| 00:00.29 | russellb | no, nothing straight forward comes to mind |
| 00:00.40 | cusco | what if I was to use SIP' |
| 00:00.41 | cusco | ? |
| 00:00.49 | cusco | err |
| 00:00.51 | cusco | IAX2 |
| 00:01.03 | russellb | same limitations there |
| 00:01.13 | cusco | so nogo for iaxvar |
| 00:01.34 | russellb | yeah, IIRC, works kind of like sending custom SIP headers |
| 00:01.39 | russellb | lets you add stuff to pass along on call setup |
| 00:02.07 | cusco | makes sense since lesl control data is transfered with iax.. |
| 00:02.27 | cusco | argh, I have no experience with agi |
| 00:03.34 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 00:03.50 | cusco | and my problem is that DIALEDTIME, ANSWEREDTIME or CDR duration/billsec on local-asterisk is not the same as in gw-asterisk |
| 00:04.02 | cusco | and I would like to have DAHDI's on this one... |
| 00:04.53 | cusco | any other way that you could think of transmitting DIALEDTIME, ANSWEREDTIME from GW's Dial ? |
| 00:11.50 | *** join/#asterisk kaushal (~kaushal@14.97.11.152) |
| 00:11.54 | kaushal | Hi |
| 00:12.32 | kaushal | where do i look into under /etc/asterisk/ conf files to disable software echo cancellation ? |
| 00:12.33 | cusco | hi |
| 00:12.49 | cusco | you mean on dahdi channels ? |
| 00:12.54 | kaushal | yeah |
| 00:13.04 | cusco | in /etc/dahdi/system.conf |
| 00:13.07 | kaushal | ok |
| 00:13.12 | cusco | mg2 is default echo canceller |
| 00:14.03 | kaushal | cusco: shall i pastebin /etc/dahdi/system.conf ? |
| 00:15.45 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 00:16.17 | cusco | no, I never actually disabled echo canceller |
| 00:16.26 | kaushal | echocanceller=mg2,63-77,79-93 |
| 00:16.32 | cusco | hw echo chanceller is expensive |
| 00:16.35 | cusco | lol |
| 00:16.47 | cusco | well |
| 00:16.56 | cusco | did u use dahdi genconf? |
| 00:17.02 | kaushal | yes |
| 00:17.06 | kaushal | /usr/sbin/wancfg_dahdi |
| 00:17.24 | cusco | /etc/dahdi/genconf_parameters |
| 00:17.26 | kaushal | basically i dont need it |
| 00:17.38 | cusco | #echo_can none |
| 00:17.46 | cusco | uncomment that |
| 00:17.49 | cusco | and re-generate them |
| 00:18.00 | cusco | I guess it replaces 'mg2' with 'none' |
| 00:18.27 | kaushal | whats mg2 ? |
| 00:18.46 | cusco | one of several available echocancellers |
| 00:18.51 | kaushal | ok |
| 00:19.22 | kaushal | so uncomment #echo_can none in /etc/dahdi/genconf_parameters and then run again /usr/sbin/wancfg_dahdi ? |
| 00:19.43 | cusco | I would say so |
| 00:19.46 | kaushal | ok |
| 00:19.55 | kaushal | cusco: can i pvt message you ? |
| 00:19.59 | cusco | if you used dahdi_genconf in the past |
| 00:20.03 | cusco | I would rather not |
| 00:20.09 | cusco | lol |
| 00:20.29 | cusco | dunno what wancfg_dahdi is |
| 00:20.30 | cusco | tho |
| 00:20.59 | kaushal | cusco: ? |
| 00:21.23 | cusco | I used /usr/sbin/dahdi_genconf |
| 00:21.36 | cusco | to generate system.conf |
| 00:22.14 | kaushal | ok |
| 00:22.21 | kaushal | cusco: can i pvt message you ? |
| 00:22.32 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 00:22.40 | kaushal | since there are some critical things |
| 00:22.51 | cusco | I have my own problems at the moment... |
| 00:22.58 | kaushal | oh ok |
| 00:22.59 | kaushal | np |
| 00:23.06 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 00:28.16 | cusco | is there a variable that can tell me the briged channel name? |
| 00:28.53 | cusco | or nvm, that one I can send trough sip headers |
| 00:28.54 | rdegges | hey p3nguin, you around by any chance? |
| 01:13.07 | p3nguin | rdegges: Yes. |
| 01:20.09 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
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| 03:57.21 | rdegges | p3nguin: still around? :p I figured out my problem! Thought you'd be interested ^^ |
| 03:57.33 | rdegges | It took forever, but I debugged the entire thing down to a specific line =/ |
| 03:58.04 | rdegges | extenpatternmatchnew=yes was on, and apparently--using that options breaks the X option of ChanSPy in 1.8.x, but not in my older 1.6.x code. =/ |
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| 05:02.01 | p3nguin | rdegges: File a bug report on it. |
| 05:41.50 | *** join/#asterisk ziggyfish (~brendan@123-243-163-103.static.tpgi.com.au) |
| 05:42.10 | ziggyfish | ahhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhh, why does this thing work sometimes and not others |
| 05:42.57 | ziggyfish | still having problems with voice |
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| 07:21.37 | ChannelZ | problems how |
| 07:47.04 | irroot | morning |
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| 09:35.45 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 09:35.47 | devil_evoxxx | hi all |
| 09:36.49 | devil_evoxxx | hi irroot |
| 09:37.05 | devil_evoxxx | i've got some problem with rtptimeout |
| 09:37.12 | devil_evoxxx | it does not work properly |
| 09:39.42 | devil_evoxxx | i've specified it on global section |
| 09:39.54 | devil_evoxxx | i've got to speicy it on single peer ? |
| 09:40.44 | kaldemar | devil_evoxxx: how is it not working? do you see rtp with rtp debug? |
| 09:44.13 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 09:47.28 | devil_evoxxx | no rtp stream between my asterisk and ip of my |
| 09:47.49 | devil_evoxxx | supplier |
| 09:47.56 | devil_evoxxx | i've got two appended channels |
| 09:47.59 | devil_evoxxx | :( |
| 09:51.31 | devil_evoxxx | i've added rtpkeepalive=0 |
| 09:53.09 | devil_evoxxx | but nothing change, i've observed that a call between two internal phone, automatically hangup due to lack of activity |
| 09:53.46 | devil_evoxxx | but, when a call comes or go out from my provider the problem still persist. I've to put rtptimeout on definition of peers? |
| 09:55.04 | devil_evoxxx | p.s. i'm using asterisk 1.8.7 |
| 09:58.53 | dirkD | devil_evoxxx: are you using media transfer? |
| 09:59.27 | dirkD | (assuming this is a sip channel, i missed part of your problem description because my connection dropped) |
| 10:00.24 | devil_evoxxx | i'm using an asterisk boxes 1.8.7 where my sip provider give me the abilty of terminating and receiving call. The connection with provider is directly in SIP. |
| 10:00.33 | devil_evoxxx | on every peer with my provider i've set canreinvite=no |
| 10:00.53 | devil_evoxxx | no nat bewteen my box and provider, all server have public ip |
| 10:02.02 | devil_evoxxx | the problem is that sometime , channels remain up (incoming and outgoing) , but the call is not active (somethimes the ATA of my client is offline) |
| 10:02.26 | devil_evoxxx | i've placed in general section this directive : rtpkeepalive=0 rtptimeout=120 rtpholdtimeout=350 |
| 10:04.16 | Gugge | what state is the hanging call in? |
| 10:04.24 | Gugge | a dial() thats never answered maybe? |
| 10:05.12 | devil_evoxxx | Guge the state is AppDial((Outgoing Line)) |
| 10:05.26 | devil_evoxxx | and state = UP |
| 10:06.22 | devil_evoxxx | state =up and application = AppDial((Outgoing Line)) |
| 10:06.59 | dirkD | did you set directmedia=no? |
| 10:07.17 | dirkD | i am not sure if canreinvite still is a valid option in 1.8 |
| 10:08.31 | devil_evoxxx | no, i not use directmedia=no |
| 10:08.40 | dirkD | try it :) |
| 10:08.46 | devil_evoxxx | i use canreinvite=no still from porting my sip.conf from 1.4 to 1.8.7 |
| 10:08.54 | devil_evoxxx | ...mm..good question dirk :) |
| 10:09.22 | devil_evoxxx | now i go eathing something..and next i try this.. |
| 10:10.21 | devil_evoxxx | but, is correct setting directmedia=no , or canreinvite=no, directly with my provider? i think no, because, i've to set canreinvite/directmedia on peer=friend of my client |
| 10:10.24 | devil_evoxxx | mmm.. |
| 10:11.48 | dirkD | i think it should work |
| 10:12.04 | dirkD | yes, on the client |
| 10:25.15 | *** join/#asterisk dym (~patrick@netsplit.me) |
| 10:25.30 | dym | greetings! |
| 10:28.13 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 10:31.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:35.52 | irroot | devil_evoxxx dirk canreinvite is now directmedia |
| 10:36.15 | irroot | dirkd ^^ |
| 10:36.38 | dirkD | ok, thought so :) |
| 11:52.46 | devil_evoxxx | irroot: but can this wrong setting make rtptimeout not working? |
| 11:54.03 | irroot | devil_evoxxx what problem you have with rtptimeout i made patch last week for it |
| 11:54.14 | irroot | the per peer settings were not working |
| 11:54.39 | irroot | what you using rtptimeout for |
| 12:05.05 | devil_evoxxx | i'm using rtptimeout for close channel that remain ip |
| 12:05.06 | devil_evoxxx | up |
| 12:05.13 | devil_evoxxx | remains appended sorry |
| 12:07.07 | irroot | devil_evoxxx where is the setting in global or peer ?? |
| 12:07.28 | irroot | the per peer setting is broken and will be fixed in 1.8.8 |
| 12:07.55 | devil_evoxxx | i've setted in global section |
| 12:08.16 | devil_evoxxx | is broken in all trunk or just in your svn trunk? |
| 12:09.42 | irroot | devil_evoxxx in all |
| 12:09.50 | irroot | if its set in global should work |
| 12:09.51 | devil_evoxxx | i've to set rtptimeout both on peer and global? |
| 12:10.10 | irroot | remember that if the rtp is flowing it will stay up |
| 12:10.25 | irroot | so if the rtp is still going the call wont cut |
| 12:10.37 | irroot | rtp debug the ip to see |
| 12:11.59 | devil_evoxxx | ok, but for example, i've got two asterisk (the main with 1.8.7) and the client machine (your last svn trunk 1.8) |
| 12:12.20 | devil_evoxxx | last night two call remain appended between main machine ( 1.8.7) and the client machine |
| 12:12.37 | devil_evoxxx | but on the client machine , core show channels say "0 call" |
| 12:12.59 | devil_evoxxx | first of all i've to fix canreinvite with directmedia |
| 12:13.41 | devil_evoxxx | and next, wait for the problem again.. |
| 12:15.47 | *** join/#asterisk robinsmidsrod (~robin@apache.smidsrod.no) |
| 12:16.14 | irroot | devil_evoxxx ok intresting if it happens buzz me if im arround |
| 12:16.27 | irroot | can look at the stats |
| 12:19.21 | devil_evoxxx | ok :) |
| 12:20.19 | robinsmidsrod | I just bought a new Siemens Gigaset DX800A, and I'm trying to sync my google contacts to it - while using the gigaset quick sync software I noticed that it does some kind of AT Hayes + OBEX communication over port 650 (obex) |
| 12:20.36 | robinsmidsrod | does anyone have any experience with this sync process? |
| 12:20.50 | robinsmidsrod | is this on-topic for this channel? |
| 12:45.37 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 12:47.05 | devil_evoxxx | irroot: ..today i'm going to be mad.. |
| 12:47.12 | devil_evoxxx | i've got a ip phone |
| 12:47.17 | irroot | ... |
| 12:47.22 | devil_evoxxx | directly registered on my asterisk boxes.. |
| 12:47.37 | devil_evoxxx | natted ip..i set rtp debug ip 87.xx.xx.xx |
| 12:47.55 | devil_evoxxx | i try to make a call from the phone to my cell |
| 12:48.01 | devil_evoxxx | but no rtp stream show in cli |
| 12:48.15 | devil_evoxxx | and i've set in logger.conf debug con console |
| 12:49.13 | irroot | mmm |
| 12:50.08 | devil_evoxxx | directmedia on peer of this phone ( type friend) |
| 12:50.13 | devil_evoxxx | is setted to directmedia=no |
| 12:54.38 | irroot | and the nat = ?? |
| 12:54.52 | devil_evoxxx | yes |
| 12:54.54 | devil_evoxxx | nat=yes |
| 13:02.36 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 13:02.53 | darkbasic | do someone know what's an alarm 4? log is PRI got event: Alarm (4) on D-channel of span 1 |
| 13:03.12 | darkbasic | Immediately after I have Detected alarm on channel 1: Red Alarm and Detected alarm on channel 2: Red Alarm |
| 13:03.18 | darkbasic | but it work flawlessly! |
| 13:03.52 | darkbasic | in fact as soon as I receive a call I get PRI got event: No more alarm (5) on D-channel of span 1 and Detected alarm on channel 1(and 2): Red Alarm |
| 13:05.23 | darkbasic | sorry I meant "Alarm cleared on channel 1/2" |
| 13:05.41 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 13:05.42 | WIMPy | It's a BRI? |
| 13:10.59 | darkbasic | WIMPy: yes, sangoma A500 with dahdi (latest wanpipe) |
| 13:11.20 | WIMPy | Sounds like the good old power saving issue. |
| 13:11.47 | darkbasic | WIMPy: didn't know about that issue, can you please give me more info? |
| 13:12.18 | WIMPy | Dahdi doesn't like when layer 2 gets deactivated. |
| 13:12.48 | WIMPy | Doues it get up agin if you try to call out? |
| 13:13.05 | darkbasic | didn't try to call out, I will check |
| 13:13.12 | darkbasic | but it does if I call in |
| 13:13.43 | WIMPy | That's the easy one. |
| 13:18.21 | darkbasic | WIMPy: no it doesn't: Unable to create channel of type 'DAHDI' (cause 17 - User busy) |
| 13:19.01 | devil_evoxxx | irroot: in 1.4 if i set rtp debug ip 87.x.x.x it work.. |
| 13:19.35 | devil_evoxxx | in 1.8.7 , i set rtp set debug ip 87.x.x.x. and i can't see the rtp stream |
| 13:19.55 | WIMPy | darkbasic: That's bad then. Do you run current versions? |
| 13:20.20 | darkbasic | WIMPy: latest firmware, asterisk-1.8.7, dahdi-2.5 and libpri-2.4.12 |
| 13:20.42 | darkbasic | but I tried with older libpri (2.4.11.5 and 2.4.11.3) and with older dahdi |
| 13:21.03 | WIMPy | Really bad then. |
| 13:21.08 | darkbasic | WIMPy: in the log there is also a PRI Span: 1 Unable to receive TEI from network in state 2(Assign awaiting TEI)! |
| 13:21.39 | darkbasic | with the legacy smg (woomera) and asterisk 1.8.3.3 it does work flawlessly |
| 13:21.51 | WIMPy | Looks like it's unable to reactivate L2. But these kind of issues are far from new :-( |
| 13:22.15 | WIMPy | Ok, if you know a working combination, use that. |
| 13:22.56 | darkbasic | WIMPy: it isn't working, woomera is bugged and unmantained and I need t38 gateway (and so asterisk >= 1.8.5) |
| 13:23.46 | WIMPy | Try the working versions of libpri and dahdi with the current Asterisk. |
| 13:23.58 | darkbasic | in particular woomera has a huge bug with asterisk 1.8: it doesn't work when you convert between codecs |
| 13:24.38 | darkbasic | WIMPy: is there any way to disable power management? |
| 13:24.59 | darkbasic | WIMPy: I already tried the working version of libpri, it doesn't work too |
| 13:25.13 | WIMPy | Yes. |
| 13:25.29 | WIMPy | You can try to see if your telco would disable it for you. |
| 13:29.51 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:31.13 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 13:32.18 | darkbasic | WIMPy: I'mg going to send an e-mail to sangoma's support, can you please tell me how I can collect some useful debug info for them?something like pri intense debug span 1? |
| 13:32.53 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:33.14 | WIMPy | I would expect them to know the issue. |
| 13:34.12 | darkbasic | WIMPy: I'm not so sure, they have DAHDI support for BRI only since 1 month |
| 13:36.17 | WIMPy | Really? What did they use before? |
| 13:36.58 | WIMPy | I'd go for whatever it is :-) |
| 13:37.00 | darkbasic | WIMPy: a proprietary gateway solution, something called woomera befora (now legacy) and a _SIP_ gatway now |
| 13:37.46 | WIMPy | Hmm. Ok. Maybe not. |
| 13:38.19 | darkbasic | I dont' like it too, especially because it's overkill for BRI |
| 13:38.36 | darkbasic | they create it for SS7 I think |
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| 13:45.13 | *** join/#asterisk bowzak (~bowzak@95.170.203.162) |
| 13:46.09 | bowzak | anyone know the peer details settings for reberworld? |
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| 13:47.50 | *** join/#asterisk sassyn (~sassyn@bzq-82-80-242-217.cablep.bezeqint.net) |
| 13:47.55 | sassyn | hi all |
| 13:48.26 | sassyn | maybe someone has the same problem |
| 13:48.36 | sassyn | when running version 1.8.x on debign/ubuntu |
| 13:48.49 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 13:48.58 | sassyn | I get 100% CPU on one of my cores when using asterisk_1.8.7.0-1digium1 100% CPU |
| 13:49.25 | sassyn | asterisk-h323 |
| 13:49.41 | sassyn | it seems when having h323 codec on |
| 13:49.47 | sassyn | it get's into 100% cpu |
| 13:50.10 | WIMPy | H.323 is not a codec |
| 13:50.48 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 13:51.34 | dym | *sigh* good old netmeeting .) |
| 13:51.36 | dym | :) |
| 13:53.36 | sassyn | WIMPy, i mean channel |
| 13:54.37 | russellb | echo "noload => chan_h323.so" >> /etc/asterisk/modules.conf |
| 13:54.44 | russellb | *CLI> module unload chan_h323.so |
| 13:55.02 | sassyn | russellb, well but I need this to be on |
| 13:55.08 | russellb | oh, heh. |
| 13:55.31 | sassyn | version 1.6.x works find |
| 13:55.33 | sassyn | fine* |
| 13:55.40 | sassyn | any idea? |
| 13:55.54 | russellb | not without diving into backtraces and code, which I don't have time to do right now |
| 13:56.15 | russellb | only other idea is if your config isn't too complex, you could try switching to chan_ooh323 |
| 13:57.12 | devil_evoxxx | guy's i think i've found something really strange..i'm using ast 1.8.7 and when i set rtp set debug ip 87.x.x.x. asterisk say me this : RTP Debugging Enabled for address: 87.x.x.x:0, but nothing shown in asterisk cli. But if i set rtp set debug on i can see all rtp stream, also the stream from my ip 87.x.x.x, particularly i saw that one of the rtp port for my ip is 16474 and, if i set rtp set debug ip 87.13.67.31:16474 i can see the 1-way rtp stream but |
| 13:59.06 | irroot | russellb yo dude how hangs ... |
| 13:59.54 | irroot | sassyn h323 is not supported well enough to get things moving |
| 14:00.05 | irroot | devil_evoxxx that is be odd |
| 14:00.59 | russellb | i'd say that's a bug. |
| 14:01.07 | russellb | probably something related to the IPv6 conversion |
| 14:01.15 | russellb | waves to irroot |
| 14:01.18 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:02.01 | irroot | russellb sassyn i commited 2 fixes for h.323 ipv6 recently check out branches/1.8 |
| 14:02.57 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 14:03.25 | irroot | double bugger !!!! |
| 14:03.30 | devil_evoxxx | irroot: in my last paragraph i want to say: " why if is :0 i can't see the rtp bi-directioinally? |
| 14:03.42 | irroot | there is a little bug in T.38 turning gateway off |
| 14:04.10 | irroot | devil_evoxxx ill need to see the SDP + invite of this call |
| 14:04.26 | sassyn | irroot, well do u think it is a ipv6 problem? |
| 14:04.53 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 14:05.41 | irroot | sassyn its possible id recomend using ooh323 its awesome and it does faxing and with t38modem / hylafax you can have a fax services |
| 14:06.38 | devil_evoxxx | irroot: http://pastebin.com/2n6pUzDa |
| 14:06.51 | *** join/#asterisk master_of_master (~master_of@p57B53863.dip.t-dialin.net) |
| 14:07.26 | sassyn | infobot, OK |
| 14:07.26 | infobot | fine |
| 14:08.22 | sassyn | irroot, OK so if I used asterisk-ooh323_1.8.4.4 I don't need the asterisk-h323? |
| 14:08.40 | irroot | no not at alll |
| 14:08.51 | irroot | ooh323 is contained all in one |
| 14:09.05 | irroot | chan_h323 requires C++ libs openh323 |
| 14:11.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:11.37 | irroot | o=root 1910572649 1910572649 IN IP4 94.230.64.37 devil_evoxxx what ip is this |
| 14:15.18 | devil_evoxxx | main asterisk boxes |
| 14:15.20 | *** part/#asterisk robinsmidsrod (~robin@apache.smidsrod.no) |
| 14:15.21 | devil_evoxxx | asterisk 1.8.7 |
| 14:15.55 | devil_evoxxx | directly from main asterisk trunk ( i've downloaded from asterisk.org) |
| 14:16.13 | bowzak | noob here... when i make a call through a trixbox, it sends like 5 NOTIFY messages to the provider, but nothing comes back. Registration is fine on the trixbox. how do i troubleshoot? |
| 14:16.37 | dym | #trixbox |
| 14:18.03 | bowzak | thanks |
| 14:18.38 | irroot | ~trixbox |
| 14:18.38 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 14:18.43 | irroot | love the reply |
| 14:19.12 | irroot | oops its been edited used to say sh1tbox :P |
| 14:19.52 | irroot | devil_evoxxx where is the nat ?? |
| 14:20.01 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:20.04 | devil_evoxxx | the nat is on 87.x.x.x |
| 14:20.10 | irroot | mmm |
| 14:20.22 | irroot | have you set locallan in sip.conf ?? |
| 14:20.29 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 14:20.31 | [TK]D-Fender | bowzak: pastebin the SIP debug for your call |
| 14:20.31 | *** part/#asterisk bowzak (~bowzak@95.170.203.162) |
| 14:20.33 | [TK]D-Fender | ~pb |
| 14:20.33 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:20.35 | [TK]D-Fender | ^^ |
| 14:20.45 | [TK]D-Fender | Oh well.. there that went... |
| 14:20.46 | devil_evoxxx | the ip 94.230.64.37 is setted directly on asterisk machine |
| 14:20.48 | devil_evoxxx | no i've not set locallan |
| 14:28.55 | irroot | devil_evoxxx it a public ip maybe putting it on local lan will help fix this |
| 14:29.59 | devil_evoxxx | ok..i try |
| 14:30.03 | devil_evoxxx | but i'ts a bug? |
| 14:32.38 | irroot | devil_evoxxx not really look in the conf file you need to set your locallan to know what is nat when using real ip's |
| 14:32.44 | *** join/#asterisk cyborg-one (1000@85-238-111-153.broadband.tenet.odessa.ua) |
| 14:39.38 | irroot | ; + whether it is talking to someone "inside" or "outside" of the NATted network. |
| 14:39.40 | irroot | ; This is configured by assigning the "localnet" parameter with a list |
| 14:39.42 | irroot | ; of network addresses that are considered "inside" of the NATted network. |
| 14:39.44 | irroot | ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY. |
| 14:39.46 | irroot | ; Multiple entries are allowed, e.g. a reasonable set is the following: |
| 14:44.21 | pabelanger | sassyn: does the same problem happen if you compile 1.8.0 from source? |
| 15:00.02 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
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| 15:13.14 | darkbasic | WIMPy: I found a FIX!!!!!!!!!! :D |
| 15:13.16 | darkbasic | WIMPy: http://svnview.digium.com/svn/libpri?view=revision&revision=2273 |
| 15:13.21 | darkbasic | still didn't test it tough |
| 15:22.00 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 15:38.19 | darkbasic | WIMPy: yeah it works 8) |
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| 15:49.41 | devil_evoxxx | irroot: you intend the locallan on definition of peer=friend ? |
| 15:50.04 | irroot | devil_evoxxx its localnet sorry its a global setting |
| 15:50.42 | irroot | also look at the extended directmedia options not only the main one |
| 15:52.04 | *** join/#asterisk ggd (~ggd@pool-173-72-204-39.clppva.fios.verizon.net) |
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| 16:00.15 | devil_evoxxx | irroot: i'v set localnet of the phone.. |
| 16:00.17 | devil_evoxxx | same issue |
| 16:04.46 | [TK]D-Fender | You don't |
| 16:04.55 | [TK]D-Fender | Localnet is a server side option, not a peer option. |
| 16:05.00 | [TK]D-Fender | this is [general] stuff. |
| 16:22.18 | devil_evoxxx | yes, i've set in general section |
| 16:22.25 | devil_evoxxx | but, my server have not a local-lan |
| 16:22.29 | devil_evoxxx | is directly with public ip |
| 16:22.45 | devil_evoxxx | and i still can not make rtp set debug ip [ip-of-client] |
| 16:22.53 | devil_evoxxx | registerd on server as type=friend |
| 16:23.12 | darkbasic | devil_evoxxx: I do have public ip too, never set localenet |
| 16:24.01 | devil_evoxxx | ok.. |
| 16:24.10 | devil_evoxxx | my problem was that i cant debug rtp stream |
| 16:24.17 | devil_evoxxx | from a client |
| 16:24.27 | devil_evoxxx | that make a call trought a peer ( my sip provider) |
| 16:26.29 | *** join/#asterisk wesphillips (~wphill04@adsl-99-161-156-160.dsl.hstntx.sbcglobal.net) |
| 16:26.32 | *** part/#asterisk wesphillips (~wphill04@adsl-99-161-156-160.dsl.hstntx.sbcglobal.net) |
| 16:26.39 | devil_evoxxx | rtp debug still "mute" |
| 16:26.53 | devil_evoxxx | i'm using 1.8.7 |
| 16:27.17 | saxa | ~book |
| 16:27.17 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 16:30.34 | *** join/#asterisk irroot (~irroot@41.54.136.34) |
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| 16:53.22 | *** part/#asterisk wesphillips (~wphill04@adsl-99-161-156-160.dsl.hstntx.sbcglobal.net) |
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| 17:24.13 | [sr] | people a small offtopic |
| 17:24.29 | [sr] | for who's on the US, the verizon 4G, is WIMAX, is this correct? |
| 17:33.25 | devil_evoxxx | darkbasic: you are using ast 1.8.x? if yes, if you try to make rtp set debug ip ip-of-your-registered-phone |
| 17:33.45 | devil_evoxxx | you are able to see rtp stream trought phone and your server? |
| 17:34.52 | darkbasic | devil_evoxxx: I'm currently upgrading to 1.8.7, I will check and let you know |
| 17:37.09 | p3nguin | [sr]: Verizon 4G is LTE. |
| 17:38.07 | [sr] | i see |
| 17:38.32 | [sr] | 4G can be in many forms, depending on what the mobil operator decides to use |
| 17:38.44 | *** join/#asterisk bmg505 (~leon@196-209-44-105.dynamic.isadsl.co.za) |
| 17:38.59 | [sr] | if i'm not wrong |
| 17:39.50 | p3nguin | I'm not sure who is using WiMAX for 4G in the US. As far as I've seen, no one else has advertised 4G services. |
| 17:41.49 | [sr] | in the US verizon is the only one for what i see |
| 17:44.04 | carrar | haha |
| 17:44.30 | carrar | Now if only you could truely have the full speed without limitations |
| 17:44.43 | carrar | or hell, just the full speed |
| 17:45.22 | carrar | Depending where you live will determine the best and fastest wireless provider |
| 17:45.49 | [sr] | non-cord stuff will never be perfect |
| 17:45.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:45.55 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 17:45.59 | [sr] | but where i live i'll have 4G in about 10years :) |
| 17:46.08 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 17:46.23 | [sr] | they are more interest in providing FTTH... that i'll have in a few weeks |
| 17:47.27 | p3nguin | I don't use my wireless phone for anything other than phone calls, so I couldn't care less what speeds they have. |
| 17:48.02 | carrar | I have a AT&T HSPDA express card and it works great |
| 17:48.47 | [sr] | p3nguin: wireless on the phone is nice when we're out, or out of the country to make call using a sip client |
| 17:48.57 | carrar | Merlin X card |
| 17:49.01 | carrar | works great |
| 17:49.04 | [sr] | i spend 2 for each minute in the US :| expensive |
| 17:49.10 | [sr] | spent |
| 17:56.21 | *** part/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
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| 18:03.09 | *** join/#asterisk VoipForces (~Adium@modemcable090.69-59-74.mc.videotron.ca) |
| 18:04.49 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 18:06.49 | VoipForces | Hi all, I have a strange issue, here is the setup, this is a dialer project that transfers calls to agents. Agents are ZoiperBiz softphone under WIndoze set for auto-answer. What I see is that the softphone answer delay is between 288ms and 64 seconds !⦠The long answer delay is not conststant to some stations. I don't want to turn full SIP debug in the risk of adding more delay. any hints ? |
| 18:07.45 | p3nguin | I'm not sure how enabling sip debug would add delay. |
| 18:09.13 | p3nguin | That would be like saying web pages take longer to load because the web server is logging all the traffic. |
| 18:12.00 | [sr] | VoipForces: what's your machine hardware specs? |
| 18:12.24 | VoipForces | [sr] Dual Quad Xeon HP Proliand ML350 |
| 18:12.39 | [sr] | here's your problem, you have an HP |
| 18:12.39 | VoipForces | [sr]: 12Gb or RAM. |
| 18:12.40 | [sr] | :p |
| 18:12.48 | [sr] | ok jokin (ya i don't like hp) |
| 18:13.06 | VoipForces | [sr] :-P HP has always worked great for me. |
| 18:13.35 | [sr] | i build servers my own, with less () i have more |
| 18:13.54 | [sr] | about your issue, no idea |
| 18:14.01 | VoipForces | [p3nguin]: it is all internal lan. Only the SIP trunk is on the internet |
| 18:14.25 | p3nguin | ~trunk |
| 18:14.25 | infobot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 18:15.28 | VoipForces | p3nguin: The carrier link is over the internet (AllStream). All ZoiperBiz softphone are on the same internal lan as the asterisk server. |
| 18:16.06 | VoipForces | Any way to have the call PID logged with the SIP traces? |
| 18:16.14 | p3nguin | I'd select a phone to debug. sip set debug peer <that phone's name> |
| 18:17.18 | p3nguin | If your method isn't working out like you'd hoped, maybe it's time to learn how to use chan_agent and app_queue. |
| 18:17.45 | VoipForces | p3nguin: Not easy in a 60 seat callcenter when I don't know which seat will be used a given night. |
| 18:18.52 | VoipForces | p3nguin: Yeah I am thinking about that or use meetme rooms. |
| 18:18.57 | p3nguin | Then I'd advise you to learn how to use app_queue and chan_agent. That seems to be the correct way to handle it. |
| 18:19.08 | p3nguin | There's no reason to use MeetMe in that case. |
| 18:19.14 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:19.51 | VoipForces | p3nguin: Well, rememver this is a outbound dialer. Not sure how quques would help. |
| 18:20.26 | p3nguin | Why can't the calls be dropped into a queue where agents are standing by? |
| 18:20.30 | [TK]D-Fender | VoipForces: SIP spec has nothing to do with the delay |
| 18:20.49 | VoipForces | p3nguin: I was thinking of havingeach agent in his own meetme room and show the detected human to the corresponding agent meetme. This dialer design needs to work in a way where calls are assigned specific agent. |
| 18:21.08 | VoipForces | [TK]D-Fender: Windoze latency ? |
| 18:21.09 | p3nguin | Sounds like a bother. |
| 18:21.21 | [TK]D-Fender | VoipForces: "something else" |
| 18:21.36 | VoipForces | p3nguin: This is a requirement in order to pull the case file. This a a gvt agency survey program. |
| 18:21.53 | VoipForces | [TK]D-Fender: Yeah, but what? |
| 18:22.54 | [TK]D-Fender | VoipForces: windows / Zoiper. Pick one |
| 18:23.43 | [TK]D-Fender | VoipForces: If you disable AA on Zoiper and can accept instantly all the time manually, then it's zoiper. If you weren't thorough with your tests and all calls are variable, then it's windows |
| 18:23.51 | VoipForces | [TK]D-Fender: yeah. right now I have Zoiper using it's own auto-answer. Do you think it would be better using the SIP CallInfo method? |
| 18:24.14 | [TK]D-Fender | VoipForces: Why should any method be "slow"/ |
| 18:24.16 | [TK]D-Fender | ? |
| 18:24.32 | [TK]D-Fender | It has issues. Try another |
| 18:25.44 | VoipForces | [TK]D-Fender: Yeah. Wish it was as simple as just changing the method. It's 60 PC each with a possible 70 user accounts to change. But I'll ask the customer to change the ZoiperBiz configuration script to add the CallInfo method. |
| 18:26.21 | [TK]D-Fender | VoipForces: The app is defective if it's doing this. It isn't the mothod, its the programmer |
| 18:27.20 | VoipForces | [TK]D-Fender: If it was doing it constantly I would agree. But it's like 25% of the calls that go above 700ms |
| 18:29.07 | [TK]D-Fender | You'r going to have to trace it to place the blame |
| 18:41.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:42.26 | VoipForces | [TK]D-Fender: Yeah. DO you know any way to have SIP traces logged with the corresponding call PID ? |
| 18:47.00 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 18:47.10 | doolittlework | hi ther epeople |
| 18:48.09 | doolittlework | i am having problem with one way speech, incomming call on sip trunk i can hear the caller, outgoing call from sip trunk the calee can hear me |
| 18:48.19 | WIMPy | If we were ipeople we'd have to pay to Apple. |
| 18:48.29 | doolittlework | lol |
| 18:48.33 | VoipForces | doolittlework: firewall ? |
| 18:48.51 | VoipForces | doolittlework: nat or external network settings in sip configuration |
| 18:50.25 | doolittlework | VoipForces: woulld natting not affect the internal calls as well? |
| 18:50.53 | VoipForces | doolittlework: not if they are on the same network as the asterisk server |
| 18:51.38 | VoipForces | doolittlework: your asterisk is behind a firewall? natted? |
| 18:52.31 | doolittlework | there is a nic connected to a vlan 10.168.1.0 and a 192.168.200.0 connected to a wifilink |
| 18:52.54 | doolittlework | 10.168 is the asterisk and snom phone network |
| 18:53.20 | WIMPy | Neither of them is public, so there's obviousely NAT involved. |
| 18:53.42 | doolittlework | so i take it there is some sort of nat on the 192.168.200 |
| 18:54.03 | WIMPy | If you can connecy to the internet there must be NAT. |
| 18:54.35 | doolittlework | no just via a radiolink to an asterisk server in the other building |
| 18:54.49 | doolittlework | but they are on two separate networks |
| 18:55.04 | WIMPy | Ok, so no Internet involved? |
| 18:55.25 | doolittlework | nope jsut one asterisk box talking to another over wifi |
| 18:56.42 | VoipForces | doolittlework: And that first asterisk box, you have one-way audio also ? |
| 18:56.55 | VoipForces | doolittlework: how's that first asterisk server cottecting to the pstn? |
| 19:01.59 | doolittlework | the telko is across the road from us the have an astriks box, they provide us with a sip trunk |
| 19:02.07 | doolittlework | over the wifi link |
| 19:02.29 | doolittlework | their network is on 192.168.200.0 network |
| 19:02.36 | doolittlework | this is on my eth0 |
| 19:03.09 | doolittlework | my eth1 is 10.168.1.20(asterisk) and 10.168.1.0 for my snom phones |
| 19:03.49 | doolittlework | if i recive a call over their sip trunk i can hear the caller but he can not hear me |
| 19:04.16 | doolittlework | if i make a call the callee can hear me but i can not hear him |
| 19:07.06 | *** join/#asterisk irroot (~irroot@41.53.33.45) |
| 19:15.00 | [sr] | website still has 10.0 beta1 |
| 19:15.06 | [sr] | on asterisk.org |
| 19:15.43 | p3nguin | Do you want it deleted? |
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| 19:26.30 | doolittlework | sorry to bug with network questions what is the valid range of ip addresses for the 41.221.230.0/255.255.255.192 subnet |
| 19:27.51 | WIMPy | 0-63 |
| 19:28.56 | ChannelZ | get 'ipcalc' |
| 19:29.53 | *** join/#asterisk hugogee (~hugogee@cpe-76-175-210-224.socal.res.rr.com) |
| 19:30.04 | p3nguin | Or use any of the hundreds of online subnet calculators. |
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| 19:35.08 | *** join/#asterisk loconut (~loconut@173-16-61-8.client.mchsi.com) |
| 19:35.39 | loconut | hello- I'm about to implement a pause/unpause deal on our intranet page, and I'm wondering if when someone is paused, what their status will show as in QueueStatus? |
| 19:35.49 | loconut | or ExtensionStatus ? |
| 19:36.06 | loconut | (eg is there a way to tell if someone is paused?) |
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| 19:58.43 | *** join/#asterisk mtbf (~ewilded@n0life.pl) |
| 20:00.38 | mtbf | I'd like to read the value from DTMF, I assume it to be only digits, can retrieve if from some variable or do i have to use exten => _pattern to catch this? |
| 20:02.56 | p3nguin | It will only be in a variable if you use Read(). |
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| 20:05.10 | doolittlework | thanks all for the help figured it out |
| 20:05.32 | mtbf | Thanks p3nguin. |
| 20:05.40 | doolittlework | cheers go well all |
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| 21:05.10 | devil_evoxxx | darkbasic: ok :) let me know if rtp debug work |
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| 22:06.46 | carrar | http://uncrunched.files.wordpress.com/2011/10/brutallyhonest.jpg |
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| 22:22.28 | GreatSUN | rehi all |
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