00:05.18 | manxpower | ~answers |
00:05.19 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
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00:55.01 | jelly-bean1 | i have a poly 320 and 330 and 501. the 320 and 330 work file downloading their firmware and getting IPs etc. the 501 cannot even get an IP via DHCP. any ideas what i have to do to the 501? i have *468 already |
00:55.46 | thehar | ~switchvox |
00:56.13 | manxpower | jelly-bean1: usually that means you pluged the lan cable into the PC port on the phone |
00:56.36 | jelly-bean1 | hehe i did do that once to the 320 but i just double-checked and its in the LAN port |
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00:57.00 | raden_work | [TK]D-Fender, dude reg issues with vitelity again tonight :( |
00:57.24 | raden_work | restarted asterisk didnt go away |
00:57.51 | raden_work | rebooted router same BS |
00:58.03 | raden_work | shutdown asterisk for 5 min all was good |
00:58.12 | manxpower | I never have registration issues with Vitelity |
00:58.24 | manxpower | maybe your router just sucks? |
00:58.32 | raden_work | is there a way when a registration fails for it to take like a 2 min break before it tries again |
00:58.41 | raden_work | manxpower, ill know when i get my asa 5505 in place |
00:58.52 | raden_work | I think part of this is router related |
00:59.23 | manxpower | raden_work: I assume you have qualify=yes for that peer. |
00:59.27 | raden_work | there any downfall to stacking HP procurve switches |
00:59.36 | raden_work | manxpower, yes but the registration itself is failing |
00:59.43 | raden_work | was failing |
00:59.57 | manxpower | raden_work: go to wal-mart. spend $60 on a linksys |
01:00.16 | manxpower | 1) the problem is fixed or 2) you now have a spare "router" |
01:00.28 | manxpower | sounds like win-win to me |
01:02.43 | manxpower | jelly-bean1: there are only a few things that could be wrong. The connection from the phone to the DHCP server might have a problem (wiring) or the boot setup is set to use a specific VLAN or is set to use CDP and you don't have CDP setup. After checking those things it's time to RMA the phone and get on with your life. |
01:02.49 | raden_work | manxpower, I have a Cisco ASA-5505 here just need to get it configured |
01:03.10 | manxpower | raden_work: you have spent weeks on this problem. |
01:03.29 | jelly-bean1 | manxpower: it is set to use CDP. wtf is cdp? |
01:03.30 | raden_work | manxpower, yes i know :( |
01:03.48 | raden_work | manxpower, goal next week get cisco in place |
01:03.49 | manxpower | jelly-bean1: CDP = method to automatically select the right VLAN for the phone to use. |
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01:03.51 | raden_work | then groupwise |
01:04.00 | raden_work | then ups software for linux |
01:04.01 | raden_work | then done |
01:04.30 | raden_work | grr i have to go bbl |
01:17.33 | jaytee | loves CDP |
01:17.50 | manxpower | jaytee: me too, but if you don't understand it...... |
01:18.02 | thehar | giggles |
01:18.21 | jaytee | also loves Option 66 for DHCP to point to my server that handles FTP provisioning |
01:19.06 | jaytee | now if I can just get SIP TCP working with Exchange Unified Messaging I can dump running sipX on a VM |
01:19.10 | manxpower | jaytee: me too. Deploying a phone should just require plugging it into the LAN and let the network take care of it. |
01:20.06 | jaytee | manxpower, yeah well I've written a bash script that copies config templates and uses SED to edit them with parameters I pass on the command line |
01:21.04 | jaytee | so for a 330 I just type ./prepphone.sh "MAC ADDRESS" XXXX and I'm done where XXXX is the 4 digit phone extension |
01:21.06 | manxpower | jaytee: I'm working on a cgi script to parse an uploaded .CSV and bulk build extensions and config files |
01:21.26 | jaytee | I'm working on a web interface to run the scripts from PHP |
01:21.47 | manxpower | jaytee: no matter what I do I always seem to get dragged into FreePBX |
01:22.22 | jaytee | not me, I'd rather write my own custom GUI. I've got too many things in my dialplan that FreePBX would kill |
01:22.54 | manxpower | jaytee: Unfortunately I can't dictate what a customer gets anymore since I'm not a consultant anymore |
01:24.06 | jaytee | this is due to setting up my dialplan originally to be able to have 4 digit dialing between my now decommissioned Nortel Option 11C and Asterisk and modeling certain Nortel features in the * dialplan. |
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02:16.43 | DJCharlie | Evening all. I have a small question about setting up a dialplan. I think I have it laid out properly, but when I try choosing from my main menu, I get hung up on with an error of "WARNING[4974]: pbx.c:2437 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension '2' in context 'incoming', but no invalid handler" |
02:17.02 | DJCharlie | A log (and part of my dialplan) are at http://pastebin.com/d2920025c |
02:17.10 | DJCharlie | Can someone help me fix this? |
02:19.51 | DJCharlie | Hello? |
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02:24.27 | DJCharlie | Anyone awake? |
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02:26.33 | scalex000 | good evening! |
02:26.42 | DJCharlie | Evening. :) |
02:26.44 | scalex000 | I have couple question about sip.conf |
02:26.56 | scalex000 | who can help me? |
02:27.01 | DJCharlie | Join the club. I'm still waiting. |
02:27.12 | scalex000 | let me paste the packet |
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02:35.41 | scalex000 | http://pastebin.ca/1579866 |
02:36.04 | scalex000 | line 53 |
02:36.59 | DJCharlie | Wish I could help you scale, but I'm still trying to figure out a dialplan. |
02:37.15 | DJCharlie | And I don't believe anyone else in here is really here. |
02:37.50 | scalex000 | they are here, maybe very tired to type. |
02:38.16 | superbeef | it's crazy its like people might do othe rshit on a friday night |
02:38.40 | DJCharlie | Well, I don't understand why anyone would idle in a support channel of all places. |
02:39.19 | superbeef | support channel or not, IRC is historically a presence based environment |
02:39.33 | superbeef | besides people can scroll back and look for something interesting if they have to |
02:40.33 | DJCharlie | Well would you care to scroll back and see if you could help scale or myself please? |
02:43.43 | superbeef | so it's choking on this? |
02:43.44 | superbeef | lets call Pennsylvania Pennsylvakia |
02:43.46 | superbeef | no |
02:43.48 | superbeef | wrong paste lol |
02:43.55 | superbeef | it's choking on this |
02:43.56 | superbeef | exten => t,2,Goto(menu2,2,1) |
02:43.57 | superbeef | right? |
02:44.00 | scalex000 | superbeef: do you know where I can put how to interconnect BCM and asterisk |
02:44.13 | superbeef | what's BCM? |
02:44.28 | scalex000 | superbeef: nortel BCM |
02:44.40 | superbeef | scalex000: probably a SIP trunk |
02:44.42 | DJCharlie | superbeef: Actually, it chokes no matter which menu option I choose. |
02:45.24 | scalex000 | superbeef: I want to share my experience ohh days and days reading so another not take too long to do it |
02:46.35 | superbeef | DJCharlie: so what's this guy doing exactly |
02:47.09 | superbeef | scalex000: not working so easily eh? |
02:47.20 | DJCharlie | superbeef: From the first answering message [incoming] you should be able to dial 2, 4, 5, 9, or 777. |
02:47.41 | DJCharlie | BUT, no matter what you dial, it hangs up immediately, with that error message. |
02:47.48 | scalex000 | superbeef: well for now yes |
02:48.28 | scalex000 | superbeef: I have a little inconvinience when dial a mobile phone through BCM |
02:49.31 | scalex000 | superbeef: If I need to dial 11 digit, nortel not say anything only ring and ring |
02:49.35 | superbeef | DJCharlie: which line is supposed to listen for input for digits? |
02:50.04 | DJCharlie | superbeef: Doesn't Background() do that? |
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02:54.29 | superbeef | try changing exten => t,2,Goto(menu2,2,1) to exten => t,2,Goto(menu2,s,1) |
02:54.33 | superbeef | http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
02:55.28 | superbeef | scalex000: what's the call path like? |
02:55.49 | DJCharlie | superbeef: No joy, same error. |
02:56.36 | scalex000 | ok |
02:57.18 | superbeef | try changing exten => t,2,Goto(menu2,2,1) to exten => t,2,Goto(menu2,s,2) |
02:58.19 | DJCharlie | superbeef: But that will point it to Read(REPEAT) and not to the outgoing message for menu 2. |
02:58.26 | superbeef | scalex000: it's okay? lol... like what's the topology[expected] of a complete call |
02:59.38 | scalex000 | superbeef: what do you need exactly? I don't know what to paste or say. |
03:00.08 | superbeef | ex: sip phone -> Nortel -> SIP trunk -> asterisk -> T1 PSTN |
03:00.40 | superbeef | DJCharlie: are you sure? |
03:01.02 | scalex000 | superbeef: well I create a extension _8X9XXXXXXX, 1, Dial(SIP/81${EXTEN},45) |
03:01.12 | superbeef | DJCharlie: either your goto is amiss, or the say you're defining the extensiion in the macro is amiss |
03:01.37 | scalex000 | superbeef: well I create a extension _8X9XXXXXXX, 1, Dial(SIP/81${EXTEN}@outbound-asterisk,45) |
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03:02.00 | superbeef | scalex000: lets simplify and think about the flow of the call outside of asterisk syntax |
03:03.09 | scalex000 | superbeef: :( |
03:03.49 | superbeef | scalex000: lol common man, you can't articulate how you want it to work? |
03:04.03 | DJCharlie | superbeef: I'm certain. Just tried it. |
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03:04.18 | scalex000 | superbeef: No I not american people |
03:06.53 | scalex000 | superbeef: its working now |
03:07.14 | superbeef | scalex000: cool.. how did you fix it? |
03:08.13 | Sandheaver | found an IRC log of this channel for the past 8 years. reading it is like speed-lurking. |
03:08.17 | scalex000 | superbeef: I mean i can dialout through nortel without problem, but when i need know if nortel is saying a message asterisk not hear |
03:09.24 | superbeef | scalex000: you want nortel to call asterisk? |
03:09.58 | superbeef | Sandheaver: pipe it through grep and wc and see how who says the most profanities |
03:10.14 | scalex000 | superbeef: well I connect Asterisk to Nortel and is already don |
03:10.16 | scalex000 | done |
03:11.00 | superbeef | scalex000: see if you can say this a different way "if nortel is saying a message asterisk not hear" |
03:11.33 | scalex000 | superbeef: right, you are in the point |
03:12.03 | scalex000 | for example: the number you have dial is disconnect at this moment |
03:12.17 | scalex000 | superbeef: asterisk only ring and ring |
03:13.13 | superbeef | scalex000: what does /var/log/asterisk/full say when its ring ring? |
03:13.24 | superbeef | DJCharlie: don't stop believing |
03:13.30 | scalex000 | superbeef: I don't know |
03:13.38 | scalex000 | superbeef: ha ha ha |
03:14.04 | superbeef | scalex000: Are you allowed to see the logs? |
03:14.52 | scalex000 | superbeef: which one? |
03:14.59 | superbeef | scalex000: asterisk |
03:15.43 | superbeef | scalex000: you need to see the asterisk logs when you Nortel -> Asterisk |
03:16.47 | scalex000 | superbeef: I mean what filename you need to see, I have 4 |
03:18.01 | superbeef | scalex000: you need to watch /var/log/asterisk/full while you call asterisk with nortel.... (tail -f /var/log/asterisk/full) |
03:18.29 | superbeef | scalex000: we want to know what asterisk sees when nortel calls |
03:19.06 | scalex000 | superbeef: well I not have full file I have messages |
03:19.47 | superbeef | scalex000: pastebin.com |
03:20.26 | DJCharlie | superbeef: Any more ideas? |
03:22.05 | scalex000 | superbeef: too long |
03:22.30 | superbeef | scalex000: you need to find the part this is just frmo when you call |
03:22.33 | superbeef | scalex000: not the whole file |
03:22.53 | superbeef | scalex000: you need to "see" in the logs where nortel is ringing asterisk |
03:23.05 | superbeef | scalex000: those errors will help you figure out what is happening |
03:23.16 | scalex000 | ok |
03:26.05 | superbeef | DJCharlie: running out of steam sorry |
03:27.26 | superbeef | slumber |
03:27.57 | DJCharlie | Anyone else? |
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03:28.52 | scalex000 | superbeef: http://pastebin.ca/1579898 |
03:30.54 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
03:31.48 | superbeef | scalex000: is nortel a phone or a PBX? |
03:32.17 | scalex000 | superbeef: PBX |
03:34.14 | superbeef | scalex000: that's tough |
03:34.17 | superbeef | https://issues.asterisk.org/view.php?id=15102 |
03:34.27 | superbeef | scalex000: i have to sleep...good luck |
03:34.45 | scalex000 | superbeef: thank you for your time |
03:34.53 | scalex000 | superbeef: see you later |
04:16.38 | *** join/#asterisk raden (n=chatzill@66-168-4-200.dhcp.stpt.wi.charter.com) |
04:17.38 | raden | anyone have logs that can tell me who was helping me with my asa 5505 the other night i dont know if i was on as raden or raden_work |
04:20.18 | raden | hello |
04:24.09 | *** join/#asterisk Greek-Boy (n=greek@41.188.154.137) |
04:26.10 | raden | Sandheaver: ? |
04:26.21 | Sandheaver | yes? |
04:26.28 | Sandheaver | oh asa |
04:26.38 | Sandheaver | what's up |
04:26.44 | Sandheaver | try virtualization? |
04:27.48 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
04:35.12 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
04:39.05 | Sandheaver | sleepytime |
04:42.11 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
04:46.07 | psilikon | exit |
05:08.00 | raden | lol |
05:27.41 | *** join/#asterisk thansen (n=thansen@199.126.167.187) |
05:28.56 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
05:29.35 | verywiseman | can i build load-balance cluster for asterisk? |
05:34.35 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
05:38.28 | drmessano | um yeah |
05:38.46 | *** join/#asterisk Grof (n=dule@89.201.165.226) |
05:42.39 | Grof | need help |
05:42.45 | Grof | asterisk is dropping SIP call |
05:47.40 | TJNII | Grof: Post some details or you'll never get an answer. |
05:53.56 | *** join/#asterisk Zackery (n=me@74-131-188-77.dhcp.insightbb.com) |
05:55.46 | p3nguin | What exactly is SIP forwarding? Does it require a trunk from one ITSP to another or what? |
05:57.14 | nextime | 'morning |
06:00.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
06:05.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@81.218.155.148) |
06:08.46 | *** join/#asterisk imcdona (n=t@24.22.222.115) |
06:31.11 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
06:37.30 | Grof | [Sep 26 08:27:52] DEBUG[30200] chan_sip.c: Got unsupported a:fmtp:101 0-16 in SDP offer |
06:37.30 | Grof | [Sep 26 08:27:52] NOTICE[30200] chan_sip.c: No compatible codecs, not accepting this offer! |
06:37.32 | Grof | ? |
06:38.04 | Grof | what is a:fmtp:101 ? |
06:38.10 | Grof | and why does it bother asterisk? |
06:45.00 | kaldemar | Grof: http://www.rfc-editor.org/rfc/rfc2198.txt |
06:48.49 | coppice | Grof: 0-16 probably means 101 is RFC2833. if that's the case * shouldn't complain |
06:51.15 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
06:51.21 | Grof | tnx |
06:52.03 | *** join/#asterisk thansen (n=thansen@d199-126-167-187.abhsia.telus.net) |
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07:30.05 | sunzofman | calls are being rejected b/c extensions are not found. |
07:32.39 | sunzofman | extensions.conf -> http://pastebin.com/d1ee8951a |
07:34.36 | kaldemar | sunzofman: what is calling into asterisk? what context have you defined for the caller? what number is called? |
07:36.03 | kaldemar | these questions would be answered by a CLI output of a call and sip.conf. |
07:36.49 | sunzofman | <PROTECTED> |
07:38.40 | kaldemar | CLI output of the whole call, not a single line. it was clear that you don't have extension 3132630611 in your dialplan. and why are you calling yourself? |
07:40.10 | [TK]D-Fender | sunzofman: And you clearly have no match forr that # |
07:41.24 | sunzofman | kaldemar: just wanted to test my new install |
07:41.26 | kaldemar | deja vu, btw |
07:42.04 | sunzofman | sip.conf -> http://pastebin.com/d50f627c3 |
07:42.45 | sunzofman | kaldemar: the output shared from CLI was the only output which appeared after the call was made. |
07:42.59 | kaldemar | sunzofman: kind of a useless test, use app Echo for example. |
07:43.34 | kaldemar | sunzofman: "core set verbose 10" and try again. you'll get more output. |
07:44.23 | sunzofman | kaldemar: ok |
07:49.00 | kaldemar | hmm. i remember telling you to add a from-broadvoice context in your dialplan earlier this week. |
07:50.36 | sunzofman | kaldemar: must i replace the existing context=internal under [general] in my dialplan? |
07:51.17 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
07:51.27 | sunzofman | kaldemar: ok i see add the [from-broadvoice] context and then populate the necessary action under neath |
07:51.36 | kaldemar | sunzofman: no, replace the context under general with something that leads to a dead end. use valid context for defined peers only. |
07:52.10 | kaldemar | but yes, add [from-broadvoice] to extensions.conf |
07:53.34 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
07:59.17 | [TK]D-Fender | [03:40]<[TK]D-Fender>sunzofman: And you clearly have no match forr that # |
07:59.43 | regan40 | hi |
07:59.48 | kaldemar | and i clearly don't have a pastebin to look at. |
08:00.33 | sunzofman | [TK]D-Fender: I obsfucated the number in the earlier pastebin. |
08:00.53 | sunzofman | kaldemar:sip.conf -> http://pastebin.com/d50f627c3 |
08:01.09 | sunzofman | kaldemar: extensions.conf -> http://pastebin.com/d1ee8951a |
08:02.09 | kaldemar | sunzofman: you posted those already. i'm waiting for the CLI output of a call. |
08:02.19 | [TK]D-Fender | sunzofman: context=from-broadvoice <--- and where is this context in your DIALPLAN? |
08:03.44 | sunzofman | [TK]D-Fender: i just added it to extensions.conf |
08:04.27 | [TK]D-Fender | sunzofman: I don't see that |
08:08.11 | regan40 | I know it is probably a FAQ but what cards are recommended for asterisk,,,> |
08:08.14 | regan40 | ..? |
08:09.32 | [TK]D-Fender | regan40: Digium & Sangoma are the front runners |
08:09.32 | regan40 | looking for analog cards and australai PSTN compatible |
08:09.54 | regan40 | googles Digium & Sangoma |
08:10.28 | sunzofman | kaldemar: latest CLI output -> http://pastebin.com/d85edb9e |
08:11.15 | sunzofman | [TK]D-Fender: i didn't repost extentions.conf after the new context [from-broadvoice] was added. i'll do that now. |
08:11.41 | kaldemar | sunzofman: that's not the whole call |
08:12.10 | [TK]D-Fender | '3132630611' <- nothing to match this |
08:12.49 | sunzofman | kaldemar: verbosity is set to 10, that is the resulting output. |
08:13.53 | kaldemar | sunzofman: go add extensions to broadvoice-in |
08:16.15 | [TK]D-Fender | kaldemar: I'll leave this one to you.. best of luck with that... |
08:16.29 | sunzofman | kaldemar: so, [from-broadvoice] context should have 'exten=> 101,x,x' and exten=> 102,x,x' underneath.. |
08:17.32 | kaldemar | sunzofman: if you're going to dial 101 and 102 from something that uses the context, yes. |
08:18.02 | kaldemar | sunzofman: but those still won't match 3132630611. |
08:19.09 | sunzofman | kaldemar: gotcha.. regarding matching the bvoice #.. not sure how to do that. the number behaves as an authentication component. |
08:19.33 | sunzofman | kaldemar: bvoice requires this authID. |
08:20.14 | kaldemar | exten => 3132630611,1,... |
08:22.56 | sunzofman | kaldemar: 3132630611 is the # that allows the outside to reach my internal extentions.. |
08:24.10 | kaldemar | sunzofman: the call comes in by that number, and you need to send it somewhere. you do that with e.g. exten => 3132630611,1,Dial(SIP/101) |
08:25.01 | sunzofman | kaldemar: understood |
08:29.04 | sunzofman | kaldemar: updated extension.conf -> http://pastebin.com/d305907d0 |
08:33.25 | sunzofman | kaldemar: still stubbing my knuckles ;-) |
08:34.24 | sunzofman | kaldemar: i've not changed sip.conf |
08:36.31 | kaldemar | did you call in already? |
08:39.57 | sunzofman | kaldemar: yep.. i get a different CLI output and the generic bvoicemail is gone. i'm making progress the [from-broadvoice] context seems to have fixed it. |
08:40.12 | sunzofman | kaldemar: -- Executing [3132630611@from-broadvoice:1] Answer("SIP/3132630611-7c416000", "") in new stack |
08:40.13 | sunzofman | <PROTECTED> |
08:43.36 | *** join/#asterisk trentster (n=marks@d58-110-225-234.sun6.vic.optusnet.com.au) |
08:45.14 | trentster | Hey have setup asterisk on a intel nano platform, if at compile time I choose all the audio greetings in all the formats like g729 etc will asterisk automatically choose the corerct version and not transcode greetings etc/ |
08:45.18 | trentster | ? |
08:46.44 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
09:05.53 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
09:10.17 | *** join/#asterisk Milad (n=milad@unaffiliated/slackark) |
09:12.19 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
09:12.33 | Milad | hi, is any command to transfer a call ? I have extension number and channel name and wana transfer this call to another extension |
09:27.28 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
09:44.16 | Gugge | Milad transfer maybe? |
09:45.05 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
09:45.33 | Milad | Gugge, I check out transfer, but I should call it in dialplan, but I want put a button on 3party app and when click that transfer call, so I think I need another command to run az AMI |
09:47.52 | *** join/#asterisk znh (n=hans@unaffiliated/znh) |
09:47.58 | znh | Hello. |
09:48.32 | znh | I configured two phones, one hard the other soft. If I call the echo service I won't hear echo from the hard phone |
09:49.18 | znh | Any ideas? |
09:49.38 | znh | I can ring the hardphone from the softphone, but also no sound |
09:49.56 | znh | the hardphone is connected via a router. on the same lan |
09:50.02 | znh | both SIP |
09:50.12 | Gugge | Milad, i press transfer on my phone, and then the number to the phone i want to transfer to |
09:50.13 | Gugge | thats it |
09:53.58 | Milad | Gugge, yes I know that way , I want to create another way to transfer call for our CTI |
09:55.48 | Milad | anyway tnx for your clues |
09:57.21 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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10:49.50 | dustybin | something strange is going on, if i ring my polycom using my softphone, my polycom makes the ringing sound, if i dial in from outside, my polycom is silent, only a green light flashes to indicate the call |
10:50.25 | dustybin | i think i know what it is |
10:56.08 | dustybin | yep, i knew it, i have a virus |
11:12.12 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
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11:15.38 | *** mode/#asterisk [+o denon] by ChanServ |
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11:28.45 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
11:28.52 | DonAlex | Hey guys.... |
11:29.55 | DonAlex | can someone help me debug why the gui is playing up. I have tried running asterisk -d -vvvvv etc. but I cannot see any debug information at all.. yet I know it is listening to port 8088 because lsof -i tcp:8088 shows it is.. |
11:30.12 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
11:30.47 | DonAlex | The symptoms are that no matter what url I put in is cannot seem to find any files. |
11:30.54 | Milad | DonAlex, can you run asterisk -r ? |
11:31.00 | DonAlex | yes |
11:31.08 | Milad | and you have a console ? |
11:31.12 | DonAlex | I am running it in debug mode atm |
11:31.39 | Milad | ow -d option :) yes |
11:31.52 | DonAlex | yes I have a console. |
11:32.03 | DonAlex | core ser verbose 5 shows nothing either.. |
11:32.14 | DonAlex | I justr cannot see any http request at all.. |
11:32.33 | DonAlex | even though I know it is accepting them because of the error pages.. |
11:33.07 | Milad | DonAlex, do you test core set debug 10 ? |
11:33.23 | DonAlex | can do.. will that show http stuff? |
11:33.54 | Milad | ow sorry you want to use GUI |
11:34.43 | DonAlex | yes.. the server is running but I cannot figure out what is going on.. the files all seem inthe right place.. right permissions but no url can access any files.. so I want to debug what is actually going on |
11:35.03 | DonAlex | am I really going to have to use strace? ;) |
11:35.08 | DonAlex | bloody hope not :P |
11:35.12 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
11:35.27 | Milad | ye ! always my problem solve with strace ;) |
11:35.54 | DonAlex | Yeah but on Asterisk ? multithreaded as it is? *whimpers* |
11:39.09 | DonAlex | Is there really no way to debug the http server from within asterisk? |
11:39.16 | DonAlex | surely that cannot be right? |
11:40.55 | DonAlex | hmmmmm |
11:41.01 | DonAlex | something does not look quite right here.. |
11:41.24 | DonAlex | http://pastebin.com/m6916c3cc |
11:41.54 | superbeef | what doesn tlook right in that |
11:42.05 | DonAlex | now I know there is no phoneprov configures.. so that is why there is .. before the path.. but why does static have that too? |
11:42.26 | DonAlex | also the default dir is static-http |
11:42.33 | DonAlex | could it just be misnamed? |
11:44.07 | DonAlex | hmm the manager url works.. |
11:44.54 | DonAlex | as does the mxml one.. |
11:44.56 | DonAlex | hmmm |
11:45.06 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
11:47.13 | *** join/#asterisk scalex000 (n=chatzill@50puntacana02.codetel.net.do) |
11:48.44 | DonAlex | Ok so where the hell are those default url defined?! |
11:48.49 | DonAlex | in the source?! |
11:55.15 | zamba | i have a cisco ata-188.. when dialing internal extensions (only four digits), it takes several seconds before the numbers are sent to the * and actually dialed |
11:55.30 | zamba | does anyone know where i can tweak this in the cisco adapter? |
11:55.31 | DonAlex | DOH!!!! |
11:55.32 | DonAlex | ok ok.. |
11:55.34 | DonAlex | solved it.. |
11:55.45 | DonAlex | Good old Debian shifting things around again.. ;) |
11:56.13 | DonAlex | by default asterisk-gui puts static-http into /var/lib/asterisk.. |
11:56.47 | DonAlex | however the ast_config_AST_DATA_DIR is /usr/share/asterisk in Debian. |
11:57.00 | DonAlex | made a symlink and the files are found.. Cool. |
11:57.01 | DonAlex | ;) |
12:09.00 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
12:18.49 | znh | I configured two phones, one hard the other soft. If I call the echo service I won't hear echo from the hard phone |
12:19.00 | znh | Any suggestions? |
12:21.53 | zamba | i have a problem with my adapter.. i can only call devices attached to it for the first few minutes after registering.. then i'm unable to dial them.. my adapter is a ata-188 |
12:22.24 | zamba | the adapter is behind NAT |
12:22.36 | zamba | it works just fine with my softphones |
12:27.03 | Gugge | set the adapter to register every minute or something like that |
12:38.30 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
12:40.51 | zamba | Gugge: yeah, i did.. that worked |
12:40.58 | zamba | Gugge: but what causes it? |
12:41.23 | zamba | Gugge: and why don't i get the same behavior with my softphones? |
12:49.37 | *** join/#asterisk Adr3nalin3 (n=afink@204.26.87.226) |
12:49.50 | Gugge | you nat device closes the forwarded udp port after xxx seconds idle |
12:50.01 | Gugge | maybe your softphone sends something every xxx seconds |
12:50.06 | Gugge | and your adapter dont |
12:50.16 | Gugge | a register renews the udp forward |
12:50.22 | Gugge | (any traffic does) |
12:50.34 | Gugge | so another sollutions is to make a new call every minute :) |
12:52.50 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
12:59.02 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
12:59.28 | sircolin | I likes jux |
12:59.42 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
13:12.02 | zamba | Gugge: hehe |
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13:24.44 | *** join/#asterisk znh (n=hans@unaffiliated/znh) |
13:24.48 | znh | Hello. |
13:24.57 | znh | I configured two phones, one hard the other soft. If I call the echo service I won't hear echo from the hard phone. |
13:25.41 | znh | sip show peers shows the phones are connected. The hardphone has my external IP (phone is connected to internet router). Others have an internal ip address. |
13:25.52 | znh | Has that anything to do with it? |
13:26.11 | znh | Gugge: any idea? |
13:27.15 | znh | It's not a codec issue for sure. I forced a codec which is supported |
13:32.24 | znh | "exited non-zero on" is the error message if I hang up |
13:40.58 | Gugge | fix it so you hard phone has an internal ip too |
13:41.24 | Gugge | i guess you asterisk cant send the sound back to the phone when its shown as the external ip |
13:41.35 | Gugge | but without total knowledge about you network setup, its only a guess |
13:42.05 | znh | Yes I figured that asterisks connect to the external IP |
13:42.13 | znh | Is it possible to force asterisk to connect to it's local IP? |
13:42.44 | Gugge | give the phone an internal ip, and set it to connect to the asterisk internal ip |
13:42.52 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
13:43.13 | znh | It has an internal IP. Asterisk is router's NAT |
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13:48.43 | *** join/#asterisk DJCharlie (n=DJCharli@h193.196.30.71.dynamic.ip.windstream.net) |
13:48.57 | DJCharlie | Morning all. Anyone awake? |
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13:55.03 | trentster | Hey have setup asterisk on a intel nano platform, if at compile time I choose all the audio greetings in all the formats like g729 etc will asterisk automatically choose the correct version and not transcode greetings etc/ |
14:01.18 | DJCharlie | I'm wondering why it won't execute a script using the System() function.... |
14:04.23 | znh | Gugge: ? |
14:18.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
14:18.30 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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14:35.46 | DJCharlie | Ok folks. Why isn't this working? |
14:35.57 | DJCharlie | exten => 301,5,System(/mnt/vm/incoming/xt/vmsend.sh "record301" "user@mydomain.com") |
14:36.20 | DJCharlie | The script runs perfectly if I execute it by hand. |
14:44.06 | Gugge | my guess it a path problem with the script |
14:44.40 | Gugge | try calling the script with an empty PATH variable, and see if it works |
14:46.02 | DJCharlie | Nope, the paths are absolute in the script, and it's running as the same user as asterisk. |
14:47.20 | Gugge | set verbose to something high, and check what happens on the asterisk console |
14:48.19 | DJCharlie | Good idea. Setting it to 10. |
14:49.10 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
14:50.03 | DJCharlie | Nothing. |
14:50.16 | DJCharlie | Plays the goodbye message, and hangs up. |
14:52.04 | DJCharlie | -- Executing [301@xt301:5] System("Zap/1-1", "/mnt/vm/incoming/xt/vmsend.sh") in new stack |
14:54.24 | kaldemar | CLI won't give any output when you run something with System |
14:54.45 | kaldemar | how is it not working? is the script not receiving the arguments? |
14:55.56 | DJCharlie | kaldemar: No, I've even gone so far as to try hard coding the variables into the script for testing. It never actually executes. |
14:57.05 | DJCharlie | Running the script by hand, it works perfectly. |
14:58.33 | kaldemar | does the script have execute permissions? |
14:58.55 | DJCharlie | Yes. |
14:59.17 | kaldemar | you should be seeing the arguments in the "Executing..." line. |
14:59.43 | DJCharlie | For testing, I have temporarily hard-coded the args. |
15:00.09 | kaldemar | is the script calling a binary? |
15:00.40 | DJCharlie | yes. |
15:00.42 | kaldemar | have you tried to echo $1 and $2 to a file to verify that it actually executes? |
15:00.53 | kaldemar | is the script calling the binary with full path? |
15:01.01 | DJCharlie | Yes. And it doesn't output anything. |
15:02.13 | kaldemar | this doesn't do a thing: http://pastebin.ca/1580300 ? |
15:02.37 | DJCharlie | That's similar to what I have in my script, and the answer is no. |
15:03.53 | kaldemar | well, there's obviously something wrong with it. if you're willing to pastebin "ls -la /mnt/vm/incoming/xt/vmsend.sh" and the script contents, i can take a look if i spot something. |
15:06.28 | DJCharlie | http://pastebin.com/d25a8405b |
15:06.55 | DJCharlie | That's with the script hard-coded |
15:08.17 | kaldemar | are you running asterisk as root? |
15:08.24 | DJCharlie | Yes. |
15:09.31 | raden | Sandheaver: morning |
15:09.38 | Sandheaver | morning |
15:09.40 | *** join/#asterisk misteranonymous (n=tyler@udp115943uds.hawaiiantel.net) |
15:09.42 | raden | guess what |
15:10.15 | raden | I picked up a procurve 1800-24G for my desk for $125 on ebay last night |
15:10.30 | raden | the guy has 2 left if you need any |
15:10.31 | Sandheaver | nice |
15:10.41 | Sandheaver | link? |
15:10.55 | Sandheaver | that a PoE switch? |
15:11.17 | Sandheaver | did you try virtualization? |
15:12.46 | raden | Sandheaver: no didnt try yet gf wants to go out of town for weekend :( |
15:12.59 | Sandheaver | fair enough |
15:13.05 | Sandheaver | GF > computers |
15:13.12 | raden | i want to thought |
15:13.24 | Sandheaver | unless it's just a getting to know you better weekend, those usually suck |
15:13.49 | DJCharlie | Enjoy those while you can. After the wedding, everything changes. |
15:13.49 | kaldemar | DJCharlie: i can't see a why the script wouldn't execute, if all you've said is correct. |
15:13.55 | Sandheaver | it doesn't take long, but if i were you i would be thinking about the GF |
15:14.13 | raden | GF = PITA |
15:14.19 | DJCharlie | kaldemar: I can't either. I'm not exactly new to scripting. |
15:14.40 | raden | DJCharlie: i dont plan on getting married we have a kid together thats enough |
15:15.22 | raden | Sandheaver: plus all monitors at work ATM trying to get everything done , you get the linky i offered 125 |
15:15.47 | Sandheaver | yup got link |
15:17.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
15:17.08 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
15:17.09 | raden | cheap for a L2, 24 port, gigabyte w/ lifetime warranty |
15:25.15 | DJCharlie | Okay, NOW the echo statements in the script show the script is running. |
15:25.26 | DJCharlie | BUT, nothing is actually happening. |
15:26.00 | *** join/#asterisk TimToady_ (n=moi@adsl116-19.kln.forthnet.gr) |
15:31.29 | misteranonymous | TimToady_: there more than one way to do it |
15:34.58 | znh | I have a phone connected to a router. The Asterisk server is on the LAN behind the router. If I try to dial my echo service, I can't hear anything. I did rtp debug and it showed the external IP. Is this what it should be? |
15:40.26 | sircolin | my asterisk box is not local to me, when I do the same rtp debug it shows my external ip too so im guessing that is correct |
15:50.05 | p3nguin | znh: Maybe you need to check out the NAT workaround. |
15:50.24 | p3nguin | But then again, maybe not. |
15:54.40 | *** join/#asterisk garymc (n=garymc@host86-163-43-91.range86-163.btcentralplus.com) |
16:15.27 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
16:24.10 | *** join/#asterisk Tim_Toady (n=moi@adsl30-17.kln.forthnet.gr) |
16:24.50 | *** join/#asterisk scalex000 (n=chatzill@250puntacana02.codetel.net.do) |
16:24.57 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
16:27.17 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
16:45.33 | *** join/#asterisk DJCharlie (n=DJCharli@h193.196.30.71.dynamic.ip.windstream.net) |
16:45.57 | DJCharlie | Ok folks, I'm back. System is working great, but I'm thinking towards future expansion... |
16:46.28 | DJCharlie | Now, correct me if I'm wrong, but an FXS is what allows you to plug in a standard analog telephone, right? |
16:46.47 | kaldemar | right |
16:47.31 | DJCharlie | Okay. Say I need 3 phones, all 3 set to ring when a certain extension is dialed. I'd need a 3 port FXS? |
16:48.31 | kaldemar | or 3 FXS ports, yes. however, if you're thinking about getting phones, i'd go for VoIP ones. |
16:49.34 | DJCharlie | And those are rather expensive... I'm working on an extremely shoestring budget here. |
16:49.48 | *** join/#asterisk nightrid3r (n=kvirc@78.20.232.172) |
16:50.25 | DJCharlie | Or would 3 VoIP phones be cheaper? |
16:50.39 | kaldemar | proper fxs cards aren't cheap either. |
16:50.59 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
16:51.08 | DJCharlie | Well, my FXO card is a X1000P, but it works fine. :) |
16:52.03 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
16:52.41 | hardwire | moo |
16:53.00 | DJCharlie | So which would be cheaper? |
16:53.01 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
16:53.30 | kaldemar | http://store.digium.com/productview.php?product_code=1TDM440EF <-- for that price, you can get 4-5 decent VoIP phones and you'll have a different grade of reliability and scalability. |
16:53.32 | [TK]D-Fender | DJCharlie: Do those phones have to act independantly and only ring together on this one occasion? |
16:53.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
16:54.22 | DJCharlie | While that'd be nice, not really. It's perfectly fine if they all work together 24/7. |
16:54.34 | *** join/#asterisk imcdona (n=t@c-24-22-222-115.hsd1.wa.comcast.net) |
16:54.42 | [TK]D-Fender | DJCharlie: I mean you could plug 3 phones onto 1 ATA's jack |
16:54.48 | DJCharlie | They'll be incoming calls only, and rarely used at that. |
16:55.24 | [TK]D-Fender | DJCharlie: Otherwise just get yourself 2 x Linksys PAP2T-NA's for < $100 total |
16:55.42 | DJCharlie | Ok, [TK]D-Fender, you've lost me. Is an ATA the same as the FXS? |
16:55.57 | [TK]D-Fender | ~ATA |
16:55.58 | infobot | from memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
16:56.41 | DJCharlie | Okay, so I get one of those, put a 3 way splitter on it, and plug in the 3 phones, right? |
16:57.22 | nightrid3r | DJCharlie: doesn't that defeat the whole idea of using a pbx |
16:57.47 | DJCharlie | nightrid3r: We're mainly using it for voicemail to be emailed to our DJs. |
16:57.57 | nightrid3r | oh k |
16:58.10 | DJCharlie | The only calls that would ring those phones is station emergencies. |
16:58.55 | [TK]D-Fender | DJCharlie: You could get away with just 1 then |
16:59.22 | DJCharlie | So what's a good low-cost brand/model for that? |
16:59.23 | [TK]D-Fender | DJCharlie: they'd be like having 3 phones in your house ont he same line. So anyone picking up the phone can listen in on the others |
16:59.33 | [TK]D-Fender | DJCharlie: I just gave you a specific model |
16:59.36 | DJCharlie | That's not an issue. :) |
16:59.45 | [TK]D-Fender | DJCharlie: http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-PAP2T-NA |
17:02.02 | DJCharlie | That's a nice price. :) Just have to convince the Bosslady. |
17:05.00 | DJCharlie | Ok, I'm off to a staff meeting to squeeze more money from the turnip. |
17:05.05 | DJCharlie | Thanks folks! |
17:11.14 | *** join/#asterisk luca`gervasi (n=ashura@host147-165-dynamic.55-79-r.retail.telecomitalia.it) |
17:11.17 | luca`gervasi | Hallo |
17:14.57 | Dovid | hello |
17:19.19 | fofware | hello guys |
17:19.53 | Dovid | hallo |
17:22.04 | Sandheaver | hillo |
17:22.20 | Sandheaver | hollo |
17:22.22 | Sandheaver | hullo |
17:22.28 | Sandheaver | hyllo |
17:22.34 | Sandheaver | all vowels covered |
17:24.23 | Sandheaver | someone should make some sort of adapter that you can plug your cell phone into when you get home, so that when your cell phone rings all the phones in the house ring, and you can just pick up any phone and talk via your cell phone |
17:24.28 | Sandheaver | though i suppose call forwarding could do this |
17:24.57 | Sandheaver | but i'm talking about a completely standalone phone system at home, that is useless for anything other than intercom when a cell phone isn't providing service to the system |
17:25.18 | Sandheaver | guess it doesn't make a lot of sense if you use your phone for anything other than talking though |
17:29.05 | fofware | Sandheaver: take a look http://www.google.com/googlevoice/about.html |
17:29.21 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
17:29.48 | Sandheaver | yeah i have a google voice account, but i want to make calls out of the home phone system, via the cell phone as well |
17:30.15 | Sandheaver | i seem to recall a product like this for nokia phones, using nokia's popport |
17:32.23 | *** part/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
17:33.40 | [TK]D-Fender | Sandheaver: This device already exists |
17:33.48 | Sandheaver | gasps |
17:33.52 | [TK]D-Fender | Sandheaver: lets you use your cell as an FXO |
17:33.57 | Sandheaver | really |
17:34.00 | Sandheaver | nice |
17:34.14 | Sandheaver | what's it called |
17:34.21 | [TK]D-Fender | http://www.phonelabs.com/prd05.asp |
17:34.57 | Sandheaver | oh bluetooth |
17:35.04 | Sandheaver | why the F didn't i think of that |
17:35.10 | Sandheaver | awesome |
17:35.11 | drmessano | loves chan_mobile in Asterisk |
17:35.21 | Sandheaver | ooOOoohh |
17:35.22 | drmessano | OOPS, sorry, that was on topic |
17:35.26 | Sandheaver | hah |
17:36.14 | drmessano | chan_mobile works great and costs me existing_system+$3 dongle |
17:37.08 | fofware | [TK]D-Fender: Hello, I did changed mailcmd in voicemail.conf for my own script, and now I need monitoring and debug it, can you tellme how I can do that? |
17:40.09 | [TK]D-Fender | fofware: What is there to monitor? Is your script not getting called? |
17:42.18 | fofware | [TK]D-Fender: look like it's not called, for that I want to get some error or something to find the issue |
17:42.18 | p3nguin | When using an IAX trunk, calls do not go through. I see the following message on the console: chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response. Where do I need to start in order to solve this? |
17:42.47 | fofware | [TK]D-Fender: when i put mailcmd=/usr/share/asterisk/asteriskmail |
17:42.58 | fofware | [TK]D-Fender: nothing happend |
17:44.15 | fofware | [TK]D-Fender: the script supose write in file the parameters that receive, so after I can write my own mail-script like you told me |
17:47.15 | [TK]D-Fender | fofware: And where do I see you leaving a VM, and your config showing me that your script should get called? or seeing the permissions ont he script itself? |
17:52.29 | haryv | is there a test script that can run though my asterisk system to locate problems with it? Seems i get random complaints such as, pressing one extention and the user experiances dead air. I can press it though the ivr and it will work fine. |
17:52.37 | *** join/#asterisk scalex000 (n=chatzill@206puntacana02.codetel.net.do) |
17:53.07 | fofware | [TK]D-Fender: http://pastebin.ca/1580465 |
17:58.00 | fofware | [TK]D-Fender: script is working I test it from command line and work, so have permissions and everything that need |
17:59.32 | [TK]D-Fender | fofware: you have not shown me HALF of what I ask for and I will not take your word for it that the users match |
18:03.20 | fofware | [TK]D-Fender: sorry I don't understand, I can paste voicemail.conf if you want and anything that you need, only let me know |
18:04.27 | [TK]D-Fender | fofware: I want to see the LS of the script. voicemail.conf CLI output of you LEAVING a VM |
18:06.24 | fofware | [TK]D-Fender: Ok, what you mena with LS of script? |
18:06.55 | [TK]D-Fender | fofware: "ls -la /usr/share/asterisk/" |
18:07.08 | fofware | ok |
18:08.43 | fofware | [TK]D-Fender: voicemail.conf http://pastebin.ca/1580481 |
18:10.30 | fofware | [TK]D-Fender: ls http://pastebin.ca/1580484 |
18:12.34 | fofware | [TK]D-Fender: CLI output http://pastebin.ca/1580486 |
18:15.23 | fofware | [TK]D-Fender: in system.log after leave VM each time i see |
18:15.23 | fofware | Sep 26 15:14:02 shila asterisk[9718]: rc_avpair_new: unknown attribute 1490026597 |
18:15.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:15.49 | fofware | use or not the script |
18:16.22 | [TK]D-Fender | fofware: did you restart * completely after changing voicemail.conf? |
18:17.05 | fofware | [TK]D-Fender: yes, I did try both, reload from CLI and stop and start from command line |
18:17.17 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
18:17.45 | *** join/#asterisk linuxviewer (n=example@ip68-110-115-3.ph.ph.cox.net) |
18:17.54 | fofware | [TK]D-Fender: Asterisk is 1.4 over debian, only to let you know |
18:18.48 | fofware | [TK]D-Fender: SMTP postfix, pop3 and IMAP Dovecot |
18:19.03 | [TK]D-Fender | fofware: go prove the user * is running as |
18:20.16 | fofware | [TK]D-Fender: ?, in other world, please... |
18:21.24 | linuxviewer | My asterisk server is no longer sending voicemail notifications via email. Is there a place that has the logs for email to diagnose why it is no longer sending email notifications? |
18:21.36 | luca`gervasi | does anybody use Cisco 7940 phones with asterisk? |
18:22.28 | linuxviewer | luca'gervasi - Yes |
18:22.49 | [TK]D-Fender | LinuCheck your mailer |
18:23.01 | [TK]D-Fender | linuxviewer: Check your mailer |
18:23.24 | fofware | linuxviewer: your mail server can send mail to same mail-box that you have in asterisk? |
18:24.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
18:24.41 | fofware | linuxviewer: remember from some months ago all mail providers reject mails from unknow mailservers |
18:25.00 | linuxviewer | fofware - I do not understand your question. It has been working, and then out of no where, I no longer receive the emails with the voicemail.s Can I just try by typing "mail" in CLI and put in information? |
18:25.32 | linuxviewer | fofware - I am aware that this may be the case, so I am trying to figure out if the machine is actually sending the email, or if it is running into an error of some sort (of course, if it shows it is sending the mail, but no mail is received, then we know it is a provider problem) |
18:26.07 | linuxviewer | I assume by default, asterisk uses mail not sendmail? |
18:26.11 | fofware | linuxviewer: I'm verry new in asterisk, but that I did is, check your mail server |
18:26.38 | linuxviewer | fofware - you say to check mail server, but I am unsure what you mean by that. How do I go about checking mail server? |
18:26.52 | fofware | linuxviewer: what mail server that you have installed |
18:26.59 | linuxviewer | Uses sendmail if not mistaken, |
18:27.02 | [TK]D-Fender | linuxviewer: The sample configs all call sendmail. If your install is using that script for something else, well you should already know where to look. otherwise its sendmail |
18:27.12 | linuxviewer | ./var/logs/maillog/ shows that it is sending the email, but it is not received. |
18:27.32 | linuxviewer | So possibly ISP blocking the outbound email? |
18:27.51 | p3nguin | Use the mail relay they provide for you. |
18:27.52 | fofware | linuxviewer: Yes, that is the cause |
18:28.08 | p3nguin | often relay.yourisp.net or similar |
18:28.35 | linuxviewer | p3nguin - I see. I put that relay information in sendmail configuration file somewhere? |
18:28.41 | p3nguin | yeah |
18:28.51 | fofware | linuxviewer: yeah |
18:29.21 | linuxviewer | Thank you for your assistance everyone. |
18:29.36 | linuxviewer | I will contact ISP, get their relay information, put it in /etc/mail/sendmail or google where it needs to go. |
18:29.49 | fofware | no problem linuxviewer |
18:30.15 | [TK]D-Fender | Linutons of ISPs prevent you from sending mail. You may have to configure sendmail to use their outbound relay with auth, etc. |
18:30.26 | [TK]D-Fender | linuxviewer: Rather common |
18:31.07 | fofware | linuxviewer: lastnight I see in internet some configuration of Asterisk voicemail to use one account of gmail, maybe you can find it |
18:31.46 | fofware | linuxviewer: I did lost the link but is there in some place :o) |
18:35.19 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
18:35.47 | e4 | Are parentheses allowed in the username field in iax.conf? |
18:36.03 | [TK]D-Fender | e4: Don't |
18:36.35 | e4 | [TK]D-Fender: It's the setting that Teliax gave us :/ |
18:38.34 | e4 | It seems at the very best like a bad idea to set that and it's not connecting properly, but I wondered if it's an invalid setting entirely. |
18:39.55 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
18:39.59 | e4 | I'd be open to suggestions for a better host if that's the issue. |
18:41.10 | [TK]D-Fender | e4: You haven't shown us the problem. |
18:41.22 | [TK]D-Fender | e4: you took 1 blind stab and have no backup in hand |
18:43.19 | *** join/#asterisk denon (i=denon@sassinak.net) |
18:43.19 | *** mode/#asterisk [+o denon] by ChanServ |
18:45.03 | fofware | [TK]D-Fender: so what you mean with "go prove the user * is running as" |
18:47.55 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
18:48.49 | *** join/#asterisk e4 (n=e4@76.79.48.214) |
18:49.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
18:50.08 | e4 | [TK}D-Fender: iax2 show registry shows it registered to the provider, but when I try to call into the service I get 'chan_iax2.c:10507 socket_process: ... Registration refused' from the provider. |
18:52.04 | fofware | [TK]D-Fender: I did find the issue, for some reazon, when i call lua script from asterisk do not execute it, maybe don't read the interpreter |
18:52.32 | fofware | [TK]D-Fender: I did write other in Shell and it Work |
18:52.53 | fofware | [TK]D-Fender: thanks |
18:55.31 | fofware | [TK]D-Fender: Asterisk export to shell VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE ? |
18:55.55 | box2 | what the balls |
18:56.18 | box2 | dpkg-reconfigure tzdata sets the new timezone in /etc/timezone and /etc/localtime |
18:56.30 | box2 | but when i echo $TZ it still says my old timezone |
18:56.38 | box2 | and when i use date it also still shows my old timezone |
18:57.13 | box2 | i hate everything |
18:57.54 | [TK]D-Fender | box2: .... wrong channel |
18:58.08 | [TK]D-Fender | box2: Unless you want to abstractly state that you hate * as well |
18:58.17 | [TK]D-Fender | box2: at which point the door is on your left :) |
18:59.55 | p3nguin | When using an IAX trunk, calls do not go through. I see the following message on the console: chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response. Where do I need to start in order to solve this? Debug output from the time I dial the number until I end the call: http://pastebin.ca/1580538 |
19:18.47 | box2 | [TK]D-Fender: heh |
19:46.32 | *** join/#asterisk scalex000 (n=chatzill@96puntacana02.codetel.net.do) |
19:55.37 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
20:01.56 | *** join/#asterisk cilkay (n=cilkay@CPE00d0b743a22f-CM0011ae01fcbe.cpe.net.cable.rogers.com) |
20:04.38 | cilkay | Hello. I'm new to all this VOIP stuff but not new to telephony. In the '90s, I did quite a bit of development using Dialogic analog and digital phone cards using their C toolkit. I'm going to order some DIDs. Is the caller ID that is displayed on outgoing calls a feature of the DID provider or something I configure at my end from Asterisk? |
20:05.08 | cilkay | If it's at my end, can that ID be changed on a per call basis? |
20:05.32 | [TK]D-Fender | cilkay: * can tell the provider what to show, but they have to allow it |
20:05.43 | [TK]D-Fender | cilkay: Just as many allow it as not |
20:06.03 | [TK]D-Fender | cilkay: You'll have to ask. Tryincally those who offer more bulk-like termination support this |
20:06.14 | [TK]D-Fender | silespecially the per-minute providers |
20:06.42 | cilkay | What would be an example of one or a couple that do? |
20:07.08 | luca`gervasi | i need a flat to call italian numbers and a local (+39) phone number...do you know any? |
20:07.52 | p3nguin | cilkay: VoIP.ms and Vitelity both allow passing of your own CID. |
20:08.27 | cilkay | p3nguin: I was on the voip.ms page, ironically. |
20:08.40 | p3nguin | CallCentric does not. They require you to verify you have control over the number and then you can select it from the web interface. |
20:09.22 | p3nguin | cilkay: I use VoIP.ms for my home and office termination since my call volume is low. At 1 cent per minute, it's hard to beat. |
20:10.23 | cilkay | My use case: I'm associated with multiple companies. When I return or make calls from numbers associated with those companies, I want the names and phone numbers of those companies to show. |
20:10.42 | cilkay | 1 cent per minute is close enough to free that it doesn't make much difference at low volumes. |
20:10.53 | p3nguin | You'll be able to set the CID number, but the name is done in another way. |
20:10.55 | cilkay | And probably not at high volumes either. |
20:11.13 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
20:12.05 | cilkay | p3nguin: How is the name set then? |
20:12.19 | p3nguin | CNAM lookups are done on the receiving end. |
20:13.01 | cilkay | And the name is associated with the number by the ITSP? |
20:13.25 | p3nguin | When you call Customer1, you only send the CID number. Customer1's telco does a lookup of the number in their LIDB. If there is a name associated with it, the name will display on the caller ID unit. |
20:13.26 | cilkay | If so, that would be fine, as long as I can set that name once and forget about it. |
20:14.02 | p3nguin | For example, I might call you, and your CID box will say Illinois Call rather than my name. |
20:14.10 | cilkay | Where does the name get associated with the number? |
20:14.22 | p3nguin | in the telco's database(s) |
20:15.03 | p3nguin | Think of it like a reverse DNS lookup. |
20:15.15 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
20:15.19 | cilkay | So, if your company is Penguin Consulting, is there a way for you to specify to your ITSP that you want that name displayed when my telco does a CNAM lookup when you call me? |
20:15.47 | cilkay | I understand the analogy. I'm still not clear who does that association. |
20:16.09 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
20:16.18 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
20:16.18 | p3nguin | Sometimes the ITSP can submit your CNAM (Caller id NAMe) to a large database, but there is nothing to guarantee that the called phone company uses that db. |
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20:17.17 | cilkay | That doesn't seem very robust. |
20:17.53 | p3nguin | On the other hand, if you are using a phone number of a company that has a regular land line number already, they most likely have a CNAM already in place. If you dial out using that phone number, the name lookup will be successful and will show the name of the company. |
20:18.48 | p3nguin | Maybe I can give you a working example. |
20:19.18 | p3nguin | If you receive a call from your local city hall, your caller ID will display something like CITY HALL plus their number, right? |
20:19.32 | cilkay | correct |
20:19.47 | p3nguin | Their number has the name already associated with it, so any lookup will be successful. |
20:20.11 | p3nguin | If I set my CID number to that of your local city hall and then I call you, you will see CITY HALL and their numbre on your caller ID. |
20:20.29 | p3nguin | Make sense? |
20:21.08 | cilkay | Sure, but what if you set your CID number to the actual VOIP number you got from voip.ms? |
20:21.18 | p3nguin | I send their number. Your telco does a name lookup based on that number, it succeeds. |
20:21.24 | cilkay | Say it was a previously unallocated number. |
20:21.52 | p3nguin | If the number does not have any name in the telco's database, it will show something like OUT OF AREA plus your number. |
20:22.09 | p3nguin | Might say ILLINOIS CALL like when I call you from my IL number. |
20:22.25 | p3nguin | You've probably seen out-of-state calls on your ID before. |
20:22.42 | cilkay | ... which to many people signals "telemarketer". |
20:22.54 | p3nguin | Unfortunately, yes. |
20:23.58 | p3nguin | You can always ask your ITSP to enter your name in the database, and then be very hopeful that it will eventually propagate to the db that your callee's telco uses. |
20:24.16 | cilkay | Maybe the "trick" it to associate with a POTS number that is "aged", i.e. has a name associated with it. |
20:24.58 | p3nguin | If they have a name in their db, that's what will show up when you use it to call someone. |
20:25.07 | p3nguin | even if it's not your name. :) |
20:25.19 | cilkay | You'd have to port that number to your ITSP eventually. It's a more expensive way to go but it's probably more reliable. |
20:26.16 | p3nguin | Even porting some existing number to a VoIP service isn't a guarantee that the name will persist in the previous telco's database. |
20:26.37 | p3nguin | They could drop it once they release the number. |
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20:26.47 | cilkay | true |
20:26.55 | p3nguin | I don't know if it's normal for them to drop them or not, though. |
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20:27.04 | cilkay | So I'll just have to impersonate "IBM" then. :) |
20:27.14 | p3nguin | That's the easy way. |
20:27.31 | cilkay | "Hello. I'm calling from the White House." |
20:27.38 | p3nguin | yeah :) |
20:28.05 | fofware | I'm in Pink House, hello cilkay |
20:28.07 | p3nguin | The problem is that they won't have a callback number on their cid for you, though. |
20:28.54 | cilkay | fofware: Hello. I'm in a light tan brick house but I have white trim :) |
20:29.18 | _trine | I'm in the dog house according to my wife |
20:30.57 | drmessano | _trine: Lucky. My wife added a new level called the "outhouse" and it makes the doghouse look like a stay at the Ritz |
20:31.11 | _trine | he he he |
20:31.36 | fofware | drmessano: lol |
20:32.29 | cilkay | p3nguin: CNAM Caller name Lookup - 1.25¢ per query <== from https://www.voip.ms/specifications.php I'm not sure what that means. |
20:32.49 | _trine | well I got my asterisk sending me my voicemail nicely |
20:33.01 | drmessano | I knew I was in the outhouse when I woke up on wednesday and had a crescent moon drawn on my chest with a sharpie |
20:33.08 | drmessano | :( |
20:34.07 | _trine | asterisk works very well off my linksys router |
20:35.17 | _trine | anyone that runs asterisk off a computer is a big girls blouse :P |
20:35.43 | drmessano | Anyone who runs Asterisk off a router doesn't have many friends |
20:36.05 | _trine | these north west of Englands sayings always make me laugh |
20:36.46 | _trine | drmessano: theres not many of us yet |
20:37.20 | drmessano | _trine: Yet? What you're doing is not new.. Asterisk has been run on Linksys boxes for at least 4 years now |
20:37.36 | drmessano | Its just not a great idea in general |
20:37.41 | _trine | yes but not with everything included |
20:37.44 | drmessano | Little horsepower, limited feature set |
20:37.58 | _trine | I have everything loaded |
20:38.20 | drmessano | You have g729, chan_skype, chan_mobile with a bluetooth dongle, and you're using ODBC logging for CDR? |
20:38.22 | _trine | and I have only used 3% of my flash space |
20:39.05 | _trine | I still have 97% of my flash space unused |
20:39.14 | _trine | on my router |
20:39.29 | *** join/#asterisk parker (n=ti@189.24.120.122) |
20:39.37 | parker | boa tarde |
20:42.22 | parker | estou tento um problema com o asterisk 1.4.21.1 |
20:42.22 | parker | tenho uma fila de atendimento, a fila funciona normalmente com os ramais internos, porem quando a ligação é externa e vai para fila, a mesma so tem duração de 1:18 min |
20:42.22 | drmessano | What exactly is "everything"? |
20:42.23 | parker | tanto do LINK E1 quanto do tronco SIP Vono |
20:42.23 | _trine | parker: we all have problems |
20:42.23 | _trine | drmessano: everything I can find |
20:42.23 | parker | _trine: my problem is with queue trougth external trunks |
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20:43.15 | parker | _trine: they only have a time of 1:18 min, aswered or not |
20:43.15 | drmessano | _trine: Considering there are some things I KNOW will not run on that platform, what is "everything"? |
20:43.15 | _trine | drmessano: routers if you are a single user as I am are now more than equal to the task |
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20:43.16 | _trine | drmessano: as i said I have only used 3% of my 8 gigs of flash space on my router |
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20:43.24 | drmessano | You're not explaining much of the everything part |
20:43.35 | parker | _trine: any suggestion? |
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20:43.46 | drmessano | Like for example, how is meetme working out for ya? |
20:44.10 | _trine | drmessano: everything I have found for asterisk I have found I have loaded into my router |
20:44.31 | drmessano | From the Openwrt repo? |
20:44.43 | _trine | parker: sorry I have no suggestions for you but maybe one of the other guys could help |
20:44.59 | parker | _trine: tanks |
20:45.04 | parker | _trine: thanks |
20:45.47 | _trine | drmessano: from everywhere,, but of course that means there will be more to add but so long as it does not take more than 8 gigs of space I'll be ok |
20:46.28 | p3nguin | cilkay: Those are lookups on your end when someone calls you. They have to query that database I was telling you about. Without paying per query, when someone calls you, you'll only get the CID number on your display. |
20:46.34 | drmessano | _trine: Some of that stuff is not even working.. so just loading a bunch of random crap doesnt imply any real functionality |
20:46.52 | _trine | drmessano: which stuff isn't working |
20:47.39 | parker | can someone help me with my queue problem? (asterisk 1.4.21.1) |
20:48.02 | cilkay | p3nguin: That could get more expensive than the phone company's overpriced visual caller ID if you get a lot of incoming calls. |
20:48.24 | p3nguin | cilkay: It's the difference between "Caller ID" and "Caller ID with name" services. |
20:48.43 | cilkay | What my monopolist telco calls "Visual Caller ID". |
20:48.50 | p3nguin | ah |
20:49.05 | cilkay | Actually, even at 1.25 cents per lookup, it's still a bargain by comparison. |
20:49.06 | p3nguin | AT&T calls it Caller ID with name. |
20:49.34 | _trine | drmessano: I also have freeswitch loaded on my router too |
20:49.46 | cilkay | I think my rip-off telco charges $15/month just for that feature. |
20:50.32 | p3nguin | I think if you wanted to by the 'with name' part a la carte from the telco, it's probably around $5 per month for unlimited queries. |
20:50.39 | p3nguin | here, I mean |
20:51.04 | p3nguin | But most plans include the name lookups in the rate. |
21:01.11 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
21:02.28 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
21:03.43 | ChannelZ | Anyone using Skype for Asterisk? |
21:07.11 | p3nguin | iax2-provision.c:518 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. What exactly do I need to configure to provision it? |
21:08.07 | ChannelZ | iaxprov.conf |
21:12.42 | p3nguin | The IAX provisioning is required for a trunk to the provider? |
21:12.49 | ChannelZ | not necessarily. |
21:14.04 | ChannelZ | provisioning is like automatic configuration |
21:14.17 | p3nguin | I have termination service and I currently use SIP for that. I have SIP phones on the inside of the Asterisk box. I wanted to change from SIP to IAX2 for the termination, but keep the phones as they are. |
21:15.14 | ChannelZ | that's fine. You just need to get IAX info from your service provider and set that up |
21:15.38 | ChannelZ | and then change your dialplan so it dials out on the right channels etc |
21:15.51 | p3nguin | I got the config info and put it all in. After I switch over to IAX, calls don't complete. Here's a debug from the time I dial the number until the time I hangup: http://pastebin.ca/1580538 |
21:16.20 | p3nguin | When I reload iax2, it spat out that notice about no provisioning. |
21:16.30 | retentiveboy | I'm looking at the Polycom ACD login/logout features and am wondering if there's any support for it in * 1.6. Anyone using it successfully? Or, should I just make a login/out dial-plan extension and a speed-dial? |
21:16.47 | p3nguin | iax2 show peers does show my ITSP peer. |
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21:18.25 | ChannelZ | It looks like a communications error, like the traffic isn't getting through. Is your Asterisk box a or behind a firewall? |
21:18.52 | p3nguin | The asterisk box is on a public IP address and it does have iptables on it. |
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21:19.28 | ChannelZ | is port 4569 open? |
21:19.31 | p3nguin | All outbound traffic is allowed, and the iax port is also open for inbound (although I don't know if that's important for outbound calls). |
21:19.36 | p3nguin | yeah |
21:20.02 | e4 | Finally have everything else hooked up. I'm getting 'CallToken Support required' on incoming calls even though the setting to disable the new security measures is set as per the pdf release. Is anyone else had issues with it? |
21:20.26 | ChannelZ | hmmm |
21:20.51 | ChannelZ | do you get incoming calls? |
21:21.12 | p3nguin | Asking me or e4? |
21:21.18 | ChannelZ | you, sorry |
21:21.41 | ChannelZ | not sure what e4 is talking about, sounds like you had to have been here earlier for the first part of the story :) |
21:22.06 | p3nguin | I have a SIP trunk from another provider for incoming calls. This IAX configuration will be only for outbound calling. Using SIP on the same provider, the calls go out normally. |
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21:23.32 | p3nguin | I just wanted to tinker with IAX in place of SIP. |
21:25.56 | p3nguin | The problem, as you may have noticed in the call debug was chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response. |
21:26.05 | ChannelZ | yeah |
21:26.28 | ChannelZ | which seems like a block of communication in one direction or another; whether completely or massive packet loss or what |
21:26.34 | p3nguin | From the looks of it, since I don't have iax devices, I don't need to worry with provisioning. |
21:26.56 | ChannelZ | right that provisioning warning is fine, I get it too and I setup an IAX connection between home and work |
21:27.20 | ChannelZ | What 'type' is your account in iax.conf ? Peer, friend.. |
21:27.24 | p3nguin | I really expected it would be like SIP -- create a peer, create a dialplan that is compatible, make calls. |
21:27.42 | p3nguin | friend |
21:28.28 | ChannelZ | it pretty much is like SIP |
21:29.10 | ChannelZ | So all UDP traffic out is allowed, and incoming UDP 4569 is allowed? |
21:29.19 | p3nguin | correct |
21:30.26 | ChannelZ | hmm |
21:35.21 | p3nguin | The packet count on port 4569 increases when I try to make a call, so something's getting back from the provider. |
21:35.35 | ChannelZ | if you only have outgoing call support should your type be 'peer' and not a friend? |
21:36.03 | ChannelZ | I'm not sure why that would matter but.. |
21:36.08 | p3nguin | I don't know that it affects it, but I am willing to change for testing purposes. |
21:36.56 | p3nguin | same problem |
21:37.36 | ChannelZ | so you hear nothing and then just get disconnected? |
21:38.11 | p3nguin | Takes about 2-3 seconds, then I get a fast busy (congestion). I have to manually end the call. |
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21:41.48 | p3nguin | According to that debug, it's 5 seconds, actually. :) |
21:42.36 | ChannelZ | well 4? 4001ms |
21:43.03 | ChannelZ | All I can think is something is not setup right on the provider side fully |
21:43.18 | p3nguin | I'll try several of their other servers, then. |
21:43.30 | p3nguin | They run Asterisk one some and OpenSER on some. |
21:43.31 | ChannelZ | you said the packet count on 4569 increases when you make a call, on the INPUT table I'm assuming? |
21:43.38 | p3nguin | yes |
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21:44.10 | ChannelZ | huh. Stumped |
21:45.49 | p3nguin | Same thing on all four US servers. |
21:45.51 | ChannelZ | I haven't quite used IAX like you, I've got a softphone using IAX into my * server |
21:47.47 | ChannelZ | what does your entry in iax.conf look like |
21:49.47 | p3nguin | http://pastebin.ca/1580691 |
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21:57.46 | ChannelZ | I really dunno. Seems like a problem on their end, if it's showing up as registered you've got communication going both ways |
21:57.51 | ChannelZ | what version of Asterisk are you running? |
21:58.03 | p3nguin | 1.4.24.1 |
22:01.50 | ChannelZ | do you have voicemail through them or any other 'local' extension you can dial other than an outside number? |
22:06.37 | louben | paw |
22:07.36 | p3nguin | Yeah, I can set up their echo test extension real quick. |
22:09.54 | p3nguin | Even calling those internal numbers gives me that auto-congestion message and a fast busy. |
22:20.51 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
22:20.55 | fofware | hey guys, May I define a varible in voicemail.conf and send it to my customized mail script? |
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22:33.29 | luca`gervasi | how can i setup a ring group? |
22:34.59 | ChannelZ | as in ringing multiple extensions? |
22:35.50 | luca`gervasi | yes |
22:36.32 | ChannelZ | just glob them together with & in your Dial command. Like Dial(SIP/joe&SIP/bob&SIP/gina) |
22:37.34 | manxpower | ~answers |
22:37.35 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
22:40.54 | luca`gervasi | thanks |
22:40.55 | luca`gervasi | bye bye |
22:46.53 | ChannelZ | argh this damn skypeforasterisk is making me mad |
22:55.30 | fofware | ChannelZ: do you know if May I define a varible in voicemail.conf and send it to my customized mail script? |
22:57.00 | manxpower | fofware: I doubt that you can. |
22:57.33 | manxpower | but if you read up on the docs of what variables ARE passed to the mailscript you might be able to set one of those to the value you want |
22:57.52 | fofware | manxpower: thanks verry much, maybe that the cause I have 2 days trying and nothing :o) |
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22:58.46 | manxpower | or you can save the value of the variable in AstDB or a file on the disk or something like that and have your script read the value |
22:58.58 | fofware | manxpower: I guess I tryed that too, I will try againg, thanks |
23:01.00 | fofware | manxpower: other way is read the value from table of SQL but I want evite 2 reads thanks |
23:01.55 | ChannelZ | what is the data/variable you're wanting to define? Something unique to each mailbox or what would be the use of configuring something in the .conf file |
23:03.07 | fofware | ChannelZ: I want send mail in spanish to spanish people in english to english poeople and .... |
23:03.42 | fofware | ChannelZ: so I have a context of eachone |
23:03.47 | ChannelZ | ah |
23:04.18 | fofware | ChannelZ: but mailbody only accept setting in general context |
23:04.49 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp209-68.adsl.forthnet.gr) |
23:05.07 | ChannelZ | so you need to write a completely different mailbody |
23:05.08 | Ad-Hoc | hi ppl |
23:06.17 | fofware | ChannelZ: the only I need send some value to an extenal script to know if laguage is english/french/spanish/portugues/etc |
23:06.41 | fofware | but I don't find the way to do that |
23:07.03 | ChannelZ | yeah built-in I don't think there is a way |
23:07.36 | ChannelZ | you'd have to just make your 'wrapper' program read the contents of the mail and parse it and do something else |
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23:07.59 | fofware | the other problem I found is my script done before asterisk voce mail done to send data |
23:08.19 | ChannelZ | huh? |
23:08.22 | fofware | ChannelZ: yes |
23:08.53 | manxpower | I suspect the VM context is passed to the script. |
23:09.00 | [TK]D-Fender | Or do the easy thing and mod something you CAN send to endoe that data |
23:09.05 | fofware | ChannelZ: no problem with the time, script can wait |
23:09.34 | [TK]D-Fender | But of course that would require some actual creativity... and we wouldn't want that, now would we? |
23:09.46 | [TK]D-Fender | is off for a while |
23:11.37 | ChannelZ | the mailcmd is assumed to be a delivery agent, like sendmail - my understanding is the code is constructing the entire mail (including the attachment if that's turned on) and that's just being fed to the 'mailcmd' over stdin |
23:11.50 | NovceGuru | anybody know if vitelity charges for toll free termination? |
23:11.53 | ChannelZ | so even if you could put a variable on the commandline being called, it wouldn't necessarily help you |
23:15.26 | fofware | ChannelZ: yes it send all by stdin and by default to sendmail, but you can change it to other script, and read stdin and make customized mail |
23:15.57 | ChannelZ | right but you have to parse that mail to replace the bits you want and keep the rest |
23:16.29 | fofware | but look like in voicemail process is not possible add any data |
23:17.53 | ChannelZ | Short of patching the code to add that capability, the "easier" way might be to just construct the emailsubject and emailbody in such a way that you can parse it easily in your wrapper script, and use the VM_MAILBOX as a switch to choose what language your script does |
23:17.58 | ChannelZ | like: |
23:18.37 | ChannelZ | emailsubject=[MSGNUM:${VM_MSGNUM}][MAILBOX:${VM_MAILBOX}] |
23:18.49 | *** join/#asterisk WinZ (n=winz@82.146.61.218) |
23:19.46 | WinZ | hello guys, anyone using asterisk with net2phone here? what are the settings? |
23:19.47 | fofware | [TK]D-Fender: My english is too poor, but that you did say sond like you don't like I make questions in the channel, If i'm wrong I'm sorry, If not thanks anyway |
23:19.53 | ChannelZ | your script can then search/parse for something looking like [MSGNUM:*][MAILBOX:*] and pull that out |
23:20.09 | fofware | ChannelZ: yes I have all data |
23:20.23 | fofware | inclusive the file attached |
23:21.17 | ChannelZ | so then parse the mailbox number in your script and use that to determine what language to reconstruct the message in |
23:21.53 | fofware | ChannelZ: but I did to modify one of that vars or add another one but i can't do that |
23:22.14 | ChannelZ | in splits your configuration to be in two places (voicemail.conf for the actual mailboxes and your script for the language) but again, short of writing a bunch of new support into the actual asterisk voicemail code to support an extra variable passed on the commandline.... |
23:22.35 | manxpower | "The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox. These arguments are passed to the program that you set in the externnotify variable. " |
23:23.16 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
23:26.38 | manxpower | You could also encode the information in the e-mail address i.e. fr^timmy@lassie.com then in side your script remove the fr^ from the e-mail address before sending the message |
23:26.49 | ChannelZ | externnotify would work but then you have to write a bunch more code to read the voicemail it's self and encode it to attach to an email |
23:27.10 | fofware | thanks manxpower and ChannelZ I will make another file to read langue value of the estensions |
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23:27.38 | ChannelZ | yeah manxpower that would be easiest actually |
23:29.29 | fofware | yes, that is a good point, manxpower but I don't want do that because is not for my own instalation |
23:29.54 | manxpower | fofware: there is no easy way to do what you want. There are only not-easy not-pretty ways. |
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23:33.44 | fofware | manxpower: thanks verry much, I'm only looking for one way that don't interfere with standart configurations files |
23:34.29 | ChannelZ | what are you writing the script in |
23:34.45 | fofware | so I guess one good thinks can be read with script the language of extension from other conf files |
23:35.31 | fofware | i'm writing in shell and lua, because I make web interfaces to routers |
23:38.06 | fofware | and both scripts work very well |
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23:45.59 | WinZ | !seen aurax |
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