31.12 | jzaw | how might i best handle calls to non existant extensions ... random attempts ... currently ppl tell me they get number busy |
| 21:31.23 | jzaw | wouldnt sit tones number not in service be better? |
| 21:31.30 | jzaw | answer() |
| 21:31.37 | jzaw | zapateler() |
| 21:31.42 | jzaw | hangup() |
| 21:31.45 | jzaw | something like that? |
| 21:31.57 | jzaw | in an i ext ? |
| 21:32.08 | loggiew | wouldn't you create a pattern and use it to match anything not the extension you want? Then redirect it to the appropriate sound or back to the original menu? |
| 21:32.37 | jzaw | well it's a v simple context ... just goes to one extension if you call the right uri |
| 21:33.10 | jzaw | so EVERYTHING bar 1 call is going to fail |
| 21:34.24 | jzaw | i guess you confirmed my own answer .. except id prefer not to answer ... just to give SIT tones |
| 21:34.40 | loggiew | ah. makes sense. |
| 21:34.49 | loggiew | Im not familiar enough to answer if there is a better way |
| 21:34.57 | jzaw | i consider it rude (and imposing a potential cost on innocent callers) to answer only to say nahnahahana no one here |
| 21:35.17 | loggiew | agreed. it's a fair point. |
| 21:35.23 | jzaw | will zapateler play over early audio? |
| 21:35.54 | loggiew | not sure. |
| 21:36.03 | jzaw | mind you ... sip uri calls arent normally from pstn ... so there's prob no cost involved |
| 21:36.10 | loggiew | ive literally told you probably near everything I know about it at this point :D |
| 21:36.44 | jzaw | is cool ! that's what chat is for ... edging forward by sharing even meagre bits of info |
| 21:36.48 | jzaw | :) |
| 21:37.06 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 21:37.09 | loggiew | couldnt agree more |
| 21:37.23 | loggiew | ive been teaching some very computer illiterate people about linux and programming |
| 21:37.26 | loggiew | its taken a year |
| 21:37.29 | loggiew | but i always tell them |
| 21:37.35 | loggiew | the key, is casual conversation. |
| 21:38.20 | jzaw | tiny steps ... even steven hawkin (brain the size of a minor planatoid </hhgttg>) started with zero knowledge |
| 21:39.29 | loggiew | existance is the presense of knowledge and the lack of existance the lack of information to be known |
| 21:39.38 | loggiew | we all feel like we start with zero |
| 21:39.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 21:40.30 | loggiew | we dont think about the idea that our very existance is proof that it can be learned. whatever it is |
| 21:41.47 | marceloamorim | guys, I don`t know if there was any change on this, but I fix the problem with my dahdi modules on tdm400p with x100m FXO |
| 21:42.26 | marceloamorim | I load other modules and this other modules fix the problem on /usr/local/etc/rc.d/dahdi start |
| 21:42.53 | loggiew | marceloamorim: Im glad you got it fixed |
| 21:43.05 | *** join/#asterisk tris- (tristan@camel.ethereal.net) |
| 21:43.48 | *** join/#asterisk ivan`_ (~ivan@unaffiliated/ivan/x-000001) |
| 21:43.55 | marceloamorim | me too loggiew =) |
| 21:44.55 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:45.46 | *** join/#asterisk CeBe (~CeBe@port-92-206-38-251.dynamic.qsc.de) |
| 21:47.28 | ipengineer | I am having trouple using the Pickup CMD.. I am getting: No target channel found for 200@a_extensions. Do I need to do something special to make pickup work? I am setting __SOURCE_CONTEXT=a_extensions on the extension that is ringing (200) |
| 21:48.32 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 21:52.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:53.50 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 21:53.54 | ipengineer | [TK]D-Fender: Notice you just jumped in.. Do you have any thoughts on this? |
| 21:53.55 | ipengineer | I am having trouple using the Pickup CMD.. I am getting: No target channel found for 200@a_extensions. Do I need to do something special to make pickup work? I am setting __SOURCE_CONTEXT=a_extensions on the extension that is ringing (200) |
| 21:54.55 | [TK]D-Fender | Show actual backup and configs |
| 21:54.56 | ipengineer | wait let me clarify on the “on the DEVICE that is ringing” |
| 21:56.44 | ipengineer | [TK]D-Fender: https://gist.github.com/zconkle/35e6d6e28782b1a6623f |
| 21:59.16 | [TK]D-Fender | ipengineer: just like it says... there is no CHANNELNAME like that ringing |
| 21:59.29 | [TK]D-Fender | SIP/dispatch2-0000005f <------------------------------- |
| 21:59.43 | [TK]D-Fender | ^- Channel |
| 22:00.02 | [TK]D-Fender | reaches for his second quiver |
| 22:00.14 | [TK]D-Fender | </legolas> |
| 22:00.52 | ipengineer | How would I declare that in the dialplan since that channel is dynamic? |
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| 22:03.36 | *** join/#asterisk wolrah_ (~wolrah@50.54.48.90) |
| 22:06.04 | [TK]D-Fender | ipengineer: Clearly that app is not the right one to target that way |
| 22:07.13 | ipengineer | Alright.. I will do some more digging and see what I can find. Thanks for taking a look |
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| 22:27.15 | *** part/#asterisk mjordan (~matt@nat/digium/x-ozjipyuzryfulrpn) |
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