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| 03:16.48 | Phunteltek | I have been getting calls from an "extension" that isn't defined in my extensions cfg file. there's never any audio, it just rings my phone. |
| 03:17.59 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.140.97) |
| 03:18.00 | Phunteltek | I'm wondering where to investigate these odd calls. |
| 03:19.00 | p3nguin | core set verbose 3 |
| 03:19.16 | p3nguin | Are you allowing anonymous SIP calls? |
| 03:20.47 | Phunteltek | maybe |
| 03:20.55 | Phunteltek | I'll check |
| 03:22.54 | Phunteltek | yeah anonymous is turned on. |
| 03:25.51 | jbroome | That's kind of you |
| 03:26.29 | Phunteltek | they called me every 41 minutes until I blocked czecheslovakia. |
| 03:28.11 | *** part/#asterisk wesphillips (~wphillips@75.53.137.6) |
| 03:28.25 | Phunteltek | then the connections came from amsterdam. they both showed up as extension 110. |
| 03:29.12 | Phunteltek | Then i blocked port 5060 from everywhere but my ITSP. |
| 03:31.49 | Phunteltek | what can people do if i allow anonymous connections? |
| 03:34.18 | jbroome | exactly what was just happening |
| 03:36.45 | Phunteltek | I was having my phone ring every 41 minutes |
| 03:37.32 | jbroome | Because any a-hole that found your PBX was able to have an extension on it |
| 03:38.15 | Phunteltek | no one was on the other end. no one made any calls through it. |
| 03:44.43 | Phunteltek | But IPs from all over the world are hitting my firewall. |
| 03:55.48 | Phunteltek | is there a good asterisk hardening guide? |
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| 04:05.45 | deltaray | I've setup a IP phone at home that connects over VPN to our asterisk server at the office. It is connecting and registering itself and I can receive audio, but it isn't sending for some reason. |
| 04:06.14 | deltaray | I did a tcpdump at the router at my house and it doesn't even seem to be sending udp packets that far to get back to the asterisk server. What might I be doing wrong? |
| 04:06.35 | drmessano | Allowing SIP guests is NOT THE SAME as allowing anyone to make a call through your PBX, or an indication that anymore more than a simple SIP attack was occuring |
| 04:06.37 | deltaray | Maybe the phone is misconfigured? But I've checked the settings and things seem pretty much the same as on my office phones. |
| 04:23.24 | tyman | What is the best channel to ask a Polycom phone centralized boot issue? |
| 04:36.59 | tyman | Is there anyone who has experience with Polycom phones that can help me out off channel? |
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| 04:57.05 | deltaray | Any ideas as to why my SPA941 IP phone won't send sip/udp out? |
| 04:58.21 | deltaray | wait, nevermind. I'm dumb, I was tcpdumping the wrong interface on my router. |
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| 05:02.52 | deltaray | Yeah, its working now. |
| 05:03.12 | deltaray | Didn't realize I still had some very old NAT config in Asterisk. |
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| 05:41.16 | Janky | any one ever used agile bill |
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| 05:48.03 | Phunteltek | What are the possible consequences of allowing anonymous inbound SIP calls? |
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| 07:23.49 | tish | what could be the problem if i can't login any user registered in asterisk? i'm using xlite with register set to always. thanks! |
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| 08:05.51 | z4nD4R | hi guys, how i can direct the unknow number ( extesion ) to the property domain?? Example if my asterisk becomme invite to 200@example.com, how make to asterisk redirect the invite to example.com??? |
| 08:11.38 | kaldemar | z4nD4R: what do you mean by example.com? you need to send the call to some specific place in your dialplan. |
| 08:12.25 | kaldemar | tish: lots of reasons, e.g. network issues, invalid configurations etc. do you see any REGISTER messages in asterisk CLI with sip debug enabled? |
| 08:12.32 | z4nD4R | kaldemar: my user want to call user in example.com... is impossible that i have all domain in the world in my dial plan |
| 08:13.21 | kaldemar | z4nD4R: so they want to dial a URI with their clients? |
| 08:13.37 | z4nD4R | kaldemar: i want somting like that: if the extension is unknov _X ( or i dont know ), so call the domain... which is requested |
| 08:14.51 | z4nD4R | kaldemar: yes |
| 08:15.00 | kaldemar | "which is requested" means that the incoming "number" is not just a number, but something like <numbers>@<domain>@<yourserver>, which means that you need an extension that matches to <numbers>@<domain>. |
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| 08:18.33 | z4nD4R | kaldemar: sometinth... how to define that.. this is not my user... send invite to the domain... |
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| 08:21.20 | schmidts | good morning |
| 08:22.11 | kaldemar | z4nD4R: well, set up a pattern like _X. that matches everything that starts with a digit, then check that there is a @ in the extension (with func CUT for example) and do whatever syntax checks you want and then dial. |
| 08:24.18 | z4nD4R | kaldemar: but this i must do for each domain? ... is impossible |
| 08:26.18 | z4nD4R | i want that the domain is filled dynamicaly ... example like variable or something |
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| 08:27.26 | kaldemar | z4nD4R: no, why would you? |
| 08:28.41 | kaldemar | z4nD4R: simplest example would be _X.,1,Dial(SIP/${EXTEN}), but you'd want to do atleast some validity checks with CUT, regex or how you want to do it. |
| 08:30.27 | z4nD4R | kaldemar: but in this case have to record to each domain which i want to call? right? |
| 08:32.10 | kaldemar | z4nD4R: why? |
| 08:32.28 | kaldemar | z4nD4R: you don't have to. you can if you want to. |
| 08:33.25 | kaldemar | z4nD4R: there is no need for anything domain specific in _X. or any cheks that you do in the extension. |
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| 09:10.52 | gvgi | Hi all, could anyone help me with a configuration problem? I'm a beginner.... |
| 09:12.23 | ectospasm | gvgi: don't ask to ask, just ask |
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| 09:16.45 | z4nD4R | kaldemar: thx i use _X.,1,Dial(SIP/${EXTEN}@${SIPDOMAIN}) |
| 09:16.59 | gvgi | ok :) I've an ubuntu lucid, I'mm trying to configure asterisk and one voip "client". The client can be Ekiga on the same asterisk's host or an audiocode mp-202. I configured sip.conf in this way: [1000] type=friend context=phones host=dynamic port=5061 secret=1234 ... but the voip softphone doesn't register. |
| 09:17.25 | ectospasm | gvgi: use pastebin, your formatting is all off |
| 09:17.29 | ectospasm | ~pb |
| 09:17.30 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 09:18.15 | gvgi | Let's have a look ;) |
| 09:18.27 | z4nD4R | gvgi: this problem have to.... solution: use other softphone... ( i use sip-communicator ) |
| 09:18.49 | kaldemar | gvgi: don't run asterisk and ekiga on the same host. |
| 09:18.54 | ectospasm | yeah, running a softphone on the same host as Asterisk is not a good idea |
| 09:19.21 | ectospasm | running X windows at all on the same host as Asterisk will be problematic |
| 09:20.37 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
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| 09:26.43 | gvgi | ok, ekiga successfull reister by another sistem... and what abount audiocodes mp-202? |
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| 09:49.43 | z4nD4R | kaldemar: Q: when asterisk look on extensions.conf on the right ext., is the same proces like in routing table ( from up to down? ) |
| 09:52.38 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:54.20 | kaldemar | z4nD4R: no. extensions are matched first with best match order, then includes in the order they are in the context. |
| 09:54.57 | z4nD4R | kaldemar: so my ext. are matched as last? |
| 09:59.46 | kaldemar | z4nD4R: of extensions in the context, yes. unless you have _., which you shoudn't. |
| 10:01.19 | z4nD4R | kaldemar: i have only 1 context ... and this _X.,1,Dial(SIP/${EXTEN}@${SIPDOMAIN}) want to use as default gw.... |
| 10:04.54 | marl_scot | hi there, i know i came accross this a while ago, but now i cant find it :( i have asterisk 1.4 + mISDN +LCR, and my DTMF tones are VERY short, how can i increase the length? |
| 10:07.41 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 10:09.52 | *** join/#asterisk Muty (~amir@unaffiliated/muty) |
| 10:10.11 | Muty | Hello guys.. I need help with asterisk and SIP |
| 10:10.54 | Muty | I have an SIP account (sipgate.de) and need to create outgoing calls with asterisk |
| 10:11.38 | Muty | I don't need incoming calls. Asterisk should just call out to different numbers in Germany. I found call files and I think thats the key for me |
| 10:12.09 | Muty | but I don't believe that I understood asterisk and its configuration (in special SIP configuration) right |
| 10:13.02 | WIMPy | Muty: http://das-asterisk-buch.de/2.1/ |
| 10:13.06 | Muty | Does anyone have a good tutorial that describes how to create outgoing calls with Asterisk over SIP from the terminal via call files? I don't want users with softphones or anything like this |
| 10:23.51 | ectospasm | Muty: check out the section on call files in the book |
| 10:23.57 | ectospasm | ~thebook |
| 10:23.58 | infobot | well, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 10:34.23 | *** join/#asterisk a_bug (~abhinav@122.180.97.198) |
| 10:34.36 | Muty | ectospasm: thanks for your help |
| 10:35.05 | Muty | I think I know how to create call files. But I get the following error when I try to execute them: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 10:35.14 | Muty | Maybe you know what this means? |
| 10:35.22 | a_bug | hi guys, with libpri's latest version ( and also with earlier versions), I get "Can't destroy call xxxxx, cref:xxxxx !" error messages, why is this ? Thanks ... |
| 10:35.33 | ectospasm | Muty: sounds like your trunk isn't available |
| 10:36.00 | Muty | Sorry for asking.. but whats "my trunk" :-)? |
| 10:36.15 | ectospasm | Muty: SIP, or DAHDI, or IAX2, or... etc... |
| 10:36.35 | ectospasm | actually, that's your "technology" |
| 10:36.39 | Muty | ok. Acutally it is available for incoming calls (that works) |
| 10:36.39 | WIMPy | a_bug: Anything special happening before? |
| 10:36.51 | Muty | But maybe it's a configuration problem for outgoing calls |
| 10:36.56 | ectospasm | Muty: what kind of trunk is it? |
| 10:36.59 | Muty | SIP |
| 10:37.03 | WIMPy | Muty: Turn up verbose and debug and watch. |
| 10:37.25 | ectospasm | SIP trunks inbound and outbound are independent. Look at the CLI debug and see if there are any clues |
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| 10:39.46 | *** join/#asterisk gvgi (~gvgi@94.163.128.187) |
| 10:40.14 | Muty | I see some messages like "Really destroying SIP Dialog ....@...." INVITE.. than REGISTER and than OPTIONS |
| 10:40.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 10:40.30 | ectospasm | that just means the SIP call was torn down |
| 10:41.30 | Muty | but those messages appear repeatedly |
| 10:41.34 | WIMPy | Looks like there is no call going on. |
| 10:41.49 | WIMPy | Better check the call file syntax again. |
| 10:41.54 | WIMPy | Or ~pb it. |
| 10:42.22 | WIMPy | Hmm. Didn't I see that working in a sentence once? |
| 10:42.27 | WIMPy | ~pb |
| 10:42.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 10:42.32 | WIMPy | Maybe not. |
| 10:47.14 | Muty | OK I've got the Channel setting in the cal files wrong |
| 10:47.55 | Muty | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out I read this wiki page here |
| 10:48.23 | Muty | And there it is mentioned that the Channel is SIP/sip-out/externalnumber |
| 10:48.38 | Muty | in my sip.conf I have saved my sip configuration as [sip-out] |
| 10:48.46 | Muty | But how can I set the number that should be called? |
| 10:49.11 | WIMPy | What do you think "externalnumber" stands for? |
| 10:49.29 | Muty | the number that is part of the public telephony number :-)? |
| 10:49.46 | Muty | ehh public telephony network I mean |
| 10:49.59 | WIMPy | yes |
| 10:50.41 | ectospasm | Muty: should be something like this: Channel: SIP/8675309 |
| 10:51.01 | ectospasm | or SIP/8675309@host |
| 10:51.20 | WIMPy | sip/probider/0123456789 |
| 10:51.33 | ectospasm | that'll work too |
| 10:51.59 | ectospasm | ...assuming you have "probider" as a SIP trunk in sip.conf et al. |
| 10:52.09 | *** join/#asterisk nfi|ermes (~sdff@host54-26-static.93-94-b.business.telecomitalia.it) |
| 10:52.41 | a_bug | WIMPy: can I interest you with a PRI q931 dump of the call ? |
| 10:52.42 | WIMPy | If not there would be a password missing. |
| 10:53.17 | Muty | and 0123456789 is the number that is called then right? |
| 10:53.20 | WIMPy | a_bug: Preferrably if its intensive debug, or debug 2 as it's called now. |
| 10:53.26 | WIMPy | ye |
| 10:53.30 | WIMPy | s |
| 10:54.23 | WIMPy | throws a ' up to the message to a_bug before p3nguin hits me. |
| 10:54.42 | a_bug | WIMPy: http://pastebin.freeswitch.org/14585 |
| 10:54.46 | a_bug | please advise |
| 10:54.47 | a_bug | thanks |
| 10:55.03 | gvgi | Hi all, another problem: i have an account messagenet. I found ad How-to and configured asterisk following it. It registers successfully but when i trie to call the number, asterisk says: No compatible codecs, not accepting this offer! |
| 10:55.15 | ectospasm | a_bug: why use a pb with a password? |
| 10:55.51 | gvgi | any suggestions? ^^ |
| 10:56.01 | ectospasm | gvgi: what codecs are disallowing/allowing in sip.conf? |
| 10:56.11 | ectospasm | s/are d/are you d/ |
| 10:56.28 | Muty | maybe you'll have a look? http://pastebin.ca/1998768 |
| 10:56.51 | gvgi | disallow all allow alaw g729 and ilbc |
| 10:57.52 | ectospasm | Muty: that's formatted the way your telco/ITSP expects the number to be formatted? |
| 10:58.04 | ectospasm | gvgi: in which section(s)? |
| 10:58.11 | WIMPy | a_bug: "Authotisation required" |
| 10:58.14 | gvgi | [general] |
| 10:58.29 | *** join/#asterisk PoTe (~PoTe@rev-200-40-119-222.netgate.com.uy) |
| 10:59.08 | Muty | and here is my sip.conf: may you have look too: http://pastebin.ca/1998771 |
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| 11:00.31 | gvgi | the how-to is at http://www.messagenet.it/voip/sw/asterisk.html |
| 11:01.05 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 11:03.19 | ectospasm | ~pb sip.conf | gvgi |
| 11:04.00 | Muty | ectospasm: did you find an error in my sip.conf? |
| 11:04.07 | ectospasm | I didn't look |
| 11:05.01 | Muty | ok |
| 11:05.16 | ectospasm | register line looks funny to me |
| 11:05.28 | ectospasm | try without the /XXXXXX |
| 11:06.02 | WIMPy | Muty: What does your console say when you try? |
| 11:06.25 | WIMPy | No, that register is ok. |
| 11:07.27 | Muty | -- Attempting call on sip/sipaccount/015256822841 for application Playback(hello-world) (Retry 1) |
| 11:07.31 | Muty | [Nov 22 12:07:20] NOTICE[5139]: pbx_spool.c:357 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 11:08.25 | ectospasm | Muty: what does "sip show peers" say? |
| 11:09.16 | Muty | ame/username Host Dyn Nat ACL Port Status |
| 11:09.19 | Muty | sipaccount/1112285 217.10.79.9 N 5060 OK (7 ms) |
| 11:09.31 | Muty | 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 11:09.31 | ectospasm | don't paste here |
| 11:09.31 | Muty | Name/username Host Dyn Nat ACL Port Status |
| 11:09.31 | WIMPy | The nat=yes for sipaccount is certainly wrong, but shouldn't cause the call to fail. |
| 11:09.31 | Muty | sipaccount/1112285 217.10.79.9 N 5060 OK (7 ms) |
| 11:09.31 | Muty | 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 11:09.33 | Muty | ups, sorry! |
| 11:09.37 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 11:09.38 | Muty | ectospasm: won't to again |
| 11:09.42 | gvgi | http://pastebin.com/Dh82pRpg |
| 11:10.01 | WIMPy | Muty: sip set debug on. |
| 11:10.11 | ectospasm | gvgi: don't paste sip.conf comments |
| 11:10.33 | WIMPy | Than pb the output or maybe you will spot an expaining message yourself. |
| 11:10.41 | ectospasm | gvgi: strip them with this command: egrep -v '^[[:space:]]*;|^$' sip.conf |
| 11:11.28 | a_bug | WIMPy: pastebin:freeswitch is the username:password |
| 11:11.35 | WIMPy | gvgi: An get a new password. |
| 11:11.46 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 11:12.23 | gvgi | http://pastebin.com/k2Uw2Tbn |
| 11:12.46 | nfi|ermes | i m installing asterisk on centos 5.5 in a server with HFC-S card . I installed dahdi-linux-complete-2.4.0+2.4.0 and asterisk-1.6.2.6 . I can't load module zaphfc. Where can i find it ? |
| 11:12.53 | gvgi | sorry WIMPy???? |
| 11:13.32 | ectospasm | gvgi: in [sip.messagenet.it] you disallow all codecs and allow none |
| 11:13.35 | WIMPy | gvgi: You should have removed your password. Noy you should change it ASAP. |
| 11:13.45 | WIMPy | now |
| 11:14.26 | ectospasm | gvgi: and you should also do as WIMPy demands, pasting your passwords like that is extremely insecure |
| 11:14.53 | shamelessn00b | lol |
| 11:14.59 | WIMPy | a_bug: Maybe that message is only misleading. You get a release complete directely after the setup, so there is not really a call, yet. |
| 11:15.07 | gvgi | ok |
| 11:16.21 | Muty | Can I post a one line message here? Or should I pastebin it? |
| 11:16.38 | WIMPy | No. Post it here. |
| 11:16.51 | Muty | [Nov 22 12:15:47] NOTICE[3746]: chan_sip.c:12718 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s) |
| 11:17.13 | kaldemar | nfi|ermes: zaphfc is a zaptel module, not DAHDI. why are you expecting to have zaphfc? |
| 11:17.16 | WIMPy | That's harmless. |
| 11:17.28 | WIMPy | Actually you don't eed to register at all to call out. |
| 11:17.47 | Muty | allright |
| 11:17.59 | WIMPy | kaldemar: I think the dahdi version is still called zaphfc. |
| 11:18.17 | nfi|ermes | kaldemar , i read that zaptel has became dadhi |
| 11:19.06 | nfi|ermes | i read everywhere that for HFC-S cards need dahdi |
| 11:19.41 | kaldemar | WIMPy: ah, it's the junghanns module. |
| 11:19.47 | WIMPy | nfi|ermes: There are quite some options. Firs of all, you might want to consider using Asterisk 1.8. That has much better ISDN integration. |
| 11:20.47 | WIMPy | And then you have several option for channel drivers: zaphfc with dahdi, misdn1 with chan_misdn or misdn2 with lcr and chan_lcr. |
| 11:21.06 | nfi|ermes | but with asterisk 1.8 , do i still nedd to use dahdi ? |
| 11:21.20 | kaldemar | nfi|ermes: try wcb4xxp first. |
| 11:21.45 | WIMPy | wcb4xxp only works for multiport cards. |
| 11:22.09 | WIMPy | NFI, why. |
| 11:26.31 | *** join/#asterisk [netman] (~netman@83.39.227.240) |
| 11:26.59 | nfi|ermes | WIMPy, you say that with with 1.8 i have several options for drivers (zaphfc also) . But is it included in asterisk or i should install dahdi anyway ? In this case where i can find zaphfc module ? And dahdi 2.4.0 is compatible with asterisk 1.8 ? |
| 11:27.57 | ectospasm | nfi|ermes: DAHDI 2.4.0 is definitely compatible with 1.8, not sure about the HFC module, though |
| 11:28.09 | WIMPy | It depends on wich of the three options you choose. They are all awailable for bith Asterisk 1.6 and 1.8. |
| 11:28.21 | WIMPy | both |
| 11:29.55 | nfi|ermes | i have kernel 2.6 and i read that misdn is no more compatible wiht this kernel |
| 11:30.04 | WIMPy | The 4th optio is capi and chan_capi, but that doesn't make sense for that hardware. |
| 11:30.16 | WIMPy | What version exactely? |
| 11:30.49 | nfi|ermes | Note: chan_misdn does not yet work with the new mISDN code that's included in the Linux kernel (2.6.27). You need chan_lcr if you need to use that. |
| 11:30.55 | nfi|ermes | http://www.voip-info.org/wiki/view/chan_misdn |
| 11:30.59 | kaldemar | zaphfc is a part of bristuff, you can get it from here: http://www.junghanns.net/downloads/ |
| 11:31.26 | WIMPy | nfi|ermes: I know, but you still have all three options. |
| 11:31.34 | kaldemar | if installed as is, it will make quite a few changes to asterisk too. |
| 11:31.59 | WIMPy | There is even a special misdn1 fork for the chan_misdn included with Asterisk 1.8. |
| 11:33.06 | nfi|ermes | ok i think i ll try to install bristuff 0.4, hoping it is compatible with 1.6 or 1.8 |
| 11:33.31 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 11:33.33 | WIMPy | For maximum features, I'd try the misdn1 fork with chan_misdn. Haven't tested that myself, yet. Otherwise misdn2/lcr/cahn_lcr has been running very smoothly for me so far. |
| 11:34.07 | WIMPy | I haven't looked at bristuff for ages. It used to require rather old versions of Asterisk. |
| 11:35.14 | kaldemar | nfi|ermes: 0.4.0-RC7 is the newest version. bristuff was last updated to asterisk 1.4.35.rc1. most of the patches will probably not work with 1.6.2 and 1.8.0. |
| 11:36.09 | WIMPy | I think I read about a standalone version of zaphfc somewhere for use with dahdi. |
| 11:37.03 | nfi|ermes | i was looking for that |
| 11:37.13 | nfi|ermes | but i didn t find it |
| 11:37.33 | WIMPy | didn't succeed, either. |
| 11:38.04 | nfi|ermes | so u suggest me misdn2 with lcr and chan_lcr |
| 11:39.07 | WIMPy | It does not support the new features introduced with 1.8 (yet?). But it works. |
| 11:39.14 | Muty | OK, calling out works now. But I can't set the CallerID |
| 11:39.24 | WIMPy | You need the git version for 1.8, however. |
| 11:39.59 | WIMPy | Muty: Do you have an account that allows you doing so? |
| 11:40.39 | Muty | Sipgate.de says that this is allowed yes |
| 11:40.40 | WIMPy | I thought that only works via the web interface. |
| 11:41.23 | Muty | You can set the value to "The device thats the number" |
| 11:41.29 | Muty | "the device sets the number" |
| 11:42.32 | Muty | the format is "CallerID: "forename surename" <Number>" right? |
| 11:43.24 | WIMPy | No "s |
| 11:43.31 | Muty | ok |
| 11:43.43 | WIMPy | But if you want to call real phones, setting a name is pretty pointless anyway. |
| 11:48.05 | Muty | WIMPy and ectospasm: Thank you for your support! Your help is really apprechiated |
| 11:49.45 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 11:50.52 | *** join/#asterisk LorenzBarth (~Miranda@62.216.165.71) |
| 11:51.43 | LorenzBarth | hi, i just play around with res_fax_digium and have the problem that the maxrate is ignored |
| 11:52.00 | ectospasm | LorenzBarth: how are you setting maxrate? |
| 11:52.05 | LorenzBarth | i try to limit it to 9600 but all faxe go with 14400 |
| 11:52.13 | ectospasm | LorenzBarth: how are you setting maxrate? |
| 11:52.19 | LorenzBarth | Set(FAXOPT(maxrate)=9600) |
| 11:52.35 | LorenzBarth | from the given sample |
| 11:52.51 | ectospasm | is there a reason you can't run it at 14400? |
| 11:53.05 | LorenzBarth | i have a lot drops at the moment |
| 11:53.18 | ectospasm | you have a FAX or FFA key? |
| 11:53.28 | LorenzBarth | with the hylafax / iaxmodem setup we could solve this by lowering speed |
| 11:53.43 | ectospasm | the two aren't comparable |
| 11:54.23 | LorenzBarth | as we are still waiting for our german distributor to deliver the channel licence we currently test with the FFA / Free licence |
| 11:54.44 | LorenzBarth | so what woul you do to get a better success rate? |
| 11:55.01 | ectospasm | I'd need to see debug of the failure |
| 11:55.09 | LorenzBarth | ok |
| 11:55.21 | ectospasm | with a FAX license, you're entitled to support directly from Digium |
| 11:55.55 | LorenzBarth | I start fax debugging and will create some logs |
| 11:56.05 | LorenzBarth | problem is, that my test are always success full |
| 11:56.34 | ectospasm | how many pages is the test fax? |
| 11:56.37 | LorenzBarth | 1 |
| 11:56.45 | LorenzBarth | the ones that fail |
| 11:56.53 | LorenzBarth | my test setup is a 1, 4 and 15 pages |
| 11:57.05 | a_bug | WIMPy: thanks :) |
| 11:58.02 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:58.10 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 11:58.27 | a_bug | Another question please people - If I understand correctly, libpri only communicates with the d-channel of the PRI line, while zaptel drivers do the entire TDM work by communicating with all the other channels in the card ? Thanks - please confirm this or correct me .. |
| 11:58.27 | shamelessn00b | guys, I need help with redictAction using AMI |
| 11:59.18 | shamelessn00b | I'm getting this message when I redirect a channel Spawn extension (abc, s, 1) exited non-zero on 'SIP/3001-0000002b' |
| 12:00.29 | WIMPy | a_bug: zaptel or dahdi provide accedd to the hardware. Signalling is done by chan_dahdi with the help of libpri. Data is processed by both dahdi and chan_dahdi. |
| 12:00.34 | shamelessn00b | == Manager 'manager' logged on from 192.168.10.36 |
| 12:00.35 | shamelessn00b | == Spawn extension (abc, s, 1) exited non-zero on 'SIP/3001-0000002b' |
| 12:00.37 | shamelessn00b | == Manager 'manager' logged off from 192.168.10.36 |
| 12:01.09 | LorenzBarth | ectospasm: http://pastebin.com/njw7cMLs |
| 12:01.18 | LorenzBarth | a sample with T2_Timeout |
| 12:01.27 | a_bug | WIMPy: so chan_dahdi does all the TDM work ? |
| 12:01.32 | kaldemar | shamelessn00b: what is the problem? |
| 12:01.50 | WIMPy | a_bug: No dahdi also does. |
| 12:01.52 | shamelessn00b | I can't redirect |
| 12:02.01 | shamelessn00b | whenever I redirect the call drops |
| 12:02.03 | a_bug | oh |
| 12:02.09 | a_bug | ok thanks |
| 12:02.13 | a_bug | a lot |
| 12:02.17 | shamelessn00b | exited non-zero on 'SIP/3001-0000002b' |
| 12:02.35 | shamelessn00b | I have an application at that context/extension/priority |
| 12:02.41 | WIMPy | a_bug: I *think* cahn_dahdi does littel, if anything, to voice. |
| 12:03.16 | shamelessn00b | basically I'm trying to implement barge-in functionality using AMI call |
| 12:03.30 | a_bug | oh |
| 12:03.38 | a_bug | damnit |
| 12:03.42 | a_bug | telephony is so involved |
| 12:03.46 | shamelessn00b | if my java code detects speech and the user is on a Playback() command in dialplan |
| 12:03.48 | a_bug | thanks for your help |
| 12:03.49 | *** part/#asterisk a_bug (~abhinav@122.180.97.198) |
| 12:04.24 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 12:04.44 | shamelessn00b | I'm trying to forcefully move it to the wait application defined at abc,s,1 |
| 12:04.47 | ectospasm | LorenzBarth: T2_TIMEOUT usually means (*I think*) that some side didn't respond |
| 12:04.59 | shamelessn00b | terminating the playback() application prematurely |
| 12:05.26 | shamelessn00b | damn I'm stuck at the simplest of all things |
| 12:05.48 | WIMPy | shamelessn00b: What Asterisk version? |
| 12:05.55 | shamelessn00b | latest trunk |
| 12:06.04 | ectospasm | why are you using trunk? |
| 12:06.17 | WIMPy | shamelessn00b: I think I saw something about AMI redirects on the issue tracker. |
| 12:06.20 | shamelessn00b | Asterisk SVN-trunk-r292970 |
| 12:06.34 | shamelessn00b | should I roll back to older version |
| 12:06.36 | shamelessn00b | ? |
| 12:06.52 | ectospasm | use something supported, like 1.8.0 or 1.6.2.14 |
| 12:06.58 | WIMPy | I fear, older would mean 1.6 then. |
| 12:07.03 | shamelessn00b | yeah |
| 12:07.14 | ectospasm | only use trunk if it provides a feature not in 1.8.0 |
| 12:07.25 | LorenzBarth | 1.6.2.14 |
| 12:07.28 | ectospasm | ...that you need |
| 12:07.38 | ectospasm | 1.8.0 is the current LTS release |
| 12:07.58 | shamelessn00b | what should I try, 1.6.x.x or 1.8 |
| 12:08.06 | ectospasm | ...you should migrate to 1.8.0 ASAP since 1.6.2 will go into security maintenance only mode in April |
| 12:08.29 | *** join/#asterisk mintos (~mvaliyav@114.143.162.219) |
| 12:08.30 | shamelessn00b | I hope it rids me of this redirect issue |
| 12:08.36 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 12:08.36 | ectospasm | if this is a new install, 1.8.0 will have the longest active support |
| 12:09.00 | shamelessn00b | all our production servers use 1.6.2.14 though |
| 12:09.33 | shamelessn00b | but since this is going to be a new service, imma try 1.8 for a change |
| 12:09.35 | shamelessn00b | :) |
| 12:09.37 | ectospasm | go with 1.6.2.14, but begin planning your migration now |
| 12:09.51 | ectospasm | ...do whatever you feel necessary |
| 12:10.10 | shamelessn00b | well, idk, how can I make my work available to the community |
| 12:10.21 | shamelessn00b | asterisk- sphinx 4 integration |
| 12:10.46 | shamelessn00b | with confidence score, noise reduction, decent barge-in |
| 12:11.12 | WIMPy | Sounds cool. |
| 12:11.42 | shamelessn00b | its basically a couple of perl scripts, some C code using Jack interface, a JAVA server hosting sphinx 4 |
| 12:11.53 | shamelessn00b | and jack_hook |
| 12:12.20 | shamelessn00b | reverst to 1.6.2.14 |
| 12:12.24 | shamelessn00b | brb |
| 12:14.34 | WIMPy | shamelessn00b: There is a patch on the issue tracker. |
| 12:15.06 | shamelessn00b | can you link me to it? |
| 12:15.08 | shamelessn00b | thanks |
| 12:15.13 | WIMPy | Issue #18230 |
| 12:18.25 | shamelessn00b | so should I just checkout latest trunk and it will be solved? |
| 12:18.40 | shamelessn00b | or do I need to patch it myself? |
| 12:19.01 | WIMPy | You need to patch yourself. |
| 12:20.31 | shamelessn00b | ok |
| 12:21.00 | E-bola | wonders if there's a list of patches for 1.8 somewhere |
| 12:25.39 | ectospasm | E-bola: svn.asterisk.org/view |
| 12:27.36 | E-bola | ectospasm: That doesnt list any patches? |
| 12:28.10 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 12:31.35 | ectospasm | E-bola: see the Changelogs there, or on downloads.asterisk.org |
| 12:31.50 | ectospasm | ...unless you mean unreleased patches |
| 12:32.01 | ectospasm | ...then the SVN server is your only shot |
| 12:32.29 | E-bola | ectospasm: I meana list that will show me what patches i should deploy, i knwo you can always manualy scover changelog or svn manualy, but its a pain and its easy to miss things |
| 12:32.47 | z4nD4R | who is here syntax error? http://pastebin.com/ypkSKknH |
| 12:34.03 | WIMPy | zamba: Your " and $[ are mixed up. |
| 12:35.35 | z4nD4R | WIMPy: right is exten => 123,1,GotoIf($"[${CHANNEL(secure_signaling)}" = ""]?fail) |
| 12:35.37 | z4nD4R | ? |
| 12:37.17 | WIMPy | That does not make sense. |
| 12:37.32 | WIMPy | Ever heared anything about nesting? |
| 12:37.39 | ectospasm | E-bola: you should only apply a patch if there's a specific bug that it addresses for you. You should wait until the next release if you have no pressing issues. |
| 12:39.01 | z4nD4R | WIMPy: syntax error: syntax error, unexpected '<token>', expecting $end; Input: |
| 12:39.02 | z4nD4R | 1" = "" |
| 12:40.41 | WIMPy | z4nD4R: You have close all kinds of containers in the reverse order they were opend. Always. That's neither Asterisk specific nor even computer related. |
| 12:41.21 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 12:41.24 | WIMPy | z4nD4R: In your case the closing part is the correct one. |
| 12:43.37 | *** join/#asterisk uzz75 (~uzz75@95.171.32.220) |
| 12:43.51 | uzz75 | Can someone help me with a Patton device ? |
| 12:46.31 | *** join/#asterisk jkroon (~jkroon@dsl-241-240-112.telkomadsl.co.za) |
| 12:52.44 | pabelanger | ~ask |
| 12:52.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:57.47 | *** join/#asterisk |Physis| (~|Physis|@187.58.65.48) |
| 12:59.58 | uzz75 | I have a Patton 4634 with 2 ISDN provider, incoming call everything ok, but if i try to call using bri 0/1, the call exit from bri 0/0. if bri 0/0 is busy, the asterisk return busy circuit |
| 13:00.17 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:00.25 | uzz75 | if i swap the isdn cable, the both isdn provider works. |
| 13:00.41 | uzz75 | but exit call only from bri 0/0 |
| 13:01.13 | uzz75 | asterisk don't report anything.. |
| 13:01.15 | ectospasm | uzz75: how are you dialing out in the dialplan |
| 13:01.18 | ectospasm | ? |
| 13:01.25 | uzz75 | Called 1004/327711xxxx but the call exit from 1001 |
| 13:01.42 | ectospasm | uzz75: what does the Dial() statement look like? |
| 13:02.20 | uzz75 | ectospasm, dial (from patton or asterisk) ? |
| 13:02.35 | ectospasm | outbound is from Asterisk, yes? |
| 13:02.43 | *** join/#asterisk |Physis| (~|Physis|@187.58.65.48) |
| 13:02.55 | ectospasm | uzz75: what does Dial() look like? |
| 13:03.42 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 13:04.17 | uzz75 | dialout-trunk,4,3277115872, |
| 13:05.44 | uzz75 | <PROTECTED> |
| 13:06.44 | uzz75 | The problem seems to be in the patton. |
| 13:06.55 | uzz75 | asterisk is using SIP trunk 1004 |
| 13:08.22 | schmidts | uzz75 do you use a huntgroup on the patton for outgoing calls? |
| 13:08.53 | uzz75 | no |
| 13:08.59 | uzz75 | the patton config is easy |
| 13:09.14 | schmidts | uzz75 could you pastebin your patton config, then i will take a look |
| 13:09.23 | schmidts | -pb |
| 13:09.29 | uzz75 | just 4 sip interface, 4 gateway |
| 13:09.48 | uzz75 | #----------------------------------------------------------------# |
| 13:09.48 | uzz75 | # # |
| 13:09.48 | uzz75 | # SN4638/5BIS/UI # |
| 13:09.48 | uzz75 | # R3.20 2006-10-04 H323 SIP BRI # |
| 13:09.48 | uzz75 | # 1970-03-02T23:34:38 # |
| 13:09.48 | uzz75 | # Generated configuration file # |
| 13:09.48 | uzz75 | # # |
| 13:09.48 | schmidts | pastebin! |
| 13:09.48 | uzz75 | #----------------------------------------------------------------# |
| 13:09.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:09.48 | uzz75 | cli version 3.20 |
| 13:09.51 | uzz75 | administrator admin password mHj6gZAZClmgca4w796Wuw== encrypted |
| 13:09.52 | uzz75 | dns-relay |
| 13:09.55 | uzz75 | webserver port 80 language en |
| 13:09.59 | uzz75 | sntp-client |
| 13:09.59 | uzz75 | sntp-client server primary 192.168.0.200 port 123 version 4 |
| 13:10.01 | uzz75 | system hostname SmartNode4634 |
| 13:10.02 | uzz75 | system |
| 13:10.04 | uzz75 | <PROTECTED> |
| 13:10.06 | uzz75 | <PROTECTED> |
| 13:10.08 | uzz75 | system |
| 13:10.12 | uzz75 | <PROTECTED> |
| 13:10.14 | uzz75 | profile napt NAPT_WAN |
| 13:10.16 | uzz75 | profile ppp default |
| 13:10.18 | uzz75 | profile tone-set default |
| 13:10.20 | uzz75 | profile voip default |
| 13:10.22 | uzz75 | <PROTECTED> |
| 13:10.25 | uzz75 | <PROTECTED> |
| 13:10.26 | uzz75 | <PROTECTED> |
| 13:10.28 | uzz75 | <PROTECTED> |
| 13:10.30 | uzz75 | <PROTECTED> |
| 13:10.33 | uzz75 | profile pstn default |
| 13:10.34 | uzz75 | profile sip default |
| 13:10.36 | uzz75 | profile dhcp-server DHCPS_LAN |
| 13:10.38 | uzz75 | <PROTECTED> |
| 13:10.42 | uzz75 | <PROTECTED> |
| 13:10.44 | uzz75 | <PROTECTED> |
| 13:10.46 | uzz75 | <PROTECTED> |
| 13:10.49 | uzz75 | <PROTECTED> |
| 13:10.50 | uzz75 | profile aaa default |
| 13:10.52 | uzz75 | <PROTECTED> |
| 13:10.55 | uzz75 | <PROTECTED> |
| 13:10.57 | uzz75 | context ip router |
| 13:11.01 | uzz75 | <PROTECTED> |
| 13:11.01 | uzz75 | <PROTECTED> |
| 13:11.02 | uzz75 | <PROTECTED> |
| 13:11.08 | uzz75 | <PROTECTED> |
| 13:11.08 | uzz75 | <PROTECTED> |
| 13:11.09 | uzz75 | <PROTECTED> |
| 13:11.12 | uzz75 | <PROTECTED> |
| 13:11.16 | uzz75 | <PROTECTED> |
| 13:11.18 | uzz75 | <PROTECTED> |
| 13:11.18 | uzz75 | <PROTECTED> |
| 13:11.20 | uzz75 | context cs switch |
| 13:11.25 | uzz75 | <PROTECTED> |
| 13:11.26 | uzz75 | <PROTECTED> |
| 13:11.26 | uzz75 | <PROTECTED> |
| 13:11.28 | uzz75 | <PROTECTED> |
| 13:11.30 | uzz75 | <PROTECTED> |
| 13:11.32 | uzz75 | <PROTECTED> |
| 13:11.35 | uzz75 | <PROTECTED> |
| 13:11.36 | uzz75 | <PROTECTED> |
| 13:11.38 | uzz75 | <PROTECTED> |
| 13:11.42 | uzz75 | schmidts, ? |
| 13:11.44 | uzz75 | http://pastebin.com/5BEu6LtF |
| 13:11.47 | uzz75 | sorry |
| 13:11.50 | schmidts | thx |
| 13:13.11 | uzz75 | schmidts, line 109 it's a copy-paste error, not present on the config file |
| 13:13.25 | schmidts | uzz75 thats not the best way to use a patton ;) |
| 13:13.37 | schmidts | give me 5 mins and i will write you a nice config which works |
| 13:14.05 | uzz75 | schmidts, I can imagine that, but that's what i found on the web.. |
| 13:14.44 | schmidts | ok ;) |
| 13:14.59 | uzz75 | schmidts, many thanks |
| 13:20.38 | Muty | Anyone uses a sipgate account for his asterisk server? |
| 13:21.12 | WIMPy | Muty: You don't expect to be the first, do you? |
| 13:21.32 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 13:21.51 | Muty | Ofcourse not, just asking. I can't set the CallerID.. at least not setting it working actually |
| 13:23.44 | WIMPy | Wrong format? |
| 13:24.03 | WIMPy | Do they say somewhere what they expect? |
| 13:24.57 | schmidts | uzz75 http://pastebin.com/B7b3uPHd i have only add the huntgroup but i dont know if this is exactly what you want |
| 13:25.11 | Muty | No, I am already searching along time know, but I don't find anything about that. They just say that it should be possible. |
| 13:25.54 | uzz75 | schmidts, i don't know what's the huntgroup..... |
| 13:26.17 | schmidts | uzz75 look at the config: service hunt-group HUNT_OUT this part |
| 13:26.43 | uzz75 | yes, it will try to call using 1,2,3 or 4 |
| 13:27.15 | schmidts | as i said it may not be what you want, but it should give you an idea how it works ;) |
| 13:27.25 | uzz75 | ok |
| 13:27.29 | uzz75 | i will try now |
| 13:27.57 | uzz75 | i'm reading the config file |
| 13:34.53 | *** join/#asterisk kartooon (~kartoon@unaffiliated/kartooon) |
| 13:35.00 | kartooon | hello |
| 13:37.47 | Muty | WIMPy: OK I found the format: CallerID: 49XXYY <SIPID> |
| 13:38.05 | Muty | Now it works beautifully |
| 13:38.22 | WIMPy | That looks strange |
| 13:38.36 | Muty | Jup. But it is required from Sipgate |
| 13:41.12 | *** join/#asterisk sbszulu (~dundubala@41.14.25.31) |
| 13:43.01 | *** join/#asterisk Holos (~chad@ip-70-38-12-204.static.privatedns.com) |
| 13:43.18 | Holos | Are people still using OSLEC for echo cancelling? |
| 13:43.54 | Holos | I'm setting up a Asus REVO with Xorcom Astribank for the FXO's and I'd like to get some reasonable echo cancellation setup, so I was thinking OSLEC since I've heard good things about it. |
| 13:44.13 | uzz75 | schmidts, thanks, thats work |
| 13:44.23 | tzafrir | Holos, hi. What distro do you use? |
| 13:44.43 | uzz75 | but there is alse the possibily to make a call using (forcing) the bri 0/1 ? |
| 13:45.19 | schmidts | uzz75 not with this config, you have to create a second huntgroup to make this |
| 13:45.31 | schmidts | uzz75 thats what i said about, it gives you an idea ;) |
| 13:45.35 | Holos | tzafrir: Centos 5.5 |
| 13:45.39 | tzafrir | uzz75, hmm... what channel driver do you use (what adapter?) |
| 13:46.32 | uzz75 | adapter ? |
| 13:46.53 | uzz75 | I'm from Italy, the first is Telecom nt1 |
| 13:47.06 | uzz75 | the second is something from Brennercom |
| 13:48.50 | uzz75 | schmidts, yes, you give the idea... i'm looking where's something tolding to use the huntgroup.... |
| 13:48.58 | uzz75 | but i try to do... |
| 13:50.10 | uzz75 | schmidts, found... |
| 13:52.51 | tzafrir | Holos, http://docs.tzafrir.org.il/extra/dahdi_linux_extra.diff is a ptch that includes OSLEC and a bunch of other things |
| 13:53.09 | tzafrir | (Generated from http://gitorious.org/dahdi-extra ) |
| 13:53.26 | tzafrir | So basically just apply it on your dahdi directory |
| 13:53.54 | tzafrir | patch -p1 <path/to/dahdi_linux_extra.diff --dry-run |
| 13:54.07 | tzafrir | patch -p1 <path/to/dahdi_linux_extra.diff # for real now |
| 13:54.22 | *** join/#asterisk lera_zed (~valera@88.211.48.174) |
| 13:54.25 | uzz75 | schmidts, don't work... |
| 13:54.48 | uzz75 | i did other group, using route call 1 dest-table FROM_ASTERISK3 and route call 1 dest-table FROM_ASTERISK4 |
| 13:55.04 | uzz75 | but the call goes out from bri 0/0 |
| 13:55.17 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 13:55.20 | tzafrir | uzz75, what BRI adapter do you use? |
| 13:55.33 | uzz75 | SN4634/3BIS |
| 13:55.51 | lera_zed | hello, guys. I am having strange problem - when I am making outgouing calls using PSTN - recipient sees number as Unknown instead of the PSTN one ... any directions ? e.g. I am not trying to replace callerid with something, just get PSTN line number displayed |
| 13:55.52 | Holos | tzafrir: Thanks.. I'll give that a whirl... |
| 13:56.12 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 13:56.28 | uzz75 | tzafrir, SN4634/3BIS/EUI |
| 13:57.53 | Holos | lera_zed: Are you using PRI/ISDN/BRI or Analog? If it's Analog, the telco will set it, if it's ISDN/PRI then it will be set by you on the outgoing call |
| 13:58.00 | *** join/#asterisk underdog (~underdog@72.46.208.205) |
| 13:59.44 | lera_zed | Holos: hm, to be honest I am not aware of details, we are renting office and and just ordered phone line, which is PSTN on my side - but apparently its connected to some internal system... |
| 13:59.53 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 13:59.53 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 14:00.04 | Scorpio2007 | any know of a way to do ALL PAGE and PAGE PICKUP ? |
| 14:00.48 | Scorpio2007 | for example: page all .. and say "xxx please pick up" and then when the xxx hits a button it bridges the 2 calls together |
| 14:01.06 | *** join/#asterisk iulhk (~iulhk@119.73.46.56) |
| 14:01.24 | Holos | lera_zed: So it sound like you're connecting to an analog line, there is no mechanism to "set" caller ID on an analog line, so it will be your upstream provider that needs to set it up. |
| 14:02.01 | Scorpio2007 | if i can do something like .. page all and wait 5 secs and then do something .. itw ould work .. is there a way to do .. timers in asterisk? |
| 14:02.17 | lera_zed | Holos: hm... strange - when I am plugging simple phone and making a call - number get displayed (PSTN one), and does not with asterisk ... |
| 14:02.55 | lera_zed | Holos: I've expected it to "just work" (c) e.g. be setted up on provider side too |
| 14:05.13 | tzafrir | lera_zed, Is the analog phone connected to Asterisk? Through what adapter? |
| 14:05.16 | Scorpio2007 | any help on page pickup in asterisk? |
| 14:05.31 | tzafrir | Caller ID never works there? |
| 14:06.08 | Scorpio2007 | is there a way to do timers in asterisk .. like do "something" and wait for 5 seconds ? |
| 14:06.27 | *** join/#asterisk chasing`Sol (~rc4@62.114.155.145) |
| 14:08.29 | lera_zed | tzafrir: yes connected to Asterisk its OpenVox A400P22 |
| 14:08.35 | Holos | Scorpio2007: Wait(5) |
| 14:09.07 | Scorpio2007 | thanks |
| 14:09.36 | WIMPy | Muty: Thanks for the info about <SIPID>. Just tried it :-) |
| 14:09.53 | lera_zed | tzafrir: no problems with incoming calls at all, and the only problem with outgoing is the missing PSTN number on the other end |
| 14:10.26 | lera_zed | tzafrir: while call to that PSTN number is forwarded as described in the calling plan |
| 14:11.03 | tzafrir | lera_zed, well, the PSTN sets the caller ID assagined to that call |
| 14:11.21 | tzafrir | You don't send out any |
| 14:11.47 | lera_zed | so I should play around with Set(CALLERID(all)=blah) ? |
| 14:12.28 | tzafrir | lera_zed, again, it makes no difference for a call you send out to the PSTN through FXO |
| 14:13.16 | xheliox | Stupid question (1.4), if you do an attended transfer, does the callid remain the same? |
| 14:13.28 | lera_zed | tzafrir: thats what I am thinking too, the only bit of a puzzle which I am missing why - PSTN number when calling using simple phone is setted up, and with asterisk does not |
| 14:13.49 | tzafrir | Where do you see the callerid? |
| 14:14.19 | lera_zed | I am unplugging asterisk box and pluggin in simple phone an calling to my mobile |
| 14:14.26 | *** join/#asterisk sbszulu (~dundubala@41.14.131.42) |
| 14:14.56 | lera_zed | then calling using sip client through the asterisk and getting Unknown |
| 14:24.54 | *** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca) |
| 14:28.38 | *** join/#asterisk a_bug (~abhinav@122.180.97.198) |
| 14:29.36 | a_bug | hi, a question please - when I restart my application that uses libpri, (asterisk/freeswitch), then libpri should get a q931 message RESTART that restarts all the channels of a span, or just the first ? |
| 14:30.05 | a_bug | I'm getting a signal RESTART, but only 1:1 is getting restarted by libpri |
| 14:30.18 | a_bug | shouldn't 1:1, 1:2, 1:3, 1:4 ... 1:31 be restarted ? |
| 14:31.24 | WIMPy | a_bug: It could actually send an restart for the entire interface. But as far as I observed it restarts them one by one with some seconds in between. |
| 14:32.47 | a_bug | WIMPy: can you please have a look at this q931 dump and tell me why it's restarting only 1:1 - http://pastebin.com/mNeYngQB |
| 14:32.51 | a_bug | pretty please :P |
| 14:33.01 | a_bug | I've highlighted the relevant q931 RESTART messages |
| 14:33.14 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 14:33.38 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 14:36.34 | WIMPy | Ok, so that's yet another possibility. You get a restart with a list of all channels. |
| 14:37.03 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:37.12 | WIMPy | And then you transmit an acknowledge without channel list and one with only the first channel. |
| 14:37.27 | WIMPy | That smells like a bug . |
| 14:45.44 | zamba | WIMPy: my what and what now? :) |
| 14:46.14 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 14:46.17 | *** join/#asterisk ming404 (~ming@dsl-173-248-231-71.acanac.net) |
| 14:47.10 | WIMPy | zamba: Sorry. Meant z4nD4R |
| 14:48.07 | *** join/#asterisk AlHafoudh (~alhafoudh@195.46.69.4) |
| 14:48.08 | AlHafoudh | hi all |
| 14:48.34 | AlHafoudh | is it possible to have two asterisks with same sip friends database? |
| 14:50.49 | espiceland | AlHafoudh: Look at realtime config. https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 14:51.04 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 14:57.53 | a_bug | damnit |
| 14:57.57 | a_bug | thanks WIMPy |
| 14:58.04 | a_bug | smells like a bug in ? |
| 14:58.06 | a_bug | libpri ? |
| 14:58.15 | WIMPy | yes |
| 14:58.22 | a_bug | damnit |
| 14:58.24 | a_bug | thanks a lot man |
| 15:00.08 | WIMPy | And i'm still trying to import it. I hat these variations in the logs :-( |
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| 15:05.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:16.50 | mintee | hey all, just updated from an older 1.4 install to 1.6.2 over the weekend, and now my SIP callerid names are being pushed to the sip phones... |
| 15:16.54 | mintee | exten => office,1,Set(CALLERID(name)="Hearing Solutions") |
| 15:17.27 | mintee | that's still the proper way set it from what I'm reading. |
| 15:18.02 | wdoekes2 | yes |
| 15:18.17 | WIMPy | no |
| 15:18.29 | wdoekes2 | or without double quotes |
| 15:18.29 | WIMPy | You don't use quotes. |
| 15:18.34 | mintee | damnit, just found out the voicemail email notifications aren't being emailed out too. |
| 15:18.41 | mintee | oh, ok, i'll remove the double quotes |
| 15:19.27 | WIMPy | So maybe it does work for you. I have seen several sip phones truncating names on non alphanumeric characters. |
| 15:20.05 | wdoekes2 | hm.. it depends on whether asterisk escapes them properly.. which it doesn't at the moment |
| 15:20.27 | wdoekes2 | I s/"/''/ everywhere, to make things look right |
| 15:21.02 | WIMPy | Well, " may be a problem, but what about ( and ) or even : ? |
| 15:21.16 | mintee | ok, ya, that fixed that |
| 15:21.33 | mintee | now i gotta determine the freaking voicemail problem :( |
| 15:21.57 | wdoekes2 | I had to replace () as well because of an issue with sipaddheader |
| 15:23.58 | mintee | rofl |
| 15:24.11 | mintee | maybe because I didn't replace the original voicemail.conf |
| 15:24.12 | Scorpio2007 | anyone know how i can use Page but then 5 secs later hang up the page ? |
| 15:24.58 | WIMPy | Scorpio2007: You want to limit pages to 5s? |
| 15:25.03 | Scorpio2007 | yah |
| 15:25.49 | WIMPy | Add a local cahnnel to the list that waits 5s and then does an answer and a hangup. |
| 15:26.05 | Scorpio2007 | not sure how to do that |
| 15:26.56 | WIMPy | You put that into an extension. And you add that extension to your page list via a local channel. |
| 15:27.19 | Scorpio2007 | alright brb |
| 15:27.22 | Scorpio2007 | ill give that a try |
| 15:28.14 | Scorpio2007 | well i want whoever paged |
| 15:28.17 | Scorpio2007 | to be limited @ 5 secs |
| 15:29.01 | Scorpio2007 | basically what im trying to do is do an all page but then give the option of the other side to do a page pickup .. so for example i can send out an all page "hey mike page pickup" and whereever hes in the building he can hit a button and then we are talking to each other |
| 15:30.03 | WIMPy | Why hit a button? What about simply picking up the phone? |
| 15:31.59 | Scorpio2007 | well what if someone wants to make an outbound call |
| 15:32.08 | Scorpio2007 | the accidently pick up the phone and now they are talking to each other |
| 15:32.15 | Scorpio2007 | they* |
| 15:32.29 | WIMPy | That could happen in the first 5s as well. |
| 15:32.46 | WIMPy | Especially in the first fraction of a second. |
| 15:33.35 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 15:33.37 | Scorpio2007 | true |
| 15:33.45 | Scorpio2007 | i dont know .. the best way to really to do the page pickup |
| 15:33.46 | WIMPy | But you should have some option to cancel a page on a particular phone, I'd assume. |
| 15:35.01 | Scorpio2007 | right now we have a toshiba phone system and the page option works great .. u pick up the phone hit a button "page al" u page .. then whoever wants to pick it up hits the "page pickup" and its good to go |
| 15:35.11 | Scorpio2007 | iam trying to mimic that function to some extent in asterisk |
| 15:35.49 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:36.38 | ChannelZ | (pushes page) Someone prrease talk to meee, I so ronrey! |
| 15:43.40 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 15:52.56 | Weazel | hey guys, i want to buy a digium analog 4port FXO, now i'm a total noob with all that, and if i understood correctly, i need to buy the AEX410 and then modules on it ? or how does that work ? |
| 15:53.49 | Weazel | anyone ? |
| 15:53.50 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 15:53.54 | cusco | check supported cards |
| 15:53.58 | cusco | buy one |
| 15:54.20 | cusco | and for every card there is a module ("driver") software so it is recognised and functional |
| 15:54.40 | Weazel | so do i need 4 modules or just one for a 4port analog card to work ? |
| 15:55.05 | Holos | Weazel: 4 FXO modules and 1 HW echo if you want it |
| 15:55.18 | Weazel | echo = card ? |
| 15:56.26 | Holos | Weazel: http://store.digium.com/productview.php?product_code=1AEX404EF |
| 15:56.30 | cusco | oh, I understood module as software |
| 15:56.57 | Holos | It seems that they call it an "HW echo can" which looks like hardware |
| 15:58.26 | Weazel | Holos: so that link u gave me is the whole shabang i need in order to put an analog card in my pbx |
| 15:58.35 | Weazel | so its 715... this whole business is so confusing :D thanks |
| 15:59.00 | Holos | Weazel: yah. it's the card, 4x fx0 modules and the hw echo all together |
| 15:59.20 | Weazel | Oh Hw echo might be the echo canceler probably |
| 15:59.42 | Holos | Weazel: yes. |
| 15:59.48 | Weazel | thanks a lot |
| 16:00.50 | mintee | anyone very familiar with hylafax and iaxmodem? |
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| 16:18.22 | thehar | yawns |
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| 16:24.27 | jaytee | mornin everyone |
| 16:25.01 | thehar | herros |
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| 16:28.52 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:28.53 | wcselby | o/ |
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| 16:35.36 | bbeers | Is there an easy way to disable RTCP packets? |
| 16:36.30 | *** join/#asterisk michael-i (~michael-i@141.41.40.127) |
| 16:37.45 | michael-i | I'm having trouble dialing into a meetme conference with a pin using an isdn telephone attached locally to an asterisk 1.6.1.20+libpri 1.4.11.4+dahdi 2.2.0.2. Any tips on chan_dahdi configuration needed to send/understand those digits? |
| 16:38.13 | michael-i | have transfer=yes and facilityenable=yes already. BRI NT PTMP mode was enabled with a simple patch to chan_dahdi |
| 16:40.08 | timholum | does anyone know if there as a way to an ami user to just one context? |
| 16:40.52 | michael-i | dialing the same conference from the outside world through a bri isdn line works... |
| 16:41.01 | WIMPy | What do ami users have to do with contexts? |
| 16:41.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 16:41.44 | timholum | WIMPy: I would like an ami user to be able to transfer calls, but only from a particular context |
| 16:42.22 | timholum | I am looking at running 2 company's phone's through just one server and separate them by context's |
| 16:42.35 | WIMPy | I don't think there is an option. |
| 16:42.59 | jaytee | ~seen beek |
| 16:43.03 | infobot | beek <~klinebl@pdpc/supporter/bronze/beek> was last seen on IRC in channel #asterisk, 10d 16h 50m 50s ago, saying: 'KavanS: Look for asterisk.pdf in the source directory.'. |
| 16:43.46 | tzafrir | michael-i, Do digits work elsewhere? |
| 16:43.56 | mintee | curses iaxmodem |
| 16:44.08 | tzafrir | On the VoiceMailMain menu? In a simple IVR? |
| 16:44.15 | michael-i | tzafrir: nope...i thought it was a transfer bug until i realized that digits weren't working in this simple test case |
| 16:44.17 | timholum | ok, I was pritty sure there wasnt either I just figured id ask, |
| 16:45.06 | tzafrir | If the digits don't work: do they get sent as DTMFs in the first place? |
| 16:46.03 | michael-i | nothing shows up in the debug output but somehow, sometimes I am able to activate a transfer with ** |
| 16:47.49 | wwalker | anyone know of a product (other than iptty) that supports TTY/TDD over SIP? (meaning 45.5 baudot like the TDD Mode that apparently was zap only and is pre-alpha) |
| 16:49.25 | wwalker | an asterisk add-on, another open source pbx, a reasonably priced commercial tool. (iptty is a possibility, but appears to only be sold as part of a full blown pbx deployment) |
| 16:52.54 | timholum | WIMPy: I just found out I might be able to do it with eventfilter, just thought I would pass the info along |
| 16:55.44 | wcselby | mintee - what's wrong with iaxmodem? |
| 16:56.31 | mintee | wcselby: been working on an upgraded asterisk+iaxmodem+hylafax setup for 2 days now |
| 16:56.45 | wcselby | heh |
| 16:56.49 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 16:56.57 | mintee | and the modems are not hearing anything from RECV |
| 16:57.18 | wcselby | try setting up asterisk+iaxmodem+interceptty+hylafax(ifax) |
| 16:57.26 | mintee | i've setup record=Yes |
| 16:57.34 | mintee | i can send faxes fine |
| 16:57.42 | wcselby | it's the receiving of the faxes? |
| 16:57.50 | mintee | ya |
| 16:57.55 | wcselby | let me see dialplan |
| 16:57.56 | mintee | via PRI |
| 16:58.15 | mintee | it's not the dialplan, cause * hands the call off to IAX2 without a problem |
| 16:58.30 | wcselby | hmmm |
| 16:58.39 | mintee | hyla pics up, then either |
| 16:58.40 | wcselby | so it passes it off to iaxmodem, but hylafax isn't answering? |
| 16:58.43 | mintee | SEND FAILED: JOB 3 DEST 8885216667 ERR No receiver protocol (T.30 T1 timeout) {E126} |
| 16:58.43 | wcselby | hmm |
| 16:58.55 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 16:59.23 | wcselby | do you have your config.iaxmodem's setup properly in hylafax? |
| 16:59.27 | mintee | hum... can't find the other error, but it's about silence |
| 16:59.33 | mintee | ya |
| 16:59.34 | mintee | 1 sec |
| 16:59.54 | mintee | http://groups.google.com/group/fa.hylafax/browse_thread/thread/926aeab996acb12a# |
| 16:59.58 | mintee | ^ me |
| 17:00.18 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 17:00.24 | mintee | had this running for over 2 years, and then decided to update, and all the sudden, notta |
| 17:00.26 | wcselby | looks |
| 17:00.56 | wcselby | i thought you said you could send faxes okay, it was receiving that didn't work? |
| 17:01.15 | mintee | Ya, that long is sending to itself |
| 17:01.44 | mintee | sendfax -> PRI -> * -> IAX2 -> hylafax |
| 17:02.01 | mintee | but i've been trying from an analog fax machine and still nothing |
| 17:02.33 | wcselby | how do the faxes come in if you use the analog fax machine |
| 17:03.00 | mintee | * detects the tone and hands it to IAX2/modemx |
| 17:03.06 | mintee | i have 6 iax's setup |
| 17:03.28 | wcselby | i mean, what's the typical setup going to be? PSTN -> PRI -> ASTERISK -> IAXMODEM -> HYLAFAX ? |
| 17:03.49 | mintee | oh, yup |
| 17:03.51 | mintee | exactly |
| 17:04.26 | wcselby | do you have special DID's set aside just for faxing or do these come in and route based on DAHDI fax tone detection? |
| 17:04.45 | mintee | DAHDI fax tone detection. |
| 17:04.53 | mintee | and that works fine... |
| 17:06.03 | mintee | i'm building hylafax 6.0.5 now |
| 17:06.18 | mintee | tried debian's apt pkg, and build the latest 3 |
| 17:06.24 | wcselby | k |
| 17:06.27 | mintee | hoping maybe its something in that |
| 17:07.05 | mintee | during faxaddmodem it queries the speed fine |
| 17:07.08 | mintee | Probing for best speed to talk to modem: 38400 OK. |
| 17:07.16 | mintee | which makes me think it's not iaxmodem |
| 17:08.06 | wcselby | hmmm |
| 17:08.30 | mintee | ya, exactly |
| 17:08.46 | mintee | i spend 8 hours trying to figure it out yesterday, and so far 3 more today |
| 17:11.02 | *** part/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 17:12.24 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 17:12.48 | mintee | Modem ttyIAX0 (+1.999.555.1212): Receiving facsimile |
| 17:12.49 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 17:17.27 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:18.22 | p3nguin | HEY, that's my number! |
| 17:21.03 | wcselby | p3nguin - heh |
| 17:36.04 | *** join/#asterisk r0d3nt (~astrutt@cheshire.telephreak.org) |
| 17:47.44 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 17:48.03 | *** join/#asterisk simplydrew (~simplydre@adsl-074-171-198-201.sip.owb.bellsouth.net) |
| 17:50.09 | *** join/#asterisk TicoTuco (~TicoTuco@201.55.145.226) |
| 17:53.56 | carrar | http://vimeo.com/17057322 |
| 17:55.41 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-4354d13c.dyn.optonline.net) |
| 17:55.46 | FuriousGeorge | hey all |
| 17:56.24 | FuriousGeorge | i set up an IVR, and set a timeout to be around 20 seconds... however, as soon as the message stops playing the channel is closed |
| 17:56.26 | FuriousGeorge | == Auto fallthrough, channel 'Zap/2-1' status is 'UNKNOWN' |
| 17:56.41 | wcselby | FuriousGeorge - show us the dialplan of your IVR |
| 17:56.55 | wcselby | use pastebin though |
| 17:56.59 | wcselby | ~pb |
| 17:56.59 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:58.13 | FuriousGeorge | wcselby: http://pastebin.ca/1999141 |
| 17:58.38 | simplydrew | hey guys. I was thinking about something today and wondered if this would be possible. some of my family members with landlines complain about extra fees with AT&T for voicemail. could I setup voicemail boxes for them and have the landline forward over for busy or no answer? |
| 17:58.53 | wcselby | FuriousGeorge - add a "WaitExten(20)" on line 6 |
| 17:59.09 | wcselby | FuriousGeorge - and then introduce a 't' extension |
| 18:00.02 | *** join/#asterisk tyman (~tyler@mail.cityofmadera.com) |
| 18:00.09 | p3nguin | simplydrew: You could, but some telcos charge a fee for each no answer or busy transfer. Check for those fees before you do it. |
| 18:00.20 | FuriousGeorge | wcselby: i added the waitexten, but can you be more specific about what the t extension should do besides hangup? |
| 18:00.58 | wcselby | well, the t extension is what happens when you timeout |
| 18:01.07 | _Corey_ | simplydrew: Also look into RDNIS, as you may want to grab this from your incoming telco if available to identify who's forwarding the calls in |
| 18:01.16 | simplydrew | p3nguin: ah, yeah, I remembered a long time ago it was a feature you could just add. didn't think of that |
| 18:01.19 | p3nguin | YOu can also set an absolute timeout value and use the T extension. |
| 18:01.21 | wcselby | you could also just add options after the WaitExten(20), such as a Goto() loop, or whatever |
| 18:01.21 | FuriousGeorge | wcselby: so exten => t,1,hangup |
| 18:01.35 | wcselby | FuriousGeorge - if that's what you want to do when you timeout, yeah |
| 18:01.43 | tyman | What's a good cli debug to test whether the ILEC has cutover numbers to my PRI? |
| 18:02.13 | FuriousGeorge | wcselby: yup, thanks |
| 18:03.31 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 18:03.57 | p3nguin | furiousgeorge: I like to play some sounds explaining what happened rather than just hanging up. |
| 18:04.23 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 18:04.26 | wcselby | p3nguin - yeah, I usually try to get my customers to do that as well |
| 18:04.32 | FuriousGeorge | p3nguin: yeah im gonna have them record a "goodbye" message |
| 18:04.36 | p3nguin | Playback(sorry-youre-having-problems&silence/1&vm-goodbye) then Hangup() |
| 18:07.03 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:07.16 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 18:07.42 | FuriousGeorge | also, my hint extensions stopped triggering lights on the phones at some point, and I'm not sure why. the hint extension needs to be defined in the context of the sip user, correct? that's the only requirement? |
| 18:09.02 | wcselby | FuriousGeorge - I usually define an entire context named [hints], then just include that in whatever context the phone is in |
| 18:09.08 | wcselby | makes administration easier on my part |
| 18:10.20 | FuriousGeorge | wcselby: that makes sense, but it wouldn't resolve this... something else is breaking blf, and i'm not sure what it could be |
| 18:10.38 | p3nguin | I alsways include it right before priority 1, which means if you can dial that particular extension you have access to the hint. |
| 18:10.45 | wcselby | FuriousGeorge - what changed when it stopped working? |
| 18:11.05 | FuriousGeorge | wcselby: not sure... i was just made aware of it now. apparently it hasn't worked in quite some time |
| 18:11.31 | wcselby | p3nguin - an interesting way to do it. i usually dial my extensions by pattern matching though, and I never got the pattern matching with hints down correctly |
| 18:11.49 | wcselby | FuriousGeorge - well, "how" is it not working? what types of phones? |
| 18:12.24 | FuriousGeorge | wcselby: they're snoms, and as to "how": blf is completely non-functional |
| 18:12.42 | wcselby | hmmm |
| 18:12.44 | *** join/#asterisk |Physis| (~|Physis|@187.60.42.210) |
| 18:12.48 | wcselby | only one client with snoms, and they don't use blf |
| 18:12.50 | wcselby | sorry |
| 18:12.57 | FuriousGeorge | perhaps it has something to do with another context I added which also defines hints |
| 18:13.18 | wcselby | if you do "core show hints" at the command line, do you get any information? |
| 18:14.06 | p3nguin | I have a hard-coded extension for every local phone, so it's not an issue for me. I can see how using a single pattern for a bunch of phones won't work with my method of hints, though. |
| 18:15.19 | FuriousGeorge | wcselby: i do, and it looks right. i may have just fixed it |
| 18:15.36 | wcselby | p3nguin - there's supposed to be a way to do it, i just haven't taken the time to really look at it |
| 18:18.28 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 18:19.52 | ruben23 | hi guys i have hosted server on a hosting company and install asterisk but the problem is- im facing huge echo even a single call, my bandiwdht is high, i get low latency, no packet loss---> any idea guys.. |
| 18:20.13 | wcselby | ruben23 - are you on a dedicated server or a VPS or ? |
| 18:20.15 | *** join/#asterisk ming404 (~chatzilla@dsl-173-248-231-71.acanac.net) |
| 18:20.25 | wcselby | how are you calls coming in, over SIP trunks? |
| 18:20.39 | FuriousGeorge | if i have an IVR with the option to dial extensions 1, 2, or 3 and an option to dial an arbitrary "if you know your party's extension", how can i stop "100" from instead triggering "1"? |
| 18:20.42 | ruben23 | <PROTECTED> |
| 18:20.44 | p3nguin | barfs |
| 18:21.06 | wcselby | FuriousGeorge - you add a timeout(digit) (i think) |
| 18:21.27 | p3nguin | furiousgeorge: If 100 exists, it won't match 1 unless you have _1! as the exten. |
| 18:21.28 | tyman | How do you reload the voicemail.conf without restarting asterisk |
| 18:21.29 | FuriousGeorge | wcselby: i got that set to (10) or something high like that |
| 18:21.42 | wcselby | yeah, that's what we did in 1.4.x -> exten => s,n,Set(TIMEOUT(digit)=2) |
| 18:21.44 | FuriousGeorge | oh right, i didn't put the _ |
| 18:21.59 | p3nguin | furiousgeorge: You don't need to make a pattern for 1 or for 100. |
| 18:22.09 | p3nguin | exten => 1,1,Do_stuff() |
| 18:22.13 | p3nguin | exten => 100,1,Do_stuff() |
| 18:22.25 | wcselby | what p3nguin said :) |
| 18:22.34 | ruben23 | im using voip for every calls |
| 18:22.39 | p3nguin | If you have _1!, 100 will never work. |
| 18:23.01 | ruben23 | wcselby:its dedicated and my calls are uing voip |
| 18:23.02 | wcselby | ruben23 - not sure why you're getting echo, but your bandwidth may be from the codec you're using. i'm guessing g711 |
| 18:23.21 | wcselby | ruben23 - are you getting echo, or are you getting breaks and pauses in your audio? |
| 18:23.25 | FuriousGeorge | p3nguin: it was an even sillier problem... instead of exten => _XXX, which i was hoping would match 100, I had omitted the underscore, so it would never hit |
| 18:23.36 | p3nguin | furiousgeorge: oh |
| 18:23.45 | wcselby | FuriousGeorge - heh, done that before too |
| 18:24.24 | wcselby | FuriousGeorge - got to the point where I was putting the _ in front of explicit extensions, just so I wouldn't forget to do it in front of variable matches |
| 18:24.25 | wcselby | lol |
| 18:24.36 | p3nguin | :) |
| 18:24.37 | wcselby | of course, that introduces issues of it's own |
| 18:24.39 | FuriousGeorge | heh |
| 18:24.44 | p3nguin | its own! |
| 18:24.50 | wcselby | haha |
| 18:24.55 | ruben23 | wcselby: the other side can hear his/her voice when he calls me or when i call him i can hear my voice.. |
| 18:24.58 | wcselby | my fingers type faster than my brain can think, p3nguin |
| 18:25.17 | wcselby | just the fact that I'm putting coherent words together without a million typos is good for me today |
| 18:25.34 | ruben23 | wcselby: ther are also breaks and pause on the calls aside form echo. |
| 18:25.53 | wcselby | ruben23 - do you have dahdi installed at all, at least for dahdi_dummy? |
| 18:26.20 | ruben23 | <PROTECTED> |
| 18:26.30 | wcselby | did you start it? |
| 18:27.06 | wcselby | actually, it's time for lunch |
| 18:27.10 | wcselby | i'll be back later |
| 18:27.11 | wcselby | :) |
| 18:27.53 | p3nguin | hahaha ... I love the scene in American Dad where he pulls out his gun and points it at some woman's face and says, "What if I told you I'm about to set the record for shooting a woman in the face the most times in a row?" |
| 18:28.42 | tyman | Anyone know what could be causing this? http://pastebin.com/taCwxeX7 |
| 18:34.54 | ruben23 | <PROTECTED> |
| 18:38.56 | ruyo | Is there any way to insert someone on the top of a queue? |
| 18:49.32 | WIMPy | What's going on when I get TE4XXP: Version Synchronization Error! ? |
| 18:52.08 | frigidzephyr | WIMPy: the card is possibly not seated well, re-seat it |
| 18:52.26 | frigidzephyr | WIMPy: I would also contact Digium technical support |
| 18:52.50 | frigidzephyr | be sure you are using the latest DAHDI driver |
| 18:54.18 | WIMPy | Hmm. Will take a look. |
| 18:57.28 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 19:01.53 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:03.26 | *** join/#asterisk a_bug (~abhinav@122.163.21.48) |
| 19:05.36 | a_bug | hi guys, libpri's q931.c, line 892 assumes that only T1 lines use slot maps |
| 19:05.45 | a_bug | Can we please expand it to E1 lines as well ? |
| 19:06.24 | a_bug | Also, how is it assuming that only T1 lines use slot maps ? By & -ing data[1] with 0x10 ? |
| 19:06.30 | a_bug | 0x10 has something to do with T1 lines ? |
| 19:15.21 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 19:23.38 | russellb | a_bug: the most likely person to respond to a question like that isn't on IRC. He'll see it if you post to the asterisk-dev mailing list, though |
| 19:24.52 | a_bug | ok russellb thanks a lot, will do |
| 19:24.59 | a_bug | :) |
| 19:25.18 | russellb | np |
| 19:28.38 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 19:30.58 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
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| 19:45.23 | carrar | moof |
| 19:48.36 | pabelanger | foom |
| 19:52.08 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 19:52.36 | thehar | orly |
| 19:52.58 | kalimc | Can anyone recommend a Speech-to-text application that would work with asterisk? Want to enable voice recognition on prompts in addition to numeric |
| 19:56.44 | _Corey_ | kalimc: Check out Lumenvox |
| 19:56.51 | kalimc | ok ty, will do now :D |
| 19:57.03 | Qwell | speech-to-text and voice recognition are very different things |
| 19:58.48 | kalimc | I am looking for being able to accept user's voice from asterisk menu's |
| 19:59.01 | Qwell | that would be speech recognition |
| 19:59.06 | kalimc | However... |
| 19:59.20 | kalimc | I am also looking for transcription software to convert voice mails |
| 19:59.42 | _Corey_ | kalimc: Lumenvox is good for the former, not the latter :( |
| 19:59.52 | kalimc | Ok thanks, it looks pretty good so far |
| 20:00.18 | *** join/#asterisk paultech (paultech@shell.vaerchi.com) |
| 20:00.50 | paultech | Hi, looking for a way to play audio thru Page() using SPA942 and asterisk 1.6 |
| 20:01.00 | paultech | Need to automatically stream a gsm via the Page application |
| 20:05.51 | *** join/#asterisk beek_ (~klinebl@pdpc/supporter/bronze/beek) |
| 20:07.54 | *** join/#asterisk omish-man (~shlomo@IGLD-84-228-71-4.inter.net.il) |
| 20:10.07 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 20:10.50 | *** part/#asterisk omish-man (~shlomo@IGLD-84-228-71-4.inter.net.il) |
| 20:12.00 | *** join/#asterisk srini (~sri@219.91.201.74) |
| 20:15.09 | sawgood | In * 1.4, what directory should I place my announcements in (is it still /var/lib/asterisk/sounds) |
| 20:15.35 | *** join/#asterisk Ceeriael (~ceeriael@62.61.146.182.generic-hostname.arrownet.dk) |
| 20:16.39 | srini | I am facing a problem with recorded calls. They are feable! Any idea where to set the volume? |
| 20:17.55 | Ceeriael | Hi, i have a working Snom 300 phone with a working SIP configuration that i would like Asterisk to take care of. I have the following settings on my Snom phone and need to know what kind of Incomming / outgoing and Registration string I should be using: user_name1: a | user_pname1: b | user_ppass1: bpass | user_pass1: apass | user_outbound1: urla.dk | user_host1: urlb.dk | user_server_type1: |
| 20:17.55 | Ceeriael | broadsoft |
| 20:18.01 | Ceeriael | thx in advance |
| 20:18.15 | Ceeriael | srini 2 sec, i have volume boost on my server. ill try to locate the setting |
| 20:18.24 | _Corey_ | paultech: Look up the Local channel and explore that. It should help you achieve what you want. |
| 20:19.04 | *** join/#asterisk CaneToad (~CaneToad@CPE-58-174-135-10.mjcz1.woo.bigpond.net.au) |
| 20:22.03 | Ceeriael | srini my mistake. i had Gain on voicemail |
| 20:22.50 | sawgood | exten => _XXXX,1,Dial(SIP/${EXTEN}) |
| 20:23.10 | ChannelZ | paultech: SIPAddHeader(Call-Info: \;answer-after=0) and then dial |
| 20:23.21 | sawgood | If instead of having an incoming extension to extension call dial the phone (I want it to go to an announcement) what would I change in the above string? |
| 20:24.02 | ChannelZ | sawgood: I don't know if I understand the question, but you'd Playback(whatever) |
| 20:24.19 | sawgood | Is the syntax like this: |
| 20:24.28 | sawgood | exten => _XXXX,1,PlayBack(SIP/${EXTEN}) |
| 20:24.35 | ChannelZ | no |
| 20:24.48 | ChannelZ | You said you wanted it to play an announcement instead? |
| 20:24.49 | p3nguin | You can't playback a SIP device. |
| 20:24.50 | *** join/#asterisk srini (~sri@219.91.201.74) |
| 20:24.53 | srini | Ceeriael: Does that effect the call recording volume? |
| 20:25.14 | Ceeriael | srini i dont think so... but srini i think i found something: http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Asterisk_/Q_23082762.html |
| 20:25.17 | sawgood | Is the command called Playback (or) PlayBack? |
| 20:25.18 | Ceeriael | srini in the bottom. |
| 20:25.22 | sawgood | I read about it on the Internet |
| 20:25.25 | Ceeriael | srini last answer. |
| 20:25.36 | p3nguin | core show application playback ... see what you get. |
| 20:25.37 | ChannelZ | Playback(some-sound-file-you-want-played) |
| 20:25.54 | sawgood | ty! |
| 20:26.54 | sawgood | How does Asterisk 'know' where the file I want playback (is located at) |
| 20:27.01 | sawgood | Is it /var/lib/asterisk/sounds by default? |
| 20:27.14 | ChannelZ | generally |
| 20:27.30 | sawgood | wow ...this seems rather easy! |
| 20:27.39 | ChannelZ | Pretty much |
| 20:27.41 | sawgood | thank you |
| 20:31.40 | *** join/#asterisk MiserySoft (~Lee@195.80.4.101) |
| 20:35.18 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 20:39.27 | wcselby | unless you're using 1.6.2.x or newer, then it's /var/lib/asterisk/sounds/en/ whatever |
| 20:39.38 | wcselby | or whatever language you're using |
| 20:40.01 | tzanger | hmm, how does the transcoder card fit into the asterisk architecture? |
| 20:40.27 | tzanger | i.e. how do you tell Asterisk/DAHDI that it exists and how to make use of it for call processing? |
| 20:42.13 | ChannelZ | I think there's a driver for it "and it just does" |
| 20:42.51 | ChannelZ | wctc4xxp |
| 20:42.56 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:43.09 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 20:45.11 | russellb | there is a transcoding interface in DAHDI, and then a codec_dahdi module in Asterisk that can use it. |
| 20:45.48 | tzanger | russellb: so SIP/IAX2/etc channel drivers "test the water" and request it if it exists? |
| 20:45.58 | tzanger | er sorry |
| 20:46.01 | tzanger | you said codec_dahdi |
| 20:46.06 | tzanger | so it just registers as a codec |
| 20:46.07 | russellb | nah ... nothing else in Asterisk knows it is going on |
| 20:46.08 | russellb | right |
| 20:46.19 | tzanger | possibly with a higher priority than the host-based ones |
| 20:46.21 | russellb | it's just a normal "I need a translation path from X to Y" operation going one |
| 20:46.26 | tzanger | right right |
| 20:46.31 | russellb | yup |
| 20:46.37 | tzanger | makes sense |
| 20:48.11 | tzanger | thanks guys :-) |
| 20:48.18 | thehar | flails |
| 20:49.26 | russellb | thehar: you would |
| 20:49.39 | thehar | watch it |
| 20:49.40 | thehar | buddy |
| 20:50.32 | *** join/#asterisk tyman (~tyler@mail.cityofmadera.com) |
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| 20:55.36 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 20:58.38 | sawgood | If I use the Playback(filename) option, does the filename need to be Ulaw format (.wav files will not work)? |
| 20:58.40 | marl_scot | please, please, am getting really desperate with this one, i have * 1.4 misdn/LCR and am trying to extend the length of dtmf tones when dialing out sip->lcr |
| 20:59.04 | mintee | fyi, regarding fax detection... just to throw it in here for the logs. http://pastebin.ca/1999331 |
| 21:01.10 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 21:04.00 | sawgood | exten => _9980,1,Playback(Recording_Hotline) |
| 21:04.10 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 21:04.19 | sawgood | Will * know to go to /var/lib/asterisk/sounds to find Recording_Hotline)? |
| 21:04.43 | marl_scot | when ever i hit a number to create a dtmf tone, i just get a click on the remote end :( |
| 21:04.48 | bmoraca_work | sawgood: that directory is configured in a config file in your asterisk config directory |
| 21:04.55 | sawgood | thank you! |
| 21:05.36 | bmoraca_work | sawgood: it will attempt to play files of any extension, first trying files that are the same codec as the channel...if it can't find that, it will attempt to find any file that it CAN decode and will transcode it into the appropriate codec |
| 21:05.52 | sawgood | nice information! |
| 21:06.20 | sawgood | I see [directories] listed in /etc/asterisk/asterisk.conf |
| 21:06.29 | sawgood | but I'm not sure what the 'title' of the sounds directory should be |
| 21:06.39 | sawgood | astetcdir => /etc/asterisk |
| 21:06.46 | sawgood | is an example I see in the file |
| 21:07.02 | wcselby | night all |
| 21:07.51 | bmoraca_work | sawgood: i believe that it interprets the path from astvarlibdir, though I could be wrong. |
| 21:08.18 | p3nguin | astvarlibdir/sounds perhaps |
| 21:08.45 | sawgood | I'll move the sound file to that dir (so as not to disturb other variables), and I'll test |
| 21:11.24 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 21:13.22 | sawgood | I have my file in .wav format, and I think maybe I need to save-as using different settings |
| 21:13.56 | sawgood | I set the owner and group permissions to asterisk, and I chmod 664 to the file |
| 21:15.13 | p3nguin | Your wav files have to be 16-bit mono 8 kHz. |
| 21:15.23 | p3nguin | Anything else can't be expected to play. |
| 21:15.33 | sawgood | If that is the case, I'm sure the .wav file is correct then |
| 21:15.41 | sawgood | I'll paste bin the error I get |
| 21:15.54 | sawgood | I'm going to use another filename for testing to be sure |
| 21:16.57 | p3nguin | Playback(demo-congrats) |
| 21:17.29 | sawgood | http://pastebin.com/f07Veccj |
| 21:18.14 | p3nguin | What does ls -l /var/lib/asterisk/sounds/Disconnected_Out.* show? |
| 21:18.51 | jaytee | beek ping |
| 21:19.01 | p3nguin | peek bing! |
| 21:19.07 | jaytee | :-) |
| 21:19.34 | sawgood | wc: /var/lib/asterisk/sounds/Disconnected_Out.*: No such file or directory |
| 21:19.50 | p3nguin | wc? |
| 21:20.02 | sawgood | word count? |
| 21:20.07 | beek | jaytee... Hello! |
| 21:20.19 | p3nguin | Where did you put Disconnected_Out? |
| 21:20.20 | jaytee | how are ya? |
| 21:20.33 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 21:20.34 | beek | jaytee doing well my friend. How about you? It's been a long time. |
| 21:20.45 | jaytee | someone asked about you last night in the ##vyatta channel |
| 21:21.05 | beek | I didn't know that you were hanging there now. |
| 21:21.13 | jaytee | occassionally |
| 21:21.18 | jaytee | and only recently |
| 21:21.29 | beek | Have some V deployments? |
| 21:21.42 | sawgood | I have Disconnected_Out (in two directories) /var/lib/asterisk and /var/lib/asterisk/sounds |
| 21:21.49 | jaytee | been real busy but getting into doing some more * deployments for some of our clients. |
| 21:21.50 | *** join/#asterisk bbeers (~bbeers@static-96-254-222-162.tampfl.fios.verizon.net) |
| 21:22.00 | *** join/#asterisk carrar (~tim@2604:5000:11:1::3) |
| 21:22.08 | beek | Sounds good. |
| 21:22.29 | jaytee | and looking at setting up our customers with small Vyatta systems instead of Cisco appliances. |
| 21:22.35 | *** part/#asterisk wwalker (~wwalker@208.92.232.27) |
| 21:22.36 | p3nguin | sawgood: There's no reason it needs to be in /var/lib/asterisk, so move that instance of it to /var/lib/asterisk/sounds/en/ if you have it. |
| 21:22.46 | beek | I kicked Cisco out the door here and have been running Vyatta for a couple of years now. |
| 21:22.47 | sawgood | I do have that directory |
| 21:23.30 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:23.36 | jaytee | well, that was odd |
| 21:23.49 | beek | I thought you might have accidently hit ^W |
| 21:23.57 | jaytee | nope |
| 21:24.30 | jaytee | but I wish I had Xchat instead of mIRC on this system. then I could look back at who was asking for you on the Vyatta channel. |
| 21:24.47 | p3nguin | Because mIRC doesn't have logging? |
| 21:25.20 | jaytee | it does but I don't think it's turned on by default and I know I didn't set it up myself. |
| 21:25.37 | sawgood | with Disconnected_Out in /var/lib/asterisk/en (and owner, group, file permissions set), I get the same error messages |
| 21:26.02 | beek | jaytee: I'll drop over and say hello over there. Whoever it was will likely reply. |
| 21:26.18 | p3nguin | sawgood: You realize that you've told me you put the file into the directory where I asked you to run ls -l on it, right? |
| 21:26.19 | jaytee | ah, found it! it was bahadunn |
| 21:26.28 | p3nguin | (1518.14) <p3nguin> What does ls -l /var/lib/asterisk/sounds/Disconnected_Out.* show? |
| 21:26.35 | beek | Gotcha. Nice guy, moved to Tenn from the Bahamas. |
| 21:26.47 | p3nguin | If you had it in the directory, it wouldn't say No such file. |
| 21:27.27 | p3nguin | sawgood: So please get your story straight and then give me the information I'm asking for in order to help you solve the problem. |
| 21:27.42 | sawgood | -rw-rw-r-- 1 asterisk asterisk 415K Nov 22 13:23 Disconnected_Out |
| 21:27.52 | sawgood | That is the file in /var/lib/asterisk/sounds/en |
| 21:27.53 | p3nguin | ls -l /var/lib/asterisk/sounds/Disconnected_Out.* says that? |
| 21:28.00 | jaytee | and be grateful that p3nguin doesn't have a ClueBat(tm) like TK does :-) |
| 21:28.08 | sawgood | here is the output of your command when I run it from / |
| 21:28.17 | beek | jaytee: So what have you been dabbling with of late (besides * and routers) |
| 21:28.31 | Qwell | ...wtf is the file extension? |
| 21:28.34 | jaytee | Exchange 2010 migration upgrades |
| 21:28.36 | p3nguin | sawgood: Your file does not have an extension to indicate to asterisk that it is a wav. |
| 21:28.38 | Qwell | of course it's going to say it doesn't exist in any format. |
| 21:28.53 | p3nguin | mv /var/lib/asterisk/sounds/Disconnected_Out /var/lib/asterisk/sounds/Disconnected_Out.wav |
| 21:28.57 | sawgood | I'll fix that |
| 21:28.59 | beek | suddenly has a sympathetic headache. |
| 21:29.04 | jaytee | :-) |
| 21:29.05 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:29.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:29.10 | p3nguin | and/or mv /var/lib/asterisk/sounds/en/Disconnected_Out /var/lib/asterisk/sounds/en/Disconnected_Out.wav |
| 21:29.30 | sawgood | works perfectly now! |
| 21:29.35 | sawgood | played the file with no problem |
| 21:29.37 | p3nguin | sawgood: There's no reason you needed to be sitting in / either. |
| 21:29.53 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:30.08 | p3nguin | sawgood: Now delete one and figure out if it plays from sounds/ or sounds/en/ |
| 21:30.52 | leifmadsen | libpri 1.4.11.5 and 1.4.12-beta3 are now available. Please see the release announcements at http://www.asterisk.org/node/51461 and http://www.asterisk.org/node/51462 respectively. |
| 21:30.56 | sawgood | yes, I will do that now |
| 21:31.17 | jaytee | I love that xkcd comic but this one is my all time favorite: http://xkcd.com/418/ |
| 21:31.57 | beek | :D |
| 21:33.48 | Ceeriael | Is there some kinf of fromUser_secret i can use in my Sip trunk ? |
| 21:34.10 | Ceeriael | i have username and fromuser (wich both have passwords, it works on my snom300, need it to work in asterisk) :( |
| 21:34.27 | sawgood | p3nguin: plays only if the file is in /var/lib/asterisk/sounds (and it needs to have .wav at the end of the file) |
| 21:34.31 | sawgood | Thank you for your help! |
| 21:35.14 | p3nguin | sawgood: So now you know where your sounds directory is. Good luck! |
| 21:36.04 | beek | jaytee: I'm trying to correlate a date to the correct xkcd.com that's one of my favorites. |
| 21:36.38 | jaytee | beek, so you know the date but not the comic number? |
| 21:36.49 | beek | jaytee, yes. |
| 21:37.03 | jaytee | what's the date? |
| 21:37.03 | beek | June 23 '10 |
| 21:37.56 | jaytee | this one? http://xkcd.com/756/ |
| 21:38.23 | p3nguin | sawgood: Asterisk reads the basename of the file (which you write as app data in dial plan), and then appends various file extensions as it looks for files to play using corresponding codecs. When it tries to play a ulaw file, it will be using yourfile.ulaw; when it wants to play a wav, it will use yourfile.wav. |
| 21:38.43 | *** join/#asterisk Alagar (~avaithiya@193.220.61.145) |
| 21:38.50 | beek | jaytee: next day. We actually used those things at our office for a fire alarm. |
| 21:40.32 | jaytee | hahaha! it was also worth it to me to carry around a very small bottle of Grey Poupon from the room service at a fancy hotel I stayed at. In came in perfectly handy on two occassions within a year. :-) |
| 21:41.14 | beek | there's a story, isn't there? |
| 21:42.15 | jaytee | just the face on someone whose jaw looks like it hit the floor when they think they're being clever and say, "Pardon me, but do you have any Grey Poupon?" and I whipped out the bottle and said, "But of course!" |
| 21:42.40 | beek | jaytee. That is excellent! |
| 21:42.58 | jaytee | I still have that bottle somewhere but I don't carry it anymore. |
| 21:43.33 | beek | That's hilarious. The reactions would have been priceless. |
| 21:43.35 | Qwell | jaytee: I used to carry scissors on me. Those got used quite frequently. |
| 21:45.33 | *** join/#asterisk anthm][ (~anthm@cpe-184-58-189-244.wi.res.rr.com) |
| 21:46.24 | beek | Qwell: I hope that you don't run with those! |
| 21:46.28 | Qwell | always |
| 21:47.22 | Qwell | they only saw blood once or twelve times |
| 21:47.57 | beek | Qwell: Your's or someone elses? |
| 21:48.02 | Qwell | yes |
| 21:53.10 | beek | thinks that Qwell was the type of kid to purpose color outside the lines. |
| 21:53.20 | beek | purposely |
| 21:55.28 | *** join/#asterisk chasecrum (~chase@adsl-065-082-196-004.sip.asm.bellsouth.net) |
| 21:55.41 | chasecrum | I've got a good one for you guys.... |
| 21:56.02 | chasecrum | Anyone ever clone an asterisk box while it was live ? with any success ? |
| 21:56.13 | p3nguin | of course. |
| 21:56.13 | thehar | chasecrum: what do you mean |
| 21:57.01 | chasecrum | I mean I've cloned zillions of servers in this lifetime, but I can't for the life of me clone the primary ast box here. And because it's the *primary* I can't take it offline to do it |
| 21:57.04 | p3nguin | dd or cp are both quite capable. |
| 21:57.15 | chasecrum | ok, lets walk through it |
| 21:57.29 | beek | gets his boots. |
| 21:57.31 | chasecrum | Should I install the base ast on the clone first and then copy over the partitions ? |
| 21:57.45 | p3nguin | That really doesn't make any sense. |
| 21:57.46 | thehar | copy over certain directories |
| 21:57.48 | chasecrum | the partitions seem to be whats kicking my butt here |
| 21:57.52 | thehar | specifically /etc/asterisk |
| 21:58.11 | p3nguin | You can isntall any OS you want, then just copy over the necessary directories and save a LOT of trouble. |
| 21:58.14 | Ceeriael | have you guys ever set up an SIP trunk with 2 passwords ? (one for username and one for authuser) ? |
| 21:58.16 | thehar | yup |
| 21:58.30 | thehar | ^_^ to what p3nguin said |
| 21:59.13 | chasecrum | hang on... brb |
| 22:02.01 | jaytee | beek, talk with ya later! gotta run. |
| 22:02.06 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:02.33 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 22:05.01 | *** join/#asterisk sbszulu (~dundubala@41.14.65.133) |
| 22:06.17 | Ceeriael | 2 set of usernames and 1 password in registry string, and 2 usernames and the other password in the outging context ;) |
| 22:09.25 | Alagar | hi any one can help me |
| 22:09.42 | Alagar | how to findout this line is trunk or not? |
| 22:11.16 | p3nguin | alagar: What line are you talking about? |
| 22:11.21 | frigidzephyr | Alagar: "trunk" its just a term used for various types of lines. typically meaning a connection from a customers system to a telecommunications provider, or connections between two sites |
| 22:11.36 | p3nguin | ~trunk |
| 22:11.36 | infobot | it has been said that trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 22:59.17 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:59.17 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 23:02.20 | *** join/#asterisk Alagar (~avaithiya@193.220.61.145) |
| 23:08.02 | bbeers | I am having interop problems with SRTP. who knows anything about it? |
| 23:23.16 | *** join/#asterisk infobot_ (~infobot@rikers.org) |
| 23:23.16 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 23:30.56 | Ceeriael | hi, im having problems with incomming calls. When the Sip trunk sends the Invite i challenge the Authentication. Ive tried using my Snom300 and it just seems to send a "ringing" back... how do i make my asterisk do the same ? |
| 23:33.47 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:33.47 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 23:38.47 | *** join/#asterisk luckman212_phone (~luckman21@mobile-166-137-138-054.mycingular.net) |