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| 03:11.04 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 03:11.14 | phix | hey hey |
| 03:13.13 | phix | I have an issue with analog phones connected to my TDM400, it seems I need to press each key for longer in order for asterisk it pick it up as being a press, any way to reduce this so it picks up a shorter press? |
| 03:13.56 | phix | It worked before I upgraded asterisk to version 1.8 (it was running 1.6 from squeeze before that) |
| 03:15.06 | phix | It does the same thing with the hook button, I need to hold it down for at least a second or two otherwise it is picked up as a flash |
| 03:15.59 | phix | Motherboards have been changed to so maybe a timing issue :\ any way, any thoughts would be useful, thnx |
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| 03:32.30 | *** join/#asterisk dorphalsig (c86ada96@gateway/web/freenode/ip.200.106.218.150) |
| 03:32.43 | dorphalsig | Hey |
| 03:32.55 | dorphalsig | Hello :) |
| 03:33.32 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 03:34.29 | dorphalsig | I have a question... I think I'm missing something when reading how queues work, and I was kinda worndering if anybody could help me understand. You see, I want to have a webpage monitor a specific agent's channel and depending on the queue and the caller id it would open a certain webpage |
| 03:34.46 | dorphalsig | I am not sure if I want to look at the manager events |
| 03:34.59 | dorphalsig | or if there is a cleaner way to look at that,,, |
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| 04:02.15 | *** join/#asterisk wtf911 (~wtf911@cpe-67-251-103-137.stny.res.rr.com) |
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| 04:20.50 | wtf911 | hmm everyone is sleeping? :P |
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| 04:27.39 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 04:35.26 | p3nguin | wtf911: Did you fix the port stuff? |
| 04:37.27 | wtf911 | working on it right now! got called to do something earlier |
| 04:37.34 | wtf911 | i just ran iptables -I INPUT -p udp --dport 10000:20000 -j ACCEPT |
| 04:37.43 | wtf911 | and iptables -I INPUT -p udp --dport 5060 -j ACCEPT |
| 04:37.51 | wtf911 | going to see if it fixes ipkall |
| 04:39.00 | jpsharp | I'm not sleeping, I'm writing code! |
| 04:39.24 | phix | so any ideas gang? |
| 04:39.49 | wtf911 | well i'll be it fixed it |
| 04:39.54 | phix | wtf911: must be |
| 04:39.59 | phix | or playing Diablo 3 |
| 04:40.09 | wtf911 | <--- SIP read from UDP:66.54.140.46:5060 ---> |
| 04:40.09 | wtf911 | ACK sip:808@wtf911.hopto.org SIP/2.0 |
| 04:40.09 | wtf911 | Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK1d991b15;rport |
| 04:40.09 | wtf911 | From: "BINGHAMTON NY" <sip:XXXXXXXXXX@66.54.140.46>;tag=as27cc2efb |
| 04:40.09 | wtf911 | To: <sip:808@wtf911.hopto.org>;tag=as2ebed27f |
| 04:40.10 | wtf911 | Contact: <sip:XXXXXXXXXX@66.54.140.46> |
| 04:40.10 | wtf911 | Call-ID: 105793731f72475005f13a4762de7fd8@66.54.140.46 |
| 04:40.11 | wtf911 | CSeq: 102 ACK |
| 04:40.11 | wtf911 | User-Agent: IPKall |
| 04:40.12 | wtf911 | Max-Forwards: 70 |
| 04:40.12 | wtf911 | Content-Length: 0 |
| 04:40.15 | phix | wtf911: no! bad! |
| 04:40.18 | phix | use pastbin for that |
| 04:40.50 | wtf911 | sorry |
| 04:41.22 | phix | any longer than 3 or 4 lines use pastebin.ca or whatever pasting site you use |
| 04:43.00 | wtf911 | do i need to do something to save iptables? (i was reading something like "iptables save" but that doesnt work) |
| 04:43.37 | jpsharp | try "Service iptables save". Depends on the distro, really. |
| 04:44.00 | wtf911 | ah there we go |
| 04:47.17 | Dovid | wtf911: In general it's a bad idea to psot ur IP and phone number publicly |
| 05:01.03 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 05:07.21 | wtf911 | how would i use authenticate to automatically continue after 4 digit pin? exten => 808,Authenticate(1111,,0)? |
| 05:10.32 | citywok | wtf999, http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate |
| 05:10.57 | wtf911 | that's the exact page i'm going off of |
| 05:11.07 | wtf911 | just checking |
| 05:12.33 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:13.15 | wtf911 | if i want to have text to speech is it easier to use Playback? or something else? |
| 05:15.27 | kaldemar | Playback is not used for TTS |
| 05:21.15 | wtf911 | is there a quick easy command to just have it play a beep? |
| 05:22.53 | wtf911 | p3nguin: btw i want to say thank you for sticking with me through everything, i've got a grasp on asterisk now, and everything works :D |
| 05:32.39 | Kobaz | axeterisk |
| 05:34.10 | kaldemar | wtf911: Playback(beep) |
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| 05:44.33 | *** part/#asterisk mpe (~mpe@94.127.49.1) |
| 05:46.50 | wtf911 | what does this mean? |
| 05:46.52 | wtf911 | [May 16 05:46:13] WARNING[7885]: pbx_config.c:1495 pbx_load_config: Invalid priority/label 'Authenticate' at line 856 |
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| 05:51.38 | kaldemar | wtf911: you probably have a syntax error in a Goto or GotoIf |
| 05:52.50 | wtf911 | i don't have any goto or gotoif |
| 05:53.02 | kaldemar | show what you have. |
| 05:53.32 | kaldemar | oh, it can also be an error in extension syntax. e.g. exten => 123,Authenticate() |
| 05:53.47 | kaldemar | that is missing a priority or label. |
| 05:54.01 | wtf911 | http://pastebin.com/VdWJkzC9 |
| 05:54.47 | kaldemar | exten => 808,Authenticate(1111,,0) is it. you must have a priority between 808 and Authenticate. |
| 05:55.04 | kaldemar | "n," needs to be added. |
| 05:55.18 | wtf911 | oops :o |
| 05:59.16 | wtf911 | [May 16 05:58:49] WARNING[8049]: pbx.c:3695 pbx_extension_helper: No application 'Authenticate' for extension (ipkall-inbound, 808, 2) |
| 05:59.34 | wtf911 | my fault hang on |
| 05:59.52 | *** join/#asterisk ghghz (~ton@kluonis.kvb.lt) |
| 06:00.07 | wtf911 | i forgot to load app_authenticate.so |
| 06:01.54 | wtf911 | ok now i'm having this problem http://pastebin.com/bnvpbQpp |
| 06:02.27 | wtf911 | i need to get sound files? |
| 06:02.29 | ghghz | Hey. If I do asterisk -rx 'channel originate SIP/blbla/number application..' asterisk calls to number, but when client drop the call, asterisk calls automaticaly again. |
| 06:02.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:02.33 | ghghz | any ideas? |
| 06:02.33 | schmidts | good morning |
| 06:04.50 | jpsharp | wtf911: Yes, you need to get sound files. |
| 06:08.39 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 06:10.09 | schmidts | can anyone of you give me a hint about a really good integratet CTI client for windows/mac, doesnt have to be free but it should work nice with asterisk |
| 06:18.50 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:23.26 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:28.23 | wtf911 | all is well now :D |
| 06:28.24 | wtf911 | http://pastebin.com/n51DPb9A |
| 06:36.31 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:36.34 | wtf911 | well one problem |
| 06:37.16 | wtf911 | i tested it and it works good when i call in from my ipkall number...now i linked my google voice number to my ipkall number and when i call in that way it just rings and rings then hits google voicemail |
| 06:37.30 | wtf911 | i turned on verbose 10 and debugging and asterisk is recieving the call |
| 06:38.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:38.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 06:38.33 | p3nguin | Calls via Google Voice are probably screened. That means you have to answer it and press 1 to accept the call. |
| 06:39.06 | p3nguin | You can try to turn off screening for forwarding, but I can't promise it will be disabled. |
| 06:39.32 | wtf911 | ok just turned it off i'll try now |
| 06:41.32 | wtf911 | no go..i'll have it turn the call over to my xlite and hear whats going on |
| 06:41.40 | p3nguin | I encountered that, regardless of the screening setting, screening was always enabled. I accepted this fact and made asterisk send the required 1 to take the call. |
| 06:43.26 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 06:44.20 | wtf911 | hmm when i call xlite it has my proper caller id and it just connects, no button pressing needed i just talk and hear myself |
| 06:44.50 | p3nguin | debug time! |
| 06:46.00 | wtf911 | if i did want to test adding the 1 in how would i do that? |
| 06:46.12 | wtf911 | just Dial(1)? |
| 06:46.33 | p3nguin | If call screening is enabled, you typically need to wait like 4-8 seconds, then SendDTMF(1). |
| 06:47.27 | *** join/#asterisk mpe (~mpe@office.ipvision.dk) |
| 06:48.18 | p3nguin | You'll want wait time after the Answer() and before the SendDTMF(). |
| 06:48.34 | p3nguin | Answer(4000) ; SendDTMF(1) |
| 06:48.39 | p3nguin | maybe? |
| 06:48.41 | p3nguin | Probably. |
| 06:48.57 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:49.20 | wtf911 | ok i'll try it |
| 06:50.17 | p3nguin | I'm actually waiting quite a bit longer than that before I send 1. |
| 06:50.43 | wtf911 | man you know your stuff :D |
| 06:51.13 | wtf911 | i just added the SendDTMF(1) (i didn't put the delay for answer) and voila instant "please enter your passkey" |
| 06:51.46 | wtf911 | weird it has to be done that way even after turning off call screening |
| 06:52.16 | p3nguin | I would set it to enabled and send the 1 every time. |
| 06:52.53 | p3nguin | Sending the call to an actual phone with screening enabled will help you understand how long you have to send the DTMF after the call is answered. |
| 06:53.30 | wtf911 | but it works now wih no delay, screening off, sending the 1? |
| 06:53.31 | p3nguin | I've seen people wait as short as 2 seconds and as long as 12 seconds before sending the 1 for accepting the call. |
| 06:54.08 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:54.14 | p3nguin | I don't know what else you have going on, but I would say the 1 isn't doing much for you at this point. |
| 06:54.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:54.41 | *** join/#asterisk hariom (~hariom@115.118.173.77) |
| 06:58.52 | wtf911 | my google voice settings http://i.imgur.com/bdWW2.png ...my extensions.conf http://pastebin.com/6BqjDDKp ...my debugging log http://pastebin.com/X0gWNtzB |
| 06:59.24 | wtf911 | with that configuration when i call it just rings until it hits google voicemail |
| 07:00.30 | wtf911 | just by adding a line after answer for exten => 808,n,SendDTMF(1) no more ringing and it goes straight to me hearing the authenticate message |
| 07:04.12 | hariom | My sip client is sends DTMF when I use professional SIP service provider but when I use my asterisk system, it is able to play and record audio but DTMF is missing. What could be the reason? |
| 07:05.57 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 07:08.15 | *** join/#asterisk PbxMan (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 07:10.39 | hariom | My sip client sends DTMF when I use professional SIP IVR service provider but when I use my asterisk system, it is able to play and record audio but DTMF is missing. What could be the reason? |
| 07:13.10 | *** join/#asterisk din3sh (~din3sh@41.212.202.98) |
| 07:13.15 | din3sh | mrning all |
| 07:14.11 | kaldemar | hariom: did you try rfc2833 and info as dtmfmode? |
| 07:15.45 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:15.58 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
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| 07:17.21 | hariom | kaldemar: Hi, yea I tried all three. inband, rfc2833 and info but still not luck |
| 07:18.45 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 07:20.47 | hariom | kaldemar: Does that depands on codec? My sip client is able to call and navigate the professionally hosted IVR. But same sip client does all the activity expect DTMF when it connects with my asterisk IVR application |
| 07:23.44 | kaldemar | inband really only works on ulaw/alaw. rfc2833 and info are signaling, so codec does not matter. |
| 07:23.55 | hariom | kaldemar: Another thing, If use the same sip client and use Wifi access point, I see DTMF coming to my asterisk box. But on GPRS network it works for professional ivr but not for my asterisk box |
| 07:24.02 | kaldemar | did you get the dtmf enabled in logger? still not seeing any incoming digits? |
| 07:24.36 | hariom | kaldemar: yea, I have enable logger. I don't see anything coming |
| 07:25.16 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:25.18 | hariom | kaldemar: btw, I am using AGI so agi debug set on mentions (timeout) even when I am pressing digits |
| 07:30.27 | hariom | kaldemar: what does relaxdtmf do? |
| 07:34.04 | wtf911 | once asterisk is running does it need to be able to access the harddrive? or is it in ram and good to go? |
| 07:39.20 | *** join/#asterisk AlHafoudh (~alhafoudh@85.248.11.120) |
| 07:39.22 | AlHafoudh | hi all |
| 07:39.41 | kaldemar | hariom: it is supposed to loosen the criteria for detecting dtmf tones in audio streams. |
| 07:39.57 | AlHafoudh | is it possible to put "@" character in sip register username? according to docs, it parses like fromuser@fromdomain :( |
| 07:40.00 | kaldemar | wtf911: depends on what your asterisk does and what you make it do. |
| 07:44.55 | kaldemar | AlHafoudh: try quoting the username portion. it works in some versions at least. |
| 07:46.48 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
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| 07:56.08 | AlHafoudh | kaldemar: thanks, i am going to try |
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| 09:11.39 | qakhan | i m getting this message when i try to get data from MSSQL 2008 using ODBC |
| 09:11.50 | qakhan | http://pastebin.com/XUX8ZzKQ |
| 09:11.58 | qakhan | http://pastebin.com/4XH0cES3 |
| 09:16.45 | AlHafoudh | after successfull registration to provider with "register =>" I have set up type=friend configuration for that provider, in SIP messages I can see that asterisk is sending OPTIONS request, but no response, in "sip show peers" the peer is unreachable, how can I troubleshoot this? |
| 09:17.57 | *** join/#asterisk alexscott (~alexscott@37.122.202.194) |
| 09:24.12 | qakhan | i m getting this message when i try to get data from MSSQL 2008 using ODBC |
| 09:24.15 | qakhan | http://pastebin.com/XUX8ZzKQ |
| 09:24.17 | qakhan | http://pastebin.com/4XH0cES3 |
| 09:26.19 | *** join/#asterisk jkroon (~jkroon@dsl-244-39-150.telkomadsl.co.za) |
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| 09:55.24 | Dovid | hia ll |
| 09:55.38 | teloniusz | hi |
| 09:55.44 | teloniusz | strange problem |
| 09:56.02 | teloniusz | two asterisks, one routes SIP to the other one |
| 09:56.47 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 09:57.02 | teloniusz | one asterisk shows SIP channel active and with ID indicating that it is a channel to the other one |
| 09:57.12 | teloniusz | the other one shows no active channels |
| 09:58.19 | teloniusz | 'hangup request <channel>' on the first one doesn't work (that is, it shows "Requested Hangup on channel 'SIP/ast-bt-2-00000884'", but channel remains active) |
| 09:58.26 | teloniusz | two questions |
| 09:58.36 | teloniusz | 1) how to destroy this channel? |
| 09:58.41 | teloniusz | 2) how to avoid this? |
| 10:00.56 | *** join/#asterisk jkroon (~jkroon@dsl-244-32-73.telkomadsl.co.za) |
| 10:03.33 | kaldemar | qakhan: "Invalid object name 'voicemessages'" |
| 10:05.23 | kaldemar | qakhan: "Function ODBC_confid not registered" <-- is your function really named "confid"? |
| 10:07.01 | *** join/#asterisk timahvo1 (~rogue@41.80.11.224) |
| 10:10.10 | *** part/#asterisk cstachris (~chrismylo@202.182.147.84) |
| 10:10.12 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 10:11.44 | qakhan | kaldemar yes |
| 10:12.05 | qakhan | i even changed it to test but same result |
| 10:12.33 | qakhan | i dont understand where voicemessages are coming from |
| 10:14.10 | wdoekes2 | app_voicemail.c:strcpy(odbc_table, "voicemessages"); |
| 10:16.02 | qakhan | wdoekes2 what i need to do with this? |
| 10:17.00 | wdoekes2 | make you understand where "voicemessages" is coming from |
| 10:17.50 | qakhan | wdoekes2 sorry i didnt get you |
| 10:17.51 | *** join/#asterisk wonderworld (~ww@dsdf-4db502f3.pool.mediaWays.net) |
| 10:18.04 | qakhan | are y talking about voicemail.conf? |
| 10:18.15 | qakhan | you* |
| 10:19.16 | kaldemar | qakhan: if you take a look at voicemail.conf.sample, you'll see where it comes from. |
| 10:19.35 | qakhan | kaldemar do i need to disable it? |
| 10:21.28 | qakhan | kaldemar what about Function ODBC_confid not registered" ? |
| 10:21.56 | kaldemar | qakhan: you should know if you want to use ODBC storage for voicemail messages. so you should know if you need to disable it. |
| 10:22.14 | kaldemar | qakhan: pastebin your func_odbc.conf |
| 10:22.25 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 10:22.47 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 10:23.12 | elliot98 | politely enteres |
| 10:23.34 | elliot98 | where does call presentation show up in an INVITE packet? |
| 10:24.06 | elliot98 | if call presentation is set to prohibited, how is it shown in a SIP packet? |
| 10:24.40 | wdoekes2 | elliot98: depends on your sendrpid settings |
| 10:25.06 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 10:25.34 | wdoekes2 | =no => not, =rpid => in the Remote-Party-IDm, =pai => *not* shown in from P-Asserted-Identity |
| 10:27.16 | mirelab | hi all, is there any way I can avoid getting this Warning when hangup on playing audio: |
| 10:27.18 | mirelab | <PROTECTED> |
| 10:27.20 | mirelab | [May 16 12:24:19] WARNING[23323]: file.c:756 ast_readaudio_callback: Failed to write frame <<--- |
| 10:30.07 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:30.15 | angryuser | Hello people |
| 10:30.27 | din3sh | why are you angry? |
| 10:30.55 | WIMPy | He's (trying to) use Asterisk. |
| 10:31.00 | WIMPy | scnr |
| 10:31.08 | angryuser | I have an E1 which receives calls and not getting the number entirely, i.e calling from the number 0123456, asterisk shows 123456 |
| 10:31.32 | angryuser | its only the presentation number, but i wonder if some kind of a dialplan can strip it ? |
| 10:31.47 | *** join/#asterisk nutxase (~nutxase@unix-02.lair.co.za) |
| 10:31.51 | nutxase | hi guys |
| 10:31.53 | angryuser | not the asterisk dialplan |
| 10:31.53 | WIMPy | The 0 isn't part of the number. See chan_dahdi.conf. |
| 10:32.08 | nutxase | anyone have any issues with *8(pickupexten) in 1.8? |
| 10:32.49 | angryuser | WIMPy, i suspected that, and i have enabled the pri debug, i see: Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) |
| 10:33.10 | angryuser | Presentation: Presentation allowed of network provided number (3) '123456' ] |
| 10:33.16 | angryuser | when it should be 0123456 |
| 10:33.26 | WIMPy | "national number" |
| 10:33.32 | angryuser | WIMPy, yep |
| 10:33.52 | angryuser | WIMPy, what in chan_dahdi is stripping that ? |
| 10:34.17 | WIMPy | As you can see thare's nothing stripped. |
| 10:34.24 | angryuser | WIMPy, http://pastebin.ca/2149285 here is mine |
| 10:34.27 | WIMPy | But you can ad prefixes. |
| 10:34.28 | elliot98 | wdoekes2: ok, so there's P-Asserted-Identity, Remote-Party-ID, and Sendrpid |
| 10:34.42 | wdoekes2 | no.. sendrpid is the sip.conf option |
| 10:34.59 | angryuser | WIMPy, i add nothing, where do you see prefixes ? |
| 10:35.07 | wdoekes2 | and there's From, which is used when sendrpid is no, but it's not prohibited |
| 10:35.12 | WIMPy | angryuser: *prefix |
| 10:35.24 | elliot98 | wdoekes2: aha...so where does it get prohibited? |
| 10:35.49 | wdoekes2 | rpid is short for Remote-Party-ID, and the option that is now 'rpid' used to be 'yes' |
| 10:35.51 | angryuser | WIMPy, i am not sure what conf line is affecting that :( |
| 10:36.33 | wdoekes2 | elliot98: I don't understand the question.. CALLERPRES()/CALLERID(num-pres)? |
| 10:36.35 | WIMPy | angryuser: <type>prefix. Take a look at the sample config. |
| 10:37.06 | elliot98 | wdoekes2: by setting the call presentation using CALLERPRES, where do I see reflected in the SIP packet? |
| 10:37.20 | wdoekes2 | depends on the sendrpid setting |
| 10:37.27 | wdoekes2 | like I said |
| 10:37.55 | WIMPy | angryuser: And take a look at the *dialplan lines as well, you might need them to call out. |
| 10:38.29 | wdoekes2 | if no, then From gets Anonymous@anonymous.invalid, if rpid, then the CLI info is in the RPID header, if pai then Anonymous@anonymous.invalid is in the PAI header (which is debatable, btw) |
| 10:38.54 | angryuser | WIMPy, i have no problems calling out, its just the presentation number when someone is calling is, that damn 0 is stripped by something, it is not asterisk dialplan |
| 10:39.11 | angryuser | s/calling is/calling out |
| 10:39.17 | wdoekes2 | ( elliot98: https://reviewboard.asterisk.org/r/1803/ ) |
| 10:39.23 | angryuser | s/calling is/calling in ;) |
| 10:39.24 | WIMPy | angryuser: There is nothing stripped. It's not part of the caller ID. |
| 10:39.47 | angryuser | WIMPy, so you think 123456 is sent by provider without 0 ? |
| 10:39.51 | elliot98 | is the privacy parameter set in the headers? |
| 10:40.07 | nutxase | any ideas? |
| 10:40.17 | WIMPy | angryuser: Yes. That's what you have seen in your pri debug as well. |
| 10:40.24 | wdoekes2 | elliot98: depends on the sendrpid ;) |
| 10:40.49 | WIMPy | angryuser: Just because you have to dial a 0 as a escape/access prefix doesn't make it part of the number. |
| 10:41.20 | angryuser | WIMPy, it is not for outgoing but incoming |
| 10:41.31 | WIMPy | Doesn't matter. |
| 10:41.55 | WIMPy | You don't have to dial the 0 for outgoing, either, if you set the type to national. |
| 10:41.59 | angryuser | WIMPy, when i dial from my mobile 05555555 number 132564654, pri rings, and presents number 5555555 to the called person |
| 10:42.12 | WIMPy | But that's probably not fitting in with Asterisk very well. |
| 10:43.05 | angryuser | WIMPy, its our nation dialing, everybody dials 0 heres to call out |
| 10:43.19 | elliot98 | wdoekes2: basically, from what I see, since the rpi is being sent, all that changes is the privacy parameter from full to oof |
| 10:43.20 | elliot98 | *off |
| 10:44.05 | angryuser | WIMPy, i will play with pridialplan |
| 10:44.07 | WIMPy | angryuser: Yes, but still it's not part of the number. And you could see the 0 as a workaround for not being able to set the type of number. |
| 10:44.09 | elliot98 | so the number is being sent, just terminating server needs to honor the privacy parameter |
| 10:44.38 | angryuser | WIMPy, i will try to set to unknown, we will see |
| 10:45.05 | WIMPy | unknown is the safest (and usually only sensible) choice. |
| 10:45.19 | WIMPy | Well, 'dynamic' is also ok, if you can make use of it. |
| 10:45.41 | WIMPy | But it's the prefixes you're after. |
| 10:45.42 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 10:45.55 | wdoekes2 | elliot98: that depends on your level of trust |
| 10:46.16 | wdoekes2 | if you don't trust the peer to honor the flag, you cannot use =rpid |
| 10:49.06 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 10:54.15 | angryuser | WIMPy, i suppose i need to set nationalprefix = 0 to add that |
| 10:54.51 | angryuser | PRI caller ID prefixes based on the given TON/NPI (dialplan) his is especially needed for EuroISDN E1-PRIs |
| 10:55.27 | WIMPy | No, that's general. |
| 10:55.36 | WIMPy | Exept for countries without area codes. |
| 10:56.15 | angryuser | WIMPy, well i am getting the presentation number without 0 and i need that 0 so people can call back, where should i add that? |
| 10:56.34 | WIMPy | wonders if there are any countries left, not using euroisdn, except for NA. |
| 10:56.53 | angryuser | WIMPy, half of the world ;( |
| 10:56.55 | WIMPy | Like you just said. |
| 10:57.06 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 10:57.22 | angryuser | My client is arguing that i do not present the chain correctly |
| 10:57.38 | angryuser | i present the chain sent by the provider |
| 10:58.05 | WIMPy | But the TON gets lost. |
| 10:58.37 | angryuser | WIMPy, yes, the question who is loosing that, me or provider |
| 10:58.44 | angryuser | i am not sure 100% |
| 10:59.05 | WIMPy | I assume you're forwarding to SIP phones? |
| 10:59.12 | angryuser | WIMPy, right |
| 10:59.29 | WIMPy | And sip doesn't know that concept. So that's were it gets lost. |
| 10:59.51 | angryuser | WIMPy, i dont think so |
| 10:59.58 | WIMPy | And that's why you have the *prefix options in chan_dahdi.conf to move the information elsewhere. |
| 11:00.51 | angryuser | WIMPy, you just said that dialplan prefix is not what i need, or i missed some other prefixes ? |
| 11:01.10 | WIMPy | What? |
| 11:01.36 | WIMPy | No. Exactely what you wrote. 'nationalprefix = 0'. |
| 11:01.53 | WIMPy | And 'internationalprefix = 00'. |
| 11:02.22 | WIMPy | Just as the sample config suggests. |
| 11:07.08 | din3sh | WIMPy if you are able to help angryuser, he might change his nick to lessangryuser |
| 11:07.39 | angryuser | well dialplan setting works |
| 11:08.25 | angryuser | WIMPy, i am also tired of national ;( need to check outgoing now |
| 11:09.44 | WIMPy | fails to parse that one. |
| 11:10.32 | qakhan | kaldemar here is my func_odbc.conf http://pastebin.com/JDuYyrPr |
| 11:10.56 | qakhan | <PROTECTED> |
| 11:11.17 | qakhan | i just want to query data from DB |
| 11:13.13 | kaldemar | qakhan: "Function ODBC_confid not registered" <--- you don't have a function named confid, so don't try to use one in your dialplan. |
| 11:13.42 | wtf911 | i'm just watching....you guys gives some awesome support! :D |
| 11:13.46 | qakhan | i told you it was confid then i changed it to test |
| 11:14.26 | kaldemar | qakhan: we're getting nowhere if you take hour answering questions and change configs in between. |
| 11:14.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:15.00 | qakhan | here same result |
| 11:15.01 | qakhan | http://pastebin.com/bJTc2598 |
| 11:15.14 | qakhan | read my previous chat |
| 11:16.00 | kaldemar | qakhan: did you reload? |
| 11:16.04 | qakhan | yes |
| 11:16.16 | qakhan | ofcourse |
| 11:17.34 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 11:19.40 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 11:19.54 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 11:20.38 | kaldemar | qakhan: show what you did. |
| 11:22.01 | qakhan | let me show you my .conf |
| 11:22.03 | qakhan | ok? |
| 11:22.23 | kaldemar | no, don't show just the config. show everything you did to add the function. |
| 11:22.36 | kaldemar | every command and its output. |
| 11:22.39 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 11:22.59 | kaldemar | qakhan: hold on with that, you have a typo. |
| 11:23.10 | kaldemar | qakhan: you have dns where you should have dsn. |
| 11:23.42 | din3sh | kaldemar helping qakhan |
| 11:23.44 | din3sh | deja vue |
| 11:23.46 | din3sh | lol |
| 11:24.00 | kaldemar | you would have seen an error reloading func_odbc.so if you looked. |
| 11:24.30 | kaldemar | din3sh: don't push it. |
| 11:28.03 | qakhan | kaldemar you are The MAN |
| 11:28.04 | qakhan | WOW |
| 11:28.06 | qakhan | :) |
| 11:31.51 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 11:37.01 | Ice_Strike | 50% of the time DTMFSend() don't work and how do I find out what went wrong. |
| 11:39.35 | Ice_Strike | SendDTMF() |
| 11:39.55 | *** join/#asterisk RogerH (5686e675@gateway/web/freenode/ip.86.134.230.117) |
| 11:46.07 | RogerH | any suggestions as to why voicemail left from a dahdi trunk is failing to record. Logs show "-- Recording was 2 seconds long but needs to be at least 3 - abandoning" but message was 10 seconds long |
| 11:46.21 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 11:48.51 | *** join/#asterisk LooserOuting (~paolo@ip-62-143-162-249.unitymediagroup.de) |
| 11:53.02 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 11:56.18 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-4-22.bois.qwest.net) |
| 11:56.28 | LooserOuting | i am runnign isto an issue trying to send a fax useing asterisk 10-svn as a gateway. ERROR[16561]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL. can sombody pls help me with this. |
| 11:56.32 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-sspbmskagujinfuv) |
| 11:57.55 | *** join/#asterisk teknoprep (~chrisrawl@unaffiliated/teknoprep) |
| 11:57.56 | teknoprep | hi all |
| 11:58.11 | teknoprep | i am not getting a reinvite for audio on my lan side? let me explain |
| 11:58.22 | teknoprep | i have an inbound call? this is how the call gets routedx |
| 11:58.46 | teknoprep | ITSP -> Asterisk B2BUA -> SIP PBX -> Polycom Phones |
| 11:58.56 | teknoprep | on the asterisk side i have a peer for the ITSP and the SIP PBX |
| 11:59.09 | teknoprep | audio is still streaming like this |
| 11:59.57 | teknoprep | ITSP -> Asterisk B2BUA -> SIP PBX -> Polycom Phones ---- while that is fine for SIP messaging i would expect once the call is established to the phone it should be ? ITSP -> Asterisk -> Polycom phone? the SIP PBX should not stay in the middle of the RTP Stream |
| 12:00.39 | teknoprep | what configuration on the peer for my SIP PBX should i have to allow for reinvites? canreinvite=yes is already there? i was wondering if there might be a [general] option that might be preventing this |
| 12:01.00 | LooserOuting | tell the SIP PBX to not stay in the media path |
| 12:01.05 | teknoprep | i already did |
| 12:01.18 | teknoprep | normally the SIP PBX with any other B2BUA will not stai in the path |
| 12:01.35 | teknoprep | its asterisk that is forcing it to stay there by either not accepting the re-invite or by some other means |
| 12:01.46 | teknoprep | SipXecs always trys to get out of the way of the media path |
| 12:04.43 | LooserOuting | Asterisk B2BUA is just doing what he was told unless it is configured to handle with some nat scenarios |
| 12:05.12 | teknoprep | it is configured to handle NAT |
| 12:05.23 | teknoprep | but not on the LAN side of the peer |
| 12:05.43 | teknoprep | the peer i have for the PBX is inside a local subnet |
| 12:05.46 | teknoprep | localnet |
| 12:06.08 | qakhan | kaldemar now fucn_odbc resgister with DB |
| 12:06.24 | qakhan | but it doesnt get data |
| 12:07.09 | qakhan | Executing [1001@outgoing:1] Wait("SIP/3001-00000002", "1") in new stack |
| 12:07.09 | qakhan | <PROTECTED> |
| 12:07.09 | qakhan | <PROTECTED> |
| 12:07.09 | qakhan | <PROTECTED> |
| 12:09.59 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:12.14 | *** join/#asterisk paryl (~par@168.215.170.202) |
| 12:18.39 | LooserOuting | @teknoprep: that make no sense. if the PBX and the clients are in the same local network the PBX should stay in the path. |
| 12:19.40 | LooserOuting | the asterisk can send reincite to establish the call between ITSP and SIP PBX |
| 12:19.46 | nutxase | anyone have any issues with *8(pickupexten) in 1.8? |
| 12:22.07 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:23.41 | teknoprep | LooserOuting, why would you want the PBX to handle the media path if it doesn't have to |
| 12:23.44 | teknoprep | it does make sense |
| 12:24.11 | teknoprep | LooserOuting, the SIP path would be handled ITSP -> Asterisk B2BUA -> SIP PBX -> Polycom phone |
| 12:24.34 | teknoprep | LooserOuting, the Media Stream would be as followoing ---- ITSP -> Asterisk B2BUA -> Polycom Phone |
| 12:25.01 | teknoprep | no if the a phone is registered to the PBX that is NAT'd |
| 12:25.15 | teknoprep | then of course the media stream would ahve to be handled by the PBX to compensate for NATing |
| 12:25.41 | teknoprep | so my question? why is asterisk not allowing the re-invite to the polycom phone |
| 12:26.26 | teknoprep | can a peer be a range of IP's ? |
| 12:26.38 | kaldemar | teknoprep: are there re-invites? have you allowed it to do so? |
| 12:26.54 | teknoprep | kaldemar, yes my PBX sends the re-invites.. i do have canreinvite=yes |
| 12:26.58 | teknoprep | for this peer |
| 12:27.20 | teknoprep | yes if i watch the RTP stream it always sends media from Asterisk B2BUA <-> SIP PBX |
| 12:27.29 | LooserOuting | @teknoprep, the PBX have to handle because you will run into nat issues |
| 12:28.26 | kaldemar | teknoprep: do you see the re-invites in asterisk? how does asterisk react to them? |
| 12:29.42 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 12:31.00 | *** part/#asterisk RogerH (5686e675@gateway/web/freenode/ip.86.134.230.117) |
| 12:31.10 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:34.37 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 12:35.50 | teknoprep | LooserOuting, not it doesn't ? nevermind i don't need to argue this with your LooserOuting |
| 12:36.18 | teknoprep | kaldemar, i'll run a SIP Trace? but i think it might actually be because i am setting up the asterisk box as a SIP Trunk and not an Unmanaged Gateway in my PBX |
| 12:36.47 | teknoprep | the issue with this is that if i set it up as a SIP Trunk it actually has to pass through the SBC on my PBX instead of just passing from the PBX to the GW which is asterisk |
| 12:36.57 | teknoprep | i think this is actually what is causing the no reinvite of media |
| 12:37.12 | teknoprep | i'll set it up later as an Unmanaged Gateway on my test rig and see if this fixes it |
| 12:42.29 | *** join/#asterisk Naikrovek (~sbrule@unaffiliated/naikrovek) |
| 12:44.49 | qakhan | kaldemar i want to get db data in veriables in dialplan |
| 12:44.58 | qakhan | plz help |
| 12:46.05 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:46.47 | kaldemar | qakhan: don't target me like that please. |
| 12:47.00 | [TK]D-Fender | ~book |
| 12:47.00 | 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 |
| 12:47.03 | [TK]D-Fender | qakhan, ^^^^^^^ |
| 12:49.14 | Ice_Strike | 50% of the time DTMFSend() don't work and how do I find out what went wrong ? |
| 12:49.15 | qakhan | kaldemar i m sorry for that, you was the only one who helped me. so i asked you |
| 12:49.29 | Ice_Strike | SendDTMF() |
| 12:52.24 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 13:12.10 | *** join/#asterisk dorphalsig (c86ada96@gateway/web/freenode/ip.200.106.218.150) |
| 13:12.36 | dorphalsig | hello |
| 13:14.50 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 13:14.52 | dorphalsig | I need to trigger an event just after an agent has picked up a call from a queue, and I was wondering if there is any way to have asterisk write something in a db |
| 13:14.56 | dorphalsig | when that happens |
| 13:15.30 | [TK]D-Fender | dorphalsig, "membermacro" <- check the queue config |
| 13:15.38 | [TK]D-Fender | dorphalsig, Also AMI throws off events |
| 13:16.04 | leifmadsen | dorphalsig: using a Local channel to call an agent, you can then use Dial() inside that with the M() or U() flags to trigger something to happen just before the channels are bridged. Then you can use DB() (for AstDB) or func_odbc to write to a relational database |
| 13:16.05 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fspvuksdcwdkspjt) |
| 13:16.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:21.05 | *** join/#asterisk danfromuk (~IceChat77@2.26.250.118) |
| 13:32.20 | Katty | hello my asterisk does not work at all how to fix plz |
| 13:33.28 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:34.02 | *** join/#asterisk serafie (~erin@nat/digium/x-lgutzllvhnkpdogb) |
| 13:35.28 | WIMPy | kands Katty a stick of TNT |
| 13:36.01 | dorphalsig | now blow it up. |
| 13:38.19 | [TK]D-Fender | 'Caude I'm T-N-T! Dynomite! |
| 13:38.23 | [TK]D-Fender | "Cause* |
| 13:38.24 | [TK]D-Fender | gah |
| 13:39.07 | Katty | watches TNT |
| 13:39.11 | WIMPy | Take shelter! The Fender is going to explode! |
| 13:39.56 | dorphalsig | Katty is holding the stick of [TK]D-Fender ? |
| 13:40.03 | Katty | no |
| 13:40.05 | dorphalsig | GET A ROOM! |
| 13:40.15 | [TK]D-Fender | This ... is a room ;) |
| 13:40.23 | Katty | i'm holding a stick of c6H2(NO2)3CH3 |
| 13:40.26 | dorphalsig | gah |
| 13:41.13 | Katty | and [TK]D-Fender is likely more explosive than TNT |
| 13:41.34 | dorphalsig | Hey tahts too much information |
| 13:41.50 | WIMPy | can see the news: Katty made [TK]D-Fender explode. |
| 13:42.02 | dorphalsig | hasnt this channel a strict anti pr0n policy? |
| 13:42.45 | WIMPy | We're in the internet and the internet is for porn. |
| 13:42.54 | dorphalsig | LOL |
| 13:43.00 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:43.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:44.03 | dorphalsig | [TK]D-Fender: ... where do I code the macro for membermacro? in extensions.conf like any other macro? |
| 13:44.30 | [TK]D-Fender | dorphalsig, correct |
| 13:44.30 | dorphalsig | WIMPy: now because of you I have to go see AvenueQ |
| 13:44.39 | dorphalsig | sweet! |
| 13:44.54 | WIMPy | You're welcome :-) |
| 13:46.00 | dorphalsig | and here I was thinking of doing a Java manager client that would monitor for events and then write them off to a queue where I would have php read them and show something to the agent... |
| 13:46.24 | dorphalsig | I always say, why make things easy if you can charge extra? |
| 13:47.23 | *** join/#asterisk darrenlooby (~darrenloo@host81-149-164-13.in-addr.btopenworld.com) |
| 13:48.28 | darrenlooby | We're getting high packet loss when making calls, but not with pings. We're on a high latency system, so wondering if Asterisk would drop packets if waiting too long and how to change... any help? |
| 13:49.03 | darrenlooby | Additionally, the loss is all on the upload, and not download. |
| 13:49.42 | WIMPy | Get TC/QoS/whatever set up. |
| 13:49.52 | dorphalsig | "The internet is for porn [for porn]..." hahahahaha |
| 13:53.31 | dorphalsig | I have this weird behaviour... I have a SIP trunk connected to a remote asterisk, when I make calls thru that remote asterisk, I can hear the other end, but they cant hear me... |
| 13:53.54 | darrenlooby | dorphalsig, a NAT issue? |
| 13:54.23 | WIMPy | NAT or general firewall. |
| 13:54.32 | dorphalsig | darrenlooby: I dont think so... both of them are in the same subnet |
| 13:55.04 | dorphalsig | and its not all of the calls |
| 13:55.21 | [TK]D-Fender | dorphalsig, "both"? Something tells me there is more than 2 *'s involved here... and I'm not hearing about those ends... |
| 13:56.03 | dorphalsig | I have my main * box, which has my extensions (SIP) |
| 13:56.39 | paryl | dorphalsig: you checked your mute button, i assume? :) |
| 13:56.47 | dorphalsig | and then this box connects to another box whose only mission in this world is to handle the SIP trunk |
| 13:57.10 | dorphalsig | paryl: Ohhh so that's what THAT button does... lol |
| 13:57.14 | paryl | haha |
| 13:58.07 | dorphalsig | paryl: when the whole thing started, that was my first thought. "Damn users, can't they know what the mute button is for???" |
| 13:59.10 | paryl | haha, it's amazing how often that happens. that, and "voicemail won't let me record a message!"... again, because of mute |
| 13:59.36 | paryl | i've seen that before, but it was a routing issue on my remote router |
| 14:03.17 | dorphalsig | but anyway, as I was saying, the SIP trace on my provider's end looks OK |
| 14:04.35 | dorphalsig | any ideas what could it be or what can I do to know what can it be? |
| 14:04.41 | paryl | you say it's not on all of the calls... is there anything that doesn't match between the good and bad attempts? |
| 14:04.50 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:05.15 | dorphalsig | I get this weird message from time to time on my cli |
| 14:05.25 | dorphalsig | stopped sounds on xxxxxxxx |
| 14:05.41 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 14:06.03 | dorphalsig | but its not even a notice |
| 14:06.44 | paryl | there are IAX calls, right? |
| 14:07.03 | dorphalsig | yes |
| 14:07.14 | paryl | do you have Trunk=yes in the conf? |
| 14:07.18 | dorphalsig | both * are IAXED |
| 14:07.30 | dorphalsig | lemme seee |
| 14:07.32 | paryl | because it shouldn't be set, and can cause the issue, iirc |
| 14:08.06 | Ice_Strike | Does asterisk have firewall or something? |
| 14:08.14 | Ice_Strike | Cant connect to asterisk anymore |
| 14:08.47 | Ice_Strike | softphone work all day |
| 14:08.54 | Ice_Strike | and all sudden stop working |
| 14:09.15 | [TK]D-Fender | Ice_Strike, No. |
| 14:10.09 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 14:10.09 | Ice_Strike | Really strange |
| 14:10.36 | paryl | dorphalsig: i found the post is what i was thinking of, btw: http://www.eflo.net/VICIDIALforum/viewtopic.php?t=7881 |
| 14:10.49 | paryl | don't know if it's exactly your sitch though |
| 14:13.33 | *** join/#asterisk bobman (~bobman@2001:470:7:3b::2) |
| 14:15.20 | dorphalsig | paryl: I have control over both IAX servers, my provider is SIP |
| 14:16.49 | *** join/#asterisk b1rkh0ff (~b1rkh0ff@178.77.20.215) |
| 14:16.52 | dorphalsig | paryl: Its more or less like this Phone<---SIP--->[myMainAsterisk]<----IAX---->[mySecondaryAsterisk]<--SIP-->[provider's gateway] |
| 14:17.43 | dorphalsig | myMainAsterisk and mySecondaryAsterisk are both on the same subnet, in the same location with nothing but a 3com switch between them |
| 14:18.55 | [TK]D-Fender | And there are 2 more ends you have not described proper at all an no details about testing done at each leg of the call. |
| 14:21.13 | Ice_Strike | Any idea why softphone wont connect to asterisk |
| 14:21.20 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 14:21.25 | Ice_Strike | I tried diffent software and same issue |
| 14:21.31 | Ice_Strike | nothing appear in CLI debug |
| 14:21.42 | Ice_Strike | for attempting to connect |
| 14:22.09 | dorphalsig | [TK]D-Fender: I've only checked my provider's end |
| 14:22.42 | [TK]D-Fender | dorphalsig, You run 3 of those and can debug each leg . You should already ahve done this. |
| 14:23.07 | [TK]D-Fender | dorphalsig, And you still haven't told us anything other than IAx2 between your two servers, which is one THIRD of the legs involved |
| 14:23.19 | Ice_Strike | Firewall on Windows is OFF |
| 14:23.40 | [TK]D-Fender | Ice_Strike, SIP debug enabled? Checked your firewalls on the server? routers in between? Forwarding/etc? |
| 14:23.52 | [TK]D-Fender | Ice_Strike, * SIDE TOO |
| 14:24.30 | dorphalsig | [TK]D-Fender: The first leg Phone-->sip->mainBox is OK, since I can make calls thru another trunk |
| 14:24.38 | dorphalsig | and I have no issue |
| 14:24.49 | dorphalsig | there |
| 14:24.55 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:26.43 | Ice_Strike | [TK]D-Fender Yes SIP debug is ON and nothing appear about connecting from sipphone to asterisk |
| 14:26.50 | Ice_Strike | firewalls on the server is empty |
| 14:27.04 | [TK]D-Fender | Ice_Strike, then packets jsut aren't arriving |
| 14:27.56 | Ice_Strike | Really strange |
| 14:28.07 | Ice_Strike | Do I need to add port on the router firewall? |
| 14:28.14 | Ice_Strike | asterisk installed on VPS though |
| 14:28.39 | [TK]D-Fender | This is basic networking. |
| 14:29.00 | [TK]D-Fender | you should know what forwarding/etc is required for your environment |
| 14:29.19 | Ice_Strike | It work well all day though |
| 14:30.36 | Ice_Strike | What forwarding port I should add? |
| 14:31.52 | [TK]D-Fender | Ice_Strike, * needs 5060 + rtp.conf's range |
| 14:32.22 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:32.32 | LooserOuting | i am runnign ito an issue trying to send a fax using asterisk 10-svn as a gateway. ERROR[16561]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL. both call legs are established. the Callee ends the Call(may be silence ?). can sombody pls help me with this. |
| 14:33.14 | LooserOuting | Callee is the t38 part. caller g711 |
| 14:37.24 | *** join/#asterisk Neptu (~Neptu@c-6c96e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 14:39.19 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 14:40.18 | Ice_Strike | Is 5060 UPD? |
| 14:40.58 | [TK]D-Fender | yes |
| 14:41.36 | Ice_Strike | rtpstart=10000 |
| 14:41.36 | Ice_Strike | rtpend=20000 |
| 14:45.09 | LooserOuting | should i write a bug report ? |
| 14:47.33 | leifmadsen | Ice_Strike: on 1.8 it is udp by default but can be tcp if enabled |
| 14:48.39 | *** join/#asterisk din3sh (din3sh@41.136.242.123) |
| 14:49.11 | alexscott | LooserOuting: do you use FAXOPT(gateway) ? |
| 14:50.31 | *** join/#asterisk kessius (~cassio@189.101.122.101) |
| 14:51.31 | *** join/#asterisk khronos (~tom@74.8.233.210) |
| 14:51.34 | khronos | Hi guys. |
| 14:52.08 | khronos | I'm running a call center on Freepbx and queuemetrics and I'm experiencing a call recording problem. |
| 14:52.32 | khronos | If a call comes in and is put on hold after an agentanswers the second part of the call is not recorded. |
| 14:52.51 | bulkorok | ~freepbx |
| 14:52.51 | 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 |
| 14:53.25 | [TK]D-Fender | That is not a FreePBX issue |
| 14:53.34 | khronos | Ok, I thought this issue might be related to the monitor application but I'll post there. |
| 14:53.37 | [TK]D-Fender | khronos, What ver of *? |
| 14:53.50 | [TK]D-Fender | khronos, No, the issue belongs here. |
| 14:55.37 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
| 14:56.15 | LooserOuting | alexscott, sure thing. the caller (g711) first responsed with 415 Unsupported media Type. Asterisk the reinvited the caller an ok'ed the callee |
| 14:56.34 | alexscott | you need to give the optional second parameter to FAXOPT(gateway), the timeout, to not have : ERROR[16561]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL. |
| 14:57.33 | LooserOuting | alexscott: something like FAXOPT(gateway,10) |
| 14:57.37 | alexscott | yup |
| 14:57.40 | alexscott | no |
| 14:57.44 | LooserOuting | :-) |
| 14:57.48 | alexscott | FAXOPT(gateway)=yes,10 |
| 14:57.48 | alexscott | ;) |
| 14:58.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:59.17 | LooserOuting | i didn't saw it in the doc. i'll try it |
| 15:00.23 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 15:01.10 | treborsux | I have openvox cards. On one of my systems that is oslec after running a few hours i get echo. Reboot and seems fine. How do i stop this? |
| 15:02.13 | WIMPy | treborsux: We need more information. |
| 15:02.30 | WIMPy | HWEC? SWEC? What drivers? |
| 15:02.47 | treborsux | how do i tell? |
| 15:03.20 | WIMPy | You should know what you bought and installed. |
| 15:04.44 | treborsux | Openvox ap1200 It installed drivers auto Elastix |
| 15:05.01 | treborsux | with hardware echo canceling |
| 15:05.35 | WIMPy | If it's hardware. you should contact openvox. |
| 15:07.20 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 15:08.32 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 15:10.42 | jeffspeff | My console is spamming the folling error. ERROR[1841]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("69.7.96.1360", "(null)", ...): No address associated with hostname The IP address is a valid one, except the extra 0 that's somehow getting attached at the end of the 136. |
| 15:15.36 | RubyRails | quick question - are recordings done by something like MixMonitor() recorded directly to /var/spool/asterisk/monitor/ directory, or are they recorded somewhere else and then moved there when the call is finished? |
| 15:16.17 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 15:20.47 | treborsux | I dont think it is all my cards do it |
| 15:21.16 | treborsux | what should be configured. Under span in hardware detect is OSLEC |
| 15:21.33 | WIMPy | You said with _hardware_ echo cancelling. |
| 15:21.34 | khronos | Running freepbx 5.5.2.4 |
| 15:21.58 | ZenMaster | Good morning guys. |
| 15:22.06 | ZenMaster | Another day of Diablo 3. The game is great. |
| 15:22.07 | WIMPy | So that's something the vendors support should sort out. |
| 15:24.56 | jkroon | RubyRails, depends on your configuration. |
| 15:25.54 | *** join/#asterisk Neptu (~Neptu@c-6c96e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 15:33.45 | *** join/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 15:33.50 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:34.57 | rampage73 | anyone had any experience getting a nec ip3na-tixh to work with asterisk |
| 15:36.31 | *** join/#asterisk capt-rogers (~IceChat77@mail1.decisioningsolutions.com) |
| 15:37.08 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:37.09 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:44.58 | Qwell | ~poll |
| 15:44.59 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 15:45.02 | Qwell | ~polls |
| 15:45.02 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 15:46.13 | leifmadsen | wow.... Fidonet polls? |
| 15:46.15 | leifmadsen | that's crazy old |
| 15:46.55 | jpsharp | A friend of mine still feverishly maintains his Fidonet node on his dialup BBS. |
| 15:47.01 | leifmadsen | crazy |
| 15:47.09 | leifmadsen | I ran an RA BBS for a few years |
| 15:47.37 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 15:47.40 | paryl | in those days, i was a pretty excellent leach |
| 15:48.13 | jpsharp | And he runs in on OS/2 as well. One of these days, he'll screw up and fall into the 21st century. |
| 15:51.11 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:51.21 | *** part/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
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| 16:17.12 | angryuser | leifmadsen, i still got the node adress ... |
| 16:17.26 | angryuser | and my point |
| 16:18.59 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 16:20.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:20.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 16:58.15 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 16:59.42 | p3nguin | rubyrails: The recordings are made in the monitor directory as the call is recorded. |
| 17:00.12 | RubyRails | thanks p3nguin |
| 17:01.12 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 17:02.29 | *** part/#asterisk teknoprep (~tek@cpe-static-bluecloudcons-rtr.cmts.eph.ptd.net) |
| 17:03.13 | *** join/#asterisk teknoprep (~tek@unaffiliated/teknoprep) |
| 17:03.15 | teknoprep | hi all |
| 17:03.26 | teknoprep | how do i setup a peer to only need to authenticate by IP address only |
| 17:04.16 | teknoprep | no registration needed... if the source IP is the correct IP it is considered allowed to connect |
| 17:04.27 | teknoprep | if a call is routed to that peer it sends to that peers IP address |
| 17:04.45 | teknoprep | is this possible ? |
| 17:05.49 | TechSmurf | Any ideas why a gxp2000 would go into a reboot loop when trying to provision? |
| 17:07.11 | teknoprep | shitty phone |
| 17:09.08 | TechSmurf | Thanks! That helped fix my problem! :) |
| 17:09.24 | [TK]D-Fender | teknoprep, Set the host, insecure=port,invite |
| 17:09.29 | teknoprep | ty |
| 17:09.32 | teknoprep | i'll try that |
| 17:10.34 | teknoprep | do i also need to set the host=IPaddr |
| 17:10.43 | teknoprep | IPaddr its actual ip address |
| 17:11.15 | [TK]D-Fender | <[TK]D-Fender> teknoprep, Set the host, insecure=port,invite <----------------- |
| 17:12.43 | teknoprep | ok done |
| 17:12.50 | teknoprep | still no reinvite for media |
| 17:12.55 | ZenMaster | Hers a question for you guys. |
| 17:12.56 | teknoprep | let me check a few other things |
| 17:13.03 | teknoprep | [TK]D-Fender, ty for the help again |
| 17:13.04 | ZenMaster | When I use Xlite to Xlite works fine. |
| 17:13.19 | ZenMaster | When I use 6755i to Xlite rings, but when I answer via xlite it hangs up. |
| 17:13.27 | ZenMaster | Log says that it was congested or busy... |
| 17:14.06 | teknoprep | check your nat settings |
| 17:14.14 | teknoprep | alot of the time that will cause issues |
| 17:14.27 | teknoprep | you may be sending information to the wrong location based on which NAT setting you are using |
| 17:15.03 | p3nguin | If you want to reinvite, you'll have to set directmedia=yes AND make sure there are no other factors that force asterisk to remain in the media path (such as certain dial options). |
| 17:15.29 | teknoprep | no i understand |
| 17:15.38 | teknoprep | ty for the directmedia let me check if i do not already have that |
| 17:16.50 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:18.29 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:18.30 | teknoprep | no still same issue |
| 17:18.39 | teknoprep | i should paste my configs |
| 17:20.01 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:20.29 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 17:20.44 | teknoprep | wooooooooooooooooohooooooooooooooooooooooooooooooooooo |
| 17:20.48 | teknoprep | WHOOOOOOOOOOOOOOOOOOOHOHOHHWOHOWHOWHOWHOW |
| 17:20.49 | teknoprep | lol |
| 17:20.55 | SuperNull | any way to look up 'bugs' for a certain version # ? |
| 17:21.03 | teknoprep | sorry.. it worked... i don't know why i had to reload the core but that made it work with those changes |
| 17:21.09 | teknoprep | TY ALL SO F';N MUCH |
| 17:23.58 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:25.10 | _Corey_ | SuperNull: Check the changelog for the version in question... |
| 17:25.32 | teknoprep | nvm |
| 17:25.37 | teknoprep | now i have the transfer issue |
| 17:26.19 | teknoprep | ahh i know why |
| 17:26.20 | teknoprep | lol |
| 17:26.21 | SuperNull | _Corey_ so the changelog of the version i want.. is going to know the bugs for that version ? wouldn't it be the changelog after the one im using ? |
| 17:26.42 | ZenMaster | :D |
| 17:27.04 | _Corey_ | SuperNull: Yeah, get the latest changelog for the branch you're using and you can see what was fixed on whatever you're currently running |
| 17:28.47 | SuperNull | okays. |
| 17:29.40 | SuperNull | seems like someone is registering when we have an explicit host entry, and with a different ip .. is there anything that would allow that? |
| 17:30.58 | teknoprep | [TK]D-Fender, can i define a peer under [default] in the sip.conf ? |
| 17:32.15 | [TK]D-Fender | that's where all peers go |
| 17:33.07 | teknoprep | here check this out |
| 17:33.12 | teknoprep | this is why i am confused |
| 17:33.30 | teknoprep | http://pastebin.com/RsYKMM06 |
| 17:33.38 | teknoprep | line 16 |
| 17:33.58 | teknoprep | i am actually trying to REFER-to 11@pbx1.bluecloudconsultants.com ---- not 11@default |
| 17:36.00 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 17:39.07 | *** join/#asterisk DukeOfURL (4a5de83a@gateway/web/freenode/ip.74.93.232.58) |
| 17:39.19 | DukeOfURL | hello |
| 17:39.46 | [TK]D-Fender | teknoprep, * is NOT a SIP PROXY |
| 17:40.06 | teknoprep | i want it to act as a B2BUA |
| 17:40.12 | teknoprep | which it is |
| 17:40.18 | [TK]D-Fender | teknoprep, calls hit dialplan. That's what * does |
| 17:40.25 | DukeOfURL | can someone answer a call waiting question? |
| 17:40.30 | [TK]D-Fender | * doesn't give a crap on the idea of passing it a URI |
| 17:40.41 | teknoprep | ok |
| 17:40.42 | [TK]D-Fender | you call into the DIALPLAN' |
| 17:41.09 | teknoprep | could i forward everything comming from 10.100.1.18 back to itself ? |
| 17:41.18 | teknoprep | so if any request comes in for lets say 11@default |
| 17:41.26 | [TK]D-Fender | teknoprep, if you're extremely lucky maybe you could try to parse the TO: header |
| 17:41.27 | teknoprep | asterisk knows to send it to a specific IP address ? |
| 17:41.31 | [TK]D-Fender | teknoprep, To: <sip:6107413324@pbx1.bluecloudconsultants.com;user=phone>;tag=as2f338f55 |
| 17:41.56 | [TK]D-Fender | teknoprep, Then again, last I heard the header only was taken from the initial invite.. but you might have a prayer that a refer resets something. |
| 17:42.02 | [TK]D-Fender | teknoprep, .... at best |
| 17:42.19 | teknoprep | ok |
| 17:42.35 | [TK]D-Fender | <teknoprep> asterisk knows to send it to a specific IP address ? <-.... what part of "dialplan" is unclear? There is no address. Thre is only extensions.conf |
| 17:42.35 | DukeOfURL | Asterisk 1.8/FreePBX. Polycom phone with multiple line appearances. User is on a call on the first line. A second call arrives, and the call waiting tone is played. The user cannot hear the audio on the first call until the second call terminates. |
| 17:42.52 | DukeOfURL | Is this normal? |
| 17:43.14 | [TK]D-Fender | DukeOfURL, did you answer the second call? |
| 17:43.17 | teknoprep | can i use a dial plan to route all requests to a peer ? |
| 17:43.22 | DukeOfURL | no it is ringing |
| 17:43.40 | [TK]D-Fender | teknoprep, you should already know what yuo can do in extensions.conf ... |
| 17:43.53 | DukeOfURL | ? |
| 17:43.54 | teknoprep | well i don't know everything i can do |
| 17:44.38 | [TK]D-Fender | teknoprep, there is no "route". There is DIAL(), etc.... "core show applications" |
| 17:44.45 | [TK]D-Fender | DukeOfURL, No, this is not normal... |
| 17:44.49 | teknoprep | ok |
| 17:45.09 | teknoprep | ahh i know what the issue is |
| 17:45.15 | [TK]D-Fender | DukeOfURL, and would appear to be a symptom of your phone's configuration. |
| 17:45.32 | DukeOfURL | phone or Asterisk? |
| 17:45.48 | [TK]D-Fender | Phone |
| 17:46.06 | DukeOfURL | Polycom Soundpoint 430 |
| 17:46.23 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:46.38 | Ice_Strike | What file format do you recommend for recording? monitor() |
| 17:46.53 | [TK]D-Fender | I wasn't asking the model. I was clarifying that it was the configs on the phone itself, not the *-side configs to match the phone |
| 17:46.56 | teknoprep | the request comes from 10.100.1.18 which is my [pbx] peer... the pbx peer doesn't have a context assigned so when calls come into the pbx from that they are sent to the general context.. in the general context i have a matching DP but it sends it out the gateway |
| 17:47.05 | teknoprep | i think i could accomplish this with some fancy DP rules |
| 17:47.08 | DukeOfURL | ok, thanks |
| 17:47.28 | [TK]D-Fender | Ice_Strike, Whatever you want to use |
| 17:47.37 | [TK]D-Fender | Ice_Strike, Ideally the same format as the caqll itself |
| 17:48.11 | Ice_Strike | [TK]D-Fender .wav is kinda huge |
| 17:48.30 | [TK]D-Fender | <[TK]D-Fender> Ice_Strike, Whatever you want to use <------------- |
| 17:48.46 | Ice_Strike | [TK]D-Fender Did I ask you repeat again? |
| 17:48.59 | Ice_Strike | [TK]D-Fender .wav is kinda huge <------------- |
| 17:49.27 | [TK]D-Fender | Ice_Strike, I tell you you can use whatever you want and you go on complaining about what you DON'T like. So don't use it. |
| 17:49.40 | Ice_Strike | [TK]D-Fender When did I complain? |
| 17:49.46 | Ice_Strike | I just said it is huge |
| 17:49.53 | [TK]D-Fender | Yes, which implies "bad" |
| 17:50.03 | [TK]D-Fender | So if it's too big for you, don't do it. |
| 17:50.20 | p3nguin | That's what she said. |
| 17:50.40 | [TK]D-Fender | "and then he woke up" |
| 17:52.52 | TechSmurf | hrm |
| 17:53.07 | TechSmurf | gxp 2000 reboot loop caused by cfg file over 4KB |
| 17:55.11 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:55.41 | p3nguin | ice_strike: I prefer .WAV files. They are small and portable. |
| 17:55.44 | *** join/#asterisk dhill (~dhill@fog.mindcry.org) |
| 17:56.05 | Ice_Strike | p3nguin cool, thanks for being polite. |
| 17:56.09 | dhill | can i have asterisk bind to multiple IPs? |
| 17:56.15 | dhill | but not all. |
| 17:56.24 | p3nguin | No. |
| 17:56.26 | p3nguin | One or all. |
| 17:57.30 | dhill | ok, if it binds to all.. asterisk doesn't use the correct interface/IP to send packets based on source.. is that known, or do i have something configured wrong? |
| 17:58.05 | p3nguin | Could be a routing issue. |
| 17:58.14 | ZenMaster | So my boss just handed me some paperwork with SoftSwitch Ip Domain DID and TFN stuff. Where do I put that information in asterisk? |
| 17:58.26 | ZenMaster | sip.conf? |
| 17:58.35 | p3nguin | maybe |
| 17:58.45 | ZenMaster | This is for configuring their trunk. I am sick of trying to deal with their GUI. |
| 17:58.51 | [TK]D-Fender | dhill, * uses standard kernel routing rules. |
| 17:59.01 | p3nguin | If I knew what you meant by "SoftSwitch Ip Domain DID and TFN stuff," it would help. |
| 17:59.02 | dhill | ok, let me try something |
| 17:59.23 | ZenMaster | Here forcefully use our GUI but then we don't tell what stuff means since we change the names of EVERYTHING@ |
| 17:59.35 | ZenMaster | p3nguin: The goal for today is to try and get a dialtone. |
| 17:59.49 | ZenMaster | The ability to make a outbound phone call from a softphone on the computer here in the office. |
| 18:00.09 | p3nguin | Configure a peer. Register the device to asterisk. |
| 18:00.15 | p3nguin | That's how you get a dial tone. |
| 18:00.26 | ZenMaster | I have devices and agents setup. |
| 18:00.30 | ZenMaster | Currently I can call localy.. |
| 18:00.37 | ZenMaster | One machine to another. |
| 18:00.43 | p3nguin | Agents are not relevant at this point. |
| 18:00.50 | dhill | asterisk listens on say, public ip 2.2.2.2, and private IP 10.0.0.1. i have vpn users that connect to the machine. if vpn user 10.2.2.2 connects to * private ip 10.0.0.1, asterisk sends packets back from public ip 2.2.2.2 :| |
| 18:01.09 | dhill | and not 10.0.0.1 |
| 18:01.19 | ZenMaster | p3nguin: Googling configuring a peer in asterisk. |
| 18:01.28 | p3nguin | ~book |
| 18:01.29 | 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 |
| 18:01.37 | p3nguin | It's ALLLLLLLLL in the book. |
| 18:01.50 | p3nguin | dhill: Sounds like a bad NAT configuration in Asterisk. |
| 18:01.52 | ZenMaster | Really? I thought it was all in the hips. :) |
| 18:02.24 | dhill | i think asterisk does it wrong |
| 18:03.12 | [TK]D-Fender | dhill, Kernel rules pick the interface, not * |
| 18:03.36 | [TK]D-Fender | Asterisk does not do this at all. |
| 18:03.42 | sp00kz | ^ its your underlying system being misconfigured. |
| 18:03.46 | p3nguin | If asterisk says to send a packet to a specific address, the routing table determines how the packet gets out. |
| 18:03.47 | Kobaz | all borken |
| 18:03.57 | dhill | hmm, ok, let me try one more idea |
| 18:03.59 | Kobaz | p3nguin: i thought it was the gnomes |
| 18:04.10 | p3nguin | Packet gnomes ruin everything. |
| 18:04.13 | Kobaz | i have the gnomes do all my routing |
| 18:04.19 | Kobaz | they usually do a good job |
| 18:04.29 | Kobaz | if they are bad, they go out on the lawn |
| 18:04.35 | [TK]D-Fender | They man the light in my fridge quite dilligently |
| 18:04.49 | p3nguin | Underpants gnomes are the worst. |
| 18:04.51 | Kobaz | my fridge gnomes are slacking. the light doesn't work anymore |
| 18:06.35 | p3nguin | I think my frige has gremlins as opposed to gnomes. |
| 18:06.36 | ZenMaster | LOL. |
| 18:07.25 | ZenMaster | This asterisk book is going to look great next to my apache definitive books. :D |
| 18:07.35 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 18:08.29 | *** join/#asterisk RypPn2 (~RypPn2@unaffiliated/ryppn) |
| 18:16.23 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 18:17.29 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 18:19.31 | bent_screwdriver | on a couple of occasions I've had our polycom IP650's all reboot at the same time (with the exception of 1 or two). I don't have any tasks or extensions set to reboot the phones so not sure what it could be. non-poe log output: http://pastebin.com/0xJNgPx0 |
| 18:22.14 | jeffspeff | I'm having an issue getting a sip phone to accept a call. Somehow during the connection process, asterisk adds a "13" to the end of the remote phones IP address and then it all just starts failing from there. Here's the sip debug. http://pastebin.com/5Da5qbu2 The call was from extension 1605 to 1101, and the error occurs when 1605 answers the call. |
| 18:23.49 | jeffspeff | i have 4 other remote offices using the same phones with the same system, and they're all working fine. |
| 18:24.08 | mjordan | jeffspeff: your SDP specified that as the IP address to send RTP to |
| 18:24.09 | mjordan | c=IN IP4 69.7.96.13613 |
| 18:24.13 | *** join/#asterisk kessius (~cassio@189.101.122.101) |
| 18:24.47 | [TK]D-Fender | jeffspeff, SIP/1101 = bad Reliably Transmitting (no NAT) to 192.168.251.113:5060: Contact: <sip:1605@12.49.160.154:5060> |
| 18:25.03 | [TK]D-Fender | jeffspeff, bad NAT settings for that subnet. |
| 18:25.42 | *** join/#asterisk kessius (~cassio@189.101.122.101) |
| 18:26.06 | [TK]D-Fender | jeffspeff, "sip show peer 1101" |
| 18:26.34 | [TK]D-Fender | -- Registered SIP '1101' at 192.168.251.113:5060 |
| 18:26.47 | jeffspeff | [TK]D-Fender, everything else is working fine though. The phones are able to register with * and pull their tftp files; and all the computers there are able to function properly. |
| 18:26.55 | [TK]D-Fender | jeffspeff, You also conveniently left off the actual SIP debug of the registration... |
| 18:27.21 | jeffspeff | [TK]D-Fender, 192.168.251.113 is that phones local ip |
| 18:27.36 | jeffspeff | i'll reboot the phone and try to grab its registration |
| 18:27.42 | [TK]D-Fender | Where is the extra 13? |
| 18:28.01 | mjordan | [TD]D-Fender: jeffspeff: regardless of the NAT settings for the signaling, the SDP sent from the polycom is wrong. It contains an invalid address. If the polycom sends Asterisk something that isn't an IP address, Asterisk isn't going to be able to turn it into one :-) |
| 18:29.04 | mjordan | [TK]D-Fender: look at line 349 in the pastebin |
| 18:29.34 | jeffspeff | mjordan, thanks, i was just digging for that |
| 18:29.49 | newtonr | jeffspeff: are you manually configuring that particular phone, or is it being provisioned remotely? |
| 18:29.58 | jeffspeff | provisioned |
| 18:30.10 | p3nguin | Why is the phone controllings its own NAT settings? |
| 18:30.22 | p3nguin | s/controllings/controlling/ |
| 18:30.59 | jeffspeff | p3nguin, i didn't set it to, |
| 18:31.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:31.56 | jeffspeff | this phone is currently in a FreePBX environment, I'm trying to slowly migrate all the remote offices off of their local freepbx to a centralized * server that i setup. |
| 18:32.10 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 18:32.22 | jeffspeff | i didn't do the original freepbx deployment, i'm just having to (trying to) clean up behind it. |
| 18:32.29 | [TK]D-Fender | mjordan, Yup, I see it... somehow I'd scrolled to an odd spot in that debug... |
| 18:33.05 | [TK]D-Fender | jeffspeff, set NAT=YES for that phone |
| 18:33.25 | [TK]D-Fender | it's SDP is learly b0rqed |
| 18:33.30 | [TK]D-Fender | clearly even |
| 18:33.35 | [TK]D-Fender | its* |
| 18:35.52 | newtonr | A factory default of the phone and re-provisioning would be interesting to see if it continues. |
| 18:36.14 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 18:36.21 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:36.23 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 18:38.56 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 18:40.32 | jeffspeff | [TK]D-Fender, p3nguin, mjordan, here's the sip debug of everything, including the phones registration... sorry there are few other calls showing in there, so it's kinda long http://pastebin.com/ZQ7rLnyJ |
| 18:42.22 | jeffspeff | the register starts on line 3415 |
| 18:43.32 | *** join/#asterisk vastina (~johnny@70.99.103.122) |
| 18:43.33 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 18:45.46 | ZenMaster | So my sip provider auths me by ip address. |
| 18:45.59 | ZenMaster | According to the book it is telling me aboutg user names and passwords. Do I just omit that? |
| 18:46.02 | p3nguin | What I'm seeing is an inconsistency in IP address for a device. |
| 18:46.15 | jeffspeff | p3nguin, where at? |
| 18:47.09 | jeffspeff | ZenMaster, yes, and omit the register part in [general] in sip.conf |
| 18:47.13 | ZenMaster | So for instance adding register => 1.1.1.1@proivder.com does not work for me. :*( |
| 18:47.22 | ZenMaster | Thanks jeffspeff. |
| 18:47.47 | ZenMaster | jeffspeff: However that seems to be putting me back at square one with getting this system to dial an outside phone number I.E. like a cell phone. |
| 18:48.04 | p3nguin | A device by the name of 1101 uses both 192.168.251.113 and 69.7.96.136. |
| 18:48.17 | p3nguin | This must be from a bad NAT setting. |
| 18:48.37 | jeffspeff | p3nguin, i can't figure out where it's getting the 69.7.96.136 from because that's it public ip. |
| 18:48.55 | jeffspeff | there's a vpn tunnel between 192.168.251.0 and 10.200.100.1 |
| 18:48.58 | jeffspeff | there's a vpn tunnel between 192.168.251.0 and 10.200.100.0 |
| 18:48.59 | p3nguin | Ensure that the nat traversal setting ON THE PHONE is not enabled. |
| 18:49.03 | [TK]D-Fender | ZenMaster, so don't register |
| 18:49.42 | p3nguin | zenmaster: Don't register. Provide the correct defaultuser (the username) and secret (the password) in the peer entry. |
| 18:49.44 | jeffspeff | p3nguin, i'm going to have the EU reset the phones configs and then see if that helps. |
| 18:50.01 | [TK]D-Fender | jeffspeff, It doesn't look like your phone is going through the VPN.. or your implementation is whacked |
| 18:50.29 | jeffspeff | [TK]D-Fender, the register is going through the vpn but then some parts aren't |
| 18:50.31 | ZenMaster | Guys: Here is something I got from my provider. :) |
| 18:51.09 | ZenMaster | " NOTE: XCast authenticates based on your IP address(es) and not on a username/password". |
| 18:51.20 | [TK]D-Fender | ZenMaster, So don't register |
| 18:51.43 | ZenMaster | Ok, but still input peer entry stuff in my sip.conf? |
| 18:53.22 | [TK]D-Fender | yes... * still need to take calls in and send out |
| 18:56.42 | p3nguin | If they authenticate only by IP address and don't care about the user/pass AT ALL, you can leave out the defaultuser and secret. |
| 18:56.59 | ZenMaster | p3nguin: Thanks that is what I was thinking. |
| 18:58.24 | p3nguin | They will send calls to you, which your peer entry will match by the host setting. Calls that you send to them via the peer entry will be sent to the host defined, and if they don't accept it, they didn't give you accurate information. |
| 18:59.01 | ZenMaster | p3nguin: You are awesome. Actually you are all awesome! :D |
| 18:59.36 | p3nguin | I don't think IP address only is a good way to authenticate, but that's just my opinion. |
| 19:00.33 | ZenMaster | This is not my choice. I was thrown into this world about two days ago. |
| 19:01.37 | ZenMaster | Ever heard of PBX,yes, ever ran it, no. Well here is a system, that we have been developing on for two years and that is completely alpha, and we want you to finish it and make it work. |
| 19:01.37 | ZenMaster | That has been my task for the last two days. :D |
| 19:01.37 | [TK]D-Fender | ZenMaster, Don't worry, the sharp rocks at the bottom will break your fall! |
| 19:02.09 | ZenMaster | Hehe yeah. |
| 19:03.28 | ZenMaster | Xcastlabs is now reachable. Thats a good sign right? :D |
| 19:04.12 | talntid | http://pastebin.com/zMdFTsHu |
| 19:04.29 | talntid | Im doing something wrong.... I need that to wait and listen to someone pressing "1" for whatever, 2 for whatever else. |
| 19:04.52 | [TK]D-Fender | ZenMaster, it's "something". Now place calls. That is the only real proof |
| 19:04.56 | talntid | just by adding exten => s,1,1,dothis() isn't working through it. it just hangs up |
| 19:05.16 | ZenMaster | Not working. |
| 19:05.17 | [TK]D-Fender | talntid, Nothing in there imakes that an IVR, nor takes in input direct... |
| 19:05.27 | [TK]D-Fender | ZenMaster, Show us the CLI w/ SIP DEBUG enabled |
| 19:05.33 | [TK]D-Fender | ~pb |
| 19:05.33 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:05.34 | [TK]D-Fender | ^^^ |
| 19:06.01 | [TK]D-Fender | talntid, IVR's are described in the book |
| 19:06.03 | [TK]D-Fender | ~book |
| 19:06.03 | 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 |
| 19:06.04 | [TK]D-Fender | ^^^^^6 |
| 19:06.20 | ZenMaster | [TK]D-Fender: What the error is saying now, call from 'xlite02' to extension '3012344562' rejected becasue extension not found in context 'internal'. |
| 19:07.03 | talntid | even after adding.. exten => 1,1,agi(googletts.agi,"test",en) to the end, it doesn't make it into an IVR. I'm missing something small, because I have done this before.. long time ago.. :\ |
| 19:07.12 | [TK]D-Fender | ZenMaster, Your softphone is landing in that context with the # you dialed and you do not have an exten in there to match that number and tell * what to do with it |
| 19:07.31 | talntid | oh waitexten |
| 19:07.33 | talntid | that's it |
| 19:07.35 | [TK]D-Fender | talntid, "core show application waitexten |
| 19:07.52 | talntid | =] thanks. I got it right before you said it |
| 19:07.54 | ZenMaster | [TK]D-Fender: So what does this mean? |
| 19:08.16 | [TK]D-Fender | ZenMaster, it means you have to tell * what to process from your phone... |
| 19:08.23 | [TK]D-Fender | ZenMaster, extensions.con <--- |
| 19:08.25 | [TK]D-Fender | ~book |
| 19:08.25 | 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 |
| 19:09.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:10.17 | ZenMaster | Does this mean taht I am going to setup an extension for every number that is ever to be dialed? |
| 19:10.28 | ZenMaster | I havce not felt this n00b in a very long time. LOL. |
| 19:10.31 | [TK]D-Fender | ZenMaster, patterns .... |
| 19:10.37 | [TK]D-Fender | ZenMaster, Time to learn the 101's |
| 19:12.46 | TechSmurf | hrm |
| 19:12.55 | TechSmurf | asterisknow + provisioner = endpoint nightmare :\ |
| 19:13.31 | [TK]D-Fender | TechSmurf, That'd really only be "EPM nightmare" |
| 19:13.45 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:13.55 | p3nguin | talntid: A typical IVR uses Read() and GotoIf(). If you are using WaitExten(), you're probably making an auto-attendant. |
| 19:14.12 | TechSmurf | the gxp 2000 config file was bogus because it was about 4.5k and the phone only accepts configs up to 4k... |
| 19:14.29 | TechSmurf | the spa962 initial config is corrupt, according to the phone... |
| 19:14.41 | p3nguin | It's the difference between a response to an inquiry and a new extension being executed. |
| 19:17.04 | talntid | http://pastebin.com/rWegshCp -- with this, this works as expected, but during the playback of northwest3, they get instructions on #'s they can push. But they can't push them while the recording is playing, or it doesn't respond to them. If I put the waitexten in front of the background() it waits 60 seconds then plays... do'h? |
| 19:17.36 | p3nguin | For example: An IVR might consist of "For yes, press 1. For no, press 2," and the routine changes based on the response. An auto-attendant might consist of "For sales, press 1. For billing, press 2," and extension 1 or extension 2 Dials a phone or Queues the call for the dept. |
| 19:18.27 | p3nguin | talntid: If you need the caller to be able to enter the extension during the playback, use BackGround() instead of Playback(), followed by the WaitExten() to add additional time after the playback finishes. |
| 19:18.51 | talntid | p3nguin, basically, I'm calling out, and giving my customers information, and giving them the option to opt out of calling, email the document to them, or repeat the message.. (well, thats the goal) |
| 19:19.09 | p3nguin | That sounds like IVR, and should use Read and GotoIf. |
| 19:19.15 | TechSmurf | My, look at that. I chopped the SPA962 config down to under 64k and it works fine too |
| 19:19.15 | talntid | roger that |
| 19:19.50 | p3nguin | If the callee pressed a certain option, Goto a specific spot in the routine. |
| 19:21.44 | p3nguin | While it is possible to build IVR with BackGround() and WaitExten(), along with additional extensions, it isn't my preferred method. |
| 19:22.03 | talntid | yeah, I gotcha |
| 19:22.14 | talntid | I'm currently reading the docs to figure out how I'm gonna do this |
| 19:24.29 | p3nguin | like... the sections in the book on IVR and auto-attendant? |
| 19:24.48 | jeffspeff | lol |
| 19:25.23 | p3nguin | I was just wondering if that was the documentation to which was referred. |
| 19:25.36 | talntid | nah, reading http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 19:25.42 | talntid | and goto |
| 19:25.50 | talntid | and gotoif, and such |
| 19:26.05 | p3nguin | The book has a really easy to understand write-up on it. Let me see if I can find it in the html version. |
| 19:26.17 | talntid | i'd appreciate it |
| 19:28.07 | p3nguin | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-IVR.html |
| 19:29.23 | p3nguin | Conversely, http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-AA.html |
| 19:29.26 | talntid | offtopic: sick, a curl module |
| 19:29.55 | p3nguin | haha |
| 19:30.47 | *** join/#asterisk kilgorex (~chatzilla@82-69-63-23.dsl.in-addr.zen.co.uk) |
| 19:30.59 | p3nguin | The term IVR is often misused to refer to an automated attendant, but the two are very different things.[150] |
| 19:31.14 | talntid | i'v always considered them the same too |
| 19:31.36 | p3nguin | [150] We suspect this is because "IVR" is much easier to say than "automated attendant." |
| 19:32.36 | p3nguin | I think, in your case, you actually do want an IVR. |
| 19:32.44 | p3nguin | So your usage of the term would have been correct. |
| 19:33.06 | talntid | never seen usage of same =>, either |
| 19:33.17 | talntid | man i'v been out of the dialplans for wayyyy too long |
| 19:33.28 | p3nguin | I'm not seeing the examples of IVR in the HTML version of the book. :/ |
| 19:33.49 | talntid | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/IVR_id247054.html |
| 19:33.54 | talntid | that's all I'm seeing so far |
| 19:34.36 | p3nguin | That is actually using a combination of both methods. |
| 19:34.57 | p3nguin | You'll notice the extensions 1 and 2, along with the Read()s and GotoIf()s. |
| 19:35.57 | jeffspeff | this problem is about to drive me crazy. |
| 19:36.12 | jeffspeff | if you look at the packet starting at line 3990. On line 3994 it says To: "Renee Decker" <sip:1101@192.168.251.113:5060>;tag=67F73C3B-CC618F00 then on line 3997 Contact: <sip:1101@69.7.96.136:5060> |
| 19:36.48 | jeffspeff | then on line 4453 it gets everything wrong and says c=IN IP4 69.7.96.13613 |
| 19:38.27 | ZenMaster | [TK]D-Fender: So in my extension.conf I need to have something like xlite02,pirority,outbound ? ANd it should work? |
| 19:39.03 | p3nguin | I guess I just remembered the IVR example incorrectly. |
| 19:39.24 | p3nguin | But there is a reasonable auto-attendant example at http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Autoattendant_id287976.html#Autoattendant_id288137 |
| 19:39.32 | jeffspeff | ZenMaster, take a look at this http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 19:39.48 | ZenMaster | jeffspeff: Thank you. |
| 19:40.14 | ZenMaster | Currently the extensions.conf I am looking at has a couple hundred lines in it. |
| 19:40.19 | p3nguin | zenmaster: Delete it. |
| 19:40.28 | p3nguin | If you didn't create it, it is the SAMPLE file. |
| 19:40.37 | p3nguin | Oh, wait... |
| 19:40.42 | p3nguin | You didn't set up this system new. |
| 19:41.01 | p3nguin | Disregard the suggestion to delete the sample file. |
| 19:41.27 | *** join/#asterisk cyborg-one (1000@85.238.116.122) |
| 19:41.33 | *** join/#asterisk cyborg-one (1000@85-238-116-122.broadband.tenet.odessa.ua) |
| 19:41.51 | ZenMaster | p3nguin: I hear ya. |
| 19:42.21 | ZenMaster | Yeah there is a whole GUI in front of this thing. It is a telemarkting system you can feed numbers into and have it dial out. |
| 19:43.22 | ZenMaster | I think I am going to install asterisk on my home linux server and play with that. Becasue this is crazy. |
| 19:43.37 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 19:44.44 | p3nguin | That would help you learn your way around. |
| 19:45.14 | talntid | thanks, p3nguin. That example will help |
| 19:45.17 | ZenMaster | I left to go get a soda and I have two boss's asking me if I have dial tone on this thing yet. |
| 19:45.51 | p3nguin | Yeah. I would have asked, too. |
| 19:46.13 | p3nguin | Configure a peer entry in sip.conf for the phone. Register the phone to asterisk. Dial tone will come with that registration. |
| 19:46.19 | jeffspeff | ZenMaster, technically you do have a dial-tone |
| 19:46.24 | p3nguin | maybe not |
| 19:46.41 | ZenMaster | Well did you guys see the error I posted earlier from the log file? |
| 19:46.45 | p3nguin | Most SIP phones won't provide a dial tone without being associated to some type of system. |
| 19:46.53 | ZenMaster | When I tried to dial my number from xlite? |
| 19:46.54 | jeffspeff | p3nguin, oh |
| 19:47.20 | p3nguin | I did not see the log. |
| 19:47.46 | ZenMaster | Ok this is a machine that I synergy to. It is two machines over. So i am going to type this hold on. |
| 19:48.20 | p3nguin | You can't copy and paste the URL? |
| 19:48.37 | ZenMaster | Note[19756] chan_sip.c: Call from 'xlite02' to extension '951294547' rejected becasue extension not found in context 'internal'. |
| 19:48.45 | p3nguin | oh, that. |
| 19:49.01 | p3nguin | You don't know how to decode that message? |
| 19:49.07 | ZenMaster | I have three computers linked via synergy for a nice little five screen setup. My job is very diversified. |
| 19:49.30 | ZenMaster | Well someone in here gave me a good start. And I am looking in my exstensions file. I am also reading the book. |
| 19:49.38 | p3nguin | A call from a device named 'xlite02' sent a call into asterisk... |
| 19:49.51 | ZenMaster | Now asterisk doe snot know what to do with it. |
| 19:50.25 | p3nguin | The call went into context 'internal', either because it matched a peer entry in sip.conf or because it matched no entry and 'internal' is actually the default context for unauthenticated calls. |
| 19:50.32 | ZenMaster | That is why I posted something earlier saying do I need to add a line to extensions.conf saying something like exten => 'xlite02',1,Outbound or something? |
| 19:50.54 | ZenMaster | Hmmm. |
| 19:51.08 | jeffspeff | ZenMaster, look back at the _XXXXXXXX thing i sent you |
| 19:51.11 | p3nguin | When the call arrived in the internal context, it tried to match an extension of '951294547', which did not exist. |
| 19:51.14 | p3nguin | Contexts and extensions are in extensions.conf |
| 19:51.39 | p3nguin | exten => 951294547,1,SometimesUseful() |
| 19:51.51 | jeffspeff | ZenMaster, also look at the patern matching in that link too |
| 19:52.05 | p3nguin | extension 951294547. Priority 1. Application in this example is SomethingUseful(). |
| 19:52.23 | ZenMaster | Ok, daddy used to call me sunny becuase he thought I was bright. |
| 19:52.24 | ZenMaster | Brb. |
| 19:52.25 | p3nguin | Extensions do not have to dial phones. |
| 19:52.38 | ZenMaster | That is what the book is saying. |
| 19:52.39 | p3nguin | Extensions do not have to dial other systems. |
| 19:52.51 | p3nguin | Extensions can do whatever you want them to do. |
| 19:53.42 | p3nguin | The reason for your error message was that you do not have an extension of 951294547 in the context where the call was sent. |
| 19:53.53 | ZenMaster | How are people able make whatever outbound numbers they want? |
| 19:54.03 | p3nguin | huh? |
| 19:54.08 | p3nguin | Make... what? |
| 19:54.22 | p3nguin | Caller ID numbers? |
| 19:55.14 | ZenMaster | I am getting confused. |
| 19:55.39 | ZenMaster | I need to create a extension number letter whatever that lets asterisk know what to do right? |
| 19:55.50 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 19:55.50 | p3nguin | There is somewhat of a disconnection between a phone call and caller ID numbers. People who only use a telephone company think that phone numbers are tied to phone lines, and that the caller ID number is always what your phone number is. Is that what you're asking about? |
| 19:56.10 | p3nguin | Yes, you have to create an extension for something to happen. |
| 19:56.22 | p3nguin | Extensions are "dialing rules." |
| 19:56.44 | ZenMaster | Ok I think I may know what is going on. |
| 19:57.06 | ZenMaster | Since there are so many lines in this extensions.conf I am sure there is probably already one setup for outbound right? Should be? |
| 19:57.17 | p3nguin | You created the 'xlite02' peer in sip.conf, yes? |
| 19:57.40 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:58.35 | ZenMaster | I have device and and agent in the front end. I think it is storing it in mysql db. |
| 19:58.37 | dhill | odd, if i connect to the public IP on asterisk, it authenticates. If i switch the IP to the private IP, I get unauthorized.. |
| 19:59.36 | ZenMaster | p3nguin: Excuse me. |
| 19:59.56 | ZenMaster | I have XCasdtLabs as peer in the sip.conf That is my sip provider. |
| 20:00.16 | ZenMaster | xlite02 is my user auth user etc I used to connect to the asterisk box via Xlite. |
| 20:00.18 | p3nguin | okay... |
| 20:00.22 | p3nguin | How that does affect the xlite02 peer we were talking about? |
| 20:00.33 | [TK]D-Fender | ZenMaster, Since there are so many lines in this extensions.conf I am sure there is probably already one setup for outbound right? Should be? <- no |
| 20:00.39 | [TK]D-Fender | ZenMaster, it is yoru job to invent these. |
| 20:00.42 | ZenMaster | XCastLabs. |
| 20:00.46 | p3nguin | Concentrate on the xlite02 peer entry. |
| 20:00.46 | ZenMaster | [TK]D-Fender: Roger. |
| 20:00.52 | p3nguin | We aren't talking about the ITSP yet. |
| 20:01.08 | [TK]D-Fender | p3nguin, Yeah he's got peers for both ends, just no dialplan |
| 20:01.15 | ZenMaster | p3nguin: Ok.:) |
| 20:01.22 | p3nguin | It's a running, operating system, so there has to be dial plan. |
| 20:01.42 | p3nguin | Look at the context you assigned for the phone. |
| 20:01.51 | p3nguin | You chose internal for the context. |
| 20:02.00 | [TK]D-Fender | He doesn't understand patterns, contexts or anything yet. |
| 20:02.07 | [TK]D-Fender | Fresh off the newb truck... |
| 20:02.11 | ZenMaster | Well gentlemen. |
| 20:02.13 | ZenMaster | Here is what is hard. |
| 20:02.22 | p3nguin | Are there other phones already on the system? Do they use the internal context? |
| 20:02.23 | ZenMaster | Half of this system is done on a front end GUI. |
| 20:02.25 | [TK]D-Fender | ZenMaster, Go read up on those chapters in the book... |
| 20:02.34 | ZenMaster | [TK]D-Fender: Currently doing so. |
| 20:02.40 | [TK]D-Fender | ZenMaster, We have no idea what that thing of yours made if anything |
| 20:02.42 | p3nguin | I really couldn't care less about this GUI. |
| 20:03.03 | p3nguin | I just want you to concentrate on the peer entry for the phone. |
| 20:03.21 | ZenMaster | Ok, |
| 20:03.26 | ZenMaster | In the sip.conf right? |
| 20:03.38 | p3nguin | Where ever you configured it. |
| 20:03.38 | ZenMaster | xcastlabs needs to be the name of the phone> |
| 20:03.43 | ZenMaster | ? |
| 20:04.11 | p3nguin | Patience is wearing thin. |
| 20:04.57 | p3nguin | If you can't stay with me on this, I have something else I could be working on. |
| 20:05.07 | ZenMaster | Ok, sorry brother. |
| 20:05.17 | ZenMaster | I made that peer entry in my sip.conf |
| 20:05.24 | p3nguin | Are you going to stay with me and answer my question? |
| 20:05.30 | ZenMaster | That is were I am at. |
| 20:05.32 | ZenMaster | Yes. |
| 20:05.39 | p3nguin | You created xlite02 in sip.conf? |
| 20:05.47 | ZenMaster | Yes. |
| 20:05.51 | p3nguin | You gave it a context value of "internal". |
| 20:06.10 | p3nguin | Are there other PHONES configured in sip.conf? |
| 20:06.19 | ZenMaster | No. |
| 20:06.31 | ZenMaster | As a matter of fact I left that blank. Because at the time. |
| 20:06.35 | dhill | thanks everyone here.. it was indeed a route issue :).. |
| 20:06.36 | ZenMaster | I did not know what that meant. |
| 20:06.45 | p3nguin | Do you have a way to check the context for the other phones that are configured? |
| 20:06.50 | Ice_Strike | is this correct |
| 20:06.53 | Ice_Strike | Monitor(wav,/root/recording/bsc/${calr_id},m) |
| 20:06.59 | Ice_Strike | I wanna save it in the root |
| 20:06.59 | p3nguin | Either in the DB or on the CLI? |
| 20:07.29 | ZenMaster | p3nguin: No phones currently configured. So no. :&*( |
| 20:07.41 | p3nguin | Oh, so only the one entry for the ITSP? |
| 20:08.01 | ZenMaster | Yes I believe so. |
| 20:08.06 | p3nguin | Okay, that's fine. What was this system doing before you got to it? |
| 20:08.23 | ZenMaster | Nothing, it was just deployed. My task was to configure phones and users for it. |
| 20:08.27 | p3nguin | Having a PBX with only an ITSP and no phones doesn't make much of a phone system. |
| 20:08.36 | ZenMaster | Under Development for two years, but one of this companies side projects. |
| 20:08.55 | p3nguin | *sigh* So what have they been doing with it? |
| 20:09.08 | p3nguin | You said it has all kinds of dial plan already on there. |
| 20:09.22 | ZenMaster | Using it as a dialing platform. |
| 20:09.24 | ZenMaster | Outbound. |
| 20:09.30 | ZenMaster | They would like to use it now as a p1. |
| 20:09.45 | p3nguin | Without any phones? That doesn't make a very good phone call to not have a phone. |
| 20:10.11 | ZenMaster | Ehehe. |
| 20:10.18 | p3nguin | Okay, forget about that part. |
| 20:10.24 | ZenMaster | Ok. |
| 20:10.35 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 20:10.38 | p3nguin | Let's pretend for a minute that it is a new system and it needs to have phones and extensions for the phones to be able to do things. |
| 20:10.49 | ZenMaster | 10-4. |
| 20:10.54 | p3nguin | I always configure all of my phones with a context of 'phones'. |
| 20:11.26 | p3nguin | In extensions.conf I create a hierarchy of what other contexts a phone can "reach" by using includes. |
| 20:11.40 | ZenMaster | Ok. |
| 20:11.50 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 20:12.03 | p3nguin | So if phones should have access to the internal extensions, I include internal. If the phones need access to outbound extensions, I include outbound. |
| 20:12.52 | p3nguin | Never overlap inbound contexts and outbound contexts because that leaves a possibility for someone to call into the system and then make calls back out... on the company dime. |
| 20:13.44 | *** part/#asterisk dhill (~dhill@fog.mindcry.org) |
| 20:13.49 | ZenMaster | Just reading about that. |
| 20:13.57 | p3nguin | Here is an example of an extensions.conf using this concept: http://pastebin.com/Piqv4Egj |
| 20:14.10 | p3nguin | Look at lines 155-159. |
| 20:15.17 | ZenMaster | Looks pretty straight forward. |
| 20:15.43 | p3nguin | You can see that a phone, which was assigned the 'phones' context, would have access to dial extensions which reach other phones as well as being able to call off-site. |
| 20:15.46 | talntid | seems to be all working as expected, p3nguin =] |
| 20:15.53 | p3nguin | Awesome. |
| 20:16.05 | ZenMaster | Ok one second. |
| 20:17.19 | *** join/#asterisk fubada (~fubada@ool-4573155d.dyn.optonline.net) |
| 20:17.22 | fubada | hi |
| 20:17.32 | fubada | i just installed asterisk-gui from epel, enabled http, but gettin a 403 |
| 20:17.41 | fubada | how do i access and what is the default use/rpass |
| 20:18.45 | *** part/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 20:19.00 | [TK]D-Fender | fubada, #asterisk-gui <- |
| 20:19.05 | fubada | ty |
| 20:19.06 | ZenMaster | p3nguin: Thank you thus far, I have to take a short break as to make some changes to our e-mail server. |
| 20:19.20 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 20:22.17 | talntid | ZenMaster, do you have dialtone yet? ;) |
| 20:22.43 | ZenMaster | Not at the moment. |
| 20:22.50 | p3nguin | Does X-Lite provide off-hook dialing? |
| 20:22.53 | ZenMaster | I have had to take a short break from that for some other administrative tasks. |
| 20:22.57 | p3nguin | If not, there will never be a dial tone. |
| 20:22.58 | ZenMaster | I am not sure. |
| 20:23.10 | ZenMaster | Well either way, I can't make an outbound call at the moment. |
| 20:23.32 | ZenMaster | p3nguin: Alas I have not messed with extensions.conf yet. I will be back in five. |
| 20:23.44 | p3nguin | Once you create correct context and extension, then you can dial outbound. |
| 20:24.08 | ZenMaster | p3nguin: I think you are correct. You I am sure know you are correct. Thank you once again. |
| 20:29.22 | [TK]D-Fender | checkout time, BBIAB |
| 20:30.58 | *** join/#asterisk mpe (~mpe@mobile-80-62-116-164.broadband.tdc.dk) |
| 20:35.07 | ZenMaster | Ok now that all of that stuff is done, now to look at my extensions.conf |
| 20:40.29 | ZenMaster | No way! |
| 20:40.32 | ZenMaster | It is working now? |
| 20:40.34 | ZenMaster | WTF? |
| 20:40.40 | ZenMaster | Over. |
| 20:41.55 | p3nguin | Magically, or did you change something? |
| 20:42.12 | ZenMaster | Think when I changed the context. |
| 20:42.39 | p3nguin | Changed it to what? |
| 20:42.59 | ZenMaster | p3nguin: Let me check my logs. |
| 20:50.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:52.59 | ZenMaster | p3nguin: It looks like there are a lot of cron scripts that are running. And whatever I have done the sip.conf it added the appropiate config to extension.conf |
| 20:53.06 | ZenMaster | p3nguin: This is what I see in the log. |
| 20:54.13 | ZenMaster | pastebin.com/afw3nb6b |
| 20:54.31 | p3nguin | empty paste |
| 20:56.46 | ZenMaster | pastebin.com/afw3nb6B |
| 20:56.50 | ZenMaster | Case sensitive? |
| 20:57.28 | p3nguin | Aren't those only hints? |
| 20:57.44 | p3nguin | dialplan show aphone01@internal |
| 20:58.29 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 20:58.45 | ZenMaster | Hmmm. |
| 20:58.50 | ZenMaster | One second. |
| 21:00.43 | *** join/#asterisk wtf911 (~wtf911@cpe-67-251-103-137.stny.res.rr.com) |
| 21:01.39 | ZenMaster | The container in my sip.conf is now named [outbound]. And there was another one made under neath it with the same information [outbount1]. |
| 21:01.53 | p3nguin | automatically? |
| 21:03.36 | ZenMaster | The other admin is logged in. |
| 21:03.46 | p3nguin | haha |
| 21:03.46 | ZenMaster | He is explaining to me why what I did did not work. |
| 21:03.50 | ZenMaster | Makes sense. |
| 21:03.56 | ZenMaster | I was 99% right he says minus one name. |
| 21:04.11 | jeffspeff | ok, so... i got that other stuff figured out... now i'm having an issue with audio. that office has 4 Polycom phone which are able to send and recieve audio. However, they also have a few Aastra's that can only recieve audio, but not send. Any ideas? |
| 21:05.09 | ZenMaster | jeffspeff: My next step is to finish getting these 6755i phones working. :D Think I have that handled though.. |
| 21:05.30 | jeffspeff | i was talking about my own problems. lol |
| 21:05.49 | p3nguin | Group therapy. |
| 21:05.59 | jeffspeff | p3nguin, true |
| 21:06.09 | jeffspeff | any ideas on my aastras? |
| 21:06.32 | ZenMaster | p3nguin: Thank you for all the excellent support in a system I totally stepped into not knowing a thing. It has been an interesting three days. |
| 21:07.11 | Ice_Strike | Monitor(wav,/root/recording/bsc/${calr_id},m) |
| 21:07.14 | Ice_Strike | thats correct? |
| 21:07.30 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:07.35 | jeffspeff | Ice_Strike, is that your callerid variable nam? |
| 21:07.37 | jeffspeff | *name |
| 21:07.56 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:07.58 | [TK]D-Fender | Running * as root is not advised |
| 21:08.01 | p3nguin | Asterisk user should not have permissions to write into that directory, so expect failure. |
| 21:08.40 | p3nguin | If asterisk is able to write there, you're probably doing something wrong. See [tk]d-fender's remark. |
| 21:09.19 | Ice_Strike | I have not tested it |
| 21:09.35 | Ice_Strike | Where do you think I should save it |
| 21:09.50 | Ice_Strike | jeffspeff ${calr_id} is custom variable yea |
| 21:09.58 | p3nguin | What is wrong with the standard location? |
| 21:10.12 | p3nguin | Monitor(wav,${calr_id},m) |
| 21:10.21 | Ice_Strike | p3nguin Because sfpt can only go to /root when i logged in |
| 21:10.38 | p3nguin | What is an sfpt? |
| 21:10.39 | Ice_Strike | sftp* |
| 21:10.44 | p3nguin | You're doing it wrong, then. |
| 21:11.09 | p3nguin | If you are logged in as root, you should be able to go almost anywhere unrestricted. |
| 21:11.28 | Ice_Strike | on sftp? |
| 21:11.36 | Ice_Strike | let me check again |
| 21:11.57 | Ice_Strike | ahh you are right |
| 21:12.01 | Ice_Strike | i've clicked on / |
| 21:12.20 | Ice_Strike | why saving in /root is not recommended? |
| 21:12.29 | [TK]D-Fender | not what I said |
| 21:12.50 | Ice_Strike | would that work |
| 21:12.51 | Ice_Strike | Monitor(wav,bsc/${calr_id},m) |
| 21:14.33 | p3nguin | You probably should create the bsc subdir manually. |
| 21:14.39 | jeffspeff | i have 4 Polycom phone which are able to send and recieve audio. However, there are also a few Aastra's that can only recieve audio, but not send. Any ideas? |
| 21:14.40 | Ice_Strike | Yep I will |
| 21:14.40 | p3nguin | /var/spool/asterisk/bsc |
| 21:14.48 | p3nguin | Then it should be okay. |
| 21:15.06 | Ice_Strike | /var/spool/asterisk/bsc ? |
| 21:15.11 | p3nguin | But you still should not run asterisk as root. To check, run the command: ps -C asterisk u |
| 21:15.24 | p3nguin | Uh, no... I mistyped the path... |
| 21:15.31 | p3nguin | /var/spool/asterisk/monitor/bsc |
| 21:15.33 | Ice_Strike | yep |
| 21:15.43 | p3nguin | Sometimes you have to debug what I type. |
| 21:15.50 | Ice_Strike | 2286 root 20 0 2435m 29m 4084 S 6.0 2.9 34:17.59 asterisk |
| 21:16.03 | Ice_Strike | asterisk running as root i think |
| 21:16.03 | p3nguin | Running it with root, I see. |
| 21:16.11 | Ice_Strike | thats wrong? |
| 21:16.11 | p3nguin | You should fix that soon. |
| 21:16.27 | Ice_Strike | hmmm is how it was done when i install asterisk |
| 21:16.42 | [TK]D-Fender | All in the book |
| 21:16.42 | Ice_Strike | then i typed service asterisk start |
| 21:16.50 | Ice_Strike | cool |
| 21:16.56 | p3nguin | CentOS? |
| 21:16.58 | Ice_Strike | Yep |
| 21:16.58 | [TK]D-Fender | service defines who it is really running as |
| 21:17.38 | p3nguin | Fix your /etc/sysconfig/asterisk AST_USER and AST_GROUP. |
| 21:17.49 | Ice_Strike | Cool will do |
| 21:18.04 | p3nguin | And also the user and group settings in asterisk.conf. |
| 21:18.24 | Ice_Strike | do i need to create user and group first |
| 21:18.31 | p3nguin | If you don't have it, yes. |
| 21:18.40 | Ice_Strike | what did you set to? |
| 21:18.43 | p3nguin | getent passwd asterisk |
| 21:18.53 | p3nguin | Does it exist already? |
| 21:19.06 | Ice_Strike | nope |
| 21:20.45 | p3nguin | groupadd -g 5060 asterisk |
| 21:20.47 | p3nguin | useradd -c Asterisk -d /var/lib/asterisk -g asterisk -s /bin/false -u 5060 asterisk |
| 21:20.53 | p3nguin | (I think) |
| 21:21.01 | Ice_Strike | ty |
| 21:21.27 | p3nguin | I typically do not have to do this manually because the package manager takes care of it with install scripts. |
| 21:21.43 | p3nguin | I guess I should consult the script. |
| 21:22.25 | p3nguin | Looks okay to me. |
| 21:22.54 | p3nguin | After you add the group and user, then lock the password: passwd -l asterisk |
| 21:23.30 | p3nguin | Then change the ownership of the files so asterisk can deal with them accordingly: chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk |
| 21:25.23 | *** join/#asterisk schultza (~allen@rc1.rcherbals.com) |
| 21:25.35 | schultza | where can i get help with zaptel? |
| 21:26.54 | *** join/#asterisk mpe (~mpe@mobile-80-62-116-164.broadband.tdc.dk) |
| 21:28.03 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:28.05 | [sr] | yellow |
| 21:28.07 | [sr] | WIMPy: hi |
| 21:28.25 | [sr] | people, which is the directory for voicemail? don't remember :S |
| 21:29.12 | p3nguin | /var/spool/asterisk/voicemail |
| 21:29.13 | [TK]D-Fender | [sr]: asterisk.conf |
| 21:29.41 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 21:29.49 | [sr] | [TK]D-Fender: ya, it's there, tks |
| 21:29.52 | [sr] | p3nguin: tks also |
| 21:29.54 | *** join/#asterisk b1rkh0ff (~b1rkh0ff@178.77.20.215) |
| 21:30.13 | p3nguin | And a tiki torch to you, as well. |
| 21:30.25 | WIMPy | hi [sr] |
| 21:33.31 | [sr] | p3nguin: whats that? :p |
| 21:34.11 | p3nguin | Prezactly. |
| 21:34.30 | *** join/#asterisk gusto (~gusto@88.128.14.113) |
| 21:36.08 | Ice_Strike | p3nguin thanks man |
| 21:39.14 | [sr] | so news? |
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| 22:00.02 | *** join/#asterisk usam (54d03a48@gateway/web/freenode/ip.84.208.58.72) |
| 22:00.11 | p3nguin | |
| 22:03.04 | usam | HEllo, Im having problem with my asterisk, Im running 1.8.11.1with latest libpri and latest dahdi,... I have had sangoma to remote debug my box and they are saying that PRI D-Channel is being reported as down when it is not. Im loosing the connection with my e1 gw 4-5 times a day.. Where can I get help? checked the mailing-list, there is not much activity there. Please advice. |
| 22:04.32 | WIMPy | What does "D-ch is reported as down when it is not" mean? We need more information. |
| 22:04.44 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 22:04.46 | *** part/#asterisk schultza (~allen@rc1.rcherbals.com) |
| 22:06.04 | usam | WIMPy: Im getting this error in the log: asterisk[25929]: WARNING[25956]: sig_pri.c:1112 in pri_find_dchan: Span 1: D-channel is down! then I have to restart asterisk to have it working again. |
| 22:06.44 | WIMPy | That's all you're getting? |
| 22:07.05 | usam | Yes,... |
| 22:07.33 | usam | should i be getting more errors? |
| 22:08.10 | [sr] | going to sleep |
| 22:08.18 | WIMPy | I certainly feel unable to comment on only that line. |
| 22:08.19 | usam | I have a script that monitors the syslog and restart the asterisk as soon as it sees 'No D-channel' |
| 22:08.45 | WIMPy | You only restart Asterisk? Or dahdi as well? |
| 22:08.56 | usam | only asterisk |
| 22:09.23 | usam | sometimes i have to do it several times to have it up and working again.. |
| 22:10.03 | sruffell | usam: BRI, T1, or E1? |
| 22:10.32 | usam | E1 |
| 22:11.03 | usam | Im using sangoma A102D |
| 22:12.28 | sruffell | And Sangoma tech support said the problem is with your provider? Otherwise, the alarm state comes from the driver, and the "up/down" on the span comes from libpri / layer2. What was their recommendation for you? |
| 22:15.03 | usam | Im using a Portech GSM PRI gateway. http://pastebin.com/XqVgufeP <--- last email from sangoma |
| 22:16.25 | sruffell | usam: If the driver is not reporting any errors / alarms, then most likely you will see why it's going down if you use enable pri intense debug. |
| 22:16.38 | zamba | i'm looking for a cheap interface card to get my four isdn lines up and running.. any suggestions? |
| 22:17.06 | WIMPy | zamba: BRI or PRI? How many? |
| 22:17.16 | WIMPy | Do you need EC? |
| 22:17.25 | zamba | i'm actually not sure if it's BRI or PRI.. how do i know the difference? |
| 22:17.39 | usam | sruffell: OK ... I will disable the autorestart script and use your recommendation |
| 22:18.32 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fspvuksdcwdkspjt) |
| 22:19.01 | WIMPy | zamba: Bad situation. You should know what yu need. BRI=2 channels, PRI=30/24 channels |
| 22:19.59 | zamba | it's 2x BRI then |
| 22:21.01 | WIMPy | If you have two spare PCi slots you can save lots of money by getting just two off the shelf cards. |
| 22:33.52 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 22:48.59 | zamba | do you have an example of an "off the shelf card"? |
| 22:50.45 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:50.53 | *** join/#asterisk JamKo (~JamKo@mail.jaguardata.com) |
| 22:50.56 | fubada | hi |
| 22:51.00 | fubada | can someone tell me what this error means |
| 22:51.02 | fubada | [May 16 18:50:19] WARNING[5975]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 22:51.08 | fubada | when calling a local extension |
| 22:51.12 | fubada | another local phone |
| 22:51.26 | WIMPy | The peer isn't registered. |
| 22:51.32 | fubada | how do i register the peer |
| 22:52.45 | WIMPy | By configuring that device. |
| 22:53.16 | fubada | its the inseucre setting |
| 22:53.19 | fubada | one sec |
| 23:00.58 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-todopjnmpmwfiybq) |
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| 23:05.32 | *** part/#asterisk teknoprep (~chrisrawl@unaffiliated/teknoprep) |
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| 23:13.25 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 23:14.17 | sawgood | Hi: yesterday, while troubleshooting voicemail to email from the Asterisk CLI (when voicemails were being processed after a call (I could see the smtp commands flow by on the Asterisk CLI) ... I am not getting that anymore ... |
| 23:14.23 | sawgood | any tips on why that is the case? |
| 23:14.53 | sawgood | I would see the 200, 220, and 250 SMTP messages inside the Asterisk CLI ... |
| 23:15.09 | sawgood | voicemail to email is still working though |
| 23:19.35 | *** join/#asterisk looserouting (~paolo@ip-62-143-235-156.unitymediagroup.de) |
| 23:20.05 | *** join/#asterisk greenwolf (a6f807bb@gateway/web/freenode/ip.166.248.7.187) |
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| 23:24.18 | talntid | http://pastebin.com/x3KEbfxk |
| 23:24.35 | talntid | [May 16 16:23:29] NOTICE[9827]: pbx.c:4110 pbx_extension_helper: Cannot find extension 'fanout_call' in context 'fanout_call' |
| 23:24.36 | talntid | [May 16 16:23:29] WARNING[9827]: pbx.c:10172 pbx_parseable_goto: Priority 'playmessage' must be a number > 0, or valid label |
| 23:25.04 | talntid | when I tried goto(playmessage), I got [May 16 16:20:24] NOTICE[9796]: pbx.c:4118 pbx_extension_helper: No such label 'playmessage' in extension '3' in context 'fanout_call' |
| 23:25.41 | talntid | trying s,playmessage |
| 23:25.57 | talntid | that worked. do'h |
| 23:29.27 | sawgood | no tips or ideas why the SMTP commands were visable from the Asterisk CLI (I sort of liked seeing those) |
| 23:30.39 | p3nguin | Goto(playmessage) would only work if you have a label 'playmessage' in the current extension. |
| 23:31.04 | greenwolf | yes correct p3nguin |
| 23:31.14 | p3nguin | ? |
| 23:31.28 | greenwolf | you must already establish or delcare that label within the current context |
| 23:31.36 | WIMPy | ...and you are lucky. |
| 23:31.39 | p3nguin | Do I look like I was asking? |
| 23:31.40 | greenwolf | i was agreeing wtih you p3nguin sorry |
| 23:31.44 | p3nguin | og |
| 23:31.45 | p3nguin | oh |
| 23:31.49 | greenwolf | excuse me im sorry damn |
| 23:32.41 | greenwolf | you know im really getting tired of ppl in this community with their condescending attitudes |
| 23:32.48 | greenwolf | im out |
| 23:33.40 | sawgood | tired ... of a forum full of minds which speak? |
| 23:33.50 | sawgood | somebody's having a bad day |
| 23:34.42 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:34.43 | greenwolf | no sawgood i was agreeing with someone |
| 23:34.45 | greenwolf | and they bit at me |
| 23:34.59 | sawgood | like a pitbull dog? |
| 23:35.06 | greenwolf | yes exactly |
| 23:35.11 | sawgood | I've had my ass bit more than a dozen times here |
| 23:35.26 | greenwolf | yea its totally un called for and rude |
| 23:35.42 | greenwolf | actually ppl are talkin about that on a forum about this community |
| 23:35.56 | greenwolf | how ppl are just so SUPERIOR than others and totally walk around acting like it |
| 23:36.15 | greenwolf | its getting out of hand..i mean this a an opensource community full of ideas and collaboration |
| 23:36.43 | greenwolf | this is not proprietary software so everyone is entitled to their opinion |
| 23:43.12 | fubada | hey guys on Incoming rules, if I want to use Local Extension by DID, can my DID on sip prvider side be a 7 digit number, but a local extension be a 4 digit number? |
| 23:43.26 | fubada | I dont want to write an incoming rule for each extension |
| 23:43.39 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
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| 23:44.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:44.53 | WIMPy | You can use Goto or switch. |
| 23:45.45 | fubada | can you explain please |
| 23:46.30 | WIMPy | If you ask a more specific question, I will give a more specific answer. |
| 23:46.43 | fubada | ok one sec :) |
| 23:46.53 | p3nguin | If you don't want to create an extension for every different thing, you'll have to use an extension pattern. |
| 23:48.11 | fubada | my pattern: _1NXXNXXXXXX |
| 23:48.20 | p3nguin | That's a good NANP pattern. |
| 23:48.20 | fubada | i want to du local extension by did pattern |
| 23:48.45 | fubada | but only use the last 4 in ${EXTEN: |
| 23:48.56 | fubada | my local extensions arre the last 4 digits of my did |
| 23:49.03 | WIMPy | Which way round do you want to do it? |
| 23:49.28 | p3nguin | I understand the part about last four digits of the DID for local phones, but what are you trying to do? |
| 23:49.47 | fubada | im trying to set up an incoming rule, but avoid having to define one for each did/extension |
| 23:50.04 | fubada | i want to setup a pattern, and ring the last 4 digits of that pattern on incoming calls |
| 23:50.12 | p3nguin | We have no concept of "incoming rule." You just create dial plan as needed. |
| 23:50.13 | WIMPy | The in your ITSPs context just do a Goto(localextensipns,${EXTEN:4:4},1) |
| 23:50.41 | fubada | im using asterisk gui :) |
| 23:50.49 | p3nguin | We don't support that here. |
| 23:51.04 | fubada | heh |
| 23:51.13 | p3nguin | We can help you with extensions and patterns, but we can't tell you how to do it with a GUI. |
| 23:51.31 | fubada | ok, can I show ya my extensions.conf |
| 23:52.07 | fubada | too long to pastebin |
| 23:52.15 | greenwolf | yea gui's just limit you anyways |
| 23:52.17 | p3nguin | Good grief. |
| 23:52.24 | greenwolf | cant do much except what is written in freepbx |
| 23:52.29 | greenwolf | useless for me |
| 23:52.35 | greenwolf | no creativity in gui |
| 23:52.55 | p3nguin | We don't need the entire extensions.conf. We just need to know what you want to do with the calls. |
| 23:54.41 | p3nguin | If you are simply trying to send calls which go to a 10-digit DID number to a phone via 4-digit extension number, the method described by wimpy will do it, sans any syntax error he may have included. |
| 23:56.23 | fubada | thanks, jsut not sure where to place it, limited by asterisk-gui |
| 23:57.08 | WIMPy | You shouldn;t take for garnted it will be possible at all. |
| 23:57.29 | p3nguin | Goto(locals,${EXTEN:-4},1); will send the call to the last four digits of the current extension in the locals context. |
| 23:58.16 | WIMPy | Right. There was a - missing. |
| 23:58.33 | p3nguin | In the case of exten => 3149691077,1,Goto(locals,${EXTEN:-4},1); the call would go to locals context extension 1077. |
| 23:59.20 | p3nguin | The same will be true when using the NANP pattern. |
