| 00:03.22 | securevoip | i believe sipgate is in only. ekiga doesn't appear to give you a number. |
| 00:04.17 | Dr1 | securevoip: does ekiga do outbound and do you think i could use SIPGate for inbound call testing and ekiga for outbound? |
| 00:06.31 | securevoip | ekiga only appears to let you call their numbers: 500, 501, 520, etc.; https://www.ekiga.net/index.php?page=services |
| 00:09.12 | *** join/#asterisk jpsharp (n=jsharp@24.224.45.160) |
| 00:09.40 | jpsharp | Can Asterisk relay a voicemail pending notify it gets from another server it is registered to? |
| 00:10.39 | Dr1 | securevoip: u think that's good enough for testing? are there no longer any ITSPs offering limited/reduced quality outbound service? |
| 00:11.59 | securevoip | Dr1: good enough for testing for sure... i don't know of any offering outbound but I am no guru... |
| 00:12.54 | Dr1 | securevoip: are there instructions or support for setting this up with asterisk? |
| 00:14.41 | securevoip | Dr1: http://www.sipgate.com/faq/show/article/id/394 |
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| 00:15.45 | securevoip | http://www.ipcomms.net/Products/ipcfaq.htm |
| 00:16.50 | Dr1 | securevoip: thank you very much, I'm going to give it a whirl |
| 00:17.04 | securevoip | Dr1: good luck! |
| 00:17.12 | [TK]D-Fender | Dr1: "testing" What does "testing" prove with any requirement of PSTN? |
| 00:21.23 | dwschool | where can i get a script for email to fax gateway for hylafax |
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| 00:42.52 | dshap | b14ck: hey what's up, u in here by any chance? |
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| 01:00.49 | dshap | can someone here help me with using the Record() application? how do i set up an extension to record a voicemail message so that the user can just hang up as opposed to pressing pound and or staying on the line for the rest of the extension to finish? |
| 01:01.05 | dshap | i'd like to make it so someone can call up, leave a message, and just hang up |
| 01:01.19 | dshap | and then the dialplan will continue to execute and correctly update the database and stuff |
| 01:03.52 | dshap | i need to process a recording when the user hangs up during the recording procedure |
| 01:03.57 | dshap | anyone know how to do this? |
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| 03:40.19 | WindowsUser | is there a list of non voice sounds? i see beep, beeperr silence/1 through 10 |
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| 03:50.07 | luch | if i run faxmail to take apart an email message, how do i get information to use in sendfax from faxmail form hylafax |
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| 03:57.13 | ricko73 | wonders when this became the hylafax support forum? |
| 03:59.36 | Nugget | next I'm going to get everyone to teach me how to install linux. |
| 03:59.57 | WindowsUser | fromfromfrom |
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| 04:50.53 | the_unknown | so... where might I look to find assistance getting a cisco 7940 to run the sip firmware |
| 04:51.24 | WindowsUser | cisco forum? general voip forum? |
| 04:52.03 | WindowsUser | aquiring the firmware for a cisco device is usually the hardest part |
| 04:52.19 | the_unknown | I have the firmware |
| 04:52.36 | the_unknown | and i'm following the instructions from cisco |
| 04:52.48 | WindowsUser | so where does everything go wrong? |
| 04:54.17 | the_unknown | it starts looking for files SEP and CTLSEP <mac> (which fails obviously,) it then loads the .cnf files like it should, but starts looking for a .sbn with a from a different version |
| 04:54.33 | the_unknown | I'm realizing that made very little sense |
| 04:54.59 | WindowsUser | is the version the current firmware version thats on the thing? |
| 04:55.22 | the_unknown | yes |
| 04:55.25 | the_unknown | rather |
| 04:55.28 | Nugget | make sure your.conf and the OS9XX file agree on which version is current |
| 04:55.41 | the_unknown | the sbn is the current version |
| 04:55.48 | the_unknown | the OS97XX has the right file |
| 04:55.54 | Nugget | and the .cnf? |
| 04:56.02 | Nugget | and SIPDefault? |
| 04:56.33 | the_unknown | SIPDefault has the version i'm trying to load |
| 04:56.33 | Nugget | it's been a while since I dealt with a 79x0 but I remember it can easily get into a death loop of conflicting version loads if any files disagree with what load to use |
| 04:56.36 | the_unknown | as does OS97XX |
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| 05:01.38 | suseGeekTW | Do I need a trunk so that I can use x-lite connecting to server to make a phone call from x-lite out to PSTN. |
| 05:02.38 | WindowsUser | x-lite would be a SIP user |
| 05:03.00 | WindowsUser | whats going from asterisk -> PSTN? SIP provider or a card of some kind? |
| 05:03.33 | suseGeekTW | actually I am using Elastix but I guess I can get some help here, too. |
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| 05:09.30 | Dr1 | "Access Denied" copying to D:\<subdir> as admin -- removed all permissions and gave admin & administrators FC and took ownership and rolled to subdirectories -- what gives? |
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| 05:17.32 | [TK]D-Fender | Dr1: Pardon? |
| 05:20.15 | carrar | looks at the channel name we're in of #Windows_Admin_Support |
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| 05:38.08 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 05:38.08 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25.1 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.2.0, dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
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| 06:01.01 | WindowsUser | carrar: there is a ##windows on freenode |
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| 08:37.15 | cjk | hi, i use asterisk as a pri gateway and i noticed that i get "Ext: 1 Channel: 28" on span 1 but i send Ext: 1 Channel: 1" on span 2. any idea how to set/change this "Channel" parameter |
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| 09:06.32 | joobie | sup ladies |
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| 09:50.07 | dexteruk | question about asterisknow if you have a PRI with 200 number you use the Incomming call routing to map them to the extenstions |
| 09:50.42 | dexteruk | you choice the trunk and then it says EXTEN: and it looking for a number |
| 09:50.55 | dexteruk | I take it that number is the extenstion you want the call to goto |
| 09:51.27 | dexteruk | so your did 3333100 goto 100 have i got that right? |
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| 09:54.42 | joobie | dexteruk, i find it really hard to understand you |
| 09:54.45 | joobie | maybe it's me.. |
| 09:55.19 | dexteruk | No its probably me :-) i havent got it infrount of me thats why im a little vag |
| 09:55.34 | dexteruk | let me see if i can get connected and i will tell you the exact options |
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| 10:11.20 | |dexteruk| | Hi joobie |
| 10:11.27 | |dexteruk| | I just got access |
| 10:11.47 | |dexteruk| | Ok we have a PRI installed with 200 channels |
| 10:12.12 | |dexteruk| | we want to set the first did 3333100 to ring extenstion 100 |
| 10:12.29 | |dexteruk| | so I went to Incomming Call routing |
| 10:12.58 | |dexteruk| | there is an option in there Local Extenstion by DID |
| 10:13.31 | joobie | something like.. exten => 3333100,1,Dial(SIP/100,30) exten => 3333100,n,Hangup |
| 10:13.41 | |dexteruk| | yes |
| 10:13.51 | |dexteruk| | exactly, i know how to do it myself |
| 10:14.01 | |dexteruk| | but throught the gui im not sure :-D |
| 10:14.18 | joobie | ahh |
| 10:14.21 | joobie | sry, i dont use the gui |
| 10:14.43 | |dexteruk| | me neither :-D I just have a client that does |
| 10:14.45 | joobie | doesnt the gui just amend the files? |
| 10:14.59 | |dexteruk| | yes but it all macros |
| 10:15.11 | joobie | so check the macros |
| 10:15.16 | joobie | see that they are all setup right |
| 10:15.18 | joobie | from the files |
| 10:15.20 | |dexteruk| | and it makes me look stupid why i try to understand whats going on |
| 10:15.25 | joobie | ahh |
| 10:15.27 | joobie | duno man |
| 10:15.33 | joobie | i dont know the gui |
| 10:15.48 | joobie | so if it was me, i'd try understand the files and figure out what's missing, then go to the gui to add it? |
| 10:15.49 | |dexteruk| | no problem, no not many people do :-D it seems |
| 10:15.50 | |dexteruk| | hehehehe |
| 10:15.58 | joobie | well |
| 10:16.03 | joobie | there are probably asteriskgui chans |
| 10:16.11 | |dexteruk| | reall |
| 10:16.20 | joobie | most of the people here (if not all) that i see usually talk about CLI |
| 10:16.26 | joobie | yea |
| 10:16.31 | |dexteruk| | ok i will check it out |
| 10:16.37 | |dexteruk| | than for the info |
| 10:16.41 | |dexteruk| | thanks |
| 10:16.46 | joobie | there is #asterisknow |
| 10:16.49 | joobie | just not as many users |
| 10:16.56 | joobie | probably because gui's are gay |
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| 10:17.09 | joobie | and there are fewer gay than straight people |
| 10:17.27 | |dexteruk| | Well being gay I would be so sure about that |
| 10:17.31 | |dexteruk| | ;-) |
| 10:18.46 | |dexteruk| | but thanks for the infro anyhow.. I know we get everywhere these days ;-) |
| 10:19.25 | joobie | cool man |
| 10:20.23 | zeeesh | how to check gtalk is enabled ? |
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| 10:28.45 | BugKhaM | I am having a problem where "AGI GET DATA" returns fialure with "200 result=-1" and then it AGI stops its execution and hangs up |
| 10:29.20 | BugKhaM | is there a way to have it continue the execution like wen it times out |
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| 10:52.45 | mort_gib | Anyone have time to give a hand with a dahdi issue?? |
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| 11:12.02 | ickmund | Anyone with an Aastra 51i know how to setup the Directory key to do a server call? |
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| 11:28.47 | lftsy | Hello all, I was using Asterisk 1.4.25.1 and I was affected by the chan_sip.c bug, with a seg fault that has been corrected on svn revision 201380 and I was wondering if it's better to apply only this patch to my 1.4.25.1 version or to use 1.4.26-r4 which is not a final revision? What would you recommand me? |
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| 12:06.50 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:43.37 | *** join/#asterisk pisco (n=pisco@cl-1305.dus-01.de.sixxs.net) |
| 12:44.55 | pisco | hello folks, anyone experienced with res_ldap? |
| 12:45.58 | pisco | the sip clients from the ldap directory can connect and echotest works, but from one client to an other gives dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 12:46.44 | pisco | any hints how fix this behavior? |
| 12:49.00 | pisco | connectivity itself is ok, I checked successful with 'traditional' entries in the text files. |
| 12:49.07 | *** join/#asterisk intralanman (n=lanman@173-132-22-66.pools.spcsdns.net) |
| 12:50.29 | [TK]D-Fender | pisco: Go look at the actual peer you are trying to dial. SIP DEBUG is your friend.... and when in doubt, so is PASTEBIN |
| 12:50.50 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:51.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:55.43 | pisco | [TK]D-Fender: http://paste2.org/p/298514 |
| 12:56.32 | [TK]D-Fender | pisco: "sip show peer user2" <- |
| 12:57.01 | pisco | user2 is also located at the 192.168.0.0/24 net |
| 12:57.21 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 12:57.36 | [TK]D-Fender | pisco: "sip show peer user2" <- |
| 12:57.40 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 12:58.10 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 12:59.21 | pisco | as far as i know in a realtime setup the command should be 'sip show peer user2 load' http://paste2.org/p/298518 |
| 12:59.48 | pisco | otherwise 'sip show peer user2' is empty |
| 13:00.17 | [TK]D-Fender | pisco: Addr->IP : (Unspecified) Port 0 <--- the phon hasn't registered and * has nowhere to call to reach this peer |
| 13:00.24 | *** join/#asterisk jicksta (n=jicksta@c-67-169-183-106.hsd1.ca.comcast.net) |
| 13:04.57 | pisco | looks like something is wrong in the registration process http://paste2.org/p/298522 |
| 13:05.50 | [TK]D-Fender | pisco: Bad LDAP setup so * can't store the reg contact. |
| 13:06.40 | *** join/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net) |
| 13:07.06 | *** join/#asterisk LemensTS (n=customgt@207.160.41.191) |
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| 13:10.07 | pisco | [TK]D-Fender: finally the Addr->IP should hold the peer ID, right? |
| 13:10.20 | pisco | s/ID/IP |
| 13:10.29 | [TK]D-Fender | pisco: and rport |
| 13:11.38 | LemensTS | hey all. im using phpagi, and i have a file that i run that uses the originate command. First it calls the caller, when they answer it connects them to something in my extensions.conf....If they do not answer tho, it never makes it to extensions.conf. But i get response=Success and Message=Originate successfully queued no matter if they answer or not to my bash....the problem im having is that it has 3 tries to make the call, but it tries to call it w |
| 13:12.32 | LemensTS | id mark it as inuse right before it does the originate cmd btw... |
| 13:18.20 | LemensTS | http://pastebin.com/m6f9a9694 i need to do the mysql query to update the inuse to 'no' if the call is not answered....i dont know how to tell if the call is not answered tho |
| 13:19.17 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:19.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:22.38 | rudeboy_xix | so u've build a class for the asterisk manager api |
| 13:23.34 | rudeboy_xix | LemensTS: i think that's not an agi |
| 13:24.11 | LemensTS | i dont think i can do it like i was thinking, since it sends it off to the originate spool. |
| 13:24.40 | *** join/#asterisk ShurV (n=ShurV@62-90-161-33.barak.net.il) |
| 13:24.44 | ShurV | Hello! |
| 13:24.53 | LemensTS | i can mark the timestamp of it sending it, then if that number comes up again see if it is 5 minutes within calling it last time before calling again. that would work |
| 13:25.13 | ShurV | Could you tell me please how to supress 'Remote UNIX connection' messages in CLI ? |
| 13:25.22 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 13:25.36 | LemensTS | why? |
| 13:25.48 | [TK]D-Fender | ShurV: reduce you verbose to "useless" levels or mod the source |
| 13:26.02 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:26.21 | ShurV | I have a script that checks every second thye channels tabel and make a soft hangup in some cases |
| 13:26.48 | LemensTS | ok cool just curious |
| 13:27.32 | beek | What is the general conversion factor for rings to seconds on a dial statement? If someone wants 'three rings', how much time is that? IIRC, I saw someone say five seconds per ring... |
| 13:28.24 | rudeboy_xix | ShurV: i dont understand the purpose of your script |
| 13:28.42 | [TK]D-Fender | beek: 15s/4r |
| 13:28.55 | [TK]D-Fender | rudyCheck for lingering channels. |
| 13:28.59 | beek | [TK]D-Fender: Thanks |
| 13:31.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:33.32 | joobie | http://xkcd.com/327/ |
| 13:33.36 | joobie | tee hee :) |
| 13:33.37 | joobie | g'night |
| 13:33.39 | Jumpie | hi guys, one of my customers is getting occaisional echo that only she can hear, form callers calling into her |
| 13:33.49 | Jumpie | its random and hard to see if its the callers connection/provider or what |
| 13:33.56 | Jumpie | whats a good way to really nail this down? its not common |
| 13:34.09 | *** join/#asterisk salvia (n=aurax@bzq-218-145-213.cablep.bezeqint.net) |
| 13:34.23 | salvia | I just finished connecting two asterisk machines via AIX trunk, dialplan is 8|XXX to this trunk. but the user i call to see my internal extension number, any way to add 8 ? |
| 13:34.29 | salvia | can anyone help me out ? |
| 13:34.33 | florz | Jumpie: it is the caller's connection or telephone. |
| 13:34.47 | *** join/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 13:35.41 | ShurV | [TK]D-Fender: thank you, 2 is sufficient |
| 13:36.04 | zeeesh | calling from sip client to gtalk user,call is connecting, as well getting echo, just sip user hearing his own voice other end is unable to hear anything ? |
| 13:36.17 | salvia | any idea folks ? |
| 13:36.30 | [TK]D-Fender | salvia: FreePBX is NOT supported here, ask in their channel please. |
| 13:36.56 | aethelrick | hi all, I've written some code to integrate our CRM and CMS with asterisk, I would like to know what the most efficient way to query asterisk is to get status of existing channels |
| 13:37.07 | salvia | alright :) |
| 13:37.55 | aethelrick | I'm using AMI at the moment to drive asterisk from the outside and I see methods there to find out channel status, but I'm looking for something cheap, like maybe surfing sa log or similar so that I don't over-tax the asterisk box |
| 13:38.11 | aethelrick | any ideas? |
| 13:39.52 | [TK]D-Fender | aethelrick: AMI is what you've got. Either poll the channel events or do a mass polling on interval |
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| 13:42.18 | tzafrir_laptop | salvia, you have Asterisk running on AIX? nice :-) |
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| 13:44.17 | aethelrick | [TK]D-Fender: thanks for the response. I was theorizing that maybe I could possibly surf a log somewhere, but not sure if the right detail is logged and if so where |
| 13:48.42 | [TK]D-Fender | aethelrick: Logs don't show open channels |
| 13:48.53 | [TK]D-Fender | aethelrick: And parsing that is horrific |
| 13:49.00 | *** join/#asterisk ariel_ (n=chatzill@198.211.95.37) |
| 13:49.34 | ariel_ | hello everyone |
| 13:50.40 | aethelrick | thanks again :) AMI it is then... (which is good because that's what my code does now) |
| 13:50.47 | Jumpie | florz |
| 13:50.51 | Jumpie | its aastra 55i |
| 13:50.55 | Jumpie | so far been really good |
| 13:51.16 | Jumpie | anyway i can verify? |
| 13:51.21 | pisco | [TK]D-Fender: with the help of the ldap log I found the issue. Regarding to https://issues.asterisk.org/view.php?id=13725 the ldap schema doesn't ship a field for AstAccountUserAgent, thats why the registration fails |
| 13:51.33 | *** part/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 13:51.47 | pisco | [TK]D-Fender: thank for pointing me to the registration ... |
| 13:51.59 | [TK]D-Fender | pisco: You're welcome |
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| 13:55.13 | Katty | joy. a co-worker has caught menegitis |
| 13:56.44 | Katty | worries. |
| 14:01.30 | *** join/#asterisk lipatden (n=ldennehy@sbpofw1.schubergphilis.com) |
| 14:02.41 | lipatden | Hi all, I'm having trouble figuring out what steps I should take to get my SIP connections working, the only guides I can find are ground-up (1. locate the source and dependent libraries) or maddeningly complex (installing call cards etc). |
| 14:03.14 | lipatden | I have an account with a VOIP provider (actually more than one), and one VOIP device at my home. Where's a good place to get started on hooking these guys up via asterisk? |
| 14:04.02 | SuPrSluG | ~book |
| 14:04.03 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:04.41 | SuPrSluG | follow the instructions in the book |
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| 14:06.10 | SuPrSluG | and you should be up and running in a couple of hours |
| 14:06.31 | [TK]D-Fender | lol |
| 14:06.55 | [TK]D-Fender | SuPrSluG: Wouldn't put down something that solid so quick... |
| 14:07.18 | [TK]D-Fender | lipatden: read this for some "inspiration on what a sinple setup might look like : |
| 14:07.21 | [TK]D-Fender | ~jerjerguide |
| 14:07.22 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 14:07.22 | SuPrSluG | optimist |
| 14:08.31 | lipatden | [TK]D-Fender: Ah ha! That looks like just what I want! |
| 14:09.11 | lipatden | And thanks for the online book, obviously a desk reference is crucial in these sorts of things, so I'll definitely my thumbing (virtually) through that thing |
| 14:10.03 | lipatden | I'm in the Netherlands, and it would appear a lot of these guides use american dialling rules |
| 14:10.25 | [TK]D-Fender | lipatden: TFOT isn't so much a reference (though it does have appendices for this) as it is an outline of the various pieces of * |
| 14:10.43 | lipatden | We dial internationally using 00(countrycode), so 0044... for UK, 001... for USA/Canada, I'm assuming I replace _1N with _0N for international in this case? |
| 14:10.55 | SuPrSluG | N=2-9 X=0-9 adjust for your dialing pattern |
| 14:11.19 | [TK]D-Fender | lipatden: Dialplan patterns is something you need to master and is core. * = %90 Dialplan, 10% other |
| 14:11.39 | lipatden | [TK]D-Fender: Exactly, which is perhaps why I'm not getting it, I don't live in the USA |
| 14:12.17 | [TK]D-Fender | lipatden: Well the concept of patterns is well documented. Look at the numbers you dial and you should know which is which |
| 14:12.32 | lipatden | E.g., "exten => _NXXNXXXXXX", what kind of call would that match in North America? Local (same area-code) or long-distance (inside USA)? |
| 14:13.06 | [TK]D-Fender | lipatden: Not necessarily long-distance |
| 14:13.22 | lipatden | Urgh :S |
| 14:13.57 | [TK]D-Fender | lipatden: Just includes an area code. Many ITSP's force you to dial 11-digit as 1NXXNXXXXXX Even though they are not all LD to you. |
| 14:14.23 | [TK]D-Fender | ~nanpa |
| 14:14.24 | infobot | from memory, nanpa is North American Numbering Plan Administration; the organization responsible for administering the integrated telephone numbering plan serving 19 North American countries. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.com/ |
| 14:14.27 | [TK]D-Fender | ^^^^^ |
| 14:16.45 | lipatden | [TK]D-Fender: Ah, that much I get, since VoIP is non-geographic, I actually use a UK VoIP account from the Netherlands to receive inbound calls from landlines at (UK) local rates, but again, the examples in pretty much all the documentation is US-specific. I just need a numeric translation to arrive at generic rules. |
| 14:17.18 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:17.23 | lipatden | Another question, "sip show peers" shows my currently described peers in sip.conf, how do I know if the connection is active? |
| 14:17.32 | [TK]D-Fender | lipatden: go read the book and play around |
| 14:17.42 | lipatden | np, thanks for the help |
| 14:18.52 | [TK]D-Fender | lipatden: there isn't such a thing as "active" really. "show channels" will show you want actual calls are in progress. "sip show peer [peername]" will show you if a device is registered to that peer (for phones, not ITSPs, etc) |
| 14:22.49 | *** join/#asterisk blkry (n=blkry@mail.firstasheville.com) |
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| 14:27.08 | lipatden | That makes sense, really I just wanted way to look at the status without having to tail a logfile |
| 14:27.39 | [TK]D-Fender | lipatden: Logs are pretty worthless. the best wayt o check depends on when & why you want to. |
| 14:29.19 | nkohh | lipatden: terminal monitor |
| 14:30.08 | lipatden | nkohh: Oh yes? a simple command-line to get in I assume? |
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| 14:31.24 | *** part/#asterisk tfrew (n=tfrew@m495a36d0.tmodns.net) |
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| 14:36.21 | blkry | How I send callers to my voicemail while the the phone is still ringing. I know I don't want to talk with them. I can reject it but the ring group continues to ring. |
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| 14:45.55 | [TK]D-Fender | blkry: Nothing you can do with FreePBX |
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| 15:00.48 | blkry | OK thank you |
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| 15:07.08 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:20.38 | Katty | [TK]D-Fender: oh goto if time |
| 15:20.42 | Katty | [TK]D-Fender: what's Thursday? |
| 15:20.48 | Katty | [TK]D-Fender: the abbreviation |
| 15:21.31 | jaytee | thu |
| 15:21.33 | AlmightyOatmeal | lol |
| 15:21.54 | Katty | jaytee: shanks. |
| 15:22.19 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:23.18 | jaytee | Katty: velcum |
| 15:23.33 | zeeesh | sip client to gtalk, call is connecting . but there is no voice at both end ? could not found any error msg at console, can anybody has any clue ? |
| 15:23.50 | [TK]D-Fender | Katty: thu |
| 15:27.00 | carrar | that comes after fri? |
| 15:27.19 | AlmightyOatmeal | your mom |
| 15:27.22 | AlmightyOatmeal | (or sat) |
| 15:27.53 | carrar | Saturday is earth day |
| 15:28.12 | carrar | douyoubi |
| 15:28.23 | carrar | so yeah |
| 15:28.29 | carrar | mother earth |
| 15:28.30 | carrar | could be |
| 15:28.42 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 15:29.35 | [TK]D-Fender | carrar: Mother Earth huh..... guess that means you're either a FRUIT or a VEGETABLE :p |
| 15:29.42 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 15:29.58 | carrar | haha |
| 15:30.11 | carrar | I'm a OAK |
| 15:30.27 | rudeboy_xix | im testing the AMI. im using the originate action. is it posible to answer the call automatic? |
| 15:30.33 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:30.47 | [TK]D-Fender | carrar: Witches are made of wood... since you'll float we'll need to burn you at the stake now :P |
| 15:31.07 | [TK]D-Fender | rudeboy_xix: Doepends what you're CALLING |
| 15:31.14 | Katty | [TK]D-Fender: apparently the co-worker's test came back negative for meningitis. |
| 15:31.25 | Katty | [TK]D-Fender: i'm so relieved i'm crying. |
| 15:31.27 | hi365 | rudeboy_xix: one way to do that is to start your dialplan with Answer() |
| 15:31.37 | rudeboy_xix | ohh, yah |
| 15:35.17 | rudeboy_xix | do i have to use wait, to make a ring first or if i use that it will still ring |
| 15:39.52 | lipatden | Hmmm, "username mismatch" trying to pass a call from my sip device to a second * server |
| 15:40.00 | [TK]D-Fender | rudeboy_xix: If you answer, ringing will stop |
| 15:40.51 | rudeboy_xix | i mean on the dialplan |
| 15:42.59 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 15:46.28 | andres833 | hi pals |
| 15:46.31 | [TK]D-Fender | rudeboy_xix: Same answer |
| 15:46.51 | andres833 | i need help with a fax |
| 15:47.35 | rudeboy_xix | ok so i have to insert a wait before the answer |
| 15:48.08 | tzafrir_laptop | andres833, is it heavy? |
| 15:49.00 | *** join/#asterisk chrisb (n=chrisb@pool-98-111-169-71.phlapa.east.verizon.net) |
| 15:50.09 | andres833 | tzafrir_laptop, thanks but was a problem with internet :P |
| 15:50.17 | *** join/#asterisk rgsteele||work (n=rgsteele@207.106.239.76) |
| 15:50.35 | andres833 | tzafrir_laptop, i am installing digium free fax fora sterisk |
| 15:51.08 | tzafrir_laptop | i try to avoid proprietary software |
| 15:52.07 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 15:53.32 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 15:54.54 | eppigy | 8[] |
| 15:54.55 | [T]ank | where in the configs does it specify what directory voicemail messages are stored in? I am looking at an install of asterisk 1.4.21.2 |
| 15:55.05 | [T]ank | on ubuntu |
| 15:55.10 | [T]ank | and it doesnt have the usual directory |
| 15:55.12 | chrisb | anyone from xmission here? |
| 15:55.23 | [TK]D-Fender | [T]ank: asterisk.conf <- |
| 15:55.30 | [T]ank | ahh.. thats right. |
| 15:55.31 | [T]ank | thanks |
| 15:58.53 | andres833 | tzafrir_laptop, i am using a spanish guie |
| 15:59.06 | andres833 | tzafrir_laptop, http://voztovoice.org/?q=node/177 |
| 16:00.15 | tzafrir_laptop | andres833, as I mentioned, that specific product is not my cup of tea. I suggest you ask others |
| 16:01.18 | andres833 | tzafrir_laptop, ok thanks |
| 16:01.29 | andres833 | tzafrir_laptop, sorry |
| 16:01.48 | tzafrir_laptop | [T]ank, /var/spool/asterisk/voicmail , as usual |
| 16:04.16 | mltlnx | <PROTECTED> |
| 16:06.39 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 16:07.22 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:08.05 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 16:08.58 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:10.23 | *** join/#asterisk hi365 (n=hi365@94.159.178.170) |
| 16:11.13 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 16:12.04 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:14.19 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
| 16:16.40 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:16.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:18.49 | lipatden | I'm having a problem trying to get a SIP connection through two * servers. |
| 16:19.26 | lipatden | I have a SIP phone (windows Messenger), registered with server1, server1's plan forwards the call to server2, which has a plan to dial out using my VoIP provider |
| 16:19.54 | lipatden | As soon as I try to dial, server1 complains 'Failed to authenticate on INVITE...' |
| 16:19.58 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 16:20.05 | lipatden | any clues where to start? |
| 16:25.58 | lipatden | Anyone? I've got a really simple setup here, this should be working |
| 16:28.49 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 16:29.45 | *** join/#asterisk ultrav1olet (n=forgotte@92.255.141.115) |
| 16:29.56 | ultrav1olet | are there any asterisk developers here? |
| 16:30.16 | ultrav1olet | we have a terrible problem since upgrading from asterisk 1.4 to 1.6 |
| 16:30.49 | ultrav1olet | Our asterisk is connected to a usual telephone landline via Wildcard TDM400P REV E/F Board 5 |
| 16:31.12 | ultrav1olet | Since we upgraded asterisk in 50% of cases it dials a wrong number |
| 16:31.23 | ultrav1olet | I have no idea why it is happening |
| 16:31.36 | ultrav1olet | this wasn't a problem with asterisk 1.4 |
| 16:33.35 | jameswf | someone may find this useful.. http://dontcallmyboss.com/projects/dahdi-compat links old commands to their replacements with an added kick in the head... |
| 16:34.05 | ultrav1olet | any ideas? |
| 16:35.46 | vAd0r | what file do i edit for dtmf |
| 16:36.45 | jameswf | ultrav1olet: add a w to your dial ex dial(DAHDI/g0/w18004664411) |
| 16:37.02 | ultrav1olet | already tried that |
| 16:37.14 | ultrav1olet | I've even added three w before dialing |
| 16:37.15 | jameswf | add 2 w's |
| 16:37.18 | ultrav1olet | like www |
| 16:37.20 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 16:37.24 | ultrav1olet | nothing has really changed |
| 16:37.25 | jameswf | lol ummm nm |
| 16:37.52 | ultrav1olet | I'm absolutely disappointed |
| 16:38.04 | jameswf | what number are you dialing what number gets dialed |
| 16:38.29 | ultrav1olet | whatever number I dial, I can get to a absolutely random number |
| 16:38.44 | ultrav1olet | like I call 12345678 and get to 792167312 |
| 16:39.21 | ultrav1olet | I've just ugraded dahdi linux modules to the latest stable version to no avail |
| 16:40.32 | ultrav1olet | I now wonder if it's worth trying a pulse dial mode |
| 16:41.01 | ultrav1olet | BTW, a usual phone connected to the same landline always calls "right" numbers |
| 16:41.59 | ultrav1olet | where should I put "pulsedial=yes" with dahdi? there's no file zapata.conf :( |
| 16:42.36 | [TK]D-Fender | ultrav1olet: And I don't see you providing a pastebin of the failed call with full debug, your configs, detailed version info on what you're running, etc. |
| 16:44.27 | ultrav1olet | [TK]D-Fender: what are the exact files and debug logs you'd like to see? |
| 16:44.50 | [TK]D-Fender | ultrav1olet: CLI of the failure at verbose 10, dahdi configs, etc |
| 16:45.08 | ultrav1olet | I'm running asterisk 1.6.0.9 with dahdi-linux-2.2.0.1 |
| 16:45.17 | ultrav1olet | wait a minute |
| 16:46.28 | [TK]D-Fender | ~topic |
| 16:46.49 | [TK]D-Fender | hrm |
| 16:48.30 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 16:49.11 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25.1 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.2.0.1 (2009/06/30), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 16:49.39 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 16:54.45 | ultrav1olet | http://pastebin.ca/1482081 |
| 16:54.54 | ultrav1olet | this time it hit the wrong number |
| 16:55.26 | ultrav1olet | next time with *the absolutely same* log output, asterisk called the right number |
| 16:56.20 | ultrav1olet | as I've said in roughly 50% of cases I hit absolutely *random* number |
| 16:56.59 | ultrav1olet | BTW busydetect=yes doesn't work at all :( |
| 16:57.09 | ultrav1olet | and hanguponpolarityswitch=yes either |
| 16:57.12 | [TK]D-Fender | ultrav1olet: What number did it actually dial there? And your rxgain is PSYCHO |
| 16:57.34 | *** part/#asterisk LemensTS (n=customgt@207.160.41.191) |
| 16:57.42 | [TK]D-Fender | ultrav1olet: -- Executing [89028305702@g_users:1] Dial("IAX2/birdie-14205", "DAHDI/1/89028305702") in new stack <-- so you saying it didn't actually dial "89028305702"? |
| 16:57.55 | ultrav1olet | yes, that's what I'm saying |
| 16:58.51 | [TK]D-Fender | ultrav1olet: Do a dial without passing a number and dial the digits manually. Also check how long it takes to get a tone |
| 16:59.04 | [TK]D-Fender | ultrav1olet: And your gains are still psycho |
| 16:59.20 | *** join/#asterisk [acer]lanman (n=lanman@67.76.163.226) |
| 16:59.25 | ultrav1olet | you mean fxorxgain=25? |
| 16:59.51 | ultrav1olet | our asterisk is very quiet, so I increased fxorxgain a bit ;) |
| 17:00.06 | ultrav1olet | now I can actually can hear the party I'm calling |
| 17:00.26 | ultrav1olet | how can I dial manually? Can you explain that part? |
| 17:00.43 | ultrav1olet | I'm using zoiper - I have no idea how to do that |
| 17:02.21 | [TK]D-Fender | [12:58]<[TK]D-Fender>ultrav1olet: Do a dial without passing a number and dial the digits manually. Also check how long it takes to get a tone |
| 17:02.39 | [TK]D-Fender | ultrav1olet: Remove the 3 from what is dialed and DTMF it YOURSELF |
| 17:03.03 | ultrav1olet | "Remove the 3 from what is dialed" - Sorry, I don't get it :( |
| 17:05.46 | [TK]D-Fender | ultrav1olet: AHDI/1/89028305702 <----- remove the bloody PHONE NUMBER that follows that "/" |
| 17:07.45 | ultrav1olet | you mean removing everything in exten => ? |
| 17:08.23 | [TK]D-Fender | ultrav1olet: What is difficulat about this? Dial the dahi channel and don't pass it a number! |
| 17:08.35 | [TK]D-Fender | DAHDI* |
| 17:09.19 | ultrav1olet | I get it :) |
| 17:09.48 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 17:09.56 | ultrav1olet | [TK]D-Fender: but I need an empty extension for that, right? |
| 17:10.08 | [TK]D-Fender | ultrav1olet: you need any exten you can reach that will do this. |
| 17:10.25 | ultrav1olet | OK |
| 17:10.27 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:12.05 | ultrav1olet | I have added exten => 0,1,Dial(DAHDI/1/) |
| 17:12.24 | ultrav1olet | then called 0, then dialed manually my own number. I hit the wrong number again :( |
| 17:12.48 | [TK]D-Fender | ultrav1olet: I'm betting gains are distorting your dtmf. |
| 17:13.09 | ultrav1olet | ok, let's reload the module |
| 17:13.12 | [TK]D-Fender | ultrav1olet: Make an IVR and dial in and see how well detection works |
| 17:14.00 | suma | why I'm not finding vxml application on my asterisk ? |
| 17:14.17 | ultrav1olet | I'm sorry what IVR is? |
| 17:14.31 | ultrav1olet | and what detection you are talking about? |
| 17:15.50 | [TK]D-Fender | ..... |
| 17:15.52 | [TK]D-Fender | ~ivr |
| 17:15.52 | infobot | from memory, ivr is Interactive Voice Response |
| 17:16.07 | [TK]D-Fender | ultrav1olet: You know.. AUTOATTENDANT. * answers, prompts for INPUT? |
| 17:16.36 | [TK]D-Fender | suma: AFAIK * doesn't COME with one. |
| 17:17.34 | suma | [TK]D-Fender: http://www.voip-info.org/wiki/view/Asterisk+cmd+Vxml |
| 17:17.42 | suma | says there is Vxml |
| 17:17.56 | suma | i don't know where to grab the source from |
| 17:18.01 | ultrav1olet | [TK]D-Fender: we don't use/have AUTOATTENDANT to my knowledge ... |
| 17:18.05 | Corydon76-dig | It's a Wiki... ANYBODY can edit it |
| 17:18.12 | ultrav1olet | where have you spotted it? |
| 17:18.27 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 17:18.56 | Corydon76-dig | I can add a page to the Wiki that says usage of Asterisk is limited to those who believe in the Flying Spaghetti Monster -- doesn't mean that it actually applies to Asterisk |
| 17:19.09 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 17:19.14 | [TK]D-Fender | ultrav1olet: [13:13]<[TK]D-Fender>ultrav1olet: Make an IVR and dial in and see how well detection works <---- WAKE UP! I said go MAKE ONE. get off your ass and MAKE a friggen menu that will take single digit output and read it BACK to you as proof that * is even WORKING. |
| 17:19.21 | suma | Corydon76-dig: I discuss with kpfleiming in the asterisk-dev he was saying about it |
| 17:19.43 | suma | *discussed |
| 17:19.59 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 17:20.16 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 17:21.02 | suma | Is there a way to integrate vxml with asterisk as of now ? |
| 17:21.13 | Corydon76-dig | suma: I read the same conversation, and I don't see where Kevin said anything about vxml already being in Asterisk |
| 17:22.17 | Corydon76-dig | suma: of course there is, but it probably requires you to write code |
| 17:22.26 | ultrav1olet | [TK]D-Fender: I'm sorry |
| 17:22.34 | Corydon76-dig | vxml is a technology, not a product |
| 17:22.36 | ultrav1olet | [TK]D-Fender: will try to do that |
| 17:23.05 | suma | Corydon76-dig: I was in an assumption that is already in |
| 17:23.22 | Corydon76-dig | suma: then you are operating under a false assumption |
| 17:24.04 | suma | Corydon76-dig: yes |
| 17:25.20 | [TK]D-Fender | ultrav1olet: tone sounds horribly broken |
| 17:25.25 | [TK]D-Fender | ultrav1olet: check your lines |
| 17:26.21 | ultrav1olet | you think that's the problem? |
| 17:26.37 | ultrav1olet | the usual phone connected to the same landline works just fine ... |
| 17:26.39 | [TK]D-Fender | ultrav1olet: plug a normal phone on them to test. Next you may have to restart for your gain change to take place. Also those starting tones sound like 1/2 the DTMF isn't making it |
| 17:26.41 | ManxPower | "Dear, "http://www.i6net.com/" provides voicexml browser for asterisk." |
| 17:26.48 | ManxPower | I don't know anything about it. |
| 17:27.17 | ManxPower | [TK]D-Fender: zaptel gains are adjusted on the fly |
| 17:27.28 | [TK]D-Fender | ManxPower: Apparently the "big print" needs a giant flashing neon sign pointing towards it |
| 17:27.36 | ultrav1olet | > Also those starting tones sound like 1/2 the DTMF isn't making it |
| 17:27.52 | ultrav1olet | I'm dialing using zoiper and tones sound just fine to me |
| 17:28.03 | [TK]D-Fender | ManxPower: on the fly? rxgain,e tc sure aren't. Also he's doing his at the module level, not even in configs. |
| 17:28.14 | ManxPower | [TK]D-Fender: When you reload chan_zap.so (or chan_dahdi.so) a list of options that were not processed is provided. |
| 17:28.21 | [TK]D-Fender | ultrav1olet: "cat /proc/interrupts" <- |
| 17:28.41 | ManxPower | the rxgain and txgain are not listed when you do the reload. I've personally adjusted gains on the fly on zap channels |
| 17:28.47 | [TK]D-Fender | ManxPower: Yeah, if you reload the channel-driver, however his gain is ont he module, not * |
| 17:29.02 | devyll | can somebody explain why does my hardware echo cancelation module is not working? dmesg: http://pastebin.com/m6240243b dahdi 2.1.0.2 dahdi-linux 2.1.0.4 dahdi-modules-2.6.26 2.1.0.4 , asterisk 1.4.24 ... and digium card TE220B |
| 17:29.18 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:29.32 | Qwell | devyll: should contact Digium support if you're having issues |
| 17:29.34 | ultrav1olet | [TK]D-Fender: btw, our CPU is pseudo SMP CPU - P4 with HT |
| 17:29.45 | eppigy | TRANAJO |
| 17:29.48 | eppigy | TRABAJO |
| 17:29.53 | [TK]D-Fender | eppigy: PHAIL |
| 17:29.56 | eppigy | yes |
| 17:30.04 | eppigy | AQUE |
| 17:30.04 | ManxPower | ultrav1olet: turn off HT when using Zaptel |
| 17:30.08 | ManxPower | or DAHDI |
| 17:30.09 | eppigy | QUE |
| 17:30.12 | ultrav1olet | http://pastebin.ca/1482124 |
| 17:30.14 | ariel_ | work, work, work, well it's just plain, work |
| 17:30.19 | [TK]D-Fender | ultrav1olet: I'd suggest removing that gain option, and restarting and trying again |
| 17:30.31 | ultrav1olet | gain option has been already removed |
| 17:30.37 | ultrav1olet | about 10 minutes ago |
| 17:30.50 | ultrav1olet | I also removed all modules and restarted asterisk |
| 17:30.52 | [TK]D-Fender | ultrav1olet: I don't believe it will take effect until the kernel module is reloaded |
| 17:30.57 | [TK]D-Fender | ultrav1olet: Not jsut * |
| 17:30.59 | ultrav1olet | surely :) |
| 17:31.02 | ManxPower | [TK]D-Fender: what is ultrav1olet's problem. I have quite a but of experience with DTMF + ZAP |
| 17:31.32 | [TK]D-Fender | ManxPower: His like sounds like grap, he upped RX to 25 just to hear the other side, and outbound DTMF is getting cut to pieces. |
| 17:31.38 | ultrav1olet | ManxPower: 1) exceptionally bad signal on the line 2) asterisk calls wrong numbers |
| 17:31.45 | [TK]D-Fender | ManxPower: Feel free to take this one on :) |
| 17:32.02 | [TK]D-Fender | ManxPower: And line apparently tests OK with an analog phone. |
| 17:32.07 | ManxPower | ultrav1olet: first set toneduration=500 |
| 17:32.16 | ManxPower | [TK]D-Fender: sounds like what I had at the campground. |
| 17:32.21 | ultrav1olet | ManxPower: should I disable HT at first? |
| 17:32.33 | ultrav1olet | I meant CPU HT |
| 17:32.47 | ManxPower | ultrav1olet: no. HT should not cause these issues, but if the simple fixes don't help..... |
| 17:32.56 | ManxPower | ultrav1olet: you're not running this in a VM are you? |
| 17:33.01 | ultrav1olet | no |
| 17:33.04 | ultrav1olet | plain hardware |
| 17:33.16 | ultrav1olet | with CentOS 5.3 on top of it |
| 17:33.47 | ultrav1olet | besides it's impossible to passthrough PCI devices to a VM |
| 17:33.58 | ManxPower | ultrav1olet: you'd be amazed how many people try. |
| 17:34.51 | ManxPower | ultrav1olet: What echo cancel tech are you using? hardware (on card) or software. If software, which one? |
| 17:35.05 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 17:35.10 | [TK]D-Fender | ultrav1olet: there's hope for you yet. Work with ManxPower here for a bit and perhaps he'll be able to help more. If your card is under warranty I'd suggest giving Digium Support a call as well if its not working by then |
| 17:35.21 | [TK]D-Fender | ManxPower: MG@ + TMD400P |
| 17:35.23 | [TK]D-Fender | MG2 |
| 17:35.53 | [TK]D-Fender | ManxPower: [12:54]<ultrav1olet>http://pastebin.ca/1482081 |
| 17:35.55 | ManxPower | I'd recommend KB2 (I think that's the one I'm thinking of) |
| 17:36.35 | ultrav1olet | ManxPower: and you have to listen to this: http://rapidshare.com/files/251125620/result.wav.html |
| 17:36.37 | ManxPower | ultrav1olet: turn OFF echo cancel and see if it starts working as expected (if so we know where to look) |
| 17:36.45 | ultrav1olet | wait a minute |
| 17:37.10 | ManxPower | you're going to make me enable javascript on my browser, aren't you? |
| 17:37.19 | ultrav1olet | sorry :) |
| 17:37.30 | ManxPower | rapidshare is the AOL of file sharing |
| 17:37.37 | ultrav1olet | <PROTECTED> |
| 17:37.57 | ultrav1olet | I can upload that file anywhere you want - just give me the URL, please |
| 17:37.58 | ManxPower | ultrav1olet: I think so. I don't have all the zaptel -> dahdi chances memorized yet. |
| 17:38.31 | ultrav1olet | asterisk also have echo cancellation in asterisk/chan_dahdi.conf |
| 17:38.36 | ultrav1olet | so I'm asking this question |
| 17:38.52 | ManxPower | I'll have to find a machine an look it up. pastebin the two files. |
| 17:38.59 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 17:38.59 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:39.05 | ultrav1olet | http://pastebin.ca/1482124 |
| 17:39.22 | ultrav1olet | wrong post, sorry |
| 17:39.51 | ultrav1olet | this one contains all my configs http://pastebin.ca/1482081 |
| 17:40.16 | ultrav1olet | ManxPower: have you downloaded my sound file? |
| 17:40.45 | ManxPower | trying to |
| 17:41.25 | ManxPower | ultrav1olet: you're not in the USA are you? |
| 17:41.57 | ManxPower | the sound file tells me nothing other than the fact you might not be in the USA or that you are connected to a PBX |
| 17:42.11 | ultrav1olet | ManxPower: Russia |
| 17:42.34 | ManxPower | I bet your dahdi drivers are not set for russion line impediance. |
| 17:42.35 | ultrav1olet | if I'm not mistaken we have the same dialing system as in France |
| 17:43.04 | ultrav1olet | and asterisk 1.4 worked just fine here |
| 17:43.43 | *** part/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 17:43.43 | ManxPower | Then why did you switch? |
| 17:43.48 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 17:43.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:44.20 | ultrav1olet | We need some features from 1.6 |
| 17:44.37 | ManxPower | That is a good reason. |
| 17:44.40 | ultrav1olet | like MixMonitor |
| 17:44.49 | ManxPower | now, change the toneduration and turn off the echo cancel and test again. |
| 17:44.53 | ultrav1olet | big brother and everything like that |
| 17:45.15 | ManxPower | also remove busydetect too |
| 17:45.28 | ManxPower | You have MULTIPLE issues. |
| 17:45.45 | ultrav1olet | <PROTECTED> |
| 17:46.12 | ManxPower | no the chan_dahdi file should be good enough. |
| 17:46.20 | ultrav1olet | wait a minute |
| 17:46.39 | ManxPower | When I had similar problems I ended up having to use Digiums HPEC |
| 17:47.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:47.34 | ultrav1olet | 1) echo cancel - off 2) toneduration=500 3) you asked for something else ... I don't remember :( |
| 17:48.12 | ultrav1olet | busydetect :) one minute |
| 17:48.23 | ManxPower | <PROTECTED> |
| 17:48.49 | *** join/#asterisk djMax (n=chatzill@66.92.91.132) |
| 17:49.16 | ultrav1olet | no luck |
| 17:49.28 | ManxPower | ultrav1olet: exact same problems? |
| 17:49.30 | ultrav1olet | the signal quality is f*cked up |
| 17:49.32 | ultrav1olet | yep |
| 17:50.06 | ManxPower | what's the dmesg output for the dahdi stuff, pastebin it. |
| 17:50.09 | ultrav1olet | http://pastebin.ca/1482140 |
| 17:50.21 | ultrav1olet | that's a new chan_dahdi.conf |
| 17:50.50 | ManxPower | no. echocancel=no don't just remove it. |
| 17:51.09 | ManxPower | also ALL options must be listed BEFORE the channel lines. |
| 17:51.25 | ManxPower | so your toneduration and hanguponpolatrity are doing nothing. |
| 17:51.50 | ultrav1olet | dmesg: http://pastebin.ca/1482143 |
| 17:52.02 | ultrav1olet | ok |
| 17:52.46 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:52.49 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-c7bf7fdedaa0018e) |
| 17:52.49 | ManxPower | I would have to look it up, but this line " Module 3: Installed -- AUTO FXO (FCC mode)" seems to indicate the card is set to USA mode and impedience |
| 17:53.02 | ultrav1olet | Is this one OK? http://pastebin.ca/1482144 |
| 17:53.19 | ultrav1olet | ManxPower: how we can remedy that? |
| 17:53.22 | djMax | a sip softphone is saying login failed, but I don't see SIP traffic. Is there any other stage which would be rejecting the registration that would show up in sip debug? |
| 17:53.59 | ultrav1olet | djMax: check your sip phone parameters - probably you are connection to the wrong server ;) |
| 17:54.22 | ManxPower | "To see the regions supported open the wctdm.c zaptel source file in a text editor. Look for the section starting with: |
| 17:54.22 | ManxPower | fxo_modes[] = " look in that file and see what modes are available |
| 17:54.27 | djMax | yeah, this is fring on an iPhone, so very possible because I'm not sure if it direct connects or what |
| 17:54.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:55.39 | ultrav1olet | ManxPower: there are a LOT of modes |
| 17:56.03 | ManxPower | the RUSSIA mode is what you want if your are in russia |
| 17:56.17 | ultrav1olet | how should I use it? |
| 17:56.33 | ManxPower | ultrav1olet: see http://www.voip-info.org/wiki/view/TDM400P |
| 17:56.54 | ManxPower | ultrav1olet: is the server connected to a Russian telco? |
| 17:57.22 | ultrav1olet | yep, of course |
| 17:57.44 | ManxPower | then set the opermode to RUSSIA per the voip-info link I gave you |
| 17:58.03 | ultrav1olet | ok, dahdi: Registered tone zone 40 (Russian Federation) |
| 17:58.07 | ManxPower | it talks about zaptel, you'll have to figure out how to translate that into the way DAHDI does it. |
| 17:58.08 | ultrav1olet | is that OK? |
| 17:58.53 | ManxPower | No! That is the tone zone, not the opermode (tonezone just tells asterisk what dialtone ringing, etc it should play. the opermode is the hardware electrical ssettings for your country. |
| 17:59.05 | ultrav1olet | full dmesg: http://pastebin.ca/1482151 |
| 17:59.16 | ultrav1olet | I see |
| 17:59.31 | ultrav1olet | I have no idea how to change that ... let's read the manuals |
| 17:59.53 | ManxPower | http://www.voip-info.org/wiki/view/TDM400P |
| 18:00.10 | ManxPower | ultrav1olet: Did you read the voip-info link I gave you? |
| 18:00.24 | ultrav1olet | yep, I've found this "options wctdm opermode=RUSSIA" |
| 18:00.38 | ultrav1olet | I'm about to reload wctdm |
| 18:00.47 | ManxPower | ultrav1olet: did you unload / load the kernel module after that change |
| 18:01.14 | ManxPower | ultrav1olet: also what color are the modules on the card? |
| 18:01.24 | ultrav1olet | NOw it says: Module 3: Installed -- AUTO FXO (RUSSIA mode) |
| 18:01.42 | ultrav1olet | you mean port colors on the board itself? |
| 18:01.46 | ManxPower | yes. |
| 18:02.15 | ManxPower | ultrav1olet: all modules are RUSSIA mode or just one module? |
| 18:02.17 | ultrav1olet | all green (we have four ports) |
| 18:02.29 | ManxPower | well, you cannot connect green ports to the telco |
| 18:02.30 | ultrav1olet | I have the only kernel mode |
| 18:02.37 | ultrav1olet | ooops |
| 18:02.41 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 18:02.46 | ultrav1olet | so, what we will do? |
| 18:02.47 | ManxPower | you connect RED ports to the telco, you connect GREEN ports to telephones. |
| 18:03.05 | ultrav1olet | clear |
| 18:03.12 | ManxPower | Well if you plugged a green module into a telco line and that line receives a call (rings) then you have blown up the port and it must be replaced. |
| 18:03.23 | ultrav1olet | horror! |
| 18:03.46 | ultrav1olet | that means I have to change something like FXS FXO, right? |
| 18:03.48 | ManxPower | contact digium support. |
| 18:03.56 | ManxPower | you have to get a red module |
| 18:04.08 | ultrav1olet | can we do that on our own, please? |
| 18:04.26 | ultrav1olet | at least, try? |
| 18:04.37 | ManxPower | do what on your own? |
| 18:05.00 | ultrav1olet | change their colors |
| 18:05.34 | ManxPower | Maybe is you are a wizard and alchemist. They are different modules with different electronics. |
| 18:05.47 | ManxPower | There is nothing further anyone can do to help you. |
| 18:06.11 | ultrav1olet | but it's working ... a bit unreliably though ... |
| 18:06.13 | infinity1 | i just plugged a polycom 450 into my network and it doesn't work. there isn't even a registration error on the server. its configured exactly like the other 30x polycom 501's! what makes a 450 so special? |
| 18:06.41 | Qwell | ultrav1olet: If you have FXS modules (green) - it will not work at all. Double-check your hardware. |
| 18:06.43 | ultrav1olet | ManxPower: are you saying that my fxsks settings have no effect on my card? |
| 18:07.03 | ultrav1olet | you mean FXS should be RED? |
| 18:07.10 | Qwell | no |
| 18:07.26 | ultrav1olet | then I don't get it |
| 18:07.36 | ultrav1olet | I thought fxs manages the color ... |
| 18:08.21 | ManxPower | FXO modules are RED, you connect them to PHONE LINES. FXS modules are GREEN, you connect them to phones. They are different modules. |
| 18:10.05 | [TK]D-Fender | infinity1: "it doesn't work". sorry could you be a little more vague perhaps? |
| 18:10.19 | ultrav1olet | so, I have to change my dahdi configuration to FXO, right? |
| 18:10.38 | [TK]D-Fender | ultrav1olet: STOP |
| 18:10.41 | Qwell | ultrav1olet: it's a *hardware* difference. |
| 18:10.44 | *** join/#asterisk mltlnx (n=mltlnx@74.201.222.66) |
| 18:10.55 | [TK]D-Fender | ultrav1olet: the physical module on the card itself : What color is teh PCB? |
| 18:11.07 | ManxPower | ultrav1olet: YOU BOUGHT THE WRONG MODULES |
| 18:11.39 | infinity1 | [TK]D-Fender: heh. i can't dial. and when i do, nothing appears on the console. it has an IP, extension, etc. very strange. |
| 18:11.49 | ultrav1olet | the up side of the card is green, the bottom is read |
| 18:11.53 | infinity1 | [TK]D-Fender: and there are no errors or entries appearing on the console when i do dial. |
| 18:11.55 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 18:11.59 | [TK]D-Fender | infinity1: You see nothing with * SIP DEBUG enabled? |
| 18:12.06 | Qwell | ultrav1olet: take a picture of it and post it somewhere |
| 18:12.21 | [TK]D-Fender | ultrav1olet: imagebin.ca |
| 18:13.15 | ultrav1olet | http://images.fuzing.com/members/3/36/00185336/277584.300x300.jpg |
| 18:13.21 | infinity1 | [TK]D-Fender: looks like sip debug does do something! hmm |
| 18:13.27 | ultrav1olet | it's not _my_ card, but it's very similar |
| 18:13.43 | Qwell | ultrav1olet: take a picture of your card. |
| 18:14.02 | [TK]D-Fender | ultrav1olet: If the little modules are RED then they are the right ones |
| 18:14.29 | infinity1 | http://pastebin.com/d1dec8a6d |
| 18:15.23 | infinity1 | [TK]D-Fender: is this the problem "SIP/2.0 407 Proxy Authentication Required" ? |
| 18:15.26 | [TK]D-Fender | infinity1: Bad auth |
| 18:15.34 | [TK]D-Fender | SIP/2.0 401 Unauthorized <----- |
| 18:15.56 | infinity1 | hmmm |
| 18:17.13 | infinity1 | [TK]D-Fender: you're a genius. |
| 18:17.28 | ManxPower | ultrav1olet: getting the picture of the card is the single most important thing you must do today. |
| 18:17.31 | infinity1 | [TK]D-Fender: i thought i would have seen an error on the console. hrmz. |
| 18:17.56 | [TK]D-Fender | infinity1: Think again :) |
| 18:17.57 | infinity1 | anyone use 722? |
| 18:18.05 | [TK]D-Fender | infPlenty of people |
| 18:18.11 | [TK]D-Fender | infinity1: Plenty of people |
| 18:18.13 | infinity1 | now that i have a "HD" phone. LOL |
| 18:18.35 | [TK]D-Fender | infinity1: Try getting it to actually PLACE a call before worrying about how it sounds ;) |
| 18:18.49 | infinity1 | [TK]D-Fender: i'm assuming you can only do HD on your own voip network. not to the outside world? |
| 18:19.20 | [TK]D-Fender | infinity1: Anywhere up to a PSTN interface. Once you're there, its G.711 |
| 18:19.21 | infinity1 | [TK]D-Fender: yea. works fine now that i found a place i forgot to modify in the sip cofnig |
| 18:20.14 | infinity1 | [TK]D-Fender: so basically only on the voip networks i control can i get something useful from 722. bummer. any 722 providers sip gateways? lol |
| 18:20.39 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:20.45 | [TK]D-Fender | infinity1: doesn't matter. As soon as it hits PSTN it goes to G.711 |
| 18:22.13 | ultrav1olet | ManxPower: trying to upload the pictures |
| 18:22.22 | infinity1 | [TK]D-Fender: i also don't have asterisk 1.6. and i only have one phone with g722. |
| 18:22.25 | ultrav1olet | Opera Mini doesn't support images.ca :( |
| 18:22.29 | infinity1 | :) |
| 18:22.33 | ultrav1olet | aah, imagebin |
| 18:22.48 | [TK]D-Fender | ..... |
| 18:22.58 | [TK]D-Fender | You can't even hand out URLs these days... sheesh |
| 18:23.04 | ultrav1olet | :) |
| 18:23.48 | ultrav1olet | uploading via sh*tty GPRS :) |
| 18:23.59 | ultrav1olet | 4KB/sec :( |
| 18:25.55 | ultrav1olet | image1: http://imagebin.ca/view/urCEUd.html |
| 18:26.15 | ultrav1olet | image2: http://imagebin.ca/view/6v2PMCAc.html |
| 18:27.03 | ManxPower | ultrav1olet: you have RED MODULES. |
| 18:27.09 | ManxPower | assuming those are pics of your actual card |
| 18:27.09 | ultrav1olet | is that good? |
| 18:27.18 | ultrav1olet | yep, it is MY card :) |
| 18:27.28 | ManxPower | ultrav1olet: I'm very sorry, but I cannot help you further. Perhaps someone else can take over. |
| 18:27.47 | ultrav1olet | this is getting sadder :( |
| 18:28.02 | ultrav1olet | ManxPower: so, what's my current problem? |
| 18:28.14 | ultrav1olet | I'm just a bit lost |
| 18:29.49 | *** join/#asterisk jmls (n=jmls@host217-36-208-155.in-addr.btopenworld.com) |
| 18:29.59 | jmls | evening all ... |
| 18:30.08 | ultrav1olet | the sound from the card is still as shitty as it's always been with 1.6.x |
| 18:30.32 | jmls | got a question about hints |
| 18:30.49 | jmls | anyone know anything about them, so I can ask some questions ? |
| 18:32.42 | jmls | contemplates his navel, whistling some whistly tune |
| 18:35.42 | ultrav1olet | what if I change from fxsks=1-4 to fxoks=1-4? will it kill my digium card? |
| 18:38.38 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 18:39.04 | [TK]D-Fender | ultrav1olet: Don't, there is no point, and it would be wrong to use that setting. Your configs seem OK, call Digium support <--- |
| 18:39.13 | [TK]D-Fender | jmls: |
| 18:39.15 | [TK]D-Fender | ~ask |
| 18:39.16 | 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. |
| 18:39.18 | [TK]D-Fender | ^^^^ |
| 18:39.36 | jmls | eek |
| 18:39.40 | jmls | slaps himself |
| 18:39.45 | jmls | ~slap jmls |
| 18:39.46 | infobot | ACTION slaps jmls, keep your grubby fingers to yourself! |
| 18:39.58 | jmls | okies. |
| 18:40.17 | jmls | so I have |
| 18:40.23 | jmls | exten => 1236,hint,SIP/foo |
| 18:40.45 | ultrav1olet | ManxAway said I'm using the wrong signalling |
| 18:40.53 | jmls | this means if something subcribes to 1236, the state of SIP/foo is monitored ? |
| 18:41.03 | [TK]D-Fender | ultrav1olet: fxsks <---- |
| 18:41.11 | [TK]D-Fender | jmls: yes |
| 18:41.22 | jmls | right. So if I wanted to change it to SIP/baa. I would have to change the dialplan ? |
| 18:41.33 | jmls | exten => 1236,hint,SIP/baa |
| 18:41.33 | [TK]D-Fender | jmls: Yes |
| 18:41.38 | jmls | and reload the dialplan ? |
| 18:41.42 | [TK]D-Fender | jmls: Yes |
| 18:41.57 | jmls | urk. There isn't a Hint() function in 1.4 either |
| 18:42.01 | ultrav1olet | [TK]D-Fender: I suppose Digium support costs money, right? |
| 18:42.07 | jmls | that's really nasty. |
| 18:42.23 | [TK]D-Fender | ultrav1olet: If you're under warranty you're OK. |
| 18:42.28 | ultrav1olet | [TK]D-Fender: and my company will hardly agree to pay even a cent |
| 18:42.33 | jmls | we have hotdesking here, where an agent can logon to any SIP phone |
| 18:42.50 | jmls | we have a phone that has BLF's and we want to monitor the agent, not the phone |
| 18:42.55 | ultrav1olet | I've got no idea if we are still covered by the warranty - we bought this card in Russia |
| 18:43.00 | jameswf | ultrav1olet: I hear you have to have to have a Asterisk business license to buy a digium card |
| 18:43.02 | jameswf | :) |
| 18:43.18 | ultrav1olet | I've got no idea about that |
| 18:43.27 | ultrav1olet | we're using this card internally |
| 18:43.34 | [TK]D-Fender | jmls: So make a script for where you login that will issue the change to your hints in a separate INCLUDE-ed file and issue a reload |
| 18:43.53 | jameswf | ultrav1olet: it was a joke... was a comment floating around yesterday |
| 18:43.54 | [TK]D-Fender | ultrav1olet: Ignore jameswf's dumb joke :) |
| 18:44.03 | [TK]D-Fender | ~cluebat jameswf |
| 18:44.04 | infobot | ACTION pulls out a ClueBat (tm) and thwaps jameswf. |
| 18:44.19 | jmls | yeah, I have that. I am just concerned about reloading the dialplan everytime an agent logs in. We have upwards of 50 of them |
| 18:44.37 | [TK]D-Fender | jmls: No big deal. |
| 18:44.43 | jmls | hmmmm. Ok. |
| 18:45.19 | ultrav1olet | :) Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 18:45.29 | ultrav1olet | fxoks didn't work as expected |
| 18:45.33 | jmls | so you reckon it's ok to reload the dialplan tens of times in a matter of minutes ? If so, cool - coz the work is already done ;) |
| 18:45.58 | [TK]D-Fender | ultrav1olet: I expected that to fail, and it did. So don't do it |
| 18:46.27 | [TK]D-Fender | Ten times in 10 minutes = 1/minute = who cares. |
| 18:46.42 | [TK]D-Fender | ultrav1olet: Not a big deal |
| 18:46.53 | jmls | it's the bursts - we can have 30 of them logging in within a minute |
| 18:47.15 | jmls | 9:00am start => 8:59 everyone logs on :) |
| 18:47.22 | [TK]D-Fender | jmls: You could always do a time offset on them... |
| 18:47.39 | jmls | yeah, was thinking of reloading two minutes after the last login |
| 18:47.39 | [TK]D-Fender | jmls: I'm sure you can come up with a few places to chect, but its really not a big dea. |
| 18:49.02 | ultrav1olet | [TK]D-Fender: does Digium have a support over Skype? |
| 18:49.12 | ultrav1olet | I'm not in a mood of making international calls |
| 18:49.24 | ultrav1olet | My company may decide to fire me |
| 18:51.07 | ISO9001 | but they may not! |
| 18:51.54 | [TK]D-Fender | ultrav1olet: IAx2 <- its in the sample extensions.conf even |
| 18:52.25 | [TK]D-Fender | ultrav1olet: Set up a softphone or another * box so you can be on that call and still be able to reboot the server, etc as required |
| 18:52.42 | ultrav1olet | jmls: that was for you :) |
| 18:53.02 | ultrav1olet | [TK]D-Fender: wrong addressee :) |
| 18:53.40 | jmls | wuh ? |
| 18:53.46 | [TK]D-Fender | ultrav1olet: No, YOU asked babout Digium support over Skype |
| 18:54.05 | [TK]D-Fender | ultrav1olet: My answer implied NO, and gave you one that DOES exist |
| 18:54.13 | ultrav1olet | I asked if they have a support person on skype |
| 18:54.29 | [TK]D-Fender | NO |
| 18:54.43 | ultrav1olet | ok, then I'm absolutely out of luck |
| 18:55.00 | ultrav1olet | I still wonder why we did have zero problems when we ran asterisk 1.4 ... |
| 18:56.01 | [TK]D-Fender | ultrav1olet: I still wonder why it needs to be SKYPE. IAX2 works jsut fine |
| 18:56.44 | ultrav1olet | I'm not in a mood of making international calls ... |
| 18:56.55 | leifmadsen | IAX2 isn't an international call... |
| 18:56.55 | [TK]D-Fender | IAX IS A 78654ING VOIP PROTOCOL! |
| 18:56.56 | ultrav1olet | or Digium has a IAX phone? |
| 18:56.57 | leifmadsen | it is a VoIP call |
| 18:57.02 | [TK]D-Fender | ARGH!!!!!!!!!! |
| 18:57.05 | leifmadsen | giggles |
| 18:57.06 | ultrav1olet | sorry :) |
| 18:57.07 | [TK]D-Fender | assplodes |
| 18:57.24 | [TK]D-Fender | pays $.05 royalties to Katty |
| 18:57.26 | ultrav1olet | where can I find their IAX number? |
| 18:57.35 | [TK]D-Fender | [14:51]<[TK]D-Fender>ultrav1olet: IAx2 <- its in the sample extensions.conf even |
| 18:57.52 | ultrav1olet | oops |
| 18:57.55 | [TK]D-Fender | ultrav1olet: .... wake the hell up already! I keep shoving the answer in your face and you keep missing it |
| 18:58.07 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 18:58.43 | ultrav1olet | [TK]D-Fender: soorrrrrrrrry |
| 18:58.44 | *** join/#asterisk errotan (n=errotan@62.201.122.98) |
| 18:59.00 | [TK]D-Fender | :F |
| 18:59.07 | leifmadsen | ultrav1olet: try skype user 'digium' |
| 18:59.12 | ultrav1olet | I've now read the whole default extensions.conf and I see nothing resembling digium support number :( |
| 18:59.20 | ultrav1olet | ahh, ok |
| 18:59.23 | [TK]D-Fender | ultrav1olet: MISERY |
| 18:59.48 | jmls | ~calm [TK]D-Fender |
| 18:59.48 | infobot | ACTION sooths [TK]D-Fender, offers a backrub, tea, a jacuzzi and anything else that might help calm [TK]D-Fender down. |
| 18:59.50 | ultrav1olet | you mean Dial(IAX2/guest@pbx.digium.com/s@default) ? |
| 19:00.01 | [TK]D-Fender | ultrav1olet: wow! |
| 19:00.03 | leifmadsen | ultrav1olet: try 'digium' on skype |
| 19:00.07 | ultrav1olet | I should call @digium? |
| 19:00.09 | jmls | ultrav1olet: RUN, for god's sake man, RUN |
| 19:00.16 | ultrav1olet | :) |
| 19:00.22 | [TK]D-Fender | ultrav1olet: If you're already running Skype, do as leifmadsen suggests |
| 19:00.25 | leifmadsen | taps the mic... is this thing on? |
| 19:00.37 | leifmadsen | [TK]D-Fender: why did you need to freak out? oh wait... nevermind. |
| 19:00.53 | ISO9001 | lol |
| 19:00.54 | [TK]D-Fender | leifmadsen: The lights are on, the wheel is spinning, but the hamster is F-ing DEAD |
| 19:01.08 | leifmadsen | [TK]D-Fender: you seriously need less caffeine |
| 19:01.13 | jmls | hhe |
| 19:01.17 | [TK]D-Fender | leifmadsen: UNPOSSIBLE |
| 19:01.41 | [TK]D-Fender | needs to maintain his certification as NASA's only classified caffeine-based life-form |
| 19:01.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:02.01 | [TK]D-Fender | gets very cranky whn he gets too much blood in his caffeine-stream |
| 19:02.34 | ultrav1olet | [Jul 3 01:01:11] WARNING[32433]: chan_iax2.c:8944 socket_process: Call rejected by 216.207.245.8: No authority found |
| 19:02.52 | jmls | Do not try and bend the dialplan. That's impossible. Instead... only try to realize the truth. ...  There is no dialplan ... Then you'll see, that it is not the dialplan that bends, it is only yourself. |
| 19:02.52 | ultrav1olet | Executing [00@i_am_sane:1] Dial("IAX2/birdie-6826", "IAX2/guest@pbx.digium.com/s@digium") in new stack |
| 19:03.44 | ultrav1olet | any ideas why my call is being rejected? |
| 19:03.54 | [TK]D-Fender | [14:59]<ultrav1olet>you mean Dial(IAX2/guest@pbx.digium.com/s@default) ? |
| 19:04.05 | [TK]D-Fender | ultrav1olet: Why does the line not look the same as the FIRST TIME? |
| 19:04.50 | leifmadsen | ultrav1olet: didn't you ask for the skype username of digium? |
| 19:04.52 | jmls | because he is actually a phsycologist performing a social engineering experiment to see who bites ... ;) |
| 19:04.59 | ultrav1olet | I'm now calling Dial(IAX2/guest@pbx.digium.com/s@digium) |
| 19:04.59 | leifmadsen | why are you still trying IAX2? |
| 19:05.19 | ultrav1olet | leifmadsen: my skype has just broken :( |
| 19:05.22 | leifmadsen | lol |
| 19:05.28 | leifmadsen | runs off to write up some release anouncement |
| 19:05.32 | ultrav1olet | it asks for a password and i've forgotten one |
| 19:05.33 | [TK]D-Fender | utlWhy did you change the bloody CONTEXT from the first sample? And why haven't you jsut tried the skype contact leifmadsen has just confirmed is supposedly legitimate? |
| 19:05.57 | jmls | ~chill [TK]D-Fender |
| 19:05.58 | infobot | ACTION grabs [TK]D-Fender and rams a chillpill down [TK]D-Fender's throat with a toilet plunger |
| 19:06.06 | jmls | lol |
| 19:06.11 | jmls | that has to hurt .. |
| 19:06.28 | [TK]D-Fender | ultrav1olet: http://tinyurl.com/496svm |
| 19:06.42 | [TK]D-Fender | ~cluebat jmls |
| 19:06.43 | infobot | ACTION pulls out a ClueBat (tm) and thwaps jmls. |
| 19:06.52 | jmls | ooowwwwwwww. |
| 19:07.09 | jmls | ~sooth jmls |
| 19:07.23 | jmls | oh. right. I see. |
| 19:07.33 | jmls | thinks up some more mindless questions |
| 19:07.42 | ultrav1olet | LOL |
| 19:09.21 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:09.59 | jmls | anyways, thanks for the help, [TK]D-Fender. It is really appreciated . Seriously. Thanks ;) |
| 19:10.32 | jmls | oh! wait ! |
| 19:10.35 | jmls | ~soothe jmls |
| 19:10.36 | infobot | ACTION exerts a calming influence on jmls |
| 19:10.39 | jmls | yay! |
| 19:13.30 | ultrav1olet | [TK]D-Fender: what should I ask digium support person? |
| 19:13.57 | [TK]D-Fender | ultrav1olet: Maybe one of these days you'll learn how to explain your problem to someone. |
| 19:14.48 | ultrav1olet | :'( |
| 19:15.39 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 19:19.08 | ultrav1olet | my case is being escalated :) |
| 19:21.24 | [TK]D-Fender | ultrav1olet: Hope you're not wearing crocs otherwise you might get sucked into the feeder... |
| 19:22.55 | ultrav1olet | oh, no :) |
| 19:27.44 | andres833 | hi |
| 19:28.12 | andres833 | i need help with the configuration the free fax for asterisk :p |
| 19:29.11 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 19:30.08 | WindowsUser | :p |
| 19:31.36 | andres833 | please i have installing and chrge the modules but i dont know what is a form for probe the fax |
| 19:33.24 | WindowsUser | like a Fax Detection method? |
| 19:34.56 | *** join/#asterisk juanjoc (n=jcomella@200.69.219.113) |
| 19:35.34 | JackTheNipple | ;-) |
| 19:38.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:45.19 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 19:47.03 | ultrav1olet | how am I supposed to know what my digium card serial number it? |
| 19:47.13 | JackTheNipple | duuh |
| 19:47.14 | ultrav1olet | I cannot find any IDs on the card |
| 19:47.24 | ultrav1olet | nothing |
| 19:47.35 | JackTheNipple | I have some and its printed on a label |
| 19:47.41 | JackTheNipple | sitting on the card |
| 19:47.43 | ultrav1olet | my card is very similar to this one http://img.diytrade.com/cdimg/549961/4064987/0/1185954376.jpg |
| 19:47.49 | ultrav1olet | no stickers |
| 19:48.10 | ultrav1olet | except a purchase/warranty date |
| 19:48.30 | JackTheNipple | hm. you need that for login, right? |
| 19:48.50 | ultrav1olet | no, for Register/Activate Products |
| 19:49.02 | ultrav1olet | in order to get even a minimal support from Digium |
| 19:49.14 | JackTheNipple | I remeber, had to do that as well |
| 19:49.20 | ultrav1olet | Register your product to access the following benefits reserved for Digium customers: |
| 19:49.21 | ultrav1olet | Streamlined access to Digium's Technical Support Organization |
| 19:49.32 | JackTheNipple | gimme a sec, I'll check if there is a way to read it with dahdi tools |
| 19:49.37 | ultrav1olet | damn, my card has no ID :( |
| 19:49.44 | ultrav1olet | ok, thanks |
| 19:50.17 | leifmadsen | it's a digium card right... ? |
| 19:50.25 | leifmadsen | and not something like OpenVox or Sangoma or something.... |
| 19:51.10 | [TK]D-Fender | [14:25]<ultrav1olet>image1: http://imagebin.ca/view/urCEUd.html |
| 19:51.10 | JackTheNipple | I have several digiums, but none of the dahdi-tools (dahdi_cfg dahdi_hardware dahdi_registration dahdi_speed date |
| 19:51.10 | JackTheNipple | dahdi_genconf dahdi_monitor dahdi_scan dahdi_test ) give me some serial back |
| 19:51.12 | [TK]D-Fender | [14:26]<ultrav1olet>image2: http://imagebin.ca/view/6v2PMCAc.html |
| 19:51.14 | ultrav1olet | it has OpenVox sticker on it |
| 19:51.28 | ultrav1olet | same here |
| 19:51.33 | leifmadsen | ultrav1olet: OpenVox != Digium |
| 19:51.37 | leifmadsen | call OpenVox for your support |
| 19:51.44 | ultrav1olet | dahdi_scan: |
| 19:51.48 | ultrav1olet | description=Wildcard TDM400P REV E/F Board 5 |
| 19:51.53 | [TK]D-Fender | ultrav1olet: You have no support from Digium, try calling OpenVox, and.... |
| 19:51.54 | ultrav1olet | devicetype=Wildcard TDM400P REV E/F |
| 19:51.56 | [TK]D-Fender | ~wglwat |
| 19:51.57 | infobot | i heard wglwat is well, good luck with all that |
| 19:52.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:52.38 | ultrav1olet | [TK]D-Fender: does that mean that we own some Chinese unsupported cheap sh*t? |
| 19:52.40 | leifmadsen | ultrav1olet: http://www.openvox.com.cn/ |
| 19:52.45 | leifmadsen | ultrav1olet: pretty much, ya |
| 19:52.49 | leifmadsen | .cn |
| 19:52.49 | [TK]D-Fender | ^^^ |
| 19:53.03 | JackTheNipple | jeees |
| 19:53.05 | leifmadsen | guess new site is: http://www.openvox.cn/ |
| 19:53.48 | ultrav1olet | which looks like http://www.openvox.com.cn/products_detail.php?genre_id=9&id=31 |
| 19:54.16 | leifmadsen | ultrav1olet: http://www.openvox.cn/support/index.php?lang=2 |
| 19:54.20 | [TK]D-Fender | TRIXBOX CERTIFIED! Its the sh1zn1t y0! |
| 19:54.31 | [TK]D-Fender | And Lead-Free (tm)! |
| 19:54.46 | [TK]D-Fender | That means it comes bundled at no extra cost! |
| 19:54.46 | ultrav1olet | f*ck me |
| 19:54.51 | ultrav1olet | no phone support |
| 19:54.55 | JackTheNipple | dammit, what does the quad-e1 with echo-stuff cost?? |
| 19:54.57 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 19:55.15 | leifmadsen | another happy OpenVox customer :) |
| 19:55.18 | JackTheNipple | no |
| 19:55.35 | [TK]D-Fender | JackTheNipple: Depends where you buy, and what you buy |
| 19:56.47 | ultrav1olet | I wonder if it's worth trying their settings: http://wiki.openvox.cn/index.php/A400P |
| 19:58.00 | ultrav1olet | and they insist on loadzone = us; defaultzone = us |
| 19:58.25 | JackTheNipple | not reallokay, found it on your link: $1699 - I payed 1940EUR, which is $2700 for a Digium??? |
| 19:58.33 | [TK]D-Fender | ultrav1olet: A suggestion based on their larger client-base |
| 19:58.46 | ultrav1olet | ok, let's stay with ru zone |
| 19:59.06 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 19:59.17 | JackTheNipple | seems to be really cheap. |
| 19:59.32 | ultrav1olet | can anyone explain to me some of their settings? |
| 19:59.43 | ultrav1olet | like rxwink=300 |
| 19:59.57 | ultrav1olet | do I really need it? will it make any difference? |
| 20:00.24 | JackTheNipple | I don't know this param from digium E1 / BRI... |
| 20:02.03 | JackTheNipple | wink: Sets the wink timing. |
| 20:02.03 | JackTheNipple | rxwink: Sets the receive wink timing. |
| 20:02.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:04.52 | [TK]D-Fender | JackTheNipple: What link? |
| 20:05.12 | JackTheNipple | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 20:05.24 | ultrav1olet | their config is useless |
| 20:05.33 | ultrav1olet | the same audio quality, the same problems |
| 20:05.43 | [TK]D-Fender | JackI meant for you pricing inquiries... |
| 20:06.16 | JackTheNipple | ahah..;-) |
| 20:06.24 | JackTheNipple | for Digium? its beronet |
| 20:06.37 | JackTheNipple | beronet.com |
| 20:06.51 | JackTheNipple | but prices are available only after login |
| 20:07.15 | JackTheNipple | ..and including VAT |
| 20:07.49 | JackTheNipple | anybody using callfiles or AMI and suppressing the outgoing callerid? |
| 20:08.41 | *** join/#asterisk WhIteSidE (n=chatzill@209.236.250.9) |
| 20:09.41 | WhIteSidE | Hello, I just installed a new system, and I have a problem I haven't seen before. After creating a sip user, I keep seeing NOTICE[2787]: chan_sip.c:19536 handle_request_register: Registration from '"Back Office" <sip:6000@192.168.0.253>' failed for '192.168.0.165' - No matching peer found |
| 20:09.45 | WhIteSidE | Any thoughts> |
| 20:09.48 | WhIteSidE | (?) |
| 20:10.32 | WindowsUser | does sip work with spaces in usernames? and did you reload the chan_sip? |
| 20:11.02 | JackTheNipple | don't think so |
| 20:11.06 | JackTheNipple | not as username |
| 20:11.09 | WhIteSidE | My username is 6000, isn't it? |
| 20:11.18 | WhIteSidE | Yes to the chan_sip |
| 20:11.20 | JackTheNipple | hm, true |
| 20:11.33 | WhIteSidE | Hold on, I'll change that |
| 20:11.54 | JackTheNipple | hold on - I'm comming (James Brown) |
| 20:12.00 | WhIteSidE | What's weird is I've seen identical behavior with two different hardphones, a BT-100 and a Snom 300 |
| 20:13.35 | WhIteSidE | Yeah,, removing the space didn't make any difference |
| 20:14.02 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 20:14.25 | seb- | can a vpn solve issues w/ blocked ports when doing sip? |
| 20:14.49 | seb- | sip needs a lot of ports for the audio...would a vpn open up all 10000 of those ports? |
| 20:15.12 | WindowsUser | might wanna read up on what vpn does |
| 20:15.29 | JackTheNipple | ^^ |
| 20:15.53 | JackTheNipple | vpn just tunnels trough one connection |
| 20:16.22 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 20:16.37 | JackTheNipple | sip in vpn without restrictions is like a lan without restrictions |
| 20:17.38 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 20:18.39 | seb- | WindowsUser, JackTheNipple: ok..lame question perhaps but can a vpn make it appear that more ports are open to a machine then the firewall allows? |
| 20:18.56 | seb- | ...by tunneling all those ports through a smaller subset? |
| 20:18.59 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 20:19.05 | [TK]D-Fender | WhIteSidE: PASTEBIN your sip.conf masking only passwords |
| 20:19.06 | [TK]D-Fender | ~pb |
| 20:19.07 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 20:19.09 | [TK]D-Fender | ^^^ |
| 20:19.27 | WhIteSidE | Hang tight |
| 20:19.31 | p4tr0p1 | is Fax Over VoIP really a s*it?? Or there is anyway to make it works? We send about 200 fax/day, and we are missing something like 20% of it. |
| 20:19.55 | seb- | losing 20%? |
| 20:19.59 | seb- | get a better fax machine |
| 20:20.08 | seb- | ebay.com |
| 20:20.39 | suma | are you using t.38 or g711 ? |
| 20:20.43 | bijit | I have an asterisk and have remote extension connected to it through vpn.. There are time that the calls staying like if it was still active even though @ one end they hung up... any suggestions? |
| 20:21.29 | *** join/#asterisk vegbox (n=kevinle@adsl-64-161-85-234.dsl.lsan03.pacbell.net) |
| 20:23.32 | ManxAway | p4tr0p1: I would expect that failure rate for FaxOverVoiceOverIP |
| 20:24.52 | ManxAway | t.38, if you can get all the pieces to work together, should improve that rate. The problem is that T.38 seems to have interop issues with different vendors. |
| 20:25.13 | ManxAway | Coppice would know much more about T.38. |
| 20:26.02 | JackTheNipple | seb-: you have extra IPs for VPN and local-connection in case you use a local vpn-client |
| 20:26.53 | JackTheNipple | so you have extra services (IP:PORT-combination) for this VPN-IP |
| 20:28.50 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 20:29.02 | JackTheNipple | if you have lan-2-lan, its transparent to your clients. they only have a single lan-address and only 65000 ports |
| 20:30.18 | [TK]D-Fender | Ok, checkout time, heading home. Later all |
| 20:31.59 | leifmadsen | Asterisk 1.4.26-rc5 has just been released! For more information see http://www.asterisk.org/node/48606. Thanks for your continued support of Asterisk! |
| 20:33.04 | WhIteSidE | Well, it looks like I found a bug in Asterisk GUI |
| 20:33.31 | WhIteSidE | Either that or I have it set up wrong |
| 20:34.05 | *** join/#asterisk jsidhu (n=js@173-8-149-33-SFBA.hfc.comcastbusiness.net) |
| 20:34.17 | p4tr0p1 | ManxAway, that was exactly what I found about that ;o/ |
| 20:34.20 | ManxAway | WhIteSidE: Did you ask on #asterisk-gui? |
| 20:34.25 | WhIteSidE | I thought that gui added the items to sip.conf, but when I started going through sip.conf, it wasn't there, and yes, I hit Apply Changes severl times |
| 20:34.36 | jsidhu | is there a way to disable the voicemail prompt (for specific voicemail boxes) "Please leave a message after the tone, when done hangup or press the pound key" ? |
| 20:34.37 | WhIteSidE | I just realized that it was an #asteisk-gui problem |
| 20:34.41 | WhIteSidE | Thanks |
| 20:34.53 | ManxAway | jsidhu: yes. see the docs for Voicemail |
| 20:35.18 | WhIteSidE | Thanks for the help guys |
| 20:35.35 | jsidhu | ManxAway: would you please point me to the docs? Searched google but didnt get anything |
| 20:35.52 | ManxAway | jsidhu: "core show application voicemail" in the Asterisk CLI. |
| 20:35.53 | *** join/#asterisk pagec (n=cpage@96.57.210.34) |
| 20:36.00 | jsidhu | thanks |
| 20:36.12 | ManxAway | There should also be docs in the doc/ directory of the source and on voip-info.org |
| 20:36.18 | pagec | is there a way to save the current dialplan (and possibly sip users) to disk from a running asterisk session? |
| 20:36.35 | ManxAway | pagec: your question makes no sense. |
| 20:36.36 | leifmadsen | WhIteSidE: it doesn't add to sip.conf -- it adds users and such to users.conf |
| 20:36.59 | ManxAway | WhIteSidE: When using Asterisk GUI ALL issues are Asterisk GUI issues. |
| 20:37.14 | jsidhu | Thanks ManxAway: i beilive i have what i needed. |
| 20:37.17 | WhIteSidE | but shouldn't it also add the extensions to sip.conf? I haven't had trouble in the past |
| 20:37.29 | leifmadsen | WhIteSidE: no, the GUI never added extensions to sip.conf -- only users.conf |
| 20:37.46 | pagec | I have asterisk 1.6 running. I am attached with `asterisk -r`, is there a way to have asterisk save it's configuration to disk (i deleted /etc by mistake) |
| 20:38.02 | leifmadsen | pagec: heh... oops |
| 20:38.02 | ManxAway | pagec: only for extensions.conf |
| 20:38.04 | JackTheNipple | lol |
| 20:38.08 | JackTheNipple | jees |
| 20:38.28 | ManxAway | there is a dialplan save or something like that. You won't get back any of the other info, AFIK |
| 20:38.32 | JackTheNipple | I feel sorry for you. |
| 20:39.01 | WhIteSidE | nark, nark, nark. Ok, that explains it I guess |
| 20:39.17 | pagec | anyone know how to do that with extensions.conf? |
| 20:39.30 | WhIteSidE | Thanks |
| 20:42.21 | WindowsUser | pagec: dialplan show |
| 20:43.25 | JackTheNipple | pagec: I don't know. Maybe you can use "show dialplan" |
| 20:43.31 | JackTheNipple | ^^ |
| 20:43.37 | WindowsUser | oh |
| 20:43.38 | leifmadsen | how about 'dialplan save' |
| 20:43.54 | leifmadsen | dialplan<TAB><TAB> shows several options |
| 20:43.59 | WindowsUser | not on my copy of 1.6:( |
| 20:44.02 | pagec | thanks |
| 20:44.20 | leifmadsen | Example: dialplan save (/etc/asterisk/extensions.conf) |
| 20:44.20 | leifmadsen | <PROTECTED> |
| 20:44.30 | WindowsUser | cool |
| 20:45.10 | JackTheNipple | leifmadsen: great! |
| 20:45.31 | WindowsUser | i guess 1.6.1.1 is too old for that |
| 20:45.40 | leifmadsen | WindowsUser: no... I ran that on 1.4 |
| 20:45.45 | leifmadsen | core dialplan ... |
| 20:45.48 | leifmadsen | perhaps |
| 20:45.55 | JackTheNipple | I'll remeber this one in case I rm my /etc ;-) |
| 20:46.02 | JackTheNipple | me too, works on 1.4 |
| 20:46.46 | leifmadsen | 1.6.1.1 was only released like... a month ago :) |
| 20:46.50 | WindowsUser | dialplan is like the one thing not thrown into core it feels like :D |
| 20:47.13 | leifmadsen | I don't have a 1.6.x box handy to check, but the options should be there somewhere |
| 20:47.16 | leifmadsen | help dialplan |
| 20:47.35 | JackTheNipple | ...its the one big thing to put in your backups :-) |
| 20:48.58 | pagec | yeah regrettably i was using online backups and the new script.... anyway |
| 20:49.04 | WindowsUser | haha |
| 20:49.06 | pagec | back to tape |
| 20:49.11 | moa_ | dialplan save worked on 1.6. Although it did re parse my /etc/asterisk/extensions.conf after the save, don't know if he'll run into problems there since it's not there. |
| 20:49.36 | WindowsUser | damn, its just missing from mine for some unknown reason then |
| 20:49.39 | WindowsUser | everything else is there |
| 20:50.29 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 20:52.21 | leifmadsen | moa_: it'll remove all comments, etc... from the file |
| 20:52.48 | leifmadsen | I use svn to track all changes in /etc/asterisk/ |
| 20:53.20 | leifmadsen | useful when your dialplan is couple thousand lines long |
| 20:53.51 | moa_ | leifmadsen: so would it work if you didn't have one for it to parse? Or is it saving the running dialplan? |
| 20:54.27 | leifmadsen | it saves the running dialplan |
| 20:55.08 | moa_ | That's good to know in case my /etc goes up in a ball of flames. |
| 20:55.51 | WindowsUser | flame on! |
| 20:59.24 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 20:59.37 | leifmadsen | I just keep backups for that kind of thing |
| 21:02.01 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 21:02.01 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:05.59 | *** join/#asterisk jsgoecke (n=Adium@208.74.176.162.static.etheric.net) |
| 21:06.05 | jsgoecke | Hello everyone. |
| 21:06.33 | JackTheNipple | hi |
| 21:06.38 | jsgoecke | I have Asterisk 1.4.25.1 on Ubuntu Jaunty. Inspite of asterisk.conf showing the run directory as /var/run/asterisk and the init.d script the same, Asterisk still wants to right asterisk.pid and asterisk.ctl to /var/run. |
| 21:06.43 | jsgoecke | Is there a new problem I should know about? |
| 21:07.43 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:07.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:10.18 | JackTheNipple | for me its also /var/run/ |
| 21:10.26 | JackTheNipple | not /var/run/asterisk/ |
| 21:10.35 | jsgoecke | But, that is a security risk if I give Asterisk the right to write to /var/run |
| 21:10.44 | jsgoecke | I want it to only access /var/run/asterisk as I have done before, always |
| 21:10.47 | *** join/#asterisk TheSin (n=TheSin@gateway.bluefalls.ca) |
| 21:11.29 | [TK]D-Fender | jsgoecke: Your choice, do whatever you want |
| 21:11.43 | JackTheNipple | do you start via /etc/init.d ? |
| 21:12.07 | TheSin | hey guys, I'm having an odd problems, when I get an incoming call I check what number they dials and I'm trying to reset the CID Name, and I'm guessing it write cause the voice mail emails show the right name but the sip phones aren't, can anyone explain what i'm doing wrong? |
| 21:12.19 | jsgoecke | And, the fact that I have it configured in asterisk.conf and in the init.d to do /var/run/asterisk, and yet it forces back to /var/run. |
| 21:12.37 | jsgoecke | The v1.4.21.1 that comes as an Ubuntu packages handles it just fine |
| 21:12.45 | jsgoecke | Problem is, that package appears to be missing the 'System' command |
| 21:12.55 | jsgoecke | Which I need, so hence a roll my own, which then presents this issue.' |
| 21:13.01 | JackTheNipple | member:jsgoecke: in deed this is strange, I'd expect is to get what you define in asterisk.conf |
| 21:13.20 | jsgoecke | JackTheNipple Exactly, and at a minimum in the init.d using stop-start-daemon |
| 21:13.25 | jsgoecke | Pulling my hair out |
| 21:13.56 | JackTheNipple | me I'm with 1.4.25 and its the same |
| 21:14.27 | JackTheNipple | jsgoecke: did you see the comment "[directories](!) ; remove the (!) to enable this" |
| 21:14.52 | JackTheNipple | jsgoecke: on top of asterisk.conf? |
| 21:15.01 | jsgoecke | Not on top, but as astrundir I believe |
| 21:15.20 | jsgoecke | [global] |
| 21:15.25 | jsgoecke | astrundir => /var/run/asterisk |
| 21:16.28 | JackTheNipple | look at /etc/asterisk/asterisk.conf |
| 21:16.39 | jsgoecke | That is what I just posted above, from /etc/asterisk/asterisk.conf |
| 21:16.51 | JackTheNipple | if there is this comment, get rid of the (!) |
| 21:16.57 | JackTheNipple | okay |
| 21:16.58 | jsgoecke | No, no commenting out |
| 21:17.20 | JackTheNipple | with 1.4.25 it works |
| 21:17.22 | JackTheNipple | just tested |
| 21:17.37 | JackTheNipple | might be something on 1.4.25.1 |
| 21:17.46 | jsgoecke | Ok, so let me roll back to 1.4.25 |
| 21:17.51 | jsgoecke | Thanks for checking |
| 21:17.54 | jsgoecke | Will try and report back |
| 21:17.58 | JackTheNipple | or test 1.4.26rc5 |
| 21:20.45 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 21:24.27 | leifmadsen | if you look at the ChangeLog, you will see the 1.4.25.1 is just a security release from 1.4.25 |
| 21:24.38 | leifmadsen | thus there is only a single change between .25 and .25.1 |
| 21:26.43 | jsgoecke | JackTheNipple what OS are you on? |
| 21:26.53 | jsgoecke | leifmadsen I think you are right, 1.4.25 is the same |
| 21:26.56 | JackTheNipple | Debian Lenny |
| 21:27.03 | jsgoecke | Maybe this is an Ubuntu Jaunty thing.... |
| 21:27.11 | leifmadsen | jsgoecke: you think I'm right? I'm the one who made the release and announcement :) |
| 21:27.20 | leifmadsen | look at the ChangeLog :) |
| 21:27.23 | JackTheNipple | ^^ |
| 21:27.30 | jsgoecke | Ok, yes, you are right ... ;) |
| 21:28.03 | JackTheNipple | is your 1.4.25 packed with ubuntu? or did you get it yourself? |
| 21:28.04 | jsgoecke | But is there something crazy on Ubuntu Jaunty that would cause a difference between the roll your own 1.4.25/.1 and the apt-get package of 1.4.21? |
| 21:28.09 | jsgoecke | No, it is not |
| 21:28.19 | jsgoecke | 1.4.21 is, problem is it does not appear to have the System command |
| 21:28.29 | suma | Is voicexml is only for one call or it can control multiple calls ? |
| 21:29.10 | jsgoecke | VXML in what context suma |
| 21:29.51 | suma | I want to automate outgoing calls and control them |
| 21:30.05 | JackTheNipple | use AMI |
| 21:30.09 | suma | resulting landing part can be queue, or with another channel etc., |
| 21:30.18 | jsgoecke | JackTheNipples Ok, we have it running on Ubuntu Jaunty with no problem on a local install |
| 21:30.37 | jsgoecke | But, this is on Amazon with an Ubuntu image there, and a problem |
| 21:30.42 | jsgoecke | This one is crazy |
| 21:31.16 | JackTheNipple | jsgoecke: this is really strange. Did you compile yourself? |
| 21:32.45 | JackTheNipple | jsgoecke: on Amazon clouds? interesting. so you're in U.S.? Or in Europe? |
| 21:32.56 | jsgoecke | No, we used an Alestic Ubuntu Jaunty image |
| 21:35.37 | jsgoecke | JackTheNipple San Francisco, so in east-1b |
| 21:41.35 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 21:43.29 | jsgoecke | Ok, it gets weirder |
| 21:43.42 | jsgoecke | It appears, that this alestic image is the one subverting the system command |
| 21:43.48 | jsgoecke | Not the Ubuntu package |
| 21:43.54 | jsgoecke | of Asterisk.... |
| 21:44.20 | jsgoecke | Time to roll my own locally and push to S3, and hope this is not some type of weird Amazon kernel issue |
| 21:44.46 | ManxAway | Running Asterisk in a cloud. Fun for the whole family! |
| 21:45.51 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:47.28 | carrar | How high do you have to be to do that? |
| 21:48.02 | jsgoecke | Question. |
| 21:48.09 | jsgoecke | Asterisk writes files as 640 |
| 21:48.17 | jsgoecke | When you issue a record cmd |
| 21:48.32 | jsgoecke | Is there anyway to get it to do 644 without issuing a chmod via the System cmd |
| 21:48.33 | jsgoecke | ? |
| 21:52.14 | jsgoecke | I see on line 6 of app_record.c it appears to be doing a 0644, but yet it comes out as 0640... |
| 21:54.08 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 21:57.03 | jsgoecke | Ok, on OSX it writes it 644, on this Ubuntu Jaunty image it is 640, this is strange. |
| 21:58.47 | jsgoecke | Unless we run as root, then it does 644, but as asterisk no joy, 640 |
| 21:59.03 | WindowsUser | maybe look into umask? |
| 21:59.04 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:59.46 | WindowsUser | mine are 644 on ubuntu and umask returns 022 |
| 21:59.55 | WindowsUser | well. 0022 |
| 22:00.29 | JackTheNipple | me too |
| 22:03.44 | *** join/#asterisk veli_ (n=veli_@85.96.109.156) |
| 22:03.45 | veli_ | hello |
| 22:03.53 | veli_ | asterisk default password ? |
| 22:04.48 | jsgoecke | veli_ Default password for which part of Asterisk? |
| 22:05.16 | veli_ | last version (i am sory, i english bad) |
| 22:05.30 | jsgoecke | veli_ Asterisk does not really have a default password. |
| 22:05.33 | WindowsUser | what has a password? |
| 22:05.53 | jsgoecke | It would depend if you are using some specific distro, or a particular API that has been configured. |
| 22:06.03 | veli_ | first at install ? |
| 22:06.18 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:06.20 | WindowsUser | like a sip client password? |
| 22:06.35 | ManxAway | veli_: I'll bet you are using some GUI or configurator for Asterisk |
| 22:07.02 | ManxAway | But Asterisk itself has no default password |
| 22:07.14 | WindowsUser | veli_: "AsteriskNow"? |
| 22:07.18 | lanning | veli_: what did you install? |
| 22:07.56 | ManxAway | Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 22:08.27 | veli_ | thanks working i dont working keybord numeric button |
| 22:08.31 | veli_ | :) |
| 22:14.12 | *** part/#asterisk ManxAway (n=manxpowe@69.73.94.162) |
| 22:17.15 | *** join/#asterisk xpot-mobile (n=james@144.35.20.62) |
| 22:23.29 | *** join/#asterisk [netman] (n=netman@158.Red-88-24-108.staticIP.rima-tde.net) |
| 22:43.44 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
| 22:44.44 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:49.34 | Joel | how is libpri different then dahdi? |
| 22:49.43 | Joel | dahdi is to talk to hw, libpri is for the protocol? |
| 22:49.49 | WindowsUser | ~libpri |
| 22:49.52 | voxter | Anyone know how to get x-lite sound card settings to persist between reboots/detaching reattaching your usb headset? |
| 22:49.59 | WindowsUser | hrm |
| 22:50.01 | voxter | it seems to want to default back to system defaults every time its loaded |
| 22:50.24 | WindowsUser | voxter: probably a missing feature from x-lite |
| 22:51.02 | *** join/#asterisk op3r (n=op3r@114.108.206.103) |
| 22:51.06 | WindowsUser | Joel: thats probably right |
| 22:51.48 | *** join/#asterisk Jumpie (n=zz@c-76-100-241-4.hsd1.md.comcast.net) |
| 22:51.54 | Jumpie | wow freenode is about as stable as undernet |
| 22:52.02 | Jumpie | every time i come home i'm disconnected, on another nick, and unregistered :p |
| 22:52.09 | op3r | had anyone experienced not hearing asterisk sound files? I mean tt-monkeys is playing perfectly on cli but you wont hear it on the phone. extension to extension is working though :( |
| 22:52.15 | drmessano | Jumpie: Unstable |
| 22:52.18 | Joel | op3r or your connection is just as unstable |
| 22:52.21 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 22:52.26 | drmessano | Jumpie: and "youre welcome" for ruining your sarcasm |
| 22:53.52 | op3r | Joel: i dont think so as extension to extension calling is fine |
| 22:53.53 | op3r | :( |
| 22:54.05 | Joel | op3r haha I meant irc |
| 22:54.12 | op3r | oh ok |
| 22:54.14 | Joel | op3r but yes, that was recently reported in 1.6 |
| 22:54.19 | Joel | op3r what version you running? |
| 22:54.26 | op3r | but Im using 1.2.30.4 |
| 22:54.27 | op3r | :( |
| 22:54.37 | Joel | op3r no idea then, almost everyone will tell you to upgrade. |
| 22:54.45 | Joel | op3r is this a trixbox or? |
| 22:55.02 | WindowsUser | op3r: are you just setting up asterisk or is this a recently noticed problem? |
| 22:55.34 | op3r | Joel: just set'ed up asterisk for vicidial |
| 22:55.51 | Joel | op3r ah |
| 22:55.56 | Joel | skybroadband? |
| 22:56.02 | op3r | I mean when I have enabled externip on sip.conf it worked fine |
| 22:56.06 | op3r | then I rebooted |
| 22:56.11 | op3r | then it went kaput |
| 22:57.12 | op3r | although at the CLI its playing but you cant hear it on the phone. |
| 23:00.37 | WindowsUser | probably a voice path problem |
| 23:00.59 | op3r | even if you are connected to the same lan? |
| 23:01.23 | WindowsUser | if I tell you to connect to 1.1.1.1 it doesn't matter if you're right next to me |
| 23:02.42 | op3r | hmmm |
| 23:05.06 | op3r | but but but |
| 23:05.19 | op3r | on extension to extension calling it is working fine. |
| 23:06.19 | WindowsUser | is the voice going through asterisk? |
| 23:06.54 | op3r | yep |
| 23:19.34 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 23:23.46 | *** join/#asterisk dwschool (n=Dwayne@64.42.227.97) |
| 23:23.55 | dwschool | hi - outgoing call through my sip trunk appear with callerid of my internal extension - how can i set it to the number i'm calling from? |
| 23:26.25 | WindowsUser | dont set caller id? |
| 23:26.41 | WindowsUser | or i guess do haha |
| 23:27.33 | WindowsUser | Set(CALLERID(number)=604xxxxxxx) |
| 23:27.41 | *** join/#asterisk DrkShadow (n=andrew@host-72-175-240-62.static.bresnan.net) |
| 23:29.14 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 23:29.16 | DrkShadow | Please explain: [Jul 2 17:21:37] VERBOSE[25351] logger.c: -- Executing [support@co:4] GotoIfTime("SIP/wind_us-1da43b70", "07:35-17:20|mon-fri|*|*?Open") in new stack; [Jul 2 17:21:37] VERBOSE[25351] logger.c: -- Goto (co,support,7); [Jul 2 17:21:37] VERBOSE[25351] logger.c: -- Executing [support@co:7] NoOp("SIP/wind_us-1da43b70", "") in new stack <-- obviously it didn't hit 6; obviously it went, so obviously it t |
| 23:29.16 | DrkShadow | ook 17:21:37 to be in the range 07:35-17:20. This is not the first time. what's happening? |
| 23:29.46 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:46.06 | *** join/#asterisk dshap (n=dshap@216-165-39-50.DYNAPOOL.NYU.EDU) |
| 23:46.36 | *** part/#asterisk lanning (n=lanning@nat/yahoo/x-c7bf7fdedaa0018e) |
| 23:46.46 | dshap | can anyone recommend a way that i can deal with a recorded file that a user records and hangs up during the recording? |
| 23:48.26 | Qwell | dshap: delete it? |
| 23:48.49 | Qwell | what's the actual question/problem? |
| 23:49.33 | dshap | Qwell: i would like a user to record a message and then i would like to run an AGI script that does something with the audio file |
| 23:49.50 | dshap | Quell: the issue is - if the user doesn't press # and just hangs up the line, everything stops |
| 23:49.55 | dshap | it's just like a voicemail system |
| 23:50.00 | dshap | you can leave a voicemail for someone and then hang up |
| 23:50.03 | dshap | and it will still be delivered |
| 23:50.05 | dshap | you don't have to press pound |
| 23:50.08 | dshap | how can i get this functionality? |
| 23:50.08 | Qwell | h extension? |
| 23:50.15 | dshap | what is the h extension? |
| 23:50.21 | Qwell | ~h exten |
| 23:50.22 | infobot | methinks exten is ${EXTEN} in a line is *always* the same as whatever is between the => and the first comma in the line. |
| 23:50.29 | Qwell | infobot: stupid bot |
| 23:50.30 | infobot | Stupid human. |
| 23:51.12 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 23:51.15 | dshap | if i name an extension "h" in a context, and a caller is in that context and hangs up. will asterisk begin executing extension "h" priority 1? |
| 23:52.30 | dshap | Qwell: is that how it works? |
| 23:53.25 | *** join/#asterisk ajmcello (n=ajmcello@75-151-111-233-Washington.hfc.comcastbusiness.net) |
| 23:54.05 | dshap | ok looks like it |
| 23:54.12 | dshap | also |
| 23:54.20 | dshap | is ${RECORDED_FILE} deprecated in v1.6? |
| 23:56.51 | dshap | i'm confused |
| 23:56.57 | WindowsUser | you sure its a variable provided by asterisk and not provided by you? |
| 23:57.17 | ajmcello | I'm using a perl AGI script and I'm trying jump to a different context in the dialplan called Expired, but I can't get it to work. I'm using "SET CONTEXT Expired" and then "EXEC GOTO (Expired,s,1)", I get this error: Channel 'SIP/150-9c3887a8' sent into invalid extension 's' in context 'Expired', but no invalid handler |
| 23:57.18 | WindowsUser | Set(RECORDEDFILE=/tmp/something_adaslkfasf.wav) |
| 23:57.20 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 23:57.53 | WindowsUser | ajmcello: paste this Expired context |
| 23:58.07 | WindowsUser | its saying theres nothing for "s" or for "i" |
| 23:58.31 | dshap | WindowsUser: i read that it was a channel variable |
| 23:58.36 | WindowsUser | kinda like how apache tells you: 404, also theres no 404 error page available |
| 23:58.44 | dshap | WindowsUser: but it's okay, i can set it myself |
| 23:58.55 | ajmcello | [Expired] |
| 23:58.55 | ajmcello | exten => s,n,Playback(Expired) |
| 23:58.55 | ajmcello | exten => i,2,GoTo(MainMenu,s,1) |
| 23:58.55 | ajmcello | exten => s,n,Wait(1) |
| 23:58.56 | dshap | WindowsUser: can i call AGI() on a hung up channel as long as it doesn't send any commands to asterisk? |
| 23:59.15 | WindowsUser | dshap: if you're in 1.4 use DeadAGI |
| 23:59.25 | dshap | WindowsUser: 1.6 |
| 23:59.41 | WindowsUser | then call AGI and dont play any audio to the channel |
| 23:59.51 | dshap | ok |
| 23:59.52 | jaytee | ajmcello, you don't have a priority 1 for the s extension or the i extension |
| 23:59.54 | WindowsUser | ajmcello: gotta start with 1 |
