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| 02:18.53 | fubada | can someone provide me with a default sip.config for polycoms? |
| 02:18.57 | fubada | not sure where to find! |
| 02:19.55 | Igneous | example configs usually come with the firmware |
| 02:20.17 | fubada | are you then familair with what the difference is between custom.cfg and sip.cfg |
| 02:20.40 | fubada | im trying to get phoneprov going, everything works, but the phone cant find its config |
| 02:22.37 | fubada | shit it jsut worked! |
| 02:38.27 | ChannelZ | shits |
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| 04:58.49 | soapee401 | d\d |
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| 05:31.49 | mealstrom | good morning |
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| 05:32.12 | xoveruk | hi |
| 05:32.45 | xoveruk | Is it advisable to use SIP or IAX2 for a remote VOIP phone connecting to a server in another country? |
| 05:39.24 | [TK]D-Fender | Doesn't matter |
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| 06:04.13 | xoveruk | which is better over satellite links? |
| 06:06.02 | [TK]D-Fender | That part doesn't matter so much. Certain codecs do better with PL. I'd notmally suggest SIP w/ Speex for those scenarios. |
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| 06:20.19 | schmidts | good morning |
| 06:26.46 | CommaCrazy | Morning |
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| 06:53.14 | jpsharp | xoveruk: I ran G729 over SIP over geosync satellite links with no problem. |
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| 07:56.33 | din3sh | hello all |
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| 08:33.41 | jacc0 | hi |
| 08:35.33 | din3sh | its FRIDAY! |
| 08:35.34 | din3sh | :D |
| 08:47.03 | jacc0 | Yes!! |
| 08:47.23 | jacc0 | and my boss just took a day off |
| 08:47.27 | jacc0 | :P |
| 08:51.03 | ectospasm | heh, I work alone for the time being |
| 08:51.20 | CommaCrazy | I wish my boss was not here for the day |
| 08:51.36 | ectospasm | ...see people when I get here, see them when I leave, but not in between |
| 08:51.40 | ectospasm | (night shift) |
| 08:51.50 | CommaCrazy | geez |
| 08:52.02 | CommaCrazy | that kinda sux |
| 08:52.20 | CommaCrazy | on the other hand no people to hassle you |
| 08:52.57 | ectospasm | ...plus a bump in pay, and I get a four day weekend every week. |
| 08:54.27 | CommaCrazy | nice |
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| 08:54.57 | CommaCrazy | and your line of work is? (guessing something to do with VoIP or telephony) |
| 08:57.49 | din3sh | my boss is at another office |
| 08:57.51 | din3sh | :D |
| 08:58.31 | din3sh | night shift is good for a while |
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| 09:05.11 | Sheeplet | lo |
| 09:14.01 | din3sh | lo too |
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| 09:30.17 | Lantizia | How can I tell who hung up from the logs? |
| 09:30.44 | Lantizia | would this mean the 303 extension hung up? "Executing [h@macro-dialout-trunk:1] Macro("SIP/303-0914df88", "hangupcall|") in new stack" |
| 09:30.57 | ectospasm | Lantizia: no |
| 09:31.19 | ectospasm | if you're collecting SIP debug or a packet capture you can see which side sends the BYE message |
| 09:31.47 | ectospasm | packet capture would be easier to analyze |
| 09:31.55 | Lantizia | so without sip debug or packet capture - simply looking at the full log won't tell you if the trunk hung up on the extension or if the extension hung up on the trunk? |
| 09:32.07 | ectospasm | no, it won't |
| 09:32.17 | Lantizia | that's some pretty awful logging |
| 09:32.33 | ectospasm | not the way it works. Enable SIP debug and collect a packet capture. |
| 09:32.44 | Lantizia | too late now |
| 09:32.49 | ectospasm | Also, that detail *might* be in the CDR, but I don't deal with that much so I don't know. |
| 09:33.22 | ectospasm | if you've enabled CEL properly you might see it there too |
| 09:33.29 | ectospasm | ...but again, I don't deal with CEL much |
| 09:33.36 | Lantizia | nah not got around to cel |
| 09:34.41 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 09:34.43 | din3sh | ectospasm: for proper billing mechanism (e.g tracking attn xfers), we have to use both cdr and cel or would cel be enough? |
| 09:35.06 | ectospasm | din3sh: I don't know |
| 09:35.29 | ectospasm | ...but again, I don't deal with CEL or CDRs much |
| 09:38.20 | *** part/#asterisk Delido192123 (~s-mendyka@web.its-medienservice.de) |
| 09:39.56 | kaldemar | Lantizia: you can use app Dial options g and F to output something in your dialplan. |
| 09:45.36 | Ice_Strike | Hi |
| 10:01.57 | jacc0 | hi |
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| 10:16.57 | Lantizia | Hey when a customer of ours makes a call outbound through our ITSP (which we use an IAX2 link to) we see... "IAX2/Trunk-13489 is making progress passing it to SIP/210-087109a0" then "IAX2/TrunkTrunk-13489 is ringing" then "IAX2/Trunk-13489 stopped sounds" then ""IAX2/Trunk-13489 answered SIP/210-087109a0" |
| 10:18.52 | mirelab | Lantizia: Is that a problem? |
| 10:19.21 | Lantizia | well does that mean the inbound audio from the trunk has stopped prior to it being answered |
| 10:19.41 | Lantizia | "IAX2/Trunk-13489 stopped sounds" |
| 10:23.05 | mirelab | Lantizia: maybe this post can give you some pointers: |
| 10:23.07 | mirelab | http://www.eflo.net/VICIDIALforum/viewtopic.php?t=7881 |
| 10:24.37 | kaldemar | Lantizia: what is the problem with the call, or is there a problem? |
| 10:24.49 | WIMPy | It means that YOUR Asterisk has stopped sounds. |
| 10:25.47 | WIMPy | Although I'm not sure what sounds it could provide in that situation.. |
| 10:26.28 | WIMPy | Switches often give some clickediclack soung on proceeding. |
| 10:33.47 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 10:34.05 | kami | Hello. |
| 10:34.49 | kami | WIMPy: I was cheering too soon about the HFC USB card/misdn2/lcr/asterisk 1.8 combo working. |
| 10:35.05 | kami | I still have the "One-way Audio" problem. |
| 10:36.08 | *** join/#asterisk jermey_g (~anonymous@c-dea1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 10:36.57 | *** join/#asterisk alexscott (~alexscott@37.122.202.194) |
| 10:43.14 | shadebob | Hi |
| 10:44.22 | shadebob | In "sip set debug on" mode I have many many many "From: "Unknown" <sip:Unknown@10.10.x.x>;" |
| 10:44.35 | shadebob | I can't explain the origin of these calls .... |
| 10:44.53 | *** join/#asterisk jermey_g (~anonymous@c-dea1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 10:45.12 | shadebob | sample : http://pastebin.com/610NTUwN |
| 10:45.56 | din3sh | every now and then asterisk ends up in deadlock |
| 10:46.26 | din3sh | i can get in cli, phones cannot call each other |
| 10:46.32 | jermey_g | <PROTECTED> |
| 10:46.50 | din3sh | 1.4.22 |
| 10:47.21 | mirelab | din3sh: are you doing any reloads while live? |
| 10:47.37 | jermey_g | din3sh, what is preventing you from an upgrade |
| 10:48.04 | din3sh | i have a few boxes on 1.4.22 |
| 10:48.08 | din3sh | no major issues |
| 10:48.17 | din3sh | so i didnt want to upgrade |
| 10:48.56 | din3sh | asterisk on this box does accept calls from PRI |
| 10:49.08 | din3sh | but no SIP phones can call each other |
| 10:49.13 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:49.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 10:49.21 | din3sh | "restart now" doesnt not help |
| 10:49.24 | mirelab | din3sh: check the error/full log (if it's enabled) what is the few last actions before deadlock |
| 10:50.20 | din3sh | mirelab: i have, nothing peculiar in there |
| 10:50.29 | din3sh | for the server to work again |
| 10:50.47 | din3sh | i have to kill asterisk, restart dahdi, start asterisk |
| 10:51.03 | din3sh | "top" shows like 0.00 |
| 10:51.07 | din3sh | no load on server |
| 10:51.57 | kaldemar | shadebob: find out what is at the address on the "SIP read from..." line. |
| 10:52.30 | jermey_g | din3sh, i would blame on dhadi for no reason |
| 10:52.32 | mirelab | din3sh: than only debugging maybe can help:https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 10:53.14 | shadebob | kaldemar : a phone ... |
| 10:53.55 | din3sh | mirelab: thanks have to decide between debugging or rolling out the 1.8.12 |
| 10:54.28 | din3sh | if i could find a simple fix, i wouldnt have to upgrade |
| 10:54.29 | mirelab | din3sh: IF it's not much code translating , than upgrade :) |
| 10:54.30 | din3sh | hmm |
| 10:55.04 | shadebob | kaldemar : http://pastebin.com/VvJG9HuV …. |
| 10:57.32 | din3sh | i'll try recompiling everything, if problem persists, will move to 1.8.12 |
| 10:57.33 | din3sh | ;) |
| 11:00.38 | kaldemar | shadebob: it's asterisk sending OPTIONS messages to the phone because you have qualify=yes configured for it. |
| 11:06.56 | *** join/#asterisk wonderworld (~ww@dsdf-4db513f7.pool.mediaWays.net) |
| 11:12.03 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:12.30 | shadebob | kaldemar : yes. When I disable qualify on the phone… unknown call disappear …. |
| 11:12.59 | *** join/#asterisk kenshin (~kenshin@dsl-69-172-84-168.acanac.net) |
| 11:14.26 | shadebob | kaldemar : but normally qualify is configured for 60s …. In my case qualify appear every second !!! |
| 11:15.57 | shadebob | kaldemar : My asterisk manage 400 phones …. should I avoid starting Qualify on all phones? |
| 11:18.13 | din3sh | why is it better to have canreinvite=no? |
| 11:19.02 | kaldemar | shadebob: btw, those are not calls. if you enable qualify for all phones, then asterisk sends messages for all phones. see qualifyfreq option. |
| 11:19.14 | kaldemar | din3sh: who says it is? |
| 11:20.49 | mirelab | din3sh: canreinvite=no , if ast is a bridge it will create new channel and bridge it with destination, if it's Yes than it will try to bridge directly 2 channels but you may loose dtmf decoding like i did lol |
| 11:22.10 | din3sh | kaldemar: just asking :p |
| 11:23.56 | mirelab | for example, my * is reciving inband DTMF and sending it to trunk as rfc2833 , if i set it to yes , DTMF is not getting decoded |
| 11:25.10 | shadebob | kaldemar : thanks for your help ! |
| 11:25.59 | *** part/#asterisk kami (~user@unaffiliated/kami-) |
| 11:26.00 | din3sh | so in what event we might want to set it to no? |
| 11:27.01 | mirelab | not completly sure about that :P |
| 11:27.22 | din3sh | kaldemar? |
| 11:27.59 | kaldemar | din3sh: for example if you have two endpoints behind nat. they'd try to send a private address to each other. |
| 11:30.26 | kaldemar | or a phone behind the same nat as asterisk and a provider. |
| 11:30.45 | kaldemar | depends on the setup and what asterisk needs to be able to do. |
| 11:34.39 | din3sh | if on same subnet with no nat, it would be pointless to set it to =no then? |
| 11:36.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 11:36.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 11:37.13 | ectospasm | din3sh: no, if you want to keep Asterisk in the audio path |
| 11:37.20 | kaldemar | din3sh: if there is no need for the audio to go through asterisk. |
| 11:47.06 | din3sh | ok thanks |
| 11:53.26 | din3sh | your explanations saves hours of debugging pointlessly |
| 11:53.27 | din3sh | ;) |
| 12:08.09 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 12:10.02 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:18.59 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 12:20.21 | *** join/#asterisk timahvo1 (~rogue@197.179.253.17) |
| 12:22.19 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 12:24.36 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:34.20 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:40.37 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 12:40.57 | eXcAliBuR | does anyone know if the latest Ubuntu has a firewall that will prevent sip phones from connecting to asterisk? |
| 12:43.12 | kaldemar | eXcAliBuR: see what "iptables -nL INPUT" tells you |
| 12:47.04 | eXcAliBuR | just gives me this |
| 12:47.05 | eXcAliBuR | target prot opt source destination |
| 12:47.36 | eXcAliBuR | i don't know iptables :( i didn't learn that in school |
| 12:48.08 | kaldemar | that's not all it gives you. |
| 12:48.24 | eXcAliBuR | yuh |
| 12:48.37 | eXcAliBuR | and this |
| 12:48.38 | eXcAliBuR | Chain INPUT (policy ACCEPT) |
| 12:48.52 | mirelab | so no settings :) |
| 12:49.11 | eXcAliBuR | that means it is allowing everything? |
| 12:49.18 | kaldemar | so you have a policy of ACCEPT and no rules that would block any packets. you can count that firewall out of the problem equation. |
| 12:49.33 | mirelab | exactly |
| 12:50.00 | eXcAliBuR | ok, my sip phone doesn't make asterisk do anything, like see that it tried to register |
| 12:50.08 | eXcAliBuR | so where do i start trouble shooting |
| 12:50.15 | eXcAliBuR | i can ping the server |
| 12:50.26 | eXcAliBuR | i put in the ip number ofit |
| 12:50.44 | kaldemar | get yourself to asterisk's CLI (asterisk -vvvr) and enable sip debug and verbosity (sip set debug on, core set verbose 10). |
| 12:50.55 | kaldemar | you'll see any SIP messages that your asterisk receives. |
| 12:51.22 | mirelab | eXcAliBuR: what does 'sip show peers' tells you from CLI> |
| 12:51.38 | eXcAliBuR | no such command |
| 12:52.09 | eXcAliBuR | maybe my sip is broken |
| 12:52.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tlwwulstctzqowlo) |
| 12:52.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:52.36 | kaldemar | chan_sip.so, i.e. the sip module is not loaded. no wonder nothing happens. |
| 12:52.51 | kaldemar | eXcAliBuR: "module load chan_sip.so" |
| 12:53.31 | eXcAliBuR | i have to paste bin, hand on |
| 12:53.33 | eXcAliBuR | hang* |
| 12:53.45 | Sjors | can I disable the sending of 183 Session Progress messages over a SIP channel? |
| 12:54.03 | eXcAliBuR | http://asterisk.pastebin.ca/2153913 |
| 12:55.13 | mirelab | you have no chan_sip module |
| 12:55.21 | eXcAliBuR | how do i fix that? |
| 12:55.28 | eXcAliBuR | rebuild asterisk? |
| 12:55.42 | mirelab | i guess that is the best thing to do |
| 12:55.44 | kaldemar | how did you install? |
| 12:56.00 | eXcAliBuR | from source |
| 12:56.04 | eXcAliBuR | i typed make |
| 12:56.07 | eXcAliBuR | make install |
| 12:56.10 | eXcAliBuR | make menuselect |
| 12:56.11 | mirelab | make sure in menuselect that it is enabled |
| 12:56.24 | kaldemar | did you run the configure script? |
| 12:56.25 | eXcAliBuR | i just exited menuselect. it looked scary |
| 12:56.41 | kaldemar | and make menuselect is pointless if run after make and make install. |
| 12:56.55 | kaldemar | make menuselect lets you choose which modules are built and installed. |
| 12:58.11 | mirelab | make menuselect and check in Channel Drivers |
| 12:58.31 | WIMPy | Sjors: I'd like to do that as well, but I think there's no user way. |
| 12:58.33 | mirelab | [*] chan_sip it should look like this |
| 12:59.10 | Sjors | WIMPy: hmm, I'd consider patching it in |
| 12:59.14 | eXcAliBuR | do i run ./configure first? |
| 12:59.24 | kaldemar | eXcAliBuR: yes |
| 12:59.33 | mirelab | eXcAliBuR: that first |
| 12:59.41 | Sjors | WIMPy: but in this case, it's not really my problem, fixing it on my end won't really help: |
| 13:00.16 | WIMPy | Sjors: For me it's only about Asterisk sendig it when entering the dialplan. |
| 13:00.29 | WIMPy | That way I only get part of the dialled number in to the phones redial list. |
| 13:00.53 | kaldemar | eXcAliBuR: there's a nice little file called README that will walk you through the minimum steps. see "NEW INSTALLATIONS" in there. |
| 13:00.55 | Sjors | WIMPy: my operator noticed that if I send an 183 status report, an early media stream will be set up between me and the one I'm calling |
| 13:01.12 | Sjors | WIMPy: except it's the wrong way around: I can call someone, and speak to him, and as long as he doesn't pick up, that call is free |
| 13:01.38 | Sjors | WIMPy: a silly bug somewhere :D |
| 13:01.48 | WIMPy | Interesting. |
| 13:01.52 | WIMPy | obviousely. |
| 13:02.21 | eXcAliBuR | i don't see chan_sip |
| 13:02.38 | eXcAliBuR | i'm running the latest asterisk |
| 13:02.44 | eXcAliBuR | maybe it's broken |
| 13:02.46 | eXcAliBuR | :[ |
| 13:02.47 | kaldemar | eXcAliBuR: don't see where? |
| 13:02.47 | mirelab | eXcAliBuR: than listen to kaldemars edvice |
| 13:02.58 | WIMPy | eXcAliBuR: Do yo have openssl? |
| 13:03.02 | eXcAliBuR | yes |
| 13:03.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:03.19 | eXcAliBuR | i'm in menu select |
| 13:03.59 | kaldemar | you'll see chan_sip under "Channel Drivers" |
| 13:04.09 | eXcAliBuR | i found it, it's XXX |
| 13:04.25 | mirelab | eXcAliBuRyou need to ./configure first |
| 13:04.31 | eXcAliBuR | i did |
| 13:05.34 | mirelab | did ./configure returned any errors? |
| 13:05.52 | eXcAliBuR | no |
| 13:06.06 | eXcAliBuR | i might not have openssl |
| 13:06.14 | eXcAliBuR | i may have confused it for open ssh |
| 13:06.49 | kaldemar | install it and re-run the configure script. |
| 13:08.01 | *** join/#asterisk fcolista (~fcolista@93-57-120-162.ip164.fastwebnet.it) |
| 13:08.05 | *** join/#asterisk oyugik (~oyugik@41.84.143.52) |
| 13:08.47 | mirelab | eXcAliBuR: lol i suggest delete all, and download fresh package :) |
| 13:09.05 | eXcAliBuR | this is a fresh package |
| 13:09.09 | oyugik | Hey guys, yesterday I had some problems but I thank people @[TK]D-Fender for helping me out |
| 13:09.10 | eXcAliBuR | fresh install |
| 13:09.16 | eXcAliBuR | :[ |
| 13:09.35 | kaldemar | you also probably need libssl-dev |
| 13:10.08 | eXcAliBuR | the tutorial for astrisk 10 sucks |
| 13:10.10 | eXcAliBuR | :[ |
| 13:10.35 | oyugik | Guys I have a module that is not being loaded, its gsm acts as fxs |
| 13:10.49 | mirelab | yes , i guess it's for res_crypto that needs to be [*] and not [XXX] in 'Resource Modules' |
| 13:11.50 | eXcAliBuR | that got me going |
| 13:11.57 | eXcAliBuR | i was missing ssl and libssl |
| 13:12.33 | eXcAliBuR | i'm so excited, i'm so close to hearing "Hello World" |
| 13:12.43 | mirelab | :) way to go |
| 13:14.05 | *** join/#asterisk mbrit (~mbrit@186.120.97.195) |
| 13:14.15 | eXcAliBuR | i hurt myself when i put my digium card in :[, poked 2 of my fingers on solder points |
| 13:14.44 | eXcAliBuR | i dropped the card, and reacted by grabbing it |
| 13:15.04 | eXcAliBuR | should have just let it fall... my fingers still hurt |
| 13:17.59 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9cf3:ffd9:7155:edd7) |
| 13:21.33 | eXcAliBuR | ok my phones are registered, but now i don't hear music on hold, it shows that it's playing in the cli |
| 13:27.46 | wonko | now to figure out Playback() without documentation! (guy who can unblock my internet walked out of the office, of course) |
| 13:37.48 | *** join/#asterisk pithagorians (~pithagori@217.12.122.18) |
| 13:38.07 | jacc0 | use the asterisk cli |
| 13:38.16 | jacc0 | core show application playback |
| 13:38.45 | *** join/#asterisk Quanta (~student@tph104.physik.uni-leipzig.de) |
| 13:39.33 | Quanta | i would like to use sipgate for calling out. i install asterisk in my gentoo. what do i need to install for the gui? |
| 13:40.12 | WIMPy | Join the channel for whatever GUI you want to use. |
| 13:40.13 | [TK]D-Fender | Quanta, Depends on which GUI. |
| 13:40.57 | Quanta | i dont know what are the opptions.... |
| 13:41.02 | Quanta | what is the recommanded? |
| 13:41.09 | Quanta | or standard? |
| 13:41.19 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 13:42.07 | [TK]D-Fender | Quanta, FreePBX (www.freepbx.org) |
| 13:42.14 | Quanta | the GUI is of course to make the calls.... |
| 13:42.19 | [TK]D-Fender | Quanta, Docs on their site & tarballs |
| 13:42.23 | [TK]D-Fender | ~freepbx |
| 13:42.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:42.25 | [TK]D-Fender | ^^^ |
| 13:47.09 | *** join/#asterisk pithagorians (~pithagori@217.12.122.18) |
| 13:47.17 | *** join/#asterisk academy (~mccarthy@unaffilated/academy) |
| 13:47.59 | academy | I have a TDM400P connected to a pstn 'truck line' with 3 channels. What do I need to set to ensure 3 channels are available in DAHDI? |
| 13:48.06 | academy | Asterisk 1.8 |
| 13:48.10 | academy | dahdi 2.5.2 |
| 13:48.13 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:48.48 | ChannelZ | configure it? |
| 13:49.22 | academy | ChannelZ: is it the channel => option? |
| 13:49.53 | ChannelZ | well there is chan_dahdi.conf yes and also the system.conf in /etc/dahdi/ |
| 13:50.37 | academy | so channel => 1,2,3 ? |
| 13:50.50 | ChannelZ | 1-3 |
| 13:51.30 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 13:51.37 | ChannelZ | You need to setup everything prior.. things like signalling=fxo_ks etc |
| 13:52.09 | leifmadsen | everything above channel => is assigned to the channels specified in the channel => |
| 13:52.12 | ChannelZ | dahdi_cfg will, in general, write something to get you going |
| 13:52.27 | leifmadsen | what ChannelZ said pretty much the easiest way |
| 13:53.41 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:55.40 | academy | Yeah, I did that. 1-3 did the trick. Thanks! |
| 13:55.43 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 13:56.07 | omani | I have an extension "88" for voicemailbox. but when dialing 88 I get "Purely numeric hostname (88), and not a peer--rejecting! |
| 13:56.43 | omani | this extension is in context [mailbox] and I included this in [local-phones] <- where all my sip phones are in |
| 13:58.55 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-czurbogxswmoppew) |
| 14:01.27 | academy | ChannelZ / leifmadsen, I get this: chan_dahdi.c: Unable to specify channel 3: No such device or address |
| 14:02.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:02.57 | academy | dahdi_cfg only sees 2 channels |
| 14:03.13 | academy | maybe I'm using the wrong teminology - it's possible to make 3 concurrent calls on the one pstn line |
| 14:03.15 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 14:03.23 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 14:04.28 | kaldemar | academy: dahdi_cfg pretty much prints what you have in /etc/dahdi/system.conf |
| 14:04.45 | kaldemar | academy: see to that part first. |
| 14:10.18 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 14:13.32 | wonko | jacc0: oooohhh, that's NICE! |
| 14:13.40 | wonko | jacc0: thanks! |
| 14:14.55 | jacc0 | ? |
| 14:15.08 | wonko | the core show application playback thing |
| 14:15.10 | jacc0 | O , the core show application ? |
| 14:15.13 | jacc0 | hehehe |
| 14:15.13 | wonko | yeah |
| 14:15.14 | jacc0 | yw |
| 14:15.28 | wonko | that's really nice that the documentation is built in like that |
| 14:15.41 | jacc0 | but how can you be in here wihtout internetz? |
| 14:15.48 | leifmadsen | the asterisk wiki generates that documentation too from the source |
| 14:16.09 | wonko | i have internet, i just wasn't able to get the guy to unblock my web until just now |
| 14:16.15 | wonko | silly environment they have here |
| 14:16.29 | *** join/#asterisk Holos (~cosmond@dyn-dial-mb-216-168-111-227.nexicom.net) |
| 14:16.36 | *** join/#asterisk serafie (~erin@nat/digium/x-hocgwjvsmawnujkk) |
| 14:16.37 | jacc0 | hehehe, better to install teamviewer on your home pc |
| 14:16.52 | Holos | Any recommendation for an FSX ATA Adapter that does 1 port or 2 ports, that is good quality? |
| 14:16.58 | jacc0 | and surf from there (if teamveiwer isn't blocked) |
| 14:17.29 | wonko | eh, i could setup an ssh tunnel if i really wanted to, but i'm lazy. :) |
| 14:17.51 | wonko | as they don't block ssh (that's how i'm here, irssi on my colo box) |
| 14:20.45 | *** part/#asterisk Holos (~cosmond@dyn-dial-mb-216-168-111-227.nexicom.net) |
| 14:20.52 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 14:36.06 | academy | As discussed earlier, I have a TDM400P connected to 1 PSTN line. That single pstn line can support upto 3 concurrent calls. |
| 14:36.10 | academy | the config is hère: http://pastebin.com/DGs768Um |
| 14:36.13 | *** join/#asterisk din3sh (~din3sh@41.136.247.65) |
| 14:36.24 | academy | I've tried setting channel => 1-3, but asterisk won't accept it |
| 14:36.32 | academy | could someone please help me? |
| 14:36.57 | academy | If I try to make 2 concurrents calls, I get CONGESTION |
| 14:37.24 | [TK]D-Fender | PSTN lines support ONE call..... |
| 14:37.51 | [TK]D-Fender | If you use a "flash" and telco 3-way option, that is still 1 channel to the telco and they do the dirty work |
| 14:38.02 | [TK]D-Fender | But even that is a massive PITA. |
| 14:38.38 | academy | my provider has told me that a "trunk line" supports 3 concurrent calls |
| 14:38.48 | academy | if I plug a pstn phone in, I get a dialtone |
| 14:39.02 | academy | they say I can receive 3 concurrent calls too |
| 14:39.06 | academy | is this not correct? |
| 14:39.11 | jacc0 | pstn not is trunk line |
| 14:39.28 | din3sh | is not* |
| 14:39.29 | din3sh | :p |
| 14:39.35 | jacc0 | there is no way pstn can support 3 concurrentcalls |
| 14:39.37 | academy | sorry then, it's a truck line |
| 14:39.41 | academy | *trunk |
| 14:39.50 | Ice_Strike | Anyone got experence with GSM Gateaway? |
| 14:40.05 | din3sh | Ice_Strike: which brand? |
| 14:40.05 | Ice_Strike | I keep getting |
| 14:40.06 | jacc0 | with trunk line i guess they mean a sip trunk |
| 14:40.14 | jacc0 | sip trunks go over ethernet |
| 14:40.17 | Ice_Strike | din3sh Dinstar |
| 14:40.19 | jacc0 | not rj11 |
| 14:40.23 | academy | no, it's provided over rj11 |
| 14:40.30 | academy | and works on my normal phone |
| 14:40.30 | din3sh | have not tried Dinstar |
| 14:40.41 | Ice_Strike | din3sh Which one you tried? |
| 14:40.44 | din3sh | rather 2N voiceblue stuff |
| 14:40.48 | jacc0 | ? |
| 14:41.04 | jacc0 | your old style phone supports 3 concurent calls? |
| 14:41.10 | Ice_Strike | din3sh I guess confguration is similar. |
| 14:41.12 | jacc0 | lol |
| 14:41.16 | din3sh | sip trunk on rj11? |
| 14:41.22 | academy | no. The provider says the line should support 3 concurrent calls |
| 14:41.29 | academy | I get a dilatons on a traditional phone |
| 14:41.33 | [TK]D-Fender | academy, It is incorrect. Dumb analog is dumb analog |
| 14:41.43 | academy | [TK]D-Fender: ok, thanks |
| 14:41.44 | din3sh | there is some kinda call waiting at your telco's exchange then? |
| 14:42.17 | [TK]D-Fender | call-waiting is still one call at a time and a pain to attempt to switch |
| 14:42.26 | jacc0 | maybe if you send a flash they will put the current call on hold for you and let you dial some other number: but that is not concurent |
| 14:43.12 | din3sh | right [TK]D-Fender |
| 14:43.15 | jacc0 | it still only one concurent call over the rj11 |
| 14:43.38 | Ice_Strike | I have connected Asterisk with GSM Gateway.. When I am making a call I keep getting this message on the console: |
| 14:43.38 | din3sh | how does the rj11 come in your * box? |
| 14:43.47 | Ice_Strike | <PROTECTED> |
| 14:43.49 | jacc0 | and if your provider talks about 'trunk line' I guess he is not talking about the rj11 |
| 14:43.50 | [TK]D-Fender | Ice_Strike, pastebin ALL OF IT |
| 14:43.56 | [TK]D-Fender | Ice_Strike, then it's busy |
| 14:44.19 | Ice_Strike | [TK]D-Fender ok 1 sec, |
| 14:44.20 | [TK]D-Fender | Ice_Strike, if there were other channels it could use then it's your job to tell it how to roll over to them |
| 14:44.24 | *** join/#asterisk shadebob (~shadebob@196.217.244.107) |
| 14:44.26 | [TK]D-Fender | Ice_Strike, go read its manual |
| 14:44.37 | Ice_Strike | [TK]D-Fender I have tried the manual many time! |
| 14:44.43 | [TK]D-Fender | Ice_Strike, no need for PB, that error translates properly (unlike most). |
| 14:44.46 | [TK]D-Fender | busy=busy |
| 14:44.59 | [TK]D-Fender | Ice_Strike, then go call their support if you can't figure it out. |
| 14:45.23 | Ice_Strike | [TK]D-Fender The line is not busy 100% I am testing it to ring to my mobile |
| 14:45.27 | Ice_Strike | then it say busy |
| 14:45.31 | Ice_Strike | which is untrue |
| 14:45.40 | din3sh | [TK]D-Fender: what would be the difference between 480 and 486 busy? |
| 14:45.50 | [TK]D-Fender | Ice_Strike, and never jsut say "GSM Gateway". You need specific information for your specific model which is very low odds of finding someone in here for so don't beat around the bush in telling them enough so they chan judge if they have any real experience for you or not. |
| 14:45.52 | jacc0 | Ice_Strike: whene yo get a busy back from the GSM gateway everything on your side looks alright |
| 14:46.15 | [TK]D-Fender | din3sh, Not sure. What does the IETF RFC say? |
| 14:46.26 | kaldemar | din3sh: 480 is not busy |
| 14:47.03 | din3sh | ? |
| 14:47.19 | [TK]D-Fender | Ice_Strike, it is "busy"... if the device has no other resource to use and they don't feel like sending "congestion" instead. Again, confirm with your models manufacturer |
| 14:47.25 | Ice_Strike | jacc0 Yea I am trying to figure out why it say busy from the GSM gateway.. in theory its shouldnt. |
| 14:47.33 | *** join/#asterisk agisamentum (~vastina@70.99.103.122) |
| 14:47.53 | jacc0 | I've seen a couple of GSM-gateways, they where all running asterisk. you might be able to take a look in the CLI from the GSM-gateway |
| 14:48.02 | kaldemar | 480 is temporarily unavailable. for example no answer or an absent subscriber. |
| 14:48.10 | Ice_Strike | [TK]D-Fender DWG2000-8G |
| 14:48.37 | din3sh | 480 Temporarily Unavailable |
| 14:48.43 | din3sh | yep |
| 14:48.43 | jacc0 | you should look into the logs of your GSM gateway to fige out why it is responding with busy |
| 14:48.54 | Ice_Strike | jacc0 I have to do many attempt to make a cal then it work.. maybe like 5 retry. |
| 14:48.57 | Ice_Strike | very odd. |
| 14:49.16 | jacc0 | O |
| 14:49.30 | jacc0 | maybe you don't have good network coverage? |
| 14:49.53 | Ice_Strike | jacc0 Network coverage is pretty good according to GSM gateway |
| 14:49.54 | jacc0 | that is what my phone does when I have poor network coverage |
| 14:50.04 | jacc0 | :p |
| 14:50.22 | Ice_Strike | I have tried inserting sim in the mobile phone and it work well. |
| 14:50.56 | [TK]D-Fender | SIM != antenna and other bits. |
| 14:50.57 | din3sh | Ice_Strike: try calling the sim number and see your log if the phone arrives on the gateway |
| 14:51.00 | jacc0 | okay, maybe it didn't hungup the last call? |
| 14:51.26 | jacc0 | and is still busy as it reports |
| 14:51.55 | din3sh | call-limit= in sip.conf for the gateway? |
| 14:52.09 | Ice_Strike | jacc0 On the GSM - it report no hang up and busy... it just say idle |
| 14:52.25 | Ice_Strike | when gateway do make a call to my phone then it will say alert |
| 14:52.58 | jacc0 | you know what.............. |
| 14:53.02 | din3sh | what about the gateway receiving call from GSM network? |
| 14:53.11 | jacc0 | office hours are over...............bye bye |
| 14:53.16 | din3sh | hhaha |
| 14:53.16 | jacc0 | WEEKEND!! |
| 14:53.29 | din3sh | nice footie tomorrow? |
| 14:53.33 | din3sh | :D |
| 15:01.41 | omani | does anybody have some experience with confbridge? |
| 15:02.51 | [TK]D-Fender | omani, next non-meta question please :) |
| 15:03.01 | omani | ah yeah, sry. =) |
| 15:03.16 | omani | I get the following error: Conference bridge '99' could not be created. |
| 15:03.24 | omani | my dialplan: exten => 99,1,Answer |
| 15:03.25 | omani | exten => 99,n,Wait(1) |
| 15:03.25 | omani | exten => 99,n,Confbridge(${EXTEN},1234) |
| 15:03.25 | omani | exten => 99,n,Hangup |
| 15:03.47 | omani | and 1234 is the context in confbridge.conf |
| 15:03.52 | omani | [1234] |
| 15:03.52 | omani | type=bridge |
| 15:03.52 | omani | max_members=4 |
| 15:03.52 | omani | mixing_interval=10 |
| 15:03.52 | omani | internal_sample_rate=auto |
| 15:03.53 | omani | record_conference=yes |
| 15:03.59 | [TK]D-Fender | omani, PASTEBIN |
| 15:04.03 | [TK]D-Fender | omani, Stop flooding in here |
| 15:04.04 | omani | im done. :/ |
| 15:04.30 | [TK]D-Fender | omani, Well your bridge is named [1234], not [99] |
| 15:04.41 | omani | so? |
| 15:04.44 | [TK]D-Fender | omani, Looks pretty clearly like you got your parameters mixed up |
| 15:04.50 | omani | I have 1234 in there |
| 15:04.51 | omani | :/ |
| 15:04.54 | din3sh | 1234 is the pin right? |
| 15:04.56 | hypknight | Anyone have any ideas as to how to "dial" a short code on behalf of an extension through AMI? |
| 15:05.01 | omani | no the context in confbridge.conf |
| 15:05.06 | omani | is named 1234 |
| 15:05.09 | omani | I have no pin |
| 15:05.11 | [TK]D-Fender | and it is looking for 99 because that is the order you have to list the parameters for the app in |
| 15:05.35 | hypknight | *feature code |
| 15:06.05 | omani | ok so if my context is named "1234" in confbridge.conf |
| 15:06.19 | omani | the line should be: exten => Confbridge(1234) right? |
| 15:06.35 | [TK]D-Fender | omani, "core show application confbridge" <- instructions. |
| 15:06.37 | omani | but same error msg. bridge 1234 could not be created. |
| 15:07.06 | omani | ok confno is the first param |
| 15:07.21 | [TK]D-Fender | omani, Pastebin the instructions, your new configs, your new call attempt, all after having restarted * to make sure your changes even got picked up |
| 15:07.42 | omani | I reload asterisk everytime, no need to restart |
| 15:07.47 | omani | or do I have to? |
| 15:08.14 | [TK]D-Fender | I'd like a restart just because I don't the working environment too much at this point.... |
| 15:08.31 | omani | ok done |
| 15:09.21 | omani | look. I haven't changed confbridge.conf, original version now. dialplan looks like above. a call attempt show the err msgs above |
| 15:09.28 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 15:11.54 | din3sh | confnumber = ${EXTEN} = 99 |
| 15:12.10 | omani | yes |
| 15:12.22 | din3sh | so 1234 is not right |
| 15:12.27 | [TK]D-Fender | [1234] <- CONF NUMBER |
| 15:12.30 | omani | no 1234 is the bridge profile |
| 15:13.18 | omani | like core show apps confbridge says. params are: confno,bridgeprofile,etc. |
| 15:13.30 | omani | I have context 1234 in bridgeprofile, so I said 1234 in dialplan |
| 15:13.48 | omani | but anyway, in both cases, it doenst work |
| 15:14.47 | omani | also a simple Confbridge(1) in dialplan doenst work. normally it should create a conference with a basic bridge_profile out of confbridge.conf (like core show apps confbrigde explains) |
| 15:15.55 | *** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca) |
| 15:16.02 | omani | but, to make it clear. just paste my a valid config for confbridge.conf and extensions.conf pls |
| 15:17.49 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
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| 15:21.34 | din3sh | what |
| 15:21.50 | din3sh | is the main advantage of conf bridge as compared to meetme? |
| 15:22.53 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 15:26.33 | Jasnejac | din3sh: which version? |
| 15:26.39 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
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| 15:51.54 | din3sh | 1.8.x |
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| 16:23.16 | wonko | File tt-weasels does not exist in any format |
| 16:23.17 | wonko | Unable to open tt-weasels (format (gsm)): Success |
| 16:23.26 | wonko | [root@phone asterisk]# ls -l /var/lib/asterisk/sounds/en/tt-weasels.gsm |
| 16:23.27 | wonko | -rw-r--r--. 1 asterisk asterisk 4257 Oct 5 2011 /var/lib/asterisk/sounds/en/tt-weasels.gsm |
| 16:23.30 | wonko | so, uhm, what? |
| 16:27.07 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:28.01 | *** join/#asterisk xpot-station (~xpot@166-70-100-198.ip.xmission.com) |
| 16:29.51 | paulc | wonko: what if you supply the full path rather than just the filename - any difference? |
| 16:31.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:31.31 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:37.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 16:40.38 | wonko | <PROTECTED> |
| 16:40.41 | wonko | File /var/lib/asterisk/sounds/en/tt-weasels does not exist in any format |
| 16:40.44 | wonko | Unable to open /var/lib/asterisk/sounds/en/tt-weasels (format (gsm)): Success |
| 16:41.27 | paulc | wonko: Hmm. Odd. What about permissions on all the directories in the chain? |
| 16:42.48 | Jasnejac | din3sh: sorry - was elsewhere. don't bother on 1.8 - the one on 10 is excellent though, much better than meetme |
| 16:45.12 | wonko | paulc: let me check |
| 16:46.02 | wonko | asterisk owns /usr/lib/asterisk and everything below it |
| 16:46.27 | wonko | also, /usr and /usr/lib are good |
| 16:47.55 | paulc | or /var/lib even? ;-) |
| 16:48.03 | wonko | right, that. :) |
| 16:48.43 | paulc | Hmm.. it's a curious one.. I'm scratching my head.. and still feeling a bit fuzzy, just got into work and I'm not fully plugged in yet.. |
| 16:48.58 | wonko | yeah, i'm a bit confused myself |
| 16:49.09 | wonko | unless i'm missing a module it needs |
| 16:49.14 | wonko | let me turn autoload on and see |
| 16:49.17 | paulc | Asterisk's running as user "asterisk", sound directories and files have appropriate ownership and permissions, as does the file.. |
| 16:49.30 | paulc | what about a "show formats" ? |
| 16:50.40 | wonko | which module would that be run from? |
| 16:52.29 | wonko | ah ha! |
| 16:52.38 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:52.39 | wonko | it's a missing module |
| 16:52.41 | wonko | now which one |
| 16:52.43 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:53.26 | paulc | which format_xxxx do you have loaded? |
| 16:53.38 | paulc | or codec_xxxx |
| 16:53.47 | paulc | Never had this problem as I go with autoloading and "out the box" usually :) |
| 16:56.10 | wonko | i'm trying to do a miminal system not loading everything. :) |
| 16:57.26 | wonko | i'm starting to think that's a silly idea |
| 16:57.27 | wonko | :) |
| 16:57.40 | [TK]D-Fender | Only as silly as the pilot :) |
| 16:59.07 | wonko | heh |
| 16:59.15 | wonko | ok, so that's all sorted out now |
| 16:59.20 | wonko | now to work on the gtalk stuff |
| 17:02.30 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-vxnfytchwbnhlwjw) |
| 17:18.38 | *** join/#asterisk Beta2K (~Beta2K@d24-36-128-84.home1.cgocable.net) |
| 17:18.42 | Beta2K | Hello all |
| 17:23.47 | *** join/#asterisk navaismo (~navaismo@201.123.94.189) |
| 17:26.54 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:29.03 | Beta2K | I know this isn't the right way to do it but I'm kinda in a bind. Anyone around tried to run a asterisk box behind a pfsense nat? |
| 17:30.19 | Beta2K | I got my trunk working fine but any extenstions connecting from the internet don't pass any rtp back to the pbx |
| 17:31.35 | *** join/#asterisk Keisuke (~Keisuke@modemcable201.185-200-24.mc.videotron.ca) |
| 17:32.46 | *** join/#asterisk navaismo (~navaismo@201.123.77.102) |
| 17:33.37 | Keisuke | hello, my asterisk server (1.8.7.0) is set to automatic reload, and i d'ont know how to stop that. (It's reload every about 6 min) |
| 17:34.08 | newtonr | Beta2K: if you can get a packet trace at the endpoint and in front of the asterisk system then possibly someone here will be able to help out if they have time |
| 17:34.30 | newtonr | Keisuke: asterisk reloads, or the whole process restarts? |
| 17:34.47 | navaismo | Keisuke, maybe the reload comes from manager user, do you see at the top some line like 'manager authenticated' |
| 17:35.18 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 17:35.31 | Beta2K | I can't from the endpoint it's a ata, but I've run tcpdump on the asterisk box and see nothing coming from the ata's ip between 10000 and 20000 |
| 17:35.46 | nny | what's the way to clear out the astdb cache for sip entries? Have a phone registering as a new username but asterisk is showing it as the old one still |
| 17:35.49 | Beta2K | sip goes through fine, and rtp makes it from the asterisk machine to the ata |
| 17:35.50 | Keisuke | i think is the whole process restart, (reloading configuration files...) |
| 17:36.33 | Keisuke | even if I block with firewall all ports, the asterisk server restart |
| 17:36.45 | newtonr | Beta2K: then you'll want to look at the SDP values being sent out from asterisk to the ATA |
| 17:37.11 | newtonr | Beta2K: in SDP is where asterisk will tell the ATA where to send back the RTP too, IP and port |
| 17:38.14 | Beta2K | newtonr: can I debug that from the console or do I have to dump the packets? |
| 17:38.22 | Keisuke | navaismo: that is my view at the top of reloading: http://pastebin.com/1abCxswY |
| 17:39.19 | Keisuke | but the manager admin, and remote UNIX connection are usual |
| 17:40.00 | newtonr | Beta2K: eh, you'd probably want to get a packet trace and use wireshark |
| 17:40.51 | navaismo | Keisuke, if you see this line "Manager 'admin' logged off from 127.0.0.1" every reload i think you have a process |
| 17:41.07 | navaismo | that do it with the admin manager user |
| 17:41.35 | Keisuke | navaismo: where are there process ? |
| 17:42.12 | Keisuke | how can i check, if i have some automatic process ? |
| 17:42.33 | navaismo | don't know, but you need to check your crontab, php scripts etc. |
| 17:43.31 | Keisuke | ok, nothing on asterisk or configuration files, can have action on a automatic restart ? |
| 17:45.05 | navaismo | nope as far i know, do you use a gui or plain asterisk? |
| 17:45.35 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
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| 17:46.26 | Keisuke | navaismo: freepbx |
| 17:47.17 | navaismo | even with freepbx i dont see any setting to reload |
| 17:47.52 | nny | what's the way to clear out the astdb cache for sip entries? Have a phone registering as a new username but asterisk is showing it as the old one still |
| 17:48.47 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 17:49.08 | Keisuke | navaismo: thx, for your help |
| 17:49.41 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 17:49.45 | navaismo | no problem |
| 17:51.12 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:59.50 | *** part/#asterisk Keisuke (~Keisuke@modemcable201.185-200-24.mc.videotron.ca) |
| 18:02.37 | [TK]D-Fender | nny, database del family key |
| 18:03.00 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 18:03.41 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 18:04.23 | volga629 | asterisk 1.8 and res ldap what will correct value in ldap for transport mapping ? |
| 18:05.48 | nny | [TK]D-Fender: thanks |
| 18:06.41 | nny | [TK]D-Fender: phone*CLI> database del family key = Database entry does not exist. |
| 18:06.48 | nny | deltree? |
| 18:06.53 | [TK]D-Fender | nny, PB the dump & attempt |
| 18:07.06 | [TK]D-Fender | deltree = dangerous and probably overkill |
| 18:07.41 | [TK]D-Fender | And you have to change "family" and "key" .. I hope you didn't actually jsut cut & paste that as it appeared... |
| 18:08.09 | nny | [TK]D-Fender: roger, ahh, i dunno. I am just trying to resolve oddness with sip credentials being cached. What's the field requirements for "family" and key? |
| 18:08.24 | [TK]D-Fender | you'll see which ones when you dump the DB |
| 18:09.31 | nny | need to install the berkley db tool to do so |
| 18:09.33 | nny | stand by |
| 18:10.45 | nny | doesn't seem to be in repos. I am going to advise the client not to switch peers on devices, this seems like a can of worms |
| 18:11.08 | nny | the registration update/attempt doesn't even appear in asterisk after I switched. So god damn odd |
| 18:11.52 | fubada | hey guys, whenever I leave a voicemail on one extension, all my phones get the new voicemail indicator |
| 18:11.56 | fubada | what could be wrong? |
| 18:12.10 | nny | [TK]D-Fender: ok i see the entries |
| 18:12.16 | fubada | as soon as I play it, the indicator on all phones goes away |
| 18:12.41 | nny | [TK]D-Fender: /SIP/Registry/6759 |
| 18:12.46 | WIMPy | You've got the same mailbox configured for all of them? |
| 18:13.02 | [TK]D-Fender | database del SIP Registry/6759 |
| 18:13.05 | [TK]D-Fender | ^^ |
| 18:13.14 | fubada | WIMPy: no |
| 18:13.23 | volga629 | Is available encryption field for res ldap ? |
| 18:13.38 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 18:13.52 | omani | how can I disable transport tls for only one sip peer in sip.conf? |
| 18:13.54 | nny | [TK]D-Fender: i'll write that down somewhere, seems a bit.. ham handed if changing a device out, but eh, at least there's a solution |
| 18:14.04 | omani | when transport=tls and encryption=yes is enabled globally |
| 18:14.16 | WIMPy | fubada: Are you sure? |
| 18:14.25 | fubada | WIMPy: chekcing :) |
| 18:14.40 | fubada | [TK]D-Fender: i had to go back to using users.conf because 'phoneprov' relies on that 100% |
| 18:14.56 | [TK]D-Fender | fubada, Then you should ditch phonprov |
| 18:15.00 | fubada | whats the deal with users.conf geting such bad rep |
| 18:15.07 | nny | [TK]D-Fender: may as well pick your brain about one other thing I am wrestling with. Any reason why RTCP stats would shit the bed on some channels and show an incorrect loss (i.e. packet count jumps from 200 to 60,000)? I assume there's something odd about the sequence numbers |
| 18:15.09 | fubada | phoneprov is insanely valuable |
| 18:15.18 | nny | sorry for the language :S |
| 18:15.22 | nny | reflexive term |
| 18:15.32 | [TK]D-Fender | There are all sorts of scripts out there to generate varios model configs from * datat |
| 18:15.53 | [TK]D-Fender | nny, Not sure really... |
| 18:16.24 | nny | [TK]D-Fender: one sec I have a pastebin of it in action, caused some panic at first, i may need to tcpdump the interface and find the RTp sequence for both sides |
| 18:16.53 | nny | [TK]D-Fender: http://pastebin.com/AxdFgUF7 is the results of sip show channelstats, 2 seconds apart |
| 18:16.56 | WIMPy | fubada: Is it? How flexible is it? i've never seen it in use. |
| 18:17.00 | [TK]D-Fender | nny, I've never really debugged RTP like that... |
| 18:17.11 | nny | [TK]D-Fender: was going to use RTCP for some reporting on channel quality |
| 18:17.19 | nny | [TK]D-Fender: but right now it's throwing false positives |
| 18:17.53 | nny | my gut says something odd is happening with RTP sequence numbers |
| 18:18.03 | nny | and it's doing the math/ reporting the loss |
| 18:18.05 | fubada | WIMPy: i am able to take a brand new polycom phone out of the box, punch its mac address as a 'macaddress' property under a [user] entry in users.conf, and phoneprov will automatically push the latest firmware, bootrom, and dynamic configuration to the new phone |
| 18:18.11 | fubada | without having to do ANYTHING on the phone |
| 18:18.47 | WIMPy | fubada: Ok, but what about all the other options you may want to set? |
| 18:18.57 | [TK]D-Fender | fubada, And as I said there are plenty of 3rd party tools that do this and more. |
| 18:18.58 | fubada | you do that in your tempaltes |
| 18:19.20 | fubada | [TK]D-Fender: can you point me to some ? Id like to read about that |
| 18:19.44 | fubada | WIMPy: for example, ill set the ntp server in polycom.xml in /var/lib/asterisk/phoneprov |
| 18:19.46 | WIMPy | And can you control what settings are set only once and which are sent every time? |
| 18:19.50 | fubada | (my dhcp doesnt hand out ntp) |
| 18:20.17 | novceguru | Is it possible to look up who owns a number? I deleted a number before it was ported out (yeah, I know) and now voip.ms is telling me it's gone, and I think what i'm hearing is "there's literally nothing we can do about it" |
| 18:20.23 | [TK]D-Fender | fubada, p3nguin has one, there is another that is made for FreePBX that may be convertible, more still that you can google, or dummy up one yourself and SED it |
| 18:20.42 | novceguru | When I call it, it rings busy |
| 18:20.45 | fubada | WIMPy: i havent figured that one out yet...im thinking it will override the changes done on phone if they collide with whats in the templates |
| 18:20.56 | [TK]D-Fender | fubada, So set the NTP in your provisioning directly. |
| 18:21.55 | WIMPy | fubada: Yes, but there are thing you only want to set the first time a phone is taken in to use while other may need updates while it's already in use. |
| 18:22.02 | fubada | [TK]D-Fender: i was just giving that as an example to WIMPy |
| 18:22.07 | WIMPy | I fiund that to be the tricky part. |
| 18:23.47 | volga629 | omani in ext settings transport=udp ? |
| 18:24.54 | nny | novceguru: http://www.reversephonelookup.com ? |
| 18:25.40 | novceguru | nny: I mean, it got deleted 3 days ago, and magically ported away from their network in that time |
| 18:25.44 | nny | novceguru: chances are the provider still owns the number but has provisioned it to someone else |
| 18:25.47 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 18:25.52 | novceguru | gotcha |
| 18:26.02 | novceguru | they said they as asking their carrier |
| 18:26.06 | nny | novceguru: you may have to set some things on fire to get that resolved |
| 18:26.08 | novceguru | I assume that means like, it went back to level 3? |
| 18:26.15 | nny | novceguru: they're probably a reseller |
| 18:26.22 | nny | novceguru: so the short answer is they don't know |
| 18:26.26 | novceguru | I mean, went back to level3/whatever, yeah I understand |
| 18:26.28 | novceguru | they are |
| 18:26.56 | nny | novceguru: you probably need to escalate the issue until you get something other than a support drone |
| 18:27.07 | nny | novceguru: it's friday, they just want to gtfo ;) |
| 18:27.20 | fubada | tfom? |
| 18:28.33 | novceguru | they keep saying they've asked the provider, they will update the ticket as necessary, blah blah |
| 18:29.11 | nny | novceguru: good luck, i have dealt with resellers before, (well I guess they all are to some degree) and some are awesome, some are horrible |
| 18:30.30 | novceguru | thanks, yeah, it's been less then stellar as far as my issue being important, at the same time, I'm the idiot that deleted it |
| 18:30.48 | novceguru | thought it was ported out and was just a stale number in the account but it was just forwarded to the new service's temp number...sigh |
| 18:31.59 | *** join/#asterisk shadebob (~shadebob@197.253.182.145) |
| 18:41.19 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:41.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:44.46 | *** join/#asterisk tm1000_ (~tm1000@provisioner.net) |
| 18:54.59 | omani | volga629: yes. worked for me. thyx |
| 18:56.40 | volga629 | np |
| 18:56.42 | volga629 | :-) |
| 19:05.42 | pabelanger | Anybody ever have an issue with a polycom 331 not generating ring tone. The actual phone, when you get an inbound call to your phone |
| 19:06.04 | [TK]D-Fender | Tried raising the ring volume above 0? ;) |
| 19:06.13 | pabelanger | it is |
| 19:06.27 | [TK]D-Fender | pabelanger, Tested selecting the ring tone in the menu? |
| 19:07.01 | pabelanger | Seems to happen randomly, the phone get a call, line1 blinks, but the phone is not generating the audio ring tone to the user |
| 19:07.22 | pabelanger | not yet, but sounds like a simple test to try |
| 19:08.23 | tm1000 | pabelanger: thats usually a firmware issue |
| 19:08.38 | tm1000 | eg firmware is not matching the configuration file |
| 19:10.05 | pabelanger | tm1000: Ya, that is what I am thinking too |
| 19:10.29 | pabelanger | Which is odd, since this site has 35 phones and is the only one |
| 19:10.40 | pabelanger | Will schedule an upgrade next week of the firmware |
| 19:14.56 | citywok | [TK]D-Fender: I'm using sipp & have 80 concurrent SIP calls, transcoding from gsm to ulaw, and being recorded, and the box is 15% idle, so pretty close to the limit |
| 19:21.20 | *** join/#asterisk newtonr (~newtonr@173-25-121-44.client.mchsi.com) |
| 19:24.45 | *** join/#asterisk schultza (~allen@rc1.rcherbals.com) |
| 19:31.06 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 19:36.46 | *** join/#asterisk ancientz (~ancientz@201-92-241-213.dsl.telesp.net.br) |
| 19:36.47 | ancientz | hello |
| 19:37.06 | ancientz | not sure if anyone cal help me here, I have a asterisk 10.4.0 infrastructure |
| 19:37.53 | pabelanger | ~ask |
| 19:37.53 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:38.48 | *** join/#asterisk timahvo1 (~rogue@197.179.253.17) |
| 19:38.50 | ancientz | I'm having some problem with transfers using features, both attended and blind, as soon as I hit #1 it says "Transfering", then I type one number and the transfer starts already |
| 19:39.27 | ancientz | i`m using dtmf rfc2833 |
| 19:39.34 | ancientz | and g729 codecs |
| 19:39.50 | ancientz | and three grandstream gxp1200 phones |
| 19:39.53 | WIMPy | Did you use a ! pattern in your dialplan? |
| 19:40.10 | ancientz | no |
| 19:40.23 | ancientz | at least not that i know, let me check it |
| 19:40.28 | WIMPy | Are you in the right context? |
| 19:40.41 | ancientz | yes, the problem is I cant even type the full number |
| 19:40.46 | ancientz | the extension is 300 |
| 19:40.54 | ancientz | i type 3 and it already starts transfering |
| 19:41.18 | ancientz | confirmed, no ! |
| 19:41.30 | ancientz | ! pattern* |
| 19:41.37 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 19:41.57 | WIMPy | I'm not sure if overlapping patterns work with feature transfers. I don;t have them enabled. |
| 19:42.20 | WIMPy | But I guess it has a match and thus doesn't wait for further input. |
| 19:42.47 | pzn | when someone dials to extension 800, then I want to "play an mp3" while person at phone 800 still didn't answer. is there any way? |
| 19:42.55 | ancientz | well there is 4 extensions in the config, 300, 301, 302 and 303 |
| 19:43.05 | ancientz | and i`m using ldap realtime |
| 19:43.10 | WIMPy | pzn: core show application dial. |
| 19:43.39 | *** part/#asterisk schultza (~allen@rc1.rcherbals.com) |
| 19:44.10 | WIMPy | Oh, no idea, how realtime influences the picture, but it likely does. |
| 19:45.09 | ancientz | oh well, problem is I need this thing working soon |
| 19:45.45 | ancientz | i even tried to change the dtmf to info |
| 19:45.50 | ancientz | but same problem |
| 19:45.52 | WIMPy | Jus have a little patience. Probably someone will be able to tell you more. |
| 19:46.45 | pzn | WIMPy, tks! |
| 19:47.16 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
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| 19:48.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 19:52.48 | *** part/#asterisk woleium (~woleium@208.53.145.169) |
| 19:54.39 | *** part/#asterisk schultza (~allen@rc1.rcherbals.com) |
| 20:00.10 | volga629 | ancientz Is normal dialing working with res ldap ? |
| 20:00.22 | ancientz | yes |
| 20:00.23 | ancientz | perfect |
| 20:00.26 | ancientz | everything is working |
| 20:00.33 | ancientz | except transfering |
| 20:00.55 | ancientz | even if I transfer using the phone function it works, just does not work when I try to transfer using features |
| 20:02.32 | ancientz | its like it is ignoring transferdigittimeout |
| 20:03.28 | volga629 | if it pooling from ldap it should have mapped record for it |
| 20:03.38 | ancientz | yeah |
| 20:03.53 | volga629 | For example I was need add encryption staff |
| 20:03.53 | ancientz | i think it does, but the problem is I cant type the whole number |
| 20:04.46 | ancientz | It only accept one number, if I type really fast it accept two lol |
| 20:05.11 | nny | Any reason why RTCP stats would be off on some channels and show an incorrect loss (i.e. packet count jumps from 200 to 60,000)? I assume there's something odd about the sequence numbers? I assume since RTCP requires both ends to play nice this could be a provider issue as well |
| 20:05.28 | volga629 | Which ldap schema record it is ? |
| 20:05.40 | ancientz | the original asterisk 10.4.0 |
| 20:05.46 | ancientz | the one that comes with it |
| 20:06.02 | *** join/#asterisk navaismo (~navaismo@201.123.60.170) |
| 20:07.02 | nny | https://issues.asterisk.org/print_bug_page.php?bug_id=9554seems related, but thats 5 years old |
| 20:07.05 | nny | er |
| 20:07.06 | nny | https://issues.asterisk.org/print_bug_page.php?bug_id=9554 |
| 20:07.12 | volga629 | check ldap if you can add attribute astaccountcancalforward under user account |
| 20:07.42 | ancientz | yes i did, its mapping on CanCallForward |
| 20:07.47 | ancientz | and on ldap its set to yes |
| 20:09.09 | ancientz | also on my dialplan I have set tT |
| 20:09.10 | ancientz | options |
| 20:10.01 | ancientz | I saw somebody saying there was a bug on 1.6.1 that having tT would cause my problem, but I have tried setting only t or T and same. |
| 20:12.23 | volga629 | do you see anything in debug ? |
| 20:12.37 | WIMPy | ancientz: Dialling and Dialling are not the same. And the one kind isn;t that easy to get right. |
| 20:14.55 | ancientz | strangely after updating to 10.4.0 i see ast_channel_bridge: Bridge stops bridging channel |
| 20:16.16 | ancientz | also I just got a ast_bridge_call: Timed out for feature! |
| 20:17.29 | ancientz | a question, i`m using g729, when I use the feature to transfer does it convert to slin? |
| 20:18.20 | ancientz | oh i think i know whats happening |
| 20:18.41 | ancientz | i`m using the extenpatternmatchnew |
| 20:19.00 | WIMPy | Oh, that faild for me as well. |
| 20:19.29 | ancientz | because i was facing some problems with asterisk trying to match more than one extension and eventually crash |
| 20:19.44 | ancientz | lets see if it was fixed now on 10.4.0 |
| 20:20.56 | ancientz | its like its searching on ldap each number I type |
| 20:21.48 | ancientz | and ldap returns an error of course because the extension does not exist |
| 20:22.13 | ancientz | asterisk is ignoring the transferdigittimeout option |
| 20:23.46 | volga629 | in mapping you using sipuser or sipeer ? |
| 20:24.15 | volga629 | possible to do config to get extension in place |
| 20:25.00 | ancientz | both |
| 20:25.12 | ancientz | i`m using both sipuser and peer |
| 20:25.16 | WIMPy | Yes, that's the way it should be. |
| 20:25.38 | ancientz | sipusers => ldap,"ou=Person,ou=Entities,dc=pecia,dc=neture,dc=com,dc=br",sip sippeers => ldap,"ou=Person,ou=Entities,dc=pecia,dc=neture,dc=com,dc=br",sip |
| 20:25.40 | WIMPy | The constant lookup that is. |
| 20:26.12 | omani | can I set more than one variable in one line with cmd Set in dialplan? |
| 20:26.22 | omani | like Set(foo=1,bar=2)? |
| 20:26.27 | WIMPy | omani: See ARRAY |
| 20:27.13 | omani | Array can only be written to, not read from |
| 20:27.32 | omani | I want to use these variables in further programming |
| 20:27.58 | omani | or is that not meant with that sentence? |
| 20:28.11 | ancientz | WIMPy strangely it looks up only once and returns an error |
| 20:28.27 | ancientz | the moment it does, the feature cancel and say the extension does not exist |
| 20:28.50 | ancientz | as I have understand the feature should search only once |
| 20:29.03 | WIMPy | omani: Maybe I misunderstood your question. And what sentence? |
| 20:29.41 | WIMPy | ancientz: It might be a misfeature in the ldap switch. |
| 20:29.42 | ancientz | thats why it has the transferdigittimeout, there is a time between each typed number so that after you typed it all it searchs only once |
| 20:29.58 | ancientz | thats bad lol |
| 20:30.00 | ancientz | =[ |
| 20:30.17 | WIMPy | It is perfectly normal that it will do a lookup after ech digit received. That's the way it finds out when you're done. |
| 20:31.02 | WIMPy | Might be interesting to try if overlap dialling generelly works with the ldap switch. |
| 20:31.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:33.06 | ancientz | maybe if i disable overlap |
| 20:33.17 | WIMPy | Where? |
| 20:33.36 | ancientz | on general |
| 20:33.36 | ancientz | Asterisk sip allowoverlap |
| 20:33.46 | ancientz | just saw that there is an option for that |
| 20:34.05 | WIMPy | Feature transfers have nothing to do with SIP. |
| 20:34.14 | ancientz | aw |
| 20:34.36 | volga629 | transferdigittimeout => 3 |
| 20:34.42 | WIMPy | Or any channel. |
| 20:34.49 | ancientz | yeap thats what i did volga629 |
| 20:34.54 | ancientz | problem is it seems to ignore that |
| 20:35.16 | WIMPy | I don;t think it's meant to do a timeout on th destination input. |
| 20:35.40 | *** join/#asterisk jawhn (~jawhn@rrcs-24-123-106-250.central.biz.rr.com) |
| 20:36.36 | ancientz | hm, just noticed something strange |
| 20:36.45 | ancientz | if I type 0 while on features the call drops |
| 20:37.04 | WIMPy | Is it the code for hangup? |
| 20:37.31 | ancientz | no |
| 20:37.36 | WIMPy | You should try without ldap. |
| 20:37.42 | ancientz | tho it is on my dialplan |
| 20:38.33 | ancientz | yeah I think it got a match of 0 from my dialplan |
| 20:39.23 | ancientz | translate.c:773 matrix_rebuild: Discovered 1450000 cost path from slin192 to g722, via slin16 |
| 20:39.35 | ancientz | moment I type #1 to features i get a flood of that |
| 20:39.51 | ancientz | seems like its trying to translate |
| 20:40.00 | ancientz | maybe features were not made for g729? |
| 20:40.09 | WIMPy | What's in your 0 extension? |
| 20:40.26 | ancientz | its the exit number to my provider |
| 20:40.58 | nny | anyone using a reseller program for sip "trunks"? |
| 20:42.04 | ancientz | just created a one number extension |
| 20:42.05 | ancientz | number 1 |
| 20:42.09 | ancientz | and tried to transfer, worked |
| 20:42.12 | ancientz | perfectly |
| 20:42.31 | ancientz | the problem is that I cant type more than one number lol |
| 20:43.58 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 20:44.44 | WIMPy | Well, being able to dial numbers is not exactely Asterisks strengths. |
| 20:44.57 | ancientz | lol |
| 20:46.21 | WIMPy | waits how log it takes for that laugh to be over... |
| 20:48.22 | *** part/#asterisk jawhn (~jawhn@rrcs-24-123-106-250.central.biz.rr.com) |
| 20:48.38 | *** join/#asterisk gusto (~gusto@80.187.231.189) |
| 20:49.13 | gusto | WIMPy: hi |
| 20:49.21 | WIMPy | lo |
| 20:53.11 | volga629 | you can try in you ldap under [config] in res_ldap.conf and on ldap set this option |
| 20:54.03 | ancientz | what option? =o |
| 20:57.20 | volga629 | timeout |
| 20:57.47 | volga629 | transferdigittimeout |
| 20:58.51 | ancientz | nah wouldnt work |
| 20:58.57 | ancientz | this option is from features |
| 20:59.07 | ancientz | the res_ldap only maps sip and extension configs |
| 21:00.14 | *** join/#asterisk Ralcala (~demo@201.132.5.123) |
| 21:00.32 | Ralcala | hi |
| 21:01.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:02.01 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 21:02.17 | Ralcala | hey guys can you please tell where i can find basic sip.conf and extensions.conf files for ipbx with 4 lines and 12 extensions |
| 21:02.36 | Ralcala | and of course a decent ivr menu...please |
| 21:03.03 | WIMPy | ipbx? |
| 21:03.18 | Ralcala | sorry Asterisk server |
| 21:03.37 | WIMPy | Well, that's something different. |
| 21:04.00 | Ralcala | with CentOS 5.4 and * 1.8.13 |
| 21:04.13 | Ralcala | and 2 GS gateways |
| 21:04.16 | WIMPy | For sip.conf, you can work from the sample. The dialplan is entirely up to you. |
| 21:04.42 | [TK]D-Fender | ? book |
| 21:04.47 | [TK]D-Fender | ~book |
| 21:04.48 | 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 |
| 21:04.53 | Ralcala | mmm to complex for my first installation |
| 21:05.47 | [TK]D-Fender | only as complex as you make it |
| 21:06.12 | Ralcala | i mean the file its to large with samples |
| 21:06.59 | Ralcala | iam on analog phone not even ip phones |
| 21:07.45 | WIMPy | As for the dialplan, it's just a demonstration of possibilities. You really have to write your own. |
| 21:07.50 | WIMPy | ~book |
| 21:07.50 | 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 |
| 21:08.23 | Ralcala | ok i try the online book |
| 21:08.26 | Ralcala | thanks |
| 21:09.15 | Ralcala | bye |
| 21:09.42 | volga629 | transferdigittimeout is not feature of parking ? |
| 21:09.56 | [TK]D-Fender | Ralcala: Plenty of beginner sample guides in blogs etc that can show what a small setup could look like |
| 21:10.35 | ancientz | volga629 dont think so, it says transfer lol |
| 21:10.57 | volga629 | check it ? |
| 21:11.09 | ancientz | well it might, but either way it is wrong that the feature tries to overlap |
| 21:11.25 | ancientz | realtime database doesnt accept overlap, they want only the exact number |
| 21:11.43 | WIMPy | ancientz: What's wrong about that? |
| 21:12.10 | ancientz | WIMPy the realtime database returns an error for the overlap query |
| 21:12.19 | ancientz | then the transfer feature cancel the search |
| 21:12.44 | WIMPy | Yes, but that's a problem of the realtime stuff. |
| 21:12.56 | WIMPy | You can run into that in more situations. |
| 21:13.51 | ancientz | well sure |
| 21:14.38 | volga629 | so what you expect for call transfer / |
| 21:14.40 | volga629 | ? |
| 21:15.24 | ancientz | well it should work with a send key |
| 21:15.32 | ancientz | you type the extension number |
| 21:15.38 | ancientz | and press # or something |
| 21:15.41 | ancientz | it commits and call |
| 21:15.53 | ancientz | like a normal transfer button works lol |
| 21:15.56 | WIMPy | And how would you dial a # then? |
| 21:16.12 | ancientz | what do you mean? |
| 21:16.25 | ancientz | the string would look like #1numberofextension# |
| 21:16.34 | ancientz | # => end of string |
| 21:17.26 | WIMPy | Yes, but what if you want to dial an extension containing a #? |
| 21:17.49 | ancientz | oh well, then you have a problem |
| 21:17.49 | ancientz | lol |
| 21:17.59 | WIMPy | And I certainly hate to have to press an extra key that is completely unneccessary. |
| 21:18.11 | ancientz | or maybe it should wait for a timeout |
| 21:18.23 | ancientz | like you type the number and then it waits 2 seconds |
| 21:18.24 | ancientz | and commit |
| 21:18.37 | WIMPy | I hate waiting at least as much. |
| 21:18.38 | ancientz | for every typed key it waits 2 seconds |
| 21:18.41 | ancientz | lol |
| 21:19.01 | WIMPy | And it regularly fails because ppl are too slow. So you have to wait longer and longer. |
| 21:19.40 | WIMPy | Seriousely that has worked for many decades. |
| 21:24.31 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 21:25.41 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 21:25.43 | ancientz | well i have remebered how to enable the ldap overlapping |
| 21:25.51 | ancientz | however I also remembered why i disabled it |
| 21:25.52 | ancientz | realtime_multi_ldap: realtime retrieval requires at least 1 parameter and 1 value to search on |
| 21:26.02 | ancientz | that error happens and stops my call |
| 21:28.42 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tlwwulstctzqowlo) |
| 21:32.55 | ancientz | so.. |
| 21:33.07 | ancientz | seems i have fixed it by adding to my extensions _X. |
| 21:33.23 | ancientz | lol |
| 21:33.35 | WIMPy | Ha |
| 21:33.48 | WIMPy | Now that you mention it, I think I've done that onve as well. |
| 21:34.01 | WIMPy | once |
| 21:36.28 | ancientz | really |
| 21:36.33 | ancientz | youve got to be kidding me |
| 21:36.34 | ancientz | lol |
| 21:37.43 | WIMPy | But I don't have it any more. |
| 21:38.38 | ancientz | lol |
| 21:38.40 | ancientz | exten => _X.,1,NoOp(Do nothing - Workaround) |
| 21:38.45 | ancientz | solved the problem |
| 21:38.46 | ancientz | well |
| 21:39.17 | WIMPy | I'd put a Hangup(1) there, in case you ever get there. |
| 21:39.55 | ancientz | sure, did |
| 21:39.56 | ancientz | lol |
| 21:40.02 | ancientz | but thans WIMPy |
| 21:40.04 | ancientz | thanks * |
| 21:40.15 | ancientz | and volga629 |
| 21:40.19 | ancientz | thanks as well |
| 21:41.09 | volga629 | np |
| 21:49.20 | ancientz | well cya |
| 21:49.22 | *** part/#asterisk ancientz (~ancientz@201-92-241-213.dsl.telesp.net.br) |
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| 22:05.13 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 22:05.24 | nny | ok stupid question time! |
| 22:05.45 | nny | localnet=10.1.0.0/255.255.0.0 localnet=10.2.0.0/255.255.0.0. If a phone has a 10.1.1.75 address, shouldn't asterisk ignore nat=yes? |
| 22:06.55 | nny | Hosts falling within the network ranges specified by the localnet option will be excluded from any NATing efforts by Asterisk. As a result, the source IP address within the SIP requests/responses will use the internal IP address of the network interface associated with bindaddr . |
| 22:15.35 | nny | hi anyone out there? |
| 22:15.43 | nny | feel like i'm taking crazy pills on this one |
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| 22:29.45 | nny | ok new question, putting nat=yes/no under each peer doesn't seem to work with localnet=XXX. It's being forced regardless, is this "workin as intended"? |
| 22:31.00 | navaismo | well, as far i know |
| 22:32.23 | navaismo | nat = yes tell asterisk to use te externip for return the sip and rtp, nat=no use the internal ip address(that the best explaination that my english permit) |
| 22:32.23 | [TK]D-Fender | It isn't directly tied to it |
| 22:32.34 | nny | [TK]D-Fender: explain? |
| 22:32.36 | navaismo | but surely someone came and correct them |
| 22:32.42 | *** join/#asterisk sixohquad (~sixohquad@184.65.142.249) |
| 22:32.46 | navaismo | here is it LOL |
| 22:33.11 | nny | [TK]D-Fender: i had nat=yes on a new phone under peer and localnet defined above it. It wouldn't even register (for obvious reasons) |
| 22:33.14 | navaismo | now im feel high stupid drugs |
| 22:33.20 | nny | [TK]D-Fender: the phone was on the localnet |
| 22:33.26 | sixohquad | hey guys, im having issues getting my voicemail working. I'm trying to call into it, and im entering the password, and the password is not working. I can assure you I have set the password, and I am entering it properly. Any theories on why it may not work? |
| 22:33.34 | nny | [TK]D-Fender: removed the nat=XXX from the peer and reset the phone.. worked |
| 22:33.45 | [TK]D-Fender | nat = yes tells * to ignore the IP the CALLER offers in SDP and answers back to the IP the packet came from. |
| 22:33.52 | [TK]D-Fender | localnet is for what * reports ITSELF as |
| 22:34.03 | [TK]D-Fender | two halves |
| 22:35.18 | nny | [TK]D-Fender: so there's no real way to have a universal nat setting even when asterisk knows that the IP of the CALLER in SDP matches the localnet subnet |
| 22:35.24 | sixohquad | http://pastebin.com/FRchRviJ |
| 22:36.16 | [TK]D-Fender | nny you're still over associating what * thinks bout itself vs what it thinks about the caller |
| 22:36.21 | [TK]D-Fender | DIFFERNT halves. |
| 22:36.34 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 22:36.46 | nny | [TK]D-Fender: yeah i guess I thought from all descriptions * used localnet to describe the peers relationship, not it's own |
| 22:36.53 | nny | its/it's |
| 22:37.30 | nny | the goal was to have a setup that allowed users to move from local to remote without sip.conf changes. so much for that :S |
| 22:37.44 | nny | seems like something people would want though |
| 22:39.16 | nny | [TK]D-Fender: maybe I should try a different question, is there a way to setup asterisk to allow users to switch from local to remote (the system has a public IP/interface) where remote can be behind NAT (i.e. a home router) and not have to edit sip.conf each time? |
| 22:40.47 | nny | sixohquad: is that vanilla asterisk? |
| 22:40.56 | nny | sixohquad: or freepbx/trixbox etc? |
| 22:40.57 | sixohquad | freepbx |
| 22:41.03 | nny | sixohquad: ~freepbx |
| 22:41.05 | nny | gah |
| 22:41.06 | nny | i fail |
| 22:41.15 | nny | but yeah, can't help ya here, freepbx looks like it uses some meta layer to auth vm |
| 22:41.19 | sixohquad | i asked in freepbx yesterday and nobody answered, so i figured i'd give this chan a shot |
| 22:41.25 | nny | Executing [*98@from-internal:16] NoOp("SIP/sativagold-00000003", "app-dialvm: Asking for password so people can't probe for existence of a mailbox") in new stack |
| 22:41.31 | nny | it not using normal auth methods |
| 22:41.35 | nny | so not much can be done here |
| 22:41.52 | nny | and the app is written by the freepbx dev team and nestled in tons of lines of code... |
| 22:41.55 | sixohquad | oh ok, well i hope someone somewhere, somtime can help me ;) |
| 22:41.58 | sixohquad | ok cool |
| 22:42.01 | [TK]D-Fender | nny: treat them as NAT all the time and still set your localnets |
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| 22:42.20 | navaismo | nny, use your domain instead IP |
| 22:42.33 | nny | [TK]D-Fender: ok so that idea still has merit, I was just associating what localnet does wrong? |
| 22:42.38 | nny | navaismo: already got that one done |
| 22:42.45 | [TK]D-Fender | nny: Correct |
| 22:43.29 | nny | [TK]D-Fender: i.e. it's not about the peer definition, but rather when asterisk decides to adjust for NAT? I can understand that. By that issue then, I assume setting NAT=XXX on a per peer basis (apart from being a security risk according to logs) is not wise for this kind of setup |
| 22:44.24 | nny | i'll find a day to test with a fresh setup phone and NAT=yes in general only + localnet. I suspect this was the fix, just wanted to confirm. It's obvious I had the idea of what localnet is wrong if I though it would work per peer |
| 22:45.57 | volga629 | in res_ldap.conf is available option do not validate cert or it should be set in ldap.conf ? |
| 22:47.56 | volga629 | some think like this ssl/novalidate-cert this from Imap |
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| 23:27.16 | gusto | volga629: i thaught you re from russia, but seems to be the opposite way |
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| 23:39.34 | volga629 | what is mean opposite way ? :-) |
| 23:39.37 | Ice_Strike | Hey |
| 23:40.21 | WIMPy | The opposite way is the same way, just another distance. |
| 23:42.51 | volga629 | yes, North America |
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