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| 00:42.29 | volga629 | Hello Everyone, I am trying set trunk with TLS |
| 00:42.49 | volga629 | but getting with self sign problem tcptls.c:209 handle_tcptls_connection: Certificate did not verify: self signed certificate in certificate chain |
| 00:43.36 | volga629 | tried put in config tlsdontverifyserver=no but didn't worked |
| 00:44.26 | volga629 | Dow I need remove password from private key ? |
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| 01:27.44 | mae_tae | good morning philippines... |
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| 04:03.46 | Penguin | Does asterisk 1.8 have the security logger level? I see reference to it in 11, but that doesn't help me with 1.8. |
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| 04:11.59 | Penguin | Looks like it is new in 10 and 11. |
| 04:27.22 | ChannelZ | Yeah |
| 04:29.23 | Penguin | That's too bad. I'm getting tired of seeing the "Sending fake auth rejection for device" messages. |
| 04:37.15 | mae_tae | Penguin: I had already fix the softphone problem, its just an authorization type issue |
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| 04:52.02 | Penguin | [Oct 1 00:49:49] NOTICE[2698]: chan_sip.c:9893 copy_header: No field 'Call-ID' present to copy |
| 04:52.11 | Penguin | What does this mean? |
| 04:52.21 | Penguin | It was right in the middle of a registration attack. |
| 04:53.12 | ChannelZ | Never seen that one before, but probably broken SIP on their part. |
| 04:53.57 | Penguin | They got four attempts in, then that message, then five more attempts before fail2ban acted on it. |
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| 05:06.08 | volga629 | look like somebody tried spoof SDP entry |
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| 06:53.31 | v0lZy | Hello |
| 06:54.02 | v0lZy | guys, im seeing an issue i think is probably network related, but i have not enough understanding to pin the cause down |
| 06:55.05 | v0lZy | When I call from my cell phone to a sip phone, if the person im calling isnt there and the call gets picked up by their coworker from another extension, i get one way audio.. I as the caller hear them, but they dont hear me. |
| 06:55.35 | v0lZy | However, if we repeat the procedure internally (me calling from my own sip phone thats on the same asterisk), audio is normal |
| 06:56.33 | v0lZy | so its only with picking up calls that are coming from outside callers |
| 06:57.20 | v0lZy | im having a hard time understanding this issue and why it happens |
| 06:57.29 | v0lZy | but its consistant over all the phones |
| 06:59.08 | v0lZy | I find it weird because the call from ITSP comes to my asterisk through the firewall just fine |
| 06:59.29 | v0lZy | if the person im calling actually picks up and transfers me to someone else, everythign works fine as well |
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| 07:01.30 | v0lZy | the only time this happens is under the conditions that the caller is calling from outside (isnt registered to the pbx), and coworkers pickup the extension the caller is calling... the caller can hear the person that made the pickup, but the person that made the pickup cant hear the caller |
| 07:01.35 | ChannelZ | are you using directmedia |
| 07:01.50 | v0lZy | ... how do i check? |
| 07:01.55 | ChannelZ | sip.conf |
| 07:02.57 | ChannelZ | oh wait.. you said the cell you're calling from when this happens is not registered to asterisk, it's a standard mobile call coming through your ITSP or something? |
| 07:03.17 | ChannelZ | or is it anonymous SIP to the server? |
| 07:03.24 | v0lZy | yes |
| 07:03.27 | v0lZy | coming through my ITSP |
| 07:03.49 | v0lZy | no, its hitting my asterisk through the trunk the ITSP routes to me. |
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| 07:04.26 | v0lZy | i find this bizzare because obviously, the call makes it to the PBX just fine and audio is normal if the callee picks up as expected |
| 07:04.35 | ChannelZ | ok. Well you'd have to check a SIP debug but it sounds like when the other person remote pickups the ringing extension it's sending a reinvite or something to the ITSP and giving it bogus IPs perhaps |
| 07:04.49 | v0lZy | however, if their peer pickusp from the callee, the peer no longer hears the caller coming in through ITSP |
| 07:05.18 | ChannelZ | When the call comes into your system, is the channel ever Answered() first, like for an IVR? |
| 07:05.28 | ChannelZ | Answer()ed rather |
| 07:05.34 | v0lZy | mmm hold on |
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| 07:08.14 | v0lZy | ChannelZ: this is the relevant dialplan: http://bpaste.net/show/9DFksFTn90K81CdHLqYA/ |
| 07:09.37 | v0lZy | no answer in that plan |
| 07:14.10 | v0lZy | ChannelZ: also, here is my verbose version of whats going on |
| 07:14.11 | v0lZy | http://bpaste.net/show/w2ZOBPvLyA9sQvAn8SRQ/ |
| 07:14.37 | v0lZy | You can see that the first call where 48 answered directly, things are ok |
| 07:14.43 | v0lZy | but on the second call, where a pickup was made |
| 07:14.46 | v0lZy | it throws -- Executing [*848@DialPatterns:1] Gosub("SIP/47-000047e5", "AppDirectPickup,DirectPickup,1(48)") in new stack -- Executing [DirectPickup@AppDirectPickup:1] Pickup("SIP/47-000047e5", "48@PICKUPMARK") in new stack == Spawn extension (AppDirectPickup, DirectPickup, 1) exited non-zero on 'SIP/48-000047e4<ZOMBIE>' |
| 07:15.00 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 07:16.04 | v0lZy | why zombie? |
| 07:17.15 | v0lZy | ChannelZ: this is my AppDirectPickup btw: http://bpaste.net/show/6G7miMv0QKF8zCuO7Gtu/ |
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| 07:19.29 | ChannelZ | well you can chuck an Answer in there somewhere before you dial and see if it magically starts working; if so then chances are you've got a funny reinvite. a SIP debug will probably reveal all |
| 07:22.17 | ChannelZ | wanders off to bed |
| 07:23.16 | bombev | guys I am trting to pass some variables within Macro context but I doesnt work, I am missing something here it is the source code and the output , http://pastebin.com/B21wG1Lh , http://pastebin.com/frBvYE3j |
| 07:25.42 | bombev | I thing I am passing the variables correct |
| 07:25.45 | bombev | exten => _XX.,n,Dial(${DIALGROUP(BG)},40,mM(dendax-registration,${CASENUMBER},${names})) |
| 07:26.25 | bombev | and the output shows that is correct |
| 07:26.26 | bombev | Executing [160@from-dendax-registration:18] Dial("SIP/160-00000001", "SIP/375,40,mM(dendax-registration, 2113812, Dimitar Kudev)") in new stack |
| 07:27.24 | bombev | but within the macro context I cant retrieve the variables , -- Executing [s@macro-dendax-registration:10] SayDigits("SIP/375-00000002", "") in new stack |
| 07:29.29 | ChannelZ | M runs the macro on the CALLED channel, which is a different channel than you set all those variables in... which are channel variables. You can pass the macro arguments, but you use them inside the macro as ${ARG1}, ${ARG2}, etc |
| 07:31.11 | ChannelZ | you also need to use ^ as a delimiter to your macro arguments, not , -- see 'core show application Dial' |
| 07:31.19 | bombev | ChannelZ so within Macro context I have to change this exten => s,n,SayDigits(${CASENUMBER}) to ----> exten => s,n,SayDigits(${ARG1}) |
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| 07:31.56 | ChannelZ | yes |
| 07:32.22 | ChannelZ | and ${names} would be ${ARG2} |
| 07:32.41 | bombev | ok now it makes sense |
| 07:32.49 | bombev | but about the delimiter |
| 07:34.02 | ChannelZ | Dial(${DIALGROUP(BG)},40,mM(dendax-registration^${CASENUMBER}^${names})) |
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| 07:42.15 | bombev | ChannelZ ok I will test it right now |
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| 07:50.48 | bombev | ChannelZ it works, well with delimiter , and ^ |
| 07:52.26 | v0lZy | ChannelZ: im talking to tech support and |
| 07:52.43 | v0lZy | apparently, my codec specification is being reported funny |
| 07:53.56 | v0lZy | I had disallow all and then allow ulaw allow alaw and allow gsm |
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| 07:54.11 | v0lZy | we now limited 2 phones for testing to alaw |
| 07:54.22 | v0lZy | and also on asterisk, i commented out ulaw and gsm |
| 07:54.25 | v0lZy | and it works ok now |
| 07:54.31 | v0lZy | on those 2 phones |
| 07:54.44 | v0lZy | on other phones it doesnt work |
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| 07:56.13 | v0lZy | i dont really fancy swiching all the phones this way |
| 07:56.13 | v0lZy | but |
| 07:56.42 | v0lZy | whats the root cause of this.. mismatching codecs... i thought the point was to specify all the codecs you are willing to handle |
| 07:56.47 | v0lZy | and recode |
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| 08:23.19 | bulkorok | hi |
| 08:33.32 | davlefou | hi |
| 08:36.22 | davlefou | in sip.conf, we put register line like that : register => user[:secret[:authuser]]@host[:port][/extension], for sql it done in sipregs, but how a put /extenion in sipregs? |
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| 08:54.41 | bulkorok | davlefou: regexten ? |
| 08:57.19 | davlefou | i should be good, i ll try |
| 08:58.15 | davlefou | bulkorok, thank |
| 09:03.55 | davlefou | And how i do for iax register in sql? |
| 09:04.19 | davlefou | i look no table about that. |
| 09:04.41 | bulkorok | check tha asterisk-sources: |
| 09:05.07 | davlefou | sorry? |
| 09:05.18 | bulkorok | ~yoursourcedirectory/asterisk-1.8.23.0/contrib/realtime/mysql/iaxfriends.sql |
| 09:05.39 | bulkorok | there are also sql's for some other tables |
| 09:06.12 | davlefou | ok |
| 09:12.42 | davlefou | but there are no iax sql for postgresql |
| 09:13.40 | davlefou | i can use sip table? |
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| 09:18.49 | bulkorok | setup the fields like the mysql table... |
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| 10:28.30 | v0lZy | i have a weird situation |
| 10:28.39 | v0lZy | where if i calla phone directly, i get a lot of gitter |
| 10:28.43 | v0lZy | but if the person presses transfer |
| 10:28.47 | v0lZy | and then cancels the transfer |
| 10:28.52 | v0lZy | the jitter goes away |
| 10:29.02 | v0lZy | anyone know what the cause of the jitter could be here? |
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| 12:15.38 | Neoti | hey all.. |
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| 12:43.59 | jmetro | Katty: ADORABLE |
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| 12:50.00 | v0lZy | could anyone explain how transfering with moh works... im having a weird issue. |
| 12:50.10 | v0lZy | When i receive a call, im getting a lot of noise |
| 12:50.48 | v0lZy | if i put the user on hold so that moh is played, then cancel the hold so I can again speak with the user, the noise is gone |
| 12:51.42 | v0lZy | happens only with calls routed through ITSP |
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| 12:54.06 | Neoti | v0lZy: could it be being introduced from the itsp? |
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| 12:55.02 | v0lZy | Neoti: well, the ITSP ran a wireshark on my local network and on his |
| 12:55.18 | v0lZy | and he noticed that my asterisk is sending ulaw, alaw etc codecs |
| 12:55.25 | v0lZy | according to him, its supposed to send only one codec |
| 12:55.32 | v0lZy | so we fixed sip.conf to have only alaw now |
| 12:55.57 | v0lZy | and i set some phones to alaw as primary and ulaw as secondary |
| 12:56.14 | v0lZy | and he also said to increase some codec parameters from 20ms to 30 ms |
| 12:56.24 | Neoti | v0lZy: its normal for the asterisk server to offer some codecs so that both parties can pick a condec etc... |
| 12:56.28 | v0lZy | that fixed the problem of users only having 1 way audio when doing a call pickup |
| 12:56.43 | v0lZy | but introduced this other problem with noise, which funny, happens only when the user answers |
| 12:56.45 | v0lZy | ringing is clear |
| 12:56.48 | v0lZy | moh is clear |
| 12:56.54 | v0lZy | but user talking is not |
| 12:57.16 | v0lZy | funnier still, when the user answering the call puts the callee on hold then resumes the conversation, the noise is gone |
| 12:57.29 | v0lZy | so right now everyone's answering their phone, putting the other line on hodld and swithcing back to them |
| 12:57.38 | v0lZy | its stupid, but at least it makes the noise go away |
| 12:58.06 | v0lZy | this i think is an asterisk issue as far as i know, it asks as a back to back user agent not telling the other leg what its doing with the opposite leg |
| 12:58.11 | Neoti | which path is the audio on,,.... from caller to callee or callee to caller ? |
| 12:58.27 | v0lZy | so if its an ITSP thing, what happens with the swithcing to moh and back that fixes things? i think this is something else |
| 12:58.44 | v0lZy | both hear noise |
| 12:58.50 | v0lZy | but in case of one way audio problem |
| 12:59.04 | v0lZy | callee could not hear the caller, but caller could hear the callee just fine. |
| 12:59.53 | Neoti | set the packet size from 30ms to 20 ms again and see what happens ? |
| 13:00.21 | v0lZy | i dont think thats the case either because i only did that on 2 phones |
| 13:00.33 | v0lZy | but the problem is also on the phones that all the while have been set to 20sm |
| 13:01.51 | Neoti | is there any transcoding going on IE from Alaw to gsm etc... ? |
| 13:02.56 | v0lZy | could be, but the machine is more than capbale of handling that |
| 13:02.58 | v0lZy | used to work ok |
| 13:03.03 | v0lZy | i have to run now |
| 13:03.06 | v0lZy | but ill be back |
| 13:03.09 | v0lZy | thanks for the tips neoti |
| 13:03.17 | Neoti | when did it start |
| 13:03.30 | Neoti | what changes, only when the codec change et? |
| 13:04.30 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 13:05.40 | Ice_Strike | Is it possible to use same SIP account on two servers? |
| 13:05.54 | Ice_Strike | to two servers connected to same ISP. |
| 13:06.55 | [TK]D-Fender | Ice_Strike: that's up to your ISP |
| 13:07.19 | Katty | jmetro: i thought so too ^_^ |
| 13:07.28 | Katty | jmetro: still waiting to hear back from the shelter/vet tho :/ |
| 13:07.39 | Katty | jmetro: they have to test him for All The Things |
| 13:13.35 | Neoti | Ice_Strike: normally you can not have two deivces and 1 sip account.... as the asterisk server would not no where to send the call etc... it might send it to A when your at B etc..... |
| 13:13.44 | giovani | can anyone recommend a source for international dids other than voxbone, didlogic, didww, didx (excluding resellers)? |
| 13:14.05 | giovani | any other did exchanges, or, owners of dids (particularly in china)? |
| 13:15.00 | Katty | also, g'morning |
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| 13:44.19 | saxa | hi,I do not know if its the right channel here, but anybody has some experience with Sangoma cards ? |
| 13:44.26 | saxa | I plan to buy one for ISDN |
| 13:44.46 | saxa | the A500RA, does it work under linux/asterisk ? |
| 13:45.24 | saxa | or is it better to go with a Digium one ? |
| 13:46.17 | Neoti | saxa: go with an openvox card... i love them ... |
| 13:47.47 | Neoti | saxa: http://www.novavox.co.uk/asterisk-cards-isdn-cards-pri/e1-(isdn30)-openvox-d110p-port-isdn-prie1-pci-p-43.html |
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| 13:49.11 | saxa | Neoti: thx, let me check if I can get one for a good price. As the distributor where I'm buying it have Digium and Sangoma, need to check if they have those too. |
| 13:49.16 | [TK]D-Fender | Neoti: Wrong tech. he's looking for BRI, not T1/E1/J1 |
| 13:49.34 | [TK]D-Fender | saxa: that is also just an expansion daughtercard, not a " |
| 13:49.37 | [TK]D-Fender | complete" card |
| 13:49.40 | TriJetScud | [TK]D-Fender: can't a PRI card perform the same functions of a BRI? |
| 13:50.07 | saxa | [TK]D-Fender: yep i see, i need basically to set up one isdn incoming line |
| 13:50.18 | saxa | where we will have 10 numbers |
| 13:50.23 | [TK]D-Fender | TriJetScud: I've never heard the DAHDI cards handling that, nor would it be cost-effective per-port |
| 13:50.28 | saxa | any suggestion ? |
| 13:50.51 | [TK]D-Fender | saxa: You're sure it's a single (dual-channel) BRI line? |
| 13:52.22 | saxa | [TK]D-Fender: to be honest no |
| 13:52.45 | saxa | basically they call that in Brasil Digitronco |
| 13:52.45 | Neoti | saxa: [TK]D-Fender: ok sorry for bri http://www.novavox.co.uk/asterisk-cards-isdn-cards-bri-(isdn2)-openvox-b100p-port-isdn-bri-pci-p-40.html i use these cards all time and know they work and are good for the price. |
| 13:53.38 | [TK]D-Fender | saxa: You need to be sure for us to advise sensible products. |
| 13:53.40 | saxa | so afaik this should be a newer isdn line |
| 13:53.58 | [TK]D-Fender | saxa: No such thing as "newer" really. It's a question of the type. |
| 13:54.01 | saxa | yeah i will inform better myself |
| 13:54.13 | [TK]D-Fender | saxa: Go call them and ask them, "is this BRI, or PRI" |
| 13:54.20 | saxa | they told its a R2 type or something |
| 13:54.42 | Neoti | saxa: BRI = ISDN2e = two channels | PRI = ISDN30 Primary rate etc ... |
| 13:54.54 | saxa | ok, thanks for the tips |
| 13:54.59 | saxa | will ask them |
| 13:55.13 | saxa | l8r i'm back on that. |
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| 14:07.07 | kazak1377_ | Hello everybody. Can someone, pls, say me how can i block playbackskip for certain button. For example, i'm playing some sound and any button is skipping it, but not an 5 button |
| 14:08.18 | [TK]D-Fender | kazak1377_: pastebin your latest call attempt |
| 14:08.20 | [TK]D-Fender | ~pb |
| 14:08.20 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:08.22 | [TK]D-Fender | ^^^ |
| 14:10.26 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:10.34 | kazak1377_ | thereis no call attempt. I'm only developing that feature. |
| 14:11.13 | [TK]D-Fender | kazak1377_: what kind of "devloping"? |
| 14:11.30 | kazak1377_ | writing macros for it |
| 14:11.49 | [TK]D-Fender | kazak1377_: sounds like basic dialplan. So show us what you're doing. |
| 14:11.54 | kaldemar | define playbackskip |
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| 14:13.30 | kazak1377_ | [TK]D-Fender: i need to allow to skip playing current audio with all buttons,except one, that i specified. |
| 14:14.19 | [TK]D-Fender | kazak1377_: So show us how you're trying to do it now. |
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| 14:14.47 | kazak1377_ | that is the problem: i don't have any idea how to do that. |
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| 14:15.22 | kaldemar | only way to do that in dialplan is to use the m option for app Background and make one-digit extensions in the context. |
| 14:15.59 | Katty | looks in |
| 14:16.24 | kazak1377_ | kaldemar: hm... I'll try that. Thanks. |
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| 14:18.50 | [TK]D-Fender | kazak1377_: what "current audio"? |
| 14:19.03 | [TK]D-Fender | kazak1377_: because so far I'm suspecting that you have other options... |
| 14:21.01 | kazak1377_ | it doesn't metter. That is IVR main-menu soud, which is loks like basic IVR, but ignoring certain button, until Playback finished |
| 14:21.08 | kazak1377_ | sound* |
| 14:21.38 | [TK]D-Fender | kazak1377_: then you SHOULD be using "Background" to play that back. |
| 14:22.08 | [TK]D-Fender | kazak1377_: and that isnt using a digit to stop the playback.. that's using a digit to actually change extensions |
| 14:22.41 | [TK]D-Fender | kazak1377_: are you just trying to fix a basic IVR that was waiting until the playback finished, or were you doing something bigger? |
| 14:23.40 | kazak1377_ | Hm. I'm adding some new features in big IVR. |
| 14:24.12 | [TK]D-Fender | kazak1377_: then you just used the wrong tool to play the audio, and "background" should be all you need to do instead |
| 14:25.50 | kazak1377_ | Hm... I know, but i'm not the first developer. Some macroses was writed before me. |
| 14:26.14 | kazak1377_ | And previous developer used Playback :-/ |
| 14:28.19 | [TK]D-Fender | kazak1377_: macro's are not used to make IVR's, and I suspec you are mixing some terms, or the actual usage up in your description. |
| 14:28.24 | kaldemar | playback can't be exited with a key. |
| 14:30.08 | [TK]D-Fender | kaldemar: So far his need is vague between needing a key JUST to exit the playback... or if he has a real active IVR he wants to be interuptable |
| 14:30.15 | [TK]D-Fender | (by using background) |
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| 14:33.57 | kazak1377_ | [TK]D-Fender: hm. Maybe, i'm mixing some terms, but its an something like IVR for an mobile operator. Where we have to get user's info from post respones, play lots of files with dozens of variables... So, without macroses it would be an terrible code.;) |
| 14:34.37 | kazak1377_ | I need to option opposite to m in Background |
| 14:35.03 | kazak1377_ | that option is interrupting, but i need it to be ignorred. |
| 14:35.03 | kaldemar | [TK]D-Fender: that's why i said he should use Background. other apps don't have a controlled exit with a key. |
| 14:35.21 | [TK]D-Fender | kaldemar: Incorrect. There is. |
| 14:35.33 | kaldemar | [TK]D-Fender: the m option allows break only if the context has a matching one-digit extension. |
| 14:35.56 | [TK]D-Fender | kazak1377_: So you want to play back a SERIES of files, where you can skip from one to the next to the next by key? |
| 14:35.57 | kaldemar | [TK]D-Fender: which one? |
| 14:37.50 | kazak1377_ | [TK]D-Fender: macro plays only one file. But i need to ignore an pressing of certain button. And extension could be more than one digit |
| 14:38.03 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-111-87.user.veloxzone.com.br) |
| 14:38.06 | [TK]D-Fender | kazak1377_: I just asked what you want to do. Not "what is it doing now" |
| 14:38.18 | [TK]D-Fender | kazak1377_: Is what I described what you want to do? |
| 14:38.48 | kazak1377_ | Yep. There is right description |
| 14:38.54 | [TK]D-Fender | kazak1377_: "core show application controlplayback" <--------------- |
| 14:38.57 | [TK]D-Fender | kaldemar: ^^^ |
| 14:41.22 | kazak1377_ | [TK]D-Fender: hm... That's sounds good. )) |
| 14:41.27 | kazak1377_ | Thanks) |
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| 14:42.20 | kaldemar | [TK]D-Fender: i don't see how that can be exited with a set of keys that is "all but one". unless the documentation lacks the option of using multiple digits as a stop. |
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| 14:43.38 | *** join/#asterisk proute (~proute@146.255.170.33) |
| 14:45.10 | [TK]D-Fender | kaldemar: does allow at least the skip, and the working is loose enough to suggest it might allow multiple. at the very least it will give him the rest and he can train his monkeys :) |
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| 14:46.49 | proute | Hi, I use Asterisk 1.8.23.1 with srvlookup=yes. I'm registering Asterisk to a provider with multi IP address via SIP SRV. So I red on Internet that Asterisk is not able to perform the srv lookup. |
| 14:47.01 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 14:47.16 | proute | So How can I configure Asterisk to register to other IP address if the first failed? |
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| 14:50.10 | qakhan | can we pass data to store procedure in func_odbc.conf? |
| 14:56.38 | boom^time | qakhan, yes. |
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| 14:59.43 | anonymouz666 | OK, guys. The time to virtualize Asterisk has come. |
| 14:59.57 | jmetro | already done |
| 15:00.06 | anonymouz666 | yeah I know I am late |
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| 15:00.43 | _Corey_ | anonymouz666: At least you waited until it's easy and free of exotic kernel patches |
| 15:01.36 | crankyadmin | Hi. Got a strange issue. Outbound calls will sometimes go right to the users **cell** phone voicemail, other times rings the as normal. This happens **with** full cell service. Any thoughts? |
| 15:01.40 | Neoti | proute: srv is there so you can give an address like something.com to some one and asterisk then lookups up the IP(s) for that domain etc.... asterisk will register against only 1 of these IPs if it trys to register again and fails i will query the srv record and find anonther IP to register to .. |
| 15:02.20 | jmetro | crankyadmin the user declines the call, etc etc etc. have them confirm the call and it will come back to your system. |
| 15:03.31 | crankyadmin | We don't get a option to accept or decline the call. It just goes right to it. Only seems to happen in the US as well... |
| 15:03.32 | proute | Neoti: Thanks for your answer. But When I put srvlookup=yes,, if Asterisk can't to register to the first IP address, Asterisk doesn't try to register to the second entry of DNS SRV... |
| 15:04.00 | SuperNull | offtopic: anyone know if adtran ta900 series mass ATA mark rtp packets? |
| 15:04.12 | qakhan | boom^time can you give me an example of this |
| 15:04.32 | qakhan | in func_odbc.ini and extensions.conf |
| 15:04.34 | Neoti | proute: humm.... could be the address or incorrectly configured srv on there DNS .... my asterisk 1.8 cert 10 does it. |
| 15:06.17 | jmetro | crankyadmin: so give them an option to accept/decline and it wont happen again |
| 15:06.34 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 15:06.56 | proute | Neoti; When I type host -t SRV _sip._udp.mydomain.com, the result show me 2 entries. |
| 15:07.21 | proute | Neoti: How can I know if my DNS entries are right? |
| 15:08.11 | Neoti | proute: i followed this http://www.voip-info.org/wiki/view/DNS+SRV |
| 15:08.43 | crankyadmin | I can't. Asterisk auto dials out (this works fine). But sometimes the call goes to their (end users) cell phone voice mail (not the asterisk voicemail) others the call goes through as normal. Asterisk has no control over a remote cell phone sitting on AT&T. Its almost like AT&T just won't let the call go through. |
| 15:09.16 | Neoti | proute: if i switch one asterisk server off it fails over ... works for me.... |
| 15:09.26 | proute | Neoti: Thanks, I will read tis topic |
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| 15:11.33 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:13.16 | [TK]D-Fender | crankyadmin: So dial them with a confirmation macro |
| 15:13.42 | [TK]D-Fender | crankyadmin: to validate that it's a human |
| 15:14.29 | crankyadmin | I'd really like to work out why it happens... as its random as random gets. But that maybe my best option. |
| 15:14.32 | Neoti | crankyadmin: [TK]D-Fender: yep do that .... "Please press 1 to accept the call" etc... then you know a human answered it ... or it not it went to vm |
| 15:15.20 | [TK]D-Fender | crankyadmin: Easily variable based on cell tower response time, etc impacting your "total time to dial" vs dial timeout |
| 15:15.59 | crankyadmin | [TK]D-Fender: Does that roughly translate into "You'll probably never find out"? |
| 15:16.14 | Neoti | ^ lolz |
| 15:16.16 | [TK]D-Fender | crankyadmin: pick up a phone and TRY |
| 15:16.29 | [TK]D-Fender | crankyadmin: brain + math = answer |
| 15:16.53 | jmetro | crankyadmin: the reason it happens is user declining the call, cell phone is off, SOMETHING didnt ring the phone or couldnt reach it - but was able to hit the carrier and ring the VM they have. |
| 15:17.09 | coppice | pick up a phone and TRY throwing it at Ballmer |
| 15:18.13 | crankyadmin | jmetro: yep. I get it. The thing is my phone is sat in front of me and it does it as well. But I can ring it from my desk phone fine. |
| 15:18.31 | [TK]D-Fender | coppice: chairs would be best for imapct & irony |
| 15:18.35 | [TK]D-Fender | impact* |
| 15:18.46 | jmetro | [TK]D-Fender: Developers Developers Developers Developers |
| 15:18.53 | coppice | nah. it was phones that finished him off |
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| 15:21.11 | anonymouz666 | jmetro: hahah |
| 15:21.14 | anonymouz666 | developers! |
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| 15:21.35 | Katty | waves to coppice |
| 15:22.23 | anonymouz666 | last week I was reading a post by coppice about audio codecs from 2004... asterisk-users |
| 15:23.49 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 15:25.14 | coppice | Its the same thing that brought down Kodak: Developers Developers Developers Developers, and a bit of film and printing paper |
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| 15:27.09 | monsterco | Hi everyone - on an Aastra SIP phone, when does a Backup Server configuration kick in? I have set that but it seems to not take effect even after 5 minutes that the main server is not available |
| 15:28.02 | [TK]D-Fender | monsterco: Easy guess : after a registration failure or failed call attempt |
| 15:28.07 | *** join/#asterisk gtj (~george.jo@ernie.fairview5.com) |
| 15:28.12 | [TK]D-Fender | monsterco: So set the registration interval smaller |
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| 15:38.08 | monsterco | Registration Failed Retry Timer |
| 15:38.09 | monsterco | Registration Timeout Retry Timer |
| 15:38.09 | monsterco | Registration Renewal Timer |
| 15:38.19 | monsterco | I guess the first one ^^^^ [TK]D-Fender> |
| 15:38.33 | monsterco | There is no mention of Backup register try interval |
| 15:38.47 | [TK]D-Fender | THIRD |
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| 15:39.42 | Penguin | It's not going to have a timer for the backup. It has a timer for how often it tries again. At failure on the primary, it will try the backup. |
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| 16:59.34 | boom^time | I'm doing a really simple gsm playback of "en/tt-weasels&&en/silence/10&&en/tt-weasels" and it's first attempting them with ulaw and giving me warnings. Is there anyway to remove these or it is it trying to avoid transcoding by looking for a ulaw version and I just need to put up with them? |
| 16:59.46 | wltjr | dude polycom is a joke, this idiot at level 2 is saying my tcpdump output is no good he wants wireshark output... fool doesn't realize both use libpcap and its maintained by tcpdump.... |
| 17:00.25 | wltjr | their phone/firmare is sending empty stuff Authorization: Digest username="", realm="asterisk", |
| 17:00.52 | wltjr | whats a good vendor, no cisco no polycom, snom? digium? grandstream? |
| 17:02.16 | boom^time | wait nm, tt-weasels.ulaw does exist. why would I be getting this warning... not to mention it's playing the file. |
| 17:03.38 | wltjr | tried 3 diff firmware, all same thing, but at one point the original firmware allowed a few lines to register, tried 4.1.4.7430, 4.1.5.3071, and 5.1.4.0844 |
| 17:04.08 | boom^time | haha, I had my head in C/C++ mode. Twas trying to && them. Oh well. |
| 17:04.17 | qakhan | what is the store procedure syntax in func_odbc.conf |
| 17:05.00 | _Corey_ | wltjr: Well, Digium's phones are good if you're going for a change. They're priced well and buying them pretty directly pays for Asterisk development.. Polycoms will work fine too. |
| 17:05.08 | boom^time | https://www.google.com/#q=func_odbc |
| 17:11.12 | wltjr | _c_c |
| 17:11.34 | jmetro | boom^time: http://lmgtfy.com/ |
| 17:11.36 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
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| 17:12.09 | wltjr | _Corey_: I might check them out, can't believe the issues with Cisco and Polycom, Cisco was my fault with going with SPA series stuff, junk, by with Polycom just shocked, and need to start buying digium stuff haven't since analog cards years ago |
| 17:12.41 | boom^time | jmetro, thank you. |
| 17:12.42 | wltjr | was using sangoma wanpipe cards after digium, I liked them for non-voip purposes a bit more, but do need to send some $ digiums way to keep * going |
| 17:14.55 | _Corey_ | wltjr: Well, Digium makes legitimately good phones. The others you describe, including the Cisco SPA stuff, run pretty mature firmware and are very widely used. |
| 17:16.48 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 17:17.38 | jmetro | Aastras have my vote for favorite, but nothing beats the speakerphone on a polycom |
| 17:18.00 | *** join/#asterisk diatonic (~diatonic@unaffiliated/diatonic) |
| 17:18.02 | [TK]D-Fender | Cisco SPA are fairly decent. NON-SPA = trouble Cisco did not care about SIP when designing & maintaining those. Polycom is pretty damn rock solid. Digium's phones are really only interesting to me for very specific applications |
| 17:18.38 | *** join/#asterisk Guest99659 (~Freeaqing@91.214.168.110) |
| 17:19.02 | wltjr | SPA IMHO is crap, it has a feature called call waiting, if you disable that the phone goes from > 1 line, to 1 line only... I spent 1 month diagnosing that and providing Cisco all kinds of information, logs, tcpdump, configs, etc they took a feature request I consider it a bug |
| 17:19.33 | wltjr | also if phone is ringing and one extension stops and another starts, if you pick up, sometimes you have to press the button for the ringing line vs just picking up the phone, there are lots of issues |
| 17:19.35 | jmetro | SPA's arent really cisco, so im not surprised cisco doesnt know the phone. |
| 17:19.56 | wltjr | not to mention SPA-501g has NO lcd, so when you get 2 calls on 1 line, you have no way to manage |
| 17:20.20 | wltjr | yes this SPA problem goes back long time to Sipura, I have a linksys phone spa-942 all the same issues, Cisco refuses to address |
| 17:20.52 | wltjr | on a clients 7960 I disabled call waiting on all lines and the phone remains a multi-line phone... |
| 17:21.10 | wltjr | I even sent that over to Cisco showing that SPA phones do not behave as real cisco ones, they care less |
| 17:21.40 | wltjr | but this polycom problem with registration out of the box is amazing, and googling, there are people with the same problem on 3.x version of the firmware so its clearly recurring with polycom sadly |
| 17:21.54 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 17:22.18 | [TK]D-Fender | [13:19]wltjrSPA IMHO is crap, it has a feature called call waiting, if you disable that the phone goes from > 1 line, to 1 line only... I spent 1 month diagnosing that and providing Cisco all kinds of information, logs, tcpdump, configs, etc they took a feature request I consider it a bug <- that was your misinterpretation. |
| 17:22.18 | wltjr | Google polycom and username mismatch, have <21>, digest has <> |
| 17:22.39 | [TK]D-Fender | wltjrnot to mention SPA-501g has NO lcd, so when you get 2 calls on 1 line, you have no way to manage <- yes, that is a sucky low-end model. No-one is forcing you to buy it |
| 17:22.54 | wltjr | you will see others with same issue sadly... one guy with a 301 polycom was like its EOL basically go screw yourself... |
| 17:23.05 | [TK]D-Fender | [13:20]wltjron a clients 7960 I disabled call waiting on all lines and the phone remains a multi-line phone...<- 79XX - SIP = Afterthought |
| 17:23.18 | wltjr | [TK]D-Fender: big mistake with the 501g clearly, but its the call waiting feature that makes everything problematic |
| 17:23.26 | [TK]D-Fender | wltjryou will see others with same issue sadly... one guy with a 301 polycom was like its EOL basically go screw yourself...What was wrong with it? I've had 2 and they work fine... |
| 17:23.42 | wltjr | [TK]D-Fender: yes but sip works great on the 7960, but I know you have to get firmware for that, I messed with sccp/skinny stuff before ugh... |
| 17:24.01 | wltjr | [TK]D-Fender: the polycom? damn thing is sending empty username in digest |
| 17:24.12 | [TK]D-Fender | wltjr: What model is doing that? |
| 17:24.16 | wltjr | [TK]D-Fender: Authorization: Digest username="", realm="asterisk" |
| 17:24.24 | wltjr | [TK]D-Fender: vvx 300 |
| 17:24.45 | [TK]D-Fender | wltjr: Ah, I've used all the SoundPoint IP's, not VVX. I know they have been a little q"quirky" |
| 17:24.51 | wltjr | [TK]D-Fender: tried 3 diff firmware same issue, I have open ticket with polycom, lvl 2 tech pissed me off so called in to make some noise, about to return them |
| 17:25.13 | wltjr | [TK]D-Fender: again I failed should have just gone with soundpoint, but the vvx phones seem really nice |
| 17:25.34 | wltjr | the web interface is nice, I much like Polycom uc/provisioning to Cisco, much cleaner xml and less |
| 17:25.47 | *** join/#asterisk BriGuy (~BriGuy@74.115.41.6) |
| 17:25.55 | wltjr | though not many options to control provisionining, keeps looking for a bunch of other files I will never have |
| 17:26.12 | wltjr | really want ot use the vvx 300, if they ever register :( |
| 17:27.00 | BriGuy | can anyone tell me why a queue with local channel members will not retry again if all members are busy (rrmemory, retry set to 5, timeout set to 10) |
| 17:27.06 | wltjr | lesson learned if you going with polycom or cisco, spend the $ and get the good stuff $200+ |
| 17:27.30 | [TK]D-Fender | wltjr: Probably just a configuration misinterpretation... |
| 17:27.30 | jmetro | BriGuy: do you have ringbusy set to yes |
| 17:27.49 | jmetro | or ringinuse whatever it is |
| 17:27.52 | [TK]D-Fender | BriGuy: Show us. Also... some details would be nice... |
| 17:28.15 | BriGuy | I was trying to keep that to off, as queue members were getting multiple calls when they were free |
| 17:28.21 | [TK]D-Fender | BriGuy: And if they are busy.. it SHOULDn'T retry by default. |
| 17:28.43 | BriGuy | the calls sit in the queue though, as soon as I run "queue show" a free members phone will ring |
| 17:28.55 | jmetro | BriGuy: so what you are saying is "I have a queue full of members, ringinuse set to no, when the members are in use, they wont get rung" |
| 17:28.58 | jmetro | working as expected. |
| 17:29.42 | BriGuy | it is like it wont get retriggered until I run queue show - the member is free |
| 17:30.04 | BriGuy | as in after the queue member hangs up, they do not get a new call, even if they are available - until I do that |
| 17:30.20 | BriGuy | the members are not in use |
| 17:31.51 | BriGuy | so what would make the queue application retry the members then after they hang up and have no active call then? |
| 17:32.42 | BriGuy | running debug in the console - the queue application never seems reset the local channel member after they are determined busy by another call |
| 17:33.33 | diatonic | Hmm.. I'm using ChanSpy on asterisk 1.8 and the x option is supposed to allow me to specify a DTMF digit to exit the application, but it only works if there is an acrive call on the channel I'm spying on :/ |
| 17:34.02 | BriGuy | is there any other information I could provide? or is someone familiar with it? |
| 17:40.45 | diatonic | My dialplan code: http://pastie.org/8370215 |
| 17:40.53 | BriGuy | jmetro: just tested ringinuse and it didn't change a thing |
| 17:43.30 | *** join/#asterisk undecided (~kvirc@49.156.43.10) |
| 17:46.30 | undecided | is it possible to use ExtenSpy() with a spygroup (list of extensions), or just bind it to a specific context? |
| 17:50.24 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 17:51.46 | diatonic | undecided, you can use enforced mode to add a colon delimited list of extensions or specify a group or groups to limit it to |
| 17:54.00 | diatonic | undecided, http://pastebin.com/wPcD6bgs |
| 17:54.00 | undecided | diatonic: thanks. Would you mind linking me to a synax example. On asteriskdocs I can't see reference to such a mod |
| 17:56.05 | undecided | isn't ${SPYGROUP} only work with channel names (as part of chanspy app)? |
| 17:56.11 | *** join/#asterisk adnc (~akif@unaffiliated/adnc) |
| 17:57.07 | undecided | will it work at all with extensions names? |
| 17:57.56 | adnc | hello, is it possible to rewrite caller id and place +49 etc. before the callerid when country code is missing |
| 17:58.04 | [TK]D-Fender | adnc: Sure |
| 17:58.23 | [TK]D-Fender | adnc: It's your dialplan... do whatever you want |
| 17:59.24 | jmetro | hey i rewrite callerID to let my users know which ringgroup its coming from, why not |
| 18:00.54 | adnc | incomming aswell? |
| 18:01.21 | [TK]D-Fender | adnc: is there any other kind? |
| 18:01.27 | adnc | and how do I catch missing country codes? |
| 18:01.31 | adnc | outgoing? |
| 18:01.54 | adnc | is [TK]D-Fender a bot? |
| 18:01.54 | [TK]D-Fender | adnc: Call comes to *. It has CID. Dialplan processes and maybe you change it... STILL has CID. Dialpln happens to dial out... uses whatever the CID happens to be right then and there |
| 18:02.40 | jmetro | adnc: he is a bot, he just googles your question if it has a ? and provides the first answer |
| 18:03.00 | [TK]D-Fender | GoogleProxy++ |
| 18:03.02 | adnc | why don't you just simply close that channel and write google |
| 18:03.26 | *** part/#asterisk adnc (~akif@unaffiliated/adnc) |
| 18:03.44 | [TK]D-Fender | adnc: I've answered your question extremely directly. What's the issue? |
| 18:04.03 | [TK]D-Fender | Some people just can't take "yes" for an answer... |
| 18:04.07 | [TK]D-Fender | *sheesh* |
| 18:04.14 | jmetro | He left |
| 18:04.17 | [TK]D-Fender | I know |
| 18:04.32 | jmetro | i was going to say you were heuristically programmed to respond in a hostile manner to anyone asking to shut you down. |
| 18:04.44 | [TK]D-Fender | Notice tafter the line I was typing and he left it wasn't addressed to him anymore |
| 18:05.05 | [TK]D-Fender | jmetro: Nothing I've said should have any reason to be interpreted as "hostile". |
| 18:05.14 | undecided | OK. I just verified ${SPYGROUP} isn't working with extensions names in ExtenSpy (since it expects channel names) |
| 18:05.44 | [TK]D-Fender | jmetro: Asked if he coud. I said yes. he asked "inbound", as it there were any other kind. I told him, yes, that's how it works, and that is what is use for outbound after. |
| 18:06.09 | [TK]D-Fender | jmetro: Some people just have persecution complexes |
| 18:06.31 | jmetro | It was a pretty simple fix too. |
| 18:07.24 | undecided | let me try another approach: |
| 18:07.28 | undecided | Is it possible to exclude a channel in Chanspy() ?? |
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| 18:19.25 | navaismo | can anybody answer to undecided ? |
| 18:21.12 | boom^time | That was entertaining to back-read. |
| 18:21.13 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 18:21.34 | boom^time | I had to after '<jmetro> i was going to say you were heuristically programmed to respond in a hostile manner to anyone asking to shut you down.' |
| 18:23.50 | [TK]D-Fender | I'M SORRY DAVE, I CAN'T LET YOU DO THAT |
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| 18:24.41 | jmetro | Lol =) |
| 18:25.07 | Katty | i miss dave. |
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| 18:32.00 | undecided | it seems people don't use these applications much.. |
| 18:43.42 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.13) |
| 18:46.51 | navaismo | many people use chanspy |
| 18:47.04 | navaismo | but what are you trying to do? |
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| 18:52.04 | undecided | navaismo: I'm trying to create an extensions list for ExtenSpy() applications |
| 18:52.16 | undecided | it seems I found my answer |
| 18:52.36 | undecided | @dialtonic was close to the answer |
| 18:53.30 | undecided | need to use the e(ext) option for Extenspy() |
| 18:53.54 | undecided | to get the enforce mode enabled |
| 18:54.19 | undecided | and specify list of extensions inside brackets |
| 18:55.14 | undecided | but still not clear to me how to form the syntax |
| 18:55.47 | undecided | I mean |
| 18:56.56 | undecided | if I enter a list of extensions for example e(123:234:123) as an OPTION for extenspy |
| 18:57.28 | undecided | what should I enter as a main argument for the application? |
| 18:57.35 | undecided | can it be left empty? |
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| 18:59.40 | undecided | ? |
| 19:00.01 | undecided | even on Google I see very few people use that app |
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| 19:18.21 | navaismo | have your tried with ExtenSpy(SIP/,e(1000:1001:1002)) |
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| 19:22.14 | undecided | I'll try now |
| 19:22.51 | undecided | so far I tried exten => 123,1,ExtenSpy(,e(11@default:12@default)) - but didn't work |
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| 19:24.04 | navaismo | those extens exists? |
| 19:24.37 | undecided | of course |
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| 19:28.07 | undecided | navaismo: no, it didn't work |
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| 19:33.13 | boom^time | [TK]D-Fender, my coffee cup next to me has HAL on it and he is saying "Stop, Dave. I'm afraid." |
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| 19:34.12 | [TK]D-Fender | boom^time: http://weknowmemes.com/wp-content/uploads/2013/08/im-afraid-i-cant-brew-that.jpg |
| 19:34.59 | boom^time | haha, aw my keurig has no dot matrix lcd. |
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| 19:55.05 | qakhan | i m working on IVR. i am take date from caller as MM/DD/YYYY |
| 19:55.34 | qakhan | is there any way i can check caller entered correct date format |
| 19:55.58 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 19:56.44 | [TK]D-Fender | qakhan: Go look at the input. |
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| 20:05.49 | qakhan | [TK]D-Fender can you give me more detail on it |
| 20:06.56 | [TK]D-Fender | qakhan: What is there to give more detail on? Go look at what the user entered. |
| 20:07.11 | [TK]D-Fender | qakhan: "core show application gotoif" <------------- |
| 20:09.26 | qakhan | i understand this, ok let say i ask caller to enter date as MM/DD/YYYY |
| 20:09.39 | qakhan | but he enter DD/MM/YYYY |
| 20:10.18 | qakhan | here i want to check and ask him to enter correct date format |
| 20:10.31 | qakhan | as MM/DD/YYYY |
| 20:10.33 | [TK]D-Fender | qakhan: So go look at what he entered |
| 20:12.45 | qakhan | ok thanks |
| 20:15.02 | navaismo | ok this snom is drive me crazy---> set nat settings to force_rport,comedia, registered with public address but in the sdp the phone use the local ip address |
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| 20:18.49 | [TK]D-Fender | navaismo: PB it |
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| 20:23.06 | navaismo | here http://pastebin.com/cvbbK8qq |
| 20:26.55 | [TK]D-Fender | navaismo: and with "nat=yes"? |
| 20:27.19 | navaismo | same |
| 20:27.54 | [TK]D-Fender | very odd |
| 20:28.32 | navaismo | yes |
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| 20:28.56 | navaismo | the other is if I enable the stun on the phone side, i cant call, the invite never reach the server |
| 20:29.06 | navaismo | time to upgrade firmware i guess |
| 20:29.21 | [TK]D-Fender | <PROTECTED> |
| 20:29.25 | [TK]D-Fender | give it a shot |
| 20:29.47 | [TK]D-Fender | And it shouldn't be the phone's fault... You should be able to tell * to ignore the SDP IP. Something buggered up |
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| 20:29.57 | navaismo | checking... |
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| 20:32.08 | [TK]D-Fender | checkout time, heading home, BBIAB |
| 20:32.14 | [TK]D-Fender | Let me know when you see me log back |
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| 21:17.58 | lnb_ | problem with faxing. ATA <-> ITSP inbound/outbound OK, ATA <-> PBX <-> ITSP Outbound OK/Inbound Not working, ATA -> PBX -> ITSP -> PBX internal fax2email Inbound / Outbound OK. How can i find out what is screwing up inbound from PBX to ATA ? |
| 21:17.59 | *** join/#asterisk pfui (c6d3635d@gateway/web/freenode/ip.198.211.99.93) |
| 21:20.13 | pfui | do I need to enable a special setting on an asterisk SIP client, or perhaps the server, to allow SIP dialing by IP? calling Dial(SIP/ipaddr) doesn't seem to work for me |
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| 21:24.08 | WIMPy | Try sip/user@host |
| 21:34.08 | pfui | WIMPy: doesn't work either |
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| 21:35.22 | WIMPy | sip set debug on and show us |
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| 21:40.24 | pfui | WIMPy: http://pastebin.com/0JwBjvrC |
| 21:40.39 | pfui | WIMPy: and nothing comes up on the server side |
| 21:41.46 | WIMPy | And where does that not work? |
| 21:41.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.103) |
| 21:42.00 | pfui | WIMPy: excuse me? |
| 21:42.03 | WIMPy | It clearly calls N7@198.211.99.93 |
| 21:42.20 | pfui | WIMPy: I could see that, but then nothing happens at the server |
| 21:42.36 | pfui | WIMPy: but if I create a peer for that device, calls go through fine |
| 21:42.51 | WIMPy | Then check waht's going on on that server or in between. |
| 21:43.22 | WIMPy | So what's the difference then? |
| 21:44.00 | pfui | WIMPy: that's why I'm asking |
| 21:44.15 | pfui | WIMPy: if it were a network issue, calls wouldn't go through either way |
| 21:44.31 | pfui | WIMPy: but for one kind to work and for the other to fail is just weird... |
| 21:44.34 | WIMPy | You configured that peer. |
| 21:44.50 | [TK]D-Fender | pfui: Contact: <sip:01069672650@192.168.1.3:5060> |
| 21:45.11 | pfui | [TK]D-Fender: ? |
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| 21:45.14 | [TK]D-Fender | pfui: Your outbound call is also passing your server's LOCAL IP because it was misconfigured |
| 21:46.29 | pfui | [TK]D-Fender: server side has nat=force_rport,comedia |
| 21:46.38 | [TK]D-Fender | pfui: you ar missing necessary SIP settings |
| 21:47.14 | pfui | [TK]D-Fender: even then, shouldn't I expect to see anything on the server about the client's connection attempts even if the server can't call back? |
| 21:47.58 | [TK]D-Fender | pfui: I haven't seen the firewalls in between... or dumps from the receiving end yet |
| 21:48.25 | navaismo | [TK]D-Fender, same with auto_comedia also i rollbakc to 11.4.0 same maybe im overlooking a bad setting-->http://pastebin.com/HFQTErfb |
| 21:48.30 | pfui | [TK]D-Fender: absolutely nothing registers on the receiving end :| |
| 21:50.06 | [TK]D-Fender | pfui: Retransmitting #3 (no NAT) to 198.211.99.93:5060: <- is that the right IP? Is this an exposed server? What about forwarding? Firewall on the server itsself>? |
| 21:51.08 | pfui | [TK]D-Fender: yep, correct server and port. It's weird because if I configure a peer in sip.conf with that IP address as a host and then Dial(SIP/peer) it just works... |
| 21:51.26 | pfui | which makes absolutely no sense whatsoever to me. |
| 21:51.56 | [TK]D-Fender | show us |
| 21:52.06 | pfui | hmm? |
| 21:53.00 | pfui | [TK]D-Fender: try registering, tcp://rpi:foobar@198.211.99.93 |
| 21:53.43 | [TK]D-Fender | pfui: Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK07b1181d |
| 21:53.49 | [TK]D-Fender | pfui: that call is going out ***UDP*** |
| 21:54.00 | [TK]D-Fender | pfui: Whyh are you even suggesting TCP there? |
| 21:54.07 | pfui | [TK]D-Fender: damn it... |
| 21:54.11 | [TK]D-Fender | pfui: BIG PRINT |
| 21:54.16 | [TK]D-Fender | \o/ |
| 21:54.37 | pfui | [TK]D-Fender: I thought Dial would respect the transport directive in sip.conf |
| 21:54.49 | [TK]D-Fender | pfui: Oh? Which? |
| 21:54.59 | [TK]D-Fender | pfui: Because UDP is clearly a valid one... |
| 21:55.02 | pfui | [TK]D-Fender: yeah, big print in the middle of a tonne of seemingly gibberish text |
| 21:55.17 | pfui | [TK]D-Fender: I have "transport=tcp" in my sip.conf |
| 21:55.30 | [TK]D-Fender | pfui: Show your whole config there.,.. |
| 21:55.39 | wltjr | anyone know why * would send 403 in registration? |
| 21:56.02 | [TK]D-Fender | wltjr: "forbidden", just like it says. PB the debug and peer |
| 21:56.08 | [TK]D-Fender | Mask ONLY the secret |
| 21:56.18 | pfui | [TK]D-Fender: http://pastebin.com/v8C59w7h |
| 21:56.50 | wltjr | [TK]D-Fender: huh :) whats pb |
| 21:57.03 | pfui | [TK]D-Fender: so, how do I tell Dial to use tcp when dialing a SIP uri? |
| 21:58.07 | [TK]D-Fender | pfui: transport=udp ; Set the default transports. The order determines the primary default transport. ; If tcpenable=no and the transport set is tcp, we will fallback to UDP. |
| 21:58.09 | wltjr | oh duh pastbins sry fried |
| 21:58.18 | ChannelZ-Wk | You don't.. it uses whatever SIP is configged to do |
| 21:58.21 | [TK]D-Fender | pfui: You should specify both in order to be explicit |
| 21:58.33 | ChannelZ-Wk | what he said |
| 21:59.29 | pfui | [TK]D-Fender: so there's no way to tell it to only use tcp? |
| 22:00.00 | wltjr | [TK]D-Fender: http://pastebin.com/9d32DYsN ty also on phone with polycom.... |
| 22:01.38 | [TK]D-Fender | pfui: I've never heard of anyone trying to use ONLY TCP |
| 22:01.52 | [TK]D-Fender | pfui: But I would go follow the rest of th rules... |
| 22:02.45 | [TK]D-Fender | wltjr: I also asked for your PEER config in there... And while you're at #2... add "sip show peers" as well |
| 22:03.19 | wltjr | [TK]D-Fender: sry one sec |
| 22:03.52 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:04.40 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 22:06.04 | wltjr | [TK]D-Fender: peers and config http://pastebin.com/fBf9W0Ut |
| 22:06.28 | pfui | [TK]D-Fender: Dial(SIP/user:pwd:::tcp@ipaddr) work! :D |
| 22:06.33 | pfui | [TK]D-Fender: thanks! |
| 22:06.38 | pfui | *works |
| 22:07.01 | [TK]D-Fender | pfui: You should REALLY make a proper peer for this instead. |
| 22:07.13 | pfui | [TK]D-Fender: why so? |
| 22:07.19 | [TK]D-Fender | wltjr: 21/21 10.4.2.5 D a 5060 OK (23 ms) <- this phone seems to be registering to ANOTHER extension as well..... |
| 22:07.37 | [TK]D-Fender | pfui: becasue the password is printed right in CLI and is one more level of exposure, including your log files. |
| 22:07.57 | wltjr | [TK]D-Fender: huh? it has 6 lines/extensions 21-26 only 21 registers the others fail |
| 22:08.04 | [TK]D-Fender | wltjr: I'd look really close at the polycom configs as well |
| 22:08.22 | wltjr | [TK]D-Fender: I am on the phone with them sending them all sorts of packet output |
| 22:08.23 | [TK]D-Fender | wltjr: probably because 21 makes it through so a digest gets set |
| 22:08.49 | wltjr | [TK]D-Fender: I can register 1 line on each phone but not more than 1, though when I had 6 I think 2 registered |
| 22:09.27 | [TK]D-Fender | wltjr: PB your [general] section as well. |
| 22:10.04 | wltjr | [TK]D-Fender: sip.conf general |
| 22:10.21 | [TK]D-Fender | yes |
| 22:11.27 | wltjr | [TK]D-Fender: http://pastebin.com/RY7tnq9D |
| 22:11.48 | wltjr | [TK]D-Fender: ty for your help, pulling my hair out here, and polycom seems to have no clue as well... quite surprised should be basic stuff |
| 22:12.35 | [TK]D-Fender | wltjr: First set each of your entries as "insecure=no" |
| 22:12.37 | wltjr | [TK]D-Fender: fyi I am replacing cisco spa501g phones with the polycom vvx 300, existing system |
| 22:13.19 | [TK]D-Fender | wltjr: retest. Upon failure, go pastebin your polycom configs for that phone |
| 22:14.53 | wltjr | [TK]D-Fender: done, here is a reset config... :) http://pastebin.com/EWFCcqYH |
| 22:15.18 | *** join/#asterisk jetlag (~jetlag@pool-71-168-241-76.cmdnnj.east.verizon.net) |
| 22:15.41 | wltjr | [TK]D-Fender: its really basic config, really seems like there is some bug polycom refuses to acknowledge or something, I HOPE its something off on my end with * but what I have no idea, things work fine with Cisco 501gs, except they suck the phones |
| 22:16.17 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.103) |
| 22:17.20 | wltjr | their best answer is wrong username password.... username is extension and password I am copy and pasting and I have in provision files, I know password is correct, provision files fail 100% if I take that same file and push it to the phone 21 likely will not register |
| 22:18.37 | wltjr | oh cisco spa web stuff has no means to reboot phone what a pita.. having to call someone and say unplug this phone... |
| 22:18.46 | ChannelZ-Wk | yeah they do... uhm |
| 22:18.49 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 22:19.22 | wltjr | ChannelZ-Wk: where? maybe removed from 501g web config? its there on like spa112 |
| 22:19.42 | ChannelZ-Wk | I think you just hit the URL /admin/reboot |
| 22:19.46 | wltjr | but that is a totally different interface, I can upload firmware to spa112 via web interface, I have to run some windows tool to update spa501g |
| 22:19.56 | wltjr | ChannelZ-Wk: hmm good to know, but then firmware? |
| 22:20.04 | ChannelZ-Wk | whatever it has |
| 22:20.06 | wltjr | love that the vvx 300 can check for new firmware download and upgrade itself :) |
| 22:20.29 | ChannelZ-Wk | if you do provisioning, you can give it a new config to fetch which can have firmware in it |
| 22:20.35 | wltjr | ChannelZ-Wk: I have had to update firmware a few times, its a pita with the spa's, except ones that have ability to upload firmware via web interface, but still have to go download from cisco |
| 22:21.01 | ChannelZ-Wk | /admin/resync?tftp://whatever/spa9xx.cfg |
| 22:21.45 | wltjr | ChannelZ-Wk: no firmware not config |
| 22:21.45 | [TK]D-Fender | wltjr: reg.2.auth.useLoginCredentials="1" <- you have this for 2 & 3, and 1 is the only one that works. make it match |
| 22:22.02 | wltjr | [TK]D-Fender: we just noticed that ourselves their config is adding that |
| 22:22.11 | ChannelZ-Wk | then that spa9xx.cfg can look like http://pastebin.com/WfLV47Uu |
| 22:22.15 | [TK]D-Fender | "their config"? |
| 22:22.15 | navaismo | [TK]D-Fender, is my firewall!! If i stop iptables then rtp goes through public IP :S |
| 22:22.30 | wltjr | [TK]D-Fender: this is generated using polycom web interface, it clearly is creating a bad config |
| 22:22.31 | navaismo | if i start iptables then it goes to private ip |
| 22:22.47 | [TK]D-Fender | navaismo: Go stand on that big plastic sheent on the floor in the corner and I'll get to you in a minute... |
| 22:22.53 | wltjr | ChannelZ-Wk: I am not talking about pushing config to the spa phones I do that already, I am talking about sofware updates |
| 22:22.59 | [TK]D-Fender | calls for a clean-up crew |
| 22:23.06 | navaismo | weird thing is that I have used that iptables before |
| 22:23.11 | navaismo | brb |
| 22:23.30 | navaismo | scratching the corner |
| 22:24.23 | ChannelZ-Wk | wltjr: so am I |
| 22:24.54 | ChannelZ-Wk | You push them a config frag to tell them how to fetch the firmware, and they do. |
| 22:25.16 | ChannelZ-Wk | This is how I provision new phones without having to even touch the web interface. I just need to know their IP. |
| 22:25.44 | wltjr | ChannelZ-Wk: firmware is a zip file... not cfg... |
| 22:25.59 | wltjr | ChannelZ-Wk: I am NOT talking about provisioning.... http://software.cisco.com/download/release.html?mdfid=282724524&softwareid=282463651&release=7.5.4&flowid=5298 |
| 22:26.11 | ChannelZ-Wk | It's a .bin - unless you are talking about a different class of phone. |
| 22:26.29 | wltjr | ChannelZ-Wk: yes not all spas are the same I think the spa112 is bin |
| 22:26.31 | ChannelZ-Wk | And look at the pastebin... You give it a tftp location to fetch the firmware bin from. |
| 22:26.44 | navaismo | time to meld it! |
| 22:27.33 | ChannelZ-Wk | Look inside the zip. |
| 22:27.37 | wltjr | ChannelZ-Wk: I do not think you are failiar with the 501gs and that web interface, there is no means to do firmware you have to use the Cisco tool |
| 22:27.54 | ChannelZ-Wk | I have a 504g at home |
| 22:28.33 | Penguin | One of these days you'll learn to stop giving people the exact solution they are asking for. |
| 22:30.47 | ChannelZ-Wk | I know |
| 22:31.22 | Penguin | Gosh! |
| 22:33.45 | wltjr | [TK]D-Fender: right on the money buddy, that was the problem, the stupid web interface is adding that parameter on its line config |
| 22:34.23 | wltjr | [TK]D-Fender: got a paypal address? polycom pointed it out as well, but I had to go through all kinda crap with them, so they deserve nothing, but I do appreciate your rubber ducking, and you did find the issue as well, damn near within the same minute |
| 22:34.55 | [TK]D-Fender | wltjr: PM'd |
| 22:35.21 | [TK]D-Fender | wltjr: Also glad that was it. I'd hate to hear Polycom devices have REAL problems ;) |
| 22:36.01 | [TK]D-Fender | wltjr: Also, those who use Polycom's web interface for configuring phones should be shot. Survivors should be shot AGAIN :p |
| 22:36.47 | [TK]D-Fender | wltjr: They should just burn it out of their firmware to make room for more, actually useful, juicy features :) |
| 22:37.29 | wltjr | [TK]D-Fender: I was being lazy, I wanted web config to make me an initial config so I could copy to server, duplicate customize and move on with life :) |
| 22:38.04 | [TK]D-Fender | wltjr: SPIP history shows this to be unfortunately counter-productive |
| 22:38.05 | wltjr | [TK]D-Fender: FYI you HAVE to use the web interface or one in phone to configure tftp server for some reason, it defaults to ftp not tftp, and ignores next server from dhcp :( |
| 22:38.25 | wltjr | [TK]D-Fender: yes I shot myself in the foot, but I did find a problem for Polycom to fix.... 2 days later... |
| 22:38.29 | [TK]D-Fender | wlyon SPIP's you can choose it right in the the bootrom |
| 22:38.57 | wltjr | [TK]D-Fender: FYI their simple setup for first line and server works, the individual per line config even for line 1 adds that nasty setting |
| 22:39.41 | wltjr | [TK]D-Fender: also you have to name the file mac-web.cfg :P |
| 22:40.39 | [TK]D-Fender | wltjr: Come back around once you've had a chance to get a handle on them a bit more and let us know how you find them. |
| 22:40.55 | [TK]D-Fender | wltjr: Not a lot of feedback, on them yet |
| 22:41.29 | wltjr | [TK]D-Fender: thus far loving it short of the bug in web interface, the web interface is VERY nice easy to update firmware, get configs logs etc |
| 22:41.55 | wltjr | [TK]D-Fender: phones are great to, tell users when they are being upgraded, rebooting, etc, very eager to use them and see how stafff likes it, they hate me |
| 22:41.59 | [TK]D-Fender | wltjr: All of the things I love provisioning for! |
| 22:42.23 | [TK]D-Fender | wlyMy phones tell ME when they are upgrading and don't NEED rebooting ;) |
| 22:42.49 | wltjr | [TK]D-Fender: yes only way to go will pp ya later when I pay for ebay stuff, now I got phones to deploy finally, then 4 more projects for same client... |
| 22:43.15 | wltjr | [TK]D-Fender: well I am off site will they email me so I know :) |
| 22:43.16 | [TK]D-Fender | wltjr: Cool. Wish you continued success with them. See you around |
| 22:43.21 | [TK]D-Fender | packs up for studio time... |
| 22:43.38 | wltjr | [TK]D-Fender: I just love having a button to see if there is new firmware, selecting which one, and hitting save or what ever, best firmware update short of android stuff |
| 22:44.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gekbxfrrpnqcrmni) |
| 23:01.38 | navaismo | aaaaaaa I use this in the firewall for the RTP--> iptables -A INPUT -p udp -m udp --dport 19000:20000 -j ACCEPT |
| 23:01.49 | navaismo | im missing something? |
| 23:07.01 | ChannelZ-Wk | assuming you're not talking about port forwarding no.. though what is -m ? |
| 23:08.37 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:08.37 | *** mode/#asterisk [+o file] by ChanServ |
| 23:10.07 | navaismo | stand as match |
| 23:11.06 | ChannelZ-Wk | ?? |
| 23:11.26 | *** join/#asterisk redotis (~redotis@nc-184-3-102-139.dhcp.embarqhsd.net) |
| 23:11.50 | redotis | What is the minimum you need for configuring confbridge? |
| 23:12.05 | ChannelZ-Wk | an internet connection and a pulse |
| 23:12.42 | navaismo | jesus im surely overlooking this iptables issue.... |
| 23:13.11 | navaismo | why the hell is masquerading the rtp port ah ah ah |
| 23:13.34 | WIMPy | What do youneed an internet connection for? |
| 23:14.12 | redotis | lol |
| 23:14.23 | redotis | What must you have in the config file? |
| 23:15.20 | WIMPy | A general and a default_user section. |
| 23:15.37 | WIMPy | Not sure they need any contents |
| 23:16.11 | redotis | thanks wimpy |
| 23:16.37 | redotis | how can you keep one context from not getting in a conference? |
| 23:17.30 | WIMPy | By not havong an extension that calls it? |
| 23:18.09 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 23:18.18 | redotis | ok...so if I have an extension in two contexts that dial the same conference number |
| 23:18.23 | redotis | will they be in the same conference/ |
| 23:18.28 | redotis | can you give me an example |
| 23:18.32 | redotis | I'm confused |
| 23:18.35 | WIMPy | sure |
| 23:19.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:19.23 | *** join/#asterisk n3hxs (~n3hxs@pool-108-16-94-145.phlapa.fios.verizon.net) |
| 23:25.51 | redotis | so if they dial the same conference number, they'll be in the same conference? |
| 23:28.28 | WIMPy | No. |
| 23:28.47 | WIMPy | If you call ConfBridge with the same name. |
| 23:28.50 | ChannelZ-Wk | The extension and the 'conference number' are unrealted |
| 23:29.02 | ChannelZ-Wk | s/unrealted/unrelated/ |
| 23:29.02 | redotis | http://hastebin.com/tovomecuqo.coffee |
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| 23:29.20 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 23:29.26 | ChannelZ-Wk | Those would be the same conference, yes. |
| 23:29.30 | redotis | Would dialing these extensions in different contexts put them in the same conference? |
| 23:29.32 | redotis | ok |
| 23:29.35 | redotis | thank you |
| 23:35.43 | *** join/#asterisk TheGuy_ (~cgrosskop@vpn.oreilly.com) |
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| 23:37.56 | TheGuy_ | weird issue just starting happening with our asterisk system which has been established for years and no incident. We have people on the east coast that dial our meet me # and when they enter the pin the pin is invalid, the logs look like they are entering double digits but I've setup a sip phone and dialed myself with the same issue, where do I start with troubleshooting? |
| 23:46.33 | ChannelZ-Wk | Are the calls coming in to you SIP? |
| 23:47.30 | TheGuy_ | PRI's |
| 23:48.21 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 23:48.43 | ChannelZ-Wk | But the sip phone you tested from was direct into the system, not calling out/back in the same PRI? |
| 23:51.32 | *** join/#asterisk Draecos (~Draecos@106-68-16-143.dyn.iinet.net.au) |
| 23:52.26 | navaismo | this doesn't work either -->iptables -t nat -A PREROUTING -i eth0 -p udp -m udp--dport 19000:20000 -j ACCEPT |
| 23:52.39 | navaismo | anyone has hints for my stupidity? |
| 23:52.49 | ChannelZ-Wk | Are you really doing tunnelling? |
| 23:52.57 | ChannelZ-Wk | IE, is Asterisk the same box as the firewall, or behind it? |
| 23:53.01 | redotis | Does the confbridge have to be a number? |
| 23:53.08 | ChannelZ-Wk | redotis: no |
| 23:55.09 | snadge | would you guys recommend compiling the latest unmolested asterisk from source on fedora? |
| 23:55.21 | snadge | or is it recommended to just install the packaged version that comes with fedora |
| 23:55.46 | snadge | or use a digium / other rpm repository |
| 23:55.50 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:56.40 | snadge | my brain is exploding basically.. ive been asked to compile asterisk from source, on our new test/development network |
| 23:56.53 | snadge | what is this.. 1990? :P |
| 23:57.12 | navaismo | ChannelZ-Wk, iptables is in the same machine with asterisk |
| 23:57.59 | ChannelZ-Wk | so then you don't want prerouting (port forwarding) because you aren't forwarding anything anywhere. |
| 23:58.07 | navaismo | enabling the iptables, asterisk sent RTP packets to private IP, disable iptables asterisk sent to the correct public IP |
| 23:58.26 | eirirs | sounds like you are missing some iptables rules |
| 23:58.41 | ChannelZ-Wk | did you properly configure externaddr and localnet in sip.conf? |
| 23:58.46 | navaismo | yes |
| 23:59.13 | ChannelZ-Wk | And the peer outside the firewall you're presumably calling from has nat turned on for it? |
| 23:59.34 | navaismo | yep |
| 23:59.54 | ChannelZ-Wk | so let's see a sip debug of a test call with the firewall up |
