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| 03:42.53 | rue_house | http://paste.debian.net/224765/ <-- this is my mgcp.conf, if I enable both gateways to try to route calls betwen them, I get the same errors I would if one of them were unplugged from the network |
| 03:43.21 | rue_house | if I enable either one of them, to route calls between its ports, it works fine |
| 03:43.42 | rue_house | as soon as that config file sees two configured, the second one defined stops working |
| 03:43.48 | rue_house | that is my mgcp.conf problem |
| 03:43.52 | rue_house | can anyone help |
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| 03:44.40 | [TK]D-Fender | rue_house: Stop doing half the job and show us the actual problem. |
| 03:44.51 | rue_house | what do you want |
| 03:44.54 | rue_house | I dont have logs |
| 03:45.20 | rue_house | its simple, if both those configs are enabled, the second one dosn't work |
| 03:45.27 | rue_house | I'm sure someone knows something |
| 03:45.34 | rue_house | does nobody use mgcp? |
| 03:45.52 | rue_house | is there a glarring error in the config file? |
| 03:47.28 | rue_house | I cant write an online system emulator just to demonstrate the problem |
| 03:48.54 | [TK]D-Fender | Go make logs |
| 03:49.01 | rue_house | ugh |
| 03:49.28 | rue_house | took me a week to get the config files so I could post them |
| 03:49.39 | [TK]D-Fender | And why is that? |
| 03:49.54 | rue_house | time and location |
| 03:49.59 | rue_house | I cant be everywhere all the time |
| 03:50.05 | [TK]D-Fender | You don't have to. |
| 03:50.24 | [TK]D-Fender | You just have to be HERE at the time when you can actually show us something we can debug. |
| 03:50.51 | rue_house | if you unplug a mgcp gateway from the network, it throws an error something like... arg |
| 03:51.10 | rue_house | when I'm infront of the bug there is nobody here |
| 03:51.30 | rue_house | I cant sit at work for 5 hours at a desk waiting for a reply to a question |
| 03:51.53 | [TK]D-Fender | You just said you could immediately trigger that outcome just be enabling the other peer. |
| 03:51.54 | rue_house | I have to go and verify fire alarms and install closet lights |
| 03:52.03 | [TK]D-Fender | tht doesn't sound like you have to sit there and wait. |
| 03:52.05 | rue_house | yes |
| 03:52.11 | [TK]D-Fender | You can trigger it on demand, or so you said |
| 03:52.21 | [TK]D-Fender | so come back when you're in a position to di so. |
| 03:52.25 | [TK]D-Fender | You have nothing to show us. You know full well what a complete waste of time that is... |
| 03:52.27 | rue_house | its "no, we cant look at your problem till you bring us this one more thing" |
| 03:52.34 | [TK]D-Fender | One more thing? |
| 03:52.38 | [TK]D-Fender | You can't SHOW us the problem. |
| 03:52.39 | rue_house | you want logs now |
| 03:52.50 | rue_house | I should make a video then |
| 03:52.54 | rue_house | post it on youtube |
| 03:52.56 | rue_house | ok |
| 03:53.00 | [TK]D-Fender | Don't be a douche... |
| 03:53.09 | [TK]D-Fender | this is a basic CLI dump jsut like any other problme. |
| 03:54.16 | rue_house | is there a way from the console to insert a message into the log? |
| 03:54.28 | rue_house | from the asterisk console? |
| 03:54.59 | [TK]D-Fender | Why woud you have to? |
| 03:55.04 | [TK]D-Fender | +l |
| 03:55.31 | rue_house | because then I can say "and now I'm switching to the other gateway" before I generate the next error |
| 03:56.17 | rue_house | it seems to me that mgcp.conf cannot operate two gateways, and I'm sure if someone took a look they could confirm this for me |
| 03:56.41 | [TK]D-Fender | You know... when you're shoving it in a PASTEBIN for us to look at ... you could just HAND TYPE that bit in.. |
| 03:57.11 | rue_house | yea, so I need to makea webpage thats a hybrid of the log and hand notes |
| 03:57.25 | rue_house | I cant just do it all on the console and paste the whole log |
| 03:57.32 | [TK]D-Fender | Do YOU see something about Hand type 1 stupid line while pastebinning. |
| 03:57.53 | rue_house | it shouldn't be hard to make a 'user console generated log message' hmmmm |
| 03:57.55 | [TK]D-Fender | wow, that mashed up pretty bad.. |
| 03:58.10 | [TK]D-Fender | Look at the grade of crap you're going on about. |
| 03:58.29 | rue_house | no, I need to go thru combinations of about 3 different factors to display the whole problem |
| 03:58.30 | [TK]D-Fender | copy. Paste. hit enter 3 times. Type your 1 line. SUBMIT |
| 04:02.13 | rue_house | its not that simple because |
| 04:04.11 | rue_house | .. I'm still typing... |
| 04:04.34 | rue_house | I have to stop and pastebin this.. |
| 04:06.17 | rue_house | http://paste.debian.net/224766/ |
| 04:06.49 | rue_house | I'm sure until I'v gone over the hardware/software reconfigurations of each of those situations you guys aren't gonna try to start to help me with the problem |
| 04:07.10 | rue_house | and I'm 100% confident that after I have the answer will be "yup, you have a problem" |
| 04:08.12 | rue_house | much like the problem with the polycom audio levels (which still isn't solved, nobody understands the questions of what the volume controls in the xml file control or what the valid range of their values is) |
| 04:08.49 | rue_house | half the time the office is still shouting "WHAT" into the phone with their finger in their non-phone ear |
| 04:09.46 | rue_house | and if nobody had ever told me to go polycom instead of aastra, I'd never known that the sip levels are impossable to debug |
| 04:10.12 | rue_house | sorry :) I'm about 110% frustration |
| 04:10.29 | [TK]D-Fender | And 0% practical |
| 04:10.57 | rue_house | do you see why it will be a problem for me to just generate the webpage describing the problem? |
| 04:11.16 | [TK]D-Fender | "generate a webpage? |
| 04:11.18 | rue_house | how I have to do about 14 hardware/configuration changes to demonstrate the problem? |
| 04:11.24 | [TK]D-Fender | Stop making pastebins sound like voodoo |
| 04:11.42 | [TK]D-Fender | Apparently you can do ONE and show us, and a dozen different ways. |
| 04:11.52 | rue_house | I prolly have to have it posted longer than 90 days, so I might as well make it a webpage |
| 04:11.54 | [TK]D-Fender | Maybe you should show us ONE of those. |
| 04:11.56 | [TK]D-Fender | But you don't |
| 04:12.04 | *** join/#asterisk hariom (~hariom@117.231.255.242) |
| 04:12.19 | [TK]D-Fender | And you came in telling us we aren't going to get it. |
| 04:12.27 | rue_house | I cant do that, buttons to bisvuits that I'm asked for them all one at a time over the course of 2 months |
| 04:12.34 | rue_house | I'll get it |
| 04:13.04 | rue_house | but its only for educational use, may name is already mud for not having this working already |
| 04:13.08 | hariom | Hi, I am getting error while loading chan_alsa.so It is enabled in make menuselect. But when I try to load it, it says can not read chan_alsa.conf |
| 04:13.37 | rue_house | hariom, does the chan_alsa.conf file exist in /etc/asterisk/ ? |
| 04:13.56 | hariom | yea. I copied it from /usr/share/... |
| 04:14.06 | hariom | rue_house |
| 04:14.14 | rue_house | ok, what are the permissions on the file? |
| 04:14.31 | rue_house | to find out, ls -l /etc/asterisk/chan_alsa.conf |
| 04:15.00 | hariom | rue_house: it has local user permission as I am running asterisk in non super user mode. I change its permission via chown |
| 04:15.08 | rue_house | I *THINK* they have to be readable by the "asterisk" user |
| 04:15.34 | rue_house | is the permission the same as the other config files? |
| 04:15.38 | hariom | rue_house: yea, I change the user and run group in asterisk.conf to local user |
| 04:15.42 | hariom | yea |
| 04:16.16 | hariom | rue_house: asterisk is running fine. Other modules are working good |
| 04:16.19 | rue_house | ok, now this one you need to check carefully. Is there any chance you made a typo in the filename }:) |
| 04:18.18 | hariom | rue_house: CLI> module load chan_alsa.so Unable to load module chan_alsa.so Command 'module load chan_alsa.so' failed. == Parsing '/etc/asterisk/alsa.conf': == Found |
| 04:18.29 | rue_house | oooh |
| 04:18.37 | rue_house | cant load the module, not the config file |
| 04:18.40 | rue_house | ok |
| 04:19.19 | rue_house | hariom, what do you get when you do the command locate chan_dahdi.so |
| 04:20.07 | hariom | rue_house: I don't have dahdi installed as I am only using sip. So only asterisk package is installed not the dahdi and libpri |
| 04:20.25 | rue_house | ok, just a sec |
| 04:20.40 | rue_house | locate chan_sip.so |
| 04:20.44 | rue_house | what do you get |
| 04:21.21 | hariom | rue_house: no output. I just gives command prompt again |
| 04:21.29 | rue_house | interesting |
| 04:21.39 | rue_house | did it complain about database not up to date? |
| 04:21.45 | rue_house | how long ago did you isntal asterik? |
| 04:22.16 | hariom | rue_house: about a week back |
| 04:22.42 | rue_house | odd there is no chan_sip.so |
| 04:23.07 | hariom | rue_house: ok, I found chan_alsa.so inside /usr/lib/asterisk/modules |
| 04:23.24 | rue_house | hmm |
| 04:23.29 | rue_house | thats the right place |
| 04:23.52 | rue_house | so if you do locate chan_sip.so does it show you that location? |
| 04:24.05 | rue_house | er locate chan_alsa |
| 04:24.30 | rue_house | heh |
| 04:24.39 | rue_house | sorry I'm ill tonight, not thinking striaght |
| 04:24.45 | rue_house | locate chan_also.so |
| 04:24.47 | rue_house | a |
| 04:24.49 | rue_house | arg1 |
| 04:26.03 | hariom | rue_house: no output with or without '.so' . In asterisk.conf file, I see the modules directory is mentioned /usr/lib/asterisk/modules |
| 04:26.16 | rue_house | run the command updatedb |
| 04:26.21 | rue_house | wait for it to finish |
| 04:26.27 | rue_house | then the locate command should work |
| 04:27.06 | rue_house | did you compile asterisk yourself? |
| 04:27.12 | rue_house | you must have |
| 04:27.27 | rue_house | wonders if you missed an install step |
| 04:27.33 | rue_house | make install ? |
| 04:28.15 | hariom | rue_house: yea, after updatedb, I see locate command working fine. It shows chan_sip.so in both the asterisk source directory and /usr/lib/asterisk/modules |
| 04:28.27 | rue_house | ok |
| 04:28.27 | dpilon | hariom: check your private msg |
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| 04:28.41 | rue_house | and it shows chan_alsa in the same place? |
| 04:29.32 | rue_house | how about permissions on chan_alsa.so? |
| 04:30.31 | rue_house | thought it felt like your in other conversations, goodnight |
| 04:30.45 | dpilon | not really |
| 04:30.58 | hariom | rue_house: complete directory has user ownership |
| 04:32.00 | rue_house | if both files exist and are readable/executable by the user asterisk is running as, I'm out of leads |
| 04:32.15 | rue_house | sorry |
| 04:35.35 | hariom | rue_house, dpilon: wait, I am pasting the complete output in pastebin |
| 04:35.56 | dpilon | cool |
| 04:37.32 | hariom | rue_house, dpilon: http://pastebin.mozilla.org/2060294 |
| 04:38.04 | dpilon | pastebin your alsa.conf |
| 04:38.12 | hariom | Could there by any way to regenerate alsa.conf? I suspect that that is corrupt |
| 04:38.39 | [TK]D-Fender | maybe look at the sample config * comes with .... |
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| 04:40.03 | hariom | rue_house, dpilon, [TK]D-Fender: http://pastebin.mozilla.org/2060300 |
| 04:40.53 | [TK]D-Fender | That sure isn't an * config file at all... |
| 04:41.08 | hariom | wow, what I have in * is way too different than this |
| 04:41.53 | [TK]D-Fender | Perhaps you tried copying your actuall dirver confi over *'s as thought it were somehow related.... |
| 04:41.57 | [TK]D-Fender | driver* |
| 04:45.14 | hariom | rue_house, dpilon, [TK]D-Fender: It worked. I can load chan_alsa.so now. Just copied from *. |
| 04:45.28 | hariom | Thank you very much guys. |
| 04:49.07 | hariom | [TK]D-Fender, rue_house, dpilon: Need guidance on how to dial from CLI. I am using sip (localsystem with no dahdi and libpri). Sip user is 1234 and extension context is LocalNum with extension as 1005 |
| 04:53.18 | hariom | I got chan_alsa.so loaded but I think I still don't have dial command |
| 04:53.43 | [TK]D-Fender | hariom: Chan ALSO is pretty crap and is a tool you should only use if yuo have no choice. |
| 04:54.00 | [TK]D-Fender | Why not jsut run a proper SIP client on your box and have it connect locally? |
| 04:54.08 | [TK]D-Fender | ALSA* |
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| 04:55.25 | hariom | [TK]D-Fender: I have Twinkle softphone on my ubuntu 10.04 system but it hangs after every call completion. I have to kill it to make another call |
| 04:55.57 | ruben231 | hi guys is this correct how i add 99900 on teh dialplan ----> exten => _801160.,n,Dial(${TRUNK2VOIP}/99900${EXTEN:1},50,tTor) |
| 04:55.58 | [TK]D-Fender | get another. |
| 04:56.01 | [TK]D-Fender | there are dozens |
| 04:56.51 | hariom | [TK]D-Fender: what softphone you use that has been working good? |
| 04:57.23 | [TK]D-Fender | Ekiga. Linphone. |
| 04:58.27 | hariom | [TK]D-Fender: I have tried Ekiga as well but it doesn't bring any sound when I make call |
| 04:58.27 | [TK]D-Fender | Make sure you change the SIP PORT on the softphone to something other than 5060 if it's on the server itself so it doesn't conflict with *'s binding of the port. |
| 04:58.46 | [TK]D-Fender | maybe something else is wrong. |
| 04:58.50 | hariom | [TK]D-Fender: yea, I use 5061 for softphone |
| 04:59.00 | [TK]D-Fender | You haven't told us anything by way of details or provided debug |
| 04:59.46 | hariom | [TK]D-Fender: It gives options for selecting audio devices but default option doesn't bring any audio. I tried other options as well. |
| 05:00.30 | [TK]D-Fender | hariom: Audio comes from some other side as well. your description is still weak and we don't have anythign to go on. |
| 05:00.47 | [TK]D-Fender | Make sure to NOLOAD chan_alsa, etc in your modules.conf so they aren't in the way |
| 05:01.05 | hariom | [TK]D-Fender: ok |
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| 08:11.58 | schmidts | good morning |
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| 08:37.10 | R1ck | [Jan 15 09:36:47] WARNING[18723]: chan_iax2.c:3488 __attempt_transmit: Max retries exceeded to host 80.101.65.254 on IAX2/ijsselburcht-432 (type = 6, subclass = 11, ts=57598426, seqno=62) |
| 08:37.15 | R1ck | what does that mean? |
| 08:42.32 | schmidts | Rick that the other side doesnt answer in a specified amount of time and asterisk will not retry to send something there for this call |
| 08:48.56 | ChannelZ | network troubles |
| 08:49.36 | ChannelZ | packets aren't making it out of your network to the other side, or their replies aren't making it back perhaps |
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| 09:02.56 | R1ck | weird.. this has been working fine for weeks |
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| 09:36.05 | fling | updated to latest opal, ptlib and ekiga, still do not have anything but alaw and ulaw! how to enable some nifty codecs like speex and x264? |
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| 10:16.17 | jkroon | hi guys, i'm looking for a way to accurately set the rx gains on a dahdi channel |
| 10:17.13 | jkroon | what I have is one system, connected to a pri, that can play milliwat on the one side |
| 10:18.16 | *** join/#asterisk SirFat (~medge@124-170-26-132.dyn.iinet.net.au) |
| 10:18.29 | SirFat | Hi - got a question if someone can help |
| 10:18.33 | SirFat | ;-) |
| 10:18.34 | jkroon | on the other side i can obviously send a call out to that system (over analog line, via exchange to the PRI), do I have a way of measuring the incoming "level" for a period? |
| 10:18.36 | SirFat | cause I'm clearly a newbie |
| 10:18.52 | jkroon | SirFat, nobody is going to help unless you actually ask a question. |
| 10:19.04 | SirFat | I thought I'd patiently see if I was being rude ;-) |
| 10:19.18 | SirFat | I am in the process of converting my asterisk 1.4.9 installation from being ISDN into SIP |
| 10:19.39 | SirFat | I have setup the peer as best I can guess, but when i perform an inbound call, I am noticing that the number it's looking up in my sip_line table, is actually the CALLER, not the CALLED number |
| 10:19.44 | SirFat | so, of course, it never works |
| 10:19.56 | jkroon | if there is an app that can measure the level then obviously I can measure what i'm receiving for eg 2s, adjust a little, then listen again, until I get just the right level ... |
| 10:20.30 | jkroon | then for adjusting the TX gains it gets a bit trickier but basically I can then somehow loop the call onto the machine itself (will have to require multiple lines) |
| 10:21.54 | SirFat | DEBUG[32731] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_lines WHERE name = '61439367205' is what appears (versus the number I dialed) |
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| 10:24.55 | wdoekes | SirFat: if sip_lines is your realtim sipusers/sippeers table, then it would look up the bit in the From, indeed |
| 10:25.21 | wdoekes | asterisk looks up 'who' is calling, not where it is calling |
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| 10:26.05 | wdoekes | you'll need separate functionality for that. extensions != accounts |
| 10:26.08 | SirFat | wdoekes: so i guess I'm configuring the sip peer part in the wrong location? |
| 10:26.17 | SirFat | ok, that's the extconfig.conf bit? |
| 10:26.26 | SirFat | which is blah,asterisk,sip_lines |
| 10:26.30 | SirFat | for peers and users |
| 10:26.51 | wdoekes | that is the who-is-calling (and who-is-registering) bit |
| 10:27.09 | wdoekes | the where-am-i-calling is in your extensions.conf |
| 10:27.53 | wdoekes | which would be a static file in the simplest of cases. but you can spice it up with e.g. func_odbc |
| 10:28.28 | wdoekes | exten => X!,1,Set(account=${ODBC_LOOKUP_FUNC(${EXTEN})}) |
| 10:28.33 | SirFat | Yeah. this is being defined in the sip.conf |
| 10:28.38 | SirFat | ok, I shall look |
| 10:29.03 | wdoekes | exten => X!,n,Dial(SIP/${account}) |
| 10:29.16 | wdoekes | note that you'll have to create ODBC_LOOKUP_FUNC yourself |
| 10:29.19 | wdoekes | ~book |
| 10:29.19 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:29.31 | SirFat | Yeah. Cool. |
| 10:29.40 | SirFat | then lsatly |
| 10:29.52 | SirFat | how do I turn on super debug mode to see how the call is routing itself through the config |
| 10:29.52 | SirFat | ? |
| 10:30.06 | wdoekes | set verbose 20 |
| 10:30.09 | SirFat | oki |
| 10:30.18 | SirFat | thankyou. I will continue plugging away and see if I can figure it out |
| 10:30.25 | *** join/#asterisk PbxMan (c335d959@gateway/web/freenode/ip.195.53.217.89) |
| 10:30.26 | wdoekes | enjoy |
| 10:30.31 | wdoekes | and consider upgrading to asterisk 11 |
| 10:30.33 | PbxMan | Morning |
| 10:30.34 | SirFat | heh |
| 10:30.37 | SirFat | I did that earlier |
| 10:30.41 | SirFat | but the thing shat the bed ;) |
| 10:30.41 | wdoekes | because you'll get better support |
| 10:30.59 | SirFat | I was trying to salvage the ivr stuff the last fellow put in |
| 10:31.41 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 10:38.04 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 10:41.44 | fling | are you using ekiga |
| 10:41.58 | linocisco | hi all |
| 10:43.24 | PbxMan | I do use it fling |
| 10:44.25 | fling | PbxMan: I do not have any codecs but alaw and ulaw, idk how to fix it |
| 10:44.37 | linocisco | I have Avaya BCM450 but which run out of VOIP phone licenses . Can we extend it with asterisk server and sip phones? |
| 10:44.48 | fling | PbxMan: I've tried to enable everything in ptlib, opal, still nothing, no speex and x264, etc |
| 10:45.35 | linocisco | I have Avaya BCM450 but which run out of VOIP phone licenses . Can we extend it with asterisk server and sip phones? with BCM450 as main PBX in place and another route or trunk to Asterisk to let asterisk's SIP clients to use BCM access |
| 10:48.03 | PbxMan | what OS are you using fling ? |
| 10:48.10 | fling | PbxMan: gentoo gnu/linux |
| 10:48.42 | PbxMan | I never had that problem I work with Ubuntu |
| 10:49.34 | PbxMan | Have you tried with another Sip Client? |
| 10:50.08 | fling | PbxMan: yes, have codecs with linphone, because it is using mediastreamer instead of tplib&opal |
| 10:53.03 | fling | PbxMan: am I missing some useflag? > http://dpaste.com/877740/ |
| 10:54.08 | PbxMan | fling: what Ekiga VO are you using? |
| 10:55.11 | fling | PbxMan: tried 3.2.7 and 4.0.0 |
| 10:55.35 | PbxMan | try 2.0.12 |
| 10:56.37 | fling | PbxMan: trying |
| 10:59.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:59.28 | fling | PbxMan: no, it depends on gnome and pulseaudio, I do not want it |
| 10:59.40 | *** join/#asterisk RZero (~RZero@85.118.159.250) |
| 11:00.18 | PbxMan | have you verified this? http://packages.gentoo.org/package/net-voip/ekiga |
| 11:00.42 | RZero | Hi all need some help, Ive just upgrade asterisk 1.6 to 1.8 all working fine, but how to restore the fax license ? |
| 11:01.06 | fling | PbxMan: http://dpaste.com/877743/ |
| 11:01.20 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:02.14 | PbxMan | fling: I never had to do this to make it work on ubuntu, I cannot help you sorry |
| 11:02.28 | fling | PbxMan: ok, thank for help :] |
| 11:03.38 | fling | PbxMan: hmm hmm |
| 11:03.55 | fling | PbxMan: don't I need opal plugins? |
| 11:04.25 | linocisco | I have Avaya BCM450 but which run out of VOIP phone licenses . Can we extend it with asterisk server and sip phones? with BCM450 as main PBX in place and another route or trunk to Asterisk to let asterisk's SIP clients to use BCM access |
| 11:05.10 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.30.138) |
| 11:05.31 | PbxMan | It comes with a few codecs out of the box and the dependencies are installed from the Repo, besides you could get it from the ubuntu software center |
| 11:08.42 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-187-53.red.bezeqint.net) |
| 11:08.45 | linocisco | I have Avaya BCM450 but which run out of VOIP phone licenses . Can we extend it with asterisk server and sip phones? with BCM450 as main PBX in place and another route or trunk to Asterisk to let asterisk's SIP clients to use BCM access |
| 11:10.41 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 11:18.24 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:22.15 | fling | PbxMan: fixed enabling opal plugins useflag |
| 11:22.37 | PbxMan | congrats fling |
| 11:25.25 | fling | PbxMan: :p |
| 11:25.48 | fling | how to enable silk in asterisk? |
| 11:44.21 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-187-53.red.bezeqint.net) |
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| 12:33.33 | *** join/#asterisk c4software (~vbrosseau@46.255.52.118) |
| 12:33.37 | c4software | Hi |
| 12:34.01 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 12:34.43 | c4software | I'm trying to play the mute/unmute sound to a channel in conf. The Mute/unmute event is sent to via the cli interface. |
| 12:34.50 | c4software | The is a way to do that ? |
| 12:38.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:42.48 | *** join/#asterisk c4software (~vbrosseau@46.255.52.118) |
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| 13:02.19 | *** join/#asterisk nicknam12322 (021f9722@gateway/web/freenode/ip.2.31.151.34) |
| 13:03.14 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:05.02 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 13:05.10 | bombev | hi all |
| 13:05.59 | fling | so tell me how to enable silk |
| 13:08.52 | *** join/#asterisk bananapie (~david@75-119-253-200.dsl.teksavvy.com) |
| 13:09.52 | bananapie | Hello, I just ran make menuselect and customized my build. I want to back up my customisation. When I compiled the kernel, I would copy .config to a USB key. But I can't find the comparable file in asterisk. Any suggestions ? |
| 13:11.07 | wdoekes | menuselect.makeopts, I think |
| 13:11.58 | bombev | I just made that context: http://pastebin.ca/2302589 |
| 13:12.46 | bombev | the goal is to restrict calls to fixed and mobile phones |
| 13:13.01 | bombev | and it works as a charm, but I got other issue |
| 13:13.19 | bananapie | wdoekes, that was the first place I checked. But it looks like menuselect.makeopts has all the default settings |
| 13:13.49 | bombev | when somebody try to call to real extensions it works good, but when somebody try to call conferene extensions such as 100, 101 the got failed |
| 13:15.01 | [TK]D-Fender | bombev, We don't see this other context, or where the call lands, or any other context we don't see that is involved because this isn't the actual original target |
| 13:15.10 | [TK]D-Fender | bombev, Show us the call |
| 13:15.14 | [TK]D-Fender | ~pb |
| 13:15.14 | 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. |
| 13:15.34 | [TK]D-Fender | bombev, Actually.... |
| 13:16.03 | [TK]D-Fender | bombev, exten => _XXX,1,NoOp() <-- this will match your 100 & 101 immediately, it has no need to look in that included context at all... |
| 13:16.13 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:16.13 | bananapie | menuselect.h says that menuselect.makeopts should have my options |
| 13:16.24 | [TK]D-Fender | bombev, It will always match in the original context and only look in includes if there isn't a match |
| 13:18.21 | bananapie | is it possible that menuselect.makeopts MENUSELECT_ADDONS and MENUSELECT_APPS contain the list of things to NOT compile ? |
| 13:19.03 | bombev | [TK]D-Fender but If i remove exten => _XXX,1,NoOp() ... the call got busy signal |
| 13:20.11 | *** join/#asterisk mirela666 (~mirela666@212.200.146.253) |
| 13:20.15 | [TK]D-Fender | bombev, Don't tell us... SHOW us. And we don't see what's in that other context either. You talk about "conferences" but we have no proof anything useful is there |
| 13:20.18 | bananapie | That's my problem, menuselect.makeopts is the file I want. But the items listed in MENUSELECT_ADDONS and MENUSELECT_APPS are the things that are not being built. |
| 13:20.19 | bananapie | thanks |
| 13:24.59 | bombev | [TK]D-Fender here you go: http://pastebin.ca/2302592 |
| 13:25.06 | *** join/#asterisk mirela666 (~mirela666@212.200.146.253) |
| 13:25.33 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
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| 13:26.16 | bombev | -- Executing [105@from-internal-conf:2] Dial("SIP/375-00000710", "SIP/105,25") in new stack |
| 13:26.22 | bombev | Purely numeric hostname (105), and not a peer--rejecting! |
| 13:27.23 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ckprrlxckjrfrjbr) |
| 13:27.54 | bombev | in my dialplan i have this: exten => _XXX,2,Dial(SIP/${EXTEN},25) |
| 13:28.11 | bombev | that includes 105 extension as well |
| 13:28.12 | bombev | right |
| 13:28.27 | WIMPy | Yes. |
| 13:28.39 | WIMPy | And what do you make of the message you posted before |
| 13:28.46 | WIMPy | ? |
| 13:29.02 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 13:29.07 | bombev | cuz, i dont understand that line: Purely numeric hostname (105), and not a peer--rejecting! |
| 13:29.29 | WIMPy | not a peer |
| 13:29.41 | WIMPy | 105 doesn't exist in your sip.conf. |
| 13:30.05 | bombev | yes 105 is not real peer |
| 13:30.25 | WIMPy | So what do you expect? |
| 13:30.31 | bombev | it is conferece extension |
| 13:31.08 | WIMPy | You just said something different. |
| 13:31.29 | *** join/#asterisk aurs (~aurs@110.84-49-69.nextgentel.com) |
| 13:31.32 | WIMPy | A Dial is not a conference. |
| 13:31.51 | bombev | well |
| 13:32.16 | bombev | any idea how to make this work with not real extension |
| 13:32.33 | kaldemar | all extensions are real. |
| 13:32.37 | WIMPy | What do you mean by "not real extension"? |
| 13:33.03 | WIMPy | You write your dialplan to do what you want it to do. |
| 13:33.15 | kaldemar | use a Goto or Dial(Local/exten@context) if you want the call to go to another extension. |
| 13:34.02 | WIMPy | Or get the extensions in the right order. |
| 13:34.51 | WIMPy | If you want an extension fom an included context to take preceedence, you have to create another context and include the others in the order you want. |
| 13:34.57 | bombev | well this 105 is Conferencing with MeetMe() |
| 13:35.15 | WIMPy | Obviousely not. |
| 13:35.42 | kaldemar | what the extension does is completely irrelevant at this point. a call does not go to another extension if you dial using chan_sip. |
| 13:36.13 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:40.32 | [TK]D-Fender | bombev, You just dialed a SIP device. That is not an extension. |
| 13:40.53 | [TK]D-Fender | bombev, There is no conference there. |
| 13:40.57 | [TK]D-Fender | ~book |
| 13:40.58 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:40.59 | [TK]D-Fender | ^^^ |
| 13:41.38 | [TK]D-Fender | bombev, Your pattern catches everything with 3 digits and you treat them all the same... |
| 13:41.43 | [TK]D-Fender | this is clearly no good. |
| 13:42.19 | [TK]D-Fender | bombev, you have nothing in there to treat 100 & 101 any differently\ |
| 13:43.04 | bombev | so i have to include few lines to treat 100,101...108 diff way |
| 13:43.59 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:49.01 | bombev | thanks guys for the help |
| 13:52.43 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
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| 14:05.26 | c4software | Its possible to know if a user leave a conference by himself or if he has been kiked ? |
| 14:06.34 | c4software | kicked |
| 14:19.51 | *** join/#asterisk sustav (~vpp@nat/digium/x-rbjgldxhcmgixoql) |
| 14:20.20 | *** join/#asterisk deo_ (~deo@112.198.82.15) |
| 14:21.44 | leifmadsen | might be an AMI event that you can monitor |
| 14:23.32 | c4software | hm |
| 14:25.53 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:29.34 | Katty | hello my asterisk does not work at all how to fix?? answer plz is urgent thx. |
| 14:30.19 | leifmadsen | Katty: step one, get a scotch |
| 14:30.28 | leifmadsen | Katty: step two, get a scotch |
| 14:30.50 | leifmadsen | Katty: step two(b): if you don't like scotch, just replace with shots of tequila |
| 14:30.50 | Katty | but cordial is so much tastier than scotch. |
| 14:30.59 | Katty | tequila is worse than scotch! >.< |
| 14:31.04 | WIMPy | Are most Asterisk Experts living in Scottland? |
| 14:31.10 | leifmadsen | Katty: I didn't say this was going to be pleasant -- your asterisk is down. It's going to be hard work. |
| 14:31.12 | Katty | and the only thing worse than tequila is Gin. |
| 14:31.42 | Katty | leifmadsen: *hee* |
| 14:34.28 | *** join/#asterisk seik0 (b244e46b@gateway/web/freenode/ip.178.68.228.107) |
| 14:36.32 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-aonzcochuwopjifn) |
| 14:36.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:37.18 | seik0 | Hi everybody. To the point ). We bumped into a problem with SIP. We accept incoming from sip provider, so we need to keep registration with "register => ..." option in sip.conf. Once we failed with internet connection and, after some time, all sip-communication within aterisk server was just blocked! I found some info here: https://issues.asterisk.org/jira/browse/ASTERISK-18930 so it's probably issue with srvlookup, but i'm not sur |
| 14:38.02 | seik0 | said, that domain lookup is synchronous, but... |
| 14:38.17 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 14:38.56 | qakhan | [TK]D-Fender i am use softphone on my system and i am still getting beep on page app |
| 14:39.23 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 14:39.25 | seik0 | but we have local dns server, so it seems to work even without internet connection (exceptions possible, but seem to be rare, m?) |
| 14:40.04 | WIMPy | seik0: You need to tell us what exactely is happening. |
| 14:41.01 | [TK]D-Fender | qakhan, And you aren't showing a complete call with SIP debug etc. |
| 14:41.17 | seik0 | WIMPy, exactly: when using sip registration and losing internet connection, then in short time we can't even register sip agent |
| 14:41.42 | *** join/#asterisk deo_ (~deo@112.198.82.15) |
| 14:41.47 | [TK]D-Fender | seik0, They suggested running a DNS caching proxy as an interim solution |
| 14:41.52 | WIMPy | seik0: That is you can't register TO Asterisk? |
| 14:42.04 | WIMPy | seik0: What is Asterisk telling you at that time? |
| 14:42.10 | seik0 | WIMPy exactly |
| 14:43.07 | seik0 | WIMPy it's not telling anyting in attempts to register. just nothing |
| 14:44.16 | seik0 | [TK]D-Fender, yes, i see it, but i wonder why dns server on local net can't help |
| 14:45.00 | [TK]D-Fender | seik0, It should. |
| 14:45.23 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
| 14:46.22 | seik0 | [TK]D-Fender, or asterisk doesn't use local dns, or problem is still is with local dns, or local dns couldn't resolve domain at that moment |
| 14:46.44 | [TK]D-Fender | seik0, DNS is based on your server itself, not *. |
| 14:46.50 | [TK]D-Fender | It uses whatever you configured locally |
| 14:47.07 | [TK]D-Fender | So I'd go check that you actually pointed your resolv.conf there |
| 14:47.11 | [TK]D-Fender | and test it |
| 14:47.22 | seik0 | resolv.conf is ok |
| 14:47.37 | [TK]D-Fender | go test |
| 14:53.08 | seik0 | without nameserver in resolv.conf it doesn't work, but we need to check how it works without internet connection, i think |
| 14:53.30 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
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| 14:57.29 | seik0 | [TK]D-Fender it surely works correctly now, ok, i further try to modele connection lost |
| 14:57.32 | seik0 | thanks |
| 15:02.36 | bombev | [TK]D-Fender can you take a look on this: http://pastebin.ca/2302621 |
| 15:03.00 | bombev | now if I call 105, and it works :) |
| 15:03.22 | [TK]D-Fender | bombev, exten => _10[0-8],1,Goto(ext-meetme,${EXTEN},1) <- stop shoving patterns in the middle of the definitions of other patterns. |
| 15:03.46 | [TK]D-Fender | bombev, exten => _XXX.,1,Playback(custom1/all-outgoing-lines) <- same here |
| 15:04.06 | [TK]D-Fender | bombev, You are going to end up breaking things like priority "n"'s real fast... |
| 15:04.34 | bombev | so I should use "n" |
| 15:04.49 | [TK]D-Fender | ....... |
| 15:05.00 | [TK]D-Fender | <PROTECTED> |
| 15:05.02 | bombev | dont get mad i am newbie |
| 15:05.11 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:05.54 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 15:07.09 | *** join/#asterisk PbxMan (c335d959@gateway/web/freenode/ip.195.53.217.89) |
| 15:07.45 | [TK]D-Fender | I told you what you were putting in the middle of places you shouldn't be doing it, and what it might break. You then took that as meaning you shouldn't use "n". This has nothing to do with being a "newbie" and more to do with reading what I just told you. You don't need to use "n", and you don't need to stop using it either. But you other bad habits are going to break things. Do your exten processing in order. |
| 15:09.40 | bombev | aha, what is the best possible order of that context |
| 15:17.58 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-73.lv.lv.cox.net) |
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| 15:23.01 | solitude88 | Hi guys I have a digium switchvox that I'm having trouble with would I be able to get help in the channel? |
| 15:23.41 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-gyhhcunwmwwkqfhm) |
| 15:23.54 | _Corey_ | solitude88: Even though Switchvox has Asterisk under its "hood", you'll need to contact Digium for support on it |
| 15:25.14 | solitude88 | _Corey_ is it possible that the errors could be an easy fix based on an asterix solution? Reason Im asking is Digium has paid support which I don't have an option to do right now |
| 15:25.20 | [TK]D-Fender | bombev, I'm not sure what part of "don't interrupt patterns with other patterns" is unclear.... |
| 15:25.27 | bombev | guys how to find the reason when I call from my extension to another one, when other side pick up the call, there is delay of 3 secs before we can hear each other |
| 15:25.59 | bombev | [TK]D-Fender i will read first the asterisk book thanks for helping me |
| 15:26.05 | [TK]D-Fender | solitude88, Depends. You haven't shown us your problem at all yet |
| 15:26.20 | [TK]D-Fender | bombev, You don't need to read the book to follow what I just said. |
| 15:26.29 | _Corey_ | solitude88: What kind of Switchvox subscriptions do you have? Silver/Gold, etc? You should have some access to support. I couldn't really answer your question as it stands though. |
| 15:26.53 | Katty | fat little cardinal on the feeder ^___^ all is right in the world today. |
| 15:26.57 | bombev | [TK]D-Fender how i interrupt those patterns with other patterns |
| 15:27.51 | [TK]D-Fender | bombev, <- What the fuck is this doing in the MIDDLE of your definition of the _XXX pattern? |
| 15:27.58 | [TK]D-Fender | dangit |
| 15:28.01 | [TK]D-Fender | bombev, http://pastebin.ca/2302635 |
| 15:28.47 | bombev | yes |
| 15:29.28 | [TK]D-Fender | bombev, http://pastebin.ca/2302636 |
| 15:29.34 | solitude88 | _Corey_ I have Silver support |
| 15:29.51 | _Corey_ | solitude88: You have e-mail access then |
| 15:29.58 | solitude88 | I did have paid support but expired towards the middle of the year |
| 15:29.58 | bombev | [TK]D-Fender |
| 15:30.08 | bombev | thanks now I got it |
| 15:30.14 | solitude88 | _Corey_ I can access this via digium? |
| 15:30.32 | solitude88 | do you have a link to support by chance |
| 15:30.40 | Qwell | digium.com/support |
| 15:30.45 | _Corey_ | solitude88: Well, if your license subscriptions have expired, probably not... |
| 15:32.31 | *** join/#asterisk angryuser (~Angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:34.06 | angryuser | Hello, can someone point me to the configuration manual of chan_dahdi.conf the one on the wiki is pretty old, i cant find all the variables, i am insterested in Pridialplan/ prilocaldialplan values and its behaviour, thank you |
| 15:34.55 | bombev | [TK]D-Fender do you have any idea about this: how to find the reason when I call from my extension to another one, when other side pick up the call, there is delay of 3 secs before we can hear each other |
| 15:34.56 | WIMPy | chan_dahdi.conf.sample |
| 15:34.58 | navaismo | angryuser, the original chan_dahdi has all the information |
| 15:35.36 | WIMPy | angryuser: And unless you are sure you need something else, set them to "unknown". |
| 15:36.37 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:36.37 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:38.26 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:38.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:43.00 | [TK]D-Fender | bombev, Slow RTP setup from your endpoints (, plus possible network issues. I recommend making sure that reinvites are disabled. |
| 15:43.53 | bombev | [TK]D-Fender how to disable the reinvites |
| 15:44.15 | [TK]D-Fender | go read the sip.conf sample |
| 15:44.26 | [TK]D-Fender | goes to move his computer to his new office |
| 15:46.54 | *** join/#asterisk benlangfeld (~Adium@unaffiliated/benlangfeld) |
| 15:47.14 | benlangfeld | Hey, I need a ticket reopening. Can someone do that for me? https://issues.asterisk.org/jira/browse/ASTERISK-18639#comment-201518 |
| 15:47.39 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:47.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:51.25 | Qwell | benlangfeld: #asterisk-bugs, but it's incredibly unlikely that anything will happen until 12. |
| 15:51.45 | benlangfeld | Thanks, cross-posting there |
| 15:52.08 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.206.253) |
| 15:55.07 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 15:57.04 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:57.04 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:59.02 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 15:59.43 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.206.253) |
| 16:07.02 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 16:10.24 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:14.54 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 16:15.05 | bombev | :) |
| 16:15.58 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 16:16.06 | jeev | is it possible my PRI card is having problems if i have something like this "Span 1: Channel 0/1 got hangup request, cause 16" mid ring group, it's already trying to call a third hunt attempt, what's happening is the call seems like it's hangingi up and going through the ring group process again, it is not following extensions_additional that has been configured through frepbx, it's going to |
| 16:16.06 | jeev | first attempt, then after that times out, second hunt, then third, immediately drops in the console and comes back as a new call.. then it complets the entire ring group sequence. |
| 16:16.51 | bombev | [TK]D-Fender that context work as charm: http://pastebin.ca/2302636 , when I place a call with that context, the other extension can not see the caller id or my extension, but see only "device" |
| 16:17.06 | bombev | what variable should I use to show my caller id in that context |
| 16:17.21 | WIMPy | jeev: That description doesn't make sense to me. But for FreePBX support go to #freepbx. |
| 16:17.44 | [TK]D-Fender | bombev, You don't CallerID comes from your device definition, not the dialplan |
| 16:18.10 | bombev | hm strange |
| 16:18.18 | jeev | WIMPy, i dont think it's a freepbx issue, i think this PRI card is going bad, call is coming in, going through the process of calling extensions, at some point it hangs up in asterisk and then goes through the process again without a problem. |
| 16:18.21 | bombev | because when i am using other context they can see my caller iD |
| 16:19.04 | WIMPy | jeev: The card doesn't hang up calls. Asterisk does that. |
| 16:19.51 | jeev | yea you know what, i just realized that dahdi wasn't running and incoming calls were coming in SIP while i was testing last night. |
| 16:23.09 | WIMPy | I wonder how you get a cause 16 that way. |
| 16:27.58 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-gayvxxkashhgzrkt) |
| 16:28.00 | raden | hugs Katty |
| 16:28.40 | *** part/#asterisk benlangfeld (~Adium@unaffiliated/benlangfeld) |
| 16:28.45 | jeev | WIMPy, excellent question. |
| 16:28.58 | jeev | i wonder the same thing |
| 16:29.05 | jeev | it's not giving me that error now though since dahdi broke. |
| 16:29.36 | Katty | hugs radic |
| 16:29.38 | Katty | oh |
| 16:29.39 | jeev | it is doing the same thing through, pretty much rings the extension in the pattern for a second, then looks like the call is hung up entirely, comes back again and goes through. |
| 16:29.41 | Katty | hugs raden too |
| 16:29.49 | radic | :o |
| 16:29.57 | raden | hi Katty how aare you ? |
| 16:29.57 | Katty | radic: free hugs for all! |
| 16:30.07 | Katty | raden: am goodly, you? |
| 16:30.15 | Katty | raden: watching the squirrels. |
| 16:30.29 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:30.40 | radic | Katty: there are 231 remaining :P |
| 16:31.06 | raden | got rid of the ice queen so yea ... doing better :P working on business plan to make big $$$$ this year ! |
| 16:31.32 | Katty | raden: i'm glad you're working towards a brighter, happier future (= |
| 16:31.36 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 16:31.52 | Merlin | does digium maintain res_speech_lumenvox, and is the source code available? |
| 16:32.01 | Qwell | We do, and it is not. |
| 16:32.18 | Merlin | is that some agreement with lumenbox? |
| 16:32.21 | Merlin | lumenvox |
| 16:32.22 | *** part/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 16:32.30 | Qwell | who knows |
| 16:32.45 | Merlin | haa |
| 16:33.06 | Merlin | any idea why lumenvox 10 requires asterisk 1.6.2 |
| 16:33.20 | raden | Katty, yea its all good :) the WISP is growing faster than I imagined , my electronics business is doing insanely well , and the company I work with in florida signing over 10% of the company to retain me with them :) I cant complain :) |
| 16:33.28 | Merlin | did 1.6.2 introduce some code hook they need? |
| 16:33.47 | Katty | raden: sounds like celebration is in order! |
| 16:34.11 | raden | Katty, (= |
| 16:34.28 | raden | \=D/ |
| 16:34.40 | Qwell | Merlin: What, are you trying to use something older than 1.6.2? |
| 16:35.14 | raden | going to try for 100 security camera installs this year ..... we did like 20 last year with no advertising of any sort most were 12 - 24 cameras a few upwards of 100 cameras |
| 16:35.22 | Merlin | Qwell: well fonality is still on 1.6.0, and my customer doesn't want to lose support |
| 16:35.34 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:35.38 | Qwell | rolls his eyes and walks away |
| 16:35.48 | Merlin | haha i knew that was coming |
| 16:36.13 | Qwell | My son, in 2nd grade, is older than the crap they use. |
| 16:36.23 | Qwell | wait |
| 16:36.25 | Qwell | younger |
| 16:36.26 | Qwell | that one |
| 16:36.48 | Merlin | they got rid of asterisk 1.2 |
| 16:36.53 | Merlin | we should be thankful for that |
| 16:38.40 | WIMPy | Smoke signs, jungle drums. What's better? |
| 16:40.08 | navaismo | jungle drums |
| 16:40.25 | [TK]D-Fender | WIMPy, Depends on the operating environment. |
| 16:41.07 | Qwell | smoke drums |
| 16:41.12 | WIMPy | should have know that this type of question will be take seriousely in here. |
| 16:44.10 | Merlin | mastedon horn drumsticks on a bald eagle hide drum |
| 16:45.50 | *** part/#asterisk c4software (~vbrosseau@46.255.52.118) |
| 16:47.27 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:50.46 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:a8a3:12ab:700f:bcfe) |
| 16:54.17 | *** join/#asterisk keith__ (~keith@udp278022uds.hawaiiantel.net) |
| 16:59.49 | *** part/#asterisk PbxMan (c335d959@gateway/web/freenode/ip.195.53.217.89) |
| 17:08.41 | ideaman55 | Anyone: I had an apt-get install of dahdi/libpri and Asterisk on a remote site. For an upgrade, I did a compiled 1.8, libpri and dahdi. Now after booting up, Asterisk doesn't load for about 2-3 minutes, along with dahdi_scan reporting back Unable to open /dev/dahdi/ctl: No such file or directory. However dahdi_hardware does show the card. I can't see any errors in dmesg with the card. The |
| 17:08.41 | ideaman55 | odd thing is that if I wait about 3 minutes, Asterisk is up, and dahdi_scan works just fine, then lsmod shows the module now. Any suggestions on what I can do to dig into where the confliction is for those first few minutes and fix? |
| 17:17.43 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 17:18.49 | ulogic | Does anybody have some tips on getting ./configure to recognize that gmime is installed? |
| 17:19.22 | Qwell | To where did you install it? |
| 17:20.35 | ulogic | I just unpacked gmime-2.6 into /usr/src, then just did a straight configure, make, and make install without any additional command line arguments |
| 17:23.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:26.54 | ulogic | the libraries for gmime are in /usr/local/lib/ |
| 17:27.03 | *** join/#asterisk curfont (~q@ytmd.ath.cx) |
| 17:27.33 | curfont | If you register a SIP trunk from two asterisk boxes, who gets the incoming? The last one who registered? |
| 17:28.27 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 17:28.39 | ulogic | asterisk configure uses pkg-config --exists --print-errors gmime-2.6 to see if it exists |
| 17:31.16 | Qwell | What does config.log say about it? |
| 17:31.18 | Qwell | ~pastebin |
| 17:31.18 | 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. |
| 17:31.57 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 17:32.06 | gavimobile | dear channel I missed you! |
| 17:32.26 | ulogic | config.log show the same thing as if I run the command directly (2 lines to follow) |
| 17:32.43 | ulogic | Package gmime-2.6 was not found in the pkg-config search path. |
| 17:32.53 | ulogic | Perhaps you should add the directory containing `gmime-2.6.pc' to the PKG_CONFIG_PATH environment variable |
| 17:32.55 | gavimobile | I have 1 remote peer which is a softphone on my iphone. lately it won't register with my server unless I restart my router where my pbx is |
| 17:33.21 | Qwell | and did you add it to the PKG_CONFIG_PATH env variable? |
| 17:33.24 | gavimobile | can anything think of a reason why this might be happening? |
| 17:33.30 | gavimobile | anyone* |
| 17:33.30 | ulogic | That file is in /usr/local/lib/pkgconfig/gmime-2.6.pc |
| 17:33.49 | Qwell | So that's a no then. |
| 17:34.11 | [TK]D-Fender | gavimobile, You've messed up your networking configs |
| 17:34.34 | ulogic | Do I need to edit configure to do that or just give it as a command line argument to ./configure ? |
| 17:34.38 | [TK]D-Fender | gavimobile, Or your router is just just messed up on it's own |
| 17:35.42 | gavimobile | [TK]D-Fender: where would I start to diagnose something like this? investing in a better router would be a good start. I always find things about my router I don't like! |
| 17:35.55 | Qwell | ulogic: Do what the message told you to do. |
| 17:36.14 | [TK]D-Fender | gavimobile, You could rty looking at your calls... your router config. your * config.... |
| 17:36.16 | gavimobile | [TK]D-Fender: btw, its only 1 remote extention |
| 17:36.27 | [TK]D-Fender | gavimobile, It isn't magic. |
| 17:37.09 | gavimobile | [TK]D-Fender: well in the past I needed to speak with the d-link technical support team and wasn't satisfied |
| 17:37.28 | gavimobile | their service over here is inadequate |
| 17:37.35 | [TK]D-Fender | gavimobile, ok/fine/sure |
| 17:38.20 | gavimobile | how would looking at my calls help me? |
| 17:39.00 | ulogic | I issued PKG_CONFIG_PATH=/usr/local/lib/pkgconfig/ followed by export PKG_CONFIG_PATH and it now works. Thanks |
| 17:40.14 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 17:40.29 | [TK]D-Fender | gavimobile, How is sitting here NOT looking at what's actually happening working out for you? |
| 17:43.17 | Katty | [TK]D-Fender: manners, sir. |
| 17:43.22 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 17:43.22 | Katty | [TK]D-Fender: you could at least be polite. |
| 17:43.51 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:44.29 | [TK]D-Fender | Katty, I'm perfectly polite about it. I'm concerned for his mental well-being ;) |
| 17:44.47 | Katty | frowns. |
| 17:44.51 | Katty | no, you're frustrated and annoyed. |
| 17:45.09 | Katty | but they're even more frustrated than you are, so be nice. |
| 17:45.15 | [TK]D-Fender | My calls don't work. Why should I look at my calls?!? <- could be a sign of a stroke. |
| 17:45.23 | [TK]D-Fender | Remember to watch for the early signs! |
| 17:45.46 | [TK]D-Fender | #themoreyouknow |
| 17:46.37 | Katty | careful that you don't turn people off from the asterisk project. |
| 17:47.14 | Merlin | the project already has too many people |
| 17:47.35 | Merlin | china already implemented a one-developer policy |
| 17:47.52 | [TK]D-Fender | lol |
| 17:50.10 | gavimobile | [TK]D-Fender: the problem isn't with the call. the problem is it doesn't register with my server until I rebout my router |
| 17:50.36 | SuperNull | anyone know of a way to verify jitter buffer length ? (in ms) |
| 17:51.26 | [TK]D-Fender | gavimobile, Same thing |
| 17:52.32 | gavimobile | well sip debug doesn't show anything |
| 17:52.46 | gavimobile | the only time it shows is once the registration happenes |
| 17:53.17 | [TK]D-Fender | AS it happens, not "once" it happens. |
| 17:53.50 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:53.50 | [TK]D-Fender | Well when you decide you have something you'd like me to look at, let me know. |
| 17:54.04 | Merlin | gavimobile: how about tcpdump before you reboot? |
| 17:57.10 | gavimobile | Merlin: that sounds like the direction I need to go |
| 17:57.31 | gavimobile | but that's for my technical support to break their heads with not me |
| 17:57.37 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 17:59.08 | igcewieling | I have a sip type=peer with a host= line, deny 0.0.0.0/0.0.0.0 and a permit 192.168.1.0/255.255.255.0. I thought outbound calls would use the host= and inbound would use the permit/deny. However, with this setup Asterisk rejects calls from any IP except for the one in the host= line. does anyone have any suggestions on accepting calls from all IPs in the permit= range? |
| 17:59.48 | WIMPy | type=peer matches by host |
| 18:00.12 | igcewieling | WIMPy: should I use type=friend? |
| 18:00.39 | igcewieling | previously we only sent calls from the ip in the host= line, so this is a bit new to me. |
| 18:00.57 | WIMPy | Do you have username/password or do you want to accept anything? |
| 18:01.55 | igcewieling | no. We are auth based on IP. Changing to friend made no difference. |
| 18:02.51 | [TK]D-Fender | igcewieling, What is this for? |
| 18:03.12 | WIMPy | Not sure there's a short cut for generating a peer per IP. |
| 18:03.22 | [TK]D-Fender | There isn't |
| 18:04.12 | igcewieling | [TK]D-Fender: we have a cluster of asterisk servers which will be sending calls. |
| 18:04.34 | igcewieling | I suppose I could allow access from ALL IPs and block at the iptables level |
| 18:04.53 | igcewieling | So, exactly what is permit/deny used for. Looks to me like they are not used for anything. |
| 18:05.02 | [TK]D-Fender | igcewieling, How are you attempting to auth them? You can't make a peer to account for mutliple IP's like that. you could use a "user" and give them all the same name for incoming though |
| 18:05.14 | WIMPy | You might be able to use permit/demy with allowguests. |
| 18:05.15 | igcewieling | [TK]D-Fender: we are authing by IP |
| 18:05.16 | Qwell | igcewieling: order matters |
| 18:05.28 | [TK]D-Fender | igcewieling, It is... it's used to restrict where a device can REGISTER from. |
| 18:05.30 | igcewieling | Qwell: deny first, then permit? |
| 18:05.41 | Qwell | oh, that. hang on |
| 18:05.45 | [TK]D-Fender | igcewieling, No, host is a 100% restriction. Dead End |
| 18:05.57 | [TK]D-Fender | igcewieling, you'll have to make multiple peers. |
| 18:06.02 | igcewieling | I even tried it with no host line. |
| 18:06.12 | [TK]D-Fender | igcewieling, You need something in the host. |
| 18:06.13 | Qwell | contactpermit/contactdeny is for register |
| 18:06.18 | igcewieling | Asterisk seems to sweet, then she pushed you in front of truck. |
| 18:06.23 | [TK]D-Fender | igcewieling, Either IP or dynamic. There is no "none" |
| 18:06.31 | WIMPy | What happens with host=dynamic? |
| 18:07.01 | Qwell | WIMPy: contactpermit/contactdeny |
| 18:07.14 | igcewieling | WIMP Dialing as SIP/peer/exten/host (new in 1.8) breaks, but I've not tried it since I updated to the latest asterisk. |
| 18:07.21 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 18:07.37 | igcewieling | Qwell: can you think of any solution to my problem which does not involve creating a new peer on 56 systems every time we add a new voice GW? |
| 18:07.54 | Qwell | igcewieling: no, tl;dr, I just saw permit/deny failing |
| 18:08.09 | SuperNull | i was doing a tcpdump on this ast server.. and some how managed to lose a packet outbound .. .. how the heck.. ? literally capturing off the local machine and it lost something |
| 18:08.40 | igcewieling | I HAVE outbound dialing using SRV (with priorities and random weights just like the RFC) using AEL. This is my one known remaining issue |
| 18:09.00 | igcewieling | Qwell: do you have any ideas on how to make asterisk accept connections from more than one IP using only one entry in sip.conf? |
| 18:10.26 | Qwell | SRV |
| 18:10.38 | Qwell | skip IPs altogether |
| 18:10.46 | igcewieling | Qwell: How do you mean? Asterisk's SRV support is totally broken. |
| 18:10.56 | igcewieling | Qwell: I'm not following you. |
| 18:11.06 | Qwell | What's the issue #? |
| 18:11.31 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 18:11.34 | igcewieling | Qwell: since forever. SRV only using the first host returned, not failing over, not supporting weights, etc. |
| 18:13.12 | igcewieling | Qwell: Are you saying that if _sip._udp.domain.com has three hosts in SRV records, asterisk will accept calls from all three hosts? That would solve virtually every issue I'm having. |
| 18:13.32 | igcewieling | assuming host=domain.com of course. |
| 18:16.01 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 18:28.13 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 18:30.12 | igcewieling | "If srvlookup is turned on, Asterisk supports DNS SRV lookups partially. Currently, Asterisk only reads the first SRV entry without bothering with priorities and weights. " |
| 18:30.31 | igcewieling | THAT is what I'm taking about when I say "SRV is broken on Asterisk" |
| 18:33.33 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:44.44 | Katty | what a gloomy cold day. |
| 18:45.06 | Katty | i guess it makes the arrival of spring all the more awesome |
| 18:46.03 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 18:47.02 | talntid | anyone know of a feature list of asterisk, and when the feature was implemented? like a grid or something? just looking for an easy way to see if there are new useful things in asterisk I can implement :) |
| 18:47.27 | Qwell | CHANGES.txt |
| 18:48.58 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.72.33) |
| 18:50.26 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:50.54 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 18:56.40 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:57.24 | mjordan | talntid: since 1.8, we've tried to get better about putting things up on the wiki as well. There's always room for improvement, but you can find info about each version and the new features on the wiki as well |
| 18:57.39 | mjordan | https://wiki.asterisk.org/wiki/display/AST/New+in+11 |
| 18:57.49 | igcewieling | Qwell: setting host=oursrvhost.mydomain.com did not allow incoming calls. |
| 18:58.44 | igcewieling | Qwell: any specific change. I don't see anything which says "fixed broken SRV priority and weights" |
| 18:59.23 | igcewieling | I see some dnsmgr changes and the addition of SRVLOOKIP and some IAX related SRV changes. |
| 19:03.17 | mjordan | igcewieling: based on my read through, I think the article is wrong. It *appears* as if we will honor weights/priorities. It is correct however that we only track a single srv entry |
| 19:04.09 | igcewieling | mjordan: that is actually good, means I didn't waste my weekend writing real support in AEL (which was somewhat interesting doing all the weights, etc). |
| 19:04.53 | igcewieling | mjordan: My only issue now is trying automatically accept calls from a block of IPs (ip auth, no username) without creating a peer for each one. Do you have any suggestions? |
| 19:05.16 | mjordan | igcewieling: the weight stuff is a bit interesting. There's even a comment in the source that says "/* Do the bizarre SRV record weight-handling algorithm involving sorting and random number generation... See RFC 2782 if you want know why this code does this" |
| 19:05.26 | mjordan | igcewieling: nope, that isn't implemented |
| 19:05.53 | mjordan | it goes beyond just DNS and SRV records - channel drivers have to have the semantics of understanding and using multiple records |
| 19:05.56 | igcewieling | insecure=very turns off all auth, correct. |
| 19:06.06 | mjordan | it's a good feature request, but no one has implemented it yet |
| 19:06.17 | *** join/#asterisk j4m3s (~j4m3s@pdpc/supporter/active/j4m3s) |
| 19:06.20 | igcewieling | mjordan: dialing out using SRV is a problem I have solved. It is the inbound I need to get working |
| 19:06.38 | j4m3s | is it possible to set a peer's registration expiry? |
| 19:06.54 | WIMPy | I'm back to basics once again and somehow stuck. I don't see a way to get the callerid ot someone doing a blind transfer. Am I blind or is it just not possible? |
| 19:07.01 | igcewieling | mjordan: the issue has existed since srv support was initially added in 1.2 or 1.4, I have no illusion it will ever be fixed. |
| 19:07.09 | mjordan | yup. Most people who are working around this limitation use insecure and no registration. I would say it is a work around and not a proper solution obviously, but getting the inbound stuff to work is a pretty large feature request. |
| 19:07.13 | WIMPy | s/ot/of/ |
| 19:07.15 | mjordan | shrugs |
| 19:07.30 | mjordan | you never know. The developer community does do some pretty awesome work :-) |
| 19:07.59 | igcewieling | mjordan: would using insecure=very (or whatever the 1.8 version is) with permit/deny allow access only from the hosts in the permit= range? |
| 19:08.44 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 19:09.36 | [TK]D-Fender | igcewieling, Proxy time <- |
| 19:09.43 | igcewieling | [TK]D-Fender: hush. |
| 19:09.53 | j4m3s | Is it possible in 1.4 to set a peer's registration expiry? |
| 19:10.04 | mjordan | it should. IIRC, ACLs are applied separately from the insecure settings. |
| 19:10.08 | Qwell | ~upgrade asterisk |
| 19:10.08 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:10.09 | igcewieling | we tried the proxy route, ended up sending all calls to asterisk for routing anyway. |
| 19:10.10 | Qwell | j4m3s: ^ |
| 19:10.56 | j4m3s | Qwell, lol this customer ain't gonna want to pay for that |
| 19:11.01 | igcewieling | mjordan: I'll give it a try when we have lower call volume. |
| 19:11.30 | j4m3s | Qwell, but did that functionality get introduced in a later version of *? |
| 19:11.42 | mjordan | igcewieling: let me know how it works... the named ACL feature in 11 would potentially help a lot there too |
| 19:12.15 | igcewieling | mjordan: *nod* That will be helpful when we move to 11 in a few years |
| 19:13.46 | igcewieling | mjordan: just tested, still rejected the call |
| 19:14.13 | mjordan | hm. |
| 19:14.20 | mjordan | what version? |
| 19:14.31 | igcewieling | I guess we'll preconfigure all 58 boxes with each potential future server. |
| 19:14.41 | igcewieling | mjordan: 1.8.20-rc-something |
| 19:14.57 | mjordan | k. There aren't huge differences between 1.8 and 11 in that area. |
| 19:15.00 | file | it only matches based on the value of host, the ACL doesn't influence that - ACLs are more for dynamic configurations |
| 19:15.16 | igcewieling | file: is there any way to make it NOT match on host? |
| 19:15.39 | file | it's host, user/pass authentication, or strictly user matching without authentication |
| 19:16.29 | igcewieling | file, what I'm looking for is for all calls from devices within the permit range to be allowed with no other authenication. |
| 19:17.12 | file | there is no present ability to do that, short of allowing anonymous authentication and then dialplan logic to examine the received IP |
| 19:17.12 | igcewieling | I can do user auth with no passwords if the same user can be in multiple peers |
| 19:17.44 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 19:17.46 | file | a user can only match against a single entry in sip.conf, but you can specify that an outgoing INVITE authenticate as the same user in multiple peer entries |
| 19:18.16 | igcewieling | file, looks to me that asterisk cannot accept calls for a peer from more than one host no matter what you do. Is that correct? |
| 19:18.28 | file | correct. |
| 19:18.44 | igcewieling | file: I have a solution for the outgoing, it is incoming I'm having issues with. |
| 19:18.51 | file | although there's nothing to say you can't decrease the amount of work required to make multiple peers with different IPs using templates |
| 19:19.03 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.72.33) |
| 19:19.12 | igcewieling | file: these are all servers we would like to keep the config in the GUI. |
| 19:19.23 | WIMPy | It's an issue everyone has, that's using an ITSP with lots of servers. |
| 19:19.59 | igcewieling | WIMPy: surprising after all these years it is still an issue. |
| 19:20.27 | WIMPy | I think there are worse ones than that. |
| 19:21.25 | WIMPy | The one I just mentioned about blind transfers for example. |
| 19:21.33 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:22.05 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:22.44 | WIMPy | I guess, I could do some AMI magic, though. |
| 19:23.07 | igcewieling | WIMPy: makes no sense. The only time callerid is passed on a transfer is when it is a blind transfer. Attended transfers don't pass the CallerID. Have you tried the "o" option to Dial? |
| 19:23.32 | WIMPy | No, Attended transfers are fine. |
| 19:23.54 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 19:24.14 | danfromuk | Whats the command to hangup a channel from the CLI in 1.8? |
| 19:24.15 | WIMPy | And I need the transferrers caller ID in the dialplan as well so I can send the call back there if it isn't answered. |
| 19:24.25 | WIMPy | channel request hangup |
| 19:25.14 | danfromuk | Thanks. I need to make a note of that! |
| 19:25.45 | *** join/#asterisk ghghz (~ton@kluonis.kvb.lt) |
| 19:26.45 | igcewieling | WIMPy: how are you doing the transfers, phone based or asterisk (dtmf) based? |
| 19:26.49 | ghghz | Hello. Is it possible to know if signal was reached destination? |
| 19:27.04 | WIMPy | On the phone. |
| 19:27.20 | igcewieling | danfromuk: there is a cli_aliases.conf.sample in the Asterisk source code with common aliases to make the transition from the old to the new asterisk easier |
| 19:27.27 | igcewieling | WIMPy: phone brand/ |
| 19:27.44 | WIMPy | any |
| 19:29.25 | WIMPy | I'm curently using a Digium phone, but this is an Asterisk thing, not a phone thing. |
| 19:31.08 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 19:32.57 | WIMPy | Even the caller ID would be some half thing. I'd really know the exact phone, but that doesn't work anyway. |
| 19:33.56 | WIMPy | err. |
| 19:34.02 | igcewieling | we've never had that problem. Original CallerID is passed on blind transfers, not passed on attended transfers. |
| 19:34.03 | ghghz | Hello. Is it possible to know if signal was reached destination? |
| 19:34.04 | WIMPy | should use |
| 19:34.31 | igcewieling | ghghz: the destination will reply with an ACK |
| 19:34.35 | WIMPy | Yes, I get that, but I don't get the transferrers caller ID. |
| 19:34.56 | WIMPy | ghghz: What signal? To where? |
| 19:35.23 | ghghz | Wait, I will explain with example |
| 19:35.52 | igcewieling | WIMPy: you won't it is one or the other, though you could infer the callerid from the peer which made the call, can't you? |
| 19:36.44 | WIMPy | Not really, no. |
| 19:36.47 | SuperNull | hey guys anyone know why a rtp analysis taken directly off a server would show local ip -> outward as lost packets? like.. how do you lose rtp packets at the box they came from |
| 19:37.01 | Qwell | SuperNull: have an example? |
| 19:37.02 | WIMPy | And I definitely want both. |
| 19:38.23 | *** join/#asterisk cyborg-one (~cyborg-on@130-0-33-92.broadband.tenet.odessa.ua) |
| 19:38.41 | igcewieling | something like ${SIPPEER(${CHANNEL(peername)},callerid)} |
| 19:38.59 | WIMPy | Will only work for sip. |
| 19:39.05 | igcewieling | correct. |
| 19:39.24 | SuperNull | Qwell i do have a large example.. |
| 19:39.25 | WIMPy | And I need the peername first. |
| 19:39.26 | SuperNull | butttt |
| 19:39.41 | SuperNull | its got legit customer call audio on it for a lawyers firm .. which could contain actual customers BS. |
| 19:39.44 | Qwell | What is showing it as "lost"? |
| 19:40.01 | SuperNull | tcpdumped to pcap.. dumped into wireshark to do analysis |
| 19:40.06 | WIMPy | Which is currently part of the channel name for sip, but I wouldn't want to assume it will always stay that way. |
| 19:40.29 | SuperNull | if i do 'rtp stream analysis' or on all streams.. i get lost from the machine it self.. |
| 19:42.08 | SuperNull | one sec i can screen shot ya |
| 19:43.27 | Qwell | forget the stream analysis. are there packets there? |
| 19:43.51 | SuperNull | Qwell : http://i.imgur.com/6SFXw.jpg is screen of the analysis.. 111.196 is the server and 110.244 is an Adtran TA924 |
| 19:44.00 | SuperNull | ... 'packets there' ? |
| 19:44.08 | SuperNull | uhm. there certainly are packets ? |
| 19:44.19 | Qwell | In the capture. Do you see packets going out? |
| 19:44.38 | SuperNull | sure ? |
| 19:44.44 | Qwell | So then what's the problem? |
| 19:45.00 | SuperNull | out of no where it jumps sequence number .. indicating lost packets ? |
| 19:45.31 | Qwell | Show me that it's skipping a sequence number on an outgoing packet. |
| 19:46.56 | ghghz | igcewieling: Look at this call file http://p.defau.lt/?v7I1rcrBHxBwf6jqTf8hdA |
| 19:47.07 | SuperNull | one sec Qwell |
| 19:47.14 | ghghz | is it possible to know if signal was reached destination? |
| 19:48.40 | WIMPy | If you want to get a result from an originate, either use AMI, look at your CDRs or write some information from the dialplan when the call ends. |
| 19:52.26 | SuperNull | Qwell: http://i.imgur.com/TqDl8.jpg |
| 19:52.43 | Qwell | <Qwell> forget the stream analysis |
| 19:53.01 | SuperNull | so your expecting me to manually go through each RTP packet ? |
| 19:53.09 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 19:54.08 | SuperNull | Qwell what are you trying to have me do here .. im lost |
| 19:54.23 | [TK]D-Fender | ghghz, And what is it that you're dialing there? |
| 19:54.48 | ghghz | WIMPy: I just want to make a lighthouse |
| 19:55.09 | [TK]D-Fender | ...? |
| 19:55.44 | [TK]D-Fender | <ghghz> Hello. Is it possible to know if signal was reached destination? <- When? How? |
| 19:55.52 | WIMPy | What? |
| 19:56.18 | ghghz | [TK]D-Fender: I want to make a short call, just one signal and hangup |
| 19:56.30 | ghghz | but need to know if that one signal was ringed. |
| 19:56.53 | [TK]D-Fender | ghghz, First, never use the term "signal". It's vague and doesn't tell us anything specific. Second, in your call-file what precisely are you calling? |
| 19:57.06 | WIMPy | You can only do that via AMI and if the channel you're dialling supports it. |
| 19:57.29 | ghghz | WIMPy: AMI you mean what? |
| 19:57.41 | ghghz | http://marcelog.github.com/articles/php_asterisk_manager_interface_protocol_tutorial_introduction.html ? |
| 19:57.42 | [TK]D-Fender | ghghz, Again, WHEN do you need to know this? HOW do you want to do that check? |
| 19:57.45 | WIMPy | ~ami |
| 19:57.45 | infobot | AMI is the Asterisk Manager Interface, a way to control an Asterisk server (and retrieve information) via a TCP/IP socket. More information is available at http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html and http://voip-info.org/wiki/view/Asterisk+manager+API |
| 19:58.37 | ghghz | [TK]D-Fender: I need to know as quick as possible. How I want, I don't know, maybe asterisk returns some result? |
| 19:58.41 | ghghz | dunno |
| 20:01.00 | [TK]D-Fender | ghghz, That doesn't tell us what is going to look at this "result". We don't know what tools it has at its disposal |
| 20:01.17 | [TK]D-Fender | ghghz, Please answer my question ato to precisely what it is you are dialing in that call-file. |
| 20:01.23 | [TK]D-Fender | as to* |
| 20:02.27 | ghghz | [TK]D-Fender: there should be SIP/trunk/destination-number |
| 20:03.10 | [TK]D-Fender | ghghz, There probably should ... but that's not what we see. |
| 20:03.23 | [TK]D-Fender | ghghz, So at least you seem to see this is probably wrong as-written |
| 20:03.50 | [TK]D-Fender | ghghz, So you still have to figure out what your process to validate who got answered should look like. |
| 20:04.14 | [TK]D-Fender | ghghz, And you should probably consider how many calls you'll be generating in the end, at what speed, etc. |
| 20:04.38 | ghghz | I don't need status if it's answered. I need status, if one ring was made on destination |
| 20:07.08 | [TK]D-Fender | ghghz, You can count ringing time at best, not a "quantity" |
| 20:07.28 | WIMPy | >>You can only do that via AMI and if the channel you're dialling supports it. |
| 20:07.46 | [TK]D-Fender | ghghz, No, what are you using to monitor this? how real-time is this process? Is this something you'll run like a report only after placing a number of calls? |
| 20:08.17 | [TK]D-Fender | WIMPy, AMI isn't the only answer. He has not been specific about how he needs to collect the info or analyse it. |
| 20:08.55 | WIMPy | As I understood it when it happens. |
| 20:09.21 | ghghz | yes |
| 20:09.37 | ghghz | instantly |
| 20:09.38 | [TK]D-Fender | WIMPy, He's been dangerously vague from the start and you've seem my repeat attempts at clarification. I'm sure you don't have to imagine my trut & assumption levels right now... |
| 20:09.44 | [TK]D-Fender | ghzBy what? |
| 20:10.06 | [TK]D-Fender | ghghz, By what? What is doing this lisening in? What will it do with that information? |
| 20:10.36 | WIMPy | He obviousely wants to do ping calls, even if he didn;t use that term. |
| 20:10.56 | ghghz | [TK]D-Fender: yes, WIMPy has understood me correctly :) |
| 20:11.03 | [TK]D-Fender | as in? |
| 20:11.34 | Qwell | verifying his telemarking list |
| 20:11.45 | Qwell | telemarketing, too |
| 20:11.49 | [TK]D-Fender | Spam to validate a spam list |
| 20:11.57 | [TK]D-Fender | Classic asshole move :) |
| 20:11.58 | Qwell | pretty much |
| 20:12.15 | _Corey_ | I had a customer attempt to patent that |
| 20:12.17 | _Corey_ | (no joke) |
| 20:12.24 | WIMPy | Or use a 900 callerid and hope some idiot calls back. |
| 20:12.33 | WIMPy | nice |
| 20:12.49 | Qwell | WIMPy: or any of the various island country NPAs |
| 20:13.08 | [TK]D-Fender | ghghz, You still haven't confirmed what you are doing "instantly" with this information. |
| 20:13.21 | WIMPy | hang up |
| 20:13.27 | [TK]D-Fender | ... |
| 20:13.54 | [TK]D-Fender | No, that act is happening regardless. He wants NOTIFICATIOn of the result of the attempt "instantly". |
| 20:13.59 | [TK]D-Fender | THAt is what I'm asking about. |
| 20:14.16 | [TK]D-Fender | What doe he do with the result of the attempt?\ |
| 20:14.16 | WIMPy | That's not what I read. |
| 20:14.25 | Katty | looks at Qwell |
| 20:14.28 | [TK]D-Fender | Does that have to be a "live" reaction with real processing? |
| 20:14.37 | Qwell | stares at Katty |
| 20:14.50 | ghghz | [TK]D-Fender: just putting notification to database |
| 20:15.05 | [TK]D-Fender | WIMPy, If he wants to know if it rings once, then that';s just a dial with about 3-4 seconds of dial time. No need to monitor to get that answer. |
| 20:15.26 | [TK]D-Fender | ghghz, then you DON'T need it "immediately" and DON'T need a monitoring process. |
| 20:15.42 | ghghz | [TK]D-Fender: sometimes it doesn't ring, that's a key |
| 20:15.48 | ghghz | sometimes OK |
| 20:15.52 | [TK]D-Fender | This can all be done with boring dialplan logic, no AMI or other nonsense. |
| 20:16.19 | WIMPy | No, he wants to know when it rigs so he can hang up then. |
| 20:16.35 | [TK]D-Fender | WIMPy, No, he wants to know that it rings to validate the number |
| 20:16.52 | [TK]D-Fender | Hanging up if it DOEWS because he doesn't actually want to TALK to anyone or play any message |
| 20:17.22 | [TK]D-Fender | Validate that it rings. Not "keep ringing or do more stuff", just "does it ring?" |
| 20:18.07 | [TK]D-Fender | Dial for 3 seconds. If you hit timeout, then it's good. If you don't get an error code, then it's good. If it answer, then it's good. |
| 20:18.14 | [TK]D-Fender | All boring dialplan. |
| 20:18.36 | ghghz | Understand |
| 20:23.24 | WIMPy | Just that you don't know how long it takes until it starts to ring. |
| 20:25.05 | [TK]D-Fender | WIMPy, Depends on how confident you feel on call setup delays. |
| 20:25.22 | [TK]D-Fender | WIMPy, that could probably be narrowed down to a reasonable level. |
| 20:25.33 | WIMPy | Since the invention of mobile phones and VOIP round about not at all. |
| 20:25.54 | [TK]D-Fender | WIMPy, Also the telco would have to report back status promptly, etc |
| 20:26.12 | WIMPy | Yes |
| 20:26.22 | [TK]D-Fender | WIMPy, And imperfect world. Then again he might as well ring for more than jsut a few seconds and hang up them anyway. |
| 20:26.48 | [TK]D-Fender | If you're going to piss people off well .... don't know what line will really get crossed one way or the other |
| 20:27.27 | navaismo | what if he use an ISDN to validate the numbers, without dialing the target customer? Still annoying but only for validated numbers and the DB are cleaned with real active numbers |
| 20:27.45 | WIMPy | If you only want to validate a number you usually use a data service, but that's a no go when using SIP. |
| 20:28.24 | WIMPy | navaismo: dial without dialing? |
| 20:29.14 | navaismo | you dial for a short time, only to get response for the telco about the number, but not to dial to the customer |
| 20:30.31 | WIMPy | 1. you have to be really fast to do that and 2. it might fail of there's a PBX at the called end. |
| 20:31.29 | navaismo | yes time between 20 -50ms |
| 20:31.43 | [TK]D-Fender | WIMPy, Fil? Depends on your point of view. Answering = confirmation to and has to be accounted for. |
| 20:31.58 | [TK]D-Fender | He just wants confirmation of validity as quickly as possible. |
| 20:32.09 | igcewieling | there are a number of commercial phone number validation services which telemarketers, and other scum use. |
| 20:32.10 | [TK]D-Fender | If it answers instantly then so be it. |
| 20:32.57 | navaismo | in the other hand, dont know if SIP headers can be handled by dialplan, if so then in a loop he can looking for a trying or ringing response and hangup immediately |
| 20:33.24 | WIMPy | I think navaismo was just going to wait for a proceeding message and that doen't neccessarily mean the number exists. |
| 20:33.48 | WIMPy | That's why I said AMI. |
| 20:34.08 | navaismo | brb |
| 20:34.14 | igcewieling | When dial is running you can't do anything else. |
| 20:34.41 | igcewieling | you best bet is enable sip debug for the peer you are dialing out from and tail the asterisk log file to see the messages |
| 20:35.09 | WIMPy | Again: AMI - the cure for everything |
| 20:35.16 | WIMPy | (that can be cured with Asterisk) |
| 20:41.16 | SuperNull | anyone use 'voipmonitor?' not looking for help or anything just a review of it |
| 20:42.08 | talntid | looks neat |
| 20:46.15 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
| 20:48.13 | SuperNull | voipmonitor looks pretty sweet.. pcap saving of all calls.. that is sexy. |
| 20:50.53 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 20:54.39 | *** part/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 21:07.46 | *** join/#asterisk rbd (6245a223@gateway/web/freenode/ip.98.69.162.35) |
| 21:08.47 | rbd | hey guys... is the asterisk manager api meant for high performance applications (e.g. we may have a few hundred connections per minute per asterisk server made over the manager api for making outbound calls) |
| 21:10.04 | WIMPy | For doing what? |
| 21:10.22 | rbd | they will make a call into a remote IVR for callflow testing |
| 21:10.45 | WIMPy | What do you want to do via AMI? |
| 21:12.02 | rbd | issue the dial commands, and the other commands as well to do the interation with the remote dialplan... as an alternative to using .call files and doing something in extensions.conf |
| 21:12.25 | talntid | any idea what voipmonitor costs, SuperNull? |
| 21:12.45 | *** join/#asterisk fibres (~no@5acee3dd.bb.sky.com) |
| 21:12.49 | SuperNull | free. |
| 21:12.51 | SuperNull | orrr |
| 21:12.53 | fibres | Evening people. |
| 21:12.56 | SuperNull | i dunno about the commercial one |
| 21:12.59 | WIMPy | Sounds reasonable. |
| 21:13.17 | talntid | it's "free" but I don't think the web interface is free. |
| 21:13.49 | SuperNull | thats the thing.. |
| 21:13.49 | SuperNull | but |
| 21:13.57 | SuperNull | we should be 'LAMP' experts.. right ;) |
| 21:14.21 | talntid | sure, someone could write another front-end, but I'd be more curious what it costs :) |
| 21:14.33 | fibres | Wondering if anyone can help me. I am having an issue with asterisk 1.8 in that it has a delay in closing channels. I am using sipp to debug and make test calls. I see the calls finish on sipp, I see asterisk respond to the bye message sipp sends, however it takes a few seconds after that for asterisk to clear the channel. when I do sip show channels I see more than should be open. |
| 21:14.34 | talntid | looks like they did a great job |
| 21:15.47 | WIMPy | fibres: That's normal. That's sip dialogs, not Asterisk channels you list there. |
| 21:16.31 | fibres | Hi wimpy. Ok what is the reason for the delay in dropping the dialogs? I dont see this behaviour in 1.4 |
| 21:17.59 | WIMPy | Probably to be able to detect duplicates. Some chan_sip guru will surely know better. |
| 21:18.37 | SuperNull | telntid i think the ideal deploy of this for existing would be using a centralized server for all other sip/audio servers (if your a multi sip server environment) |
| 21:18.51 | fibres | I am wondering if this is responsible for a major decrease in throughput on any asterisk version above 1.4 |
| 21:19.49 | fibres | On 1.4 I can handle over 80cps with 1200 concurrent calls. on 1.6 and above I have problems after a short time when I go over 20cps and 300 concurrent. |
| 21:22.22 | fibres | After a short time of sending calls I start to see __sip_autodestruct: Autodestruct on dialog '941-3642@xxx.xxx.xxx.xxx' with owner in place (Method: BYE) |
| 21:32.12 | [TK]D-Fender | <igcewieling> When dial is running you can't do anything else. <- no need to. These are one-shot validations and can be run in parallel. |
| 21:32.17 | [TK]D-Fender | And... it's checkout time.... |
| 21:48.46 | *** join/#asterisk nunne (~nunne@c-70f0e355.021-109-73746f46.cust.bredbandsbolaget.se) |
| 21:51.59 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.247.160) |
| 22:06.20 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 22:08.24 | jpsharp | Is there a hook inside the SIP registration code to do a dynamic update of an e.164 record? |
| 22:14.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:22.13 | pabelanger | jpsharp, no |
| 22:22.22 | pabelanger | unless you write something vi AMI |
| 22:28.58 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 22:43.38 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-187-53.red.bezeqint.net) |
| 22:43.44 | curfont | iax between 10.3 and 10.5 says "Unable to negotiate codec" |
| 22:43.50 | curfont | even though both are configured for ulaw |
| 22:43.57 | curfont | two other 10.5 with the same config work |
| 22:44.05 | curfont | hmm :/ |
| 22:46.39 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:48.32 | igcewieling | curfont: often when the happens the incoming call is not matching a peer in iax.conf and so is using the settings under [default]. |
| 22:48.51 | igcewieling | try adding the ulaw codec to your [general] section |
| 22:49.01 | *** part/#asterisk tamiel (~tamiel@208.66.27.62) |
| 22:50.15 | curfont | igcewieling: already done, 3 boxes are using the same exact general config (copy paste) |
| 22:50.19 | curfont | deny all and allow ulaw |
| 22:50.25 | curfont | i even see in the debug its using ulaw |
| 22:50.30 | curfont | the two 10.5 work between them |
| 22:50.35 | curfont | the 10.3 doesnt work with them though |
| 22:50.53 | igcewieling | you have one of those "non-obvious fix" problems. best of luck. |
| 22:51.06 | curfont | i am trying to put 10.5 on the 10.3 box, hopefully this is it :( |
| 22:53.18 | curfont | 10.3 and 10.5 have some big differences in the console, 10.5 forces you to use the "core" syntax |
| 22:53.29 | curfont | hmm |
| 23:01.20 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.58 | curfont | mp3 support isnt by default right? you need to compile it in through the menu config? |
| 23:02.52 | keith__ | yes, and you will also have to run contrib/scripts/get_mp3_source.sh before the make |
| 23:05.18 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 23:14.08 | curfont | ok so it wasnt what i thought |
| 23:14.19 | curfont | apparently its trying to use GSM, and i havent enabled GSM anywhere |
| 23:19.05 | *** join/#asterisk tzica (~na@unaffiliated/tzica) |
| 23:19.33 | tzica | using AsteriskNOW - if I want to backup asterisknow folder/files where are these located ? |
| 23:21.07 | tzica | I want to create a backup configuration script |
| 23:26.21 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 23:38.39 | *** join/#asterisk Schmee (~zaphod@ppp100-124.static.internode.on.net) |
| 23:40.49 | Schmee | hi all. I have a relatively simple asterisk 1.8 setup, involving a single SIP triunk and a bunch of Cisco 79xx handsets. I'm not using FreePBX or any other configuration helpers and it's working fine for the moment. However, I have realised that I can no longer use the *xx feature codes from the SIP provider. Is there some way I can forward all *xx codes to the trunk, or am I stuck using locally generated features only? |
| 23:47.32 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 23:58.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:59.06 | ChannelZ | Schmee: assuming you have no other local features that are consuming those codes being dialed (but I think all of Asterisk's features codes are in-call) then it's more a matter of writing some dialplan to pass them along to your provider. |
